| 00:01.24 | C6Vette | tessuer__, If I use the xfer button on a Grandstream its unattended, but on the Cisco its attended. So I guess its phone dependant. |
| 00:01.44 | C6Vette | To some degreee |
| 00:02.47 | Dr-Linux | anybody is using spa3000? |
| 00:04.05 | tessier__ | ah, I need to enable it in features.conf... |
| 00:04.28 | tessier__ | C6Vette: Right. Phone dependent and then you need to use a feature code in asterisk to implement which ever one your phone does not do. |
| 00:05.54 | *** part/#asterisk rnovotny22 (n=rnovonty@198.57.19.126) |
| 00:08.15 | C6Vette | That make sense.. |
| 00:11.57 | *** join/#asterisk okdo (n=goldenol@65.171.196.18) |
| 00:11.59 | f0urtyfive | hey |
| 00:12.12 | f0urtyfive | can anyone point me to some "valid" e164 info for asterisk |
| 00:12.12 | okdo | is there a simple way of making it so a user can't call out but can belong to the normal default context? |
| 00:12.19 | f0urtyfive | seems all the stuff I can find has been outdated |
| 00:13.38 | f0urtyfive | okdo: I believe thats the point of contexts ;) |
| 00:14.28 | *** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.itb.ac.id) |
| 00:16.18 | *** join/#asterisk aaugustf (n=aaugustf@copper-14.dynamic2.rpi.edu) |
| 00:16.50 | aaugustf | hi everyone |
| 00:16.54 | aaugustf | in working on an extensions.conf, i have the following: exten => t,4,Dial(SIP/102,20,m) |
| 00:17.16 | aaugustf | instead of going to an extension, i'd like to to send the call to an IVR - what is the syntax for doing that? |
| 00:19.09 | Nugget | clarify "an IVR" |
| 00:20.43 | aaugustf | i have a named IVR (digital receptionist) that i use for incoming calls |
| 00:21.01 | aaugustf | i'd like to direct calls in this script (which are incoming) to the IVR |
| 00:21.21 | Nugget | How does it plug into your asterisk box? |
| 00:21.55 | aaugustf | well, this is actually part of a script that takes incoming calls from a SPA-3000 and sends them to Asterisk |
| 00:22.00 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
| 00:22.04 | shmaltz | hi everyone |
| 00:22.15 | aaugustf | right now, it directs those calls to extension 102, but i'd like it to go directly to an IVR rather than a particualr extension |
| 00:22.25 | shmaltz | who here has perl odbc experience? |
| 00:23.24 | Nugget | ...and will admit it? :) |
| 00:23.49 | shmaltz | Nugget, if you are afraid to admin it, PM me |
| 00:23.53 | Nugget | aaugustf: I still don't understand how the IVR is connected to your asterisk install, which obviously affects the answer. |
| 00:24.09 | shmaltz | it can be any other language as long as it does the job |
| 00:24.24 | aaugustf | the IVR is the one built into asterisk - i created it via the freepbx interface - is that unusual? |
| 00:24.35 | Nugget | oh, ok. so it's just a context in your dialplan? |
| 00:24.48 | Nugget | just use the Goto() application in lieu of that Dial() |
| 00:24.51 | aaugustf | i added options, recordings, etc - now i simply want the script to connect to it - but the IVR doesn't have an extension |
| 00:25.03 | Nugget | it has to have an extension if it's in the dialplan |
| 00:25.21 | aaugustf | so it would be Goto(Corp_IVR1) for instance? |
| 00:25.35 | Nugget | Depends. What is "Corp_IVR1"? |
| 00:25.39 | f0urtyfive | lolk |
| 00:25.55 | aaugustf | that would be the name of the IVR |
| 00:26.00 | aaugustf | as i named it in freepbx |
| 00:26.01 | Nugget | clarify "the name" |
| 00:26.08 | Nugget | freepbx? |
| 00:26.13 | Nugget | This is #asterisk. |
| 00:26.33 | *** join/#asterisk DrukenHME (n=jdumais@CPE0040f43870d3-CM00137189cb0c.cpe.net.cable.rogers.com) |
| 00:26.54 | aaugustf | well, i think this is more of an asterisk scripting issue (though the IVR was generated via freepbx) |
| 00:26.58 | DrukenHME | evening everyone, anyone got a working machine on rogers cable? |
| 00:27.11 | Nugget | Perhaps. If the freepbx gui is inadequate to do this sort of thing, it is. |
| 00:27.22 | Nugget | but you're like nine steps away from being able to ask that sort of question in here. |
| 00:27.22 | aaugustf | incidentally, using the Goto command simply made asterisk drop the call |
| 00:27.48 | Nugget | It's assumed you have the fundamental understanding of how extensions.conf works before we can really help out |
| 00:28.33 | Nugget | nor do the people in here have any real experience with freepbx, so there's little understanding on our part how your extensions.conf has been set up/mangled/molested by the demands of the freepbx gui. |
| 00:28.47 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 00:28.53 | aaugustf | well given that everything else is setup, its really my ignorance of asterisk coding that's hurting |
| 00:28.58 | *** join/#asterisk techie (n=techie@66-81-136-77.nocal.dialup.o1.com) |
| 00:28.58 | DrukenHME | </rant> ? |
| 00:29.01 | Nugget | minimally you should read up on the distinctions between contexts, extensions, and steps. |
| 00:29.56 | *** join/#asterisk Samoied (n=Samoied@201.21.232.88) |
| 00:30.06 | shmaltz | DrukenHME, why the block SIP? |
| 00:30.13 | DrukenHME | anyone know is rogers is being a bunch of dinks and blocking 4569 ? |
| 00:30.15 | Nugget | since freepbx apparently allowed you to create a whole IVR, it stands to reason that it doesn't lack the capability you're seeking. I'd suggest giving it another attempt at resolving this within the freepbx framework. |
| 00:30.31 | DrukenHME | shmaltz: huh? block SIP ? |
| 00:30.31 | Nugget | that's what they'd want you to do, I'm sure. |
| 00:30.51 | shmaltz | whats 4569? IAX? |
| 00:30.56 | DrukenHME | yeah |
| 00:31.34 | shmaltz | DrukenHME, I doubt that blocking IAX is on their list, but are you talking inbound or outbound? |
| 00:31.45 | DrukenHME | inbound... |
| 00:31.47 | *** join/#asterisk heison (n=heison@CPE000a01d49e6f-CM00111a59bec2.cpe.net.cable.rogers.com) |
| 00:32.13 | shmaltz | DrukenHME, as per their AUP are you allowed to host anything? |
| 00:32.33 | DrukenHME | fuct if i know... who actually reads it? |
| 00:32.48 | DrukenHME | they gonna tell me i can't host my own telephone? |
| 00:33.16 | shmaltz | DrukenHME, it might be so, and that might be the reason for them blocking it |
| 00:33.27 | shmaltz | DrukenHME, try changing the port number and test it |
| 00:34.14 | DrukenHME | uhg.... |
| 00:34.46 | DrukenHME | i can send iax calls FROM my home server to my work server, works fine, but work server says my home server is unreachable |
| 00:35.06 | *** join/#asterisk riddlebox (n=james@24-171-10-102.dhcp.stls.mo.charter.com) |
| 00:36.54 | shmaltz | DrukenHME, changing the port number should take more than 2 minutes just do it and it will confirm your doubts |
| 00:37.15 | DrukenHME | yeah... |
| 00:37.21 | DrukenHME | rogers is such a pain in my ass |
| 00:39.01 | hmmhesays | anyone every play poker at partypoker.com? |
| 00:41.07 | shmaltz | hmmhesays, I don't think here, try #pokerplayers |
| 00:41.24 | bkw_ | ok what did I miss? |
| 00:41.42 | shmaltz | bkw_ nothing interesting, sos |
| 00:41.46 | shmaltz | ~sos |
| 00:41.56 | jbot | Verilog Design Data Management Product. URL: http://www.cliosoft.com |
| 00:44.44 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 00:44.44 | *** mode/#asterisk [+o mog] by ChanServ |
| 00:44.45 | litage | what's BLF? |
| 00:45.24 | shmaltz | ~blf |
| 00:45.36 | jbot | from memory, blf is Busy Lamp Field, aka little lights next to speed dials that light up when the person is on the phone and blink when that line is ringing. hint extensions are static mapped to SIP or other channels. |
| 00:45.36 | shmaltz | ~google blf |
| 00:46.21 | litage | thanks |
| 00:54.04 | *** join/#asterisk poonj (n=poonj@c-67-172-183-153.hsd1.ca.comcast.net) |
| 00:57.49 | *** join/#asterisk TripleFFFF (n=TripleFF@145-27.mc.cite.net) |
| 00:58.24 | *** join/#asterisk Dovid (n=dovi5988@pool-71-250-59-102.nwrknj.east.verizon.net) |
| 00:58.28 | TripleFFFF | question.. can we match but starting with the end ? as in _X6232225555,1, ? |
| 00:58.32 | TripleFFFF | to mathc the one or not ? |
| 00:58.37 | TripleFFFF | or no wat to do that |
| 01:01.57 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 01:04.31 | shmaltz | TripleFFFF, what are you trying to do? |
| 01:05.55 | TripleFFFF | match ALL 555-555-1212 and 1-555-555-1212 |
| 01:07.13 | *** join/#asterisk kio (n=kio@ool-4577ae5e.dyn.optonline.net) |
| 01:08.16 | shmaltz | TripleFFFF, how does this relate to your previous question? |
| 01:08.22 | TripleFFFF | yeas |
| 01:08.33 | TripleFFFF | so it can match on the right part |
| 01:08.42 | TripleFFFF | instead of having all my ii12943239408 inbound number duped |
| 01:08.57 | TripleFFFF | for 4445551234,1, blah |
| 01:08.57 | TripleFFFF | <PROTECTED> |
| 01:08.58 | TripleFFFF | et |
| 01:09.44 | *** join/#asterisk DrukenHME (n=jdumais@CPE0040f43870d3-CM00137189cb0c.cpe.net.cable.rogers.com) |
| 01:11.00 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.216.2) |
| 01:11.08 | *** join/#asterisk SwK_ (n=Silik0nJ@12-218-74-89.client.mchsi.com) |
| 01:11.26 | shmaltz | TripleFFFF, I wish you would expain yourself |
| 01:12.08 | *** join/#asterisk UForgotten (i=uforgott@laurel.dreamhost.com) |
| 01:13.05 | *** join/#asterisk asdjkakj1 (n=aaaazz@pool-71-245-225-219.bstnma.fios.verizon.net) |
| 01:15.49 | TripleFFFF | want to be able do match a number with or without the 1 in front in the file extensions.conf.. |
| 01:16.22 | [TK]D-Fender | TripleFFFF : you'll need to do each seperately. There is no regex like that with any practicality to it |
| 01:17.57 | *** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.itb.ac.id) |
| 01:18.22 | UForgotten | doesn't 1|nxxnxxx do that? |
| 01:18.30 | UForgotten | or is that just a freepbx thing? |
| 01:19.20 | shmaltz | UForgotten, even 1nxxnxxx wont match anything |
| 01:21.44 | *** join/#asterisk x86 (n=x86@p3m/member/x86) |
| 01:34.16 | *** join/#asterisk bkruse (n=bkruse@69.73.127.92) |
| 01:34.24 | bkruse | hello there. |
| 01:34.58 | TripleFFFF | ya |
| 01:36.06 | bkruse | hello is all. |
| 01:36.17 | *** part/#asterisk UForgotten (i=uforgott@laurel.dreamhost.com) |
| 01:40.20 | droops | any reason music on hold would whistle |
| 01:40.33 | *** join/#asterisk Strom_C (n=strom@netblock-66-159-243-60.dslextreme.com) |
| 01:40.40 | droops | hey strom |
| 01:40.45 | Strom_C | hello hello |
| 01:40.46 | benjk | um, you gotta select your music differently |
| 01:41.20 | Strom_C | whats up, droops? |
| 01:41.39 | droops | just asking questions in asterisk |
| 01:41.52 | droops | and thanks benjk that worked, i just changed it to files and it works fine |
| 01:42.17 | droops | mode=files not mode=quietmp3 |
| 01:44.44 | bkruse | or loudmp3 if i remember right? |
| 01:44.52 | bkruse | ive never actually tried it |
| 01:45.05 | *** part/#asterisk smackus (n=ckwall@63.149.122.93) |
| 01:46.26 | droops | nope loudmp3 doesnt work |
| 01:46.29 | droops | i jsut tried |
| 01:46.40 | bkruse | hmm |
| 01:47.02 | bkruse | i know it was refernced in oreillys dry manually but, i havent gotten the chance to mess with the newest trunk |
| 01:47.05 | bkruse | but im excited :D |
| 01:50.42 | *** join/#asterisk zotz (n=zotz@24.244.163.225) |
| 01:54.36 | caio1982 | i believe i've asked it in the wrong channel (asterisk-dev) so here it's: is somehow possible to make asterisk send in all dtmfs right after opening a channel, without waiting for the answer signal? |
| 01:55.15 | Strom_C | huh? |
| 01:55.44 | bkruse | caio1982: why would you want to do this(just wondering) |
| 01:56.03 | caio1982 | yeah, weird question... but it's the current situation with a very old "leucotron" pabx |
| 01:56.07 | *** join/#asterisk Qwell (n=north@unaffiliated/qwell) |
| 01:56.07 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 01:56.41 | bkruse | hmm |
| 01:57.02 | Strom_C | so...DID, essentially? |
| 02:01.27 | caio1982 | if it's not possible for some design reason i'll get it :) i'm trying to debug and understand how this "leucotron" is working |
| 02:01.42 | *** join/#asterisk yxa (n=diablo@58.185.90.101) |
| 02:01.47 | caio1982 | i thought that loud about the dtmfs, after reading a email suggesting it |
| 02:01.59 | yxa | given a string of countrycode+areacode+number, how does one query for the phone rate from a table in mysql? |
| 02:02.17 | bkruse | caio1982: im sure u could edit the wait string in asterisk source code to do it |
| 02:02.32 | bkruse | but if you dont know to much C then, good luck, i would do it for you, but i myself dont know enough :[ |
| 02:03.21 | caio1982 | bkruse: is this just a guess or it can be done for real? i'll look for it, but i'm just checking to not waste too much time testing hehe |
| 02:03.48 | bkruse | um, just a guess, but sort of educated |
| 02:04.16 | bkruse | im sure it waits for the return of the off-hook status to send the DTMF, or you know what, i might have a physical representation(depends if its pots, aka electrical, or something like sip) |
| 02:04.26 | caio1982 | i didnt think changing the source code of asterisk because that wouldn't be a very clean solution but... well, i'm updating my sources :) |
| 02:04.59 | bkruse | it would not be clean at all, unless you were a good programmer :] |
| 02:05.05 | caio1982 | heheh |
| 02:05.10 | bkruse | not to mention it probably uses the off-hook type function for alot more so |
| 02:05.14 | bkruse | it would in fact become dirty |
| 02:05.33 | caio1982 | well, dirty things can be fun :) |
| 02:05.41 | caio1982 | thanks for the help bkruse |
| 02:07.16 | *** join/#asterisk spr1te (i=spr1te@194.187.130.227) |
| 02:09.15 | bkruse | np |
| 02:09.30 | bkruse | i think its possible, and i need to brush up on C, so i might look into it :] |
| 02:10.37 | caio1982 | \,,/ |
| 02:16.32 | caio1982 | " D([called][:calling]) - Send the specified DTMF strings *after* the called\n" |
| 02:16.32 | caio1982 | " party has answered, but before the call gets bridged. |
| 02:16.35 | brimstone | ha |
| 02:16.44 | caio1982 | probably that's a good shot |
| 02:16.52 | bkruse | indeed. |
| 02:17.09 | *** join/#asterisk freeepbxxnoobbb (n=chatzill@rrcs-67-52-187-18.west.biz.rr.com) |
| 02:17.17 | Qwell | freeepbxxnoobbb: see topic |
| 02:17.31 | freeepbxxnoobbb | i know |
| 02:17.45 | freeepbxxnoobbb | but this is an asterisk question |
| 02:17.50 | freeepbxxnoobbb | can someone help me out with a problem. My phones wont answer any calls. I can pick up the calls but i can still hear it ringing. |
| 02:18.32 | bkruse | oh great, how generic can u get. |
| 02:19.05 | Strom_C | HELP IT DOESNT WORK OH NO |
| 02:19.20 | Strom_C | i think that's slightly more generic |
| 02:19.21 | Strom_C | :) |
| 02:19.24 | freeepbxxnoobbb | I dont know what i did wrong |
| 02:19.37 | bkruse | Strom_C: u win. |
| 02:19.53 | freeepbxxnoobbb | ??? :( |
| 02:20.38 | freeepbxxnoobbb | incoming calls from outside then there is completely no audio |
| 02:20.58 | bkruse | timing source? |
| 02:21.03 | freeepbxxnoobbb | internal calls then itll still be ringing on the phone when picked up |
| 02:21.07 | bkruse | u have digium hardware/ztdummy? |
| 02:21.07 | freeepbxxnoobbb | zaptel |
| 02:21.10 | *** part/#asterisk TripleFFFF (n=TripleFF@145-27.mc.cite.net) |
| 02:21.15 | freeepbxxnoobbb | x100p |
| 02:21.20 | bkruse | ahh x100p |
| 02:21.21 | bkruse | zaptel is configured correctly? |
| 02:21.34 | freeepbxxnoobbb | 1 channel slave |
| 02:21.59 | freeepbxxnoobbb | is it a zaptel problem |
| 02:22.05 | freeepbxxnoobbb | ? |
| 02:22.09 | bkruse | man it could be alot of things, this is no detail, lets stop flooding here, open a new chat |
| 02:27.13 | *** join/#asterisk ast_freak (n=ast_frea@68-112-130-237.dhcp.stcd.mn.charter.com) |
| 02:28.45 | benjk | freepbxnoobbb, if it is related to freepbx, you will find more help in #freepbx |
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| 02:49.06 | ComputerWarm | hello all anyone here using A2B and actually have it billing the for using a toll free num? |
| 02:51.37 | *** join/#asterisk kratzers (n=kratzers@kratzers.static.pa.net) |
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| 03:16.30 | *** join/#asterisk brodiem (n=brodiem@cpe-66-69-222-36.austin.res.rr.com) |
| 03:17.22 | brodiem | I have a question about re-invites. I know re-invites and NAT don't get along, but is there a problem if there are two SIP endpoints on the same local LAN, with the PBX sitting externally across a NAT? So the re-invite would make the two SIP endpoints on the local LAN on the inside of the NAT talk directly to each other |
| 03:18.20 | bkruse | maybe not, if the port gets poked throught he firewall(or nat) |
| 03:18.28 | bkruse | its the returning packet that is going to get rejected |
| 03:18.39 | bkruse | so you can prolly connect, then ull get owned on the answering part/data transfer |
| 03:18.54 | Strom_M | heheh |
| 03:18.57 | bkruse | there prolly are ways to do it, get creative with ur network, its fun :] |
| 03:18.58 | Strom_M | owned by sip |
| 03:19.00 | bkruse | or be lame and forward 5060 and 10000-20000 |
| 03:19.12 | bkruse | sip > brodiem |
| 03:19.29 | brodiem | lol |
| 03:19.40 | brodiem | sip > brodiem's users |
| 03:19.49 | bkruse | ouch |
| 03:19.56 | bkruse | ya, from what your saying, try what i said |
| 03:20.04 | bkruse | becuase by themself it might poke the nat to do the transaction |
| 03:20.21 | bkruse | but i think the problme is going to be on the answer, the phone will prolly ring, but sending its off-hook status will get owned. |
| 03:20.26 | bkruse | why is it external? |
| 03:20.46 | Strom_M | bkruse: i have a setup like that |
| 03:21.11 | Strom_M | asterisk box is on a public ip, and sip endpoints are behind a nat router |
| 03:21.34 | bkruse | port forwarding do the trick Strom_C/ |
| 03:21.36 | bkruse | ?* |
| 03:22.01 | bkruse | um, without port forwarding, the asterisk box couldnt call on the sip phone, being called, i believe |
| 03:22.10 | Strom_M | ive never bothered with reinvites |
| 03:22.13 | brodiem | I'm still not fully understanding... both SIP endpoints negotiate SIP with the PBX -- the PBX uses its NAT workarounds to talk SIP back to each endpoint and tells each endpoint how to find the other, and the media addresses would then be the private IPs correct? |
| 03:22.23 | Strom_M | i just keep the ports open with qualify=yes |
| 03:22.42 | brodiem | I wanted to use re-invites to prevent local calls from going out into the cloud |
| 03:23.27 | bkruse | Strom_M: in that case, it indeed would work :] |
| 03:23.49 | bkruse | try qualify=yes |
| 03:24.07 | bkruse | from what Strom_M's saying, sounds like, logically, it should work |
| 03:24.15 | bkruse | i would try first without nat, then with |
| 03:24.26 | Strom_M | makeitworkplease=yes |
| 03:24.43 | brodiem | natisapita=always |
| 03:24.51 | Strom_M | haha |
| 03:25.06 | bkruse | natisgreatherthanbrodiem= from yes to no |
| 03:25.06 | Strom_M | er sorry |
| 03:25.08 | bkruse | and u should be good. |
| 03:25.10 | Strom_M | lol=very |
| 03:25.24 | bkruse | honestly, sip isnt meant for nat |
| 03:25.27 | *** join/#asterisk [shodan] (n=shodan@ip016.96-113-216.pppoe1.joliette.intermonde.net) |
| 03:25.28 | bkruse | its lame. |
| 03:27.00 | brodiem | well I'll just give it a try and see what happens |
| 03:28.59 | hmmhesays | eh, sip and nat can work just fine if you do it right |
| 03:29.32 | hmmhesays | or you can sit and complain about it ... which is more fun ;) |
| 03:30.50 | bkruse | haha |
| 03:30.59 | bkruse | hmmhesays: agreed it can, with a lil port forwarding majic. |
| 03:31.05 | bkruse | and some sip.conf tweaking. |
| 03:32.44 | hmmhesays | hello set your sip registrations to the right time and you don't even need that |
| 03:32.51 | hmmhesays | most routers are stateful these days |
| 03:33.08 | hmmhesays | so toss out your netgear from 1997 pay 40 bucks and get something newer if you are having nat issues |
| 03:33.36 | bkruse | its not nat issues |
| 03:33.52 | bkruse | its port problems with qualify, so yes if u set ur timing right, and they kept a connect |
| 03:33.57 | bkruse | the basis is, no inbound connections |
| 03:34.01 | bkruse | unless already connected. |
| 03:34.25 | hmmhesays | if find setting your sip re-register to 60 works way better than qualify |
| 03:34.33 | bkruse | agreed |
| 03:34.56 | bkruse | but it has to hit the asterisk box first, so in this case(asterisk is acessible by the sip phones without nat on the asterisk box) its possible |
| 03:38.06 | hmmhesays | i see |
| 03:38.25 | *** part/#asterisk DasTech (n=DasTech@d47-69-168-46.col.wideopenwest.com) |
| 03:38.31 | bkruse | i think. |
| 03:38.33 | bkruse | :] |
| 03:38.45 | bkruse | but yours right, your way is in fact 1337 for this situation |
| 03:38.48 | bkruse | and frankly, more secure |
| 03:40.03 | Strom_M | what abut nub-proof |
| 03:43.20 | bkruse | def not Strom_M |
| 03:43.36 | Strom_M | hehhe |
| 03:44.54 | hmmhesays | lol |
| 03:45.01 | hmmhesays | the nubs hav egone tooo bed |
| 03:45.07 | hmmhesays | ur 2 funneh |
| 03:47.29 | hmmhesays | who wants to re-design my website? |
| 03:47.41 | hmmhesays | it pretty much blows the proverbial goat |
| 03:47.57 | hmmhesays | www.thelostpacket.org |
| 03:49.19 | hmmhesays | ruh roh |
| 03:49.21 | *** join/#asterisk Ciber311 (n=Ciber311@user-1087e94.cable.mindspring.com) |
| 03:49.21 | *** join/#asterisk jjhall (n=chatzill@72.24.119.202) |
| 03:49.22 | Strom_M | well the questions are |
| 03:49.38 | Strom_M | 1. does the goat have rabies |
| 03:49.42 | bkruse | looking. |
| 03:49.52 | hmmhesays | i think the server got shut off |
| 03:50.07 | hmmhesays | Strom_M: yes |
| 03:50.08 | Strom_M | 2. is the blowing being done for quarters, nickels, or foodstamps? |
| 03:50.14 | hmmhesays | Quarters |
| 03:50.21 | *** join/#asterisk harlequin516 (n=sham@dsl01-ppp-4444.fastq.com) |
| 03:50.25 | hmmhesays | are you calling my goat a cheap whore? |
| 03:50.39 | Strom_M | no no |
| 03:50.45 | jjhall | is there a way to control registration of a sip user via the dialplan? For example, press *11 to make asterisk login to an account, and *12 to logout? |
| 03:50.49 | Strom_M | im calling the /site/ a cheap whore |
| 03:50.59 | hmmhesays | jjhall: yeah you could probably do it |
| 03:51.02 | bkruse | jjhall: im sure there is |
| 03:51.10 | hmmhesays | via the db command |
| 03:51.14 | bkruse | right |
| 03:51.17 | bkruse | make it execute commands |
| 03:51.27 | jjhall | Hmm. Didn't think of that. |
| 03:51.34 | hmmhesays | I make asterisk talk dirty to me |
| 03:51.40 | bkruse | voip-info it |
| 03:51.44 | hmmhesays | cc pick up that guitar and talk to me |
| 03:51.44 | harlequin516 | What can I do to read a single dtmf char to a variable from the channel from the dialplan? Read will not let me read a '#', but works for everything else. |
| 03:51.47 | bkruse | hmmhesays:i designed that app... |
| 03:52.03 | hmmhesays | bkruse: haha |
| 03:52.12 | hmmhesays | that's because read is # terminated |
| 03:52.17 | jjhall | Basically I have a call queue I want to log into. It isn't a real queue, just a SIP account that allows multiple logins. They all ring at once. I want to be able to login and logout on a whim. |
| 03:52.28 | bkruse | harlequin516:cant you just do dtmf(whatever asterisk uses for variables) |
| 03:52.46 | bkruse | jjhall: hmm should be possible |
| 03:52.48 | *** join/#asterisk r0d3nt (n=RatMan@foster.stonedcoder.org) |
| 03:52.50 | bkruse | with some dialplan majic |
| 03:53.02 | hmmhesays | jjhall: just use a real queue |
| 03:53.10 | hmmhesays | with ringall strategy |
| 03:53.31 | bkruse | i wish i could help more but, i have to go do pre-cal homework, and jjhall and harlequin516: get here earlier and i might be able to design a rela basic, proof of concept dialplan |
| 03:53.54 | jjhall | I don't have control over it, otherwise I would. I use Asterisk at home, and want to login to the remote queue instead of just programing an extra port on my ATA as they suggested I do. |
| 03:54.20 | hmmhesays | just because you're using aah doesn't mean you don't have control |
| 03:54.21 | hmmhesays | wtf |
| 03:54.34 | jjhall | bkruse: Thanks for the help. Hopefully I'll be able to come up with something and save you the trouble. Have fun with the homework! |
| 03:54.45 | hmmhesays | the *_custom.conf files give you all the control you need to do anything |
| 03:55.04 | bkruse | jjhall: thanx, high school is a drag, a@h sucks, svn on a stable debian system, save yourself trouble |
| 03:55.11 | bkruse | i should be able to help u a lil tomorrow |
| 03:55.14 | hmmhesays | a@h doesn't suck |
| 03:55.25 | hmmhesays | it sucks for nubs that don't know the full potential |
| 03:55.29 | jjhall | hmmhesays: You aren't understanding. I use real asterisk for my phone system at home. The remote "queue" is not something I have admin over. I just register as a user when I need to. See what I am saying? This isn't a "can't configure without GUI" issue. |
| 03:55.35 | hmmhesays | ahh |
| 03:56.03 | hmmhesays | so the remote queue sends you a call when you are registered to the far end box? |
| 03:56.03 | bkruse | gotcha. |
| 03:56.09 | harlequin516 | Anyone have an idea? The agi commands can do this easily.. I wonder why asterisk doesn't provide these functions fromthe dialplan? |
| 03:56.18 | EzWayz | cmd system still work in 1.2.11 asterisk ? |
| 03:56.27 | bkruse | lol, should, i hope |
| 03:56.32 | harlequin516 | I just want to read an # with a 2 second timeout. |
| 03:56.36 | jjhall | hmmhesays: Exactly. |
| 03:56.47 | bkruse | jjhall: agentlogin, and agentmemberlogin? or w;e the 2 differnet times |
| 03:57.44 | hmmhesays | harlequin516 exten => _X.,1,Background(tt-monkeys); exten => _X.,n,WaitExten(2); exten => #,1,NoOP(DARN DEM DUKE BOYS) |
| 03:58.11 | harlequin516 | Is there maybe a bridge command that allows you to execute and retrive the result of an agi command? |
| 03:58.32 | hmmhesays | i just showed you how to read a # and do something with it |
| 03:58.57 | harlequin516 | hmmhesays: Ah I see.. |
| 03:59.20 | harlequin516 | hmmhesays: Not as elegant as I was hoping for, but I think it does what I asked. |
| 03:59.20 | hmmhesays | jjhall: just set your dial plan so that if global var ${GOAWAY} = 1 go to congestion |
| 03:59.43 | hmmhesays | what's not elegant? the screaming monkeys sound file or the reference to dukes of hazzard |
| 04:00.05 | hmmhesays | you see smell what i'm cooking jjhall? |
| 04:00.14 | jjhall | hmmhesays: Will that leave the calls to go to another user currently logged in or will it give the caller a fast-busy? |
| 04:00.14 | hmmhesays | geebus its getting late |
| 04:00.25 | jjhall | And yes I see where you are going with it. :-) |
| 04:00.40 | hmmhesays | jjhall if the far end box gets congestion it is not going to connect the call |
| 04:00.47 | harlequin516 | So WaitExten and Read are the only two commands that can retrieve DTMF signalls? |
| 04:01.08 | hmmhesays | harlequin516: i have no idea, thats just how I would do it |
| 04:01.11 | harlequin516 | sorry I meant "the only two applications" |
| 04:01.54 | hmmhesays | jjhall: if they are using app queue or just multiple endpoints in the dial command if you return congestion or busy or anything except answer they will just ignore your endpoint |
| 04:02.19 | jjhall | hmmhesays: and it should failover to another working extension. The only issue I could see there is the remote user only accepts a limited number of registrations at a time and I could potentially keep someone else from logging in. |
| 04:02.43 | jjhall | I'll have to do some more brainstorming. |
| 04:02.49 | hmmhesays | so you want to actually un=register from there end |
| 04:02.58 | hmmhesays | comment out your register line and sip reload |
| 04:03.23 | hmmhesays | that would be cake if you are using an include |
| 04:03.28 | hmmhesays | with one line it it |
| 04:03.42 | jjhall | hmmhesays: Yes, I need to unregister. |
| 04:03.54 | hmmhesays | you could just unregister and sip reload |
| 04:04.22 | jjhall | hmmhesays: Interesting idea. |
| 04:04.28 | hmmhesays | two 2 line bash scripts and 2 cmd systems in your dp |
| 04:04.49 | EzWayz | there is a good link talking integration nortel with asterisk |
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| 04:05.39 | EzWayz | ? |
| 04:05.42 | hmmhesays | have a sip_register.conf include it in sip.conf and just overwrite it with a blank file when you want to unregister and aterisk -rx 'sip reload' |
| 04:05.44 | hmmhesays | bam done |
| 04:05.46 | jjhall | hmmhesays: I could use the entire registration for this user in its own include called "queue.on" The batch file could rename it to queue.off and sip reload, and vise versa to logon. |
| 04:06.00 | hmmhesays | why rename |
| 04:07.05 | hmmhesays | well yeah... my bad... have sip_register.conf included in sip.conf and in your bash script rm sip_register.conf; cp queue.on sip_register.conf; asterisk -rx 'sip reload' |
| 04:07.12 | hmmhesays | purdy simple |
| 04:07.18 | jjhall | Because I want to be able to logon using the same method. Asterisk should ignore (well, throw a warning) if it can't find the correct file, right? |
| 04:07.43 | jjhall | See, we were on the same page, just a different paragraph. :-) |
| 04:07.46 | hmmhesays | so you have queue.on with register line and queue.off blank |
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| 04:08.10 | jjhall | That should work just fine. Thanks for the idea! |
| 04:08.15 | hmmhesays | so when you wanted to log out rm sip_register.conf; cp queue.off sip_register.conf; asterisk -rx 'sip reload' |
| 04:08.25 | hmmhesays | np, feel free to donate money to me for it |
| 04:08.27 | hmmhesays | :D |
| 04:08.47 | J4k3 | is it my imagination or is trixbox really broken when it comes to handling fxo cards? |
| 04:08.49 | jjhall | If I weren't currently inbetween jobs due to a "corporate right-sizing" I would. :-) |
| 04:09.19 | hmmhesays | yeah I'm in between jobs because the company I worked for was insane and kept on contracting me out for free |
| 04:09.30 | jjhall | Nice! |
| 04:09.45 | jjhall | I had an interview today that I'm pretty confident about, but only time will tell. |
| 04:09.56 | hmmhesays | I have some contracts i'm working on |
| 04:10.06 | bkruse | hmmhesays: oh rlly. |
| 04:10.14 | bkruse | jjhall: where at? for what position/ |
| 04:10.21 | sephiro499 | Man I'm looking for a job too... |
| 04:10.41 | hmmhesays | bkruse yeah |
| 04:10.57 | jjhall | bkruse: I was in the custom engineering dept for MPC Computers (formerly MicronPC.) I'm going after an IT position for a local farm supply store. |
| 04:11.17 | hmmhesays | jjhall: kind of a kick down |
| 04:11.29 | hmmhesays | jjhall: you ever work with uclinux on mipsel? |
| 04:11.41 | hmmhesays | its kicking my arse right now |
| 04:11.56 | bkruse | yay for linux! |
| 04:12.01 | bkruse | hmmhesays: whats the problem? |
| 04:12.22 | hmmhesays | the guy i'm working with is semi-retarded when it comes to human contact |
| 04:12.36 | hmmhesays | so I keep getting mis information about the platform |
| 04:12.45 | hmmhesays | which leads to a lot of segfaults in my binaries |
| 04:12.47 | jjhall | hmmhesays: Not really a kickdown in my opinion. It appears to actually pay better, and sounds like it will give me more flexibility to play with various technology than I had before. |
| 04:12.58 | jjhall | hmmhesays: And no, never worked with that at all. |
| 04:13.20 | hmmhesays | which intern results in me drinking more beer and playing more guitar instead of working |
| 04:13.36 | jjhall | Plus I miss the days when I worked for a smaller company without the corporate attitudes in over-abundance. |
| 04:13.47 | bkruse | sweet |
| 04:13.49 | benjk | dont know, Monica Lewinsky perhaps |
| 04:13.49 | hmmhesays | I worked for a dinky company that thought they were big |
| 04:13.50 | bkruse | sounds like a good plan |
| 04:14.22 | hmmhesays | now they are going to fail miserably because no one in fargo can even come close to my level of expertise (which isn't really that hard) lol |
| 04:14.32 | jjhall | I also applied to be a restaurant manager for the local Taco Bell franchisee. When i got laid off I was just tired of the tech industry in general, but I've come to my senses. :-) |
| 04:14.47 | hmmhesays | jjhall: I am burned out, that's why i'm starting a bar band |
| 04:15.02 | bkruse | jjhall: im glad u did. |
| 04:15.04 | jjhall | There you go! |
| 04:15.09 | hmmhesays | I can subside on 500 bucks a week for 10 hours of work |
| 04:15.12 | hmmhesays | for awhile |
| 04:15.22 | jjhall | This company has been around since the early 50s, and seems to have level heads in charge. |
| 04:15.44 | jjhall | I grew up on a farm and remember making constant trips to their stores as a kid. |
| 04:15.48 | hmmhesays | especially when that work involves me (24) playing guitar for many screaming young drunken college hotties |
| 04:15.59 | jjhall | Fringe benefits anyone? LOL |
| 04:16.38 | hmmhesays | free beer, chicks screaming at you (in a good way) and getting paid to do it.... downside? sometimes you have to play in shit holes |
| 04:16.51 | jjhall | Yeah, there are some real dives out there for sure. |
| 04:16.56 | hmmhesays | the pros FAR out weigh the cons though |
| 04:17.06 | hmmhesays | yeah but then sometimes you get gigs in places like this |
| 04:17.10 | hmmhesays | www.playmakersfargo.com |
| 04:17.17 | jjhall | I worked in radio while in high school and shortly after. I know that scene all to well. :-) |
| 04:18.25 | jjhall | I'm actually pretty excited about this job. The office is about 2.5 miles from home, so I can ride my bike to work. It involves some travel to support their stores, but they are all within a 3 hour drive and they provide the vehicle. |
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| 04:18.47 | hmmhesays | but that said, i'm out for the night, bkruse, nice chatting with you, jjhall keep in touch, I do a lot of inter office telephony type stuff for multi site locations |
| 04:18.50 | *** join/#asterisk r0d3nt (n=RatMan@foster.stonedcoder.org) |
| 04:19.07 | jjhall | hmmhesays: Good to know. Have a good night and don't wake the neighbors! :-) |
| 04:19.27 | hmmhesays | haha, its one more beer and an episode of smallville for me |
| 04:19.45 | jjhall | I'm waiting for the next season (if they do one) of Beauty and the Geek. |
| 04:19.56 | *** join/#asterisk Vco (n=Vco@S01060050da6df072.sc.shawcable.net) |
| 04:20.03 | jjhall | Eureka and the Stargate series are my favorites right now. |
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| 04:29.59 | nvzn | using trixbox, incoming calls do not work unless i enable anonymous sip callers in freePBX |
| 04:30.22 | nvzn | i guess its a problem with my incoming section in the Trunk settings |
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| 04:30.43 | hads | <PROTECTED> |
| 04:31.11 | nvzn | cool |
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| 05:22.02 | Strom_C | woah woah woah |
| 05:22.08 | Strom_C | since when did you become a Sprint salesperson? |
| 05:22.23 | Qwell | grr |
| 05:22.33 | Qwell | now I have to remember those stupid commercials |
| 05:22.55 | Juggie | hah |
| 05:23.08 | Strom_C | 1-800-PIN-DROP |
| 05:24.28 | Strom_C | heh...it's a CAS T1 |
| 05:24.41 | Strom_C | so really its more like "so whiny, you can't hear a pin drop" |
| 05:28.40 | *** join/#asterisk mitcheloc (n=mitchelo@titaniumsoft.net) |
| 05:28.44 | Qwell | I love how that number takes 1:40 before you timeout for not pressing keys |
| 05:34.39 | sx-wks | Strom_C: lol |
| 05:35.08 | sx-wks | Strom_C: what's a CAS T1 ? |
| 05:35.55 | Strom_C | channel associated signaling T1 |
| 05:36.17 | sx-wks | aka bit robbing ? |
| 05:37.48 | sx-wks | ah HA... |
| 05:38.06 | Strom_C | a-ha? |
| 05:38.11 | file | take on me! |
| 05:38.39 | sx-wks | when I pick up my phone on Zap/1 , the background to the 1 on the screen turns green. and back to blue when I hangup :D |
| 05:38.44 | x86 | omg |
| 05:38.50 | x86 | i think i should call the FBI or something |
| 05:39.06 | x86 | i think there's a terrorist in my kitchen by the name of Pepper al-Habenero |
| 05:39.10 | sx-wks | yippie |
| 05:39.24 | sx-wks | x86: wtf ? |
| 05:39.26 | x86 | terrorized my insides |
| 05:39.50 | file | x86: uh huh |
| 05:40.27 | file | x86: http://www.hotsauceworld.com/satansblood.html |
| 05:40.27 | file | ...go for that |
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| 05:40.34 | sx-wks | ROTFL !! http://yro.slashdot.org/article.pl?sid=06/09/05/2344247 |
| 05:45.16 | Qwell | sx-wks: good ole Mark Stumpf |
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| 05:52.48 | *** join/#asterisk rift0r (i=[U2FsdGV@207.44.158.6) |
| 05:53.33 | rift0r | Hi, I am looking for an analog adapter with 1 or more FXS ports that is stand alone... anyone have any recommendations? |
| 05:53.52 | rift0r | i see tons of them out here I was just curious if anyone had any opinions on them |
| 05:57.21 | rift0r | is the HandyTone 496 any good? |
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| 05:59.18 | *** mode/#asterisk [+o Corydon-w] by ChanServ |
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| 06:01.11 | RoyK | mrnng |
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| 06:07.13 | x86 | rift0r: i've had good luck with the HT386 |
| 06:07.57 | x86 | rift0r: a lot of people will tell you to get a Linksys / Sipura adaptor though |
| 06:08.22 | x86 | the PAP2 is supposed to be really nice, but it's a bit more than the grandstream adapters |
| 06:08.24 | Qwell | I want to get my hands on an ATA186 |
| 06:09.59 | rift0r | x86 what advantages do the linksys/sipura adapters have over the grandstream? |
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| 06:18.28 | shodan | in canada , is roger wireless cell phones , gsm or cdma ? |
| 06:19.45 | Juggie | both |
| 06:19.53 | Juggie | all new phones are gsm |
| 06:19.59 | Juggie | but they still have a cdma or tdma network |
| 06:20.03 | Juggie | i'm not sure which |
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| 06:22.42 | shodan | k, my dad got a razr from them a while ago , wasn't sure if it would work with those gsm gateway thingies |
| 06:29.30 | shodan | are there any other gsm service provider in canada ? wikipedia says they're the only one ? |
| 06:29.57 | Strom_C | what, telus / bell canada arent gsm? |
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| 06:41.21 | shodan | nope they're both cdma |
| 06:41.48 | *** join/#asterisk pbx1 (i=pbx1@netblock-66-245-193-38.dslextreme.com) |
| 06:41.50 | shodan | are there any cdma gateways ? |
| 06:41.54 | docelmo | Pull up gsm.com |
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| 06:42.50 | docelmo | http://www.gsmworld.com/roaming/gsminfo/cou_ca.shtml |
| 06:42.57 | docelmo | GSM providers in Canda |
| 06:42.58 | *** join/#asterisk kwagga (n=corne@196.211.32.90) |
| 06:43.01 | docelmo | Canada |
| 06:48.40 | docelmo | Roger's Cell plans suck ass |
| 06:52.01 | shodan | thanks |
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| 06:52.41 | docelmo | no prob |
| 06:53.07 | docelmo | I love my Tmobile.. $100 a month 3000 minutes 3 lines unlimited cell to cell free unlimited nights/weekends |
| 06:53.36 | nick125_lappy | lol |
| 06:53.41 | nick125_lappy | my cell phone bill is $190 :/ |
| 06:53.50 | shodan | o_O |
| 06:53.58 | docelmo | You're getting screwed by someone |
| 06:54.06 | tengulre | anybody can give me a free g729 codecs? |
| 06:54.08 | docelmo | I just upgraded to 3000 and saved like 200 a month |
| 06:54.11 | nick125_lappy | 4 lines, 2000 minutes, unlimited n/w |
| 06:54.19 | docelmo | tengulre ya go download one |
| 06:54.43 | tengulre | docelmo, where can download one? |
| 06:54.45 | docelmo | nick125_lappy who's yer provider? |
| 06:54.49 | nick125_lappy | docelmo: t-mobile |
| 06:54.54 | docelmo | there was a post about it a couple days ago |
| 06:54.59 | docelmo | nic seriously? |
| 06:55.07 | docelmo | wow.. I would rethink that one |
| 06:55.13 | docelmo | I just upgraded to 3000 |
| 06:55.25 | shodan | why do you need minutes if you have unlimited cell to cell ? just use a gsm gateway ? |
| 06:55.34 | nick125_lappy | I got unlimited SMS ($10) and unlimited GPRS ($20), plus insurance on my line ($6) |
| 06:56.11 | docelmo | shodan if I could get my hands on one I would |
| 06:57.29 | shodan | http://cgi.ebay.ca/Tri-Band-GSM-Cellular-Terminal-Gateway-IP-PBX-VOIP-GPRS_W0QQitemZ110027481018QQihZ001QQcategoryZ61839QQssPageNameZWDVWQQrdZ1QQcmdZViewItem |
| 06:57.44 | shodan | 230$usd buy it now |
| 06:59.53 | shodan | why do you need insurance ?! |
| 07:00.06 | docelmo | for my reason.. My PEBL fell in water.. |
| 07:00.11 | docelmo | Parts of my phone work now |
| 07:00.18 | docelmo | its a $400 USD phone |
| 07:00.31 | nick125_lappy | I think insurance on my $600 treo would be a good idea.. |
| 07:00.34 | shodan | oh ok it's the phone that's insured |
| 07:02.57 | docelmo | You have to do it in 15 days of handset purchase |
| 07:06.05 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 07:10.01 | hads | You guys in the US are charged for minutes when people call you correct? |
| 07:10.31 | Strom_C | on mobile phones? yes |
| 07:11.03 | Qwell | except, commonly, people on the same provider |
| 07:11.12 | hads | Yeah sorry, mobiles. That seems odd :) |
| 07:11.32 | Strom_C | but, conversely, the calling party is charged the same rate as landline calls |
| 07:11.32 | JT | america is odd |
| 07:11.37 | *** join/#asterisk Aurs (n=Aurs@host-81-191-123-189.bluecom.no) |
| 07:12.29 | hads | Strom_C: Ah, that means it makes slightly more sense. |
| 07:13.16 | *** join/#asterisk juanjoc (n=juanjoc@248-32-235-201.fibertel.com.ar) |
| 07:13.39 | Strom_C | so after my extensive scientific comparison of the audio quality of a Cisco 7960 and a Polycom IP430, I have found the winner |
| 07:13.41 | *** join/#asterisk oomph (n=oomph@69-175-199-236.chvlva.adelphia.net) |
| 07:13.52 | Qwell | yay cisco |
| 07:13.56 | Strom_C | CISCO!!!! |
| 07:14.01 | oomph | hello |
| 07:14.02 | Qwell | of course |
| 07:14.04 | pbx1 | Is there a way to have the queue_log relect the time an agent puts a user on hold? |
| 07:14.09 | Qwell | polycom probably cheaped out, heh |
| 07:14.12 | Strom_C | yep |
| 07:14.16 | oomph | anyone know of a good billing/calling card system that works with asterisk? |
| 07:14.26 | Strom_C | a2billing? astcc? |
| 07:14.45 | oomph | Strom_C have you used any of them? i heard a2billing as kinda buggy |
| 07:14.50 | pbx1 | I'm parsing the queue_log myself with a python script |
| 07:14.51 | oomph | not tried it yet though |
| 07:14.57 | Strom_C | havent tried either |
| 07:15.02 | pbx1 | I just need to know where it is on the log |
| 07:15.24 | shodan | hmm , the only gsm provider in canada are rogers and fido(owned by rogers) |
| 07:15.29 | shodan | not much of a choice |
| 07:16.43 | pbx1 | I see this COMPLETEAGENT(holdtime|calltime|origposition) but that hold time is the wait time before a caller answers |
| 07:17.03 | pbx1 | I didn't find a place that has the time when the agent actually puts the user on hold |
| 07:17.21 | Strom_C | i dont know if thats a metric you can even obtain, pbx1 |
| 07:17.33 | pbx1 | that's what I was afraid of |
| 07:17.48 | Strom_C | of course, i may be wrong |
| 07:18.07 | pbx1 | yeah, I"m stuck on that now |
| 07:18.14 | pbx1 | and one of my clients wants that badly |
| 07:18.38 | Strom_C | i suppose there is someone at digium who knows |
| 07:19.32 | pbx1 | I guess I could try emailing digium |
| 07:19.54 | Strom_C | that works |
| 07:20.12 | pbx1 | thanks :) |
| 07:20.31 | docelmo | You may have to do something custom. I dont think app_queue reports when a call goes on hold. Just start and end |
| 07:20.52 | pbx1 | so app_queue.c would be the file to mess with? |
| 07:20.53 | docelmo | I would imagine it might be fairly simple to do it. |
| 07:21.02 | docelmo | yes |
| 07:21.11 | docelmo | you're wanting queueing information right? |
| 07:21.18 | pbx1 | yes |
| 07:21.33 | pbx1 | like what's in queue_log |
| 07:21.41 | docelmo | then yes you want app_queue.. |
| 07:21.47 | docelmo | Do you know C |
| 07:22.02 | pbx1 | I'm a bit rusty, but It'll come back to me |
| 07:22.46 | docelmo | Well if you need help Im on here almost 24 hours a day |
| 07:23.01 | docelmo | Im also one of the more inexpensive consultants.. :P |
| 07:23.11 | pbx1 | cool |
| 07:23.17 | Qwell | s/inexpensive/cheap/ |
| 07:23.20 | pbx1 | well if it comes down to it I'll put money down |
| 07:23.43 | docelmo | Storm is one of the more expensive.. :) |
| 07:23.55 | Strom_C | I also destroy houses |
| 07:23.56 | Strom_C | NOT |
| 07:23.58 | pbx1 | oh, good to know :) |
| 07:24.01 | Strom_C | I'M NOT STORM |
| 07:24.04 | Strom_C | gah |
| 07:24.05 | Qwell | yeah, app_queue doesn't log hold |
| 07:24.15 | docelmo | sorry type too fast |
| 07:24.16 | pbx1 | oh well |
| 07:24.19 | Strom_C | :) |
| 07:24.45 | pbx1 | Guess back to emailing digium |
| 07:24.59 | *** join/#asterisk DrukenHME (n=jdumais@CPE0040f43870d3-CM00137189cb0c.cpe.net.cable.rogers.com) |
| 07:25.01 | docelmo | hehe or a consultant that couple probably do it now.. :) |
| 07:25.11 | docelmo | I accept paypal.. :P |
| 07:25.31 | pbx1 | ok, well let me know if you can pull it off and if the price is right, we can talk more |
| 07:25.33 | pbx1 | :) |
| 07:25.41 | docelmo | question is how much of a hack would it take to make it happen |
| 07:25.55 | docelmo | lemme go find the code.. Do you want this for 1.2 or 1.4? |
| 07:26.18 | docelmo | Cause just so you know 1.2 modules are not compatible with 1.4 |
| 07:26.18 | pbx1 | <PROTECTED> |
| 07:26.35 | docelmo | ok.. I think I have 1.2.10 I will have to download that one |
| 07:26.41 | docelmo | but lemme have a look see at the one I have' |
| 07:27.52 | *** join/#asterisk inspired (n=mikael@85.221.0.46) |
| 07:32.36 | *** join/#asterisk Niklas- (n=Niklas@213.237.44.34) |
| 07:33.05 | Niklas- | Hi. Isn't it possible to use variabels for an extension? Like "exten => $FOOBAR,1,...." ? |
| 07:33.09 | Qwell | no |
| 07:33.15 | Niklas- | ok |
| 07:33.17 | Niklas- | thanks |
| 07:33.27 | docelmo | ok I found the code.. Just need to figure out how to pull the hold state out now.. |
| 07:33.48 | pbx1 | how's it look? |
| 07:34.04 | docelmo | I will refrain from how I feel about the asterisk code base |
| 07:34.11 | docelmo | but overall promising |
| 07:34.23 | pbx1 | heh |
| 07:37.15 | shodan | hmm just how hard is it to read/write/clone sim cards ? |
| 07:37.34 | docelmo | with the right hardware simple |
| 07:37.39 | docelmo | the software is the kicker |
| 07:37.53 | Strom_C | hell, the hardware costs $30 |
| 07:38.02 | Strom_C | not that I know anything about smartcards, of course |
| 07:39.38 | shodan | is that forbidden ? (to clone a sim card and use it in another phone) |
| 07:39.40 | Qwell | Strom_C: we need to find a bar that uses smartcard credits |
| 07:39.49 | Strom_C | hahaha |
| 07:39.58 | Strom_C | "Excuse me, I'd like to open a tab on my Kinko's card" |
| 07:40.03 | shodan | like if both phones are mine and I'll be the only one using them |
| 07:40.24 | Qwell | not QUITE what I was thinking |
| 07:40.30 | Strom_C | oh? well what quite was you thinking? |
| 07:40.43 | Qwell | Joes Bar card |
| 07:40.59 | Qwell | I doubt there are any places that do that though, heh |
| 07:42.24 | Strom_M | ooh, this is still going |
| 07:42.26 | Strom_M | sweet |
| 07:42.47 | Strom_M | i can stay on the channel AND go get a cold drink |
| 07:42.53 | Strom_M | muahahaha |
| 07:44.16 | docelmo | pbx1 well here is what I can tell so far.. app_queue isnt where the hold state would be found. It would have to be somewhere else. It would be such a hack its not worth it for me to do it. I suggest getting a quote from digium |
| 07:44.55 | docelmo | after it bridges the call it exits the queue.. So it would be interesting to say the least on how to make that happen |
| 07:45.15 | docelmo | But they actually wanna know how long they were on hold for? Ive never heard that request for |
| 07:45.17 | *** join/#asterisk juanjoc (n=juanjoc@248-32-235-201.fibertel.com.ar) |
| 07:45.18 | docelmo | before* |
| 07:45.48 | Strom_C | yeah, thats odd |
| 07:45.59 | Strom_C | even when I worked in a call center, they never cared about that metric |
| 07:46.00 | stoffell | anyone familiar on receiving telco error messages instead of just hangup_causes? (recently posted to mailing list also) |
| 07:46.11 | *** join/#asterisk A-Tuin (n=a-tuin@steves.ip.v4.me.uk) |
| 07:46.14 | Strom_C | it was just counted as part of your call time |
| 07:46.19 | Strom_C | stoffell: what do you mean |
| 07:46.21 | docelmo | stoffell care to explain? |
| 07:46.27 | docelmo | Strom_C yep |
| 07:46.29 | pbx1 | docelmo: ah, I see. That's too bad |
| 07:46.59 | pbx1 | <PROTECTED> |
| 07:47.14 | docelmo | pbx1 Im not saying it cant be done.. but in all honesty its not worth what digium would charge for for it. I would guess as a rough guesstimate somewhere around 500-1000 buks to make it happen |
| 07:47.30 | stoffell | i'm happy to explain; i would like to receive the telco error when dialing wrong numbers on an E1 (or ISDN BRI), now I just get the hangup_cause and I should play my own voice prompts. with the traditional pbx, I always heard the telco error messages.. |
| 07:47.37 | docelmo | tell em to monitor the calls.. :) |
| 07:47.38 | pbx1 | docelmo: I see |
| 07:47.54 | stoffell | i have tried playing with priindication, but that made no difference.. |
| 07:47.59 | pbx1 | yeah, that's what we said so far |
| 07:48.01 | Strom_M | stoffell: oh, you want the RECORDING |
| 07:48.07 | docelmo | haha.. he's talking about early media |
| 07:48.11 | docelmo | which asterisk sucks at |
| 07:48.12 | stoffell | Strom_M: yes, the telco- recording |
| 07:48.13 | pbx1 | docelmo: but they'd like a hard number |
| 07:48.44 | pbx1 | oh well, thanks. I'll have to see what digium tells me and see if it's worth it. |
| 07:48.55 | docelmo | pbx1 see how much its worth to them.. :) |
| 07:49.47 | pbx1 | right :) |
| 07:49.49 | docelmo | stoffell what your looking for is early media pass thru.. Asterisk will not do it based on how it does RTP |
| 07:50.06 | docelmo | Man I have to be at my office in 5 hours.. This sucks |
| 07:50.12 | Strom_M | docelmo, what are you talking about |
| 07:50.17 | Strom_M | docelmo' |
| 07:50.19 | Strom_M | er |
| 07:50.34 | Strom_M | it worls just fine on pris here in california |
| 07:50.39 | docelmo | the 'recordings' he is looking for is called early media |
| 07:50.50 | docelmo | its the "Telco" voice |
| 07:50.50 | Strom_M | i know that |
| 07:51.05 | docelmo | over sip it doesnt |
| 07:51.10 | stoffell | docelmo: hm, a new word to me, i'll try google on that also |
| 07:51.19 | docelmo | what early media? |
| 07:51.27 | Strom_M | um, worls fine for me on my box here with sip phones... |
| 07:51.38 | docelmo | hmmm |
| 07:51.52 | pbx1 | thanks again docelmo |
| 07:51.57 | docelmo | maybe they fixed it.. My cluster is still running 1.2.4 I think |
| 07:52.08 | stoffell | i also use sip phones to call out to the zap channels, but I guess it's an asterisk thingy/setting/..? |
| 07:52.10 | docelmo | it doesnt work there |
| 07:52.36 | stoffell | i'm running 1.2.10, and it doesn't work, unless i need a setting to make it work :) |
| 07:52.53 | Strom_M | the problem is that if the network is sending a hangupcause all the way back to the cpe rather than doing a Proceeding and a recording followed by a disconnect... |
| 07:53.10 | Strom_M | i blame the pstn |
| 07:53.27 | docelmo | stoffell got a firebird? |
| 07:53.44 | docelmo | err Tbird? |
| 07:53.50 | docelmo | I need to goto bed |
| 07:53.58 | stoffell | docelmo: thunderbird yes (email) :) |
| 07:54.10 | docelmo | no tbird |
| 07:54.14 | docelmo | BIG DIFFERENCE |
| 07:54.20 | stoffell | docelmo: nope.. not that i know of :) |
| 07:54.41 | *** join/#asterisk my007ms (n=my007ms@217.139.224.194) |
| 07:54.47 | docelmo | cause with that you could generate a call and see everything they are sending you to see if you need to call the telco and bitch at them |
| 07:55.12 | my007ms | good day all |
| 07:55.13 | docelmo | but yes.. Normally you would get progress(ring) then early media if there is a problem |
| 07:55.23 | stoffell | docelmo: ah, i see.. but wiring up the 'traditional' pbx gives the expected behaviour (telco-voice) |
| 07:55.27 | Strom_M | progress != ring |
| 07:55.30 | docelmo | well progress could be ring or early media |
| 07:55.35 | Strom_M | alerting == ring |
| 07:55.45 | docelmo | really? |
| 07:55.49 | stoffell | maybe i can see them if I do zap debugging? |
| 07:55.52 | docelmo | I always saw progress |
| 07:55.58 | docelmo | yep |
| 07:56.01 | docelmo | should be able to |
| 07:56.07 | docelmo | PRI? |
| 07:56.20 | dorel__ | "/bin/sh /usr/sbin/safe_asterisk -p -U asterisk" is the process that runs asterisk right? |
| 07:56.22 | Strom_M | well, in q931 it should be an alerting message for ring |
| 07:56.42 | my007ms | i have make 2 asterisk server one primary and othere backup one abut as i need to run asterisk in havy duty system i need to test mine asterk for how man call i can make over sip at the same time is there tool do that |
| 07:56.43 | stoffell | docelmo: tested on BRI and PRI (different locations) |
| 07:56.43 | docelmo | dorel__ dude.. just run safe_asterisk |
| 07:56.46 | docelmo | make life simple |
| 07:56.56 | docelmo | stoffell enjoy q931 debugging |
| 07:57.04 | dorel__ | docelmo: what do you mean "just" safe_asterisk? |
| 07:57.12 | stoffell | docelmo: thanks for the tips, will try playing with that a bit |
| 07:57.20 | docelmo | type safe_asterisk and pound the enter key |
| 07:57.33 | docelmo | tank Strom_C also.. :) |
| 07:57.52 | Strom_M | youre welcome :) |
| 07:58.00 | docelmo | ok Im going to bed.. |
| 07:58.06 | dorel__ | docelmo: oh, that's obvious but its running through init.d scripts |
| 07:58.26 | docelmo | add it to rc.local |
| 07:58.39 | docelmo | your running a redhat flavor right? |
| 07:58.39 | *** join/#asterisk THX2000 (i=AgentFLY@adsl-66-51-192-221.dslextreme.com) |
| 07:58.48 | stoffell | yeah, Strom_M too ;) |
| 07:58.53 | my007ms | is there tool make many dumy call |
| 07:59.04 | docelmo | .call or dial |
| 07:59.08 | docelmo | you choose |
| 08:00.06 | THX2000 | Is there a way to change the value of the variables in [globals] from within the dial plan, and have them stick on a reload? |
| 08:00.45 | *** join/#asterisk arcy (n=arcanum@ppp171-77.adsl.forthnet.gr) |
| 08:00.58 | docelmo | THX2000 no |
| 08:00.59 | Strom_M | yeah |
| 08:01.03 | Strom_M | er |
| 08:01.04 | Strom_M | no |
| 08:01.08 | THX2000 | :( |
| 08:01.09 | Strom_M | ) |
| 08:01.20 | docelmo | unless you write them to the extensions.conf file |
| 08:01.21 | THX2000 | hehe, i liked strom's first answer better :P |
| 08:01.27 | Qwell | losing banking benefits sucks...immensely |
| 08:01.36 | Strom_M | ?? |
| 08:01.39 | Qwell | no more free..everything..at WF |
| 08:01.41 | docelmo | ya.. ?? |
| 08:01.46 | docelmo | WF? |
| 08:01.52 | Qwell | my former employer :D |
| 08:01.54 | Strom_M | wells fargo |
| 08:01.58 | docelmo | ohh |
| 08:02.03 | docelmo | use usaa.com |
| 08:02.04 | Qwell | I got all my banking stuff for free |
| 08:02.05 | docelmo | :) |
| 08:02.06 | docelmo | I do |
| 08:02.13 | Qwell | like...everything |
| 08:02.17 | docelmo | ya.. USAA also |
| 08:02.18 | docelmo | :) |
| 08:02.20 | Qwell | notary services, cashiers checks, etc |
| 08:02.27 | docelmo | I have a 100% free account |
| 08:02.43 | Qwell | but I could go into a branch, and say "I want you to do this", and they'd do it..free |
| 08:02.46 | Strom_M | ive been a happy wells customer since 1983 |
| 08:02.57 | Qwell | Strom_M: aren't you only like...20ish? |
| 08:03.03 | Strom_M | back when i was a first interstate customer :) |
| 08:03.04 | docelmo | You choose to work for Digium.. :) |
| 08:03.07 | Strom_M | yes |
| 08:03.33 | Qwell | but yeah, I've been real happy with them - it's gonna suck to start paying though |
| 08:03.57 | Assid | start paying? |
| 08:04.07 | docelmo | ok well on that note Im going to bed.. cya in the morning |
| 08:04.12 | Qwell | Assid: for services I got free |
| 08:04.35 | Assid | well.. if you work there.. why would they charge you |
| 08:04.41 | Qwell | Assid: because I don't anymore :P |
| 08:04.45 | Assid | ohhhh |
| 08:04.55 | Assid | whyd you quit |
| 08:04.59 | Qwell | I quit, to go work for Digium :D |
| 08:05.05 | Assid | you just joined like a month or so back rigth |
| 08:05.13 | Qwell | Assid: like...5 years ago |
| 08:05.41 | Assid | oh yeah.. hell confused |
| 08:05.59 | Qwell | Assid: I worked for a bank for 5 years, and got everything free. Now that I don't, I don't get it free anymore. |
| 08:06.33 | Assid | hell i get tons of free service from my bank.. they even send someone to collect paperwork from me ..... such as signatures |
| 08:06.51 | Qwell | neat |
| 08:08.57 | Strom_C | whats funny, Qwell, is how I was musing earlier today about how every B of A branch I've walked into has always seemed like a really miserable, depressing place |
| 08:09.06 | Qwell | ugh |
| 08:09.13 | Qwell | Don't get me started on BofA |
| 08:09.22 | Assid | bofa? |
| 08:09.24 | Assid | oh |
| 08:09.24 | Qwell | I am *NEVER* going to bank with them again |
| 08:09.26 | Strom_C | former customer? |
| 08:09.30 | Qwell | indeed |
| 08:09.41 | Strom_C | yeah, i've heard horror stories |
| 08:09.55 | Qwell | got charged a fee one month (when I never got charged before), and the fee caused...overage...which caused...you guessed it...overdraft fees |
| 08:09.56 | Qwell | WHICH |
| 08:10.00 | Qwell | overdrafted |
| 08:10.00 | florz | .o( already sounds like BofH =:-) |
| 08:10.05 | Qwell | and caused overdraft fees |
| 08:10.24 | Strom_C | so basically a $2 fee cascaded into like $500 worth of problems |
| 08:10.27 | Qwell | ended up somewhere around $90 |
| 08:10.46 | Strom_C | that's horrid |
| 08:10.57 | Qwell | yeah, I paid them, and said "close my accounts immediately" |
| 08:11.39 | Strom_C | smart move |
| 08:11.46 | Strom_C | I've never had an issue at wells ever ever |
| 08:11.59 | Qwell | I have, but they've always been fixed rather quickly |
| 08:12.00 | Assid | you guys need better banks |
| 08:12.41 | *** join/#asterisk vlt (n=dm@p54B34065.dip0.t-ipconnect.de) |
| 08:14.11 | Strom_C | who do you bank with? |
| 08:17.00 | x86 | what do you guys think of my website: https://voip.shellshark.net/ |
| 08:17.18 | x86 | is it clear? |
| 08:17.28 | x86 | any recommendations? |
| 08:17.44 | Qwell | well, I can't see my xchat window behind it, so I'd say it's opaque |
| 08:17.50 | Assid | buy a ssl cert |
| 08:18.03 | Strom_C | you actually expect people to /buy/ things from this hackjob? |
| 08:18.10 | Assid | and fill it up a bit |
| 08:18.36 | x86 | Strom_C: what's wrong with it? besides the temporary cert? |
| 08:18.48 | *** join/#asterisk techie (n=techie@ppp-69-239-205-253.dsl.frs2ca.pacbell.net) |
| 08:19.16 | x86 | Assid: by fill it up you mean give some more details about each plan? |
| 08:19.16 | Strom_C | if I were a potential customer, i'd look at it for half a second, go "they have no idea what they're doing," and move on |
| 08:19.32 | x86 | Strom_C: any suggestions? |
| 08:19.57 | Strom_C | x86: your layout is not at all conducive to easy comprehension |
| 08:20.30 | x86 | are you suggesting I go with more of a horizontal layout than the current vertical? |
| 08:20.52 | Strom_C | partially, and also dont represent pricing plans with pictures of phones |
| 08:20.55 | Qwell | hire a designer...seriously |
| 08:20.58 | Strom_C | yeah |
| 08:21.03 | Strom_C | hire a designer |
| 08:21.14 | Strom_C | stop wasting your time :) |
| 08:21.18 | x86 | yeah i'm all technical... this artsy stuff really isnt my thing ;) |
| 08:21.41 | x86 | Strom_C: i dont represent the pricing, i represent the plan with the phones |
| 08:21.50 | x86 | bad idea though? |
| 08:22.13 | Strom_C | and hire a good designer - someone who understands UI design and ergonomics, not just some artfag who makes things all pretty |
| 08:22.22 | Strom_C | x86: yes |
| 08:22.25 | x86 | if i was representing pricing i'd have a BT101 for the cheapest plans, and a 7985 for the most expensive plans ;) |
| 08:22.58 | x86 | suggestions on designers? |
| 08:23.02 | Strom_C | no clue |
| 08:23.10 | x86 | hmm |
| 08:23.10 | Strom_C | i'm not in the webpage business |
| 08:23.13 | MrChimpy | or you make the page as simple as possible. you can make something reasonably elegant by not wigging out on trying to make it pretty |
| 08:23.28 | Strom_C | this is my personal site though: |
| 08:23.30 | Strom_C | www.stromcarlson.com/ |
| 08:23.51 | x86 | ouch |
| 08:23.58 | x86 | that pains me to look at man |
| 08:24.06 | Strom_C | wha? |
| 08:24.15 | Strom_C | too orange? |
| 08:24.49 | Strom_C | or am I not using enough of the latest and greatest liquid ajax soap xml shit for your liking? :) |
| 08:25.04 | JT | hey at least it's visible |
| 08:25.10 | MrChimpy | mine is here: http://www.i-r-genius.com |
| 08:25.11 | JT | x86's site keeps looping or something |
| 08:25.24 | Qwell | You're all nubs |
| 08:25.24 | Qwell | http://visualbasicpro.com/ |
| 08:25.27 | JT | the ie symbol keeps stoppimg and starting |
| 08:25.29 | Qwell | perfect design |
| 08:25.54 | Strom_C | hahahaha |
| 08:25.57 | Strom_C | <3 qwell |
| 08:26.04 | x86 | haha |
| 08:26.14 | x86 | JT: bug in IE, I presume |
| 08:26.16 | Qwell | plz2be not adding pr0n/ at the end of that url, thank you |
| 08:26.42 | JT | pretty awesome <TITLE> |
| 08:26.42 | JT | visualbasicpro.com |
| 08:26.42 | JT | </TITLE> |
| 08:26.51 | JT | doesn't even have <html> |
| 08:26.53 | MrChimpy | pfft. wot no pr0n? |
| 08:26.59 | JT | x86: or your site? :P |
| 08:27.03 | Qwell | wtf, did somebody use curl? |
| 08:27.33 | Strom_C | http://www.stromcarlson.com/audio/test-call-to-911.mp3 |
| 08:30.02 | *** join/#asterisk benjk (n=benjamin@f8a01-0357.din.or.jp) |
| 08:33.42 | Qwell | anyways, bed |
| 08:34.15 | x86 | JT: nope, my site validates ;) |
| 08:34.20 | x86 | JT: it's a bug in IE |
| 08:34.31 | JT | that only your site causes, right |
| 08:34.44 | JT | so there is likely something you can do to fix it, in any case :P |
| 08:35.04 | x86 | ah cool, i've never met someone that's been to every site on the Internet before... that's amazing |
| 08:35.16 | x86 | how long did it take you? |
| 08:35.16 | JT | no probs |
| 08:35.16 | Strom_C | x86: stop being a smartass |
| 08:35.26 | x86 | Strom_C: lol |
| 08:35.38 | x86 | Strom_C: you have similar issues? |
| 08:35.44 | JT | anyway, the point you should take from this is that your site is in the minority with the looping thing, and some potential customers may not be able to view it as a result |
| 08:35.46 | Strom_C | I'm using firefox |
| 08:36.15 | x86 | JT: you said it was "looping" you did not claim you were unable to view it |
| 08:36.25 | x86 | is that the case though? |
| 08:36.29 | JT | yeah, white page |
| 08:36.32 | JT | unable to view |
| 08:36.37 | x86 | what version of IE? |
| 08:37.03 | JT | the version that has the problem with the site is v5 |
| 08:37.05 | JT | old i know |
| 08:37.14 | x86 | ah |
| 08:37.24 | x86 | yeah i'm not even worried about IE5 compatibility ;) |
| 08:37.43 | x86 | any modern system will get v6 or soon v7 from windows update automagically anyway :P |
| 08:38.08 | JT | well it does it in version 6 too |
| 08:38.13 | JT | so you can stop being smug now :) |
| 08:38.36 | x86 | i just said i'm not worried about ie5 compatibility, you said nothing about v6 until just now |
| 08:39.28 | JT | yeah i just tested |
| 08:40.03 | JT | but it seems your attitude is a bit of "not worried"/"screwem" :P when you're in business, they're potential customers too |
| 08:40.19 | JT | happens on http and https btw |
| 08:40.28 | x86 | if they are running IE5, that is my attitude indeed |
| 08:40.37 | x86 | now IE6 compatibility does worry me, however |
| 08:40.53 | x86 | can you trace down why it's happening? |
| 08:41.21 | Strom_C | my guess is you used the wrong lol. |
| 08:41.41 | JT | i can view it now by viewing the source on the cycling page, and using that url myself |
| 08:41.54 | JT | my guess is it's meta-refreshing the front page instead of to that url |
| 08:42.10 | JT | i could only view enough source to see that in ie6 |
| 08:42.24 | x86 | https://voip.shellshark.net/ meta refreshes to https://voip.shellshark.net/cgi-bin/plans.pl, yes |
| 08:42.41 | JT | https://voip.shellshark.net/ meta refreshes to https://voip.shellshark.net/ in ie :P |
| 08:42.48 | x86 | weird |
| 08:42.54 | Strom_C | metalol! |
| 08:43.28 | x86 | <meta http-equiv="refresh" content="0;https://voip.shellshark.net/cgi-bin/plans.pl"> |
| 08:43.39 | x86 | have no idea where IE would get something else... |
| 08:43.52 | x86 | unless you tried http://voip.shellshark.net first, then it's expected to break ;) |
| 08:44.28 | JT | why? |
| 08:44.53 | x86 | well because that particular server is not supposed to support direct http requests, just https |
| 08:45.13 | x86 | the main site properly hands it off control to this site via https |
| 08:45.20 | JT | a bit annoying, it should at least redirect |
| 08:45.22 | JT | x86: http://webdesign.about.com/od/metataglibraries/a/aa080300a.htm |
| 08:45.24 | x86 | s/hands it/hands/ |
| 08:45.40 | JT | you need to use the url= argument to meta refresh |
| 08:45.51 | JT | not put the refresh time and url in content= |
| 08:46.10 | JT | oh wait |
| 08:46.12 | JT | it was both |
| 08:46.19 | Strom_C | x86: seriously, hire a designer |
| 08:46.20 | JT | but your one still doesn't work, hmm |
| 08:47.00 | JT | so that's not it |
| 08:47.05 | x86 | yeah man |
| 08:47.06 | JT | well, might not be it |
| 08:47.07 | x86 | you're right |
| 08:47.07 | JT | i dunno |
| 08:47.09 | phearless | the schema "Asterisk as a SIP client behind nat, connecting to outside SIP " works ? |
| 08:47.12 | x86 | i needed url= in front of the url |
| 08:47.15 | x86 | try now |
| 08:47.27 | x86 | phearless: sure |
| 08:47.35 | JT | works |
| 08:47.35 | x86 | Strom_C: yeah, will do soon :) |
| 08:47.43 | x86 | JT: cool man, thanks for the assistance :) |
| 08:47.47 | JT | yeah i missed that it still didn't have url= |
| 08:47.52 | phearless | it is written on http://www.voip-info.org/wiki-Asterisk+SIP+NAT+solutions that it can be broken |
| 08:48.06 | phearless | "Every setup works somewhere, but it depends on the client, the NAT, the server and many other factors. In most cases, 1 and 3 is broken" |
| 08:48.56 | x86 | works fine here with a cisco pix 501, asterisk 1.2.11 inside the NAT, and remote asterisk 1.2.10 |
| 08:49.02 | JT | callcalculator doesn't work for me, and the tos opens a new window |
| 08:49.12 | Strom_C | wait, you're using SIP to trunk asterisk boxes? |
| 08:49.18 | Strom_C | why not use IAX2 to trunk? :) |
| 08:49.21 | x86 | JT: callcalculator is busted right now, working on importing new rates |
| 08:49.26 | phearless | x86: you just had to forward some ports ? |
| 08:49.37 | x86 | phearless: yeah... both SIP and IAX2 work fine for me |
| 08:49.55 | x86 | phearless: but really you should use IAX2 if at all possible, like Strom_C is saying |
| 08:50.17 | x86 | phearless: you'll have far less headaches and firewall hole-punching sessions ;) |
| 08:50.31 | x86 | JT: yeah the ToS should open a new window |
| 08:50.33 | *** join/#asterisk _omer (n=omer@203.128.20.175) |
| 08:50.38 | _omer | hi |
| 08:50.46 | x86 | JT: at least for now... it will be moved to lightbox shortly |
| 08:50.57 | phearless | oh yes I have not seen the message of Strom_C |
| 08:51.01 | _omer | what does Asterisk-addons contain ?? |
| 08:51.17 | phearless | in fact I would like to use the services of http://www.voxbone.com/ |
| 08:51.19 | Strom_C | _omer: it contains 85% LOL and 15% OMFG |
| 08:51.32 | JT | "lightbox" sounds like something that will not work in a text based browser |
| 08:51.39 | phearless | they provide virtual numbers that are forwarded via SIP |
| 08:52.00 | _omer | Strom_C: :) whats OMFG? |
| 08:53.00 | _omer | Strom_C: can I install "format_mp3" player without installing asterisk-addons? |
| 08:53.07 | x86 | JT: yeah it works, but it makes it seem as all one big page |
| 08:53.33 | x86 | JT: lightbox is just a way for you to make a <div> class hidden, and use javascript to make it appear, essentially... pretty cool stuff |
| 08:53.47 | JT | i see |
| 08:53.59 | x86 | for example |
| 08:54.00 | Strom_C | x86: ive got blind friends who would want to kill you over that one |
| 08:54.02 | x86 | click on VoIP Login |
| 08:54.05 | JT | sounds like it makes things unbookmarkable |
| 08:54.06 | *** join/#asterisk ree (n=ree@3e70c83a.adsl.enternet.hu) |
| 08:54.10 | JT | which also irritates me :] |
| 08:54.36 | x86 | JT: the line of thought there is to bookmark the main page, where the ToS is one click away |
| 08:54.48 | x86 | instead of bookmarking the ToS directly |
| 08:54.48 | Strom_C | let me hook you up with this chick I know named "accesibility" |
| 08:55.04 | x86 | is she hot? |
| 08:55.06 | JT | yeah, when i click on voip login, othing happens |
| 08:56.48 | JT | ok something happens in ie6 |
| 08:57.19 | x86 | like i said, it may not be 100% IE5 compatibile, but it should be decent with IE6 |
| 08:57.19 | JT | why not make a seperate page for it, like a normal web site? |
| 08:57.33 | x86 | trying to do some "Web 2.0" stuff |
| 08:57.36 | JT | people don't like unlinkable content |
| 08:57.49 | Strom_C | Web 2.0 can go fuck a lit firecracker for all I care |
| 08:57.56 | x86 | rofl |
| 08:58.33 | x86 | I'll make an old version of the site and allow the user to use either one |
| 08:59.02 | Strom_C | or how about just making something that WORKS without all your awful "wizardry" nonsense |
| 08:59.02 | _omer | can I get install format_mp3 instead of installing asterisk-addons? |
| 08:59.15 | x86 | Strom_C: it does not work? |
| 08:59.33 | Strom_C | x86: too-clever-by-half garbage rarely does |
| 08:59.44 | Assid | x86: whats the login page |
| 08:59.46 | JT | it does, as long as you use one of a narrow set or browsers |
| 08:59.59 | ree | Hi all, I count as newbie with asterisk and I have an annoying problem. Use case is that from my own asterisk I connect to a friend's asterisk like a client via "register". Registration ok however when he tries to call me, I got "Authentication rejected" on his call |
| 09:00.00 | Assid | oh wait |
| 09:00.04 | JT | imho, web sites should work in as much as possible, instead of needing the latest mainstream browser |
| 09:00.05 | tzafrir | _omer, basically, yes |
| 09:00.37 | Assid | how do you make the rest dark ? |
| 09:00.48 | tzafrir | _omer, though it would be much more efficient to convert the mp3s to wavs/slin/gsm offline |
| 09:00.51 | x86 | Assid: check out lightbox, it's sweet |
| 09:01.02 | ree | which I don't understand: why does it authenticate his call, when I act as a client in this game? iow, all calls that come through that registration could be just accepted on my server without further auth |
| 09:01.09 | x86 | Assid: it's not really magic, just the combination of some crafty CSS and a tiny bit of javascript |
| 09:01.34 | Assid | yeah |
| 09:01.47 | _omer | tzafrir : ok great then from where do I get format_mp3 from ??? |
| 09:02.08 | tzafrir | hmmm.... not sure.... |
| 09:02.20 | ree | Boy, am I on the right channel? |
| 09:03.05 | _omer | tzafrir : then I think i need to install asterisk-addons...actually i dont want to put more load on my asterisk by installing asterisk-addons... |
| 09:03.44 | tzafrir | ree, yeah. We're discussing the new http support of asterisk, right? |
| 09:04.02 | x86 | VoXML |
| 09:04.11 | ree | tzafrir: Sure, that's actually something that I understand..... |
| 09:04.43 | Assid | dont see the code you trigger on login click |
| 09:06.11 | x86 | Assid: it's handled somehow by the class of the anchor |
| 09:06.44 | _omer | tzafrir : any idea about converting MP3s into RAW ? |
| 09:06.54 | _omer | tzafrir : sox doesnt do that.. |
| 09:08.05 | Strom_C | mplayer! |
| 09:08.08 | Strom_C | and then sox |
| 09:08.18 | *** join/#asterisk fnordus (n=dnall@24.85.128.203) |
| 09:08.45 | _omer | hmmm...thanks.. |
| 09:12.28 | Aurs | mp3->lame->wav->sox->raw |
| 09:12.51 | *** join/#asterisk rikstah (n=rick@c-71-227-234-92.hsd1.or.comcast.net) |
| 09:13.16 | Strom_C | or even better, just downsample the wav to 8khz 16-bit wav |
| 09:13.26 | Strom_C | and use that |
| 09:13.30 | Strom_C | instant LOL |
| 09:13.58 | tzafrir | sox with mp3 support does that... |
| 09:14.04 | _omer | I convert MP3 into WAV (mpg123) then WAV to RAW (SOX) ....a 9MB MP3 is now 18MB RAW :-0 |
| 09:14.25 | tzafrir | _omer, but you forgot to downsample it |
| 09:14.54 | tzafrir | asterisk will need and play the file as 8kHz |
| 09:15.13 | tzafrir | and mono |
| 09:15.31 | _omer | sox -c 1 alwaysbemybaby.wav -t raw -r 16000 -c 1 -s -w 111.raw |
| 09:15.36 | tzafrir | this will slice it to its original size or smaller |
| 09:15.50 | *** join/#asterisk hermuli (n=Eladamri@a88-112-252-73.elisa-laajakaista.fi) |
| 09:16.02 | _omer | sox -c 1 alwaysbemybaby.wav -t raw -r 8000 -c 1 -s -w 111.raw <--- should I try this? |
| 09:16.24 | tzafrir | -r 8000, yes |
| 09:16.48 | _omer | yes....now its back to it's normal size.....thanks :) |
| 09:34.23 | shodan | anyone familiar with these analog phones => http://cgi.ebay.ca/Nortel-350-LCD-Display-Phone-w-Handset-Power-Supply_W0QQitemZ150029773358QQihZ005QQcategoryZ58335QQssPageNameZWDVWQQrdZ1QQcmdZViewItem |
| 09:34.45 | shodan | that "service" button , is that an ADSI thingy ? |
| 09:35.36 | *** part/#asterisk ree (n=ree@3e70c83a.adsl.enternet.hu) |
| 09:36.12 | *** join/#asterisk Sonderblade (n=mah@static-213.131.147.169.addr.tdcsong.se) |
| 09:49.17 | *** join/#asterisk techie (n=techie@ppp-69-239-205-253.dsl.frs2ca.pacbell.net) |
| 09:52.13 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 10:00.33 | phearless | anybody use http://www.voxbone.com/? |
| 10:00.34 | phearless | anybody use http://www.voxbone.com/ ? |
| 10:06.11 | RoyK | anybody use http://www.voxbone.com/ ? |
| 10:08.44 | shodan | http://www.cannabisculture.com/forums/uploads/1161537-answer%20is%20no.jpg |
| 10:08.50 | fourcheeze | nah, nobody uses any of them |
| 10:09.25 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 10:10.13 | shodan | is there something like adsi for voip phones ? this thing looks usefull |
| 10:10.24 | *** join/#asterisk Seb7 (n=sebast@host217-34-0-169.in-addr.btopenworld.com) |
| 10:12.58 | phearless | nice pic, shodan |
| 10:13.23 | shodan | yeah , seemed appropriate ;) |
| 10:16.06 | *** join/#asterisk [shodan] (n=shodan@ip059.96-113-216.pppoe1.joliette.intermonde.net) |
| 10:19.00 | *** join/#asterisk backblue (n=igor@82.102.1.42) |
| 10:19.02 | backblue | hi |
| 10:19.21 | *** join/#asterisk tsurk0 (n=tsurko@85.187.160.157) |
| 10:22.44 | *** join/#asterisk A-Tuin (n=a-tuin@steves.ip.v4.me.uk) [NETSPLIT VICTIM] |
| 10:24.20 | MrChimpy | anyone here using sangoma a104d? |
| 10:26.55 | [shodan] | questions like "anyone here using X ?" rarely get answers in here ! ;) |
| 10:29.03 | *** join/#asterisk A-Tuin (n=a-tuin@steves.ip.v4.me.uk) [NETSPLIT VICTIM] |
| 10:30.37 | MrChimpy | other options seem to give device not supported by kernel when I do wanrouter start |
| 10:37.23 | *** join/#asterisk ghenry (n=ghenry@suretecsystems.plus.com) |
| 10:47.19 | *** join/#asterisk A-Tuin (n=a-tuin@steves.ip.v4.me.uk) |
| 10:49.22 | backblue | ppl, anyone have connected b2b any E1 cards? (for testing purposes) |
| 10:53.54 | *** join/#asterisk RoyK (n=roy@ti211210a080-1761.bb.online.no) [NETSPLIT VICTIM] |
| 10:53.54 | *** join/#asterisk alawguy (n=mike@85-124-232-191.work.xdsl-line.inode.at) [NETSPLIT VICTIM] |
| 10:55.48 | *** join/#asterisk Bert- (n=bert@LAubervilliers-151-12-81-84.w193-252.abo.wanadoo.fr) |
| 10:55.50 | Bert- | hello there |
| 10:57.00 | Bert- | can someone explains me what exactly mean http://pastebin.ca/162286 please ? |
| 10:57.41 | Bert- | and what should I do to avoid theses notices ? |
| 10:58.50 | Strom_C | get the other end of the call to turn off VAD |
| 10:58.52 | Bert- | is it about silence suppression ? |
| 10:58.57 | Strom_C | yes |
| 10:58.58 | Bert- | ok |
| 10:59.08 | Bert- | so it is not from my conf, but from remote one ? |
| 10:59.20 | Bert- | ok |
| 10:59.21 | mut | strom: you know if wildcards work for includes? |
| 10:59.26 | mut | so i can do like |
| 10:59.38 | mut | #include </etc/asterisk/sip_peers/*.conf> |
| 10:59.50 | mut | to include all conf files in the folder, say 1 per peer |
| 11:00.03 | Strom_C | no idea, but that sounds like a brain-damaged way of doing configuration |
| 11:00.06 | mut | i don't have any test machine setup to try it |
| 11:00.21 | mut | um why? |
| 11:00.39 | mut | its easier to maintain than a single file with 300 peers |
| 11:00.48 | *** join/#asterisk svenadh (n=sven@213.217.93.246) |
| 11:00.53 | Strom_C | well, give it a shot and let me know what happens |
| 11:01.32 | mut | explain to me your reasoning behind the brain damaged way |
| 11:01.42 | Strom_C | it's 4 AM and i'm exhausted |
| 11:01.52 | Strom_C | therefore, lol-in-a-box |
| 11:01.52 | mut | cause, maybe it is and i should do it some other way i dunno |
| 11:05.46 | *** join/#asterisk shap (n=shap@c-68-33-84-43.hsd1.md.comcast.net) |
| 11:17.59 | *** join/#asterisk spr1te (i=spr1te@194.187.130.227) |
| 11:19.40 | *** join/#asterisk scage_ (n=xzen@202.63.226.41) |
| 11:23.39 | *** join/#asterisk Ebola (i=1000@81-86-155-65.dsl.pipex.com) |
| 11:28.31 | *** join/#asterisk willy_1234 (n=icechat5@62.231.36.101) |
| 11:28.37 | willy_1234 | help |
| 11:28.58 | willy_1234 | how do i change the max waiting time of a queue |
| 11:29.30 | willy_1234 | i looked in queues.conf and queues_additional.conf but cant find |
| 11:32.32 | *** join/#asterisk tsurk0 (n=tsurko@85.187.160.157) |
| 11:44.07 | *** join/#asterisk RoyK (n=roy@ti211210a080-1761.bb.online.no) |
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| 12:04.44 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 12:15.17 | *** join/#asterisk af_ (n=af@ip-173-161.sn1.eutelia.it) |
| 12:16.19 | *** join/#asterisk _deg_ (n=deg@201-24-224-220.ctame704.dsl.brasiltelecom.net.br) |
| 12:16.27 | *** join/#asterisk basty (n=basty@212.218.65.195) |
| 12:16.29 | basty | Hi |
| 12:17.39 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 12:17.53 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 12:18.30 | basty | I am using Asterisk and Zaptel - so I am trying to configure a hotstandby with drdb and heartbeat. As a matter of fact the zaptel drivers for my digium card should be loaded with heartbeat. I need to find a way to disable loading these modules by startup the machine. I am using Debian 3.1 - anyone could give me a hint ? :-) |
| 12:18.35 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 12:23.35 | *** join/#asterisk MikeJ (n=mikej@d14-69-8-30.try.wideopenwest.com) |
| 12:24.04 | MikeJ | anyone know of a phone, or softphone that supports g722? |
| 12:25.08 | tzafrir | basty, what triggers loading those modules? hotplug? |
| 12:25.24 | shodan | MikeJ, probably polycoms 301 and up ? |
| 12:25.53 | tzafrir | basty, why disable loading the modules? What's wrong with having the modules loaded? |
| 12:28.26 | MikeJ | shodan, do you know that for sure, I didn't think the polycoms did at all |
| 12:29.39 | *** join/#asterisk [Yatta] (n=noe@65.183.3.229) |
| 12:30.10 | *** join/#asterisk stagiaire (n=stagiair@LSt-Amand-152-31-13-31.w82-127.abo.wanadoo.fr) |
| 12:30.54 | [Yatta] | morning pppl.... how do i try out * 1.4 |
| 12:33.24 | [TK]D-Fender | [Yatta]: Have you compiled and loaded chan_fluxcapacitor.so firt? It's required.... |
| 12:33.30 | [TK]D-Fender | first* |
| 12:33.42 | tzafrir | [Yatta], 1.4 is not released yet. The closest thing to 1.4 is current svn trunk |
| 12:33.49 | [Yatta] | i haven't done a thing.. i'm in the #freepbx channel now sinc e i installed that last night |
| 12:33.59 | [Yatta] | i want to try h.264.. |
| 12:34.24 | [Yatta] | i have a Granstream video phone i want to give a try |
| 12:37.23 | DrukenHME | [TK]D-Fender: gotta load chan_fluxcapacitor if you want any hope in hell on calling SIP/11-05-1955 ..... |
| 12:38.33 | [TK]D-Fender | Polycom's only support G.711 & G.729 |
| 12:40.01 | *** join/#asterisk S^P (n=ss@203.81.196.20) |
| 12:42.22 | basty | tzafrir: Well - I want to connect 2x Cards (Master/Slave) to 1x Phone-Uplink. |
| 12:43.05 | basty | tzafrir: if there is a failover on the Master-Asterisk - heatbeat should start the zap-driver automaticly. |
| 12:43.53 | basty | tzafrir: And if I connect 2 active cards to one phone-uplink I will run into problems with the clocking :-\ |
| 12:43.56 | tzafrir | Why would you restart the zaptel driver? What's the point in it? You don't have to restart asterisk to change a line |
| 12:44.25 | tzafrir | You can also use ztcfg in an intelligent manner to change clocking parameters and such |
| 12:44.48 | tzafrir | basty, what cards? Why two seaparate cards? |
| 12:45.12 | *** join/#asterisk jmls (n=asterisk@host81-159-195-120.range81-159.btcentralplus.com) |
| 12:45.13 | shodan | MikeJ, no I'm not sure , can't find it |
| 12:45.22 | *** join/#asterisk DasTech (n=DasTech@d47-69-168-46.col.wideopenwest.com) |
| 12:45.23 | tzafrir | two separate cards on the same system still leaves too many points of failure, I believe |
| 12:45.27 | basty | tzafrir: TP410 Quard E1 - because I have two Servers (Master and a Slave for failover) |
| 12:45.41 | shodan | MikeJ, the snom 360 does it |
| 12:45.46 | DasTech | ok any one here have the 2.0.1 polycom firmware ? |
| 12:48.37 | [TK]D-Fender | DasTech: Go ask your reseller for it |
| 12:50.40 | pablus | morning from stgo of chile |
| 12:53.09 | tzafrir | basty, does http://voip-info.org/wiki/view/Xorcom+Astribank#zap_restart (the patch, not the product) help? |
| 12:53.57 | *** join/#asterisk Sonderblade (n=mah@static-213.131.147.169.addr.tdcsong.se) |
| 12:54.55 | *** join/#asterisk tld (n=tld@82.196.214.14) |
| 12:55.08 | tzafrir | basically: configure both cards. keep two zapata.conf-s : one for master and one for slave. On failover, swap them, and run zap restart |
| 12:55.22 | tld | Any recommendations for a US web store selling cordless phones? (I'm looking to order a DECT phone, not a WiFi VoIP phone) |
| 12:56.04 | DasTech | I got the phone from a former client as a bonus |
| 12:56.06 | tzafrir | (though I have not really tested it with digital cards) |
| 12:56.20 | DasTech | when I finished a contract |
| 12:57.51 | basty | tzafrir - okay thanks! I will try to test that. |
| 12:57.53 | basty | bye |
| 12:59.15 | *** join/#asterisk DrukenLPY (n=jdumais@CPE0040f43870d3-CM00137189cb0c.cpe.net.cable.rogers.com) |
| 13:00.33 | *** join/#asterisk inspired (n=mikael@85.221.7.59) |
| 13:03.27 | kmilitzer | Hello everyone. Can someone give me a hint how to built a transfer option in the dialplan, that generates two CDRs. One CDR for the "coming" Call leg, with original CID as src and dialed number as dst and one CDR with the dialed number as src and the number to which is transfered as dst ... any ideas how to do this? |
| 13:03.41 | *** join/#asterisk cstomi (n=chatzill@22-36.adsl.etel.hu) |
| 13:04.08 | stoffell | if I call a foreign country, I hear the foreign ringtone (uk/us/europe, all different) on my SIP phone. is this ringtone generated by asterisk or the telco? |
| 13:06.24 | [TK]D-Fender | stoffell: No doubt its just audio down the stream.... |
| 13:07.45 | *** join/#asterisk xyklopz (n=xyklopzi@86.122.8.28) |
| 13:08.00 | xyklopz | I know this is probably a retarded question, but is there anyway to flash the line in Asterisk over DTMF |
| 13:08.09 | xyklopz | my cordless handset doesn't support it ... |
| 13:08.24 | Muck- | does anyone know a tutorial how to set up an asterisk for about 10 people to connect to the main asterisk in the company... possibly best connected via iax2? |
| 13:08.32 | xyklopz | the idea would be similar to the way blind transfer works using # |
| 13:08.39 | dorel__ | ehh |
| 13:08.52 | dorel__ | i didnt know mark spencer was also the original author of gaim. how leet :) |
| 13:09.01 | Muck- | because they should get some of the numbers of the two ISDN BRIs here |
| 13:09.39 | Muck- | but i would like to make encryption through the internet via iax2 |
| 13:09.55 | xyklopz | Muck-, OpenVPN |
| 13:10.21 | [TK]D-Fender | Muck-: http://www.voip-info.org/wiki-Asterisk+-+dual+servers |
| 13:11.15 | xyklopz | I'm following the dual servers setup through a VPN (one server in US one in EU and works perfectly) + all voice traffic is encrypted between the two boxes |
| 13:13.12 | *** join/#asterisk [Yatta] (n=noe@65.183.3.229) |
| 13:13.22 | xyklopz | so I guess flashing the line (call waiting) must be done in the phone |
| 13:13.23 | xyklopz | not asterisk |
| 13:14.18 | *** join/#asterisk rados___ (n=rados@c-68-62-71-76.hsd1.mi.comcast.net) |
| 13:14.20 | *** join/#asterisk aixa (n=Miranda@www.crediweb.lv) |
| 13:14.27 | aixa | Hi everyone |
| 13:14.29 | *** join/#asterisk darkskiez (n=mbryars@194.247.78.146) |
| 13:14.59 | aixa | I just stumbled across the macro sample which uses Dial(Local/..) |
| 13:15.25 | aixa | what does this Local/.. stand for? |
| 13:15.33 | aixa | local context? |
| 13:15.59 | [TK]D-Fender | xyklopz: If you are referring to flashing an analog trunk then NO, * must do it. * does not pass on a flash jsut because you do it from an analog phone. Look at the "flash" application. Might do the job though I've never heard of a successful implementation of it. |
| 13:16.35 | [TK]D-Fender | aixa: Local channel. It connects the call to another part of the dialplan as though it were the starting point of the call. |
| 13:16.40 | rados___ | is ooh323 module/channel installed with the default trixbox installation? |
| 13:16.43 | *** part/#asterisk MikeJ (n=mikej@d14-69-8-30.try.wideopenwest.com) |
| 13:16.57 | [TK]D-Fender | rados___: Please read the channel topic.... |
| 13:17.06 | Muck- | [TK]D-Fender thank you! |
| 13:17.15 | rados___ | sorry |
| 13:17.27 | [TK]D-Fender | Muck-: So you want to set up a "satelite" office basically? |
| 13:17.45 | aixa | which par of dialplan it exactly connects to? |
| 13:18.01 | aixa | can i control the context from which the number gets dialed |
| 13:18.06 | aixa | ? |
| 13:19.29 | [TK]D-Fender | aixa: Dial(Local/exten@context) |
| 13:19.36 | aixa | okay :) |
| 13:19.39 | *** join/#asterisk Nebukadneza (n=daddel9@i3ED6F509.versanet.de) |
| 13:19.44 | aixa | many thanks, I somehow missed it |
| 13:20.05 | [TK]D-Fender | aixa: Like :Dial(Local/2000@contextwithextensformysipphones) |
| 13:20.43 | aixa | ye yes yes |
| 13:20.43 | *** join/#asterisk littleball (n=littleba@cm201.omega152.maxonline.com.sg) |
| 13:20.45 | aixa | :))) |
| 13:20.51 | aixa | fixed and working already |
| 13:21.01 | aixa | many thanx once again |
| 13:21.13 | stoffell | does anyone know where I can find more info on q931? (still struggling with the telco-voice stuff) |
| 13:21.17 | [TK]D-Fender | aixa: How are you using it now? |
| 13:21.51 | aixa | Local/${temp}@fw-dial |
| 13:21.52 | Dr-Linux|work | [TK]D-Fender, do you have experties with spa3000? |
| 13:22.48 | aixa | i had to achieve transfer to forwarded numbers with different dial logic than simple internal calls |
| 13:23.01 | [TK]D-Fender | awe6: Ah... recursive call-forwarding... excellent place to use it. |
| 13:23.07 | [TK]D-Fender | aixa: rather* |
| 13:23.27 | [TK]D-Fender | Dr-Linux|work: Yes, I've owned one before. |
| 13:24.18 | aixa | [TK]D-Fender: it is used to terminate recursions |
| 13:24.26 | kmilitzer | aixa: How do you handle your billing with this Local dials? |
| 13:24.33 | [TK]D-Fender | aixa: Thats the "easy" way to let a user forward to any number they can dial and have it do so recusively. So if A forwards to B, and B to C, and C to and external number it'll look through pretty much transparent to the originating caller |
| 13:24.39 | aixa | I dont need to handle billing |
| 13:24.41 | kmilitzer | aixa: Do you get correct CDRs? |
| 13:25.33 | Dr-Linux|work | [TK]D-Fender, great, |
| 13:26.16 | kmilitzer | Recursiv Call Forwarding is cool, I am playing with it right now, too, but it kills your CDRs :( |
| 13:26.52 | aixa | [TK]D-Fender: so that in my setup if A changed seats with B, A could froward his calls to B and B could forward his calls to A |
| 13:27.14 | aixa | and nboth would receive their calls at their new places respectively |
| 13:27.48 | aixa | thats why i needed diferent context than "default" |
| 13:28.11 | willy_1234 | how do u restart asterisk from the CLI |
| 13:28.25 | [TK]D-Fender | aixa: Ok, perhaps the term "nested" would be more appropriate.... you could always set an inherited var to check for nesting depth as a safeguard... |
| 13:28.31 | aixa | as in default direct call to B would be transfered to A and from A to B and sp neverending stroy |
| 13:28.47 | [TK]D-Fender | aixa: Trust me... segfaults end EVERYTHING ;0 |
| 13:29.03 | aixa | [TK]D-Fender okay now you got me really calm |
| 13:29.12 | [TK]D-Fender | :D |
| 13:33.14 | aixa | back |
| 13:33.37 | aixa | i already have crashed that box with txfaxf for a number of times |
| 13:33.48 | aixa | still trying to figure out whats the cause |
| 13:33.58 | aixa | spandsp, libtiff or glibc |
| 13:34.21 | aixa | so no big worries for crashing it again with forward :PP |
| 13:36.11 | aixa | ok, jokes aside - I`ll let this config a try and see what happens, if it is really that unstable then I'll have to lok for workaround |
| 13:36.43 | shodan | hmm , sip phone calls outside => * => cheap fxo ,,. sip phone hears ringing , callee picks up , caller and callee can talk , but caller can still hear ringing ! |
| 13:36.50 | shodan | what's up with that !! |
| 13:37.34 | shodan | work hours start in 20 mins ! crap :(( |
| 13:38.15 | fourcheeze | shodan: was it working before? |
| 13:39.37 | aixa | anyway - anybody else has these issues with txfax? That it crashes the * if it gets *.tiff file it can't read? |
| 13:41.19 | *** join/#asterisk coppice (n=chatzill@229.166.17.210.dyn.pacific.net.hk) |
| 13:42.31 | *** join/#asterisk }btorch{ (n=kvirc@adelphi.geofocus.com) |
| 13:42.49 | shodan | fourcheeze, yes ! |
| 13:43.03 | shodan | I didn't change anything |
| 13:43.20 | aixa | at first - with older libtiff asterisk gave at least some error output before it went belly up |
| 13:43.38 | aixa | now its silently dies |
| 13:44.48 | shodan | but all of a sudden echo cancelling works ?!?! |
| 13:45.23 | *** join/#asterisk phearless (n=phear@host81-138-68-106.in-addr.btopenworld.com) |
| 13:46.09 | phearless | when I left, I used /quit or /part ? |
| 13:46.32 | *** join/#asterisk Rahail (n=rahail1@209.19.88.243) |
| 13:46.39 | *** join/#asterisk `Kevin (n=Kevin@64.243.236.20) |
| 13:47.10 | *** join/#asterisk Bambr (n=Bambr@213-35-237-23-dsl.end.estpak.ee) |
| 13:48.51 | shodan | ok apparently my x100p isn't detecting that the callee has answered and it keep ringing , what could cause that ? how does it know that the callee has answered ? |
| 13:49.03 | *** join/#asterisk my007ms (n=my007ms@217.139.224.194) |
| 13:49.07 | *** join/#asterisk iulius (n=iulius@mail1.technologieshq.com) |
| 13:51.10 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 13:51.10 | *** mode/#asterisk [+o anthm] by ChanServ |
| 13:53.23 | rados___ | is there a way I can verify that ooh323 is installed properly? |
| 13:54.31 | xyklopz | I know this is probably a retarded question, but is there anyway to flash the line in Asterisk over DTMF |
| 13:54.33 | xyklopz | my cordless handset doesn't support it ... |
| 13:55.07 | xyklopz | the idea would be similar to the way blind transfer works using # but this is for call waiting |
| 13:56.29 | [TK]D-Fender | xyklopz: Look at "dynamic features" on the WIKI, and "show application flash" in * CLI |
| 13:56.33 | *** join/#asterisk DrukenHME (n=jdumais@CPE0040f43870d3-CM00137189cb0c.cpe.net.cable.rogers.com) |
| 13:58.14 | xyklopz | thanks [TK]D-Fender |
| 13:59.49 | xyklopz | I can't find "dynamic features" on the wiki |
| 14:02.15 | xyklopz | I assume it would be something like adding ... |
| 15:39.15 | *** join/#asterisk jbot (i=ibot@pdpc/supporter/active/TimRiker/bot/apt) |
| 15:39.15 | *** topic/#asterisk is Asterisk: The Open Source PBX -=- http://www.asterisk.org -=- Asterisk 1.2.11, Asterisk-addons 1.2.4, and Zaptel 1.2.8 released! (August 22, 2006) -=- FreePBX/AMP/Asterisk@Home/Trixbox support in #freepbx -=- Please use a pastebin when needing to show others large amounts of data. That is all. |
| 15:39.18 | SplasPood | maybe it won't crash |
| 15:39.42 | blitzrage | SplasPood: 8.3 (I think thats what I have) just locks up randomly on my 7960 |
| 15:39.46 | SplasPood | yep |
| 15:39.54 | vader-- | im still using 7.x on mine |
| 15:39.54 | SplasPood | 8.4 or some other more recent revision just came out |
| 15:39.57 | SplasPood | rumor has it, it fixes |
| 15:40.04 | SplasPood | vader--: no compelling reason to upgrade |
| 15:40.16 | vader-- | nope |
| 15:40.26 | blitzrage | file: any idea how I can determine if a module is in the middle of a reload so I don't start reloading another module too soon inside my script? |
| 15:40.29 | SplasPood | yea I'm saying there isn't |
| 15:40.31 | vader-- | only thing i wanna get rid of is the IP address crap it displays on the screens |
| 15:40.33 | file | not off the top of my head |
| 15:40.44 | SplasPood | yea thats annoying... |
| 15:40.47 | vader-- | but i wouldn't risk stability for something like that |
| 15:40.49 | blitzrage | SplasPood: I'm actually thinking of going back to 7.3 -- it's the only version that has ever worked consistantly for me |
| 15:40.58 | blitzrage | file: doh'eth :) |
| 15:40.59 | SplasPood | 7.5 was rock solid for me |
| 15:41.14 | blitzrage | I think I even had problems with 7.5... |
| 15:41.14 | BlackNTan | i'm having a problem getting my asterisk box to 'register' with Broadview.... and the logs aren't very clear as to what the problem may be (just timing out)... if I change the 'secret' I get an authentication error earlier in the log which leads me to believe that I am communicating with Broadview but for some reason I just can't register... any one know what this could be? http://paste.lisp.org/display/25537 |
| 15:47.36 | *** join/#asterisk GeeJay (n=gerry@195.69.91.144) |
| 15:48.16 | file | blitzrage: it would probably be possible to add the ability |
| 15:51.43 | *** join/#asterisk sb_mx (n=sb_mx@200.78.229.18) |
| 15:53.51 | *** join/#asterisk dasenjo (n=dasenjo@208.195.215.104) |
| 15:54.31 | *** join/#asterisk Winkie (i=slain@cpc2-stre3-0-0-cust344.bagu.cable.ntl.com) |
| 15:55.54 | GeeJay | I have got an issue with cdr in Asterisk. It appears to me that Asterisk logs CDRs even if the call has NOT been established by a 200 ok under certain circumstances. This seems to happen when the remote party CANCELs its INVITE (before it has been okayed by Asterisk) and also sends a BYE after the cancel. (Some clients fire off BYEs after a Cancel to make sure that in case the INVITE has been already 200 Okayed by Asterisk that such established conversation |
| 15:59.47 | SomeJ | ok, sorry about the spam earlier, logs and configs along with the issue can be seen here : http://pastebin.ca/162488 |
| 16:00.30 | [Yatta] | i'm trying to add h.264 support to my * box |
| 16:00.33 | [Yatta] | i did svn checkout http://svn.digium.com/svn/asterisk/trunk asterisk-addon |
| 16:00.51 | [Yatta] | do i need to configure asterisk-addon then confgiure asterisk?? |
| 16:01.14 | [Yatta] | or just confgiure asterisk and it will look into the addon?? |
| 16:03.43 | *** join/#asterisk s0lid (n=jlq@61.28.161.132) |
| 16:04.39 | *** join/#asterisk leejohn (n=johnlee@210.213.240.166) |
| 16:05.49 | leejohn | good day guys, has anyone from you encounter dtmf problems in recent trunk with regards to rfc2833 ? |
| 16:06.15 | file | leejohn: come to #asterisk-bugs |
| 16:06.24 | leejohn | my setup is PAP2-NA -> SIP -> Asterisk -> TDM400 -> PSTN |
| 16:06.25 | leejohn | thanks |
| 16:07.48 | kumbalae | leejohn: what is your problem? |
| 16:10.54 | leejohn | kumbalae: rfc2833 doesn't work well on trunk :) |
| 16:11.40 | file | no, there's a zaptel issue with vldtmf that we are trying to get solved - rfc2833 reception and transmission should be fine |
| 16:11.46 | *** join/#asterisk rift0r (i=[U2FsdGV@207.44.158.6) |
| 16:21.23 | blitzrage | file: oh yah? How complicated do you think it would be? |
| 16:21.41 | blitzrage | file: would be very handy for those of us doing 'asterisk -rx "reload chan_sip.so" |
| 16:21.57 | file | erm that's more complicated |
| 16:22.11 | blitzrage | even if I could check it from an AMI first though |
| 16:22.41 | file | chan_sip doesn't reload stuff when reload is called, it sets a variable that gets checked in the main thread for it |
| 16:22.43 | blitzrage | if there was like a Reload event or something that triggered a start and end of reload to the AMI? |
| 16:22.49 | blitzrage | ah |
| 16:22.58 | *** join/#asterisk af_ (n=af@ip-164-15.sn2.eutelia.it) |
| 16:22.58 | file | I think. |
| 16:23.03 | rift0r | so any of you guys use analog adapters with FXS? |
| 16:23.49 | blitzrage | file: heh :) |
| 16:25.44 | *** join/#asterisk LoneShadow (n=duh@59.92.149.34) |
| 16:30.51 | *** join/#asterisk wunderkin (i=kev@ip68-226-113-228.ph.ph.cox.net) |
| 16:31.49 | *** join/#asterisk Givemelove2k (n=foo@208.57.229.162) |
| 16:34.39 | *** join/#asterisk zaptel (n=just@nat1.inalambrica.net) |
| 16:34.54 | zaptel | hello everybody |
| 16:35.34 | zaptel | has anyone tried to connect an asterisk box to a Toshiba Strata PBX with a T1 card? |
| 16:38.10 | *** join/#asterisk knarfly_wk (n=bwatson@12.42.132.26) |
| 16:39.32 | *** join/#asterisk Vlasis_ (n=Vlasis@194.219.121.194) |
| 16:40.22 | *** join/#asterisk redder86 (n=lee@gateway.howardsilvan.com) |
| 16:40.37 | *** part/#asterisk redder86 (n=lee@gateway.howardsilvan.com) |
| 16:40.59 | Vlasis_ | hello all |
| 16:41.01 | Vlasis_ | :) |
| 16:43.39 | wunderkin | voipsupply is on the ball, too bad i did not call them yesterday |
| 16:44.08 | *** join/#asterisk sangee (n=rkuru@206.191.114.66) |
| 16:48.04 | mountainm2k | OK, so Voicemail() will transfer back to the "a" extension if the caller hits star... How can I set up the "a" extension differently for each user? |
| 16:48.22 | blitzrage | what do you mean? |
| 16:48.32 | Strom_C | how about this |
| 16:48.40 | mountainm2k | I want *MY* voicemail to transfer to my cell, but nobody else's... |
| 16:48.57 | Strom_C | exten => a,1,Goto(whee,${CALLERID(num)},1) |
| 16:49.10 | Strom_C | or some kind of conditional branching logic |
| 16:50.29 | blitzrage | if you can save the value of ${EXTEN} or something, then yah, you could just use a Goto() like Strom_C said |
| 16:50.37 | mountainm2k | so I have a macro for building users -- I want to simply pass in the number/extension I want it to call... Can I have it set a channel variable $COVEREXTEN, then a,1,Dial($COVEREXT) ? |
| 16:50.37 | Strom_C | yeah, there we go |
| 16:51.01 | Strom_C | just save the name of the voicemail box to some temporary channel variable |
| 16:51.07 | shodan | is there some doc about ADSI & asterisk ? I have a nortel telecom vista 350 and I'd really like to give that a try ! |
| 16:51.11 | *** join/#asterisk kannan (n=kannan@59.144.4.228) |
| 16:51.21 | Strom_C | and then...instant lol |
| 16:51.30 | Qwell | Strom_C: instant lol? |
| 16:51.34 | kannan | hello all |
| 16:51.36 | Qwell | Does that come in a tin can? |
| 16:52.02 | Vlasis_ | bye all :) |
| 16:52.04 | *** part/#asterisk Vlasis_ (n=Vlasis@194.219.121.194) |
| 16:52.07 | Strom_C | actually, it comes from the lol-in-a-box company of City of Industry, CA |
| 16:52.44 | Qwell | lol-in-a-box? |
| 16:52.48 | Strom_C | yes |
| 16:52.54 | Qwell | seriously? |
| 16:52.55 | *** join/#asterisk bjohnson_ (n=bjohnson@jecinc.tor.istop.com) |
| 16:53.08 | *** join/#asterisk LoneShadow (n=duh@59.92.142.128) |
| 16:53.18 | Strom_C | well, what do you want? lol-in-a-packet? lol-on-a-plane? |
| 16:53.27 | Qwell | I prefer my lol-in-a-cup |
| 16:53.34 | Strom_C | ahh, i see |
| 16:53.43 | shodan | hey there's 4 match on google for lol-in-a-box |
| 16:53.47 | [TK]D-Fender | shodan: If you own it already, its locked. |
| 16:53.52 | Strom_C | hahahaha, is there? |
| 16:54.46 | shodan | [TK]D-Fender, what about those lock codes or just removing the "cmos" battery in the lcd module ? |
| 16:55.28 | [TK]D-Fender | shodan: I do believe that this is non-volatile. You'd need to flash the eeprom or something. |
| 16:55.39 | *** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com) |
| 16:55.56 | [TK]D-Fender | shodan: Ma Bell doesn't want you having fun with their rip-off services |
| 16:56.07 | shodan | all those adsi phones on ebay ? also locked ? |
| 16:56.31 | [TK]D-Fender | shodan: Some may be unlocked, and would normally be advertised as such. |
| 16:56.57 | [TK]D-Fender | shodan: Hey... near Joliette, QC? |
| 16:57.09 | *** join/#asterisk madero (n=rbc@62-101-126-214.ip.fastwebnet.it) |
| 16:57.19 | Juggie | [TK]D-Fender, DAMN QC! |
| 16:57.28 | shodan | [TK]D-Fender, yeah , just like my xbox, my dish receiver, my nintendo ds and just about anything else with a warranty sticker , but I showed them ! |
| 16:57.29 | madero | hello ... do you think it would be possible to use asterisk to broadcast over the internet? |
| 16:57.30 | shodan | [TK]D-Fender, yep |
| 16:57.31 | [TK]D-Fender | Juggie: Too late!!!! ;) |
| 16:57.45 | Juggie | [TK]D-Fender, someone in 'QC' |
| 16:57.54 | Juggie | managed to get a copy of my bank card somewhere. |
| 16:58.02 | Juggie | and empty a little less then 1000$ out of a bank machine |
| 16:58.03 | [TK]D-Fender | shodan: I've got an Aastra PT390 ADSI unlocked phone that I've used with * that is for sale if you're interested. I'm in Montreal. |
| 16:58.05 | Juggie | outside montreal. |
| 16:58.06 | madero | basically, we need to have a speaker, talking almost-realtime ... questions being asked using a web-chat , and |
| 16:58.07 | Juggie | DAMN QC. |
| 16:58.26 | Strom_C | Instant LOL: http://www.aastratelecom.com/telephones/residential/screenphones/pro_77.asp |
| 16:58.27 | madero | lot of clients connecting using something like xmms, media-player or similar... andy idea? |
| 16:58.44 | shodan | [TK]D-Fender, yes probably , unless I can manage to hack this one , $? |
| 16:59.18 | [TK]D-Fender | shodan: Lets say $75. And It is unlocked and ready for * (U ised it with a TDM22B) |
| 16:59.49 | [TK]D-Fender | Strom_C: Whats so funny about that phone? |
| 16:59.58 | madero | any suggestion? |
| 17:00.22 | Qwell | That is quite possibly the ugliest phone I've ever seen |
| 17:00.22 | Strom_C | [TK]D-Fender: the joke is that it isn't funny; it's just a quick and simple solution to the problem of "I want ADSI" |
| 17:00.26 | rift0r | so any of you guys use analog adapters with FXS? |
| 17:00.33 | [TK]D-Fender | Qwell: I've seen FAR worse.... |
| 17:00.40 | shodan | [TK]D-Fender, any idea if it'd work with a spa-2102 ? |
| 17:00.42 | rift0r | i am looking at the 496 |
| 17:00.54 | [TK]D-Fender | rift0r: Yes, most of us. Got a more specific question? |
| 17:01.04 | [TK]D-Fender | rift0r: Avoid GrandSUCK at all costs. |
| 17:01.11 | rift0r | well which ones do you use and what are your opinions on the good ones |
| 17:01.27 | rift0r | [TK]D-Fender really, it said it won best of voip show 2006 or something |
| 17:01.30 | rift0r | looked pretty featureful |
| 17:01.37 | [TK]D-Fender | rift0r: For general use get an SPA-2002. Best general purpose one for the money. Tons of options and prettys imple to use. |
| 17:01.39 | Strom_C | which voip show was that? |
| 17:01.42 | rift0r | what should i go for then? linksys? digium iax? |
| 17:01.44 | mountainm2k | OK, in my cover-extension macro, I'm doing Set(CoverExt=${ARG2} -- Should I do that with a ExecIf (IE if there _is_ a ${ARG2} ? |
| 17:01.53 | Qwell | Strom_C: grandstreamcon |
| 17:01.57 | Strom_C | hahahaha |
| 17:02.04 | [TK]D-Fender | rift0r: Linksys SPA-2002 is your best bet. |
| 17:02.06 | knarfly_wk | [TK]D-Fender: I use Grandstream and it's great as far as I'm concerned....also only cost $58 |
| 17:02.23 | Strom_C | winning best of show at grandstreamcon is like being the tallest midget |
| 17:02.24 | [TK]D-Fender | knarfly_wk: And for $10 more you get a REAL one..... |
| 17:02.36 | rift0r | what about the iax |
| 17:02.39 | knarfly_wk | :) |
| 17:02.39 | rift0r | from digium |
| 17:02.50 | Qwell | rift0r: I like my iaxy. It kinda...just works |
| 17:03.00 | [TK]D-Fender | Strom_C: Also like running in the Special Olympics : Even if you win... YOU'RE STILL A RETARD! |
| 17:03.05 | Qwell | (disclaimer: I work for Digium) |
| 17:03.08 | rift0r | lol |
| 17:03.12 | shodan | 58$ ?! crap , my spa-2102 cost me 90$ :( I got ripped ! |
| 17:03.17 | rift0r | i am still bent on the whole sip / iax issue |
| 17:03.17 | bkw_ | "kinda" is the key word here |
| 17:03.21 | knarfly_wk | [TK]D-Fender: It really is a good phone. U must have had a bad experience |
| 17:03.24 | rift0r | i am new to this about to set up a pbx |
| 17:03.26 | Qwell | bkw_: troll :P |
| 17:03.39 | [TK]D-Fender | shodan: Well you picked the router-included model which I am NOT suggesting as well... |
| 17:03.58 | shodan | [TK]D-Fender, they didn't have the 2002 :( |
| 17:04.00 | [TK]D-Fender | bkw_: s/just// |
| 17:04.07 | rift0r | [TK]D-Fender does the linksys one you recommended have tftp config support |
| 17:04.39 | [TK]D-Fender | rift0r: Yes, as well as HTTP as I recall in addition to a really nice web interface (which if its just 1 or two you'll want to use probably). |
| 17:04.44 | shodan | [TK]D-Fender, where do you get your voip gear in mtl ? |
| 17:05.09 | [TK]D-Fender | shodan: Depends. Gentek has great pricing and is on-island. Aside from there depends what I'm wanting. |
| 17:05.19 | rift0r | [TK]D-Fender hrm, ya i saw the linksys one it actually was similarly priced to the 496 |
| 17:05.27 | rift0r | thx for the input |
| 17:05.42 | [TK]D-Fender | rift0r: np |
| 17:06.10 | shodan | k, I'll give them a visit next time instead of online stores ! |
| 17:06.26 | [TK]D-Fender | shodan: They are a wholesaler only, no storefront. |
| 17:06.28 | rift0r | [TK]D-Fender one more thing, did you unlock the SPA2002? |
| 17:06.40 | [TK]D-Fender | rift0r: No need. never locked |
| 17:06.55 | rift0r | k i thought i read something about that |
| 17:07.10 | [TK]D-Fender | rift0r: That''d be the PAP2, many of which were locked to Vonage, etc.... |
| 17:07.32 | rift0r | ahh i see |
| 17:07.35 | bkw_ | all I can say is ... AAA meetings are fun |
| 17:07.38 | [TK]D-Fender | rift0r: Also the PAP2 has a more limited featureset IIRC... so SPA-2002 is the A1 choice for you for starters. |
| 17:07.39 | *** join/#asterisk MikeJ (n=mikej@d14-69-8-30.try.wideopenwest.com) |
| 17:07.46 | rift0r | ok thx |
| 17:07.49 | [TK]D-Fender | bkw_: Road trip? |
| 17:07.49 | bkw_ | Asterisk Alcoholic Anonymous |
| 17:07.50 | rift0r | ill pick one of those up |
| 17:08.20 | [TK]D-Fender | rift0r: If you want a GODLY one (and know you need it) then look at the MediTrix 2102. |
| 17:08.25 | [TK]D-Fender | MediaTrix |
| 17:08.34 | rift0r | how much is that |
| 17:09.27 | shodan | even better :) |
| 17:09.48 | rift0r | hmm $110 |
| 17:09.53 | [TK]D-Fender | rift0r: Much more. how/where do you intending on using your ATA? |
| 17:12.38 | SplasPood | Anyone know of a vendor within china that will sell the digium FXO cards? (need to interface to two chinese pots lines) |
| 17:12.51 | *** join/#asterisk LoneShadow (n=duh@59.92.146.88) |
| 17:13.29 | *** join/#asterisk X-Rob_ (n=rob-x@dsl-202-173-151-24.qld.westnet.com.au) |
| 17:14.59 | *** join/#asterisk wunderkin (i=kev@ip68-226-113-228.ph.ph.cox.net) |
| 17:15.32 | madero | one more question ... has anyone tryed any "web phone"? something in flash/activex/javascript/whatever that would allow someone to connect to an asterisk server? |
| 17:16.12 | shodan | [TK]D-Fender, thanks for the tip, I had been searching for a wholesaler for a while now :) |
| 17:17.01 | *** join/#asterisk redder86 (n=lee@gateway.howardsilvan.com) |
| 17:17.12 | *** part/#asterisk redder86 (n=lee@gateway.howardsilvan.com) |
| 17:18.07 | *** join/#asterisk spatulamaan (n=ggilmore@207.188.8.252) |
| 17:18.35 | *** join/#asterisk Avalone (n=Avalone_@dial-324.vl-cen-as1.avtlg.ru) |
| 17:19.31 | *** join/#asterisk IronMan2000 (n=kent@65.124.236.24) |
| 17:19.58 | mut | is there a prerecorded sound for like |
| 17:20.00 | IronMan2000 | I am looking to hire someone that can help me setup my asterisk server.. |
| 17:20.10 | mut | "this feature is unavailable" or something to that effect |
| 17:20.18 | BlackNTan | another plea for help: |
| 17:20.19 | BlackNTan | i'm having a problem getting my asterisk box to 'register' with Broadview.... and the logs aren't very clear as to what the problem may be (just timing out)... if I change the 'secret' I get an authentication error earlier in the log which leads me to believe that I am communicating with Broadview but for some reason I just can't register... any one know what this could be? http://paste.lisp.org/display/25537 |
| 17:20.56 | mut | broadview? |
| 17:20.58 | mut | why not call them.. |
| 17:21.28 | tessier__ | http://pastebin.ca/162565 |
| 17:21.34 | tessier__ | Anyone see anything wrong with this? |
| 17:21.40 | tessier__ | I am trying to dial long distance on my PRI and it fails. |
| 17:21.44 | IronMan2000 | anyone for hire to help with asterisk setup? |
| 17:21.48 | tessier__ | I can dial local numbers even with 10 digit dialing. |
| 17:21.55 | tessier__ | IronMan2000: Lots of people are. |
| 17:22.22 | tessier__ | My telco insists that long distance is working properly |
| 17:23.02 | IronMan2000 | well, I guess everyone must be out to lunch |
| 17:23.08 | Strom_C | IronMan2000: I'll gladly consider it :) |
| 17:23.32 | IronMan2000 | Strom, thank you. |
| 17:23.33 | carrar | IronMan2000, whats your pay? |
| 17:24.49 | IronMan2000 | $150.00 for the features I need setup. Can pay via PayPal |
| 17:25.02 | [TK]D-Fender | IronMan2000: PM |
| 17:25.22 | Strom_C | IronMan2000: yes, PM here as well :) |
| 17:25.24 | tessier__ | IronMan2000: It usually helps to say how much you are paying and what you need up fron |
| 17:25.24 | tessier__ | t |
| 17:25.44 | tessier__ | If you want help setting up H323 I'm not interested. If you need help setting up other things I might help. |
| 17:25.45 | carrar | $150/hr right? |
| 17:25.56 | tessier__ | But it looks like you already have helpers so good luck. |
| 17:27.10 | carrar | IronMan2000, personaly I think you should attempt to try to do it yourself |
| 17:28.05 | *** join/#asterisk harlequin516 (n=sham@dsl01-ppp-4444.fastq.com) |
| 17:28.06 | carrar | tessier__ |
| 17:28.15 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 17:28.17 | carrar | You need to have your telco watch as you dial LD |
| 17:28.43 | tessier__ | carrar: I have done that |
| 17:28.43 | Qb3rt | on my asterisk server i need to forward calls to another country in one avaya system... is there a manner for me to bring back the call in canada and forward it to another location on keypress by the guy on answered the call in the other country?? |
| 17:28.49 | harlequin516 | The Dial cmd has the options h, and H. How can you turn off both? |
| 17:29.01 | *** join/#asterisk SeicherlBoB (n=seicherl@85-126-76-170.work.xdsl-line.inode.at) |
| 17:29.12 | harlequin516 | Can you specify -h? |
| 17:29.39 | Qwell | harlequin516: Just don't add them |
| 17:29.57 | SeicherlBoB | hi there! it seems the only extension i can catch a call via my voip-provider is with "_." though i registered the connection to extension 1 any ideas? |
| 17:30.00 | tessier__ | carrar: When I dial a local long distance number with 11 digits (like 1858XXXXXXX or 1619XXXXXXX) it works fine. And they see the number dialed. If I dial a long distance number like 16319240517 (which is a number they told me to test with) I get the above error. |
| 17:30.06 | BlackNTan | mut: broadVOICE sorry about that |
| 17:30.37 | harlequin516 | Are they all off by default? |
| 17:30.54 | carrar | tessier, what about if you call that with your cell? |
| 17:31.14 | carrar | what do they say when you dial the number that does not work when they are watching? |
| 17:31.25 | carrar | assuming it works with your cell |
| 17:31.25 | Qwell | harlequin516: yes, options are only enabled when you add them |
| 17:31.48 | tessier__ | So now I have a fingerpointing situation with my telco it seems. |
| 17:31.55 | tessier__ | carrar: It works if I call it with my cell. |
| 17:32.00 | carrar | well drop them if they can't help you |
| 17:32.15 | carrar | they can watch as you dial |
| 17:32.17 | mut | anyone used chan_ss7? |
| 17:32.22 | carrar | bbl (meeting) |
| 17:32.36 | *** join/#asterisk freebsd_fan (n=ebola@catagiuri305.giuri.unige.it) |
| 17:32.37 | tessier__ | carrar: When I dial a long distance number they say they see the line flash and that's it |
| 17:32.55 | SeicherlBoB | on extension s my machine wont answer. it only answers when using _. any suggestions why s extension is not working? |
| 17:33.29 | [TK]D-Fender | SeicherlBoB: What interface are your calls coming in under? |
| 17:33.44 | *** part/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 17:33.51 | SeicherlBoB | [TK]D-Fender: its a voip-only system. |
| 17:34.37 | [TK]D-Fender | SeicherlBoB: Well most SIP calls come in TARGETING a number (usually the same as a DID you have registered to them. Since the target is always know, "s" does not come into play. |
| 17:35.15 | SeicherlBoB | [TK]D-Fender: ok. this is all rather new to me. i just did what o'reiley told me ^^ can you explain that a bit more to me? |
| 17:35.16 | [TK]D-Fender | SeicherlBoB: You shoul use a simple exten like : exten => 2135551212,1,Goto(mymainmenu,s,1) or something to cath that incoming call. |
| 17:35.46 | [TK]D-Fender | SeicherlBoB: The only device that needs "s" for incoming calls is an analog line on a Zaptel interface. |
| 17:35.58 | SeicherlBoB | [TK]D-Fender: ahhh |
| 17:36.15 | [TK]D-Fender | SeicherlBoB: Zaptel analog channels have no idea WHY they are ringing, just that they ARE. |
| 17:36.51 | [TK]D-Fender | SeicherlBoB: All digital channels know what resource the caller is trying to access (DID's fall on PRI's, etc...) |
| 17:36.59 | SeicherlBoB | [TK]D-Fender: so if i have my SIP-account registered in some context like sip_in i should put it on a extension like 12345 and catch that extension in the dialplan? |
| 17:37.10 | [TK]D-Fender | SeicherlBoB: Exactly |
| 17:37.27 | SeicherlBoB | [TK]D-Fender: ok. i'll try that. hold on |
| 17:37.38 | [TK]D-Fender | SeicherlBoB: Your catch-all really does just that, but you should be able to put a SPECIFIC value for them. |
| 17:38.30 | [TK]D-Fender | SeicherlBoB: Make the first priority of your catch-all do something like this : exten => _.,1,NoOp(The actual exten that was called was ${CALLERID(number)}) |
| 17:39.11 | [TK]D-Fender | SeicherlBoB: Then you'll have a confirmation on what they are sending so you can make the change. It should also be the last things after the last "/ in your "register" line in sip.conf |
| 17:39.47 | SeicherlBoB | ahhh... ok. wait a moment |
| 17:39.54 | mountainm2k | Hey, wow, I caused a loop... Nifty... |
| 17:40.06 | Qwell | mountainm2k: That's easy |
| 17:40.11 | Qwell | exten => s,1,Goto(s,1) |
| 17:40.26 | mountainm2k | Yeah, well, I'm still trying to figure out where mine is looping... |
| 17:40.36 | Qwell | ahh, accidently...yeah, that's different :p |
| 17:40.50 | mountainm2k | in my "user extension" macro, I have Set(CoverExt=${ARG2}) |
| 17:41.23 | mountainm2k | then back in the parent context, I have exten => a,1,Goto(extension-context,${CoverExt},1) |
| 17:41.48 | mountainm2k | so then in Voicemail(), I should be able to hit star, and go to $CoverExt |
| 17:41.49 | SeicherlBoB | [TK]D-Fender: and that output should be stated on the console? |
| 17:41.52 | mountainm2k | but that caused a loop |
| 17:41.55 | [TK]D-Fender | mountainm2k: Careful with global vars like that... protect your scope... |
| 17:42.05 | [TK]D-Fender | SeicherlBoB: Exactly. Then place a call to it and watch. |
| 17:42.11 | mountainm2k | Should be a channel var, yes? |
| 17:42.20 | mountainm2k | as in, specific only to this call, not global? |
| 17:42.30 | [TK]D-Fender | mountainm2k: Pastebin your entire context and everything related to it. |
| 17:42.32 | [TK]D-Fender | ~pb |
| 17:42.36 | jbot | rumour has it, pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/ (BROKEN AND SUCKING NUTS), or http://pastebin.ca, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com, or http://bzflag.pastebin.ca/, or http://paste.lisp.org/ for the lisp/scheme nuts, or http://paste-it.net |
| 17:42.45 | mountainm2k | hey, wow, jbot works again! |
| 17:42.47 | SeicherlBoB | [TK]D-Fender: what you told me just echoed the calling number, not the called extension |
| 17:43.00 | [TK]D-Fender | SeicherlBoB: OOPS |
| 17:43.04 | SeicherlBoB | hehe |
| 17:43.10 | [TK]D-Fender | SeicherlBoB: Make the first priority of your catch-all do something like this : exten => _.,1,NoOp(The actual exten that was called was ${EXTEN}) |
| 17:43.21 | [TK]D-Fender | SeicherlBoB: Yeah... silly reflex made me type that! :) |
| 17:43.28 | SeicherlBoB | np |
| 17:43.59 | SeicherlBoB | well, makes more sense now - i should start thinking aswell |
| 17:45.23 | SeicherlBoB | [TK]D-Fender: ok. first i get "The actual eyten that was called was +43720505187" then i get "The actual eyten that was called was t" |
| 17:45.35 | SeicherlBoB | oh... t is for terminated? |
| 17:46.26 | mountainm2k | OK, Here it is... http://pastebin.ca/162584 |
| 17:46.39 | SeicherlBoB | so do i have to register that account to that extension AND catch it with that extension on the dialplan? including the "+"? |
| 17:46.44 | mountainm2k | Not the entire thing, but it should be what's required for this... |
| 17:46.53 | mountainm2k | (I can do the entire thing if this part isn't clear) |
| 17:48.01 | [TK]D-Fender | SeicherlBoB: Thats because the exten eventually runs out and your catchall is cathcing things twice |
| 17:48.22 | [TK]D-Fender | SeicherlBoB: Now that you know exactly how the call comes in remove your catchall and replace it witrh the fixed value. |
| 17:48.59 | *** join/#asterisk voipman (i=distorti@junipero.3sheep.com) |
| 17:49.15 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 17:49.50 | [TK]D-Fender | mountainm2k: "a" should be in your macro, not outside. And should look more like : exten => a,1,Goto(internal-ld,${MACRO_EXTEN},1) |
| 17:49.58 | *** part/#asterisk Avalone (n=Avalone_@dial-324.vl-cen-as1.avtlg.ru) |
| 17:50.01 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 17:50.27 | [TK]D-Fender | mountainm2k: And that variable Hint you have there does NOT work. |
| 17:50.30 | mountainm2k | Oh? But then it'll define the a extension for every time the macro is called... |
| 17:51.31 | [TK]D-Fender | mountainm2k: Actually more like : exten => a,1,Goto(internal-ld,${ARG2},1) |
| 17:51.47 | *** join/#asterisk seicherl (n=seicherl@85-126-76-170.work.xdsl-line.inode.at) |
| 17:51.58 | *** join/#asterisk h3x (n=h3xor@64.192.116.17) |
| 17:51.58 | seicherl | so, now i terminated ^^ |
| 17:51.59 | mountainm2k | that's inside my macro? |
| 17:52.09 | [TK]D-Fender | mountainm2k: it is only important local to the dialplan being called and is not GLOBAL. its limited to the channel calling it. |
| 17:52.11 | seicherl | [TK]D-Fender: sorry, lost connection |
| 17:52.42 | seicherl | [TK]D-Fender: first i got "The actual eyten that was called was +43720505187" then i got "The actual eyten that was called was t" (because the call was terminted ?) |
| 17:53.28 | [TK]D-Fender | mountainm2k: http://pastebin.ca/162589 |
| 17:53.37 | mountainm2k | <PROTECTED> |
| 17:53.37 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 17:53.37 | *** mode/#asterisk [+o mog] by ChanServ |
| 17:53.49 | mountainm2k | Oh... hehe, crossed in the mail... |
| 17:54.14 | [TK]D-Fender | mountainm2k: Yeah, virtually identical. You are learning... |
| 17:54.51 | mountainm2k | See, I thought when I first started all this that the macro would define a seperate "a" for each actual user phone (IE each time I say the macro) |
| 17:54.59 | mountainm2k | But that just isn't the case, is it... |
| 17:55.03 | [TK]D-Fender | mountainm2k: Why do all phones have 4 versions? |
| 17:55.05 | *** join/#asterisk juanjoc (n=juanjoc@200.73.189.82) |
| 17:55.31 | seicherl | [TK]D-Fender: so do i have to register that account to that extension AND catch it with that extension on the dialplan? including the "+"? |
| 17:55.42 | [TK]D-Fender | mountainm2k: No you should call the macro with unique parameter indicating where to go. |
| 17:55.42 | mountainm2k | eh? Not sure what you mean... I have internal, internal-local, internal-ld, and internal-intl |
| 17:56.01 | mountainm2k | Right... Well, if ARG2 isn't set, that means don't do anything... |
| 17:56.11 | mountainm2k | In other words, I'm guessing some people won't want this "feature"... |
| 17:56.15 | [TK]D-Fender | Seicherl : the "+" is a relevent char. so exten => +43720505187,1,GetMoving() |
| 17:56.55 | [TK]D-Fender | mountainm2k: You'll need to pass a set of parameters that is relevent to the specific call on a case-by-case basis |
| 17:57.58 | [TK]D-Fender | mountainm2k: like say Macro(mydialmacro,2000,internal-ld,8005551212) for instance. in your macro you'd test to see if parm is set and then act accordingly. |
| 17:58.33 | *** join/#asterisk darkskiez (i=mhb@bb-87-81-62-203.ukonline.co.uk) |
| 17:58.51 | mountainm2k | http://pastebin.ca/162588 -- OK, so in exten => a the NoOp becomes Goto -- then I reload pbx_config and it totally ignores the star |
| 17:58.56 | mountainm2k | when I call |
| 17:59.17 | mountainm2k | and zero, too, for that matter |
| 18:00.36 | [TK]D-Fender | mountainm2k: Thats just not normal.... works great for me... |
| 18:00.51 | Qwell | You need to enable...something...somewhere |
| 18:01.20 | *** join/#asterisk seicherl (n=seicherl@85-126-76-170.work.xdsl-line.inode.at) |
| 18:01.20 | [TK]D-Fender | mountainm2k: Chow CLI output while you're at it. |
| 18:01.20 | [TK]D-Fender | show* |
| 18:01.51 | seicherl | [TK]D-Fender: great. got disconnected again.... can you help me with that extension stuff again? |
| 18:02.00 | Qwell | operator=yes, is it? |
| 18:02.16 | [TK]D-Fender | Seicherl : Seicherl : the "+" is a relevent char. so exten => +43720505187,1,GetMoving() |
| 18:02.44 | seicherl | [TK]D-Fender: and do i have to register the SIP-account under that extension aswell? |
| 18:03.10 | [TK]D-Fender | Seicherl : Well thats just what they send you. I take it thats the # you pay them for? |
| 18:03.29 | *** join/#asterisk unmanaged (n=unmanage@64.89.118.139) |
| 18:03.35 | mountainm2k | Here's my CLI: http://pastebin.ca/162601 |
| 18:04.02 | mountainm2k | I dial an extension that prompts for the mailbox, then dial my mailbox number, and I get my greeting... star and zero are both ignored... |
| 18:04.04 | seicherl | [TK]D-Fender: i mean in sip.conf where i register my SIP account. do i have to add the /+43720505187 ? |
| 18:04.22 | [TK]D-Fender | mountainm2k: I don't see your macro being called anywhere in there.... |
| 18:04.39 | [TK]D-Fender | Seicherl : Just do the extensions.conf part I mentioned first |
| 18:04.42 | mountainm2k | hahaha, of course, damn... One sec... |
| 18:05.10 | unmanaged | hmm When trying to push about 500 calls to test a system I am hitting what seems to be a limit of 126 channels, everything else after 126 calls go to UNAVAILABLE... what are some tips for high volume of calls.... |
| 18:05.13 | [TK]D-Fender | mountainm2k: My apples are CLEARLY superior to your ORANGES.... |
| 18:05.27 | mountainm2k | hahaha... |
| 18:05.36 | *** join/#asterisk smackus (n=ckwall@63.149.122.93) |
| 18:05.43 | mountainm2k | my "go to voicemail" ext doesn't actually call the macro, that'd be why... |
| 18:05.49 | *** part/#asterisk IronMan2000 (n=kent@65.124.236.24) |
| 18:06.26 | smackus | is it possible to run ztcfg on just one span of my t1 card? i have only one span that I need to change, but dont want to affect the calls on the other spans |
| 18:06.52 | mountainm2k | ;smacks himself upside the face... |
| 18:07.17 | mountainm2k | OK, it works for outside calls, anyhow... |
| 18:07.37 | tessier__ | Anyone familiar with snom phone dialplans able to tell me why this doesn't work for dialing international from the US: |^(9011[0-9]*)|sip:+\1@\d| |
| 18:07.39 | mountainm2k | And perhapps it's best that the "go to voicemail" function doesn't allow zeroing / staring back out... |
| 18:07.57 | tessier__ | It looks like it should begin with 9 then 011 then any number of digits 0-9 but the phone never matches it. |
| 18:07.57 | mountainm2k | [TK]D-Fender: Thanks for the help... :-) |
| 18:07.58 | [TK]D-Fender | mountainm2k: Ok, either way you've got some confirmation and a better feel for macros. Congrats. |
| 18:08.14 | *** join/#asterisk toerkeium (i=oo@201.216.206.221) |
| 18:08.15 | seicherl | [TK]D-Fender: ok, if i have no extension set in sip.conf and only in extensions.conf (as you told me) i get no answer |
| 18:08.26 | tessier__ | Note that I do not have a trailing d which would mean it is a complete number since we do not know how long the international number might be. |
| 18:08.47 | [TK]D-Fender | seicherl: PM me your "register" line from sip.conf |
| 18:08.56 | seicherl | [TK]D-Fender: typo! ^^ |
| 18:09.04 | tessier__ | The rest of the dialplan works fine. |
| 18:09.26 | mountainm2k | Hmmm, any way to only define the a extension if ${ARG2} is set to something? |
| 18:09.35 | mountainm2k | Perhapps a GotoIf |
| 18:09.37 | mountainm2k | ? |
| 18:09.57 | seicherl | [TK]D-Fender: looks good ^^ |
| 18:10.09 | mountainm2k | GotoIf{${ARG2}!=''|blah ? |
| 18:10.23 | [TK]D-Fender | seicherl: So all is goo with the fixed #? |
| 18:11.00 | [TK]D-Fender | mountainm2k: GotoIf($["${ARG2}"=""]?blah) |
| 18:11.04 | seicherl | [TK]D-Fender: yeah. no extension set in sip.conf but in the dialingplan ^^ seems to work. i'll try to call my softphone |
| 18:11.59 | [TK]D-Fender | seicherl: They maye ALWAYS target the exten related to your account regardless of whats sent. Depends on them a bit.... |
| 18:11.59 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 18:12.13 | mountainm2k | exten => a,1,GotoIf($["${ARG2}"=""]?internal-ld,${ARG2},1) |
| 18:12.16 | mountainm2k | should work... |
| 18:12.49 | mountainm2k | and then the next priority dumps them back to VoiceMail -- ? |
| 18:13.29 | *** join/#asterisk DasTech (n=DasTech@d47-69-168-46.col.wideopenwest.com) |
| 18:13.41 | DasTech | ok anyone here work with pix firewalling |
| 18:13.45 | seicherl | [TK]D-Fender: looks good. ^^ |
| 18:14.04 | DasTech | I have the page from the wiki but it does not say where to make the protocols |
| 18:14.14 | seicherl | [TK]D-Fender: yeah, i just thought i could assign an internal extension to the sip-account |
| 18:14.38 | DasTech | http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 18:14.41 | [TK]D-Fender | seicherl: Well you use your dialplan to make it do whatever you want. |
| 18:14.56 | DasTech | I did the sip and rtp stuf |
| 18:15.01 | [TK]D-Fender | seicherl: Its not a "pass-through. * isn't a proxy, its a B2BUA (back To Back User Agent) |
| 18:15.44 | seicherl | [TK]D-Fender: yeah. gotta get used to that a bit more. and how can i use that for external calls now? (or should i better as mr o'reily again |
| 18:15.50 | *** join/#asterisk boch (n=root@201.216.241.97) |
| 18:15.53 | DasTech | where do you create the protocol |
| 18:16.16 | boch | is it possible to do: host=x.x.x.x,x.x.x.x in sip.conf ? |
| 18:17.30 | tessier__ | Sometimes I really hate voip. :( |
| 18:17.34 | [TK]D-Fender | seicherl: Depends how you set that guy up. Yeah, give it a few tries and come back here after you've failed at a whole bunch of different ways :) |
| 18:17.40 | mountainm2k | exten => a,1,GotoIf($["${ARG2}"=""]?internal-ld,${ARG2},1) |
| 18:17.41 | mountainm2k | didn't work |
| 18:17.54 | mountainm2k | WARNING[10494]: pbx.c:2357 __ast_pbx_run: Channel 'SIP/603-697a' sent into invalid extension 'a' in context 'internal-ld', but no invalid handler |
| 18:17.57 | tessier__ | So many other people to deal with, weird hardware, weird software... |
| 18:18.11 | tessier__ | This should work! It is copied straight from the example. |
| 18:18.37 | [TK]D-Fender | mountainm2k: Read that line carefully.. there is a tragic error in it. |
| 18:20.18 | *** join/#asterisk ToTo (n=ToTo@host212-207.pool872.interbusiness.it) |
| 18:21.22 | mountainm2k | Hmmm... It would appear it's going all the way out to internal-ld extension 'a' -- that's not what I want... D'oh... |
| 18:21.51 | [TK]D-Fender | mountainm2k: Read it again... obvious error in there.... |
| 18:22.09 | mountainm2k | pulling out my book -- looks like I missed the : |
| 18:22.30 | vader-- | does anyone know if there is a way to build a gotoiftime where it's a file filled with dates? |
| 18:22.36 | vader-- | i want to fill a file with dates that we are not in the building |
| 18:22.44 | mountainm2k | well, not from what you wrote... |
| 18:22.46 | mountainm2k | Hmmm... |
| 18:23.05 | mountainm2k | Yeah, no : after the ? |
| 18:23.12 | [TK]D-Fender | mountainm2k: Break up that gog and ask yourself what variables are filled with and what will happen.... |
| 18:23.13 | mountainm2k | or else I need a != as my expr |
| 18:23.27 | [TK]D-Fender | mountainm2k: That might help in a way..... |
| 18:23.40 | [TK]D-Fender | mountainm2k: read the logic :) |
| 18:23.46 | mountainm2k | It seems the GotoIf was incorrect -- bassackwards logic |
| 18:24.00 | mountainm2k | I reversed the logic in the GotoIf, and it does what I wanted... |
| 18:24.04 | [TK]D-Fender | mountainm2k: :) |
| 18:24.37 | mountainm2k | [TK]D-Fender: heh, guess I should learn to actually read the errors... I usually am pretty good at that... |
| 18:24.40 | *** join/#asterisk zeppelin_ (n=a@200.213.49.77) |
| 18:25.46 | seicherl | [TK]D-Fender: sorry, i was AFK. thanks for your help. that was an importain lesson to learn. |
| 18:25.59 | [TK]D-Fender | seicherl: np |
| 18:26.18 | *** join/#asterisk c4t3l (n=c4t3l@69.15.174.114) |
| 18:26.25 | [TK]D-Fender | vader--: well you can make a seperate file with the pile of dates in it to merge into your main extensions.conf if you like... |
| 18:27.45 | *** join/#asterisk new2voip (n=new2voip@secure.kayacorp.com) |
| 18:29.58 | *** join/#asterisk dijungal (n=dijungal@64.86.52.254) |
| 18:30.37 | dijungal | What calling card program can i use with trixbox. Let's say if i want to issue calling cards..? |
| 18:31.03 | boch | is it possible to do: host=x.x.x.x,x.x.x.x in a peer entry in sip.conf ? |
| 18:31.07 | Strom_C | if you want to run a calling card company, don't use trixbox |
| 18:31.11 | [TK]D-Fender | boch: Yes |
| 18:31.18 | dijungal | use what.? |
| 18:31.30 | [TK]D-Fender | dijungal: Normal * like the rest of us |
| 18:31.32 | *** join/#asterisk spatulamaan (n=ggilmore@207.188.8.252) |
| 18:31.41 | dijungal | Storm_C: I wanna do an internet cafe with voip service also |
| 18:31.50 | Strom_C | my name is not storm |
| 18:31.56 | dijungal | but trixbox is soo much easier to manage |
| 18:31.58 | *** part/#asterisk spatulamaan (n=ggilmore@207.188.8.252) |
| 18:32.03 | *** join/#asterisk budairc (n=chatzill@200.215.57.173) |
| 18:32.08 | dijungal | oooh crap... Strom_C... lol |
| 18:32.21 | [TK]D-Fender | dijungal: Yes, an idiot interface for idiot level functionality. |
| 18:32.30 | Strom_C | dijungal: if you can't use vi, you're not really in a position to be administering a production PBX |
| 18:32.30 | dijungal | so what calling card service can i add to asterisk...? |
| 18:32.47 | [TK]D-Fender | dijungal: Go look on www.voip-info.org first then come back. |
| 18:32.51 | dijungal | i don't like vi ... i use nano |
| 18:33.06 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 18:33.08 | [TK]D-Fender | dijungal: Good then use nano and lean * thr proper way |
| 18:33.08 | budairc | hi.. |
| 18:33.11 | dijungal | i'm always on voip-info.org... that's how i found this IRC channel |
| 18:33.45 | dijungal | i've tried the proper way month ago.. by configuring the .confs but it's too tedious for what i wanna do |
| 18:34.00 | dijungal | the asterisk box is on;y gonna be running prolly 4 or 5 phones.. nothing big |
| 18:34.13 | dijungal | then i'm thinking of running the calls through vonage or skype out |
| 18:34.17 | Strom_C | ~hafc |
| 18:34.26 | jbot | i guess hafc is hire a freaking consultant. Look, if you're having difficulty understanding what you're doing and need a solution soon, you will be far better off hiring a competent consultant than continuing to pull your hair out. |
| 18:34.26 | [TK]D-Fender | dijungal: Then you clearly missed the obvious page : http://www.voip-info.org/wiki/view/Asterisk+Prepaid+Applications |
| 18:34.26 | dijungal | cause asterisk out seems to be more expensive these days |
| 18:34.51 | [TK]D-Fender | dijungal: Which I GOOGLED in like 10 seconds flat in response to your saying you looked there first and felt you had to come here for the answer.... |
| 18:35.06 | dijungal | great |
| 18:35.46 | [TK]D-Fender | dijungal: Indeed, HIFC..... |
| 18:35.50 | [TK]D-Fender | HAFC* |
| 18:35.51 | budairc | it's possible change the ring?! one type for internal context and other for external context?? |
| 18:36.19 | *** join/#asterisk flashnet (i=flashnet@kbhn-vbrg-sr0-vl212-213-185-15-166.perspektivbredband.net) |
| 18:36.24 | [TK]D-Fender | budairc: try rephrasing that into an actual sentence please.... |
| 18:37.10 | budairc | [TK]D-Fender: sorry.. but my english is sux |
| 18:37.29 | *** join/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 18:37.39 | *** part/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 18:37.44 | [TK]D-Fender | budairc: I'll respond to french as well.... take your pick. |
| 18:38.08 | Strom_C | Mauvais numero! |
| 18:38.17 | Strom_C | and that concludes my entire knowledge of French |
| 18:39.18 | seicherl | [TK]D-Fender: great. i can also make my own public extensions and redirect them to internal users ^^. thanks man. i think i now have a whole lot more of a clue about that than before |
| 18:40.03 | DasTech | ok I need help understanding what is to be done for pix firewall I am fallowing the wiki page but it says create a protocal where what file |
| 18:40.10 | DasTech | its not very clear |
| 18:40.21 | DasTech | http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 18:40.38 | [TK]D-Fender | Strom_C: Add "Va-t'en mon ostie, et retournera jamais TABARNAC!" to your repetoire then ;) |
| 18:40.48 | Strom_C | what does that mean? |
| 18:40.52 | [TK]D-Fender | seicherl: Great to hear. |
| 18:41.03 | [TK]D-Fender | Strom_C: "Hello newb!" ;) |
| 18:41.05 | mountainm2k | Can I call a macro from within a macro? |
| 18:41.09 | Strom_C | haahah |
| 18:41.32 | [TK]D-Fender | mountainm2k: Yes |
| 18:41.34 | Strom_C | what's the literal translation? |
| 18:41.45 | seicherl | [TK]D-Fender: may i ask you one last thing? i need a procedure to show a client which extension or number was called (not calling). you know how to do that? |
| 18:42.14 | *** join/#asterisk variable_office (n=variable@Adv-Proprietary-Systems.s7-0-0.2-15-0.ar4.CHI1.gblx.net) |
| 18:42.18 | [TK]D-Fender | Strom_C: Just because of the nesting I'd read it back as "Get the fuck outta here and never come back (asshole)". |
| 18:42.26 | mountainm2k | in http://pastebin.ca/162588 -- the zero-out (to "o") isn't working... |
| 18:42.30 | mountainm2k | just ignores the digits... |
| 18:42.37 | [TK]D-Fender | Strom_C: The french swearing doesn't translate literally as one might think... |
| 18:42.38 | variable_office | does anyone here use voipjet with asterisk? |
| 18:42.45 | seicherl | [TK]D-Fender: something like rewriting the CallerID to the number that was called |
| 18:42.54 | Strom_C | heh |
| 18:42.57 | Strom_C | seicherl: easy |
| 18:43.09 | seicherl | Strom_C: tell me |
| 18:43.16 | Strom_C | seicherl: Set(CALLERID(num)=${EXTEN}) |
| 18:43.29 | [TK]D-Fender | seicherl: Yes, that'd be one way, but there are some discreet ways. Like prfixing the CID name with a MINIMAL set of chars to indicate the channel the call came in on. |
| 18:43.49 | Strom_C | that too; mine is the "lol bruteforce" method |
| 18:44.02 | *** join/#asterisk J4k3 (i=jsuter@dhcp-12-197-128-61.intrastar.net) |
| 18:44.05 | Strom_C | simple and totally not what you'd actually want to do |
| 18:44.07 | [TK]D-Fender | seicherl: I suggest using only a prefix to the NAME rather than losing all of the original CID info. |
| 18:44.30 | budairc | [TK]D-Fender: so.. can i change the ring (type)? for external context one type of ring.. and for internal context other!? (understand?) |
| 18:44.35 | [TK]D-Fender | Strom_C: Brute is highly effective, but so very little difference to improve it vastly. |
| 18:44.52 | seicherl | [TK]D-Fender: the thing is, all SID-accounts should lead to one client phone and the secretary there shall know what company is beeing called |
| 18:45.03 | [TK]D-Fender | budairc: depends, but likely yes. |
| 18:45.22 | mitcheloc | does anyone have the bt500? |
| 18:45.25 | [TK]D-Fender | seicherl: how many incoming DID's do you need to uniquely ID? |
| 18:45.26 | Strom_C | seicherl: better idea: set distinctive ringing for different companies, or have the different DNIS route to different line appearances |
| 18:45.32 | mitcheloc | *jabra bt500 |
| 18:45.39 | *** part/#asterisk sigwerk (n=sigwerk@cyclone.sigterm.net) |
| 18:46.06 | seicherl | [TK]D-Fender, Strom_C: there are about 20-50 different SID-accounts ^^ |
| 18:46.17 | seicherl | [TK]D-Fender, Strom_C: different ringtones are a bad idea |
| 18:46.25 | Strom_C | maybe you should look into doing screenpops or something then |
| 18:46.35 | Strom_C | running an answering service, are you? :) |
| 18:46.43 | seicherl | Strom_C: depends on the client i'm using |
| 18:46.52 | seicherl | Strom_C: somehting like that |
| 18:47.17 | Strom_C | call center? |
| 18:47.31 | seicherl | Strom_C: no. but never mind. |
| 18:47.44 | Strom_C | hey, I was just guessing :) |
| 18:47.52 | *** part/#asterisk variable_office (n=variable@Adv-Proprietary-Systems.s7-0-0.2-15-0.ar4.CHI1.gblx.net) |
| 18:47.54 | Strom_C | one option is this: |
| 18:47.56 | seicherl | can i do some string-manipulation to the Calleid? |
| 18:48.23 | Strom_C | use queues. Have a separate queue with agent callbacks for each company, and have it play the operator a recording before patching the caller through |
| 18:48.52 | seicherl | Strom_C: hmm.... got no clue about queues yet |
| 18:48.54 | seicherl | brb |
| 18:49.01 | Strom_C | so (s)he picks up, hears "Super Happy FunCo" and then knows to answer "Super Happy FunCo, can i help you?" |
| 18:49.45 | smackus | is there a way in one line to do RemoveQueueMember <member name> from <ALL QUEUES>? is there a variable or some other phrase so that I do not have to go through and do them all one at a time? |
| 18:50.44 | smackus | seicherl: i just use the caller id and they like it alot |
| 18:50.47 | seicherl | Strom_C: well... sounds not bad. dont know if they want it like that. |
| 18:51.07 | Strom_C | that seems like the best way to do it in lieu of screen pops |
| 18:51.09 | seicherl | smackus: you got something like that? |
| 18:51.12 | LoneShadow | anyone in here who is good with Sipura 3000 PSTN setup ? |
| 18:51.19 | Strom_C | altering the caller ID information is generally a Bad Idea(tm) |
| 18:51.20 | *** join/#asterisk C6Vette (n=info@72-166-37-114.dia.static.qwest.net) |
| 18:51.25 | smackus | yes |
| 18:51.49 | seicherl | Strom_C: well, its a bloody workaround... nasty hack. but if it works, noone asks ^^ |
| 18:52.15 | Strom_C | seicherl: having the recording is a better idea |
| 18:52.24 | seicherl | can anyone gimme a codesnipplet? or tell me where to put that statement? |
| 18:52.37 | Strom_C | so you have one queue for Super Happy FunCo, another for Free Candy Athletic Shoes Inc., and so on |
| 18:52.39 | seicherl | Strom_C: one after another. i'm just learning |
| 18:53.24 | dijungal | exit |
| 18:53.41 | seicherl | Strom_C: and i will have to ask my customer anyway what they want. but before i must have at least one solution |
| 18:55.21 | *** join/#asterisk justinu|laptop (n=Justin@12.44.122.130) |
| 18:55.27 | C6Vette | ${STRFTIME(${EPOCH},,%I%M%S)} reflects GMT time in dialplan is there a setting somewhere to compensate for timezones? |
| 18:55.38 | SplasPood | Other than the fact that almost every device on earth supports RADIUS, any other benefits to using it to log my CDRs rather than direct to mysql, etc? |
| 18:56.20 | MikeJ | radius loggin would typically be realtime cdrs, so if your box takes a dump mid call, you would still get a record... |
| 18:56.37 | MikeJ | I don;t know that there is a realtime radius cdr for asterisk |
| 18:57.08 | MikeJ | all the major billing packages generally uses radius |
| 18:57.40 | seicherl | Strom_C: although its no good idea ^^ can i write " exten => 1234,n,Set(CALLERID(num)=somestring) "? |
| 18:58.12 | *** join/#asterisk crich1999 (n=crich@port-212-202-201-41.dynamic.qsc.de) |
| 18:58.29 | SplasPood | MikeJ: Well wouldn't using cdr_odbc or _mysql to a remote db server be the same diff.. oh mid call... you mean two records.. start/end rather than just 1 at the end? |
| 18:59.24 | Nugget | See? This is exactly why I log to /dev/null. No messy ambiguity. |
| 19:01.10 | seicherl | smackus: you manipulate the callerid, you said. how would i do that? can you gimme a codesnipplet or some explaination? |
| 19:01.31 | mountainm2k | http://pastebin.ca/162650 -- extension "a" works (Thanks TK!), but extension "o" does not... Any ideas? |
| 19:01.35 | *** join/#asterisk spr1te (n=spr1te@194.187.130.229) |
| 19:01.48 | SplasPood | Nugget: :) |
| 19:01.53 | dlynes_laptop | telnet |
| 19:01.55 | Nugget | telnet is eeeeeeevil! |
| 19:01.57 | sb_mx | C6Vette, yes. this link might help you http://www.voip-info.org/wiki/index.php?page=Asterisk+func+strftime |
| 19:02.24 | *** join/#asterisk vlt (n=daniel@dslb-088-073-199-192.pools.arcor-ip.net) |
| 19:03.38 | [TK]D-Fender | mountainm2k: For "o" to work you need to set that up in voicemail.conf on a per-box basis with "|operator=yes" |
| 19:04.37 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 19:05.22 | mountainm2k | Ahhhhhh, of course it does... Thanks again TK... (slaps himself in the face... again...) |
| 19:06.23 | mountainm2k | Yup, that sure fixed it... |
| 19:07.28 | rift0r | what's the cheapest place online to get voip stuff... is voiplink seems pretty cheap, are they reliable? |
| 19:07.53 | Nugget | mountainm2k is making is "o" face. |
| 19:08.29 | *** join/#asterisk hotroot (n=michael@pD9E96F35.dip.t-dialin.net) |
| 19:08.45 | *** join/#asterisk nortex (n=breeves@snapper.titanspecialties.com) |
| 19:08.53 | Adrian__ | hallo michael |
| 19:08.56 | seicherl | Strom_C: how would i manipulate the CALLERID? i tried but i still get the original callerid at the client phone |
| 19:09.29 | Strom_C | pastebin extensions.conf |
| 19:09.42 | wunderkin | rift0r, the only advice i can give right now is not to order from redorbit, i placed an order today from voipsupply and so far it seems good |
| 19:09.56 | *** join/#asterisk spr1te (i=spr1te@194.187.130.229) |
| 19:10.10 | Nugget | I've bought several phones from voipsupply with no problems at all. |
| 19:10.33 | wunderkin | my only problem with voipsupply right now is that it took them an hour before anyone actually got in from sales |
| 19:10.40 | seicherl | Strom_C: http://pastebin.ca/162671 |
| 19:11.05 | justinu|laptop | i don't like voipsupply... got a big runaround with some cisco 7960s i bought from them |
| 19:11.05 | Strom_C | try Set(CALLERID(name)= |
| 19:11.43 | justinu|laptop | not in stock, after they said they were... shipped bad phone/power supply, took over 30 days to complete RMA process, etc. |
| 19:11.58 | Strom_C | yeah, voipsupply dicked around with a client of mine with regards to credit card processing |
| 19:12.02 | Strom_C | delayed the install by a week |
| 19:12.06 | Strom_C | never again with voipsupply |
| 19:12.09 | [TK]D-Fender | Cisco is eeeeevil! |
| 19:12.09 | justinu|laptop | agreed |
| 19:12.19 | bkw_ | cisco isn't evil they do actually work |
| 19:12.22 | kannan | hi to all |
| 19:12.27 | vlt | Hello. I just discovered features.conf and the transfer functions. |
| 19:12.29 | seicherl | hehe |
| 19:12.29 | vlt | Where can I define the AT timeout? |
| 19:12.34 | seicherl | Strom_C: works! |
| 19:12.38 | justinu|laptop | yah, phones are fine once I got non DOA hardware |
| 19:12.43 | [TK]D-Fender | bkw_: Doesn't preclude their being evil :) |
| 19:13.11 | rift0r | ok so I am looking at the SPA2002 and Grandstream HandyTone 496 |
| 19:13.15 | wunderkin | vlt, i was looking for that too, it didnt seem to be in the example conf, i think i just had to change it in the source, i dont think there is a setting for it yet |
| 19:13.15 | rift0r | i may just pick up both =/ |
| 19:13.31 | Strom_C | rift0r: RUN, do not walk, from grandstream |
| 19:13.41 | rift0r | ok you are the second person to say that |
| 19:13.42 | rift0r | :P |
| 19:13.44 | *** join/#asterisk RoyK (n=roy@ti211210a080-1761.bb.online.no) |
| 19:13.45 | seicherl | Strom_C: man, thank you! that was one of the most importaint things to do today! |
| 19:13.48 | rift0r | so SPA2002 it is |
| 19:13.56 | Strom_C | you're welcome |
| 19:13.57 | [TK]D-Fender | rift0r: Yup |
| 19:14.06 | rift0r | hehe |
| 19:14.07 | kannan | i am build one asterisk server on linux slackware for my friend,I would like to support him remotly from windows xp.how I get the xwindow of linux slack on my windows xp |
| 19:14.11 | justinu|laptop | only customers i have who like Granstream are NOCs |
| 19:14.13 | vlt | wunderkin d: Thank you. |
| 19:14.18 | vlt | How can I stop calling the phone I try to transfer the call to (would be flash on an ordinary PBX)? |
| 19:14.18 | bkw_ | [TK]D-Fender, so qualify that satement? Why is cisco evil? |
| 19:14.19 | seicherl | Strom_C: is it hard to get fax supported on asterisk? |
| 19:14.33 | wunderkin | vlt, the hangup feature code |
| 19:14.36 | Strom_C | seicherl: faxing is like 10% skill, 20% knowledge, and 70% voodoo |
| 19:14.42 | seicherl | lol |
| 19:14.50 | seicherl | Strom_C: is it worth trying? |
| 19:14.54 | Qwell | only 70? you must have years of fax experience |
| 19:14.59 | Strom_C | seicherl: if you're in an all-TDM environment it will work |
| 19:15.08 | seicherl | all-TDM |
| 19:15.10 | seicherl | ? |
| 19:15.13 | Strom_C | no voip |
| 19:15.15 | Strom_C | all T1s |
| 19:15.21 | vlt | wunderkin: The *0 by default? |
| 19:15.22 | seicherl | lol. i have VOIP only |
| 19:15.24 | c4t3l | time division multiplexing |
| 19:15.41 | wunderkin | vlt, not sure, do a show features |
| 19:16.09 | seicherl | something like fax-to-mail working? |
| 19:16.18 | [TK]D-Fender | bkw_: They resist working with the more open standards, etc.... |
| 19:16.34 | c4t3l | fax-to-email does work |
| 19:16.40 | bkw_ | [TK]D-Fender, didn't they just drop SCCP in favor of SIP? |
| 19:16.48 | Qwell | bkw_: allegedly, but... |
| 19:16.53 | seicherl | c4t3l. from within asterisk? |
| 19:16.54 | Qwell | no clue how true that is |
| 19:16.57 | bkw_ | they did |
| 19:17.07 | Qwell | bkw_: I plan on having a chat with a cisco sales guy at VON |
| 19:17.10 | vlt | wunderkin: There's a "Disconnect Call * *0" line |
| 19:17.10 | bkw_ | the fact that I have a 7970 on my desk right now running SIP sums it up |
| 19:17.12 | Qwell | or a tech, if there is one |
| 19:17.15 | Qwell | bkw_: true |
| 19:17.31 | c4t3l | seicher, no. using a pair of programs known as iaxmodem and hylafax |
| 19:17.47 | c4t3l | asterisk does help tho :( |
| 19:17.52 | wunderkin | vlt, yes |
| 19:18.10 | seicherl | c4t3l: ok... i will check that the other day. now i start celebrating the manipulation of the callerid ^^ |
| 19:18.22 | vlt | How can I set up the following: Hitting *2 and a target number dials a phone. If I wait I can talk to the 3rd person and then transfer the call by hanging up (that's how it works now), but when I hangup while the other phone still rings I want the behavoir of hitting #1: Connecting the other two phones. |
| 19:18.23 | vlt | In short: a mix of Attended and Blind Transfer |
| 19:18.23 | c4t3l | hehe |
| 19:18.25 | justinu|laptop | SCCP is pretty cool.. an open specification would have helped it out |
| 19:18.32 | bkw_ | true |
| 19:18.37 | Qwell | justinu|laptop: ...it would have helped immensely |
| 19:18.42 | seicherl | c4t3l: just have to check if it works on the hardware-phone they use aswell^^ |
| 19:18.44 | Qwell | and yes, sccp as a protocol is very cool |
| 19:19.11 | c4t3l | seicher, ahh. Good luck |
| 19:19.16 | Qwell | justinu|laptop: glad to see I've infected you, btw ;) |
| 19:19.20 | justinu|laptop | heh |
| 19:19.20 | Strom_C | ~faxing |
| 19:19.21 | jbot | it has been said that faxing is 8% knowledge, 5% skill, 11% luck, and 76% voodoo |
| 19:19.21 | seicherl | bye |
| 19:19.32 | c4t3l | faxing sucks! |
| 19:19.35 | justinu|laptop | i like thin clients |
| 19:19.36 | seicherl | hehe. |
| 19:19.42 | seicherl | bye mates |
| 19:19.54 | Qwell | sccp is so incredibly flexible |
| 19:19.56 | file | every time you fax... the fax machine takes a piece of your soul |
| 19:20.04 | bkw_ | c4t3l, faxing rocks |
| 19:20.08 | bkw_ | when you do it with cisco gear |
| 19:20.09 | Qwell | every time you fax over voip, god kills a kitten |
| 19:20.14 | c4t3l | why do poeple STILL insist on using it???? |
| 19:20.31 | Qwell | c4t3l: because it's difficult to sign emails? |
| 19:20.57 | c4t3l | we should all just move back into caves and use paper cups and string! |
| 19:21.04 | bkw_ | because it works |
| 19:21.07 | bkw_ | faxing has been reliable for ages |
| 19:21.18 | c4t3l | ! |
| 19:21.26 | bkw_ | yes it has |
| 19:21.34 | bkw_ | I do millions of faxes a month on cisco gear |
| 19:21.43 | justinu|laptop | fax is a really funky mix of ancient protocols |
| 19:21.44 | bkw_ | and have very few errors |
| 19:22.14 | c4t3l | i've gotten fax to work reliably with *, but I just hate it so much! |
| 19:22.16 | *** join/#asterisk anthonyl (n=Default@c-71-57-41-193.hsd1.il.comcast.net) |
| 19:22.42 | bkw_ | things like Digium hardware cause fax not to work on TDM |
| 19:22.52 | justinu|laptop | because there's no PLC? |
| 19:22.54 | tzanger | bkw_: I have no trouble with digium hardware and faxing |
| 19:23.06 | tzanger | at least not digitally, I haven't had an analog line in a long time now though ot test taht |
| 19:23.09 | tzanger | justinu|laptop: no |
| 19:23.11 | bkw_ | tzanger, we have had issues with frame slips |
| 19:23.13 | justinu|laptop | no jitter buffers? |
| 19:23.15 | Qwell | my faxing solution is the best |
| 19:23.22 | tzanger | Qwell: fax machine? :-) |
| 19:23.29 | Qwell | I go to Kinkos, hand them a piece of paper, and a dollar, and it gets to the recipient...maybe |
| 19:23.32 | bkw_ | and doesn't DTMF cause zaptel drama? |
| 19:23.52 | justinu|laptop | popcopy! |
| 19:24.03 | tzanger | bkw_: I haven't had a chance ot revisit that... it *did* but I odn't think it does that any more |
| 19:24.41 | vlt | How can I let my phone ring again after an unsuccessful blind transfer (#1)? |
| 19:24.47 | *** join/#asterisk rajiv|work (n=rajiv@gentoo/developer/rajiv) |
| 19:25.28 | rajiv|work | anyone know why * might stop aswering zaptel calls ? |
| 19:25.29 | *** join/#asterisk angom (n=angom@red-corp-200.79.133.82.telnor.net) |
| 19:26.06 | rajiv|work | i have a sangoma a200 and it was working fine for 33 days. but it just stopped asnwering ports 2, 3, and 4. |
| 19:26.59 | *** join/#asterisk DarKnesS_WolF (n=wolf@196.218.41.120) |
| 19:27.09 | *** join/#asterisk eKo1 (n=eKo1@190.4.7.90) |
| 19:27.27 | *** join/#asterisk hmmhesays (n=hmmhesay@24-117-135-28.cpe.cableone.net) |
| 19:27.38 | *** join/#asterisk Toadyus (i=Toadyus@S010600121746f9fe.mh.shawcable.net) |
| 19:28.01 | *** join/#asterisk Egonis (n=Egonis@207.245.14.10) |
| 19:28.19 | Egonis | How do I allow incoming callers to exit Voicemail and return to the main menu? |
| 19:29.35 | wunderkin | Egonis, show application voicemail |
| 19:30.10 | vlt | How can I park a call? I thought by pressing 700 while active call but that didn't work. Do I have to transfer the call to 700 by pressing #1 700? |
| 19:30.24 | hotroot | yes |
| 19:30.30 | Strom_C | you must do an attended transfer to 700 |
| 19:30.33 | vlt | hotroot: Thanks |
| 19:31.09 | vlt | But then how can I hear the number it is parked to? |
| 19:31.22 | Egonis | wunderkin: so by making an extension 'exten => a,1,Goto(blah,s,1) would allow VoiceMail(u0) to return? |
| 19:31.22 | hotroot | astersik will tell you |
| 19:31.25 | Strom_C | that's why you do an ATTENDED transger |
| 19:31.31 | Strom_C | er, transfer |
| 19:31.50 | vlt | Strom_C: by pressing *2 (in default conf)? |
| 19:32.06 | Strom_C | vlt: beats me; i always use the transfer feature on my sip phone |
| 19:32.10 | vlt | Think I got it know ;-) |
| 19:32.59 | vlt | Strom_C: And when you use sip transfer you can hear asterisk telling you the parking lot? |
| 19:33.26 | *** part/#asterisk hotroot (n=michael@pD9E96F35.dip.t-dialin.net) |
| 19:34.50 | [TK]D-Fender | vlt: You'd have to use a catch-all cotext that would attempt to pass-on the call and return on failure |
| 19:35.29 | *** part/#asterisk shap (n=shap@c-68-33-84-43.hsd1.md.comcast.net) |
| 19:35.59 | vlt | Is there an example extensions.conf somewhere? |
| 19:36.08 | *** join/#asterisk kronenpj (n=chatzill@rrcs-71-41-238-3.se.biz.rr.com) |
| 19:36.14 | *** join/#asterisk frenzy (n=frenzy@196.46.104.119) |
| 19:36.20 | DarKnesS_WolF | [TK]D-Fender: where i can read about asterisk codecs ? and if i want asterisk to use a codec based on the client .. i mena the client chose the codecs not astersik . any idea ? |
| 19:36.32 | frenzy | hello all... |
| 19:37.22 | rajiv|work | hmm. i'm not the only one: http://lists.digium.com/pipermail/asterisk-users/2006-May/152961.html |
| 19:38.10 | frenzy | I have an issue with my phone provider... when a call comes in and the caller hangs up the phone provider isnt sending any tones or hangup messages. |
| 19:38.50 | [TK]D-Fender | vlt: There are no "magic examples" for such things. |
| 19:38.55 | [TK]D-Fender | DarKnesS_WolF: Read the WIKI and the BOOK, aside from that, no idea. |
| 19:39.01 | frenzy | I have to hangup for the call to really hangup... Is it possible for asterisk to hangup on silence? |
| 19:39.11 | DarKnesS_WolF | [TK]D-Fender: all the wiki :P? |
| 19:39.21 | DarKnesS_WolF | [TK]D-Fender: thx i'll dig the book as i think :-) |
| 19:39.40 | [TK]D-Fender | DarKnesS_WolF: Get off your ass and read the resources you know are there... |
| 19:40.04 | DarKnesS_WolF | [TK]D-Fender: dah why u assum i'm not geting my ass and reading ? i do ! but sometimes i have problem to know what i need to search for |
| 19:42.15 | *** join/#asterisk X-Gen (n=X-Gen@dsl-145-216-31.telkomadsl.co.za) |
| 19:43.20 | Strom_C | frenzy: are you using VoIP or POTS? |
| 19:43.26 | *** join/#asterisk hotroot (n=michael@pD9E96F35.dip.t-dialin.net) |
| 19:44.10 | frenzy | Zap (Analog Pots) -> Asterisk -> SIP |
| 19:44.32 | *** join/#asterisk Vorondil (n=vorondil@64.191.168.244) |
| 19:44.34 | Strom_C | frenzy: get your telco to add disconnect supervision to your phone line |
| 19:45.07 | frenzy | i dont think they would be willing to do that :) |
| 19:45.12 | Strom_C | why not? |
| 19:45.13 | frenzy | whats hanguponpolarityswitch? |
| 19:45.23 | Strom_C | who is your telco? |
| 19:45.25 | frenzy | thick folks :( |
| 19:45.38 | Adrian__ | i seam to have some problems with ENUM/direct IP calls. they only work one way. the other party hears me but i do not hear them. i am behind a NAT - i checked the ports and firewall rules, ports are forwarded according to my rtp conf (10000-10100) and i have according firewall rules to allow that traffic - any ideas? |
| 19:45.40 | frenzy | some african monopoly teleco |
| 19:46.01 | *** join/#asterisk phalacee (n=Sunforge@202.3.110.33) |
| 19:46.02 | eKo1 | Adrian__: What happens if you turn of the firewall? |
| 19:46.47 | Adrian__ | eKo1 - there is no easy way to do that... :/ |
| 19:48.38 | [TK]D-Fender | Adrian__: Pastebin the [general] section from your sip.conf |
| 19:48.47 | [TK]D-Fender | ~pb |
| 19:48.48 | jbot | from memory, pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/ (BROKEN AND SUCKING NUTS), or http://pastebin.ca, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com, or http://bzflag.pastebin.ca/, or http://paste.lisp.org/ for the lisp/scheme nuts, or http://paste-it.net |
| 19:50.16 | *** join/#asterisk dswillia74437 (n=me@199.3.247.34) |
| 19:50.24 | *** join/#asterisk |dennis| (n=dennis@shc.edu.bz) |
| 19:50.45 | dswillia74437 | what is the maximum numbers of stored messages comedian will handle per box? |
| 19:50.51 | *** part/#asterisk Egonis (n=Egonis@207.245.14.10) |
| 19:51.05 | Qwell | dswillia74437: 9999, unless you change a #define in app_voicemail.c |
| 19:51.23 | *** join/#asterisk willy1234 (n=IceChat7@62.231.36.194) |
| 19:51.25 | Qwell | however, there is an option in voicemail.conf, that limits it also |
| 19:51.53 | dswillia74437 | great, has anyone heard of using asterisk just for voicemail off an avaya pbx? |
| 19:52.02 | Qwell | sure |
| 19:52.22 | willy1234 | how do u set up fax to email |
| 19:52.31 | bkw_ | use a cisco AS5300 |
| 19:52.35 | Qwell | willy1234: That is a loaded question |
| 19:52.54 | mountainm2k | willy1234: a VERY loaded question |
| 19:53.09 | [TK]D-Fender | willy1234: Read the WIKI... obvious samples there : Canaplus YTD 2006-08-31 vs YTD 2005-08-31.xls |
| 19:53.14 | mountainm2k | willy1234: I got it working using a program called iaxmodem and HylaFAX... |
| 19:53.14 | willy1234 | what do u mean by loaded |
| 19:53.19 | [TK]D-Fender | willy1234: http://www.voip-info.org/wiki/view/Asterisk+Fax+to+email |
| 19:53.22 | [TK]D-Fender | oops* |
| 19:53.38 | [TK]D-Fender | mountainm2k: No, he's definately shooting blanks ;) |
| 19:53.50 | mountainm2k | lol |
| 19:54.14 | willy1234 | looks hard |
| 19:54.35 | [TK]D-Fender | willy1234: Life is like a dick... if it gets too hard FUCK IT. |
| 19:54.41 | Adrian__ | D-Fender - http://pastebin.ca/162716 |
| 19:54.46 | willy1234 | lol |
| 19:54.56 | mountainm2k | lol |
| 19:55.20 | [TK]D-Fender | willy1234: And not that hard... go read up on SpanDSP while you're at it. start there just being able to RECEIVE a fax before you start worrying about how to spit it back out... |
| 19:55.24 | willy1234 | i have hang up issues on a tdm04B |
| 19:55.36 | *** join/#asterisk anonymouz666 (n=anonymou@200.218.193.4) |
| 19:55.49 | [TK]D-Fender | Adrian__: Thats not quite right..... |
| 19:56.02 | mountainm2k | willy1234: I had a hard time getting the rxfax() and txfax() (the applications provided by SpanDSP -- google that) to work |
| 19:56.05 | willy1234 | ive a fax machine for sending |
| 19:56.25 | mountainm2k | willy1234: So I found iaxmodem (google that too), and HylaFAX (google that too)... |
| 19:56.28 | *** part/#asterisk rados___ (n=rados@c-68-62-71-76.hsd1.mi.comcast.net) |
| 19:56.37 | Adrian__ | D-Fender - what part? |
| 19:56.38 | [TK]D-Fender | Adrian__: http://pastebin.ca/162718 |
| 19:56.55 | [TK]D-Fender | Adrian__: Missing the canreinvites, and its host, not IP and you need the matching part... |
| 19:57.15 | mountainm2k | willy1234: In our company, we gave every single employee a dedicated fax-to-email number... With only a few mods to the HylaFAX script, I had it read from * Realtime for the email address... |
| 19:57.21 | [TK]D-Fender | willy1234: what are you looking to do? Use your fax machine like a scanner? |
| 19:58.14 | willy1234 | i have a fax mackhine connected to a pap2 but it sometimes fails |
| 19:58.25 | mountainm2k | I should document what I did to make it work... |
| 19:58.54 | willy1234 | i would like the pbx to email a copy aswell as send it to the fax |
| 19:59.17 | willy1234 | any tips on tdm04b hang up issues |
| 19:59.31 | mountainm2k | willy1234: Unless your fax machine is also a network printer, I'd say that won't work real well... Again, in my experience... |
| 19:59.41 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 19:59.42 | willy1234 | it wont dectect the pstn hang up signal |
| 20:00.00 | willy1234 | it is a newtork printer |
| 20:00.03 | dswillia74437 | Ok so my biss is ready to throw away our audix system that we use on our avaya, wanting to replace it with asterisk/comedian i dont think i would be able to light the mwi on our avaya phones. Am I wrong? |
| 20:00.04 | mountainm2k | willy1234: But keep in mind -- if you're using VoIP for the incoming phone lines, it's not going to work at all... |
| 20:00.13 | Adrian__ | [TK]D-Fender - ok i filled that stuff in, still doesnt work |
| 20:00.53 | willy1234 | the incomming line is a pstn line it is then sent to the sip fax extension |
| 20:01.28 | [TK]D-Fender | Adrian__: Check your headers on a SIP debug |
| 20:01.30 | mountainm2k | willy1234: OK... I've found that my fax machine, connected to a TDM400 analog port works reliably... I have not tried a SIP ATA... |
| 20:01.53 | [TK]D-Fender | willy1234: Doesn't work that way. * can't spy passively on a fax like that |
| 20:01.59 | mountainm2k | willy1234: I use that for outbound, and the iaxmodem / HylaFAX solution for inbound (incoming faxes are only emailed, NOT printed) |
| 20:02.10 | willy1234 | ok |
| 20:02.19 | willy1234 | thanks |
| 20:02.23 | [TK]D-Fender | willy1234: * would have to receive it electronically through something like SpanDSP/hylafax and then FORWARD it internally. |
| 20:03.10 | [TK]D-Fender | willy1234: I would suggest using SpanDSP for inbound fax in that case and use your fax machine for only outbound if you NEED to have fax-2-email. |
| 20:03.16 | willy1234 | if i can get it to email then they could print it out |
| 20:03.31 | [TK]D-Fender | willy1234: Otherwise I'd say get * the hell away from your fax machine and leave it on its own line. |
| 20:03.57 | [TK]D-Fender | willy1234: Upon receipt there is no reason you couldn't have your * print it directly itself. |
| 20:04.27 | willy1234 | yeah i was tying to use line hunting to use the fax line as a fall over to the main line when it was busy |
| 20:05.11 | mountainm2k | Heh, I set up 3 fax ports... no more busies... lol |
| 20:05.44 | willy1234 | ok the TDM04B hang up problem is more important to solve |
| 20:05.49 | [TK]D-Fender | I use SpanDSP on my PRI with WAY more channels that I can forsee needing... same here ;) |
| 20:06.04 | [TK]D-Fender | willy1234: Ask your telco to add "disconnect supervision" to the line. |
| 20:06.13 | willy1234 | ok |
| 20:06.21 | willy1234 | what does that do |
| 20:06.33 | *** join/#asterisk _deg_ (n=deg@200.163.193.247) |
| 20:06.50 | file | willy1234: it adds a way for the card to know that someone hung up |
| 20:06.57 | file | polarity reversal I believe |
| 20:07.04 | mountainm2k | willy1234: Basically the PSTN will signal your * box that the guy on the other end hung up... |
| 20:07.13 | mountainm2k | willy1234: Usually drops the talk-battery for 500ms |
| 20:07.17 | mountainm2k | but sometimes it is reversed |
| 20:07.31 | willy1234 | cool |
| 20:07.32 | [TK]D-Fender | willy1234: 2 popular options : momentary circuit-cut, or polarity reversal. |
| 20:07.38 | willy1234 | thanks so much |
| 20:07.49 | kannan | i am build one asterisk server on linux slackware for my friend,I would like to support him remotly from windows xp.how I get the xwindow of linux slack on my windows xp |
| 20:07.53 | file | [TK]D-Fender: I'm calling today... Sleepy Wednesday |
| 20:08.07 | willy1234 | so will the provider be able to do that? |
| 20:08.10 | [TK]D-Fender | kannan: Who needs xwindows? :) Get Putty :) |
| 20:08.30 | [TK]D-Fender | file: Calling for what? Your new system still? |
| 20:08.31 | kronenpj | kannan: Look at Exceed (purchase) or Cygwin for an X server for your windows box. |
| 20:08.44 | file | [TK]D-Fender: no, I just slept really horribly lastnight |
| 20:08.46 | kronenpj | kannan: Then use PuTTY to tunnel the X connection securely through SSH. |
| 20:09.11 | kannan | i need linux slack xwindow(server) from my windows xp(client) |
| 20:09.32 | kannan | in putty its show only command promt |
| 20:09.48 | kannan | somebody says vnc server |
| 20:10.01 | kannan | show xwindow of linux |
| 20:10.10 | Qwell | kannan: nx is WAY better than vnc |
| 20:10.12 | kannan | but i dont know how its work |
| 20:10.32 | Qwell | kronenpj: That's almost exactly what nx does |
| 20:10.45 | kronenpj | Qwell: And a little easier to set up, in my experience. |
| 20:10.46 | Qwell | except it's got a lot of tweaks to make it really fast |
| 20:10.53 | kannan | where i get that |
| 20:10.59 | mountainm2k | kannan: Incidently, I've been administering Unix/Linux boxes for a long time using only PuTTY (without xwindows) |
| 20:11.11 | kannan | ok |
| 20:11.23 | [TK]D-Fender | besides, who installs Slackware to run X? :) |
| 20:11.27 | *** join/#asterisk JunK-Y (n=junky@modemcable205.175-81-70.mc.videotron.ca) |
| 20:11.30 | kannan | i would like work on my xwindow |
| 20:11.30 | mountainm2k | kannan: and Asterisk is text-only -- there isn't much of a GUI to it... |
| 20:11.34 | [TK]D-Fender | (aside from me) ;) |
| 20:11.49 | kannan | admin pages |
| 20:12.02 | mountainm2k | what admin pages? |
| 20:12.10 | [TK]D-Fender | kannan: Again X doesn't help you with * in any meaningful way... |
| 20:12.39 | [TK]D-Fender | (prepare for FreePBX/Trixbox revelation) |
| 20:13.06 | mountainm2k | kannan: If you insist on using X, I recommend X-Win32 -- it has an SSH client built in, and is less expensive than Xceed (but more than Cigwin)... |
| 20:13.15 | mountainm2k | kannan: www.starnet.com |
| 20:13.23 | Qwell | mountainm2k: Then you haven't looked at nx :) |
| 20:13.26 | kannan | if i do any changes on vicidial,astguiclient admin pages to change the settings,in command promt it dosnt show properly |
| 20:13.42 | Qwell | mountainm2k: I swear, if you use it, you'll never use anything else again, heh |
| 20:13.51 | mountainm2k | Qwell: Send URL pls? |
| 20:13.58 | Qwell | nx.com? dunno |
| 20:14.02 | mountainm2k | Qwell: Incidently, I never use Xwin32 either, although I did buy it... |
| 20:14.05 | Qwell | It's semi-GPLed |
| 20:14.13 | kannan | ok |
| 20:14.15 | Qwell | There is freenx, and the official nx client is free |
| 20:14.19 | mountainm2k | ahhha, nomachine.com ? |
| 20:14.23 | Qwell | that's the one |
| 20:14.32 | *** join/#asterisk new2voip (n=new2voip@secure.kayacorp.com) |
| 20:14.34 | IOscanner | I have inbound/outbound termination from an external provider. I don't have to register. They just route the calls into my PBX. I have a trunk setup for outbound and it works. Inbound is not working. I have a trunk the system answers the call but doesn't seem to know what to do with the call and plays a noservice msg. |
| 20:14.41 | Qwell | it's very, very cool, and very, very fast |
| 20:14.44 | IOscanner | I am on the CLI and I see the DID is set to the correct number. I have an inbound route setup, but it is not using that route. |
| 20:14.53 | Qwell | mountainm2k: for real, I watched a movie in totem over it |
| 20:15.00 | Qwell | and it looked GREAT |
| 20:15.05 | IOscanner | when the call transfers to me I do see: Received incoming SIP connection from unknown peer to 8661111111 |
| 20:15.17 | IOscanner | any ideas what I might be missing? |
| 20:15.20 | file | IOscanner: are you a FreePBX/Trixbox user? |
| 20:15.29 | mountainm2k | Qwell: Interesting, I'm going to give it a try... |
| 20:15.31 | IOscanner | Yes FreePBX |
| 20:15.36 | Qwell | mountainm2k: iirc, it can do sound and printers and stuff too :) |
| 20:15.47 | kannan | yes |
| 20:16.05 | file | IOscanner: yeah... #freepbx - they can help you hopefully get it configured properly... |
| 20:16.19 | mountainm2k | Qwell: Not like I have any machines requiring X |
| 20:16.25 | mountainm2k | Qwell: but still... |
| 20:17.24 | IOscanner | I think it is just a trunk configuration issue. I know the system Just not sure what asterisk needs to allow this trunk to be passed correctly |
| 20:22.59 | *** join/#asterisk lehel (n=mey@86.125.118.244) |
| 20:23.07 | lehel | hello |
| 20:28.11 | *** join/#asterisk hmmhesays (n=hmmhesay@24-117-135-28.cpe.cableone.net) |
| 20:28.15 | hmmhesays | well I got my 2102 |
| 20:28.17 | hmmhesays | rock |
| 20:28.32 | _deg_ | Is this possible to have Asteterisk 1.2 and Realtime Static? |
| 20:31.01 | [TK]D-Fender | ok, heading home, BBIAB |
| 20:38.06 | c4t3l | does anyone know of a way to find out if a phone is forwarded to itself? |
| 20:39.03 | MikeJ | in a pure sip environment, that is what loop checking is for... most protocols should implement somthing like a max-forwards |
| 20:39.24 | *** join/#asterisk [Yatta] (n=noe@65.183.3.229) |
| 20:39.56 | MikeJ | when you get into a b2bua behavior like asterisk, I am not sure what the sip spec says to do on that, it may be to pass the max forwards (decreased by 1) to the other end in the case of a forward... |
| 20:40.08 | MikeJ | I do not beleive that is acutally done right now... |
| 20:49.16 | *** join/#asterisk flashnet (i=flashnet@kbhn-vbrg-sr0-vl212-213-185-15-166.perspektivbredband.net) |
| 20:51.15 | *** join/#asterisk Serbaniaotic (n=mikep@206.124.12.162) |
| 20:52.09 | Serbaniaotic | quick question, is there any BLA/SLA built with any of the recent releases? |
| 20:54.16 | bkw_ | BLA/SLA? |
| 20:54.29 | Qwell | ~bla |
| 20:54.30 | jbot | foo |
| 20:54.31 | Qwell | ~sla |
| 20:54.32 | jbot | methinks sla is service level agreement. if they're down for more then XX minutes, they pay *YOU* |
| 20:54.43 | Qwell | shared line appearances |
| 20:55.01 | c4t3l | does anyone know of a way to find out if a phone is forwarded to itself? |
| 20:55.02 | Qwell | and I think he means BLF |
| 20:55.06 | Qwell | ~blf |
| 20:55.07 | jbot | it has been said that blf is Busy Lamp Field, aka little lights next to speed dials that light up when the person is on the phone and blink when that line is ringing. hint extensions are static mapped to SIP or other channels. |
| 20:55.15 | Serbaniaotic | bridged line appearance / shared line appearance... the bane of my existence since parking is too many keystrokes according to my users |
| 20:55.21 | *** join/#asterisk volp (n=volp@201.210.82.39) |
| 20:56.02 | new2voip | I am trying to connect an IAX soft phone to my server internally and the client tells me the server cannot be contacted. SIP works fine, any clues as to why IAX client will not connect? |
| 20:56.43 | CtRiX | watch at the logs... may be helpful |
| 21:01.09 | *** join/#asterisk dashu (n=dashu@p549C62CA.dip.t-dialin.net) |
| 21:01.17 | Serbaniaotic | has any one been able to figure out a way to repark to the same parking spot after an accidental pickup? |
| 21:01.25 | mercestes | c4t3l: Check out the polycom -phone.cfg files. |
| 21:03.57 | dashu | :o got some questions just have seen that there is a windows version of asterisk can u tell me if i can do the same stuff with the asterisk for the windows version that i could do with the linux version ? my boss would be happy about it cause we are running some asterisk linux server for quiet some time now but he isnt really good with linux and wanted to be able to use asterisk too |
| 21:04.50 | c4t3l | i've gotta box with a S!@T load of asterisk processes. is there a command line tool that can help me trace which phone may be forwarded to itself? |
| 21:05.13 | Qwell | c4t3l: show channels might give you a clue |
| 21:05.53 | *** join/#asterisk andresmujica (n=andresmu@201.244.243.154) |
| 21:06.55 | *** part/#asterisk volp (n=volp@201.210.82.39) |
| 21:10.21 | dashu | :p i know my english is bad but an answer would still be nice |
| 21:11.00 | MikeJ | dashu... |
| 21:11.11 | MikeJ | the only windows version I know of is asteriskwin32.com stuff... |
| 21:11.41 | MikeJ | I ported most of that stuff to the 1.2 branch in tree, but some of the build stuff was sacraficed in the recent build system reorg in trunk |
| 21:12.41 | MikeJ | the only windows stuff is cygwin stuff.. and it has never been gotten to really work on anything after 1.0 branch... and no one was interested in helping maintain it.. |
| 21:13.17 | dashu | :o |
| 21:13.50 | MikeJ | sorry.. |
| 21:14.15 | dashu | hehe XD not ur fault |
| 21:14.27 | MikeJ | the code is pretty close to being fine for cygwin, the build will have to be re-worked to get the linking right again |
| 21:14.38 | dashu | oh |
| 21:14.51 | MikeJ | but it should be fairly trivial.. the problem i had was I could never properly debug anything in cygwin as gdb never worked well |
| 21:15.35 | MikeJ | the gdb for mingw is supposed to work better, but getting asterisk to compile with mingw would be a decent chunk more work.. |
| 21:16.12 | Serbaniaotic | i'm tweaking around getting colinux to run on my xp machine then using gentoo build to install asterisk |
| 21:16.31 | MikeJ | don't bother.. |
| 21:16.55 | MikeJ | it's a waste of time, you might as well use vmware at that point |
| 21:16.57 | dashu | oh well :p thanks for the info guess the linux server will have to do for a little longer ^^ |
| 21:17.01 | Strom_C | do you people just love pain or something? |
| 21:17.07 | nick125_lappy | lol |
| 21:17.15 | SplasPood | wow.. the Zimbra asterisk zimlet is INSANE |
| 21:17.28 | Serbaniaotic | we are geeks, we're prone to pain and punishment |
| 21:17.52 | nick125_lappy | wtf, my associate just told me that iax.cc just suspended our account because we made a 5 hour phone call...wtf |
| 21:20.07 | CunningPike | dashu: Why don't you want your linux server? License cost too high? :) |
| 21:20.30 | nick125_lappy | lol |
| 21:22.57 | Toadyus | <PROTECTED> |
| 21:23.18 | Strom_C | yes |
| 21:23.25 | Strom_C | dynamic meetme conferencing |
| 21:24.01 | nick125_lappy | I think its a flag you pass to app_meetme, but, I don't remember it off hand |
| 21:24.35 | bkw_ | d |
| 21:24.50 | bkw_ | how can you not remember that? |
| 21:24.54 | bkw_ | :P |
| 21:25.34 | Toadyus | strom_c - is that a addon? |
| 21:25.45 | Strom_C | no |
| 21:25.58 | Strom_C | you pass the d flag to Meetme() |
| 21:26.20 | Toadyus | ok |
| 21:26.26 | Toadyus | never set up meetme before |
| 21:26.35 | Strom_C | it |
| 21:26.40 | Strom_C | it's balls-easy |
| 21:26.53 | Toadyus | balls?? |
| 21:27.35 | *** join/#asterisk keith80403 (n=keith804@24-56-189-80.co.warpdriveonline.com) |
| 21:28.54 | nick125_lappy | unless you are running in a xen VM which doesn't have a RTC device, which causes zaptel to cry and pout |
| 21:29.05 | *** join/#asterisk km- (n=pgrace@aeneas.fierymoon.com) |
| 21:29.39 | km- | hey guys, got a quick question (I hope) |
| 21:29.43 | km- | Sep 6 16:27:53 NOTICE[3084]: chan_iax2.c:5258 authenticate: Asked to authenticate to 10.0.0.5 with an RSA key, but they don't allow RSA authentication |
| 21:30.07 | km- | Is that error produced when the key is wrong too? I'm trying to figure out why I can't originate a RSA call |
| 21:30.23 | nick125_lappy | km-: that usually means the host you are trying to connect do doesn't like RSA |
| 21:30.39 | *** join/#asterisk Defraz (n=t0tal@fw.centrisys.com) |
| 21:30.42 | km- | nick125_lappy: I'm trying to figure out why :) |
| 21:31.13 | nick125_lappy | what kind of host are you trying to connect to? you might want to look in the logs on that box as well |
| 21:33.50 | km- | the remote system doesn't show any connect attempt that I can see |
| 21:34.59 | *** join/#asterisk bkruse (i=bkruse@nat/digium/x-a5bcb2813ea59f7e) |
| 21:35.08 | *** join/#asterisk frenzy (n=frenzy@196.46.104.89) |
| 21:35.48 | frenzy | how do I enabled Call waiting on BT102 I have the option enabled in the admin section and rebooted it however asterisk still says call waiting disabled |
| 21:37.39 | Vorondil | hey y'all, quick question: what do the letters output by 'iax2 jb debug' mean? (i.e. - vvvvvllvvvvlsvvv, etc) |
| 21:37.52 | *** join/#asterisk anthonyl (n=anthonyl@c-71-57-43-221.hsd1.il.comcast.net) |
| 21:37.54 | *** join/#asterisk techie (n=techie@ppp-69-239-205-253.dsl.frs2ca.pacbell.net) |
| 21:37.54 | anthonyl | j #blackfin |
| 21:38.34 | km- | rsa keying is a pain in the butt I guess |
| 21:41.44 | kannan | any recommendations for a DID number is USA that will be supported by my asterisk ox, I need one DID number that has 20 simultanous lines at least |
| 21:41.52 | bkruse | Vorondil: dont take my word, but jitter buffer debug? |
| 21:42.32 | Vorondil | bkruse: :-P indeed. any idea what each letter signifies? |
| 21:45.11 | *** part/#asterisk mountainm2k (n=mountain@216.87.64.218) |
| 21:45.16 | Adrian__ | i am having a problem with ENUM - normal sip calls work. ENUM calls work, but i do not get any audio (the other person can hear me but i cannot hear them) |
| 21:45.24 | Adrian__ | any ideas? |
| 21:46.44 | Ryushin | How many people are using tftp for their polycom phones instead of ftp? |
| 21:47.01 | Strom_C | i set up tftp this weekend |
| 21:47.35 | Ryushin | I'm just wondering if that is why my ip430's are having problems. I'm using vsftpd. |
| 21:47.55 | CunningPike | What is the name of the PBX appliance that Digium is involved in? |
| 21:47.57 | Strom_C | i just used atftpd |
| 21:48.17 | *** join/#asterisk f0urtyfive (i=f0urtyfi@c-67-165-5-232.hsd1.ct.comcast.net) |
| 21:48.27 | blitzrage | Adrian__: thats odd because ENUM should just be a lookup scheme I'm pretty sure... ? |
| 21:48.31 | Ryushin | Yea, I just installed atftpd. |
| 21:48.37 | blitzrage | although I've not done much (really any) work with ENUM |
| 21:48.43 | blitzrage | but I don't think it's a transport |
| 21:48.49 | Strom_C | what is the problem you're having, Ryushin? |
| 21:49.17 | *** part/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 21:49.54 | droops | hey strom, ive almost got it working |
| 21:50.00 | *** join/#asterisk basty (n=basty@mail.sunblast.de) |
| 21:50.01 | basty | Hi |
| 21:50.06 | Strom_C | droops: cool |
| 21:50.10 | Strom_C | whats your solution? |
| 21:50.11 | droops | i just need to use SET() from my agi |
| 21:50.16 | basty | Anyone familar with using Asterisk with Zaptel on Heartbeat? |
| 21:50.18 | droops | im using meetme, thats working fine |
| 21:51.33 | Ryushin | The ip430's go into this constant reboot. I told formatted the file system on three of them, and those can no longer fine the application and go into a reboot loop. I have about 8 others that kept rebooting, and three of them fixed themselves by turning of vsftpd. I'm thinking that somehow vsftpd might be the issue, but the 601 worked fine with it. It's supports case sensitivity, and the phones have been reading and dumping files to |
| 21:51.33 | Ryushin | it just fine. |
| 21:52.32 | basty | I need to find a way on how to stop zaptel in case of a failover. Problem is that when I setup haresources on heartbeat like "... zaptel asterisk ..." Zaptel is getting starting before asterisk....Thats okay...but in case of a failover it first tries to stop zaptel while asterisk is still running. |
| 21:52.55 | basty | well - and that doesnt work.. ;-) |
| 21:53.06 | Ryushin | I'm wondering if I should just use tftp until the ip430's have the new 2.0.1 firmware and 2.3.2 bootroms. |
| 21:53.45 | Ryushin | The phones aren't pulling anything from the tftpd daemon though. Even after pressing 1357, to force them to reset. |
| 21:56.03 | Ryushin | I don't have onsite access to the phones right now. Is there a way to force a polycom phone to pull from tftp instead of ftp? |
| 21:56.20 | CunningPike | Ryushin: It's in the phone itself |
| 21:56.39 | CunningPike | Ryushin: ftp is better - you can poll for updates etc |
| 21:58.03 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-88-155-199-172.red.bezeqint.net) |
| 21:59.26 | *** join/#asterisk RoyK (n=roy@ti211210a080-1761.bb.online.no) |
| 21:59.28 | *** join/#asterisk techie (n=techie@ppp-69-239-205-253.dsl.frs2ca.pacbell.net) |
| 22:00.44 | *** join/#asterisk dieno2 (n=dienno2@58.65.193.77) |
| 22:00.59 | dieno2 | can any one tell me how to listen live Calls on Asterisk |
| 22:01.44 | Ryushin | CunningPike: That's why I wanted to use FTP. But for right now, I just want to get these ip430's to work. |
| 22:02.27 | dieno2 | can any one tell me how to listen live Calls on Asterisk\ |
| 22:02.32 | dieno2 | please |
| 22:02.39 | Ryushin | It's just to remove vsftpd as the culprit. |
| 22:04.36 | CunningPike | Ryushin: Hmmm - we use vsftp without issues |
| 22:05.24 | *** join/#asterisk saftsack (n=saftsack@p54A7DC29.dip.t-dialin.net) |
| 22:08.12 | Ryushin | Yea, the 601's don't have any issues, but I'm sure having issues with the ip401's. |
| 22:10.44 | *** join/#asterisk backblue (n=moo@87-196-11-214.net.novis.pt) |
| 22:12.10 | *** join/#asterisk budairc (n=chatzill@200.215.57.174) |
| 22:12.44 | *** join/#asterisk _DAW (n=_DAW@adsl-222-12-239.msy.bellsouth.net) |
| 22:12.52 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 22:14.27 | _DAW | hello mates |
| 22:15.13 | CunningPike | Ryushin: I'm wondering if the firmware on the 430s is newer than what is on your vsftp server |
| 22:22.14 | *** part/#asterisk Vorondil (n=vorondil@64.191.168.244) |
| 22:22.49 | *** join/#asterisk dieno (n=Dieno@58.65.193.77) |
| 22:23.01 | dieno | can any 1 tell me how to listen LIVE CAlls |
| 22:23.07 | dieno | plz |
| 22:23.25 | C6Vette | dieno, look up ChanSpy |
| 22:23.48 | *** join/#asterisk Ebola (i=1000@81-86-155-65.dsl.pipex.com) |
| 22:23.50 | C6Vette | and/or ZapScan |
| 22:24.07 | droops | anyone ever set a variable with a call file? i cant find an example of the syntax |
| 22:24.08 | *** part/#asterisk Serbaniaotic (n=mikep@206.124.12.162) |
| 22:24.23 | droops | Set:prop_id=4 isnt right |
| 22:24.25 | C6Vette | droops, yes. hold on |
| 22:24.43 | dieno | ok thnx |
| 22:24.56 | C6Vette | SetVar: var=607554 |
| 22:25.00 | Ryushin | CunningPike: Well, I installed 3.2.2 and 2.0.1 yesterday. They don't seem to be picking it up. |
| 22:25.10 | droops | thanks C6Vette |
| 22:25.24 | Ryushin | I think I'm going to have to go onsite and hard code the problem phones to use tftp. See if that fixes it. |
| 22:25.55 | CunningPike | Ryushin: I'd be interested to hear how it goes.... |
| 22:26.23 | dieno | and now how to install Zapscan or Chanspy |
| 22:27.30 | *** join/#asterisk teknoprep (n=teknopre@unaffiliated/teknoprep) |
| 22:27.37 | teknoprep | i am looking for an ata that supports asterisk and GSM |
| 22:27.40 | teknoprep | anyone know of one? |
| 22:27.49 | CunningPike | dieno: Do you have a specific question about the procedure, having read all the available materials? |
| 22:29.04 | Ryushin | I'll let you know. If it fixes the problem, I'll add it to the ip430 info on the the voip-info wiki. |
| 22:29.35 | CunningPike | Ryushin: Sure :) |
| 22:29.41 | budairc | what is the best codec?!? low bandwidth and good quality |
| 22:29.52 | CunningPike | G.729 |
| 22:29.59 | budairc | and is free? |
| 22:30.03 | teknoprep | gsm is great |
| 22:30.10 | teknoprep | aprox 33kbit/sec |
| 22:30.24 | CunningPike | budairc: Ah. You didn't specify ;) |
| 22:30.26 | droops | C6Vette, that worked like a charm, thanks |
| 22:30.44 | C6Vette | droops, glad I could help |
| 22:31.06 | Cresl1n | g.729 sucks |
| 22:31.09 | Cresl1n | it sounds awful |
| 22:31.14 | *** part/#asterisk basty (n=basty@mail.sunblast.de) |
| 22:31.16 | budairc | CunningPike: and ata supports g.729? |
| 22:31.41 | CunningPike | Sipura does - at least the SPA-3000 |
| 22:32.42 | budairc | hmm.. G.729 is good or is sucks? ehhe |
| 22:32.50 | budairc | Cresl1n: why? |
| 22:32.59 | teknoprep | sooo |
| 22:33.03 | teknoprep | gsm codec with ata? |
| 22:33.05 | *** join/#asterisk trevarthan (n=trevarth@c-71-226-190-251.hsd1.ga.comcast.net) |
| 22:33.06 | teknoprep | anyone know of one? |
| 22:33.12 | Cresl1n | budairc: is just does |
| 22:33.31 | *** join/#asterisk rollergrrl (n=0x3e44d@71-213-6-123.slkc.qwest.net) |
| 22:33.39 | Cresl1n | listen to ulaw on a good speaker, and listen to g.729 and there's a world of difference |
| 22:33.40 | budairc | Cresl1n: for u.. what is best? |
| 22:33.53 | trevarthan | hello. Can someone here recommend GUIs for office use and call center use? |
| 22:34.06 | Cresl1n | budairc: g.729 works for most people. it just doesn't sound that great |
| 22:34.12 | Adrian__ | SIP calls over my sip provider work, but SIP direct IP or ENUM calls do not work (no incoming audio) |
| 22:34.15 | Cresl1n | well, it sounds good enough for most people though |
| 22:34.44 | trevarthan | I prefer open source, but if there is a commercial GUI that is clearly superior for a particular purpose then I'd like to know about it. |
| 22:34.46 | teknoprep | is it possible to have asterisk behind a nat... and a sip phone behind a nat.. and have them talk properly... even after forwarding ports it did not work properly... do i have to edit any conf files forthis? |
| 22:34.58 | budairc | Cresl1n: but.. low bandwidt |
| 22:35.06 | teknoprep | budairc use GSM |
| 22:35.08 | Cresl1n | yeah, that's the problem :-) |
| 22:35.09 | teknoprep | GSM OWNS YOU |
| 22:35.17 | *** join/#asterisk Strom_M (n=pocketir@m610e36d0.tmodns.net) |
| 22:35.22 | budairc | teknoprep: :D |
| 22:35.22 | teknoprep | budairc even my phones on the lan use GSM .. with a GSM trunk |
| 22:35.34 | budairc | i'm using GSM |
| 22:35.34 | Strom_M | yech |
| 22:35.35 | *** join/#asterisk vosque (i=e40djoee@69.50.222.162) |
| 22:35.44 | trevarthan | teknoprep: I made asterisk talk to a SIP phone while both were behind a firewall. You have to publish the ports on the asterisk server and use a STUN server though. |
| 22:36.02 | teknoprep | publish the ports on the asterisk server? |
| 22:36.06 | teknoprep | wtf does publish ports mean? |
| 22:36.10 | vosque | Is there anyway to make the ZAP channel not pick up right away? I've still got a POTS phone that I would like to have an opportunity to answer. |
| 22:36.37 | Strom_M | vosque: buy a terminal adapter |
| 22:36.38 | rollergrrl | How would one combine gsm files, on the fly, so you can use them in one Read? |
| 22:36.44 | trevarthan | teknoprep: publish the asterisk ports on your firewall. "punch a hole in the firewall" might be a better term. |
| 22:36.55 | teknoprep | that would be port forwarding or dnat |
| 22:36.59 | vosque | Strom_M: eventually, yes, but is there anything I can do in the meantime? |
| 22:36.59 | teknoprep | destination nat |
| 22:37.01 | trevarthan | vosque: yes. use wait(), I think. |
| 22:37.18 | teknoprep | punching a hole into the firewall is still even a bad term but ty for the help |
| 22:37.30 | teknoprep | now the stun ... do i have to do that on asterisk or do i do that on the phone? |
| 22:37.42 | trevarthan | teknoprep: port forwarding. right. you're seriously arguing with me over terminology? |
| 22:38.00 | trevarthan | teknoprep: STUN is for the phone. |
| 22:38.15 | teknoprep | trevarthan no... i am arguing for the fact that if you at least said somethign with similar viability towards port forwarding i wouldn't have said anything |
| 22:38.15 | budairc | teknoprep: do u know how i change the ring type.. differents for internal extension.. and external extension? |
| 22:38.19 | trevarthan | teknoprep: the asterisk server needs a static IP on the firewall's public interface. |
| 22:38.21 | teknoprep | but i had no idea what you were tlaking about |
| 22:38.40 | budairc | teknoprep: sorry for my english.. i know that sucks.. |
| 22:38.41 | teknoprep | budairc its settings on the phone |
| 22:38.42 | Adrian__ | any ideas about how to fix one way audio with direct IP/ENUM calls? |
| 22:38.58 | teknoprep | budairc you can also doit with asterisk but i don't know |
| 22:39.09 | Strom_M | adrian: are the calls sip? |
| 22:39.31 | Adrian__ | Storm_M - yes |
| 22:39.31 | teknoprep | personally i would never put a voip server behind a NAT firewall.. but i was just asking for gerneral knowledge.. as i have tryed to doit and it won't work |
| 22:39.33 | trevarthan | again: can anyone recommend GUIs for asterisk in a call center and/or office environment? |
| 22:39.35 | budairc | teknoprep: hmm.. its settings for softphone too?! (x-lite) |
| 22:39.55 | Strom_M | adrian: behind a nat? |
| 22:40.18 | Adrian__ | Storm_M - SIP calls work as long as they run over my SIP provider - but when i try to call a SIP adress directly i do not get audio (the other person can hear me) |
| 22:40.25 | Adrian__ | Storm_M - ya behind NAT |
| 22:40.36 | trevarthan | teknoprep: I assure you that it does work. I can take a preconfigured hardware VoIP phone and plug it into anyone's cable/dsl and get dialtone to my asterisk server. |
| 22:40.36 | Strom_M | adrian: the problem is the nat |
| 22:40.54 | Strom_M | adrian: put the asterisk server in front of the nat |
| 22:41.11 | Strom_M | and stop calling me storm |
| 22:41.30 | teknoprep | trevarthan is your Asterisk box behind a NAT ? |
| 22:41.34 | budairc | Adrian__: i'm solve my problem using in front of of the nat |
| 22:41.36 | Adrian__ | Strom_M - oops sorry... |
| 22:42.20 | teknoprep | trevarthan did you have to edit any conf files to add the ip to? |
| 22:42.38 | Adrian__ | budairc/Strom_M - i am behind a ADSL router - and i only get one IP - so i need NAT don't I? |
| 22:42.43 | trevarthan | teknoprep: "behind" is a loose term. Yes, it's physically connected to a NAT. No, it doesn't use NAT to get to the outside world because the ports are forwarded. |
| 22:42.45 | teknoprep | trevarthan i would think you would need to fake the external ip of the trixbox to SIP phones |
| 22:43.04 | trevarthan | teknoprep: just a moment. |
| 22:43.11 | teknoprep | trevarthan yes actually you still use NAT if you take Ext.IP and forward it to an Intern.IP |
| 22:43.14 | teknoprep | that is still nat |
| 22:43.14 | teknoprep | now |
| 22:43.20 | Adrian__ | i forwarded the pords, set up the firewall rules, etc. it sould work :( |
| 22:43.36 | teknoprep | if you just Forward through a Firewall... then no its just a port forwarding Packet Filter that allows said ports to IP |
| 22:43.44 | teknoprep | like an ACL in a Router/Pix |
| 22:43.48 | wunderkin | docelmo = nubb |
| 22:43.52 | Strom_M | adrian: double nat does not work well with sip |
| 22:43.54 | budairc | Adrian__: hmm.. use DMZ for the server.. all ports send to the server |
| 22:43.54 | wunderkin | :) |
| 22:44.05 | trevarthan | teknoprep: do you want me to help you, or do you want to keep correcting me? |
| 22:44.37 | teknoprep | trevarthan i was just informing you.. i would never take the standpoint to correct you in how your setup is... as its working.. so no i am not doing either in your question |
| 22:44.45 | teknoprep | trevarthan yes i do want help |
| 22:44.49 | Adrian__ | Strom_M - double NAT? |
| 22:44.54 | Adrian__ | i only have one... |
| 22:45.07 | Strom_M | nat on the other end of the call |
| 22:45.14 | Adrian__ | ah |
| 22:45.21 | budairc | in and out |
| 22:45.23 | trevarthan | teknoprep: just a moment. I have to find the config. |
| 22:45.29 | teknoprep | nat <-> Inet <-> Nat |
| 22:45.39 | teknoprep | phone - nat - inet - nat - * |
| 22:46.06 | budairc | teknoprep: thnx for help me.. ;) |
| 22:46.14 | Adrian__ | but that doesnt make sense - i called a guy who also has a nat - he can hear me but i cant hear him :/ |
| 22:46.55 | teknoprep | budairc np |
| 22:46.55 | Strom_M | adrian: please go read up on sip |
| 22:46.56 | budairc | Adrian__: yeahh.. is a nat problem.. with rtp ports |
| 22:47.17 | *** join/#asterisk Beighto (n=chatzill@64.160.113.130) |
| 22:47.26 | trevarthan | teknoprep: in sip.conf you need to set externip and localnet. |
| 22:48.57 | trevarthan | teknoprep: I believe I also have the 'nat' option commented out. I think that defaults to nat=no. |
| 22:49.29 | *** join/#asterisk ivanfm (n=ivanfm@c93481ec.virtua.com.br) |
| 22:49.32 | teknoprep | ok |
| 22:49.46 | trevarthan | teknoprep: I tested this setup with two Grandstream hardware phones (the ATA and the one that came out before the GXP. I can't remember the model.) |
| 22:50.01 | trevarthan | teknoprep: STUN is absolutely necessary. |
| 22:50.06 | teknoprep | ok |
| 22:50.09 | teknoprep | its not for me |
| 22:50.13 | teknoprep | its for someone in #freepbx |
| 22:50.20 | teknoprep | i don't hook up voip servers behind nat |
| 22:51.11 | trevarthan | teknoprep: well, as long as you have a static IP for the server it works pretty darn well. You might try it sometime. |
| 22:51.16 | teknoprep | trevarthan is localnet=192.168.1.0/24 look right? |
| 22:51.24 | *** join/#asterisk zotz (n=zotz@24.244.163.225) |
| 22:51.33 | teknoprep | also where do you comment out your nat=no at? |
| 22:52.12 | trevarthan | teknoprep: I'm not sure if it accepts the short subnet form. I used /255.255.255.0 on mine. |
| 22:52.17 | teknoprep | ok |
| 22:52.27 | trevarthan | teknoprep: nat=no is in sip.conf also. |
| 22:53.55 | trevarthan | now.... would anyone on the list be able to recommend a GUI for office or call center use? Anything at all that works nicely? I've got non linux people who need to admin an asterisk box and I need a workable solution... |
| 22:54.20 | teknoprep | trevarthan hold up trying to get jhs here.. the guy that needs help |
| 22:55.00 | trevarthan | teknoprep: :) *I* need help. You just happen to be asking something that I have an answer to. :) |
| 22:55.09 | teknoprep | lol |
| 22:55.12 | teknoprep | what help you need? |
| 22:55.21 | teknoprep | phsycological? |
| 22:55.41 | trevarthan | Yeah, sorta. I need a GUI recommendation for a linux server application called asterisk. :) |
| 22:55.42 | budairc | why nat as comment by default.!? if the option don't change for users in front nat... |
| 22:55.58 | Beighto | Two issues for someone to take a whack at: When someone dials in my dialplan sometimes their dtmf is not registering correctly to the server, for example: 123 will show up as 1233. Second issue: When people are conferencing in meetme their words will be sparatically cut off for a split second about every 30 seconds or so. |
| 22:56.09 | Beighto | trevarthan: why don't you just use Trixbox? |
| 22:56.45 | trevarthan | beighto: will that work well for an office with up to 50 people and a PRI line? |
| 22:56.49 | teknoprep | phsycological?? |
| 22:56.51 | teknoprep | trevarthan how do you comment it out.. with the normal # ? |
| 22:56.57 | *** join/#asterisk RoyK (n=roy@ti211210a080-1761.bb.online.no) |
| 22:56.57 | Beighto | trevarthan absolutely |
| 22:57.06 | trevarthan | teknoprep: semicolon. |
| 22:57.25 | trevarthan | beighto: how hard are security upgrades? |
| 22:58.10 | Beighto | trevarthan: don't know, never got that far with the thing. I hear ever since it went from @home to trixbox it got very easy |
| 22:58.11 | trevarthan | bieghto: also, does the webgui allow full configuration of extensions, voicemail, etc... ? Or is it only a subset? |
| 22:58.24 | teknoprep | trevarthan i don't use a tab completion that adds a : |
| 22:58.38 | teknoprep | trevarthan i am on mirc |
| 22:58.54 | teknoprep | trevarthan i have alot of windows only applications for admin'ing a very large ammount of remote offices |
| 22:59.00 | Beighto | trevarthan: I think it is full configuration, if not you can change things the old fashioned way if you can find the correct conf file |
| 22:59.29 | *** join/#asterisk dalekurt (n=DaleKurt@65.183.3.229) |
| 22:59.33 | trevarthan | teknoprep: I don't understand. You asked how to comment out a line. Use don't use a hash (#), you use a semicolon (;) |
| 22:59.51 | teknoprep | oh |
| 23:00.00 | trevarthan | beighto: ok. that's the sort of info I'm looking for. I'll check it out. Anyone else have any recommendations? |
| 23:00.36 | *** join/#asterisk knarfly (n=bdavis@c-65-34-177-3.hsd1.fl.comcast.net) |
| 23:05.23 | Adrian__ | could somebody test call my ENUM? |
| 23:06.03 | teknoprep | what is is? |
| 23:06.05 | teknoprep | it? |
| 23:06.07 | trevarthan | I'm gonna head over to freepbx and ask some trixbox questions. If you need me teknoprep, I'll be there for a few minutes. |
| 23:06.15 | *** part/#asterisk trevarthan (n=trevarth@c-71-226-190-251.hsd1.ga.comcast.net) |
| 23:06.43 | teknoprep | Adrian__ what is the number? |
| 23:08.07 | justinu|laptop | someone should tell trevarthan about joining more than one channel at a time |
| 23:08.22 | tessier__ | Well fuck me gently with a chainsaw |
| 23:08.31 | tessier__ | Turns out our PRI provider won't let us set our own callerid |
| 23:08.43 | teknoprep | hmmm |
| 23:08.43 | tessier__ | Only for long distance calls |
| 23:08.54 | tessier__ | That's why my system has been jacked for the past few days |
| 23:09.00 | teknoprep | i have CID set to a number they own but not on our asterisk box |
| 23:10.56 | *** join/#asterisk jeebusmobile (n=jeebusmo@100.sub-75-215-178.myvzw.com) |
| 23:12.02 | knarfly | tessier__: Have a look at http://myvoice.splitinfinity.com |
| 23:12.46 | tessier__ | knarfly: splitinfinity? I have one of your coffee mugs in my kitchen... |
| 23:12.50 | tessier__ | knarfly: You based in San Diego? |
| 23:13.34 | knarfly | tessier__: No, Miami...but my simple setup may not be as complex as yours....however, I can set my CID to whatever I want |
| 23:14.17 | knarfly | tessier__: Makes it easy for certain tasks in my life |
| 23:17.08 | tessier__ | knarfly: I have worked with a few T-1 and DS-3 PRI's before and they always let me set my CID. I have heard of ones that don't but this is the first time I have run into it. |
| 23:18.12 | knarfly | tessier__: Spoofing my CID was what got me into * in the 1st place. |
| 23:18.26 | CunningPike | Can anyone remember the name of the turnkey Asterisk box that Digium was working on? |
| 23:19.09 | wunderkin | CunningPike, are you thinking of poundkey? i think that is just a linux distro |
| 23:19.23 | CunningPike | Aha - that's it! Thanks :D |
| 23:21.07 | *** join/#asterisk RoyKa (n=roy@gprs-ggsn6-nat.mobil.telenor.no) |
| 23:28.21 | *** join/#asterisk rrivas (n=rrivas@200.68.91.21) |
| 23:28.53 | *** part/#asterisk rrivas (n=rrivas@200.68.91.21) |
| 23:31.20 | tessier__ | They changed it. They used to allow us to set our own callerid, we tested everything, signed off on it, then things broke and we find out they no longer allow us to set our own CID. |
| 23:31.49 | rollergrrl | How would one combine gsm files, on the fly, so you can use them in one Read? |
| 23:37.45 | Strom_C | tessier__: who is your PRI provider? |
| 23:38.51 | tessier__ | Cox Communications |
| 23:39.11 | Strom_C | well, duh |
| 23:39.20 | Strom_C | you're getting your phone service from a TV company |
| 23:41.30 | Strom_C | I've looked at the way they do things, and it horrifies me |
| 23:42.40 | justinu|laptop | true dat |
| 23:42.40 | justinu|laptop | TV/cable companies don't have experience providing reliable service, and it shows |
| 23:42.54 | Strom_C | Like I told one of my clients once: who would you rather get your phone service from - a company that has, in one form or another, been providing telephone service for the last 130 years and really knows what they're doing, or a cable company that went "LOL Phones!" a few years ago? |
| 23:42.55 | *** join/#asterisk freeepbxxnoobbb (n=chatzill@rrcs-67-52-187-18.west.biz.rr.com) |
| 23:43.08 | *** join/#asterisk plainvoip (n=root@gw01.plainvoip.com) |
| 23:43.27 | linagee | Strom_C: cable companies can offer single hop service though. :-D |
| 23:43.50 | Strom_C | and that means balls if the plant is garbage |
| 23:44.35 | linagee | Strom_C: all companies should offer their customers SLAs. you should get lots of money back from cox if they have downtime on their phones. |
| 23:44.39 | *** join/#asterisk Frogdude (i=Frogdude@c-24-16-72-159.hsd1.wa.comcast.net) |
| 23:44.56 | *** join/#asterisk dprevite (n=dprevite@c-67-162-110-89.hsd1.il.comcast.net) |
| 23:45.07 | Strom_C | well sure, they can pay you back but the idea is to avoid the stupidity in the first place |
| 23:46.17 | linagee | Strom_C: who's been providing phone service for the past 130 years? ma bell? |
| 23:46.47 | Strom_C | yes |
| 23:47.16 | freeepbxxnoobbb | can someone help me out with my phones are not able to communicate with each other sip phones cant dial each others extensions |
| 23:47.45 | CunningPike | freeepbxxnoobbb: Check the topic...... ;) |
| 23:47.48 | Strom_C | freeepbxxnoobbb: are you running real asterisk now, or are you still running freepbx? |
| 23:48.11 | freeepbxxnoobbb | freepbx ....But no one helps me in there |
| 23:48.26 | CunningPike | freeepbxxnoobbb: Good luck finding help here too :) |
| 23:48.28 | Strom_C | well, it's time to buckle down and learn how things are REALLY done |
| 23:48.42 | Strom_C | get rid of freepbx and start fresh with asterisk and nothing else :) |
| 23:49.03 | CunningPike | That's how we did it when I were a lad - uphill, both ways |
| 23:49.06 | freeepbxxnoobbb | I will on the next system i build |
| 23:49.33 | freeepbxxnoobbb | but this one i have to fix and im gonna get fired |
| 23:49.42 | Strom_C | wait |
| 23:49.44 | linagee | you're going to fix it and get fired? |
| 23:49.58 | Strom_C | your first foray into asterisk is a PRODUCTION system? |
| 23:50.14 | freeepbxxnoobbb | yea bit off more then i can chew |
| 23:50.21 | Strom_C | sigh |
| 23:50.25 | Strom_C | ~hafc |
| 23:50.32 | jbot | well, hafc is hire a freaking consultant. Look, if you're having difficulty understanding what you're doing and need a solution soon, you will be far better off hiring a competent consultant than continuing to pull your hair out. |
| 23:50.32 | *** join/#asterisk bjohnson (n=bjohnson@i216-58-9-214.cybersurf.com) |
| 23:50.39 | freeepbxxnoobbb | my first was actually astguiclient on slackware |
| 23:50.39 | CunningPike | ~glwat |
| 23:50.43 | linagee | freeepbxxnoobbb: find someone to contract it to and present it to your boss about how great it is, or find a new job. :-D |
| 23:50.56 | *** join/#asterisk viler (i=1000@200.114.70.228) |
| 23:52.02 | CunningPike | jbot, wglwat is well, good luck with all that |
| 23:52.03 | jbot | CunningPike: okay |
| 23:52.16 | CunningPike | ~wglwat |
| 23:52.18 | jbot | methinks wglwat is well, good luck with all that |
| 23:52.37 | nick125_lappy | yeah, I agree with ~hafc |
| 23:52.55 | nick125_lappy | because, you'll end up hiring someone *anyways*, _AND_ you'll have to buy a wig ;) |
| 23:53.01 | wunderkin | heh, freepbxnubbbbbbbbbb |
| 23:53.13 | Strom_C | hehehh |
| 23:53.43 | wunderkin | my favorite word on irc is nubb :D |
| 23:53.56 | linagee | freeepbxxnoobbb: tell your boss that you can have asterisk up and running, but it will take a few years. :-D |
| 23:54.15 | freeepbxxnoobbb | never mind i got it |
| 23:54.28 | freeepbxxnoobbb | i just had to unregister the phones |
| 23:54.34 | CunningPike | Mine is asshat |
| 23:55.47 | linagee | freeepbxxnoobbb: don't forget to tell your boss how much the current system sucks. :-D |
| 23:56.01 | freeepbxxnoobbb | i know |
| 23:56.20 | nick125_lappy | lol |
| 23:56.29 | freeepbxxnoobbb | ill probably start from scratch next time since i know what to do |
| 23:56.44 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 23:57.08 | *** join/#asterisk `Kevin (n=Kevin@64.243.236.20) |
