| 00:01.02 | *** join/#asterisk kb3ien (n=kb3ien@ool-45766a2d.dyn.optonline.net) |
| 00:01.13 | kb3ien | how does gotoiftime get the timezone? |
| 00:01.57 | Katty | hi. |
| 00:02.19 | *** join/#asterisk Xetrov` (n=xetrov@unaffiliated/xetrov/x-827361) |
| 00:03.31 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 00:05.31 | *** join/#asterisk manxpower (n=EWieling@69.73.94.162) |
| 00:06.41 | manxpower | ~answers |
| 00:06.42 | infobot | i heard answers is Asterisk Book: http://astbook.asteriskdocs.org & SIP w/ NAT: http://www.aocomputing.net/?p=3 & Wiki: http://www.voip-info.org & docs: /path/to/src/asterisk/doc & 1.4 .vs. 1.6: /path/to/src/asterisk/UPGRADE*.txt |
| 00:08.02 | *** join/#asterisk seanbright (n=sean@asterisk/contributor-and-bug-marshal/seanbright) |
| 00:09.46 | LiNeTuX | Katty: I'll say it. Hello! |
| 00:10.14 | Katty | (= |
| 00:10.38 | LiNeTuX | just got done reading about the MOH fiasco. |
| 00:10.46 | LiNeTuX | Gotta love the lawyers. |
| 00:13.53 | Katty | scowls. |
| 00:14.05 | Katty | paging eppigy |
| 00:14.23 | rob0 | Guy walks into a bar with his alligator. "Do you serve lawyers here?" he asks. Barkeep says, "Sure." "Okay, I'll have a beer, and a lawyer for my alligator." |
| 00:15.23 | Katty | grins. |
| 00:16.33 | *** join/#asterisk joako (n=ston3d@opensuse/member/joak0) |
| 00:16.39 | Nugget | -- |
| 00:16.39 | Nugget | On the first day of the trial, the judge addressed the courtroom: "I have to declare an interest in this case. Last week, the plaintiff sent me a check for $10,000 to find in his favor. |
| 00:16.43 | Nugget | Two days later, I received $20,000 from the defendant to find in his favor. |
| 00:16.46 | Nugget | I have therefore sent $10,000 back to the defendant and can now try the case without bias." |
| 00:16.49 | Nugget | -- |
| 00:16.56 | Katty | hehehehehe |
| 00:18.09 | joako | Nuigget: If you need to paste more than 3 lines please use a pastebin |
| 00:18.20 | Katty | infobot: seen seanmh? |
| 00:18.24 | infobot | seanmh <n=johndoe@198.59.129.24> was last seen on IRC in channel #asterisk, 133d 9h 37m 1s ago, saying: 'I'll let you know when we have 1.6 support and then I will await my cookie :D'. |
| 00:18.24 | Nugget | take your ritalin, joako. |
| 00:18.30 | Katty | woah |
| 00:18.33 | Katty | 133 days ago?! |
| 00:18.50 | joako | Nugget: I use other remedies for my ADHD |
| 00:19.15 | Katty | i would say Nugget has seniority here. |
| 00:19.19 | Katty | so..yeah. |
| 00:19.30 | Katty | please go troll somewhere else (= |
| 00:19.43 | joako | I wasn't even being serious |
| 00:19.47 | Katty | excellent. |
| 00:20.10 | Katty | jaytee: poke. |
| 00:20.26 | jaytee | Katty: poke poke |
| 00:20.28 | Nugget | totally ignoring the fact that I didn't even past "over 3 lines" too! :) |
| 00:20.39 | Katty | jaytee: :> |
| 00:20.42 | Katty | jaytee: herroes. |
| 00:20.51 | jaytee | how are ya hon? |
| 00:20.58 | Katty | exhaustimicated :< |
| 00:21.04 | Katty | spent all day in software training. |
| 00:21.06 | Katty | how's you? |
| 00:21.31 | jaytee | I'm packing for a trip to MA. |
| 00:21.40 | joako | Well it shows 5 lines here... but anyways, how about them asterisks? |
| 00:21.59 | Katty | jaytee: oh? amish visitation? |
| 00:22.31 | *** join/#asterisk matt_d (n=matt@70.134.79.103) |
| 00:22.32 | jaytee | no, not many Amish in Massachusetts. Just vacation and my mom's 91st birthday |
| 00:22.41 | Katty | goodness gracious. |
| 00:22.44 | matt_d | hello everyone |
| 00:22.45 | Katty | i hope i live to be that old. |
| 00:22.54 | Katty | jaytee: is she in good health? |
| 00:23.00 | jaytee | she's still going strong and her mind is still sharp. |
| 00:23.38 | Katty | that's wonderful! i'm so happy for her. |
| 00:23.44 | jaytee | we have good genes and longevity in my fam, plus above average intelligence. |
| 00:23.45 | LiNeTuX | jaytee: curious if you don't mind me asking. what does she do to keep her mind active? |
| 00:24.25 | jaytee | LiNeTuX, she reads alot, walks often and loves crossword puzzles......and arguing about politics :-) |
| 00:24.30 | manxpower | The infobot never forgets |
| 00:24.40 | jaytee | ~botsnack |
| 00:24.40 | infobot | thanks, jaytee |
| 00:24.54 | jaytee | he's a good bot |
| 00:29.50 | Katty | i let the ferrets out to play :> |
| 00:29.57 | Katty | the basement is a mad house. |
| 00:30.59 | drfreeze | Hey, can someone help jog me out of this sip registration problem |
| 00:31.26 | drfreeze | I have a standard setup, but can't get the sip phone to register. It's got to be something simple I'm missing |
| 00:31.29 | drfreeze | No matching peer found |
| 00:31.46 | jaytee | borrows [TK]D-Fender's ClueBat(tm) and whacks drfreeze upside the head. |
| 00:31.51 | jaytee | Did that help? |
| 00:32.28 | jaytee | sip hard phone or softphone? |
| 00:32.53 | drfreeze | here is the sip.conf: http://pastie.textmate.org/private/eghtrlpmfsfkioqydsefrg |
| 00:33.13 | jaytee | yes, thanks for that but what about an answer to my question |
| 00:33.15 | drfreeze | jaytee: need harder hit please |
| 00:33.28 | drfreeze | jaytee: polycom 550 |
| 00:33.34 | [TK]D-Fender | takes his ClueBat (tm) back and thwaps drfreeze for himself... |
| 00:34.13 | jaytee | drfreeze, same network and is this the only phone you've got? |
| 00:34.14 | drfreeze | [TK]D-Fender: it's ok. I put my qtrback padding on before I came into the room. :) |
| 00:34.21 | drfreeze | yes |
| 00:34.34 | drfreeze | Aug 18 19:32:11] NOTICE[15479]: chan_sip.c:15933 handle_request_register: Registration from '<sip:510@10.0.0.160>' failed for '10.0.0.175' - No matching peer found |
| 00:34.35 | jaytee | did you use the web interface? |
| 00:34.44 | drfreeze | for what? |
| 00:34.54 | jaytee | for configuring the phone |
| 00:35.04 | drfreeze | jaytee: I did some tweeking with the web interface |
| 00:35.08 | drfreeze | did that mess me up? |
| 00:35.57 | jaytee | did you do a sip reload after adding the phone to sip.conf? |
| 00:36.49 | drfreeze | yes |
| 00:36.55 | drfreeze | the overrides have the following: <OVERRIDES reg.1.callsPerLineKey="3" reg.1.lineKeys="1" saf.2="SoundPointIPWelcome.wav" tcpIpApp.sntp.gmtOffset="-21600" up.welcomeSoundOnWarmBootEnabled="1" up.backlight.idleIntensity="1" up.backlight.onIntensity="3"/> |
| 00:37.17 | jaytee | ah, so you're using FTP |
| 00:37.52 | drfreeze | jaytee: yes. vsftpd. set the sntp server in there |
| 00:39.25 | *** join/#asterisk bbryant (n=brett@c-68-59-22-114.hsd1.sc.comcast.net) |
| 00:40.20 | jaytee | drfreeze, do you have a local-settings.cfg file in your FTP directory for the phones? |
| 00:41.37 | drfreeze | jaytee: I have localcfg set in one of my cfg files |
| 00:41.51 | drfreeze | not file named specifically local-setting.cfg |
| 00:41.55 | drfreeze | *not |
| 00:41.56 | drfreeze | *no |
| 00:42.08 | manxpower | Losing local settings can be a hassle in the short run, but I think it is a good idea in the long run. |
| 00:42.09 | *** join/#asterisk coppice (n=chatzill@52.204.17.210.dyn.pacific.net.hk) |
| 00:43.04 | drfreeze | manxpower: not sure what file that is. I just downloaded the 3.1.3cfg files from polycom and have a few to poing to the right server and set the date zone |
| 00:43.24 | jaytee | ok, well you probably followed a different guide then I used. You need a statement in sip.cfg or local-settings.cfg voIpProt.server.1.address="your asterisk server IP address" |
| 00:44.22 | jaytee | drfreeze, here's a copy of mine to look at. http://pastebin.ca/1534519 |
| 00:45.38 | *** join/#asterisk talntid (n=asd@c-98-247-117-146.hsd1.wa.comcast.net) |
| 00:45.51 | drfreeze | this looks interesting |
| 00:45.51 | drfreeze | <--- Transmitting (no NAT) to 10.0.0.175:5060 ---> |
| 00:45.51 | drfreeze | SIP/2.0 404 Not found |
| 00:46.08 | jaytee | and I include that file on the CONFIG_FILES= line in the master cfg file for that phone. the file that is only the mac address of the phone with a .cfg extension |
| 00:46.33 | jaytee | what IP is 175? the server or the phone? |
| 00:46.40 | jaytee | the phone right? |
| 00:46.53 | drfreeze | jaytee: thanks for the link. |
| 00:47.00 | drfreeze | yes |
| 00:47.26 | jaytee | is your server firewall open on port 5060? |
| 00:47.45 | drfreeze | should be totally disabled, will double check |
| 00:49.43 | drfreeze | no selinux |
| 00:49.47 | drfreeze | no iptables |
| 00:50.22 | *** join/#asterisk rene- (n=rene@200.79.232.176.cable.dyn.cableonline.com.mx) |
| 00:50.29 | *** part/#asterisk rene- (n=rene@200.79.232.176.cable.dyn.cableonline.com.mx) |
| 00:51.07 | drfreeze | Via: SIP/2.0/UDP 10.0.0.175;branch=z9hG4bKdc64dbc96C94513A;received=10.0.0.175 |
| 00:51.32 | matt_d | brb |
| 00:52.17 | jaytee | drfreeze, check the whitepaper titled Configuration File Management on Soundpoint IP Phones on this page, it has a good guide in it for setup. http://www.polycom.com/products/resources/white_papers/index.html |
| 00:52.28 | jaytee | gotta run, bbiab |
| 00:52.42 | drfreeze | thx |
| 00:56.54 | LemensTS | From reading for a few more hours and testing, it seems you can not ORIGINATE a g729 call to an itsp, and connect it to your diaplan without doing decoding because it will be in slin....? |
| 01:01.56 | *** join/#asterisk Failrar (n=Failrar@coffee.ipv6.goatse.be) |
| 01:02.23 | *** join/#asterisk s14ck (n=s14ck@190-76-67-197.dyn.movilnet.com.ve) |
| 01:04.20 | *** join/#asterisk chendy (n=chatzill@58.251.101.21) |
| 01:05.08 | LemensTS | here is a sip show channel of that call http://pastebin.com/m437e5c86 |
| 01:07.03 | *** join/#asterisk ArchGT (n=archgt@unaffiliated/archgt) |
| 01:07.09 | *** join/#asterisk joat (n=joat@ip70-174-79-200.hr.hr.cox.net) |
| 01:09.34 | KyleK | LemensTS: sounds like you should post it to a bug tracker |
| 01:12.38 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-uadkqgmhubyhghzx) |
| 01:13.12 | KyleK | hey can asterisk transcode wideband codecs without cutting down to narrowband internally? |
| 01:15.40 | *** join/#asterisk MrNaz (n=mrnaz@ppp121-44-213-216.lns10.mel4.internode.on.net) |
| 01:17.49 | *** join/#asterisk BeeBuu (n=beebuu@218.13.68.23) |
| 01:26.54 | *** join/#asterisk sjobeck (n=Adium@208-151-246-203.dq1sn.easystreet.com) |
| 01:27.24 | *** part/#asterisk sjobeck (n=Adium@208-151-246-203.dq1sn.easystreet.com) |
| 01:37.29 | *** join/#asterisk Kumbang (n=semprul@167.205.24.69) |
| 01:38.22 | KyleK | huh |
| 01:38.26 | manxpower | What the hell happened with MOH? |
| 01:38.38 | KyleK | skype for asterisk will now log into personal accounts? |
| 01:39.46 | manxpower | LemensTS: Transcoding is required if you want to do any processing of the audio like listening to DTMF, MeetMe (mixing), "r" option to Asterisk (injects ringing sound), t/T (has to listen to DTMF), transport to PSTN (has to transcode). In short if you can't transcode, most Asterisk and VoIP features are lost. |
| 01:40.51 | manxpower | KyleK: 1.6 should be able to handle wideband internally. 1.4 will always convert to narrowband internally if I recall correctly |
| 01:41.17 | manxpower | LemensTS: pastebin the CLI for the call. All of the call. |
| 01:41.39 | manxpower | then say my nick so I notice and I can stop watching me movie and look at it |
| 01:41.47 | *** join/#asterisk ecret (n=ecret@CPE001e9002348e-CM001225d8ab30.cpe.net.cable.rogers.com) |
| 01:44.40 | KyleK | okay cool |
| 01:45.13 | *** join/#asterisk KavanS (n=KavanS@static-71-117-242-28.ptldor.dsl-w.verizon.net) |
| 01:46.02 | LemensTS | manxpower: i posted it to the bug tracker like KyleK said: https://issues.asterisk.org/view.php?id=15743 |
| 01:46.08 | LemensTS | has all the info there |
| 01:46.59 | KavanS | problem: I have a hylafax daemon I use for outgoing faxes...I think I need to setup a different context to solve a problem...the problem is that I have a current failover dial and when the pstn/zaptel line 1 is in use, it goes to SIP provider from there...is there anyway someone can suggest a way I can send a "busy signal" to hylafax to sort of force it to keep redialing, and then sooner or later it gets through on pstn? |
| 01:52.32 | LiNeTuX | manxpower: MOH = license hell |
| 01:53.23 | KavanS | LiNeTuX, what about classical music? |
| 01:53.43 | LiNeTuX | http://blogs.digium.com/2009/08/18/asterisk-music-on-hold-changes/ <-- for those who haven't seen it |
| 01:54.54 | LiNeTuX | KavanS: Can be safe. Not always. |
| 01:55.31 | drmessano | Digium has this shit all wrong |
| 01:55.54 | LiNeTuX | drmessano: go on... |
| 01:55.56 | manxpower | LiNeTuX: stop answering the call |
| 01:56.05 | drmessano | If they want real MOH, royalty free.. They need to call Trent Reznor |
| 01:56.07 | manxpower | that blog post does not say WHY |
| 01:56.12 | drmessano | NiN MOH |
| 01:56.15 | LiNeTuX | heh! right o! |
| 01:56.24 | KavanS | drmessano, lol |
| 01:56.38 | drmessano | He would probably compose some badass MOH for asterisk |
| 01:56.45 | LiNeTuX | manxpower: my read is the French g'mnt, but I'm probably wrong. |
| 01:57.30 | drmessano | If you dont mind the occasional reference to pigs, violence, or fucking.. I would say NiN MOH would be a big hit amongst the overly tattooed Asterisk crowd |
| 01:57.44 | rob0 | Probably just lawyers, gov't wouldn't likely get involved. |
| 01:57.47 | drmessano | and it would be FREE |
| 01:57.50 | LiNeTuX | drmessano: I guess you haven't heard Ghosts I-IV |
| 01:58.08 | drmessano | Nope |
| 01:58.37 | KavanS | god I had that ghosts album |
| 01:58.43 | KavanS | every other album I like... |
| 01:58.54 | drmessano | I would love to be able to use all 3 versions of "Sin" for MoH |
| 01:58.59 | LiNeTuX | drmessano: mostly classical |
| 01:59.02 | LiNeTuX | ish |
| 01:59.22 | drmessano | ah |
| 01:59.23 | KavanS | drmessano, I should include internal MoH music for employees |
| 01:59.28 | KavanS | they would probably appreciate NiN :P |
| 02:00.18 | drmessano | and for Queue's, Radiohead |
| 02:01.55 | drmessano | thinks he needs a good stereo desk phone to appreciate the NiN Sin MoH |
| 02:03.16 | LiNeTuX | I just wish the PSTN would support "HD" audio... or at least "HQ" :) |
| 02:04.51 | drmessano | As an audio engineer and asterisk fanboy, I take great offense at your inference the PSTN supports "audio" |
| 02:05.00 | LiNeTuX | heh |
| 02:05.54 | KavanS | lol |
| 02:06.03 | drmessano | Perhaps "Amplitude Modulated White Noise with Noise Cancellation" |
| 02:06.45 | KavanS | can anyone suggest how to send a busy signal to my hylafax daemon when my analog line is in use? |
| 02:07.51 | *** join/#asterisk blkry (n=chatzill@96.37.27.72) |
| 02:08.34 | KyleK | odd the SFA faq kinda says that personal accounts can be used just not for skypeout |
| 02:09.09 | *** join/#asterisk Magicblaze0071 (n=sony@fl-67-235-212-177.dhcp.embarqhsd.net) |
| 02:09.10 | drmessano | oh? |
| 02:09.38 | KyleK | Will Skype for Asterisk support personal Skype accounts? |
| 02:09.39 | KyleK | For calls to landlines and mobiles (SkypeOut), only accounts created in a company's Business Control Panel will be supported. For online numbers (SkypeIn), all users will be supported. |
| 02:09.39 | Magicblaze0071 | is there someone who can give me $3/month a number using asterisk + unlimited calls to the US like skype? |
| 02:10.32 | KyleK | i guess they're circling around it, but i wish they'd just say skype to skype is or isn't supposed to work |
| 02:10.49 | KavanS | so personal will work for incoming |
| 02:10.54 | KavanS | but no outgoing unless you have a biz acct? |
| 02:11.16 | KyleK | it sounds like it but i just upgraded to the latest s/w available and no go |
| 02:12.55 | Magicblaze0071 | can one comepete in costs using asterisk with skype? |
| 02:12.56 | drmessano | ouch |
| 02:13.16 | KavanS | not sure if I wanna use them for outgoing shit |
| 02:15.37 | drmessano | I wonder why they require the accounts to fall in the BCP |
| 02:15.56 | Magicblaze0071 | drmessano: I've flushed the ignore list. I can see you and you obviously can hear me. |
| 02:16.09 | Magicblaze0071 | I've to go, will be back. |
| 02:17.41 | drmessano | Im guessing perhaps the need for an account set up in BCP lessens the abuse vector.. dunno |
| 02:17.42 | leifmadsen | you just need to make an account with the biz portal |
| 02:17.42 | leifmadsen | those accounts are still free |
| 02:18.10 | drmessano | Im still curious why |
| 02:18.11 | leifmadsen | I'm sure it has something to do with Skype's policy |
| 02:18.21 | drmessano | policy? |
| 02:18.30 | leifmadsen | policy or technology implementation |
| 02:18.41 | KyleK | policy meaning they can be as silly as they want |
| 02:18.43 | LiNeTuX | Skype probably sees Asterisk as a business tool, not personal. |
| 02:18.49 | leifmadsen | that too |
| 02:19.30 | drmessano | Maybe so... but obviously they would have some reason to have such a trivial requirement, which means its not so trivial to them |
| 02:20.13 | drmessano | Perhaps there's a higher SLA on business accounts |
| 02:20.31 | drmessano | Little bit of CYA for quality |
| 02:20.36 | LiNeTuX | If I was really after free calls, I'd be using Google Voice. But I just use my low-rate SIP provider and pay my $5/mo for my phone use. |
| 02:20.50 | leifmadsen | I'd guess it's something silly like the way the system is implemented |
| 02:20.58 | KyleK | can everyone get google voice yet? |
| 02:21.05 | leifmadsen | only in the US, and only with an invite |
| 02:21.06 | drmessano | They probably dont want to support a bunch of residential users with linux boxes complaining of quality issues on outbound calls |
| 02:21.12 | leifmadsen | which you can apply for on the google site, and get within a week |
| 02:21.13 | LiNeTuX | yeah. supposedly takes about a week to get an invite back |
| 02:21.22 | LiNeTuX | jynx |
| 02:21.23 | leifmadsen | took me like 3-4 days |
| 02:21.29 | leifmadsen | LiNeTuX: you owe me a beer |
| 02:21.34 | KyleK | damn |
| 02:21.40 | LiNeTuX | leifmadsen: I'll buy you one at Astricon. |
| 02:21.43 | KyleK | they can tell im in canada? :( |
| 02:21.48 | LiNeTuX | Yard House. |
| 02:21.48 | leifmadsen | yes |
| 02:21.51 | drmessano | Google could pwn Skype if they put Google Voice with Google Talk |
| 02:21.58 | leifmadsen | is in Canada, but got a Google Voice account anyways |
| 02:22.01 | drmessano | PSTN in and out |
| 02:22.11 | KyleK | leifmadsen: do tell |
| 02:22.14 | drmessano | Asterisk can already handle Jingle |
| 02:22.14 | leifmadsen | I'm a sneaky bastard |
| 02:22.21 | leifmadsen | chan_gtalk |
| 02:22.23 | LiNeTuX | leifmadsen: I responded to your twitter req for an invite, but you must not have gotten it.. I had an extra one. |
| 02:22.26 | drmessano | So done and done |
| 02:22.31 | *** join/#asterisk OrNix (n=ornix@78.40.81.34) |
| 02:22.35 | leifmadsen | LiNeTuX: oh really? odd |
| 02:22.43 | drmessano | Im not sure why they threw the switch |
| 02:22.46 | leifmadsen | LiNeTuX: sweet -- I like beers at astricon :) |
| 02:22.56 | leifmadsen | drmessano: eh? |
| 02:22.58 | drmessano | Im not sure why they havent threw the switch |
| 02:23.01 | *** join/#asterisk ecret (n=ecret@CPE001e9002348e-CM001225d8ab30.cpe.net.cable.rogers.com) |
| 02:23.01 | drmessano | Sorry |
| 02:23.01 | LiNeTuX | leifmadsen: Heh. I might have screwed up the DM. oh well. |
| 02:23.05 | KyleK | does the service itself track canadians or just the signing up? |
| 02:23.09 | drmessano | GV <> Google Talk |
| 02:23.17 | leifmadsen | KyleK: well, it's kinda useless unless you have a US DID |
| 02:23.33 | leifmadsen | since it is just a forwarding service, and only forwards to US numbers |
| 02:23.40 | KyleK | oh |
| 02:24.00 | leifmadsen | if you must know though, you can't sign up from Canada, so I had a friend in the US setup a web proxy for me while I did the sign up thing |
| 02:24.11 | leifmadsen | then I signed up for a free inbound DID number from washington |
| 02:24.16 | drmessano | Would seem simple to allow GoogleTalk accounts as a destination.. They just need a keypad or dialbox in one of the apps to complete it.. |
| 02:24.17 | leifmadsen | www.nerdvittles.com has all the info |
| 02:24.23 | rob0 | A US DID from ipkall costs ... yes, nothing. |
| 02:24.28 | leifmadsen | drmessano: I'm sure they are working on all of it |
| 02:24.51 | drmessano | leifmadsen: Google is slower than fucking Microsoft, and I have no idea why.. Is it the bouncy balls? |
| 02:24.52 | LiNeTuX | I use my ipkall DID w/GV. it's just a pain. |
| 02:25.09 | leifmadsen | drmessano: HUGE user base. I'm sure that's kinda an issue. |
| 02:25.12 | KyleK | drmessano: google tests everything extensively |
| 02:25.17 | KyleK | including colors |
| 02:25.20 | drmessano | hah |
| 02:25.23 | KyleK | i loved that article |
| 02:25.52 | leifmadsen | When you're Google, and you've got a huge brand name to keep in the good books, I'm sure you take things slower than everyone would like |
| 02:25.57 | drmessano | Gmail was in beta for 12 years.. Im sure they had to have gotten to a point about 11 years ago when it sucked less than Yahoo.. Thats a good sign to take it out of beta |
| 02:26.20 | drmessano | thinks "...or was it 15 years" |
| 02:26.20 | leifmadsen | I don't even want to know how much work is involved with things like Gmail and such |
| 02:26.24 | leifmadsen | 12 years? right... |
| 02:26.29 | leifmadsen | probably closer to like 3 |
| 02:26.34 | drmessano | No way |
| 02:26.34 | KyleK | 2005 yea |
| 02:26.38 | LiNeTuX | it was quite a while |
| 02:26.40 | heedly | yep |
| 02:26.42 | *** join/#asterisk chendy (n=chatzill@58.251.101.21) |
| 02:26.43 | drmessano | 2004 or 2003 |
| 02:26.45 | leifmadsen | hell, I've worked on Asterisk systems for over 2 years before |
| 02:26.47 | heedly | gmail hasn't been around long |
| 02:27.03 | KyleK | early 2004 probably |
| 02:27.04 | leifmadsen | I think people over-estimate how long these web services have been around |
| 02:27.16 | leifmadsen | I'd bet no more than 5 years |
| 02:27.19 | drmessano | My ex and I had Gmail accounts when people actually paid for invites.. that would have been 2003-2004 |
| 02:27.19 | leifmadsen | at the most |
| 02:27.41 | leifmadsen | 2003 was hardly 12 years ago |
| 02:27.49 | drmessano | Um, joke? |
| 02:27.54 | leifmadsen | NO FUCKING JOKES |
| 02:28.02 | drmessano | No shit, this is @asterisk |
| 02:28.04 | leifmadsen | </joke> |
| 02:28.07 | drmessano | or #asterisk |
| 02:28.12 | drmessano | or umm |
| 02:28.14 | LiNeTuX | There once was this guy and a sheep. |
| 02:28.30 | rob0 | Walked into a bar? And said, "Ouch!" |
| 02:28.31 | leifmadsen | So a Rabbi, a Priest, and Turtle walk into a bar |
| 02:28.58 | KyleK | between jan 15th and april 6th 2004 |
| 02:29.06 | KyleK | horray for internet archive |
| 02:29.09 | leifmadsen | nice :) |
| 02:29.17 | leifmadsen | so ummm.... 5 years :) |
| 02:29.41 | drmessano | Which domain name? |
| 02:29.43 | leifmadsen | I bet someone just forgot to change the icon last year; some intern |
| 02:30.01 | KyleK | drmessano: gmail.com |
| 02:30.01 | leifmadsen | anyways, I'm going to watch the end of my movie |
| 02:31.05 | drmessano | gmail.com wasn't there from day1 |
| 02:31.16 | drmessano | Trying to remember how we got to it |
| 02:31.24 | KyleK | googlemail.com? |
| 02:31.38 | drmessano | nah.. Was a google.com/somecrap |
| 02:31.43 | manxpower | I assume it was google.com/mail |
| 02:31.47 | LiNeTuX | gmail.google.com = 4/04 |
| 02:32.19 | LiNeTuX | http://web.archive.org/web/20040401041817/http://gmail.google.com/ |
| 02:32.26 | *** part/#asterisk LemensTS (n=customgt@adsl-70-238-169-241.dsl.stlsmo.sbcglobal.net) |
| 02:32.34 | heedly | actually gmail.com wasn't google.com when it started |
| 02:32.40 | heedly | archive.org shows '96 |
| 02:32.48 | drmessano | http://en.wikipedia.org/wiki/Gmail |
| 02:32.50 | heedly | so maybe 12 years is right |
| 02:32.52 | heedly | kk |
| 02:33.16 | drmessano | I guess mar 21, 2004 could be correct |
| 02:33.21 | LiNeTuX | launched on april 1st? really? |
| 02:33.28 | LiNeTuX | from google? |
| 02:33.32 | LiNeTuX | and people believed it? |
| 02:35.46 | drmessano | Oldest mail message I have is from 6/21/04.. I know I deleted some earlier ones |
| 02:36.03 | LiNeTuX | drmessano: Google didn't :) |
| 02:36.15 | drmessano | Except that one time? |
| 02:41.12 | *** join/#asterisk phunyguy (n=phunyguy@69.130.65.44) |
| 02:42.53 | drmessano | I cant remember or find a google link to it.. but I remember very early on there was some incident with random users having lost their entire inbox |
| 02:43.06 | drmessano | and after that, the net was flooded with gmail backup apps |
| 02:43.17 | manxpower | I used my gmail account for 2 days |
| 02:43.34 | manxpower | I'll stick to using folders, thankyouverymuch |
| 02:44.56 | *** join/#asterisk rossand (n=rossand@CPE000c413a19a3-CM00159a025ad4.cpe.net.cable.rogers.com) |
| 02:45.10 | drmessano | Labels aren't difficult to get used to, you just need to take the time to set them up, the same as you did your folder structure. I was turned off by it too, and it really took us moving to apps at work before I spent real time learning the gmail interface |
| 02:45.23 | drmessano | Before then it was just another toy and I hardly ever used my @gmail.com account |
| 02:45.35 | manxpower | Maybe so, but I prefer IMAP and a real e-mail client. |
| 02:45.39 | drmessano | But I have a bunch of domains on Apps now, using Chrome for access |
| 02:45.47 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 02:46.00 | KyleK | I like the ajaxy refreshing webmail part |
| 02:46.04 | drmessano | It was like quitting an addiction, but i've gotten used to it |
| 02:46.19 | manxpower | shudders in horror at ajax |
| 02:46.29 | drmessano | The only thing that sucks in Gmail, horribly, is the contact management |
| 02:46.46 | drmessano | Gmail is a mail client and client app, but no contact manager |
| 02:46.49 | drmessano | err |
| 02:46.54 | drmessano | Gmail is a mail client and calendar app, but no contact manager |
| 02:47.33 | LiNeTuX | drmessano: Damn - they have your domains and you're using Chrome... they must know everything about you - even your d*** size! |
| 02:47.40 | LiNeTuX | :) |
| 02:47.53 | drmessano | I dare say that the early abortion of a rolodex that MS called "Cardfile" that was the great grandfather to Outlook had better contact management than Gmail currently features |
| 02:48.45 | KyleK | they have his domains? you just have to make some MX records and a CNAME for the webmail access to use thier webmail |
| 02:48.49 | drmessano | digs up Cardfile.exe and Schedule+, then shakes his head |
| 02:49.31 | LiNeTuX | KyleK: I know. I have one of my domains pointing there too I used as an experiment when they came out with that service |
| 02:49.39 | manxpower | I use runmail.com SSL/TLS, IMAP, an OK web interface when I need it, even hosted outside the United "Warrentless Wiretap" States. |
| 02:49.57 | *** join/#asterisk moy (n=moy@CPE001cdfec4cee-CM00080dab8485.cpe.net.cable.rogers.com) |
| 02:50.16 | manxpower | not going to stop a govt wiretap, but at least they should work for it a little |
| 02:50.29 | LiNeTuX | just has his own server at a colo |
| 02:50.52 | KyleK | I don't think I'd sent anything worth wirehax0rating via email |
| 02:50.55 | Magicblaze0071 | can google voice numbers be directed to ring extensions on an asterisk server? |
| 02:50.57 | drmessano | I've grown past the whole "I need to host all my own shit" phase.. took me a long time |
| 02:51.20 | manxpower | drmessano: I liked hosting my own stuff, but it is a lot of work |
| 02:51.31 | drmessano | When my home network became 4 homes, 12 servers, VPNs, and too much time.. |
| 02:51.42 | LiNeTuX | A lot of work? WTF were you guys doing? I haven't seen my server in over 2 years :) |
| 02:51.52 | drmessano | Now my file server and my Asterisk box hum in the corner.. |
| 02:52.14 | drmessano | LiNeTuX: Actually using them? |
| 02:52.22 | LiNeTuX | drmessano: Every day :) |
| 02:52.25 | drmessano | Hosting 20 domains |
| 02:52.35 | drmessano | or closer to 25 I guess |
| 02:52.45 | LiNeTuX | doesn't do a lot of web hosting (no money in that) |
| 02:52.47 | drmessano | Had like 75 users |
| 02:52.59 | drmessano | Of course not |
| 02:52.59 | Magicblaze0071 | LiNeTuX: Can you get this to work for 25 phone numbers from google voice? |
| 02:53.00 | KyleK | LiNeTuX: servers with direct access to the internet need to be patched occasionally :) |
| 02:53.08 | Magicblaze0071 | receiving and sending calls? |
| 02:53.08 | LiNeTuX | just provides enough services to clients to pay his colo bill ($45/mo) |
| 02:53.11 | Magicblaze0071 | using astersk. |
| 02:53.18 | drmessano | Even less doing it for friends and offering free hosting for clubs you get involved in |
| 02:53.19 | manxpower | not to mention updating spam filters |
| 02:53.46 | LiNeTuX | SPAM filters are better left to the pros. |
| 02:53.58 | drmessano | Never, ever mention you know anything other than Geocities-level tech when joining a club |
| 02:54.02 | manxpower | I got less spam when I ran my own spam filters |
| 02:54.05 | drmessano | or starting one |
| 02:54.31 | drmessano | Proper answer --> "We can probably host it on google" |
| 02:54.47 | LiNeTuX | yep |
| 02:54.51 | drmessano | Google Groups, Google Apps, nothing but net |
| 02:55.54 | KyleK | drmessano: haha yea |
| 02:55.55 | drmessano | What started off 8 years ago as a learning environment turned into GoogleZilla |
| 02:56.24 | KyleK | my ham radio club is hosting its website off of a windows XP box kind of inside the municipal network |
| 02:56.40 | drmessano | I would NEVER.. NEVER host a ham club site |
| 02:56.52 | LiNeTuX | KyleK: That's a crime, isn't it (hosting on XP) |
| 02:57.02 | manxpower | it should be. |
| 02:57.07 | KyleK | hehehehehe |
| 02:57.26 | manxpower | On the other hand I'm going to hell just for running Server 2003 |
| 02:57.39 | KyleK | its apache on XP Pro |
| 02:57.58 | drmessano | Ham club hosting its site on XP.. that sounds about right.. By the way, a mouse just knocked over the coke bottle holding up your receiver in the repeater cabinet |
| 02:58.09 | drmessano | :( |
| 02:58.36 | drmessano | I guess I shouldnt talk.. My repeater is an abomination of a box |
| 03:00.57 | drmessano | Had thermal issues in the 3 foot motorola repeater cabinet we were using, so got a 25 watt motorola repeater exciter and a 250 motorola repeater exciter, running 15 watts out of the 25 watter, which is the minimum to drive the 250 watter to about 100 watts.. The whole thing is now in a 6 foot rack and weighs about 500 lbs |
| 03:01.36 | LiNeTuX | drmessano: I think that explains a lot about you. |
| 03:01.57 | drmessano | Oh? |
| 03:02.24 | LiNeTuX | drmessano: it's not really the folks in IRC that irritate you, it's the 1.21 gigawatts going into your head |
| 03:02.31 | drmessano | lol |
| 03:03.30 | drmessano | I think I tend to overengineer things |
| 03:03.39 | manxpower | Too much RF will really fuck you up. |
| 03:03.42 | LiNeTuX | join the club! |
| 03:04.17 | drmessano | There was a story a few years ago about some tower climbers almost being fried by an overzealous program director |
| 03:05.49 | drmessano | Station had tower work scheduled on their 100,000 watt stick and the program director didnt know.. So they bumped the power down to like 10% and sent the guys up.. they got within 50 feet of the antenna.. PD wakes up, cant hear the station.. Calls the transmitter site, turns the plates back on, which bumps it back to 100% by default |
| 03:05.49 | manxpower | someone turned it up to 11? |
| 03:06.07 | drmessano | Their radiation badges go off, instantly.. Radiation suits started SMOKING |
| 03:06.39 | drmessano | They decended like 250 feet in less than 30 seconds, barely came out alive |
| 03:07.02 | LiNeTuX | nice. did they take the PD and hoist him up by his balls to the top? |
| 03:07.22 | manxpower | I don't even like to be near a HughNet dish. |
| 03:07.36 | *** join/#asterisk beilabs (n=beilabs@121.216.245.186) |
| 03:07.40 | drmessano | PD's are the single point of failure for most BS at a radio station, yet they NEVER get any shit for it |
| 03:07.46 | Magicblaze0071 | drmessano: Any plans for doing constructive work in the next one month? :) |
| 03:07.51 | kb3ien | its the tissue boxes on its feet? |
| 03:08.23 | *** join/#asterisk jplank (n=gbove@cpe-098-024-169-236.carolina.res.rr.com) |
| 03:09.10 | LiNeTuX | unplanned downtime is almost directly associated with lack of change management control |
| 03:09.18 | kb3ien | cool just disabled the DND button on my polycom. my operator keeps hitting it when she wants the speaker phone. |
| 03:09.24 | LiNeTuX | erm, almost always that is |
| 03:09.43 | kb3ien | whats the good word on GoToIfTime, what timezone does it like to be in? |
| 03:09.52 | manxpower | make dial the overhead paging |
| 03:10.30 | Magicblaze0071 | Is there a chance that all the small voip users unite and compete with skype/GV? |
| 03:10.34 | KyleK | are you sure thats a mistake? ;) |
| 03:10.54 | *** join/#asterisk mindCrime (n=chatzill@cpe-076-182-103-085.nc.res.rr.com) |
| 03:11.18 | Magicblaze0071 | who is the biggest sip provider? |
| 03:11.40 | manxpower | Apparently many companies compete with the big ones. |
| 03:11.58 | drmessano | If everyone was using SIP URI, there would be nothing to compete with |
| 03:12.10 | rob0 | Anheuser Busch. |
| 03:12.13 | drmessano | </utopian rhetoric> |
| 03:12.15 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 03:12.30 | Magicblaze0071 | Here is my question: Skype can give a DID and free outbound calls for $3/month. GV can do that for free. SIP providers like Voicepulse charge $11! |
| 03:12.46 | Magicblaze0071 | drmessano: absolutely. |
| 03:12.50 | rob0 | AB is the biggest sip provider and a pretty big gulp provider too. |
| 03:12.55 | drmessano | HAHA |
| 03:13.10 | manxpower | I deal with Verizon SIP most of the time these days |
| 03:13.18 | rob0 | GV's not doing free outbound, those are limited calls. |
| 03:13.43 | drmessano | Skype has a limit of 10,000 mins/6 hours a day |
| 03:13.56 | Magicblaze0071 | rob0: I could call anyone in the US using my grandcentral -> Google voice. (At least the ones I needed to calls, I never had problems) |
| 03:13.58 | kb3ien | https://issues.asterisk.org/view.php?id=14028#c95943 is that added to the mainline yet, or not? |
| 03:13.59 | drmessano | Skype has a limit of 10,000 mins/mo and 6 hours a day |
| 03:14.03 | rob0 | drmessano, I found because of SIP attack bots, it's best to keep SIP closed to the world. |
| 03:14.16 | rob0 | (only allow known peers from outside the firewall) |
| 03:14.29 | drmessano | rob0: Just need decent security |
| 03:14.42 | Magicblaze0071 | drmessano: 10000 min/mo = $3! Are you kidding me, that is bad? |
| 03:15.00 | rob0 | The last one that hit me was a DoS, ate up all my bandwidth with registration attempts. |
| 03:15.04 | *** join/#asterisk DrCron (i=rszasz@saxonco.com) |
| 03:15.18 | drmessano | Didnt say it was bad, its just not unlimited.. and once you hit 10,000 minutes, you pay $.02 a min |
| 03:15.19 | manxpower | Your target customer should not be the ones that want the lowest price. |
| 03:15.44 | drmessano | The 6 hours a day kinda irks me |
| 03:16.08 | LiNeTuX | rob0: fail2ban |
| 03:16.34 | rob0 | LiNeTuX: yeah, or iptables -m recent |
| 03:16.43 | Magicblaze0071 | drmessano: For $3, is there a voip company that will allow me that on asterisk? |
| 03:17.06 | Magicblaze0071 | or even comes close? |
| 03:17.08 | KyleK | Magicblaze0071: you could get chan_skype if you really like the $3 price point |
| 03:17.16 | KyleK | I haven't been happy with skypeout calls |
| 03:17.31 | *** join/#asterisk obnauticus (n=obnautic@about/windows/regular/obnauticus) |
| 03:17.32 | Magicblaze0071 | KyleK: I hate skype. The only thing I like is their $3 pricepoint. |
| 03:17.47 | rob0 | MB, free DIDs at ipkall. Many cheap termination providers exist. I rarely reach $3/month in calling. |
| 03:18.02 | rob0 | ~itsp-us |
| 03:18.40 | KyleK | Magicblaze0071: how much calling do you do a month on average? |
| 03:18.47 | rob0 | ~itsplist |
| 03:18.55 | KyleK | ~wb |
| 03:18.56 | infobot | thx |
| 03:19.03 | Magicblaze0071 | 25 users x 1000 minutes/person. |
| 03:19.04 | shido6 | ~botsnack |
| 03:19.04 | infobot | shido6: aw, gee |
| 03:19.46 | KyleK | and you can get all that for $3 without violating thier ToS? |
| 03:19.56 | *** join/#asterisk TJNII (n=TJNII@207.189.199.58) |
| 03:19.59 | drmessano | Nope |
| 03:20.05 | drmessano | You need 3 accounts |
| 03:20.12 | drmessano | Minimum |
| 03:20.17 | LiNeTuX | drmessano: and a lot of maint. overhead |
| 03:20.46 | *** join/#asterisk matt_d (n=matt@70.134.79.103) |
| 03:20.47 | matt_d | hello all |
| 03:20.48 | Magicblaze0071 | KyleK: of course I have the option of telling my users....go to "skype ...."! |
| 03:21.10 | LiNeTuX | A word to the wise: sometimes it's not all about 'cheap'. At some point you want 'good' or even 'functional' |
| 03:21.26 | drmessano | LiNeTuX: Absolutely |
| 03:21.28 | shido6 | pic 2 |
| 03:21.41 | drmessano | Cheap, Fast, Good, pick 2 out of the 3 |
| 03:23.12 | TJNII | Evening gentlemen. Someone just tried to call through my * box using channel SIP/127.0.0.1-0851f960. This bothers me. I don't have a 127.0.0.1 account set in sip.conf. How did this happen? |
| 03:23.19 | rob0 | I'm using Vitelity. I don't worry at all about the cost of calls, and I think it's pretty reliable. |
| 03:23.41 | matt_d | TJNII: that is strange. 127.0.0.1 is your loopback (im sure u know that already) |
| 03:23.48 | manxpower | I also use vitelity. Load it up with $25 every three months or so and not worry about it. |
| 03:23.53 | rob0 | 127.0.0.1 is localhost, only routable on the loopback ... it could not have come from outside. |
| 03:23.55 | Magicblaze0071 | I'll talk to vitelity tomorrow...i wish these companies were competing with GV or skype, in cheap/fast and good |
| 03:24.02 | TJNII | matt_d: I know. It is one of the reasons I am very nervous right now. |
| 03:24.17 | drmessano | Who said GV or Skype were good? |
| 03:24.40 | drmessano | They're cheap and have nice brand names.. |
| 03:24.47 | manxpower | drmessano: I can't imagine anyone saying that here. |
| 03:24.56 | rob0 | I suspect TJNII is worrying about nothing. Prepare a pastebin with all the details? |
| 03:25.09 | TJNII | rob0: Aye. |
| 03:25.10 | Magicblaze0071 | drmessano: Lets say I can live with the quality of skype...can I get its price on asterisk? |
| 03:25.33 | Magicblaze0071 | I think we both know the answer. |
| 03:25.38 | KyleK | skype to skype is good |
| 03:26.04 | KyleK | Magicblaze0071: i think we've collectively dug a 1 mile trench around No'svillve while discussing this with you |
| 03:26.04 | drmessano | First off, Skype is gonna cost you $11 a month the first year |
| 03:26.06 | Magicblaze0071 | my asterisk - asterisk is great too :) |
| 03:26.12 | drmessano | Considering you need a $66 license |
| 03:26.13 | TJNII | rob0: This is what I spotted on the console: http://pastebin.com/m27b877b9 |
| 03:26.18 | rob0 | I'm telling you, I doubt I spend as much as that per month. Granted, I don't spend a lot of time on the phone ... |
| 03:26.24 | manxpower | Telecom, no matter what sort, is not for the cheap |
| 03:26.48 | KyleK | I only need to call like two people longdistance these days and one of them has free long distance to me |
| 03:27.52 | drmessano | Right now we're at an impasse.. Big companies like Google have the bandwidth to start competing with 100+ years of copper, but they dont have 100 years experience.. The telecoms have 100+ years of infrastructure.. Google wants to give you $1 a month voice, but they really need to charge you a bit more so they can ensure SLA and actually build a quality network |
| 03:28.41 | drmessano | The telecoms have 100+ years of infrastructure to maintain and cant offer you $5 a month service.. but they can offer you quality if you dont buy flash-in-pan offerings |
| 03:29.16 | KyleK | my dad somehow got my local phone line down to $25 and caller id |
| 03:29.19 | drmessano | There is middle ground.. its closer to about $18 to $25 a month per channel |
| 03:29.38 | manxpower | I paid like $12/month for POTS + callerid |
| 03:29.57 | LiNeTuX | I got direct on-net SIP service - 24 channels for $350/mo |
| 03:29.58 | drmessano | But ZOMG Skype/Gizmo/Google fanboys wont pay that for business service because thats too much |
| 03:30.06 | Magicblaze0071 | drmessano: as a user, if you have bandwidth, what I don't understand is that why telecom companies don't learn to operate at a lesser price...are they waiting for GV or skype to kill them for ever? |
| 03:30.18 | KyleK | manxpower: the extra money goes towards maintaining the igloos up here ;) |
| 03:30.23 | drmessano | You miss the point |
| 03:30.24 | manxpower | LiNeTuX: which provider? |
| 03:30.29 | matt_d | LiNeTuX: unlimited? |
| 03:30.34 | drmessano | Google and Skype wont kill them with what they have to offer.. never |
| 03:30.40 | LiNeTuX | manxpower: Brighthouse (Florida) |
| 03:30.48 | rob0 | Funny thing is, GV might actually encourage me to get a local PSTN number. |
| 03:30.57 | LiNeTuX | matt_d: no. some mins included, (7K?), then $0.03 |
| 03:31.11 | drmessano | Google and Skype need to up the SLA by about 10x to get to where traditional telecoms are at |
| 03:31.16 | matt_d | LiNeTuX: Use Teliax. unlimited trunks. |
| 03:31.28 | KyleK | Magicblaze0071: POTS works longer than an hour after the power goes out typically |
| 03:31.34 | rob0 | It might be a bit more reliable than ipkall, and independent of whether or not my server's online. |
| 03:31.40 | LiNeTuX | matt_d: Yeah, but not native service. I'm on BH's fiber - not their internet. |
| 03:32.10 | manxpower | ponders getting a kitten |
| 03:32.23 | shido6 | please get a GV account, it will make my searches so much easier when looking for gossip |
| 03:32.24 | manxpower | LiNeTuX: my new employer does only on-net SIP |
| 03:32.47 | LiNeTuX | manxpower: It's the shit. |
| 03:32.58 | LiNeTuX | Setup takes 2 seconds. |
| 03:32.59 | Magicblaze0071 | LiNeTuX: Whom did you buy 24chanells for $350? |
| 03:33.18 | LiNeTuX | Magicblaze: Brighthouse. |
| 03:33.26 | TJNII | rob0: Does that look suspicious to you? It sure looks suspicious to me. |
| 03:33.34 | matt_d | LiNeTuX: it doesn't matter "which" internet you are hooked too. teliax offeres unlimited trunks. you only pay by usage. they have a good model. |
| 03:33.45 | TJNII | rob0: And I don't know why * allowed that sip channel in the first place. That bothers me more. |
| 03:34.07 | LiNeTuX | matt_d: My point is that I don't have to worry about bandwidth issues or QoS or any of that crap with an on-net provider. |
| 03:34.09 | matt_d | TJNII: What bothers me is that they don't fix the whisper function in ChanSpy!!!!! hehe |
| 03:34.23 | drmessano | The problem with "VoIP service" is the direction they are headed in.. While your telecoms are offering variations on their services over different conduits, like pulling fiber and offering SIP vs a PRI, your ITSPs are offering you the fucking moon... over your existing shitty Comcast/DSL connection.. Sure its cheap.. because they give you the milk, and who cares if the carton is full of holes |
| 03:34.42 | matt_d | LiNeTuX: ah.. much cheeper than bundling voice T1's :) |
| 03:34.53 | *** join/#asterisk ReDNeQ (n=ReDNeQ@70.114.229.58) |
| 03:35.39 | *** join/#asterisk alrs (n=lars@170.206.224.54) |
| 03:35.39 | LiNeTuX | matt_d: cheaper all around. if I need more, I call them. it's done. no more hardware, no new equipment, no 'real' provisoning... etc |
| 03:36.10 | Magicblaze0071 | drmessano: I guess I'm stuck with vitelity/VP or the likes for now...at their price point. |
| 03:37.01 | LiNeTuX | matt_d: and as I'm sure you know, you don't have to fuck with echo cancellation crap with SIP service |
| 03:37.29 | Magicblaze0071 | any suggestions for a good ip phone < $100? |
| 03:37.36 | LiNeTuX | oh god. |
| 03:37.37 | matt_d | LiNeTuX: Im still trying to decide if I want to pull in voice T1's. Megapath quoted me unlimited minutes (within the states) if I get one or three from them. |
| 03:37.47 | drmessano | Magicblaze0071: Nothing wrong with that.. who are you gonna call when google voice goes down? Larry? Sergey? |
| 03:37.56 | LiNeTuX | matt_d: I have PRI's I'm phasing out now |
| 03:38.33 | drmessano | Thats the other thing.. Gizmo seems interesting too, but I hear their tech support is worse than AOL |
| 03:38.40 | LiNeTuX | matt_d: If you have access to a provider who will offer on-net SIP, do that. |
| 03:38.41 | drmessano | If I have a problem, I have a problem |
| 03:39.25 | drmessano | Biggest pain in the ass is residential.. With business at least you are required in some ways to get a specific level of service, which comes with that support.. |
| 03:39.43 | Magicblaze0071 | drmessano: What is the sweet price point for ip phones now? functionality/maintainance/reliability vs price? |
| 03:39.55 | drmessano | Who the hell wants to spend HOURS or DAYS getting service for their home phone when it goes down, because you picked "cheap", as its "only my home phone" |
| 03:40.26 | drmessano | I have Comcast Business at home because I can call those bitches out 3 times in one day.. thank you 4-hour SLA |
| 03:40.28 | LiNeTuX | drmessano: WAF is very low with that plan. |
| 03:40.31 | *** join/#asterisk phuff (n=user@sourceforge/staff/luapffuh) |
| 03:40.31 | matt_d | who has home phones anymore? |
| 03:40.32 | matt_d | hehe |
| 03:40.39 | KyleK | i do :) |
| 03:41.06 | drmessano | Magicblaze0071: Guess it depends... about $90 to $400 |
| 03:41.06 | matt_d | i do as well :( only becuase my work requires it. there is only an answering machine hooked up to it though. |
| 03:41.22 | rob0 | TJNII, no idea, but I would strongly suspect it's something normal, maybe a quirk in your dialplan. |
| 03:41.23 | drmessano | I dont want a home phone, but even for work it helps having two phones |
| 03:41.26 | Magicblaze0071 | a good reliable ip phone for $100? $120 ? any recommendations? |
| 03:41.45 | drmessano | I call Comcast when I am on-call, and thats 30 mins I am using my BlackBerry to listen to hold music |
| 03:41.48 | drmessano | Eff that |
| 03:41.58 | LiNeTuX | here we go again. good, reliable, cheap |
| 03:42.13 | shido6 | http://www.callminer.com + Google Voice - makes news , gossip, and peoples behavior after an "event" so much easier to log and then predict for certain areas |
| 03:42.58 | TJNII | rob0: Well, I highly doubt it is the dialplan since nobody legitimate was calling at that time, plus the wacko sip account. This * box has been up for a year and a half now, I have never seen this before. |
| 03:43.08 | TJNII | Brute force SIP attacks, yes. This, no. |
| 03:43.47 | drmessano | shido6: i think you're being overly generous.. Ma Bell has been sifting all calls through NSA run NOC's since 2001.. |
| 03:43.55 | drmessano | They have much more experience |
| 03:44.01 | drmessano | Google is the new kid |
| 03:44.14 | Magicblaze0071 | LiNeTuX: K, remove the cheap. How about good and reliable ip phone recommendation? I don't need something complicated. Just to call, receive, voicemail. Simple. |
| 03:44.19 | matt_d | I have a question. I am using ChanSpy with whisper function. The whisper is delayed when its working though my IAX2 service. However there is no delay on recv. audio nor is there a delay between the otehr two phones being spied on. the only delay happens when i whisper to the main spied on phone. when i try this on the local network though softphones there is no delay. its not a bandwidth issue.. seems like it only happens though the IAX2 provider. any idea' |
| 03:44.31 | matt_d | i have already made sure that the codec's for all connections is ulaw (to avoid transcoding) |
| 03:45.09 | LiNeTuX | Magicblaze: Why not use a Linksys/Cisco SPA and some analog phones then? |
| 03:45.22 | drmessano | SPA-941 is a good phone |
| 03:45.25 | drmessano | Not too expensive |
| 03:45.45 | drmessano | Low end business/home phone |
| 03:45.57 | drmessano | But not as low as a grandstream :) |
| 03:46.20 | Magicblaze0071 | LiNeTuX: Wouldn't that mean I will have to run wires ? I have rooms across places with only ethernet. |
| 03:46.34 | *** part/#asterisk manxpower (n=EWieling@69.73.94.162) |
| 03:46.42 | KyleK | single or double ethernet lines? |
| 03:46.54 | drmessano | You run ethernet to the ATA |
| 03:47.06 | drmessano | No difference |
| 03:47.13 | *** join/#asterisk nkohh (n=justin@unaffiliated/kohh) |
| 03:47.17 | Magicblaze0071 | ATA is too much to maintain...i wud like to have an ip phone...I've done enough fishing with ATAs. |
| 03:47.26 | drmessano | I made my suggestion |
| 03:47.31 | nkohh | hey everyone, can anyone recommend a wholesale tollfree provider w/ an API that allows for adding of DIDs? |
| 03:47.31 | Magicblaze0071 | drmessano: SPA-941 looks nice |
| 03:47.57 | Magicblaze0071 | anything above SPA-941 in reliability / functionality (one up from that phone?) |
| 03:48.17 | drmessano | What exactly do you want? |
| 03:48.26 | [TK]D-Fender | Magicblaze0071: Its already overkill <- |
| 03:48.28 | drmessano | I gave you one model, now you want something one up? |
| 03:48.41 | Magicblaze0071 | perfect. I'll stick with that then. |
| 03:48.45 | drmessano | Maybe 3 down from 5 up and 2 up from 2 down? |
| 03:48.48 | Magicblaze0071 | I've heard good things about that phone. |
| 03:49.13 | LiNeTuX | top-shelf taste with a well-shelf wallet |
| 03:50.19 | drmessano | LiNeTuX: "Ham is too expensive!" "Get SPAM then." "What about one up from SPAM?" "You mean, HAM?" |
| 03:50.28 | LiNeTuX | heh |
| 03:50.42 | LiNeTuX | spamwich |
| 03:50.50 | drmessano | Oh oh oh |
| 03:50.57 | drmessano | Was at Wally World the other day |
| 03:51.00 | drmessano | SPAM WITH BACON |
| 03:51.04 | *** part/#asterisk Magicblaze0071 (n=sony@fl-67-235-212-177.dhcp.embarqhsd.net) |
| 03:51.07 | drmessano | OH OH OH.. I had to change my pants when I got home |
| 03:51.35 | LiNeTuX | spam spam spam spam spam spam spam spam... spammety spam, spammety spam. |
| 03:52.23 | drmessano | I almost got pulled over.. I had 101 uses for the new spam with bacon before I left the parking lot, and had to scribble them all on whatever exposed skin areas I had, so I wouldnt forget |
| 03:52.48 | LiNeTuX | heh. i'm spammed out. gotta get up early. later folks. |
| 03:52.54 | drmessano | I almost lost my recipe for spam bacaroni when I dropped the pen scribbling on my ankle |
| 03:53.01 | drmessano | :( |
| 03:53.11 | drmessano | Nevermind that Boxwood I ran over |
| 03:53.37 | drmessano | Later |
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| 04:03.08 | a1 | Is it possible to call a PBX and get the billing number, without the receiving call ringing? |
| 04:04.07 | matt_d | a1: explain further |
| 04:05.10 | *** join/#asterisk ramindia (n=balajibh@96-10.southernonline.net) |
| 04:05.55 | a1 | matt_d: I want to query a phone number for the billing # and then look it up on LIDB/CNAM for verification purposes |
| 04:06.28 | matt_d | what do u mean without the receiving call ringing? |
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| 04:30.15 | vader-- | hello |
| 04:31.31 | vader-- | question for you guys. We are currently running asterisk 1.2.x.x with cisco 7940G phones. We have 60 of them. We are looking to pick up new phones and was looking at the new cisco line but not sure what works well. Any suggestions? |
| 04:31.35 | matt_d | hi vader |
| 04:32.03 | a1 | matt_d: I mean by getting call info without ringing the phone |
| 04:32.30 | matt_d | a1: im still not sure what you are talking about. but i would venture to say yes. |
| 04:32.50 | matt_d | vader--: i havent used any of the new cisco ones :*( |
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| 05:14.50 | maour | i edited meetme.conf , now which module should i reload to aware asterisk ?! |
| 05:15.30 | maour | Asterisk 1.6.1.4 |
| 05:15.35 | *** join/#asterisk semaries (n=martin@stgt-5f7084d0.pool.einsundeins.de) |
| 05:17.13 | maour | seems Currently neither a reload or restart is required for Asterisk to notice changes in meetme.conf. |
| 05:17.21 | maour | :p |
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| 05:40.24 | drmessano | 1.6.1.4 isnt even out yet |
| 05:42.24 | kaldemar | say what? |
| 05:43.32 | kaldemar | http://www.asterisk.org/node/48615 |
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| 06:12.56 | drmessano | craziness |
| 06:18.33 | *** join/#asterisk larin (n=larin@mail.cs-service.by) |
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| 06:31.51 | mchou | drmessano: how does the polycom ip500 match up against spa-941? |
| 06:32.26 | mchou | drmessano: in terms of audio quality, not necessarily features |
| 06:36.15 | box2 | craziness |
| 06:38.47 | *** join/#asterisk hajvan (n=hsehic@tux.econtec.de) |
| 06:38.51 | hajvan | hi |
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| 06:44.46 | alunca | I installed AsteriskNow, and having an error apr_sockaddr_info_get() failed for hostname ... how can I fix this? |
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| 07:13.00 | fcois93 | hello all |
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| 07:13.19 | fcois93 | is it possible to lock a conference IF no admin are connected ? |
| 07:13.27 | fcois93 | http://forums.digium.com/viewtopic.php?t=70817 |
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| 08:27.18 | alunca | whenever I call out, it will wait for a while, and then it ring myself ... where should I look for this issue? |
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| 08:32.56 | TommyBotten | alunca: From the command line interface, enable debugging and verbose. And look for log entries when you make the call. |
| 08:33.51 | alunca | TommyBotten I'm in CLI> now; how can I enable debugging? udptl debug ? |
| 08:33.57 | TommyBotten | core set debug 8 |
| 08:34.00 | TommyBotten | core set verbose 8 |
| 08:34.06 | alunca | both ? |
| 08:34.16 | TommyBotten | Yes |
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| 08:44.24 | alunca | TommyBotten how can I get the log now? |
| 08:44.43 | TommyBotten | If you make the call, it should appear in the CLI |
| 08:44.53 | alunca | it runs too fast |
| 08:44.57 | fcois93 | is it possible to set a system command result in a var ? |
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| 09:07.38 | alunca | TommyBotten this is what exactly my issue |
| 09:07.41 | alunca | http://pbxinaflash.com/forum/showthread.php?p=31978 |
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| 09:20.18 | mattboll | Hi, I want to redirect a number to my mobile phone, it rings but there is no sound so it could be some firewall or RTP port problem |
| 09:20.32 | mattboll | but when i directly call the mobile it works |
| 09:20.46 | mattboll | and if i redirect to a local phone it works |
| 09:21.31 | mattboll | and if i use the transfer button from my sip phone it works as well |
| 09:22.57 | jgoo | echo - I've read a lot of theory, but found little 'best practice' firs steps to stop echo, I have a 4 port openvox isdn card, what is the easiest change I can do to see if echo is alleviated? |
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| 09:39.49 | jgoo | after I set txgain in zapata.conf - to reload, I need to stop / restart asterisk ? |
| 09:40.02 | jgoo | is there a quicker reload so I can test multiple variables? |
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| 09:45.05 | jgoo | http://kb.digium.com/entry/1/ << does anyone have a link to the docs to recompile zaptel source? |
| 09:47.14 | kaldemar | see README in the source |
| 09:48.19 | jgoo | kaldemar, I am using trixbox, is the source already there? what a cf |
| 09:48.58 | jgoo | Why can't they compile all the options in and have configuration that changes runtime behaviour? why have compile time configuration? anyway... |
| 09:49.42 | jgoo | kaldemar, when I've edited zapata.conf - what is the shortest path to reload it for testing? |
| 09:49.53 | kaldemar | i don't know about trixbox. i don't use it. |
| 09:50.05 | kaldemar | there's #trixbox for it |
| 09:50.16 | jgoo | kaldemar, that is the worst response I get in here, it has nothing to do with trixbox |
| 09:50.35 | jgoo | loading zapata.conf into asterisk has nothing to do with trixbox |
| 09:50.36 | kaldemar | depends on the asterisk version you're using. either reload chan_zap.so or restart asterisk. |
| 09:50.58 | jgoo | yeah, I'll restart asterisk :-/ |
| 09:51.06 | kaldemar | it has everything to do with trixbox, since trixbox modifies all configurations and has generated them. |
| 09:52.47 | jgoo | hrm, yeah - perhaps, fuck trixbox - so I have txgain set to 0 - but some people get echo, when they talk too close to the mouthpiece (because they are blunt idiots who use a loud telephone voice) |
| 09:53.34 | jgoo | (I hate people who talk louder on phones... you give them a phone and its who is it? OH HELLO? YEAH, IM ON A PHONE, IN AN ART GALLERY, YEAH RUBBISH) |
| 09:59.06 | jgoo | can someone update this faildoc to include illustrative configuration snippets, where it says 'set training' and then talks for 900 chars about delays, put a code block echotraining=xx http://www.voip-info.org/wiki/view/Asterisk+echo+cancellation |
| 09:59.22 | *** join/#asterisk [netman] (n=netman@80.30.61.220) |
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| 10:00.42 | skrusty | morning all |
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| 10:05.24 | kaldemar | jgoo: you can update it too |
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| 10:21.15 | Bach | Hi all, need a little guidance |
| 10:21.20 | *** part/#asterisk TheOpenSourcerer (n=TheOpenS@81-178-65-1.dsl.pipex.com) |
| 10:22.24 | Bach | trying to setup an asterisk as a gateway server between a voip provider an another asterisk. Anyone who has some info regarding this. |
| 10:22.57 | Bach | The last asterisk will control phone numbers ect. The gateway should only pass the correct numbers through |
| 10:23.20 | larin | Wildcard |
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| 10:25.41 | Bach | My problem is that i cant find any info how to configure the sip trunk on the gateway to connect the two asterisk together |
| 10:31.11 | *** part/#asterisk fcois93 (n=Administ@bagnolet.acropolistelecom.net) |
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| 10:34.52 | frk2 | Im having the most interesting issue with my Cisco 7912G phones and Asterisk |
| 10:35.19 | box2 | interesting like communication with aliens? |
| 10:35.30 | frk2 | box2, something like that seriously :) |
| 10:36.22 | frk2 | randomly, for no apparent reason, when I dial from my cisco the phone would simply 'freeze' on the 'Enter number' dialog. I can see it sends a single invite to the asterisk box, receives a reply, but the call just never proceeds. The phone keeps trying to send invites and asterisk never receives them |
| 10:36.40 | frk2 | i can confirm from wireshark that the sip packets from the phone never arrive on the server |
| 10:36.55 | frk2 | this happens totally randomly |
| 10:36.59 | frk2 | with no obvious pattern |
| 10:37.21 | frk2 | cisco debug logs (on the phone) simply keep on saying retrying xmit sip packet... |
| 10:37.26 | frk2 | retries 10 times and the phone gives up |
| 10:38.00 | box2 | does it keep happening until you reboot the phone or something? |
| 10:38.01 | frk2 | is that interesting enough, box2 ? :) |
| 10:38.05 | frk2 | no no |
| 10:38.10 | frk2 | the next call it may randomyl work |
| 10:38.13 | frk2 | and will keep working |
| 10:38.24 | frk2 | the call after that it may randomly do this again |
| 10:38.35 | box2 | hm |
| 10:38.42 | box2 | are you using the lastest firmware for the phones |
| 10:38.48 | frk2 | its 8.3.1 right? |
| 10:38.48 | *** join/#asterisk Hatrix (n=Hatrix@213.201.24.127.static.user.ono.com) |
| 10:38.53 | frk2 | for 7912G ? |
| 10:39.03 | box2 | i'm not sure what it would be now |
| 10:39.30 | Hatrix | hello, i want to give asterisk the CAP_FOWNER=ep capability, with setcap I can set it for the file (in this case /usr/sbin/asterisk), with getcap on this file I see it is set, but if i start the process and check with getpcap the only thing set is: Capabilities for `7601': = cap_net_admin+ep, so why does the CAP_FOWNER does not work? And can I set the capabillity of a running asterisk process? |
| 10:39.53 | frk2 | im wondering if its a network bug |
| 10:40.20 | kaldemar | Bach: see "Connecting Two Asterisk Boxes Together via SIP" in the book |
| 10:40.24 | box2 | that's strange that after the first invite packet no more invites reach asterisk |
| 10:40.25 | kaldemar | ~book |
| 10:40.26 | 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 |
| 10:40.44 | frk2 | box2, exactly |
| 10:40.50 | frk2 | the first invite ALWAYS reaches asterisk |
| 10:40.59 | frk2 | but subsequent invites may not |
| 10:41.30 | box2 | where are you wiresharking from? |
| 10:41.38 | frk2 | the asterisk server |
| 10:41.53 | frk2 | from the phone side i just look at cisco logs on the phone itself |
| 10:42.37 | box2 | do you have other brand/model phones that work fine on the same leg of the network? |
| 10:42.45 | frk2 | yeah |
| 10:42.55 | frk2 | linksys pap2s, grandstream, linksys spa 962 |
| 10:42.58 | frk2 | all work excellent |
| 10:44.04 | box2 | sounds like your 7912G's are malfunctioning |
| 10:44.12 | frk2 | thats what i would think |
| 10:44.13 | frk2 | but why |
| 10:44.28 | frk2 | there are 300 of them in a building and they work fine |
| 10:44.35 | frk2 | another 150 in this building and they cause issues |
| 10:44.51 | frk2 | another 500 of them dispersed in remote locations and they cause no problems |
| 10:45.15 | box2 | all running the same version firmware? |
| 10:45.22 | frk2 | exactly the same |
| 10:45.54 | box2 | do you have the same network infrastructure in these different locations? |
| 10:46.02 | box2 | such as same brand/model switches etc |
| 10:46.05 | ickmund | I'm debugging a dialplan that I didn't set up myself. It's an IVR, and when you choose a single digit you get in a queue. Once in the queue, there's a context so a single digit breaks you out. Problem is, it doesn't go to this context, it goes to the main IVR context again... any thoughts? |
| 10:46.24 | frk2 | box2, all cisco 3560's |
| 10:46.53 | frk2 | i promised it'll be interesting! :) |
| 10:47.00 | box2 | lol, it is... |
| 10:47.39 | box2 | and your 3560's are all running same version firmware i assume as well |
| 10:48.24 | frk2 | i dont know that |
| 10:48.31 | frk2 | they arent my 3560s :) |
| 10:48.57 | box2 | ah |
| 10:49.34 | box2 | i suppose i would personally start with that, then |
| 10:49.39 | frk2 | really? |
| 10:49.53 | box2 | it's the first thing we've discussed that could be any different |
| 10:49.54 | frk2 | could the switches have anything to do with this? |
| 10:50.07 | frk2 | im gonna try isolating a couple of phones and see if they work |
| 10:51.11 | box2 | yea, i once had 3com switches that by default blocked all layer 2 traffic |
| 10:51.26 | box2 | so no phones could get their boot config from tftp |
| 10:51.37 | box2 | i was scratching my head about that one for a long time |
| 10:52.08 | frk2 | hmm |
| 10:52.12 | frk2 | but why random? |
| 10:52.25 | frk2 | do these switches have some intelligent shit in them that can block in realtime |
| 10:53.43 | box2 | perhaps they are configured with a strange QoS setting |
| 10:54.02 | box2 | if a lot of people are talking it will start banhammering new attempts |
| 10:54.10 | box2 | or something even stranger |
| 10:54.53 | hershel | odd question here: I am building my first IVR and the box is in my office. I have a SIP box which I can treat as the 'office' but I need a phone line to plugin. i know there are VOIP services that have internal calls free. like if i had 2 accounts i could call from a softphone to my voip box. |
| 10:55.05 | hershel | any advice regarding that? I need a way to test my IVR, as I build it. |
| 10:56.23 | hajvan | ok i need some help here, i have asterisk 1.4 (debian lenny) Diva eincon + chan_capi working, i'v done "core set verbose 5" and "capi debug", i see the call to 2945721 on CLI however my softphone isn't ringin error message "ISDN1#02: did not find exten for '2945721', i |
| 10:56.27 | hajvan | gnoring call." |
| 10:56.30 | box2 | hershel: you need a number to dial into your asterisk? |
| 10:56.35 | box2 | i didn't quite understand |
| 10:56.38 | hershel | box2: that's right. |
| 10:56.44 | box2 | hershel: try ipkall |
| 10:56.48 | box2 | it is free |
| 10:57.11 | box2 | DID numbers to SIP/IAX (numbers are all in Washington state) |
| 10:57.39 | hershel | ok, maybe freeworlddialup will work. i am not in the US so washington is long distance for me. :) |
| 10:57.48 | frk2 | box2, that is my guess |
| 10:58.00 | hershel | freeworlddialup if i have 2 accounts i could do one softphone and then call my SIP box. I will try that. Thanks! |
| 10:58.05 | hajvan | capi.conf incomingmsn=2945721 and isdn-in as context, extensions.conf exten => 2945721,2,Dial(SIP/4242,30) |
| 10:58.14 | frk2 | box2, but once the conversation starts there is no issue |
| 10:58.17 | hajvan | under [isdn-in] |
| 10:58.20 | hajvan | any hints?!? |
| 10:59.10 | box2 | frk2: QoS can be defined in many ways, a SIP session setup packets are easily differentiated between a channel that is already in progress |
| 11:00.25 | frk2 | box2 i hope its that |
| 11:00.27 | frk2 | makes sense though |
| 11:00.41 | box2 | it's the only thing i can think of, anyway |
| 11:01.47 | *** join/#asterisk chnuebi (n=patrick@102-68.0-85.cust.bluewin.ch) |
| 11:01.56 | frk2 | box2, thanks. will check it out |
| 11:02.53 | box2 | frk2: np |
| 11:04.22 | *** join/#asterisk garymc (n=gar@host81-134-0-102.in-addr.btopenworld.com) |
| 11:04.34 | garymc | Hi, can i plug an ISDN cable right into my server? |
| 11:04.37 | *** join/#asterisk djin (n=djin@84-104-110-179.cable.quicknet.nl) |
| 11:04.42 | garymc | to allow some testing of phone lines etc? |
| 11:05.47 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 11:07.32 | *** join/#asterisk M_Red (n=ty@97-119-253-80.spkn.qwest.net) |
| 11:09.28 | Chainsaw | garymc: With an ISDN telephony adapter, yes. |
| 11:11.11 | *** part/#asterisk M_Red (n=ty@97-119-253-80.spkn.qwest.net) |
| 11:13.03 | garymc | so i need an adapter? |
| 11:13.06 | garymc | in my server? |
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| 11:32.53 | garymc | right im pretty much stuck |
| 11:32.56 | garymc | i need help |
| 11:33.01 | garymc | here is my setup |
| 11:33.42 | garymc | <PROTECTED> |
| 11:33.42 | garymc | <PROTECTED> |
| 11:33.42 | garymc | <PROTECTED> |
| 11:33.42 | garymc | <PROTECTED> |
| 11:33.44 | garymc | so what do i need, hardware ? |
| 11:34.02 | garymc | did that make sense |
| 11:34.06 | garymc | Ive got two servers |
| 11:34.12 | garymc | one Asterisk and an LTSP server |
| 11:34.23 | garymc | both linked to my ethernet switch |
| 11:35.10 | *** join/#asterisk ariel_ (n=chatzill@c-71-196-99-26.hsd1.fl.comcast.net) |
| 11:36.05 | kaldemar | no special hardware needed for the voip phone. but for an ISDN line, you need hardware as Chainsaw said. |
| 11:36.12 | ariel_ | Morning folks |
| 11:36.40 | kaldemar | you can't just plug a calbe in to whatever NIC and expect it to work. |
| 11:40.56 | Chainsaw | kaldemar: It would actually destroy the NIC. |
| 11:41.46 | *** part/#asterisk larin (n=larin@mail.cs-service.by) |
| 11:42.11 | tzafrir_laptop | What ISDN adapter do you have? |
| 11:42.42 | tzafrir_laptop | garymc, ==^ |
| 11:43.14 | Chainsaw | kaldemar: For fun, pick up an old 3Com PCI NIC (Vortex or Boomerang). It actually has a picture etched into a metal. A phone horn with a big "don't do that!" cross through it. |
| 11:43.36 | Chainsaw | s/a metal/the metal/ |
| 11:44.34 | kaldemar | it won't like ringing voltage. :) |
| 11:47.33 | hershel | my agi script is fighting with asterisk process for control of my CPU. I made a test call and now each is 50% CPU usage. :( |
| 11:48.10 | *** join/#asterisk thisismyname (n=quassel@212.77.162.22) |
| 11:48.35 | Hatrix | hershell, that's great, better than asterisk freaking fucking eating 100% of it |
| 11:48.50 | Hatrix | hershel: don't take me seriosly, i am a bit on the edge right now |
| 11:48.59 | hershel | lol |
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| 11:52.16 | surrogat2000 | hi, i'm trying to write a callfile without a channel, because i want to call a robot and send some dtmf-digits |
| 11:52.29 | surrogat2000 | how can i do that? |
| 11:52.45 | surrogat2000 | without the "channel:" it gives an error |
| 11:54.27 | garymc | tzafrir_laptop: I have no adapter |
| 11:54.30 | garymc | im new to this |
| 11:57.25 | surrogat2000 | nobody any idea? |
| 11:57.29 | kaldemar | surrogat2000: dial the robot with Channel and send the DTMF with exten, context and priority. |
| 11:58.11 | kaldemar | you can't use call files without Channel |
| 12:01.49 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 12:02.21 | surrogat2000 | but with wich channel, i dont want to let a telephone ring |
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| 12:34.38 | surrogat2000 | ok, solved :) |
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| 12:49.40 | hajvan | anyone who can give me a hint howto configure dial-out usinf capi/isdn? |
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| 12:49.51 | hajvan | s/usinf/using |
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| 13:37.39 | Katty | blah. |
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| 13:37.48 | Katty | so much cranky today |
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| 13:41.23 | manxpower | ~answers |
| 13:41.24 | infobot | i heard answers is Asterisk Book: http://astbook.asteriskdocs.org & SIP w/ NAT: http://www.aocomputing.net/?p=3 & Wiki: http://www.voip-info.org & docs: /path/to/src/asterisk/doc & 1.4 .vs. 1.6: /path/to/src/asterisk/UPGRADE*.txt |
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| 13:44.24 | *** join/#asterisk Deeewayne (n=dwayne@75.76.254.162) |
| 13:44.24 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
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| 13:47.39 | manxpower | waves to Deeewayne |
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| 13:48.07 | Deeewayne | manxpower, hey! How's it going ? |
| 13:48.59 | manxpower | Deeewayne: Pretty good. Working for a CLEC now. |
| 13:49.04 | geneticx | hello. I have some users complaining that when they call the remote office they hear choppy voice but the remote office hears them perfectly...what could be the root cause of this? latency is kind of high and fluctuates from 603 ms to 1056ms |
| 13:49.22 | manxpower | geneticx: that would be the root cause |
| 13:50.25 | geneticx | manxpower: that's what I thought but how would I know if it's me the problem or them? |
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| 13:50.59 | [TK]D-Fender | geneticx: Get calls from other people |
| 13:51.08 | manxpower | geneticx: it's just basic networking issues at that point. |
| 13:51.09 | [TK]D-Fender | geneticx: If it happens to them, then its you. |
| 13:51.53 | geneticx | [TK]D-Fender: ok, let me try that now. |
| 13:52.34 | manxpower | geneticx: the problem is sending the audio from you to them, so your upspeed or their downspeed. If you send calls across the internet you should expect at least occasional problems |
| 13:52.39 | *** join/#asterisk blackest_mamba (n=blackest@c-71-239-160-143.hsd1.il.comcast.net) |
| 13:53.09 | manxpower | or any router between your router and their router. |
| 13:53.36 | geneticx | manxpower: yup, but these aren't occasional problems when they been going on for the past 3 days =( |
| 13:54.09 | *** join/#asterisk jplank (n=gbove@cpe-098-024-169-236.carolina.res.rr.com) |
| 13:54.36 | grandpapadot | Hey, can someone please explain to me what MS Communication Server actually is, is it a PBX of sorts? I swear the website is vague as can be. |
| 13:54.41 | *** join/#asterisk Victor_Yure (n=victor@unaffiliated/victoryure/x-837844) |
| 13:54.55 | manxpower | grandpapadot: It is your worst nightmare. |
| 13:55.28 | grandpapadot | manxpower: lol, no doubt, I'm not considering it, just curious what all the chatter is about. |
| 13:56.14 | grandpapadot | manxpower: A PBX with some presence support? |
| 13:57.10 | manxpower | It's a unified platform leveraging Microsoft technologies in a paradigm shift to generate customer facing synergy. As I understand it, it's a glorified PBX |
| 13:58.21 | grandpapadot | manxpower: lol, thanks... So it's just basically terrible? |
| 13:58.32 | hajvan | i know you guys are master of *^universe and newbie questions are boring, but can someone help a newbie on a ISDN outbound issue ...? |
| 13:58.56 | ruben23 | anyone have idea how do i correct this error-->http://pastebin.com/mdcf52a5 i got this with my inbound calls for my asterisk server. |
| 13:58.59 | manxpower | grandpapadot: it's the only SIP system that I know of that does not support UDP for SIP. They only support TCP for SIP. |
| 13:59.14 | ickmund | With Chanspy in * 1.4, how can I do a Whisper on an extension that I currently have a Barge on? Any ideas? |
| 13:59.41 | manxpower | ruben23: remove the file /var/lib/asterisk/mohmp3/LICENSE |
| 13:59.50 | *** join/#asterisk spck (n=spck@unioncab.com) |
| 13:59.51 | *** join/#asterisk tpeers (n=peers@sysadmin-cnc.malagasy.com) |
| 13:59.58 | *** join/#asterisk InfoNutz (n=infonutz@204.50.209.225) |
| 14:00.15 | *** part/#asterisk tpeers (n=peers@sysadmin-cnc.malagasy.com) |
| 14:00.35 | ruben23 | manxpower: that would be all..? |
| 14:00.40 | ruben23 | ill try now |
| 14:00.54 | [TK]D-Fender | ruben23: You clearly have a file called "LICENSE" in yuor MoH folder that isn'ta SOUND FILE. so remove the excess junk |
| 14:00.57 | grandpapadot | manxpower: So how does it interop with asterisk? |
| 14:01.09 | ickmund | Can I somehow save the extension-number in * somewhere, hang up, hit a Whisper-key, and then retrieve that extension-variable? |
| 14:01.30 | hajvan | incomming calls are woring, but by out calls i get chan_sip.c:14035 handle_request_invite: Call from '4242' to extension 'FOONUMBERXXXX' rejected because extension not found. |
| 14:02.26 | *** join/#asterisk putnopvut (n=putnopvu@asterisk/master-of-queues/mmichelson) |
| 14:02.26 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:02.42 | hajvan | and i have exten => _0X.,n,Dial(CAPI/ISDN1#02/${EXTEN}) |
| 14:02.55 | hajvan | any hints? |
| 14:03.16 | grandpapadot | manxpower: rather, how well does it interop and how are folks getting UDP->TCP for SIP? (generalization if you have one, not looking for details) |
| 14:04.17 | [TK]D-Fender | hajvan: Here's a hint : you are showing an exten with an "n" priority. You need tos tart with '1" before you can follow with "n"'s |
| 14:05.05 | hajvan | [TK]D-Fender: 10x |
| 14:05.57 | *** join/#asterisk mweichert_ (n=mweicher@216.13.154.21) |
| 14:05.58 | manxpower | grandpapadot: 1.6 was the first version to support SIP/TCP, the only system that requires SIP/TCP is Microsoft. You do the math |
| 14:06.50 | mweichert_ | hello, when I try to dial an internal extension from a registered soft phone, I get the message "extension XXX is on the phone" and see this in the log: "dialparties.agi: Extension XXX has do not disturb enabled, or followme pre-ring returned busy" |
| 14:06.56 | mweichert_ | how do I turn off DND? |
| 14:06.57 | grandpapadot | manxpower: ahh.. I haven't even compiled 1.6 yet.. (thanks for the feedback) |
| 14:07.08 | [TK]D-Fender | mweichert_: FreePBX is NOT supported here. |
| 14:07.14 | [TK]D-Fender | ~freepbx |
| 14:07.15 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 14:07.25 | *** join/#asterisk PuroOsso (n=PuroOsso@189.74.141.10) |
| 14:07.27 | mweichert_ | I've tried *79 but that doesn't work |
| 14:07.42 | mweichert_ | [TK]D-Fender, you think this problem is specific to FreePBX? |
| 14:07.50 | [TK]D-Fender | mweichert_: YES |
| 14:08.01 | mweichert_ | [TK]D-Fender, ok, thanks. |
| 14:08.15 | PuroOsso | [TK]D-Fender, hi |
| 14:09.49 | PuroOsso | [TK]D-Fender, need some help... my english is very bad, but I'll try to explain my problem... you might remember.. |
| 14:10.29 | PuroOsso | I'm getting mute call... not all time, and the person can hear me, but i can't hear nothing... |
| 14:10.51 | PuroOsso | only my voice is recorded... |
| 14:11.01 | *** join/#asterisk davevg (n=davevg__@75.97.64.33) |
| 14:11.28 | PuroOsso | RTPC debug shows me that I'm having packet loss... |
| 14:12.16 | *** join/#asterisk ccesario (n=ccesario@189-19-6-236.dsl.telesp.net.br) |
| 14:12.24 | PuroOsso | sip debug shows me that, sometimes, there's no route to destination |
| 14:12.31 | [TK]D-Fender | PuroOsso: pastebin a failed call with SIP debug enabled. |
| 14:12.33 | [TK]D-Fender | ~pb |
| 14:12.34 | 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://bin.cakephp.org/ , or apt-get install pastebinit |
| 14:12.37 | PuroOsso | and then, it becomes reachable again... |
| 14:13.15 | PuroOsso | ok, but there's many numbers here, so the logs are very confuse |
| 14:14.48 | PuroOsso | just a minute |
| 14:15.29 | *** join/#asterisk jmacz (n=mcorb@190.144.75.22) |
| 14:15.58 | PuroOsso | http://pastebin.com/d44592fa4 |
| 14:16.13 | *** join/#asterisk andres833 (n=andres83@190.144.75.22) |
| 14:16.46 | hajvan | how comes to "chan_zap.c:11900 build_channels: Unable to reconfigure channel '1'" Error on Debian lenny even if i didn't changed zapata.conf? |
| 14:16.56 | hajvan | Debian bug, or PEBCAK? |
| 14:17.04 | manxpower | ~freepbx |
| 14:17.05 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 14:17.11 | *** part/#asterisk manxpower (n=EWieling@69.73.94.162) |
| 14:17.35 | PuroOsso | [TK]D-Fender, the only warning is at line 96 |
| 14:17.50 | *** join/#asterisk KavanS (n=KavanS@71.117.242.28) |
| 14:19.05 | *** join/#asterisk outtolunc (n=me@c-98-248-96-110.hsd1.ca.comcast.net) |
| 14:20.24 | PuroOsso | [TK]D-Fender, any idea? |
| 14:21.00 | [TK]D-Fender | PuroOsso: Here's one : provide what I asked for. |
| 14:22.00 | PuroOsso | [TK]D-Fender, http://pastebin.com/d44592fa4 |
| 14:22.44 | PuroOsso | sip debug was enable... |
| 14:22.47 | [TK]D-Fender | PuroOsso: All I see are local channels. there is NOTHING to debug in there. that pastebin is of no value |
| 14:22.51 | *** join/#asterisk geneticx (n=ethicx@adsl-074-169-015-252.sip.mia.bellsouth.net) |
| 14:22.57 | [TK]D-Fender | PuroOsso: And that warning you see is irrelevent |
| 14:22.59 | *** join/#asterisk Naikrovek (n=jjohnson@63-252-251-77.ip.mcleodusa.net) |
| 14:23.37 | [TK]D-Fender | PuroOsso: there is no SIP DEBUG as I requested |
| 14:23.56 | [TK]D-Fender | PuroOsso: And I don't feel like beating this to death. |
| 14:23.56 | PuroOsso | but it was on |
| 14:24.12 | PuroOsso | i'll try again... on next mute call |
| 14:25.14 | *** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose) |
| 14:25.14 | PuroOsso | but, the log that I did show you is a call from outside |
| 14:25.20 | PuroOsso | not an internal call |
| 14:26.11 | [TK]D-Fender | PuroOsso: I don't see the start of the call, there is no originating tech, there is no SIP DEBUG for the spawned call, NOTHING. |
| 14:26.18 | [TK]D-Fender | goes off to do something productive. |
| 14:26.48 | PuroOsso | [TK]D-Fender, I'll come back with more info later... |
| 14:26.51 | PuroOsso | thanks |
| 14:29.41 | garymc | ok, im totally lost as to what hardware i need EXACTLY |
| 14:30.03 | garymc | Im told i need a Sangoma A500 BRI to connect my ISDN 2e to it |
| 14:30.11 | garymc | Now i think i need to purchase module too? |
| 14:30.18 | garymc | can anyone tell me what i need> |
| 14:30.21 | garymc | ? |
| 14:30.39 | *** join/#asterisk jasonwoot (n=some@69.73.89.233) |
| 14:30.40 | garymc | also what is 8 channel , 22 channel etc |
| 14:30.49 | garymc | and ports needed and so on? |
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| 14:33.24 | *** join/#asterisk shido6 (n=shido6@74-132-202-71.dhcp.insightbb.com) |
| 14:34.04 | mweichert_ | what's the difference between "pick up parked call" and "directed call pickup" ? |
| 14:37.00 | Vec | Hi, I have a SIP phone (Avaya) which registers with Asterisk fine, can make calls etc, then when the re-registration timeout occurs, it attempts to register again, and fails but not from asterisk point of view. I guess I need to go reading the RFCs but maybe someone can give me some pointers: The first REGISTER from the phone results in a 401, the phone then sends another REGISTER which results in 200 ok. When the phone sends the re-register it ju |
| 14:37.53 | grandpapadot | Vec: We've had similar issues with Nortel 6804, 6812, 6830 (which I think are now Avaya?), they work great as long as they stay registered. |
| 14:37.55 | *** join/#asterisk msetim (n=setim@189.32.44.21) |
| 14:37.58 | msetim | hello |
| 14:38.02 | jasonwoot | who manages the bot for this channel? |
| 14:38.28 | *** join/#asterisk rene- (n=renemend@200.34.66.137) |
| 14:39.13 | msetim | it's safe to use asterisk rx to get information to snmp monitoring? |
| 14:39.26 | Vec | grandpapadot : I can change the registration timeout to 1 day, but then the phone will "break" after a day, there is no way to set it longer than that. |
| 14:39.43 | Vec | msetim : we use it all the time with no probs, with zabbix |
| 14:39.59 | grandpapadot | Vec: I'm not sure if or how we ever solved it ... |
| 14:40.19 | Vec | grrrr :( |
| 14:40.25 | jdnWEST | Anyone know why ipSEC would screw with a voice channel? Do I need to adust a MTU or something? |
| 14:40.35 | msetim | Vec, Do you already get issues with deadlock or cpu load? |
| 14:40.58 | Vec | jdnWEST : it would increase latency and possibly throughput, what u mean screw, whats happening. |
| 14:41.16 | *** join/#asterisk moy (n=moy@bas1-unionville55-1177733667.dsl.bell.ca) |
| 14:41.20 | grandpapadot | jdnWEST: We use ipsec tunnels for one of our secure voice products, the generic jitter buffer set to 100-200ms fixed our problems on sip channels. |
| 14:41.47 | Vec | msetim : no issues with cpu load, sometimes the asterisk -rx "blah blah" command can fail to return a value, but that doesn't matter much for snmp and its very rare |
| 14:42.20 | Vec | grandpapadot : do u perhaps know the model of nortel phone u where having the problem with ? |
| 14:42.41 | grandpapadot | Vec: afaik all the ones I listed had the same issue |
| 14:42.59 | Vec | grandpapadot : thanks will check them out |
| 14:43.42 | garymc | Not much help on here? |
| 14:44.03 | grandpapadot | garymc: eh? |
| 14:44.14 | Vec | grandpapadot : those phones "look" very diffirent to the one I am using, they don't seem too closely related. Avaya 96xx series |
| 14:44.36 | grandpapadot | Vec: How many of them do you have in service? |
| 14:44.45 | garymc | well, i need to know what hardware i need and nobody seems to know much |
| 14:45.01 | grandpapadot | garymc: What's your question, please restate. |
| 14:45.09 | grandpapadot | garymc: and you get what you pay for, lol |
| 14:45.16 | garymc | i have ISDN2e |
| 14:45.25 | garymc | i want to link it to my Asterisk Server |
| 14:45.33 | garymc | what extra hardware is needed? |
| 14:45.34 | jdnWEST | hmmm, Grandpapadot: I'm only getting like 1ms of jitter, but I'll see if I can do that |
| 14:45.38 | garymc | im using AsteriskNOW |
| 14:46.00 | garymc | see just to test this out, seems like i gotta spend loads of money |
| 14:46.18 | Vec | grandpapadot : we have about 80, but they are not in service, they are from an old Avaya system, I need to get them working with asterisk |
| 14:47.03 | drmessano | garymc: If you want to "test out" BRI, then yeah, you need to spend money for the real deal.. No one said this was a free ride |
| 14:47.35 | grandpapadot | jdnWEST: The jitter buffer will help even if there isn't much actual jitter as it will stack, order, and send the packets .. essentially buffer them .. just a thought |
| 14:47.37 | jdnWEST | Drmessano: People s till use BRI circuts? |
| 14:47.53 | drmessano | jdnWEST: Apparently overseas.. |
| 14:47.55 | grandpapadot | Vec: ah, I was going to suggest getting some Polycoms |
| 14:48.01 | jdnWEST | grandpapadot: One of the routers is a cisco 2801, other one is Pfsense (BSD) |
| 14:48.38 | Chainsaw | jdnWEST: Yes, especially in the UK where PRI is rather expensive. |
| 14:48.56 | Chainsaw | jdnWEST: So instead of getting a PRI with only a few channels active, it's actually cheaper to get 2x BRI. |
| 14:49.11 | grandpapadot | jdnWEST: Watch that pfsense, i've always sen problems with nat and pfsense/bsd. |
| 14:49.41 | garymc | im in UK |
| 14:49.46 | garymc | where are you guys? |
| 14:49.51 | grandpapadot | USA |
| 14:49.54 | Chainsaw | garymc: Peterborough (Cambridgeshire). |
| 14:49.55 | Katty | raindrops keep fallin on my head :< |
| 14:49.59 | garymc | well your always ahead of us |
| 14:50.12 | Chainsaw | garymc: USA ahead of the UK? In what please? |
| 14:50.13 | InfoNutz | wuts a quick dialplan to send a called extension to a PRI? |
| 14:50.15 | garymc | peterbrough your normally 10 years behind us :) |
| 14:50.29 | *** join/#asterisk servilio (n=servilio@unaffiliated/servilio-ap) |
| 14:50.29 | garymc | Chainsaw telecoms tech |
| 14:50.39 | Chainsaw | garymc: You're kidding right? |
| 14:50.44 | garymc | nope |
| 14:51.00 | garymc | even yhour average broadband speeds are way ahead of ours |
| 14:51.27 | garymc | Chainsaw the average Broadband speed in uk is about 2mb Download |
| 14:51.29 | Chainsaw | garymc: You'll learn about contention ratio one day. And then you'll see it's not quite as it seems. |
| 14:51.30 | garymc | if that |
| 14:51.33 | grandpapadot | We (US) are behind most countries in broadband speeds ... |
| 14:51.47 | Katty | let's not even talk about how behind our healthcare is. |
| 14:51.49 | *** part/#asterisk hajvan (n=hsehic@tux.econtec.de) |
| 14:51.55 | garymc | yeah? whats the average speed in US |
| 14:52.00 | InfoNutz | Up here in Canada we have all the bandwidth we need |
| 14:52.12 | Katty | InfoNutz: and cold weater. |
| 14:52.14 | jdnWEST | grandpapadot:thats why we're using ipsec, doing have to muck with forwarding sip (although we don't really have problems with it, just have to have enough cpu power for those boxes) |
| 14:52.27 | InfoNutz | Yeah the colder the T1's the faster they run :) |
| 14:52.32 | grandpapadot | Dunno, but I have an 18MB/2MB connection at my house and that's at the top of the list. |
| 14:52.43 | Katty | InfoNutz: hehe |
| 14:53.29 | grandpapadot | That's ok though, because our current administration is going to push us into the dark ages, stay tuned, we'll be a third rate, third world country in 50 years. |
| 14:53.39 | Katty | grandpapadot: it won't take that long. |
| 14:53.52 | *** join/#asterisk seanbright (n=sean@asterisk/contributor-and-bug-marshal/seanbright) |
| 14:53.56 | garymc | your already communist |
| 14:54.08 | garymc | you execute people still |
| 14:54.08 | grandpapadot | graymc: who's communist? |
| 14:54.10 | Katty | no, we're not. |
| 14:54.12 | garymc | US |
| 14:54.16 | Katty | we're a bit socialist |
| 14:54.17 | InfoNutz | is dial-up connections still widely used down in the US? |
| 14:54.27 | grandpapadot | we don't execute anyone, wtf? |
| 14:54.36 | grandpapadot | InfoNutz: no |
| 14:54.38 | Katty | executions have nothign to do with it. |
| 14:54.46 | Chainsaw | InfoNutz: With most dial plans in the US offering free local calls, it lasted longer then in other countries. |
| 14:54.48 | *** join/#asterisk coppice (n=chatzill@52.204.17.210.dyn.pacific.net.hk) |
| 14:54.50 | grandpapadot | We're not even a bit socialist yet, although we're trying to get there quick. |
| 14:54.56 | Katty | yes we. |
| 14:54.58 | Chainsaw | InfoNutz: But it's pretty much gone now. |
| 14:55.23 | grandpapadot | garymc: Who have we executed? Do you mean prisoners? Oh, well, in that case, we don't do enough of that, IMHO. |
| 14:55.33 | InfoNutz | Chainsaw: that makes sense... WAN links are more and more common up north here in canada, its cutting into our business |
| 14:55.34 | Katty | we already ahve socialized healthcare for those over the age of 65, i believe. |
| 14:55.40 | grandpapadot | Katty: Yep. |
| 14:55.46 | grandpapadot | Katty: Broken at that. |
| 14:55.47 | Katty | i consider Public School to be socialist. |
| 14:55.56 | grandpapadot | Katty: And they are broken too. |
| 14:55.58 | Katty | yes. |
| 14:56.00 | Katty | they are. |
| 14:56.06 | Katty | but it's still socialism :P |
| 14:56.07 | grandpapadot | Katty: Basically, anything our government does ends up broken. |
| 14:56.12 | Katty | personally, i like socialism. |
| 14:56.16 | grandpapadot | Katty: Except for the US military. |
| 14:56.21 | Katty | i think the private sector needs competition. |
| 14:56.30 | InfoNutz | were socialism/capitalism |
| 14:56.38 | servilio | hi all! I am having a mismatch between the configuration automatically generated for an Asterisk server and the configuration a system I am setting up expects it to be in, can someone give me hand at troubleshooting it? I need to decide if I need to tweak the Asterisk config or the PHP code of the system |
| 14:56.44 | Katty | grandpapadot: well, that may be true. but we still need government. |
| 14:57.27 | grandpapadot | Katty: Yes, but small federal government with the power pushed back to the states (that way if you don't like your states policies, taxes, views, etc, you can move to another) |
| 14:58.00 | grandpapadot | Katty: I think this is what was originally intended (The United STATES of America) |
| 14:58.11 | Katty | grandpapadot: i definately agree with that. |
| 14:58.20 | Katty | grandpapadot: but there are still things that need to be federally controlled. |
| 14:58.36 | grandpapadot | Katty: The fed has no business in business and why the f*ck is the federal government even involved in things like abortion, healthcare, etc, it should all be state level functions. |
| 14:59.00 | grandpapadot | Katty: Yep, the fed should control intl relations + trade, military, and interstate commerce and currency. |
| 14:59.02 | Katty | grandpapadot: we're currently seeing what happens when the government stays out of healthcare. |
| 14:59.05 | Katty | grandpapadot: it's eating us alive. |
| 14:59.23 | Katty | grandpapadot: or perhaps, killing us. ^_- |
| 14:59.27 | Katty | quite literally. |
| 14:59.32 | grandpapadot | katty: No way, I was the CIO for a chain of hospitals for 10 years, federal regulation of healthcare is what has f*cked up our healthcare system. |
| 15:00.02 | Katty | grandpapadot: well i'm not going to get into an arguement over which way is best. |
| 15:00.08 | grandpapadot | katty: we may want to take this off channel, lol, if you want to hear my view from the inside |
| 15:00.08 | Katty | grandpapadot: but it is quite obviously that the status quo is el sucko |
| 15:00.10 | [TK]D-Fender | Katty: the real money to be made is in the middle.... with people who are alive (sort of), but with one or more chronic conditions to make them the health-care cash-cows that they are... |
| 15:00.14 | *** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler) |
| 15:00.22 | grandpapadot | katty: I agree that the status quo is el sucko |
| 15:00.24 | *** join/#asterisk galeras (n=galeras@186.80.181.115) |
| 15:00.32 | servilio | grandpapadot, Katty: my opinion is that they should regulate, but not *administer* the healthcare |
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| 15:01.52 | Katty | servilio: well, that's not the plan anyway. |
| 15:01.53 | grandpapadot | servilio: I think they should let health care business complete without regulation, the dirt bags will loose business to the guys that are doing it right and the problems will work them selves out. |
| 15:02.00 | Katty | servilio: they're not wanting to take OVER the current healthcare system. |
| 15:02.11 | Katty | servilio: they want a complimentary, public, option... to the private option |
| 15:02.13 | grandpapadot | Katty/servilio: Yet. |
| 15:02.18 | Katty | servilio: would would bring a little competition to the market. |
| 15:02.37 | Katty | servilio: maybe make the current healthcare industry redo a few things. |
| 15:02.38 | grandpapadot | Katty/servilio: It's still a turn down the road to socialism which inevitably leads to communism. |
| 15:02.55 | servilio | Katty, they can't make a drastic change, the whole system might rebel :) |
| 15:02.57 | Katty | grandpapadot: would you rather keep the status quo? |
| 15:03.01 | garymc | still didnt find out what hardware i needed then |
| 15:03.10 | coppice | yeah, because the whole of western europe went commie years ago, didn't it :-) |
| 15:03.19 | Katty | coppice: it sooooo did. |
| 15:03.25 | grandpapadot | Katty: no, I think we need to deregulate healthcare and let the businesses compete. Competition yeilds lower prices and higher quality, history has proven tham. |
| 15:03.34 | servilio | grandpapadot: no regulation at all -> they do what they wish with us |
| 15:03.46 | servilio | grandpapadot: really bad |
| 15:03.56 | Katty | i'm not sure i like the idea of deregulating healthcare. |
| 15:04.03 | grandpapadot | servilio: "They" are just business owners. The bad apples will fail, the good ones will success. |
| 15:04.07 | grandpapadot | s/success/succeed. |
| 15:04.39 | Katty | i'm sure some of the regulations are good. |
| 15:04.41 | Katty | others, not so much. |
| 15:04.46 | grandpapadot | Katthy: have you worked in healthcare and seen what regulation does? This is fact: regulating healthcare forces the underlying business to make choices based on MARGIN and not what's in the best interest of the patient. |
| 15:05.06 | grandpapadot | Katty: We spent $4M on software that let us work the regulatory system to our advantage. |
| 15:05.23 | servilio | grandpapadot: I don't mean tight grip, but something that makes the ones running/administering health care businesses *responsible* and *accountable* |
| 15:05.24 | grandpapadot | Katty: This is how healthcare businesses compete today. |
| 15:05.30 | Katty | no. i'm not in the healthcare industry. but quite frankly something must be done. |
| 15:05.44 | Katty | i dont' claim to know the right way. |
| 15:05.52 | Katty | only that if we do nothing, we're screwed. |
| 15:05.59 | grandpapadot | Katty: I agree completely that something needs to be done. |
| 15:06.02 | Katty | so i'm up for doing something, even if it leads down the wrong road for awhile. |
| 15:06.10 | Katty | until we find whatever works. |
| 15:06.19 | servilio | Katty: agreed |
| 15:06.40 | Katty | and i'm not going to scream socalism anytime the government wants to step in and try something new. |
| 15:07.48 | *** join/#asterisk Rico29 (n=Rico@lns-bzn-48f-81-56-220-28.adsl.proxad.net) |
| 15:07.49 | servilio | grandpapadot, Katty: what I know for sure is that total deregulation won't make them care for the patient either, and furthermore, the patients might not have a way of making them do the right thing |
| 15:08.03 | Katty | well complicated problems are never fixed with one thing. |
| 15:08.07 | grandpapadot | Katty: here's my opinion, deregulate, here's why (hypothetical): I'm a health care entreprenuer, I'm going to do *everything* in my power to offer higher quality care at a lower price to ensure I get your business back or at least referrals and a good reputation. I'm going to try to stomp my competition by being better than they are, this includes giving free care to those who can't afford it. With the current state of re |
| 15:08.59 | grandpapadot | Katty: You get a bunch of hungry entreprenuers competing, we'll end up with affordable patient visits and incredible quality of care. |
| 15:09.40 | Katty | grandpapadot: that may be all fine and dandy, and maybe you're right, but we haven't gotten to that point yet. |
| 15:09.55 | *** join/#asterisk s14ck (n=s14ck@190-76-66-252.dyn.movilnet.com.ve) |
| 15:10.21 | grandpapadot | Katty: But at the end of the day, it's great to live in this country, and to be able to have an opinion. Thanks for the debate. |
| 15:10.24 | Katty | grandpapadot: so you really only have two options. |
| 15:10.32 | Katty | grandpapadot: keep status quo, or change. |
| 15:10.45 | Katty | grandpapadot: yesh. |
| 15:10.51 | grandpapadot | Katty: I 100% agree that change is necessary. |
| 15:10.56 | *** join/#asterisk ArchGT (n=ArchGT@unaffiliated/archgt) |
| 15:10.58 | *** join/#asterisk bmoraca (n=chatzill@66.242.174.254) |
| 15:11.10 | grandpapadot | Katty: I just don't think having the government involved will help, only hurt. |
| 15:11.24 | grandpapadot | Katty: big government is bad, m'kay |
| 15:11.33 | Katty | well, look at it this way. |
| 15:11.39 | Katty | without getting them involved, we stay status quo |
| 15:11.44 | [TK]D-Fender | grandpapadot: No, dumb gov't = bad |
| 15:11.46 | servilio | grandpapadot: I've seen with my own eyes how much damage can do total control from the gov |
| 15:11.59 | *** join/#asterisk Defraz (n=T0tal@corp.fuzecore.com) |
| 15:12.02 | Katty | at some point, the government HAS to step in. |
| 15:12.06 | Katty | whether for the good, or the bad. |
| 15:12.13 | grandpapadot | TK: I don't think they are mutually exclusive, big is slow and slow is dumb |
| 15:12.17 | [TK]D-Fender | Size just allows dumb to do more damage. Put the blame where it belongs |
| 15:12.35 | beek | <PROTECTED> |
| 15:12.40 | servilio | grandpapadot: but I don't believe that deregulation to the point where businesses/adminstrators aren't accountable and responsible is a good way |
| 15:13.10 | [TK]D-Fender | grandpapadot: Big is slow? Go piss off a hippo or carcharodon carcharias, and see what "slow is ;) |
| 15:13.18 | grandpapadot | servilio: They would be accountable, but to their customers, not to a regulating entity. If customers leave, they go out of business and aren't a problem anymore. |
| 15:13.42 | InfoNutz | lol Fender |
| 15:14.09 | bmoraca | the US government was not conceived to control our lives, it was designed to provide common currency and common foreign policy between the states and provide for interstate trade regulation |
| 15:14.32 | bmoraca | civil matters were supposed to be handled expressly by the states |
| 15:14.36 | grandpapadot | servilio: Perfect example, General Motors. The customers spoke, we don't want their cars (at the price stated). What happened? Our government stopped the natural cirle of communism from stomping them out. |
| 15:14.40 | servilio | grandpapadot: you are assuming that customers can properly assess they are receiving the proper service |
| 15:14.42 | grandpapadot | capitalism! capitalism! |
| 15:14.43 | grandpapadot | yikes |
| 15:14.58 | Katty | watches the conversation go by, sips redbull |
| 15:15.11 | Katty | i just want to survive the next 10 years. |
| 15:15.12 | *** join/#asterisk marv[work] (n=timr@router.asteriasgi.com) |
| 15:15.15 | [TK]D-Fender | grandpapadot: Indeed the bailouts = BS. |
| 15:15.26 | coppice | the US would probably have a healthy car industry if GM had been allowed to collapse in the early 80s |
| 15:15.37 | grandpapadot | servilio: well, there's no accounting for stupidity I guess. In todays modern information everywhere age, there's no excuse for not researching what your doc is doing to you. |
| 15:15.50 | grandpapadot | coppice: I agree completely with that. |
| 15:16.07 | Katty | well unfortunately the government is still ran by /people/ |
| 15:16.12 | Katty | and people know people, and care for people. |
| 15:16.21 | Katty | and are subject to the same personality flaws. |
| 15:16.22 | servilio | grandpapadot: then you are asking everybody to consult two doctors and be knowledgable on medical procedures? |
| 15:16.35 | coppice | the japanese hammer the car industry and the semiconductor industry. semis got no government help, and look where the US and the japanese are in semis today |
| 15:16.35 | grandpapadot | bmorca: here! here! |
| 15:16.44 | Katty | i'm sure someone knew someone at GM. |
| 15:16.45 | coppice | s/hammer/hammered |
| 15:16.47 | Katty | and made a deal. or whatever. |
| 15:16.50 | bmoraca | grandpapadot: you're assuming that GM had that amount of control over the price of their cars. however, when labor costs only go up and cannot be brought down even by laying off workers, GM was basically fucked. yes, management had problems too, but there was little they could have done about it while the UAW was still in power |
| 15:16.57 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 15:18.09 | coppice | bmorca: GM went the same way as the UK car industry. decades of antagonist between workers and managers, while places like germany and japan managed some level of cooperation |
| 15:18.09 | grandpapadot | bmoraca: No, i didn't assume that. I understand how the labor (unions) and their chain of businesses contributed to the cost of the vehicles. I was simply stating that, as a taxpayer, it was never *my* problem. I didn't *choose* to work at GM, I didn't *choose* to buy GM stock, why should I be *forced* to bail them out? |
| 15:19.05 | Katty | If GM collapsed, and all those people lost their jobs.... |
| 15:19.13 | grandpapadot | bmoraca: Actually, how *dare* they take taxpayer money to fix their problems. There would have been 20 other auto maker startups bounce into existance to take their place. Some would have succedded, some would have failed. |
| 15:19.17 | Katty | and i don't know how many people that would be with all the dealerships. |
| 15:19.25 | bmoraca | grandpapadot: i absolutely disagreed with the bailout, and i think that they fucked up the restructuring, too. but GM's demise was not because their cars (over the past 3-4 years) were inferior, but because their labor costs were more than triple that of competing manufacturers |
| 15:19.33 | Katty | don't you think ultimately everyone would be effected by it? |
| 15:19.36 | grandpapadot | Katty: Like I said, I didn't *choose* to work at GM, it's not *my* problem. |
| 15:19.36 | *** join/#asterisk sjobeck (n=Adium@208-151-246-203.dq1sn.easystreet.com) |
| 15:19.37 | coppice | Katty: why would all those people loose their jobs? |
| 15:19.57 | Katty | coppice: if GM goes under, don't GM dealerships go under? |
| 15:20.02 | grandpapadot | Katty: Not that I don't care, I meant it's not *my* problem from a tax liability perspective. |
| 15:20.21 | [TK]D-Fender | Katty: Maintaining the lie is expensive and it directly hits the posckets of people that didn't want to buy cars from them in the first place. that is ouright theft. |
| 15:20.24 | Katty | grandpapadot: no perhaps not your personal problem, but my father works for GM |
| 15:20.34 | coppice | Katty: if there is a demand for cars big enough to support so many people selling them, they'll find some cars to sell |
| 15:20.35 | Katty | grandpapadot: therefor, it effects me. |
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| 15:20.40 | bmoraca | grandpapadot: make no mistake, i'm a libertarian and a constitutionalist. i am vehemently opposed to government-run business and any government intervention in business as a whole. that's not their purpose. |
| 15:20.45 | [TK]D-Fender | Katty: time for a "market-based career change" |
| 15:20.59 | grandpapadot | All: At the end of the day, there is still a HUGE demand for vehicles, if GM goes under, there WILL be other companies pop up to take their place and employee folks. Not to mention the increased business to Ford and the others, they would hire as well to meet demand. |
| 15:21.06 | rob0 | Telephonists: I know I've seen here that no legitimate CID would be a US-tollfree number, is that right? Seems I am getting spam on my ipkall DID, and I am thinking about blocking US-tollfree numbers. |
| 15:21.08 | Katty | coppice: yes, perhaps (= |
| 15:21.19 | Katty | coppice: that is a good point. but difficult for someone 3 years away from retiring. |
| 15:21.32 | Katty | coppice: just a bad situation. |
| 15:21.43 | grandpapadot | Katty: Again, that was that persons choice to work for GM and count on their retirement, it's all about personal accountability. |
| 15:21.49 | bmoraca | GM is now owned by the gov't and the UAW, and because of that, I will likely never buy another GM car. it's sad, because the new Malibus and Camaros and some of the SUVs are excellent cars. |
| 15:22.06 | grandpapadot | Katty: in absolutely NO way should I be forced to fix someone elses poor choice through taxation. |
| 15:22.32 | *** part/#asterisk sjobeck (n=Adium@208-151-246-203.dq1sn.easystreet.com) |
| 15:22.37 | rob0 | NoOp("SIP/66.54.140.46-007ebe30", "ipkall-222 "TOLL FREE TF" <8007324504>") |
| 15:22.41 | grandpapadot | I own a 2007 GMC Sierra, best vehicle I've ever owned, too bad my next one will be a Ford. |
| 15:22.43 | bmoraca | Katty: what about the fact that GM workers were paid WELL above market average and competetive salary ranges for the last 40+ years? has no one managed to save any of that money? Michigan is not an expensive state to live in. |
| 15:23.17 | Katty | the point isn't whether or not you're being 'fined' for something you didn't 'do' |
| 15:23.17 | grandpapadot | s/entitlement/personal responsibility |
| 15:23.29 | drmessano | bmoraca: No, they bought GM's overpriced cars and then had to put fuel in them |
| 15:23.33 | rob0 | But granpappy, don't you know, that's what taxation is all about, financing poor choices?!? |
| 15:23.38 | Katty | at least in my opinion it's not. |
| 15:23.56 | Katty | life's not fair. get over that (= |
| 15:23.57 | rob0 | Hell, you pay for the drug war, and if that wasn't a poor choice, what is? |
| 15:23.58 | grandpapadot | rob0: apparently |
| 15:24.06 | coppice | the GMs they make in China seem a lot better than the ones they make in Europe and the US. anyone with such a disjointed product line deserved to fail |
| 15:25.56 | grandpapadot | Katty: That's the whole point isn't it? This country was setup for life to be fair, we've just messed it all up by making it not fair. |
| 15:26.25 | Katty | grandpapadot: life will never be fair, no matter where you go. |
| 15:26.47 | coppice | but life wasn't set up for life to be fair |
| 15:26.48 | grandpapadot | Katty: I don't mind paying taxes at all, but I do mind paying an unfair percentage. Whether you make $15K or $1.5M a year, the percentage of taxes paid should be equal. |
| 15:27.00 | Katty | grandpapadot: yes, i think we've all figured that out. |
| 15:27.23 | Katty | grandpapadot: but no matter what you want, and what you think is right, it's not going to happen. |
| 15:27.26 | Katty | grandpapadot: it doesn't work that way. |
| 15:27.30 | Katty | grandpapadot: you are given choices. |
| 15:27.32 | grandpapadot | lol, ok, jumping off my soap box, I love you all :) |
| 15:27.35 | Katty | grandpapadot: and of those choices, you must choose. |
| 15:27.46 | Katty | grandpapadot: so. yeah. |
| 15:27.47 | rob0 | Well, I hate to be argumentative, but I really believe that this gov't was set up (and is maintained) for the same reasons that every other gov't throughout all human history was: to preserve privilege for the privileged. |
| 15:28.10 | Katty | grandpapadot: life is what it is, and no matter how much you don't like it...it still is. |
| 15:28.33 | Katty | grandpapadot: and some point you just have to sit there, make your choice, and hope for the best. |
| 15:28.45 | rob0 | And I also hate to be almost on-topic, but any comment on blocking tollfree CID numbers like the spam caller I pasted above? |
| 15:29.09 | *** join/#asterisk ariel_ (n=chatzill@63.214.236.169) |
| 15:29.25 | Katty | rob0: i think i missed it. something about blacklist? |
| 15:29.31 | drmessano | rob0: 95% of tollfree calls are SPAM.. but there is that 5% |
| 15:29.45 | drmessano | I wouldnt feel 100% about blocking tollfree |
| 15:29.52 | ariel_ | hello |
| 15:30.03 | rob0 | Maybe redirect them to a menu? |
| 15:30.17 | drmessano | Redirect them to Barack |
| 15:30.41 | drmessano | If Barack talks to Barack, the echo cancellation should get them |
| 15:31.08 | superbeef | Is anybody aware of any write-ups somewhere discussing large asterisk deployents with multiple branches and a centralized dialplan or soething along those lines? |
| 15:31.09 | drmessano | A menu wouldnt be a bad idea |
| 15:31.12 | rob0 | Does echo cancellation cancel out excess buzzwords? |
| 15:31.22 | drmessano | Just needs to be a short menu |
| 15:31.52 | grandpapadot | superbeef: how many sites and how many endpoints at each site? |
| 15:32.48 | superbeef | 8-10 sites with 10-30 users, and then probably 1 or 2 sites with 200 (call center) |
| 15:33.05 | servilio | can someone point me out docs on that would help me figure out why SIP peers don't come up in with "Action: Status" in the AMI? |
| 15:34.09 | *** join/#asterisk aethelrick (n=richard@mail.mexuar.net) |
| 15:34.11 | superbeef | grandpapadot: basically we want extensiosn to connect to local PBX, but be able to manage extensions from a single box |
| 15:34.51 | [TK]D-Fender | superbeef: "manage" in what capacity? |
| 15:35.41 | superbeef | [TK]D-Fender: provisioning, queue assignment |
| 15:35.50 | aethelrick | greetings all. I'm using record() in my dialplan to allow a user to record their voice and dump it to a file on disk. This works well and has been up and running for a couple of weeks now. Today I started to get users complaining of the call being dropped before they have pressed the # key. I tried myself and verified the problem. Has anyone encountered this? |
| 15:36.51 | [TK]D-Fender | superbeef: centrally managed separate PBX's.... never heard of a system that does that. |
| 15:37.00 | [TK]D-Fender | superbeef: Best of luck with that... |
| 15:37.08 | aethelrick | my record command looks like this... record(somefile.wav|0|180) |
| 15:37.09 | superbeef | [TK]D-Fender: haha |
| 15:37.38 | *** join/#asterisk arpu (n=arpu@chello062178159144.10.14.univie.teleweb.at) |
| 15:37.39 | [TK]D-Fender | aethelrick: You say they start recorrding then get dropped... so it isn't your app call. |
| 15:37.45 | [TK]D-Fender | athSo go look at an actual failure |
| 15:37.49 | *** join/#asterisk brian (n=brian@unaffiliated/brian) |
| 15:37.51 | [TK]D-Fender | aethelrick: So go look at an actual failure |
| 15:38.13 | superbeef | [TK]D-Fender: i'm going off imaginary requirements from my boss who thinks in Avaya..... anyway I think I'm mainly looking for some good examples of larger deployments |
| 15:38.43 | [TK]D-Fender | superbeef: If the branches are independent then thats all there is to say for that. |
| 15:39.13 | aethelrick | [TK]D-Fender: yes, they start the recording process ok and the file gets written correctly (I have used playback() to verify this) but recording is abbruptly halted, a beep is played (I see this in the asterisk console) and the line is hungup |
| 15:39.20 | [TK]D-Fender | superbeef: Feel free to invent whatever you want of course. |
| 15:39.34 | [TK]D-Fender | aethelrick: And we see none of it |
| 15:41.40 | *** join/#asterisk propellerhead (n=yogurt2u@host197.190-227-252.telecom.net.ar) |
| 15:44.51 | rob0 | looks in the mirror when looking for failure! |
| 15:45.13 | rob0 | Ah, there she is, standing behind me. ;) |
| 15:45.48 | aethelrick | [TK]D-Fender: here's the dialplan http://pastebin.com/d643aea08 (I'm getting logs) I've blanked out actual phone numbers and passwords with X's |
| 15:47.10 | *** join/#asterisk CunningPike (n=CunningP@204.239.8.157) |
| 15:47.49 | aethelrick | [TK]D-Fender: here's logs http://pastebin.com/d327dfbdf (once again, phone numbers and passwords X'd out) |
| 15:49.12 | [TK]D-Fender | aethelrick: -- Executing [XXXXXXXXXXX@internal:7] Record("SIP/voxige-00775050", "bd-intro.wav|0|180") in new stack ,-- 0 secods of silence before aborting? Yeah, thats SMRT... |
| 15:49.18 | spck | anyone have any experience hooking up spans in dahdi? i'm using a redfone bridge to convert to TDMoE and am having trouble configuring the channels... |
| 15:49.34 | aethelrick | [TK]D-Fender: if you look at the second last line of the log you can see the spontaneous beep and then the hangup. This is occurring while the user is speaking into the handset. |
| 15:50.04 | aethelrick | [TK]D-Fender: according to asterisk manual 0 means "disabled" |
| 15:50.04 | [TK]D-Fender | aethelrick: The moment you gt any silence even in the middle. You know... like pausing the breath... |
| 15:50.16 | aethelrick | http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Record |
| 15:50.34 | *** join/#asterisk Moc (n=mochouin@pbx.moctel.com) |
| 15:50.46 | [TK]D-Fender | aethelrick: the WIKI is NOT a manual |
| 15:51.14 | lirakis | Are there any add ons that enable asterisk to send CDR via radius? |
| 15:51.25 | lirakis | ive seen some that do authentication |
| 15:51.53 | [TK]D-Fender | aethelrick: So either the silence detection tripped, the chose to stop recording with the exit key, or the max duration was reached |
| 15:52.06 | [TK]D-Fender | aethelrick: Go look at another call with FULL CORE DEBUG enabled |
| 15:52.32 | *** part/#asterisk Moc (n=mochouin@pbx.moctel.com) |
| 15:52.48 | aethelrick | I originally had it without the 0 for silence detection and the 180 for max duration just using record(somefile.wav) the behaviour is the same |
| 15:53.35 | *** join/#asterisk jaytee (n=jaytee@unaffiliated/jaytee) |
| 15:53.36 | *** join/#asterisk Moc (n=mochouin@pbx.moctel.com) |
| 15:53.45 | Moc | how does one reopen a ticket on mantis ? |
| 15:53.56 | [TK]D-Fender | aethelrick: the output you did show indicates it leaves normally. one of those 3 conditions was likely met |
| 15:54.23 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
| 15:55.02 | jaytee | free airport wifi FTW!!! |
| 15:55.37 | Moc | jaytee: sniffing your data |
| 15:55.57 | *** join/#asterisk puzzled (n=foobar@188.90.100.47) |
| 15:56.06 | Moc | is dhubbard here ? |
| 15:56.07 | Qwell | Moc: Are you the reporter? |
| 15:56.26 | aethelrick | [TK]D-Fender: I know silence wasn't the problem because I myself was cut of in the middle of saying a word. I also know that the max duration wasn't the problem because I was cut off at between 15 and 46 seconds into my recording. So I'm guessing that asterisk thinks the line was hung up or that the # key was pressed (neither of which is true) |
| 15:56.30 | Moc | Qwell: nope. As normal, bug get closed before they are resolved |
| 15:56.37 | Qwell | What bug? |
| 15:56.44 | Moc | 3450 |
| 15:56.47 | aethelrick | [TK]D-Fender: maybe the network connection got bad/congested?? |
| 15:57.15 | Moc | the original reporter probably given up on it to get it in |
| 15:57.24 | [TK]D-Fender | aethelrick: get rid of your silence detection risk |
| 15:57.27 | Qwell | Moc: ...it was closed because the code was committed. |
| 15:57.49 | Moc | Qwell: it was commited to *, but not libpri |
| 15:57.59 | Qwell | Yes it was. |
| 15:58.18 | Qwell | It says so on the notes.. Did you read them? |
| 15:59.13 | Nugget | is reminded of "English, ***********, do you speak it?" |
| 15:59.16 | aethelrick | [TK]D-Fender: will do |
| 15:59.17 | *** join/#asterisk xpot-mobile (n=james@66-182-88-85.static.sdyl010.digis.net) |
| 15:59.55 | Moc | Qwell: I just downloaded libpri-1.4.10.1 and code is not in |
| 16:00.02 | drmessano | .... |
| 16:00.04 | Qwell | what's your point? |
| 16:00.15 | drmessano | SVN? Can you grep it? |
| 16:00.41 | Nugget | heh |
| 16:00.58 | Nugget | git ain't no country I ever heard of. They do diffs in git? |
| 16:01.22 | Moc | my point is it doesn't make sence |
| 16:02.09 | drmessano | Moc: Being commited and there actually being a release including said commit are two different things |
| 16:02.13 | jaytee | "Sense, ************, can you spell it?" |
| 16:02.34 | drmessano | If you download from SVN, you'll get your patched code |
| 16:03.59 | Qwell | Moc: In the future, make your trollish comments elsewhere, please. |
| 16:04.41 | Moc | it commited to 1.4 branch in april... 1.4.10.1 was release june... |
| 16:05.08 | *** join/#asterisk HenrikBe (n=zapphir@d83-183-226-89.cust.tele2.se) |
| 16:05.33 | Katty | jaytee: lunchtime soon. |
| 16:05.39 | Qwell | Moc: That doesn't change anything. The releases weren't made in a way you might have wanted. That doesn't mean it wasn't committed, and it doesn't mean the issue isn't resolved. |
| 16:05.42 | Katty | jaytee: Arroz con Pollo |
| 16:05.53 | HenrikBe | Hi, is there any way to get the dialed number (${exten}) in a dialplan where s is used? I only get zero(0) when I try it? |
| 16:06.01 | *** part/#asterisk chnuebi (n=patrick@102-68.0-85.cust.bluewin.ch) |
| 16:06.10 | jaytee | Katty, rice with chicken huh? sounds good |
| 16:06.13 | Katty | jaytee: a spicey rice and chicken dish |
| 16:06.26 | Katty | jaytee: yes, the photo is lovely. |
| 16:06.36 | Moc | This is the kind of attitude and thinking why I quit using * |
| 16:06.37 | Katty | jaytee: and it doesn't look like it will take forever to make. |
| 16:06.52 | jaytee | i had some awesome tamales the other day from this little hole in wall tamale restaurant. |
| 16:06.58 | *** part/#asterisk Moc (n=mochouin@pbx.moctel.com) |
| 16:07.00 | Qwell | Moc: attitude? you wanted an explan |
| 16:07.05 | rob0 | awww |
| 16:07.09 | *** mode/#asterisk [+b *!*@*.moctel.com] by Qwell |
| 16:07.12 | Qwell | peace |
| 16:07.15 | Katty | i wish people were more positive :< |
| 16:07.22 | [TK]D-Fender | HenrikBe: ${EXTEN} is where you ARE. I'f you're in "s", then it's "s". |
| 16:07.23 | rob0 | That's so SAD. |
| 16:07.25 | Katty | jaytee: ooh, beef? |
| 16:07.29 | jaytee | so if the retard is looking for a patch why'd he quit using *? |
| 16:07.29 | Katty | jaytee: or chicken tamales? |
| 16:07.43 | Katty | jaytee: maybe he got tired of the negative atmosphere? |
| 16:07.47 | Katty | jaytee: who knows. |
| 16:07.49 | HenrikBe | TK-Fender: thanks!! |
| 16:07.49 | rob0 | Yeah, I was wondering the same thing ... |
| 16:08.07 | jaytee | Katty, I had one of each, shredded beef with mild red sauce, spicy chipotle beef and one chipotle chicken |
| 16:08.07 | ickmund | Is it possible to Chanspy via AGi? |
| 16:08.21 | [TK]D-Fender | ickmund: AGI is jsut another way to call dialplan apps. |
| 16:08.28 | Katty | jaytee: sounds delicious. i should make tamales or enchiladas soon. |
| 16:08.35 | Katty | jaytee: always been a fan. |
| 16:08.48 | jaytee | Qwell wasn't being negative, he gave him the info he needed, the little dick just didn't like what he heard. |
| 16:09.11 | Qwell | jaytee: worse, the way he started the question was accusatory |
| 16:09.18 | Qwell | simply because he didn't understand (or read...) the issue. |
| 16:09.20 | jaytee | Qwell, very true |
| 16:09.37 | Qwell | whatever - really don't care |
| 16:09.53 | jaytee | start off on that foot with me and I'm gonna stomp on it with one of my size 12's |
| 16:09.59 | Katty | Qwell: do you need a hug? |
| 16:10.04 | Katty | Qwell: i think you need a hug. |
| 16:10.08 | Katty | hugs on Qwell |
| 16:10.08 | Qwell | Katty: so over it :p |
| 16:10.09 | jaytee | I think he dows |
| 16:10.11 | jaytee | does |
| 16:10.23 | *** join/#asterisk ChkDigit (n=mike@static24-72-71-175.regina.accesscomm.ca) |
| 16:10.37 | Katty | maybe he's just having a bad day. |
| 16:10.46 | Katty | i don't think we need to resort to name calling. |
| 16:10.57 | [TK]D-Fender | Question : What's worse, ignorance or apathy? |
| 16:10.58 | [TK]D-Fender | Answer : I don't know, and I don't care... |
| 16:11.01 | jaytee | who? Moc? |
| 16:11.01 | aethelrick | [TK]D-Fender: I've changed the silence duration as you suggested. But I have also had a look in source code for the record.c app. silence=0 does indeed disable silence detection as far as I can tell and it is also the default. The paper manual is uninformative regarding defaults for this setting and as it happens the wiki was a better source of info. Ultimately though the code does not lie :) |
| 16:11.19 | Katty | let's shift this conversation to cookies! |
| 16:11.22 | Katty | and happy fluffy ducks. |
| 16:11.33 | [TK]D-Fender | aethelrick: Maybe your eyes do. Maybe I don't see a quality attempt to look at the problme. |
| 16:11.36 | Katty | and all things bright and cheery, that makes life worth living. |
| 16:11.42 | ickmund | [TK]D-Fender: From an Aastra phone I call a server script. Can I Chanspy from this (or run any other Application that requires a channel)? |
| 16:12.16 | [TK]D-Fender | ickmund: "call a server script". Could you actually frame this in some meaningful way. |
| 16:12.26 | rob0 | Fluffy duck cookies! |
| 16:12.28 | [TK]D-Fender | ickmund: And of course you need a channel |
| 16:12.33 | jaytee | a puppy licking your face!!! |
| 16:12.36 | HenrikBe | [TK]D-fender: I get 0 from s too |
| 16:13.05 | Katty | jaytee: feeding elephants little debbie cakes. |
| 16:13.06 | [TK]D-Fender | rob0: If it walks like a duck and quacks like a duck.... grab the chocolate chips, and GET HIM!!!!!!! |
| 16:13.19 | jaytee | Katty :-) |
| 16:13.31 | Katty | and napping in the afternoon, when it's raining. |
| 16:13.32 | [TK]D-Fender | HenrikBe: And I don't see you showing me your code, and the CLI output highlighting the failure. |
| 16:13.47 | jaytee | Katty, one of our white rhinos is pregnant! |
| 16:13.52 | [TK]D-Fender | HenrikBe: *'s variables work, your implementation does not. Pastebin is your friend <- |
| 16:13.54 | [TK]D-Fender | ~pb |
| 16:13.54 | 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://bin.cakephp.org/ , or apt-get install pastebinit |
| 16:13.56 | [TK]D-Fender | ^^^^^^^^^^^ |
| 16:14.03 | jaytee | and we just had a baby giraffe saturday |
| 16:14.15 | aethelrick | [TK]D-Fender: thank you for your suggestions so far. not sure if I've done something to offend you, but you're very abrasive, rude and about as helpful as a kick in the bollocks. |
| 16:14.24 | Katty | jaytee: see...if the world cared more about baby giraffes... |
| 16:14.33 | Katty | jaytee: and less about who can make the most profit over a cancer patient |
| 16:14.33 | [TK]D-Fender | jaytee: I'm so proud of you... how long have you been seeing the mother? ;) |
| 16:14.54 | Chainsaw | aethelrick: Fender will get more direct if you fail to answer questions asked of you. |
| 16:14.56 | jaytee | [TK]D-Fender, ever since I bought that stepladder on sale at Home Depot |
| 16:15.15 | [TK]D-Fender | jaytee: m1dg3t pr0n |
| 16:15.26 | *** join/#asterisk AndrewKT (n=chatzill@user-0c8h5rq.cable.mindspring.com) |
| 16:15.47 | jaytee | aethelrick, there are few people as knowledgeable and as helpful in this channel as [TK]D-Fender and if he seems abrasive it's most likely because you're not providing information he's asked you for. |
| 16:16.45 | jaytee | Katty, Jon Stewart made a real good point the other nite. In the US we are forced by law to have car insurance but not health insurance. |
| 16:17.05 | SuPrSluG | sorry |
| 16:17.35 | coppice | jaytee: you're not forced to have car insurance, as you're not forced to have a car |
| 16:17.51 | ickmund | [TK]D-Fender: Hmm... I'm trying here, but I can't find a more meaningful way of describing the event. Could you frame the question in a more meaningful way? |
| 16:18.13 | [TK]D-Fender | ickmund: How are you calling your AGI if you aren't in a channell? |
| 16:18.33 | jaytee | coppice, why don't you go take a 20 mile hike and then come back and tell me how your feet feel? |
| 16:18.41 | HenrikBe | [TK]D-fender: http://pastie.org/588665 |
| 16:18.46 | coppice | I don't have a car. |
| 16:18.57 | Katty | jaytee: see? |
| 16:19.01 | jaytee | coppice, but you live in Europe |
| 16:19.02 | Katty | jaytee: more about giraffees |
| 16:19.05 | *** join/#asterisk Carlos_PHX (n=carlos@ip68-108-193-174.ph.ph.cox.net) |
| 16:19.08 | Katty | jaytee: and less about...yeah. |
| 16:19.08 | aethelrick | jaytee: I answer questions, I wasn't asked any... I guess I caught [TK]D-Fender on a bad day, all I got for my questions was scorn. All I'm saying is a little polite goes a long way. |
| 16:19.13 | coppice | jaytee: no I don't |
| 16:19.18 | jaytee | no? |
| 16:19.19 | ickmund | [TK]D-Fender: Via one of many wrappers, specifically PHPAGI |
| 16:19.45 | *** part/#asterisk aethelrick (n=richard@mail.mexuar.net) |
| 16:19.49 | jaytee | coppice, HK? |
| 16:20.21 | coppice | yes. A city that has public transport. If you're american you might not be familiar with the term |
| 16:20.26 | [TK]D-Fender | HenrikBe: agents, s, 2, SayNumber, s <- you put "s" in there... "s" is not ${EXTEN} |
| 16:20.41 | [TK]D-Fender | henrWhy did you put a single letter "s" as the data? |
| 16:20.50 | grandpapadot | Hey will this crontab run every 3 hours -> * */3 * * * /some/cmd |
| 16:20.57 | [TK]D-Fender | ickmund: What is making that script get called? |
| 16:21.24 | ickmund | [TK]D-Fender: My phone makes a http request |
| 16:21.30 | mweichert_ | how can I force DND off for a particular extension? |
| 16:21.49 | SuPrSluG | coppice: that'd be socialist like the public option. lol |
| 16:21.52 | HenrikBe | [TK]: I tried ${exten} first, but it returned zero, so I thought that s would contain the number dialed |
| 16:22.03 | [TK]D-Fender | ickmund: You can't just go call an AGI script manually without a channel. AGI = CALL PROCESSING. you aren't in a call,. |
| 16:22.14 | Katty | oh boy here we go again. |
| 16:22.17 | [TK]D-Fender | HenrikBe: What you have shown me is in fact clearly wrong. |
| 16:22.17 | Kobaz | hmm |
| 16:22.33 | HenrikBe | [TK]: Hmm, okay |
| 16:22.33 | Kobaz | in the new dahdi 2.2.0.2, make install doesn't install the dahdi perl helper modules |
| 16:22.35 | Katty | you guys can scream socialism all afternoon (= |
| 16:22.38 | Katty | i'm going to go make lunch. |
| 16:22.45 | Kobaz | Loading DAHDI hardware modules: |
| 16:22.46 | Kobaz | Can't locate Dahdi/Span.pm in @INC |
| 16:22.46 | [TK]D-Fender | mweichert_: Same answer as before. FreePBX is NOT supported here. |
| 16:22.47 | Kobaz | ...etc |
| 16:22.51 | ickmund | [TK]D-Fender: Sure I can, I have several scripts that does it. Executing 'sip show peers' doesn't require a channel to mention one examples, which is an excellent way of creating a directory of online peers |
| 16:22.59 | coppice | SuPrSluG: it can't socialist. we don't have any death squads for people who can't fit on the train during rush hour |
| 16:23.00 | SuPrSluG | ? |
| 16:23.13 | [TK]D-Fender | ickmund: "sip show peers" is a CLI command. chanspy is a dialplan app <- |
| 16:23.54 | [TK]D-Fender | ickmund: And being an AGI implies using AGI commands. If you use some AMI methods in the PHP-AGI class that does not make your script an AGI |
| 16:23.59 | Kobaz | hi Katty |
| 16:24.16 | [TK]D-Fender | ickmund: Nor can you go calling dialplan apps at random |
| 16:24.40 | mweichert_ | [TK]D-Fender, sorry I thought maybe there was some asterisk-specific way of turning off DND |
| 16:25.17 | [TK]D-Fender | mweichert_: * can't tell phones what to do about that, and any DND implemented in the dialplan is back to FreePBX |
| 16:25.27 | bmoraca | mweichert_: DND is not a part of asterisk. there are various implementations of it, from the phone to dialplan and database driven apps, but asterisk, in and of itself, does not do DND |
| 16:25.58 | [TK]D-Fender | bmoraca: Except analog ZAP/DAHDI FXS |
| 16:26.12 | bmoraca | right |
| 16:26.20 | [TK]D-Fender | Which I never advise |
| 16:26.22 | bmoraca | but who uses those anyway? :P |
| 16:26.28 | [TK]D-Fender | bmoraca: Exactly :p |
| 16:27.00 | spck | does anyone have an example chan_dahdi.conf? |
| 16:27.04 | ickmund | [TK]D-Fender: I was hoping there was a way to create a channel somehow, much in the same way a dailer connects both ends of the conversation |
| 16:27.29 | bmoraca | ickmund: you can program your PHP app to issue an AMI originate |
| 16:27.42 | bmoraca | or drop a call file in the spool directory |
| 16:27.47 | [TK]D-Fender | ickmund: What "both ends"? You are calling this php script manually from some other process. What 2 calls? |
| 16:28.23 | *** join/#asterisk CunningPike (n=CunningP@204.239.8.157) |
| 16:28.30 | [TK]D-Fender | spck: your * tarball has one |
| 16:28.46 | [TK]D-Fender | spck: And its virtually identical to zapata.conf |
| 16:28.53 | ickmund | bmoraca: I'll check it out, thanks |
| 16:29.34 | bmoraca | ickmund: to be fair, i have no clue what your end goal is...i was just providing two different ways to create an asterisk channel from a PHP script |
| 16:30.13 | [TK]D-Fender | bmoraca: Yeah, I keep my "psychic hotline" grade advice to a minimum |
| 16:30.33 | ickmund | bmoraca: A client wants an easy way to do barge with chanspy and then a one-hit-wonder button to whisper on the same extension, without upgrading to 1.6 which I understand can do this |
| 16:30.48 | [TK]D-Fender | bmoraca: If they can't express they full intent I'm not going to waste time and effort developing solutions for guesses I have to come up with myself |
| 16:30.52 | bmoraca | ickmund: set him up with iSymphony |
| 16:31.11 | [TK]D-Fender | their* |
| 16:32.25 | bmoraca | totally understandable...i didn't really feel like reading up in the conversation and AMI would do what he needs... |
| 16:32.33 | bmoraca | just not in the way he expects |
| 16:33.05 | ickmund | thanks bmoraca, I'll check into that product. Not what this client is looking for thou. |
| 16:33.36 | bmoraca | ickmund: iSymphony does what you're asking for. only difference is that it runs on the computer instead of the phone. |
| 16:34.02 | ickmund | bmoraca: Yea, that's a show stopper thou |
| 16:34.27 | bmoraca | well, good luck with that then :) |
| 16:34.54 | ickmund | bmoraca: I'll work around it with a two button solution, I don't see a feasible way to do this without upgrading. Thanks for your help thou |
| 16:40.34 | TSM | anyone here out in the LA area? |
| 16:40.50 | *** join/#asterisk Shaun2222 (n=Shaun222@ip68-5-154-233.oc.oc.cox.net) |
| 16:43.45 | spck | [TK]D-Fender: guess i was hoping for one that was actually working |
| 16:43.58 | Shaun2222 | anybody help me out with my failover setup. It's not working, right now my voicepulse account is disabled and returning '-- SIP/sjc-primary.voicepulse.com-026fed40 is circuit-busy' and my extensions setup is not jumping to the next path out. I understand why, but i'm curious how i should rewrite this section of my extensions.conf to fix the problem... http://pastebin.ca/1535281 |
| 16:44.41 | Shaun2222 | i was thinking that if there was a channel error CHANUNAVAIL would be all i would need to check for but obviously thats not the case. |
| 16:45.18 | [TK]D-Fender | spck: Redfone doesn't have a sample for you? |
| 16:45.35 | [TK]D-Fender | spck: because they are almost the only people how use TDMoE. |
| 16:47.00 | [TK]D-Fender | Shaun2222: I don't see you looking at the dialstatus in there... |
| 16:47.15 | coppice | radsys seem to have a lot of TDMoE stuff. I wonder how well it sells |
| 16:47.16 | Shaun2222 | exten => _X.,3,GotoIf($[${DIALSTATUS}!=CHANUNAVAIL]?11) |
| 16:48.59 | Shaun2222 | voicepulse sample configs check for CHANUNAVAIL so thats why i did, but they are now returning BUSY which in my case would not go on. |
| 16:49.36 | *** join/#asterisk manxpower (n=EWieling@69.73.94.162) |
| 16:49.50 | manxpower | ztmonitor won't tear down existing calls will it? |
| 16:50.04 | *** join/#asterisk Daviey (n=Daviey@ubuntu/member/pdpc.gold.Daviey) |
| 16:50.35 | [TK]D-Fender | Shaun2222: You are testing the value without LOOKING at it |
| 16:50.55 | Shaun2222 | what do you mean? |
| 16:51.11 | spck | it's odd redfone has example configurations for almost everything but asterisk |
| 16:51.14 | spck | i really don't get it |
| 16:51.15 | [TK]D-Fender | Shaun2222: NoOp <- LOOK AT IT |
| 16:51.56 | Shaun2222 | why do i need to look at it? |
| 16:52.25 | [TK]D-Fender | Shaun2222: Because you are assuming the value that is coming back |
| 16:53.09 | Shaun2222 | i can see in the asterisk out put it telling me the circuit is busy |
| 16:53.18 | Shaun2222 | <PROTECTED> |
| 16:53.24 | *** join/#asterisk |Cybex| (n=John@80.100.126.176) |
| 16:53.30 | [TK]D-Fender | Shaun2222: "busy" !"chanunavail" |
| 16:53.39 | Shaun2222 | yes, i know. |
| 16:53.43 | [TK]D-Fender | Shaun2222: the message is worthless trash |
| 16:53.56 | servilio | what might be the cause of connected SIP clients not appearing in the list of open channels? |
| 16:54.02 | Shaun2222 | i understand why it's doing that, question was needing help with re-writing this so it did true failover. |
| 16:54.02 | [TK]D-Fender | Shaun2222: Please open you eyes and look at the variable |
| 16:54.30 | Shaun2222 | i was figuring BUSY would be sent if you called sombody and there number was busy |
| 16:54.33 | Shaun2222 | is that not the case? |
| 16:54.35 | [TK]D-Fender | Shaun2222: It is failing. and YOU are failing to examine what failure LOOKS LIKE |
| 16:54.37 | rob0 | Variablize with variable eyes |
| 16:54.49 | tzafrir_laptop | manxpower, no, it won't |
| 16:55.22 | *** join/#asterisk PuroOsso (n=PuroOsso@189.74.141.10) |
| 16:57.10 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 16:57.31 | Shaun2222 | [TK]D-Fender: -- Executing [19494124186@dialout:3] NoOp("SIP/222-0af52f00", "CONGESTION") in new stack |
| 16:57.38 | Shaun2222 | ok, so it's conjestion.. |
| 16:57.51 | ickmund | bmoraca: Solved it the old fashioned way... told the client it was impossible, so now he's happy doing it just about any way that works ;) |
| 16:58.08 | Shaun2222 | still brings me right back to my needing help with re-writing this so the failover works... what status's should i be looking for |
| 16:59.52 | [TK]D-Fender | Shaun2222: anything EXCEPT "busy" sounds like a failure to me. |
| 17:00.23 | InfoNutz | Hey, has anyone implemented a credit card authorization module for the IVR to pick up credit information, authorize the payment for a certain amount/min and then pass the call through asterisk? |
| 17:01.06 | [TK]D-Fender | InfoNutz: Almost certainly |
| 17:01.21 | Shaun2222 | what about ANSWER? |
| 17:01.33 | [TK]D-Fender | Shaun2222: Will never happen |
| 17:01.48 | Shaun2222 | why? |
| 17:01.51 | SuPrSluG | http://www.voip-info.org/wiki/view/Asterisk+variable+DIALSTATUS |
| 17:02.22 | SuPrSluG | http://www.voip-info.org/wiki/view/Asterisk+voicepulse+connect |
| 17:02.25 | [TK]D-Fender | Shaun2222: because you aren't looking at how you call Dial and you're forgetting how it works. |
| 17:03.22 | InfoNutz | Fender, you hear of anyone using paypals API with a AGI Perl script... i noticed one was on voip-info.org |
| 17:04.12 | [TK]D-Fender | InfoNutz: not specifically.. but a script is a script... |
| 17:05.33 | [TK]D-Fender | InfoNutz: I'm sure AGI doesn't care what odds & ends you do on the side |
| 17:06.30 | InfoNutz | true.. i'm finding very little information on the net about this, so it may prove difficult for me. paypal is changing their authentication gateway over to https webservice which is screwing up our pulse system, and they want 10 G plus to implement the change. i'm thinking asterisk as an alternative. |
| 17:07.11 | servilio | I am trying to integrate Asterisk with another systems that expects all the clients (connected or not) to be listed with the Status command of the AMI, is this a correct assumption? |
| 17:07.33 | servilio | I've been reading the online docs but haven't found any clue so far |
| 17:08.00 | servilio | neither by googling for the problem |
| 17:08.29 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 17:10.10 | InfoNutz | if you bittorrent "Asterisk the future of Telphony" you'll find a section in there about the AMI... there is also AMI Connector .NET that provides the interface for you to issue commands and get the information you need. |
| 17:11.27 | servilio | InfoNutz: thanks, will take a look! |
| 17:11.42 | Qwell | bittorrent? |
| 17:11.52 | Qwell | You're going to pirate a free book? seriously? |
| 17:11.53 | Qwell | ~book |
| 17:11.54 | 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 |
| 17:11.57 | [TK]D-Fender | InfoNutz: * has nothing to do with their scripts & API's |
| 17:12.09 | [TK]D-Fender | Torrenting a 3 meg download? |
| 17:12.13 | Qwell | and that |
| 17:12.14 | [TK]D-Fender | CRAZY PEOPLE |
| 17:12.30 | InfoNutz | lol i torrent everything |
| 17:12.51 | spck | any idea what causes this error: chan_dahdi.c:8748 mkintf: Unable to get parameters: Inappropriate ioctl for device? |
| 17:13.24 | spck | i get it when i try to load chan_dahdi.so |
| 17:17.15 | *** join/#asterisk generalhan (n=asd@about/windows/staff/generalhan) |
| 17:21.48 | *** join/#asterisk jcape (n=jcape@209.120.251.81) |
| 17:32.43 | InfoNutz | has anyone implemented a 3rd party merchant over voice with asterisk that will check the credit card and then continue on in the dialplan if the 3rd party merchant responds with yes/no and then complete the call? |
| 17:33.18 | Naikrovek | InfoNutz: i'm sure someone with a 1-900 number has figured that out |
| 17:33.44 | grandpapadot | InfoNutz: Try CyberSource w/their simply php gateway and then just do a php agi script, should be pretty easy. |
| 17:36.22 | InfoNutz | grandpapadot: thanks for that, i'll take a look. |
| 17:36.55 | *** join/#asterisk martyndev (i=be18869a@gateway/web/freenode/x-hbpgsyhnozgaqiff) |
| 17:36.57 | martyndev | Hi |
| 17:36.58 | martyndev | :d |
| 17:37.03 | *** join/#asterisk lost_soul (i=shawn@cpe-74-71-234-29.twcny.res.rr.com) |
| 17:38.00 | Naikrovek | is having real trouble with his inter-office VPN. what the deuce? |
| 17:38.37 | [TK]D-Fender | infoas I said, if they offer an API it functioning has nothing to do with * |
| 17:38.52 | *** join/#asterisk Samyray_ (n=chatzill@95.133.228.45) |
| 17:39.16 | Samyray_ | ÐобÑÑй Ð´ÐµÐ½Ñ |
| 17:39.24 | Samyray_ | Ðне Ð¼Ð¾Ð¶ÐµÑ ÐºÑо-Ñо на ÑÑÑÑком оÑвеÑиÑÑ? |
| 17:40.48 | bmoraca | InfoNutz: Authorize.net is a very easy-to-use payment gateway as well |
| 17:40.54 | martyndev | I have a analog card with a two gsm gateways. When i try recive a call and just detect with "Starting simple switch on 'Zap/249-1'" and if I hangup before Answer() will be execute, the call still active forever.. |
| 17:40.55 | martyndev | :'( |
| 17:40.57 | bmoraca | actually, they're owned by cybersource now |
| 17:41.03 | martyndev | Wha do you think ? |
| 17:41.12 | rob0 | Holy Unicode Batman! I can't read Samyray_ !! |
| 17:42.06 | rob0 | is now known as Robin |
| 17:42.06 | martyndev | in voip-info says that busydetect is necesaryin that case, but i have busydetect=yes and busycount=3 |
| 17:42.06 | martyndev | what can i do? |
| 17:42.06 | Samyray_ | It is very bad |
| 17:42.06 | Samyray_ | It was my russian words |
| 17:42.32 | InfoNutz | bmoraca: looks like its my best option so far, just going over the api's user guide. looks swift |
| 17:42.34 | rob0 | Ne goboriu po-russki! |
| 17:43.03 | Samyray_ | Ya yge ponal |
| 17:43.16 | Samyray_ | Tu ya tak ponal rysskoyazu4nui? |
| 17:43.18 | bmoraca | InfoNutz: i've implemented it a few times. works really well and is very easy to use |
| 17:44.09 | InfoNutz | bmoraca: you use PHP for authorization? |
| 17:44.33 | [TK]D-Fender | martyndev: "callprogress=yes" |
| 17:44.35 | *** join/#asterisk vgster (n=vgster@94-194-190-189.zone8.bethere.co.uk) |
| 17:44.52 | [TK]D-Fender | martyndev: You are receiving a call, not placing one. There is no "busy" |
| 17:45.07 | bmoraca | InfoNutz: i actually used ColdFusion the times I implemented it, but the idea and process is identical, and they provide you with functional demo code right out of the gate. plus, developer accounts are free for Authorize.net |
| 17:45.25 | Samyray_ | Can someone help noob? |
| 17:46.06 | Samyray_ | I want to try learn Asteriks at home,and I want to know,what hardware i must hawe to start it? |
| 17:46.16 | Samyray_ | *have |
| 17:46.30 | errr | Samyray_: get the free book |
| 17:46.37 | errr | ~book |
| 17:46.38 | 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 |
| 17:47.40 | Samyray_ | Okey,I will look,thanks |
| 17:48.01 | *** join/#asterisk trebaum (n=trebaum@ip68-8-175-208.sd.sd.cox.net) |
| 17:48.19 | InfoNutz | bmoraca: yeah i see that, was able to download the client for php, which is awesome. I'll just have to re-learn php and i'm new to asterisk, so that will be a hurdle. If asterisk was a .net application this would be installed today lol |
| 17:50.04 | generalhan | anyone here familiar with the dahdi setup using Xorcom Astribanks? when i use dahdi_genconf it sets my PRI T1 as an E1, but the dahdi services start up perfectly. when i change the span to represent the T1 dahdi wont start up anymore, i get the: DAHDI_SPANCONFIG failed on span 1: Invalid argument (22) |
| 17:50.58 | generalhan | this is my first attempt at building/running and Asterisk system without the cards inside the server itself. maybe im just missing something all together ?! :( |
| 17:51.37 | *** join/#asterisk phuff (n=user@sourceforge/staff/luapffuh) |
| 17:53.54 | bmoraca | InfoNutz: it can be done fairly easily with AGI...shouldn't take you long to do...the PHP isn't that advanced |
| 17:54.03 | *** join/#asterisk sah-work (n=Bawbatos@65-119-47-100.dia.static.qwest.net) |
| 17:54.56 | spck | generalhan: i've been struggling trying to configure dahdi for the past couple of days |
| 17:55.10 | spck | so you have my sympathy |
| 17:55.15 | grandpapadot | InfoNutz: Make the first line of your php file #!/bin/php and then don't forget to chmod +x and call it directly with agi. |
| 17:55.43 | generalhan | spck: well i cant do much with your sympathy ... so lets see if we cannot figure this out !? |
| 17:55.57 | grandpapadot | InfoNutz: err... #!/usr/bin/php |
| 17:56.06 | generalhan | spck: you seeing the same results as i am ? or slightly different ? |
| 17:56.34 | *** join/#asterisk Tar-Get (n=Tar-Get@83.101.83.53) |
| 17:57.19 | spck | i get chan_dahdi.c:8748 mkintf: Unable to get parameters: Inappropriate ioctl for device |
| 17:57.24 | spck | prolly a separate issue |
| 18:01.29 | *** join/#asterisk jcape (n=jcape@209.120.251.81) |
| 18:03.10 | *** join/#asterisk hammerzone45 (n=hammer15@c-71-229-108-12.hsd1.fl.comcast.net) |
| 18:03.23 | *** join/#asterisk DarkRift (n=dark@65.92.166.135) |
| 18:03.52 | hammerzone45 | question: how can i forward an incoming call to my asterisk server to another asterisk server located in another network so they can take the call? |
| 18:04.42 | [TK]D-Fender | hammerzone45: Dial() <- |
| 18:04.46 | grandpapadot | hammerzone45: Dial() |
| 18:05.12 | bmoraca | through the ether, man |
| 18:05.13 | *** join/#asterisk |Cybex| (n=John@80.100.126.176) |
| 18:05.32 | InfoNutz | bmoraca: grandpapado: thanks alot for your guys input, you helped me make a solid descision on this avenue. now for the proposal :) |
| 18:05.50 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 18:07.49 | hammerzone45 | like Dial(SIP\IP address) ?? |
| 18:09.08 | grandpapadot | hammerzone45: Setup sip.conf entries for each box on the other, then just Dial() like you would any other time, it's that easy. |
| 18:09.20 | nkohh | hello everyone. my asterisk server doesn't seem to be reading modules.conf on load. I am using my own config files, not the supplied defaults. Is there something I might be missing that causes it not to read that file on startup? |
| 18:09.39 | grandpapadot | nkohh: pastebin yuor modules.conf and also check the file permissions |
| 18:10.29 | grandpapadot | nkohh: also pastebin your asterisk.conf |
| 18:10.55 | *** part/#asterisk Tar-Get (n=Tar-Get@83.101.83.53) |
| 18:11.17 | *** join/#asterisk errotan (n=errotan@5403E5E1.catv.pool.telekom.hu) |
| 18:11.26 | hammerzone45 | grandpapadot: will a sip.conf entry in the "destination" server will be enough if i only want to transfer calls to it? |
| 18:12.45 | grandpapadot | hammerzone45: It will work, but I like to be a bit more explicit. |
| 18:13.38 | *** join/#asterisk |Cybex| (n=John@80.100.126.176) |
| 18:14.31 | manxpower | hammerzone45: you need an entry in the destination and a context with something to match the incoming call in extensions.conf |
| 18:15.32 | hammerzone45 | great, thanks grandpapadot, manxpower and D-Fender |
| 18:16.28 | *** join/#asterisk drakeman (n=drakeman@201.227.92.170) |
| 18:17.19 | drakeman | hey, , i want to install asterisk on debian to test and learn, can it be installed in a virtual machine and it works the same as if it was installed in a real machine? |
| 18:19.09 | *** join/#asterisk ctaloi (n=ctaloi@nat-66-218-1-155.usadatanet.com) |
| 18:19.09 | drakeman | cause, a friend of mine, is going to isntall it too, in another country a we want to make tests,\ |
| 18:19.10 | drakeman | call tests! |
| 18:19.10 | *** join/#asterisk scalex000 (n=chatzill@180.120.88.200.f.sta.codetel.net.do) |
| 18:19.10 | [TK]D-Fender | drakeman: VM's may have audio issues due to timeslive allocation. |
| 18:19.11 | shido6 | drakeman: yes , I would recommend vserver |
| 18:19.40 | eppigy | donde? |
| 18:20.13 | scalex000 | hello |
| 18:20.24 | carrar | harro |
| 18:20.51 | eppigy | hello |
| 18:20.52 | scalex000 | I need help to setup polycom IP phones |
| 18:20.55 | *** join/#asterisk lanning (n=lanning@173.8.187.197) |
| 18:20.58 | scalex000 | who know about it |
| 18:21.03 | carrar | sounds HOT++ |
| 18:21.19 | carrar | Did you read the Admin guide? |
| 18:21.35 | drmessano | Ojabart? |
| 18:21.37 | carrar | (Polycom Admin Guide) |
| 18:21.47 | scalex000 | Im readign |
| 18:21.54 | carrar | You're all set then |
| 18:21.56 | scalex000 | but i not clearly |
| 18:22.04 | drmessano | No ur not |
| 18:22.28 | scalex000 | in the admin guide say to unzip template etc |
| 18:22.34 | scalex000 | but I don't have it |
| 18:22.39 | carrar | the configs for them are in the firmware you downloaded from polycom web site |
| 18:22.59 | manxpower | www.fnords.org/~eric/polycom-config-examples |
| 18:23.03 | scalex000 | they have many link |
| 18:23.03 | carrar | configs being "templates" |
| 18:23.06 | drakeman | thanks guys! well, im going to install frist in virtualbox, and if everything is fine, then im going to instlal it in my machine! |
| 18:23.32 | scalex000 | carrar but they have differents link |
| 18:23.54 | carrar | explain? |
| 18:23.59 | scalex000 | iM not sure what to download. I not see if they have greater firmware |
| 18:24.06 | scalex000 | than I have |
| 18:24.11 | carrar | what model |
| 18:24.11 | shido6 | no |
| 18:24.18 | [TK]D-Fender | scalex000: http://downloads.polycom.com/voice/voip/sip_sw_releases_matrix.html |
| 18:24.48 | [TK]D-Fender | scalex000: Go download the latest appropriate firmware pack for your phone. It includes basic configs you can personalize |
| 18:25.08 | scalex000 | ok |
| 18:25.12 | scalex000 | I will |
| 18:25.30 | drmessano | AND GET OFF MY LAWN! |
| 18:25.42 | *** join/#asterisk hugorebelo (n=hugorebe@200-171-132-124.completo.com.br) |
| 18:25.52 | drmessano | Not you |
| 18:25.56 | drmessano | But thats ok too |
| 18:26.01 | scalex000 | fender let me try again |
| 18:26.03 | scalex000 | thanks for now |
| 18:26.10 | [TK]D-Fender | drmessano: Can you hear that? |
| 18:26.35 | drmessano | Hear what? |
| 18:26.37 | [TK]D-Fender | drmessano: Its the sound of no-one disagreeing with you :D |
| 18:26.42 | drmessano | lol |
| 18:29.42 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 18:31.04 | *** part/#asterisk hammerzone45 (n=hammer15@c-71-229-108-12.hsd1.fl.comcast.net) |
| 18:31.14 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.static.bezeqint.net) |
| 18:32.24 | generalhan | if anyone has a chance to look at my configs for my astribank device, the dahdi service will at least start if i leave my configs this way: < http://pastebin.com/d350d5da7 > but sets my PRI T1 port as an E1 port. by trying to change the system.conf file to represent my T1 settings the dahdi service will no longer start properlly all i changed was: < http://pastebin.com/d36525297 > can someone shed a little light on this for me ? |
| 18:32.50 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 18:33.25 | tzafrir_laptop | generalhan, you need to set it in /etc/dahdi/xpp.conf |
| 18:33.51 | tzafrir_laptop | basically: pri_protocol T1 |
| 18:33.56 | generalhan | tzafrir_laptop: i saw reference to that file in your docs ... but i dont even have that file at all. is this something that i need to create myself ? |
| 18:34.19 | generalhan | ... or should it be created by default |
| 18:34.27 | tzafrir_laptop | if that file does not exist, the defaults (E1 for that value) are used |
| 18:34.56 | generalhan | tzafrir_laptop: i just KNEW it was going to be something stupid that i missed. ill create that file now and give it a shot. thanks for the kick in the head ! |
| 18:37.37 | *** join/#asterisk mweichert_ (n=mweicher@216.13.154.21) |
| 18:38.30 | mweichert_ | how do I transfer a call in asterisk? When on the phone, I have been putting the call on hold and then pressing *2 but nothing happens - the person on the other end just hears the tones |
| 18:39.09 | *** join/#asterisk ehsjoar (n=ehsjoar@c-24-9-91-203.hsd1.co.comcast.net) |
| 18:39.38 | [TK]D-Fender | mweichert_: On what phone? |
| 18:40.07 | mweichert_ | Aastra 9480i |
| 18:40.23 | [TK]D-Fender | mweichert_: it has its own transfer buttons. * should not be doing this |
| 18:43.16 | *** join/#asterisk andresmujica1 (n=andresmu@ubuntu/member/andresmujica) |
| 18:43.40 | generalhan | tzafrir_laptop: i think that did it, i just created a file called xpp.conf and put pri_protocol T1 on the first line. saved it, reloaded dahdi and no errors this time., |
| 18:44.04 | generalhan | though i expected to see the lights stay on the FXS modules once it was loaded properlly |
| 18:44.51 | tzafrir_laptop | you should see lights on a properly-configured fxs port only if it is in use |
| 18:45.01 | tzafrir_laptop | (and flashing if it is ringing) |
| 18:45.13 | PuroOsso | [TK]D-Fender, http://pastebin.com/d743c5327 |
| 18:45.23 | PuroOsso | just happened |
| 18:45.38 | generalhan | tzafrir_laptop: cool! and a blinking sync light or a solid one ? |
| 18:47.07 | [TK]D-Fender | PuroOsso: Reliably Transmitting (NAT) to 10.100.100.42:20276: <-- you did not set up your localnet's properly. this should not be (NAT) |
| 18:48.10 | PuroOsso | and how do I do that? I'm really new... don't know nothing yet |
| 18:48.26 | [TK]D-Fender | PuroOsso: Go ask for support in FreePBX. |
| 18:48.28 | *** join/#asterisk zorro1 (n=zorro@69.77.169.14) |
| 18:48.40 | [TK]D-Fender | PuroOsso: You are in the wrong channel |
| 18:51.06 | PuroOsso | [TK]D-Fender, but if my localnet's isn't set up properly every call would be mute |
| 18:51.23 | [TK]D-Fender | PuroOsso: I also have no idea what to make of your "DGV" |
| 18:51.41 | [TK]D-Fender | PuroOsso: Sorry, This is not something I'm going to be able to help you with. |
| 18:52.32 | *** join/#asterisk |Cybex| (n=John@80.100.126.176) |
| 18:52.34 | PuroOsso | [TK]D-Fender, any idea? do u know where I can find more help? |
| 18:52.47 | manxpower | ~freepbx |
| 18:52.47 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 18:53.09 | PuroOsso | I have to solve this problem... |
| 18:53.23 | PuroOsso | as fast as "I" can.. |
| 18:53.47 | Nugget | ahh http://macnugget.org/stuff/asterisk-irc.txt |
| 18:54.01 | manxpower | PuroOsso: then ak on #FreePBX |
| 18:54.12 | manxpower | or contact the FreePBX company and use their paid support. |
| 18:54.34 | PuroOsso | manxpower, ok.. thanks! |
| 18:57.27 | bmoraca | fonality are a bunch of tards |
| 18:57.52 | bmoraca | we looked at fonality when we first got in to asterisk and the impression was not good |
| 18:59.21 | *** join/#asterisk theHub (n=theHub@69.177.93.21) |
| 19:00.40 | *** part/#asterisk zorro1 (n=zorro@69.77.169.14) |
| 19:02.02 | *** join/#asterisk phunyguy (n=phunyguy@h69-130-68-217.kgldga.dsl.dynamic.tds.net) |
| 19:02.55 | *** join/#asterisk xpot-mobile (n=james@66-182-88-85.static.sdyl010.digis.net) |
| 19:03.10 | *** join/#asterisk Olobola (i=Olobola@7.sub-70-212-133.myvzw.com) |
| 19:03.42 | *** join/#asterisk ehsjoar (n=ehsjoar@c-24-9-91-203.hsd1.co.comcast.net) |
| 19:05.29 | *** join/#asterisk asteriskmonkey (n=philip@69.77.169.14) |
| 19:05.41 | PuroOsso | manxpower, no help at #FreePBX =/ |
| 19:06.47 | *** join/#asterisk meesterarend (n=frans@vc-41-192-94-177.umts.vodacom.co.za) |
| 19:07.32 | *** join/#asterisk martyn-dev (i=be18869a@gateway/web/freenode/x-chdrnwfjddedqszz) |
| 19:07.37 | martyn-dev | Hi |
| 19:08.08 | manxpower | PuroOsso: there is help here either. |
| 19:08.20 | martyn-dev | [TK]D-Fender: thank you, i'm going to test with "callprogress=yes", but i can to do this change on zapata.conf at night, cause on day the operation still workiing.. |
| 19:12.21 | drmessano | http://macnugget.org/stuff/asterisk-irc.txt <--- Nugget, you just put me out of business. |
| 19:12.25 | drmessano | :( |
| 19:12.42 | drmessano | I...... Um. My work is done here. |
| 19:12.47 | *** join/#asterisk asterwiki (n=asterwik@69.77.169.14) |
| 19:13.51 | PuroOsso | manxpower, do u have any idea for me? |
| 19:15.26 | *** join/#asterisk lasko (n=lasko@70.102.15.210) |
| 19:15.48 | manxpower | PuroOsso: no. Nobody here uses FreePBX. |
| 19:15.59 | carrar | hahah |
| 19:16.01 | *** join/#asterisk |Cybex| (n=John@80.100.126.176) |
| 19:16.02 | manxpower | The people that use FreePBX are on the #FreePBX channel. |
| 19:16.10 | manxpower | Just how many times do people have to tell you this? |
| 19:16.29 | *** join/#asterisk geneticx (n=ethicx@host-208-88-126-198.biznesshosting.net) |
| 19:16.43 | carrar | People still use FreePBX? |
| 19:16.49 | grandpapadot | drmessano: lol, that's freakin' great |
| 19:16.55 | lasko | What would be the easiest way or disabling a sip device in sip.conf without disallowing it to register? I was thinking disallow=all and setting allow=None would this work as a way to disable a device temporarily? |
| 19:17.00 | drmessano | ~nat |
| 19:17.01 | infobot | nat is probably Network Address Translation Usable in Asterisk sip.conf file with externip, localnet, and localmask setup properly. See docs. |
| 19:17.05 | drmessano | ~sipnat |
| 19:17.06 | infobot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 19:17.18 | drmessano | ^^^^^^^^^^^^^^^^^^^ PuroOsso |
| 19:17.19 | PuroOsso | manxpower, =/ |
| 19:17.23 | *** join/#asterisk marhbere (i=c8319add@gateway/web/freenode/x-pufyoxsmvizwfaub) |
| 19:17.40 | manxpower | lasko: point the device to a context that can't do anything context=wheverever for that device |
| 19:17.41 | [TK]D-Fender | lasko: Depends on your idea of "disable" |
| 19:17.43 | PuroOsso | drmessano, I am not behind a NAT |
| 19:17.44 | rob0 | drmessano/Nugget, lol, that's excellent! |
| 19:18.03 | lasko | manxpower aha.. That is simple.. don't know why I didn't think of that lol |
| 19:18.29 | drmessano | I just retweeted... thats 100% Nugget.. and he just made me feel like an unfunny bastard |
| 19:18.36 | manxpower | lasko: that would also allow you to send all calls from the device to a message saying "Pay your bill, idiot" or something like that |
| 19:18.38 | drmessano | bows to Nugget |
| 19:18.47 | lasko | manxpower: Thanks |
| 19:18.51 | manxpower | drmessano: you are an unfunny bastard. |
| 19:19.15 | TJNII | snorts at asterisk-irc.txt |
| 19:19.33 | drmessano | manxpower: I don't take compliments well, so please, stop |
| 19:20.24 | drmessano | I would have tweeked a little more drama out of the urgency |
| 19:21.23 | drmessano | 1. I need help 2. Heres all the wacked out newb shit I just did to get to where I am now 3. My boss is on my ass 4. You guys are laughing at me, and this is very serious, WTF, HALP ME NOW |
| 19:22.30 | TJNII | Has anyone run asterisk in vmware with any success? I'm considering it because I want to virtualize my server with vmware, but I'm thinking I might run into timing issues. Can anyone confirm / deny? |
| 19:22.38 | Nugget | :) |
| 19:23.35 | grandpapadot | TNJII: Seems to work great with VMWare Workstation 6, debian 4/5, asterisk 1.4.25+, but we just use it for testing/dev purposes, never tried to load it up beyond 10-15 calls. |
| 19:24.05 | TJNII | grandpapadot: That's good to know. I use it for a home PBX, so low volume. I do need it to be reliable, however. |
| 19:24.30 | TJNII | This is also why I want to virtualize, I want my setup to look like 4 servers, but I don't want to pay to power 4 servers. |
| 19:25.08 | *** join/#asterisk Deeewayne (n=dwayne@75.76.254.162) |
| 19:25.08 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 19:25.29 | grandpapadot | Anyone have one of those problems with their significant other towhere there is not actually a problem but the fact that there could be is a problem and it's a problem? Wow. |
| 19:26.15 | *** join/#asterisk andres833 (n=andres83@190.144.75.22) |
| 19:30.14 | Nivex | I just moved my home PBX (read: very few calls) into a KVM. Heavy I/O loads on the host make it a bit choppy, and I don't have MeetMe since DAHDI doesn't work right |
| 19:30.24 | *** join/#asterisk crazybyte (n=crzp@unaffiliated/crazypenguin/x-000001) |
| 19:32.00 | denon | Nivex: might try it under ESXi |
| 19:32.05 | denon | I know a few people happy with it |
| 19:32.16 | *** join/#asterisk arpu (n=arpu@chello062178159144.10.14.univie.teleweb.at) |
| 19:32.29 | RypPn | I have a working asterisk instance on vmware-server, it needed a few things doing to make it work right, no cpuspeed or powernow enabled on the host, irqbalance installed and amd cool n quiet disabled, works fine with dadhi/confs, but its not under any kind of load |
| 19:32.55 | *** join/#asterisk asterwiki (n=asterwik@69.77.169.14) |
| 19:33.20 | asterwiki | testing |
| 19:34.59 | Nivex | denon: it's a home server for tinkering on. I don't have the dedicated hardware to throw ESXi at |
| 19:35.11 | Nivex | denon: I figure once 1.6.2 is relased I'll start using ConfBridge |
| 19:35.31 | *** join/#asterisk Tim_Toady (n=moi@adsl52-231.kln.forthnet.gr) |
| 19:35.35 | Nivex | upgrades to the host OS/KVM will probably help in the future too |
| 19:41.24 | Naikrovek | is there any way to tell what's up with an IAX2 trunk when both ends show each other as "UP" but when dialling i get "all circuits are busy" |
| 19:46.17 | [TK]D-Fender | Naikrovek: Go look at the call in CLI |
| 19:46.34 | [TK]D-Fender | Naikrovek: enable IAX2 debug, look for Dial parm errors, etc |
| 19:47.56 | Naikrovek | <PROTECTED> |
| 19:48.12 | [TK]D-Fender | Naikrovek: that is after the face |
| 19:48.24 | [TK]D-Fender | Naikrovek: And does not include IAX2 debug |
| 19:48.27 | [TK]D-Fender | ~pb |
| 19:48.28 | 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://bin.cakephp.org/ , or apt-get install pastebinit |
| 19:48.29 | [TK]D-Fender | ^^^^^^^^ |
| 19:48.31 | [TK]D-Fender | fact* |
| 19:48.33 | drmessano | grandpapadot: When you have a significant other, every day of your life you live with the potential that though you may be the nicest man on the planet, you could also have been a mean, spiteful bastard like her ex/dad/brother/first pet, and she's keeping a close eye on you, man |
| 19:48.39 | Naikrovek | one line doesn't require pastbin... |
| 19:49.35 | drmessano | grandpapadot: Just because nothing is wrong, doesnt mean whatever you did or didnt do couldnt have gone horribly wrong, you jerk |
| 19:50.54 | rob0 | Hey, I resemble that. |
| 19:51.45 | [TK]D-Fender | Naikrovek: IAX2 debug will generate a LOT more unless * is deciding not to even try |
| 19:51.57 | drmessano | rob0: And damn you for just thinking it, even though you may or may not have! |
| 19:52.00 | [TK]D-Fender | Naikrovek: And I don't see what you're dialing. |
| 19:52.15 | Naikrovek | well gimme a minute to finish putting this on pastebin |
| 19:53.31 | *** join/#asterisk frk2 (n=frk2@zivios/member/fkhan) |
| 19:54.08 | Naikrovek | [TK]D-Fender: http://pastebin.com/d2be49ca2 |
| 19:54.51 | [TK]D-Fender | Naikrovek: CAUSE : No authority found <-- rejected |
| 19:55.20 | Naikrovek | no authority? wtf does that mean |
| 19:55.35 | [TK]D-Fender | Naikrovek: Means they don't like you're request |
| 19:55.43 | Naikrovek | clearly, but why |
| 19:55.44 | Naikrovek | hrm. |
| 19:55.52 | Naikrovek | and why didn't i see that when i looked at that paste |
| 19:55.54 | [TK]D-Fender | Naikrovek: http://www.voip-info.org/wiki/view/Asterisk+No+authority+found |
| 19:55.55 | Naikrovek | wtf is up with ME? |
| 19:56.02 | Naikrovek | ty |
| 19:56.29 | citywok | [TK]D-Fender: i figured out my problem with the sip-info header. i had to add a pattern for both legs of the call in the context, and add the hedaer under the internal leg of the call |
| 19:56.34 | *** join/#asterisk |Cybex| (n=John@80.100.126.176) |
| 19:56.56 | [TK]D-Fender | citywok: OK/fine/sure |
| 19:58.11 | *** join/#asterisk heliosj (n=jeff@178.199.8.67.cfl.res.rr.com) |
| 19:58.43 | *** join/#asterisk pgarcia (n=chatzill@yoda.kanatek.com) |
| 19:59.40 | heliosj | Good afternoon. Quick question about 'wraptime' for queues. If you have two queues, and the same member in both queues, will the longer wrap time prevail regardless if the second queue has a shorter wraptime? |
| 20:01.01 | pgarcia | Hi, I have a weird need here. For testing purposes I need to put the FXO port offhook and start a dialplan, which will wait for digits. My problem is that the FXS side (another equipment) cannot generate the ring to the FXO port. I'm trying to start an outgoing call through this FXO port, but since I don't have any progress tones, the call drops after 30 secs. Anybody has an idea what I can do? |
| 20:02.17 | [TK]D-Fender | pgarcia: FXS devices don't generate ringing, they accept it. |
| 20:03.10 | Naikrovek | [TK]D-Fender: not sure how i fixed it but i did. thanks for help |
| 20:03.20 | Naikrovek | didnt change the passwords, just retyped them, and works. |
| 20:03.32 | pgarcia | FXS ports generates rings to FXS devices (or to FXO ports), right? |
| 20:04.33 | manxpower | ~fxofxs |
| 20:04.34 | infobot | fxofxs is, like, An FXO port (red Digium module) expects to receive dialtone and receive ring voltage. You can connect it to a PSTN line from the telco. An FXS port (green Digium module) expects to provide dialtone and provide ring voltage. You can connect a phone or a fax to it. |
| 20:06.05 | pgarcia | so, to get my FXO detecting the answer (outgoing call) I always have to have dialtone, etc...? |
| 20:06.20 | *** join/#asterisk bradf (n=bradf@citireg.com) |
| 20:07.13 | [TK]D-Fender | pgarcia: What are you trying to do? You sense of direction and depth of description as jumbled up |
| 20:07.27 | bradf | i have an asterisk/freepbx server, two linksys PAP2, connected to Polycom phones. apparently both of those lines have fuzzy quality - how can i troubleshoot that? |
| 20:07.56 | manxpower | pgarcia: you cann't have asterisk determine when the far end picks up for outgoing calls from an FXO port. |
| 20:08.54 | *** join/#asterisk vgster (n=vgster@94-194-190-189.zone8.bethere.co.uk) |
| 20:09.51 | *** join/#asterisk vgster (n=vgster@94-194-190-189.zone8.bethere.co.uk) |
| 20:10.23 | pgarcia | [TK]D-Fender, let me try to explain better: I have asterisk with a FXO port which normally is connected to a PSTN. However, I've connected it to another computer with a FXS card installed. I'd like to "originate" the call from the asterisk+fxo side but of course, the fxs acting as a PSTN. |
| 20:10.35 | *** part/#asterisk xheliox (n=jeff@178.199.8.67.cfl.res.rr.com) |
| 20:11.30 | manxpower | pgarcia: people do that all the time. nothing special about it |
| 20:11.47 | [TK]D-Fender | pgarcia: then you just dial out. |
| 20:11.49 | pgarcia | So, my question could be: Could I pickup my phone (asterisk+fxo) and wait for digits instead of dialout ? |
| 20:12.36 | pgarcia | I could dial out, but since I don't know if the other side answered, I cannot start a dialplan. Or can I? |
| 20:12.54 | [TK]D-Fender | pgarcia: Originate a call that dials without passing a number, and on the other end run an IVR. |
| 20:13.19 | manxpower | ~book |
| 20:13.20 | 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 |
| 20:13.24 | [TK]D-Fender | pgarcia: and you ARE asking for trouble since the side has no reason to know that you are calling |
| 20:13.24 | pgarcia | [TK]D-Fender: you mean, play something? |
| 20:13.31 | [TK]D-Fender | ~ivr |
| 20:13.32 | infobot | hmm... ivr is Interactive Voice Response |
| 20:13.57 | TJNII | infobot? Did jbot get renamed in my absence? |
| 20:14.11 | Qwell | jbot is dead. RIP jbot. |
| 20:15.49 | pgarcia | [TK]D-Fender, I tried dial without a number but after 30 secs, the call drops. Maybe I did something wrong... |
| 20:16.20 | s14ck | Can I do one alias for a voice mail? |
| 20:16.51 | s14ck | 2 sip peers with the seems voice mail |
| 20:16.56 | s14ck | how can I do it? |
| 20:17.07 | manxpower | s14ck: put the same mailbox= line for both peers |
| 20:17.12 | [TK]D-Fender | s14ck: You can specify whatever and however many boxes you want per peer |
| 20:18.18 | [TK]D-Fender | checkout time, BBL |
| 20:18.35 | bradf | anybody have some pointers on finding out why these two conference lines I have are fuzzy? |
| 20:19.43 | manxpower | bradf: what kind of conference lines. POTS, T-1 CAS, T-1 PRI, SIP, IAX2, MGCP, Skinny/SCCP? |
| 20:19.53 | drmessano | Genetics? |
| 20:20.04 | manxpower | remember fuzzy questions get fuzzy answers |
| 20:20.04 | bradf | maxpower: sip |
| 20:20.25 | bradf | manxpower: was less fuzzy the first time i asked it :-p |
| 20:20.51 | manxpower | Ah, I see why nobody answered |
| 20:21.20 | bradf | manxpower: i just guessed that maybe i asked it wrong, so i tried again with less detail |
| 20:21.35 | manxpower | you really asked on the wrong channel |
| 20:22.15 | manxpower | tough to determine if your problem is Asterisk or FreePBX. I guess nobody wants to go thru the work to find out. |
| 20:23.27 | bradf | darn, can you recommend any professional services company that might want to go through the work to find out? |
| 20:23.56 | drmessano | bradf: What manxpower is saying is, if anyone had the answer, they would have given it to you, but since you're running freepbx, and no one has the answer, here's some free passive aggressiveness. |
| 20:24.10 | drmessano | Thank you, drive through |
| 20:25.22 | bradf | drmessano: i appreciate your having spelled it out for me, and am a bit curious as to how you got your ability to read minds! i'd like to subscribe to your newsletter |
| 20:25.43 | *** join/#asterisk savant42 (n=savant42@c-76-102-97-60.hsd1.ca.comcast.net) |
| 20:26.02 | *** join/#asterisk |Cybex| (n=John@80.100.126.176) |
| 20:26.18 | drmessano | Hmmm.. he didn't stick around long enough for the URL :( |
| 20:26.40 | savant42 | greetings, #asterisk |
| 20:26.49 | drmessano | goes back to another FreePBX 2.6b1 install |
| 20:28.08 | *** join/#asterisk R0dya (n=dvorak@78.16.207.80) |
| 20:30.14 | *** join/#asterisk sjobeck (n=Adium@32.153.125.37) |
| 20:30.57 | *** part/#asterisk sjobeck (n=Adium@32.153.125.37) |
| 20:32.27 | *** part/#asterisk lasko (n=lasko@70.102.15.210) |
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| 20:38.28 | *** join/#asterisk mercutioviz (n=michaelc@freeswitch/developer/msc) |
| 20:38.30 | mercutioviz | ~book |
| 20:38.31 | 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 |
| 20:38.36 | *** join/#asterisk |Cybex| (n=John@80.100.126.176) |
| 20:42.23 | *** join/#asterisk |Cybex| (n=John@80.100.126.176) |
| 20:44.34 | rob0 | "ast_get_srv: SRV lookup for '_sip._UDP.proxy01.sipphone.com' ..." this debug output in console just started without me doing anything. Why? |
| 20:45.04 | rob0 | (every 01:45) |
| 20:47.26 | manxpower | rob0: srvlookup=off |
| 20:47.34 | *** join/#asterisk |Cybex| (n=John@80.100.126.176) |
| 20:47.53 | rob0 | yeah, but what I'm wondering, is why this just began out of the blue? |
| 20:48.16 | rob0 | I just did "sip set debug off", maybe will shush it? |
| 20:48.25 | *** join/#asterisk catpants (i=catpants@c-71-228-179-232.hsd1.al.comcast.net) |
| 20:48.36 | rob0 | nope |
| 20:48.52 | rob0 | maybe they just added a SRV record? |
| 20:52.02 | rob0 | srvlookup=off |
| 20:52.42 | rob0 | oh well, afk, bbl |
| 20:59.59 | *** join/#asterisk leifmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 20:59.59 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 21:06.59 | *** part/#asterisk mercutioviz (n=michaelc@freeswitch/developer/msc) |
| 21:08.46 | *** part/#asterisk fiXXXerMet (n=kjohnson@206.205.38.66) |
| 21:11.03 | *** join/#asterisk mrsci (n=mrsci@unaffiliated/mrsci) |
| 21:12.00 | Naikrovek | how do i tell asterisk to forward extensions that don't exist on that server to another * server that does know of the extensions |
| 21:13.34 | *** join/#asterisk scud (n=scud@69.73.19.194) |
| 21:14.18 | scud | one time at band camp, i put an oem switchvox in my pussy. |
| 21:15.21 | Naikrovek | ugh |
| 21:17.34 | *** join/#asterisk lanning (n=lanning@173.8.187.197) |
| 21:19.18 | *** mode/#asterisk [+b *!*n=scud@69.73.19.*] by Qwell |
| 21:19.18 | *** kick/#asterisk [scud!i=north@pdpc/sponsor/digium/Qwell] by Qwell (cool story bro) |
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| 21:23.30 | jgoo | How would it be possible: have several phones that show the incoming lines (4 isdn lines) and have the lines flash when ringing - what feature is this and which phones support it? |
| 21:24.17 | jgoo | Or do you have to have the lines forward to that extension (as I want to have two devices that do this feature, I could have a ring-group that rings both, and pickup on other devices, but I figured there might be another way) |
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| 21:30.02 | manxpower | jgoo: do you want to be able to pick up the lines on any phone? |
| 21:30.21 | manxpower | oh, ISDN. Nevermind. |
| 21:31.45 | jgoo | manxpower, does it really make a difference? |
| 21:32.04 | jgoo | also, may be isdn, but someone is saying that the line is ringing but asterisk isn't picking it up (openvox card) |
| 21:32.15 | jgoo | says happening infrequently, never happens when I test |
| 21:33.11 | jgoo | you know, this whole industry is a pile of shit, from what I've seen every card, version, front end, device or adapter has its own problems and unreliability... is anything working reliably? (not reliable... I know how to troubleshoot it when it goes wrong.. but like the old stuff) |
| 21:34.30 | bmoraca | Sangoma cards are great. openvox are piles of shit. |
| 21:34.43 | bmoraca | don't think Sangoma makes a BRI card, though. |
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| 21:35.18 | manxpower | jgoo: I assume you are using ISDN phones |
| 21:35.55 | jgoo | 'isdn phones' ? why would i be using 'isdn phones' am I not running asterisk? |
| 21:36.05 | jgoo | does asterisk have a magic isdn port card I don't know about? |
| 21:36.26 | jgoo | asterisk handles the isdn, everything else is sip... *checks channel* |
| 21:36.51 | jgoo | bmoraca, where can I have read about openvox being shit before? where is all this knowledge collected? |
| 21:37.03 | jgoo | certainly not that voip-wiki, that is a horredously poorly written pile of crap |
| 21:37.30 | generalhan | ... wow |
| 21:37.33 | KyleK | is there a POTS caller id method thats immediate? |
| 21:38.07 | bmoraca | KyleK: no. |
| 21:38.16 | bmoraca | not that i've heard of, anyway |
| 21:38.29 | jgoo | generalhan, don't wow, you know it is true - you search and you get the impression, either so few people are actually making real scale systems, or if they are they aren't writing about the problems they are having... |
| 21:38.50 | jgoo | how much are sangoma cards, and why are they great? |
| 21:38.52 | manxpower | jgoo: ISDB BRI or ISDB PRI? |
| 21:39.01 | jgoo | 4 line isnd |
| 21:39.09 | jgoo | BRI |
| 21:39.19 | generalhan | jgoo: have you made *any* contributions to the documentation ? |
| 21:39.41 | manxpower | The easiest thing to do is Dial(SIP/phone1&SIP/phone2&SIP/phone3) |
| 21:39.54 | jgoo | generalhan, not the way you should be looking at it, the whole thing has already lost if you want to reflect the blame onto me |
| 21:40.19 | jgoo | generalhan, don't be a gimp (the mindset of gimp developers) |
| 21:40.35 | generalhan | jgoo: not putting the blame on you. just saying, if you arent doing anything about it, who are you to knock the efforts of others |
| 21:40.49 | jgoo | generalhan, my complain about documentation and people sharing real world knowledge in some place isn't less valid because I haven't shared |
| 21:41.03 | heedly | sure it is |
| 21:41.08 | jgoo | generalhan, the 'who are you' is a completely random and unjustified point you are making |
| 21:41.21 | heedly | it means you have no idea what you talk about from a production stand point |
| 21:41.38 | jgoo | heedly, did that make sense? |
| 21:41.46 | heedly | how didn't it? |
| 21:41.52 | jgoo | because, it didn't to me - what are you trying to say? |
| 21:42.02 | jgoo | that I can / can't that it is a justified point, or isn't? |
| 21:42.04 | heedly | you don't make docs |
| 21:42.20 | heedly | how are you a position to criticiz others for not making them |
| 21:42.28 | heedly | unless you pay them or something |
| 21:42.30 | jgoo | right, and why would that make me less entitled to understand the docs that are there? |
| 21:43.02 | jgoo | heedly, there is no entitlement to judge, I am not criticising people, your small minded logic is tiring. You don't earn the right to say whether docs are good or bad, they just are good or bad. |
| 21:43.17 | heedly | < jgoo> generalhan, my complain about documentation and people sharing real world knowledge in some place isn't less valid because I haven't shared |
| 21:43.34 | heedly | my statement was a direct response to that |
| 21:43.48 | generalhan | jgoo: its a matter of opinion. i dont think the docs are "bad". but if you do, and you think that things need to be changed, do something about it. |
| 21:43.50 | jgoo | right heedly, and how, and I beg of you to make a complete and well rounded fucking argument, and not just point to it like you just have, how does that illustratye yout point? |
| 21:44.10 | heedly | logic |
| 21:44.16 | Qwell | jgoo: enough |
| 21:44.17 | heedly | 1 statement = 1 more statement |
| 21:44.22 | heedly | etc.. etc.. |
| 21:44.22 | jgoo | generalhan, depends, what docs are you talking about? I am talking about specific ones I've seen - I read a lot, I guess i missed some parts that gave 'current best practices' |
| 21:44.29 | jgoo | and analogues to real world scenarios |
| 21:44.33 | heedly | until someone makes an ass of them selves |
| 21:44.37 | heedly | is the way of IRC |
| 21:44.38 | heedly | duh |
| 21:44.39 | jgoo | heedly, go away |
| 21:44.44 | heedly | kk |
| 21:45.18 | heedly | seriously, the docs are fine for me. |
| 21:45.43 | heedly | as far as "best practices" they tend to be good business plans and are not shared much |
| 21:45.45 | jgoo | Qwell, don't enough me, this is a perfectly valid discussion about places we can store docs - if you want to add something constructive and not be confrontational I invite you to do that. heedly, I appreciate what you are saying (wether you contrib. or not!) ;-) what docs are you reading? |
| 21:45.53 | jgoo | heedly, you are right there |
| 21:46.20 | KyleK | damn i guess it'll take two rings to get callerid going through ata->asterisk->ata |
| 21:46.24 | jgoo | but, at least something like 'product x sucks' would be a good start -- where can I get an overall feeling about that, without polling everyone live in irc :p |
| 21:46.57 | heedly | I found everthing I needed here: http://www.voip-info.org/wiki/view/Asterisk |
| 21:47.39 | jgoo | heedly, bearing in mind, the asterisk part is fully grokked... I have no problems with the software... just the hardware / what works / why sometimes the line rings, but asterisk doesn't know about it... |
| 21:47.52 | jgoo | stuff that basically negates any worth in a system, even if it fails 1 in 200 calls |
| 21:47.53 | Qwell | Wikis are editable by everybody. Including device manufacturers. If you want to know what products are good, you need direct user experience. |
| 21:48.18 | jgoo | Hrm, would be nice to know something... it is too much of a minefield... |
| 21:48.26 | heedly | jqoo: propable in the Hardware part |
| 21:48.31 | jgoo | two openvox cards, 600 euros |
| 21:48.32 | heedly | also what Qwell said |
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| 21:48.57 | jgoo | one is fine, one seems to miss calls, not sure why, I'll debug it tomorrow... perhaps isdn signal needs some adjustments from telco |
| 21:49.15 | Qwell | YGWYPF |
| 21:49.28 | jgoo | Qwell, how much are better cards? |
| 21:49.38 | jgoo | how much should you pay for isdn cards? |
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| 22:31.08 | jgoo | Ok, point - openvox bri cards have 3 possible drivers. No effort is made to actually give any form of guidance as to which one to use. What. I'll try all three? maybe troubleshoot and solve 4-5 critical issues in each one, and then decide? I am using misdn right now, but I've used bristuff on other isdn cards, but not misdn V2... |
| 22:31.16 | *** join/#asterisk LiNeTuX (n=LiNeTuX@119.153.205.68.cfl.res.rr.com) |
| 22:31.19 | jgoo | anyone else have some real information on which is the better to use? |
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| 22:32.10 | Katty | hello. |
| 22:33.39 | jgoo | So, back to my original question - how can I see the activity on the trunk, from a sip device itself - what feature is that called? |
| 22:34.10 | jgoo | Right now, the 4 lines light up when sip calls are placed to the device, but what if they are ringing elsewhere? |
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| 22:48.13 | citywok | has anybody built an xml voicemail applicaiton like the aastra one for trixbox? i'm modifying their trixbox app right now to work with standalone asterisk, and have it working now that it plays the audio file, but when it calls the phone to play it, it resets the screen so it isnt in the voicemail app anymore, not very helpful |
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| 22:57.32 | hardwire | anybody have a huge list of international test numbers? |
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| 23:01.24 | jgoo | So, how can I have the activity of external lines show up on a device (bonus, you get to chose from any voip device in the world) |
| 23:01.26 | voipmonk | hardwire - google the hotel contact country code |
| 23:02.00 | *** join/#asterisk vcampos (n=campos@200-158-26-119.dsl.telesp.net.br) |
| 23:02.02 | hardwire | I'm already headed that way :) |
| 23:02.09 | hardwire | python-mechanize to the rescue. |
| 23:02.11 | voipmonk | hola? |
| 23:04.13 | jgoo | voipmonk, this must be a common thing - when a line is ringing, to show it on a device |
| 23:04.50 | voipmonk | u dont like seeing that the line is ringing on your device? |
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| 23:05.02 | voipmonk | :) |
| 23:05.04 | jgoo | to show it |
| 23:05.16 | jgoo | not - to not show it |
| 23:05.27 | *** join/#asterisk SaiSoma (n=SaiSoma@adsl-074-167-136-030.sip.mob.bellsouth.net) |
| 23:05.37 | voipmonk | 's face turns upside down and morphs into a big glowing question mark |
| 23:06.03 | voipmonk | what device? |
| 23:06.04 | jgoo | 4isdn -pbx --- phones... several of which would like to see what lines (channels) are in use and to pickup any line |
| 23:06.18 | voipmonk | what kinda phones? |
| 23:06.20 | jgoo | what device - exactly, that is my challenge, show me any voip device that can do it |
| 23:06.24 | jgoo | any |
| 23:06.25 | jgoo | any |
| 23:06.33 | jgoo | ffs any, I am googling this bitch for an hour |
| 23:06.39 | voipmonk | so u dont have the phones and are looking for phones with BLA ? |
| 23:06.42 | jgoo | every single old pbx I've come across has and uses this feature |
| 23:06.51 | bmoraca | jgoo: asterisk isn't a key system...why are you trying to turn it into one? |
| 23:06.54 | jgoo | I can't find any .. analogue...ahem... in the digital |
| 23:07.02 | jgoo | define key system |
| 23:07.22 | voipmonk | now if we could get you to arrange your words like yoda... |
| 23:07.22 | jgoo | voipmonk, I hae one with bla, but that isn't what I mean |
| 23:07.25 | jgoo | that is for extensions |
| 23:07.33 | voipmonk | its like talking to the old batman :) |
| 23:07.42 | voipmonk | cant get ....... to my utility belt |
| 23:07.51 | voipmonk | just messin with ya |
| 23:07.51 | jgoo | yeah? interesting, what qualities of my prose affords you that opinion? |
| 23:07.58 | jgoo | aaah, ellipses |
| 23:08.03 | jgoo | yeah :p I do that |
| 23:08.06 | bmoraca | a key system is a system where each of the lines shows up as a KEY on the phone. PBX > key system. and, if you can't define it yourself, you shouldn't be here. |
| 23:08.07 | voipmonk | hehe |
| 23:08.14 | voipmonk | ~botsnack |
| 23:08.14 | infobot | :), voipmonk |
| 23:08.36 | jgoo | bmoraca, stop snobbing, I've read about key system... is pbx > key system - why can't it solve the real world need? |
| 23:08.45 | voipmonk | http://www.voip-info.org/wiki/view/Asterisk+SLA |
| 23:09.00 | bmoraca | because you're trying to make asterisk do something it's not supposed to do. if you want a key system, go buy a key system. |
| 23:09.32 | leifmadsen | jgoo: you want to use SLA (Shared Line Appearance) |
| 23:09.32 | jgoo | so !(PBX > key system) ; PBX != key system |
| 23:09.46 | leifmadsen | as voipmonk pointed out |
| 23:09.58 | jgoo | aaah, I thought SLA service level agreement, thanks leifmadsen |
| 23:10.01 | jgoo | and voipmonk |
| 23:10.03 | voipmonk | well.... Asterisk is a toolkit - u can make it do what it do or what you want it to do , may the source be with you :) ur gonna spend ca$h if cash = time |
| 23:10.05 | jgoo | (and I am batman) |
| 23:10.26 | voipmonk | oh hell... |
| 23:10.37 | bmoraca | no, a PBX is much more flexible than a key system. there's a reason that key systems aren't used anymore in large-scale installations. |
| 23:10.41 | leifmadsen | jgoo: no you're not -- The_Boy_Wonder is batman |
| 23:10.50 | leifmadsen | ~The_Boy_Wonder |
| 23:10.58 | voipmonk | ~nice_try |
| 23:11.00 | leifmadsen | infobot: The_Boy_Wonder is batman |
| 23:11.01 | infobot | okay, leifmadsen |
| 23:11.06 | voipmonk | lol |
| 23:11.09 | bmoraca | asterisk's implementation of SLA is a hack and doesn't work all that well. but, you're welcome to try. make sure you as rue_mohr about it. he can tell you all about it. |
| 23:11.20 | leifmadsen | ~The_Boy_Wonder |
| 23:11.21 | infobot | the_boy_wonder is, like, batman |
| 23:11.25 | jgoo | bmoraca, but you are insisting that asterisk can't do this... and if it can't ... then in all situation where you need this feature in addition to ... you know.. phones.. then it is missing a feature |
| 23:11.36 | jgoo | no matter what the old tech used to be called |
| 23:11.41 | jgoo | I can do core show channels |
| 23:11.51 | jgoo | I just what that on the phone, as blinkenlights |
| 23:11.53 | bmoraca | jgoo: the POINT of a PBX is that you don't need physical line buttons. |
| 23:12.12 | jgoo | bmoraca, I know that! (and I am glad you pointed out that was the point you were making now) |
| 23:12.17 | jgoo | but the customer doesn't. |
| 23:12.27 | bmoraca | then the customer needs a key system, not a pbx. |
| 23:12.40 | jgoo | I tried to explain: when more than one line is ringing, how do you chose which one you take? *silence*, right, so just press 8... its the same... |
| 23:12.49 | jgoo | no, they need a pbx, but they like seeing the lights |
| 23:13.46 | bmoraca | uhm...there's no such things as "lines" in voip. you can have call appearances on your phones, sure. if you can't figure out how to answer more than one call on a voip phone, you shouldn't be using them. |
| 23:14.15 | bmoraca | and if your question is regarding distinguishing between different inbound lines, you can do that easily by modifying the callerid information before the call is sent to the phones. |
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| 23:15.12 | The_Boy_Wonder | BATMAN! |
| 23:15.20 | jgoo | bmoraca, these people are just used to this way - I guess there is one thing though, they want to pickup multiple calls, say 'one moment' and then go back to the first call - you need a 4 line device to handle that right? |
| 23:15.38 | bmoraca | no... |
| 23:15.38 | TJNII | jgoo: You're looking for hints, iirc |
| 23:15.56 | bmoraca | TJNII: yes, hints are part of asterisk's SLA implementation |
| 23:16.39 | TJNII | Maybe I'm not sure what he's asking, I'm going between skimming the channel and resroting a '70s trimline.... |
| 23:16.54 | TJNII | s/resroting/restoring |
| 23:17.11 | jgoo | reading up on SLA - hints sounds interesting, thanks TJNII |
| 23:17.12 | bmoraca | jgoo: i use Cisco 7940 and Polycom 330 phones all the time...they only have two "call appearance" buttons (you could potentially equate these to "line" buttons if you wanted to). however, i can configure each of those call appearances to have multiple calls come in on them, pretty much as many as i want. |
| 23:17.32 | leifmadsen | The_Boy_Wonder: na na na na na na na na, na na na na na na na na |
| 23:17.57 | bmoraca | and by prefixing the callerid name, I can tell which phone number was dialed, which queue the call came from, etc. |
| 23:18.44 | TJNII | needs a ATA that works with rotary phones.... |
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| 23:19.09 | The_Boy_Wonder | leifmadsen: I counted the na's just to make sure it was correct |
| 23:19.36 | generalhan | haha |
| 23:20.22 | laggo | how can some providers offer outgoing calls without registration - isn't this tantamount to offering free pstn calls? how do they know what user's account to debit? |
| 23:21.08 | jgoo | bmoraca, I see - that was the cognitive gap - how to 'answer' the real ringing lines... to make them become flashing lines on that device... I guess I need to add that device to a ring group... and set it to a mute ring tone? |
| 23:21.38 | leifmadsen | The_Boy_Wonder: was it? :) |
| 23:21.40 | bmoraca | jgoo: no... |
| 23:21.43 | jgoo | bmoraca, the old setup - a phone would ring, and they would see this ringing (like a busy lamp, but on the line) and pick it up... |
| 23:22.15 | leifmadsen | The_Boy_Wonder: I think I counted right :) each of the na's lands on an 8th I'm pretty sure (in 4/4) |
| 23:22.48 | *** part/#asterisk savant42 (n=savant42@c-76-102-97-60.hsd1.ca.comcast.net) |
| 23:23.07 | bmoraca | jgoo: your cognitive gap is that you don't realize the difference between a PBX and what you're trying to do. you don't map trunk lines to phones in a PBX. you can somewhat simulate it in asterisk with hints (emulates SLA), but it really doesn't work very well. |
| 23:23.09 | The_Boy_Wonder | leifmadsen, well, sort of, i would have started each measure with a Da |
| 23:25.27 | leifmadsen | The_Boy_Wonder: ah yes, you are correct sire |
| 23:26.18 | jgoo | bmoraca, not map them, they just want to see calls coming in, and keep answering them - with one touch |
| 23:26.52 | jgoo | bmoraca, you are sat there. phone rings. you don't want a loud device, so you want a flashing light. I don't care how it is flashing, this is the user, they are right, they want a flashing light (and a ringer, in the other room) |
| 23:27.00 | Nivex | wishes he had a PBX at work |
| 23:27.15 | jgoo | they want to pickup and press that light to answer (because they might pickup to make another call, and ignore it) |
| 23:27.43 | bmoraca | jgoo: ....... you're not listening. i don't help people who don't listen. |
| 23:28.23 | Naikrovek | anyone know of a good example of the switch statement in extensions.conf. I have two sites and I need to route incoming calls from one to the other, but right now if a caller enters an extension for site B while calling site A, they get an error that no such extension exists |
| 23:29.23 | Naikrovek | carved the Bat Signal into his pubes once when he was drunk. |
| 23:29.33 | jgoo | bmoraca, I am listening, I understand what you are saying, and I am trying to tell you, I can't figure out how to configure it to be like this |
| 23:30.11 | jgoo | I know the lines are virtual in the pbx, but, the callers aren't - they want... a visual queue... a bank that shows their calls on hold, and the ability to answer them in one touch |
| 23:30.23 | Naikrovek | oh, to be young again, and also an idiot |
| 23:30.35 | jgoo | because - that is how it works now... they see a light, they press, they talk, they see another, press, hi, please hold, press, back to original call |
| 23:30.47 | jgoo | bmoraca, is that unreasonable what I am saying? |
| 23:37.21 | Naikrovek | jgoo: you can stack these onto certain Polycom phones: http://www.cngadget.com/images//20090521/d36a842bdca74df9.jpg |
| 23:38.21 | Naikrovek | They can show you who's on the phone, but I don't think they show what line/trunk they're using. |
| 23:38.42 | Naikrovek | jgoo: the polycom phones I am using at my employer are two-line phones, and they're two separate lines. you can place a caller on hold and answer the other "line" if you like. |
| 23:38.47 | drmessano | They can show you whatever you want them to |
| 23:39.04 | drmessano | You CAN dumb down Asterisk to key system level |
| 23:39.20 | jgoo | Naikrovek, yes, that sounds good, the GXP2000 has 4 lines, that is what I am thinking - I'd just need to have the incoming call ring that phone... but with a mute ringtone... that would work |
| 23:40.28 | *** join/#asterisk Naikrovek (n=jeremiah@98.214.112.102) |
| 23:41.35 | drmessano | How many phones? |
| 23:41.39 | drmessano | and how many lines? |
| 23:42.12 | drmessano | If youre going to use Asterisk like some 4 line phones with POTS lines on them, why not cut out the middle man and get soem 4 line phones from Wal Mart? |
| 23:42.46 | TJNII | hehe. Just use the ethernet drop as 4 POTS lines.... |
| 23:43.55 | drmessano | jgoo: Knock knock? |
| 23:44.27 | Qwell | jgoo: remember what I said earlier about getting what you pay for? |
| 23:44.32 | Qwell | ~phones |
| 23:44.33 | infobot | While personal preference will dictate which phone works best for you, general consensus on a rough order of quality and suggestibility is as follows: Polycom (any), Aastra 480i, Aastra 5i Series, Cisco 7940+, Linksys SPA-9XX, Snom, and finally everything else. Do not consider Grandstream phones. Ever. |
| 23:45.36 | drmessano | Selective hearing? |
| 23:48.13 | jgoo | drmessano, just 4 isdn lines (I was reading about SLAs) |
| 23:48.47 | jgoo | I've used polycom in the US, they are good (some power issues on a few) but yeah, good |
| 23:49.00 | Qwell | jgoo: The last sentence. |
| 23:49.09 | KyleK | drmessano: walmart has 4 line phones? |
| 23:49.19 | Qwell | KyleK: sure |
| 23:49.25 | jgoo | I know. grandstream suck. I didn't know this info was only available on a bot on irc... nowhere else |
| 23:49.42 | Qwell | yet you're still considering them |
| 23:49.58 | KyleK | jgoo: lawyers dont go on irc yet ;) |
| 23:50.12 | KyleK | oh nose someone said a product sucks SUE SUE SUE |
| 23:50.18 | jgoo | drmessano, no, this is a voip system with some other features, but two desks want to be able to pickup any incoming call with one push of a button, keep that call, and then pickup another call, fast, one click. it seems they have to be in the ring group (but not have a ringtone set for that type of call) |
| 23:50.31 | drmessano | http://www.grandstreamsucks.com/ |
| 23:50.32 | jgoo | Qwell, no, I bought some a while back, I know they suck |
| 23:50.38 | Qwell | yet you're still considering them |
| 23:51.52 | drmessano | Grandstream sucks so bad, someone dropped a 10 spot (25% of the retail cost of a GS phone!) down to register that domain and put that page up! |
| 23:52.21 | jgoo | Qwell, no I am not considering them, I am saying, I know that the gxp2000 had 4 lines, and they light up when they get a call (which means the incoming has to ring that device) |
| 23:52.23 | TJNII | Someone paid $80 a piece for Grandstreams? They got ripped off hard. The only reason I like Grandstreams is because they cost $20. I would _never_ install them in a office, but in an area with rare use, for $20 ... okay. |
| 23:52.46 | jgoo | Just backing up my thinking over how this one click answer / hold / swap will work, for upto 4 incoming lines (4 channels is enough I think) |
| 23:53.00 | drmessano | $80? |
| 23:53.05 | jgoo | TJNII, $20? what model? what site? |
| 23:53.15 | TJNII | jgoo: ebay (buyer beware) |
| 23:53.23 | jgoo | right, ok :-p |
| 23:53.32 | TJNII | drmessano: That's roughly what the math comes out to on that site you posted. |
| 23:53.49 | drmessano | Why dont you jsut get a grandstream PBX |
| 23:54.53 | jgoo | drmessano, ? I have an asterisk box, all working (several installs, I've written a lot of database linked apps for them) |
| 23:55.06 | drmessano | Asterisk will do what you want it to do, but it will be like putting a 4 yr old boy in a pink dress |
| 23:55.11 | jgoo | but, how to get the lights blinking just the way these people insist they want them, needed a bit of thinking |
| 23:55.13 | Kobaz | i have a queue, that's a ringall.. if someone was on the phone while the queue was hit, they don't get the call... and 60 seconds later they will get the call... no matter what timeout settings i use |
| 23:55.29 | drmessano | jgoo: If youre so skilled with Asterisk, this should be a no-brainer |
| 23:55.43 | jgoo | drmessano, I am not sure you understand what I am trying to do, or accept that this is a legitimate 'thing' that these people want the phones to do... |
| 23:55.50 | jgoo | drmessano, stop being confrontational |
| 23:56.16 | TJNII | This is the internet. You're supposed to be confrontational. |
| 23:56.21 | jgoo | I am trying to find which is the best solution, bar adding a 4 line sip client to a ring group on all incoming, with no ringtone, so the lights flash, but it rings for internal calls |
| 23:56.28 | Kobaz | with rrandom, if someone rejects the call... it will come back to their phone immediatly if there is no one else in thw queue |
| 23:56.36 | Kobaz | i can't get the same behavoir for the ringall queue |
| 23:56.40 | drmessano | jgoo: I know exactly what youre trying to do.. its called BLF, and its basically making Asterisk and 4 phones that have suddenly become overpriced for the task, mimic that of a bunch of wal mart multiline phones |
| 23:57.10 | jgoo | drmessano, you seem to think you know better than the people who sit in the office, and both yourself and bmoraca seem to think the users care what you call little bits of tech, they don't they want - quiet phone that flashes, and they can ***optionallY**** answer the call |
| 23:57.43 | jgoo | drmessano, so what feature makes asterisk systems not overpriced? |
| 23:58.14 | jgoo | drmessano, come on, what feature is every other asterisk installation used that makes them not overpriced. which one is it. tell me. I'll be sure to use it, so I know this isn't over priced. |
| 23:58.18 | drmessano | I know _exactly_ what they want.. They want little blinky line appearances |
| 23:58.29 | drmessano | Are you able to read? |
| 23:58.33 | jgoo | drmessano, should I spell it out for you exactly, and then you give me your honest opinion |
| 23:58.40 | jgoo | as to wether you were right or wrong? |
| 23:58.48 | drmessano | Who is being confrontational now? |
| 23:59.03 | jgoo | drmessano, don't ask me if I am able to read, it does nothing to move the debate forward, if I've missed apoint, reitterate it |
| 23:59.14 | drmessano | I have, 3 times.. |
| 23:59.22 | drmessano | [19:57] <drmessano> jgoo: I know exactly what youre trying to do.. its called BLF, and its basically making Asterisk and 4 phones that have suddenly become overpriced for the task, mimic that of a bunch of wal mart multiline phones |
| 23:59.24 | jgoo | asking you to decide if you are right or wrong isn't confrontational, I'll try and be terse and outline the requirements |
| 23:59.46 | jgoo | drmessano, to which I specifically replied to you, didn't you read what I said? |
