| 00:06.56 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.120) |
| 00:07.07 | bmoraca | i get my TA924 IAD this week for testing fun! woo! |
| 00:18.51 | rjg | ok I'm stuck |
| 00:19.23 | rjg | can anyone figure out why since installing and configuring an FXO card my local IAX extensions can't call each other (and can't be called from the PSTN)? |
| 00:19.47 | rjg | (any pointers in the right direction would be great) |
| 00:20.13 | rjg | <PROTECTED> |
| 00:20.16 | rjg | <PROTECTED> |
| 00:24.02 | rjg | [2009-05-11 01:23:25] WARNING[3309] app_dial.c: Unable to create channel of type 'IAX2' (cause 3 - No route to destination) |
| 00:25.20 | VaGoNeTaS | pastebin n' shit? |
| 00:26.24 | rjg | well what do you want? |
| 00:26.51 | rjg | bear in mind they've been created initially by freepbx so may be somewhat spammy |
| 00:32.55 | [TK]D-Fender | forget showing configs |
| 00:33.06 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 00:33.20 | [TK]D-Fender | rjg: pastebin "iax2 show peers" from * CLI |
| 00:33.47 | [TK]D-Fender | ~pb |
| 00:33.47 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste , or , http://bin.cakephp.org/ |
| 00:37.28 | rjg | http://pastebin.com/m6d48951b |
| 00:38.23 | [TK]D-Fender | rjg: 2002 (Unspecified) (D) 255.255.255.255 0 UNKNOWN |
| 00:38.35 | [TK]D-Fender | rjg: Your device has not registered to * and cannot be called |
| 00:39.34 | rjg | heh. |
| 00:39.46 | rjg | I'm a muppet, thanks [TK]D-Fender |
| 00:39.51 | rjg | although not entirely my fault |
| 00:40.00 | rjg | I was testing with DIAX, which turns out to be a bit quirky |
| 00:40.38 | rjg | no idea why I was sure the problem was with asterisk |
| 00:41.26 | [TK]D-Fender | rjg: Denial : It ain't jsut a river in Egypt |
| 00:42.45 | rjg | :) |
| 00:42.53 | rjg | right bed time for me, 01:42 here |
| 00:44.26 | eppigy | hello |
| 00:44.29 | eppigy | i am dave |
| 00:48.36 | jaytee | he is dave |
| 00:51.05 | eppigy | yes |
| 00:51.08 | eppigy | how are you jaytee |
| 00:51.30 | jaytee | fine, just got back from the new Star Trek movie |
| 00:51.36 | eppigy | I see |
| 00:51.39 | eppigy | did you enjoy it? |
| 00:51.48 | jaytee | yes, i did |
| 00:51.50 | eppigy | good |
| 00:52.17 | eppigy | to what degree do they show the desireable african american female unlcothed? |
| 00:52.38 | *** join/#asterisk nix8n82 (n=nate@mo-65-41-196-62.sta.embarqhsd.net) |
| 00:54.10 | jaytee | there's a little bit of a teaser at the beginning of the movie but not much. it's PG-13 after all |
| 00:55.25 | eppigy | :[ |
| 01:02.21 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-53fb1cda17cb91ae) |
| 01:02.32 | *** join/#asterisk nighty^ (n=nighty@210.188.173.245) |
| 01:13.18 | *** join/#asterisk blkry (n=blkry@24-241-119-056.dhcp.gnvl.sc.charter.com) |
| 01:15.16 | *** join/#asterisk freakazoid0223 (n=matt@pool-71-246-17-74.phlapa.fios.verizon.net) |
| 01:19.11 | *** join/#asterisk Kumbang (n=unknown@125.163.83.153) |
| 01:20.46 | *** join/#asterisk Aiatek (n=munoz@190.94.5.219) |
| 01:21.42 | *** join/#asterisk simprix (n=simprix@c-71-205-52-252.hsd1.mi.comcast.net) |
| 01:28.27 | jdblack | they didn't ruin star trek. I was so happy. |
| 01:34.26 | [TK]D-Fender | that's good to hear |
| 01:35.09 | jaytee | I liked it too |
| 01:37.55 | [TK]D-Fender | jaytee: Typical cheesy bad-guy? |
| 01:39.22 | *** join/#asterisk chendy (n=chatzill@58.251.100.254) |
| 01:40.11 | jaytee | [TK]D-Fender, I like to avoid details when a movie first comes out. some people get irritated about spoilers and such. |
| 01:40.33 | jaytee | but it ain't your grandma's Star Trek |
| 01:40.40 | [TK]D-Fender | jaytee: I'm sure you can answer without wrecking it. |
| 01:40.53 | [TK]D-Fender | jaytee: Was the bad guy overly cartoony or worthless? |
| 01:41.23 | jaytee | no, he was actually quite convincingly villainous |
| 01:41.50 | [TK]D-Fender | Insurrection = OMG, waste of good actor on Bad Guy. The last one with TNG.. ANOTHING f-ing Troi-mind-rape-this-time-as-awhole-f-ing-movie? |
| 01:43.14 | [TK]D-Fender | jaytee: Decent execution? viable motivation and follow-through? |
| 01:43.42 | jaytee | I liked First Contact and I did think the Sona role for F. Murray was a bit of a stretch he did an over the top job on that stinker but Nemisis? That turd should never have made it into post-production but then they have to try to salvage at least some box office cash once a turd is dumped. |
| 01:44.23 | [TK]D-Fender | jaytee: Yeah, Nemisis... BAH. |
| 01:44.38 | jaytee | in this one there are some slightly "cheesy" elements but it's more like a brie or a gouda, not some limburger or worse. |
| 01:45.02 | jaytee | and the tiresomely overused element of "time-travel" |
| 01:45.08 | [TK]D-Fender | jaytee: Any characters you felt they damaged? |
| 01:45.33 | jdblack | [TK]D-Fender: They did great. Seriously. |
| 01:46.16 | jdblack | Thogh there's a certain relationship that surprised me. Almost makes sense though. |
| 01:47.00 | jaytee | well, the new guy who plays Kirk lacks the sputtery speech delivery of over enunciated verbs and nouns that Shatner was famous for. Always reminded me of a car with a dying battery that just couldn't quite crank quickly and would never quite start. |
| 01:47.30 | jaytee | jdblack, yeah that romantic angle caught me off guard |
| 01:47.53 | jdblack | I kinda buy it though, upon reflection. How about you? |
| 01:48.04 | jdblack | I like this kirk better, btw. |
| 01:48.05 | jaytee | I really liked Karl Urban's portrayal of McCoy |
| 01:48.25 | jdblack | Yeah? Seemed like they pulled the McCoy cliche's a little too hard for me. |
| 01:48.55 | jaytee | jdblack, me too. One, he's not acting like he wants to make love to himself at every moment, doesn't hog the camera view and can actually act. |
| 01:49.22 | jdblack | Sulu was _fantastic_. Chekov.. DUnno. what do you think? |
| 01:50.02 | jaytee | Chekov was very different in this one but I liked the change. |
| 01:50.02 | jaytee | Sulu was ok |
| 01:50.02 | jdblack | Oh, they even killed a redshirt. :) |
| 01:50.02 | jaytee | nothing really special there. |
| 01:50.05 | jdblack | I almost forgot about that. |
| 01:50.08 | [TK]D-Fender | RED-SHIRT!!!!!!!!!!! |
| 01:50.10 | [TK]D-Fender | \o/ |
| 01:50.30 | [TK]D-Fender | Checkov & Sulu were always rather underdeveloped |
| 01:50.32 | jaytee | ya gotta have at least one red shirt die or it ain't a real star trek movie. everyone knows that. |
| 01:51.33 | jdblack | Yeah. I was really happy. They paid close attention to the traditions. JJ is either a trekie, or he talked to a lot of 'em. |
| 01:51.59 | jaytee | somewhere in some program God wrote that created the Universe is a rule-base and in it are random laws of the Universe like. "Electrons have negative charge, protons have positive charge, red shirt guy on away team dies everytime" |
| 01:52.11 | jdblack | And google says' he ain't a trekie. |
| 01:52.32 | jaytee | JJ said in early episodes that he'd never been a fan or really watched the whole series before making the movie. |
| 01:52.47 | jaytee | oops, meant early interviews |
| 01:53.08 | jdblack | Yeah. just read that. |
| 01:53.26 | *** join/#asterisk xuser (n=xuser@unaffiliated/xuser) |
| 01:53.35 | jdblack | I will be so pissed if they go back and save the whales in the next trek movie. |
| 01:54.44 | jaytee | supposedly they've already got a tentative green light for the sequel based on box office projections. |
| 01:56.48 | jaytee | jdblack, it's worst than that! They go back in time to get some humpback whales to take to the center of the galaxy to meet God and have lunch with Spock's half-brother. |
| 01:57.09 | jdblack | Ohhh. Will they recreate life on a planet on the way? |
| 01:57.25 | jaytee | yes, in order to revive Spock of course |
| 01:58.03 | jdblack | That reminds me.. I don't remember seeing Kahn |
| 01:58.19 | jaytee | Of the original movies I actually liked the one with the whales better than any of the odd numbered films. |
| 01:58.40 | jaytee | Khan was like the best of the original series movies |
| 01:59.03 | jaytee | or did you mean in this new film? |
| 01:59.09 | jdblack | Oh, and thank the gods that they didn't drag in too much DS-9 crap |
| 01:59.40 | jaytee | Someone was complaining that the whole bridge set looked too much like an Apple Store |
| 01:59.48 | [TK]D-Fender | jdblack: DS-9 is a far better platform for a movie actually. They HAD character development as opposed to all the others. |
| 02:00.09 | [TK]D-Fender | jdblack: Though Polluting THIS movie would jsut be stupid |
| 02:00.19 | jaytee | DS-9 had to have better character developement because DS-9 never "went" anywhere. |
| 02:00.35 | [TK]D-Fender | jaytee: They did, but they always wen't home :) |
| 02:00.39 | jdblack | didn't it travel through time a few times? |
| 02:01.25 | jdblack | Let's make a Voyager movie series! |
| 02:01.40 | jdblack | we can play 'em to captured terrorists |
| 02:01.45 | jaytee | still would have been nice if they'd thrown in the one redeeming part of Voyager as a "walk-through" and by that I mean Jeri Ryan's ass in a skin tight suit. |
| 02:02.16 | jdblack | I despired that character. |
| 02:02.43 | jdblack | despised too. |
| 02:03.31 | jaytee | to borrow a line from Chasing Amy, "all that girl needs is some serious deep dicking" |
| 02:03.49 | *** join/#asterisk terracon (n=greisky@CPE001cf097a750-CM0012254076d6.cpe.net.cable.rogers.com) |
| 02:03.55 | jdblack | [TK]D-Fender: btw, still quit smoking. |
| 02:04.02 | jaytee | congrats! |
| 02:04.09 | jdblack | I almost have my high priced hooker money saved. :) |
| 02:04.30 | jaytee | Qwell was going to try the electronic smokes but so far he's been mum on the subject |
| 02:04.38 | Qwell | they work |
| 02:04.44 | jdblack | Chantix did it for me. |
| 02:04.48 | jaytee | yeah? you liking them? |
| 02:04.53 | Qwell | jaytee: more or less |
| 02:05.06 | *** join/#asterisk maxagaz (n=maxagaz@soho2.i-xanadu.com) |
| 02:05.14 | jaytee | Chantix? I'm real leary of anything that messes with the brain chemistry |
| 02:05.33 | jdblack | Few consumed things don't. |
| 02:06.26 | *** join/#asterisk werdan7 (n=w7@freenode/staff/wikimedia.werdan7) |
| 02:10.53 | Aiatek | hi im trying to connect two Asterisk PBXs using a T1 Trunk with ami,d4 signalling coding |
| 02:11.06 | Aiatek | and with e&m_w |
| 02:11.44 | *** join/#asterisk ingenius (n=alektro@host176.190-230-72.telecom.net.ar) |
| 02:12.22 | jaytee | because PRI is just too damn hard! |
| 02:12.45 | Aiatek | do i need to add something like pri_net and pri_cpe |
| 02:12.46 | Aiatek | ? |
| 02:12.53 | [TK]D-Fender | jdblack: Congrats.... |
| 02:13.15 | [TK]D-Fender | Aiatek: Why would you connect 2 *'s via PRI? |
| 02:13.17 | jaytee | no, not if they're using ami,d4 and e&m_wink |
| 02:13.32 | [TK]D-Fender | Aiatek: Sad waste of expensive cards and limited channels as well |
| 02:13.43 | jaytee | yeah? why not just plain SIP or IAX over ethernet? |
| 02:13.52 | Aiatek | its in my lab testing |
| 02:14.17 | [TK]D-Fender | Aiatek: AMI,D4 sounds like RBS not PRI |
| 02:14.25 | Aiatek | exactly |
| 02:14.29 | [TK]D-Fender | Aiatek: And what is the point of this "test"? |
| 02:14.49 | Aiatek | but i want to know if there are some lines like pri |
| 02:15.45 | Aiatek | because in pri you need to add whos going to be pri_net or pri_cpe |
| 02:16.24 | [TK]D-Fender | Aiatek: For RBS there aren't |
| 02:16.27 | carrar | Just like Dom/Sub |
| 02:16.35 | Aiatek | <[TK]D-Fender> to connect a legacy PBX for example |
| 02:17.05 | Aiatek | ok |
| 02:19.06 | VaGoNeTaS | tk |
| 02:19.12 | VaGoNeTaS | i can make local calls |
| 02:19.20 | VaGoNeTaS | and longdistas but not to cellphones |
| 02:19.36 | VaGoNeTaS | and i can receive incoming calls |
| 02:20.06 | VaGoNeTaS | weird thing coz the installer he just called me from the E1 line with some kind of instrument |
| 02:20.13 | VaGoNeTaS | to my cell |
| 02:20.18 | VaGoNeTaS | so is not blocked from pstn |
| 02:20.24 | Aiatek | can you paste your dial plan |
| 02:20.28 | VaGoNeTaS | yes |
| 02:20.31 | VaGoNeTaS | its very simple |
| 02:20.34 | jaytee | headdesks |
| 02:20.40 | Aiatek | not here |
| 02:20.42 | VaGoNeTaS | yes |
| 02:20.43 | VaGoNeTaS | pb |
| 02:20.44 | VaGoNeTaS | i know |
| 02:22.17 | VaGoNeTaS | http://pastebin.ca/1418838 |
| 02:22.20 | eppigy | TRABAJO |
| 02:22.22 | VaGoNeTaS | that's my dialplan |
| 02:22.37 | VaGoNeTaS | i can make local, and longdistance on my country |
| 02:22.50 | VaGoNeTaS | but not to cellphones from asterisk |
| 02:23.31 | *** part/#asterisk [TK]D-Fender (n=joe@64.235.218.194) |
| 02:23.44 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194) |
| 02:24.14 | [TK]D-Fender | WTF.. that was wierd |
| 02:24.21 | VaGoNeTaS | what |
| 02:24.30 | [TK]D-Fender | VaGoNeTaS: So whats this new PB I just missed? |
| 02:24.35 | [TK]D-Fender | (client issues) |
| 02:24.43 | VaGoNeTaS | no, is just the dialplan from extensions.conf |
| 02:24.50 | *** join/#asterisk ZX81 (n=matt@202.20.97.211) |
| 02:25.02 | VaGoNeTaS | what i was saying is that im able to make local calls |
| 02:25.07 | ZX81 | hi all, anyone have any ideas on a good way to measure Post Dial Delay? |
| 02:25.12 | VaGoNeTaS | and longdistance on my country but not to cellphones |
| 02:25.25 | VaGoNeTaS | but im almost sure that is not blocked from my provider |
| 02:25.43 | VaGoNeTaS | coz the installer called me to my cell from his instrument the day he just installed the E1 line on the office |
| 02:26.01 | ZX81 | pridialplan? |
| 02:26.06 | VaGoNeTaS | is unknown |
| 02:26.16 | Aiatek | when you dial a cellphone what output do you have |
| 02:26.17 | Aiatek | ? |
| 02:26.19 | ZX81 | have you tried others |
| 02:26.22 | VaGoNeTaS | CHANUNAVAIL |
| 02:26.40 | VaGoNeTaS | i dont know more... |
| 02:27.01 | ZX81 | are there not examples in the config file? |
| 02:28.14 | VaGoNeTaS | yes, the example says pridialplan=national |
| 02:28.33 | VaGoNeTaS | when i set up the pridialplan to that i got CHANUNAVAIL when i make local calls too |
| 02:28.51 | [TK]D-Fender | VaGoNeTaS: "pridialplan" and "prilocaldialplan" are often best set as "unknown" |
| 02:29.05 | ZX81 | which he has it set to and cellphone calls are failing |
| 02:29.25 | VaGoNeTaS | i have only one, pridialplan=unknown |
| 02:29.28 | [TK]D-Fender | VaGoNeTaS: And the problem has nothing to do with dialplan unless you are missing required prefixes. All of this is stuff a telco tech should be able to answer for you in a handful of seconds. |
| 02:29.31 | VaGoNeTaS | im gona paste the chan_dahdi.conf |
| 02:29.36 | VaGoNeTaS | so u can see |
| 02:29.37 | ZX81 | trying adding or removing an extra 0 |
| 02:29.44 | [TK]D-Fender | VaGoNeTaS: there are only these 2 parms to play with |
| 02:29.47 | ZX81 | before the cell number |
| 02:29.57 | VaGoNeTaS | k just a sec |
| 02:30.03 | [TK]D-Fender | VaGoNeTaS: there is "unknown" , "national", and "international" AFAIK. |
| 02:30.08 | [TK]D-Fender | VaGoNeTaS: Call your telco |
| 02:31.01 | VaGoNeTaS | but when he called me from his instrument the call went through which is the weird thing |
| 02:31.02 | *** join/#asterisk simprix (n=simprix@c-71-205-52-252.hsd1.mi.comcast.net) |
| 02:31.26 | ZX81 | I assume his instrument was connected directly to the PRI |
| 02:31.30 | VaGoNeTaS | yes |
| 02:31.37 | VaGoNeTaS | i mean |
| 02:31.41 | VaGoNeTaS | to the E1 line |
| 02:31.52 | VaGoNeTaS | which is now connected to my Redfone quad box |
| 02:32.04 | ZX81 | so, say your cell number is 012 345, try calling 0012 345 and 12 345 |
| 02:32.11 | VaGoNeTaS | i did |
| 02:32.15 | VaGoNeTaS | dint go throguh |
| 02:32.17 | ZX81 | same result |
| 02:32.18 | VaGoNeTaS | through* |
| 02:32.18 | ZX81 | ok |
| 02:32.19 | [TK]D-Fender | VaGoNeTaS: You know all of this "but it worked with his miracle device" don't help you one bit. Just stop saying it OK? Ask your telco what they don't like about what you're sending |
| 02:32.20 | VaGoNeTaS | yep |
| 02:32.52 | VaGoNeTaS | im not saying that it was a miracle device im saying that the cellphone calls are available on this line |
| 02:32.58 | VaGoNeTaS | they are not being blocked |
| 02:33.08 | ZX81 | :) as they are on most lines :) |
| 02:33.18 | ZX81 | but he presented it differently from how you are doing now |
| 02:33.20 | VaGoNeTaS | haha yep |
| 02:33.25 | ZX81 | the telco will be able to say |
| 02:33.34 | ZX81 | man you should have dialed 00000000 first |
| 02:33.36 | ZX81 | or something |
| 02:33.43 | ZX81 | or that some setup is wrong |
| 02:33.47 | VaGoNeTaS | hahahah maybe |
| 02:33.55 | VaGoNeTaS | but, we have more lines with the same company here |
| 02:34.06 | ZX81 | usually when one type of call works and the other doesn't, it's the dialplan |
| 02:34.20 | VaGoNeTaS | actually i have another new callcenter with the same provider and the same same dialplan and the local and cellphones goes through |
| 02:34.26 | ZX81 | they could have disabled cell calls after they left too |
| 02:34.28 | ZX81 | yeah |
| 02:34.35 | ZX81 | but every line can be set up different |
| 02:34.38 | VaGoNeTaS | the only differences between them is the pri card |
| 02:34.54 | VaGoNeTaS | one its a 4 ports sangoma and the other one a 4 ports redfone |
| 02:35.17 | ZX81 | normally that wouldn't make a difference - although if you wanted to test you could swap them |
| 02:35.22 | VaGoNeTaS | yes, that could've be a reason |
| 02:35.25 | VaGoNeTaS | that makes sense |
| 02:35.39 | VaGoNeTaS | they maybe just blocked the cellcalls after they left |
| 02:36.06 | VaGoNeTaS | yep, but i cant |
| 02:36.12 | VaGoNeTaS | coz that call center is 24/7 |
| 02:36.14 | ZX81 | yeah - I've had telcos do some strange things :) the problem is getting them to respond to you |
| 02:36.20 | VaGoNeTaS | its a private hospital |
| 02:36.26 | ZX81 | right |
| 02:36.40 | ZX81 | meh - hospitals don't need phones :D |
| 02:36.56 | VaGoNeTaS | im in Chile, south america, belive me, they does |
| 02:37.04 | ZX81 | :D |
| 02:37.11 | VaGoNeTaS | xD |
| 02:38.28 | WindBack | VaGoNeTaS, hola VaGoNeTaS |
| 02:38.37 | VaGoNeTaS | hi |
| 02:38.38 | WindBack | VaGoNeTaS, from chile |
| 02:38.41 | VaGoNeTaS | yes |
| 02:38.50 | VaGoNeTaS | tu eres argentino |
| 02:39.04 | WindBack | VaGoNeTaS, and are you a lazy person? |
| 02:39.06 | WindBack | ha ha |
| 02:39.34 | [TK]D-Fender | WindBack: So far can't seem to bother calling the telco for support.... could be... |
| 02:39.59 | WindBack | WindBack, yes I'm from argentina |
| 02:40.43 | Aiatek | did you tried doing a call from a console |
| 02:41.03 | VaGoNeTaS | Aiatek you're talking to me? |
| 02:41.05 | VaGoNeTaS | if you do |
| 02:41.06 | VaGoNeTaS | yes i did |
| 02:41.10 | VaGoNeTaS | several tests |
| 02:41.13 | Aiatek | ok |
| 02:41.22 | Aiatek | change your provider |
| 02:41.25 | Aiatek | :P |
| 02:41.26 | VaGoNeTaS | ahahaha |
| 02:41.32 | WindBack | [TK]D-Fender, sorry, i didnt understand you |
| 02:41.42 | VaGoNeTaS | if that would've be easy i would |
| 02:42.20 | VaGoNeTaS | and im pretty sure that it has something to do with the asterisk configuration and not with the provider |
| 02:42.29 | VaGoNeTaS | coz we have it with 3 callcenters that we own |
| 02:42.33 | VaGoNeTaS | and they works properly |
| 02:42.45 | VaGoNeTaS | that's whay pisses me off |
| 02:43.11 | [TK]D-Fender | VaGoNeTaS: It takes 2 to communicate, and they are getting what you are saying, and not liking it. Well holy shit they'd beeter be able to tell you WHY. |
| 02:43.16 | [TK]D-Fender | better* |
| 02:43.41 | [TK]D-Fender | VaGoNeTaS: Stop fucking guessing , get off your ass and ASK THEM. What is so damn hard about this? |
| 02:43.52 | WindBack | Someboy undertand why, the h extensions is not executed if the called party doesnt pick up the phone inside a macro in asterisk 1.6.1?? |
| 02:44.02 | *** join/#asterisk esaym (n=user@cpe-24-174-186-34.satx.res.rr.com) |
| 02:44.40 | jaytee | because it wasn't a hangup, it was a timeout, you want the t extension |
| 02:45.25 | [TK]D-Fender | WindBack: Show us |
| 02:46.16 | WindBack | jaytee, yes, but if I want to execute something when the caller hungup the phone, now I can't |
| 02:46.31 | WindBack | [TK]D-Fender, to show what?? |
| 02:46.50 | jaytee | WindBack, you're contradicting yourself |
| 02:47.07 | [TK]D-Fender | WindBack: Your dialplan and a call being processed that doesn't work as you're expecting / hoping |
| 02:47.28 | WindBack | [TK]D-Fender, |
| 02:47.30 | WindBack | ok |
| 02:47.38 | WindBack | let me pastebin it |
| 02:48.02 | jaytee | sees [TK]D-Fender readying his grip on the infamous ClueBat |
| 02:49.52 | WindBack | [TK]D-Fender, http://www.pastebin.ca/1418859 |
| 02:50.24 | WindBack | [TK]D-Fender, at the moment, i'm not needing this behaviour |
| 02:50.25 | [TK]D-Fender | jaytee: WTF is it really? I don't get how people are so unfathomably stupid about what to show and have to be bludgeoned into doing so? "Hi my car radio doesn't work, whats wrong with it?". "Show it to me", "You mean open the trunk? Or the Glove-box?" |
| 02:50.58 | jaytee | lol |
| 02:51.10 | [TK]D-Fender | jaytee: And then give you HALF of what you asked for. |
| 02:52.57 | jaytee | well, I got confused about the whole h extension thing when he clearly stated, "the h extensions is not executed if the called party doesnt pick up the phone". Of course it wouldn't! the call was never properly completed in the first place, ergo the t extension for a timeout would be more appropriate but that's just me making a wild ass guess. |
| 02:54.58 | WindBack | jaytee, in previos versions of asterisk, when the called doesnt pick up the phone, and the caller hangup, the extension h also exectues |
| 02:55.15 | *** join/#asterisk saftsack (n=saftsack@p579244CA.dip.t-dialin.net) |
| 02:55.36 | WindBack | jaytee, now, it only execute if the called pick up the phone and some of the hung up |
| 02:55.59 | jaytee | ah, when the caller hangs up the phone. you didn't specify that earlier. |
| 02:56.19 | WindBack | jaytee, I dont undertand why this new behaviour |
| 02:57.37 | [TK]D-Fender | WindBack: Where's the rest? |
| 02:57.48 | WindBack | jaytee, I also saw in I dev-mailist that now, if the caller hung up the phone before the called pick up, only execute h in the main context |
| 02:58.11 | WindBack | for me, this behaviour is not consistent |
| 02:59.16 | WindBack | [TK]D-Fender, http://www.pastebin.ca/1418864 |
| 02:59.34 | WindBack | [TK]D-Fender, here, the context where I execute my macro |
| 03:00.04 | jaytee | and I don't see an h extension in there anywhere |
| 03:00.29 | WindBack | jaytee, since my h exten is in the macro and not in the main context |
| 03:01.48 | *** join/#asterisk simprix (n=simprix@c-71-205-52-252.hsd1.mi.comcast.net) |
| 03:02.22 | WindBack | jaytee, I repeat, At the moment, i'm not needing this behaviour, (you can see my line: exten => h,1,GotoIf($[$[${VALIDCALL} = 0] & $["${DIALSTATUS}" = "ANSWER"]]?2:3) |
| 03:02.26 | [TK]D-Fender | WindBack: Where's the rest? |
| 03:03.17 | WindBack | [TK]D-Fender, do you want al the dialplan?? |
| 03:03.53 | [TK]D-Fender | WindBack: .... I see I'm just wasting my time here.... |
| 03:04.33 | WindBack | [TK]D-Fender, why you said that?? |
| 03:05.48 | WindBack | [TK]D-Fender, I was trying to understand that change in the release 1.6.1, why you are wasting your time? |
| 03:19.38 | nkohh | hi everyone. I can play sound digits with Playback, but when parking calls, they don't play. what can I do to troubleshoot this? I'm using 1.4.24.1, and I see in console it playing the sound files. |
| 03:24.59 | *** join/#asterisk CunningPike (n=CunningP@S01060014bf81366b.vc.shawcable.net) |
| 03:26.12 | *** join/#asterisk neuro9 (n=test@c-67-167-49-106.hsd1.il.comcast.net) |
| 03:26.24 | neuro9 | hey everyone! |
| 03:26.30 | nkohh | hey!!!! |
| 03:26.52 | neuro9 | okay, anyone familiar with the AMI Redirect command? |
| 03:29.21 | neuro9 | I have an issue where I try to redirect to a different context and it instantly hangs up the channel |
| 03:30.46 | neuro9 | documentation is a little lax for manager debugging, could anyone be of assistance? |
| 03:45.54 | neuro9 | i get a Success response but a hangup immediatly after it |
| 03:48.16 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 03:48.42 | *** join/#asterisk ToyotaGuy23 (n=matt@cpe-075-183-017-135.triad.res.rr.com) |
| 03:48.46 | neuro9 | hey |
| 03:48.59 | neuro9 | you guys good asterisk users? |
| 03:50.42 | drmessano | Nah, we suck |
| 03:50.46 | ToyotaGuy23 | LOL! |
| 03:50.47 | drmessano | Most of us, anyway |
| 03:50.58 | neuro9 | hmm |
| 03:51.00 | neuro9 | dang |
| 03:51.13 | neuro9 | any chance of assistance with the Redirect issue? |
| 03:51.30 | neuro9 | or am i on my own w/ this one |
| 03:51.55 | [TK]D-Fender | neuro9: Ever heard of PASTEBIN before? |
| 03:52.06 | neuro9 | no, whats that? |
| 03:52.06 | drmessano | Your association between your problem and us being "good asterisk users" is puzzling |
| 03:52.13 | [TK]D-Fender | ~pb |
| 03:52.14 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste , or , http://bin.cakephp.org/ |
| 03:52.16 | [TK]D-Fender | ^^^^^^^ |
| 03:52.32 | drmessano | neuro9: Do you own a weedwacker? |
| 03:52.47 | neuro9 | used to. i sold it for a snack |
| 03:54.05 | drmessano | http://www.lewishillbillies.com/phpbb/viewtopic.php?p=37119#p37119 |
| 03:55.57 | neuro9 | http://pastebin.com/d21c560d1 |
| 03:56.44 | [TK]D-Fender | neuro9: Dialplan please |
| 03:57.50 | ToyotaGuy23 | is a complete pbx / asterisk newbie himself |
| 03:58.16 | ToyotaGuy23 | is *NOT* however a logic loving redneck =) |
| 03:58.24 | neuro9 | http://pastebin.com/m4fe7fb18 |
| 03:58.38 | drmessano | ToyotaGuy23: Do you own a weedwacker? |
| 03:58.42 | neuro9 | i use mysql realtime, so those are the corresponding lines |
| 03:58.45 | ToyotaGuy23 | ABSOLUTELY! |
| 03:58.55 | drmessano | ToyotaGuy23: hell yesh |
| 03:59.00 | ToyotaGuy23 | 4 stroke |
| 03:59.13 | drmessano | ToyotaGuy23: Thats what she said! |
| 03:59.14 | neuro9 | I use originate to connect to extension 80 temporarily |
| 03:59.34 | neuro9 | then at a code defined moment, redirect to 82 |
| 03:59.38 | ToyotaGuy23 | does asterisk get along with ubuntu distros? |
| 03:59.48 | drmessano | ToyotaGuy23: Pretty much.. |
| 04:00.26 | drmessano | ToyotaGuy23: Asterisk gets along with anything the person installing/admining is comfortable with. |
| 04:01.37 | neuro9 | a telling line is the Hangup cause txt: "Cause-txt: Normal Clearing" |
| 04:01.43 | neuro9 | im research that atm |
| 04:01.44 | ToyotaGuy23 | drmessano, i was going to use pbx in a flash, but it comes with CentOS, and learning debian / ubuntu / gOS was a recent thing for me. I don't mind learning rpm systems, but now is just not the time. |
| 04:01.59 | [TK]D-Fender | neuro9: neuro9 Where is that last PB from? |
| 04:02.15 | ToyotaGuy23 | installing asterisk is going to be a 'compile from source job' then |
| 04:02.17 | ToyotaGuy23 | ? |
| 04:02.44 | neuro9 | TK: from my mysql database |
| 04:02.45 | [TK]D-Fender | ToyotaGuy23: Always suggested regardless. |
| 04:02.47 | drmessano | ToyotaGuy23: If youre gonna use something like PIAF, then no |
| 04:03.05 | drmessano | ToyotaGuy23: It does the work.. and your mileage will vary on that |
| 04:03.11 | [TK]D-Fender | neuro9: Show me the rest of the configs that show that * will refer to that DB for the extens in the first place |
| 04:03.25 | [TK]D-Fender | neuro9: And what ver of *? |
| 04:03.26 | neuro9 | key, context, extension, prio, app, data |
| 04:03.33 | neuro9 | 1.6 |
| 04:03.36 | drmessano | ToyotaGuy23: Asterisk can be installed from packages or source on a variety of platforms. Its up to you which suits you best |
| 04:04.13 | neuro9 | [realtime] |
| 04:04.13 | neuro9 | switch => Realtime/@ |
| 04:04.30 | [TK]D-Fender | neuro9: Ok, looking mostly sane so far |
| 04:04.47 | neuro9 | yea, everything works fine w/ the realtime db |
| 04:04.52 | neuro9 | i use it for other extensions |
| 04:04.53 | [TK]D-Fender | neuro9: Do other extens work in there? Can you originate the exten you are trying to redirect to directly? |
| 04:05.01 | neuro9 | originate works |
| 04:05.07 | [TK]D-Fender | neuro9: Against your targer? |
| 04:05.10 | [TK]D-Fender | target* |
| 04:05.13 | neuro9 | i even tried redirecting to a hardcoded context |
| 04:05.16 | neuro9 | same result |
| 04:05.23 | neuro9 | yea |
| 04:05.43 | [TK]D-Fender | neuro9: Ok, Your testing is looking pretty solid so far. Show us an actual call attempt & transfer |
| 04:05.56 | neuro9 | k just a sec |
| 04:09.38 | neuro9 | http://pastebin.com/d699bcbb4 |
| 04:09.54 | neuro9 | theres a redirect success in there but it didnt copy |
| 04:10.29 | *** join/#asterisk MrNaz (n=mrnaz@ppp118-208-187-219.lns10.mel4.internode.on.net) |
| 04:11.03 | neuro9 | im still confused what the normal call clearing hangup reason is for |
| 04:11.17 | neuro9 | i'm not finding docs on that response other than its an exit code |
| 04:12.18 | ToyotaGuy23 | drmessano, so if i get the packages from the ubuntu repos, I will need an IP phone (or adapter) and then an SIP provider and I will be set... or close? |
| 04:13.26 | neuro9 | my guess is thats a normal exit code |
| 04:13.42 | neuro9 | it doesnt correspond to an "error" per se in the RFC |
| 04:13.48 | neuro9 | its the only one that doesnt |
| 04:14.06 | drmessano | ToyotaGuy23: Thats pretty close |
| 04:14.21 | drmessano | ToyotaGuy23: Thats the blocks anyway |
| 04:15.05 | [TK]D-Fender | neuro9: Please provide the CLI output at verbose 10 as well. |
| 04:16.22 | neuro9 | i have it at verbose 100, i assume that would cover 10? |
| 04:17.18 | ToyotaGuy23 | drmessano, my wife is PO'ed at me . We work from home and need a phone for our business. I bought a MagicJack and two cordless phones. Well, the magic jack sucked terribly, and she is leery of me spending any more $$$ on hardware that she believes may not "work" |
| 04:17.26 | ToyotaGuy23 | it's a real problem =) |
| 04:17.50 | rob0 | Wives often are. |
| 04:17.51 | neuro9 | TK: no CLI output |
| 04:18.39 | [TK]D-Fender | neuro9: You're redirecting a call, I do want to see the call itself, a channel dump, etc. If it drops the call it'd show a drop as well |
| 04:18.47 | [TK]D-Fender | neuro9: Don't tell me there's NO evidence |
| 04:20.26 | ToyotaGuy23 | ok guys what else will i need on my grocery list? *phone or adapter / *SIP service / *Asterisk on ubuntu / <insert the rest here> |
| 04:21.04 | neuro9 | how can i channel dump? |
| 04:21.41 | [TK]D-Fender | neuro9: "core show channels concise". Also SIP debug would possibly help. and you HAVE to see the call get dropped off even with basic verbose... |
| 04:24.56 | ToyotaGuy23 | !help toyotaguy23 |
| 04:25.43 | [TK]D-Fender | ToyotaGuy23: Try in #drphil |
| 04:26.07 | drmessano | ToyotaGuy23: Asterisk / Provider / Phones or adapters / skill |
| 04:26.35 | neuro9 | SIP/1001-096eac08!realtime!80!2!Up!Wait!30000!8**5*2838!!3!2!(None)!1242016032.12 |
| 04:26.43 | neuro9 | thats when from core show channels conise |
| 04:26.51 | neuro9 | *from |
| 04:27.03 | *** join/#asterisk propellerhead (n=yogurt2u@host1.190-30-31.telecom.net.ar) |
| 04:27.05 | neuro9 | when its waiting in the first extension before the transfer |
| 04:27.39 | neuro9 | sip debugging yields nothing out of the ordinary |
| 04:27.51 | neuro9 | a BYE packet is sent upon disconnect |
| 04:28.50 | ToyotaGuy23 | drmessano, im pretty good with *nix in general, like the highest level of the newbie rank. what does the skill level look like really? Once it's working, i won't have to make adjustments often will I? that was a BIG problem with the magic jack. I was always messing with it. Drivers here, software there, it was a time issue |
| 04:29.21 | drmessano | ~book |
| 04:29.22 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 04:29.49 | drmessano | Do some reading.. you will need to learn how to make the dialplan work.. doesn't matter what your *nix skill is, you need to learn the app |
| 04:30.12 | drmessano | I know *nix dumbasses that can write badass dialplans, and vice versa |
| 04:30.35 | ToyotaGuy23 | have many people had success with debian or debian-like distros (ubuntu, gOS, etc)??? |
| 04:30.40 | drmessano | Lots |
| 04:30.44 | drmessano | Its not about the DISTRO |
| 04:30.47 | neuro9 | TK: nothing from core set debug channel SIP/1001-096eac08 either |
| 04:30.47 | drmessano | Its about the APP |
| 04:32.09 | drmessano | You obviously need to know your distro in order to install and make this work. Its more than just "yum install apache" |
| 04:32.18 | drmessano | But then you need to learn the app itself |
| 04:32.36 | drmessano | Probably 80/20 |
| 04:32.38 | ToyotaGuy23 | well my pal sets this up for companies for a living. and i know how annoying newbie questions can be, so im trying to bother you guys instead =) But he seems to prefer PBIAF. I just dont have the time or patience right now to learn CentOS |
| 04:33.04 | drmessano | ok |
| 04:33.23 | drmessano | I doubt your friend could help you much |
| 04:33.33 | ToyotaGuy23 | whys that? |
| 04:33.42 | drmessano | If hes setting up PIAF, he probably knows less about the raw asterisk config than you do now |
| 04:33.57 | drmessano | Its all point and click unless hes doing anything real custom |
| 04:34.12 | drmessano | You dont really touch the command line with a GUI install |
| 04:34.25 | ToyotaGuy23 | is asterisk not much gui? |
| 04:34.28 | ToyotaGuy23 | mostly command? |
| 04:34.31 | drmessano | Asterisk is 0 GUI |
| 04:34.37 | ToyotaGuy23 | ahh |
| 04:34.40 | neuro9 | TK: could using ExtraChannel make a difference? |
| 04:35.01 | ToyotaGuy23 | so its 5,000 new commands to learn / lookup |
| 04:35.04 | neuro9 | Toyota: don't jump on the asterisk gui bandwagon, learn the command line.. its very powerful |
| 04:35.23 | neuro9 | to rival most professional PBX's, like only 10-20 |
| 04:35.26 | drmessano | ToyotaGuy23: 4,986 |
| 04:35.38 | ToyotaGuy23 | would like to point out that neuro9 does *NOT* own a weed wacker |
| 04:36.18 | drmessano | You could use Asterisk to rival a Cisco system by learning less than 1/4 of the book |
| 04:36.26 | drmessano | Like, dumbass newb level |
| 04:36.31 | [TK]D-Fender | neuro9: Shouldn't be needed. Check the channel dump before and after |
| 04:36.49 | ToyotaGuy23 | neuro9, im just kidding bro |
| 04:37.17 | neuro9 | lol |
| 04:37.33 | drmessano | Weedwacker() |
| 04:37.55 | neuro9 | should i need another Answer() on the new extension even though the channels already answered? |
| 04:38.18 | neuro9 | cause i would believe Redirect functions like the Goto |
| 04:38.28 | ToyotaGuy23 | asks google about asterisk commands |
| 04:38.43 | [TK]D-Fender | neuro9: You should see dialplan execute instantly |
| 04:39.41 | neuro9 | how about the fact that i'm holding my extension in a WAIT? |
| 04:39.48 | neuro9 | ill try a park just to test |
| 04:40.10 | [TK]D-Fender | neuro9: I didn't want to point fingers at that... might be involved... change your test around a little |
| 04:41.29 | ToyotaGuy23 | just downloaded that book |
| 04:43.18 | ToyotaGuy23 | http://www.downloads.oreilly.com/books/9780596510480.pdf |
| 04:44.30 | ToyotaGuy23 | sorry take out www |
| 04:44.48 | ToyotaGuy23 | drmessano, how long have you used asterisk? |
| 04:45.19 | drmessano | 3 years |
| 04:45.28 | *** join/#asterisk thelordmortis (n=lordmort@203-8-160-250.secure.com.au) |
| 04:45.37 | drmessano | You dont need to link it |
| 04:45.41 | drmessano | ~book |
| 04:45.42 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 04:45.48 | drmessano | I linked you earlier |
| 04:45.54 | drmessano | Guess you didnt see that |
| 04:46.15 | neuro9 | TK: that was more telling.. i got some Rename events before it failed |
| 04:46.22 | neuro9 | that looks more like its supposed to i think |
| 04:46.26 | neuro9 | let me try something else |
| 04:47.39 | neuro9 | okay that worked :) i was in a park.. and i redirected to voicemail |
| 04:49.14 | ToyotaGuy23 | so there is no gui for asterisk on a *nix system? the windows version must be full gui (like every other windows app) |
| 04:51.14 | ToyotaGuy23 | off to bed thanks guys |
| 04:51.30 | ToyotaGuy23 | [TK]D-Fender, thanks for the #drphil =) |
| 04:51.33 | ToyotaGuy23 | lol |
| 04:52.31 | neuro9 | TK: okay whats your advice.. I want to open a channel with AMI originate.. hold it until needed.. and interject using redirect with different functions (like text to speech.. etc) |
| 04:53.12 | neuro9 | if i could put the originating extension on hold or park without hold music that would accomplish the first half |
| 04:53.34 | neuro9 | can i disable music on hold on a per channel basis? |
| 04:54.21 | drmessano | ..... |
| 04:55.27 | neuro9 | wait found that command |
| 04:56.57 | neuro9 | oh thats nice.. Park ignores it |
| 05:01.16 | [TK]D-Fender | neuro9: Send it to an exten that sets a null MoH class and then park |
| 05:01.37 | [TK]D-Fender | neuro9: Or a channel that does Wait(5) or other small enough increment in a loop |
| 05:02.36 | neuro9 | hmm k, i will try that |
| 05:04.06 | neuro9 | TK: wait didnt work. it jumps properly out of park |
| 05:04.55 | neuro9 | but i set the moh class to be "Set(CHANNEL(musicclass)=null)"? |
| 05:05.13 | neuro9 | and it says "701" when i park it |
| 05:05.16 | neuro9 | can i turn that off? |
| 05:07.01 | [TK]D-Fender | neuro9: Not afaik |
| 05:07.28 | [TK]D-Fender | neuro9: And for MoH, make an actual class whose folder has no sound files to play |
| 05:09.13 | neuro9 | okay |
| 05:09.16 | neuro9 | a dev/null deal |
| 05:20.03 | [TK]D-Fender | checkout time, later all |
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| 06:36.41 | stix_ | Hi guys. I remember that meetme once had an option where it should prompt the users to say their name when entering the conference. I can't seem to find that on voip-info.org anymore - anybody knows what I am talking about? |
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| 06:43.53 | tzafrir_laptop | http://www.villagetelco.org/2009/05/a-phone-mesh/ |
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| 06:52.48 | stix_ | Hi guys. I remember that meetme once had an option where it should prompt the users to say their name when entering the conference. I can't seem to find that on voip-info.org anymore - anybody knows what I am talking about? |
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| 08:06.09 | gpuk3 | hi all. I have a TDM411B digium card and would like to disable software echo cancellation on Channel 1 (the FXS port). In /etc/dahdi/system.conf I currently have echocanceller=mg2,1 for Channel 1. Should I set this to echocanceller= to disable echo cancellation? |
| 08:07.57 | *** join/#asterisk mintos (n=mvaliyav@nat/redhat-in/x-c55c8ae38760f53f) |
| 08:21.19 | gpuk3 | nm commenting out echocanceller=mg2,1 seems to do it |
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| 08:47.22 | ^shark_ | my company requested that we have a voip solution, i am looking for a business proposal example for the same to present to CEO |
| 08:47.44 | Superbartt | and your question being? |
| 08:49.22 | ^shark_ | where can i find a good business proposal for the asterisk voip solution |
| 08:50.12 | Superbartt | look at digium.com |
| 08:50.31 | Superbartt | at the frontpage there is like a huge header saying "VoIP Telephony Solutions for Business" |
| 08:50.40 | Superbartt | contact them :p |
| 08:51.04 | ^shark_ | ok let me give it a try |
| 08:52.03 | plundra | Are there any sip -> iax proxys out there? (Just protocol conversion, shouldn't be aware of any accounts or whatnot) |
| 08:52.41 | Superbartt | just setup asterisk to do it? :p |
| 08:54.04 | ^shark_ | ok i am not looking for any appliances or solutions from digium, cos i can setup asterisk with no problem, i am only looking for a business proposal written by some so that i can copy to suit our needs |
| 08:54.06 | plundra | I'd rather not, I would like to run it on slow alix boxes with limited everything. |
| 08:54.29 | plundra | And can you really just proxy with asterisk, without being aware of the accounts? (user/secret etc.) |
| 08:55.04 | Superbartt | well uhmm ^shark_, It seems to me you are trying to outsource the work you should be doing :p |
| 08:55.32 | Superbartt | yes plundra, It's quite possibly to allow anonymous accounts etcetera and just let them dial the same number over IAX |
| 08:55.42 | *** join/#asterisk lpccoder_ (n=lpccoder@81.203.255.90.dyn.user.ono.com) |
| 08:55.47 | lpccoder_ | hello |
| 08:56.00 | lpccoder_ | I have a question... I'm new to Asterisk |
| 08:56.10 | plundra | Superbartt: But can they be authenticated and registered? |
| 08:56.13 | lpccoder_ | My question is quite long... |
| 08:56.14 | ^shark_ | Superbartt: i do not know how to write a business proposal, i am looking for one then edit it |
| 08:57.03 | Superbartt | hmm plundra, registering would always require the users to be known to the system |
| 08:57.08 | lpccoder_ | Is possible to create a local VoIp network and make te outgoing calls go trough a mobile device? |
| 08:57.38 | lpccoder_ | Who can help me? |
| 08:58.08 | Superbartt | lpccoder_ that isn't a that long question, and yes |
| 08:58.15 | plundra | Superbartt: Yeah, but I want the end-system to be aware of it, not the middle man. Since that would give me either redundant configurations or atleast spread out, which is not prefereable if I want to move the users around. |
| 08:59.08 | Superbartt | hmm plundra, what exactly are you trying to achieve for the conversion? |
| 08:59.30 | lpccoder_ | well, wich mobile device can I use for this? |
| 08:59.56 | plundra | Superbartt: Not have to deal with sip behind nat problems, mainly. |
| 08:59.58 | Superbartt | junghanns has PCI GSM devices, look at http://www.voipango.com/isdn-interfaces/junghannsnet/pci-gsm/?lang=1 or http://www.junghanns.net/en/produkte.html |
| 09:00.52 | Superbartt | hmm plundra, I hope that's not the only reason, because on limited hardware as ALIX bord (planning to play with them too myself soon) a conversion to IAX is "cpu intensive" |
| 09:00.55 | lpccoder_ | is not possible to use a Nokia connected via USB, for example? |
| 09:02.00 | *** join/#asterisk intralanman (n=lanman@174-159-116-27.pools.spcsdns.net) |
| 09:02.03 | plundra | Superbartt: Really? Even with no codec-conversion (of course... :-P) And it's for small scale only, at most five users and not all making calls at the same time. |
| 09:02.38 | Superbartt | lpccoder_ You should go and google that... But I'm afraid not, It would require specific linux drivers (which I don't think nokia would make) and it would require the nokia to be able to initiate the call and send the call itself over usb... It has a reason why those GSM cards are that expensive |
| 09:03.03 | plundra | Superbartt: Maybe I'll try figuring out how to handle being behind nat. I havn't done much testing, but one time I was able to get back through the nat (inbound call), but at another time, I wasn't. |
| 09:04.13 | lpccoder_ | well... thanks for yous answers, I'll investigate a little and come back for any questions.. thanks a lot, again... |
| 09:04.39 | Superbartt | well plundra, I'm not sure if there aren't any protocol conversions the board just passes trough the sip+rtp to an iax channel or how that atually works |
| 09:04.48 | Superbartt | no problem lpccoder_ |
| 09:05.19 | wdoekes | Hi! In my realtime-dynamic setup with asterisk 1.4.24, asterisk sometimes forgets that registrations of some phones exist (including the phone that is calling!) yielding a busy tone. Asterisk then doesn't send out the username in the To: header which causes the INVITE'd phone to return a 404. My rt-settings are: rtcachefriends=no / rtupdate=yes / rtautoclear=yes. The question: what causes asterisk to forget both sip_pvt.fullcontact and sip_pvt.username? (I |
| 09:05.26 | Superbartt | and about the nat thing, I haven't had any problem when having my SIP port + a range of rtp's opened up... But none of my clients were behind nat, just the SIP telco |
| 09:06.35 | plundra | Superbartt: Sipproxys seem more common (just started googling that), I'll see about that or try without it again. |
| 09:06.39 | plundra | Superbartt: Thanks anyways :-) |
| 09:07.22 | *** join/#asterisk kamanashisroy (n=kamanash@119.30.36.11) |
| 09:08.56 | Superbartt | testing always is the best solution :p I'm not an expert (yet, waiting for my boss to pay for the Asterisk Advanced, and DcAP) but I would be glad if my experience would help someone out =) |
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| 09:19.31 | wdoekes | .. or perhaps I should've filled in the username field in the sip_friends table *tries* |
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| 10:22.35 | proxium | gm everyone, how to debug manager.conf please, it has a strange behaviour? |
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| 10:33.03 | HenrikBe | is there a way to pass the agent number of the agent who gets a call in a queue to php? |
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| 11:09.37 | mort_gib | Slow day |
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| 11:44.37 | IPGHOST | hi |
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| 11:51.09 | j_kroon | hi all, it seems that when iax2 auth fails on registration request there starts a loop of registartion attempts and auth required stuff. what is the common fix for this? a delay on the server before rejecting? |
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| 12:18.04 | j_kroon | hi, is it possible to configure dahdi to use software echo cancellation in spite of a hardware echo canceller being installed? If so, which one is the current recomendation? |
| 12:18.58 | *** join/#asterisk macros73 (n=cs_@dsl093-063-232.pit1.dsl.speakeasy.net) |
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| 12:39.05 | SuPrSluG | j_kroon:hardware is better most of the time |
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| 12:39.51 | j_kroon | SuPrSluG, i'm getting really bad voice breakups at at least two clients. as soon as i switch off the hardware echo can it's all nice and clear, but there is echo. |
| 12:39.59 | j_kroon | it's like the hardware thing is too effective. |
| 12:40.33 | j_kroon | i've fooled with different echocancel= settings, from 32 right through to 1024, 1024 is the best, at 32 it's like you're talking with a voice grater. |
| 12:41.15 | j_kroon | unless there is some paramaters I need to also pass around? |
| 12:41.16 | SuPrSluG | with hardware you shouldn't need any settings. |
| 12:41.38 | [TK]D-Fender | j_kroon: What are your gains set to? |
| 12:42.20 | j_kroon | [TK]D-Fender, at the one place that's having the issue quite badly +4.0 both on tx and rx, any lower and you can't hear the people (too soft) |
| 12:42.45 | [TK]D-Fender | j_kroon: What card? |
| 12:42.48 | j_kroon | i did run the fxotune stuff and my opermode is set to SOUTHAFRICA |
| 12:43.05 | j_kroon | in the worst of the cases a TDM800 with 2 x Quad FXO's, all digium. |
| 12:43.13 | Great_Anta_Baka | anyone here use ATCOM bri cards? |
| 12:43.47 | [TK]D-Fender | j_kroon: Do you know which VPM module? |
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| 12:45.01 | *** join/#asterisk |403Forbidden| (i=noneavai@freecode-project/coder/HttpError) |
| 12:45.22 | j_kroon | dmesg | grep VPM says revision 01, U-law mode, DTMF threshold set to 1250, enabling VPM100 gain adjustments on any FXO ports found, Present and operational (Rev B) |
| 12:45.48 | j_kroon | does that tell you what you wanted to know or is there something else I should be looking at? |
| 12:47.35 | j_kroon | the other client has VPMADT032: Present and operational (Firmware version 107) |
| 12:47.52 | *** join/#asterisk jstrom (i=johan@core.stromnet.se) |
| 12:47.54 | [TK]D-Fender | j_kroon: Yeah the VPM100 is the older crappy one (pre-Otasic) |
| 12:48.50 | *** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee) |
| 12:49.16 | j_kroon | ok, the one in the office (working very well) is also ADT032. however, my other client also using a VPMADT032 is having the same problem as the other client (restarting asterisk this morning fixed it for them) |
| 12:50.38 | j_kroon | in fact, those two clients are currently the only ones who have reported the issue, but if I listen very, very carefully I can hear it too on our line, but it's very, very slight. I need to really concentrate to hear the cut-out point. |
| 12:54.39 | *** join/#asterisk mindi (n=MinD@xdsl-87-78-161-229.netcologne.de) |
| 12:54.42 | mindi | hi |
| 12:55.12 | mindi | my asterisk doesnt start anymore since I installed chan_oss |
| 12:55.53 | mindi | "WARNING[11116]: chan_capi.c:5933 cc_init_capi: CAPI not installed, chan_capi disabled!" |
| 12:56.19 | mindi | im trying to use a avm b1pci-card |
| 12:56.41 | mindi | can anyone help me? |
| 12:56.58 | j_kroon | and that needs chan_oss why? |
| 12:57.25 | mindi | sry, its chan_capi |
| 12:57.31 | j_kroon | and thats a warning you're quoting there, not an error. ah ok. |
| 12:57.52 | mindi | its the last thing it says before it stops |
| 12:58.04 | j_kroon | well, looks like you're missing CAPI support somewhere, either in userspace, or maybe in the kernel? I've never worked with CAPI so guessing here. |
| 12:58.05 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 12:59.25 | mindi | ive got some probs to get the card workin properly either |
| 13:00.10 | mindi | is there a shell command to find out which IRQ my card is using? |
| 13:01.13 | SuPrSluG | cat /proc/interrupts |
| 13:01.53 | mindi | thx! |
| 13:02.21 | tzafrir_laptop | j_kroon, if the hardware echo canceller is used, it overrides the software one |
| 13:02.36 | tzafrir_laptop | so basically disable the hardware one first |
| 13:02.42 | j_kroon | tzafrir_laptop, so i need to physically go remove it in other words ... |
| 13:02.42 | *** join/#asterisk lesouvage (n=lesouvag@82.73.69.76) |
| 13:03.06 | j_kroon | or can i do it via some command? |
| 13:03.10 | tzafrir_laptop | j_kroon, no. There should be a module parameter or something. Or maybe even something in chan_dahdi.conf |
| 13:03.20 | tzafrir_laptop | (sorry, not really familiar with it) |
| 13:03.26 | j_kroon | goes poking around |
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| 13:04.28 | *** join/#asterisk sHoZaIb (n=sHoZaIb@dxb-as19540.alshamil.net.ae) |
| 13:07.06 | mindi | hm i still cant use capiinit |
| 13:07.32 | mindi | I need to configure /etc/isdn/capi.conf |
| 13:08.40 | mindi | maybe this is the reason why i cant get capi working |
| 13:09.29 | *** join/#asterisk theHub (n=theHub@69.177.93.21) |
| 13:09.50 | mindi | i need the cards I/O-adress too |
| 13:10.36 | mindi | Is there a shell-command for this? |
| 13:11.37 | Superbartt | lspci -vvv? |
| 13:13.05 | mindi | yes |
| 13:13.46 | mindi | gives me two I/O-adresses, region 1 & region 2 |
| 13:13.56 | mindi | which is the one i need? |
| 13:14.57 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
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| 13:15.48 | Superbartt | I have no idea mindi :) |
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| 13:15.55 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 13:16.47 | mindi | stupid question I guess :9 |
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| 13:17.35 | Superbartt | well, I just know lspci :P never worked with capi |
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| 13:56.37 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 13:57.15 | leifmadsen | Asterisk 1.6.2.0-beta2 is now available for download! |
| 13:58.21 | *** join/#asterisk blebleble (n=godie@s7.n194.n222.n216.static.myhostcenter.com) |
| 13:58.43 | blebleble | anyone ever run into a user in a queue transfers a call to another ring group (billing) then does not get a call again in the queue until the billing person hangs up? |
| 13:58.45 | *** join/#asterisk jmacz (n=jmacz@190.144.75.22) |
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| 14:05.29 | jaytee | 1.6.2.0-beta2? 1.6.1.0 was just released a little while back. how come no 1.6.1.1 or something like that? seems like a strange version jump |
| 14:06.13 | beek | morning jaytee |
| 14:06.19 | jaytee | morning beek |
| 14:06.39 | file | jaytee: they are different. |
| 14:06.45 | beek | jaytee: there's no 1.6.1.1 because there are no bugs in 1.6.1.0 ;-) |
| 14:07.40 | *** join/#asterisk ajohnson (n=ajohnson@63.147.46.186) |
| 14:07.50 | jaytee | file, obviously. I guess I'll never get used to linux versioning |
| 14:08.10 | *** join/#asterisk Beirdo (n=gjhurlbu@unaffiliated/beirdo) |
| 14:09.36 | mindi | Hi |
| 14:10.25 | *** join/#asterisk Nilzao (n=urubuh@200-161-87-81.dsl.telesp.net.br) |
| 14:10.47 | Nilzao | sup asterisk hackers |
| 14:11.31 | jaytee | same old same old |
| 14:11.32 | mindi | anyones got a clue of capi? |
| 14:11.32 | Nilzao | trying to find a way to make attended transfers with qutecom/wengophone softphone |
| 14:12.24 | *** join/#asterisk ingenius (n=alektro@netsolution.com.ar) |
| 14:12.43 | Nilzao | any bookmark? |
| 14:13.01 | jaytee | "So there I was talking to Wendy and then I whipped out my wengophone and she fainted!" |
| 14:13.24 | [TK]D-Fender | blebleble: This is a common issue. Tog et around it you need to do features.conf DTMF transfers. * doesn't detect the transfer and consider the member freed |
| 14:13.40 | Nilzao | lol |
| 14:15.01 | Nilzao | well as far i got |
| 14:15.06 | Nilzao | i put the atxfer=*2 |
| 14:15.20 | Nilzao | in the [featuremap] |
| 14:15.35 | Nilzao | nothing happens, and nothing show in the log file |
| 14:15.46 | Nilzao | restarted asterisk too |
| 14:16.13 | [TK]D-Fender | Nilzao: Know what else we don't see? The CLI output of your failed call attempt..... |
| 14:16.15 | [TK]D-Fender | ~pb |
| 14:16.16 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste , or , http://bin.cakephp.org/ |
| 14:16.18 | [TK]D-Fender | ^^^^^^^^ |
| 14:16.23 | *** join/#asterisk jasonwoot (n=some@bookit-dev.com) |
| 14:17.21 | Nilzao | d-fender, i will search more about dtmf transfers then i bother u again |
| 14:17.24 | *** join/#asterisk KavanS (n=KavanS@static-71-117-242-28.ptldor.dsl-w.verizon.net) |
| 14:17.33 | Nilzao | tx for now |
| 14:18.13 | [TK]D-Fender | Nilzao: Just PASTEBIN a failed call attempt. |
| 14:21.26 | *** part/#asterisk ^shark_ (n=Administ@41.222.2.65) |
| 14:21.37 | Aiatek | ~RBS |
| 14:21.52 | Aiatek | ~ RBS |
| 14:23.41 | Aiatek | <infobot> rbs |
| 14:27.51 | Nilzao | [TK]D-Fender: nothing at the CLI output about it too |
| 14:28.10 | [TK]D-Fender | Nilzao: ... PASTEBIN |
| 14:28.10 | Nilzao | [TK]D-Fender: just the ring, then the call pick up |
| 14:28.27 | Nilzao | [TK]D-Fender: man 2 lines ring, pick up, hang up |
| 14:28.45 | Nilzao | [TK]D-Fender: all ok, when press the *2 nothing changes |
| 14:29.44 | Nilzao | i will try to transfer the call in the cli console |
| 14:31.13 | Aiatek | how it works infobot |
| 14:31.15 | Aiatek | ? |
| 14:31.30 | Aiatek | i mean asking something to infoboot |
| 14:32.39 | Nilzao | Aiatek: i guess... its ~question |
| 14:32.46 | Nilzao | ~help |
| 14:33.56 | tzafrir_laptop | ~ask |
| 14:33.56 | infobot | 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. |
| 14:35.07 | *** join/#asterisk awk_r (n=awk_r@nat/digium/x-527a62add5ea5d36) |
| 14:35.26 | [TK]D-Fender | Aiatek: It doesn't have anything on RBS, doesn't matter how many different ways you ask |
| 14:37.29 | Aiatek | hehehehehhee |
| 14:37.30 | Aiatek | ok |
| 14:37.34 | Aiatek | thx |
| 14:41.41 | *** join/#asterisk kippi (n=chriso@83-244-164-130.cust-83.exponential-e.net) |
| 14:42.29 | kippi | is there away I can use Background and read together? I want to create a message that asks for your username but allows you to enter it before the message is finshed |
| 14:44.12 | [TK]D-Fender | kippi: No. Does Read not allow you to interrupt a sound file it plays? |
| 14:44.42 | *** join/#asterisk Deeewayne (n=dwayne@75.76.254.162) |
| 14:44.42 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 14:44.52 | kippi | read isn't picking up the digits |
| 14:45.12 | kippi | if I wait until background is finished it works |
| 14:45.17 | [TK]D-Fender | kippi: Or simply doesn't care until its done |
| 14:45.39 | [TK]D-Fender | kippi: No, Background BEFORE read is not going to work. I said using READ to play backa file first |
| 14:46.14 | kippi | so you can use read to play a file? |
| 14:46.38 | [TK]D-Fender | kippi: ..... |
| 14:47.53 | *** join/#asterisk casix (n=casix@pbxedifici.adamvozip.es) |
| 14:49.11 | casix | hello, anyone know if it is possible in the originate action from manager to call to more than one extension in the channel var? like channel: SIP/101&SIP/102&SIP103. Thank you |
| 14:49.51 | [TK]D-Fender | casix: Directly no, via a Local channel, yes |
| 14:50.40 | *** join/#asterisk axisys (n=axisys@155.70.141.45) |
| 14:53.12 | casix | thx |
| 14:53.15 | *** join/#asterisk gr0mit (n=tim@extrt.txrx.org.uk) |
| 14:53.50 | fun330 | is there a way to display realtime call duration on a web page? |
| 14:57.02 | *** join/#asterisk tzafrir_laptop (n=tzafrir@local.xorcom.com) |
| 14:57.21 | *** part/#asterisk gego (n=rick@b238085.customer.hansenet.de) |
| 14:58.19 | [TK]D-Fender | fun330: AMI & CLI both offer ways to check up on calls so, yes |
| 14:59.05 | eppigy | hello |
| 14:59.07 | eppigy | i am dave |
| 15:00.49 | plundra | Superbartt: Not sure if you care, but siproxd worked like a charm :-) |
| 15:01.53 | Superbartt | ok nice plundra, I'll keep it in mind if I ever need it :P |
| 15:02.20 | [TK]D-Fender | eppigy: These are the Daves I know I know, these are the Daves I know! |
| 15:04.08 | *** join/#asterisk neurosys (n=neurosys@173-9-159-182-miami.txt.hfc.comcastbusiness.net) |
| 15:07.22 | jaytee | and these are the Daves I know I don't know and the Daves I don't know that I don't know. |
| 15:08.06 | eppigy | \o/ |
| 15:12.08 | casix | if in originate action from manager I call to a queue (via LOcal) will asterisk call the destination when the agent responds or when the queue make the answer and start moh? |
| 15:16.36 | tzafrir_laptop | Anybody actually confirmed that eppigy is actually a Dave? |
| 15:16.38 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 15:17.06 | [TK]D-Fender | casix: It will bridge instantly. Queue plays audio, etc, and answers the call |
| 15:17.55 | *** join/#asterisk seanmh (n=johndoe@198.59.129.24) |
| 15:22.32 | casix | ok |
| 15:23.17 | *** join/#asterisk proxium (n=proxium@196.203.51.238) |
| 15:24.00 | proxium | Hello, did Asterisk-1.6.0.9 work with Vicidial-2.0.5 ?? |
| 15:24.10 | [TK]D-Fender | casix: In 1.6.1 you can use Dial options to force diaplan redirects for both legs to actually generate the outbound call. |
| 15:24.56 | leifmadsen | jaytee: because you don't understand the different between the 1.6.1 and 1.6.2 branches |
| 15:25.02 | proxium | More explanation is here: http://www.eflo.net/VICIDIALforum/viewtopic.php?t=7958 |
| 15:25.12 | leifmadsen | 1.6.feature_release.bug_release |
| 15:25.21 | *** join/#asterisk jelly-bean (n=jelly@c-76-23-46-62.hsd1.ut.comcast.net) |
| 15:25.22 | leifmadsen | 1.6.1 <-- feature release |
| 15:25.32 | leifmadsen | 1.6.1.0, 1.6.1.1 <-- bug release |
| 15:25.40 | leifmadsen | 1.6.2 <-- feature release |
| 15:25.49 | leifmadsen | 1.6.2.0, 1.6.2.1 <-- feature releases |
| 15:25.51 | leifmadsen | etc.... |
| 15:25.55 | jelly-bean | is there any good gui for asterisk that shows calls waiting (in queue)? |
| 15:25.57 | jaytee | leifmadsen, ah! thank you for that clarification. now it makes perfect sense. |
| 15:26.10 | leifmadsen | 1.6.0.x, 1.6.1.x, 1.6.2.x <-- all separate branches, maintained at the same time |
| 15:26.30 | leifmadsen | jelly-bean: freepbx and asterisk-gui? |
| 15:26.46 | leifmadsen | doesn't use gui's, and simply is guessing that is true |
| 15:27.10 | jaytee | it's getting really hard for an old fart like me to keep up with all the changes and new stuff that gets added in lately. |
| 15:29.33 | [TK]D-Fender | jelly-bean: Go look on the WIKI GUI list |
| 15:30.34 | jaytee | been awhile since I messed with FreePBX. I think it had FOP and that would show calls in queues but I can't remember for certain. Don't use FOP anyways, I'm a Dapper Dan man :-) |
| 15:31.19 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 15:32.41 | jelly-bean | has anyone tried the RoR Queue-Tip? |
| 15:32.57 | proxium | PLZ you told me to upgrade my Asterisk server 1.4 to 1.6, now, no one is sure about the support of Vicidial-2.0.5 |
| 15:33.29 | leifmadsen | proxium: you upgraded blindly because someone told you to on IRC? |
| 15:33.40 | wilsonj | lol |
| 15:34.09 | *** join/#asterisk captiancrash (n=captianc@70.159.118.70) |
| 15:34.39 | proxium | not so blindly, but they advice me to do !! |
| 15:36.00 | proxium | Now I have problems on manager file and I want to ask you about this, any idea ? |
| 15:38.17 | *** join/#asterisk simprix (n=simprix@adsl-76-220-119-66.dsl.klmzmi.sbcglobal.net) |
| 15:40.17 | Nilzao | brothers, someone have a bookmark with a how to do attended transfers? |
| 15:40.32 | Nilzao | i got good things about it at voip-info |
| 15:40.57 | Nilzao | but cant make it with qutecom/wengophone |
| 15:42.46 | Nilzao | i put the atxfer=*2 at [featuresmap] features.conf |
| 15:43.20 | Nilzao | when pressing *2 at dialpad qutecom, nothing happens, no logfiles nothing on CLI |
| 15:46.09 | Nilzao | one good thing, i can hold the call, call the other extension, but can't transfer |
| 15:47.01 | casix | Nilzao: in logger.conf you have to add dtmf to see its in log file. And try to do it more quikly, maybe this is the problem |
| 15:47.04 | *** join/#asterisk PMantis (n=sswitzer@cpe-74-65-58-229.rochester.res.rr.com) |
| 15:48.10 | PMantis | Hello, asterisk 1.6.1 on Ubuntu 9.04, compiled. astrundir => /var/run/asterisk, but still getting "Unable to connect to remote asterisk (does /var/run/asterisk.ctl exist?)" Whaaa? |
| 15:48.42 | Nilzao | casix: making it, brb tx |
| 15:50.30 | proxium | PMantis: in /etc/asterisk/asterisk.conf try to verify the line (astrundir => /var/run/asterisk) |
| 15:50.37 | *** join/#asterisk CunningPike (n=CunningP@204.239.10.119) |
| 15:51.10 | PMantis | proxium: # grep rundir /etc/asterisk/asterisk.conf |
| 15:51.10 | PMantis | astrundir => /var/run/asterisk |
| 15:52.24 | *** join/#asterisk edgars (n=edgars@ns.dtg.lv) |
| 15:52.27 | proxium | <PROTECTED> |
| 15:52.30 | edgars | hello! |
| 15:52.49 | edgars | on which distribution is build asterisknow? |
| 15:52.52 | *** join/#asterisk ariel_ (i=3fd6eca9@gateway/web/ajax/mibbit.com/x-e40a0515f4289dd8) |
| 15:53.40 | PMantis | proxium: Yup, owned by asterisk:asterisk (recursively) at /etc/asterisk and mode 775 |
| 15:53.57 | PMantis | proxium: == Parsing '/etc/asterisk/asterisk.conf': == Found |
| 15:54.40 | proxium | PMantis: you have freepbx with it? |
| 15:55.03 | PMantis | proxium: Not yet, just compiled asterisk so far |
| 15:55.25 | *** join/#asterisk lasko_ (n=lasko@70.102.15.210) |
| 15:56.17 | *** join/#asterisk marv[work] (n=timr@router.asteriasgi.com) |
| 15:56.29 | *** join/#asterisk ruben23 (n=AGENT@122.55.48.242) |
| 15:56.32 | Nilzao | casix: added dtmf worked, tx, now i can see what happens |
| 15:56.34 | *** part/#asterisk lasko (n=lasko@70.102.15.210) |
| 15:56.56 | *** join/#asterisk Defraz (n=T0tal@fw-poky.fuzecore.com) |
| 15:57.35 | PMantis | proxium: Here, a few things I ran for you to see: http://pastebin.ca/1419317 |
| 15:57.41 | [TK]D-Fender | edgars: Depends which version |
| 15:57.55 | *** join/#asterisk lost_soul (i=shawn@cpe-67-241-68-104.twcny.res.rr.com) |
| 15:58.08 | casix | :) |
| 15:58.39 | ruben23 | hi anyone have idea on this error installing zaptel 1.4 http://pastebin.com/m126b3216 my distro si ubuntu server... |
| 15:58.41 | lost_soul | afternoon everyone |
| 15:59.02 | [TK]D-Fender | PMantis: You haven't shown us that * is RUNNING |
| 15:59.41 | PMantis | [TK]D-Fender: True, but it's looking in /var/run, *NOT* /var/run/asterisk |
| 15:59.54 | Nilzao | nice, now i know the qutefone is sending the dtmf. But asterisk is not interpreting the *2 command |
| 16:00.13 | Nilzao | the *qutecom/whengophone |
| 16:01.02 | casix | Nilzao: try to push it fast, if not you can try with only * to see if it works |
| 16:01.11 | [TK]D-Fender | Nilzao: Wheres the pastebin I asked for 2 HOURS agio? |
| 16:01.32 | PMantis | [TK]D-Fender: If I chown asterisk /var/run ... then start asterisk, it works. but astrundir is not /var/run. Check the config excerpt I pasted, too. |
| 16:01.51 | PMantis | [TK]D-Fender: It's all in the one paste |
| 16:02.13 | PMantis | [TK]D-Fender: I tried with a trailing /, too/ |
| 16:03.43 | Nilzao | casix: i tryed as faster as i can push... the cli show the * started the passtrought 480ms passed, how i increase this time? |
| 16:04.12 | Nilzao | casix: im looking it at features.conf |
| 16:05.20 | [TK]D-Fender | PMantis: pastebin your ENTIREE asterisk.conf |
| 16:05.37 | Nilzao | casix: nevermind... 480ms is the dtmf time sorry |
| 16:11.10 | casix | Nilzao: try with only one digit first |
| 16:11.50 | PMantis | [TK]D-Fender: ok... coming up. |
| 16:12.22 | edgars | [TK]D-Fender: last one 1.5 |
| 16:14.53 | PMantis | [TK]D-Fender: You made me look closer, and i found it... didn't read the first line: [directories](!) ; remove the (!) to enable this |
| 16:15.09 | defswork | what are you guy's opinion on putting all of a business' line on voip over adsl ? |
| 16:15.48 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 16:17.55 | Nilzao | casix: one or two digits, dtmf works and the CLI show them |
| 16:18.25 | Nilzao | casix: show the '*' dtmf, the '2' dtmf |
| 16:18.34 | Nilzao | casix: but nothing more happens |
| 16:18.45 | Nilzao | casix: i wanna just transfer the call |
| 16:18.49 | Nilzao | to other extension |
| 16:19.17 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 16:19.28 | ruben23 | hi anyone have idea on this error installing zaptel 1.4 http://pastebin.com/m126b3216 my distro si ubuntu server... |
| 16:19.44 | ruben23 | i already installed the linux kernel headers.. |
| 16:19.55 | Nilzao | casix: added one line to features.conf in [featuresmap] atxfer=*2 |
| 16:20.22 | Nilzao | casix: and in [general] featuredigittimeout=1000 |
| 16:21.20 | casix | Nilzao: then I don't know why don't work... sorry |
| 16:22.16 | edgars | [TK]D-Fender: ok, centos :) |
| 16:22.25 | *** join/#asterisk HeXiLeD (n=H3X@unaffiliated/hexiled) |
| 16:22.45 | Nilzao | casix: i did nothing on extensions.conf about it |
| 16:22.50 | Nilzao | casix: i have to? |
| 16:23.59 | *** join/#asterisk wonderworld (n=ww@ip-62-143-16-28.unitymediagroup.de) |
| 16:27.57 | [TK]D-Fender | Nilzao: WHERE IS THE PASTEBIN I ASKED FOR **2 HOURS AGO** ? |
| 16:28.39 | Nilzao | [TK]D-Fender: i was trying to make it, the CLI was showing nothing |
| 16:29.01 | [TK]D-Fender | Nilzao: I said to pastebin the complete call attempt. |
| 16:29.31 | Nilzao | [TK]D-Fender: but the call attempt works |
| 16:29.50 | [TK]D-Fender | Nilzao: PASTEBIN the damn call. |
| 16:29.50 | Nilzao | [TK]D-Fender: casix helped me to make the dtmf show in the CLI |
| 16:30.07 | Nilzao | [TK]D-Fender: ok, u will see there is nothing in the pastebin i paste it |
| 16:33.08 | tzafrir_laptop | with the debian 1.6.1 packages I keep getting the canary killed when I start Asterisk at boot |
| 16:33.19 | tzafrir_laptop | but I can't seem to reproduce it otherwise |
| 16:33.38 | Nilzao | [TK]D-Fender: Here it is http://paste.lisp.org/display/80030 |
| 16:33.42 | tzafrir_laptop | anybody else noticed something similar? |
| 16:33.58 | [TK]D-Fender | Nilzao: Executing [6005@default:1] Dial("Zap/1-1", "SIP/6005&IAX2/6005") in new stack |
| 16:34.22 | [TK]D-Fender | Nilzao: You have to use the "t" or "T" OPTIONS to allow transferring! Read the damn instructions. |
| 16:34.27 | [TK]D-Fender | Nilzao: THAT is your problem |
| 16:34.54 | [TK]D-Fender | Nilzao: And you'd have had this answer 2 hours ago if you just did as I asked |
| 16:35.00 | [TK]D-Fender | Nilzao: "core show application dial" <- |
| 16:35.14 | Nilzao | i read this at voip-info: Note: You MUST use the T and/or t options in the command Dial() in order to allow the caller and/or callee to use any transfer feature |
| 16:35.22 | Nilzao | but wtf is that? |
| 16:35.41 | [TK]D-Fender | Dial(SIP/6005&IAX2/6005,,tT) <-- |
| 16:36.12 | [TK]D-Fender | Nilzao: Go rad the app's INSTRUCTIONS |
| 16:36.14 | [TK]D-Fender | read* |
| 16:36.15 | *** join/#asterisk buzzyd (n=psp-man@86.54.239.113) |
| 16:36.30 | buzzyd | Hi All, |
| 16:36.55 | buzzyd | I've been playing with followme and just noticed that the CDR doesn't match with reality |
| 16:37.13 | Nilzao | great, lets go the confs, the insctuctions says nothing |
| 16:37.18 | buzzyd | billsec shows as 0 when I've been on call for 60 seconds |
| 16:37.28 | buzzyd | any ideas? |
| 16:38.49 | [TK]D-Fender | [12:34]<[TK]D-Fender>Nilzao: "core show application dial" <- instructions |
| 16:42.43 | tzafrir_laptop | it doesn't make sense. I see the message right after Asterisk starts. It should take at least two minutes before Asterisk should think the canary is dead |
| 16:43.59 | buzzyd | actually what it seems to do is create a new CDR line which has no userfield or accountcode with lastapp filled as followme is there an option to fill in these bits? |
| 16:45.56 | buzzyd | anyone got experience with followme and asterisk cdr? |
| 16:57.34 | *** join/#asterisk propellerhead (n=yogurt2u@host105.190-136-230.telecom.net.ar) |
| 17:01.27 | Nilzao | dunno where put it, i grep "Dial(" * in /etc/asterisk, will put this ,tT in all Dial instructions |
| 17:06.22 | casix | bye |
| 17:07.02 | buzzyd | digging further I notice if I put call forward on in the handset it does the same thing creates a new CDR line with no userfield or accountcode any ideas anyone? |
| 17:12.56 | ariel_ | buzzyd: asterisk has issues with the billing part your looking for. In most of our call centers and any place we use cdr's for billing add an new uniqueID to the records and follow them that way. Via our report we then tie them all together. |
| 17:17.31 | *** part/#asterisk Shazaum`afk (n=Shazaum@unaffiliated/shazaum) |
| 17:21.17 | *** join/#asterisk stephank (n=traveler@2002:52c5:cf78:7:21c:c4ff:fece:ea94) |
| 17:22.34 | buzzyd | Ariel, how can you link the uniqueID for calls that generate multiple entries? |
| 17:22.36 | stephank | Hello! Short question.. I'm looking at how to store telephone numbers in a database. Wikipedia tells me there's not country code (starting with) 0. So, am I correct in saying that I can safely store telephone numbers in E.164 format as an integer? |
| 17:23.10 | *** join/#asterisk sah-work (n=Bawbatos@65-119-47-100.dia.static.qwest.net) |
| 17:26.03 | mmlj4 | stephank: URL? |
| 17:26.23 | *** part/#asterisk PMantis (n=sswitzer@cpe-74-65-58-229.rochester.res.rr.com) |
| 17:26.42 | stephank | mmlj4: The wikipedia URL? http://en.wikipedia.org/wiki/List_of_country_calling_codes |
| 17:27.37 | *** join/#asterisk Iamnacho (i=Iamnacho@ip68-103-153-140.ks.ok.cox.net) |
| 17:31.06 | *** part/#asterisk awkfu (n=awkfu@66.162.90.56) |
| 17:33.38 | jelly-bean | does asterisk support multi-thread |
| 17:33.42 | jelly-bean | or what is asterisk coded in? |
| 17:34.16 | Jacke | take a look at the source. It was mostly C last time i've checked. |
| 17:35.20 | *** join/#asterisk bmoraca (n=chatzill@66.242.174.254) |
| 17:37.47 | *** join/#asterisk grEvenX (n=even@apb9hb.ip.ssc.net) |
| 17:38.43 | ariel_ | buzzyd: via dial plan entries, We created context for all our incoming calls that actually take the calls UniqueID and adds the dialed in number and then we created a cdr table called UID. This is how we tracked it. But since we did it under contract I can't give out the code. But it's something you can create via the dial plan. |
| 17:40.07 | leifmadsen | jelly-bean: yes, asterisk is multi-threaded |
| 17:40.40 | jaytee | whew! |
| 17:41.21 | Nilzao | [TK]D-Fender: i found this exten=s,3,Goto(default|6005|1) on extensions.conf |
| 17:41.35 | Nilzao | [TK]D-Fender: but i cant find the Dial() |
| 17:42.08 | Nilzao | [TK]D-Fender: i'm reading something about the Goto command and the default extension |
| 17:42.32 | [TK]D-Fender | Nilzao: Let me guess.... using the Asterisk-GUI? |
| 17:42.42 | Nilzao | [TK]D-Fender: bingo! |
| 17:42.50 | [TK]D-Fender | Nilzao: SMRT |
| 17:43.21 | [TK]D-Fender | Nilzao: those extens are generated on CONFIG LOAD, they are not physically in extensions.conf |
| 17:43.41 | [TK]D-Fender | Nilzao: They are generated based on users.conf , and I'm not sure whree/how you get to specify dial options for them |
| 17:43.54 | [TK]D-Fender | Nilzao: Go ask in their channel as GUI's aren't supported in here |
| 17:44.59 | *** part/#asterisk jelly-bean (n=jelly@c-76-23-46-62.hsd1.ut.comcast.net) |
| 17:44.59 | Nilzao | [TK]D-Fender: ok, i will try a little more before that |
| 17:45.27 | Nilzao | [TK]D-Fender: thanks a lot, i will stay here |
| 17:46.03 | Nilzao | [TK]D-Fender: gui's aren't nice? |
| 17:46.29 | [TK]D-Fender | Nilzao: They do things their way. If you don't like it, tough luck |
| 17:47.51 | *** join/#asterisk esaym (n=user@cpe-24-174-186-34.satx.res.rr.com) |
| 17:48.21 | Nilzao | [TK]D-Fender: so lets vi the confs and forget the gui for a while |
| 17:48.21 | nkohh | hi everyone. I can play sound digits with Playback, but when parking calls, they don't play. what can I do to troubleshoot this? I'm using 1.4.24.1, and I see in console it playing the sound files. |
| 17:48.54 | [TK]D-Fender | Nilzao: users.conf is ALSO a problem |
| 17:49.20 | [TK]D-Fender | Nilzao: It was built with the GUI in mind. You'd be advised to dump that as well and configure each channel driver separately |
| 17:51.29 | Nilzao | [TK]D-Fender: i will... soon, when i undertand better, i promise |
| 17:51.57 | *** join/#asterisk jelly-bean (n=jelly@c-76-23-46-62.hsd1.ut.comcast.net) |
| 17:52.12 | [TK]D-Fender | Nilzao: Don't worry, I don't expect any real change, but at least you know why. |
| 17:53.07 | jelly-bean | on http://store.digium.com/products.php?category_id=2 what is the dif. between the: "Four (4) span digital T1/E1/J1/PRI PCI 5.0V card and hardware echo cancellation" and the "Four (4) span digital T1/E1/J1/PRI PCI card and hardware echo cancellation" ? |
| 17:53.10 | jelly-bean | what does the 5.0V mean? |
| 17:53.16 | nkohh | voltage |
| 17:53.28 | *** join/#asterisk donnib (n=donnib@0x555281d0.adsl.cybercity.dk) |
| 17:53.29 | jelly-bean | ya but why would anyone need it |
| 17:53.41 | jelly-bean | and how is it dif. from the other |
| 17:53.51 | Chainsaw | jelly-bean: 5.0V PCI is the old standard. |
| 17:53.57 | bmoraca | jelly-bean: one's meant to be used in a PCI-X slot, the other in a PCI slot...although most slots of both kinds run at both 3.3V and 5V anymore |
| 17:53.57 | Chainsaw | jelly-bean: The notch is on the other side. |
| 17:54.08 | Chainsaw | jelly-bean: Unless you know that your machine is a Pentium III or older, you want 3.3V |
| 17:54.19 | jelly-bean | so if im building a new system i should not get 5.0 |
| 17:54.28 | jelly-bean | k |
| 17:54.32 | Chainsaw | jelly-bean: Indeed, you want a 3.3V card, or maybe even PCI-Express. |
| 17:54.36 | Chainsaw | jelly-bean: Definitely not PCI 5.0V |
| 17:55.08 | jelly-bean | thnx Chainsaw |
| 17:55.18 | Chainsaw | :) |
| 17:55.38 | donnib | can anybody help with a incoming route which won't match my incomming DID eventhough i can see the correct DID in the log ? i am using freepbx. |
| 17:57.43 | *** join/#asterisk tzafrir_laptop (n=tzafrir@local.xorcom.com) |
| 17:58.09 | [TK]D-Fender | donnib: Not supported here. You're already in #freepbx , go ask them |
| 17:58.20 | donnib | oki |
| 17:58.24 | donnib | thx |
| 17:58.46 | [TK]D-Fender | jelly-bean: Verify with your MOTHERBOARD first |
| 17:59.31 | [TK]D-Fender | jelly-bean: Sangoma's cards are 3.3 / 5 V interchangeable |
| 18:00.00 | jelly-bean | theres a pci one that is interchangable and then a pciexpress one |
| 18:00.08 | jelly-bean | i think i'll just get a mobo w/ the pciexpress |
| 18:02.17 | *** join/#asterisk kerx (n=root@adsl-69-105-171-127.dsl.irvnca.pacbell.net) |
| 18:03.00 | kerx | Hi, I am trying to setup a dynamic featuremap that does a automatic 3-way call. So the 'agent' presses *2, and it initiates a 3-way call and plays audio to both endpoints. Is this possible? |
| 18:04.26 | *** join/#asterisk lanning (n=lanning@nat/yahoo/x-e5484939f47eaca4) |
| 18:08.13 | *** join/#asterisk GeekBoy (n=kvirc@200.94.227.203) |
| 18:09.10 | GeekBoy | anyone heard of getting an ATA to dial * up once the phone is picked up..or should I say,, askerisk answer the line as it answered a call to it´s numer |
| 18:10.36 | [TK]D-Fender | GeekBoy: Many ATA's support "dial on pickup" or "batphone" service like this. Check the MANUAL for it to see how. |
| 18:11.18 | jelly-bean | are there some major improvements to latest asterisk since 1.4.21 |
| 18:11.28 | jaytee | I setup an fxs port on a Linksys for a "Hotline" to do just that |
| 18:11.48 | GeekBoy | thanks fender |
| 18:12.46 | *** join/#asterisk sjzzalx (n=jeff@c-76-23-46-62.hsd1.ut.comcast.net) |
| 18:13.07 | jelly-bean | anything major? or should i stick with my package manager version above? is it easy enough to install? |
| 18:14.18 | *** join/#asterisk darkdrgn2k3 (n=darkdrgn@CPE000c419e662f-CM0011aea0fa16.cpe.net.cable.rogers.com) |
| 18:14.24 | darkdrgn2k3 | Hi, i just inatalled AsteriskNow but im a little list |
| 18:14.34 | darkdrgn2k3 | the web gui i see is FreePBX, is this right? |
| 18:14.50 | [TK]D-Fender | darkdrgn2k3: Comes with that and the AsteriskGUI by Digium |
| 18:15.02 | darkdrgn2k3 | how do i access asteriskgui? |
| 18:15.53 | [TK]D-Fender | dakdgo as in that disto's support channel listed in the /topic |
| 18:16.04 | [TK]D-Fender | darkdrgn2k3: ask in that disto's support channel listed in the /topic |
| 18:16.34 | darkdrgn2k3 | *sigh*] |
| 18:16.36 | [TK]D-Fender | jelly-bean: there have been important securit updates since the, bug-fixes, etc |
| 18:16.40 | kerx | Anyone know the best way I can accomplish a 3-Way Dynamic Featuremap? I'm trying to accomplish the following scenario: UserA and UserB on a call, UserA presses *2, and immediately "Bridge's" in a 3'rd call playing audio to both UserA and UserB. |
| 18:17.43 | [TK]D-Fender | kerx: Give your user's phones that can do 3-way calling on their own. |
| 18:18.18 | darkdrgn2k3 | LOL nm |
| 18:18.25 | kerx | [TK]D-Fender: Hrm, they want to do 1-touch 3-way dialing. Or else I would just give them the X-lite version that supports 3-way :) |
| 18:18.37 | kerx | That's why I'm digging into the featuremap |
| 18:18.50 | kerx | To create a dynamic featuremap, but I've noticed it's not as simple as I thought it would be :-( |
| 18:20.17 | kerx | Doesn't even seem with a dyanmic featuremap, I can do audio to both UserA and UserB, you can only do to 'self' or 'peer' |
| 18:20.32 | kerx | Any suggestions TK or others would be appreciated :) |
| 18:22.20 | [TK]D-Fender | kerx: You haven't described exactly what "#3" is going to be doing. |
| 18:23.22 | kerx | [TK]D-Fender: Caller #3 is just a number to dial |
| 18:23.44 | Brixius | Any idea why ael insists on wrapping stuff in a $[] ie I have a line like var=${CUT("from"|\;|3)} in my extensions.ael and pbx_ael builds it as Set(olistr=$[${CUT("from"|\;|3)}]) |
| 18:24.23 | [TK]D-Fender | Brixius: because it assumes basic math |
| 18:24.31 | [TK]D-Fender | Brixius: Use Set DIRECTLY yourself |
| 18:25.38 | kerx | [TK]D-Fender: I noticed online someone achieved this successfully using a AGI script executing Originate and Redirect's |
| 18:25.41 | Nilzao | tx, bye |
| 18:25.51 | [TK]D-Fender | kerx: In 1.6+ yuo can use Bridge, I'd do this triggered off an AGI. the agi would see the originating channel, then do an AMI originate with the actual code to execute targeting the appropriate channel; |
| 18:26.02 | *** part/#asterisk Nilzao (n=urubuh@200-161-87-81.dsl.telesp.net.br) |
| 18:26.03 | [TK]D-Fender | rexBridge, not Redirect |
| 18:26.10 | [TK]D-Fender | kerx: Bridge, not Redirect |
| 18:26.24 | kerx | I see. I did upgrade to 1.6 since I noticed Bridge(), |
| 18:26.31 | Brixius | the problem I'm having is "from" has the following "isup_oli=0;useragent=phone" so it was doing weird stuff because it was evaluating the from field instead of just setting it. |
| 18:26.55 | Brixius | I'll have to change some of my statements to use the Set command myself. |
| 18:27.07 | *** join/#asterisk mog (n=mog@c-68-62-169-246.hsd1.al.comcast.net) |
| 18:27.07 | *** mode/#asterisk [+o mog] by ChanServ |
| 18:27.09 | kerx | [TK]D-Fender: So just do this on the originating channel, would only bridge the audio to the initiating channel, and UserB wouldn't hear audio right ? |
| 18:27.24 | kerx | Or would it pass audio to all 3? |
| 18:27.29 | [TK]D-Fender | kerx: Should work both ways IIRC |
| 18:28.12 | kerx | Ok, I'll give that a shot... If not, I'll maybe stick 'em in a dynamic meetme room |
| 18:28.30 | kerx | [TK]D-Fender: Thanks again |
| 18:31.40 | darkdrgn2k3 | extensions are users right? |
| 18:34.08 | [TK]D-Fender | darkdrgn2k3: Only in terminology used by GUI's. |
| 18:34.28 | jelly-bean | i'm installing asterisk-1.6.1.0. what ver of libpri is the latest stable and compatible with this? 1.4? |
| 18:34.53 | [TK]D-Fender | darkdrgn2k3: An extension is a number you DIAL. What it does is irrelevant |
| 18:35.11 | [TK]D-Fender | jelly-bean: look at the grouping listed at asterisk.org |
| 18:36.09 | jelly-bean | [TK]D-Fender: not seeing any. link? |
| 18:36.30 | [TK]D-Fender | jelly-bean: LOOK HARDER |
| 18:36.51 | jelly-bean | im on asterisk.org |
| 18:36.53 | jelly-bean | front page |
| 18:37.19 | [TK]D-Fender | jelly-bean: LOOK HARDER <-- clearly not on the FIRST page. |
| 18:37.27 | darkdrgn2k3 | tk: so how would i create a sip user accuont? |
| 18:37.36 | Brixius | darkdrgn2k3: I would more associate devices with users, but I don't define users necessarly as people. nor a 1:1 corralition. |
| 18:37.49 | *** join/#asterisk lost_soul (i=shawn@cpe-67-241-68-104.twcny.res.rr.com) |
| 18:37.54 | [TK]D-Fender | darkdrgn2k3: What pasrt of "GUI's not supported here" was not clear? |
| 18:37.58 | Brixius | darkdrgn2k3: look at sip.conf |
| 18:38.34 | jelly-bean | i'm not finding it |
| 18:38.36 | *** join/#asterisk e-Reggae (n=aramos@selena.electrisa.com) |
| 18:38.37 | darkdrgn2k3 | tk: ooo so you wannt me to swich back to asterisk-gui just so you can throuw your 2 cents in there...... its not really a questiona botu the gui.. trying to get my had around the bloodhy terminology! |
| 18:38.41 | darkdrgn2k3 | Brixius: thank you |
| 18:39.21 | e-Reggae | hi, i have a TE122B card with echo cancellation module, i use trixbox 2.6.x... question: i hear echo in the beggining of the conversation, any solution to this? |
| 18:40.05 | jelly-bean | http://www.asterisk.org/downloads |
| 18:40.22 | Brixius | darkdrgn2k3: define a sip device and for context you put in where you want them to start out in the dialplan. But if your using a gui, I'm not sure where you put that in a gui. |
| 18:41.06 | darkdrgn2k3 | brian |
| 18:41.14 | darkdrgn2k3 | Brixius: fair enough |
| 18:41.15 | [TK]D-Fender | darkdrgn2k3: "how do I do X" typically means your GUI channel supports you, not this one. |
| 18:41.33 | [TK]D-Fender | darkdrgn2k3: their terminology, their rules. |
| 18:42.09 | darkdrgn2k3 | so what is this channel for |
| 18:42.55 | Brixius | darkdrgn2k3: I would agree with [TK]. I view this channel is for those who are editing there conf files directly with vi and setting stuff up that way. |
| 18:42.56 | bmoraca | darkdrgn2k3: support of asterisk in its native form...configuration files. |
| 18:42.56 | [TK]D-Fender | darkdrgn2k3: For people using * directly on their own. |
| 18:43.45 | darkdrgn2k3 | baaa why am i botheirng to argue.. waste of my time and yours |
| 18:43.56 | darkdrgn2k3 | thank you very much for all your time guys.. ill go play with it and see what i can do |
| 18:43.56 | darkdrgn2k3 | bye |
| 18:43.57 | *** part/#asterisk darkdrgn2k3 (n=darkdrgn@CPE000c419e662f-CM0011aea0fa16.cpe.net.cable.rogers.com) |
| 18:44.12 | bmoraca | lol |
| 18:46.27 | Brixius | I've used trixbox, it has some nice features, but I like the flexibility I get by editing the conf files myself better. :) |
| 18:47.09 | [TK]D-Fender | Lesson to all on configuring * yourself VS using a GUI : Don't complain that a car drives like crap when YOU aren't the one behind the wheel <- "How do I make my idiot chauffeur slow down?!" |
| 18:47.44 | Brixius | lol |
| 18:47.53 | kerx | [TK]D-Fender: So you would suggest a GUI is better than a person who doesn't know how to configure Asterisk manually ? |
| 18:48.32 | [TK]D-Fender | kerx: No I mean no amount of direction means ANYTHING when you aren't in F-ING CONTROL OF THE CAR :p |
| 18:48.56 | bmoraca | kerx: the simple solution is that people who don't know shouldn't be trying to configure asterisk in a production scenario. they should hire a consultant to do it or learn everything they'll need BEFORE they start in live. |
| 18:48.59 | kerx | roger, agreed. i've been worried about the GUI * world |
| 18:49.31 | [TK]D-Fender | kerx: You're being chauffeured around, you don't do shit. Sit back and if you don't like the ride, TFB :) |
| 18:49.40 | [TK]D-Fender | kerx: doesn't worry me in the least :) |
| 18:50.02 | kerx | I'm not using a GUI though |
| 18:50.37 | kerx | Unless you mean asking for support in here is being chauffeured around |
| 18:50.37 | [TK]D-Fender | GUI's are fine if you are happy with the box they put you in. Just don't complain to 3M plasitcs jsut because Sony used them making the stereo you bought and dislike. |
| 18:50.49 | [TK]D-Fender | kerx: No, asking for support FOR a GUI |
| 18:51.05 | kerx | I see. Yes, GUI support should be directed in #trixbox |
| 18:51.06 | kerx | :P |
| 18:51.08 | bmoraca | a GUI isn't inherently bad...as long as you understand what's going on behind the scenes and set your expectations appropriately. |
| 18:52.00 | [TK]D-Fender | bmoraca: understanding whats going behind isn't really important. As long as you are happy with what it provides without much question, then you're fine |
| 18:52.15 | [TK]D-Fender | bmoraca: Just don't nag the OEM providers. |
| 18:52.25 | *** join/#asterisk beherit (n=albert@netsys.bts.corp.amdatex.net) |
| 18:52.54 | beherit | how to troubleshoot a muffled sound in voip? any tips on how and where will I start |
| 18:54.22 | Brixius | muffled sound? like quiet sound, or garbled sound? |
| 18:54.41 | jelly-bean | what about the zaptel driver does that need to be something specific or is zaptel-1.4-current.tar.gz good? |
| 18:55.02 | [TK]D-Fender | jelly-bean: Same answer as before... |
| 18:55.12 | jelly-bean | no its not listed in the group |
| 18:55.27 | [TK]D-Fender | ~dahdi |
| 18:55.27 | infobot | [~dahdi] Digium/Asterisk Hardware Device Interface (DAhdi). The new name of zaptel More info at http://www.asterisk.org/zaptel-to-dahdi , and is pronounced "dah-dee" with a short A, or pronounced like http://www.russellbryant.net/dahdi.wav |
| 18:55.36 | jelly-bean | oh ok cool |
| 18:55.38 | jelly-bean | thnx |
| 18:55.54 | jelly-bean | then this asterisk book i was following is way out of date |
| 18:55.58 | jelly-bean | is there a newer install guide? |
| 18:56.12 | [TK]D-Fender | jelly-bean: in the tarball |
| 18:56.18 | [TK]D-Fender | (as always) |
| 18:56.27 | jelly-bean | README or the doxygen |
| 18:56.40 | kerx | [TK]D-Fender: This is w/ a Bridge inside the AGI script: WARNING[21858]: features.c:4455 bridge_exec: Bridge failed because channel SIP/flowroute/72578030#18172885756,p does not exists or we cannot get its lock |
| 18:56.44 | jelly-bean | make progdocs ? |
| 18:56.58 | nkohh | hi everyone. I can play sound digits with Playback, but when parking calls, they don't play. what can I do to troubleshoot this? I'm using 1.4.24.1, and I see in console it playing the sound files. |
| 18:57.14 | [TK]D-Fender | kerx: ",p"? I think you've trimmed poorly |
| 18:57.32 | [TK]D-Fender | jelly-bean: How about the obvious README and INSTALL files? |
| 18:58.06 | kerx | WARNING[21868]: features.c:4455 bridge_exec: Bridge failed because channel SIP/flowroute/72578030#18172885756 does not exists or we cannot get its lock |
| 18:58.10 | kerx | That's without the p |
| 18:58.34 | jelly-bean | [TK]D-Fender: that is specific to the asterisk package |
| 18:58.44 | jelly-bean | what about everything else--libpri, etc. |
| 18:58.45 | kerx | What kind of channels can you make? |
| 18:58.50 | kerx | w/ Bridge() |
| 18:58.55 | kerx | or does the channel have to be up already? |
| 18:59.00 | [TK]D-Fender | jelly-bean: They all have docs... |
| 18:59.16 | jelly-bean | you assume i've done this before and know all that needs to be installed in what order etc |
| 18:59.25 | [TK]D-Fender | jelly-bean: same order |
| 18:59.43 | jelly-bean | ya but some things are diff like zaptel |
| 18:59.48 | jelly-bean | so you're telling me there's no docs or tutorials |
| 18:59.53 | jelly-bean | that you know of |
| 19:00.13 | jelly-bean | relevant to 1.6.x.x |
| 19:00.42 | [TK]D-Fender | jelly-bean: the compile style & order hasn't really changed and the isntall instructions are ALL in their respective tarballs |
| 19:01.42 | jelly-bean | ok i'll continue w/ the asterisk book and asking in here |
| 19:03.01 | beherit | brixius garbled sound sometimes star wars sound like darth vader |
| 19:03.48 | beherit | for you GURU's are what are the recommended IP Phones for asterisk with very good voice quality. Cisco, polycom or grandstream |
| 19:03.53 | *** join/#asterisk Victor_Yure (n=victor@unaffiliated/victoryure/x-837844) |
| 19:04.49 | [TK]D-Fender | beherit: Polycom > All |
| 19:05.31 | beek | beherit: Polycom |
| 19:05.39 | beherit | fender, what model are the cheapest among polycom? |
| 19:05.50 | beek | IP320 |
| 19:05.52 | beek | IP330 |
| 19:06.00 | beherit | ok i will look into it |
| 19:06.08 | beek | <$100 |
| 19:07.36 | *** join/#asterisk simprix (n=simprix@65.209.144.66) |
| 19:09.05 | beherit | beek, does that IP330 are capable of using headset? |
| 19:09.11 | beek | Yes |
| 19:10.19 | beherit | what are the compatible headset for it, as I have check the site it says product has been remove or something |
| 19:11.12 | beek | I use the Plantronics series. |
| 19:11.55 | beek | I have a Polaris on my head right now. |
| 19:11.57 | beherit | beek what model of plantronicx? |
| 19:12.02 | beek | Polaris |
| 19:12.14 | beek | You'll need the quick-disconnect with the 1/8" plug |
| 19:12.16 | beherit | is that headset is audio jack or USB? |
| 19:12.43 | beek | 1/8" stereo |
| 19:12.46 | [TK]D-Fender | IP 330 does not use Polaris |
| 19:12.57 | [TK]D-Fender | it uses 2.5mm standard |
| 19:13.04 | [TK]D-Fender | same with IP 320 |
| 19:13.27 | rob0 | beek, is that the submarine-based nuclear missile? |
| 19:13.41 | rob0 | If so, that must hurt. |
| 19:13.42 | [TK]D-Fender | load chan_nuke.so ! |
| 19:13.46 | beek | rob0: Yes. |
| 19:14.19 | beek | looks for the quick-disconnect cable with the appropriate plug for my 330 |
| 19:15.12 | beherit | fender- what are the compatible headset for IP330? |
| 19:15.28 | buzzyd | shouldn't the uniqueid in the cdr be unique? |
| 19:18.06 | *** join/#asterisk Defraz (n=T0tal@fw-poky.fuzecore.com) |
| 19:19.16 | *** join/#asterisk MindTheGap (n=MindTheG@mail.lpj.com.br) |
| 19:19.17 | [TK]D-Fender | beherit: 2.5mm STANDARD. Every boring 2.5mm headset you've ever seen at radio Shack |
| 19:21.43 | nkohh | this issue is blowing me away. I feel so stupid :-( |
| 19:21.51 | *** join/#asterisk scooby2 (n=scooby2@pdpc/supporter/active/scooby2) |
| 19:22.30 | beherit | fender, does the vioce quality will not be affected if I use a cheap headset? |
| 19:22.45 | nkohh | beherit: it can be affected |
| 19:22.57 | scooby2 | would Set(CALLERID(all)=) work for setting callerid to unknown if the provider allows it? |
| 19:23.06 | nkohh | but for voice only you're usually okay with anything above bottom of the barrel |
| 19:23.13 | rob0 | That question seems so ... odd. Of course a cheap headset might be ... cheap. |
| 19:23.17 | brad_mssw | anyone familiar with the polycom soundstation IP 6000 ? having an asterisk registration issue with one here, getting 'chan_sip.c:8553 check_auth: username mismatch, have <polycomsip>, digest has <>' |
| 19:23.39 | brad_mssw | 'polycomsip' is the name of the section in sip.conf so that's right ... not sure what this blank digest is |
| 19:23.45 | brad_mssw | just configuring it using the web interface |
| 19:24.11 | beherit | nkohh, I can't afford to have a ppor voice quality :( thats the main reason why we are going to invest in ip phone. right now we are using eyebeam and computer and we have a poor voice quality |
| 19:24.36 | nkohh | well, the answer is yes, a bad headset can fuck up your voice quality, obviously. |
| 19:24.37 | buzzyd | Anyone know why the uniqueid in the CDR isn't always unique? |
| 19:25.33 | nkohh | buzzyd: from the "Asterisk cdr mysql" page on voipinfo: Attention! The uniqueid field is not guaranteed to be unique across the different CDR entries, even though the name suggests exactly that. |
| 19:25.33 | scooby2 | it will be unique among the current calls |
| 19:26.43 | buzzyd | damn ok anyone got some tips for billing :) |
| 19:26.59 | jelly-bean | should i install asterisk before asterisk-addons? |
| 19:27.06 | scooby2 | jelly-bean: yes |
| 19:28.16 | buzzyd | scooby2: Not true I have two calls 20 secs apart with same notsounquieID :) |
| 19:29.13 | [TK]D-Fender | scooby2: "core show application setcallerpres" |
| 19:29.26 | nkohh | buzzyd: I just assign them a unique ID that I index on as I insert into DB.... |
| 19:30.11 | [TK]D-Fender | beherit: Of course it will. for quality I recommend an IP 4XX or higher class phone with a Plantronics M22 Amp + good headset. My CSR's love the H261 Binaural one personally |
| 19:30.16 | kerx | quit |
| 19:30.51 | brad_mssw | anyone familiar with polycoms? |
| 19:30.52 | *** join/#asterisk davidf88 (n=davidf88@92.2.162.2) |
| 19:31.04 | jelly-bean | scooby2: thnx |
| 19:31.36 | buzzyd | nkohh, can you assign it to the asterisk generated CDR or do I need to create a new table and import that data with a unique index? |
| 19:31.42 | *** join/#asterisk grEvenX (n=even@cB78D5AC1.dhcp.bluecom.no) |
| 19:32.00 | beherit | H261 is a headset right? |
| 19:32.13 | jelly-bean | does asterisk-gui come with asterisk or do i install it seperately? |
| 19:32.16 | [TK]D-Fender | beherit: yes |
| 19:32.24 | beherit | is that a plantronics? |
| 19:32.28 | [TK]D-Fender | jelly-bean: seperately |
| 19:32.32 | [TK]D-Fender | beherit: yes |
| 19:32.36 | beherit | ok thanks |
| 19:32.45 | jelly-bean | i am just doing a test install with softphones. should i try make samples or can i do all the config i need from within asterisk-gui? |
| 19:33.54 | jelly-bean | does asterisk come with its own web-based gui or is that what asterisk-gui is for? |
| 19:34.20 | [TK]D-Fender | jelly-bean: Latter. All are bolt-on, not "native" |
| 19:34.24 | *** join/#asterisk Aiatek (n=Asterisk@75.112.88.200.m.sta.codetel.net.do) |
| 19:34.36 | [TK]D-Fender | jelly-bean: Each is supported in their own channel as listed in the /topic |
| 19:34.40 | jelly-bean | i c |
| 19:34.42 | jelly-bean | thnx |
| 19:36.35 | *** join/#asterisk bijit (n=benji@190.10.122.14) |
| 19:40.26 | nkohh | [TK]D-Fender: you got any ideas on my 'numbers not playing' issue? |
| 19:41.05 | nkohh | i assume you'd have offered them if you had them, but I generally regard you as the individual most likely to be able to assist me, so I figured I would ask outright |
| 19:41.14 | [TK]D-Fender | nkohh: From what you've shown.... No. |
| 19:41.42 | nkohh | I don't really know what more I can show. I see them playing, they seem to be going out the proper channel.. I just don't hear them. |
| 19:41.46 | *** join/#asterisk Nilzao (n=nils@200-168-146-103.dsl.telesp.net.br) |
| 19:42.09 | [TK]D-Fender | nkohh: I have no description of how you are doing this, what hardware, what process, no CLI output, nothing. |
| 19:42.19 | nkohh | fair enough. |
| 19:45.26 | nkohh | [TK]D-Fender: I apologize for that. Here is the CLI output of the call coming in, connecting, and being parked. That's all I see (except for the "user got tired of waiting" when I hang up). It does successfully park it on 701, so I can retrieve the call. |
| 19:46.00 | nkohh | this is on a freebsd 7.0 1.4.24.1 with no modifications (aside from freebsd patches) and everything is over sip. there is no other equipment in the box. |
| 19:46.21 | *** join/#asterisk ManxPower (n=manxpowe@user-24-236-95-236.knology.net) |
| 19:46.30 | *** part/#asterisk beek (n=klinebl@pdpc/supporter/professional/beek) |
| 19:47.22 | nkohh | and I simply include parkedcalls in my default context. |
| 19:47.47 | nkohh | and I've made no changes to the stock features.conf. |
| 19:48.06 | nkohh | and playing digits with Playback(digits/7) (etc) works. |
| 19:48.28 | *** join/#asterisk beek_ (n=klinebl@pdpc/supporter/professional/beek) |
| 19:49.56 | [TK]D-Fender | ... |
| 19:51.08 | nkohh | sorry, did I miss/omit something? |
| 19:51.25 | [TK]D-Fender | nkohh: Yeah.... EVERYTHING I asked for. |
| 19:51.39 | beherit | fender - any recommended online store for the ipphones and headset that ship in the philippines |
| 19:52.16 | [TK]D-Fender | beherit: No idea |
| 19:52.17 | nkohh | oops, forgot the link... http://pastebin.com/d15e00e9 |
| 19:53.28 | [TK]D-Fender | nkohh: And the rest? |
| 19:54.28 | nkohh | nevermind. |
| 19:54.54 | nkohh | you have my apologies. |
| 19:55.49 | [TK]D-Fender | o>o |
| 19:56.22 | [TK]D-Fender | just never seems to get the simple things he asks for. |
| 19:56.37 | [TK]D-Fender | figures people really don't want help. |
| 19:57.10 | nkohh | you are probably right :-) |
| 19:57.40 | eppigy | 8[] |
| 20:01.44 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 20:05.08 | *** join/#asterisk Micho123 (n=mcho123@77.42.135.87) |
| 20:05.34 | Micho123 | Hi al, is there a way to receive a call in a special context and continue the call flow using another context ame? |
| 20:05.39 | Micho123 | *name? |
| 20:08.12 | [TK]D-Fender | Micho123: "core show application goto" <- |
| 20:08.51 | jelly-bean | i keep getting |
| 20:08.55 | jelly-bean | [May 11 14:08:38] NOTICE[30774]: chan_sip.c:19536 handle_request_register: Registration from '<sip:6000@10.1.10.99>' failed for '10.1.10.22' - No matching peer found |
| 20:09.03 | jelly-bean | trying to login with x-lite |
| 20:09.14 | jelly-bean | i created a user and gave it an extension |
| 20:09.25 | jelly-bean | i configured x-lite to have username and authorization user of the extension (6003) |
| 20:09.58 | [TK]D-Fender | jelly-bean: Well that shows 6003, not 6000 Looks like you missed something. |
| 20:10.04 | jelly-bean | thats another user |
| 20:10.07 | jelly-bean | but it says the same thing |
| 20:10.10 | jelly-bean | we're all 3 trying to connect at once |
| 20:11.05 | *** part/#asterisk donnib (n=donnib@0x555281d0.adsl.cybercity.dk) |
| 20:11.08 | [TK]D-Fender | jelly-bean: well you're getting a "not found" for 6000. It simply is not a valid # at all. |
| 20:11.14 | *** join/#asterisk Slade- (i=user@255.85.204.68.cfl.res.rr.com) |
| 20:11.28 | Slade- | hey after asterisknow is first installed. what type of access is initially available? ssh? |
| 20:11.34 | jelly-bean | [May 11 14:11:11] NOTICE[30774]: chan_sip.c:19536 handle_request_register: Registration from 'Mike <sip:6003@10.1.10.99>' failed for '10.1.10.23' - No matching peer found |
| 20:12.24 | *** join/#asterisk sjzzalx (n=jeff@c-76-23-46-62.hsd1.ut.comcast.net) |
| 20:13.53 | VaGoNeTaS | hey buddys |
| 20:13.55 | VaGoNeTaS | http://pastebin.ca/1419623 |
| 20:14.12 | VaGoNeTaS | that's the error that i get when i try to reload "chan_agent.so" |
| 20:14.23 | VaGoNeTaS | coz i've made an agents.conf file |
| 20:14.27 | jelly-bean | doo i have to install anything special to get SIP to work? |
| 20:15.08 | [TK]D-Fender | jelly-bean: SIP is working, your configs jsut don't have mathing accounts |
| 20:15.32 | jelly-bean | can i create the sip accounts via asterisk-gui or do i need to manually add them to /etc/asterisk/sip.conf |
| 20:15.35 | [TK]D-Fender | matching* |
| 20:15.42 | [TK]D-Fender | jelly-bean: either |
| 20:15.52 | jelly-bean | where in asterisk-gui, do you know? |
| 20:15.55 | [TK]D-Fender | jelly-bean: *GUI does them all in users.conf |
| 20:16.01 | VaGoNeTaS | what could it be? |
| 20:16.05 | jelly-bean | ok so i created a uer |
| 20:16.06 | jelly-bean | user |
| 20:16.56 | [TK]D-Fender | VaGoNeTaS: Could be it wasn't loaded for your first command, and it gave you a warning because it wasn't there to unload first, and then when you tried to load it, it was already loaded by the 2nd step of your RELOAD |
| 20:16.59 | *** join/#asterisk kerx (n=root@adsl-69-105-171-127.dsl.irvnca.pacbell.net) |
| 20:17.15 | [TK]D-Fender | jelly-bean: Maybe you didn't actually APPLY your changes. |
| 20:17.15 | kerx | [TK]D-Fender: you here, I've got as far as we discussed w/ the AGI and the AMI bridging |
| 20:17.25 | kerx | Oh there you are :) |
| 20:17.44 | *** part/#asterisk Slade- (i=user@255.85.204.68.cfl.res.rr.com) |
| 20:17.53 | kerx | I'm going to pastebin, one sec |
| 20:18.13 | Pan3D | quick question on hung calls in asterisk: If a conference call gets hung up on the server side, could it potentially continue to send SIP messages or RTP to a client machine that had been connected? |
| 20:18.23 | Pan3D | hung as in fail*( |
| 20:18.27 | Micho123 | [TK]D-Fender, the goto tag will make the job? |
| 20:19.11 | jelly-bean | [TK]D-Fender: yep clicked Apply Changes |
| 20:19.12 | Micho123 | [TK]D-Fender, witch1*CLI> core show application got |
| 20:19.12 | Micho123 | Your application(s) is (are) not registered |
| 20:19.12 | Micho123 | switch1*CLI> |
| 20:19.30 | sjzzalx | So, in a new setup, are we supposed to connect directly to the asterisk server as a SIP thing, or should there be an intermediate? The docs I have mention OpenSER but it's place isn't really clear to me; does Asterisk need something like that to be its homie? |
| 20:19.33 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 20:19.38 | jelly-bean | should i be able to see them with: sip show peers from console? |
| 20:19.58 | [TK]D-Fender | Micho123: TRY AGAIN |
| 20:20.10 | [TK]D-Fender | jelly-bean: Should |
| 20:20.29 | kerx | http://pastebin.ca/1419636 |
| 20:20.52 | jelly-bean | 0 sip peers [Monitored: 0 online, 0 offline Unmonitored: 0 online, 0 offline] |
| 20:20.55 | kerx | Please check the above. What's interesting is the minute teh Bridge() initiate's from the AMI, UserA get's hungup(), and UserB stay's with the bridge'd call |
| 20:21.00 | jelly-bean | so its creating users but not peers? |
| 20:21.10 | jelly-bean | i restarted the asterisk server |
| 20:21.17 | VaGoNeTaS | no, is not loaded |
| 20:21.21 | jelly-bean | i see the extensions listed on system status page of asterisk-gui |
| 20:21.21 | VaGoNeTaS | i cant see the module |
| 20:21.30 | VaGoNeTaS | i did an show agents online |
| 20:21.39 | VaGoNeTaS | and it says not found |
| 20:21.42 | VaGoNeTaS | command not found bla bla |
| 20:21.46 | [TK]D-Fender | jelly-bean: Must have done them wrong. |
| 20:22.09 | *** join/#asterisk Victor_Yure (n=victor@unaffiliated/victoryure/x-837844) |
| 20:22.13 | [TK]D-Fender | jelly-bean: type=peer <- whAT YOU SHOULD HAVE |
| 20:22.47 | jelly-bean | sip show objects lists all the extensions i created |
| 20:22.52 | sjzzalx | [TK]D-Fender: type: peer shows in sip show objects |
| 20:22.52 | jelly-bean | and type: peer is set |
| 20:23.20 | [TK]D-Fender | jelly-bean: Well something is wrong with them |
| 20:23.38 | *** join/#asterisk captiancrash (n=captianc@70.159.118.70) |
| 20:26.17 | kerx | So the Bridge() immediately Hung up on me, and UserB remained connected w/ the Bridge'd party :-( |
| 20:26.19 | VaGoNeTaS | no i just cant load or unload the chan_agent.so |
| 20:26.49 | [TK]D-Fender | checkout time, heading home... |
| 20:26.56 | kerx | bye 4 now |
| 20:28.36 | *** join/#asterisk profXavier (n=jezus@unaffiliated/neverblue) |
| 20:28.48 | profXavier | bought new Polycom IP601 phones |
| 20:28.53 | profXavier | and I want to set them up |
| 20:29.06 | profXavier | what do I need to configure on asterisk's side |
| 20:29.12 | profXavier | is it just the context ? |
| 20:30.09 | russellb | that is a very open ended question :-) |
| 20:30.09 | nkohh | you will need to add sip users as well as extensions |
| 20:30.13 | nkohh | etc.. |
| 20:30.16 | russellb | as it depends on what you want to be able to do with it :-) |
| 20:30.31 | profXavier | i have the user, setup on my grandstream |
| 20:30.38 | profXavier | i want to use the Polycom instead |
| 20:30.55 | *** join/#asterisk n3hxs (n=HAMming@pool-173-66-115-234.washdc.fios.verizon.net) |
| 20:31.01 | profXavier | sorry, im a bit of an idiot/noob, when it comes to * |
| 20:31.25 | VaGoNeTaS | anybody knows why i cant... b http://pastebin.ca/1419623 |
| 20:31.39 | VaGoNeTaS | is not being well loaded |
| 20:32.30 | profXavier | Registration from '<sip:*.*.*.*@*.*.*.*>' failed for '192.168.90.*' - No matching peer found |
| 20:32.34 | profXavier | i get this error |
| 20:32.45 | profXavier | i have hidden the IPs |
| 20:35.57 | brad_mssw | what does this error mean: [May 11 16:35:02] WARNING[23184]: chan_sip.c:8553 check_auth: username mismatch, have <polycomsip>, digest has <> |
| 20:35.57 | brad_mssw | <PROTECTED> |
| 20:36.52 | brad_mssw | if I comment out the sip.conf section, I get the 'no matching peer error' ... but with it, I get that error, then a 'Username/auth name mismatch' error |
| 20:37.45 | VaGoNeTaS | there is some way to force the unload of one module? |
| 20:39.32 | profXavier | russellb: can you direct me a bit please? |
| 20:39.45 | russellb | sorry, busy .. |
| 20:39.48 | profXavier | np |
| 20:39.53 | profXavier | nkohh ? |
| 20:41.02 | brad_mssw | oh geez, polycom, give me a break... seems like it only likes numeric usernames |
| 20:43.22 | profXavier | oh |
| 20:43.31 | profXavier | another person with Polycom issues ? |
| 20:45.17 | ariel_ | brad_mssw: I have allot of polycom's and they do use other names then just numbers. I set them up via the ftp config files |
| 20:45.38 | *** join/#asterisk lordvadr (n=lordvadr@otis.ctc.biz) |
| 20:46.27 | profXavier | ariel_ that seems like a great way to manage them |
| 20:46.42 | profXavier | how do you setup the configs for different users? |
| 20:50.33 | *** part/#asterisk Nilzao (n=nils@200-168-146-103.dsl.telesp.net.br) |
| 20:51.05 | ariel_ | phones boot off ftp with there mac address, in your dhcp service you setup option 66 for them. In the http://www.voip-info.org/wiki/view/Polycom%20Phones |
| 20:51.45 | profXavier | sweet |
| 20:51.49 | profXavier | lots to read O_o |
| 20:54.45 | profXavier | too many question |
| 20:54.57 | profXavier | what are common options in codecs? |
| 20:55.17 | profXavier | in NA, ulaw (G.711) right.. but anything else? |
| 20:55.38 | brad_mssw | ariel_: well, we use linksys 942's in the rest of the office, only 1 polycom, figured the web interface would be sufficient since I don't need to do mass provisioning |
| 20:56.00 | brad_mssw | ariel_: so perhaps the issue is with the web interface, or possibly the firmware revision this thing shipped with (which appears to be 3.0.2 I think) |
| 20:56.36 | brad_mssw | (it's the ip6000) |
| 20:58.18 | *** join/#asterisk RHN (n=Roger@stjhnf0140w-142162065076.pppoe-dynamic.nl.aliant.net) |
| 20:58.59 | RHN | I have a fresh install of asterisknow running, but i cannot get logged in with the default admin/amp111, can anyone offer advice ? |
| 21:07.00 | pmhaddad-lappy | just so everyone knows, the dCAP is crazy tough :) |
| 21:09.11 | Qwell | pmhaddad-lappy: the Digium developers won't take it. we don't want to fail :p |
| 21:09.43 | pmhaddad-lappy | Qwell, i took it with a dev today |
| 21:09.50 | pmhaddad-lappy | i don't know who it was though |
| 21:09.52 | profXavier | what server offers firmware updates/provisioning for the Polycom phones ? |
| 21:10.02 | pmhaddad-lappy | i'll be super pumped if i pass |
| 21:10.08 | pmhaddad-lappy | but not too bummed out if i don't |
| 21:10.15 | pmhaddad-lappy | i had no idea it was that NUTS |
| 21:10.19 | *** part/#asterisk RHN (n=Roger@stjhnf0140w-142162065076.pppoe-dynamic.nl.aliant.net) |
| 21:12.52 | sjzzalx | Hello. I get http://pastebin.com/m1e019963 and this breaks my heart |
| 21:14.13 | *** join/#asterisk [netman] (n=netman@203.Red-81-36-132.dynamicIP.rima-tde.net) |
| 21:15.12 | *** join/#asterisk therealcircut (i=circut@smoke.dope.org) |
| 21:15.27 | therealcircut | hey all, anyone have a document pertaining to zapata -> dahdi configuration? |
| 21:15.45 | therealcircut | upgrading my asterisk install and I'm having trouble configuring the channels |
| 21:15.47 | ariel_ | brad_mssw: I have never used the web for setting up the polycom's. I have always used the ftp way |
| 21:15.49 | therealcircut | incomign & outgoing |
| 21:16.54 | sjzzalx | How can I get Speex to work given http://pastebin.com/m1e019963 ? Thing isn't working ;( |
| 21:21.02 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194) |
| 21:26.47 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 21:30.19 | *** part/#asterisk jelly-bean (n=jelly@c-76-23-46-62.hsd1.ut.comcast.net) |
| 21:30.22 | *** join/#asterisk jelly-bean (n=jelly@c-76-23-46-62.hsd1.ut.comcast.net) |
| 21:31.37 | Kobaz | how do i reset the sangoma t1 counters in wanpipemon |
| 21:32.17 | zchaos | anyone know how to rename a file in linux using the MC editor |
| 21:33.40 | beherit | zchaos: its in the menu |
| 21:36.43 | wilsonj | zchaos, what's wrong with vim? |
| 21:44.07 | *** join/#asterisk jameswf (n=james@unaffiliated/jameswf-home) |
| 21:46.28 | sjzzalx | so guys, how are you all? If I remove a thing from sip.conf that lives in user.conf, I can no longer register. Is this supposed to happen? Word on the street is that users.conf should be the only place I have to fill in these things. Is this a build flag or something I'm missing? Using Asterisk 1.6.1.0 |
| 21:48.13 | *** join/#asterisk telecos (n=sergio@66.167.219.87.dynamic.jazztel.es) |
| 21:50.10 | *** join/#asterisk asteriskmonkey (n=philip@69.77.169.14) |
| 21:50.37 | asteriskmonkey | has anyone had an issue where they get a 20 second audio blankout after during a call that has gone on for > 8 mins? |
| 21:51.21 | Kobaz | i wouldn't recommend using users.conf... set up your users in sip.conf and iax.conf respectively |
| 21:51.21 | asteriskmonkey | i seem to be having a number of ppl complain about an audio gap thats occuring on long conversations |
| 21:51.48 | [TK]D-Fender | ~users.conf |
| 21:51.49 | infobot | users.conf is a flaming pile of sh1t that takes the fine control of several perfectly usable * config files and reduces them to the lowest common denominator and makes your system behave like a "toaster grade" PBX system. |
| 21:52.16 | *** join/#asterisk qdk (n=qdk@0x573d8dd1.bynqu1.dynamic.dsl.tele.dk) |
| 21:52.31 | Kobaz | good way to put it |
| 21:52.46 | *** join/#asterisk jeffspeff (n=jeffspef@c-98-240-118-231.hsd1.ky.comcast.net) |
| 21:53.17 | jeffspeff | hey, can anybody help me with a price quote for my first * voip sale? |
| 21:53.29 | Kobaz | jeffspeff: vonage.com |
| 21:53.48 | jeffspeff | Kobaz, looking to offer a lot more than vonage |
| 21:54.00 | bmoraca | jeffspeff: cost + 20% + whatever labor @ whatever rate |
| 21:54.08 | sjzzalx | [TK]D-Fender: is that to say, use sip.conf instead of users.conf? |
| 21:54.10 | drmessano | ~sipnat |
| 21:54.10 | infobot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 21:56.12 | bmoraca | sjzzalx: using users.conf removes quite a bit of granularity from Asterisk and makes it very difficult to diagnose problems. so...yes...don't use users.conf. |
| 21:57.46 | jeffspeff | bmoraca, * hardware is going to be $800 USD (my cost), plus $1600 for all the phones... going to setup Teliax reseller account... i want to be competetive for my area, but not charge too much... currently the client leases the voip equipment for $500 p/mo. and they are very unhappy with it... i'm looking to just sell them service and teach their in-house IT how to use it |
| 21:58.37 | sjzzalx | bmoraca: thanks. |
| 21:59.27 | *** join/#asterisk Nugget (i=nugget@carrera.macnugget.org) |
| 22:01.58 | *** join/#asterisk nullable_type (n=nullable@hq.verbx.net) |
| 22:02.13 | *** join/#asterisk ingenius (n=alektro@OL77-237.fibertel.com.ar) |
| 22:02.27 | nullable_type | What is the best tool to convert stereo .wav files to best quality gsm possible |
| 22:03.40 | bmoraca | jeffspeff: like I said... cost + 20% + whatever labor @ whatever rate. unless you're looking to lease the equipment to them...if that's the case, you'll need to figure out what kind of lease, etc. if you're wanting to sell them the voip trunking, that's something that depends on what your costs are and what your client wants to pay |
| 22:03.59 | jelly-bean | what's the easiest way to setup the Polycom FTP dir |
| 22:08.16 | *** join/#asterisk mazpe (i=mazpe@c-75-74-199-171.hsd1.fl.comcast.net) |
| 22:10.30 | [TK]D-Fender | jelly-bean: Go download an appropriate firmware from : http://downloads.polycom.com/voice/voip/sip_sw_releases_matrix.html |
| 22:10.37 | [TK]D-Fender | jelly-bean: And get the admin guide. |
| 22:11.01 | [TK]D-Fender | jelly-bean: The WIKI has a provisioning handbook that can be used to help you learn what you need to configure them |
| 22:11.04 | [TK]D-Fender | ~wikis |
| 22:11.05 | infobot | [~wikis] VoIP Wiki covering Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE. http://www.voip-info.org (c) Arte Marketing Inc / CommPartners |
| 22:11.12 | *** part/#asterisk asteriskmonkey (n=philip@69.77.169.14) |
| 22:11.59 | *** part/#asterisk ManxPower (n=manxpowe@user-24-236-95-236.knology.net) |
| 22:14.51 | drmessano | Anyone know what packages I need for CentOS to get IMAP support working in Asterisk? |
| 22:15.14 | *** join/#asterisk Nugget (i=nugget@carrera.macnugget.org) |
| 22:15.25 | drmessano | The app mentioned by Asterisk doesnt tell me a lot |
| 22:15.53 | drmessano | imap_tk doesn't mean anything to me |
| 22:16.33 | KyleK | so you need to figure out which imap client libraries? |
| 22:17.39 | drmessano | yeah |
| 22:19.27 | drmessano | I did a yum list | grep imap and got cyrus-imapd, cyrus-imapd-devel, ..-perl, and ..-tools, along with php-imap.. So I am guessing I need to go out and find it |
| 22:20.14 | drmessano | Unless its staring me in the face and i've no idea what I am looking at |
| 22:21.14 | KyleK | are you sure they didn't just write thier own imap client? |
| 22:21.24 | KyleK | libc-client |
| 22:21.59 | sjzzalx | Hello! Is asterisk-addons supposed to match asterisk version? |
| 22:22.07 | sjzzalx | Or should I always be using the latest asterisk-addons? |
| 22:22.14 | drmessano | make menuconfig lists an extenal dependency called "imap_tk" |
| 22:23.02 | exothermc | anyone have any idea what would cause this: http://pastebin.com/mc3b1670 |
| 22:23.24 | drmessano | Ah, hang on.. getting closer, I think |
| 22:23.29 | KyleK | http://forums.digium.com/viewtopic.php?p=41932&sid=e380c817cd17f740f25617bd2030e063 |
| 22:23.44 | KyleK | did you try installing libc-client-devel |
| 22:24.53 | *** join/#asterisk timeshell (n=chatzill@gw.lusi.on.ca) |
| 22:26.58 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 22:29.09 | drmessano | humps KyleKs leg |
| 22:29.25 | drmessano | That did it |
| 22:29.26 | drmessano | Thanks! |
| 22:35.30 | drmessano | 1.6.2 beta 2.. Damn.. Slow down guys |
| 22:36.38 | exothermc | KyleK: Was that in reference to my post? |
| 22:36.48 | therealcircut | hey all, im having a problem with WaitExten() |
| 22:36.49 | drmessano | No it wasnt |
| 22:37.29 | KyleK | nope just google whoring |
| 22:38.09 | exothermc | Then anyone have any idea why asterisk won't compile? http://pastebin.com/m350d0b5b |
| 22:38.13 | drmessano | KyleK: I guess the idea was to name it as far away from being an IMAP client library as possible |
| 22:39.25 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 22:39.35 | exothermc | [TK]D-Fender: any insight? |
| 22:39.50 | drmessano | exothermc: Are you using SVN? |
| 22:40.07 | exothermc | drmessano: Nope that was just plain tarball |
| 22:40.31 | drmessano | Grab an earlier tarball and try it |
| 22:41.03 | exothermc | drmessano: downgrade? |
| 22:41.29 | drmessano | Since you haven't installed anything, its hardly a downgrade |
| 22:41.34 | drmessano | Incorrect term |
| 22:41.41 | drmessano | Get an earlier tarball |
| 22:42.08 | exothermc | drmessano: got one you suggest? |
| 22:42.20 | drmessano | Anything other than the one you have |
| 22:43.17 | *** part/#asterisk jelly-bean (n=jelly@c-76-23-46-62.hsd1.ut.comcast.net) |
| 22:46.14 | therealcircut | is it the expected behavior of WaitExten() to read digits one at a time |
| 22:46.15 | therealcircut | ? |
| 22:50.09 | [TK]D-Fender | therealcircut: no |
| 22:50.58 | exothermc | ok must have been something I picked in menuselect since starting from fresh works. |
| 22:51.05 | *** join/#asterisk styelz (n=yoohoo@m0o0.mooo.com) |
| 22:51.27 | exothermc | drmessano: Your suggestion makes no sense for a software platform as popular as this one. |
| 22:52.02 | *** join/#asterisk timeshell (n=chatzill@gw.lusi.on.ca) |
| 22:52.06 | drmessano | exothermc: Fine, let me hold your damn hand as you count backwards from the current release |
| 22:52.37 | drmessano | If LATEST doesnt work, go BACK ONE |
| 22:52.38 | *** join/#asterisk voxter (n=voxter@190.10.13.241) |
| 22:52.43 | drmessano | How hard is that? |
| 22:52.49 | exothermc | drmessano: No your suggestion that an earlier build would likely solve the problem. |
| 22:52.56 | timeshell | drmessano, courtesy please. |
| 22:53.01 | drmessano | It has often times before |
| 22:53.16 | drmessano | timeshell: Yawn |
| 22:53.44 | exothermc | drmessano: So just to clarify asterisk often releases stable builds that don't compile? |
| 22:53.45 | timeshell | drmessano, This is a public channel where people come for help, not to be put down |
| 22:53.45 | *** join/#asterisk davidf88 (n=davidf88@92.2.162.2) |
| 22:53.51 | exothermc | smells bullshit. |
| 22:54.52 | drmessano | timeshell: If you plan to lecture me, be prepared to do so for all parties involved. Thank you for your courtesy |
| 22:55.12 | timeshell | drmessano, when I'm monitoring this channel, I will do so. |
| 22:55.15 | drmessano | exothermc: it wouldnt be the first time there was a BUG in a RELEASE.. this is why they make MORE releases |
| 22:58.31 | [TK]D-Fender | exothermc: Often, no, but its has happened |
| 22:59.02 | [TK]D-Fender | exothermc: And what drmessano suggested is a common, and perfectly sane debugging & testing method |
| 23:01.14 | exothermc | [TK]D-Fender: Ya I know it makes sense, but I just thought it wasn't likely to still be a existing bug 1.5 months later. Anything is possible but that stretches out into the far reaches of the probable. |
| 23:01.54 | exothermc | [TK]D-Fender: bugs can exist a lot longer than that, but not standard compile bugs. |
| 23:02.07 | exothermc | *usually* |
| 23:02.18 | [TK]D-Fender | ....usually |
| 23:04.29 | *** join/#asterisk wimt_ (i=wimt@freenode/staff/wikipedia.wimt) |
| 23:12.45 | *** join/#asterisk arthax0r (i=arthax0r@gateway/shell/blinkenshell.org/x-858439566dd0f064) |
| 23:14.11 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 23:15.46 | *** join/#asterisk kerx (n=root@adsl-69-105-171-127.dsl.irvnca.pacbell.net) |
| 23:16.44 | kerx | Hi, anyone know if it's not possible to do a AMI Redirect on a channel that is inside a AGI? It looks like for me it's creating some lock or something that's not allowing the channel to Redirect properly. This is Asterisk 1.4.23.1 |
| 23:21.10 | *** join/#asterisk Foriskak__ (n=Foriskak@69.38.211.98) |
| 23:21.47 | Foriskak__ | what is the recommended way of placing calls/playing wavs with perl and SIP? |
| 23:22.07 | Foriskak__ | I have found only this site: http://community.moertel.com/ss/space/Talk+-+Fun+with+Asterisk+and+Perl |
| 23:22.18 | bmoraca | Foriskak__: depends on exactly what you want to do |
| 23:23.04 | Foriskak__ | bmoraca: basically, have a standalone perl script register itself to a SIP account (know how to do this), place a call (know how to do this too), and then, figure out if someone picks up, and if so, play a wav |
| 23:23.20 | Foriskak__ | the last two parts are not clear with net::sip::simple module |
| 23:23.42 | Foriskak__ | was wondering if someone has already done this rather simplistic thing |
| 23:24.04 | Foriskak__ | ideally, this should be possibe without involving asterisk, as it may be down/not have lines out when this script ought to run |
| 23:24.07 | bmoraca | Foriskak__: pretty silly to do that with Perl. or, if you have to use Perl, use it only to connect to AMI and originate the call to an Asterisk extension that does everything else. |
| 23:24.30 | [TK]D-Fender | Foriskak__: huh? If someone picks up what? |
| 23:24.41 | Foriskak__ | [TK]D-Fender: if someone picks up the call |
| 23:24.42 | *** part/#asterisk sjzzalx (n=jeff@c-76-23-46-62.hsd1.ut.comcast.net) |
| 23:24.46 | [TK]D-Fender | Foriskak__: WHAT call? |
| 23:24.56 | Foriskak__ | I do not want to involve Asterisk...it may be down, point of this sript is to alert that condition |
| 23:25.07 | Foriskak__ | [TK]D-Fender: the call the perl script places via the SIP account |
| 23:25.38 | [TK]D-Fender | Foriskak__: ... |
| 23:25.42 | bmoraca | Foriskak__: might want to ask in a perl chat room then... |
| 23:25.53 | [TK]D-Fender | Foriskak__: Ok, try to start witha complete and coherent plan... |
| 23:26.10 | Foriskak__ | [TK]D-Fender: ? Do you not understand what I typed above? |
| 23:26.20 | [TK]D-Fender | Foriskak__: Little bits. |
| 23:26.57 | [TK]D-Fender | Foriskak__: if this has nothing to do with *, why did you reference that AGI script and why are you here? |
| 23:28.24 | kerx | Watch out with AGI and AMI, it looks like I've just experienced that you can't really Redirect (transfer) a call using AMI while a call is inside an AGI. |
| 23:28.35 | Foriskak__ | I reference the AGI code because that is all I could google up about this task |
| 23:28.39 | kerx | You must hack it out another way like I'm about to try to :) |
| 23:29.06 | [TK]D-Fender | Foriskak__: Well AGI has nothing to do with SIP programming whatsoever |
| 23:29.12 | Foriskak__ | I did not find anyone doing this without involving *. but since you people must have thought about this before... maybe there is a SIP channel |
| 23:29.40 | [TK]D-Fender | Foriskak__: Don't go to a dog show and expect to hear people talking about CATS. |
| 23:30.19 | Foriskak__ | what other frequented channels involve SIP? |
| 23:30.36 | *** join/#asterisk axisys (n=axisys@ip68-98-177-71.dc.dc.cox.net) |
| 23:30.58 | kerx | Foriskak__: You can try #sip, #freeswitch, #kamailio, #openser, #trixbox |
| 23:31.04 | kerx | Most likely they will know how to speak SIP |
| 23:31.26 | *** join/#asterisk The_Boy_Wonder (n=davidvos@asterisk/batman-developer/dvossel) |
| 23:31.36 | *** join/#asterisk putnopvut (n=putnopvu@asterisk/master-of-queues/mmichelson) |
| 23:31.36 | bmoraca | I can speak SIP... |
| 23:31.36 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 23:31.44 | bmoraca | 0101001010010101001001010101010011110010100101001001 |
| 23:31.46 | *** join/#asterisk mnicholson (n=mnichols@nat/digium/x-376ab605af0942f0) |
| 23:31.47 | *** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler) |
| 23:32.21 | KyleK | Foriskak__: if you've gotten the sip packets down, you need to figure out RTP streams |
| 23:32.22 | *** join/#asterisk BadHAL (n=nn@66.194.174.11) |
| 23:32.24 | *** join/#asterisk angler (n=angler@pdpc/sponsor/digium/angler) |
| 23:32.24 | *** mode/#asterisk [+o angler] by ChanServ |
| 23:33.01 | drmessano | RTP is easy |
| 23:33.05 | drmessano | --> |
| 23:33.07 | drmessano | Err ummm |
| 23:33.10 | drmessano | --> |
| 23:33.12 | drmessano | Shit, NAT |
| 23:33.40 | eppigy | u a nat |
| 23:34.57 | *** join/#asterisk fish-bulb (n=cstewart@nat/digium/x-41da6ace7c4ace3b) |
| 23:38.26 | *** join/#asterisk Qwell (i=north@pdpc/sponsor/digium/Qwell) |
| 23:38.26 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 23:40.58 | *** join/#asterisk Aiatek (n=munoz@190.94.5.219) |
| 23:43.38 | *** join/#asterisk ghento (n=ghento@CPE0014bfaf7d7a-CM00159a03312c.cpe.net.cable.rogers.com) |
| 23:50.07 | exothermc | exit |
| 23:50.12 | *** part/#asterisk exothermc (n=miles@74.85.89.146) |
| 23:50.49 | *** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim) |