| 00:03.14 | *** join/#asterisk chopp (~chopp@unaffiliated/chopp) |
| 00:05.29 | chopp | I'm trying to use msmtp instead of sendmail. Shouldn't putting "mailcmd=/usr/bin/msmtp" in voicemail.conf be working? |
| 00:27.15 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 00:27.15 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 00:30.11 | leifmadsen | Juggie: ping! |
| 00:33.53 | GlobeTrotterz | Hi Guys, I want to use the queueaddmember application to add users to the Queue. I am using the extensions.ael. it does now pass the value for ${MACRO_EXTEN}. Is there a way to do pass ${MACRO_EXTEN} with gosub ? |
| 00:36.01 | GlobeTrotterz | does NOT pass |
| 00:40.09 | ChannelZ | pass gas! |
| 00:41.04 | pabelanger | collect $200? |
| 00:44.03 | raden | LMAO |
| 00:55.49 | leifmadsen | GlobeTrotterz: uhhh... GoSub() is not a Macro() |
| 00:56.16 | leifmadsen | thus there is no ${MACRO_EXTEN} channel variable created. It's just ${EXTEN} like in the rest of the dialplan. |
| 00:57.12 | ritztech | anyone in the dialplan can you have the ability on a recording RECORD it and hang up without the # key |
| 00:57.23 | ritztech | or IF they do quickly force the pound key |
| 00:57.53 | leifmadsen | ritztech: what dialplan application? |
| 00:58.51 | ritztech | just custom |
| 00:59.02 | ritztech | exten => 1222,n,record(asterisk-recording:ulaw) |
| 00:59.04 | leifmadsen | that is not the correct answer |
| 00:59.10 | ritztech | HAHA sorry |
| 00:59.12 | leifmadsen | what version? |
| 00:59.16 | ritztech | 1.6 |
| 00:59.23 | leifmadsen | 1.6 is not a version |
| 00:59.31 | *** join/#asterisk KavanS (~KavanS@unaffiliated/kavans) |
| 00:59.32 | leifmadsen | 1.6.0, 1.6.1, and 1.6.2 are branches |
| 00:59.35 | leifmadsen | there is no 1.6 branch |
| 00:59.41 | KavanS | sipdroid with a htc incredible - anyone else using this combo with asterisk? |
| 00:59.46 | leifmadsen | anyways, did you do 'core show application record" ? |
| 00:59.57 | leifmadsen | ritztech: if you had, you'd see this: k: Keep recorded file upon hangup. |
| 01:00.17 | ritztech | im trying to not have it need a # sign Just a hangup only because my PBX on the FXO returns me a Busy tone after the hangup is used |
| 01:00.22 | ritztech | Asterisk: 1.6.2.11 |
| 01:00.28 | leifmadsen | ritztech: see answer above |
| 01:00.43 | leifmadsen | ritztech: you didn't really try very hard... |
| 01:00.44 | leifmadsen | <PROTECTED> |
| 01:00.45 | leifmadsen | <PROTECTED> |
| 01:01.25 | *** join/#asterisk GameGamer43|Mac (~GameGamer@CPE-65-27-76-78.new.res.rr.com) |
| 01:01.33 | ritztech | HOLY SHIT theres core application and examples wow nice |
| 01:01.39 | leifmadsen | lol |
| 01:01.41 | leifmadsen | wow |
| 01:01.55 | ritztech | im on crack with a tint of crack |
| 01:02.23 | leifmadsen | http://www.asteriskdocs.org |
| 01:04.06 | ritztech | so with the k do i do this |
| 01:04.07 | ritztech | exten => 1222,k,record(asterisk-recording:ulaw) |
| 01:04.18 | ritztech | exten => 1222,n,k,record(asterisk-recording:ulaw) |
| 01:04.43 | leifmadsen | LOL |
| 01:04.54 | leifmadsen | Record(filename.format[,silence[,maxduration[,options]]]) |
| 01:05.21 | ritztech | ohh , option let me try agin |
| 01:05.26 | leifmadsen | exten => 1222,1,Record(asterisk-recording.ulaw,,kx) |
| 01:05.56 | ritztech | yay :) just id it but not the 2 commas |
| 01:06.38 | leifmadsen | look at the syntax |
| 01:06.55 | *** join/#asterisk ChannelZ (~bobm@burner.com) |
| 01:07.04 | leifmadsen | in fact, there should be 3 commas |
| 01:07.19 | leifmadsen | Record(filename.format[,silence[,maxduration[,options]]]) |
| 01:07.29 | leifmadsen | ,silence,maxduration,options |
| 01:07.51 | leifmadsen | you need commas as separators if you're not passing anything to the fields |
| 01:10.13 | *** join/#asterisk Tech_Travis (~Administr@cpe-76-168-191-127.socal.res.rr.com) |
| 01:11.50 | ritztech | crapola soo it did work after i went 1222,n,Record but now it wont continue through the rest of the dialplan |
| 01:11.50 | ritztech | http://pastebin.com/mJFvcDJa |
| 01:15.23 | nny | hrrm |
| 01:15.34 | nny | so I have a directory with probably over a million files in it |
| 01:15.35 | nny | at least |
| 01:15.56 | nny | someone turned "record always" on with a freepbx system 9 months ago, and let it run on 100 concurrent channels |
| 01:16.13 | nny | i want to rm -rf it, (I can't ls it or anything) |
| 01:16.22 | nny | ls never spits output back |
| 01:16.26 | nny | just hangs and gets angry |
| 01:16.48 | nny | i am worried there may be a useful symbolic link inside of it though that point back at files I actually need |
| 01:17.29 | nny | any advice? |
| 01:17.47 | nny | hrrm |
| 01:18.01 | nny | <PROTECTED> |
| 01:20.47 | ritztech | leifmadsen what if theres more on the Dial plan does it mean it cant continue once i hangup |
| 01:22.00 | leifmadsen | ritztech: once you hangup, what would continue? |
| 01:22.14 | leifmadsen | the dialplan is related to a channel which is what is processing your call |
| 01:22.27 | ritztech | the rest of the Dialplan after the recording was recorded |
| 01:22.28 | ritztech | http://pastebin.com/mJFvcDJa |
| 01:22.48 | leifmadsen | ritztech: see the 'h' extension' |
| 01:22.56 | leifmadsen | ritztech: you really should read the documentation |
| 01:24.17 | leifmadsen | nny: I've had that problem before -- I ended up using xargs |
| 01:25.07 | leifmadsen | nny: in my particular case I was trying to scp all contents of a directory where 'ls' would not work (and thus I couldn't just scp * remotelocation:/home/lmadsen |
| 01:25.15 | ritztech | h extension ? |
| 01:25.22 | leifmadsen | ritztech: yes, the 'h' extension |
| 01:25.22 | ritztech | researching.... |
| 01:25.29 | leifmadsen | nny: here is an example: find . -maxdepth 1 -type f -print0 | xargs -0 -IFILES scp FILES files@192.168.1.4:/var/www/html/ |
| 01:25.51 | leifmadsen | ritztech: it's in the documentation at the link I provided |
| 01:25.54 | leifmadsen | ~docs |
| 01:25.54 | infobot | somebody said docs was for basic documentation of Asterisk ask see http://voip-info.org/ (~voip-info) and TheBook (~book) |
| 01:25.59 | leifmadsen | ~answers |
| 01:25.59 | infobot | [~answers] Asterisk Book: http://astbook.asteriskdocs.org & SIP w/ NAT: http://www.aocomputing.net/?p=3 & Wiki: http://www.voip-info.org & Docs: /path/to/src/asterisk/doc & 1.4 .vs. 1.6.0 .vs. 1.6.1: /path/to/src/asterisk/UPGRADE*.txt |
| 01:26.03 | ritztech | haha thats where im at |
| 01:26.13 | leifmadsen | read the chapters about dialplan |
| 01:26.26 | leifmadsen | voip-info is quite a bit out of date |
| 01:26.37 | leifmadsen | (but then again, so is the book, which is getting an update for 1.8) |
| 01:26.53 | leifmadsen | but it'll still talk about the 'h' extension as it is a fundamental aspect of dialplan programming |
| 01:27.06 | ritztech | i got taht asteriskbook like 2 years ago the oriely one |
| 01:27.37 | leifmadsen | heh |
| 01:27.42 | ritztech | so am i taking it the wrong way on that whole dialplan IT WORKS good but have to press pound |
| 01:27.43 | leifmadsen | guess you skipped a couple of chapters |
| 01:27.51 | ritztech | quite a few months haha |
| 01:28.11 | leifmadsen | <ritztech> so am i taking it the wrong way on that whole dialplan IT WORKS good but have to press pound <-- does not parse |
| 01:28.25 | ritztech | im like working on hour 70 since sunday |
| 01:28.34 | leifmadsen | anyways, enjoy the documentation |
| 01:28.36 | leifmadsen | I'm going to bed |
| 01:46.14 | *** join/#asterisk guilhermebr (~Guilherme@189.63.50.214) |
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| 02:02.48 | *** join/#asterisk kevcox (~kevin.cox@ip72-209-187-251.ks.ks.cox.net) |
| 02:03.17 | kevcox | Anyone have a good recommendation for a standard PCI board that will accomodate 4 analog lines? |
| 02:12.02 | *** part/#asterisk kevcox (~kevin.cox@ip72-209-187-251.ks.ks.cox.net) |
| 02:20.39 | ChannelZ | TDM400p |
| 02:21.51 | ChannelZ | or sorry - it's the 410 not 400 |
| 02:22.14 | ChannelZ | or get the 800P for future expansion |
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| 03:49.08 | ectospasm | Kevin`: Digium TDM410 + 4xX100M |
| 03:49.11 | ectospasm | oop |
| 03:49.14 | ectospasm | s |
| 03:49.21 | ectospasm | they've left |
| 03:49.29 | ectospasm | Kevin`: tab fail, sorry |
| 04:03.16 | *** join/#asterisk Sipster (~Sipster@modemcable045.5-200-24.mc.videotron.ca) |
| 04:43.00 | joobie | hmm.. anyone know how to improve echo issues? |
| 04:43.32 | joobie | sip phone connected to asterisk via g729.. asterisk then sends the call out pennytel as a sip provider using g729 and pennytel send the call to the 3rd party |
| 04:51.07 | *** join/#asterisk p3nguin_ (gpz5GvdFkf@cpe-0210-4bff-fe2b-9074.dhcp.a2infotech.com) |
| 04:51.37 | ectospasm | joobie: sounds like acoustic echo to me |
| 04:51.41 | ectospasm | get a better phone |
| 05:08.20 | p3nguin_ | Turn down the volume and see if that helps. |
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| 05:13.05 | joobie | ta |
| 05:13.08 | joobie | trying to get more info from the end user |
| 05:13.14 | joobie | i got a polycom 321 |
| 05:13.19 | joobie | mics are ok in them |
| 05:14.23 | ectospasm | joobie: are they using speaker phone when they get the echo? |
| 05:14.58 | ectospasm | Polycom 3xx series are about the cheapest they make, IIRC |
| 05:16.12 | p3nguin_ | It is usually that they need to turn down the volume. |
| 05:22.02 | Kevin` | joobie: have you tried echo cancelling on the asterisk to phone side? although iirc the pure software side of that is a bit limited |
| 05:22.21 | Kevin` | oh, sip phone you said, nevermind |
| 05:23.08 | Kevin` | joobie: do you have echo problems if you call a perfect phone like a softphone from it? |
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| 05:24.45 | *** part/#asterisk nny (~Scott@174.107.201.103) |
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| 06:29.59 | pwnguin | silly question, can i use asterisk to route video conferences? |
| 06:30.49 | pwnguin | im kinda guessing yes, since I think SIP allows it, but i haven't tried it so I thought i'd ask |
| 06:31.09 | ectospasm | I don't know about MeetMe or ConfBridge, but they probably do allow it |
| 06:31.24 | ectospasm | I know Asterisk can't record video, though, or at least couldn't the last I checked |
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| 07:52.50 | hyriand | how would you estimate the chance the deadlocks we're experiencing (http://pastebin.com/S870RSi3) could be related to using a tc400b card? |
| 07:53.27 | *** join/#asterisk schmidts (~schmidts@lmlo.sil.at) |
| 07:53.31 | schmidts | good morning all |
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| 08:07.34 | joobie | sorry Kevin` p3nguin_ ectospasm, just saw your responses but gotta run |
| 08:07.39 | joobie | will jump online from home |
| 08:07.40 | joobie | cheers |
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| 08:30.48 | UQlev | are there any options to use https authentication by users to add their IP to the white list to allow access to SIP/IAX? |
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| 09:26.06 | joobie | burp |
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| 10:08.58 | xheliox | Quiet in here. |
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| 10:16.46 | schmidts | its friday ;) |
| 10:18.50 | Ikarus | hrm, I'd almost order one http://www.dealextreme.com/details.dx/sku.27535 |
| 10:22.24 | schmidts | not exactly what i like but if it fits your needs, why not ;) |
| 10:23.21 | *** join/#asterisk ahowlader (~Adnan@180.234.49.161) |
| 10:24.21 | Chainsaw | schmidts: Cheap & nasty, just how he likes it. |
| 10:24.42 | schmidts | :D |
| 10:24.58 | ahowlader | hi tzafrir_laptop |
| 10:26.19 | tzafrir_laptop | ahowlader, hi, what's up? |
| 10:26.56 | ahowlader | tzafrir_laptop, nothing |
| 10:28.11 | ahowlader | tzafrir_laptop, i shall discuss with you later, first let me establish to serve test |
| 10:38.25 | Ikarus | schmidts: it's mostly just to see how far chinese engineering has gotten |
| 10:39.48 | Ikarus | though why Chainsaw just had to make that remark.... |
| 10:43.11 | *** join/#asterisk c0rnoTa (~c0rnoTa@109.188.48.74) |
| 10:43.29 | c0rnoTa | Hello everyone |
| 10:46.52 | c0rnoTa | I have one little question. If i recompile libpri, should i recompile dahdi-linux and asterisk? |
| 10:47.19 | c0rnoTa | and if i recompile dahdi-linux should I recompile asterisk? |
| 10:47.40 | c0rnoTa | recompilation wired with version upgrade |
| 10:54.21 | schmidts | but atleast they are more beautifull than snoms :D |
| 10:55.29 | *** join/#asterisk hrhrhr (~c1@unaffiliated/hrhrhr) |
| 10:55.31 | schmidts | c0rnota i would recompile all, lipri, dahdi and asterisk each of them use functions from the other, so if you change something on one point you should make sure it would work on the other end too |
| 11:02.20 | c0rnoTa | schmidts: thx |
| 11:04.39 | *** join/#asterisk Martinblr (~Miranda@61.12.17.170) |
| 11:04.50 | Martinblr | how to set dialplan restriction for each extensions |
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| 11:24.47 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 11:25.28 | henk | is there a ael2 reference somewhere? |
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| 11:52.49 | *** join/#asterisk mallchin (~mallchin@mail.dataproservices.co.uk) |
| 11:52.52 | mallchin | hi guys |
| 11:52.54 | mallchin | I'm having some trouble with voicemail, it goes to play a message to the caller stating they have gone through to voicemail then appears to hang -- the call sits there with no audio? |
| 11:53.17 | mallchin | it works from a sip phone, but fails on an incoming call via iax |
| 11:59.24 | mallchin | bbs |
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| 12:21.09 | *** join/#asterisk [TK]D-Fender (~chatzilla@216.191.106.163) |
| 12:53.49 | henk | is there a ael2 reference somewhere? |
| 12:55.07 | [TK]D-Fender | henk: checked your tarball? |
| 12:55.59 | henk | [TK]D-Fender: hm, no tarball, debian binary package... i'll check. |
| 12:58.02 | henk | [TK]D-Fender: doesn't seem so. are you implying that there is one in the tarball? if so, i'll just download it and extract it... do you happen to know (part of) the filename? |
| 12:59.10 | [TK]D-Fender | henk: there are teh SAMPLE CONFIGS, and a DOC folder. Maybe you should look a little |
| 13:01.05 | mallchin | back |
| 13:01.46 | *** join/#asterisk denysonique (~dennis@unaffiliated/dennisonicc) |
| 13:01.49 | mallchin | anyone got any ideas on my voicemail issue please -- why would the dialplan suddenly stop running and it sit doing nothing? |
| 13:02.48 | *** join/#asterisk ariel_ (~chatzilla@63.214.236.169) |
| 13:03.03 | henk | [TK]D-Fender: there is doc/tex/ael.tex which looks like a syntax reference. now i just need to make a pdf of it or something... thanks |
| 13:05.27 | henk | ah, it's read by asterisk.tex, so pdflatex asterisk.tex works. it throws errors but it works. and the resulting pdf has a complete chapter about AEL. |
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| 13:07.50 | russellb | Katty: who is this person you suggested I be friends with? |
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| 13:11.07 | schmidts | me ? :D :D |
| 13:12.01 | russellb | heh, are you on facebook? |
| 13:12.24 | schmidts | sure i am |
| 13:12.49 | russellb | k, then add me :-p ... http://www.facebook.com/russellbryant |
| 13:12.57 | schmidts | done ;) |
| 13:13.16 | c0rnoTa | Another question for you, asterisk guru: sometimes i'v got "channel.c: Got a FRAME_CONTROL (8) frame on channel DAHDI/.." What's mean FRAME_CONTROL (8) |
| 13:13.39 | russellb | control frames are call signaling information ... 8 is *looks it up...* |
| 13:13.55 | russellb | Congestion. |
| 13:14.25 | c0rnoTa | where you have looked ? :) i hadn't ffound it in source |
| 13:14.41 | russellb | i have the crystal ball that tells me |
| 13:14.50 | c0rnoTa | russellb: ^)) |
| 13:14.50 | russellb | include/asterisk/frame.h |
| 13:15.00 | russellb | enum ast_control_frame_type |
| 13:15.02 | c0rnoTa | thanks a lot |
| 13:15.07 | russellb | AST_CONGROL_CONGESTION = 8 |
| 13:15.09 | [TK]D-Fender | suspects it's really just a frosted light-bulb... |
| 13:15.51 | c0rnoTa | ^)) okay. |
| 13:15.52 | russellb | schmidts: i'm having trouble reading your facebook page, haha... |
| 13:16.32 | c0rnoTa | a have asked, because a there was call drops after FRAME_CONTROL (8) |
| 13:17.16 | c0rnoTa | but no "got hangup request, cause .." |
| 13:17.36 | c0rnoTa | so, will see. Thank you, guys |
| 13:19.43 | mallchin | :( |
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| 13:32.23 | Katty | peeks in |
| 13:33.22 | Katty | GOOD MORNING BEAUTS! |
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| 13:37.48 | c0rnoTa | Katty: morning |
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| 13:39.20 | *** part/#asterisk d00gster (~dt@2.90.43.20) |
| 13:41.30 | stix | Is a2billing the best and easiest solution if I want to bill users on my asterisk? |
| 13:41.50 | [TK]D-Fender | stix: MS Excel <- |
| 13:42.05 | c0rnoTa | One of my friends (Cache) makes attrafax (zoiper) patch for asterisk and want to do it more flexible and sensitive for dependants (like libtiff or libt30). Any suggestions or advices on this irc channel? Or we should try another one? :) |
| 13:42.17 | schmidts | russellb i wonder if you can read it, allmost its german :D |
| 13:42.22 | stix | [TK]D-Fender, and how is the export from asterisk to excel working? |
| 13:42.42 | [TK]D-Fender | stix: Excel reads CSV. The end |
| 13:43.00 | russellb | schmidts: that's what I mean. I can't read German. :-) |
| 13:43.47 | stix | [TK]D-Fender, but a2billing can create invoices automatically and probably a lot of other fancy features. It is free and Excel costs money. |
| 13:44.05 | [TK]D-Fender | stix: OpenOffice |
| 13:44.12 | c0rnoTa | ^)) |
| 13:44.40 | stix | [TK]D-Fender, also my users can sign into a2billing and have a look at their usage |
| 13:45.35 | *** join/#asterisk d00gster_ (~dt@86.51.184.26) |
| 13:46.30 | stix | let's just say that I don't wanna do manual work and therefore Excel and OO spreadsheet is not options. What would then be the best free tool? |
| 13:47.57 | drmessano | stix, you already sold us on A2Billing |
| 13:48.06 | drmessano | stix: Are you asking or telling? |
| 13:48.11 | [TK]D-Fender | stix: We'll take 2. |
| 13:48.12 | drmessano | s/telling/selling/ |
| 13:48.12 | *** join/#asterisk t_dot_zilla (~chatzilla@firewall-a.buf.ny.i-evolve.net) |
| 13:48.37 | stix | I havn't tried any and I don't know if any other than a2billing exists - therefore I am asking |
| 13:48.59 | drmessano | I would go with A2Billing, or code your own |
| 13:49.45 | stix | okay |
| 13:50.46 | drmessano | russellb coded an entire billing system for Asterisk, in an airport mens room, with his iPhone, in 3.4 minutes once |
| 13:50.50 | drmessano | It can't be THAT hard |
| 13:51.19 | russellb | it's true |
| 13:51.43 | stix | wow :) |
| 13:52.15 | file | never say "it can't be that hard" when billing is involved |
| 13:52.51 | [TK]D-Fender | Viagra <- |
| 13:52.56 | fauxalliance | prorated? |
| 13:53.10 | schmidts | it could be that easy. every call costs 10 $ even if there is no answer |
| 13:53.31 | fauxalliance | sounds like Deutsche Telekom |
| 13:53.53 | drmessano | schmidts: and even if you only thought about making the call |
| 13:55.19 | drmessano | "Man, I need to call my mom.. it's been weeks!" "Wait.. Damnit, there goes 10 bucks" |
| 13:55.34 | schmidts | sorry mum, its too expensive |
| 13:55.50 | drmessano | app_thought is pretty scary |
| 13:55.54 | fauxalliance | "Mrs. Messano, you have a collect call, do you accept the charges?" |
| 13:56.11 | schmidts | app_thought OMG *rofl* |
| 13:56.40 | drmessano | fauxalliance: My mother would probably keel over if I called her, keel over twice if I called collect |
| 13:57.14 | hyriand | anybody here going to sfd2010 in the hague tomorrow? |
| 13:57.23 | hyriand | (.nl) |
| 13:58.44 | *** join/#asterisk guilhermebr (~Guilherme@200.103.96.98) |
| 13:59.41 | drmessano | I always wanted to visit The Hague. I just like the name.. Sounds like the 3rd and final album from the eurotrash popsynth-industrial-death-metal band "Guess 'N Height" |
| 14:01.06 | [TK]D-Fender | drmessano: Bless you :p |
| 14:03.34 | drmessano | It's really a find album. Not as great as their non-denominational winter holiday album "Tan 'N Baum" |
| 14:03.39 | drmessano | fine* |
| 14:07.17 | *** join/#asterisk frigidzephyr (~rnewton@nat/digium/x-ytsuabqdjeohmfpc) |
| 14:07.58 | BarthezZ | hmm, I can't get a clue on hte following error... " chan_dahdi.c:12826 pri_dchannel: Ring requested on unconfigured channel 0/0 span 1" |
| 14:08.38 | *** join/#asterisk wcselby (~wcselby@pdpc/supporter/professional/wcselby) |
| 14:08.46 | wcselby | o/ |
| 14:09.35 | *** join/#asterisk rrb3942 (~rbullock@208.34.105.161) |
| 14:14.12 | *** join/#asterisk GVT (~Gabor@pop1.isgroup.sr) |
| 14:14.46 | [TK]D-Fender | BarthezZ: Means you got a call on a channel you didn't set up for * to use in chan_dahdi |
| 14:15.33 | BarthezZ | but I don't see a channel 0/0 if i do a cat /proc/dahdi/1 |
| 14:17.47 | *** join/#asterisk thecardsmith (~doug@pool-72-92-131-254.burl.east.myfairpoint.net) |
| 14:18.14 | *** join/#asterisk Tim_Toady (~moi@77.49.122.124.dsl.dyn.forthnet.gr) |
| 14:18.24 | *** join/#asterisk odenkos_ntbk (~odenkos@ip-212-081-019-170.static.nextra.sk) |
| 14:19.07 | BarthezZ | quite sure there shouldn't be a channel 0 |
| 14:21.19 | [TK]D-Fender | BarthezZ: Considered the possibility that it is reporting is ZERO based? |
| 14:22.18 | *** join/#asterisk bsaxon (~bsaxon@12.107.149.61) |
| 14:23.00 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
| 14:24.24 | *** join/#asterisk myster (~myster@74.198.148.18) |
| 14:24.40 | GVT | anyone know what the most popular phone systems are running asterisk? hardware solution wise |
| 14:25.31 | [TK]D-Fender | GVT: What fits your description of "phone system"? |
| 14:25.45 | [TK]D-Fender | GVT: Some might say "Intel" |
| 14:25.50 | GVT | [TK]D-Fender, the pbx server |
| 14:26.35 | GVT | GVT, I was thinking something more like switchvox, but that seems rather expensive in comparison to Aastra http://www.voiplink.com/Aastra_AastraLink_s/259.htm |
| 14:26.38 | [TK]D-Fender | GVT: GVT> anyone know what the most popular "PBX SERVER" are running asterisk? hardware solution wise? <- as substituted still doesn't clarify |
| 14:26.41 | *** join/#asterisk l2trace99 (~jr@74.118.41.1) |
| 14:27.04 | [TK]D-Fender | GVT: So bundled hardware w/ * as sold as a consumer product? |
| 14:27.20 | GVT | [TK]D-Fender, I suppose :) |
| 14:27.32 | [TK]D-Fender | GVT: Helps when you know what you're asking for... |
| 14:27.56 | [TK]D-Fender | GVT: Switchvos, Rhino's Ceros line, ScopServ, plenty of others |
| 14:28.03 | GVT | [TK]D-Fender, I am looking for bundled hardware, I'll need at least 6 fxo ports, and at least 1 fxs port |
| 14:30.02 | *** join/#asterisk tengulre (~tengulre@221.182.46.141) |
| 14:30.27 | tengulre | does the asterisk-1.8 support codecs g729? |
| 14:31.03 | c0rnoTa | I'v found General issue #18000 in mantis ^) |
| 14:31.04 | *** join/#asterisk Kobaz (~kobaz@its.kobaz.net) |
| 14:33.09 | kaldemar | tengulre: yes, with non-free licenses: http://www.digium.com/en/products/g729codec.php |
| 14:33.14 | stevo1664 | i have a problem with my otherwise awesome asterisknow setup. I have setup Queues and enabled announce hold time, when it tries to do this it says your estiamted wait time is 1 <hang up> can anyone advise me how I may diagnose this have run asterisk -r -vvvvv and nothing obvious there (obvious to me anyway), I have checked the queue-minutes file is there.. Im running 2.7.0.8 with freepbx, but no one on their IRC is helping me. Th |
| 14:36.49 | *** join/#asterisk p3nguin (gpz5GvdFkf@cpe-0210-4bff-fe2b-9074.dhcp.a2infotech.com) |
| 14:37.52 | *** join/#asterisk trixboxer (~Trixboxer@office.supportdepartment.net) |
| 14:38.16 | [TK]D-Fender | tengulre: Not yet |
| 14:38.43 | [TK]D-Fender | tengulre: for the actual codec itself. Passthrough yes, but I see no binary |
| 14:39.12 | [TK]D-Fender | stevo1664: PASTEBIN the failed call. |
| 14:39.32 | *** join/#asterisk coppice (~chatzilla@112.203.17.210.dyn.pacific.net.hk) |
| 14:40.58 | Kobaz | hmm |
| 14:41.06 | c0rnoTa | People. Another question :) what can you say about 'span 2 got hangup request, cause 102'? Where i can see, what timer issue i have? |
| 14:41.10 | Kobaz | is there a way to run dialplan on channel 'making progress' |
| 14:46.35 | GlobeTrotterz | Hi Guys, AgentCallbackLogin is now depriciated in 1.6. Anyone know how to log in and out ouf queues via the dialplan? |
| 14:47.27 | stevo1664 | will do thanks |
| 14:47.38 | GlobeTrotterz | I am usung the extension.ael and the supplied instruction in queues-with-callback-members.txt . I have one problem |
| 14:48.40 | GlobeTrotterz | ael uses gosun istead on macro, so the MACRO_EXTEN does not pass the value. |
| 14:49.28 | [TK]D-Fender | ~pb |
| 14:49.28 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude. |
| 14:49.31 | [TK]D-Fender | stevo1664: ^^^^^^^^ |
| 14:50.00 | [TK]D-Fender | GlobeTrotterz: then pass the exten as a paramteter in yoru Gosub |
| 14:50.25 | [TK]D-Fender | [10:41]<Kobaz>is there a way to run dialplan on channel 'making progress' <- no |
| 14:50.51 | [TK]D-Fender | Kobaz: Short of a lot of direct new code. Very messy of course |
| 14:52.19 | Kobaz | hmm |
| 14:52.41 | stevo1664 | OK i have pastebinned my queue hold time problem at http://pastebin.com/qggBVtVT it is line 619 that is the last file played before hangup |
| 14:52.48 | Kobaz | well... it's either i modify asterisk, or adtran modifies their code, or comes up with something magical |
| 14:52.53 | Katty | wts headache. 2 dollah. |
| 14:55.31 | [TK]D-Fender | stevo1664: Perhaps your queue kicks out callers if nobody is taking calls. I see ti distribute to several |
| 14:55.39 | *** join/#asterisk aidinb (~Aidin@unaffiliated/aidinb) |
| 14:56.49 | [TK]D-Fender | stevo1664: And actually I don't see your actual call's end in there |
| 14:57.29 | stevo1664 | i have two calls in progress |
| 14:57.31 | [TK]D-Fender | stevo1664: the one ending in "134" is your inbound caller. I do not see THEIR hangup |
| 14:57.43 | stevo1664 | 134 is the first call |
| 14:58.03 | *** join/#asterisk ruben23 (~RLACUMBA@121.97.111.142) |
| 14:58.13 | stevo1664 | 136 is the one that gets booted, it will only play hold time if there is at least two people in the queue |
| 14:58.17 | [TK]D-Fender | Ah, looking at now |
| 14:58.22 | stevo1664 | thanks |
| 14:59.31 | *** join/#asterisk bsaxon (~bsaxon@12.107.149.61) |
| 14:59.48 | GlobeTrotterz | I need to capture and store the agents Caller id when the clerk first signs, and store it into a variable, ie; ${Agent1} .. then when agent 2 logs in store the caller id in a varialbe ie. ${AGENT2}.. then the dialplan will call the Agent with this... 7799 => &callagent(SIP/${Agent1}) macro callagent(device) |
| 15:00.16 | ruben23 | hi guys any opensource video conferencing..? |
| 15:00.30 | [TK]D-Fender | stevo1664: -- <SIP/ccm-county-02-00000136> Playing 'digits/1.ulaw' (language 'en') == Spawn extension (from-internal, 50005, 9) exited non-zero on 'SIP/ccm-county-02-00000136' <- It is looking like their end killed the call. |
| 15:01.18 | [TK]D-Fender | GlobeTrotterz: So change your Gosub to ALSO pass the 7799 |
| 15:01.37 | stevo1664 | right, it always happens after 'your estimated wait time is 1 |
| 15:02.43 | stevo1664 | so I think it must be triggered by asterisk somehow. I am running with freepbx so the config files are a bit weird, so I thought I might try to build a normal asterisk server, I have several queue config files with Freepbx |
| 15:03.21 | GlobeTrotterz | Thanks D-Fender: Can you give me a tip as how to do this please. I am have basic programming skills |
| 15:03.22 | [TK]D-Fender | stevo1664: I don't recall any issues about queues like that.. I would think your client has an issue. Was that YOU calling in yourself? |
| 15:03.34 | GlobeTrotterz | less than basic :) |
| 15:03.36 | [TK]D-Fender | GlobeTrotterz: See how you pass that 1st parm? Pass the EXTEN as well |
| 15:03.54 | GlobeTrotterz | ah ok |
| 15:04.21 | GlobeTrotterz | i got you,, thanks alot D-Fender im gonna try to figure it out,, iwilllet you know how it goes |
| 15:04.25 | stevo1664 | yes, my phone is on cisco call manager, dialing the queue on asterisk, the queue member/agent is a hunt group on call manager |
| 15:04.51 | stevo1664 | i could try to call in from a sip client registered to asterisk perhaps? |
| 15:05.35 | *** join/#asterisk bsaxon (~bsaxon@12.107.149.61) |
| 15:05.41 | *** join/#asterisk aidinb (~Aidin@unaffiliated/aidinb) |
| 15:07.18 | mallchin | anyone help with my voicemail stopping unexpectedly? it works phone -> sip -> asterisk voice system2 -> voicemail but fails phone -> pri -> asterisk voice system 1 -> iax -> asterisk voice system 2 -> voicemail |
| 15:08.44 | [TK]D-Fender | stevo1664: Good idea to test. Confirm if it is isolated to one calling source |
| 15:09.40 | *** join/#asterisk bsaxon (~bsaxon@12.107.149.61) |
| 15:12.28 | mallchin | have I got voicE? |
| 15:15.25 | *** join/#asterisk silvestre_id (~silvestre@200-204-158-49.dsl.telesp.net.br) |
| 15:17.50 | Martinblr | i have a isdn bri router and connected the output to asterisk via fxo port, when the lines are busy after 30 minutes the call is automatically disconnected, becuase of a busy tone is detected.... |
| 15:21.28 | thecardsmith | i've got my entire asterisk built from yum repos in centos... and i want to run patlooptest, but, for the life of me i can't figure out what package provides it, does anyone know where it is? thanks :) |
| 15:23.52 | *** join/#asterisk freckle (~viperdude@viperdudeuk.broker.freenet6.net) |
| 15:24.14 | *** join/#asterisk t_dot_zilla (~chatzilla@firewall-a.buf.ny.i-evolve.net) |
| 15:24.37 | t_dot_zilla | what does 'Diversion' mean in a SIP Message Header |
| 15:25.12 | *** join/#asterisk T3CHKOMMIE (~juleshoeh@207.86.14.138) |
| 15:25.22 | T3CHKOMMIE | hello everyone! |
| 15:26.58 | T3CHKOMMIE | i was wondering if anyone has experience setting up a simple asterisk box and tieing it directly to a google voice account. i have been reading alot of tutorials but they all reaquire a gizmo 5 account or a sipgate account. i was wondering since google can now my calls from gmail is there not a way to tie in the server driectly to this service now? |
| 15:30.50 | *** join/#asterisk darkskiez_ (~dz@195-11-205-216.suip.mezzonet.net) |
| 15:33.15 | *** join/#asterisk BANSAL (~bansal@117.199.114.171) |
| 15:33.21 | bougyman | i don't think asterisk has a channel driver for it, yet. |
| 15:33.40 | bougyman | FreeSWITCH does, you could run an fs instance as a proxy for it. |
| 15:34.22 | T3CHKOMMIE | bougyman, i'm a pbx noob still. is freeswitch like asterisk? |
| 15:35.13 | bougyman | it's another open source voice (and some video) engine. |
| 15:35.18 | bougyman | so yes, in that way it's like it. |
| 15:35.33 | russellb | Asterisk has had a channel driver for google talk for many years now |
| 15:35.40 | Qwell | google voice |
| 15:35.42 | bougyman | russellb: does it support voice calls? |
| 15:35.47 | russellb | oh sorry, didn't read clearly |
| 15:35.48 | russellb | bougyman: yes |
| 15:36.02 | bougyman | russellb: oh neat, last week this convo was had and someone said it did not. |
| 15:36.12 | russellb | google*talk* though |
| 15:36.17 | bougyman | no, google voice |
| 15:36.17 | Qwell | it's been there since 1.4.. |
| 15:36.20 | russellb | which is something different than the google voice gmail thing |
| 15:36.26 | T3CHKOMMIE | im just trying to set up somthing in my appartment to make and recieve calls, i have some sip phones laying arround and i dont have a "real" phoneline to my hosue, i just wanna get something landline like. any ideas? |
| 15:36.28 | russellb | right, i dunno about that :-) |
| 15:36.31 | bougyman | i think he's talking about the google voice, isn't he? |
| 15:36.37 | bougyman | that's the channel driver i'm talkin about |
| 15:36.45 | russellb | probably, sometimes i jump into the middle of stuff and don't read backlog ... |
| 15:37.00 | leifmadsen | same here :) |
| 15:37.09 | thecardsmith | i've made my * take a call from the PSTN and then bridge to make a free outbound LD call to the PSTN with GV |
| 15:37.28 | thecardsmith | just toying with it, the GV just seemed like a sweet telephony toy |
| 15:37.35 | Qwell | it is just google talk apparently |
| 15:37.36 | bougyman | anyway, it could even run on the same box, T3CHKOMMIE, i know people doing it (asterisk as the pbx with a stripped down FreeSWITCH just for the google voice stuff) |
| 15:38.47 | T3CHKOMMIE | bougyman, so i understand corretly, i get a * box running with freeswitch (for the GV drivers) then i connect * box to interenet and gv account and i can make sip calls from home throught gv? |
| 15:39.08 | Qwell | it's not SIP. |
| 15:39.08 | bougyman | you'd connect * to fs via standard sip |
| 15:39.17 | bougyman | and fs to GV with your uname/pass |
| 15:39.41 | bougyman | it'd just hand everything off to asterisk over sip. |
| 15:39.43 | bougyman | just a dumb proxy |
| 15:39.51 | Qwell | T3CHKOMMIE: Do you want google voice, or the gmail voice thing? |
| 15:39.58 | *** join/#asterisk Bubu (~chatzilla@p4FDDE48E.dip.t-dialin.net) |
| 15:40.58 | T3CHKOMMIE | i just wanna be able to make calls with gv |
| 15:40.59 | T3CHKOMMIE | ? |
| 15:41.13 | T3CHKOMMIE | and get my home sip phones to ring when somone calls my gv number. |
| 15:41.24 | T3CHKOMMIE | looking to make a landline replacement (comcast is gay) |
| 15:41.44 | bougyman | http://freeswitch.org/node/281 (receiving) and http://freeswitch.org/node/280 (originating) -> http://wiki.freeswitch.org/wiki/Google_Voice docs on how to do it. |
| 15:42.16 | T3CHKOMMIE | bougyman, awsome, that makes more sence. do i need to have two server boxes? or can i run * and FS on the same system? |
| 15:42.29 | Qwell | gtalk != google voice... |
| 15:42.35 | bougyman | yes, you can run them both on the same system. |
| 15:42.42 | Qwell | that wiki details how to use gtalk. which Asterisk has supported for a very long time. |
| 15:42.44 | bougyman | you simply have to change some default ports. |
| 15:42.54 | bougyman | Qwell: and google voice. |
| 15:43.04 | Qwell | it says Google Voice, but no. It's gtalk. |
| 15:43.22 | bougyman | it's my google voice account |
| 15:43.33 | bougyman | i guess they use the gtalk protocol on that. |
| 15:43.46 | bougyman | so your gtalk driver should support it, no? |
| 15:44.05 | Qwell | yes. |
| 15:44.15 | Qwell | all you have to do is check the "Forward to gtalk" option in google voice. |
| 15:45.09 | T3CHKOMMIE | must be missing something in the difference between gtalk and gv.... |
| 15:45.29 | *** join/#asterisk b11d` (~no@234-200-29-134.hcc.mnscu.edu) |
| 15:45.33 | bougyman | google voice allows you to send your calls to gtalk |
| 15:45.47 | bougyman | that's what you have to do to use the dingaling (google talk) driver explained above. |
| 15:46.09 | *** join/#asterisk sol (~sol@unaffiliated/sol) |
| 15:46.10 | T3CHKOMMIE | ok, so when i make a call from "gmail" am i using gv or gtalk? |
| 15:46.15 | Qwell | gtalk. |
| 15:46.28 | bougyman | and if it's gtalk asterisk should be able to do it with it's gtalk channel driver? |
| 15:46.33 | Qwell | yes |
| 15:46.35 | T3CHKOMMIE | but when i get a vm from my gv number thats gv? |
| 15:46.39 | bougyman | so no need for two systems, T3CHKOMMIE |
| 15:47.45 | T3CHKOMMIE | all the tutorials i have been reading from NERD show alot of different things to set up. it seems like it wold be easy or "simple" to connect * to gv if both can use sip. but im not telephony expert. |
| 15:47.53 | Qwell | T3CHKOMMIE: Don't read his garbage. |
| 15:47.56 | Qwell | it's complete crap. |
| 15:48.02 | Qwell | ignore everything he says - he's wrong. |
| 15:48.13 | Qwell | where he = nerdvittles guy |
| 15:48.16 | T3CHKOMMIE | ah, ok. it seemed like alot of stuff to do. |
| 15:48.50 | T3CHKOMMIE | so, in a nut shell if i want to get my SIP landline working throught Google services , i can just used * and thats it? |
| 15:48.55 | Qwell | NO |
| 15:48.57 | Qwell | It's not SIP. |
| 15:49.17 | Qwell | Please stop confusing protocols... It's very important that you get it right. |
| 15:49.28 | T3CHKOMMIE | hu, i was pretty sure i read that * can do sip. |
| 15:49.33 | Qwell | of course it can |
| 15:49.38 | T3CHKOMMIE | ok.... |
| 15:49.42 | Qwell | but you went from talking about jingle to SIP.. |
| 15:49.52 | T3CHKOMMIE | jingle? |
| 15:49.58 | Qwell | gtalk |
| 15:50.25 | T3CHKOMMIE | oh, well crap, i was under the impresseing that gtalk used SIP to make its calls. |
| 15:50.38 | T3CHKOMMIE | i think i read that from NERD? |
| 15:50.45 | Qwell | <Qwell> ignore everything he says - he's wrong. |
| 15:50.52 | *** join/#asterisk dforbu (~dforbu@216.54.131.253) |
| 15:51.19 | T3CHKOMMIE | ok, so then phone => sip => asterisk box => jingle => gtalk? |
| 15:51.29 | Qwell | pretty much |
| 15:51.36 | freckle | hi, is there a way to inject audio into a one channel of a call in progress? |
| 15:51.42 | T3CHKOMMIE | ok, thanks, im glad i got atleast that figured out. |
| 15:52.13 | T3CHKOMMIE | so then my * box should be able to jingle to gtalk. and that is how i can make in/outbound calls? |
| 15:52.20 | Qwell | sure |
| 15:52.28 | Qwell | or you could just use a simple ITSP, of which there are hundreds. |
| 15:52.34 | Qwell | ~itsp |
| 15:52.35 | infobot | [~itsp] An ITSP is an Internet Telephony Service Provider (or VoIP telephone company). They allow you to either SEND calls to the PSTN (this is 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. |
| 15:53.07 | T3CHKOMMIE | AH! now, is gv not a itsp? |
| 15:53.13 | Qwell | No it is not. |
| 15:53.25 | T3CHKOMMIE | hu. so how is it that i have DID with them? |
| 15:53.40 | Qwell | They provide DIDs. That doesn't make them an ITSP. |
| 15:53.42 | *** part/#asterisk c0rnoTa (~c0rnoTa@109.188.48.74) |
| 15:53.47 | T3CHKOMMIE | ok |
| 15:53.52 | Qwell | at least, not an open one |
| 15:54.15 | T3CHKOMMIE | so an itsp would be like... sipgate or gizmo5? |
| 15:54.22 | Qwell | ~itsplist-us |
| 15:54.22 | infobot | Here are some popular ITSPs (USA) starting with the more respected ones: http://www.teliax.com , http://www.voicepulse.com/connect/ , http://www.broadvoice.com , http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net , http://voip.ms and http://flowroute.com |
| 15:54.23 | Qwell | but yes |
| 15:54.35 | Qwell | of course, gizmo5 is Google now.. |
| 15:54.42 | freckle | Qwell: since when has being open been a requirement to be a ITSP? |
| 15:55.00 | Qwell | freckle: kinda pointless if you can't send calls through them |
| 15:55.04 | T3CHKOMMIE | ok. and its possible for me to set up a sip based landline at home with out an itsp if i jingle to gtalk correct? |
| 15:55.18 | freckle | Qwell: you can |
| 15:55.24 | Qwell | T3CHKOMMIE: you're confusing terms again, but syre |
| 15:55.27 | Qwell | sure* |
| 15:55.33 | Qwell | freckle: as of what, 2 weeks ago? |
| 15:55.38 | freckle | maybe not all destinations, but it is still internet telephony |
| 15:55.55 | freckle | even if only to other providers it is internet telephony |
| 15:55.56 | T3CHKOMMIE | Qwell, what im i confusing. im not trying to be "smart" i just really want to understand this :( |
| 15:56.09 | T3CHKOMMIE | thanks for the help btw. |
| 15:56.09 | freckle | /s/providers/subscribers |
| 15:56.09 | Qwell | T3CHKOMMIE: "sip based landline" makes no sense whatsoever |
| 15:56.19 | T3CHKOMMIE | ah, ok |
| 15:56.28 | dforbu | so.. back in the old days, on my * 0.9 system, I had voicemail with options in mysql table, one of the options was outgoing=yes - which made * play the unavail.wav and hang up. that option has a different syntax in 1.6? |
| 15:56.45 | freckle | Qwell: i only know Gtalk can call other Gtalk... thats internet telephony |
| 15:56.45 | T3CHKOMMIE | i want to have sip phones in my house, running over cat5... with just interent to my house.... so thats what i was implying :/ |
| 15:56.51 | Qwell | freckle: gv != gtalk |
| 15:57.13 | freckle | Qwell: sure, I thought it was Gtalk you were talking about |
| 15:58.55 | freckle | hi, is there a way to inject audio into a one channel of a call in progress? |
| 15:59.42 | T3CHKOMMIE | Qwell, i seem to have this habbit of starting a project and not completly understand something, and wasting 50 hrs of my life realizing i was trying to do something that wasnt possible. so im just want to make sure i understand it well before i start. for example. i spent 3 months not knowing what "packet injection" was and trying to crak my WEP... boy was that frustrating and very unsuccessful. |
| 16:03.09 | T3CHKOMMIE | so im going to set up my * box to connect to my gtalk account and then i should be able to make out/inbound calls. right? |
| 16:03.27 | Qwell | sure |
| 16:03.33 | Qwell | but I still say you should just use an ITSP. |
| 16:03.34 | p3nguin | Once you figure out how, let me know. |
| 16:03.58 | *** join/#asterisk VanClone (~whatever@kernel-panic/vandyke) |
| 16:04.15 | VanClone | hello voipers |
| 16:04.15 | T3CHKOMMIE | ya, i understand its easier, but i dont want to deal with that, i would rather learn to directly link the two. |
| 16:04.48 | T3CHKOMMIE | Qwell, thanks for the help and clarification, now that i know how its supposed to work maybe i will be able to google away from NERDs suggestions! |
| 16:05.02 | Qwell | sighs |
| 16:05.16 | VanClone | question... does asterisk store logs for what people dialed during an IVR? |
| 16:05.40 | Qwell | VanClone: I believe CEL covers that. |
| 16:05.47 | Qwell | similar to CDR, but not quite |
| 16:06.02 | VanClone | separate module, I assume |
| 16:06.08 | drmessano | Is Gtalk+Gvoice even working in * right now? |
| 16:06.10 | VanClone | I'm using trixbox CE atm |
| 16:06.23 | Qwell | drmessano: gvoice isn't relevant. it's just gtalk |
| 16:07.12 | Qwell | VanClone: yeah, you're screwed then. |
| 16:07.26 | drmessano | Qwell: Yes, it is.. GVoice is working with the chat applet, and supposedly to Gtalk, but from I understood there is another issue with calls hitting Asterisk.. Maybe an API change or something with Jingle |
| 16:07.50 | Qwell | drmessano: The gtalk client is infinitely backwards compatible. |
| 16:08.00 | Qwell | So, no API change Google makes should ever break chan_gtalk |
| 16:08.16 | Qwell | or, rather... |
| 16:08.29 | Qwell | the gtalk servers are infinitely backwards compatible to all versions of the gtalk client. |
| 16:08.47 | VanClone | Qwell: the problem I'm having is that for a certain extension, someone had to try three times before getting directed to that extension, and got a message saying the extension didn't exist in the two failures |
| 16:08.56 | Qwell | VanClone: #trixbox |
| 16:10.01 | VanClone | Qwell: thanks |
| 16:10.41 | drmessano | Well, let me put it this way.. I have calls routed to Gtalk. I have had Gtalk in * for some time. Calls never hit Asterisk via Gtalk.. I know they're being routed to Gtalk because Pidgin tried to answer them, but they apparently don't support google's variant of XMPP/Voice, so the audio doesn't set up properly. |
| 16:10.59 | drmessano | I also know someone else had the SAME issue a few days ago |
| 16:11.25 | b11d` | someone mentioned that gtalk was broken in asterisk atm.. |
| 16:11.31 | b11d` | maybe that was in -stable or something |
| 16:12.01 | drmessano | Well, maybe that's it |
| 16:12.59 | drmessano | I haven't tried making any gtalk calls with * in some time, so I guess it could be completely broken and not just some unimplemented change |
| 16:13.16 | drmessano | hmm |
| 16:16.10 | drmessano | I wonder if Gtalk was kneecapped in Asterisk so Gvoice+gtalk wouldn't undermine the Skype compatibility. Gosh, that's a pretty good conspiracy, even for me. |
| 16:16.11 | *** join/#asterisk NeonLevel (~NeonLevel@189.169.140.136) |
| 16:16.17 | drmessano | Now to get it down to 130 characters |
| 16:20.05 | b11d` | yeah that's a good conspiracy.. probably all done by the CIA who operates Skype |
| 16:20.36 | b11d` | i once worked on a prog!*@U(*@U(A NO CARRIER |
| 16:22.51 | drmessano | The CIA operates Skype which they use for branch office trunking to their ECHELON monitoring locations |
| 16:23.17 | Katty | what in the snickerdoodles is going on in here. |
| 16:23.31 | carrar | waiting to be EATIN!!! |
| 16:23.33 | Katty | i leave for 5 minutes and everyone's talking about the CIA?! |
| 16:23.44 | drmessano | Which happen to be the closets AT&T's C.O.s and the basements of all public high schools |
| 16:23.45 | carrar | The snickerdoodles ARE NOW DIAMONDS!!! |
| 16:23.46 | Katty | come on now, don't talk behind my back |
| 16:23.55 | Qwell | Katty: CIA came in and arrested everybody while you were gone :( |
| 16:24.05 | Katty | Qwell: i am the CIA. |
| 16:24.11 | Qwell | exactly |
| 16:24.24 | drmessano | A lot of people don't realize the CIA uses high schools for recruiting and free office space |
| 16:24.39 | Katty | they have some interesting tests. |
| 16:24.46 | Katty | and that's all i'm allowed to say on the topic (= |
| 16:25.38 | Letoric | hello folks. Is it possible to assign multiple IP addresses to a single SIP provider? |
| 16:25.39 | carrar | hands everyone a National Security Letter and orders everyone to not talk about it |
| 16:25.52 | carrar | heh |
| 16:25.52 | Letoric | IE, our backup SIP provider has 3 IP's they want us to use |
| 16:26.17 | Katty | anywho! it is time for luncheries. |
| 16:26.22 | carrar | Whats good |
| 16:26.23 | Katty | poofs. |
| 16:26.24 | carrar | err |
| 16:26.32 | carrar | Thats good letoric |
| 16:27.48 | carrar | round robin between them all |
| 16:27.48 | carrar | MCI is the same way exect they have 4 IP's |
| 16:27.54 | Letoric | carrar: How? |
| 16:27.55 | p3nguin | Hmm, Internet Explorer and SIP? |
| 16:28.06 | carrar | Use Asterisk from Source! |
| 16:28.09 | carrar | and get rolling! |
| 16:28.26 | carrar | Write a very simple AGI |
| 16:31.17 | carrar | That just passes the call to the next IP thats in order or used least or however you want to write it |
| 16:31.17 | carrar | I myself put the IP's in a db so I can change them at anytime and look at hit stats on them |
| 16:32.17 | carrar | You can too!! |
| 16:32.17 | carrar | With the power of Asterisk from Source |
| 16:32.17 | carrar | Nothing is impossible! |
| 16:34.15 | ccomp5950 | BitchSlapOverSIP() |
| 16:34.31 | ccomp5950 | sorry, been at it for the last few years, seems impossible =( |
| 16:34.52 | carrar | Thats impossible!! |
| 16:42.11 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 16:42.16 | tzanger | Katty: I didn't think girls did that |
| 16:42.56 | Martinblr | I have a Netmod ISDN BRI router and from the router I have connected the analog port in Asterisk via FXO card. When the lines are established, after 31 minutes the call is automatically disconnected.While checking the log it shows as busy tone is detected |
| 16:44.42 | tzanger | wtf |
| 16:45.17 | drmessano | I think it takes 31 minutes for hardware to realize BRI is extinct |
| 16:45.28 | tzanger | how do vlans isolate anything when the computer is plugged into the phone and the phone into the wall? even without vlans, then phone's MAC won't see the computer's traffic since the phone's switch will not broadcast the computer's packets to the phone's internal switch port |
| 16:45.36 | drmessano | "OMG, why didn't you tell me someone BRI'ed in this?" |
| 16:47.19 | drmessano | tzanger: The IF on the LAN port of the phone is in front of the internal IF of the phone and the PC port |
| 16:47.59 | tzanger | drmessano: the phone would have a 3 port switch. 1 to the upstream (wall), one to the phone internal ethernet and one to the computer ethernet jack |
| 16:48.27 | fauxalliance | http://archive.networknewz.com/networknewz-10-20030725IntroductiontoVLANs.html |
| 16:48.48 | fauxalliance | I'm always amazed how people get themselves all tied in a knot when the topic of VLANs comes up... |
| 16:48.51 | drmessano | tzanger, correct |
| 16:48.55 | tzanger | I mean the only thing I could see woudl be that the phoen wouldn't see broadcast traffic for the computer's vlan, but come on, that's almost zero these days |
| 16:49.17 | VanClone | meh |
| 16:49.19 | drmessano | tzanger, it IS 3 ports |
| 16:49.23 | tzanger | fauxalliance: I'm not tied in a knot, I'm questioning the utility of going to all the trouble to set up vlans when you're plugging the computer into the phone |
| 16:49.32 | VanClone | looks like not even CEL would help, which is only available on * 1.8 |
| 16:49.41 | fauxalliance | tzanger, SOP |
| 16:49.59 | drmessano | Phone IF <----> LAN <---> PC Port |
| 16:49.59 | carrar | tzanger, the phone tags it's traffic |
| 16:50.04 | carrar | the pc is untagged |
| 16:50.26 | p3nguin | Yes, the phone tags it is traffic. |
| 16:50.41 | drmessano | lol |
| 16:51.06 | carrar | if setup that way that is |
| 16:51.25 | tzanger | p3nguin: ok, what is the advantage of this? |
| 16:51.33 | fauxalliance | isolation |
| 16:51.35 | tzanger | so the phone doesn't see SMB broadcast traffic? I don't get it |
| 16:51.38 | carrar | traffic segregation |
| 16:51.55 | tzanger | carrar: yes, but switches already do that |
| 16:52.12 | carrar | yes? |
| 16:52.16 | fauxalliance | tzanger, not quite as abstractly as a switch. |
| 16:52.34 | fauxalliance | switches are almost as dumb as hubs. |
| 16:53.02 | fauxalliance | and hubs are about as smart as a pack of paperclips. |
| 16:53.23 | fauxalliance | vlan tagging fills the gap between switch and router |
| 16:53.37 | fauxalliance | router = smart |
| 16:53.38 | tzanger | fauxalliance: no |
| 16:53.44 | tzanger | fauxalliance: you have your core switch |
| 16:53.50 | tzanger | you have your phoen switch |
| 16:53.53 | fauxalliance | maybe I'm too abstract. |
| 16:53.54 | tzanger | the phone switch is only receiving 3 things |
| 16:53.59 | Martinblr | drmessano: if we put some call delay in asterisk that will help... |
| 16:54.07 | tzanger | 1) broadcast traffic. 2) traffic for the computer's MAC and 3) traffic for the phone's MAC |
| 16:54.28 | tzanger | so again, other than getting rid of broadcast traffic which is already next to nil, why go to the trouble of installing a VLAN? |
| 16:54.29 | carrar | broadcast traffic stays within a vlan |
| 16:55.02 | tzanger | carrar: yes but I'm not really seeing the advantage of preventing a phone from seeing the oh... 5 packets a minute of broadcast traffic that any modern network has? |
| 16:55.02 | drmessano | Martinblr: Make Asterisk wait about 45 mins before it realizes there's a BRI connected to it? Brilliant |
| 16:55.38 | carrar | then you probably have no need for VLANS :) |
| 16:55.46 | fauxalliance | tzanger, scale up.... quickly... see what happens... |
| 16:55.47 | drmessano | Maybe this will all be clear when we switch to IPv6 |
| 16:55.55 | tzanger | VLANs make great sense when you aren't plugging your computers into your phones. you set up the VLANs on the core switches and then the phone traffic stays on the phone network and the computer traffic stays on the computer network |
| 16:56.02 | carrar | hahah drmessano |
| 16:56.04 | fauxalliance | drmessano, you haven't yet? |
| 16:56.14 | Martinblr | drmessano: is that really help...:) |
| 16:56.34 | tzanger | fauxalliance: sure, but again a small office (<50ish people) has no real need for VLANs if the computers are already being plugged into thep hones |
| 16:56.36 | carrar | tzanger, thats how it works when you plug the PC into the phone also |
| 16:56.37 | drmessano | I've tried to explain this to you dumbasses time and time again, but maybe I will sit back and just wait til you see IPv6 and how it fixes all these problems.. ALL of them |
| 16:56.39 | Martinblr | drmessano: but if i answer call after 45 minutes nobody will be on the call |
| 16:56.42 | tzanger | that's what I'm trying to understand |
| 16:56.46 | carrar | if you configure it correctly |
| 16:56.50 | tzanger | drmessano: :-) |
| 16:57.04 | tzanger | carrar: ok, but again |
| 16:57.17 | drmessano | IPv6 promises to make SIP secure, HTTP secure, and speed up GOPHER |
| 16:57.17 | tzanger | the traffic for BOTH computer+phone is already hitting the physical switch in the phone |
| 16:57.18 | ccomp5950 | I'm waiting on the first service pack for IPv6, I hate being an early adopter =( |
| 16:57.28 | b11d` | hhaa |
| 16:57.30 | b11d` | ipv6sp1 |
| 16:57.31 | carrar | one is tagged and one isn't |
| 16:57.36 | tzanger | you're not saving any bandwidth. you're not saving any switching capacity becuase the little switch already has ot handle the traffic for both |
| 16:57.39 | carrar | (at the phone) |
| 16:57.51 | fauxalliance | tzanger, then for you... non issue |
| 16:57.52 | tzanger | it's not saving you a damn thing nor making things any easier on the (oftentimes) crappy switch in the phone |
| 16:57.54 | carrar | When it hits the switch the swich tags them |
| 16:58.03 | carrar | well tags the pc |
| 16:58.07 | drmessano | ccomp5950: I am using TCP/IP Automatic Updates. I need to run a net dist-upgrade and go to ipv6 I guess |
| 16:58.21 | *** join/#asterisk Deeewayne (~dwayne@75-150-14-49-Atlanta.hfc.comcastbusiness.net) |
| 16:58.21 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 16:58.44 | fauxalliance | drmessano, nope, you just plug in the IP/4 to 6 adaper... just like HDTV |
| 16:58.49 | tzanger | carrar: Listen, I understand switching architecture and I understand VLANs. I'm not sure you understand my question. You're not isolating very much since ALL traffic for phone+computer has to hit the 3-port switch in the phone |
| 16:59.09 | drmessano | fauxalliance: I am still waiting for my IPv6 coupon from Microsoft for Windows 2000 |
| 16:59.11 | tzanger | whereas the main point of VLAN is isolating traffic |
| 16:59.13 | carrar | you are isolate by vlan tags |
| 16:59.14 | tzanger | you're not isolating very much |
| 16:59.22 | tzanger | drmessano: hahaha |
| 16:59.28 | carrar | because the phones traffic is tagged |
| 16:59.31 | ccomp5950 | sounds right, my TCP/IP implementation is still stuck on IPv3.5beta4 Release Candidate 2. I sit in my office wondering why no one emails me =( |
| 16:59.34 | carrar | the pc is not |
| 16:59.54 | fauxalliance | ccomp5950, ARP was really fishy in that release. |
| 17:00.00 | *** join/#asterisk Tim_Toady (~moi@77.49.122.124.dsl.dyn.forthnet.gr) |
| 17:00.03 | ccomp5950 | I have to telnet into IRC, do you know how many PONG requests IRC servers send? |
| 17:00.03 | Nugget | telnet is eeeeeeevil! |
| 17:00.08 | tzanger | carrar: but the point is that the 3 port switch STILL sees traffic for both |
| 17:00.09 | ccomp5950 | telnet |
| 17:00.09 | fauxalliance | SSH is secure. |
| 17:00.10 | fauxalliance | !@ |
| 17:00.25 | fauxalliance | ahem, reverse netcat sometin sometin |
| 17:00.34 | carrar | If you think it |
| 17:00.42 | carrar | If you think its a switch try moving the plugs around |
| 17:01.08 | drmessano | ccomp5950: I ran the IPv3.11 with the IPv4 "Community Technology Preview" addon... I could DHCP an address, but not actually communicate with anyone.. I could send email to myself, but then again, I do that now. |
| 17:01.44 | fauxalliance | to:root from:root re:root |
| 17:01.47 | ccomp5950 | I didn't like that one, didn't work well with BeOS =( |
| 17:01.59 | fauxalliance | the Amiga loved it. |
| 17:02.17 | fauxalliance | and I still run that on OS/2 warp 4 |
| 17:02.47 | drmessano | ccomp5950: Didn't work with the NeXT machine I had. When I found out they turned NeXT into OSX and still nothing worked, I already knew why. |
| 17:02.49 | ccomp5950 | there was a community patch where it would work with BeOS but only on full moons and you had to install a light sensor. I think the developers were picking on me. |
| 17:02.49 | tzanger | carrar: I am almost positive that it's a 3 port switch and not a 3 port hub. Switches are stupid cheap now (and yes, you're right, they are only smart to the extent that they can match MAC ID) and I'd be very surprised if aastra or polycom phones had hubs |
| 17:03.18 | Poincare | tzanger: you're absolutely right, besides those phones hqve 100mbit switches and youre network/computers probably have gbit |
| 17:03.41 | drmessano | 10mbit should be fast enough for anyone |
| 17:03.43 | tzanger | Poincare: that too |
| 17:03.45 | fauxalliance | tzanger, does it have an ARP cache/ |
| 17:03.46 | fauxalliance | ? |
| 17:03.55 | tzanger | fauxalliance: possibly a VERY small one |
| 17:04.03 | tzanger | but then again it's only watching for (most cases) two MAC addresses |
| 17:04.04 | fauxalliance | then it's a switch |
| 17:04.20 | Kobaz | drmessano: 640kb should be enough ram for anyone |
| 17:04.23 | drmessano | "His" and "Hers" |
| 17:04.30 | fauxalliance | can you see traffic destined to other ports on the device? |
| 17:04.45 | fauxalliance | yes? stupid hub. |
| 17:04.52 | tzanger | fauxalliance: and in fact if they wanted to save money they could... they only need to watch for ONE MAC address... the phone itself. Any traffic not for the phone or broadcast goes to the other interface |
| 17:05.05 | drmessano | Kobaz: Glad you got the joke. Please donate to my legal defense fund |
| 17:05.12 | Kobaz | drmessano: heh |
| 17:05.16 | fauxalliance | tzanger, perhaps we can incrperate that into IPv7 |
| 17:05.41 | b11d` | no its IPv2010 |
| 17:05.55 | b11d` | or IPvISTA |
| 17:06.00 | drmessano | hahah |
| 17:06.00 | ccomp5950 | haha |
| 17:06.07 | fauxalliance | <haiku> TCP/IP, learn how it fits together, there is no escape</haiku> |
| 17:06.10 | drmessano | I'm stealing that one |
| 17:06.19 | b11d` | :P |
| 17:06.22 | drmessano | IPvISTA SP2 |
| 17:06.32 | fauxalliance | now with *nix interconnectivity |
| 17:06.45 | fauxalliance | and VLAN support ;-P |
| 17:06.54 | b11d` | now with options that'll make you drool but which will be removed from the actual final release |
| 17:07.05 | drmessano | "Microsoft Transmission Control Protocol / Internet Explorer Protocol for Workgroups 2011" |
| 17:07.11 | b11d` | hahaha |
| 17:07.30 | ccomp5950 | switches back to FDDI |
| 17:07.35 | fauxalliance | w00t! |
| 17:07.42 | fauxalliance | thats an interconnect for ya! |
| 17:07.53 | ccomp5950 | token ring...ON CRACK!~ |
| 17:07.55 | b11d` | fsck that.. lets to back to digiboards and 1200 bps modems |
| 17:07.58 | drmessano | RS232 ZIP cord LAN with Diodes |
| 17:08.06 | fauxalliance | we had dual FDDI's with countercurrent tokens between bldgs... fkcing wicked |
| 17:08.27 | drmessano | grabs the frozen yellow hose and taps |
| 17:08.49 | b11d` | there IS always IP over Avian Carrier |
| 17:08.56 | b11d` | the RFC on that is pretty serious stuff |
| 17:09.10 | ccomp5950 | they haven't implemented error correction in it yet =( |
| 17:09.11 | drmessano | b11d`: Sure, until you get Avian Flu and need to reboot |
| 17:09.16 | b11d` | haha |
| 17:10.08 | fauxalliance | drmessano, they keep me up all night.. fighting with the other misdelivered packets... best effort my ass |
| 17:10.09 | drmessano | "WARNING: VERY SERIOUS ERROR 0x73672363 You may be infected with Avian Flu. If this is the first time you're seeing this message, you're probably fscked anyway." |
| 17:10.47 | drmessano | "DO NOT IGNORE THIS WARNING, IT'S VERY SERIOUS" |
| 17:10.53 | drmessano | "That is all." |
| 17:11.53 | fauxalliance | anyone want to trade a VRRP license for a Zircom 9 pin serial to ethernet adaper? |
| 17:12.16 | carrar | You need a license to do VRRP? |
| 17:12.24 | fauxalliance | sure |
| 17:12.25 | carrar | What is that crap |
| 17:12.28 | b11d` | VRRP is patent encumbered by Cisco |
| 17:12.40 | b11d` | there is CARP now which is free |
| 17:12.40 | VanClone | oh another Cisco protocrap |
| 17:12.49 | Martinblr | drmessano: is there any other way to overcome with the call disconnection.. |
| 17:12.50 | carrar | HSRP in cisco |
| 17:13.01 | fauxalliance | Hot-Standby my ass |
| 17:13.12 | VanClone | cisco owns both HSRP and VRRP |
| 17:13.21 | VanClone | 0wnag3 |
| 17:13.26 | drmessano | I want to patent HLRP |
| 17:13.27 | VanClone | not really |
| 17:13.28 | fauxalliance | the IETF had a hand in it. |
| 17:13.49 | VanClone | drmessano: what's HLRP? |
| 17:13.50 | drmessano | Highly Litigious Routing Protocol .. "Use it and we'll sue you to death, like SCO" |
| 17:14.00 | VanClone | hahahaha |
| 17:14.07 | fauxalliance | ftp://ftp.openbsd.org/pub/OpenBSD/songs/song35.mp3 |
| 17:14.27 | drmessano | "How well does it work?" "We don't know.. can't implement it without suing our own asses off" |
| 17:14.31 | fauxalliance | ^^^ too funny... courtesy of OpenBSD. |
| 17:14.39 | carrar | use CARP! |
| 17:14.49 | drmessano | Can't Always Route Properly? |
| 17:14.57 | fauxalliance | carrar, PF |
| 17:15.25 | b11d` | never had any issues with CARP |
| 17:15.27 | b11d` | personally |
| 17:15.48 | drmessano | IGMP = I Guide MOST Packets |
| 17:15.55 | fauxalliance | CARP is awesome... and free. and truly redundant and unencumbered. |
| 17:16.08 | fauxalliance | MPLS? |
| 17:16.31 | b11d` | Many Packets Lost Suddenly |
| 17:16.38 | drmessano | HAHAHAH |
| 17:16.41 | coppice | Internet's Got My Packet |
| 17:16.43 | fauxalliance | b11d`, hear the CARP tune above? |
| 17:17.01 | b11d` | yeah |
| 17:17.07 | b11d` | i've followed openbsd for like 10 years |
| 17:17.12 | drmessano | BGP = Bitbuckets Gate Poorly |
| 17:17.27 | fauxalliance | People look at me strange listening to the tunes with the top down... |
| 17:17.48 | drmessano | ISDN = I Should Die Now |
| 17:18.20 | fauxalliance | BRI = better ring indefinitely. |
| 17:18.37 | Kobaz | ISDN = I Still Don't Need |
| 17:19.07 | Kobaz | ISDN = Instantly Spending Dollars Needlessly |
| 17:19.13 | b11d` | lol |
| 17:19.15 | drmessano | BRI = Budgeting Reduces Income |
| 17:20.19 | drmessano | ADSL = Always Down, Significant Latency |
| 17:20.23 | b11d` | lol |
| 17:20.46 | *** join/#asterisk slacker775 (~dhollis@rrcs-24-129-189-194.se.biz.rr.com) |
| 17:20.49 | drmessano | GSM = Grandma Sounds Muddy |
| 17:20.50 | fauxalliance | ISDN = intentionally slow digital network |
| 17:21.30 | drmessano | CDMA = Called, Deaf Man Answered |
| 17:21.52 | drmessano | That sounds like Verizon |
| 17:21.56 | drmessano | "Hello? hello?" |
| 17:21.57 | b11d` | Called Dad, Mom Answered |
| 17:22.00 | drmessano | HAHA |
| 17:22.21 | drmessano | Called Dad, Mongolia Answered |
| 17:22.59 | b11d` | China Dialed, Mongolia Answered |
| 17:23.06 | drmessano | hah |
| 17:23.39 | fauxalliance | CPE = carrier pigeon equiptment |
| 17:23.44 | coppice | Clinton Did Marketing in Asia |
| 17:24.34 | VanClone | hahahahahahahahaha |
| 17:25.09 | fauxalliance | Oracle - A proprietary system used to convert human souls into cash. |
| 17:25.10 | heffer | UMTS is called "Unerwartete Mehreinnahmen zur Tilgung von Staatsschulden" (Unexpected earnings for the repayment of national debts) in Germany. |
| 17:25.30 | drmessano | iphone = i'm probably holding old nextel equipment |
| 17:26.01 | heffer | because they made 50 Billion € by auctioning UMTS frequency licenses |
| 17:27.05 | Katty | hhhhhhhhhellllloooooooooooooooo nurse. |
| 17:29.49 | *** join/#asterisk mpe (~mpe@0xd99d3f8f.customer.cybercity.dk) |
| 17:30.59 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 17:32.09 | fauxalliance | drmessano, How many telecoms consultants does it take to change a light bulb? |
| 17:32.14 | fauxalliance | None, no one will do it but theyÂ’re all willing to talk about the prospect of change. |
| 17:32.24 | drmessano | lol |
| 17:32.38 | fauxalliance | Two mobile phones walk up to a pub. One says to the other: I’ll go inside and check things out. After a few loud noises inside, the mobile phone gets thrown out the front door. He says to his pal, “You can’t go in there. Bad reception.” |
| 17:32.57 | b11d` | boooooooooo |
| 17:33.02 | drmessano | hahah |
| 17:33.11 | fauxalliance | zero stars of five... agreed |
| 17:33.15 | b11d` | :) |
| 17:33.29 | b11d` | Bacon and Eggs walk into a bar.. bartender turns to them and says "Hey! We dont serve breakfast here!" |
| 17:33.31 | drmessano | Two Trixboxes walk into a bar.. the third one ducks |
| 17:34.01 | tzanger | wow, the comedians are out in full force today |
| 17:34.02 | *** part/#asterisk ruben23 (~RLACUMBA@121.97.111.142) |
| 17:34.02 | drmessano | Ok, no, the third one didn't duck.. Kerry Garrison just says it did |
| 17:34.16 | drmessano | TRIXBOX ALWAYS DUCKS |
| 17:34.23 | drmessano | I MEAN TRIXBOX NEVER DUCKS |
| 17:34.30 | drmessano | Oh, the good old days |
| 17:35.04 | leifmadsen | tzanger: ohai |
| 17:35.06 | *** join/#asterisk Defraz (~Defraz@c72co-edge-router.fuzecore.com) |
| 17:35.10 | drmessano | "trixbox contains absolutely no spyware.. and by that, i mean, we know what phones you're using and who is banging your wife when you're at work" |
| 17:35.11 | *** join/#asterisk b0ot (~Jinxed---@147.177.19.101) |
| 17:36.07 | b0ot | Any idea how my phones in my network are able to talk to one another fine, but my sj softphones are only able to recieve audio not send it |
| 17:36.47 | fauxalliance | NAT gone wonky? |
| 17:37.03 | b11d` | had that happen once, was an incorrect default gateway |
| 17:38.07 | Katty | leifmadsen: ohai2u |
| 17:38.36 | fauxalliance | <PROTECTED> |
| 17:38.41 | dforbu | so.. back in the day, on my old system, I had voicemail with options in mysql table, one of the options was outgoing=yes - which made * play the unavail.wav and hang up. that option has a different syntax in 1.6? I dumped the db into my new * install and it has no effect anymore.. |
| 17:39.36 | *** join/#asterisk odenkos_web (d45113aa@gateway/web/freenode/ip.212.81.19.170) |
| 17:40.51 | dforbu | b0ot - firewall blocking traffic in 1 direction? |
| 17:41.02 | fauxalliance | doubtful |
| 17:41.25 | *** join/#asterisk trelane (~trelane@funtoo/staff/trelane) |
| 17:41.33 | fauxalliance | one-way auido screams NAT foo. |
| 17:43.09 | odenkos_web | hi all, I'm compiling * on a low-end machine.. it doesn't have gtk+ installed for menuselect to work.. is there any other way (some file) to edit the list of features it should compile? |
| 17:43.18 | *** join/#asterisk myster (~myster@207.148.172.210) |
| 17:44.02 | fauxalliance | odenkos_web, how low end? do you plan on accomplishing much with it? |
| 17:44.36 | b0ot | fauxalliance, NAT? |
| 17:44.53 | fauxalliance | b0ot, yes NAT |
| 17:44.57 | Tim_Toady | odenkos_web you dont need gtk, only ncurses |
| 17:45.10 | b0ot | fauxalliance, what do you think the issue is |
| 17:45.11 | *** join/#asterisk aidinb (~Aidin@unaffiliated/aidinb) |
| 17:45.31 | fauxalliance | b0ot, firewall |
| 17:46.16 | b0ot | fauxalliance, i have the firewalls both disabled |
| 17:46.23 | *** join/#asterisk jinxed (~chatzilla@CPE0016b6eddb9a-CM001404dc5032.cpe.net.cable.rogers.com) |
| 17:47.01 | odenkos_web | fauxalliance: sparc procesor, 256 MB ram, *really* low end.. and I will only use it as a home system with < 10 peers |
| 17:47.20 | odenkos_web | no dahdi, no meetme, no pri. |
| 17:47.28 | fauxalliance | barebones for sure. |
| 17:48.21 | odenkos_web | so is there any file the menuselect options are saved to or something? |
| 17:51.56 | ectospasm | menuselect needs ncurses, last I checked... |
| 17:52.14 | b11d` | also gonna need to be in an X environment otherwise the console screen is too narrow :) |
| 17:52.19 | b11d` | failed for me anyways |
| 17:53.18 | odenkos_web | b11d`: I'm sure I don't have X on there ;) |
| 17:53.30 | b11d` | theres a .h file you can modify to change the width req.. |
| 17:53.35 | b11d` | dont remember it though :) |
| 17:53.50 | odenkos_web | b11d`: thanks for the tip, I'll look into it |
| 17:54.24 | *** join/#asterisk javar (~javier@186.83.27.216) |
| 17:54.42 | *** part/#asterisk javar (~javier@186.83.27.216) |
| 17:54.50 | b11d` | np |
| 17:58.45 | asilva | any help with this :: http://pastebin.com/rFf5nM4T |
| 18:07.14 | trelane | is there a sip switch that can terminate a large amount of t1's (more than 4) and provide SIP to an asterisk PBX? |
| 18:07.44 | Kobaz | audiocodes makes one i think |
| 18:07.49 | b11d` | get a bunch of t1 cards ;) |
| 18:08.20 | Kobaz | trelane: for a sip switch like that from something like audiocodes is going to run you about $30k |
| 18:08.27 | WIMPy | Last time we came to the conclusion that 8xPRI cards give you the highest possible port density. |
| 18:08.48 | trelane | Kobaz, and if I hate audiocodes with an insane burning passion? (anything from Cisco or Adtran?) |
| 18:09.06 | trelane | WIMPy, elaborate? |
| 18:09.07 | Kobaz | trelane: then you'll need to roll your own |
| 18:09.12 | Kobaz | personally i love audiocodes and adtran |
| 18:09.13 | trelane | Kobaz, noted |
| 18:09.35 | Kobaz | i just wish audiocodes were a little cheaper |
| 18:09.47 | ritztech | anyone know where to obtain Asterisk Support im having the worst luck with all these consulting companies |
| 18:09.52 | trelane | Kobaz, I love adtran and would prefer to use them, I had a REALLY bad experience with getting an audiocodes product configured, a manual written in incredibly fluent engrish, and non-helpful customer no-service |
| 18:09.53 | ritztech | http://www.voip-info.org/wiki/view/Asterisk+consultants+USA |
| 18:09.56 | WIMPy | trelane: Sangoma |
| 18:10.06 | trelane | ritztech, what problem are you having? |
| 18:10.14 | trelane | WIMPy, I do in fact love them, they're good folk |
| 18:10.15 | Kobaz | trelane: 904 series, but it's single pri |
| 18:10.21 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
| 18:10.41 | ritztech | its an advanced thing that im trying to get done with QUEING outbound paging |
| 18:11.01 | Kobaz | they are making a pure multispan pri->sip sometime next year |
| 18:11.19 | b11d` | no |
| 18:11.20 | ritztech | Anyone know a good Asterisk support Company that actually is legit |
| 18:11.23 | b11d` | the sangoma people SUCK now |
| 18:11.38 | b11d` | they used to be great, then within 6 months they just dropped the ball BIG time |
| 18:11.46 | trelane | b11d`, what happened? |
| 18:11.48 | b11d` | no support people return calls or email, no sales.. nothing |
| 18:11.52 | b11d` | forced me to buy digium cards |
| 18:11.57 | b11d` | i used to LOVE them |
| 18:12.01 | b11d` | bought a dozen pri cards from them |
| 18:12.14 | b11d` | tried for over a week to contact them without any success |
| 18:12.25 | b11d` | phone.. email... nothing |
| 18:12.37 | b11d` | so i have nothing good to say about them now |
| 18:13.45 | trelane | that's sad :( |
| 18:13.55 | b11d` | yeah, i think so too.. |
| 18:13.57 | trelane | they failed to make the channelized T3 card I wanted |
| 18:13.58 | b11d` | i friggin loved them |
| 18:14.19 | trelane | because being able to terminate 28 t1's on a PCI slot in linux would have made me happy beyond belief |
| 18:14.34 | b11d` | no way the PCI port could handle that |
| 18:14.49 | trelane | b11d`, sure it will, there's more than enough bandwidth |
| 18:14.55 | b11d` | PCIe ? |
| 18:14.56 | b11d` | yeah |
| 18:15.00 | trelane | PCI yeah |
| 18:15.05 | trelane | but you could only do 1 in a system |
| 18:15.09 | b11d` | yeah |
| 18:15.10 | trelane | PCIE... that would let you do one per x1 |
| 18:15.26 | trelane | notes that even 1 in a system is cheaper than a 7206 |
| 18:16.04 | ectospasm | ritztech: Digium (-; |
| 18:16.42 | ectospasm | ritztech: they offer Open Source Subscriptions, including for queuing support |
| 18:17.14 | ritztech | they are like charing $8000 for this type of customizaitin |
| 18:17.43 | trelane | ritztech, that is actually a fairly complex customization. I was considering quoting it, but you have to queue the pages somewhere and have an AGI app play 'em one at a time |
| 18:17.51 | ectospasm | ritztech: Oh, I wasn't aware this was classified as consultative support. |
| 18:18.38 | trelane | ectospasm, sadly what he's asking for is fairly complex, you'd either have to write an AGI script (which I don't do), or write/modify a module to handle it |
| 18:18.43 | trelane | it's a cool idea though |
| 18:18.50 | ritztech | which im not sure how to Que the pages at all unless any of you can quote me for AGI support since my dialplan actually works |
| 18:19.01 | ectospasm | yeah, and Digium won't help you write the AGI, even with an L4 subscription. |
| 18:19.13 | ritztech | thats sucks haa |
| 18:19.23 | trelane | ritztech, check the extension of the pager, record the page to a file, and run an agi or something that checks the folder, plays the pages in order, and deletes after playing |
| 18:19.55 | trelane | you might even be able to do the last bit with a bash script, cron, and a call file |
| 18:19.56 | ritztech | how would i look at a agi (is taht some sort of perl thing) |
| 18:20.05 | ritztech | i did read of a .call file |
| 18:20.26 | trelane | ritztech, in the end I think you're going to have to generate a call file based on a spool of recorded pages |
| 18:20.27 | ectospasm | AGI: Asterisk Gateway Interface, essentially a dialplan script to be executed in any language you so choose |
| 18:20.53 | ectospasm | you could have a call file execute the AGI |
| 18:21.19 | ritztech | i have no idea on .call files Though i did see this |
| 18:21.19 | ritztech | http://www.russellconsultants.com/information/how-to-mainmenu-15/21-networking/18-queued-asterisk-call-files.html |
| 18:21.20 | ectospasm | ...and use cron or some other shell program copy the call file to the outgoing spool |
| 18:21.35 | ectospasm | ritztech: it's in the book |
| 18:21.44 | WIMPy | Don't copy! |
| 18:22.03 | ectospasm | WIMPy: ah, yeah, but I never have problems with it that way |
| 18:22.15 | WIMPy | Luck. |
| 18:22.17 | ectospasm | WIMPy: but of course I only copy from the CLI prompt |
| 18:22.20 | WIMPy | Pure luck. |
| 18:22.30 | WIMPy | That doesn't matter. |
| 18:22.52 | ectospasm | ...and my call files are rather short, just channel, application, and data. |
| 18:23.08 | ectospasm | ...but I concede your point. |
| 18:25.17 | ectospasm | actually, in all my years of doing this, I have yet to see a call file fail when I used cp rather than mv |
| 18:25.29 | ectospasm | ...but there is always the potential |
| 18:25.32 | *** join/#asterisk ManxPower (~manxpower@24.236.78.136) |
| 18:25.38 | ManxPower | Is anyone here familiar with the phpagi add_event_handler function? |
| 18:27.48 | ManxPower | I can't seem to make it keep waiting for events without exiting. wait_response seems to expect to wait for a response to a sent event and sleep() just blocks |
| 18:30.58 | *** join/#asterisk Letoric (~Letoric@tpi-dfw-uc-f1-69-94-238-22.totalprocess.net) |
| 18:31.49 | ManxPower | There are exactly 11 hits on google for: phpagi add_event_handler |
| 18:32.09 | Letoric | <--- Windows background, so forgive the next question please ;P |
| 18:32.20 | Letoric | What is the proper way to 'upgrade' Asterisk on a linux system |
| 18:32.37 | Letoric | do I remove the old binaries/ |
| 18:32.42 | ManxPower | Letoric, reinstall, but do NOT run "make samples" |
| 18:32.49 | ManxPower | Letoric, you don't remove the old binaries. |
| 18:33.02 | Letoric | ok. Now here's another question that ties in |
| 18:33.16 | Letoric | What happens when you were on an older version that loaded modules that the new one shouldn't load |
| 18:33.31 | Letoric | does the make install do some kind of clean-up first? |
| 18:33.53 | Letoric | just thinking outside the box, to make sure I do things correctly each time |
| 18:33.54 | Letoric | ;) |
| 18:33.55 | ManxPower | Letoric, your pbx blows up until you fix the issue. If you want uninstall features then you have to use an Asterisk package for your distro, and that is not recommended. |
| 18:34.16 | Letoric | heh, ok ;) |
| 18:34.42 | Letoric | my CEO is a hardcore unix guy, but he just removes all the old stuff and reconfigures things each time |
| 18:34.46 | Letoric | that seemed like the hard path to me |
| 18:34.47 | ManxPower | Letoric, unless you are moving between major varions it should not be a problem. |
| 18:35.07 | ManxPower | Letoric, Asterisk is one of the only software that I do NOT recommend using package management with |
| 18:35.36 | ectospasm | Letoric: you'll have to clear out /usr/lib/asterisk/modules, then Asterisk should start OK |
| 18:35.49 | ectospasm | Letoric: do NOT run "make samples" as ManxPower stated |
| 18:35.51 | ManxPower | Letoric, most of the time the process of upgrading asterisk is: ./configure && make install && asterisk -rx "restart now" |
| 18:36.07 | Letoric | Thanks. I'll keep it in mind for future upgrades |
| 18:36.14 | ectospasm | I've just started doing something at work, passing --prefix="/opt/asterisk-<version>" to configure |
| 18:36.29 | ectospasm | there all Asterisk related stuff for that version exists in /opt |
| 18:36.31 | Letoric | so far I've been running a test box and a production box, and just alternating them each time after I test so it's always clean |
| 18:36.37 | ManxPower | ectospasm, you must be using 1.6.x |
| 18:36.44 | ectospasm | ManxPower: works for 1.4, too |
| 18:36.49 | ectospasm | ...for the most part |
| 18:37.20 | ectospasm | Actually, I'm not at work so I can't verify that statement |
| 18:38.48 | Katty | hehe. |
| 18:38.50 | Katty | ecto spasm. |
| 18:38.55 | wcselby | o/ |
| 18:39.00 | Katty | ohai wcselby!!! |
| 18:39.03 | Katty | hugs wcselby |
| 18:39.06 | wcselby | howdy |
| 18:39.07 | wcselby | :) |
| 18:39.14 | Katty | what's blazin |
| 18:39.26 | wcselby | not much |
| 18:39.30 | wcselby | yourself? |
| 18:39.44 | ManxPower | I've like to blaze Javascript into ashes |
| 18:43.02 | ectospasm | ManxPower: yep, I confirmed: ./configure --prefix="/opt/asterisk-1.4.35" works, all directories in asterisk.conf are OK. |
| 18:43.02 | Katty | wcselby: pretty plum day |
| 18:43.02 | wcselby | js ain't too bad |
| 18:43.40 | ManxPower | wcselby, it is not bad if you drink the "everything is an object" model. I do not. |
| 18:44.07 | ManxPower | I like PHP because it tries to be useful, not pedantically perfect. |
| 18:44.40 | wcselby | heh |
| 18:44.44 | wcselby | i didn't say I used js |
| 18:44.53 | ectospasm | heh, Bjarn Stroustroup said you should only use OOP when necessary |
| 18:44.53 | florz | PHP? useful? muahaha! |
| 18:44.54 | wcselby | just that it ain't the worst one out there |
| 18:45.02 | wcselby | not sure which one I'd call the worst one |
| 18:45.03 | ManxPower | wcselby, I would not use it if I had a choice. |
| 18:45.14 | wcselby | i always end up mostly doing perl scripts for my needs |
| 18:45.58 | wcselby | but you know, according to a prominent poster on the -users mailing list, you can initialize xxx number of C programs in the time it takes to load the perl / php / whatever compiler to run one of those scripts |
| 18:46.19 | ectospasm | wcselby: it's an interpreter, not a compiler |
| 18:46.22 | wcselby | of course, it takes a week to write one of those programs, but I digress |
| 18:46.39 | wcselby | ectospasm - whatever, I'm not a programmer, obviously. i write scripts in perl |
| 18:46.41 | wcselby | :) |
| 18:46.51 | ectospasm | Perl/PHP/etc. are generally interpreted, not compiled. Although compilers do exist. |
| 18:47.11 | ectospasm | You could compile a Perl program, and it'd run almost (or just) as fast as a C program |
| 18:47.22 | florz | no, you could not |
| 18:47.33 | wcselby | uh oh |
| 18:47.35 | wcselby | what did i start |
| 18:47.38 | ectospasm | There are Perl compilers. |
| 18:47.42 | wcselby | not a programming relgious war |
| 18:47.42 | *** join/#asterisk imox1234 (~imox1234@p4FC5C512.dip0.t-ipconnect.de) |
| 18:47.44 | florz | there are not |
| 18:47.52 | Tim_Toady | perlcc is deprecated :P |
| 18:48.15 | Tim_Toady | unmaintained and not in the def distribution anymore |
| 18:48.24 | ectospasm | Tim_Toady: distribution of what? |
| 18:48.27 | Tim_Toady | of perl |
| 18:48.37 | ectospasm | There can't be commercial compilers? |
| 18:48.47 | florz | plus, even if there was, that wouldn't make it run as fast as C |
| 18:49.05 | florz | which is kindof a bogus comparison anyway |
| 18:49.27 | ectospasm | it would run faster than the interpreted Perl |
| 18:49.40 | florz | not necessarily |
| 18:49.42 | Katty | wcselby: programming jihad on you! |
| 18:49.49 | Katty | wcselby: <3 |
| 18:49.51 | ectospasm | florz: depends on the compiler, though. |
| 18:49.58 | florz | and on the interpreter |
| 18:50.15 | wcselby | Katty - apparently |
| 18:50.17 | ectospasm | in general, though, compiled programs will always be faster than interpreted ones. |
| 18:50.25 | florz | no |
| 18:50.51 | ManxPower | speed kills |
| 18:51.09 | ectospasm | how can you get optimization in an interpreter like you can with a compiler? |
| 18:51.19 | Katty | ManxPower: only if you hit something |
| 18:51.28 | wcselby | Katty - or it hits you |
| 18:51.34 | Katty | that too. |
| 18:51.47 | florz | ectospasm: by building it into the interpreter |
| 18:51.49 | b0ot | Ok, I have a weird issue with my softphones. I can only get one way audio where the softphone can recieve audio but not talk when I dial out. BUT when I am dialed the softphone is able to recieve/transmit audio any ideas? |
| 18:52.09 | florz | ectospasm: also, you are assuming that there is much that can be optimized |
| 18:53.31 | ectospasm | florz: I let the compiler/interpreter determine that. But then again, I'm not a programmer, either (-; |
| 18:57.17 | Katty | wcselby: adreneline++ |
| 18:58.23 | Kobaz | okay so i have a weird problem |
| 18:58.25 | Kobaz | Got RTP RFC2833 from 192.168.15.1:16066 (type 101, seq 009324, ts 1620315480, len 000004, mark 1, event 00000001, end 0, duration 00160) |
| 18:58.34 | Kobaz | i have a background() and a 1 exten |
| 18:58.50 | Kobaz | i get a dtmf digit of 1... but it doesn't do anything |
| 18:58.58 | Kobaz | and asterisk doesn't report any dtmf recieved in the dialplan |
| 18:59.04 | Kobaz | if i call from a different phone... it works fine |
| 19:01.29 | ectospasm | Kobaz: are you sure the phone is set for RFC2833? |
| 19:01.33 | Kobaz | yes |
| 19:01.41 | Kobaz | both phones are rfc2833 |
| 19:01.48 | Kobaz | something is different somewhere though |
| 19:01.51 | Kobaz | otherwise both would work |
| 19:01.53 | ectospasm | are they otherwise identical phones? |
| 19:02.02 | Kobaz | yes |
| 19:02.05 | Kobaz | polycom 331 |
| 19:02.21 | Kobaz | what's weird. is i see the rfc2833 going through |
| 19:02.24 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 19:03.07 | ectospasm | I dunno then, that is weird. |
| 19:03.28 | Kobaz | yeap |
| 19:05.02 | ectospasm | I'd suggest going over the problem phones settings, in the phone, not sip.conf et al. |
| 19:05.30 | ectospasm | I can't remember the last time I saw this, but I want to say it was because the phone DTMF mode was set for inband |
| 19:05.40 | b0ot | Ok, I have a weird issue with my softphones. I can only get one way audio where the softphone can recieve audio but not talk when I dial out. BUT when I am dialed the softphone is able to recieve/transmit audio any ideas? |
| 19:05.46 | b0ot | Here is my sip debug log: http://pastebin.ca/1943043 |
| 19:05.53 | b0ot | or at least a good portion |
| 19:06.32 | ectospasm | b0ot: is there a NAT between the softphone and Asterisk? |
| 19:06.45 | b0ot | shouldn't be ectospasm |
| 19:06.58 | b0ot | its all a private network |
| 19:07.20 | b0ot | There should be no NAT |
| 19:07.35 | ectospasm | sounds like a problem with the softphones |
| 19:07.46 | carrar | or pc firewall |
| 19:08.12 | ectospasm | I always forget about that part. |
| 19:08.20 | ectospasm | ...I don't deal with softphones much |
| 19:09.31 | *** join/#asterisk odenkos_ntbk (~odenkos@ip-212-081-019-170.static.nextra.sk) |
| 19:09.45 | ManxPower | Here is the code I'm having problems with: http://pastebin.ca/1943048 |
| 19:10.07 | ManxPower | if ANYONE has experience with phpagi's add_event_handler I'd really appreciate help. |
| 19:12.30 | b0ot | ectospasm, carrar I have completly turned off the firewall on both pc's and they are running two different versions of SJphone |
| 19:13.02 | b0ot | Are there any free softphones that are recommended for compatiability for asterisk? |
| 19:13.05 | ManxPower | b0ot, turn off reinvites and to a tcpdump to see what is going on |
| 19:13.13 | *** join/#asterisk ruben23 (~ITadmin@125.212.40.2) |
| 19:13.47 | b0ot | ManxPower, how would i do that? |
| 19:13.59 | Chainsaw | b0ot: Ekiga comes to mind. |
| 19:14.21 | *** join/#asterisk retentiveboy (~pdugas@74-95-28-33-Atlanta.hfc.comcastbusiness.net) |
| 19:15.27 | Katty | stares at clock |
| 19:15.29 | Katty | pouts. |
| 19:15.44 | retentiveboy | Likely it's a silly questions but... Is there a way to detect an inbound SIP/T.38 fax call before answering a call? Wondering if I can get away with using a single inbound number if my provider supported T.38. |
| 19:16.11 | b0ot | Chainsaw, do you have something that more just a phone for an interface and works with windows. I dont need/want any of the chatting features |
| 19:16.34 | wcselby | retentiveboy - if you're on 1.6.2 there's faxdetect=yes on sip.conf |
| 19:16.51 | wcselby | or maybe "on" instead of "yes" |
| 19:16.53 | wcselby | check the sample file |
| 19:17.04 | Chainsaw | b0ot: X-Lite in that case. |
| 19:17.13 | Chainsaw | b0ot: Or it's paid more fully-featured brother Eyebeam. |
| 19:17.22 | Chainsaw | s/it's/its/ |
| 19:18.02 | ManxPower | b0ot, set canreinvite=no in sip.conf [general] then execute tcpdump on the asterisk server |
| 19:18.18 | retentiveboy | wcselby: I remember that from back when I was using zap channels. It only worked once the call was answered for obvious reasons. Is T.38 signaled early enough like that? |
| 19:19.27 | retentiveboy | duh, it's described in the config file. nm |
| 19:21.56 | odenkos_ntbk | hey guys.. I'm getting this error when compiling chan_sip.. any ideas? http://pastebin.com/T2UNWp2k |
| 19:23.17 | *** join/#asterisk NiugeS (~NiugeS32@5e0c4557.bb.sky.com) |
| 19:23.42 | WIMPy | odenkos_ntbk: Truncated file? |
| 19:24.44 | odenkos_ntbk | WIMPy: think so? |
| 19:25.42 | ManxPower | odenkos_ntbk, download the source from downloads.digium.com again |
| 19:25.48 | NiugeS | hi all... i am trying to build a low level report in Excel.. at the moment I download and copy and past the CDR however someone mentioend I can connect direclty to the CDR.. can anyone point me to some documentation or let me know how best to proceed? Thanks |
| 19:26.07 | odenkos_ntbk | ManxPower: ahhh.. great |
| 19:27.26 | *** join/#asterisk netmax (~netmax@is.linux-administrator.com) |
| 19:29.33 | Kobaz | ectospasm: now both phones don't work with dtmf |
| 19:30.45 | *** join/#asterisk NiugeS (~NiugeS32@5e0c4557.bb.sky.com) |
| 19:30.45 | Kobaz | ectospasm: actually it seems random... sometimes dtmf works, and sometimes it doesnt |
| 19:30.49 | *** part/#asterisk retentiveboy (~pdugas@74-95-28-33-Atlanta.hfc.comcastbusiness.net) |
| 19:32.36 | *** join/#asterisk p3nguin_ (gpz5GvdFkf@cpe-0210-4bff-fe2b-9074.dhcp.a2infotech.com) |
| 19:32.51 | b0ot | X-lite looked promissing but it wanted me to install some c++ package or something. Has anyone here used firefly, (it also is a lot smaller than x-lite) |
| 19:33.27 | ManxPower | Kobaz, sounds like you are connecting to an ITSP. |
| 19:33.46 | odenkos_ntbk | WIMPy, ManxPower: the sha1sum is correct on the download |
| 19:33.59 | ManxPower | b0ot, Understand that "all softphones suck". |
| 19:34.31 | b0ot | ManxPower, my network consists of 4 phones... two soft phones and two polycoms. 3/4 phones are on the same subnet, and the other is 1 jump |
| 19:34.40 | *** join/#asterisk imcdona (~imcdona@valium.voicebyip.net) |
| 19:35.30 | b0ot | I don't get how It can have transmit/recieve audio on softphones when you call out from a real phone to the soft phone, but not when you place a call with the softphone |
| 19:36.17 | *** join/#asterisk bsaxon (~bsaxon@12.107.149.61) |
| 19:36.49 | WIMPy | odenkos_ntbk: You must have something fundametally incompatible (or missing?) then. However such kind of errors can also be caused by hardware defects. |
| 19:36.56 | ManxPower | b0ot, there is no difference |
| 19:37.31 | ManxPower | b0ot, have you done a tcpdump to see the ACTUAL RTP audio packets reaching the server? |
| 19:38.00 | ManxPower | (the reason you want canreinvite=no is so the data continues to be sent via the server and not directly between the two phones. |
| 19:38.15 | odenkos_ntbk | WIMPy: could you define "hardware defects" please? |
| 19:38.20 | ManxPower | b0ot, Oh! Do you have g729 or g723.1 or the "all" codec enabled in sip.conf? |
| 19:38.32 | ManxPower | odenkos_ntbk, memory corruption |
| 19:39.08 | WIMPy | odenkos_ntbk: Memory (incl. cache), CPU, bus. |
| 19:39.10 | b0ot | ManxPower, codec=all |
| 19:39.20 | ManxPower | b0ot, DO NOT EVER DO THAT! |
| 19:39.27 | b0ot | well allow=all |
| 19:39.36 | ManxPower | don't do that either 8-) |
| 19:39.46 | b0ot | ok |
| 19:39.51 | ManxPower | disallow=all and allow=ulaw when you are testing |
| 19:40.07 | ManxPower | b0ot, otherwise Asterisk could select a codec that IT DOES NOT SUPPORT |
| 19:40.43 | ManxPower | Usually your call would drop if that was the case, but better to keep your codecs simple when you have problems |
| 19:40.47 | odenkos_ntbk | ManxPower, WIMPy: my chan_sip.c file isn't corrupt in any way.. it must be something with the hardware, but since I'm compiling this on a virtual machine.. I don't know what could be wrong |
| 19:41.37 | b0ot | ManxPower, so it should be. [general] | port = 5060 | bindaddr = 0.0.0.0 | disallow=all | allow=ulaw | context = bogon-calls ? |
| 19:41.51 | ManxPower | odenkos_ntbk, if chan_sip.c is not corrupted then there are only two other things it could be. 1) compiler bug 2) on disk file corruption of some part of the compiler or 3) memory or hardware issue. |
| 19:42.06 | ManxPower | b0ot, don't use port or bindaddr either |
| 19:42.46 | b0ot | how will my phones connect to it then? |
| 19:42.56 | NiugeS | does anyone have a suggestion on how to remotely connecto the the CDR database? |
| 19:43.16 | ManxPower | b0ot, Asterisk will pick the correct defaults |
| 19:43.29 | odenkos_ntbk | ManxPower: as much as I know it could be any of them |
| 19:44.42 | WIMPy | odenkos_ntbk: You could try to re-install your toolchain. Otherwise a memort test might be a good idea. |
| 19:45.59 | WIMPy | The best memory test is compiling something big with gcc, however. Like compiling gcc itself. If that succeeds the memory is ok. |
| 19:46.34 | b0ot | ManxPower, now I can transmit audio from the softphone, but can't recieve audio from the softphone |
| 19:49.51 | Kobaz | aww |
| 19:49.54 | Kobaz | my computer locked up |
| 19:51.13 | odenkos_ntbk | WIMPy: how much memory does gcc need to compile *? |
| 19:56.03 | *** join/#asterisk csnook (~chris@209-6-38-66.c3-0.smr-ubr1.sbo-smr.ma.cable.rcn.com) |
| 20:00.52 | *** join/#asterisk Alagar (~Administr@122.164.179.78) |
| 20:04.10 | ManxPower | b0ot, pastebin your tcpdump |
| 20:06.09 | *** join/#asterisk ccesario_ (~ccesario@189-29-53-173-ac.cpe.vivax.com.br) |
| 20:06.48 | b0ot | ManxPower, I don't know how to do a tcpdump here is my sip.conf: http://pastebin.com/wxUAV5kB extentions.conf: http://pastebin.com/TMfg2r6m and sip debug: http://pastebin.com/pL6eTsEU |
| 20:09.32 | ectospasm | b0ot: man tcpdump or man wireshark |
| 20:10.41 | Katty | ectospasm: i just can't get over your /nick |
| 20:10.55 | ectospasm | Katty: you like it? |
| 20:10.56 | Katty | ectospasm: it induces giggling every time. |
| 20:11.05 | ectospasm | I'm glad I amuse |
| 20:11.08 | Katty | ectospasm: <3 |
| 20:11.26 | b0ot | ectospasm, i can capture an interface with wireshark... should i do it making calls? |
| 20:11.53 | ectospasm | b0ot: yeah, see if you get RTP flowing in both directions on the failed call |
| 20:12.32 | [TK]D-Fender | b0ot: Your SIP debug did not contain a CALL |
| 20:13.33 | odenkos_ntbk | ManxPower, WIMPy: I changed the virtual machines memory from 128 to 256 and chan_sip compiles as intended.. thanks guys! |
| 20:15.38 | b0ot | aaaaahhhh what the heck now it works! |
| 20:15.41 | b0ot | I didn't change anything |
| 20:20.56 | ManxPower | b0ot, it started working because [TK]D-Fender showed up. He is called the Asterisk Whisperer around here. |
| 20:21.55 | b0ot | Lol [TK]D-Fender has helped me multiple times :) but I would be hesitant to ever use anything with whispering to describe it :p |
| 20:22.19 | [TK]D-Fender | Whiper-Page <- very useful |
| 20:22.29 | [TK]D-Fender | Chanspy w/ whisper |
| 20:27.19 | *** join/#asterisk _pepo_ (~kvirc@p153.etapanet.net) |
| 20:27.51 | carrar | whispers |
| 20:28.29 | *** join/#asterisk cesurasean (~Sean_Brad@server.simplewebhosting.us) |
| 20:28.43 | cesurasean | can someone explain the advanage of using a TDM card? |
| 20:28.47 | cesurasean | vs SIP? |
| 20:29.05 | Kobaz | as usual, depends what you are doing |
| 20:29.43 | carrar | if you don't have internet |
| 20:29.58 | carrar | faxes work better |
| 20:30.06 | [TK]D-Fender | checkout time..... |
| 20:30.33 | carrar | probably better connectiviy quality |
| 20:30.56 | cesurasean | so a TDM card is actually better? |
| 20:31.09 | carrar | heh |
| 20:31.23 | carrar | over what |
| 20:31.27 | cesurasean | SIP |
| 20:31.36 | carrar | not necessarly |
| 20:31.37 | cesurasean | i already have a SIP for vonage setup |
| 20:31.43 | *** part/#asterisk slacker775 (~dhollis@rrcs-24-129-189-194.se.biz.rr.com) |
| 20:31.46 | carrar | You haven't given anydetails |
| 20:31.47 | cesurasean | cant i get faxes using that? |
| 20:32.04 | cesurasean | http://cesurasean.cjb.net - you can see my config here |
| 20:32.26 | cesurasean | <PROTECTED> |
| 20:32.29 | ManxPower | cesurasean, SIP will be as reliable as the sum total of how reliable every connection and router between you and your SIP provider. A TDM card will be as reliable as your telephone line. You decide. |
| 20:32.31 | carrar | you will be limited to 23 concurrent calls with a PRI |
| 20:32.34 | cesurasean | i don't quite udnerstand this shit cause im a newbie |
| 20:32.58 | carrar | then go with a PRI |
| 20:33.12 | carrar | less for you to potentially worry about |
| 20:33.19 | carrar | but again, no details |
| 20:33.19 | cesurasean | ManxPower, thx. that's what i needed to hear. |
| 20:33.39 | ManxPower | TDM is almost always more expensive, but is generally more reliable |
| 20:34.05 | carrar | yuppers |
| 20:34.06 | cesurasean | i will start as SIP and maybe migrate once i learn asterisk a bit |
| 20:34.10 | ManxPower | Unless you are in the NYC area, then who knows how reliable TDM will be. Verizon seems to like randomly screwing up our customer's phone lines. |
| 20:34.19 | carrar | heh |
| 20:34.21 | ManxPower | cesurasean, that is the best idea I've heard. |
| 20:34.41 | cesurasean | thanks guys |
| 20:34.57 | carrar | and, build your asterisk from Source!! |
| 20:35.13 | cesurasean | i used debian repo to install it. :) |
| 20:35.16 | cesurasean | works fine! |
| 20:35.42 | ManxPower | carrar, we sometimes get "commit" dates from Verizon for a problem being fixed that are 2 weeks in the future |
| 20:36.16 | carrar | heh |
| 20:36.18 | carrar | nice |
| 20:36.30 | ManxPower | they also love to break some other line at the same customer when they are there to repair a line. |
| 20:36.40 | Kobaz | heh |
| 20:37.20 | ManxPower | Today we had Verizon claim that the customer denied them access to take the line down, but somehow Verizon can't actually tell us the tech's name. |
| 20:37.49 | ManxPower | And the building manager, who would have the key to the telcom closet and the customer claiming Verizon never showed up. |
| 20:37.53 | ManxPower | </rant> |
| 20:38.10 | carrar | yeah |
| 20:38.13 | carrar | That happens |
| 20:38.25 | carrar | or they put it in the wrong building |
| 20:38.26 | odenkos_ntbk | :se noai |
| 20:38.27 | odenkos_ntbk | a:se ai |
| 20:38.32 | thecardsmith | one trick i use when i file a ticket with verizon... |
| 20:38.34 | odenkos_ntbk | sorry |
| 20:38.39 | thecardsmith | ...is i always flag the ticket as "d-channel down" |
| 20:38.40 | thecardsmith | for t's |
| 20:38.45 | thecardsmith | and that freaks 'em out, for some reason |
| 20:38.50 | thecardsmith | then, i just say "oops." |
| 20:38.53 | thecardsmith | so I just get someone on the phone |
| 20:39.58 | thecardsmith | cause usually it takes 'em like 4-8 hours to respond to my tickets, otherwise. although we have gear at a vz data center, and those guys are actually "ok", like satisfactory |
| 20:40.38 | ManxPower | thecardsmith, usually these are either data lines (T-1 or EoC) or POTS |
| 20:41.31 | thecardsmith | good ole copper! nice. i think we have data from vz too, but, yeah pris from 'em mostly |
| 20:43.22 | ManxPower | I love PRIs and would only use PRI if I had a choice. |
| 20:43.58 | Kobaz | yeap |
| 20:43.59 | thecardsmith | i dig PRIs myself... although I'd use SS7 if I had a choice |
| 20:44.08 | ManxPower | Our customers with lots of DIDs usually use a PRI because it is quite a bit cheaper per-DID than using SIP from Verizon Wholesale. |
| 20:45.15 | thecardsmith | no kidding! |
| 20:45.18 | thecardsmith | i would've guessed otherwise |
| 20:46.34 | ManxPower | thecardsmith, So would I, but VZ considers ALL SIP DIDs to be "foreign" and are billed as though they are pulled out of a remote CO |
| 20:48.39 | thecardsmith | ahhh ha, alright, that does make sense |
| 20:48.52 | thecardsmith | doesn't usually count the beans... although he has gotten the short end of the stick from the bean counters! |
| 20:49.17 | ManxPower | Recently Verizon has started leaving orders in a pending state after the install completes. We can't open trouble tickes on the lines when they are still in an "install" state. |
| 20:49.47 | ManxPower | happened twice this week |
| 20:49.50 | thecardsmith | rolls his eyes |
| 20:49.52 | *** join/#asterisk mpe (~mpe@0xd99d3f8f.customer.cybercity.dk) |
| 20:49.52 | thecardsmith | sounds like vz |
| 20:50.17 | ManxPower | thecardsmith, the ONLY way we get that issue fixed is to conference the repair people and the install people and let them fight it out between themselves |
| 20:51.03 | thecardsmith | it's fun kickin' telephony tires :) ...my company is -finally- switching from Windows/Apex/Dialogic platform to a Linux/Asterisk/Digium platform, and it's a psych up... but I did get in trouble for spending too much time on the project right now XD |
| 20:51.08 | ManxPower | I am assuming incompetence rather than malice, but the customers are the ones that pay the price. |
| 20:51.25 | thecardsmith | "cock up before conspiracy" the saying goes |
| 20:51.40 | thecardsmith | it's usually the case, totally |
| 20:51.52 | thecardsmith | the only line telephony companies know: "it's the other end" |
| 20:52.37 | thecardsmith | alright, my weekend starts now... :) I'll be back! nice kickin' tires *thumbs up* |
| 20:53.25 | lirakis | is away: off to the pub |
| 20:53.49 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 21:07.46 | Katty | MARCO |
| 21:08.06 | carrar | POLO |
| 21:08.10 | Katty | carrar: <3 |
| 21:08.30 | carrar | POOL PARTY! |
| 21:08.36 | Katty | carrar: man i'm feelin punked |
| 21:08.45 | Katty | carrar: 5 needs to hurry up and get here already |
| 21:08.52 | carrar | COFFEE UP!! |
| 21:09.26 | carrar | Being that I am from Seattle I am drinking a latte as I type |
| 21:09.37 | carrar | fresh ground beans, home made espresso! |
| 21:11.34 | Katty | coffee :< |
| 21:11.41 | Katty | it tastes funny. |
| 21:12.33 | carrar | do what I do |
| 21:12.36 | carrar | add hazzlenut |
| 21:12.42 | carrar | and soy milk |
| 21:12.50 | Katty | :>>> |
| 21:12.55 | Katty | soy milk is win! |
| 21:13.04 | Katty | milk tastes fun :< |
| 21:13.15 | Katty | s/fun/funny/ |
| 21:13.30 | carrar | Milk can also be fun |
| 21:13.39 | Katty | how |
| 21:13.46 | carrar | like with oreos |
| 21:13.52 | Katty | soymilk is better with oreos |
| 21:13.55 | carrar | heh |
| 21:13.58 | carrar | just saying |
| 21:13.59 | carrar | I can be |
| 21:14.01 | carrar | it |
| 21:14.01 | Katty | k |
| 21:14.19 | carrar | Maybe milk has lost that loving feeling from you |
| 21:14.25 | carrar | which is ok |
| 21:14.28 | Katty | yes :< |
| 21:14.30 | Katty | it has :< |
| 21:14.40 | Katty | we are not very close these days. sniffle. |
| 21:14.45 | carrar | heh |
| 21:17.42 | Corydon76-dig | Katty: try almond milk? |
| 21:18.14 | Corydon76-dig | Doesn't have the gritty texture of soymilk |
| 21:21.51 | Katty | Corydon76-dig: i have not. |
| 21:21.57 | Katty | Corydon76-dig: not a huge fan of rice milk. |
| 21:22.09 | Katty | Corydon76-dig: could have been the brand i tried tho. i will see if i can find some almond milk. |
| 21:25.19 | *** join/#asterisk dandre (~daniel@ble59-2-81-56-122-47.fbx.proxad.net) |
| 21:31.12 | ManxPower | Katty, Almond Milk (at least the brand I tried) does not have much protein in it (unlike real milk or soymilk) |
| 21:33.50 | Katty | unfortunate. |
| 21:34.48 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 21:35.56 | Katty | hi fender bender. |
| 21:37.15 | [TK]D-Fender | Katty: Mew. |
| 21:40.20 | Corydon76-dig | ManxPower: if you're getting all your protein from milk, you have other problems |
| 21:42.14 | *** join/#asterisk bsaxon (~bsaxon@12.107.149.61) |
| 21:42.42 | ManxPower | Corydon76-dig, maybe I'm getting too much protein |
| 21:43.30 | Corydon76-dig | ManxPower: that actually is possible, especially if your protein intake is excluding carbohydrates |
| 21:44.02 | Corydon76-dig | Atkins diet increases your chances of dying prematurely |
| 21:44.49 | Corydon76-dig | Dr. Atkins is not around to defend his diet, as he died prematurely |
| 21:44.50 | *** join/#asterisk fofware (~fabian@186.124.148.94) |
| 21:45.35 | ManxPower | I'm not vegan or anything like that. I just think that eating dead animals is somewhat disgusting. |
| 21:46.02 | ManxPower | I do try to be aware of my protein intake. |
| 21:46.06 | Katty | i just think most meat tastes funny |
| 21:46.09 | [TK]D-Fender | LIVE is where it's at.... |
| 21:46.11 | Katty | tacos are good tho |
| 21:46.18 | Katty | and chicken strips |
| 21:46.20 | [TK]D-Fender | "It's not the kill.. it's the THRILL OF THE CHASE!!!!!!" |
| 21:46.22 | [TK]D-Fender | rocks out |
| 21:46.34 | ManxPower | Katty, as long as I can keep from remembering where it came from, I do occasionally eat meat. |
| 21:46.39 | Katty | [TK]D-Fender: alient ant farm, smooth criminal |
| 21:46.53 | Katty | [TK]D-Fender: what i'm listening to ;) |
| 21:47.44 | Corydon76-dig | ManxPower: you don't break a chicken's neck with your bare hands, then pluck and cook it yourself? |
| 21:48.00 | ManxPower | somehow I think there would be more vegetarians if everyone had to go on a tour of a working slaughterhouse in gradeschool. |
| 21:48.21 | Katty | probably, ManxPower |
| 21:48.33 | Katty | ManxPower: but i don't know. this country is very fond of cheap meat. |
| 21:48.38 | Corydon76-dig | only among kids who couldn't stomach it |
| 21:48.39 | [TK]D-Fender | Katty: I play "Movies" by AAF |
| 21:48.49 | ManxPower | Katty, they are welcome to eat all the meat they want. |
| 21:48.56 | Corydon76-dig | Other kids would slaughter their pet dogs |
| 21:49.29 | ManxPower | I did not say "everyone would be come vegetarian". |
| 21:49.40 | ManxPower | Corydon76-dig, my family slaughtered some of our own animals. |
| 21:49.44 | [TK]D-Fender | ManxPower: "you are what you eat" :p |
| 21:50.01 | Katty | omnomnom. |
| 21:50.01 | ManxPower | [TK]D-Fender, then I am a pizza. |
| 21:50.15 | Katty | i am.......not answer that one. |
| 21:50.20 | Katty | s/answer/answering/ |
| 21:50.39 | Corydon76-dig | Katty is a soybean? |
| 21:50.47 | Katty | Giggity. |
| 21:50.57 | Corydon76-dig | or just soylent green? |
| 21:50.58 | Katty | what is the dealio with today. |
| 21:51.04 | Katty | come on 5, hurry up |
| 21:52.23 | odenkos_ntbk | another problem at compile-time.. *sigh* .. http://pastebin.com/ZN2Pca3f any clues? |
| 21:52.28 | *** join/#asterisk jmacz (~jmacz@190.144.75.22) |
| 21:53.03 | ManxPower | odenkos_ntbk, what version of Asterisk |
| 21:53.12 | odenkos_ntbk | 1.6.2.13 |
| 21:53.23 | *** join/#asterisk Cain (~Geek@unaffiliated/cain) |
| 21:56.57 | *** join/#asterisk bsaxon (~bsaxon@12.107.149.61) |
| 22:00.05 | *** join/#asterisk fskrotzki_ (~fskrotzki@cpe-74-74-245-250.rochester.res.rr.com) |
| 22:01.08 | odenkos_ntbk | ManxPower: I'm clueless guess I should try again in the morning.. I thought I could compile it by midnight atleast.. |
| 22:02.26 | *** join/#asterisk netmax (~netmax@is.linux-administrator.com) |
| 22:05.44 | Nugget | Can you spot the IT guy? http://www.deansproperty.com.au/Home/Profiles No cheating! |
| 22:09.23 | odenkos_ntbk | ahh.. ok, good night |
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| 22:44.07 | kerx | Nugget, lol |
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| 23:08.33 | *** part/#asterisk ruben23 (~ITadmin@125.212.40.2) |
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| 23:44.52 | ariel_ | evening folks |
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