| 00:01.50 | *** join/#asterisk joako (n=ston3d@opensuse/member/joak0) |
| 00:02.09 | joako | can I change the location of /var/spool/asterisk/outgooing but leave everything else in /var/spool/asterisk the same? |
| 00:02.21 | Katty | so it's ground buffalo? |
| 00:02.25 | Katty | in bratwurst shape? |
| 00:02.33 | Katty | with bits of jalepeno |
| 00:02.55 | joako | Am I in #asterisk or #cooking? |
| 00:03.10 | p3nguin | joako: I'm not sure that you can break astspooldir into pieces like that. |
| 00:03.17 | dcabot | I'm wondering myself |
| 00:03.22 | Katty | yeah i'm not really sure i care, joako |
| 00:03.49 | joako | p3nguin: Can I remove /var/spool/asterisk/outgooing and then create a symlink? |
| 00:04.04 | p3nguin | I don't see why that wouldn't work. Try it. |
| 00:04.10 | saghul | joako: sure you can |
| 00:04.13 | hardwire | Katty: and cheese |
| 00:04.21 | Katty | mmmmmmmmmmmmmmmmmmmmm cheese |
| 00:04.23 | hardwire | smoked |
| 00:04.27 | hardwire | in brat links |
| 00:04.32 | hardwire | very fun |
| 00:04.35 | hardwire | nicely packed |
| 00:04.37 | hardwire | full of goo |
| 00:04.40 | hardwire | very yum |
| 00:04.41 | hardwire | bbl |
| 00:04.47 | Katty | neat. |
| 00:04.47 | p3nguin | mkdir some-other-place ; ln -s some-other-place /var/spool/asterisk/outgoing |
| 00:04.48 | dcabot | I'll take 5 lbs |
| 00:05.00 | hardwire | dcabot: that'll be $600 |
| 00:05.21 | dcabot | Wow, what a business! |
| 00:05.26 | hardwire | I aim to please. |
| 00:05.29 | hardwire | bbl |
| 00:07.13 | joako | saghul: and as long as asterisk can read/write the files permission & ownership do not matter, correct? |
| 00:08.05 | dcabot | So, does anyone know of any issues with IAX2 and binfone on 1.6.0.14 or 1.6.0.17 or 1.4.26.2? |
| 00:08.05 | saghul | joako: right |
| 00:12.00 | saftsack | dcabot, did you have a look in the bugtracker? |
| 00:13.22 | *** part/#asterisk saghul (n=saghul@ip51ccb640.speed.planet.nl) |
| 00:15.14 | ruben23 | p3nguin: just info, it worked well now |
| 00:15.20 | *** join/#asterisk saghul (n=saghul@ip51ccb640.speed.planet.nl) |
| 00:15.47 | *** part/#asterisk saghul (n=saghul@ip51ccb640.speed.planet.nl) |
| 00:15.49 | ruben23 | i test it..but with inbound im testing it outside my network |
| 00:16.09 | *** join/#asterisk saghul (n=saghul@ip51ccb640.speed.planet.nl) |
| 00:18.54 | *** part/#asterisk saghul (n=saghul@ip51ccb640.speed.planet.nl) |
| 00:19.15 | *** join/#asterisk saghul (n=saghul@ip51ccb640.speed.planet.nl) |
| 00:20.26 | p3nguin | ruben23: Are you experiencing a problem with something? |
| 00:22.14 | ruben23 | p3nguin: its just i tried to dial my did# within my local phone busy..i dont have other tester outside my network |
| 00:22.46 | p3nguin | ruben23: You don't have another phone that isn't connected on asterisk? |
| 00:23.17 | p3nguin | ruben23: no cell phone, no POTS? |
| 00:24.41 | Katty | http://tinyurl.com/yzrrxet <- the gas station i frequent. |
| 00:24.44 | ruben23 | p3nguin:office are close, just for now..but tomorrow ill do it |
| 00:26.55 | Katty | http://tinyurl.com/yd3z2xf <- my favorite chinese place. |
| 00:28.26 | Katty | http://tinyurl.com/yal3xnq <- where i get groceries |
| 00:29.53 | dcabot | saftsack: Sorry, didn't mean to ignore you. I was deep in wireshark. asterisk is sending a reject, but without any decent debug statements, can't tell why. |
| 00:30.23 | dcabot | What's the link for asterisk's bugtracker? |
| 00:30.57 | saftsack | https://issues.asterisk.org/main_page.php |
| 00:31.23 | Katty | infobot: asterisk bug tracker? |
| 00:31.44 | Katty | infobot: Asterisk Issue Tracker is https://issues.asterisk.org |
| 00:31.44 | infobot | Katty: okay |
| 00:32.02 | Katty | infobot: Asterisk Issue Tracker? |
| 00:32.02 | infobot | somebody said asterisk issue tracker was https://issues.asterisk.org |
| 00:32.57 | ruben23 | p3nguin: anything serious with this--->[Nov 8 08:15:06] NOTICE[3316]: chan_sip.c:15933 handle_request_subscribe: Received SIP subscribe for peer without mailbox: 202 |
| 00:34.24 | saftsack | are there some alternatives to asterisk tapi and siptapi for integrating asterisk into m$ outlook? |
| 00:37.49 | Xetrov` | how do you link someone directly to a spot? |
| 00:38.15 | Xetrov` | oh nm |
| 00:38.19 | Xetrov` | <--idiot |
| 00:40.21 | hardwire | baka |
| 00:42.33 | p3nguin | ruben23: That means you have created a sip peer or user with a voicemail line, but you didn't configure voicemail for that user in voicemail.conf. |
| 00:44.08 | p3nguin | ruben23: Once you're done with configuring files, paste your extensions.conf and sip.conf in the pastebin. |
| 00:44.24 | ruben23 | ok im done |
| 00:48.31 | ruben23 | p3nguin: this is my extensions.conf ----->http://pastebin.com/m21a2ae4c and this is my sip.conf ---->http://pastebin.com/m59261c73 |
| 00:52.02 | p3nguin | I have a couple concerns. |
| 00:52.23 | ruben23 | p3nguin: yes what is it..? |
| 00:53.06 | p3nguin | You're dialing SIP/201 and SIP/211 in the match for 201, but you don't have a 211 peer/user in sip.conf. |
| 00:53.58 | p3nguin | And your IVR is going to break, because you probably do not have the local/mm-this-call-monitor-record file that will try to play on an incoming call. |
| 00:54.21 | dcabot | How can I get more debug out of asterisk to determine what's going on with this IAX2 connection. I tried core set debug 100 and core set verbose 100, but it doesn't say anything about rejecting the incoming IAX2 call. |
| 00:55.04 | p3nguin | iax2 set debug |
| 00:55.15 | ruben23 | p3nguin:ill add it up..but what do you mean dialing SIP/201 and SIP/211 in the match for 201---> what is this function..? |
| 00:55.25 | p3nguin | # |
| 00:55.26 | p3nguin | exten => 201,n,Dial(SIP/201&SIP/211,50,km) |
| 00:55.37 | ruben23 | i can dial both at the same time..? |
| 00:55.46 | p3nguin | If someone calls 201, that says to Dial SIP/201 and SIP/211 at the same time. |
| 00:55.54 | p3nguin | You took that out of my samples. |
| 00:56.07 | p3nguin | 201 is my desk phone, 211 is my cordless phone. |
| 00:56.27 | ruben23 | ok very usefull config |
| 00:56.34 | p3nguin | So if someone calls my desk phone at 201, this causes my desk phone to ring and my cordless phone to ring at the same time. |
| 00:56.34 | dcabot | did that and iax2 set debug trunk on as well. It doesn't even say that it's rejecting the call. |
| 00:57.57 | ruben23 | hmm..so my mm-this-call-monitor-record file is just then lacking |
| 00:58.19 | p3nguin | That's a file that I recorded for my IVR. I doubt you have a file by the same name. |
| 00:58.38 | p3nguin | How about... |
| 00:58.39 | ruben23 | yeah..i even dont know how to do a record |
| 00:59.06 | p3nguin | BackGround(thanks-for-calling-today) |
| 00:59.10 | p3nguin | Maybe you have that file. |
| 00:59.20 | ruben23 | ill try |
| 00:59.34 | p3nguin | You could just look and see if you have the file. |
| 01:00.40 | p3nguin | Ideally, you'll end up making your own recording (or have someone do it for you), and use it for your attendant. |
| 01:01.25 | ruben23 | p3nguin: i dont have it, what directory do voice prompt resides..? |
| 01:01.33 | p3nguin | Also, exten => _NXXZXXXXXX,3,MixMonitor ... this is going to record every incoming call. |
| 01:01.48 | p3nguin | On my systems, /var/lib/asterisk/sounds/ |
| 01:02.40 | p3nguin | If you don't want to record every incoming call, consider removing that line. |
| 01:03.35 | ruben23 | can i make it that ill be the one to do it if i want to record a particular call |
| 01:03.54 | p3nguin | If you do want to record all your calls, I assume TX has similar laws as ours stating that you are required to advise callers that you will be recording. |
| 01:04.34 | p3nguin | If you want to record some calls, you'll do that on a per-call basis. |
| 01:04.42 | ruben23 | yeah, but the dial plan record automatically right..? |
| 01:05.01 | *** join/#asterisk garrythefish (n=garrythe@unaffiliated/garrythefish) |
| 01:05.39 | garrythefish | its time to kick ass and chew bubble gum... |
| 01:05.40 | garrythefish | ... |
| 01:05.42 | garrythefish | and i'm all out of GUM |
| 01:05.44 | *** part/#asterisk garrythefish (n=garrythe@unaffiliated/garrythefish) |
| 01:06.39 | p3nguin | If you leave MixMonitor() in there, every call will be recorded. |
| 01:07.11 | ruben23 | ok ill omit it..ill do it on a per call basis. |
| 01:08.35 | ruben23 | where can i get more asterisk-sounds package |
| 01:09.56 | p3nguin | On my system, "yaourt -Ss asterisk" will list a bunch of possibilities. I'm sure yours is different. |
| 01:10.48 | p3nguin | Another thing. Once you remove that line where MixMonitor is, your priorities will no longer be correct. So fix the numbering, and then fix where 'i' takes you. |
| 01:11.36 | *** join/#asterisk jtodd (i=zhggneov@asterisk/community-director-and-tie-dye-shirt-lover/jtodd) |
| 01:11.36 | *** mode/#asterisk [+o jtodd] by ChanServ |
| 01:11.50 | ruben23 | yeah probably replaced it with background() abd download voice prompt form astersik |
| 01:12.15 | p3nguin | That could work. |
| 01:13.33 | ruben23 | <PROTECTED> |
| 01:13.46 | ruben23 | and also for meetme conference.. |
| 01:15.04 | p3nguin | park is park; has nothing to do with transfers. |
| 01:15.23 | p3nguin | You can enable parking in features.conf if you haven't already. |
| 01:15.56 | ruben23 | p3nguin: but need to have config on extensions.conf also../ |
| 01:17.58 | p3nguin | negative |
| 01:18.10 | p3nguin | parking is not done in extensions.conf. |
| 01:18.17 | p3nguin | features.conf |
| 01:18.39 | ruben23 | p3nguin: ok |
| 01:19.56 | ruben23 | <PROTECTED> |
| 01:23.05 | [TK]D-Fender | p3nguin: [20:15]<p3nguin>park is park; has nothing to do with transfers. <- sure it does |
| 01:23.52 | *** join/#asterisk wierdo (n=jimmy@77.78.3.197) |
| 01:25.18 | Godfather_ | [TK]D-Fender, this is a valid register??? http://pastebin.com/m457ade3f |
| 01:25.51 | [TK]D-Fender | Godfather_: No, that is a NOTIFY, not a register... just like it says |
| 01:26.57 | p3nguin | Is there a way to configure automon to use MixMonitor instead of Monitor, and a way to specify the file naming convention? |
| 01:27.24 | *** join/#asterisk jasonpr (n=jasonpr@216.58.247.235) |
| 01:29.15 | Godfather_ | [TK]D-Fender, can you revise my files? i advanced, it seems that im registered (xlite says that), but cant call to another extension |
| 01:29.17 | Godfather_ | :/ |
| 01:33.34 | Godfather_ | http://pastebin.com/m16ef4a0a ? |
| 01:38.48 | dcabot | Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW |
| 01:38.48 | dcabot | <PROTECTED> |
| 01:38.48 | dcabot | <PROTECTED> |
| 01:38.48 | dcabot | <PROTECTED> |
| 01:38.48 | dcabot | <PROTECTED> |
| 01:38.48 | dcabot | <PROTECTED> |
| 01:38.50 | dcabot | <PROTECTED> |
| 01:38.52 | dcabot | <PROTECTED> |
| 01:38.54 | dcabot | <PROTECTED> |
| 01:38.56 | dcabot | <PROTECTED> |
| 01:38.58 | dcabot | <PROTECTED> |
| 01:39.01 | dcabot | <PROTECTED> |
| 01:39.02 | dcabot | <PROTECTED> |
| 01:39.04 | dcabot | <PROTECTED> |
| 01:39.06 | dcabot | <PROTECTED> |
| 01:39.08 | dcabot | <PROTECTED> |
| 01:39.10 | dcabot | FreePBX-1*CLI> |
| 01:39.19 | dcabot | Shouldn't the TON, presntn, and transit be other than zero? |
| 01:40.57 | p3nguin | dcabot: Was there something wrong with the pastebin that you HAD to paste all that here in the channel? |
| 01:41.10 | *** join/#asterisk Mw3_ (i=mw3@mw3.hu) |
| 01:42.03 | _ShrikE | ! |
| 01:42.18 | Godfather_ | [TK]D-Fender, this is what i get when im tryin to call 'cc102' from 'cc101', http://pastebin.com/m64fcaa30 |
| 01:42.47 | carrar | sexy |
| 01:43.16 | dcabot | Oops. Sorry. http://pastebin.com/d615f0d5c |
| 01:44.02 | [TK]D-Fender | Godfather_: Where do you have a "cc102" in your dialplan? |
| 01:44.22 | *** join/#asterisk ardor (n=IceChat7@ip72-193-201-128.lv.lv.cox.net) |
| 01:45.14 | Godfather_ | [TK]D-Fender, http://pastebin.com/m77996b4e |
| 01:45.19 | Godfather_ | at the bottom |
| 01:50.06 | Godfather_ | wich command says active registered clients ? sip show subscriptions ? |
| 01:51.07 | Godfather_ | [TK]D-Fender, http://pastebin.com/meb561c8 |
| 01:51.30 | [TK]D-Fender | Godfather_: that is SIP.CONF. I said in your DIALPLAN |
| 01:52.22 | [TK]D-Fender | Godfather_: extensions.conf <------------------------- |
| 01:52.30 | *** join/#asterisk jthurman42 (n=jthurman@c-67-169-218-181.hsd1.wa.comcast.net) |
| 01:52.41 | Godfather_ | ok |
| 01:54.47 | Godfather_ | [TK]D-Fender, actually, i didnt change that file . |
| 01:55.50 | [TK]D-Fender | Godfather_: Looking for cc102 in default (domain 10.1.1.101) SIP/2.0 404 Not Found --- Well the call is looking for "cc102" in [default] and it doesn't find a match. Go fix it |
| 01:56.40 | Godfather_ | Hummm... |
| 01:58.09 | *** join/#asterisk chendy (n=chatzill@116.30.194.230) |
| 01:58.39 | Godfather_ | [TK]D-Fender, http://pastebin.com/mcbb7710 |
| 01:58.57 | dcabot | I don't see any password in the IAX2 new call packet. How does the incoming call authenticate? |
| 01:59.44 | Godfather_ | [TK]D-Fender, actually, i dont know exactly how to write it, can you put me an extension for the cc102 .. |
| 02:00.58 | [TK]D-Fender | Godfather_: exten => cc102,1,NoOp(Hopefully a clue isn't far away...) |
| 02:01.04 | carrar | you don't need authentication, you can trust the innerweb |
| 02:03.18 | *** join/#asterisk chuckf (n=chuckf@ubuntu/member/chuckf) |
| 02:03.53 | *** join/#asterisk voipmonk (n=voipmonk@69.172.69.111) |
| 02:04.42 | Godfather_ | [TK]D-Fender, this is my extensions.conf http://pastebin.com/m3ac475ee , its correct? |
| 02:05.17 | Godfather_ | (the entire file), i removed the examples now. |
| 02:05.18 | [TK]D-Fender | Godfather_: why don't you go try and SEE |
| 02:05.34 | Godfather_ | Call failed, Not found |
| 02:05.51 | carrar | heh |
| 02:06.16 | *** join/#asterisk brut- (n=brut-@h66-173-4-254.mntimn.dedicated.static.tds.net) |
| 02:06.27 | carrar | flips the ham raido on for additional entertainment |
| 02:06.36 | [TK]D-Fender | Godfather_: "dialplan show" <- pastebin it along with your failed call attempt |
| 02:08.00 | Godfather_ | [TK]D-Fender, http://pastebin.com/m48668aa7 |
| 02:08.55 | p3nguin | carrar: What band? |
| 02:09.01 | [TK]D-Fender | Godfather_: your extensions.conf is not being loaded |
| 02:09.08 | carrar | 75 |
| 02:11.05 | Godfather_ | [TK]D-Fender, http://pastebin.com/m487e8a96 |
| 02:11.09 | carrar | listening to some net on 3.937 |
| 02:11.46 | Godfather_ | i think something is missing in my extensions.conf :s |
| 02:11.56 | *** join/#asterisk JAMMAN2110 (n=james@unaffiliated/jamman2110) |
| 02:12.06 | [TK]D-Fender | Godfather_: 'cc102' => 1. NoOp(Hopefully a clue isn't far away...) [pbx_config] |
| 02:12.15 | [TK]D-Fender | Godfather_: No, I think you forgot to APPLY your changes |
| 02:12.20 | [TK]D-Fender | Godfather_: its there now |
| 02:13.06 | Godfather_ | [TK]D-Fender, i dial again. |
| 02:13.41 | Godfather_ | [TK]D-Fender, changed to: Call failed, declined |
| 02:13.54 | Godfather_ | i go to the sip set debug |
| 02:13.55 | [TK]D-Fender | Godfather_: PASTEBIN... |
| 02:15.37 | Godfather_ | http://pastebin.com/m7937c7eb |
| 02:16.31 | [TK]D-Fender | Godfather_: "core set verbose 10 |
| 02:16.34 | [TK]D-Fender | Godfather_: then retry |
| 02:17.46 | Godfather_ | [TK]D-Fender, http://pastebin.com/m6cdaa5 |
| 02:18.02 | [TK]D-Fender | Godfather_: -- Executing [cc102@default:1] NoOp("SIP/cc101-024c2168", "Hopefully a clue isn't far away...") in new stack |
| 02:18.24 | [TK]D-Fender | Godfather_: congratulations, this is the first call you've even placed that wasn't completely ignored by * |
| 02:18.37 | [TK]D-Fender | Godfather_: now go DO SOMETHING with it |
| 02:18.55 | Godfather_ | i want call cc101 to cc102... |
| 02:19.10 | [TK]D-Fender | Godfather_: Fine... this one I'll just hand to you.... |
| 02:19.28 | [TK]D-Fender | Godfather_: exten => cc102,1,Dial(SIP/cc102,20) |
| 02:20.00 | dcabot | saftsack: http://www.camrivox.com/products/flexor-cti-outlook/ |
| 02:21.43 | Godfather_ | [TK]D-Fender, impresive!! |
| 02:22.08 | Godfather_ | :)) |
| 02:22.25 | Godfather_ | now is working :D |
| 02:23.04 | [TK]D-Fender | Godfather_: Last piece before I head out for a while : All call processing = extensions.conf. The dialplan says what happens when you call an extension. there is no "just call XYZ". You have to tell it exactly what instructions to follow to do ANYTHING at all. |
| 02:23.25 | [TK]D-Fender | Godfather_: The dialplan is 95% of asterisk. Master ir |
| 02:23.35 | [TK]D-Fender | Godfather_: Now go read the book. |
| 02:23.37 | [TK]D-Fender | it* |
| 02:23.39 | [TK]D-Fender | ~book |
| 02:23.40 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 02:23.41 | [TK]D-Fender | ^^^^^^ |
| 02:23.50 | [TK]D-Fender | heads off for a bit |
| 02:24.19 | Godfather_ | [TK]D-Fender, thx ^ 999 |
| 02:37.52 | *** join/#asterisk joako (n=ston3d@opensuse/member/joak0) |
| 02:39.06 | *** join/#asterisk [8none1] (n=[8none1]@c-68-52-180-102.hsd1.tn.comcast.net) |
| 02:39.34 | *** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler) |
| 02:40.10 | *** part/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler) |
| 02:44.21 | *** join/#asterisk superbeef (n=superbee@74.84.194.4) |
| 02:46.01 | *** join/#asterisk Iamnach0 (i=Iamnacho@ip98-186-180-143.ks.ks.cox.net) |
| 03:03.06 | *** join/#asterisk Qwell (i=north@pdpc/sponsor/digium/Qwell) |
| 03:03.06 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 03:22.06 | *** join/#asterisk coppice (n=chatzill@171.168.17.210.dyn.pacific.net.hk) |
| 03:27.38 | *** join/#asterisk sizzers (n=sizzers@75.119.125.63) |
| 03:28.25 | sizzers | hi all, 1.6.0.17 is newer than 1.6.0.5 correct? i'm settling an argument here |
| 03:33.32 | Godfather_ | 17 > 5, then yes, it isnt ? |
| 03:34.45 | *** join/#asterisk voipmonk (n=voipmonk@69.172.69.111) |
| 03:35.20 | sizzers | his argument was that 5 was greater than 1 |
| 03:35.33 | Qwell | it is |
| 03:35.41 | sizzers | i told him already, my hair is a bird , your argument is invalid |
| 03:35.44 | p3nguin | but 5 is less than 17. |
| 03:35.51 | Qwell | depends how you sort |
| 03:36.11 | coppice | we all know about your sort |
| 03:36.21 | Qwell | sizzers: Keep him away from your servers. |
| 03:36.24 | sizzers | right qwell, that could be interpreted both ways |
| 03:41.07 | [TK]D-Fender | sizzers: yes.. he probably goes both ways.... |
| 03:41.38 | [TK]D-Fender | sizzers: Both lead to getting fucked :) |
| 03:41.38 | ricko73 | gah! what a fucked up week |
| 03:41.38 | sizzers | LMAo lool |
| 03:41.40 | sizzers | the sense of humor is strong in here tonight, i like it |
| 03:41.41 | ricko73 | is anyone else in here going to admit they're using Teliax? |
| 03:41.55 | sizzers | no |
| 03:41.57 | ricko73 | for some reason what worked a few days ago, is yet again broken |
| 03:41.59 | sizzers | looks around warily |
| 03:42.02 | p3nguin | I'll admit I've considered it in the past. |
| 03:43.40 | ricko73 | what I had working with asterisk 1.4 now won't work with 1.6.1.9 |
| 03:43.48 | ricko73 | (even though it worked before) |
| 03:44.03 | ricko73 | I see the sip traffic, but rtp is not being passed |
| 03:44.15 | p3nguin | Check your ports. |
| 03:44.20 | ricko73 | I hear 'white noise' on my phone |
| 03:46.44 | *** join/#asterisk Linuturk (n=linuturk@unaffiliated/linuturk) |
| 03:48.02 | ricko73 | nope. firewall has ports the ports open |
| 03:49.22 | jblack | One (or both) of two things are happening, one one (or both) of the two sides.... a firewall is eating the packets, or the sip packets lied about where packets should be sent. |
| 03:49.49 | ricko73 | well my asterisk box has a public IP address there is no nat involved |
| 03:50.00 | ricko73 | I have 5060 and 10000-2000 udp open |
| 03:50.33 | jblack | Dont' "well" me. That's it, regardless of any attempts at reason on your part. |
| 03:50.45 | jblack | Unless you want to argue someone stole your speaker and/or microphone. |
| 03:50.53 | *** join/#asterisk Linuturk (n=linuturk@unaffiliated/linuturk) |
| 03:51.03 | ricko73 | lose the attitude |
| 03:51.16 | jblack | Or else what? |
| 03:51.31 | ricko73 | it's just not becoming ;) |
| 03:51.38 | p3nguin | Children, please... |
| 03:51.43 | jblack | Good, because I don't want to become anything. |
| 03:52.00 | jblack | well, almost. I'd like to become laid, but that's not in the cards tonight. |
| 03:52.12 | ricko73 | well... ;) |
| 03:52.23 | jblack | Though I seem to be fucking you pretty good. |
| 03:52.26 | jblack | Next time, wear a skirt |
| 03:52.51 | ricko73 | wow. I guess the high school kids are out now |
| 03:53.04 | jblack | They probably are, it being saturday night in the states. |
| 03:53.05 | ricko73 | thanks for the laugh. |
| 03:53.34 | jblack | Seriously though. Excluding truly silly reasons, either the packets are being dropped on the floor, or misdirected. |
| 03:54.41 | ricko73 | I understand that |
| 03:54.52 | jblack | cool |
| 03:55.14 | ricko73 | I was simply stating that I had the ports open and rtp.conf is configured to use those ports AND I do not have a nat issue |
| 03:55.17 | ricko73 | that's it |
| 03:55.21 | jblack | can I go back to having a surly attitude, wearing all black, and bitching that life ain't no fair? |
| 03:55.27 | ricko73 | you're the one who call all up in my face about don't 'well' me |
| 03:56.44 | [TK]D-Fender | ricko73: Describe the complete call path |
| 03:57.20 | ricko73 | Teliax --> Asterisk --> Polycom |
| 03:57.37 | ricko73 | I can make outbound calls successfully |
| 03:57.48 | jblack | So, your polycom, that has a publicly addressible ip? |
| 03:57.55 | ricko73 | no the Asterisk box does |
| 03:58.00 | [TK]D-Fender | ricko73: Try again... |
| 03:58.32 | ricko73 | how do you want me to describe the call path? |
| 03:58.56 | jblack | sigh. ricko... like I said twice already.... Try this out.. for the polycom sip account, try canreinvite=no |
| 03:59.36 | jblack | (to alleviate confusion, I said the issue was nat/firewall twice) |
| 03:59.55 | [TK]D-Fender | ricko73: Apparently there is no netowrking inbetween anything in your world... |
| 04:00.03 | ricko73 | jblack: the problem is before it's even getting to my polycom |
| 04:00.08 | jblack | [TK]D-Fender: What do you expect? The guy think's there's no nat. |
| 04:00.29 | jblack | Well, maybe there's not. maybe the phone isn't natted, and it's a straight network. |
| 04:00.45 | jblack | dual homed *, non-natted for the local net. Same difference though. |
| 04:01.01 | ricko73 | I have an background message that's played on inbound calls |
| 04:01.12 | ricko73 | I do no hear that played back in my cell phone when I call in |
| 04:01.17 | jblack | ricko73: Again, turn on canreinvite=no in the appropriate polycom account in sip.conf |
| 04:01.29 | ricko73 | jblack: the polycom doesn't even come into play here |
| 04:01.40 | ricko73 | forget about the polycom |
| 04:01.49 | ricko73 | the problem is between Teliax and Asterisk |
| 04:01.55 | [TK]D-Fender | ricko73: Going to show us anything anytime soon? |
| 04:01.58 | ricko73 | or something between the two |
| 04:02.10 | ricko73 | [TK]D-Fender: what exactly do you want me to show you |
| 04:02.18 | [TK]D-Fender | ricko73: You can't think of anything? |
| 04:02.36 | jblack | [TK]D-Fender: Are we role reversing tonight? Are you the patient one? |
| 04:02.50 | [TK]D-Fender | jblack: No, p3nguin filled that role earlier |
| 04:03.02 | ricko73 | when you want me to describe the call path, what exactly do you want me to show you? |
| 04:03.16 | jblack | !debug |
| 04:03.28 | jblack | I must be thinking of another channel |
| 04:03.45 | [TK]D-Fender | ricko73: the broken call, configs, firewall dump, etc... |
| 04:03.56 | jblack | Why don't you turn on debugging for sip, collect an entire call, and paste that at pastebin. Alon... |
| 04:04.02 | ricko73 | ok |
| 04:04.03 | [TK]D-Fender | ricko73: Seriously.. no clue? Show me the damned dead body if you want an autopsy |
| 04:04.23 | ricko73 | lol |
| 04:04.28 | ricko73 | you guys are something tonight |
| 04:04.58 | jblack | ricko73: You're about 5 seconds from being ignored by everyone in the channel. I suggest you not be critical of the help you receive until you feel done receiving it. |
| 04:05.48 | jblack | [TK]D-Fender: My bet is 192.168.2.0/24 . Which network do you think he's using? |
| 04:05.54 | ricko73 | it's not my first time in the channel |
| 04:06.26 | [TK]D-Fender | ricko73: Which means we should have to tell you to wake up and show us your settings and errors.. right? |
| 04:06.42 | jblack | typo? |
| 04:06.42 | [TK]D-Fender | ricko73: module unload chan_codependence.so |
| 04:06.50 | ricko73 | no but I misunderstood what you wanted for a call path description |
| 04:06.53 | ricko73 | sorry |
| 04:07.10 | jblack | why are you still talking? You should be busy |
| 04:07.25 | jblack | Hello, world. |
| 04:07.29 | [TK]D-Fender | ricko73: Oh, I'm past your first "oversight" and am on to lambasting you for this greater one :) |
| 04:08.32 | jblack | I bet that he doesnt' have reinvite off, and packets from teliax are being sent to 192.168.1.0/24 |
| 04:09.18 | jaytee | lambasting........mmmmmm, nothing tastier than basted lamb |
| 04:09.34 | jblack | Lamb does sound good. |
| 04:11.11 | sizzers | asking for help here is like asking for soup from the soup nazi. if you don't got your order straight and perfect when your turn is up, NO SOUP FOR YOU!!! |
| 04:11.33 | sizzers | that's with IRC in general ricko, so resubmit your question in 1 hour |
| 04:11.45 | ricko73 | jblack: I bet I do have 'canreinvite=no' |
| 04:12.22 | sizzers | it's very unlikely that you'll get any help right now ricko, best fade into obscurity and try again when everybody forgets about the past 10 minutes |
| 04:13.14 | jblack | He'll probably get somewhere once he pastes a call trace and his configs. |
| 04:13.21 | dcabot | Looks like IAX2 changed behavior somewhere after 1.4.24. Problem DNE there. |
| 04:14.31 | *** join/#asterisk voipmonk (n=voipmonk@69.172.69.111) |
| 04:16.36 | Godfather_ | [TK]D-Fender, i dont want to be boring, but i'm just able to be make from cc101 to cc102,on the other way doesnt work (isnt available), any idea? http://pastebin.com/m1cf44353 http://pastebin.com/m675198e5 |
| 04:17.26 | [TK]D-Fender | Godfather_: Because you have no concept on PRIORITIES |
| 04:18.27 | Godfather_ | [TK]D-Fender, its true, but i tried exten => cc101,2,Dial(SIP/cc101,20) with just this line and doesnt work |
| 04:18.34 | dcabot | Definitely happened between 1.4.24 and 1.4.26. IAX2 calls aren't accepted anymore in 1.4.26 with the simple config that I have. |
| 04:18.50 | [TK]D-Fender | Godfather_: 1, not 2 <- |
| 04:19.29 | Godfather_ | [TK]D-Fender, dialing to cc102 works, but i want to dial to cc101 |
| 04:19.49 | [TK]D-Fender | Godfather_: 1, not 2 <- |
| 04:20.06 | [TK]D-Fender | Godfather_: your CC101 exten does not have a PRIORITY 1 |
| 04:20.15 | p3nguin | lol |
| 04:27.18 | Godfather_ | [TK]D-Fender, sorry but nope... see that http://pastebin.com/m171a3d1b |
| 04:27.33 | Godfather_ | i did dialplan show before,as you can see |
| 04:28.24 | [TK]D-Fender | Godfather_: you seem to have extreme difficulty APPLYING YOUR CHANGES |
| 04:28.50 | dcabot | I'm trying to revert the asterisk packages from 1.4.26 to 1.4.24, but yum keeps saying 1.4.24 is still installed. |
| 04:29.20 | Godfather_ | [TK]D-Fender, ? |
| 04:30.02 | Godfather_ | tessio*CLI> dialplan reload |
| 04:30.03 | Godfather_ | Dialplan reloaded. |
| 04:30.03 | [TK]D-Fender | Godfather_: ..."reload" <- |
| 04:30.26 | Godfather_ | its reloaded, and i tried stoping astering and starting it. |
| 04:30.39 | [TK]D-Fender | Godfather_: I and I told you 3 times already.... |
| 04:30.49 | [TK]D-Fender | Godfather_: exten => cc101,2,Dial(SIP/cc101,20) <---------- this needs to be priority ONE |
| 04:30.54 | [TK]D-Fender | Godfather_: not TWO |
| 04:31.08 | Godfather_ | root@tessio:/etc/asterisk# cat extensions.conf |
| 04:31.08 | Godfather_ | [default] |
| 04:31.08 | Godfather_ | ;exten => cc102,1,Dial(SIP/cc102,20) |
| 04:31.09 | Godfather_ | exten => cc101,1,Dial(SIP/cc101,20) |
| 04:31.09 | Godfather_ | ;exten => cc102,1,Playback(rock) |
| 04:32.40 | Godfather_ | just changing 101 to 102 and dialplan reload WORKS, but this way NO |
| 04:32.55 | [TK]D-Fender | Godfather_: and there you figgen commented out the other 2 lines |
| 04:33.21 | [TK]D-Fender | [23:31]<Godfather_>;exten => cc102,1,Dial(SIP/cc102,20) <- Not gonna get loaded! |
| 04:33.22 | Godfather_ | I erase it if you want Ok |
| 04:38.40 | *** join/#asterisk h00k (n=anthonyr@unaffiliated/h00k) |
| 04:38.42 | *** part/#asterisk h00k (n=anthonyr@unaffiliated/h00k) |
| 04:39.43 | Godfather_ | [TK]D-Fender, http://pastebin.com/m4199efb1 |
| 04:39.45 | *** join/#asterisk Iamnacho (i=Iamnacho@ip98-186-180-143.ks.ks.cox.net) |
| 04:40.10 | *** join/#asterisk Nugget (i=nugget@carrera.macnugget.org) |
| 04:41.07 | [TK]D-Fender | Godfather_: sip show peers <- |
| 04:41.34 | Godfather_ | [TK]D-Fender, http://pastebin.com/m280313a0 |
| 04:43.00 | Godfather_ | if i change the extensions.conf to exten => cc102,1,Dial(SIP/cc102,20) and reload, it works |
| 04:45.33 | [TK]D-Fender | Godfather_: cc101/cc101 (Unspecified) D 5060 Unmonitored <--- not registered. You can;t call this device |
| 04:46.21 | *** join/#asterisk cellofellow (n=cellofel@69-71-167-33.mammothnetworks.com) |
| 04:48.12 | Godfather_ | [TK]D-Fender, on x-lite it says 'registered' |
| 04:48.26 | Godfather_ | and says Your username is: cc101 |
| 04:48.27 | [TK]D-Fender | * begs to differ |
| 04:48.53 | Godfather_ | the ip-address is 10.1.1.2 (cc101), can i add this manually? |
| 04:51.09 | Godfather_ | [TK]D-Fender, http://img509.imageshack.us/img509/8738/xlite.png |
| 04:51.54 | [TK]D-Fender | Godfather_: restart X-Lite |
| 04:57.50 | Godfather_ | [TK]D-Fender, the same, i tried also downloading a newer version, same result |
| 04:58.02 | Godfather_ | sip show peers not showing the cc101 ip |
| 04:58.09 | Godfather_ | but says 2 clients online |
| 04:58.53 | [TK]D-Fender | Godfather_: restart it, check the SIP DEBUG, and then "sip show peers" |
| 04:59.19 | Godfather_ | i restarted |
| 04:59.37 | *** join/#asterisk moy (n=moy@CPE001cdfec4cee-CM00080dab8485.cpe.net.cable.rogers.com) |
| 05:00.48 | Godfather_ | [TK]D-Fender, there is something strange... |
| 05:00.58 | Godfather_ | i disable both accounts on xlite |
| 05:01.07 | Godfather_ | and then i stopped asterisk service |
| 05:01.17 | Godfather_ | and still disabled cc101 and cc102 |
| 05:01.30 | Godfather_ | then i started asterisk, and did sip show peers |
| 05:01.58 | p3nguin | It's hard for me to dial letters on my phone. Care to explain why you chose such an odd schema? |
| 05:02.06 | Godfather_ | http://pastebin.com/m57dbcb6d |
| 05:02.22 | Godfather_ | that is the previuos sip show peers |
| 05:02.33 | Godfather_ | not chaning... |
| 05:02.56 | [TK]D-Fender | Godfather_: Are all 3 "host=dynamic" in sip.conf? |
| 05:03.13 | Godfather_ | [TK]D-Fender, yes |
| 05:03.40 | Godfather_ | http://pastebin.com/m5b17a24b |
| 05:03.43 | [TK]D-Fender | Godfather_: database deltree SIP |
| 05:03.49 | [TK]D-Fender | Godfather_: then "sip show peers" |
| 05:04.25 | Godfather_ | http://pastebin.com/m847ffeb |
| 05:04.48 | Godfather_ | both account still disabled |
| 05:05.12 | *** part/#asterisk cellofellow (n=cellofel@69-71-167-33.mammothnetworks.com) |
| 05:05.57 | Godfather_ | would you like to 'prove' if i put host=10.1.1.2 (in cc101) if it work? |
| 05:06.09 | Godfather_ | instead of dynamic |
| 05:07.14 | Godfather_ | [TK]D-Fender, i disabled the firewall too |
| 05:08.56 | [TK]D-Fender | Godfather_: no |
| 05:10.24 | *** join/#asterisk bbt (n=sam@180.189.138.55) |
| 05:14.07 | *** join/#asterisk teknoprep (n=Chris@unaffiliated/teknoprep) |
| 05:14.20 | teknoprep | hi all... can the polycom ip 650 usb port be used for a bluetooth adapter ? |
| 05:14.34 | teknoprep | if so where the hell do i buy one that works with it ? |
| 05:15.15 | Godfather_ | [TK]D-Fender, then? |
| 05:15.37 | teknoprep | [TK]D-Fender, he would know.. he knows everything |
| 05:15.38 | teknoprep | lol |
| 05:26.14 | *** join/#asterisk Hyphenex (n=Hyphenex@vps17.vz7.ramhost.us) |
| 05:30.36 | *** join/#asterisk Faithful (n=Faithful@115.147.205.62) |
| 05:31.04 | *** join/#asterisk Trupsalms (n=shawndel@adsl-68-20-35-251.dsl.chcgil.ameritech.net) |
| 05:31.20 | Trupsalms | help |
| 05:32.38 | Godfather_ | [TK]D-Fender, was a problem of X-lite |
| 05:32.51 | Godfather_ | i used sjsoftphone instead and this one works |
| 05:33.01 | teknoprep | what issue are you having ? |
| 05:33.09 | teknoprep | i have never had an issue with x-lite |
| 05:33.17 | Trupsalms | i have asterisk installed on ubuntu |
| 05:33.24 | teknoprep | cheers |
| 05:33.54 | Godfather_ | teknoprep, 2 peers, and can make calls only in one way |
| 05:34.00 | Trupsalms | i need help installing the asterisk-gui |
| 05:34.05 | Trupsalms | on ubuntu |
| 05:34.14 | teknoprep | Godfather_, sounds like a NAT issue |
| 05:34.33 | Godfather_ | teknoprep, nope, i installed sjsoftphone and now works on both |
| 05:34.41 | Trupsalms | help |
| 05:34.42 | Godfather_ | (both softphones are on the same subnet) |
| 05:34.42 | teknoprep | Godfather_, there is too much NAT compensation going on these days |
| 05:34.43 | Trupsalms | help |
| 05:34.53 | teknoprep | Godfather_, where is the PBX > |
| 05:35.02 | Godfather_ | teknoprep, in the same subnet :P |
| 05:35.14 | teknoprep | Godfather_, don't know what to tell you.. wish i was there |
| 05:35.20 | [TK]D-Fender | Trupsalms: GUI's are NOT supported in this channel -> #asterisk-gui |
| 05:35.32 | Godfather_ | [TK]D-Fender, thx... |
| 05:35.34 | teknoprep | Trupsalms, stfu |
| 05:35.35 | Trupsalms | comeon |
| 05:35.36 | ricko73 | [TK]D-Fender: fyi, I just had someone call me via direct sip uri to my asterisk box and the call was passed properly. Whatever is happening on my incoming calls from Teliax is likely a configuration issue on their side |
| 05:35.36 | teknoprep | lol |
| 05:35.38 | Godfather_ | sorry to be so boring |
| 05:35.45 | teknoprep | dude spamming help will not get you help |
| 05:36.00 | Trupsalms | what make gui no supported |
| 05:36.06 | Trupsalms | what channel then |
| 05:36.07 | teknoprep | Trupsalms, you could try asking the quest... and waiting for an answer |
| 05:36.15 | Trupsalms | because it is still asterisk |
| 05:36.26 | ricko73 | Trupsalms: #asterisk-gui |
| 05:36.48 | Godfather_ | teknoprep, yep, this was very strange, the same version of xlite on both pcs, and calls just go in one way, but with sjphone works ok |
| 05:37.12 | Hyphenex | hey, my friends dad has 3 telephone lines atm, and he wants to move to an Asterisk box. What phone cards would I be looking at getting to support these three lines? |
| 05:37.13 | teknoprep | probably something over compensating for NAT |
| 05:37.26 | teknoprep | just a bad check box somewhere |
| 05:37.42 | teknoprep | did u try turning stun off ? |
| 05:37.46 | Godfather_ | teknoprep, i revised about 20 times.. and reinstalled it jeje |
| 05:37.48 | Godfather_ | no idea :| |
| 05:37.55 | Godfather_ | humm |
| 05:38.16 | teknoprep | i don't think i ever called another local x-lite client local to me before |
| 05:38.21 | teknoprep | i only use that when on the road |
| 05:38.33 | ricko73 | Hyphenex: a tdm410 card would work |
| 05:38.35 | Godfather_ | teknoprep, well, i am a noob. |
| 05:38.41 | teknoprep | i am too |
| 05:38.43 | teknoprep | we all are |
| 05:38.49 | teknoprep | except for [TK]D-Fender |
| 05:39.04 | Godfather_ | he is a guru |
| 05:39.07 | Hyphenex | Thanks ricko73 :) |
| 05:39.09 | Hyphenex | I'll look into it |
| 05:39.17 | Godfather_ | teknoprep, stun server is actived yes |
| 05:39.27 | teknoprep | try turning it off on both clients |
| 05:39.35 | teknoprep | and then exit the program and reopen it |
| 05:39.39 | teknoprep | then make the call |
| 05:39.51 | Godfather_ | the program you are saying is xlite on both pcs no? |
| 05:39.57 | teknoprep | yup |
| 05:40.08 | Godfather_ | Stun server -> set to 'Discover server' ? |
| 05:40.12 | Trupsalms | help |
| 05:40.18 | Trupsalms | asterisk-gui |
| 05:40.19 | teknoprep | i would want it off |
| 05:40.24 | Trupsalms | help asterisk-gui |
| 05:40.31 | teknoprep | since you are local |
| 05:41.13 | Godfather_ | teknoprep, two options, discover server or use specific server. |
| 05:41.22 | *** join/#asterisk newbieeeeeee (n=root@201-42-8-65.dsl.telesp.net.br) |
| 05:41.36 | teknoprep | use specific server and put the IP address of your PBX in it |
| 05:41.40 | teknoprep | see if that works |
| 05:41.41 | newbieeeeeee | hello there!!! |
| 05:41.57 | teknoprep | you probably are not running a stun server but at least it might time out |
| 05:41.57 | Godfather_ | teknoprep, thats what i have now |
| 05:41.58 | newbieeeeeee | trying to compile asterisk, but still stuck on libpri... |
| 05:42.08 | teknoprep | Godfather_, then put discover server |
| 05:42.13 | newbieeeeeee | is there any pre-requiriment i am missing? |
| 05:42.14 | Godfather_ | ok |
| 05:42.32 | teknoprep | Godfather_, brb need a smoke |
| 05:44.05 | newbieeeeeee | getting some compiler error while trying to compile libpri.. some erros like gcc -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -g -fPIC -MD -MT copy_string.o -MF .copy_string.o.d -MP -c -o copy_string.o copy_string.c |
| 05:44.16 | Hyphenex | ok, I don't get the difference between FXS and FXO ports |
| 05:44.21 | newbieeeeeee | has anyone experienced such a errors? |
| 05:44.25 | Godfather_ | teknoprep, nope, doesnt work hhe |
| 05:44.26 | Godfather_ | ok |
| 05:44.29 | [TK]D-Fender | ~fxofxs |
| 05:44.30 | infobot | i heard fxofxs is An FXO port (red Digium module) expects to receive dialtone and receive ring voltage. You can connect it to a PSTN line from the telco. An FXS port (green Digium module) expects to provide dialtone and provide ring voltage. You can connect a phone or a fax to it. |
| 05:45.03 | Hyphenex | oh cool, Thanks :) |
| 05:47.08 | Godfather_ | [TK]D-Fender, thx for supporting, best support i've ever had seriusly |
| 05:48.08 | Hyphenex | ricko73: so the TDM410 can be both FX0 and FXS?? |
| 05:48.26 | teknoprep | Godfather_, i can help you out another time man.. i am beat.. going to bed... i'll load up x-lite and try it at home |
| 05:48.36 | *** join/#asterisk ethicx (n=geneticx@adsl-10-104-182.mia.bellsouth.net) |
| 05:48.45 | Godfather_ | teknoprep, no problem, seems to be a X-lite problem.. |
| 05:48.59 | Godfather_ | i'll search another version and install it to another directory |
| 05:49.05 | Godfather_ | dont worry about it |
| 05:49.23 | Godfather_ | i have bigger problems than chossing a sip client jeje |
| 05:49.24 | Godfather_ | :) |
| 05:50.06 | Godfather_ | here is 6:49 am, no sleep |
| 05:50.09 | ricko73 | Hyphenex: it depends which modules you put on the card |
| 05:50.30 | ricko73 | you can put up to 4 modules in any combination of fxs or fxo |
| 05:50.39 | Godfather_ | teknoprep, goodnight! |
| 05:51.15 | Hyphenex | ricko73: So if I wanted to forward the three physical lines to VoIP lines, and have one telephone connected to the card, I'd get the TDM413E (1 FXS Ports, 3 FXO Ports / Hardware Echo Cancellation)? |
| 05:51.44 | ricko73 | well you wouldn't really be 'forwarding' them to voip lines... |
| 05:52.02 | ricko73 | you'd be connecting them to the TDM410 card on fxo ports |
| 05:52.54 | ricko73 | the HW echo cancellation is optional. It's nice if you can afford it, but not required in all cases |
| 05:53.41 | [TK]D-Fender | Hyphenex: And forego getting FXS modules and get an ATA for your phones. Cheaper and more flexible |
| 05:53.42 | [TK]D-Fender | ~ATA |
| 05:53.43 | infobot | somebody said ata was Analogue Terminal Adapter which provides an FXS and/or FXO and ethernet, see http://www.voip-info.org/wiki/view/ATA |
| 05:53.56 | [TK]D-Fender | Hyphenex: eg : Linksys PAP2T-NA |
| 05:58.27 | Hyphenex | I'm so confused right now |
| 06:01.08 | *** part/#asterisk Trupsalms (n=shawndel@adsl-68-20-35-251.dsl.chcgil.ameritech.net) |
| 06:01.23 | Hyphenex | we want to replace their phones with VoIP phones (so we can set up an IAX trunk to their home), so if they've got three telephone lines from their provider, throwing out their current phones, I'd need three FS0 ports, right? |
| 06:06.08 | [TK]D-Fender | Hyphenex: You mean replace their LINES, no? |
| 06:06.45 | Hyphenex | nahh, they want to keep their physical lines, just go into VoIP so they can forward their calls and stuff |
| 06:07.49 | [TK]D-Fender | Hyphenex: You mean be able to work from home basically then? |
| 06:08.01 | Hyphenex | yep, that too :) |
| 06:09.01 | [TK]D-Fender | Hyphenex: Ok, well first you won't be wanting IAX2 phones... all the ones out there suck. Second, You won't be needing any FXS it seems, just FXO (not ZERO) |
| 06:09.17 | *** join/#asterisk geneticx (n=geneticx@adsl-10-104-182.mia.bellsouth.net) |
| 06:10.10 | Hyphenex | okies, so just a card that supports FXO to do inbound and outbound on the physical lines :) |
| 06:10.31 | Hyphenex | we'd use SIP for the phones, but an IAX trunk between home and office (so we'll install two asterisk boxes in) |
| 06:11.13 | [TK]D-Fender | Hyphenex: Putting extra boxes at the remote sites jsut to support 1 phone is crazy BTW |
| 06:12.30 | Hyphenex | it's cool, it's a mighty big house, they want 6 phones at their home, and three phones at their office |
| 06:17.34 | Hyphenex | if we were to do the same thing with his home (but there's only one physical line here), could we get a card or box or something with one FX0 port? oh, and can I still make phone calls through the card/ATA with an FX0 port? |
| 06:17.54 | carrar | You'll need a server for each phone at the house |
| 06:17.57 | jblack | oh man.. they're gonna want the wiring in the walls. Your life is gonna suck |
| 06:18.40 | Hyphenex | wiring is already all through their house (it's pretty new), why a server for each phone? (they're SIP phones) |
| 06:18.57 | Hyphenex | at the moment, they've just got one phone in the kitchen, but the plan was to use the VoIP phones |
| 06:19.00 | jblack | no, you'd just want one server. :) |
| 06:19.09 | carrar | Why do you even need 1 |
| 06:19.30 | jblack | All the phones hit * via sip on the local network (preferably an isolated network), and * talks to upstream. |
| 06:19.49 | Hyphenex | yep, that's it :) |
| 06:20.01 | carrar | whats 'upstream'? |
| 06:20.07 | Hyphenex | and upstream would be via their landline telephone? |
| 06:20.22 | jblack | You -might- want a second server, but that's probably overkill |
| 06:20.38 | jblack | In this day and age, they can use their cell phones for redundancy |
| 06:20.39 | carrar | run DRDB with Linux HA! |
| 06:22.16 | Hyphenex | so how to get the 'upstream' part working with their pstn line is the hard part now.. I could go a tdm410 like we'll be putting in at their work, but that's super expensive, and their home has only one number |
| 06:22.34 | Hyphenex | is there something like an ATA that works with a FSO port? |
| 06:22.47 | carrar | Why do a SIP peer with a SIP Provider |
| 06:22.52 | carrar | err WHy not |
| 06:23.10 | Hyphenex | they get capped a lot :P |
| 06:23.19 | Hyphenex | and they like to torrent :P |
| 06:23.28 | carrar | get a second internet for your voice |
| 06:23.38 | carrar | DSL, whatever |
| 06:23.53 | *** join/#asterisk |Cybex| (n=John@atwork-20.r-212.178.82.atwork.nl) |
| 06:24.00 | Hyphenex | they still need the pstn for their pay TV service (it does some freaky stuff I think??) |
| 06:24.05 | Hyphenex | so might as well just use that |
| 06:24.09 | jblack | Ohhh. You better go for a dedicated DSL line then. |
| 06:24.11 | carrar | use a ATA |
| 06:24.15 | jblack | or use the ATA. |
| 06:24.17 | carrar | or get a pots line |
| 06:24.38 | Hyphenex | would the ATA work if it works like a dialup modem? |
| 06:24.42 | jblack | pardon, pstn, with an ATA. I'm sure linksys has a cheapo one, like the old NANPA natpa, or something like that. |
| 06:24.45 | carrar | probably now |
| 06:24.48 | carrar | not |
| 06:24.50 | jblack | they work with pstn lines. |
| 06:25.14 | carrar | probably not gonna work through asterisk |
| 06:25.22 | Hyphenex | hmm, that kind of defeats the purpose :P |
| 06:25.43 | carrar | purpose of asterisk is for your phones I thought |
| 06:25.55 | Hyphenex | yep :) |
| 06:25.56 | jblack | what's "the purpose" ? |
| 06:26.21 | Hyphenex | so we can use the one phone line, and make all their phones ring, transfer calls, then to their office, etc |
| 06:28.05 | jblack | sure, but for the cost of that one phone line, you can get a DSL line that can handle a half dozen concurrent calls. |
| 06:29.38 | Hyphenex | yeah, but we need that one phone line anyway for their pay TV service |
| 06:30.04 | carrar | get free satellite tv |
| 06:30.34 | Hyphenex | nahh, they're happy with their TV |
| 06:30.53 | carrar | You can get ION and NASA for free |
| 06:30.59 | carrar | what else do you need to watch |
| 06:31.04 | Hyphenex | I'll still need to get a TDM410 for their home, right? |
| 06:31.21 | carrar | and a module or two |
| 06:31.35 | Hyphenex | and a module?? |
| 06:31.56 | carrar | yeah, read what a TDM410 is |
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| 06:33.18 | carrar | You plugging your TV into asterisk via a FXO port? |
| 06:33.30 | Hyphenex | oh, you mean a FXO module?? |
| 06:33.31 | Hyphenex | yeah |
| 06:33.46 | Hyphenex | nahh, the TV is going to be on the phone line directly |
| 06:33.48 | Hyphenex | wasn't going to touch that |
| 06:34.19 | carrar | carry on then! |
| 06:34.28 | carrar | 1 line is gonna suck |
| 06:34.31 | carrar | seriously |
| 06:35.09 | Hyphenex | is there anything cheaper then the TDM's for 1 line? (and I'd configure those with Zaptel, right) |
| 06:35.50 | carrar | always cheap ways, and witch cheaper come their own set of issues |
| 06:35.55 | carrar | with |
| 06:36.19 | carrar | could use a x100p clone card |
| 06:36.26 | carrar | good luck with that |
| 06:36.53 | carrar | or a ATA with FXS port |
| 06:37.38 | carrar | sorry FXO |
| 06:41.09 | Hyphenex | I might look into the ATA with FXO, any recommendations? |
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| 06:57.24 | ChannelZ | the SPA3102 is 'cheap' but I'm not particularly impressed with mine |
| 06:58.55 | drmessano | I think the OpenVOX single port FXO trumps the SPA-3102, TBH |
| 07:01.55 | ChannelZ | are those Digium clones? |
| 07:02.42 | drmessano | Yeah |
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| 07:07.58 | Hyphenex | sorry, kicked the ethernet cable :P |
| 07:08.44 | Hyphenex | ChannelZ: Why arn't you happy with yours? |
| 07:10.22 | coppice | the SPA3102 is one of the best ATAs you can get. sadly its still a piece of junk. such is the state of the VoIP market |
| 07:10.34 | Hyphenex | what makes you say that? |
| 07:10.37 | mchou | indeed |
| 07:10.53 | ChannelZ | I have always had massive echo problems when using it, softphone->FXS |
| 07:11.12 | ChannelZ | or really softphone->FXO rather, calling out |
| 07:11.22 | Hyphenex | yuck, yeah, I wouldn't like that |
| 07:11.28 | coppice | reliability, bad echo cancellation, T.38 for loony tunes, etc |
| 07:11.52 | Hyphenex | so what's worth getting for a single channel FXO with asterisk? |
| 07:12.26 | coppice | if you want a single channel FXO, why would you look at an SPA3102? |
| 07:13.25 | ChannelZ | because they are < $100 |
| 07:13.25 | Hyphenex | I was looking at the spa3000, for a cheap dial-out/in solution to my VoIP phones |
| 07:15.08 | coppice | the SPA3102 is expensive compared to other ATAs, as it has an FXS port as well as the FXO one. a PAP2T is far cheaper for FXO |
| 07:15.34 | Hyphenex | would that have the same echo problems your having? |
| 07:16.06 | coppice | the echo problems are mostly on the FXS port |
| 07:16.23 | Hyphenex | is this with your SPA3102? |
| 07:16.39 | coppice | its with anyone's SPA3102 |
| 07:17.38 | Hyphenex | ahhk, but if I were using my own SIP phones and just using that as an outbound/inbound to the pstn (FXO), echo shouldn't be a problem? |
| 07:18.18 | ChannelZ | you (and I) mean FXS actually, I'm delirious |
| 07:19.07 | ChannelZ | You're wanting to plug a phone line into the widget and use VoIP phones yes? |
| 07:19.20 | mchou | lol |
| 07:19.27 | mchou | you guys are funny |
| 07:19.34 | Hyphenex | yeah, but FXO is for connecting it to a pstn network, right? |
| 07:19.42 | mchou | who's on first? |
| 07:20.11 | mchou | Hyphenex: correct. "O" for central Office |
| 07:20.33 | Hyphenex | so I wouldn't even need an FXS port? right? |
| 07:21.07 | mchou | depends what you want to do :) |
| 07:21.12 | ChannelZ | it depends on your point of view |
| 07:21.33 | Hyphenex | gosh, now you're being confusing :P |
| 07:21.47 | mchou | Hyphenex: no, there is no relativity |
| 07:21.50 | Hyphenex | ok, lets say I have no physical phones, everyone is going to be using Xlite :) |
| 07:22.09 | Hyphenex | but I still want to have my softphones make calls through my pstn line in my home |
| 07:22.34 | mchou | Hyphenex: then yiou need both fxs and fxo |
| 07:22.41 | ChannelZ | You want FXS |
| 07:22.51 | Hyphenex | are you sure? |
| 07:22.55 | ChannelZ | he says he doesn't want analog phones |
| 07:23.15 | mchou | he wants to make calls thru PSTN |
| 07:23.22 | ChannelZ | yeah |
| 07:23.35 | mchou | I'm assuming that's not thru an ITSP |
| 07:23.54 | Hyphenex | I don't know what an ITSP is |
| 07:24.08 | mchou | list pstn provisioned to his business or whereever |
| 07:24.16 | mchou | ~itsp |
| 07:24.17 | infobot | [~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 |
| 07:24.22 | ChannelZ | Local Softphone -> Local * box -> ATA -> FXS -> phone company |
| 07:24.47 | mchou | ChannelZ: you forgot the FXO part |
| 07:25.13 | mchou | Hyphenex: let's step back |
| 07:25.15 | ChannelZ | sighs |
| 07:25.43 | mchou | you have a (PSTN) phone line now? |
| 07:25.45 | DND | wait. im confused also on what are you guys talking about.. :P |
| 07:25.57 | mchou | you guys are FUNNY |
| 07:26.08 | ChannelZ | I'm always mixing this up because from the zaptel/asterisk point of view it's like the signalling type or whatever |
| 07:26.16 | mchou | your credibility has been blown |
| 07:26.42 | mchou | Hyphenex: you have a (PSTN) phone line now? |
| 07:26.45 | ChannelZ | When did I claim credibility? |
| 07:26.49 | Hyphenex | are we on the same page? I'd need to get a card/ATA with a FXO port to connect to my pstn network? :P |
| 07:26.58 | Hyphenex | yep, I have a pstn phone line now :) |
| 07:27.12 | mchou | and you want to keep that pstn line? |
| 07:27.17 | Hyphenex | yep |
| 07:27.50 | mchou | and lets say when your intenet goes out, you can still call out via that said phone line? |
| 07:28.03 | mchou | (PSTN) |
| 07:28.08 | Hyphenex | yeah |
| 07:28.21 | Hyphenex | just not from my VoIP system, only a physically attached phone |
| 07:28.22 | mchou | yes, then you need both FXS and FXO |
| 07:28.34 | mchou | oh wait a sec |
| 07:28.46 | Hyphenex | ok, lets say I don't even care about making a phone call on blackout :P |
| 07:28.54 | Hyphenex | I just need an FXO port, right? |
| 07:29.04 | mchou | who is blacked out? |
| 07:29.16 | Hyphenex | oh, my internet |
| 07:29.24 | mchou | ok |
| 07:29.36 | mchou | then you dont need anything :) |
| 07:29.49 | mchou | cause you dont need to make phone call :) |
| 07:29.49 | Hyphenex | just an FXO port (I'm not using s SIP Broker) |
| 07:30.35 | mchou | Hyphenex: please. unless you understand the terminolgy dont use terms like FXS and FXO |
| 07:31.16 | mchou | your inet is blacked out. You still need calls connected or not? |
| 07:31.29 | Hyphenex | lets assume I don't even have the internet :P |
| 07:31.34 | mchou | if yes, go with FXO. If no, FX only |
| 07:31.43 | mchou | if yes, go with FXO. If no, FXS* only |
| 07:32.00 | ChannelZ | Hyphenex: You want an FXO port. |
| 07:32.05 | Hyphenex | I want to be making and receiving phone calls on my pstn line |
| 07:32.10 | Hyphenex | yeah, an FXO port :) |
| 07:32.19 | DND | then an FXO port |
| 07:32.25 | mchou | about time |
| 07:32.40 | DND | easy. then install iso bundle and setup extensions and install x-lite |
| 07:32.43 | DND | thats all |
| 07:32.45 | DND | right? |
| 07:32.46 | Hyphenex | so my question was, does the SPA3102 have echo on the FXO port? :P |
| 07:32.57 | Hyphenex | yep, right :) |
| 07:32.59 | ChannelZ | yes |
| 07:33.12 | ChannelZ | It's very slow to cancel |
| 07:33.22 | Hyphenex | so, is the SPA3102 has echo on the FXO port, what would I be best off getting? |
| 07:33.39 | ChannelZ | A carrier pigeon |
| 07:33.50 | Hyphenex | *sigh* seriously? |
| 07:33.50 | DND | but is it much more efficient to echo cancel on the pbx itself? |
| 07:35.39 | DND | hey, hosw can i do that pink text thing? |
| 07:35.42 | DND | *how |
| 07:36.39 | ChannelZ | DND: ?? how do you get * to echo cancel non-dahdi hardware? |
| 07:38.50 | mchou | Hyphenex: go to audicodes, http://www.audiocodes.com/products/mediapack-1xx |
| 07:39.03 | mchou | Hyphenex: expensive though |
| 07:40.00 | DND | i knew it. i was not reading properly :D |
| 07:40.03 | Hyphenex | nahh, just want an asterisk->pstn bridge that won't echo :P |
| 07:41.44 | mchou | Hyphenex: then go with OpenVOX single port FXO |
| 07:45.03 | Hyphenex | what about this? :) http://www.digiumcards.com/tdm01b.html |
| 07:45.43 | ChannelZ | If you want to spend that much. http://www.voiplink.com/OpenVox_A400P01_1_FXO_p/openvox-a400p01.htm |
| 07:45.43 | ChannelZ | (clone) |
| 07:46.30 | Hyphenex | it's just as good at the TDM410's? |
| 07:46.34 | ChannelZ | you can but an FXS module down the road if you need one, in either case |
| 07:46.49 | ChannelZ | Can't help you there, I have a TDM |
| 07:47.02 | Hyphenex | might as well just get a TDM then :) |
| 07:47.08 | Hyphenex | and no echo with those? |
| 07:48.26 | ChannelZ | eh, it depends. Echo has more to do with the actual connection than the hardware; It's that the SPA 3102 is lousy at dealing with it. |
| 07:49.30 | ChannelZ | I generally don't have a problem on my digium card but I'll get echo for the first 10-15 seconds of a call sometimes when someone is calling me on a VoIP<->PSTN setup of their own |
| 07:49.36 | ChannelZ | I guess because the echo is much longer in those cases so it takes it awhile to learn |
| 07:50.24 | Hyphenex | ahhk, so if I wanted to cheap out for home, I'd get the OpenVox cards, but the one you linked me doesn't have any echo cancelation on it, so that could be an issue right? |
| 07:50.39 | mchou | Hyphenex: just go a head and buy a spa3102 |
| 07:50.55 | Hyphenex | nahh, I don't want to live with echo |
| 07:51.04 | mchou | the folks complaining about echo dont know what they are talking about |
| 07:51.20 | mchou | just like how the messed up FXS/FXO |
| 07:51.40 | mchou | spa-3102 has weak spots but the echo is a non issue |
| 07:51.46 | ChannelZ | mchou: You should write a guide since you're so brilliant |
| 07:52.14 | ChannelZ | The amount of jackass-ego in this channel astounds me |
| 07:52.22 | mchou | spa-3102 weak spot is firmware |
| 07:52.37 | mchou | doubt hook fash, etc |
| 07:52.43 | mchou | flash* |
| 07:52.55 | mchou | double hook flash, etc |
| 07:53.44 | mchou | ChannelZ: what exactly did you do to try minimze the echo on spa-3102? |
| 07:54.00 | mchou | minimize* |
| 07:54.13 | ChannelZ | I prayed to the unicorns |
| 07:54.21 | mchou | wrong answer |
| 07:54.45 | mchou | that's why you think there's ego involved |
| 07:54.58 | mchou | nothing to do with ego at all |
| 07:55.27 | ChannelZ | Well, I don't know what I'm talking about, afterall. I guess my doubt hook fash settings are wrong. |
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| 07:56.02 | AppleBoy | hello |
| 07:56.04 | mchou | oh, seize upon my typing mistakes |
| 07:56.25 | mchou | it makes your ego so much superior |
| 07:56.27 | AppleBoy | hi file |
| 07:56.42 | Hyphenex | hmm, so you'd recommend going the SPA-3102?? And what happens when I get echo? (I'm not a know it all genius by any means) :P |
| 07:56.45 | ChannelZ | scratches his head |
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| 08:07.16 | twanny796 | zaptel problem, any ideas? http://pastebin.com/m50f508b3 |
| 08:10.17 | twanny796 | want to use kernel 2.4.36 |
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| 08:30.52 | _Samael_ | Anyone alive here? |
| 08:30.52 | Katty | for the moment |
| 08:30.52 | _Samael_ | Cah you help me with merging vars along att. transfer? |
| 08:32.44 | Katty | no. i'm going to sit right here and be miserable. |
| 08:32.51 | Katty | possibly get more vicks. |
| 08:33.56 | _Samael_ | Ok. I'll try to find another way ) |
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| 08:39.14 | Hyphenex | do you think I'd get echo on this? http://www.voiplink.com/OpenVox_A400P01_1_FXO_p/openvox-a400p01.htm |
| 08:39.59 | mchou | no |
| 08:40.25 | mchou | thats got on board echo canceller that works well |
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| 08:44.38 | DND | just a question, what's the downside of the digium cards? |
| 08:44.48 | DND | specifically aex800/tdm800 |
| 08:48.44 | DND | woo!! mass kill!! |
| 09:00.15 | Hyphenex | Thanks mchou |
| 09:00.19 | Hyphenex | I'm just going to get one of those for home |
| 09:00.29 | Hyphenex | and if it works well, get another one for work with a few extra modules :) |
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| 09:36.58 | b14ck | i wish i had more freetime to add stuff to it |
| 09:36.58 | Katty | i wish i hadn't had that last drink. |
| 09:37.07 | Katty | i /really/ wish i hadn't had that caffeinated tea earlier today. |
| 09:37.07 | b14ck | caffine is awsome |
| 09:37.09 | Katty | or the hot dog. |
| 09:37.21 | Katty | caffeine is horrible. |
| 09:37.21 | b14ck | i just had some jalepeno poppers and a couple beers |
| 09:37.36 | b14ck | actually---it was a lot of beers :X |
| 09:38.17 | tzafrir | Is why you say that caffeine is awesome? |
| 09:38.35 | b14ck | no way caffeine is always awesome |
| 09:38.36 | tzafrir | Is this* |
| 09:38.47 | b14ck | i drink a couple amp / full throttle drinks everyday |
| 09:38.56 | Katty | caffeine is responsible for everything from high blood pressure to miscarriages. |
| 09:39.35 | b14ck | tzafrir, are you also tshiff? oO |
| 09:39.37 | Katty | it also dehydrates you. |
| 09:40.19 | tzafrir | b14ck, it's 11:40 AM in this time zone |
| 09:40.34 | *** part/#asterisk Hyphenex (n=Hyphenex@vps17.vz7.ramhost.us) |
| 09:40.35 | tzafrir | and my work week is Sun-Thu |
| 09:40.37 | b14ck | ouch you in Australia or something? |
| 09:40.55 | b14ck | So you're at work? |
| 09:41.15 | Katty | of course, the vodka was probably even worse for me. |
| 09:41.39 | ChannelZ | arrgh * 1.4.26.3's voicemail is messed up |
| 09:41.45 | tzafrir | TZ=Australia/Sydney date |
| 09:41.45 | tzafrir | Sun Nov 8 20:41:33 EST 2009 |
| 09:41.54 | b14ck | nice |
| 09:41.58 | b14ck | what kinda work do you do? |
| 09:42.11 | tzafrir | Something with Asterisk :-) |
| 09:42.18 | Katty | would estimate BAC from .03 to .12 right now. |
| 09:42.36 | Katty | probably more like .09 to .12 |
| 09:42.40 | b14ck | tzafrir, do you work for digium? |
| 09:42.50 | tzafrir | No . Xorcom |
| 09:42.58 | b14ck | that sounds familiar |
| 09:43.00 | b14ck | *googles* |
| 09:43.44 | b14ck | ah you guys run elastix? |
| 09:43.49 | b14ck | that's pretty cool |
| 09:43.52 | tzafrir | giggles |
| 09:45.42 | *** join/#asterisk dan__t (n=dant@72.233.89.95) |
| 09:45.55 | Katty | well. i'm gonna try to sleep again. later. |
| 09:45.57 | b14ck | i think i spoke to a xorcom engineer a few days ago at my work |
| 09:46.02 | dan__t | Good morning. |
| 09:46.02 | b14ck | later Katty |
| 09:46.09 | b14ck | thanks for the venting help :D |
| 09:46.37 | dan__t | I've dabbled in Asterisk a bit, about six months ago, and I'll be embarking on another project with it soon here with some advanced stuff that I've not worked with yet. I wanted to make sure i had some terminology down first, however. |
| 09:47.04 | dan__t | What's the real name for like a conference, if I were to place two channels on the same channel, or conference? |
| 09:47.11 | dan__t | Combining two calls etc etc |
| 09:47.16 | ChannelZ | MeetMe |
| 09:47.24 | b14ck | bridging calls? oO |
| 09:47.29 | ChannelZ | thats the app anyway :) |
| 09:47.49 | tzafrir | there's also app_conference and app_confbridge |
| 09:47.50 | dan__t | Bridge? |
| 09:48.09 | tzafrir | bridge is connecting two (or more) channels together |
| 09:48.23 | tzafrir | A channel is one leg of a call |
| 09:48.30 | dan__t | Got it. So an established call would be on a channel. |
| 09:48.32 | dan__t | Ok, cool. |
| 09:48.48 | dan__t | I was doing a litlte bit of this but I was flying completely blind, I felt uneasy about it all haha |
| 09:49.13 | tzafrir | which implies that 3-way calling is a tripod |
| 09:49.35 | b14ck | tzafrir, I have to give it to you. Xorcom's website and products look awesome. |
| 09:49.43 | b14ck | If I was a business owner I would be purchasing one |
| 09:50.01 | dan__t | Cool, thank you. |
| 09:50.13 | dan__t | Is there a way that I can get something like Background() to interact with a bridge? |
| 09:50.17 | dan__t | Does it work that way? |
| 09:50.38 | b14ck | you can do certain things to make it do that |
| 09:50.45 | b14ck | but it doesn't work that way, no |
| 09:51.13 | tzafrir | Background works with a single channel, basically. It may or may not be bridged. Though it is quite pointless to play sounds if it isn't going anywhere |
| 09:51.19 | dan__t | I guess what I'm looking for is to find out what options are available to me, how I can interact with a bridge, maybe even just with one channel connected with that bridge. I'm interested in this part. |
| 09:51.35 | b14ck | tzafrir, http://www.xorcom.com/catalog/xr2020.html what form factor is this? Is this a rack server form factor type appliance or what? |
| 09:51.43 | dan__t | I'm talking about using Background and intercepting DTMF to engage AGI. |
| 09:52.24 | b14ck | dan__t, no (not the way you are thinking of it) |
| 09:52.33 | b14ck | If you are using MeetMe, anyhow, it is a full application. |
| 09:52.47 | b14ck | Once callers enter the conference you cannot run arbitrary dialplan like you could on a normal call. |
| 09:52.55 | b14ck | (without using AMI or other tricks) |
| 09:53.20 | tzafrir | b14ck, yes, rack cmounted, 2U |
| 09:53.21 | dan__t | Yeah I suspected I'd need to use AMI. |
| 09:53.31 | b14ck | tzafrir, that looks awesome |
| 09:54.34 | tzafrir | dan__t, alternatively, have a look at features.conf |
| 09:54.57 | tzafrir | However: is this in an IVR or in the middle of a call? |
| 09:55.08 | tzafrir | Background seems to imply this is an IVR |
| 09:55.16 | _Samael_ | features.conf can help with a little set of features |
| 09:55.36 | dan__t | I guess Background was abad example to use. |
| 09:55.50 | _Samael_ | WaitExten? |
| 09:56.09 | dan__t | IN particular, I'm playing around with LumenVox and grammars. Long story short, I'd like to have LumenVox sit on a bridge and do things based on some keyword command. |
| 09:56.34 | dan__t | Say for example one channel was some sort of moderator for the bridge, he'd prefix a command using a key word, yada yada |
| 09:57.08 | dan__t | Looks like my options for that are fairly limited, specifically to SpeechBackground() or SpeechStart() |
| 09:57.12 | dan__t | From what I can tell so far, anyway. |
| 09:57.36 | _Samael_ | dan__t: May be use WebMeetMe? |
| 09:57.52 | _Samael_ | dan__t: For moderating conf |
| 09:58.31 | dan__t | hrm. |
| 10:00.28 | dan__t | I don't want to use something like that just yet. My history with Asterisk has proven that I might not understand the inner workings of what i'm actually doing before I deploy this stuff. |
| 10:01.35 | _Samael_ | ))) |
| 10:02.03 | dan__t | eh? |
| 10:02.47 | b14ck | dan__t, do you know tnid.org? |
| 10:03.05 | dan__t | I'm familiar. |
| 10:03.28 | dan__t | Why, what's up? |
| 10:03.37 | b14ck | do you work for them? |
| 10:03.41 | dan__t | I don't. |
| 10:03.51 | b14ck | or one of their other companies? |
| 10:03.55 | b14ck | like tnis or w/e |
| 10:04.14 | dan__t | I don't. |
| 10:04.28 | b14ck | ok, just wondering |
| 10:04.32 | dan__t | What made you think I did? |
| 10:04.47 | b14ck | i know one of their engineers a little bit |
| 10:04.53 | b14ck | he was working on something similar |
| 10:05.01 | b14ck | to what you've just described |
| 10:05.24 | dan__t | Ahh ok. |
| 10:05.43 | dan__t | Bet he had a clue about what he is doing heh |
| 10:07.11 | dan__t | So are SpeechBackground() or SpeechStart() specific to LumenVox, or are those Asterisk internals? |
| 10:08.19 | dan__t | Oh, guess they're part of Asterisk |
| 10:10.16 | b14ck | they're part of asterisk |
| 10:10.20 | b14ck | lumenvox has its own stuff |
| 10:12.05 | dan__t | jea |
| 10:12.50 | dan__t | I want to take that speech stuff and apply it to a bridge. |
| 10:13.14 | dan__t | Or, like I said, a particular channel of that bridge (I would say "a channel of a bridge" to refer to one individual call, right?" |
| 10:13.33 | b14ck | dan__t, what you wanna do is gonna be complex |
| 10:13.46 | b14ck | it's probably easier if you just mod the conference app in C yourself to be honest |
| 10:13.53 | dan__t | Ouch. |
| 10:17.22 | dan__t | So Apache has documentation that breaks down each directive and tells in which context it can be used and stuff. Is there a similar matrix for Asterisk functions? |
| 10:19.17 | b14ck | dan__t, best reference is withing asterisk CLI |
| 10:19.19 | b14ck | *CLI |
| 10:19.20 | b14ck | eg: |
| 10:19.24 | b14ck | core show functions |
| 10:19.27 | b14ck | core show applications |
| 10:19.35 | b14ck | for more information: |
| 10:19.40 | b14ck | core show function xxxxxx |
| 10:19.43 | b14ck | etc |
| 10:19.56 | dan__t | Oh, right. |
| 10:22.20 | b14ck | also though, there are several tricky things you can do in conferences to add on functionality |
| 10:22.27 | b14ck | without making your life too much harder |
| 10:22.36 | dan__t | Do you have a few minutes to elaborate on some of those things? |
| 10:22.50 | b14ck | ya (someone else correct me if i'm wrong) |
| 10:22.58 | b14ck | create a context in dialplan |
| 10:23.02 | b14ck | to do all the logic you want |
| 10:23.15 | b14ck | eg: listen for a dtmf keypress, then play a soundfile (whatever you wanna do) |
| 10:23.29 | b14ck | create a conference room, and have all your users join it and start talkikng / doing their thing |
| 10:23.38 | dan__t | yeah |
| 10:23.44 | dan__t | that's how it normally would go. |
| 10:23.49 | b14ck | create a callfile which dials INTO the conference then set the payload to hit the asterisk context that you created with all the logic |
| 10:23.50 | *** join/#asterisk af_ (n=getsmart@88-149-230-138.dynamic.ngi.it) |
| 10:24.03 | b14ck | that way that single extension which is now in the conference has normal dialplan / agi / whatever features |
| 10:24.11 | b14ck | so it can listen for keypresses and do cool stuff |
| 10:24.27 | dan__t | oooh |
| 10:24.31 | b14ck | also-- |
| 10:24.32 | dan__t | like a shadow or alias or something..? |
| 10:24.36 | b14ck | I'm not sure what language you are doing this in. |
| 10:24.48 | b14ck | But I wrote a nifty call-file library which can help (in python) if you are interested =p |
| 10:25.10 | b14ck | dan__t, its like registering your custom code as a user in the system |
| 10:25.18 | dan__t | yeah |
| 10:25.20 | b14ck | instead of at the system level |
| 10:25.21 | dan__t | I get it. |
| 10:25.24 | b14ck | ;) |
| 10:25.30 | b14ck | http://github.com/comradeb14ck/pycall |
| 10:25.32 | dan__t | that's a good one. |
| 10:26.25 | b14ck | =) |
| 10:26.39 | dan__t | A real good one, actually. |
| 10:26.48 | b14ck | ya, im surprised i thought of that while drunk |
| 10:26.53 | b14ck | woot |
| 10:27.00 | dan__t | Tonight is recovery/geek out night. |
| 10:27.04 | dan__t | Friday night hurt. |
| 10:27.13 | b14ck | heh |
| 10:27.29 | b14ck | so what are you developing this app for? |
| 10:27.35 | b14ck | you work for a lec or something? |
| 10:27.44 | b14ck | party-line stuff? |
| 10:28.53 | dan__t | Just f'ing around honestly. I always wanted to learn Asterisk so I was just trying to come up with some practical examples. |
| 10:29.15 | b14ck | conference bridges are a big business for a lot of carriers |
| 10:29.19 | b14ck | especially the small, local ones |
| 10:29.31 | dan__t | I bet. |
| 10:29.33 | b14ck | since they get paid for termination they will setup free party-lines and stuff |
| 10:29.44 | b14ck | and supposedly theres like a lot of competition for the best conferencing apps |
| 10:29.53 | b14ck | you should make it and sell it to some of those guys |
| 10:29.54 | b14ck | <3 |
| 10:30.09 | dan__t | hah |
| 10:31.36 | b14ck | well, goodnight everyone |
| 10:31.40 | b14ck | im gonna crash |
| 10:32.44 | dan__t | Hey, thanks for yoru help. I'll pop in later on today, I want to get a better idea of how your idea would work. |
| 10:32.51 | dan__t | I get the basics ubt I want to read up on it some more. |
| 10:33.55 | b14ck | i actually wrote this application before |
| 10:33.59 | b14ck | almost exactly what u wanted oO |
| 10:34.17 | b14ck | anyways, im gonna sleep. ill be on in ~15 hours or so |
| 10:34.18 | b14ck | nite! |
| 10:35.25 | dan__t | haha ok cya, thanks again. |
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| 11:05.32 | *** join/#asterisk TJNII (n=TJNII@207.189.199.62) |
| 11:17.19 | DND | guys |
| 11:17.37 | DND | i need to extend the time for inputting a feature code during a call |
| 11:18.08 | DND | whenever i type *1 for recording on-demand, it doesnt get recorded |
| 11:20.43 | *** join/#asterisk Tim_Toady (n=moi@ppp206-119.adsl.forthnet.gr) |
| 11:28.24 | *** join/#asterisk dandate2 (n=mangy@112.202.226.219) |
| 11:29.19 | dandate2 | i wanted to share my experience. i hooked up asterisk to a fiber optic connection in america and made an ext->ext call between 2 international clients and the result was choppy and quiet. when the same 2 international clients made a skype call to eachother the volume and quality was 10x better |
| 11:29.26 | dandate2 | is this because skype has backbones in taiwan? |
| 11:31.08 | *** join/#asterisk bbt (n=sam@180.189.138.55) |
| 11:32.53 | *** join/#asterisk viq (n=viq@unaffiliated/viq) |
| 11:33.18 | Gugge | dandate2: thats because something went wrong with the data from minimum one of the clients to the asterisk box |
| 11:34.36 | *** join/#asterisk dkirker (n=dkirker@gateway0.openmobl.com) |
| 11:38.24 | dandate2 | i did a series of test calls |
| 11:39.45 | dandate2 | test call 1. eyebeam to eyebeam internationally with server in america = crap : test call 2. skype to skype internationally = great : test call 3. skype to american landline forwarded to skype = great : test call 4. skype to american DID provider forwarded to skype = crap |
| 11:46.58 | *** join/#asterisk baijum (n=baiju@122.167.109.214) |
| 11:49.05 | DND | dandate2, that hurts, terminated |
| 11:49.18 | dandate2 | lol |
| 11:49.25 | dandate2 | sall gravy |
| 11:50.04 | dandate2 | u linux guys are offended by anything windows heh |
| 11:50.57 | DND | how much did you bought for eyebeam? |
| 11:51.07 | dandate2 | i just downloaded the torrent to be honest |
| 11:51.20 | DND | hah! :P |
| 11:51.32 | DND | san ka work |
| 11:51.56 | dandate2 | purchasing eyebeam is not practical for work at home call centers |
| 11:52.08 | dandate2 | because we have people trying it out all the time, eating up licesnes, and never coming back |
| 11:52.26 | DND | work from home. really nice. |
| 11:52.46 | DND | if only my IT manager would allow us like that |
| 11:54.03 | DND | dandate2, you were unbanned :D |
| 11:54.06 | dandate2 | yeah its tough cuz u can have people passing sales |
| 11:54.13 | dandate2 | i've cought people doing that already |
| 11:59.53 | *** join/#asterisk sulex (n=sulex@host-78-14-173-96.cust-adsl.tiscali.it) |
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| 12:24.54 | *** join/#asterisk toffifee666 (n=kahlb@p508D759D.dip.t-dialin.net) |
| 12:25.12 | toffifee666 | hi there |
| 12:49.30 | *** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim) |
| 12:55.38 | *** join/#asterisk Athahira (n=chatzill@80.249.209.114) |
| 13:02.35 | *** join/#asterisk tlir (n=tlir@77.126.35.195) |
| 13:03.11 | Athahira | No audio format found to offer ....when the peer definition is clearly have the codecs allowed |
| 13:03.29 | Athahira | any suggestions..... |
| 13:04.46 | *** join/#asterisk shinao1 (n=shinao1@41.219.249.42) |
| 13:05.19 | Athahira | the calling client side have 3 codec allowed ...but software only have 1 codec....and sip.conf only have 2 of the allowed codec |
| 13:07.11 | Athahira | i am using * 1.6.1.9 |
| 13:07.20 | *** join/#asterisk dcabot (n=dcabot@c-69-254-32-199.hsd1.fl.comcast.net) |
| 13:11.30 | Athahira | hello |
| 13:12.19 | Athahira | anyone could suggest y * refuse to send invite....and display this No audio format to offer !! |
| 13:12.33 | Athahira | y * refuse this invite.... ? |
| 13:23.49 | Athahira | plz...anyone !!Plz...any one suggest the error no audio format found to offer ??? |
| 13:25.13 | dandate2 | are you trying to use g729 |
| 13:25.49 | dandate2 | try using a different codec like ulaw or gsm and see if that works, i had the same problem on a fresh pbxinaflash installation that did not have g729 libraries installed |
| 13:26.26 | DND | and try to pastebin the actual * logs |
| 13:30.12 | Athahira | thanx guys |
| 13:30.14 | *** join/#asterisk e4 (n=e4@rrcs-76-79-48-214.west.biz.rr.com) |
| 13:32.06 | Athahira | well ..its really strange ....ii have 2 clients....one only support codec g723 ...the other gsm |
| 13:32.29 | Athahira | and provider definition codecs allowed is g729,g723,gsm |
| 13:33.18 | Athahira | when calling from the g723 client .....call is passed to provider using gsm !! using sip show channels ...i see tx and rx as gsm ...so no voice ius heard |
| 13:33.41 | Athahira | but when calling from gsm client....call is passing correctly as gsm..voice is really good |
| 13:34.58 | *** join/#asterisk Iamnacho (i=Iamnacho@ip98-186-180-143.ks.ks.cox.net) |
| 13:35.03 | Athahira | so...to force use g723 for the g723 client ..i delete gsm allowed from provider --------->Pingo!! error, no audio format found to offer !! it just don't try to send the call to provider..i enabled sip debug |
| 13:36.03 | Athahira | any clue y is that ?? i use adoresofphone free software ..which only have g723 codec |
| 13:41.33 | Athahira | Adding codec 0x2 (gsm) to SDP it always add this codec !! y not add other enabled codecs ?!!! |
| 13:42.37 | dandate2 | i dont even know what g233 is |
| 13:43.01 | dandate2 | just get zoiper or something, use gsm. contact contact-sales@didforsale.com they can install g729 libraries for you |
| 13:43.34 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 13:43.50 | Athahira | yes..but y * don't send the allowed codec inside client definition inside the invite message ??? |
| 13:44.07 | Athahira | can anyone explain how * decide to send codec capabilities ? |
| 13:44.22 | Athahira | from where it get codecs when it send invite message ? |
| 13:46.41 | *** join/#asterisk retentiveboy (n=pdugas@74-95-28-37-Atlanta.hfc.comcastbusiness.net) |
| 13:48.36 | *** join/#asterisk clart001 (n=clart@host155-18-dynamic.6-87-r.retail.telecomitalia.it) |
| 13:49.36 | clart001 | i get in my log this: http://pastebin.com/d32c03de9 |
| 13:49.45 | clart001 | can you help me? |
| 13:53.08 | Athahira | anyone plz explain how asterisk send codec capabilities within invite sip messages ?? from where it get it ? sip.conf general section or client defined section ? |
| 13:54.54 | dandate2 | athahira just get skype call center |
| 13:55.10 | dandate2 | it uses 1000x more resources but u wont ahve any codec problems at least |
| 13:55.30 | dandate2 | to host 30 calls for example you'll need a quad core processor |
| 13:55.53 | *** join/#asterisk Godfather_ (n=godfathe@187.Red-83-38-229.dynamicIP.rima-tde.net) |
| 13:56.04 | dandate2 | it literally opens 30 skype windows on your desktop and directs calls that way, pretty funny stuff |
| 13:57.00 | Athahira | ok...but we rae talking about * ...and i love opensource....skype is not open ...even they give it to me fre with 1 millione functions...i just don't want it !!! |
| 13:57.44 | Athahira | * is great...and open...keep ur mind working ....make u smarter....skype make u dump !! |
| 14:00.42 | clart001 | that is a iax problem |
| 14:01.30 | Athahira | anyone guys...? |
| 14:01.49 | *** join/#asterisk knctrnl (n=aembrey@user-69-1-12-224.knology.net) |
| 14:06.32 | drmessano | Skype is just a service.. how is the application being not open source even an issue |
| 14:08.22 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 14:11.52 | TSM2 | anyone of a free easy f11 dial app for asterisk |
| 14:11.58 | TSM2 | windows desktop pref |
| 14:14.16 | clart001 | i self fix my issue |
| 14:14.21 | clart001 | thank to all |
| 14:15.15 | *** part/#asterisk clart001 (n=clart@host155-18-dynamic.6-87-r.retail.telecomitalia.it) |
| 14:16.25 | *** join/#asterisk mchou (n=mchou@unaffiliated/mchou) |
| 14:19.18 | *** join/#asterisk Pan3D (n=Pan3D@streaming.sensoryresearch.net) |
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| 14:32.46 | *** join/#asterisk PanicMan (i=panicman@122.102.33.67) |
| 14:32.57 | PanicMan | still in big panic with IAX :'( |
| 14:33.13 | PanicMan | getting one sided compression |
| 14:33.18 | PanicMan | other side is not working :( |
| 14:33.23 | PanicMan | help meeeeeeeeeeeeeeeeeeeeeeeeeeeeeee |
| 14:34.10 | PanicMan | hello sir, please help help :( |
| 14:34.50 | Pan3D | your nick is apropos |
| 14:35.49 | PanicMan | :( |
| 14:36.19 | PanicMan | Pan3D, can u help me out ! :'( |
| 14:38.36 | *** part/#asterisk PanicMan (i=panicman@122.102.33.67) |
| 14:46.49 | *** join/#asterisk JT (n=j@unaffiliated/jt) |
| 14:56.07 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194) |
| 14:59.22 | Pan3D | you might say "he left in a panic" |
| 15:02.50 | mchou | Don't have that dude anywhere near a launch base |
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| 16:07.32 | Katty | it's a beautiful day in the neighborhood |
| 16:07.36 | Katty | a beautiful day for a neighbor |
| 16:07.50 | Katty | would you be mine? :> |
| 16:13.05 | [TK]D-Fender | 668 ... the neighbour of the Beast |
| 16:13.12 | ChannelZ | Only if you don't make a lot of noise. |
| 16:15.50 | Katty | no, no noise this morning. tho i will be making french toast here shortly. |
| 16:17.09 | [TK]D-Fender | Katty: You mean FREEDOM TOAST! |
| 16:17.18 | Katty | yes, of course. |
| 16:17.33 | Katty | later on, we'll be having liberty cabbage. |
| 16:17.40 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 16:19.24 | *** join/#asterisk tamiel (n=tamiel@ip-1.net-81-220-19.versailles.rev.numericable.fr) |
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| 16:30.04 | *** join/#asterisk debuggerboy (n=debugger@117.196.164.36) |
| 16:30.05 | *** join/#asterisk Godfather_ (n=godfathe@187.Red-83-38-229.dynamicIP.rima-tde.net) |
| 16:34.29 | *** join/#asterisk Athahira (n=chatzill@81.10.117.50) |
| 16:36.06 | Athahira | hello everyone |
| 16:37.06 | *** join/#asterisk mchou (n=mchou@unaffiliated/mchou) |
| 16:38.18 | Athahira | can anyone plz confirm the outgoing sip channel, invite include codecs availability...form where those codec info inserted inside the invite message ? |
| 16:39.43 | Athahira | based on outgoing peer definition ? or the originating client allowed codecs ? or based on client incoming invite message which contains codecs supported ? |
| 16:41.22 | Athahira | anyone pls could help me understand this ? |
| 16:41.25 | ChannelZ | I'm not positive what you're really asking, but if it's "can I restrict codecs on a peer-by-peer basis", then yes, just put those definitions under each peer |
| 16:42.25 | Katty | returns with french toast and banana. |
| 16:43.08 | Athahira | ChannelZ: yes...but i see different behavoiur on the outgoing invite ...for example i have 2 defined clients with 3 codecs...outgoing peer also have 3 codecs... |
| 16:43.47 | Athahira | client 1 issue invite --->outgoing peer : this have 3 codecs inside the invite mesage ... |
| 16:44.26 | Athahira | but client 2 when issue invite ------->outgoing peer : here the invite only have 1 available codec !! |
| 16:45.04 | ChannelZ | depends on what your sip.conf looks like |
| 16:45.12 | ChannelZ | pastebin it |
| 16:45.22 | ChannelZ | ~pastebin |
| 16:45.23 | infobot | [~pastebin] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 16:45.32 | Athahira | Invite here is the second leg invite ....which asterisk originate to the outgoing peer |
| 16:45.35 | [TK]D-Fender | Athahira: For your peer entry when you dial |
| 16:45.52 | [TK]D-Fender | Athahira: thats what determines it. |
| 16:46.07 | [TK]D-Fender | Athahira: * is not a SIP PROXY, it is a B2BUA |
| 16:46.08 | [TK]D-Fender | ~B2BUA |
| 16:46.09 | infobot | extra, extra, read all about it, b2bua is a back 2 back user agent |
| 16:46.13 | p3nguin | katty: You reminded me of this http://www.youtube.com/watch?v=4kK6F7t-x6E |
| 16:46.20 | *** part/#asterisk debuggerboy (n=debugger@117.196.164.36) |
| 16:46.36 | [TK]D-Fender | Athahira: So the call that spawns the dial has nothing to do with the call it causes * to place. |
| 16:46.44 | [TK]D-Fender | Athahira: that's what TRANSCODING is for |
| 16:46.57 | Katty | p3nguin: ^_- |
| 16:47.03 | [TK]D-Fender | Athahira: the 2 legs need not have anything in common. |
| 16:47.11 | Athahira | Fender: yes i understand that B2BUA , the peer u mentioned is the outgoing peer, right ? |
| 16:47.23 | [TK]D-Fender | Athahira: When * dials, the peer defines how it calls |
| 16:47.26 | Katty | p3nguin: i'm not sure i know how to feel about that |
| 16:47.35 | Katty | p3nguin: but you don't want to eat at denny's. trust me. |
| 16:47.39 | ChannelZ | p3nguin: huh I can't say I've ever seen that |
| 16:47.50 | p3nguin | katty: I hate Denny's food, actually. |
| 16:48.03 | Katty | good. because it is just plain awful for you. |
| 16:48.40 | Katty | there are 820 calories (on average) in the grand slam |
| 16:48.47 | Athahira | Fender....i will show u 2 invite messages going to the peer (provider defined which have 3 allowed codecs), these 2 invite coming from different clients(friends) |
| 16:49.20 | Athahira | and the 2 invite have different annouced available codecs inside them !! |
| 16:49.48 | Katty | p3nguin: http://www.dennys.com/LiveImages/enProductImage_690.pdf |
| 16:49.54 | Athahira | Fender: is it ok ? |
| 16:50.32 | *** join/#asterisk xpot (n=xpot@70-91-210-233-BusName-Utah.hfc.comcastbusiness.net) |
| 16:51.11 | [TK]D-Fender | Athahira: just do it, and make them separate PB's with the configs included in each |
| 16:51.55 | Athahira | Fender: ok...i will make 2 calls now...and copy sip debug invite going to the defined peer ....one min |
| 16:54.51 | Katty | p3nguin: http://42ndrecipestreet.blogspot.com/2009/07/dennys-style-french-toast.html |
| 16:56.12 | p3nguin | mmmm... French toast... |
| 16:56.55 | Katty | p3nguin: the french toast platter at denny's is 1010 calories :/ |
| 16:57.09 | Katty | p3nguin: and 52g of fat :< |
| 16:57.10 | Athahira | Fender...done , i have the 2 logs ....where i can paste them to show u ? |
| 16:57.23 | Katty | p3nguin: let's not even talk about the salt content |
| 16:57.50 | p3nguin | katty: *sadface* |
| 16:57.58 | p3nguin | katty: I thought you wanted me to have French toast! |
| 16:58.07 | Katty | p3nguin: just make it yourself. |
| 16:58.10 | Katty | p3nguin: much healthier. |
| 16:58.30 | Katty | p3nguin: i honestly don't understand how they manage to cram so many calories into things. |
| 16:58.41 | p3nguin | lard? |
| 16:58.46 | Katty | who knows. |
| 16:59.12 | Katty | the two slices of bread would be 160, the egg 70, milk is probably about 30... |
| 17:00.31 | Katty | the flour has 25 calories. |
| 17:00.59 | Katty | if the bread absorbed ALL of the batter, and ALL of the 1 T of oil you used to cook it in... |
| 17:01.29 | Katty | it's about 350 calories |
| 17:01.36 | Katty | 1 T of honey is 60 calories. |
| 17:01.36 | *** join/#asterisk TimToady_ (n=moi@adsl52-245.kln.forthnet.gr) |
| 17:01.44 | [TK]D-Fender | ~pb |
| 17:01.45 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , or apt-get install pastebinit |
| 17:01.49 | [TK]D-Fender | Athahira: ^^^ |
| 17:01.51 | Katty | so not really sure how denny's goes from 400ish calories to 1010 |
| 17:02.00 | p3nguin | Maybe they're using that "new math" or something. |
| 17:02.03 | Katty | it's MADNESS. |
| 17:02.36 | [TK]D-Fender | Katty: HFCS, more oil, etc |
| 17:02.46 | Katty | yes probably |
| 17:03.03 | [TK]D-Fender | Katty: And an assumed amount of syrup |
| 17:03.11 | [TK]D-Fender | Katty: Butter, etc |
| 17:03.13 | Katty | and probably more than 2 pieces. |
| 17:03.45 | Athahira | i opened pastbin link...now what ? |
| 17:03.49 | Katty | tho the note says if you add any sides, like sausage...it's not included |
| 17:03.59 | Athahira | i paste the infor there ...then ? |
| 17:04.17 | Katty | heh |
| 17:04.22 | Katty | that's hilarious |
| 17:04.38 | Katty | READ the page, i promise you will figure out what to do with pastebin. |
| 17:04.42 | Athahira | http://pastebin.com/m2c912c0d |
| 17:04.50 | Athahira | is this correct ? |
| 17:06.25 | *** join/#asterisk manxpower (n=ewieling@24.42.221.26) |
| 17:06.28 | Athahira | http://pastebin.com/m41dc74d8 |
| 17:06.40 | Athahira | this also have peer info |
| 17:07.11 | Athahira | so this clearly confirm that outgoing leg is not based on peer definition !? |
| 17:07.44 | Athahira | or client definition |
| 17:07.56 | Katty | hi manx. |
| 17:08.01 | Athahira | so its based on what ?? any clue ?? |
| 17:09.00 | Athahira | i used same originating friend account ..but different software which have different codec capability |
| 17:09.28 | *** join/#asterisk iCEBrkr (n=icebrkr@72.251.206.106) |
| 17:09.35 | Katty | hi iCEBrkr |
| 17:10.01 | Athahira | [TK]Fender: u got the logs ? |
| 17:11.22 | *** join/#asterisk Godfather_ (n=godfathe@62.43.134.46.dyn.user.ono.com) |
| 17:11.28 | Athahira | i am using * 1.6.1.9 ...the latest |
| 17:11.35 | Katty | hi Godfather_ |
| 17:11.42 | Godfather_ | hi Katty |
| 17:11.48 | Katty | Athahira: patience is a virtue. |
| 17:12.08 | Athahira | oops...sorry:-) |
| 17:12.43 | [TK]D-Fender | Athahira: Your peer lists : Codec Order : (g723:30,g729:20,gsm:20) |
| 17:13.09 | *** join/#asterisk Iamnacho (i=Iamnacho@ip98-186-180-143.ks.ks.cox.net) |
| 17:13.10 | [TK]D-Fender | Athahira: And the call did : Adding codec 0x1 (g723) to SDP , Adding codec 0x100 (g729) to SDP , Adding codec 0x2 (gsm) to SDP |
| 17:13.27 | *** join/#asterisk voipmonk (n=voipmonk@69.172.114.76) |
| 17:13.29 | [TK]D-Fender | Athahira: So how is it that * is not doing EXACTLY as it said it would? |
| 17:14.20 | Athahira | did u see the second call ? from same sip client...different software ...but it only added gsm !!! |
| 17:14.31 | [TK]D-Fender | Athahira: And you are not showing a COMPLETE call with verbose, etc |
| 17:14.54 | [TK]D-Fender | Athahira: What "same SIP client"? How can that be "different software"? |
| 17:15.32 | Athahira | i use adorsoftphone to register..made the call ...close it ....then used sjphone ..registered ..made the same call |
| 17:15.50 | [TK]D-Fender | Athahira: To register to what? |
| 17:15.56 | Athahira | to * |
| 17:16.18 | [TK]D-Fender | Athahira: Please show somethign COMPLETE. |
| 17:16.20 | Athahira | and my dialplan would send calls to peer provider ...shown in the link |
| 17:16.55 | Athahira | this clearly prove that * doesn't put the peer allowed codecs !! |
| 17:17.15 | [TK]D-Fender | Athahira: I don't see a proper sip.conf entry, or [general] section, I do not see verbose of your actual call, etc |
| 17:17.30 | [TK]D-Fender | Athahira: You are masking all sorts of stuff, and YOU are making things unclear |
| 17:18.04 | TSM2 | is it possable to send a SIP Header back to the originater when i call an extention on the system?, ive tried using SIPAddHeader but cant see what im trying to send back |
| 17:18.38 | manxpower | TSM2: I don't think that is possible. |
| 17:19.41 | Athahira | hmm.ok...clients definition are on realtime database ( all have codec allowed : g729,g723,gsm) .... general section: g729,g723,gsm |
| 17:19.48 | TSM2 | without it how can asterisk support incall CID updates or say messages to phone screens |
| 17:20.06 | manxpower | TSM2: It doesn't. |
| 17:20.18 | TSM2 | i hope it will soon |
| 17:20.22 | Athahira | the peer provider as seen have this order: g723,g729,gsm |
| 17:20.31 | manxpower | TSM2: and I hope I win a million dollars |
| 17:20.40 | TSM2 | manxpower: thats what im saying, its missing a major function |
| 17:20.47 | manxpower | Athahira: don't allow g723 or g729 it won't work for you. |
| 17:20.56 | [TK]D-Fender | TSM: SIPAddHeader only adds to the initial invite |
| 17:21.09 | manxpower | TSM2: Write the SIP spec to do this and I'm sure Asterisk will support it eventually. |
| 17:21.17 | [TK]D-Fender | Athahira: Wake up and show the complete call debug, SIP & Verbose |
| 17:21.29 | TSM2 | manxpower: from what i read i thought it was a SIP spec |
| 17:21.48 | Athahira | My question is : from where * put codec availability on the seconf leg ?? |
| 17:21.52 | *** join/#asterisk uqlev (n=yuriy@91.184.221.31) |
| 17:21.58 | manxpower | [TK]D-Fender: sometimes it's better to just give up on the user and try to help people that are willing to do what needs to be done to solve the problem. |
| 17:22.01 | Athahira | second leg |
| 17:22.06 | [TK]D-Fender | Athahira: In the outbound peer |
| 17:22.20 | Athahira | no it is not !!! |
| 17:22.31 | manxpower | TSM2: You can always use something like sipsak to send custom messages. |
| 17:22.44 | Athahira | my link proved it ...u can try it ur self !! |
| 17:22.44 | *** join/#asterisk ToaSty (n=inickX@c-24-18-78-155.hsd1.wa.comcast.net) |
| 17:22.46 | [TK]D-Fender | Athahira: Provide what I have requested |
| 17:22.48 | manxpower | Athahira: [TK]D-Fender is one of the most knowledgeable people in the WORLD. He is not wrong. |
| 17:22.55 | manxpower | (about Asterisk) |
| 17:22.56 | [TK]D-Fender | Athahira: You're link is INCOMPLETE. |
| 17:23.01 | ToaSty | hello |
| 17:23.09 | *** join/#asterisk voipmonk (n=voipmonk@dsl-67-204-22-217.acanac.net) |
| 17:23.09 | [TK]D-Fender | your* |
| 17:23.25 | ToaSty | I am downloading for the first time the iso to install cant wait |
| 17:23.38 | Athahira | so..there is no simple answer .... |
| 17:23.38 | *** join/#asterisk Trupsalms (n=shawndel@adsl-68-20-35-251.dsl.chcgil.ameritech.net) |
| 17:23.42 | manxpower | ToaSty: If you are downloading an ISO this is the wrong channel. |
| 17:23.46 | ToaSty | what should I get for a small business . PCI CARD wise?.... |
| 17:23.49 | [TK]D-Fender | Athahira: There is. You aren't providing it |
| 17:23.49 | ToaSty | haha |
| 17:23.59 | manxpower | ~asteriskgui |
| 17:24.00 | [TK]D-Fender | ToaSty: Depends on your needs |
| 17:24.10 | Trupsalms | Unable to connect http://localhost:8088/asterisk/static/config/index.html |
| 17:24.12 | ToaSty | well I have 2 employees |
| 17:24.14 | Trupsalms | help |
| 17:24.25 | TSM2 | manxpower: idont think its enough, realy it needs to be inside astersik to mint off the full PBX system, its a bit of a pain that you dont get CID info when you pickup parked calls etc.. or atten transfer calls |
| 17:24.30 | [TK]D-Fender | Trupsalms: Same answer as last night... gui'S ARE not SUPPORTED IN THIS CHANNEL <- |
| 17:24.39 | ToaSty | I have 2 cell phones and I will be setting up a magicjack at my office for SIP |
| 17:24.48 | Athahira | if u said depend on ougoing peeer definition...then this is incorrect !! |
| 17:24.50 | Trupsalms | well direct me to a location please |
| 17:25.03 | [TK]D-Fender | Athahira: You aren't showing what I requestend. |
| 17:25.04 | manxpower | ~freepbx |
| 17:25.05 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 17:25.12 | manxpower | ~asterisk-gui |
| 17:25.12 | infobot | [~asterisk-gui] The Asterisk-GUI is an open source project lead by Digium. It's client-driven, and its only dependency is Asterisk itself. Check out the new version from svn here: $ svn co http://svn.digium.com/svn/asterisk-gui/branches/2.0 asterisk-gui-2.0. For support go to #asterisk-gui |
| 17:25.12 | Athahira | ok..have to leave ..will be back...don't know if really anyone have a clue !! |
| 17:25.16 | manxpower | ~asterisknow |
| 17:25.17 | infobot | i heard asterisknow is based on Asterisk, but is difficult to support in #asterisk for a number of reasons. Please seek support in #asterisknow instead. |
| 17:25.17 | [TK]D-Fender | Trupsalms: I did last night : #asterisk-gui |
| 17:25.31 | *** join/#asterisk joat (n=joat@ip70-174-79-200.hr.hr.cox.net) |
| 17:25.34 | Athahira | maybe i have to study c language to be able to answer this simple question |
| 17:25.43 | Athahira | bye all.......... |
| 17:25.58 | Athahira | be back |
| 17:26.02 | manxpower | [TK]D-Fender: Does Athahira remind you of Sarah Palin? |
| 17:26.10 | [TK]D-Fender | Athahira: Listen up I asked you to show me the ENTIRE fucking call with full SIP DEBUG and VERBSOE, now get off your ass and DO IT |
| 17:26.29 | Godfather_ | loool XD |
| 17:26.34 | odenkos | :) |
| 17:26.41 | Godfather_ | so funny |
| 17:26.41 | manxpower | i.e. Thinking they can make facts change just my wanting them to change. |
| 17:27.46 | manxpower | Athahira: Maybe Asterisk is not for you. |
| 17:27.54 | ToaSty | do i need a card to get started with this pbx idea? |
| 17:28.00 | manxpower | ToaSty: no. |
| 17:28.03 | ToaSty | or just use sip and cell |
| 17:28.05 | ToaSty | right on! |
| 17:28.07 | ToaSty | thanks |
| 17:28.10 | ToaSty | good info |
| 17:28.12 | manxpower | ToaSty: But you should read the Asterisk book |
| 17:28.13 | manxpower | ~book |
| 17:28.14 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 17:28.14 | [TK]D-Fender | ToaSty: You need a card if you want to connect physical lines to your * server |
| 17:28.26 | manxpower | or a SIP ATA |
| 17:28.39 | [TK]D-Fender | ToaSty: You have not definied your NEEDS before asking about the hardware that MIGHT relate to it. |
| 17:29.35 | ToaSty | just voicemail,press 1 for english. so far |
| 17:29.46 | Trupsalms | ok |
| 17:29.47 | ToaSty | I am going to get that book pdf today |
| 17:29.55 | Trupsalms | how to test my asterisk installation |
| 17:30.16 | [TK]D-Fender | Trupsalms: USE IT. Testing means doing. |
| 17:30.22 | *** join/#asterisk jtodd (i=sb1a9tu0@asterisk/community-director-and-tie-dye-shirt-lover/jtodd) |
| 17:30.23 | *** mode/#asterisk [+o jtodd] by ChanServ |
| 17:30.37 | Trupsalms | don't know the commands for everything |
| 17:30.52 | [TK]D-Fender | Trupsalms: [12:28]<infobot>[~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 17:30.57 | Trupsalms | test script to see if everything is running ok |
| 17:31.12 | [TK]D-Fender | Trupsalms: there is no such thing as a "test script" for * |
| 17:31.14 | manxpower | Trupsalms: We don't know either since you are not running something that is supported here. |
| 17:31.25 | ToaSty | thank you for the help |
| 17:31.26 | Trupsalms | whjat is support |
| 17:31.27 | [TK]D-Fender | Trupsalms: because there is no "standard" or means to judge "success" |
| 17:31.28 | ToaSty | I am going to do some reading |
| 17:31.35 | Trupsalms | i have ubuntu with asterisk installed |
| 17:31.40 | manxpower | Trupsalms: You are using an GUI for Asterisk, aren't you? |
| 17:31.43 | Trupsalms | this is a asterisk channel correct |
| 17:31.44 | ToaSty | i'm sure i'll be back looking for a new challenge |
| 17:32.01 | Trupsalms | no gui |
| 17:32.05 | Trupsalms | i have asterisk |
| 17:32.07 | manxpower | Trupsalms: then you are on the right channel |
| 17:32.26 | manxpower | Trupsalms: Remember Asterisk is NOT a PBX. Asterisk is a TOOLKIT that allows you to build a PBX. |
| 17:32.29 | [TK]D-Fender | Trupsalms: This is the "how to build a motor channel", not "how to drive a race-car channel" |
| 17:32.30 | Trupsalms | i'm trying to put the gui on top of its since i don't know much of the commands to configure iot |
| 17:32.48 | [TK]D-Fender | [12:31]<Trupsalms>no gui |
| 17:32.52 | Trupsalms | ok |
| 17:33.02 | [TK]D-Fender | [12:32]<Trupsalms>i'm trying to put the gui on top of its since i don't know much of the commands to configure iot |
| 17:33.15 | Trupsalms | help can i recieve help on getting just asterisk up and running with a test |
| 17:33.27 | [TK]D-Fender | Trupsalms: You are a terrible liar, avoid political careers. You don't meet the minimum requirement |
| 17:33.30 | manxpower | [TK]D-Fender: The nice thing about a full time 40+ hr/week job is that I no longer have time to help these nutjobs |
| 17:33.50 | manxpower | Trupsalms: have you read the Asterisk book? |
| 17:33.58 | [TK]D-Fender | Trupsalms: Go read the book. So far you seem to be NOWHERE. |
| 17:34.43 | manxpower | To go from nothing to a working prototype PBX you should expect to spend a 1 - 3 weeks. To go from a working prototype to a full production PBX you should expect to spend 1 - 3 months. |
| 17:34.45 | [TK]D-Fender | Trupsalms: We are not here to type out everything that it teaches to you line by line BY HAND. |
| 17:34.45 | ToaSty | I just got my ebox working nd all setup. I just wanted to ass asterisk to it ;) |
| 17:34.53 | Trupsalms | what command can i run in the terminal to show you that i have just asterisk install |
| 17:35.02 | Trupsalms | the maybe you should help me |
| 17:35.21 | manxpower | Trupsalms: pastebin your extensions.conf (pastebin.ca) |
| 17:35.28 | [TK]D-Fender | manxpower: No need yet |
| 17:35.42 | [TK]D-Fender | manxpower: he doesn't know how to tell if its even RUNNING |
| 17:35.46 | manxpower | [TK]D-Fender: it was the easiest way to know if he's running GUI configs or not. |
| 17:35.49 | [TK]D-Fender | manxpower: Don't jump the gun now. |
| 17:36.00 | [TK]D-Fender | mand we KNOW HE IS or at least planning to. |
| 17:36.17 | [TK]D-Fender | manxpower: He jsut installed it, on UBUNTU (tell-tale sign ofno pre-installed GUI). |
| 17:36.29 | [TK]D-Fender | manxpower: And he doesn't even know how to start or connect to *. |
| 17:36.32 | Trupsalms | i'm honest enought to let you know i'm planning to, but if i can not |
| 17:36.37 | manxpower | [TK]D-Fender: then he's screwed. |
| 17:36.39 | Trupsalms | i will have to go command line |
| 17:36.46 | [TK]D-Fender | manxpower: Believe me that means he has configured precisely nothing. |
| 17:37.20 | manxpower | Trupsalms: the problem is you don't even know enough to know what you know and what you don't know. Nothing wrong with that as long as you are willing to read the documentation. The Asterisk book is something you should read. |
| 17:37.29 | [TK]D-Fender | Trupsalms: go read the BOOK on the basics of starting the * process and conencting to a running instance. |
| 17:37.50 | [TK]D-Fender | Great.... attempting to flood us as well... |
| 17:38.04 | manxpower | I bet he tried pasting his extensions.conf to the channel |
| 17:38.18 | [TK]D-Fender | manxpower: Which you asked him for... silly goose! |
| 17:38.21 | p3nguin | hahaha lame |
| 17:38.29 | manxpower | No, I asked him to put it on pastebin.va |
| 17:38.37 | ToaSty | lol |
| 17:38.41 | ToaSty | *.va |
| 17:38.49 | [TK]D-Fender | manxpower: You seriously think he knows the difference on that either? |
| 17:39.06 | manxpower | [TK]D-Fender: If he doesn't he can't be helped anyway. |
| 17:40.05 | ToaSty | anyone have a setup working ebox with Asterisk? |
| 17:40.12 | manxpower | Hello everyone! I want to build a house. What is a nail? |
| 17:40.24 | manxpower | ToaSty: what the hell is an "ebox" |
| 17:40.28 | p3nguin | manxpower: What is a house? |
| 17:40.30 | ToaSty | haha |
| 17:40.31 | ToaSty | :) |
| 17:40.52 | ToaSty | http://www.ebox-platform.com/ |
| 17:40.56 | ToaSty | its a great tool |
| 17:41.10 | ToaSty | if you hate your router and enjoy linux |
| 17:41.48 | manxpower | p3nguin: I should have said something like "What is joist?" |
| 17:42.16 | ToaSty | it replaces your router |
| 17:42.33 | manxpower | ToaSty: I wish you the BEST of luck. |
| 17:42.35 | p3nguin | toasty: Software can replace my hardware? |
| 17:42.43 | p3nguin | AMAZING |
| 17:42.46 | ToaSty | thank you :) |
| 17:42.53 | ToaSty | I am going to keep on reading |
| 17:49.06 | *** join/#asterisk saghul (n=saghul@ip51ccb640.speed.planet.nl) |
| 17:56.51 | *** join/#asterisk saghul (n=saghul@ip51ccb640.speed.planet.nl) |
| 17:56.59 | *** join/#asterisk JofleyUK (n=JofleyUK@80.229.150.188) |
| 18:00.26 | dandate2 | what is ideal upstream on a comcast cable modem for pbx? |
| 18:00.35 | *** join/#asterisk saghul (n=saghul@ip51ccb640.speed.planet.nl) |
| 18:01.06 | manxpower | dandate2: The ideal would not to be using a cablemodem |
| 18:02.16 | [TK]D-Fender | [13:01]<[TK]D-Fender>dandate2: An amount greater than what you are going to place on it. |
| 18:04.53 | p3nguin | What's wrong with cable modem access for the phone system? |
| 18:05.27 | manxpower | p3nguin: no QoS, no SLA for one thing. |
| 18:05.48 | p3nguin | Not on residential service, no... |
| 18:06.16 | p3nguin | If you have business class services, you'll get an SLA. |
| 18:06.36 | *** join/#asterisk Iamnach0 (i=Iamnacho@ip98-186-180-143.ks.ks.cox.net) |
| 18:06.48 | *** join/#asterisk voipmonk (n=voipmonk@dsl-67-55-17-41.acanac.net) |
| 18:07.06 | [TK]D-Fender | manxpower / p3nguin : this topic has shifted to #freepbx ... feel free to escape it :) |
| 18:07.27 | manxpower | [TK]D-Fender: so the blind leading the blind. |
| 18:08.18 | [TK]D-Fender | manxpower: No, I'm roasting him there... just sparing all of you |
| 18:10.49 | *** join/#asterisk sizzers (n=sizzers@75.119.125.63) |
| 18:11.17 | sizzers | hey guys, i got some interesting feedback about skype for asterisk. anybody give a shit? |
| 18:11.41 | dandate2 | whats that sizzers |
| 18:12.01 | manxpower | sizzers: nopt. |
| 18:12.03 | manxpower | ..er.. nope |
| 18:12.19 | sizzers | because your first attempt at a joke failed, i will be ignoring you manxpower |
| 18:12.55 | manxpower | sizzers: no loss here. |
| 18:13.16 | sizzers | just stfu instead of being a wiseguy |
| 18:13.32 | sizzers | first MAJOR limitation, outbound calls coming from the skype for asterisk module can't use "skype out unlimited subscriptions" |
| 18:13.55 | sizzers | you're always billed per minute with S4A |
| 18:16.00 | dandate2 | skyp outbound rates are retarded, even with the ulimmited plan |
| 18:16.01 | sizzers | Since these unlimited outbound plans were the most attractive feature of "skype trunks", this is a major blemish, but ovbviously intentional and strategic |
| 18:16.11 | dandate2 | every call after 50 in a day will cost you at least 10 cents |
| 18:16.18 | dandate2 | if it was only 1 minute long |
| 18:16.38 | *** join/#asterisk e4 (n=e4@lnk2-themill-gw.binary.net) |
| 18:16.56 | sizzers | that's interesting dandate2, i read about the 10000 minute / month limit, and the 600 minute per day maximums, but did not see the 50 calls per day max |
| 18:17.04 | sizzers | is that documented somewhere? |
| 18:18.18 | dandate2 | yes |
| 18:18.25 | dandate2 | go to order unlimmited subscription in business control panel |
| 18:18.31 | dandate2 | it says that its subject to the terms of fair use |
| 18:18.33 | dandate2 | click on terms of fair use |
| 18:18.45 | dandate2 | says that any calls after 50 in a day are subject to normal rates and connection charges |
| 18:18.59 | dandate2 | then check out the rates, its 4 cents per minute and 6 cent connection charge |
| 18:19.26 | dandate2 | its to prevent large outbound call centers wtih predictive dialers i guess |
| 18:19.45 | sizzers | good catch, but that's not exactly what it says |
| 18:19.59 | sizzers | "no more than 50 different numbers in total can be called per day" |
| 18:20.45 | sizzers | now that could be interpreted one of two ways. either the way you described it, or it could allow much more than 50 calls, as long as some of the numbers were called over and over, such as a remote office or a vendor/client |
| 18:20.46 | dandate2 | well |
| 18:21.00 | dandate2 | yeah i guess that would be the case |
| 18:21.01 | sizzers | i'm betting you're correct, since that would favor them the most |
| 18:21.13 | dandate2 | i was thinking along the lines of my predictive dialer schemes are ruined |
| 18:21.14 | dandate2 | heh |
| 18:21.23 | sizzers | right, me to lol.... hang on.... |
| 18:21.30 | sizzers | dials up legal department |
| 18:21.34 | sizzers | shit, closed |
| 18:21.36 | dlynes | dandate2: well, it's designed to stop people like you, not help people like you :) |
| 18:21.53 | dlynes | but still |
| 18:21.59 | dlynes | I don't understand it |
| 18:22.08 | dandate2 | the only people that will suffer are the callees, as we jack up our rates to cover skype costs bahahahahhaah |
| 18:22.37 | dlynes | You'd think they'd like people like you..more long distance calls...thus higher call volume, thus lower termination rates (bulk discount), and thus higher profits in the end for them |
| 18:22.42 | sizzers | wow, as i reread that fair use paragraphs, i find it to be extremely vague and open to interpretation |
| 18:24.02 | sizzers | it does lower their total rates, but if they give customers the unabridged magic power of the word "unlimited" , they'll get raped |
| 18:24.33 | dlynes | Has anybody here set up hylafax to be used to send out faxes via sendfax()/txfax() on an ASP basis? i.e. use the hylafax virtual printer on windows, and still be able to account for each fax going out on a subscriber by subscriber basis? |
| 18:24.57 | dlynes | sizzers: oh...that's a different story....they would be quite stupid to do unlimited then |
| 18:25.16 | [TK]D-Fender | dlynes: Hylafax has absolutely nothing to do with SendFax() / ReceiveFax() |
| 18:25.26 | [TK]D-Fender | dyAnd the older fax equiv's |
| 18:26.04 | dlynes | [TK]D-Fender: hylafax would receive the print job, and then you would have some other process to dump it out into /var/spool/asterisk/outgoing, and create a call file |
| 18:26.24 | *** join/#asterisk edguy3 (n=edguy3@c-98-221-27-224.hsd1.nj.comcast.net) |
| 18:26.46 | [TK]D-Fender | dlynes: And why would you not use Hylafax itself to send and track the accounting on it? |
| 18:27.08 | dlynes | [TK]D-Fender: because then i'd have to use iaxmodem, which doesn't work so well |
| 18:27.36 | dlynes | [TK]D-Fender: and besides...iaxmodem would still ultimately need to use sendfax() to send it out t.38 |
| 18:27.43 | [TK]D-Fender | dlynes: Well... good luck trying to hack Hylafax to half of its job... |
| 18:28.04 | [TK]D-Fender | dlynes: And this would be a question fo #hylafax, not here... |
| 18:28.20 | dlynes | [TK]D-Fender: yeah...I'm only wanting to use the print feature of it, not the fax send/receive portion of it |
| 18:28.46 | dlynes | [TK]D-Fender: well...I just asked here, because i figured someone else here might have had such a use for hylafax |
| 18:33.39 | *** join/#asterisk voipmonk (n=voipmonk@dsl-67-55-17-41.acanac.net) |
| 18:50.53 | Godfather_ | wich application uses asterisk to send emails? |
| 18:52.13 | manxpower | Godfather_: sendmail |
| 18:52.19 | [TK]D-Fender | Godfather_: Only thing to do with e-mailing is the voicemail system. And the sample config calls teh sendmail binary and whatever you have on your system with that pointing to it is what gets used |
| 18:53.45 | *** join/#asterisk riddlebox (n=user@75-132-225-75.dhcp.stls.mo.charter.com) |
| 19:02.58 | *** join/#asterisk Athahira (n=chatzill@81.10.116.234) |
| 19:04.00 | Godfather_ | [TK]D-Fender, i touched something because i didnt recive the emails, and know just calling but voicemail() doesnt arrive |
| 19:04.05 | Godfather_ | http://pastebin.com/m946090f |
| 19:04.24 | Godfather_ | just dialing to 101 (name=ruiz) |
| 19:05.47 | Godfather_ | Connected to Asterisk 1.6.2.0~rc2-0ubuntu1 currently running on virtual (pid = 3101) |
| 19:05.47 | Godfather_ | [Nov 8 20:05:28] WARNING[3148]: app_voicemail.c:1419 create_dirpath: ast_mkdir '/var/spool/asterisk/voicemail/default/ruiz/tmp' failed: Permission denied |
| 19:05.47 | Godfather_ | [Nov 8 20:05:28] WARNING[3148]: app_voicemail.c:5122 leave_voicemail: Failed to make directory (/var/spool/asterisk/voicemail/default/ruiz/temp) |
| 19:05.47 | Godfather_ | virtual*CLI> |
| 19:08.02 | ChannelZ | that just looks like disk permissions |
| 19:08.31 | *** join/#asterisk Malkor (n=marco@hlle-d9ba181b.pool.mediaWays.net) |
| 19:09.49 | Godfather_ | root@virtual:/var/spool/asterisk/voicemail/testmail# ls -l |
| 19:09.49 | Godfather_ | total 4 |
| 19:09.50 | Godfather_ | drwxr-xr-x 4 asterisk asterisk 4096 2009-11-08 16:44 ruiz |
| 19:09.50 | Godfather_ | root@virtual:/var/spool/asterisk/voicemail/testmail# |
| 19:10.07 | saghul | check which user asterisk is running with and directory permissions |
| 19:10.34 | *** join/#asterisk slinksh0t (n=slinksh0@c-76-108-31-194.hsd1.fl.comcast.net) |
| 19:10.40 | Godfather_ | saghul, directory permissions seems to be ok no? |
| 19:11.27 | manxpower | Godfather_: you can't know if the directory permissions are correct until you know what user Asterisk is running as. |
| 19:11.29 | ChannelZ | it depends on who asterisk is running as |
| 19:11.29 | saghul | which is the user running asterisk? |
| 19:11.33 | Godfather_ | how i see wich user is asterisk running? |
| 19:11.40 | saghul | ps aux | grep asterisk |
| 19:11.46 | manxpower | Godfather_: one way is "ps -aux | grep asterisk" |
| 19:11.59 | manxpower | Godfather_: you might want to get a book on Linux |
| 19:12.12 | Godfather_ | asterisk 3101 0.5 4.8 47060 18604 ? Ssl 20:00 0:03 /usr/sbin/asterisk -p -U asterisk |
| 19:12.13 | Godfather_ | asterisk 3102 0.0 0.1 2868 740 ? S 20:00 0:00 astcanary /var/run/asterisk/alt.asterisk.canary.tweet.tweet.tweet |
| 19:13.17 | Godfather_ | then? |
| 19:13.22 | Godfather_ | seems ok too |
| 19:13.59 | saghul | what about voicemail/default? |
| 19:14.11 | saghul | do ls -lh /var/spool/asterisk/voicemail/ |
| 19:14.12 | *** join/#asterisk clive- (i=ident@dsl-242-186-71.telkomadsl.co.za) |
| 19:14.35 | Godfather_ | drwxr-xr-x 3 asterisk asterisk 4,0K 2009-11-08 16:44 testmail |
| 19:15.03 | saghul | and /var/spool/asterisk/? |
| 19:15.05 | clive- | hi guys, can I use asterisk 1.6 to send out multiple faxes using t.38 ? |
| 19:15.14 | ChannelZ | but what about the directory it's self |
| 19:15.17 | saghul | it's trying to write to voicemail/default |
| 19:15.36 | Godfather_ | saghul, http://pastebin.com/m23c80070 |
| 19:16.07 | ChannelZ | so it can't write into the voicemail directory |
| 19:16.08 | saghul | voicmeial folder is owned by 'godfather' |
| 19:16.13 | saghul | that's the point |
| 19:16.30 | saghul | do chown -R asterisk.asterisk /var/spool/asterisk/voicemail |
| 19:17.03 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 19:18.24 | [TK]D-Fender | Godfather_: ruiz => test,Ivan Ruiz,myemail@gmail.com <- "ruiz" = BAD. These are supposed to be NUMBERs |
| 19:18.40 | Godfather_ | it might be 101 |
| 19:18.48 | [TK]D-Fender | chown -R asterisk.asterisk /var/spool/asterisk/ |
| 19:18.53 | [TK]D-Fender | Godfather_: there is no "might" |
| 19:19.34 | [TK]D-Fender | And the rights are dependent on WHO * is running as |
| 19:20.01 | p3nguin | dares godfather_ to dial "ruiz" from his touchtone keypad |
| 19:20.03 | [TK]D-Fender | This did assume that * is running as user "asterisk", but I don't have any particular reason to believe such |
| 19:20.14 | Godfather_ | saghul, [TK]D-Fender know works the permisions. |
| 19:20.18 | [TK]D-Fender | p3nguin: he dialed CC102 yesterday |
| 19:20.30 | p3nguin | I know. I was really dumbfounded. |
| 19:20.46 | Godfather_ | p3nguin, i dial 101 wich translated to "ruiz" |
| 19:20.49 | [TK]D-Fender | p3nguin: Don't be... he's using a softphone. May as well have dialed "FRED" |
| 19:20.52 | p3nguin | I've seen softphone with A,B,C, and D. |
| 19:20.56 | ChannelZ | full size keyboard hooked up to his phone |
| 19:21.06 | Godfather_ | exten => 101,2,Dial(SIP/ruiz,5) |
| 19:21.14 | ChannelZ | Extensions wit umlats! |
| 19:21.14 | Godfather_ | no keyboard needed |
| 19:21.55 | Godfather_ | [TK]D-Fender, now im using softphones and ip-phones |
| 19:21.55 | ChannelZ | Godfather_: Yes but the mailbox should be 101 then. Unless you only care that voicemail boxes are accessable only by being hardcoded into the dialplan |
| 19:22.34 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 19:22.43 | Godfather_ | i replaced it to |
| 19:22.47 | Godfather_ | [default] |
| 19:22.48 | Godfather_ | 101 => test,Ivan Ruiz,myemail@gmail.com |
| 19:22.58 | Godfather_ | i try. |
| 19:24.35 | Godfather_ | exten => 101,5,VoiceMail(ruiz) |
| 19:24.40 | Godfather_ | then this should be 101 no? |
| 19:24.50 | Godfather_ | exten => 101,5,VoiceMail(101) |
| 19:25.40 | [TK]D-Fender | Godfather_: Better |
| 19:25.42 | p3nguin | Perhaps you need to not dial letters. |
| 19:25.59 | Godfather_ | p3nguin, I NOT dial letters |
| 19:26.07 | p3nguin | (1321.05) <Godfather_> exten => 101,2,Dial(SIP/ruiz,5) |
| 19:26.21 | p3nguin | That is. |
| 19:26.21 | Godfather_ | p3nguin, yes i dial 101 |
| 19:26.21 | *** join/#asterisk voipmonk (n=voipmonk@dsl-67-55-17-41.acanac.net) |
| 19:26.22 | Godfather_ | no dial "ruiz" |
| 19:26.28 | Godfather_ | username = ruiz |
| 19:26.30 | p3nguin | Dial(SIP/ruiz,5) |
| 19:26.33 | Godfather_ | its the login on the sip client |
| 19:26.33 | ChannelZ | so what |
| 19:26.36 | p3nguin | What do you think that is doing? |
| 19:26.53 | Godfather_ | its dialing the user who has logged as "ruiz" |
| 19:26.56 | *** join/#asterisk DelphiWorld (n=Miranda@196.20.95.121) |
| 19:27.04 | *** part/#asterisk DelphiWorld (n=Miranda@196.20.95.121) |
| 19:27.10 | Godfather_ | i tried it, just calling from 102 to 101 and calls ruiz |
| 19:27.24 | Godfather_ | with a analogphone (pap2t) |
| 19:27.26 | p3nguin | Suit yourself. I have more important things to do than worry about your fucked up dialplan. |
| 19:27.35 | Godfather_ | haha |
| 19:27.37 | ChannelZ | SIP/text is not fucked up |
| 19:28.00 | Godfather_ | i read it on a manual, not inveted by me of course |
| 19:28.03 | *** join/#asterisk JayTee52 (n=jforde05@unaffiliated/jaytee) |
| 19:28.44 | Godfather_ | and the voicemail also |
| 19:29.20 | Godfather_ | p3nguin, http://www.asteriskguru.com/tutorials/asterisk_voicemail.html |
| 19:29.30 | Godfather_ | you must read this page i think |
| 19:32.43 | manxpower | your SIP account has NOTHING to do with your extension. |
| 19:33.19 | manxpower | If you set them the same that doesn't mean they are the same. The relation is purely in your head, not in Asterisk. |
| 19:33.49 | manxpower | The same goes for voicemail. |
| 19:34.02 | Godfather_ | http://pastebin.com/m63d98d77 |
| 19:34.12 | Godfather_ | [TK]D-Fender, i hear the voicemail well |
| 19:34.22 | Godfather_ | but im not getting the message in my email |
| 19:34.34 | [TK]D-Fender | Godfather_: Yes, well your ISP could be CLOCKING YOU <- |
| 19:34.55 | [TK]D-Fender | Godfather_: many prevent you from just acting as an SMTP server to prevent you from becoming a BOT. |
| 19:35.00 | ChannelZ | sendmail setup and otherwise doing something? (sendmail DIE DIE DIE) |
| 19:35.05 | Godfather_ | how can i test the program sendmail ? |
| 19:35.07 | [TK]D-Fender | BLOCKING* |
| 19:35.08 | [TK]D-Fender | gah |
| 19:35.25 | [TK]D-Fender | Godfather_: Look at your sendmail queue and see if the VM e-mails are in it |
| 19:36.22 | manxpower | Godfather_: once Asterisk hands the message to your Mail Transfer Agent (usually sendmail or something that acts like sendmail), then Asterisk is completely out of the equation. |
| 19:36.42 | Godfather_ | [TK]D-Fender, i got the messages in /var/spool/asterisk/voicemail |
| 19:36.47 | manxpower | Godfather_: NO! |
| 19:36.51 | Godfather_ | i can hear that |
| 19:36.55 | manxpower | /var/spool/mail |
| 19:37.02 | Godfather_ | manxpower, nope.. |
| 19:37.14 | [TK]D-Fender | Godfather_: go check SENDMAIL's QUEUE |
| 19:37.34 | [TK]D-Fender | Godfather_: and pastebin your ENTIRE voicemail.conf |
| 19:37.44 | manxpower | Godfather_: yes. /var/spool/asterisk/voicemail is where Asterisk stores stuff for you too check via VoiceMailMain. It has nothing to do with your e-mail. |
| 19:38.11 | manxpower | Godfather_: now if you don't want to try to troubleshoot your e-mail system then we really can't help. |
| 19:38.37 | Godfather_ | manxpower, i want to configure it, but no idea where |
| 19:38.45 | [TK]D-Fender | Godfather_: PASTEBIN |
| 19:38.53 | manxpower | Godfather_: that is totally outside this channel. |
| 19:38.58 | Godfather_ | [TK]D-Fender, i dont know how to see sendmail-queu |
| 19:39.05 | Godfather_ | i pastebin ok |
| 19:39.31 | manxpower | Godfather_: what does the command "mailq" give you? |
| 19:40.35 | manxpower | Godfather_: you are going to fail with Asterisk if you don't understand Linux. |
| 19:40.57 | *** join/#asterisk gr00t (n=em@78.33.161.206) |
| 19:41.23 | Godfather_ | [TK]D-Fender, http://pastebin.com/m38756cd |
| 19:41.44 | Godfather_ | manxpower, i cant do 3 thinks at the same time |
| 19:41.46 | [TK]D-Fender | Godfather_: "man sendmail" |
| 19:41.53 | TSM2 | ive seen iaxmodem but it does not seem to be for windows, is there any app that does virtual modems? |
| 19:42.33 | [TK]D-Fender | Godfather_: Don't kid yourself... you can't do TWO things at the same time :p |
| 19:42.33 | Godfather_ | haha |
| 19:42.33 | Godfather_ | xd |
| 19:42.44 | Godfather_ | i asked it before, i dont have SENDMAIL, im downloading it now |
| 19:42.51 | Godfather_ | thats the reaseon i can recieve emails |
| 19:43.07 | manxpower | Godfather_: you could be using any one of a dozen MTAs. Most distros use postfix |
| 19:43.49 | Godfather_ | manxpower, i see it on voicemail.conf, but i'll try sendmail |
| 19:44.00 | manxpower | Godfather_: what distro are you using? |
| 19:44.16 | Godfather_ | freebsd |
| 19:44.25 | manxpower | Godfather_: I'm sorry to hear that. |
| 19:44.29 | jblack | dude, sendmail is a bad call. reqally. stick with postfix |
| 19:44.30 | gr00t | Hi all - can anyone give me a bit of background on Asterisk vs AsteriskNOW vs Asteris Business Edition? We have been running various versions of Asterisk on top of a CentOS base for a few years now, but are wondering about the right route forwards in reliability, support, testing of versions prior to release, etc. of course, all this bearing in mind associated costs. Aside from Asterisk Business Edition, which we have used in the guise of various Asterisk appli |
| 19:44.32 | Godfather_ | was kidding.. ubuntu 9.10 |
| 19:44.33 | Godfather_ | :P |
| 19:45.35 | manxpower | gr00t: Asterisk is a TOOLKIT that allows you to build a PBX. AsteriskNOW is a full ISO PBX w/GUI (not supported here) and Business Edition is a pre-built Asterisk based PBX with no access to the OS or the source code. |
| 19:45.51 | manxpower | Business Edition is only supported by Digium. |
| 19:46.09 | manxpower | Godfather_: chances are you are using Postfix |
| 19:46.13 | *** join/#asterisk bn-7bc (n=bjarne-i@mac.lan.noare-1.holmedal.net) |
| 19:46.28 | ChannelZ | uses msmtp as a simple SMTP agent on his * box |
| 19:46.33 | *** join/#asterisk Tim_Toady (n=moi@adsl52-245.kln.forthnet.gr) |
| 19:46.34 | Godfather_ | manxpower, i have declare it to voicemail.conf ? |
| 19:46.37 | manxpower | And if you had responded to "what does mailq give you" we would know tht. |
| 19:46.53 | [TK]D-Fender | Godfather_: voicemail.conf calls the script YOU tell it to in there |
| 19:46.53 | Godfather_ | ok |
| 19:47.15 | manxpower | Godfather_: postfix has a "sendmail" program |
| 19:47.20 | [TK]D-Fender | Godfather_: and most MTA's have an app-switcher that subs out sendmail with a script that passes the data to their own backend binary |
| 19:47.26 | Godfather_ | root@virtual:/etc/asterisk# mailq |
| 19:47.26 | Godfather_ | MSP Queue status... |
| 19:47.39 | manxpower | That's not postfix. |
| 19:48.05 | manxpower | In any case, I not doing any more Linux support on #asterisk. I wish you the BEST of luck trying to learn Linux and Asterisk at the same time. |
| 19:48.13 | *** part/#asterisk manxpower (n=ewieling@24.42.221.26) |
| 19:49.07 | Godfather_ | [TK]D-Fender |
| 19:49.09 | Godfather_ | MSP Queue status... |
| 19:49.09 | Godfather_ | /var/spool/mqueue-client is empty |
| 19:49.10 | Godfather_ | Total requests: 0 |
| 19:49.10 | Godfather_ | MTA Queue status... |
| 19:49.10 | Godfather_ | /var/spool/mqueue is empty |
| 19:49.10 | Godfather_ | Total requests: 0 |
| 19:49.28 | gr00t | manxpower: Thanks (not sure what that was such a struggle to find online). How does everyone feel about the upgrade path/patching of AsteriskNOW? Can it be updated with normal asterisk releases? or have they cooked a few customisations in to support the GUI, "special features", etc.? |
| 19:49.35 | *** join/#asterisk Xetrov` (n=xetrov@unaffiliated/xetrov/x-827361) |
| 19:49.40 | Godfather_ | gr00t, he left. |
| 19:49.58 | jblack | Godfather_: yeah, because you're an ass |
| 19:50.03 | Godfather_ | lol |
| 19:50.05 | *** part/#asterisk jblack (n=jblack@pool-96-243-97-134.sctnpa.east.verizon.net) |
| 19:50.09 | gr00t | poop :p |
| 19:50.15 | gr00t | anyone else? :) |
| 19:51.20 | *** join/#asterisk ToaSty (n=inickX@c-24-18-78-155.hsd1.wa.comcast.net) |
| 19:51.25 | [TK]D-Fender | groot : NO |
| 19:52.21 | [TK]D-Fender | gr00t: ABE is its own branch and you buy into a "dead end" for the thought of the right to PAY for Digium support and the warm and fuzzy feeling CIO's get thinking that paying for it really makes it "better" |
| 19:52.50 | gr00t | lol I see - so ABE is not developed from the same trunk as vanilla *? |
| 19:53.09 | [TK]D-Fender | gr00t: No, it is always an older release. IIRC ABE is only up to * 1.4 |
| 19:53.17 | gr00t | (that fuzzy feeling is great though isnt it :) |
| 19:53.35 | gr00t | ok - counts ABE out then |
| 19:59.41 | gr00t | just downloading an asterisknow iso to have a look at the GUI - its labeled 1.5 - i take it that trails as well then.. (although i suppose we could just install vanilla asterisk straight over) |
| 20:01.13 | gr00t | asterisknow homepages mentions a skype gateway as a feature of asterisknow. interesting concept, as skype is great for firewall/router traversal in hotels, public wifi on iphones, etc. have yet to see a skype channel driver in addons or elsewhere without going commercial |
| 20:01.20 | *** join/#asterisk TimToady_ (n=moi@adsl138-29.kln.forthnet.gr) |
| 20:01.29 | TSM2 | gr00t: the gui's just simulate writing the dialplans |
| 20:01.55 | TSM2 | gr00t: i fyou wanted, you could use anything, freepbx, create config then remove freepbx and it would become vanilla asterisk |
| 20:03.04 | gr00t | its not the config generating so much, which is mostly automated in our deployment anyways, its the ongoing little extension changes, etc. that you could train local admin staff to make without requiring more experienced staff to ssh in or visit on-site at off-net isntalls |
| 20:04.01 | gr00t | TSM2 - intruiging (just sounds like too much of a good thing if it is just so sideways added on :) will stick it on a vm and give it a go. |
| 20:04.16 | gr00t | intriguing? (oh dear..) |
| 20:04.59 | [TK]D-Fender | gr00t: .... |
| 20:05.03 | [TK]D-Fender | SFA *is* commercial; |
| 20:05.08 | [TK]D-Fender | ~SFA |
| 20:05.16 | [TK]D-Fender | ~skypeforasterisk |
| 20:05.17 | infobot | methinks skypeforasterisk is a Digium-made channel driver that allows Asterisk to connect to the Skype network using a Skype-supported connection method. Unlike all other solutions it does not require X11, kernel modules, or a Windows machine. See http://www.digium.com/skype for details |
| 20:05.17 | hardwire | wonders what it is. |
| 20:05.18 | hardwire | oh./ |
| 20:05.20 | hardwire | hehe |
| 20:05.21 | hardwire | WOO |
| 20:05.28 | hardwire | punches [TK]D-Fender and runs off |
| 20:06.02 | *** join/#asterisk jblack (n=jblack@pool-96-243-97-134.sctnpa.east.verizon.net) |
| 20:06.20 | [TK]D-Fender | lets hit 1D-TEN-T seeking ClueBat (tm) Track'N'Thwap hardwire |
| 20:06.26 | [TK]D-Fender | his* |
| 20:06.50 | hardwire | is that anything like the hunter seekers in dune? |
| 20:07.18 | hardwire | hears a low humm of id10tid10tid10t |
| 20:07.19 | jblack | its more like a wiflebat |
| 20:07.41 | ChannelZ | whoa i read that as 'wifebat' first |
| 20:07.47 | hardwire | you would! |
| 20:07.51 | hardwire | calls the authorities. |
| 20:08.01 | hardwire | wait.. I'm supposed to not be here. |
| 20:09.16 | gr00t | ah.. I thought skypeforasterisk was one of those perpetual development products - didnt realise it was released :( |
| 20:09.27 | gr00t | thanks for your pointers by the way guys - much appreciated |
| 20:09.39 | *** join/#asterisk [8none1] (n=[8none1]@c-68-52-180-102.hsd1.tn.comcast.net) |
| 20:09.55 | [TK]D-Fender | gr00t: And AsteriskNOW is just another trixbox... minus at least a good portion of the crap |
| 20:10.08 | hardwire | hah |
| 20:10.12 | [TK]D-Fender | gr00t: CentOS + * + FreePBX |
| 20:10.35 | [TK]D-Fender | gr00t: No magic, no special extras, no extra support, etc |
| 20:11.01 | *** join/#asterisk slinksh0t (n=slinksh0@c-76-108-31-194.hsd1.fl.comcast.net) |
| 20:11.21 | gr00t | heh - in that case itll just be easier to add on freepbx if the deployment merits it. |
| 20:11.33 | gr00t | (and i was hoping for some magic, too!) |
| 20:12.03 | [TK]D-Fender | gr00t: No.. it'll simply come with it, making doing non-FreePBX installs more of a PITA for having to PULL IT OUT |
| 20:12.39 | *** join/#asterisk vader (n=me@204.183.88.101) |
| 20:12.41 | vader | hola |
| 20:12.57 | vader | have any of you guys seen a working cisco 7940G phone come up unprovisioned? |
| 20:13.17 | gr00t | not a 7940 - plenty of 7941's though.. |
| 20:13.38 | vader | it was working, there is a valid configuration file on the server for it, the tftp log shows it sending the configuration file, ive factory reset the phone, but it shows up unprovisioned |
| 20:13.46 | vader | i can ping the phone but i can't telnet into it |
| 20:13.47 | Nugget | telnet is eeeeeeevil! |
| 20:14.14 | gr00t | vader - has the firmware on the phone changed? cisco has a habbit of changing the odd field or parameters with firwmare updates |
| 20:14.34 | vader | i don't think it has |
| 20:15.39 | gr00t | does the 7940 have a status webpage like the 41's do? if so, have a look at the debug/messages output of the phone - it often shows the point where the configuration fails to load |
| 20:16.51 | gr00t | sorry vader, console logs |
| 20:16.55 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 20:18.17 | gr00t | oh while i'm here - if anyone needs a particular firmware rev for cisco phones let me know... |
| 20:19.59 | vader | hmm |
| 20:25.43 | p3nguin | vader: Did you enable telnet in the config? It is disabled by default. |
| 20:26.46 | gr00t | (if the configuration doesnt parse correctly then telnet will not be set to enabled either...) |
| 20:27.09 | vader | weird i just forced the phone to configure from my desktop's tftp server and it loaded everything fine |
| 20:27.20 | vader | i switched it back to my asterisk server and tada everything is fine |
| 20:27.25 | vader | phone works again |
| 20:27.26 | vader | weird |
| 20:27.32 | vader | atftd is really weird |
| 20:27.38 | vader | my polycom phone didn't like it either |
| 20:27.55 | gr00t | vader - maybe check if the phone actually downloads from your ast tftp by changing something in the config, and isnt just retaining that from your desktop |
| 20:27.55 | p3nguin | I think polycom hates all tftpds. |
| 20:28.19 | hardwire | tftpd-hpa works well |
| 20:28.23 | vader | gr00t i did it pulled it from the server |
| 20:28.27 | vader | checked the logs and all |
| 20:28.33 | gr00t | weird... |
| 20:29.02 | p3nguin | hardwire: That's what I have! version 5.0 |
| 20:29.56 | gr00t | p3nguin and hardwire - does -hpa give any benefits of note? |
| 20:30.50 | JofleyUK | have any of you guys had probs with really choppy audio on the system prompts after installing 2.6 upgade? |
| 20:32.12 | vader | i would love to know why this one guy gets echoy phone calls every day |
| 20:32.17 | vader | ive changed his phone |
| 20:32.17 | gr00t | JofleyUK: only in VM environments (resolved by changing clocksource and divider) |
| 20:32.36 | *** join/#asterisk coil (i=imgay@unaffiliated/coil) |
| 20:32.40 | coil | how is babby formed? |
| 20:32.48 | JofleyUK | aha!! I was running it on *cough *cough MS Virtual PC environment. |
| 20:32.56 | gr00t | vader: just the one guy? does he call/get called by anyone unique? (if so, maybe far-end echo?) |
| 20:33.00 | JofleyUK | Just for testing I might add :P |
| 20:33.03 | coil | They need to do way instain mother> who kill thier babbys. becuse these babby cant frigth back it was on the news this mroing a mother in ar who had kill her three kids . they are taking the three babby back to new york too lady to rest my pary are with the father who lost his chrilden ; i am truley sorry for your lots |
| 20:33.26 | gr00t | JofleyUK: we run some production boxes in 2008 r2 with hyper-v, but have not tried virtual pc.. |
| 20:33.30 | vader | people on the other end can't hear it |
| 20:33.32 | vader | only hime |
| 20:33.34 | gr00t | the same fix might apply though (digs up link) |
| 20:34.00 | JofleyUK | just to be a bit of a n00b but how do i change them gr00t |
| 20:34.11 | JofleyUK | in Asterix? |
| 20:34.32 | gr00t | no, sorry - they are kernel switches |
| 20:34.37 | gr00t | just grabbing the relevant link |
| 20:34.54 | gr00t | this is a VMWare link, but the suggestions are relevant :) |
| 20:34.54 | gr00t | http://kb.vmware.com/selfservice/microsites/search.do?language=en_US&cmd=displayKC&externalId=1006427 |
| 20:35.04 | JofleyUK | hmmm... might have to use vmware then as MS Virt PC is extremely simplistic in configuration |
| 20:35.13 | gr00t | watch out for x64/x86 differences |
| 20:35.22 | JofleyUK | i will :) |
| 20:35.29 | JofleyUK | Thanks btw |
| 20:35.50 | gr00t | pleasure |
| 20:35.58 | gr00t | (but not as much as this beer) |
| 20:36.42 | vader | from our core switch we convert from copper to fiber then it's a fiber run to his room where he has another converter and then the phone |
| 20:36.49 | vader | tried different phones, different converters |
| 20:36.54 | vader | no error's on his line |
| 20:36.57 | gr00t | vader: is this an ip-only call? or does it use a telco line? |
| 20:37.01 | *** join/#asterisk confI (n=config@ip68-12-95-243.ok.ok.cox.net) |
| 20:37.05 | vader | we have a pri |
| 20:37.12 | vader | it's a cisco 7940g on his end |
| 20:37.24 | gr00t | any hardware echo cancellation? |
| 20:37.35 | Katty | hello folks |
| 20:38.12 | vader | on the pri i guess |
| 20:38.13 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 20:39.12 | gr00t | is that external pri or add-in card under asterisk? (if ast, what make?) |
| 20:47.39 | dan__t | 'morning. |
| 20:49.42 | b14ck | morning |
| 20:49.55 | dan__t | did some hacking last night on what you recommended and I almost got what I was looking for. |
| 20:50.06 | Katty | i did some sleeping last night |
| 20:50.06 | dan__t | i still have to initiate it via DTMF actions. |
| 20:50.13 | dan__t | Next step is omitting that part. |
| 20:50.19 | dan__t | haha nice, me too. |
| 20:51.07 | b14ck | dan__t, ya i actually wrote a full conferencing app which allows users to do just that =p |
| 20:51.14 | b14ck | it only has basic funcitons programmed in |
| 20:51.19 | b14ck | like my voice up |
| 20:51.21 | b14ck | my voice down |
| 20:51.24 | b14ck | mute me |
| 20:51.26 | b14ck | unmute me |
| 20:51.27 | b14ck | etc |
| 20:51.56 | dan__t | Vitelity locked my account because I forgot my password. |
| 20:51.58 | dan__t | Not happy. |
| 20:52.09 | b14ck | use flowroute |
| 20:52.10 | b14ck | ! |
| 20:52.13 | b14ck | www.flowroute.com! |
| 20:52.24 | b14ck | they are cheaper / cooler |
| 20:52.45 | dan__t | I'm not going to start wars or anything but I've called Vitelity a few times for some questions and I think there's just one engineer working there. |
| 20:52.46 | dan__t | He's a dick. |
| 20:52.59 | dan__t | holy f flowroute is cheap. |
| 20:53.34 | gr00t | (unholy cheap!) |
| 20:53.58 | dan__t | Seriously. |
| 20:53.59 | TSM2 | how much are their trunks? do they have call packages |
| 20:54.12 | TSM2 | IAX or SIP |
| 20:54.14 | b14ck | flowroute is the best sip provider |
| 20:54.18 | b14ck | they only do sip, no iax |
| 20:54.30 | b14ck | I know this because I know their CEO personally, and they do everything right. |
| 20:54.42 | b14ck | They are some of the smartest people i know |
| 20:54.48 | b14ck | And their goal is to never ever have to do work ever |
| 20:54.51 | dan__t | Bah, see that's the thing, I want IAX |
| 20:55.29 | dan__t | Because I'm a BOFH and that's what I want. |
| 20:56.16 | b14ck | =/ |
| 20:56.29 | dan__t | Guess technically speaking it doesn't matter |
| 20:56.45 | dan__t | I've never used sip with a provider before, suppose IAX2 is the difficult of the two eh? |
| 20:57.08 | *** join/#asterisk tzafrir (n=tzafrir@bzq-218-155-145.cablep.bezeqint.net) |
| 20:57.11 | b14ck | SIP is a little easier, but works almost the same way. |
| 20:57.16 | dan__t | Yeah. |
| 20:57.22 | b14ck | It just uses different ports. |
| 20:57.34 | gr00t | iax2 is a crapload easier on firewall/routing |
| 20:57.43 | TSM2 | ~iax |
| 20:57.44 | infobot | it has been said that iax is port 5036 for the original (deprecated) IAX protocol. Port 4569 is for the the current IAX2 protocol. IAX is pronounced "Eeks". stands for Inter-Asterisk Exchange |
| 20:57.49 | TSM2 | ~iax2 |
| 20:57.49 | infobot | rumour has it, iax2 is http://www.voip-info.org/wiki-IAX |
| 20:57.53 | b14ck | I don't really see how its easier to be honest |
| 20:58.05 | dan__t | Yeah I'm familiar with IAX2. I am worried about punching through this BS firewall. |
| 20:58.13 | TSM2 | IAX is much easier on firewalling as it uses one tunnel in and out for all calls |
| 20:58.19 | b14ck | The difference is forwarding 1 port (4569UDP) vs. 10,001 ports (5060UDP + 10k->20k UDP) |
| 20:58.28 | TSM2 | so unlike SIP it does not try opening loads of ports |
| 20:58.38 | dan__t | I know :) |
| 20:58.43 | dan__t | I guess I could just dmz this machine. |
| 20:58.48 | *** join/#asterisk fnordus (n=dnall@70.70.0.215) |
| 20:58.49 | TSM2 | also IAX saves banwidth |
| 20:58.50 | dan__t | Get it over with. |
| 20:58.53 | b14ck | dont dmz! |
| 20:58.55 | dan__t | !! |
| 20:58.58 | b14ck | !! |
| 20:58.58 | TSM2 | when there is more than 2 concurrent calls |
| 20:59.04 | dan__t | Reminds me, I need to call Cox and get a second IP. |
| 20:59.10 | TSM2 | why dont DMZ i do, but its a firewalled DMZ |
| 20:59.30 | b14ck | That's precisely why (most people don't know how to handle their network setup) |
| 20:59.57 | b14ck | So when they DMZ they just wait to get raped by some brute force ssh bots =/ |
| 20:59.58 | gr00t | dmz gets around a lot - could even port forward all and then blacklist with a firewall - the killer is if you have any natp traffic over the same link, which can really screw with your sip traffic (its that 10000-20000 range) |
| 21:00.04 | dan__t | Ok, signing up for flowroute. b14ck, can I use a code from you or something for you to get an affiliate kickback or whatever? Do they do something like that? |
| 21:00.11 | b14ck | dan__t, nope |
| 21:00.22 | b14ck | dan__t, you can create an account with them with no cc info |
| 21:00.27 | b14ck | and test it out first |
| 21:00.30 | dan__t | Yeah. Ask them when they'll do IAX2. |
| 21:00.32 | b14ck | it's all auto-provision |
| 21:00.56 | drmessano | DMZ on most "routers" does some nasty packet hackery and isn't a real DMZ |
| 21:00.59 | TSM2 | b14ck: ive got a dual sonicwall 2040 units, one port for LAN, one for DMZ and one for WAN, one for HA, everything blocked to the DMZ and only certian ports open |
| 21:01.06 | drmessano | Proper port forwarding trumps all |
| 21:01.11 | TSM2 | naa mine is a proper DMZ |
| 21:01.18 | TSM2 | with transparent firewall |
| 21:01.24 | dan__t | I trust this old cisco 806 :/ |
| 21:01.27 | dan__t | what a piece of s. |
| 21:01.28 | b14ck | just uses pfsense |
| 21:01.31 | b14ck | =p |
| 21:01.47 | gr00t | drmessano: any tips? (with sip) some pretty elaborate forwardings/triggers, but still get some crappy situations... |
| 21:02.13 | TSM2 | b14ck: yup but i like some hardware kit for things like firewalls etc |
| 21:02.22 | gr00t | (noone likes 10000 ports open :p |
| 21:02.28 | drmessano | gr00t: Elaborate forwardings and triggers? 5060 + RTP range the proper settings in SIP.conf |
| 21:02.30 | drmessano | ~sipnat |
| 21:02.31 | infobot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 21:02.38 | drmessano | Thats all there is to it.. Nothing elaborate |
| 21:03.20 | drmessano | 5060 + RTP range + the proper settings in SIP.conf, rather |
| 21:03.28 | dan__t | I'm not completely retarded, I promise. |
| 21:03.48 | gr00t | drmessano: do you tune your rtp range in sip.conf any particular way? (never bothered tbh!) |
| 21:03.58 | drmessano | gr00t: Nope |
| 21:03.59 | TSM2 | ~rtp |
| 21:04.00 | infobot | from memory, rtp is The Internet-standard protocol for the transport of real-time data, including audio and video. RTP is used in virtually all voice-over-IP architectures, for videoconferencing, media-on-demand, and other applications. A thin protocol, it supports content identification, timing reconstruction, and detection of lost packets. |
| 21:04.05 | TSM2 | ~rtp.conf |
| 21:04.20 | drmessano | Why bother.. 10000 ports or 20 ports.. same Daemon.. |
| 21:04.44 | drmessano | If you're gonna attack the listener, who cares how many ports it's using? |
| 21:04.52 | gr00t | true, but sometimes there might be other traffic on the same wan link that likes to use that (large) range? |
| 21:05.29 | gr00t | drmessano: re attack - fair point |
| 21:05.37 | drmessano | gr00t: from the same box? |
| 21:05.48 | drmessano | or a different box? |
| 21:06.05 | drmessano | If its another box that needs to have ports forwarded in that range, duh, change the config |
| 21:06.30 | TSM2 | single IP for voip server |
| 21:06.55 | drmessano | apples are red, but not always? |
| 21:07.00 | dan__t | Awesome, my GF locked her keys in her car. |
| 21:07.13 | dan__t | Guess I'll be back in a bit. Thanks again for the help guys. |
| 21:07.26 | TSM2 | one single voip box can need a fair ammount of ports, ftp/tftp/dns/ssh/http/https/sip/iax/rtp/AMI |
| 21:07.31 | gr00t | drmessano: random clients in an office kind of thing |
| 21:07.53 | drmessano | gr00t: If the ports are forwarded.. guess what? |
| 21:08.49 | drmessano | NAT 101.. Also known as that scene from "Better off Dead" .. "Go that way, really fast.. if something gets in your way, turn..." |
| 21:10.47 | gr00t | drmessano: youre right of course, but for some unexplained reason we get occasional spates of retransmissions. i guess the key is to insist on some decent wan hardware that does things properly. still, without excusing my own incompetence, i do like the simplicity of iax port usage :) |
| 21:11.12 | drmessano | IAX is a waste of time |
| 21:11.25 | TSM2 | why dr? |
| 21:12.14 | drmessano | You're NOT going to find someone using a decent implementation of IAX on the far end unless the other box is YOURS.. Beyond that, they could easily be using some far buggier Asterisk implementation or god help you, someone elses |
| 21:13.26 | TSM2 | yeh but for interoffice links then its ok |
| 21:13.32 | drmessano | IAX is nice for interoffice where you can guarantee both boxes are using something stable, but for interconnect to an ITSP with some unknown implementation, good luck |
| 21:13.58 | gr00t | drmessano - that something you see often? (non-asterisk stable iax2 implementations) |
| 21:13.59 | TSM2 | also from the angle that a lesser used protocall is usualy less prone to attack |
| 21:14.00 | drmessano | Case in point: Use IAX with LES.net |
| 21:14.25 | TSM2 | i would not use LES.net, i dont like their website |
| 21:14.53 | drmessano | TSM2: You must be new here.. ITSP sites that are more flashy are usually more fishy.. |
| 21:15.15 | drmessano | TSM2: The crap sites usually have the best service offerings |
| 21:15.25 | drmessano | Well, to a point |
| 21:15.41 | TSM2 | i dont use ITSP much as our PSTN is cheeper than nearly all voip providders in the UK |
| 21:16.02 | drmessano | gr00t: better question is, are there a lot of stable IAX2 implementations? Later 1.4, 1.6 asterisk.. beyond that, hit and miss |
| 21:16.42 | drmessano | I can grab about a dozen 1.4 releases and find major IAX2 issues, and I dont mean security-wise, though we have those too |
| 21:16.51 | gr00t | drmessano: got to be honest - not had enough experience to have an opinion on that yet, but will bear it in mind |
| 21:16.53 | Katty | omnomnomnomnoms orange |
| 21:18.10 | gr00t | TSM2 - cheaper pstn? do you mean call rates or line rentals? (or both) and if so, which company? |
| 21:18.17 | *** join/#asterisk correcaminos (n=laguilar@201.199.12.190) |
| 21:18.20 | thehar | steals Katty's delicious orange |
| 21:18.26 | Katty | cries |
| 21:18.31 | thehar | takes a bite and gives it back |
| 21:18.33 | thehar | there there |
| 21:18.35 | Katty | :> |
| 21:18.37 | TSM2 | gr00t: we dont pay for line rental :p |
| 21:18.39 | Katty | omnomnomnomnoms orange |
| 21:18.42 | thehar | hehe |
| 21:18.46 | *** part/#asterisk coil (i=imgay@unaffiliated/coil) |
| 21:18.58 | gr00t | TSM2 - bah not fair :P |
| 21:19.02 | Katty | i have an AMAZING alchamaholic beverage. |
| 21:19.07 | Katty | it is called Apple Pie |
| 21:19.17 | thehar | are you drinking it |
| 21:19.23 | Katty | 2 gallons apple cider, 2 gallons apple juice, 6 cinnamon sticks, 4 cups sugar. simmer 4 hours. |
| 21:19.27 | TSM2 | gr00t: we are a image company, used to have massive ISDN call bills, so they gave line rental for free |
| 21:19.30 | Katty | dump in bottle of everclear. enjoy. |
| 21:19.41 | TSM2 | gr00t: thats all gone down now but they never changed therates |
| 21:19.43 | *** join/#asterisk gr0mit (n=tim@2001:8b0:3d9:0:9d8b:c40b:60d7:b426) |
| 21:19.48 | thehar | ohmai Katty |
| 21:20.05 | Katty | it's gooooooooooooooooooooooooooood. |
| 21:20.05 | TSM2 | gr00t: still its not bad when you get 9p/m for some mobiles |
| 21:20.17 | Katty | but it creates like 4 gallons. |
| 21:20.20 | Katty | so.. |
| 21:20.27 | Katty | not very potent with 1 bottle of everclear |
| 21:20.29 | thehar | drink it all |
| 21:20.40 | Katty | you'd get the diabetus. |
| 21:20.45 | thehar | and die |
| 21:21.04 | gr00t | TSM2 - nice - mobile rates too. the reason we often choose voip over pstn is line charges/isdn hardware outlay... mobile costs always suck unless you do some real quantity - even from overseas providers |
| 21:21.05 | Katty | and probably have to pee a lot |
| 21:21.08 | thehar | lots |
| 21:21.36 | Katty | but it's very dreamy, in the evening... |
| 21:21.41 | Katty | warm. with a little ice cream. |
| 21:22.16 | TSM2 | gr00t, i worked out i can still get cheeper if i wanted with a mix of a ISTP for some calls etc.. but i worked out that if keep cutting the calls from the telco they may finaly get totaly annoyed |
| 21:22.20 | thehar | i think you're dreamy |
| 21:22.59 | gr00t | TSM2: i think youre likely right there |
| 21:23.32 | thehar | is playing |
| 21:24.16 | TSM2 | gr00t, for the sake of mabey saving £20-30pm out of £200 then im not gona worry |
| 21:24.37 | TSM2 | gr00t, also the ISDN is supplied by BT, so atleast its got the fix time we require |
| 21:24.55 | TSM2 | gr00t, i have an account with voipfone just incase there is any problems |
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| 21:59.55 | Katty | thehar: <3 |
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| 22:37.41 | ardor | Hi, how is everyone's weekend going |
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| 22:47.54 | ChannelZ | sucking ass, coming down with a cold. |
| 22:52.35 | ardor | I am passing a macro a ARG1 = to "blah blah blah VER blah blah blah blah" |
| 22:53.04 | ardor | I want to replace "VER" inside the marco with "yes" if the user presses 1 and "no" if the user presses 2 |
| 22:53.22 | ardor | But I dont know how. |
| 22:53.35 | ardor | is there a function that will do a find and replace? |
| 22:53.45 | *** join/#asterisk KavanS (n=KavanS@static-173-50-141-22.ptldor.fios.verizon.net) |
| 22:53.45 | ardor | inside a string |
| 22:56.10 | *** join/#asterisk manxpower (n=ewieling@24.42.221.26) |
| 22:56.21 | voipmonk | thats what "Set" is for |
| 22:57.23 | ardor | http://www.voip-info.org/wiki/view/Asterisk+cmd+Set |
| 22:57.31 | ardor | I dont see anything that lets me find and replace. |
| 22:57.49 | ardor | voipmonk: do you have a better url, i could see |
| 22:58.49 | voipmonk | you dont want to send a new string, you want to replace a word inside a string with YES, right? |
| 22:59.33 | ardor | yes, turn "blah blah ver blah blah" to "blah blah yes blah blah" |
| 23:00.07 | voipmonk | blah blah ${answer} blah blah |
| 23:00.38 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 23:00.51 | ardor | the problem is ${ARG1} = "blah blah ver blah blah" |
| 23:01.23 | ardor | so i need to do set(new=${ARG1},replace,ver,${answer}_ |
| 23:02.22 | voipmonk | heh |
| 23:02.25 | voipmonk | you can use "CUT" |
| 23:02.40 | voipmonk | or you can rewrite the arg with you new answer |
| 23:03.00 | voipmonk | if they push 1 then change it to say yes |
| 23:03.50 | ardor | well the marco that is getting ${ARG1} only knows somewhere inside it has "ver" other then that its clueless as to what ${ARG1} means |
| 23:04.04 | ardor | with that understanding, can i rewrite the arg with my new answer? |
| 23:05.58 | ardor | this should work i think. |
| 23:05.58 | ardor | http://www.voip-info.org/wiki/view/Asterisk+func+regex |
| 23:07.42 | saghul | AFAIK regex just matches the string, it doen't do the replace itself. |
| 23:08.21 | voipmonk | oh good lord |
| 23:08.47 | manxpower | ardor: there is no equiv of the perl s/// in Asterisk. |
| 23:09.14 | manxpower | ardor: you can either find another way to do what you want or write an AGI script to do what you want. |
| 23:09.34 | ardor | Thanks Manxpower |
| 23:10.08 | *** join/#asterisk JAMMAN2110 (n=james@unaffiliated/jamman2110) |
| 23:10.18 | manxpower | ardor: I was also disappointed that ${REGEX} did not do what I wanted. |
| 23:10.40 | manxpower | I ended up using CUT() with a FOR loop. |
| 23:10.41 | *** join/#asterisk johnakabean (n=john@static-173-50-101-14.nrflva.east.verizon.net) |
| 23:11.07 | ardor | Yeah, I am going to go with a shellscript or an AGI |
| 23:11.13 | johnakabean | do you have to buy more than 1 license to get more than 1 call at a time? |
| 23:11.27 | [TK]D-Fender | REGEX does not do actual substitution, it only confirms a match |
| 23:11.34 | johnakabean | for g.729 |
| 23:11.35 | johnakabean | sorry |
| 23:11.46 | [TK]D-Fender | johnakabean: 1 license = 1 transcode |
| 23:11.48 | manxpower | johnakabean: yes |
| 23:12.00 | johnakabean | can you put more than one license on one system? |
| 23:12.25 | johnakabean | that's going to be complicated to have 400 channels |
| 23:12.43 | [TK]D-Fender | johnakabean: the licens you pay for includes the QTY |
| 23:13.00 | [TK]D-Fender | johnakabean: and at 400 this is too much for 1 box... |
| 23:13.17 | [TK]D-Fender | johnakabean: Consdier Digiums Transcoder card |
| 23:13.33 | johnakabean | ahhh i knew there had to be a cheaper alternative |
| 23:13.38 | manxpower | You buy 1, 2, 3, or 10 licenses or whatever number of licenses you need. |
| 23:13.42 | [TK]D-Fender | johnakabean: 2 tops per server.... this is going to get nasty |
| 23:13.51 | manxpower | johnakabean: no, the Transcoder card would be more expeneive. |
| 23:14.07 | johnakabean | it can't cost more than 4000 |
| 23:14.09 | johnakabean | can it? |
| 23:14.30 | [TK]D-Fender | johnakabean: So speaks the person who talks first and looks only after repeated badgering |
| 23:15.10 | manxpower | VoIP Transcoding PCI-Express x1 card for 120 G.729a or 92 mixed channels of G.729a/G.723.1 Price: $1424.50 |
| 23:15.54 | manxpower | With the transcoder card it comes out to be a little under $12 per channel |
| 23:15.59 | *** join/#asterisk Maliuta (n=scooby@kiev.lusan.id.au) |
| 23:16.28 | manxpower | johnakabean: I doubt you'll find anything out there than can transcode 400 g729 channels |
| 23:16.32 | [TK]D-Fender | manxpower: And don't worry about bus or CPU load.. his deluxe Optiplex servers can handle it! |
| 23:17.29 | johnakabean | if it uses pci express 2 then it could |
| 23:17.32 | manxpower | [TK]D-Fender: the point of a transcode card is to offload the processing from the CPU. A 64K ulaw channel is not going to flood the bus unless you are doing massive numbers of transcodes |
| 23:18.11 | johnakabean | i have no problem with the cpu: xeon |
| 23:18.17 | johnakabean | 4 gigs of ram |
| 23:18.34 | johnakabean | well, at the moment |
| 23:18.36 | [TK]D-Fender | johnakabean: 2 servers, 2 cards each... |
| 23:18.39 | johnakabean | don't know about doing g 729 |
| 23:18.54 | johnakabean | i have no virtualization; centos gets the whole system to itself |
| 23:19.04 | manxpower | johnakabean: Exactly what sort of transcoding are you planning on doing? |
| 23:19.21 | johnakabean | i just want to reduce bandwidth so i could max out the server's potential |
| 23:19.32 | johnakabean | right now its 400 ulaw calls averagely |
| 23:19.37 | JAMMAN2110 | You want to max out your PBX? |
| 23:19.42 | johnakabean | 400 ulaw maxes a T1 |
| 23:19.52 | [TK]D-Fender | johnakabean: Switching to G.729 only HURTS your potential |
| 23:19.58 | johnakabean | i want it to be able to get its fully potential |
| 23:20.06 | manxpower | johnakabean: 24 calls maxes out a T-1 |
| 23:20.12 | johnakabean | no, 400 calls |
| 23:20.19 | [TK]D-Fender | johnakabean: BAD MATH |
| 23:20.26 | johnakabean | i don't use the T1 voice |
| 23:20.28 | manxpower | johnakabean: If you want to max your server don't transcode. It's as simple as that |
| 23:20.30 | johnakabean | every channel is data |
| 23:20.49 | [TK]D-Fender | johnakabean: T1 = 1.544mbit : SIP + ULAW =85kbit |
| 23:20.57 | johnakabean | yes |
| 23:20.57 | [TK]D-Fender | johnakabean: Twit :p |
| 23:21.07 | johnakabean | i used 86 in my math |
| 23:21.15 | hardwire | meh? |
| 23:21.17 | manxpower | It is a tradeoff. More CPU used = less bandwidth used. more bandwidth used = less CPU used. |
| 23:21.21 | [TK]D-Fender | johnakabean: So how do you fit 400 calls into a T1? |
| 23:21.30 | johnakabean | compression |
| 23:21.34 | [TK]D-Fender | johnakabean: ..... |
| 23:21.49 | manxpower | johnakabean: You are never going to compress 400 calls. |
| 23:21.55 | manxpower | (on one server) |
| 23:22.17 | johnakabean | well with 200 calls at any given moment, its still capable of more |
| 23:22.27 | johnakabean | right now there is 176 |
| 23:22.38 | manxpower | johnakabean: what codec are you transcoding to? |
| 23:22.42 | johnakabean | ulaw |
| 23:23.01 | johnakabean | what would be better? |
| 23:23.04 | manxpower | johnakabean: Um, when you transcode you convert from one codec to another codec. ulaw<->ulaw is NOT transcoding |
| 23:23.06 | johnakabean | with great call quality |
| 23:23.13 | johnakabean | then i'm not transcoding |
| 23:23.23 | manxpower | There ya go! That was simple wasn't it. |
| 23:23.25 | [TK]D-Fender | johnakabean: And you're also not pumping 176 ULAW channels over a T1 |
| 23:23.27 | johnakabean | all endpoints use ulaw |
| 23:23.40 | johnakabean | core show channels |
| 23:23.52 | [TK]D-Fender | johnakabean: Your enpoints aren't plugged directly to your T1 |
| 23:24.04 | johnakabean | i use a full duplex switch |
| 23:24.22 | manxpower | Like that matters |
| 23:24.35 | [TK]D-Fender | johnakabean: And you aren't getting more than 18 calls over it |
| 23:24.37 | johnakabean | all channels, up to 24, get 1.5 mb/s |
| 23:24.49 | johnakabean | each |
| 23:25.02 | [TK]D-Fender | johnakabean: johnakabean In DATA. In voice, 24 if you're on CAS, which sucks |
| 23:25.20 | johnakabean | fender, i'm not using pri |
| 23:25.32 | manxpower | Even telemarketing and call centers that use asterisk don't try to push more than about 250 calls over 1 server. |
| 23:25.37 | [TK]D-Fender | johnakabean: So if its CAS, you get all 24 instead of 23 |
| 23:25.50 | *** join/#asterisk jtodd (i=ay4lx0vo@asterisk/community-director-and-tie-dye-shirt-lover/jtodd) |
| 23:25.51 | *** mode/#asterisk [+o jtodd] by ChanServ |
| 23:25.58 | manxpower | johnakabean: don't talk about T-1 channels when you talk about data. |
| 23:25.59 | johnakabean | I took a windows sub mainframe and turned it into this |
| 23:26.07 | manxpower | or people will be confused. |
| 23:26.19 | [TK]D-Fender | [18:25]<johnakabean>I took a windows sub mainframe and turned it into this <- massive joke |
| 23:26.29 | manxpower | Ah! Now I understand. You're a troll. |
| 23:26.38 | [TK]D-Fender | Putting the words "Windows" and "mainframe" in the same sentence... LOL |
| 23:26.39 | manxpower | Glad we got that cleared up. |
| 23:26.40 | johnakabean | joke? i do small to medium business IT work |
| 23:26.46 | johnakabean | its a server that was relinquished |
| 23:26.58 | manxpower | [TK]D-Fender: we're feeding the trolls. |
| 23:27.24 | [TK]D-Fender | johnakabean: You bought a fucking DELL OPTIPLEX for use as a server and tried to scam the place out of it, are looking to pump more calls through a T1 than it supports. |
| 23:27.34 | johnakabean | i don't do huge companies, so a mainframe in small and medium business only uses one quad core cpu |
| 23:27.43 | [TK]D-Fender | johnakabean: And lets not even talk about your "enhanced security strategies" of old... |
| 23:28.03 | johnakabean | security strategies? try getting in the server |
| 23:28.07 | [TK]D-Fender | johnakabean: You clearly have no sense of scope as to what qualifies as a "mainframe" |
| 23:28.14 | johnakabean | i do |
| 23:28.20 | johnakabean | but i nickname things |
| 23:28.30 | [TK]D-Fender | johnakabean: A Power6 IBM AS/400 is MID-RANGE. |
| 23:28.36 | *** join/#asterisk Godfather_ (n=godfathe@187.Red-83-38-229.dynamicIP.rima-tde.net) |
| 23:28.43 | [TK]D-Fender | johnakabean: More like you bullshit your way through every term you use |
| 23:28.58 | hardwire | aka bean? |
| 23:29.04 | hardwire | is bean your nickname? |
| 23:29.11 | johnakabean | fender, i hope you have a degree in your office to demote me |
| 23:29.33 | manxpower | I just hope we get some non-idiots on the channel that need help. |
| 23:29.49 | Chainsaw | johnakabean: Insulting Fender never ends well. |
| 23:29.55 | [TK]D-Fender | manxpower: Roll those odds baby! |
| 23:30.03 | [TK]D-Fender | Chainsaw: \o/ |
| 23:30.04 | johnakabean | not insulting; he's insulting without a point to stand on |
| 23:30.17 | voipmonk | grabs the pop corn |
| 23:30.44 | manxpower | waves a rubber chicken at voipmonk |
| 23:30.46 | *** join/#asterisk jmworx__ (n=jeval@mail.octasic.com) |
| 23:30.47 | [TK]D-Fender | johnakabean: Shall we go back to your thoughts of RAID1 in software for a corporate server, and changing the username for the "root" account? |
| 23:30.57 | *** join/#asterisk jackal (n=jackal@pool-96-247-204-238.clppva.fios.verizon.net) |
| 23:30.57 | TJNII | disagrees with manxpower and pages DrClue and me|ong |
| 23:31.06 | *** join/#asterisk rdahlin_1 (n=rdahlin_@78-73-17-198-no168.tbcn.telia.com) |
| 23:31.21 | hardwire | am I about to witness a "critical event"? |
| 23:31.22 | johnakabean | you think changing root was the ONLY way I wanted to implement security? |
| 23:31.29 | [TK]D-Fender | johnakabean: Anyway I'd love you to show me 176 calls of ULAW over a T1.... |
| 23:31.33 | manxpower | hardwire: lets hope not. It never ends well. |
| 23:31.34 | *** join/#asterisk rajiv (n=rajiv@gentoo/developer/rajiv) |
| 23:31.49 | johnakabean | its was one of the miniscule things |
| 23:31.52 | hardwire | manxpower: well.. I don't see how it would last long. It's not hard to outclass johnakabean. |
| 23:31.55 | [TK]D-Fender | johnakabean: Heck... show me another protocol and codec you'd use to acheive this... |
| 23:32.05 | [TK]D-Fender | johnakabean: And then the server that could survive it. |
| 23:32.13 | johnakabean | just ulaw, with all endpoints doing ulaw |
| 23:32.41 | [TK]D-Fender | johnakabean: not getting 176 over a link that supports 24 |
| 23:32.47 | hardwire | johnakabean: whats 64 * 24? |
| 23:32.54 | [TK]D-Fender | johnakabean: At CAS to telco. |
| 23:36.11 | hardwire | I bet he's using that old trick where he just runs the t1 at a faster clock rate, in this case 16x. |
| 23:36.19 | TSM2 | haaa |
| 23:36.21 | voipmonk | no man |
| 23:36.23 | voipmonk | its overclocked |
| 23:36.25 | voipmonk | :) |
| 23:36.36 | hardwire | and water cooled |
| 23:36.45 | voipmonk | damn right, joo didnt know? |
| 23:36.48 | hardwire | stike that.. liquid nitrogen. |
| 23:36.57 | TSM2 | did he replace the copper too for string |
| 23:37.12 | TSM2 | you all read about string as being the next big thing in communication |
| 23:37.19 | voipmonk | Quantum Comp goes so fast it moves backward |
| 23:37.22 | manxpower | hardwire: at those temps why not use a superconductor and run it at infinite speeds. |
| 23:37.23 | TSM2 | and two plastic cups |
| 23:37.25 | hardwire | TSM2: well.. to get this trick to work you need 16x the copper pairs too. |
| 23:37.48 | hardwire | manxpower: ah.. that brings me back to ringworld. |
| 23:37.54 | TSM2 | hardwire: naa string, loads of strands, so its x16 compatable |
| 23:38.02 | johnakabean | hardwire: there are 48 wires, 24 channels, you can choose to have a minimum of ONE channel data, the rest voice; now, combining these channels, with all being data, gives 1.5 mb/s full duplex. There is a pri converter(supplied by the telco) that converts to 12 X 3mb/s ethernet, with each getting 2.9mb/s. Using 4 other centos boxes for ONLY load balancing to the 5th, asterisk server, you get 36 mb/s synchronous. |
| 23:38.10 | manxpower | hardwire: been 20 years since I read that series |
| 23:38.28 | hardwire | manxpower: you're an old toot. |
| 23:38.31 | JAMMAN2110 | Im tempted to comment, but Im going to follow voipmonks example |
| 23:38.36 | manxpower | hardwire: aye, I am. |
| 23:38.49 | johnakabean | there is more than one kind of T1 |
| 23:39.15 | manxpower | "(05:38:49 PM) johnakabean: there is more than one kind of T1" <-- I think that belongs in the /topic |
| 23:39.16 | TSM2 | sort your mb MB Mb out |
| 23:39.22 | johnakabean | the small boxes have 2.4 ghz athlon, and only do packet forwarding |
| 23:39.37 | hardwire | am I insane for thinking he mentioned 400 calls over 1.5Mbps? |
| 23:39.38 | p3nguin_ | johnakabean: full for fractional |
| 23:39.49 | johnakabean | fully |
| 23:39.55 | johnakabean | 700 a month |
| 23:39.56 | manxpower | hardwire: no. He's insane for mentioning that. |
| 23:39.57 | TSM2 | hardwire: hees talking 1.5 MBps |
| 23:40.07 | TSM2 | hees got it messed up |
| 23:40.12 | hardwire | sure |
| 23:40.18 | hardwire | even then it's only 200 calls. |
| 23:40.21 | johnakabean | no, tsm2, i am not saying million bits per second |
| 23:40.27 | johnakabean | i'm saying megabytes per second |
| 23:40.32 | *** join/#asterisk JAMMAN2110 (n=james@unaffiliated/jamman2110) |
| 23:40.34 | TSM2 | thats MBps |
| 23:40.40 | johnakabean | or MB/s |
| 23:40.42 | *** join/#asterisk cnu (n=cnu@113.80-203-44.nextgentel.com) |
| 23:40.47 | JAMMAN2110 | Well that sucked |
| 23:40.57 | *** join/#asterisk IBC_jkenney (n=jkenney@ip65-44-169-66.z169-44-65.customer.algx.net) |
| 23:41.24 | TSM2 | you were saying T1 so people went and took the standard 1.5Mbps |
| 23:41.41 | p3nguin_ | johnakabean: Connection rate is always measured in bits (or bauds). Never in Bytes -- only transfer is rated in Bytes. |
| 23:41.44 | hardwire | yeh.. ethernet != isdn |
| 23:41.55 | manxpower | TSM2: I can't imagine why. Could it be because a T-1 runs at 1.544Mbps? |
| 23:41.56 | *** join/#asterisk b14ck (n=comradeb@cpe-24-24-136-239.socal.res.rr.com) |
| 23:42.38 | JAMMAN2110 | manxpower - Your level of sarcasm amuses me |
| 23:42.42 | TSM2 | manxpower: duno we dont have T1 in the UK |
| 23:42.44 | TSM2 | :) |
| 23:42.45 | manxpower | It doesn't really matter HOW the T-1 is delivered, it's handed off to the end user as a DSX-1 1.544Mbps |
| 23:43.40 | manxpower | Maybe johnakabean should use E-1s, they run at about 2Mbps. |
| 23:43.43 | p3nguin_ | tsm2: Look up E1 |
| 23:43.50 | hardwire | moves johnakabean's decimal places around |
| 23:43.55 | TSM2 | yup i know what they are |
| 23:44.06 | TSM2 | we have it for our tel here ISDN30 |
| 23:44.29 | TSM2 | we also have 100Mb fiber straight to the main datacenter in london |
| 23:44.44 | hardwire | I'd like that kind of bdubs. |
| 23:44.48 | [TK]D-Fender | hardwire: VerizonMath approved! |
| 23:45.05 | [TK]D-Fender | T1 = 1.544 mega-BIT |
| 23:45.06 | TSM2 | so 400 channels of G729 8k requires abt 6.7 Mbps |
| 23:45.11 | hardwire | [TK]D-Fender: I really hated the mathz on that whole issue. |
| 23:45.21 | *** join/#asterisk tzafrir__laptop (n=tzafrir@212.179.75.202) |
| 23:45.21 | hardwire | It reminds me of the guy that handles our master rate sheet. |
| 23:45.26 | manxpower | TSM2: Does that include packet overhead? |
| 23:45.26 | hardwire | who.. is.. insane. |
| 23:45.40 | [TK]D-Fender | TSMbut thats only the CODEC... see telco doesn't carry that in raw frames. |
| 23:45.56 | hardwire | iax2ftw |
| 23:45.57 | [TK]D-Fender | TSM: when you consider that you're going to see that number get SMASHED |
| 23:45.58 | TSM2 | woops no 3.6Mbps |
| 23:46.29 | TSM2 | im tired, look here http://www.bandcalc.com/ |
| 23:46.37 | [TK]D-Fender | hardwire: And when you try gang-piling those channels onto IAX2... you'll watch your entire pipe jump up and down like a Jack Russell fucking terrier! |
| 23:46.42 | hardwire | TSM2: if it were all to/from the same peers.. it would be easier. |
| 23:46.44 | JAMMAN2110 | If I go away for 30 minutes and come back will this converastion still be going? |
| 23:46.56 | [TK]D-Fender | watches the calls drop like flies |
| 23:46.58 | hardwire | JAMMAN2110: did you have something more important to say? |
| 23:47.07 | TSM2 | [TK] on E1/T1 voice connections overhead is on the D chan |
| 23:47.29 | [TK]D-Fender | TSM2: No... on **PRI** |
| 23:47.33 | JAMMAN2110 | hardwire, Would you care if I did? |
| 23:47.33 | hardwire | [TK]D-Fender: Have you evidence of a magic number of calls per trunk that iax2 can handle flawlessly? |
| 23:47.41 | [TK]D-Fender | TSM2: Straight T1/E! doesn't have a "D-Chan" |
| 23:47.41 | hardwire | JAMMAN2110: that all depends on what it is. |
| 23:47.49 | manxpower | JAMMAN2110: if it's not, an equally stupid conversation will be going on. |
| 23:47.54 | TSM2 | [TK] you guys call it PRI and ISDN30 her |
| 23:48.11 | TSM2 | [TK] i keep getting confused about it |
| 23:48.16 | [TK]D-Fender | TSM2: No. PRI is a signalling protocol over a carrier like T1/E1/J1 |
| 23:48.18 | [TK]D-Fender | ~pri |
| 23:48.19 | infobot | well, pri is [~pri] Primary Rate Interface, often called T1 or E1 (European Standard). E1 offers 30 ISDN B-Channels a 64kBit/s + 1 D-Channel with 64kBit/s. The T1 has 23 B-Channels + 1 D-Channel. Cards to use with *: T100P, E100P, TE410P, R1T1,R2T1,R4T1, etc. |
| 23:48.27 | [TK]D-Fender | EW |
| 23:48.31 | [TK]D-Fender | bad botlet |
| 23:48.36 | hardwire | haha |
| 23:48.43 | hardwire | somebody misinformed the computAR! |
| 23:48.46 | [TK]D-Fender | TSM: t1 is a carrier circuit, same like E1 |
| 23:48.48 | [TK]D-Fender | ~t1 |
| 23:48.49 | infobot | [~T1] T1 is the basic digital telephony circuit used in North America. T1 runs at 1.544 Mbps. It can be an unstructured channel for data. It can be channelized, to provide 24 time slots of voice or data, each of 64kbps. Time slot 24 is used for D-Chan when used with PRI signalling. |
| 23:48.50 | [TK]D-Fender | ~e1 |
| 23:48.51 | infobot | [~E1] E1 is the basic digital telephony circuit used everywhere except the US,Japan,Taiwan & Hong Kong (who use T1 and similar variants). E1 runs at 2.048Mbps. It can be an unstructured channel for data. It can be channelized, to provide 31 time slots of voice or data, each of 64kbps. Time slot 16 is used for D-Chan when used with PRI signalling. |
| 23:49.11 | saftsack | the infobot doesn't like sangome, heh? ^^ |
| 23:49.12 | TSM2 | [TK] ok got it now, there is also J1 for japan etc |
| 23:49.15 | TSM2 | @J1 |
| 23:49.17 | TSM2 | :J1 |
| 23:49.20 | TSM2 | ~J1 |
| 23:49.21 | [TK]D-Fender | hardwire: Yeah someone mucked with my botlet... |
| 23:50.13 | JAMMAN2110 | Hmm.. Ive forgotten something |
| 23:50.51 | manxpower | JAMMAN2110: I left earlier when someone started telling [TK]D-Fender that he didn't know what he is talking about. I came back to this. |
| 23:50.53 | [TK]D-Fender | manxpower: So... a 1.544 megaBYTE connection huh? Sill me for considering T1, T2, T3 (28x T1). johnakabean seems to have invented a new standard |
| 23:50.56 | hardwire | so.. I'm actualyl only half informed.. whats johnakabean up to?' |
| 23:51.33 | JAMMAN2110 | manxpower - it was your original question though right? |
| 23:51.37 | [TK]D-Fender | manxpower: 1.5x 10base-T! |
| 23:51.39 | [TK]D-Fender | \o/ |
| 23:51.49 | hardwire | \\o |
| 23:51.51 | hardwire | o// |
| 23:51.51 | JAMMAN2110 | 10BaseT is awesome |
| 23:52.12 | manxpower | [TK]D-Fender: these days much of my life is dealing with "ethernet over copper", which I think is a silly name for something that's not actually ethernet. |
| 23:52.25 | JAMMAN2110 | pats his stack of 9 Cisco Catalyst 1900s |
| 23:52.32 | manxpower | JAMMAN2110: Hmm? No. |
| 23:52.33 | [TK]D-Fender | manxpower: its just a "nickname" :) |
| 23:52.50 | manxpower | JAMMAN2110: I love catalysts. I have a 5509 in my atic. |
| 23:53.13 | JAMMAN2110 | CC1900s are 10BaseT with 100BaseT uplinks |
| 23:53.21 | manxpower | [TK]D-Fender: like me calling sip peers "sip trucks" |
| 23:53.21 | JAMMAN2110 | 24 port |
| 23:53.28 | *** join/#asterisk correcaminos (n=laguilar@201.199.12.190) |
| 23:53.29 | hardwire | [TK]D-Fender: kinda like how one of my coworker calls "cause codes" "cost codes" |
| 23:53.38 | hardwire | as in "response codes" |
| 23:53.46 | manxpower | or calling a web browser "the internet" |
| 23:53.49 | [TK]D-Fender | hardwire: Charge him every time he uses the wrong term :p |
| 23:53.57 | hardwire | and he keeps telling me that when our providers give us 5xx.. it could mean anything. |
| 23:53.58 | manxpower | User: The internet is down! Me: Even in China? |
| 23:54.01 | hardwire | it makes me wanna hurt |
| 23:54.21 | JAMMAN2110 | [TK]D-Fender, We charge people for that at work... we make a few thousand a year at $1 per fine... donate it all to charity :) |
| 23:54.26 | hardwire | [TK]D-Fender: you should hear him talk about resporgs. |
| 23:54.33 | TJNII | God, it is getting dark outside. I just woke up. My sleep schedule is so off.... |
| 23:54.37 | hardwire | and wiresharks. |
| 23:54.54 | hardwire | JAMMAN2110: seriously? |
| 23:55.04 | JAMMAN2110 | hardwire, Seriously |
| 23:55.05 | manxpower | hardwire: do the wiresharks have frickin' lasers? |
| 23:55.06 | hardwire | who is the official know it all that passes judgement? |
| 23:55.14 | *** join/#asterisk jmacz (n=jmacz@190.27.181.213) |
| 23:55.22 | hardwire | manxpower: only if you compile it with that flag. |
| 23:55.40 | manxpower | ./configure --with-frickin-lasters |
| 23:55.46 | manxpower | and lasers too! |
| 23:56.04 | JAMMAN2110 | "Wiresharks, now with that fricken long lasting effect" |
| 23:56.11 | hardwire | and lasers. |
| 23:56.14 | hardwire | too. |
| 23:56.21 | [TK]D-Fender | How much do they cost? |
| 23:56.26 | hardwire | I don't know what anybody thinks happens in here on Sundays. |
| 23:56.30 | [TK]D-Fender | One MEEEELION DOLLARS! |
| 23:56.47 | JAMMAN2110 | Its monday here |
| 23:57.02 | hardwire | JAMMAN2110: lies. |
| 23:57.31 | JAMMAN2110 | Monday November 9 - 12:57 PM |
| 23:57.49 | hardwire | You could be making that up. |
| 23:58.18 | JAMMAN2110 | I live in New Zealand |
| 23:58.22 | JAMMAN2110 | GMT +12 |
| 23:58.24 | JAMMAN2110 | +1 for DST |
| 23:59.13 | JAMMAN2110 | New Zealand, also known as "as far away from most of the people in this room as you can get" |
| 23:59.18 | JAMMAN2110 | :) |
