| 00:01.19 | drmessano | No really, if they took the cast of Stand By Me and put them in the Asterisk move, all of you would be in awe of just how skilled casting directors are.. Sure, the choice of Wil Wheaton to play Russell is a little bit of a stretch, but you'll see he can pull it off well |
| 00:01.25 | drmessano | movie* |
| 00:02.21 | drmessano | http://i.cnn.net/v5cache/TCM/Images/Dynamic/i29/StandByMe_WP_1024x768_012720051535.jpg <-- Early morning dev meeting? You decide. |
| 00:08.23 | drmessano | What sort of 24/7 support response is available for Asterisk from Digium? |
| 00:09.13 | jaytee | zip, nada, nuthin |
| 00:09.26 | jeev | is excite!!! yes, excite!!! |
| 00:09.38 | jaytee | excite or excited? |
| 00:09.48 | jeev | EXCITE |
| 00:09.50 | jeev | like borat |
| 00:09.55 | jaytee | ok |
| 00:10.29 | jeev | my failover script rules |
| 00:10.31 | drmessano | Oh really? |
| 00:10.31 | jeev | openvpn is working great |
| 00:10.35 | jaytee | cool |
| 00:10.37 | jeev | asterisk redundancy is working perfect |
| 00:10.42 | jeev | my tmobile and att wifi hotspots both work |
| 00:10.56 | jaytee | did you teach yourself to script or take some classes? |
| 00:10.57 | jeev | only thing that's left is mccain and palin admit that they're scum of the earth |
| 00:11.02 | jeev | and economy goes back up |
| 00:11.05 | jeev | teached self |
| 00:11.10 | jeev | part of that script i stole |
| 00:11.12 | drmessano | jaytee: He bought a coder for a month |
| 00:11.12 | jeev | but the functions i did |
| 00:11.22 | jeev | scripting is fun but my main issue is that i always doubt myself |
| 00:11.26 | jaytee | first one's never gonna happen, second one might with a minor miracle |
| 00:11.28 | drmessano | jaytee: Like Richard Pryor in "The Toy" |
| 00:11.58 | jaytee | actually he scripts very good. better than I can |
| 00:12.03 | drmessano | Thats his "toy" |
| 00:12.21 | jeev | jaytee |
| 00:12.24 | Carlos_PHX | Wonders if any politician has ever admitted he's a lying weasel. |
| 00:12.26 | jeev | you must be talking to my arch nemesis |
| 00:12.30 | drmessano | He jabs him through the cage and makes him write perl for onions |
| 00:12.31 | jeev | on my ignore lits |
| 00:12.48 | jaytee | Carlos_PHX, yes there was one back in the 60's but he was on acid at the time and he didn't get reelected |
| 00:14.05 | drmessano | jaytee: Just started a project today to replace $65000 worth of pbx hardware in a mobile operations center with Asterisk.. for about 1/10th the cost |
| 00:14.07 | jaytee | in the words of Rodney King, "Can't we all just get along?" BIFF, SOCK, POWEE, THUD! BANG! "Guess not then" |
| 00:14.34 | Carlos_PHX | Wonders if we will ever again elect any politician who isn't a lying weasel. |
| 00:14.35 | drmessano | Its AMAZING what people will pay for hardware |
| 00:14.37 | drmessano | and services |
| 00:14.57 | drmessano | $7500 for a cell phone module for a PBX |
| 00:15.02 | Carlos_PHX | WTF |
| 00:15.21 | jaytee | drmessano, and I'm sure they'll be singing the praises of whatever upper management asshole that manages to steal all the credit for your efforts for at least a week over that one. |
| 00:15.28 | drmessano | lol |
| 00:15.29 | Carlos_PHX | So I was talking to Sprint PCS the other day about doing some integration between customer's cell phones and our service. |
| 00:15.29 | drmessano | No |
| 00:15.39 | drmessano | Its a volunteer project for emergency managment |
| 00:15.40 | Carlos_PHX | $30k for a 50-user box to do something magic. |
| 00:16.04 | drmessano | Carlos_PHX: Its INSANE what they charge |
| 00:16.24 | Carlos_PHX | I ask: So just tell me what signalling this sends you, and I'll duplicate it in Asterisk. |
| 00:16.41 | Carlos_PHX | Them: Oh now, this box uses a special protocol, you can't do it without that. |
| 00:16.54 | *** join/#asterisk osiris (n=osiris@c-71-205-27-88.hsd1.mi.comcast.net) |
| 00:16.54 | Carlos_PHX | Me: Tell me what you see on your end of the PRI and I'll duplicate it. |
| 00:17.04 | drmessano | They had 4 analog phone modules in the truck to make it flexible, at $7500 each.. guess where those went earlier in the year? |
| 00:17.05 | Carlos_PHX | Them: It's special signalling, you can't do it. |
| 00:18.19 | jaytee | Carlos_PHX, sounds like the same bullshit about getting callerid to analog phones on Nortel, "Oh, well you need a CLASS modem card in the PBX and the associated licensing per phone" |
| 00:18.42 | Carlos_PHX | It's definitely BS, because it's not a dedicated connection. |
| 00:18.49 | Carlos_PHX | It's either ISDN or SS7 signalling. |
| 00:18.53 | drmessano | How about them charging $32 a month (DISCOUNTED) for unlimited incoming and 200 outgoing minutes.. per line |
| 00:18.58 | Carlos_PHX | "Just tell me WTF to send you" |
| 00:19.08 | Carlos_PHX | Oh, and their "engineer" asked me what SS7 is. |
| 00:19.12 | drmessano | HA |
| 00:19.23 | LoRez | that's scary |
| 00:19.38 | *** join/#asterisk seanmh (n=seanmh@c-68-35-21-64.hsd1.nm.comcast.net) |
| 00:19.44 | Carlos_PHX | It was the most wasted hour of my life. |
| 00:20.35 | drmessano | So they were paying $256 a month service for 8 cell phone modules with unlimited incoming and 1600 minutes outgoing.... when the in/out ratio of an average incident is going to be 80% in/20% out |
| 00:20.45 | drmessano | So those 1600 minutes would be gone in less than a day |
| 00:20.52 | jaytee | Nortel sells you Callpilot voicemail hardware and software. The hardware is a Win2K3 embedded server on a card in the PBX. Has a nice 120GB HD. You get so much storage to start and more than half the drive capacity is unused. You want more voicemail storage? gotta buy the license keys for the mailboxes and another fee to "unlock" the storage for the drive you already paid for. What a racket! |
| 00:21.11 | jeev | i wish it were easy to use an existing nortel system |
| 00:21.14 | jeev | and hook it up to ITSP |
| 00:22.45 | jaytee | Nortel Norstar or Meridian? |
| 00:24.56 | Carlos_PHX | The legacy vendors are soooo screwed when the masses figure out the alternatives. |
| 00:25.27 | Carlos_PHX | Nortel ... Yeah, it's a VoIP system. Um no, you can't connect it to any other VoIP system. |
| 00:25.51 | tzanger | jeev: part of my hobby is turning those systems inside out |
| 00:26.23 | Carlos_PHX | In a week or two I get to take a Tadiran out to the desert for target practice. |
| 00:27.09 | drmessano | We had our discussion today about setting up VoIP for the Mobile Operations Center and the fire dept.. and I was asked if the CCM the county uses could be used in the mix |
| 00:27.16 | drmessano | Oh yeah, sure |
| 00:27.24 | jaytee | if your talking about the BCS 1000 that's VOIP but you can get a standard SIP compliant gateway card for it and you can also use a T1 card and interface to * that way. Same goes for more recent Meridian PBXs like my Option 11C. |
| 00:27.35 | drmessano | When I think Cisco, i think unified |
| 00:27.46 | drmessano | All for one, one for all, everyone give me a dollar |
| 00:27.53 | jaytee | lol |
| 00:27.57 | tzanger | if you need a right now solution, rip out the chassis and wire all the extensions up to Citel gateways |
| 00:29.44 | Carlos_PHX | Cisco is like that bum downtown that won't take a "no" when he asks for a dollar or five thousand. |
| 00:31.53 | jaytee | "Give me 5 bucks!" "No!" "Damn it, give me some money! I'm a NAM vet!" "What? you couldn't be more than 25! GTFO!" |
| 00:33.31 | kerx | weird |
| 00:33.38 | kerx | even now with AGI, my CDR's billsec is set to 0 |
| 00:33.53 | kerx | so it's not just the .call file's that do this, it's also the AGI |
| 00:33.59 | kerx | i wonder if it's something wrong on my end ? |
| 00:34.18 | Carlos_PHX | Holy sh*t, I just got a mass-mailing from Tech Data with all 2671 people in the clear in TO: |
| 00:34.27 | ManxPower | kerx: did you reply to my two questions? |
| 00:34.30 | Carlos_PHX | Mmmm...spam to my company mail account. |
| 00:34.46 | X-Rob | Carlos_PHX, hit 'reply all' and send 'Tech Data sucks. Don't buy stuff from them, as they've just released your email address to spammers - like me!' |
| 00:34.47 | kerx | ManxPower, Yes sir, I have them placed in my cdr.conf |
| 00:35.14 | ManxPower | My questions? |
| 00:35.41 | kerx | You asked if I had those two variables set in my configuration, and I've placed them in, but it still does the same exact thing. |
| 00:35.45 | X-Rob | <ManxPower> kerx: 1) do you have callprogress=yes or busydetect=yes? |
| 00:35.45 | X-Rob | <ManxPower> kerx: 2) what type of interface are you dialing out of? analog, T-1, SIP, IAX2? |
| 00:35.54 | kerx | I'm calling out using SIP |
| 00:35.54 | ManxPower | X-Rob: thanks. |
| 00:36.05 | X-Rob | ManxPower, np 8) |
| 00:36.17 | Carlos_PHX | X-Rob: I pretty much said nearly that. |
| 00:36.17 | kerx | X-Rob, Thanks also :) |
| 00:36.17 | ManxPower | so callprogress and busydetect don't apply to SIP. |
| 00:36.28 | ManxPower | kerx: Does this happen with multiple providers? |
| 00:36.36 | kerx | "","","s","gafachi-incoming","","SIP/GAFACHI-090d2218","","AGI","/opt/asterisk/scripts/custom/testagi.agi","2008-10-27 00:32:47",,"2008-10-27 00:33:10",23,0,"NO ANSWER","DOCUMENTATION","1225067567.5","18183453041-m1d" |
| 00:36.39 | kerx | Here is how the record looks like |
| 00:36.58 | kerx | The minute I pick up the call, that is placed in the Master.csv, and I'm continuously still on the phone. |
| 00:37.00 | ManxPower | kerx: do you answer in your dialplan or run something that auto-answers? |
| 00:37.36 | kerx | I have in extensions.conf to SetCDRUserField, then Answer(), then run the AGI() |
| 00:37.46 | kerx | I use a .call file to cp and paste it into my /var/spool/asterisk/outgoing/ directory |
| 00:37.59 | ManxPower | drmessano: If Digium would just run pre-release versions of Asterisk on their corporate PBX, many of these bugs would have been found before release. |
| 00:38.06 | kerx | Before, I was not using the AGI and I was doing a Background() and then a WaitExten() |
| 00:38.11 | *** join/#asterisk freakazoid0223 (n=mattc@pool-71-242-211-196.phlapa.east.verizon.net) |
| 00:38.21 | ManxPower | kerx: I'll keep that in mind as I start to use 1.6 |
| 00:38.40 | ManxPower | Background normally answers the channel |
| 00:38.42 | kerx | I've switched over to 1.4.22 now also, and it does the same thing, so it's not just 1.6.0.1 |
| 00:39.00 | ManxPower | kerx: I have never heard of that problme. |
| 00:39.44 | kerx | Yeah, it seems like it exists now, for the CDR's, the billsec is always 0, and it always states "NO DOCUMENTATION" |
| 00:39.51 | kerx | err, * "NO ANSWER" |
| 00:40.07 | ManxPower | kerx: You CANNOT be the only one with this problem. |
| 00:40.32 | kerx | what happens if you copy a .call file in ur /var/spool/asterisk/outgoing using a SIP, do you have correct CDR fields for billsec and disposition? |
| 00:40.55 | ManxPower | kerx: I guess I could ssh into one of my servers. |
| 00:41.06 | *** join/#asterisk mog (n=mog@c-68-62-219-86.hsd1.al.comcast.net) |
| 00:41.06 | *** mode/#asterisk [+o mog] by ChanServ |
| 00:41.07 | kerx | If you dare :P |
| 00:41.15 | kerx | It would be really appreciated. |
| 00:41.28 | kerx | s/It/I |
| 00:41.31 | kerx | err! |
| 00:41.37 | kerx | damn, I can't speak today |
| 00:41.41 | kerx | I would really appreciate it :) |
| 00:43.10 | ManxPower | "","9852463509","prompt","vm-notify","""Voicemail Notify"" <9852463509>","Local/check@vm-notify-552c,1","","Wait","1","2008-10-26 16:58:31",,"2008-10-26 16:58:31",0,0,"ANSWERED","DOCUMENTATION" |
| 00:44.37 | ManxPower | Of course my .call files just chan_local to handle everything in the dialplan. |
| 00:45.59 | kerx | I see, now it seems that might be a good try for me |
| 00:46.17 | kerx | Can I use the .call file to send it to Local then have in extensions.conf that handle the SIP connection ? |
| 00:46.38 | ManxPower | I use it to check if the user still has new messages in their mailbox right before it dials |
| 00:47.00 | ManxPower | actually sip.conf and chan_sip handle sip connections. |
| 00:47.01 | kerx | Can you make a outbound SIP connection by any chance? |
| 00:47.44 | ManxPower | extensions.conf parser really doesn't know what you are dialing just that chan_sip said it handles SIP/ destinations so pass the parameters to that channel driver. |
| 00:48.26 | kerx | Well, what I meant is maybe the CDR is not working because I'm passing it the SIP info immediately |
| 00:48.38 | *** join/#asterisk seanmh (n=seanmh@c-68-35-21-64.hsd1.nm.comcast.net) |
| 00:48.39 | ManxPower | kerx: try it and see. |
| 00:48.52 | ManxPower | Maybe your provider just sucks |
| 00:48.52 | kerx | Hehe, I don't know if I would know how to do that in my .call and dialplan :) |
| 00:49.37 | kerx | ManxPower, I tried with Asterisk 1.2.27 straight off of a older VicidialNow CD and I made the same SIP connection and I did a .call file |
| 00:49.41 | kerx | The disposition worked |
| 00:49.42 | ManxPower | read up on .call files and localchannel.txt |
| 00:50.22 | kerx | So I wonder if it's something newly introduced in Asterisk 1.4.x and crept up into the Asterisk 1.6.x |
| 00:50.28 | kerx | Or it could just be my dumb-ass ? |
| 00:50.29 | kerx | heh |
| 00:50.50 | drmessano | ...... |
| 00:51.08 | drmessano | Sorry, dropped a chicken wing on the keyboard.. |
| 00:51.20 | ManxPower | kerx: all major changes are documented un upgrade-1.2.txt and upgrade-1.4.txt and upgrade.txt |
| 00:51.31 | Carlos_PHX | Mmm....chicken wings |
| 00:51.38 | ManxPower | all minor changes are in Changes or Changelog (I don't recall the exact name) |
| 00:52.09 | kerx | ManxPower, k |
| 00:52.24 | kerx | Ok, I think I can do a quick test here w/ a local channel and then use Dial() with the sip info in the dialplan itself |
| 00:56.33 | drmessano | When you come that close to being shivved, you learn that even though a package of crackers may say "Nabisco", in prison, it could have anyone's name on it.. especially your cellmate |
| 00:56.38 | drmessano | Ah crap, wrong window |
| 00:58.45 | kerx | AHAH! |
| 00:58.52 | kerx | Ok, at least something! |
| 00:59.01 | kerx | ok, let me get a pastebin up |
| 00:59.02 | kerx | !topic |
| 00:59.03 | kerx | err |
| 00:59.14 | kerx | ~pastebin |
| 00:59.14 | jbot | [~pastebin] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 00:59.43 | Carlos_PHX | drmessano: Have you ever been in a Turkish prison? |
| 00:59.55 | Carlos_PHX | Wonders if drmessano smoked his dinner. |
| 01:03.53 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 01:03.53 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 01:03.57 | kerx | http://pastebin.ca/1237399 |
| 01:04.01 | drmessano | No, i've never been in a turkish prison.. I'm sure it can't be as bad as Syria. When I worked as a umm.. coffee broker, I was throw into a Syrian prison Hafiz al-Assad's personal security |
| 01:04.08 | drmessano | Worst week of my life, unofficially |
| 01:04.24 | drmessano | by* |
| 01:04.35 | Katty | ello. |
| 01:05.11 | drmessano | ehlo |
| 01:06.04 | jaytee | hi Katty |
| 01:06.05 | *** join/#asterisk Kumbang (n=dsp@rusnas.paume.itb.ac.id) |
| 01:06.12 | Carlos_PHX | Whoa, Cox just upgraded us again at home. 22mbps down, 2.4 up. |
| 01:06.17 | Carlos_PHX | I mean 3.4 up. |
| 01:06.17 | kerx | So what do you think about my CDR's on the pastebin? |
| 01:06.18 | Katty | jaytee: herro, how be? |
| 01:06.27 | kerx | Carlos_PHX, Wow, nice rate |
| 01:06.31 | tzanger | Carlos_PHX: fuck off. I'm stuck at 1728 down, 384 up |
| 01:06.34 | *** join/#asterisk IsUp (n=nocturne@unaffiliated/isup) |
| 01:06.35 | tzanger | cries |
| 01:06.41 | Carlos_PHX | Damn, that would suck. |
| 01:06.43 | kerx | tzanger, same here |
| 01:06.47 | tzanger | yep |
| 01:06.52 | Carlos_PHX | That's not bytes I take it. |
| 01:06.52 | lmadsen | tzanger: you suck! :) |
| 01:06.54 | lmadsen | 10/1 here |
| 01:06.56 | jaytee | Katty, I'm fine how's you 'n da Riddick? |
| 01:06.58 | tzanger | lmadsen: only when asked |
| 01:07.06 | lmadsen | tzanger: o.O |
| 01:07.24 | lmadsen | btw: if you have a home office, and no plants, go do yourself a favour and get something for your desk |
| 01:07.26 | tzanger | lmadsen: get your mind out of the gutter; I reserve the right to determine elligibility |
| 01:07.42 | lmadsen | tzanger: phew! I was slightly more worried than usual |
| 01:07.48 | jaytee | I have Comcast and I get like 3-4 down and maybe a packet a week up |
| 01:07.52 | tzanger | hah |
| 01:07.59 | [TK]D-Fender | tzanger: No DSL in your area? |
| 01:08.10 | lmadsen | that sounds like DSL speeds |
| 01:08.12 | [TK]D-Fender | tzanger: Actually... that IS DSL isn't it? |
| 01:08.20 | [TK]D-Fender | tzanger: What ISP? |
| 01:08.21 | Katty | jaytee: i feel... odd, to be honest. |
| 01:08.28 | lmadsen | especially if you're not like 10 meters away from the CO |
| 01:08.33 | Katty | jaytee: every time i think i'm getting ill, i have a mild form of panic attack. |
| 01:08.36 | tzanger | [TK]D-Fender: doesn't matter what hte ISP is, it's DSL |
| 01:08.51 | [TK]D-Fender | tzanger: I get 5000/800 on mine... |
| 01:08.52 | Katty | jaytee: been having this odd pain on the right side of my head everytime i cough |
| 01:08.54 | tzanger | DSL = Damn Slow Link |
| 01:09.08 | tzanger | [TK]D-Fender: yeah, I'm in a bad neighbourhood for DSL; won't go Rogers cable though, they have caps |
| 01:09.11 | [TK]D-Fender | tzanger: Mind you I'd gotten down-clocked to 3000/800, but I'd trade down for up in a heartbeat |
| 01:09.20 | Katty | jaytee: ryan thinks it's just sinus related... |
| 01:09.32 | [TK]D-Fender | tzanger: Yeah, cable contracts are garbage... |
| 01:09.36 | Katty | jaytee: me, being a hypochondriac, makes me wonder if it's not something more. but i'm too afraid to go to the doctor. |
| 01:09.37 | Carlos_PHX | Qwest has been claiming they have 20mb DSL available, but it's still less than 1mb up. |
| 01:09.37 | tzanger | yep |
| 01:09.40 | ManxPower | lmadsen: A plant would die if it was on my desk, it would not get enough light |
| 01:09.45 | Katty | jaytee: and everytime i think about it, it tends to freak me out a good bit. |
| 01:09.56 | [TK]D-Fender | ManxPower: Change your bulbs :) |
| 01:10.01 | tzanger | Katty: I'm about as anti-hypochondriac as they come... my wife keeps the kids clean, I teach them how to get dirty, everyone wins :-) |
| 01:10.09 | lmadsen | ManxPower: that's too bad.... I face south at the water, so I have sun all day long |
| 01:10.16 | Katty | tzanger: lucky for you. |
| 01:10.20 | ManxPower | Quit y'er whining. The only service available where I live is Satellite, Dialup, or EVDO (Verizon "aircard") |
| 01:10.22 | kerx | Fawk this issue.... |
| 01:10.31 | Katty | tzanger: i think about getting sick, and i start shaking uncontrolably :/ |
| 01:10.31 | kerx | ManxPower, did you have a chance to look? |
| 01:10.34 | kerx | http://pastebin.ca/1237399 |
| 01:10.37 | kerx | here is my pastebin |
| 01:10.42 | lmadsen | I start getting sick, and get myself a drink |
| 01:10.44 | tzanger | ManxPower: actually I get better speeds on my 3G connection with Rogers |
| 01:10.50 | kerx | It's weird how it does all 3 logs now that I use local |
| 01:10.52 | jaytee | Katty, are your sinuses stuffed up? |
| 01:10.57 | kerx | but it actually says "ANSWER" now on the disposition |
| 01:11.22 | lmadsen | thanks god he doesn't have to rely on CDRs in Asterisk |
| 01:11.24 | kerx | but it makes 3 records in the Master.csv when I pick up the phone |
| 01:11.34 | kerx | lmadsen, what do you use!! ? please tell me, I will switch |
| 01:11.47 | lmadsen | kerx: no, I don't worry about CDRs in my applications, sorry :( |
| 01:11.47 | kerx | i need to have really good records, for everything |
| 01:11.50 | kerx | oh ok |
| 01:11.52 | ManxPower | tzanger: The EVDO stuff on the local tower has been flaky the past few days, at random times I just switch to 1xRTT (114K) then a few hours, it switches back to EVDO |
| 01:11.53 | kerx | cries |
| 01:11.58 | lmadsen | kerx: what I've done is create my own records using func_odbc |
| 01:12.18 | tzanger | urf |
| 01:12.22 | ManxPower | kerx: wait for responses to your mailing list post. |
| 01:12.22 | lmadsen | kerx: but you will run into a problem when you do an attended transfer with SIP because you won't know it's a transfer |
| 01:12.28 | kerx | ManxPower, ok |
| 01:12.37 | jaytee | I've had no problems with CDR using mysql in 1.4 |
| 01:12.38 | lmadsen | asterisk requires patience... you'll go insane if you're not |
| 01:12.43 | Carlos_PHX | ManxPower: You should move. |
| 01:12.56 | Carlos_PHX | Or just carry your packets to the closest access point and send them from there. |
| 01:13.24 | ManxPower | I live in a cabin on a mountain at an Intentional Community. I'm not moving. |
| 01:13.54 | tzanger | ManxPower: it's called a highly directional antenna |
| 01:13.55 | kerx | I just know it's obviously something stupid, when I use a local channel it begins the correct disposition |
| 01:13.56 | jaytee | he's that crusty old man at the top of the hill yelling "Get off my lawn!!! damn kids!!" all the time :-) |
| 01:13.59 | ManxPower | If Helium were not so expensive I'd try an AP hanging off a weatherbaloon |
| 01:14.00 | kerx | But anyways, I will be patient.... |
| 01:14.02 | tzanger | point it at the nearest decent tower |
| 01:14.12 | tzanger | what is an intentional community? |
| 01:14.20 | jaytee | I was just going to ask that |
| 01:14.24 | ManxPower | tzanger: the Wikipedia is your friend |
| 01:14.29 | kerx | I'll just be patient and begin reading asterisk source code |
| 01:14.42 | kerx | How long has asterisk been around? |
| 01:14.46 | tzanger | kerx: don't read asterisk source code, you'll *certainly* go insane |
| 01:14.50 | Carlos_PHX | Sounds like a bunch of dirty hippies in a commune. :-p |
| 01:15.00 | kerx | Is it like a 1-2 year old project? |
| 01:15.05 | lmadsen | kerx: 1.4 has been around for like... 2+ years now |
| 01:15.09 | tzanger | ManxPower: interesting; what are your community's specific interests |
| 01:15.10 | lmadsen | project is aroudn 6 years old |
| 01:15.17 | kerx | wow, ok, it must just be the CDR stuff is a puppy |
| 01:15.19 | lmadsen | I've been using Asterisk for over 5 years |
| 01:15.32 | ManxPower | I just switched back to 1xRTT *grumble* |
| 01:15.32 | jaytee | it's an ashram sortof, he's channeling Shirley McClaine |
| 01:15.45 | ManxPower | tzanger: www.bluffcreekfalls.com |
| 01:16.00 | lmadsen | kerx: the CDR stuff was never designed initially to handle the complexness that Asterisk has developed into. codefreeze-lap has been doing a lot to help get it to a "logical" working order, but unfortunately when you change one thing in CDR, another mole pops up somewhere else |
| 01:16.08 | Carlos_PHX | ManxPower: Seriously, I use a power booster and antenna on my boat, HUGE improvement in EV-DO coverage. |
| 01:16.12 | ManxPower | lmadsen: when did you start? I started late 2001 or early 2002 |
| 01:16.16 | kerx | lmadsen, heh, ok |
| 01:16.20 | lmadsen | kerx: there is something called CEL that is in testing |
| 01:16.21 | kerx | how can I contact codefreeze-lap ? |
| 01:16.26 | lmadsen | codefreeze-lap: ping! :) |
| 01:16.35 | lmadsen | kerx: search for him as murf on the bug tracker |
| 01:16.38 | tzanger | are one of the guyus on the cliff you? |
| 01:16.52 | lmadsen | ManxPower: late 2002 |
| 01:16.53 | kerx | lmadsen, do you mean CELL |
| 01:16.54 | kerx | ? |
| 01:16.59 | lmadsen | kerx: I do not |
| 01:17.02 | kerx | ok |
| 01:17.03 | ManxPower | lmadsen: something needs to be done with CDRs, it is holding Asterisk back. |
| 01:17.14 | kerx | google asterisk cel doesn't bring up too much |
| 01:17.15 | lmadsen | ~cel |
| 01:17.22 | ManxPower | tzanger: nope. |
| 01:17.25 | lmadsen | jbot: cel is Channel Event Logging |
| 01:17.26 | jbot | lmadsen: okay |
| 01:17.54 | kerx | I feel that either the CDR's is broken, or maybe everyone is doing Call Detail Recording another way, and I'm just stuck on CDR right now |
| 01:17.55 | tzanger | sounds like an interesting place, that's for sure |
| 01:18.00 | ManxPower | this intentional community is more for profit and less about being a commune |
| 01:18.09 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 01:18.21 | Carlos_PHX | Actually sounds like a damn good idea as I read about it. |
| 01:18.22 | jaytee | with lots of nekkid people wandering around |
| 01:18.25 | Carlos_PHX | How do you find them? |
| 01:18.26 | kerx | But definitely if Asterisk had stronger logging, it would be nice |
| 01:18.28 | Carlos_PHX | Heh, bonus. |
| 01:18.57 | Carlos_PHX | ManxPower: Ever thought of bringing something like Wi-MAX into the place to resell? |
| 01:19.11 | ManxPower | There are lots of the commune type of intentional communities in the hills, near the Jack Daniels company, IIRC. |
| 01:19.36 | ManxPower | Carlos_PHX: some of the personalities here would cause that to not work. |
| 01:19.38 | *** join/#asterisk IsUp (n=nocturne@unaffiliated/isup) |
| 01:20.14 | jaytee | not the "technawlagee is the seed of the devil" types I hope? |
| 01:20.15 | riddlebox | figured out my problem with not having any sip clients over the internet being able to connect, my dd-wrt i forwarded 5060-5062, and as soon as I made it 5060-5060 it started working |
| 01:20.28 | lmadsen | kerx: here is a post I think you should read: http://www.asterisk.org/node/48358 |
| 01:20.39 | kerx | lol, i'm on it! :) |
| 01:20.43 | IsUp | hello |
| 01:20.48 | kerx | after you told me about CEL |
| 01:20.50 | kerx | thanks |
| 01:21.18 | *** join/#asterisk RA25 (n=RA25@c-65-96-173-21.hsd1.ma.comcast.net) |
| 01:22.03 | kerx | http://svn.digium.com/svn/asterisk/team/murf/CDRfix5 |
| 01:22.04 | kerx | :-( |
| 01:22.37 | lmadsen | kerx: that branch has been merged into recent asterisk |
| 01:22.39 | Carlos_PHX | ManxPower: I use one of these with my Verizon card and it's all the difference between 1x and EV-DO in marginal areas: http://www.digitalantenna.com/prods/cellbooster_DA4000_directconnectamplifier.html |
| 01:22.48 | lmadsen | kerx: that is why it does not exist |
| 01:23.15 | ManxPower | Carlos_PHX: Oh, I get 3 bars |
| 01:23.27 | kerx | ok |
| 01:23.30 | Carlos_PHX | I found the bars to be a lie. |
| 01:23.36 | kerx | I definitely need to find a way to speak to 'mashup' |
| 01:23.37 | jaytee | just like the cake |
| 01:23.40 | kerx | Does he come on IRC ever? |
| 01:23.49 | Carlos_PHX | I can get 4 bars and get 1x, switch on the booster and still get 4 bars but EV-DO. |
| 01:24.11 | Carlos_PHX | I think it means that it's just switching from a 1x cell to an EV-DO cell. |
| 01:24.11 | kerx | Wow someone responded to my asterisk-users email |
| 01:24.19 | lmadsen | kerx: he is on IRC quite often, I'm sure he is hanging out with his family this evening |
| 01:24.20 | kerx | "I have the same problem for Disposition when I use call files. The duration is correct but the Disposition is always NO ANSWER. I also am using 1.6.0.1. I did not have the problem when I was using 1.4.21" |
| 01:24.36 | kerx | Weird how he even was able to get the billsec correct |
| 01:24.55 | lmadsen | what version are you using? |
| 01:25.31 | drmessano | Trying to use a USR robotics winmodem for asterisk is a PITA.. I cant crimp the RJ45s on the end of the serial cable :( |
| 01:25.47 | drmessano | s/USR/US |
| 01:25.56 | drmessano | :( |
| 01:26.02 | kerx | lmadsen, i've tried both 1.6.0.1 and 1.4.22 |
| 01:26.11 | lmadsen | drmessano: lol |
| 01:26.19 | jaytee | shakes his head......danny, danny, danny, whatever are we gonna do with you? |
| 01:32.30 | *** join/#asterisk simprix (n=simprix@c-71-205-52-252.hsd1.mi.comcast.net) |
| 01:33.08 | *** join/#asterisk dmz (n=dmz@64.203.203.232.dyn-cm-pool-64.hargray.net) |
| 01:43.10 | jaytee | wow, this room went from busy to all of a sudden like being in #asterisknow |
| 01:43.53 | *** join/#asterisk chunkxzor (n=chunkxzo@c-68-62-254-17.hsd1.fl.comcast.net) |
| 01:44.06 | jeev | wow, the room went from being busy to asking sarah palin an easy question. |
| 01:44.10 | jeev | silenec |
| 01:44.12 | jeev | silence |
| 01:45.38 | Carlos_PHX | Any second it will go to being like asking Obama how he's going to fix the economy. |
| 01:45.42 | Carlos_PHX | Piles of bs... |
| 01:46.20 | jeev | lets ask Sarah palin what she knows about foreign policy |
| 01:46.32 | jeev | and how her and mccain are mavericks |
| 01:46.44 | Carlos_PHX | Let's ask Obama what he knows about anything. |
| 01:46.46 | jeev | and what kind of maverick fires someone over a divorce |
| 01:46.48 | jeev | LOL |
| 01:46.54 | jeev | jaytee |
| 01:46.55 | jeev | .. |
| 01:47.00 | jeev | ok dood |
| 01:47.11 | jeev | you go vote for someone who came in bottom 5% of his MILITARY class |
| 01:47.14 | jeev | and i'll be back in a bit |
| 01:47.21 | Carlos_PHX | Lest you think I'm a McCain fan, however, I'm just responding to your incessant trolling. |
| 01:47.38 | Carlos_PHX | Fanboyism for any of these worthless scumbags is unbecoming. |
| 01:48.29 | jaytee | I'm a member of the Whig party and we haven't had anyone in the White House since Millard Fillmore so this year's election is no more disturbing than the last one for me. |
| 01:53.42 | *** join/#asterisk coolthreads (n=coolthre@203-97-238-71.cable.telstraclear.net) |
| 01:57.35 | *** join/#asterisk mayo (n=mayo@d206-116-26-86.bchsia.telus.net) |
| 02:03.03 | coolthreads | been trying for ages to get the answering system to produce clear enough sound, but sounds broken, choppy and static. anyone been able to solve this problem? |
| 02:04.01 | jaytee | answering system? |
| 02:04.06 | coolthreads | my phone calls sound great, no complains there. |
| 02:04.28 | coolthreads | interactive voice menu i mean i guess |
| 02:04.42 | Carlos_PHX | Do you have some telephony hardware installed or using ztdummy? |
| 02:04.57 | [TK]D-Fender | ~gsmbug |
| 02:04.58 | jbot | [~gsmbug] there is a bug compiling Asterisk with GCC 4.2 where optimization errors cause GSM transcoding to distort heavily. Compile with GCC 4.1 or lower, or follow this patch : http://bugs.digium.com/view.php?id=11243 |
| 02:04.59 | [TK]D-Fender | ^^^^^^ |
| 02:05.14 | [TK]D-Fender | coolthreads: Above is highly suspect |
| 02:06.02 | coolthreads | thanks |
| 02:06.11 | coolthreads | no telephony hardware |
| 02:12.45 | *** join/#asterisk chunkxzor (n=chunkxzo@c-68-62-254-17.hsd1.fl.comcast.net) |
| 02:14.33 | jeev | Carlos_PHX, it's not being a fanboy, get a clue |
| 02:26.06 | drmessano | Someone is voting for McCain? |
| 02:26.11 | drmessano | Cool.. can I touch you? |
| 02:35.44 | chunkxzor | you can touch me ;) |
| 02:42.41 | *** join/#asterisk UnixDawg (n=UnixDawg@155.129.204.68.cfl.res.rr.com) |
| 02:42.59 | Carlos_PHX | Heh, that was a fanboy test. You passed. Not in a good way. |
| 02:43.41 | Carlos_PHX | Let's see, we've covered politics, cell phones, and drmessano's strange prison fantasies (of all those I like the latter most). Anything else for tonight? |
| 02:44.04 | Carlos_PHX | Feels like I'm on CIS chat in the 90s. |
| 02:45.33 | jeev | was he talking about gaining weight and playing with his own titties? |
| 02:45.40 | jaytee | hmmm, how about trading recipes for flan? |
| 02:45.48 | Carlos_PHX | Mmmmm...flan. |
| 02:46.00 | jeev | ew |
| 02:46.10 | jaytee | I'll accept tiramasu as a substitute |
| 02:46.20 | Carlos_PHX | I'm Cuban, so it's just natural for me. |
| 02:46.30 | Carlos_PHX | Like cigars and building boats out of 50s pickups. |
| 02:46.37 | jaytee | ooooh, I'd kill for a Cohiba |
| 02:46.38 | jeev | jaytee, order this: http://www.mashtimalones.com/merchant2/merchant.mvc?Screen=CTGY&Category_Code=250 |
| 02:46.40 | jeev | delish |
| 02:46.45 | jeev | ahmadinejad style ice cream |
| 02:47.02 | jeev | wow, wholefoods sells it now |
| 02:47.03 | jaytee | iranian food gives me gas |
| 02:47.15 | Carlos_PHX | Hmm, there's a war joke in there somewhere. |
| 02:47.20 | jaytee | they do have the best caviar though |
| 02:47.42 | jeev | ew seafood |
| 02:47.45 | jeev | jaytee, order that ice cream |
| 02:47.45 | jaytee | war joke? I was talking about a cuban cigar |
| 02:47.47 | jeev | you will not be disappointed |
| 02:50.21 | *** join/#asterisk Zizou (n=zizou@190.75.194.51) |
| 02:55.47 | *** join/#asterisk CrashSys (i=Kumba@azrael.crashsys.com) |
| 02:56.46 | CrashSys | Anyone seen the G729 codec not install cause "Cannot restore segment prot after reloc: permission denied"? |
| 02:57.27 | CrashSys | it tosses that up when asterisk tries to laod |
| 02:58.31 | file | selinux |
| 02:58.39 | CrashSys | !(JFOJSDAL:FJ)#*$%)!*40581 |
| 02:58.41 | CrashSys | :( |
| 03:00.35 | Carlos_PHX | You don't by any chance have an "unlimited" user license for g729 do you... |
| 03:00.45 | CrashSys | Nope... 2-seats... |
| 03:05.19 | CrashSys | OMG... this guy used users.conf... |
| 03:06.16 | jameswf | A positive attitude may not solve all your problems, but it will annoy enough people to make it worth the effort.\ |
| 03:06.44 | jaytee | lol |
| 03:06.56 | CrashSys | Can you even specify codec options in users.conf? |
| 03:07.02 | Katty | ninite. |
| 03:07.30 | *** join/#asterisk ionix (n=ionix@122x220x154x138.ap122.ftth.ucom.ne.jp) |
| 03:09.40 | jeev | yum, peanut butter sammich |
| 03:14.24 | Carlos_PHX | Mmmm...Glenlivet on ice... |
| 03:17.29 | *** join/#asterisk osiris (n=osiris@c-71-205-27-88.hsd1.mi.comcast.net) |
| 03:18.25 | *** join/#asterisk magic_hat (n=geoffdou@h-66-167-66-201.chcgilgm.dynamic.covad.net) |
| 03:19.50 | magic_hat | hey all. I'm having a devil of a time sorting something out... I'm getting an ast_streamfile failed on my greeter file on inbound calls. The file exists in /var/lib/asterisk/sounds, and it's chmod 777. |
| 03:20.21 | [TK]D-Fender | magic_hat: pastebin everything |
| 03:21.44 | magic_hat | http://pastie.org/301153 |
| 03:22.29 | magic_hat | it's a new server and I'm sure something is not quite wired right... but damned if I know what it is. |
| 03:22.38 | [TK]D-Fender | magThat includes folder dumps, full dialplan exectuion at verbose 10, core debug 10, etc |
| 03:23.15 | Carlos_PHX | Have those greeting files worked before? Format? |
| 03:23.33 | magic_hat | Carlos_PHX: yeah, the greeting file has worked w/ this dialplan on another server. |
| 03:23.49 | magic_hat | That's what's making me crazy. |
| 03:24.13 | [TK]D-Fender | magic_hat: New PB please... |
| 03:24.24 | Carlos_PHX | Can you play any stock files? And as [TK]D-Fender said, more info. |
| 03:24.42 | [TK]D-Fender | screw stock files... show us the PROBLEM |
| 03:27.02 | magic_hat | brb... may have found something. |
| 03:32.24 | Gopher_77 | is it possible to create sound devices with asterisk? |
| 03:32.48 | [TK]D-Fender | Gopher_77: As in? |
| 03:33.06 | Gopher_77 | [TK]D-Fender: as in an alsa device |
| 03:33.19 | [TK]D-Fender | Gopher_77: to do what exactly? |
| 03:34.16 | Gopher_77 | [TK]D-Fender: to allow skype to communicate through a phone |
| 03:34.34 | [TK]D-Fender | ~skype |
| 03:34.38 | jbot | [~skype] Skype is a free VoIP software and service using a closed client and propritary protocol. Only commercial channel drivers exist, with most solutions being complex, complicated, and hack-ish . Digium's SkypeForAsterisk (see ~SkypeForAsterisk) is a new solution that is a cleaner non-dependent option. |
| 03:34.38 | [TK]D-Fender | ~skypefor asterisk |
| 03:34.39 | Gopher_77 | [TK]D-Fender: or another application |
| 03:34.41 | [TK]D-Fender | ~skypeforasterisk |
| 03:34.42 | jbot | [~skypeforasterisk] is a Digium-made channel driver that allows Asterisk to connect to the Skype network using a Skype-supported connection method. Unlike all other solutions it does not require X11, kernel modules, or a Windows machine. See http://www.astricon.net/skype for beta details. |
| 03:34.57 | [TK]D-Fender | Gopher_77: *'s only also interface takes it over completely... |
| 03:35.02 | [TK]D-Fender | alsa* |
| 03:35.22 | Gopher_77 | [TK]D-Fender: interesting |
| 03:37.35 | *** join/#asterisk CunningPike (n=arodgers@S010600095b33697f.vc.shawcable.net) |
| 03:37.36 | Gopher_77 | [TK]D-Fender: hmm.... isn't skype closed source? |
| 03:40.49 | [TK]D-Fender | Gopher_77: Yes |
| 03:41.02 | Gopher_77 | [TK]D-Fender: so it's a joint venture then? |
| 03:41.09 | *** join/#asterisk jjg (n=jjg@76.21.4.40) |
| 03:41.26 | jjg | is it possible to make a sip call from the cli? |
| 03:42.12 | ionix | and that to the phone? :) |
| 03:42.14 | ionix | chat |
| 03:42.30 | ionix | not from Asterisk |
| 03:42.45 | ionix | but yes, you can find sip utils to connect to asterisk and establish sip. |
| 03:42.55 | jjg | ionix, talking to me? |
| 03:42.59 | ionix | Or just copy a correct file in asterisk's spooler |
| 03:43.13 | ionix | yeh |
| 03:43.26 | jjg | ahh, i remember the spooler thing now |
| 03:43.43 | jjg | ok, thanks for the info .. |
| 03:44.19 | [TK]D-Fender | jjg: Yes you can dial from * CLI. |
| 03:44.34 | [TK]D-Fender | jjg: look at "dial" or "console dial" |
| 03:44.44 | jjg | [TK]D-Fender, ok will do, thanks |
| 03:44.48 | ionix | uh since when |
| 03:44.54 | [TK]D-Fender | jjg: However you should really call through another device set up through * |
| 03:45.03 | [TK]D-Fender | ionix: sice over 5 years. |
| 03:45.19 | ionix | with the special chan_oss |
| 03:45.24 | [TK]D-Fender | (since thats how long I've been using *) |
| 03:45.47 | ionix | then it goes through the sound card |
| 03:45.55 | ionix | but no console dial out of the box |
| 03:46.17 | [TK]D-Fender | ionix: huh? |
| 03:46.27 | [TK]D-Fender | ionix: What does "out of the box" imply here? |
| 03:46.52 | jjg | i'm testing an embedded ua that is connecting to * on my laptop ... if I run another ua ( ie., ekiga ) will i see issues? |
| 03:47.05 | ionix | it implies after a normal ./configure make make install |
| 03:47.26 | ionix | without the need to compile the extra chan_oss module |
| 03:47.29 | [TK]D-Fender | ionix: OSS is available after a basic install... |
| 03:48.08 | [TK]D-Fender | jjg: if you want to test a UA on your laptop you can always install another SIP device on your laptop on a different signalling port, ot use an IAX2 client, etc |
| 03:48.11 | ionix | to be compiled separatly or by editing the makefile, no/ |
| 03:48.18 | jjg | that is to say ... if i run a ua on my laptop while * is running will I see problems? I'm building the ua now, anyway .. but just curious if anyone knows for sure |
| 03:48.27 | [TK]D-Fender | ionix: No. |
| 03:48.28 | jjg | [TK]D-Fender, got it |
| 03:49.51 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 03:49.56 | ionix | Alright, maybe I had a weird build. I had to build chan_oss manually to enable intercom |
| 03:50.42 | [TK]D-Fender | ionix: first guess one might venture for your situation is a lack of prerequisites when you first did your build. |
| 03:51.13 | jjg | anyone using h.264? |
| 03:52.00 | *** join/#asterisk jjshoe (i=jjshoe@cpe-76-175-157-237.socal.res.rr.com) |
| 04:20.20 | *** join/#asterisk drmessano (n=nonya@pdpc/supporter/active/drmessano) |
| 04:22.31 | *** join/#asterisk CunningPike (n=arodgers@S010600095b33697f.vc.shawcable.net) |
| 04:34.12 | *** join/#asterisk jameswf (n=james@ip68-109-160-72.ph.ph.cox.net) |
| 04:34.44 | Gopher_77 | does asterisk work with hisax? |
| 04:36.19 | WimpMan | no |
| 04:36.45 | WimpMan | You have to use zaptel(dahdi), mISDN or visdn. |
| 04:37.14 | WimpMan | Well, you could use i4l with chan_modem, but you don't want that. |
| 04:37.50 | *** join/#asterisk tobias (n=tobias@user-0ce2hvs.cable.mindspring.com) |
| 04:40.33 | *** join/#asterisk sbc383 (n=stuff@S0106000fea3b527f.cg.shawcable.net) |
| 04:42.48 | *** join/#asterisk mmlj4 (n=jkelly@ip70-171-94-246.no.no.cox.net) |
| 04:44.25 | sbc383 | What's the proper term for when a group of analog lines are tied to 1 phone number? Eg. My office has 8 lines, when someone dials our number, the call will be connected on whatever line is free. So up to 8 people may call at the same time. I heard someone once call this an 8-line rotary, but I'm not sure if that's correct. |
| 04:45.54 | justdave | sounds like a ring group to me |
| 04:45.59 | trelane | that's a really old term for it, it's also called a hunt group or ring group |
| 04:46.24 | justdave | hunt group with "ring all" as the hunt method |
| 04:47.27 | *** join/#asterisk implicit (n=bayan@unaffiliated/implicit) |
| 04:47.50 | sbc383 | ah, I see. Thanks very much. |
| 04:48.53 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.176) |
| 04:49.58 | justdave | so what's the proper way to get bugs fixed in the Asterisk::Manager Perl module? I see one bug ever mentioning it on bugs.digium.com, and two bugs on CPAN's RT that are pretty old |
| 04:50.23 | justdave | I have a patch I can submit |
| 04:51.25 | [TK]D-Fender | sbc383: Typical term used by telco's is "hunt group" or "line hunting" |
| 04:52.03 | [TK]D-Fender | sbc383: the first (or primary) line is typically called the "pilot" and is the number that will hunt against the others in order. |
| 04:54.26 | X-Rob | justdave, if the project is abandoned, you can either create a new one and fork it, or apply to take it over on the cpan mailing list |
| 04:57.55 | justdave | hmm, that's not actually distributed by Digium is it... |
| 04:58.04 | justdave | I was thinking that was from asterisk-addons or something |
| 04:58.07 | justdave | but I don't see it in there |
| 04:58.13 | justdave | must have gotten it out of cpan :) |
| 05:01.45 | *** join/#asterisk jc_yyz2bkk (n=jc@ppp-58-8-249-134.revip2.asianet.co.th) |
| 05:01.50 | tzafrir_laptop | Gopher_77, hisax is an isdn4linux (i4l) driver |
| 05:02.28 | tzafrir_laptop | What card is it? |
| 05:03.31 | tzafrir_laptop | The Asterisk-perl modules are now in CPAN |
| 05:03.39 | tzafrir_laptop | and are actively maintianed |
| 05:03.50 | tzafrir_laptop | Though Asterisk::Manager never really worked |
| 05:04.16 | justdave | yeah, that's what I discovered tonight. |
| 05:04.25 | *** join/#asterisk russellb (n=russell@asterisk/digium-open-source-team-lead/russellb) |
| 05:04.25 | *** mode/#asterisk [+o russellb] by ChanServ |
| 05:04.28 | justdave | think I got it working though. |
| 05:04.41 | tzafrir_laptop | patches would be welcomed |
| 05:04.41 | jc_yyz2bkk | hi all, ive opened up all the required ports, added my internip info, and i cant hear anything when i call via sip to the asterisk voicemail using xlite or SIP to SIP. If i use skype or gizmo i can talk with people(not through asterisk)... im running asterisk 1.4 |
| 05:05.02 | jc_yyz2bkk | any ideas? |
| 05:05.21 | justdave | sendcommand had a hash as the second parameter followed by a scalar, that doesn't actually work unless the hash is the last parameter. have to make it a hash reference otherwise |
| 05:05.46 | justdave | changing it to take a hash reference and fixing the callers seems to have gotten it working |
| 05:08.05 | justdave | jc_yyz2bkk: have nat=yes on all of the sip registrations in sip.conf as well? |
| 05:08.27 | justdave | (at least for the ones that connect from outside your firewall) |
| 05:08.56 | justdave | also, did you open ports for RTP and tell which ports you opened for it in rtp.conf? |
| 05:09.37 | [TK]D-Fender | ~sipnat |
| 05:09.38 | jbot | [~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 |
| 05:09.42 | [TK]D-Fender | ^^^ read VERY closely |
| 05:10.00 | [TK]D-Fender | bed time.. later all |
| 05:10.01 | Gopher_77 | tzafrir_laptop: and * doesn't support isdn4linux anymore right? |
| 05:10.24 | tzafrir_laptop | Gopher_77, no. mISDN instead. What card is it? |
| 05:10.35 | *** join/#asterisk ManxPower (n=manxpowe@248.sub-75-202-132.myvzw.com) |
| 05:10.50 | Gopher_77 | tzafrir_laptop: Tiger Jet 3XX (FXO) |
| 05:11.05 | tzafrir_laptop | use zaptel for that |
| 05:11.15 | Gopher_77 | tzafrir_laptop: thanks |
| 05:11.26 | tzafrir_laptop | it's not an ISDN card. |
| 05:12.13 | Gopher_77 | tzafrir_laptop: lspci says 01:09.0 Communication controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface |
| 05:13.06 | tzafrir_laptop | that's because it has the same vendor ID and product id |
| 05:13.20 | tzafrir_laptop | zaptel_hardware will tell you something different :-) |
| 05:13.29 | jc_yyz2bkk | justdave: yes, yes, yes, and yes |
| 05:14.09 | Gopher_77 | tzafrir_laptop: how do I get zaptel to use this device instead of hisax? |
| 05:14.11 | tzafrir_laptop | justdave, so please send your patches to the list |
| 05:14.24 | tzafrir_laptop | do you have zaptel installed? |
| 05:14.31 | Gopher_77 | tzafrir_laptop: now I do |
| 05:14.54 | justdave | tzafrir_laptop: yup, found the maintainers website and was working on subscribing to the list as we speak :) |
| 05:14.59 | tzafrir_laptop | I'm not sure if you actually need to blacklist hisax |
| 05:15.05 | justdave | do there happen to be archives for that list anywhere? |
| 05:15.52 | tzafrir_laptop | http://asterisk.gnuinter.net/mailinglists.html I'm not aware of any archives, but it's ezmlm, so you can ask it for the previous messages |
| 05:18.16 | *** join/#asterisk rrrobert (n=rrrobert@202.125.156.122) |
| 05:22.45 | rrrobert | Hi i want to set a channel variable such that i can be access in multiple context within dialplan for each channel its value should be unique ....I have tried Set(GLOBAL(var)=${EXTEN}) but its not unique for multiple channels...and simple set does not work over multiple contexts ..any suggession |
| 05:27.09 | justdave | using Asterisk::Manager to send $ami->command() is ever so much faster than system("asterisk -rx 'command'") :) |
| 05:28.18 | *** join/#asterisk coolthreads (n=coolthre@203-97-238-71.cable.telstraclear.net) |
| 05:37.28 | *** join/#asterisk axisys (n=axisys@117.18.231.197) |
| 05:37.40 | jeev | shit |
| 05:37.43 | jeev | i forgot one of my root passwords |
| 05:37.49 | *** part/#asterisk axisys (n=axisys@117.18.231.197) |
| 05:38.17 | russellb | jeev: actually, i hacked your box and changed it. |
| 05:38.27 | jeev | russellb, the box was down for a month, foo. |
| 05:38.27 | Gopher_77 | lol |
| 05:38.34 | jeev | dont make me p0ke your eyes out with my boogery fingers. |
| 05:38.38 | *** join/#asterisk [gquit]bombadil (n=dana@CPE-70-94-44-157.wi.res.rr.com) |
| 05:38.49 | Gopher_77 | night elf mohawk is a reality |
| 05:40.56 | *** join/#asterisk ElCheapo (n=elcheapo@d199-126-36-20.abhsia.telus.net) |
| 05:42.28 | drmessano | foo? |
| 05:42.36 | drmessano | ..that's a ban |
| 05:44.25 | jeev | wow |
| 05:44.26 | jeev | # grep -c failure messages |
| 05:44.26 | jeev | 32 |
| 05:44.28 | jeev | i guessed the pass |
| 05:44.30 | jeev | 32 failed tries |
| 05:45.22 | drmessano | If I had a MasterCard Black Plutonium Card, I would have bought a new root password before trying 32 times. |
| 05:53.11 | Gopher_77 | didn't those accounts get hacked? |
| 05:55.03 | jeev | no |
| 05:55.09 | jeev | that was me, trying to guess my root password |
| 05:55.23 | jeev | amazing, i have 32 passwords i use |
| 05:55.23 | jeev | lol |
| 05:55.26 | jeev | i shouldn't have given that info out |
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| 06:19.12 | *** join/#asterisk rrobert (n=rrobert@202.125.156.122) |
| 06:35.20 | rrobert | How to set a channel variable within a dial plan, so that it could be accessable in different contexts |
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| 07:35.48 | ghostknife | I have an odd sitution. Our switchboard phone can't accept calls anymore. Doing a direct IP call doesn't work. But dialing it's extension doesn't. It also doesn't accept incoming calls (they goto the failover extension lists). I get this in the log when trying to make a call from extension 16 to extension 11. Though IT can dial extension 16 just fine: http://rafb.net/p/JZ1bLc85.html |
| 07:35.54 | ghostknife | Any ideas? |
| 07:37.40 | ghostknife | This is a successful call from 11 to 16: http://rafb.net/p/eNI0wt68.html |
| 07:44.02 | *** join/#asterisk IsUp (n=nocturne@unaffiliated/isup) |
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| 07:50.55 | *** join/#asterisk Math` (n=mrene@modemcable048.248-130-66.mc.videotron.ca) |
| 07:51.13 | Math` | would the "convert" CLI command work with codec_g729 ? |
| 07:57.49 | *** join/#asterisk blogbasti (n=blogbast@calypso.planet-ic.de) |
| 08:04.19 | styelz | i think convert is for image files |
| 08:07.28 | encode | yep, convert is part of imagemagick |
| 08:07.41 | styelz | oh the asterisk CLI convert |
| 08:07.47 | styelz | my bad |
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| 08:11.28 | kaldemar | Math`: have you tried it? why wouldn't it work? |
| 08:12.00 | Math` | licensing stuff... I wasnt sure of the internals either |
| 08:12.14 | Math` | I need a command line 729 converter, wanted to ask for advice before buying a license for that |
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| 08:26.11 | *** part/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
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| 08:38.12 | *** join/#asterisk Chris-NB (n=chris@nfw.ecos.at) |
| 08:38.14 | Chris-NB | hi |
| 08:38.28 | Chris-NB | anyone familiar with national/international calls in spain? |
| 08:42.43 | *** join/#asterisk defswork (n=andy@mx2.3gcomms.co.uk) |
| 08:43.30 | *** join/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 08:48.55 | *** join/#asterisk mav3rick (n=t0m@trankil.biz) |
| 08:49.16 | mav3rick | hello |
| 08:50.22 | mav3rick | I have a simple question. I have boxes A and B running Asterisk. A dials B. if A crashes, B does not detect the hangup, channel runs indefinitely. How can I prevent this ? How can B discover the crash/hangup ? |
| 08:57.01 | mav3rick | I think I found (rtptimeout) |
| 08:57.04 | hesco | ~pb |
| 08:57.05 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 08:59.00 | *** join/#asterisk Rico29 (n=Rico@lns-bzn-48f-81-56-220-28.adsl.proxad.net) |
| 09:05.34 | yang | hi Rico29 |
| 09:07.35 | *** join/#asterisk angryuser (n=gdobrovo@LPuteaux-151-42-35-99.w193-251.abo.wanadoo.fr) |
| 09:09.03 | Rico29 | hi |
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| 09:50.13 | Daejeo | During the native sip bridging, pbx is dropping the call. what could be the reason? |
| 09:55.57 | angryuser | can someoneone help me with sipsak ? |
| 09:57.04 | angryuser | i am able to generate register sring with sipsak, but instead of configured ip in string it uses the local ip adress, here is the string: |
| 09:57.12 | angryuser | <PROTECTED> |
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| 10:18.19 | angryuser | found it |
| 10:20.58 | *** join/#asterisk magronez (n=eusei@unaffiliated/magrao/x-2903) |
| 10:20.59 | rrrobert | I am trying to write a dial out application, asterisk dial out successfully, but it never connect to my output stream from the dialplan, I am stucked, need some pointers.. |
| 10:22.45 | *** join/#asterisk tkbeat (n=tk@80.64.182.204) |
| 10:24.22 | Daejeo | 701xxx-xxx-xxxx# |
| 10:29.45 | *** join/#asterisk festr_ (n=festr@ns.hiro.cz) |
| 10:29.52 | festr_ | hi |
| 10:30.11 | festr_ | i cannot compile trunk chan_dahdi.so with libpri support. any suggestion? |
| 10:30.21 | festr_ | i've downloaded trunk libpri, make install |
| 10:31.21 | festr_ | i've tried libpri 1.4 without success |
| 10:41.20 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
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| 11:26.20 | *** join/#asterisk DoDaT69 (n=DoDaT69@internal.digitalson.com) |
| 11:26.33 | alexandrekeller | good morning, as I am in Brazil and it's onlye 9:30 am |
| 11:26.35 | DoDaT69 | Does anyone have a link handy for nvfax source? I cant seem to find it.. |
| 11:26.43 | alexandrekeller | has anyone ever seen this message: rc_avpair_new: unknown attribute 1490026597 |
| 11:26.44 | alexandrekeller | ? |
| 11:28.43 | DoDaT69 | never mind.. looks like they are included in agx-ast-addons ;) |
| 11:30.31 | *** join/#asterisk coppice (n=chatzill@33.155.17.210.dyn.pacific.net.hk) |
| 11:34.13 | DoDaT69 | WARNING[8787] app_dial.c: Unable to create channel of type 'SIP' (cause 20 - Unknown) |
| 11:34.37 | DoDaT69 | wtf? anyone caught that before? google does not appear to have any results for cause 20- |
| 11:38.14 | stintel | DoDaT69: I am having that too in a very specific case |
| 11:38.21 | *** join/#asterisk DarkRift (n=dark@65.92.170.95) |
| 11:38.23 | DoDaT69 | faxing? |
| 11:38.41 | stintel | DoDaT69: nope. ringgroup with strategy firstavailable (freepbx) |
| 11:38.55 | DoDaT69 | hmm.. all your extensions registered? |
| 11:39.03 | DoDaT69 | and what version are you running? |
| 11:39.08 | DoDaT69 | I am on 1.4.22 |
| 11:39.11 | stintel | DoDaT69: using deviceanduser mode in freepbx |
| 11:39.16 | stintel | DoDaT69: also 1.4.22 |
| 11:39.37 | DoDaT69 | I am about to downgrade back to previous version I was using.. just came about from an upgrade this weekend.. |
| 11:40.04 | stintel | DoDaT69: ok. can you let me know if you still see that warning after downgrading ? |
| 11:40.19 | DoDaT69 | 1.4.19 was working good, just had problems w/ faxes making asterisk seg fault :( |
| 11:40.27 | stintel | if so I can report that back to freepbx developer - he suggests it's a problem with asterisk |
| 11:40.29 | DoDaT69 | not every time either.. just here and there.. |
| 11:40.44 | DoDaT69 | 10-4, working on downgrade now. |
| 11:40.52 | stintel | DoDaT69: thanks! |
| 11:41.44 | DoDaT69 | np ;) |
| 11:44.05 | *** join/#asterisk Daviey (n=Daviey@ubuntu/member/daviey) |
| 11:45.17 | Daviey | Hey, on a BT PRI/ISDN30 - what number format should i pass for PATS services like 118XXX ? |
| 11:45.55 | coolthreads | I am getting distortion on playback of .gsm sound files |
| 11:45.59 | DoDaT69 | I thinapp_dial.c: Unable to create channel of type 'SIP' (cause 3 - No route to destination) |
| 11:46.03 | DoDaT69 | new error after downgrade |
| 11:46.07 | coolthreads | still not able to resolve issue, has anyone found any solutions to this? |
| 11:46.31 | DoDaT69 | what kind of distortion? |
| 11:46.43 | stintel | DoDaT69: :/ |
| 11:46.45 | DoDaT69 | I experienced something like that this weekend during an upgrade. |
| 11:47.28 | coolthreads | a static, slow, breaking up sound |
| 11:47.57 | DoDaT69 | hmm.. that is kinda like what I had.. I had to recompile my core in a certain order |
| 11:48.14 | coolthreads | its fine when making phone calls |
| 11:48.17 | DoDaT69 | pri or sip trunks? |
| 11:48.27 | coolthreads | sip |
| 11:48.52 | DoDaT69 | hmm |
| 11:49.10 | DoDaT69 | not sure on that one.. logs say anything special? |
| 11:49.33 | kaldemar | coolthreads: there's an issue with the gsm codec and gcc 4.2. |
| 11:49.35 | coolthreads | i understand this is going around ref: http://bugs.digium.com/view.php?id=11243 |
| 11:50.15 | kaldemar | what codec are you using with your phones? |
| 11:50.16 | Daejeo | During the native sip bridging between my two VOIP, pbx is dropping the call. what could be the reason? |
| 11:50.40 | Daejeo | packet2 packet is ok |
| 11:50.43 | coolthreads | g711u |
| 11:51.38 | Daejeo | any educated advise? |
| 11:51.55 | kaldemar | coolthreads: use ulaw sounds then. |
| 11:52.29 | kaldemar | you'll get rid of the distortion and don't have to do transcoding either. |
| 11:52.55 | coolthreads | its works great when im in a conversation, only when i use playback() |
| 11:53.06 | DoDaT69 | how frustrating.. fax will not work for SQUAT!! GRR |
| 11:55.04 | kaldemar | coolthreads: http://downloads.digium.com/pub/telephony/sounds/ |
| 11:56.41 | coolthreads | thanks |
| 11:57.10 | DoDaT69 | trying 1.4.21.2 |
| 11:57.25 | DoDaT69 | I am running that version, not having this issue.. very strange.. |
| 11:57.34 | coolthreads | do you know any apps to convert wav to ulaw? |
| 11:59.48 | DoDaT69 | http://www.freedownloadmanager.org/downloads/wav_ulaw_conversion_info/ |
| 12:00.02 | DoDaT69 | sorry dont know of any free ones off hand.. |
| 12:00.12 | kaldemar | coolthreads: try sox |
| 12:00.33 | coolthreads | sox seems to be the one |
| 12:01.20 | coolthreads | how do you find sox? |
| 12:02.56 | stintel | anyone knows an option to pass to genzaptelconf so that it will not add crc4 to the span= line in /etc/zaptel.conf ? |
| 12:04.30 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 12:05.50 | coolthreads | thanks for your input kaldemar, gives me direction again |
| 12:05.55 | tzafrir_laptop | stintel, doesn't it? |
| 12:06.10 | tzafrir_laptop | stintel, what card is it? E1 or T1? |
| 12:06.52 | stintel | tzafrir_laptop: it's TE110P with E1 |
| 12:07.31 | tzafrir_laptop | stintel, I guess you should add your own option to do that :-( |
| 12:07.56 | stintel | tzafrir_laptop: too bad :( I will just call telco to enable crc4 on the line ;) |
| 12:11.46 | *** join/#asterisk BBHoss (n=bbhoss@c-68-62-170-33.hsd1.al.comcast.net) |
| 12:12.34 | JT | not running CRC4 on an e1 is silly |
| 12:14.31 | stintel | JT: care to elaborate ? or point me to a doc that explains that ? |
| 12:14.48 | coppice | non running CRC4 on an ISDN E1 is silly (though common), but its pretty much standard on non-ISDN E1s |
| 12:15.28 | stintel | it's an isdn e1 |
| 12:15.36 | stintel | so I better call telco and enable it anyway :) |
| 12:15.39 | coppice | where? |
| 12:15.52 | stintel | belgium, kpn |
| 12:16.10 | coppice | europeans should know better than that :-) |
| 12:16.42 | stintel | ahh well forgive me but I am pretty new to this stuff :) |
| 12:17.43 | *** join/#asterisk propellerhead (n=yogurt2u@200.41.65.114) |
| 12:19.49 | coolthreads | kaldemar: works great, thanks |
| 12:20.56 | kaldemar | np |
| 12:22.00 | alexandrekeller | any ideas about this message: rc_avpair_new: unknown attribute 1490026597 ?!?!?! |
| 12:22.25 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 12:26.08 | coppice | stintel: I was referring to the telco rather than you |
| 12:28.41 | stintel | coppice: :) lol I am such a noob :P calling the telco over voip then stopped asterisk :P |
| 12:29.25 | stintel | coppice: but they will enable crc4 and let me know when that is done |
| 12:30.04 | *** join/#asterisk Daboone72 (n=daniel_l@cpc2-hem15-0-0-cust435.lutn.cable.ntl.com) |
| 12:31.27 | *** join/#asterisk zpnd (n=tim@212.175.20.8) |
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| 12:35.12 | ghostknife | When our switchboard operator answers, and transfers a call, and that phone just keeps on ringing, is it possible to add a timeout that takes it back to ringing on the switchboard? |
| 12:36.10 | write_erase | Hi. I don't understant what is best for CISCO phones support : chan_skinny or chan_sccp ? What are the differences ? |
| 12:36.17 | kaldemar | ghostknife: yes, it is. |
| 12:37.04 | kaldemar | ghostknife: add a timeout for the dial command and send the call back to the operator in the following priorities in your dialplan. |
| 12:37.26 | [TK]D-Fender | write_erase: Skinny is being maintained, sccp isn't. |
| 12:38.29 | ghostknife | kaldemar: :/ |
| 12:40.59 | RypPn | http://sourceforge.net/projects/chan-sccp-b last svn update... 2 days ago r352 |
| 12:44.12 | write_erase | Can I use Asterisk Realtime Extention to replace all configuration files or just some of them ? |
| 12:44.58 | *** join/#asterisk zpnd (n=tim@212.175.20.8) |
| 12:45.07 | zpnd | hi again. |
| 12:45.51 | zpnd | i need little help. actually i want to ask something about that i want to do. |
| 12:46.10 | hi365 | seems like you need help asking |
| 12:46.35 | zpnd | =) |
| 12:46.51 | zpnd | can you check out this ? http://forums.digium.com/viewtopic.php?t=65095 |
| 12:46.56 | zpnd | i wrote there. |
| 12:47.23 | *** part/#asterisk alexandrekeller (n=alexandr@201.47.59.175.adsl.gvt.net.br) |
| 12:47.45 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 12:48.16 | [TK]D-Fender | zpnd: What does * have to do with maintaining a phone list? |
| 12:48.22 | viraptor | what might cause nativeaudioformat to not be set? |
| 12:48.45 | [TK]D-Fender | zpnd: * isn'ta phone-book... anything concerning this is completely separate. What could * offer for this at all really? |
| 12:48.57 | viraptor | on a sip channel I don't see any of native, read, write :/ |
| 12:49.00 | *** join/#asterisk PTorres (n=PTorres@200.68.87.146) |
| 12:49.24 | *** join/#asterisk pikachu2000 (n=pikachu2@pikachu.cyberdesigns.co.za) |
| 12:50.57 | Daejeo | i want to dial the queue number 239. how can i write the script? Local/239@from-internal? |
| 12:52.17 | viraptor | ok - let me rephrase that -> is it normal that audionativeformat is not set? if yes -> how do I force it to be, or how do I obtain the codec in dialplan on hangup? |
| 12:53.22 | *** part/#asterisk PTorres (n=PTorres@200.68.87.146) |
| 12:57.09 | [TK]D-Fender | Daejeo: what is a "queu number" exactly? And why should we assume that the context is correct? |
| 12:58.34 | *** join/#asterisk jer (n=jer@unaffiliated/jer) |
| 12:59.58 | zpnd | sorry, i was on the phone. help desk and unrelated customers with hosting is my hobby(!) =P |
| 13:00.49 | zpnd | ok, my english is not fluent but i will try to explain =) sorry for that. |
| 13:02.05 | zpnd | we are using asterisk. can asterisk help to us about caller id identity ? if possible we want to run together asterisk and our cutomer db. |
| 13:02.30 | [TK]D-Fender | zpnd: * already provides you the Caller ID. It can't do any more than that. |
| 13:02.47 | *** join/#asterisk PTorres (n=PTorres@200.68.87.146) |
| 13:03.00 | [TK]D-Fender | zpnd: If you can match that CID with your database you could perhaps push more info to the phone, but this isn't *'s job, this would be a scrip of your won making |
| 13:03.04 | [TK]D-Fender | own* |
| 13:03.04 | zpnd | so asterisk can't match phone number and names on the db ? |
| 13:03.52 | zpnd | get it. is there any document that i can read about this ? |
| 13:03.55 | zpnd | may be samples |
| 13:05.03 | [TK]D-Fender | zpnd: No such thing. What you want to do is custom. It means that you should be a competant programmer |
| 13:06.00 | *** join/#asterisk TOrrIeri (n=torrieri@nelug/crew/torrieri) |
| 13:06.03 | [TK]D-Fender | zpnd: Read up on AGI as that is like what you'll call to try to match up your CID to your external DB (in a script you make), and "core show application sendtext" as this might be useful in pushing any data you do find out to the phone |
| 13:06.07 | zpnd | coding shouln't a problem. just i must read the documents but i don't want to =) |
| 13:07.13 | [TK]D-Fender | zpnd: Too bad for you I guess. If you can't read I hope you're a prodigy with "blind luck". |
| 13:08.39 | zpnd | i can read. just i don't want to. i'm a little tired =) |
| 13:09.25 | [TK]D-Fender | zpnd: You've got your answer. Go sleep onit. |
| 13:09.28 | [TK]D-Fender | on it* |
| 13:09.59 | Katty | reminder to self: do NOT eat sweets for breakfast. |
| 13:10.46 | zpnd | too bad for me, i'm at the office. but if you wish i can logging out =) |
| 13:12.37 | [TK]D-Fender | Katty: Mew. |
| 13:13.32 | DoDaT69 | I am getting a whining sound whenever I call the asterisk system. |
| 13:13.45 | *** join/#asterisk feeds (n=feeds@85-135-235-5.adsl.slovanet.sk) |
| 13:13.45 | DoDaT69 | no errors at all in the logs |
| 13:16.08 | DoDaT69 | hmm |
| 13:16.12 | DoDaT69 | one way audio too |
| 13:17.19 | Katty | tkbeat: mew. |
| 13:17.20 | Katty | oh |
| 13:17.22 | Katty | [TK]D-Fender: mew. |
| 13:18.05 | Katty | [TK]D-Fender: would you like half a tummy ache, with a side of nausea? |
| 13:18.38 | [TK]D-Fender | Katty: No thanks, I slept pretty good last night... must be the drugs :) |
| 13:20.43 | Katty | [TK]D-Fender: i'm glad you got some sleep. |
| 13:21.14 | Katty | [TK]D-Fender: had strange dreams last night. |
| 13:23.03 | Katty | [TK]D-Fender: civil war. |
| 13:23.21 | [TK]D-Fender | Katty: You mean a premonition then ;) |
| 13:23.41 | Katty | i sure hope not |
| 13:24.11 | Katty | dreamt we were stock piling ammunition, canned food, and dog food. |
| 13:24.22 | Katty | and riddick had turned into a kill on command pet |
| 13:24.40 | [TK]D-Fender | Katty: A German Sheppard? Unheard of... |
| 13:24.44 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
| 13:24.46 | Katty | grins |
| 13:24.49 | Katty | no, of course not |
| 13:24.52 | Katty | that'd be absurd. |
| 13:24.54 | Katty | hugs [intra]lanman |
| 13:24.55 | [TK]D-Fender | Katty: thats what you get for picking a breed like that... |
| 13:25.03 | Katty | [TK]D-Fender: one of the reasons i picked the breed, actually |
| 13:25.22 | [intra]lanman | hugs Katty back |
| 13:25.23 | [TK]D-Fender | Katty: 'course it could have beena doberman, pitbull, or some other warm & cuddly breed... |
| 13:25.32 | Katty | [TK]D-Fender: went to the park yesterday around 1... |
| 13:25.49 | Katty | [TK]D-Fender: some random person decided to invite me to a church picnic |
| 13:25.58 | Katty | [TK]D-Fender: i blame the dog being a puppy |
| 13:26.21 | *** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee) |
| 13:27.17 | Katty | [TK]D-Fender: the girl had a horendiously noisy and aggorant daughter. |
| 13:27.24 | Katty | hugs jaytee |
| 13:27.33 | Katty | [TK]D-Fender: she didn't quite know what to think when i told her i was an athiest |
| 13:27.39 | jaytee | mornin Katty *hugs* |
| 13:27.49 | jaytee | Katty, sinuses any better? |
| 13:27.52 | Katty | [TK]D-Fender: and then she was more caught off when i told her no, i did not want to talk about it. |
| 13:28.04 | Katty | jaytee: oh no dear. i've been having crazy sinus pressure for a couple weeks now. :< |
| 13:28.13 | [TK]D-Fender | Katty: And why did you decide to share this little bit with her int he first place? |
| 13:28.13 | Katty | jaytee: starting to worry that maybe i need a CT scan. |
| 13:28.26 | Katty | [TK]D-Fender: because she would not stop asking me which church i went to |
| 13:28.36 | Katty | [TK]D-Fender: and i love reactions. |
| 13:28.37 | jaytee | Katty, are your sinuses congested so you can't breath through your nose? |
| 13:28.41 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.199.65) |
| 13:28.44 | Katty | jaytee: nose is perfectly fine :/ |
| 13:28.46 | Katty | hugs anonymouz666 |
| 13:28.56 | jaytee | hmmm, I'd see a doc |
| 13:29.07 | anonymouz666 | Katty: good morning! |
| 13:29.22 | Katty | jaytee: yeah if it last much longer, i'm going to |
| 13:29.53 | [TK]D-Fender | Katty: For real fun you should learn just enough Arabic to scare people... and for full effect you need the blank stare & conviction to back it ;) |
| 13:29.59 | Katty | jaytee: just don't wanna think about it right now. |
| 13:30.07 | jaytee | first thing I noticed when I came in was Corydon's new nick :-) PeachesBoytoy. lol |
| 13:30.10 | Katty | [TK]D-Fender: hehe |
| 13:30.12 | [TK]D-Fender | Katty: Al-aquab jihad jihad!!! I mean... pass the PB&J! |
| 13:30.35 | Katty | [TK]D-Fender: you are truly terrible. |
| 13:30.39 | Katty | [TK]D-Fender: i love it. |
| 13:31.05 | [TK]D-Fender | Katty: Go get yourself a box of Sudafed.. and not the BS verion... the one with 120mg of straight-up pseudophedrine hydrochloride... |
| 13:31.22 | [TK]D-Fender | Katty: Katty Its the shiznit y0 |
| 13:31.32 | jaytee | it's not really a "blank" stare. it's more of a "I don't see you. You are an infidel. I look right through you." stare |
| 13:31.47 | jaytee | but blank is nice and short. brevity is the soul of wit |
| 13:31.54 | Katty | actually have some--not sure why i haven't taken it yet. maybe i'm afraid it won't affect me and then i'll know it's something horribly scary like a tumor |
| 13:32.03 | [TK]D-Fender | jaytee: Yeah, that's it... |
| 13:32.05 | Katty | is hopeless |
| 13:32.15 | [TK]D-Fender | Katty: 11 steps to go! |
| 13:32.29 | Katty | someone get me a shrink |
| 13:32.57 | coppice | cold water makes most things shrink |
| 13:33.04 | jaytee | Katty, you don't need one of those. Just ask Tom Cruise, he'll tell ya! |
| 13:33.05 | Katty | not quite what i meant. |
| 13:33.12 | asim- | hello all |
| 13:33.15 | Katty | just need scientology in my life eh? |
| 13:33.20 | [TK]D-Fender | Katty: Know how many psychologists it takes to change a lightbulb? |
| 13:33.25 | jaytee | coppice, except cotton fabrics :-) |
| 13:33.33 | Katty | [TK]D-Fender: how many? |
| 13:33.40 | asim- | i'm trying to configure dahdi with asterisk for the dahdi dummy timer. i've got it all to load however musiconhold still chops like crazy. any help? |
| 13:33.45 | jaytee | Katty, Xenu rocks baby!!!!! |
| 13:33.52 | [TK]D-Fender | Katty: Just one, but the lightbulb has to really, really want to change :) |
| 13:34.09 | jaytee | C'mon! Drink the Kool-Aid. Whaddaya got to lose? |
| 13:34.24 | coppice | xenu's not unix |
| 13:34.24 | [TK]D-Fender | asim-: What are you using for MoH currently? |
| 13:34.36 | jaytee | sometimes a cigar is JUST a cigar |
| 13:34.49 | [TK]D-Fender | jaytee: Or we're really happy to see you ;) |
| 13:34.51 | asim- | moh in sip and iax |
| 13:34.56 | jaytee | hahaha |
| 13:35.01 | asim- | just default moh for when people are on hold |
| 13:35.04 | [TK]D-Fender | asim-: I mean the SOUREC |
| 13:35.07 | [TK]D-Fender | SOURCE* |
| 13:35.15 | asim- | hmm, wav or gsm i think |
| 13:35.21 | Katty | what's Xenu? |
| 13:35.25 | [TK]D-Fender | asim-: Less think, more look |
| 13:35.33 | [TK]D-Fender | Katty: Scientology <- |
| 13:35.38 | Katty | oh |
| 13:35.40 | asim- | ah sorry |
| 13:35.41 | Katty | jbot: xenu |
| 13:35.42 | jbot | His name was Xenu, he used renegades. Various misleading data by means of circuits was placed in the implants. the Arch enemy of Scientology, or xenu.net (cool site! aka Operation Clambake) |
| 13:35.43 | [TK]D-Fender | Katty: Do follow along now! |
| 13:35.50 | asim- | by source you mean like hardware? |
| 13:36.01 | Katty | [TK]D-Fender: too busy dreaming about civil war, apparently |
| 13:36.09 | [TK]D-Fender | asim-: I asked you exactly what you were using for MoH. This is not a complicated question... |
| 13:36.26 | Katty | and now i'm worrying about my head again |
| 13:36.33 | Katty | dangit |
| 13:36.35 | asim- | well you make it seem confusing cause it seems self explanatory, moh for music on hold. |
| 13:37.18 | *** join/#asterisk af_ (n=getsmart@88-149-230-89.dynamic.ngi.it) |
| 13:37.44 | asim- | default source |
| 13:37.59 | [TK]D-Fender | asim-: What is playing? navite MoH? Some external player app? What is it using as the source? What format of files? Where did they come from? |
| 13:38.00 | Katty | jaytee: you can't talk to the hypochondriac about her heath |
| 13:38.07 | [TK]D-Fender | asim-: you know... DETAILS <--- |
| 13:38.09 | Katty | jaytee: it just causes unneeded stress |
| 13:38.10 | asim- | lol |
| 13:38.11 | asim- | sorry |
| 13:38.23 | asim- | well i tried playing mp3 which was choppy, then wav, then gsm |
| 13:38.38 | asim- | then just defaulted back to native files in /var/lib/asterisk/moh |
| 13:38.40 | [TK]D-Fender | asim-: Don't jsut say the type, I asked for the PRIGIN of the files. |
| 13:38.42 | asim- | everything is choppy |
| 13:38.44 | [TK]D-Fender | ORIGIN* |
| 13:38.54 | asim- | asterisk-sounds i guess |
| 13:38.59 | [TK]D-Fender | asim-: MP3's don't fall out of thin ait |
| 13:39.06 | [TK]D-Fender | asim-: Stop guessing and go look. |
| 13:39.11 | [TK]D-Fender | air* |
| 13:39.23 | [TK]D-Fender | damn... typing getting choppy... how do I fix that in *? |
| 13:39.29 | Katty | Jen ai Marre!!! |
| 13:39.33 | [TK]D-Fender | chuckles |
| 13:40.00 | [TK]D-Fender | Katty: She's a hottie, isn't she? |
| 13:40.07 | asim- | asterisk provides a default moh directory with sound files right? thats where its playing them from |
| 13:40.11 | Katty | yes. i'm also fed up. |
| 13:40.14 | [TK]D-Fender | Katty: And that's "J'en" |
| 13:40.18 | Katty | oh? |
| 13:40.18 | Katty | k |
| 13:40.18 | asim- | otherwise i created my own mp3, wav and gsm which all had the same issue |
| 13:40.51 | [TK]D-Fender | asim-: MP3 must be non VBR, 128kbit suggested, no ID3 tags, etc |
| 13:41.07 | [TK]D-Fender | asim-: And it'd be nice to know the environment your testing this in... |
| 13:41.39 | *** join/#asterisk n3hxs (n=HAMming@63.68.135.4) |
| 13:41.40 | asim- | linux 2.6.18-92, dell server, 1gb ram, 2ghz, onboard sounds, no digium hardware |
| 13:42.07 | asim- | basically moh cuts in an out even with default moh files provided with asterisk 1.6 |
| 13:42.31 | [TK]D-Fender | asim-: What are you listening to it on? |
| 13:42.49 | asim- | a voip phone, an n810, xlite on a pc, xlite on a laptop |
| 13:42.54 | asim- | all the same issue |
| 13:43.03 | [TK]D-Fender | asim-: you keep mentioning half, or less with every answer |
| 13:43.04 | asim- | running through gig switches in the office |
| 13:43.13 | [TK]D-Fender | asim-: Networked how? etc |
| 13:43.13 | asim- | sorry i just thought you'd have some suggestions |
| 13:43.41 | [TK]D-Fender | asim-: If you want your mechanic to fix your car it helps when you properly describe the problem. |
| 13:43.42 | asim- | cisco routers, dell 48 port switches, through patch panel, then out to netgear 8port switches |
| 13:44.13 | [TK]D-Fender | asim-: What codec is the call in? |
| 13:44.28 | asim- | ulaw or alaw i think. |
| 13:45.05 | asim- | in sip.conf i do: allow=alaw;ulaw |
| 13:45.14 | asim- | will it default to something else in codecs.conf ? |
| 13:46.24 | ronr | is it possible to create a dailplan to intercept a Dial? what I want is to have an incoming call Dial(A&B&C) and allow D&E&F to call some number to 'pick up' |
| 13:46.50 | [TK]D-Fender | asim-: Please go look at this and let us know when you have collected the complete picture... |
| 13:47.02 | *** join/#asterisk didz_ (n=dsad@201.19.199.65) |
| 13:47.10 | WimpMan | ronr: Correct. It's called pickup. |
| 13:47.16 | asim- | i'm trying to get some assistance here. i'm setting up asterisk for our office solution with the majority done, and i dont have all the info |
| 13:47.17 | [TK]D-Fender | ronr: go lookup "pigup groups" on the WIKI. |
| 13:47.20 | asim- | which is why i'm here |
| 13:47.28 | asim- | i thought someone would be able to help |
| 13:47.40 | asim- | cause google has a scatter of info and not all of it helps |
| 13:47.51 | [TK]D-Fender | asim-: You seem uncertain of what codec your calls is taking place in, you can't give a definitive answer on exactly what format your MoH is in. |
| 13:47.56 | ronr | WimpMan: [TK]D-Fender: tx |
| 13:47.58 | ronr | *thx |
| 13:47.59 | asim- | its like, maybe someone woule be able to point me in a direction |
| 13:48.23 | asim- | .... |
| 13:48.32 | asim- | really unhelpful. thanks |
| 13:48.32 | [TK]D-Fender | asim-: Go in your MoH folder, empty it out and load up ULAW format MoH off the Asterisk HTTP server |
| 13:48.48 | asim- | thanks |
| 13:48.49 | asim- | will do that |
| 13:50.29 | asim- | as for codec. which is the best formats to be in? |
| 13:51.37 | Katty | hmm. google clains my weird left side of head thing is indeed most likely sinuses. lots of people claim it can go on for months, and that sometimes psuedophed wont' phase it. |
| 13:51.40 | [TK]D-Fender | asim-: Its best if your MoH is in the same codec |
| 13:52.00 | asim- | ah i see |
| 13:52.10 | asim- | downloaded some now. will try it out thanks |
| 13:53.08 | [TK]D-Fender | Katty: Right now you're fearing nothing can help you. That's called "paranoia" and borders on "hypochondria" |
| 13:53.08 | [TK]D-Fender | Katty: Go do something about it. |
| 13:53.08 | [TK]D-Fender | Katty: Drugzzzzzzzzzzz |
| 13:53.08 | Katty | [TK]D-Fender: you're right. |
| 13:53.11 | Katty | [TK]D-Fender: i'm not going to calm down until a guy in a white coat says, yep sinus infection |
| 13:53.20 | Katty | calls doctor |
| 13:53.26 | [TK]D-Fender | Katty: Sudafed & Ibuprophen... the only 2 drugs I need... |
| 13:53.34 | asim- | and i assume the same goes for voicemail playback, menus, etc? |
| 13:53.39 | Katty | [TK]D-Fender: i want my problem fixed. not masked |
| 13:53.47 | [TK]D-Fender | Katty: And coloured mucous discharge? |
| 13:54.04 | Katty | [TK]D-Fender: no. no stuffiness either |
| 13:54.16 | [TK]D-Fender | Katty: So what is it exactly? |
| 13:54.31 | Katty | [TK]D-Fender: just pain on the left side of my head when i cough |
| 13:55.29 | jaytee | Hey!! I thought we weren't supposed to talk about that? |
| 13:55.44 | Katty | we're not |
| 13:57.34 | Katty | goes to doctor Right Now |
| 13:58.19 | jaytee | but while the topic is open, I had pain that was behind my right ear to the back of the head whenever I coughed and it felt like it was right below the scalp and not under the bone. Lasted about 2 weeks and went away. |
| 13:58.38 | *** join/#asterisk tcseke (n=chatzill@217.20.134.239) |
| 13:59.31 | jaytee | I called the clinic and they said it would require lots of tests and at least 3 to 4 hundred to misdiagnose my problem. |
| 14:00.46 | *** join/#asterisk gbr_ (n=gbr@200.103.96.98) |
| 14:02.59 | *** join/#asterisk Zeeek (n=Zeeek@pdpc/supporter/active/Zeeek) |
| 14:03.08 | jaytee | other than the extensions.sample.ael file and the WIKI can anyone point me to other resources for AEL2 scripting? |
| 14:04.59 | *** join/#asterisk mog (n=mog@nat/digium/x-19973f26fcf44cc4) |
| 14:04.59 | *** mode/#asterisk [+o mog] by ChanServ |
| 14:10.33 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 14:11.32 | [TK]D-Fender | jaytee: You can always nag codefreeze-lap directly ;) |
| 14:12.44 | jaytee | [TK]D-Fender, I'm just looking for some better documentation than the sample file. The Wiki is sparse and I don't trust half the stuff on there to be current or always accurate. |
| 14:13.02 | *** join/#asterisk tobias (n=tobias@user-0c8hj2l.cable.mindspring.com) |
| 14:13.22 | [TK]D-Fender | jaytee: Check your doc folder. |
| 14:13.34 | *** part/#asterisk telcohitman (n=telcohit@tn-76-5-147-175.dhcp.embarqhsd.net) |
| 14:13.53 | *** join/#asterisk apocn (n=apo@unaffiliated/apocn) |
| 14:14.59 | jaytee | [TK]D-Fender, already looked in there but I just looked again. Don't see anything regarding AEL2 in the tarball for 1.6.0 |
| 14:15.28 | [TK]D-Fender | jaytee: then wait for him to come around... |
| 14:15.58 | jaytee | [TK]D-Fender, you mean codefreeze-lap? |
| 14:16.06 | jaytee | he's logged in atm. |
| 14:16.07 | apocn | Hello all, I have 2 queues with timeouts, if a caller waits too long in the first is automatically sent to the second and so on. My problem is that on my statistics appear like the caller is hanging up/leaving and my efficiency is poor in both queues. (using queuemetrics). How can I solve that? |
| 14:16.26 | jaytee | codefreeze-lap, PING |
| 14:16.33 | *** join/#asterisk riddlebox (n=adfad@75-128-170-26.static.stls.mo.charter.com) |
| 14:21.59 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 14:21.59 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 14:22.27 | *** join/#asterisk ManxPower (n=manxpowe@201.sub-70-222-31.myvzw.com) |
| 14:26.04 | [TK]D-Fender | apocn: If you're dumping them from queues, you can't |
| 14:26.29 | [TK]D-Fender | apocn: Unless you do something to clean up the logs. Queumetrics can't be smart on your behalf for this |
| 14:33.21 | apocn | ok |
| 14:33.28 | *** join/#asterisk Hamburglr (n=hamburgl@c-71-199-226-249.hsd1.fl.comcast.net) |
| 14:33.42 | apocn | what if I specify the same queue name (for the queuemetrics reports) for both asterisk queues? |
| 14:34.03 | [TK]D-Fender | apocn: they are separate calls. |
| 14:34.08 | apocn | right |
| 14:34.11 | [TK]D-Fender | apocn: they will not be seen any other way |
| 14:34.34 | [TK]D-Fender | apocn: You're baicallys crewed unless you do some clean-up manipulation of the queue log. |
| 14:34.45 | apocn | ok, I'll do that |
| 14:35.04 | apocn | thanks [TK]D-Fender |
| 14:35.08 | *** join/#asterisk fskrotzki (n=fskrotzk@host198.textwise.com) |
| 14:35.48 | codefreeze-lap | jaytee: pong! what can I do to you, er **for** you? |
| 14:35.52 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 14:38.55 | jaytee | codefreeze-lap, I was wondering where I might find more detailed info or documentation on AEL2. I've looked in the docs folder and read the extensions.sample.ael in 1.6.0 |
| 14:39.31 | ManxPower | jaytee: I suspect you have read all the docs on AEL2 8-)| |
| 14:40.01 | lmadsen | codefreeze-lap: morning! |
| 14:40.21 | seanbright | voip-info has a page |
| 14:40.58 | jaytee | codefreeze-lap, thanks for the info |
| 14:41.52 | codefreeze-lap | jaytee: Well, I'm infamously bad about reading what I wrote, so I can't really say much about it, but yes, seanbright is correct, my best doc is on http://voip-info.org/wiki/view/Asterisk+AEL2 |
| 14:42.10 | ManxPower | codefreeze-lap: you wrote that page? |
| 14:42.40 | ManxPower | If so we may have discovered the only not-wrong page on the Wiki! |
| 14:42.44 | jaytee | codefreeze-lap and seanbright, thanks! I'll give that a read. |
| 14:43.06 | codefreeze-lap | ManxPower: Yep. I did! :) |
| 14:43.31 | ManxPower | codefreeze-lap: any reason you did not include that in the official Asterisk docs |
| 14:43.57 | ManxPower | *shiver* *while* It's cold outside. |
| 14:44.04 | *** join/#asterisk ddunavant (n=David@75.145.240.14) |
| 14:46.03 | codefreeze-lap | Actually, there is a similar doc in the Asterisk tree, but prefer the wiki, because I can keep enhancement requests, bug histories, etc, there. |
| 14:46.38 | *** part/#asterisk UnixDawg (n=UnixDawg@155.129.204.68.cfl.res.rr.com) |
| 14:48.00 | jaytee | bookmarks the page while grumbling about the devs deprecating AgentCallbackLogin |
| 14:48.36 | lmadsen | jaytee: that conversation is over a year old |
| 14:49.21 | *** join/#asterisk sp00k3y (n=sp00k3y@wsip-98-190-136-194.ph.ph.cox.net) |
| 14:49.33 | jaytee | lmadsen, really? well sorry but I wasn't involved in the conversation. I doubt if you or anyone else wants or even cares about my 2 cents opinion but thanks for the info :-) |
| 14:50.40 | *** part/#asterisk pikachu2000 (n=pikachu2@pikachu.cyberdesigns.co.za) |
| 14:51.12 | sp00k3y | this is prolly a dumb question but can u put more than one FXO card on a single server? |
| 14:51.36 | lmadsen | sp00k3y: no point really when you can get up to 24 FXO channels on a single card |
| 14:51.43 | lmadsen | (depending on the card and module configuration) |
| 14:51.47 | jaytee | the only dumb questions are the ones people are too afraid to ask for fear of looking ignorant |
| 14:51.53 | lmadsen | if you're talking about X101P... then don't bother |
| 14:52.19 | sp00k3y | well i mean, I have a client who has 6 analog lines that he wants to use on his server |
| 14:52.39 | sp00k3y | not a PRI |
| 14:52.51 | ManxPower | sp00k3y: You can put multiple cards in the same server. 1 or 2 cards should work. More than that may or may not work well. |
| 14:53.05 | sp00k3y | i see |
| 14:53.16 | sp00k3y | is there anything special I have to define in zaptel? |
| 14:53.24 | lmadsen | sp00k3y: use a TDM2400P with 2 FXO modules, then you have up to 8 lines on a single card |
| 14:53.33 | ManxPower | sp00k3y: each card generates 8,000 interrupts/second. |
| 14:53.53 | sp00k3y | ah ok |
| 14:55.53 | [TK]D-Fender | sp00k3y: keep in mind that card is huge... |
| 14:56.00 | sp00k3y | yes im seeing that now lol |
| 14:56.17 | [TK]D-Fender | sp00k3y: TDM800P might be a better choice, or a Sangoma equivalent |
| 14:57.16 | lmadsen | oh right, I totally forgot there was an 8 port version |
| 14:57.23 | lmadsen | doesn't use hardware usually |
| 14:57.32 | ManxPower | You could go with a TDM400P w/4 modules, that will give you 4 lines on the server, you can add some VoIP service to handle the overflow. |
| 14:57.33 | sp00k3y | yeah that looks like it fits more to what he wants |
| 14:57.43 | *** join/#asterisk ibm2 (n=Administ@196.203.192.179) |
| 14:57.47 | *** join/#asterisk jameswf (n=james@dsl093-157-131.phx1.dsl.speakeasy.net) |
| 14:57.57 | ManxPower | lmadsen: I don't normally use analog 8-| |
| 14:58.08 | lmadsen | that's good |
| 14:58.13 | lmadsen | I just use SIP |
| 14:58.44 | sp00k3y | yeah we tried to get him to go for a PRI but he already has those lines in place |
| 14:58.56 | *** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler) |
| 14:59.00 | sp00k3y | i think i'll reccomend the TDM 800 |
| 14:59.07 | jameswf | morning kids |
| 14:59.17 | ManxPower | sp00k3y: Analog is never even close to as reliable as PRI |
| 14:59.25 | sp00k3y | i know |
| 14:59.29 | sp00k3y | we tried to explain that to him |
| 14:59.48 | jameswf | analog is okay, not as fast but okay |
| 14:59.49 | sp00k3y | he was very adament about staying with his analog lines |
| 14:59.52 | ManxPower | Your telco will normally he happy to upgrade your analog lines to PRI with no install cost if you push for it. |
| 15:00.11 | *** join/#asterisk russellb (n=russell@asterisk/digium-open-source-team-lead/russellb) |
| 15:00.12 | *** mode/#asterisk [+o russellb] by ChanServ |
| 15:00.17 | ManxPower | sp00k3y: That's OK. He'll change his mind or fire you once he sees just how much problem analog is. |
| 15:00.22 | sp00k3y | LOL |
| 15:01.06 | ManxPower | Put your PRI recommendation in writing so you have something to show when there are problems |
| 15:01.07 | sp00k3y | i'm going to have him sign a scope of work before we even start, so if he hates it it's his own fault and i have proof |
| 15:01.26 | sp00k3y | it will have the PRI recommendation in it as well |
| 15:01.29 | ManxPower | sp00k3y: VERY smart. |
| 15:01.30 | jameswf | I have no issues with nalog other then the one issue you get with all pstn lines, the fact they are controlled by the telco |
| 15:02.17 | ManxPower | jameswf: Glare will be a problem, as well as issues with CPC (far end disconnect indication) and slow dialing. |
| 15:02.30 | jameswf | why? |
| 15:03.23 | sp00k3y | thanks for the suggestions everyone |
| 15:03.24 | ManxPower | why? Well Glare happens for the same reason glare happens on other pbxs with analog. |
| 15:03.34 | jameswf | I am guessing you are talking about bell canada. glare is fixed in dialing order and CPC is fixed by proper hardware |
| 15:03.50 | ManxPower | CPC may or may not be a problem, depending on your telco. |
| 15:04.07 | ManxPower | Glare can be REDUCED with dialing order, not resolved. |
| 15:04.21 | ManxPower | CPC is fixed by the telco side, not the Asterisk side. |
| 15:04.58 | jameswf | pri has just as many problems if not more like i said the telco is the factor that makes it all a pain in the explicitive |
| 15:05.05 | *** join/#asterisk Math` (n=mrene@64.254.252.151) |
| 15:05.20 | ManxPower | Lets say you are dialing 11 digits, asterisk collects the digits from you, then dials out the analog line, transmitting the DTMF for all 11 digits. Now if your DTMF time is 300ms,. you would wait about 4 seconds for each call |
| 15:05.27 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 15:05.27 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 15:05.54 | ibm2 | hello, can anyone tell me how to activate standard jabber in my asterisk |
| 15:07.02 | magronez | is away: fui almocar |
| 15:08.03 | *** part/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 15:15.25 | *** join/#asterisk DukeOfURL (n=chatzill@introspect.com) |
| 15:15.32 | *** part/#asterisk ibm2 (n=Administ@196.203.192.179) |
| 15:15.41 | *** join/#asterisk kfife (n=Miranda@home.chicagoventure.com) |
| 15:15.51 | *** join/#asterisk ibm2 (n=Administ@196.203.192.179) |
| 15:16.18 | DukeOfURL | is this the place to aks about DAHDI? |
| 15:16.26 | DukeOfURL | ask |
| 15:18.14 | jaytee | yes, both DAHDI and MOHMI |
| 15:18.47 | tzanger | hahahaha |
| 15:19.03 | *** join/#asterisk casix (n=casix@pbxedifici.adamvozip.es) |
| 15:19.06 | casix | hello |
| 15:19.15 | kfife | welcome |
| 15:24.20 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
| 15:26.34 | Katty | [TK]D-Fender: i have survived a trip to the doctor. |
| 15:26.42 | Katty | jaytee: you'll be happy to know i don't have a brain tumor. |
| 15:26.51 | Katty | jaytee: we can now talk about the pain on the left side of the head without inducing butterflies. |
| 15:26.55 | Zeeek | {{{Katty}}} |
| 15:26.56 | tzanger | ?it's naaht ah tooomah" |
| 15:27.02 | Katty | hugs Zeeek |
| 15:27.04 | jaytee | Katty, very happy but then I didn't think you had a brain tumor. |
| 15:27.19 | Katty | jaytee: well hypochondriacs go for the worst |
| 15:27.22 | Zeeek | Katty: but presumably you do have a brain? |
| 15:27.35 | Katty | Zeeek: no, no brain either. |
| 15:27.38 | Katty | Zeeek: which is handy |
| 15:27.48 | Zeeek | easier to fall asleep at night, anyway |
| 15:27.49 | coppice | I think hypochondriacs have something seriously wrong with them |
| 15:27.59 | tzanger | coppice: :-) |
| 15:28.03 | jaytee | coppice, lol |
| 15:28.46 | Katty | i have to agree. |
| 15:29.01 | Katty | there is no reason whatsoever that i should have mild panic attacks everytime something doesn't feel quite right. |
| 15:29.20 | [TK]D-Fender | Katty: Sure there is |
| 15:29.24 | Katty | it is an unacceptable response, which is at times overwhelming to deal with. |
| 15:29.25 | casix | it is possible to change the codec in the middle of a conversation? for example the * recieve an incoming call, it plays a locutions and after that * redirects the call to an extension. I want asterisk in the middle of the conversation but working on bridge mode with this call. Is possible to change the codec of the incoming call to match the codec of the phone that recieve that call? |
| 15:29.40 | coppice | *mild* panic attacks are for wimps. |
| 15:29.41 | ManxPower | Katty: Once you accept the fact that you will die, and have no idea when, you can be much more zen about it all. |
| 15:29.46 | [TK]D-Fender | Katty: This has been accumulating with your for a LONG time now. |
| 15:29.56 | Katty | ManxPower: hmm yes. the dying bit doesn't bother me. the suffering bit does. |
| 15:30.07 | ManxPower | Katty: Me too. |
| 15:30.12 | [TK]D-Fender | Katty: And sounds rather psychosomatic |
| 15:30.13 | Katty | [TK]D-Fender: yeah, ever since that UTI in the middle of the night that scared the crap out of me :/ |
| 15:30.52 | ManxPower | Katty: I figure I'll just off myself if suffering becomes too much. |
| 15:31.00 | Katty | coppice: yeah i just get the blood pressure problems, racing heart, shaking, and stomach hurts. |
| 15:31.15 | Katty | coppice: not the full 9 yards, as it were. |
| 15:31.32 | ManxPower | Fortunaty, that should not be required for 10 - 20 years |
| 15:31.35 | Katty | [TK]D-Fender: the doc put me on anti-biotics for a sinus infection and an ear infection. |
| 15:31.36 | coppice | arsenic can cure that |
| 15:31.45 | [TK]D-Fender | coppice: Or bleach... |
| 15:31.51 | [TK]D-Fender | coppice: Cures AIDS you know... |
| 15:31.51 | Katty | [TK]D-Fender: he claims i will be perfectly fine in about 2 weeks. |
| 15:31.52 | jaytee | what's a UTI? |
| 15:31.59 | Katty | jaytee: urinary track infection. |
| 15:32.04 | Katty | jaytee: hurts like hell. mostly harmless. |
| 15:32.07 | coppice | or granite workshops, but they're a bit slower |
| 15:32.13 | jameswf | phone sex will give you hearing aids |
| 15:32.22 | kfife | Quick question: I have a dialplan event that needs to trigger an outbound call (open a channel, dial some dtmf tones, play message, hang up). I'm getting 'stuck' because everything in extensions.conf is predicated on the idea of receiving a call, or stitching together two calls together. I know I can create a .call file, but how do I trigger it from the dialpan? |
| 15:32.31 | Katty | jaytee: if left alone, it can cause a kidney infection which is slightly more serious |
| 15:32.39 | ManxPower | kfife: see .call files |
| 15:32.40 | jaytee | yep |
| 15:32.44 | kfife | Also it seems a bit kludgy to need drop a file into the filesystem just to trigger a call. Is there a better way to trigger this from the dialplan, or concept that I'm not getting? |
| 15:32.57 | ManxPower | documented (poorly, where else but the doc directory of the asterisk source code. |
| 15:32.59 | Katty | jaytee: usually painful painful stomach cramping, and over time a pain on the left side |
| 15:33.10 | Katty | jaytee: which will make you worry about all sorts of things |
| 15:33.22 | Katty | jaytee: within about 48 hours tho, i felt perfectly fine again |
| 15:33.31 | Katty | jaytee: the pain was horrid |
| 15:33.32 | ManxPower | kfife: there are 2 ways to initiate calls in an automated way. .call files and the Manager Interface. |
| 15:33.40 | Katty | jaytee: i'd put it above the wisdom teeth coming out. |
| 15:33.57 | Katty | jaytee: and far above the appendix recovery |
| 15:34.08 | *** join/#asterisk sw (n=sw@unaffiliated/sw) |
| 15:34.08 | kfife | ManxPower: Thanks for the tip. What's the best way to trigger these events from the dialplan? |
| 15:34.30 | coppice | Katty: its sounds like you get excellent value for money from your health insurance |
| 15:34.31 | Katty | jaytee: easily fixed by anti-biotics tho. |
| 15:34.34 | ManxPower | kfife: the best way is a .call file or manager interface using AGI |
| 15:34.41 | Katty | coppice: oh yes. |
| 15:34.43 | jaytee | or sulfa based drugs |
| 15:34.46 | Katty | coppice: i have a specialist for everything >.< |
| 15:35.18 | Katty | coppice: appendix is the only major problem tho |
| 15:35.33 | rednode | does anything know if Asterisk can be integrated with NICE??? |
| 15:35.34 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 15:35.36 | [TK]D-Fender | Katty: Your appenix will be then end of you ;) |
| 15:35.42 | [TK]D-Fender | the* |
| 15:35.47 | coppice | I had a pre-med for that once, but skipped the operation. the pre-med just seemed enough |
| 15:35.47 | Zeeek | *end |
| 15:35.48 | [TK]D-Fender | addendix* |
| 15:35.54 | [TK]D-Fender | dammit, can't type for beans today... |
| 15:35.59 | Zeeek | *nix |
| 15:36.03 | rednode | does anything know if Asterisk can be integrated with NICE??? |
| 15:36.05 | rednode | does anything know if Asterisk can be integrated with NICE??? |
| 15:36.24 | Zeeek | bad echo |
| 15:36.53 | ManxPower | rednode: you don't want to /bin/nice the Asterisk application. the -r option will do that for you if you need it. |
| 15:37.47 | ManxPower | Zeeek: I call it Internet Tourtettes where people shout out the same questions multiple times pissing off everyone in the channel. |
| 15:38.10 | Zeeek | that was probably just a case of up arrow recall |
| 15:38.22 | ManxPower | on the plus side it's obvious that english is not rednode's native language. |
| 15:38.23 | Zeeek | call call call |
| 15:38.58 | kfife | lol |
| 15:39.02 | kfife | ManxPower: This is really helpful. Correct me if I'm wrong here but the existence of the .call file triggers execution. |
| 15:39.18 | coppice | ManxPower: but I thought Tourette's was saying what you *think* without filtering. |
| 15:39.22 | ManxPower | kfife: yes. |
| 15:39.37 | Zeeek | kfife: the convenience of the .call file is brought on only if you're good enuf at some language to generate them easily |
| 15:39.52 | PTorres | hi everyone, would this the be right place to ask some questions about CAS timing in zaptel 1.4.10 ?? |
| 15:40.12 | *** join/#asterisk Deeewayne (n=Deeewayn@nat/digium/x-7b5e29ba9802b00b) |
| 15:40.12 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 15:40.13 | ManxPower | coppice: that's why I added "internet" in the front of it. Maybe it should be called "Tourette's 2.0" instead? |
| 15:40.49 | ManxPower | kfife: if you set the timestamp of the .call file to be in the future then asterisk will wait for that date/time before processing the file. |
| 15:40.52 | coppice | this sounds a bit like "cooking sashimi" |
| 15:41.08 | [TK]D-Fender | coppice: :) |
| 15:41.13 | ManxPower | PTorres: CAS timing? I think you mean "T-1 sync source" |
| 15:41.56 | ManxPower | coppice: I know! The condition should be known as "Tourette's! Tourette's!" |
| 15:42.22 | PTorres | ManxPower:I have a E1 , with R2 protocol, but we are having issues at the "cas level" |
| 15:42.46 | coppice | what issues? |
| 15:43.05 | *** join/#asterisk jicksta (n=jicksta@c-24-6-87-187.hsd1.ca.comcast.net) |
| 15:43.16 | jameswf | tzafrir_laptop: goofd job on the piaf forums, it seems like your pulling scark teeth with tweasers, most would have given up... |
| 15:43.18 | kfife | ManxPower: that's great! |
| 15:43.35 | jameswf | *good, *shark |
| 15:43.47 | ManxPower | PTorres: "CAS" doesn't really mean the same thing for E-1s as for T-1s |
| 15:44.10 | *** join/#asterisk Firass-VC22 (n=firass@rza.vikcomm.wwu.edu) |
| 15:44.22 | ManxPower | PTorres: now DESCRIBE the problem and what you have done to try to fix it. |
| 15:44.42 | PTorres | the telco is telling us we are sending a distant multiframe alarm |
| 15:45.07 | PTorres | as in the bit 6 of the 16th timeslot |
| 15:45.10 | ManxPower | coppice: any ideas on that? |
| 15:45.21 | anonymouz666 | PTorres: are you using DGV? |
| 15:45.28 | PTorres | dgv ? |
| 15:45.39 | anonymouz666 | nevermind then |
| 15:45.42 | ManxPower | PTorres: what R2 library are you using? |
| 15:45.43 | coppice | PTorres: does your end show any alarms? |
| 15:45.51 | jameswf | PTorres: are you using the right country and idle state |
| 15:46.00 | coppice | this is nothing to do with R2. its below that |
| 15:46.12 | PTorres | yes, we have many trunks like this, but this is the first one with this telco |
| 15:46.20 | ManxPower | coppice: odd that it is on channel 16 isn't it? |
| 15:46.38 | coppice | no. channel 16 is where an E1 puts all the CAS signals |
| 15:46.50 | jameswf | configure 16 as clear it isnt used in r2 right? |
| 15:47.19 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
| 15:47.32 | PTorres | abcd bits are correct , mfcr2 with chan_unicall , we have to test openr2 yet, but this looks like a zaptel thing |
| 15:47.41 | coppice | if he has installed multiple R2 E1s, I assume he will get something as basic as that right |
| 15:48.06 | coppice | are you using the same cards and drivers as usual? |
| 15:48.31 | ManxPower | coppice: Ah. I always thought "CAS" for an E-1 is silly. It's not Channel Associated afterall |
| 15:48.50 | coppice | its very much channel associated |
| 15:48.51 | PTorres | yes.. digium boards |
| 15:49.17 | coppice | I don't know when the digium drivers will raise that alarm bit. |
| 15:49.40 | PTorres | I asked them to send some information or trace or something but I don't think they will :( |
| 15:50.14 | PTorres | on our end all looks green, we can even receive incoming calls !! |
| 15:51.25 | coppice | I seem to remember someone recently complaining about some of the spare bits in the CAS channel being wrong |
| 15:53.47 | casix | it is possible to change the codec in the middle of a conversation? for example the * recieve an incoming call, it plays a locutions and after that * redirects the call to an extension. I want asterisk in the middle of the conversation but working on bridge mode with this call. Is possible to change the codec of the incoming call to match the codec of the phone that recieve that call? |
| 15:54.21 | coppice | PTorres you said incoming calls work. what about outgoing? surely if one works, the other works |
| 15:55.00 | PTorres | outgoing calls are terminated by the telco because of this 'alarm' |
| 15:55.05 | *** join/#asterisk LeddyHM (i=leddy@polar.artica.net) |
| 15:56.22 | ManxPower | casix: the answer is no |
| 15:57.05 | casix | ManxPower: :( thanks |
| 15:57.17 | *** join/#asterisk CunningPike (n=arodgers@204.239.8.157) |
| 15:59.19 | LeddyHM | exten => 2,1,Goto(local-internal,100,1 what's the proper way to configure it to try 100, then 200 if no answer then go back to 100's vm if 200 doesn't answer? |
| 15:59.25 | *** join/#asterisk iratik (n=itariki@209.248.216.146.nw.nuvox.net) |
| 15:59.40 | iratik | Have any of you had any luck with that "gizmo5 backdoor" thing? |
| 16:00.44 | trelane | ? |
| 16:01.39 | iratik | Well if you haven't heard of it ... it sounds really good ...i guess they have peering agreements with all the cell phone providers except for at&t .... which means that they can offer free calls to all cell phones ... can't get it to work worth a crap though |
| 16:01.52 | trelane | cool |
| 16:01.57 | trelane | that's a shame that it doesn't work though |
| 16:02.04 | PTorres | I asked another telco about this and they had no clue ( we have many e1/r2 working ok with them ) :D |
| 16:02.14 | iratik | was hoping someone else here has heard of it ..... so i could get a test number |
| 16:02.24 | iratik | i mean... i get the same message on every single number i try |
| 16:02.41 | iratik | "this number is not available for free calling... bla bla bla.." |
| 16:03.25 | coppice | PTorres: well, the bit they are complaining about is an alarm to indicate an upstream E1 link has problems, but I don't know if the zaptel or dahdi drivers have any way to control it. |
| 16:03.43 | outtolunc | try any 'comcast triple play' user phone number |
| 16:04.59 | iratik | outtolunc: they are 866 number when i googled that |
| 16:05.24 | outtolunc | i am talking the 'home users phone number' not comcasts.. sheesh |
| 16:05.48 | *** join/#asterisk zamolxes (n=zamolxes@82.76.1.167) |
| 16:06.21 | iratik | Times like this i wish i had a list of my friends organized by CLEC |
| 16:07.19 | coppice | PTorres: if you have Digium cards, I think you should ask them. |
| 16:07.21 | zamolxes | hello. I get this warning when using call files in /var/spool/asterisk/outgoing. WARNING[2538] pbx_spool.c: Unable to set utime on /var/spool/asterisk/outgoing/1.call: Operation not permitted . The call works ok, but how can I eliminate the warning? asterisk has full rights to the file. |
| 16:09.21 | PTorres | coppice : I see... |
| 16:10.23 | anonymouz666 | PTorres: what telco? |
| 16:10.55 | PTorres | <PROTECTED> |
| 16:11.19 | coppice | PTorres: maybe I am thinking of wrong alarm. A quick check says that bit informs the far end that you are getting too many multiframe alignment errors from them. the distant alarm is something different. I wonder if you really are getting a lot of errors from them |
| 16:12.04 | PTorres | <PROTECTED> |
| 16:12.38 | PTorres | coppice: cat /proc/zaptel/1 shows nothing ... zttool is fine too... what else can I check ? |
| 16:14.44 | coppice | I forget how Digium map the T1 coloured alarms to the E1 alarms, but I think you would expect a yellow alarm, if you are sending that bit to the far end |
| 16:15.16 | PTorres | right... me too , but its all green :D |
| 16:16.00 | *** join/#asterisk mayo_ (n=mayo@a221-smpafs01.blockb-142.stargate.ca) |
| 16:16.27 | hi365 | does the new polycom call-pickup feature work with asterisk? |
| 16:17.41 | *** join/#asterisk propellerhead (n=yogurt2u@host38.190-136-115.telecom.net.ar) |
| 16:18.46 | *** join/#asterisk telecos (n=sergio@153.166.219.87.dynamic.jazztel.es) |
| 16:19.55 | sw | i'm putting a call file in /var/spool/asterisk/outgoing and the call is not initiated.. do i need to add a specific module.. i'm working on a pretty slimmed down asterisk |
| 16:20.09 | Qwell | sw: pbx_spool.so |
| 16:20.23 | *** join/#asterisk cesar_CR (n=cesar@200.91.75.66) |
| 16:21.46 | sw | Qwell, thx :D |
| 16:24.52 | *** join/#asterisk russellb (n=russell@asterisk/digium-open-source-team-lead/russellb) |
| 16:24.52 | *** mode/#asterisk [+o russellb] by ChanServ |
| 16:36.56 | *** join/#asterisk RMod (n=nicolasj@unaffiliated/rmod) |
| 16:36.59 | *** join/#asterisk Defraz (n=T0tal@63.228.246.250) |
| 16:40.55 | *** join/#asterisk Ericounet (n=Ericoune@ACaen-151-1-78-94.w86-215.abo.wanadoo.fr) |
| 16:46.33 | *** join/#asterisk dlynes (n=dlynes@S01060016b68219f1.vs.shawcable.net) |
| 16:47.48 | ManxPower | zamolxes: you have a permissions problem. You are not root when you set the future timestamp |
| 16:47.57 | dlynes | Is dahdi-linux-complete 2.0.0+2.0.0(?) compatible with both Asterisk 1.4 and Asterisk 1.6? |
| 16:48.01 | ManxPower | maybe you need to set ctime, rather than utime? |
| 16:48.19 | ManxPower | dlynes: yes! (for recent versions of 1.4 and all versions of 1.6) |
| 16:48.36 | casix | we have many small virtual pbx with asterisk. There is any way to share the licences of g729 with all installations? that when asterisk need a g729 licence connect to a "server licences" or something to catch a licence and when hangup free it |
| 16:48.39 | dlynes | cool...so no more hardware drivers being tied to specific versions of asterisk, then? |
| 16:48.52 | ManxPower | sw: yes you need the pbx_spool module, I think. |
| 16:48.59 | [TK]D-Fender | casix: No. |
| 16:49.09 | *** part/#asterisk ibm2 (n=Administ@196.203.192.179) |
| 16:49.14 | ManxPower | dlynes: Huh? There were almost never hardware versions tied to Asterisk. |
| 16:49.36 | ManxPower | the only reason it's in 1.4 is because is under an agreement to stop using the zaptel name |
| 16:50.03 | dlynes | ManxPower: oh....I was told some time ago that I should download the version of zaptel released at the same time as a particular version of asterisk if I wanted to be sure they would work together |
| 16:50.04 | casix | [TK]D-Fender: thanks |
| 16:50.32 | ManxPower | dlynes: that is and continues to be correct, EXCEPT for late 1.4 versions and DAHDI |
| 16:50.44 | dlynes | ManxPower: ok...cool |
| 16:50.51 | dlynes | ManxPower: makes it so much easier that way, then |
| 16:51.00 | dlynes | ManxPower: thanks for clearing that up |
| 16:51.11 | ManxPower | dlynes: Zaptel/DAHDI is not undergoing many changes as it has matured in recent years. |
| 16:52.50 | zamolxes | ManxPower: of course i'm not, asterisk runs as asterisk. looking at http://www.asterisk.org/doxygen/1.2/pbx__spool_8c.html the code is if (utime(o->fn, &tbuf)) . why would it need root? this doesn't always happen, jsut somtimes |
| 16:53.13 | ManxPower | Most of the "You need a recent zaptel for your recent asterisk" was because of API changes and there are not many API changes happening antmore. |
| 16:53.24 | Qwell | jameswf: nice quote about the rumors |
| 16:54.33 | ManxPower | zamolxes: Is the /var/spool/asterisk owned by the "asterisk" user? |
| 16:54.47 | [TK]D-Fender | Qwell: When does * 1.2 maintenance drop off completely? |
| 16:54.59 | Qwell | [TK]D-Fender: for security? dunno |
| 16:55.24 | ManxPower | [TK]D-Fender: when 1.4 becomes stable. 8-| |
| 16:55.24 | [TK]D-Fender | Qwell: Just wondering how that impacts Zaptel/DAHDI in the case of the next release |
| 16:55.46 | Qwell | there only releases that will be made for 1.2 will be security |
| 16:55.55 | Qwell | so, it own't |
| 16:56.01 | anonymouz666 | ManxPower: did you see the transnexus testing ? |
| 16:56.11 | [TK]D-Fender | Qwell: Just a question of perpetuating the verboten name... |
| 16:56.14 | ManxPower | anonymouz666: is that a place where drag queens gather? |
| 16:57.01 | anonymouz666 | ManxPower: they used the asterisk 1.4, the results were good, IMHO. |
| 16:57.56 | ManxPower | anonymouz666: now 1.4.0 |
| 16:57.59 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 16:58.03 | ManxPower | NOT 1.4.0 |
| 16:59.39 | mort_gib | Hi, I have the same old problem with ONE installation that keeps dropping calls, like once a week. I need to sort it out, so any ideas |
| 17:00.07 | ManxPower | mort_gib: remove busydetect and callprogress from you zap config |
| 17:00.27 | mort_gib | Yeah, hang on |
| 17:01.26 | anonymouz666 | ManxPower: 1.4.21.1 |
| 17:01.27 | *** join/#asterisk TenJack (n=chatzill@c-71-197-166-145.hsd1.or.comcast.net) |
| 17:01.34 | anonymouz666 | or 21.2 don't remember |
| 17:01.56 | TenJack | whats the deal with these complie errors when installing app_swift? |
| 17:02.03 | TenJack | anyone experienced this? |
| 17:02.09 | mort_gib | ManxPower: Calls are done via BRI, Sangoma A500 card |
| 17:02.19 | *** join/#asterisk fskrotzki (n=fskrotzk@host.textwise.com) |
| 17:02.20 | ManxPower | TenJack: you seem to think this is a common problem |
| 17:02.26 | mort_gib | So I shouldn't need to touch ZAP files |
| 17:03.01 | ManxPower | mort_gib: Can't help you then |
| 17:03.37 | mort_gib | Damn, it's a bit frustrating! |
| 17:03.45 | TenJack | ii get this error: error: swift/swift.h: No such file or directory |
| 17:03.57 | TenJack | then: [app_swift.o] Error 1 |
| 17:04.04 | ManxPower | TenJack: install swift-devel |
| 17:04.15 | ManxPower | or whatever you need to do to install Cepstral |
| 17:05.30 | zamolxes | ManxPower: drwxrwx--- 2 asterisk asterisk 4096 2008-10-27 17:57 outgoing |
| 17:05.38 | *** join/#asterisk drumkilla (n=russell@asterisk/digium-open-source-team-lead/russellb) |
| 17:05.38 | *** mode/#asterisk [+o drumkilla] by ChanServ |
| 17:05.42 | ManxPower | (swift is the engine for Cepstral |
| 17:05.43 | zamolxes | ManxPower: anyway, it doesn't happen to all the call files, just once in a while, it's rather suspect |
| 17:06.01 | ManxPower | zamolxes: I suspect sometimes you are not user asterisk when copying the files |
| 17:06.09 | *** join/#asterisk TedC (n=cabeen@milli.chem.ucsb.edu) |
| 17:06.17 | *** part/#asterisk TedC (n=cabeen@milli.chem.ucsb.edu) |
| 17:06.42 | ManxPower | zamolxes: Or maybe you are creating the file in /var/spool/outgoing rather than creating it elsewhere on the partition and then mv'ing it to the correct directory. |
| 17:06.43 | zamolxes | ManxPower: clearly i am not. i'm www-data. but i make sure they're 666 .. and move them in outgoing |
| 17:07.00 | zamolxes | i'm moving them (using the rename syscall , it's atomic :) |
| 17:07.26 | zamolxes | hmm. will debug that possibility further though, you may be right, I could be missing something :) |
| 17:07.33 | ManxPower | *nod* Something is happening that you are not aware of. |
| 17:07.54 | ManxPower | I have never in my 6 years of using Asterisk seen a randomly not working .call files |
| 17:08.12 | jeev | in my 6months of asterisk, i dunno what the hell a .call file is |
| 17:08.14 | zamolxes | well the thing is they work great, i just get the warning. didn't see any missing funcs |
| 17:08.27 | zamolxes | s/funcs/calls/ |
| 17:08.40 | zamolxes | heh |
| 17:08.41 | ManxPower | But the fact that you understand that rename is an atomic operation indicates to me you'll be able to figure it out. |
| 17:08.42 | zamolxes | jbot++ |
| 17:09.14 | zamolxes | thanks for the confidence :) |
| 17:10.33 | ManxPower | zamolxes: Most people using Asterisk have no clue as to what an atomic system call is. |
| 17:10.44 | ManxPower | Heck most of them can't even figure out Linux. |
| 17:10.52 | ManxPower | Especially the newbies. |
| 17:11.06 | ManxPower | What? You mean I need a space after "cd"???? |
| 17:11.55 | Katty | dear lord |
| 17:11.56 | *** join/#asterisk kfife (n=Miranda@home.chicagoventure.com) |
| 17:11.58 | Katty | someone get me a pillow stat |
| 17:12.06 | [TK]D-Fender | ManxPower: sounds nukular to me, hyuk! |
| 17:12.13 | Katty | they must have given me tranqs not anti-biotics |
| 17:12.54 | TenJack | ManxPower: how do i install asterisk-devel? and what is that exactly? its not installed with the regular version of asterisk? |
| 17:13.20 | [TK]D-Fender | Katty: Nothing like an expensive and timely misdiagnosis to put a dent in your day... |
| 17:13.47 | *** join/#asterisk RypPn_OuT (i=TuMbL@rosscom.demon.co.uk) |
| 17:16.06 | Katty | [TK]D-Fender: well i did take some pseudophed awhile back |
| 17:16.18 | Katty | [TK]D-Fender: i'm going to blame it instead. |
| 17:16.55 | [TK]D-Fender | Katty: thats what scapegoats are for.... |
| 17:17.19 | TenJack | anyone know what asterisk-devel is and/or how to get it? it seems you can do a yum install, but is there any other way? |
| 17:17.33 | Qwell | TenJack: what distro is this? |
| 17:17.34 | [TK]D-Fender | TenJack: Where'd you get that name from? |
| 17:18.02 | TenJack | ManxPower was saying i needed it, then i saw it in a forum comment |
| 17:18.08 | [TK]D-Fender | TenJack: [13:04]<ManxPower>TenJack: install swift-devel <- Someone can't read |
| 17:18.15 | TenJack | im having trouble installing app_swift |
| 17:18.16 | Katty | randomly passes out |
| 17:18.29 | nido | revives Katty |
| 17:18.30 | TenJack | hehe |
| 17:18.38 | nido | katty -r; reload |
| 17:18.58 | TenJack | for some reason im getting errors when installing app_swift 1.4.2 on mac os x |
| 17:19.12 | Katty | i'm thinkin i'm gonna have to nap |
| 17:20.32 | [TK]D-Fender | TenJack: Oh now you're not even using linux... |
| 17:20.43 | [TK]D-Fender | TenJack: And you're wonding if compile issues are common? |
| 17:21.09 | TenJack | yea, well can you be of any assistance? |
| 17:21.55 | [TK]D-Fender | TenJack: Certainly not on OS/X |
| 17:22.46 | TenJack | so i dont know much about linux at all,what is the best way to run it? |
| 17:23.28 | ManxPower | TenJack: NO! I said you need swift-devel |
| 17:23.37 | Katty | poor TenJack |
| 17:23.40 | Katty | i've been there. |
| 17:23.47 | ManxPower | Swift is a 3rd party PAY text-to-voice solution |
| 17:23.47 | Katty | it's hard not knowing what on earth you're doing :< |
| 17:24.00 | ManxPower | so either buy it and install it or stop trying to use it. |
| 17:24.01 | Qwell | jbot: tell Katty about roflmao |
| 17:24.18 | Katty | Qwell: that is my ringtone. |
| 17:25.20 | Katty | Qwell: would like a sinus and/or ear infection? |
| 17:25.24 | TenJack | ManxPower: i know oh so you have to pay for app_swift and the cepstral voices? |
| 17:25.30 | Qwell | Katty: ear please |
| 17:25.56 | Katty | Qwell: <3 |
| 17:26.54 | *** join/#asterisk stewbaby (n=stewart@ip-217-204-65-78.easynet.co.uk) |
| 17:27.14 | StephenF[W] | Anyone know why my phones presence will change when im on interoffice calls, but it will not change when I dial outside? |
| 17:27.26 | Katty | Qwell: these pills are huge. |
| 17:27.33 | StephenF[W] | show hints, also does not change status when I call an outside number |
| 17:27.46 | jeev | http://www.purplehat.org/few/future_ex_wife.jpg |
| 17:27.48 | jeev | is that real? |
| 17:28.29 | *** join/#asterisk magic_hat (n=geoffdou@h-66-167-66-201.chcgilgm.dynamic.covad.net) |
| 17:28.49 | Katty | jeev: looks like it. |
| 17:28.53 | TenJack | ManxPower: so i think swift and cepstral are the same, i am trying to get them working with adhearsion which is an api for ruby on rails, but ive read that i need something called app_swift to bridge these two. is this correct? |
| 17:28.58 | Katty | jeev: no obvious contact ring |
| 17:29.58 | jeev | weird |
| 17:30.11 | Katty | not really |
| 17:30.31 | Katty | the pattern is a bit odd |
| 17:30.38 | Katty | but the color is perfectly normal |
| 17:31.28 | jeev | hh |
| 17:31.35 | *** join/#asterisk blitzrage (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 17:31.35 | *** mode/#asterisk [+o blitzrage] by ChanServ |
| 17:31.39 | Katty | blitzrage: ohai |
| 17:31.42 | blitzrage | HI! |
| 17:31.47 | Katty | blitzrage: Qwell took my ear infection |
| 17:31.52 | Katty | blitzrage: would you like my sinus infection? |
| 17:31.53 | blitzrage | good! |
| 17:31.55 | Katty | blitzrage: selling very cheap |
| 17:31.55 | blitzrage | I would not |
| 17:31.58 | Katty | :< |
| 17:31.59 | Katty | k |
| 17:32.03 | blitzrage | I already have a bit of a sinus thing myself |
| 17:32.06 | Katty | :<< |
| 17:32.14 | Katty | applies steam to blitzrage |
| 17:32.15 | Qwell | blitzrage: a little more won't hurt thenn |
| 17:32.16 | blitzrage | and the g/f has terrible sinuses, so I'm all sinused out |
| 17:32.22 | Katty | )_= |
| 17:34.41 | blitzrage | thx for the offer though |
| 17:34.43 | blitzrage | but I must decline |
| 17:34.52 | Katty | you're just too modest |
| 17:35.59 | blitzrage | I've never heard that before :) |
| 17:36.04 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 17:36.23 | Katty | first time for everything |
| 17:36.41 | *** part/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 17:37.23 | blitzrage | after installing my transition piece between the hardwood and the tile last night, the kitchen/living room transition looks fantastic! |
| 17:37.44 | TenJack | would you guys say ubuntu is the best way to use linux? |
| 17:38.09 | Katty | personally, i like debian |
| 17:38.14 | Katty | but you'll get into distro wars here... |
| 17:38.22 | Katty | everyone has their own favorite. go with the one you're most comfortable with |
| 17:38.31 | [TK]D-Fender | dons his asbestos digs.... |
| 17:38.42 | blitzrage | ~bestquestions |
| 17:38.45 | blitzrage | ~thebest |
| 17:38.49 | blitzrage | ~thebestquestions |
| 17:38.50 | jbot | methinks thebestquestions is Whenever you ask a "what is the best..." or "who is the best..." type questions, you're asking for trouble, and possibly may be called a troll. These types of questions do not have answers. Your best bet is to rephrase the question as, "What kind of experience do people have with..." or "Who has experience with...". |
| 17:38.52 | blitzrage | aha! |
| 17:39.03 | [TK]D-Fender | blitzrage: 15th times the charm! |
| 17:39.10 | blitzrage | totally |
| 17:39.13 | [TK]D-Fender | high-5's blitzrage |
| 17:39.25 | blitzrage | high-8's [TK]D-Fender |
| 17:39.34 | Katty | scandalious. |
| 17:39.43 | [TK]D-Fender | blitzrage: Dude... lay off the plutonium! |
| 17:40.04 | blitzrage | I got a few extra fingers installed so it was easier to work in base-8 counting |
| 17:41.14 | *** join/#asterisk musse- (n=musse@static-212.214.40.123.addr.tdcsong.se) |
| 17:41.36 | Katty | [TK]D-Fender: what is sit in french? |
| 17:41.40 | Katty | [TK]D-Fender: to sit. |
| 17:41.45 | Katty | [TK]D-Fender: Dog, Sit! |
| 17:42.44 | [TK]D-Fender | Katty: Assier-toi |
| 17:43.19 | Katty | [TK]D-Fender: and how do you pronounce that? |
| 17:43.40 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 17:43.43 | [TK]D-Fender | Katty: ass-ee-ay t-wah |
| 17:43.52 | [TK]D-Fender | +/- |
| 17:44.10 | Katty | ay and is eh |
| 17:44.11 | Katty | eh? |
| 17:44.24 | Katty | hey minus the h |
| 17:44.32 | [TK]D-Fender | Katty: Yup.. you're ready to emmigrate now ;) |
| 17:44.44 | Katty | k |
| 17:44.47 | [TK]D-Fender | Katty: Cannuckianland welcomes yoU! |
| 17:45.08 | Katty | mccain gets elect i might head north |
| 17:45.12 | Katty | civil war doesn't sound pleasant |
| 17:46.17 | [TK]D-Fender | Katty: So you're in one of those "Pro-American" parts of the country? |
| 17:46.48 | Katty | nods |
| 17:46.54 | Katty | mccain's on every bumpersticker |
| 17:46.58 | Katty | along with Support our Troops |
| 17:47.00 | Katty | American Pride |
| 17:47.03 | Katty | and pro-life stickers |
| 17:47.05 | Nugget | W'04! |
| 17:47.20 | Nugget | I can't believe people still have those stickers |
| 17:47.31 | Katty | welcome to Misery |
| 17:47.31 | Nugget | they're all over down here |
| 17:47.48 | [TK]D-Fender | Katty: Is that how's its actually pronounced? ;) |
| 17:48.00 | *** join/#asterisk RMod_ (n=nicolasj@unaffiliated/rmod) |
| 17:49.45 | ManxPower | Cthulhu 2008! Why settle for the lesser of two evils? |
| 17:50.32 | nido | lol |
| 17:54.31 | kfife | Can someone explain the concept behind "priority and priority+1" when using the G dial flag? Why would you ever want to transfer the call legs to x and x+1 instead of simply x and y? |
| 17:54.49 | Katty | [TK]D-Fender: if you have a funny accent. |
| 17:54.57 | Katty | [TK]D-Fender: or if you say it real fast ;) |
| 17:55.24 | *** join/#asterisk tkbeat (n=tk@p54B9485B.dip.t-dialin.net) |
| 17:55.29 | [TK]D-Fender | Katty: From "state of mind" to "state in the union" in one hurried slur! |
| 17:56.42 | kfife | is the idea simply: ..100,Goto(this) ..101,Goto(that)? It seems like a workaround for what should be simply x and y. |
| 17:56.50 | *** join/#asterisk edoceo (n=edoceo@c-71-197-244-147.hsd1.or.comcast.net) |
| 17:58.10 | edoceo | How would I configure to ring extA four times then extB four times then if no answer goto VM for ExtA? |
| 17:58.11 | *** join/#asterisk Kate-o (n=vajda1ka@unaffiliated/kateo/x-334924) |
| 17:58.13 | kfife | I suspect that I'm missing a concept |
| 17:58.21 | Katty | [TK]D-Fender: sorry, brain too fuzzy right now. please leave a message after the beep and try again later. BEEP. |
| 17:59.03 | [TK]D-Fender | edoceo: Dial 1 for 15s, the other for 15s, and then go to VM. Priorities 1-3. |
| 17:59.40 | [TK]D-Fender | Katty: "You have 5000 new messages. 5000 marked 'urgent' First message....." |
| 18:00.06 | [TK]D-Fender | kfife: No, that's pretty much it |
| 18:00.13 | Katty | exten => s,1,Wakeup ; exten => s,2,Eat ; exten => s,3,Sleep exten => s,4,(goto,s,1) |
| 18:00.26 | Katty | or goto(s,1) |
| 18:00.53 | [TK]D-Fender | or just Goto(1) |
| 18:00.56 | Katty | hmm. i need a wait in there |
| 18:01.02 | Katty | wait(999999999999999999999999999999( |
| 18:01.03 | Katty | ) |
| 18:01.09 | [TK]D-Fender | Katty: Wait comes bundled with the apps ;) |
| 18:01.18 | Katty | k |
| 18:01.27 | *** join/#asterisk SQLDarkly (n=nospam@p10-162.dsl.ecentral.com) |
| 18:01.35 | [TK]D-Fender | Katty: As to other essential services like "Bathroom()" tec, to keep you from ass-ploding ;) |
| 18:01.38 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 18:01.57 | kfife | [TK]D-Fender: :-) Too bad. I was hoping there'd be a cool concept wedged in there. |
| 18:01.59 | Katty | assier toy ploding |
| 18:02.01 | Katty | toi |
| 18:02.04 | Katty | can't type today. |
| 18:02.09 | SQLDarkly | How is MeetMe() in 1.6 vs 1.4.22 I heard there was a bug of somesort that broke MeetMe() in 1.6 |
| 18:02.39 | [TK]D-Fender | SQLDarkly: I'm sorry, could you be a little more vague? |
| 18:02.40 | Nugget | WORKSFORME |
| 18:03.06 | Katty | wished isymphony worked |
| 18:03.29 | Katty | then i could upgrade for real |
| 18:04.03 | SQLDarkly | I suppose you want more specifics. I appreciate sarcasm heh. Well the bug had to do with timing and DHADI. I dont remember the specifics. That is why I am asking as people do. I am curious if anyone has had any issue with MeetMe() without any telephony hardware on 1.6 |
| 18:04.35 | *** join/#asterisk implicit (n=bayan@unaffiliated/implicit) |
| 18:04.35 | *** join/#asterisk grEvenX (n=even@cB78D5AC1.dhcp.bluecom.no) |
| 18:05.54 | [TK]D-Fender | SQLDarkly: 1.6 series has 2 "releases" and there is CVS. Care to narrow this down some more? And perhaps tell us what exact problem you are experiencing.... |
| 18:06.13 | Katty | well i sure got a problem |
| 18:06.16 | Katty | i'm sleepy |
| 18:06.18 | Katty | this is just No Good |
| 18:06.56 | kfife | [TK]D-Fender: It would have expected it to be designed like syntax of D([called][:calling]) in other words, something like: G([called][:calling]), where called/calling=context^exten^pri. That could easily be designed to allow backward compatibility. |
| 18:07.00 | *** join/#asterisk RMod (n=nicolasj@unaffiliated/rmod) |
| 18:07.23 | kfife | In other words if no called party was speciifed, it would model the legacy functionality. |
| 18:07.33 | [TK]D-Fender | kfife: .... huh? |
| 18:07.51 | kfife | I would have expeced G() to work like D() |
| 18:08.14 | *** join/#asterisk ddunavant (n=David@75.145.240.14) |
| 18:08.22 | [TK]D-Fender | kfife: Ah, I misinterpreted your last question |
| 18:08.28 | [TK]D-Fender | kfife>Can someone explain the concept behind "priority and priority+1" when using the G dial flag? Why would you ever want to transfer the call legs to x and x+1 instead of simply x and y? |
| 18:08.46 | kfife | Except that the called / calling parametrs could be formatted with carets to specify context, extension, priority |
| 18:08.50 | [TK]D-Fender | kfife: with "G" dialplan continues on right away. |
| 18:09.27 | [TK]D-Fender | kfife: oops. thats "g", not "G" |
| 18:09.36 | kfife | Right. The current implementation requires two GOTO's to do anythint productive |
| 18:09.40 | kfife | That' |
| 18:09.52 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 18:10.15 | [TK]D-Fender | kfife: You can specify the explicit place to jumpt to... no need for a "goto" |
| 18:10.38 | kfife | But it's the n, and n+1 that throws me. |
| 18:10.40 | SQLDarkly | I am not experiencing a problem. I am simply trying to discover a potential hazard of upgrading. I am interested in some of the new candy that is provided however I am working with a cluster of servers that is using 1.4.22 only for conferences so upgrading needs to be kept in the back of my mind especially if a future version improves on what I am using the cluster for. So I am not here looking for any solution to a given problem or issue, b |
| 18:10.43 | kfife | I want each leg to be different. |
| 18:10.52 | [TK]D-Fender | kfife: yeah, I see your point now... |
| 18:11.08 | [TK]D-Fender | kfife: What on earch are you looking at this function for in the firswt place? ;) |
| 18:11.18 | kfife | :-) |
| 18:12.01 | [TK]D-Fender | SQLDarkly: Worry about problems you have... you are LOOKING for trouble without any details, which of couse since you don't HAVE a problem and something to show us means you feel more than comfortable going on wild-goose chases on your whim |
| 18:12.09 | kfife | I'm just looking for a very simple way to: on the end of a call, dial a number, play some DTMF, say some stuff, play some more DTMF, and hangup. |
| 18:12.25 | Daejeo | can anyone point me to moh? nice music for free? |
| 18:12.28 | kfife | Say some stuff as in Playback() |
| 18:12.36 | [TK]D-Fender | kfife: use "h" and issue a call-file |
| 18:12.40 | SQLDarkly | Ill agree with that ;) I am known for loving to tinker. |
| 18:13.02 | [TK]D-Fender | SQLDarkly: Fine, just don't drag us into wsting time along with yout like that... |
| 18:13.18 | SQLDarkly | Theory is never a waste my friend |
| 18:13.25 | kfife | [TK]D-Fender: Excellent. Let me think about that |
| 18:13.41 | SQLDarkly | The hypothetical should always be discussed and prodded at |
| 18:13.54 | [TK]D-Fender | SQLDarkly: In theory I could travel back and recover the last 10 minutes I spent caring about this.... but then again, I flunked the theory... |
| 18:13.54 | kfife | [TK]D-Fender: The called party is a machine. |
| 18:14.01 | *** join/#asterisk dennisharrison (n=dennisha@97.80.39.152) |
| 18:14.26 | [TK]D-Fender | is currently debugging chan_fluxcapacitor.so |
| 18:14.29 | kfife | [TK]D-Fender: so the h would be lost on its sensibilities :-) |
| 18:14.37 | dennisharrison | anyone ever heard of ztdummy causing a system to not boot due to random kernel panics at runlevel 3? |
| 18:14.38 | *** join/#asterisk Hadi- (n=AlanS@cmr-208-124-195-107.cr.net.cable.rogers.com) |
| 18:14.41 | Daejeo | can anyone point me to moh? nice music for free? |
| 18:14.47 | Hadi- | hello.. anyone here used avantfax? |
| 18:14.55 | [TK]D-Fender | Daejeo: * comes wish some already |
| 18:15.01 | kfife | [TK]D-Fender: Thanks for your help by the way. |
| 18:15.11 | Katty | [TK]D-Fender: mmm, fluxcapacitors. |
| 18:15.13 | [TK]D-Fender | Daejeo: and there are links on the WIKI for this. |
| 18:15.25 | kfife | [TK]D-Fender: How would you trigger leaving a call file in the dialplan? |
| 18:15.40 | [TK]D-Fender | kfife: "h"<---- |
| 18:16.12 | kfife | [TK]D-Fender: in other words: h extension triggers what application/function? |
| 18:16.22 | *** join/#asterisk subdolus (n=subdolus@subby.afraid.org) |
| 18:16.29 | [TK]D-Fender | kfife: DeadAGI, System, etc |
| 18:17.23 | kfife | It sounds like that's a more 'proper' way to do this than to the kludge with the G() flag |
| 18:17.46 | *** join/#asterisk StephenF[W] (n=none@198.144.201.106) |
| 18:18.44 | kfife | [TK]D-Fender: Still, it seems a bit odd that there's no application that is the equivalent of dropping a call file. Kind of like Dial except not designed specifically to bridge channels. |
| 18:19.10 | Katty | jbot: ode to joy |
| 18:19.44 | Katty | jbot: :< |
| 18:19.45 | jbot | < is probably redirection of stdin to a program |
| 18:19.52 | Katty | jbot: odetojoy |
| 18:20.03 | Katty | jbot: habanera? |
| 18:20.03 | jbot | methinks habanera is bork Bork BORK Bork! http://www.youtube.com/watch?v=EDFgtFXfnv0 |
| 18:20.10 | Katty | jbot: but not ode to joy? |
| 18:20.26 | Katty | jbot: you make me sad. |
| 18:21.07 | [TK]D-Fender | kfife: System(/usr/sbin/asterisk -rx "originate......") |
| 18:21.33 | [TK]D-Fender | Katty: You keep asking jbot the same thing and expecting different results... |
| 18:21.41 | Hadi- | hello.. anyone here used avantfax? |
| 18:22.19 | kfife | [TK]D-Fender: Thanks for the tip. From the 10,000' view, do you think there is any merit to making G have explicit called:calling parametrs in some future release of asterisk? |
| 18:22.20 | [TK]D-Fender | Hadi-: No more than there were when you asked about 5 minutes ago... |
| 18:22.24 | subdolus | jbot: sup? |
| 18:22.24 | jbot | Yo subdolus, how's it going eh? |
| 18:22.30 | subdolus | :D |
| 18:22.43 | [TK]D-Fender | kfife: Didn't we have this discussion last week? |
| 18:22.46 | Katty | [TK]D-Fender: and your surprised, why? |
| 18:22.51 | Katty | jbot: Danny Boy? |
| 18:22.52 | jbot | i heard danny boy is http://www.youtube.com/watch?v=OCbuRA_D3KU -- *sniffle* Oh Danny Boy!!! *sniffle* |
| 18:23.05 | subdolus | haha |
| 18:23.13 | kfife | [TK]D-Fender: wasn't me. |
| 18:23.47 | [TK]D-Fender | kfife: I see all sorts of good ideas for changes... |
| 18:24.04 | *** join/#asterisk RMod_ (n=nicolasj@unaffiliated/rmod) |
| 18:24.37 | kfife | [TK]D-Fender: is there an official digium 'suggestion box' do we just hope that the digium guys are listening? |
| 18:24.51 | kfife | It's interesteing that this very issue came up last week. |
| 18:26.34 | [TK]D-Fender | kfife: thats what the bounty list is for on the WIKI... or you can't ask if anyone is interested in picking it up in #asterisk-dev perhaps... unsure about there though |
| 18:27.34 | jjshoe | s/can't/can/ |
| 18:31.52 | *** part/#asterisk Math` (n=mrene@64.254.252.151) |
| 18:32.08 | kfife | [TK]D-Fender: Great info. Thanks. Is using the System () funciton 'expensive' in terms of memory etc. For example does it open up a shell and keep it open for the duration of the call or does it close as soon as the command is given to Ast? |
| 18:32.51 | [TK]D-Fender | kfife: All you're going to do is issue a quick command... very light, esp compared to AGI |
| 18:32.58 | Katty | [TK]D-Fender: have you seen star wars according to a 3 year old? |
| 18:33.07 | [TK]D-Fender | Katty: I believe so |
| 18:33.47 | Katty | [TK]D-Fender: don't talk back to darth vader. he'll getcha |
| 18:33.57 | gcbirzan | Hm. I have these two asterisk machines, one of which is older than me, most likely, and at one point today, the very old one stopped calling the other one over IAX. It just says "Called rndsoft/742", and I get nothing on the other side... ANy idea what I can check? |
| 18:33.58 | kfife | [TK]D-Fender: Beautiful. That seems like absolutely the best way to do this. I really appreciate you sharing your expertise!! |
| 18:36.01 | *** part/#asterisk Hadi- (n=AlanS@cmr-208-124-195-107.cr.net.cable.rogers.com) |
| 18:37.49 | [TK]D-Fender | gcbirzan: Address changed perhaps? Firewall issue? |
| 18:37.50 | *** join/#asterisk IsUp (n=nocturne@unaffiliated/isup) |
| 18:37.56 | IsUp | hey ya |
| 18:38.55 | IsUp | i've just download Asterisk 1.6, how can i use SS7 with 1.6? |
| 18:39.03 | IsUp | should i download libss7 from trunk? |
| 18:41.54 | scooby2 | Is there any way to let a caller sit in queue if an agent is available but if all agents are busy for > 60 seconds transfer them. I know I can use the queue timeout to jump out after 60 seconds but that would not let a caller wait if an agent is idle. |
| 18:43.38 | *** join/#asterisk mchou (n=mchou@unaffiliated/mchou) |
| 18:44.15 | *** join/#asterisk jdnWEST (n=jdn@mx1.westparkcom.net) |
| 18:44.50 | IsUp | any ideas? |
| 18:44.54 | jameswf | for those of you who actualy use linux outside of asterisk : http://lameduck.codeweavers.com/press/20081027/ |
| 18:45.37 | gcbirzan | [TK]D-Fender: Doubtful. It 'sometimes' works, though I haven't been able to see a pattern. And from the new one, calling the old one works perfectly all the time. |
| 18:46.09 | jdnWEST | Anyone know if OSLEC is better than the echo cancelation for the sangoma cards, or do I need to splurge on the cards? |
| 18:46.28 | Katty | jdnWEST: i buy the cards. |
| 18:46.29 | *** join/#asterisk nikko (n=nikko@69.57.49.100) |
| 18:47.11 | jdnWEST | I'm looking to bridge 2 PRI's from one state to another over 3 connections (BGP dark magic) for a emergency call center, so quality is actually a concern.. |
| 18:47.17 | IsUp | hey Katty, ive just get two A101 :D going to test them |
| 18:47.31 | Katty | IsUp: nice (= |
| 18:47.34 | jaytee | they'll get my Octasic chip when they pry it from my cold dead fingers |
| 18:47.50 | jdnWEST | jaytee: Octasic? |
| 18:48.09 | jaytee | the ec chip on Digium cards |
| 18:48.30 | gcbirzan | [TK]D-Fender: And tcpdump doesn't really help me. I just don't see any traffic coming my way, from what I can figure. Though, I might be wrong, since I'm calling from the new one and trying to call back. |
| 18:48.35 | jdnWEST | Who makes better voip cards, Digium or Sangoma? |
| 18:49.02 | IsUp | Katty, do you have any idea about libss7 trunk? i am using chan_ss7 but i am going to switch libss7 if i can setup. |
| 18:49.07 | jaytee | never tried Sangoma, but Digium makes better T1 cards than Nortel, or at least my users say the line "sounds" clearer now |
| 18:49.10 | Katty | IsUp: sorry, no :< |
| 18:49.10 | IsUp | Sangoma always the best |
| 18:49.15 | scooby2 | Sangoma |
| 18:49.21 | [TK]D-Fender | jaytee: Sangoma was the first to use Otasic actually... by well over a year |
| 18:49.33 | Katty | jdnWEST: gotta agree with the mob. sangoma. |
| 18:50.29 | IsUp | i am using all Sangoma stuff. a104 a102 a108... sangoma is great |
| 18:50.37 | jaytee | [TK]D-Fender, didn't know what they used really. Looked at pricing, read stuff on the wiki and decided I liked the configuration scheme for Digium T1 cards than I did for Sangoma. |
| 18:50.59 | IsUp | Katty: btw thank youy |
| 18:51.01 | IsUp | *you |
| 18:51.10 | [TK]D-Fender | jaytee: Indeed less to configure for what little that means. |
| 18:52.29 | jaytee | [TK]D-Fender, I'm not arguing about which is better. I really have no personal basis for comparison. Lots of people in here seem to like Sangoma products. I don't dislike them. I just went in one direction and so far I'm satisfied with the results. |
| 18:53.14 | [TK]D-Fender | jaytee: I'm glad that you are satisfied with your solution then :) |
| 18:53.29 | jameswf | wooohooo fanboy war |
| 18:53.53 | gcbirzan | Ah. Okay. I found something. When it works, it says -- Executing Dial("IAX2/172.16.32.42:1062-12", "IAX2/rndsoft/742|30") in new stack, when it does it says "IAX2/rndsoftblahblah". Though, I can see an "-- Accepting AUTHENTICATED call from 172.16.32.42:" before the call that doesn't work |
| 18:54.06 | jaytee | jameswf, no it's not a fanboy war. |
| 18:54.09 | jdnWEST | hmmm, something tells me that sangoma vs. Dig... isn't going to be apple vs. windows any time soon |
| 18:54.45 | jameswf | jdnWEST: you dont have to use either :) |
| 18:54.51 | *** join/#asterisk oush (n=rdn@dehghany.demon.co.uk) |
| 18:54.57 | jdnWEST | Is anyone using cards Without the ECHO cancelation and just using OSLEC? |
| 18:54.59 | oush | any VoIP experts that know cisco around? |
| 18:55.03 | StephenF[W] | What kind of desktop integration do you guys have running on your asterisk installs? |
| 18:55.10 | IsUp | i am using without E/C |
| 18:55.19 | jameswf | EC on T1 is rarely needed |
| 18:55.20 | jdnWEST | oush: there are a couple in the #cisco chan. |
| 18:55.37 | StephenF[W] | I've seen thinkgs like HUDlite out there, what other things are asterisk peoples doin? |
| 18:56.34 | jameswf | StephenF[W]: asterisk installs shouldnt have desltops |
| 18:56.43 | StephenF[W] | i mean user account |
| 18:56.46 | jameswf | *desktops |
| 18:56.58 | StephenF[W] | user dekstops |
| 18:57.13 | jameswf | alot of people like xlite |
| 18:57.27 | jaytee | is Snap still around? |
| 18:57.28 | Katty | is that the one that does video? |
| 18:57.28 | jameswf | you wont see em in here but some like HUC |
| 18:57.31 | jameswf | *HUD |
| 18:57.52 | StephenF[W] | im talking about integration with like Outlook, and other desktop applications |
| 18:57.54 | jameswf | xlite does video.... so i hear xlite for linus sucks |
| 18:58.01 | StephenF[W] | not a softphone |
| 18:58.02 | jaytee | Snap does that |
| 18:58.14 | Katty | zoiper girl myself (= |
| 18:58.27 | jameswf | zoiper for linux kinda sucks too |
| 18:58.29 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 18:58.33 | jameswf | i use moziax |
| 18:59.01 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 18:59.12 | *** join/#asterisk bmg505 (n=leon@196-209-8-72-ndn-esr-2.dynamic.isadsl.co.za) |
| 18:59.31 | jaytee | I'm playing with trying to get MS Office Communicator going with out Microsoft's mediation server using just Exchange and * 1.6 Haven't had much time to play with it though. Too many priorities ahead of it in the queue |
| 19:01.06 | jeev | wow |
| 19:01.08 | jeev | microsoft? |
| 19:01.09 | nikko | we're an OS X shop, and only have address book dialing for X-Lite integration, which is weak, but better than duplicate address books everywhere |
| 19:01.35 | *** join/#asterisk russellb (n=russell@asterisk/digium-open-source-team-lead/russellb) |
| 19:01.35 | *** mode/#asterisk [+o russellb] by ChanServ |
| 19:01.47 | magic_hat | [TK]D-Fender: still wrestling w/ that audio file not being found from last nite. here's the pastie: http://pastie.org/301642 |
| 19:01.53 | jaytee | jeev, it's only cuz we're mostly Windows here and we use Exchange UM for voicemail instead of Comedian Mail in Asterisk |
| 19:02.29 | jaytee | russellb, quick question if you have the time |
| 19:02.50 | [TK]D-Fender | magic_hat: Should have shown me this yesterday... it'd have saved you a lot of time |
| 19:03.08 | IsUp | magic_hat: go to line 18 and remove .gsm extension |
| 19:03.08 | [TK]D-Fender | mag BackGround("SIP/dailynews-09b47760", "/var/lib/asterisk/sounds/cdngreeting.gsm") <-- NEVER specify the file extension |
| 19:03.15 | magic_hat | [TK]D-Fender: I agree -- had to deal w/ another emergency. lol |
| 19:03.41 | magic_hat | I've tried it with just BackGround(cdngreeting) too. Same prob. |
| 19:03.44 | [TK]D-Fender | magic_hat: 8 will pick the best format based on what's available |
| 19:04.03 | russellb | jaytee: um, maybe |
| 19:04.03 | [TK]D-Fender | magic_hat: try showng us something that isn't clearly wrong then. |
| 19:04.05 | russellb | you can tree |
| 19:04.17 | *** join/#asterisk skyphyr (n=quassel@92-234-8-140.cable.ubr07.hari.blueyonder.co.uk) |
| 19:04.36 | gcbirzan | [TK]D-Fender: I think it was something NAT related. But, meh. I didn't change the firewall. *looks at coworker and plans a murder* |
| 19:04.55 | russellb | s/tree/try/ |
| 19:05.20 | *** join/#asterisk seanmh (i=HydraIRC@216.31.101.121) |
| 19:05.26 | gcbirzan | That being said, thanks. |
| 19:05.27 | jaytee | russellb, is there a way to change the order that Comedian Mail plays messages so that it's LIFO instead of FIFO? a guy was in here on Saturday night asking that. I couldn't find anything in the mail configs at home where I'm using Comedian Mail. Seems to only be FIFO order. |
| 19:05.50 | russellb | good question. |
| 19:05.55 | jeev | really good question. |
| 19:05.56 | jeev | lol |
| 19:05.58 | russellb | unless you find it in voicemail.conf, then no |
| 19:06.00 | jaytee | Exchange UM defaults to LIFO but it can be configured |
| 19:06.04 | russellb | but I seem to remember someone adding that option |
| 19:06.07 | russellb | probably in 1.6 only |
| 19:06.17 | jaytee | russellb, nope, not listed in voicemail.conf |
| 19:06.24 | russellb | did you look in 1.6? |
| 19:06.33 | jaytee | I'll have to look in the configs for 1.6 |
| 19:06.36 | russellb | http://svn.digium.com/svn/asterisk/trunk/configs/voicemail.conf.sample |
| 19:06.37 | russellb | :) |
| 19:06.40 | jaytee | I looked in 1.4 |
| 19:07.04 | russellb | i don't see it at quick glance |
| 19:07.05 | russellb | shrugs |
| 19:07.54 | magic_hat | [TK]D-Fender: see the update... http://pastie.org/301642 |
| 19:09.47 | jaytee | neither do I but there's a messagewrap settting I don't recall seeing in 1.4 |
| 19:10.08 | jaytee | lol, andrew dufresne and ellis redding mailboxes. |
| 19:10.17 | jaytee | get busy livin or get busy dying |
| 19:10.20 | [TK]D-Fender | magic_hat: please provide a call with SIP debug enabled |
| 19:15.58 | *** join/#asterisk skyphyr (n=quassel@92-234-8-140.cable.ubr07.hari.blueyonder.co.uk) |
| 19:16.31 | magic_hat | [TK]D-Fender: http://pastie.org/301642 |
| 19:16.37 | ReDNeQ | anyone here familiar with snom phones? |
| 19:17.12 | jameswf | ~ask |
| 19:17.12 | jbot | ask is, like, Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 19:17.31 | ReDNeQ | jameswf: my question was specific! |
| 19:18.12 | jameswf | argh.... the point was simply ask! if someone knows they will pipe up |
| 19:18.21 | ReDNeQ | no since in wasting peoples time if there is nobody here that knows snom phones! |
| 19:19.18 | jameswf | hears betwean [TK]D-Fender and google [TK]D-Fender knows all |
| 19:19.27 | jaytee | now that I've read at least 3 lines of irrelevant text I'd like to just go on record that I really, really like cheese of almost any variety. |
| 19:19.50 | stintel | :P |
| 19:19.57 | stintel | jaytee: so do I |
| 19:20.02 | jameswf | jaytee: iven those that smell bad |
| 19:20.11 | jameswf | *even |
| 19:20.12 | stintel | especially those that smell bad ;) |
| 19:20.34 | jameswf | stintel: sounds french.. |
| 19:20.52 | jaytee | jameswf, I ask Google first and then if I can't find it there I ask [TK]D-Fender, mainly just so if he asks "did you bother to google it?" I can answer honestly. |
| 19:21.54 | [TK]D-Fender | magic_hat: and with core debug 10 please... |
| 19:21.55 | jameswf | they say America is the country with 1,000 religions and 1 cheese, france has 1 religion and 1.000 cheeses |
| 19:21.58 | *** join/#asterisk JohnnyScarlet (n=JohnnySc@user-12hdmrj.cable.mindspring.com) |
| 19:22.01 | jaytee | I bought this cheese that looks kinda marbled that's an irish mild cheddar made with Guinness. can't remember the name though. |
| 19:22.06 | jaytee | it was awesome |
| 19:22.21 | JohnnyScarlet | Chuiness? |
| 19:22.26 | JohnnyScarlet | Guiddar? |
| 19:22.27 | jaytee | nope |
| 19:22.30 | stintel | roquefort is one of my favorites |
| 19:22.31 | jameswf | s/./,/ |
| 19:22.48 | *** join/#asterisk Carlos_PHX (n=Carlos@64.129.235.200) |
| 19:22.59 | jameswf | im a simple swiss kinda guy |
| 19:23.05 | magic_hat | [TK]D-Fender: stand by :) |
| 19:23.07 | JohnnyScarlet | Having some broadvoice setup issues on my first ever Asterisk installation |
| 19:23.20 | jameswf | likes swiss cheese, knives, women |
| 19:23.27 | jaytee | aha! found it. it's called Cahill's Porter Cheddar |
| 19:23.29 | JohnnyScarlet | I followed the directions verbatim, of which [TK}D-Fender sent me |
| 19:23.38 | *** join/#asterisk korihor (n=korihor@201.211.174.97) |
| 19:24.15 | JohnnyScarlet | however when I ran /usr/sbin/asterisk -cvvv I got a, " chan_sip.c:19700 set_insecure_flags: Unknown insecure mode 'very'" |
| 19:25.11 | jaytee | if an operating system was compared to swiss cheese with the holes representing security vulnerabilities then Windows would be Alsace Lorraine |
| 19:25.24 | JohnnyScarlet | And i also got a " chan_sip.c:21361 reload_config: Failed to bind to 0.0.0.0:5060: Address already in use" I'm following the instructiosn from broadvoice AND the asterisk O'Rielly book |
| 19:25.32 | *** join/#asterisk hi365_m (n=hi365@213.151.45.106) |
| 19:25.40 | JohnnyScarlet | thats too generous JayTee |
| 19:25.49 | magic_hat | [TK]D-Fender: updated. sorry about that -- thought I had verbosity set at 10 |
| 19:26.21 | JohnnyScarlet | whats the proper insecure mode flag fro Broadvoice? |
| 19:26.57 | JohnnyScarlet | "very" as it's instructions state, is giving me Warnings |
| 19:26.57 | hi365_m | is it posible to use the polycom call pickup feature with asterisk? |
| 19:27.27 | [TK]D-Fender | magic_hat: I said core debug, not just verbose |
| 19:28.32 | magic_hat | sorry... misunderstood. how do i set core debug? |
| 19:32.34 | JohnnyScarlet | Can someone give me an idea of the types of flags I can use then? |
| 19:33.30 | *** join/#asterisk skyphyr (n=quassel@92-234-8-140.cable.ubr07.hari.blueyonder.co.uk) |
| 19:34.25 | StephenF[W] | Any ideas why outgoing calls are not updating my presence to busy? |
| 19:34.50 | StephenF[W] | incoming calls work, but if I dial out to the PSTN my presence is not updated, on Polycom phones |
| 19:34.58 | StephenF[W] | also it is not updated in show hints |
| 19:36.36 | StephenF[W] | ~book |
| 19:36.36 | jbot | i guess book is Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com or see ~buybook |
| 19:37.55 | *** join/#asterisk skyphyr (n=quassel@92-234-8-140.cable.ubr07.hari.blueyonder.co.uk) |
| 19:38.21 | *** part/#asterisk gbr_ (n=gbr@200.103.96.98) |
| 19:38.36 | magic_hat | [TK]D-Fender: updated the pastie |
| 19:39.18 | *** part/#asterisk jer (n=jer@unaffiliated/jer) |
| 19:39.31 | *** join/#asterisk jer (n=jer@unaffiliated/jer) |
| 19:39.47 | jdnWEST | Can I run * in ESXi? |
| 19:41.31 | TalkRadio | i doubt it |
| 19:42.25 | russellb | should be fine |
| 19:42.38 | blitzrage | I run it in VMware Server with no problems |
| 19:43.39 | *** join/#asterisk skyphyr (n=quassel@92-234-8-140.cable.ubr07.hari.blueyonder.co.uk) |
| 19:43.55 | jeev | russellb |
| 19:44.10 | russellb | jeev: |
| 19:44.14 | russellb | stop saying my name |
| 19:44.40 | jeev | it was a typo |
| 19:44.44 | TalkRadio | sry i thought you meant running it natively on the exsi box not in vmware machine |
| 19:45.00 | *** join/#asterisk voxter (n=voxter@rrcs-67-53-210-146.west.biz.rr.com) |
| 19:47.56 | *** join/#asterisk skyphyr (n=quassel@92-234-8-140.cable.ubr07.hari.blueyonder.co.uk) |
| 19:50.48 | *** join/#asterisk famicom (i=famicom@5ED2FF2D.cable.ziggo.nl) |
| 19:50.54 | famicom | lo there |
| 19:51.14 | magic_hat | [TK]D-Fender: any luck w/ that? |
| 19:51.29 | [TK]D-Fender | magic_hat: You mean with the new link you didn't provider to me? |
| 19:52.01 | famicom | quick question |
| 19:52.14 | *** join/#asterisk edibrac (n=edibrac4@206.173.193.34.ptr.us.xo.net) |
| 19:52.16 | famicom | i'm thinking of setting up the following |
| 19:52.17 | famicom | 1 direct phoneline |
| 19:52.17 | famicom | 2 phone numbers that are hooked up to an automated menu |
| 19:52.17 | famicom | 1 number for incoming/outgoing faxes |
| 19:52.37 | famicom | is asterisk the right choice? |
| 19:53.07 | magic_hat | [TK]D: i just updated the old one; http://pastie.org/301642 |
| 19:53.08 | russellb | asterisk is always the right choice |
| 19:53.47 | magic_hat | famicom: others in here know way more than I, but you might want to read up on asterisk and faxes. last i checked it didn't work well. |
| 19:54.10 | famicom | yup, i know that voip lines are generally trouble for faxing |
| 19:54.29 | magic_hat | but then again, that's what your fax machine is for :) |
| 19:54.31 | famicom | but most commercial fax2email solutions out there are shit |
| 19:54.45 | famicom | so it's probably easier to roll my own |
| 19:55.04 | famicom | I've constantly had faxes being bounced etc etc |
| 19:55.19 | magic_hat | we use maxemail. works okay. |
| 19:55.19 | famicom | and these were HOSTED services, mind you |
| 19:55.54 | famicom | yeah, but those are US only |
| 19:55.57 | famicom | i'm dutch |
| 19:56.03 | famicom | out here there's Efax |
| 19:56.12 | famicom | which is obscenely expensive |
| 19:56.58 | famicom | and 2 others, one of which ignored my repeated sign ups and another which kept dropping faxes back to senders |
| 20:01.57 | *** join/#asterisk jameswf (n=james@dsl093-157-131.phx1.dsl.speakeasy.net) |
| 20:06.39 | cesar_CR | hello guys I am having problems with queues over dahdi channel... |
| 20:06.51 | cesar_CR | caller cant hear musiconhold |
| 20:07.20 | cesar_CR | or the messages from the asterisk |
| 20:07.37 | famicom | here's a solution |
| 20:07.40 | famicom | PICK UP THE GODDAMN PHONE |
| 20:08.06 | cesar_CR | here is the output from the * http://pastebin.ca/1238065 |
| 20:08.10 | *** join/#asterisk dlynes_office (n=dlynes@S01060016b68219f1.vs.shawcable.net) |
| 20:08.26 | cesar_CR | famicom, ???? |
| 20:08.39 | famicom | I hate being put on hold |
| 20:08.39 | edibrac | do GSM phones interfere with Cisco 7940/60's? |
| 20:10.01 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 20:12.24 | cesar_CR | famicom... jajaja you where right!!! |
| 20:12.40 | cesar_CR | there was a Answer ... missing |
| 20:12.47 | magic_hat | [TK]D-Fender: u see that yet? |
| 20:13.02 | cesar_CR | thanks :D jajaja |
| 20:14.25 | *** join/#asterisk skyphyr (n=quassel@92-234-8-140.cable.ubr07.hari.blueyonder.co.uk) |
| 20:14.46 | *** join/#asterisk pepse (n=pepse@71-223-125-121.phnx.qwest.net) |
| 20:14.55 | cesar_CR | 88729999 |
| 20:14.59 | pepse | greetings, ladies and germs. |
| 20:14.59 | cesar_CR | SORRY |
| 20:15.39 | pepse | Has anyone had success using SIP credentials from a MagicJack with Asterisk? I am getting 400 Bad Request on dialing out, and registration timed out on registering. |
| 20:16.22 | *** join/#asterisk skyphyr (n=quassel@92-234-8-140.cable.ubr07.hari.blueyonder.co.uk) |
| 20:16.50 | jdnWEST | Anyone use the Rhino Cards? |
| 20:19.42 | [TK]D-Fender | magic_hat: try writing out the path explicitly minus only the extension... |
| 20:20.26 | magic_hat | k hang on |
| 20:21.27 | blitzrage | pepse: use a pastebin and show us the SIP dialog |
| 20:22.45 | magic_hat | [TK]D-Fender: bingo. damn, I thought I tried that last nite. |
| 20:22.55 | magic_hat | any idea why that's happening? |
| 20:23.29 | [TK]D-Fender | magic_hat: I'd check your config paths again... you at one point also showed me dialplan that didn't match CLI... |
| 20:23.42 | [TK]D-Fender | magic_hat: Frankly I'm not sure what to trust from what you show me. |
| 20:24.00 | [TK]D-Fender | magic_hat: Maybe talking configs from one session, and CLI from another, etc |
| 20:24.37 | [TK]D-Fender | magic_hat: But you've gotten to taking multiple simultaneous step which makes debugging a PITA |
| 20:24.57 | Linuturk | is there a particular channel or area to get support for Asterisk on FreeBSD? |
| 20:25.19 | blitzrage | here is the place |
| 20:25.23 | [TK]D-Fender | Linuturk: This is an appropriate place to ask |
| 20:25.24 | *** join/#asterisk lanning (n=lanning@66.151.128.195) |
| 20:25.24 | blitzrage | but there is not much support |
| 20:25.34 | blitzrage | depends who's on I suppose |
| 20:25.40 | [TK]D-Fender | Linuturk: Indeed as a much lower % use it |
| 20:25.51 | magic_hat | [TK]D: lol, you're right. |
| 20:25.59 | [TK]D-Fender | Linuturk: One of the more active maintainers is here regularly though |
| 20:26.09 | Linuturk | well, i'm reading up on it, for my net5501, and the server I'm replacing has a Wildcard TDM400P driver |
| 20:26.17 | Linuturk | er, card* |
| 20:26.32 | Linuturk | just want to make sure I'm not heading down a dead end |
| 20:26.51 | blitzrage | Linuturk: have you looked at astlinux for your soekris? |
| 20:26.52 | *** join/#asterisk Carlos_PHX (n=Carlos@64.129.235.200) |
| 20:26.58 | Linuturk | I've got a few regualar phone lines coming in. Nothing like a T1 or anything. |
| 20:27.29 | blitzrage | Linuturk: I suggest you look at the external devices and connect it via SIP to the net5501 |
| 20:27.45 | blitzrage | then you're not limited on the hardware, or the drivers being compilable against your distro of choice |
| 20:28.05 | Linuturk | blitzrage: actually, I've got that on there right now, but the project doesn't seem to be that well organized. I don't want to put something on there that's hard to support. At least freebsd and vanilla asterisk have docs. astlinux is skinny on docs atm |
| 20:28.19 | Linuturk | "external devices" ? |
| 20:28.25 | blitzrage | analog--<device>--SIP--net5501 |
| 20:28.35 | *** join/#asterisk Katty (n=asterisk@mail.copi-rite.com) |
| 20:28.39 | [TK]D-Fender | checkout time... heading home... later all |
| 20:28.51 | blitzrage | like say... one of those SPA3102 devices |
| 20:28.53 | Linuturk | blitzrage: so, replace the PCI card with an external device? |
| 20:29.07 | magic_hat | anyone know how to guesstimate how many POTS lines i'd need for 8 users? |
| 20:29.09 | blitzrage | Linuturk: correct, then you just need to connect it to the same switch as the net5501 |
| 20:29.14 | Katty | hai |
| 20:29.22 | blitzrage | magic_hat: depends number of simultanous calls, length of calls, etc... |
| 20:29.57 | blitzrage | magic_hat: then you can spread that over some sort of standard deviation vs. time or something... (I suck at math, but this sounds like a statistics problem) |
| 20:30.10 | magic_hat | blitzrage: yeah, of course. Is there any rule of thumb to do a reasonable estimate though? |
| 20:30.10 | Katty | magic_hat: what type of business is it? |
| 20:30.12 | blitzrage | Katty: omghi2u |
| 20:30.20 | Katty | blitzrage: hewwoes. |
| 20:30.24 | Linuturk | blitzrage: I've got more than one line coming in for said office :-/ |
| 20:30.25 | Katty | blitzrage: icanhazhug? |
| 20:30.34 | magic_hat | Katty: newsroom. So a lot of calls. But not like a call center or anything. |
| 20:30.51 | blitzrage | Linuturk: I think there should be a device that does multiple lines... mediatrix does, but that is kinda overkill for what you're trying to do |
| 20:30.54 | Katty | magic_hat: would you say mostly inbound, or outbound traffic? |
| 20:31.14 | magic_hat | Katty: roughly equal. or 60-40 favoring outbound. |
| 20:31.16 | Linuturk | yeah, which is why I sorta want to stick with the pci device blitzrage. puts it all in one box |
| 20:31.31 | Katty | magic_hat: i'd say 5 or 6 lines then |
| 20:31.43 | Katty | magic_hat: you could probably do with 5 |
| 20:31.49 | tzanger | Linuturk: that's why I like the sangoma ADSL cards |
| 20:32.01 | blitzrage | I have one of those... |
| 20:32.02 | tzanger | ADSL card + TDM400/800/whatever |
| 20:32.05 | blitzrage | no DSL though... need to sell it |
| 20:32.08 | tzanger | haha |
| 20:32.20 | magic_hat | cool... i can always add more. just need a way to compare costs across a couple of different setups. |
| 20:32.35 | Katty | magic_hat: are you doing analog lines or sip trunks? |
| 20:32.39 | Katty | magic_hat: integrated t1? |
| 20:32.44 | Linuturk | well, we've got a DSL line for our internet at this office. blitzrage tzanger How would I fit that in? |
| 20:32.57 | magic_hat | Katty: I'm looking at the possibility of switching from sip to analog. |
| 20:33.04 | Katty | nods |
| 20:33.04 | magic_hat | and trying to figure how much it'll cost. |
| 20:33.05 | Linuturk | besides an IAX trunk back to the main office ;p |
| 20:33.17 | blitzrage | Linuturk: you won't be able to with the net5501 since it only has a single PCI slot |
| 20:33.24 | Linuturk | ah |
| 20:33.44 | blitzrage | but DAHDI (driver beyond Zaptel) should build on FreeBSD afaik |
| 20:34.35 | Linuturk | http://www.voip-info.org/wiki/view/FreeBSD+zaptel << blitzrage well, this page says support is currently beta. I'm just curious how "beta" it is |
| 20:34.55 | blitzrage | that page is probably quite out of date... |
| 20:34.59 | Katty | magic_hat: an 8 port sangoma card with echo cancelation will run you about a grand. |
| 20:35.04 | *** join/#asterisk Trionnis (i=Trionnis@s233-51-251.nap.wideopenwest.com) |
| 20:35.07 | Nugget | the path of least resistance is to just suck it up and tolerate a linux box for asterisk |
| 20:35.10 | tzafrir_laptop | wonders if they actually bother using dahdi-tools |
| 20:35.19 | tzafrir_laptop | (for dahdi/freebsd) |
| 20:35.23 | blitzrage | tzafrir_laptop: I was just gonna ask you if you know if DAHDI builds on FreeBSD? |
| 20:35.31 | Katty | magic_hat: not including your phone bill, of course. |
| 20:35.42 | blitzrage | doesn't want to spread anymore disinformation than is necessary :) |
| 20:35.42 | Katty | magic_hat: i believe sip trunks will run you about 30 a month, give or take, free long distance. |
| 20:35.44 | Nugget | I wouldn't expect dahdi to build on freebsd. |
| 20:35.50 | Linuturk | blitzrage: does that mean they've come a long way with that hardware, or that devel stalled? |
| 20:35.54 | Katty | magic_hat: per trunk, of course. no additional hardware required. |
| 20:35.54 | tzafrir_laptop | UnixDawg keeps talking about it |
| 20:36.18 | blitzrage | ah... ok, for some reason I thought it was common now for some devs to build on FreeBSD, but maybe that is still just asterisk |
| 20:36.35 | blitzrage | ya, I keep seeing talk from him about building it... no idea if anything has come from it |
| 20:36.38 | Nugget | asterisk builds fine, but dahdi is a linux driver. |
| 20:36.41 | blitzrage | welp, back off to do some testing |
| 20:37.00 | Nugget | zaptel for freebsd is an independent codebase |
| 20:37.13 | Nugget | I assume they'll have to do the same for dahdi |
| 20:37.38 | Trionnis | someone around that can point me in a direction for info on ajam Originate, and passing in variables? |
| 20:37.49 | Trionnis | not a lot of docs out there it seems... |
| 20:38.38 | Nugget | 23-Oct-2008 08:50 <UnixDawg_> well we are about done with dahdi on bsd |
| 20:38.42 | Nugget | ^ the last word |
| 20:41.41 | Linuturk | so, is asterisk on bsd a dead end? |
| 20:42.33 | *** join/#asterisk gr0mit (n=tim@81.187.32.146) |
| 20:44.41 | Nugget | imho, yes, it's pain you don't have to endure for very little benefit. |
| 20:45.28 | Nugget | if you need anything that requires dahdi (app_page, app_meetme, app_flash, etc) you'll probably be frustrated (at best) |
| 20:46.03 | Nugget | and if you ever have any problems you need help from the community to solve you'll be met by a blinking chorus of shrugging shoulders where people say "dunno, it works in linux" |
| 20:46.06 | *** join/#asterisk Maxous (n=Maxous@168.9.44.2) |
| 20:46.26 | *** part/#asterisk Maxous (n=Maxous@168.9.44.2) |
| 20:46.36 | Nugget | asterisk development is just moving too fast for portability to even register on digium's radar, and it shows |
| 20:47.03 | *** join/#asterisk xuser (n=JJ@unaffiliated/xuser) |
| 20:47.14 | Nugget | just suck it up, install the linux distro that you find least offensive, and treat it as a bootloader for asterisk, which it sort of is. :) |
| 20:48.12 | Nugget | <-- not a digium employee or asterisk coder, but I gave it a shot for a year before I gave up and went linux |
| 20:49.06 | blitzrage | Linuturk: fyi astlinux is actively developed by a couple of people. Check out #astlinux, they are putting out releases every couple of months from what I've been seeing |
| 20:49.13 | blitzrage | just an fyi |
| 20:50.11 | Linuturk | well, i guess I'll keep it astlinux then, per you guy's suggestions |
| 20:50.27 | Linuturk | now I need to mirror the configs, and hope this card works in this box |
| 20:50.32 | *** join/#asterisk Tebi (n=user@support.ccxtech.fi) |
| 20:50.36 | Nugget | is astlinux on 1.6 yet? |
| 20:50.43 | blitzrage | Nugget: oh I doubt it |
| 20:50.55 | Trionnis | ajam docs? anyone? please? :) |
| 20:51.00 | blitzrage | took a bit moving it to 1.4, but I'm pretty sure they are releasing a lot more often and keeping up with 1.4. releases |
| 20:51.05 | blitzrage | ajam docs? |
| 20:51.26 | Linuturk | I got 1.4 a couple of weeks ago on ast |
| 20:51.35 | Linuturk | astlinux-0.6.1 - Asterisk 1.4.21.2 |
| 20:51.42 | Linuturk | tis what I have :) |
| 20:51.47 | Trionnis | yes... having issues passing variables into Originate |
| 20:51.56 | Trionnis | looking for some clear-cut docs on it |
| 20:54.07 | Trionnis | don't get me wrong, your section in TFOT wasn't bad, but it was a bit light on the details ;) |
| 20:56.55 | *** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194) |
| 20:57.09 | *** join/#asterisk plik (i=gorph@phalse.2600.COM) |
| 20:57.48 | Yourname` | What's that file one needs to change to be able to see DTMF in CLI? |
| 20:59.16 | *** part/#asterisk PTorres (n=PTorres@200.68.87.146) |
| 20:59.37 | *** join/#asterisk skyphyr (n=quassel@92-234-8-140.cable.ubr07.hari.blueyonder.co.uk) |
| 21:00.04 | *** join/#asterisk johnakabean (n=none@pool-72-82-112-211.nrflva.east.verizon.net) |
| 21:00.33 | jaytee | quittin time, be back later |
| 21:00.37 | Katty | what goes well with baked beans? |
| 21:00.46 | Linuturk | hotdogs |
| 21:00.47 | jaytee | hot dogs or hamburgers |
| 21:00.49 | johnakabean | hey room, anyone know how to setup Qos using TC? I have setup 3 que priorities so far and tried to add ports to the que but they are being ignored. |
| 21:00.50 | Linuturk | sliced |
| 21:00.53 | Linuturk | in the beans |
| 21:01.04 | Katty | hmm. |
| 21:01.07 | Katty | that might work |
| 21:01.15 | Linuturk | tis good |
| 21:01.16 | Linuturk | :) |
| 21:01.21 | Katty | i'm starving |
| 21:01.31 | *** part/#asterisk jaytee (n=jforde@unaffiliated/jaytee) |
| 21:01.32 | _ShrikE | beanie weenies |
| 21:01.49 | pepse | blitzrage: will do, just seeing if someone has it already ocnfigured/working and maybe i'm just missing an option. |
| 21:01.49 | Linuturk | beanie weenies FTW!! |
| 21:02.09 | blitzrage | pepse: hard to say without information regardless :) |
| 21:02.18 | JohnnyScarlet | whats the proper insecure mode flag for Broadvoice? The one listed in broadvoice instructiosn gets me an "Unknown flag" for "Insecure=very" |
| 21:02.19 | Linuturk | thanks blitzrage :) |
| 21:02.38 | johnakabean | johnny i use BV |
| 21:02.42 | johnakabean | what's up. |
| 21:02.49 | blitzrage | Katty: ya, pieces of tofu dogs in brown beans is good |
| 21:02.54 | JohnnyScarlet | or can I just eliminate the insecure=very line altogether? |
| 21:03.08 | blitzrage | JohnnyScarlet: on 1.6.x it is insecure=invite,port |
| 21:03.29 | blitzrage | this is in sip.conf btw |
| 21:03.35 | JohnnyScarlet | johnakabean, any advice? it's my first asterisk box ever, and i'm having setup warnings :(. I haven't even gotten to programmign the AAStra 57i phone yet |
| 21:04.01 | Trionnis | Broadvoice? |
| 21:04.03 | JohnnyScarlet | blitzrage you mean when i run make samples... it's in sip.conf? |
| 21:04.04 | johnakabean | yeah the parameters on the site are incomplete....i'll look for where i posted the right ones |
| 21:04.13 | Trionnis | perhaps the gods are trying to tell you something if it's not working... ;) |
| 21:04.17 | blitzrage | JohnnyScarlet: yes... how much documentation have you read? |
| 21:04.27 | Katty | eats hershey's bar |
| 21:04.31 | blitzrage | has a feeling JohnnyScarlet may have put the "option" in the wrong spot... |
| 21:04.36 | JohnnyScarlet | Well i'm actually reading the Asterisk O'rielly guide, but it covers 1.4 |
| 21:04.45 | johnakabean | ohhhh i' on 1.4 |
| 21:04.47 | blitzrage | steals Katty's chocolate and runs to the corner nibbling on it |
| 21:04.48 | JohnnyScarlet | and mixing that with the broadvoice instructions |
| 21:04.51 | johnakabean | don't use 1.6 with Bv |
| 21:04.58 | Trionnis | don't use Bv period |
| 21:05.01 | Trionnis | ;) |
| 21:05.07 | blitzrage | don't. |
| 21:05.09 | pepse | blitzrage: i can post my config now, just not at home to try making calls and such. |
| 21:05.10 | Katty | blitzrage: cookies n cream. |
| 21:05.12 | johnakabean | don't use BV if you don't want unlimited channels and DID |
| 21:05.17 | johnakabean | ;-) |
| 21:05.19 | blitzrage | Katty: delicious!! |
| 21:05.25 | Katty | blitzrage: i'd trade you for something less sweet |
| 21:05.25 | *** part/#asterisk blitzrage (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 21:05.27 | johnakabean | i have a workaround in my box to have unlimited DiD's and channels |
| 21:05.29 | JohnnyScarlet | Trionnis I have to use what the boss makes me use :) I still have to setup Vonage, and two other BV lines |
| 21:05.47 | *** join/#asterisk russellb (n=russell@asterisk/digium-open-source-team-lead/russellb) |
| 21:05.47 | *** mode/#asterisk [+o russellb] by ChanServ |
| 21:05.55 | Katty | russellb: i'm hungry. |
| 21:05.58 | Katty | russellb: feed me. |
| 21:05.59 | Trionnis | he's here! |
| 21:06.01 | Trionnis | ssshhh |
| 21:06.06 | JohnnyScarlet | blitzrage where is the right spot for the "option"? |
| 21:06.15 | russellb | Katty: kinky |
| 21:06.29 | Katty | russellb: not quite the same train of thought. |
| 21:06.34 | Katty | russellb: i was thinking protein. |
| 21:06.39 | Katty | russellb: maybe carbs. |
| 21:06.43 | russellb | O.O |
| 21:06.48 | Trionnis | lol |
| 21:06.49 | russellb | offers Katty a bagel |
| 21:06.52 | Katty | :> |
| 21:06.57 | johnakabean | Anyone know how to setup qos with TC? |
| 21:07.05 | johnakabean | johnny you're going to need to do this as well |
| 21:07.09 | Katty | russellb: my life is doom for the next 3 weeks. |
| 21:07.16 | russellb | Katty: i'm sorry |
| 21:07.16 | JohnnyScarlet | blitzrage: do you have a link to an proper example for sip.conf using broadvoice? |
| 21:07.31 | johnakabean | i am going to give you that johnny to try |
| 21:07.35 | Katty | russellb: me too :< |
| 21:07.46 | JohnnyScarlet | awesome, thanks johnakabean |
| 21:07.55 | pepse | blitz: http://pastebin.ca/1238118 |
| 21:08.19 | russellb | ~hug Katty |
| 21:08.20 | jbot | ACTION hugs Katty tightly until Katty turns slightly blue |
| 21:08.23 | Katty | :> |
| 21:08.45 | JohnnyScarlet | crap, I have to tend a meeting right now, but i'll leave my window open for the updates johnakabean and blitzrage . be back ina few |
| 21:09.00 | Trionnis | wonders how many more people are going to keep talking to blitz before they realize he left the channel a while ago |
| 21:09.09 | Katty | i was wondering th esame thing. |
| 21:09.14 | Katty | but being amused by it. |
| 21:09.41 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 21:10.00 | JohnnyScarlet | johnakabean: btw, what is tc? I know QoS. Don't I jsut set that up within my router? or no? |
| 21:10.00 | Yourname` | Lindt! |
| 21:10.14 | [TK]D-Fender | JohnnyScarlet: http://www.broadvoice.com/support_install_asterisk.html |
| 21:10.38 | JohnnyScarlet | [TK}D-Fender thanks, You gave me that on Firday. I printed it out and followed it Verbatim. |
| 21:10.50 | Trionnis | Andrew!! |
| 21:10.54 | JohnnyScarlet | however insecure=very is not a valid flag according to * 1.6 |
| 21:11.00 | [TK]D-Fender | JohnnyScarlet: Then feel free to show us your config and the SIP debug of your failures |
| 21:11.00 | Trionnis | tackles [TK]D-Fender and gives him a noogie |
| 21:11.07 | JohnnyScarlet | Johnakabean said he ahd a workaround, but he uses 1.4 |
| 21:11.09 | Trionnis | long time no see |
| 21:11.10 | [TK]D-Fender | JohnnyScarlet: "insecure=port,invite |
| 21:11.11 | Katty | [TK]D-Fender: next 3 weeks is DOOM |
| 21:11.11 | Trionnis | :) |
| 21:11.35 | [TK]D-Fender | Trionnis: y0 |
| 21:11.49 | JohnnyScarlet | which port though? 5060? |
| 21:12.02 | JohnnyScarlet | or port is not a variable? |
| 21:12.16 | [TK]D-Fender | JohnnyScarlet: Is your * behind NAT? |
| 21:12.25 | JohnnyScarlet | yes |
| 21:12.27 | [TK]D-Fender | ~sipnat |
| 21:12.27 | jbot | [~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 |
| 21:12.29 | [TK]D-Fender | ^^^^^^^^^ |
| 21:12.30 | pepse | oh, blitz signed off |
| 21:12.59 | JohnnyScarlet | ok, i will work with that. thank you. I ahve another error to address :)... |
| 21:13.02 | JohnnyScarlet | <PROTECTED> |
| 21:13.20 | JohnnyScarlet | no idea what that is referring to |
| 21:14.37 | johnakabean | johnnyscarlet: http://pastebin.centos.org/22418 |
| 21:15.16 | johnakabean | compare mine to broadvoices you will see a difference.......now ANYONE KNOW HOW TO SETUP QOS FOR ME AND JOHNNY USING TC? |
| 21:15.59 | Katty | johnakabean: I DON'T THINK TYPING IN ALL CAPS IS GOING TO MAKE ANYONE MORE RESPONSIVE. |
| 21:16.09 | Katty | johnakabean: in fact, it just annoys. |
| 21:16.15 | Katty | johnakabean: i'd recommend not doing it. |
| 21:16.17 | [TK]D-Fender | johnakabean: Or repeating yourself every 10-15 minutes |
| 21:16.37 | ManxPower | johnakabean: You cannot do real QoS over the internet. Is that what you are trying to do? |
| 21:16.52 | johnakabean | no, i have a nat on my asterisk box.... |
| 21:16.54 | johnakabean | i'm behind it |
| 21:17.10 | johnakabean | i want my phone calls to get priority of course |
| 21:17.18 | ManxPower | johnakabean: That was not my question. |
| 21:17.33 | ManxPower | NAT has nothing whatsoever to do with QoS |
| 21:17.53 | johnakabean | ok my connect has 1 ms jitter at max and 98 percent Quality at lowest. |
| 21:18.18 | johnakabean | but it is the reason I want qos for when i'm downloading something and the phones are in use |
| 21:18.21 | ManxPower | johnakabean: What is providing the quality info? |
| 21:18.37 | ManxPower | johnakabean: you can only do QoS on TRANSMITTED data, not RECEIVED data. |
| 21:18.53 | ManxPower | It sounds like you only need what I call "fake QoS" |
| 21:19.25 | ManxPower | just give ports 5060 and whatever ports are listed in /etc/asterisk/rtp.conf (defaults to 10,000 - 20,000). All UDP of course. |
| 21:19.54 | ManxPower | Bittorrent and other UDP based applications will just breeze right thru your "QoS", but TCP based apps should do pretty good. |
| 21:20.13 | johnakabean | manx, i do that but everything ends up priority 3 when in a phone call. |
| 21:20.26 | ManxPower | johnakabean: then I guess you are not doing it right. |
| 21:20.47 | ManxPower | SIP uses the ports listed. It uses no other ports. Remember these are all DEST ports, not SOURCE ports. |
| 21:21.16 | johnakabean | that's what i need is my transmitted data as i have Adsl |
| 21:21.23 | johnakabean | i use dport in the tc line |
| 21:21.45 | ManxPower | johnakabean: I know nothing about the implementation specifics of "TC". But I do know QoS. |
| 21:21.57 | Yourname` | Oh, wow, trying to get DTMF working with Aastra 57i, Voicenetwork.ca on 1.4.22 is killing me!!! |
| 21:22.09 | johnakabean | here is my tc http://pastebin.centos.org/22419 |
| 21:22.18 | *** join/#asterisk hfb (n=hfb@pool-96-247-49-20.lsanca.dsl-w.verizon.net) |
| 21:22.20 | ManxPower | Yourname`: Use 1.6, then you can deal with all the other issues and forget about the DTMF issues. |
| 21:22.38 | johnakabean | your name, disable DTMF processing on the aastra device so it just sends audible signals for asterisk |
| 21:22.44 | ManxPower | johnakabean: You must have missed it when I said " I know nothing about the implementation specifics of "TC". " |
| 21:22.45 | Yourname` | Man, that would be nice. Gonna be doing that real soon |
| 21:22.55 | [TK]D-Fender | lol |
| 21:22.57 | johnakabean | if you still have problems, its not setup for asterisk and your provider |
| 21:22.59 | pepse | ManxPower: the service I'm trying to use (and am having problems with) uses 5070. Any reason I should need that port forwarded to my * or anything? |
| 21:23.11 | [TK]D-Fender | johnakabean: Total waste.. you are trying to prioritize SIP.... |
| 21:23.22 | pepse | * being behind a router |
| 21:23.23 | johnakabean | so, fender, what should i do |
| 21:23.37 | *** join/#asterisk xuser (n=JJ@unaffiliated/xuser) |
| 21:23.39 | johnakabean | i am not behind a router.....my asterisk box IS the router |
| 21:23.45 | [TK]D-Fender | johnakabean: You should wake up and realize that your VOICE is carried over RTP, not SIP |
| 21:23.56 | ManxPower | pepse: You need to brush up on your networking. Every connection has 2 sides, the port on the server side is 5060, the other port can be any random number from 1024 - 6535 |
| 21:23.57 | johnakabean | so yes, I MYSELF ON THIS COMPUTER am behind the ASTERISK BOX router |
| 21:24.05 | johnakabean | get it? |
| 21:24.14 | [TK]D-Fender | johnakabean: If you are trying to do QoS, its on the wrong PORTS. |
| 21:24.33 | johnakabean | 5060 is sip right? |
| 21:24.33 | ManxPower | [TK]D-Fender: He must have missed me telling him about 10,000 - 20,000 UDP. |
| 21:24.35 | [TK]D-Fender | JohnYou do not need to prioritize SIP, you need to prioritize RTP |
| 21:24.35 | johnakabean | and 5061 |
| 21:24.45 | ManxPower | Perhaps he is hard of hearing^H^H^H^H^H^Hseeing |
| 21:24.59 | johnakabean | ok so how do you add a port range to TC |
| 21:24.59 | pepse | ManxPower: i'm just trying to figure out why i'm having problems.. all my other connections work fine, but MagicJack is a little funky. |
| 21:25.01 | [TK]D-Fender | johnakabean: I'm going to repeat this in an attempt to to acheive some final clarity... |
| 21:25.02 | ManxPower | I give up. TK, you can abuse him, I give up. |
| 21:25.05 | johnakabean | for 100000 through 200000 |
| 21:25.15 | [TK]D-Fender | johnakabean: SIP does not carry your actual coversation traffice <---- |
| 21:25.21 | johnakabean | it just invites it |
| 21:25.22 | johnakabean | yeah |
| 21:25.38 | johnakabean | rtp is 100000 through 200000 but i cant get a freakin port range in TC.....THAT'S WHY I'M HERE |
| 21:25.45 | [TK]D-Fender | johnakabean: Go find another channel or site to support it. |
| 21:25.57 | ManxPower | johnakabean: why ARE you here rather than in some TC support forum? |
| 21:25.58 | [TK]D-Fender | johnakabean: is this #tc? No. |
| 21:26.08 | johnakabean | because this is dealing with asterisk |
| 21:26.23 | ManxPower | johnakabean: no, it is dealing with TC |
| 21:26.24 | johnakabean | no, but it looks like #idiots |
| 21:26.28 | [TK]D-Fender | johnakabean: Do you sue Dow Plastics when your bumper falls off your new car? |
| 21:27.03 | johnakabean | i thought most of the poeple in here had prioritized their voice traffic |
| 21:27.08 | johnakabean | but i guessed wrong |
| 21:27.18 | ManxPower | johnakabean: you did guess wrong. |
| 21:27.27 | pepse | I use QOS on my crappy old wired-only linksys router |
| 21:27.43 | pepse | i just set 5060 udp and 10000-20000 udp to my * machine to have priority |
| 21:27.44 | ManxPower | Most people eventually realize that "QoS" on a *DSL connection just isn't going to live up to the hype. |
| 21:27.46 | pepse | seems to work fine |
| 21:27.49 | [TK]D-Fender | johnakabean: Most of us try to avoid using the internet for VoIP. Of those who do, how many do you think use TC? |
| 21:28.03 | pepse | what's TC, anyway? |
| 21:28.18 | ManxPower | I use QoS with Cisco routers and private dedicated T-1s |
| 21:28.27 | johnakabean | if you don't use the internet for voip, how do you get trunks........ZAPTEL? |
| 21:28.35 | johnakabean | pstn |
| 21:28.38 | [TK]D-Fender | johnakabean: Yes |
| 21:28.53 | johnakabean | great, pay 1000 bucks for a freakin T1 for 24 channels |
| 21:29.13 | ManxPower | johnakabean: all my customers use PRI with Digium or Sangoma cards. That way they don't call me whining about bad call quality when their internet gets flaky or overloaded. |
| 21:29.15 | pepse | i use pstn too, but i have one of those cheapo 1-line zaptel cards |
| 21:29.16 | pepse | :) |
| 21:29.33 | [TK]D-Fender | johnakabean: I pay $600 and don't have to worry about QoS or a failure along a long # of points. |
| 21:29.34 | johnakabean | i don't have bad quality, i'm just making sure i don't run into it |
| 21:29.34 | pepse | as well as a 4-port zaptel card (WDM400? i forget the model #) |
| 21:29.41 | pepse | both of them have horrible echo tho :( |
| 21:29.55 | johnakabean | crystal clear with 20 active phone calls |
| 21:30.02 | johnakabean | i only have a 512 upload |
| 21:30.06 | [TK]D-Fender | johnakabean: Well go right ahead, just don't expect us to have all the answers for your personal QoS methodology |
| 21:30.10 | johnakabean | and 3 mbps down |
| 21:30.13 | ManxPower | pepse: you can get a software commercial EC for free if your Digium card is under warrenty. If it's not then it costs $10/channel. Called HPEC |
| 21:30.34 | johnakabean | high precision echo canceller |
| 21:30.37 | [TK]D-Fender | pepse: If you're off-warranty, try OSLEC first |
| 21:30.40 | johnakabean | why does this have to do only with pstn? |
| 21:30.42 | johnakabean | lol |
| 21:30.42 | pepse | ManxPower: Cool, I'll look into that. I'm sure they are not under warranty. I have been using them for 2+ years. |
| 21:30.54 | pepse | OSLEC and HPEC. will do. |
| 21:30.57 | Katty | pulls hair out. |
| 21:30.59 | [TK]D-Fender | ~oslec |
| 21:30.59 | jbot | methinks oslec is Open Source Line Echo Canceller. See http://www.rowetel.com/ucasterisk/oslec.html . |
| 21:31.05 | X-Rob | ManxPower, or you can use OSLEC, which I've found is better than HPEC |
| 21:31.16 | ManxPower | pepse: then you can buy them for $10/channel, but be sure you have a hefty machine, the really good software EC sucks up a lot of CPU |
| 21:31.23 | johnakabean | why does pstn have echo if it does't go over the internet |
| 21:31.32 | [TK]D-Fender | X-Rob: YMMV may vary with either, but I'd rather try the free one first :) |
| 21:31.33 | pepse | i think it's a 1.8ghz machine, dedicated to * |
| 21:31.43 | Katty | johnakabean: that's the way ole analog lines work |
| 21:31.43 | ManxPower | X-Rob: I've never tried OSLEC. I switched to hardware tellabs EC and T-1/PRI for everything before OSLEC existed. |
| 21:31.46 | Katty | johnakabean: they have echo |
| 21:32.04 | johnakabean | when my jitter spikes once in a blue moon i get echo |
| 21:32.04 | tzafrir_laptop | Speaking of OSLEC, brave testers are welcomed: http://docs.tzafrir.org.il/dahdi-linux/#_oslec |
| 21:32.09 | [TK]D-Fender | johnakabean: "internet" has nothing to do with echo |
| 21:32.14 | pepse | johnakabean: voip-info.org has an excellent explanation of what causes echo. |
| 21:32.15 | johnakabean | latency does |
| 21:32.22 | X-Rob | ManxPower, Fair enough. It's surprisingly good. .au has some rather strange line conditions, which is possibly why the us-designed HPEC doesn't work so well over here. OSLEC is really good. |
| 21:32.35 | [TK]D-Fender | johnakabean: It'd have to be pretty terrible... delay yes, echo... no |
| 21:32.50 | ManxPower | X-Rob: I love the tellabs stuff so much I'm selling it rather than just installing it at customer locations |
| 21:32.59 | X-Rob | [TK]D-Fender, it HPEC might be better, I first tried it when it was released |
| 21:33.16 | johnakabean | well i'm about 200 miles from broadvoices local relay. |
| 21:33.16 | X-Rob | ManxPower, Yup. Hardware EC is the best. When you've got a whole damn DSP set aside for EC, it's going to win 8) |
| 21:33.25 | johnakabean | 19 ms ping |
| 21:33.36 | ManxPower | X-Rob: I use Tellabs stuff that was removed from telcos. |
| 21:34.06 | ManxPower | "If it's good enough for the telcos it's good enough for me" |
| 21:35.00 | johnakabean | you guys are trying to help johnny with freakin broadvoice and you don't use voip |
| 21:35.02 | johnakabean | oook |
| 21:35.10 | johnakabean | i guess what i gave him helped |
| 21:35.14 | *** part/#asterisk johnakabean (n=none@pool-72-82-112-211.nrflva.east.verizon.net) |
| 21:35.19 | Katty | kbainow |
| 21:35.49 | pepse | whatamaroon. |
| 21:35.55 | X-Rob | ManxPower, yeah, well you win with T1's. You can't buy any second hand E1 hardware for cheap. |
| 21:38.21 | [TK]D-Fender | X-Rob: Nope, but nice E1 cards w/ EC... easily come by :) |
| 21:38.35 | *** join/#asterisk ManxPower (n=manxpowe@183.sub-75-202-105.myvzw.com) |
| 21:38.42 | ManxPower | I find that you can either fight Asterisk's and VoIP's oddities and live a miserable pointless life, or you can embrace Asterisk's and VoIP's oddities and be happy and content. |
| 21:38.47 | X-Rob | Easily come by, definately. Pleasant on the wallet, definately not. |
| 21:38.54 | ManxPower | I try to do the latter |
| 21:39.14 | [TK]D-Fender | X-Rob: No moreso than T1.... |
| 21:39.55 | *** join/#asterisk jer (n=jer@unaffiliated/jer) |
| 21:40.38 | X-Rob | [TK]D-Fender, they were about 40% more expensive, last time I looked. Which is slightly acceptable, as there's 30 channels as oppsed to 24, but that's only 5/6ths more. Not 40%. |
| 21:40.50 | X-Rob | uh 1/6th even |
| 21:41.16 | [TK]D-Fender | X-Rob: I don't know any T1 Cards w/e HWEC that don't do E1 as well |
| 21:41.40 | [TK]D-Fender | X-Rob: X-Rob that makes them the same with my paltry math stills |
| 21:42.04 | [TK]D-Fender | skills even ;) |
| 21:42.04 | X-Rob | [TK]D-Fender, Well. There you go. The ones I was looking at were T1 only. That's what happens when I stop paying attention for 24 months 8) |
| 21:42.35 | [TK]D-Fender | X-Rob: I'd love to know what you were loking at because I've been using my Samngoma A104d for 3 years now <- |
| 21:42.58 | ManxPower | [TK]D-Fender: he's talking about traditional telephony gear |
| 21:43.09 | ManxPower | not the software DSP based cards for PCs |
| 21:43.12 | pepse | what would cause asterisk to not be able to register, giving "registration timed out" over and over? At the same time, I can register to the same account with a softphone. |
| 21:43.17 | X-Rob | [TK]D-Fender, I spent _ages_ searching for a reasonable E1 hw ec, and couldn't find 'em. Plenty of T1 ones for about US$1500 |
| 21:43.28 | X-Rob | the E1s were all 2500+ |
| 21:43.29 | [TK]D-Fender | pepse: wrong NAT settings |
| 21:43.41 | pepse | is there anything besides "NAT=yes"? |
| 21:43.50 | [TK]D-Fender | X-Rob: Sangoma's to both w/ HWEC |
| 21:43.52 | [TK]D-Fender | ~sipnat |
| 21:43.52 | jbot | [~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 |
| 21:43.53 | [TK]D-Fender | ^^^^^^^^^^^^ |
| 21:43.54 | ManxPower | Bah! My 4 x T-1 HW EC box sold yesterday for $249 8-) |
| 21:43.59 | [TK]D-Fender | pepse: YES. Read up |
| 21:50.05 | *** part/#asterisk AndyML (n=quassel@pool-96-227-91-204.phlapa.fios.verizon.net) |
| 21:50.10 | *** join/#asterisk telecos (n=sergio@153.166.219.87.dynamic.jazztel.es) |
| 21:53.37 | jks | anyone seen a utility that can extract the audio from captured iax2 packets? |
| 21:55.37 | *** join/#asterisk neobsd (n=neobsd@190.81.184.1) |
| 21:55.39 | neobsd | hi |
| 21:55.41 | neobsd | please |
| 21:55.49 | pepse | i wonder if magicjack's SIP servers are behind NAT |
| 21:55.56 | pepse | i would guess no, but you never know |
| 21:56.05 | neobsd | how i can configure etsi please ? |
| 21:56.07 | neobsd | . |
| 21:56.37 | pepse | can I set externalIP= to a hostname? will Asterisk resolve it and fill in proper IP? |
| 21:57.07 | [TK]D-Fender | pepse: "externhost=" + "externrefresh" |
| 21:57.26 | pepse | externrefresh=yes or something? |
| 21:57.49 | neobsd | sorry, can you help me with ETSI please? |
| 21:57.49 | neobsd | . |
| 21:58.58 | pepse | [TK]D-Fender: I didn't know the register line had anything to do with that area.. |
| 21:59.39 | [TK]D-Fender | pepse: * has to know what IP's to tell the other end to answer back on |
| 21:59.39 | pepse | na, externalip= didn't make a difference :/ |
| 21:59.48 | [TK]D-Fender | pepse: pastebin your config |
| 21:59.56 | [TK]D-Fender | pepse: I already suspect 1 error... |
| 21:59.59 | pepse | chan_sip.c:6984 sip_reg_timeout: -- Registration for '<my user>@<myproxy>' timed out, trying again" |
| 22:00.01 | [TK]D-Fender | (specifically) |
| 22:00.07 | [TK]D-Fender | pepse: your CONFIG <--- |
| 22:00.10 | [TK]D-Fender | ~pb |
| 22:00.11 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 22:00.12 | [TK]D-Fender | ^^^^^^^ |
| 22:00.21 | pepse | http://pastebin.ca/1238118 |
| 22:00.49 | pepse | sorry, that doesn't include my externalip= line tho |
| 22:02.07 | pepse | oh, i already have externip in the top of my sip.conf. Heh |
| 22:02.19 | pepse | but i had set it to a hostname |
| 22:03.28 | pepse | changed it to externhost, no diff. |
| 22:04.47 | scooby2 | can Asterisk distinguish between free agents and agents on calls on new incoming calls? |
| 22:04.58 | [TK]D-Fender | pepse: And your ability to follow instructions also failing... |
| 22:05.33 | [TK]D-Fender | scooby2: Typically yes |
| 22:06.40 | scooby2 | [TK]D-Fender: i have a weird request. They want the caller to stay in the queue forever if an agent is available. Otherwise if no agent becomes available, leave after 60 seconds. |
| 22:07.13 | pepse | [TK]D-Fender: I know you enjoy belittling and all, but I'm not -that- new at this. There's one particular service giving me these problems, while all other service I've been using for some time work fine |
| 22:07.20 | pepse | and I've always been behind NAT |
| 22:07.24 | [TK]D-Fender | scooby2: there is no "leave after X time", only leave period |
| 22:07.40 | JohnnyScarlet | ok i'm back :p |
| 22:08.00 | [TK]D-Fender | pepse: And I asked to see your config and you showed me only a PIECE of it |
| 22:08.20 | JohnnyScarlet | {TK}D-Fender: If I never specified bind:0.0.0.0 port 5060 ... why am I getting the warning "Failed to bind to 0.0.0.0:5060: Address already in use" when running * |
| 22:08.50 | [TK]D-Fender | JohnnyScarlet: Let me guess... running a softphone on your server as well? |
| 22:10.03 | pepse | [TK]D-Fender: http://pastebin.ca/1238160 happy? |
| 22:10.16 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 22:10.33 | JohnnyScarlet | not running a softphone |
| 22:10.43 | [TK]D-Fender | pepse: you are missing "externrefresh" like I told you |
| 22:10.49 | JohnnyScarlet | this is jsut a simple ubuntu install, followed by my * install |
| 22:11.03 | [TK]D-Fender | JohnnyScarlet: Well something else is preventing * from binding to that port.... |
| 22:11.10 | pepse | [TK]D-Fender: i asked how to use externrefresh, but I didn't get an answer |
| 22:11.15 | pepse | [TK]D-Fender: is it externrefresh=yes? |
| 22:11.30 | [TK]D-Fender | pepse: "externrefresh=howmanysecondstorecheck" |
| 22:11.39 | [TK]D-Fender | pepse: 120 is "healthy" |
| 22:11.42 | pepse | i see |
| 22:11.47 | JohnnyScarlet | and the insecure=port, invite got me unknown flag error I wrote "insecure=5060,invite" isn't that the default port? Yes I am behind NAT |
| 22:11.57 | [TK]D-Fender | pepse: Next, what have you forwarded to *? |
| 22:12.02 | JohnnyScarlet | ahhh, thx, lemme check that out TK |
| 22:12.03 | *** join/#asterisk voxter (n=voxter@rrcs-67-53-210-147.west.biz.rr.com) |
| 22:12.12 | [TK]D-Fender | JohnnyScarlet: port the WORD, not a NUMBEr |
| 22:12.15 | pepse | hm, what do you mean by forwarded to? |
| 22:12.20 | pepse | oh ports |
| 22:12.26 | pepse | 5060 and 10000-20000 udp |
| 22:12.28 | [TK]D-Fender | JohnnyScarlet: "insecure=port,invite" <- literally... |
| 22:12.54 | pepse | i can use softphone clients from the outside connecting into my local net ok |
| 22:12.58 | [TK]D-Fender | pepse: Once you're done, pastebin SIP debug of your failed attempts |
| 22:13.04 | JohnnyScarlet | I will correct the insecure thing TK, thank you. I just ran an NMAP of the server, only thing in the 5000 range is VNC |
| 22:13.28 | [TK]D-Fender | JohnnyScarlet: who are your running * as? |
| 22:13.37 | pepse | and what debug level should I use? |
| 22:13.41 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 22:13.41 | pepse | err -and |
| 22:13.48 | [TK]D-Fender | pepse: "sip debug" <--- |
| 22:13.52 | JohnnyScarlet | r00t :( |
| 22:13.54 | pepse | k |
| 22:14.18 | [TK]D-Fender | JohnnyScarlet: thats good. pastebin your problems. |
| 22:14.56 | edibrac | i'm getting a "fuzzy" quality when I hear people speak - the other side can hear this too |
| 22:15.34 | edibrac | also over the last 5 months we've been having 3-4 PRI red alarms that last a few seconds |
| 22:15.57 | edibrac | er, 3-4 red alarms per month |
| 22:16.02 | *** join/#asterisk sp00k3y (n=sp00k3y@wsip-98-190-136-194.ph.ph.cox.net) |
| 22:16.08 | *** join/#asterisk flohack (n=fhackenb@chello084115131198.3.graz.surfer.at) |
| 22:17.04 | pepse | [TK]D-Fender: It says Retransmitting #5 (NAT) to <ip of proxy>:5060... but my register line clearly states 5070 |
| 22:17.33 | *** join/#asterisk [gquit]bombadil (n=dana@CPE-70-94-44-157.wi.res.rr.com) |
| 22:17.45 | pepse | anywhere else to specify the port to register on? |
| 22:18.27 | [TK]D-Fender | pepse: Ok, I am tired of not seeing proper configs and debug information for your situation. Perhaps someone else will assist you. |
| 22:18.48 | [TK]D-Fender | moves on to more productive things |
| 22:19.24 | pepse | [TK]D-Fender: I'm just trying to read the stuff before I go pasting all my goddamn info onto public paste sites, jeez. |
| 22:20.34 | pepse | would you want to paste all of your sip debug info for all the world to see? |
| 22:20.35 | *** join/#asterisk Kate-o (n=vajda1ka@unaffiliated/kateo/x-334924) |
| 22:20.37 | pepse | I would guess not. |
| 22:20.52 | [TK]D-Fender | pepse: Yes, I would, and have. |
| 22:21.22 | [TK]D-Fender | pepse: I've asked for this repeatedly. |
| 22:21.27 | pepse | as if trying to troubleshoot this stuff wasn't bad enough. |
| 22:21.47 | [TK]D-Fender | pepse: Try asking your mechanic whats wrong with your cal and not letting him start the engine. |
| 22:21.55 | [TK]D-Fender | car* |
| 22:22.10 | flohack | Hi! Can someone tell me why I do not get a correct call-time when using dynamic queue members (e.g. SIP/3901)? |
| 22:22.12 | [TK]D-Fender | pepse: Either way maybe someone else can tell without actually seeing whats going on |
| 22:22.17 | pepse | [TK]D-Fender: um, should i pastebin my lastlog of our conversation for you? :) |
| 22:22.18 | flohack | The call time is always set to 0 and I get an AgentComplete event as soon as my queue member picks up the call. |
| 22:22.38 | pepse | [TK]D-Fender: I'm trying to let you know what's going on, can you give me a second? |
| 22:23.23 | [TK]D-Fender | flohack: Have you checked that maybe the agent is indeed hanging up on yrou caller. I had one guy doing that for moths... was a bitch to track to human error |
| 22:23.37 | pepse | it's silly to pretend like you want to help and at the same time be so stand-offish |
| 22:23.57 | JohnnyScarlet | [TK}D-Fender here is my pastebin http://pastebin.centos.org/22420 these are the warnings I get |
| 22:23.57 | flohack | [TK]D-Fender: He is not, I'm testing the system and I impersonate the agent :-) |
| 22:24.12 | [TK]D-Fender | flohack: We still can't be sure if he wasn't dodging his job or just trigger happy on the headset button to pickup/hangiup back to back |
| 22:24.29 | stintel | hehe. reminds me of my 1st line days :P |
| 22:24.43 | pepse | [TK]D-Fender: I would also like to read it all so that next time I wouldn't need to ask anyone for help, and understand the problem myself |
| 22:24.48 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 22:25.48 | pepse | it's weird that the first time it does say "reliably transmitting nat" to the ip on 5070, but then later doesn't. I guess that's just a fallback feature or something. |
| 22:26.23 | flohack | [TK]D-Fender: I'm calling the queue from my mobile, click the pickup button on my softphone. As soon as I pick up the call I get an AgentComplete event on my AMi connection. The queue log is filled with the appropriate event (completecaller) and a call time of 0, the hold time is fine though |
| 22:26.52 | *** join/#asterisk murdock_ut (n=chatzill@mail.kimballequipment.com) |
| 22:26.57 | [TK]D-Fender | flohack: Ok, thats AMI, whats the queue dump say about the agent and call? |
| 22:27.11 | flohack | [TK]D-Fender: Give me a second |
| 22:28.02 | pepse | it does say Contact: <sip:s@myexternalip>.. |
| 22:28.05 | flohack | [TK]D-Fender: The agent is marked Busy and channels are like this: |
| 22:28.07 | flohack | SIP/3903-083a2878 (None) Up Bridged Call(Zap/19-1) |
| 22:28.07 | flohack | Zap/19-1 1002@default:1 Up Dial(local/9002) |
| 22:28.59 | [TK]D-Fender | flowpastebin it, and thats a queue dump I asked for, not a channel dump |
| 22:29.11 | pepse | my callid has my external ip as well |
| 22:29.35 | pepse | from line, contact line, everything. |
| 22:29.36 | JohnnyScarlet | [TK}D-Fender any idea how I screwed up TK? |
| 22:30.16 | pepse | i suppose you wouldn't want to give me a clue on what to look for, beyond just "show me all your stuff" |
| 22:31.17 | [TK]D-Fender | JohnnyScarlet: Looks like a conflict with USERS.CONF. |
| 22:31.34 | JohnnyScarlet | ahhh ok i will tend to that right away TK, thank you |
| 22:32.00 | [TK]D-Fender | pepse: I have no intention on flying blind. |
| 22:32.17 | flohack | [TK]D-Fender: Sorry, here is the pastebin: http://pastebin.com/m5e84db48 |
| 22:33.02 | *** join/#asterisk drumkilla (n=russell@asterisk/digium-open-source-team-lead/russellb) |
| 22:33.02 | *** mode/#asterisk [+o drumkilla] by ChanServ |
| 22:33.31 | flohack | [TK]D-Fender: Hope you meant 'queue show QUEUE' with 'queue dump' |
| 22:33.36 | [TK]D-Fender | flohack: ok, so they got the call. |
| 22:33.41 | *** join/#asterisk KingOfDos (n=irssi@ht1-ix.kingofdos.net) |
| 22:34.06 | flohack | [TK]D-Fender: Yes, I took the call myself and can hear me speaking from my mobile |
| 22:34.06 | [TK]D-Fender | flohack: Looks pretty normal.. maybe its an AMI bug... |
| 22:34.25 | JohnnyScarlet | HEY! there isn't a setting up USERS section in the table of contents |
| 22:35.09 | JohnnyScarlet | I call shenanigans on this book! |
| 22:35.45 | [TK]D-Fender | Hey, there was a chapter on motor-derby driving with my Prius manual! |
| 22:35.53 | [TK]D-Fender | goes and crashes his car |
| 22:35.58 | flohack | [TK]D-Fender: I'm not concerned with the AMi message, the problem is that the queue log is wrong. The AMi message is sent right after writing to the queue log as it is evident from app_queue.c. I suspect that asterisk bridges the call to the SIP device and terminates the channel which executed Queue() and therefore thinks that the call is complete. |
| 22:36.43 | [TK]D-Fender | flohack: I DO hav a suspicion about your use of chained Local channels on this. Call them with "/n" on the end so the don't "rebod" on answer |
| 22:37.02 | [TK]D-Fender | rebond* |
| 22:37.24 | *** join/#asterisk jer (n=jer@unaffiliated/jer) |
| 22:37.53 | flohack | [TK]D-Fender: But the queue dials the SIP device directly, I cannot pass anything. |
| 22:38.27 | *** join/#asterisk StooJ (n=stooj@stooj.plus.com) |
| 22:38.31 | [TK]D-Fender | flohack: -- Executing [9002@default:1] Answer("Local/9002@default-b6a7,2", "") in new stack |
| 22:38.45 | [TK]D-Fender | flohack: LOCAL... |
| 22:38.54 | [TK]D-Fender | flohack: change up that dial like I suggested |
| 22:39.04 | [TK]D-Fender | floIn your AGI |
| 22:39.19 | flohack | [TK]D-Fender: Ok you mean I should call the queue with /n appended |
| 22:40.17 | [TK]D-Fender | flohack: When you sue the local channel to GET to the queue. |
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| 22:45.45 | flohack | [TK]D-Fender: I'm really sorry, but I'm still unsure what to do. I do a Dial(Local/9002) from my agi script. If I understood you correctly, I should to a Dial(Local/9002/n)? I just could not find anything about appending /n in the docs. Just the n option to dial, which seems to be something else. |
| 22:46.09 | [TK]D-Fender | flohack: Yes |
| 22:46.41 | flohack | [TK]D-Fender: Or would a goto be better that a dial? |
| 22:47.05 | [TK]D-Fender | flohack: That too unless you need to come back to the AGI |
| 22:47.15 | *** join/#asterisk tobias (n=tobias@user-0ce2hvs.cable.mindspring.com) |
| 22:47.21 | JohnnyScarlet | ugh, i'm about to give up |
| 22:48.09 | JohnnyScarlet | no change, not to mention after this is runnign smooth, I still have to figure out hwo to properly configure this frickin' aastra phone(which obviously wouldn't be covered by the book) |
| 22:49.21 | neobsd | hi |
| 22:49.22 | StephenF[W] | Do you guys normally start your extensions with a higher number like 8 or 9 to avoid overlapping with IVRs? |
| 22:49.48 | neobsd | sorry how i can do a trunk between asterisk-gui and trixbox ? |
| 22:49.48 | neobsd | . |
| 22:52.29 | flohack | [TK]D-Fender: Thanks! I'll give it a try |
| 22:53.11 | *** join/#asterisk rcy` (n=rcy@S01060002553240a8.vc.shawcable.net) |
| 22:53.41 | JohnnyScarlet | thanks for tryign to assist [TK]D-Fender but i'm obviously out of my league here, too much to cram in my skull with all my other responsibilities. I jsut simply don't udnerstand all the jargon and protocols |
| 22:53.50 | flohack | [TK]D-Fender: Ok, I'm pretty sure that solves it, as calling the queue directly works as expected. Thanks a lot! |
| 22:54.00 | JohnnyScarlet | goodnight all |
| 22:56.15 | flohack | [TK]D-Fender: BTW, could you please point me to the docs explaining /n, I just can't find anything. |
| 22:56.29 | *** join/#asterisk nikko (n=nikko@12-218-250-137.client.mchsi.com) |
| 22:57.00 | [TK]D-Fender | flohack: When a locl channel is answered without it, it tries to rebridge and that can lose one of the call-id's |
| 22:57.17 | [TK]D-Fender | flohack: So while the call isn't lost, the remapping FUBAR's logs, etc |
| 22:59.21 | StephenF[W] | whats the deal with using exten = blahblah and exten => blahblah |
| 22:59.26 | StephenF[W] | which one is best practice? |
| 22:59.38 | [TK]D-Fender | StephenF[W]: No functional difference |
| 22:59.41 | StephenF[W] | ok |
| 22:59.43 | flohack | [TK]D-Fender: Ok, I see. That would then be a specific feature of chan_local.c ? |
| 23:02.21 | *** join/#asterisk jjg (n=jjg@12.40.200.74) |
| 23:02.56 | jjg | has anyone tried using the Qt Extended voip application that integrates the iaxclient library? |
| 23:03.05 | [TK]D-Fender | flohack: Not sure exactly where that is spread out |
| 23:03.40 | flohack | [TK]D-Fender: Ok I found the docs on it: doc/localchannel.txt |
| 23:05.50 | [TK]D-Fender | flohack: Almost looks like big print ;) |
| 23:06.38 | *** join/#asterisk blitzrage (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
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| 23:07.13 | lmadsen | OT: I have a friend who is selling a Polycom IP501, located in Georgetown, ON, Canada. Let me know if interested. |
| 23:10.16 | [TK]D-Fender | lmadsen: I've got one for sale here :) |
| 23:13.25 | *** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim) |
| 23:14.32 | flohack | [TK]D-Fender: Thanks for your help, gotta get some sleep. See you |
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| 23:16.44 | *** join/#asterisk hi365_m (n=hi365@213.151.62.217) |
| 23:17.38 | pepse | guh. crappy day is almost over. |
| 23:27.46 | *** join/#asterisk Thorn (n=thorn@unaffiliated/thorn) |
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| 23:36.56 | jameswf | OT: froen snickers are the greatest thing in the worls |
| 23:37.14 | jameswf | *frozen *world |
| 23:38.03 | unpaidbill | what about naked chicks |
| 23:38.13 | unpaidbill | or naked chicks with computers |
| 23:38.28 | unpaidbill | or just some computers |
| 23:39.58 | jameswf | I am a geek and I find myself annoying, if i dated a geek chick I would probably wanna punch her in the face, I prefer my women to be non techie |
| 23:40.33 | unpaidbill | i prefer my women to be dead but not yet cold |
| 23:40.53 | jameswf | I like my women like my coffee |
| 23:40.55 | unpaidbill | like a realdoll heated in a shower for a few weeks |
| 23:40.57 | jameswf | cold and bitter |
| 23:41.07 | unpaidbill | i thought you were going to say black and hot |
| 23:41.14 | jameswf | heh |
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| 23:55.07 | Katty | dumdedum |
| 23:55.36 | jeev | DOOD |
| 23:55.39 | jeev | RED ALERT 3, COMING SOON MOFOS |
| 23:55.48 | Katty | hmm. |
| 23:55.54 | Katty | russellb: ohai |
| 23:56.34 | jameswf | ping Qwell |
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| 23:57.12 | StephenF[W] | how do you guys prevent having to wait for timeout for overlapping extensions? For example I have a main IVR that all incoming calls are sent to, it has press 1 for this, press 2 for this, etc. and the option to dial an extension at any time. Our extensions are 2XX so selecting menu option 2 takes 5 secs to timeout... |
| 23:57.42 | StephenF[W] | I've thought of moving our extension to 8XX and never using option 8 in the menu, or making another menu item like press 3 to dial an extension... |
| 23:57.47 | StephenF[W] | any other cool ideas? |
