| 00:03.18 | *** join/#asterisk jksM (jks@193.189.93.254) |
| 00:07.22 | jaytee | wow, 3 killed in a shooting at University of Alabama, Huntsville. |
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| 00:11.52 | Snoogan | can anyone shed some light on a small problem I have registering a sip trunk |
| 00:12.26 | Snoogan | when i use the line externip=..... in my sip_nat.conf my sip trunk will not register |
| 00:12.32 | Snoogan | without it, it registers fine |
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| 00:56.48 | ManxPower-work | jaytee: Yup. It's all over the local news. |
| 00:57.57 | jaytee | sad stuff |
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| 01:09.41 | *** join/#asterisk cracoucas83 (~IceChat7@83.194.194-77.rev.gaoland.net) |
| 01:10.15 | trelane | if only they had had firearms wtih which to legally defend themselves from the crazy woman |
| 01:10.18 | cracoucas83 | hi erveybody |
| 01:10.26 | trelane | alas they were totally defenseless, and made so by law |
| 01:10.44 | cracoucas83 | I need to know to get caller number in asterisk extension conf |
| 01:13.11 | cracoucas83 | can someone tell me how to disoplay caller number with variables inside extensions.conf ? |
| 01:13.13 | *** join/#asterisk Caplain (shayne@caplain.loves.boys.fbi.gov.silverelitez.org) |
| 01:13.58 | Pan3D | Caplain: ! |
| 01:15.47 | *** join/#asterisk etfonhomey (~chatzilla@74-131-159-160.dhcp.insightbb.com) |
| 01:19.13 | jaytee | cracoucas83, try NoOp(${CALLERID(num)}) as the first priority in your incoming context |
| 01:19.30 | cracoucas83 | thank you jaytee |
| 01:19.42 | cracoucas83 | can you give me the exact line |
| 01:20.36 | etfonhomey | cracoucas83, for an incoming DAHDI channel - exten => s,1,NoOp(${CALLERID(name)}) |
| 01:20.54 | cracoucas83 | thank you |
| 01:21.04 | cracoucas83 | I also need the called number |
| 01:21.09 | cracoucas83 | in the same line |
| 01:21.15 | etfonhomey | exten => s,2,NoOp(${CALLERID(num)}) |
| 01:21.27 | etfonhomey | oops |
| 01:21.55 | etfonhomey | cracoucas83, for an incoming DAHDI channel - exten => s,1,NoOp(${CALLERID(name)} + ${CALLERID(num)}) maybe? |
| 01:22.04 | cracoucas83 | I am tryning thank you |
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| 01:42.22 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
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| 01:47.24 | corretico | hello people |
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| 02:25.50 | sier | asterisk is reallt hard |
| 02:25.52 | sier | really^ |
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| 03:11.40 | p3nguin | hard? |
| 03:20.35 | russellb | soft? |
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| 03:26.58 | Katty | hi |
| 03:27.12 | Katty | i don't want to know where that conversation was going |
| 03:34.31 | *** join/#asterisk pentanol (~pentanol@77-35-10-184.pppoe.primorye.net.ru) |
| 03:43.56 | *** join/#asterisk jmcdowell (~nooe@174-156-218-136.pools.spcsdns.net) |
| 03:44.00 | jmcdowell | hola all! |
| 03:44.24 | jmcdowell | Anyone have any idea how to solve this problem ? |
| 03:44.35 | jmcdowell | When a call is answered at one phone then transferred to another phone it'd be nice to see the caller ID of the person calling on the new phone rather than the caller ID of the extension the initial phone call is coming from. |
| 03:48.01 | p3nguin | (2025.50) <sier> asterisk is reallt hard |
| 03:48.02 | p3nguin | (2025.52) <sier> really^ |
| 03:52.29 | jmcdowell | Well, not everyone at once! |
| 03:52.35 | jmcdowell | Heres another one. |
| 03:52.53 | jmcdowell | like every time a call comes in to the phone system the phones that don't actually pick up the phone call get a "missed call" |
| 03:52.59 | russellb | jmcdowell: what you're looking for is connected party ID support, which is already done, and will first be in a release in Asterisk 1.8. |
| 03:53.00 | jmcdowell | can I avoid that? |
| 03:53.26 | jmcdowell | I fixed that one, for now. Enabled blind transfer. |
| 03:53.39 | jmcdowell | But the missed call thing, is truly chapping my ass. |
| 03:54.30 | russellb | what version are you using |
| 03:54.51 | jmcdowell | 1.4 |
| 03:55.06 | russellb | ah. As of 1.6.0, the 'c' option to Dial() solves that. |
| 03:55.16 | jmcdowell | blind transfer is a feature of the polycom |
| 03:56.46 | jmcdowell | i am getting ready to do a ubuntu install of the latest version of asterisk though. |
| 03:57.13 | *** join/#asterisk coppice (~chatzilla@94.201.17.210.dyn.pacific.net.hk) |
| 03:57.16 | russellb | " c - If DIAL cancels this call, always set the flag to tell the channel\n" |
| 03:57.16 | russellb | " driver that the call is answered elsewhere.\n" |
| 03:57.27 | russellb | then the phone won't treat it as missed |
| 03:57.41 | jmcdowell | ohh |
| 03:57.42 | jmcdowell | I c |
| 03:57.52 | jmcdowell | I thought you were talking about the transfer |
| 03:58.07 | russellb | nah, that's not taken of until 1.8 |
| 03:58.15 | russellb | the 'c' thing for the missed call issue is there starting in 1.6.0 |
| 03:58.35 | jmcdowell | nice |
| 04:01.16 | jmcdowell | What about this one, I'd like to be able to see who is on what line from any phone. Currently I am unaware if anyone is on the phone just by looking at the screen. |
| 04:02.30 | russellb | i'm sorry, the first 2 questions are free, but the 3rd one costs $100. |
| 04:02.38 | jmcdowell | lol |
| 04:03.25 | russellb | well, in a vague answer ... you are looking for "dialplan hints" |
| 04:03.33 | jmcdowell | I know about the polycom "buddy list", I am just trying to figure out how to make it work. |
| 04:03.37 | jmcdowell | Dial plan? |
| 04:03.42 | russellb | extensions.conf |
| 04:03.49 | jmcdowell | To display who is on the phone at any given time? |
| 04:04.04 | jmcdowell | Really? |
| 04:04.07 | russellb | you put exten 1234 in your polycom buddy list or whatever |
| 04:04.35 | russellb | in extensions.conf, you provide a "hint" that says the state of extension 1234 is mapped to SIP device 1234 ... exten => 1234,hint,SIP/1234 |
| 04:04.46 | *** join/#asterisk came0 (~came0@34.124.188.72.cfl.res.rr.com) |
| 04:04.47 | russellb | so when SIP/1234 is on the phone, a phone watching the state of 1234 gets notified |
| 04:04.54 | jmcdowell | I created the 0000000000000-directory.xml that lists each extension |
| 04:05.04 | jmcdowell | I haven't heard anything about what you are talking about. |
| 04:05.09 | russellb | now you have! |
| 04:05.11 | jmcdowell | hmmmm... |
| 04:05.34 | russellb | and hearing about it is 3.14159% of the battle |
| 04:05.43 | jmcdowell | that will make line 1 say "102 - Dave" or something like that if 102 is on the phone? |
| 04:06.14 | russellb | um, I don't know |
| 04:06.21 | russellb | I don't know what oyu mean by "line 1" |
| 04:06.26 | russellb | and I don't know who Dave is |
| 04:06.40 | jmcdowell | So the 601 has 6 line displays |
| 04:06.41 | drmessano | That does not compute |
| 04:06.41 | russellb | in any case, I need to pass out |
| 04:06.54 | russellb | drmessano can help you from here |
| 04:06.55 | russellb | :-p |
| 04:07.07 | drmessano | Nice |
| 04:07.07 | jmcdowell | I am trying to make it so that if anyone is on a phone call, the given line will say "102" or Dave or what ver. |
| 04:07.13 | jmcdowell | so I know they are on the phone. |
| 04:07.29 | jmcdowell | That's unfourtanate, the doc is on permenent ignore. |
| 04:07.51 | drmessano | Dev-waits-for-reg-to-show-up-so-he-can-pass-the-buck FTW |
| 04:08.04 | jmcdowell | That can never change |
| 04:08.15 | drmessano | lol |
| 04:08.20 | *** join/#asterisk jantman (~jantman@ool-4352721f.dyn.optonline.net) |
| 04:08.51 | russellb | drmessano: lol ... |
| 04:09.16 | jmcdowell | Yep, the doc is out... I will figure it out, just like I did all the other things I needed to know. |
| 04:09.39 | jantman | sorry if this is a bit dense, but I can't seem to find an answer in the docs or online... in the output of "iax2 show peers" what is the ms value in the status column? |
| 04:09.44 | jmcdowell | I am really anxious to build the CVS version of asterisk and freepbx and see where I can take it. |
| 04:10.01 | jmcdowell | Isn't that Milliseconds ? |
| 04:10.14 | *** join/#asterisk coppice (~chatzilla@94.201.17.210.dyn.pacific.net.hk) |
| 04:10.22 | jantman | @jmcdowell - yes, but what is it - jitter? latency? |
| 04:11.02 | jmcdowell | it means that there is N amount of latency between the phone and the host. |
| 04:11.13 | jantman | ok, thanks |
| 04:11.16 | p3nguin | drmessano: It's odd how he ignores many of the people willing to help him. |
| 04:11.30 | drmessano | Indeed.. not sure what i did to earn mine |
| 04:11.57 | p3nguin | drmessano: I think he ignores the people that make him feel the most inferior. |
| 04:12.07 | jmcdowell | p3nguin, the doc is hardly willing to help anyone without running his smart ass mouth |
| 04:12.12 | jmcdowell | I don't need that. |
| 04:12.16 | drmessano | Huh? |
| 04:12.16 | jmcdowell | LOL |
| 04:12.18 | jmcdowell | Inferior |
| 04:12.20 | jmcdowell | LOL |
| 04:12.22 | jmcdowell | that's funny |
| 04:12.46 | drmessano | Most of the time I can tell if I made someone feel like crap.. This one.. Log pls |
| 04:12.48 | jmcdowell | I certainly don't feel that, I work and live in an environment that promotes helping others, and sharing knowlege. |
| 04:12.58 | jmcdowell | WITHOUT being a dick about it. |
| 04:13.08 | jmcdowell | Just as I have ignored you |
| 04:13.12 | drmessano | lol |
| 04:13.16 | jmcdowell | (just now) |
| 04:13.20 | p3nguin | And yet you ignore those who have the knowledge to solve the issues you present. Doesn't add up for me. |
| 04:13.22 | jmcdowell | It's because I have better things to do |
| 04:13.26 | jmcdowell | nothing else.. |
| 04:13.54 | drmessano | He has better things to do.. like whine on IRC about how everyone is so mean to him and how he's going to ignore them |
| 04:13.58 | drmessano | Sounds like a busy guy |
| 04:14.15 | drmessano | Just sayin |
| 04:14.28 | p3nguin | Someone who truly had better things to do would likely take the help to solve the problems being presented. |
| 04:14.52 | drmessano | Yep.. he would have gotten his answer and went back to helping people |
| 04:15.02 | drmessano | "just sayin" |
| 04:16.15 | russellb | gah, i leave you kids alone for 5 minutes and it turns into a flame war |
| 04:16.22 | jmcdowell | LOL |
| 04:16.24 | drmessano | No flame war here |
| 04:16.28 | jmcdowell | no war here |
| 04:16.34 | jmcdowell | just an adult dealing with kiddies |
| 04:16.36 | drmessano | ^^^ passive troll |
| 04:16.47 | russellb | ~due |
| 04:16.48 | infobot | due are u a f*cken bot or something? |
| 04:16.49 | russellb | err... |
| 04:17.20 | p3nguin | Give someone a driver's license and a job, and suddenly they think they're an adult. |
| 04:20.26 | *** join/#asterisk TimRiker (~timr@bzflag/projectlead/TimRiker) |
| 04:20.30 | drmessano | still doesn't know what he did to that dude, but feels even less guilty than if he did know |
| 04:24.13 | *** join/#asterisk pentanol (~pentanol@77-35-10-184.pppoe.primorye.net.ru) |
| 04:25.55 | p3nguin | I'm surprised he didn't start ignoring russellb after the first two answers. |
| 04:27.07 | jaytee | ignoring people is a great way of obtaining help |
| 04:28.04 | jaytee | he's either a troll or one of the most dim-witted bastards that've ever come in here. |
| 04:29.35 | drmessano | Internet Trolling for Dummies, page 17, paragraph 4.. "How not to get banned"... the secret is to convince the channel ops that even if you're calling someone a mindless, monkeyfaced, poo-poo eater, you're still taking the high ground. Ignoring the OPs limits you to what they see and now what they can convince. |
| 04:30.11 | *** join/#asterisk Jasonwert (~jasonwert@97-83-97-13.dhcp.trcy.mi.charter.com) |
| 04:30.45 | russellb | heh, i try to do my little part and help someone, and you guys make it sound like I chose the worst time possible |
| 04:30.46 | russellb | :-) |
| 04:30.58 | drmessano | lol |
| 04:31.01 | russellb | goes back into his hole |
| 04:31.10 | p3nguin | It's not that... you just chose the wrong person to help. |
| 04:31.28 | jaytee | russellb, you're help is always appreciated by most and those that don't appreciate it can suck eggs. |
| 04:31.32 | russellb | now i'm just trolling on you folks. |
| 04:31.38 | drmessano | Apparently I chose something that got me an ignore, but gosh I still don't remember it |
| 04:31.57 | p3nguin | I bet you'll lose sleep over it, too. |
| 04:32.14 | jaytee | but that guy puts everyone on ignore for the slightest little thing and it took the 'tard more than a week and a half to figure out Polycom provisioning |
| 04:32.23 | drmessano | I am slighty OCD.. This is seriously bothering me |
| 04:32.35 | jaytee | would have made faster progress if he hadn't had me and p3nguin on ignore |
| 04:32.36 | p3nguin | hmm |
| 04:32.38 | drmessano | I need closure or I am going to have lock and unlock the front door 64 times |
| 04:32.48 | p3nguin | lol |
| 04:32.51 | jaytee | hehe |
| 04:32.52 | drmessano | Oh no, lost count.. 65.. Need to keep going to 128 |
| 04:32.59 | drmessano | :( |
| 04:33.18 | jaytee | hmmm, doesn't that picture hanging over there look a bit crooked? |
| 04:33.19 | drmessano | jaytee, he put you on ignore too? |
| 04:33.24 | jaytee | whistles shamelessly |
| 04:33.25 | p3nguin | I've never encountered someone IRL who does that. |
| 04:34.15 | jaytee | drmessano, yeah he'd had me on ignore and I didn't know it and I'd been posting pastebins of config info for about an hour. |
| 04:34.22 | drmessano | LOL |
| 04:34.41 | drmessano | That settles it, because jaytee never gets an ignore.. This guy is officially wackballs |
| 04:35.02 | *** join/#asterisk Jasonwert (~jasonwert@97-83-97-13.dhcp.trcy.mi.charter.com) |
| 04:35.58 | jaytee | i'd tried to help him three nights in a row. when he came in again the next night whining about Polycom's documentation I just said, "if you haven't gotten it by now, then you probably never will. Maybe VoIP just isn't your thing." |
| 04:36.01 | russellb | I have something to admit ... I have been addicted to a script related to automated testing of Asterisk for the past few hours ... and it's Friday night ... |
| 04:36.07 | jaytee | that must have got me the ignore |
| 04:36.42 | russellb | i'm glad I'm not the only one up and on here right now, heh. |
| 04:37.12 | jaytee | don't worry, russell, none of us have real lives either |
| 04:37.22 | drmessano | LOL |
| 04:37.37 | p3nguin | waits on more beer to arrive |
| 04:37.41 | russellb | yay beer |
| 04:37.48 | drmessano | russellb, Asterisk actually saves me money by having a distracting support channel |
| 04:38.03 | russellb | nice way to look at it |
| 04:38.20 | p3nguin | I've sent my gopher to fetch some Noble Pils. I hope it's good. |
| 04:38.39 | drmessano | It's probably saved me $30,000 or so in failed marriages that I thankfully never got into because I was too busy on here to meet them.. |
| 04:38.57 | drmessano | Open Source DOES pay |
| 04:39.31 | russellb | so the question still remains ... |
| 04:39.39 | russellb | can the pickle gain more fans than Nickleback? |
| 04:39.51 | drmessano | HAHA |
| 04:39.59 | drmessano | Im rooting for it\ |
| 04:40.08 | p3nguin | pickle? |
| 04:40.20 | drmessano | i think it can.. |
| 04:40.55 | russellb | drmessano: I just did my part. |
| 04:41.18 | drmessano | p3nguin, google: Can a pickle get more fans than nickleback |
| 04:41.29 | drmessano | Yes, the typo is correct |
| 04:41.36 | drmessano | Follow the link to facebook |
| 04:41.48 | ManxPower-work | I've been married. It's nothing special. |
| 04:42.14 | drmessano | Thats what I told Wife #7 |
| 04:42.23 | drmessano | or was it #9 |
| 04:42.26 | drmessano | GAH |
| 04:42.32 | ManxPower-work | Stick with boyfriends/girlfriends. |
| 04:43.10 | jaytee | i keep having intermittent dizzy spells that last about 3 to 5 seconds. had an EEG today that didn't look too good. now I've gotta have an EKG next week |
| 04:43.29 | jaytee | getting old sucks big time |
| 04:44.20 | *** join/#asterisk drako (~luisjose@nelug/coreteam/luisjose) |
| 04:45.32 | *** part/#asterisk jantman (~jantman@ool-4352721f.dyn.optonline.net) |
| 04:46.35 | drmessano | jaytee, human beings are like pieces of abandoned code.. They can make some really neat patches for it, but in the end, unless we learn enough to take over the whole project, we can only patch so much |
| 04:47.04 | jaytee | hahahaa |
| 04:47.22 | *** part/#asterisk ManxPower-work (~EWieling@216.186.151.147) |
| 04:47.33 | *** join/#asterisk ManxPower-work (~EWieling@216.186.151.147) |
| 04:47.51 | jaytee | I'll drop dead and all my Windows developer friends will be going on about....."he was always healthy until he got into open-source!" |
| 04:47.52 | drmessano | "We dont know how the hell this works, but if we keep looping this value through this value here, it makes this routine last longer before it crashes" <--- Medical Science |
| 04:47.53 | p3nguin | Looks like the pickle only has about 379,000 more fans to go. |
| 04:48.09 | jaytee | I became a fan of the pickle |
| 04:48.23 | jaytee | hmmm, that doesn't sound quite right does it? :-) |
| 04:48.26 | p3nguin | I don't do facebook, or I probably would. |
| 04:48.31 | russellb | jaytee: lol.. |
| 04:48.37 | russellb | quick, someone quote that out of context |
| 04:48.50 | drmessano | jaytee, women have been known to drive men that far |
| 04:49.16 | ManxPower-work | jaytee: if I had ops you would be in the topic of one of the channels I'm on. |
| 04:49.23 | drmessano | LOL |
| 04:49.26 | ManxPower-work | well the /topic at least |
| 04:49.28 | jaytee | hahaha |
| 04:51.10 | *** topic/#asterisk by russellb -> Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.2.2, 1.6.1.14, 1.6.0.22 (2010/02/02), 1.4.29 (2010/01/15), *-Addons 1.6.2.0 (2009/12/18), 1.6.1.2, 1.6.0.4, 1.4.10 (2009/12/02), dahdi-linux 2.2.1 + dahdi-tools 2.2.1 (2010/01/20), Libpri 1.4.10.2 (2009/10/20) -=- <jaytee> I became a fan of the pickle |
| 04:51.23 | drmessano | HAHAH |
| 04:51.28 | jaytee | nooooooooooooooooooo! |
| 04:52.08 | drmessano | 1.2.39 is out.. What happens when we get to 1.2.99.. release party for 1.3? |
| 04:52.13 | jaytee | I laughed so hard I got dizzy again....wheeeeeeee |
| 04:52.17 | russellb | 1.2.100 |
| 04:52.43 | drmessano | Is 1.2 following the former release cycle of the 0.99 Wine betas? |
| 04:52.46 | coppice | 1.2.99.1 |
| 04:52.51 | jaytee | is 1.2.39 stable? |
| 04:52.55 | *** topic/#asterisk by russellb -> Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.2.2, 1.6.1.14, 1.6.0.22 (2010/02/02), 1.4.29 (2010/01/15), *-Addons 1.6.2.0 (2009/12/18), 1.6.1.2, 1.6.0.4, 1.4.10 (2009/12/02), dahdi-linux 2.2.1 + dahdi-tools 2.2.1 (2010/01/20), Libpri 1.4.10.2 (2009/10/20) -=- Related channels: #asterisknow,#switchvox,#freepbx,#asterisk-dev,#asterisk-bugs,#asterisk-c |
| 04:52.55 | jaytee | ducks for cover |
| 04:53.09 | p3nguin | #asterisk-c ? |
| 04:53.12 | *** topic/#asterisk by russellb -> Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.2.2, 1.6.1.14, 1.6.0.22 (2010/02/02), 1.4.29 (2010/01/15), *-Addons 1.6.2.0 (2009/12/18), 1.6.1.2, 1.6.0.4, 1.4.10 (2009/12/02), dahdi-linux 2.2.1 + dahdi-tools 2.2.1 (2010/01/20), Libpri 1.4.10.2 (2009/10/20) -=- Related channels: #asterisknow,#switchvox,#freepbx,#asterisk-dev,#asterisk-bugs |
| 04:53.16 | russellb | ran out of space ... |
| 04:53.23 | p3nguin | damn those limits! |
| 04:53.35 | russellb | perhaps it's the 928374923 version numbers in the topic |
| 04:53.38 | *** join/#asterisk pawz (~pawz@ppp118-208-188-17.lns20.bne4.internode.on.net) |
| 04:54.01 | jaytee | maybe if you used mmddyy date instead of "In the year of our Lord....." |
| 04:54.01 | drmessano | snickers something evil about there being 18 different "current" Asterisk releases |
| 04:54.24 | russellb | yeah yeah .. |
| 04:54.39 | drmessano | hey, I was a fan of the 1.6 release cycle |
| 04:55.34 | drmessano | I just didnt think 1.2 would never die, and never expected the drinking binge that turned into 1.4 becoming the LTS release |
| 04:55.45 | drmessano | :) |
| 04:55.56 | russellb | 1.2 will be dead by the end of the year, heh |
| 04:56.02 | jaytee | 1.4 is LTS? I thought that wouldn't happen till 1.8 |
| 04:56.06 | russellb | and 1.4 will be in security only by the end of the year |
| 04:56.51 | drmessano | Yeah yeah |
| 04:56.55 | drmessano | I've heard that before |
| 04:56.57 | russellb | :-) |
| 04:57.03 | drmessano | $year will be the year of the Linux Desktop |
| 04:57.33 | jaytee | roflcopter |
| 04:57.39 | russellb | lollerblades |
| 04:58.00 | drmessano | I guess I understand the 1.4 thing |
| 04:58.24 | drmessano | Based on deployment.. |
| 04:59.04 | drmessano | I'll still never forgot how the 1.6 releases ushered Asterisk into its manhood |
| 04:59.17 | drmessano | *sniff* |
| 04:59.49 | russellb | i'll never forget that one time at band camp |
| 05:00.18 | drmessano | I'm gonna start a facebook page.. "We'll miss you Akerisk 1.6.x" |
| 05:00.35 | drmessano | (name altered for copyright purposes) |
| 05:00.48 | p3nguin | hehe |
| 05:01.09 | p3nguin | Lots of people call it Asterix, so you could use that. |
| 05:01.55 | jaytee | thinks it's cool that the shuttle crew was woken to the theme from Firefly |
| 05:01.55 | drmessano | "Bon Voyage Octothorpe 1.6.x" |
| 05:02.17 | ManxPower | I so much want to go to 1.6, but we have too much legacy stuff deployed on 1.4 |
| 05:02.37 | drmessano | I guess I better upgrade to a 1.6.2.x release before russellb kills it |
| 05:02.40 | drmessano | Just sayin |
| 05:03.13 | jaytee | I want to go to 1.6.1.10 so I can get rid of sipX but my boss is afraid |
| 05:03.36 | drmessano | jaytee, you're going about this all wrong |
| 05:04.04 | jaytee | enlighten me, oh wise one! |
| 05:04.11 | drmessano | Set up a trixbox on an old desktop PC... and tell him "Look, I know you're afraid of 1.6.1.10, but it COULD be worse.. SEE" |
| 05:04.18 | jaytee | lol |
| 05:04.37 | ManxPower | Trixbox looks cool until you try to do much of anything with it. |
| 05:05.03 | russellb | meh |
| 05:05.46 | drmessano | Digium should pick a Dell server style, like the 2900 series and make a replacement front cover for rebranding |
| 05:06.11 | drmessano | Then you can eliminate the fools who want trixbox appliances |
| 05:06.22 | jaytee | I've got about 70% of my custom web gui done. it will allow some brainless, dickless admin to add users and extensions and that's it while preserving the flexibility of asterisk because it doesn't use mysql or any other db, it just edits portions of the extensions.conf and sip.conf files |
| 05:06.30 | russellb | heh, we build our own appliances, but only sell them with switchvox for the most part |
| 05:06.38 | russellb | they're pretty sexy.. |
| 05:06.57 | ManxPower | jaytee: I've been working on the asterisk part of a GUI my boss is writing for clients. |
| 05:07.11 | ManxPower | That and making our polycoms sit up and do tricks. |
| 05:07.49 | jaytee | that jmcdowell dude figured out how to make his Polycoms play dead. |
| 05:08.29 | ManxPower | jaytee: I think he's at like -1 million in polycom karma. |
| 05:08.52 | jaytee | hehe |
| 05:09.09 | drmessano | russellb, I like the Digium boxes, and I can see why they're not sold by themselves.. but seriously, some people buy trixbox appliances to roll their own, or because they're sexy.. |
| 05:09.26 | ManxPower | We've been fighting with polycom about an LLDP bug. Polycom says it is an Adtran bug, Adtran says it is Polycom's bug |
| 05:09.31 | jaytee | chartreuse is sexy? since when? |
| 05:09.41 | drmessano | and they don't claim to be using an specific, optimal components in any way... could be all consumer junk inside for all we know |
| 05:09.47 | *** join/#asterisk Cain` (~Geek@unaffiliated/cain) |
| 05:10.24 | drmessano | Oh, they do have those ATX redundant power supplies.. |
| 05:10.38 | drmessano | *cough* ebay *cough* |
| 05:11.10 | jmcdowell | It's likely an Adtran issue |
| 05:11.11 | russellb | I don't think we could make money on them to justify selling them standalone. "I bought your box and installed shitty linux distro XYZ from 1994 and it won't work" |
| 05:11.22 | jmcdowell | LOL |
| 05:11.25 | drmessano | and that sexy 4-port "Its a NIC!" "No, ITS A SWITCH".. "NO, ITS A NIC!!" "NO, DANGIT, ITS A SWITCH" PCI cards |
| 05:11.36 | drmessano | russellb, exactly |
| 05:11.43 | jmcdowell | I have had SO many issues with adtran switches and they always blame it on someome else. |
| 05:11.52 | drmessano | We dont care |
| 05:11.58 | russellb | drmessano: harsh |
| 05:12.04 | drmessano | Im on ignore |
| 05:12.43 | jmcdowell | We had one rebooting randomly, and they tried to blame it on the power company. Until it woudln't power up, they maintained that. |
| 05:12.44 | drmessano | I could tell you his mom wears combat boots, and its like a fart in a windstorm |
| 05:12.56 | jmcdowell | Finally it died, and we got it replaced and got rid of it. |
| 05:12.58 | jaytee | wonders if "Richard" still has me on ignore |
| 05:13.49 | drmessano | would rather hear if the pickle has any Asterisk issues |
| 05:14.29 | drmessano | My Blackberry keeps rebooting itself |
| 05:14.38 | drmessano | This likely isn't good |
| 05:14.48 | russellb | laptop battery is just about dead ... that's my sign to go to bed for real this time |
| 05:14.50 | russellb | g'night folks |
| 05:15.15 | jaytee | nite russell |
| 05:15.40 | drmessano | Night man |
| 05:15.46 | jmcdowell | l8r |
| 05:22.40 | drmessano | l8r? |
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| 05:37.13 | makafre | hey guys, how do we tell astmanproxy likes our http command or not? |
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| 05:54.36 | p3nguin | drmessano: Yeah, you know... leightr. |
| 05:57.32 | *** part/#asterisk makafre (~makafre@206-248-174-197.dsl.teksavvy.com) |
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| 06:10.22 | dougsk | I was kinda curious what people were using for session border controllers at about 2000 sip trunks (peak). |
| 06:21.15 | *** join/#asterisk troy- (c1917b3ebd@bgp4.us) |
| 06:21.25 | troy- | anyone know where i can buy SMS enabled DIDs? |
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| 07:27.50 | pentanol | hey, any one alive? |
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| 07:55.46 | ChannelZ | no |
| 08:13.22 | carrar | no one here |
| 08:18.57 | pentanol | how you did radius auth with asterisk? which module works well? |
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| 10:01.02 | juancferrer | Is there any way to get the username of the sip client making a call inside extensions.con? Theres variables available for the domain, URI, etc, but there's not one for the username...I could cut it from the URI i guess |
| 10:57.49 | *** join/#asterisk Igramul (~black@p5497C350.dip.t-dialin.net) |
| 10:58.39 | Igramul | Hi, if I trigger a call with a call-file, how can I access the called number in the context (e.g. is there a variable)? |
| 11:02.21 | Gugge | ${EXTEN} maybe ? |
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| 11:13.11 | Igramul | If I want to create a fax send report (fax sent via SendFAX application), should I use system or DeadAGI? |
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| 11:36.28 | [sr] | howdy! |
| 11:36.31 | [sr] | i'm back :P |
| 11:37.09 | [sr] | people, i already have sip extensions configured, but my question is, this HAS to be done fot each extension the the files sip.conf and extensions.conf, correct? |
| 11:37.16 | [sr] | that's the normal procedure? |
| 11:45.02 | kaldemar | yes |
| 11:45.27 | fenrus | well, you can write a macro that match your numbers..? |
| 11:46.25 | kaldemar | or use regcontext and regexten |
| 11:47.44 | kaldemar | along with autocreatepeer=yes if you want an open system |
| 11:55.43 | Pimmetje | I like to have sort number dialing like *12 for a number out of a database or so. Say 001234567890 Mr x has *12. So when i dail *12 it should dail 001234567890 and when Mr x calls (number 001234567890 comes in) it should replace the callerid with Mr x. I like those data to be stored in a database or so something i can easy change. Are there any plugin's for this or do i have to make a program for this? |
| 11:56.50 | Pimmetje | I know softphone's can do this client side but i like to do it server side |
| 11:57.07 | Pimmetje | because i than only have to maintain 1 database |
| 12:00.24 | Gugge | Pimmetje, you _could_ use astdb ... or realtime .... or odbc functions |
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| 12:24.37 | Rajmohan | hi |
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| 12:41.30 | orly_owl | would an IAX ATA work with SIP? |
| 12:42.32 | *** join/#asterisk mnick86 (~mnick86@95-90-234-223-dynip.superkabel.de) |
| 12:43.44 | mnick86 | I have hardware echo cancelling from digium and echocancel=yes ... If i type "dahdi show channel 1" it says "echo cancellation 128 taps \n currently off" |
| 12:48.12 | kaldemar | orly_owl: an IAX ATA works with IAX... but asterisk of course does SIP. |
| 12:48.36 | orly_owl | kaldemar: im using ekiga actually, which is sip, but #ekiga is dead |
| 12:54.57 | orly_owl | kaldemar: so should i not buy it? |
| 12:55.10 | kaldemar | orly_owl: what do you want to do with it? |
| 12:56.04 | orly_owl | kaldemar: use it for ekiga or other voip |
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| 12:56.49 | orly_owl | other voip that im yet to sign up for |
| 12:56.52 | kaldemar | you're not saying much. what ATA is it? is it used to connect to an analog line or a phone? |
| 12:57.01 | orly_owl | http://cgi.ebay.com.au/ws/eBayISAPI.dll?ViewItem&item=140380328245 |
| 12:57.03 | kaldemar | are you using asterisk? |
| 12:57.15 | orly_owl | to connect to a phone |
| 12:57.24 | orly_owl | im not using asterisk |
| 12:58.32 | kaldemar | you can only dial with IAX with that. so no, you can't directly connect to ekiga with it. if you had asterisk in between, you could. |
| 13:04.27 | leifmadsen | IAX --> IAX. SIP --> SIP. IAX--X SIP |
| 13:10.10 | orly_owl | ah |
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| 13:44.41 | squeeb | Hey |
| 13:45.05 | squeeb | Having a bit of a weird problem here, albeit a slightly weird setup, I'm hoping you can help me debug something |
| 13:45.52 | squeeb | We purchased the skype plugin for asterisk, and it works with most skype accounts, we have this 3G Skype Phone which uses g729 as far as I can tell and with the g729 license and codec installed, we can make calls to it from our dialplan |
| 13:46.09 | squeeb | however, when it rings in to the pbx via skype, the phones ring for a split second and then hang up |
| 13:46.13 | squeeb | http://pastebin.org/91136 |
| 13:46.23 | squeeb | this is the cycle, I can't for the life of me figure out what's causing it to hang up |
| 13:47.35 | *** part/#asterisk orly_owl (~DavoDinku@c122-108-50-15.sunsh1.vic.optusnet.com.au) |
| 13:52.13 | squeeb | anybody any idea |
| 13:52.15 | squeeb | ? |
| 13:55.03 | squeeb | [Feb 13 13:55:21] NOTICE[4356]: core.cpp:2138 sfa_call_hangup: ending call |
| 13:55.06 | squeeb | `every time |
| 13:55.09 | squeeb | `every time |
| 13:55.13 | squeeb | any ideas? |
| 14:08.34 | jmcdowell | anyone know of a low cost Voip door phone? |
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| 14:29.27 | errr | what choices for decent sounding text to speach are there? I just tested festival and that sounds worse than microsoft sam.. |
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| 14:47.08 | mnick86 | When I dial to the phone's extension, the phone is NOT ringing, but If I go off hook I get the call established. Any ideas why the phone is not ringing ?? |
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| 15:14.06 | plundra | How do I make the queue actually try members with higher penalty, when the callers QUEUE_MAX_PENALTY have been increased (with help from queuerules) beyond what is required? |
| 15:15.17 | plundra | Do I need to drop back to the dialplan and execute Queue() again? |
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| 15:34.49 | Dibri | hi there, my aserisk seems to eat a lot of resources, 56% cpu etc. op_server.pl takes up the other 40% or so |
| 15:35.02 | Dibri | is this "normal" behaviour? |
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| 15:43.28 | DelphiWorld | hi |
| 15:45.12 | DelphiWorld | anyone using spa901? |
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| 16:34.31 | Dibri | meh it seems a memory issue rather than cpu issue |
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| 16:43.23 | p3nguin | plundra: The penalty on queue members is explained here: http://www.voip-info.org/wiki/view/Asterisk+call+queues |
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| 17:13.39 | Skeeter- | I had a very bad time with the Sangoma A200 this week, when we restart wanrouter, we need to unplug the fxo lines from the card and plug em back to make it work |
| 17:17.23 | *** join/#asterisk Squeeb (squeeb@eggwee.co.uk) |
| 17:17.42 | Squeeb | How can I take the output from an application and put it into a variable? |
| 17:18.03 | Squeeb | IE, I want the return of AddQueuemember in a variable |
| 17:18.09 | Squeeb | so i can test it later |
| 17:19.24 | russellb | There aren't really return values from applications |
| 17:19.41 | russellb | some applications set a SOMETHING_STATUS variable for you |
| 17:19.42 | Squeeb | it says here that AddQueueMember will return ADDED or ALREADY |
| 17:19.51 | Squeeb | ah |
| 17:19.58 | russellb | what does it say exactly? |
| 17:20.31 | Squeeb | Once run, this also sets a variable of AQMSTATUS which is set to one of the following: (1.0, 1.2+) |
| 17:20.33 | russellb | AQMSTATUS it appears |
| 17:20.34 | Squeeb | ADDED |
| 17:20.35 | Squeeb | oh |
| 17:20.38 | Squeeb | MEMBERALREADY |
| 17:20.40 | Squeeb | NOSUCHQUEUE |
| 17:20.42 | russellb | yeah, so, just check ${AQMSTATUS} |
| 17:20.45 | Squeeb | too much coffee, it's making me read every other word :/ |
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| 18:12.53 | rethus | what is a channel in extensions.conf??? like [incoming] [authentication] etc? |
| 18:13.36 | rethus | i try to set a variable |
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| 18:14.35 | rethus | caller comes from [general] to [incoming] from [incoming] to [auth] and in auth to phpagi (test.php) |
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| 18:15.24 | rethus | test.php checks for pin-entry... and set variable "pinentry"... but everytime i try to get PINENTRY it doesn't change |
| 18:16.18 | *** join/#asterisk cuco (~Diego@95.35.116.180) |
| 18:26.01 | [TK]D-Fender | rethus: Would help if you SHOWED US. |
| 18:27.12 | rethus | one moment |
| 18:28.21 | [TK]D-Fender | [13:12]<rethus>what is a channel in extensions.conf??? like [incoming] [authentication] etc? <- and this makes no sense |
| 18:28.48 | *** join/#asterisk xmitter (~xmitter@c-24-21-213-242.hsd1.or.comcast.net) |
| 18:29.56 | rethus | http://pastebin.com/d579b43bc |
| 18:30.08 | *** join/#asterisk af_ (~getsmart@88-149-230-64.dynamic.ngi.it) |
| 18:30.27 | rethus | so the incomming point to context [auth] and in aut, the agi test.php is called |
| 18:30.31 | *** part/#asterisk errr (~errr@fedora/errr) |
| 18:30.58 | rethus | my big problem is, that the variable PINENTRY not stored for next "auth-Loop |
| 18:31.10 | [TK]D-Fender | rethus: Show a call |
| 18:31.20 | rethus | i only wan't to increment the var to check, if 3 login-attemps are over |
| 18:31.24 | rethus | call? |
| 18:31.33 | *** join/#asterisk luke-jr (~luke-jr@2002:62b3:1d4c:0:20e:a6ff:fec4:4e5d) |
| 18:31.37 | rethus | you mean agi-debug-output? |
| 18:31.50 | [TK]D-Fender | rethus: * cli showing how this processes. |
| 18:32.10 | rethus | i have activated agi debug.. should i turn off before? |
| 18:32.49 | [TK]D-Fender | rethus: What do you think? |
| 18:33.01 | luke-jr | so 3:40 AM this morning, some IP in Beijing, China (117.41.229.104) spam-rang my phone -.- |
| 18:33.06 | luke-jr | anyone else seen something like this? |
| 18:33.07 | *** join/#asterisk atis_work (~atis_work@193.238.212.171) |
| 18:33.10 | rethus | http://pastebin.com/d3a8ae58 |
| 18:33.16 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 18:33.23 | rethus | thats with verbose 6 |
| 18:34.30 | rethus | so i get confused about this easy increment for this var |
| 18:34.46 | [TK]D-Fender | rethus: WOW... and that shows NOTHING. Where do I see a variable being set? Where do I see it being refernced? Where do I see any input?> |
| 18:34.59 | [TK]D-Fender | rethus: What was the point of showing me that pastebin? |
| 18:35.43 | rethus | do no what u mean. u say post cli output and i've done |
| 18:35.57 | rethus | i could activate agi debug ? |
| 18:36.32 | [TK]D-Fender | rethus: you asked if you should leave AGI debug on.... without its worthless trash. |
| 18:36.47 | [TK]D-Fender | rethus: C'mon... seriously |
| 18:36.50 | rethus | http://pastebin.com/d780856f4 |
| 18:37.31 | rethus | so what exactly u need for information? |
| 18:37.31 | [TK]D-Fender | rethus: <SIP/dev-b6eefb40>AGI Tx >> 510 Invalid or unknown command <---- wow, does this look good to YOU? |
| 18:37.49 | rethus | thats are only echo outputs |
| 18:38.05 | rethus | i've try also without them, but didn_t work |
| 18:38.30 | rethus | cause i didn't know a way to output a var on cli without this error |
| 18:38.41 | rethus | noop seems not to work out of agi |
| 18:39.02 | [TK]D-Fender | rethus: You can call whatever dialplan app you want from AGI. Thats the POINT |
| 18:39.30 | Skeeter- | I had a very bad time with the Sangoma A200 this week, when we restart wanrouter, we need to unplug the fxo lines from the card and plug em back to make it work |
| 18:45.04 | rethus | [TK]D-Fender: here i have do 3 Pinentrys... and have this agi-debug-output... can u see something out of this? http://pastebin.com/d6eb18b17 |
| 18:45.35 | rethus | on line 34. SET VARIABLE PINENTRY "1" |
| 18:46.11 | rethus | with result "1" on line 35... so i think the var mus be set ?! |
| 18:46.13 | Skeeter- | [TK]D-Fender, i got a coworker online with me, TiCPU , he knows linux much more then me, he is probably going to help you help us more then I would do\ |
| 18:46.25 | rethus | than came the pin-request |
| 18:46.35 | rethus | i enter 11111 |
| 18:47.02 | rethus | an get back to re-enter my pin (auth 1,1) |
| 18:47.41 | rethus | here the PINENTRY seems to be "1" (Line 70) |
| 18:48.16 | [TK]D-Fender | ..... |
| 18:49.04 | rethus | aaarrggghh i see my mistake :-( |
| 18:49.26 | [TK]D-Fender | retuYou're only setting it to "1" |
| 18:49.29 | rethus | i have to use $pin_entry['data'] instead of $pin_entry[result] |
| 18:49.53 | rethus | return is only the flac if request happends or nut |
| 18:50.17 | [TK]D-Fender | rethus: Yes, definitely nuts |
| 18:55.47 | rethus | so how can i output a var on cli without getting an error? |
| 19:01.15 | [TK]D-Fender | rethus: ..NoOp.. |
| 19:03.03 | rethus | thanks... now my pin request works |
| 19:07.08 | TiCPU | I'm trying to debug Skeeter-'s problem, and it's not that simple, as I can see, I'm pretty new in telephony, but can get my way around debugging using the source, it's just that I really don't know what source to look at, it's just like the wanpipe drivers copies sources from the kernel and dahdi and patches them. Is there any tip as to where to look at to know what the driver answers to asterisk, any tool to get line status? |
| 19:09.03 | TiCPU | it always tells me CHANUNAVAIL even with multiple line plugged in, then for it to use a line it must me replugged in the interface, just like it didn't detect the line first, it seems to be a driver problem as when the wanrouter driver is removed and re'insmoded the problem is back |
| 19:09.04 | *** join/#asterisk Wildy (~simba@178.176.30.173) |
| 19:22.20 | TiCPU | when running wanrouter restart it even says in messages that 'Module 1: Line connected on span 1!' |
| 19:24.36 | TiCPU | is there any trace to see what happens when a line is replugged in a module? |
| 19:25.27 | jmcdowell | anyone know of a GOOD door phone ? |
| 19:26.08 | rethus | can i set maximum number of users for a channel? |
| 19:26.50 | jmcdowell | s |
| 19:26.50 | jmcdowell | yes |
| 19:26.58 | [TK]D-Fender | rethus: a channel is a CALL. |
| 19:27.07 | jmcdowell | when you setup the outbound route I beleive you can set the max # of channels. |
| 19:27.33 | [TK]D-Fender | jmcdowell: And what precisely is an "outbound route"? |
| 19:28.28 | rethus | ok i mean a conference |
| 19:29.21 | [TK]D-Fender | rethus: Its your dialplan... limit them there |
| 19:29.31 | rethus | how |
| 19:29.40 | TiCPU | dahdi_monitor -vm on the only line doesn't show a busy tone neither :/ |
| 19:30.03 | [TK]D-Fender | rethus: I don't know.. how about looking at how many callers are IN that MeetMe |
| 19:31.21 | rethus | isn't there an confic-value which i can use |
| 19:31.47 | [TK]D-Fender | rethus: No. MeetMe is a single dialplan app, not an entire conference center solution |
| 19:31.54 | [TK]D-Fender | rethus: No more than * is actually a PBX |
| 19:32.03 | [TK]D-Fender | rethus: its what YOU make it to be |
| 19:32.34 | rethus | ok, thanks |
| 19:32.51 | rethus | but can i set the used codec via agi? |
| 19:33.06 | rethus | or can i change it on runtime... |
| 19:33.29 | rethus | so if i see to many people are connected i switch down to gsm to save bandwith |
| 19:34.10 | [TK]D-Fender | rethus: No. When you're in the dialplan its already too late |
| 19:34.17 | p3nguin | [tk]d-fender: Careful. If you try to teach jmcdowell anything, he'll put you on ignore. |
| 19:35.03 | *** join/#asterisk fofware (~chatzilla@190.7.25.160) |
| 19:36.54 | [TK]D-Fender | p3nguin: I'd still like to hear his answer to my question... |
| 19:37.09 | rethus | ah, found it |
| 19:37.13 | rethus | in asteris.conf |
| 19:37.19 | rethus | maxcalls is the parm |
| 19:37.49 | [TK]D-Fender | retuasterisk.conf won't limit MeetMe specifically... |
| 19:38.14 | rethus | someone have experiance with load average setting on asterisk? |
| 19:38.27 | rethus | how can i see the current load average? |
| 19:38.49 | p3nguin | w, top, htop, or any of various other system tools. |
| 19:40.41 | TiCPU | [TK]D-Fender: Skeeter- told me that you could probably be knowledgable helping me with this specific problem but looking at what type of question you answer, it looks like you have experience with asterisk itself but maybe not with this specific hardware I'm talking about, is it possible? |
| 19:40.43 | *** join/#asterisk smooth_penguin (~smoove@59.95.6.74) |
| 19:46.00 | [TK]D-Fender | TiCall Sangoma support <- |
| 19:48.17 | rethus | how can i fugure out which maxload i should use for my asterisk |
| 19:48.24 | rethus | and where can i check the macload? |
| 19:48.29 | rethus | maxload |
| 19:48.52 | rethus | is this the data i get if i run top on shell? |
| 19:49.30 | TiCPU | they seem to be closed for the week-end :( I'll see what I can do then, I'll probably learn something while searching |
| 19:50.34 | TiCPU | wow, just found out wanpipemon was able to tell line voltage haha |
| 19:55.52 | *** join/#asterisk csababus (~chatzilla@92-249-236-117.pool.digikabel.hu) |
| 19:59.01 | *** join/#asterisk xmitter (~xmitter@c-24-21-213-242.hsd1.or.comcast.net) |
| 20:02.18 | *** join/#asterisk Alagar (~Administr@122.164.38.193) |
| 20:03.34 | *** join/#asterisk Heretic (~fallen@dsl-246-69-127.telkomadsl.co.za) |
| 20:03.44 | Heretic | hi all |
| 20:09.35 | rethus | TiCPU: means u can see the adjustment of a mic ? |
| 20:10.09 | *** join/#asterisk mpe_ (~mpe@0x4dd624b2.adsl.cybercity.dk) |
| 20:12.19 | TiCPU | rethus: the line voltage itself on idle |
| 20:15.37 | TiCPU | getting deeper, found in a regdump that the 4th register is the status and is set to 0x29 after wanrouter restart and after replugging the line it always stay 0xA9 even when line unplugged... getting closer |
| 20:17.12 | rethus | have to gone... thanks for supporting me |
| 20:17.17 | rethus | see ya |
| 20:17.23 | *** part/#asterisk rethus (~contio@p5087397B.dip.t-dialin.net) |
| 20:18.36 | b14ck | Hi everyone! |
| 20:18.43 | b14ck | Beautiful day here in CA :D |
| 20:19.01 | *** join/#asterisk Dibri (~gavit@pop1.isgroup.sr) |
| 20:19.17 | TiCPU | cloudy with snow here :( I want summer now! |
| 20:19.27 | Dibri | when one uses tftp to boot the phone from, it reads my phonemodel.cfg |
| 20:19.45 | Dibri | I specified there that it should download spa$MA.cfg |
| 20:20.08 | Dibri | but it doesn't seem to replace $MA with its mac address |
| 20:20.12 | b14ck | TiCPU, where are you? :) |
| 20:20.26 | TiCPU | Québec, Canada |
| 20:20.51 | b14ck | ah |
| 20:23.21 | TiCPU | Dibri: I'm new to telephony but I do know though that tftpd-hpa is able to substitute MAC address of requester to filename, is that what you're trying to do? |
| 20:27.56 | *** join/#asterisk juancferrer (~juan@adsl-0-91-106.jan.bellsouth.net) |
| 20:28.05 | *** join/#asterisk mchou (~quassel@unaffiliated/mchou) |
| 20:28.47 | Dibri | TiCPU: well the phone is getting my spa922,cfg file in which I secify to download spa$MA or spa$MAC which are diff versions of how a mac address is written(one with : and another without) |
| 20:29.09 | Dibri | TiCPU: for some reason the phone does notdownload this file or it tries t download this file and cannot parse it |
| 20:29.26 | Dibri | so I'm wondering what the spa$ma file should look like |
| 20:29.41 | TiCPU | you mean filename? |
| 20:30.08 | TiCPU | best would be to check using tcpdump or simply the tftpd server log file, you'll see what the phone tries to get |
| 20:32.07 | p3nguin | dibri: Where did you get the sample config files? |
| 20:32.16 | ChannelZ | You can go http://PHONEIP/admin/spacfg.xml to see the entire thing |
| 20:32.30 | p3nguin | dibri: With Cisco phones, there is no : in the file name. |
| 20:32.32 | ChannelZ | where PHONEIP is the phone's IP obviously |
| 20:32.50 | Dibri | p3nguin: with $MA it's without : and with $MAC it's WITH |
| 20:33.07 | ChannelZ | For the configs you make, you don't have to include EVERY option, just the ones you want to change. Mine are like a dozen lines or so to hit the major differences between a running config and the factory defaults |
| 20:33.24 | Dibri | p3nguin: my config files are created with a script called createprov.sh |
| 20:34.00 | Dibri | p3nguin: its from a website wwww.howto.gr/wp/provisioning-linksys-voip-phones/ |
| 20:36.43 | juancferrer | anyone gotten festival working on ubuntu 9.10 server? I see in the festival log that asterisk connection is accepted and then disconnected. But in the asterisk CLI, any time the dialplan gets to the Festival command, it just says "Executing Festival" and "Parsing festival.conf", and then nothing else happens and it won't go to the next dialplan command. It just gets stuck at Festival |
| 20:38.39 | juancferrer | and of course, I never hear the festival voice |
| 20:39.03 | *** join/#asterisk oej (~olle@ns.webway.se) |
| 20:39.23 | p3nguin | juancferrer: Paste the context of your extensions.conf where you are using Festival(). |
| 20:39.28 | p3nguin | pastebin.com |
| 20:39.44 | juancferrer | ok |
| 20:40.16 | Dibri | p3nguin: can u provide me with an example cfg file? |
| 20:41.03 | juancferrer | http://pastebin.com/m238eada4 |
| 20:41.25 | ChannelZ | Dibri: RE: You can go http://PHONEIP/admin/spacfg.xml to see the entire thing |
| 20:41.53 | ChannelZ | Looking at that script it's using some syntax I'm not sure what it means or if it's valid.. all of the ua="na" stuff |
| 20:42.00 | p3nguin | juancferrer: Did you start the festival server? |
| 20:42.36 | juancferrer | it's running, and I'm seeing connection accepted and disconnected in the festival log |
| 20:42.48 | p3nguin | juancferrer: After you configure festival on the system, you'll need to start it with festival --server. You did that already? |
| 20:43.06 | *** join/#asterisk fibres (~no@cpc2-nfds1-0-0-cust1021.lei3.cable.ntl.com) |
| 20:43.38 | juancferrer | yes, festival --server |
| 20:44.03 | juancferrer | but if I use the sample festival init file, I get nothing in the logs |
| 20:44.09 | juancferrer | so that's kinda weird |
| 20:44.11 | p3nguin | juancferrer: Maybe the problem is your festival configuration. Here's what my ~/.festivalrc looks like: http://pastebin.com/d5dbce0ec |
| 20:44.48 | juancferrer | ok, I guess i might need that audio stuff in there |
| 20:45.36 | p3nguin | I also remember having a problem if I ran festival server as root. If I ran it as a regular user, asterisk didn't have any trouble using festival. |
| 20:46.06 | juancferrer | I'll try that as well |
| 20:46.15 | juancferrer | I guess I need to get alsa on this server first |
| 20:46.36 | troy- | anyone know where i can get SMS enabled DIDs? |
| 20:47.04 | juancferrer | I just thought festival made a wav file in memory and then passed it to asterisk to play, I didn't know it played the audio itself |
| 20:47.24 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 20:47.24 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 20:47.39 | p3nguin | I also gave up festival and started recording my own sound files. |
| 20:48.03 | juancferrer | I've got read out stuff that I retrieve from a server |
| 20:59.17 | leifmadsen | festival has awful sunds |
| 20:59.19 | leifmadsen | sounds* |
| 20:59.26 | leifmadsen | cepstral is a lot better |
| 21:02.29 | juancferrer | yeah, i'm going the free route for now |
| 21:02.45 | juancferrer | but still no luck with asterisk |
| 21:03.56 | juancferrer | I can even see the cached files in the cache dir, but it just gets stuck at Festival, and no audio |
| 21:05.37 | thansen | how can I get...core show channels verbose...to show the whole channel? it's getting cut off |
| 21:07.49 | *** join/#asterisk outtolunc (~me@m345636d0.tmodns.net) |
| 21:10.36 | *** join/#asterisk lesouvage (~lesouvage@82.73.69.76) |
| 21:11.54 | [TK]D-Fender | thansen: core show channels conciser |
| 21:11.55 | [TK]D-Fender | thansen: core show channels concise |
| 21:12.10 | ChannelZ | hehe thats what they need.. concise, and conciser |
| 21:13.30 | thansen | [TK]D-Fender: anything else since that is deprecated? |
| 21:13.45 | [TK]D-Fender | thansen: is it GONE? |
| 21:13.55 | thansen | no |
| 21:15.25 | thansen | [TK]D-Fender: but I don't want to program for it if it's going to be GONE in the future :( |
| 21:17.06 | [TK]D-Fender | thansen: Live for now, plan for the future, and cross your fingers and be prepared to kiss your sorry ass goodbye when even the best laid plans fall in ruins :) |
| 21:17.30 | thansen | that was deep |
| 21:17.43 | thansen | still isn't a solution though :D |
| 21:18.18 | ChannelZ | Here's something deep: there are no guarantees |
| 21:19.35 | ChannelZ | (and where is it deprecated? 1.6.2?) |
| 21:19.51 | *** join/#asterisk werdan7 (~w7@freenode/staff/wikimedia.werdan7) |
| 21:21.02 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
| 21:27.00 | leifmadsen | thansen: doesn't matter -- anything beyond 1.6.2 will have cli_aliases so you can always create an alias if required. Additionally, your approach is incorrect. You should be using something like the manager for getting information programatically |
| 21:27.06 | leifmadsen | heads off to get ready for THE KEG! |
| 21:30.49 | thansen | ChannelZ: yes, 1.6.2 |
| 21:31.42 | thansen | ChannelZ: and yes, that was deep too |
| 21:34.19 | thansen | how does a cli_alias help if there's no call in the cli that allows me to see the full channel? oh well |
| 21:36.04 | Raden | <PROTECTED> |
| 21:43.08 | *** part/#asterisk arj__ (~a@82-169-95-126.ip.telfort.nl) |
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| 22:18.01 | markit | hi, I've to file an issue against core-sounds-en.txt of 1.4.17 set, but in mantis, asterisk version, I don't have that version. Will I use 1.4.19 (the nearer version) or am I missing something? category: sounds |
| 22:19.00 | markit | mmm maybe #asterisk-bugs is more apropriate place |
| 22:21.31 | *** join/#asterisk smooth_penguin (~smoove@59.95.6.74) |
| 22:23.25 | *** join/#asterisk Warp4 (~robert.wo@cpe-69-204-101-172.buffalo.res.rr.com) |
| 22:26.20 | *** part/#asterisk markit (~marco@88-149-177-66.static.ngi.it) |
| 22:31.24 | p3nguin | Does anyone even care about bugs that existed on 1.4.17 anymore? |
| 22:33.30 | *** part/#asterisk Gnewt (~hackerlet@li57-94.members.linode.com) |
| 22:36.32 | russellb | (no) |
| 22:39.20 | carrar | haha |
| 22:57.05 | crochat | Hi guys |
| 22:57.29 | crochat | Have a little problem compiling Asterisk with chan_misdn :-( |
| 22:57.44 | crochat | I have compiled and installed mISDN, mISDNuser... |
| 22:58.13 | crochat | mISDNuser headers seems to be installed in /usr/include/mISDNuser |
| 22:59.46 | crochat | chan_misdn (part of Asterisk 1.4.29 channels) needs mISDNif.h, isdn_net.h and l3dss1.h, all present in /usr/include/mISDNuser... |
| 23:00.12 | crochat | But in Asterisk 1.4.29 sources, ./configure can't find it |
| 23:01.20 | crochat | I also tried with --with-misdn=/usr/include/mISDNuser --with-isdnnet=/usr/include/mISDNuser --with-suppserv=/usr/include/mISDNuser but it doesn't work |
| 23:01.40 | crochat | I tried with just /usr path, it's the same problem |
| 23:02.22 | crochat | It keeps telling me something like: configure: *** The mISDN User Library installation on this system appears to be broken. |
| 23:03.00 | crochat | So after that, in "make menuselect", I can't select chan_misdn... normal :-( |
| 23:03.46 | crochat | I'm on this problem since many hours... so please, does anybody have an idea how to get rid of that ? |
| 23:04.24 | russellb | it should find it without any arguments to configure |
| 23:04.35 | russellb | config.log will show what the error is in looking for it |
| 23:05.13 | *** join/#asterisk Failrar (~Failrar@5ED66E6D.cable.ziggo.nl) |
| 23:05.45 | crochat | configure:13979: checking for mISDN_open in -lmISDN |
| 23:06.16 | crochat | configure:14004: gcc -o conftest -g -02 conftest.c -lmISDN >&5 |
| 23:06.26 | crochat | /usr/bin/ld: cannot find -lmISDN |
| 23:07.10 | crochat | It doesn't tell me more about what's going wrong :-( |
| 23:08.36 | ChannelZ | you are missing some isdn library |
| 23:08.48 | ChannelZ | or it's devtools half |
| 23:09.39 | crochat | It just needs those three libraries in /usr/include/mISDNuser |
| 23:09.55 | *** join/#asterisk mpe (~mpe@0x4dd624b2.adsl.cybercity.dk) |
| 23:11.07 | ChannelZ | lsconfig |
| 23:11.09 | ChannelZ | err ldconfig |
| 23:11.32 | crochat | I already ran ldconfig... |
| 23:13.50 | *** join/#asterisk nix8n82 (~AndChat@63.162.27.14) |
| 23:16.28 | russellb | the error is that "libmISDN" can not be found |
| 23:16.37 | russellb | ls /usr/lib/libmISDN* |
| 23:19.09 | crochat | /usr/lib/libmisdn.a /usr/lib/libmisdn_pic.a /usr/lib/libmisdn.so |
| 23:23.46 | crochat | mmm, I tried to symlink those with libmISDN syntax... the errors are now different |
| 23:24.54 | crochat | configure:13979: checking for mISDN_open in -lmISDN |
| 23:24.54 | crochat | configure:14004: gcc -o conftest -g -O2 conftest.c -lmISDN >&5 |
| 23:24.54 | crochat | /tmp/ccSMfeNk.o: In function `main': |
| 23:24.54 | crochat | /usr/local/src/asterisk-1.4.29/conftest.c:168: undefined reference to `mISDN_open' |
| 23:24.54 | crochat | /usr/lib/gcc/i486-linux-gnu/4.3.2/../../../../lib/libmISDN.so: undefined reference to `pthread_create' |
| 23:24.55 | crochat | /usr/lib/gcc/i486-linux-gnu/4.3.2/../../../../lib/libmISDN.so: undefined reference to `pthread_cancel' |
| 23:24.58 | crochat | /usr/lib/gcc/i486-linux-gnu/4.3.2/../../../../lib/libmISDN.so: undefined reference to `pthread_join' |
| 23:41.37 | *** join/#asterisk ruben23 (~AGENT@122.55.48.243) |
| 23:42.44 | ruben23 | hi, i got this warning, does my dahdi broken not properly installed..? ---->http://pastebin.com/m46c4b0b8 |
| 23:45.03 | Squeeb | q |
| 23:45.24 | ruben23 | Squeeb:..? |
| 23:46.45 | *** join/#asterisk tzafrir_laptop (~tzafrir@212.179.75.202) |
| 23:47.25 | *** join/#asterisk Tech_Travis (~Administr@cpe-76-168-191-127.socal.res.rr.com) |
| 23:49.21 | *** join/#asterisk jakent (~john@soleil.johnkent.mooo.com) |
| 23:53.43 | Raden | anyone know where to get 15-17" lcd monitors used cheap ? |
| 23:55.33 | crochat | mmm, it seems that chan_misdn does not work with the new mISDN code (v2) :-( |