| 00:00.13 | outtolunc | mikealeonetti: as martha stated before, the context assigned to a zap channel is for when you pickup that line.. not when some inbound somewhere comes in |
| 00:00.14 | outtolunc | (unless it comes in that channel) |
| 00:00.14 | [TK]D-Fender | mikealeonetti: indeed strange. |
| 00:00.15 | Ward1983 | [TK]D-Fender, i cannot smell that |
| 00:00.36 | Ward1983 | [TK]D-Fender, in that case: how do i delete a user from the CLI ? |
| 00:00.52 | [TK]D-Fender | Ward1983: they're in users.conf |
| 00:01.09 | mikealeonetti | outtolunc: the context says that it should be answered automatically |
| 00:01.11 | Ward1983 | [TK]D-Fender, so there are no commands to do this? |
| 00:01.25 | [TK]D-Fender | Ward1983: No, * is configured by TEXT FILES. get used to it. |
| 00:01.26 | Ward1983 | [TK]D-Fender, by that i mean in asterisk cli |
| 00:01.37 | mikealeonetti | plus asterisk doesn't even know when the line is ringing |
| 00:01.45 | Ward1983 | [TK]D-Fender, you are arrogant and i don't want any help froom you |
| 00:01.51 | [TK]D-Fender | mikealeonetti: pastebin your zaptel.conf. Just for a sanity check. |
| 00:02.01 | mikealeonetti | sure |
| 00:02.33 | [TK]D-Fender | Ward1983: I answered your question rather directly. This is not like a cisco router for example where you configure things from a CLI 1 command at a time. Its all fed from config files and parsed. |
| 00:02.35 | mikealeonetti | http://pastebin.com/d2a8bd4d8 <-- /etc/zaptel.conf |
| 00:02.46 | Ward1983 | think i care about text files? you think everyone is a noob and you're the smartest person on earth, at least thats how you talk to me |
| 00:03.01 | mikealeonetti | imagine he was the smartest person on Earth |
| 00:03.02 | FinboySlick | [TK]D-Fender: The config itself hasn't changed from the previous working setup, but what do you mean by comprehensive backup everything in /etc/asterisk? I think some library got messed up by an update, but I wouldn't know which. I suspected sox, rolling back didn't work. Anyway, this wasn't to impose support on you, I'm grateful enough for the attention, was just wondering if you'd come across something like that before. |
| 00:03.05 | mikealeonetti | then you'd just sound stupid |
| 00:03.08 | Ward1983 | liek i never saw a text config before... |
| 00:03.14 | unpaidbill | hahah mike |
| 00:03.18 | [TK]D-Fender | Ward1983: You don't have to care about text files. Unfortunately thats what is required to configure *. |
| 00:03.27 | jaytee | liek I can't even spell like |
| 00:03.32 | Ward1983 | i know i was just asking IF there were commands |
| 00:03.33 | Ward1983 | thats all |
| 00:03.37 | Ward1983 | and you answer my in caps |
| 00:03.41 | Ward1983 | me |
| 00:03.44 | mikealeonetti | <3 |
| 00:03.49 | outtolunc | *breath* |
| 00:03.57 | [TK]D-Fender | mikealeonetti: Ok, your setup really does check out. Is there anything special on this line? Something that might suppress a ring? |
| 00:04.17 | Ward1983 | jaytee, nothing better to do? i guess |
| 00:04.19 | exothermc | Ward1983: Please do go tell someone who cares. |
| 00:04.37 | exothermc | [TK]D-Fender: Where can I find RTCP data? |
| 00:04.49 | Ward1983 | exothermc, i dont force you to read, plus you can ignore me |
| 00:04.52 | mikealeonetti | [TK]D-Fender: I can't imagine. It's a phone provided by cable and hooked into the cable modem. |
| 00:04.52 | jaytee | exothermc, were you looking for documentation about RTCP or actual data on the console |
| 00:04.54 | [TK]D-Fender | Ward1983: Ok, let go already. I've told you where to look. Go in and wipe out the offending entry and move on with things. |
| 00:05.04 | mikealeonetti | there are so special options enabled, at least nothing I changed |
| 00:05.22 | Ward1983 | [TK]D-Fender, i will im jsut shocked by the way you peopel treat me |
| 00:05.33 | [TK]D-Fender | mikealeonetti: lets see something else : pastebin "zap show channels" and then "zap show channel 3" |
| 00:05.42 | exothermc | jaytee: data on the console would be nice to play with, but more along the lines of historical data for a call. |
| 00:05.44 | mikealeonetti | sure |
| 00:05.57 | drmessano | Ward1983: I hope a tree falls on your cat |
| 00:06.02 | [TK]D-Fender | Ward1983: Stop puching. I made 1 dry comment and you're making a federal case out of it and raining it all over everybody. |
| 00:06.03 | jaytee | hahahahaa |
| 00:06.07 | [TK]D-Fender | pushing* |
| 00:06.14 | exothermc | Ward1983: I suggest you put a comment in writing and submit it to the complaint department. |
| 00:06.19 | unpaidbill | haha |
| 00:06.25 | jaytee | Ward1983, hey buddy, grow some thicker skin and come back later |
| 00:06.29 | [TK]D-Fender | Ad wow, feel the backlash. |
| 00:06.34 | [TK]D-Fender | and* |
| 00:06.36 | drmessano | Write it on a piece of paper, fold it 9 times, and .. . . crap, brb |
| 00:06.47 | [TK]D-Fender | marvels at how his typing slowly degrades.... |
| 00:06.59 | unpaidbill | it must be karma |
| 00:07.00 | mikealeonetti | http://pastebin.com/d5490e10f here yar |
| 00:07.05 | drmessano | ~karma |
| 00:07.05 | jbot | drmessano has karma of 1 |
| 00:07.09 | [TK]D-Fender | unpaidbill: My karma ran over your dgma :p |
| 00:07.09 | FinboySlick | is to blame, it's his despair rubbing off on everyone ;) |
| 00:07.11 | drmessano | ~karma Ward1983 |
| 00:07.11 | jbot | ward1983 has neutral karma |
| 00:07.16 | drmessano | ~karma -20 Ward1983 |
| 00:07.16 | jbot | -20 ward1983 has neutral karma |
| 00:07.30 | drmessano | ~karma Ward1983 -20 |
| 00:07.30 | jbot | ward1983 -20 has neutral karma |
| 00:07.34 | drmessano | crapola |
| 00:07.39 | unpaidbill | what have you done |
| 00:07.55 | drmessano | doesn't know how the karma works on the bot.. I truly fail |
| 00:07.59 | exothermc | so anyone have any idea how to get some RTCP stats out of asterisk? |
| 00:08.00 | [TK]D-Fender | ~karma [TK]D-Fender |
| 00:08.00 | jbot | [tk]d-fender has karma of 9 |
| 00:08.03 | [TK]D-Fender | :D |
| 00:08.09 | jaytee | watches as the fabric of the universe becomes unraveled. |
| 00:08.10 | unpaidbill | well arent you just peachy |
| 00:08.34 | [TK]D-Fender | begins to crochet a sweater out of loose strands of universe... |
| 00:08.35 | coppice | ~karma drmessano |
| 00:08.35 | jbot | drmessano has karma of 1 |
| 00:08.37 | jaytee | exothermc, stats no |
| 00:08.38 | drmessano | ~karma buddha |
| 00:08.38 | jbot | buddha has neutral karma |
| 00:08.42 | coppice | overrated |
| 00:08.51 | Ward1983 | the offtopicness :s |
| 00:08.51 | *** join/#asterisk guilherme-jorge (n=guilherm@189.100.240.34) |
| 00:08.56 | FinboySlick | New meaning to string theory. |
| 00:09.03 | exothermc | jaytee: but......? |
| 00:09.06 | drmessano | ~karma gandhi |
| 00:09.06 | jbot | gandhi has neutral karma |
| 00:09.18 | drmessano | Irony |
| 00:09.32 | [TK]D-Fender | drmessano: Don't iron my sweater.... it'll shrink! |
| 00:09.35 | drmessano | ~karma karma |
| 00:09.35 | jbot | karma has karma of 6 |
| 00:09.39 | guilherme-jorge | hello all, someone konws if its possible to call skype throught asterisk using some API or something? |
| 00:09.50 | drmessano | ~karma comcast |
| 00:09.50 | jbot | comcast has karma of -4 |
| 00:09.55 | mikealeonetti | [TK]D-Fender: shows the same thing even when it's supposed to be ringing |
| 00:10.32 | [TK]D-Fender | ~karmakarma |
| 00:10.32 | jbot | Karma Chameleon! |
| 00:10.40 | [TK]D-Fender | :p |
| 00:10.40 | Ward1983 | hmmm i got some questions but i've grown scared to ask them (good job!) |
| 00:10.42 | mikealeonetti | kik |
| 00:10.45 | coppice | comcast is highly overrated |
| 00:11.04 | unpaidbill | ~karma unpaidbill |
| 00:11.04 | jbot | unpaidbill has neutral karma |
| 00:11.09 | unpaidbill | hey at least im not n the hole! |
| 00:11.17 | mikealeonetti | too bad |
| 00:11.18 | *** join/#asterisk craigk (n=craigk@58.174.150.119) |
| 00:11.19 | coppice | ~karma grey |
| 00:11.19 | jbot | grey has neutral karma |
| 00:11.22 | [TK]D-Fender | Ward1983: If you can't ask, its your loss. |
| 00:11.22 | mikealeonetti | you write good songs down in the hole |
| 00:11.32 | drmessano | coppice: Who overrates comcast? |
| 00:11.44 | [TK]D-Fender | drmessano: Un-possible! |
| 00:11.53 | coppice | a karma of -4 seems to overrate it |
| 00:12.05 | drmessano | Ward1983: The passive whinging is unimpressive |
| 00:12.08 | unpaidbill | -666 ! devilservice! |
| 00:12.13 | Ward1983 | [TK]D-Fender, just to make sure: is this the official channel? |
| 00:12.19 | Ward1983 | i must have misread |
| 00:12.22 | drmessano | "I want so badly to ask something.. *sigh*" <---- EMO |
| 00:12.24 | [TK]D-Fender | mikealeonetti: can you put the cable modem on a splitter with a regular ohone & * and listen to it ring... just to be sure. |
| 00:12.42 | mikealeonetti | [TK]D-Fender: yes. lemme fetch a regular phone. |
| 00:12.44 | [TK]D-Fender | Ward1983: For GUI specific stuff, yes it is. For the "normal" stuff it is. |
| 00:12.48 | unpaidbill | ward, you gotta be tough to roll with phone dudes. no sensitive skin |
| 00:13.06 | unpaidbill | fender makes me cry every night, buti come back for more |
| 00:13.06 | Ward1983 | [TK]D-Fender, you mean not for GUI i presume |
| 00:13.09 | coppice | "I want to ask something so badly" |
| 00:13.11 | coppice | "Well you'll never see things asked worst than they are asked here" |
| 00:13.13 | drmessano | unpaidbill: Thats goes for IRC in general.. Crying on IRC is pathetic |
| 00:13.18 | [TK]D-Fender | ~emo |
| 00:13.19 | jbot | /wrists |
| 00:13.19 | unpaidbill | haha true |
| 00:13.34 | [TK]D-Fender | Ward1983: Ask away |
| 00:13.48 | drmessano | hangs his head down low and kicks a pebble with pent up frustration |
| 00:13.59 | unpaidbill | no dont! he'll just tell you to go away if you ask about the gui! |
| 00:14.02 | drmessano | Awww shucks mom, you always love the beaver more than me |
| 00:14.02 | unpaidbill | it's atrap! |
| 00:14.12 | [TK]D-Fender | drmessano: 1 mediocre step for passive-aggressivity! |
| 00:14.13 | drmessano | Ackbar? |
| 00:14.19 | Ward1983 | drmessano, i dont know if you noticed, retard, but you type more then me... |
| 00:14.25 | [TK]D-Fender | drNow with brine-shrimp! |
| 00:14.28 | Ward1983 | i dont want to ask about the damn gui :s |
| 00:14.35 | drmessano | Ward1983: WTF does that have to do with anything? |
| 00:14.48 | unpaidbill | im listening to 'because tonight belongs to lovers' |
| 00:14.49 | [TK]D-Fender | Ward1983: just spill it already! |
| 00:14.56 | Ward1983 | drmessano, it means you whine more then me, retard! |
| 00:14.57 | unpaidbill | it's not really working out for me. |
| 00:15.02 | Ward1983 | well if this is the official IRC channel then asterisk is the very first piece of software i don't want to use because of the people in its official IRC channel |
| 00:15.13 | Ward1983 | be fucking proud of it, idiots |
| 00:15.15 | drmessano | Ward1983: Keep whinging |
| 00:15.17 | unpaidbill | i hear callmanager is really good |
| 00:15.18 | *** part/#asterisk Ward1983 (n=ward@91.178.105.217) |
| 00:15.19 | jaytee | Ward1983, you should quit using the term retard, it's not nice |
| 00:15.19 | [TK]D-Fender | Ward1983: Congratulations on your newfound validation. Now are you going to stop and jsut ask your question? |
| 00:15.35 | [TK]D-Fender | unpaidbill: And what else do your Rice Crispies say to you? ;) |
| 00:15.43 | unpaidbill | they told me someone pissed in them! |
| 00:15.47 | drmessano | 1983.. thats.. 25ish |
| 00:15.58 | drmessano | How sad.. Id expect full grown skin by then |
| 00:15.59 | [TK]D-Fender | unpaidbill: No.... those were your Corn Flakes ;) |
| 00:16.17 | unpaidbill | ah well, all those kellogs cereals look the same to me |
| 00:16.17 | jaytee | I remember when Eisenhower was President |
| 00:16.19 | drmessano | YOU GUYS MAKE ME WANT TO USE SKYPE! |
| 00:16.31 | unpaidbill | i dont even know who eisenhower is |
| 00:16.41 | jaytee | drmessano, check this site out www.pfingo.com |
| 00:16.53 | drmessano | Ohhh.. pfingo.. sounds cool |
| 00:17.16 | drmessano | ZOMG My browsar froze |
| 00:17.17 | *** join/#asterisk LiNeTuX|Home (n=LiNeTuX@171.116.8.67.cfl.res.rr.com) |
| 00:17.19 | mikealeonetti | [TK]D-Fender: rang |
| 00:17.21 | drmessano | Hack! Hack! |
| 00:17.39 | FinboySlick | Hmmm, which part of asterisk actually handles the bridging between zap and sip/iax? |
| 00:17.40 | drmessano | Wait, no.. shitty site |
| 00:17.46 | [TK]D-Fender | mikealeonetti: Ok, that IS freaky. How long have you had the card? |
| 00:17.48 | drmessano | FinboySlick: Asterisk |
| 00:18.03 | mikealeonetti | mikealeonetti: six months roughly |
| 00:18.04 | [TK]D-Fender | FinboySlick: channel.c IIRC |
| 00:18.12 | mikealeonetti | should be a Digium TDM400 |
| 00:18.13 | [TK]D-Fender | mikealeonetti: Stop talking to yourself! |
| 00:18.15 | [TK]D-Fender | :p |
| 00:18.20 | *** part/#asterisk exothermc (n=miles@74.85.89.146) |
| 00:18.21 | mikealeonetti | lol |
| 00:18.31 | [TK]D-Fender | mikealeonetti: I'd say call up Digium support and get ready to walk them through all of this. This IS a wierd one. |
| 00:18.39 | nick125 | mikealeonetti: Yeah, we're getting really concerned about you there. |
| 00:18.45 | coppice | jaytee: The opening screen was as much as I could take |
| 00:19.12 | drmessano | I fingo'ed you out |
| 00:19.16 | drmessano | You are your plan |
| 00:19.22 | drmessano | s/are/and/ |
| 00:19.25 | mikealeonetti | [TK]D-Fender: would digium support charge me forever? |
| 00:19.34 | jaytee | coppice, you should have been here .....um....saturday or sunday. it was hilarious. I wish I had the logs to this chat from the weekend. |
| 00:19.49 | [TK]D-Fender | mikealeonetti: Not if the carkd is under warranty |
| 00:20.06 | drmessano | jaytee: If you don't stop pfingo'ing me, I am gonna compile |
| 00:20.10 | jaytee | and sometimes even if it isn't |
| 00:20.49 | mikealeonetti | [TK]D-Fender: hahaha! I fooled myself! I bought it a year ago. Time ... really flies. |
| 00:21.02 | jaytee | mikealeonetti, were you having fun? |
| 00:21.07 | nick125 | mikealeonetti: ouch, sounds like you're pfingoed there. |
| 00:21.16 | mikealeonetti | yes |
| 00:21.25 | mikealeonetti | I did have a power outage |
| 00:21.28 | jaytee | lucky |
| 00:22.21 | mikealeonetti | well |
| 00:22.27 | mikealeonetti | I suppose there is always crying myself to sleep at night |
| 00:22.35 | [TK]D-Fender | mikealeonetti: I might try asking on the mailing list. Be VERY thorough in your info posting. Do max debug tests. And NOW might be a goot time to run through "zttest" |
| 00:22.49 | [TK]D-Fender | mikealeonetti: OH.. perhaps the module fried? |
| 00:22.54 | mikealeonetti | it has two modules |
| 00:23.01 | mikealeonetti | and both aren't working |
| 00:24.01 | mikealeonetti | could my cable company be sending some ridiculous signal for ringing instead of what asterisk might expect? |
| 00:24.02 | jaytee | I'm reading the documentation for Office Communications Server 2007 Speech Server and thinking......."hmmm, if I mix these sleeping pills with those muscle relaxers and a fifth of Jack Daniels......." |
| 00:24.09 | FinboySlick | Okay, so logically (as was pointed out), the fact that asterisk can talk and record from zap just fine strongly suggests that my zaptel is working okay... That added to the fact that asterisk talks/records to my sip stuff means my sip is also okay. So (again as pointed out) this is some sort of codec issue. Or would this happen in the raw audio inbetween the decoding on one end and encoding on the other end? |
| 00:25.36 | [TK]D-Fender | FinboySlick: Screams codec issue. Now stop wastebing time and PASTEBIN IT <---- |
| 00:25.47 | [TK]D-Fender | mikealeonetti: No, if it rings a normal phone, the card should be fine. |
| 00:26.12 | Qwell | wait, what, phone rings, line doesn't...huh? |
| 00:26.24 | Qwell | surely you're referring to two separate ports? |
| 00:26.33 | FinboySlick | [TK]D-Fender: Pastebin the sip.conf and zapata.conf? I must have missed your previous answer to what I should actually pastebin. |
| 00:26.34 | mikealeonetti | no.... |
| 00:27.13 | [TK]D-Fender | FinboySlick: SIP configs, SIP debug & CLI for a failed call. What else do you think I'd be asking for? Show em evidence of the damn murder! |
| 00:27.27 | Qwell | then I'm confused. how does a phone come into play? |
| 00:27.39 | [TK]D-Fender | Qwell: Can call out on Zap, max debug deoes register voltage change. |
| 00:27.39 | FinboySlick | [TK]D-Fender: There is none. As far as asterisk is concerned, the call works just fine. |
| 00:27.40 | unpaidbill | hrm, this REFER should work right? http://pastebin.com/m6ba2d410 ... it's giving me a 603 declined. 9905 is a MeetMe extension with no password set, it should just ring those extensions and dump the calls into the conference, right? i must be missing some configuration option in the sip user? |
| 00:27.47 | Qwell | deoes? |
| 00:27.56 | Qwell | oh |
| 00:27.58 | [TK]D-Fender | FinboySlick: ...... PASTEBIN IT. |
| 00:28.23 | jaytee | unpaidbill, canreinvite=no? |
| 00:28.40 | Qwell | mikealeonetti: how old is the card? |
| 00:29.00 | mikealeonetti | Qwell: a year and amonth |
| 00:29.07 | Qwell | yeah, call support |
| 00:29.07 | mikealeonetti | modules are the same |
| 00:29.16 | Qwell | (tomorrow) |
| 00:29.19 | mikealeonetti | from digium |
| 00:29.24 | mikealeonetti | or the people I bought it from? |
| 00:29.28 | Qwell | Digium |
| 00:29.39 | unpaidbill | canreinvite is yes |
| 00:29.40 | Qwell | just jot down the serial or something |
| 00:29.40 | mikealeonetti | I hope I don't sound dumb on the phone |
| 00:29.45 | mikealeonetti | alrighty |
| 00:30.01 | *** join/#asterisk ChrisN (n=ChrisN@tz.tztalk.com) |
| 00:30.07 | jaytee | unpaidbill, set it to no and do a sip reload and try again |
| 00:30.12 | mikealeonetti | I'm off to dinner then |
| 00:30.33 | mikealeonetti | thanks everybody! |
| 00:31.00 | [TK]D-Fender | mikealeonetti: You're welcome and good luck. |
| 00:31.05 | mikealeonetti | thanks! |
| 00:31.08 | *** part/#asterisk mikealeonetti (n=mike@ool-457b9930.dyn.optonline.net) |
| 00:31.37 | FinboySlick | [TK]D-Fender: Okay, okay. Sheesh.... ;) I'm working on it. Now in the meantime, sip to iax works fine. (and iax to sip) |
| 00:32.32 | unpaidbill | ok... |
| 00:34.50 | unpaidbill | still got declined, 603 |
| 00:34.50 | unpaidbill | hrm |
| 00:36.25 | FinboySlick | [TK]D-Fender: How debug do you want the debug? Or is 'set verbose 10' enough? I'm not entirely sure how to get real debugging output. |
| 00:36.36 | unpaidbill | http://pastebin.com/ma7a4ced yeah no go... i dunno |
| 00:38.11 | jaytee | unpaidbill, pastebin your sip.conf |
| 00:40.08 | [TK]D-Fender | FinboySlick: I said verbose 10, SIP DEBUG |
| 00:40.41 | ChrisN | I have two servers running Asterisk 1.4.19, s1 and s2. I can connect to a SIP extension on s2 which uses "insecure=port,invite" from a softphone. When I use "dial(sip/ext@s2)" on s1 to connect to s2, s1 gets a "chan_sip.c:12253 handle_response_invite: Failed to authenticate on INVITE to s1" error. SIP debug shows that s2 is expecting proxy authentication. How should I fix this? |
| 00:40.57 | unpaidbill | http://pastebin.com/m330ddbf6 |
| 00:42.08 | jaytee | unpaidbill, so 9999 is the extension that can't join meetme? |
| 00:42.57 | unpaidbill | they can all join, 9999 is just sending out the invite (refer i guess it's called in the logs) to the 3 phones who should ring to join it |
| 00:43.29 | jaytee | and they all register to the same * server? |
| 00:43.34 | unpaidbill | yes |
| 00:43.41 | jaytee | hrm |
| 00:43.50 | unpaidbill | i can do a Refer-To: <sip:8805@10.1.1.8:5060> |
| 00:44.07 | unpaidbill | but when i add multiple extensions separated by semicolon that's when i get the declined hammer |
| 00:44.22 | jaytee | use an & |
| 00:44.26 | unpaidbill | right in the face bam bam bam |
| 00:44.59 | unpaidbill | i know that works for dials... im actually sending sip headers though.. i will try |
| 00:45.02 | unpaidbill | if this works i kiss you |
| 00:45.17 | jaytee | oh no!!! none of that! I've only had 3 potatoes |
| 00:45.32 | unpaidbill | i hope they were fermented |
| 00:45.40 | jaytee | nope |
| 00:45.49 | unpaidbill | you may vomit twhen i'm finished then |
| 00:46.01 | jaytee | I prefer my alcohol in liquid not solid form |
| 00:47.58 | *** join/#asterisk snapple42 (n=snapple4@CPE00184d8cadc3-CM0013718c1e2c.cpe.net.cable.rogers.com) |
| 00:48.24 | unpaidbill | ok restarting services..about to try this with & |
| 00:48.39 | FinboySlick | [TK]D-Fender: http://pastebin.com/de40980f I've likely forgotten something in there, but I'm sure you'll remind me. |
| 00:50.46 | unpaidbill | hmm, it did not like & at all. |
| 00:50.47 | unpaidbill | hehe |
| 00:50.49 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-d6fb60ac5e0a5055) |
| 00:51.09 | [TK]D-Fender | FinboySlick: ok, at linux CLI do "rmmod ztdummy. stop *. Re-init zaptel, restart *, retest |
| 00:51.48 | FinboySlick | [TK]D-Fender: I never had a ztdummy loaded. |
| 00:52.07 | unpaidbill | disappointing. |
| 00:52.15 | unpaidbill | oh well maybe i'll just give up for a few d ays |
| 00:52.20 | [TK]D-Fender | FinboySlick: sometimes it can linger |
| 00:52.21 | unpaidbill | then suddenly it will hit me |
| 00:52.36 | jaytee | unpaidbill, you're using Refer-To: ? |
| 00:52.41 | unpaidbill | yeah |
| 00:52.43 | FinboySlick | [TK]D-Fender: I have wctdm24xxp and zaptel, that's all. |
| 00:52.57 | [TK]D-Fender | FinboySlick: Nuke it just to be sure |
| 00:53.26 | unpaidbill | i'm trying to make some stupid auto conferencing bs work, but i'm bastardizing someone elses work that uses some java sip client to initiate calls |
| 00:53.38 | unpaidbill | i will probably end up just rewriting it to use the manager interface, i dunno |
| 00:53.47 | unpaidbill | i'd rather take the easy way out though |
| 00:53.55 | FinboySlick | [TK]D-Fender: Tried that. Same result. Also tried loading wctdm24xxp with vpmsupport=0 to make sure my echo canceller wasn't the one gone bezerk. |
| 00:54.18 | unpaidbill | OH i know why ampersand messed up so bad |
| 00:54.30 | unpaidbill | i passed a friggen & to a web address.. duh, let me fix |
| 00:54.42 | [TK]D-Fender | FinboySlick: Good call... ok, next, check the firewall on your server. "iptables --list". |
| 00:54.51 | unpaidbill | %26 baby! |
| 00:54.58 | [TK]D-Fender | FinboySlick: I do presume that your phone is on the local LAN with your * as it appears. |
| 00:55.01 | jaytee | unpaidbill, one other thing to try for the refer although it may make no difference is to add insecure=port,invite for each user in sip.conf |
| 00:55.07 | FinboySlick | [TK]D-Fender: All fine and dandy for the firewall. |
| 00:55.20 | jaytee | I've only had to use refer for click to dial |
| 00:55.43 | unpaidbill | yeah that's what this is essentially, just with multiple users |
| 00:55.48 | jaytee | not conferencing, for that I use the Page application in a macro |
| 00:55.53 | FinboySlick | [TK]D-Fender: And yes, they're practically on the same switch. |
| 00:55.57 | unpaidbill | i can do the click to dial with 1 person fine, connects them to my ext |
| 00:56.13 | FinboySlick | Actually, they *are* on the same switch in this case. |
| 00:56.21 | [TK]D-Fender | FinboySlick: your firewall is CLEAR? |
| 00:56.33 | unpaidbill | well, im gonna try this again with & that work, then give your insecure line a shot |
| 00:56.38 | unpaidbill | then move on to better things |
| 00:56.39 | FinboySlick | [TK]D-Fender: Empty with policy at ACCEPT on all filter tables. |
| 00:56.43 | unpaidbill | thanks for the help! |
| 00:56.51 | unpaidbill | i hope this works... fingers are crossed |
| 00:56.52 | jaytee | np, hope you get it working |
| 00:56.57 | [TK]D-Fender | FinboySlick: Ok, running out of possibilities here... |
| 00:57.11 | [TK]D-Fender | FinboySlick: So sip>sip = OK. 1 Zap channel out another = fine as well? |
| 00:57.17 | unpaidbill | oh i will, it's just going to be a painful process of figuring out how to connect to the manager interface with java heh |
| 00:57.20 | unpaidbill | javascript! |
| 00:57.27 | unpaidbill | oh wait it will have to be with java.. fuck |
| 00:57.35 | jaytee | rails? |
| 00:57.50 | unpaidbill | yeah im just going to pass it off to a php script i think |
| 00:57.56 | FinboySlick | [TK]D-Fender: Might be hard to do zap to zap... I can try to mess with my extensions to manage that though. |
| 00:57.57 | unpaidbill | let that do the connection, screw java |
| 00:58.35 | jaytee | unpaidbill, this is * you're running right? not trixbox or some other derivative |
| 00:58.41 | unpaidbill | yeah just asterisk |
| 00:58.56 | unpaidbill | this script im using is some zimbra crapped out experimental thing |
| 00:59.04 | unpaidbill | for drag/drop conferences from my contacts list |
| 00:59.16 | unpaidbill | select 5 users, drag to 'conference' icon, win |
| 00:59.20 | unpaidbill | except in my case it's lose |
| 01:00.12 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-5f303c13fad497a5) |
| 01:00.41 | jaytee | check out the page application in *. You might figure out a way to pass the info to * to the Page app. |
| 01:01.38 | FinboySlick | [TK]D-Fender: Okay, I think I can comandeer two cell phones and manage a zap to zap link, I'll give you an update in about 15 minutes. |
| 01:02.16 | unpaidbill | i'll check that out |
| 01:02.16 | jaytee | hungry, be back later |
| 01:02.16 | unpaidbill | now to test this again with & working.. |
| 01:02.20 | jaytee | I only suggested & for the dialplan, I didn't know you were using java |
| 01:02.39 | unpaidbill | ahh, yeah it doesnt work |
| 01:02.55 | unpaidbill | failed sip transfer to non-existing extension! boo |
| 01:05.32 | jaytee | rollback to using the semi-colon. |
| 01:05.38 | unpaidbill | yeah i did |
| 01:05.59 | unpaidbill | im just going to make it post to a perl cgi that connects to the manager interface |
| 01:06.01 | unpaidbill | screw it |
| 01:06.17 | unpaidbill | at least it will keep me busy for a while |
| 01:06.35 | jaytee | when I have a refer call from Exchange Unified Messaging to another SIP client on my * box I have to have insecure=port,invite in sip.conf or the call fails with 603 |
| 01:06.47 | unpaidbill | oh yeah |
| 01:06.49 | unpaidbill | i need to try that |
| 01:06.58 | unpaidbill | on each client? |
| 01:07.03 | jaytee | but I'm routing calls from Exchange UM through sipX as a proxy. |
| 01:07.08 | unpaidbill | or just the one doing the refer |
| 01:07.09 | jaytee | yeah, on each client |
| 01:07.41 | *** join/#asterisk Gwayne (n=Gwayne@cm196.kappa245.maxonline.com.sg) |
| 01:08.33 | unpaidbill | nah same thing, 603 |
| 01:14.12 | FinboySlick | [TK]D-Fender: Okay... zap to zap works fine. |
| 01:14.29 | *** join/#asterisk mosty (n=mosty@ppp191-34.static.internode.on.net) |
| 01:14.31 | [TK]D-Fender | <PROTECTED> |
| 01:16.16 | FinboySlick | [TK]D-Fender: I'm usually pretty good at figuring stuff out and I'm totally baffled too. I may manage to get another TDM800 card tomorrow. And I have a sangoma too, but those are a pain to compile drivers for. Though from all I can see, the problem isn't the card itself. |
| 01:17.03 | [TK]D-Fender | FinboySlick: on the zap leve. make some dialplan to play a sound, record a message, and play ti back. |
| 01:17.08 | mosty | wanpipe is easy to compile |
| 01:18.46 | FinboySlick | [TK]D-Fender: That's roughly equivalent to leaving a voicemail and checking it, no? |
| 01:18.57 | [TK]D-Fender | FinboySlick: Yes |
| 01:19.18 | FinboySlick | [TK]D-Fender: That works on sip, zap and iax. |
| 01:19.25 | [TK]D-Fender | FinboySlick: :/ |
| 01:19.33 | [TK]D-Fender | FinboySlick: I'm stuck at this point... |
| 01:20.25 | FinboySlick | I haven't tried zap to iax yet. |
| 01:21.57 | [TK]D-Fender | FinboySlick: You just did this between a hardphone & *... that it fairly trustworthy. Now you say that SIP > * = fine, Zap > * = fine, and Zap > * > SIP = bad. I'd say call Digium support. |
| 01:22.09 | *** part/#asterisk martha (n=martha@bzq-179-135-226.static.bezeqint.net) |
| 01:23.02 | mosty | i want to allow toggling of call recording, but with some audio feedback so you can tell if a call is being recorded or not. is this possible with automon in features.conf, or will i need to use a custom applicationmap? |
| 01:23.14 | FinboySlick | [TK]D-Fender: Yeah, I did... They're stumped and won't support 1.2 anymore anyway. I'm working on an emergency upgrade to 1.4 now (it is much needed anyway), but my distro doesn't like it yet. |
| 01:24.50 | [TK]D-Fender | FinboySlick: Umm.... I didn't think 1.2 supported the TDM800.... |
| 01:26.03 | FinboySlick | [TK]D-Fender: Well, it worked fine for me 'till today's messup. I'm suspecting foul on something like a new version of glibc or what not. |
| 01:26.33 | FinboySlick | [TK]D-Fender: But it's sadly not messed up enough to give me any kind of error loading some library or even a coredump. |
| 01:26.42 | [TK]D-Fender | FinboySlick: well that has been linked with GSM issues, but nothing else I've heard of. 4.1 = trouble |
| 01:26.46 | FinboySlick | Heh, it's bad when you wish you had a coredump ;) |
| 01:27.20 | FinboySlick | 4.1? |
| 01:27.49 | unpaidbill | 2001 wants its freebsd back |
| 01:28.31 | *** join/#asterisk [cfdisk] (n=cfdisk@68-116-156-85.dhcp.ftwo.tx.charter.com) |
| 01:29.17 | [TK]D-Fender | OpenBSD : Keeping up with yesterday. |
| 01:29.30 | JT | that's pretty unfair |
| 01:29.37 | JT | openbsd is ahead of linux in many aspects |
| 01:29.39 | JT | but not all |
| 01:29.39 | [TK]D-Fender | NetBSD : Tonright we're gonna party like its 1999 |
| 01:30.14 | FinboySlick | [TK]D-Fender: You're so mean tonight ;) |
| 01:30.23 | *** join/#asterisk BeeBuu (n=beebuu@219.135.42.236) |
| 01:30.50 | asdx | Linux FTW |
| 01:31.36 | unpaidbill | haha |
| 01:31.43 | unpaidbill | i'll stick with sco |
| 01:31.47 | unpaidbill | at least it's secure |
| 01:31.57 | asdx | it doesn't have anything of sco |
| 01:32.27 | unpaidbill | sco 5.0.6, oh man, pkgadd, sysconfig, woooooooo |
| 01:32.48 | FinboySlick | knew sco had unpaid bills, but he didn't know unpaidbill had sco. |
| 01:32.56 | unpaidbill | haha |
| 01:33.44 | FinboySlick | blames SCO for a lot of people moving to windows instead of linux. |
| 01:33.46 | unpaidbill | luckilly not anymore.. last job i had i was admining sco boxes, an unfortunately large amount of them |
| 01:34.03 | unpaidbill | it was torture |
| 01:34.23 | outtolunc | i blame people who actually paid that blackmail <G> |
| 01:34.28 | asdx | why didn't you rm -rf / and put linux instead |
| 01:34.44 | unpaidbill | we started doing that eventually |
| 01:34.53 | asdx | cool |
| 01:34.54 | unpaidbill | but when you have 150 servers scattered around the country, it takes a while :/ |
| 01:35.08 | asdx | heh |
| 01:35.15 | unpaidbill | basically we'd wait for a server to shit the bed, or get so old that we had to upgrade it because it couldnt handle the load put on it |
| 01:35.27 | FinboySlick | if they were old enough to run sco, why didn't you replace them all with one new linux box? ;) |
| 01:35.48 | unpaidbill | the customers we had... you wouldnt believe |
| 01:35.52 | FinboySlick | That way you get the added benefit of a single point of failure! |
| 01:37.05 | unpaidbill | the only benefit to paying for sco was the support |
| 01:37.12 | unpaidbill | haha just kidding. there was no benefit. |
| 01:40.27 | *** join/#asterisk Kumbang (n=dsp@167.205.24.69) |
| 01:43.57 | asdx | unpaidbill: what do you server in all these servers, voip/asterisk stuff? |
| 01:44.23 | asdx | serve |
| 01:44.45 | unpaidbill | the sco boxes were running a proprietary app to manage mobile fleets |
| 01:44.52 | *** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194) |
| 01:44.53 | unpaidbill | they used digium cards and a custom PBX |
| 01:44.56 | unpaidbill | err, dialogic |
| 01:45.35 | asdx | hehe |
| 01:45.38 | unpaidbill | the software could be rewritten in php|perl, using asterisk and mysql and be 1000x better |
| 01:46.12 | JT | python ;) |
| 01:46.24 | asdx | ruby ;) |
| 01:46.29 | unpaidbill | i left there 4 years ago and at that point we had only been migrating to linux for about 1.5 years and just starting to use postgres instead of some crappy flatfile DB |
| 01:46.49 | unpaidbill | yeah whatever ruby python anything but C |
| 01:47.01 | unpaidbill | actually the newer interface used java on the web browser |
| 01:47.04 | unpaidbill | but only supported IE |
| 01:47.22 | unpaidbill | the old version used .... sco X |
| 01:47.44 | unpaidbill | you ahd people connecting in to the server with exceed and logging into an X session! |
| 01:47.58 | unpaidbill | it was pretty hideous |
| 01:50.33 | asdx | i did an asterisk installation for a company here... it does inbound/outbound, conference, ivr, call forwarding, etc |
| 01:50.53 | asdx | it's working quite well now |
| 01:51.12 | asdx | i'm waiting for 1.6 though xD |
| 01:51.12 | unpaidbill | yeah, i use it here |
| 01:51.19 | unpaidbill | what's 1.6 got |
| 01:51.22 | unpaidbill | t.38 native? |
| 01:51.39 | asdx | app_confbridge |
| 01:51.44 | asdx | soft based conferencing |
| 01:52.10 | unpaidbill | different from meetme? |
| 01:52.33 | asdx | looks like it has the same features but doesn't need zaptel/ztdummy |
| 01:52.52 | asdx | so it can work quite well on xen environments, etc, i think |
| 01:52.57 | unpaidbill | ah |
| 01:53.00 | asdx | without much trouble |
| 01:53.09 | unpaidbill | im about to test out a quad xeon with 4 xen asterisk servers |
| 01:53.14 | unpaidbill | well, in the next few weeks |
| 01:53.16 | asdx | cool |
| 01:53.40 | asdx | http://www.asterisk.org/node/48491 |
| 01:53.45 | asdx | that stuff is interesting too |
| 01:53.54 | unpaidbill | ooh multiple parking lots is nice |
| 01:54.01 | asdx | http://www.asterisk.org/node/48492 |
| 01:55.00 | unpaidbill | sweet |
| 01:57.45 | *** join/#asterisk Strom_C (n=strom@208.127.172.112) |
| 01:59.18 | *** part/#asterisk ecrist (n=ecrist@snipe.secure-computing.net) |
| 02:02.27 | *** join/#asterisk mwalling (n=mwalling@you.dontlike.us) |
| 02:06.56 | asdx | can you connect trough sockets in asterisk and control the entire thing trough it... from languages |
| 02:06.58 | unpaidbill | anyone know who that company is that was suing people for fax->email email->fax services because of some bullshit IP patent? yesterday someone got me all riled up talking about it |
| 02:07.01 | asdx | if not that would be cool to have |
| 02:07.25 | Sargun_screen | asdx: AGI! |
| 02:07.34 | Sargun_screen | er, AMI! |
| 02:07.39 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 02:07.40 | asdx | lol nice |
| 02:07.46 | Sargun_screen | anyways, voip-info.org |
| 02:07.51 | jaytee | I think russellb said earlier today that he'd added the softconferencing to the 1.4 trunk so it might be available now. |
| 02:08.26 | asdx | jaytee: app_confbridge? |
| 02:08.38 | Sargun_screen | they are an amazing resource that will allow yoyh great joy! |
| 02:08.40 | jaytee | yeah, I think russell said it was added to trunk |
| 02:08.49 | Sargun_screen | meetme! |
| 02:09.33 | unpaidbill | or can someone point me to the irc logs for this # |
| 02:09.36 | unpaidbill | for some reason i cant find them |
| 02:10.06 | unpaidbill | ~irclogs |
| 02:10.11 | unpaidbill | ~logs |
| 02:10.12 | jbot | [logs] apt/ibot/infobot/jbot/purl all log daily to http://ibot.rikers.org/<channelname>/ where channelname is html encoded ie: %23debian | lines that start with a space are not shown | some channels have stats at http://ibot.rikers.org/stats/<channelname>.html.gz |
| 02:10.20 | unpaidbill | awwww jbot you sexy bot |
| 02:10.27 | *** join/#asterisk bintut (n=bintut@222.127.192.118) |
| 02:10.48 | bintut | waves to all.. |
| 02:11.46 | bintut | i'm confused with the definition of the agent feature in asterisk.. anyone can explain it to me in detail? thanks.. |
| 02:11.56 | asdx | does the manager interface works with sockets? |
| 02:12.00 | unpaidbill | catch curve |
| 02:12.02 | unpaidbill | thank you jbot. |
| 02:12.02 | jbot | unpaidbill: bitte |
| 02:12.03 | asdx | AMI/AGI |
| 02:12.13 | jaytee | [TK]D-Fender, ManxPower beat you by 10 lines of text in the channel stats |
| 02:13.28 | [TK]D-Fender | wow, I'm nearly relevent! |
| 02:14.05 | jaytee | more than relevant, you're #2 |
| 02:14.38 | jaytee | which is odd because you're here from like 8 am ET to 11 pm or later most days |
| 02:15.26 | jaytee | and you're almost always in here helping people for free, you arrogant so and so! :-) |
| 02:18.50 | jaytee | kinda pissed me off when Ward1983 was going off on you earlier, if I'd been an op in this channel I would have kicked him. |
| 02:19.58 | lmadsen | wishes he was here :) |
| 02:20.10 | jaytee | who, Ward? |
| 02:20.25 | jaytee | he was being a total douche |
| 02:20.37 | FinboySlick | [TK]D-Fender: Don't worry, I still love you. |
| 02:20.44 | jaytee | and TK kept trying to help him anyway even with the insults. |
| 02:20.44 | asdx | O-o |
| 02:20.46 | FinboySlick | But ManxPower is better. |
| 02:22.16 | ChrisN | Any way to have "dial(sip/ext@somewhere) not include my username for authentication? I happen to have the same username as one on the destination server but I want to connect using the account in sip.conf which doesn't require authentication. |
| 02:22.17 | [TK]D-Fender | lmadsen: Nah, if he was kick-worthy, I'd have done it :) |
| 02:22.44 | [TK]D-Fender | lmadsen: Caustic, but not a total loss. |
| 02:22.49 | jaytee | [TK]D-Fender, I'm amazed at your patience with idiots like that |
| 02:23.31 | lmadsen | I meant i wish I was hear to enjoy reading it :) |
| 02:23.55 | FinboySlick | is pretty glad for that patience with idiots. |
| 02:24.09 | unpaidbill | anyone have a cisco 7935 working with chan_sccp |
| 02:24.23 | unpaidbill | mine locks up on registration every god damn time |
| 02:24.28 | unpaidbill | with every version of the firmware |
| 02:24.28 | jaytee | didn't russellb say this morning that the latest 1.4 trunk had the new app_confbridge in it |
| 02:24.40 | lmadsen | I doubt it |
| 02:24.47 | lmadsen | 1.4 doesn't get new features |
| 02:24.52 | lmadsen | unless it fixes a bug |
| 02:25.00 | jaytee | oh, so only 1.6. I must have misunderstood |
| 02:25.10 | lmadsen | right |
| 02:26.16 | jaytee | I can hardly wait till 1.6 is no longer beta so I can dump sipX as a proxy to Exchange UM. It works but not having to maintain that as well as * will be nice. |
| 02:26.43 | *** join/#asterisk SteveTotaro (n=Administ@pool-70-17-234-155.balt.east.verizon.net) |
| 02:26.55 | lmadsen | you can use 1.6 as soon as you want -- the label means nothing -- if it works for you, it works for you |
| 02:27.08 | lmadsen | just because a label is applied to 1.6 doesn't make it immediately work for you |
| 02:27.30 | lmadsen | in fact, you'll be happier with 1.6 if you actually test ahead of time and report the bugs you run into |
| 02:28.16 | jaytee | I know and I've setup 1.6 as a test server but I'm just now getting done with other priorities so I should be able to focus on that over the next couple months. |
| 02:28.39 | asdx | lmadsen: is 1.6 already usable for production? |
| 02:29.00 | lmadsen | asdx: that is an unanswerable question |
| 02:29.31 | lmadsen | any checkout of asterisk from SVN is usable in production depending on what you're doing with it |
| 02:29.54 | lmadsen | some people don't think 1.4 is usable for production, other people like me have been using it for nearly 2 years |
| 02:30.05 | *** join/#asterisk mwalling (i=mwalling@you.dontlike.us) |
| 02:30.09 | lmadsen | depends what you're doing with it, and what your debugging skills entain |
| 02:30.09 | jaytee | 1.4 is rock solid for me |
| 02:30.13 | lmadsen | entail |
| 02:30.35 | asdx | s/usable/stable |
| 02:30.38 | asdx | i mean |
| 02:30.45 | jaytee | I'm on 1.4.15 and the only reason I haven't upgraded to a later 1.4 release is because I don't need to. |
| 02:30.48 | lmadsen | my original answer to you still stands |
| 02:31.04 | lmadsen | there is no such thing as "stable" |
| 02:31.16 | asdx | ok i'm going to upgrade to 1.6 and see if it works for me |
| 02:31.21 | mosty | ...when it comes to asterisk (hehe) |
| 02:31.22 | lmadsen | that sounds like a bad idea |
| 02:31.33 | lmadsen | mosty: when it comes to any software |
| 02:31.49 | mosty | lmadsen, tell that to donald knuth |
| 02:31.53 | asdx | lmadsen: bad idea, why |
| 02:31.59 | lmadsen | asdx: make sure you test on a development box extensively -- do NOT upgrade production servers |
| 02:32.00 | *** join/#asterisk tobias (n=tobias@user-0ce2hvs.cable.mindspring.com) |
| 02:32.04 | mosty | asdx, you have it backwards. see if it works for you, and then upgrade |
| 02:32.22 | asdx | ok |
| 02:32.30 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 02:32.32 | asdx | 1.4.21 is working great for us |
| 02:33.07 | lmadsen | I NEVER upgrade a working box -- I always take a 2nd box, install a newer version of software, and utilize the existing configuration and test extensively |
| 02:33.50 | jaytee | lmadsen, you have a valid point. I've been working in various IT jobs supporting Windows Server and Windows clients for over 15 years now and every release of Windows in shrink wrap I still consider beta until at least the first service pack. |
| 02:34.15 | jaytee | lmadsen, that's the approach I'm taking with 1.6 |
| 02:34.17 | lmadsen | I don't believe in stable software -- I believe in 'stable enough that it works for me without my interaction very often' |
| 02:35.08 | jaytee | nothing is 100% perfect and there always those unusual configurations that reveal things that didn't crop up in beta testing...sometimes months or even years later. |
| 02:35.25 | lmadsen | exactly |
| 02:35.31 | FinboySlick | lmadsen: Today I wish I had your discipline about upgrading. |
| 02:35.41 | lmadsen | FinboySlick: you'll learn quickly enough |
| 02:36.04 | FinboySlick | lmadsen: Heh... I wish. |
| 02:36.20 | jaytee | FinboySlick, touch the stove burner when it's on enough times and it becomes clear to you sooner or later that it's not a good idea |
| 02:36.41 | lmadsen | NEVER EVER EVER upgrade a working system |
| 02:36.42 | drmessano | Wait until you screw up something with an untested upgrade and almost lose a job |
| 02:36.45 | lmadsen | ESPECIALLY a phone system |
| 02:36.56 | lmadsen | drmessano: s/almost// |
| 02:37.01 | drmessano | lol |
| 02:37.05 | drmessano | that too |
| 02:37.29 | FinboySlick | I sadly always manage to fix stuff way too quick to remember just how much misery I was in *before* it got fixed. |
| 02:37.30 | lmadsen | I've never lost my job -- exactly because I learned not to upgrade production systems without extensive testing |
| 02:37.41 | jaytee | yeah, because the users (customers) always take their phones for granted and when they pick up the handset and don't hear dialtone all hell breaks loose. |
| 02:37.41 | FinboySlick | But tonight is one of those moments where I'm in misery. |
| 02:37.43 | lmadsen | and it shocks me to see how many people don't understand that |
| 02:38.19 | FinboySlick | lmadsen: Very true. |
| 02:39.02 | asdx | lmadsen: doesn't newer version of X software fixes bugs that could make the production servers more "stable" ? |
| 02:39.11 | asdx | well it could introduce more bugs too xD |
| 02:39.25 | lmadsen | exactly |
| 02:39.31 | lmadsen | TEST TEST TEST TEST TEST TEST |
| 02:39.50 | lmadsen | for some reason people think just upgrading blindly is a good idea |
| 02:40.23 | jaytee | my boss is one of those "early adopter zealots" who wants to upgrade right when something comes out. We're an IT shop of 6 people and we don't have near enough time and resources to test thoroughly. |
| 02:40.59 | *** join/#asterisk guilherme-jorge (n=guilherm@189.100.240.34) |
| 02:41.02 | lmadsen | sometimes you have to save bosses from themselves |
| 02:41.09 | jaytee | it drives me crazy because almost always there is no driving need to upgrade. |
| 02:41.15 | asdx | bosses suck |
| 02:41.32 | lmadsen | I have been very firm with "superiors" when they are in the wrong :) |
| 02:41.41 | jaytee | lmadsen, yeah and it usually comes with alot of arguing and him being pissed off that he doesn't get his way. |
| 02:42.08 | guilherme-jorge | Is it possible to call some skype contact using asterisk? For example, I want to call DID and then talk to my skype contacts. Is there a way? |
| 02:42.13 | lmadsen | that's when I change jobs :) |
| 02:42.26 | lmadsen | guilherme-jorge: check google -- there is a way, but it's not elegant |
| 02:42.32 | lmadsen | skype is a closed protocole |
| 02:42.36 | lmadsen | protocol* |
| 02:42.37 | jaytee | but I tell him, "Look, we can upgrade if you want and I'll be happy to help you pack your pictures and painting in your car when you no longer have a job since that'll be the end result" |
| 02:42.53 | guilherme-jorge | lmadsen: Ok. Thanks |
| 02:44.11 | jaytee | wow, all this time I thought Skype was a virus or a trojan, not a protocol. I really need to get caught up on the technology :-) |
| 02:45.25 | asdx | is the asterisk book going to be updated after 1.6 is released? |
| 02:45.57 | jaytee | ask lmadsen |
| 02:46.02 | lmadsen | that's the plan |
| 02:46.06 | lmadsen | as far as I know |
| 02:46.07 | lmadsen | :) |
| 02:46.08 | drmessano | Oh hell no |
| 02:46.12 | asdx | cool |
| 02:46.13 | drmessano | I just got my 1.4 book |
| 02:46.20 | drmessano | I WANT AN ADDENDUM |
| 02:46.21 | lmadsen | don't worry... you got at least a year |
| 02:46.28 | lmadsen | it takes a long time to update a book |
| 02:46.35 | jaytee | gee, Leif you might make enough off the book sales to get your Chevy Metro a tuneup! |
| 02:46.38 | drmessano | 3 pages to staple in the back for 1.6 changes |
| 02:46.53 | lmadsen | jaytee: might make enough to buy groceries for a couple months :) |
| 02:47.13 | drmessano | lmadsen: How much do you make off the folks that take the PDF to Kinkos? |
| 02:47.14 | lmadsen | jaytee: and luckily I don't have a car |
| 02:47.25 | lmadsen | drmessano: more money than selling the book :) |
| 02:47.32 | jaytee | that's cuz you need to write smutty romance novels if you want to make the NY Times Bestseller list and rake in the real dough. |
| 02:47.33 | drmessano | lol |
| 02:47.45 | lmadsen | I'm not that creative :) |
| 02:48.00 | lmadsen | I just live a soap opera... that seems to be enough for me :) |
| 02:48.18 | drmessano | "Dahdi lowered his dialplan on the bed.. " |
| 02:48.31 | asdx | lol |
| 02:48.36 | jaytee | you don't need to be, my sister used to read all those Harlequin romance books, I swear they were all ghost written by a computer. Just change the names and locations and title and bam, new book. |
| 02:49.07 | jaytee | is it DAHDI or DADHI? |
| 02:49.22 | drmessano | "_____ slowly _____ his _______, as _______ gasped in ______ at the sight of his __________" |
| 02:49.27 | drmessano | Thats how they work |
| 02:49.41 | drmessano | Madlibs |
| 02:49.49 | jaytee | and throw in a "Uuuunnnngggghhhh" and "aaaaaahhhhhh" here and there |
| 02:50.21 | jaytee | so does it stand for Digium Asterisk Hardware Device Interface? |
| 02:50.31 | Strom_C | _Pants_ slowly _wanged_ his _cheese_, as _dongles_ gasped in _toast_ at the sight of his _jellyfish_. |
| 02:50.50 | jaytee | now that's talent!!!! |
| 02:50.54 | *** join/#asterisk dxm (n=dxm@200.146.116.191.adsl.gvt.net.br) |
| 02:51.03 | jaytee | Strom, you missed your true calling! |
| 02:51.11 | drmessano | lol |
| 02:51.26 | Strom_C | I'm especially proud of the phrase "dongles gasped in toast" |
| 02:51.53 | Strom_C | I think I'm going to carve that into a polished granite block and erect it at a scenic overlook in Griffith Park |
| 02:52.03 | jaytee | the line from yesterday about slapping purple cock head against ass and legs had me nearly pissing my pants laughing |
| 02:52.08 | drmessano | Heh... he said "park" |
| 02:52.28 | Strom_C | heh, I forgot what exactly I said |
| 02:53.42 | drmessano | They need more action/adventure/romance novels based on the lives of PBX installers |
| 02:53.47 | drmessano | Thats an untapped market |
| 02:55.30 | Strom_C | "He whipped out his majestic impact tool and punched it into her connecting block." |
| 02:56.20 | jaytee | Strom_C> chapter 8 devolves into graphic gay porn with huge purple cocks slapping your ass and legs |
| 02:56.28 | Strom_C | hehehhe |
| 02:56.29 | jaytee | that was from yesterday afternoon |
| 02:57.35 | jaytee | when I said to MrNaz to read Chapter 5 of the book and then said something about in Chapter 7 Mike leaves Laura for another man and it gets really interesting. |
| 02:58.19 | drmessano | "My god, you have such a large beta" |
| 02:58.30 | jaytee | I loved the joke about the chocolate and the potatoes |
| 02:58.44 | Strom_C | "She admired the curves of his buttset" |
| 02:59.10 | Strom_C | "The ANAC didn't respond, so he dialed it harder and harder" |
| 02:59.16 | jaytee | "Did he say he has a hairy butt?" "No, he said he had a Harris buttset" |
| 03:00.55 | drmessano | "As he leaned over, she couldn't help but to quiver at the sight of his glistening sweat pooling at the top of hairy buttcrack, exposed by the soft tug of his worn jeans by the load of his heavy, expensive test equipment" |
| 03:01.07 | jaytee | I just realized that once I switch everyone from the Nortel 11C over to Asterisk I'll be able to retire my punchdown tool |
| 03:01.35 | MrNaz | morning jaytee |
| 03:01.50 | jaytee | my MDF looks like it was wired by Ray Charles after a night of heavy drinking |
| 03:01.57 | *** join/#asterisk Entranced (n=Entrance@mobile-166-214-038-029.mycingular.net) |
| 03:02.03 | Strom_C | jaytee: why haven't you taken the time to clean it up? |
| 03:02.20 | MrNaz | yea i'm reading the book, chapter 7 gets really hot and steamy... previously i was thinking the book was ok for kids, but now i think it deserves an MA rating |
| 03:02.55 | jaytee | Strom, gee whiz. I inherited it. The mess was the work of then Ameritech that got gobbled up by SBC that got gobbled up by AT&T |
| 03:03.04 | MrNaz | jaytee i ripped apart the mdf in my house the other night out of total frustration with the obscene wiring job... i then ran around the house with a cable tester and labelling machine hehe |
| 03:03.17 | Strom_C | jaytee: no no no no and no. |
| 03:03.21 | jaytee | it would take me at least a solid week |
| 03:03.24 | Strom_C | ABD ourchased AT&T and then renamed itself. |
| 03:03.26 | drmessano | "Her heart was pounding as his jelly donut covered fingers clumsily pounded the buttons on his glowing buttset" |
| 03:03.27 | Strom_C | er |
| 03:03.32 | Strom_C | SBC purchased AT&T and then renamed itself. |
| 03:03.48 | jaytee | ok |
| 03:04.00 | jaytee | it was merger mania there |
| 03:04.25 | drmessano | AT&T != AT&T, AT&T is SBC+ everyone they bought |
| 03:04.35 | jaytee | I've got over 800 pairs in a mix of 66 and 110 blocks |
| 03:05.35 | jaytee | and most of that runs out of the MDF and into a pedestal outside with harmonica splices and runs underground throughout the campus. Most of it is in piss poor shape. |
| 03:05.49 | drmessano | I need to do a "Guide to industry acronyms" |
| 03:06.04 | drmessano | CPE = Customer Promised Everything |
| 03:06.56 | jaytee | have you ever worked on a Nortel Meridian system? |
| 03:07.03 | drmessano | MDF = Mechanically Dense Fuckup |
| 03:07.08 | drmessano | Nope |
| 03:07.18 | jaytee | talk about acronyms, sheesh |
| 03:07.36 | jaytee | SCPW=Station Control Pass Word |
| 03:08.01 | jaytee | DSC=Distant Steering Code |
| 03:08.11 | drmessano | DOCSIS = Dumb Old Comcast Service Increasingly Sucks |
| 03:08.25 | jaytee | it goes on forever. 12 manuals with tons and tons and tons of acronyms |
| 03:09.08 | Strom_C | jaytee: I've got a client with a Norstar |
| 03:09.28 | drmessano | SLC-96 Shorted Line Container x 96 |
| 03:10.45 | jaytee | all my outbound calls now from Nortel route through my * box with a TE212 card. I've had ZERO problems with it handling the bridging between PRI channels. It runs like a champ and when I asked several people whom I knew made some long duration calls from looking in the CDR records if they noticed anything 3 of them said it sounded clearer than it used to :-) |
| 03:10.49 | Strom_C | drmessano: and i'm sure you know what GTE stands for |
| 03:10.53 | jaytee | Way to go, Digium!!! |
| 03:11.09 | Strom_C | or stood for, anyway |
| 03:11.12 | drmessano | Nope, go for it lol |
| 03:11.20 | Strom_C | the Great Telephone Experiment |
| 03:11.24 | drmessano | LOL |
| 03:11.26 | Strom_C | Guaranteed Trouble Everytime |
| 03:11.33 | Strom_C | Get Telephone Elsewhere |
| 03:11.39 | jaytee | MCI= Most Calls Interrupted |
| 03:11.42 | drmessano | NEC = Not Everyone Communicates |
| 03:12.18 | jaytee | or MCI Sprint = "We double bill everything because we know most of you aren't paying attention." |
| 03:12.31 | drmessano | MCI = Must Call Information |
| 03:13.06 | jaytee | at my last job they were trying to bill 5 bucks for a 1 minute fax call from Trenton, NJ to Indianapolis. |
| 03:13.20 | drmessano | MCI = My Circuits Intermittent |
| 03:13.36 | jaytee | we danced around for a few months and then I just dumped it on the Corporate Attorney |
| 03:14.28 | jaytee | Strom_C, those little Norstars aren't half bad for a key system but now that * is making major inroads I don't think they'll be around too much longer. |
| 03:14.52 | Strom_C | yeah, the norstar came with the building, so they said "Hey, can you make this work?" |
| 03:15.13 | jaytee | I hate the Norstar programming though |
| 03:15.31 | jaytee | almost as bad as a Toshiba DK Strata |
| 03:16.17 | Strom_C | it's clumsy but it works |
| 03:16.59 | jaytee | yeah, and Nortel does produce hardware with high MTBF |
| 03:17.26 | jaytee | except for their phonesets, especially the M3900 series |
| 03:19.05 | *** join/#asterisk CunningPike (n=arodgers@S010600095b33697f.vc.shawcable.net) |
| 03:19.26 | FinboySlick | Can I use zaptel 1.4 with asterisk 1.2? (don't ask, I'm desperate ;) |
| 03:19.33 | jaytee | that's where I think *'s strength really lies, it has incredible flexibility and it runs on commodity hardware. For a tenth of the cost of a Meridian switch you could have two heavyweight Quad Xeon * boxes with one as a standby spare. |
| 03:19.38 | Strom_C | FinboySlick: uhm, I really doubt it |
| 03:20.12 | FinboySlick | ztmonitor gives me some very strange stuff. |
| 03:20.20 | *** join/#asterisk jeev (i=jeev@naptime.net) |
| 03:20.57 | mosty | FinboySlick, yes you can |
| 03:21.21 | JT | jaytee: what do you need that much cpu for? |
| 03:21.53 | *** join/#asterisk rcy` (n=rcy@S010600003981572c.vc.shawcable.net) |
| 03:21.56 | jaytee | so that I can run WoW in Wine while everyone is using their phone? |
| 03:22.13 | drmessano | I think he was doing a comparison of a higher end system compared to the competition |
| 03:22.19 | jaytee | I would have been happy with a Core 2 Duo |
| 03:22.21 | drmessano | Nothng wrong with that |
| 03:22.29 | FinboySlick | when you guys use ztmonitor, does the RX bar jump all the way past the TX mark? I think my card might be defective. |
| 03:22.54 | drmessano | Saying you could buy 5000 WRT54Gs for the cost of a meridian isn't a reasonable comparison |
| 03:23.04 | *** join/#asterisk GReggFT (n=GReggFT@fw1-p1.lexi.com) |
| 03:23.41 | drmessano | FinboySlick: Your desperation of "Let me just start throwing this and that at is" is going to cost you in the end.. fix your problem |
| 03:23.45 | jaytee | yeah, my boss chose the server, Quad Xeon with 4GB of ram. I've only got 12 SIP clients and the calls from the Nortel 11C going through it and it's almost idling. One core might hit 6% utilization but that's about it. |
| 03:26.37 | FinboySlick | drmessano: I appreciate the wisdom... But I'm not sure how I would even approach the problem. Were you following earlier? |
| 03:28.38 | jaytee | anyone know if the times in the IRC stats for the channel are Eastern time or some other timezone? |
| 03:29.13 | FinboySlick | jaytee: What are the times now? |
| 03:29.39 | FinboySlick | Oh nevermind. |
| 03:29.59 | jaytee | it's a graph of 24 hour period |
| 03:30.17 | jaytee | and a table that shows the most active users in the channel |
| 03:30.27 | FinboySlick | jaytee: Then I wouldn't know. |
| 03:32.15 | *** join/#asterisk Iamnacho (i=Iamnacho@ip68-103-153-140.ks.ok.cox.net) |
| 03:32.38 | *** join/#asterisk stkn__ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 03:32.53 | drmessano | FinboySlick: It sounds like you're grasping at straws.. there is always a solution |
| 03:33.16 | FinboySlick | Okay, this is definitely strange. if I ztmontior my zap channel, RX goes through the roof every time there is TX, then it goes absolutely silent. |
| 03:33.28 | drmessano | FinboySlick: No, I wasn't here earlier, nor do I have the fix for everything.. My name isn't Russell Freaking Bryant for diety's sake. |
| 03:33.33 | Strom_C | FinboySlick: what kind of card? |
| 03:33.57 | FinboySlick | Strom_C: TDM800 with HWEC. |
| 03:34.05 | Strom_C | FXO or FXS ports? |
| 03:34.09 | FinboySlick | one 4FXO module. |
| 03:34.31 | FinboySlick | Sounds like some sort of fault, doesn't it? |
| 03:34.38 | Strom_C | what's the problem, exactly? |
| 03:36.10 | FinboySlick | Strom_C: Hmmm. Well, everything seems to work peachy with the exeption of a zap call going to an sip phone (or the reverse, sip to zap). Zap to asterisk works fine, sip to asterisk works fine. |
| 03:36.26 | Strom_C | what's the actual problem though? |
| 03:36.56 | FinboySlick | If I connect a zap channel to a sip channel, the sip hears very very loud white noise, but zap hears the sip dude just fine. |
| 03:37.26 | FinboySlick | Sounds like a codec problem, only stuff like iax to sip works fine. |
| 03:37.53 | FinboySlick | and now I just noticed that if TX is active on the card, RX goes through the roof. |
| 03:38.06 | FinboySlick | Which would sound like that white noise. |
| 03:38.47 | FinboySlick | I had [TK]D-Fender pretty stumped when we went around the problem earlier. And I'm still stumped myself. |
| 03:39.25 | Strom_C | what kind of phone? |
| 03:39.38 | Strom_C | what version of asterisk / zaptel? |
| 03:40.11 | FinboySlick | Strom_C: Hard and soft phohes give me the same result. This is oldie zaptel 1.2.26 and asterisk 1.2.27. |
| 03:40.36 | FinboySlick | Strom: The whole kit worked fine and dandy prior to a reboot, doesn't now. No config change. |
| 03:40.50 | Strom_C | have you tried disabling echo cancellation? |
| 03:41.07 | FinboySlick | Strom_C: Yup. No difference at all. |
| 03:41.44 | Strom_C | have you tried calling Digium? :) |
| 03:42.36 | obnauticus | Is anyone here good at socket programming with perl |
| 03:42.47 | obnauticus | i need help real fast for part of a script im writing for asterisk |
| 03:42.49 | FinboySlick | Strom_C: Yes, though I hadn't noticed the RX when TX problem then. The only thing they could offer was disabling HWEC and saying that they won't support 1.2 anymore. Which I can't blame them for, really. |
| 03:43.28 | Strom_C | well, just for kicks, is there anything preventing you from upgrading to 1.4? |
| 03:43.56 | *** join/#asterisk Segnale007 (n=Segnale0@cheetah-tiv-ppp664.bmts.com) |
| 03:44.34 | drmessano | Upgrade to 1.4? |
| 03:44.36 | drmessano | HEATHEN |
| 03:44.40 | FinboySlick | Strom_C: Well, nothing outside of the fact that I have no distro packages for it which is a maintenance annoyance. I was about to try that then. |
| 03:44.49 | drmessano | 1.2 4everNuSuck4SayingNot |
| 03:45.11 | drmessano | Which distro? |
| 03:45.26 | lmadsen | I don't like to say 'upgrade' -- a better term is 'migrate' |
| 03:45.34 | FinboySlick | drmessano: Ricer-foo. |
| 03:45.41 | drmessano | Ricer-Foo? |
| 03:45.42 | ManxPower | FinboySlick: your PBX will blow up if you don't read the upgrade.txt file in 1.4 source tree |
| 03:45.42 | FinboySlick | Or gentoo if you prefer. |
| 03:45.46 | drmessano | Oh |
| 03:45.50 | drmessano | Compile from source |
| 03:45.55 | lmadsen | upgrade makes people think of updating a currently working system -- migration implies a 2nd system |
| 03:46.10 | FinboySlick | ManxPower: Actually, my dialplan is 1.4 ready. I did some tests a while back. |
| 03:46.18 | lmadsen | someone give me something practical to build in Asterisk right now |
| 03:46.29 | ManxPower | FinboySlick: then you should be fine |
| 03:46.33 | drmessano | Even though Gentoo is like being caught naked with a goat, it works fine |
| 03:46.46 | lmadsen | if you could see a full systems dialplan and configuration for a common 'system' -- what would you choose? |
| 03:46.50 | FinboySlick | drmessano: I find your words heartwarmingly comforting. |
| 03:46.51 | ManxPower | I thought it was like being handcuffed to a zebra? |
| 03:47.53 | drmessano | ManxPower: You need to include the part about being naked.. it MAKES the reference |
| 03:47.58 | Strom_C | lmadsen: as to the function of the system? |
| 03:48.09 | lmadsen | Strom_C: I want to build something and document it |
| 03:48.15 | lmadsen | I'm looking for a problem to solve |
| 03:48.19 | Strom_C | build a sample PBX for a fifty-person office. |
| 03:48.32 | lmadsen | Strom_C: ok -- and what features does said office require? |
| 03:48.42 | FinboySlick | Gentoo still have an overlay with 1.4 in it somewhere? |
| 03:49.24 | Strom_C | the usual slew of things -- voicemail, conferencing, the standard CLASS features pack |
| 03:49.35 | Strom_C | ISDN PRI service from the telco |
| 03:49.44 | lmadsen | queues, MoH, voicemail, conferencing |
| 03:49.54 | lmadsen | ~google class features pack |
| 03:49.55 | Strom_C | yeah |
| 03:50.09 | Strom_C | lmadsen: well, you know...call waiting, call forwarding, caller ID, etc etc |
| 03:50.17 | lmadsen | gotcha |
| 03:50.24 | jaytee | subspace radio |
| 03:50.54 | lmadsen | Strom_C: I'll be back in a bit I guess :) |
| 03:52.10 | drmessano | Asterisk does callerID now? |
| 03:52.15 | jaytee | I thought I had an * problem earlier today because a couple of Budgetone 200 phones only show the callerid number and not the name. Then I realized it's cuz the Grandstream Budgetone is a piece of shit. |
| 03:52.43 | jaytee | drmessano, yeah! didn't you get the memo? |
| 03:53.10 | drmessano | Budge-tone or Budget-one |
| 03:53.14 | drmessano | You decide |
| 03:53.33 | Strom_C | I pronounce it "flaming bag of poop" |
| 03:53.40 | jaytee | hahaha |
| 03:54.00 | jaytee | I love the Polycoms. They are sooo much nicer. |
| 03:54.15 | jaytee | just ordered a 501 and a 550 |
| 03:54.43 | [TK]D-Fender | "BarbieTone" |
| 03:55.46 | drmessano | I was there for the development of the BT101.. All the product developers were sitting in a room |
| 03:55.50 | drmessano | and one of them stood up |
| 03:55.54 | drmessano | and yelled |
| 03:56.17 | lmadsen | I have a Mitel 5220, Polycom 501, and Cisco 7960, and I like them all for different reasons |
| 03:56.17 | drmessano | "Alright chums, let's do this.. LEEEEEROOY Jenkins" and ran into the lab |
| 03:56.17 | rabelais | not it? |
| 03:56.26 | drmessano | 6 months later, BT101 |
| 03:58.18 | jaytee | hahahaha |
| 03:58.25 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 03:58.36 | unpaidbill | for what reason do you like the 7960 |
| 03:58.48 | unpaidbill | because i have those, 7912s and 7902 and i hate them all |
| 03:58.53 | drmessano | I like the 7960 because it makes my wallet lighter |
| 03:59.44 | unpaidbill | haha |
| 03:59.51 | unpaidbill | yep, they're real good for that |
| 04:00.00 | lmadsen | I like the 7960 because it is easy to change around, and is very reliable |
| 04:00.21 | lmadsen | I've been using it as a development/test phone for 5 years on Asterisk |
| 04:00.26 | unpaidbill | it's easy to change around!? |
| 04:00.36 | unpaidbill | i cant figure out how to move soft switches, change their functions, or anything |
| 04:00.40 | unpaidbill | appears to all be hard coded |
| 04:00.43 | lmadsen | I wouldn't deploy them to a site, but as a "phone" it works remarkably well |
| 04:01.04 | unpaidbill | oh you mean move to a differnet server |
| 04:01.05 | *** join/#asterisk tuxd00d (n=tuxd00d@128.187.132.25) |
| 04:01.07 | lmadsen | unpaidbill: I don't do more than make calls, and perform transfers and sometimes the odd 3 way call |
| 04:01.20 | unpaidbill | yeah well, a 7960 for that function is way overkill imo |
| 04:01.21 | lmadsen | I mean to change the username/password/server |
| 04:01.24 | unpaidbill | i'd just use zoiper :P |
| 04:01.28 | lmadsen | I use that too |
| 04:01.32 | lmadsen | the 7960 is nice when it's free though |
| 04:01.43 | unpaidbill | i want to move around the soft buttons on the damn 7960, but they're hard coded soft buttons, what the hell |
| 04:01.56 | unpaidbill | and the differences between SCCP and SIP firmware are small and annoying |
| 04:01.56 | lmadsen | it was my first VoIP phone, so it's got nostalgic properties too |
| 04:02.06 | lmadsen | I only use SIP |
| 04:02.21 | FinboySlick | I wish there were more iax hardphones. |
| 04:02.26 | lmadsen | meh |
| 04:02.30 | lmadsen | isn't all the big a fan of IAX |
| 04:02.37 | lmadsen | s/the/that |
| 04:03.00 | unpaidbill | i recently moved over to SIP on them.. do you have to pick up the receiver before dialing #s, or hit the speaker phone first, or can you just dial a num and pick up the handset and have it dial? |
| 04:03.18 | lmadsen | you have to take the phone off hook (either physically, or via speakerphone) |
| 04:03.19 | unpaidbill | i havent looked around much yet for it, but i HAVE to either pick up the handset, or hit speakerphone before i can input #s on the damn SIP firmware |
| 04:03.26 | unpaidbill | yeah see that's goddamn retarded |
| 04:03.30 | unpaidbill | kicks cisco |
| 04:03.31 | lmadsen | it has it's annoyances, I agree |
| 04:03.40 | lmadsen | but it's rock solid hardware |
| 04:03.40 | unpaidbill | esxpecially because the SCCP firmware doenst work like that |
| 04:03.50 | lmadsen | I would never recommend someone to actually deploy them though |
| 04:03.53 | *** join/#asterisk RipeR-81 (i=RipeR81@dynamic13-254.MAN-B2-1.cablenet.com.ni) |
| 04:03.55 | unpaidbill | haha yeah |
| 04:03.58 | RipeR-81 | good evening |
| 04:04.04 | unpaidbill | i had mine thrust upon me when i took a new job |
| 04:04.07 | RipeR-81 | anybody used touchstar ? |
| 04:04.14 | lmadsen | good morrow RipeR-81 |
| 04:04.20 | unpaidbill | would have been polycom or something otherwise |
| 04:04.21 | lmadsen | neverheard of it :) |
| 04:04.30 | lmadsen | unpaidbill: well I have 2 of those too :) |
| 04:04.30 | unpaidbill | assuming THEY let me arrange soft keys as i like |
| 04:04.39 | lmadsen | they don't... not really anyways |
| 04:04.41 | RipeR-81 | lmadsen is a predictive dialer... that uses asterisk to connect to a remote IP PBX |
| 04:04.46 | unpaidbill | well fuck polycom too then :) |
| 04:04.58 | unpaidbill | i've used vicidial |
| 04:05.04 | unpaidbill | it works pretty well i guess |
| 04:05.08 | RipeR-81 | lmadsen using dialogic cards |
| 04:05.13 | unpaidbill | haha dialogic |
| 04:05.15 | lmadsen | runs :) |
| 04:05.20 | unpaidbill | i used to use dialogic cards |
| 04:05.24 | unpaidbill | in sco boxes and linux boxes |
| 04:05.30 | unpaidbill | and let me tell you, they're a gigantic waste of money |
| 04:05.31 | unpaidbill | :P |
| 04:06.06 | unpaidbill | well, for everything i have seen them used for they are |
| 04:06.07 | jaytee | Voiceguide uses dialogic boards, so does Nuance |
| 04:06.15 | unpaidbill | ok well nuance is cool |
| 04:06.20 | RipeR-81 | unpaidbill i realize the same thing |
| 04:06.23 | FinboySlick | While they are crappy, poor quality phones with dreadful firmware (though I guess I was lucky on the versions I used), I kind of like the 4-line thing on the GXP-2000. Don't see a lot of phones in a relatively close price range that takes the same approach. |
| 04:06.27 | RipeR-81 | :p thats why im ditching that dialer |
| 04:06.33 | unpaidbill | riper check vicidial out |
| 04:06.38 | unpaidbill | it doesnt look pretty, but it works |
| 04:06.44 | RipeR-81 | lol |
| 04:06.45 | unpaidbill | and it's easy to modify |
| 04:06.54 | unpaidbill | so if you have a web developer worth anything he can make it look better |
| 04:06.55 | jaytee | yeah, we're running Liaison which was from Locus but got bought by Nuance for our IVR with voice recognition |
| 04:07.36 | unpaidbill | jaytee have you played with sphinx at all? |
| 04:07.51 | unpaidbill | or lumenvox? |
| 04:08.06 | jaytee | I'm looking at getting an upgrade to a SIP enabled IVR from them because I haven't heard enough good things about Lumenvox although they offer a Nuance package. |
| 04:08.24 | RipeR-81 | unpaidbill im more looking into oracle contact center anywhere |
| 04:08.32 | RipeR-81 | unpaidbill or aspect |
| 04:08.32 | unpaidbill | ah, i was just wondering how they compare as far as accuracy goes |
| 04:08.40 | unpaidbill | i have never had the fortune of messing with a nuance system |
| 04:09.14 | jaytee | I've only read up about sphinx but from what I've read it seems to be too immature a product at this point. |
| 04:09.30 | unpaidbill | well, using it with a pbx isnt exactly plug and play |
| 04:09.35 | unpaidbill | you kind of have to do it yourself |
| 04:10.20 | unpaidbill | what does a nuance driven voice ivr run you |
| 04:10.24 | jaytee | I'd rather have a separate voice recoginition IVR system than have to configure * for it. |
| 04:13.09 | lmadsen | ~ == tilde, yes? |
| 04:13.13 | unpaidbill | yes |
| 04:13.18 | lmadsen | (i.e. correct spelling?) -- alternative names for the ~ character? |
| 04:13.22 | unpaidbill | tilde, yes |
| 04:13.31 | unpaidbill | there are no alternatives |
| 04:13.37 | lmadsen | ya, just making sure it wasn't like # :) |
| 04:13.40 | unpaidbill | unless you're retarded and call it squiggly line or something :P |
| 04:13.45 | lmadsen | pound, hash, octothorpe, etc.... |
| 04:13.49 | lmadsen | unpaidbill: lol |
| 04:14.25 | unpaidbill | yeah octothorpe is definitely the coolest name for a character |
| 04:14.43 | unpaidbill | makes me want to punch the sky and scream FUCK YEAH! |
| 04:15.00 | jaytee | Americuh! Fuck Yeah! |
| 04:15.53 | RipeR-81 | lol |
| 04:15.55 | mosty | how can i set a channel variable for the callee? |
| 04:16.06 | [TK]D-Fender | mosty: .... huh? |
| 04:16.10 | RipeR-81 | blitzrage isgoing nuts XD |
| 04:16.50 | blitzrage | mosty: A --caller--> Asterisk --callee--> B ? |
| 04:17.56 | jaytee | I wonder what Lumenvox means when they say 500 pronunciations per interaction or 12000 pronunciations per interaction? |
| 04:18.03 | mosty | i have a custom application in features.conf that i want to enable for both A and B to use |
| 04:19.18 | mosty | in blitzrage's diagram, it works for A, but not for B because B does not have the correct DYNAMIC_FEATURES channel variable set |
| 04:19.25 | FinboySlick | JOOOOOOOOY! |
| 04:19.36 | FinboySlick | zaptel 1.4 fixed it! |
| 04:19.41 | blitzrage | mosty: Set(__DYNAMIC_FEATURES=foo) |
| 04:19.53 | FinboySlick | I am sooOOoooo happy. |
| 04:20.13 | unpaidbill | high five |
| 04:20.39 | blitzrage | mosty: I also think you need to set options in features.conf that tells the system which channels can utilize the feature, and which side it applies to |
| 04:20.47 | FinboySlick | I'm thinking it might have been a firmware messup. |
| 04:21.04 | FinboySlick | Anyway. What counts is that I can sleep now. |
| 04:21.20 | mosty | blitzrage, in features.conf you can set caller or callee, but that just says which channel to take control of once the event is triggered |
| 04:21.48 | blitzrage | ; ActivatedBy -> This is which channel is allowed to activate this feature. Valid |
| 04:21.49 | blitzrage | ; values are "caller", "callee", and "both". "both" is the default. |
| 04:21.49 | blitzrage | ; The "caller" is the channel that executed the Dial application, while |
| 04:21.50 | blitzrage | ; the "callee" is the channel called by the Dial application. |
| 04:21.51 | mosty | blitzrage, and Set(DYNAMIC_FEATURES=foo) doesn't work for the called end, does it? |
| 04:22.12 | blitzrage | mosty: I didn't say that -- I said Set(__DYNAMIC_FEATURES=foo) |
| 04:22.24 | *** join/#asterisk Juggie (i=Juggie@CPE001601df17fb-CM001a6682d7b6.cpe.net.cable.rogers.com) |
| 04:23.03 | blitzrage | there is also the ActivateOn option which is used to control which side of the call (originating or terminating channel) the feature gets applied to |
| 04:24.11 | mosty | blitzrage, oh perhaps i can specify "caller/callee" in the application map |
| 04:24.35 | blitzrage | mosty: no you can't |
| 04:24.37 | Juggie | blitzrage! he's back! |
| 04:24.38 | blitzrage | do it like this |
| 04:24.39 | blitzrage | stopMonitorSrc => #1,self/caller,StopMonitor ; allow the source caller to stop monitoring |
| 04:24.40 | blitzrage | stopMonitorDst => #1,peer/callee,StopMonitor ; allow the destination caller to stop monitoring |
| 04:24.46 | Juggie | lmadsen is on vacation? |
| 04:24.57 | blitzrage | then you have to also set the DYNAMIC_FEATURES like I told |
| 04:25.26 | mosty | blitzrage, so Set(__DYNAMIC_FEATURES=foo) sets the variable on the called side? |
| 04:25.33 | *** part/#asterisk dxm (n=dxm@200.146.116.191.adsl.gvt.net.br) |
| 04:25.47 | blitzrage | yes |
| 04:25.59 | blitzrage | __ means "make this variable transitive" |
| 04:26.24 | FinboySlick | Thanks for all the support guys. I'll try to be back and help instead of leech next time ;) |
| 04:26.30 | FinboySlick | Now I want bed. |
| 04:26.44 | blitzrage | mosty: this is how I use it: |
| 04:26.45 | blitzrage | exten => s,n,Set(_DYNAMIC_FEATURES=${IF($[${ISNULL(${DYNAMIC_FEATURES})}]?stopMonitorDst:${DYNAMIC_FEATURES}#stopMonitorDst)}) ; allow the destination channel to stop recording |
| 04:27.49 | blitzrage | all that is doing is determining where to apply the # before the stopMonitor in case I had previously set something |
| 04:28.12 | blitzrage | or rather, whether to apply it at all |
| 04:28.25 | blitzrage | Juggie: sssssssssssh -- you'll give away my secret |
| 04:28.27 | mosty | using __ seems to work- thanks |
| 04:28.44 | blitzrage | so many good secrets |
| 04:35.42 | Juggie | :> |
| 04:37.42 | *** join/#asterisk BBHoss (n=hoss@c-68-62-175-86.hsd1.al.comcast.net) |
| 04:43.51 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 04:44.00 | jaytee | ~book |
| 04:44.00 | 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 |
| 04:45.28 | *** join/#asterisk TrentCreek (n=TrentCre@cpe-70-117-198-98.rgv.res.rr.com) |
| 04:45.49 | TrentCreek | who beez here? |
| 04:46.06 | JT | :o |
| 04:46.37 | TrentCreek | who knows about disabling SE Linux even though it is already set for disable? |
| 04:49.44 | jaytee | Trent, what distro are you running? |
| 04:52.14 | jaytee | TrentCreek, if you're running RHEL or CentOS edit your /etc/selinux/config file so it has SELINUX=disabled in it and restart. |
| 04:53.41 | jaytee | or you can set it to SELINUX=permissive and then turn it off by typing echo > 0 /selinux/enforce |
| 04:54.13 | jaytee | and turn it back on again with echo 1 > /selinux/enforce |
| 04:54.25 | *** join/#asterisk denon (n=denon@tooth.decay.org) |
| 04:54.25 | *** mode/#asterisk [+o denon] by ChanServ |
| 04:54.37 | Strom_C | look look |
| 04:54.39 | Strom_C | vim restaurant |
| 04:54.40 | Strom_C | http://www.flickr.com/photos/stromcarlson/2421795454/in/set-72157604594227967/ |
| 04:55.09 | jaytee | ooops, just noticed, it should be echo 0 > /selinux/enforce not echo > 0. I had them transposed |
| 04:56.17 | jaytee | Strom_C, so when you're done with your meal do you just shout to the waiter, "SHIFT ZZ"? |
| 04:56.44 | Strom_C | i think you shout "wq!" |
| 04:57.04 | jaytee | hehe |
| 04:57.15 | jaytee | I love Thai |
| 04:57.53 | jaytee | ok, time for bed, nite all |
| 04:58.17 | drmessano | Latero |
| 04:59.45 | TrentCreek | oops |
| 05:00.03 | TrentCreek | WTFU |
| 05:02.41 | TrentCreek | jaytee: I am running Fedora and that is already set in the config file |
| 05:03.31 | drmessano | Then you're doing something wrong |
| 05:03.34 | TrentCreek | no |
| 05:04.19 | TrentCreek | I installed Firestarter, and that is when things started to go wrong |
| 05:04.40 | drmessano | facepalms |
| 05:05.58 | drmessano | Sounds to me like you need to UNDO what you've done..... |
| 05:05.59 | TrentCreek | even the GUI for the firewall/SELinux has both SE Linux and fire wall ghoested out because they are both disabled in the system, but yet I am getting SE Linux security warnings |
| 05:06.37 | TrentCreek | So it is preventing Asterisk from using the DNS |
| 05:06.47 | drmessano | Get rid of firestarter |
| 05:07.47 | TrentCreek | yeah..... |
| 05:08.05 | drmessano | The latest version is over 3 years old |
| 05:08.09 | drmessano | WTF |
| 05:08.13 | drmessano | Dump it |
| 05:09.34 | drmessano | Never, ever use abandonware |
| 05:14.39 | TrentCreek | I am hoping to use somthing for sharing the internet |
| 05:14.39 | TrentCreek | since I don't know xxxxxxx very well |
| 05:14.44 | denon | drmessano has obviously never played Outrun! |
| 05:14.48 | TrentCreek | okay its uninstalled |
| 05:14.56 | TrentCreek | anyone got a homemade Wifi antenna? |
| 05:15.00 | TrentCreek | "Cantenna?" |
| 05:16.05 | TrentCreek | of firestarter? |
| 05:16.35 | drmessano | lol |
| 05:16.45 | drmessano | Well, I was thinking more along the lines of... |
| 05:16.45 | TrentCreek | lost my connection for a moment |
| 05:17.34 | drmessano | Never use an application that digs into the core of your system that is 3 years old and unsupported as hell |
| 05:17.37 | drmessano | Thats common sense |
| 05:18.03 | drmessano | I normally won't touch anything that hasn't been updated in a year |
| 05:18.23 | denon | drmessano: so, how current is the firmware on your switch? |
| 05:18.29 | denon | or home router |
| 05:18.35 | drmessano | 1 week |
| 05:18.42 | drmessano | Maybe 2 |
| 05:18.51 | denon | sounds like you don't like stability :) |
| 05:19.16 | drmessano | It's DD-WRT, and the new release has been tested for like 2 years |
| 05:19.19 | drmessano | WTF |
| 05:19.23 | denon | I dont mind letting stuff get old if it's not accessible to the public, and if it's solid |
| 05:19.50 | denon | like internal switch firmware |
| 05:19.54 | drmessano | FAIL, try again |
| 05:21.08 | denon | Im not going to cycle core switches every months because cisco fixed some weird issue with some crazy ancient realtek nic or something |
| 05:21.13 | denon | -s |
| 05:21.48 | drmessano | Actually, when it comes to routers, I update as often as necessary without going way out of date.. if I havent updated router software or distro in a year, its a security issue waiting to happen |
| 05:21.50 | denon | but yeah, in general, I agree .. best not to touch old stuff .. |
| 05:22.04 | denon | it's especially true on the open source world, where so much stuff gets half-ass developed, then dumped |
| 05:22.45 | denon | sourceforge is full of descriptions like, "This product will be a .." |
| 05:22.48 | denon | from 2002 |
| 05:22.57 | denon | "pre-Alpha" |
| 05:23.52 | drmessano | Flite is good example |
| 05:24.07 | drmessano | I like Flite.. it isn't perfect or great, but it works |
| 05:24.15 | drmessano | Doesn't work on 1.6 |
| 05:24.33 | drmessano | Personally, I should have saw it coming.. It's development broke 2 or 3 of my rules |
| 05:24.43 | drmessano | But there was < 12 month old code |
| 05:24.47 | drmessano | I FAIL'ed |
| 05:40.16 | nick125 | major fail. |
| 05:41.43 | TrentCreek | That's colonel fail |
| 05:46.09 | RipeR-81 | im looking for a 1800 number provider in mexico... anybody has info? |
| 05:47.26 | TrentCreek | VOIP I presume? |
| 05:47.45 | unpaidbill | jose cueeeeeeeeeeeeeervo! |
| 05:47.47 | RipeR-81 | TrentCreek you bet!!! |
| 05:47.56 | unpaidbill | oh you said voip, not tequila |
| 05:48.00 | RipeR-81 | unpaidbill uhmm.. what is that? lol |
| 05:48.15 | unpaidbill | that's me being a jackass. i apologize |
| 05:48.26 | RipeR-81 | unpaidbill no regret m8t |
| 05:48.28 | RipeR-81 | :D |
| 05:48.42 | TrentCreek | I recall seeing a company that did not actually provide toll free service, but calls are allowed to be made from Mexico on it |
| 05:48.45 | unpaidbill | have you checked the list on voip-info? |
| 05:48.59 | TrentCreek | but at normal rates |
| 05:49.10 | RipeR-81 | unpaidbill silly me... no i havent :) |
| 05:49.27 | unpaidbill | dont know if it will help, but there are a ton of voip providers listed there, you may get lucky! |
| 05:49.45 | unpaidbill | i have found that it's a real pain in the ass to find a decent provider |
| 05:49.50 | unpaidbill | you really have to sift through a lot of shit |
| 05:50.06 | TrentCreek | I think it is Voxbone, but I can't recall |
| 05:50.42 | TrentCreek | Or it is another....I have to look it up |
| 05:53.12 | TrentCreek | RipeR-81: While I am looking, you can check this guy out. aztelco.net He is in DF |
| 05:56.00 | TrentCreek | I think this could be it www.didx.net/ |
| 05:56.47 | TrentCreek | now where is that Auxiliary SELinux disable button? |
| 05:57.22 | mihr | in /etc |
| 05:58.02 | RipeR-81 | TrentCreek |
| 05:58.03 | RipeR-81 | thx |
| 05:58.39 | TrentCreek | sure |
| 05:59.02 | *** join/#asterisk techie (n=techie@adsl-76-214-9-106.dsl.lsan03.sbcglobal.net) |
| 06:02.08 | RipeR-81 | TrentCreek found a carrier... metrored.com.mx |
| 06:02.31 | TrentCreek | Firefox can't find the server at metrored.com.mx. |
| 06:03.05 | RipeR-81 | www.metrored.com.mx ? perhaps |
| 06:03.14 | TrentCreek | yep... |
| 06:03.22 | RipeR-81 | TrentCreek hate when apache only listens to www!!! |
| 06:03.35 | TrentCreek | that make me wonder why they can;t fix a simple header problem |
| 06:04.13 | TrentCreek | i think it is a header issue...they need to rout both xxx.com & www.xxx.com |
| 06:05.25 | TrentCreek | RipeR-81: Are you sure that is correct? Seems they do retail |
| 06:05.27 | mihr | TrentCreek: its in the DNS setting. |
| 06:05.36 | TrentCreek | ahh |
| 06:06.01 | TrentCreek | in IIS you just make the headers like I showed you |
| 06:06.15 | TrentCreek | and it point it to the same site |
| 06:06.57 | mihr | how can it reach the IIS if no A record for that ip. |
| 06:06.58 | TrentCreek | RipeR-81: Okay I see..trunks and other goodies |
| 06:07.40 | TrentCreek | DNS you just pointing the IP with the major domain |
| 06:07.48 | drmessano | There is no A record for the domain |
| 06:07.52 | TrentCreek | oh |
| 06:08.07 | drmessano | ping is your friend |
| 06:08.19 | *** join/#asterisk dominic1 (n=dob@213.221.82.242) |
| 06:08.27 | TrentCreek | 10, 20 o 30 troncales. |
| 06:09.12 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
| 06:09.19 | TrentCreek | RipeR-81: I se no prices |
| 06:09.36 | TrentCreek | and knowing Mexico, PLUS no prices means big $$$$ |
| 06:10.21 | mihr | drmessano, in your opinion whats the best and cheapest voip provider today? |
| 06:10.38 | drmessano | You're looking got a Unicorn |
| 06:10.40 | drmessano | for* |
| 06:10.47 | drmessano | is sleepy |
| 06:10.48 | mihr | ohhh |
| 06:11.12 | mihr | so looking for a good provider is also like looking for the holy grail. |
| 06:11.22 | drmessano | When someone asks me for best and cheap, I immediately discount the fact they are serious about finding decent service |
| 06:11.49 | drmessano | I like LES.net.. It works well.. |
| 06:11.56 | drmessano | Others like it as well |
| 06:12.06 | mihr | drmessano: correct |
| 06:12.12 | drmessano | But 10 others will show you someone else with a cheaper per min rate.. or more rate centers |
| 06:12.45 | TrentCreek | voicetrading.com has 3 levels of service/pricing |
| 06:12.48 | drmessano | Les.net is up.. they're not expensive, and they seem to care about service |
| 06:12.55 | RipeR-81 | TrentCreek prices.. i´ll quote them tomorrow .. |
| 06:13.11 | drmessano | I don't like the "fleamarket" sites |
| 06:13.16 | TrentCreek | I would be scared to ask |
| 06:13.30 | nick125 | Trust me, you more than likely don't want to go the cheap route. |
| 06:13.36 | drmessano | They're like throwing darts at a dartboard, blindfolded |
| 06:13.59 | mihr | nick125: oh okay. |
| 06:14.03 | drmessano | "Oh.. DIDFun.com has a $3 DID for my area.. @ 11 cents per minute" |
| 06:14.10 | TrentCreek | well on Voicetrading you dont liek the cheap route, then switch to their higher priced/better quality |
| 06:14.33 | drmessano | "Oh.. DIDYou.com has a $7 DID for my area.. @ 1 cent per minute.. but you need 10000 minutes to qualify" |
| 06:14.34 | mihr | well, we are only planning to have 50 seats of call center agent. |
| 06:14.36 | TrentCreek | what area? |
| 06:14.38 | drmessano | Most of those are BS |
| 06:14.40 | unpaidbill | callcentric.com dirt cheap DID 2.95/mo unlimited! |
| 06:14.42 | unpaidbill | high five! |
| 06:14.56 | drmessano | Look at their outbound rates |
| 06:15.00 | *** join/#asterisk trnzmeta (n=bleh@iinet.guard.com.au) |
| 06:15.01 | drmessano | Callcentric is terrible |
| 06:15.25 | mihr | pure outgoing no incoming calls. |
| 06:15.30 | nick125 | needs to look at porting (or getting new ones) his numbers.. |
| 06:15.50 | drmessano | mihr: Les.net will do bulk discounts.. E-mail them |
| 06:16.00 | TrentCreek | That price is not correct |
| 06:16.06 | unpaidbill | i dont use outbound |
| 06:16.10 | unpaidbill | their dirt cheap did is great |
| 06:16.15 | drmessano | $2.95 + 911 |
| 06:16.22 | unpaidbill | no 911 for me |
| 06:16.23 | mihr | drmessano: yeah I will thanks. |
| 06:16.28 | unpaidbill | i told them in not in the usa |
| 06:16.40 | drmessano | $4 a month from Les.net.. and it works |
| 06:16.42 | nick125 | doesn't get the point of e911 with having 5 cell phones in his house. |
| 06:16.56 | JT | nick125: how so? |
| 06:17.07 | drmessano | Callcentric tried to port a number for me.. and it took them 2 months to tell me they couldnt |
| 06:17.10 | drmessano | They can go to hell |
| 06:17.21 | mihr | drmessano: do les.net have something like $xx for unlimited call for 50 users? or something similar? |
| 06:18.14 | nick125 | JT: If I have 4 cell phones in the house (four Verizon and a dozen GSM devices without service) and a POTS line, I'm pretty sure I'll have something to call 911 with. |
| 06:18.34 | nick125 | mihr: In bulk, you'll have a really hard time finding places that offer unlimited anything. |
| 06:18.49 | JT | nick125: probably, if you have POTS |
| 06:18.51 | nick125 | well, anything above "consumer" grade. |
| 06:21.41 | mihr | nick125: is 50 seats all outgoing are above "consumer" grade? |
| 06:22.11 | *** join/#asterisk _gm (n=gmustafa@202.133.78.60) |
| 06:22.24 | drmessano | mihr: NO ONE has unlimited outbound for bulk users |
| 06:22.28 | nick125 | mihr: How many minutes are you using? |
| 06:22.33 | nick125 | drmessano: bingo. |
| 06:22.40 | drmessano | mihr: VoIP is metered for a reason.. same reason AT&T doesnt meter it |
| 06:22.46 | nick125 | Even if there was somewhere, they'd have usage limitation. |
| 06:22.51 | drmessano | Its different.. You can see huge discounts |
| 06:23.01 | drmessano | But you have to shop them out |
| 06:23.47 | drmessano | Nobody is dumb enough to offer unlimited service to a callcenter.. if you're going to save money with VoIP, you need to think different |
| 06:23.53 | drmessano | First off, it's not "Almost free" |
| 06:24.30 | drmessano | Secondly, don't expect canned plans to amount to crap.. they are put together for stupid sheep consumers who are used to paying Bell |
| 06:24.39 | drmessano | $24.99 for Vonage? Yah, right |
| 06:25.18 | mihr | drmessano: ok. |
| 06:26.07 | drmessano | Getting a cheap or Free DID is easy.. getting better service is worth a few extra bucks sometimes.. |
| 06:26.18 | nick125 | You might be able to see if you can find a provider that'll give you a usage break for that much usage. |
| 06:26.19 | drmessano | When it comes to termination... shop around.. It jumps all over |
| 06:26.38 | drmessano | Like I said, Les.net offers discounts they don't publish |
| 06:26.56 | drmessano | I know I got an e-mail sometime back that they offer discounts for bulk users.. |
| 06:27.00 | drmessano | But you have to ask |
| 06:27.04 | TrentCreek | $24.99? That's it?? I am signing up right now! |
| 06:27.13 | nick125 | I mean, I'm guessing that you're going to be using least 125k minutes a month. |
| 06:27.18 | digitalirony | JT: you can always program in for 911 service yourself |
| 06:27.24 | digitalirony | its just another extension |
| 06:27.28 | nick125 | (50 seats with 2500 minutes each) |
| 06:27.36 | drmessano | How so? |
| 06:27.45 | *** join/#asterisk fnordus (n=dnall@70.71.224.2) |
| 06:27.59 | drmessano | You can't just "Program in" e911 |
| 06:28.00 | digitalirony | just make a 911,ext right |
| 06:28.23 | digitalirony | kinda |
| 06:28.31 | digitalirony | you can have it call the police |
| 06:28.34 | digitalirony | lol |
| 06:28.37 | drmessano | Thats NOT |
| 06:28.38 | drmessano | NOT |
| 06:28.39 | drmessano | 911 |
| 06:28.44 | digitalirony | i understand |
| 06:28.51 | digitalirony | and its NOT pots |
| 06:29.02 | digitalirony | but its a solutions to an answer |
| 06:29.07 | digitalirony | and its not a no you can't do it |
| 06:29.09 | drmessano | Thats non-emergency numbers.. which are NOT NOT given priority or feature E911 info |
| 06:29.14 | drmessano | E = Enhanced |
| 06:29.20 | drmessano | As in, your address, etc |
| 06:29.43 | digitalirony | its a signal of some kinda |
| 06:30.03 | Strom_C | "Someone's dying!!" "What's your address?" "Oh...uh...Carol, what's our address here? Hmm, ok, I'll go look for it on our stationery, hang on......." |
| 06:30.09 | drmessano | Tell that to me later when someone is breaking into your house and you got a BUSY SIGNAL |
| 06:30.12 | Strom_C | dumb dumb dumb dumb dumb dumb dumb dumb dumb dumb dumb dumb dumb dumb dumb dumb dumb dumb dumb dumb dumb dumb dumb dumb dumb dumb dumb dumb dumb dumb dumb dumb dumb dumb dumb dumb dumb dumb |
| 06:30.13 | drmessano | or get PUT ON HOLD |
| 06:30.19 | drmessano | You have NO fucking idea what youre talking about |
| 06:30.24 | Strom_C | dumb dumb dumb dumb dumb dumb dumb dumb dumb dumb dumb dumb dumb dumb dumb dumb dumb dumb dumb dumb dumb dumb dumb dumb (filling up screen) |
| 06:30.49 | digitalirony | drmessano: its not dumb, if someone doesn't have 911 service and needs something worth doing |
| 06:30.56 | digitalirony | instead they would just not have it |
| 06:30.56 | trnzmeta | errr press the any key |
| 06:30.58 | drmessano | ITS NOT WORTH DOING |
| 06:30.59 | drmessano | Thats it |
| 06:31.08 | drmessano | You are better off NOT expecting it to work |
| 06:31.12 | drmessano | Because it wont |
| 06:31.27 | digitalirony | well its like a hand granade man |
| 06:31.38 | digitalirony | you pull the pin throw it in and see who blows up |
| 06:31.50 | digitalirony | it was just a suggestion |
| 06:31.53 | Strom_C | perhaps you should learn to spell "grenade" |
| 06:32.04 | drmessano | What you're saying makes no sense. . |
| 06:32.17 | drmessano | Non-emergency numbers are non-emergency |
| 06:32.24 | drmessano | They are not handled by dispatchers in a lot of cases |
| 06:32.51 | drmessano | They are also prone to busy signals and feature no e911 info |
| 06:32.54 | Strom_C | "Someone's breaking in!!" "Sir, this is the fire department." |
| 06:33.22 | digitalirony | i have done it.... |
| 06:33.27 | drmessano | Your suggestion is not only wrong, but it's dangerous.. because it adds some presumption |
| 06:33.56 | digitalirony | i have had to dial 911 before and not had 911 service |
| 06:34.02 | digitalirony | so i called the police |
| 06:34.03 | drmessano | digitalirony: You are talking to someone who has been involved with public safety for years.. you are WRONG, WRONG |
| 06:34.24 | creativx | I AM RIGHT |
| 06:34.26 | creativx | i just wanted to add that. |
| 06:34.58 | drmessano | Yeah, and I have called the non-emergency number to report a drunk walking down the middle of the road and been put on hold for 10 mins |
| 06:35.09 | drmessano | Do you want that for your fake 911 setup? |
| 06:35.30 | digitalirony | drmessano: Thank you for telling me that. You are right, I am totally wrong for not saying exactly what you are saying, but im saying here in real life, when you have no 911 service its better to have the ghostbusters on speed dial than to watch some one die or get robbed |
| 06:35.51 | digitalirony | and not for fake 911 |
| 06:35.56 | digitalirony | won't ever take place of that |
| 06:36.05 | digitalirony | i just used to be poor |
| 06:36.09 | digitalirony | lol |
| 06:36.10 | drmessano | digitalirony: No, in real life.. your idea is adding some assumption it will be there when you need it |
| 06:36.34 | drmessano | and it wont be |
| 06:37.23 | drmessano | Your idea is not "better than nothing".. you may as well put animal control on your 911 extension.. you have just as much of a chance of getting help that way |
| 06:37.39 | drmessano | or a pizza place.. maybe they can dial 911 for you |
| 06:37.56 | digitalirony | holy crap man, chill out i don't want to argue with you. im aloud to speak my idea's and your aloud to say a better way you could do that instead of wasting 10 mins of anyones time who else is reading this and say a better solutions of being in a place where you don't have 911 service on your phone(s) and want a a back up |
| 06:38.06 | JT | "allowed" |
| 06:38.14 | digitalirony | thanks |
| 06:38.26 | drmessano | Your solution is dangerous.. and you're not fucking getting that |
| 06:38.29 | JT | a better solution is to not have the extension and to have POTS and/or cell phones |
| 06:38.31 | *** join/#asterisk fnordus (n=dnall@70.71.224.2) |
| 06:38.56 | drmessano | It's not "better" than nothing |
| 06:39.00 | jblack | What is the difference between a broken clock and an economist? |
| 06:39.00 | digitalirony | i do get that its dangerouse as in it might not work or there is a chance that it will take to long |
| 06:39.07 | jblack | a broken clock is right twice a day. |
| 06:39.09 | drmessano | It's worse.. it's setup under false pretenses |
| 06:39.13 | digitalirony | but its SOMETHING beter than nothing |
| 06:39.20 | nick125 | Anyone here use voicepulse? I'm thinking of adding them for outbound calling. |
| 06:39.26 | Strom_C | drmessano: what do you expect from someone who can barely spell words in the one language he's actually expected to know? |
| 06:40.02 | drmessano | digitalirony: Good.. I hope that one day when you need an emergency response, you'll take the extra 2 or 3 minutes to use your pseduo 911 versus getting real help |
| 06:40.14 | drmessano | Maybe then Darwin will have his way |
| 06:40.23 | jblack | digitalirony: I set up dialing to 911 to get forwarded to the local police station. |
| 06:40.27 | digitalirony | drmessano: I cn spll any wy i wnt 2 fi you utsanerd em |
| 06:40.43 | drmessano | digitalirony: I said nothing about your spelling |
| 06:40.49 | drmessano | digitalirony: Learn to read too |
| 06:40.56 | *** join/#asterisk fnordus (n=dnall@70.71.224.2) |
| 06:41.18 | jblack | digitalirony: Heh. If you're looking for answers, then being as clear as possible is in your interests. |
| 06:41.26 | digitalirony | drmessano: im srry it wsnt ouy |
| 06:41.30 | jblack | Our interests mostly revolve around staving off boredom. |
| 06:41.59 | drmessano | My interest is not to see someone spewing off something that is going to end up with someone getting hurt or killed for following the suggestion |
| 06:42.06 | drmessano | Sorry, but that bothers the shit out of me |
| 06:42.14 | jblack | Anyways, set up a local 911 extension that dials the local police station, if you lack 911 and e911 |
| 06:42.28 | unpaidbill | is there any app that exists that allows you to detect a beep tone from an answering machine? or specify a frequency range or anything? |
| 06:42.34 | *** join/#asterisk shinao1 (n=shinao1@62.173.61.17) |
| 06:42.37 | drmessano | The idea of doing that is dangerous |
| 06:42.39 | jblack | unpaidbill: look at AMD |
| 06:42.49 | Strom_C | "Thank you for calling Jizz City Police. This office is currently closed. Please call back during our normal business hours. Thank you!" |
| 06:42.51 | unpaidbill | yeah i've looked at AMD, it's nice and all |
| 06:42.54 | Strom_C | URRK HUK DEAD |
| 06:42.55 | unpaidbill | but it doesnt detect a beep |
| 06:43.05 | digitalirony | jblack: i wasn't looking for the answer, just threw a suggestions out, and drmessano decided he didn't like me, and instead of improving me on it and just saying a better way to do this, he starts telling me im dumb because I actually don't know as much as him because im probabaly atleast half his age and had a life |
| 06:43.17 | jblack | dude, drmessano doesn't like anyone. |
| 06:43.29 | drmessano | Strom_C: What about when you need an ambulance, and the person answering the non emergency number at the police station doesnt have the number for the ambulance service |
| 06:43.43 | jblack | I have no doubt that he leaves a bag of flaming poo on his mother's doorstep every haloween. |
| 06:43.48 | drmessano | Strom_C: ..or if you need a fire truck, and you live in a rural fire district, and she doesnt know which fire dept to call |
| 06:43.50 | unpaidbill | something like background that would play a message 'hey asshole you owe us money p-' *BEEP* 'hey asshole you owe us money pay up' |
| 06:44.20 | drmessano | jblack: I like people that don't suggest something completely dangerous |
| 06:44.26 | Strom_C | digitalirony: you may think drmessano is being a jerk, but you're also being a tool for not actually taking his suggestions and instead getting all butthurt that he isn't just blowing sunshine up your ass. |
| 06:44.41 | unpaidbill | lots of butthurt people here today! |
| 06:44.42 | drmessano | jblack: Dismiss what I am saying with personal comments all you want.. it doesnt make it any less of a BAD idea |
| 06:45.07 | digitalirony | Strom_C: doesn't have to blow sun shine up my ass, but he doesn't have to be rude either |
| 06:45.12 | TrentCreek | seems there are more people here now..maybe another can answer my question |
| 06:45.18 | Strom_C | unpaidbill: automatically calling people with recordings like that tends to be illegal |
| 06:45.21 | drmessano | digitalirony: I am not being rude.. I am just not agreeing with you, and youre acting like a child |
| 06:45.28 | Strom_C | digitalirony: it's IRC...grow a thicker skin. |
| 06:45.30 | TrentCreek | How does one Disable SELinux that has been already disabled? |
| 06:45.43 | Strom_C | TrentCreek: that question makes no sense |
| 06:45.50 | Strom_C | why would you disable it if it's already disabled? |
| 06:45.54 | jblack | well, I was overthinking it at the time, but I set mine up so that if you press any number while a short Background() plays telling the caller what is going on ("You are at address blah, your number is blah"), then it goes into endless loop playing something like 'A person at address has dialed 911. The person at address is hurt badly enough they can not talk". and again and again and again |
| 06:45.54 | TrentCreek | i know, bu tit is happening |
| 06:46.08 | unpaidbill | strom: haha well obviously it wouldnt call anyone an asshole, but for reminders for people to pay there bill it would be nice |
| 06:46.14 | TrentCreek | SELinux is disables on my server, but yet it is functioning |
| 06:46.26 | Strom_C | unpaidbill: any recording, not just offensive ones |
| 06:46.29 | unpaidbill | assuming they sign up for the service.. whatever |
| 06:46.30 | drmessano | digitalirony: I made the mistake of trying to tell you why your idea is a bad one, as I actually have some insight as an insider how the process really works.. If you don't want to take my advice, then so be it. |
| 06:46.32 | *** join/#asterisk angryuser (n=angryuse@d02m-213-44-187-97.d4.club-internet.fr) |
| 06:46.39 | jblack | I don't think it's a great idea; just a mitigation for the fact that my local police chief has ignored 5 emails and 2 calls asking for information of how to get registered with 911. |
| 06:46.59 | digitalirony | drmessano: i never actually got your advice other than mine was bad |
| 06:47.01 | Strom_C | jblack: so contact some other authority |
| 06:47.04 | TrentCreek | Strom_C: SELinux is disabled on my server, but yet it is functioning |
| 06:47.05 | digitalirony | that was kind of the whole point |
| 06:47.15 | jblack | strom: The police station _is_ the city authority. |
| 06:47.18 | drmessano | digitalirony: You were too busy arguing and calling me a jerk.. read what I have written |
| 06:47.30 | jblack | I dived into the 911 stuff, found the local whatchamacallit, and it's them. |
| 06:48.18 | Strom_C | jblack: the police is a department of the city government, is it not? |
| 06:48.26 | drmessano | jblack: You said "police" which usually means "town" or "city" try the county |
| 06:48.41 | Strom_C | so therefore, if the police is unresponsive, you go up one level |
| 06:48.46 | Strom_C | i.e. city government |
| 06:48.50 | Strom_C | or county government |
| 06:49.00 | drmessano | County usually handles 911, even if the city/township is a PSAP |
| 06:49.14 | jblack | The one for the city is in the wilkes-barre city police department. |
| 06:49.47 | digitalirony | and so that means what |
| 06:49.47 | jblack | Yeah, I should push it harder, strom. Chain of command and all. |
| 06:50.11 | jblack | dremessalo: The list for where the 911 stations are is available online. |
| 06:50.16 | jblack | And that's where mine is. |
| 06:50.40 | jblack | Wilkes-Barre is the seat for county government too. |
| 06:51.34 | drmessano | The 911 center may be in Wilkes-Barre, but that has nothing to do with who runs it |
| 06:51.42 | jblack | Fair enough. |
| 06:51.43 | drmessano | It's likely not the Police dept |
| 06:51.46 | digitalirony | and when you used to live in a place that just got 911 service 3-5 years ago and not everyone still had it.... you can call the local police station cause thats what your gonna get anyway |
| 06:51.54 | mihr | trentcreek, how did you know its already disabled? |
| 06:51.55 | drmessano | It would be the sheriffs office or EMA in a lot of cases |
| 06:52.07 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 06:52.13 | drmessano | digitalirony: No.. They DO NOT answer the non emergency lines the same, AT ALL |
| 06:52.30 | TrentCreek | mihr: Because I manually disabled it because it was screwing up my apps |
| 06:52.41 | digitalirony | i don't know, maybe not where your from or where everyone else is from. but where im from they did |
| 06:52.50 | digitalirony | I DID it once |
| 06:52.58 | digitalirony | when some one was trying to break in |
| 06:53.00 | drmessano | digitalirony: If you have NO 911 in town, and thats the only access number, then obviously they have lots of lines and rollover.. if they do have 911, thats where the money is, and your non emergency number may just be a single pots line |
| 06:53.13 | jblack | Well, there is the luzerne county ema, and I registered with them too (they have a web page). That didn't do any good hte last time I called 911 |
| 06:53.26 | mihr | trentcreek, did you restart the box after you disabled it? |
| 06:53.35 | TrentCreek | mihr: now it is back on, but everything indicates it is diabled |
| 06:53.47 | TrentCreek | mihr: I disabled it MONTHS ago |
| 06:54.12 | drmessano | digitalirony: I call my local non-emergency number twice a week on average.. I am lucky to get through without being put on hold or getting a busy.. Sure, I have gotten RIGHT THROUGH a time or two.. but is that good enough for SAVING MY LIFE? No! |
| 06:54.38 | digitalirony | drmassano: your life sucks |
| 06:54.40 | drmessano | In an emergency, it will fail |
| 06:54.44 | jblack | what drives me crazy about my police department is that any time you cann the non emergency number, they want you to dial 911. |
| 06:54.46 | drmessano | digitalirony: Why does it suck? |
| 06:54.48 | mihr | trentcreet: how may I know if you don't restart the service after you disabled it even after a MONTH ago. |
| 06:54.53 | drmessano | digitalirony: Who is being the jerk now? |
| 06:55.11 | digitalirony | drmessano: i was making a pun on how often you call the police |
| 06:55.13 | mihr | jblack: correct! |
| 06:55.24 | jblack | Looks like someone is robbing the place across the street, call 911. Drunk guy driving over curbs, call 911. Person blocking your driveway, call 911 |
| 06:55.27 | TrentCreek | It was fine until I installed Firestarter |
| 06:55.37 | drmessano | digitalirony: I call in drunk drivers, road hazards, etc, all the time.. I do it as a public service |
| 06:55.48 | digitalirony | damn |
| 06:56.05 | digitalirony | that sucks.... |
| 06:56.12 | TrentCreek | I looked at the configs, even GUIs are greys out because it is diables, but yet there it is affectign my apps |
| 06:56.12 | digitalirony | now i know people really do that |
| 06:56.17 | jblack | They even took the police numbers out of the phone book and replaced every damned one with huge "911" numbers. |
| 06:56.20 | digitalirony | cause sometimes i do drive drunk |
| 06:56.24 | friendly12345 | ...people who use Internet Explorer, people who don't wash.... |
| 06:56.25 | digitalirony | but i don't do it on purpose |
| 06:56.28 | creativx | any way to purge all dynamic queue members from a given queue? |
| 06:56.30 | digitalirony | im just blacked out |
| 06:56.44 | drmessano | facepalms |
| 06:56.52 | drmessano | Are you like, 17? |
| 06:57.04 | jblack | wtf? Isn't that supposed to be for emergencies only? Aren't you supposed to get into trouble if you call them for something that isn't an emergency, because someone that's dying, could... die? |
| 06:57.06 | digitalirony | no, im just really tired |
| 06:57.25 | unpaidbill | ~logs |
| 06:57.28 | jbot | i guess logs is apt/ibot/infobot/jbot/purl all log daily to http://ibot.rikers.org/<channelname>/ where channelname is html encoded ie: %23debian | lines that start with a space are not shown | some channels have stats at http://ibot.rikers.org/stats/<channelname>.html.gz |
| 06:57.38 | jblack | You're uh... blacked out? |
| 06:58.02 | TrentCreek | j blacked out |
| 06:58.03 | digitalirony | i have been sleeping on a couch in the living room of a new apt, and working 3rd shift all of a sudden i get no sleep in the room once the sun comes out, so i sleep maybe 3 hours a night, so i am a bit lucid right now |
| 06:58.04 | drmessano | 911 is for emergencies.. but most places enforce it very little |
| 06:58.09 | drmessano | They do enforce it though |
| 06:58.18 | digitalirony | drmassano: im sure |
| 06:58.21 | mihr | TrentCreek: can you uninstall the firestarter? |
| 06:58.23 | drmessano | My sister had a 1 year old who kept dialing 911 |
| 06:58.30 | jblack | That's rather my point. The police department no longer publishes any number _but_ 911. |
| 06:58.33 | drmessano | They showed up knowing it was a kid. |
| 06:58.56 | digitalirony | yeah but when you live in a small town its not hard to get the number to anything |
| 06:59.09 | TrentCreek | mihr: I did a while ago |
| 06:59.16 | digitalirony | some one you know, knows some one who works where you want to call |
| 06:59.22 | digitalirony | and everyone orders pizza |
| 06:59.24 | jblack | Ironically, that's the only number I ever had trouble finding.. the non-emergency number to the police department. |
| 06:59.30 | digitalirony | and i delivered pizza |
| 06:59.36 | jblack | It's not that small of a city anyways. perhaps 40k |
| 06:59.41 | drmessano | jblack: Part of that too is a liability issue |
| 06:59.49 | mihr | TrentCreek: then try to restart selinux |
| 06:59.51 | jblack | well, too small to be a real city, too large to be a town. You know the sort |
| 06:59.59 | TrentCreek | mihr: Clearly it changed something, but the firewall and SE Linux are clearly disabled |
| 07:00.00 | digitalirony | yep |
| 07:00.06 | drmessano | jblack: Someone calls that number, dies.. sues becuase no one answered.. Thats part of my argument here. |
| 07:00.23 | TrentCreek | mihr: I cannot restart it..it is disabled |
| 07:00.25 | jblack | I watch rush hour go by my house every night.. it lasts from 4:55 pm to 5:00 pm. |
| 07:00.37 | mihr | treentcreek: what applicationare affected anyway? |
| 07:01.02 | jblack | Hell. I even live on main street, and have my very own maple tree. |
| 07:01.15 | digitalirony | drmessano: mine is, that its the best thing to have if you made it yourself and you know who you are calling, and its a way to call, hell you could make a 912 for the ems number or something |
| 07:01.39 | digitalirony | drmessano: and you could know better when to call and not call that and run next door |
| 07:01.54 | Strom_C | i hope you're kidding about 912, given that it's an area code and a prefix and all that |
| 07:01.55 | digitalirony | its giving yourself a way that you wouldn't have any other time |
| 07:01.56 | MrNaz | ok so i have asterisk running and installed and i think i have a valid dialplan... how do i start it so i can dial in and see if it works? |
| 07:02.02 | JT | run next door, when there is an armed attacker in your house? |
| 07:02.03 | JT | good idea |
| 07:02.09 | Strom_C | MrNaz: "asterisk -cvvvvg" |
| 07:02.21 | TrentCreek | mihr: with the funny command? |
| 07:02.33 | drmessano | Who has time to go .. "Hmm.. should I try the non emergency number... or.... go next door so I wont die.. Hmmmm" |
| 07:02.39 | *** join/#asterisk bmg505 (n=leon@196-209-76-219-tbnb-esr-2.dynamic.isadsl.co.za) |
| 07:02.43 | digitalirony | Strom_C: I understand you, it was just something like a formula without any math |
| 07:03.00 | drmessano | I know *I* always take 30 or 45 seconds to assess immediate life or death situations.. |
| 07:03.02 | mihr | TrentCreek: how did you disable your selinux? |
| 07:03.08 | digitalirony | Strom_C: I will try to be more clear next time |
| 07:03.13 | Strom_C | digitalirony: sigh...don't be more of a moron than is necessart |
| 07:03.16 | Strom_C | s/rt/ry/ |
| 07:03.44 | MrNaz | Strom_C when i type that, it says asterisk is already running... it also says this: Parsing '/etc/asterisk/extconfig.conf': Found is that the file my dialplan needs to go into, or extensions.conf ? |
| 07:03.46 | drmessano | Strom_C: Needs to be a LOLCAT that says "Where is your 912 now?" |
| 07:04.08 | Strom_C | MrNaz: if asterisk is already running, try "asterisk -r" |
| 07:04.33 | creativx | is there any string function to check for ${str1] inside ${str2} ? |
| 07:04.47 | MrNaz | Strom_C yea that seems to just output the status... what else will it do? |
| 07:05.02 | digitalirony | MrNaz: your dialplan can go into /etc/asterisk/extensions.conf usually |
| 07:05.02 | Strom_C | MrNaz: it should connect you to the CLI... |
| 07:05.26 | drmessano | I any case.. Wasted enough time in here tonight.. digitalirony, for the record, I never called you stupid.. but I do think your idea is flawed.. tried to give you some insight into why, but hey, as jblack says, I am just a troll.. so nevermind me </emo> |
| 07:05.29 | jblack | There are 1 types of people. Those that know boolean, and those that don't. :) |
| 07:05.32 | MrNaz | Strom_C oh yea... it has... what can i do at the cli? |
| 07:05.49 | Strom_C | MrNaz: watch what happens as you place calls. |
| 07:05.58 | MrNaz | aah |
| 07:05.59 | TrentCreek | mihr: echo 1 > /selinux/enforce ? |
| 07:06.11 | MrNaz | Strom_C well lemme paste my dialplan |
| 07:06.17 | Strom_C | oh god |
| 07:06.18 | Strom_C | hang on |
| 07:06.24 | digitalirony | MrNaz: you need to learn to set the verbose in the CLI so you can get different info |
| 07:06.25 | Strom_C | do you have any phones connected? |
| 07:06.29 | jblack | MrNaz: Oh, set debug 9, and verbose 9 |
| 07:06.32 | digitalirony | try the show command with tab |
| 07:07.02 | MrNaz | i have the pstn line connected into an fxo port of the asterisk box |
| 07:07.27 | Strom_C | MrNaz: so call it and see what happens |
| 07:07.44 | MrNaz | Strom_C nothing |
| 07:07.50 | MrNaz | do i need to turn verbosity up ? |
| 07:08.00 | Strom_C | set verbose to 10 and then tell us what happens |
| 07:08.07 | MrNaz | also... lemme get a pstn handset and make sure the line was patched properly |
| 07:08.13 | MrNaz | how do i change verbosity ? |
| 07:08.20 | Strom_C | "set verbose 10" |
| 07:08.42 | Strom_C | MrNaz: you'll need more than just the handset; you'll need the whole telephone. |
| 07:09.07 | digitalirony | also when you start the asterisk cli using "asterisk -r" you can do "asterisk -rvvvvvvvvvv" where the number of v = the verbose level |
| 07:09.11 | MrNaz | Strom_C yeap... i was just going to grab a pstn phone and put it on the line and ring it to make sure the line is working |
| 07:09.15 | mihr | TrentCreek: vi /etc/selinux/config then edit the line "selinux=disabled" then restart your box to be sure that it will take effect |
| 07:09.51 | MrNaz | hmm... i call the pstn still no movement at the asterisk CLI... lemme get that phone |
| 07:10.18 | digitalirony | MrNaz: do you even see it when you pick up the phone? |
| 07:10.41 | digitalirony | MrNaz: if your using analog that is |
| 07:10.48 | TrentCreek | mihr: It's been that way Since Sept 2007, and still is |
| 07:11.01 | TrentCreek | but you want to me restart it |
| 07:11.10 | MrNaz | the line is working... i get a dialtone and it rings when i call from my cell |
| 07:11.27 | digitalirony | but you don't see anything on the asterisk console? |
| 07:11.35 | MrNaz | digitalirony i'm using analog... this is an asterisk box with 4 fxo ports |
| 07:11.36 | Strom_C | MrNaz: ok -- what happens when you type "zap show channels"? |
| 07:12.11 | MrNaz | localhost*CLI> zap show channels| Chan Extension Context Language MOH Interpret | pseudo default default |localhost*CLI> |
| 07:12.14 | mihr | TrentCreek: by doing echo 1 to enforce will turn the selinux only in permissive mode but it will not be totally disabled it |
| 07:12.21 | MrNaz | that's it |
| 07:12.27 | Strom_C | MrNaz: ok, so you didnt configure that correctly |
| 07:12.30 | Strom_C | or at all |
| 07:12.33 | digitalirony | MrNaz: use pastebin or they might complain |
| 07:12.34 | MrNaz | hmm |
| 07:12.34 | MrNaz | ok |
| 07:12.42 | Strom_C | pastebin your /etc/zaptel.conf and /etc/asterisk/zapata.conf |
| 07:12.53 | unpaidbill | i already wrote a letter. |
| 07:12.56 | MrNaz | digitalirony yes i know... i've gotten the fine art of treading the line between short pastes in the channel and pastebin down to a fine art ;) |
| 07:13.25 | Strom | gotten the fine art [...] down to a fine art |
| 07:13.36 | TrentCreek | mihr: It's been "selinux=disabled" Since Sept 2007 |
| 07:13.39 | MrNaz | Strom_C well i thought the channels were configured properly... is there a way to check the status with zapata ? |
| 07:13.47 | Strom | quietly, the English language crawled into a corner and wept itself to sleep |
| 07:14.03 | Strom | MrNaz: please pastebin those files |
| 07:14.07 | MrNaz | Strom oh... yea obviously i dont have the avoidance of tautology down to a fine art |
| 07:14.10 | digitalirony | TrentCreek: how do you know its SeLinux causing the problem |
| 07:14.40 | digitalirony | MrNaz: just do this ok |
| 07:14.51 | digitalirony | pastebin those files |
| 07:14.54 | *** join/#asterisk rootlogin (n=root@saturn2.franken.de) |
| 07:15.17 | TrentCreek | mihr: Because I keep getting SELinux warning messages that tell me it prevented certain actions |
| 07:15.53 | Strom | TrentCreek: perhaps you may get more help on an SELinux related channel |
| 07:15.56 | digitalirony | TrentCreek: what distro are you using |
| 07:16.21 | TrentCreek | Fedora 7 |
| 07:16.48 | digitalirony | TrentCreek: Strom is right |
| 07:16.58 | mihr | TrentCreek: then its not been disabled yet. it is it running. |
| 07:17.22 | TrentCreek | no..it is clearly disabled |
| 07:17.31 | MrNaz | Strom : zaptel.conf - http://rafb.net/p/n7IBkI42.html zapata.conf - http://rafb.net/p/3xljqB98.html |
| 07:17.34 | digitalirony | TrentCreek, Not if its running |
| 07:18.10 | TrentCreek | Services show it disabled..GUI shows it Grayed out....config file has DISBALED |
| 07:18.15 | TrentCreek | what is runningit? |
| 07:18.44 | digitalirony | umm |
| 07:18.45 | MrNaz | Strom wow i'm using the wrong zapata.conf |
| 07:18.55 | digitalirony | MrNaz thats all commented |
| 07:19.09 | Strom | MrNaz: did you even touch zapata.conf? this looks like the default one that "make samples" shits into your /etc/asterisk directory |
| 07:19.39 | Strom | (ah, such lovely analogies) |
| 07:19.46 | MrNaz | Strom yes it is.... i have a proper one that i went through with jaytee yesterday and i dont know where that went |
| 07:20.05 | digitalirony | context=something |
| 07:20.05 | digitalirony | signalling =>fxs_ks |
| 07:20.05 | digitalirony | channel =>1-4 |
| 07:20.27 | Strom | well, you know what to do |
| 07:20.30 | Strom | go find it |
| 07:20.34 | Strom | points |
| 07:20.40 | Strom | plays heroic music |
| 07:21.10 | Strom | god damnit, who replaced my heroic music with britney spears |
| 07:21.13 | digitalirony | * signalling = fxs_ks |
| 07:21.46 | MrNaz | Strom / digitalirony : http://rafb.net/p/Stf1kU20.html <-- that's the zapata.conf that i just put there.... look ok? |
| 07:22.04 | Strom | yes |
| 07:22.34 | digitalirony | that looks ok..... |
| 07:22.35 | MrNaz | Strom britney spears *is* heroic music, if you're a 13 year old with adhd... so look for the 13 year old female staff members with adhd if you want your list of suspects |
| 07:22.43 | MrNaz | Strom ok so how do i restart asterisk ? |
| 07:22.52 | Strom | at the CLI: "restart now" |
| 07:23.02 | digitalirony | asterisk -rx "restrt now" |
| 07:23.11 | digitalirony | asterisk -rx "restart now" |
| 07:23.25 | Maliuta | or kill -HUP, or pkill, or sysvinit |
| 07:23.37 | Strom | echo "balls" |
| 07:23.42 | Strom | init hellp |
| 07:23.45 | Strom | s/p/o/ |
| 07:23.49 | Strom | atdt |
| 07:23.51 | MrNaz | ok done |
| 07:23.55 | MrNaz | set verbosity to 10 |
| 07:23.58 | MrNaz | lets make a call |
| 07:24.20 | TrentCreek | Sure Dave |
| 07:24.24 | Strom | 555-2368 riiiing hello click if you'd like to make a call honk honk honk honk honk |
| 07:24.25 | MrNaz | zap show channels shows 4 channels eg: 4 incoming default |
| 07:24.29 | MrNaz | here goes nothing... |
| 07:24.35 | TrentCreek | let's call Mayor Bloomburg |
| 07:25.41 | MrNaz | w00t |
| 07:25.45 | MrNaz | life at the CLI |
| 07:26.00 | digitalirony | MrNaz: ok. does the call work? |
| 07:26.03 | MrNaz | ye |
| 07:26.05 | MrNaz | yea |
| 07:26.08 | MrNaz | it answered the call |
| 07:26.12 | MrNaz | but now it wont hang it up heh |
| 07:26.15 | digitalirony | MrNaz: congrats |
| 07:26.20 | MrNaz | time for dialplanning |
| 07:26.49 | digitalirony | yeah |
| 07:27.08 | MrNaz | so from here on out, all i need to do is make the dialplan for any system ? |
| 07:27.28 | digitalirony | MrNaz: what do you mean |
| 07:27.44 | Strom | pricks bonking |
| 07:27.57 | MrNaz | well the hardware is answering calls etc, i just need to tell it what to do when calls come in etc right? |
| 07:28.01 | MrNaz | there's no more set up involved? |
| 07:28.02 | digitalirony | MrNaz: you have to set the context for each line |
| 07:28.17 | digitalirony | MrNaz: and you can make it change based on callerid all sorts of neat stuff |
| 07:28.46 | *** join/#asterisk mikkel (n=mikkel@130.226.36.170) |
| 07:29.23 | digitalirony | MrNaz: /etc/asterisk/zapata.conf you can also make other context for each FXO line |
| 07:29.44 | Strom | digitalirony: jesus, don't overwhelm the guy just yet |
| 07:29.57 | digitalirony | Strom: sorry |
| 07:30.01 | MrNaz | yea... but this is my first system, and the ultimate home is on a 4 line office with just 4 lines in a line hunt configuration |
| 07:30.03 | digitalirony | Strom: he asked |
| 07:30.15 | digitalirony | Well this is all i ment |
| 07:30.17 | Strom | digitalirony: not really |
| 07:30.22 | MrNaz | so there's no need for anything fancy on this box... this was a learning experience for me to get * up and running... i'll get creative once i get my head properly around all this shit |
| 07:30.40 | *** join/#asterisk af_ (n=getsmart@88-149-230-57.dynamic.ngi.it) |
| 07:30.47 | digitalirony | sec |
| 07:31.09 | unpaidbill | periodograms make my head hurt |
| 07:31.12 | unpaidbill | aieeee |
| 07:31.21 | Strom | who's up for half-chewed starbursts!? |
| 07:31.37 | unpaidbill | im up for some half cocked C |
| 07:31.43 | MrNaz | Strom ok so now that i have a working * box, i need to write a dialplan that answers the line, routes the call to a connected SIP phone, and then hangs up when either party hangs up... that's all dialplan right ? |
| 07:31.49 | *** join/#asterisk Gwayne (n=Gwayne@cm196.kappa245.maxonline.com.sg) |
| 07:32.07 | Strom | MrNaz: pretty much...although if you're just dialing a phone, better to let the phone do the answering |
| 07:32.27 | Strom | don't answer() unless you have an explicit reason to do so |
| 07:32.38 | MrNaz | Strom right... so i can use the dialplan to make all the SIP phones ring, and then route the call to whichever one picks up first right ? |
| 07:32.48 | MrNaz | and all that logic goes in the dialplan ? |
| 07:32.48 | digitalirony | MrNaz: simply do this |
| 07:32.49 | Strom | basically |
| 07:33.56 | MrNaz | is the book a good reference for dialplan skills or should i google for a tutorial ? |
| 07:34.09 | Strom | the book should be fine for now :) |
| 07:34.20 | unpaidbill | the book is kick ass. |
| 07:34.26 | digitalirony | http://rafb.net/p/OBZCda87.html |
| 07:35.24 | MrNaz | digitalirony what's with the first 19 lines of integers? |
| 07:35.33 | digitalirony | lol, i don't know..... |
| 07:35.43 | Strom | digitalirony: what the fuck |
| 07:35.46 | Strom | honestly |
| 07:35.47 | digitalirony | MrNaz: I didn't put them there |
| 07:35.56 | Strom | don't overcomplicate shit for him at this point |
| 07:36.04 | Strom | MrNaz: ignore this crap :) |
| 07:36.11 | Strom | do your dialplan thing |
| 07:36.11 | MrNaz | hehe |
| 07:36.11 | digitalirony | Strom: thats not overcomplicated |
| 07:36.20 | Strom | facepalms |
| 07:36.59 | digitalirony | Strom: thats how they started us off at the bootcamp :( |
| 07:37.30 | Strom | must not be the bootcamp I wrote |
| 07:37.45 | digitalirony | iunno it just went quick |
| 07:37.46 | MrNaz | digitalirony forgive me if i choose to listen to Strom :) he's been a fantastic help to my travails thus far |
| 07:37.51 | Strom | mine was a bit simpler :) |
| 07:38.51 | digitalirony | MrNaz: go for it. |
| 07:39.18 | digitalirony | MrNaz: but it wouldn't hurt you to try to make since of the difference of what i gave you and what you have |
| 07:39.42 | MrNaz | digitalirony yeap... in fact i'm doing that now |
| 07:40.09 | MrNaz | if i'm interpretting it correctly, you're putting each line onto a different context so it can be referred to individually |
| 07:40.10 | MrNaz | right ? |
| 07:41.50 | digitalirony | yep |
| 07:42.00 | digitalirony | so maybe you can have a number just you know |
| 07:42.25 | digitalirony | MrNaz: i would still put auth on it....but thats complicated |
| 07:43.08 | digitalirony | MrNaz: it just gives you a way to split things. and understand the call flow a little better |
| 07:43.39 | MrNaz | ok... yeap i do get that conceptually |
| 07:44.09 | MrNaz | but i'll stick to a single context until i understand things more thoroughly and i'm familiar with dialplan creation |
| 07:44.25 | digitalirony | yeah |
| 07:44.46 | digitalirony | try something simple |
| 07:44.46 | digitalirony | try this |
| 07:45.04 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
| 07:46.10 | digitalirony | http://rafb.net/p/LBmr1a77.html |
| 07:46.21 | digitalirony | /etc/asterisk/extensions.conf |
| 07:46.42 | *** join/#asterisk synthetiq (n=roger@unl201395.nl.customer.alter.net) |
| 07:47.04 | digitalirony | Put that in there and try calling yourself |
| 07:49.36 | TrentCreek | no..it's"put that in your pipe and smoke it!" |
| 07:50.10 | digitalirony | Actually: that would be if i wanted to say it in a rhetorical way |
| 07:50.14 | digitalirony | and i don't |
| 07:50.22 | TrentCreek | lol |
| 07:52.16 | *** join/#asterisk MrNaz (n=naz@ppp59-167-151-3.lns4.mel6.internode.on.net) |
| 07:54.19 | digitalirony | heh oh yeah MrNaz: don' |
| 07:54.34 | digitalirony | don't forget to do a reload in asterisk |
| 07:56.52 | *** join/#asterisk gr0mit (n=tim@dhcp3.zuk40.mot-tools.co.uk) |
| 08:01.43 | *** join/#asterisk LuisTorres (n=chatzill@bl6-209-130.dsl.telepac.pt) |
| 08:01.53 | LuisTorres | Morning |
| 08:02.07 | Strom | but seriously, what about y3k and y4k |
| 08:02.07 | unpaidbill | Evening |
| 08:02.12 | TrentCreek | Morgen |
| 08:02.19 | TrentCreek | Buenas dias |
| 08:02.52 | unpaidbill | ahiahi |
| 08:03.54 | TrentCreek | no hawaiian |
| 08:04.08 | unpaidbill | yes. |
| 08:04.28 | Strom | bluhahaubhblablbalblaalhlaulhbah |
| 08:04.53 | unpaidbill | yes that is how hawaiians sound |
| 08:04.57 | unpaidbill | pidgin is a hell of a vernacular |
| 08:05.59 | *** join/#asterisk absd (n=a@124-168-95-240.dyn.iinet.net.au) |
| 08:06.06 | synthetiq | /em contemplates how to play a custom ring sound of 180/183 one way sdp |
| 08:06.12 | synthetiq | ../me |
| 08:06.25 | unpaidbill | what is sdp |
| 08:06.36 | unpaidbill | session description protocol? |
| 08:06.39 | Strom | small dicks poking |
| 08:06.50 | unpaidbill | hey lets not make fun of asians now |
| 08:07.14 | Strom | asians? ive seen small dicks of all colors. |
| 08:07.37 | unpaidbill | yes but the mean for asian dicks is way down there |
| 08:07.38 | Strom | i'm singling no one group out |
| 08:07.45 | Strom | um...sure. |
| 08:07.48 | unpaidbill | haha |
| 08:08.00 | Strom | i'm going to wager that I have more experience with dicks than you do |
| 08:08.12 | unpaidbill | i have a lot of experience with one dick |
| 08:08.17 | absd | I've got a problem -- got an analog asterisk setup (both fxo and fxs analogs) ... when I use the 'redial' key from my analog handset, it dials the number (plus all the extra numbers I entered from the IVR prompts) and then fails to connect the call... The same analog phone straight into a pstn line the redial works ... |
| 08:08.20 | synthetiq | yes basicly want to play a custom ring tone on one way audio session so call isnt answered |
| 08:08.37 | Strom | synthetiq: play music on hold that sounds like a custom ring tone |
| 08:08.44 | Strom | or screw around with indications.conf |
| 08:09.33 | Strom | absd: pastebin your extensions.conf |
| 08:10.01 | synthetiq | i tried musiconhold |
| 08:10.29 | synthetiq | but then calls arent answered when i used a goto via perl agi |
| 08:14.50 | *** join/#asterisk ^shark_ (n=^shark_@41.222.2.65) |
| 08:14.58 | Strom | absd: helloooooooo |
| 08:15.13 | absd | Strom: how do I pastebin (sorry for newb question) |
| 08:15.20 | Strom | ~pb |
| 08:15.21 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 08:15.34 | *** join/#asterisk phix (n=threat@123-243-44-131.tpgi.com.au) |
| 08:17.09 | absd | http://pastebin.com/d25e62b9b |
| 08:17.51 | Strom | absd: your pattern matching is kind of brain-dead |
| 08:17.55 | Strom | who wrote this nonsense? |
| 08:18.10 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 08:18.12 | absd | Not me *lol* ... i'm the idiot trying to fix it though. |
| 08:18.22 | unpaidbill | stake -> heart |
| 08:18.30 | absd | If you could lead me in the right direction I'd be much appreciative |
| 08:18.41 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-196c35eb9d4c0f0b) |
| 08:18.51 | unpaidbill | N is the shit |
| 08:18.58 | Strom | absd: instead of having _X. as your outbound pattern match, you should write a pattern match that fits your country's numbering plan |
| 08:19.48 | absd | Umm, the _x one is commented out, isn't it? |
| 08:20.09 | absd | and the others seem to vaguely match at least some of australias number plans |
| 08:20.26 | unpaidbill | the fact that they exist, commented out or not, is a travesty |
| 08:20.39 | unpaidbill | strom is doing everything he can possibly do to keep himself from bashing you in the face |
| 08:20.55 | absd | Whatever helps |
| 08:21.02 | unpaidbill | also, outgoing_test :P |
| 08:21.51 | Strom | this dialplan would be best if rewritten |
| 08:21.55 | unpaidbill | # |
| 08:21.56 | unpaidbill | exten = _[123456789]XXXXXXX!,1,Macro(trunkdial,Zap/g1/${EXTEN}) |
| 08:22.01 | unpaidbill | that is totally wacky though |
| 08:22.14 | unpaidbill | are you in the usa? |
| 08:22.19 | absd | australia |
| 08:22.21 | unpaidbill | oh, nm then |
| 08:22.26 | unpaidbill | it's probably not wacky then |
| 08:22.28 | unpaidbill | hehe |
| 08:22.31 | absd | lol |
| 08:22.33 | Strom | who the hell writes [123456789] instead of Z? |
| 08:22.45 | unpaidbill | well |
| 08:22.52 | unpaidbill | i was just wondering about a 10 digit starting with 1 |
| 08:22.53 | unpaidbill | but yeah |
| 08:22.55 | unpaidbill | <PROTECTED> |
| 08:23.07 | Strom | seriously though |
| 08:23.10 | unpaidbill | haha |
| 08:23.21 | *** join/#asterisk mikkel (n=mikkel@130.226.36.170) |
| 08:23.33 | absd | sure, so how about my original question |
| 08:23.33 | Strom | this dialplan is somewhat ridiculous |
| 08:24.06 | Strom | get rid of all the silly "match everything under the sun" pattern matches and replace them with patterns that mirror the AU numbering plan |
| 08:24.44 | absd | line's 96 through 100 seem to do that already |
| 08:25.29 | Strom | nope |
| 08:25.32 | Strom | they terminate in a ! |
| 08:25.51 | Strom | which will match anything from "" to "98198120953187319175938730784731958134573145931" |
| 08:25.55 | absd | that ! termination has been put in there directly in response to this redial issue -- they never used to exist |
| 08:25.56 | Strom | etc etc etc |
| 08:26.16 | Strom | the ! means "match zero or more characters" so I don't see how you thought that might solve the problem |
| 08:27.04 | absd | eg, analog phone calls 132381 ... hits an ivr and dials 123456 ... then hangs up... Then the person hits "redial" on that analog line... the number asterisks now receives is 132381123456 ... I want that to actually dial 132381 again |
| 08:27.06 | unpaidbill | i thought it was like, EXTENSION! VERY EXCITING EXTENSION! |
| 08:27.22 | Strom | absd: I know what the problem is |
| 08:27.32 | Strom | unpaidbill: heh |
| 08:27.44 | unpaidbill | no really i did |
| 08:27.48 | unpaidbill | i wasnt trying to be funny |
| 08:27.57 | Strom | ok then |
| 08:28.21 | absd | so even if I remove all the !'s ... how is 132381123456 going to hit the pstn with the right number dialing? |
| 08:28.26 | Strom | absd: seriously -- clear out all the outbound pattern matching and then start fresh |
| 08:28.58 | Strom | because if you have _NXXXXXX and dial 55523687777, asterisk /should/ match once you hit "8" |
| 08:29.04 | absd | Strom: I would do that if I had enough clue as to what I'd be doing differently... My current (limited) understanding of this is that most of that looks reasonable |
| 08:30.25 | *** join/#asterisk jicksta (n=jicksta@75-101-5-48.dsl.static.sonic.net) |
| 08:31.00 | absd | umm... so australia has "13xxxx" numbers... and 1300xxxxxx numbers... if it did what you're saying, how would I get line 97 to ever match rather than line 96 stealing it as a 13xxxx number? |
| 08:31.23 | Strom | does 13xxxx ever have a zero immediately follow the 3? |
| 08:32.18 | absd | Not sure, do you know if there's somewhere that contains a list of rules for australia? |
| 08:32.29 | unpaidbill | anpa? |
| 08:32.44 | absd | anpa? |
| 08:32.53 | Strom | www.dbce.gov.au |
| 08:32.55 | Strom | er |
| 08:32.59 | Strom | www.dbcde.gov.au |
| 08:33.23 | *** join/#asterisk KermitTheFragger (n=siepkes@118-197.bbned.dsl.internl.net) |
| 08:33.45 | Strom | unpaidbill: NANPA doesn't have jurisdiction over australia |
| 08:33.48 | unpaidbill | here's a dialplan from a linksys.. easily convertable |
| 08:33.49 | unpaidbill | (1xxS0|0,000S0|0,13xxxxS0|0,1[389] 00xxxxxxS0|0,0[23478]xxxxxxxxS0|0, [1-9]xxxxxxxS0|0,00111xxxxxxxxxxS0 |0,0011x.) |
| 08:34.07 | unpaidbill | yea strom, na = north american, a = australian! it's flawless logic |
| 08:34.16 | Strom | facepalms |
| 08:34.35 | unpaidbill | http://whirlpool.net.au/wiki/?tag=LinksysPAP2_AusPBX_dialplan ! |
| 08:34.39 | absd | not suprisingly that linksys one looks remarkably close to what's there |
| 08:35.01 | unpaidbill | well the guy could be a retard, i just took the first link i found |
| 08:35.50 | Strom | found it |
| 08:35.52 | Strom | http://www.comlaw.gov.au/ComLaw/Legislation/LegislativeInstrumentCompilation1.nsf/all/search/F51C1D3514017E6DCA2573CA0020C24F |
| 08:35.55 | absd | i can't seem to find dialplan info on that dbcde site -- any idea where it is in there (or a deep link?) |
| 08:36.42 | ^shark_ | ~CLAControl |
| 08:36.53 | ^shark_ | ~ CLAControl |
| 08:37.01 | Strom | don't tell anyone that the Californian found the document for you ;) |
| 08:37.17 | Strom | ^shark_: please do that in a pm to the bot |
| 08:37.20 | unpaidbill | and dont ever call it cali |
| 08:37.23 | unpaidbill | for the love of god |
| 08:37.35 | Strom | we don't need you and the bot making out in public |
| 08:37.38 | Strom | unpaidbill: heh |
| 08:37.48 | ^shark_ | hi friends, i have tried the voip site for CLAControl but i cant find it anywhere!? |
| 08:37.53 | absd | still, even with that, not enntirely sure how it's going to help me get around the redial issue |
| 08:38.10 | ^shark_ | can someone give me some url i can study this app. |
| 08:38.15 | *** join/#asterisk tengulre (n=tengulre@125.71.208.16) |
| 08:38.16 | Strom | absd: because you're specifying specific matches instead of big general ones |
| 08:38.20 | ^shark_ | Strom, how do i do that |
| 08:38.46 | absd | strom, so is there anything I'm changing other than ripping out the !'s? |
| 08:38.48 | Strom | ^shark_: read your IRC client's manual |
| 08:39.14 | Strom | absd: um, i would recommend taking it all out and then building from scratch |
| 08:39.32 | unpaidbill | wtf, does australia have a different charset than usa? |
| 08:39.44 | Strom | unpaidbill: ? |
| 08:39.45 | unpaidbill | did you look at the html link to their numbering plan |
| 08:40.04 | absd | Strom: great, happy to do that -- can you give me an example of how you'd write one line of it? |
| 08:40.37 | absd | unpaidbill: nah, aussies use american standard 104 keyboards, why? |
| 08:40.48 | unpaidbill | i dunno, something isnt right on their page |
| 08:40.54 | Strom | absd: i dont know the AU numbering plan |
| 08:40.56 | unpaidbill | i'm getting all sorts of messed up characters |
| 08:41.05 | unpaidbill | the pdf is OK though |
| 08:41.12 | Strom | but for a north american number like +1 311 555 2368 |
| 08:41.20 | Strom | _1NXXNXXXXXX |
| 08:42.24 | Strom | http://www.flickr.com/photos/stromcarlson/1765974872/in/set-72157602726615878/ |
| 08:42.27 | ^shark_ | Strom: ok-- |
| 08:42.29 | absd | Surely someone has already written an aussie dialplan to match pstn numbers like this before... Why am I trying to reinvent the wheel, surely there's a db of this stuff somehwere |
| 08:43.03 | ^shark_ | Strom: anywhere i can find info on the CLAControl app.? |
| 08:43.10 | unpaidbill | people dont like to share their secrets |
| 08:43.18 | unpaidbill | such as beep detection :P |
| 08:43.30 | Strom | ^shark_: i havent ever heard of it |
| 08:43.35 | absd | beep detectioN? |
| 08:43.39 | Strom | is it supposed to be standard in asterisk? |
| 08:44.41 | ^shark_ | Strom: it stands for call control e.g outgoing calls |
| 08:45.54 | ^shark_ | Strom: i am trying to figure out how i can set passwords to specific extensions when making outgoing calls, any extension with no proper password should not make outgoing calls |
| 08:46.27 | Strom | and for this you need a whole app? |
| 08:46.31 | Strom | pffft |
| 08:46.33 | Strom | do it in dialplan |
| 08:46.38 | BeeBuu | anyone know is asterisk support H248？ |
| 08:46.42 | *** join/#asterisk railsmunky (n=nick@82-70-72-101.dsl.in-addr.zen.co.uk) |
| 08:46.52 | unpaidbill | authenticate, shark? |
| 08:47.03 | *** join/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 08:47.10 | unpaidbill | vmauthenticate? |
| 08:47.39 | railsmunky | Hi people. I'm having BT reporting that we have stuck channels on our PRI. Would that happen at the card end (sangoma a102) or is it a BT switch problem? |
| 08:48.02 | Strom | railsmunky: might be a card thing |
| 08:48.03 | ^shark_ | Strom: any resource i can find abut this? |
| 08:48.15 | Strom | ^shark_: perhaps you should read the book |
| 08:48.18 | railsmunky | Strom: how would i check that? |
| 08:48.34 | Strom | also, jesus christ, starting your name with a ^ makes typing it an incredibly irritating pain in the ass |
| 08:48.49 | Strom | railsmunky: perhaps run "zap show channels"? |
| 08:48.54 | absd | hmm.... so is a catchall like exten => _XXXXXXX.,1,Dial(iax2/donie/${EXTEN}) a better idea for catching those numbers that are 'really long because someone used an IVR after dialing the number' |
| 08:49.00 | Strom | or "core show channels"? |
| 08:49.11 | Strom | absd: no no no no no no no and no |
| 08:49.16 | Strom | you didnt listen to me |
| 08:49.32 | absd | I listened, just didn't understand. Big difference :) |
| 08:49.33 | Strom | i told you to avoid big fat generic pattern matches like that one |
| 08:49.51 | railsmunky | Strom: zap show is giving all the configured channels core show is giving me 9 channele (is that active or available?) |
| 08:50.09 | Strom | core show channels should be active calls |
| 08:52.30 | *** join/#asterisk xacatecas (n=jkroon@dsl-242-222-63.telkomadsl.co.za) |
| 08:52.53 | xacatecas | ok, here is one to munch on, iax2 registration between asterisk 1.4.19.1 and 1.4.21 seems to be broken, it keeps on deregistering. |
| 08:53.00 | xacatecas | i'll be back later to see if I can figure that one out. |
| 08:53.12 | Strom | what the fuck was the point of that |
| 08:53.17 | Strom | don't do that, ever |
| 08:53.19 | Strom | idiots |
| 08:53.20 | Strom | grr |
| 08:53.21 | unpaidbill | haha |
| 08:53.26 | unpaidbill | he's going to check the irc logs |
| 08:53.55 | Strom | yells something highly offensive in Afrikaans and throws rotten boerwors |
| 08:54.17 | absd | ok.. hmm... so avoid big fat pattern match... so when asterisks sees 13238156789 with a patter match of 13xxxx it should pass the whole thing to pstn as soon as the "132381" part is ddialed, and have the whole 13238156789 dialed on pstn? |
| 08:54.18 | unpaidbill | !xa kx? |
| 08:54.36 | BeeBuu | is asterisk support H.248? |
| 08:55.09 | Strom | absd: no |
| 08:55.19 | Strom | you should read up on how pattern matching works, i think |
| 08:55.21 | Strom | ~book |
| 08:55.22 | 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 |
| 08:55.24 | unpaidbill | mgcp ? |
| 08:56.18 | unpaidbill | http://www.voip-info.org/wiki-Asterisk+MGCP+channels |
| 08:57.26 | *** join/#asterisk railsmunky (n=nick@collaboration.capuk.org) |
| 08:59.04 | *** join/#asterisk jarod14 (n=jarod14@LMontsouris-152-63-1-19.w80-12.abo.wanadoo.fr) |
| 08:59.45 | absd | Strom: ok, I've read the pattern matching section. How are you suggesting 13238156789 get's sent to the pstn (where the actual phone number is the 132381 part, and the 56789 is a random part based on what the person dialed into the IVR) if I don't use a big fat catch-all? |
| 08:59.48 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
| 09:00.12 | Strom | absd: the point is that you DON'T dial the IVR prompts back at the PSTN |
| 09:00.19 | Strom | you have to allow time for the call to set up |
| 09:00.30 | Strom | and then you let the user re-enter the prompts |
| 09:01.08 | *** join/#asterisk oilinki (n=oil@ppp-124-120-4-61.revip2.asianet.co.th) |
| 09:01.13 | absd | but I'm hitting "redial" on my analog phone handset which has memorised "13238156789" ... when I have that handset plugged into pstn, it connects me to "132381" and ignores the rest of the stuff |
| 09:01.48 | Strom | yes |
| 09:01.56 | Strom | the PSTN is ignoring everything past 81 |
| 09:02.05 | Strom | as your PBX should be doing |
| 09:02.52 | absd | I'm totally confused as to how to do that... |
| 09:03.42 | Strom | I've told you how three times now |
| 09:03.48 | Strom | write specific pattern matches |
| 09:03.54 | Strom | not big fat generic ones |
| 09:04.24 | absd | so you're saying 13xxxx will match 13238156789? |
| 09:04.29 | Strom | NO |
| 09:04.31 | Strom | grr |
| 09:04.32 | Strom | listen |
| 09:04.44 | Strom | 13XXXX will match 132381 |
| 09:04.46 | *** join/#asterisk jks (i=jks@193.189.93.254) |
| 09:04.48 | Strom | the rest will be DISCARDED |
| 09:05.04 | Strom | the call will go through after the sixth digit is dialed |
| 09:05.08 | BeeBuu | no one know asterisk support h.248 or not? |
| 09:05.15 | absd | ohhhh... really? cool. |
| 09:05.36 | *** join/#asterisk Mavvie (n=edwin@ppp121-44-49-136.lns10.syd7.internode.on.net) |
| 09:06.47 | ^shark_ | thinking about dinner |
| 09:06.56 | absd | Strom: thankyou for being patient enough to explain (again... and again...) I do get it now, I'll be able to make that work. Thanks again :) |
| 09:07.53 | Strom | fat queer angry cocks from outer space that glow and make a noise |
| 09:08.22 | absd | I'll see what I can arrange. |
| 09:09.15 | unpaidbill | what |
| 09:09.24 | unpaidbill | personifying cocks is just weird |
| 09:12.43 | *** join/#asterisk wonderworld (n=ww@ip-62-143-31-149.hsi.ish.de) |
| 09:19.47 | *** join/#asterisk trnzmeta (n=bleh@123-243-201-39.static.tpgi.com.au) |
| 09:31.30 | *** join/#asterisk pa (n=paolo@unaffiliated/pa) |
| 09:35.33 | digitalirony | Assholifying people isn't wierd? |
| 09:35.36 | digitalirony | lols |
| 09:36.10 | digitalirony | needs sleep |
| 09:37.48 | *** part/#asterisk BeeBuu (n=beebuu@219.135.42.236) |
| 09:53.01 | ^shark_ | has gone for lunch |
| 09:58.33 | *** join/#asterisk _ys (i=yuri@91.151.196.254) |
| 10:01.06 | *** join/#asterisk Maliuta (n=nikolai@kiev.lusan.id.au) |
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| 10:06.41 | *** join/#asterisk ido- (n=wtf@212.199.189.65) |
| 10:07.04 | ido- | hello. how can i transfer a call from one extension to another (while in a call) |
| 10:09.44 | *** join/#asterisk Sjeemz (i=sjeemz@nox.sjeemz.nl) |
| 10:11.56 | *** join/#asterisk RoyK (n=roy@fw.fortel.no) |
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| 10:16.28 | *** join/#asterisk pa (n=paolo@unaffiliated/pa) |
| 10:17.37 | tzafrir_laptop | ido-, what type of extension? sip? zap? |
| 10:19.30 | *** join/#asterisk angryuser (n=angryuse@d02m-213-44-187-97.d4.club-internet.fr) |
| 10:22.43 | Nasra | Hi ...any1 in here offer private support to a newbie |
| 10:22.56 | Nasra | ? |
| 10:24.44 | tzafrir_laptop | Nasra, I don't . But try being more specific |
| 10:25.21 | tzafrir_laptop | And there's of course: |
| 10:25.24 | tzafrir_laptop | ~thebook |
| 10:25.26 | 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 |
| 10:25.45 | *** join/#asterisk nefertum (n=alumno@b100853.xa.bi.ehu.es) |
| 10:26.08 | Nasra | I am reading that book since two months ago... |
| 10:26.44 | Nasra | learning alot...but... |
| 10:26.51 | nefertum | hi, i'm using asterisk 1.4, do you know where can I find doc to configure something like "three way calling" ? |
| 10:28.08 | E-bola | Hmm Asteriskwin32.com ..... |
| 10:28.15 | E-bola | have anybody tried running asterisk on windows? |
| 10:28.29 | Nasra | tzafrir_laptop: I have the SPA3102...and I going to buy the pci card as well...with a ip phone ....to get started |
| 10:28.31 | E-bola | wonders if its rubish or "ok" |
| 10:29.20 | Nasra | TDM410P... |
| 10:29.56 | tzafrir_laptop | E-bola, it's a bit dated, isn't it? |
| 10:30.11 | tzafrir_laptop | I think you can build a later asterisk on cygwin |
| 10:30.14 | E-bola | tzafrir_laptop: Version 0.66 released 02/01/2008 |
| 10:30.19 | E-bola | its based on 1.2 though |
| 10:30.22 | tzafrir_laptop | But then again: why?? |
| 10:30.44 | E-bola | tzafrir_laptop: trying to save the cost of an additional asterisk box (its a really cheap costumer) |
| 10:30.56 | E-bola | and our own box cant host them as we do not ahve enough bandwith |
| 10:31.11 | *** join/#asterisk d-k-t-2 (n=dt@125.120.133.213) |
| 10:33.31 | *** join/#asterisk ido- (n=wtf@212.199.189.65) |
| 10:33.33 | tzafrir_laptop | vmware / virtualbox is probably a better option |
| 10:33.48 | E-bola | ya i would think so too |
| 10:33.50 | nefertum | anyone knows how to configure in asterisk 1.4 the "three way calling" ? |
| 10:33.52 | tzafrir_laptop | Unless you have some extra hair |
| 10:33.54 | ido- | hello tzafrir_laptop |
| 10:33.56 | E-bola | althought i dont like the idea of asterisk sharing hw with other services |
| 10:34.21 | ido- | how can i transfer a call from one extension to another (while in a call) ? |
| 10:39.01 | Nasra | tzafrir_laptop:are you still here ?? |
| 10:47.18 | *** part/#asterisk dominic1 (n=dob@213.221.82.242) |
| 10:48.13 | nefertum | sorry, something about threewaycalling? |
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| 10:51.32 | tzafrir_laptop | Nasra, did you have a specific question? That specific hardware seems reasonable |
| 10:52.07 | *** part/#asterisk friendly12345 (n=friendly@ppp59-167-89-42.lns2.mel6.internode.on.net) |
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| 11:18.55 | MrNaz | hey al |
| 11:19.03 | MrNaz | all |
| 11:21.08 | ^shark_ | is srtp now available in the new source code |
| 11:26.06 | MrNaz | srtp? |
| 11:27.18 | *** join/#asterisk pa (n=paolo@unaffiliated/pa) |
| 11:28.20 | redax | hi, |
| 11:28.33 | redax | is there anybody using chan_alsa ? |
| 11:28.44 | kaldemar | ^shark_: doesn't seem to be yet |
| 11:29.16 | MrNaz | how do you restart asterisk ? |
| 11:29.47 | *** join/#asterisk sysreq` (n=sysreq@unaffiliated/sysreq) |
| 11:31.42 | kaldemar | MrNaz: with an init script or by "restart now" in the CLI, for example. |
| 11:31.52 | MrNaz | yea found it already |
| 11:31.52 | MrNaz | thanks |
| 11:32.04 | MrNaz | i'm about to call in my first asterisk call :) |
| 11:32.14 | MrNaz | i'm going to see if it plays the congrats message in my dialplan |
| 11:34.06 | Sjeemz | Hi, we've been using asterisk 1.2.17 with a TE110P & lots of Cisco 7960's (chan_sccp 20060408) for a few years now. Everything seems to be working fine, except for one thing: when a caller, who is also using asterisk, is transfered to another extension, that extension hears nothing but silence... has anybody ever seen something like this before? |
| 11:35.25 | ido- | d |
| 11:36.27 | ido- | oh my |
| 11:38.45 | *** join/#asterisk Mavvie (n=edwin@ppp121-44-81-220.lns10.syd6.internode.on.net) |
| 11:48.54 | *** join/#asterisk iceyp (n=icepick@firewall.unix.co.nz) |
| 11:49.35 | iceyp | hey guys, I have a cisco 7940 behind a shared internet connection where we dont have access to the router which is doing NAT, so we're unable to port forward 5060 to the phone, which means the registration times out and the user cannot receive calls. |
| 11:50.02 | iceyp | Will setting the registry expires value from 3600 to a smaller value fix this? and if so what value is fare |
| 11:51.23 | kaldemar | iceyp: you could also try to keep the session up with parameter qualify in sip.conf. |
| 11:52.03 | MrNaz | is there a minimum dialplan i can use as a base? the one that the asterisk setup leaves there is a huge mess, and i'd like to start with the minimum files necessary |
| 11:52.10 | iceyp | kaldemar mmm, let me test that thanks |
| 11:52.48 | *** join/#asterisk pa (n=pa@unaffiliated/pa) |
| 11:53.01 | iceyp | mmm, i already have qualify=yes |
| 11:53.24 | redax | hehh, mz Alsa subsys working well, still chan_alsa keep saying "chan_alsa.c: snd_pcm_open failed: No such device" |
| 11:53.39 | redax | don't you have any idea? |
| 11:53.41 | *** join/#asterisk PoWeRKiLL (n=powerkil@82.166.242.254) |
| 11:57.40 | mihr | iceyp, where does your asterisk, before the router going to the internet or after the router going to the internet? |
| 11:59.33 | *** join/#asterisk angryuser (n=angryuse@d02m-213-44-187-97.d4.club-internet.fr) |
| 12:00.40 | kaldemar | iceyp: the "yes" is a default hard coded value, you could also try giving the parameter a numeric value, which will be in milliseconds. |
| 12:02.52 | *** join/#asterisk DagMoller (n=aguirre@unaffiliated/dagmoller) |
| 12:14.15 | redax | is alsa support working at all in asterisk 1.4 ? |
| 12:16.33 | *** join/#asterisk shido6 (n=shido6@209.114.208.192) |
| 12:20.07 | kaldemar | redax: it is |
| 12:21.27 | *** join/#asterisk pbx800 (n=hel@59.90.16.163) |
| 12:21.49 | pbx800 | hello all - whats the minimum commitment to knock level3? |
| 12:22.37 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 12:22.58 | blitzrage | sounds like a question L3 could answer :) |
| 12:23.41 | pbx800 | blitzrage - no, I meant lot of them have interacted with them. Wanted to check whats the rate for level3? Last time, I checked it was 25k per month. NOt sure what is it now |
| 12:25.20 | blitzrage | pbx800: and L3 wouldn't tell you that if you asked them? |
| 12:25.24 | blitzrage | I knew what you meant |
| 12:25.33 | *** join/#asterisk juanjoc (n=juanjoc@host190.190-225-197.telecom.net.ar) |
| 12:27.22 | pbx800 | blitz - its long route if I contact them. anyway, thanks |
| 12:27.55 | pbx800 | is there an integration with voicexml to asterisk as commercial level. any tips on the integration? |
| 12:28.29 | blitzrage | is it really? I had no idea it took that long just to find out how much the min. level is... I suppose it could be a sales tactic. Used to be 10k a month back in the day.. but that was like 2 years ago |
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| 12:56.35 | pbx800 | can anyone recommend terimination providers to US other than voicepulse who can offer reliability. we have around 2k usage every month. |
| 12:57.06 | [TK]D-Fender | pbx800: Give Teliax a try |
| 12:57.51 | gr0mit | 2kwhat? |
| 12:58.05 | gr0mit | mins or $ ?! |
| 12:58.08 | pbx800 | teliax - pretty unstable. does not connect. I wouldnt recommend |
| 12:58.10 | pbx800 | $ |
| 12:58.33 | [TK]D-Fender | pbx800: les.net. Never heard anything bad about them. |
| 12:58.42 | gr0mit | pbx800, which country are you based in? |
| 12:58.55 | pbx800 | US |
| 12:59.01 | gr0mit | ah ok |
| 12:59.47 | pbx800 | les.net has audio chops.. We use les.ne and voipjet as backups. But seriously, never wanted to use them as primary carriers. |
| 13:00.23 | [TK]D-Fender | pbx800: You've knocked out 3 of them better ones there. One might think you are the one with issues, and not them. |
| 13:02.47 | pbx800 | Fender - lol. nope its just that our usage is little high and I have used teliax, voipjet, voicepulse, callwithus, les.net, vitelity exhausitively. Voicepulse is by far the best but their rates are little high. I wanted to check if someone had a better recommendation. Thanks. |
| 13:03.52 | [TK]D-Fender | pbx800: 2k min/mo isn't much really... |
| 13:04.00 | pbx800 | thats not 2k mins, thats $2000 every month termination |
| 13:04.42 | [TK]D-Fender | pbx800: You should show minutes, not dollars. Gives us no idea how much you're over-paying. |
| 13:05.08 | [TK]D-Fender | ~itsplist-us |
| 13:05.09 | jbot | [~itsplist-us] Here are some popular ITSPs (USA) starting with the more respected ones : http://www.teliax.com , http://connect.voicepulse.com , http://www.nufone.net , http://www.broadvoice.com, http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net |
| 13:05.24 | [TK]D-Fender | pbx800: Three, go peruse. |
| 13:05.36 | x86 | [TK]D-Fender: here at work they force me to use AT&T for everything, and our long distance bills alone are about $15-20k per month |
| 13:05.58 | x86 | [TK]D-Fender: that's before voice T1 circuit charges, or any data T1 or Internet T1 charges too |
| 13:06.13 | pbx800 | thats heavy usage. |
| 13:06.16 | [TK]D-Fender | x86: You poor sod.... |
| 13:06.25 | pbx800 | isn't t1 subsidized now? |
| 13:06.32 | x86 | we pay $0.0293 per minute... rediculous |
| 13:06.33 | mocker | I use vitelity, haven't had any real problems w/ them |
| 13:07.03 | pbx800 | 3c outgoing is pretty high but I can understand the toll quality. |
| 13:07.32 | x86 | there are places offering $0.008 per minute over a T1 |
| 13:07.36 | x86 | 9les.net is one) |
| 13:07.46 | x86 | (les.net is one) |
| 13:08.00 | [TK]D-Fender | x86: And you're bosses don't because....? |
| 13:08.24 | x86 | [TK]D-Fender: dude was that english? ;) |
| 13:08.25 | pbx800 | their bosses do, I guess :) |
| 13:08.41 | x86 | and you are bosses don't..... wtf |
| 13:08.56 | [TK]D-Fender | x86: "Don't switch because?" <- |
| 13:09.03 | x86 | ah ;) |
| 13:09.18 | x86 | they know the name, and they prefer to pay a premium to stay with AT&T |
| 13:09.31 | x86 | because, they're the best, of course |
| 13:09.41 | [TK]D-Fender | x86: tahts 15$K / MONTH wasted. They aren't too bright it seems. |
| 13:10.31 | x86 | [TK]D-Fender: 0how do you figure that's $15K per month wasted? any other provider would charge me something... only thing wasted is the difference |
| 13:11.02 | [TK]D-Fender | x86: Meant to say $10K. |
| 13:11.23 | x86 | ah |
| 13:11.36 | x86 | yeah prolly somewhere closer to $8k I'd rekon |
| 13:11.58 | x86 | $96,000 per year flushed down the drain |
| 13:12.13 | pbx800 | I am have a ride on that $ |
| 13:12.14 | UnixDog | morning kids |
| 13:12.32 | pbx800 | unixdog - morning mom |
| 13:12.39 | UnixDog | lol |
| 13:12.45 | UnixDog | funny . |
| 13:13.08 | *** join/#asterisk KenLee (n=k3leland@bg-fw2out.monmouth.com) |
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| 13:17.53 | UnixDog | this week sucks. |
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| 13:25.09 | UnixDog | russellb blog site is major slow to respond |
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| 13:36.50 | Sargun_screen | what would be tthe best way to handle a fsm that can have exrernally triggered uhm, forks? |
| 13:37.04 | Sargun_screen | like, can an fsm temporarily fork |
| 13:37.07 | jmacz | Hi everyone, anybody knows how to obtain the date of a call from a pri debug file? |
| 13:37.25 | Sargun_screen | sort of like schrondinger's cat¥' |
| 13:39.46 | Sargun_screen | look a the mtime of the file? |
| 13:40.49 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 13:40.58 | *** join/#asterisk Sniper_VOIP (n=michofr@mail.splendor.net) |
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| 13:42.53 | jmacz | In the debug file, one cand find q.931 messages and several information, but it doesn't shows the date. I wonder if the numbers in brackets below the "Message type" and "Call State" directives carry the date information of a call |
| 13:44.13 | *** join/#asterisk fskrotzki (n=fskrotzk@host198.textwise.com) |
| 13:46.41 | jaytee | Sargun_screen, I saw a tshirt on Thinkgeek the other day that says, "Schroedinger's cat is dead!" |
| 13:48.49 | Sniper_VOIP | Hi all...I would like to know if the following can be done with Asterisk and how |
| 13:48.50 | Sniper_VOIP | I need to have a call test server where a customer dial a test number that we give out, he will hear a test message, record a short message for him and then play it back to hear the audio quality |
| 13:48.57 | errr | In the security.txt file it mentions securing the manager with SSH.. how is that done? |
| 13:49.28 | [TK]D-Fender | Sniper_VOIP : easily, and go read ... THE BOOK. |
| 13:49.30 | [TK]D-Fender | ~book |
| 13:49.31 | 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 |
| 13:50.23 | [TK]D-Fender | Sniper_VOIP: There is no "how-to" for Asterisk. Its like finding a guide for Play-Doh. You can do whatever you want with it so there is no way for a book to have every idea implemented specifically to show you |
| 13:50.46 | [TK]D-Fender | Sniper_VOIP: You'll need to lean about various forms fo connectivity, dialplan logic, etc. |
| 13:51.10 | Sniper_VOIP | [TK]D-Fender, this could be done with VOice Mail? |
| 13:51.35 | [TK]D-Fender | Sniper_VOIP: Not the right too for your needs. Use "record" instead |
| 13:52.00 | [TK]D-Fender | tool* |
| 13:52.31 | *** join/#asterisk __ys (i=yuri@91.151.196.254) |
| 13:53.33 | *** join/#asterisk RobH (n=RobH@rob.tech.wikimedia.org) |
| 13:54.19 | Sniper_VOIP | [TK]D-Fender, Is there any Playback option inside Asterisk? |
| 13:54.20 | *** join/#asterisk dominic1 (n=dob@213.221.82.242) |
| 13:54.29 | errr | heh |
| 13:54.29 | lmadsen | eh? |
| 13:54.37 | errr | dude read teh book |
| 13:54.39 | lmadsen | Sniper_VOIP: you mean... the Playback() application? |
| 13:54.48 | [TK]D-Fender | Sniper_VOIP: Yes... its called "Playback". Go read the dialplan application and function list. |
| 13:54.57 | dominic1 | I know I already asked many times..... but how can I add the opportunity to talk internal with g722 and external with g711 without transcoding? |
| 13:55.00 | lmadsen | 'core show applications' -- 'core show functions' |
| 13:55.02 | dominic1 | is anybody using that? |
| 13:55.22 | dominic1 | I know the channelvar but that's not working |
| 13:55.59 | *** join/#asterisk Serge8888 (n=Serge@76.66.214.130) |
| 13:56.02 | lmadsen | dominic1: make it so that the internal phones are configured to be allowed to use both g722 and g711, and the external peer just g711, and it should negotiate to g711 |
| 13:56.08 | lmadsen | goes off to do real work now |
| 13:57.15 | *** join/#asterisk Rem| (n=remi@bg-fw2out.monmouth.com) |
| 13:57.15 | *** join/#asterisk SuPrSluG (n=SuPrSluG@24.75.47.130) |
| 13:58.36 | coppice | stand firm. never negotiate! |
| 13:58.53 | Damin | Anyone in Minneapolis? |
| 13:58.53 | [TK]D-Fender | Terrists! |
| 13:59.00 | eric2 | yo have to negotiate, it's not an option in the real world |
| 13:59.04 | eric2 | *you |
| 13:59.24 | Serge8888 | Hi All, I have find out that when sip phone went down agent still registered in the queue. is any way to fix? Thnks |
| 13:59.39 | eric2 | negotiation will get you farther, faster!!! :) |
| 13:59.40 | pbx800 | Sniper_VOIP: use echotest application |
| 13:59.51 | *** join/#asterisk dawebber (n=dawebber@209.136.161.135) |
| 14:00.01 | *** part/#asterisk dawebber (n=dawebber@209.136.161.135) |
| 14:00.05 | dominic1 | the problem is my external peer is a isdn card |
| 14:00.11 | brodiem | Serge8888: why is that a problem? |
| 14:00.52 | Sniper_VOIP | pbx800, this application should help me to record a voice and pay it back to the user? |
| 14:01.14 | [TK]D-Fender | dominic1: Unless you can set it in the dialplan I don't think it'll work. |
| 14:01.41 | [TK]D-Fender | pbx800: No, not echo test. Fixed recording for quality. You can't jsut that with it echoing back to you instantly. |
| 14:02.00 | [TK]D-Fender | Sniper_VOIP: Record + Playback. That is all. |
| 14:02.10 | Serge8888 | brodiem: I use quemetrics for reporting some agents might shutdown softphone but according report they still in queue. |
| 14:02.48 | Sniper_VOIP | [TK]D-Fender, yes but each user will have its own recording file...How to specify that in the PlayBack? |
| 14:02.50 | ManxPower | Serge8888: one option is to use qualify=yes |
| 14:03.03 | [TK]D-Fender | loads eric2 onto his trebuchet and sends him farther, faster. |
| 14:03.25 | [TK]D-Fender | Sniper_VOIP: Use the uniqueid of the call as the filename. |
| 14:03.46 | [TK]D-Fender | Sniper_VOIP: Of keep a counter and use that. |
| 14:04.00 | brodiem | ManxPower: that wouldn't remove a channel from queue though would it? It would just report the sip channel as unavailable |
| 14:05.17 | Serge8888 | ManxPower: I have tried it does not affect on queue, sip peer in UNKNOWN status but still in queue for 3h |
| 14:05.37 | brodiem | Serge8888: are you using chan_agent or chan_local? |
| 14:07.33 | Serge8888 | brodiem: I am not shure but in queue my agent looks like Local/1401@from-internal |
| 14:08.37 | *** join/#asterisk railsmunky (n=nick@82-70-165-142.dsl.in-addr.zen.co.uk) |
| 14:08.53 | brodiem | Serge8888: with chan_agent it would work with autologoffunavail=yes, but chan_agent is less preferable than chan_local |
| 14:09.37 | [TK]D-Fender | Or* |
| 14:10.41 | brodiem | Serge8888: otherwise you could do a simple shell script in cron to periodically check for unknown channel states and remove the members from queue |
| 14:19.01 | *** join/#asterisk aksyn (n=aksyn@78.86.127.226) |
| 14:20.31 | Serge8888 | brodiem: Shell script will work, thank you |
| 14:21.04 | ManxPower | If anyone knows a source for 180 degree coax barrels, please /msg me. |
| 14:21.21 | ManxPower | or even 90 degree ones, or 90 decree coax connectors (F-type) |
| 14:22.25 | *** join/#asterisk xacatecas (n=jkroon@dsl-242-222-63.telkomadsl.co.za) |
| 14:22.47 | xacatecas | hi guys, i need suggestions. I'm urgently looking for alternatives to trixbox. |
| 14:22.53 | xacatecas | what are my options? |
| 14:23.19 | ManxPower | xacatecas: there is only one alternative here on this channel. |
| 14:23.30 | ManxPower | Can you guess what it is? |
| 14:24.03 | synthetiq | alternative is not to use a qui |
| 14:24.08 | synthetiq | gui |
| 14:24.11 | coppice | I didn't know americans used F-type connectors |
| 14:24.12 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 14:24.15 | ManxPower | We only install from source on this channel. |
| 14:24.29 | ManxPower | coppice: *nod* Standard coax connectors |
| 14:24.32 | synthetiq | apt-get install asterisk |
| 14:24.33 | synthetiq | =] |
| 14:24.42 | Serge8888 | is any way to convert AEL .conf to extentions .conf? |
| 14:24.48 | xacatecas | ManxPower, source-compiled asterisk :) |
| 14:24.50 | ManxPower | synthetiq: then you would need to talk to the person that built the package. |
| 14:24.56 | ManxPower | Serge8888: Yes! |
| 14:25.42 | ManxPower | Serge8888: Asterisk does not process AEL, Asterisk converts AEL into regular extensions.conf on load. I think there is a utility in Asterisk to just run the conversion outside of Asterisk |
| 14:25.47 | xacatecas | ok, so back in the real world where I need to roll out twenty or thirty of the things and allow clients to configure their own extensions? |
| 14:25.52 | *** join/#asterisk theHub (n=theHub@69.177.93.21) |
| 14:26.09 | ManxPower | xacatecas: If you don't want to install from source then go to a channel where they use GUIs |
| 14:26.28 | ManxPower | We. Don't. Use. GUIs. Here. |
| 14:27.19 | xacatecas | ok. |
| 14:27.26 | brodiem | xacatecas: maybe switchvox will do what you want, try #switchvox |
| 14:28.21 | [TK]D-Fender | xacatecas: Go learn *-GUI maybe, or prepare for find another trixbox like solution that will shove you in a tight corner. |
| 14:28.39 | [TK]D-Fender | xacatecas: What part about it aren't you satisfied with? |
| 14:28.57 | ManxPower | Asterisk GUIs are like living in a straight jacket, it seems cool at first, but you realize just how limited you are. |
| 14:29.25 | brodiem | hmm I wouldn't think it would be that cool to wear a straight jacket at first :) |
| 14:30.01 | ManxPower | Actually using Asterisk GUIs is like living in a straight jacket with "Down with Digium! printed on it. It seems cool at first, but you soon realize how limited you are and how much everyone hates you. |
| 14:30.04 | ManxPower | ~zeeeek |
| 14:30.08 | *** join/#asterisk SteveTotaro (n=Administ@pool-70-17-234-155.balt.east.verizon.net) |
| 14:30.11 | ManxPower | ~zeeek |
| 14:30.11 | jbot | from memory, zeeek is someone who once said "learning asterisk using a GUI is like learning sex through masturbation. You'll never get to the good stuff." |
| 14:30.30 | d-k-t-2 | ManxPower, using ABE seems to be the same :( |
| 14:30.45 | ManxPower | d-k-t-2: ABE is something we CAN'T support here. |
| 14:31.10 | ManxPower | But at least with ABE you get support from Digium. With GUIs people mostly just point and laugh. |
| 14:31.13 | *** join/#asterisk ming_zym (n=ming_zym@123.103.29.247) |
| 14:32.01 | *** join/#asterisk codehaxor (n=hellowor@222.127.250.100) |
| 14:32.30 | codehaxor | hello guys |
| 14:34.20 | *** join/#asterisk af_ (n=getsmart@88-149-230-57.dynamic.ngi.it) |
| 14:34.26 | ManxPower | Whoo! Whoo! Mozulla 3 being released at 10amPDT |
| 14:34.36 | ManxPower | Mozulla? |
| 14:34.44 | brodiem | ManxPower: today is Firefox 3 Download Day |
| 14:35.07 | ManxPower | brodiem: *nod* I'll be downloading it this evening |
| 14:35.11 | brodiem | http://www.spreadfirefox.com/en-US/worldrecord |
| 14:35.21 | ManxPower | I just wish they would hurry up and release the next major release of Thunderbird. |
| 14:35.28 | coppice | will there be 3.0.1 for tea? |
| 14:35.50 | ManxPower | APPARENTLY, the developers don't use it with multiple remote IMAPS server. |
| 14:36.20 | [TK]D-Fender | ManxPower: Tried Evolution yet? |
| 14:36.34 | coppice | the thunderbird in fedora 9 has some huge delays. dunno what its doing |
| 14:36.41 | brodiem | ugh, evo |
| 14:36.42 | ManxPower | [TK]D-Fender: I used to use that. I'm mostly in Windows these days and try to use cross-platform stuff. |
| 14:37.08 | [TK]D-Fender | ManxPower: Theres a windows build of it. |
| 14:37.12 | ManxPower | GAIM/Pidgin, Firefox, Thunderbird, OpenOffice, jEdit, etc. |
| 14:37.20 | coppice | last time I tried evolution it treated chinese like it was some kind of foreign language |
| 14:37.32 | ManxPower | [TK]D-Fender: Evolution was worse than Firefox the last time I used it. |
| 14:38.09 | *** join/#asterisk CunningPike (n=arodgers@vpn.dnv.org) |
| 14:38.27 | tzanger | I just use Kontact. I find the mozilla stuff pretty heavy |
| 14:38.42 | ManxPower | It is pretty heavy. |
| 14:38.48 | *** join/#asterisk spokra (n=spokra@host093-179-178.sea0.speakeasy.net) |
| 14:39.44 | brodiem | <== starting to make the switch to Zimbra |
| 14:40.23 | brodiem | also heavy but since it runs a local sql server if you have a ton of mail like I do it's extremely fast for searching |
| 14:40.44 | lmadsen | brodiem: welcome to the club :) |
| 14:41.37 | brodiem | lmadsen: heh just finished installing the ZCS open-source edition, just wish it had some better import options =/ |
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| 15:11.26 | dominic1 | I have two isdn pri's. Am I able to spoof the callerid from one to the other? That I am always dialing out with one number? |
| 15:13.11 | lmadsen | dominic1: depends if your service provider allows you to do that -- usually with PRI they allow you to control the CID |
| 15:13.30 | lmadsen | might want to check with your service provider though because sometimes they don't like that |
| 15:14.25 | waverly360 | Anyone here use asterisk in an environment where a lot of call recordings are archived for historical/legal reasons? |
| 15:15.30 | lmadsen | I do |
| 15:16.01 | Kobaz | would anyone know why asterisk would completely become unresponsive after doing a sip reload... i tried to do another sip reload and it complained that the previous reload never finished... and since the reload ast was completely dead and stopped handling calls |
| 15:16.07 | waverly360 | How many recordings do you average a month? |
| 15:16.13 | waverly360 | or better question |
| 15:16.17 | *** join/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
| 15:16.27 | waverly360 | how much space do you require a month for accumulated recordings? |
| 15:17.10 | Kobaz | take your codec bit rate, multiply it out by the amount of seconds of call time you plan to record per month, and you'll get your number |
| 15:17.32 | brodiem | waverly360: 20GB over 27,000 recordings here.. |
| 15:18.33 | waverly360 | Well, what I'm trying to get at here..I have an asterisk box that's accumulated about 95 gigs worth in 3 months |
| 15:18.34 | brodiem | wav w/gsm encoding |
| 15:19.21 | waverly360 | The asterisk box can't keep them..so I'm trying to think of ways to back them up automatically on a windows server |
| 15:19.33 | *** join/#asterisk cesar_CR (n=cesar@200.91.75.45) |
| 15:19.47 | brodiem | just use ftp or rsync |
| 15:19.48 | Kobaz | rsync works nicely |
| 15:19.51 | waverly360 | I know there are several options...but wanted to see how others were doing it |
| 15:20.23 | brodiem | rsync for reliability |
| 15:20.41 | waverly360 | well..rsync works well to synchronize, but I'm looking to remove them from the pbx as this is happening..does it have that capability? |
| 15:20.48 | brodiem | yes |
| 15:21.19 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 15:21.19 | *** mode/#asterisk [+o russellb] by ChanServ |
| 15:22.54 | waverly360 | hmm |
| 15:22.57 | waverly360 | that might work |
| 15:23.16 | brodiem | indeed |
| 15:24.07 | waverly360 | so no one here would recommend any of the pay-for backup apps then? |
| 15:24.17 | waverly360 | It's my experience that most all of them suck |
| 15:24.23 | waverly360 | but maybe I'm just biased :) |
| 15:24.34 | ManxPower | waverly360: It really depends on your needs. Nothing Asterisk specific about it. |
| 15:24.49 | waverly360 | Oh I know..I just wasn't sure what forum I could ask that in :) |
| 15:24.57 | ManxPower | waverly360: I hate ALL backup systems, they work great...until you need to restore something |
| 15:25.22 | coppice | has wct4xxp become more flaky recently, or do I have a card which has become flaky? :-\ |
| 15:25.34 | brodiem | waverly360: rsync will do everything you need it to do, if you want more functionality like e-mail reporting check out RIBS |
| 15:25.38 | waverly360 | ManxPower: that's been my experience mostly. |
| 15:25.39 | ManxPower | coppice: I think all of 1.4 has become flaky |
| 15:25.42 | brodiem | w/o writing your own script that is |
| 15:26.51 | waverly360 | I think the part that's tricky for me, is that I've provided this company the pbx..but the server is theirs. |
| 15:27.00 | Kobaz | ManxPower: very flaky |
| 15:27.35 | ManxPower | At least they seem to be making progress on the longstanding deadlock issues. |
| 15:27.45 | Kobaz | manxpower: i made a call out to an audiocodes fxo gateway and then asterisk proceeded to become completely unresponsive |
| 15:27.47 | *** join/#asterisk artz22 (n=artz@190.189.0.33) |
| 15:27.49 | *** join/#asterisk UnixDog (n=UnixDog@181.128.204.68.cfl.res.rr.com) |
| 15:27.50 | artz22 | hi all |
| 15:28.06 | ManxPower | Kobaz: what version of 1.4 are you using? |
| 15:28.17 | Kobaz | 1.4.14 on this particular box |
| 15:28.20 | waverly360 | ManxPower: If not 1.4...what version do you prefer? |
| 15:28.26 | ManxPower | waverly360: I use 1.2.x |
| 15:28.33 | waverly360 | ManxPower: same here |
| 15:28.35 | ManxPower | I think 1.2.24.1 |
| 15:28.51 | Kobaz | maybe we should move to 1.2 |
| 15:28.56 | ManxPower | waverly360: I had high hopes for 1.6 until I started to understand their new release cycle. |
| 15:28.59 | waverly360 | ManxPower: but we're only at 1.2.14...been meaning to upgrade, but I'm so afraid it'll break something we've custom written |
| 15:29.02 | UnixDog | 1.2 is eol |
| 15:29.09 | UnixDog | time to move to 1.4 |
| 15:29.13 | Kobaz | we've been crashing asterisk with our call generator |
| 15:29.17 | ManxPower | Kobaz: You could lose functionality if you downgrade. |
| 15:29.17 | UnixDog | come over to the dark side |
| 15:29.23 | Kobaz | ManxPower: yeah |
| 15:29.28 | ManxPower | Kobaz: try upgrading to the latest 1.4 first. |
| 15:29.44 | coppice | EOL is the magic term that makes stuff work. most things finally become usable a little after EOL is declared |
| 15:29.57 | *** join/#asterisk putnopvut (n=putnopvu@216.207.245.1) |
| 15:29.57 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 15:30.03 | cesar_CR | hi guys, if I use a server rack stile, is it easy to install the digium cards there ??? |
| 15:30.06 | Kobaz | ManxPower: yeah, we crash 1.4.20 too |
| 15:30.11 | ManxPower | UnixDog: Not according to the reports I see here which tend to be "I upgraded to 1.4.whatever (usually latest) and Asterisk started crashing." |
| 15:30.31 | artz22 | i'm implementing a channel driver for a custom usb-phone (FXO). what should I call when someone calls to the FXO? I don't know how to tell asterisk that there is an incoming call |
| 15:30.32 | waverly360 | The cool thing about open source EOL software is just that..it's open source..it's only EOL as long as developers aren't modding code any longer. |
| 15:30.39 | UnixDog | I dont use latest and greatest I use 1 to 2 versions back |
| 15:30.50 | ManxPower | I usually tell them to work to debug the issue if it's not a production system, if it is a production system then get a backtrace of the crash and then revert to the working release. |
| 15:30.57 | *** join/#asterisk hfb (n=hfb@pool-71-118-254-245.lsanca.dsl-w.verizon.net) |
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| 15:31.07 | waverly360 | ManxPower: Do you experience asterisk crashes much at all? |
| 15:31.16 | ManxPower | waverly360: once or twice a year at most. |
| 15:31.22 | Kobaz | ManxPower: yeah, i need to enable the debug flags |
| 15:31.37 | anonymouz666 | ManxPower: how many calls/day? |
| 15:31.38 | waverly360 | ManxPower: That sounds about average for us as well. |
| 15:31.44 | ManxPower | Kobaz: nobody will listen to a bug report unless you use the latest. |
| 15:31.48 | UnixDog | I have yet to have a 1.4 crash on me |
| 15:32.01 | UnixDog | a few small issues. but never crash |
| 15:32.18 | ManxPower | anonymouz666: I have no idea. I'd have to look, but I would guess several hundred per day |
| 15:32.38 | *** join/#asterisk outtolunc (n=me@66.218.53.172) |
| 15:32.38 | anonymouz666 | small setup then |
| 15:32.48 | ManxPower | UnixDog: The 1.4 problems seem, based on what I've seen here, to be usage specific race conditions -- a real bitch to diagnose and fix. |
| 15:32.50 | brodiem | 1.4 crashed for me by trying to add a PSTN number to a queue, and when I specified an invalid file name with CLI "file convert", but never during normal daily operation |
| 15:32.58 | waverly360 | We have several asterisk boxes installed. 1.2 is pretty stable in that most boxes can go 200+ days without crashing, but it's still frustrating when it does. |
| 15:33.12 | ManxPower | anonymouz666: as I said, a guess. |
| 15:33.27 | Kobaz | what i want, is less restarts for config changes |
| 15:33.42 | Kobaz | *CLI> module reload chan_zap.so |
| 15:33.47 | Kobaz | [Jun 17 11:30:58] WARNING[31878]: chan_zap.c:11129 process_zap: Ignoring signalling |
| 15:33.50 | Kobaz | !!! |
| 15:33.55 | ManxPower | Kobaz: Huh? You almost never need to restart for config changes, EXCEPT for some zap options. |
| 15:34.01 | Kobaz | exactly |
| 15:34.03 | ManxPower | Kobaz: did you change the signalling? |
| 15:34.06 | Kobaz | yes |
| 15:34.07 | Kobaz | heh |
| 15:34.21 | ManxPower | *nod* That is one of the things you need to restart chan_zap for. |
| 15:34.30 | ManxPower | addind/removing channels, signalling. |
| 15:34.48 | ManxPower | Kobaz: and really, if you change the signaling you'll have to re-run ztcfg and ztcfg terminates all calls. |
| 15:34.51 | Kobaz | i'm spoiled by this avaya that's lounging around in the corner |
| 15:35.25 | Kobaz | ManxPower: and plus we have a sangoma card, so it requires a reflash of the firmware to change the clock source |
| 15:35.36 | Kobaz | well we've switched to sangoma for everything |
| 15:35.37 | ManxPower | Kobaz: I never knew that. |
| 15:35.54 | *** join/#asterisk LiNeTuX (n=LiNeTuX@proxy.fmpa.com) |
| 15:35.57 | waverly360 | Kobaz: I've never had to reflash the firmware on any of our sangoma cards |
| 15:36.02 | waverly360 | Kobaz: what kind are you using? |
| 15:36.07 | Kobaz | change the clock source and you need to do: asterisk stop; /etc/init.d/wanpipe restart; asterisk start |
| 15:36.14 | ManxPower | Apparently your PBX is much more dynamic than mine. We NEVER change zap stuff during production hours. |
| 15:36.14 | Kobaz | a10x |
| 15:36.34 | [TK]D-Fender | Kobaz: that is NOT "reflashing the firmware" |
| 15:36.35 | Kobaz | ManxPower: well we're in the setup phase of a production box |
| 15:36.40 | ManxPower | My users would castrate me. |
| 15:36.49 | Kobaz | [TK]D-Fender: well, you know... reuploading the firmware |
| 15:36.55 | Kobaz | not flashing the boot rom or anythning |
| 15:37.06 | [TK]D-Fender | Kobaz: You just did it again... |
| 15:37.13 | waverly360 | XD |
| 15:37.15 | Kobaz | ManxPower: yeah i dont change pri stuff during production hours either, heh |
| 15:37.18 | coppice | all these computer based E1 and T1 cards are pretty pathetic at handling clock sources, alarms and monitoring error rates |
| 15:37.21 | [TK]D-Fender | Kobaz: Say "A", and then redescribe it as "B" |
| 15:37.25 | Kobaz | [TK]D-Fender: did what again? |
| 15:37.30 | Kobaz | oh |
| 15:37.34 | Kobaz | semantics |
| 15:37.42 | *** join/#asterisk ecrist (n=ecrist@snipe.secure-computing.net) |
| 15:37.46 | [TK]D-Fender | Kobaz: No, LANGUAGE <- |
| 15:37.47 | waverly360 | oh snap |
| 15:38.02 | ecrist | which ports do I need open so an SIP phone can pull a config from a server via FTP? |
| 15:38.06 | ecrist | I'm not seeing anything blocked on my firewall, but the phone complains that it couldn't contact the boot server for it's config. |
| 15:38.13 | Kobaz | semantics! :P |
| 15:38.13 | anonymouz666 | coppice: alarms and monitoring error rates I would agree with you. |
| 15:38.15 | waverly360 | the ftp port |
| 15:38.15 | [TK]D-Fender | ecrist: TCP 21 |
| 15:38.20 | waverly360 | 21 |
| 15:38.33 | brodiem | I would rather have to deal with zap reloads then listening to users complain about jitter when using SIP-only over public internet along with their data.. |
| 15:38.49 | waverly360 | ecrist: but you need to make sure you have an ftp server running on the pbx |
| 15:38.55 | *** join/#asterisk martha (n=martha@bzq-179-135-226.static.bezeqint.net) |
| 15:39.03 | waverly360 | ecrist: or rather wherever you're pulling the config from... |
| 15:39.28 | ecrist | k, thanks |
| 15:39.28 | *** join/#asterisk korihor (n=korihor@190.199.171.145) |
| 15:39.29 | martha | so, anyone having problem with IAX2 in 1.4.21? |
| 15:39.39 | martha | I just upgraded and no iax calls seem to go through now |
| 15:39.55 | Kobaz | ManxPower: it would be nice to be able to change signalling and etc without bringing down the pbx if you for instance had a multi port t1 card with more than one free, and you want to bring up another line during production hours |
| 15:40.16 | Kobaz | ManxPower: but yeah, i like that most everything can be reloaded in 1.4 |
| 15:40.18 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.137) |
| 15:42.57 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.137) |
| 15:43.34 | ManxPower | points at martha and says "Look, another report!" |
| 15:43.46 | Kobaz | hehe |
| 15:44.07 | *** part/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 15:44.42 | martha | ManxPower: so, a lot of people? I guess I should revert to 1.4.20 |
| 15:45.25 | Kobaz | revert to 1.4.10 just to be safe |
| 15:45.44 | Kobaz | heh |
| 15:46.10 | Kobaz | it's weird, the problems i have always seem to pop up no matter what version i'm using |
| 15:47.43 | martha | actually, I was at 1.4.9 before this |
| 15:48.03 | Kobaz | i've been on 1.4.14 for a while |
| 15:48.09 | *** join/#asterisk km2 (n=x@c-24-23-255-173.hsd1.ca.comcast.net) |
| 15:50.49 | martha | 1.4.9 crashed ~1x/day for me, which is why I wanted to upgrade |
| 15:51.17 | *** join/#asterisk ccesario (n=ccesario@189.20.219.10) |
| 15:52.04 | cesar_CR | hi guys, if I use a server rack stile, is it easy to install the digium cards there ??? |
| 15:52.38 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.137) |
| 15:52.53 | *** join/#asterisk sack (n=sack@249.Red-81-32-160.dynamicIP.rima-tde.net) |
| 15:52.58 | [TK]D-Fender | cesar_CR: Its a PCI card. Just like any other. Look at the size, use your brain. |
| 15:53.09 | anonymouz666 | [TK]D-Fender: hey, take easy. |
| 15:53.51 | cesar_CR | [TK]D-Fender: sorry man.... |
| 15:54.10 | Kobaz | martha: i haven't had 1.4.14 crash under normal working conditions, only when stressed with many calls in acd's |
| 15:54.17 | [TK]D-Fender | anonymouz666: that was plenty "easy". |
| 15:55.06 | [TK]D-Fender | anonymouz666: Next people will ask how much volume a cubic-foot box takes up. |
| 15:55.21 | [TK]D-Fender | anonymouz666: say.... what colour was Napoleon's white horse? |
| 15:55.23 | brodiem | fender is like the Judge Judy of asterisk lol |
| 15:56.27 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.137) |
| 15:56.32 | LiNeTuX | thinks Rhino support is a few cans short of a 6-pack |
| 15:57.42 | Kobaz | heh |
| 15:57.47 | Kobaz | LiNeTuX: yeah |
| 15:57.59 | Kobaz | LiNeTuX: i had to help them debug their own hardware |
| 15:58.30 | LiNeTuX | Kobaz: It's about there now. I'm not sure if I should tell him why he's broken my box, or let him figure it out on his own. |
| 15:59.15 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.137) |
| 16:00.17 | Kobaz | LiNeTuX: yeah, i was watching one of the support guys terminals and the first thing he did was clobber all of my configs |
| 16:01.05 | LiNeTuX | Kobaz: this guy doesn't even know how to read their own support docs on blacklisting modules |
| 16:01.43 | Kobaz | heh |
| 16:01.47 | [TK]D-Fender | 1Hr to Firefox 3.0 :) |
| 16:02.02 | Kobaz | yay |
| 16:02.04 | LiNeTuX | May the flood begin |
| 16:02.41 | *** join/#asterisk viperdude (n=jon@sokar.enta.net) |
| 16:03.40 | viperdude | should command "logger rotate" rotate cdr-customer/Master.csv ? |
| 16:03.53 | viperdude | cdr-custom even :-) |
| 16:03.53 | Qwell | no |
| 16:04.03 | viperdude | Qwell: how do i do it then>? |
| 16:04.22 | Qwell | use the logrotate daemon if you need to do that |
| 16:04.59 | viperdude | aha, I seem to be comsuming swap space because of the size of Master.csv so this hopefully will do the trick |
| 16:06.33 | Qwell | it's opening in append mode. I don't think that uses much ram |
| 16:09.20 | viperdude | hmm well at the moment I delete the Master.csv once imported into MySQL, as soon as I do that the RAM usage starts to grow. 512Mb so far today |
| 16:09.31 | viperdude | restarting * clears the RAM |
| 16:16.05 | artz22 | I'm trying to make an auto attendant with asterisk so when someone calls to a FXO asterisk answers and plays a welcome message, how should I modify extensions.conf? |
| 16:16.48 | viperdude | artz22: in the context the call comes into s,n,Playback(myanswerfile) |
| 16:17.25 | artz22 | I see.. so if I'm making a channel driver myself I have to call 's', right? |
| 16:17.38 | artz22 | or is it just by default? |
| 16:17.59 | BCS-Satori | I programed a user at extension 307, but have yet to setup a phone with for that user. When I reloaded asterisk and did "sip show peers" it said that 307 was just registered to an ip address of a phone. But I know for a fact that that phone that asterisk claims the peer is registered to does not contain any registration information for 307? Any idea whats going on? The user is host=dynamic too |
| 16:19.38 | *** part/#asterisk dominic1 (n=dob@213.221.82.242) |
| 16:19.52 | *** join/#asterisk aksyn (n=aksyn@78.86.127.226) |
| 16:20.39 | codefreeze-lap | Serge8888: look for aelparse in 1.4, 1.6.0, and trunk |
| 16:22.24 | viperdude | artz22: s is the default extension in a context |
| 16:22.45 | [TK]D-Fender | BCS-Satori: if it says something registered, then it probably has. * doesn't make that stuff up. |
| 16:22.54 | [TK]D-Fender | BCS-Satori: pastebin backup for your situation. |
| 16:22.58 | artz22 | viperdude: I think I got it, thanks |
| 16:23.22 | [TK]D-Fender | artz22: No, you don't need to use "s". |
| 16:23.39 | [TK]D-Fender | viperdude: and no, not "default" in the strictest sense. |
| 16:24.49 | *** join/#asterisk servettas (n=usta@88.249.71.190) |
| 16:24.52 | *** join/#asterisk msetim (n=msetim@200.195.161.164) |
| 16:26.08 | [TK]D-Fender | artz22: What kind of channel driver are you thinking of creating? |
| 16:26.12 | artz22 | I got this: -- Executing [s@default:1] Wait("SatTest/aaa", "1") in new stack |
| 16:26.14 | artz22 | a FXO one |
| 16:26.26 | artz22 | SatTest/aaa is the device now |
| 16:26.35 | artz22 | it's a prototype for a usb phone |
| 16:27.01 | *** join/#asterisk jjshoe (n=jjshoe@72.37.252.50) |
| 16:27.25 | flush | yo |
| 16:29.00 | flush | theres a service here in montreal its number 811 for info sante and shits.. i set up extensions correctly for 811 but when i dial it, it starts monitor, then it stops music on hold but wont ring, it just times out with fast tone... but when i press "9811" cause 9 is to have the outbound, it works and will ring |
| 16:29.10 | flush | any idea why it doesnt when i dont do "9811" |
| 16:29.39 | *** join/#asterisk dominic1 (n=dob@213.221.82.242) |
| 16:30.40 | penguinFunk | can asterisk use more than one isdn30 channel for video calling? |
| 16:31.10 | [TK]D-Fender | flush: PASTEBIN is your friend. We can't see your call or code. We're not psychic you know... |
| 16:31.21 | flush | im doing so right now |
| 16:31.39 | penguinFunk | its works brilliantly office > office through our vpn, just wondering if we can use some of our isdn30 channels instead of eating up our leased line bandwidth ? |
| 16:31.54 | servettas | i need voip trunk and i do not have any idea can anyone help me about it ? |
| 16:31.59 | flush | http://pastebin.com/m742d16b |
| 16:32.07 | BCS-Satori | [TK]D-Fender: http://pastebin.com/m7a8a5a03 I have also checked the web interface of the Linskys SPA962 and Ext1: is 305, Ext2: Non, Ext3: 305B3, Ext4: 305B4, Ext5: 305B5, Ext6: 305B6. No where in the phone does it have 307 defined. |
| 16:32.36 | penguinFunk | im thinking we need 256kbps which would mean (4 * 64kbps isdn channels) |
| 16:32.45 | penguinFunk | any way I can bond these together to make one call? |
| 16:33.07 | penguinFunk | can this be done with asterisk ? |
| 16:33.13 | flush | zomg |
| 16:33.17 | [TK]D-Fender | BCS-Satori: SOMETHING at that IP reg'd against it at one point, and * didn't hit the renewal point. It only "refound" a device to respond to the Options packets. |
| 16:33.27 | [TK]D-Fender | BCS-Satori: Or you're just wrong and missed it. |
| 16:33.54 | [TK]D-Fender | flush: pastebin your failed call at verbose 10. |
| 16:35.18 | [TK]D-Fender | servettas: Please try to ask a more specific question. |
| 16:37.22 | viperdude | [TK]D-Fender: sorry about my "default" comment regarding s extension, its more like the extension of last resort |
| 16:37.41 | [TK]D-Fender | viperdude: Only as implemented by certain channel drivers, etc. |
| 16:37.53 | viperdude | sure |
| 16:37.56 | [TK]D-Fender | viperdude: PRI doesn't care about it, generally neither does SIP, etc. |
| 16:38.03 | BCS-Satori | [TK]D-Fender: i also see in my database old SIP/NAME that no longer exist or old extensions, do i need to manually remove these as well? |
| 16:38.06 | [TK]D-Fender | viperdude: Loose concepts really. |
| 16:38.20 | [TK]D-Fender | BCS-Satori: .... DATABASE? |
| 16:38.32 | flush | [TK]D-Fender i think its because of the EXTEN:1 on the "811" extension.. how to i tell to compose all the extension but not remove first digit |
| 16:38.39 | flush | {EXTEN:} ? |
| 16:39.26 | *** join/#asterisk JerJer (n=PhatJ@pdpc/supporter/bronze/jerjer) |
| 16:39.27 | [TK]D-Fender | flush: ${EXTEN} is the WHOLE # you dialed |
| 16:39.36 | flush | copy thanks |
| 16:39.53 | [TK]D-Fender | flush: exten => 811,3,Dial(${OUTBOUNDTRUNK}/${EXTEN:1},,mwW) <-- yup |
| 16:40.02 | flush | nice |
| 16:40.06 | *** join/#asterisk ^sandro^ (n=lksjdsf@dsl-67-204-9-136.acanac.net) |
| 16:40.10 | ^sandro^ | hello everyone.. good day |
| 16:40.20 | flush | [TK]D-Fender you see its dumb cause it was the same settings for my 911 |
| 16:40.31 | flush | so i would have had to do "9911" |
| 16:40.33 | flush | bad |
| 16:40.37 | [TK]D-Fender | flush: No, I think the 911 one is dumber ;) |
| 16:40.44 | flush | rofl |
| 16:40.56 | ^sandro^ | i have a problem wtih caller ids.. can someone help me? |
| 16:40.57 | ^sandro^ | exten => _xxxxxxxxxx,1,Set(CALLERID(name)=Anonymous) |
| 16:40.57 | ^sandro^ | exten => _xxxxxxxxxx,2,Set(CALLERID(num)=) |
| 16:40.57 | ^sandro^ | exten => _xxxxxxxxxx,3,Dial(SIP/1${EXTEN}@host) |
| 16:40.58 | JerJer | I have just been tasked with deploying a 'scalable' voicemail situation. anyone have comments on the 'right' way to make such a beast happen? I like the concept of IMAP storage, but last time i tried it there was problems :( |
| 16:41.08 | BCS-Satori | [TK]D-Fender: in "database show" inside asterisk -vr |
| 16:41.15 | ^sandro^ | for some reason i keep getting incorrect formatted characters.. im not sending the right information to the isp |
| 16:41.29 | ^sandro^ | what is the correct way to forward the blocked caller id.. if received |
| 16:41.37 | [TK]D-Fender | BCS-Satori: Ah. Well if you want to knock a reg off the map, just nuke the key. |
| 16:41.47 | [TK]D-Fender | BCS-Satori: and do a "sip reload" |
| 16:41.59 | viperdude | JerJer: i wrote my own agi vmail app that stores in MySQL, over 3000 users without a issue |
| 16:42.17 | JerJer | agi or fast agi ? |
| 16:42.22 | viperdude | either |
| 16:42.27 | viperdude | i use agi |
| 16:42.36 | JerJer | agi in my world croaks badly |
| 16:42.52 | JerJer | i am talking like perhaps a thousand simultaneous calls at any given time :) |
| 16:42.59 | viperdude | really I have a whole ITSP built on AGI |
| 16:43.10 | JerJer | fastagi is better - but not optimal - imho |
| 16:43.13 | viperdude | try res_perl |
| 16:43.21 | JerJer | oh god no |
| 16:43.46 | viperdude | so all you need is enough * boxes to handle the traffic and timely db access |
| 16:43.57 | ^sandro^ | version 1.4.18 |
| 16:44.19 | anonymouz666 | JerJer: how many asterisk boxes you are talking about? |
| 16:44.45 | jaytee | I just bought the Lumenvox starter kit to evaluate. hope this doesn't turn out to be a collosal waste of time. |
| 16:45.00 | LiNeTuX | Do most folks still use the old "dial 9" for the outside lines when setting up *? |
| 16:45.44 | viperdude | LiNeTuX: no i auto detect if the number dialled is a local exten or a external call, plus if the caller is allowed to dial it |
| 16:46.19 | *** join/#asterisk vgster (n=vgster@93-96-221-240.zone4.bethere.co.uk) |
| 16:46.21 | JerJer | anonymouz666: at this point two dual quad cores - with openser acting as a load balancer |
| 16:46.37 | JerJer | but we may add a third dual quad core |
| 16:46.42 | [TK]D-Fender | ^sandro^: "core show application setcallerpres" |
| 16:46.46 | viperdude | JerJer: 2 boxes for 1000's of concurrent calls? |
| 16:47.35 | JerJer | viperdude: sure - its just i/o |
| 16:47.59 | JerJer | they are going to be virtualized - of course |
| 16:48.02 | viperdude | JerJer: not if processing vmail request |
| 16:48.10 | viperdude | more then i/o |
| 16:48.17 | *** join/#asterisk Vco (n=rbowler@S0106000db905c200.cg.shawcable.net) |
| 16:48.41 | JerJer | two dual core boxes will run maybe 4-8 instances of asterisk |
| 16:48.44 | JerJer | each |
| 16:48.54 | *** join/#asterisk jack_sparo (n=eddy@91.73.203.98) |
| 16:48.59 | viperdude | good luck i wish you well |
| 16:48.59 | JerJer | er |
| 16:49.05 | JerJer | two dual quad core |
| 16:50.15 | JerJer | viperdude: virtualized is cake - i do it all day long already |
| 16:50.23 | anonymouz666 | JerJer: ast 1.4? running stable? |
| 16:50.38 | viperdude | JerJer: cool |
| 16:50.59 | ^sandro^ | hey so if i set SetCallerPres(unavailable) it should work? |
| 16:51.10 | JerJer | only thing I don't have a solid understanding of is database storage over IMAP storage of voicemail |
| 16:51.51 | viperdude | JerJer: i would suggest with that level of usage you want something fast, MySQL? |
| 16:52.57 | [TK]D-Fender | ^sandro^: go TRY and see... |
| 16:53.07 | ^sandro^ | did.. no luck |
| 16:53.51 | [TK]D-Fender | ^sandro^: pastebin is your friend. |
| 16:55.14 | jaytee | I thought Google is your friend? |
| 16:55.48 | JerJer | anonymouz666: you mean running asterisk 1.4 stable in a virtualized environment? absolutely |
| 16:56.18 | JerJer | been using virtualized situations for well over a year now |
| 16:56.40 | anonymouz666 | XEN? |
| 16:56.41 | russellb | if anyone feels like causing trouble ... http://216.237.124.26/russellbryant.net/ |
| 16:56.42 | JerJer | in fact its the only way I know how to scale an asterisk box - due to per process thread limitations |
| 16:56.50 | ^sandro^ | hello... thank you for your help.. well the problem i am having is as follows.. i make a call with *67 off it shows me the caller id without a problem.. however when i make a call with *67 on it it shows a 412-222-2222 number |
| 16:57.06 | ^sandro^ | i want it to just say private caller.. which is fine.. but it shows me random numbers.. its not the caller number which is good but .. |
| 16:57.16 | ^sandro^ | i want to basically send to the devices that its private caller |
| 16:58.08 | JerJer | russellb: define what kind of trouble you lookin to cause :) |
| 16:58.32 | ^sandro^ | http://pastebin.com/m35087c3b |
| 16:59.17 | russellb | JerJer: hehe ... people ringing phones and blowing up their inbox until they fix it i guess :) |
| 16:59.39 | JerJer | ooh call files |
| 17:00.41 | JerJer | pbx_spool.so to the rescue |
| 17:00.45 | ^sandro^ | doesn't matter if i set SetCallerPres(prohib) or not |
| 17:00.51 | russellb | JerJer: lol |
| 17:02.24 | [TK]D-Fender | Oh boy... Mozilla is getting hit now :) |
| 17:02.36 | [TK]D-Fender | keeps pounding for 3.0... |
| 17:04.01 | *** join/#asterisk bminish (n=bminish@rb.brenbox.home.minish.org) |
| 17:04.48 | hi365 | hit is not the word. do they get their world record if they crash in the midlle? |
| 17:05.55 | JerJer | anonymouz666: I have used both xen and linux-vserver - depending on the situation |
| 17:06.32 | [TK]D-Fender | hi365: semantically speaking... could be possible :) |
| 17:06.38 | *** join/#asterisk Ebola (n=Ebola@unaffiliated/ebola) |
| 17:06.56 | hi365 | well, they seem to be overloaded, so no cookies for them |
| 17:07.20 | anonymouz666 | JerJer: it's an interesting setup you did. I also saw the kpfleming presentation about Asterisk Clustering |
| 17:10.23 | anonymouz666 | the great openser is always mentioned |
| 17:10.24 | ^sandro^ | hello |
| 17:11.00 | ^sandro^ | i am having some trouble with call blocking.. when i make the call it shows intead of private caller it shows some random numbers 412-222-2222 on the users display.. information can be found here : http://pastebin.com/mf6de61f |
| 17:12.26 | ^sandro^ | the privacy=full is set when blocked but still shows random numbers on the terminating caller iD |
| 17:16.42 | *** join/#asterisk gr0mit (n=tim@extrt.txrx.org.uk) |
| 17:19.57 | *** join/#asterisk FuriousGeorge (n=brian@ool-457f216e.dyn.optonline.net) |
| 17:20.12 | FuriousGeorge | hey all |
| 17:20.28 | *** join/#asterisk mintee (n=mintone@75.150.132.150) |
| 17:20.33 | mintee | g'day all |
| 17:20.48 | FuriousGeorge | my 'well-behaved' asterisk server is rebelling |
| 17:21.29 | FuriousGeorge | for some reason, every 24 hours or so I see this on the console: chan_zap.c:1625 zt_set_hook: zt hook failed: Device or resource busy |
| 17:21.35 | mintee | 'twas wondering if there was any very simple CDR web app for billing out there. I don't need anything crazy, just basically to be able to search via CID, or CALLEDID |
| 17:21.50 | FuriousGeorge | it alternates between my two fxs every few seconds |
| 17:21.52 | *** part/#asterisk aliver (n=aliver@ip-216-17-160-99.rev.frii.com) |
| 17:22.36 | orionr | ~book |
| 17:22.37 | 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 |
| 17:23.01 | FuriousGeorge | when this happens, i believe my fxo can't take incoming calls. i haven't really left it in that state because my users make me aware of it when they notice they cant make outbound calls |
| 17:23.04 | mintee | was book for me? |
| 17:23.39 | *** join/#asterisk grEvenX (n=even@pc107-130.ktv.no) |
| 17:23.51 | FuriousGeorge | sadly, ive seen this behavior with asterisk 1.2 in the past, and thought it was behind me. |
| 17:25.44 | FuriousGeorge | its kinda demoralizing, because of the two busy asterisk server i administrate, this one was the relatively 'low-maintenance' one |
| 17:32.12 | FuriousGeorge | http://pastebin.ca/1049269 <--thats what it looks like from the console |
| 17:34.23 | FuriousGeorge | vrap its happening angain, ETF |
| 17:34.32 | FuriousGeorge | WTF :) |
| 17:35.40 | FuriousGeorge | omg, this is terrible, i cant sit and watch this server all day |
| 17:35.45 | *** join/#asterisk trapa (n=2264581@207.230.238.94) |
| 17:36.21 | FuriousGeorge | im gonna try to upgrade it on the fly |
| 17:37.00 | unpaidbill | mintee: asterisk-stat |
| 17:37.03 | unpaidbill | is what you want |
| 17:37.52 | artz22 | I have an FXO which works with u-law only.. can that be a problem for music-on-hold and playing the gsm demo files though it? when I try that it just stops or says a frame write error |
| 17:38.13 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.137) |
| 17:39.46 | unpaidbill | asterisk should transcode whatever audio you use assuming you hvve the codecs available artz.. |
| 17:40.01 | unpaidbill | i have gsm audio files and play them through zap channels without an issue |
| 17:40.11 | artz22 | yes i suppose |
| 17:40.25 | artz22 | maybe my driver is wrong although calls work perfectly |
| 17:41.24 | unpaidbill | couldnt tell without seeing your config |
| 17:41.31 | unpaidbill | and even then i probably couldnt tell |
| 17:48.08 | artz22 | well thanks I will see what happens |
| 17:48.14 | artz22 | g2g now. cyas |
| 17:48.16 | *** part/#asterisk artz22 (n=artz@190.189.0.33) |
| 17:52.25 | *** part/#asterisk ecrist (n=ecrist@snipe.secure-computing.net) |
| 17:59.08 | Nasra | I have a newbie question in here please....here we go ...: if I a sapura 3102 ...does that mean I still need a digium card TDM410P or the sap3102 will suffice it? |
| 17:59.45 | [TK]D-Fender | Nasra: SPA-3102 has a fully independent FXO port as well as a fully independant FXS port. |
| 17:59.48 | *** join/#asterisk beek (n=klinebl@65.211.106.242) |
| 18:00.10 | [TK]D-Fender | Nasra: If you need no more than 1 of either then that device would be enough to server BOTH needs. |
| 18:00.27 | Nasra | so I don't need the pcicard ...is this correct? |
| 18:01.08 | Nasra | okay...I think you answerd my question..... |
| 18:01.28 | trapa | If i have s,1,Ringing then s,2,Answer ... and then S,3,Queue(GSS_Queue) ,,, Why don't i hear any rining or background when somone calls in? |
| 18:01.38 | *** join/#asterisk [Akemi] (n=akemi@206-248-159-70.dsl.teksavvy.com) |
| 18:02.12 | Nasra | [TK]-Fender: can I ask you another question? |
| 18:03.27 | Nasra | why when i want to start Asterisk to compile it is telling me something ...like the card ? |
| 18:03.37 | Nasra | is missing |
| 18:07.42 | [TK]D-Fender | Nasra: Nowhere does the installation of * ever complain about not having a card. |
| 18:08.17 | *** join/#asterisk lanning (n=lanning@ip67-152-85-190.z85-152-67.customer.algx.net) |
| 18:08.43 | [TK]D-Fender | trapa: Because your queue answers the call and plays MoH like its supposed to and you didn't tell it to do anything else instead. |
| 18:11.15 | *** join/#asterisk mikkel (n=mikkel@84-238-113-66.u.parknet.dk) |
| 18:14.19 | trapa | Fender: But i don't hear the MOH |
| 18:14.31 | trapa | Not until after the first "Your first in line" |
| 18:16.16 | [TK]D-Fender | trapa: And after that you hear music? |
| 18:18.08 | [TK]D-Fender | Brb |
| 18:18.21 | trapa | Yeah i think so .. *calls to check* |
| 18:23.11 | trapa | Now it's gotten worse, now it provides dialtone ... WTF ... |
| 18:23.26 | *** join/#asterisk ccesario_ (n=ccesario@linux.unialco.com.br) |
| 18:23.36 | *** part/#asterisk martha (n=martha@bzq-179-135-226.static.bezeqint.net) |
| 18:25.05 | *** join/#asterisk MrNaz (n=naz@ppp121-44-207-206.lns3.mel4.internode.on.net) |
| 18:25.28 | trapa | Yeah .. like 10+ seconds of silence, then the "Your call is first in line" then you get MOH |
| 18:25.47 | trapa | No ringing, no nutthin |
| 18:29.19 | *** join/#asterisk JenniferAkemi (n=akemi@69-196-132-4.dsl.teksavvy.com) |
| 18:32.26 | *** join/#asterisk Gareth (i=gareth@www.wiked.org) |
| 18:33.08 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 18:33.45 | [TK]D-Fender | yay, Firefox 3.0 greatness... |
| 18:34.17 | unpaidbill | firefox 3 yea! |
| 18:34.19 | unpaidbill | i just saw that |
| 18:34.37 | unpaidbill | actually i loaded it right as the change went.. and it said firefox 3 download but the link was still to 2.0.0.14! |
| 18:34.40 | unpaidbill | bastards |
| 18:34.43 | kaldemar | "Http/1.1 Service Unavailable" |
| 18:34.56 | unpaidbill | oh shit it went back to 2.0.0.14 |
| 18:35.13 | unpaidbill | fricken a |
| 18:35.28 | DagMoller | Linux: http://download.mozilla.org/?product=firefox-3.0&os=linux&lang=en-US Windows: http://download.mozilla.org/?product=firefox-3.0&os=win&lang=en-US |
| 18:35.28 | Gareth | Anyone seeing issues with IAX2 and 1.4.21? |
| 18:36.12 | unpaidbill | whoa 7mb |
| 18:36.13 | unpaidbill | wtf |
| 18:36.29 | unpaidbill | did they forget to strip debugging out of it |
| 18:38.43 | *** join/#asterisk [Akemi] (n=akemi@206-248-159-80.dsl.teksavvy.com) |
| 18:38.47 | russellb | Gareth: please join #asterisk-bugs and ask for help with debugging it |
| 18:38.51 | russellb | Gareth: so we can get it fixed |
| 18:39.42 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 18:39.47 | [TK]D-Fender | *boing* |
| 18:42.48 | trapa | Fender: Is there a way to make the queue play the "Your first in line" like the instant somone joins .. INstead of playing Moh for a little while then playing that? |
| 18:42.58 | jeev | lol |
| 18:43.01 | jeev | first in line eh.. bastards |
| 18:43.22 | jets | Has anyone noticed odd audio issues in 1.4.21? |
| 18:43.42 | Gareth | russellb: ah cool. thanks. |
| 18:44.21 | jets | (I know, vague, but that's all I have to work with so far.) |
| 18:44.24 | *** join/#asterisk lannm (n=lann@67.in-addr.hurricanelabs.com) |
| 18:45.24 | [TK]D-Fender | trapa: Nothing short of cracking open app_queue.c |
| 18:46.33 | lannm | manager api quesion: can multiple actions be outstanding (i.e. can two actions be sent before a response is received)? |
| 18:47.38 | trapa | fender: Hrm .. any other suggestions to avoid the long pause? (Call 888-685-7732) if you'd like to hear it, if that helps. |
| 18:48.29 | lannm | the wiki entry claims it both ways |
| 18:48.54 | *** join/#asterisk harryv (n=harry@67-207-147-205.slicehost.net) |
| 18:49.24 | harryv | what exactly does cause this: http://pastie.org/216704 .. regular calls onZAP/g1 goes through all fine |
| 18:51.39 | [TK]D-Fender | trapa: Horrible breakup, but yes, silence before MoH. remove "ringing", from the start, then test. Upon failure, pastebin your queue definition and dialplan. |
| 18:52.04 | trapa | thx, okie. |
| 18:52.54 | [TK]D-Fender | harryv: what signalling is that? |
| 18:53.43 | *** join/#asterisk jsmith (n=jsmith@72.21.36.138) |
| 18:53.43 | *** mode/#asterisk [+o jsmith] by ChanServ |
| 18:54.05 | harryv | pri_cpe |
| 18:54.45 | [TK]D-Fender | harryv: Why are you passing CLASS codes to a PRI? These things don't exist on those. |
| 18:55.35 | *** join/#asterisk DagMoller (n=aguirre@unaffiliated/dagmoller) |
| 18:55.55 | harryv | oh. |
| 18:56.14 | unpaidbill | i surprises me how many people at phone companies have never heard of asterisk |
| 18:56.21 | unpaidbill | it.. |
| 18:56.30 | Qwell | unpaidbill: oh, they're aware of it |
| 18:56.34 | harryv | it's a private number that has recently been moved to our ISDN, so someone was trying to use a service normmally provided by the telco. |
| 18:56.37 | unpaidbill | then they play dumb very well |
| 18:56.54 | Qwell | well, if you're talking to a sales guy, or a newb tech.. |
| 18:57.14 | unpaidbill | i have asshat engineers calling me asking what i have, because they're not sure i can accept callerid |
| 18:57.23 | *** join/#asterisk JenniferAkemi (n=akemi@69-196-130-112.dsl.teksavvy.com) |
| 18:57.27 | unpaidbill | haha |
| 18:57.59 | unpaidbill | assholes. |
| 18:58.25 | unpaidbill | even a sales guy should know, if you work in the industry you should be well versed on all this shit imo |
| 18:58.55 | trapa | Fender: http://pastebin.com/d29fc0085 Contains queues.conf, agents.conf, and extensions.conf |
| 18:59.31 | *** part/#asterisk jsmith (n=jsmith@72.21.36.138) |
| 19:02.16 | Kobaz | hmm |
| 19:02.25 | *** part/#asterisk lannm (n=lann@67.in-addr.hurricanelabs.com) |
| 19:04.22 | *** join/#asterisk moy (n=moyhu@nat/ibm/x-913882c10d68a9eb) |
| 19:04.31 | *** join/#asterisk MrNaz (n=naz@ppp121-44-207-206.lns3.mel4.internode.on.net) |
| 19:06.25 | [TK]D-Fender | trapa: exten => 2020,n,Ringing <- told you to remove all of these. |
| 19:06.57 | trapa | oh .. 2020 is just what i use for internal call testing .. I don't think that applies to external calls coming in, Does it? |
| 19:07.22 | [TK]D-Fender | trapa: you pasted way to much useless dialplan. |
| 19:07.24 | trapa | Oh and when i internal dial 2020, that one does ring, then moh, then "your first" ... |
| 19:07.30 | [TK]D-Fender | trapa: Should have jsut shown the relevent bits |
| 19:08.04 | trapa | Sorry :( I'll try not to in future. I figured you'd want it all in case there was a mistake elsewhere that i didn't think was relevant but was. |
| 19:11.32 | [TK]D-Fender | harryv: SMRT |
| 19:15.07 | *** join/#asterisk sniper_sniper (i=michofr@62.84.81.49) |
| 19:18.37 | harryv | [TK]D-Fender: smrt? |
| 19:20.01 | *** join/#asterisk peterpan746 (n=wopulnt@dsl-242-47-184.telkomadsl.co.za) |
| 19:20.11 | peterpan746 | hi everyone |
| 19:20.57 | peterpan746 | i need to install the additional sounds files, but i'm unable to locate them on the web. Does ne know where I can find them? |
| 19:25.15 | [TK]D-Fender | peterpan746: You select them during "make menuconfig" |
| 19:25.40 | peterpan746 | i'm running the selintra SARK |
| 19:26.31 | *** join/#asterisk makkksimal (n=makkksim@e177209126.adsl.alicedsl.de) |
| 19:26.35 | peterpan746 | on sme server |
| 19:27.34 | [TK]D-Fender | peterpan746: http://downloads.digium.com/pub/telephony/sounds/ |
| 19:27.47 | [TK]D-Fender | peterpan746: Next time you might want to actually look at asterisk.org |
| 19:30.24 | *** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee) |
| 19:31.25 | peterpan746 | i was there, looked at svn.digium.com, but got totally confused |
| 19:32.36 | peterpan746 | this was my last resort, but thank you very much |
| 19:33.25 | *** join/#asterisk dioedu (n=dioedu@201.7.117.114) |
| 19:39.44 | *** join/#asterisk l0verb0y (n=l0verb0y@119.111.96.120) |
| 19:43.43 | l0verb0y | anyone have any experience with autodialers? |
| 19:44.58 | sniper_sniper | Hi All...I need to use asterisk server to make a voice test...the user will dial an extension number record his voice and the system will payit back...so I added the following lines to extensions_custom.conf..http://pastebin.com/m6b4bde0a |
| 19:45.14 | sniper_sniper | When dialing the extension I'm getting Serice unavailable |
| 19:45.21 | sniper_sniper | can someone help me plz? |
| 19:45.53 | [TK]D-Fender | sniper_sniper: pastebin the complete call attempt from beginning to end with sip debug enabled |
| 19:47.04 | *** join/#asterisk SteveTotaro (n=Administ@pool-70-17-234-155.balt.east.verizon.net) |
| 19:52.05 | sniper_sniper | [TK]D-Fender, http://pastebin.com/m3b765521 |
| 19:52.46 | sniper_sniper | [TK]D-Fender, It seems from the trace that the Asterisk server is sending the call to the PSTN gateway |
| 19:52.59 | [TK]D-Fender | sniper_sniper: This is a FreePBX dialplan problem and is not supported in this channel. |
| 19:54.56 | *** join/#asterisk Defraz (n=T0tal@fw.fuzecore.com) |
| 19:55.37 | *** join/#asterisk spokra (n=spokra@host093-179-178.sea0.speakeasy.net) |
| 19:55.38 | [TK]D-Fender | sniper_sniper: And however you tried setting it up, it seems its trying to treat the 5678 you dialed as if it were a "phone extension" (by their definition). See : -- Executing [s@macro-dial:7] Dial("SIP/222-084aaa90", "SIP/5678||tr") in new stack |
| 19:55.59 | *** join/#asterisk znoG (n=gs@host238.190-139-62.telecom.net.ar) |
| 19:56.38 | sniper_sniper | [TK]D-Fender, should 5678 be defined as extension or not? |
| 19:57.09 | [TK]D-Fender | sniper_sniper: What you showed is perfectly valid, however it never gets there because of how FreePBX is set up. |
| 19:57.41 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.131) |
| 19:57.59 | sniper_sniper | [TK]D-Fender, I see...what could be the solution?Remove freePBX? |
| 19:57.59 | ManxPower | sniper_sniper: What part of "This is a FreePBX dialplan problem and is not supported in this channel." did you not understand? |
| 19:58.13 | ManxPower | sniper_sniper: The solution is to go ask on the #freepbx channel |
| 19:58.31 | [TK]D-Fender | sniper_sniper: Lean how to use FreePBX, or stop using it and configure * yourself. That retty much sums it up. |
| 19:58.36 | [TK]D-Fender | pretty* |
| 19:58.45 | [TK]D-Fender | learn* |
| 19:58.59 | [TK]D-Fender | yikes... missing keystrokes all over today. |
| 20:01.15 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 20:05.08 | trapa | Fender: Should i re-pastebin stuff or are there other steps i could take to see why the MOH isn't playing right away |
| 20:05.30 | *** join/#asterisk gsiener (n=gsiener@209.169.48.66) |
| 20:06.07 | [TK]D-Fender | trapa: Remove the bit I told you to and pastebin a complete call from beginning to end at verbose 10, core debug 10 |
| 20:06.18 | gsiener | Hi all. I'm evaluating "turnkey" * appliances and wanted to know if anyone would recommend the one they're using |
| 20:06.44 | trapa | asterisk -rvvvvvvvvvvvvvvvv will do verbose 10, how do i get "core debug" ? |
| 20:06.58 | [TK]D-Fender | trapa: Do it at CLI |
| 20:09.22 | *** join/#asterisk AC-Jay (n=Jay@rrcs-24-106-28-178.west.biz.rr.com) |
| 20:09.40 | ManxPower | gsiener: nobody here uses turnkey appliances |
| 20:09.51 | [TK]D-Fender | ManxPower: Not entirely accurate. |
| 20:09.55 | *** join/#asterisk RobH (n=RobH@cpe-72-184-129-136.tampabay.res.rr.com) |
| 20:10.07 | ManxPower | gsiener: this channel is for help using building, configuring Asterisk from source. |
| 20:10.26 | AC-Jay | Hi folks. I have an agi call to a php file on my server. Unfortunately, it says the file does not exist, despite it being there and both readable and executable. Any thoughts why I'd get that error? |
| 20:10.49 | AC-Jay | *and being both* |
| 20:11.24 | [TK]D-Fender | gsiener: http://www.williamsglobal.com/products.asp?type=Detail&id=124103 |
| 20:13.50 | [TK]D-Fender | AC-Jay: pastebin is your friend. |
| 20:16.16 | AC-Jay | [TK]D-Fender: http://pastebin.com/d2ecdddaf |
| 20:16.30 | gsiener | Thanks |
| 20:16.45 | AC-Jay | I snipped the irrelevant parts off. However, if our definitions of "relevant" differ, let me know ;) |
| 20:17.46 | [TK]D-Fender | AC-Jay: And your script please? I also dislike it being root:root |
| 20:18.11 | [TK]D-Fender | AC-Jay: make sure your permissions are right if you're running non-root. |
| 20:18.17 | trapa | fender: The verbose debug is tripping pastebin's spam filter .. Do you need to see the "REGISTER attempt # " Lines? |
| 20:18.24 | [TK]D-Fender | AC-Jay: But do PB your script up |
| 20:18.41 | [TK]D-Fender | trapa: No. |
| 20:18.43 | AC-Jay | yeah I don't like running it as root either, but I couldn't get it to work properly running as a different user |
| 20:18.54 | [TK]D-Fender | AC-Jay: thats fine. PB the script. |
| 20:19.26 | AC-Jay | sure, before I do so, are the contents of the actual script relevant if it says the file isn't even there? |
| 20:20.04 | [TK]D-Fender | AC-Jay: Yes |
| 20:20.10 | AC-Jay | roger. one sec |
| 20:21.17 | *** part/#asterisk peterpan746 (n=wopulnt@dsl-242-47-184.telkomadsl.co.za) |
| 20:22.52 | *** part/#asterisk gsiener (n=gsiener@209.169.48.66) |
| 20:23.55 | ^sandro^ | hello |
| 20:24.54 | *** join/#asterisk angryuser (n=angryuse@d02m-213-44-187-97.d4.club-internet.fr) |
| 20:25.09 | trapa | Fender: Couldn't get it to get past the pastebin spam problem .. so here's the doc http://asterisk.codesavanah.com/asterisk-log.txt |
| 20:25.57 | angryuser | who is using chan_mobile here ? i got majog problem with codec,"unable to find codec translation path from unknown to slin" then asterisk crashed during call |
| 20:26.00 | AC-Jay | [TK]D-Fender: script - http://pastebin.com/d7fdc178b |
| 20:26.21 | ^sandro^ | hello, can anyone help with caller iD ? |
| 20:26.29 | [TK]D-Fender | AC-Jay: kame sure your path the php is correct (which php" |
| 20:26.30 | ^sandro^ | i can't seem to figure out what I am doing wrong and I can't get it working |
| 20:26.38 | AC-Jay | yup. checked that. it's right |
| 20:28.37 | [TK]D-Fender | AC-Jay: Ok, out of ideas ATM. |
| 20:28.42 | AC-Jay | yeah same |
| 20:28.44 | [TK]D-Fender | GTG, heading home, BBIAB |
| 20:29.30 | AC-Jay | is it just me or does it seem like, despite the huge population of this channel, fender is the only one who provides help? |
| 20:29.30 | *** join/#asterisk smth (n=chatzill@CPE000d48380389-CM00195eda2522.cpe.net.cable.rogers.com) |
| 20:29.54 | trapa | Yeah thats prettymuch the way it goes. |
| 20:30.09 | AC-Jay | why the hell isn't he an op then? and why are the ops ops? |
| 20:30.20 | trapa | I try to help out when i can, but i'm a new just like so many others here. Tho there is somone else who helps out some |
| 20:30.39 | trapa | Well probably because the others help when they are online and when they aren't at their computer they just sit and leave the channel window open.. |
| 20:31.08 | trapa | They may not be in the timezone, because i only come here during my working hours ... so it could be beliveable that fender is hte only one who helps out during this timezone |
| 20:31.19 | *** join/#asterisk LeddyHM (i=leddy@da-club.with.my.beerandcondoms.com) |
| 20:31.19 | AC-Jay | it's just misleading to have so many people in that position and no one helping |
| 20:31.21 | trapa | and that the others do lotsa helping other times. |
| 20:31.48 | trapa | Perhaps .. But it's not like they are getting paid to do it. |
| 20:31.57 | AC-Jay | I realize that, and I don't want to seem ungrateful |
| 20:32.23 | AC-Jay | It's just disheartening for a newbie like me to come into a room with almost 300 people (and 11 ops) and have only one person providing help |
| 20:33.22 | trapa | Yeah ... Belive me i'm still a newb too .. and it's funny everyone says asterisk is easy .. I'm sure it is if you work with it on a daily basis .. but i don't and it's not easy for me .. I'm understaning dialplans more now tho .. which gives me hope |
| 20:34.02 | angryuser | and dont forget about gmt, there are pls all over the world here ;) |
| 20:34.20 | AC-Jay | I know. This is a globally-used software |
| 20:34.29 | angryuser | some are sleeping or going to sleep, like me |
| 20:34.34 | AC-Jay | Not a fan of 300 people idling though. |
| 20:35.17 | AC-Jay | As I said, for newbies like me, when I see 300 people in a chat room, ask a question and don't get an answer, it's extremely frustrating |
| 20:35.32 | angryuser | The script, you writed it by yourself ? |
| 20:35.33 | harryv | when i dial a number asterisk waits for 30 seconds before calling it unles i terminate with a # --- any chance there is a way to configure that amount of time? |
| 20:35.47 | AC-Jay | angryuser: me? |
| 20:35.57 | angryuser | <AC-Jay> yes |
| 20:36.18 | AC-Jay | yup. php-agi does the brunt of the work, but I wrote the script myself |
| 20:36.26 | *** join/#asterisk jmardonesk (n=jmardone@200-126-99-94.bk7-dsl.surnet.cl) |
| 20:37.07 | jmardonesk | hello, where is necesary put the wav files for playback or background commands? |
| 20:37.14 | angryuser | <AC-Jay> why do you call yourself a newbie? a newbie ask's about sip config in major ;) |
| 20:38.15 | trapa | harryv: Trying to search for that. Are you using a regular phone? |
| 20:38.17 | AC-Jay | I'm an asterisk newbie; not a PHP newbie :) |
| 20:38.58 | angryuser | <AC-Jay> and i think when you can write some scripts, there is many ways to test, add some vars to see them in logs or in cli |
| 20:38.58 | ManxPower | trapa: Asterisk IS easy -- compared to most PBXs. However Telecom in general and VoIP specifically is very complicated, hard to do right, and poorly documented. |
| 20:39.18 | AC-Jay | angryuser: the problem is that asterisk can't seem to find the php file, despite it being there |
| 20:39.26 | harryv | trapa: it's an analog phone connected to a sipura box |
| 20:39.30 | AC-Jay | "No such file or directory" |
| 20:39.41 | harryv | might actually be a configuration of the sipura .. |
| 20:39.58 | trapa | harry: Is 30 seconds a guess or exact? |
| 20:40.07 | harryv | a guess.. |
| 20:40.16 | trapa | harry: Could be actually. Time it out, find out how many it really is .. |
| 20:40.30 | angryuser | <AC-Jay> hm i am not pro of php or agi, never needed anyway |
| 20:40.42 | angryuser | <AC-Jay> but i sense it is a small thing ;) |
| 20:40.46 | trapa | harry: also .. try going int the asterisk command shell (asterisk -rvvvvvvvv) and see if the asterisk server is recieveing the dial digits ... |
| 20:40.50 | ManxPower | AC-Jay: Can your run the script outside of Asterisk? (even if it just generates some error other than what you are getting) |
| 20:41.05 | trapa | harry: if it is then it might be a asterisk problem, if it's not recieving the digits it's probablyt thye sipura box |
| 20:41.36 | ManxPower | harryv: ALL SIP devices have their dialplan built into the device and then the device sends the whole number to Asterisk at once. |
| 20:41.53 | AC-Jay | yes I can |
| 20:42.02 | AC-Jay | ManxPower, yes. |
| 20:42.04 | AC-Jay | sorry |
| 20:42.08 | ManxPower | AC-Jay: what error do you get? |
| 20:42.12 | AC-Jay | no errors |
| 20:42.32 | ManxPower | AC-Jay: I assume it just hangs? That would be normal for an AGI script run outside of Asterisk |
| 20:42.58 | ManxPower | AC-Jay: and you are ONLY executing the file name, not something like "php /path/to/my/script", right? |
| 20:43.48 | *** join/#asterisk bmg505 (n=leon@196-209-8-145-ndn-esr-2.dynamic.isadsl.co.za) |
| 20:43.50 | ManxPower | AC-Jay: What is the file size listed in "ls -l /usr/bin/php" |
| 20:44.10 | AC-Jay | ManxPower: yes, it hangs, just like it should. and I am trying to execute to both a full path (specified in my script) as well as letting asterisk prepend it with its agi directory |
| 20:44.29 | AC-Jay | 2932732 |
| 20:45.02 | ManxPower | AC-Jay: The number reason for "file not found" is that the program listed on line 1 of the script is not found, so yes, the contents are importan. |
| 20:45.35 | ManxPower | And what are the permissions listed by "ls -l /etc/asterisk/agi/create_filename.php" |
| 20:45.56 | AC-Jay | rwxr-xr-x |
| 20:46.01 | AC-Jay | 0755 |
| 20:46.09 | AC-Jay | whatever's easier for you to read :P |
| 20:46.22 | ManxPower | You don't have something stupid like extra spaces in the AGI line in extensions.conf, right? |
| 20:46.30 | *** join/#asterisk masus (n=ethemc@85.110.227.80) |
| 20:47.03 | ManxPower | AC-Jay: try it with a very simple AGI script |
| 20:47.25 | *** join/#asterisk angom (n=angom@201.170.65.143) |
| 20:47.54 | AC-Jay | nope, checked that too. trying with a simple agi script at the moment. get out of my head :P |
| 20:49.10 | trapa | ManxPower: Do you know much about dialplans and queues. I'm having trouble where the MOH doesn't play till after the "Your first in line" and that results in a very long period of silence (Since no ringing sounds play either) |
| 20:49.20 | UnixDog | anyone here writen or know of a priority queuing agi script |
| 20:51.14 | *** part/#asterisk jicksta (n=jicksta@75-101-5-48.dsl.static.sonic.net) |
| 20:52.00 | UnixDog | looking for a way to put a paid client into a q ahead of other callers |
| 20:52.27 | Strom_C | you don't need an AGI script to do that |
| 20:52.34 | Strom_C | just have two different queues |
| 20:52.39 | Strom_C | one with a higher weight than the other |
| 20:55.25 | trapa | Does anyone know what the "ringing" soundfile is .. or if there is one ? |
| 20:55.54 | smth | hi, how can I collect the "--" leading message of asterisk. are they only available on cli? |
| 20:56.30 | AC-Jay | ManxPower: even the simple agi script fails |
| 20:56.47 | unpaidbill | smth you can use the manager interface |
| 20:57.18 | AC-Jay | simple "echo 'hello world';" script. agi says the script cannot be found |
| 20:57.18 | AC-Jay | "file or directory not found" |
| 20:57.18 | trapa | I found that if i do s,1,Playback(beep) s,2,Queue(QSS_Queue) then the moh plays just fine .. and so that means that if i initate by sending a playback file (instead of s,1,Ringing() because that doesn't work) then it should be good... But it's be nice to send that first sound file as ringing noise. |
| 20:57.33 | AC-Jay | trapa, are you answering? |
| 20:57.45 | AC-Jay | s,1,Answer() |
| 20:58.38 | *** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194) |
| 20:58.39 | trapa | AC: yeah . |
| 20:59.06 | trapa | ac: I had ringing, answer, queue then i had just answer, queue ... now i have answer, playback(beep), queue |
| 20:59.14 | smth | unpaidbill, thanks, |
| 20:59.50 | trapa | adn finally the moh starts playing right away, whereas before it would only start playing moh after the first "you are first in line" which was like 30 seconds into the call .. |
| 21:01.14 | UnixDog | wel well the idea is to have only the base queues but be able to put someone in with a higher priority |
| 21:02.00 | *** part/#asterisk GReggFT (n=GReggFT@fw1-p1.lexi.com) |
| 21:02.17 | AC-Jay | wb fender |
| 21:02.47 | *** join/#asterisk Superbartt (n=bart@ipd50a21c9.speed.planet.nl) |
| 21:04.16 | trapa | Fender: I found that if i did playeback(beep) after answer, and before queue it will feed the moh right away ... But now teh question is why, and whether the ringing noise is a sound file i could playabck to send that one ring before popping somone into the quue |
| 21:04.39 | *** join/#asterisk javawizard2539 (n=javawiza@c-76-23-28-244.hsd1.ut.comcast.net) |
| 21:05.23 | [TK]D-Fender | trapa: you'd have to answer first, playback something like "silence/1", the do "ringing(5) or something. |
| 21:05.47 | [TK]D-Fender | trapa: Though why you'd dump someone in queue with no announcement is beyond me... |
| 21:06.59 | voxter | Whats the story on cpu usage for asterisk 1.6/1.4 vs 1.2? Reduced cpu usage overall, or no? I'm running into a bottleneck trying to Page() 60 phones on a Celeron 2.8ghz box |
| 21:07.03 | voxter | Asterisk 1.2 |
| 21:07.24 | trapa | fender: Good point ... maybe somthing we should be doing. |
| 21:12.18 | [TK]D-Fender | voxter: problem paging that many is just managing the SIP response timeouts for channel creation. mass paging can be a real mess. |
| 21:12.52 | voxter | [TK]D-Fender: Indeed. |
| 21:12.58 | smth | unpaidbill, I just took a look the action of AMI. but still not sure which one can help get the "--" leading mesages? |
| 21:13.29 | voxter | [TK]D-Fender: but, of course, paging is a critical component for these guys, so i have to find -some- resolve. |
| 21:13.50 | *** join/#asterisk LoOoD (n=gman@64.201.247.2) |
| 21:14.36 | LoOoD | how do I tell how long a call is? |
| 21:15.14 | Strom_C | long as in duration? |
| 21:15.18 | Strom_C | or long as in distance from end to end |
| 21:15.39 | LoOoD | duration? |
| 21:15.44 | LoOoD | time |
| 21:15.50 | Strom_C | look in the call detail records |
| 21:16.13 | voxter | [TK]D-Fender: do you know any directives in sip.conf that may be changed to reduce load during a SIP call? Like turning setting nat=no instead of yes, or qualify=no instead of yes, etc? |
| 21:16.31 | unpaidbill | smth, make a manager login, telnet to localhost 5038, login, make a call |
| 21:16.39 | unpaidbill | everything you want is right there |
| 21:16.53 | [TK]D-Fender | voxter: nope, and each phone may react at its own speed as well. |
| 21:17.09 | voxter | [TK]D-Fender: yeah.. they're all either polycom 430, 501, or 601's |
| 21:18.11 | smth | unpaidbill, oh I am going to try it. many thanks |
| 21:18.14 | voxter | This is amusing. asterisk is going between taking 1-2% cpu and 9% cpu during the same call session. no new calls set up or torn down |
| 21:18.20 | [TK]D-Fender | voxter: 1 idea : record the "page" first, then spool up a pile of call-files, or AMI Originates to do this on a more individualistic basis |
| 21:19.17 | jets | Have people been reporting a lot of transcoding problems when compiling with gcc 4.2? |
| 21:19.38 | jets | I only say that because it was a known bug that was closed, but it is now the main compiler in a lot of distros like Ubuntu's latest release |
| 21:19.50 | jets | I just spent a lot of the day tracking down the problem |
| 21:19.58 | voxter | [TK]D-Fender: yeah. The report i got was that some people do not receive the page.. But clearly this is because the machine hits a cpu limit and the last joiners dont make it. |
| 21:20.28 | ManxPower | voxter: "clearly"? |
| 21:20.46 | ManxPower | I doubt it would work with 60 phones even if you doubled the CPU |
| 21:21.36 | [TK]D-Fender | voxter: its not CPU, its phone response timing |
| 21:22.12 | voxter | Uh, I am watching the CPU of the server hit 100% during the Page() |
| 21:22.27 | [TK]D-Fender | jets: I knew of a few with 4.1 |
| 21:22.32 | voxter | and stay there until the page finishes. |
| 21:22.48 | [TK]D-Fender | voxter: Whart zaptel timer are you using? |
| 21:23.09 | voxter | [TK]D-Fender: sangoma a200d, wanrouter driver. |
| 21:23.15 | [TK]D-Fender | voxter: Hrm. |
| 21:23.23 | [TK]D-Fender | voxter: and * / zaptel? |
| 21:23.38 | voxter | [TK]D-Fender: 1.2.17 / 1.2.26 |
| 21:23.54 | voxter | [TK]D-Fender: Interestingly enough, it behaves BETTER with chan_zap.so unloaded in *. |
| 21:24.12 | [TK]D-Fender | voxter: might want to consider upgrading. |
| 21:24.21 | lmadsen | migrating! |
| 21:24.22 | ManxPower | voxter: that is impossible as Page() uses Meetme() and Meetme() requires a timer. |
| 21:24.24 | lmadsen | never upgrade! :) |
| 21:24.38 | ManxPower | It may use the zap timer directly not using chan_zap, I guess. |
| 21:24.41 | Qwell | chan_zap isn't a timer |
| 21:24.41 | russellb | MeetMe does not use a timer! |
| 21:24.41 | voxter | [TK]D-Fender: hence my first question. Is asterisk 1.4 better at managing CPU? :) |
| 21:24.44 | *** join/#asterisk Nugget (i=nugget@carrera.macnugget.org) |
| 21:24.50 | voxter | Everyone is correct. :) |
| 21:24.54 | ManxPower | Qwell: I sit corrected. |
| 21:24.58 | russellb | MeetMe uses zaptel/dahdi for conference mixing, it's not just timing |
| 21:25.02 | ManxPower | russellb: This is no time to be funny. |
| 21:25.07 | russellb | i'm not being funny |
| 21:25.12 | russellb | i'm being accurate |
| 21:25.25 | lmadsen | russellb isn't funny during work hours |
| 21:25.25 | ManxPower | russellb: Dude, this is Asterisk. Nobody is accurate here. |
| 21:25.26 | jets | Hrm. I wonder how we can be more proactive about this 4.2 bug. Perhaps mention it upon 'make' based on the version of gcc? |
| 21:26.00 | Qwell | jets: what bug? |
| 21:26.29 | russellb | ManxPower: oh, then I guess it's ok |
| 21:26.36 | voxter | Is it expected that doing a Page() on 48 phones would max out the CPU on a Celeron 2.8ghz dell? |
| 21:26.36 | jets | Qwell: this is RE: closed bug 10226, which was closed without a resolution. |
| 21:26.59 | jets | Compiling asterisk with gcc 4.2+ results in a lot of transcoding problems. |
| 21:26.59 | jets | A lot of distros are moving towards gcc 4.2 too |
| 21:27.31 | ManxPower | looks at russellb and says "SIP Trunk" |
| 21:27.42 | jets | Oh I love "sip trunking!" |
| 21:27.54 | ManxPower | It's just like a SIP Peer only the same! |
| 21:28.07 | Qwell | voxter: lots of things would max out a celeron |
| 21:28.10 | russellb | I don't have a problem with people using that term :) |
| 21:28.57 | ManxPower | whois russellb |
| 21:29.04 | russellb | ha |
| 21:29.09 | russellb | is an imposter |
| 21:29.34 | ManxPower | russellb: Aren't you supposed to be on#Asterisk-NOW and Asterisk-GUI too? |
| 21:29.38 | *** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk) |
| 21:29.43 | jets | Qwell: this is awful because most of the prompts installed by default are GSM so EVERYTHING sounds like crap |
| 21:29.43 | russellb | ManxPower: nah. |
| 21:29.44 | jets | lol |
| 21:29.56 | Qwell | jets: upgrade, gsm was fixed |
| 21:29.59 | ManxPower | russellb: nobody else from digium either, I assume, except bkruse |
| 21:30.05 | russellb | heh |
| 21:30.11 | russellb | i have nothing useful to contribute in either channel |
| 21:30.25 | ManxPower | russellb: neither does anyone else |
| 21:30.27 | *** join/#asterisk Yourname`` (i=chatzill@unaffiliated/yourname/x-837320) |
| 21:30.27 | jets | Qwell: In trunk? The problem still exists in asterisk 1.4.21 |
| 21:30.38 | bkruse | ManxPower: :) |
| 21:30.41 | russellb | ManxPower: pari should be in the gui channel ... but he stays pretty busy, he might avoid IRC |
| 21:30.42 | Qwell | then it's probably a different issue.. |
| 21:30.51 | UnixDog | ok |
| 21:30.53 | UnixDog | bbiab |
| 21:30.55 | jets | ya I think its specifically compiling with gcc 4.2 |
| 21:31.00 | ManxPower | russellb: I would think that Digium would support those products |
| 21:31.31 | russellb | IRC is not an official support channel for digium products |
| 21:31.40 | russellb | people that bought a digium product can contact digium tech support |
| 21:31.40 | AC-Jay | hmm. does AGI worth with PHP5? |
| 21:31.49 | AC-Jay | s/worth/work |
| 21:31.53 | ManxPower | Why even have those other channels then? |
| 21:31.56 | AC-Jay | whoops |
| 21:31.58 | jets | asterisk-now was mostly a marketing vehicle anyways, wasnt it? |
| 21:32.02 | AC-Jay | hmm. does AGI work with PHP5? |
| 21:32.05 | russellb | ManxPower: for user to user support, for the same reason we have this one |
| 21:32.18 | lmadsen | AC-Jay: of course |
| 21:32.19 | jets | ac-jay: yes, agi just uses stdin/stdout |
| 21:32.21 | brodiem | AC-Jay: AGI doesn't care what it is |
| 21:32.32 | ManxPower | jets: I thought it was to get people to use a GUI then not have anyone to ask for help so they come to this channel |
| 21:32.34 | AC-Jay | I can't find a php5 equivalent of the -q flag, however |
| 21:32.55 | ManxPower | russellb: There doesn't seem to BE any user-user support on those channels, is there? |
| 21:34.03 | russellb | no, why would there be |
| 21:34.07 | russellb | it's a commercial product |
| 21:34.11 | russellb | not an open source thingy |
| 21:34.34 | russellb | the GUI and asterisknow are open source projects |
| 21:34.34 | Qwell | s/BE/<emphasis>be</emphasis> |
| 21:34.34 | lmadsen | you buy BE so you can get support directly from Digium |
| 21:34.39 | russellb | very big difference |
| 21:34.48 | lmadsen | and not the same project even |
| 21:34.51 | Qwell | russellb: ^^ :D |
| 21:34.55 | lmadsen | GUI != AsteriskNow |
| 21:35.03 | mintee | if zap show channels only shows one open channel... doesn that mean that there is only one open channel? |
| 21:35.16 | *** part/#asterisk brodiem (n=brodiem@rrcs-24-227-171-186.sw.biz.rr.com) |
| 21:35.21 | mintee | i'm calling into a channel, that DIAL()'s out to another number using the pri |
| 21:35.27 | mintee | then bridges the calls together. |
| 21:35.33 | mintee | I'm just wondering about usage... |
| 21:35.56 | mintee | zap show channels only shows the original inbound call, and not the bridged outbound connection |
| 21:36.08 | [TK]D-Fender | mintee: "core show channels" |
| 21:36.12 | mintee | right |
| 21:37.09 | mintee | ok, i see them both now |
| 21:37.11 | ManxPower | Thank you, Qwell |
| 21:37.26 | Qwell | ManxPower: yes, there is some. obviously, it isn't a lot |
| 21:37.31 | mintee | and it says they are bridges, but does that mean there are 2 legs that will be charged for the usage? |
| 21:37.38 | mintee | *bridged |
| 21:37.56 | ManxPower | Qwell: The blind leading the deaf, leading the mute, as far as I can tell. |
| 21:38.37 | [TK]D-Fender | mintee: 2 channels is 2 channels. |
| 21:38.41 | Qwell | well, it's a GUI. If something goes wrong, then...well...there's a reason they're using a GUI in the first place. |
| 21:38.59 | mintee | [TK]D-Fender, yeah, that's what I was thinking... Just trying to figure out the CDR's |
| 21:39.04 | ManxPower | Qwell: Then they come here and INSIST on help |
| 21:39.10 | voxter | Hmm. I have a theory on why the cpu usage is going through the roof for paging. The phones are running in g729 mode, but they actually transcode to slin for Meetme, dont they... |
| 21:39.23 | Qwell | there's not much we can do about that |
| 21:39.31 | [TK]D-Fender | Qwell: Yes... they're hoping the GUI is smarter than they are dumb ;) |
| 21:39.33 | mintee | because the CDR's just show that there was an outbound destination, but doesn't report on the times. |
| 21:39.35 | *** join/#asterisk Strom_C (n=strom@208.127.172.112) |
| 21:39.36 | Qwell | people insist for help with freepbx and trixbox as well |
| 21:39.39 | ManxPower | Qwell: cancel the project |
| 21:39.42 | Qwell | insist for? insist on |
| 21:39.45 | AC-Jay | how do you disable agi debugging? |
| 21:39.48 | [TK]D-Fender | voxter: OMG!! |
| 21:39.51 | [TK]D-Fender | voxter: YES <- |
| 21:39.56 | [TK]D-Fender | voxter: that is HORRID! |
| 21:39.56 | voxter | [TK]D-Fender: Durr! right? :P |
| 21:39.57 | ManxPower | I am (partially) joking. |
| 21:39.59 | Qwell | ManxPower: why? It's far better than some "other" things. |
| 21:40.04 | AC-Jay | nm, got it |
| 21:40.15 | voxter | [TK]D-Fender: Can I allow=slin in the polycoms paging extension in sip.conf and have it stop transcoding? or does the polycom even support slin? |
| 21:40.24 | [TK]D-Fender | voxter: MeetMe (the back end) has to mix, and that will to my awareness decode & encode across a LOT. |
| 21:40.27 | ManxPower | Qwell: Because the users are too inexperience to help each other and nobody that knows anything wants to use a GUI. |
| 21:40.35 | UnixDog | you will need transcoding |
| 21:40.38 | [TK]D-Fender | voxter: Are these phones in on a local LAN? |
| 21:40.41 | voxter | [TK]D-Fender: or I guess run it in whatever codec is easiest on the cpu to transcode to slin |
| 21:40.50 | jets | Shouldn't jbot have a standard message of 'that's not what this channel is really about, call the vendor of trixbox/asterisknow/3com branded asterisk box" |
| 21:40.54 | voxter | [TK]D-Fender: they are but i run them in g729 so that when calls go out the IAX trunk they dont need to transcode. |
| 21:40.57 | Qwell | jets: does |
| 21:40.58 | UnixDog | as that most the asterisk sounds are in gsm and you need to transcode them |
| 21:41.06 | voxter | [TK]D-Fender: IAX only, no Zap |
| 21:41.14 | ManxPower | Qwell: In all fairness, it seems like Asterisk GUI sucks a lot less than the others out there. 8-) |
| 21:41.17 | [TK]D-Fender | voxter: welcome to the price you pay. |
| 21:41.20 | Qwell | ManxPower: indeed |
| 21:41.34 | voxter | [TK]D-Fender: ha. So. polycom do slin? or should i transcode on the trunk? :) |
| 21:41.37 | [TK]D-Fender | voxter: Meetme is the culprit and you're pretty screwed |
| 21:41.38 | ManxPower | dialedparties.agi needs to rot in hell. |
| 21:41.46 | UnixDog | lol |
| 21:41.50 | [TK]D-Fender | voxter: Yeah, you should transcode on the trunk.. |
| 21:42.05 | [TK]D-Fender | ManxPower: It's been doing that for a long time already... |
| 21:42.23 | UnixDog | freepbx shold rot in hell |
| 21:42.50 | voxter | [TK]D-Fender: Well, actually, since these polycoms have a DIFFERENT extension that they get paged on, I could always just set the paging extensions to a better codec. What is the least cpu intensive to be translated to slin? probably ulaw i guess. |
| 21:42.52 | UnixDog | trixbox should go to its own special hell |
| 21:42.54 | outtolunc | what? i can't hear you ? <G> your are breakin up <G> |
| 21:43.07 | UnixDog | g711 |
| 21:43.23 | jets | y'know.... i don't understand what trixbox is, other then a few php apps wrapped around freepbx? |
| 21:43.26 | jets | am i missing something or is that it? |
| 21:43.33 | Qwell | jets: that's it |
| 21:43.35 | voxter | jets: thast pretty much it. |
| 21:43.48 | [TK]D-Fender | voxter: Actually... you MIGHT be able to set a codec in some extended dialplan to try to force selection to ulaw if yuou set it all up right. |
| 21:44.12 | jets | hrm I did like their endpoint manager, but i'm certain someone these days has something better and less whorified |
| 21:44.13 | AC-Jay | AGI problem still. AGI call to a php file, asterisk reports the file does not exist. It is there, and php runs it just fine from the CLI |
| 21:44.16 | [TK]D-Fender | jets: FreePBX is the real offender, trixbox is just something that includes it. |
| 21:44.25 | UnixDog | well I find all the guis to be a pain |
| 21:44.35 | ManxPower | Maybe Digium needs to send out letters. Dear Developer, your product/project sucks so much we are no longer allowing you to use the terms zaptel, dadhi, Asterisk, or Digium anywhere in any reference to your product/project. Thank you." |
| 21:44.39 | AC-Jay | Have tried php and php-agi binaries |
| 21:44.52 | AC-Jay | erm |
| 21:44.56 | UnixDog | druid/asterisk-now/freepbx/voiceone/askozia/ |
| 21:44.58 | jets | ac-jey have you tried a php5 agi class? |
| 21:45.03 | ManxPower | If users don't know about Asterisk, they can't come here, can they? |
| 21:45.05 | jets | there's several notes on www.voip-info.org for php5. |
| 21:45.07 | UnixDog | what ones did I miss |
| 21:45.07 | voxter | [TK]D-Fender: For example, if you were extension 105, there are two extensions configured on the phone. 105 = g729 only, I have another, 5105, which i could set to ulaw only. That is of course unless the polycoms do native slin, which would remove the necessity for transcoding altogether. |
| 21:45.09 | AC-Jay | php-cgi |
| 21:45.13 | UnixDog | I know there are a few more |
| 21:45.15 | AC-Jay | is there one, jets? |
| 21:45.17 | AC-Jay | I am using phpagi |
| 21:45.25 | ManxPower | voxter: Polycoms CANNOT do native SLN |
| 21:45.32 | AC-Jay | hmm. I must have missed those. thanks |
| 21:45.50 | voxter | ManxPower: ok that solves that problem then. Any idea which codec is least cpu intensive transcoding to slin? ulaw? |
| 21:46.06 | codefreeze-lap | mintee: I only saw your last comment; everyone: if you are having trouble with CDR's in 1.4 or 1.6.0; please try http://svn.digium.com/svn/asterisk/team/murf/CDRfix4 or http://svn.digium.com/svn/asterisk/team/murf/CDRfix6 (1.4 vs 1.6.0) and see any of your problems are fixed. If not, add your comments to bug 12724, 10927, or 11093, whichever seems more appropriate. |
| 21:46.33 | codefreeze-lap | (or open a new one). |
| 21:46.33 | russellb | even if it does, please provide feedback |
| 21:46.36 | UnixDog | the 3.0.0 firmware haz sln support in it |
| 21:46.47 | UnixDog | but I have not tested it |
| 21:46.50 | voxter | Hmmmmm |
| 21:47.04 | UnixDog | it also has g722 |
| 21:47.07 | voxter | I wonder if i used sln on my paging appearances if meetme would still have to mix |
| 21:47.10 | voxter | er, transcode |
| 21:47.23 | ManxPower | voxter: Why don't you try it and get back to us. |
| 21:47.29 | codefreeze-lap | russellb speaks the truth. If any problems are fixed, positive feedback is definitely welcome as well. |
| 21:47.31 | voxter | ManxPower: I just might do that. |
| 21:48.19 | AC-Jay | jets: do you have a specific link handy? |
| 21:49.02 | trapa | UHm .. Does anyone know what the torture menu's are? |
| 21:50.29 | voxter | trapa: pretty sure those are jokes/examples for telemarketers |
| 21:51.04 | trapa | Okay i'm just listening to all the sound files looking for one that says the call will be recoreded .. and i heard twice now .. "press 3 to send this caller to the torture menu's" |
| 21:51.09 | *** join/#asterisk lateil (n=oz@88.239.82.46) |
| 21:51.22 | trapa | I'd like to activate them and listen to them ,i assume they are endless loop menus |
| 21:52.33 | lmadsen | trapa: you can look for examples on the voip-info wiki -- they are just dialplans |
| 21:52.36 | lmadsen | not built in |
| 21:52.41 | lmadsen | you can create your own if you wish |
| 21:52.55 | trapa | Cool :P |
| 21:57.21 | voxter | wheeee's |
| 21:57.31 | voxter | Changed all the intercom extensions to ulaw, and now i only hit 80% cpu usage. |
| 21:57.42 | voxter | Next to try native slin. |
| 21:59.24 | *** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk) |
| 21:59.51 | [TK]D-Fender | voxter: Again, not possible. |
| 22:00.51 | *** join/#asterisk john_fbac (n=john_fba@181.sub-70-220-116.myvzw.com) |
| 22:01.45 | voxter | [TK]D-Fender: supposedly it is past firmware 3.0 |
| 22:04.07 | voxter | Hmm. I wish there was a way to play a beep to the paged phones inside the meetme, rather than having to use a ringtone on the phone itself. |
| 22:04.08 | *** part/#asterisk AC-Jay (n=Jay@rrcs-24-106-28-178.west.biz.rr.com) |
| 22:08.22 | trapa | is there a site somewhere where i can download more sound files than that which was provided with asterisk? |
| 22:09.01 | lateil | hi guys, i was wondering if it is wise to run asterisk with -p... anyone got idea? |
| 22:10.29 | smth | unpaidbill, I did telnet to AMI and logged in. than made a call to voicemail box. but I did not find any response info about the "--" leading messages which were available on cli. only response the info about the request action. |
| 22:10.30 | Nugget | telnet is eeeeeeevil! |
| 22:12.28 | *** join/#asterisk npratt (n=npratt@pool-71-183-32-116.nycmny.fios.verizon.net) |
| 22:13.37 | npratt | I have Trixbox installed with HudAdmin. Im unable to connect to my running HudAdmin from my HudLite client. UID/PWD and Server passwords have all been verified. Port is open from local machine to server (port 6600). What else can I check? |
| 22:13.49 | *** join/#asterisk javawizard2539_ (n=javawiza@c-76-23-28-244.hsd1.ut.comcast.net) |
| 22:14.01 | [TK]D-Fender | voxter: in your provisioning change the ring-tone to a WAV you set. |
| 22:14.42 | [TK]D-Fender | StanAccy: You can check the channel topic because trixbox / FreePBX are not supported in here |
| 22:14.55 | voxter | [TK]D-Fender: yeah, right now I have it do auto-answer simply because doing a ring-answer can make the page start a few seconds later, leaving a receptionist having to wait 5-6 seconds before speaking. |
| 22:15.09 | StanAccy | Apologies. |
| 22:15.11 | *** part/#asterisk StanAccy (n=npratt@pool-71-183-32-116.nycmny.fios.verizon.net) |
| 22:45.22 | *** join/#asterisk grandpapadot (n=no@adsl-074-185-089-046.sip.bhm.bellsouth.net) |
| 22:46.19 | grandpapadot | Hi all. What would cause a message that says "incompatable codec" after a call has been initiated for a few seconds? The call is initiated with g729 and the audio is fine until the message pops up in the console. My first guess is a re-invite is being issued. |
| 22:48.10 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 22:48.34 | *** join/#asterisk wonderworld (n=ww@ip-62-143-31-149.hsi.ish.de) |
| 22:51.38 | *** join/#asterisk obnauticus (n=obnautic@c-67-160-185-165.hsd1.wa.comcast.net) |
| 22:54.53 | riddlebox | tzafrir_laptop, you around? |
| 22:55.06 | tzafrir_laptop | syes |
| 22:56.15 | riddlebox | hey, you package oslec right? |
| 22:56.47 | tzafrir_laptop | Debian packages include oslec, right |
| 22:57.21 | riddlebox | and do you have them on the current zaptel drivers? |
| 22:59.31 | tzafrir_laptop | I have a version for 1.4.11 in the SVN |
| 23:00.25 | *** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk) |
| 23:01.24 | riddlebox | what kind of patch do you use for those versions, when I compile it I used the 1.4.9.2 patch in the source, but I would think a newer patch needs to be created but I am not sure what it does |
| 23:04.09 | riddlebox | or if the patch is included in the newer zaptel drivers and we dont have to patch it anymore |
| 23:08.48 | *** join/#asterisk km2 (n=x@c-24-23-255-173.hsd1.ca.comcast.net) |
| 23:09.11 | tzafrir_laptop | First off, I omit most of the patch. Most of it is the zaptap device. It is useful ofr debugging, but I'm not really sure how robust it really is |
| 23:09.25 | tzafrir_laptop | And thus the patch becomes much simpler |
| 23:09.35 | tzafrir_laptop | I also build oslec in the same tree as zaptel |
| 23:09.39 | grandpapadot | Anyone know what causes this: Got SIP response 482 "Loop Detected" back from <ip address> ?? |
| 23:11.32 | _ShrikE | Are you getting that back from your provider? |
| 23:11.38 | *** join/#asterisk jameswf-home (n=james@dsl093-157-131.phx1.dsl.speakeasy.net) |
| 23:11.58 | grandpapadot | No, from myself, i.e., the box that's generating the message is the IP being displayed ... |
| 23:12.07 | grandpapadot | Never seen it before, this is a high traffic server ... |
| 23:14.51 | grandpapadot | Could it be related to a re-invite message? |
| 23:15.04 | *** join/#asterisk xacatecas (n=jkroon@dsl-240-142-203.telkomadsl.co.za) |
| 23:15.46 | *** join/#asterisk Quentusrex (n=nobody@ip72-219-211-71.dc.dc.cox.net) |
| 23:15.49 | Quentusrex | Hello all |
| 23:15.50 | xacatecas | hi guys, on an iax2 trunk i've located the notransfer option again, however, if i recall there was an additional option other than a single yes/no, however, I don't seem to be able to locate it at the moment. |
| 23:16.36 | xacatecas | would anybody be able to refresh my memory? specifically, i'd like the actual voice data to go direct between the clients but I'd still like to receive the meta information (eg, call duration) |
| 23:16.40 | jameswf-home | heh save the whales... trade em in for cash and prizes |
| 23:17.30 | Quentusrex | My company started using asterisk 3 years ago. And since then we have grown to 3 separate offices, with a peak call volume of around 100 calls. All of this is(and has been) going through a single off site Asterisk server, but now we need to know how to handle growing call volume. Can anyone point me in the direction of a good tutorial or white paper? |
| 23:19.52 | Quentusrex | Is there a way to have Asterisk load balance calls? |
| 23:20.20 | Quentusrex | but have all of the servers offer the same features and services? such as A2Billing? |
| 23:21.29 | xacatecas | Quentusrex, i've started looking at (ab)using dundi for this purpose. |
| 23:22.09 | xacatecas | and yes, as long as you run everything on top of the same database there should be no reason why you can't have all those features. |
| 23:22.12 | *** join/#asterisk Cobra2599 (i=user@cpe-69-205-158-227.stny.res.rr.com) |
| 23:22.12 | Quentusrex | I've heard of people using openser, but I haven't looked into it yet. |
| 23:22.34 | xacatecas | i've never heard of openser. |
| 23:22.36 | Cobra2599 | Is this the right place to get some help with a new asterisk installation? |
| 23:23.08 | xacatecas | Cobra2599, in theory :) |
| 23:23.12 | Cobra2599 | lol |
| 23:23.14 | lmadsen | Quentusrex: you won't find a tutorial on this subject -- it is infinitely vast and dependent upon your own situation (not to mention complex). However, you can check out my presentation on the subject at http://www.leifmadsen.com/presentations/Why%20Cluster?%20Introduction%20to%20Asterisk%20Clustering..pdf |
| 23:23.32 | Cobra2599 | [Jun 17 12:26:03] NOTICE[17713] chan_sip.c: Registration from '"Daniel"<sip:201@74.201.52.21>' failed for '69.205.158.227' - No matching peer found |
| 23:23.36 | Cobra2599 | thats what i get |
| 23:23.46 | lmadsen | Cobra2599: you don't have a peer [201] |
| 23:23.48 | Cobra2599 | whenever i try to connectto the server with Xlite |
| 23:24.03 | *** join/#asterisk rvhi (n=chatzill@udp255518uds.hawaiiantel.net) |
| 23:24.17 | Cobra2599 | im using freepbx to add the extension so shouldnt that take care of it all? |
| 23:24.27 | Cobra2599 | if not how do i add the peer |
| 23:24.29 | lmadsen | it should -- see topic though |
| 23:24.37 | rvhi | hi, with 'hint' extension for presence, it shows unavailable when the phone rings |
| 23:24.38 | Quentusrex | lmadsen, the link doesn't work |
| 23:24.38 | lmadsen | there is a #freepbx channel |
| 23:25.03 | Cobra2599 | that link you sent earlier is a 404 |
| 23:25.30 | rvhi | that cause some problem, e.g. people wants to know if the receptionist is on the phone or not, if yes, they will pick up the call, |
| 23:25.50 | rvhi | but with the presence, it shows receptionist busy even if she is not busy |
| 23:25.54 | rvhi | just ringing |
| 23:28.33 | xacatecas | lmadsen, that link seems to be a 404. |
| 23:28.52 | lmadsen | xacatecas: I heard the first time .. I was fixing it |
| 23:28.53 | lmadsen | http://www.leifmadsen.com/presentations/Why_Cluster_Introduction_to_Asterisk_Clustering.pdf |
| 23:28.58 | lmadsen | the ? was f'n it up |
| 23:29.23 | xacatecas | oh sorry, didn't see that somebody already told you. |
| 23:29.33 | xacatecas | just skimmed for an update from you ... sorry. |
| 23:29.56 | lmadsen | np |
| 23:32.38 | Cobra2599 | hmm no one is talking over on the freepbx channel |
| 23:32.50 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 23:32.51 | Cobra2599 | can you guys tell me where to add a peer ? |
| 23:33.10 | lmadsen | Cobra2599: we don't know freepbx here |
| 23:33.35 | Cobra2599 | well if i cant use freepbx cause its not working im going to remove it and use just asterisk |
| 23:33.40 | lmadsen | it does some funky stuff that even if we told you were to put it, which would work in standard asterisk, would probably get overwritten by the freepbx gui |
| 23:33.47 | Quentusrex | I've heard of being able to 'connect' multiple asterisk servers with just IAX2, but will that solve my issues? |
| 23:33.49 | lmadsen | great idea |
| 23:33.51 | Cobra2599 | i knwo how to add the extensions in the config files but nothing else |
| 23:34.02 | lmadsen | reading docs will help you :) |
| 23:34.02 | Quentusrex | Will I be able to just toss in an extra server as volume increases? |
| 23:34.02 | *** join/#asterisk NovceGuru (n=NovceGur@oh-65-40-70-180.sta.embarqhsd.net) |
| 23:34.07 | lmadsen | www.asteriskdocs.org |
| 23:34.12 | xacatecas | Cobra2599, freepbx is useful ... until you try to do something that doesn't fit into it's framework. |
| 23:34.32 | xacatecas | and my experience says that that tends to happen the moment you try to use asterisk. |
| 23:35.09 | jeev | lmadsen, curious.. do you take big jobs offered from companies? |
| 23:35.17 | lmadsen | jeev: I do! |
| 23:35.21 | jeev | damn |
| 23:35.23 | jeev | wonder how much you charge |
| 23:35.25 | lmadsen | contact Professional Services at Digium :) |
| 23:35.29 | xacatecas | Quentusrex, not quite. the updated link works and the content is quite interesting, even though it def does NOT tell you "how" to do all. |
| 23:35.33 | jeev | ahh, i'm just curious |
| 23:35.37 | jeev | i could imagine a 100k bill, easy |
| 23:35.37 | jeev | heh |
| 23:35.39 | xacatecas | IAX2 seems to be working pretty well for me. |
| 23:35.51 | lmadsen | jeev: been there, done that, got the scars to prove it |
| 23:35.55 | jeev | haha |
| 23:35.57 | jeev | that's cool man |
| 23:36.36 | *** join/#asterisk implicit (n=implicit@ip68-4-97-211.oc.oc.cox.net) |
| 23:37.08 | jeev | god i hate my life right ow! |
| 23:37.09 | jeev | now |
| 23:37.21 | lmadsen | loves Leif |
| 23:37.23 | lmadsen | :) |
| 23:37.37 | jeev | leif ericson? |
| 23:37.37 | lmadsen | that's p. life, not leaf, fyi... which makes the pun make more sense |
| 23:37.41 | jeev | h |
| 23:37.42 | jeev | bastard |
| 23:37.56 | jeev | i'm <12 miles from my datacenter and broke one of my servers.. too lazy to drive down there. |
| 23:38.00 | lmadsen | the viking is p. layf ericsson |
| 23:38.02 | jeev | my stocks aren't doing well |
| 23:38.11 | jeev | my gf's cousin owes me at least 55k right now |
| 23:38.19 | jeev | i hate looking at my bank accounts |
| 23:38.23 | jeev | i hate it |
| 23:38.38 | lmadsen | looks around and wonders what room he wandered into |
| 23:38.47 | lmadsen | pats jeev on the head |
| 23:38.48 | unpaidbill | ~imapvoicemail |
| 23:38.51 | unpaidbill | ~imap voicemail |
| 23:38.54 | unpaidbill | ~imapvm |
| 23:38.58 | unpaidbill | curses. |
| 23:38.59 | seanbright | settle down |
| 23:39.01 | jeev | kik |
| 23:39.05 | jeev | where is fender to yell at me |
| 23:42.43 | xacatecas | lmadsen, the voicemail in imap idea still messes with my brain. absolutely awesome, but messed. |
| 23:42.57 | xacatecas | my brain just doesn't want to accept that such cool things are possible. |
| 23:43.06 | lmadsen | it isn't |
| 23:43.14 | lmadsen | it's just a figment of your imagination |
| 23:43.39 | xacatecas | is on p47 ... so it _has_ to be true :p |
| 23:44.39 | *** join/#asterisk SteveTotaro (n=Administ@pool-70-17-234-155.balt.east.verizon.net) |
| 23:44.49 | *** join/#asterisk kmshanah (n=kmshanah@cubit.disenchant.net) |
| 23:45.52 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 23:46.01 | esaym | anyone know why asterisk is adding a 1 to the beginning of outgoing pstn numbers? |
| 23:46.21 | seanbright | esaym: because you told it to? |
| 23:46.22 | xacatecas | because you're telling it to? |
| 23:46.36 | seanbright | i win |
| 23:46.41 | esaym | yea that is what I don't get, maybe it is my ata that is doing it |
| 23:48.05 | xacatecas | hmm, random question - for any given call, is it possible to gain access to the codec(s) for the call in the dialplan? |
| 23:48.09 | esaym | actually the ata puts a "+1" at the beginning of the numbers |
| 23:48.40 | xacatecas | uses a nasty hack to strip that crap off should client phone 0027... |
| 23:48.53 | _ShrikE | finally! has asterisk 1.6 B9 playing nicely with exchange 2007 UM |
| 23:49.15 | xacatecas | basically as the first line of all client contexts I've got exten => 0027X.,1,Goto(???,1,0${EXTEN:4}) |
| 23:49.32 | xacatecas | ok, anyway, i'm off |
| 23:52.05 | *** join/#asterisk coppice (n=chatzill@179.202.17.210.dyn.pacific.net.hk) |
| 23:52.11 | harryv | doing some agi stuff, and the file i request to played is playing fine, but the result i get back is: "200 result=0" -- 0 indicates failure, no? |
| 23:54.00 | lmadsen | pretty sure 0 means ok |
| 23:54.39 | harryv | http://www.voip-info.org/wiki-Asterisk+AGI -- "where X is generally 1 for success and 0 for failure. Some commands return additional name=value pairs. Some return a string value in parentheses. |
| 23:54.51 | harryv | but, it seems like it is the other way round. |
| 23:55.07 | seanbright | harryv: what agi command specifically? |
| 23:55.46 | harryv | actauly, it returns 0 no matter the given sound file exists or not. |
| 23:55.56 | seanbright | stream file? |
| 23:55.57 | harryv | seanbright: i'm trying to implement a record-function.. |
| 23:56.23 | harryv | but, i just tried out Playback. it returns 0 even if the file it should play doesnt exist |
| 23:56.34 | *** join/#asterisk pa (n=pa@unaffiliated/pa) |
| 23:56.38 | harryv | seanbright: no, Recordhttp://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Record |
| 23:56.50 | seanbright | record file |
| 23:57.22 | harryv | yep |
| 23:58.33 | seanbright | jeebus |
| 23:58.40 | seanbright | the res_agi code in 1.4 is ugly as sin |