| 00:00.08 | mvanbaak | AgentLogin is deprecated |
| 00:00.31 | mvanbaak | but that doesn't matter |
| 00:00.39 | mvanbaak | set the callerid on the calling channel |
| 00:00.46 | mvanbaak | like the snippets I gave you |
| 00:01.06 | Yourname`` | I know it's depricated.. I love it though |
| 00:01.21 | Yourname`` | Agents try to say stupid shit like "oh i couldn't get to the call on time" |
| 00:01.22 | mvanbaak | put a 'Set(CALLERID(name)=foo)' before you call the Queue app |
| 00:01.31 | Yourname`` | Increased our call answer rate by 20% by putting them on AgentLogin |
| 00:01.53 | Yourname`` | I do use that! But the PHONE doesn't get a call.. the call comes in their ear right after the beep. |
| 00:02.03 | Yourname`` | If the phone rang, yes they'd see the CALLERID(name) |
| 00:02.13 | Yourname`` | But agentLogin doesn't let the phone ring is what I'm saying |
| 00:02.22 | irieKen | nhuilsman_work: xten is echoing like crazy, and I think that it is a software issue. Do you recommend another softphone? |
| 00:02.39 | mvanbaak | irieKen: sjphone |
| 00:02.58 | Yourname`` | Since the time the agent does the AgentLogin, he is "permanently" logged in. Meaning he'll have to have the rcvr to his head as long as he's logged in. To logout, he puts the rcvr back down in the cradle. |
| 00:03.32 | mvanbaak | Yourname``: isn't there some setting in the queue to playback a soundfile ? |
| 00:03.38 | mvanbaak | and let them ack the call |
| 00:04.08 | Yourname`` | mvanbaak: Instead of letting them ack the call, it has this announce= feature for each queue. |
| 00:04.24 | mvanbaak | and that's not working ? |
| 00:04.27 | Yourname`` | mvanbaak: I was looking at it last night, and seanbright recommended it earlier today so I think Illl be using that (havent tested it yet) |
| 00:04.30 | Yourname`` | BUT. |
| 00:04.54 | Yourname`` | There are other "sources" of these calls too that don't go into the queue. I mean from 2-3 DIDs. |
| 00:05.08 | mvanbaak | ah |
| 00:05.18 | Yourname`` | Thats what is perplexing moi :( |
| 00:05.19 | mvanbaak | they get them using a Dial ? |
| 00:05.44 | mvanbaak | you can run a macro |
| 00:05.54 | mvanbaak | with the M flag to the Dial application |
| 00:06.06 | Yourname`` | Get what using a dial? |
| 00:06.21 | mvanbaak | the calls not going through a queue |
| 00:06.28 | Yourname`` | http://pastebin.ca/1092972 -> for example |
| 00:06.44 | mvanbaak | I'm not going to look at that now |
| 00:06.50 | mvanbaak | it's 2:06 here |
| 00:06.59 | Yourname`` | Its not a paste.. just something I typed in |
| 00:07.00 | Yourname`` | lol |
| 00:07.03 | mvanbaak | at 6:00 my alarm will wake me up |
| 00:07.09 | mvanbaak | 4 hours of sleep |
| 00:07.12 | mvanbaak | ugh |
| 00:07.19 | mvanbaak | sorry |
| 00:07.22 | Yourname`` | Oh shucks, I know how that is.. sorry man, have a good night thanks a lot tho |
| 00:07.30 | mvanbaak | no problem |
| 00:07.34 | mvanbaak | hope you get it fixed |
| 00:07.38 | mvanbaak | latero all |
| 00:07.46 | Yourname`` | Just a matter of finding a way.. |
| 00:07.50 | Yourname`` | latero el mvanbako |
| 00:08.06 | irieKen | mvanbaak: Seems like it's a combination of Grandstream phone and incoming calls:S |
| 00:11.23 | *** join/#asterisk russellb_ (n=russellb@asterisk/developer-and-stable-maintainer/drumkilla) |
| 00:11.23 | *** mode/#asterisk [+o russellb_] by ChanServ |
| 00:13.23 | Nugget | http://macnugget.org/stuff/asterisk-irc.txt <-- Grandstream phone |
| 00:13.57 | Katty | hai Nugget |
| 00:14.03 | Nugget | huggles katty! |
| 00:15.02 | jblack | Nugget: This channel doesn't cover trixbox, because it's too different from asterisk. They add extra stuff in. |
| 00:15.20 | Nugget | that noise you hear, jblack, is the joke flying over your head |
| 00:15.45 | jblack | I'm pretty sure the noise I hear is the air conditioner. |
| 00:16.09 | Nugget | are you under the impression that I was asking for help with trixbox? |
| 00:16.35 | jblack | Yes, becuase you pasted a link to what is presumably your site, which contains questions. |
| 00:16.43 | Nugget | wow. |
| 00:16.43 | Nugget | just. |
| 00:16.44 | Nugget | wow. |
| 00:17.18 | Nugget | do you have aspergers or something? |
| 00:17.22 | jblack | At best, your career as a comedian is FAIL. |
| 00:17.43 | jblack | I might. |
| 00:18.15 | *** join/#asterisk russellb (n=russellb@asterisk/developer-and-stable-maintainer/drumkilla) |
| 00:18.15 | *** mode/#asterisk [+o russellb] by ChanServ |
| 00:18.27 | Yourname`` | The first time I heard the word aspergers, I thought wtf is assburgers.. |
| 00:20.54 | angryuser | we have hot dogs why not assburgers ? |
| 00:21.20 | jblack | the first time a shrink accused me of possibly having it, I first read it as asperagus. |
| 00:21.33 | *** join/#asterisk ZenBSDi (n=zenbsdi@unaffiliated/ZenBSDi) |
| 00:21.48 | jblack | I got really annoyed. "This guy doesn't know shit. I HATE asperagus". It made much more sense the the second reading. ;) |
| 00:24.21 | nhuisman_work | jblack: see our first clue is when someone admits to being to a shrink |
| 00:24.21 | nhuisman_work | hehe |
| 00:24.44 | *** join/#asterisk wonko2 (n=wonko@wiggum.4amlunch.net) |
| 00:25.46 | jblack | The funny thing is, sometimes they get spookily close to the truth. |
| 00:26.33 | Yourname`` | And if you're from India, the minute that truthness comes closer, the guy leans over to the shrink and the minute it's 99% true just stands up in joy saying "YES YOU ARE RIGHT!" |
| 00:26.45 | Yourname`` | However, the guesswork only said something about the past and the current. NOT the future. |
| 00:27.07 | Yourname`` | And that, my friends as they say, gets lost in translation. |
| 00:27.13 | jblack | Isn't diagnoses mostly about identifying the current? |
| 00:30.46 | Yourname`` | Is it? |
| 00:30.52 | Yourname`` | I don't think so.. |
| 00:31.04 | Yourname`` | Nvmd, what am I sayin', I've never even been to one. |
| 00:31.50 | *** join/#asterisk SanityIO (n=SanityIO@77.242.105.93) |
| 00:31.52 | nhuisman_work | anyone here using cisco 7940 phones? |
| 00:31.59 | nhuisman_work | i have a question about timezone settings |
| 00:32.04 | nhuisman_work | my are always off by one hour. |
| 00:32.12 | jblack | Sounds like dst to me. |
| 00:32.16 | nhuisman_work | me too |
| 00:32.25 | jblack | Would "always" be 'since spring' ? |
| 00:32.27 | Yourname`` | I'm willing to sell a 7960 to somebody for only one hundred dollars. |
| 00:32.36 | nhuisman_work | I can't figure out what timezone is hawaii |
| 00:32.39 | nhuisman_work | they all seem to be wrong |
| 00:32.39 | Yourname`` | Never used if it looks like it, but that's only because it was collecting dust. |
| 00:32.50 | Yourname`` | timeanddate.com nhuisman_work |
| 00:33.05 | nhuisman_work | Yourname``: it seems like the cisco phones don't obey that |
| 00:33.29 | nhuisman_work | HST != hawaii |
| 00:33.57 | Yourname`` | lol so calculate it in GMT or UTC and add or subtract the offsets? |
| 00:35.04 | nhuisman_work | yeah I can't figure out what syntax to use to do say GTM -10 |
| 00:37.08 | Yourname`` | ah |
| 00:40.06 | Katty | dumdedum. |
| 00:40.10 | Katty | twiddles thumbs. |
| 00:40.30 | nhuisman_work | suggestions on the gmt bit anyone? |
| 00:41.01 | Yourname`` | That's the only one I could think of man, sorry. |
| 00:41.19 | nhuisman_work | yeah that's what I came up with I Tried "GMT -10", "GMT-10" ,e tc |
| 00:41.57 | nhuisman_work | i tried disabling dst too |
| 00:42.36 | wonko2 | wait, HST10 is hawaii, why doesn't that work? |
| 00:42.38 | Yourname`` | I'm not sure about how those phones work, but why don't you try setting an ntp server through your dhcp server? |
| 00:42.43 | nhuisman_work | i do have one |
| 00:44.16 | nhuisman_work | hmm |
| 00:44.23 | *** join/#asterisk implicit (n=bayan@unaffiliated/implicit) |
| 00:44.32 | nhuisman_work | maybe I just need to use GMT |
| 00:44.38 | nhuisman_work | and then set the offset with a different flag |
| 00:44.43 | wonko2 | does Cisco understand that newfangled Country/City notation? |
| 00:45.00 | nhuisman_work | like "Alaskan Standard/Daylight Time " |
| 00:45.11 | nhuisman_work | the 7970 apparently does |
| 00:46.41 | wonko2 | the only other thing i can find for hawaii is HAST |
| 00:46.46 | wonko2 | which may or may not work |
| 00:46.51 | nhuisman_work | yeah |
| 00:47.14 | nhuisman_work | i tried AHST too |
| 00:47.52 | wonko2 | i'm fresh out of ideas then, i think you have the right idea with GMT+offset |
| 00:48.00 | wonko2 | good luck making the phone do that |
| 00:48.08 | wonko2 | that'll teach you to live in Hawaii. ;) |
| 00:48.09 | nhuisman_work | fucking cisco |
| 00:49.17 | wonko2 | hmmmm |
| 00:49.24 | *** join/#asterisk [T]ank (n=ckwall@c-71-199-25-239.hsd1.co.comcast.net) |
| 00:49.25 | wonko2 | i don't think my grandstreams are stable |
| 00:49.30 | [T]ank | if anyone is in the market for them, I am selling linksys spa 942s. I have aprox 100 for sale. Asking anywhere from $50 to $70 depending on quantity purchased. |
| 00:52.21 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 00:54.04 | nhuisman_work | i wonder if just using "GMT" and then it uses the tftp properly |
| 00:54.07 | nhuisman_work | err ntp |
| 00:56.05 | wonko2 | ntp isn't going to change your time zone |
| 00:56.25 | Qwell | [T]ank: I'll give you $20. |
| 00:56.31 | nhuisman_work | yeah but I see lots of configs that use GMT |
| 00:56.37 | nhuisman_work | and no offsets at all, how can they be working |
| 00:56.58 | wonko2 | maybe people just learned to live with it? ;) |
| 00:57.45 | [T]ank | how many do you want for that price ;-) |
| 00:57.48 | nhuisman_work | Qwell: you have any idea how I might find the current type via telnet on a phone? |
| 00:57.48 | Nugget | telnet is eeeeeeevil! |
| 00:57.53 | nhuisman_work | Timer List - Current Time: 289079849 is what I get from show time |
| 00:57.57 | nhuisman_work | which isn't in unix time |
| 00:58.04 | nhuisman_work | not sure what the hell it is |
| 00:58.53 | wonko2 | it's cisco time! |
| 00:58.54 | wonko2 | ;) |
| 00:59.00 | nhuisman_work | break it down |
| 00:59.03 | nhuisman_work | ON THE FLOOR! |
| 00:59.06 | nhuisman_work | *smash* |
| 00:59.17 | wonko2 | unix time - 12 + the current cost of gas - the mpg your car gets = cisco time |
| 01:00.20 | wonko2 | dammit, i'm going to have to go reset that ht286 as it's stopped responding at all |
| 01:00.23 | wonko2 | bah |
| 01:01.05 | nhuisman_work | just to throw you off |
| 01:01.16 | nhuisman_work | cisco "HST" = Beijing, hong kong |
| 01:01.18 | nhuisman_work | nice huh |
| 01:02.36 | nhuisman_work | if that's not smoking crack, i dunno what is. |
| 01:05.43 | thing1 | i'm trying to load cisco 7971 up with sip firmware, but the tftp logs only show Sent term71.defaults.loads to phone_ip, the phone is blank and just keeps rebooting over and over |
| 01:06.20 | wonko2 | according to the docs i find on cisco's web site: |
| 01:06.21 | wonko2 | # The following parameters setup time zone and |
| 01:06.21 | wonko2 | # daylight savings settings. |
| 01:06.21 | wonko2 | # Supported time zones are : |
| 01:06.21 | wonko2 | # EST, AST, NST, BST, AT, WAT, GMT, HST, YST |
| 01:06.24 | wonko2 | so, wtf, over? |
| 01:06.38 | nhuisman_work | that's competely wrong |
| 01:06.45 | nhuisman_work | i go into the phone and check the timezones |
| 01:06.54 | nhuisman_work | and it has one for every GMT offset |
| 01:07.34 | nhuisman_work | AHST is correct for hawaii, i just need to figure out why dst won't friggin work |
| 01:07.36 | nhuisman_work | i disabled it |
| 01:09.47 | wonko2 | ok, i don't understand how to update the firmware on these ht286s, anyone familiar with them at all? |
| 01:09.50 | wonko2 | oh wait |
| 01:09.53 | wonko2 | i'm being retarded |
| 01:09.55 | wonko2 | nevermind |
| 01:10.52 | wonko2 | forgot to switch it from http to tftp for the firmware update |
| 01:11.19 | wonko2 | now to figure out which codec i want to use |
| 01:14.26 | *** part/#asterisk korihor (n=korihor@201.211.168.130) |
| 01:24.35 | wonko2 | hmmmm |
| 01:24.51 | wonko2 | audio is good, but we get an echo back on what we say |
| 01:25.13 | *** part/#asterisk [T]ank (n=ckwall@c-71-199-25-239.hsd1.co.comcast.net) |
| 01:27.01 | Nugget | ssoouunnddss ffiinnee ttoo mmee |
| 01:28.30 | Katty | i just took some dorvaset. |
| 01:28.42 | Katty | it appears that nugget took some about 30 minutes ago. |
| 01:28.51 | Nugget | heh |
| 01:29.22 | Katty | Nugget: did i tell you my appendix decided to leave me? |
| 01:29.29 | Nugget | eek, no. |
| 01:29.30 | *** join/#asterisk BeeBuu (n=beebuu@218.13.83.170) |
| 01:29.45 | Katty | Nugget: aye. hence dorvaset. |
| 01:29.52 | wonko2 | I'm not using a zaptel, so what are my echo cancellation options? |
| 01:29.59 | BeeBuu | anyone tell me how to get DTMF when talking?thanks. |
| 01:30.07 | wonko2 | Katty: that sounds painful |
| 01:30.07 | Qwell | wonko2: on the phones |
| 01:30.15 | Katty | wonko2: not really. |
| 01:30.30 | nhuisman_work | isn't there an echo module asterisk can have added to it? |
| 01:32.07 | wonko2 | what i've found so far is for zaptel, but i have only just started looking |
| 01:32.37 | *** join/#asterisk codefreeze-lap (n=murf@216.166.159.235) |
| 01:32.50 | drmessano | Why do you need echo cancellation on something other than Zaptel? |
| 01:32.53 | *** part/#asterisk codefreeze-lap (n=murf@216.166.159.235) |
| 01:33.05 | nhuisman_work | obviously because it's echoing :P |
| 01:33.05 | drmessano | For $1000 I will code a nice IAX2 echo canceller |
| 01:33.20 | wonko2 | too bad i use SIP |
| 01:33.21 | wonko2 | ;) |
| 01:33.30 | drmessano | I'll make a SIP one too |
| 01:33.32 | nhuisman_work | drmessano: how would you code that? |
| 01:34.09 | drmessano | Actually, I could do it in under 100 lines of code probably |
| 01:34.16 | BeeBuu | drmessano: how can i catch DTMF when phone talking? |
| 01:34.32 | Nugget | feh, you could do it easy in just one line of perl. while(<VOIP>) { $_ =~ s/echo//g; print $_; } |
| 01:34.46 | drmessano | Perl makes me sick.. |
| 01:35.04 | drmessano | I bet this is a legit perl application: 3902jrfin49t98n3ffi3nt98n34( |
| 01:35.13 | nhuisman_work | yeah people write horrible perl code because they love to make things shorter then they need to be |
| 01:35.23 | nhuisman_work | built in fucking variables and crap make it impossible to read later on |
| 01:35.37 | nhuisman_work | perl is great if you follow a coding standard |
| 01:35.39 | nhuisman_work | imo |
| 01:35.46 | drmessano | Perl has a standard? |
| 01:35.48 | Nugget | I duno, "great" may be a bit strong. |
| 01:35.51 | wonko2 | ok, let's see what these ATAs can do for echo cancellation |
| 01:35.58 | drmessano | LOL |
| 01:36.01 | nhuisman_work | drmessano: no but they can be written |
| 01:36.03 | Nugget | it's better than php at least. :) |
| 01:36.09 | drmessano | wonko2: SIP doesnt echo |
| 01:36.18 | drmessano | Im sorry to be the one to tell you this |
| 01:36.21 | BeeBuu | anyone tell me how to get DTMF when talking?thanks. |
| 01:36.24 | nhuisman_work | drmessano: one of our coding projects uses perl and they have a 90 page spec |
| 01:36.44 | drmessano | I feel like I just spilled the beans on 'The birds and the bees' to a 4th grader |
| 01:36.49 | drmessano | But dude, SIP doesn't echo |
| 01:37.04 | drmessano | The easter bunny isn't real either |
| 01:37.08 | wonko2 | drmessano: so i'm imagining the echo then. :) |
| 01:37.23 | drmessano | Yes |
| 01:37.27 | drmessano | SIP doesn't echo |
| 01:37.29 | wonko2 | and so is my wife? |
| 01:37.39 | WhiteWolf | look at other causes |
| 01:37.40 | drmessano | SIP _ doesn't _ echo |
| 01:38.04 | wonko2 | ok, so it's either the 5' of phone cable, or what? |
| 01:38.12 | JT | echo is only induced in analogue circuits |
| 01:38.37 | klb | or poorly-designed IP phones |
| 01:39.54 | wonko2 | ok, so analogue handset(s) <-> Grandstream HT286 <-> Asterisk <-> VoicePulse <-> other land phone |
| 01:40.02 | drmessano | All phones have sidetone |
| 01:40.09 | drmessano | Excessive sidetone levels are NOT echo |
| 01:40.22 | wonko2 | also: analogue handset(s) <-> Grandstream HT286 <-> Asterisk <-> Grandstream HT286 <-> analogue handset(s) |
| 01:40.43 | wonko2 | so, i say something, and i hear myself repeated back to me is not echo |
| 01:40.46 | wonko2 | it's sidetone? |
| 01:41.15 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-b17f9dd761b463fc) |
| 01:41.29 | drmessano | You are right, wonko2.. SIP does indeed echo.. You caught us |
| 01:41.42 | drmessano | Shucks, fooled again |
| 01:41.54 | Katty | ^_- |
| 01:42.02 | Qwell | wonko2: No, it's a junk ATA. |
| 01:42.06 | Katty | stuffs drmessano in the closet. |
| 01:42.09 | *** join/#asterisk PeterFA (n=Peter@unaffiliated/peterfa) |
| 01:42.22 | drmessano | It's SIP echo |
| 01:42.36 | drmessano | or maybe it's IAX and SIP mixing with H323 |
| 01:42.43 | drmessano | O.o |
| 01:42.57 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 01:43.01 | lanning | reverb = no, in sip.conf |
| 01:43.03 | lanning | :) |
| 01:43.10 | drmessano | <JT> echo is only induced in analogue circuits <--- Said it best |
| 01:43.17 | drmessano | Simply, and truthfully |
| 01:43.31 | *** join/#asterisk chendy (n=chatzill@58.251.229.10) |
| 01:44.23 | wonko2 | that's all well and good, but doesn't quite help me now does it? |
| 01:44.37 | JT | klb: IP phones have analogue sections too |
| 01:44.50 | JT | wonko2: grandstreams are junk and known for echo |
| 01:44.53 | JT | just get a linksys |
| 01:44.58 | drmessano | wonko2: I wasn't trying to help |
| 01:45.06 | klb | JT, I never said they didn't |
| 01:45.14 | wonko2 | drmessano: i noticed |
| 01:45.41 | *** join/#asterisk JimVanM (n=jimvanm@bas1-toronto63-1088810630.dsl.bell.ca) |
| 01:45.44 | drmessano | wonko2: I can run over your grandstreams with my giant SUV if you like |
| 01:46.00 | drmessano | wonko2: So long as you park a couple small children next to them |
| 01:46.15 | drmessano | wonko2: I wouldn't want to be accused of singling out Grandstreams |
| 01:46.32 | AiT | wonko2, how do you like voicepulse? |
| 01:47.22 | *** part/#asterisk JimVanM (n=jimvanm@bas1-toronto63-1088810630.dsl.bell.ca) |
| 01:50.32 | wonko2 | AiT: so far so good, except that it's not good to use just yet till i get this ECHO fixed. :) |
| 01:51.16 | klb | have you replaced the analog phones? |
| 01:55.12 | wonko2 | i'm going to try the other phone just as soon as i can find my cell phone |
| 01:58.31 | *** join/#asterisk korihor (n=korihor@190.78.32.60) |
| 01:58.37 | wonko2 | hmmm, no echo there |
| 01:58.43 | wonko2 | it must be the crappy wireless phone |
| 01:59.01 | wonko2 | anyway |
| 01:59.04 | wonko2 | it's movie time |
| 02:05.47 | drmessano | Not SIP echo? |
| 02:05.49 | drmessano | o.o |
| 02:07.21 | JerJer | anyone got an example (if possible) how I can use the value of a channel variable to call a function? |
| 02:08.44 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 02:08.49 | *** join/#asterisk nn (n=nn@unaffiliated/nn) |
| 02:09.22 | JerJer | ugh - that idea won't work :( |
| 02:09.29 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.135) |
| 02:10.26 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 02:10.47 | JerJer | Set(CDR(tech)=${CHANNEL(channeltype)}) CDR(tech) is gonna return IAX2 .....the function is IAXPEER :( |
| 02:10.51 | *** join/#asterisk SwK (n=SwK@freeswitch/developer/swk) |
| 02:25.07 | [TK]D-Fender | JerJer: How does it evaluate now? |
| 02:25.21 | *** join/#asterisk N9URK (i=IceChat7@167.sub-75-249-231.myvzw.com) |
| 02:26.34 | *** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.itb.ac.id) |
| 02:27.18 | JerJer | I need IAX versus IAX2 to call IAXPEER function |
| 02:27.26 | JerJer | so i'm doing an IF function :( |
| 02:27.56 | JerJer | does vi (or anything else) do syntax highlighting for this asterisk function mess? |
| 02:28.02 | N9URK | anybody here have any luck installing asterisk-gui? |
| 02:28.21 | JerJer | i'm missing something and i don't see what :( |
| 02:29.08 | [TK]D-Fender | JerJer: ${CHANNEL(channeltype):0:3} |
| 02:29.21 | JerJer | ohhh - fun |
| 02:29.41 | JerJer | i so do not know much about how everything like that fits together |
| 02:30.10 | [TK]D-Fender | JerJer: trim nothing from the front, return 3 chars. Works for SIP/ZAP/IAX |
| 02:30.24 | [TK]D-Fender | JerJer: Fails for pretty much the rest |
| 02:30.26 | JerJer | that's good enough for this project |
| 02:30.52 | Corydon76-dig | works for any dialplan function |
| 02:31.58 | JerJer | so now how does one use that value to call another function ? ${CHANNEL(channeltype):0:3}PEER(ip) ? |
| 02:33.21 | JerJer | close, but no cigar |
| 02:33.22 | [TK]D-Fender | JerJer: ${EVAL(${CHANNEL(channeltype):0:3}PEER(ip))} |
| 02:33.28 | JerJer | "CDR(device_ip)=SIPPEER(ip)") in new stack |
| 02:33.39 | JerJer | kick ass |
| 02:34.11 | *** join/#asterisk implicit- (n=bayan@unaffiliated/implicit) |
| 02:36.23 | JerJer | hmmm |
| 02:37.45 | JerJer | not quite right yet |
| 02:37.59 | JerJer | i need ${SIPPEER(ip)} don't i? |
| 02:38.10 | JerJer | exten => s,n,Set(CDR(device_ip)=${EVAL(${CHANNEL(channeltype):0:3}PEER(ip))}) |
| 02:38.39 | [TK]D-Fender | JerYou know what... this is entirely too much to bother cramming in one line. make it a few |
| 02:38.55 | JerJer | lol - i'm a perl monger |
| 02:39.00 | [TK]D-Fender | JerJer: 1 or 2 GotoIf's would cover it nivcely |
| 02:44.05 | *** join/#asterisk uluatu (n=uluatu@200.195.161.164) |
| 02:44.24 | JerJer | ugh - this is in a macro, so i'm trying to limit the amount of branching around |
| 02:47.50 | *** join/#asterisk LeddyHM (i=leddy@da-club.with.my.beerandcondoms.com) |
| 02:51.11 | Qwell | JerJer: there should probably be an IAX2PEER function as well |
| 02:51.21 | JerJer | its IAXPEER |
| 02:51.24 | Qwell | I mean...the channel driver name is IAX2, as you pointed out |
| 02:51.28 | Qwell | I mean as a new feature |
| 02:51.37 | JerJer | oh - perhaps |
| 02:51.44 | Qwell | makes sense to me :D |
| 02:51.51 | Qwell | it's probably like 3 lines of code |
| 02:52.04 | JerJer | i have another problem with this though - I need to know the peer name :( |
| 02:52.04 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 02:56.11 | [TK]D-Fender | JerJer: CAT + Channel |
| 02:56.14 | [TK]D-Fender | CUT* |
| 02:57.01 | JerJer | yeah i am pondering that now |
| 02:57.10 | *** join/#asterisk Paige (n=Paige@208.89.241.9) |
| 02:57.14 | JerJer | realllly - segfault |
| 02:57.41 | JerJer | exten => s,n,Set(peername=${CHANNEL()}) |
| 02:57.55 | JerJer | segfaults 1.4.21 |
| 03:00.36 | [TK]D-Fender | JerJer: exten => s,n,Set(step1=${CUT(CHANNEL,/.2)) exten => s,n,Set(peername=${CUT(step1,-.1)) |
| 03:00.50 | *** join/#asterisk uluatu (n=uluatu@200.195.161.164) |
| 03:01.02 | [TK]D-Fender | JerJer: exten => s,n,Set(step1=${CUT(CHANNEL,/,2)}) exten => s,n,Set(peername=${CUT(step1,-.1)}) |
| 03:01.08 | [TK]D-Fender | JerJer: exten => s,n,Set(step1=${CUT(CHANNEL,/,2)}) exten => s,n,Set(peername=${CUT(step1,-,1)}) |
| 03:01.12 | [TK]D-Fender | 3rd times the charm |
| 03:01.51 | JerJer | nice - although these ppl use a dash in many of their peers :( |
| 03:02.02 | JerJer | but thanks - this is great |
| 03:02.18 | [TK]D-Fender | JerJer: ok, work backwards |
| 03:02.24 | JerJer | yup |
| 03:02.34 | [TK]D-Fender | JerJer: exten => s,n,Set(step1=${CUT(CHANNEL,/,2)}) exten => s,n,Set(peername=${step1:-5}) |
| 03:02.51 | [TK]D-Fender | JerJer: SIP/IAX peers use 4 digits suffixes IIRC |
| 03:02.56 | [TK]D-Fender | (plus dash) |
| 03:03.18 | JerJer | more than 4 here |
| 03:03.52 | [TK]D-Fender | JerJer: Ok, plenty of ways to skin this cat, don't get whiny with me! |
| 03:04.05 | [TK]D-Fender | :p |
| 03:04.06 | JerJer | not whining at all |
| 03:07.59 | file | setvar |
| 03:08.25 | file | set a variable in sip.conf using that, can also be used for technology... |
| 03:08.48 | Qwell | oh, nice |
| 03:08.51 | JerJer | ohh - hmm |
| 03:09.10 | Qwell | static int func_channel_read(struct ast_channel *chan, char *function, char *data, char *buf, size_t len) { if !strcasecmp(data, ... |
| 03:09.13 | JerJer | so i can set 'custom' values on peer? fun |
| 03:09.19 | Qwell | JerJer: the reason for your crash above is...pretty obvious :D |
| 03:09.50 | file | JerJer: you can set dialplan variables that will get set when a channel is created |
| 03:09.51 | Qwell | if I had my cert here, I'd go ahead and fix that... I'll try to remember tomorrow |
| 03:10.49 | JerJer | file: not following you yet |
| 03:10.49 | drmessano | I came up with a great idea |
| 03:10.53 | JerJer | googles |
| 03:10.55 | Qwell | that sscanf in the .write looks sketchy too |
| 03:11.05 | drmessano | Strap a PAP2 to to a 250GB external hard drive |
| 03:11.11 | [TK]D-Fender | JerJer: SetVar=peername=thename |
| 03:11.23 | file | JerJer: so you have a sip friend called dave... it has setvar=HELLO=world, when dave places a call then ${HELLO} will contain world in the dialplan |
| 03:12.00 | Qwell | setvar=file=POTATO |
| 03:17.32 | JerJer | hell yeah |
| 03:17.37 | JerJer | that works great |
| 03:25.57 | *** join/#asterisk Segnale007 (n=Segnale0@host68-30-dynamic.19-79-r.retail.telecomitalia.it) |
| 03:27.25 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 03:41.00 | *** join/#asterisk LemensTS (n=matthew@adsl-70-238-163-192.dsl.stlsmo.sbcglobal.net) |
| 03:42.41 | LemensTS | .. |
| 03:43.41 | LemensTS | ~freepbx |
| 03:43.41 | jbot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 03:44.19 | LemensTS | ~gui |
| 03:44.20 | jbot | somebody said gui was (Graphical User Interface) Any system that uses graphics to represent the functions of a program. All Windows operating systems are GUI. or point-and-drool, double-click, drag-and-drop, ... or at http://pla-netx.com/linebackn/guis/index.html. Of course Real Programmers use the command line interface. See cli |
| 03:44.49 | LemensTS | ~cli |
| 03:44.49 | jbot | cli is probably a Command Line Interface, the best form of interface around, of course Call-Level Interface, originally developed by SQL Access Group, but now known as Microsoft's ODBC standard. an x86 assembly instruction Common Language Infrastructure (See mono or .net) |
| 03:45.07 | LemensTS | ~.net |
| 03:45.08 | jbot | when .NET system goes online, human decisions are removed from the office environment. It contains Application Center 2000, SQL server 2000, Exchange 2000, Host Integration Server 2000, BizTalk 2000, Commerce Server 2000. They seperate every single bit of simple operation into many buzzword-servers. Don't you hate your money? |
| 03:46.15 | JT | spaam |
| 03:46.22 | jblack | jbot: please ignore lemensts |
| 03:46.36 | jblack | If only that were a real command. :) |
| 03:46.38 | LemensTS | heh it was to quite in here |
| 03:46.52 | [TK]D-Fender | LemensTS: Let the voices keep you company then. |
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| 04:04.15 | drmessano | ~~~~~~~~~~ |
| 04:04.19 | drmessano | ~~~~~~~~~ |
| 04:04.20 | jbot | I'm ignoring you now. |
| 04:04.35 | drmessano | slaps jbot |
| 04:04.47 | drmessano | slaps jbot with a worm out mIRC alias |
| 04:04.51 | drmessano | errr |
| 04:04.54 | drmessano | slaps jbot with a worn out mIRC alias |
| 04:05.06 | drmessano | ~xyzzy |
| 04:05.07 | jbot | twice as much happens |
| 04:05.10 | drmessano | yay |
| 04:05.24 | drmessano | xyzzy ftw |
| 04:05.41 | LemensTS | quit spamming |
| 04:05.46 | LemensTS | heh |
| 04:05.52 | drmessano | Are you threatening me? |
| 04:06.16 | drmessano | I am the great cornholio |
| 04:06.24 | LemensTS | do you need tp |
| 04:06.32 | drmessano | for my bunghole, yes |
| 04:06.40 | drmessano | NICARAGUA! |
| 04:07.21 | drmessano | ~fire |
| 04:07.22 | jbot | Bender : Light a fire for a man and he's warm for a night. Light a man on fire and he's warm for the rest of his life... |
| 04:08.18 | drmessano | ~hardon collider |
| 04:08.25 | drmessano | Oh come on.. |
| 04:08.28 | drmessano | Nobody keeps up |
| 04:11.48 | *** part/#asterisk wonko2 (n=wonko@wiggum.4amlunch.net) |
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| 04:24.21 | jblack | I hated that show. |
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| 04:34.26 | [TK]D-Fender | 'nite all |
| 04:34.43 | thing1 | i'm trying to load cisco 7971 up with sip firmware, but the tftp logs only show Sent term71.defaults.loads to phone_ip, the phone is blank and just keeps rebooting over and over |
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| 04:57.17 | prg3 | Where would I start looking to find out how to make this scenario work: I want call center users to call into a number, activate their phone nuber to take calls, hang up, and the system adds that number they supplied to a pool that Dial() will use for an incoming call? When they are done their shift, they call in the same "management" number, and deactivate themselves.. Having a website that they can activate and deactivate would be a cool secon |
| 05:03.48 | Corydon76-dig | prg3: replace Dial with Queue, use AddQueueMember and RemoveQueueMember, and you've just described queueing |
| 05:04.09 | nix8n82 | prg3 check out vicidial |
| 05:04.35 | Corydon76-dig | nix8n82: he's looking for incoming, not outgoing |
| 05:04.50 | nix8n82 | can do both |
| 05:04.55 | Corydon76-dig | vicidial is a predictive dialer |
| 05:05.56 | nix8n82 | it's very easy to set it up for inbound |
| 05:06.14 | prg3 | Actually, it'll drive external calls in the end.. since the users who add themselves to the queue are external numbers.. |
| 05:07.18 | prg3 | Queue sounds like what I was thinking.. I'll do some reading. |
| 05:07.23 | prg3 | I gotta run, but thanks for the tip! |
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| 07:40.24 | miloux | If the last of an extension is: exten => XX11-TIMEOUT,2, Dial(XXX/X/9999) |
| 07:40.50 | miloux | Should i put a XX11-TIMEOUT,2, Hangup after that one? |
| 07:41.06 | miloux | i mean if, it dials, it gets answered and that person hangs up but the caller doesnt |
| 07:41.12 | miloux | does it go back to the dialplan then? |
| 07:41.18 | miloux | so i should hang it up |
| 07:45.32 | miloux | Answer: if the other person hangs up, it does not return to dialplan and keeps the call "alive" |
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| 08:16.13 | defswork | What's extension state 16 ? |
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| 08:23.56 | dominic1 | can I write a if statement in ael in one line? |
| 08:24.25 | rvhi | hi, i got an irq sharing between sangoma card and nic. How do i move one to another irq? |
| 08:25.31 | defswork | rvhi: bios |
| 08:25.45 | defswork | rvhi: but generally today you don't need to worry about it |
| 08:26.24 | rvhi | defswork: got some 3-10 second silence during a call for no reason |
| 08:26.36 | defswork | that wont be interrupts |
| 08:26.38 | rvhi | defswork: goolge around and saw irq sharing is a possible reason |
| 08:26.43 | defswork | nah |
| 08:26.55 | defswork | irq sharing isnt the problem it was when you only had 15 to play with |
| 08:26.56 | rvhi | defswork: what might be the cause, then? |
| 08:27.06 | defswork | both parties had nothing to say ? :) |
| 08:27.49 | rvhi | what should i look for now? |
| 08:28.07 | defswork | check for dropped packets on your nic |
| 08:28.18 | defswork | if you aren't getting any it's not that |
| 08:28.30 | defswork | what sangoma card ? |
| 08:28.41 | rvhi | ifconfig didn't show any error |
| 08:28.43 | rvhi | a200 |
| 08:29.11 | rvhi | cat /proc/interrupts shows, |
| 08:29.27 | rvhi | <PROTECTED> |
| 08:30.33 | dominic1 | has anybody experience with func_odbc.conf? |
| 08:30.53 | *** join/#asterisk obnauticus (n=obnautic@about/windows/regular/obnauticus) |
| 08:31.11 | dominic1 | I have a very big statement and want to paste it into this config. Is it possible to write into the config over more than one line? |
| 08:32.19 | tzafrir_laptop | dominic1, what config? |
| 08:32.43 | tzafrir_laptop | asterisk .conf files? |
| 08:32.47 | dominic1 | func_odbc.conf |
| 08:32.53 | tzafrir_laptop | not AFAIK |
| 08:33.09 | dominic1 | my sql statement is very big and not readable if I paste it in one line |
| 08:33.49 | dominic1 | oooooooooooh.... |
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| 09:09.24 | DarKnesS_WolF | tzafrir_laptop: there? |
| 09:09.46 | tzafrir_laptop | DarKnesS_WolF, yes |
| 09:09.54 | *** join/#asterisk Mr_Lonely (n=u@202.56.7.140) |
| 09:09.55 | DarKnesS_WolF | i did disable busydetect and added teh country tones ton zonedata.c and indecations.conf and everything seems to work perfect now no more disconnection |
| 09:09.56 | Mr_Lonely | <PROTECTED> |
| 09:10.03 | Mr_Lonely | hii all |
| 09:10.05 | DarKnesS_WolF | sure with zaptel-1.4.11 |
| 09:10.15 | Mr_Lonely | i`m newbiw |
| 09:10.18 | Mr_Lonely | newbie |
| 09:10.24 | Mr_Lonely | would u help me? |
| 09:10.29 | tzafrir_laptop | ~ask |
| 09:10.30 | jbot | it has been said that ask is 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. |
| 09:10.42 | tzafrir_laptop | jbot's a nice guy |
| 09:11.03 | Mr_Lonely | i`m interested about asterisk |
| 09:11.16 | Mr_Lonely | nd i`m an undergraguate engineering student |
| 09:11.23 | creativx | and you are lonely |
| 09:11.25 | creativx | great match! |
| 09:11.30 | tzafrir_laptop | ~docs |
| 09:11.30 | jbot | [~docs] Asterisk documentation can be found at http://www.digium.com/index.php?menu=documentation , http://www.asteriskdocs.org , http://www.asteriskguru.com , the WIKI (~wiki), or the BOOK (~book) |
| 09:11.33 | Mr_Lonely | but i don`t know how to start nd from where? |
| 09:11.38 | tzafrir_laptop | bah, obsolete |
| 09:11.42 | creativx | ~tfot |
| 09:11.42 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com |
| 09:11.55 | Mr_Lonely | tzafrir_laptop |
| 09:11.57 | Mr_Lonely | lolz |
| 09:11.59 | Mr_Lonely | creativx |
| 09:12.18 | Mr_Lonely | i`m unknown to linux |
| 09:12.39 | Mr_Lonely | what can i do with asterisk? |
| 09:12.48 | creativx | you can start by reading the book |
| 09:12.55 | Mr_Lonely | is it appliable in switching exchange? |
| 09:13.01 | Mr_Lonely | yah i`ve downloaded that book |
| 09:13.06 | creativx | what is switching exchange |
| 09:13.14 | tzafrir_laptop | Mr_Lonely, what do you have in mind to do with it? |
| 09:13.18 | tzafrir_laptop | ~wiki |
| 09:13.26 | Mr_Lonely | telephone switching exchange |
| 09:13.32 | tzafrir_laptop | ~voip-info |
| 09:13.32 | jbot | voip-info is probably the Voice Over IP wiki. It is a community resource which will answer all of your questions, from Asterisk to ZTDummy. You can find it over at http://www.voip-info.org - well worth bookmarking |
| 09:13.55 | Mr_Lonely | true to say i don`t know what i can do with asterisk |
| 09:14.10 | creativx | do you know what you _want_ to do with it |
| 09:14.42 | Mr_Lonely | for that i`ve to know what can i do for that? |
| 09:14.58 | Mr_Lonely | what r the uses of asterisk? |
| 09:15.36 | creativx | a plethora |
| 09:15.43 | Mr_Lonely | ? |
| 09:15.47 | tzafrir_laptop | jbot, no docs is for basic documentation of Asterisk ask see http://voip-info.org/ (~voip-info) and TheBook (~book) |
| 09:15.48 | jbot | tzafrir_laptop: okay |
| 09:16.38 | tzafrir_laptop | An Asterisk is a useful wildcard in regular experssions, methinks |
| 09:17.04 | Mr_Lonely | hey bro..not fair to kid with newbie |
| 09:17.22 | Mr_Lonely | i come here to get help |
| 09:17.23 | creativx | nobody is kidding |
| 09:17.26 | creativx | you just got a lot of help |
| 09:17.47 | creativx | if you failed to recognize that.. well |
| 09:18.07 | Mr_Lonely | creativx what do u do? |
| 09:18.15 | *** join/#asterisk roxlu (n=Roxlu@90-145-42-196.wxdsl.nl) |
| 09:18.16 | tzafrir_laptop | take a look at http://voip-info.org, http://asterisk.org/ , |
| 09:18.28 | creativx | Mr_Lonely: i do a lot |
| 09:18.34 | Mr_Lonely | like? |
| 09:18.39 | tzafrir_laptop | This channel is best for focused questions |
| 09:18.39 | Mr_Lonely | thanks tzafrir_laptop |
| 09:18.53 | creativx | i integrated our CMS with asterisk using AMI |
| 09:19.10 | creativx | bringing click2dial/cid lookup/db integration to the users screens et al |
| 09:19.29 | tzafrir_laptop | creativx, you're throwing some buzzwords and not all of them are familiar :-) |
| 09:19.40 | creativx | i know ;) it was on purpose |
| 09:19.41 | creativx | hehe |
| 09:19.45 | Mr_Lonely | ufffffffffffffffffff |
| 09:19.52 | creativx | but tzafrir_laptop is right.. focused questions would be your best bet here |
| 09:20.04 | Mr_Lonely | for newbie? |
| 09:20.05 | Mr_Lonely | ?????// |
| 09:20.11 | creativx | like "why doesnt exten => _X.,1,noop(jeje) fire???" |
| 09:20.17 | tzafrir_laptop | Asterisk is basically a telephony application server |
| 09:20.33 | tzafrir_laptop | or a sort of tool-kit for creating PBXes |
| 09:21.02 | tzafrir_laptop | It's named Asterisk because, like the * in regular expressions, it can stand for anything |
| 09:21.33 | Mr_Lonely | thanks..then |
| 09:21.35 | tzafrir_laptop | (well, good thing they used basic RE . Otherwise we would have ended up with Dot) |
| 09:22.05 | *** join/#asterisk verywiseman (n=verywise@unaffiliated/verywiseman) |
| 09:22.55 | Mr_Lonely | i think i can use that in switching exchange too |
| 09:23.04 | Mr_Lonely | wow..i got a thesis sub |
| 09:23.07 | Mr_Lonely | gr8888 |
| 09:24.19 | Mr_Lonely | u dere? |
| 09:25.20 | *** part/#asterisk Mr_Lonely (n=u@202.56.7.140) |
| 09:28.22 | *** join/#asterisk pukkita (n=pukkita@137.Red-80-59-10.staticIP.rima-tde.net) |
| 09:28.28 | pukkita | hiya |
| 09:29.11 | pukkita | any known iax2 interoperatibility problems between asterisk 1.12 and 1.14??? I'm getting authority not found and username/pass are right... |
| 09:37.32 | *** join/#asterisk mikkel (n=mikkel@84-238-113-66.u.parknet.dk) |
| 09:38.19 | DarKnesS_WolF | tzafrir_laptop: with busdetect i am getting random disconnection but with busydetect=no it works fine bt when i hang up it keep running |
| 09:38.35 | DarKnesS_WolF | for like 8 secounds then hangup |
| 09:39.14 | tzafrir_laptop | pukkita, try messing with jitter-buffer related parameters. That's the black-magic tip I can give you |
| 09:39.31 | tzafrir_laptop | I'm not even sure in what context it was relevant |
| 09:39.58 | tzafrir_laptop | DarKnesS_WolF, any chance you get polarity reversal or powerdenial (ks)? |
| 09:40.58 | DarKnesS_WolF | tzafrir_laptop: from the centeral office ? |
| 09:41.06 | DarKnesS_WolF | PSTN i mean |
| 09:41.18 | DarKnesS_WolF | not sure but it is soo messy and i can't use busydetect at all |
| 09:42.24 | tzafrir_laptop | try a different progzone maybe (uk?) |
| 09:42.38 | tzafrir_laptop | or also try playing with busypattern |
| 09:43.29 | *** join/#asterisk fnordus (n=dnall@70.71.225.48) |
| 09:50.15 | DarKnesS_WolF | tzafrir_laptop: i did add busycount 8 and will test |
| 09:50.24 | DarKnesS_WolF | i have the country in the zonedata.c |
| 09:51.27 | tzafrir_laptop | zonedata is not used for busy detection |
| 09:51.49 | dominic1 | why am I not able to make select statements longer than 2035 cols? |
| 09:52.21 | tzafrir_laptop | dominic1, I probably won't be able to answer you, but, |
| 09:52.35 | tzafrir_laptop | it would help to rephrase your question as: |
| 09:53.19 | tzafrir_laptop | I try a select statement with more than 2035 cols and get the error "<insert text here>". What's the problem? |
| 09:54.59 | DarKnesS_WolF | tzafrir_laptop: mmmm progzone ? in zapata.conf ? |
| 09:55.46 | DarKnesS_WolF | ah ok got it |
| 09:55.47 | DarKnesS_WolF | let me try |
| 09:56.41 | pukkita | tzafrir_laptop: jitter-buffer? the two servers are on the same LAN... |
| 09:56.53 | dominic1 | I think I found something in func_odbc.c: char sql_read[2048]; |
| 09:57.06 | *** join/#asterisk zydoon (n=zydoon@41.225.153.114) |
| 09:58.17 | *** part/#asterisk zydoon (n=zydoon@41.225.153.114) |
| 09:59.20 | tzafrir_laptop | DarKnesS_WolF, yes, in zapata.conf |
| 10:00.23 | DarKnesS_WolF | tzafrir_laptop: one more question what doe the busypattern format like ? i mean for the tone ? i have my busy is 425/500,0/500 |
| 10:01.15 | *** join/#asterisk C4colo (n=DJpyro@66.185.107.193) |
| 10:02.03 | tzafrir_laptop | in that case you'de use busypattern=500,500 |
| 10:02.14 | C4colo | I have a client asking for a SIP softphone suggestion for Mac, any suggestions? |
| 10:02.20 | tzafrir_laptop | but also try progzone=uk . It might even work |
| 10:02.51 | DarKnesS_WolF | tzafrir_laptop: ok will d |
| 10:02.53 | DarKnesS_WolF | do * |
| 10:03.41 | pukkita | tzafrir_laptop: solved, did use permit= based on ip instead of username/secret |
| 10:04.04 | pukkita | funny worded fine between two 1.12.x and not between 1.1.2 and 1.14 |
| 10:04.29 | pukkita | between 1.12.x and 1.14.x sorry |
| 10:13.33 | dominic1 | Is it possible to add a var like ${DIALSTATUS} in a statement in func_odbc and tell the statement that it should not parse the var? |
| 10:14.51 | *** join/#asterisk yang (i=yang@CAcert/Assurer/yang) |
| 10:15.08 | yang | Does anyone use hylafax? |
| 10:17.52 | *** join/#asterisk Great_Anta_Baka (i=c4219f6b@gateway/web/ajax/mibbit.com/x-b70384e9dc58a5b7) |
| 10:21.38 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 10:23.19 | DarKnesS_WolF | tzafrir_laptop: same if u have busydetect=yes random disconnect |
| 10:23.27 | DarKnesS_WolF | if i left it the call will take like 10 secounds to hangup |
| 10:27.26 | *** join/#asterisk benneton (n=DELL@89.111.209.75) |
| 10:27.38 | benneton | Hi guys! |
| 10:28.25 | benneton | Need help.. How to connect 3 asterisk PBX with all 3 dynamic DNS. |
| 10:28.29 | benneton | Thanks |
| 10:28.55 | benneton | OpenVPN? Or there is integrated solution? |
| 10:31.24 | benneton | Anyone? |
| 10:32.51 | *** join/#asterisk shinao1 (n=shinao1@smtp.gtbank.com) |
| 10:37.21 | *** join/#asterisk roxlu_ (n=Roxlu@90-145-42-196.wxdsl.nl) |
| 10:37.48 | *** join/#asterisk SwK (n=SwK@freeswitch/developer/swk) |
| 10:39.42 | BBHoss | benneton: yeah, you could use openvpn |
| 10:45.36 | *** join/#asterisk roxlu (n=Roxlu@90-145-42-196.wxdsl.nl) |
| 10:49.28 | tzafrir_laptop | why would you need a VPN? |
| 10:49.35 | tzafrir_laptop | are the boxes behind NAT? |
| 10:49.46 | tzafrir_laptop | A VPN adds latency, generally |
| 10:58.43 | *** join/#asterisk nikolaev (n=nikolaev@78.83.145.98) |
| 10:59.41 | DarKnesS_WolF | ~book |
| 10:59.42 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 11:01.36 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 11:02.07 | *** join/#asterisk linuxstb (n=linuxstb@rockbox/developer/linuxstb) |
| 11:04.31 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 11:11.52 | *** join/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 11:14.45 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 11:16.04 | *** join/#asterisk JenniferAkemi (n=akemi@conference/cluecon/x-374a7160c490c509) |
| 11:16.15 | JenniferAkemi | morning |
| 11:17.48 | *** join/#asterisk DagMoller (n=aguirre@unaffiliated/dagmoller) |
| 11:19.21 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 11:23.35 | benneton | how to connect 3 ast pbx |
| 11:23.40 | benneton | with dyndns |
| 11:23.54 | benneton | no static ip for me |
| 11:23.59 | benneton | :) |
| 11:24.36 | tzafrir_laptop | use a local DNS server that will resolve them specifically, maybe? |
| 11:24.37 | tzafrir_laptop | not sure |
| 11:25.30 | benneton | if i enable register => ... in iax.con, then reload iax |
| 11:25.55 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 11:26.02 | benneton | and then disable comment register in iax.conf and reload asterisk |
| 11:26.08 | benneton | everything is fine |
| 11:26.24 | benneton | how to register, without register => line |
| 11:26.25 | benneton | :D |
| 11:27.07 | benneton | now, all i got is host=dyndns.address |
| 11:27.13 | *** join/#asterisk zydoon (n=zydoon@41.225.153.114) |
| 11:27.13 | benneton | and this works. |
| 11:27.37 | zydoon | I have 15 cisco ipphones running with chan_sccp |
| 11:28.22 | zydoon | from time to time they hang and get into registering loop |
| 11:28.35 | zydoon | it seems that their firmwares are buggy |
| 11:28.47 | zydoon | is it beeter to use the SIP firmware §? |
| 11:28.50 | benneton | after restart of adsl connection, ip is changed. so I need to do uncomment "register => ", and restart asterisk and comment it again and restart |
| 11:33.09 | defswork | installed his 4th install last night :o |
| 11:35.36 | *** join/#asterisk BhaalWK (i=bhaal@freenode/staff-emeritus/bhaal) |
| 11:35.43 | *** join/#asterisk rabby (n=rabby@p4FCE9826.dip0.t-ipconnect.de) |
| 11:35.45 | *** join/#asterisk roxlu_ (n=Roxlu@90-145-42-196.wxdsl.nl) |
| 11:35.49 | *** join/#asterisk angryuser (n=sldf@88.140.123.21) |
| 11:35.49 | rabby | hi |
| 11:36.51 | rabby | although all install went well, i wonder what is going on with my isdn fritz card (capi/fcpci). the capiinfo does not show any error but lots of zeroes :( http://nopaste.info/e28d17ac5a.html |
| 11:37.50 | *** part/#asterisk zydoon (n=zydoon@41.225.153.114) |
| 11:38.44 | *** join/#asterisk Dovid (n=chatzill@tony09-118-62.inter.net.il) |
| 11:39.30 | Dovid | any one here mess with cmd amd ? looked at it on the wiki but dont gully understand it |
| 11:40.34 | *** join/#asterisk scampbell (n=scampbel@mail.scampbell.net) |
| 11:41.10 | Dovid | fully* |
| 11:43.56 | *** join/#asterisk abatkin (n=abatkin@81.130.203.231) |
| 11:51.22 | *** join/#asterisk ccesario (n=ccesario@189-19-9-100.dsl.telesp.net.br) |
| 11:52.19 | *** join/#asterisk coppice (n=chatzill@27.202.17.210.dyn.pacific.net.hk) |
| 12:08.05 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 12:14.08 | *** join/#asterisk BhaalWK (i=bhaal@freenode/staff-emeritus/bhaal) |
| 12:14.10 | *** join/#asterisk russellb (n=russellb@asterisk/developer-and-stable-maintainer/drumkilla) |
| 12:14.10 | *** mode/#asterisk [+o russellb] by ChanServ |
| 12:15.17 | *** join/#asterisk shtoom (n=shtoom@121.246.167.147) |
| 12:15.49 | *** join/#asterisk theHub (n=theHub@69.177.93.21) |
| 12:18.58 | *** join/#asterisk phix (n=threat@123-243-44-131.tpgi.com.au) |
| 12:19.02 | *** join/#asterisk kaii (n=kai@ciphron.de) |
| 12:21.30 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.140) |
| 12:29.24 | Dovid | any one here mess with cmd amd ? looked at it on the wiki but dont gully understand it |
| 12:29.40 | creativx | impressive you even managed to repeat your typo :-) |
| 12:30.07 | Dovid | i thought i fixed it. this is the result of sleep deprivation |
| 12:31.10 | *** join/#asterisk neuwald (n=neuwald@189.61.96.60) |
| 12:31.27 | neuwald | hi folks. I have one asterisk just running SIP, with no digium boards. Can I run MeetMe application? |
| 12:33.37 | russellb | if you install zaptel/dahdi and load the ztdummy/dahdi_dummy kernel module, then yes |
| 12:34.10 | lesouvage | I'm trying to download the videocaps branche of Asterisk using svn. I tried "svn co http://svn.digium.com/svn/asterisk/branches/team/oej/videocaps" but got the message that the URL doesn't exist. Can anybody help me out? |
| 12:35.10 | [TK]D-Fender | russellb: dahdi_dummy <-- OMG. |
| 12:35.12 | neuwald | russellb I'm running asterisk on freebsd. Installed zaptel, but I still can't see the meetme application |
| 12:35.46 | russellb | neuwald: did you re-run the asterisk configure script and rebuild asterisk after installing zaptel? |
| 12:35.47 | russellb | [TK]D-Fender: ;) |
| 12:36.35 | creativx | dahdid a |
| 12:36.39 | [TK]D-Fender | russellb: Somebody had better go and fix all of this... (expletive deleted) naming |
| 12:37.05 | russellb | i'm not sure what you'd like me/someone to do |
| 12:37.15 | russellb | the name is ... set, unfortunately |
| 12:38.10 | russellb | I will not argue that it's kind of silly. But, we're stuck at this point |
| 12:38.43 | neuwald | russellb I didin't rebuild asterisk after installing zaptel. I'll do it now. |
| 12:38.48 | russellb | ok. |
| 12:39.18 | creativx | hehe, how much alcohol went into figuring out that name russellb? :) |
| 12:39.23 | creativx | or how many marketers |
| 12:39.34 | russellb | lesouvage: take out "branches" in that URL |
| 12:39.48 | russellb | I think it came down to finding something that could be trademarked ... |
| 12:40.00 | russellb | and the response was "well, it's kind of lame, but I can't think of anything better" |
| 12:40.11 | creativx | hehe well I like it |
| 12:40.22 | creativx | each time i read it i feel like continuing saying dahdi-da |
| 12:40.33 | russellb | heh |
| 12:40.37 | russellb | well glad somebody does |
| 12:40.40 | russellb | i'm sort of neutral on it |
| 12:41.25 | creativx | hehe oh well its just a name |
| 12:42.05 | [TK]D-Fender | creativx: Oh? http://www.switched.com/2008/08/04/verizon-bans-libshitz-family-from-using-last-name-in-e-mail-addr/ |
| 12:43.01 | *** join/#asterisk JenniferAkemi (n=akemi@conference/cluecon/x-d183c388e87a7ff1) |
| 12:43.45 | creativx | haha |
| 12:44.34 | creativx | poor fella |
| 12:45.16 | *** join/#asterisk vale-ICS (n=vale@icsnet.demon.co.uk) |
| 12:45.32 | vale-ICS | join eve-radio |
| 12:46.04 | neuwald | russellb I rebuild asterisk and there is no meetme application |
| 12:46.23 | russellb | did you re-run configure before re-building? |
| 12:46.36 | [TK]D-Fender | neuwald: You need to have zaptel configured first and should ahve ztdummy loaded |
| 12:47.10 | russellb | you don't have to do that to get it to compile ... |
| 12:48.58 | neuwald | wheel, I'm using freebsd on these box. What should I do? deinstall asterisk and install it again? |
| 12:49.55 | [TK]D-Fender | neuwald: And have you confirmed ztdummy is being loaded? |
| 12:50.08 | k-man | if i have asterisk behind NAT, do i need to do port forwarding to make outbound calls? |
| 12:50.15 | k-man | err... sip calls i mean |
| 12:50.17 | russellb | no |
| 12:50.24 | [TK]D-Fender | k-man: ... |
| 12:50.26 | neuwald | take a look: kmod_load="zaptel.ko qozap.ko tau32pci.ko wcfxo.ko wcfxs.ko wct1xxp.ko wct4xxp.ko wcte11xp.ko wcte12xp.ko" |
| 12:50.26 | [TK]D-Fender | ~sipnat |
| 12:50.27 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 12:50.34 | [TK]D-Fender | k-man: Told you before... READ THE GUIDE |
| 12:51.01 | k-man | [TK]D-Fender, i'm trying to read all that stuff but i'm just trying to understand where the connections go |
| 12:51.04 | neuwald | ztdummy isn't on freebsd ports... maybe I'll have to install it manually |
| 12:51.05 | [TK]D-Fender | neuwald: do "ztcfg -vvvv" |
| 12:51.20 | [TK]D-Fender | k-man: the FIRST one. this isn't Raw-Cat Science. |
| 12:51.30 | [TK]D-Fender | k-man: is SHOWS you where things go. |
| 12:51.55 | neuwald | http://pastebin.com/m23a8a8ff |
| 12:52.01 | neuwald | [TK]D-Fender there is the result |
| 12:52.32 | [TK]D-Fender | neuwald: neuwald Not sure under BSD, but PB : cat /proc/interrutps |
| 12:52.37 | [TK]D-Fender | neuwald: neuwald Not sure under BSD, but PB : cat /proc/interrupts |
| 12:52.59 | neuwald | I have no digium boards on that server |
| 12:53.01 | *** join/#asterisk Rico29 (n=Rico@ARennes-257-1-22-209.w81-53.abo.wanadoo.fr) |
| 12:53.26 | [TK]D-Fender | neuI KNOW. |
| 12:54.45 | neuwald | [TK]D-Fender I think it will help me: http://www.mercenary.ca/articles/zaptel_asterisk.php |
| 12:55.45 | [TK]D-Fender | neuwald : Looks relevant. |
| 12:56.30 | rwaite | what's the best way to diagnose an iax2 peer that is unreachable, yet pingable? |
| 12:56.37 | neuwald | [TK]D-Fender 30 1 0xcafe1000 2000 ztdummy.ko |
| 12:56.44 | rwaite | this is a voip provider... would it be most likely on their end? |
| 12:56.47 | neuwald | seems that ztdummy is loaded |
| 12:56.55 | neuwald | now I'll try to deinstall, clean and install asterisk again |
| 12:57.22 | [TK]D-Fender | neuwald: Now trash your * install and recompile from scratch and it should see zaptel and compile in support along with MeetMe |
| 12:57.42 | [TK]D-Fender | rwaite: Could very well be |
| 12:57.45 | creativx | rwaite: call them and ask why they arent letting in iax connections |
| 12:57.49 | *** part/#asterisk pukkita (n=pukkita@137.Red-80-59-10.staticIP.rima-tde.net) |
| 12:57.54 | rwaite | did, they're gonna call back. |
| 12:57.59 | neuwald | [root@br-poa-01 /usr/ports/net/asterisk]# make deinstall && make clean && make install |
| 12:58.05 | rwaite | just wondering if there's anything i can check on my end in the mean time. |
| 12:58.45 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 12:58.55 | [TK]D-Fender | neuwald: For 1.4 under linux you need to do "./configure" first, and verify in "make menuconfig" |
| 12:59.13 | [TK]D-Fender | rwaite: Trya passing another call and watch the packets go by |
| 12:59.15 | *** join/#asterisk benneton (n=DELL@89.111.209.75) |
| 12:59.21 | neuwald | [TK]D-Fender make install on ports will do it |
| 12:59.31 | rwaite | k |
| 12:59.58 | [TK]D-Fender | neuwald: If you say so, don't know the process changes for BSD |
| 13:02.21 | *** join/#asterisk BhaalWK (i=bhaal@freenode/staff-emeritus/bhaal) |
| 13:02.35 | *** part/#asterisk benneton (n=DELL@89.111.209.75) |
| 13:04.07 | *** join/#asterisk flush (n=SYN_SENT@ip216-239-88-163.vif.net) |
| 13:08.08 | neuwald | [TK]D-Fender yes man. thank u very much, now it's running :-) |
| 13:09.06 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 13:10.15 | lesouvage | Is there a sip video softphone available for macosx to test my videocaps branche of asterisk. |
| 13:11.01 | russellb | eyebeam, i guess |
| 13:12.07 | vale-ICS | is now away: gone |
| 13:12.41 | seanbright | eyeBeam does video? |
| 13:13.13 | russellb | no, i'm just making crap up |
| 13:13.16 | neuwald | seanbright the paid version yes |
| 13:13.22 | seanbright | interesting. |
| 13:13.33 | [TK]D-Fender | seanbright: EYEbeam. |
| 13:13.52 | seanbright | [TK]D-Fender: eyeBeam |
| 13:13.52 | vale-ICS | is back from: gone (been away for 1m) |
| 13:13.53 | [TK]D-Fender | jabs seanbright with a pointy stick |
| 13:14.27 | seanbright | i guess the video options are completely missing if you don't have a camera plugged in |
| 13:14.39 | seanbright | i don't even see the video sidecar |
| 13:15.04 | russellb | you have to turn the nublock off |
| 13:15.40 | seanbright | :) |
| 13:16.01 | *** kick/#asterisk [seanbright!n=russellb@asterisk/developer-and-stable-maintainer/drumkilla] by russellb (payback is a bitch!) |
| 13:16.01 | *** join/#asterisk seanbright (n=sean@asterisk/contributor-and-bug-marshal/seanbright) |
| 13:16.34 | seanbright | good times, good times. |
| 13:19.16 | k-man | cool, i got outbound calls working |
| 13:22.31 | *** join/#asterisk Peri (n=redanti@xtreme-6-241.dyn.aci.on.ca) |
| 13:22.58 | Peri | morning lads and lassies |
| 13:24.04 | seanbright | bobs your uncle |
| 13:24.24 | *** join/#asterisk moy (n=moy@nat/ibm/x-280993c5f279dbdf) |
| 13:24.54 | k-man | where can i read up on how inbound sip calls works with asterisk? |
| 13:25.54 | [TK]D-Fender | k-man: they get authed against your peer typically. Enable SIP DEBUG and look at the packets and you'll see what its matching |
| 13:26.15 | [TK]D-Fender | k-man: Your REGISTER should tell them what exten to hit, and the peer they match which context. |
| 13:26.48 | lesouvage | I noticed that EYEbeam is $ 60,- and I need two otherwise there nothing to test. Is there another option to play around with video before buying some serious hardware? |
| 13:27.05 | [TK]D-Fender | lesouvage: Ekiga |
| 13:27.28 | [TK]D-Fender | lesouvage: And I believe X-Lite does video now with the low-end codecs |
| 13:28.41 | k-man | [TK]D-Fender, what i want to understand is, how does the register line tell the sip provider where to send calls and does it require port 5060 or can the register line ask calls to be sent to a different port? |
| 13:29.08 | *** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee) |
| 13:29.08 | [TK]D-Fender | k-man: your BINDPORT tells which port. |
| 13:29.23 | [TK]D-Fender | k-man: and yes, thats the entire point of registereing |
| 13:29.25 | [TK]D-Fender | ~sipregister |
| 13:29.26 | jbot | [~sipregister] SIP Registration is to tell your provider as to what IP address & EXTEN to send INCOMING calls to. Some ITSP's let you use a fixed IP or host rather that register. Registration is NOT normally needed to PLACE a call, as those are typically auth'd independently. Others accept UN-AUTH'D calls once you are registered (saves on negotiation BW) |
| 13:30.16 | lesouvage | [TK]D-Fender: thanks, I look into it. |
| 13:30.29 | k-man | [TK]D-Fender, ok thanks |
| 13:31.33 | shtoom | Hi, I am facing a 20 , 20secs audio loss, on an asterisk installation which worked fine for some time but started giving this problem |
| 13:31.56 | shtoom | I've tried to upgrade to latest asterisk (1.4) version |
| 13:32.05 | shtoom | still the problem is not solved |
| 13:32.20 | [TK]D-Fender | shtoom: And your description is weak. |
| 13:33.29 | *** join/#asterisk JenniferAkemi (n=akemi@conference/cluecon/x-f0a1a63329e76721) |
| 13:33.35 | k-man | so if i set bindport to 5070 say, then register with my sip provider, the provider will try to connect to port 5070? |
| 13:33.35 | *** join/#asterisk gbr_ (n=gbr@200.103.96.98) |
| 13:34.11 | shtoom | [TK]D-Fender: the problem happens for all the calls that are handled by asterisk the machine has two network interfaces sharing same IRQ is that problem ? |
| 13:37.30 | *** join/#asterisk shido6 (n=shido6@209.114.208.192) |
| 13:39.02 | *** join/#asterisk chandoo (n=chandra@ool-4353b4c7.dyn.optonline.net) |
| 13:39.07 | Katty | herroes. |
| 13:39.50 | *** join/#asterisk jasonwoot (n=jasonrot@user-69-73-40-171.knology.net) |
| 13:40.10 | [TK]D-Fender | shtoom: You can do a lot better. |
| 13:40.15 | [TK]D-Fender | Katty: Mew. |
| 13:40.24 | [TK]D-Fender | k-man: Yes |
| 13:41.11 | *** join/#asterisk c4t3l (n=rcallico@notes.questia.com) |
| 13:43.16 | verywiseman | which is bracket using with $ ,{} or [] , for expression? |
| 13:43.29 | *** join/#asterisk scampbell (n=scampbel@mail.scampbell.net) |
| 13:44.18 | *** join/#asterisk ScarEye (n=scareye@12.27.87.66) |
| 13:45.34 | *** part/#asterisk JenniferAkemi (n=akemi@conference/cluecon/x-f0a1a63329e76721) |
| 13:47.38 | [TK]D-Fender | verywiseman: [] |
| 13:47.50 | [TK]D-Fender | verywiseman: Go lookup "asterisk expressions" on the WIKI |
| 13:47.55 | [TK]D-Fender | ~wikis |
| 13:47.56 | jbot | [~wikis] VoIP Wiki covering Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE. http://www.voip-info.org (c) Arte Marketing Inc / CommPartners |
| 13:47.58 | *** join/#asterisk n3hxs (n=HAMming@static-151-196-93-200.balt.east.verizon.net) |
| 13:48.18 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 13:48.33 | verywiseman | [TK]D-Fender, ok , i ask because i see difference btw asterisk book & voip-info |
| 13:48.49 | [TK]D-Fender | verywiseman: What page? |
| 13:49.56 | verywiseman | [TK]D-Fender, http://voip-info.org/wiki/view/Asterisk+variables |
| 13:50.27 | [TK]D-Fender | verywiseman: Ok, where in there? |
| 13:51.04 | verywiseman | [TK]D-Fender, see Using $ section |
| 13:51.26 | [TK]D-Fender | verywiseman: I don't see the error. Paste it. |
| 13:51.40 | verywiseman | MY_VAR=${SS}{EPOCH}-${SS}{EXTEN}.gsm |
| 13:52.31 | [TK]D-Fender | verywiseman: that is not an expression. |
| 13:53.02 | [TK]D-Fender | verywiseman: and is a very nifty trick |
| 13:53.40 | *** join/#asterisk LemensTS (n=matthew@adsl-70-238-187-205.dsl.stlsmo.sbcglobal.net) |
| 13:54.54 | k-man | hmm... i changed the bindport in asterisk to 5070, changed my sip phone to connect to port 5070 but it fails to connect |
| 13:55.10 | k-man | its an spa942 |
| 13:55.47 | [TK]D-Fender | k-man: You desperately need to learn to enable SIP DEBUG and friggen pastebin stuff. |
| 13:58.07 | k-man | [TK]D-Fender, actually, i don't think asterisk is receiving anything from the spa942 |
| 13:58.24 | [TK]D-Fender | k-man: Stop thinking and start SHOWING |
| 13:58.25 | k-man | [TK]D-Fender, there are no messages in sip debug from it |
| 13:58.35 | k-man | thers no messages to show |
| 13:58.53 | [TK]D-Fender | k-man: Configs then. Proof as to the port being bound. Your firewall setup, etc |
| 14:01.37 | mandd | any reason why when I put a sound on speaker phone it is breaking up, also many people are claiming that I am braking up |
| 14:01.50 | mandd | could it be some codec problem? |
| 14:02.02 | mandd | i haven't set anything, and just using default once |
| 14:02.14 | k-man | [TK]D-Fender, i set the spa942 to connect to asterisk on port 5070, and left asterisk listening on 5060, restarted the phone, and it connects - ie, its like the spa942 is ignoring the port setting |
| 14:03.04 | [TK]D-Fender | k-man: "left asterisk listening on 5060" <- I told you to change the BINDPORT. |
| 14:03.58 | [TK]D-Fender | mandd: Is is only when placeing the call on speakerphone? What about the same phone using the handset? |
| 14:05.01 | k-man | i did - and the spa942 wouldn't connect. changed bindport back to 5060 and it connects |
| 14:05.32 | k-man | even though the spa942 config says sip port 5070 |
| 14:05.34 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 14:06.26 | [TK]D-Fender | k-man: Set it to 5070 and SHOW US |
| 14:07.02 | *** join/#asterisk brodiem (n=brodiem@rrcs-24-227-171-186.sw.biz.rr.com) |
| 14:08.19 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 14:08.41 | *** join/#asterisk [Akemi] (n=akemi@conference/cluecon/x-c4032ae4758a225e) |
| 14:09.03 | *** part/#asterisk [Akemi] (n=akemi@conference/cluecon/x-c4032ae4758a225e) |
| 14:11.36 | *** join/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 14:14.58 | jblack | [TK]D-Fender: I sincerely hope you get pleasure out of having the same question five times a day, day after day. |
| 14:15.20 | jblack | With the same responses, five times a day, day after day. |
| 14:15.43 | [TK]D-Fender | jblack: "Why do we always come here? I guess we'll never know. It's like a kind of torture... to HAVE TO WATCH THIS SHOW!" |
| 14:15.50 | [TK]D-Fender | "Animal"'s out |
| 14:16.37 | jblack | some day, you're gonna get a wife taht's really going to appreciate you. |
| 14:16.49 | jblack | She'll blah blah blah, got a new skirt, and you'll say "SHOW ME!" |
| 14:17.12 | *** join/#asterisk seanmh (i=seanmh@216.31.101.25) |
| 14:17.12 | *** join/#asterisk JenniferAkemi (n=akemi@conference/cluecon/x-b1615982db1f22a8) |
| 14:17.31 | k-man | anyway, its late here |
| 14:17.42 | [TK]D-Fender | k-man: How.. PRDUCTIVE |
| 14:17.44 | k-man | i'll come back tomorrow to ask you the same questions again |
| 14:18.17 | k-man | thanks for your help [TK]D-Fender |
| 14:20.07 | *** part/#asterisk JenniferAkemi (n=akemi@conference/cluecon/x-b1615982db1f22a8) |
| 14:22.24 | *** join/#asterisk spokra (n=spokra@host093-179-177.sea0.speakeasy.net) |
| 14:22.27 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 14:22.55 | *** join/#asterisk _mwoodj_ (n=mwoodj@pdpc/sponsor/digium/hyper-eye) |
| 14:23.03 | *** join/#asterisk ltd (n=z@pat.transact.net.au) |
| 14:26.20 | *** join/#asterisk codefreeze-lap (n=murf@216.166.159.235) |
| 14:27.21 | defswork | any sangoma guys around ? |
| 14:27.48 | Qwell | defswork: what, like employees? |
| 14:27.51 | Qwell | not a chance |
| 14:27.58 | brodiem | Anyone have a sample h263 they can send me so that I can rule out my files being the problem trying to get playback to work? |
| 14:28.21 | Katty | hugs [TK]D-Fender |
| 14:28.43 | Katty | [TK]D-Fender: could i interest you in a few staples? |
| 14:29.45 | [TK]D-Fender | Katty: I prefer sugical tape personally... |
| 14:29.50 | [TK]D-Fender | surgical* |
| 14:31.16 | Katty | [TK]D-Fender: fresh out. sorry. |
| 14:31.35 | Katty | [TK]D-Fender: i made the mistake at looking at it this morning |
| 14:31.51 | Katty | [TK]D-Fender: looks like the tragic victim of a medival torture scene :< |
| 14:31.59 | [TK]D-Fender | Katty: Kinda what I figured |
| 14:31.59 | km- | [tk]d-fender: hey, did you happen to give three-way g729 a shot while you were home? |
| 14:32.11 | [TK]D-Fender | km-: Perv. |
| 14:32.19 | km- | haha |
| 14:32.20 | km- | you know it |
| 14:32.30 | [TK]D-Fender | km-: Rather I didn't ;) |
| 14:32.38 | [TK]D-Fender | km-: And no. I said "remind me" <- |
| 14:32.44 | dominic1 | isn't that possible: Gotoif($[${DIALSTATUS} = BUSY]?dial,0123456,begin) |
| 14:32.57 | brodiem | I believe you said "ask me again in 3 hours" hahah |
| 14:33.06 | km- | [tk]d-fender: yeah, didn't get much time to sneak away last night with the newborn being a whiner |
| 14:33.22 | km- | waiting for the day when he can hold his own bottle and STFU :P |
| 14:33.36 | [TK]D-Fender | dominic1: Why don't you show use your dialplan and the CLI output, and you tell US. |
| 14:34.43 | *** join/#asterisk Assid (n=assid@unaffiliated/assid) |
| 14:34.45 | defswork | I've got a A500 BRI card and it's stopped working - was ok last night |
| 14:34.54 | Assid | howdy |
| 14:35.00 | defswork | debug shows Unknown in the prot column |
| 14:35.01 | [TK]D-Fender | doody |
| 14:35.10 | Assid | [TK]D-Fender: was waiting for that ;) |
| 14:36.10 | dominic1 | *g* thanks, I think with gotoif it's not possible to got to a named label |
| 14:36.29 | [TK]D-Fender | dominic1: Yes it is |
| 14:36.57 | [TK]D-Fender | dominic1 : WTF are labels for if you can't GOTO them? Thats retarded |
| 14:36.59 | Assid | okay i plan to run asterisk in an openvz container.. im thinking.. shoudl i run postgres within the container as well or the hardware node?. |
| 14:38.44 | *** join/#asterisk fogo (n=fogo@rs-69-169-132-200-0003.broadweave.net) |
| 14:39.37 | *** join/#asterisk bijit (n=benji@190.241.15.48) |
| 14:40.58 | *** join/#asterisk chantanito (n=santiago@200-71-150-172.static.telcel.net.ve) |
| 14:42.17 | chantanito | hi, i need some help... |
| 14:42.45 | chantanito | does anyone knows what this error means: 'Could not get all 512 bytes of the header' |
| 14:43.08 | chantanito | i got this error from the log of a Polycom 501.... |
| 14:44.14 | *** join/#asterisk fogo (n=fogo@rs-69-169-132-200-0003.broadweave.net) |
| 14:44.15 | [TK]D-Fender | chantanito: What actual problem does it cause? |
| 14:44.56 | chantanito | [TK]D-Fender: an error on the phone after booting: 0x10020 |
| 14:45.15 | chantanito | [TK]D-Fender: it says Error is 0x10020 |
| 14:45.33 | hi365 | can you overide a global variable by setting it again? (i.e. in [globals], if you have foo=bar and then foo=green, what will foo be set to?) |
| 14:45.36 | [TK]D-Fender | chantanito: Possibly your firmware file is corrupted. I'd suggest re-extracting |
| 14:45.56 | chantanito | [TK]D-Fender: ok, i'll try |
| 14:46.26 | [TK]D-Fender | hi365: no, it keeps its value until you restart *. |
| 14:46.37 | [TK]D-Fender | hi365: So you need to chage it while executing the dialplan. |
| 14:47.07 | hi365 | [TK]D-Fender: so it will take and keep the first value? |
| 14:47.24 | [TK]D-Fender | hi365: what "first value"? |
| 14:47.47 | hi365 | (i.e. in [globals], if you have foo=bar and then foo=green, what will foo be set to?) |
| 14:48.17 | *** part/#asterisk fogo (n=fogo@rs-69-169-132-200-0003.broadweave.net) |
| 14:48.36 | [TK]D-Fender | hi365: on initial load it should be the 2nd. then onwards should be unchanged until you change it in the dialplan. |
| 14:49.32 | *** join/#asterisk fogo (n=fogo@rs-69-169-132-200-0003.broadweave.net) |
| 14:49.59 | mandd | [TK]D-Fender hand set work fine |
| 14:50.06 | mandd | only speakerphone |
| 14:50.36 | mandd | works* |
| 14:50.47 | hi365 | [TK]D-Fender: k. thanks |
| 14:52.16 | JSnoz | did someone already managed to use the module conntrack_sip for iptables ? |
| 14:52.27 | JSnoz | my iptables is on the same machine than my asterisk box |
| 14:52.31 | JSnoz | -box |
| 14:53.00 | *** join/#asterisk merkurie (n=merkurie@192.153.163.44) |
| 14:54.52 | merkurie | there isn't a way to specify multiple databases in extconfig.conf, right? i'm probably stuck using some type of load balance if i want to do failover? |
| 14:55.01 | *** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler) |
| 14:55.23 | [TK]D-Fender | mandd: then the speakerphone sucks |
| 14:55.39 | [TK]D-Fender | merkurie: Correct |
| 14:56.05 | [TK]D-Fender | JSnoz: To do what exactly? |
| 14:56.32 | *** part/#asterisk c4t3l (n=rcallico@notes.questia.com) |
| 14:56.41 | drako | I have a 4port ISDN and there is only working 2 lines out of 4 |
| 14:56.42 | *** join/#asterisk tobias (n=tobias@66.152.121.39) |
| 14:58.34 | *** join/#asterisk waverly360 (n=waverly@ns2.dalcon.com) |
| 14:59.17 | merkurie | [TK]D-Fender, thanks |
| 14:59.52 | *** join/#asterisk fskrotzki (n=fskrotzk@host198.textwise.com) |
| 15:00.18 | *** join/#asterisk [Akemi] (n=akemi@conference/cluecon/x-5e3e353d2231ab56) |
| 15:03.04 | *** join/#asterisk Firass-z0r (n=Firass@ead224-222.housing.wwu.edu) |
| 15:03.29 | *** join/#asterisk JenniferAkemi (n=akemi@conference/cluecon/x-d169ffa9b5af0902) |
| 15:07.52 | drako | nvm |
| 15:08.12 | drako | how can I limit a sip user to only be able to make 2 calls at same time. |
| 15:08.18 | drako | limiting sip ports |
| 15:08.47 | *** join/#asterisk [Akemi] (n=akemi@72.60.168.132) |
| 15:08.53 | *** join/#asterisk CanWood (n=chatzill@24.108.64.80) |
| 15:09.24 | jasonwoot | why did I convert to asterisk? |
| 15:09.52 | *** part/#asterisk [Akemi] (n=akemi@72.60.168.132) |
| 15:09.54 | [TK]D-Fender | 42 |
| 15:10.15 | waverly360 | heh |
| 15:10.25 | *** join/#asterisk galeras (n=galeras@201.245.54.165) |
| 15:10.57 | jasonwoot | did you ever try throwing yourself at the ground and miss? |
| 15:11.43 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 15:11.43 | mort_gib | -and completely miss |
| 15:11.47 | *** join/#asterisk l2cache (n=l2cache@179.190.204.68.cfl.res.rr.com) |
| 15:11.59 | waverly360 | I'll bet that happens to skydivers sometimes... |
| 15:12.15 | galeras | Dear sirs, i have a E1 TE121p Digium Card, i can receive calls fine, but outgoing calls got this error: "Channel 0/1, span 1 got hangup request, cause 1" Any suggestion? |
| 15:12.18 | waverly360 | I think it's possible anyhow. |
| 15:12.45 | waverly360 | galeras: Are you setting the callerid to anything in particular? |
| 15:13.04 | [TK]D-Fender | jasonwoot: Nope, 100% success |
| 15:13.35 | *** join/#asterisk iulius (n=iulius@mail1.technologieshq.com) |
| 15:13.47 | l2cache | I have agents logging in to a queue (dynamic SIP) via addqueuemember, every other day, a few random phones do not take any calls at all. There is no pattern yet. I am using rrmemory and the queue is getting full of calls, except for the random phones. 3 days ago it was 3 phones, yesterday 3 different phones, today - 6 different phones. Any guesses? |
| 15:14.25 | galeras | waverly360: no, calleird is not setted |
| 15:14.58 | [TK]D-Fender | l2cache: pastebin dumps of member status, queue status, CLI output, etc |
| 15:15.11 | *** join/#asterisk putnopvut (n=putnopvu@conference/asterisk/x-9c25c68ca4978296) |
| 15:15.11 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 15:15.24 | [TK]D-Fender | galeras: enable PRI debug if its PRI signalled |
| 15:15.27 | waverly360 | galeras: I suppose it could be a number of things. Invalid zapata.conf settings maybe? I've just experienced a lot of problems trying to make calls when I have the callerid set to something that the carrier deems invalid |
| 15:16.00 | galeras | pls, give me a sec, i will post a PRI debug. |
| 15:16.27 | *** join/#asterisk Paige_ (n=Paige@208.89.241.31) |
| 15:19.21 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 15:20.26 | galeras | Pls, take a look of http://pastebin.com/m63e75d42 |
| 15:20.48 | *** join/#asterisk implicit (n=bayan@unaffiliated/implicit) |
| 15:22.37 | *** join/#asterisk ta^3 (n=tacvbo@conference/cluecon/x-f4843577ed94baa7) |
| 15:22.42 | [TK]D-Fender | galeras: 7-digits numbers aren't valid in your area, are they? |
| 15:22.47 | waverly360 | galeras: Ext: 1 Cause: Unallocated (unassigned) number (1) |
| 15:23.48 | galeras | [TK]D-Fender: yes, here 7-digits are valid |
| 15:24.04 | [TK]D-Fender | galeras: -- Executing [86538400@from-internal:1] Dial("SIP/199-b7600590", "ZAP/1/6538400") in new stack <--- is 6538400 valid? |
| 15:24.04 | galeras | waverly360: Which mean? |
| 15:24.08 | minthome | how can the Hangup() app run for 79 seconds? |
| 15:24.33 | galeras | <PROTECTED> |
| 15:24.42 | [TK]D-Fender | galeras: Ask your telco what they see |
| 15:24.50 | minthome | i have a user that has used 23 minutes worth of Hangup() this month... as least that's what it looks like in the CDR's |
| 15:25.39 | [TK]D-Fender | galeras: "Unallocated (unassigned) number" as waverly360 mentioned doesn't make it look valid... |
| 15:25.48 | waverly360 | galeras: Well, [TK]D-Fender knows more about some of these debug lines than I do..but that looks like an unallocated number being the reason it was disconnected |
| 15:25.59 | [TK]D-Fender | minthome: thats the last app executed. Doesn't mean they spent TIME in it |
| 15:26.15 | minthome | ah, yeah, just figuring that out... thanks [TK]D-Fender |
| 15:27.38 | waverly360 | galeras: is the callerid being set anywhere before your dial command? |
| 15:28.40 | galeras | waverly360: no, however i will try to call setting callerid to something. let's try... |
| 15:28.47 | *** join/#asterisk JenniferAkemi (n=akemi@72.60.168.132) |
| 15:30.05 | waverly360 | Hey [TK]D-Fender, if I have a polycom phone with 3 lines..all tied to the same sip id, is there a way to prevent a queue from sending a call to an agent who is already on the phone? |
| 15:30.26 | waverly360 | ..or to anyone who knows how. |
| 15:30.42 | [TK]D-Fender | waverly360: numerous ways |
| 15:30.58 | *** join/#asterisk BBHoss (n=hoss@72.146.23.145) |
| 15:31.06 | waverly360 | [TK]D-Fender: I figured there were a few. Right now, we're just using the last line on the phone as the queue line..and that line is only allowed to accept one call at a time. |
| 15:31.18 | [TK]D-Fender | waverly360: if going through the dialpla, check first. If using AQM with the device, the device itself should reports its being in-use (theres a doc for this) |
| 15:31.36 | galeras | :( same result : http://pastebin.com/d695968fc |
| 15:32.04 | [TK]D-Fender | galeras: Presentation: Presentation permitted, user number not screened (0) '199' ] <- you have NOT set the CID before your call. |
| 15:32.46 | galeras | sorry, i miss "extensions reload" .... |
| 15:36.47 | galeras | Please check: http://pastebin.com/d45377c8e |
| 15:37.30 | *** join/#asterisk ChkDigit (n=mike@static24-72-71-175.regina.accesscomm.ca) |
| 15:37.39 | waverly360 | is 3139000 a valid number assigned to that pri? |
| 15:38.04 | waverly360 | galeras: you might need to put the area code in front of it as well. |
| 15:38.41 | galeras | waverly360: yes that is the number |
| 15:39.29 | *** join/#asterisk SuPrSluG (n=SuPrSluG@24.75.47.130) |
| 15:39.30 | [TK]D-Fender | waverly360: Stop thinking NANPA. |
| 15:39.36 | waverly360 | galeras: It's very possible that your carrier has something wrong on their end. You should do what [TK]D-Fender suggested and get with them. Make sure that DID is assigned to your pri. |
| 15:39.41 | waverly360 | [TK]D-Fender: NANPA? |
| 15:39.46 | [TK]D-Fender | ~nanpa |
| 15:39.47 | jbot | nanpa is probably North American Numbering Plan Administration; the organization responsible for administering the integrated telephone numbering plan serving 19 North American countries. Regulatory authorities in each participating country have plenary authority over numbering resources, but the participating countries share numbering resources cooperatively. http://www.nanpa.com/ |
| 15:41.11 | waverly360 | [TK]D-Fender: Sorry, it's just what I'm accustomed to dealing with. |
| 15:41.25 | *** join/#asterisk nikolaev (n=nikolaev@78.83.145.98) |
| 15:42.04 | [TK]D-Fender | waverly360: You should be accustomed to looking where people are and what they're using :) |
| 15:42.09 | nikolaev | Is there a way to bill the extensions registered into the asterisk ? |
| 15:42.21 | galeras | thanks guys, i will check with the telco again (BTW: same result with area code as well: http://pastebin.com/d6eb5b49) |
| 15:42.49 | [TK]D-Fender | nikolaev: Nothing that's *'s job |
| 15:43.28 | [TK]D-Fender | nikolaev: You can look at the CALLS placed on your system via CDRs of course. |
| 15:43.46 | nikolaev | oh okay |
| 15:44.07 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 15:44.16 | nikolaev | so whatever I want to do with the calls, I should look into the cdr table ? ( using DB ), correct ? |
| 15:44.52 | [TK]D-Fender | nikolaev: You should look at the CDRs by whatever way you have set them up to be recorded |
| 15:45.38 | waverly360 | galeras: Sorry. That's about as far as I can go...I'm apparently not familiar with your dialing area, so I doubt I can help much more. All I can say is that I've had experiences with some telcos where inbound calls work fine, but outbound calls don't because the DID(s) wasn't associated with the PRI, so no outbound number was deemed valid. That's just one possible situation though. |
| 15:45.48 | nikolaev | okay. Any documentations available related to this case ? |
| 15:46.15 | [TK]D-Fender | nikolaev: Yes, the "docs" folder that came with the source, and THE BOOK |
| 15:46.17 | [TK]D-Fender | ~book |
| 15:46.18 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 15:46.25 | [TK]D-Fender | nikolaev: And for everything else : |
| 15:46.27 | [TK]D-Fender | ~wikis |
| 15:46.27 | jbot | [~wikis] VoIP Wiki covering Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE. http://www.voip-info.org (c) Arte Marketing Inc / CommPartners |
| 15:46.42 | nikolaev | okay, thanks a lot |
| 15:47.33 | *** join/#asterisk wasabi (n=wasabi@ubuntu/member/wasabi) |
| 15:48.05 | wasabi | So I would really like to integrate my VoIP systems into my existing single signon infrastructure. So people can actually log into and out of their phones like they do their desktops. Is there any basis for this? |
| 15:48.29 | [TK]D-Fender | wasabi: Show me a phone that lets you "sign on" as a user period. |
| 15:48.50 | wasabi | Yeah. Hence my question. |
| 15:48.53 | wasabi | Are there any such phones? |
| 15:49.13 | Qwell | you could "disable" the phone until somebody dials some exten to "login" |
| 15:49.23 | [TK]D-Fender | wasabi: You can effectively do this with plenty of dialplan to check which device you have allocated to a "user" |
| 15:49.34 | waverly360 | [TK]D-Fender: referring to my queue question earlier...I'm using the dialplan..so you're saying when the queue app tries to call an agent, have my dial plan determine whether they're on the phone first, and then reject the call if so? |
| 15:49.58 | wasabi | Hmm. That's true. I could manipulate the phone from the desktop itself. |
| 15:49.58 | [TK]D-Fender | waverly360: Yes. "core show application chanisavail" |
| 15:50.15 | [TK]D-Fender | wasabi: "manipulate"? |
| 15:50.18 | waverly360 | [TK]D-Fender: wtf..why didn't I think of that >< |
| 15:50.25 | wasabi | Assign the user to a specific extension, in some way. |
| 15:50.28 | waverly360 | [TK]D-Fender: lemme look more into that...thanks :) |
| 15:50.45 | wasabi | But I guess I'd like roaming extensions too. |
| 15:51.06 | [TK]D-Fender | wasabi: Pretty much the same thing |
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| 16:11.16 | [TK]D-Fender | whee |
| 16:11.16 | [TK]D-Fender | ~whee |
| 16:11.16 | jbot | [~whee] Weeeeeeeee! http://www.albinoblacksheep.com/flash/weeee |
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| 16:20.29 | darkskiez | have debian taken ahold of asterisk releases :P I've just realised 1.6-beta1 was released nearly 8 month ago :) |
| 16:20.36 | *** join/#asterisk mog (n=mog@c-68-62-219-86.hsd1.al.comcast.net) |
| 16:20.36 | *** mode/#asterisk [+o mog] by ChanServ |
| 16:20.56 | Qwell | darkskiez: hold you breath. |
| 16:21.02 | Qwell | your* |
| 16:21.30 | darkskiez | Qwell: as opposed to 'dont' hold ... |
| 16:21.35 | Qwell | correct |
| 16:21.58 | darkskiez | im in no hurry, still running a 1.2 alpha release in successsful production |
| 16:22.48 | lesouvage | darkskiez: how hard is "./configure" "make" "make install". You don't need a debian package, just download the sourcecode and build Asterisk. |
| 16:23.07 | *** join/#asterisk Assid (n=assid@unaffiliated/assid) |
| 16:23.35 | darkskiez | lesouvage: hey, chillax! I'm not using debian releases |
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| 16:24.56 | bijit | ~wikis |
| 16:24.56 | jbot | [~wikis] VoIP Wiki covering Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE. http://www.voip-info.org (c) Arte Marketing Inc / CommPartners |
| 16:25.28 | lesouvage | darkskiez: Sorry, but what is your question |
| 16:25.50 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 16:26.16 | darkskiez | lesouvage: it was rhetorical, just a comment that i released that asterisk 1.6 has been in beta release for a long long time, like debian. |
| 16:26.31 | darkskiez | lesouvage: not saying its a bad thing either. |
| 16:26.53 | jpcansa | Hi, I´m using SPA3102 as my pstn trunks to dial out, how can i check in my dialplan which one is available to dial out? |
| 16:27.38 | [TK]D-Fender | jpcansa: Dial out one. If its busy, jsut dial the other right after it |
| 16:28.26 | defswork | can you pass cli through IAX2 ? |
| 16:28.31 | lesouvage | darkskiez: I will rhetoricaly stop reacting to your rhetorical comments ;-) |
| 16:28.59 | *** join/#asterisk hfb (n=hfb@pool-71-118-254-245.lsanca.dsl-w.verizon.net) |
| 16:29.02 | jpcansa | thx [TK]D-Fender, i´ll try that |
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| 16:29.29 | darkskiez | lesouvage: hah :) well, there goes my trolling career. |
| 16:32.00 | lesouvage | darkskiez: I totally chillax right now. Thanks for the advice. |
| 16:34.08 | drako | how can I limit a sip user to only be able to make 2 calls at same time. limiting sip channels. |
| 16:36.36 | [TK]D-Fender | drako: Go look at the same sip.conf |
| 16:36.40 | [TK]D-Fender | sample* |
| 16:36.50 | *** join/#asterisk ptimmins (n=paul@gw.corp.clearrate.com) |
| 16:38.06 | ptimmins | I recently upgraded from asterisk 1.4.11 to 1.4.21.2, and my connection to global crossing's sonus switches has DTMF issue if I set canreinvite=no |
| 16:38.06 | ptimmins | they use rfc2833 |
| 16:38.06 | ptimmins | any ideas? I've tried searching the bugtracker |
| 16:38.06 | bijit | anyone has configured aastra hardphone to autoconf via ftp? |
| 16:38.16 | ptimmins | it worked in 1.4.11 and it works if I set canreinvite=yes, but then reinvites happen which global doesn't support and audio path disappars |
| 16:38.29 | ptimmins | oh, and the dtmf issue is FROM the sonus, TO it works fine |
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| 16:50.00 | zippytech | whats the command to dail from cli |
| 16:51.01 | [TK]D-Fender | zippytech: type "help" and read. |
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| 16:55.51 | Emilis | anybody know where I can get a copy of x-ten pro for my ppc 2003, or know of an alternative program? |
| 16:57.42 | [TK]D-Fender | Emilis: www.xten.com |
| 16:58.30 | Emilis | tkd: they don't seem to carry x-ten pro anymore |
| 16:58.58 | jaytee | it's called Eyebeam now |
| 16:59.44 | Emilis | eyebeam doesn't support pocket pc 2003 |
| 17:02.34 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 17:02.59 | [TK]D-Fender | Emilis: They have mobile solutions listed. Go read. |
| 17:06.24 | *** part/#asterisk discHead (n=larry@wsip-70-183-82-162.sd.sd.cox.net) |
| 17:07.18 | *** join/#asterisk phl4kx (n=root@200.60.139.218) |
| 17:07.28 | phl4kx | hi all friends :) |
| 17:08.11 | *** join/#asterisk wiseoldowl (n=wiseoldo@75-128-118-24.dhcp.aldl.mi.charter.com) |
| 17:09.37 | wiseoldowl | Quick question if someone could help - I have a couple experimental trunks to Free World Dialup. On their trunks (and only theirs), I get repetitive message on the CLI that say this: |
| 17:09.44 | phl4kx | the Digium cards can use in Ubuntu 64 bits? |
| 17:09.58 | wiseoldowl | -- ast_get_srv: SRV lookup for '_sip._udp.fwd.pulver.com' mapped to host fwd.pulver.com, port 5060 |
| 17:10.24 | *** part/#asterisk oej (n=olle@ns.webway.se) |
| 17:10.26 | zippytech | does this look right Executing [s@macro-dialout-trunk:20] Dial("SIP/200-b7003580", "ZAP/4/4251054|300|") in new stack |
| 17:10.28 | wiseoldowl | Could someone explain what that message is telling me (and maybe how to get rid of it)? |
| 17:10.51 | zippytech | 4 being the trunk what the 300? |
| 17:11.08 | [TK]D-Fender | zippytech: "core show application dial" <- |
| 17:11.29 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 17:11.29 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 17:11.31 | [TK]D-Fender | wiseoldowl: tahts a DNS SRV record lookup. Its perfectly normal. Leave it alone |
| 17:11.45 | *** join/#asterisk tmccrary (n=tmccrary@68.78.185.227) |
| 17:11.58 | *** part/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 17:12.01 | tmccrary | Is it possible to get any kind of T1 card for the Asterisk Appliance products? |
| 17:12.13 | wiseoldowl | Okay. But I get the message about every ten or 15 seconds, so I thought it was a problem. Thanks. |
| 17:12.13 | [TK]D-Fender | tmccrary: No. |
| 17:12.25 | tmccrary | [TK]D-Fender: thanks |
| 17:12.45 | [TK]D-Fender | wiseoldowl: Guess they've got a really widespread HA DNS setup |
| 17:12.57 | [TK]D-Fender | tmccrary: Its a toaster. Don't expect too much. |
| 17:13.55 | wiseoldowl | Probably. Just was worried it was telling me I needed to change something in my configuration and I wasn't getting it. :-) Thanks. |
| 17:14.08 | tmccrary | [TK]D-Fender: is switchvox equipment better? |
| 17:14.39 | [TK]D-Fender | tmccrary: The full-server bundles are.... well just that. |
| 17:14.58 | waverly360 | There should be a bot here that keeps up with experience :P |
| 17:15.27 | waverly360 | [TK]D-Fender's experience goes up 230 points. [TK]D-Fender reaches level 53! |
| 17:16.18 | phl4kx | the Digium cards can use in Ubuntu 64 bits? |
| 17:16.36 | [TK]D-Fender | swings his ClueBat at a random moron for 3D20+ 5 Billion damage! |
| 17:16.47 | waverly360 | XD |
| 17:16.58 | waverly360 | Perfect. |
| 17:17.19 | *** part/#asterisk wiseoldowl (n=wiseoldo@75-128-118-24.dhcp.aldl.mi.charter.com) |
| 17:21.49 | phl4kx | the Digium cards can use in Ubuntu 64 bits? |
| 17:24.25 | [TK]D-Fender | phl4kx: unload chan_brokenrecord.so |
| 17:24.36 | klb | <PROTECTED> |
| 17:24.52 | *** join/#asterisk Turno (n=rfelder@216.12.247.171) |
| 17:25.03 | tmccrary | modprobe LOLLERSKATES |
| 17:25.20 | phl4kx | [TK]D-Fender: unload? for what? |
| 17:25.31 | Turno | is there a site where I can compare call center type software for asterisk... like... agent software for a tech support queue? |
| 17:25.31 | phl4kx | I have AEX800P |
| 17:25.36 | [TK]D-Fender | phl4kx: Means stop repeatin yourself every 5 minutes. |
| 17:25.46 | tmccrary | Turno: Asterisk has support for this built in |
| 17:26.06 | [TK]D-Fender | Turno: What do you mean by "agent software"? |
| 17:26.14 | Turno | tmccrary: but the software for an agent to login, be ready/not ready |
| 17:26.22 | Katty | [TK]D-Fender: do you like mufins? |
| 17:26.27 | Katty | [TK]D-Fender: also, muffins. |
| 17:26.41 | [TK]D-Fender | Katty: <--- Omnivore :) |
| 17:27.08 | [TK]D-Fender | Turno: You don't need any special software for that |
| 17:27.11 | tmccrary | Turno: you dial an extension to login |
| 17:27.12 | Turno | i wonder if I can help the company I work for migrate away from their current proprietary solution which uses cisco ip phones and a cisco application on the pc |
| 17:27.25 | keith4 | phl4kx: if you already have the card, why don't you try it yourself and see if it works? |
| 17:27.31 | tmccrary | but if they have cisco they're already invested like 200k into the system |
| 17:27.44 | phl4kx | keith4: I like to buy the card |
| 17:27.51 | Turno | so I can use any asterisk compatible soft phone and have a similar experience? |
| 17:27.57 | keith4 | so by "have"... you mean "don't have" |
| 17:27.58 | Katty | [TK]D-Fender: i'm glad to know you're an omnivore. do you like muffins, mister omnivore? |
| 17:28.05 | phl4kx | keith4: ok |
| 17:28.14 | Turno | tmccrary: probably a lot more than 200k |
| 17:28.16 | phl4kx | keith4: do you use a cards in 64 bits? |
| 17:28.19 | [TK]D-Fender | Katty: Kinda inclusive. "Yes". |
| 17:28.31 | keith4 | phl4kx: no |
| 17:28.32 | [TK]D-Fender | phl4kx: JFGI |
| 17:28.35 | Katty | [TK]D-Fender: do you think bits of meat would be good in a muffin? |
| 17:28.39 | keith4 | go ask digium |
| 17:28.52 | Katty | [TK]D-Fender: mister omnivore. |
| 17:29.17 | outtolunc | bets he eats that paper cup the muffin comes in also (mr omnivore <G>) |
| 17:29.23 | [TK]D-Fender | Katty: I'm somewhat particular about the mix :) |
| 17:29.50 | [TK]D-Fender | outtolunc: I should reprhase that.... I'm finding it really hard to swallow your BS ;) |
| 17:30.13 | outtolunc | drat <G> |
| 17:30.33 | Katty | [TK]D-Fender: mm, i see. k |
| 17:30.33 | [TK]D-Fender | Katty: Meat in a muffin? I supposed it could head towards "meatloaf" territory. |
| 17:30.53 | Katty | [TK]D-Fender: i was thinking more like oatbran muffin or whole wheat muffin with bits of turkey sausage, or ham or something |
| 17:31.00 | [TK]D-Fender | likes everything louder than everything else. |
| 17:31.21 | [TK]D-Fender | Katty: Difficult for me to picture. Who are you trying to kill? |
| 17:31.32 | *** join/#asterisk LND (n=Lee@89.192.154.254) |
| 17:31.36 | Katty | [TK]D-Fender: wouldn't you like to know >:) |
| 17:31.41 | phl4kx | [TK]D-Fender: JFGI?????????? |
| 17:31.47 | [TK]D-Fender | ~jfgi |
| 17:31.47 | jbot | http://www.google.com/search?q=jfgi |
| 17:32.13 | [TK]D-Fender | reaches for his ClueBat (tm) |
| 17:32.17 | phl4kx | hahaha |
| 17:32.48 | Katty | puts ClueBat(tm) into muffin as well. |
| 17:33.07 | file | tickles Katty |
| 17:33.31 | [TK]D-Fender | file: Want a muffin? *WHAM*! |
| 17:35.20 | Katty | now introducting ClueMuffin(tm) |
| 17:36.11 | [TK]D-Fender | Katty: FTW! |
| 17:36.20 | Katty | FTM. |
| 17:36.27 | [TK]D-Fender | Mmmmm |
| 17:36.49 | Katty | i was going to bash that. |
| 17:36.50 | *** join/#asterisk JenniferAkemi (n=akemi@72.60.168.132) |
| 17:37.02 | Katty | sadly. bash.org has taken a vacation |
| 17:37.03 | Nugget | heh |
| 17:37.10 | Nugget | send it to qdb.us instead |
| 17:37.37 | *** part/#asterisk tmccrary (n=tmccrary@68.78.185.227) |
| 17:41.11 | Katty | http://qdb.us/205360 |
| 17:41.45 | roxlu_ | Is there a 'solid' solution to get asterisk/sip working when I'm behind nat? |
| 17:42.02 | ptimmins | roxlu_, yes, make sure all your clients aren't. |
| 17:42.07 | [TK]D-Fender | roxlu_: Read. The. Guide. |
| 17:42.09 | [TK]D-Fender | ~sipnat |
| 17:42.10 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 17:42.14 | [TK]D-Fender | ^^^^^^^^^^ |
| 17:42.24 | Katty | i'm going to bake another batch of cluemuffins i think |
| 17:42.38 | [TK]D-Fender | Katty: High in iron! |
| 17:42.38 | roxlu_ | okay .. I was just wondering if there is solid solution... as I found some articles which arent really promising |
| 17:43.00 | [TK]D-Fender | roxlu_: Go read. |
| 17:43.13 | Katty | and eat. |
| 17:43.29 | [TK]D-Fender | Katty: Eating is entirely optional. |
| 17:43.37 | Katty | [TK]D-Fender: i dunno, a cluemuffin a day... |
| 17:43.40 | Katty | [TK]D-Fender: might keep google away. |
| 17:43.45 | *** join/#asterisk unpaidbill (i=bill@420nugs.info) |
| 17:44.33 | *** join/#asterisk undrdawg (n=steve@unaffiliated/underdawg) |
| 17:44.34 | [TK]D-Fender | kneeds up some more batter(y) |
| 17:44.59 | Katty | this conversation's getting way out of line. |
| 17:45.14 | Katty | might i suggest bigger muffin liners. |
| 17:45.14 | *** join/#asterisk korihor (n=korihor@201.211.168.130) |
| 17:45.35 | undrdawg | what does HDLC_MAINTAINERS_ARE_MORE_STUPID_THAN_I_THOUGH do? |
| 17:45.49 | Qwell | are you an HDLC maintainer? |
| 17:45.50 | undrdawg | in the file zconfig.h |
| 17:46.00 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 17:46.09 | undrdawg | heh no |
| 17:46.21 | undrdawg | i just hate silly macro names |
| 17:47.01 | Qwell | undrdawg: look in kernel/zaptel-base.c |
| 17:47.01 | Katty | [TK]D-Fender: did i mention i was going to ask the doctor for a copy of the surgery/camera recording? |
| 17:47.21 | Katty | [TK]D-Fender: im thinking of youtubing it if it doesn't violate those hipa regulations. |
| 17:47.21 | undrdawg | i knew you were going to say something like that :P |
| 17:47.47 | undrdawg | i'll check it out later |
| 17:48.31 | *** join/#asterisk lucidblue (n=lucidblu@ip72-197-81-172.sd.sd.cox.net) |
| 17:48.46 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 17:49.08 | lucidblue | hola all, I need some help troubleshooting a flash player error in the FOP, would this be an appropriate place for that? |
| 17:50.02 | Qwell | lucidblue: No it wouldn't. |
| 17:50.10 | Katty | lucidblue: fop has their own support, and they're quite helpful. |
| 17:50.20 | Katty | lucidblue: i've contacted them a few times for similiar issues. |
| 17:50.44 | Katty | lucidblue: very nice people. just shoot them an email. |
| 17:50.56 | Katty | Qwell: are you a muffin fan? |
| 17:51.22 | Qwell | Katty: I'm allergic |
| 17:51.33 | Katty | pats Qwell |
| 17:51.47 | Katty | Qwell: i'll mourn for your muffiny loss. |
| 17:51.54 | lucidblue | okay, thanks, I'll try and find their email on the website.. |
| 17:53.16 | *** part/#asterisk CanWood (n=chatzill@24.108.64.80) |
| 17:53.37 | *** join/#asterisk RobH (n=RobH@rob.tech.wikimedia.org) |
| 17:56.00 | waverly360 | [TK]D-Fender: at the risk of being swatted by the cluebat...the question I was asking earlier about queues, I'm having a difficult time figuring out how to have the queue dial a user based on some dialplan criteria..were you referring to some sort of functionality that I don't have available in 1.2? |
| 17:58.55 | [TK]D-Fender | waverly360: What are you using for member right now? |
| 18:01.00 | waverly360 | [TK]D-Fender: to add agents to the queue? or how am I sending a caller to the queue? I'm just using the queue function for that...I guess I'm not entirely sure what you're asking. |
| 18:01.47 | [TK]D-Fender | waverly360: QUEUES have MEMBERS. What are you using for MEMBERS? |
| 18:02.43 | *** join/#asterisk kaldemar (n=kalde@vipunen.hut.fi) |
| 18:02.53 | waverly360 | AddQueueMember and RemoveQueueMember |
| 18:03.11 | waverly360 | [TK]D-Fender: those are the two functions that I'm using to add members (which I've been referring to as agents) |
| 18:03.21 | *** join/#asterisk jnials (n=jnials@cuervo.unwiredbuyer.com) |
| 18:04.17 | [TK]D-Fender | waverly360: Ok, then what are you adding as devices since those aren't static members? |
| 18:04.23 | *** part/#asterisk Turno (n=rfelder@216.12.247.171) |
| 18:05.44 | *** part/#asterisk JenniferAkemi (n=akemi@72.60.168.132) |
| 18:06.39 | waverly360 | [TK]D-Fender: Sorry, you lost me again. |
| 18:07.00 | [TK]D-Fender | waverly360: ..... do you seriously have any clue about these dialplan applications you're using? |
| 18:07.23 | [TK]D-Fender | waverly360: You provide them a DEVICE to dial. What is it? |
| 18:07.37 | waverly360 | [TK]D-Fender: Well I obviously have something working here, but the terminology you're using doesn't always coincide with mine. |
| 18:07.50 | outtolunc | he means interface |
| 18:07.51 | waverly360 | [TK]D-Fender: I provide them a sip/num |
| 18:08.05 | [TK]D-Fender | waverly360: Good. SIP/123 is a DEVICE |
| 18:08.15 | waverly360 | [TK]D-Fender: Well I'm sorry. |
| 18:08.32 | [TK]D-Fender | waverly360: In 1.4 minimum I know there is a way to limit hitting agents that may be on other calls. I don't believe thats available for 1.2 |
| 18:08.59 | waverly360 | [TK]D-Fender: It looks like there some sort of AGI addition in 1.4+ |
| 18:09.04 | [TK]D-Fender | waverly360: to do the dialplan-based limit you'd have to point to a local channel whose exten would perform the check. |
| 18:09.11 | [TK]D-Fender | waverly360: No need for AGI |
| 18:09.20 | [TK]D-Fender | waverly360: And AGI has nothing to do with this. |
| 18:09.39 | waverly360 | [TK]D-Fender: My dialplan is controlled almost entirely by AGI ...that's the only reason I was mentioning it. |
| 18:10.16 | waverly360 | [TK]D-Fender: but that could work...change the way I'm defining a member..instead of using a sip device, could use a local one like you mentioned..that would give me the capability I think... |
| 18:10.27 | [TK]D-Fender | waverly360: not quite, but close enough for you to consider it so. |
| 18:10.45 | [TK]D-Fender | waverly360: You shold seriously get off of 1.2 |
| 18:11.09 | waverly360 | [TK]D-Fender: I seriously want to... |
| 18:11.42 | waverly360 | [TK]D-Fender: I'm going to have to start stabbing sales guys... |
| 18:12.16 | waverly360 | [TK]D-Fender: it's like trying to build a spaceship with a steam engine. |
| 18:12.27 | *** join/#asterisk CrashSys (n=kumba@216-199-37-76.tpa.fdn.com) |
| 18:12.36 | waverly360 | [TK]D-Fender: but I do appreciate the help...thank you. |
| 18:12.54 | Qwell | sales guys produce a lot of steam |
| 18:12.57 | Qwell | throw more into the engine |
| 18:13.08 | waverly360 | Couldn't I just burn them instead? |
| 18:13.11 | [TK]D-Fender | waverly360: Go use local channels for the moment. |
| 18:13.22 | [TK]D-Fender | waverly360: whats what he said |
| 18:13.25 | CrashSys | Is it possible to use voicemailmain but pass a flag that doesn't allow the user to over-write the voicemail greetings? I have a customer that uses a VM box as a general company box and they dont want people to over-write the greeting... |
| 18:14.01 | Nugget | chmod 444 the raw recording files in /var |
| 18:14.07 | Nugget | it's ugly but it'll work |
| 18:14.11 | Qwell | but then nobody can change them |
| 18:14.11 | mchou | haha, that would be funny |
| 18:14.33 | mchou | I can picture disgruntled employees now |
| 18:14.34 | Qwell | (and, if Asterisk is running as root, it won't matter) |
| 18:14.47 | Nugget | ah, good point |
| 18:15.01 | waverly360 | There are other ways to set permissions on a file to prevent even root from modifying them |
| 18:15.06 | waverly360 | look for the immutable flag |
| 18:15.10 | *** join/#asterisk jpcansa (n=jpbenavi@201.198.58.30) |
| 18:15.15 | CrashSys | well I have asterisk running as asterisk, but they would have to call me everytime they wanted to change the greetings |
| 18:15.24 | [TK]D-Fender | CrashSys: symlink them to a read-only mounted FS <- |
| 18:15.39 | waverly360 | That's even better...still leaves the initial problem. |
| 18:15.48 | [TK]D-Fender | CrashSys: First you don't want them to change it. Then you DO. MAke up your mind. |
| 18:16.08 | waverly360 | *smacks himself* |
| 18:16.22 | waverly360 | gives [TK]D-Fender a cookie. |
| 18:16.35 | [TK]D-Fender | reaches for his ClueBat (tm) again |
| 18:16.41 | CrashSys | The ideal scenario is they dial the general VM box extension to check messages but can not change the greeting... then for administrative purposes they dial a different extension, enter a pin, and then can change it... |
| 18:16.48 | *** join/#asterisk lanning (n=lanning@66.151.128.195) |
| 18:16.48 | waverly360 | XD |
| 18:17.28 | CrashSys | I can spend some time using a cron script that runs every 5 minutes, looks for a file, if it exists it copies it over the old greeting... that would work... |
| 18:17.42 | waverly360 | But still.. |
| 18:17.46 | CrashSys | I was just wondering if there was something simple like a flag to pass to the voicemailmain cmd |
| 18:17.56 | waverly360 | if it's changed by someone authorized it'll overwrite that too.. |
| 18:18.13 | keith4 | wow |
| 18:18.29 | CrashSys | That's fine. If you give someone the pin # for the administrative extension then that's their problem... |
| 18:18.59 | waverly360 | wait... |
| 18:19.13 | waverly360 | so how will normal users check the voicemail? |
| 18:19.27 | CrashSys | dialing extension 100 = voicemailmain(s100) |
| 18:19.41 | CrashSys | dialing extension 101 = administrative stuff to change voicemail greeting |
| 18:19.56 | CrashSys | Just for example |
| 18:20.40 | CrashSys | So dialing 100 puts you right into the VM box... dialing 101 would ask for a PIN #, then give you the option to change the VM greetings... |
| 18:20.48 | CrashSys | guess i'll just do it that way |
| 18:21.03 | waverly360 | that's what you want..but you don't have a way to accomplish that yet do you? I've not used it..but won't using the s option just skip the voicemail password and give them full access anyways? |
| 18:21.03 | klb | personal identification number number? |
| 18:21.09 | waverly360 | that's what the documentation seems to suggest. |
| 18:21.10 | CrashSys | Yup |
| 18:21.52 | [TK]D-Fender | .... |
| 18:22.07 | CrashSys | Waverly: if you send someone to voicemailmain(s100) they can freely change the voicemail greeting unless you change the greeting file to read-only and have asterisk run as non-root |
| 18:22.34 | CrashSys | See, we dont want employee's recording "Fuck Off" on the main company catch-all voicemailbox... |
| 18:23.10 | keith4 | then you could have another extension that authenticates, changes the files to read/write, and then sends to voicemailmain. and changes them back to read-only afterwards |
| 18:23.32 | keith4 | might require some AGI |
| 18:23.35 | CrashSys | keith: No, hard to do that as non-privileged user... |
| 18:23.51 | waverly360 | I understand that..I think I misunderstood your intentions. So you're going to create two extensions...and two voicemailboxes |
| 18:24.03 | CrashSys | Nevermind |
| 18:24.10 | keith4 | CrashSys: it's not hard to do |
| 18:24.11 | waverly360 | and have extension 101's voicemail greeting auto copied over to 100 every 5 minutes |
| 18:24.34 | waverly360 | so that if you change it on 101, it will change on 100..and just hide the fact that 101 exists? |
| 18:24.50 | CrashSys | I was just going to record a file called "100-greet.wav" and have a cron script copy the file |
| 18:25.04 | keith4 | i like my idea better |
| 18:25.30 | waverly360 | keith4: Your idea is more solid, just a bit more to implement. |
| 18:25.36 | [TK]D-Fender | CrashSys: Thats viable. |
| 18:25.42 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 18:25.44 | *** join/#asterisk pikachu2000 (n=pikachu2@196-209-198-179-rrdg-esr-2.dynamic.isadsl.co.za) |
| 18:26.25 | keith4 | you could also not give anyone the voicemail pin, and have all voicemail sent out via email instead |
| 18:26.36 | waverly360 | The only problem is that he'll have to re-record a new greeting and copy it by hand anytime it needs to be changed. |
| 18:26.39 | keith4 | don't let the plebes into voicemailmain. just give the pin to the admins |
| 18:27.24 | CrashSys | They dont want to check e-mail. They want to use the phone. |
| 18:27.48 | Alan_Hicks | Howdy. Some one just hit me with an interesting question that I have no idea how to asnwer, so I'm here to ask the gurus. :^) I've been asked to give a quote on a small asterisk system for a truck lot. Mostly it's just standard stuff, but they need external paging. |
| 18:27.50 | waverly360 | Well...sounds like about three good ideas there. |
| 18:27.58 | Qwell | Voicemail(1111&1112) |
| 18:28.02 | Qwell | give newbs the pin to 1112 |
| 18:28.11 | Qwell | problem solved |
| 18:28.45 | Alan_Hicks | Basically, they have purchased and installed speakers around their lot and want to hook these up so they can press a key or dial an extension and page sales personel out on the lot. Any ideas what might be needed to do this, or where I can find more information? |
| 18:28.47 | waverly360 | Excuse me...4 good ideas. I like that one too |
| 18:28.50 | Qwell | OR, don't play a prompt in Voicemail() and just use a Playback beforehand |
| 18:28.54 | [TK]D-Fender | Qwell: Except for the incremental waste of disk space. |
| 18:29.00 | Qwell | [TK]D-Fender: hardlink |
| 18:29.05 | [TK]D-Fender | Qwell: Which I suppose there are ways to shunt |
| 18:29.20 | keith4 | Alan_Hicks: you just need a sound card and a big-ass amp |
| 18:29.22 | CrashSys | Qwell: Yeah, i'm looking at that now... seems like it's the least invasive option |
| 18:29.27 | waverly360 | Alan_Hicks: there are several paging systems that you can connect to an FXO or FXS port on an analog card. |
| 18:29.28 | *** join/#asterisk linuxstb (n=linuxstb@rockbox/developer/linuxstb) |
| 18:29.33 | CrashSys | requires the least amount of set-up and head-ache |
| 18:29.40 | waverly360 | Alan_Hicks: or the soundcard types too... :P |
| 18:29.44 | Qwell | the latter is probably the best |
| 18:29.53 | Alan_Hicks | Hmm.... I hadn't thought about the sound card idea. |
| 18:30.11 | Alan_Hicks | Is there documentation online for piping to a sound card? /me googles. |
| 18:30.20 | waverly360 | Alan_Hicks: yep. |
| 18:30.21 | Qwell | Alan_Hicks: chan_alsa |
| 18:30.36 | Alan_Hicks | Thanks guys. |
| 18:31.05 | rob0 | Alan_Hicks? |
| 18:31.12 | rob0 | I've heard of him. |
| 18:31.13 | Alan_Hicks | rob0: The one and only. |
| 18:31.48 | Alan_Hicks | rob0: What're you doin' here? |
| 18:32.03 | rob0 | mostly just lurking |
| 18:33.21 | Alan_Hicks | Anyone care to recommend any of those paging systems that connect to FXO or FXS ports in case I decide to go that route? |
| 18:34.02 | [TK]D-Fender | Alan_Hicks: Go lookup Viking paging system |
| 18:34.07 | Alan_Hicks | Thanks. |
| 18:35.20 | waverly360 | Crazy asterisk people..... >.> |
| 18:36.28 | *** part/#asterisk galeras (n=galeras@201.245.54.165) |
| 18:38.01 | jasonwoot | indeed |
| 18:40.12 | Kobaz | so i'm working on a thing to simulate paging... my idea is to originate a bunch of calls (using auto answer) direct into a meetme |
| 18:40.48 | Kobaz | from the dialplan, what's a good way to start spawning up new calls... i don't see how you can do it with a regular dial |
| 18:41.45 | [TK]D-Fender | Kobaz: : "core show application page" |
| 18:41.49 | Kobaz | oh |
| 18:41.54 | Kobaz | i must be blind |
| 18:42.06 | Kobaz | haha |
| 18:42.12 | Kobaz | wow, thats exactly what i need |
| 18:47.56 | LemensTS | b2bua have the rtp traffic flow thru them, not just the sip signaling. Is this correct? |
| 18:48.09 | [TK]D-Fender | LemensTS: not necessarliy |
| 18:49.06 | LemensTS | Only when it is transcoding a codec? |
| 18:49.36 | [TK]D-Fender | LemensTS: No. |
| 18:52.13 | *** join/#asterisk kuto (n=kuto@75.152.131.174) |
| 18:54.07 | *** join/#asterisk axisys (n=axisys@155.70.141.45) |
| 18:58.25 | paige | hwo do i install the php aig for 1.6.0-beta9? |
| 18:58.28 | paige | agi |
| 18:59.53 | [TK]D-Fender | paige: No such thing as "install AGI". |
| 19:00.05 | [TK]D-Fender | paige: PHP-AGI is its own project on sourceforge. |
| 19:00.10 | paige | ok |
| 19:00.12 | [TK]D-Fender | paige: Go download it there |
| 19:00.19 | paige | thank you |
| 19:00.58 | rob0 | Download Moses ... waaaaay down, in Egypt land, tell, ol' Pharoah ... |
| 19:01.28 | kuto | hi, i have a new internet with only 1 static ip, this includes only internet, my question is it possible that i install asterisk with it? |
| 19:01.44 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 19:01.44 | paige | looks like sourceforge is borked |
| 19:02.44 | [TK]D-Fender | kuto: Yes |
| 19:03.36 | kuto | what happen to asterisk if i have 5 concurrent user using internet then? |
| 19:03.49 | kuto | im using telus |
| 19:04.27 | [TK]D-Fender | kuto: Go lookup "codec bandwidth" on the WIKI |
| 19:04.29 | [TK]D-Fender | ~wikis |
| 19:04.30 | jbot | [~wikis] VoIP Wiki covering Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE. http://www.voip-info.org (c) Arte Marketing Inc / CommPartners |
| 19:05.32 | *** join/#asterisk stencil (n=stencil@206-248-163-157.dsl.teksavvy.com) |
| 19:05.52 | [TK]D-Fender | paige: Look just fine to me |
| 19:06.06 | *** join/#asterisk astassistant (n=skip@65-126-63-1.dia.static.qwest.net) |
| 19:10.51 | paige | there it goes |
| 19:11.49 | *** join/#asterisk vgster (n=vgster@93-96-221-240.zone4.bethere.co.uk) |
| 19:12.37 | km- | surprised at how easy that was; I just wrote a quick lil .NET app to change my msn messenger status when I'm on the phone, uses asterisk manager interface and msn's com components |
| 19:12.49 | *** join/#asterisk vgster (n=vgster@93-96-221-240.zone4.bethere.co.uk) |
| 19:14.51 | *** join/#asterisk implicit (n=bayan@unaffiliated/implicit) |
| 19:15.00 | jasonwoot | you're hired |
| 19:15.13 | jasonwoot | I've never written a quick lil anything |
| 19:15.37 | Alan_Hicks | has written a quick little rubber check. |
| 19:15.59 | jasonwoot | hopes it wasn't to him |
| 19:16.53 | jasonwoot | I got terrible grades in C++ and XHTML because i don't indent my code what so ever... it's all in a nice, straight line |
| 19:21.41 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 19:23.22 | *** join/#asterisk SwK (n=SwK@freeswitch/developer/swk) |
| 19:27.45 | *** join/#asterisk Blackvel (n=blackvel@dslb-088-065-099-173.pools.arcor-ip.net) |
| 19:29.22 | *** join/#asterisk Assid (n=assid@unaffiliated/assid) |
| 19:31.59 | Blackvel | who uses asterisk call forwarding to mobile phone over a sip provider? I am interested to set the special redirect flag in sip so my mobile phone writes in display "call forwarded on number ..." like it does with real call forwarding by telco. I hope and believe that there is some special flag which sip providers might use to set also for their telco switch? |
| 19:32.30 | Blackvel | i do not have any issue with clip no screening. it works with the provider |
| 19:33.03 | *** join/#asterisk justdave (n=dave@unaffiliated/justdave) |
| 19:35.04 | stencil | Hello, Would there be a easy way to test what Caller ID people are receiving when I phone them? |
| 19:35.33 | [TK]D-Fender | stencil: What are you calling out via? |
| 19:35.57 | stencil | snom 300 through Asterisk |
| 19:36.20 | stencil | [TK]D-Fender: you mean itsp? |
| 19:36.36 | paige | does anyone here who uses phpagi? |
| 19:36.48 | [TK]D-Fender | stencil: I mean how are you getting to the PSTN obviously |
| 19:37.11 | stencil | sip through les.net |
| 19:37.18 | [TK]D-Fender | paige: You shouldn't ask leading questions. Just go and ask |
| 19:37.33 | [TK]D-Fender | stencil: And are you setting the CID before you call out? |
| 19:37.48 | paige | [TK]D-Fender, sorry, beena long day and it is only 12:37 |
| 19:37.50 | stencil | yes |
| 19:38.10 | [TK]D-Fender | stencil: Then they should receive the #, if not the name. Name is not normally configurable |
| 19:38.10 | paige | i am trying to figure out how to install it to show me active calls |
| 19:38.16 | *** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk) |
| 19:38.38 | stencil | ok thanks [TK]D-Fender |
| 19:39.11 | keith4 | stencil: uh... call someone and ask "what CID came up?" |
| 19:39.16 | [TK]D-Fender | paige: You don't "install it". Its just a library to issue a few petty in/out calls for the basics. And AGI can't provide that data. AMI can. |
| 19:39.16 | keith4 | or call your own cell phone |
| 19:39.33 | waverly360 | [TK]D-Fender: that totally worked |
| 19:39.47 | waverly360 | [TK]D-Fender: the chanisavail command with the queues I mean. |
| 19:39.48 | [TK]D-Fender | waverly360: Local channel? |
| 19:40.01 | [TK]D-Fender | waverly360: Yes, I've done it myself many years back |
| 19:40.21 | waverly360 | [TK]D-Fender: Yep. I lose my custom ring tones, so I need to figure that out, but still..is a step in the right direction. Many thanks. |
| 19:40.25 | *** join/#asterisk SwK (n=SwK@freeswitch/developer/swk) |
| 19:40.30 | Assid | okay so i finally got asterisk working with zaptel in a openvz container |
| 19:40.36 | Assid | now i gotta decide on the fate of the database |
| 19:41.03 | stencil | I would really like to thank all the developers for all their hard work, Asterisk is just the most fun program I have ever dealt with! |
| 19:41.37 | *** join/#asterisk rabelais (n=blank@unaffiliated/rabelais) |
| 19:41.59 | *** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk) |
| 19:42.54 | *** part/#asterisk SwK (n=SwK@freeswitch/developer/swk) |
| 19:43.37 | *** join/#asterisk Blackvel (n=blackvel@dslb-088-065-099-173.pools.arcor-ip.net) |
| 19:43.45 | waverly360 | [TK]D-Fender: ok...mind if I pick your brain a little bit? |
| 19:44.14 | waverly360 | [TK]D-Fender: actually...nevermind..I need to look into something first. |
| 19:44.38 | *** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee) |
| 19:45.00 | [TK]D-Fender | k |
| 19:47.35 | paige | ls |
| 19:47.44 | *** join/#asterisk hi365_m (n=hi365@213.151.56.70) |
| 19:48.46 | *** join/#asterisk datachomper (n=russ@h-66-134-255-227.nycmny83.covad.net) |
| 19:49.30 | Blackvel | there there some special call forwarding flag to indicate redirection on mobile phones? :) |
| 19:51.01 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 19:52.50 | *** join/#asterisk dexthageek (n=root@68.236.214.95) |
| 19:53.09 | phl4kx | the Digium cards can use in Ubuntu 64 bits? |
| 19:53.30 | klb | heh |
| 19:53.40 | phl4kx | klb: do you use |
| 19:53.40 | phl4kx | ? |
| 19:54.03 | [TK]D-Fender | phl4kx: JFGI <- Or FFS just call Digium sales. |
| 19:54.38 | dexthageek | I am using phpagi. Is it possible to play music (background,musiconhold) while waiting for response from database, or another script? |
| 19:55.20 | [TK]D-Fender | dexthageek: I believe there is a stream command that will queue up sounds to play while you do other things |
| 19:56.15 | dexthageek | i have tried stream_file but the ivr will not continue until the stream has finished |
| 19:58.08 | keith4 | so that's probably not the stream command he means |
| 19:58.25 | [TK]D-Fender | dexthageek: it has an "escape digits" parameter you can set to let it exit on DTMF |
| 19:58.44 | *** join/#asterisk hubguruJR (n=hubguruJ@mail.ntegratedsolutions.com) |
| 19:59.27 | dexthageek | yea i am using that for other things. |
| 19:59.59 | dexthageek | here is an example. I need to retrieve data from a database. I would like to play music while i wait for the database to return the rowset |
| 20:00.11 | [TK]D-Fender | dexthageek: use it for this then. you want ti to be able to exit, there you go. |
| 20:01.01 | [TK]D-Fender | dexthageek: Sorry, I missed a detail there... my bad |
| 20:01.05 | [TK]D-Fender | dexthageek: Looking... |
| 20:01.24 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 20:02.32 | [TK]D-Fender | dexthageek: Indeed I do not see a way |
| 20:03.20 | seanbright | can't set MOH right before doing other stuff? |
| 20:04.09 | [TK]D-Fender | seanbright: No way to terminate. |
| 20:04.12 | dexthageek | but it waits until the song is finished befre it moves on |
| 20:04.19 | dexthageek | and i can't terminate it correct |
| 20:04.28 | seanbright | moh blocks? |
| 20:04.37 | [TK]D-Fender | dexthageek: Only option I see is to write a full-on application. |
| 20:04.48 | dexthageek | well thast what I am doing using php |
| 20:04.59 | seanbright | "full-on application" = c app |
| 20:05.03 | [TK]D-Fender | dexthageek: No, I mean a COMPILED full-on * applicaiton. |
| 20:05.13 | [TK]D-Fender | dexthageek: not AGI |
| 20:05.36 | seanbright | back up a second |
| 20:06.04 | seanbright | i'm confused as to why setting music on hold (from agi 'set music') and then doing something else wouldn't work? |
| 20:06.05 | *** join/#asterisk tc3driver (n=tc3drive@adsl-75-49-241-185.dsl.irvnca.sbcglobal.net) |
| 20:06.27 | Juggie | huh? |
| 20:06.38 | seanbright | ohhh |
| 20:06.42 | seanbright | because they aren't on hold |
| 20:07.08 | Juggie | shoudn't set music just set the music for the next time they go on hold, if they go on hold? |
| 20:07.17 | seanbright | yes, see my last message |
| 20:07.37 | seanbright | Juggie: look up from the keyboard before hitting enter ;) |
| 20:07.46 | datachomper | Can you put a channel on hold from inside the AGI and do other stuff, then recover that channel? |
| 20:08.07 | [TK]D-Fender | datachomper: No, AGI AI your channel. |
| 20:08.09 | [TK]D-Fender | IS* |
| 20:09.33 | Juggie | seanbright, i havn't looked at my keyboard to type since i was like 15 :) |
| 20:09.40 | Juggie | i just dont read :> |
| 20:09.56 | seanbright | Juggie: you probably haven't looked at your keyboard to type since you were like 15 |
| 20:10.00 | seanbright | woops |
| 20:11.10 | Juggie | anyway, i cant think of a solution to his problem off the top of my head. |
| 20:11.29 | Juggie | lets say i need to do an operation in agi which can take super long (lets say a mainframe operation which can take 5-10 seconds) |
| 20:11.42 | seanbright | Juggie: can you think of a solution to this problem off the top of your head? |
| 20:11.55 | *** join/#asterisk codebanshee (n=chris@194.164.236.240) |
| 20:11.55 | Juggie | no :) |
| 20:12.19 | Juggie | how would you enable moh, preform the operation, then disable.. is that possible w/ agi? |
| 20:12.29 | seanbright | based on the code i'm looking at... yes |
| 20:12.33 | seanbright | you should be able to: |
| 20:12.38 | seanbright | SET MUSIC ON |
| 20:12.47 | seanbright | ... insert processing magic here ... |
| 20:12.49 | seanbright | SET MUSIC OFF |
| 20:13.05 | russellb | what happens if you don't provide an AGI command after some period of time? |
| 20:13.08 | russellb | will it hang up? |
| 20:13.12 | *** part/#asterisk tc3driver (n=tc3drive@adsl-75-49-241-185.dsl.irvnca.sbcglobal.net) |
| 20:13.21 | russellb | i.e., what's wrong with exec startmusiconhold |
| 20:13.25 | Juggie | oh ok, i thought set music just told asterisk what music to use should the call go on hold. |
| 20:13.28 | russellb | and then going off and doing stuff |
| 20:13.31 | seanbright | right |
| 20:13.37 | seanbright | SET MUSIC ON is the same thing |
| 20:13.42 | seanbright | dexthageek: ^^^ |
| 20:14.05 | seanbright | dexthageek: SET MUSIC ON YourMusicOnHoldClassHere |
| 20:14.06 | [TK]D-Fender | seanbright: Of course at best you'd have to set up an entire class per file you'd want to treat this way. |
| 20:14.20 | *** part/#asterisk codebanshee (n=chris@194.164.236.240) |
| 20:14.25 | Juggie | russellb, i dont think agi has an idle timeout. |
| 20:14.28 | seanbright | [TK]D-Fender: correct, but it's a hell of a lot better than building a new asterisk app |
| 20:14.56 | seanbright | from a level-of-effot standpoint, anyway. |
| 20:15.01 | seanbright | s/effot/effort/ |
| 20:17.20 | *** join/#asterisk grandpapadot (n=anonymou@mail.heavylogic.com) |
| 20:17.48 | seanbright | pretends to go off and test it... |
| 20:17.57 | seanbright | yup! works like a charm! |
| 20:18.09 | grandpapadot | Hi all. Is there some trick to getting externnotify and externpass to work with 1.4's voicemail.conf? Is there some pre-requisite? When I have my debug level set, it doesn't even show it trying to fire for the events. |
| 20:18.37 | seanbright | grandpapadot: pastebin your voicemail.conf |
| 20:18.39 | seanbright | ~pb |
| 20:18.40 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 20:20.18 | grandpapadot | seanbright: http://pastebin.com/m6c0ea500 |
| 20:20.44 | seanbright | interesting |
| 20:20.49 | *** join/#asterisk Superbartt (n=bart@ipd50a21c9.speed.planet.nl) |
| 20:21.22 | seanbright | i have to be honest with you... from a system management perspective |
| 20:21.37 | seanbright | you should probably not name paths "path/to/my" and scripts as "script.php" |
| 20:21.50 | seanbright | i'm guessing over time it would make it difficult to determine what those things are |
| 20:21.51 | grandpapadot | lol, I replaced the real path since it's an anon pastebin ... |
| 20:21.57 | seanbright | grandpapadot: ohhhh |
| 20:22.19 | seanbright | looks back to see where he asked for a censored copy of grandpapadot's voicemail.conf... |
| 20:22.20 | grandpapadot | I'm thinking the space is throwing it off: /usr/bin/php /path/to/my/script.php |
| 20:22.37 | seanbright | grandpapadot: just use the shebang in the script |
| 20:22.44 | seanbright | #!/usr/bin/php |
| 20:22.47 | grandpapadot | Yea, that's where I was going. |
| 20:22.51 | grandpapadot | .. next... |
| 20:22.56 | seanbright | super |
| 20:22.58 | seanbright | we'll wait here |
| 20:22.59 | Qwell | #!$@ |
| 20:23.00 | dexthageek | seanbright: im gonna try the set music |
| 20:23.03 | dexthageek | i will let you know |
| 20:23.04 | Qwell | wonders what would happen |
| 20:23.08 | Qwell | tries it |
| 20:23.08 | dexthageek | thanks |
| 20:23.14 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.static.bezeqint.net) |
| 20:23.30 | seanbright | Qwell: an angel will lose it's wings |
| 20:23.47 | Qwell | lame, it don't work |
| 20:23.57 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.static.bezeqint.net) |
| 20:23.59 | seanbright | Qwell: silliness. |
| 20:24.21 | Superbartt | Hmmfg, I seem to be haveing a strange problem with my queues and voicemail. I want to use queues to do a ringall on some phones, but if it hasn't been picked up after 45 seconds it should go to the voicemail. Now it goes to voicemail after about 1:10 minute on my phone... The relevant configuration is at: http://paste.barthezz.name/?show=305 |
| 20:24.58 | seanbright | Superbartt: where is the '45' part in any of that? |
| 20:25.09 | grandpapadot | Nothing .. It's like it's just ignoreing externnotify |
| 20:25.15 | Superbartt | uhmm, i replace the 45 by 20 because it wasn't doing what i want seanbright |
| 20:25.40 | Superbartt | but with the current config (20 in the extensions) it still takes 1minute 10 before it goes to the voicemail |
| 20:26.32 | waverly360 | Superbartt: might need to hard-set the timeout option as well as a few others. I'm not sure what the defaults are for a queue because I always set them, but they may have large defaults that are causing your queue to hold onto the call a bit longer. |
| 20:26.58 | Superbartt | so in the queues.conf also set the timeout to 20 (45 eventually)? |
| 20:27.13 | seanbright | grandpapadot: strange. |
| 20:27.27 | waverly360 | Superbartt: not necessarily. It's sometimes trick to get the numbers correct..hang on..lemme refresh my memory. |
| 20:27.42 | grandpapadot | Yep |
| 20:28.20 | *** join/#asterisk SamuraiDio (n=diovani@201.41.41.235) |
| 20:28.22 | SamuraiDio | hi |
| 20:28.23 | waverly360 | Superbartt: try setting timeout to something like 15 seconds. |
| 20:28.28 | seanbright | Superbartt: if we call the timeout option in queues.conf T1 and the timeout argument in the Queue app T2, the queue app will check every T1 seconds to see if T2 has elapsed. |
| 20:28.41 | seanbright | (in a nutshell) |
| 20:28.53 | putnopvut | seanbright: well....not if you get a recent checkout of 1.4 |
| 20:28.53 | waverly360 | seanbright: much better than I would have put it :) |
| 20:29.03 | seanbright | putnopvut: well then speak up |
| 20:29.06 | putnopvut | by recent, I mean like from last week. |
| 20:29.23 | [TK]D-Fender | Checkout time... heading home. Later all |
| 20:29.24 | SamuraiDio | i'm tryng to authenticate a sip server to my asterisk server just by its ip address, but when calling i always receive a 407 error. |
| 20:29.26 | Superbartt | so seanbright, for example i need to set the timeout in queues to 5, and in extensions to 45, so that the queue app wil check every 5 seconds if the timeout has exceeded? |
| 20:29.29 | putnopvut | The T2 timeout is checked much more often so that the app-specified timeout may occur at just about any point. |
| 20:29.32 | waverly360 | [TK]D-Fender: night |
| 20:29.33 | SamuraiDio | what should i do to accept that call? |
| 20:29.51 | seanbright | Superbartt: make the one in queues.conf 15 |
| 20:30.00 | Superbartt | ok, gimme a sec to test :) |
| 20:30.04 | seanbright | Superbartt: make the one in the call to Queue() be 45 |
| 20:30.33 | Superbartt | yes :) |
| 20:30.41 | waverly360 | Superbartt: awesome |
| 20:30.41 | seanbright | putnopvut: and that change is in 1.4 now? |
| 20:30.48 | putnopvut | seanbright: correct. |
| 20:30.48 | Superbartt | calling :p |
| 20:30.52 | putnopvut | I can look up the svn rev... |
| 20:30.59 | Superbartt | voip01*CLI> show version |
| 20:30.59 | Superbartt | Asterisk 1.2.29-BRIstuffed-0.3.0-PRE-1y-s built by root @ voip01 on a i686 running Linux on 2008-06-25 14:04:13 UTC |
| 20:31.00 | Superbartt | btw |
| 20:31.02 | seanbright | putnopvut: option to make it work the 'old' way? |
| 20:31.08 | seanbright | Superbartt: YIKES |
| 20:31.17 | putnopvut | seanbright: what do you mean? |
| 20:31.25 | seanbright | putnopvut: that's a behavior change, no? |
| 20:31.30 | putnopvut | seanbright: it was a bug fix. |
| 20:31.43 | Superbartt | did the trick seanbright :D thanks :) |
| 20:31.48 | seanbright | Superbartt: no sweat. |
| 20:31.55 | seanbright | putnopvut: hmm. k. |
| 20:31.59 | putnopvut | It was because the queue wouldn't time out when it was supposed to, and if a periodic announcement caused the time to exceed the timeout, then you'd end up ringing a queue member forever. |
| 20:32.01 | Superbartt | yes have been staring blind on this and couldn't find anything usefull :x |
| 20:32.10 | Superbartt | yes->just* |
| 20:32.16 | putnopvut | I think that last sentence of mine probably made a lot less sense than it could have. |
| 20:32.20 | seanbright | putnopvut: yeah. |
| 20:32.31 | waverly360 | later on guys..heading home myself. |
| 20:32.34 | seanbright | putnopvut: just seems like a behavior change that could (possibly) bite someone |
| 20:32.34 | putnopvut | seanbright: let me find the bug number... |
| 20:32.46 | *** part/#asterisk waverly360 (n=waverly@ns2.dalcon.com) |
| 20:32.53 | seanbright | putnopvut: someone who was maybe using the timeout values incorrectly |
| 20:33.00 | putnopvut | http://bugs.digium.com/view.php?id=13186 |
| 20:33.00 | seanbright | someone like me. |
| 20:33.01 | seanbright | heh |
| 20:33.29 | putnopvut | seanbright: I'm not sure how making it work as it's documented is a behavior change that could bite someone :) |
| 20:33.52 | seanbright | putnopvut: really? |
| 20:34.07 | keith4 | because we've all been using it how it works, not how it's documented |
| 20:34.13 | seanbright | ^^^ |
| 20:34.56 | *** join/#asterisk galeras (n=galeras@201.245.54.165) |
| 20:34.57 | putnopvut | Okay, so can you tell me of a situation where this new change could actually cause a problem? |
| 20:35.06 | seanbright | putnopvut: of course not |
| 20:35.19 | putnopvut | lol |
| 20:35.50 | putnopvut | It's a case where the bug in question made it very clear that the app's timeout was implemented very half-assedly. |
| 20:35.56 | *** join/#asterisk Shotygun (n=thorn@82.166.64.119) |
| 20:36.36 | keith4 | true |
| 20:36.39 | seanbright | putnopvut: timeout=45 and Queue(...|60) |
| 20:36.49 | seanbright | putnopvut: old way, we time out in 90 seconds |
| 20:36.56 | seanbright | putnopvut: new way, we time out in 60? |
| 20:37.00 | putnopvut | not exactly. |
| 20:37.09 | putnopvut | Both will time out in 60. |
| 20:37.14 | seanbright | oh |
| 20:37.16 | putnopvut | Here's what was changed with the bugfix. |
| 20:37.24 | seanbright | well then i'm talking out of my ass... again |
| 20:37.27 | putnopvut | Let's say you have that situation you just put before me. |
| 20:37.29 | dexthageek | set_music on/off worked, however music did not play |
| 20:37.31 | keith4 | so, it was more like "how-often-to-check-for-timeout", before? |
| 20:37.48 | seanbright | dexthageek: if the music didn't play, how did it "work"? |
| 20:37.54 | putnopvut | But let's say you also had a periodic announcement that would last about 20 seconds. |
| 20:38.45 | putnopvut | Then you'd call a queue member for 45 seconds, then the periodic annoucement would play for about 20 seconds. |
| 20:38.46 | putnopvut | The timeout check occurred between the attempt to call the member and the periodic announcement playback. |
| 20:38.51 | dexthageek | well asterisk cli said start music on hold. It allowed me to process when that was finished. Music on hold was disabled. |
| 20:39.00 | dexthageek | but music did not play |
| 20:39.09 | putnopvut | Now there's also a timeout check after the periodic announcement playback so that we don't try to call a queue member after we've already exceeded the timeout. |
| 20:39.43 | putnopvut | If a call is placed with time still left on the clock, but the time left is less than the timeout specified in queues.conf, we go with whatever time is remaining before the app should timeout. That's how it was before too. |
| 20:39.44 | seanbright | dexthageek: turn on agi debugging ('agi debug' in CLI) and turn on verbose at the CLI (core set verbose 10) and pastebin a call attempt |
| 20:40.17 | seanbright | putnopvut: doesn't all of that seem extraordinarily complex to you? |
| 20:40.50 | putnopvut | seanbright: yes :) |
| 20:40.50 | seanbright | the dual timeouts have always confused me |
| 20:40.55 | putnopvut | me too. |
| 20:41.01 | putnopvut | I think the one in queues.conf should be renamed. |
| 20:41.07 | seanbright | and sadly (i think) for no good reason |
| 20:41.24 | putnopvut | Something like "ringtime" to indicate that it's the amount of time to try ringing a queue member. |
| 20:41.42 | seanbright | putnopvut: commit away :) |
| 20:41.48 | putnopvut | heh |
| 20:42.09 | seanbright | the timeout in the queue app call should be independent of everything |
| 20:42.20 | seanbright | just need to fire up 1 thread per caller that starts a clock |
| 20:42.23 | seanbright | done and done |
| 20:42.23 | seanbright | heh |
| 20:42.35 | heedly | How would I print out times from one event to another? |
| 20:43.05 | seanbright | heedly: ${EPOCH} |
| 20:43.12 | seanbright | heedly: and a little math |
| 20:43.32 | seanbright | Set(Now=${EPOCH}) |
| 20:43.38 | heedly | that works, thanks. |
| 20:43.38 | seanbright | ; do stuff |
| 20:43.44 | seanbright | perfect |
| 20:43.54 | dexthageek | http://pastebin.com/d1b057769\ |
| 20:43.56 | dexthageek | http://pastebin.com/d1b057769 |
| 20:44.25 | seanbright | dexthageek: is the channel answered? |
| 20:44.30 | seanbright | i.e. Answer() |
| 20:44.53 | dexthageek | yes |
| 20:45.43 | seanbright | add a stream file in there and see if you hear that |
| 20:45.47 | *** join/#asterisk lanning (n=lanning@66.151.128.195) |
| 20:45.57 | *** join/#asterisk angryuser (n=sldf@88.140.128.235) |
| 20:46.08 | grandpapadot | seanbright: permissions... asterisk (bin) couldn't traverse the entire path to the script..but oddly,nothing in debug .. |
| 20:46.25 | seanbright | grandpapadot: strange indeed. give this a shot: |
| 20:46.40 | seanbright | grandpapadot: chown asterisk /path |
| 20:46.46 | seanbright | heh |
| 20:47.02 | seanbright | runs mkdir -p /path/to/my |
| 20:47.41 | grandpapadot | Yea, got it fixed once I figured it out .. I keep everything in /path/to/my ... I just chmod -R 777 * and addgroup asterisk root and addgroup www-data root and have my ftp path set to "/", makes everything a lot easier |
| 20:47.57 | grandpapadot | My root passwords are easy to remember too, just "password" |
| 20:48.20 | seanbright | grandpapadot: hot |
| 20:48.30 | seanbright | grandpapadot: is your IP 127.0.0.1? |
| 20:48.32 | dexthageek | seanbright: stream_file did not work during the music on hold |
| 20:48.38 | grandpapadot | seanbright: How did you know? |
| 20:48.48 | seanbright | dexthageek: meaning it threw an error or you didn't hear it? |
| 20:48.51 | *** join/#asterisk astassistant (n=skip@65-126-63-1.dia.static.qwest.net) |
| 20:48.53 | seanbright | grandpapadot: i'm l33t |
| 20:49.47 | dexthageek | seanbright: did not hear it |
| 20:49.57 | grandpapadot | Later, all. |
| 20:50.01 | seanbright | dexthageek: ok... have you ever heard audio on this device? |
| 20:51.21 | dexthageek | yes many times |
| 20:51.29 | seanbright | dexthageek: do me a favor... throw a 'CHANNEL STATUS' call at the beginning of your agi script and get another pastebin with agi debug turned on |
| 20:51.32 | seanbright | will brb |
| 20:51.45 | *** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194) |
| 20:57.19 | datachomper | is musiconhold application built into the core? I don't see it in the apps directory. |
| 20:57.46 | *** join/#asterisk Assid (n=assid@unaffiliated/assid) |
| 20:57.53 | Blackvel | is there a SET command for call redirection indicator (so handy shows "call forward")? |
| 20:58.28 | seanbright | datachomper: res/res_musiconhold.c |
| 20:58.29 | Assid | is there a way to FORCE asterisk to connect to postgres using a non TCP socket connection to connect to postgres? |
| 20:58.50 | seanbright | dexthageek: any luck? |
| 21:00.04 | seanbright | Assid: don't include a hostname |
| 21:00.18 | galeras | Thanks for any help: in a PRI, incoming calls are fine, outgoing calls fails. Telco can make calls in both ways using a tester. I have a Te120p Card. Please take a look of: http://pastebin.ca/1093164 |
| 21:00.32 | galeras | same results with latest 1.4 and 1.2 versions |
| 21:00.42 | [TK]D-Fender | galeras: what do they say when they SEE your call atttempt? |
| 21:01.00 | datachomper | I'm thinking that "set music on" only toggles flags for the actually musiconhold application. The application is what's playing the music and blocking the channel. |
| 21:01.09 | dexthageek | seanbright: http://pastebin.com/d29a87ccf |
| 21:01.29 | seanbright | datachomper: take a look at the code in res/res_agi.c |
| 21:01.44 | seanbright | datachomper: it specifically starts moh on the channel |
| 21:01.59 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 21:02.04 | seanbright | dexthageek: all of the CLI output, please |
| 21:02.06 | ptimmins | galeras, send the whole 10 digit number |
| 21:02.18 | ptimmins | also, fix your ANI, they may be hating on it |
| 21:02.29 | ptimmins | # |
| 21:02.30 | ptimmins | > Calling Number (len= 7) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) |
| 21:02.30 | ptimmins | # |
| 21:02.30 | ptimmins | > Presentation: Presentation permitted, user number not screened (0) '199' ] |
| 21:02.33 | ptimmins | not valid, ya know |
| 21:03.08 | galeras | [TK]D-Fender: unfortunately they aren't helpful, the just show me they can make calls in both ways and tellme the PRI is OK. |
| 21:03.28 | [TK]D-Fender | galeras: No, told you to have the WATCH while YOU made the call. |
| 21:03.40 | seanbright | them* |
| 21:03.48 | [TK]D-Fender | ^^ |
| 21:04.22 | galeras | ^^ |
| 21:04.34 | seanbright | << |
| 21:04.38 | datachomper | seanbright, Ya, you're right |
| 21:04.45 | seanbright | datachomper: consistently! |
| 21:04.47 | seanbright | :) |
| 21:06.22 | Assid | hrmm i still have to use odbc for postgresql ? |
| 21:06.41 | seanbright | Assid: for what? realtime? cdrs? |
| 21:06.49 | Assid | seanbright: realtime |
| 21:06.57 | seanbright | Assid: res_config_pgsql |
| 21:07.06 | galeras | i will try with a red-fone |
| 21:07.08 | Assid | so i dont need them addons? |
| 21:07.23 | seanbright | Assid: not sure what addons you mean, but you shouldn't |
| 21:07.59 | seanbright | dexthageek: pb your musiconhold.conf file too |
| 21:08.56 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 21:11.13 | dexthageek | seanbright: http://pastebin.com/d6b13f3cf |
| 21:11.40 | errr | is ${UNIQUEID} an asterisk global variable? |
| 21:11.56 | *** join/#asterisk nn (n=nn@unaffiliated/nn) |
| 21:12.02 | *** join/#asterisk foexle (n=heiko@88-134-242-24-dynip.superkabel.de) |
| 21:12.10 | seanbright | errr: it's a channel variable |
| 21:12.34 | errr | seanbright: ok, I guess what I mean is it defined by asterisk, and not by me |
| 21:12.35 | dexthageek | seanbright: i am using the sample musiconhold.conf: http://pastebin.com/d28e65141 |
| 21:13.28 | seanbright | dexthageek: and you have files in /var/lib/asterisk/mohmp3? |
| 21:13.41 | seanbright | what version of asterisk is this? |
| 21:14.00 | dexthageek | seanbright: yes. I have exec MusicOnHold i hear music agi->exec('MusicOnHold') |
| 21:14.10 | seanbright | sweet |
| 21:14.12 | seanbright | so you're good? |
| 21:14.38 | dexthageek | seanbright: 1.4.19.1 |
| 21:14.54 | seanbright | errr: yes |
| 21:15.06 | dexthageek | seanbright: no if i run $agi->set_music(true) it says starting Music on hold. but i do not hear any music |
| 21:15.21 | errr | seanbright: great thanks |
| 21:15.25 | seanbright | dexthageek: but if you run $agi->exec('MusicOnHold') you hear music? |
| 21:15.31 | dexthageek | seanbright: yes |
| 21:15.44 | km- | russelb: hey, is there any way to access your devstate stuff via ami? |
| 21:16.03 | seanbright | dexthageek: how about $agi->exec('StartMusicOnHold') |
| 21:16.28 | dexthageek | seanbright: let me try it on sec |
| 21:16.51 | km- | russell: I'm trying to tweak my presence script to not rely on ExtensionStatus, I've got it working with Link/Hangup events but would prefer the extended info of ExtensionStatus. Any ideas? |
| 21:17.16 | *** part/#asterisk hubguruJR (n=hubguruJ@mail.ntegratedsolutions.com) |
| 21:17.18 | km- | only reason why I dont wanna use extensionstatus is I dont want to have to require users to hint their extensions |
| 21:17.50 | dexthageek | seanbright: the application runs but no music is heard |
| 21:18.07 | seanbright | dexthageek: than something is jacked. |
| 21:18.09 | Corydon76-dig | km-: you can use AMI GetVar with the associated dialplan function |
| 21:18.26 | seanbright | dexthageek: i would just *try* to throw an $agi->Answer() in there |
| 21:18.32 | seanbright | dexthageek: and see if that does anything |
| 21:18.33 | km- | Corydon: i.e., devstate? |
| 21:18.41 | Corydon76-dig | DEVICE_STATE |
| 21:18.44 | km- | corydon: but in that case I'm polling for status, not trapping an event. |
| 21:18.49 | dexthageek | seanbright: in the MOH block? |
| 21:19.04 | seanbright | dexthageek: as the first thing your AGI does |
| 21:19.18 | dexthageek | seanbright: its already in there |
| 21:19.32 | Corydon76-dig | km-: correct, polling only |
| 21:19.54 | km- | corydon: I dont suppose that's the end of the world, but it'd be nicer to get an event. I wonder if there's a way to merge russell's devstate magic with ExtensionStatus events |
| 21:20.01 | kfife | Hey folks, FRIDAY I was mentioning an über-geeky project wehre people were connecting old Strowger Step-By-Step switches using Asterisk and ATA |
| 21:20.01 | seanbright | dexthageek: that is is... crap |
| 21:20.21 | *** join/#asterisk franhp (n=franhp@75.Red-88-24-201.staticIP.rima-tde.net) |
| 21:20.25 | kfife | ...using 'virtual tie-lines'. I was asked for the URL and I couldn't find it. |
| 21:20.34 | kfife | ...but I have it now. |
| 21:20.35 | seanbright | dexthageek: i am fresh out of ideas |
| 21:20.36 | Corydon76-dig | km-: we already have problems with devicestate changes happening too fast and confusing watchers |
| 21:20.55 | kfife | The URL is http://home.comcast.net/~kirtley.stanfield/ |
| 21:21.13 | km- | corydon: I wonder how badly the manager interface could get pounded for polls |
| 21:21.23 | dexthageek | seanbright: I appreciate the help! Thanks |
| 21:21.28 | Corydon76-dig | km-: some of that may be addressed in 1.6, since we queue devstate changes with the actual devicestate, instead of asking every single time. |
| 21:21.39 | seanbright | dexthageek: i hope you figure it out. i think the idea is sound, just not sure why you aren't getting audio. |
| 21:21.41 | *** join/#asterisk ta^3 (n=tacvbo@conference/cluecon/x-105b6b0aab5f0aea) |
| 21:21.56 | kfife | ...Check it out if you get a moment. It's one of the geekiest, impractical outstanding things I've seen |
| 21:22.02 | km- | kfife: that's awesome. |
| 21:22.06 | Corydon76-dig | km-: it would be nice, but it's not going to happen until the races get addressed |
| 21:22.06 | seanbright | dexthageek: you don't hear the "goodbye" at the end? |
| 21:22.08 | km- | kfife: next step, cordboard? |
| 21:22.23 | kfife | LOL |
| 21:22.46 | kfife | There's actuall a private namespace for the 7 digit dialing that exists in the 'network' |
| 21:22.47 | km- | corydon: you can dynamically add extensions via AMI, right? |
| 21:23.16 | kfife | You can join the 'club' and get an exchange in that private namespace |
| 21:23.28 | kfife | ...I recommend you check it out. Very interesting read. |
| 21:23.54 | kfife | ...Many dialtones on the switches are generated by a MECHANICAL dial tone generator |
| 21:24.05 | km- | kfife: I definitely dont have that much hardware |
| 21:24.21 | seanbright | km-: with UpdateConfig |
| 21:24.22 | kfife | KM: and perhaps not enough space for it if you did :-) |
| 21:24.50 | *** join/#asterisk grandpapadot (n=no@adsl-074-185-089-046.sip.bhm.bellsouth.net) |
| 21:24.52 | grandpapadot | dundi rocks ... |
| 21:24.55 | kfife | ...One of the best things to do is listen to the audio recording taken from the middle of a CO populated with strowger switches |
| 21:24.55 | km- | kfife: that's for sure. |
| 21:25.09 | km- | I was about to click it but didnt want my head to explode |
| 21:27.44 | km- | corydon: so you're saying that hint/ExtensionStatus can sometimes cause races under high load? |
| 21:27.50 | km- | corydon: or just device updates in general |
| 21:27.51 | *** join/#asterisk Yourname` (i=Yourname@unaffiliated/yourname/x-837320) |
| 21:28.31 | kfife | Speaking of DUNDI Does anyone here know if the enum.org public trial is still open? I am listed at e.164.org, but BETTER would be e164.arpa. Apparently end users and companies were allowed to populate (or receive delegation) of their zone in e164.arpa. I signed up but haven't heard anything. Is it closed? |
| 21:28.48 | km- | not sure, I use freenum.org now though |
| 21:29.27 | km- | isn*itad is better imho only because I know e164 had some random rules about who could get in on it, but freenum has iana assigning numbers for it |
| 21:30.08 | kfife | I use freenum too, but I would estimate that most ITSPs interconnects and organizations looking for a cheap or NETWORK AGNOSTIC route to the destination, enabling a myriad of things |
| 21:30.23 | kfife | ...would be looking in e164.arpa |
| 21:30.32 | kfife | ...first before any other enum tree |
| 21:30.34 | km- | e164 isn't really network agnostic though, is it? |
| 21:30.40 | km- | I mean, it requires e.164.org to maintain it |
| 21:30.42 | kfife | I mean the transport is network agnostic |
| 21:31.16 | km- | you're saying that given the two choices, providers would be more likely to choose the one that more closely mimics their number assignment paradigm |
| 21:31.49 | kfife | In other words, as long as the PSTN is the clearinghouse for e164 addresses, we will never be able to have wideband calls, video etc |
| 21:32.14 | kfife | KM: to your question, I'm saying that the larger the enum tree, the more traction it will get. |
| 21:33.11 | kfife | so you're right, but I think that if people and orgainizations can have their zones delegated it stands the greatest chance of assisting real-time voice toward the end goal of being network agnistic |
| 21:33.20 | kfife | ...by |
| 21:33.31 | km- | yeah, but that's really a chicken-and-the-egg -- people wont use it until it's populated, providers wont use it till users are using it |
| 21:33.31 | kfife | 'it' I mean enum.arpa |
| 21:34.16 | kfife | KM: Right again, but there's a nice migration path. In other words, my routing is: Query enum.arpa, use the route. If none found, terminate on the PSTN |
| 21:35.01 | kfife | If I have to query five different enum trees, enum becomes less of a problem-solver, and more of a neat-o technolgoy |
| 21:35.57 | kfife | Poll: do you think there's a market for a goldentree.org type business that is an enum directory that consolidates all of the known enum trees? |
| 21:37.14 | kfife | ...That's a poll, not a response to a user named poll :-) |
| 21:37.57 | km- | sorry, ADD kicked in and I was watching the youtube of obscura digital's holographic display |
| 21:38.48 | km- | goldentree.org resolves to a school |
| 21:39.06 | km- | ah |
| 21:39.17 | *** part/#asterisk datachomper (n=russ@h-66-134-255-227.nycmny83.covad.net) |
| 21:39.20 | *** join/#asterisk ricko73 (n=dhartman@wilug/newlug/ricko73) |
| 21:39.26 | kfife | sorry, that was a hypothetical name to describe a potential business |
| 21:39.35 | km- | I think the golden tree idea is the right one, everyone needs to stop forking and make one central database |
| 21:40.00 | km- | I was part of that problem with enum.fierymoon.com too, totally should have just proselytized for a central store somewhere |
| 21:40.08 | ricko73 | I have a question/issue related to automon that I'm not finding documentation to help with |
| 21:40.28 | ricko73 | it works fine on sip only channels, but is not working at all if a zap channel is involved |
| 21:40.50 | km- | I think there's a different monitoring for zap |
| 21:40.55 | ricko73 | Is there anything special that needs to happen to record calls using automon with zap channels? |
| 21:40.56 | km- | dunno what automon is though so I'll STFU now :) |
| 21:41.10 | kfife | A consolodation script would be a ten-minute script for someone proficient. A few hours for me:-) |
| 21:41.13 | ricko73 | km see /etc/asterisk/features.conf |
| 21:41.28 | *** join/#asterisk macros73_ (n=cs@c-67-163-224-69.hsd1.pa.comcast.net) |
| 21:42.16 | ricko73 | Uses the dial() command option w or W to record |
| 21:42.26 | km- | kfife: yeah, but then someone is tasked with making sure someone maintains the list of all the providers |
| 21:42.46 | kfife | You are right. |
| 21:43.24 | km- | plus you'd need to write a dns server app that took the request for enum resolution then enum'd all the sources |
| 21:43.30 | km- | what do you do with collisions? |
| 21:43.44 | km- | I have 5.1.1.e164.arpa and then joe has 5.1.1.e164.arpa on another provider |
| 21:43.47 | km- | Who wins? |
| 21:44.05 | *** join/#asterisk RobH (n=RobH@cpe-72-184-129-136.tampabay.res.rr.com) |
| 21:44.30 | km- | Every time I get in this convo it just seems more and more important that if we want ENUM to survive as a technology, everyone has to agree to one unified registrar |
| 21:44.50 | kfife | I would envision the pseudocode being something like: query all known enum trees, return top response to original query ordered by not-null, then ordered by subscriber's individual preference order given at time of signup, or a logical order if no preference given. |
| 21:45.21 | km- | that's rube-goldberging it though, no offense |
| 21:45.48 | kfife | KM: Enum.org only delegates e.164 numbers which are formatted as +1npanxxxxxx |
| 21:45.56 | kfife | so 511 would not be delegated |
| 21:46.02 | km- | it was an example, though. |
| 21:46.12 | km- | you have two enum providers with the same registered item |
| 21:46.19 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 21:46.28 | km- | the magic-parser would need to decide for the user which one they wanted |
| 21:46.40 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.95.130) |
| 21:46.44 | kfife | you're the master of your own switch. You can route according to whichever enum tree you like. |
| 21:46.45 | km- | and no matter how smart you make it's decision tree, the chance is always there that he's calling Joe Internet rather than Gramma |
| 21:47.14 | km- | right, that's the situation we're in now, you can enum query against any server you want |
| 21:47.22 | dexthageek | quit |
| 21:47.24 | km- | if you wanted a unified solution |
| 21:47.31 | km- | you'd need to deal with this issue |
| 21:48.00 | ricko73 | hmmm, perhaps it's a dtmf issue |
| 21:49.07 | km- | I would imagine that large companies would utilize their own enum trees |
| 21:49.28 | kfife | KM: That's why I like e164.arpa. It's overseen to ensure that the rightful owner of +13122977000 is the only party able to update that zone. No different than the DNS. Only the right people can update the microsoft.com zone to prevent domain hijacking. |
| 21:50.07 | km- | I thought e164.arpa's rules were more than that at one point |
| 21:50.14 | km- | more than "you have to own this number" |
| 21:50.20 | km- | that's why I shied away from it |
| 21:50.45 | kfife | That's the purpose of the public enum trial. You would ask for a delegation, and obtain a registrar, just like a domain owner. |
| 21:51.27 | kfife | I don't knwo what the process is for proving that you're a number's RESPORG. Probably no different than the rules for LNP. |
| 21:51.47 | km- | reading e164.org's page |
| 21:51.55 | km- | looks like they have a script that calls the number and gives you a pin |
| 21:51.55 | kfife | There's no doubt that this is the reason why enum languishes |
| 21:51.58 | km- | pretty straightforward |
| 21:52.14 | km- | hehe resporg |
| 21:52.31 | kfife | e164.org is not the same as enum.org. |
| 21:52.41 | kfife | e164.org is a community based enum tree. |
| 21:52.43 | km- | pats his sms/800 credentials |
| 21:52.45 | km- | hmm |
| 21:53.27 | kfife | enum.org is the web site for the IANA-overseen e164.arpa |
| 21:53.33 | kfife | which is the 'golden tree' |
| 21:53.39 | km- | ah yeah, that enum.org looks like you gotta be a carrier to get it |
| 21:54.04 | kfife | Again, that's the point of the public trial. You do not have to be a carrier for the trial. |
| 21:54.21 | kfife | Read this: http://www.enum.org/information/trial.cfm |
| 21:54.54 | kfife | On that page it concurrs with your 'gotta be a carrier' |
| 21:55.23 | *** join/#asterisk ta^3 (n=tacvbo@conference/cluecon/x-e84ea50658e2e9b8) |
| 21:55.23 | kfife | but other pages I've read that it's not so strict. |
| 21:55.39 | km- | ah well |
| 21:56.09 | km- | ok, 6pm, time for me to leave 30 mins ago |
| 21:56.20 | km- | kfife: keep up the good fight for enum :) |
| 21:56.23 | km- | & |
| 21:56.29 | kfife | Thansk! |
| 21:57.51 | bijit | how can I make aastra phones autoconf vlan id? |
| 21:58.07 | *** join/#asterisk awannabe (n=brad@ip24-251-152-67.ph.ph.cox.net) |
| 21:58.08 | *** join/#asterisk implicit (n=bayan@unaffiliated/implicit) |
| 21:58.30 | awannabe | Hey guys, is it possible to run a Digium FXO card and a Sangoma PRI card in same system? |
| 21:58.34 | *** join/#asterisk dogmeat (n=Bob@unaffiliated/dogmeat) |
| 22:00.43 | *** join/#asterisk rmayorga_ (n=rmayorga@unaffiliated/rmayorga) |
| 22:01.46 | *** part/#asterisk ptimmins (n=paul@gw.corp.clearrate.com) |
| 22:04.28 | Blackvel | is there a list of good isdn BRI cards for asterisk (and good drivers) on voip-info or somewhere? |
| 22:04.47 | *** join/#asterisk lanning (n=lanning@66.151.128.195) |
| 22:06.12 | Qwell | Blackvel: Digium B410 |
| 22:06.18 | Qwell | B410p? I forget. |
| 22:08.33 | awannabe | BRI cards, fun! |
| 22:09.53 | Corydon76-dig | Qwell: no p suffix on new cards |
| 22:10.14 | Qwell | define 'new'. |
| 22:10.53 | ricko73 | automon works fine... I wasn't pushing the key sequence fast enough |
| 22:11.12 | Qwell | ricko73: You can change the timeout in features.conf. default is 500ms |
| 22:11.31 | ricko73 | I changed that to 1000ms and will test |
| 22:11.40 | ricko73 | (well changed it to 1000 ) |
| 22:12.42 | ricko73 | Qwell: will asterisk need to be stopped and started for that to change or is there a reload that can handle a change in features.conf? |
| 22:12.48 | Blackvel | well its a 4 s0 port :) didn't someone mention Sangoma as well as? How about OpenVOX? I do not run a big company with many people and telephones but I may require a two s0 BRI card. who could tell me the difference in quality comparing to a cheap Zaphfc/fritz passive isdn card? I need to run this * with pstn connect / ivr and pbx integration (phones) |
| 22:12.50 | *** join/#asterisk jm|laptop (n=jm|home@zen.jamiem.com) |
| 22:12.58 | *** join/#asterisk ltd (n=z@pat.transact.net.au) |
| 22:14.34 | Blackvel | problems seem to be all the time the drivers for BRI. misdn/bristuff...visdn... this system should work 99,99% o.k during the week (no space for bristuff errors and not picking up calls because of errors) |
| 22:15.20 | Blackvel | i may be able to fire away my analog/isdn telephone, but surely not my isdn pstn number connection |
| 22:15.31 | Blackvel | e.g to go for fxs module |
| 22:15.49 | Blackvel | so, where are we standing in 2008? :) |
| 22:16.58 | Assid | err sometimes i see a whole lot of invites in sip show channel |
| 22:17.08 | Assid | is there a way to timeout those invites so they dont wait forever |
| 22:21.36 | Assid | i see a lots of Init: INVITE |
| 22:23.25 | Assid | i guess i should update it |
| 22:27.47 | C4colo | is not equal != or <> in the dialplan? |
| 22:28.09 | *** join/#asterisk astassistant (n=skip@65-126-63-1.dia.static.qwest.net) |
| 22:28.53 | Assid | alrite im out.. gnight |
| 22:29.34 | km- | [tk]d-fender: hey, you home? |
| 22:29.52 | *** join/#asterisk davidstrauss (n=straussd@wikimedia/davidstrauss) |
| 22:30.30 | davidstrauss | Is there a way to override configuration files created by AsteriskNow? |
| 22:30.41 | davidstrauss | It seems like changes get overwritten by the GUI periodically. |
| 22:32.31 | *** join/#asterisk craigk (n=craigk@58.174.150.119) |
| 22:33.44 | *** join/#asterisk nnY_2 (n=nny_1@64.203.237.47) |
| 22:34.01 | nnY_2 | so I am tracking down an issue and some advice would be helpful |
| 22:34.26 | nnY_2 | I seem to have some SIP channels that are "open" even though no one is using the phone |
| 22:34.26 | davidstrauss | nnY_2: Rule 1 of technical IRC channels: Don't ask to ask. |
| 22:34.39 | nnY_2 | davidstrauss I know this, it was a preface |
| 22:35.14 | nnY_2 | 10.0.0.228 13 6f8631a4570 00102/00101 0x3f0004 (ulaw| No Rx: BYE |
| 22:35.35 | davidstrauss | Doesn't Rx: BYE indicate a hangup? |
| 22:35.42 | nnY_2 | I would have thought so |
| 22:35.57 | davidstrauss | nnY_2: have you enabled verbose logging? |
| 22:36.01 | Katty | jbot: cluemuffin |
| 22:36.13 | Katty | [TK]D-Fender: :< |
| 22:36.19 | davidstrauss | nnY_2: you can watch asterisk close out old SIP channels |
| 22:37.04 | nnY_2 | davidstrauss i just enabled sip debug for that peer |
| 22:37.19 | davidstrauss | does debug include verbose logging? |
| 22:37.48 | *** join/#asterisk Gat0rvean (n=gredish@64.191.128.145) |
| 22:38.33 | nnY_2 | davidstrauss looks like i can enable it in logging.conf |
| 22:38.53 | davidstrauss | yes |
| 22:38.59 | davidstrauss | you'll see things like: |
| 22:39.01 | davidstrauss | [Aug 5 22:17:29] VERBOSE[539] logger.c: Scheduling destruction of SIP dialog '749977346efd315f2968c0657b62cb27@192.168.69.1' in 32000 ms (Method: REGISTER) |
| 22:39.19 | davidstrauss | so, my server is keeping channels open at least 32 seconds after they close |
| 22:39.47 | davidstrauss | and then later: |
| 22:39.48 | davidstrauss | [Aug 5 22:16:52] VERBOSE[539] logger.c: Really destroying SIP dialog '66813f5250c0fc9e78356e1773eaf044@10.0.0.200' Method: NOTIFY |
| 22:40.05 | nnY_2 | ok i get that dialog in the console from debugging sip fwiw |
| 22:40.09 | davidstrauss | (examples only, not from the same call) |
| 22:40.16 | nnY_2 | Really destroying SIP dialog '25e94ae049802275238c620a1d030133@10.0.0.1' Method: OPTIONS |
| 22:40.23 | nnY_2 | thats the ID of the open channel/call |
| 22:40.28 | nnY_2 | but it remains :\ |
| 22:41.04 | nnY_2 | in logger.conf is suggests not enabling verbose logging for production systems |
| 22:41.15 | nnY_2 | as it produces a metric ton of output |
| 22:41.41 | nnY_2 | well unless you are debugging so ok there i go |
| 22:42.15 | nnY_2 | so can someone enlighten me as to what kinds of things would cause this so I know what to look for? |
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| 22:47.31 | nnY_2 | so all i see is http://pastebin.com/m2e94ad45 and the nonce expiration over and over |
| 22:48.38 | *** join/#asterisk dmz (n=dmz@64.203.203.232.dyn-cm-pool-64.hargray.net) |
| 22:50.08 | nnY_2 | I have enabled sip history as well |
| 22:50.13 | *** join/#asterisk LND (n=Lee@92-233-208-244.cable.ubr08.gate.blueyonder.co.uk) |
| 22:50.21 | nnY_2 | sadly just now so I may have to let the logs fill up for 24 hours |
| 22:53.43 | nnY_2 | hmm let's try a different approach.. Anyone want to make some dough assisting me with this issue? |
| 22:55.46 | C4colo | the problem is that sip channels are staying open even after "really destroying"? |
| 22:56.19 | nnY_2 | it appears that way |
| 22:56.21 | C4colo | upgrade to the latest asterisk, there was a version that had this issue |
| 22:56.30 | nnY_2 | 1.4.21.2 atm |
| 22:56.35 | C4colo | it does happen on some systems even with the latest asterisk |
| 22:56.51 | C4colo | another option that some have implemented is running "restart when convenient" at midnight from a cron job |
| 22:56.57 | nnY_2 | well |
| 22:57.00 | nnY_2 | it screws up the hints |
| 22:57.03 | nnY_2 | and the sidecar |
| 22:57.07 | C4colo | grandstreams? |
| 22:57.12 | nnY_2 | ha hell nah |
| 22:57.23 | nnY_2 | well nto that much better in some peoples opinion, but linksys 962 |
| 22:57.51 | C4colo | I haven't worked with those ... is there a setting for subscription expiration? |
| 22:58.02 | nnY_2 | I believe so |
| 22:58.10 | C4colo | the problem with the grandstreams is that they do not re-issue the subscription for notifications |
| 22:58.15 | C4colo | unless rebooted |
| 22:58.57 | C4colo | how quickly do they build up? 100 per day? 100 per hour? |
| 22:59.23 | nnY_2 | nah only 1 or 2 a day it seems |
| 22:59.36 | nnY_2 | this is after a day of use, and there is one phone that reports "busy" on the blf |
| 22:59.51 | C4colo | poe switch or wall adapters? |
| 23:00.03 | nnY_2 | yesterday it was two, although I haven't got enough historical data to see if it's the same phones or diff ones |
| 23:00.05 | nnY_2 | poe switch |
| 23:00.39 | C4colo | have it run the cron job to restart on monday morning at 12:01 am then cycle the poe switch |
| 23:00.51 | C4colo | fairly rough workaround I know |
| 23:01.07 | Qwell | grandstream? might as well run the script daily |
| 23:01.12 | nnY_2 | can't really, this is a client site |
| 23:01.16 | C4colo | no linksys |
| 23:01.16 | nnY_2 | not grandstream |
| 23:01.22 | nnY_2 | and the phone doesn't think it is being used |
| 23:01.42 | C4colo | do the linksys support remote reboot triggers? |
| 23:01.47 | C4colo | like sipura atas do |
| 23:01.52 | nnY_2 | yeah but this is not a workaround |
| 23:02.03 | nnY_2 | they need it to work properly for the full day |
| 23:02.08 | C4colo | yea |
| 23:02.13 | nnY_2 | a busy blf when not is gonna screw them up big time |
| 23:02.27 | C4colo | I have never gotten a solid answer on why that happens |
| 23:03.02 | C4colo | on one of our servers it happens from the voicemail application once or twice a month |
| 23:03.11 | C4colo | a restart of asterisk is required to clear it |
| 23:03.20 | drmessano | Grandstream is trying to suck just enough to keep the price down, and to make them easy to buy out :) |
| 23:03.21 | C4colo | have tried many forms of clearing the call |
| 23:04.02 | nnY_2 | drmessano i dunno how grandstream got involved in this conversation |
| 23:04.17 | nnY_2 | so the phone is keeping the channel open or asterisk is? |
| 23:04.33 | nnY_2 | cause there is no line usage reported on the phone |
| 23:04.35 | C4colo | I asked if you were using grandstreams when you said a restart of asterisk breaks BLF |
| 23:04.43 | nnY_2 | oh yeah heh |
| 23:04.43 | C4colo | because that is a common issue with grandstreams |
| 23:04.57 | nnY_2 | ok good, wondering if it's the phone or * |
| 23:04.59 | C4colo | never heard of it on the linksys phones, but not surprised |
| 23:05.32 | nnY_2 | why you think linksys has a seperate division for phones than cisco? |
| 23:05.37 | nnY_2 | cause I don't |
| 23:05.40 | C4colo | yes |
| 23:05.57 | C4colo | the linksys phones are closer to the sipura line of products than the cisco 7xxx phones |
| 23:06.05 | drmessano | I dunno why people piss on the Linksys stuff so much |
| 23:06.08 | drmessano | It works |
| 23:06.10 | nnY_2 | yeah me neither |
| 23:06.12 | nnY_2 | i love it |
| 23:06.21 | nnY_2 | cough* more than polycom* cough |
| 23:06.25 | C4colo | I don't like cisco or linksys |
| 23:06.34 | drmessano | ~polycommunist |
| 23:06.34 | jbot | A polycommunist is someone who believes Polycom phones can do no wrong.. that Polycom's are so over and above anything else, that what you are using is surely crap, and the mere fact you mention another brand name is of great insult to the channel, the community, and the world. They may also be getting a 10% kickback. |
| 23:06.34 | C4colo | I like Aastras better than polycoms |
| 23:06.40 | nnY_2 | hahahaha |
| 23:07.01 | C4colo | lol, that's pretty good |
| 23:07.03 | drmessano | One of my finest works |
| 23:07.12 | C4colo | polycoms are nice, but have a crappy user interface |
| 23:07.13 | nnY_2 | I use the 480i although I have had to rma a couple of aastras for wierd crap |
| 23:07.17 | drmessano | "They may also be getting a 10% kickback." |
| 23:07.18 | nnY_2 | weird too |
| 23:07.28 | C4colo | is it just me or is 3 programmable buttons crap? |
| 23:07.28 | *** join/#asterisk `Sean (i=Un1x@CPE001d451b875f-CM00111ade88b6.cpe.net.cable.rogers.com) |
| 23:07.59 | C4colo | the 5i series seems to be much better than the older hardware |
| 23:08.19 | C4colo | the 9143i is a good "standard" phone built on the 5i hardware |
| 23:08.52 | C4colo | anyway, back to the point |
| 23:09.03 | C4colo | those rogue channels are just a thing with asterisk at this point |
| 23:09.13 | C4colo | nobody has ever explained where they come from or how to fix them |
| 23:09.24 | C4colo | some systems have more than others |
| 23:09.33 | nnY_2 | I am looking at the linksys web interface right now for any indication |
| 23:09.40 | drmessano | I think there is a certain price baseline where all IP phones hit a crossroad of price vs quality.. Then above that is about how much money you want to spend on a damn phone |
| 23:09.41 | nnY_2 | so is it the phone that is causing this or the server? |
| 23:09.54 | C4colo | we process 15,000 calls per day on average and only have about 4 or 5 per month, while some people have up to 100 per day show up |
| 23:10.27 | drmessano | I had problems with the channels getting stuck |
| 23:10.31 | drmessano | But pre 1.4.20 |
| 23:10.35 | nnY_2 | all the line statuses here are idle in the phone |
| 23:10.40 | nnY_2 | next registration in 1902s |
| 23:10.43 | C4colo | yea, we are running 1.4.21 I beleive |
| 23:10.45 | C4colo | actually let me check |
| 23:11.01 | drmessano | I'm on some 1.4.21ish SVN |
| 23:11.03 | C4colo | subscription expiration is more important for the BLF functions |
| 23:11.08 | C4colo | for notifications |
| 23:11.32 | C4colo | oh, hmm, 1.4.19 on the server I saw it most commonly |
| 23:12.23 | C4colo | drmessano: <nnY_2> 1.4.21.2 atm |
| 23:12.33 | C4colo | so it is post .20 |
| 23:12.40 | nnY_2 | yeah if not for the blfs a nightly restart of * would clear things like this up, but right now it is a major issue with this install and basicaly i'll be working on it till it's gone |
| 23:13.16 | Blackvel | n8 |
| 23:14.03 | C4colo | how much googling have you done on the issue? |
| 23:14.18 | nnY_2 | been trying, can't seem to find the right keywords yet |
| 23:14.19 | C4colo | last time I researched it I didn't find much but supposedly "fixed" bug reports and such |
| 23:14.30 | C4colo | and more people saying "wtf is this?" |
| 23:14.45 | nnY_2 | so if i reboot the phone, should the problem go away temporarily? |
| 23:16.21 | C4colo | is it a channel between the phone and asterisk? |
| 23:16.55 | C4colo | or do you mean the BLF light being "busy"? |
| 23:17.05 | C4colo | because asterisk is telling the phone that the other extension is "busy" |
| 23:17.11 | C4colo | because there is an active chanel for it |
| 23:17.19 | C4colo | ... as far as asterisk knows |
| 23:18.06 | nnY_2 | ha from an irclog "shit or get off the POTS." |
| 23:18.12 | nnY_2 | i want that as a bumper sticker |
| 23:18.13 | C4colo | haha |
| 23:18.35 | nnY_2 | yeah i figured that |
| 23:19.00 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 23:19.17 | *** join/#asterisk zippytech (n=zippytec@mail.zippytech.com) |
| 23:19.33 | zippytech | where is the setting for where the pid is placed? |
| 23:19.39 | nnY_2 | so I guess the next question is why does * think the channel is still active |
| 23:19.49 | C4colo | the call duration was too short? |
| 23:19.49 | nnY_2 | especially if it's last sip was BYE |
| 23:19.57 | C4colo | that was one suggestion from someone who had the problem |
| 23:20.09 | zippytech | mine is l tring /var/run/asterisk.pid and i think it should be /var/run/asterisk/asterisk.pid |
| 23:21.01 | nnY_2 | i'll check the cdr and see what crappened last |
| 23:21.21 | C4colo | sip show channels should show the last (aka current) application |
| 23:24.04 | nnY_2 | well there is a 1 second call to extenion 10 at 3 pm this afternoon |
| 23:24.10 | nnY_2 | any way to see how long the channel has been open? |
| 23:24.26 | C4colo | start - end |
| 23:24.33 | nnY_2 | k |
| 23:24.35 | C4colo | oh |
| 23:24.37 | C4colo | wait |
| 23:24.43 | C4colo | how long the channel has been open ... |
| 23:24.54 | C4colo | yea, hangup the channel it will clear the CRD record |
| 23:25.06 | nnY_2 | hmm good point |
| 23:25.11 | C4colo | however I don't know if restarting asterisk would clean up the CDR, probably |
| 23:25.14 | C4colo | I would guess so |
| 23:25.22 | nnY_2 | any other way to force the channel down? |
| 23:25.38 | C4colo | but stupider things have happened than CDR not clearing ... for example, channels hanging open |
| 23:25.42 | C4colo | yea, reboot the system |
| 23:26.04 | C4colo | like I said, I have tried many many ways to clear those channels |
| 23:26.09 | C4colo | soft hangup does not do it |
| 23:26.22 | C4colo | however feel free to try, your results may vary |
| 23:27.39 | C4colo | a sip [purge|destroy|kill|bend|fold|spindle|mutilate] function is sorely needed in asterisk |
| 23:27.51 | nnY_2 | heh |
| 23:30.21 | nnY_2 | well crap |
| 23:30.26 | nnY_2 | longest cdr is 27 minutes |
| 23:30.34 | nnY_2 | so it didn't capture the specifics |
| 23:31.05 | nnY_2 | another annoyance |
| 23:31.06 | nnY_2 | <PROTECTED> |
| 23:31.06 | nnY_2 | <PROTECTED> |
| 23:31.13 | nnY_2 | over and over and over after a restart >< |
| 23:32.46 | viraptor | my g729 license doesn't show up in console (show g729), but it's listed in transcoding paths during the start - how can I verify whether it works or not? |
| 23:34.39 | C4colo | make a call from a g729 endpoint to a ulaw device/trunk |
| 23:34.44 | C4colo | if it transcodes, it work |
| 23:34.56 | C4colo | if it says 408 unaccptable here then it doesn't |
| 23:35.20 | viraptor | ok - how can I test it if I don't have any g729 devices nearby? |
| 23:35.56 | nnY_2 | can always try xlite, doesn't it support g729? |
| 23:36.22 | nnY_2 | crap nm |
| 23:36.27 | nnY_2 | don't see it in codecs |
| 23:37.31 | viraptor | I see "== Registered translator 'lintog729' from format slin to g729, cost 8" all right, but 'show g729' gives me only 'no such command' |
| 23:41.27 | *** join/#asterisk sack (n=sack@221.Red-81-34-35.dynamicIP.rima-tde.net) |
| 23:41.39 | *** join/#asterisk torrikft (i=torrikft@77.208.27.43) |
| 23:44.51 | viraptor | ehh... ok - I was loading 729 not digium's 729a |
| 23:45.17 | torrikft | hey viraptor |
| 23:45.18 | *** join/#asterisk outtolunc (n=me@c-67-170-211-86.hsd1.ca.comcast.net) |
| 23:55.49 | *** join/#asterisk LemensTS (n=matthew@adsl-70-238-187-205.dsl.stlsmo.sbcglobal.net) |
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