00:00.21 | asterboy | ok, I pointed at the gs site and the thing still won't upgrade. |
00:00.23 | bzbw | ok, here is the related config: http://pastebin.ca/52124 |
00:00.30 | asterboy | Where is the self destruct button. |
00:00.39 | asterboy | this crap is going to the garbage bin. |
00:00.46 | asterboy | fucking garbage |
00:00.57 | bzbw | I don't understand why it is not working:( |
00:01.25 | ghost99 | Manxpower: .. when you said that milliwatt thing how does that go ... I type in in at centos promt or CLI and it doesn't like it .... |
00:01.30 | bzbw | asterboy: what's the symptom on your upgrade, I happened to have a few GXPs. |
00:01.32 | [TK]D-Fender | exten => _2.,3,Dial(sip/BroadVoice/${EXTEN:1}) |
00:01.44 | dlynes | tainted-, prices have gone up since the last time i looked |
00:01.54 | asterboy | it just won't even initate an upgrade |
00:01.56 | bzbw | D-Fender: same thing, I change it to that, same story |
00:01.59 | ManxPower | ghost99, you REALLY need to read The Book |
00:02.04 | ManxPower | ~thebook |
00:02.06 | jbot | hmm... thebook is Asterisk: The Future of Telephony, released under a Creative Commons license and available at http://www.asteriskdocs.org << Read the book online! |
00:02.06 | asterboy | it's ignoring the files in my tftp root |
00:02.22 | [TK]D-Fender | bzbw : pastebin sip debug of a complete call attempt |
00:02.27 | asterboy | bzbw, what version do you have installed? |
00:02.37 | bzbw | asterboy: u have a trace file(in ethereal)? |
00:02.38 | ManxPower | ghost99, you can't run apps from the Asterisk CLI |
00:02.54 | dlynes | asterboy, what ftp server are you using? |
00:02.59 | dlynes | asterboy, erm tftp i mean |
00:03.03 | CunningPike | asterboy: I upgraded one once when were trialing them, and I think all I did was what you said |
00:03.26 | asterboy | ha |
00:03.33 | asterboy | I mean tftp-ha |
00:03.47 | dlynes | asterboy, yeah...you shouldn't have any problems with that one |
00:04.16 | asterboy | Is there a way to get more logging out of tftp-ha |
00:04.17 | asterboy | ? |
00:04.25 | dlynes | asterboy, is the tftp server's time synced to the same source as your polycoms? |
00:05.11 | dlynes | asterboy, you mean besides -vvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvv? |
00:05.14 | asterboy | yes |
00:05.26 | asterboy | no I mean with tftpd |
00:05.37 | dlynes | asterboy, yeah...specify more v's for more verbosity |
00:05.37 | bzbw | asterboy: I've heard that they are releasing the newer one soon, maybe next Tuesday:) |
00:05.48 | asterboy | I |
00:05.56 | asterboy | I'll be kicked out of here by then |
00:06.12 | asterboy | Besides it won't matter if I can't upgrade |
00:08.21 | dlynes | asterboy, one of the asterisk developers lives in Calgary |
00:08.29 | dlynes | asterboy, maybe he'll help you for a hefty fee? |
00:09.35 | bzbw | asterboy: call their LA office now, they are still working, I know a couple guys there can help a lot:) |
00:09.39 | dlynes | He used to live in Langley, but he moved to Calgary recently |
00:10.07 | dlynes | bzbw, He bought Polycoms as an end user |
00:10.50 | dlynes | asterboy, did you not buy the polycoms from a polycom channel partner? |
00:10.55 | asterboy | (626) 956 0260 |
00:11.01 | asterboy | nope, ebay |
00:11.09 | dlynes | ah...suckage |
00:11.23 | asterboy | lol, they have "PLease DIAL..." on there answer service. |
00:11.27 | asterboy | says it all |
00:11.33 | asterboy | who the hell DIALS anything |
00:12.51 | [TK]D-Fender | asterboy : Whats the problem with them now? |
00:13.02 | CunningPike | Won't upgrade, apparently |
00:13.06 | asterboy | yep |
00:13.19 | [TK]D-Fender | Polcom's? |
00:13.54 | CunningPike | You're about 5th in line [TK]D-Fender lol |
00:14.20 | [TK]D-Fender | TFTP doesn't upgrade based on file date IIRC.... |
00:14.37 | [TK]D-Fender | Part of why I always use FTP personally. |
00:14.55 | dlynes | [TK]D-Fender, ah...thought it might because my sipuras autoupgrades were flaky as hell, too |
00:15.20 | dlynes | not to mention sipura autoprovisioning |
00:15.34 | [TK]D-Fender | TFTP = slightly dumb. FTP=better |
00:16.30 | *** join/#asterisk franck (n=franck@tikiwiki/franck) |
00:16.30 | asterboy | ooooo...Grandstream is giving me the latest firmware! |
00:16.35 | CunningPike | [TK]D-Fender: Correct about tftp |
00:16.38 | asterboy | and yes ftp is cooler |
00:16.54 | asterboy | but no choice with GXP |
00:16.55 | dlynes | polycoms don't do https? |
00:17.10 | [TK]D-Fender | dlynes : Certainly they can. |
00:17.22 | dlynes | just ftp is easier to set up? |
00:17.59 | [TK]D-Fender | dlynes : That too. |
00:18.20 | franck | Hi all |
00:19.00 | franck | I'd like to make a script that sends me an e-mail each time a SIP phone register outside a predefined range of IP addresses, How to? |
00:22.26 | Vahram | Anybody can help me with asterisk development, i whant to patch chan_h232.c to set variable tat will have Q931cause |
00:22.53 | Vahram | actyally i need only know the puction that is putting channel vars |
00:23.08 | Vahram | * |
00:24.04 | justinu | good afternoon #asterisk |
00:24.43 | Vahram | I guess everybody is sleepin)) |
00:33.00 | *** join/#asterisk MikeJ[Laptop] (n=vircuser@64.241.37.140) |
00:34.35 | CunningPike | Well, it's been a slice |
00:36.05 | *** join/#asterisk De_Mon (n=de_mon@fl-69-69-140-189.dyn.sprint-hsd.net) |
00:37.01 | *** join/#asterisk BadPacket (n=BadPacke@unaffiliated/badpacket) |
00:39.53 | bzbw | asterboy: did you get the new firmware?:) |
00:39.58 | asterboy | yep |
00:40.04 | bzbw | what was it? |
00:40.06 | asterboy | latest greatest and it is far more impressive |
00:40.14 | bzbw | what version is it? |
00:40.40 | asterboy | 1.1.0.10 |
00:41.04 | *** join/#asterisk Thazza (n=me@229.9.233.220.exetel.com.au) |
00:41.05 | asterboy | and Bootloader 1.1.0.1 |
00:41.12 | bzbw | man, it's almost the latest one, but they said it should be 1.1.0.11:) |
00:41.18 | *** join/#asterisk yxa (n=diablo@58.185.90.101) |
00:41.19 | asterboy | the menu is the way it should be now. |
00:41.34 | asterboy | let me know if you want it? |
00:41.38 | *** join/#asterisk marv (n=marv@12-219-145-181.client.mchsi.com) |
00:42.17 | bzbw | Is it working? |
00:44.37 | bzbw | k, my issue is yet to resolved:( |
00:45.02 | asterboy | haven't tested yet |
00:45.12 | Thazza | anyone got a Sipura SPA-3000 working without echo on the FXO? |
00:45.31 | bzbw | man it's Dial(SIP/${EXTEN:1}@BroadVoice), it just did NOT resolve to the right host:(!!! |
00:46.59 | bzbw | anyone knows how to step through * for debugging? Into the Context translation level? |
00:47.24 | asterboy | yippeee!!! |
00:47.35 | asterboy | the phone was causing the transfer problem |
00:47.42 | asterboy | new upgrade fixed it. |
00:48.09 | asterboy | I have a natural knack for finding those problems no one else finds...drive me nuts |
00:50.25 | DoktorGreg | You just need to think more like everyone else:P |
00:50.35 | [TK]D-Fender | asterboy : no, plenty of other people got chumped into buying GS' :D |
00:51.25 | asterboy | well, this is the first time I caved in cause the client couldn't afford the Polycom |
00:51.31 | asterboy | so now I' |
00:51.40 | asterboy | I'm the one who pays in the end. |
00:52.23 | *** join/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
00:52.47 | paolob | Hi guys! How do I put a call on hold? is there a special key? |
00:53.01 | [TK]D-Fender | paolob : depends on your phone. What do you have? |
00:53.05 | De_Mon | asterboy so ture |
00:53.20 | paolob | [TK]D-Fender, normal phones connected to pap2 |
00:53.33 | asterboy | more grey hair....just great. |
00:53.51 | [TK]D-Fender | paolob : then its up to the PAP2 to put the call on "hold" there may be a flash + *code feature in the ATA for that. |
00:53.54 | asterboy | they'll be pilling dirt on my early at this rate |
00:54.20 | paolob | [TK]D-Fender, doesn't asterisk use a special key, like # to transfer calls? |
00:54.23 | [TK]D-Fender | paolob : The SPA series from them has a code for it, but not sure about the PAP2 specifically. |
00:55.05 | [TK]D-Fender | paolob : # is just a way so that shit phones and analog devices which don't have special signalling capabilities can tell # of its intent. |
00:55.09 | [TK]D-Fender | * |
00:57.24 | paolob | [TK]D-Fender, explain better, I don't understand... |
00:58.30 | jake1932 | # also works on cell phones |
00:59.05 | [TK]D-Fender | Real SIP phone had hold/transfer/conference buttons, etc. just like digital PBX phones. You are plugging a DUMB analog phone int an ATA. So you need to tell the ATA to put the call on hold with the only tools at your disposal : Flash and DTMF. |
00:59.21 | [TK]D-Fender | Now thats assuming the PAP even OFFERS you the capability of telling it to do that. |
00:59.28 | sevard | http://video.google.com/videoplay?docid=-3857855347623051125 |
00:59.38 | [TK]D-Fender | jake1932 : to transfer calls? |
00:59.43 | jake1932 | yes |
00:59.53 | [TK]D-Fender | jake1932 : I presume you mean coming in on a Zap cahnnel of course... |
01:00.06 | [TK]D-Fender | jake1932 : Either way its an * mechanism. |
01:00.09 | jake1932 | well sort of - VIOP term to cell phone |
01:00.10 | *** join/#asterisk Abydos313 (i=abydos31@adsl-71-129-51-118.dsl.irvnca.pacbell.net) |
01:00.49 | *** join/#asterisk wunderkin (i=kev@ip68-226-113-228.ph.ph.cox.net) |
01:00.57 | jake1932 | just wanted to point out that it's more than just crappy phones and analog |
01:01.24 | *** join/#asterisk mbrooks (n=mbrooks@gateway.digium.com) |
01:01.36 | jake1932 | it's anything except IP phones connected directly to asterisk |
01:02.40 | *** join/#asterisk Abydos313 (i=abydos31@adsl-71-129-51-118.dsl.irvnca.pacbell.net) |
01:06.12 | bzbw | asterboy: u can't transfer the call with new firmware? |
01:06.32 | *** part/#asterisk franck (n=franck@tikiwiki/franck) |
01:07.05 | DoktorGreg | anybody know of a way to get adobe illustrator out of this danged MDI interface? |
01:07.14 | bzbw | speaking of video, anyone knows whether 1.2.7.1 support H.264? |
01:07.28 | *** join/#asterisk zeppelin__ (n=zeppelin@201.66.166.237) |
01:08.56 | skydog | sevard: pretty cool stuff |
01:09.48 | jake1932 | yep - i could be at two places at once finally |
01:09.56 | sevard | skydog: speculation that it's what is inside of the nintendo revolution |
01:09.59 | ManxPower | bzbw, Did any other 1.2x support it? |
01:10.17 | skydog | cool |
01:10.25 | sevard | s/it\'s/is/g |
01:10.47 | sevard | skydog: speculation that it's what is inside of the nintendo revolution |
01:10.51 | sevard | s/it's/is/g |
01:11.00 | sevard | :| |
01:11.03 | paolob | Guys, can I use a variable this way in extensions.conf : exten => ${VAR},1,Dial(${VAR},20,Tt) ? |
01:11.13 | sevard | i'm tired. |
01:11.32 | [TK]D-Fender | paolob : No, you can't use a variable as an EXTEN, only a CONSTANT. |
01:11.39 | skydog | anybody running asterisk as non root? |
01:11.56 | jake1932 | paolob: use Goto |
01:12.04 | sevard | skydog: i am |
01:13.05 | skydog | I think ive got it...but the start up is simply starting it by - su asterisk -c /usr/sbin/safe_asterisk or dropping into asterisk to start? |
01:13.45 | ManxPower | skydog, read the wiki page on running Asterisk as non-root |
01:13.57 | skydog | yep im on it now.. |
01:14.09 | bzbw | ManxPower: Mark told me at one point it does:( |
01:14.13 | sevard | skydog: I thought safe_asterisk was a script that also ran asterisk as a different user, i don't know. i don't use that script. |
01:14.28 | ManxPower | bzbw, 1.2 gets no NEW features. |
01:14.35 | skydog | asterisk -U asterisk -G asterisk ? maybe?? |
01:14.39 | ManxPower | if it's always been a 1.2 feature, then the latest should have it. |
01:14.52 | sevard | skydog: man su |
01:15.02 | ManxPower | skydog, yes, but then you also need to change the logging permissions, the zaptel device permissions, the database permissions. READ THE WIKI PAGE |
01:15.03 | sevard | and man sudo |
01:15.19 | skydog | http://www.voip-info.org/wiki/index.php?page=Asterisk+non-root is what im referring to. |
01:15.24 | ManxPower | you do NOT need to sudo or su. safe_asterisk can be told to run asterisk as non-root |
01:15.39 | ManxPower | skydog, if those instructions don't work, get back to us. |
01:16.01 | paolob | [TK]D-Fender, how do I define a CONSTANT? |
01:16.12 | skydog | good enough..:) ..so far it has been successful minus these parts - |
01:16.15 | paolob | and how do I reference it? |
01:16.38 | skydog | no such location for /usr/local/share/asterisk .. |
01:16.46 | *** join/#asterisk MacDome (n=eseidel@c-71-198-177-144.hsd1.ca.comcast.net) |
01:16.48 | skydog | on centos install that is. |
01:17.02 | [TK]D-Fender | paolob : What are you really trying to DO? |
01:17.22 | skydog | and im guessing since ztdummy is installed /dev/zap doesnt exist either...right? |
01:17.44 | *** part/#asterisk poisoner (i=poisoner@tauceti.poisoner.de) |
01:18.01 | paolob | [TK]D-Fender, I'm trying to write extensions.conf in a way that permit me change the ext. number without changing all the occurrences of each ext. in extensions.conf |
01:18.19 | [TK]D-Fender | paolob : And what is this an exten TO? |
01:19.07 | skydog | perhaps the use of macros will help? with the extensions ?? |
01:20.05 | paolob | [TK]D-Fender, for example, I define 701 as my phone number, but I want to have the freedom to change "my" phone to 702, and I want to write extensions.conf in a transparent way |
01:20.35 | paolob | in a way that a change 701 to 702 only once, and I reload the confs, and all is ok |
01:21.02 | [TK]D-Fender | paolob : how big a PBX setup are we talking about? Thi just doesn't seem worth it.... |
01:21.21 | paolob | [TK]D-Fender, 30 extensions |
01:21.37 | [TK]D-Fender | paolob : Not worth it.... |
01:21.44 | [TK]D-Fender | seriously |
01:22.04 | *** join/#asterisk miguel3239 (n=chatzill@ns1.nashuacs.com) |
01:22.11 | paolob | but how do I define a constant in extensions.conf ? |
01:22.33 | paolob | MYPHONE=701 ? |
01:22.45 | paolob | MYPHON E => 701 ? |
01:22.53 | [TK]D-Fender | paolob : look in the sample and check the WIKI |
01:23.04 | paolob | [TK]D-Fender, ok, thank you! |
01:23.15 | [TK]D-Fender | paolob : but that looks about right. |
01:25.46 | natmlt | paolob in the book "Aterisk, The Future of Telephony" that you can download for free, check out Global Variables on page 91 |
01:25.55 | jake1932 | you could do 701,1,Goto(myphone,1) |
01:26.08 | jake1932 | then have myphone,1,Blah |
01:26.16 | paolob | natmlt, but are constant only predefined, or can i define my own ones? |
01:27.18 | natmlt | paolob you can define your own |
01:27.19 | paolob | natmlt, how? |
01:27.19 | natmlt | paolob their example JOHN=zap/1 for dialing a user John over zap/1 |
01:27.19 | wunderkin | must be experiencing packet loss |
01:27.34 | paolob | natmlt, and how do I reference that constant? |
01:27.47 | natmlt | Put it in the context [globals] |
01:27.57 | natmlt | it is reserved strictly for global variables |
01:28.15 | natmlt | and the will apply to all contexts in extensions.conf |
01:28.31 | natmlt | You should download the book |
01:28.38 | natmlt | it will help a bunch |
01:29.03 | [TK]D-Fender | jake1932 : That sample is HORRIBLY wrong... |
01:29.24 | natmlt | http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
01:29.30 | [TK]D-Fender | natmlt : much better sample... |
01:30.09 | [TK]D-Fender | paolob : per natmlt's sample it'd be - exten => ${JOHN},1,Dial(SIP/myphone,30) |
01:30.11 | jake1932 | hmm - should work |
01:30.19 | jake1932 | lemme try |
01:30.26 | skydog | another good write up on configuring * as non-root : http://www.dynx.net/ASTERISK/AMP/INSTALL |
01:30.32 | [TK]D-Fender | jake1932 : Yes it'd work, but thats got nothing to do with "constants" |
01:30.53 | jake1932 | no - but it solves what he asked originally - trying to alias extensions |
01:31.22 | [TK]D-Fender | jake1932 : not to alias an exten, then he'd have to change the "701" in a ton of places. |
01:31.45 | jake1932 | not true |
01:31.52 | jake1932 | only in the goto statement |
01:33.28 | jake1932 | no? |
01:34.13 | [TK]D-Fender | jak he wants to change the very fact of "701" leading to his phone. and when he has 10 occurences of 701 ledaing to him, what would he do to change it to 702? |
01:34.45 | jake1932 | it would be myphone leading to him |
01:34.50 | jake1932 | 701 would be the alias |
01:34.56 | skydog | probably a little complex, but he could put his stuf in a database and change it that way...just a thought..but that would make things just a hell of alot more complicated for 30 callers... |
01:35.11 | jake1932 | realtime would be overkill |
01:35.17 | skydog | agreed |
01:35.33 | skydog | but again a solution of sorts.. |
01:35.42 | jake1932 | yes |
01:36.01 | [TK]D-Fender | jake1932 : he's talking about have many "exten => 701,1," all over his dialplan. He wants to replace ALL OCCURENCES of 701 to something else. You are talking about changing that action taken by 701, he want 701 ITSELF to change. |
01:36.38 | jake1932 | ok - i'm going to reread what he asked for |
01:37.24 | jake1932 | <PROTECTED> |
01:37.26 | skydog | well it is flat file and he could simply grep the file out and change the occurences of 701 wih one shot...but im truly confused now as to what the original requirement was...lol |
01:37.31 | [TK]D-Fender | jake1932 : Mind you I can hardly imagine why he'd have more that 2 references to a given "phone extension number". I always do cascaded contexts which inherit everything... |
01:38.23 | [TK]D-Fender | jake1932 : and the exten # is 701, not the action it should take on being dialed. |
01:38.41 | skydog | thanks to peeps for the reference of the non-root install works like butta! ;) |
01:39.48 | *** join/#asterisk OloBola (n=not@netblock-68-183-67-158.dslextreme.com) |
01:40.31 | jake1932 | regarding more than 1 reference, isn't that what you're doing by defining MYEXTEN anyways? |
01:40.39 | jake1932 | as a constant |
01:42.07 | jake1932 | he's probably not even here anymore |
01:42.08 | jake1932 | :) |
01:43.13 | [TK]D-Fender | but you only have to change that constant at the top ONCE. |
01:43.27 | [TK]D-Fender | And all 10 times it appears follow along... |
01:43.46 | jake1932 | you're saying with my example you'd have to change it more than once? |
01:44.37 | [TK]D-Fender | seet this? <jake1932> you could do 701,1,Goto(myphone,1) |
01:44.52 | jake1932 | yepp - that's exactly what i have |
01:44.57 | [TK]D-Fender | he's saying he'll have lots of 701's all over the place. its a FIXED value there! |
01:45.08 | jake1932 | oh |
01:45.14 | jake1932 | i think it follow you |
01:45.15 | [TK]D-Fender | and when he wants to change them ALL to 702?! |
01:45.32 | jake1932 | he would change them to myexten (not 702) |
01:45.47 | jake1932 | i'm going to pastebin |
01:46.44 | *** join/#asterisk nortex (n=breeves@adsl-69-149-172-106.dsl.amrltx.swbell.net) |
01:47.21 | [TK]D-Fender | he wants to do : exten => ${MYPHONE},1,Dial(Zap/1,20) and so on in 50 differnt places and for the MOMENT use 701 and be able to change his mind later and not have to change 50 lines.... just the constant declared at the top. Getting it now? |
01:47.38 | jake1932 | yes |
01:47.56 | jake1932 | http://pastebin.ca/52153 |
01:48.57 | *** join/#asterisk MacDome (n=eseidel@c-71-198-177-144.hsd1.ca.comcast.net) |
01:50.20 | [TK]D-Fender | jake1932 : Thats nice. but he'll have 100 occurances of the FIRST LINE like 701 throughout his plan. Do you get it? Each one would have to be changed. |
01:50.40 | jake1932 | 30 |
01:50.53 | jake1932 | but i don't think there's any way around that |
01:51.03 | jake1932 | you need to specify it somewhere |
01:51.13 | [TK]D-Fender | not 30. FOR 30 people with potentially multiple occurences of each throughout his plan. |
01:51.33 | [TK]D-Fender | So 30 * # of occurences |
01:51.51 | jake1932 | somewhere, whether you use a constant, or the method i put out there, you need to specify what myexten points to |
01:52.59 | jake1932 | now - if he goes and changes his phone user ids, vm boxes, etc |
01:53.06 | jake1932 | then - that's a different issue |
01:53.12 | [TK]D-Fender | jake1932 : Your's has 701 HARD CODED. Do you get it? thats not a universally updateable CONSTANT. if he puts a 701 in 10 contexts he'll have to chage all 10. if he uses a constant he need only chage the constant declaration up top. |
01:53.40 | ManxPower | [TK]D-Fender, so basically he wants to create a complex, hard to understand dialplan just so he doesn't have to put in extensions. |
01:54.02 | [TK]D-Fender | ManxPower : Quite possibly, but that wasn't my point :) |
01:54.33 | *** join/#asterisk konfuzed (n=Konf@H211.C18.B96.tor.eicat.ca) |
01:54.33 | ManxPower | Dialplans are complex, no matter how hard you try to make them simple. You soon find that simple does not work in the real world. |
01:54.43 | [TK]D-Fender | ManxPower : just trying to get across that he wants the EXTENSION modified globally, not the APPLICATION called NOR its PARAMETERS. |
01:55.03 | [TK]D-Fender | ManxPower : Sure it can work, its just simply "not worth it" |
01:55.23 | ManxPower | [TK]D-Fender, I know. Totally stupid thing to do and may or may not work in the current verison of asterisk but could EASILY not work in future releases. |
01:55.34 | [TK]D-Fender | ManxPower : My dialplans are DAMN clean things as are the macros and other methods that support it. |
01:55.53 | jake1932 | mine too |
01:55.55 | ManxPower | [TK]D-Fender, mine are CLEAN, but not SIMPLE. |
01:55.56 | [TK]D-Fender | ManxPower : SHHH!!!! Don't destroy his hopes! Let him "live for the moment" |
01:56.13 | bzbw | hmm, I've been troubleshooting with D-Fender on this, looks like broadvoice now require * to use "sip.broadvoice.com" in From, To and Contact header, if not, it will reject the call:( |
01:56.15 | *** join/#asterisk bjohnson (n=bjohnson@i216-58-59-27.cybersurf.com) |
01:56.53 | [TK]D-Fender | bzbw : Not really, thats copied right from a client using it right now.... |
01:57.05 | [TK]D-Fender | bzbw : PM |
01:57.21 | jake1932 | "if he puts a 701 in 10 contexts he'll have to chage all 10" - nowhere did i see talk about multiple contexts - i can see what you're saying on that |
01:57.48 | [TK]D-Fender | <jake1932> paolob: [TK]D-Fender, I'm trying to write extensions.conf in a way that permit me change the ext. number without changing all the occurrences of each ext. in extensions.conf |
01:58.00 | [TK]D-Fender | 701 is the ext, not the ACTION it would take. |
01:58.20 | [TK]D-Fender | therefor this statement implies multiple occurances of 701 <- |
01:58.51 | jake1932 | 701,1 701,2 ,etc |
01:58.55 | bzbw | D-Fender: I compare the sip registration, which uses all "sip.broadvoice.com" in all 3 headers with the Invite, registration goes through, but invite gotten 604. |
01:58.56 | jake1932 | maybe i read it wrong |
01:59.18 | [TK]D-Fender | jake1932 : those are PRIORITIES of ext 701 in a given context..... |
01:59.42 | jake1932 | paolob: are you still here? |
01:59.54 | [TK]D-Fender | bzbw : In what you pastebin'd for me I never saw a reject, it kept RETRYING only. |
02:00.03 | paolob | jake1932, yes |
02:00.04 | De_Mon | I need help using dbg to track down this codec 128 WARNING I keep getting... I've connected gdb to the asterisk process, now what? |
02:00.32 | bzbw | D-Fender: I'll take your advise and try more, got to go now, thanks anyway:) |
02:00.34 | jake1932 | paolob: were you using multiple contexts in which you need the dynamic extensions? |
02:01.34 | jake1932 | or were all your extensions in a single context? |
02:01.36 | paolob | jake1932, yes, and it was for a matter of trasparency: someone that would have to modify asterisk configuration should understand clearly what are those ext. numbers referring to |
02:03.30 | jake1932 | but that was a fun debate! |
02:04.03 | ManxPower | nobody needs dynamic extensions |
02:04.10 | paolob | guys, have config files changed their syntax from 1.0.9 to 1.2? |
02:04.21 | Qwell | paolob: some, yes, of course |
02:04.45 | tainted- | Qwell! |
02:04.52 | paolob | Qwell, if I copy a 1.0.9 sip and extensions.conf to a 1.2, what should I change? |
02:05.02 | Qwell | paolob: Everything that the README files says to change |
02:05.13 | Qwell | everything that the CHANGES file says to change |
02:05.21 | paolob | Qwell, ok, thank |
02:05.27 | Qwell | Everything that was discussed on the various mailing lists, and forums, and emails, etc |
02:05.40 | jake1932 | yes - read it all |
02:05.43 | jake1932 | haha |
02:05.55 | jake1932 | there will be a wuiz |
02:05.58 | jake1932 | qiz |
02:06.00 | jake1932 | ah |
02:06.02 | Qwell | You fail |
02:06.25 | jake1932 | should've paid attention in typing class |
02:06.42 | [TK]D-Fender | ManxPower : I should remember a timeless addage "Arguing on the internet is like running in the Special Olympics. Even if you win, you're still a RETARD" |
02:07.04 | ManxPower | [TK]D-Fender, hence my extensive /ignore list |
02:07.31 | [TK]D-Fender | ManxPower : Fortunately I don't get repeat offenders..... |
02:07.43 | jake1932 | i wasn't trolling |
02:08.02 | [TK]D-Fender | jake1932 : Buy you we're caught ; hook, line, and sinker! |
02:08.08 | dlynes | Does anyone happen to have a better alternative to Text::CSV_XS (Perl) for importing old asterisk csv files? |
02:08.19 | dlynes | I've encountered some csv lines iwht binary data in them |
02:08.48 | [TK]D-Fender | jake1932 : Wow I typed that out SO wrong.. hehe |
02:08.52 | [TK]D-Fender | its getting late... |
02:08.53 | *** join/#asterisk MikeJ[Laptop] (n=vircuser@d14-69-8-30.try.wideopenwest.com) |
02:09.01 | *** join/#asterisk Splas (n=jwb@brooklyn.paravolve.net) |
02:09.01 | jake1932 | yes - see it's catchy |
02:09.10 | [TK]D-Fender | adfgafgsdfhgkljhsjhfdkjtew;klnrg |
02:09.17 | [TK]D-Fender | *blarg* |
02:11.26 | [TK]D-Fender | ManxPower : 501 display clearly isn't as nice as the 601, but still better than pretty much everything else except a Cisco 7940+ |
02:11.46 | [TK]D-Fender | I'm betting its probaby close to tie with Cisco. |
02:11.54 | [TK]D-Fender | Wish I had one for testing... |
02:12.08 | jake1932 | is there a backlight on the +? |
02:12.25 | [TK]D-Fender | jake1932 : is no "+" i mean that model and higher |
02:12.33 | jake1932 | ok |
02:12.40 | [TK]D-Fender | 7960, etc |
02:12.44 | jake1932 | right |
02:12.55 | jake1932 | how come they couldn't fit a backlight on these phones |
02:12.56 | jake1932 | ? |
02:13.26 | jake1932 | i even have a cheapo 9417 nortel with one |
02:14.02 | [TK]D-Fender | jake1932 : Dunno.... its amongst the most requested features. Maybe they'll all wake up soon and pay attention. |
02:14.13 | jake1932 | let's hope |
02:14.54 | [TK]D-Fender | jake1932 : the Aastra 480i has a backlight but a chacter based display, not pixel. Pretty much ever phone has a clearly counterbalancing factor against it. |
02:15.26 | jake1932 | how much would a backlight up the phone cost though? |
02:15.35 | *** join/#asterisk Tusker (n=tusker@203.117.94.152) |
02:16.17 | [TK]D-Fender | Grandstream GXP-2000 is the PERFECT example of this. Has back-light, multiple keys for presence PoE and more for $85USD! great value, right? NO! Shit speakerphone and handset. NASTY echo... ona SIP PHONE!! Flakey ass firmware. FEELS like something fit for Ken & Barbie... |
02:16.29 | jake1932 | haha |
02:16.41 | [TK]D-Fender | jake1932 : Probably not too much more... they just cut corners thinking the market they were targeting. |
02:17.08 | jake1932 | the aastra is a business phone though |
02:17.19 | jake1932 | same market |
02:17.33 | [TK]D-Fender | I'd HAPPILY pay a bit more for Full PoE and backlight options on all Polycom phones. |
02:17.58 | jake1932 | well even with that - i have a 7960 on my desk - because of the general quality of the phone |
02:18.00 | DoktorGreg | wooo hooo the pvr is done transcoding and commercial skipping doktor who! |
02:18.02 | [TK]D-Fender | jake1932 : Yewah, but they just built that phone off their old analog model..... just changed the board. not a "new" phone like all the others designed. |
02:18.27 | jake1932 | right |
02:18.51 | jake1932 | have you used the aastra phone before? |
02:19.29 | Tusker | heya guys... sorry for such a newby question, but I was wondering if it is possible to call one #, and then dial the requested number, within a dialplan/extension... ie, like when you dial using a normal modem, and you have to put the , (delay) there... imagine a "normal" key or pbx system, which asks "please dial the extension now" |
02:20.28 | [TK]D-Fender | jake1932 : A little, yes. |
02:20.56 | jake1932 | how does it compare with the polycom? |
02:21.01 | [TK]D-Fender | Tusker : Sure you can. |
02:21.04 | jake1932 | or cisco? |
02:21.42 | Tusker | what "feature" is this called? everything I search on, such as extension, refers to asterisk internal extension :) |
02:21.55 | [TK]D-Fender | jake1932 : Only advantage to Aastra is the backlight. Polycom wins on configurability, LCD usability (backlight aside), and physical.audio quality |
02:22.31 | jake1932 | can you get polycom support through normal channels without spending a fortune? |
02:22.47 | jake1932 | with a small number of phones? |
02:22.48 | [TK]D-Fender | jake1932 : never needed support :) |
02:23.06 | jake1932 | i think they come with SIP loaded - right? |
02:23.07 | [TK]D-Fender | jake1932 : you get it from your reseller typicall, and cost is just fine. |
02:23.27 | wunderkin | [TK]D-Fender = polycom support |
02:23.51 | wunderkin | and sales |
02:23.57 | [TK]D-Fender | wunderkin : wel I AM planning on getting cert'd :) |
02:24.07 | wunderkin | i know, what a surprise |
02:24.23 | Tusker | [TK]D-Fender: can you share your wisdom on the way to write the "dialplan" ? |
02:24.28 | [TK]D-Fender | and NO, I don't sell them :) I'm just promoting what I honestly feels is a WORTHY product, and people of repute agree. |
02:24.38 | wunderkin | promoter, sorry |
02:24.44 | [TK]D-Fender | Tusker : depends what you're trying to acheive. |
02:25.47 | [TK]D-Fender | Polycom IP 501 = $170, Aastra 480i = $200+. |
02:26.48 | jake1932 | with SIP? |
02:27.04 | [TK]D-Fender | jake1932 : Yes, both do. |
02:27.11 | *** join/#asterisk MoutaPT (n=MoutaPT@85.139.183.56) |
02:27.21 | Tusker | well, it's a sip account... so, I'd like to put it as one of the outgoing connections in asterisk... so if I connect to my asterisk account, and dial say, 00 xxxx xxxx, so asterisk takes that, dials 55,,,xxxx xxxx |
02:27.23 | [TK]D-Fender | jake1932 : They can also be flashed with MGCP (both brands) |
02:27.23 | jake1932 | "SIP Software - Certified VoIP Resellers can download from the Polycom Resource Center (PRC)." |
02:27.31 | jake1932 | that line is a little scary |
02:27.38 | MoutaPT | Hi does any one could advice me how to have incoming calls pop up with SugarCRM |
02:27.48 | MoutaPT | I'm planing an app for call center |
02:27.53 | [TK]D-Fender | Tusker : no need for pauses in dialing sip #'s |
02:28.21 | Tusker | the sip service has a redirect call service |
02:28.26 | [TK]D-Fender | Tusker : you woul dimsple prefix the # and route based on the prefix, filtering it off and adding any other digits as required. |
02:28.43 | Tusker | any number I dial directly, before it answers, goes to the voice prompt |
02:28.55 | Tusker | once it answers, then I can dial the destination number |
02:28.56 | [TK]D-Fender | Tusker : hmmm so you want it to use thier timed interface to do changes? |
02:29.06 | jake1932 | like a LD pin code? |
02:29.16 | [TK]D-Fender | Tusker : Sounds like a pain in the ass.... |
02:29.28 | [TK]D-Fender | jake1932 : Yeah, sounds like that kind of style... |
02:29.41 | Tusker | yeah, it is a bit of a pain... they have their LD service running on the same server it seems |
02:30.20 | jake1932 | you can't just get the extensions authorized - i wouldn't rely on sending tones afterwards |
02:30.46 | jake1932 | only plan Z |
02:30.46 | [TK]D-Fender | I'm not sure how to do an out-bound macro like that... I know you can do pauses in dialing out an ANALOG interface, but your situation is outside of my experience. |
02:31.17 | Tusker | ok, I'll have a look at out-bound macro then, cheers! |
02:35.22 | [TK]D-Fender | Tusker : Take a look at the m() parameter of Dial. Might do what you're looking for if you time it right. Use along with a command to send DTMF. |
02:35.25 | MoutaPT | Hi does any one could advice me how to have incoming calls pop up with SugarCRM?? |
02:35.57 | [TK]D-Fender | MoutaPT : First questio to ask yorself : how to identify the caller.... |
02:36.03 | *** join/#asterisk Spy000007 (n=Spy007@ool-44c045b0.dyn.optonline.net) |
02:36.11 | [TK]D-Fender | can't type again.. sheesh |
02:36.34 | Tusker | ok, cool, cheers |
02:36.38 | jake1932 | it's pseudo spanish |
02:37.02 | Spy000007 | anyone use SOAP (XML) and asterisk? |
02:37.31 | MoutaPT | based on RT request ticket id |
02:37.34 | MoutaPT | or caller id |
02:38.21 | MoutaPT | i may set caller id based on DTM from caller or simply use Request ticket id to forward to SugarCRM and pop up this call, don't know if it is easy... |
02:38.33 | MoutaPT | dTM=DTMF |
02:39.16 | MoutaPT | No body has done this before? |
02:39.23 | Spy000007 | no one? everyone still using perl to hack together sip.conf? |
02:39.38 | MoutaPT | what is more usual for call centers? Agents need an app to handle calls... |
02:39.39 | [TK]D-Fender | MoutaPT : I'd say have them enter the identfying field with a Read, then have a PC call a URL based on it. |
02:39.54 | MoutaPT | like YACC? |
02:40.05 | [TK]D-Fender | MoutaPT : Depends on what you can call to do something usefull. |
02:40.16 | [TK]D-Fender | Don;'t know Yacc offhand... |
02:40.53 | MoutaPT | YACC throws an URL with some parameters from asterisk call, like caller ID and UNIQUEID |
02:41.07 | MoutaPT | i was looking for a complete solution already |
02:41.17 | MoutaPT | even if it is not opensource |
02:42.11 | [TK]D-Fender | MoutaPT : Sorry, such a thing should be Google-able |
02:42.32 | Spy000007 | MoutaPT: doesn't fonality have some sort of crm integration? |
02:42.37 | *** join/#asterisk asterboy (n=kevin@S010600485480f4be.ed.shawcable.net) |
02:42.48 | MoutaPT | i'm going to look for it |
02:43.01 | MoutaPT | i've been googling SIP CRM phone |
02:43.06 | MoutaPT | Sip crm calls |
02:43.11 | MoutaPT | nothing usefull |
02:44.35 | Spy000007 | MoutaPT: see www.fonality.com -- it mentions crm integration... but says salesforce.com, not sugarcrm |
02:44.48 | MoutaPT | thank you |
02:44.49 | Spy000007 | MoutaPT: it's only an example though, so maybe it supports more than one |
02:44.58 | MoutaPT | i'm there already |
02:45.00 | MoutaPT | :) |
02:45.44 | asterboy | gotta give Grandstream an A+ for customer tech support. |
02:45.53 | asterboy | Big fat F for Polycom |
02:45.55 | Spy000007 | actually it seems like there's a guy who's on the board of directors for both sugarcrm and fonality, so i'm sure integration is on the way |
02:46.10 | [TK]D-Fender | MoutaPT : What kind of URL could you call with a ticket # or similar to pull up something sueful in a browser? |
02:47.09 | [TK]D-Fender | asterboy : Grandstream greatly supporst a crappy product, Polycom crappily supports a great product. And thus the Universe's balance is restored! |
02:47.17 | asterboy | lol |
02:47.22 | asterboy | that is exactly it |
02:47.24 | file | universe... balanced... |
02:47.25 | file | never! |
02:47.30 | [TK]D-Fender | asterboy : Whats your Polycom issue now? |
02:47.50 | asterboy | none, just that when I tried to get tech support, the buck was passed. |
02:47.56 | [TK]D-Fender | file : My karma ran over your dogma :D |
02:47.58 | MoutaPT | i may have a cgi runing with ticket number argument... so all the customer info is there... |
02:48.09 | file | muahahahaha |
02:48.12 | [TK]D-Fender | asterboy : Fine, but do you actually still have a problem right now? |
02:48.15 | asterboy | two rings, 1 push of a button and Grandstream tech support HUMAN was online. |
02:48.27 | asterboy | not that I know of. |
02:48.33 | [TK]D-Fender | file : I've banked so many here you can't faze me :) |
02:48.39 | asterboy | The grandstream upgrade fixed the polycom issue. |
02:48.50 | asterboy | err...stopped screwing up the polycom |
02:48.52 | file | this is true |
02:49.03 | file | next time I'm in Montreal, breakfast/lunch/dinner/whatever's on me! |
02:49.16 | [TK]D-Fender | asterboy : Sounds more like GS had a problem and its fixed until something ELSE breaks :) More like your Polycom's REPORTED that your GS' suck :) |
02:49.18 | asterboy | Montreal is beautiful |
02:49.28 | asterboy | lol |
02:49.33 | asterboy | yep that was it. |
02:49.59 | asterboy | If this upgrade works though, the Grandstream will be a good phone. |
02:50.11 | [TK]D-Fender | file : just get yourself up here, we'll grab JunK-Y and go for a beer. No debts between us! |
02:50.15 | jake1932 | ERR: Cheapo Phone detected on network - Aborting! |
02:50.27 | file | :D |
02:50.30 | asterboy | Montreal is a great place for beer and food. |
02:50.34 | [TK]D-Fender | asterboy : I preffer the term "less shit" where GS is concerned ;) |
02:50.42 | asterboy | true |
02:51.04 | asterboy | I won't be selling them |
02:51.49 | [TK]D-Fender | Imagine I payed for my own IP 301 & IP 501 for home. Thats about $330 + tax CDN. Thats the price of 3 GXP's, and I'm very glad to have payed more. |
02:52.12 | asterboy | yep, at home I have IP 300, 500 and 600 |
02:52.20 | asterboy | no gxps |
02:52.40 | file | I've got an IP600 at my desk... works well |
02:52.43 | [TK]D-Fender | file : Once I confirm that item we discussed as being constant between 1.2.4 & 1.2.7.1 think you could lend a hand in patching? :) |
02:52.52 | asterboy | especially with the micro browser |
02:52.59 | asterboy | I get digg.com feeds and weather |
02:53.04 | file | [TK]D-Fender: yessir, I can give you something so you can figure out the flow |
02:53.09 | file | and see what's going kaboom |
02:53.12 | [TK]D-Fender | asterboy : Did I help you with those at one point? |
02:53.36 | [TK]D-Fender | file : cool. will let you know once I've got proof :) |
02:53.42 | file | yay proof |
02:54.19 | Beirdo | 151 proof? |
02:54.21 | [TK]D-Fender | file : Well I know for sure about the intermittint one in 1.2.4, I'll jsut make sure its the same in current as well as the other mode. |
02:54.22 | asterboy | yep, pointed me towards the wiki to show what the semantics were |
02:54.46 | asterboy | I almost have a calendar ported over to the phone now. |
02:54.56 | [TK]D-Fender | asterboy : picky-ass thing isn't it? I HATE that it doesn't have tables.... |
02:55.20 | asterboy | really picky, and the wiki is horribly insufficient |
02:55.34 | asterboy | It's like programming in old Pascal |
02:55.48 | asterboy | one wrongly placed space and boom |
02:55.49 | [TK]D-Fender | asterboy : No... I LIKED Pascal... |
02:56.03 | asterboy | old Pascal was way too picky |
02:56.07 | [TK]D-Fender | and Pascal was not white-space sensitive. Fortran and Cobol often were though... |
02:56.46 | asterboy | one missing space before a semicolon and the whole program fwould not run and you got an error message which told you nothing of where the problem was |
02:56.49 | [TK]D-Fender | I was a TurboPascal GOD in my day (you know... when swooping pteradactyl's were the greatest threat to man...) |
02:57.03 | asterboy | mine was |
02:57.17 | [TK]D-Fender | asterboy : I did mine on an IMB 360 and in DOS... never had whitespace issues before... |
02:57.57 | [TK]D-Fender | IBM* |
02:57.58 | asterboy | I did mine on a TI |
02:58.14 | [TK]D-Fender | TI? No accounting for their flakeyness :) |
02:58.25 | asterboy | heh |
02:59.08 | asterboy | cobol should be wiped from history |
02:59.27 | asterboy | Fortran was good, but not very fun |
02:59.33 | [TK]D-Fender | One thing I'd really like to know : If only the 60x has the microbrowser, what is the "Services" button on my 501 for? |
03:00.11 | asterboy | didn't they get the microbrowser in the latest updates |
03:01.43 | [TK]D-Fender | Don't recall that.... |
03:01.59 | [TK]D-Fender | I'll go DL the release notes... |
03:02.18 | asterboy | I had a C64 with a 300 baud modem to upload my fortran code. |
03:02.27 | asterboy | to the university computers |
03:02.57 | asterboy | can't remember if I was using a fortran emulator or terminal emulator |
03:03.33 | [TK]D-Fender | asterboy : I've had worse :) not as a modem... but I think I may have mentioned my earliesst 300 baud mode required you to hold a toggle switch to initiate carrier :) |
03:03.38 | asterboy | anyway, I remember sitting in my little room till 3am pushing buttons. |
03:03.52 | asterboy | oh ya...lol |
03:04.29 | [TK]D-Fender | Yay, you can use DHCP to set the SIP server address now, cool... |
03:09.33 | *** join/#asterisk cryptnix (n=andrew@64.25.198.126) |
03:11.13 | [TK]D-Fender | asterboy : nope, nothing about the MB for 50x. Tried adding it into the config, still no activity.... |
03:14.29 | Tusker | [TK]D-Fender: is D(digits) in asterisk 1.0.7 ? |
03:14.47 | [TK]D-Fender | Tusker : I believe so..... taht might do it... |
03:15.38 | *** join/#asterisk websae_ (n=websae@CPE-24-167-206-22.wi.res.rr.com) |
03:16.54 | websae_ | quiet friday night out there |
03:20.37 | websae_ | jake: how are you doing? |
03:20.48 | jake1932 | doing well - how bout you? |
03:28.13 | *** join/#asterisk znoG (n=gs@109-130-89-200.fibertel.com.ar) |
03:31.11 | websae_ | come one...come all |
03:33.00 | *** join/#asterisk shido6 (n=shido6@d38-45-81.commercial1.cgocable.net) |
03:36.57 | websae_ | welcome back cunningpike |
03:37.37 | CunningPike | Thank you :D |
03:37.49 | websae_ | you bet |
03:38.16 | [TK]D-Fender | Well.. I'm done... veg night concludes and back to thr grind tomorrow... |
03:40.12 | [TK]D-Fender | ater |
03:45.05 | asterboy | dam that sucks...I thought the 501 could do that |
03:45.19 | asterboy | I know the 500 can not |
03:45.39 | asterboy | service call for me. |
03:45.49 | *** join/#asterisk shuri (n=shuri@64.235.209.226) |
03:45.52 | asterboy | I just got off work and now I have to go back |
03:45.55 | asterboy | yuk |
03:49.04 | websae_ | why? |
03:49.17 | asterboy | money |
03:49.31 | asterboy | why else does anyone work? |
03:49.35 | websae_ | ohh |
03:49.35 | websae_ | ok |
03:49.37 | websae_ | there you go |
03:49.40 | websae_ | but are there issues? |
03:49.55 | asterboy | nothing on the telco side. |
03:50.15 | asterboy | bbl |
03:50.23 | websae_ | bye |
03:53.04 | *** join/#asterisk bmg505 (n=leon@dsl-146-51-61.telkomadsl.co.za) |
03:53.10 | *** join/#asterisk pageus (n=FreePBX0@ip70-190-19-6.ph.ph.cox.net) |
03:53.48 | *** join/#asterisk Gamercjm (n=chris@pool-71-254-177-36.lsanca.fios.verizon.net) |
03:56.22 | Gamercjm | Anyone know about VirtualHosts in apache, cant seem to get mine to work |
03:56.38 | websae_ | #Apache |
03:57.10 | CunningPike | :) |
03:57.24 | websae_ | cunningpike: still alive heh? |
03:57.36 | CunningPike | Oh yes |
03:59.41 | *** join/#asterisk angom_h (n=angom@red-corp-200.79.134.173.telnor.net) |
04:02.25 | dlynes | Anyone play with MixMonitor? |
04:02.37 | *** join/#asterisk jimmy_deanPB (n=jhodapp@cpe-24-166-23-17.indy.res.rr.com) |
04:02.37 | dlynes | Erm ControlPlayback, I mean? |
04:05.52 | *** join/#asterisk alephcom (n=alephcom@host75.net14.mcsnet.ca) |
04:11.19 | dlynes | Actually, I guess the problem I'm having is with MixMonitor |
04:11.25 | dlynes | It seems to record an empty gsm file |
04:11.37 | dlynes | Or empty raw file, or whatever format I'm recording |
04:12.32 | dlynes | I'm using MixMonitor(filename.gsm,a) |
04:12.37 | tainted- | why not just use Monitor |
04:12.51 | dlynes | tainted-, because it doesn't do call leg mixing without using an external process |
04:13.00 | tainted- | soxmix |
04:13.06 | tainted- | what's wrong with that? |
04:13.08 | dlynes | soxmix is an external process |
04:13.10 | Qwell | well, try using Monitor, and see if either/both are empty |
04:13.18 | tainted- | Qwell! |
04:13.20 | Qwell | If they aren't, you know it's likely to be a bug in mixmon |
04:13.21 | tainted- | stop trolling |
04:13.33 | dlynes | yeah...good idea |
04:13.53 | tainted- | Qwell u know much manager api? |
04:13.59 | Qwell | not really |
04:14.52 | *** part/#asterisk angom_h (n=angom@red-corp-200.79.134.173.telnor.net) |
04:17.13 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
04:17.45 | jake1932 | <PROTECTED> |
04:17.55 | jake1932 | we use it in a production system |
04:18.08 | tainted- | what platform do u use |
04:18.13 | tainted- | to connect to it |
04:18.21 | jake1932 | we use a .NET app |
04:18.25 | tainted- | 2.0? |
04:18.57 | dlynes | Yeah...works fine with monitor |
04:18.59 | jake1932 | .net 2003 |
04:19.16 | tainted- | are u using the asterisk.net library? |
04:19.21 | jake1932 | oh |
04:19.30 | jake1932 | no - we wrote are own routines to parse |
04:19.42 | tainted- | oh nice |
04:19.47 | jake1932 | it was pretty straightfoward |
04:20.01 | tainted- | yea i just got into |
04:20.02 | tainted- | it |
04:21.01 | jake1932 | we're going to move over some agi code in a few weeks so everything is using manager |
04:21.13 | tainted- | wow |
04:21.14 | jake1932 | should be cleaner |
04:21.15 | tainted- | same here |
04:21.41 | tainted- | agi does have good uses though |
04:21.56 | tainted- | do u write all your apps in 1.1? |
04:22.51 | jake1932 | 1.1 .NET SDK? |
04:22.59 | jake1932 | yes |
04:23.03 | jake1932 | all current apps |
04:23.37 | tainted- | i wish there was a channel bridge app |
04:24.05 | dlynes | There is |
04:24.09 | dlynes | It's called the chunnel |
04:24.37 | dlynes | It bridges England and France across the channel :) |
04:24.45 | dlynes | Good application of modern architecture :) |
04:25.08 | tainted- | groan |
04:25.12 | dlynes | channel...bridge...app... |
04:25.13 | tainted- | someone spiked your bubble tea |
04:25.17 | jake1932 | i think you can hack something with meetme |
04:25.29 | tainted- | meetme is a bucket of ass |
04:25.45 | jake1932 | haha |
04:25.56 | tainted- | yea u can originate calls and drop into meetme() context |
04:26.13 | jake1932 | right |
04:26.16 | tainted- | meetme(b) is broken |
04:27.09 | jake1932 | don't think you need that though |
04:27.33 | jake1932 | as long as you can still retain control of the call using manager |
04:27.53 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
04:27.55 | tainted- | is it possible to get meetme attendee list? |
04:28.06 | tainted- | from manager api or otherwise? |
04:28.11 | tainted- | for kicking/muting etc |
04:28.17 | *** join/#asterisk gursikh (n=guriskh1@adsl-209-30-245-73.dsl.hstntx.swbell.net) |
04:28.52 | jake1932 | i know you can access the CLI commands |
04:29.00 | jake1932 | (from manager) |
04:29.09 | jake1932 | looking if there is one |
04:29.17 | tainted- | there's meetmeadmin() |
04:29.24 | tainted- | and i think meetmecount() |
04:29.28 | jake1932 | that gives you a count |
04:29.39 | jake1932 | but i'm talking about cli cmds |
04:29.42 | jake1932 | those are apps |
04:29.48 | tainted- | hmm |
04:29.50 | tainted- | good idea |
04:29.56 | tainted- | wonder if there is |
04:30.08 | jake1932 | yep |
04:30.11 | CunningPike | dlynes: You've got mail |
04:30.17 | jake1932 | http://www.voip-info.org/wiki-Asterisk+cmd+MeetMe |
04:30.21 | jake1932 | looks like you can |
04:30.42 | dlynes | CunningPike, thx |
04:30.49 | CunningPike | np |
04:31.08 | dlynes | Just learning a new perl module atm |
04:32.21 | tainted- | hmm |
04:32.26 | *** join/#asterisk Assid (n=assid@203.115.64.12) |
04:32.37 | tainted- | is it possible to set meetme to use 729 instead of ulaw? |
04:33.08 | russellb | meetme can not mix g729 natively, no. |
04:33.25 | russellb | you can call into it with g729, of course |
04:33.33 | tainted- | and it transcodes? |
04:33.37 | russellb | yes |
04:33.49 | russellb | if you have codec_g729 |
04:33.52 | tainted- | no wonder |
04:34.53 | tainted- | i don't have 'Meetme' in CLI |
04:35.27 | jake1932 | do you have the app though? |
04:35.45 | tainted- | yea |
04:35.48 | tainted- | oops |
04:35.49 | tainted- | wrong box |
04:35.51 | tainted- | lol |
04:36.32 | pageus | heya tainted |
04:36.52 | tainted- | hey didn't recognize u w/o the capital P |
04:37.01 | pageus | ROFL |
04:37.04 | Pageus | lol |
04:37.13 | tainted- | how's it going |
04:37.42 | Pageus | was just playing some metroid prime hunter with my son before i came in and worked on this.. turns out i won't get my cable drop till monday at 8am.. but i did get the specs on the line |
04:37.49 | tainted- | is voicepulse still acting weird? |
04:38.07 | tainted- | what kind of cable? |
04:38.24 | *** join/#asterisk fjean (n=fjean@201.29.130.118) |
04:38.42 | fjean | hi guys |
04:38.46 | fjean | tell me |
04:38.47 | Pageus | nope.. that apparently was a misconfiguration on my end.. since i had hunt on and that extension was offline it went to vm |
04:38.57 | Pageus | heya dlayn |
04:39.38 | fjean | is there anything special to be done in order to make zaptel 1.2.5 ? I get *** No rule to make target `modules' |
04:39.41 | Pageus | it's a T1.. esf coding, b8zs framing, 6 2 way DID trunks. the start signaling is E&M Wink signal type is dtmf |
04:39.51 | russellb | fjean: just "make ; make install |
04:39.53 | Pageus | tg direction (??) is 2 way |
04:40.05 | fjean | russelb, let me try |
04:40.19 | tainted- | Pageus how much is that running u per month? |
04:40.40 | fjean | russelb - nope... |
04:41.08 | Pageus | 220 |
04:41.11 | tainted- | fjean u don't need to 'make modules' |
04:41.11 | Pageus | plus ld |
04:41.20 | Pageus | i have a total of 20 DID's |
04:41.26 | russellb | fjean: then you need the kernel headers installed |
04:41.53 | Pageus | this looks like it's all the info i need to configure that line though, right? |
04:42.06 | fjean | russelb - ok, I installed kernel-source already, maybe it's missing a link |
04:42.46 | russellb | fjean: you need /usr/src/linux-`uname -r` |
04:42.54 | fjean | ah |
04:43.02 | russellb | or linux-headers-`uname -r` |
04:43.03 | tainted- | Pageus u'll need some hardware |
04:43.04 | russellb | something like that :) |
04:43.12 | fjean | hehe, let me see |
04:43.35 | tainted- | 6 channels for 220, that's pricey |
04:43.49 | tainted- | are the other channels data? |
04:44.45 | russellb | fjean: which should also give you /lib/modules/`uname -r`/build/ |
04:45.27 | Pageus | i have the hardware.. |
04:45.35 | Pageus | that isn't an issue |
04:45.52 | Pageus | and the 220 is all business line.. |
04:46.04 | Pageus | we were paying 350 a month |
04:46.07 | fjean | russelb - your good, i think I installed the kernel sources for 2.4 instead of 2,6 |
04:46.24 | russellb | fjean: ;) |
04:46.32 | Pageus | but since it never leaves the facility, we have no gov taxes |
04:52.38 | fjean | russelb - way to go, working |
04:52.49 | Pageus | the good part is that it's all local calling.. so i don't have to pay the ld fee for outbound unless is really is ld |
04:53.10 | russellb | fjean: awesome |
04:55.53 | fjean | by the way, I am no expert at linux but one thing I know is that it's difficult to get the modprobes to be performed at boot time..I thouht make install would do that part.. |
04:56.08 | russellb | fjean: it's specific to the distribution of linux ... |
04:56.17 | fjean | right.. |
04:56.28 | russellb | fjean: there are init scripts you can use for most |
04:56.41 | *** join/#asterisk jerlique (n=jerlique@lnk6.adl5.adsl.esc.net.au) |
04:57.25 | *** join/#asterisk nrw (n=nrw@CPE000024c1cfca-CM0014e8b564ec.cpe.net.cable.rogers.com) |
04:58.34 | fjean | russelb: any script included in zaptel ? |
04:58.54 | russellb | yeah, what distro are you using |
04:59.04 | fjean | mandrake 10.1 :-) |
04:59.50 | CunningPike | Does mandrake use /etc/init.d? |
05:00.02 | fjean | yes.. |
05:00.02 | russellb | i have no idea if the included one will work |
05:00.23 | russellb | it looks like it's set up to work for debian or redhat. |
05:00.33 | CunningPike | If it does, the one that gets installed doesn't have paths to modprobe (e.g. /sbin/modprobe), so you need to modify it to make sure |
05:00.42 | fjean | well i know make install would not work, if that is what its supposed to do.. |
05:00.55 | fjean | cunning: ok |
05:01.00 | russellb | no, make install does not install it |
05:01.04 | russellb | "make config" installs the init script |
05:01.10 | fjean | ah, right |
05:01.14 | fjean | I forgot |
05:01.17 | fjean | ok |
05:04.47 | *** part/#asterisk fjean (n=fjean@201.29.130.118) |
05:11.34 | *** join/#asterisk subdolus (n=subby@subby.afraid.org) |
05:49.14 | *** join/#asterisk kristalino (n=kristali@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr) |
06:04.27 | *** join/#asterisk c4t3l (n=robert@cpe-24-175-57-117.houston.res.rr.com) |
06:39.16 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
06:41.03 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219) |
06:42.07 | *** join/#asterisk tparcina (n=tparcina@83.131.135.135) |
06:42.51 | tparcina | good morining |
06:43.03 | websae_ | good morning |
06:43.07 | websae_ | how are you? |
06:44.54 | tparcina | sleapy :) |
06:45.11 | tparcina | it's quite... |
06:45.40 | websae_ | it sure is |
06:45.44 | websae_ | no activity for hours |
06:46.18 | tparcina | and you are from australia, or are you doing night shift? :)) |
06:46.31 | websae_ | from United States |
06:46.37 | websae_ | Milwaukee, WI |
06:46.38 | websae_ | :) |
06:47.17 | tparcina | do, what's the time over there? |
06:47.24 | websae_ | where are you from? |
06:47.29 | websae_ | it's 1:45AM |
06:47.48 | tparcina | croatia (hr) |
06:48.06 | websae_ | what do you use asterisk for? |
06:48.10 | *** join/#asterisk OloBola (n=not@netblock-68-183-67-158.dslextreme.com) |
06:48.26 | tparcina | I work in company that makes pbx's |
06:48.48 | tparcina | we curently use one, and we have sell one call centar |
06:48.59 | websae_ | do you use sip or pri? |
06:49.00 | tparcina | now, we re selling another two... |
06:49.05 | websae_ | very interesting |
06:49.09 | websae_ | people terminating over SIP? |
06:49.15 | websae_ | or PRIs? |
06:49.23 | tparcina | right now pri, and h323 |
06:49.29 | websae_ | sure |
06:49.50 | tparcina | but those two new * will be conected together, probably over sip |
06:50.11 | tparcina | u work with *? |
06:50.49 | websae_ | yes |
06:50.51 | websae_ | quite a bit |
06:51.06 | websae_ | here private msg me |
06:51.14 | tparcina | hawe you used cisco phones? |
06:51.17 | *** join/#asterisk Kernel_Core (n=I@193.251.135.118) |
06:51.39 | websae_ | a couple |
06:51.41 | websae_ | times |
06:52.12 | tparcina | here private msg me - don't get it |
06:52.18 | tparcina | what phones? |
06:52.24 | tparcina | sip or sccp? |
06:52.32 | websae_ | sip |
06:52.35 | websae_ | 7960 |
06:52.46 | *** join/#asterisk MGSsancho (n=user@adsl-67-125-156-130.dsl.irvnca.pacbell.net) |
06:52.48 | websae_ | do you have msn or yahoo? |
06:56.20 | *** join/#asterisk nigelr (n=nigelr@ninja.nobiscuit.com) |
06:57.04 | nigelr | anyone got time for a curly question on tranfer capabilities for a PRA? |
06:58.49 | *** join/#asterisk MacDome (n=eseidel@c-71-198-177-144.hsd1.ca.comcast.net) |
06:59.27 | CunningPike | Sorry, websae_ - didn't see your PM until now |
06:59.46 | CunningPike | Stupid Colloquy always pops the PM window under the main one |
07:00.13 | *** join/#asterisk MacDome (n=eseidel@c-71-198-177-144.hsd1.ca.comcast.net) |
07:04.25 | *** join/#asterisk IOscanner (n=IOscanne@c-67-164-154-209.hsd1.tx.comcast.net) |
07:08.32 | tparcina | cisco, sip, hinting (multiple line appereanses) - has anybody done it? |
07:08.52 | tparcina | i have check on wiki, but I couldn't find anything about hinting on sip |
07:18.47 | *** part/#asterisk startled (i=startled@d58-105-31-172.dsl.vic.optusnet.com.au) |
07:23.33 | dlynes | tparcina, I'm working on a page for that, tparcina |
07:23.39 | dlynes | tparcina, it's using aastra phones |
07:23.55 | tparcina | hi dlaynes, how are you today? |
07:24.46 | dlynes | good |
07:24.48 | tparcina | don't get it. what aastra phones have to do with cisco 7960 sip hinging? |
07:24.56 | tainted- | dlynes finish the billing system yet? lol |
07:24.59 | dlynes | aastra's do sip hinting, too |
07:25.09 | websae_ | oooo---dirty |
07:25.18 | tainted- | websae_ what games do u have? |
07:25.37 | tainted- | we could play halo 2 and stuff |
07:25.51 | websae_ | they cancelled my xbox live aaccount |
07:26.29 | tainted- | that's terrible |
07:27.02 | websae_ | im depressed |
07:27.11 | tparcina | dlynes, have you done anything so far? (cisco hinting) |
07:27.23 | dlynes | Aastra hinting, yes |
07:28.41 | tparcina | and you are sure that hinting is posible on 7960 with sip firmware? |
07:28.52 | dlynes | Nope |
07:28.57 | tparcina | do you have link to any instructions (even partial) |
07:29.03 | dlynes | I don't know what the 7960 supports |
07:29.07 | dlynes | One sec |
07:32.59 | dlynes | tparcina, http://www.voip-info.org/wiki/view/480i+Busy+lamp+field+'BLF'+support |
07:33.03 | {zombie} | I didn't think the 7960 supported hinting even via SCCP - you need the 7961 for that |
07:34.19 | {zombie} | or the 7914 expansion panel for the 7960 |
07:34.49 | dlynes | tparcina, btw...you did say "but I couldn't find anything about hinting on sip" |
07:35.06 | dlynes | tparcina, you didn't say you were looking for hinting on sip, specifically for the cisco 7960 |
07:35.28 | {zombie} | dlynes: he did in the line above |
07:35.35 | {zombie} | cisco, sip, hinting |
07:35.50 | dlynes | Yeah, but not originally, which is what i had replied to |
07:36.56 | *** join/#asterisk poisoner (i=poisoner@tauceti.poisoner.de) |
07:37.02 | tparcina | ok, sorry if I have mislead you :) |
07:37.07 | dlynes | yeah...hinting isn't supported on the 7960, tparcina |
07:37.42 | tparcina | zombie, have you set up hinting on 7970 with sip? |
07:37.54 | dlynes | doesn't look like 7961 supports it either...you need 7971, and maybe 7970 |
07:38.01 | websae_ | does anyone know tasker here? |
07:38.08 | websae_ | I am looking for tasker |
07:38.26 | {zombie} | tparcina: no I only have 7960, 7940 and 7910 cisco phones |
07:38.32 | tparcina | dlynes, i though so, but evry while, sombody mentions it, so i though... anyway, too bad |
07:38.37 | {zombie} | although I have set up hinting with grandstream gxp2000 and all snom phones |
07:39.03 | tparcina | dlynes, i have 7970, but i can't find instructions |
07:39.17 | dlynes | tparcina, that's what I said...I don't think the 7970 supports it |
07:39.22 | dlynes | I think you need the 7971 |
07:39.30 | tparcina | i hate cisco phones, they have 100 books for ccme but nothing for sip general... |
07:39.44 | dlynes | voip-info isn't clear on that, but they seem to indicate the 7970 doesn't do hinting |
07:39.52 | dlynes | the 7971 does hinting for sure |
07:40.37 | {zombie} | 7961 does too, via SCCP at least |
07:40.49 | *** join/#asterisk xMOe (n=Blade@62.149.93.73) |
07:40.50 | {zombie} | http://www.voip-info.org/wiki-chan_sccp2 |
07:40.54 | xMOe | hola guys, i've call center system based on Asterisk with Digium TDM400P but calle id dosnt work out fine can anyone tel if its support caller ID in Saudi Arabia or not , and if yes any online reference for zapata.conf configuration settings are needed |
07:43.35 | tparcina | xmoe, check ith your telco does it support caller id |
07:44.03 | dlynes | xMOe, your telco and/or your line might not support it |
07:44.27 | dlynes | it's not a country-specific thing |
07:44.50 | dlynes | i can buy analog lines here with caller id, and without caller id |
07:45.18 | dlynes | you generally pay extra for a line if it has caller id |
07:45.46 | dlynes | tainted-, do you ever go to sleep? |
07:46.12 | tainted- | no |
07:46.18 | dlynes | thought not |
07:46.23 | tainted- | it's gotten so bad, i dream in code |
07:46.59 | dlynes | 0x22 0x43 0x44 0x45 0x22 |
07:47.24 | tainted- | not that bad |
07:47.28 | dlynes | lol |
07:47.39 | tainted- | sometimes i'll solve real life problems in code |
07:47.42 | dlynes | or better yet |
07:47.59 | dlynes | A9 22 43 44 45 22 A9 C9 D8 |
07:48.09 | dlynes | Run that, and tell me what you get |
07:48.15 | dlynes | It's 6502 opcode :) |
07:48.22 | tainted- | if (takeGarbageOutNow() { ... smellFactor += 1; ) // hmmm |
07:48.35 | tainted- | oops missing ) |
07:48.43 | tainted- | damn it now i'll never get to sleep |
07:48.45 | dlynes | It prints out the string, 'ABC' |
07:48.48 | dlynes | heh |
07:49.08 | *** join/#asterisk af_ (n=af@ip-143-220.sn1.eutelia.it) |
07:49.10 | tainted- | yea you're pretty cool |
07:49.17 | tainted- | 010011001011110100101 |
07:50.49 | dlynes | Is the search feature on voip-info broken? |
07:51.37 | xMOe | tparcina its support 100% |
07:52.01 | xMOe | dlynes yes its do and i used to get called id with asterisk and sipura |
07:52.11 | dlynes | ah |
07:52.29 | xMOe | but now with TDM400P i just got " Astrerisk" |
07:52.51 | dlynes | Yeah...that's what you get when it can't get a caller id |
07:53.03 | dlynes | Is this TDM400P a Digium TDM400P, or is it a clone? |
07:53.39 | xMOe | <PROTECTED> |
07:53.47 | dlynes | one second |
07:54.14 | xMOe | ok |
07:56.02 | L|NUX | can some one tell me how can i goto another context once i got timeout ? |
07:56.32 | dlynes | xMOe, do you get an error like this: messages.7:Apr 27 00:07:33 WARNING[1569] chan_zap.c: CallerID returned with error on channel 'Zap/1-1'? |
07:56.40 | Pageus | i hear you on that one tainted.. dreaming in code has written many a program for me |
07:58.40 | xMOe | dlynes i got no error msg :) |
07:59.11 | websae_ | anyone know tasker around here? |
07:59.18 | dlynes | L|NUX, Look at the following page: http://www.voip-info.org/wiki/view/Asterisk+standard+extensions |
07:59.34 | dlynes | L|NUX, also, check the documentation for the Dial() application |
07:59.47 | dlynes | xMOe, wow...no idea then |
08:00.05 | dlynes | xMOe, do you have warning level messages disabled? |
08:03.32 | *** join/#asterisk Pageus (n=FreePBX1@ip70-190-19-6.ph.ph.cox.net) |
08:03.47 | xMOe | dlynes nope.. |
08:04.11 | dlynes | xMOe, so you double checked logger.conf? |
08:06.19 | *** join/#asterisk Alystair (n=bob@CPE001109c15241-CM00407b8794db.cpe.net.cable.rogers.com) |
08:08.18 | xMOe | yes i ddd |
08:26.32 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219) |
08:30.41 | *** join/#asterisk VeNoMouS_ (n=jj@202.162.177.196) |
08:31.52 | *** join/#asterisk frenzy (n=frenzy@196.45.144.41) |
08:32.02 | frenzy | hey all |
08:32.27 | frenzy | is there an advanced carrier routing tool available for asterisk ? |
08:35.01 | *** join/#asterisk [hC] (n=hardcore@S01060004e21ea953.vc.shawcable.net) |
08:36.07 | *** join/#asterisk Dr-Linux (n=Linux@202.59.73.131) |
08:38.12 | frenzy | ? |
08:40.49 | dlynes | frenzy, i think you're looking for a carrier grade linux |
08:41.00 | dlynes | ? |
08:41.26 | frenzy | a carrier routing based on mysql |
08:41.38 | frenzy | bascially to do LCR failover |
08:42.05 | dlynes | Oh...you mean a dialplan application? |
08:42.25 | dlynes | I think there might be something out there already for that |
08:42.28 | frenzy | something like that |
08:42.37 | dlynes | but if you mean a separate tool that runs from the command line then probably not |
08:43.08 | frenzy | no |
08:43.26 | frenzy | what kind of applications are availble to do that? |
08:43.32 | dlynes | http://www.voip-info.org/wiki/view/LCR+tool+for+i4l |
08:45.34 | *** join/#asterisk rkr245 (n=ravi@office.callsat-telecom.com) |
08:54.33 | *** join/#asterisk ghost99 (n=neville@222-152-219-77.jetstream.xtra.co.nz) |
08:54.42 | *** join/#asterisk UlbabraB (n=caplaz@host241-43.pool8172.interbusiness.it) |
09:00.28 | *** join/#asterisk boddy (n=boody@85.103.15.201) |
09:02.03 | *** join/#asterisk abatista (n=Ariel@dsl-20-177.cofs.net) |
09:02.05 | boddy | hii all local phone that on Meridian 1C when call the sip client which codec it will use ? asterisk and meridian connected via E1 |
09:02.21 | *** join/#asterisk jomo2005 (n=jomo2005@c-24-98-66-60.hsd1.ga.comcast.net) |
09:02.43 | *** part/#asterisk jomo2005 (n=jomo2005@c-24-98-66-60.hsd1.ga.comcast.net) |
09:03.14 | boddy | ? |
09:10.50 | *** join/#asterisk treobruce (n=jomo2005@c-24-98-66-60.hsd1.ga.comcast.net) |
09:12.29 | *** join/#asterisk speedwagon (n=Ariel@dsl-20-177.cofs.net) |
09:13.05 | *** part/#asterisk treobruce (n=jomo2005@c-24-98-66-60.hsd1.ga.comcast.net) |
09:13.55 | boddy | anbody help me? |
09:13.55 | boddy | zzzzz |
09:14.19 | *** join/#asterisk MGSsancho (n=user@adsl-67-125-156-130.dsl.irvnca.pacbell.net) |
09:29.23 | *** join/#asterisk stoffell_h (n=stoffell@d5153F9E0.access.telenet.be) |
09:30.59 | dlynes | boddy: probably whatever its preferred codec is |
09:31.41 | *** join/#asterisk Dr-Linux (n=Linux@202.59.73.131) |
09:33.21 | *** join/#asterisk markit (n=konversa@host119-245.pool8172.interbusiness.it) |
09:34.04 | *** join/#asterisk jeffik (n=Jeff@Crimson-111.085.ADSL.NetSurf.Net) |
09:38.32 | *** join/#asterisk CrummyGummy (n=wayne@dsl-145-99-210.telkomadsl.co.za) |
09:39.21 | boddy | dlynes I didnt understand |
09:39.32 | *** join/#asterisk frk2 (n=kvirc@202.141.251.102) |
09:39.44 | frk2 | is there any such thing as a cheap IP phone that DOESNT hang? |
09:39.53 | dlynes | your sip client has a preferred codec; the codec it prefers to use over other codecs |
09:40.02 | dlynes | that's the codec it'll probably use |
09:40.18 | frk2 | i dont understand. I have tested all IP phones and almost ALL of them hang.. the low end ones |
09:40.29 | dlynes | frk2: I haven't had any hanging issues iwth the budgetone 102's |
09:41.02 | frk2 | dlynes |
09:41.08 | frk2 | i had high hopes on those two |
09:41.13 | frk2 | but one hung today only :( |
09:41.17 | dlynes | ah |
09:41.22 | frk2 | i upgraded the firmware to 1.0.8.16 |
09:41.24 | dlynes | Yeah...don't go for the ACT phones |
09:41.29 | frk2 | i hope that didnt screw it up |
09:41.29 | dlynes | They hang regularly |
09:41.36 | dlynes | they suck total dog doo |
09:41.40 | frk2 | ACT? |
09:41.48 | dlynes | Advantage Century Telecom |
09:41.54 | boddy | dlynes:so if my sip clients use g729 all conversation is g729 |
09:41.59 | boddy | ? |
09:42.04 | dlynes | also known as GVC, Azatel, ... |
09:42.05 | frk2 | arent they the pa1688 guys? |
09:42.11 | *** join/#asterisk bmg505 (n=leon@dsl-146-51-61.telkomadsl.co.za) |
09:42.12 | dlynes | frk2: correct |
09:42.21 | frk2 | dude. funny enough |
09:42.24 | dlynes | frk2: i've never used that particular product though |
09:42.27 | frk2 | the pa1688 phones hang the LEAST |
09:42.32 | frk2 | but they have other issues |
09:42.34 | jeffik | dlynes: Is there a way to hard reset an ACT phone? |
09:42.51 | frk2 | dlynes- what firmware on those non hanging 102s? |
09:42.59 | dlynes | jeffik: 255*0*0*0, assuming you've got a hardware revision that actually works on |
09:43.44 | dlynes | frk2: 1.0.1.0 |
09:43.51 | jeffik | dlynes: well mine worked, i loaned it to a client and now that i have it back when i plug in power all lights come on and display shows blocks |
09:43.51 | frk2 | thats the beta |
09:44.02 | dlynes | if it works, it works |
09:44.08 | dlynes | I've never upgraded the firmware on it |
09:44.10 | frk2 | dlynes-- dont you also have the GXP 2000s? |
09:44.14 | dlynes | nope |
09:44.38 | dlynes | Grandstream BT-102, Azatel IPCall104, Aastra 9133i |
09:45.03 | dlynes | And a bunch of polycom 500's that will only work with Artisoft Televantage |
09:45.09 | markit | gxv-3000 amatorial video from CeBIT 2006: http://video.google.com/videoplay?docid=6420459719167336518&pl=true |
09:45.28 | *** part/#asterisk liran_ (n=liran@80.178.14.98.adsl.012.net.il) |
09:45.42 | dlynes | jeffik: i would guess the firmware is fried on it |
09:46.06 | dlynes | jeffik: if the display is all scrambled i doubt the keypad reset sequence will work |
09:46.47 | jeffik | dlynes: just tried your suggestion 255*0*0*0 and nothing |
09:47.02 | boddy | dlynes e1 uses ulaw/alaw ? |
09:47.42 | dlynes | jeffik: like i said...i didn't think it would work |
09:47.56 | dlynes | boddy: no...e1 uses telephone event |
09:48.03 | dlynes | boddy: it doesn't use a codec |
09:48.15 | jeffik | dlynes: thanks for the suggestion anyway, it was worth a try |
09:48.31 | boddy | I confused :( |
09:48.36 | dlynes | jeffik: yeah, but at least you know the reset sequence if you have other phones you've lost the password for |
09:49.03 | dlynes | jeffik: it works for firmware 2.0.8 and higher |
09:49.19 | dlynes | jeffik: it might work for 2.0.7, but definitely not 2.0.6 |
09:50.20 | dlynes | I'm going to be calling those bastards on Monday to see if I can get some support out of them |
09:50.25 | dlynes | Azatel's gone out of business |
09:50.27 | *** join/#asterisk mutante (i=mutante@s23.org) |
09:50.56 | boddy | dlynes:Please explain to me when local phone calls to sip client will it use any codec ? |
09:51.12 | dlynes | boddy: local doesn't use a codec, sip client does |
09:51.39 | frk2 | dlynes i think the firmware upgrade did it |
09:51.42 | dlynes | boddy: asterisk will by default try to use the least processor intensive codec, unless the client overrides it |
09:51.46 | *** join/#asterisk treobruce_ (n=jomo2005@c-24-98-66-60.hsd1.ga.comcast.net) |
09:52.07 | dlynes | boddy: so, if your client supports g729, ulaw, alaw, ... |
09:52.18 | stoffell_h | so if u use u/alaw on your sip client, the * server does not have to do transcoding.. |
09:52.21 | dlynes | boddy: and ulaw is your default, it'll use ulaw |
09:52.45 | Assid | err.. anyone have a poly501 ? |
09:53.11 | Assid | err.. 301 |
09:53.14 | frk2 | i wonder if i can downgrade the GS-102 |
09:53.15 | stoffell_h | Assid, almost everyone has 1 ;) |
09:53.28 | treobruce_ | hey guys... i'm trying to setup asterisk on debian |
09:53.31 | *** join/#asterisk apardo (n=apardo@87.217.145.245) |
09:53.31 | Assid | is it any good? |
09:53.33 | treobruce_ | can anyone help |
09:53.44 | Assid | whats thedifference between 301 and 501 majorly |
09:53.48 | stoffell_h | Assid, i have 501, it's great. (601 has microbrowser) |
09:53.58 | boddy | I am wondering this local phones dosent use codec but how session is compatible with g729 asterisk do this ? |
09:54.16 | Assid | i was supposed to get a 501.. they sent me a 3013 |
09:54.20 | Assid | 301 rather |
09:54.23 | stoffell_h | I believe the 301 has no speaker phone, 501 does, and the sound of the handsfree is great |
09:54.34 | dlynes | and the 301 is half duplex speaker phone |
09:54.41 | stoffell_h | aah, listen only.. |
09:54.55 | Assid | so cant speak and talk at the same time |
09:55.02 | dlynes | Assid: correct...only listen |
09:55.03 | Assid | anythuing else? |
09:55.35 | treobruce_ | any willing helper here to help a newbie with asterisk? |
09:55.42 | treobruce_ | msg me if u are willing :) |
09:55.43 | treobruce_ | thanks |
09:56.03 | dlynes | treobruce_: just ask your question in the channel |
09:56.27 | treobruce_ | oh |
09:56.36 | treobruce_ | sorry, new to freenode too :) |
09:56.48 | treobruce_ | well i'm trying to learn asterisk |
09:56.56 | treobruce_ | so i installed debian on my system |
09:56.56 | boddy | dlynes:do you understand my question |
09:57.06 | Assid | please tell me it atleast supports ftp |
09:57.23 | dlynes | boddy: yes, and so did stoffel_h |
09:57.23 | boddy | I am trying to learn logic |
09:57.30 | dlynes | boddy: we both answered you already |
09:57.42 | iDunno | erm - logic shouldn't need to be learnt. logic should just happen. it's logical! |
09:57.44 | stoffell_h | Assid, I think the 301 also supports central provisioning through ftp (but check voip-info / polycom 2 be sure) |
09:57.57 | treobruce_ | how do i install asterisk on debian |
09:58.05 | stoffell_h | treobruce_, you needs docs, i will give them, just a sec.. |
09:58.08 | stoffell_h | ~docs |
09:58.11 | jbot | it has been said that docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
09:58.26 | Assid | stoffell_h: whats better? polycom301 or linksys pap2 ? |
09:58.36 | treobruce_ | wow that's alot |
09:58.47 | stoffell_h | happy reading ;) |
09:58.52 | *** join/#asterisk kore (i=kore@mindwipe.org) |
09:58.57 | boddy | If I use g729 on asterisk then asterisk will transcoding |
09:58.58 | stoffell_h | Assid, don't know that linksys.. |
09:59.11 | dlynes | Assid: You're comparing a phone to an ata |
09:59.16 | dlynes | Assid: that does not compute |
09:59.26 | Assid | feature vs/ quality |
09:59.34 | dlynes | Assid: no comparison |
09:59.41 | dlynes | Assid: polycom wins hands down by a long shot |
09:59.50 | Assid | hrmm |
09:59.51 | Assid | okay |
10:00.14 | dlynes | Assid: the pap2 has more features, but who cares? all those features are available in asterisk, too |
10:00.43 | dlynes | Assid: a polycom is much better than using an analog phone hooked up to a pap2 |
10:00.44 | boddy | dlynes: |
10:01.17 | Assid | PAP2 has features? |
10:01.20 | Assid | man i didnt know |
10:01.33 | frk2 | so anybody using the GS - 102 with the newer firmware? |
10:01.35 | stoffell_h | hm, Assid, you don't want that linksys thingie, you want that polycom ;) |
10:01.38 | frk2 | 1.0.8.16? |
10:01.55 | frk2 | damnit.. i shouldn't have upgraded! |
10:02.00 | dlynes | stoffell_h: the pap2 is the new two port fxs ata from linksys, it's a drop-in replacement for the sipura 2002, and the sipura 2000 |
10:02.28 | dlynes | stoffell_h: it's the crap that vonage is giving away for free |
10:02.30 | boddy | dlynes: If I use g729 on asterisk then asterisk will transcoding |
10:02.37 | dlynes | boddy: correct |
10:02.38 | boddy | ? |
10:02.38 | stoffell_h | dlynes, thanks.. yeah, we all want native sip phones, not rubbish :) |
10:02.41 | boddy | ok |
10:02.45 | boddy | thanks alot |
10:02.45 | dlynes | boddy: and your performance will suffer |
10:02.47 | Assid | actually i got both next to me |
10:02.59 | Assid | i have to decide what i want |
10:03.14 | boddy | I will use good machine has p4 cpu |
10:03.18 | dlynes | Assid: the sipura 2000 aka linksys pap2 has a whole bunch of vertical service codes predefined |
10:03.34 | frk2 | dlynes - is there a way to downgrade the GS-102? |
10:03.36 | dlynes | other than that, and the ability to do faxing |
10:03.46 | dlynes | frk2: maybe install older firmware? :) |
10:03.54 | stoffell_h | boddy, all depends on the amount of calls |
10:03.54 | frk2 | no on |
10:04.04 | frk2 | thats supposed to be impossible (according to grandstream) |
10:04.16 | dlynes | that would seem kinda silly |
10:04.21 | boddy | stoffell_h yes you right 10-15 calls is simultane |
10:04.23 | dlynes | try it and see what happens :) |
10:04.30 | dlynes | it's only a $60 phone, anyways |
10:04.49 | boddy | in same time |
10:05.31 | dlynes | Assid: but other than, unless you really need to do faxing, feel like pulling out all your hair, and then get frustrated when faxing is heavily unreliable |
10:05.41 | dlynes | Assid: i wouldn't suggest the pap2 |
10:05.48 | boddy | ok thanks all |
10:15.40 | *** join/#asterisk Assid (n=assid@203.115.64.12) |
10:16.56 | frk2 | so i guess the best low cost phone is the GS-102 |
10:18.09 | stoffell_h | frk2, the "lowest" I go is the thomson st2030 |
10:18.28 | frk2 | The lowest I went was the atcom 323 - pa1688 based |
10:18.35 | frk2 | thomson? |
10:18.38 | frk2 | where do you get that from? |
10:19.00 | stoffell_h | frk2: http://www.voip-info.org/wiki/view/Thomson+ST2030 |
10:19.09 | stoffell_h | they are starting to get pretty popular in europe |
10:19.45 | frk2 | checking them out |
10:19.48 | frk2 | how much for a phone? |
10:20.21 | stoffell_h | frk2, check the e-commerce sites on the bottom, but we do them for 125 EUR without vat |
10:20.59 | frk2 | nice looking phone |
10:21.02 | frk2 | damn dude |
10:21.05 | frk2 | thats not 'low end' |
10:21.06 | frk2 | :) |
10:21.24 | frk2 | not for a third world country im in |
10:21.35 | stoffell_h | well, i've been using gxp-2000, it's cheaper, but sound quality is not so good |
10:21.46 | frk2 | gxp 2000 hangs man |
10:21.51 | frk2 | with too many simultaneous calls |
10:22.00 | stoffell_h | it's buggy yes :) |
10:22.07 | frk2 | put it as the operator phone at one of my clients |
10:22.08 | stoffell_h | but latest firmware makes it a bit more stable |
10:22.10 | frk2 | had to remove it |
10:22.34 | frk2 | see if a IP phone HANGS.. its total loss |
10:22.41 | frk2 | thats the worst thing that can happen to a IP phone |
10:22.57 | stoffell_h | there's also a thomson 2020, it's cheaper, but doesn't have central provisioning |
10:23.02 | stoffell_h | yeah, it is :) |
10:23.37 | Assid | stoffell_h: do you have a 301 with you at the moment? |
10:24.08 | stoffell_h | no Assid, only have used 501 |
10:25.35 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219) |
10:26.06 | frk2 | for a third world country, i need a cHEAPP ass phone with decent voice quality and one that DOES NOT hang |
10:26.16 | frk2 | cheap meaning in the $40-$60 range |
10:26.17 | Assid | frk: ata |
10:26.49 | frk2 | even ATAs are more expensive man |
10:27.28 | Dr-Linux | what is third world country? :S |
10:28.12 | Assid | is it me.. or polycom site slow? |
10:29.13 | Assid | err.. anyone remember that site.. whihc had the polycom resources |
10:29.25 | Assid | a third party site |
10:30.53 | Assid | got it .. freedomphones |
10:33.53 | Assid | how do i come to know the sip version? |
10:33.54 | *** join/#asterisk _Vile (n=vile@90.b160.bendtel.net) |
10:34.39 | Assid | and what version can i upgrade a 301 to? |
10:36.39 | Assid | stoffell_h: you there? |
10:37.02 | _Vile | ask your question again |
10:37.25 | Assid | what version of sip /bootrom can i upgrade to with 301 ? |
10:37.41 | _Vile | 301? |
10:37.54 | frk2 | anybody using pa1688 based phones? |
10:37.58 | Assid | polycom 301 |
10:38.04 | _Vile | thought so |
10:38.19 | _Vile | I can't answer that, I play cisco |
10:38.53 | frk2 | but anybody? ive been using atcom's for a while |
10:38.54 | _Vile | but gimme a sec to search google |
10:40.40 | *** join/#asterisk saftsack (n=saftsack@p54A7F649.dip.t-dialin.net) |
10:41.13 | _Vile | 3.1 |
10:41.18 | Assid | i guess i can use 2.6.2 and 1.6.x |
10:41.25 | Assid | err. 3.1 they suggested not to use |
10:41.31 | Assid | you cant downgrade again |
10:41.40 | _Vile | sec |
10:42.00 | _Vile | 2.6.1 |
10:42.04 | _Vile | go with that |
10:42.07 | _Vile | it seems clean |
10:42.21 | Assid | yeah thats what i have |
10:42.34 | Assid | but sip is old |
10:42.34 | Assid | 0104173809|so |*|00|Application, main: Label=SIP, Version=1.4.1.0040 14-Dec-04 11:49 |
10:42.37 | Assid | 1.4.x |
10:42.57 | _Vile | *shrug*, I can't suggest more than is documented |
10:43.17 | Assid | any idea how 1.6.2 is? |
10:43.42 | _Vile | none, do you have any problems? |
10:44.02 | Assid | havent got it up |
10:44.07 | Assid | but i need the files anyways.. to provision |
10:44.25 | Assid | my 501's are running on 1.6.2 |
10:45.19 | *** part/#asterisk Pageus (n=FreePBX1@ip70-190-19-6.ph.ph.cox.net) |
10:45.40 | *** join/#asterisk zotz (n=zotz@24.231.32.85) |
10:45.42 | _Vile | 2.6.1 is apparently the newest, bug free thing you're going to find |
10:46.18 | stoffell_h | Assid, on my 501's i'm running the latest |
10:47.01 | _Vile | stoff, what version is the latest? |
10:47.11 | stoffell_h | 1.6.5 |
10:47.24 | stoffell_h | bootrom can be 2.6.x or 3.x |
10:48.06 | Assid | any difference between 1.6.5 and 1.6.2? |
10:48.25 | _Vile | yep |
10:48.50 | Assid | worth going in fro 1.6.5 ? |
10:48.50 | _Vile | I can't help here, I don't have a phone and am going from undocumentation |
10:49.01 | _Vile | stoff has the right info |
10:49.07 | _Vile | verified |
10:49.11 | stoffell_h | Assid, difference, yes, you can read it in the zip file (of 1.6.5) I hope :) |
10:49.36 | stoffell_h | my poly's were 1.6.2 also, but I wanted to have the latest to be sure of stability.. :) |
10:49.50 | Assid | just wondering if its worth upgrading to 1.6.5 |
10:49.52 | stoffell_h | _Vile you using cisco's ? |
10:49.56 | Assid | since its a new phone already |
10:50.04 | _Vile | 7940s 7960s yes |
10:50.11 | stoffell_h | Assid, if you don't have a reason, skip the upgrade.. you can always do it later.. |
10:50.21 | Assid | dont have a reason NOT TO |
10:50.36 | _Vile | yes you do |
10:50.43 | _Vile | do you have bugs? |
10:50.44 | stoffell_h | _Vile, it's correct that the latest 7960 firmware is free downloadable? (I 've downloaded it a few days ago) |
10:50.48 | _Vile | that's a reason not to |
10:50.57 | _Vile | if your answer is no |
10:51.13 | _Vile | stoff, I dunno, I have a cisco pass :) |
10:51.13 | stoffell_h | Assid, like _Vile says, if it's not broke, don't try to fix it ;) |
10:51.43 | Assid | never tried this phone.. dont know if its buggy or anything.. i remember the 501 being buggy till i upgraded it to 1.6.2 |
10:51.52 | _Vile | well |
10:51.58 | _Vile | first step > try it |
10:52.10 | _Vile | second step > ask |
10:52.24 | _Vile | I take that back |
10:52.29 | _Vile | second step > research |
10:52.33 | _Vile | third step > ask |
10:53.12 | stoffell_h | :) |
10:53.26 | _Vile | is the 7960 fw releases? |
10:53.29 | _Vile | released? |
10:53.52 | _Vile | maybe cisco is pushing for hardware now, I need to look |
10:54.13 | _Vile | they're big on hardware, software &&& support.. hm |
10:54.43 | stoffell_h | _Vile, i thought so (P003-8-2 or something) |
10:55.41 | _Vile | <PROTECTED> |
10:55.48 | _Vile | my phones are currently running it |
10:55.51 | _Vile | was internal |
10:56.06 | stoffell_h | yeah, that's the one.. I downloaded it last week (without having cisco pass) |
10:56.08 | _Vile | hmm, last month anyway |
10:56.25 | _Vile | interesting |
10:56.28 | stoffell_h | maybe it's released then.. |
10:57.12 | _Vile | could be a good way for them to push their hardware |
10:57.29 | _Vile | and *hah* support contracts |
10:57.58 | stoffell_h | hehe, yeah, would make life easier and phones (even) more widespread |
10:58.28 | *** join/#asterisk hads|home (n=hads@203.109.245.87) |
10:59.36 | _Vile | I wonder if that's a marketing thing, those downloads are usually restricted... someone slashdot it, Cisco, releasing upgrades for their IP phones |
10:59.45 | _Vile | *shrug* |
10:59.50 | _Vile | I'm going to bed |
11:00.21 | stoffell_h | indeed... cu l8er ;) |
11:06.26 | asterboy | bed? |
11:06.46 | asterboy | plenty of time to sleep when your DEAD! |
11:07.14 | asterboy | you know....shovles full of dirt being dumped on top of your body. |
11:07.26 | luke-jr_ | Anyone else use SellVoIP? |
11:07.56 | asterboy | nope, just BuyVoIP |
11:08.16 | asterboy | night |
11:12.16 | *** join/#asterisk Mavvie (n=edwin@252-131-222-203.rev.techex.net.au) |
11:13.34 | Assid | i think i screwed something up |
11:13.42 | Assid | its stuck at Checking application |
11:13.44 | Assid | :( |
11:14.55 | Assid | ftp shows its trying to transfer my <mac>.cfg file |
11:15.02 | stoffell_h | does the .cfg file exist? |
11:15.06 | Assid | yep |
11:15.13 | stoffell_h | it could be in the wrong format then.. |
11:15.25 | Assid | wrong format? |
11:15.27 | stoffell_h | just wait :) you won't brake it (unless you power off during provisioning) |
11:15.50 | Assid | 401 /home/p301/0004f20265fb.cfg b _ o r p301 ftp 0 * c |
11:15.54 | stoffell_h | well, any typo in the file could cause errors.. |
11:16.19 | stoffell_h | in that file you define where to get the 'other' .cfg files, do they exist? |
11:16.56 | Assid | <APPLICATION APP_FILE_PATH="sip.ld" CONFIG_FILES="0004f20265fb-phone.cfg, 0004f20265fb-sip.cfg" MISC_FILES="" LOG_FILE_DIRECTORY="log/" OVERRIDES_DIRECTORY="" CONTACTS_DIRECTORY=""/> |
11:17.02 | *** join/#asterisk coppice (n=chatzill@153.192.17.210.dyn.pacific.net.hk) |
11:17.03 | luke-jr_ | EWW |
11:17.31 | stoffell_h | do the files exist? (sip.ld, blabla.cfg, etc..) |
11:17.36 | Assid | yes |
11:18.02 | Assid | okay how do i stop it and make it start up again? kill the ftp server? |
11:18.25 | stoffell_h | well, if the ftp server says it's doing nothing, then you can do that, or restart the phone |
11:18.34 | Assid | power on/off ? |
11:18.48 | stoffell_h | yep |
11:18.58 | stoffell_h | unless it's reading the sip.ld file ;) |
11:19.23 | Assid | well.. as long as the bootrom is not being updated.. dont see why it could die |
11:19.40 | Assid | just curious.. whats the buttons for hardware rebooting |
11:19.45 | Assid | like you press a few keys right? |
11:20.06 | stoffell_h | hm, i only know of the reboot-through-menu (wich you can't right now), don't know any other |
11:21.54 | Assid | damn... sip.ld is soo freaking big |
11:22.53 | Assid | does it always download the file every reboot etc? |
11:23.15 | stoffell_h | no, only if it's changed |
11:24.07 | Assid | still stuck at checking application on the phone :( |
11:24.54 | stoffell_h | I guess you better reboot it |
11:25.07 | Assid | i did |
11:25.10 | Assid | 2nd time its stuck |
11:26.14 | stoffell_h | did you follow the procedure on uhm, voip-info? (there's a big article on poly provisioning) |
11:26.42 | Assid | err.. ive provisioned over 10 phones in 501 |
11:26.47 | Assid | first time im doing a 301 |
11:26.55 | stoffell_h | aaaaah, ok:) |
11:26.56 | Assid | but they were all connected to lan |
11:27.04 | Assid | im doing it on a slowwwwwwww broadband |
11:27.50 | Assid | sip.ld is like 11MB |
11:27.59 | Assid | its gonna takes a while to download |
11:28.33 | Assid | if i ftp the file locally .. and have it connect.. and then change the ftp back.. will it download sip.ld from that remote ftp again ? |
11:29.19 | *** join/#asterisk Corydon76-home (i=three@pdpc/supporter/sustaining/Corydon76-home) |
11:29.28 | stoffell_h | Assid, good question. If the date/time on the file remains the same, I guess it won't |
11:30.34 | Assid | hrmm.. maybe i should get snmp or something to check if my router is rx/tx any packets |
11:30.36 | Assid | and their speed |
11:31.38 | *** join/#asterisk Corydon76-home (i=purple@pdpc/supporter/sustaining/Corydon76-home) |
11:34.52 | *** join/#asterisk sulex (n=sulex@pdpc/supporter/active/sulex) |
11:38.22 | *** join/#asterisk Assid (n=assid@203.115.64.12) |
11:38.27 | Assid | sorry |
11:38.28 | Assid | got cut |
11:38.45 | Assid | okay its showing me a new screen... downloading new application |
11:39.07 | Assid | i hope it doesnt need to do this again and again |
11:46.18 | *** join/#asterisk Lino` (n=Lino@i577BF510.versanet.de) |
11:57.00 | *** join/#asterisk QbY (i=user@cm-12-146-225-117.dhcp.geo-sc.southerncoastalcable.net) |
11:57.30 | QbY | Anyone know of documentation that shows how you can fully utilize the display on Polycom phones? |
11:59.28 | stoffell_h | QbY, talking about 601? |
12:00.59 | QbY | stoffell_h.. yes |
12:02.22 | *** join/#asterisk Dr-Linux (n=Linux@202.59.73.131) |
12:10.27 | codebreaker | somebody know how to connect n boxes together and route calls to any number between them. no matter on wich box the client has registered. without use of dundi. also like a active-active setup. i know there was an easy way but i forgot |
12:13.07 | Assid | hrmm |
12:13.19 | Assid | how do put up a ringer of your own |
12:13.30 | Assid | i put it in saf.. i see it download |
12:13.32 | jerlique | can anyone suggest the a way to "link" 3 or more asterisk servers together, so that coupled with ser each * server knows about users which have registered on other * servers via the dispatcher module in ser |
12:15.56 | codebreaker | jerlique: if you get an answer please also forward it to me or juast give me a hilight |
12:16.27 | Assid | i cant get a new ringtone :( |
12:16.35 | jerlique | sure ;) |
12:16.58 | saftsack | is there anywhere a tutorial to install capi? |
12:18.30 | *** join/#asterisk Cybertoy (n=maxim@dsl254-123-241.nyc1.dsl.speakeasy.net) |
12:18.50 | Assid | does the 301 even support custom ringtones? |
12:23.07 | *** join/#asterisk xtr (i=Analogyo@S0106000c41ed11e1.vf.shawcable.net) |
12:24.09 | Assid | err.. i keep getting this noise every few mins.. i think its for the registration.. is there a way to stop it from making that noise every time it registers ? |
12:25.56 | *** join/#asterisk stuartcw (n=chatzill@softbank221025056004.bbtec.net) |
12:26.05 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219) |
12:26.49 | *** join/#asterisk Dr-Linux (n=Linux@202.59.73.131) |
12:28.40 | saftsack | hi has anyone of you chan_capi` |
12:28.41 | saftsack | ? |
12:31.46 | *** join/#asterisk n4y (n=tmalkut@host-ip2-24.crowley.pl) |
12:32.39 | stoffell_h | codebreaker, hm, also interested in your "n boxes" question.. |
12:32.53 | stoffell_h | Assid, it has something to do with the voicemail maybe? (chirping sound) |
12:38.41 | *** join/#asterisk Dr-Linux (n=Linux@202.59.73.131) |
12:39.47 | Assid | nah..not voicemail |
12:39.47 | Assid | its around the time i have my timeout/registration |
12:39.47 | Assid | 240 secs |
12:45.29 | codebreaker | stoffell_h: im asking me if nobody have "n boxes " connected together and as exmaple all clints a connecting to voip.mydomain and always get a different ip vi rrd-dns |
12:46.51 | codebreaker | there is so much written about failover trunks with switches etc.. but not how to setup all the boxes to do sharing the knowledge |
12:50.00 | stoffell_h | codebreaker, i'm not sure if it can be done in a good way, without dundi |
12:55.07 | *** join/#asterisk Dr-Linux (n=Linux@202.59.73.131) |
12:56.41 | codebreaker | stoffell_h: i think a had such a the time of asterisk ver 1.0.... and ist was really easy |
12:56.54 | *** join/#asterisk marv (n=marv@12-219-145-181.client.mchsi.com) |
12:57.40 | *** part/#asterisk Alystair (n=bob@CPE001109c15241-CM00407b8794db.cpe.net.cable.rogers.com) |
13:07.22 | *** part/#asterisk QbY (i=user@cm-12-146-225-117.dhcp.geo-sc.southerncoastalcable.net) |
13:10.28 | *** join/#asterisk juanjoc (n=juanjoc@248-32-235-201.fibertel.com.ar) |
13:30.23 | *** join/#asterisk austinnichols102 (n=austinni@dsl-10-169.cofs.net) |
13:34.42 | Assid | argh..how the hell do you get the last nmbder dialled? |
13:35.50 | [Airwolf] | Does anyone has some experience with call forwarding ? |
13:36.01 | [Airwolf] | Because I mad this extention: |
13:36.02 | [Airwolf] | http://pastebin.com/688553 |
13:36.06 | saftsack | does someone know the fritz isdn card? |
13:36.11 | [Airwolf] | And it seems to be working perfectly |
13:36.24 | [Airwolf] | But, if I dial the number it just isn't forwarded. |
13:37.32 | [Airwolf] | And I can't figure out why it isn't |
13:41.37 | *** join/#asterisk darby_t (i=darby_t@aapn121.neoplus.adsl.tpnet.pl) |
13:45.34 | *** join/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.233.134.Dial1.SanJose1.Level3.net) |
13:48.55 | *** join/#asterisk darby_t (i=darby_t@aapn121.neoplus.adsl.tpnet.pl) |
13:50.00 | *** join/#asterisk joat (n=joat@ip70-160-147-169.hr.hr.cox.net) |
13:52.31 | Assid | how do i know if i have a poe cable or no |
13:52.46 | *** join/#asterisk wunderkin (i=kev@ip68-226-113-228.ph.ph.cox.net) |
13:54.04 | austinnichols102 | Here's the URL for reference: http://www.voip-info.org/wiki-Asterisk+phone+cisco+79xx |
13:54.41 | Assid | i got this black fat patch cord |
13:54.55 | austinnichols102 | They're saying that 8.2 works it just that the CID displays IP addy |
13:55.17 | austinnichols102 | damn - wrong window |
13:56.05 | Assid | and some extra thing.. which seems that it should be clipped onto a a cable |
13:59.24 | *** join/#asterisk Lino` (n=Lino@i577BD976.versanet.de) |
14:00.54 | Assid | oh.. i need a special kind of switch to use PoE ? |
14:03.45 | coppice | well, power doesn't appear by magic, or the oil companies would be out of business :-) |
14:05.15 | file | coppice: it doesn't?!? someone lied to me!!! |
14:05.47 | *** part/#asterisk darby_t (i=darby_t@aapn121.neoplus.adsl.tpnet.pl) |
14:06.36 | luke-jr_ | coppice: oil needs to be made somehow, so oil has its source too ;) |
14:06.46 | luke-jr_ | coppice: eventually, it all goes back to the Sun, which might as well be magic |
14:07.25 | coppice | the oil companies just steal the oil from the dinosaurs it really belongs to |
14:07.43 | luke-jr_ | pfft, oil isn't from dinosaurs =p |
14:08.10 | luke-jr_ | that's fiction |
14:08.36 | luke-jr_ | not that I could tell you where it *is* from in reality o.o |
14:09.05 | ManxPower | My car gets 358 dead dinos to the mile! |
14:09.13 | russellb | luke-jr_: magic |
14:09.21 | luke-jr_ | russellb: could be |
14:09.26 | luke-jr_ | for certain definitions of magic |
14:09.45 | russellb | where magic is defined as the explanation for things you don't understand |
14:10.19 | luke-jr_ | or potentially where it's defined as things which are outside the realm of science |
14:10.34 | russellb | oil is not outside of the realm of science |
14:11.34 | luke-jr_ | its origin could be, but I don't know |
14:11.43 | luke-jr_ | could be part of the original creation |
14:12.04 | russellb | asterisk is powered by oil, you know ... |
14:12.12 | luke-jr_ | cool |
14:12.40 | luke-jr_ | it runs fine w/o oil for me tho |
14:13.48 | coppice | it creaks a bit without lubrication, though |
14:14.19 | *** join/#asterisk mutilator (i=WebChat@65.111.201.122) |
14:14.35 | ManxPower | Yay! The ditch witch works! |
14:14.44 | mutilator | mornin everyone |
14:14.50 | mutilator | how goes saturday |
14:14.53 | ManxPower | Almost too well, actually, within 2 hrs of getting it they cut one of the water lines. |
14:15.20 | luke-jr_ | hm, wonder if a digital camera can take pics w/o internal storage via Wifi... |
14:15.33 | Qwell | luke-jr_: Sure, why not? |
14:15.38 | *** join/#asterisk VeNoMouS_ (n=jj@202.162.177.196) |
14:15.39 | Qwell | Just don't expect it to be very fast |
14:15.41 | *** join/#asterisk ApEtc (i=apetc@ip70-162-216-7.ph.ph.cox.net) |
14:15.46 | VeNoMouS_ | asterisk in the news in nz again |
14:15.48 | VeNoMouS_ | http://computerworld.co.nz/news.nsf/news/3E42037A86813A51CC25715D00105063 |
14:16.02 | VeNoMouS_ | tsk tsk nawty phisers using asterisk |
14:16.05 | ManxPower | VeNoMouS_, Great, more newbies. |
14:16.16 | VeNoMouS_ | VoIP services are appealing because they allow customers to set up numbers anywhere in the globe. And because they can be combined with telephone software like the open-source Asterisk PBX (Private Branch Exchange) product, it can be inexpensive for thieves to set up a professional-sounding line. |
14:17.20 | *** join/#asterisk shido6 (n=shido6@d38-45-81.commercial1.cgocable.net) |
14:17.31 | mutilator | i dun think newbs do that sort of thing.. |
14:17.41 | mutilator | newbs can;t even make a dial plan |
14:17.52 | VeNoMouS_ | lol |
14:18.15 | shido6 | help them out. |
14:18.28 | luke-jr_ | Qwell: my actual goal is to do GPS-capable photos but have the GPS usable from a laptop ;) |
14:18.33 | VeNoMouS_ | heh man, ive pretty much finshed most of my managment frontend for our cdr shit |
14:18.35 | Qwell | shido6: I think in this case, helping them would be a bad idea. :p |
14:18.36 | Qwell | <VeNoMouS_> http://computerworld.co.nz/news.nsf/news/3E42037A86813A51CC25715D00105063 |
14:18.37 | Qwell | <VeNoMouS_> tsk tsk nawty phisers using asterisk |
14:18.55 | shido6 | thanks qwell |
14:18.58 | shido6 | where's my foot |
14:19.05 | Qwell | in your mouth, I imagine. ;) |
14:19.09 | shido6 | :) |
14:19.09 | VeNoMouS_ | write mixmonitor to file, transcode to mp3, insert mp3 into mysql, stream mp3 on the fly from mysql via php |
14:20.10 | VeNoMouS_ | heh so i can list all the calls for the day, and u can play back each call |
14:20.41 | VeNoMouS_ | Qwell wanna see? |
14:20.55 | Qwell | see what? |
14:21.00 | Qwell | oh |
14:21.11 | Qwell | nah, too early |
14:21.15 | VeNoMouS_ | lol |
14:21.16 | VeNoMouS_ | Sun Apr 30 02:20:47 NZST 2006 |
14:21.19 | VeNoMouS_ | its late for me |
14:24.21 | VeNoMouS_ | wow its so talkie in here! |
14:25.37 | *** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239) |
14:26.13 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219) |
14:29.59 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
14:34.25 | Assid | err.. i keep having this irritating sound .. every few mins.. i think its of the phone registering every 240 seconds |
14:34.30 | blitzrage | anyone have experience with polycom phones? I'm wondering if it's possible to block the call forwarding feature on the phone itself -- or, alternatively -- to detect a 302 redirect in asterisk (1.2.x) |
14:34.32 | Assid | but can i have it disable that noise |
14:37.11 | *** join/#asterisk ComputerWarm (n=donc@HS196-230-97.nt.net) |
14:37.23 | shido6 | my spa blinks at me everytime it registers |
14:37.27 | shido6 | so i set it for 3600 |
14:37.32 | ComputerWarm | Hello everyone |
14:37.40 | Assid | err.. okay its ever 3 odd minds |
14:37.54 | Assid | which still isnt 240 seconds |
14:38.25 | ComputerWarm | I have a question I installed h323 that comes with asterisk, i got no errors while i was compiling but its still not installed.... I did as the instructions said |
14:38.37 | ComputerWarm | But when i try to load it. there is no apps for it |
14:38.48 | shido6 | then you didnt follow the directions |
14:38.50 | shido6 | :) |
14:39.09 | ComputerWarm | ya i did everything the Readme and install file said |
14:39.10 | Assid | <PROTECTED> |
14:39.23 | shido6 | most ppl say that and miss a line or two :) |
14:39.41 | ComputerWarm | let me start from the beginning again |
14:39.46 | shido6 | blow away all remnance of h323 and read the README again |
14:40.18 | nrw | blitzrage: do you want to disable the ability to do the redirect, or force the phone to take the call even with call forwarding on |
14:40.49 | Assid | whats lineSeize for? |
14:41.29 | ManxPower | Assid, are you sure it's not the MWI indication ring? |
14:41.50 | Assid | how do i know? |
14:42.12 | ManxPower | well, does the SIPura only do a partial ring when you have voicemail? |
14:42.21 | Assid | its polycom 301 |
14:42.33 | ManxPower | Assid, so what is the problem? |
14:42.44 | *** join/#asterisk MacDome (n=eseidel@c-71-198-177-144.hsd1.ca.comcast.net) |
14:42.48 | Assid | it keeps beeping every minute or so |
14:42.54 | ManxPower | does the annoying sound only happen when the MWI light is blinking? |
14:43.33 | Assid | you mean the red light on top |
14:43.40 | Assid | i thought thats power light |
14:43.59 | ManxPower | noname32, that's the MWI light. |
14:44.05 | ManxPower | <PROTECTED> |
14:44.07 | *** join/#asterisk MacDome (n=eseidel@c-71-198-177-144.hsd1.ca.comcast.net) |
14:44.14 | Assid | damn |
14:44.14 | Assid | yes |
14:44.15 | ManxPower | that "annoying sound" is to tell you that you have voicemail. |
14:44.18 | Assid | its cause of that |
14:44.26 | Assid | just curious tho |
14:44.47 | ManxPower | I managed to get rid of the annoying sound, but it took 2 days and reading the Admin guide a zillion times. |
14:45.04 | Assid | where do i program a button for voicemail.. |
14:45.19 | ManxPower | Assid, read the admin guide. |
14:45.31 | ManxPower | and the button for voicemail is only for CALLING voicemail. |
14:46.02 | ManxPower | It's for people too lazy to dial the voicemail extension on your Asterisk system. |
14:46.30 | Assid | yeah i know |
14:46.53 | Assid | it works for a 501 i remotely configured |
14:46.57 | *** join/#asterisk Zeeek (n=IceChat7@pdpc/supporter/active/Zeeek) |
14:47.00 | Assid | just dont know what to do on my 301 |
14:47.13 | Assid | never seen a 501 in hand .. was configuring my friends box |
14:47.15 | ManxPower | IN FACT, programming the voicemail button was discussed on -users within the past 14 days. I guess you must have missed that thread. |
14:48.06 | Zeeek | gentlemen... |
14:48.31 | ManxPower | Zeeek, someone in Paris was looking for an Asterisk consultant. |
14:48.38 | Zeeek | yes, we know him |
14:48.52 | Zeeek | not that I'm a "consultant" anyway ;) |
14:49.03 | ManxPower | Ah. OK. |
14:49.11 | Zeeek | we have an asterisk group now meeting every month |
14:49.15 | Zeeek | but thanks! |
14:49.46 | Zeeek | I'm having tollfree DID woes at the moment |
14:50.08 | Zeeek | how's thing with you, Manx ? |
14:51.10 | *** join/#asterisk yxa (i=lonari@cm121.gamma228.maxonline.com.sg) |
14:51.10 | ManxPower | Zeeek, Pretty good. Living on a mountian now. 8-) |
14:51.19 | Zeeek | heh, understandable |
14:51.27 | ManxPower | No more fleeing Storms of Doom |
14:51.40 | Zeeek | now you'll be struck by lightning ! |
14:51.40 | ManxPower | I hope to have my cabin within 2 months. |
14:51.52 | Assid | hrmm.. 301 doesnt have a seperate button for messages |
14:52.13 | coppice | "No more fleeing Storms of Doom" sounds like you stopped playing some addictive computer game |
14:52.36 | ManxPower | coppice, we started calling Katrina "Katrina: Storm of Doom" after a while. |
14:52.49 | nrw | assid: no it does not |
14:52.57 | nrw | you can assign the second line key to it if you so wish |
14:53.18 | coppice | ManxPower: typically american. in the UK it would have been referred to as "the breeze" |
14:53.35 | Zeeek | chin wot |
14:53.43 | ManxPower | Uh uh. |
14:54.12 | Zeeek | in new jersey it might have been know as Da Breeze |
14:54.24 | ManxPower | I'll bet they play sports without padding and eat rare meat too! |
14:54.52 | Assid | i wonder whats the price difference between 301/501 |
14:55.05 | ManxPower | Assid, that would depend on where you get them from. |
14:55.19 | nrw | assid: about 150 bucks |
14:55.21 | Splas | hrm... my 7960 seems to append the domain of the SIP server to the callerid num of inbound calls... |
14:55.28 | nrw | actually about 120 |
14:55.45 | [TK]D-Fender | nrw : 120 WHAT? |
14:55.52 | coppice | come on. the UK calls the atlantic ocean the pond, and some dislike that kind of overstatement. storm of doom would only be good for laugh |
14:55.55 | Assid | 120 bucks for polycom301? |
14:56.05 | nrw | v<Assid> i wonder whats the price difference between 301/501 |
14:56.11 | nrw | <nrw> assid: about 150 bucks |
14:56.18 | nrw | <nrw> actually about 120 |
14:56.22 | [TK]D-Fender | nrw : The difference isn't 120$ in USD or CAD.... |
14:56.27 | [TK]D-Fender | not by a long shot |
14:56.27 | ManxPower | coppice, well, until you live thru one. |
14:56.39 | Assid | so its $80 for a 301 ? |
14:56.51 | [TK]D-Fender | IP 301 = $115 USD, IP 501 = $170 USD |
14:56.55 | nrw | wow |
14:57.01 | ManxPower | Assid, call Gauston at Avenue Computer Supply, |
14:57.01 | coppice | the US will probably live through some more this year |
14:57.03 | nrw | where the heck are they those prices? |
14:57.18 | Assid | hrmm |
14:57.19 | [TK]D-Fender | nrw : Leanr to shop around.... www.atacomm.com |
14:57.20 | ManxPower | coppice, yup. |
14:57.26 | ManxPower | Let me find his contact info. |
14:57.28 | Assid | they shoulda bought me a 501 :( |
14:57.42 | [TK]D-Fender | I got mine for $135 CAD and $200 CAD respectively. |
14:58.13 | Assid | now the only thing i gotta get working is this ringtone |
14:58.19 | Assid | i wanna get the CTU ringtone in there |
14:58.20 | *** join/#asterisk jeebusroxors (n=jeebusro@29palms-cuda1-68-170-36-65.losaca.adelphia.net) |
14:58.31 | Assid | it just doesnt work :( |
14:59.00 | nrw | my 5012 cost 250 bucks |
14:59.04 | nrw | 501s even |
14:59.07 | nrw | sigh |
14:59.23 | ManxPower | Assid, call Gaston Gureau at Avenue Computer Supply, 225-615-7297 Tell him "Eric" sent you. I do not get a comission, but he is a friend. |
14:59.41 | ManxPower | I *THINK* he has decent prices on Polycoms, you should shop around. |
14:59.58 | ManxPower | nrw, got them on the greymarket, eh? |
15:00.07 | Assid | nice.. will keep it in mind |
15:00.14 | Assid | im actually in india.. |
15:00.40 | ManxPower | I doubt he'll ship outside the USA |
15:01.07 | *** join/#asterisk Kernel_Core (n=I@193.251.135.118) |
15:01.08 | Assid | yeah |
15:01.13 | Assid | thats wheree alot of my time went |
15:01.16 | Assid | i was waiting for the phone |
15:01.23 | nrw | manxpower: im not sure where things really come from. we have a purchasing guy that handles all that stuff. i just have to count the cost against my budget |
15:01.24 | Assid | for someone to get it |
15:01.40 | nrw | and i could have gotten a few more phones if i knew they were 80 bucks less than we got them for |
15:01.58 | ManxPower | nrw, Avenue has decent prices on a lot of stuff. |
15:02.24 | ManxPower | but for the really cheap stuff, best to go with someone else. |
15:02.33 | *** join/#asterisk Abydos313 (i=abydos31@ppp-71-133-210-73.dsl.irvnca.pacbell.net) |
15:03.12 | Assid | hrmm.. gonna keep that avenues in mind |
15:03.17 | Assid | for the guys i work for |
15:03.22 | Assid | they always need more phones |
15:03.26 | ManxPower | As I said, I doubt he will ship outside the USA |
15:03.44 | Assid | yeah.. those guys are in the US |
15:04.07 | ManxPower | We get special pricing from Avenue, so I can't tell you what is standard price is for polycoms |
15:04.55 | ManxPower | I doubt they will provide support, other than to maybe provide firmware updates on request. |
15:05.33 | Abydos313 | anyone know how to setup/check voicemail with telasip? |
15:13.39 | Zeeek | Manxpower didn't voipsupply once have good prices on Poly? |
15:14.00 | Zeeek | they have good support and do ship internationally |
15:14.20 | [TK]D-Fender | Zeeek : compared to others "worse" prices, yes. www.antonline.com has the lowest I've seen to date. |
15:14.42 | [TK]D-Fender | VoIP Supply is good for the all-around service though. |
15:14.59 | Zeeek | I can testify that they did a good job for us on a return |
15:15.11 | Zeeek | and make firware available to customers |
15:15.13 | *** join/#asterisk heka (n=heka@82.114.68.123) |
15:15.19 | Zeeek | kinky, that firware |
15:15.29 | Zeeek | wooden trousers and all that |
15:16.28 | Zeeek | does anyone have tollfree DID and if so who is providing them? |
15:17.20 | ManxPower | Zeeek, no idea. |
15:17.27 | heka | didx had something similar! but Im not so sure. |
15:17.35 | Zeeek | didx is a broker |
15:17.43 | ManxPower | Zeeek, teliax for me |
15:18.01 | ManxPower | Of course, all ITSPs suck. Teliax seems to suck less than most. |
15:18.19 | Zeeek | I had an account at one time. They do tollfree DID? |
15:18.40 | Zeeek | <going to look> |
15:19.00 | *** join/#asterisk telenieko (n=marc@167.Red-80-35-144.staticIP.rima-tde.net) |
15:19.09 | Zeeek | yeah they do |
15:19.14 | telenieko | anybody here knows how does the bristuff code work ? :o) |
15:21.06 | jake1932 | ManxPower: that's misleading to say unlimited on https://www.teliax.com/newaccount/?r=1&cp=default |
15:21.49 | ManxPower | jake1932, they disclose their softlimits. Like ALL ITSPs, unlimited is just a marketing gimick |
15:23.11 | jake1932 | i can understand an "Unlimited" Plan - but they're showing a table of what is included - and the softcaps below - seems pretty cheezy |
15:24.10 | *** join/#asterisk gmaruzz (n=gmaruzz@217-133-80-112.b2b.tiscali.it) |
15:26.01 | *** part/#asterisk gmaruzz (n=gmaruzz@217-133-80-112.b2b.tiscali.it) |
15:26.05 | *** join/#asterisk gmaruzz (n=gmaruzz@217-133-80-112.b2b.tiscali.it) |
15:26.42 | jake1932 | here's a new marketing idea - advertise what you actually sell - and stop with the bait and switch |
15:27.23 | jake1932 | (not taking away from the service - which from what you said is good |
15:28.07 | jake1932 | ) |
15:33.32 | *** join/#asterisk Flosoft (n=admin@213.219.160.100) |
15:33.33 | Flosoft | hey |
15:33.45 | Flosoft | I am looking for an Asterisk Frontend |
15:33.52 | Flosoft | which I can install on a Plesk Server |
15:34.06 | Qwell | I hear vi is a good frontend |
15:34.12 | [TK]D-Fender | Try vi, emcs, pico, nano, mc, |
15:34.24 | Flosoft | well ... maybe something which is nicer ;) |
15:34.31 | Flosoft | like: Iritgo |
15:34.41 | [TK]D-Fender | gedit, kedit, OOo Writer? |
15:34.47 | jake1932 | ultraedit - on windows |
15:34.57 | Flosoft | a Non-texteditor interface |
15:34.59 | Qwell | MS Word? |
15:35.03 | Qwell | Excel? |
15:35.05 | [TK]D-Fender | hehehe, ok, I'm outta here |
15:35.11 | [TK]D-Fender | Qwell : No... Word sucks :) |
15:35.23 | [TK]D-Fender | later! |
15:35.36 | Flosoft | any constructive help? |
15:36.06 | jake1932 | AMP - try #freepbx |
15:36.20 | Flosoft | well ... it is difficult on plesk I was told |
15:36.33 | jake1932 | it's a PIA on anything |
15:36.41 | Flosoft | PIA? |
15:36.48 | jake1932 | easiest to just edit the files you need with a text editor |
15:36.59 | Qwell | PITA? |
15:37.10 | jake1932 | ~PITA |
15:37.11 | jbot | somebody said pita was pain in the ass |
15:37.38 | Zeeek | wrong! It's bread |
15:38.03 | Flosoft | hehe |
15:38.13 | jake1932 | ~PIA |
15:38.15 | jbot | pia is, like, ask me about pita |
15:38.17 | Qwell | jbot: pita is also a bread-like food |
15:38.19 | jbot | Qwell: okay |
15:38.42 | *** join/#asterisk Samoied (n=Samoied@201-35-214-13.fnsce703.dsl.brasiltelecom.net.br) |
15:39.07 | Flosoft | well ... I really like Freepbx |
15:39.19 | Flosoft | but to install it on Plesk is quite difficult it seems |
15:39.22 | coppice | Qwell: isn't that pitta? |
15:39.29 | Qwell | coppice: no, don't think so |
15:39.32 | Qwell | jbot: define pita |
15:39.45 | Qwell | at least, not in English |
15:39.47 | Flosoft | Talima / Iritgo seems to be perfect ... but I can't find it anymore |
15:40.26 | Qwell | Pita (also called pitta or pita bread) is a round flat wheat bread made with yeast. It is traditional in many Middle Eastern and Mediterranean cuisines and is believed to have originated in Ancient Greece. |
15:40.42 | Qwell | well, I guess both are acceptable...the former is the preferred spelling though |
15:40.47 | *** join/#asterisk pardove (n=pardove@217.219.250.24) |
15:41.00 | pardove | is this command valid in extensions.conf: switch => SIP/user:secret@server/context |
15:41.02 | jake1932 | flosoft - you probably won't get any help in here on that - check http://www.voip-info.org/wiki-Asterisk+GUI |
15:41.18 | Qwell | pardove: I don't think so, no |
15:41.20 | Flosoft | thx... i'll take a look |
15:41.32 | coppice | Qwell: a google search seems to conflict with you. pitta bread gets massive hits, like www.nutrition.org.uk/upload/bread%20pitta.pdf |
15:41.32 | Qwell | pretty sure you can't do SIP switch... |
15:42.02 | Qwell | http://googlefight.com/index.php?lang=en_GB&word1=pita&word2=pitta |
15:42.07 | Qwell | sorry, pita wins :P |
15:42.09 | pardove | can i have video calls when using iax trunks? |
15:43.19 | coppice | Qwell: you didn't qualify it with bread. you got all the GWBush references with pita :-) |
15:43.25 | *** join/#asterisk Laureano (n=Laureano@host172046.metrored.net.ar) |
15:43.28 | Qwell | :P |
15:43.29 | Laureano | Hello |
15:43.57 | Qwell | pita bread still wins, by a significant amount |
15:44.05 | pardove | Qwell: can i have video calls when using iax trunks? |
15:44.17 | Laureano | Did anyone uses VoipTalk with IAX? |
15:44.22 | Zeeek | yes |
15:44.23 | Qwell | pardove: I don't know. and if I did, I would have answered the first time you asked, less than 2 minutes ago |
15:44.35 | Zeeek | Laureano might be off today |
15:44.42 | Zeeek | there was a recent change |
15:44.42 | pardove | Qwell: sorry ;-) |
15:44.58 | Laureano | I supose that. |
15:44.59 | *** join/#asterisk gr0mit (n=guest@extrt.txrx.org.uk) |
15:45.07 | Zeeek | you have problems? |
15:45.17 | Laureano | I see in the support page that we have to change the server to iax5.voiptalk.org |
15:45.28 | Zeeek | I chenged it and now nothing works |
15:45.43 | Zeeek | I'm guessing they are doing some work now or something |
15:45.49 | Laureano | Ohhh |
15:45.56 | Zeeek | Just a guess |
15:46.04 | Laureano | I guess the same. |
15:46.15 | Zeeek | well now we know there are 2 of us :) |
15:46.26 | Laureano | Because yesterday, for some moments, I can't do outgoing calls. |
15:46.30 | Zeeek | aha |
15:46.42 | Zeeek | I don't use them much, only for europe |
15:46.48 | Laureano | Now it works. But the incoming calls are broken. |
15:47.00 | Zeeek | it works since when? |
15:47.08 | Laureano | When I try to pass audio in the call, I get an "Invalid Call" from IAX. |
15:47.47 | Laureano | I guess that the outgoing calls are working since last hours of Yesterday. |
15:47.59 | Laureano | (Please note that I'm in Argentina) |
15:48.02 | Zeeek | ok |
15:48.16 | Laureano | But, its good to see that I'm no the only one with problems. |
15:48.22 | Laureano | Thank you Zeeek |
15:48.34 | Zeeek | well if it continues, I'll have to contact the |
15:48.35 | heka | anybody can help me modify an perl agi? |
15:48.46 | Zeeek | Laureano no problem |
15:48.55 | *** join/#asterisk n4y (n=tmalkut@host-ip2-24.crowley.pl) |
15:49.01 | Laureano | I try to contact them, but apparently they work Mon-Fri,9:00-18:00 |
15:49.17 | Laureano | A least the service number that I can dial from here. |
15:49.33 | Laureano | Maybe the UK customers can call right now, but not me. |
15:49.45 | Zeeek | I've never tried to call them |
15:50.05 | Laureano | Well, that was the first time I tried. |
15:50.29 | Laureano | BRB |
15:52.31 | Laureano | Back |
15:52.44 | *** join/#asterisk darby_t (i=darby_t@aapc207.neoplus.adsl.tpnet.pl) |
15:52.59 | Laureano | Well, I think that its just a matter of time to be working again. Lets wait. |
15:53.20 | Zeeek | Laureano is outgoing working in IAX right now for you? |
15:54.13 | Laureano | Yes. |
15:54.31 | Laureano | I use the VoipTalk account to make calls to UK, and works fine. |
15:54.37 | Zeeek | ok |
15:54.44 | Zeeek | I'll check later |
15:54.54 | Laureano | Aren't working for you? |
15:55.09 | jake1932 | heka - although i probably can't help you with that, usually the guys on here respond better if you ask a very specfic question about your perl agi - like, what you're trying to do with it |
15:55.17 | Zeeek | it was not working a few hours ago. I can't check now because the phones are in use |
15:56.20 | Laureano | But... you can't check it because you can dial to VoipTalk? |
15:56.35 | *** join/#asterisk vittogio (n=vittogio@host173-40.pool8259.interbusiness.it) |
15:56.50 | vittogio | hi all |
15:57.14 | vittogio | i have a critical issue on my asterisk server, it is there someone that could hep me? |
15:57.26 | *** join/#asterisk salviadud (n=ralfalfa@dsl-200-78-64-10.prod-infinitum.com.mx) |
15:57.36 | heka | jake1932: did it by myself. thank you. |
15:58.29 | Laureano | Tell us about your problem I we will see if we can help you vittogio |
15:58.51 | Laureano | Zeeek, Why don't you use the Asterisk Auto Dialer? |
15:59.28 | vittogio | ok, i have 2 asterisk server connected via the PRI port, very often this link get a reset and all calls are dropped |
15:59.49 | jake1932 | vittogio: did you swap the cable yet? |
16:00.21 | vittogio | yes, i did and i checked the signalling, master/slave also |
16:00.36 | jake1932 | irq issues maybe? |
16:01.04 | Laureano | When the servers drop the calls. Do you see something in the Asterisk console or in the full log? |
16:01.27 | vittogio | i made a check and "unfortunately" there were no irq issue |
16:01.45 | vittogio | there is RELEASED |
16:02.18 | Laureano | The span goes down and then up? |
16:02.25 | vittogio | it is like the D-channel has some issue, also because it seems that the calls move to full duplex to half duplex audio |
16:02.36 | vittogio | yes the span do down and then up |
16:02.52 | vittogio | i have no IP trunk on the system |
16:03.17 | Laureano | Ok. I think that I know what can be, but I can't find the link with the solution. Give me 5 minutes. |
16:03.27 | vittogio | ok |
16:04.31 | jake1932 | vittogio - while he's looking - how come you used two PRI cards to link the servers, and not IP? |
16:05.29 | vittogio | we have a point to point connection between 2 sites via a Clear channel |
16:05.43 | jake1932 | ok |
16:06.08 | mutilator | does echo can settings in zapata reload when you reload |
16:06.11 | mutilator | or does it need a restart? |
16:06.37 | vittogio | in the zapata the echocancel is set to no |
16:07.40 | Laureano | mutilator, You need to restart. |
16:08.20 | vittogio | i.e. i have no problem with the connection to the PSTN network, i mean no random restarts |
16:10.01 | Laureano | vittogio, Do you use the "pri intese debug" command? |
16:10.10 | Laureano | That always help. |
16:10.12 | *** join/#asterisk Ariel_ (n=Ariel@adsl-068-157-125-248.sip.mia.bellsouth.net) |
16:10.16 | vittogio | yes |
16:11.34 | Laureano | And no clues with that? |
16:12.11 | vittogio | i mean you can see that a reset occurred by the remote party |
16:12.37 | vittogio | we saw the involved timer, the T308 |
16:13.11 | vittogio | but if i connect 2 legaycy pbx over the same link, everythink is working fine |
16:14.23 | vittogio | i have also connected 1 asterisk server to an Alcatel PBX usingthe same link and wha i saw is that the system has been able to locate channel over the 30s regularly used by an E1 |
16:15.01 | vittogio | it seems to be something related to the software hdlc implementation |
16:16.46 | Laureano | Do you see any HDLC abort errors? |
16:16.56 | vittogio | yes |
16:17.14 | vittogio | HDLC Frame(8) |
16:17.21 | vittogio | sorry FCS(8) |
16:17.28 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
16:17.47 | coppice | what makes you think it is anything to do with the software HDLC? |
16:19.00 | Laureano | vittogio, please check this: http://lists.digium.com/pipermail/asterisk-users/2005-September/118021.html |
16:21.18 | vittogio | currently the resetinterval =never |
16:21.39 | vittogio | as i saw that the range of the value starts from 60s |
16:21.53 | vittogio | do you think that "0" is accepted? |
16:22.50 | vittogio | coppice: i tried many setting over the zapata and zaptel, so i thought that should be something related to the software |
16:23.41 | coppice | you have a problem which everyone else doesn't, so the software must be wrong. well, that seems a logical deduction. :-) |
16:24.44 | vittogio | when you have issue you try to figure out "all" the possible causes"..... |
16:25.38 | coppice | its quicker if you start with the likely ones. commonest reason for link errors is frame slips. are you slaving to the clock from the PBX? |
16:26.12 | vittogio | i tried 2 different situation |
16:26.16 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219) |
16:26.20 | Laureano | vittogio, This link mayba also can help: http://www.ctunion.com/node/95 |
16:26.42 | vittogio | the first is to connect asterisk to a legacy pbx and hget the clock from the last one |
16:26.54 | vittogio | the other is having 2 asterisk server connected together |
16:27.01 | vittogio | the issue is the same |
16:27.27 | Flosoft | has anyone got experience with voiceone |
16:27.29 | Flosoft | ??? |
16:27.39 | coppice | what did you put in zaptel.conf? |
16:27.40 | Laureano | Flosoft, No, sorry. |
16:29.18 | vittogio | Change the server... not so easy... |
16:29.58 | vittogio | i read the article in the past, i was was also thinking the same, but is something i cannot perform any action |
16:30.05 | vittogio | in the zaptel? |
16:30.29 | Laureano | I know that isn't easy to change a server. But, did you test with a loopback cable, for example? |
16:31.31 | vittogio | with a loopback cable you get that: you are master and also the other party ... or slave |
16:31.41 | *** join/#asterisk alephcom (n=alephcom@host75.net14.mcsnet.ca) |
16:31.46 | vittogio | i made the following |
16:32.00 | vittogio | i ran some calls using the outgoing |
16:32.04 | Laureano | I know that. But you didn't see any error? |
16:32.50 | vittogio | the error is occurring only when the channel is fully busy and other calls are trying to access to it |
16:33.05 | Laureano | Aha |
16:33.32 | vittogio | i tried to replicate the errors on local making auto calls using the .call file |
16:33.38 | vittogio | well no issue |
16:34.18 | Laureano | So. If you have all the 30 channels of the trunk in use, and then you try to place a new call; the trunk goes down and up? |
16:34.24 | vittogio | the span in the master server is set: span=1,0,2,ccs,hdb3 |
16:34.39 | vittogio | after 15/20 minutes |
16:35.41 | vittogio | on the debug what is missing is the time: when the error occurs.... |
16:35.56 | Laureano | And this didn't happen when you connect * to the Legacy PBX? |
16:36.44 | vittogio | this is happening only with 2 asterisk server connected each other or when as asterisk server is connected to a legacy pbx |
16:37.13 | Laureano | Aha |
16:37.16 | vittogio | is i connect 2 legacy pbx together (Alcatel and Ericsson) no disconnection |
16:37.42 | vittogio | i mean it is very very strange |
16:37.47 | Qwell | sounds like your timing is pretty wrong |
16:37.56 | vittogio | yeah, it is |
16:38.12 | Laureano | If you were only using *2* connection, I would suggest you to control the used channels with CheckGroup(). But... |
16:38.26 | coppice | vittogio: you are whining a lot, but not providing any information. |
16:38.57 | vittogio | information like? |
16:39.21 | coppice | what is in your zaptel.conf. you only erplied with part of the answer |
16:39.27 | vittogio | ok |
16:40.57 | vittogio | span=1,1,2,ccs,hdb3 - span=2,1,2,ccs,hdb3 - span=3,1,2,ccs,hdb3 - span=4,0,2,ccs,hdb3 |
16:41.23 | vittogio | #1 and # 2 connected to the PSTN |
16:41.36 | Zeeek | Laureano are you using the new server for voiptalk? It still isn't working for me |
16:41.37 | vittogio | #3 cnnected to Alcatel (that is master) |
16:41.52 | *** part/#asterisk alephcom (n=alephcom@host75.net14.mcsnet.ca) |
16:41.52 | coppice | so, you have 3 spans all set to be the primary clock source. that doesn't make much sense |
16:42.07 | coppice | you don't have crc4 set, which is normal for ISDN E1s |
16:42.29 | vittogio | crc4 is not required by the provider, so i cannot set it up |
16:42.37 | Laureano | Zeeek, Yes, I'm using iax5.voiptalk.org. But, the 2 domains point to the same IP address. |
16:42.56 | vittogio | you see 1 in al the spans as Digium told me to set them in this way |
16:43.00 | Zeeek | interesting, I didn't even check that :) |
16:43.03 | coppice | well, E1s without CRC4 are used in many places, and they are quirky. CRC4 was added to the E1 spec for a reason |
16:43.27 | *** part/#asterisk ComputerWarm (n=donc@HS196-230-97.nt.net) |
16:43.28 | vittogio | i know but the provider is not using it |
16:44.42 | coppice | change your zaptel.conf so you don't have multiple E1 set to the same priority. I don't know if matters now, but it used to. quirky things used to happen |
16:45.43 | vittogio | this is what i knew but i got a mail from digium saying to use 1 (slave) and 0 for master, avoiding the use of 2..3 and so on |
16:46.17 | vittogio | you are suggesting to use 2 for the span#3, right? |
16:46.35 | wunderkin | vittogio, who at digium told you to do that? |
16:46.44 | vittogio | the support |
16:46.45 | coppice | of course not. you want that to be your clock source, so it needs to be 1 |
16:47.11 | *** join/#asterisk IceManRISK (n=kart@200.138.91.1) |
16:47.20 | vittogio | the clock source should come from the provider |
16:47.26 | wunderkin | heh, man this is rough, who at support |
16:47.52 | coppice | I thought you said 3 was to be your clock master? |
16:48.04 | vittogio | solution but not the man, it is not correct |
16:48.52 | *** join/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.21.26.Dial1.SanJose1.Level3.net) |
16:49.28 | vittogio | the clock source is the alcatel, but it is preferred to use the PSTN as clock source |
16:49.46 | vittogio | so do you think that i should better mange the clock between the PBX? |
16:51.07 | vittogio | the problem is coming from the fact that at 1 side i have the PSTN and from the other i have the other PBX |
16:51.32 | *** join/#asterisk relateness (n=relatene@81.52.161.78) |
16:52.01 | coppice | you need something like: |
16:52.03 | coppice | span=1,1,0,ccs,hdb3 |
16:52.04 | coppice | span=2,2,0,ccs,hdb3 |
16:52.06 | coppice | span=3,3,0,ccs,hdb3 |
16:52.08 | coppice | span=4,0,0,ccs,hdb3 |
16:52.37 | relateness | Hi, can any one help me to find a good introduction to asterisk |
16:52.43 | Qwell | ~docs |
16:52.44 | jbot | methinks docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
16:52.53 | vittogio | same issue with that it was my first configuration |
16:53.31 | coppice | is the PBX set to slave to you? |
16:53.53 | vittogio | no |
16:54.17 | coppice | is the PBX locked to the PSTN? |
16:54.29 | vittogio | yes |
16:54.35 | *** join/#asterisk apardo (n=apardo@87.217.145.245) |
16:54.48 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219) [NETSPLIT VICTIM] |
16:54.49 | *** join/#asterisk florz (n=florz@2001:1a50:503c:0:0:0:0:1) [NETSPLIT VICTIM] |
16:54.49 | *** join/#asterisk luke-jr_ (n=luke-jr@2002:1891:f657:0:20e:a6ff:fec4:4e5d) |
16:54.49 | *** join/#asterisk Sedorox (n=Brandon@smartserv/cna/Sedorox) [NETSPLIT VICTIM] |
16:54.49 | *** join/#asterisk flynux (i=glyng5y@cl-8.bru-01.be.sixxs.net) [NETSPLIT VICTIM] |
16:54.49 | *** join/#asterisk RaYmAn-Bx (i=rayman@skumler.dk) [NETSPLIT VICTIM] |
16:54.49 | *** join/#asterisk Ironhand (i=arjen@meek.xs4all.nl) [NETSPLIT VICTIM] |
16:54.50 | *** join/#asterisk macTijn (i=martijn@linda.net.insecure.nl) [NETSPLIT VICTIM] |
16:55.09 | relateness | thank you ! |
16:56.03 | vittogio | before having asterisk connected, i had 2 pbx connected, i have repliacted the same timing setting but no results |
16:56.07 | *** join/#asterisk Kokey (n=ubunture@dsl-200-78-65-27.prod-infinitum.com.mx) |
16:57.28 | gaupe | how is the current (not beta) firmware on the GXP-2000 now? |
16:57.54 | vittogio | i think that i am bothering boring you with this case |
16:58.01 | gaupe | I've got a installation with 16 phones, just considering if I should replace them or not |
16:58.05 | asterboy | gaupe, awesome |
16:58.18 | Qwell | If you have to ask that question, it's time to switch phones |
16:58.30 | gaupe | asterboy: compared to previous versions, or compared to other phones? |
16:58.40 | asterboy | both |
16:59.11 | gaupe | Qwell: I haven't taken over the installation yet, but I'm going to - and since it's not more than 16 phones - they can be replaced quite easily |
16:59.13 | vittogio | coppice,Laureano, thanks for your suggestions. if i find out a solution i will inform |
16:59.15 | asterboy | the new menu and entire retrofit gives the phone what it should have had from the start |
16:59.45 | gaupe | asterboy: I looked at the software when in beta-stages, and it looked a lot better :) |
17:00.07 | asterboy | its a MUST upgrade |
17:00.12 | gaupe | :) |
17:01.06 | gaupe | I've got 10 Thomson ST2030 in at my office, considering using them - since I've already got a autoprovision setup ready for them |
17:01.12 | a1fa | hello hello |
17:01.13 | a1fa | hello |
17:01.32 | a1fa | hey, i need to make a extension for i event, if it goes to i 3 times, to dial a specific extension |
17:01.39 | a1fa | can i just use vars and gotoif |
17:01.41 | a1fa | ? |
17:01.45 | vittogio | bye |
17:02.36 | a1fa | how can i add +1 to a variable? |
17:03.47 | codebreaker | a1fa: of course you can do this. if somebody tries 3 times to select the rigth ivr and fails let call the operator |
17:04.21 | a1fa | codebreaker : yeah.. do you have the code allready so i dont have to reinvent the wheel? |
17:05.12 | codebreaker | one moment i have read tonight something about this in the asterisk book. and i also know there was a extension on my system. i will poste du pastebin few minutes |
17:05.28 | a1fa | sweet |
17:05.29 | a1fa | thanks man |
17:05.33 | Qwell | Set(VAR = $[${VAR} + 1]) |
17:05.40 | a1fa | i was goint to use set(var... |
17:05.44 | a1fa | cool |
17:06.16 | a1fa | exten => s,1,Set(SOMEVAR=${MATH(${SOMEVAR}+1)}) ; increment |
17:06.24 | a1fa | so you dont have to use math? |
17:06.31 | Qwell | no... |
17:06.43 | codebreaker | http://pastebin.com/688858 |
17:06.47 | a1fa | codebreaker : thanks |
17:06.49 | Zeeek | doesn't the [] do the evaluation |
17:06.52 | Qwell | Zeeek: yes |
17:07.09 | Zeeek | hence the non need for math |
17:07.41 | a1fa | sweet |
17:07.54 | a1fa | codebreaker : thanks man.. i am going to simplify this and repaste it |
17:08.27 | codebreaker | a1fa: http://www.asteriskdocs.org/modules/news/ read there the asterisktfot book. really good |
17:09.19 | Flosoft | has anyone got experience with voiceone? |
17:10.59 | a1fa | done |
17:11.27 | a1fa | here is my version |
17:11.35 | a1fa | "basterdized" version of timeout |
17:11.40 | a1fa | and invalid re-entry |
17:11.50 | a1fa | http://pastebin.com/688866 |
17:12.47 | a1fa | works great |
17:14.21 | a1fa | any objections? |
17:15.11 | codebreaker | yup. often i only need a starting point |
17:20.39 | *** join/#asterisk vittogio (n=vittogio@host173-40.pool8259.interbusiness.it) |
17:22.05 | *** part/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.21.26.Dial1.SanJose1.Level3.net) |
17:22.20 | Flosoft | can anyone help me installing an Webinterface? |
17:22.48 | a1fa | no |
17:22.55 | a1fa | Flosoft : apm? you mean? |
17:22.57 | a1fa | apm sucks |
17:23.13 | Flosoft | well I am looking for one that I can run on my Plesk Server |
17:23.16 | *** join/#asterisk xtr (n=94752345@S0106000c41ed11e1.vf.shawcable.net) |
17:23.26 | Flosoft | I thought of acami or voiceone |
17:23.47 | *** join/#asterisk inv_arp[work] (i=junya@adsl-10-153-159.mia.bellsouth.net) |
17:23.59 | Flosoft | freepbx is nice .. but I can't use it afaik |
17:24.01 | *** part/#asterisk Laureano (n=Laureano@host172046.metrored.net.ar) |
17:24.06 | *** join/#asterisk Laureano (n=Laureano@host172046.metrored.net.ar) |
17:25.49 | asterboy | For a great GUI * .... |
17:25.49 | asterboy | ...use... |
17:25.49 | Flosoft | yes ......... |
17:25.49 | asterboy | vim and X Windows |
17:25.49 | salviadud | yeah |
17:25.49 | Flosoft | already know that joke |
17:25.49 | salviadud | the cli rocks! |
17:25.50 | asterboy | yep |
17:25.50 | salviadud | air guitar |
17:25.50 | Flosoft | I really need one |
17:25.57 | a1fa | :P |
17:26.19 | Flosoft | please ... I really need some help |
17:26.50 | *** join/#asterisk ManxPower (n=ewieling@dpc67142183150.direcpc.com) |
17:27.40 | a1fa | aww. poor baby |
17:27.45 | a1fa | why do you want to use a gui? |
17:27.50 | a1fa | isnt there a readme or install? |
17:27.51 | Flosoft | I want an interface to manage asterisk on a shared webhost |
17:28.02 | a1fa | why not use vi or pico |
17:28.02 | Flosoft | the problem is plesk |
17:28.10 | a1fa | i dunno what plesk is man |
17:28.10 | Flosoft | I can't fuck up the Webserver configs |
17:28.15 | ManxPower | Flosoft, Let us know when you write the interface. |
17:28.20 | Flosoft | Plesk = cPanel |
17:28.28 | a1fa | dude |
17:28.31 | a1fa | just ssh to the box |
17:28.37 | Flosoft | well I can |
17:28.38 | a1fa | i once wanted to use APM |
17:28.41 | a1fa | biggest mistake ever |
17:28.41 | Flosoft | it is a dedicated server |
17:28.48 | a1fa | it takes control over your configs |
17:28.49 | ManxPower | a1fa, It's some silly WEB/GUI thing for people that can't or don't know how to admin a box. |
17:28.51 | a1fa | and it needs mysql |
17:29.07 | a1fa | mysql takes too much memory and cpu |
17:29.09 | a1fa | ;) |
17:29.10 | coppice | dedicated? I like hard working servers :-) |
17:29.14 | *** join/#asterisk gaiadg (n=gaiadg@cpe-74-64-42-87.nyc.res.rr.com) |
17:29.19 | Flosoft | hehe |
17:29.35 | Flosoft | well ... Freepbx looks nice ... but I don't know how to install it. |
17:29.57 | Flosoft | I would need some help setting it up |
17:30.17 | a1fa | hehe |
17:30.21 | a1fa | this is * not freepbx |
17:30.21 | *** join/#asterisk MikeJ__ (n=vircuser@d14-69-8-30.try.wideopenwest.com) |
17:30.27 | Flosoft | well |
17:30.32 | Flosoft | I found Voiceone |
17:30.34 | a1fa | why dont you get the source |
17:30.35 | salviadud | freepbx is a nightmare |
17:30.36 | Flosoft | or Acami |
17:30.49 | salviadud | you see, with * |
17:30.53 | gaiadg | hello, quick question. I have a context in sip.conf for three extenions. What is the best way to setup outbound calleid to be the cid of the sip account? |
17:30.54 | salviadud | YOU can be the artist |
17:31.06 | salviadud | i see a extensions.conf file made in freepbx |
17:31.18 | salviadud | it's a jungle of macros and nonsense |
17:31.35 | *** join/#asterisk hohum (i=corbe@snoop.burghcom.com) |
17:31.51 | salviadud | i prefer raw asterisk, for i can bake it the way i want it to |
17:32.00 | salviadud | medium rare |
17:32.06 | salviadud | or... sizzling hot |
17:32.11 | Flosoft | well ... I don't know how to configure asterisk without Webui |
17:32.14 | salviadud | you get the idea |
17:32.22 | salviadud | flosoft, how old are you? |
17:32.25 | Flosoft | 16 |
17:32.28 | *** join/#asterisk ast_freak (n=jesse@12.28.106.2) |
17:32.46 | salviadud | dude... |
17:32.55 | salviadud | i remember when i was 16 |
17:33.09 | salviadud | back then... the internet pr0n was traded, not leeched |
17:33.19 | Flosoft | lol |
17:33.53 | salviadud | my recommendation is that you read the asterisk handbook |
17:34.13 | salviadud | another question |
17:34.17 | salviadud | what distro are you using? |
17:34.22 | Flosoft | Debian 3.1 |
17:34.49 | Flosoft | please ... I really just want to setup an Webinterface that doesn't conflict with Plesk |
17:35.06 | salviadud | i honestly don't know man |
17:35.11 | *** join/#asterisk dahunter3 (n=dahunter@pool-71-110-89-49.lsanca.dsl-w.verizon.net) |
17:35.26 | salviadud | i doubt you'll find someone that would want to help you out with that, it is very off topic |
17:35.35 | *** join/#asterisk n4y (n=tmalkut@host-ip2-24.crowley.pl) |
17:35.54 | Flosoft | where could I find some then? |
17:35.58 | Flosoft | *someone |
17:36.17 | *** join/#asterisk darby_t (i=darby_t@aapa56.neoplus.adsl.tpnet.pl) |
17:36.47 | *** join/#asterisk alephcom (n=alephcom@host75.net14.mcsnet.ca) |
17:37.14 | ManxPower | Flosoft, Try the asterisk-biz mailing list. |
17:37.30 | ManxPower | most of here have no interest whatsoever in doing custom CGI programming for interfacing to Asterisk. |
17:37.45 | Flosoft | well not custom |
17:38.06 | Flosoft | I just need help installing a Webinterface |
17:38.13 | ManxPower | It's hard to do, complex, and not very rewarding. Also since the customer seldom actually knows what they want, poor project management frequently makes such projects a total disaster. |
17:38.23 | ManxPower | Flosoft, which web interface? |
17:38.47 | ManxPower | Asterisk does not come with a web interface. |
17:39.39 | Flosoft | Voiceone, Acami ... maybe even freepbx if it can work without changing the config of the webservers |
17:40.07 | ManxPower | I've never heard of the first two, the last one is supported in #FreePBX. NONE of these come with Asterisk. |
17:40.24 | Flosoft | well there is none that comes with asterisk |
17:40.25 | *** part/#asterisk gmaruzz (n=gmaruzz@217-133-80-112.b2b.tiscali.it) |
17:40.47 | ManxPower | Try #FreePBX. |
17:41.12 | Flosoft | just did |
17:42.39 | salviadud | the guys at #freepbx are there, sometimes |
17:42.44 | salviadud | i can understand your grief |
17:42.55 | salviadud | still. we encourage you to read and stay in school |
17:43.07 | salviadud | that way, you'll really learn |
17:44.48 | Flosoft | hehe |
17:45.57 | coppice | still. we encourage you to read and stay in school, as it keeps down the length of the dole queue :-) |
17:49.22 | codebreaker | somebody know how to connect n boxes together and route calls to any number between them. no matter on wich box the client has registered. without use of dundi. also like a active-active setup. i know there was an easy way but i forgot |
17:50.05 | ManxPower | Flosoft, Where are you located? |
17:50.23 | ManxPower | codebreaker, use Dundi or ENUM |
17:51.51 | codebreaker | ManxPower: how high is the load for every request on the dundi/dnsserver? all this servers are connected to the same network i only ave them for loadbalancing/failover etc |
17:53.20 | ManxPower | codebreaker, I have no idea, but no matter what you're going to have to have some method of querying extensions. Read up on the dundi website. Dundi is designed for this sort of stuff, that is it's advantage. ENUM is based on DNS so people know what issues to expect in DNS, that is it's advantage. |
17:53.45 | ManxPower | codebreaker, there is also a Wiki page on fail over, loadbalancing, etc. |
17:55.02 | codebreaker | ManxPower: i know. but the all use some ser in front of it. but nowhere(or i dont found it) it is described how to have more servers with the same iax.con extensions.conf etc.. |
17:55.38 | ManxPower | none of the clients I have are willing to spend the money on failover, and none of them are large enough for load balancing. I've used ENUM to reduce the amount of work to manage the servers. |
17:56.05 | ManxPower | codebreaker, look into Realtime support in Asterisk. |
17:56.13 | *** part/#asterisk alephcom (n=alephcom@host75.net14.mcsnet.ca) |
17:57.15 | codebreaker | ManxPower: there is one way i think to go. when a clint registeres then this asteriskserver is updating the bind and all other will then use this setting until a new update is triggered |
17:58.44 | codebreaker | ManxPower: for me it is not a question of money. for me it is the question how to do it myself with opensource not for money. just to learn something(or lern more) :) |
17:59.07 | ManxPower | codebreaker, My point is I cannot help yu further. |
17:59.33 | codebreaker | i understand. this was only to clarify my point :) |
18:05.13 | *** join/#asterisk tiCo89 (i=mario@debian.uid0.ch) |
18:05.18 | tiCo89 | hello |
18:06.30 | tiCo89 | i would like to setup asterisk on my gateway, it should make a connection to a sip-provider. if there are clients in my lan (e.g kphone) it should ring on all these phones, if not it should go on a mailbox, has somebody a good howto? |
18:10.07 | *** join/#asterisk juanjoc (n=juanjoc@248-32-235-201.fibertel.com.ar) |
18:12.20 | asterboy | start here: |
18:12.22 | asterboy | ~docs |
18:12.23 | jbot | docs is, like, probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
18:12.33 | asterboy | lots of howto |
18:13.18 | codebreaker | tiCo89: http://pastebin.com/688975 |
18:13.29 | ManxPower | ~thebook |
18:13.30 | jbot | somebody said thebook was Asterisk: The Future of Telephony, released under a Creative Commons license and available at http://www.asteriskdocs.org << Read the book online! |
18:13.31 | ManxPower | ~mailinglist |
18:13.32 | jbot | Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search. Browse the mailing list archive at http://lists.digium.com/ or search through it at http://www.asteriskguru.com/archives, or and there is also the Macintosh Asterisk mailing list at http://www.astmasters.net/maml.htmm |
18:15.27 | *** join/#asterisk timscott (n=a@d198-53-19-216.abhsia.telus.net) |
18:15.34 | *** part/#asterisk darby_t (i=darby_t@aapa56.neoplus.adsl.tpnet.pl) |
18:16.22 | *** join/#asterisk Vyeperman (n=Vye@ip68-6-130-59.sd.sd.cox.net) |
18:23.48 | Flosoft | how easily is asterisk configurable? |
18:24.50 | russellb | it's so easy that it configures itself! |
18:24.55 | russellb | ta-da! |
18:25.09 | *** join/#asterisk x86 (n=x86@p3m/member/x86) |
18:26.08 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219) |
18:26.24 | Flosoft | well .. .what about IVR and and ringroups etc? |
18:26.48 | russellb | that's all very common configuration, there are many examples available |
18:26.50 | russellb | ~docs |
18:26.51 | jbot | [docs] probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
18:30.28 | file | yay nice quiet #asterisk weekend |
18:30.53 | *** join/#asterisk tasat (n=tasat@c-67-180-181-221.hsd1.ca.comcast.net) |
18:31.27 | *** join/#asterisk markit (n=konversa@host119-245.pool8172.interbusiness.it) |
18:31.45 | *** join/#asterisk cryptnix (n=andrew@64.25.198.126) |
18:49.26 | Hmmhesays | heh |
18:52.18 | *** join/#asterisk BadPacket (n=root@unaffiliated/badpacket) |
18:55.27 | *** part/#asterisk Flosoft (n=admin@213.219.160.100) |
18:58.50 | *** join/#asterisk Flosoft (n=admin@213.219.160.100) |
18:59.09 | Flosoft | can anyone help me with a VoiceOne installation? |
19:00.01 | salviadud | is there a black man in the chan? |
19:00.20 | salviadud | i am looking for an african-american, maybe even an african-south-american |
19:02.04 | *** join/#asterisk stoffell_h (n=stoffell@81.83.249.224) |
19:02.35 | sevard | whatchu need bro |
19:02.36 | Flosoft | can anyone help me with a VoiceOne installation? |
19:03.40 | salviadud | i need some pointers |
19:03.43 | Qwell | Flosoft: keep asking every 2 minutes |
19:04.04 | salviadud | on how to pretend i'm black |
19:04.06 | salviadud | at least on the phone |
19:04.18 | Qwell | salviadud: Just talk normally. |
19:04.21 | sevard | say stuff like shit dawg, and That's Wack |
19:04.25 | Flosoft | well ... I am stuck with this for a week now |
19:04.37 | salviadud | i'm gonnna have fun with this |
19:04.40 | Qwell | florz: So ask voiceone |
19:05.31 | salviadud | sevard, are you authentically black, or are you just giving me some generic advice man? |
19:05.42 | sevard | salviadud: everyone on the internet is black. |
19:06.05 | salviadud | o_O |
19:06.25 | salviadud | that's wack dawg |
19:06.32 | timscott | May I ask why you would want to "sound black" on the phone? |
19:06.44 | sevard | there you go. |
19:06.51 | stoffell_h | lol |
19:06.58 | timscott | whiird dawg why u needin' that shit on the tele', yo? |
19:07.29 | sevard | timscott: black people don't belong on the telephone? |
19:07.32 | salviadud | i want some hoes :) |
19:07.37 | timscott | oh man |
19:08.03 | timscott | get off the internet, and go make friends. money plays you'll find more "hoes" that way. |
19:08.10 | salviadud | im gonna call all those laquishas |
19:08.24 | sevard | salviadud: what about the La Fan Das |
19:08.34 | sevard | erm, Duh |
19:08.40 | timscott | la fon da |
19:08.41 | salviadud | they got all kinds of cookie names |
19:08.46 | Qwell | enough, now |
19:08.58 | Qwell | this conversation ends immediately |
19:09.18 | salviadud | qwell, what's wrong with the hoes? |
19:09.32 | salviadud | you like hoes too right? |
19:09.52 | sevard | alright, i'm irish and i love irish jokes, i also love potatoes (does the bride come with potatoes?!) why is it when ever somebody makes fun of me it's cool but when i start picking on black people it's like OH SHIT MAN THE NAZIS ARE HERE. |
19:10.13 | timscott | Because you're white. |
19:10.24 | timscott | You're the prverbial "man." |
19:10.27 | sevard | that's wack dawg |
19:10.45 | timscott | I like asterisk. |
19:10.51 | timscott | What about them thurr asterisks, huh? |
19:10.55 | salviadud | i'm mexican |
19:11.25 | salviadud | yeah, those asteriks got a big booty |
19:19.29 | *** join/#asterisk cmx_DK (n=dp@cpe.atm2-0-1051183.0x503f8576.kd4nxx11.customer.tele.dk) |
19:19.31 | *** join/#asterisk RoyK (n=roy@ti211210a080-0268.bb.online.no) |
19:20.06 | cmx_DK | good afternoon guys :) (in denmark its 9:21pm) :) |
19:20.36 | timscott | Good afternoon. :) |
19:20.40 | timscott | It's 1:23 here. :D |
19:21.57 | cmx_DK | hehe :) - thats the beauty of the internet :)... |
19:22.19 | cmx_DK | i have a problem with double dtmf tones with sip trunks. |
19:22.36 | timscott | What do you mean? |
19:22.41 | timscott | DTMF is getting sent twice? |
19:23.43 | cmx_DK | yup - its not just a standalone problem - we have 100+ asterisk serveres all with the same problem, and not the same hardware, and different asterisk versions |
19:24.08 | timscott | you have 100 asterisk servers? |
19:24.12 | timscott | oof |
19:24.19 | cmx_DK | i was just wandering if any one else has that problem. |
19:24.20 | timscott | what distribution? |
19:24.31 | timscott | no, i've never had the problem |
19:24.37 | timscott | are you using a virtual asterisk setup? |
19:24.43 | timscott | like with Xen or VServers? |
19:24.45 | coppice | poisson or gaussian |
19:24.57 | cmx_DK | we uses CentOS - not the aah - but our homemade iso. |
19:25.07 | cmx_DK | nope - not virtual. |
19:25.15 | timscott | :/ |
19:25.26 | Flosoft | can anyone help me with a VoiceOne installation? |
19:25.35 | file | Flosoft: over and over and over you ask... |
19:25.49 | cmx_DK | the problem is with DISA - atleast the problem is very consistent with DISA. |
19:27.06 | jake1932 | is VoiceOne an ITSP? |
19:28.06 | Flosoft | no |
19:28.16 | Flosoft | VoiceOne is an GUI for Asterisk |
19:28.16 | jake1932 | what is it? |
19:28.19 | jake1932 | oh |
19:28.26 | *** join/#asterisk MikeJ[Laptop] (n=vircuser@adsl-69-209-162-62.dsl.sfldmi.ameritech.net) |
19:28.34 | jake1932 | i don't think anyone on here supports GUIs for asterisk |
19:28.49 | cmx_DK | anyone else that can help regarding the DTMF problem? |
19:29.19 | jake1932 | does VoiceOne have a mailing list or forum? |
19:30.26 | Flosoft | they are both quite bad |
19:30.29 | Flosoft | the only thing I have |
19:30.30 | Flosoft | http://www.voiceone.it/download/ |
19:31.18 | jake1932 | the forum seems to have some activity |
19:31.29 | *** join/#asterisk Whisk (n=whisk@82-40-184-22.cable.ubr04.croy.blueyonder.co.uk) |
19:32.03 | Flosoft | the thing is I don't know too much bout asterisk or linux |
19:32.08 | jake1932 | haha |
19:32.12 | Flosoft | so I am a bit lost in the installation procedure |
19:32.34 | jake1932 | why not start with asterisk - you may find you don't need this GUI |
19:32.34 | Flosoft | especially... Debian isn't a sudo system |
19:33.00 | Flosoft | so I would need to change the beginning of the tutorial they have there |
19:33.06 | *** join/#asterisk kaz0358 (n=kurtzogl@ES-189.telecom.ksu.edu) |
19:33.10 | jake1932 | asterisk itself is very well supported here (including the installation with Debian) |
19:33.11 | Flosoft | and I don't know what debian packages I would ned |
19:33.31 | jake1932 | there's even a page on the wiki on how to install asterisk + debian |
19:33.32 | kaz0358 | does anyone know if iaxtel is down? |
19:34.12 | VoicePulse | Flosoft: How many users is your system intended for? |
19:34.40 | Flosoft | 20 - 40 |
19:35.01 | Flosoft | don't know yet really :S |
19:35.36 | VoicePulse | Then I suggest you go with a proven plug-and-play product like Fonality or SwitchVox, it will take at least 6 months to learn enough to get things up and running reliably. |
19:35.59 | jake1932 | VP - i take issue with that |
19:36.19 | VoicePulse | or pay jake1932 to set it up for you :) |
19:36.23 | jake1932 | at least a few years! |
19:36.25 | jake1932 | haha |
19:36.55 | jake1932 | actually - rapid is pretty fast - your results may differ |
19:37.05 | jake1932 | it's prebuilt debian + asterisk |
19:37.24 | kaz0358 | could someone that has iax setup dial the echo test number as see if it works? 1-700-999-9613 |
19:37.31 | kaz0358 | err iaxtel |
19:37.48 | russellb | kaz0358: it probably doesn't work |
19:37.56 | jake1932 | http://xorcom.com/rapid/ |
19:37.57 | file | iaxtel goes up and down, a lot |
19:38.27 | *** join/#asterisk paryl (n=chatzill@216-201-177-82.res.logixcom.net) |
19:39.06 | kaz0358 | russellb and file, ahh okay.. well that might explain why i'm not able to do a iax2 register or dial any 700 numbers even when going out as user guest |
19:39.34 | jake1932 | Flosoft - although rapid runs on an older asterisk version, it's probably good enough for you to get your hands dirty a bit |
19:39.39 | file | we're mucking with it right now |
19:40.02 | Flosoft | the point is I need to integrate it into an running system |
19:40.03 | paryl | does anyone here use a tdm2400p? i assumed my zaptel.conf and zapata.conf should be similar to my machine with a tdm400p, but it doesn't seem to be liking my config :\ |
19:40.08 | Flosoft | I can't change Distro |
19:40.21 | russellb | kaz0358: try again now |
19:40.34 | jake1932 | <PROTECTED> |
19:40.37 | Flosoft | yes |
19:40.38 | kaz0358 | russellb, okay.. trying to re-register |
19:41.30 | russellb | this machine is so weak |
19:41.39 | jake1932 | Flosoft: ok, try http://www.voip-info.org/tiki-index.php?page=Asterisk+Linux+Debian |
19:41.45 | paryl | i get "ERROR[3776]: chan_zap.c:6878 mkintf: Unable to open channel 1: No such device" |
19:42.08 | paryl | but ztcfg -vvvv shows 4 channels configured properly |
19:42.10 | kaz0358 | russellb, okay.. cool. registeration was successful. and i was able to dial the echo test number. thanks! hehe. i was about ready to pull my hair out. |
19:42.30 | file | don't gauge a working Asterisk on calling iaxtel |
19:43.08 | *** join/#asterisk RoyK (n=roy@ti211210a080-0268.bb.online.no) |
19:43.18 | kaz0358 | russellb, there is only *one* machine running the iaxtel stuff? what order of users are normally registered over iax2? |
19:43.35 | file | 788 right now |
19:43.44 | file | it's 2.4GHz P4 |
19:43.51 | russellb | 512 MB ram |
19:43.53 | file | with 512MB of RAM |
19:44.09 | russellb | probably an emachines or something, lol |
19:44.12 | bsdfreak | heya file |
19:44.13 | kaz0358 | file, thanks.. maybe there should be a short addendum to voip-info.org wiki on that :) |
19:44.30 | file | hellllllllo |
19:47.55 | russellb | it's still fun to poke at it |
19:48.02 | russellb | just don't expect it to work :) |
19:48.10 | *** join/#asterisk r0d3nt (n=RatMan@foster.stonedcoder.org) |
19:48.15 | russellb | it's actually a very useful load test for us |
19:48.20 | paryl | can anyone help me configure my tdm2400p? |
19:48.28 | paryl | it's just not working |
19:48.35 | file | it's better then it was, but there's still one last problem to track down |
19:48.36 | Ariel_ | russellb, it was great 2 or 3 years ago. for testing asterisk iax2 connections. |
19:48.50 | russellb | yeah, we need a new server for it |
19:49.19 | file | russellb: we should put in a request for a new one |
19:49.25 | russellb | file: yes, we should |
19:49.33 | russellb | file: why don't you get on that |
19:49.35 | *** mode/#asterisk [+o file] by russellb |
19:49.45 | file | you have seniority! |
19:49.55 | russellb | but you're the iaxtel maintainer :-p |
19:50.12 | Qwell | ooo, bribe |
19:50.15 | file | minor detail |
19:50.18 | Qwell | now, time for blackmail :p |
19:50.19 | russellb | zing! |
19:50.20 | jake1932 | who has the janitor job? |
19:50.30 | russellb | jake1932: file |
19:50.32 | jake1932 | haha |
19:50.34 | file | pfft |
19:50.38 | russellb | he has to clean our office every night |
19:50.44 | Qwell | remotely! |
19:50.56 | russellb | it's a virtual office ... |
19:51.05 | Qwell | yeah |
19:51.16 | Qwell | You don't telecommute to your virtual office? |
19:51.19 | russellb | there's a chair on the ceiling, feel free to have a seat. |
19:51.29 | *** join/#asterisk gr0mit (n=guest@extrt.txrx.org.uk) |
19:51.49 | *** join/#asterisk Laureano (n=mdelia@host172046.metrored.net.ar) |
19:52.03 | file | we are not responsible if you fall and hurt yourself |
19:52.04 | Laureano | Hello |
19:52.22 | Ariel_ | but it would be fun to watch |
19:53.17 | kaz0358 | any idea on the number of extensions accessible on the public dundi context? i'm not for sure if that is the correct wording for it |
19:53.54 | russellb | kaz0358: a lot ... but as a percentage of the e164 number space, probably very little |
19:55.06 | kaz0358 | russellb, i thought that might be the case. |
19:55.39 | russellb | i'm not really sure, though |
19:55.46 | russellb | i'm not a member anymore |
19:56.21 | kaz0358 | if you want to participate as a tier 3, does it take much active bandwidth? |
19:56.37 | kaz0358 | thanks for the info russellb |
19:58.27 | paryl | hate to ask again... but is there anyone that can help me with my tdm2400p? |
19:58.39 | paryl | no one is answering any of my questions :\ |
19:58.46 | kaz0358 | paryl, what kind of problem are you having? |
19:58.51 | gr0mit | paryl what is the problem? |
19:59.25 | paryl | well, i've configured zaptel.conf pretty much identical to my machine with a tdm400p in it... fxsks=1-4 |
19:59.37 | paryl | and ztcfg -vvvv shows 4 channels configured |
19:59.45 | gr0mit | mkay.... |
19:59.52 | paryl | but asterisk gives me "ERROR[3843]: chan_zap.c:6878 mkintf: Unable to open channel 1: No such device" |
19:59.55 | paryl | and dies |
20:00.09 | paryl | zapata is configured the same way that it is on the other machine too... |
20:00.39 | paryl | none of the docs on digium's site have anything specific to the 2400p, so i wasn't sure if i'm configuring it wrong or not |
20:01.06 | *** join/#asterisk surfdue (n=tyler@unaffiliated/surfdue) |
20:01.08 | surfdue | hello! |
20:01.17 | surfdue | i want a pbx i think this is the best right? |
20:01.26 | surfdue | :) |
20:01.35 | luke-jr_ | ... |
20:01.38 | kaz0358 | surfdue, the best is a high subjective thing :) |
20:01.57 | surfdue | well I am looking for a pbx with a web config |
20:02.02 | surfdue | im thinking asterisk with amp? |
20:02.03 | luke-jr_ | kaz0358: not at all; just depends on what you need the best *for* |
20:02.08 | surfdue | or do you have a diffrent suggestion for me? |
20:02.15 | luke-jr_ | surfdue: see the topic |
20:02.19 | surfdue | I see it |
20:02.27 | luke-jr_ | surfdue: specifically regarding AMP |
20:02.29 | paryl | also, "WARNING[3885]: chan_zap.c:923 zt_open: Unable to specify channel 1: No such device" and " ERROR[3885]: chan_zap.c:10314 setup_zap: Unable to register channel '1-4'" |
20:02.33 | surfdue | are you telling me to not use asterisk? |
20:02.42 | surfdue | i dont use amp. |
20:02.43 | luke-jr_ | surfdue: telling you not to use AMP, sure |
20:02.45 | surfdue | and i dont know what that is? |
20:02.50 | luke-jr_ | ... |
20:02.56 | surfdue | is there a web config utility? |
20:02.59 | luke-jr_ | you just fscking said "im thinking asterisk with amp" |
20:03.03 | surfdue | in astirisk |
20:03.08 | luke-jr_ | no |
20:03.15 | surfdue | i was reading the topic when i read that sorry |
20:03.16 | kaz0358 | paryl, do a zap show status |
20:03.22 | surfdue | i meant i need asterisk with web config |
20:03.28 | Qwell | "need"? |
20:03.43 | paryl | kaz: i can't, asterisk dies on that error |
20:03.47 | surfdue | :| |
20:03.48 | gr0mit | kaz0358 i dont think paryl can get it running |
20:03.49 | luke-jr_ | well, I don't think anyone here wants to play with people using web configs |
20:03.52 | kaz0358 | surfdue, no i'm not telling you not to use it. i've never used any external web interface to configure or setup asterisk.. so i can't really comment on that aspect of it |
20:04.10 | surfdue | kaz0358, how should i configure it |
20:04.20 | gr0mit | surfdue : you are asking for trouble trying to use a web interface |
20:04.21 | *** join/#asterisk ToTo (n=ToTo@host38-109.pool8258.interbusiness.it) |
20:04.23 | luke-jr_ | surfdue: configuration files |
20:04.31 | surfdue | is it easy to write them? |
20:04.46 | kaz0358 | paryl, ahh.. well i haven't played with either the 4 or 24 cards, but i have messed around with the 2 port t1/e1 card and i ran into the same problem for a while. it turned out it was a problem with the configuration files |
20:04.49 | *** join/#asterisk Assid (n=assid@203.115.64.12) |
20:04.49 | jake1932 | not at first |
20:05.00 | jake1932 | but you'll get it - they're plenty of samples |
20:05.01 | luke-jr_ | surfdue: sure |
20:05.06 | gr0mit | surfdue you can get a call in and out with about 5 lines in extensions.conf |
20:05.11 | Assid | okay something wrong with my phone.. it keeps losing ticks for the time |
20:05.16 | paryl | kaz: what was the problem with the config files, exactly? |
20:05.19 | paryl | do you remmeber? |
20:05.21 | luke-jr_ | gr0mit: extensions.conf is crufty ;) |
20:05.36 | gr0mit | start with something simple! |
20:05.40 | gr0mit | crufty? |
20:05.49 | jake1932 | oohg - new word! |
20:05.59 | gr0mit | AEL ???? |
20:06.04 | jake1932 | crufty |
20:06.14 | Assid | it just slows down |
20:06.23 | Assid | anyone know how ot fix it? |
20:06.26 | kaz0358 | paryl, well.. there were several problems.. but one in particular was that i was specifying a logical port on the spanmap in the zapata.conf file. i never did figure out what the deal was with that, but simply removing the 3rd option made things work. |
20:07.04 | luke-jr_ | gr0mit: 2000 => { Dial(SIP/2000); switch(${DIALSTATUS}) { case BUSY: Busy(); break; default: Playback(wtf); Hangup(); }; }; |
20:08.05 | gr0mit | what is this, luke-jr_ ??? |
20:08.15 | kaz0358 | paryl, but that didn't stop asterisk from firing up. not specifying how the channels should be setup did cause problems.. or trying to specify something conflicting for the channels in both zaptel.conf and zapta.conf |
20:08.23 | luke-jr_ | gr0mit: Asterisk Extension Language |
20:08.31 | luke-jr_ | gr0mit: extensions.ael |
20:08.35 | gr0mit | hmmmm. is this something recent? |
20:08.39 | luke-jr_ | 1.2 |
20:08.43 | gr0mit | mkay. |
20:08.56 | luke-jr_ | tho buggy until next major release |
20:09.05 | luke-jr_ | cuz russellb doesn't want to fix bugs in 1.2's implementation |
20:09.23 | surfdue | is there a fedora core 4 install guide for asterisk? |
20:09.26 | gr0mit | very low WAF |
20:09.36 | luke-jr_ | well, just make sure your switch()s have a default: and fallthrus have at least one statement before the fall |
20:09.48 | luke-jr_ | the other bugs are mostly gone in latest 1.2 |
20:09.50 | snitt | \o/ |
20:09.56 | luke-jr_ | oh, and don't use 'continue;' |
20:10.05 | luke-jr_ | at least not in a for loop |
20:10.25 | paryl | ok... if you don't have anything in zapata.conf, should asterisk still recognize the zap card? |
20:10.40 | gr0mit | paryl, no! |
20:11.03 | paryl | but it does! i mean, it doesn't configure the channels, but 'zap show status' shows me the card |
20:11.08 | gr0mit | you need to define the channels in zapata.conf |
20:11.23 | gr0mit | yes but you will not be able to make any calls |
20:11.30 | luke-jr_ | gr0mit: http://www.voip-info.org/wiki-Asterisk+AEL |
20:11.46 | gr0mit | tvm, luke_jr_ |
20:12.20 | luke-jr_ | tvm? |
20:12.27 | paryl | ok, with just: |
20:12.28 | paryl | [channels] |
20:12.30 | paryl | signalling = fxs_ks |
20:12.31 | paryl | channel => 1-4 |
20:12.33 | paryl | i get the error |
20:12.43 | paryl | that's all that's in zapata.conf |
20:13.17 | gr0mit | luke-jr_ tvm = Thanks Very Much or Ta very much if, like me, you are in the UK |
20:13.44 | luke-jr_ | ah ok |
20:13.45 | luke-jr_ | np |
20:14.16 | gr0mit | paryl, you need a bit more than this in zapata.conf, i think |
20:14.27 | luke-jr_ | append a '2' to the wiki link to get info on the new version in HEAD |
20:14.41 | paryl | gr0mit: well, yeah, but should that be enough to kill asterisk? |
20:15.01 | gr0mit | but i speak from a position of ignorance as I have only used the ultra-crappy X100p and lots of BRI and E1 cards |
20:15.31 | paryl | is there anything else i can use to verify the card is working properly? |
20:15.36 | gr0mit | dumb quesstion paryl but you did compile and install zaptel....? |
20:15.50 | paryl | yes, just compiled it fresh |
20:15.53 | *** part/#asterisk surfdue (n=tyler@unaffiliated/surfdue) |
20:16.28 | gr0mit | please paste your zaptel.conf and zapata.conf into www.pastebin.ca |
20:17.29 | paryl | it's not big, three lines: fxsks=1-4, defaultzone=us, loadzone=us |
20:17.38 | gr0mit | also paryl please paste the output of ztcfg -vv |
20:17.44 | *** join/#asterisk MikeJ[Laptop] (n=vircuser@adsl-69-209-183-91.dsl.sfldmi.ameritech.net) |
20:18.30 | sevard | he got sick of your shit. |
20:18.42 | *** join/#asterisk paryl (n=chatzill@216-201-177-82.res.logixcom.net) |
20:18.50 | paryl | grr... got kicked out |
20:20.03 | gr0mit | ok paryl paste paste paste |
20:21.25 | paryl | gr0mit: http://pastebin.ca/52260 |
20:21.42 | gr0mit | got it |
20:22.47 | gr0mit | you want to remove switchtype - national |
20:22.58 | gr0mit | this only referrs to PRI/ISDN lines |
20:22.59 | Flosoft | I can't start asterisk although I did everything Voiceone told me to do |
20:23.14 | Flosoft | debian:/var/www/vhosts/debian.FLOSOFT-NET/httpdocs# /etc/init.d/asterisk start |
20:23.14 | Flosoft | Asterisk not yet configured. Edit /etc/default/asterisk first. |
20:23.22 | Flosoft | that is what it says :S |
20:23.26 | paryl | gr0mit: ok, done, but still the same error |
20:23.28 | ManxPower | Flosoft, That is not an Asterisk message. |
20:23.44 | Flosoft | http://www.voiceone.it/download/ |
20:23.45 | Flosoft | all done :S |
20:24.01 | gr0mit | hmph please paste the error you are getting to pastebin, with a bit of context above the error |
20:24.09 | ManxPower | another one for the ignore lists. |
20:25.12 | paryl | gr0mit: http://pastebin.ca/52262 |
20:25.36 | gr0mit | got it |
20:25.54 | blitzrage | nrw: I'd like to prevent the phone from performing the redirect (I know -- my reply has several hours latency :)) |
20:26.24 | gr0mit | well it looks like it cant load chan_zap |
20:27.16 | paryl | riiiight? :) |
20:27.34 | gr0mit | paryl, 2 secs.... |
20:28.40 | gr0mit | pls check that chan_zap.so is actually in /usr/lib/asterisk/modules directory |
20:29.03 | gr0mit | if not, then looks like zaptel did not compile and install correctly |
20:29.38 | codebreaker | always when a phone registers can i asterisk let run a script/externalcommand? |
20:30.02 | paryl | gr0mit: yes, it's in there |
20:30.08 | gr0mit | ho hum |
20:31.41 | paryl | it looks to me that it's something to do with either the configuration or the drivers... |
20:31.51 | paryl | i compiled from the 1.2.5 tarball on asterisk.org |
20:32.09 | gr0mit | drivers are fine. if ztcfg says it has configured the channels then it is telling the truth. |
20:32.13 | paryl | and compiled libpri also, though i don't know if it's needed with the tdm cards |
20:32.26 | paryl | so... it must be the config? |
20:34.15 | blitzrage | 1.2.5? |
20:34.19 | blitzrage | 1.2.7.1 is out :) |
20:36.17 | paryl | no, i'm talking about zaptel-1.2.5 |
20:36.28 | blitzrage | oh I see -- I gotta try and keep up |
20:36.38 | gr0mit | well i am somewhat stumped, paryl! |
20:37.24 | paryl | i just don't get it |
20:38.25 | gr0mit | oh wait! |
20:38.35 | gr0mit | doh! |
20:38.41 | paryl | ?? |
20:39.16 | gr0mit | i think you should remove all the => from zapata.conf and only use = |
20:39.41 | *** join/#asterisk surfdue (n=tyler@unaffiliated/surfdue) |
20:39.42 | surfdue | hello |
20:39.54 | surfdue | i am installing astrisk it needs to be run as the same user as apache |
20:39.59 | gr0mit | i am not sure that => is valid in zapata.conf, only extensions.conf |
20:40.06 | surfdue | but im on a shared server with cpanel so i cant make apache run as asterisk what do i do? |
20:40.12 | gr0mit | please try this |
20:40.58 | paryl | gr0mit: no dice :( |
20:41.00 | gr0mit | unless anyone else can assure me that => is valid |
20:41.03 | gr0mit | hmmm |
20:41.11 | gr0mit | well it was worth a try |
20:44.08 | gr0mit | ok, can you just start with the context signalling and channel lines in zapata.conf ? |
20:44.12 | paryl | i use => on both of my other servers with no problems |
20:44.17 | hads|home | gr0mit: In zapata.conf = is for assigning variables and => is for creating objects, i.e channels. |
20:44.19 | gr0mit | mkay |
20:44.50 | gr0mit | hads|home, yes, you are right |
20:45.11 | gr0mit | but callprogress => no is not correct |
20:45.40 | hads|home | You are correct, it should only be used for the channel entries. |
20:45.43 | gr0mit | the only place to put a => is the channels line at the bottom |
20:45.57 | gr0mit | pls try this, paryl |
20:46.05 | paryl | gr0mit: i've shortened zapata.conf to: |
20:46.06 | paryl | [channels] |
20:46.08 | paryl | signalling = fxs_ks |
20:46.09 | paryl | context = incoming |
20:46.11 | paryl | channel => 1-4 |
20:46.31 | gr0mit | and...? |
20:47.26 | *** join/#asterisk mog_home (n=achika54@68.62.237.103) |
20:47.31 | paryl | same error |
20:47.40 | hads|home | looks fine. For a single fxo port right? |
20:47.50 | paryl | i also tried channel => 1 |
20:48.13 | paryl | hads: 4 ports |
20:49.08 | hads|home | Ah, yes sorry, Just woke up :) |
20:49.24 | paryl | these will plug into phone lines from the phone company... that does mean i need to use fxs, correct? |
20:49.38 | gr0mit | yes, paryl |
20:49.57 | gr0mit | the boards are FXO but you need fxs signalling |
20:50.11 | paryl | right... just making sure :) |
20:50.33 | gr0mit | can you do a cat /proc/zaptel/* and paste for me |
20:52.06 | paryl | http://pastebin.ca/52269 |
20:53.37 | gr0mit | ok thanks |
20:53.52 | *** join/#asterisk Cybertoy (n=maxim@dsl254-123-241.nyc1.dsl.speakeasy.net) |
20:53.56 | gr0mit | looks ok to me |
20:55.03 | *** part/#asterisk Laureano (n=mdelia@host172046.metrored.net.ar) |
20:55.21 | paryl | ok, this is just crazy |
20:55.29 | paryl | how could it not work? |
20:55.33 | *** part/#asterisk surfdue (n=tyler@unaffiliated/surfdue) |
20:55.36 | gr0mit | well i hate to say this paryl but without sshing into your box to look, I really am rather puzzled |
20:55.57 | gr0mit | and as it is bedtime here i am not about to offer to do that! |
20:56.09 | paryl | haha, well, i do appreciate your trying |
20:56.44 | gr0mit | i managed to get ghost99 to the point where he was able to make a call yesterday, so i was happy |
20:57.07 | gr0mit | and so was he, despite ManxPower's unhelpful comments |
20:58.16 | gr0mit | hope you find someone in your timezone to get you running, paryl. |
20:58.37 | paryl | alright man, thanks again |
20:58.37 | gr0mit | Also, don't forget that Digium will get you to the point that the cards run |
20:58.50 | paryl | but they aren't open on the weekends are they? |
20:58.52 | gr0mit | this is part of what you pay them for the cards |
20:59.05 | gr0mit | no, they are not open at the weekend! |
20:59.18 | paryl | stupid non-workaholics ;) |
20:59.56 | gr0mit | traditional British habit, that |
21:00.52 | *** part/#asterisk gr0mit (n=guest@extrt.txrx.org.uk) |
21:05.00 | tparcina | deam, if digium was from dalmatia, they wouldn't work even during weekdays :) |
21:05.49 | *** join/#asterisk Assid (n=assid@203.115.64.12) |
21:07.04 | paryl | haha |
21:07.34 | Assid | i cant seem to access the call log list |
21:07.42 | Assid | on polycom 301 |
21:07.54 | Assid | nvm |
21:07.58 | Assid | gotta do it through menu |
21:09.12 | *** join/#asterisk websae (n=icechat5@h69-129-251-26.69-129.unk.tds.net) |
21:09.32 | *** join/#asterisk tekati (n=captain@cpe-66-75-215-63.bak.res.rr.com) |
21:09.49 | *** join/#asterisk robin_sz (n=robin@adsl.redpoint.org.uk) |
21:10.00 | websae | sure is quiet out there again :) |
21:10.40 | robin_sz | probably because people are busy |
21:10.55 | tekati | Shhhh... |
21:11.03 | robin_sz | busy cutting up their GXP200s, digging deep, lime filled pits, and burying them |
21:11.25 | websae | busy hehehe |
21:11.26 | websae | i hear ya on that one |
21:11.38 | websae | you don't like your Grandstream? |
21:13.04 | timscott | I don't mind my GXP2000. |
21:13.11 | websae | I like mine :) |
21:13.15 | websae | it works well |
21:13.25 | websae | except the speakerphone, but it has other good qualities :) |
21:13.25 | tekati | I don't have one. But am looking for a good IP phone is that a recommended phone? |
21:13.35 | tekati | Ah I need a good speakerphone. |
21:13.39 | tekati | Who has a good one? |
21:13.41 | websae | and once you put some clay in the headset so it hangsup everytime, it works well |
21:13.47 | websae | Polycome 501 |
21:13.52 | websae | $200ish |
21:13.53 | BadPacket | polycom |
21:13.56 | BadPacket | (obviously) |
21:14.01 | websae | or cisco :) |
21:14.17 | BadPacket | yeah, the cisco 79xx have good speakerphones too |
21:14.33 | websae | very good speakerphones |
21:14.38 | websae | everything cisco is good :) |
21:14.43 | websae | quality is there |
21:15.16 | timscott | $$$++ |
21:15.18 | timscott | for cisco |
21:15.24 | websae | yeah! |
21:15.39 | websae | but if it's quality you need.........it's quality they have :)---people pay for quality |
21:15.41 | Assid | cisco costs wayyy too much |
21:15.46 | Assid | polycoms got the best value for money |
21:15.47 | websae | just like in termination and origination services |
21:16.03 | BadPacket | websae: exactly |
21:16.20 | ManxPower | Polycoms can be gotten amazingly cheap and you don't have to by the power supply and you don't have to buy the firmware. |
21:16.22 | websae | BadPacket: i sense you have experience that feeling |
21:16.30 | BadPacket | definitely |
21:16.36 | BadPacket | I've tried grandstream and nufone |
21:16.46 | BadPacket | and now I use a Cisco 7960 and voicepulse |
21:16.56 | ManxPower | BadPacket, and you are still in therapy because of the grandstream? |
21:16.59 | websae | NuFone----one word....yuck |
21:17.07 | BadPacket | ManxPower: hah |
21:17.18 | BadPacket | more because of nufone |
21:17.24 | ManxPower | Nufone was good until their recent...issues. |
21:17.38 | tekati | But is the speakerphone part of the Polycoms good. |
21:17.40 | BadPacket | s/issues/MAJOR issues/ |
21:17.41 | Assid | currently im using poly301.. with sipdiscount for personal use.. and another line configured to my box's extension |
21:17.42 | websae | I use to always have issues with them |
21:17.45 | websae | *ALWAYS |
21:18.13 | BadPacket | the polycom speakerphone is even better than the 7960, but the polycom phone is such a pain in the ass to configure |
21:18.22 | Assid | not really |
21:18.30 | Assid | its one of the fastest ive configured |
21:18.31 | ManxPower | Huh? Basic config can be done via the web interface. |
21:18.37 | Assid | web interface is easy |
21:18.46 | Assid | only pain in polycom is.. the wait time for reboots |
21:18.52 | ManxPower | advanced config can be....challanging. |
21:19.00 | BadPacket | the built-in web server is buggy and slow... maybe I need newer firmware if you guys say it's good |
21:19.11 | ManxPower | Assid, Yeah, and in production enviroments you dno't normally reboot the phone |
21:19.15 | Assid | yeah |
21:19.23 | ManxPower | BadPacket, the built in web server IS buggy and slow. |
21:19.25 | Assid | normally.. you provision the phone via ftp/tftp |
21:19.41 | BadPacket | that's what I meant by it being a pain in the ass |
21:19.50 | Assid | i still wanna get this ringtone into my phone dammit.. just cant get it |
21:20.00 | BadPacket | Assid: yes, but I was testing stuff, so I was changing a bunch of stuff via the web interface |
21:20.12 | Assid | web interface frankly sucks |
21:20.28 | robin_sz | websae, sorry was AFK |
21:20.28 | Assid | youu change one thing.. and if you wanna access some other menu.. you gotta save it |
21:20.31 | Assid | thats more time gone |
21:20.36 | websae | ohh okay |
21:20.39 | robin_sz | websae, I dont know if I liek mine or not ... |
21:20.46 | websae | did you just get it? |
21:20.49 | robin_sz | its hard to tell if its any good |
21:20.56 | robin_sz | no had it 8 months |
21:21.05 | BadPacket | Assid: yeah - the sipura/linksys are better for that |
21:21.09 | Assid | yep |
21:21.20 | robin_sz | but I accidentally "upgraded" to the beta firmware and cant go back |
21:21.21 | Assid | thats why i live provisionng |
21:21.27 | Assid | you change everything you want to |
21:21.29 | Assid | and then reboot |
21:21.36 | robin_sz | display has been fscked ever since |
21:22.18 | *** join/#asterisk TripleFFFFFFFFF (n=TripleFF@147-102.mc.cite.net) |
21:22.23 | TripleFFFFFFFFF | hi all. |
21:22.33 | timscott | hello. |
21:22.36 | TripleFFFFFFFFF | im wondering why latest svn doent compile with mysql on freebsd |
21:22.41 | TripleFFFFFFFFF | anyone have a hint ? |
21:23.01 | tekati | So between the Polycom 301 and the 501 is the 501 the way to go? |
21:23.13 | tekati | And if so what are the main reasons? |
21:23.19 | Assid | okay someone try and get this wav file running for me.. i just cant get it |
21:23.24 | Assid | nnot on the 301 atleast |
21:23.48 | Assid | tekati: 3 line phone .. bigger screen.. full duplex speaker |
21:24.18 | robin_sz | TripleFFFFFFFFF, did you compile it? |
21:24.30 | TripleFFFFFFFFF | well im in ports and making as we speak |
21:24.39 | TripleFFFFFFFFF | as the /usr/src/asterisk-addons is not working |
21:24.48 | tekati | Full duplex speaker works for me. Enough right there. I have more conference calls then I care to talk about. Speaker phone on my Panasonic MultiTalk8 over a PAP2 just does not do to well all the time. |
21:25.09 | TripleFFFFFFFFF | voice# gmake |
21:25.09 | TripleFFFFFFFFF | ./mkdep -fPIC -I../asterisk -D_GNU_SOURCE -I/usr/local/include/mysql `ls *.c` |
21:25.09 | TripleFFFFFFFFF | app_addon_sql_mysql.c:25:27: asterisk/file.h: No such file or directory |
21:25.09 | TripleFFFFFFFFF | app_addon_sql_mysql.c:26:29: asterisk/logger.h: No such file or directory |
21:25.14 | TripleFFFFFFFFF | lots of that |
21:25.23 | websae | you sure have quite a few "F"s in your nick there... |
21:25.33 | Nugget | He's whiter than white! |
21:25.39 | TripleFFFFFFFFF | yeah |
21:25.48 | TripleFFFFFFFFF | can never remember my pass |
21:25.51 | TripleFFFFFFFFF | so i add a F |
21:26.03 | TripleFFFFFFFFF | so im doing the ports one |
21:26.12 | TripleFFFFFFFFF | just hope it wont pbreak my asterisk |
21:26.22 | Assid | okay so anyone wanna volunteer in seeing why this wav file just refuses to load up on poly301 |
21:27.20 | TripleFFFFFFFFF | do i need a flag for mysql on ports for freebsd ? |
21:27.20 | X-Rob | TripleFFFFFFFFF, you need to do a 'make install' in asterisk before you install addons |
21:27.22 | paryl | oh my word... i just figured out the solution to my problem |
21:27.23 | TripleFFFFFFFFF | oh well |
21:27.24 | TripleFFFFFFFFF | lol |
21:27.37 | paryl | is anyone around that was helping me before? |
21:28.18 | *** part/#asterisk tiCo89 (i=mario@debian.uid0.ch) |
21:28.46 | paryl | it seems the tdm2400p comes default with the moules installed backwards... so my one fxs module was in port 6, which meant i had to reference channels 21-24, rather than 1-4... sigh |
21:29.40 | TripleFFFFFFFFF | btw even when i got latest diium source i cant compile now |
21:29.40 | X-Rob | TripleFFFFFFFFF, you need to do a 'make install' in asterisk before you install addons |
21:29.42 | Assid | www.pienotech.com/ctu.wav .. i am trying to get this to work on polycom 301 |
21:30.05 | TripleFFFFFFFFF | yeah |
21:30.07 | mutilator | yea |
21:30.09 | TripleFFFFFFFFF | asterisk is already running |
21:30.12 | TripleFFFFFFFFF | fyi |
21:30.30 | tekati | Who has the best deal on the Polycom 501s? |
21:30.34 | nrw | ctu.wav: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono 44100 Hz |
21:30.35 | tzanger | voipsupply likely |
21:30.43 | nrw | it has to be a 8khz 8 bit file |
21:30.48 | nrw | thats why its not working |
21:31.03 | Assid | nrw: i tried .. sox ctu.wav -c1 -r8000 ...... |
21:31.07 | Assid | didnt work.. |
21:31.14 | Assid | then i tried with -r16000 didnt work |
21:31.25 | TripleFFFFFFFFF | latest svn |
21:31.26 | TripleFFFFFFFFF | gmake[1]: Entering directory `/usr/src/asterisk/apps' |
21:31.27 | TripleFFFFFFFFF | Makefile:16: *** missing separator. Stop. |
21:31.27 | TripleFFFFFFFFF | gmake[1]: Leaving directory `/usr/src/asterisk/apps' |
21:31.31 | TripleFFFFFFFFF | lol |
21:32.15 | TripleFFFFFFFFF | <<<<<<< .mine |
21:32.17 | TripleFFFFFFFFF | on this line |
21:32.20 | TripleFFFFFFFFF | in apps/Makefile |
21:32.47 | Assid | nrw? |
21:33.28 | *** join/#asterisk RoyK (n=roy@ti211210a080-0268.bb.online.no) |
21:33.51 | TripleFFFFFFFFF | <<<<<<< .mine |
21:33.56 | TripleFFFFFFFFF | what are these in the makefiles ? |
21:34.05 | russellb | TripleFFFFFFFFF: svn revert -R . |
21:34.12 | TripleFFFFFFFFF | ?? |
21:34.34 | TripleFFFFFFFFF | wow |
21:34.36 | TripleFFFFFFFFF | russe |
21:34.38 | *** join/#asterisk RoyK (n=roy@ti211210a080-0268.bb.online.no) |
21:34.39 | robin_sz | ugh ... over a month since the last (broken) gxp 2000 release, and still no sign of a fix .. im beginning to wonder if they are using the infinite-number-of-minkeys method of development |
21:34.43 | TripleFFFFFFFFF | you da man.. so what did that do ? |
21:34.47 | russellb | TripleFFFFFFFFF: :D |
21:34.48 | robin_sz | seldom have I seen such slow progress |
21:35.00 | X-Rob | robin_sz, sure you have. |
21:35.06 | russellb | TripleFFFFFFFFF: you had a conflict when you updated for some reason, so that command reverted local changes. |
21:35.06 | TripleFFFFFFFFF | it just reverted my svn to old version ? |
21:35.07 | X-Rob | think of, uh.. |
21:35.08 | X-Rob | um |
21:35.13 | TripleFFFFFFFFF | oh |
21:35.14 | X-Rob | OK, no, you win. |
21:35.18 | X-Rob | It's crap. |
21:35.19 | russellb | TripleFFFFFFFFF: earlier something was removed and then added back, so that probably caused it |
21:36.02 | TripleFFFFFFFFF | so from svn i say nothing more to make resmysql ? |
21:36.13 | X-Rob | TripleFFFFFFFFF, you need to do a 'make install' in asterisk before you install addons |
21:36.16 | TripleFFFFFFFFF | just install base.. then go addons and make mysql then remake base ? |
21:36.38 | nrw | triplef: I am curious why you dont just install the asterisk and asterisk-addons ports. |
21:36.44 | TripleFFFFFFFFF | oh |
21:36.47 | nrw | instead of trying to compile the latest svn |
21:36.49 | TripleFFFFFFFFF | theres asterisk-addons port lol |
21:36.53 | *** join/#asterisk Abydos313 (i=abydos31@ppp-71-133-210-73.dsl.irvnca.pacbell.net) |
21:37.00 | TripleFFFFFFFFF | ok doing |
21:37.22 | TripleFFFFFFFFF | ./usr/ports/net/asterisk-addons |
21:37.23 | TripleFFFFFFFFF | true |
21:37.25 | TripleFFFFFFFFF | ok |
21:37.28 | TripleFFFFFFFFF | that good then |
21:37.28 | robin_sz | X-Rob, where? |
21:37.31 | TripleFFFFFFFFF | lol |
21:37.36 | robin_sz | ok I win > |
21:37.38 | robin_sz | :) |
21:37.58 | TripleFFFFFFFFF | wow with h323 also .. nice.. now i can code my video phone |
21:38.09 | TripleFFFFFFFFF | anyone got xten for pocket pc to work ? |
21:38.16 | Assid | nrw: any luck? |
21:39.03 | nrw | assid: im not trying. I just wanted to make sure you changed its format. I'm at home i couldnt tell you if it worked even if i did do it |
21:39.06 | X-Rob | IF any of the *bsd'ers would like to do a freepbx install documentation, I'd be most appreciative. It's very much centos based at the moment |
21:39.22 | X-Rob | but there's no reason why it wouldn't work on any of the bsds |
21:39.49 | robin_sz | I am willing to wrote up the debian install documentation if you want |
21:39.59 | fourcheeze | can the MYSQL command be used with a persistent connection somehow |
21:40.01 | robin_sz | ok ... here goes ... |
21:40.11 | tekati | Anyone have a polycom 501 or cisco 79XX phone they want to trade for a TDM40B? |
21:40.15 | robin_sz | "apt-get install asterisk" |
21:40.22 | robin_sz | there, how did I do? |
21:40.35 | X-Rob | robin_sz, well that's a start. |
21:40.39 | fourcheeze | robin_sz: you missed out "apt-get update" |
21:40.40 | Assid | nrw: how did you get the info? |
21:40.48 | nrw | assid: the file command |
21:41.01 | robin_sz | dang, me and my partial documentation |
21:41.10 | X-Rob | http://aussievoip.com.au/wiki/freePBX-Debian |
21:41.30 | Assid | ctu2.wav: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono 16000 Hz -- that doesntwork |
21:41.47 | russellb | fourcheeze: yeah, use source and checkinstall |
21:41.57 | robin_sz | always remember to apt-get remove before soing a make install .... |
21:42.04 | robin_sz | or Bad Things happen |
21:42.17 | fourcheeze | russellb: what's checkinstall ? |
21:42.22 | *** join/#asterisk Druken (n=Druken@CPE00121716da99-CM00159a090acc.cpe.net.cable.rogers.com) |
21:42.31 | nrw | assid: and there you have your answer to why its not working |
21:42.32 | russellb | it runs make install but keeps track of what gets installed |
21:42.42 | russellb | so you can later remove it, just like you had installed it with apt |
21:43.02 | russellb | it's pretty nifty |
21:43.36 | Assid | nrw: doc says L16/160008 (16-bit, 16 kHz sampling rate, mono) is okay |
21:43.42 | fourcheeze | russellb: sounds cool |
21:43.47 | fourcheeze | I keep meaning to work out "stow" |
21:43.52 | Assid | when i tried 8khz .. it didnt work either |
21:53.41 | TripleFFFFFFFFF | wow |
21:53.43 | TripleFFFFFFFFF | http://brands.xten.net/x-lite/download/X-Lite_CE_Install.exe |
21:53.45 | TripleFFFFFFFFF | ;) |
21:53.54 | TripleFFFFFFFFF | freebie |
21:54.01 | TripleFFFFFFFFF | pocket pc |
21:54.09 | TripleFFFFFFFFF | sip .. ill try with bluettoth to my asterisk and see |
21:57.28 | Assid | aargh.. work you crazy pos |
21:58.09 | codebreaker | do i need zaptel complete or only the zaptel patch if i want to run asterisk without any cards? |
22:06.56 | *** join/#asterisk A-Tuin (n=a-tuin@steves.ip.v4.me.uk) |
22:07.06 | *** join/#asterisk Dr-Linux (n=Linux@202.59.73.131) |
22:10.38 | *** join/#asterisk RoyK (n=roy@ti211210a080-0268.bb.online.no) |
22:12.53 | Dr-Linux | hi gays |
22:12.55 | Dr-Linux | guys |
22:15.33 | Katty | herro. |
22:24.08 | tainted- | Dr-Linux freudian slip? |
22:25.22 | Dr-Linux | tainted-: what's freudian? |
22:25.41 | timscott | doing something accidental on purpose. |
22:26.07 | Dr-Linux | :S |
22:26.09 | Dr-Linux | and slip? |
22:26.19 | timscott | you slipped up, make a mistake |
22:26.29 | timscott | a freudian slip is a mistake that you meant to make |
22:26.33 | asterboy | your subconcious thoughts are your true intentions |
22:26.35 | Dr-Linux | timscott: you mean i said "hi gays" ? |
22:26.39 | timscott | aye |
22:26.44 | asterboy | brokeback asterisk |
22:27.14 | Dr-Linux | i c |
22:27.29 | Dr-Linux | if someone took my mistake on his heart, i'm sorry for him |
22:29.23 | *** join/#asterisk Ariel_ (n=Ariel@dsl-20-177.cofs.net) |
22:38.11 | *** part/#asterisk sulex (n=sulex@pdpc/supporter/active/sulex) |
22:52.28 | *** join/#asterisk esculapio__ (n=ESCulapi@245stb68.codetel.net.do) |
22:53.10 | *** join/#asterisk Dr-Linux (n=Linux@202.59.73.131) |
23:00.39 | blitzrage | lol |
23:00.58 | websae[A] | blitzrage: how arie ayou doing? how's that support ticket going? |
23:02.32 | blitzrage | websae[A]: is it still not resolved? not sure - unfortunately its out of my hands |
23:03.04 | blitzrage | oh I just tried it -- works for me |
23:10.10 | *** join/#asterisk b4ka (i=WinNT@200-127-198-118.cab.prima.net.ar) |
23:14.41 | *** join/#asterisk OloBola (n=not@netblock-68-183-67-158.dslextreme.com) |
23:17.17 | *** join/#asterisk dahunter3 (n=dahunter@pool-71-110-89-49.lsanca.dsl-w.verizon.net) |
23:20.56 | *** join/#asterisk Mnabil (n=Mnabil@196.205.192.21) |
23:28.41 | *** join/#asterisk MikeJ[Laptop] (n=vircuser@d14-69-8-30.try.wideopenwest.com) |
23:36.55 | *** join/#asterisk IceManRISK (n=kart@201.15.207.170) |
23:39.56 | *** join/#asterisk MstlyHrmls (n=mh@melbourne.mostly-harmless.ca) |
23:40.18 | timscott | :) |
23:43.49 | *** join/#asterisk justinu|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
23:45.07 | *** join/#asterisk TripleFFFFFFFFF (n=TripleFF@147-102.mc.cite.net) |
23:45.37 | TripleFFFFFFFFF | [func_logic.so] => (Logical dialplan functions) |
23:45.37 | TripleFFFFFFFFF | Apr 29 19:45:17 ERROR[5497]: pbx.c:1333 ast_custom_function_register: Function ISNULL already registered. |
23:45.37 | TripleFFFFFFFFF | Apr 29 19:45:17 ERROR[5497]: pbx.c:1333 ast_custom_function_register: Function SET already registered. |
23:45.37 | TripleFFFFFFFFF | Apr 29 19:45:17 ERROR[5497]: pbx.c:1333 ast_custom_function_register: Function EXISTS already registered. |
23:45.41 | TripleFFFFFFFFF | Ouch ... error while writing audio data: : Broken pipe |
23:45.43 | TripleFFFFFFFFF | ok |
23:45.46 | TripleFFFFFFFFF | that latest port |
23:45.49 | TripleFFFFFFFFF | anyidea ? |
23:46.24 | TripleFFFFFFFFF | need reboot |
23:46.24 | TripleFFFFFFFFF | brb |
23:46.27 | *** part/#asterisk TripleFFFFFFFFF (n=TripleFF@147-102.mc.cite.net) |
23:47.19 | Ariel_ | it's so slow here tonight..... |