00:00.06 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
00:01.49 | De_Mon | if it still loops, serverA is calling serverB and just having a grand time |
00:02.47 | Dr-Linux|home | De_Mon: i'd need to grab your brain for my problem :) |
00:03.32 | Dr-Linux|home | De_Mon: why i can't transfer the a call to agi() ? |
00:04.15 | *** join/#asterisk Skarmeth (n=Skarmeth@201009082231.user.veloxzone.com.br) |
00:05.17 | Dr-Linux|home | De_Mon: nevermind :) |
00:05.48 | Dr-Linux|home | heh, i don't if ever my this problem will resolved :) |
00:07.03 | *** join/#asterisk karmatronic (n=karmatro@84.77.152.248) |
00:07.39 | *** join/#asterisk Teeli (n=tili@172.Red-88-0-147.dynamicIP.rima-tde.net) |
00:08.57 | De_Mon | Dr-Linux|home your doing a SIP transfer to an extension that calls agi() and the script doesnt work or what? |
00:09.05 | *** join/#asterisk Ard0gx (n=adiaz0@200.87.243.211) |
00:09.29 | Dr-Linux|home | De_Mon: thanks for discussing this issue |
00:09.41 | *** join/#asterisk riddlebox (n=james@75-132-205-166.dhcp.stls.mo.charter.com) |
00:10.01 | Dr-Linux|home | De_Mon: correct, in that case agi() crashes |
00:10.28 | Dr-Linux|home | De_Mon: but if directly dial agi() extension that works just fine |
00:10.58 | De_Mon | why does it say it crashed? |
00:11.03 | Dr-Linux|home | De_Mon: i can transfer the call anywhere into the dialplan, but not to agi() |
00:11.10 | *** join/#asterisk Flusher (i=flusher@filer.euroserv.com) |
00:11.34 | awannabe | De_Mon, i dont have a user directive in my sip.conf |
00:12.03 | Dr-Linux|home | De_Mon: it simply reutruns complated 0 , same like when agi script is wrong |
00:12.05 | Hmmhesays | can someone send me a fax? |
00:12.33 | Dr-Linux|home | De_Mon: i tried different agi's , i tried different version, different servers, but same problem |
00:12.49 | De_Mon | awannabe awannabe any particular reason you're not using IAX for your innerasterisk communication? |
00:13.27 | Dr-Linux|home | De_Mon: it could be a channel reason maybe :S |
00:14.01 | *** part/#asterisk markit (n=konversa@host119-245-static.72-81-b.business.telecomitalia.it) |
00:14.14 | De_Mon | awannabe http://www.voip-info.org/wiki-Asterisk+-+dual+servers might help you out |
00:14.29 | De_Mon | http://www.voip-info.org/wiki/view/Asterisk+Connect+2+servers |
00:14.40 | De_Mon | that's another page, same topic |
00:14.55 | De_Mon | Dr-Linux|home have you tried a simplier agi script? |
00:15.33 | Dr-Linux|home | De_Mon: yes, it never works |
00:16.02 | Dr-Linux|home | De_Mon: maybe it's a bug |
00:16.28 | Dr-Linux|home | De_Mon: anthm said, it's "channel masqu..." problem, but i don't understand that |
00:17.38 | *** join/#asterisk TripleFFFF (n=TripleFF@145-27.mc.cite.net) |
00:17.49 | TripleFFFF | to have more then 1 email to voicmeail notif is pipe the delim ? |
00:18.47 | *** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox) |
00:21.17 | Ard0gx | Hi, doing TOP on my linux I see more than one asterisk process, is that normal ? |
00:21.38 | rudholm | it is |
00:22.48 | rene- | hello |
00:23.04 | rene- | where should a call process jump after a dial to a peer who has exceeded its call limit? |
00:24.33 | *** join/#asterisk j3g (i=nobody@189.6.32.159) |
00:25.22 | awannabe | De_Mon, yeah ive been looking at that |
00:25.45 | j3g | i am trying to get my VOIP provider to work with my asterisk box... everything seems to be set up correctly... but when I dial, I get no sound at all.. but If I Press the HOLD button on the softphone, I can hear music on the regular phone line... the codecs are set to be the same... what could be wrong? |
00:27.15 | De_Mon | TripleFFFF no, tell your sendmail app that allusers@foo is multiple recipients |
00:27.45 | De_Mon | Ard0gx yes |
00:28.27 | De_Mon | Dr-Linux|home bummer, bug anthm some more :] |
00:28.48 | `Sean | eww |
00:28.51 | `Sean | Snom phones are soo ugly |
00:28.52 | De_Mon | maybe it works better in openpbx ;) |
00:28.58 | `Sean | there like those classic operator types |
00:29.18 | De_Mon | zoom zoom zoom i'm gona go eat eat eat |
00:29.52 | j3g | and is there a easy way to get asterisk to transcode GSM(my softphone) to g729 (my voip providers best protocol) ? |
00:34.26 | De_Mon | is g729 supported? |
00:35.08 | Ard0gx | De_Mon: but have more than 10 of it with just 50 calls is normal as well ? |
00:35.26 | Ritalin2 | if you have allow=g729 (and other's disallowed) won't it automatically do whatever transcoding it needs ? |
00:36.15 | De_Mon | yeah, it will |
00:36.37 | De_Mon | Ard0gx that sentance did not make sense |
00:37.23 | j3g | De_Mon: g729 is not supported by me (i didn't buy the codec) but my VOIP provider supports it |
00:37.48 | Ard0gx | De_Mon: well, the thing I have 10 asterisk process doing TOP but, I only have 50 calls doing show channels |
00:37.48 | De_Mon | yeah, um, asterisk needs to support it for you to send or receive |
00:38.06 | De_Mon | the number of asterisk processes is not related to the number of channels/calls |
00:39.31 | Ard0gx | De_Mon: yeah, that I understand, but I have another asterisk box running and I have this 50 calls or more and I just have one asterisk process |
00:39.43 | j3g | De_Mon: i have set my sip.conf to only allow ilbc and alaw for both my extension (softphone) and my ISP.. the connection seems to go well but no sound :( |
00:40.21 | TripleFFFF | de mon no |
00:40.30 | Ritalin2 | i didn't realize you had to buy g729 |
00:40.36 | *** join/#asterisk sahafeez (n=sahafeez@67.109.14.227.ptr.us.xo.net) |
00:40.48 | `Sean | Ritalin2 you dont.. |
00:40.52 | Ard0gx | De_Mon: so, I don't understand why appears these asterisk process. Doing 'stop now' the main asterisk process dead, but all other one noT |
00:40.52 | `Sean | there is a free way search |
00:40.52 | `Sean | :P |
00:41.02 | TripleFFFF | i want ozzy on crack as MOH |
00:41.29 | *** join/#asterisk sjobeck (n=sjobeck@static-70-104-254-218.ptldor.dsl-w.verizon.net) |
00:41.58 | Ritalin2 | I wish I had VPN setup :-/ |
00:42.39 | *** part/#asterisk TripleFFFF (n=TripleFF@145-27.mc.cite.net) |
00:43.54 | *** join/#asterisk Tene (n=tene@poipu/supporter/slacker/tene) |
00:45.19 | De_Mon | Ard0gx if they arnt dying from stop now they arnt threads |
00:45.32 | De_Mon | what distribution are you using? |
00:45.42 | De_Mon | or did you compile yourself |
00:47.31 | Ard0gx | De_Mo: I'm using RHEL 4.0 |
00:48.16 | rene- | are hints useful when your phones dont support subscriptions? |
00:48.36 | Ard0gx | with 'asterisk-1.2.12.1' and 'zaptel-1.2.9.1' |
00:48.54 | *** join/#asterisk ShadowHntr (i=sentinel@wikipedia/Shadowhntr) |
00:50.33 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
00:50.33 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
00:51.51 | rene- | so basically asterisk publishes information of extensions tru hints and subscriptions capable devices blink leds to extensions subscribed to? |
00:52.50 | j3g | one thing I noticed now... If I dial from asterisk to a regular phone (using my voip provider) and I keep talking on the microphone on the PC I get a few (maybe 2) seconds of sound going to the regular phone line.. then all gets silent |
00:54.39 | *** join/#asterisk ariel_ (n=ariel_@dsl-20-177.cofs.net) |
00:56.33 | *** part/#asterisk _Thor (i=CS@user-vc8fl7l.biz.mindspring.com) |
00:57.39 | perd | anyone here use 7960 with sip? i have an easy and quick question about it |
00:59.21 | retrogradeorbit | yes. 7960, 7941 et al with sip. shoot yr question |
01:05.37 | *** join/#asterisk ManxPower (n=manxpowe@68.113.119.198) |
01:05.42 | *** join/#asterisk jeebusroxors (n=jeebusro@cpe-75-80-231-237.dc.res.rr.com) |
01:07.38 | *** join/#asterisk olsen_ (n=Diego@200.61.236.51) |
01:13.09 | *** join/#asterisk RoyK (n=roy@217-175-39.100710.adsl.tele2.no) |
01:14.06 | *** join/#asterisk rnst (i=workbenc@201.217.19.70) |
01:14.29 | *** join/#asterisk fatgoose (n=samuel@206-248-156-131.dsl.teksavvy.com) |
01:14.46 | *** part/#asterisk fatgoose (n=samuel@206-248-156-131.dsl.teksavvy.com) |
01:17.56 | Druken | anyone have experince with via sata raids ? |
01:20.53 | awannabe | zaptel can be a biatch! |
01:22.19 | *** join/#asterisk sbingner (n=sam@pdpc/supporter/sustaining/sbingner) |
01:31.15 | awannabe | anyone around to help me trouble shoot a dual server SIP issue |
01:35.16 | *** join/#asterisk oQPa (n=roque@15.Red-83-40-197.dynamicIP.rima-tde.net) |
01:35.48 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
01:35.51 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
01:35.51 | *** mode/#asterisk [+o mog] by ChanServ |
01:39.04 | Ritalin2 | awannabe: what now |
01:39.35 | awannabe | still doesnt work, something with the context I guess |
01:39.38 | Ritalin2 | what is it you are trying to accomplish |
01:39.41 | awannabe | it just does a loop |
01:40.01 | awannabe | place calls from serverA to serverB, serverA has PRIs in it |
01:40.07 | awannabe | so serverB can use the dialtone acourse |
01:40.07 | Ritalin2 | okay how do you have things setup physically? where are the two server's etc |
01:40.17 | awannabe | across a WAN link |
01:40.26 | Ritalin2 | are they in different locations or something? |
01:40.33 | *** join/#asterisk dorphalsig (n=dorphals@pcsp163-73.supercabletv.net.co) |
01:40.42 | awannabe | yeah |
01:40.53 | Ritalin2 | i think what you need to do is set up a link between the two as a trunk |
01:41.06 | awannabe | hrmm |
01:41.06 | Ritalin2 | though i dont know this for sure since i've never tried to do it. that's just my understanding of how things work |
01:41.23 | awannabe | i mean IP works fine, so i dont think it needs a trunk |
01:41.38 | awannabe | or from what ive read anyways |
01:41.49 | Ritalin2 | are they both running asterisk? |
01:41.50 | dorphalsig | Ummm hi, does anybody know if I need a gatekeeper in order to be able to place/recieve calls thru oH323? |
01:42.27 | awannabe | yeah both asterisk |
01:42.34 | awannabe | same version, same hardware, all the same |
01:42.53 | dorphalsig | I'm trying to make a Cisco speak with * 1.2.14 |
01:43.10 | dorphalsig | is it a lost case? |
01:43.17 | Ritalin2 | awannabe: i dont know how to pass an exten from one server to the other. i think what you want to do is dial like 9 or something to connect to the other server and then configure an extension on the other server so that it automatically gives a dialtone when you connect... enter your number and then it uses that to dial |
01:43.27 | Ritalin2 | i mean... that'd be one solution ... again i'm no expert |
01:43.30 | dorphalsig | (and excuse my ignorance but wtf does a gatekeeper do?) |
01:43.54 | awannabe | yeah ie seen it done several times before without all that garbage |
01:44.42 | Ritalin2 | you setup so that the remote server is registered? |
01:44.49 | awannabe | nope |
01:45.47 | Ritalin2 | how are you connecting to the other server? |
01:45.56 | ManxPower | "Gatekeeper" is an H323 term. Think of it as sort of an H323 "router" or "proxy" (neither term is really correct) |
01:45.57 | [TK]D-Fender | awannabe : You set up 1 SIP account on each side just like you would a phone. You'd create a pattern match that will then pass the call off that account to the other server and it will process it just like a normal call from a standard phone. |
01:46.14 | awannabe | [TK]D-Fender, thats what i thought, but i get a nasty loop |
01:46.17 | *** join/#asterisk Iamnacho (i=Iamnacho@ip68-103-140-124.ks.ok.cox.net) |
01:46.39 | perd | with the sip firmware on the 7960 i cant just start dialing numbers on the dialpad like i could with the skinny firmware, is there a setting for this or is it something in the dialplan.xml ? |
01:46.41 | [TK]D-Fender | awannabe : then you need to fix contexts & prefixes. |
01:46.51 | awannabe | yeah, context is the prob |
01:46.57 | dorphalsig | But, do I really need it in order to use asterisk with H323? |
01:47.02 | Ritalin2 | or you don't have the serverA->serverB sip set up? |
01:47.04 | ManxPower | If you get a loop, then you have a dialplan problem, not a SIP problem |
01:47.07 | awannabe | what should the context for the sip entries be then? |
01:47.14 | ManxPower | dorphalsig: I don't think so. |
01:47.20 | awannabe | thats whats confushing me i guess |
01:47.43 | ManxPower | dorphalsig: The only people that use H323 with Asterisk are 1) People that like pain and 2) People that have no other choice. |
01:48.11 | [TK]D-Fender | awannabe : I'm thikning you're passing an exten that happens to be the pattern match for heading the OTHER way as well... make sure they DON'T match or that your inbound context processes what its given and don't jsut feed it back in as-in (like stip prefixes, etc. |
01:49.05 | awannabe | [TK]D-Fender, yeah it is...cause that pattern it suppsoe to send it out the PRI |
01:49.06 | ManxPower | Remember, if you have two pattern matches that could match a number, any pattern match in an include =>'d context will match last and never match if there is something more specific. |
01:49.35 | bok | anyone know if there is a way to limit the calls to a sip phone to one, even if that phone has multiple lines registered. ie if someone is on the phone all lines are busy? |
01:49.47 | ManxPower | [TK]D-Fender: I'll bet he's doing the classic newbie of exten => _XXX,1,Dial(SIP/${EXTEN}) instead of an individual Dial() line for each extension |
01:50.19 | ManxPower | bok: That happens by default on almost all phones if you set up a separate SIP account for each line. |
01:50.33 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
01:50.33 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
01:50.43 | bok | ManxPower: ahh ok, so its a phone setting then, cheers! |
01:51.16 | perd | ManxPower: so it's bad to do like.. _88XX,1,Dial.... ? |
01:51.25 | awannabe | ManxPower, exten => _NXXNXXNXXX,1,DIAL(SIP/super/${EXTEN},200,tr) is bad? |
01:51.26 | perd | you enter every extension in separately? |
01:51.46 | ManxPower | perd: of course. |
01:51.57 | perd | i see, what is the reason for that? |
01:52.07 | ManxPower | How else would you allow users to do custom call routing, custom features, etc |
01:52.14 | perd | seems like entering in a billion extensions would be a bad idea, since you waste time |
01:52.29 | ManxPower | Heck, I could not even have a fax->email gateway if I didn't do it that way |
01:52.43 | ManxPower | perd: you only do it once for each extension when it is created or changed |
01:52.48 | perd | i have a fax->email gateway and i do it that way |
01:53.12 | ManxPower | perd: how many users have the fax->email gateway? |
01:53.17 | perd | 25 |
01:53.21 | perd | all of 'em |
01:53.23 | ManxPower | and how do you map and extension to an e-mail address? |
01:53.23 | perd | small office :) |
01:53.36 | perd | i use iaxmodem |
01:53.40 | perd | and hylafax |
01:54.04 | perd | the recvdfax script checks iax.conf, each persons 'modem' devined there has a ;email= line |
01:54.19 | perd | i just parse the iax.conf, get the email, convert tiff to pdf and shoot it out |
01:54.30 | perd | i couldnt get spandsp working with 1.4 |
01:54.39 | perd | well, for app_txfax |
01:54.47 | [TK]D-Fender | ManxPower : For internal extension, I wouldn't say its "bad"..... |
01:54.57 | [TK]D-Fender | ManxPower : Let them 404... who cares? |
01:55.15 | perd | when people dial a bad extension they get a message |
01:55.24 | *** join/#asterisk Juggie (n=Juggie@CPE00c049d9f271-CM000a73a18a20.cpe.net.cable.rogers.com) |
01:55.44 | perd | i use voicemail to do that.. if no voicemailbox is found it dumps out of the macro and goes to _88XX,2, which is pbx-invalid |
01:55.54 | ManxPower | part of my dialplan (domain is fake, of course) http://pastebin.ca/313754 |
01:56.26 | awannabe | oh shit |
01:56.55 | ManxPower | [TK]D-Fender: I use pattern match for dialed extensions that do not match a valid extension |
01:57.30 | ManxPower | As you can see there is SOME duplication of extension call routing, but not much |
01:57.31 | awannabe | yeah |
01:58.09 | ManxPower | Also, we do not use the extension as the SIP device ID for two reasons 1) EXTENSIONS AND DEVICES ARE NOT THE SAME THING and 2) it is easier for us. |
01:58.47 | perd | my extensions.conf is kind of fucked up |
01:59.03 | perd | heh, this is how i do my internal extensions http://pastebin.ca/313757 |
01:59.19 | perd | i have been flipping between defining each individually and using the pattern matching |
02:00.00 | ManxPower | do it individually. It was massivly more flexable |
02:00.12 | perd | yeh i probably will |
02:00.12 | Nivex | <PROTECTED> |
02:00.50 | ManxPower | macro-std-exten basically has a set of defaults, then you can override the defaults with setting dialplan channel variables |
02:01.24 | awannabe | that was both of them |
02:01.51 | ManxPower | Also I can find any specific line appearance on any phone by having the user read me the numbers off the little white sticker on the bottom of the phone (the MAC) |
02:02.15 | ManxPower | That is good when 1 extension rings 4 different line appearances on 3 different phones |
02:02.30 | perd | what are line appearances? multiple extensions on one phone? |
02:02.45 | perd | multiple phones ringing from one DID? |
02:02.47 | ManxPower | line appearance == line on the phone |
02:02.52 | perd | oh |
02:02.57 | ManxPower | i.e. a 3-line phone has 3 line appearances |
02:03.01 | perd | right |
02:03.13 | ManxPower | but since they are not actually lines..... |
02:04.34 | *** join/#asterisk shepimport (n=shep@h194.189.31.71.ip.alltel.net) |
02:05.12 | shepimport | Hey all anyone have a sec to work on a trunk registration issue? |
02:05.43 | ManxPower | ~freepbx |
02:05.46 | jbot | freepbx is, like, unable to be supported here. It's a complex piece of dialplan and GUI, and can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
02:05.49 | [TK]D-Fender | ManxPower : lol |
02:06.09 | [TK]D-Fender | ManxPower : Down boy! Let him finish incriminating himself, THEN decapitate ;) |
02:06.32 | ManxPower | he said "trunk" |
02:06.44 | [TK]D-Fender | ManxPower : There is a small-moderate chance against the "easy guess" of "trunk" :) |
02:07.09 | [TK]D-Fender | ManxPower : the term DOES get thrown around a lot... not enough to execute on for me yet... |
02:07.31 | shepimport | wow... thats the community spirit... actually i am a senior sales engineer for a session border controller company and work with sip signalling daily... but.. maybe they are more polite in the freepbx room |
02:07.49 | [TK]D-Fender | shepimport : So you are indeed using FreePBX? |
02:08.15 | shepimport | no, just writing to conf files |
02:08.53 | [TK]D-Fender | shepimport : Well then go right ahead and pastebin up your errors, and your sip config's (mask only passwords). |
02:08.57 | [TK]D-Fender | use www.pastebin.ca |
02:09.33 | shepimport | i am in the cli... created a new trunk, which is actually an ext off a broadsoft server (through an acme SBC) |
02:09.53 | shepimport | i see it register aok... than it loses the authentication... |
02:10.02 | shepimport | one sec i will paste |
02:12.42 | Hmmhesays | [TK]D-Fender your incoming fax working yet? |
02:13.03 | shepimport | http://pastebin.ca/313769 |
02:13.37 | rene- | it is funny i thought asterisk was the acme voice product |
02:14.24 | *** join/#asterisk lowlevel (n=Stuart@CPE000e0c057fad-CM000f9f7d6742.cpe.net.cable.rogers.com) |
02:14.26 | [TK]D-Fender | Hmmhesays : Didn't check... |
02:15.40 | [TK]D-Fender | shepimport : You're IP's don't match between your peer & register |
02:16.32 | shepimport | no they do... i was changing them since the SBC is not fully locked down... but i just missed a few |
02:17.55 | [TK]D-Fender | shepimport : Missinga few doesn't SOUND good to me... |
02:18.03 | Hmmhesays | i just faxed you something |
02:18.34 | shepimport | i meant when i was chaning them... they all match in whats in the conf file |
02:19.18 | [TK]D-Fender | shepimport : Look at your pastebin. the register & peer IP's simply don't match |
02:20.05 | *** join/#asterisk stkn_ (i=nobody@gentoo/developer/pdpc.active.stkn) |
02:20.48 | perd | he changed them for obfuscations sake, fender |
02:20.57 | ManxPower | shepimport: Ah, that explains the use of "trunk". |
02:20.58 | shepimport | fender: check it now... they do... i pulled that from a file i did not save |
02:21.44 | ManxPower | shepimport: is either end of the connection behind NAT? |
02:21.59 | shepimport | even off ethereal thecaptures just show a standard username/authentication mismatch... but i know it is entered correctly |
02:22.02 | [TK]D-Fender | clean it all up, and test again. Also, in no setup for BV I've every seen do you enter raw IP's like tha, esp for named domains.... |
02:22.14 | [TK]D-Fender | shepimport : Feels dangerous |
02:22.31 | retrogradeorbit | perd, there is a regular expression syntax in the phones config that deals with numbers and delays to deal with call initialisation |
02:22.33 | *** join/#asterisk _Soul_ (n=Soul@87-196-105-234.net.novis.pt) |
02:23.10 | retrogradeorbit | ie at what point the number is considered 'complete' and the SIP call begins |
02:23.11 | shepimport | BV? BW? |
02:23.28 | ManxPower | shepimport: diagram it. i.e. Asterisk -> Internet -> NAT Router -> Remote SIP server or however it is. |
02:23.33 | [TK]D-Fender | shepimport : nevermind, I misread something earlier |
02:23.56 | [TK]D-Fender | shepimport : This is your BroadSoft server... It hough it was an ITSP... |
02:24.58 | ManxPower | It sounds like Asterisk (SIP) <-> ???? <-> Broadsoft SIP Box |
02:25.04 | ManxPower | IS that correct, shepimport? |
02:25.07 | shepimport | it is behind a NAT router... but the sofswitch is behind a Session border controller... so that does the NAT transversal by creating a remote ALG on my firewall... so the externalnat entry is not required |
02:25.40 | ManxPower | shepimport: so you are SURE the issue is not with the NAT box closing the translations because of inactivity? |
02:25.40 | shepimport | Asterisk --> nat ---> Acme SBC --> Broadworks/sof |
02:26.16 | shepimport | the sbc is set to send an options every 2 seconds... |
02:26.28 | shepimport | can you use a qualify on a trunk? |
02:26.57 | ManxPower | shepimport: In Asterisk there is no such thing as a "SIP trunk" They are all just devices |
02:27.41 | ManxPower | The only people that say "SIP trunk" are the ones using some sissy GUI that complicates the config files so much we all just want to die. |
02:27.45 | shepimport | so let me add a qualify and a external NAT entry... and get back to ya... thanks for the support |
02:28.04 | ManxPower | shepimport: qualify basically sends an OPTIONS every second or so |
02:28.11 | *** join/#asterisk battini (n=inittab@cpe-24-209-36-174.neo.res.rr.com) |
02:28.31 | shepimport | yeah... qualify does one every 2 seconds unless maunally set |
02:29.07 | shepimport | ok thanks a lot.. i will get back to yall and let ya know the outcome |
02:30.37 | *** join/#asterisk zmef420 (n=zmef420@metarb3-pool4-182.mtco.com) |
02:30.49 | *** part/#asterisk Ard0gx (n=adiaz0@200.87.243.211) |
02:33.00 | *** join/#asterisk Newbie___ (n=me@60.49.23.186) |
02:33.01 | *** part/#asterisk Ritalin2 (n=dave@74-34-103-241.dsl1.pwll.tn.frontiernet.net) |
02:33.52 | awannabe | [TK]D-Fender, you have any docs on setting two server up to talk to each other? cause i am having a hell of a time!! |
02:34.22 | km- | Heh, I have a bit of a weird situation here, and I think I'm just doing something stupid. I have an ATA device. Lets say I have a call going on that ATA. I flash over and dial 700, thinking I'm parking the call in progress. Oddly enough, it parks the new call instead. |
02:34.51 | ManxPower | km-: you need to complete the transfer |
02:34.54 | *** join/#asterisk Avochelm (n=damien@gw-morphett.koalatelecom.com.au) |
02:35.32 | ManxPower | If you didn't have to do that then you would never hear the parking slot number (the person being parked would hear it) |
02:35.54 | ManxPower | Parking a call is just an attended transfer to the parking extension |
02:35.54 | km- | I think I found my error. |
02:36.03 | km- | If I hit #, it brings up allison and "Transfer!" |
02:36.05 | [TK]D-Fender | awannabe : lookup "asterisk dual servers" on the WIKI, that'll give you about 3 ways to do it |
02:36.10 | km- | If I flash, however, it opens a second line. |
02:36.36 | *** join/#asterisk Crashsys (n=kumba@72.187.211.158) |
02:36.39 | ManxPower | km-: what happens if you just hangup after you hear the parkinglot number. The call on hold should then be transfered |
02:36.51 | km- | yep, if I hit # it transfers. |
02:36.56 | km- | It was a misunderstanding on my part |
02:37.00 | km- | I thought flash = transfer |
02:37.00 | awannabe | [TK]D-Fender, yeah ive tried that |
02:37.01 | Strom_C | oh hell, not # inbound transfer |
02:37.02 | km- | but it's # = transfer. |
02:37.05 | ManxPower | km-: DO you know how to do an attended transfer using FLASH on your ATA? |
02:37.07 | Crashsys | Anyone have problems with a Sangoma A200D being quiet on rx? |
02:37.16 | km- | ManxPower: nope, have no idea. |
02:37.17 | ManxPower | km-: no, FLASH would START a transfer. You still have to complete it. |
02:37.30 | km- | how would I do an attended transfer on a sip1000? |
02:37.32 | ManxPower | km-: figure it out and you'll be able to park with FLASH |
02:37.33 | km- | err, spa1000 |
02:37.36 | km- | hmm. |
02:37.41 | ManxPower | km-: I'm sure it's in the manual |
02:38.07 | km- | I will putter with it some more |
02:38.13 | Strom_C | km-: try this |
02:38.13 | ManxPower | I thought it was FLASH + DIAL NUMBER + HANGUP |
02:38.22 | Strom_C | flash, dial number, hang oh damn you |
02:38.29 | ManxPower | km-: it is working the way it is supposed to work |
02:38.37 | km- | I'm not insinuating it isn't |
02:38.42 | *** join/#asterisk BSDTech (n=RNeese@ppp-69-238-75-25.dsl.irvnca.pacbell.net) |
02:38.42 | Strom_C | that'll teach you to type faster than me! |
02:38.45 | km- | I'm saying I'm a dumbass and need to play with it more :) |
02:38.48 | ManxPower | ouch! |
02:38.55 | *** join/#asterisk BZBW (n=wlwzhang@ip67-153-142-110.z142-153-67.customer.algx.net) |
02:39.04 | ManxPower | km-: Attended Transfer is required to do parking |
02:39.21 | Qwell | Strom_C: http://consumerist.com/consumer/sprint/cancel-sprint-account-by-writing-intelligent-wellcrafted-emails-to-the-ceo-228053.php |
02:39.23 | km- | the "#" starts an attended transfer, I believe. Since dialing 700 will indeed give me the "7 0 1" |
02:39.25 | ManxPower | Regardless of if it is a FLASH attended transfer or a # attended transfer |
02:39.27 | BZBW | strange, no one complains about BLF not working in 1.4, other than me:( |
02:39.47 | Qwell | BZBW: Did you try the latest branch 1.4? |
02:39.50 | *** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net) |
02:39.52 | ManxPower | BZBW: Guess what version of Asterisk I'm running? |
02:40.22 | BZBW | 1.2.10? |
02:40.30 | Qwell | ManxPower: 1.0.8? |
02:40.52 | ManxPower | 1.2.x |
02:40.58 | ManxPower | Do you know why I'm running 1.2.x? |
02:41.03 | *** join/#asterisk mjh001 (n=mjh001@c-68-37-78-102.hsd1.nj.comcast.net) |
02:41.04 | BZBW | Qwell: why latest branch, I download the official one, is there any different official 1.4? |
02:41.15 | Qwell | BZBW: there are bug fixes and such |
02:41.18 | ManxPower | Because 1.4.x has not been out long enough to have the critical bugs and not so critical discovered and fixed |
02:41.48 | BZBW | 1.4 has been in beta for more than 3 months:(:( |
02:41.49 | ManxPower | I call this the "I'm tired of users bitching at me" plan for upgrades |
02:42.45 | BZBW | Qwell, if I need the latest branch, I can have to use SVN, right? WHat branch is it? |
02:43.01 | Qwell | http://svn.digium.com/svn/asterisk/branches/1.4 |
02:43.35 | BZBW | I hope it doesn't introduce other serious bugs:? |
02:43.41 | *** join/#asterisk shepimport (n=shep@h194.189.31.71.ip.alltel.net) |
02:44.33 | shepimport | Hey guys that did not work (qualify or externip/localnet) will try again tommorow... thanks for the effort... night!!! need to get home before the mrs strangles me |
02:46.01 | awannabe | [TK]D-Fender, each phone doesnt have to be registed on the other server, right? |
02:46.34 | Crashsys | Are milliwatt test-tone numbers generally a default number or is it all specific to provider? |
02:46.40 | km- | OK. I think I figured it out. |
02:46.49 | Qwell | Crashsys: it's specific to a provider and area. call them and ask for it |
02:46.50 | [TK]D-Fender | awannabe : not at all. |
02:46.56 | km- | I flash. The other call connects. I flash back, both parties are connected. I then hang up. Then the transfer completes. |
02:47.08 | [TK]D-Fender | awannabe : you set up 1 account on each side, and the PBX's call each other to bridge the calls. |
02:47.13 | awannabe | [TK]D-Fender, anything else i need to check, if both server have static IPs, i dont need a regsiter =>, correct? |
02:47.18 | Crashsys | Qwell: heh, I tried calling the operator and "help desk" and they said "milli-what?" |
02:47.27 | Crashsys | maybe i'll run a line-truck off the road and ask them |
02:48.56 | infinity1 | i have a few polycom phones in the office and the pbx is outside the office so we're using NAT. Obviously we're having problems getting the polycoms to work 100%. Is there a solution? |
02:49.26 | [TK]D-Fender | awannabe Correct |
02:50.33 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
02:50.33 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
02:52.01 | awannabe | [TK]D-Fender, my dial() that is SIP/server, server has a entry [server] entry in sip.conf |
02:56.54 | awannabe | [TK]D-Fender, Forbidden - wrong password on authentication for INVITE to '"Front Desk" <sip:306@192.168.166.10>;tag=as336bd92f' i get that, but the phone is registered! |
02:59.07 | BZBW | I have a SIP service provider, only allow one call at the time, is there a function for me to detect if that peer account is used so that I will route the call to another SIP service provider? |
02:59.56 | BZBW | ChanIsAvail is not the one, that i know of. |
02:59.59 | awannabe | OMG!!! ok, it works!! |
03:03.24 | [TK]D-Fender | awannabe : yay |
03:03.42 | awannabe | hella weird |
03:03.48 | awannabe | username on the phone, i switched it, it works |
03:04.01 | awannabe | now, i cant dial inward from PSTN to other side, sayin unknown hostname, so i gotta see what the heck that is |
03:05.54 | Crashsys | Does a sangoma A200d have automatic gain control? |
03:06.09 | ManxPower | Crashsys: any reason to think it might? |
03:06.47 | Crashsys | Hoping it might :) |
03:07.12 | Crashsys | Cause I have 16.0 set for my rxgain in zapata.conf and the people in the office say it's still to quiet... |
03:07.27 | Crashsys | seems like a lot |
03:07.55 | Crashsys | for POTS |
03:08.42 | [TK]D-Fender | Crashsys : on an A200? |
03:08.52 | Crashsys | Yeah |
03:09.04 | Crashsys | Polycom IP430's... |
03:09.06 | [TK]D-Fender | Crashsys : I'm guessing not... syour lines are whacked... |
03:09.10 | danp | hmm |
03:09.14 | [TK]D-Fender | Crashsys : what firmware? |
03:09.22 | Crashsys | The A200 is v07 |
03:09.31 | [TK]D-Fender | Crashsys : the poly's |
03:09.32 | danp | i have mine set to 8.0 |
03:09.40 | Crashsys | Sip 1.6.7 |
03:10.04 | Crashsys | Tomorrow i'm going to walk around and make sure the volume is up on all the handsets... |
03:10.14 | Crashsys | Since the one computer/phone I sat at sounded plenty loud to me today |
03:10.24 | [TK]D-Fender | Crashsys : I think thats it... there were 2 releases of 1.6.7, and I found that the upgrade to 2.X helped a LOT |
03:10.28 | Crashsys | For all I know is they turned the speakerphone up, then picked the handset up... |
03:10.36 | danp | my tx was pretty hot so i had to turn it down to -5.0 |
03:10.45 | Crashsys | My TX is fine... |
03:10.49 | Crashsys | it's the RX that's lame |
03:11.03 | Crashsys | according to ztmonitor... but it looks like i'm getting good signal where it's at... |
03:11.12 | Crashsys | fender: you had low-audio issues with 1.6.7 firmware? |
03:11.44 | Crashsys | I'm trying to find Verizon Tampa's milliwatt test tone number... |
03:12.28 | *** join/#asterisk TheCops (n=henri@got.securebinary.com) |
03:12.41 | [TK]D-Fender | Crashsys : yup |
03:13.08 | in-pt | hi all |
03:13.29 | in-pt | can anyone please tell me what is the syntax for dial string of skinny channel |
03:13.40 | Qwell | Dial(line@device) |
03:13.43 | Qwell | erm |
03:13.47 | Qwell | Dial(Skinny/line@device) |
03:13.59 | Crashsys | d-fender: Any reconfig necessary for upgrading? |
03:14.03 | Crashsys | Or just the same conf's you had? |
03:14.14 | in-pt | thnx Qwell |
03:15.09 | BSDTech | your all fired |
03:15.15 | Crashsys | good |
03:15.17 | BSDTech | wrong channel |
03:15.18 | Crashsys | now I can go get drunk |
03:15.23 | BSDTech | lol |
03:15.25 | Crashsys | ohh :( |
03:15.36 | BSDTech | its taking to long to fix the current issues |
03:15.46 | BSDTech | you all must work extra hours |
03:15.47 | wunderkin | Crashsys, hey since you have an ip430, and are on 1.6.7 now and maybe upgrading (?) can you test something for me |
03:15.54 | BSDTech | fix it then your fired |
03:16.09 | Crashsys | wunderkin: I'm not at the site... but what ya got in mind? |
03:16.20 | BSDTech | ip430 being a polycom why use the 1.6.7 |
03:16.28 | BSDTech | why not move to 2.0.3 |
03:16.30 | in-pt | what is the reson for this error on asterisk cli Got SIP response 489 "Bad event" back from 192.168.0.80 |
03:16.39 | wunderkin | you don't have the phone where you're at? darn.. well how about someone else? |
03:16.41 | Crashsys | BSD: Cause I didn't think of it at the time? |
03:16.57 | BSDTech | ok |
03:17.12 | Crashsys | Do I need to re-write the config's for 2.0 or will 1.6 configs run under a 2.0 sip without issues? That is my current mission... |
03:17.22 | wunderkin | Crashsys, redo |
03:17.25 | BSDTech | 1.6 should work fine |
03:17.29 | [TK]D-Fender | Crashsys : Definately must rewrite |
03:17.31 | BSDTech | mine do |
03:17.34 | Crashsys | ... |
03:17.37 | [TK]D-Fender | bad risk... |
03:17.43 | [TK]D-Fender | many parms changed. |
03:17.44 | BSDTech | they are the same |
03:17.48 | wunderkin | so i guess i shouldnt ask BSDTech to test for me :D |
03:17.51 | [TK]D-Fender | I would NOT take chances with this |
03:18.14 | BSDTech | I will pass you a config from 2.0.3 |
03:18.19 | Crashsys | blah... polycom needs a more sane config file layout.. |
03:18.33 | [TK]D-Fender | Crashsys : Works rgeat for me.... |
03:18.33 | Crashsys | First, I gotta figure out how to download 2.0.3 :D |
03:18.37 | BSDTech | it has 3 files now |
03:19.11 | BSDTech | sip.cfg mac.cfg and phonee####.cfg |
03:19.40 | Crashsys | My 1.6.7 has that... <macaddr>.cfg, sip.cfg, and phone.cfg |
03:19.52 | BSDTech | thats it |
03:20.20 | BSDTech | there is the mac-directory~xml |
03:20.26 | BSDTech | but its not really needed |
03:20.47 | *** join/#asterisk linlin (i=will@71.194.70.13) |
03:21.12 | BSDTech | I love mty 501 and soon to get a 601 |
03:21.26 | wunderkin | while we are on the subject, can someone with an ip430 try making an outbound call (not answered), and call into the phone and answer the incoming call .. tell me if your phone stops responding and reboots..? |
03:21.54 | TheCops | Someone is using Callpickup ? I have difficulties to figure out how it is working. I only want to intercept an incoming call, but if I read the description, it intercept outbound call too. |
03:22.12 | BSDTech | what boot rom ver |
03:22.32 | wunderkin | me? the latest |
03:22.46 | BSDTech | 3.2.2 and firmware 1.6.7 |
03:23.02 | wunderkin | TheCops, no.. i dont think so.. |
03:23.54 | BSDTech | wunderkin ? |
03:24.00 | Crashsys | Anyone got the Sip2.0.3 zip from polycom? |
03:24.00 | TheCops | wunderkin, So, If I add all SIP extensions in the group 1, I can use *8 to take an incomming call? |
03:24.18 | wunderkin | yeah |
03:24.23 | BSDTech | shhhhh where di you herar about 2.0.3 |
03:24.32 | BSDTech | its a secrete |
03:25.02 | wunderkin | im using 2.0.3 now but i'm just trying to figure out if it is just me with the problem, other than polycom god |
03:25.18 | BSDTech | I know on the 2 we have at the office 2.0.3 and 3.2.2 works fine |
03:25.28 | BSDTech | never went to 21.6.7 |
03:25.32 | wunderkin | i had this problem before 2.0.3 back to 1.6.x i think |
03:26.04 | Crashsys | wunderkin: I know that 1.6.7 will let me call the phone without rebooting... |
03:26.26 | wunderkin | yeah, that's a nice feature of a phone, but that wasn't my problem... lol.. |
03:26.46 | BSDTech | we have no issues with 3.2.2 bootrom and firmware 2.0.3 on the 430 we have |
03:27.12 | BSDTech | and I think I have 2.0.3 here some where |
03:27.25 | danp | i found it on google |
03:27.35 | danp | search for polycom 2.0.3 |
03:27.37 | BSDTech | ok |
03:27.39 | danp | that is all i will say |
03:28.12 | Crashsys | Can you downgrade a phone from 2.0 to 1.6? |
03:28.19 | BSDTech | nope |
03:28.23 | Crashsys | dman |
03:28.24 | wunderkin | yeah you can |
03:28.25 | Crashsys | err damn |
03:28.29 | BSDTech | not if you have boot rom 3.2.2 |
03:28.34 | wunderkin | i have |
03:28.46 | BSDTech | and for the most part down grading can break a phone |
03:29.32 | *** join/#asterisk j0 (n=dan@S01060016b6b541d2.va.shawcable.net) |
03:29.34 | wunderkin | i don't know about that, i think mine were already broken, but you aren't supposted to use 1.6 configs on 2.0 apparantly ;P |
03:29.55 | wunderkin | i believe it did say that somewhere in the docs |
03:30.27 | BSDTech | I just did a diff on 1.6 and 2.0 and there are a few minor change sin sip.cfg thats it |
03:30.37 | wunderkin | it would be nice to have a config validator and stuff, i really thought my problems were due to bad config... it just amazes me that this many phones are bad..... |
03:30.58 | TheCops | I did an upgrade 1.6.7 to 2.0.3 and I keep the config of my 1.6.7 |
03:31.03 | [TK]D-Fender | Crashsys : yes, you can downgrade SIP versions. |
03:31.04 | TheCops | working perfectly |
03:31.18 | [TK]D-Fender | Crashsys : its Bootrom's you're stuck with from 2.x to 3.x |
03:31.34 | Crashsys | I could care less about the bootrom... it boots... :) |
03:31.38 | Crashsys | it's 3.2.2 anyways |
03:31.41 | [TK]D-Fender | Crashsys : more on the 500&300 |
03:31.42 | BSDTech | yeah the only changes are in the sip.cfg and it updates when you untar the file |
03:32.04 | Crashsys | you mean overwrites? |
03:33.06 | BSDTech | yes it puts a new ver in its place |
03:35.13 | *** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox) |
03:41.10 | *** join/#asterisk Globetrotter (n=eric@205.211.239.11) |
03:41.39 | *** join/#asterisk BSDTech (n=RNeese@ppp-69-238-75-25.dsl.irvnca.pacbell.net) |
03:41.50 | Globetrotter | hi guys,, how do i ftp to asterisk bussiness edition ??? |
03:43.49 | wunderkin | you sure are trying... |
03:44.02 | *** join/#asterisk adrianqcs23 (n=Adrian@60.52.206.125) |
03:44.08 | BSDTech | you dont |
03:44.16 | adrianqcs23 | hi |
03:44.20 | BSDTech | you have to buy it and they ship you a cd |
03:44.49 | BSDTech | and the manual |
03:45.06 | Globetrotter | i already have it,, bnut i cannot get ftp access to it to work |
03:45.21 | russellb | how does asterisk have anything to do with ftp access? |
03:45.27 | adrianqcs23 | can someone help me with my call monitoring? i tend to monitor calls with the manager cli using telnet to record calls on the fly....but the sound file is empty i.e. 44bytes |
03:45.27 | Nugget | telnet is eeeeeeevil! |
03:45.37 | BSDTech | ssh |
03:45.42 | BSDTech | its your friend |
03:45.52 | danp | does the manager support ssh? |
03:45.59 | rene- | it doesnt |
03:46.01 | danp | huh. |
03:46.03 | BSDTech | what manager |
03:46.06 | russellb | it does support SSL in trunk :) |
03:46.12 | BSDTech | wtf you talking about |
03:46.14 | rene- | heh |
03:46.25 | rene- | BSDTech: you didnt seem you understood the issue |
03:46.27 | Globetrotter | because i am trying o get my monitored recordings of the server so that we can pull up calls at request |
03:46.52 | adrianqcs23 | yes |
03:47.07 | adrianqcs23 | i am using the monitor command |
03:47.26 | adrianqcs23 | i.e. Action: Monitor |
03:47.31 | BSDTech | dog needs walking |
03:47.33 | adrianqcs23 | in telnet |
03:47.40 | BSDTech | use ssh |
03:47.58 | adrianqcs23 | i can tel net |
03:48.01 | BSDTech | or get ari |
03:48.14 | BSDTech | then your box is insecure |
03:48.29 | BSDTech | and a network secrity risk |
03:48.31 | adrianqcs23 | i can execute orignate and monitor that channel |
03:48.51 | adrianqcs23 | i can save the files |
03:49.27 | *** join/#asterisk BitBandit (i=McDonald@68-187-56-58.dhcp.stgr.ut.charter.com) |
03:49.34 | adrianqcs23 | but when monitoring a channel bridged, it saves 0 bytes of sound files |
03:50.04 | adrianqcs23 | anybody an expert in asterisk? |
03:50.08 | rene- | adrianqcs23: that is weird |
03:50.09 | adrianqcs23 | hello |
03:50.33 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
03:50.33 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
03:50.40 | rene- | russellb: how do i download branch asterisk? |
03:51.08 | rene- | BSDTech: it is recommended that manager access is disallowed outside of local net |
03:51.26 | Globetrotter | hi guys,, me again.. i want to be able to share my monitor recordings to pull up on demand... any ideas? |
03:51.31 | rene- | BSDTech: the asterisk creators recommend to firewall those boxes |
03:51.43 | rene- | Globetrotter: i would publish them over http with apache |
03:51.49 | *** join/#asterisk sethwhit (n=SethWhit@70-56-234-164.clsp.qwest.net) |
03:51.52 | rene- | why bother with ftp |
03:52.14 | Globetrotter | becasue i dont know any better :) |
03:52.44 | rene- | well i would set an alias for recordings to point to the location of asterisk in apache |
03:52.47 | adrianqcs23 | Globelrotter: how do you monitor channels on the fly? |
03:53.13 | rene- | for specifics you might want to ask on #apache |
03:53.23 | adrianqcs23 | Globletrotter: Without modifying the extensions.conf |
03:53.38 | Globetrotter | sorry,, but i am not too familiar with astrerisk yet.. still learning |
03:53.43 | adrianqcs23 | I am trying telnet and accesing the manager |
03:54.09 | adrianqcs23 | but have inconsistency in recording them.... some appear to be 44 bytes??? |
03:54.38 | TheCops | What's the relation with the callgroup and pickupgroup? I want a specific group can take call from another group, I don't understand how to configure it. |
03:54.42 | rene- | adrianqcs23: i dont know if this would help but you can use auto recording by dialing some feature code that is activated in features.ocnf |
03:54.55 | rene- | TheCops: that is tricky and i dont remember |
03:54.57 | rene- | heh |
03:55.09 | TheCops | rene-, yup seem to be complicated |
03:55.33 | rene- | but configuration is not difficult |
03:55.38 | wunderkin | callgroup sets the group .. pickupgroup gets the groups... you set pickupgroup to the callgroups that you want to allow it to pickup |
03:55.56 | rene- | oh |
03:56.06 | rene- | that is clean |
03:56.25 | TheCops | ho |
03:56.29 | *** join/#asterisk bhrobinson (n=brobinso@northtx1-static.telwestonline.com) |
03:57.03 | TheCops | wunderkin, so, if I want ext 200 to pickup call for 201,I put callgroup=1 to 201, and pickupgroup=1 to 200 |
03:57.47 | wunderkin | yeah i guess |
03:57.59 | TheCops | Nice thank you |
03:58.47 | bhrobinson | is anyone in here from iowa? |
04:00.08 | adrianqcs23 | One-touch recording |
04:00.11 | adrianqcs23 | <PROTECTED> |
04:00.11 | adrianqcs23 | <PROTECTED> |
04:00.11 | adrianqcs23 | IN and OUT audio will be split into two files, and will be available on your asterisk server in the following directory: |
04:00.19 | adrianqcs23 | what does it mean by one-touch? |
04:00.37 | rene- | that you press some combination of keys to enable instant recording |
04:00.51 | wunderkin | keys != 1 |
04:01.01 | rene- | maybe it is not one touch since most key combinations are composed of multiple keys |
04:01.25 | rene- | but that is what is called |
04:01.35 | adrianqcs23 | arghhh! |
04:01.43 | rene- | i am trying to do svn checkout http://svn.digium.com/svn/asterisk/branch asterisk but it doesnt work |
04:01.47 | adrianqcs23 | i wan to record without the user to press anything |
04:02.12 | rene- | it should be possible but i have never done it |
04:02.26 | rene- | then why not enable recording of every call? |
04:02.31 | rene- | who decides when to record? |
04:02.43 | wunderkin | asterigod |
04:02.56 | rene- | wunder any clue on how to download branch? |
04:03.49 | rene- | it seems to be svn checkout http://svn.digium.com/svn/asterisk/branches asterisk |
04:04.03 | rene- | but that got me 1.2 |
04:04.06 | rene- | :p |
04:04.09 | adrianqcs23 | how do you enable recording of every calls? do i need to go to every line extension and run the monitor command? |
04:04.19 | adrianqcs23 | that would be tedious! |
04:04.39 | wunderkin | branch? |
04:04.47 | rene- | wunder yes |
04:04.48 | adrianqcs23 | by the way the extension is maintained by fonality... so i dun wan to mess up the extension.conf |
04:05.07 | [TK]D-Fender | adrianqcs23 : GUI's are not suported here, sorry |
04:05.09 | wunderkin | ... you mean trunk, but why? |
04:05.13 | rene- | i mean branch |
04:05.17 | wunderkin | guis are too pretty for [TK]D-Fender |
04:05.27 | rene- | i have branch installed by digium |
04:05.38 | rene- | but that has a broken manager interface |
04:05.38 | adrianqcs23 | i need some way to 'record the channels' without messying with the configuration files |
04:05.53 | rene- | adrianqcs23: it will be harder that way |
04:06.02 | *** join/#asterisk raina (n=raina@pdpc/supporter/active/ro3159) |
04:06.29 | rene- | i want a newer branch install |
04:06.31 | adrianqcs23 | <[TK]D-Fender> yeah i know that.... i am trying to recording calls |
04:06.44 | wunderkin | ~ GUIs are too sexy for [TK]D-Fender's shirt.. too sexy .. it hurts! ~ |
04:06.45 | jbot | okay, wunderkin |
04:06.51 | adrianqcs23 | <[TK]D-Fender> without accessing the gui |
04:06.53 | wunderkin | lol ! jbot agrees |
04:07.07 | adrianqcs23 | <[TK]D-Fender> preferably by agi or telnet |
04:07.23 | rene- | agi is messing with the dialplan |
04:08.09 | [TK]D-Fender | adrianqcs23 : in in tru fashion chances are any changes you do manually will get blown away then next time you apply any changes to the GUI info. |
04:08.26 | [TK]D-Fender | ~lart jbot |
04:08.29 | [TK]D-Fender | :O |
04:08.32 | wunderkin | :D |
04:08.37 | [TK]D-Fender | omg pwned |
04:08.43 | [TK]D-Fender | ~jbot |
04:08.44 | jbot | extra, extra, read all about it, jbot is only marginally useful at best, He got a C- on his Turing Test |
04:08.44 | wunderkin | pwned. |
04:08.57 | [TK]D-Fender | jbot : suck THAT biotch! |
04:08.58 | jbot | ACTION sucks THAT biotch!'s lips |
04:09.04 | [TK]D-Fender | :D |
04:10.23 | *** join/#asterisk CunningPike (n=CunningP@S010600095b33697f.vc.shawcable.net) |
04:11.12 | [TK]D-Fender | ~[TK]D-Fender |
04:11.13 | jbot | hmm... [tk]d-fender is rockin' the casbah !!! |
04:11.17 | [TK]D-Fender | huzzah! |
04:11.49 | wunderkin | lock the cat box |
04:12.04 | wunderkin | ~guis |
04:12.05 | jbot | guis are too sexy for [TK]D-Fender's shirt.. too sexy .. it hurts! ~ |
04:12.09 | wunderkin | sweet |
04:13.56 | rene- | he |
04:14.00 | rene- | h |
04:14.27 | *** join/#asterisk Splat (n=Splat@220-253-104-247.TAS.netspace.net.au) |
04:15.00 | rene- | well lads this has been largely futility but as always one keeps coming for the bot |
04:15.42 | rene- | jk |
04:15.54 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
04:16.25 | wunderkin | ~seen justinu |
04:16.28 | jbot | justinu <n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net> was last seen on IRC in channel #asterisk, 31d 9h 17s ago, saying: 'yeah... they were like half price... so it was $500 instead of $1000'. |
04:16.36 | wunderkin | where's that boy been |
04:17.40 | [TK]D-Fender | ~seen justinu_laptop |
04:17.42 | jbot | i haven't seen 'justinu_laptop', [TK]D-Fender |
04:17.53 | [TK]D-Fender | he has another ID IIRC |
04:18.02 | wunderkin | ~seen justinu[laptop] |
04:18.03 | jbot | i haven't seen 'justinu[laptop]', wunderkin |
04:18.09 | wunderkin | shrug |
04:18.35 | [TK]D-Fender | somehting like that |
04:19.01 | [TK]D-Fender | ~seen justinu|laptop |
04:19.03 | jbot | justinu|laptop <n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net> was last seen on IRC in channel #asterisk, 50d 1h 21m 7s ago, saying: 'bluregard: heh, same here'. |
04:19.10 | [TK]D-Fender | eek, even LONGER |
04:19.36 | wunderkin | yeah.. :~( |
04:21.41 | rene- | mmm |
04:21.44 | rene- | SVN-branch-1.4-r50468 still has a broken manager |
04:21.47 | rene- | why oh why |
04:27.22 | Corydon76-home | Did you file a bug? |
04:30.47 | CunningPike | rene-: Ah - is that why FOP won't worl? |
04:30.53 | CunningPike | s/worl/work/ |
04:33.53 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
04:34.18 | BSDTech | ? |
04:34.25 | BSDTech | I missed some one paging me |
04:34.48 | BSDTech | ok |
04:44.13 | *** part/#asterisk bhrobinson (n=brobinso@northtx1-static.telwestonline.com) |
04:44.15 | *** join/#asterisk apardo (n=apardo@87.217.145.152) |
04:50.33 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
04:50.33 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
04:53.09 | *** join/#asterisk sjobeck (n=sjobeck@208-151-246-203.dq1sn.easystreet.com) |
04:54.36 | wunderkin | CunningPike, don't you have polycom phones? |
04:54.48 | CunningPike | wunderkin: Indeed we do |
04:54.58 | wunderkin | have an ip430? available now..? |
04:55.13 | CunningPike | wunderkin: 'Fraid not - we only have 501s and 601s |
04:55.23 | wunderkin | hmmm k |
04:55.47 | wunderkin | wish i had the $ to spend on getting a 501 or 601 for home too.. |
04:58.37 | *** join/#asterisk apardo (n=apardo@87.217.145.152) |
05:00.47 | CunningPike | wunderkin: Work for someone who uses them ;) |
05:01.30 | wunderkin | yeah well thats the only reason i had to spring for an ip430 finally, i had 2 bt101s :P |
05:01.41 | CunningPike | Ugh |
05:04.52 | wunderkin | barbietone :D |
05:14.28 | *** join/#asterisk QuickOrbit (n=Christop@host-64-234-10-249.nctv.com) |
05:15.04 | Qwell | ~lart wunderkin |
05:15.29 | russellb | Qwell: i hate moving :( |
05:15.32 | Qwell | russellb: yeah... |
05:15.35 | russellb | Qwell: but QuickOrbit rocks and is helping me :D |
05:15.41 | Qwell | sweet |
05:15.44 | *** join/#asterisk Defraz (n=t0tal@24-116-159-197.cpe.cableone.net) |
05:15.49 | Qwell | somebody buy that man a beer :D |
05:16.00 | QuickOrbit | he is paying me well |
05:16.47 | *** join/#asterisk sjobeck (n=sjobeck@208-151-246-203.dq1sn.easystreet.com) |
05:16.48 | russellb | and by well, he means with nothing |
05:17.07 | QuickOrbit | just wait till you get your credit card statement |
05:17.12 | wunderkin | special services |
05:17.37 | russellb | wunderkin: sssh! |
05:18.12 | *** kick/#asterisk [QuickOrbit!n=russell@asterisk/developer-and-stable-maintainer/drumkilla] by russellb (russellb) |
05:18.47 | *** join/#asterisk QuickOrbit (n=Christop@host-64-234-10-249.nctv.com) |
05:18.51 | wunderkin | ;) |
05:18.52 | QuickOrbit | thanks bud |
05:19.00 | *** join/#asterisk bkruse_home (n=kruz@69.73.127.92) |
05:19.18 | russellb | bkruse_home is here, everyone stop talking about him |
05:19.19 | Strom_C | hi |
05:19.26 | Qwell | Strom_C: see link above :p |
05:19.36 | Strom_C | and there was that one time where bkruse told me about his---oh ok, whatever, russel |
05:20.05 | Strom_C | the sprint link? |
05:20.08 | Qwell | mmhmm |
05:20.16 | Strom_C | yeah, i glaced at it before i went out |
05:20.17 | *** join/#asterisk cbullock81 (n=cbullock@adsl-227-62-82.jan.bellsouth.net) |
05:20.19 | *** part/#asterisk bkruse_home (n=kruz@69.73.127.92) |
05:20.33 | *** join/#asterisk bkruse_home (n=kruz@69.73.127.92) |
05:20.36 | Strom_C | hehe |
05:20.39 | Qwell | Qwell: 2, Sprint: 0 |
05:20.41 | cbullock81 | Hello all |
05:20.42 | bkruse_home | ok ive given you time to stop :D |
05:20.43 | Strom_C | just cant stay away |
05:20.50 | Strom_C | oh durh, that's you :D |
05:20.54 | bkruse_home | Qwell i was reading that in the redbull |
05:21.00 | bkruse_home | hi-larious |
05:21.05 | bkruse_home | russellb: hello ^_^ |
05:21.36 | russellb | bkruse_home: greetings |
05:22.17 | cbullock81 | I have a question... a very noobish question... I am trying to understand the differences between lines and call appearances. I'm just having a hard time understanding everything related to that... anybody have any good resources I could read? |
05:22.41 | bkruse_home | cbullock81: asterisk.........the future of IP telephony. |
05:22.48 | bkruse_home | not sure if that will help....... |
05:22.49 | bkruse_home | ~thebook |
05:22.53 | jbot | thebook is probably a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
05:22.53 | cbullock81 | i read it |
05:22.54 | bkruse_home | :P |
05:22.58 | cbullock81 | i might need to hit it up again |
05:23.16 | cbullock81 | i setup my 1st polycom phone today |
05:23.22 | Strom_C | cbullock81: well, in a modern telephone system there really isn't any such thing as a "line" |
05:23.26 | [TK]D-Fender | cbullock81 : what model? |
05:23.31 | cbullock81 | IP650 |
05:23.55 | bkruse_home | russellb: hows it going? havent talked to you in forever! |
05:24.00 | cbullock81 | when i have 3 or 4 polycom phones setup... how can the other users tell if all the "lines" are in use |
05:24.10 | russellb | bkruse_home: moving ... |
05:24.11 | cbullock81 | i have 2 wildcard x100p cards in my server |
05:25.07 | [TK]D-Fender | cbullock81 : ok you have (w/o expansion) 6 line keys. each can be for a different registration (ID), and each can handle 1-8 calls depending on your setup |
05:25.34 | cbullock81 | so its correct that that line key shows my extension? |
05:25.52 | [TK]D-Fender | cbullock81 : in what way? |
05:26.15 | cbullock81 | well like my extension is "199", and thats what it says out to the right of the line button |
05:29.34 | cbullock81 | am i totally not making any sense? |
05:31.16 | [TK]D-Fender | Hey, I'm working on a 1.4 system and am getting this message that I can't understand the origin of : [Jan 12 04:27:34] WARNING[25488]: pbx.c:776 _extension_match_core: Wrong usage of [] in the extension |
05:31.37 | [TK]D-Fender | This is after a dial has executed, but not immediately after |
05:31.49 | [TK]D-Fender | 1.4 FTP release |
05:32.16 | *** join/#asterisk lowlevel (n=Stuart@CPE000e0c057fad-CM000f9f7d6742.cpe.net.cable.rogers.com) |
05:34.04 | russellb | [TK]D-Fender: what does the pattern look like? |
05:34.16 | *** join/#asterisk dorphalsig (n=dorphals@pcsp163-73.supercabletv.net.co) |
05:34.22 | dorphalsig | Hi? |
05:34.35 | Tene | Hi! |
05:34.49 | dorphalsig | Can anybody help me get an h323 box working with asterisk? |
05:34.58 | dorphalsig | I have already oh323 |
05:35.03 | dorphalsig | installed and into asterisk |
05:35.50 | dorphalsig | I just cant get the damned box to hit the server |
05:35.52 | *** join/#asterisk niZon (i=bleh@S0106beefd4cecc3d.wp.shawcable.net) |
05:39.00 | [TK]D-Fender | russellb : Ahh.. I think I missed that |
05:40.02 | russellb | [TK]D-Fender: oh, so the message was right? missed an ending ] ? |
05:40.51 | russellb | the extension matching code was rewritten for 1.4 |
05:40.55 | russellb | so just making usre it wasn't broken ... |
05:41.03 | dorphalsig | hello? |
05:41.41 | [TK]D-Fender | russellb : I had _[3,4,6,9]11 . No ","s needed. |
05:41.59 | russellb | ahh, gotcha |
05:42.06 | [TK]D-Fender | russellb : _[3469]11 should be correct, right? |
05:42.16 | russellb | yes |
05:42.28 | [TK]D-Fender | russellb : So what changed? |
05:43.31 | russellb | well, mostly for performance and code quality in general |
05:43.39 | russellb | but one thing was changed ... |
05:43.45 | bkruse_home | russellb: YAY! |
05:43.53 | bkruse_home | russellb: yay bluemen |
05:43.55 | russellb | previously, you could have two extensions in a context ... |
05:44.03 | russellb | 12345 and _1XXXX |
05:44.15 | russellb | which one do you think would get matched first? you would think 12345 since it is more exact ... |
05:44.41 | russellb | well, it never actually worked that way, and people had to force sort order by splitting things into contexts and including them in clever order |
05:44.49 | russellb | but now 12345 is actually matched first. |
05:45.01 | bkruse_home | woot |
05:45.30 | [TK]D-Fender | russellb : Ok, so basically 2 successful matches are scored against each other for "Wildcardness"? |
05:45.41 | Qwell | wildcardosity |
05:45.49 | [TK]D-Fender | Qwell : Duly noted |
05:45.56 | bkruse_home | Qwell: your going to court for killing jbot |
05:46.11 | [TK]D-Fender | Qwell : theres a ..... truthiness about that ;) |
05:46.17 | russellb | [TK]D-Fender: sort of |
05:46.39 | russellb | the way it works internally is that they're sorted based on ... wildcardnessosity |
05:46.42 | bkruse_home | jason you going to see the bluemen? |
05:46.47 | russellb | and then to match, it just traverses them until it finds a match |
05:47.24 | [TK]D-Fender | russellb : so a fixed length match is higher ranked than one with "." for instance? |
05:48.00 | russellb | yes, i think so :) |
05:48.09 | [TK]D-Fender | russellb and a char series map higher than "x" per char? :) |
05:48.22 | [TK]D-Fender | russellb : I could devise a sick compound scoring system for this :) |
05:48.50 | russellb | exact match should always be highest priority match |
05:49.00 | [TK]D-Fender | russellb : Very naturally. |
05:49.03 | russellb | and then it starts matching patterns ... based on pattern type, and pattern length |
05:49.52 | [TK]D-Fender | russellb : Better than before... I can definately live with this. to think we could have had this ages ago if we left out AEL ;) |
05:50.29 | russellb | heh |
05:50.33 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
05:50.33 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
05:56.10 | bkruse_home | russellb: when you gona be in meh city! |
05:56.21 | bkruse_home | so you can fix my linux + the asterisk kthx |
05:56.49 | russellb | saturday |
05:57.10 | bkruse_home | !!!!!!!!!!!! |
05:57.26 | bkruse_home | strait into a house, or apartment, or kp's? |
05:57.53 | sethwhit | how do I start Asterisk |
05:58.00 | sethwhit | from the command line |
05:58.08 | russellb | bkruse_home: kp's |
05:58.10 | sethwhit | I just need to get the server runnign |
05:58.41 | Strom_C | sethwhit: /usr/sbin/asterisk -cvvvvvvg to make sure it starts up cleanly, then "stop now" and then /usr/sbin/safe_asterisk |
05:59.04 | sethwhit | what if I want to start it at boot |
05:59.13 | Strom_C | depends on the OS |
05:59.18 | Strom_C | what are you running? |
05:59.18 | sethwhit | CentOS |
05:59.28 | Strom_C | "make config" in the asterisk source directory |
06:00.07 | russellb | make samples ? |
06:00.19 | Qwell | init scripts |
06:00.24 | russellb | oh |
06:00.30 | sethwhit | here is my problem |
06:00.36 | sethwhit | I have installed Asterisk on a VPS |
06:00.45 | sethwhit | and would like to use Freepbx to manage it |
06:00.50 | bkruse_home | russellb: im going to come over and bring pizza |
06:00.55 | bkruse_home | make samples is broke in trunk |
06:01.06 | russellb | bkruse_home: lol ... want to help me unload? :) |
06:01.08 | sethwhit | when I try to start wit the amportal script I get |
06:01.09 | sethwhit | STARTING ASTERISK |
06:01.09 | sethwhit | Cannot find your TTY (9) |
06:01.21 | Strom_C | sethwhit: don't waste your time with freepbx |
06:01.38 | sethwhit | ok |
06:01.47 | Strom_C | learn config files :) |
06:01.51 | sethwhit | lol |
06:01.52 | bkruse_home | russellb: ill come over and help you unpack, really |
06:01.52 | sethwhit | yay |
06:01.53 | sethwhit | .... |
06:01.57 | bkruse_home | Strom_C: or use the gui |
06:02.00 | russellb | bkruse_home: sweet! :) |
06:02.02 | bkruse_home | ;] |
06:02.11 | Strom_C | bkruse_home: i havent used the gui enough to know whether to recommend it yet |
06:02.14 | sethwhit | gui? |
06:03.04 | russellb | sethwhit: asterisknow.org |
06:03.29 | Strom_C | I should dick around with asterisknow again |
06:03.36 | bkruse_home | saturday? |
06:03.53 | russellb | bkruse_home: yeah, i'm leaving here sometime in the morning |
06:04.56 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
06:04.57 | bkruse_home | how far does KP live from dig? |
06:04.57 | bkruse_home | the dig* |
06:04.57 | bkruse_home | Strom_C: i will not comment. |
06:04.57 | bkruse_home | lol |
06:04.58 | bkruse_home | it will be really cool, and it looks awesome |
06:04.58 | bkruse_home | :P |
06:05.03 | bkruse_home | sethwhit: its cool. |
06:05.03 | bkruse_home | Strom_C: eh.....wait for it to be non-beta |
06:05.04 | bkruse_home | its kind of.......well just wait for the non-beta |
06:05.06 | bkruse_home | lol |
06:05.08 | bkruse_home | russellb: how far does kpflem live from the atrium? |
06:05.19 | Strom_C | bkruse_home: I cant wait |
06:05.44 | russellb | bkruse_home: um, a ways ... |
06:06.00 | *** join/#asterisk bkruse_home (n=kruz@69.73.127.92) |
06:06.07 | Qwell | 800km |
06:06.11 | Qwell | obviously |
06:06.14 | Strom_C | :) |
06:06.15 | *** join/#asterisk shidan (n=chatzill@CPE0013107d30c4-CM001371871af0.cpe.net.cable.rogers.com) |
06:06.19 | bkruse_home | ouch |
06:06.56 | dorphalsig | Hey, anybody has done anything with H323 here? |
06:07.10 | dorphalsig | I'm trying to get an Ericsson switch and asterisk 2 talk to each other |
06:07.13 | *** join/#asterisk conver2 (n=marc3234@206-248-129-60.dsl.teksavvy.com) |
06:07.14 | file | kp is 800km from teh office |
06:07.21 | file | in both distance and time |
06:07.21 | dorphalsig | but I dont understand a damned thing |
06:08.14 | bkruse_home | :X |
06:08.59 | conver2 | in using odbcstorage, is the voicemail recording initially stored in file before being sent to db? |
06:09.05 | Qwell | 1897110752113 10027110752113 |
06:09.07 | Qwell | weeeeird |
06:09.28 | Qwell | I dialed this number three times. twice, it read those numbers off, and the third...somebody answered |
06:10.07 | Qwell | quite bizarre |
06:14.30 | *** join/#asterisk x86 (n=x86@p3m/member/x86) |
06:15.15 | *** part/#asterisk QuickOrbit (n=Christop@host-64-234-10-249.nctv.com) |
06:17.53 | *** join/#asterisk obnauticus (i=asd@c-24-21-116-29.hsd1.mn.comcast.net) |
06:18.03 | *** join/#asterisk sjobeck (n=sjobeck@208-151-246-203.dq1sn.easystreet.com) |
06:18.23 | *** join/#asterisk [F] (n=f@pool-71-163-132-11.washdc.fios.verizon.net) |
06:18.28 | [F] | hey, has anyone here used: http://www.jajah.com/ |
06:18.43 | [F] | and can tell me if its legit or not? (right now I have asterisk compiling on my freebsd box) |
06:18.52 | Strom_C | dead hookers |
06:19.17 | Tene | [F]: a good thing to check is if google knows about anyone complaining about it. |
06:20.18 | [F] | hrm, unless I suck at googling, it seems to be something that works. |
06:20.42 | [F] | however, would jajah be completely useless (and/or redundant) if you have your own asterisk server? |
06:22.48 | bkruse_home | [F]: im checking it out, but yes, because they prolly use asterisk |
06:23.02 | [F] | bkruse_home: i just found the site out two seconds ago |
06:23.21 | bkruse_home | [F]: thats extremelly to do in asterisk |
06:23.24 | bkruse_home | extremely * |
06:23.55 | [F] | if its something thats worth having, just setup an asterisk server so you could use your cell to call your house, and issue a script to launch the website and enter your cell number and the destination's number (which you'd feed to asterisk) |
06:24.06 | [F] | bkruse_home: what is extreme(ly) done in asterisk? |
06:24.15 | bkruse_home | lol |
06:24.16 | bkruse_home | type |
06:24.25 | bkruse_home | s/extremelly/extremely easy/ |
06:24.37 | [F] | ahhh, what, the service offered by jajah? |
06:24.43 | bkruse_home | yes |
06:24.49 | [F] | oh cool. |
06:24.55 | [F] | i'm a complete asterisk noob (its not even done compiling on my box) |
06:25.11 | bkruse_home | gl! |
06:25.18 | [F] | thanks! |
06:27.08 | *** join/#asterisk Chris-NB (n=chris@ng1.kurtkrenn.com) |
06:29.28 | *** join/#asterisk Strom_C (n=strom@netblock-66-159-243-60.dslextreme.com) |
06:32.46 | *** join/#asterisk sjobeck (n=sjobeck@208-151-246-203.dq1sn.easystreet.com) |
06:40.39 | *** join/#asterisk sjobeck (n=sjobeck@208-151-246-203.dq1sn.easystreet.com) |
06:41.58 | *** join/#asterisk shinux__ (n=shinux@196.220.24.242) |
06:45.25 | cbullock81 | has anyone had any experience with SLA in Asterisk 1.4? |
06:46.40 | *** join/#asterisk Aces1Up (n=rich@ip68-96-224-23.lv.lv.cox.net) |
06:47.25 | Aces1Up | hell i have been looking around in google the past 20min. anyone know of a good guide on how to set up a remote sip extension in asterisk to allow a remote softphone to connect to my asterisk server? |
06:48.16 | Strom_C | ~thebook |
06:48.19 | jbot | [thebook] a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
06:49.44 | x86 | Aces1Up: you may also want to check out voip-info.org |
06:50.04 | x86 | Aces1Up: it's an invaluable reference to everything about asterisk (amoung many other things) |
06:50.33 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
06:50.34 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
06:51.24 | *** part/#asterisk cbullock81 (n=cbullock@adsl-227-62-82.jan.bellsouth.net) |
07:08.33 | *** join/#asterisk abatista (n=ariel_@dsl-20-177.cofs.net) |
07:10.10 | *** join/#asterisk mjko (n=korpim@ip-58-28-130-48.ubs-dsl.xnet.co.nz) |
07:11.20 | *** join/#asterisk [F] (n=f@pool-71-163-132-11.washdc.fios.verizon.net) [NETSPLIT VICTIM] |
07:11.20 | *** join/#asterisk obnauticus (i=asd@c-24-21-116-29.hsd1.mn.comcast.net) |
07:11.20 | *** join/#asterisk conver2 (n=marc3234@206-248-129-60.dsl.teksavvy.com) [NETSPLIT VICTIM] |
07:11.20 | *** join/#asterisk jeebusroxors (n=jeebusro@cpe-75-80-231-237.dc.res.rr.com) [NETSPLIT VICTIM] |
07:11.20 | *** join/#asterisk ariel_ (n=ariel_@dsl-20-177.cofs.net) [NETSPLIT VICTIM] |
07:11.20 | *** join/#asterisk fenlander (n=neils@82.152.81.57) [NETSPLIT VICTIM] |
07:11.20 | *** join/#asterisk kFuQ (n=somedude@c-67-185-123-34.hsd1.wa.comcast.net) [NETSPLIT VICTIM] |
07:11.20 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) [NETSPLIT VICTIM] |
07:11.21 | *** join/#asterisk Bazy (n=bazy@86.125.51.251) [NETSPLIT VICTIM] |
07:11.21 | *** join/#asterisk tessier_ (n=treed@kernel-panic/sex-machines) [NETSPLIT VICTIM] |
07:11.21 | *** join/#asterisk juice (n=juice@mo-76-0-43-77.dhcp.embarqhsd.net) [NETSPLIT VICTIM] |
07:11.21 | *** join/#asterisk ixx (i=foobar@cpe-70-122-50-222.austin.res.rr.com) |
07:11.21 | *** join/#asterisk svanlund (n=dasv@slave.rixport80.se) [NETSPLIT VICTIM] |
07:11.21 | *** join/#asterisk MooingLemur (n=troy@unaffiliated/mooinglemur) [NETSPLIT VICTIM] |
07:11.21 | *** join/#asterisk `Sauron (i=sauron@dsl001-130-033.aus1.dsl.speakeasy.net) [NETSPLIT VICTIM] |
07:11.21 | *** join/#asterisk eliXier (i=GTI16V@gti.twice-irc.de) [NETSPLIT VICTIM] |
07:11.21 | *** join/#asterisk Az_au (n=az@216.127.73.119) [NETSPLIT VICTIM] |
07:11.21 | *** join/#asterisk brian (i=brian@unaffiliated/brian) [NETSPLIT VICTIM] |
07:26.00 | *** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
07:29.42 | DrCron | can asterisk set up a channel just to a soundcard? |
07:29.56 | naftali5 | chan_oss.so |
07:30.30 | naftali5 | http://www.voip-info.org/wiki/view/Setting+up+paging+with+a+sound+card |
07:31.11 | DrCron | hmm, just thinking that it would be so nice to have a wider project to add an application to asterisk to do radio routing |
07:31.41 | DrCron | replace this IRLP code disaster |
07:32.43 | *** join/#asterisk xnon (n=xnon@200.82.223.85) |
07:35.00 | *** join/#asterisk zeeesh (i=aadilism@9-237-154-202.wol.net.pk) |
07:37.09 | *** join/#asterisk tessier_ (n=treed@kernel-panic/sex-machines) |
07:37.44 | *** join/#asterisk JT (n=jon@unaffiliated/jt) |
07:38.15 | DrCron | asterisk doesnt have any support for video at this time, correct? |
07:38.24 | perd | passthrough |
07:41.09 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
07:46.05 | naftali5 | yes |
07:46.34 | *** join/#asterisk Splat (n=Splat@220-253-104-247.TAS.netspace.net.au) |
07:46.49 | naftali5 | 1.4 has h264 passthrough |
07:46.55 | naftali5 | 1.2.* can use http://www.asterisk-backports.org/wiki/index.php/Passthrough-h264 |
07:47.18 | naftali5 | may need to be manually applied, though |
07:47.29 | *** join/#asterisk xnon_ (n=xnon@200.82.222.41) |
07:50.33 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
07:50.33 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
07:59.06 | *** part/#asterisk mjko (n=korpim@ip-58-28-130-48.ubs-dsl.xnet.co.nz) |
08:08.52 | *** join/#asterisk bkruse_home (n=kruz@69.73.127.92) |
08:12.38 | CunningPike | DrCron: http://www.zapatatelephony.org/app_rpt.html |
08:13.02 | CunningPike | DrCron: What's the 'disaster' with IRLP? |
08:13.13 | CunningPike | DrCron: VA7IRL, by the way |
08:13.49 | DrCron | KI6HFB |
08:14.06 | DrCron | well, propriatary protocol, software, and hardware |
08:14.10 | CunningPike | DrCron: Pleased to meet you :) |
08:14.11 | DrCron | closed net |
08:14.32 | DrCron | sort of the oposite of asterisk |
08:14.34 | CunningPike | DrCron: Ah, yes - a disaster in general then. I thought there was a particular crisis I hadn't heard about |
08:15.01 | CunningPike | DrCron: Check out the link to Dude's work above - interesting stuff |
08:15.04 | DrCron | i was just trying to avoid foul language |
08:15.25 | DrCron | app_rpt? yhea, I looked at that a bit, interesting |
08:16.52 | CunningPike | DrCron: The other thing to check out is BCWARN - http://www.bcwarn.net/ |
08:17.45 | CunningPike | DrCron: Link looks busted atm |
08:19.07 | DrCron | the radio interface card is just a bit expensive |
08:19.22 | DrCron | i dont see why its much better then a serial port and a sound card |
08:19.48 | CunningPike | DrCron: Not familiar with it - anyway, bedtime here |
08:19.53 | CunningPike | DrCron: Great to chat - 73 |
08:20.00 | DrCron | http://qrvc.com/radiocards.html |
08:20.02 | DrCron | 73 |
08:22.37 | *** join/#asterisk Defraz (n=t0tal@24-116-159-197.cpe.cableone.net) |
08:31.53 | *** join/#asterisk Dandre (n=testdan@was59-3-82-236-48-30.fbx.proxad.net) |
08:32.09 | Dandre | hello all, |
08:33.03 | Dandre | is there some documentation about the syntax used in http manager interface in asterisk 1.4? |
08:35.47 | *** join/#asterisk SheriF_SpacE (n=sherif@chatty.shellshark.net) |
08:35.54 | *** join/#asterisk inspired (n=mikael@85.221.7.59) |
08:37.10 | *** join/#asterisk FuriousGeorge (n=brian@ool-43536ea8.dyn.optonline.net) |
08:44.28 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
08:49.01 | *** join/#asterisk xnon (n=xnon@200.82.222.41) |
08:50.33 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
08:50.33 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
09:02.27 | *** join/#asterisk Dibbler_Lap (n=Dibbler@212.248.196.179) |
09:07.09 | *** join/#asterisk pif (n=ldm@zenon.apartia.fr) |
09:07.21 | x86 | new ShellShark site is live guys! |
09:07.26 | x86 | http://www.shellshark.net/ |
09:10.28 | pif | what's the diff between cdr-custom and cdr-csv ? |
09:10.33 | pif | they look the same |
09:11.06 | *** join/#asterisk Ahrimanes (n=ma@81.7.159.2) |
09:12.30 | *** join/#asterisk jm|work (n=jamie@dilbert.jamiem.com) |
09:18.29 | *** join/#asterisk qdk (n=qdk@213.150.62.32) |
09:19.14 | *** join/#asterisk Todaro (n=chatzill@dslb-084-056-246-228.pools.arcor-ip.net) |
09:24.36 | *** join/#asterisk Gled|Work (n=sguser@LPuteaux-151-42-17-115.w193-252.abo.wanadoo.fr) |
09:27.21 | *** join/#asterisk apardo (n=apardo@87.217.145.152) |
09:38.49 | xnon | friends |
09:38.53 | xnon | good morning |
09:38.56 | *** join/#asterisk Lokiji (n=Lokiji@ip-89-102-178-195.karneval.cz) |
09:38.59 | xnon | i got a problem! |
09:39.07 | xnon | im try asterisk -vvvvvvvvvvvr |
09:39.16 | xnon | but de console say: Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?) |
09:39.33 | xnon | anybody can help me |
09:39.41 | james_ | try it without -r |
09:39.48 | james_ | that tries to connect to existing process |
09:39.53 | Mavvie | xnon: well, -r means: connect to a running asterisk server. |
09:39.54 | james_ | which you obviously dont have running |
09:40.14 | xnon | ok |
09:40.34 | xnon | ok i did it, but say the same |
09:40.36 | xnon | Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?) |
09:41.00 | xnon | my asteriisk are not running i think |
09:41.27 | xnon | :S the asterisk service not run i see it |
09:41.34 | xnon | what can i do? |
09:41.47 | xnon | my asterisk service cant init |
09:43.19 | *** join/#asterisk RoyK (n=roy@217-175-39.100710.adsl.tele2.no) |
09:43.27 | xnon | ummm its strange, because in /var/run/asterisk the file asterisk.ctl its ok! |
09:44.02 | *** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust826.leed.cable.ntl.com) |
09:45.09 | naftali5 | tail /var/log/asterisk/full |
09:47.51 | xnon | ok |
09:50.33 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
09:50.33 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
09:51.11 | xnon | ok the last line say Loading module chan_zap.so failed! |
09:53.09 | Lokiji | hello, which codec is the best when i have some lost packet? |
09:55.55 | naftali5 | xnon, your zapata.conf is probably not set properly or zaptel.conf |
10:02.24 | *** join/#asterisk speedwagon (n=ariel_@dsl-20-177.cofs.net) |
10:08.05 | *** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin) |
10:09.11 | *** join/#asterisk Teeli (n=tili@87.219.93.81) |
10:12.15 | *** join/#asterisk ariel_ (n=ariel_@dsl-20-177.cofs.net) |
10:12.51 | *** join/#asterisk stimpie (n=michiel@ip565faf27.direct-adsl.nl) |
10:13.57 | stimpie | could somebody tell my how to get the context from which a dialplan application was called? |
10:15.02 | *** join/#asterisk RoyK (n=roy@213.160.242.90) |
10:18.55 | naftali5 | http://www.voip-info.org/wiki-Asterisk+variables |
10:23.27 | RoyK | morning |
10:25.39 | stimpie | naftali5, so: context = pbx_builtin_getvar_helper(chan, "CONTEXT"); should get the current context? |
10:25.50 | *** join/#asterisk tparcina (n=tparcina@ccme.lama.hr) |
10:26.33 | tparcina | asterisk 1.4, how do I define which processor I use? In asterisk 1.2 I could edit Makefile and specify i586 processor |
10:31.10 | tparcina | anybody? |
10:31.19 | Ahrimanes | tparcina, I would think autoconf figures it out |
10:32.34 | e-ddie | I wouldnt think |
10:32.53 | Ahrimanes | e-ddie, I know.. you generally dont ;) |
10:33.01 | e-ddie | Ahrimanes: exactly |
10:33.45 | *** join/#asterisk dlynes_laptop (n=dlynes@S0106001346f7843f.vc.shawcable.net) |
10:35.26 | tparcina | Ahrimanes: it does figure it out - but it figures wrong :-( |
10:36.08 | tparcina | e-ddie: do you know how to define? |
10:36.22 | *** join/#asterisk docelm0 (n=vircuser@c-68-32-143-73.hsd1.de.comcast.net) |
10:37.18 | e-ddie | google usually works for me |
10:37.33 | Ahrimanes | yup |
10:38.29 | tparcina | i'm googling, but I can't find anything... |
10:39.18 | Ahrimanes | http://hobbes.bsd-dk.dk/~aron/funny_problem.png <- in this call, the caller doesnt get ringing, if i call out from a sip phone on the 1.2.7.1 customer server, i get ringing.. anyone seen this before? |
10:43.21 | *** join/#asterisk mitcheloc (n=mitchelo@titaniumsoft.net) |
10:45.38 | tparcina | Ahrimanes: do you have indications.conf? |
10:46.22 | Ahrimanes | tparcina, hm yes, why? |
10:47.44 | tparcina | Ahrimanes: just checking, because that could be the reason... |
10:47.54 | *** join/#asterisk reber (i=reber@gateway/tor/x-23573bb93263eb32) |
10:48.08 | Ahrimanes | tparcina, ok, well calls from sip phones to pstn and from pstn to sip phones work fine |
10:48.33 | Ahrimanes | but seems when the 1.0.9 pstn gateway * is involved with both ends of the call, it screws up |
10:48.54 | tparcina | channel, where to define processor on which I'm installing asterisk? on asterisk 1.2 i could modify Makefile and specify i586. what should I do at Asterisk 1.4? |
10:49.50 | tparcina | Ahrimanes: sorry, didn't have that kind of problem |
10:50.14 | Ahrimanes | ok |
10:50.33 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
10:50.33 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
10:51.06 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
10:58.37 | *** join/#asterisk Newbie___ (n=me@60.49.23.186) |
11:00.01 | *** join/#asterisk zapata (n=herbert@norway.ath.cx) |
11:02.21 | *** join/#asterisk svenna_ (n=svenna@p548D323B.dip0.t-ipconnect.de) |
11:04.33 | *** join/#asterisk emiquelito (n=evandro@200-155-185-1.static.spo.ifx.net.br) |
11:12.21 | Ahrimanes | anyone here got good quality connection to somalia? |
11:12.45 | e-ddie | where i come from we have a place we call 'soma lia' |
11:12.54 | e-ddie | is that good enough= |
11:12.55 | e-ddie | ? |
11:12.59 | Ahrimanes | hehe sure |
11:13.07 | e-ddie | great |
11:17.05 | x86 | guys |
11:17.14 | x86 | tell me what you think of my new site design |
11:17.20 | x86 | http://www.shellshark.net/ |
11:21.09 | Ahrimanes | x86, pretty nice |
11:21.25 | Ahrimanes | x86, but get a real ssl certificate? |
11:21.36 | x86 | lol, it's on the way :) |
11:22.16 | Ahrimanes | :) |
11:22.33 | x86 | layout and design looks nice though? |
11:23.02 | Ahrimanes | yep |
11:23.43 | x86 | you should buy service from us ;) |
11:28.28 | e-ddie | hrm |
11:29.03 | e-ddie | how come you have 'free us or canada phone number', and you dont get it with personal-sharkout? |
11:29.09 | e-ddie | doesnt sound free to me :) |
11:30.08 | *** join/#asterisk redax (n=redax@r6.hu) |
11:30.09 | redax | hi |
11:30.32 | redax | TRANSFER is not logged to queue_log, |
11:30.43 | redax | what do I messed here? |
11:35.49 | *** join/#asterisk Ebola (n=Ebola@host81-152-204-157.range81-152.btcentralplus.com) |
11:36.15 | *** join/#asterisk flenders (n=fserto@unaffiliated/flenders) |
11:36.44 | *** join/#asterisk zapp-branigan (n=zapp-bra@81-202-140-56.user.ono.com) |
11:36.49 | flenders | does anyone have settings for linksys/sipura phones? |
11:40.01 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
11:42.23 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
11:42.40 | *** join/#asterisk olivier__ (n=olivier@obs92-4-82-239-116-113.fbx.proxad.net) |
11:45.29 | *** join/#asterisk coppice (n=chatzill@129.168.17.210.dyn.pacific.net.hk) |
11:49.22 | redax | flenders: what settings? |
11:49.32 | redax | I've using SPA-942, and SPA-922 |
11:49.41 | redax | both working well. + PAP2 |
11:50.33 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
11:50.33 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
11:51.28 | *** join/#asterisk creativx (n=creadure@226.62-97-205.bkkb.no) |
11:54.27 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
11:56.07 | *** join/#asterisk myiagy (n=myiagy@200.175.61.250.static.gvt.net.br) |
12:09.25 | x86 | e-ddie: it's free with certain plans :) |
12:09.51 | x86 | e-ddie: SharkOUT is for outbound calling only, it can not recieve phone calls |
12:13.12 | *** join/#asterisk flenders (n=fserto@unaffiliated/flenders) |
12:13.19 | flenders | damn laptop |
12:13.49 | flenders | batteries could last 10 hours |
12:14.41 | Ahrimanes | hehe |
12:15.41 | flenders | if anyone replied to my previous question, could you please repost it? |
12:16.06 | Ahrimanes | flenders, i have some configs for SPA1001 and PAP2 |
12:20.31 | *** join/#asterisk voicetech (n=lotusscr@marnock.com) |
12:23.53 | voicetech | Can somebody please help with a caller id problem? When there is a polarity change I don't get the caller id. So in effect I only get every other caller id. This is driving me insane! I didn't have any problems with the x100p card. But I do with the Digium 2 FXO Port TDM card. |
12:28.41 | dlynes_laptop | voicetech: make sure your polarity isn't reversed |
12:28.49 | dlynes_laptop | voicetech: after you've checked that, use fxotune to adjust the gains |
12:30.40 | *** join/#asterisk eGov (n=sreynold@c-68-47-234-136.hsd1.tn.comcast.net) |
12:31.01 | voicetech | I don't know if I can stop the polarity from being reversed. Our telco can change it at any time. at the moment its every other call |
12:34.17 | voicetech | Can you tell me how I make sure polarity isnt reversed? |
12:45.38 | voicetech | can anyone tell me what this means and how to fix it please : chan_zap.c: Got event 17 (Polarity Reversal)... |
12:45.44 | *** join/#asterisk reber (i=reber@gateway/tor/x-50a9bd97e5ed209e) |
12:45.48 | *** join/#asterisk flenders (n=fserto@unaffiliated/flenders) |
12:46.02 | voicetech | and this: chan_zap.c: CID timed out waiting for ring. Exiting simple switch |
12:46.35 | tparcina | how to define to Asterisk 1.4 that I have i586 processor? On Asterisk 1.2 I could edit Makefile. |
12:50.33 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
12:50.33 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
12:52.29 | *** join/#asterisk delphiuk (n=richard@212.42.164.14) |
12:52.50 | delphiuk | <PROTECTED> |
12:53.07 | delphiuk | 1.4 before the GUI thinks you have ananlouge card installed? I have a x101 clone |
12:53.27 | delphiuk | sorry for the disjointed lines |
12:56.10 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
12:56.58 | *** join/#asterisk Ebola (n=Ebola@host81-152-204-157.range81-152.btcentralplus.com) |
13:05.45 | voicetech | anyone help with uk callerid problem? |
13:14.11 | *** join/#asterisk AstaWerksDotCom (n=doug@63.161.96.170) |
13:15.46 | *** join/#asterisk flenders (n=fserto@unaffiliated/flenders) |
13:19.30 | *** join/#asterisk flenders (n=fserto@unaffiliated/flenders) |
13:21.54 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
13:34.56 | *** join/#asterisk inspired (n=mikael@84.208.136.152) |
13:41.20 | *** join/#asterisk shinux__ (n=shinux@196.220.24.242) |
13:42.32 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
13:43.10 | *** join/#asterisk af_ (n=af@ip-170-44.sn1.eutelia.it) |
13:46.01 | *** join/#asterisk apardo (n=apardo@87.217.145.152) |
13:47.58 | *** join/#asterisk dhill (i=dhill@fog.mindcry.org) |
13:48.09 | dhill | does qualify= cause a lot of load with asterisk? |
13:49.11 | *** join/#asterisk IPmonger (n=ipmonger@c-68-84-208-206.hsd1.pa.comcast.net) |
13:49.59 | *** join/#asterisk phatmonkey (i=nobody@81.2.121.150) |
13:50.13 | *** join/#asterisk oQPa (n=roque@15.Red-83-40-197.dynamicIP.rima-tde.net) |
13:50.33 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
13:50.33 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
13:51.18 | phatmonkey | 1.2.13: this is a SIP call |
13:51.38 | phatmonkey | whoops |
13:53.26 | penguinFunk | voicetech: what problem? |
13:53.47 | phatmonkey | 1.2.13: this is an inbound SIP call, I cannot send audio from the phone to the server. the RTP traffic is getting through, i have checked with tcpdump, but even simple things like echo and voicemail do not receive any audio from the phone. voicemail complains of "Jan 12 13:53:26 WARNING[4932]: app.c:644 ast_play_and_record_full: No audio available on SIP/8201-0819c050??" |
13:54.10 | penguinFunk | i have yet been able to get callerid working for calls orginating on pstn coming into asterisk out to sip users phones |
13:54.27 | penguinFunk | < also in uk |
13:57.00 | *** join/#asterisk Ebola (n=Ebola@host81-152-204-157.range81-152.btcentralplus.com) |
14:03.49 | *** join/#asterisk vt3 (n=vt3@m016020.ppp.asahi-net.or.jp) |
14:04.33 | *** join/#asterisk potsboy (n=chrisg@196.211.16.202) |
14:07.01 | *** join/#asterisk lorinc (n=ang@caracas-1902.adsl.interware.hu) |
14:09.48 | *** join/#asterisk olivier__ (n=olivier@obs92-4-82-239-116-113.fbx.proxad.net) |
14:13.35 | Katty | morning. |
14:14.05 | IPmonger | good morning |
14:14.08 | phatmonkey | i've tried with two different clients now, a hard and softphone, with no success |
14:14.14 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
14:14.19 | *** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
14:15.32 | Katty | that's not peachy. |
14:16.23 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
14:17.17 | *** join/#asterisk jonta_ (n=jonta@c-f86ae155.238-1-64736c20.cust.bredbandsbolaget.se) |
14:17.55 | *** join/#asterisk luisjose (n=laptop@unaffiliated/luisjose) |
14:18.22 | phatmonkey | sigh, my mistake, the traffic being received was from the other phone that i was using to test... no traffic gets through with the echo test. back to checking the NAT settings.... |
14:19.28 | *** join/#asterisk arcadies (i=arcadies@dsl-243-0-122.telkomadsl.co.za) |
14:19.30 | vt3 | compared with commercial solutions, how well does asterisk perform conference calls? can it handle say a 100 users well? |
14:19.58 | arcadies | is it possable to add some sort of "gate opening" thing to the pbx? |
14:20.20 | arcadies | you know, to open gates electronicaly |
14:22.10 | luisjose | Who has set up 2 asterisk server in the same subnet behind a nat? |
14:22.17 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
14:23.51 | *** join/#asterisk b11d|bbl (n=no@234-200-29-134.hcc.mnscu.edu) |
14:24.04 | b11d|bbl | morning all |
14:24.47 | [TK]D-Fender | Katty: Mew. |
14:25.39 | *** join/#asterisk h0 (n=fakhir@unaffiliated/fakhir) |
14:26.01 | [TK]D-Fender | phatmonkey: Where are your phone and * relative to each other |
14:27.52 | xnon | dlynes_laptop, are you there friend? |
14:28.07 | xnon | dr0ne, are you there ! |
14:31.43 | docelmo | MEW! |
14:32.17 | phearless | hello ! |
14:32.22 | arcadies | is it possable to add some sort of "gate opening" thing to the pbx? |
14:32.25 | arcadies | you know, to open gates electronicaly? |
14:32.28 | *** join/#asterisk wunderkin (i=kev@ip72-208-3-221.ph.ph.cox.net) |
14:32.29 | phearless | I got a question for the experts ! |
14:32.56 | phearless | how can I send TWO commands, one after one, and get the output of each, to the Asterisk Manager ? |
14:33.06 | phearless | it is for a PHP script |
14:34.24 | SheriF_SpacE | evening guys |
14:34.33 | b11d | arcadies.. no doubt it is |
14:34.40 | b11d | if you wanted to do it via a serial port or something |
14:34.56 | b11d | you'd need to write a gate controller app, and then just call it from an AGI script, or even system() within extensions.conf |
14:35.15 | olivier__ | phearless> -->http://voip-info.org/wiki/index.php?page=Asterisk+manager+Example%3a+PHP |
14:35.31 | arcadies | thx b11d ill try that |
14:35.35 | b11d | good luck |
14:35.47 | arcadies | ill let you know |
14:35.47 | arcadies | :D |
14:35.49 | b11d | cool :) |
14:35.52 | phearless | olivier__: I know this page |
14:35.56 | *** join/#asterisk hohum (n=dcorbe@mercury.sunrocket.com) |
14:35.59 | phearless | they send only one command |
14:36.11 | b11d | just add a second then |
14:36.54 | *** join/#asterisk ctooley (n=ctooley@rrcs-71-42-115-242.sw.biz.rr.com) |
14:37.08 | ctooley | Is there a "!=" not equal in AEL2? |
14:37.12 | olivier__ | use fputs and fget that'it |
14:37.36 | Dandre | is there some documentation about the syntax used in http manager interface in asterisk 1.4? |
14:37.45 | ctooley | Because I'm getting "Jan 12 08:17:24 WARNING[20453]: ast_expr2.fl:321 ast_yyerror: ast_yyerror(): syntax error: syntax error, unexpected '!=', expecting $end; Input: != 0" |
14:37.59 | phearless | b11d: it is weird |
14:38.03 | phearless | I have to send : Action: Logoff |
14:38.06 | phearless | to get the output |
14:38.09 | b11d | hrm |
14:38.13 | phearless | and when I logoff, I can not send another command |
14:38.20 | b11d | oh yeah I suppose not then |
14:41.39 | wunderkin | you can send multiple... \r\n\r\n, then the next... if you are doing an originate you need async true |
14:42.13 | *** join/#asterisk reber (i=reber@gateway/tor/x-a6716cbe69f64a14) |
14:48.20 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
14:48.20 | *** mode/#asterisk [+o anthm] by ChanServ |
14:48.20 | *** join/#asterisk imesper (n=chatzill@c9066412.static.spo.virtua.com.br) |
14:49.27 | imesper | I am trying to connect to ami but it always returns me error Missing action in request, I am using asterisk 1.4, in asterisk 1.2 it works fine |
14:50.33 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
14:50.33 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
14:50.48 | imesper | Can someone give me a help? |
14:50.58 | *** join/#asterisk mtgh (n=chatzill@dsl093-001-038.det1.dsl.speakeasy.net) |
14:51.31 | voicetech | ne1 have callerid working in uk with tdm card? care to share your config? |
14:51.54 | *** join/#asterisk naitram (n=danny@216.77.58.40) |
14:52.53 | naitram | need dial command that lets me continue dial plan execution if either the caller or callee hang up |
14:53.22 | naitram | g option only works if called extension hangs up |
14:53.57 | [TK]D-Fender | naitram: use the "h" standard extension. |
14:54.46 | *** join/#asterisk inv_Arp (n=junya@c-71-206-88-100.hsd1.fl.comcast.net) |
14:54.58 | *** join/#asterisk HarryR (n=Harry@host-83-146-53-46.bulldogdsl.com) |
14:55.33 | naitram | [TK]D-Fender: if i need specific execution for each extension, how would that work? |
14:55.39 | *** join/#asterisk pif (n=ldm@zenon.apartia.fr) |
14:55.56 | [TK]D-Fender | naitram: ..HUH? |
14:56.20 | *** join/#asterisk mercestes (n=merceste@rrcs-71-41-157-70.sw.biz.rr.com) |
14:56.22 | pif | hi, what is the best ATA to connect to a doorphone (it needs to relay DTMF real good) ? |
14:56.57 | pif | I'm tryin with a sipura 3000 without success, won't relay dtmf tones |
14:57.03 | [TK]D-Fender | pif: For high quality, go with MediaTrix |
14:57.16 | pif | oki, got a url? |
14:57.41 | naitram | [TK]D-Fender: here is what i need to do, i need to execute a php script before the connection is made and after the connection is made, each extension has to provide specific information in both scripts based on its extension, can i do this with the h extension? |
14:57.49 | pif | found |
14:57.53 | ManxPower | Sometimes I get tired of people blaming Asterisk for echo. |
14:58.11 | [TK]D-Fender | naitram: Who are "each extension"? |
14:58.28 | mercestes | ManxPower: Especially when they are on a data T1 using direct sip connections to you or another provider. |
14:58.42 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
14:58.55 | [TK]D-Fender | ManxPower: I sometimes get tired of people in #asterisk echoing ..... Thus there is balance ;) |
14:59.19 | phatmonkey | [TK]D-Fender, phone is behind NAT, server has public IP. it's quite odd though - I copied the configuration from a working system in an almost identical situation. i'm guessing it's something up with the firewall configuration on the server or something like that |
15:00.00 | [TK]D-Fender | phatmonkey: Does your phones sip entry have "nat=yes"? |
15:00.08 | *** join/#asterisk alexandrepos (n=alexandr@201.21.143.130) |
15:00.15 | [TK]D-Fender | phatmonkey: And should also be "canreinvite=no" |
15:00.22 | alexandrepos | anybody can help-me ? |
15:00.32 | phatmonkey | yep, and qualify |
15:00.34 | [TK]D-Fender | alexandrepos: If you actually just ask a question, MAYBE. |
15:00.53 | phatmonkey | [TK]D-Fender, canreinvite=no is in general |
15:00.54 | [TK]D-Fender | phatmonkey: Ok, well that covers what you should need from what you described. Start checking firewalls.... |
15:01.40 | alexandrepos | [TK]D-Fender: i have two peers in sip, but only second peer receive a incomming call ! i i invert order another work |
15:01.59 | alexandrepos | [TK]D-Fender: u have ideia ? |
15:02.52 | [TK]D-Fender | alexandrepos: No enough information yet. pastebin your SIP configs for the phones, and the CLI output of a failed call. |
15:02.54 | [TK]D-Fender | ~pb |
15:02.57 | jbot | somebody said pb was a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
15:03.02 | ManxPower | Customer: Sometimes e-mail to me bounces. Me: Have the sender fax me a copy of the bounce message. Me: (I get the fax) Tell the sender to spell your damn domain correctly. |
15:03.34 | ManxPower | I am SO glad I'm not a first level support tech anymore. |
15:03.44 | [TK]D-Fender | ManxPower: or remove spaces (hard to tell sometimes), or... or..... |
15:04.18 | [TK]D-Fender | ManxPower: "Very few things in the Universe are unlimited. Amongst them is human stupidity..." |
15:04.35 | [TK]D-Fender | Katty: Hugz |
15:04.47 | ManxPower | [TK]D-Fender: I suspect this was a 1-time issue. The REAL issue might be still undiscovered. |
15:04.58 | naitram | [TK]D-Fender: each extension is the wrong semantics, I will be more specific, system(phpscriptSetBit 1 1) then Dial(sip/danslaptop)... connection is made, now regardless of who hung up system(phpscriptSetBit 1 0), now the arguments to phpscriptSetBit change based on who is being dialed. This works now, fine, if the called hangs up. But I cant get it to work if the caller hangs up. |
15:06.48 | ManxPower | naitram: see the "g" option to dial. |
15:07.10 | *** part/#asterisk delphiuk (n=richard@212.42.164.14) |
15:07.29 | ManxPower | Asterisk does different things depending on who hangs up. exten 'h' is for one side hangs up (I don't remember which one) and g option to dial is for when the other side hangs up |
15:07.30 | naitram | ManxPower: the g option only continues if the called party hangs up, not if the caller hangs up. Right? |
15:07.40 | pif | [TK]D-Fender : you using Mediatrix ata's yourself? |
15:08.03 | ManxPower | naitram: you need both "g" and exten "h" if you want to catch both types of hangups. Yes, I think it is stupid too. |
15:08.30 | coppice | Mediatrix - where bugs are a way of life :-) |
15:08.54 | naitram | ManxPower: if i use the h extension, how do i execute specific behavior for the extension that hung up? |
15:10.51 | pif | coppice : oops, I almost clicked "order" |
15:10.55 | phatmonkey | ah, it probably isn't a firewall issue. RTP traffic can reach the server if another server with a public IP sends it, so it's probably some SIP addressing problem |
15:11.21 | pif | coppice : what are your favorite ATA's (it's for a doorphone) |
15:11.24 | ManxPower | naitram: you need to run your script BOTH in the "h" extension AND in the priority after the dial to catch both instances |
15:12.24 | coppice | pif: I don't have a favourite, but I have had to work around a number of stupid bug in Mediatrix ATAs. They only seem to work with other boxes because the other boxes have been made to tolerate their crap |
15:12.57 | pif | let's say the least problematic |
15:13.05 | pif | to your knowledge |
15:13.26 | *** join/#asterisk DrkShdw (n=scorpio@unaffiliated/drkshdw) |
15:14.06 | naitram | ManxPower: yeah I got that, but when the h extension determines someone has hung up, is there information available that tells me who has hung up so that I may place the right value in my call to my cleanup script, like System(phpscript var1) where var1 is the extension that hung up |
15:14.17 | phatmonkey | do i need to configure the clients with STUN servers or something? |
15:14.24 | phatmonkey | not really sure what they do |
15:15.51 | [TK]D-Fender | pif: Used a larger gateway once, and a small ATA once. |
15:16.04 | [TK]D-Fender | pif: Somewhat complex, very powerful. |
15:16.17 | ManxPower | naitram: you don't understand. "h" will only be run if the caller hangs up, not of the callee hangs up |
15:16.40 | ManxPower | so you know that if your agi is run out of exten "h" you KNOW the caller hanged up |
15:16.54 | ManxPower | phatmonkey: I have never ever seen a real need for STUN with Asterisk. |
15:17.21 | *** join/#asterisk heh_v_water (n=heh_v_wa@71-32-211-123.hlna.qwest.net) |
15:17.24 | phatmonkey | ManxPower, yeah, i have the same setup on another server and I've had no problems |
15:17.32 | phatmonkey | i've no idea why this is causing so much hassle |
15:17.45 | [TK]D-Fender | ManxPower: Almost ass-backwards isn't it? having to split your dialplan based on who killed the call.... figure it might eb better to set an env variable like we do for dialstatus + "g" |
15:18.11 | ManxPower | [TK]D-Fender: Yes, it is stupid. |
15:18.37 | ManxPower | naitram: why do you need to know who hung up first? |
15:18.42 | ManxPower | Eventually both legs of the call will hang up |
15:21.32 | naitram | ManxPower: I dont really care who hung up, i just need to run the scripts regardless of who hung up. But I need the information specific for the called extension |
15:21.55 | naitram | I clearly dont understand how the dial plan is executed |
15:22.21 | naitram | what is the pastbin used here |
15:22.45 | *** join/#asterisk adorah (n=Michael@87.69.57.228.cable.012.net.il) |
15:23.10 | naitram | ill paste my extensions.conf and maybe you guys could tell me where I should put the h extension |
15:24.01 | *** join/#asterisk adorah (n=Michael@87.69.57.228.cable.012.net.il) |
15:24.13 | *** join/#asterisk tsurko (n=tsurko@vpn-pppoe-213-240-242-237.megalan.bg) |
15:24.26 | ManxPower | naitram: http://pastebin.ca/314207 |
15:27.11 | naitram | ManxPower: here is what i have http://pastebin.ca/314211 |
15:28.03 | ManxPower | naitram: my VERY SIMPLE example should help you |
15:28.29 | naitram | Ill read over it |
15:29.52 | *** join/#asterisk pif (n=ldm@zenon.apartia.fr) |
15:32.22 | *** join/#asterisk apardo (n=apardo@87.217.145.146) |
15:32.52 | Katty | can linux mount a tape as if it were mass storage? |
15:34.04 | ManxPower | Katty: yes. It cannot mount it as a random access device |
15:34.29 | ManxPower | "mass storage" is a generic term for ANY large amount of storage. |
15:34.47 | poller | Where "large" can be anything |
15:35.06 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
15:37.32 | sweeper | isn't mass storage the generic windows driver/moniker? |
15:38.46 | [TK]D-Fender | ManxPower: When you say "not random access" do you mean it can't SEEK live, or only that you can't WRITE to it (which is what I assumed already)> |
15:39.13 | penguinFunk | tape is a FIFO device |
15:39.23 | penguinFunk | read it sequentially |
15:39.25 | monsted | sweeper: mass storage can be pretty much anything random access |
15:39.27 | penguinFunk | not random access |
15:40.03 | ManxPower | [TK]D-Fender: I mean that if you want to read the 100'th record on the tape you need to rewind to the beginning, then seek over the first 99 records |
15:40.31 | ManxPower | also, you want to INSERT anything yo have to rewrite everything after the inserted data |
15:41.08 | penguinFunk | cant you just append to the end? |
15:41.16 | [TK]D-Fender | ManxPower: So just no tracking for "relative" movement then? So if you have read up to 200, and need to go to 100, you have to back 200, then forward 100? |
15:41.20 | ManxPower | Tapes are horribly unreliable and I think they were invented by someone that hates people |
15:41.39 | monsted | ManxPower: don't confuse shitty tape products with the real thing |
15:41.41 | ManxPower | penguinFunk: you can append to the end, just not insert into the middle |
15:41.51 | penguinFunk | lol ManxPower |
15:41.56 | penguinFunk | yeh i thought so |
15:41.59 | ManxPower | monsted: mainframe tapes were pretty reliabele |
15:42.13 | monsted | ManxPower: anything that doesn't suck is very reliable :) |
15:42.33 | ManxPower | I think think the Prime mini tapes (reel to reel) ever lost any of my data |
15:42.39 | monsted | ManxPower: the storagetek products (9840 and 9940) and LTO are great products |
15:43.12 | [TK]D-Fender | My Certance LTO3 seems to run just fine.... though too damned slow for some reason :/ |
15:43.32 | ManxPower | monsted: do the storagetek tape drives last more than WARRENTY + 1 Day? |
15:43.42 | [TK]D-Fender | Its 400/800gb size doesn't seem so impressive now that 1TB HD's are coming out :/ |
15:43.47 | monsted | ManxPower: unless they break, yeah :) |
15:44.06 | monsted | ManxPower: you'd usually have a service contract to go with your $40,000 drive :) |
15:44.16 | ManxPower | My largest customer has something like 10 tape drives, not one of them works antmore |
15:44.28 | monsted | clever :) |
15:44.42 | ManxPower | They switched to rsync + big ass storage server |
15:44.44 | monsted | i think we've got 25 drives now, all of them in perfect working order |
15:44.45 | *** join/#asterisk javar (n=javar@69.79.134.24) |
15:45.31 | Katty | should i be a draenei or a blood elf? |
15:46.11 | monsted | well, blood elves associate with those smelly cows - i'd vote draenei |
15:46.12 | ManxPower | Katty: carnivorous mushroom is not an option? That would be cool. |
15:46.12 | [TK]D-Fender | Katty: Keebler :) |
15:46.40 | [TK]D-Fender | :D |
15:46.50 | Katty | monsted: the draenei do have a pretty cute /train |
15:47.30 | ManxPower | 8 mins to conference call |
15:48.22 | monsted | Katty: haven't seen it, i just can't be arsed to maintain a guild membership on both sides - i'll roll a draenei :) |
15:49.20 | *** join/#asterisk pif (n=ldm@zenon.apartia.fr) |
15:49.29 | *** join/#asterisk cbullock81 (n=cbullock@adsl-068-213-099-052.sip.jan.bellsouth.net) |
15:50.33 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
15:50.33 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
15:50.50 | ManxPower | Looks like I'm going to have to do a Customer Smackdown soon. |
15:51.45 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
15:53.40 | *** part/#asterisk alexandrepos (n=alexandr@201.21.143.130) |
15:56.19 | Katty | ManxPower: so you can mount a tape like you would a flash drive, dump some files on it using cp in a script, and umount it? |
15:56.34 | ManxPower | Katty: correct. |
15:56.39 | Katty | hot. |
15:56.42 | Katty | why can't windows do that :< |
15:56.48 | ManxPower | That would, of course overwrite any files on the tape. |
15:56.54 | Katty | most excellent. |
15:56.57 | ManxPower | Katty: uh, windows can do that |
15:57.03 | ManxPower | Oh wait! |
15:57.04 | Katty | not the way i want it to. |
15:57.09 | ManxPower | No, not MOUNT a tape drive |
15:57.10 | monsted | i did that on windows in '96 or so :) |
15:57.38 | Katty | i want the tape to be accessible just like a hard drive would be. |
15:57.45 | ManxPower | you cp the files to /dev/tapedrivedevice |
15:57.51 | ManxPower | Katty: never gonna happen |
15:57.56 | Katty | :< |
15:57.57 | monsted | Katty: that'll suck, but it's possible |
15:58.01 | ManxPower | because tapes don't have a file system |
15:58.48 | monsted | you just glue a file system on top of it - don't remember the software that does it, but it exists |
15:59.05 | ManxPower | monsted: I would not consider that a workable solution |
15:59.10 | Katty | meh |
15:59.18 | Katty | maybe i'll just rip that tape drive out |
15:59.21 | monsted | if you don't mind having a hard drive involved, you could use a HSM solution |
15:59.37 | monsted | ManxPower: as i said, it'll suck :) |
16:01.03 | monsted | tapes good, napster bad! |
16:01.11 | monsted | or something |
16:01.20 | *** join/#asterisk file (n=file@asterisk/developer-and-muffin-lover/file) |
16:01.20 | *** mode/#asterisk [+o file] by ChanServ |
16:01.21 | Katty | you're showing your age dear. |
16:02.20 | monsted | hierarical storage management is quite cool - you'll have your most often used/newest files on fast disks, older on slow disks and oldest on tape or optical |
16:02.58 | b11d | so, what do you kids think of this "music television" ? |
16:03.05 | monsted | and with recovery-friendly tape drives, it'll be quite usable |
16:03.28 | b11d | i enjoy tape backup |
16:03.34 | b11d | 400GB LTO's for me.. |
16:03.41 | b11d | those are probably small by now though |
16:03.49 | monsted | backup to LTO is great, recovery not so much :) |
16:04.00 | b11d | i dont use any kind of automated recovery.. so it works for me |
16:04.21 | b11d | automated recovery from tape, that is. |
16:04.33 | Katty | i just mount network drives, dump data, umount |
16:04.48 | b11d | whatever works for you in your situation, thats whats best. |
16:04.53 | cbullock81 | Hey. I have some confusion about SLA in 1.4. I've seen the commads and config files in it, but i've read that it's not fully functional? Anybody have any info on this. |
16:04.55 | monsted | Katty: that might not work for our 2000ish servers ;) |
16:04.56 | b11d | tapes are nice and easy for me to take to the bank on my way home |
16:05.03 | Katty | monsted: not my problem. |
16:05.12 | Katty | monsted: that's your problem - you deal with it (= |
16:05.17 | b11d | he is :) |
16:05.26 | Katty | most excellent. |
16:05.52 | monsted | we shoot it over fibre channel to another site about 5 miles away and onto storagetek 9840 tapes |
16:06.21 | b11d | how much data do you backup? |
16:06.22 | monsted | dual-reel things that are incredibly fast at loading/recovering |
16:06.39 | *** join/#asterisk CrashSys (n=kumba@bartleby.crashsys.com) |
16:06.41 | *** join/#asterisk marv[work] (n=timr@24.214.206.254) |
16:07.14 | b11d | heh I like that -- storage life: 15-30 years |
16:07.16 | b11d | thats a big gap.. |
16:07.17 | b11d | :) |
16:08.34 | monsted | b11d: i don't have any figures, but we've got about 12000 cartridges |
16:08.42 | b11d | impressive |
16:09.06 | b11d | i always wondered about enterprise level backup systems.. never had the chance to work on gear like that. |
16:09.16 | monsted | dual-reel tapes mean very low capacity, though |
16:09.23 | b11d | yeah like 20gb/ea or somethign eh |
16:09.29 | b11d | i take it thats uncompressed too.. |
16:09.32 | cbullock81 | Hey. I have some confusion about SLA in 1.4. I've seen the commads and config files in it, but i've read that it's not fully functional? Anybody have any info on this. |
16:10.13 | b11d | http://voip-info.org/wiki/view/Asterisk+SLA |
16:10.17 | b11d | thats about it.. i think.. |
16:10.26 | cbullock81 | yea... i read that... not much help tho |
16:10.29 | b11d | yeah.. |
16:10.34 | b11d | you should ask in #asterisk-dev maybe |
16:10.38 | cbullock81 | k |
16:10.40 | *** join/#asterisk UlbabraB (n=salama@host241-43-static.72-81-b.business.telecomitalia.it) |
16:10.47 | b11d | those guys are probably going to hate me :) |
16:10.55 | b11d | i doubt the devs want to be bothered :) |
16:10.55 | b11d | haha |
16:12.01 | monsted | b11d: yeah, 20 gig each |
16:12.21 | cbullock81 | heh... i hate to bother them, but im in the dark |
16:12.47 | b11d | well it says that Asterisk doenst have the support yet.. so thats pretty much it. |
16:12.50 | b11d | end of question. |
16:13.48 | *** join/#asterisk Avochelm (n=damien@gw-morphett.koalatelecom.com.au) |
16:14.06 | *** join/#asterisk JunK-Y (n=junky@modemcable105.205-56-74.mc.videotron.ca) |
16:14.58 | cbullock81 | they told me that it's being rewritten & to use the existing stuff @ my own risk and peril |
16:15.20 | b11d | fair enough |
16:16.29 | *** join/#asterisk kirberich (n=robert@i538711D5.versanet.de) |
16:18.55 | sweeper | mmm, getting some really awful distortion on the hold music |
16:19.58 | b11d | really |
16:19.58 | b11d | Katty was yesterday |
16:19.58 | b11d | oh no that was meetme.. |
16:20.06 | b11d | hrm.. are you using mpg123 to play the MoH, or the one included with aster-addons? |
16:23.05 | sweeper | mpg 123 |
16:23.07 | Dandre | is there some documentation about the syntax used in http manager interface in asterisk 1.4? |
16:23.36 | sweeper | interestingly, there are about 20 mpg123 processes running.... |
16:23.54 | CrashSys | Is there an option in the polycom config files to make it remember volume settings? That way everytime someone picks up the handset they dont have to raise the volume? |
16:24.24 | Katty | b11d: it was everything, actually |
16:24.33 | Katty | b11d: but then mysteriously went away, until i registered another SIP phone |
16:24.41 | Katty | b11d: and then it came back for awhile........then went away |
16:27.32 | *** join/#asterisk Gunnar (n=gunnar@nat.sigmasoft.com) |
16:28.53 | *** part/#asterisk javar (n=javar@69.79.134.24) |
16:30.30 | *** join/#asterisk jmls (n=asterisk@host86-135-41-172.range86-135.btcentralplus.com) |
16:30.42 | *** join/#asterisk RoyK (n=roy@217-175-39.100710.adsl.tele2.no) |
16:31.02 | jmls | hey, I need some help with realtime queues - all the extconfig is set up, the tables have data, asterisk has no queues :( |
16:31.13 | jmls | I was able to set up realtime voicemail with no problem |
16:31.46 | *** part/#asterisk jmls (n=asterisk@host86-135-41-172.range86-135.btcentralplus.com) |
16:31.59 | *** join/#asterisk jmls (n=asterisk@host86-135-41-172.range86-135.btcentralplus.com) |
16:32.11 | jmls | bugger. pressed the wrong button ;) |
16:35.13 | [TK]D-Fender | CrashSys: Yes, look for the "persist" tag in sip.cfg. You can make any audio source "sticky" that way |
16:35.31 | b11d | weird |
16:35.37 | b11d | sweeper.. switch to the aster-addons mp3 player |
16:35.45 | b11d | it does a much better job |
16:35.49 | b11d | i had the same issues with mpg123 |
16:35.56 | Qwell[] | or just convert your moh to ulaw or something |
16:36.02 | b11d | that didnt work for me |
16:36.07 | b11d | i had to switch to formatmp3 |
16:37.08 | sweeper | mmm |
16:37.13 | *** join/#asterisk ChicagoBud (n=Bud@adsl-70-228-35-78.dsl.chcgil.ameritech.net) |
16:37.35 | b11d | 'twas the only way she cleared up.. |
16:38.09 | b11d | Katty.. that is a bizarre problem.. i hate those sporadic ones. |
16:38.39 | Hmmhesays | yeah sporadic problems suck |
16:38.45 | Hmmhesays | b11d are you enxt to a fax machine? |
16:38.48 | Hmmhesays | *next |
16:38.54 | b11d | i can be |
16:39.04 | b11d | right across the hall :p |
16:39.42 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
16:39.42 | *** mode/#asterisk [+o mog] by ChanServ |
16:41.53 | *** join/#asterisk ManxPower (n=manxpowe@2.sub-75-203-79.myvzw.com) |
16:43.11 | *** join/#asterisk oQPa (n=roque@15.Red-83-40-197.dynamicIP.rima-tde.net) |
16:44.51 | ManxPower | For some reason I thought in 1.2 Hangup() allowed you to specify a hangup cause, but "show application hangup" does not indicate this. |
16:45.09 | rene- | a gang-bang |
16:45.12 | jmls | if you use realtime queues, show queues says "no queues", but you can use it . |
16:45.19 | b11d | hehe |
16:45.46 | Qwell[] | show queues only shows loaded queues |
16:46.02 | rene- | i think it will show queues after you have used it? |
16:46.03 | Qwell[] | you CAN do show queue blah, and it will load it (which means it'll be in show queues the next time you call it) |
16:46.03 | *** join/#asterisk alexandrepos (n=alexandr@201.21.143.130) |
16:46.05 | rene- | them |
16:46.11 | emiquelito | hello asterisk hackers! When I type 'sip show channels' at the asterisk CLI I can see some channels which are not active anylonger. How can I avoid that or even drop these dead channels? |
16:46.32 | alexandrepos | what a better phone sip or iax2 for kde ? |
16:46.34 | rene- | emiquelito: i think if you reload the sip module they will go away |
16:47.08 | emiquelito | rene- hmmm, but why it happens? Is there a parameter for this behaviour? |
16:47.44 | wunderkin | emiquelito, it will also show qualifies and registrations being sent since they are also calls |
16:47.57 | rene- | nice |
16:48.14 | rene- | wunderkin: can that behaviour affect call-limit ? |
16:48.53 | wunderkin | no |
16:50.05 | rene- | wunderkin: i have some dead channels to phones that have no call and then asterisk saying that the phones are at their call-limit, the phones being members of a dynamic queue but the queue is empty at this time? is there any other reason for those dead channels? |
16:50.30 | a1fa | Does anybody know if it will be possible to get a sip client on Apple's iPhone? |
16:50.33 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
16:50.33 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
16:50.38 | Qwell[] | a1fa: call them and ask :) |
16:50.45 | Qwell[] | ANYTHING we say would be speculation |
16:50.52 | perd | a1fa just buy a laptop, it's cheaper |
16:51.02 | wunderkin | wow |
16:51.20 | perd | hey qwell, did you fix skinny yet? :P |
16:51.27 | emiquelito | wunderkin I understand. I'm developing a softphone right now and sometimes I have to take a look at asterisk to see how things are going on there. Using 'sip show channels' I found lots of INVITE lines for the same extension. is this correct? my softphone often tries to keep registered in the server... maybe that's the problem |
16:51.31 | wunderkin | rene-, i dont know, you would have to show us what you are looking at |
16:51.36 | Qwell[] | perd: You didn't email me the 10 extra hours a day I need :P |
16:51.43 | b11d | haha |
16:51.47 | b11d | yeah perd.. |
16:51.48 | perd | aww i didnt attach that? |
16:51.52 | perd | damn me |
16:51.52 | Qwell[] | perd: guess not |
16:52.01 | b11d | christ theres a bunch of people waiting for that |
16:52.03 | a1fa | Qwell[] : ring ring |
16:52.05 | b11d | and you cant attach 10 hours? |
16:52.07 | b11d | geeze :) |
16:52.13 | a1fa | perd : how much will that iphone cost ? |
16:52.20 | a1fa | i've heard $400-$800 |
16:52.21 | perd | i think it's 400+ |
16:52.22 | b11d | i understand the iPhone is a flop.. |
16:52.23 | sweeper | 500/600 |
16:52.23 | perd | yeah |
16:52.29 | sweeper | for 4gb/8gb |
16:52.33 | monsted | a1fa: well, jobs said it won't run custom apps and i very much doubt it'll have SIP by itself |
16:52.47 | perd | touchscreen lcds suck |
16:52.55 | perd | horrible idea |
16:53.00 | sweeper | snark |
16:53.13 | coppice | optical touchscreen LCDs don't suck |
16:53.21 | sweeper | the lcd itself probably isn't touchscreen |
16:53.26 | perd | they get all greasy |
16:53.29 | monsted | i think the iphone will sell well, despite it being teh sux |
16:53.33 | perd | you have to constantly wipe it off |
16:53.40 | perd | you cant feel the buttons so you have to look at the phone to dial it |
16:53.56 | a1fa | monsted : jobs is a douche |
16:53.58 | b11d | yeah.. |
16:54.01 | sweeper | some people have personal hygene, don't ahve horridly greasy hands, mr baconmitts |
16:54.04 | b11d | but Steve Ballmber is a bigger douche |
16:54.06 | b11d | Ballmer |
16:54.11 | rene- | what is a douche |
16:54.11 | a1fa | Ballsucker |
16:54.16 | a1fa | rene- : i dont know |
16:54.17 | b11d | really? |
16:54.18 | perd | <PROTECTED> |
16:54.20 | b11d | hahaha |
16:54.26 | rene- | the iraqui used to |
16:54.27 | b11d | a douche is what women use to flush out their vaginas after intercourse |
16:54.32 | a1fa | i just like how it sounds |
16:54.33 | perd | fucking baconmitts hahah |
16:54.39 | rene- | jajaaj |
16:54.43 | a1fa | skwak |
16:54.48 | rene- | then jobs is a douche |
16:54.52 | b11d | http://en.wikipedia.org/wiki/Douche |
16:54.54 | a1fa | lets write an email to jobs tell him he is an asshole |
16:54.58 | a1fa | he needs to let us run custom apps |
16:55.00 | a1fa | WTF! |
16:55.05 | a1fa | i am sure somebody will hack it |
16:55.13 | b11d | yeah you know it |
16:55.16 | rene- | well he hasnt opened ichat after so many years and nobody has hacked it |
16:55.22 | b11d | yueah well that sucked |
16:55.45 | rene- | otoh i think it does jabber now |
16:55.50 | sweeper | because ichat blows |
16:55.58 | a1fa | wtf is ichat? |
16:55.59 | a1fa | iDildo? |
16:56.03 | b11d | garbage |
16:56.07 | a1fa | iGoatSecX1 |
16:56.10 | b11d | its what women use to flush out their ears after intercourse |
16:56.10 | sweeper | thing is, ipod linux is pretty unusable |
16:56.11 | rene- | well the sms thing in the appel phone is pretty much ichat |
16:56.13 | De_Mon | sounds macish |
16:56.41 | sweeper | but maybe the iphone will inspire better work |
16:56.56 | a1fa | i like multi-touch idea |
16:57.07 | sweeper | or, they'll release some sort of api for the iphone. I mean, they said it supports widgets |
16:57.17 | *** part/#asterisk jmls (n=asterisk@host86-135-41-172.range86-135.btcentralplus.com) |
16:57.26 | a1fa | hm.. cool |
16:57.29 | a1fa | widget sip client |
16:57.34 | rene- | yeah but they didnt said third party widgets |
16:57.43 | *** join/#asterisk qdk (n=qdk@0x535eae17.boanxx9.adsl-dhcp.tele.dk) |
16:57.48 | a1fa | i am sure someone will hack it |
16:57.51 | a1fa | ipod was hacked |
16:57.55 | heh_v_water | I doubt the iphone will be called iphone |
16:57.59 | rene- | i want someone to hack the apple tv |
16:58.05 | sweeper | heh_v_water: I saw that. lol cisco |
16:58.11 | rene- | could it be used as a desktop pc? |
16:58.15 | rene- | with like linux? |
16:58.22 | sweeper | why the hell would you? |
16:58.25 | sweeper | get a mini :/ |
16:58.31 | *** part/#asterisk Bazy (n=bazy@86.125.51.251) |
16:58.39 | heh_v_water | yup mini's rock if you want mac |
16:58.47 | a1fa | mac < gay |
16:59.01 | a1fa | transvestite > mac < gay |
16:59.07 | heh_v_water | lol |
16:59.18 | a1fa | i got rage! |
16:59.34 | rene- | swepeer i have a mac |
16:59.57 | rene- | mini |
17:00.05 | a1fa | heh |
17:00.21 | rene- | but if i could get a mac for what 250~ |
17:00.38 | [TK]D-Fender | Mac = lack of volume of good free sofware. On WinXP I have TONS. On Linux I have TONS. |
17:01.07 | rene- | D-Fender: most free software (open source) can be used and compiled in a mac |
17:01.17 | [TK]D-Fender | Mac's Wifi is RETARDED and I can't get my stupid 40bit WEP working on a PowerBook G4 I'v got here... |
17:01.26 | heh_v_water | [TK]D-Fender, agreed BUT.. mac does have actual modern games that can be loaded on and played without too much heartache |
17:01.33 | [TK]D-Fender | rene-: Problem is that not enough of it HAS yet. |
17:01.38 | heh_v_water | I don't use macs myself |
17:01.45 | rene- | games are a problem |
17:01.52 | rene- | and ubuntu is so cool |
17:02.07 | a1fa | D-Fender > * |
17:02.35 | [TK]D-Fender | heh_v_water: Yes, different point though. I'm talking about all the normal productivity stuff. I haev PDF Creator, FileZilla, Scribus, InkScape, and so much more that runs very well natively on PC, just not for Mac. |
17:02.40 | monsted | [TK]D-Fender: you can compile almost anything on osx, just like on linux - what seems to be the problem? |
17:03.00 | rene- | pdf creator? just print to pdf, |
17:03.05 | sweeper | w00t, asterisk-addons ftw |
17:03.08 | rene- | inkscape have it installed |
17:03.13 | CrashSys | d-fender: You were saying to look for "Persist" to make the volume controls static? |
17:03.14 | sweeper | inkscape \o |
17:03.23 | [TK]D-Fender | monsted: I'm not a coder and don't want to compile basic destop apps. Sure for *, and a few other specialty servers, but not for joe-blow stuff |
17:03.33 | wunderkin | CrashSys, yeah... have you upgraded to 2.0.3 yet? |
17:03.37 | [TK]D-Fender | CrashSys: yes |
17:03.44 | a1fa | in other news, Beckham is coming to U.S. |
17:03.51 | monsted | [TK]D-Fender: no different than linux |
17:03.54 | CrashSys | D-Fender: Gracias :) |
17:03.57 | a1fa | gay > beckham < mac < gay |
17:04.07 | monsted | [TK]D-Fender: and there are tools to do it for you |
17:04.12 | *** join/#asterisk Nukemizer (n=Nuke@67.137.28.165) |
17:04.23 | [TK]D-Fender | monsted: Hate to say I'm expecting "Click&install). |
17:04.34 | CrashSys | whoooo... gains... |
17:04.58 | *** join/#asterisk bwlang (n=bwlang@bb-66-55-211-238.gwi.net) |
17:05.17 | [TK]D-Fender | monsted: With APT on distro's like Ubuntu, installing is more than easy. Windows is the usualOkFineSureMinusThatOneSetting standard install procesure. |
17:05.22 | rene- | inkscape was click and install for me i mean i cant apt-get it but average joe doesnt apt-get stuff |
17:05.27 | a1fa | heh |
17:05.31 | heh_v_water | you would think by now there would be some sort of binary repository of open source software for mac |
17:05.32 | [TK]D-Fender | CrashSys: Don't mess with gains unless you HAVE to. |
17:05.44 | a1fa | i prefer debian over ubuntu |
17:05.52 | [TK]D-Fender | CrashSys: Not to be mistake with VOLUMES. You can cause echo and other shit if you screw up. |
17:05.54 | a1fa | aka, the real thing |
17:05.58 | rene- | yes |
17:06.04 | heh_v_water | a1fa, amen |
17:06.11 | rene- | i am weary of using ubuntu on servers |
17:06.13 | a1fa | god bless you |
17:06.14 | CrashSys | well those gains are actual DB right? not like zapata's gains right? |
17:06.21 | rene- | for laptops is teh thing |
17:06.21 | PupenoR | would it cause any problem for queues to be named by numbers only ? |
17:06.27 | [TK]D-Fender | a1fa: Yeah I probably would too if they made an equally convenient installer disc. |
17:06.43 | a1fa | [TK]D-Fender : i use netinstall images |
17:06.54 | a1fa | netinstall in expert mode |
17:07.03 | [TK]D-Fender | CrashSys: I don't know that partuiculars, just set persist and adjust at source. You should never have to mess with "raw" gains on SIP phones. |
17:07.07 | *** join/#asterisk CunningPike (n=CunningP@dhcp-10-234.district.north-van.bc.ca) |
17:07.20 | heh_v_water | [TK]D-Fender, you may get your wish in the next few months.. when Debian Etch gets released there is going to be all kinds of new debian disks available including a live cd and such |
17:07.26 | [TK]D-Fender | a1fa: NO! I want the basics (even old), to come off CD and from there let me choose what to do. |
17:07.41 | a1fa | aha |
17:07.44 | CrashSys | I like my Slack |
17:07.46 | CrashSys | 1-cd install |
17:07.53 | CrashSys | unless you want X or other stuff |
17:07.55 | a1fa | CrashSys : i like my debian netinstall - 50mb |
17:07.55 | [TK]D-Fender | a1fa: I don't want to spend lots of time waiting for crap to download and install (or worse, compile like Gentoo..... friggen RICERS!) |
17:08.01 | heh_v_water | I install debian with a 50MB business card cd.. :P |
17:08.11 | [TK]D-Fender | CrashSys: Agreed, but I don't like it of standard desktop use. |
17:08.18 | a1fa | [TK]D-Fender : i know what you mean.. gentoo is for S&M people |
17:08.24 | a1fa | fucking masochists |
17:08.30 | CrashSys | I dont use linux as a desktop... it has a hard time playing Asheron's Call :D |
17:08.58 | *** join/#asterisk h0 (n=fakhir@unaffiliated/fakhir) |
17:09.14 | CrashSys | Plus there's no real good source of video editing software for it, short of writing your own... |
17:09.21 | heh_v_water | it took me years to finally stop using Windows as my primary desktop machine.. now I use debian almost exclusively.. when i use windows I yearn for ym debian |
17:09.31 | penguinFunk | callers complain that they find it difficult to hear people in this office because the outgoing voice is a bit too quiet, people in the office can hear the callers perfectly fine tho. How can i increase the outgoing volume just a little bit ? |
17:09.34 | a1fa | AMEN BROTHER! |
17:09.36 | monsted | it's too bad linux is so badly made |
17:09.38 | BSDTech | freebsd |
17:09.47 | Qwell[] | badly made? |
17:09.51 | BSDTech | easy fast clean and asterisk is in ports |
17:09.52 | heh_v_water | lol |
17:09.52 | a1fa | monsted : lol? |
17:09.53 | sweeper | penguinFunk: rxgain/txgain in the channel conf |
17:10.02 | a1fa | ~lart monsted |
17:10.04 | CrashSys | I like my OpenBSD! |
17:10.05 | monsted | if only people had thrown their support behind something that works nicely like freebsd, things would've been much better |
17:10.09 | penguinFunk | thanks sweeper |
17:10.10 | CrashSys | But alas, no zaptel... |
17:10.24 | CrashSys | I'm hoping for a FreePBX/LibSangoma solution some time soon :) |
17:10.27 | BSDTech | and asterisk runs great |
17:10.31 | sweeper | OpenBSD is for hippy beer makers with attitude problem |
17:10.33 | sweeper | *s |
17:10.35 | a1fa | CrashSys : i like openbsd as well.. reminds me of debian in early days |
17:10.43 | sweeper | I'M LOOKING AT YOU THEO |
17:10.45 | CrashSys | I make beer |
17:11.01 | CrashSys | well, drink it... |
17:11.03 | a1fa | Do you have any mexican in you CrashSys ? |
17:11.09 | penguinFunk | and by channel conf you mean zaptel.conf ? |
17:11.12 | BSDTech | move asterisk to freebsd and you have a stable pbx |
17:11.13 | CrashSys | No, but i've been in a mexican before... |
17:11.15 | CrashSys | does that count? |
17:11.27 | a1fa | In mexican accent, so would you like some, esse? |
17:11.45 | BSDTech | Kpasa |
17:11.45 | sweeper | CrashSys: nobody wants to hear about your gay love affair with the delivery boy :v |
17:12.01 | BSDTech | com-esta |
17:12.02 | CrashSys | awww, but he was so cute... |
17:12.05 | sweeper | besides, we all know you're the catcher >.> |
17:12.14 | CrashSys | Now that's just rude... |
17:12.16 | BSDTech | ).( |
17:12.21 | BSDTech | lol |
17:12.28 | monsted | Qwell[]: please convince your digium colleagues to move primary development from linux to freebsd ;) |
17:12.43 | Qwell[] | freebsd is crap |
17:12.45 | Qwell[] | sorry |
17:12.45 | file | ha |
17:12.52 | BSDTech | I have been trying for 3 years to get them to move |
17:12.55 | BSDTech | they wont |
17:13.01 | [TK]D-Fender | CrashSys: You looking for an automatic way to completely configure Sangoma's, or just get it working? Latter should be aeasy |
17:13.02 | BSDTech | Mark is to head strong |
17:13.18 | CrashSys | d-fender: eh? |
17:13.34 | CrashSys | bsd: You use sangoma for * on BSD? |
17:13.34 | [TK]D-Fender | As Nugget usually says "Linux is poo" |
17:13.40 | BSDTech | if you use the sangoma rpb and its a a200 card there is a setup script |
17:13.47 | BSDTech | yes |
17:13.56 | BSDTech | a100 and a200 |
17:13.59 | sweeper | Qwell so mean D: |
17:14.00 | rene- | BSDTech: linux+asterisk doesnt crash for me, when you say stable you mean less line errors or less calls dropped for you? |
17:14.29 | sweeper | loooonix |
17:14.32 | BSDTech | all the above freebsd overall has been stable for me with sangoma cards and asterisk |
17:14.41 | a1fa | * 1.4 is most stable so far |
17:14.48 | a1fa | DTMF works perfectly |
17:14.50 | BSDTech | not |
17:14.54 | rene- | yeah sangoma |
17:14.56 | CrashSys | 1.2.14 has been pretty stable for me so far |
17:15.03 | CrashSys | less "strange" things happening... |
17:15.05 | BSDTech | it has major transcoding issues |
17:15.21 | rene- | 1.4 has broken transfers for me |
17:15.26 | BSDTech | that also |
17:15.28 | rene- | so broken it takes asterisk down |
17:15.31 | sweeper | I'd like to tune it to a streaming radio, but looks like a hassle :v |
17:15.45 | BSDTech | nope just install madplayer |
17:16.10 | BSDTech | and then setup madplayer in moh.conf andpoint it at the stream you want |
17:16.16 | sweeper | madplayer bitches about a lack of soundcard |
17:16.27 | sweeper | and then dies |
17:16.43 | BSDTech | not on bsd |
17:16.47 | rene- | i have seen some roomba clones for like 30~ i wonder if those work well or what |
17:16.47 | sweeper | well, lack of /dev/esd, to be specific |
17:17.07 | wunderkin | CrashSys, have you updated to 2.0.3? |
17:17.08 | sweeper | oops |
17:17.10 | sweeper | /dev/dsp |
17:17.37 | sweeper | fucking gentoo |
17:17.37 | rene- | BSDTect: not on bsd was you talkin to me? |
17:17.38 | BSDTech | madplay works fin on my server and it has no sound card |
17:17.52 | sweeper | stop giving bsd a bad name :/ |
17:18.24 | BSDTech | gentoo-bsd |
17:18.26 | BSDTech | lol |
17:18.53 | luisjose | BSDTech, my asterisk crash when a linphone client try to log in |
17:18.56 | luisjose | BSDTech, seg fault |
17:19.19 | sweeper | luisjose: awsome |
17:19.25 | BSDTech | wow |
17:19.32 | BSDTech | are you on linux or bsd |
17:19.37 | luisjose | BSD |
17:19.44 | BSDTech | what ver of asterisk |
17:20.01 | cbullock81 | i got a newbie question. how do you guys deal with letting the phones (or users) know that all outbound lines are busy before they try a new call? |
17:20.14 | sweeper | see, I realized why you * zealots like to compile from source...very few dependancies :P |
17:20.19 | *** join/#asterisk Assid (i=assid@59.183.52.135) |
17:20.26 | BSDTech | youcansetup a roll over |
17:20.40 | BSDTech | and have it tell the user that the lines are busy |
17:20.56 | BSDTech | luis what ver of asterisk |
17:21.19 | cbullock81 | could you expound on that a little... i am totally new |
17:21.42 | BSDTech | is this for sip or zaptel setup |
17:21.46 | cbullock81 | zap |
17:21.53 | [TK]D-Fender | cbullock81: clarify the "BEFORE they try a NEW call" part please... |
17:22.56 | cbullock81 | like if i have 2 zap channels outbound & both are busy. When i pickup and try to dial an outbound line, is there a way to notify me that the lines are busy |
17:23.46 | BSDTech | he wants a check stataus |
17:23.50 | BSDTech | setup |
17:24.26 | BSDTech | where he dials the number if line 1 and line 2 are busy it then says all outbound circuts are busy please try again ltr |
17:24.27 | luisjose | sweeper, I did compile asterisk from source. |
17:24.40 | BSDTech | louis you did not answer me |
17:24.44 | BSDTech | what ver of asterisk |
17:24.55 | luisjose | BSDTech, oh sorry |
17:24.56 | cbullock81 | 1.4 |
17:25.02 | [TK]D-Fender | cbullock81: Yes, but not BEFORE they dial.... |
17:25.07 | luisjose | Asterisk 1.2.13 |
17:25.35 | cbullock81 | well, i would love for the phones to indicate the lines are busy before even trying an outbound call, but i dont know if asterisk supports that |
17:25.37 | BSDTech | do a update on your ports tree I believe it should be 1.2.14 now |
17:25.46 | [TK]D-Fender | cbullock81: They dial, * reports back "CONGENTION" in DIALSTATUS and then you play a sound file saying "try again later please". Thats how |
17:25.48 | BSDTech | and I believe there is some fixes |
17:26.19 | wunderkin | well, anyone here with an ip430? |
17:27.03 | a1fa | When is the new version planed for release? |
17:27.12 | [TK]D-Fender | wunderkin: You mean to do more tests than we've already gone through? |
17:27.35 | luisjose | BSDTech, oh ok, wondering when 1.4 is going to be on ports |
17:27.50 | BSDTech | 1.4 is a bug release it was ment to bug the hell out of you |
17:27.57 | BSDTech | when it is fixed |
17:28.11 | a1fa | like i said, i had no problems, except pitch changer wont compile |
17:28.17 | luisjose | oh i thought it was released already. |
17:28.18 | wunderkin | yeah, i just wanted to get a couple other people to test it, either its this batch of hardware in the phones or my config, but they say to rma but i dont know if that was from polycom or not |
17:28.21 | a1fa | justin is releasing new patch next week |
17:28.33 | BSDTech | try transcoding g729 to ulaw or gsm |
17:28.38 | BSDTech | <PROTECTED> |
17:28.47 | a1fa | nah :) |
17:28.50 | a1fa | i keep it simple |
17:28.53 | BSDTech | try g726 to ulaw it crashes |
17:28.55 | a1fa | ulaw across the board |
17:29.03 | Qwell[] | BSDTech: try upgrading to 1.4 svn |
17:29.13 | BSDTech | ok will do |
17:29.20 | luisjose | Hey fellas, for example, which hardware you guys will use to set up lets say an hotel with 200 extensions? |
17:29.27 | luisjose | FXS hardware. |
17:29.31 | Qwell[] | luisjose: analog? |
17:29.39 | Qwell[] | some quad T1 cards and channel banks, probably |
17:29.55 | wunderkin | anyone that tells you to use sip phones in the rooms are freaking high |
17:30.05 | luisjose | Qwell[], yes. |
17:30.10 | a1fa | wunderkin : unless its a high class hotel |
17:30.17 | a1fa | i've been in hotel rooms with cisco phones |
17:30.26 | BSDTech | why you can get grandstream 200's for 48 bucks each |
17:30.37 | BSDTech | and they work |
17:30.39 | wunderkin | ive never been in a 'high class' hotel before, 2 star for me baby |
17:30.48 | a1fa | wunderkin : cheap ass |
17:30.50 | BSDTech | but soon hotel rooms wont have phones |
17:31.03 | a1fa | I dont stay in a hotel unless its 5 * or higher |
17:31.15 | luisjose | a1fa, bastard |
17:31.18 | a1fa | BSDTech : comparing to $5 for ordinary hotel phones |
17:31.23 | *** join/#asterisk [hC] (n=hardcore@S0106000fb51cc225.vf.shawcable.net) |
17:31.23 | a1fa | luisjose : i was kidding |
17:31.28 | BSDTech | hotels are over rated impose on friends its more fun |
17:31.31 | luisjose | Ok sorry then. |
17:31.34 | a1fa | yeah |
17:31.38 | a1fa | i impose on my friends most of the time |
17:31.39 | Qwell[] | 5 star hotels are for nubs |
17:31.42 | Qwell[] | I stay in 7 star hotels |
17:31.47 | a1fa | I pay in beer |
17:31.50 | Qwell[] | of course, there is only...1, but what the hell |
17:31.57 | luisjose | Ok so, how are these channel bank? |
17:32.01 | a1fa | 24 pack of Patt's Blue Ribbon |
17:32.05 | luisjose | never seen one on sites or anywhere |
17:32.07 | BSDTech | lol |
17:32.08 | Qwell[] | luisjose: They connect to a T1 card, and split out the signalling |
17:32.11 | BSDTech | and pizza |
17:32.13 | luisjose | how many fxs ports they have? |
17:32.14 | Qwell[] | to fxs/fxo ports |
17:32.19 | Qwell[] | 24 or 32, depending |
17:32.21 | a1fa | 24 |
17:32.21 | BSDTech | and the guest room is always open |
17:32.26 | Qwell[] | 24 or 31? dunno |
17:32.33 | BSDTech | ok 200 rooms |
17:32.35 | wunderkin | 42 |
17:32.36 | a1fa | i thought T1 had 21 channels |
17:32.41 | Qwell[] | 24 channels |
17:32.49 | a1fa | yeah.. 24 channels |
17:32.50 | a1fa | sorry |
17:32.56 | luisjose | so if i need 200 extensions? |
17:32.57 | BSDTech | 23 voice and 1 data |
17:33.03 | Qwell[] | luisjose: you just get a bunch of them |
17:33.07 | Qwell[] | or a higher density channelbank |
17:33.07 | wunderkin | BSDTech, thats for pri |
17:33.08 | BSDTech | 23b and 1 d |
17:33.09 | danp | that's PRI |
17:33.12 | BSDTech | yes |
17:33.25 | Qwell[] | I think there are dual and maybe quad T1 channelbanks out there |
17:33.27 | luisjose | a t1 card for each 24 ext? |
17:33.31 | Qwell[] | with 48 and 96 channels |
17:33.35 | Qwell[] | luisjose: correct |
17:33.45 | luisjose | thats a waste, they wont be using all ext at same time. |
17:33.47 | BSDTech | a101 from sangoma |
17:33.53 | BSDTech | is a quad card |
17:33.54 | Qwell[] | luisjose: but you must split them out |
17:34.02 | wunderkin | use monkeys |
17:34.07 | Qwell[] | otherwise, when two people pick up a phone sharing a port...they can hear each other |
17:34.08 | danp | just have rooms share lines...that should be fine |
17:34.14 | Qwell[] | and if somebody calls them...BOTH rooms ring |
17:34.16 | Qwell[] | that is also bad |
17:34.32 | BSDTech | I do hotels |
17:34.34 | Qwell[] | you don't actually need T1s to the telco for as many rooms as you have |
17:34.40 | luisjose | Qwell[], in what kind of computer im gonna hook 8 t1 cards? |
17:34.40 | BSDTech | I have done 24 dyas in hotels |
17:34.51 | BSDTech | days inn that iis |
17:34.53 | monsted | luisjose: if you're using channel banks, each T1 channel corresponds to an analogue phone - you can still oversubscribe the outgoing T1 line |
17:34.54 | Qwell[] | luisjose: 2 quad span cards in 1 machine isn't unreasonable |
17:35.04 | a1fa | yeah |
17:35.05 | BSDTech | and we use a channel bank |
17:35.07 | a1fa | you can run 8 fine |
17:35.10 | [TK]D-Fender | BSDTech: You mean A104..... |
17:35.13 | monsted | hell, some servers actually have 8 pci slots :) |
17:35.14 | a1fa | dual opteron |
17:35.17 | BSDTech | sorry yes |
17:35.19 | BSDTech | the 104 |
17:35.21 | a1fa | 16 GB of ram |
17:35.23 | Qwell[] | [TK]D-Fender: you mean TE410p ;) |
17:35.27 | BSDTech | brain a little fried |
17:35.36 | BSDTech | been up for 36 straight |
17:35.39 | luisjose | Ok wait wait |
17:35.51 | [TK]D-Fender | Qwell[]: Of all Digium models, that is the LAST one I'd choose :) |
17:35.58 | luisjose | <monsted> luisjose: if you're using channel banks, each T1 channel corresponds to an analogue phone - you can still oversubscribe the outgoing T1 line <- i dont get it |
17:35.59 | BSDTech | get a rhino channel bank |
17:36.00 | Qwell[] | okay, TE412p |
17:36.09 | Qwell[] | but, for a channelbank...really, an echo can? nah |
17:36.43 | Qwell[] | luisjose: For 200 users, you'd have something like maybe 2-3 T1s to the telco |
17:36.45 | tzanger | Qwell[]: you need 'em |
17:36.55 | tzanger | Qwell[]: or rather CAN need 'em |
17:36.57 | Qwell[] | tzanger: I stand corrected then. I've never done fxs :) |
17:37.01 | tzanger | hell I need an echocan for my PRI |
17:37.03 | Qwell[] | not like that anyhow |
17:37.09 | BSDTech | depends on how many outbound /inbound lines the hotel needs/wants |
17:37.39 | tzanger | echocan for fxs? not really needed for most installs I've seen |
17:37.39 | tzanger | my mistake |
17:37.48 | tzanger | FXO needs echocan for sure (usually sw works fine but really the TE407 is amazing, I'm VERY happy) |
17:38.02 | wunderkin | luisjose, think of the channel bank as a switch i guess |
17:38.05 | tzanger | FXS needs for ecohcan are much less stiff, but I could forsee it |
17:38.13 | luisjose | Ok I chosed a bad example. no hotel, corporation with a lot of inside traffict not outgoing calls |
17:38.20 | luisjose | just on internal extensions |
17:38.29 | luisjose | corporation |
17:38.32 | BSDTech | then Polycom |
17:38.40 | luisjose | but with fxs |
17:38.44 | luisjose | analog |
17:38.46 | BSDTech | depends on how many line buttons you want |
17:38.58 | luisjose | damnit. |
17:39.00 | BSDTech | then a channelbank |
17:39.05 | luisjose | channel bank |
17:39.18 | BSDTech | look at rhino |
17:39.19 | luisjose | ok so how many analog phone can I hook to a channel bank? |
17:39.33 | danp | 24, same as in a hotel :P |
17:39.38 | BSDTech | depends on the channel bank you get |
17:39.45 | Qwell[] | luisjose: 24 if it's a single span channelbank |
17:39.53 | Qwell[] | I'm fairly certain that dual exists, and likely quad |
17:40.05 | monsted | hmm |
17:40.06 | luisjose | so i must die with a line per channel. |
17:40.09 | Qwell[] | and hell, you could probably find an 8 span if you look hard enough |
17:40.15 | rene- | asterisk seems |
17:40.20 | Qwell[] | luisjose: for the phones, yes, BUT, NOT for the "outgoing" lines |
17:40.23 | monsted | any reason not to buy a european channel bank and get 30 channels for the price of 23? :) |
17:40.24 | Qwell[] | ie; the T1's to the telco |
17:40.41 | luisjose | it does not make sense to me, what about i get ip phones and connect it to my lan |
17:40.49 | luisjose | on a giga ethernet |
17:40.49 | Qwell[] | you only really need a 1/4 or 1/3 ratio or something. That's something you'd have to just estimate at first, then expand later |
17:40.52 | rene- | i downloaded trunk and manager seems still broken to me |
17:40.55 | Qwell[] | You could, sure, BUT... |
17:41.00 | BSDTech | then get polycoms |
17:41.02 | rene- | not able to login |
17:41.05 | Qwell[] | IP phones are much easier/more valuable to steal |
17:41.10 | [TK]D-Fender | tzanger: You said Sangoma's have failed for you, have you tried Digiums new Otasic-powered lineup? |
17:41.17 | rene- | yeah but generally |
17:41.23 | Qwell[] | luisjose: If you aren't worried about theft, then absolutely, go with IP phones |
17:41.24 | luisjose | yes but why you can apply that concept to fxs lines |
17:41.30 | BSDTech | I have yet to have a sangoma fail me |
17:41.38 | Qwell[] | nobody is gonna steal a $20 analog phone, heh |
17:41.42 | BSDTech | I have had 4 digium cards go south on me |
17:41.48 | Qwell[] | but, if a polycom is sitting in there...yeah, it's more likely |
17:42.09 | [TK]D-Fender | BSDTech: I know, tzanger is the only case I've heard, and he's knowledgeable and has work with Sangoma support for a LONG time working on it. |
17:42.34 | tzanger | [TK]D-Fender: as I said, TE407 kicks serious ass |
17:42.36 | luisjose | Ok but i think you guys dont get my point |
17:42.50 | Qwell[] | luisjose: maybe not - explain it |
17:42.51 | luisjose | why i dont need to use a 64kbps per channel |
17:42.52 | [TK]D-Fender | Qwell[]: Thats why I bought 2 x Uniden UIP-200's for high damage/theft risk areas :) Phones that can take a beating, and I could afford to lose |
17:42.55 | luisjose | i mean |
17:43.02 | BSDTech | hmmmm |
17:43.05 | luisjose | i can hook as many phone as my bw allow me |
17:43.05 | BSDTech | weird |
17:43.05 | Qwell[] | [TK]D-Fender: those SIP or something? |
17:43.08 | BSDTech | what card |
17:43.15 | [TK]D-Fender | tzanger: The good way, right? So why do you need add'l echo-can? |
17:43.17 | tzanger | [TK]D-Fender: Sangoma I am not 100% failed me. It took fucking FOREVER to get it replaced, and then I specifically asked for a failure report, to know what exactly went wrong, and they coughed and replied that they didn't test it, they just sent me a new one |
17:43.27 | luisjose | maybe if i have low traffic i can hook 500 ip phones |
17:43.28 | [TK]D-Fender | Qwell[]: Yup.. They're CRAPTASTIC! |
17:43.33 | tzanger | [TK]D-Fender: TE407 has the octasic echo can, that's what I'm talking about |
17:43.40 | BSDTech | you can also get the cheap grandstream 200's and no one will steal them |
17:43.47 | BSDTech | and they are single line phones |
17:43.49 | tzanger | [TK]D-Fender: I *have* echo can, in the form of the octasic echo can in the TE407 |
17:43.51 | luisjose | in other setup ill be only able to use 50 phones but! |
17:44.06 | luisjose | why i have to waste 2 T1 on a setup with a low traffic on FXS line. |
17:44.09 | [TK]D-Fender | tzanger: Thought you just asked about getting one for PRI though... whats that about then? |
17:44.11 | Qwell[] | luisjose: Those are the tradeoffs |
17:44.14 | BSDTech | well wait |
17:44.18 | luisjose | trying to get stuff cheap. |
17:44.19 | *** join/#asterisk shepimport (n=shep@h194.189.31.71.ip.alltel.net) |
17:44.23 | Qwell[] | luisjose: like I said, one of the most important factors (in a hotel env) is theft |
17:44.29 | BSDTech | then go grandstream |
17:44.38 | Qwell[] | If you can afford to lose a $100 IP phone every couple days...then go for it |
17:44.38 | luisjose | gah |
17:44.45 | tzanger | I had TE405 on this PRI before, and had echo problems. I had TE406 on this PRI before, again with echo issues (the 406 is TRASH). A104d worked well but had audio artifacting they could NOT solve, and I haven't tried the replacement A104d yet, since the TE407 is working 100% |
17:44.47 | [TK]D-Fender | BSDTech: like Andrew Dice Clay said "Some people play hard to get.... I play hard to WANT! heeeyyyyaaa!" |
17:44.51 | b11d | sweeper.. did you switch to format_mp3? |
17:44.51 | luisjose | you can get a POTS for $9 :P |
17:44.54 | monsted | Qwell[]: you'd bill the customer if the phone has gone missing |
17:44.57 | BSDTech | lol |
17:45.10 | Qwell[] | monsted: good luck A) proving it, B) getting the money (in a timely matter) |
17:45.11 | monsted | Qwell[]: and as a bonus, you can check logs to see when the phone went offline |
17:45.12 | BSDTech | TK your to much |
17:45.15 | BSDTech | lol |
17:45.22 | [TK]D-Fender | tzanger: Not any clearer.... whats this about a CURRENT need for a PRI EC? |
17:45.23 | BSDTech | LOUIS |
17:45.23 | Qwell[] | A is easiest, but...yeah |
17:45.27 | BSDTech | get a grip |
17:45.28 | luisjose | so as conclusion, I need a channel bank and a t1 every 24 analog phones? |
17:45.30 | BSDTech | and listen |
17:45.40 | Qwell[] | luisjose: If you use analog phones, yes |
17:45.43 | tzanger | [TK]D-Fender: heh. I have a PRI and it needs EC. The TE407 is my EC for that PRI that needs it. |
17:45.51 | monsted | luisjose: you're building from scratch? |
17:45.58 | luisjose | monsted, yes |
17:45.58 | shepimport | hey all... anyone have asterisk w/ a session border controller experience? |
17:46.07 | [TK]D-Fender | tzanger: As in you'd like to take the TE407P *out* of the equation? |
17:46.09 | luisjose | monsted, well looking forward to. |
17:46.10 | monsted | luisjose: it'd be insane to buy analogue stuff then :) |
17:46.28 | luisjose | analog phone goes from $9... |
17:46.33 | BSDTech | tzangert sangoma card did you have |
17:46.36 | luisjose | ip phones at least $40 |
17:46.46 | monsted | channel banks and T1 cards are far from free |
17:47.18 | a1fa | just use ip phones |
17:47.24 | a1fa | see how many get stolen |
17:47.28 | tzanger | [TK]D-Fender: no no no no no |
17:47.30 | monsted | definitely go for IP phones |
17:47.31 | [TK]D-Fender | luisjose: tzanger: Perhaps I misinterpreted your conversations starting line : <tzanger>hell I need an echocan for my PRI |
17:47.34 | a1fa | in frist week of operation |
17:47.48 | a1fa | make sure you use budgetcrap phone |
17:48.00 | a1fa | and dont take cash for rooms.. take visa/mc |
17:48.08 | a1fa | so you can charge them if they steal your sip phone |
17:48.10 | tzanger | [TK]D-Fender: I do not want to take the TE407 out. I love it, it is working 100% |
17:48.19 | *** join/#asterisk alamantia (i=Anthony@nat/digium/x-588291bb6c886e47) |
17:48.21 | tzanger | [TK]D-Fender: I was just saying "I have a PRI that needs echo cancellation" |
17:48.29 | `Sean | a1fa |
17:48.32 | a1fa | You can get a big discount if you buy 200 IP phones at once |
17:48.35 | `Sean | then they can just dispute the charge |
17:48.40 | [TK]D-Fender | luisjose: Channel banks mean you spend a lot of money on a T1 card, the channel bank itsel, phones, power bars to support the power bricks for the phones, and then the sucky interfacs that is Zaptel FXS :( |
17:48.41 | a1fa | for like $20 per phone |
17:48.57 | rene- | look for hospitality ip phones |
17:49.01 | luisjose | ill just embed a pc into the wall and use softphones |
17:49.05 | rene- | they are expoensive but they can take a beating |
17:49.11 | [TK]D-Fender | tzanger: Yeah, that to me says I have a need... no that you FILLED it :) Clear now! |
17:49.12 | monsted | i'm hoping we land a deal that involves us getting 35000 phones :) |
17:49.20 | a1fa | luisjose : i can get you 200 ip phones for $20+ or so |
17:49.22 | BSDTech | call grandstream and strike up a deal |
17:49.23 | rene- | 35k |
17:49.26 | rene- | wheeew |
17:49.42 | a1fa | luisjose : i will cut you a good deal on ip phones |
17:49.45 | [TK]D-Fender | GrandSuck.... *shudder* |
17:49.47 | sweeper | b11d: yes I did, works awsomely now~ |
17:49.48 | BSDTech | the issue you will have is a operator console |
17:49.51 | a1fa | i take paypal ;) |
17:49.55 | rene- | a1fa |
17:49.58 | rene- | what model |
17:49.59 | rene- | make |
17:49.59 | luisjose | a1fa, good to know. |
17:50.06 | luisjose | a1fa, I'm in Venezuela. |
17:50.07 | rene- | have pictures and specs? |
17:50.12 | a1fa | ok no free shipping |
17:50.16 | a1fa | mama vuevo |
17:50.17 | luisjose | lol |
17:50.25 | luisjose | 8==D |
17:50.28 | a1fa | BT102 |
17:50.28 | BSDTech | I take paypal/visa/amex/discover/diners/left nut/first born |
17:50.33 | sweeper | it's huevo |
17:50.33 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
17:50.33 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
17:50.40 | luisjose | actualy is guevo :P |
17:50.45 | luisjose | actually* |
17:50.50 | sweeper | mmm |
17:50.59 | sweeper | I'm pretty sure it's huevo |
17:51.03 | luisjose | sweeper, on swearing context |
17:51.20 | sweeper | ecodeMP3: Junk at the beginning of frame e108bf2a |
17:51.34 | a1fa | ok 200 phones for $14k? |
17:51.34 | sweeper | well, you're still saying "egg sucker" |
17:51.37 | a1fa | sounds good? |
17:51.45 | sweeper | oops, damned paste buffer |
17:51.48 | a1fa | free shipping |
17:52.49 | a1fa | 200 BT101 for $14k + additional $1k for a configuration TFTP server |
17:52.49 | luisjose | sweeper, yap but you want to saw "di** sucker" |
17:52.54 | luisjose | say&** |
17:53.01 | BSDTech | ok the best setup for a hotel is the existing punchdown block and the sangoma a400 |
17:53.11 | BSDTech | you can have upto 4 in a systemn |
17:53.30 | `Sean | i want a 7985 :P |
17:53.31 | `Sean | from cisco |
17:53.32 | `Sean | lol |
17:53.34 | `Sean | :D |
17:53.36 | sweeper | wtf |
17:53.41 | Qwell[] | `Sean: buy me one, and I'll get it working with chan_skinny |
17:53.46 | BSDTech | sangoma a400 |
17:53.51 | `Sean | lol qwell |
17:53.54 | [TK]D-Fender | BSDTech: EW..... |
17:53.54 | sweeper | weird venezuelan slang :P |
17:53.58 | `Sean | why doesn't digium buy them for you :P? |
17:54.01 | Qwell[] | `Sean: That...wasn't a joke |
17:54.03 | ChicagoBud | BDSTech: are the grandstream 200's decent office phones? How is the speakerphone? |
17:54.06 | `Sean | digium will never cut support, for cisco |
17:54.26 | sweeper | woohoo, 66black |
17:54.26 | [TK]D-Fender | BSDTech: thats a horrid setup... SIP gateways! |
17:54.28 | sweeper | *black |
17:54.30 | sweeper | *BLOCK |
17:54.41 | `Sean | Qwell any idea how much they are? |
17:54.44 | ManxPower | a1fa: 200 BT101s don't actually cose $14k. They would cost you your job. |
17:54.49 | BSDTech | http://sangoma.com/datasheets/p_a400-specs |
17:54.50 | Qwell[] | `Sean: a lot :P |
17:54.53 | Qwell[] | > $2k |
17:54.55 | `Sean | lol |
17:54.56 | shepimport | hey--- anyone interested in helping me with some signalling issues? |
17:55.02 | luisjose | BSDTech, punchdown? |
17:55.02 | sweeper | well, depends. if you've got decent phones already, no point in replacing everything |
17:55.03 | ManxPower | You know that the BT101s can't even display non-numbers, right? |
17:55.05 | Qwell[] | MUCH > |
17:55.05 | `Sean | ya i see it |
17:55.08 | [TK]D-Fender | ManxPower: LOL.... don't get "smart" here.... its over most of their heads :) |
17:55.27 | *** join/#asterisk sahafeez (n=sahafeez@67.109.14.227.ptr.us.xo.net) |
17:55.31 | sweeper | if you're building new, then yea, run at least two lines of c5 to every room |
17:55.36 | BSDTech | its where the lines from the rooms meetup in the room with the pbx |
17:55.39 | sweeper | gigabit switch, the whole 9 yards :) |
17:55.44 | ManxPower | You have to press SEND or wait for the dialing to timeout on the BT101s, no dial pattern matching |
17:55.50 | Qwell[] | eww |
17:55.53 | Qwell[] | ManxPower: really? |
17:55.53 | [TK]D-Fender | BSDTech: See my not on Zaptel FXS. Poor deployment methodology. |
17:55.55 | `Sean | Qwell i dont think they have good encoding for the video, because it'd use alot of bandwidth |
17:56.06 | Qwell[] | `Sean: probably h263 or whatever |
17:56.15 | a1fa | ManxPower ? |
17:56.17 | ManxPower | Qwell[]: yes, really |
17:56.18 | `Sean | hrmp yea probably |
17:56.23 | [TK]D-Fender | sweeper: That'd have to be Cat5e then. |
17:56.28 | Qwell[] | cat6 |
17:56.32 | Qwell[] | ftw |
17:56.42 | Qwell[] | Run your TV over it also |
17:56.52 | Qwell[] | If you've got money to burn, might as well go all out |
17:56.58 | ManxPower | we will be putting in our first Gig Fiber link soon |
17:57.09 | sweeper | [TK]D-Fender: hard to find non-e c5 these days :P |
17:57.12 | Qwell[] | put a media system in every room, streaming on-demand video |
17:57.27 | sweeper | apple tv :D |
17:57.32 | BSDTech | ok |
17:57.39 | Qwell[] | get one of those stupid N64 controller remotes that nobody ever uses |
17:57.39 | BSDTech | TK looking |
17:57.46 | sweeper | hahaha |
17:57.50 | luisjose | BSDTech, where do you hook the pots on that card? |
17:57.54 | Qwell[] | seriously, who uses those? |
17:58.02 | ManxPower | installing Grandstream stuff is a good way to get fired. |
17:58.12 | sweeper | Qwell[]: I have, when company was footing the bill, and I was bored |
17:58.39 | BSDTech | there is a wireing diagram for it |
17:58.46 | BSDTech | that shows how |
17:59.06 | sweeper | chaaaaaannel bank |
17:59.09 | Qwell[] | is that the stupid one that uses a VGA connector or whatever? |
17:59.18 | sweeper | db13? |
17:59.20 | luisjose | BSDTech, on the PDF? |
17:59.26 | Qwell[] | sweeper: 15 |
17:59.33 | sweeper | orite |
17:59.43 | sweeper | sounds like a PITA |
17:59.50 | sweeper | t1 card + chan bank for me |
18:00.10 | sweeper | yay for hyooge oil services companies that don't mind spending bux to get the good stuff :) |
18:00.45 | [TK]D-Fender | sweeper: Meditrix 1124 + Amphenol punchdown block. |
18:00.48 | monsted | mmm, cisco gear ;) |
18:00.59 | Qwell[] | monsted: he said good stuff |
18:01.02 | sweeper | we use Adit600's |
18:01.06 | Qwell[] | not pointlessly expensive |
18:01.10 | luisjose | lol |
18:01.21 | monsted | Qwell[]: i quite like my catalyst 6500s :) |
18:01.25 | sweeper | monsted: cisco <> good stuff |
18:01.25 | BSDTech | yes |
18:01.29 | BSDTech | looking fo rit |
18:03.02 | sweeper | bah, mediatrix thing has no expandability :/ |
18:03.09 | sweeper | I can just add cards to an adit chassis |
18:03.44 | Qwell[] | sweeper: That's why the mediatrix is less expensive |
18:04.06 | [TK]D-Fender | sweeper: Instead you gain redundancy, mounting ease, easyier deployment, lower server load, and no need for a HUGE case to support those monster cards. Plus better call handling. |
18:04.54 | ManxPower | We use Adtran Total Access 750s |
18:05.13 | sweeper | monster cards? wha? |
18:05.31 | [TK]D-Fender | sweeper: "huge". |
18:05.44 | sweeper | have you SEEN an adit 600? |
18:05.51 | [TK]D-Fender | sweeper: You want a sime rack mount server? Hope it isn't 1-2U.... |
18:06.00 | [TK]D-Fender | simple/slim |
18:06.03 | luisjose | what is a punch block to wire analog phones? |
18:06.10 | sweeper | luisjose: yea, aka 66block |
18:06.22 | Qwell[] | luisjose: It's where the wires "terminate" |
18:06.25 | [TK]D-Fender | luisjose: Sounds like you need to learn a lot about phones in GENERAL |
18:06.31 | [TK]D-Fender | Strom_C: Link him! |
18:06.36 | Qwell[] | they go from the room to the punchdown block, then from there to...whereever |
18:06.38 | luisjose | [TK]D-Fender, yes, Im just starting. |
18:06.39 | sweeper | luisjose: http://www.wildtracks.cihost.com/homewire/phoneblk.jpg |
18:06.57 | BSDTech | in the us they have a plastic block with metel clips called a punch down block |
18:07.01 | [TK]D-Fender | sweeper: That'll do in the mean time. |
18:07.02 | Qwell[] | real cablers use punchdown blocks for cat3 for ethernet |
18:07.13 | Qwell[] | we had that at our old office when I worked for WF :D |
18:07.17 | [TK]D-Fender | Qwell[]: EW. |
18:07.21 | cpm | 66 or 110? |
18:07.23 | Qwell[] | "How come my jack doesn't work anymore?" |
18:07.23 | sweeper | it burns |
18:07.28 | Qwell[] | "It never worked...it's cat3" |
18:07.33 | Qwell[] | "uhh...no, it worked. fix it!" |
18:07.49 | Qwell[] | "okay, but it's cat3 and it terminates at a punchdown block" |
18:07.51 | [TK]D-Fender | Qwell[]: I got to completely GUT this new building I'm in and go Gigabit dual-LAN :) |
18:07.55 | BSDTech | I have done 4 a |
18:07.58 | BSDTech | sorry |
18:08.10 | luisjose | Ok i see is pretty much like a patch pannel but how it is connected to a DB25 port? |
18:08.12 | BSDTech | I have used 2 a400 with 4 doughtervoadrs |
18:08.21 | Qwell[] | luisjose: no, no, no |
18:08.26 | rene- | Are analog interfaces (channel bank) more reliable for office extensions than sip phones? |
18:08.32 | Qwell[] | a punchdown block is basically a patch panel, but it doesn't connect to a DB25... |
18:08.33 | BSDTech | I have used 2 a400 with 4 doughter boadrs giving 96 lines |
18:08.42 | Qwell[] | UNLESS you're doing something stupid, like using that ridiculous sangoma card |
18:08.48 | BSDTech | some do |
18:08.53 | BSDTech | some dont |
18:08.55 | Qwell[] | luisjose: sometimes they go to amphenol |
18:09.05 | Qwell[] | which is kinda like DB25, but it's not |
18:09.13 | luisjose | Qwell[], lol ok so you punch down the cables from the analog lines and it end as? |
18:09.21 | Qwell[] | it's a standard telcom jack (unlike the video card cable you need for the sangoma...*cough*) |
18:09.29 | BSDTech | ? |
18:09.32 | sweeper | luisjose: there is something called a "connectorized 66 block" |
18:09.34 | BSDTech | not for the a400 |
18:09.46 | sweeper | which has an amphenol connector, all wired into one side of the block |
18:09.48 | Qwell[] | BSDTech: maybe it's the a800 then... I don't honestly know sangoma models |
18:09.50 | BSDTech | but I wired my own jack |
18:10.04 | BSDTech | the a800 isfor ds3 |
18:10.07 | Qwell[] | I just know they have stupid connectors on some of their cards, heh |
18:10.20 | Qwell[] | BSDTech: their model numbers are just confusing |
18:10.25 | Qwell[] | which one is the 8 port analog? |
18:10.26 | sweeper | that amphenol connector is used to connect to a channel bank or other analog phone system |
18:10.40 | [TK]D-Fender | Amphenol is a longer Centronics connector. |
18:10.42 | Qwell[] | the TDM2400 uses amphenol |
18:10.43 | BSDTech | a400 uses 25 pin serisl/parallel type port |
18:10.59 | Qwell[] | BSDTech: Is that the 4 port analog, or quad T1? |
18:11.03 | [TK]D-Fender | BSDTech: ... no, not DB25... AMPHENOL.... |
18:11.04 | cpm | RJ-50 |
18:11.19 | luisjose | oh well i thought it was more simple, my home system is built with 2 SIPURA 2002 and 1 SIPURA 3000 |
18:11.19 | sweeper | http://www.voipsupply.com/product_info.php?products_id=569 <-- the adit 600 is cheeper |
18:11.24 | BSDTech | no its the 12 line ver of the a200 |
18:11.32 | ManxPower | Qwell[]: no more confusing that Digium's part numbers |
18:11.32 | Qwell[] | and the a200 is...? |
18:11.39 | [TK]D-Fender | OMG.. I'm mistaken! |
18:11.40 | luisjose | sweeper, let me see what is a 66block |
18:11.41 | ManxPower | 405, 415, 425? |
18:11.46 | BSDTech | a200 is the 4 port card |
18:11.47 | Qwell[] | ManxPower: come on now.. we have fairly standard stuff, and it actually makes sense |
18:11.51 | Qwell[] | BSDTech: 4 port what? |
18:12.02 | BSDTech | http://sangoma.com/datasheets/p_a400-specs |
18:12.06 | Qwell[] | see, their 4 port cards have a freaking 2 in the model number |
18:12.09 | BSDTech | thats the a400 |
18:12.12 | [TK]D-Fender | "Each 12 port A400 REMORA card is connected by means of a standard 12 line color coded telephone cable terminating at the card in a robust DB25 connector, and ready for hard wiring into a punch block at for the PSTN connection." |
18:12.21 | BSDTech | a200 is a 4port tdm fxs/fxo card |
18:12.21 | Qwell[] | and a 12 port card has a freaking 4 |
18:12.23 | [TK]D-Fender | z0mg! Retards! |
18:12.37 | ManxPower | Qwell[]: that is because (I think) the 4-port Sangoma T-1/E-1 cards are REALY a 2-port card with a 2-port daughter card |
18:12.41 | Qwell[] | ManxPower: surely, ours make at least a little sense :p |
18:12.43 | BSDTech | no the 12 port has a 25 pin connector |
18:12.45 | sweeper | luisjose: http://www.alliancesystems.com/Products/Images/Cables/L-CAB0550.jpg |
18:13.22 | sweeper | that black connector goes to your channel bank |
18:13.33 | sweeper | and then you punch down the phone line pairs on the other side |
18:13.42 | sweeper | and use bridge clips to connect/disconnect |
18:15.11 | [TK]D-Fender | Sangoma A400 = 12 port huge card w/o amphenol. FUGLY! |
18:15.15 | Qwell[] | ManxPower: Kenny explained the Sangoma model numbering scheme very well |
18:15.35 | Qwell[] | <kenny> Qwell: I think you have to adjust for canadian currency |
18:15.55 | [TK]D-Fender | Qwell[]: With GWB...you mean PAR ;) |
18:16.14 | Qwell[] | [TK]D-Fender: huh? |
18:16.20 | Qwell[] | ~par |
18:16.30 | jbot | Paragraph formatter for plain text. URL: http://www.cs.berkeley.edu/~amc/Par/ |
18:16.54 | BSDTech | so I like the a400 its a nice card has worked well |
18:16.54 | [TK]D-Fender | Qwell[]: Dubbaya sinking your currency's value to the point where the IS no conversion between currencies :) |
18:17.01 | Qwell[] | ahh :p |
18:17.07 | Qwell[] | figured PAR was an acronym |
18:17.12 | BSDTech | have had no issues |
18:17.31 | [TK]D-Fender | Qwell[]: No, I have no "bold" in this IRC client (that I've found), so I have to place emphasis with caps :) |
18:17.43 | Qwell[] | /par/ |
18:17.46 | Qwell[] | *par* |
18:17.55 | Qwell[] | <i>par</i> |
18:17.57 | [TK]D-Fender | Qwell[]: doesn't appear as such on my client... |
18:18.07 | Qwell[] | it's not supposed to ;) |
18:18.07 | [TK]D-Fender | Qwell[]: none of those are bolded here... |
18:18.13 | [TK]D-Fender | Qwell[]: :( |
18:18.19 | Qwell[] | but better than a non-TLA TLA |
18:18.21 | [TK]D-Fender | I miss mIRC sometimes... |
18:18.26 | Qwell[] | ~lart [TK]D-Fender |
18:18.32 | Qwell[] | ~lart [TK]D-Fender with mIRC |
18:18.42 | Qwell[] | That is an acceptable lart |
18:19.07 | [TK]D-Fender | Qwell[]: that should read "strips", not "stripes" :) |
18:19.21 | Qwell[] | [TK]D-Fender: red and flesh stripes |
18:19.22 | [TK]D-Fender | jbot: self-fornicate! |
18:19.43 | Qwell[] | but... |
18:19.46 | Qwell[] | ~lart [TK]D-Fender with mIRC |
18:20.03 | [TK]D-Fender | ~jbot |
18:20.06 | jbot | methinks jbot is only marginally useful at best, He got a C- on his Turing Test |
18:20.07 | [TK]D-Fender | ^^^^^ |
18:21.31 | tzanger | haha |
18:21.37 | tzanger | C- on his Turing test |
18:21.47 | *** part/#asterisk pythos (i=lanebob@unaffiliated/pythos) |
18:23.38 | BSDTech | to test on bsd |
18:23.43 | *** join/#asterisk naftali5 (n=naftali5@ool-44c05121.dyn.optonline.net) |
18:25.05 | luisjose | sweeper, and you hook it to a channel bank? |
18:25.10 | CrashSys | D-Fender: You, sir, are TEH MANG!!! |
18:25.22 | BSDTech | luisjose here look at this |
18:25.23 | *** join/#asterisk Ebola (n=Ebola@host81-152-204-157.range81-152.btcentralplus.com) |
18:25.25 | BSDTech | http://www.888voipstore.com/rhino-equipment-mid-21.html |
18:25.26 | luisjose | sweeper, and its identified by the order or pounching right? |
18:25.31 | [TK]D-Fender | ~[TK]D-Fender |
18:25.33 | jbot | [[tk]d-fender] rockin' the casbah !!! |
18:25.33 | BSDTech | those are channel banks |
18:25.35 | luisjose | BSDTech, checking/ |
18:26.06 | perd | boo yah |
18:26.35 | Hmmhesays | [TK]D-Fender: you ever used a macro to create an ivr menu? |
18:26.48 | CrashSys | Rhino's stuff on par with Sangoma? |
18:26.58 | [TK]D-Fender | Hmmhesays: Not a full IVR, but using Read, sorta. You shouldn't do that normally. |
18:27.16 | [TK]D-Fender | CrashSys: AVOID, and no... its super-sup-par |
18:27.19 | BSDTech | http://www.888voipstore.com/rhino-channel-bank-cb24-fxs-110v-pr-16391.html |
18:27.31 | CrashSys | sup or sub? |
18:27.34 | BSDTech | thats the one you whould use to put phones in the rooms |
18:27.39 | Hmmhesays | yeah I have a unique scenario here, I need an ivr based on entries in voicemail.conf and app directory |
18:28.19 | [TK]D-Fender | BSDTech: http://voipstore.atacomm.com/Shops/ViewItem.aspx/27934028032-29013176064.htm |
18:28.44 | [TK]D-Fender | BSDTech: Same price, don't need a T1 card, lower system load. No special cards. REDUNDANT. |
18:28.48 | rene- | there is the openvox12 port analog card |
18:28.51 | rene- | has anyone used it |
18:28.57 | rene- | openvoice sp? |
18:29.02 | [TK]D-Fender | Channel bank = only if you're cheap and got it as a GIFT |
18:29.09 | *** join/#asterisk ShadowHntr (i=sentinel@wikipedia/Shadowhntr) |
18:29.12 | [TK]D-Fender | rene-: More stuff tro avoid.... |
18:29.27 | b11d | . |
18:29.42 | [TK]D-Fender | Hmmhesays: Gimme some recordings! |
18:30.05 | rene- | D-Fender: what do you use: polycom? |
18:30.16 | Hmmhesays | [TK]D-Fender: recordings of what? |
18:30.26 | ManxPower | Channel banks are totally awesome! |
18:30.27 | [TK]D-Fender | Hmmhesays: Your stuff.... |
18:30.41 | BSDTech | ok cool |
18:30.46 | BSDTech | nice |
18:30.50 | ManxPower | ANYTIME I need analog, I go with a channel bank |
18:30.50 | [TK]D-Fender | Hmmhesays: Now that I'm feeling kinda funk leaning sweeps, soloing is just so much more fun :) |
18:31.27 | rene- | ManxPower on a green field pbx install do you go for analog or polys? |
18:31.32 | *** join/#asterisk osas (n=nnnnnnno@CABLE-72-53-75-244.cia.com) |
18:32.02 | luisjose | [TK]D-Fender, thats like a big fxs sipura? |
18:32.21 | [TK]D-Fender | luisjose: Yes, thats a close enough description. |
18:32.37 | luisjose | [TK]D-Fender, i like that solution |
18:32.40 | [TK]D-Fender | luisjose: 24 ports, uses a telecom standard Rj21 (amphenol) connector. |
18:32.40 | BSDTech | http://www.888voipstore.com/grandstream-gxp-2000ext-pr-17149.html would make the cheapest best front desk terminal for a hotel |
18:32.58 | b11d | grandstream really? |
18:33.04 | b11d | i didnt think they had ANY good use |
18:33.12 | BSDTech | but they need to make it with a shift button |
18:33.13 | [TK]D-Fender | b11d: KINDLING |
18:33.17 | danp | heh |
18:33.21 | b11d | haha |
18:33.32 | [TK]D-Fender | b11d: Get a good Charcoal filter first though... |
18:33.33 | *** join/#asterisk karmatronic (n=karmatro@84.77.152.248) |
18:33.37 | BSDTech | thats the only one I have seen with that many lines you can monitor |
18:33.40 | CrashSys | Channel Banks = The Bomb for hotels/call centers/etc... |
18:33.46 | BSDTech | yes |
18:33.46 | b11d | nah I like the plastic in the air |
18:33.50 | CrashSys | Suck for an office tho :) |
18:33.53 | BSDTech | but for the front desk |
18:33.56 | perd | you can get a nice amphenol -> 24 port patch panel from graybar |
18:34.08 | b11d | yeah i've got a few of those |
18:34.23 | BSDTech | but you need a front desk terminal |
18:34.35 | b11d | have one phone per line on the front desk |
18:34.37 | luisjose | [TK]D-Fender, and the other side it goes RJ45 to a switch/NIC right? |
18:34.37 | b11d | that'd be the best |
18:34.38 | b11d | :) |
18:34.47 | CrashSys | I use 66-blocks |
18:34.53 | CrashSys | I'm cheap :( |
18:35.03 | BSDTech | lol |
18:35.04 | luisjose | CrashSys, 66-blocks connected to? |
18:35.04 | b11d | i like the 66 over the 110 still |
18:35.08 | CrashSys | channel bank |
18:35.11 | AstaWerksDotCom | aastra has a new phone comming out in feb that has a side car for 20 extensions that would be good for a front desk |
18:35.17 | CrashSys | Amp 24-pair --> 66-block |
18:35.17 | luisjose | god damnt! |
18:35.19 | [TK]D-Fender | luisjose: Exactly. |
18:35.20 | luisjose | im a bit lost |
18:35.26 | [TK]D-Fender | luisjose: Decent web-admin on it. |
18:35.37 | CrashSys | 110 is all there is for Cat5 :( |
18:35.40 | BSDTech | but the grandstream has 56 on each |
18:35.44 | BSDTech | side cad |
18:35.47 | CrashSys | 110 seems flimsy |
18:35.52 | b11d | you *can* do cat5 on 66 blocks.. |
18:35.55 | b11d | i do it.. |
18:35.56 | luisjose | amphenol isnt 66-block? |
18:35.57 | b11d | it sucks |
18:36.01 | CrashSys | b11d: it sucks |
18:36.02 | CrashSys | err |
18:36.04 | b11d | i know |
18:36.04 | b11d | hehe |
18:36.13 | b11d | 110.. yeah flimsy is right.. |
18:36.17 | luisjose | [TK]D-Fender, so you have used it? |
18:36.20 | BSDTech | 112 |
18:36.22 | AstaWerksDotCom | web interface on grand sterams fail always |
18:36.23 | CrashSys | Amphenol 24 is a type of connector... |
18:36.35 | [TK]D-Fender | luisjose: Yup, works well. |
18:36.42 | CrashSys | I think it's actually a 25-pair connector |
18:36.44 | luisjose | BSDTech, lol that such an ugly phone. |
18:36.58 | luisjose | [TK]D-Fender, on normal or low load? |
18:37.00 | [TK]D-Fender | CrashSys: 25 pair, only 24 used. |
18:37.01 | BSDTech | sorry . |
18:37.09 | perd | http://pastebin.ca/314379 anyone see a problem with that configuration for a 7902 with chan_skinny ? |
18:37.13 | BSDTech | not every phone can look like cisco |
18:37.15 | CrashSys | I'd use a shitstream GXP2000 for a hotel/receptionist phone in a heartbeat... cheap, big, and cheap... |
18:37.23 | perd | for some reason it doesnt work and it causes my phone to act very strange |
18:37.27 | [TK]D-Fender | luisjose: normal. |
18:37.49 | [TK]D-Fender | luisjose: its industrial gear. It works, and works well. |
18:37.53 | BSDTech | the rev2's work great |
18:38.08 | *** join/#asterisk sbingner (n=sam@pdpc/supporter/sustaining/sbingner) |
18:38.15 | luisjose | Ok so im getting it slowly at least |
18:38.30 | [TK]D-Fender | luisjose: Good to hear. Progress = good |
18:38.51 | BSDTech | and the hotels I have done we used the gxp2000 as a desk phone with 2 side cars |
18:39.03 | luisjose | still dont get the link between the 66-block and channel banks. |
18:39.05 | BSDTech | being the hotels where only 100 rooms |
18:39.11 | BSDTech | <PROTECTED> |
18:39.14 | *** join/#asterisk endikos (n=endikos@gtlgateway.nmsu.edu) |
18:39.15 | *** join/#asterisk spatulamaan (n=ggilmore@207.188.8.252) |
18:39.17 | [TK]D-Fender | BSDTech: That store is kinda pricey |
18:39.22 | BSDTech | so it monitors them all |
18:39.44 | [TK]D-Fender | BSDTech: You know if you just want to monitor, use FOP or the like.... |
18:39.46 | CrashSys | Luisjose: Most phone companies will terminate POTS lines on a 66-block... it's a punchdown connector that you put raw wires into... |
18:40.28 | CrashSys | http://en.wikipedia.org/wiki/66_block |
18:40.28 | b11d | i had a bunch of 66 blocks actually "fail" on me once.. |
18:40.35 | b11d | all the metal rapidly oxidized for some reason |
18:40.40 | CrashSys | b11d: too many punches? salt water? |
18:40.50 | b11d | no.. no salt water.. and not too many punches.. |
18:40.51 | BSDTech | fop has issues I like hud |
18:40.54 | b11d | they just stopped working. |
18:41.00 | b11d | it didnt make ANY sense |
18:41.06 | CrashSys | Here in florida we get salt water issues a lot.. |
18:41.10 | Hmmhesays | making an ivr system with different entries per user just sucks |
18:41.12 | b11d | yeah I dont doubt that.. |
18:41.12 | BSDTech | and am now setting up hotels with a terminal running hudpro |
18:41.12 | [TK]D-Fender | BSDTech: Sure, whatever... |
18:41.13 | CrashSys | specially near the beach... |
18:41.49 | CrashSys | b11d: I've used a die-electric aerosol spray and actually sprayed the blocks down before if it looks like humidity is an issue... |
18:41.59 | b11d | hrm, thats cool.. |
18:42.00 | CrashSys | Sure the block is a lil sticky, but it wont effect nothing... |
18:42.03 | b11d | yeah |
18:42.14 | CrashSys | Easier then swapping a block out... |
18:42.24 | b11d | yeah it was a huge pain in the ass |
18:42.35 | BSDTech | and I thoght FOP had issues with more then 40 exten |
18:43.29 | BSDTech | ok 1.4 is comppiling |
18:43.33 | BSDTech | kon freebsd |
18:43.46 | CrashSys | drumroll |
18:44.36 | BSDTech | Istill think we wll have to do some patching |
18:44.41 | BSDTech | like we did on 1.2 |
18:44.45 | BSDTech | allthe time |
18:45.04 | BSDTech | then I can get freepbx to work |
18:45.12 | BSDTech | and Iwill have a rocking pbx |
18:45.33 | BSDTech | might look at the asterisk gui when the get it working |
18:46.01 | BSDTech | ok its installed |
18:46.10 | BSDTech | now to test it with g729 |
18:46.29 | BSDTech | I love my dual p3 1.2 ghz |
18:46.35 | BSDTech | 1 gig ram |
18:46.49 | b11d | heh. |
18:46.54 | b11d | i have no issues with asterisk on FreeBSD at all |
18:46.58 | b11d | i heart it |
18:47.04 | BSDTech | ok looks like transcoding is fixed |
18:47.19 | BSDTech | so 1,4 svn works |
18:47.29 | BSDTech | cool |
18:47.46 | ChicagoBud | <luisjose> you terminate the lines from each room to a 66-block. Then you conecct the channel bank to the 66-block via a cable with the AMP connectors |
18:48.35 | BSDTech | ok I am happy |
18:48.50 | BSDTech | now I am create a 1.4 port |
18:48.52 | b11d | well this is a cause for felicitations! |
18:48.59 | BSDTech | but zaptel is still in the works |
18:49.09 | b11d | wtf is with zap? it works fine for me |
18:49.11 | b11d | on 6.2-PRE |
18:49.21 | ChicagoBud | <luisjose> LAN < 1 - 1> CB <1 - 1> 66 <1 - 24> |
18:49.31 | BSDTech | ? |
18:49.43 | BSDTech | we are porting 1.4 zatel |
18:49.53 | BSDTech | right nwo only the 1.2 is in ports |
18:49.58 | b11d | oh yeah I dont use the ports.. |
18:50.04 | b11d | cool, i see what you mean |
18:50.09 | BSDTech | but libpri and freebpx should work now |
18:50.26 | b11d | cool |
18:50.26 | BSDTech | how did you get zaptel 1.4 to compile |
18:50.32 | b11d | i just compiled it? |
18:50.33 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
18:50.33 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
18:50.34 | BSDTech | its ment for linux |
18:50.43 | b11d | i used zaptel-bsd-trunk |
18:50.53 | BSDTech | thats whats in ports |
18:50.54 | b11d | from svn |
18:51.14 | BSDTech | from digium or the asterisk-bsd dev group |
18:51.21 | b11d | from the svn.pbxpress.com |
18:51.33 | ChicagoBud | <BSDTech> Is the Grandstream Budgetone 200 a decent office phone for a 10 user system? How is the speaker phone on them? |
18:51.40 | ManxPower | can anyone recommend a SIMPLE and free .WAV file editor for Windows? |
18:51.43 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
18:51.58 | b11d | Audacity is the best |
18:51.58 | b11d | imho |
18:51.58 | CrashSys | I've used the BT-200 |
18:52.01 | Qwell[] | ManxPower: sndrecorder |
18:52.09 | CrashSys | If money is an issue it will suffice... |
18:52.09 | Qwell[] | comes with windows since like 3.1 :p |
18:52.11 | b11d | sndrec32 |
18:52.14 | Qwell[] | whatever |
18:52.30 | BSDTech | b11d svn sting pls |
18:52.40 | b11d | ? |
18:52.44 | AstaWerksDotCom | goldwave.com |
18:52.56 | ManxPower | ChicagoBud: Grandstream has a history (years) or terrible firmware on somewhat crappy hardware. I would not give one to my ex-wife. I would give my ex-wife cyanide, but not a grandstream product. |
18:52.57 | AstaWerksDotCom | free sound player called goldwave there |
18:53.03 | AstaWerksDotCom | editor |
18:53.12 | BSDTech | the zaptel-bsd-trunk |
18:53.14 | ChicagoBud | <ManxPower>Cool Edit used to be really good but Adobe owns it now -- not sure if it is free anymore |
18:53.20 | BSDTech | Idont find the link |
18:53.22 | ManxPower | Qwell[]: sndrecorder is what comes with Windows? |
18:53.27 | b11d | hang on |
18:53.35 | Qwell[] | sndrec32 |
18:53.39 | CrashSys | Just write a record macro in * |
18:53.40 | CrashSys | call it |
18:53.41 | CrashSys | done :D |
18:53.47 | [TK]D-Fender | ManxPower: Audacity rocks... |
18:53.56 | [TK]D-Fender | ManxPower: http://audacity.sourceforge.net/ |
18:54.04 | ChicagoBud | Is there a decent under $100 phone that is generally recommeded for small offices? |
18:54.19 | AstaWerksDotCom | aastra 9112 borders 100 its a great phone |
18:54.33 | CrashSys | Chicago: You need speakerphone? |
18:54.35 | [TK]D-Fender | ChicagoBud: None. |
18:54.44 | CrashSys | Polycom IP301's for $115 |
18:54.46 | ChicagoBud | yes on the speakerphone |
18:54.48 | CrashSys | no speakerphone tho |
18:54.56 | AstaWerksDotCom | 9112 has speaker |
18:55.00 | ChicagoBud | $125? |
18:55.01 | [TK]D-Fender | ChicagoBud: Do you have or are planning on getting PoE? |
18:55.16 | ManxPower | Qwell[]: Thanks! sndrec32 was just what I needed. |
18:55.19 | ManxPower | just barely |
18:55.22 | Qwell[] | heh |
18:55.28 | ChicagoBud | Do not have POE but would consider it but |
18:55.29 | AstaWerksDotCom | $107 on my website but can get it down to $100 for you |
18:55.30 | Qwell[] | it's simple, but it works |
18:55.32 | b11d | bastard |
18:55.36 | b11d | I said sndrec32 |
18:55.36 | b11d | :) |
18:55.53 | b11d | http://www.pbxpress.com/~gonzo/zaptel-bsd-trunk.tar.gz |
18:55.57 | b11d | there it is |
18:56.02 | *** join/#asterisk olinux (n=olinux@starbucks.wellspublishing.net) |
18:56.03 | b11d | from here: i |
18:56.04 | b11d | doh |
18:56.07 | *** join/#asterisk sbingner (n=sam@pdpc/supporter/sustaining/sbingner) |
18:56.10 | b11d | http://www.voip-info.org/tiki-index.php?page=FreeBSD+zaptel |
18:56.25 | b11d | got the tarball, svn'd myself an update, and that was it |
18:56.57 | *** join/#asterisk Dr-Linux (n=Nothing@202.125.139.198) |
18:57.04 | luisjose | ChicagoBud, so is a standar connector that use 66block and channel banks |
18:57.19 | Dr-Linux | why i'm facing this problem queue module? >> ivr1*CLI> reload app_queue.so |
18:57.19 | Dr-Linux | The previous reload command didn't finish yet |
18:57.44 | BSDTech | ok thats from the asterisk-bsd dev group I am apart of and thats 1.2.4 zaptel |
18:57.49 | BSDTech | it needs updating |
18:58.28 | ChicagoBud | <luisjose> most channelbanks send out the 24 analog lines via an AMP connector (25-pair cable) |
18:58.29 | BSDTech | and we are working to move it to 1.2.12 I think is the current zap and 1.4 for zap |
18:58.29 | Qwell[] | Dr-Linux: how long ago did you reload app_queue.so previously? |
18:58.31 | BSDTech | 2 ports |
18:58.42 | BSDTech | there will be 2 branches of each |
18:58.46 | b11d | ohh |
18:58.47 | b11d | ok |
18:58.48 | BSDTech | in the ports tree |
18:58.55 | Dr-Linux | Qwell[]: 2 minutes ago |
18:58.57 | ChicagoBud | <AstaWerksDotCom>looking now |
18:58.58 | BSDTech | shower time brb |
18:58.59 | [TK]D-Fender | ChicagoBud: If you really need to save $, then Polycom IP 430. Supports PoE, and has all the features your normal users should need @ $150/ea |
18:59.00 | b11d | i've got to run.. |
18:59.00 | b11d | ttyl |
18:59.08 | [TK]D-Fender | ChicagoBud: Possibly less |
18:59.09 | Dr-Linux | Qwell[]: bcoz queue application was not answering :S |
18:59.23 | CrashSys | IP430 = Good shiznit |
18:59.25 | Dr-Linux | Qwell[]: i can't unload the module as well |
18:59.30 | CrashSys | not a receptionists phone but good stuff |
18:59.34 | Dr-Linux | ivr1*CLI> unload app_queue.so |
18:59.34 | Dr-Linux | Unable to unload resource app_queue.so |
18:59.34 | Dr-Linux | Jan 12 10:13:23 WARNING[5616]: loader.c:135 ast_unload_resource: Soft unload failed, 'app_queue.so' has use count 9 |
18:59.47 | ChicagoBud | <[TK]D-Fender>I'll take a look. Thanks. |
18:59.57 | [TK]D-Fender | ChicagoBud: www.telephonydepot.com |
19:00.12 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
19:00.28 | Dr-Linux | Qwell[]: any idea why? |
19:00.45 | Dr-Linux | or suggestions |
19:01.55 | Dr-Linux | should i restart the asterisk? |
19:02.06 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
19:02.29 | perd | is chan_skinny just broken or am i doing something wrong? it doesnt appear to be working in 1.4 at all |
19:02.37 | L|NUX | hello |
19:02.41 | perd | and that seems odd, since im pretty sure people use it :) |
19:02.48 | Qwell[] | perd: it's your phone |
19:02.54 | CrashSys | People should just avoid cisco |
19:03.00 | Dr-Linux | Qwell[]: no clue? :) |
19:03.04 | perd | qwell they all do it though, the 7902, 7912 and 7960 |
19:03.10 | Qwell[] | 7960 should work |
19:03.18 | perd | but the other two shouldnt? |
19:03.19 | L|NUX | i am getting following error on astersik 1.2.10 launch_script: Unable to create toast pipe: Too many open files |
19:03.20 | CrashSys | It's like the sony of the networking world... it's everywhere, but only plays nice with other cisco stuff... ) |
19:03.22 | CrashSys | :) |
19:03.23 | *** join/#asterisk WAudette (n=WAudette@c-71-237-146-239.hsd1.or.comcast.net) |
19:03.23 | Dr-Linux | how can i kill the app_queue.so module |
19:03.26 | Qwell[] | perd: the other two are completely untested |
19:03.29 | L|NUX | can some one tell me how can i fix this issue ? |
19:03.30 | perd | ah |
19:03.30 | Qwell[] | (by me) |
19:03.44 | Qwell[] | Dr-Linux: Do you have any queues running? |
19:03.49 | perd | well, if you want more dumps let me know, anything i can do to help you make it work :) |
19:04.05 | luisjose | BSDTech, did you test the gui on freebsd? |
19:04.06 | Qwell[] | perd: eventually I'd like to try to get "feature dumps" |
19:04.18 | Qwell[] | ie; a dump of it putting a call on hold, and doing conferencing, stuff like that |
19:04.22 | perd | sure |
19:04.30 | Dr-Linux | Qwell[]: ofcos i'm using queues for our callcenter |
19:04.48 | Qwell[] | Dr-Linux: then you can't unload it if queues are running |
19:05.06 | Dr-Linux | Qwell[]: ? |
19:05.17 | Dr-Linux | Qwell[]: "show channel" shows there is nothing |
19:05.38 | CrashSys | kill -9 will unload it... :) |
19:06.06 | CrashSys | They should let me drink at work... i'd be so much more productive... |
19:08.45 | BSDTech | not yet |
19:08.49 | BSDTech | its in the plans |
19:09.00 | BSDTech | but I know it will take some patching |
19:09.15 | BSDTech | mostlikly |
19:09.29 | perd | qwell for feature dumps do you want two of the same type of phones or does it matter? |
19:09.44 | Qwell[] | huh? |
19:09.47 | perd | not matter that is |
19:11.01 | perd | nm |
19:11.08 | BSDTech | I like the freepbx interface but it is database driven |
19:11.37 | BSDTech | and i like the 3rd lane interface |
19:11.44 | BSDTech | its more robust |
19:11.52 | BSDTech | but they all have issues |
19:11.55 | BSDTech | on bsd |
19:11.56 | *** part/#asterisk reber (i=reber@gateway/tor/x-a6716cbe69f64a14) |
19:12.36 | Dr-Linux | Qwell[]: i restarted the asterisk, and everything is fine now |
19:13.04 | Dr-Linux | Qwell[]: but not sure what was the problem with app_queue.so module |
19:13.07 | *** part/#asterisk WAudette (n=WAudette@c-71-237-146-239.hsd1.or.comcast.net) |
19:13.09 | *** join/#asterisk WAudette (n=WAudette@c-71-237-146-239.hsd1.or.comcast.net) |
19:13.12 | *** part/#asterisk WAudette (n=WAudette@c-71-237-146-239.hsd1.or.comcast.net) |
19:13.18 | Dr-Linux | and how can i avoid this problem in future |
19:20.55 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
19:28.58 | *** join/#asterisk McGee (n=rootgauf@p54957728.dip.t-dialin.net) |
19:29.38 | McGee | Hi, what is the status of call rerouting on T-Com PTP ISDN? Is it possible or is there still codeing work to be done? |
19:30.17 | *** join/#asterisk hohum (n=dcorbe@mercury.sunrocket.com) |
19:31.14 | *** join/#asterisk zapp-branigan (n=zapp-bra@81-202-140-56.user.ono.com) |
19:34.55 | *** join/#asterisk bkw__ (n=brian@ip68-0-120-100.tu.ok.cox.net) |
19:34.56 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
19:39.46 | *** join/#asterisk lters (n=tech@mrtcdsl-433.mis.net) |
19:45.16 | *** join/#asterisk pif (n=ldm@zenon.apartia.fr) |
19:45.46 | *** join/#asterisk jm|laptop (n=jamie@dilbert.jamiem.com) |
19:46.23 | *** join/#asterisk javar (n=javar@69.79.134.24) |
19:47.33 | Marty-OTT | I just received a Dul port T1 card from Sangoma - going to put it in my Asterisk box... :P |
19:47.43 | Marty-OTT | It's MY FIRST TIME.. I;m a VIRGIN at this |
19:47.53 | *** join/#asterisk bprice20 (n=brandon@216.120.224.199) |
19:48.02 | *** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net) |
19:48.23 | Marty-OTT | Anyone has any hints about when my FreeBSD box boots back up with the Sangoma card in it? |
19:48.53 | pif | sangoma is not supported on freebsd |
19:50.01 | mercestes | pif: Wow, what a mood kill |
19:50.04 | luisjose | lol |
19:50.17 | bprice20 | Has anyone gotten res_snmp under 1.4.0 to toss out number of registered sip users |
19:50.33 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
19:50.34 | pif | good cherry popping for a virgin, hey? :) |
19:50.35 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
19:50.39 | bprice20 | I can get other statistics out of it like number of sip channels |
19:51.10 | bprice20 | just not number of registrations; I think I need to right snmp OID |
19:51.12 | mercestes | pif: He didn't even get to take his clothes off. It's like, taking yoru girlfriend to a hotel room, and she whispers "i'm a virgin." and you get in there...and *all* the rooms are chaparoned. |
19:51.24 | mercestes | and their's no privacy anywhere. |
19:51.33 | pif | dude |
19:51.58 | mercestes | yea |
19:52.16 | mercestes | worst nightmare I ever had. |
19:52.21 | ManxPower | Is it "newbies answer questions they know nothing about day" on asterisk-users? |
19:52.38 | mercestes | ManxPower: Must be, you're here. |
19:52.48 | mercestes | :P |
19:52.56 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
19:54.03 | [TK]D-Fender | :O |
19:54.17 | pif | dunno, just reading "my freebsd box" made my debian juices flow |
19:54.32 | ManxPower | Wheres a gun toting Republican like JerJer when you need one? |
19:54.41 | tzanger | hahaha |
19:55.32 | *** join/#asterisk GiantPickle (n=GiantPic@S0106006008bd147d.gv.shawcable.net) |
19:55.57 | Marty-OTT | pif: As far as I;m reading here - yes it is |
19:56.10 | Marty-OTT | I checked this out with the Sangoma themselves before buying |
19:56.31 | *** join/#asterisk cmdln (n=cmdln@64-126-105-14.static.everestkc.net) |
19:56.33 | Marty-OTT | I'm reading the installation instructions under FreeBSD |
19:56.38 | *** part/#asterisk cmdln (n=cmdln@64-126-105-14.static.everestkc.net) |
19:56.42 | pif | sure, I just wanted to give a friday night adrenaline rush |
19:56.49 | pif | to a virgin |
19:56.54 | Marty-OTT | :P ... well done... lol |
19:57.09 | Marty-OTT | was concerned someone had issues with Sangoma card(s) |
19:57.26 | pif | you should install linux nevertheless |
19:57.39 | pif | get rid of that crap *bsd |
19:58.15 | ManxPower | Marty-OTT: so few people use Asterisk on anything except linux just assume you are on your own with *MSD |
19:58.57 | ManxPower | Just think of yourself as so "special" nobody will be able to help you |
20:00.07 | CrashSys | Even if you were to use Linux so few people would want to help you that you would still feel special :) |
20:00.20 | pif | lol |
20:00.24 | CrashSys | Most people will just quote you their hourly rate |
20:01.09 | [TK]D-Fender | Quick stuff I offer free. Full setups I charge cheaply for :) |
20:01.22 | cpm | btw |
20:01.23 | ManxPower | I have several 2-port Sangoma cards on Linux with Asterisk |
20:01.24 | cpm | sip sucks |
20:01.34 | [TK]D-Fender | just yesterday did a 7-phone Polycom / Sangoma setup from scratch. |
20:01.36 | ManxPower | cpm: heretic |
20:02.06 | *** join/#asterisk h0 (n=fakhir@unaffiliated/fakhir) |
20:02.15 | pif | I've been more lucky with sip to trunk my asterisks than with iax2 |
20:02.29 | pif | (random UNREACHABLE crap) |
20:02.40 | cpm | How to use sip in the *real world* 1, set up asterisk, set up a sip device, go on the road, realize you need another pocket asterisk box of some kind running iax to tunnel back to your main pbx, because, , , SIP SUCKS |
20:02.45 | ManxPower | pif: me too. |
20:03.28 | ManxPower | cpm: Um, Before Katrina I roamed between variaous nat and non-nat networks just fine and connected back to my Asterisk server, also behind nat. |
20:03.35 | ManxPower | no config changes needed at all |
20:03.46 | ManxPower | Using a SIPura |
20:03.49 | cpm | cool, glad it works well for you. |
20:04.00 | cpm | Ahh, Sipura, I hear good things there |
20:04.03 | ManxPower | Anyone that can't roam with a SIP device doesn't have things set up correctly. |
20:04.17 | cpm | tell that to Polycom |
20:04.34 | ManxPower | cpm: Polycoms should do it just fine too. |
20:04.43 | ManxPower | I had NO NAT config stuff on the SIPura |
20:04.54 | cpm | prolly 70 percent of the time |
20:04.57 | *** join/#asterisk Strom_M (n=strom@netblock-66-159-243-60.dslextreme.com) |
20:05.17 | cpm | end up fireing that hoplessly awful kiax, and bingo! I can connect. |
20:05.43 | *** join/#asterisk UlbabraB (n=salama@host241-43-static.72-81-b.business.telecomitalia.it) |
20:05.46 | cpm | not that the kiax folks are doing anything wrong, I'm grateful for that softphone. Because it pretty m uch always works, when my polycom won't. |
20:06.11 | *** join/#asterisk Chris-NB (n=chris@argos.campus-sbg.at) |
20:06.40 | cpm | I should get some sipura stuff. |
20:06.48 | ManxPower | It sucks to be you |
20:06.50 | luisjose | sipura is ok for a house |
20:06.59 | cpm | Not really. |
20:07.06 | robl^ | new polycom firmware has NAT-friendly features.. even NAT keepalive. |
20:07.20 | pif | my spa-3000 won't relay dtfm tones to a fucking door phone |
20:07.33 | luisjose | pif, how come? |
20:07.44 | pif | and it's configured "inband" all the way |
20:07.45 | cpm | good thing that relaying to door knobs isn't often needed |
20:07.53 | luisjose | door phone? |
20:07.54 | ManxPower | pif: How do you know that? |
20:07.59 | [TK]D-Fender | pif: use "INFO" |
20:08.03 | ManxPower | pif: put a phone on it and see if you hear the tones |
20:08.12 | pif | good thought |
20:08.20 | ManxPower | pif: it should have been your FIRST thought. |
20:08.26 | pif | it's a remote install |
20:08.32 | *** join/#asterisk s1gny|wrk (n=s1gny@p54914E21.dip.t-dialin.net) |
20:08.36 | pif | must use an employee to test |
20:08.53 | pif | been bugging these guys all day :) |
20:09.01 | *** part/#asterisk s1gny|wrk (n=s1gny@p54914E21.dip.t-dialin.net) |
20:10.02 | [TK]D-Fender | pif: Then it should come as no surprise to them ;) |
20:10.57 | pif | [TK]D-Fender : why info? |
20:11.04 | *** part/#asterisk bprice20 (n=brandon@216.120.224.199) |
20:11.21 | pif | I want the sound to goto the door phone in order to unlock the door |
20:11.25 | [TK]D-Fender | pif: Will sed DTMF between the ATA & * OOB, but reconstruct clean on each side. |
20:11.35 | pif | hmmm |
20:11.44 | [TK]D-Fender | pif: I'm betting its distorting |
20:12.03 | ManxPower | [TK]D-Fender: I say it's a volume issue. The tones are not loud enough. |
20:12.28 | [TK]D-Fender | ManxPower: Could very well be, and going OOB will stop gain from being a normal issue hopefully. |
20:12.40 | [TK]D-Fender | ManxPower: Gain of the standard voice channel anyways |
20:12.49 | pif | if INFO == rfc2833 ? |
20:12.53 | pif | s/if/is |
20:12.55 | ManxPower | [TK]D-Fender: yes, but OOB would make the tones pretty short, perhaps too short for the door device to work |
20:12.58 | ManxPower | pif: NO! |
20:13.08 | ManxPower | INFO == INFO. RFC2833 == RFC2833 |
20:13.24 | [TK]D-Fender | 1 + 1 = 3! |
20:13.38 | pif | there's no INFO on the spa-3000 |
20:13.43 | pif | scrap that |
20:13.52 | ManxPower | pif: they call it something else I'm sure. |
20:13.56 | pif | there's no rfc2833 on the spa-3000 |
20:14.01 | ManxPower | AVT is what they call RFC2833 |
20:14.07 | pif | yes! |
20:14.18 | pif | thanks |
20:14.34 | pif | what does avt stand for? |
20:14.46 | ManxPower | ask SIPura. |
20:14.55 | pif | sure ;) |
20:15.09 | ManxPower | I always set mine to AVT in the SIPura and set it to RFC2833 in Asterisk and NEVER EVER had a problem |
20:15.29 | pif | have tried that, will do, thanks again |
20:15.38 | pif | s/have/haven't |
20:16.07 | *** join/#asterisk shidan (n=chatzill@CPE0013107d30c4-CM001371871af0.cpe.net.cable.rogers.com) |
20:16.22 | naitram | is there a way to concat two strings like System(1ststring . ${VAR1}), does that work, the dot is a php concat |
20:16.51 | ManxPower | pif: you are wasting time. Get the box into a a test enviroment and play with the settings |
20:17.14 | ManxPower | naitram: in what l;anguage / enviroment" |
20:17.31 | ManxPower | SetVar(FNORD=${BOB}${DOBBS}) |
20:17.38 | pif | I'm fucking stupid, I let go my last unit, and now I must rely on strangers to test |
20:18.36 | naitram | ManxPower: in the dial plan scripts exten => 101,1, System(/var/www/myscript.php 1stargument 2nd 3rd . ${MYDIALPLANVARIABLE}) |
20:19.36 | ManxPower | <PROTECTED> |
20:19.53 | ManxPower | remember variables are substituted first thing |
20:20.25 | naitram | ManxPower: oh, ok. Thanks for the help. New to asterisk. |
20:20.39 | ManxPower | in your example you have 5 args, where "." is the 4th arg |
20:22.05 | naitram | ManxPower: Gotcha, thanks |
20:22.49 | pif | hmm, I see "DTMF Playback Level: -16" could that be the cluprit? |
20:23.13 | pif | and "DTMF Playback Length: .1" |
20:23.15 | *** join/#asterisk Defraz (n=t0tal@24-116-159-197.cpe.cableone.net) |
20:24.11 | ManxPower | Since we don't know what volume level or tone length your "door phone" requires, you'll just have to try it. |
20:24.31 | ManxPower | your EAR is the best thing to test with, so get your ass over to the client and try it. |
20:25.29 | pif | yep |
20:26.15 | sweeper | hmm |
20:28.01 | sweeper | time to find out if 1ghz/512mb can handle 13 channels sans transcoding |
20:28.18 | sweeper | *23 |
20:28.34 | Strom_M | probably |
20:28.56 | [TK]D-Fender | sweeper: Should have no issue |
20:29.03 | sweeper | yay |
20:30.05 | ManxPower | sweeper: I have a 1.8Ghz machine that handles 96 channels with only a small amount of transcoding |
20:30.42 | sweeper | cool \o |
20:31.10 | ManxPower | for various reasons, assume only 24 channels are ever in use at the same time |
20:31.44 | *** join/#asterisk s1gny|wrk (n=s1gny@p54914E21.dip.t-dialin.net) |
20:32.05 | *** part/#asterisk s1gny|wrk (n=s1gny@p54914E21.dip.t-dialin.net) |
20:32.32 | *** join/#asterisk Qwell[] (i=qwell@unaffiliated/qwell) |
20:32.32 | *** mode/#asterisk [+o Qwell[]] by ChanServ |
20:33.00 | CrashSys | I'm glad I haven't had to worry about transcoding yet :) |
20:34.55 | ManxPower | CrashSys: we don't do a lot of VOIP outside the LAN |
20:36.39 | cbullock81 | hey D-Fender |
20:36.53 | cbullock81 | can i hit you up one more time about polycom phones? |
20:38.17 | cbullock81 | I'm trying to setup Busy Lamp Field for the polycom ip650 phones, and i must be overlooking something. anyone have any experience with this? |
20:39.37 | wunderkin | on text:ip650:/ignore $nick |
20:39.38 | wunderkin | ;p |
20:42.56 | cbullock81 | what was that for? |
20:43.14 | Qwell[] | he's jealous :P |
20:43.18 | cbullock81 | heh |
20:43.19 | CrashSys | Envy |
20:43.33 | wunderkin | enV |
20:43.51 | cbullock81 | have any of you used BLF with polycom before? |
20:44.22 | danp | for messages or "buddies"? |
20:45.18 | cbullock81 | i guess for buddies... basically i just want it to show if the extension is in use |
20:45.23 | danp | yeah |
20:45.33 | BSDTech | no but I have on the gxp2000 |
20:45.42 | BSDTech | thats simple |
20:45.47 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
20:45.47 | *** mode/#asterisk [+o denon] by ChanServ |
20:45.59 | danp | you need to turn messaging on in your phone config, add hints in asterisk and add the things you want to watch to the directory on the phone with buddy watching turned on |
20:46.57 | cbullock81 | about adding hints... i did something like this "exten => 101,hint,SIP/101" is that correct? |
20:47.14 | danp | yep |
20:47.33 | cbullock81 | ok. then how do you tell the phone to subscribe to that hint (or however you word it) |
20:47.53 | danp | on the phone itself, hit the directories button and then go into the contact directory |
20:48.13 | danp | the contact field will be your hint name (101). be sure to turn buddy watching on near the bottom |
20:48.28 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
20:48.33 | *** join/#asterisk linlin (i=will@c-71-194-70-13.hsd1.il.comcast.net) |
20:48.38 | danp | but that only shows up if you've turned on messaging...which you can do via the XML configs. not sure if you can do that via the web interface |
20:48.47 | naitram | tyring this; exten => 101,1,Set(MEMAREA=6800 101) ; getting this error No application 'Set' for extension (sip, 101, 1). |
20:49.01 | BSDTech | ok I am building my new freebsd/asterisk server now |
20:49.12 | BSDTech | and I will use the thirdlane interface |
20:49.14 | naitram | is this not how to set a variable? |
20:49.23 | BSDTech | wich I thinkis what digium should use |
20:49.41 | cbullock81 | danp: i'm about to enable messaging and see what i can come up with. i appreciate the help. i might have to hit you up again in a bit if thats ok :) |
20:50.01 | ManxPower | naitram: what version of Asterisk? |
20:50.16 | danp | cbullock81: sure! it's pretty easy once you get messaging turned on |
20:50.33 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
20:50.33 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
20:50.33 | CrashSys | 2.0! |
20:50.44 | naitram | ManxPower: 1.0.7-BRI |
20:50.52 | Strom_M | *blink* |
20:50.57 | Strom_M | 1.0.7 |
20:50.58 | ManxPower | naitram: set is not valid in 1.x, use SetVat |
20:51.05 | ManxPower | setvar |
20:51.09 | BSDTech | the phone does the blf |
20:51.20 | *** part/#asterisk shidan (n=chatzill@CPE0013107d30c4-CM001371871af0.cpe.net.cable.rogers.com) |
20:51.22 | BSDTech | you just need to put the name and exten number |
20:51.30 | BSDTech | and it should work |
20:51.33 | naitram | ManxPower: ok, what is the latest stable version of Ast |
20:51.45 | BSDTech | it does in the grandstreams |
20:52.07 | danp | BSDTech: it's basically the same with the polycoms but there's a little extra setup to do |
20:52.18 | BSDTech | ahh ok |
20:52.27 | CrashSys | pif: Good luck |
20:52.30 | CrashSys | Wifi = Suck |
20:52.32 | [TK]D-Fender | pif: Try 2010. |
20:52.37 | pif | i know |
20:52.42 | pif | 2010? |
20:52.54 | BSDTech | I wish the cisco wifi phone was sip but its not |
20:53.01 | BSDTech | its skinny sccp based |
20:53.22 | CrashSys | Other then Aastra are there any good cordless (not wifi) phones? |
20:53.24 | pif | the 7920 ? |
20:53.30 | [TK]D-Fender | pif: Yeah, I figure maybe another 3 years or so they'll get it right :D |
20:53.45 | pif | it blows chunks (sold mine on ebay :) |
20:54.06 | De_Mon | Sweet! bleach 110 is out |
20:54.17 | De_Mon | oops, wrong channel/network |
20:54.34 | sweeper | just a bit |
20:54.38 | CrashSys | lol |
20:54.40 | pif | De_Mon : /join #teenporn |
20:55.12 | CrashSys | The sad part is I bet atleast 25% of this channel knows what he is talking about :) |
20:56.52 | BSDTech | ok I like the new thirdlane interface |
20:56.57 | De_Mon | ? teenporn since when was bleach considered teen porn? |
20:57.06 | BSDTech | its much better then the asterisk gui |
20:57.53 | De_Mon | the wifi cisco phone has a better gui than the asterisk? |
20:57.57 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
20:58.20 | BSDTech | no |
20:58.26 | danp | why does the thirdlane pbx manager have a link to google in the menu |
20:58.37 | CrashSys | $$$? |
20:59.03 | sweeper | ok |
20:59.04 | *** join/#asterisk [hC] (n=hardcore@206.108.27.93) |
20:59.13 | sweeper | why the FUCK does * use awk in the build process? |
20:59.35 | *** join/#asterisk airjump (n=airjump@p508AD998.dip.t-dialin.net) |
20:59.39 | mog | why not ? |
20:59.56 | De_Mon | because awk is better than sed |
20:59.57 | De_Mon | :P |
21:00.05 | sweeper | also, why does it have a HARD LINK to awk |
21:00.25 | sweeper | that's just fucking stupid |
21:00.26 | *** part/#asterisk mtaht4 (n=m@c-71-198-23-124.hsd1.ca.comcast.net) |
21:00.45 | CrashSys | Thirdlane is a webmin module... sounds interesting... |
21:00.49 | De_Mon | uh, okay. |
21:00.58 | mog | submit a bug sweeper |
21:01.04 | De_Mon | ya I think i've looked at 3rdlanes pbx manager before |
21:01.07 | De_Mon | and a patch |
21:01.12 | Marty-OTT | so... anyone on here who's installed a Sangoma T1 card on FreeBSD here? |
21:01.38 | *** join/#asterisk oQPa (n=roque@15.Red-83-40-197.dynamicIP.rima-tde.net) |
21:01.41 | cbullock81 | danp: ok. it now shows if an extension is registered on asterisk, but when that extension is busy, the status does not change... any ideas? |
21:01.41 | *** join/#asterisk codefreeze (n=steve_mu@216.166.159.235) |
21:02.12 | ManxPower | Marty-OTT: You are on your own, masochist. |
21:02.46 | danp | cbullock81: on your phone, does the entry show up as a silhouette or as a number pad icon? |
21:02.58 | Marty-OTT | I'm not alone in this boat.. the card's been detected but the Sangoma docs are unclear. |
21:03.11 | Marty-OTT | I already sent a support e-mail |
21:03.29 | pif | and you visa card cryptogram ? |
21:03.37 | cbullock81 | danp: silhouette |
21:03.55 | sweeper | mog: I'll put it on the todo list :v |
21:05.18 | danp | cbullock81: hmm...does 'show hints' say you have a watcher? |
21:06.14 | cbullock81 | danp: yea it does |
21:06.51 | danp | if you get that device on a call and check 'show hints' again, does it show its status as InUse? |
21:07.01 | cbullock81 | lemme see |
21:07.09 | Marty-OTT | found the answer.. |
21:07.47 | cbullock81 | danp: yea it still shows idle |
21:08.10 | danp | cbullock81: hmm, you're actually in a call, right? not just off hook? |
21:08.33 | *** join/#asterisk tsurko (n=tsurko@vpn-pppoe-213-240-242-237.megalan.bg) |
21:08.38 | cbullock81 | yea. i placed a call |
21:08.53 | danp | odd |
21:10.09 | danp | on my console, i have my verbosity set to 10 and i see messages such as: |
21:10.14 | danp | <PROTECTED> |
21:10.25 | danp | do you see anything like that? |
21:12.34 | *** part/#asterisk CrashSys (n=kumba@bartleby.crashsys.com) |
21:12.37 | *** join/#asterisk `Sean (i=Un1x@CPE000c148d127c-CM00140458831c.cpe.net.cable.rogers.com) |
21:13.33 | ManxPower | Yet another noob on the mailing list that think a phone has to be registered in order to make calls |
21:13.57 | [TK]D-Fender | ManxPower: *sigh* |
21:14.22 | ManxPower | and one that things "sip show registry" will show you devices registered to Astertisk |
21:15.17 | ManxPower | Oh and one that thinks there is no echo on the PSTN and that it is caused by VoIP |
21:17.39 | pif | if .fr our condoms are called ManX |
21:17.51 | ManxPower | pif: I can live with that |
21:17.53 | pif | coincidence? |
21:18.06 | ManxPower | as long as they are the extra large version |
21:18.34 | pif | just bought a pack in anticipation for sunday night |
21:18.56 | ManxPower | I usually just mail order mine in bulk |
21:18.58 | Qwell[] | "if .fr"? |
21:19.19 | pif | met a chick on meetic.fr |
21:22.15 | pif | viagra is always handy on first dates |
21:23.29 | pif | where pressure to perform can kill your member |
21:23.45 | Juggie | what does this have to do with asterisk |
21:24.43 | Strom_M | bonarpillz |
21:24.45 | pif | just meant to say that * users are not all faggots |
21:24.53 | pif | like its developpers |
21:24.56 | Corydon-w | Hey, now, watch it. |
21:25.08 | [TK]D-Fender | .... |
21:25.21 | *** mode/#asterisk [+b *!*n=ldm@*.apartia.fr] by Qwell[] |
21:25.21 | *** kick/#asterisk [pif!i=qwell@unaffiliated/qwell] by Qwell[] (Qwell[]) |
21:25.25 | Qwell[] | watch nothing |
21:25.38 | Qwell[] | That was inexcusable |
21:26.04 | [TK]D-Fender | Qwell[]: Yup. WAY over the line. |
21:26.09 | ManxPower | Hmm? He should be much more offensive than that to be banned. |
21:26.22 | [TK]D-Fender | ManxPower: Depends on the term... |
21:26.54 | ManxPower | Maybe in France the term is not offensive. |
21:27.02 | Corydon-w | Wouldn't that be "bundle of sticks"? |
21:27.10 | Qwell[] | ManxPower: "<pif> like its developpers" |
21:27.35 | Qwell[] | That comment absolutely shows that it was meant to be offensive |
21:28.23 | Juggie | that type of conversation has no place in here. |
21:28.57 | [TK]D-Fender | Corydon : extra double-entenre of "piles" ;) |
21:29.06 | [TK]D-Fender | Corydon : Recursive humour ;) |
21:30.15 | [TK]D-Fender | Qwell[]: http://dictionary.reference.com/browse/piles |
21:30.44 | Qwell[] | gotcha |
21:30.58 | [TK]D-Fender | :D |
21:31.11 | Qwell[] | That bit of info is unfortunately not something one can "unlearn" |
21:31.32 | [TK]D-Fender | Qwell[]: I will rest well knowing I've made an impact :) |
21:31.47 | [TK]D-Fender | bbiab |
21:31.52 | [TK]D-Fender | heading home... |
21:45.29 | *** join/#asterisk Netgeeks (n=root@pbx5.netgeeks.net) |
21:46.01 | Netgeeks | Anyone see anything wrong with this dialplan command: GotoIf($[${X} >= ${DIGLEN}]?end-loop:start-loop) |
21:46.44 | ManxPower | Netgeeks: what are you trying to ACCOMPLISH? |
21:47.26 | Netgeeks | I need to cycle through a variable lenght string of digits and break them out one at a time to playback the reading of the digits |
21:47.47 | Netgeeks | so I get the length, and loop over the variable for each digit |
21:47.59 | Netgeeks | but, that GotoIf is failing |
21:48.43 | Netgeeks | hrm, I think I figured out why |
21:50.33 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
21:50.33 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
21:52.39 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.216.2) |
21:55.24 | in-pt | hello all |
21:55.37 | b11d | go away |
21:55.39 | b11d | we're closed |
21:55.39 | BSDTech | back |
21:55.42 | b11d | hehe |
21:55.48 | in-pt | i installed asterisk-1.4.0, but dont see skinny on asterisk cli |
21:55.53 | in-pt | what could be the problem |
21:55.55 | b11d | sigh |
21:56.00 | b11d | 33 more minutes.. |
21:56.01 | b11d | lets go |
21:56.26 | BSDTech | The technical channel for wich you have reached is currently closed please logout and try back ltr. Beeeeeepppppp |
22:00.35 | perd | does option p (screening mode) for dial in 1.2 ? |
22:00.42 | perd | err, does option p work |
22:02.08 | perd | goddamn cisco phones forcing me to downgrade |
22:03.11 | BSDTech | ok kids whats going on here |
22:03.15 | BSDTech | The technical channel for wich you have reached is currently closed please logout and try back ltr. Beeeeeepppppp |
22:04.59 | b11d | ahhh |
22:05.26 | cbullock81 | when you are using hints in extensions.conf, are there any additional settings in sip.conf that must be configured (asterisk 1.4) |
22:05.28 | b11d | does Vista continually attempt to re-validate itself against MS or what.. |
22:05.37 | b11d | cbullock81.. not in my experience |
22:05.50 | danp | hmm |
22:05.54 | b11d | i had to make changes in <MAC>-sip.cfg for Polycom's to get hints working for Buddy lists. |
22:05.55 | cbullock81 | the status will not change |
22:06.24 | cbullock81 | my buddy list shows the phones that are turned on properly & has the ones that are not connected showing as away |
22:06.32 | b11d | is it a polycom? |
22:06.35 | cbullock81 | but when a call is made, the status does not change |
22:06.37 | cbullock81 | yea |
22:06.39 | cbullock81 | ip650 |
22:06.50 | b11d | are you using a provisioning server? |
22:06.52 | cbullock81 | yea |
22:07.02 | b11d | hrm.. did you edit the <MAC>-sip.cfg appropriately? |
22:07.27 | cbullock81 | i guess... i must have misseed something |
22:07.44 | b11d | yeah.. i removed my 'hints' shit because i never needed it.. |
22:07.52 | b11d | but it took a little work to get going.. once it worked, it worked.. |
22:07.59 | b11d | cant speak to the 650.. |
22:08.02 | b11d | these were on 501s |
22:08.05 | danp | yeah, i can't think of anything else i had to do. i have 601's |
22:08.08 | *** part/#asterisk naitram (n=danny@216.77.58.40) |
22:08.24 | cbullock81 | hmmm... you didnt use hints in *? |
22:08.32 | danp | i think this is all i had to add to the phone config: <feature feature.1.name="presence" feature.1.enabled="1" /> |
22:08.33 | b11d | yes |
22:08.35 | b11d | i did |
22:08.43 | b11d | but I disabled it, because i had no real use for it |
22:08.48 | cbullock81 | o |
22:08.49 | b11d | i dont just roll shit out because "its neat" |
22:09.07 | b11d | danp is right.. thats what i had to change |
22:09.18 | cbullock81 | k... i did that, but im going to try again |
22:09.23 | b11d | then i had to have hints in the extensions.conf.. on both extensions that were to be monitored |
22:09.41 | b11d | like, say 1234A wanted to monitor 1234B -- both extensions had to have HINTS |
22:09.52 | danp | both do? hmm |
22:09.55 | *** join/#asterisk ta^3 (n=tacvbo@189.146.191.134) |
22:09.58 | cbullock81 | would that setting need to be in my local-settings.cfg? |
22:09.59 | b11d | thats how it worked for me |
22:10.07 | b11d | dunno.. i dont use a 'local-settings.cfg' |
22:10.29 | sweeper | ok, so when configuring a 4-port t1 card, I do my channels like so: bchan=1-23,25-47 dchan=24,48? |
22:10.45 | b11d | check out the zaptel.conf.sample or something |
22:10.45 | sweeper | well, assuming I'm using 2 t1s |
22:11.11 | Marty-OTT | BROADSOFT |
22:11.13 | Marty-OTT | VS. |
22:11.13 | b11d | wouldnt you define seperate spans, and then define b & d chans per span? |
22:11.15 | Marty-OTT | ASTERISK |
22:11.20 | Marty-OTT | ??? |
22:11.26 | perd | OMG VS WTF |
22:11.27 | b11d | Marty.. lets drink Royal Reserve tonight.. |
22:11.28 | Strom_M | Marty-OTT, that's irritating |
22:11.37 | Marty-OTT | sorry |
22:11.47 | sweeper | ok, I needs a bettar examplu |
22:11.55 | Marty-OTT | Looking at Broadsoft solutions.. just sent an e-mail.. if I don't have to reinvent the wheel.. I won't |
22:12.10 | Strom_M | what does broadsoft sell? |
22:12.12 | perd | the wheel is obsolete |
22:12.12 | sweeper | the example zaptel.conf doesn't really explain syntax for multiple spans |
22:12.17 | b11d | hrm... |
22:12.26 | sweeper | perd: so is your mom. doesn't stop us from using her every night \o |
22:12.34 | Marty-OTT | SoftPBX solution - pretty much a leader in the industry for paid products |
22:12.53 | b11d | http://www.voip-info.org/wiki/index.php?page=Asterisk+config+zaptel.conf |
22:13.04 | sweeper | I was just heading there ;) |
22:13.09 | Strom_M | Marty-OTT, you honestly don't have to reinvent the wheel with asterisk - it's easy stuff |
22:13.14 | sweeper | between voip-info and *guru, it's all good :) |
22:13.16 | *** join/#asterisk simplton (n=root@c-24-99-119-243.hsd1.ga.comcast.net) |
22:13.22 | james_ | Marty-OTT: Broadworks is about 100000000000000x more complete than asterisk |
22:13.31 | *** part/#asterisk simplton (n=root@c-24-99-119-243.hsd1.ga.comcast.net) |
22:13.37 | Marty-OTT | Strom_M: You know what - I agree - so far it hasn't been to hard... |
22:13.39 | Marty-OTT | James??? |
22:13.50 | Marty-OTT | how's that? |
22:14.05 | Marty-OTT | I haven't talk to anyone at Broadsoft and probably won't until Monday |
22:14.11 | eGov | Maybe he means complex? |
22:14.16 | james_ | Marty-OTT: it's a complete platform, all administered via guis |
22:14.35 | rene- | like mitel |
22:14.36 | *** join/#asterisk ivanfm (n=ivanfm@c93481ec.virtua.com.br) |
22:14.37 | james_ | it's more complex, it's more complete, it has way more features, it's easier, it's more expensive |
22:14.42 | Marty-OTT | JAnes: Can someone go in and turn on/off features on a per phone basis? |
22:15.01 | james_ | on a per user basis, yeah |
22:15.02 | rene- | nothing you cant emulate with some mad SER & YATE & ASTERISK skillz |
22:15.03 | Marty-OTT | James: Interesting.. you have any idea how expensive it is? (I mean, I'll know on Monday/Tueasday anyways) |
22:15.03 | *** join/#asterisk simplton (n=root@c-24-99-119-243.hsd1.ga.comcast.net) |
22:15.19 | james_ | rene-: and maybe 5 years of time |
22:15.34 | rene- | of course wont be as polished |
22:15.35 | Marty-OTT | That's the thing.. |
22:15.48 | Strom_M | Marty-OTT, I sent you a PM |
22:15.55 | james_ | Marty-OTT: no idea how much, my guess is quite expensive... i help support our countrys biggest telcos voip network, which is broadsoft |
22:16.13 | Marty-OTT | yeesh.. |
22:16.21 | rene- | broadsoft is big iron voice session border controller stuff 100k usd and up |
22:16.24 | *** join/#asterisk bkw__ (n=brian@ip68-0-120-100.tu.ok.cox.net) |
22:16.29 | rene- | and waaay up |
22:16.30 | Marty-OTT | yow!!! |
22:16.54 | Marty-OTT | Well, yeah, I don't want to spend more than $10,000 |
22:16.57 | b11d | what country is .st? |
22:17.03 | james_ | oh, i imagine it would be out of your price range |
22:17.09 | james_ | b11d: haha, ignore that... i'm in australia |
22:17.12 | b11d | oh |
22:17.14 | b11d | :) |
22:17.20 | rene- | yup, maybe some used cisco or mitel gear |
22:17.22 | b11d | i'd move back to australia in a second.. |
22:17.26 | rene- | mitel is user friendly |
22:17.34 | rene- | dunno about cisco |
22:17.51 | rene- | i wouldnt think it was user friendly |
22:17.54 | Marty-OTT | rene: funny you say that about Mitel... I taught a lot of CIsco courses for them and they tried to get me to take a free course on the 3300 |
22:18.01 | james_ | what cisco would offer for <$10k would be a joke |
22:18.02 | Marty-OTT | just never got around to it. |
22:18.52 | rene- | well the stability problems of 1.4 in my shop is making managers want to revert back to mitel, and me taking a course to learn about the gear.. sigh... |
22:19.08 | rene- | from what i have seen it seems quite simple |
22:19.22 | b11d | what kind of stability issues? |
22:19.27 | b11d | deadlocks? |
22:19.31 | rene- | yes |
22:19.31 | *** join/#asterisk dasenjo (n=dasenjo@208.195.215.51) |
22:19.34 | b11d | still eh.. |
22:19.35 | b11d | that sucks |
22:19.42 | *** join/#asterisk syzygyBSD (n=chatzill@66.226.228.204.cpe.speedyquick.net) |
22:19.52 | Strom_M | rene-, this is why my clients are still on 1.2.x; i'm waiting until two major bugfix releases of 1.4 to put it in production |
22:19.52 | *** part/#asterisk endikos (n=endikos@gtlgateway.nmsu.edu) |
22:19.59 | Marty-OTT | rene: 1.4 for what? |
22:20.10 | eGov | I assume * |
22:20.16 | b11d | eGov would be right |
22:20.23 | b11d | eGov, you better start piping up more often :) |
22:20.27 | BSDTech | o1.4 is good in svn head |
22:20.28 | eGov | hehe |
22:20.30 | syzygyBSD | I want to upgrade to 1.4 if only for the imap voicemail |
22:20.32 | Marty-OTT | oh asterisk .. |
22:21.01 | BSDTech | wwhats asterisk and what can it do for me. how will it make me rich |
22:21.04 | Marty-OTT | Rene: did you simply use the 3300 as a Sip server? The Mitel sets are expensive |
22:21.05 | cbullock81 | b11d or danp: what options would you configure for the directory.xml anything special |
22:21.40 | b11d | i dont use it.. |
22:21.47 | b11d | not yet anyway.. the directory on polycom's sucks |
22:21.53 | b11d | especially when it gets over like 15 entries |
22:21.55 | *** part/#asterisk ctooley (n=ctooley@rrcs-71-42-115-242.sw.biz.rr.com) |
22:22.23 | syzygyBSD | BSDTech: asterisk will search for little, lets call them "widgets" and will sell them on the black market, by doing this it will slowly amass a fortune so it can enter the drug trade, shortly thereafter you will be rich! |
22:22.27 | cbullock81 | well, that is where i added the extensions that i want to monitor to the soft keys |
22:22.35 | b11d | in directory.xml ?? |
22:22.38 | b11d | wtf |
22:22.41 | b11d | ohh yeah thats right |
22:22.42 | BSDTech | lol |
22:22.44 | b11d | I forgot about that |
22:22.59 | b11d | man I need to not pipe up when im not doing somethign anymore :) |
22:23.10 | cbullock81 | heh... i appreciate your input |
22:23.11 | rene- | 1.4 for call center |
22:23.20 | cbullock81 | i need any help i can round up |
22:23.23 | b11d | yeah i hear that.. |
22:23.24 | rene- | Strom_M: yes |
22:23.29 | b11d | one sec..i'll see if i still have anything |
22:23.33 | rene- | but the way queues work is a bit slow |
22:23.36 | cbullock81 | that would be great |
22:24.01 | b11d | got it |
22:24.42 | BSDTech | ok all the bsd users to the left of the room and all the linux users to the right |
22:24.59 | BSDTech | bsd users raise your left hand |
22:25.09 | BSDTech | linux users raise your right hand |
22:25.20 | Marty-OTT | Rene: So, what were using from Mitel? |
22:25.21 | BSDTech | and if you use both grab a seat on thefloor |
22:25.36 | Marty-OTT | I'm sitting... but I'm Freebsd |
22:25.53 | BSDTech | well I have freepbx 2.2 almost working on bsd |
22:26.05 | syzygyBSD | ya, our last bsd box died a month ago, we installed linux to replace it |
22:26.14 | b11d | weak |
22:26.14 | BSDTech | bumbs |
22:26.24 | BSDTech | move back to bsd |
22:26.28 | b11d | yeah |
22:26.47 | BSDTech | they should move the asterisk project to bsd only |
22:26.48 | syzygyBSD | don't have the free time |
22:26.57 | BSDTech | free time 45 min |
22:26.58 | BSDTech | ok |
22:27.08 | rene- | Marty-OTT: Mitel is being used for queues |
22:27.18 | rene- | in some of the call center |
22:27.20 | Marty-OTT | yes, but shich box? |
22:27.22 | Marty-OTT | which box? |
22:27.22 | BSDTech | for a basse install and apache |
22:27.25 | rene- | i think we have a 330 |
22:27.26 | rene- | 3300 |
22:27.29 | syzygyBSD | free time configuring 15 servers to work with BSD... and all the applications we need, custom scripts, etc |
22:27.39 | BSDTech | ahh ok |
22:27.49 | BSDTech | build 1 and clone it |
22:28.02 | rene- | but everything is super expensive |
22:28.02 | BSDTech | thats what gmirror is for |
22:28.06 | syzygyBSD | lol, no chance, all our servers have different things running on them |
22:28.29 | BSDTech | no you clone the base install |
22:28.36 | Marty-OTT | zyaugyBSD: Hardward of software crash? I've had FreeBSD running for 2 years - Apache, SSL, GNU RAdius, Postgres, Courier Mail, PureFTPD - never a single crash |
22:28.37 | BSDTech | then add the needed pkgs |
22:28.44 | syzygyBSD | or, we could just stick with what is working... |
22:28.53 | ManxPower | Most distros let you build an autoinstall |
22:28.58 | syzygyBSD | Marty-OTT: hardware |
22:29.11 | BSDTech | then you could have just moved the drive |
22:29.12 | syzygyBSD | it was runnign for 4 years without a single problem though |
22:29.17 | BSDTech | unless the drive failed |
22:29.32 | BSDTech | and you should have had the drive raid 1 |
22:29.33 | syzygyBSD | backplane died, and we didn't have a duplicate box |
22:29.43 | Marty-OTT | Well... that's not a BSD issue then.. |
22:29.52 | syzygyBSD | I didn't say it was... |
22:29.59 | BSDTech | then just move the drive |
22:30.03 | BSDTech | it will boot |
22:30.59 | Marty-OTT | Well, so long as it's the same motherboard chipset.. |
22:31.04 | syzygyBSD | we did to pull the info off, but we didn't have a spare 2U server that would support 6 drives |
22:31.23 | b11d | but this goes up to 11.. |
22:31.51 | syzygyBSD | had to move to a 5 disk configuration |
22:31.55 | Marty-OTT | oh.. |
22:32.26 | rene- | mitel works well but it is not nearly as flexible as asterisk |
22:32.30 | syzygyBSD | ya, but setting up a server with 4 databases running on it in 2 hours is kinda fun |
22:32.31 | rene- | and it is oh so expensive |
22:33.44 | Marty-OTT | rene: cool.. |
22:33.49 | *** join/#asterisk robin_sz (n=robin@rapid2.gotadsl.co.uk) |
22:34.23 | *** join/#asterisk Aughey (n=jha@64.219.54.121) |
22:34.49 | robin_sz | hi .. using chan_mISDN, from at least a few landlines it doesnt seem to convert dtmf tones to dialled digits ... how can I make it more sensitive? |
22:35.03 | Aughey | how do I change what is displayed on the display of the phone when its extension is dialed? |
22:35.24 | *** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net) |
22:35.25 | syzygyBSD | robin_sz: change the DTMF mode |
22:35.39 | robin_sz | right |
22:35.49 | robin_sz | umm |
22:36.00 | robin_sz | err |
22:36.02 | syzygyBSD | dtmfmode=inband |
22:36.05 | syzygyBSD | or something... |
22:36.09 | b11d | Aughey.. why dont you get a little more specific eh |
22:36.26 | robin_sz | syzygyBSD, in where exactly? |
22:36.27 | b11d | oh wait no.. let me just get out my copy of "Manual for Every Kind of Phone in The World" |
22:36.29 | *** join/#asterisk [hC] (n=hardcore@mail.metrobridge.com) |
22:36.33 | syzygyBSD | I don't do anything with mISDN, but that is normally the solution to your problem |
22:36.46 | syzygyBSD | robin_sz: the top of the configuration file for mISDN? |
22:37.16 | Aughey | I have SIP telephones and they have either a 2 or 4 line display. When you dial an extension, it displays "Call from XXXXX". I want it to say something else (for extension sharing) |
22:37.20 | robin_sz | syzygyBSD, you are nto confusing this with the dtmfmode=rfc1183/inband problem are you? |
22:37.37 | syzygyBSD | sounds like that is the problem to me... |
22:37.41 | b11d | Aughey.. do you think its going to be the same procedure for every kind of SIP phone available? |
22:37.47 | b11d | or do you maybe want to tell us what kind of phones you have? |
22:38.05 | syzygyBSD | is it just a gain issue? |
22:38.21 | Aughey | I have Grandstream and snom phones |
22:38.24 | b11d | ugh |
22:38.25 | b11d | umm |
22:38.40 | b11d | i have no further contributions :P |
22:39.03 | syzygyBSD | poor people who have grandstreams, so shuned |
22:39.17 | b11d | yeah.. im an asshole.. |
22:39.25 | b11d | its actually just that i have no experience with those.. |
22:39.55 | robin_sz | syzygyBSD, err ... sureley dtmfmode is only an option on SIP channels? |
22:40.33 | robin_sz | I have a grandstream GXP2000 .. its great! I would recommend it to anyone |
22:40.53 | syzygyBSD | uhh, I only deal with SIP and PRI, so sorry if I am going in the wrong direction |
22:41.04 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
22:41.08 | robin_sz | its wonderful for stopping cars rolling on a slope |
22:41.17 | *** join/#asterisk Dr-Linux|home (n=Dreamer@DSL-202-59-73-131.nexlinx.net.pk) |
22:41.45 | robin_sz | or for putting under your pillow to provide a better back angle when relaxing reading a book in bed |
22:41.50 | *** join/#asterisk Qwell[] (i=qwell@unaffiliated/qwell) |
22:41.50 | *** mode/#asterisk [+o Qwell[]] by ChanServ |
22:41.57 | Dr-Linux|home | hi |
22:42.34 | robin_sz | I did try using it as a phone once, but that was disastorous, I dont think it was designed for that purpose |
22:43.16 | b11d | hahaha |
22:43.27 | b11d | robin_sz.. i just imagine you at a conference saying that stuff, all straight faced.. |
22:43.51 | robin_sz | nah |
22:44.11 | syzygyBSD | not the conference going type? |
22:44.19 | robin_sz | id be out in the car park, making money ... "grab a hammer, smash a grandstream, only $2 a time" |
22:45.01 | syzygyBSD | any tech conference could make a ton off that |
22:45.13 | robin_sz | unless anyone can think of a way of getting reid of GXP2000's for more than £2 a pop |
22:45.23 | [hC] | Has anyone experienced an issue with analog lines in asterisk where someone tries to place a call out, "seemingly" when a call is trying to come in at the same time, and instead of being connected to the number that the person was trying to call OUT to, they are connected to the new incoming call? |
22:45.29 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
22:45.38 | [hC] | Maybe an issue where a line does not hang up quick enough before asterisk tries to sue that channel to call out on again? |
22:45.46 | [hC] | sue=use |
22:47.39 | Strom_M | [hC], that's a problem called glare |
22:47.53 | Qwell[] | ~glare |
22:47.54 | jbot | ACTION glares at qwell[] |
22:48.02 | Strom_M | silly qwell |
22:48.03 | Qwell[] | okay then |
22:48.13 | Strom_M | [hC], how many analog trunks do you have? |
22:48.34 | b11d | it wont |
22:48.37 | b11d | its already fried |
22:48.40 | robin_sz | ok, there seems to be a "dtmf threshold" value in the misdn module config |
22:48.48 | robin_sz | but its in milliseconds |
22:48.52 | b11d | you hooked it up to the power supply again didnt you? |
22:49.00 | robin_sz | hows that work then? |
22:49.10 | robin_sz | I think this is a low gain sort of problem |
22:49.17 | b11d | messing with gain sucks |
22:49.20 | robin_sz | sending digits might be a bit quiet |
22:49.23 | sweeper | it did |
22:49.26 | sweeper | yay \o |
22:49.49 | b11d | excellent |
22:50.33 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
22:50.33 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
22:52.54 | syzygyBSD | robin_sz: are you sure they are quiet? |
22:53.09 | syzygyBSD | ie, have you recorded a call having problems? |
22:53.24 | *** join/#asterisk jvalenzuela (n=jvalenzu@wingnut.dspfl.com) |
22:53.42 | x86 | what do you guys think of my new web design: http://www.shellshark.net/ |
22:54.17 | b11d | doenst look bad |
22:54.17 | syzygyBSD | wow, a redirection page |
22:54.17 | sweeper | is nice, but redirects and bad ssl certs are bad |
22:54.29 | sweeper | jsut use mod-rewrite |
22:54.39 | syzygyBSD | loads kinda slow |
22:54.45 | syzygyBSD | but looks good |
22:55.03 | BSDTech | ok man I hate oh323 |
22:55.08 | sweeper | your "concurrent calls" thing could use some cleaning up |
22:55.19 | BSDTech | I have to kill it in the atserisk build for 1.4 |
22:55.52 | syzygyBSD | x86: I would recommend you reorder the number of concurrent calls and place them next to eachother |
22:56.13 | robin_sz | syzygyBSD, no, im not sure, but certainly some analogue lines seem to have trouble and from what i can see, mobiles don't |
22:56.24 | x86 | syzygyBSD: yeah, that's on the todo list already :) |
22:57.45 | syzygyBSD | hmm, the phone numbers seem a bit wrong too |
22:58.46 | *** join/#asterisk justdave (n=dave@unaffiliated/justdave) |
22:58.59 | *** join/#asterisk alamantia (i=Anthony@nat/digium/x-cb1f27dbaed7fec0) |
22:59.34 | syzygyBSD | I'd suggest a print link on terms of service too. |
22:59.46 | syzygyBSD | wish I could do graphic design |
22:59.58 | [hC] | Strom_M: 7. |
23:00.06 | robin_sz | ot looks nice enough |
23:00.08 | robin_sz | it |
23:00.32 | robin_sz | although I suspect its all hard-coded in some Perl scripts |
23:01.16 | *** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox) |
23:01.30 | x86 | syzygyBSD: there is a print link, down at the bottom |
23:02.06 | x86 | robin_sz: already using a template system, no HTML is hard-coded in perl ;) |
23:02.15 | x86 | well, XHTML even ;) |
23:02.18 | robin_sz | x86, mason? |
23:02.25 | x86 | home built |
23:02.39 | robin_sz | ummm ... why? |
23:02.40 | syzygyBSD | x86, where is the print link? |
23:02.53 | x86 | syzygyBSD: down by the copyright in the bottom footer |
23:03.17 | x86 | robin_sz: for reasons I can not disclose here ;) |
23:03.20 | syzygyBSD | ahh, k, I was viewing the version off the menu |
23:03.43 | x86 | robin_sz: the CMS engine ties in with everything else we have, basically |
23:04.08 | x86 | robin_sz: so we decided to build one from the ground-up to get complete integration, while maintaining maintainability ;-) |
23:05.07 | robin_sz | x86, so the template system can read the same templates as use dby your CMS? |
23:06.40 | x86 | robin_sz: what you are seeing is the CMS in action |
23:07.08 | robin_sz | right, time to go and play with my robot |
23:07.25 | sweeper | but perl D: |
23:07.40 | sweeper | ruby on rail for me prz |
23:07.43 | *** join/#asterisk sbingner (n=sam@pdpc/supporter/sustaining/sbingner) |
23:10.50 | *** join/#asterisk WAudette (n=WAudette@c-71-237-146-239.hsd1.or.comcast.net) |
23:12.30 | Strom_M | [hC], are inbound and outbound calls hunting from different ends of the group? |
23:14.13 | WAudette | Is anyone here familiar with Paging through the Sound Card? It is working but, through chan_oss. Are there any switches that I can set in (console/dsp) to tell an extention to not autoanswer in effect making it Ring? Then I could have another extention page like normal with just (console/dsp)? |
23:14.53 | Strom_M | WAudette, huh? |
23:15.13 | Strom_M | i'm not clear on what you're trying to do |
23:15.17 | WAudette | In oss.conf if I set autoanswer=no it rings but I want to set it per exention and not for the whole channel. |
23:15.26 | Strom_M | why? |
23:16.07 | WAudette | Overhead Paging in a warehouse... They want it to produce a ringing sound when a call comes in over the PA System. They want to Intercom Page via the same PA System. |
23:16.38 | Strom_M | why would calls come in via the PA system? |
23:16.41 | WAudette | I can do one or the other so far, but can't figure out how to fanagle it to do either or. |
23:16.45 | Strom_M | that makes no sense |
23:16.57 | WAudette | They woldn't actuall come in... Just produce the ring sound. |
23:17.07 | Strom_M | oh, ok, your grammar is screwy |
23:17.16 | Strom_M | so you want to use the PA as a ringer |
23:17.22 | WAudette | Yeah... Train of thought typing... sorry. |
23:17.42 | WAudette | Yes, and as an intercom. |
23:17.55 | nortex | WAudette, What paging system are you using? |
23:18.07 | Strom_M | well, paging moreso than intercom; intercom implies two-way communication |
23:18.17 | WAudette | A legacy Valcom PA from the system I am replacing w/ * |
23:18.31 | Strom_M | WAudette, i'm trying to think of what you can do... |
23:18.36 | WAudette | It has a closed input... meaning it is simple and alway off hook. |
23:18.42 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
23:19.07 | Strom_M | you could generate a .call file when the call comes in, and have that .call file call console/dsp with a ringing sound or a message saying "call from [whoever]" |
23:19.22 | WAudette | Strom_M: Good point on the terminology.... Paging is more specific and what I meant to ask. Sorry. :) |
23:20.49 | nortex | WAudette, To get a ring on the sound card you can turn off autoanswer on the chann_oss, then dial it with the incoming call. |
23:20.55 | Strom_M | nortex, but that's not going to work in his situation |
23:20.59 | Strom_M | he wants to use it as a PA as well |
23:21.03 | WAudette | Ok. There aren't any switches that you can think of like (console/dsp, ,A(custom/bosun) but one that would tell chan_oss not to answer? |
23:21.03 | Strom_M | so better to use the .call file |
23:21.11 | Strom_M | not AFAIK |
23:21.46 | nortex | So you want to use the sound card to page and be the loud ringer? |
23:21.53 | WAudette | nortex: Yes, that does work... but sets it up to no answer for all calls to (console/dsp) and I still need the Paging system output too. |
23:22.17 | Strom_M | WAudette, see my solution |
23:22.34 | WAudette | nortex: Correct Page and loud ringer... Excactly. |
23:23.18 | nortex | Ok, That is different then mine, I use ZAP to page and the sound card to ring though a music input. |
23:24.01 | *** join/#asterisk linlin (i=techpeps@71.194.70.13) |
23:24.06 | nortex | I mainly use a Viking system, but I have used a valcom with FXO and FXS inputs. |
23:24.07 | WAudette | Strom_M: Yes, to make a .call file. |
23:24.08 | mitcheloc | you could get a voip overhead speaker |
23:24.23 | WAudette | Thinking about it. |
23:25.07 | mitcheloc | probably not as cheap though |
23:25.28 | WAudette | But then I have to find a VOIP overhead speaker that does the same thing... Paging + Ringign. |
23:25.32 | justdave | I'm trying to set up an inbound IAX trunk. It registers successfully, but inbound calls generate an error in asterisk "Rejected connect attempt from <iax provider ip>, who was trying to reach '<our phone number>@' |
23:25.33 | sweeper | or get a cheep pc and put a softphone set to autoanswer on it |
23:25.34 | nortex | mitcheloc, I looked at those and was suprised at the cost, at least for small systems. The Viking one I've been using is $160 with the system and 1 8-ohm horn. |
23:25.53 | *** join/#asterisk RoyK (n=roy@217-175-39.100710.adsl.tele2.no) |
23:25.53 | BSDTech | waudette ok hire me |
23:25.56 | justdave | anyone know what I should look for? been looking at iax docs for a while and not seeing what I'm doing wrong |
23:26.04 | WAudette | Strom_M: Do you have an example .call file that I could work with. I have never made one. |
23:26.24 | WAudette | BSDTech: Ok, I can do that. <grin> |
23:26.30 | Strom_M | no; check voip-info |
23:26.35 | nortex | justdave, Does the number your calling exist in the context the IAX peer is in? |
23:26.46 | WAudette | Strom_M: Thanks for your input! |
23:27.17 | BSDTech | I will move up there in a heart beat |
23:27.23 | justdave | there's an extension defined with the DID number, if that's what you're asking |
23:27.42 | justdave | in the from-outside context, which is the context defined in iax.conf |
23:32.06 | nortex | justdave, can you pastebin the relevant iax.conf and extensions.conf sections? |
23:35.38 | *** join/#asterisk riddlebox (n=james@75-132-205-166.dhcp.stls.mo.charter.com) |
23:37.24 | justdave | nortex: http://pastebin.mozilla.org/2796 http://pastebin.mozilla.org/2797 |
23:37.39 | justdave | the xxxxxs are actually the phone number of course |
23:38.15 | *** join/#asterisk FaithX (n=faithful@ns.linuxterminal.com) |
23:38.42 | *** part/#asterisk vt3 (n=vt3@m016020.ppp.asahi-net.or.jp) |
23:39.10 | justdave | the extension stuff is identical to the from-outside-1650xxxxxxx-tl-allhours context is set up identical to one for a different inbound line (except that one comes in via SIP) |
23:43.03 | justdave | my first thought was to add the @ on the end of the extension number since it was including it there on the inbound, but it didn't like that |
23:43.08 | nortex | So the second pastebin is the context from-outside |
23:43.17 | justdave | correct |
23:45.09 | justdave | hmm, actually, that's in from-outside-redir |
23:45.14 | justdave | but from-outside has this: |
23:45.14 | justdave | exten => _X.,1,Goto(from-outside-redir,${EXTEN},1) |
23:45.14 | justdave | exten => s,1,Goto(from-outside-redir,${EXTEN},1) |
23:45.55 | nortex | Ahh, try commenting out the s |
23:47.42 | riddlebox | does anyone know why in fedora I cannot get my music on hold working? |
23:48.09 | riddlebox | all I get is some little static |
23:48.20 | b11d | can you play the mp3 manually with mpg123 ? |
23:48.25 | b11d | and hear it over your speakers? |
23:48.56 | nortex | justdave, That may not solve it, you might also try setting iax.conf context to from-outside-redir |
23:48.57 | *** join/#asterisk Globetrotter (n=eric@205.211.214.167) |
23:49.01 | justdave | the s is the default inbound isn't it? |
23:49.24 | riddlebox | blld, the problem is I dont have anything hooked up to the box, it is a stand alone machine plus on fedora it is mpg321 but there is a link to mpg123 |
23:49.29 | nortex | yeah, s should be used if nothing matches, if I remeber right. |
23:50.00 | justdave | we have other phone numbers so I'd prefer not to screw with that :) I'll try changing the context on the iax.conf |
23:50.15 | justdave | nope, that didn't work |
23:50.22 | Globetrotter | Hi Guys,, finally got my MOH towork.. how to i make it play the files at random.. mode=files, random=yes.. but no good |
23:50.25 | justdave | same error |
23:50.32 | Globetrotter | plays the same files |
23:50.33 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
23:50.33 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
23:50.49 | Globetrotter | and i have mulitles files in my folder |
23:51.33 | nortex | justdave, Does the number match exactly or is it like a DID with 10 digits or 4 instead of all 11? |
23:51.54 | riddlebox | blld, I guess I could hook up a headset and see, hold on |
23:52.00 | justdave | it's 11 digits |
23:52.08 | nortex | justdave, It is really odd to me that the number@context in the error has a blank context. |
23:52.29 | justdave | does IAX need to know the context on both ends? |
23:52.40 | justdave | maybe I need to set it somewhere on the provider's control panel |
23:53.46 | nortex | No your just telling it where to put the calls as they come to your system. |
23:55.25 | nortex | justdave, Sorry I didn't help much. |
23:56.09 | Dr-Linux|home | anybody is using agent/queue system? |
23:59.39 | WAudette | Strom_M: Do you have any hints on what to search for? .call seems to be a wildcard of sorts catch almost everyting in voip-info.org. |
23:59.54 | Strom_M | "call file" |
23:59.55 | Qwell[] | WAudette: "call file" |