| 00:01.53 | *** part/#asterisk mog (n=mog@c-71-207-200-130.hsd1.al.comcast.net) |
| 00:01.55 | *** join/#asterisk ptblank (n=MURDER1@cpe-75-84-216-188.socal.res.rr.com) |
| 00:02.53 | *** join/#asterisk nuonguy (n=john@c-24-6-175-26.hsd1.ca.comcast.net) |
| 00:04.09 | ipso | I'm trying to setup Asterisk 1.4.2 with Gtalk, and I can see the call coming in, and it looks like Asterisk is answering it (the menu starts to play) but on the GTALK client, it doesn't think anyone picked up, so it says no one answered. However in my Asterisk console I see this message: Starting Gtalk/ipso-a620 at inbound-gtalk,ipso@gmail.com,1 failed so falling back to exten 's' -- If I type: show dialplan inbound-gtalk it shows that s |
| 00:04.09 | ipso | ,1 exists as: Answer. Any ideas whats going wrong? |
| 00:19.51 | *** join/#asterisk mishehu (i=mishehu@cshells.shavedgoats.net) |
| 00:20.05 | *** join/#asterisk Dovid (n=Dovid@l192-117-114-1.broadband.actcom.net.il) |
| 00:20.08 | Dovid | hi guys |
| 00:20.14 | Dovid | how do i set a variable using asterisk -rx ? |
| 00:23.17 | drfreeze | Dovid: `asterisk -rx "database put office oncall #{oncall[1]}"` |
| 00:23.47 | drfreeze | `asterisk -rx "database get #{key}"`.split(/\n/)[0].sub(/^Value:\s+/,"") |
| 00:24.08 | Dovid | i dont want to use the DB |
| 00:24.27 | Dovid | or are all varaibles stored in the asterisk db ? |
| 00:24.40 | drfreeze | hmm... |
| 00:25.14 | drfreeze | * can use MySQL, but that is a bit much to just set a variable |
| 00:25.58 | Dovid | yea |
| 00:26.10 | drfreeze | David you may be able to do a `asterisk -rx "Set(MYVAR=value)"` |
| 00:26.18 | Dovid | i tired that |
| 00:26.26 | drfreeze | didn't work? |
| 00:26.34 | ipso | Anyone here familiar with Asterisk and Jabber integration? |
| 00:26.45 | Dovid | dpfeeze: i think i will use the asterisk db |
| 00:26.55 | Dovid | it is ? |
| 00:27.10 | Dovid | asterisk -rx 'databse put a/a=5' ? |
| 00:27.25 | drfreeze | nope |
| 00:27.48 | drfreeze | asterisk -rx 'database put family var value' |
| 00:28.20 | Dovid | thx |
| 00:28.37 | *** join/#asterisk lee_is_me (n=chatzill@12-227-176-77.client.mchsi.com) |
| 00:29.17 | Dovid | and to use it in the dial plan |
| 00:29.23 | drfreeze | also, checkout the output of: asterisk -rx 'database get family' |
| 00:29.39 | Dovid | $DB[family/var] ? |
| 00:30.45 | drfreeze | exten => s,n,Set(ONCALL=${DB(office/oncall)}) |
| 00:30.45 | drfreeze | exten => s,n,Set(OFFICESTATE=${DB(office/state)}) |
| 00:35.13 | lee_is_me | hi everyone, is anyone up for a polycom question? |
| 00:37.34 | lee_is_me | or is that off topic? |
| 00:38.07 | *** part/#asterisk infernix (i=nix@unaffiliated/infernix) |
| 00:38.25 | *** join/#asterisk kev009 (n=kev009@ip70-162-43-70.ph.ph.cox.net) |
| 00:39.14 | kev009 | I'm looking for some ATAs for my house, is IAX2 or SIP better for LAN connections? |
| 00:39.45 | *** join/#asterisk saftsack (n=saftsack@pD9E06820.dip.t-dialin.net) |
| 00:41.33 | lee_is_me | its like I just walked into a bar and everyone fell silent... |
| 00:42.05 | drfreeze | lee_is_me: ask your question |
| 00:42.21 | drfreeze | kev009: SIP |
| 00:42.36 | lee_is_me | lol, thanks. 301 has humming noise in background |
| 00:42.40 | Dovid | lee_is_me: ask away |
| 00:42.54 | drfreeze | lee_is_me: turn the volume down. :) |
| 00:43.08 | lee_is_me | tried fiddling with rx/txgain - just seems to raise and lower the volume of the hum |
| 00:43.13 | Dovid | lee_is_me: is it only with 301 ? or other phones ? are using ZAP ? |
| 00:43.19 | Dovid | it can be "echo" |
| 00:43.37 | lee_is_me | i have a budgeton on the lan and it does not have it...zap line for both |
| 00:43.40 | drfreeze | lee_is_me: all 301's or just one? |
| 00:43.50 | lee_is_me | just one of each |
| 00:44.04 | drfreeze | lee_is_me: how is the 301 provisioned? |
| 00:44.15 | lee_is_me | from the web interface |
| 00:44.22 | lee_is_me | just purchased it, first one |
| 00:44.56 | lee_is_me | checked cables to make sure they were not near interferece... |
| 00:45.21 | lee_is_me | i ran across one article suggesting to use a shield cable to a PC from the phone to ground it |
| 00:45.22 | drfreeze | lee_is_me: lee_is_me may be a bad phone |
| 00:45.40 | drfreeze | lee_is_me: how is it powered? |
| 00:45.55 | lee_is_me | from the wall |
| 00:46.01 | lee_is_me | not poe |
| 00:46.06 | drfreeze | ok |
| 00:46.22 | lee_is_me | cheap power strip from the wall... |
| 00:46.47 | lee_is_me | i just wanted to check for advise before RMAing the phone |
| 00:46.48 | drfreeze | I'd say bad configuration or bad phone |
| 00:47.02 | drfreeze | if budgetone is ok, then probably bad phone |
| 00:47.27 | lee_is_me | thanks. I'll play around a bit more before sending it back... |
| 00:47.47 | drfreeze | lee_is_me: do you know how to provision the phone via ftp? |
| 00:48.21 | lee_is_me | i was starting to read the docs the other day but, but got caught up in some other things |
| 00:48.31 | drfreeze | lee_is_me: provision the phone with the standard setup. If the buzzing persists, RMA it |
| 00:48.42 | lee_is_me | i was thinking of writing a utility for it just to get familiar with the xml markup |
| 00:49.09 | lee_is_me | standard as is LCD menu or T/FTP? |
| 00:49.41 | drfreeze | ftp and xml |
| 00:49.56 | drfreeze | there are a lot of setting you can't set via web or menu |
| 00:50.10 | lee_is_me | yeah, i noticed. many, many options |
| 01:05.59 | *** join/#asterisk rad07 (i=raca@64-126-95-37.static.everestkc.net) |
| 01:07.42 | rad07 | hi. I am gettig SIP/2.0 405 Method Not Allowed |
| 01:08.26 | rad07 | I am using SJphone/1.60.289a (SJ Labs) client and Asterisk 1.4 |
| 01:08.37 | rad07 | Any thoughts |
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| 01:13.02 | Nugget | I think that the song 'Sweet Emotion' by Aerosmith fails to deliver on the promise that it's opening 20 seconds makes. |
| 01:15.38 | *** part/#asterisk c6vette (n=khagan@71-223-199-122.phnx.qwest.net) |
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| 01:16.49 | *** part/#asterisk c6vette (n=khagan@71-223-199-122.phnx.qwest.net) |
| 01:27.34 | [TK]D-Fender | rad07, Maybe if we knew what it was RESPONDING to.... |
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| 02:21.10 | Inverted | is there a way to get the billsec info using agi? |
| 02:22.05 | Inverted | let me clarify, a realtime status during the call |
| 02:31.00 | *** join/#asterisk BBRodriguez (n=BBRodrig@p54b00a8b.dip0.t-ipconnect.de) |
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| 02:44.15 | eald_home | Inverted, I know for a friend that is possible, look for prepaid solutions, they need to cut the call |
| 02:44.24 | eald_home | but I don't know the details |
| 02:46.16 | *** join/#asterisk DaveCanoe (n=Dave@H6.C30.B96.tor.eicat.ca) |
| 02:48.08 | lee_is_me | nugget: I would have to agree with you. Now the long play version of Rare Earth's Get Ready... |
| 03:22.44 | *** join/#asterisk CCFL_Man2 (i=0b75447f@pool-71-241-109-8.scr.east.verizon.net) |
| 03:23.36 | CCFL_Man2 | hey guys, anyone here have an adit 600 manual? |
| 03:25.53 | SwK | CCFL_Man2 Not for you they dont |
| 03:27.05 | CCFL_Man2 | SwK: they want $25 for a cd with the pdf files on it, $75 for print |
| 03:27.30 | SwK | hah |
| 03:27.44 | CCFL_Man2 | carrier access this is |
| 03:28.31 | CCFL_Man2 | and aparently if nothing is provisioned i don't even get talk battery out of the fxs ports |
| 03:30.16 | CCFL_Man2 | SwK: it's hard to believe but the adit 600 runs on vxworks and it's rock solid stable |
| 03:32.46 | CCFL_Man2 | wow, it's easy to configure |
| 03:33.06 | CCFL_Man2 | SwK: which is better to provide the clock? the channel bank or asterisk? |
| 03:36.13 | CCFL_Man2 | actually it does provide talk battery |
| 03:37.54 | *** join/#asterisk sharp (n=sharp@dsl092-234-217.phl1.dsl.speakeasy.net) |
| 03:38.07 | SwK | asterisk |
| 03:38.57 | *** join/#asterisk olinux (n=olinux@ip68-107-4-202.sd.sd.cox.net) |
| 03:39.12 | CCFL_Man2 | ahh |
| 03:42.37 | *** join/#asterisk Mavvie (n=edwin@ppp2-43.lns1.syd7.internode.on.net) |
| 03:44.12 | coppice | a talking battery? interesting |
| 03:48.42 | *** join/#asterisk stkn__ (n=stkn@gentoo/developer/pdpc.active.stkn) |
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| 03:53.18 | Sweeper | :/ |
| 03:54.41 | Inverted | Why would this: my $billsec = $AGI->agi_exec("EXEC Set('BILLSEC=${CDR(billsec)}')"); be producing this: |
| 03:54.41 | Inverted | Bareword "billsec" not allowed while "strict subs" in use at /var/lib/asterisk/agi-bin/test.agi line 155. |
| 03:55.35 | CCFL_Man2 | you can provide an internet T1 with an adit600 |
| 03:59.59 | JunK-Y | Inverted: thats an error related to perl, not to * |
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| 04:39.03 | Caplain | hey |
| 04:39.13 | Caplain | i cant get inbound to work with nufone |
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| 05:18.43 | CCFL_Man2 | anyone here have an adit 600 manual? |
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| 05:33.59 | Caplain | i cant get inbound to work with nufone |
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| 07:25.58 | reber | hi |
| 07:31.23 | blitzrage | yo |
| 07:41.16 | CCFL_Man2 | apparently configuring an adit 600 channel bank for voice is quite easy |
| 07:56.07 | *** join/#asterisk thevoke (n=michiel@missioncontrol.eng.gbxs.net) |
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| 08:21.04 | webman | I've configure ODBC with asterisk 1.4 and asterisk is connected to ODBC/etc, but how do I config the voicemail.conf to tell it to pull voicemail config from the DB? |
| 08:23.20 | *** join/#asterisk oej (n=olle@apollo.webway.se) |
| 08:24.03 | webman | I've configure ODBC with asterisk 1.4 and asterisk is connected to ODBC/etc, but how do I config the voicemail.conf to tell it to pull voicemail config from the DB? |
| 08:24.58 | *** join/#asterisk ming_zym (n=ming_zym@f0-0-tep-rtr1.corp.cnb.yahoo.com) |
| 08:33.10 | sooli | Hi ! My Asterisk server answer fine ... play mp3music but when i want to call from my softphone i have congestion ... |
| 08:52.49 | *** join/#asterisk stimpie (n=michiel@ip565faf27.direct-adsl.nl) |
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| 09:20.57 | *** join/#asterisk mklebel (n=mklebel@72.2.212.77) |
| 09:21.53 | mklebel | is conference calling over http possible? |
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| 10:00.41 | sbingner | mklebel, with enough trickery, sure... why not |
| 10:01.01 | mklebel | trickery eh |
| 10:06.18 | *** join/#asterisk sooli (n=sooli@AMontpellier-156-1-180-55.w90-36.abo.wanadoo.fr) |
| 10:08.38 | *** join/#asterisk toot (n=toot@84.19.255.123) |
| 10:16.47 | tmjb | hello could some tell me how to combine channels from misdn and zapata, i know there are groups but in channel and group manual say Dial,Zap/g3 but i want something like Dail/misdn/1 if busy Dial/zap/1-8 thankyou :) |
| 10:23.00 | tmjb | anyone awake |
| 10:23.05 | tmjb | or alive |
| 10:27.20 | *** join/#asterisk saftsack (n=saftsack@pD9E06E23.dip.t-dialin.net) |
| 10:51.24 | *** join/#asterisk tzafrir (n=tzafrir@62.90.10.53) |
| 10:58.57 | *** join/#asterisk EmleyMoor (i=phil@topdeck.tinsleyviaduct.com) |
| 10:59.35 | EmleyMoor | What's a reasonably-priced IP phone that can run with an external power supply and later be converted readily to PoE? |
| 11:05.49 | hal2k | EmleyMoor: grandstream gxp2000 is one |
| 11:07.18 | *** join/#asterisk gambolputty (n=gambolpu@cblmdm72-240-182-254.buckeyecom.net) |
| 11:09.30 | *** join/#asterisk Iamnacho (i=Iamnacho@ip68-103-153-140.ks.ok.cox.net) |
| 11:17.12 | *** join/#asterisk IgorG (n=FeedomPa@host-195-162-53-193.pppoe.omsknet.ru) |
| 11:24.49 | tmjb | how do i check if some phone call goto extension |
| 11:25.50 | tmjb | ups context |
| 11:34.08 | tmjb | I need to alllow from one one special dialout context (extensions.conf) how can i check if the phone is the right phone then give him special dialout |
| 11:35.07 | tmjb | oh bloody me |
| 11:35.19 | tmjb | for the sip phone i can specify a context |
| 11:35.33 | tmjb | and probably should overide the default context |
| 11:35.36 | tmjb | nice |
| 11:42.50 | xheliox | EmleyMoor: Polycom 430 |
| 11:43.50 | xheliox | ~grandstream |
| 11:44.02 | jbot | extra, extra, read all about it, grandstream is the Yugo of VoIP hardware. Run. Run away now. |
| 11:45.28 | tmjb | is this good way to user dial out Dial(Zap/1/${EXTEN}&mISDN/1-3/${EXTEN}) |
| 11:46.01 | *** join/#asterisk ltd (n=z@202-161-14-136.dyn.iinet.net.au) |
| 11:46.04 | tmjb | jbot, not bloat yugo i am from country that make is it |
| 11:46.07 | jbot | tmjb: that's too long |
| 11:46.31 | tmjb | jbot,:) jut kidding it is a garbage car |
| 11:46.32 | jbot | tmjb: I think you lost me on that one |
| 11:47.00 | tmjb | jbot,but it still on the streets |
| 11:47.02 | jbot | So we want to be stupid, right, still on the streets ? Fine, pushups, right now. OK, side straddle hops, right now. OK, run in place, right now. I didn't say stop! |
| 11:47.05 | tmjb | hard life |
| 11:47.44 | *** join/#asterisk oej (n=olle@apollo.webway.se) |
| 11:48.40 | tmjb | anyway what did you meen buy too long |
| 11:49.01 | xheliox | tmjb: jbot isn't a person, it's a bot. hence the name. |
| 11:49.29 | tmjb | oh stupid me |
| 11:50.15 | *** join/#asterisk toot (n=toot@84.19.255.123) |
| 11:50.30 | tmjb | ~cisco |
| 11:50.32 | jbot | cisco makes the routers that move the pr0n across the internet at a very high rate of exchange. a company full of patent protected punks! |
| 11:50.44 | gambolputty | ~polycom |
| 11:50.45 | jbot | [polycom] the manufacturer of one of the best IP phones in the market. http://polycom.com - Note: Here is where you can get some downloads: http://www.polycom.com/resource_center/0,,pw-6812-12612,00.html |
| 11:50.53 | gambolputty | ~snom |
| 11:50.58 | tmjb | ~linksys |
| 11:50.59 | jbot | methinks linksys is a tool of satan |
| 11:51.03 | gambolputty | nothing yet I guess |
| 11:51.29 | xheliox | Snom is like all German products. High quality, but wacky engineering. :) |
| 11:51.34 | gambolputty | ? |
| 11:51.36 | gambolputty | how so? |
| 11:51.43 | tmjb | wacky |
| 11:52.20 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 11:52.55 | xheliox | Everything is just counter intuitive on the Snom's - just like driving a VW. :) When I drive a VW, there's knob where I expect a lever, a switch where I expect a button.. same thing with Snom. :) |
| 11:53.20 | xheliox | Metaphorically speaking. |
| 11:54.16 | xheliox | Well, I'm not actually speaking about botton arrangements -- nevermind. :) |
| 11:57.39 | tzafrir | jbot, Snom is like all German products. High quality, but wacky engineering. :) |
| 11:57.41 | jbot | okay, tzafrir |
| 11:57.44 | tzafrir | ~snom |
| 11:57.47 | jbot | from memory, snom is like all German products. High quality, but wacky engineering. :) |
| 11:57.48 | xheliox | lol |
| 11:58.02 | *** join/#asterisk Ebola (n=Ebola@host86-136-130-202.range86-136.btcentralplus.com) |
| 11:58.28 | tzafrir | ~sipura |
| 11:58.30 | jbot | sipura is probably selling out to Cisco... fags. |
| 11:58.49 | xheliox | oh c'mon now. |
| 11:58.54 | toot | i love me Snom's - 320's/360's are great phones :) |
| 11:59.13 | tzafrir | xheliox, if you want it, edit the entry |
| 11:59.17 | xheliox | in the age of nappy headed ho political correctness, I object ;) |
| 12:00.52 | xheliox | jbot, sipura has been aquired by cisco, see ~cisco |
| 12:03.02 | tzafrir | jbot, no sipura has been aquired by cisco, see ~cisco and ~linksys . |
| 12:03.45 | tzafrir | jbot, no, sipura is <reply>has been aquired by cisco, see ~cisco and ~linksys . |
| 12:03.47 | jbot | okay, tzafrir |
| 12:03.53 | tzafrir | ~sipura |
| 12:03.55 | jbot | has been aquired by cisco, see ~cisco and ~linksys . |
| 12:04.03 | xheliox | those fags. |
| 12:04.12 | tzafrir | jbot, no, sipura is <reply>Sipura has been aquired by cisco, see ~cisco and ~linksys . |
| 12:04.14 | jbot | tzafrir: okay |
| 12:04.23 | tzafrir | ~sipura |
| 12:04.25 | jbot | Sipura has been aquired by cisco, see ~cisco and ~linksys . |
| 12:05.54 | xheliox | gracias. |
| 12:09.44 | xheliox | woohoo - tornadoes. |
| 12:12.16 | gmorg | anybody have any experience with the aastra 57i ? |
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| 12:17.06 | Blackvel | hi. anyone from Darmstadt/Germany and interested in freelance consulting for a customer? |
| 12:20.36 | EmleyMoor | ~aastra |
| 12:20.58 | EmleyMoor | Hmmm... no comment, eh? |
| 12:21.05 | xheliox | Blackvel: I'd be happy to consult, but I'm no where near Germany, nor do I speak German. |
| 12:21.39 | EmleyMoor | Darmstadt as in Chesterfield's twin town? |
| 12:22.15 | Blackvel | nah germany (near Frankfurt) |
| 12:22.25 | EmleyMoor | Yes, that's the one |
| 12:22.43 | Blackvel | got an inquery for some web tools + asterisk programming |
| 12:23.05 | Blackvel | could forward to someone who is interested ;) |
| 12:23.20 | Blackvel | but its all on-site unfortunately |
| 12:23.33 | EmleyMoor | There's only two things I know about Darmstadt - that it's twinned with Chesterfield and has a chemical element named after it |
| 12:24.03 | Blackvel | btw |
| 12:24.23 | Blackvel | is there an irc bot which can announce joining ppl about the job? :) |
| 12:24.51 | Blackvel | but hmm..most ppl are from the US/UK I expect |
| 12:25.16 | xheliox | Blackvel: I'd post on the asterisk-biz list. Lots of consultants monitor that. |
| 12:25.35 | Blackvel | ahh good tip |
| 12:25.55 | EmleyMoor | How critical is the latest update to 1.2? |
| 12:26.12 | Blackvel | maybe I should ask the recruiter agency why local consulting is required |
| 12:26.45 | EmleyMoor | I'm on 1.2.13 from Debian |
| 12:26.46 | xheliox | Blackvel: Yeah, you'll find a lot more resources if you can do most of it remotely. |
| 12:31.52 | tzafrir | The Etch package still lacks the SIP fixes of 1.2.16 and .1.2.17 |
| 12:31.52 | tzafrir | Sorry |
| 12:31.52 | *** join/#asterisk ToyMan (n=Stuart@user-12lcqol.cable.mindspring.com) |
| 12:38.45 | *** join/#asterisk jmls (n=asterisk@host86-135-47-194.range86-135.btcentralplus.com) |
| 12:38.55 | jmls | hey guys |
| 12:39.29 | jmls | I want to rerecord the "1" "2" "3" ... "0" sounds, but not overwrite the defaults |
| 12:39.50 | jmls | so, I've recorded them into a custom directory /var/lib/asterisk/sounds/custom |
| 12:40.04 | jmls | I was hoping to be able to do the following: |
| 12:40.21 | jmls | exten => foo,1,Set(CHANNEL(language)=custom) |
| 12:40.37 | jmls | exten => foo,2,SayDigits(1234567890) |
| 12:41.00 | jmls | however, I get no errors, but still get the default Allison sounds for the digits. |
| 12:41.18 | jmls | anyone got any clues on what I'm doing wrong ? |
| 12:41.20 | jmls | tia |
| 12:52.38 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@cpe-24-162-48-94.houston.res.rr.com) |
| 12:54.51 | tmjb | does any know if bt200 support callerid but NAMES |
| 12:56.01 | *** join/#asterisk djs_2_6 (n=DJS@cpe-071-077-048-198.nc.res.rr.com) |
| 12:56.36 | tmjb | i only see digits |
| 13:14.02 | *** join/#asterisk Pazzo (n=ugelt@dialin-225136.rol.raiffeisen.net) |
| 13:23.40 | *** join/#asterisk p0g0 (n=pogo@madwifi/support/p0g0) |
| 13:29.29 | *** join/#asterisk coppice (n=chatzill@94.143.17.210.dyn.pacific.net.hk) |
| 13:33.32 | *** join/#asterisk _Raptor_ (i=sirasenn@faui08r.informatik.uni-erlangen.de) |
| 13:33.42 | _Raptor_ | hello |
| 13:34.06 | _Raptor_ | how can i tell my asterisk to ignore a SIP Error 415 (unsupported media)? |
| 13:35.50 | _Raptor_ | i can establish a connection but after about 40 seconds my provider (arcor) seems to send an billing information packet and the asterisk answers with 415 and then arcor sends bye |
| 13:38.51 | *** join/#asterisk tsurko (n=tsurko@77.70.24.142) |
| 13:41.05 | _Raptor_ | here some detail: http://pbot.rmdir.de/12e5ccf8f56f8c93db7b974f7af1e267 |
| 13:41.14 | _Raptor_ | any ideas? |
| 13:49.20 | *** join/#asterisk lee_is_me (n=chatzill@12-227-176-77.client.mchsi.com) |
| 13:53.46 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 13:55.43 | lee_is_me | Has anyone seen the problem where you park a call, but then feature button no longer works once that call is taken out of parking? |
| 13:56.21 | *** join/#asterisk infernix (i=nix@unaffiliated/infernix) |
| 13:56.39 | *** part/#asterisk infernix (i=nix@unaffiliated/infernix) |
| 13:56.49 | lee_is_me | whois lee_is_me |
| 14:00.35 | *** join/#asterisk `p4r14h`work (n=josh@72.22.238.36) |
| 14:03.53 | *** part/#asterisk reber (n=reber@cl-158.dub-01.ie.sixxs.net) |
| 14:04.32 | Sweeper | lee_is_me: what phone? |
| 14:04.56 | lee_is_me | Sweepter: last I tried with a grandstream |
| 14:05.19 | lee_is_me | ...budgetone |
| 14:06.45 | *** join/#asterisk tmjb (n=tmjb@212.200.241.47) |
| 14:07.09 | Sweeper | sorry, never used one \o |
| 14:08.27 | *** join/#asterisk tb0301s (n=sowa@e178036040.adsl.alicedsl.de) |
| 14:08.48 | lee_is_me | I just tried it with a polycom 301 and it exhibits the same behavior...so prolly not the phone |
| 14:09.59 | *** join/#asterisk anonymouz666 (n=anonymou@200.218.193.6) |
| 14:10.38 | anonymouz666 | is it possible to use GoogleTalk in version 1.2? is there a backport? |
| 14:25.53 | *** join/#asterisk msetim (n=msetim@200.138.105.6) |
| 14:37.24 | *** join/#asterisk eltech (i=G00Ds@ool-457c94a3.dyn.optonline.net) |
| 14:39.51 | *** join/#asterisk ManxPower (n=manxpowe@dpc67142183150.direcpc.com) |
| 14:44.44 | *** join/#asterisk wunderkin (n=kev@dslstat-ppp-95.fastq.com) |
| 14:45.14 | *** join/#asterisk Greek-Boy (n=grb@196.46.105.176) |
| 14:45.42 | Greek-Boy | so what is the difference between 1.4 and 1.2? |
| 14:45.56 | _Raptor_ | 0.2 |
| 14:46.06 | Greek-Boy | loo |
| 14:46.08 | Greek-Boy | lol |
| 14:46.15 | Greek-Boy | i mean why did ast branch off? |
| 14:49.16 | *** join/#asterisk etfonhomey (n=etfonhom@74-140-213-69.dhcp.insightbb.com) |
| 14:50.18 | Strom_M | Greek-Boy: there's an UPGRADE and README file in the 1.4 source branch which will explain the differences |
| 14:50.31 | Greek-Boy | ok thanks |
| 14:50.50 | Greek-Boy | is 1.4 suitable for a production environment? |
| 14:51.35 | Strom_M | I run it at home without much incident; my clients are all still on 1.2 |
| 14:52.12 | anonymouz666 | I wonder if I can use gtalk on 1.2 |
| 14:57.00 | *** join/#asterisk marty-ott-athome (n=me@host-209-50-87-122.dyn.295.ca) |
| 14:58.02 | marty-ott-athome | Good morning if anyone's around - wondering if my Montreal Asterisk support guy is around? |
| 14:59.47 | marty-ott-athome | well... if anyone sees this message... I have a question on how SIP operates. |
| 14:59.49 | *** join/#asterisk jmacz (n=jmacz@201.244.171.105) |
| 15:00.13 | marty-ott-athome | See.. my understadning of SIP is that if point A wants to talk to point Z - point A registers with Point B (ASterisk server first). |
| 15:00.16 | Strom_M | marty-ott-athome: I don't think we need to be in Montreal to answer questions about SIP |
| 15:00.31 | marty-ott-athome | Then, once the call is established (Strom_M - that would be great)... |
| 15:00.47 | marty-ott-athome | The call is supposed to be direct between the SIP endpoints (without going through the server) |
| 15:00.56 | Strom_M | eh, not quite. |
| 15:01.01 | marty-ott-athome | My problem is that my calls seem to be going through my server right now... ohh really.. |
| 15:01.49 | Strom_M | That's one way of doing it, but unless there's a good reason to do it that way, the calls should still go through the server. the point-to-point model, for example, will fail if both phones are behind separate NAT devices |
| 15:02.02 | Strom_M | the point-to-point model will also prevent proper call accountig |
| 15:02.12 | marty-ott-athome | because my issue right now is that my server is at a location on a fibre optics pipe but where bandwidth is maxed out - Ohh... all my end points are NOT behind nat. |
| 15:02.37 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 15:02.39 | marty-ott-athome | Strom_M - I see so all in all - it's best not to do the point-to-point model then? |
| 15:03.16 | Strom_M | if your bandwidth is maxed out, you shouldnt be doing VoIP in the first place |
| 15:03.38 | marty-ott-athome | yes - but i thought it would be used just for call setup and teardown .... |
| 15:03.47 | marty-ott-athome | What I will have to do is move the server.. |
| 15:03.52 | marty-ott-athome | pretty much it.. |
| 15:05.52 | Greek-Boy | does asterisk need to be reinstalled after a kernel upgrade? |
| 15:06.17 | marty-ott-athome | Strom_M: So, it's best not to do point-to-point model then? |
| 15:06.28 | marty-ott-athome | how do most people do it? |
| 15:06.35 | Strom_M | Greek-Boy: if you're running Zaptel, then you need to recompile Zaptel against the new kernel |
| 15:06.51 | Strom_M | marty-ott-athome: it depends |
| 15:06.54 | Greek-Boy | k |
| 15:06.56 | Greek-Boy | thanks |
| 15:07.04 | Strom_M | what function is the box serving? what are you trying to accomplish overall? |
| 15:07.27 | marty-ott-athome | Hmm... it handles voice-mail and just the call registrations and teardowns |
| 15:07.45 | marty-ott-athome | I do need cdr |
| 15:07.53 | marty-ott-athome | actually - no I don't |
| 15:08.11 | marty-ott-athome | Sorry.. we just go along with our SIP Peer's CDRs |
| 15:08.38 | lee_is_me | Anyone seen humming/background from Telco POTS lines? |
| 15:09.58 | lee_is_me | I thought it was my phones, but then plugged a regular Conair type phone directly to the jack and get the humming noise there. |
| 15:10.05 | Strom_M | marty-ott-athome: how many peers are you supporting, and what do you mean "we just go along with our SIP peer's CDRs" |
| 15:10.18 | Strom_M | lee_is_me: yeah, it happens sometimes |
| 15:10.37 | Strom_M | how many thousand feet long is your copper loop? |
| 15:10.46 | lee_is_me | I'm calling Verizon about it tomorrow, but was curious as to the cause |
| 15:11.10 | lee_is_me | Not sure, it's in my home office so standard copper line coming in from the street. |
| 15:11.23 | marty-ott-athome | Right now.. we have 1 user but we have about 1000 potential customers from serveral appartment buildings. What I mean is we don't deliver the calls to POTS.. we simply forward them to our SIP peering partner if you will. |
| 15:21.21 | ManxPower | lee_is_me: Usually a the hum is a harmonic of 60Hz and is caused by a partial short to ground of one of the pairs |
| 15:21.28 | ManxPower | the telco can fix it |
| 15:22.13 | ManxPower | sorry, one of the wires in the pair |
| 15:22.36 | Greek-Boy | Strom_M marty-ott-athome can achieve what he wants by setting up SIP reinvites? |
| 15:22.49 | lee_is_me | ManxPower: thanks. |
| 15:23.36 | *** join/#asterisk denon (n=denon@tooth.decay.org) |
| 15:23.36 | *** mode/#asterisk [+o denon] by ChanServ |
| 15:24.01 | lee_is_me | I have way too much to learn... |
| 15:24.25 | *** join/#asterisk Strom_M (n=strom@netblock-66-159-243-60.dslextreme.com) |
| 15:25.25 | blitzrage | you can setup the reinvites with Asterisk as long as the media doesn't have to hit his local network at all |
| 15:25.33 | marty-ott-athome | yes... my sip.conf already has canrevinvite = yes |
| 15:25.53 | blitzrage | in 1.4 you'll want directrtpsetup=yes as well |
| 15:26.06 | Strom_M | but what about lol=very |
| 15:26.08 | blitzrage | then verify with a 'sip debug' and 'rtp debug' |
| 15:26.18 | marty-ott-athome | blitzrage: does it need to be setup at the clients as well? |
| 15:26.21 | blitzrage | Strom_M: thought that was an obvious option to have |
| 15:26.32 | blitzrage | marty-ott-athome: yes -- and neither end can be behind NAT |
| 15:26.40 | Strom_M | blitzrage: you'd be surprised |
| 15:26.46 | marty-ott-athome | No ends are behind NAT, so that's cool |
| 15:26.56 | marty-ott-athome | and I dont' care about CDR |
| 15:27.18 | blitzrage | you'll still get CDR |
| 15:27.24 | blitzrage | that's what signalling is for |
| 15:27.24 | *** join/#asterisk Nand0 (n=Nando@unaffiliated/nand0) |
| 15:27.30 | blitzrage | you're just redirecting the media |
| 15:28.16 | marty-ott-athome | My only potential issue is you said "hit his local network"... we have a VOIP gateway at a building with 24 ports on it (it acts like 24 sip clients).. if two clients want to talk to each other off the same voip gateway.. that woulc be an issue? |
| 15:30.08 | ManxPower | marty-ott-athome: no |
| 15:31.25 | marty-ott-athome | ok cool - right now it looks like our calls are still going through the SIP Server .. which is bad for us. But we only setup canreinvite on the server.. didn't realize we had to set it up at the clients as well. |
| 15:32.03 | ManxPower | marty-ott-athome: only the audio goes direct. in 1.2 there is no easy way to tell if the audio is going direct. |
| 15:32.33 | ManxPower | also if the legs of the calls use different codecs or if asterisk thinks it needs to listen to the stream they won't go direct either |
| 15:32.37 | blitzrage | there's no 'rtp debug' in 1.2? |
| 15:32.53 | ManxPower | blitzrage: try it |
| 15:33.01 | blitzrage | or if you have a flag in Dial() which needs to listen for DTMF (like tT or wW) |
| 15:33.08 | blitzrage | ManxPower: I don't have 1.2 anymore |
| 15:33.19 | ManxPower | or if you have asterisk recording calls |
| 15:33.36 | marty-ott-athome | yeeash.. ok well there's not transcoding.. |
| 15:33.50 | marty-ott-athome | What about voicemail? |
| 15:34.00 | ManxPower | nifty, there is an rtp debug |
| 15:34.00 | marty-ott-athome | I'll have to check any special flags in Dial() |
| 15:34.07 | ManxPower | marty-ott-athome: what about it? |
| 15:34.07 | marty-ott-athome | cool.. |
| 15:34.08 | blitzrage | marty-ott-athome: think about it for 2 seconds -- anytime Asterisk needs to deal with audio, there will be media |
| 15:34.36 | blitzrage | same thing with DTMF which travels in the RTP out of band (assuming rfc2833) |
| 15:34.46 | marty-ott-athome | Well, can I do direct RTP with Asterisk doing voicemail? That shouldn't be a problem... |
| 15:35.17 | ManxPower | blitzrage: technically since INFO or rfc2833 is our of band, asterisk should not need to break reinvites for that. |
| 15:35.22 | ManxPower | it does, IIRC |
| 15:35.33 | marty-ott-athome | blitzrage: So, am I asking for trouble trying to implement reinvite - how do you guys do it? |
| 15:35.38 | ManxPower | marty-ott-athome: direct RTP between where and where. |
| 15:35.47 | blitzrage | marty-ott-athome: I do it like I said -- I don't listen for DTMF on all calls |
| 15:35.53 | marty-ott-athome | two SIP endpoints |
| 15:36.03 | ManxPower | if you want asterisk to record a message then audio will go to asterisk |
| 15:36.09 | blitzrage | ManxPower: but out of band in rfc2833, but still carried in the RTP packets |
| 15:36.24 | ManxPower | marty-ott-athome: the call has not been established if a call goes to voicemail |
| 15:36.48 | blitzrage | Asterisk isn't magic... if there is media involved that Asterisk needs to deal with, it needs to go through Asterisk |
| 15:37.13 | blitzrage | only when 2 end points which are off your Asterisk's network, and have external IPs are talking to each other directly, can you reinvite the media |
| 15:37.20 | marty-ott-athome | User at Apt. A off a vOIP gateway calls someone at a pots line, the call registers with our Asterisk server which in turn establishes a connections with our VOIP Peer (which has the PRI connectivities) ... the voip gateway at apt A. should be direct RTP to our VOIP peer.. |
| 15:37.23 | ManxPower | marty-ott-athome: direct RTP is not really going to help you much anyway. |
| 15:37.43 | blitzrage | marty-ott-athome: oh-- asterisk has the PRI? then you're dealing with media no matter what |
| 15:38.07 | blitzrage | oh -- the peer has the PRI |
| 15:38.07 | blitzrage | I misread |
| 15:38.07 | ManxPower | blitzrage: no he uses an ITSP |
| 15:38.07 | marty-ott-athome | no . our asterisk servers does not.. yeah |
| 15:38.14 | marty-ott-athome | ITSP? |
| 15:38.19 | blitzrage | ~itsp |
| 15:38.22 | jbot | somebody said itsp was Internet Telephony Service Provider. An ITSP is a "VoIP Phone Company" |
| 15:38.25 | ManxPower | Internet VoIP provider |
| 15:38.36 | marty-ott-athome | oh yes... our parnters if the ITSP |
| 15:38.46 | marty-ott-athome | but our our partners don't have access to the particular clients.. |
| 15:38.56 | marty-ott-athome | au revoir blitzrage |
| 15:39.31 | *** join/#asterisk vjfromgt (n=vijay_0@user-387g9k7.cable.mindspring.com) |
| 15:39.32 | marty-ott-athome | but again, that's 2 SIP endpoints of public IPs that could talk directly to each tohers so I would think direct RTP could work. |
| 15:39.50 | ManxPower | marty-ott-athome: I can't imagine any situation where direct audio is going to help you with your setup |
| 15:40.13 | ManxPower | unless the SIP devices are NOT on your network |
| 15:41.01 | marty-ott-athome | it bypasses our SIP server - it's direct.. put it this way... our end client is in Toronto and our SIP peer partner is in Toronto and the Internet link between them is direct (fibre optic direct link on the Bell backbone). |
| 15:41.10 | marty-ott-athome | Where it the SIP server? In Ottawa. |
| 15:41.27 | marty-ott-athome | So, why have the calls go from Toronto .. to Ottawa only to go back to Toronto? |
| 15:41.37 | ManxPower | marty-ott-athome: since the two endpoints are NOT on your network then it will help |
| 15:42.10 | marty-ott-athome | Ottawa is still the same ISP but this ISP does not have a direct link to Toronto.. so yes, 2 different networks. |
| 15:42.26 | marty-ott-athome | And, the bandwidth where our server is located is very high.. |
| 15:43.37 | marty-ott-athome | Mind you, if someone in Ottawa calls someone in Ottawa.. we have the exact opposite problem. The call has to go to Toronto first and then our SIP peer provider send the call on their network back to OTtawa to be delivered on a local PRI. :) |
| 15:44.32 | marty-ott-athome | The main thing here is I'm trying to get away from the actual calls from going through the server because the bandwidth there is very high and we don't really have anywhere else to put it. At this location, it's on a stable high performence fibre optics link. |
| 15:45.01 | marty-ott-athome | We could put the server off a 2nd DSL line dedicated to voip right in our office.. |
| 15:45.30 | marty-ott-athome | btw: asterisk-1.2.13_2 is what we are currently running. |
| 15:47.25 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 15:49.58 | ManxPower | DSL? You are going to try to run a VoIP service over DSL? |
| 15:50.14 | ManxPower | I always knew Canadians are crazy, but not THAT crazy. |
| 15:51.57 | marty-ott-athome | No, what I said is we COULD put a srever off a 2nd DSL line dedicated to voip right in the office. |
| 15:52.30 | marty-ott-athome | But, our SIP partner (a large company) has 2 DSL lines in Oshawa supporting part of their LD service with no problems. |
| 15:52.47 | marty-ott-athome | If there's not DATA on the line.. just voice, it's doable for a certain number of users of course |
| 15:53.32 | ManxPower | I was thinking more of when the DSL goes down. |
| 15:54.21 | marty-ott-athome | oh... yeah.. well, that's why we put our Server on the a main fibre optics line - just for that reason actually. But because the calls are going trought the server right now, voice is crappy 6/10 |
| 15:54.32 | marty-ott-athome | that's why I want to get RTP direct going. |
| 15:54.46 | marty-ott-athome | bandwidth on that fibre optics line is high |
| 15:55.22 | ManxPower | remove any canreinvite=no lines from sip.conf, make sure all calls use the same codec. Make sure there are no options on the Dial line that might require Asterisk to listen to audio. Done. |
| 15:56.32 | marty-ott-athome | That's what I have right now actually but the clients at both ends of a call (in this case: Mediatrix at one end and our SIP peer at the other) need to also be setup for RTP direct - correct? |
| 16:00.07 | *** join/#asterisk nighty^^ (n=nighty@sushi.rural-networks.com) |
| 16:01.03 | marty-ott-athome | shit... and I don't see any type of "canreinvite" on the Mediatrix (which is being used as a SIP gateway) |
| 16:01.38 | marty-ott-athome | It's: Mediatrix (gateway for pots clients) ------------------ Asterisk SIP Server --------------------------- SIP Peer |
| 16:03.04 | ManxPower | marty-ott-athome: reinvites are enabled by default on almost all devices. |
| 16:03.15 | ManxPower | set up a call then do a "sip show channels" to see the codecs used. |
| 16:03.40 | marty-ott-athome | I'll have to do that when I have access to my phone.. I'm at home righ tnow but I will do that though |
| 16:04.05 | marty-ott-athome | It's good to know though that it's setup by default on most |
| 16:04.55 | *** join/#asterisk monstertruck (n=monstert@74.167.124.204) |
| 16:05.35 | monstertruck | hi, does anybody know of an ip phone that can do rj11 too? |
| 16:05.52 | Strom_M | by "rj11" do you mean "analog"? |
| 16:06.12 | monstertruck | Strom_M, yes, that can dial through ip or pstn |
| 16:06.33 | monstertruck | or, at least, receive through both |
| 16:06.37 | Strom_M | no; there's no such phone as far as I'm aware |
| 16:06.47 | *** join/#asterisk eltech (i=G00Ds@ool-457c94a3.dyn.optonline.net) |
| 16:06.50 | Strom_M | get a POTS interface card for your PBX |
| 16:06.51 | ta^3 | monstertruck: almost all ATAs can do it. |
| 16:07.05 | Strom_M | ta^3: no, only those ATAs with FXO ports |
| 16:07.42 | monstertruck | what im trying to do is have the minimum amount of devices set up |
| 16:07.48 | ta^3 | monstertruck: grandstream, and CiscoATA can do that. You configure a dialplan inside them to decide which calls got trough PSTN and which ones to your SIP gw. |
| 16:08.08 | Strom_M | ta^3: he asked about an IP phone, not a terminal adapter |
| 16:08.10 | monstertruck | im deploying that at a client site that is currently using a regular telco as a provider |
| 16:08.13 | monstertruck | and me now |
| 16:08.37 | *** join/#asterisk tzafrir_laptop (n=tzafrir@62.90.10.53) |
| 16:08.39 | monstertruck | but has to use two phones, one for each provider |
| 16:08.46 | Strom_M | monstertruck: how many users are at the client's site? |
| 16:08.52 | monstertruck | 8 |
| 16:09.03 | Strom_M | and they only have one phone line? |
| 16:09.23 | monstertruck | 10 phone lines |
| 16:09.30 | Strom_M | set up a PBX |
| 16:09.31 | monstertruck | 10 sip clients |
| 16:10.35 | Strom_M | do they use all 10 phone lines simultaneously? |
| 16:10.38 | monstertruck | i could do that, but it would up the cost from $400 in 10 ip phones plus some cabling, to over $1000 |
| 16:10.53 | *** join/#asterisk mosty (i=mostyn@60-241-198-194.static.tpgi.com.au) |
| 16:11.05 | Strom_M | and how much are they spending per month on the phone lines? |
| 16:11.16 | monstertruck | at most 8 on a busy day |
| 16:11.19 | ManxPower | monstertruck: if you consider $1,000 to be too much to install a PBX then you are far beyond our help |
| 16:11.36 | monstertruck | regular days usage is 3-5 simultaneous calls |
| 16:11.41 | ManxPower | You are suffering from CAS |
| 16:11.50 | Strom_M | Cheapass Syndrome? |
| 16:11.55 | monstertruck | :D |
| 16:12.00 | ManxPower | Strom_M: correct. |
| 16:12.25 | Strom_M | woot |
| 16:12.41 | Strom_M | monstertruck: throw a TDM800P in a PBX, dump two of the analog lines, and set up IP sets for everyone. Done, done, and done. |
| 16:13.57 | monstertruck | I'll have to take an anti-CAS pill and do that |
| 16:15.13 | mosty | how can i find out why a call dropped out? i'm entering a queue, but i get hung up on approximately 5 seconds after joining the queue. the console just says Spawn extension (queue, s, 4) exited non-zero on 'IAX2/foo-1 |
| 16:16.29 | ManxPower | There is no cure for CAS. You must let it run its course, but it is seldom fatal |
| 16:19.31 | *** join/#asterisk mog (n=mog@c-71-207-200-130.hsd1.al.comcast.net) |
| 16:19.31 | *** mode/#asterisk [+o mog] by ChanServ |
| 16:19.41 | Strom_M | it's a mog |
| 16:20.22 | *** join/#asterisk wunderkin (n=kev@dslstat-ppp-95.fastq.com) |
| 16:21.51 | *** join/#asterisk reber (n=reber@cl-158.dub-01.ie.sixxs.net) |
| 16:26.10 | mosty | the wiki mentions return codes for some application commands, but how do you tell what the value is? is it put in a variable? |
| 16:26.25 | *** join/#asterisk marthag (n=martha@NaTr-net1.ser.netvision.net.il) |
| 16:28.03 | CCFL_Man2 | anyone here have an adit 600 manual? |
| 16:28.51 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 16:30.25 | Strom_M | mosty: yes |
| 16:31.27 | mosty | Strom_C: which variable? QUEUESTATUS is empty (after my queued call dies) |
| 16:34.03 | marthag | hi. how can I see where a Zap channel is forwarded to? |
| 16:34.28 | *** join/#asterisk wunderkin (n=kev@dslstat-ppp-95.fastq.com) |
| 16:34.43 | mosty | show channels |
| 16:35.27 | marthag | there is no active call on the line. |
| 16:35.42 | marthag | I want to see where it will go if I dial |
| 16:36.00 | mosty | show dialplan |
| 16:36.07 | marthag | the line is forwarded via *72, not the dialplan |
| 16:36.36 | marthag | and zap show channel doesn't show it either |
| 16:36.51 | mosty | *72 is either done by the dialplan, or by the phone itself |
| 16:37.06 | mosty | where did you setup *72? |
| 16:37.26 | marthag | I just set cancallforward=yes in zapata.conf |
| 16:38.05 | marthag | our phones are analog phones, attached to a zhone channelbank |
| 16:39.25 | mosty | i'm not sure then. what does *72 actually do? |
| 16:40.03 | Strom_M | if you dont have *72 set in the dialplan, the zapata channel driver will handle it |
| 16:40.21 | marthag | exactly. |
| 16:40.37 | marthag | but where is it putting the info and how can I see it? |
| 16:40.43 | Strom_M | that I'm not sure of |
| 16:40.47 | [TK]D-Fender | marthag, You CAN'T. No Way. No how. Period. |
| 16:40.58 | Strom_M | I've always coded my own call forward routines :) |
| 16:42.45 | marthag | ok, thanks anyway |
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| 16:59.02 | mosty | anyone know how i can figure out why my queued call dies? |
| 16:59.04 | *** join/#asterisk Dovid (n=Dovid@85.159.160.199) |
| 16:59.19 | *** join/#asterisk menfin (n=cray@LNeuilly-152-21-127-49.w193-253.abo.wanadoo.fr) |
| 16:59.36 | Dovid | i know this ISNT the php room but anyone here know php ? |
| 17:01.32 | jmls | only in a biblical sense .. |
| 17:01.40 | jmls | ;) |
| 17:02.00 | mosty | dovid: why are you asking here? |
| 17:04.04 | mog | lol jmls |
| 17:04.44 | Dovid | hehe. no one was answering in ##php and i have been tryin to figure this out all day. but i got an answer |
| 17:05.02 | Dovid | i am a newbie at php. the begining is allways fun. |
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| 17:08.19 | tzafrir | jmls, don't f** with php! |
| 17:08.59 | tzafrir | Dovid, well, you could try asking. |
| 17:09.10 | anonymouz666 | I woke up early today, to sleep after lunch. |
| 17:10.53 | [TK]D-Fender | Dovid, just f'n ask! We mays as well shoot down your ACTUAL question rather than your request to ask :) |
| 17:10.53 | jmls | tzafrir: omg, is it not safe ? |
| 17:12.29 | [TK]D-Fender | mosty, check your queue log |
| 17:12.39 | [TK]D-Fender | mosty, and it might help if you described "dies" better. |
| 17:13.09 | blitzrage | what's a PHP? |
| 17:13.19 | blitzrage | pretty hot pork? |
| 17:13.29 | blitzrage | mmm... pork |
| 17:13.57 | anonymouz666 | PHP is nice |
| 17:14.14 | xheliox | damn, it's too late in the day to throw a pork roast in the crock pot for pulled pork :( |
| 17:14.57 | mosty | [TK]D-Fender: where is the queue log? i'm looking through the syslog with all the logging types. by "dies" i mean that moh stops, and the call is hungup |
| 17:15.32 | [TK]D-Fender | mosty, queue_log is you're using text-file output. |
| 17:15.35 | blitzrage | xheliox: that sucks |
| 17:15.52 | blitzrage | I use PHP all the time; great programming language |
| 17:15.57 | blitzrage | great documentation too |
| 17:20.29 | lee_is_me | is there any decent GUI available for polycom provisioning? |
| 17:20.44 | xheliox | vi :) |
| 17:20.49 | lee_is_me | lol |
| 17:21.59 | mosty | [TK]D-Fender: http://rafb.net/p/lSgGdM94.html |
| 17:22.55 | [TK]D-Fender | mosty, how about pastbin-ing the QUEUE_LOG |
| 17:23.58 | mosty | [TK]D-Fender: i don't have a queue log, there's nothing in logger.conf about that, how do i create it? |
| 17:24.11 | [TK]D-Fender | mosty, go find the FILE. |
| 17:24.15 | [TK]D-Fender | it exists. |
| 17:24.21 | [TK]D-Fender | get off your ass and FIND IT |
| 17:24.34 | [TK]D-Fender | "man find" |
| 17:25.18 | [TK]D-Fender | mosty, typically "/var/log/asterisk/queue_log |
| 17:27.36 | mosty | sorry, i thought all logging was done through logger.conf http://pastebin.ca/441368 |
| 17:29.12 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 17:29.47 | [TK]D-Fender | mosty, you never read ANY of the WIKI pages on this did you... |
| 17:30.10 | [TK]D-Fender | Abandon..... means the caller got pissed off at waiting and HUNG UP. |
| 17:31.07 | mosty | [TK]D-Fender: except i am the caller in this case, and i did not hang up |
| 17:31.45 | mosty | i am looking at the wiki, the documentation there on call queues is very lacking |
| 17:32.53 | mosty | i call into the queue, i hear MOH for a few seconds, and it hangs up on me |
| 17:34.06 | jmls | mosty: what's your queue command ? |
| 17:34.18 | [TK]D-Fender | Apr 16 03:03:48 club asterisk[32254]: DEBUG[495]: channel.c:2414 in set_format: Set channel IAX2/foo-4 to write format g729 |
| 17:34.26 | [TK]D-Fender | mosty, Got G.729 setup right? |
| 17:34.52 | mosty | [TK]D-Fender: i tried disabling g729, same thing happens using alaw |
| 17:34.59 | jmls | and queues.conf ? |
| 17:35.08 | [TK]D-Fender | mosty, and it'd be nice to see a concise pastebin from start to finish includeing CLI output, protocal debug and so on. |
| 17:35.19 | [TK]D-Fender | yes, queues.conf entry might help a well |
| 17:35.24 | mosty | jmls: Queue(Sales) |
| 17:35.26 | etfonhomey | [TK]D-Fender, are you ever in a good mood on here? |
| 17:35.40 | [TK]D-Fender | etfonhomey, Do you ever give me reason to be? :) |
| 17:35.40 | jmls | etfonhomey: not a good question |
| 17:36.06 | mosty | it's a realtime queue, but the same thing happens if i use a regular queue |
| 17:36.12 | jmls | mosty: add a timeout |
| 17:36.54 | mosty | jmls: same thing happens with a timeout |
| 17:37.23 | [TK]D-Fender | mosty, believe me, most people including myself don't want to hear DESCRIPTIONS. We want SLI output, log files, and CODE. |
| 17:37.29 | [TK]D-Fender | CLI* |
| 17:37.44 | jmls | we use rt queues all the time, no problems |
| 17:37.53 | jmls | (1.4) |
| 17:37.53 | etfonhomey | [TK]D-Fender, surely I have once or twice... |
| 17:38.15 | [TK]D-Fender | etfonhomey, Yes, I'm sure you have. I was happy then, wasn't I? ;) |
| 17:38.42 | etfonhomey | [TK]D-Fender, not that I've ever been able to tell. |
| 17:39.07 | jmls | fight! fight! |
| 17:39.19 | mosty | [TK]D-Fender: i'm cut/pasting right now, it's taking a while because i have a broken wrist :P |
| 17:39.32 | etfonhomey | [TK]D-Fender, nah. I'm just messing with him. I've worked with him a lot lately. |
| 17:39.55 | etfonhomey | oops |
| 17:40.02 | jmls | heh |
| 17:40.08 | etfonhomey | jmls, nah. I'm just messing with him. I've worked with him a lot lately. |
| 17:40.12 | jmls | ;) |
| 17:40.38 | *** join/#asterisk eltech (i=G00Ds@ool-457c94a3.dyn.optonline.net) |
| 17:40.38 | [TK]D-Fender | Yesterday I dropped into a local music store I bought my last guitar from just to see what they had. I'd been speccing out a costom job on-line that was going to set me back like $1200 USD and instead find a really nice stock Ibanez with all the key features I was looking for for $800 CAD. And rather than haggle on the price I'm going to go in an barter services for the sale :) |
| 17:40.53 | [TK]D-Fender | So I guess I'm having a good weekend :) |
| 17:41.14 | etfonhomey | That's what I like to hear!! |
| 17:41.15 | [TK]D-Fender | And tahts barter VS haggle for downplaying the price as it actually looks pretty decent |
| 17:41.43 | etfonhomey | Now that's a [TK]D-Fender in a good mood! |
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| 17:45.23 | mosty | [TK]D-Fender: http://pastebin.ca/441386 what else can i add that would be useful? |
| 17:47.03 | [TK]D-Fender | mosty, you may want to tweak your queue definition. "leavewhenempty=no" , "joinempty=yes" |
| 17:47.08 | mDuff | I have a few core employees going overseas for a week, and I'd like to have a way to encrypt their phone conversations. I was hoping to find SRTP support in SER or some other kind of proxy which could be placed at both ends -- but a few minutes of googling hasn't found anything Open Source and Asterisk-friendly. Do I have any options other than relying on OpenVPN? |
| 17:47.23 | *** join/#asterisk hi365 (n=hi365@mail.pcgeula.co.il) |
| 17:47.39 | hi365 | hello, anyone here using a sangoma a102d? |
| 17:47.56 | mDuff | hi365, I have one of those. |
| 17:48.29 | mosty | [TK]D-Fender: same thing happens with those settings |
| 17:48.31 | hi365 | mDuff: would be as kind as to do a zap show status? |
| 17:48.35 | *** join/#asterisk dersteer (i=dersteer@pool-71-180-106-238.tampfl.fios.verizon.net) |
| 17:49.39 | dersteer | here is my question... I'm interested in testing the new Asterisk and I want to know if I have to have a multi line VoIP phone to use a PBX system. I see some low cost phones (45 bucks) that say they are asterisk compatable but I want to make sure they can transfer between extensions |
| 17:49.49 | mDuff | hi365: not in-channel (since it's multiple lines); let me do that out-of-band. |
| 17:49.54 | [TK]D-Fender | mosty, never seen a [general] section before. put it in the queue def direct please. |
| 17:50.07 | hi365 | mDuff: sure |
| 17:50.32 | [TK]D-Fender | mosty, (those settings). I don't beleive they apply to that settings. |
| 17:50.56 | jmls | [TK]D-Fender: I have a [general] in my queues.conf |
| 17:51.07 | jmls | in fact, it's the only section ;) |
| 17:51.12 | [TK]D-Fender | jmls, I corrected my phrasing |
| 17:51.46 | mDuff | dersteer: You can use features.conf to do transfers even without the phone supporting it. |
| 17:52.01 | mDuff | dersteer: it's just a little nicer/cleaner through the phone UI. |
| 17:52.10 | *** join/#asterisk eltech (i=G00Ds@ool-457c94a3.dyn.optonline.net) |
| 17:52.11 | jmls | mosty: does SIP/205 ever answer ? |
| 17:52.16 | mDuff | dersteer: that said, stay away from cheap phones if they're for business users who care about speakerphone working adequately. |
| 17:52.19 | [TK]D-Fender | dersteer, I would seriously avoid buying crap phones.... |
| 17:52.36 | jmls | and what does ${QUEUESTATUS} hold ? |
| 17:52.39 | dersteer | http://www.voipsupply.com/product_info.php?products_id=40 is this one crap? |
| 17:52.59 | etfonhomey | If it's $45 it is crap, I would say. |
| 17:53.03 | [TK]D-Fender | dersteer, Absolute garbage |
| 17:53.05 | jmls | yuk |
| 17:53.07 | [TK]D-Fender | ~gs |
| 17:53.18 | jbot | gs is, like, South Georgia and the South Sandwich islands, or ghostscript. GrandSuck phones are cheap junk which should be avoided with extreme prejudice |
| 17:53.18 | dersteer | what would you recommend me to start out with? |
| 17:53.19 | jmls | had three of them. |
| 17:53.19 | mosty | jmls: no, it's the middle of the night, nobody is at work except me |
| 17:53.19 | dersteer | I'm open to recommendations |
| 17:53.19 | [TK]D-Fender | see above ^^^^^ |
| 17:53.25 | [TK]D-Fender | dersteer, ... |
| 17:53.27 | mDuff | dersteer: We've been through GXP-2000s and SPA841s, and right now we're buying mostly Snom 360s. They're expensive, but the users like 'em. |
| 17:53.28 | [TK]D-Fender | ~phones |
| 17:53.30 | jbot | i heard phones is http://bani.anime.net/phones/. SIP Hardphones in order of quality/suggestability : Polycom (any), Aastra 480i, Cisco 7940(+), Linksys SPA-9XX, Snom |
| 17:53.35 | *** part/#asterisk hi365 (n=hi365@mail.pcgeula.co.il) |
| 17:53.38 | jmls | so, the only member in the queue doesn't answer ? |
| 17:53.41 | [TK]D-Fender | dersteer, Polycom > all |
| 17:53.50 | [TK]D-Fender | dersteer, www.telephonydepot.com |
| 17:53.51 | dersteer | thanks all :) |
| 17:53.57 | [TK]D-Fender | dersteer, www.atacomm.com |
| 17:54.06 | etfonhomey | [TK]D-Fender, is that where you buy from? |
| 17:54.26 | [TK]D-Fender | jmls, Looks like his queue is kicking because he put queue pecific settings under [general] where they don't apply |
| 17:54.36 | [TK]D-Fender | etfonhomey, I've bought from all of them before |
| 17:55.07 | etfonhomey | [TK]D-Fender, do you have a preferred vendor? Or do you just go with who is cheapest when you're ready to buy some? |
| 17:55.35 | [TK]D-Fender | etfonhomey, So far I haven't been disappointed with any of them, so I go for price. its a commodity good..... |
| 17:55.38 | mDuff | [TK]D-Fender: Dunno. Some of our users have commented on the speakerphone on our snoms being comparable if not better than that on our (analog, older) polycom conference phones. |
| 17:55.46 | mosty | [TK]D-Fender: with those settings in the queue def it did the same thing |
| 17:55.58 | [TK]D-Fender | etfonhomey, telephonydepot has the best pricing in general, but Atacomm lists the newest models. |
| 17:56.11 | [TK]D-Fender | mosty, add protocol debug.... |
| 17:56.12 | jmls | mosty, have you pasted, line by line, the exact copy of your queues.conf ?? |
| 17:56.31 | jmls | (and reloaded asterisk ?) |
| 17:56.50 | [TK]D-Fender | mosty, pastebin your new queues.conf |
| 17:56.55 | jmls | ;) |
| 17:58.20 | mosty | http://pastebin.ca/441410 yes i reloaded |
| 17:58.57 | dersteer | [TK]D-Fender: in a business setting (with 8 lines) would you recommend getting phones that all support 8 lines? or can some of them be cheaper 2 line models and still be able to perform all functions? |
| 17:59.03 | dersteer | sorry for the n00b questions |
| 17:59.57 | mDuff | dersteer: Why do you want a single phone to log onto 8 accounts? The most we've ever needed is two (so our support staff can have a personal line as well as the support line) |
| 17:59.59 | [TK]D-Fender | dersteer, You do not yet understand the CONCEPT of "lines" with regards to phones.... |
| 18:00.20 | dersteer | thats probably true... lol |
| 18:00.21 | [TK]D-Fender | dersteer, how many CALLS do you think an individual will be on at any one time? |
| 18:00.28 | dersteer | 1 |
| 18:00.41 | [TK]D-Fender | dersteer, then the phone is irrelevent to the number of lines your PBX has |
| 18:00.55 | dersteer | awesome |
| 18:00.57 | [TK]D-Fender | dersteer, How many phones do you need? |
| 18:01.01 | dersteer | 35 |
| 18:01.09 | dersteer | (with spairs) |
| 18:01.23 | dersteer | the company has 8 lines currently |
| 18:01.31 | dersteer | and a old worthless PBX system |
| 18:02.14 | [TK]D-Fender | dersteer, Can you run a unique Rj45 to each desk for their phones to use? |
| 18:02.28 | dersteer | I could but want to avoid it |
| 18:02.53 | [TK]D-Fender | dersteer, So you expect to run them in-line with PCS with a pass-through port? |
| 18:03.07 | CCFL_Man2 | anyone play with an adit 600? |
| 18:03.29 | dersteer | yes.. I expected the phone I picked to have a 2 port ethernet that could pass thru the net to the PC connected |
| 18:03.45 | dersteer | each workstation has a networked machine currently |
| 18:03.55 | CCFL_Man2 | dersteer: cisco phones can do that |
| 18:04.13 | dersteer | polycom can't? |
| 18:04.22 | CCFL_Man2 | not sure |
| 18:05.32 | [TK]D-Fender | dersteer, yes they can. All but the IP 320 |
| 18:06.17 | mDuff | That's not necessarily a good idea, though. Means that a failure on the phone can knock out the workstation. |
| 18:06.39 | mDuff | Particularly a bad idea with cheap phones -- we had that happen a lot with the GXP-2000s crashing. |
| 18:06.44 | wunderkin | the phone only needs power |
| 18:07.06 | [TK]D-Fender | dersteer, Go issues plugging a power brick at each desk for their phone? |
| 18:07.14 | wunderkin | yeah if it resets that can f you up for a sec... but... if the phones crash... you shouldn't be using them.. |
| 18:07.50 | [TK]D-Fender | dersteer, and/or can you envision $40 for a 24 port PoE switch to power them by? |
| 18:08.16 | mDuff | wunderkin: the gxp-2000s crashed mostly when they got too much network traffic. So -- put them on their own switches and not in-line with the workstations, and they *mostly* worked. |
| 18:08.19 | [TK]D-Fender | mDuff, See jbots previous message about GS |
| 18:08.28 | wunderkin | heh :D |
| 18:08.54 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 18:08.55 | dersteer | [TK]D-Fender 40 bucks :o |
| 18:09.02 | [TK]D-Fender | dersteer, ? |
| 18:09.04 | [TK]D-Fender | der for? |
| 18:09.16 | [TK]D-Fender | dersteer, err $440 <- |
| 18:09.21 | dersteer | oh... ok |
| 18:09.22 | dersteer | lol |
| 18:10.23 | [TK]D-Fender | wunderkin, cheap whore == $60/h ;) |
| 18:12.28 | [TK]D-Fender | dersteer, For your average users : http://voipstore.atacomm.com/Shops/ViewItem.aspx/27934028032-58274372864.htm |
| 18:12.55 | [TK]D-Fender | dersteer, to replace your current switches : http://voipstore.atacomm.com/Shops/ViewItem.aspx/27934028032-57838514688.htm |
| 18:14.35 | [TK]D-Fender | dersteer, You can save 25$/phone (35 * 25 = 875$) by getting this instead : http://voipstore.atacomm.com/Shops/ViewItem.aspx/27934028032-54293451520.htm |
| 18:14.51 | [TK]D-Fender | dersteer, And invest the saving in doing your wiring PROPERLY for the phone network. |
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| 18:47.47 | jmls | has anyone an example of how transfer() would work ? |
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| 18:49.02 | jmls | also, can you use a variable in features.conf ? |
| 18:49.09 | Swabby- | I'm building a Asterisk server for a small nonprofit. We'll have 8 seats..should i use asterisk or trixbox? |
| 18:49.53 | Swabby- | Also is TDM30B decent for 3 incoming lines? (Analog) |
| 18:53.16 | tzafrir_laptop | Swabby-, if you use TrixBox you use Asterisk |
| 18:53.30 | tzafrir_laptop | Hence your question is incorrect |
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| 19:17.11 | tmjb | is there some way I can enable constant sound on sip phone or disable silence |
| 19:20.20 | wunderkin | look for silence suppression |
| 19:20.22 | tmjb | the problem is when the speak is does not talk i get silence |
| 19:20.28 | tmjb | oh tnx |
| 19:21.42 | *** join/#asterisk DrukenLPY (n=jdumais@CPE000854ddcdb1-CM00137189cb0c.cpe.net.cable.rogers.com) |
| 19:26.08 | *** join/#asterisk DrukenHME (n=jdumais@74.115.68.66) |
| 19:26.10 | Swabby- | tzafrir: so trixbox is just a more packaged implementation |
| 19:26.35 | Strom_M | ~trixbox |
| 19:26.46 | jbot | Trixbox is a full linux distro that includes *, FreePBX, and other 3rd party add-ons. It is these things on top of * which make it seriously painful to support and hence you will find little help here for it. Try asking in #freepbx , or their forums at http://www.trixbox.org/modules/newbb/ |
| 19:26.46 | tmjb | wunderkin,i have Silence Suppression to no in my bt200 but i can still hear it ? any ideas |
| 19:26.46 | tmjb | ~bt200 |
| 19:27.47 | *** join/#asterisk zigman (i=zigman@irc.zigman.de) |
| 19:28.08 | *** join/#asterisk saftsack (n=oliver@p54A70CE6.dip0.t-ipconnect.de) |
| 19:28.24 | Swabby- | jbot: thats' what i thought..its probably better to use the more popular asterisk appraoch |
| 19:28.45 | Swabby- | Is the TDM30B a good approach to bringing in the Incoming lines that are analog? |
| 19:32.32 | Strom_M | Swabby-: jbot isn't a person |
| 19:32.37 | Strom_M | it's a computer program |
| 19:32.40 | Strom_M | hence "bot" |
| 19:33.15 | Strom_M | Swabby-: the TDM30B is only for connecting three telephone sets to...if you want to connect analog lines, you should use a TDM03B |
| 19:35.22 | nasls_lsa | HAhah |
| 19:35.24 | nasls_lsa | :) |
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| 19:39.29 | *** join/#asterisk bawb2 (n=bawb2@ip48200.estcmp.ku.edu) |
| 19:48.41 | tmjb | what codec is the best to setup for sip and grandstream |
| 19:49.35 | tmjb | i GSM but is this a good codec |
| 19:50.16 | Strom_M | tmjb: depends on your needs |
| 19:50.34 | Strom_M | are you going for stingy bandwidth usage, or are you going for calls that sound like real telephone calls? |
| 19:51.41 | tmjb | i have 100MB/s network and about 30 phones mostly calls are for PBX |
| 19:52.42 | Strom_M | use ulae |
| 19:52.44 | Strom_M | er, ulaw |
| 19:52.58 | Strom_M | (or alaw if you're not in North America) |
| 19:53.19 | *** join/#asterisk _Roman (n=roman@87.254.76.159) |
| 19:54.01 | tmjb | ulaw |
| 19:54.03 | tmjb | ok tnx |
| 19:54.42 | _Roman | I am wanting to use festival to read some text from within a perl agi script. (I am using the latest svn version of asterisk). Is there a better/easier way to getting festival to speak other then using stream_file and text2wave > |
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| 20:07.42 | *** join/#asterisk Dovid (n=Dovid@85.159.160.199) |
| 20:07.51 | Dovid | can anyone explain this: |
| 20:08.04 | Dovid | SELECT Apr 15 16:07:14 NOTICE[30520]: frame.c:183 __ast_smoother_feed: Dropping extra frame of G.729 since we already have a VAD frame at the end |
| 20:08.27 | Qwell | Dovid: turn off VAD |
| 20:08.56 | *** join/#asterisk Zefk (n=Zefk@81.181.249.106) |
| 20:09.02 | Dovid | where is the VAD setting ? |
| 20:09.06 | Dovid | sip.conf ? |
| 20:09.10 | Qwell | in your client |
| 20:09.14 | *** join/#asterisk Mad|Cow (n=madcow@74.94.5.97) |
| 20:12.33 | Mad|Cow | I just installed a 12 port FXO card in my PBX (that has 8 POTs lines conntected to it). I'm having trouble with my dial plan... when I use the dial command, I can only make it dial the specific channel, how do I make it automatically select a fee channel and dial? |
| 20:12.36 | Dovid | Qwell: do u know where VAD would be on the Polycoms ? |
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| 20:26.56 | tzafrir_laptop | anybody here uses res_seroconf or res_bonjour ? |
| 20:27.08 | tzafrir_laptop | any of those two usable? |
| 20:28.00 | tzafrir_laptop | astmasters.net does not seem to have any useful content any more |
| 20:29.09 | tzanger | hmm sip hinting on 1.4.x isn't working? |
| 20:29.26 | tzanger | IP501 has sip.conf entry of [201] and context=polycom |
| 20:29.40 | tzanger | core show hints hsows it but it's always idle |
| 20:29.52 | tzanger | [polycom] has 201,hint,SIP/201 |
| 20:29.55 | tzanger | should be fine |
| 20:32.04 | [TK]D-Fender | tzanger, You'll need to add "call-limit=99" to your sip.conf entry |
| 20:32.04 | tzanger | yeah, I remmber now |
| 20:32.04 | tzanger | I *just* remembered that as you wrote it :-) |
| 20:32.05 | tzanger | thank you |
| 20:32.11 | tzanger | how are things with you today? |
| 20:33.39 | [TK]D-Fender | *yawn* Family has headed back to T-O and my place is in better shape than its ever been, just taking it easy. |
| 20:33.58 | *** join/#asterisk dniel (n=daniel@host109.190-30-86.telecom.net.ar) |
| 20:34.02 | [TK]D-Fender | tzanger, Tomorrow taking a look at a guitar I saw yesterday before making an offer on it |
| 20:34.18 | tzanger | [TK]D-Fender: awesome |
| 20:34.51 | [TK]D-Fender | tzanger, not bad I guess. Still shopping for a new sword and budgeting... |
| 20:35.40 | tzanger | hmm, still showing idle, does the call-limit mean the #of active calls until it's showing as busy? |
| 20:35.43 | tzanger | [TK]D-Fender: which guitar?? |
| 20:36.14 | tzanger | .. sword? |
| 20:37.00 | [TK]D-Fender | tzanger, http://www.thomann.de/gb/ibanez_rgt42dxipt.htm?partner_id=79570 |
| 20:37.04 | *** join/#asterisk pariah (n=j0sh@unaffiliated/pariah) |
| 20:37.34 | tzanger | that looks like flame |
| 20:37.45 | tzanger | and the frets are very cool too |
| 20:38.18 | dniel | do anybody speak spanish? |
| 20:38.51 | dniel | here there are a lot of ops |
| 20:38.53 | [TK]D-Fender | tzanger, And for the sword I'm considering this : http://www.casiberia.com/product_details.asp?id=SH1018 |
| 20:39.19 | [TK]D-Fender | tzanger, big selling points are the floating trem, 24 fret, and neck-though-body |
| 20:39.26 | tzanger | my lord what on earth are you going to do with the sword? |
| 20:39.32 | tzanger | do you have a vermin problem or something? |
| 20:39.32 | Qwell | [TK]D-Fender: yes, everybody needs a katana |
| 20:39.40 | Qwell | $849? good lord |
| 20:39.47 | *** part/#asterisk dniel (n=daniel@host109.190-30-86.telecom.net.ar) |
| 20:39.50 | [TK]D-Fender | tzanger, Thought you knew I'd been studying Katori Shinto for over a year now.... |
| 20:40.19 | [TK]D-Fender | qwell, thats MSRP. I've seen it around $525 US retail |
| 20:40.27 | Qwell | still :p |
| 20:40.38 | [TK]D-Fender | qwell, and the kind of sword I'd LIKE to buy would cost FAR more than that. |
| 20:40.45 | Qwell | naturally |
| 20:40.46 | tzanger | [TK]D-Fender: no, I did not know |
| 20:41.13 | [TK]D-Fender | qwell, tzanger : My Oni Forge Bushi ($545) http://aocomputing.net/bushi/ |
| 20:41.22 | Qwell | saw |
| 20:41.39 | [TK]D-Fender | I'm looking for something with a longer tsuka (handle) length that will be a bit more agile. |
| 20:42.39 | Dovid | anyone know if there is a polycom channel on here ? |
| 20:43.06 | Dovid | nm. its #polycom |
| 20:44.39 | tzanger | ok this buddy watch is driving me nuts |
| 20:44.50 | tzanger | even call-limit=1 and when the phone's on a call it shows as idle |
| 20:44.59 | [TK]D-Fender | tzanger, and set the type to "peer" |
| 20:45.13 | tzanger | [TK]D-Fender: I'm still al long way to go before I am able to really use the features you describe on a guitar |
| 20:45.19 | tzanger | Ive kind of hit a plateau |
| 20:45.22 | tzanger | gotta get beyond it |
| 20:45.30 | [TK]D-Fender | Dovid, voice.vadEnable = "0" |
| 20:46.57 | [TK]D-Fender | tzanger, I've been playing for about 17 years now, and have had my last one for about 5 ($300 +/-). I was preparing to shell out around $1200 USD for a Carvin custom job :) |
| 20:47.30 | [TK]D-Fender | tzanger, The one I linked you will set me back about $800 |
| 20:48.12 | _Roman | I am using a perl agi script to interact with festival. The following works kind of. It will not seem to accept a string with a space in it, it will only speak the first work. I am using this command: print "EXEC festival 'test message'"; Can anyone make a suggestion how to make this work? |
| 20:50.26 | tzanger | [TK]D-Fender: indeed |
| 20:50.55 | tzanger | [TK]D-Fender: I like my V70CE but I *REALLY* liked the Taylor 110 |
| 20:51.04 | tzanger | funnily enough, I did NOT like their more expensive ones |
| 20:51.09 | tzanger | the 110 is their cheapest, about $700 |
| 20:51.45 | [TK]D-Fender | tzanger, ah, an acoustic player... |
| 20:51.54 | tzanger | *nods* |
| 20:52.03 | mrdigital | are wireless voip phones good? |
| 20:52.14 | [TK]D-Fender | tzanger, as long as it isn't C&W I can forgive you ;) |
| 20:52.24 | tzanger | [TK]D-Fender: hahahaha |
| 20:52.24 | [TK]D-Fender | ~wifisip |
| 20:52.26 | jbot | Wi-Fi SIP phones suck. All of them. HARD. Some only slightly less than others... |
| 20:52.29 | *** join/#asterisk saftsack (n=oliver@p54A7214E.dip0.t-ipconnect.de) |
| 20:52.35 | tzanger | no I want to do more bluesy-rock stuff |
| 20:53.03 | [TK]D-Fender | tzanger, I'm guessing you'd be more a Les Paul kind of guy.... |
| 20:53.45 | tzanger | my boss has a few les pauls |
| 20:53.51 | tzanger | I like the wider neck for sure |
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| 21:04.59 | blitzrage | Les Paul's are nice |
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| 21:11.07 | *** join/#asterisk jpablo (n=jpablo@linuxuanl.org) |
| 21:11.42 | jpablo | hey people, anyone knows what set of unicall libs are supposed to work ? |
| 21:11.55 | jpablo | I'm in a undefined symbol nightmare |
| 21:17.38 | jpablo | anybody ? |
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| 22:00.07 | tmjb | anyone know what is this chan_sip.c:4848 process_sdp: Unsupported SDP media type in offer: image 5004 udptl t38 |
| 22:03.51 | *** join/#asterisk sahafeez (n=sahafeez@ip68-6-223-156.sd.sd.cox.net) |
| 22:04.36 | [TK]D-Fender | tmjb : looks like you're receiving a fax via T.38 |
| 22:04.51 | [TK]D-Fender | tmjb, But are connecting the call to an endpoint that doesn't support it. |
| 22:05.03 | tmjb | yes but it does not work |
| 22:05.06 | [TK]D-Fender | tmjb, * only supports T.38 in pass-through mode |
| 22:05.38 | tmjb | i got handy 286 should i set it in pass-through mode |
| 22:06.33 | tmjb | i have setup for both faxes t38pt_udptl=yes |
| 22:06.53 | tmjb | without canreinvite = yes |
| 22:10.03 | menfin | Is asterisk able to playback mp3 file with Playback() application ? |
| 22:12.36 | tmjb | [TK]D-Fender,tnx works great in pass-trouth mode |
| 22:13.43 | [TK]D-Fender | menfin, yes |
| 22:15.00 | tmjb | [TK]D-Fender,thank you for such fast reply very much is a about 1 a clock in NIS [Serbia] i am bit tired |
| 22:15.20 | [TK]D-Fender | tmjb, np |
| 22:16.01 | *** join/#asterisk jovannotti (i=Toche@200.26.134.180) |
| 22:16.17 | jovannotti | Hi to all, I need some help with TC400B |
| 22:16.25 | jovannotti | someone of you have used that card ? |
| 22:17.10 | jovannotti | hello ? |
| 22:17.23 | *** join/#asterisk tr2x (n=alvar@80-218-185-55.dclient.hispeed.ch) |
| 22:17.26 | menfin | [TK]D-Fender, something is wrong in my extensions.conf, because Playback doesn't find my mp3 file, extension is required for this type of file ? |
| 22:18.18 | *** join/#asterisk djs_2_6 (n=DJS@cpe-071-077-048-198.nc.res.rr.com) |
| 22:19.23 | [TK]D-Fender | menfin, You need to ahve compiled the asterisk-addons package to be able to playback MP3's |
| 22:19.43 | jovannotti | some help about TC400B ? |
| 22:20.24 | menfin | thanks [TK]D-Fender |
| 22:20.38 | menfin | i'm looking some webpage about this |
| 22:21.26 | Strom_M | jovannotti: what's the trouble? |
| 22:22.34 | jovannotti | Strom, I coudl install the card, g729 works fine, when I use g723 the conversation is very bad |
| 22:22.48 | jovannotti | I installed first 1.4.1 and then 1.2.17 and the problem remains |
| 22:23.02 | Strom_M | what do you mean "the conversation is very bad" |
| 22:23.45 | jovannotti | I mean, you can understand the other person, like so much noise when the other person talks |
| 22:24.06 | Strom_M | what codecs are the other legs of the call using? |
| 22:24.13 | jovannotti | only occurs with g723, with g729 is ok |
| 22:24.20 | jovannotti | I set g729 and works fine |
| 22:24.32 | jovannotti | but when I set g723 the problem occurs |
| 22:24.52 | Strom_M | when you use g723, what codecs are the legs of the call between the phones and the box using? |
| 22:24.53 | *** join/#asterisk dj-fu (n=aj@unaffiliated/dj-fu) |
| 22:25.03 | jovannotti | I am using a cisco 2811 as the remote point, which supports both codecs |
| 22:25.14 | jovannotti | I can see g723 |
| 22:25.22 | Strom_M | will you please answer my question? |
| 22:25.49 | jovannotti | when execute :<show channell 'channelname'>, I saw g723 and g729 respectively |
| 22:25.57 | Strom_M | well thats the thing |
| 22:26.08 | Strom_M | you can't have multiple transcode steps and expect it to sound any good |
| 22:26.27 | Strom_M | because what's happening is that asterisk is transcoding g729 -> slin -> g723 |
| 22:26.42 | jovannotti | no no , let me explain |
| 22:26.44 | Strom_M | and you get multiple layers of artifacting |
| 22:26.50 | jovannotti | this is the scenario |
| 22:27.42 | jovannotti | PHONE ---> FXS(Cisco) ---> G723 ---> asterisk box ---> FXO |
| 22:27.58 | jovannotti | the other is the same with g729, there are 2 scenarios |
| 22:28.03 | jovannotti | g729 works fine |
| 22:28.09 | Strom_M | ok, and what's connected to the FXO? |
| 22:28.15 | jovannotti | but g723 doesnt |
| 22:28.59 | Strom_M | what's connected to the FXO? |
| 22:29.00 | jovannotti | I connect a TDM2406E and then a sim card to GSM |
| 22:29.08 | Strom_M | oh christ |
| 22:29.24 | jovannotti | with g729 works fine (between cisco and asterisk) |
| 22:29.27 | *** join/#asterisk sharp (n=sharp@dsl092-234-217.phl1.dsl.speakeasy.net) |
| 22:29.28 | Strom_M | just transcode the call six more times between endpoints, and then it will sound AWESOME |
| 22:29.54 | jovannotti | and I can speak very well between my phone and my cell phone |
| 22:30.09 | Strom_M | yes, I understand that g729 works well - you don't have to keep repeating that fact |
| 22:30.16 | jovannotti | sorry |
| 22:30.59 | jovannotti | but the trascoding is just between g723 and FXO |
| 22:31.07 | Strom_M | do you have a telephone which you can connect directly to the asterisk box? |
| 22:31.40 | jovannotti | the phone is connected to the FXS port from cisco 2811 |
| 22:31.51 | Strom_M | forget it - you're not listeing to me |
| 22:33.05 | jovannotti | sorry i didnt understand, but the answer is yes |
| 22:33.10 | Strom_M | ok |
| 22:33.18 | jovannotti | I can connect an phone to Zap/1 , Yes ! |
| 22:33.23 | Strom_M | so..you have a telephone on the 2811, and another on the asterisk box |
| 22:33.38 | Strom_M | place a call directly between those using g723 and see if the call quality is acceptable to you |
| 22:33.41 | jovannotti | yes I could to configure that |
| 22:33.57 | jovannotti | ok, the idea is to isolate gsm network , right ? |
| 22:33.59 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 22:34.07 | Strom_M | no, the idea is to AVOID the gsm network |
| 22:35.15 | *** join/#asterisk kev009 (n=kev009@ip70-162-43-70.ph.ph.cox.net) |
| 22:35.24 | jovannotti | I can configure and test , but the problem is that I need to teminate calls in GSM using G723 in the asterisk box |
| 22:35.54 | Strom_M | well, instead of complaining at me, why don't you test like I'm asking you to? |
| 22:36.50 | kev009 | I need some help selecting a single and a dual FXO ATA |
| 22:37.02 | kev009 | sorry, FXS |
| 22:37.22 | Strom_M | kev009: the linksys PAP2T is a good choice |
| 22:37.40 | Strom_M | or the digium IAXy if SIP will cause problems in your network design |
| 22:37.42 | *** join/#asterisk wundaboy (n=look@adsl-67-127-57-203.dsl.pltn13.pacbell.net) |
| 22:37.56 | wundaboy | anyone here have any experience with Junction Networks? |
| 22:38.00 | kev009 | Strom_C: SIP is fine, but I need a built in ethernet switch |
| 22:38.16 | Strom_M | kev009: por que? |
| 22:38.53 | kev009 | some of the rooms in my house only have one RJ45 jack, so a built in switch like my Cisco 7960 would be desirable |
| 22:39.21 | Strom_M | i dont think any have a switch per se; they tend to have a NAT gateway |
| 22:39.27 | jovannotti | Strom, do you speak spanish ? |
| 22:39.41 | kev009 | are there any like that? (that don't do routing/NAT) |
| 22:39.41 | Strom_M | jovannotti: not enough to carry out a technical discussion, no |
| 22:39.49 | Strom_M | kev009: not that I'm aware of |
| 22:40.09 | jovannotti | ok, I am configuring the scenario , to test quality |
| 22:40.29 | kev009 | cisco 188s will do it but looking for alternatives |
| 22:41.18 | *** part/#asterisk jonmoore (n=jonmoore@c-68-53-165-155.hsd1.ky.comcast.net) |
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| 22:47.05 | *** join/#asterisk ShadowHntr (i=sentinel@wikipedia/Shadowhntr) |
| 22:48.05 | Strom_M | kev009: why not just put an IP phone in each room? |
| 22:49.36 | *** join/#asterisk edguy3 (n=edguy@host-24-225-134-24.patmedia.net) |
| 22:51.02 | *** join/#asterisk mavior (n=chatzill@81-174-8-252.f5.ngi.it) |
| 22:52.04 | mavior | hello everybody.someone could explain how to load different loadzones per channel on zaptel ? tnx |
| 22:52.33 | Strom_M | jovannotti: how's that test coming? |
| 22:52.34 | *** join/#asterisk daveburr (i=Miranda@175.sub-70-193-14.myvzw.com) |
| 22:53.37 | mavior | i mean how to set use different loadzone tones for different zap channels... |
| 22:55.27 | *** join/#asterisk asterisknerds (n=asterisk@66.7.124.15) |
| 22:55.27 | asterisknerds | <PROTECTED> |
| 22:56.52 | *** join/#asterisk nybble (n=nybble@about/apple/performa/nybble) |
| 22:58.03 | mavior | anybody on this? :) |
| 22:58.39 | Strom_M | mavior: I'm not sure that's possible |
| 22:58.59 | daveburr | what the purpose ? |
| 22:59.53 | *** join/#asterisk saftsack (n=saftsack@pD9E06E23.dip.t-dialin.net) |
| 23:00.30 | *** join/#asterisk ptblank (n=MURDER1@cpe-75-84-216-188.socal.res.rr.com) |
| 23:00.30 | mavior | really don't know how to understand this extract from zaptel.conf , anyway: "# Finally, you can preload some tone zones, to prevent them from getting |
| 23:00.32 | mavior | # overwritten by other users (if you allow non-root users to open /dev/zap/* |
| 23:00.33 | mavior | # interfaces anyway." |
| 23:01.29 | mavior | Strom ,have a read here http://www.voip-info.org/wiki-Asterisk+config+zaptel.conf,under "Edit your zaptel.conf file" |
| 23:02.06 | Strom_M | mavior: yeah, you're barking up the wrong tree |
| 23:02.07 | Strom_M | why do you want to set multiple tone zones on the same card anyway? |
| 23:02.07 | mavior | sorry the url is this one http://www.voip-info.org/wiki-Asterisk+config+zaptel.conf |
| 23:02.51 | mavior | mmhh....just because i want different tones for different channels |
| 23:02.52 | uhb | can a digium x100p card be used to feed a house's regular phone line? (cut the incoming line, send that to asterisk and then back out) |
| 23:03.46 | mavior | on that page is written "You can load more than one set of tones into an interface to allow you to generate tones from different zones by giving a comma-separated list of two letter country codes"...really don't know how to load them separately anyway |
| 23:03.47 | Strom_M | uhb: no; you need an FXO port and an FXS port |
| 23:04.04 | uhb | ok, and the 100p is just FXO, right? |
| 23:04.17 | Strom_M | it's just garbage is what it is |
| 23:04.45 | *** part/#asterisk plantseeker (n=chatzill@host81-154-189-53.range81-154.btcentralplus.com) |
| 23:05.01 | Strom_M | get a TDM11B at minimum :) |
| 23:05.09 | [TK]D-Fender | Strom_M, No, tell us how you REALLY feel! ;) |
| 23:05.21 | jovannotti | strom, I could take 20 min, you'll be here ? |
| 23:05.38 | Strom_M | [TK]D-Fender: ok! |
| 23:05.48 | Strom_M | jovannotti: what's taking so long? |
| 23:06.08 | mavior | even though on the last lines of zaptel.conf is written " We are all done with our channel parameters, so now we specify what channels they apply to " and then this command "channels=1-4" i'm working with a tdm400p card |
| 23:06.35 | jovannotti | a person is moving to the site where the phone is located, I am doing remote configuration |
| 23:08.11 | Strom_M | terrific |
| 23:08.52 | mavior | Strom_M: "yeah, you're barking up the wrong tree" what ? |
| 23:13.32 | mavior | anybody please? |
| 23:14.32 | *** join/#asterisk sahafeez (n=sahafeez@ip68-6-223-156.sd.sd.cox.net) |
| 23:21.42 | *** join/#asterisk tb0301s (n=sowa@e178061088.adsl.alicedsl.de) |
| 23:28.35 | mavior | is it possible that nobody knows if is possible to have different zap channels configured with different tones ? |
| 23:36.08 | *** join/#asterisk lee_is_me (n=chatzill@12-227-176-77.client.mchsi.com) |
| 23:38.51 | *** join/#asterisk aptura (n=jondoe@S010600a0c93f6f7e.vs.shawcable.net) |
| 23:43.03 | blitzrage | anyone buy cellphones from cti-miami.com ? |
| 23:43.13 | blitzrage | looks like they have the E61i I wanna buy |
| 23:43.28 | aptura | nope |
| 23:44.53 | aptura | what makes it different |
| 23:45.03 | blitzrage | camera built in and much nicer casing |
| 23:47.42 | tmjb | is there better way to do this Dial(Zap/1/${EXTEN}&mISDN/1-3/${EXTEN}) |
| 23:48.05 | tmjb | the thing is I want to call zap1 then if bussy allow call to misdn1,2,3 .. |
| 23:48.34 | [TK]D-Fender | tmjb, that does not even do what you are thinking |
| 23:49.33 | aptura | whats a common cause of error updating boot rom for ip500. |
| 23:50.08 | [TK]D-Fender | tmjb, You would need to _XXXX.,1,[exten],1,Dial(Zap/1/${EXTEN}) then _XXXX.,2,[exten],1,Dial(mISDN/1-3/${EXTEN}) |
| 23:50.35 | [TK]D-Fender | aptura, corrupted file or interrupted file is the only way I know for BR problems |
| 23:51.20 | jpablo | stupid rhino E1 card |
| 23:51.27 | jpablo | doesn support unicall correctly |
| 23:51.31 | tmjb | [TK]D-Fender, oh tnx |
| 23:51.39 | JT | morning |
| 23:52.00 | mavior | anybody knows how to apply different options to different channels , when configuring the zaptel.conf file? |
| 23:53.02 | JT | i'd say the answer is no by now |
| 23:54.24 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 23:56.03 | mavior | people this one is from page 360 of asterisk TFOT book : http://pastebin.ca/442023...i am guessing that have different tones for different zap channels is theorically possible!! |
| 23:56.42 | mavior | ok the url is this http://pastebin.ca/442023 |
| 23:56.56 | aptura | TK okay because this is a ongoing issue. The bootrom.ld versions dont exactly match my phones version. Running 1.5.3 on my phone. Tried all bootrom.lds and using this one version stating mac.cfg is erroring out. |
| 23:58.08 | aptura | actually its 3.1.0 |
| 23:58.59 | *** join/#asterisk astranik (n=smirnoff@bas13-ottawa23-1177810290.dsl.bell.ca) |
| 23:59.58 | [TK]D-Fender | aptura, What SIP ver? |