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| 00:32.41 | aliverius | evening |
| 00:32.54 | aliverius | anyone else experiencing problems with misdn v2? |
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| 00:42.27 | *** join/#asterisk mltlnx (n=mltlnx@cpe-24-193-11-14.nyc.res.rr.com) |
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| 00:43.46 | ldsjohn | handle_request_invite: Call from '' to extension 's' rejected because extension not found. |
| 00:43.46 | ldsjohn | any clue what that means? |
| 00:44.41 | ldsjohn | I think that for some reason my incoming call isn't being matched to the right peer entry. but google isn't helping much with that error, this is in asterisk1.4 with asteriskgui 2.0 |
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| 00:46.33 | *** part/#asterisk C4colo (n=DJpyro@66.185.111.33) |
| 00:47.13 | [TK]D-Fender | ldsjohn: its a dialplan error because the extension they want doesn't exist in the context its searched for in. |
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| 00:47.46 | ldsjohn | so this is a problem with my config, and prolly an asteriskgui issue? |
| 00:48.09 | [TK]D-Fender | ldsjohn: GUI or your use of it |
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| 00:48.40 | [TK]D-Fender | ldsjohn: You'd ahve to look at SIP DEBUG to see if it matched the peer you were expecting, and to see what context the calls point to. |
| 00:49.17 | ldsjohn | okay, should i aks this in the other channel? |
| 00:49.30 | [TK]D-Fender | ldsjohn: I thikn you should go LOOK like I just told you\ |
| 00:49.42 | ldsjohn | k |
| 00:49.46 | ldsjohn | so turn on sip debug |
| 00:49.55 | ldsjohn | and then try the call again, and see what I find? |
| 00:50.31 | HaMF | hm had an similar problem here. curiously with only one phone (out of 170). (analog phone via zaptel -> voip). I was never able to track the problem down... :| |
| 00:51.37 | ldsjohn | well hopefully I can figure this out then, sip debug gave me a ton of info. |
| 00:52.10 | ldsjohn | this isn't on my phone its for inbound calls from a sip trunk |
| 00:52.24 | [TK]D-Fender | ldsjohn: Yes, I'm well aware |
| 00:54.09 | ldsjohn | okay I am seeing in the debug, the call is coming from the right ip, and it has caller id info, but I dont see that its getting passed that, unless I am missing something. do you want me to pastebin the output? |
| 00:55.17 | [TK]D-Fender | look at the peer it matches |
| 00:55.34 | [TK]D-Fender | and the "looking for " right before you get 404'd |
| 00:55.40 | [TK]D-Fender | that will tell you the context |
| 00:56.06 | [TK]D-Fender | and its looking for "s" because you did not specify a numbered extensions for them to send their call to when you get called. |
| 00:56.13 | [TK]D-Fender | this is in your REGISTER statment. |
| 00:56.26 | ldsjohn | oh, where do I put register statements in the gui? |
| 00:56.36 | ldsjohn | I was under the impression that it was building one since there isn't a spot to put it |
| 00:57.16 | [TK]D-Fender | ldsjohn: Its probably doing most of the job for you and I don't sue it so I couldn't tell you where, if anywhere |
| 00:57.33 | ldsjohn | I have never used the new gui till today, I am trying to replace a freepbx install that has worked for 3 years till the harddrive died last week. so I coudl be doing this wrong |
| 00:58.04 | [TK]D-Fender | *GUI is very much a "work in progress". It isn't near a "rollout point" that I could recommend. |
| 00:58.51 | ldsjohn | well that isn't apparant from the documentation |
| 00:59.16 | ldsjohn | maybe I should scrap it and try to get it working with the configs |
| 01:00.52 | [TK]D-Fender | ldsjohn: Actually it should be apparent from the LACK of documentation |
| 01:01.51 | ldsjohn | okay I think I see the problem |
| 01:02.00 | ldsjohn | my peer is listed as callcentric.com |
| 01:02.03 | ldsjohn | but when it comes back |
| 01:02.08 | *** join/#asterisk shmaltz (n=chatzill@mail.dmaven.com) |
| 01:02.15 | ldsjohn | it says no matching peer for 'ipaddress:580' |
| 01:02.51 | ldsjohn | so I need it to know that callcentric.com and the ip for callcentric.com are the same thing |
| 01:02.58 | shmaltz | I custom context/extensions/macros to extensions_custom.conf in freepbx, now how do I send an inbound call to that from freepbx? |
| 01:03.50 | shmaltz | I added custom context/extensions/macros to extensions_custom.conf in freepbx, now how do I send an inbound call to that from freepbx? |
| 01:04.00 | [TK]D-Fender | shmaltz: its in your inbound routes, and something you should be asking in their channel |
| 01:04.22 | shmaltz | [TK]D, just noticed and switched to theirs :P |
| 01:04.32 | shmaltz | I hate does nice tools, as I always use vanila |
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| 01:24.45 | aliverius | i think misdn2 doesnt provide libisdnnet.so |
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| 01:25.45 | aliverius | so effectively chan_misdn is broken with recent kernels? |
| 01:26.49 | ldsjohn | I think that every time I get an incoming call it comes from a different ip, that doesn't seam like normal behavior to me |
| 01:31.09 | [TK]D-Fender | ldsjohn: You rpovider has multiple signalling gateways. perfectly normal. |
| 01:31.28 | ldsjohn | so I need to figure out how to tie them all to the peer |
| 01:33.44 | [TK]D-Fender | ldsjohn: So for I don't see you proving that it isn't matching |
| 01:33.48 | [TK]D-Fender | ~pb |
| 01:34.12 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste , or , http://bin.cakephp.org/ |
| 01:34.12 | ldsjohn | okay |
| 01:34.12 | ldsjohn | Using INVITE request as basis request - 18709901-3453413396-411384@msw1.telengy.net |
| 01:34.15 | ldsjohn | that is I think the relevent line, that is in the sip packet just before the 404 |
| 01:34.41 | ldsjohn | but trunk_1/17772484176 204.11.192.22 5080 Unmonitored |
| 01:34.43 | ldsjohn | is sip show peers |
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| 01:35.31 | ldsjohn | I asked if you wanted a pastebin :-D, one sec and ill have one |
| 01:36.50 | ldsjohn | http://pastebin.ca/1451364 |
| 01:36.56 | ldsjohn | thats the 404 and the one before it |
| 01:37.43 | ldsjohn | http://pastebin.ca/1451365 is my sip show peers and sip show registry output |
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| 01:38.33 | ldsjohn | the second line of the sib debug changes every couple calls |
| 01:38.38 | ldsjohn | uses different ip's |
| 01:38.41 | riddlebox | has anyone messed with the magicjack? I am thinking of using it for my outbound services if it is worth it |
| 01:38.52 | ldsjohn | my neihbor uses it |
| 01:39.00 | ldsjohn | I wouldn't recomend it just from their experience |
| 01:39.15 | riddlebox | really |
| 01:39.20 | ldsjohn | they have a lot of dropped calls, and lots of inbound calls that dont ever get to them |
| 01:39.29 | ldsjohn | they sometimes can talk for an hour |
| 01:39.34 | [TK]D-Fender | Found no matching peer or user for '204.11.192.31:5080' <--------- |
| 01:39.34 | ldsjohn | and sometimes 5 minutes |
| 01:39.53 | riddlebox | ldsjohn, that sucks |
| 01:40.20 | ldsjohn | but the peer is at 204.11.292.22 so it dont match |
| 01:41.23 | [TK]D-Fender | ldsjohn: but I don't see your CONFIGS.... |
| 01:41.26 | ldsjohn | riddlebox: your mileage may vary, i have personally experienced magic jack as having awesome call quality when it works. I just know that they arn't happy with it |
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| 01:42.04 | [TK]D-Fender | Use of MJ is in violation of their TOS and should get you booted |
| 01:42.32 | riddlebox | [TK]D-Fender, unless you have it setup on a windows machine, and a line over to an ata.... |
| 01:42.35 | ldsjohn | wich ones do you want Fender? |
| 01:43.49 | ldsjohn | I think sip.conf and users.conf |
| 01:47.48 | ldsjohn | http://pastebin.ca/1451378 users.conf |
| 01:48.03 | ldsjohn | http://pastebin.ca/1451379 sip.conf |
| 01:48.17 | ldsjohn | the sip.conf is full of comments because it was created from the make samples |
| 01:49.53 | riddlebox | ldsjohn, I think I may try it, just to mess with it, I havent played with any *asterisk* type of things for a while and I am bored |
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| 01:58.35 | ldsjohn | would the allowexternaldomains fix this? it seams I am getting an invite from a non recognized domain? and thats what that seams to allow |
| 01:58.36 | Deadpepsi | hello all |
| 01:59.47 | ldsjohn | nevermind the default is yes... |
| 02:03.15 | Deadpepsi | i came cause i want to build a pbx wit a few features 1 i want to use sip wifi phones 2 i want them to be able to access the pbx from the internet ie (hot spots, friends and families homes with wifi and so forth) i am pretty sure this can be done also want to use voip sip lines for the in and out bound phone lines |
| 02:04.30 | Deadpepsi | i also am wondering if this will run on ubuntu |
| 02:07.10 | [TK]D-Fender | Deadpepsi: * doesn't care about your OS, and it doesn't care how packets get to it where you intend to use VoIP, just as long as they do. |
| 02:08.01 | Deadpepsi | so this is not a pipe dream yeah! |
| 02:09.16 | Deadpepsi | now i guess i need to start by setting it up in a vm and begin testing! |
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| 02:20.50 | leifmadsen | dances |
| 02:27.16 | ldsjohn | Fender do you use any gui at all. or do you just handcraft your configs? |
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| 02:34.39 | [TK]D-Fender | ldsjohn: Anything worth doing is worth doing yourself |
| 02:35.44 | ldsjohn | yeah, I think thats what im goign to attempt next |
| 02:36.18 | ldsjohn | should I stick with 1.4 or give 1.6 a shot? this is a home phone system will have 3 sip phones and 1 sip trunk |
| 02:41.15 | ldsjohn | the other question is should I start with the samples, or should I pare things down to the bare min. im not going to use half the modules |
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| 02:46.05 | [TK]D-Fender | ldsjohn: Make the samples, then trash your sip.conf & extensions.conf and start from scratch. |
| 02:46.14 | [TK]D-Fender | ldsjohn: Plenty of guides to show you the minimum requirements |
| 02:48.32 | ldsjohn | they used users.conf |
| 02:48.46 | ldsjohn | i have allways in the past seen what is in users.conf in sip.conf |
| 02:48.52 | ldsjohn | is users.conf the new way? |
| 02:50.34 | [TK]D-Fender | ldsjohn: thats the *GUI way |
| 02:50.36 | [TK]D-Fender | ~users.conf |
| 02:50.41 | infobot | users.conf is a flaming pile of sh1t that takes the fine control of several perfectly usable * config files and reduces them to the lowest common denominator and makes your system behave like a "toaster grade" PBX system. |
| 02:50.52 | ldsjohn | thats what I thought |
| 02:51.27 | RyanRR | "sh1t" |
| 02:51.30 | RyanRR | some pretty harsh language here |
| 02:51.34 | RyanRR | lets just cut that the fuck out right now |
| 02:51.34 | ldsjohn | well thanks for your help I reallly appreciate it. ive been struggling with the gui all day. thinking it was callcentrics or my fault |
| 02:51.48 | ldsjohn | yeah what the hell is up with that sh1t |
| 02:52.58 | [TK]D-Fender | "Shit looks pretty good... when compared to CRAP" |
| 02:53.03 | ldsjohn | I think im going to go watch a movie and go to bed, and start this from scratch in the morning |
| 02:53.18 | ldsjohn | yeah I agree. im tired of people telling me not to use shit or damn or hell |
| 02:53.47 | ldsjohn | im a temple going mormon even. and I love when people argue that the bible says not to cuss |
| 02:54.54 | ldsjohn | the only reason for the multiple words for each and every thing is so that high class nobles could look down on the ignorant lower class, and the sooner the world realizes it and stops giving a damn. the sooner people can just get on with their lives and stop telling others not to cuss |
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| 02:57.17 | [TK]D-Fender | ldsjohn: Cussing just gives them more ways to express their already solid disdain for idiots :) |
| 02:58.13 | ldsjohn | yeah allthough there are some phrases that make me wince |
| 02:58.23 | ldsjohn | like every time I hear someone say "Jesus Fucking Christ" |
| 02:58.27 | ldsjohn | I can't help but picture that in my head |
| 02:58.56 | ldsjohn | I mean if you take that phrase litterally like my mind does, then there are only a couple ways it can go |
| 02:59.55 | [TK]D-Fender | ldsjohn: One is best NOT taking biblical passages literally. |
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| 03:43.57 | *** join/#asterisk ramindia (n=Administ@202.63.96.10) |
| 03:44.45 | ramindia | hey i need some paid consultant programmer to write small script in Asterisk.. PM me any one intrested |
| 03:47.05 | [TK]D-Fender | ramindia: To do what? |
| 03:51.17 | ramindia | Writing a Custom and advanced Dialplan |
| 03:51.20 | [TK]D-Fender | ... |
| 03:51.27 | [TK]D-Fender | ramindia: can you be any more specific... |
| 03:52.09 | ramindia | i want to use Asterisk , |
| 03:52.38 | ramindia | Asterisk1--- Asterisk2---provider 1 and provider 2 and so on |
| 03:53.08 | ramindia | Provider1 and 2 accept the call with PIN authentication and dial to destination |
| 03:53.56 | ramindia | so asterisk2 dials 1XXXXX to provider1 and wait 2 seconds.. send DTMF ( pin) wait 3 seconds.. when the call connects.. send back the same to Asterisk1 |
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| 03:55.39 | [TK]D-Fender | <PROTECTED> |
| 03:56.42 | ramindia | for Asteriks1.. Asterisk2 is the gateway |
| 03:57.17 | ramindia | so Asterisk1 have users.......so when the user dial destination number.. the process starts |
| 03:59.27 | [TK]D-Fender | ramindia: ok, this is straight-line dialplan between #1, to #2, with an outbound DTMF to add |
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| 04:00.51 | ramindia | there is some database integration required |
| 04:01.01 | ramindia | i have 100 calling cards with 100pins |
| 04:01.21 | ramindia | if one fails it need to pick up from the database and dial |
| 04:01.54 | ramindia | anotherone |
| 04:02.50 | [TK]D-Fender | ramindia: There is no way to detect this failure |
| 04:03.36 | ramindia | so is this going to be blind dial |
| 04:06.25 | ramindia | how will i find out that call failed |
| 04:06.53 | ramindia | take example..either customer dialed wrong number or not used number i should be able to detect the call failed right |
| 04:08.17 | [TK]D-Fender | ramindia: There is no way to detect this failure <--------------------------- |
| 04:09.59 | ramindia | is this possible, if i wont get response from calling card 1.. can imeediatly intiate the call with another calling card and pin to the provider ? |
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| 04:14.26 | [TK]D-Fender | ramindia: There is no way to detect this failure <--------------------------- what part about this don't you get? The call is ACCEPTED at the PSTN level there is no secondary detection. Your plan will not work. |
| 04:15.22 | ramindia | ok |
| 04:16.23 | ramindia | thanks for suggestion |
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| 04:25.09 | Slade- | hey. not really an asterisk question.. but do most sip trunk providers do the tone detection for you and pass it on to asterisk like the rfc says? |
| 04:26.30 | Slade- | the asterisk part of that question is.. does asterisk handle that, or insist on doing its own tone detection? |
| 04:28.03 | [TK]D-Fender | Slade-: Yes |
| 04:28.12 | Slade- | yes to both? |
| 04:28.51 | [TK]D-Fender | Slade-: Depends. Usually they do it and pass OOB |
| 04:29.35 | Slade- | and asterisk eats that info up instead of doing its own tone detection? |
| 04:30.04 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 04:30.07 | Slade- | do i have to configure it in any way to do so |
| 04:30.50 | [TK]D-Fender | SlaYou configue * to match how your ITSP send it to you |
| 04:31.04 | [TK]D-Fender | Slade-: If they support OOB then you set your peer to match |
| 04:31.12 | Slade- | ok |
| 04:31.20 | [TK]D-Fender | Slade-: Some on G711 just pass audio and leave it to * to detect |
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| 04:32.11 | Slade- | ok |
| 04:32.15 | Slade- | reads up :) |
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| 05:25.56 | [TK]D-Fender | checkout time, later all |
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| 06:20.08 | phigphone | Morning |
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| 08:32.07 | TeleNet | I have installed asterisk with ss7 and dadhi, when I dial from my mobile number I cant able to hear the IVR voice, whereas I can able to see in asterisk console it is playing. Why is it so? |
| 08:39.18 | *** join/#asterisk linagee (n=jalton@about/linux/staff/linagee) |
| 08:40.16 | linagee | what's a better setup? FXS/FXO card inside a computer with 25-pair connector, or T1 card in computer with external channel bank with 25 FXO/FXS's? |
| 08:42.54 | linagee | oh wow. definitely the external channel bank I guess. It seems cheaper. |
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| 08:44.31 | TeleNet | linagee: one T1 card and external channel bank is the best |
| 08:45.51 | linagee | TeleNet: that way seems cheaper too. |
| 08:46.13 | linagee | TeleNet: I'd almost think there would be less problems putting a T1 card in a computer than a TDM2400/etc |
| 08:49.22 | TeleNet | linagee: use sangoma card it is the best |
| 08:49.39 | linagee | TeleNet: if you hook a 25-pair amphenol to a 25-pair cable to a 66-block with amphenol connector and never have to terminate 50 different wires, is that cheating? :) |
| 08:50.29 | linagee | terminating 25 pairs into an amphenol connector seems slightly nightmarish to me. lol |
| 08:54.29 | TeleNet | linagee: sometimes you will get confused by connecting the amphenol cable |
| 08:55.46 | linagee | TeleNet: sometimes? lol |
| 08:55.57 | linagee | TeleNet: i will die a happy human if I never have to terminate 25-pairs by hand. lol |
| 08:56.29 | linagee | TeleNet: oh, unless it's a charge by the hour sort of incident. :) |
| 08:56.55 | linagee | but somehow a $15 predone cable just doesn't seem worth it unless you can terminate that in like 5 minutes. lol |
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| 09:03.20 | linagee | TeleNet: FXS just doesn't seem worth it. hah |
| 09:03.44 | linagee | TeleNet: $90 of equipment to hook up a $10 phone. for that price, might as well get a voip phone |
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| 09:05.43 | linagee | true an ethernet port may also cost you $40-$50 on the switch side (especially if it's PoE), still seems more worth it thoguh |
| 09:13.13 | TeleNet | linagee: :D |
| 09:17.13 | linagee | TeleNet: I've seen an unnamed restraunt switching from T1s going into ciscos to DSL modems, and AT&T boxes with a 3G PC card and WAN port for the DSL. |
| 09:17.37 | linagee | cisco + T1 -> DSL + AT&T 3G as a backup |
| 09:18.02 | linagee | the DSL modem was consumer grade even. I guess people don't care for high end when you have a redundancy anyway. lol |
| 09:18.38 | linagee | I'd bet the T1 was costing them $1000/mo and AT&T must have cut some deal for maybe $100 + maybe $100 DSL. |
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| 09:27.39 | TeleNet | linagee: T1 card will cost you only 300$ |
| 09:28.09 | TeleNet | linagee: to 600$ |
| 09:30.09 | linagee | TeleNet: $1000/mo for the service |
| 09:30.18 | linagee | $1000 plus or minus a couple hundred |
| 09:30.57 | TeleNet | oh ok |
| 09:31.00 | linagee | TeleNet: why does a T1 cost so much when it's only taking 2 pairs of wire? ;) |
| 09:31.25 | linagee | TeleNet: is the real expense of a telephone company on the switch side or something? is wiring costs nothing to them? |
| 09:33.52 | TeleNet | normally t1 card is very cheaper |
| 09:35.39 | linagee | TeleNet: I don't think central offices have T1 cards though |
| 09:35.50 | linagee | TeleNet: they probably have something that plugs in like 8 T1s at one time or something |
| 09:40.24 | TeleNet | yes 8 T1 are costlier |
| 09:44.06 | linagee | I meant the telco likely does not have one line card just for your measley T1 in their central office. they likely have something that takes multiple T1s on one card. |
| 09:44.18 | linagee | or they multiplex it into an DS3 or something |
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| 10:00.42 | Nunners | http://pastebin.com/d45d81ecc For some reason call waiting has suddenly started. Is there an easy way of removing it? I had tried in the users.conf file, but to no avail.... |
| 10:02.20 | Nunners | QUIT (resolved missed another callwaiting line) |
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| 10:45.29 | barbacha | hey |
| 10:45.50 | barbacha | the blind transfer feature (##) stoped working for some reason. What should I investigate first ? |
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| 10:56.36 | sulex | I'm using Wiatexten in a IVR menu, I've the standard 't' and 'i' extensions sets within the context. "i" works great as the only dtmf I accept is "1". But if i just wait for the timeout to reach the "t" extension this is simply not considered and asterisk goes ahead and doesn't jump to "t" |
| 10:57.07 | sulex | <PROTECTED> |
| 10:57.15 | sulex | <PROTECTED> |
| 10:57.17 | sulex | ideas? |
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| 11:27.51 | maqr | on my dialplan, i Answer() and then Dial() my extension (since i'm the only user), and if i'm not there it goes to Voicemail() so people can leave me voicemail... but if i want to check that voicemail remotely, i'm not sure how to work that into the dial plan, adding an extension doesn't seem to work, as there's no time to dial it |
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| 11:39.13 | maqr | i think what i want is to allow the caller to dial an extension while a Dial() is in progress |
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| 11:57.10 | Blackvel | hey all. can't remember the website. there was a voice demo where a callcenter agent called in and asterisk script/robot was entertaining >20minutes. anyone with the google-keywords or the website-link? |
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| 12:00.47 | mort_gib | How stable is 1.6. ?? |
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| 12:04.15 | mort_gib | Another thing, I need to use a B410p card with Dahdi, but with point-to-multipoint is that supported ?? |
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| 12:08.00 | beek | Good morning all |
| 12:08.24 | j_kroon | if I pass a call out to a SIP channel which then gets rejected, how can I get at the SIP error code? |
| 12:09.01 | j_kroon | the HANGUPCAUSE variable gives me a PRI-like value, which I can probably work with, but I'd prefer working with the SIP codes direct. |
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| 12:16.20 | j_kroon | the HANGUPCAUSE variable gives me a PRI-like value, which I can probably work with, but I'd prefer working with the SIP codes direct. |
| 12:16.46 | j_kroon | i would like to gain access to the SIP reject cause if possible. |
| 12:18.45 | mort_gib | How stable is 1.6.1 ?? |
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| 12:22.11 | [TK]D-Fender | mort_gib: Bleeding edge... Should wait until .2 release minimum before considering for production |
| 12:22.11 | mort_gib | Thanks TK, |
| 12:22.14 | mort_gib | I have one last client that remains "unstable" and the only thing I haven't tried is another server |
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| 12:22.54 | gr0mit | so, no-one used Yealink phones?? |
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| 12:24.48 | [TK]D-Fender | gr0mit: Yay!....link |
| 12:25.08 | gr0mit | are they any good? |
| 12:25.24 | gr0mit | fears not... |
| 12:25.25 | [TK]D-Fender | gr0mit: http://www.yealink.com/en/index.asp <- GAH, blatant Cisco / GS ripoffs |
| 12:25.36 | gr0mit | hehe!!! |
| 12:25.39 | [TK]D-Fender | gr0mit: Wouldn't touch with a 10' pole |
| 12:25.57 | gr0mit | has a polycom 410 arriving soon |
| 12:26.06 | j_kroon | trust TK to be as brutally honest as always |
| 12:26.08 | gr0mit | bought of ebay for a song |
| 12:26.36 | [TK]D-Fender | gr0mit: .... 410? |
| 12:27.03 | j_kroon | ok, seeing as we've just recovered from a netsplit - after a Dial() call, if the call got rejected, is it possible to gain access to the SIP reject code, eg 404 for NOTFOUND etc ? |
| 12:27.24 | gr0mit | sorry ip430 |
| 12:27.30 | j_kroon | it seems these gets translated into PRI CAUSECODES which can be found in HANGUPCAUSE but I'd prefer to work with the SIP codes if possible. |
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| 12:52.24 | Katty | GOOD MORNING!!!!! |
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| 12:54.41 | leifmadsen | licks Katty |
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| 12:56.20 | VoipForces | Hi, anyone has come across an asterisk log parser that would be able to give the steps a call took via the asterisk log file ? I can do this by hand but it's long and and tedious to remove the extra junk and to make it "non asterisk human readeable"... |
| 12:56.41 | [TK]D-Fender | Katty: Mew. |
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| 12:57.42 | Katty | [TK]D-Fender: mew. |
| 12:57.44 | Katty | hugs leifmadsen |
| 12:57.50 | Katty | leifmadsen: no licking please! you might smudge the makeup |
| 12:58.01 | leifmadsen | Katty: pffft makeup |
| 12:58.02 | Katty | or war paint, as my mom calls it. |
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| 12:58.58 | [TK]D-Fender | "Marriage : the only war where you get to have sex with the enemy" |
| 12:59.32 | j_kroon | hi, is it possible to get at the SIP error code in the case of CONGESTION/CHANUNAVAIL DIALSTATUS? |
| 12:59.58 | leifmadsen | j_kroon: I don't believe so |
| 13:00.09 | j_kroon | leifmadsen, second question: how? |
| 13:01.08 | VoipForces | might have missed if there were reply to my question as I had to reboot my phone (which acts as my laptop switch) cause someone had taken my office phone while I was at a custoner site... |
| 13:01.10 | [TK]D-Fender | j_kroon: He just told you NO. |
| 13:01.32 | j_kroon | misread that. crap. oh well. the PRI translated codes will then have to do. |
| 13:01.50 | Katty | [TK]D-Fender: speaking of marriage |
| 13:01.56 | Katty | [TK]D-Fender: i got my sister's wedding photos! |
| 13:01.59 | [TK]D-Fender | Katty: You got "rocked"? |
| 13:02.01 | [TK]D-Fender | .... |
| 13:02.34 | Katty | [TK]D-Fender: my sister's wedding photos. |
| 13:02.40 | Katty | [TK]D-Fender: i got her boquet(sp?) tho |
| 13:02.48 | Katty | [TK]D-Fender: she sent it home with mom |
| 13:02.55 | [TK]D-Fender | Katty: "Always the bridesmaid, never the bride" |
| 13:03.05 | Katty | something like that. |
| 13:06.05 | Katty | i must be defective :< no one wants to marry me. |
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| 13:09.13 | *** join/#asterisk nighty^ (n=nighty@x122091.ppp.asahi-net.or.jp) [NETSPLIT VICTIM] |
| 13:09.13 | *** join/#asterisk VaGoNeTaS (n=debian@xen.datapartner.cl) [NETSPLIT VICTIM] |
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| 13:09.34 | mort_gib | Katty: This channel is flooded with geeks that has a very dim image of what a girl is, you should be able to get a few leads here ;-) |
| 13:11.10 | Katty | mort_gib: i don't need any leads, i'm already... leaded... |
| 13:11.13 | Katty | mort_gib: or osmething. |
| 13:11.48 | mort_gib | Katty: Yeah, then take it as a warning :-) |
| 13:12.10 | Katty | what? ^_- |
| 13:12.17 | Katty | you do not parse. |
| 13:12.38 | mort_gib | -Still trying to figure out if you can actually write that in (US) English :-) |
| 13:12.47 | mort_gib | leaded.... |
| 13:13.50 | [TK]D-Fender | mort_gib: Yes, but not in that context |
| 13:14.15 | Katty | leaded or unleaded. |
| 13:14.18 | *** join/#asterisk Victor_Yure (n=victor@unaffiliated/victoryure/x-837844) |
| 13:14.19 | mort_gib | [TK]D-Fender: It thought as much, but not being English/American.... |
| 13:14.23 | Katty | would be the appropriate context. |
| 13:16.19 | Katty | Lead = Pb |
| 13:16.44 | mort_gib | Or laden.... |
| 13:16.52 | gr0mit | or loaded! |
| 13:17.00 | Katty | or Plumbum |
| 13:17.03 | Katty | (really!) |
| 13:17.18 | [TK]D-Fender | plomb |
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| 13:40.51 | sulex | when a call reaches the "h" extension because the called party hangup, can i tranfer the caller to another context or queue? |
| 13:43.39 | SuPrSluG | yep |
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| 13:46.16 | Blackvel | who has experiences with speech delay on ISDN with Snom 370 and patton smartnode 4634? |
| 13:46.25 | [TK]D-Fender | sulex: No, the channel is dead. |
| 13:46.37 | [TK]D-Fender | sulex: If you want to continue after callee hangup use the "g" dial option |
| 13:46.40 | *** join/#asterisk bakermd (n=bakermd@38.101.225.215) |
| 13:47.03 | bakermd | What are you guys paying for outbound or 2-way SMS messaging? |
| 13:47.07 | Blackvel | the other party does not hear when i am starting to speak. I can not interrupt. or both parties speak at the same time. |
| 13:47.23 | jtexter3 | Is there a way to configure chan_dahdi to do SIT detection on a PRI? |
| 13:47.53 | Blackvel | is this a general snom problem? should not happen when my local network is interconnected to pstn (digital isdn) with patton media GW |
| 13:47.56 | jtexter3 | I have a number that provides a tri-tone, but no cause code in the disconnect message, so I'd like to be able to catch the tri-tone |
| 13:48.18 | sulex | [TK]D-Fender: and the "g" is set, but when the called pary press * the channel doesn't hang... except setting "g" in the dial string do I have to set anything else elsewhere for "g" to work? thanks? |
| 13:48.31 | [TK]D-Fender | Blackvel: So this doesn't happen witha softphone on the same subnet as the SNOM? |
| 13:48.57 | [TK]D-Fender | sulex: "g" if for any reason the calle disconnects |
| 13:49.37 | sulex | [TK]D-Fender: so if he just hangs up the caller will go ahead with the next extension? |
| 13:49.52 | [TK]D-Fender | sulex: Yes, just like the instructions say |
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| 13:50.36 | *** join/#asterisk putnopvut (n=putnopvu@asterisk/master-of-queues/mmichelson) |
| 13:50.36 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 13:51.11 | *** join/#asterisk leifmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 13:51.11 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 13:51.23 | Blackvel | [TK]D-Fender: did not test so far in local net. could not test this very satisfyingly, as i do not have a professional 200euro headset (how can I make conclusions with 20-30 euro headset hardware)... |
| 13:51.35 | Blackvel | maybe I should find a way to test... |
| 13:52.00 | [TK]D-Fender | blackYou're talking about a complete conversation dely. A 5$ PIECE OF SHIT HEADSET WILL DO. |
| 13:52.07 | [TK]D-Fender | delay* |
| 13:52.53 | Blackvel | did you guys ever experience delay problems on local net and snom firmware (some times minimal, one time more delay) |
| 13:52.55 | [TK]D-Fender | Blackvel: and only ONE side is likely to be the issue, not both |
| 13:53.04 | [TK]D-Fender | Blackvel: Go do a REAL rest to prove it |
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| 13:53.48 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 13:53.50 | rue_mohr | little snappy this morning? |
| 13:54.23 | [TK]D-Fender | rue_mohr: Only a little, and the big caps line was an accident. |
| 13:54.35 | [TK]D-Fender | rue_mohr: Key word focus isn't though |
| 13:55.06 | sulex | [TK]D-Fender: yeah, most of the time "just like instructions" say is correct... not always unfortunately with * :) |
| 13:55.28 | rue_mohr | I think I'm gonna have a ball converting my home system from 1.2 |
| 13:55.39 | [TK]D-Fender | sulex: Yes, but not following them just because they "might" be wrong is a sad cop-out. |
| 13:55.51 | Blackvel | is this usually only a problem on ONE side (like mine with snom voip and eventually patton) as the other party was detached by ISDN (not sure what the company's TK equipement was)? it it not a one-time ever problem. happens on most my calls more or less |
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| 13:58.05 | rue_mohr | so, the SVN dahdi drivers for 1.4 caused interesting effects if a call origionated from the co there was no audio either way(thru the tdm800), but it if came from inside the office, it was ok, I think it was the latest svn, anyone heard of any odd goings on? |
| 13:58.28 | rue_mohr | I didn't have time to flush it out much as I has only enough time to scramble to re-install hte old version |
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| 13:59.40 | beek | I just upgraded from 1.6.9 to 1.6.10 on one server using the patch file... no problem. |
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| 14:00.19 | beek | I have another server at 1.4.24.1 and when I try to upgrade to 1.4.25 using its patch file, I get three FAILED patches. |
| 14:00.40 | rue_mohr | can you not use the patch? |
| 14:00.48 | beek | Do I have to use the patch file for 1.4.25-rc1 to get to .25? |
| 14:00.57 | rue_mohr | shoudl I presume your suign the patch cause you changed something? |
| 14:01.44 | *** join/#asterisk mikkel (n=mikkel@84-238-113-66.u.parknet.dk) |
| 14:01.57 | beek | rue_mohr: Nope. The only edits I do are to the installed files. |
| 14:02.26 | *** join/#asterisk _Steve_ (n=Steve@unaffiliated/steve/x-520345) |
| 14:02.44 | _Steve_ | how can i tell how many concurrent calls are active on my sip trunks? |
| 14:02.45 | rue_mohr | k, so why dont you just grab the 1.4.25 src? |
| 14:03.00 | rue_mohr | sip show inuse ? |
| 14:03.17 | rue_mohr | or you jsut want a count? |
| 14:03.22 | beek | rue_mohr: I can do that, it's just a lot of additional bandwidth. |
| 14:03.33 | rue_mohr | oh |
| 14:03.52 | _Steve_ | i want to figure out why i get "all circuits are busy" |
| 14:03.56 | rue_mohr | all I can say about your patch is 'somethings different' |
| 14:04.16 | rue_mohr | _Steve_, where are they trying to dial to? |
| 14:04.28 | _Steve_ | anything, all numbers get it |
| 14:04.56 | rue_mohr | I dont remember if thats a possable result of "I cant get there" |
| 14:06.01 | beek | rue_mohr: It's wierd. I just untarred the 1.4.24.1 source into a clean directory and can't apply the patch... Oh well, I'll just pull down the source for the latest and recompile. |
| 14:06.07 | *** join/#asterisk tobias (n=tobias@cpe-071-070-219-040.nc.res.rr.com) |
| 14:06.30 | rue_mohr | beek, sorry I cant help |
| 14:06.55 | rue_mohr | someone in asterisk-dev might know |
| 14:07.04 | seanbright | what patch are we talking about? |
| 14:07.07 | beek | I'll check there as I'm curious. |
| 14:07.17 | seanbright | oh i see |
| 14:07.19 | beek | seanbright: I'm trying to patch 1.4.24.1 ---> 1.4.25 and getting problems. |
| 14:07.36 | seanbright | let me give it a shot. hold. |
| 14:07.51 | rue_mohr | hahah you been put on hold |
| 14:08.03 | rue_mohr | plays beek some hold music |
| 14:08.06 | beek | rue_mohr: I really like the MOH they're using. |
| 14:08.22 | rue_mohr | its kinda funkey eh? |
| 14:08.54 | beek | Hmmm... the deaf guy next to me is tapping his foot in time to the music. |
| 14:09.22 | leifmadsen | beek: he's been faking it, and listening to your conversations |
| 14:09.36 | leifmadsen | haxoring your emailz |
| 14:09.49 | beek | glances around nervously |
| 14:10.44 | seanbright | beek: looks like sip.conf.sample is the only problem |
| 14:10.50 | seanbright | err |
| 14:10.52 | seanbright | and ChangeLog |
| 14:11.05 | beek | I'm glad it's not me. |
| 14:11.12 | seanbright | but you're right, the patch appears jacked |
| 14:11.17 | seanbright | blame leifmadsen. |
| 14:11.24 | beek | leifmadsen: it's your fault. |
| 14:11.26 | leifmadsen | blames mkrelease |
| 14:11.34 | seanbright | that too. |
| 14:11.36 | leifmadsen | points at mkrelease |
| 14:11.50 | leifmadsen | you probably need to apply it to 1.4.24 |
| 14:11.55 | leifmadsen | not 1.4.24.1 |
| 14:12.00 | leifmadsen | see if that works |
| 14:12.14 | seanbright | tries |
| 14:12.37 | leifmadsen | w00t! google pin entered -- I can get adsense payments now |
| 14:12.42 | beek | is glad seanbright is doing it so he doesn't have to download that tarball |
| 14:12.44 | leifmadsen | probably in 6 months when it reaches $100 :) |
| 14:13.38 | seanbright | leifmadsen: that worked. |
| 14:14.05 | *** join/#asterisk angryuser (n=angryuse@LPuteaux-151-42-27-99.w193-251.abo.wanadoo.fr) |
| 14:14.09 | seanbright | so patches only work between maintenance releases |
| 14:14.12 | seanbright | not security ones |
| 14:14.42 | *** join/#asterisk moy (n=moy@74.12.123.90) |
| 14:14.58 | leifmadsen | seanbright: probably because of the way they are made -- if I don't run make release against the full previous release, the ChangeLog gets out of whack |
| 14:15.17 | seanbright | ahhh |
| 14:15.17 | *** join/#asterisk tobias (n=tobias@cpe-071-070-219-040.nc.res.rr.com) |
| 14:15.19 | seanbright | i don't download tarballs anyway |
| 14:15.21 | leifmadsen | and you lose all the commit entries between 1.4.24 and 1.4.24.1 when you make 1.4.25 |
| 14:15.23 | seanbright | cuz i'm hardcore. |
| 14:15.38 | leifmadsen | EXTREME!!!!!!! |
| 14:16.08 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
| 14:16.29 | beek | leifmadsen: For my future reference then, I'll do a cp -a to duplicate the current directory when a security .x release comes out and patch that separately. |
| 14:16.49 | leifmadsen | beek: thanks! |
| 14:16.58 | beek | Thank you guys! |
| 14:17.03 | leifmadsen | np |
| 14:17.07 | leifmadsen | thanks seanbright! |
| 14:17.43 | seanbright | no problem. |
| 14:18.45 | *** join/#asterisk ariel_ (i=3fd6eca9@gateway/web/ajax/mibbit.com/x-25e307368d2d1ade) |
| 14:19.04 | *** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler) |
| 14:19.16 | sulex | [TK]D-Fender: any idea why the "g" is not considered if the calling party reaches the agent through a queue. If i call the agent extension directly "g" works properly, when I call the queue and I'm passed to that agent, the "g" option seems not considered at all. What i wanna do is to forward the caller to an extension if an agent hangs up the call... |
| 14:26.22 | *** join/#asterisk qdk (n=qdk@0x573d8dd1.bynqu1.dynamic.dsl.tele.dk) |
| 14:26.46 | *** join/#asterisk nighty^ (n=nighty@x122091.ppp.asahi-net.or.jp) |
| 14:27.07 | [TK]D-Fender | sulex: PASTEBIN <---- |
| 14:27.08 | [TK]D-Fender | ~pb |
| 14:27.09 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste , or , http://bin.cakephp.org/ |
| 14:27.29 | *** join/#asterisk jmacz (n=jmacz@190.144.75.22) |
| 14:27.43 | jaytee | gee, last time I looked Dial() and Queue() were two different apps |
| 14:28.20 | *** join/#asterisk SlipperyChicken (n=andrew@LONDON14-1168107385.sdsl.bell.ca) |
| 14:28.28 | [TK]D-Fender | jaytee: Yes, and Queue can call all kinds of channels... do the math and think a step or two ahead :) |
| 14:29.06 | *** join/#asterisk ctp (n=ctp@brsg-d9bee35c.pool.mediaWays.net) |
| 14:29.35 | leifmadsen | depends how the Queue() is calling the agent |
| 14:29.44 | leifmadsen | the Agent channel will have no concept of 'g' |
| 14:29.55 | leifmadsen | and if you're using a Local channel, that may not work either |
| 14:30.52 | [TK]D-Fender | leifmadsen: there is another 1.6 series feature that may work as well. |
| 14:31.14 | leifmadsen | [TK]D-Fender: ? |
| 14:32.14 | [TK]D-Fender | leifmadsen: 1.6.1 IIRC... "goto on hangup" |
| 14:32.25 | [TK]D-Fender | leifmadsen: Don't ahve it installed, but recall reading about |
| 14:32.36 | leifmadsen | never heard of it :) |
| 14:32.43 | *** join/#asterisk arpu (n=arpu@chello080109017107.12.14.vie.surfer.at) |
| 14:32.56 | leifmadsen | although I don't have all the Dial() options memorized |
| 14:33.21 | [TK]D-Fender | leifmadsen: I occasionally note the cool & useful bits :) |
| 14:33.42 | sulex | [TK]D-Fender: http://pastebin.com/m7eb1004d |
| 14:34.02 | sulex | [TK]D-Fender: not all the agents entry are listed, just to give an easier read |
| 14:35.51 | [TK]D-Fender | sulex: Who hung up. The caller, or "SIP/fadda_loris"? |
| 14:36.24 | sulex | SIP/fadda_loris him |
| 14:36.31 | sulex | the called party, sorry |
| 14:36.50 | sulex | if i place a call to 666 directly "g" works |
| 14:36.58 | *** join/#asterisk ramindia (n=Administ@202.63.96.10) |
| 14:38.27 | [TK]D-Fender | sulex: Ok, could be queue related... |
| 14:38.42 | [TK]D-Fender | sulex: If you're sure the dialplan is properly loaded the rest looks OK |
| 14:39.15 | sulex | [TK]D-Fender: yep, I've tried also empting the db and restarting asterisk... this is why I was playing with "h" |
| 14:39.32 | sulex | because "g" seems not working with dial() if called from a queue |
| 14:39.33 | [TK]D-Fender | sulex: won't work though... "h" is whent he channel is dea... |
| 14:39.36 | [TK]D-Fender | dead* |
| 14:39.39 | sulex | yeah :( |
| 14:41.31 | *** join/#asterisk chendy (n=alex@58.60.218.202) |
| 14:43.18 | *** join/#asterisk feder (n=feder@host.190.15.192.172.static.itcsa.net) |
| 14:44.55 | feder | hello, does anybody know what happened with the astxs script? It's not included with asterisk 1.4.24.1 |
| 14:45.38 | *** join/#asterisk |Cybex| (n=John@80.100.126.176) |
| 14:46.12 | *** join/#asterisk Great_Anta_Baka (n=tensai@196.33.159.83) |
| 14:46.15 | tzafrir_laptop | no. it no longer works |
| 14:46.56 | ariel_ | wonders what was that script for! |
| 14:47.11 | feder | to compile and install apps |
| 14:47.22 | angryuser | hello, i need asterisk to receive a call but not say 200 OK to the proxy, i to place that call on hold and continue to send 183 Session in progress, any hints how to do that ? |
| 14:47.43 | *** join/#asterisk CunningPike (n=CunningP@204.239.10.119) |
| 14:47.51 | angryuser | proxy will take it back in 10 sec |
| 14:48.15 | feder | I need to put AMD or MachineDetect on my asterisk and I don't know how to do that without astxs, any ideas? |
| 14:48.16 | *** join/#asterisk propellerhead (n=yogurt2u@host146.190-137-3.telecom.net.ar) |
| 14:48.39 | *** join/#asterisk Pazzo (n=ugelt@reserved-225136.rol.raiffeisen.net) |
| 14:48.42 | angryuser | i need it to place* |
| 14:49.12 | [TK]D-Fender | angryuser: "vi chan_sip.c" |
| 14:49.47 | *** join/#asterisk SkywaIker (n=pirch@58.147.17.166) |
| 14:50.11 | [TK]D-Fender | feder: I seem to have app_amd on mine by default... |
| 14:50.20 | [TK]D-Fender | feder: No extra install required. |
| 14:50.49 | angryuser | [TK]D-Fender, let's assume that i dont program in C |
| 14:50.54 | *** join/#asterisk ctp (n=ctp@brsg-d9bee35c.pool.mediaWays.net) |
| 14:51.01 | [TK]D-Fender | angMy answer implies that that is the only way <- |
| 14:51.16 | [TK]D-Fender | angryuser: So if you don't program in C, go hire someone to do it for you |
| 14:52.04 | feder | <[TK]D-Fender>: great, I'm such an idiot I looked for AMD on "core show functions" not in applications :P |
| 14:52.13 | feder | <[TK]D-Fender> many many thxs |
| 14:52.26 | [TK]D-Fender | feder: np |
| 14:54.04 | *** join/#asterisk Torrieri (n=Torrieri@nelug/crew/torrieri) |
| 14:54.56 | eppigy | GUTEN MORGEN |
| 14:58.15 | *** join/#asterisk aenaus_ (n=hdgfghf@79.107.52.142) |
| 14:58.16 | gr0mit | gueten Tag |
| 14:59.33 | *** join/#asterisk malcolmd (n=malcolmd@pdpc/sponsor/digium/malcolmd) |
| 15:01.45 | *** join/#asterisk mchou (n=mchou@unaffiliated/mchou) |
| 15:05.01 | leifmadsen | godmorgen |
| 15:05.38 | *** join/#asterisk mltlnx (n=mltlnx@mail.shorefire.com) |
| 15:05.39 | coppice | æ©è¾° |
| 15:06.43 | leifmadsen | neato -- I can see that :) |
| 15:06.53 | jaytee | what is that? Kanji for Bum surfer? |
| 15:07.37 | coppice | its Hanzi for good morning |
| 15:09.03 | jaytee | ah, also known as pinyin and the origin of kanji |
| 15:09.44 | coppice | that's not pinyin |
| 15:10.13 | Nugget | ã³ã¼ãã¼ã¯ããããã§ãã |
| 15:10.50 | *** join/#asterisk hugorebelo (n=hugorebe@200-171-132-124.completo.com.br) |
| 15:11.22 | jaytee | oh, right! duh. pinyin is phonetic romanization, hanzi is ideogramattic |
| 15:11.43 | ldsjohn | no they both are just scribbles |
| 15:11.44 | *** join/#asterisk HaMF (n=hannes@nat-wh.rz.uni-karlsruhe.de) |
| 15:11.50 | jaytee | hehe |
| 15:12.11 | ldsjohn | I have a friend who served a mission in taiwan and learned mandarin |
| 15:12.33 | coppice | I have no idea if æ©è¾° comes out the same in Kanji or not. Some things are close or the same. some are very different. |
| 15:12.34 | eppigy | DI DI MAO |
| 15:13.00 | coppice | ldsjohn: so what? every friend I have knows Mandarin |
| 15:13.33 | coppice | well, almost every friend |
| 15:13.35 | ldsjohn | :-D thats why I think its scribbles, barbaric scribbles at that. he taught me some I just never really got into it |
| 15:13.42 | ldsjohn | its like, me john I work go |
| 15:13.49 | ldsjohn | you ug you fire make |
| 15:14.56 | ariel_ | argh that hurt... I am having to redo my 3 E1's. I forgot the heat shrink labels....and our tech can't figure out which cable is which... Maybe I should replace the tech's instead....;-) |
| 15:15.26 | ldsjohn | wait your the one who forgot the labels? and you want to replace him? |
| 15:15.34 | coppice | chinese is a language without tenses, which is kinda weird when you first encounter it |
| 15:16.27 | ariel_ | ldsjohn: the cable is only 3 feet long... |
| 15:17.02 | *** join/#asterisk pierrelux (n=pierrelu@IP-208-88-110-46.mtl.fibrenoire.ca) |
| 15:17.07 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 15:17.33 | *** join/#asterisk SwK (n=SwK@freeswitch/developer/swk) |
| 15:17.40 | [TK]D-Fender | coppice: Chinese too? Was hoping that was an oversight limited to Japanese :) |
| 15:17.57 | ldsjohn | well, I encountered something like that once, where we needed to know wich was wich, since there were 2 and we didn't know wich was wich, we just opened zttool and then unplugged one, on, it went red and then we knew wich was wich |
| 15:18.18 | jaytee | so there's no distinction whatsoever between past, present and future tense in verbs? |
| 15:18.36 | ldsjohn | well there isn't much distinction from what i was taught |
| 15:18.52 | ldsjohn | if I wanted to say I went to work or I am going to work it seamed to be the same words. unless I just had a bad teacher |
| 15:19.13 | coppice | jaytee: time is an illusion. lunchtime doubly so |
| 15:19.41 | eppigy | well I will be partaking in some very tangible food |
| 15:19.55 | eppigy | on my illusory lunch time |
| 15:20.00 | eppigy | YA FEEL ME |
| 15:20.25 | ariel_ | ldsjohn: I would have if they were not different colors. Besides it's just patch cables. Red ones are for E1 Crossover and Green ones for E1 normal. |
| 15:20.35 | jaytee | "The source of all unhappiness is other people. As soon as you learn to think of other people as noisy furniture, the sooner you will be happy" - Wally from Dilbert |
| 15:22.07 | ldsjohn | ariel_: so it should be easy to figure out :-D |
| 15:23.09 | ldsjohn | ariel_: I am pretty sure that zttool and zapata.conf should make it easy to figure out wich is wich though |
| 15:25.01 | *** join/#asterisk anonymouz666 (n=anonymou@189.24.175.58) |
| 15:25.22 | *** join/#asterisk jameswf (n=james@unaffiliated/jameswf-home) |
| 15:26.18 | *** join/#asterisk Dovid (n=annon@72-59-101-4.pools.spcsdns.net) |
| 15:26.18 | ariel_ | ldsjohn: even easier they just need to plug one end to the patch panel other end to the box of the asterisk box. They are installing it. Cables are loose in the bag. (there just being baby's) |
| 15:26.44 | Dovid | hi - anyone here in France ? |
| 15:29.29 | timeshell_atwork | Dovid? Est-ce que c'est important? |
| 15:29.51 | timeshell_atwork | Ou tu veux parler francais? |
| 15:30.43 | Dovid | i need some one in france to test a number |
| 15:31.08 | beek | Dovid: head over to #voip-users-conference. Zeeek lives in France. |
| 15:31.18 | Dovid | thanks |
| 15:31.22 | beek | Dovid: He's on now. |
| 15:31.29 | mort_gib | Dovid: well, I'm sure you will be surprised to hear that it's possible to call from Country to Country |
| 15:31.34 | mort_gib | :-) |
| 15:31.59 | angryuser | Dovid, depends, why ? |
| 15:33.18 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 15:35.39 | Dovid | i needed a number there tested |
| 15:35.44 | Dovid | i cant call from out of france to it |
| 15:36.09 | timeshell_atwork | Why not? |
| 15:36.28 | Dovid | TF in france |
| 15:36.39 | *** join/#asterisk viraptor (n=viraptor@87-194-164-154.bethere.co.uk) |
| 15:36.50 | timeshell_atwork | Isn't there a way to dial TF in other countries? |
| 15:37.49 | timeshell_atwork | http://mobiletechnology.suite101.com/article.cfm/how_to_dial_tollfree_numbers_abroad |
| 15:38.02 | viraptor | could someone remind me which version of asterisk was it when the problem with high IAX callno-s was discovered (resulting in massive delays)? |
| 15:38.48 | Dovid | thanks |
| 15:39.10 | Nugget | http://lnk.nu/canford.co.uk/wuq <-- rackmount |
| 15:39.53 | *** join/#asterisk jplank (n=GBove@cpe-075-181-097-208.carolina.res.rr.com) |
| 15:39.55 | Qwell | Nugget: ha |
| 15:42.15 | carrar | is that 19" or telco? |
| 15:42.42 | carrar | replace those wine bottles with something a little harder |
| 15:43.19 | carrar | suddenly spending all day in the datacenter isn't too bad of a idea |
| 15:43.57 | carrar | cooling fans on the back might be nice too |
| 15:44.23 | Qwell | carrar: cooling? wine? |
| 15:44.29 | beek | Is it just me or does 1.4.25 + patch still show 1.4.25, not 1.4.25.1, when connecting to the console with asterisk -r? |
| 15:44.35 | carrar | gonna get hot in a rack |
| 15:44.41 | carrar | need air flow |
| 15:47.48 | *** join/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 15:48.20 | *** join/#asterisk mintee (i=1000@72-165-177-67.dia.static.qwest.net) |
| 15:49.01 | mintee | there a decently easy application to create a informational extension? Like voicemail without the option to leave a message. |
| 15:49.19 | mintee | so they can call in, and change the message with a PIN |
| 15:50.11 | [TK]D-Fender | mintee: Change what message? |
| 15:50.26 | [TK]D-Fender | mintee: "informational extension"? HUH? |
| 15:50.52 | mintee | [TK]D-Fender: "Dial 103 to hear hours and store location" |
| 15:51.14 | [TK]D-Fender | mintee: this is IVR 101.... |
| 15:51.18 | mintee | which would be a Background(locationhours) |
| 15:51.23 | [TK]D-Fender | mintee: go read the book.... |
| 15:51.30 | mintee | I've read the book |
| 15:51.59 | *** join/#asterisk marv[work] (n=timr@router.asteriasgi.com) |
| 15:52.00 | leifmadsen | just sounds like dialplan... |
| 15:52.02 | [TK]D-Fender | mintee: then whats the mystery? "core show application record" <- |
| 15:52.02 | mintee | cut me a break, i was just asking if there was a premade application, or if I need to write an entire dialplan with PIN authencation |
| 15:52.24 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 15:52.41 | [TK]D-Fender | mintNo, there is no magical "IVR in 1 command" application. Thats insane. Just the layout of the dialplan alone tells you this. |
| 15:52.56 | leifmadsen | Would only be about 2-3 lines of dialplan |
| 15:53.06 | *** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose) |
| 15:53.10 | leifmadsen | utilize Playback() followed by Record() |
| 15:53.10 | [TK]D-Fender | mintee: exten => 123,1, Background(locationhours) <- there |
| 15:53.21 | leifmadsen | use Authenticate() if you need to perform authentication |
| 15:53.28 | [TK]D-Fender | leifmadsen: If you want to make a quick wayt o change the contents, yup |
| 15:53.28 | mintee | meh, to each their own. I'll just write an #include dialplan that I can call within whatever extensions |
| 15:53.45 | leifmadsen | I don't see how much simpler you're expecting that to be... |
| 15:53.48 | [TK]D-Fender | mintee: and "#include dialplan"? huh? |
| 15:53.49 | mintee | leifmadsen: well aware of the auth |
| 15:54.11 | mintee | [TK]D-Fender: nevermind, i'm sorry I've confused your shallow mind |
| 15:54.16 | leifmadsen | I don't think I quite understand the issue at hand then |
| 15:54.18 | mintee | again |
| 15:54.24 | mintee | i do, nevermind |
| 15:54.35 | mintee | sorry i bothered the 1ee7 of #asterisk |
| 15:54.35 | leifmadsen | perhaps some more information about the call flow you are expecting would be useful |
| 15:54.41 | [TK]D-Fender | mintee: shallow? Hardly. You are throwing around meaningless terms and explaining yourself quite poorly |
| 15:54.45 | leifmadsen | mintee: I'm attempting to help here... |
| 15:54.52 | leifmadsen | mintee: you're avoiding my questions though |
| 15:54.54 | mintee | you've already answered my questions |
| 15:55.07 | Slade- | wow this conversation is painful |
| 15:55.08 | leifmadsen | then why are we still arguing? |
| 15:55.18 | [TK]D-Fender | leifmadsen: res_masochism.so |
| 15:55.22 | eppigy | D: |
| 15:55.23 | mintee | cause of the douchbaggery that is #asterisk |
| 15:55.26 | leifmadsen | [TK]D-Fender: you're not helping |
| 15:55.32 | mintee | it's like an efnet channel |
| 15:55.34 | mintee | toots |
| 15:55.34 | leifmadsen | mintee: no, you mean [TK]D-Fender |
| 15:55.35 | *** part/#asterisk mintee (i=1000@72-165-177-67.dia.static.qwest.net) |
| 15:55.36 | [TK]D-Fender | leifmadsen: No really can. |
| 15:55.45 | [TK]D-Fender | no-one* |
| 15:56.00 | leifmadsen | goes back to something productive |
| 15:56.12 | eppigy | let jesus' love shine through you |
| 15:56.17 | leifmadsen | I can understand the frustration some people find in this channel |
| 15:56.24 | [TK]D-Fender | leifmadsen: He is anlother one of the "ask little pieces of a question with vague goals and invented terminology" types |
| 15:56.42 | leifmadsen | [TK]D-Fender: then help him to ask better questions instead of belittling him |
| 15:56.50 | [TK]D-Fender | leifmadsen: Your attempt fared about as well |
| 15:57.10 | [TK]D-Fender | leifmadsen: I asked him what he meant. He didn't explain, and then called me "shallow" |
| 15:57.26 | leifmadsen | your approach was confrontational |
| 15:57.36 | Slade- | you two are still at it? man what an effective troll he was |
| 15:57.52 | Blackvel | anyone using HTC TyTN II / Touch Pro / Touch Pro 2 with sip-client and asterisk? does this work as expected (sound quality, speakers, etc.)? I heared rumour that HTC's should not be used and only nokia works fine. |
| 15:57.54 | leifmadsen | Slade-: no, this conversation has happened several times between us |
| 15:58.08 | Slade- | leifmadsen: ah |
| 15:58.09 | [TK]D-Fender | leifmadsen: No, "HUH?" jsut expresses great confusion. Did not belittle him,. |
| 15:58.13 | [TK]D-Fender | Slade-: Highly :) |
| 15:58.49 | [TK]D-Fender | sigh |
| 16:00.05 | Slade- | i dunno. [TK]D-Fender has helped me on a few issues, i've always understood by the time the conversation was over. |
| 16:00.32 | eppigy | mintee was a troll |
| 16:00.41 | eppigy | the use of "toot" in hsi closing |
| 16:00.42 | eppigy | his |
| 16:00.47 | eppigy | makes that pretty evident |
| 16:00.51 | leifmadsen | Slade-: I never said [TK]D-Fender wasn't helpful |
| 16:01.03 | eppigy | I mean don't let the troll win |
| 16:01.07 | eppigy | lets all hold hands |
| 16:01.08 | leifmadsen | anyways, [TK]D-Fender have met in person, so no worries |
| 16:01.14 | eppigy | and focus on positive things |
| 16:01.25 | leifmadsen | EOL. |
| 16:01.52 | Slade- | goes back to porting phpagi |
| 16:03.12 | *** join/#asterisk layne (n=layne@ool-44c0048f.dyn.optonline.net) |
| 16:05.21 | *** join/#asterisk Defraz (n=T0tal@fw-poky.fuzecore.com) |
| 16:10.34 | Slade- | hey is fastagi not commonly in use? regular agi seems to be much more popular |
| 16:11.23 | Blackvel | Slade-: back from where? java? |
| 16:11.48 | Slade- | Blackvel: not sure i understand what you're asking |
| 16:12.48 | Blackvel | * Slade- goes back to porting phpagi |
| 16:12.59 | Slade- | back from the drama in channel :P |
| 16:13.06 | Blackvel | oh I see :) |
| 16:13.19 | Slade- | porting phpagi to php 5x |
| 16:14.50 | Slade- | from 4x |
| 16:19.35 | mort_gib | Why do I get missing closing parentis SIPAddHeader(Alert-Info:info=Direct-Nr;x-line-id=0):q! |
| 16:19.41 | mort_gib | *%"£$( |
| 16:19.46 | mort_gib | :x! |
| 16:20.09 | mort_gib | Not my day!! |
| 16:22.08 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 16:25.54 | carrar | mort_gib |
| 16:25.57 | carrar | <PROTECTED> |
| 16:26.10 | carrar | and remove the ":q!" :) |
| 16:27.01 | mort_gib | carrar: I just reinstalled my desktop, dual head, I USED to give focus to the window I had my mouse over |
| 16:27.31 | carrar | I do that |
| 16:27.38 | carrar | Know where you mouse is! |
| 16:27.41 | carrar | heh |
| 16:27.48 | mort_gib | SIPAddHeader(Alert-Info:info=Direct-Nr;x-line-id=0) won work, but SIPAddHeader(Alert-Info:info=Direct-Nr) will |
| 16:27.51 | mort_gib | Froody |
| 16:27.54 | carrar | You need to use Xeyes |
| 16:27.59 | carrar | that will help you |
| 16:28.06 | mort_gib | ?? |
| 16:28.18 | carrar | type it |
| 16:28.19 | *** join/#asterisk Toerkeium (i=Toerkeiu@201.216.206.221) |
| 16:28.21 | carrar | "Xeyes" |
| 16:28.32 | carrar | move it to the center of your work area |
| 16:29.15 | mort_gib | hang on |
| 16:30.30 | mort_gib | tuxeyes LOL |
| 16:40.09 | *** join/#asterisk lanning (n=lanning@173.8.187.197) |
| 16:46.08 | *** join/#asterisk drudge` (i=anonymou@unaffiliated/drudge/x-837452) |
| 16:48.54 | *** join/#asterisk wonderworld (n=w@ip-62-143-16-28.unitymediagroup.de) |
| 16:54.48 | *** join/#asterisk bbryant (n=brett@c-68-59-21-152.hsd1.sc.comcast.net) |
| 16:59.39 | *** part/#asterisk viraptor (n=viraptor@87-194-164-154.bethere.co.uk) |
| 17:03.17 | *** join/#asterisk batphone (n=wclayton@aus-pix-00-pat-01.corenap.com) |
| 17:03.35 | batphone | im getting a 604 back from CBeyond when trying to make outbound calls after moving the PBX to another PRI |
| 17:03.45 | batphone | it shows as being registered |
| 17:03.50 | batphone | i forget what causes this, its been so long |
| 17:04.14 | batphone | anyone know off the top of their head? if i rememeber, CBeyond has to do something on their end |
| 17:04.26 | batphone | i dont think there is anything i can do to * to make that work |
| 17:04.44 | batphone | this is a SIP 604 |
| 17:06.03 | ldsjohn | argh, if your around [TK]D-Fender im having the same problem with a bare bones config again. same error but this time I have a basic config |
| 17:06.12 | [TK]D-Fender | batphone: pastebin the entire output of the SIP communication. |
| 17:06.14 | [TK]D-Fender | ~pb |
| 17:06.15 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste , or , http://bin.cakephp.org/ |
| 17:06.16 | [TK]D-Fender | ^^^^^^ |
| 17:06.49 | ldsjohn | I am going to pastebin all my configs for you. I can't blame the GUI this time |
| 17:10.08 | ldsjohn | http://pastebin.ca/1452057 thats my sip.conf followed by my extensions.conf, I have a barebone modules.conf ( just turns on autoload, and a logging.conf that just turns on logging, and thats all I have ) |
| 17:10.41 | [TK]D-Fender | ldsjohn: And I don't see you showing the error in there |
| 17:11.11 | ldsjohn | I am getting the same chan_sip.c:14781 handle_request_invite: Call from '' to extension '17772484176' rejected because extension not found. error |
| 17:11.33 | ldsjohn | I can paste a sip debug, it looks identical to last nights |
| 17:12.20 | ldsjohn | except instead of looking for s its looking for 17772484176 |
| 17:12.33 | [TK]D-Fender | ldsjohn: SIP DEBUG |
| 17:12.46 | *** join/#asterisk tharrison (n=chatzill@70.88.150.241) |
| 17:12.51 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 17:13.53 | ldsjohn | http://pastebin.ca/1452065 |
| 17:13.54 | [TK]D-Fender | ldsjohn: And indeed there is no match for that number |
| 17:14.15 | [TK]D-Fender | ldsjohn: Looking for 17772484176 in from-callcentric (domain 192.168.42.50) SIP/2.0 404 Not Found |
| 17:14.31 | [TK]D-Fender | ldsjohn: its telling you what & where, and you have no match. go fix this. |
| 17:14.46 | tharrison | Can someone help me with my google foo? I'm trying to find info on connecting two sites, each with an asterisk server, but I'm only find simple directions on setting up a trunk between the two. I need some info on joining the two sites a little more tightly. |
| 17:15.06 | ldsjohn | so shoudl I change my register string is that the secret? the register string shoudl say /s instead of /17772484176 |
| 17:15.25 | tharrison | (ie. sharing voice mail, simultaneous ringing, etc) |
| 17:15.44 | [TK]D-Fender | ldsjohn: What should it be "secret"? |
| 17:15.45 | drudge` | hey, what does your school or work use for spam filtering? i need to find some solution |
| 17:16.11 | ldsjohn | it just says to put my phone number there in callcentrics instructions |
| 17:16.23 | *** join/#asterisk jackal (n=jackal@pool-72-73-21-209.clppva.fios.verizon.net) |
| 17:16.35 | [TK]D-Fender | ldsjohn: yes, and your register is fine, you have no exten to MATCH it however. |
| 17:18.00 | ldsjohn | changing the s in my extension from s to my 1777 number fixed it, incoming calls work |
| 17:18.17 | ldsjohn | of course you knew that |
| 17:20.07 | *** join/#asterisk rue_mohr (n=rue@24.207.122.10) |
| 17:20.32 | ldsjohn | now I just have to configure my pap2 and my polycom phone and im golden |
| 17:21.02 | rue_mohr | I'd like to know how your pap2 audio goes |
| 17:21.08 | rue_mohr | echo and volume wise |
| 17:21.30 | rue_mohr | oh wait no, the pap2 isn't co, its station... |
| 17:22.42 | *** part/#asterisk drudge` (i=anonymou@unaffiliated/drudge/x-837452) |
| 17:22.58 | *** join/#asterisk mercutioviz (n=michaelc@freeswitch/developer/msc) |
| 17:24.42 | *** part/#asterisk ramindia (n=Administ@202.63.96.10) |
| 17:29.10 | *** join/#asterisk ayeso (n=chatzill@ext-52.sagetelecom.net) |
| 17:29.23 | *** join/#asterisk smultron (n=smultron@cpe-67-9-150-163.austin.res.rr.com) |
| 17:30.39 | ayeso | I am having trouble with a .call file, when i initiate the call file by moving it to the outgoing directory, it places the outbound call. when its answered it connects it to the correct context, in the context i use the "read" application. it is not reading dtmf for some reason. but if i call directly rather than using a call file it works fine. any ideas? |
| 17:32.43 | *** join/#asterisk arpu (n=arpu@chello080109017107.12.14.vie.surfer.at) |
| 17:32.59 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 17:34.55 | ayeso | i changed the peer in sip.conf to dtmf=inband and it works now.. really weird but whatever. |
| 17:38.36 | *** join/#asterisk ttl- (n=patrick@d5153AE82.access.telenet.be) |
| 17:40.45 | *** join/#asterisk stijnbe (n=stijnbe@78-21-61-204.access.telenet.be) |
| 17:43.54 | *** join/#asterisk p4tr0p1 (n=jam@201.34.141.194) |
| 17:53.55 | rue_mohr | hmm I have two different types of alert-info I need to send different phones at the same time, can I specify more than one alt-info and the other phones will just ignore what they dont understand? |
| 17:57.40 | rue_mohr | see, the aastra phones need a xml blurb and the polycom ones take a text string, techcially I suppose I can make the polycoms' text string a xml blob |
| 18:01.04 | *** join/#asterisk n3hxs (n=HAMming@lax-wan-a-47.lax.dsl.cerfnet.com) |
| 18:02.19 | *** join/#asterisk Victor_Yure (n=victor@unaffiliated/victoryure/x-837844) |
| 18:03.40 | kaldemar | rue_mohr: fork the call to two local channels that set different headers and dial |
| 18:05.49 | rue_mohr | hu? |
| 18:06.05 | rue_mohr | its a.....ring group... |
| 18:06.06 | batphone | [TK]D-Fender: looks like it got fixed on CBeyond's end |
| 18:06.36 | batphone | sorry i didnt get that pastebin up. it was just a standard SIP604 message in the debug. |
| 18:06.51 | rue_mohr | exten => s,n,Dial(${dDENNIS}&${dVAL}&${dCHARLES}&${dALANA}&${dJAN}); |
| 18:06.59 | rue_mohr | so you saying I can split it? |
| 18:07.21 | kaldemar | Dial(Local/polycom@context&Local/aastra@context) => polycom,1,SIPAddHeader, => polycom,n,Dial(... |
| 18:07.32 | kaldemar | get what i mean? |
| 18:07.40 | [TK]D-Fender | No-one ever thinks of chan_local properly... |
| 18:08.12 | *** join/#asterisk gert (i=gert@vps202.sbp.mihos.nl) |
| 18:08.36 | rob0 | is having improper thoughts about chan_local |
| 18:08.42 | jaytee | No one ever expects the Spanish Inquisition |
| 18:08.45 | rob0 | is off for a cold shower now |
| 18:11.58 | *** join/#asterisk techie33 (n=hanford2@209.234.197.86) |
| 18:12.28 | techie33 | Hello all, I'm having an issue with a 7970 can anyone help? |
| 18:12.40 | iksik | morning |
| 18:13.11 | techie33 | how are you iksik |
| 18:13.13 | iksik | any ideas why oh323 is trying to connect with GK on all IPs i've gor? :| listenAddr is set only for 1 |
| 18:13.22 | iksik | very tierd.. ;/ |
| 18:13.37 | iksik | s/gor/got |
| 18:13.56 | techie33 | Ya, me too! I just setup an asterisk box and been pulling my hair out w/ cisco phones! LOL |
| 18:14.09 | techie33 | the 7940's work great but the 7970's sux! |
| 18:14.10 | techie33 | lol |
| 18:14.51 | iksik | ;] |
| 18:15.19 | _Steve_ | polycom > * |
| 18:15.27 | iksik | I'm trying to setup that GK account from few days now, and no results ;/ |
| 18:16.11 | techie33 | what's a GK account? |
| 18:16.19 | iksik | gatekeeper |
| 18:17.20 | iksik | from asterisk to voipswitch... and now i've got that connection, and attempt to discover GK account on voipswitch, but VS is set only for 1 of my IPs, and only for 1 port, but oh323 is trying to connect on all IPs, and on few other ports then 1720 :| |
| 18:17.58 | techie33 | humm not familliar w/ GK sorry. |
| 18:18.02 | iksik | [Jun 8 20:16:25] WARNING[22767]: chan_oh323.c:4265 oh323_gk_check: Failed to register with gatekeeper 'XXX.XXX.XXX.XXX'. |
| 18:18.03 | iksik | ;/ |
| 18:19.54 | beek | wants to buy an shrubbery |
| 18:21.46 | ariel_ | I think I broke my Cisco 3645.... it's e1 port is flashing argh.... |
| 18:21.50 | [TK]D-Fender | beek: and a NICE one this time! |
| 18:22.26 | beek | It'll go along nicely with my purloined lupens |
| 18:23.41 | ariel_ | goes to make a loop back rj45 cable to test the te220b and the cisco to make sure they both are ok.... |
| 18:27.51 | *** join/#asterisk b14ck (n=blacky@cpe-98-151-210-28.socal.res.rr.com) |
| 18:27.51 | *** join/#asterisk zamba (i=marius@flage.org) |
| 18:27.51 | *** join/#asterisk [netman] (n=netman@121.Red-83-45-37.dynamicIP.rima-tde.net) |
| 18:27.51 | *** join/#asterisk VaGoNeTaS (n=debian@xen.datapartner.cl) |
| 18:27.51 | *** join/#asterisk BobPierce (n=BobPierc@216.36.132.162) |
| 18:27.51 | *** join/#asterisk sharp (n=sharp@sauropod.org) |
| 18:29.47 | *** join/#asterisk s14ck (n=s14ck@190.38.214.204) |
| 18:31.31 | gert | hi all. I'm having a problem with 'transmit_silence_during_record'. It works okay in the dialplan, but it won't transmit silence when recording from an AGI call... is that a bug or a feature? |
| 18:32.34 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 18:37.59 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 18:39.26 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 18:43.56 | *** join/#asterisk hfb (n=hfb@pool-96-247-49-46.lsanca.dsl-w.verizon.net) |
| 18:44.02 | *** join/#asterisk elitecoder (n=liq@apollo.bullethost.com) |
| 18:45.00 | elitecoder | I need to detect if someone hungup during a Background. Anyone know how I could do that? |
| 18:46.16 | techie33 | what does it mean when a sip peer is unreachable? |
| 18:46.34 | techie33 | Name/username Host Dyn Nat ACL Port Status |
| 18:46.34 | techie33 | npgw01 172.1.2.254 5060 Unmonitored |
| 18:46.34 | techie33 | 4377/4377 10.40.2.54 D A 5060 OK (163 ms) |
| 18:46.34 | techie33 | 4372/4372 172.1.2.9 D A 5060 OK (86 ms) |
| 18:46.34 | techie33 | 4371/4371 172.1.2.15 D A 5060 UNREACHABLE |
| 18:46.34 | techie33 | 4370/7b452e87-4496-4762-e 172.1.2.17 D A 5060 UNREACHABLE |
| 18:46.42 | techie33 | 4360/4360 10.40.2.58 D A 5060 OK (145 ms) |
| 18:46.42 | techie33 | 6 sip peers [Monitored: 3 online, 2 offline Unmonitored: 1 online, 0 offline] |
| 18:47.18 | [TK]D-Fender | tehcPASTEBIN, do not spam in here |
| 18:47.20 | [TK]D-Fender | ~pb |
| 18:47.21 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste , or , http://bin.cakephp.org/ |
| 18:47.35 | [TK]D-Fender | techie33: Amd that means it failed a qualify check |
| 18:47.36 | *** join/#asterisk s14ck (n=s14ck@190.38.214.204) |
| 18:49.37 | *** join/#asterisk dajhorn (n=dajhorn@206.16.96.160) |
| 18:50.04 | eppigy | are you purposefully using public ip addresses? |
| 18:50.32 | *** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose) |
| 18:50.42 | eppigy | DONDE ESTA |
| 18:50.58 | KavanS | lol |
| 18:52.49 | elitecoder | YEAH DONDE MOFOS |
| 18:53.22 | eppigy | NO NECESITO |
| 18:53.25 | eppigy | TRABAJO |
| 18:53.38 | elitecoder | I need to detect if someone hungup during a Background. Anyone know how I could do that? |
| 18:54.00 | techie33 | what affect does setting qualify=no have by chance? |
| 18:55.01 | *** join/#asterisk Geminizer (n=whoami@cpe-74-77-1-168.buffalo.res.rr.com) |
| 18:55.16 | *** join/#asterisk xbmodder_ (n=Sargun@atarack/Staff/Sargun) |
| 18:55.58 | Aiatek | <techie33> unreachable its = not online or not network conectivity |
| 18:57.29 | Geminizer | hello, guys. Would anyone know how I would go about setting the caller id for an extension? |
| 18:57.50 | elitecoder | Geminizer: Using the SET command before it's passed to them, in your dialplan |
| 18:58.13 | elitecoder | Geminizer: http://www.voip-info.org/wiki/view/Setting+Callerid |
| 18:58.41 | elitecoder | before DIAL'ing a phone, just set the caller id |
| 18:59.19 | [TK]D-Fender | techie33: with "no" * will not check up on your hosts. |
| 18:59.19 | p4tr0p1 | hi folks! Anybody knows if it's any bug related about monitor recording only one side of a conversation, but only sometimes? |
| 19:00.16 | elitecoder | Isn't there a general way to check whether an application completed? |
| 19:00.40 | [TK]D-Fender | elitecoder: As in? |
| 19:00.57 | elitecoder | Well I noticed in the CLI it says Application completed returning 0 |
| 19:00.58 | elitecoder | and stuff |
| 19:01.11 | elitecoder | I don't remember if it does it for background but i thought there was a way to check it |
| 19:01.20 | [TK]D-Fender | elitecoder: Sounds like you already have an answer there |
| 19:01.40 | elitecoder | What kinds of commands can i use to test it? |
| 19:01.55 | [TK]D-Fender | elitecoder: I believe it spols it up and exits immediately jsut like any other app |
| 19:02.01 | [TK]D-Fender | etilWhat it your actual goal? |
| 19:02.15 | elitecoder | I just want to know if they hang up in the middle of the message |
| 19:02.25 | elitecoder | actually, at all, in a certain extension |
| 19:03.31 | [TK]D-Fender | elitecoder: Seriously doubt that possible |
| 19:03.53 | elitecoder | heh |
| 19:04.03 | [TK]D-Fender | elitecoder: What would you do with this information? |
| 19:04.21 | elitecoder | Log it, and if reliable enough use it to determine the current # of calls from a dialer script |
| 19:05.12 | [TK]D-Fender | elitecoder: detecting its it happens to occur during a background doesn't seem to relate to keeping count of anything. Ether the channel is up or it isn't. |
| 19:05.38 | elitecoder | sigh |
| 19:07.08 | elitecoder | then how do I tell when the channel goes down |
| 19:07.10 | elitecoder | same thing no? |
| 19:07.10 | elitecoder | lol |
| 19:07.16 | *** join/#asterisk websae (i=6158f542@gateway/web/ajax/mibbit.com/x-7529c96e6f2fa86b) |
| 19:07.36 | elitecoder | hmm ;x |
| 19:07.37 | [TK]D-Fender | elitecoder: Its also a question of how you want to do this. |
| 19:07.41 | elitecoder | http://www.voip-info.org/wiki/view/Asterisk+config+extensions.conf |
| 19:07.45 | elitecoder | predefined extension names |
| 19:08.05 | [TK]D-Fender | elitecoder: AMI sends out hangup notices, etc. |
| 19:08.16 | [TK]D-Fender | elitecoder: You can also do channel dumps via it, etc |
| 19:08.29 | elitecoder | Yeah I'm not going to use that unless I have to. No more time to mess with this heh |
| 19:09.09 | elitecoder | this looks right |
| 19:09.09 | elitecoder | http://www.voip-info.org/wiki/view/Asterisk+h+extension |
| 19:09.24 | [TK]D-Fender | elitecoder: not always reliable, but an option. |
| 19:09.36 | *** part/#asterisk websae (i=6158f542@gateway/web/ajax/mibbit.com/x-7529c96e6f2fa86b) |
| 19:09.50 | elitecoder | any idea when it's not? |
| 19:11.53 | elitecoder | no problem |
| 19:11.54 | [TK]D-Fender | elitecoder: not really. Another thing is you'll need to make sure its in every context that the call could be in at any given point in time |
| 19:12.04 | *** join/#asterisk jicksta (n=jicksta@67.164.0.78) |
| 19:12.06 | elitecoder | Alright |
| 19:13.43 | *** join/#asterisk stijnbe (n=stijnbe@78-22-110-114.access.telenet.be) |
| 19:14.14 | elitecoder | Can I simply exten => h, 1, AGI(...) ? or do I need to hang up after? |
| 19:15.28 | *** join/#asterisk Defraz (n=T0tal@24-117-236-174.cpe.cableone.net) |
| 19:15.56 | timeshell_atwork | What's the best analog to voip phone adapter? |
| 19:16.29 | rue_mohr | I dont actually think its digium |
| 19:16.33 | rue_mohr | but its all I'v tried |
| 19:16.44 | [TK]D-Fender | timeshell_atwork: Mediatrix |
| 19:16.48 | rue_mohr | accept for my mainstreet channelbank, which has troubles too |
| 19:17.05 | [TK]D-Fender | timeshell_atwork: as far as 1-2 port is ocncerned |
| 19:17.05 | *** join/#asterisk mltlnx (n=mltlnx@asams.mserve.com) |
| 19:17.20 | timeshell_atwork | TK : Thank you |
| 19:17.57 | [TK]D-Fender | timeshell_atwork: the nediatrix ones (higher end) supports transparent DHCP, etc and some other wild stuff, T.38, dual G.729, etc |
| 19:18.21 | timeshell_atwork | Just looking for a couple ports fornow |
| 19:18.28 | [TK]D-Fender | timeshell_atwork: How many>? |
| 19:18.37 | [TK]D-Fender | timeshell_atwork: located where relative to your server? |
| 19:18.38 | timeshell_atwork | A couple = 2 |
| 19:18.44 | timeshell_atwork | Same network. |
| 19:18.53 | timeshell_atwork | Gigabit fiber link |
| 19:19.24 | timeshell_atwork | Different physical office |
| 19:19.28 | [TK]D-Fender | timeshell_atwork: Linsys PAP2 or SPA-2102 is jsut fine then |
| 19:19.44 | rue_mohr | ok, so lets see, if I want to take caller id and change 555-1234 to 555-4321.... |
| 19:20.00 | timeshell_atwork | TK: I have PAP2's, but occasionally they them seem to act up and give garbage over the line. |
| 19:20.09 | [TK]D-Fender | timeshell_atwork: I'd suggest Mediatrix for trickier CPE requirements, but its overkill for most. |
| 19:20.22 | rue_mohr | timeshell_atwork, OOoooOOo... |
| 19:20.25 | timeshell_atwork | I want something that's simple, but solid. |
| 19:20.34 | *** join/#asterisk korihor (n=korihor@190.72.254.245) |
| 19:20.38 | rue_mohr | for office or stations? |
| 19:20.46 | rue_mohr | (fxo or fxs) |
| 19:20.48 | [TK]D-Fender | timeshell_atwork: Go with the 2102 then |
| 19:21.21 | iksik | hmm |
| 19:21.34 | iksik | Remote host can't match request NOTIFY to call '37c928ef4fbcf30248fa210c5fb1cd7d@XX.XXX.XXX'. Giving up - what does it mean? |
| 19:21.46 | iksik | this 37c928ef.... thing :| |
| 19:23.55 | *** join/#asterisk Whitor (n=Whitor@64.128.237.124) |
| 19:24.29 | timeshell_atwork | TK : Yah, the price seems ok on that. Thanks |
| 19:25.06 | *** join/#asterisk tamiel (n=tamiel@ip-120.net-81-220-18.versailles.rev.numericable.fr) |
| 19:26.47 | elitecoder | This isn't working for me and I can't find any examples online. Background(Message1, m,, extension_context) |
| 19:26.50 | timeshell_atwork | TK : I presume that it works on a LAN as well as acting as a residential gateway type device. It will not be used as a pppoe device. |
| 19:27.01 | elitecoder | It plays the message but when I dial an extension from extension_contexts nothing happens. |
| 19:30.25 | [TK]D-Fender | timeshell_atwork: It can do whatever you require |
| 19:31.02 | [TK]D-Fender | elitecoder: "core show application background", and do not add extra white-space |
| 19:33.16 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 19:34.01 | *** join/#asterisk GrantA (n=email@213.22.165.84) |
| 19:34.39 | *** join/#asterisk jtexter3 (n=jtexter3@72.242.229.213) |
| 19:35.05 | elitecoder | sigh |
| 19:35.26 | elitecoder | I've already read that like 3x |
| 19:36.45 | elitecoder | ill just keep trying |
| 19:37.04 | GrantA | does Yahoo, and Gmail neccessarily delete emails they think is spam, or dont they just put it in your spam folder |
| 19:37.36 | rob0 | TTBOMK only Hotmail is so irresponsible as to silently discard mail. |
| 19:37.54 | rob0 | hotmail/msn and their other domains |
| 19:38.25 | *** join/#asterisk af_ (n=getsmart@88-149-240-107.dynamic.ngi.it) |
| 19:38.27 | rob0 | oh hmmm, this isn't an email-related channel :) |
| 19:38.37 | Geminizer | elitecoder: is there a way to set up caller id mappings to extensions using the asterisk mysql database, as opposed to going through the dialplan config file? |
| 19:40.27 | elitecoder | Geminizer: I don't know. |
| 19:40.57 | timeshell_atwork | Can anyone comment on the Polycom SpectraLink WiFi phones? |
| 19:42.32 | Geminizer | elitecode: ok, thank you for your help :) |
| 19:43.39 | elitecoder | Looks like my only issue was a space before the context name |
| 19:43.44 | elitecoder | now it works right |
| 19:45.58 | SuPrSluG | Geminizer: I think that caller id mapping would be an entry in the sip database. |
| 19:47.28 | SuPrSluG | Geminizer: like `callerid` varchar(80) default NULL, |
| 19:53.38 | timeshell_atwork | What needs to be set to ensure that extensions on either side of 2 asterisk servers calling each other display each others actual extensions rather than the extension used to connect the asterisk servers? |
| 19:56.48 | *** join/#asterisk j_kroon (n=jkroon@dsl-240-169-134.telkomadsl.co.za) |
| 19:56.56 | ariel_ | timeshell_atwork: I have setup over 360 of those phones. |
| 19:56.56 | justdave | is there some trick to loading a new bootrom on a Polycom SoundPoint IP 4000 ? |
| 19:57.15 | justdave | according to Polycom's site, the current version certified for the IP4000 is 4.1.2revB |
| 19:57.20 | justdave | which is what we have on the tftp server |
| 19:57.23 | timeshell_atwork | ariel_ Ok.... and? |
| 19:57.30 | *** join/#asterisk werdan7 (n=w7@freenode/staff/wikimedia.werdan7) |
| 19:57.31 | justdave | all the IP501s and IP6000s picked it up fine |
| 19:57.53 | ariel_ | They work very well, but are a pain to configure |
| 19:57.53 | justdave | the IP4000s are dying with a "BootROM has changed. Error 0x00" on the screen and continuously rebooting. |
| 19:58.10 | timeshell_atwork | ariel_ I would like to know if they are reliable. I would like to know if they skip between 2 access points with the same SSID seemlessly. |
| 19:58.47 | timeshell_atwork | Do they autodetect open access points and do they ever die or lock up like the UTStarcom's F3000's |
| 19:58.55 | ariel_ | There very reliable, and there the only ones I have found that skip between 2 access points without dropping the call. |
| 19:59.02 | justdave | can't find any mention of that error/message in any of Polycom's docs |
| 19:59.42 | timeshell_atwork | ariel_ A particular model # that is recommended? |
| 20:00.08 | ariel_ | We set them up on large cruise ships for deck phones for the crew. And they travel between 5 to 9 access points. They have to use there SVP server to work correctly. We use the 8020's and the 8030's |
| 20:00.29 | timeshell_atwork | SVP server? |
| 20:01.08 | [TK]D-Fender | elitecoder: As I said... no extra white-space. Keep that in mind elswhere |
| 20:01.30 | elitecoder | you mentioned that? lol sorry |
| 20:01.40 | elitecoder | I read it in a book, thats how i remembered to try that |
| 20:02.10 | ariel_ | timeshell_atwork: reason they work so well is that they use an SVP server that registers them and service there services via QoS. |
| 20:02.57 | *** join/#asterisk Nine5102 (n=Aleksi@192.80-203-96.nextgentel.com) |
| 20:03.02 | [TK]D-Fender | [15:31]<[TK]D-Fender>elitecoder: "core show application background", and do not add extra white-space |
| 20:03.08 | ariel_ | timeshell_atwork: asterisk = sip.conf normal setting. = SVP server = keeps license for phones and registers them between the SVP and the phones. |
| 20:03.34 | j_kroon | hi, if i let a phone to one of my providers ring "indefinitely" the provider will after a time kill the call with a SIP/480 and cause code 19, which is NO ANSWER. Is there a way I can get the CDR to say NOANSWER instead of FAILED? |
| 20:04.58 | timeshell_atwork | ariel_ Ok. So how well would they work without this SVP server? |
| 20:05.15 | elitecoder | ty ty |
| 20:05.15 | ariel_ | I don't know if they started selling them without it yet |
| 20:05.27 | timeshell_atwork | Eh?? |
| 20:06.35 | ariel_ | timeshell_atwork: last batch we got it's a packaged deal. Each SVP does up to 80 phones, but it also has a really strange upgrade path as it takes 5 SVP's to get to 200 phones |
| 20:06.39 | timeshell_atwork | ariel_ Are we talking 802.11b/g phones? |
| 20:07.16 | ariel_ | Polycom Spectralink 8020's and 8030's yes |
| 20:07.28 | ariel_ | I am looking at one in my hand right now. |
| 20:07.31 | *** join/#asterisk muiro (n=muiro@cpe-173-89-177-15.neo.res.rr.com) |
| 20:07.39 | timeshell_atwork | Why wouldn't they be sold without the SVP? |
| 20:07.47 | timeshell_atwork | I'm only looking at 2 or 3 for our office. |
| 20:07.57 | timeshell_atwork | We probably would have no need for QoS |
| 20:07.57 | ariel_ | there is even a youtube of some people really trying to destroy the phone |
| 20:08.13 | muiro | Can I use any kind of a wildcard with Bridge()? |
| 20:08.22 | ariel_ | timeshell_atwork: licensing |
| 20:08.31 | timeshell_atwork | Licensing what? |
| 20:08.38 | *** join/#asterisk cc6 (n=cc6@pool-72-91-71-74.tampfl.fios.verizon.net) |
| 20:09.02 | ariel_ | the license for the phone is tied to the SVP right now. |
| 20:09.15 | Geminizer | SuPrSluG: all the callerid entries in asterisk.sip follow the form "device <ext_num>" --- I changed one of them, and still having no luck |
| 20:09.56 | SuPrSluG | Geminizer: are you trying to do realtime type config? |
| 20:11.39 | Geminizer | I'm not too sure what that means... I am very new to the whole "asterisk" realm -- currently, when I dial out, the caller id on the recipient's end shows the extension, when I want it to show a toll free number |
| 20:11.56 | timeshell_atwork | ariel_ http://www.voipsupply.com/spectralink-8002-scb |
| 20:12.15 | timeshell_atwork | ariel_ http://www.voipsupply.com/ip-phones/wi-fi |
| 20:12.27 | *** join/#asterisk phr3ak (n=nnnphr3a@gnet.hu) |
| 20:13.02 | *** join/#asterisk lanning (n=lanning@nat/yahoo/x-b51d9ff0b5207b2c) |
| 20:13.32 | cc6 | can anyone help me with a new AA50 and Poly IP330 |
| 20:13.55 | ariel_ | timeshell_atwork: I have not used those. I have only used these from them: http://www.voipsupply.com/catalogsearch/result/?q=Spectralink+8020&manufacturers=http%3A%2F%2Fwww.voipsupply.com%2F |
| 20:14.24 | timeshell_atwork | But it appears they are made to be Wifi only sans "SVP" |
| 20:16.52 | cc6 | can anyone help me with a new AA50 and Poly IP330 |
| 20:17.20 | cc6 | the phones wont register even using the auto provisioning tool |
| 20:17.25 | *** join/#asterisk Chuggs (n=Chuggs@s142-179-186-158.ab.hsia.telus.net) |
| 20:18.04 | ariel_ | timeshell_atwork: The 8020's that we have use are wifi phones, but they need the SVP Server box. The 8002 I have not used. You might have to give them a call to make sure it's a real stand along phone. |
| 20:18.21 | timeshell_atwork | Yah |
| 20:18.23 | timeshell_atwork | I'd do that. |
| 20:18.35 | timeshell_atwork | By the desc, it looks like it is. |
| 20:18.36 | Chuggs | is it typically a big deal when getting a PRI installed, time and contract wise? |
| 20:18.39 | ariel_ | timeshell_atwork: and they only update firmware and a few settings via ftp/tftp. Rest has to be done from the phone it self. |
| 20:18.57 | timeshell_atwork | ariel_ That would be expected and actually desired in this case. |
| 20:22.51 | [TK]D-Fender | checkout time, BBIAB |
| 20:23.38 | cc6 | can anyone help me with a new AA50 and Poly IP330 |
| 20:23.41 | cc6 | the phones wont register even using the auto provisioning tool |
| 20:25.53 | *** join/#asterisk ruben23 (n=AGENT@124.107.3.178) |
| 20:26.06 | ruben23 | hi |
| 20:27.44 | ariel_ | cc6: I know how to setup the IP330 but I have never used the auto provisioning tool. The AA50 has full Digium support you can give them a call. |
| 20:28.48 | cc6 | ariel_ I have contacted them and they are responsive to my time line needs |
| 20:29.20 | muiro | Can Bridge() accept wildcards? failing that, can anyone suggest another method to programatically connect bridge to various channels? |
| 20:30.02 | *** part/#asterisk jtexter3 (n=jtexter3@72.242.229.213) |
| 20:31.12 | cc6 | i need to do a simple proof of concept with one phone. We already have the trunks coming in and it routes to VM ok but it just wont ring the phone |
| 20:32.47 | *** join/#asterisk bbryant (n=brett@c-68-59-21-152.hsd1.sc.comcast.net) |
| 20:35.06 | ariel_ | cc6: what error are you getting on the cli when you dial the phone? Is the phone showing up as registered? |
| 20:35.43 | cc6 | i am not seeing any error. The phone says "Line 1 not registered" |
| 20:36.33 | ariel_ | cc6: are you doing everything via the asterisk-gui? |
| 20:36.41 | cc6 | yes |
| 20:37.49 | ariel_ | The folks at #asterisk-gui might be able to help a bit more then here. But the polycom get there setup from either a tftp, ftp or even a http setup. |
| 20:38.30 | ariel_ | they have some config files that need to be setup. And I also know that the asterisk-gui puts just about everything in the user.conf file. |
| 20:38.36 | cc6 | yes I have tried both a TFTP on a seperate server and the built in with the same results. |
| 20:39.07 | cc6 | it can dial the VM with no issues just will not register to the system |
| 20:39.40 | ariel_ | are you sure the firewall is not blocking sip registration? |
| 20:40.20 | ariel_ | have you tried to register a soft phone? like Xlite |
| 20:40.23 | cc6 | the aa50 and phone are in the same room on the same switch in the same subnet |
| 20:40.37 | cc6 | I have and xlite has the same issues as the IP330 |
| 20:41.21 | ariel_ | are you aware that the aa50 is running linux and also has I think ip_tables running (Firewall). |
| 20:42.53 | cc6 | yes I am aware but do not know which options to change in the gui. |
| 20:44.06 | ariel_ | I wonder how close that gui is to the one I have with the ABE setups. |
| 20:44.42 | cc6 | it looks pretty close to asterisknow gui |
| 20:45.09 | cc6 | i have ssh connectivity but I do not know the system so I havent played around in there |
| 20:46.38 | ariel_ | once again the folks at #asterisk-gui might be able to help you out. Switch to that chat room and ask them. |
| 20:46.48 | cc6 | k thanks |
| 20:46.49 | cc6 | :) |
| 20:47.36 | *** join/#asterisk s14ck (n=s14ck@190.38.214.204) |
| 20:49.15 | iksik | anyone can help me with incoming calls configuration? :/ |
| 20:49.50 | SuPrSluG | what's the issue? |
| 20:50.43 | *** join/#asterisk ziram19 (n=chatzill@41.226.177.171) |
| 20:58.15 | ziram19 | did someone has any solution for echo that happens even a call is established without that the other side answer? |
| 20:58.17 | ziram19 | is the hardware solution works for this kind of echo? |
| 20:58.46 | *** part/#asterisk a1fa (n=a1fa@unaffiliated/a1fa) |
| 20:59.11 | ziram19 | is there any one? |
| 20:59.29 | *** join/#asterisk icyValk77 (n=icyValk7@host217-43-31-121.range217-43.btcentralplus.com) |
| 21:00.30 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194) |
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| 21:04.30 | *** mode/#asterisk [+o jsmith] by ChanServ |
| 21:05.44 | *** part/#asterisk icyValk77 (n=icyValk7@host217-43-31-121.range217-43.btcentralplus.com) |
| 21:08.54 | *** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose) |
| 21:13.33 | *** join/#asterisk Firass-z0r (n=asadf@c-67-201-205-34.reshall.wwu.edu) |
| 21:13.48 | ziram19 | Allo? |
| 21:14.41 | *** join/#asterisk vvuja (n=vvuja@79-175-89-243.adsl-a-1.sezampro.yu) |
| 21:14.58 | *** join/#asterisk academy (n=adam@unaffilated/academy) |
| 21:15.23 | *** part/#asterisk vvuja (n=vvuja@79-175-89-243.adsl-a-1.sezampro.yu) |
| 21:15.45 | academy | I have Asterisk installed with DAHDI and a TDM400P. Outgoing calls work perfectly, but DAHDI doesn't seem to pickup incoming calls. What's the best way to debug this? |
| 21:16.06 | academy | if I do 'dahdi show channels', it lists my two fxo ports |
| 21:16.34 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 21:16.47 | [TK]D-Fender | academy: pastebin your chan-dahdi.conf & dialplan |
| 21:16.51 | *** join/#asterisk jsgoecke (n=Adium@c-71-202-25-141.hsd1.ca.comcast.net) |
| 21:16.52 | jsgoecke | Hola |
| 21:16.54 | [TK]D-Fender | ~pb |
| 21:16.55 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste , or , http://bin.cakephp.org/ |
| 21:17.00 | jsgoecke | Did something change in the installation scripts of v1.4.25? |
| 21:17.25 | jsgoecke | Now on my OSX/Leopard, suddenly Asterisk is install configs in /usr/local/etc/asterisk and the executable is not in my path |
| 21:17.32 | jsgoecke | Use to install just like on Linux... |
| 21:17.48 | wonderworld | how would i list dahdi channels that are in use atm? not all available... |
| 21:18.11 | *** join/#asterisk Greek-Boy (n=greek@41.222.89.77) |
| 21:18.59 | Greek-Boy | anybody know why func_odbc.so would cause a segmentation fault on asterisk? |
| 21:19.13 | ariel_ | core show channels |
| 21:21.37 | *** join/#asterisk drfreeze (n=Jim@207.191.114.82) |
| 21:21.39 | drfreeze | Hello |
| 21:22.12 | drfreeze | I've been searching the wiki for information on how I can compute the time on hold for a caller from the log files |
| 21:22.25 | drfreeze | So far I haven't found anything that will let me do that |
| 21:22.48 | drfreeze | Looking at the possible options: http://www.voip-info.org/wiki/view/Asterisk+cdr+csv |
| 21:23.36 | drfreeze | but don't know if MOH can be determined from lastapp or lastdata |
| 21:24.05 | drfreeze | Anyone ever try this? |
| 21:24.53 | *** join/#asterisk Defraz (n=T0tal@fw-poky.fuzecore.com) |
| 21:25.04 | ariel_ | drfreeze: your going to have to find all the records for the call you need to add up the time. As a call may have multiple records |
| 21:25.09 | jsmith | drfreeze: I'm afraid that wouldn't be possible with the current CDR engine... maybe with CEL you'll have that level of granularity |
| 21:25.31 | academy | I have Asterisk installed with DAHDI and a TDM400P. Outgoing calls work perfectly, but DAHDI doesn't seem to pickup incoming calls. What's the best way |
| 21:25.33 | academy | +to debug this? |
| 21:25.35 | ariel_ | what is cel? |
| 21:25.58 | jsmith | Call Event Log |
| 21:26.13 | ariel_ | academy: they posted a response for you to use pastebin your info |
| 21:26.15 | drfreeze | academy: thx |
| 21:26.17 | *** join/#asterisk mltlnx (n=mltlnx@asams.mserve.com) |
| 21:26.20 | jsmith | ... a more detailed call log than the current CDR engine, which is currently under development |
| 21:26.29 | ariel_ | but for starter what does the CLI say about your incoming call? |
| 21:27.17 | ariel_ | jsmith: nice. So there going to tie all the records up with a unique ID that belongs to the call and not the record? |
| 21:27.17 | academy | ahh, was that for me? ok sorry. I get nothing at all in the cli when a call comes in. One sec. |
| 21:27.37 | jsmith | ariel_: Yup... that's my understanding |
| 21:27.52 | ariel_ | academy: are you sending them to a context with a exten => s,1,answer or something like that? |
| 21:29.05 | ariel_ | jsmith: wow, it would be nice not to have it in the dial plan what we are doing now by adding our own type of uniqueID to all the cdr's |
| 21:30.28 | drfreeze | jsmith: thx |
| 21:31.02 | academy | ariel_/[TK]D-Fender: http://pastebin.ca/1452370 |
| 21:31.40 | academy | When the call comes in, nothing at all is printed to the cli |
| 21:31.58 | *** part/#asterisk muiro (n=muiro@cpe-173-89-177-15.neo.res.rr.com) |
| 21:32.31 | academy | updated the paste to show dahdi show channels |
| 21:32.57 | ariel_ | academy: setup a catch all extension and do verbose = 99, something like exten => _X.,1,NoOp, then exten => _X.,2,answer then next |
| 21:33.18 | ariel_ | core set verbose 99 |
| 21:34.09 | academy | ok |
| 21:35.18 | ariel_ | don't think your calls are posting all the numbers your setup like it might not be putting any 000 into the post. Or in some cases it might not be sending any numbers at all so you might need to just do a exten => s,1,answer |
| 21:35.54 | rue_mohr | uh |
| 21:35.57 | *** join/#asterisk mltlnx (n=mltlnx@asams.mserve.com) |
| 21:36.01 | rue_mohr | dosn't 10 include everything? |
| 21:36.22 | ariel_ | 10 include everything? |
| 21:36.31 | rue_mohr | debug / verbose 10 |
| 21:36.35 | ariel_ | wonders what he missed. |
| 21:36.40 | ariel_ | oh that |
| 21:37.10 | ariel_ | set verbose to 99 is just a habbit I have |
| 21:38.20 | ariel_ | academy: hope you get it working. I have to go home now... It's time to go and play with the wife and my little girl... |
| 21:39.19 | ziram19 | is hpec echo canceller so performant as the hardware module? |
| 21:39.19 | academy | ok ariel_. Nothing is printed still - it's almost as if dahdi is just ignoring the line. Thanks for the help anyway! |
| 21:39.37 | rue_mohr | ziram19, I dont know I couldn't get the damn thing to work |
| 21:40.03 | *** part/#asterisk elitecoder (n=liq@apollo.bullethost.com) |
| 21:40.55 | [TK]D-Fender | ziram19: No |
| 21:41.17 | ziram19 | you mean that the hardware module is the best |
| 21:41.40 | ziram19 | and what about oslec the open source one? |
| 21:42.46 | *** join/#asterisk seanmh (n=johndoe@c-69-254-131-168.hsd1.nm.comcast.net) |
| 21:43.24 | *** part/#asterisk pierrelux (n=pierrelu@IP-208-88-110-46.mtl.fibrenoire.ca) |
| 21:44.34 | [TK]D-Fender | ziram19: OSLEC is generally considered a little lower than HPEC. YMMV |
| 21:44.34 | ziram19 | TK ? |
| 21:44.59 | ziram19 | TK thanks |
| 21:45.42 | ziram19 | TK so if hpec doesn't solve the pb i must buy the hardware module |
| 21:46.00 | ziram19 | i am in this case |
| 21:46.22 | *** join/#asterisk DarkRift (n=dark@65.92.166.229) |
| 21:47.18 | [TK]D-Fender | ziram19: Have you tried OSLEC? |
| 21:47.33 | ziram19 | no not yet? |
| 21:47.35 | *** part/#asterisk jsgoecke (n=Adium@c-71-202-25-141.hsd1.ca.comcast.net) |
| 21:48.30 | *** join/#asterisk scooby2 (n=aprowant@pdpc/supporter/active/scooby2) |
| 21:49.00 | ziram19 | but i tried hpec and you told me that oslec is generally more lower so i have not much chance |
| 21:49.05 | scooby2 | Is there anyway to manually add an entry to mysql cdr? |
| 21:49.29 | *** join/#asterisk Deadpepsi (n=marc@pool-173-65-250-118.tampfl.fios.verizon.net) |
| 21:49.44 | Nugget | of course, it's just sql. |
| 21:49.52 | Nugget | insert a row |
| 21:49.54 | Deadpepsi | anyone here want to help an old slow guy build a asterisk server with gui will have 4 voip trunk lins in and the rest is for me to play around with |
| 21:49.54 | scooby2 | well from asterisk |
| 21:50.05 | *** part/#asterisk jsmith (n=jsmith@asterisk/training-and-documentation-guru/jsmith) |
| 21:50.13 | Nugget | write an AGI that inserts it. |
| 21:50.38 | Deadpepsi | i need simple and easy to understand and manage |
| 21:50.40 | *** join/#asterisk n3hxs (n=HAMming@lax-wan-a-47.lax.dsl.cerfnet.com) |
| 21:50.45 | [TK]D-Fender | ziram19: GO TRY IT |
| 21:50.53 | leifmadsen | scooby2: or use func_odbc |
| 21:51.00 | [TK]D-Fender | Deadpepsi: GUI's are not supported here. |
| 21:51.08 | Deadpepsi | ok |
| 21:51.19 | Nugget | apostrophe's aren't either! ;) |
| 21:51.26 | Deadpepsi | well then how do you manage it all? |
| 21:51.30 | scooby2 | my dst is getting overwritten so I want to make an entry with dst, unique call id, etc. |
| 21:51.38 | scooby2 | leifmadsen: thats what I was looking for |
| 21:51.41 | Deadpepsi | not aginst other options |
| 21:51.54 | Deadpepsi | just thought gui would be easier for now |
| 21:52.36 | scooby2 | I'd like to stay away from agi if possible. I actually just want my dst to stop getting overwrote. Instead of the number called it is showing as the queue number |
| 21:52.41 | leifmadsen | scooby2: dst is whatever the last executed dialplan extension was when the call ended I'm pretty sure -- Asterisk CDRs are a finicky beast. You are probably looking for something like CEL, which is not yet merged to Asterisk trunk. |
| 21:53.39 | leifmadsen | scooby2: like I said -- you need to make sure the last extension in the dialplan being executed is what you want written in the CDR |
| 21:53.40 | leifmadsen | a Goto() or anything like that is going to change the dst on the channel |
| 21:53.43 | scooby2 | leifmadsen: yeah thats what is happening |
| 21:53.54 | mercutioviz | what is CEL?! |
| 21:54.00 | scooby2 | channel event logging |
| 21:54.28 | mercutioviz | ah. is that like the xml cdr stuff in freeswitch? |
| 21:54.36 | *** join/#asterisk s14ck (n=s14ck@190.38.214.204) |
| 21:55.02 | *** join/#asterisk cvnet (n=dahitler@24.156.134.228) |
| 21:55.07 | cvnet | hello all |
| 21:55.10 | scooby2 | QueueLog ? |
| 21:56.13 | *** join/#asterisk s14ck (n=s14ck@190.38.214.204) |
| 21:59.14 | *** join/#asterisk academy (n=adam@unaffilated/academy) |
| 22:01.09 | academy | I have a TDM400P working fine outbound, but Asterisk doesn't even pick up the line with inbound calls. Even with verbosity high, nothing at all is printed to the cli. Any idea what could be wrong. Config here: http://pastebin.ca/1452424 |
| 22:01.53 | [TK]D-Fender | academy: that context is no good.. you don't have an "s" exten for your calls to land on. |
| 22:02.15 | scooby2 | leifmadsen: would a Macro also change the dst? |
| 22:02.53 | anthm | 283-171 |
| 22:02.55 | scooby2 | or just Goto/GotoIfTime/etc |
| 22:02.56 | anthm | doh |
| 22:05.37 | *** join/#asterisk propellerhead (n=yogurt2u@190.31.75.237) |
| 22:05.51 | Chainsaw | academy: Make sure you've set the zaptel/DAHDI system up for the correct country. |
| 22:06.12 | academy | Chainsaw: I have |
| 22:06.32 | Chainsaw | academy: I can't tell, you haven't told me where you are or shared that part of the config. |
| 22:09.04 | *** join/#asterisk lesouvage (n=lesouvag@82.73.69.76) |
| 22:09.06 | cvnet | if you want to do IPsec connection when you configure your vpn, should you choose 1.Cisco Compatible VPN (vpnc) 2. OpenVPN or 3. Point-to-point tunneling Protocoal (pptp) i think 3 but please correct me if im wrong |
| 22:10.22 | *** join/#asterisk mltlnx (n=mltlnx@asams.mserve.com) |
| 22:11.59 | academy | Chaisaw/[TK]D-Fender: does that give you the information you need? http://pastebin.ca/1452439 |
| 22:12.05 | academy | thanks for the help |
| 22:12.56 | [TK]D-Fender | academy: that looks better, but you'll have to restart * or reload the module to take effect |
| 22:13.02 | [TK]D-Fender | academy: A straight reload will not do it |
| 22:13.08 | *** join/#asterisk SkywaIker (n=pirch@58.147.17.166) |
| 22:13.16 | *** join/#asterisk Aiatek (n=yo@190.94.61.83) |
| 22:14.41 | academy | [TD]D-Fender: still nothing at all printed to the cli |
| 22:14.50 | academy | it just rings out |
| 22:17.21 | scooby2 | is there a variable that has the number the caller called in it? So I could do something like Set(CDR(userfield)=$DST) |
| 22:17.34 | scooby2 | (if that still works) |
| 22:18.17 | *** join/#asterisk arpu (n=arpu@chello062178159144.10.14.univie.teleweb.at) |
| 22:20.10 | scooby2 | actually something like Set(CDR(userfield)=CDR(dst)) ? |
| 22:24.24 | *** join/#asterisk juanIMP (n=Juancho@200.71.41.254) |
| 22:29.47 | leifmadsen | scooby2: _X.,1,Set(INCOMING_EXTENSION=${EXTEN}) |
| 22:30.03 | leifmadsen | Set(CDR(userfield)=${INCOMING_EXTENSION}) |
| 22:30.57 | scooby2 | thank you |
| 22:32.13 | scooby2 | ${EXTEN} would be the number the customer dialed? |
| 22:32.28 | scooby2 | yes it would |
| 22:33.36 | Deadpepsi | guys am i on my own here to figure this out? i have asterisk now installed but i do not see any gui for it so i am guessing i am just have a basic asterisk server now with no configuration! |
| 22:33.44 | *** join/#asterisk Assimilate (n=Assimila@72.22.242.66) |
| 22:34.02 | mmlj4 | right, no configuration |
| 22:34.10 | mmlj4 | a gui exists, though |
| 22:34.27 | mmlj4 | but most folks learn how to configure it by reading the book |
| 22:34.28 | Deadpepsi | but not supported here |
| 22:34.29 | mmlj4 | ~book |
| 22:34.30 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 22:35.20 | Deadpepsi | classic open source answer :( |
| 22:35.26 | *** part/#asterisk Deadpepsi (n=marc@pool-173-65-250-118.tampfl.fios.verizon.net) |
| 22:35.28 | mmlj4 | if you need a gui, install that, or use one of the plug-n-pray distros |
| 22:35.41 | scooby2 | the book is priceless |
| 22:35.51 | leifmadsen | (trixbox, AsteriskNOW, PBX in a Flash, etc...) |
| 22:36.08 | *** join/#asterisk Deadpepsi (n=marc@pool-173-65-250-118.tampfl.fios.verizon.net) |
| 22:37.06 | [8none1] | Deadpepsi: You do need to read the book to know what your doing. It's an easy read and is aplicable whether you use a GUI or not. |
| 22:37.07 | leifmadsen | Deadpepsi: you mean the step by step instructions provided to get a phone setup isn't enough? If that's the case, then I suggest you use a pre-packaged GUI distribution |
| 22:37.49 | leifmadsen | Asterisk has a steep learning curve as it integrates several technologies into a single package, is a tool-kit to build phone applications, but doesn't need to be overly complicated to get some simple calls going through it |
| 22:38.01 | *** join/#asterisk R-Guy_ (n=ron@secure.starsolutions.com) |
| 22:38.19 | leifmadsen | but you do need to read some documentation to understand what is going on, and how the .conf files all work together, or you're just going to get frustrated and lost |
| 22:38.25 | cvnet | if you want to do IPsec connection when you configure your vpn, should you choose 1.Cisco Compatible VPN (vpnc) 2. OpenVPN or 3. Point-to-point tunneling Protocoal (pptp) i think 3 but please correct me if im wrong |
| 22:38.28 | Deadpepsi | no thats not what i need but since i ask questions and get to google or read a book guess i as you say in over my head LOL |
| 22:39.07 | leifmadsen | Deadpepsi: I don't understand what you need then. Please help me, help you. |
| 22:39.23 | *** join/#asterisk Torrieri (n=Torrieri@nelug/crew/torrieri) |
| 22:40.07 | [8none1] | Deadpepsi: take advantage of the help offered. leifmadsen is one of the authors of the Asterisk book |
| 22:40.16 | leifmadsen | [8none1]: filthy lies |
| 22:40.22 | [8none1] | :) |
| 22:40.44 | mercutioviz | Deadpepsi: telephony != easy. it takes time and effort to learn everything. the TFOT book is very well written |
| 22:41.50 | R-Guy_ | I have a TDM400P card populated with a FXS module. The FXS port is connected to a security alarm panel (dialer). When the alarm panel dials-out to the central station, that the communications always fails. The information being sent from the alarm panel is FSK. I am wondering of the hardware or the zaptel driver is mucking with the FSK (thinking it is something like ADSI or callerid). |
| 22:41.50 | R-Guy_ | Has this been seen before? |
| 22:41.55 | academy | I'm receiving the error Starting DAHDI/2-1 at fxo-inbound-group-1,s,1 failed so falling back to exten 's'. Does that mean the DID I'm using does not match the incoming one? |
| 22:42.08 | academy | exten=>_X.,n,Goto(fxo-inbound-group-0,s,1) |
| 22:42.25 | leifmadsen | academy: is it really an FXO port? there is no number coming it -- that's only on a digital line |
| 22:42.32 | [8none1] | R-Guy_: It's probably trying to make a modem connection which will almost always fail through voip |
| 22:42.34 | leifmadsen | academy: Asterisk is going to try and match the 's' (start) extension |
| 22:43.20 | academy | leifmadsen: yes, it's an fxo port. What should I change? |
| 22:43.41 | leifmadsen | academy: asterisk is going to match on the 's' extension, not a pattern match (i.e. not a number) |
| 22:43.56 | leifmadsen | you should change your dialplan as appropriate |
| 22:44.05 | Deadpepsi | i never said it was easy if it were there would not be an irc chat room with 282 users in now would there humm but what i dislike is the only answer i ever seem to get with anything involving linux or open source is "go read XXXX book or google it" if i can lean linix in 3 weeks after 30 years as a ms-tech i think know about reading books and all |
| 22:44.06 | R-Guy_ | When I Monitor() the call, I can't hear much of the FXS side, but the LL side sounds file. Makes me think that the FSK signal is being intercepted somewhere. |
| 22:44.31 | leifmadsen | Deadpepsi: you haven't asked a specific question from what I can tell... what is your question? |
| 22:44.41 | *** join/#asterisk mattbUK (n=mattbUK@82-46-93-20.cable.ubr16.stav.blueyonder.co.uk) |
| 22:44.44 | leifmadsen | Deadpepsi: what part are you stuck on? what do you need help with? |
| 22:45.33 | R-Guy_ | Codec is PCMU, so it should carry a low speed modem signal okay. |
| 22:45.37 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 22:46.25 | Deadpepsi | what other way other then gui can i use to configure it? |
| 22:46.36 | leifmadsen | R-Guy_: yes, that should be ok. I've never sent a modem signal through asterisk though, but I don't think it should affect anything. The only thing I can think of is that the alarm is starting to try to communicate before the other end has actually answered |
| 22:46.48 | Deadpepsi | i was told gui was not supported here |
| 22:46.50 | leifmadsen | Deadpepsi: the vim text editor? without a GUI, asterisk is configured manually through text files |
| 22:47.01 | Deadpepsi | ok |
| 22:47.07 | academy | leifmadsen: thank you |
| 22:47.10 | leifmadsen | Deadpepsi: it's not -- this room is here to help with manual configuration |
| 22:47.10 | Deadpepsi | thats all i asked before |
| 22:47.25 | *** join/#asterisk jicksta (n=jicksta@67.164.0.78) |
| 22:47.28 | Deadpepsi | and thats fine i am all for know it both ways |
| 22:47.45 | leifmadsen | Deadpepsi: you have to grow some thick skin for the asterisk community. It's full of know-it-all tech heads -- telephony people are way worse than kernel hackers (who are in turn worse than linux people in general) |
| 22:47.47 | Deadpepsi | the world is to stuck on gui |
| 22:47.47 | mercutioviz | Deadpepsi: personally, what I did was get trixbox up and running first, looked at the stuff it did, then I went back and read the TFOT book |
| 22:48.10 | R-Guy_ | leifmadsen: I can hear both sides, and the alarm panel side only starts sending after is hears the answer tone from the central station. I am thinking that this has something to do with the TDM harware or the zaptel driver. |
| 22:48.35 | Deadpepsi | i have it installed in vbox to play and lean |
| 22:48.37 | Deadpepsi | for now |
| 22:48.42 | leifmadsen | R-Guy_: hmmmm... I'm not really a hardware guy (I tend to mostly use SIP, and when hardware, always digital circuits). I'd be surprised if Asterisk was modifying anything.... |
| 22:49.15 | leifmadsen | Deadpepsi: right -- I'm still not sure what problem you were having before, or why you thought you weren't getting help |
| 22:49.33 | Deadpepsi | its by no means ready for real stuff yet but i have a pc and all ready when i am and it is ready for me to go live with it |
| 22:49.39 | R-Guy_ | leifmadsen: Not asterisk the zaptel driver. If tis responsible for generating FSK for ADSI, so I assume it also decodes FSK as well. |
| 22:50.42 | Deadpepsi | well i asked the question about gui and was told it was not supported so i then asked the what other options cfor config i had and waited almost an hour before asking the same thing re-worded |
| 22:51.17 | Deadpepsi | and then the bashing started ok yea i brand new here and i really want to lean this i cant afford cisco call manager |
| 22:51.44 | Deadpepsi | my wife and i have found a need to setup a pbx in our home |
| 22:52.09 | Deadpepsi | its all good thought i am use to it |
| 22:52.24 | Deadpepsi | got the same thing when i dumped xp and went linux |
| 22:52.27 | R-Guy_ | I can debug it further using an SIP ATA and bypass the TDM400 and zaptel stuff all together -- just wondering if anyone as looked at this before. |
| 22:52.48 | leifmadsen | R-Guy_: I don't know enough about analog hardware, and how the driver handles the signalling stuff, sorry. |
| 22:52.55 | R-Guy_ | np |
| 22:53.45 | [8none1] | Deadpepsi: there are other channels for support of Trixbox, FreePBX, etc. You can check with those guys for any problems. But if you are just getting started and don't know anything about VoIP or Telephony you should read the book or google for getting started guides. |
| 22:54.07 | [8none1] | Deadpepsi: It's just more than can be covered in an IRC channel |
| 22:54.54 | Deadpepsi | i know a lot about voip and pots but the pbx is new and all |
| 22:55.00 | leifmadsen | Deadpepsi: for a quick setup, you should grab something like trixbox, AsteriskNOW, or PBX in a flash |
| 22:55.44 | leifmadsen | Deadpepsi: asterisk is complex -- when someone says to go read the book, they really do mean it -- it's not just some trick to try and schlub you off |
| 22:55.45 | Deadpepsi | i have asteriskNOW installed in vbox now and it booted to consol |
| 22:55.57 | Deadpepsi | ok |
| 22:56.08 | Deadpepsi | i will take the leifmadsen |
| 22:56.14 | leifmadsen | Deadpepsi: when you install AsteriskNOW 1.5, it should ask you whether you want to install FreePBX or the Asterisk GUI I'm pretty sure |
| 22:56.25 | Deadpepsi | i am use to the go read a book as the blow off |
| 22:56.28 | leifmadsen | Deadpepsi: that's why we wrote the book -- to help you get started from the ground up |
| 22:56.41 | Deadpepsi | ok |
| 22:56.50 | leifmadsen | Deadpepsi: it is not a blow off -- you really do need to know the basics to have any hope of understanding what is going on |
| 22:56.58 | Deadpepsi | i dont disagree |
| 22:57.11 | leifmadsen | ok, when you have a specific question about something, feel free to come back and ask |
| 22:57.22 | Deadpepsi | ok |
| 22:57.46 | Deadpepsi | mind if i ideal and read other problems? |
| 22:58.07 | *** join/#asterisk xfighter (n=xfighter@a95-198.adsl.paltel.net) |
| 22:58.10 | xfighter | hi |
| 22:58.22 | xfighter | whats the difference between asterisk and asterisk now???? |
| 22:58.35 | leifmadsen | Deadpepsi: I don't understand the question |
| 22:58.41 | Deadpepsi | lol |
| 22:58.45 | Deadpepsi | i am reading |
| 22:58.47 | leifmadsen | oh, idle |
| 22:58.52 | leifmadsen | of course, idle away |
| 22:58.52 | Deadpepsi | yea sorry |
| 22:59.09 | leifmadsen | xfighter: one is just asterisk, the other is asterisk as a package with GUI |
| 22:59.13 | Aiatek | hi leifmadsen the last friday i was in bogota setting and extension for you in my practical exam :P |
| 22:59.24 | Deadpepsi | also switched to a laptop some time back and still not use to the tiny keyboard |
| 22:59.27 | leifmadsen | Aiatek: :D |
| 22:59.57 | xfighter | leifmadsen : so I can use one of them or shall I use both? |
| 23:01.15 | xfighter | ?? |
| 23:01.17 | *** join/#asterisk \malex\ (i=Xz+yrHsv@unaffiliated/malex/x-000000001) |
| 23:01.48 | \malex\ | is there a comparitive review of trixbox and elastix and pbx in a flash and any other full-distro astrisk solutions? |
| 23:02.00 | xfighter | leifmadsen : I was reading throught the book and I found this sentence |
| 23:02.15 | xfighter | which gives you a complete PBX system with graphical configuration screen |
| 23:02.15 | xfighter | all built into one! |
| 23:02.47 | xfighter | does this mean that there was no need to download my own linux dist and install asterisk manually? |
| 23:03.21 | Aiatek | xfighter exactly |
| 23:03.26 | [TK]D-Fender | xfighter: Gui's for Asterisk is like selling a box of deluxe coloring pencils and telling the artist all they can do is draw pictures of a fire-truck. |
| 23:03.36 | xfighter | OH MY GOD |
| 23:03.43 | xfighter | no no am really pissed of |
| 23:03.46 | Aiatek | it will install everything |
| 23:03.57 | xfighter | now that I have downloaded more than 5 gegs of data |
| 23:04.07 | xfighter | spent more than 2 days understanding the concepts |
| 23:04.20 | xfighter | now I found out that there was no neeed to do all of that |
| 23:04.26 | xfighter | THIS IS ........... |
| 23:04.37 | xfighter | Uhhh I can't believe this |
| 23:05.08 | xfighter | you guyz should've mentioned that earlier in the book |
| 23:05.38 | [TK]D-Fender | xfighter: Meantioned what? |
| 23:05.38 | xfighter | and made it optional for new joiners to either use the gui distribution or go through the manual installation |
| 23:06.13 | [TK]D-Fender | xfighter: it was ALWAYS optional, and the books instruciton in no way tell you to use one fo the Barbie Easy-Bake Oven distros out there |
| 23:06.48 | xfighter | I know I don't like spoon feeding myself |
| 23:07.00 | xfighter | but not in my case |
| 23:07.06 | [TK]D-Fender | xfighter: Thats ok... we don't spoon-feeding people either ;) |
| 23:07.11 | [TK]D-Fender | like* |
| 23:07.13 | [TK]D-Fender | dangit |
| 23:07.15 | xfighter | am preparing for my seminar |
| 23:07.21 | xfighter | and I just need the time |
| 23:07.21 | [TK]D-Fender | xfighter: On? |
| 23:07.30 | xfighter | voip communications |
| 23:07.37 | *** join/#asterisk ecret (n=ecret@CPE001e9002348e-CM001225d8ab30.cpe.net.cable.rogers.com) |
| 23:07.40 | [TK]D-Fender | xfighter: What else have you mastered? |
| 23:07.50 | xfighter | it's a bachelor |
| 23:07.52 | xfighter | in fact |
| 23:08.12 | [TK]D-Fender | xfighter: I'm talking in terms of VoIP tech. |
| 23:08.14 | xorl | [TK]D-Fender: You'll probably know this right off the bat, when I call a new number on my new SIP provider, it's a silent outgoing, ususually here the ringing but i'm not |
| 23:08.25 | xorl | tried inband and rfc.... Still no go, any tips? |
| 23:08.44 | [TK]D-Fender | xorl: those are DTMF modes, not indications. |
| 23:08.52 | [TK]D-Fender | xorl: If they provide no progress you could try dialing with "r" |
| 23:09.10 | [TK]D-Fender | xorl: Ill-advised but potentially necessary if they don't give you a 180 on call progress |
| 23:09.19 | xorl | I get a 186 |
| 23:09.45 | xfighter | oh , well ... am thinking about implementing a server that is welling to make it possible for voip devices to communicate easily with each other in addition of a gui of my own using java to do the same job and javascript website for the same purpose also |
| 23:10.01 | xfighter | to summerize , phone2phone , phone2pc , pc2phone |
| 23:10.51 | rue_mohr | pbx.c:1642 pbx_substitute_variables_helper_full: Error in extension logic (missing '}') exten => s,n,SET(CALLERID(name)=${ODBC_MODCID(CALLERID(number),CALLERID(name))} read=SELECT newname FROM modcid WHERE number='${ARG1}' UNION all values('${ARG2}') limit 1 ???? |
| 23:11.19 | rue_mohr | oh I think I see it |
| 23:12.01 | xfighter | anyway [TK]D-Fender , may I ask u something? |
| 23:12.17 | [TK]D-Fender | xfighter: You already have |
| 23:12.25 | [TK]D-Fender | xfighter: I'm ight even permit ANOTHER qustion :p |
| 23:12.26 | xfighter | one more please :D |
| 23:12.32 | xfighter | oh thanx |
| 23:12.46 | [TK]D-Fender | xfighter: Go for it.. but you still haven't mentioned any other VoIP techs you've really worked with... |
| 23:13.15 | [TK]D-Fender | rue_mohr: Yup, couple obvious breaks |
| 23:13.28 | [TK]D-Fender | xorl: You get a RINGING from them? |
| 23:14.26 | xfighter | now that I have finished the installation manually am I ever goin to need the gui distribution of asterisk or I can continue with what I have so far , and do u think it will be easier for me to have the gui dustribution , please note that I've just finished ch3 of the book and I'll go through ch4 tonight |
| 23:14.47 | rue_mohr | couple? |
| 23:14.51 | rue_mohr | I found the missing ) |
| 23:14.57 | xorl | [TK]D-Fender: I would assume I should. |
| 23:15.03 | *** join/#asterisk BadHAL (n=nn@cpe-72-179-194-139.stx.res.rr.com) |
| 23:15.09 | [TK]D-Fender | xfighter: With most GUI installs you configure jsut about everything via their web interface and hardly touch the raw configs at all. This makes most of the book useless |
| 23:15.11 | xorl | Others (i've read online) have ringback no problem. |
| 23:15.12 | Aiatek | read until chapter 6 and you will undestand many things about asterisk |
| 23:15.23 | [TK]D-Fender | xfighter: Like getting your drivers license and then hiring a chauffeur. |
| 23:15.30 | rue_mohr | but now its giving me CALLERID(name) as the result |
| 23:15.53 | xfighter | so it's either the book or the gui ? |
| 23:16.24 | xfighter | I would love to listen to your advice |
| 23:16.33 | [TK]D-Fender | xfighter: GUI teaches you nothing except how to build a cookie-cutter PBX according to the concepts they implement. Do it yourself and you can do whatever you feel like |
| 23:17.08 | [TK]D-Fender | xfighter: My calls do XYZ if Joe calls me after 7pm on tuesdays if its raining and the Mets won their last home-game. |
| 23:17.20 | [TK]D-Fender | xfighter: jsut about ANYTHING |
| 23:17.23 | rue_mohr | ah found it ${CALLER... |
| 23:17.23 | *** join/#asterisk mjmac (n=mjmac@cpe-74-65-154-175.maine.res.rr.com) |
| 23:17.30 | [TK]D-Fender | xfi nots just a "step 1,2,3 boring crap. |
| 23:17.41 | [TK]D-Fender | rue_mohr: Good... more errors to fix :) |
| 23:17.48 | [TK]D-Fender | rue_mohr: Keep looking :) |
| 23:18.12 | rue_mohr | WHAT!? |
| 23:18.29 | rue_mohr | exten => s,n,SET(CALLERID(name)=${ODBC_MODCID(${CALLERID(number)},${CALLERID(name)})}) |
| 23:18.33 | rue_mohr | its perfect!???? |
| 23:18.34 | xfighter | then so be it , now that I have finished most of the work , I won't start over again , but I will download the gui in case I have faced problems that I will not be able to solve easily , but so far I will proceed with mission and with book :) |
| 23:18.52 | rue_mohr | isn't it!? |
| 23:18.54 | xorl | weird not even specifying r is working |
| 23:18.58 | [TK]D-Fender | rue_mohr: Yeah, that one balances |
| 23:19.11 | rue_mohr | thankyou! |
| 23:19.15 | [TK]D-Fender | xorl: pastebin a faied callw ith SIP debug |
| 23:19.19 | rue_mohr | re-applies his flipped lid |
| 23:19.34 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
| 23:19.34 | xorl | [TK]D-Fender: will do |
| 23:19.39 | [TK]D-Fender | xfighter: keep in mind there are SEVERAL GUI's for *. |
| 23:19.41 | xfighter | ok guyz gotcha go understand the channels concept |
| 23:20.19 | xfighter | for asterisk or softphones? |
| 23:20.43 | [TK]D-Fender | xfighter: And most of them completely own your ass. AsteriskGUI is a "work in progress", and has plenty of bugs and forces you to get your hands dirty. Not only do you have to understand * itself on a pretty good level, you ALSO need to know how the GUI part builds bits as well |
| 23:20.43 | Greek-Boy | anybody know why func_odbc.so would cause a segmentation fault on asterisk? |
| 23:21.22 | xfighter | MAN |
| 23:21.25 | xfighter | I LOVE YOU |
| 23:21.52 | [TK]D-Fender | xfighter: FreePBX owns your ass, but is at least largely "complete" |
| 23:22.03 | *** join/#asterisk gazzerh (n=garryh@93-97-187-150.zone5.bethere.co.uk) |
| 23:22.47 | Greek-Boy | Guys while we are on the topic of GUI's |
| 23:23.03 | Greek-Boy | what's a good billing system? |
| 23:24.25 | xorl | [TK]D-Fender: Get that notice? |
| 23:24.57 | [TK]D-Fender | xorl: no |
| 23:25.01 | xorl | weird |
| 23:26.05 | *** part/#asterisk Slade- (i=user@255.85.204.68.cfl.res.rr.com) |
| 23:33.10 | rue_mohr | exit |
| 23:33.13 | rue_mohr | \exit |
| 23:37.23 | *** join/#asterisk layne (n=layne@ool-44c0048f.dyn.optonline.net) |
| 23:37.27 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 23:41.28 | *** join/#asterisk khronos (i=tom@sip02.kineticvoip.com) |
| 23:41.38 | khronos | Evening all. |
| 23:42.27 | khronos | If I have a Cisco 2621 routerat the front of my network how many calls can I expect to be able to pass thgouh it assuming I'm using g711 as the audio codec? |
| 23:43.50 | khronos | Lets assume I have a cross-over cable between two of these routers and the servers sit on the back side of each. |
| 23:45.37 | GrantA | What is a UK-based CID readback phone #... so I can test my callerid |
| 23:49.12 | Winkie | GrantA: you can call me if you like and i'll tell you it :) |
| 23:49.20 | *** join/#asterisk blackest_mamba (n=blackest@c-71-239-160-143.hsd1.il.comcast.net) |
| 23:51.00 | *** join/#asterisk evildan7 (n=werdan7@freenode/staff/wikimedia.werdan7) |
| 23:54.58 | *** join/#asterisk PanicMan (i=Learner@122.102.33.80) |
| 23:57.23 | PanicMan | how to replace incoming prefix in Asterisk, like, if sender use 01231111111 asterisk will remove 0123 & forward 1111111 |
| 23:58.58 | PanicMan | no one is here, seems, everyone is busy :S |