| 00:00.07 | EI5GTB | ireland |
| 00:00.09 | rasterix | dont mention skype in here! |
| 00:00.22 | EI5GTB | lol, see the disregard :P |
| 00:00.26 | rasterix | i made that mistake once... got burnt alive |
| 00:00.42 | EI5GTB | heh, tnx for the heads up |
| 00:01.09 | knarfly | success is sweet...now to roll up another blunt and enjoy life as it was meant to be... |
| 00:03.01 | EI5GTB | hmm |
| 00:03.09 | rasterix | as far as i know the only way to call a cell phone is to go through a pstn gateway |
| 00:03.19 | rasterix | so i doubt there are any really cheap options |
| 00:03.22 | EI5GTB | dang |
| 00:03.31 | EI5GTB | so voip is no real advantage.. |
| 00:03.37 | *** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.itb.ac.id) |
| 00:03.45 | EI5GTB | yea |
| 00:03.52 | EI5GTB | give everyone a dam radio :P |
| 00:03.58 | EI5GTB | <-- ham radio callsign |
| 00:04.00 | rasterix | not for that... but i like i said... im probably wrong... u need to wake the sleeping blue people |
| 00:04.10 | EI5GTB | blue people? |
| 00:04.28 | rasterix | angler... bkruse... corydon etc |
| 00:04.44 | EI5GTB | ooh. their green here :P |
| 00:05.40 | r0land | any1 have any advice: http://pastebin.com/d54970e04 |
| 00:05.44 | rasterix | ei5gtb what actually is the problem? |
| 00:05.59 | rasterix | is it ur kids calling cells from the landline? |
| 00:06.40 | EI5GTB | i should clear that up....im the kids :p |
| 00:06.44 | rasterix | its probably cheaper for them to call from their own cells... |
| 00:06.47 | EI5GTB | well, i dont call cells |
| 00:06.50 | *** join/#asterisk implicit (n=bayan@unaffiliated/implicit) |
| 00:06.55 | EI5GTB | but others in the family do |
| 00:07.09 | EI5GTB | but that aside |
| 00:07.26 | EI5GTB | im wondering can i get a cheaper deal with voip than 30 euro a month for landliens |
| 00:07.47 | rasterix | how many landlines do you have? |
| 00:07.57 | EI5GTB | just the one |
| 00:08.10 | florz | how many minutes of calls do you have there? |
| 00:08.13 | rasterix | whats the broadband like in ireland? |
| 00:08.38 | EI5GTB | well, i have a 3mb package |
| 00:08.42 | RecycleBin | Would i be using the same card for voip that i would for a standard line ? |
| 00:08.42 | EI5GTB | get about 312kb up |
| 00:09.07 | florz | RecycleBin: what card and what "standard line"? |
| 00:09.14 | rasterix | wont you need the landline still for your broadband? |
| 00:09.16 | EI5GTB | florz, could be an hour or 2 |
| 00:09.22 | RecycleBin | standard land line |
| 00:09.29 | EI5GTB | rasterix, yea, but i could |
| 00:09.42 | EI5GTB | <PROTECTED> |
| 00:09.45 | EI5GTB | and only pay for bb |
| 00:09.50 | florz | EI5GTB: erm ... now, we are talking about a flat rate for calls to landline, or the costs of a land line to your house? |
| 00:10.28 | EI5GTB | prepares problem statment |
| 00:10.34 | rasterix | ei5gtb... get bt talk and everyone u know to signt up |
| 00:10.34 | RecycleBin | http://www.digium.com/en/products/analog/tdm410.php <-- that was the card i was going to get for my landline. But its going to make sence in the future to use voip i was hoping that card would work for voip |
| 00:11.02 | florz | RecycleBin: you may want to read up on what "voip" means |
| 00:11.13 | rasterix | recyclebin: you dont need a card for voip |
| 00:11.23 | RecycleBin | Hmm |
| 00:11.26 | RecycleBin | interesting |
| 00:11.46 | EI5GTB | ok, atm i pay like 60 a month for broadband and landline calls (UNLIMITED) im wondering could i cet a deal with a voip provider for calls to landlines for less than 30 or 40 a month |
| 00:12.40 | RecycleBin | Vonage does unlimited local and long distance for about 40 a month |
| 00:12.48 | florz | EI5GTB: probably - the problem is, the deal with the voip provider won't help you any without internet access |
| 00:13.20 | r0land | EI5GTB calcentric has a flat rate plan for 30 $ to 34 countries |
| 00:13.24 | rasterix | florz: i think he is saying he pays extra for the unlimited landline calls |
| 00:13.39 | EI5GTB | lul, i know, but i could stop paying 40 a month for unlimited calls to mobiles and only pay the 20 or 30 a month for internet |
| 00:13.50 | florz | EI5GTB: I've got no clue of the rates in Ireland, but this side of the north sea, you'd be able to buy 2 hours worth of calls to landlines for ~ 1.2 EUR |
| 00:13.50 | EI5GTB | r0land, that sounds decent |
| 00:14.00 | EI5GTB | wowzers.. |
| 00:14.16 | RecycleBin | So for voip, i just need a nic card is it ? |
| 00:14.29 | r0land | RecycleBin and a 512 con with low latency |
| 00:14.30 | florz | RecycleBin: no, you need IP access |
| 00:14.31 | rasterix | recyclebin: yes |
| 00:14.43 | RecycleBin | I have a broadbank connection |
| 00:14.47 | rasterix | and internet access of course |
| 00:14.53 | r0land | RecycleBin depends on ur latency |
| 00:15.06 | r0land | latency= number of switches/routers between you and ur internet gateway |
| 00:15.08 | florz | RecycleBin: you can use IP over RS232 if you like ... |
| 00:15.15 | EI5GTB | ooh |
| 00:15.17 | EI5GTB | hig speed :) |
| 00:15.31 | RecycleBin | Hmm, i dont mind using the landline, but when i get into long distance.. Yikes |
| 00:15.48 | florz | EI5GTB: Well, current incarnations get into the megabit/s range ... could put quite a few calls through that |
| 00:15.48 | r0land | theres alot of plans voip wise... |
| 00:16.03 | RecycleBin | Well, i guess it wont hurt to have the card |
| 00:16.06 | r0land | my fav is callcentric.. though @ work we use vonage |
| 00:16.16 | EI5GTB | florz, sshhh |
| 00:16.29 | EI5GTB | :P |
| 00:16.37 | rasterix | recyclebin: the card will provide a timing source for asterisk |
| 00:16.53 | rasterix | recyclebin: although u can use ztdummy |
| 00:17.13 | EI5GTB | im hungry, does asterisk have a Feed(); ? |
| 00:17.16 | RecycleBin | Its all local calling until i get a process worked out |
| 00:17.26 | r0land | EI5GTB http://www.callcentric.com/rate_plans04.php |
| 00:18.00 | RecycleBin | Thats decent |
| 00:18.19 | RecycleBin | Maybe i should scrap the card and go voip straight away, saves me a lot of money |
| 00:18.24 | r0land | anyway u guys.. |
| 00:18.32 | r0land | any1 has asterisk experience tht could help out?! |
| 00:18.48 | Qwell | ~ask |
| 00:18.48 | jbot | [ask] Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 00:19.11 | rasterix | i forgot what your question was roland |
| 00:19.27 | RecycleBin | So, in the asterisk box, if im using voip it will connect via the network ? |
| 00:19.46 | rasterix | recyclebin: yes |
| 00:19.57 | RecycleBin | Sweet, cuts out 200 bux for my project |
| 00:20.01 | RecycleBin | Always nice |
| 00:20.03 | *** part/#asterisk beek (n=klinebl@65.211.106.242) |
| 00:20.10 | Qwell | well, you've still got to pay for phone calls.. |
| 00:20.15 | RecycleBin | Yeah |
| 00:20.19 | r0land | i have the following topolog: sipphones <-> asterisk <-> sipura3102... sometimes when i call/recieve a call from sipura on my sipphone and another phone call comes through! something gets messed up and i could hear the attendant welcoming the caller and then some1 asnwering the call..! the 2 phone calls merge.. and when i ask the other person to hang up! my line breaks as well.. |
| 00:20.23 | RecycleBin | i understand that bit |
| 00:20.29 | rasterix | yes but he doesnt have to pay out for a card |
| 00:20.38 | RecycleBin | exactly |
| 00:20.45 | voxter | Anyone know an acceptable price monthly for PRI service in around the Phoenix Area? |
| 00:20.49 | RecycleBin | plus when i get into long distance its cheaper |
| 00:20.52 | voxter | I'm used to canadian prices.. :) |
| 00:20.55 | Qwell | How are you going to make calls? |
| 00:20.55 | EI5GTB | r0land, cheers for the link, should work good |
| 00:21.10 | RecycleBin | Me or voxter ? |
| 00:21.15 | voxter | You :) |
| 00:21.17 | r0land | ps: ive got callcentric configured on my htc vox, i usualy go from one hotspot to another! so im always using my callcentric account :) i hardly have to pay anything to my cell operator |
| 00:21.19 | EI5GTB | rasterix, cheers for the help, bed time here, talk again |
| 00:21.23 | *** join/#asterisk jeffspeff (i=jeff@c-98-240-113-135.hsd1.ky.comcast.net) |
| 00:21.25 | r0land | EI5GTB :) |
| 00:21.25 | RecycleBin | I want to autodial a list |
| 00:21.32 | Qwell | voxter: I bet jameswf would know |
| 00:21.41 | Qwell | he's not here though.. |
| 00:22.31 | rasterix | recyclebin: www.voip-info.org |
| 00:22.37 | RecycleBin | Ty |
| 00:24.49 | rasterix | qwell: i asked earlier about asterisk variables... ${myvariable:5} = substr(${myvariable},5) |
| 00:25.06 | r0land | Qwell any advice about my current config : http://pastebin.com/dcf539d8 <-- when i call in, my attendant doesnt pick up.. :( |
| 00:25.30 | Qwell | rasterix: uh huh? |
| 00:25.51 | rasterix | i think the syntax should change since currently you can have ${myvariable:${someothervariable}} |
| 00:26.01 | Qwell | why? |
| 00:26.06 | rasterix | so then you have a variable within a variable |
| 00:26.16 | Qwell | you can |
| 00:26.37 | rasterix | structurally its not good |
| 00:26.42 | Qwell | says you |
| 00:26.45 | Qwell | man bash |
| 00:26.49 | rasterix | lol |
| 00:26.52 | rasterix | im bitter |
| 00:27.19 | rasterix | because its messing up my attempts to do a lexical parse on extensions.conf |
| 00:27.28 | Qwell | don't do that then |
| 00:27.32 | rasterix | i cant think of another language where this could happen |
| 00:27.37 | Qwell | bash |
| 00:27.54 | rasterix | im not hating |
| 00:28.05 | *** join/#asterisk Levonk (n=lk@adsl-76-237-15-42.dsl.lsan03.sbcglobal.net) |
| 00:28.20 | rasterix | i just think removing it and using a substr() function would be more elegant |
| 00:28.48 | Qwell | ${SUBSTR(${myvariable}|5)} vs ${myvariable:5} |
| 00:28.56 | Qwell | how is that more elegant? |
| 00:30.03 | rasterix | because a variable should be just that... an entity |
| 00:30.11 | Qwell | says you |
| 00:31.31 | rasterix | if you are parsing a language it needs to be tokenised |
| 00:31.45 | Qwell | So tokenize it |
| 00:31.59 | rasterix | i have a variable within a variable |
| 00:32.55 | *** join/#asterisk mvanbaak (n=michiel@asterisk/contributor-and-bug-marshal/mvanbaak) |
| 00:33.04 | rasterix | its just a thought |
| 00:33.26 | Qwell | both bash and Asterisk are able to handle that syntax just fine... |
| 00:33.33 | Qwell | it's clearly able to be parsed |
| 00:34.15 | rasterix | ok we agree to disagree |
| 00:34.45 | r0land | so will some1 agree to give a newbie (me) a helping hand! |
| 00:35.58 | rasterix | just because a language can be handled by asterisk's interpreter doesnt make it syntactically correct |
| 00:36.45 | *** join/#asterisk EI5GTB (n=EI5GTB@193.120.192.61) |
| 00:36.53 | rasterix | and bash is an old language so i dont see how citing that as an example of sound rules is correct |
| 00:38.21 | *** join/#asterisk Strom_C (n=strom@208.127.172.112) |
| 00:42.26 | Qwell | r0land: no, please don't message me |
| 00:42.47 | Strom_C | msg qwell omg hello lol |
| 00:43.05 | jblack | heh. tk and I gave up on him earlier. Now he's msg'ing people? |
| 00:43.19 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
| 00:43.24 | rasterix | r0land is very excitable lol |
| 00:43.41 | jblack | Good thing his phone system ain't right. Otherwise, he'd start calling next. |
| 00:43.55 | r0land | :) |
| 00:44.16 | RecycleBin | Man, so much for being up and running a couple days. This is like a 2 weeks project and thats if i get my parts and voip in a week |
| 00:44.29 | EI5GTB | i think we should use asterisk to make a new phone system, that rocks. and takes over from everything else |
| 00:44.34 | jblack | RecycleBin: Well, if you go straight voip... |
| 00:44.35 | EI5GTB | cos im bored, and i like that sorta thing |
| 00:44.48 | ManxPower | RecycleBin: No, it will take you at least a month |
| 00:45.02 | rasterix | qwell: im not trying to knock asterisk btw i love it... i just think there are certain aspects of extensions.conf language that could be improved |
| 00:45.13 | RecycleBin | ManxPower, Thats likely a fair estimate |
| 00:45.33 | jblack | rasterix: Try ael. |
| 00:45.34 | RecycleBin | jblack, i am going straight voip |
| 00:46.08 | jblack | so, you shouldn't need any hardware at all, excluding headsets you can pick up at walmart tonight. |
| 00:46.13 | rasterix | jblack: looked but tbh unless its adopted by asterisk core im not going to spend time on it |
| 00:46.32 | *** join/#asterisk jeffspeff (n=jeff@c-98-240-113-135.hsd1.ky.comcast.net) |
| 00:46.42 | ManxPower | rasterix: um, AEL IS part of the core Asterisk. |
| 00:46.43 | jblack | rasterix: I must not understand you. I think ael is in asterisk today. |
| 00:46.51 | RecycleBin | True, but i still need to shop for an old box |
| 00:47.03 | RecycleBin | Not sure my old lappy is a good candidate |
| 00:47.07 | Qwell | has been for...3+ years? |
| 00:47.21 | *** part/#asterisk EI5GTB (n=EI5GTB@193.120.192.61) |
| 00:47.21 | rasterix | my bad... |
| 00:47.26 | *** join/#asterisk EI5GTB (n=EI5GTB@193.120.192.61) |
| 00:47.32 | rasterix | i googled it a while back |
| 00:47.32 | Chicago | Hmm... I have made Zork finally kinda work. |
| 00:47.37 | rasterix | and got the wrong impression |
| 00:48.05 | Chicago | There is something I have done wrong though. When I call in and hear zork talk to me, nothing responds when I talk back. |
| 00:48.31 | Chicago | Also, I understand pressing # should indicate I am done speaking, it doesn't seem the # is recognized either. |
| 00:48.32 | EI5GTB | wtf, dumb internet quitting me |
| 00:49.25 | Chicago | The only clue I have is "Use of uninitialized value in concatenation (.) or string at /usr/lib/perl5/vendor_perl/5.8.8/Asterisk/AGI.pm line 1049, <STDIN> line 28. |
| 00:49.54 | *** join/#asterisk jeffspeff (n=jeff@c-98-240-113-135.hsd1.ky.comcast.net) |
| 00:51.22 | jblack | zork on *. That's cool |
| 00:52.17 | rasterix | looking on voip-info it says AEL is still considered experimental at this stage? |
| 00:53.16 | *** join/#asterisk jeffspeff2 (i=jeff@c-98-240-113-135.hsd1.ky.comcast.net) |
| 00:53.18 | florz | you are on the safe side if you simply consider asterisk experimental as a whole |
| 00:53.19 | ManxPower | rasterix: That page is years out of date. That applies to Asterisk 1.2 |
| 00:53.23 | jblack | Heh. there were two develpers here that told you it's been in * for years, so you're gonna go off to a wiki and quote it as saying it's experimental |
| 00:53.36 | ManxPower | rasterix: much of the information on voip-info is out of date |
| 00:53.36 | rasterix | no im just asking |
| 00:54.35 | rasterix | so long term is AEL going to replace extensions.conf? |
| 00:55.30 | *** join/#asterisk jeffspeff2 (i=jeff@c-98-240-113-135.hsd1.ky.comcast.net) |
| 00:55.37 | Qwell | rasterix: never |
| 00:56.40 | rasterix | its a pretty harsh place this channel when your trying to get to learn... |
| 00:58.35 | rasterix | qwell: why will they not replace extensions.conf (asides from the problems of legacy systems that use it) |
| 00:58.52 | Qwell | because not everybody is a C developer |
| 00:58.55 | *** join/#asterisk jeffspeff (i=jeff@c-98-240-113-135.hsd1.ky.comcast.net) |
| 00:59.09 | Qwell | for people who can use it - great. for those who can't, there's extensions.conf |
| 00:59.52 | [TK]D-Fender | rasterix: First you'd have to recode channel drivers in order to determine where to start processing calls. extensions is a rather flat DB concept. |
| 01:00.01 | RecycleBin | Hmm, a friend of mine is telling me i need a card anyway to plug into the voip device ... |
| 01:00.50 | [TK]D-Fender | rasterix: all channel start off on a given extension. Would this rather point to a procedure inside which you'd always do a "switch" or regex, etc to immediately determine why you're there in the first place? Then you'd have to rethink a bunch of other stuff to. |
| 01:01.11 | [TK]D-Fender | rasterix: AEL is nothing but a parser which "compiles" down to extensions.conf logic on load |
| 01:01.27 | [TK]D-Fender | RecycleBin: Yes, we call those "nics" and "switches" :) |
| 01:01.33 | rasterix | this is a very vague question.. but would it be fair to say 95% of systems use extensions.conf the rest ael? (percentages obviously not precise) |
| 01:01.52 | Qwell | rasterix: it would be fair to say that some non-zero percentage uses ael |
| 01:02.05 | beernuts | lol |
| 01:02.08 | RecycleBin | Yeah the nic card i knew. He was saying i still needed a FOX card |
| 01:02.14 | rasterix | qwell: i think that answers my question :) |
| 01:02.21 | Qwell | or even "zero or greater percentage" |
| 01:02.44 | rasterix | now im not sure |
| 01:02.50 | [TK]D-Fender | RecycleBin: I think you're meaning FXO. |
| 01:02.50 | rasterix | is that just sarcasm |
| 01:03.06 | [TK]D-Fender | RecycleBin: Describe what you want to do and I'll correct you on it. |
| 01:03.07 | rasterix | or are you saying ael is in use by a very small minority? |
| 01:03.18 | jblack | rasterix: how, exactly, would he know? |
| 01:03.24 | [TK]D-Fender | rasterix: lets just say 95/5 is as fair a guess as any. |
| 01:03.34 | rasterix | because he deals with support all the time |
| 01:03.39 | RecycleBin | I want to use a voip line to call out, and i was trying to avoid buying a FOX card |
| 01:03.53 | rasterix | jblack: so he would know if the questions relate to ael or extensions.conf |
| 01:04.00 | [TK]D-Fender | RecycleBin: FXO <- please learn the term. |
| 01:04.03 | [TK]D-Fender | ~fxofxs |
| 01:04.03 | jbot | somebody said fxofxs was An FXO port (red Digium module) expects to receive dialtone and receive ring voltage. You can connect it to a PSTN line from the telco. An FXS port (green Digium module) expects to provide dialtone and provide ring voltage. You can connect a phone or a fax to it. |
| 01:04.25 | rasterix | jblack: sarcasm is only effective when based upon a sound argument |
| 01:04.27 | [TK]D-Fender | RecycleBin: If you want to use an **ITSP** so far no card is needed. |
| 01:04.31 | [TK]D-Fender | ~itsp |
| 01:04.32 | jbot | [~itsp] An ITSP is an Internet telephony Service Provider or "VoIP Telephone company". The allow you to either PLACE calls to the PSTN (called Termination), RECEIVE calls from the PSTN (called Origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs |
| 01:04.33 | [TK]D-Fender | ^^^ |
| 01:04.36 | jblack | Who's being sarcastic? |
| 01:04.40 | RecycleBin | Ok |
| 01:04.43 | RecycleBin | Excellent |
| 01:04.56 | rasterix | not me... ever |
| 01:05.03 | [TK]D-Fender | RecycleBin: that handles THAT end of the call. Now what do you intend to TALK into? |
| 01:05.26 | jblack | anyways, taking it back to your original question, you're not stuck with the grammar in extensions.conf. There's another grammar. |
| 01:05.41 | rasterix | ok point taken |
| 01:05.52 | rasterix | but since its in use by a small minority |
| 01:05.56 | RecycleBin | I was planning to have asterisk Autodial a phone list and play a recorded MSG, Or if i want to use the phone a headset is fine |
| 01:06.09 | rasterix | there is no harm in making suggestions about improvements to extensions.conf |
| 01:06.20 | jblack | It doesn't matter how many people are using it. It matters whether or not it'll be supported for as long as you can use it. And it will be. |
| 01:06.31 | *** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.itb.ac.id) |
| 01:07.04 | jblack | the * team is big on adding options, not on taking them away. |
| 01:07.17 | [TK]D-Fender | RecycleBin: Oh yeah, just an auto-dialer advert system. Yes... you need... A PC <- And for accessing the ITSP, my guess : Internet access. Thats it |
| 01:07.40 | RecycleBin | Thankyou |
| 01:07.53 | [TK]D-Fender | RecycleBin: No cards as you get to the PSTN via your ITSP |
| 01:08.14 | RecycleBin | Big savings there |
| 01:08.17 | rasterix | jblack: i believe certain asterisk variables got deprecated in favour of functions... |
| 01:08.34 | [TK]D-Fender | rasterix: All sorts. |
| 01:08.36 | jblack | and? |
| 01:08.38 | florz | rasterix: once you find out how the extensions.conf really works, you will see that there are far worse problems with it and that the developers basically aren't interested in a usable language ... so making suggestions for improvements is basically pointless, unless you are willing to re-implement more or less all the dialplan handling |
| 01:09.43 | rasterix | jblack: so options have been taken away... so your argument was not correct |
| 01:10.28 | [TK]D-Fender | rasterix: What options have been "taken away"? |
| 01:10.46 | ManxPower | rasterix: extensions.com got a bunch of applications for AEL. |
| 01:10.59 | ManxPower | Before AEL there was never a While or Gosub apps |
| 01:11.14 | jblack | Forget that. He's one of those stuck-in-a-boolean life types. |
| 01:11.19 | [TK]D-Fender | ManxPower: Not really bundled... |
| 01:11.22 | ManxPower | In fact AEL has DRIVEN additions and updates to extensions.conf |
| 01:11.29 | [TK]D-Fender | jblack: !true |
| 01:11.47 | jblack | "Girls tend to have long hair. My mom has short hair, so you're WRONG!" |
| 01:11.48 | rasterix | fender: variables have been replaced with functions. i agree this is not actually taking away options but then neither is introducting a substr function to replace ${variable:5,3} |
| 01:12.01 | [TK]D-Fender | ManxPower: Little I can think of was required to be brought in for AEL to work as it does. All it would change is how it would parse out. |
| 01:12.03 | rasterix | jblack: your a child |
| 01:12.05 | jblack | rasterix: Correlations can range from -1, to 0, to 1. |
| 01:12.29 | seanbright | you're |
| 01:12.33 | seanbright | not 'your' |
| 01:12.40 | rasterix | lol |
| 01:12.42 | jblack | they like to add new features, they dislike removing features. Just because the syntax for CallerID has changed, doesn't mean they're going to wipe out every codec but g.729 |
| 01:12.52 | [TK]D-Fender | rasterix: "introducting a substr function to replace ${variable:5,3}" <-- and what is this specific example? What function replaces basic substring work? |
| 01:12.59 | ManxPower | rasterix: feel free to write one. heck you could even write it in extensions.conf macro of subroutine. |
| 01:13.29 | [TK]D-Fender | sends the Grammar Rangers out to get rasterix |
| 01:13.43 | rasterix | sigh |
| 01:14.22 | [TK]D-Fender | rasterix: Can you clarify your example I jsut asked about? |
| 01:14.32 | rasterix | fender: i can try |
| 01:14.42 | ManxPower | There are really two basic types of Asterisk users. There are the people like me (most of the time), Qwell, [TK]D-Fender and most of the other long time users. We accept Asterisk's oddities and life happy productive PBX lives. Then there are the people like rasterix and others that fight Asterisk's oddities. Those people end up living pointless miserable lives frequently involving drug overdoses and crack houses. |
| 01:15.03 | ManxPower | Which do you want to be? |
| 01:15.12 | florz | rasterix: the most important reason I would see for not adding a substr function would be that the (seemingly) cleanest implementation would open most uses to even worse security problems than most asterisk installs do have anyway ... |
| 01:15.14 | rasterix | fine i give up |
| 01:15.35 | [TK]D-Fender | rasterix: Before you do you could perhaps teach me about this new function I'm unaware of <- |
| 01:16.05 | ManxPower | rasterix: learn Perl, then ask again. |
| 01:16.25 | jblack | wrap it in a cgi. |
| 01:16.29 | jblack | pardon, an agi |
| 01:16.39 | rasterix | fender: i dont really understand your question... i was just suggesting ${variable:5,3} should be replaced by substr(${variable},5,3) |
| 01:17.07 | rasterix | manxpower: and perl is such a perfect language? |
| 01:17.23 | jblack | You can write agis in any language you like. |
| 01:17.32 | ManxPower | exten => 666,1,Macro(substr,${variable},5,3) |
| 01:17.34 | [TK]D-Fender | A riddle wrapped in an enigma, covered in whipped cream, chocolate sauce & a cherry on top!~ |
| 01:17.53 | rasterix | manxpower i understand that |
| 01:17.54 | ManxPower | rasterix: not at all. It's even worse than extensions.conf when it comes to all the stuff you don't like about it. |
| 01:18.01 | jblack | considers writing a blog entry entitled "Life is analog, not digital, you twit" |
| 01:18.52 | ManxPower | New Orleans is freaking out today even more than yesterday |
| 01:18.59 | jblack | Yeah. They're doomed. |
| 01:19.06 | rasterix | but my point is just that a variable should be an entity... not contain another variable... and the only reason i say this is because i was trying to write a lexical parser for extensions.conf and this caused problems... i dont care that much! |
| 01:19.20 | ManxPower | jblack: Hush you, I'm in New Orleans right now. |
| 01:19.29 | jblack | What? You brought it up! |
| 01:19.38 | ManxPower | rasterix: dude, Asterisk uses BISON as it's parser |
| 01:19.47 | [TK]D-Fender | rasterix: It certainly could. Now here's some background : the variable notation method predates the invention of functions period. So we'd be ripping out code we already have. Does this new function add enough extra value in any way? I can't see it YET... Now is the function method SHORTER? Nope. So while it is more in line with other common languages it doesn't actually ADD to our lives. |
| 01:20.43 | [TK]D-Fender | rasterix: Do you follow my thought process? YES it could be a function. Does it add any value? Little if any in my books, but YOU could always go create it yourself. Rather easily actually since you could use CUT as a code-base |
| 01:20.45 | jblack | it would in one way.. It would get him to go away... |
| 01:20.58 | [TK]D-Fender | jblack: He's not worht that kind of effort ;) |
| 01:21.11 | rasterix | good grief |
| 01:21.19 | [TK]D-Fender | lol |
| 01:21.21 | ManxPower | Ruh roh, Shaggy! It's Rasterisk! |
| 01:21.41 | [TK]D-Fender | Scooby-dooby-dooby-dooby-dooooooooooooo!!!!!!!!!!! |
| 01:21.49 | rasterix | so this is where the great minds that are asterisk get together.... |
| 01:22.00 | ManxPower | rasterix: no, that would be in Huntsville. |
| 01:22.10 | ManxPower | It's just the freaks that hang out here. |
| 01:22.39 | jblack | Someone should implement lisp in extensions.conf syntax. |
| 01:22.57 | ManxPower | jblack: someone did it for perl. google res_perl |
| 01:23.10 | [TK]D-Fender | rasterix: So far you're running around poking us with a stick and trying to tell us "how it is". We've been doing this for years. Way to ingratiate yourself there... |
| 01:23.43 | ManxPower | How many Asterisk PBXs do you have in production anyway? |
| 01:23.45 | [TK]D-Fender | jblack: lisp? Someone'll think thats just an audio issue and expect a DSP to correct it ;) |
| 01:24.02 | rasterix | fender: if making a suggestion about syntax is poking you with a stick then you need thicker skin |
| 01:24.10 | jblack | "My lisp speaks with a lisp! We need a new func_ or app_ !" |
| 01:24.10 | ManxPower | I want extenions.rpgII |
| 01:24.46 | ManxPower | rasterix: It's not broken. It may be odd, but it's not broken. Bison can handle tokenizing extensions.conf you should be able too. |
| 01:25.03 | ManxPower | Hmmm...res_cobol |
| 01:25.21 | jblack | I know. Someone has to emulate a C=64. |
| 01:25.24 | rasterix | i can and will handle it |
| 01:25.33 | ManxPower | I actually missed COBOL the other day. Had to generate reports from CDRs and Cobol would have been perfect for that. |
| 01:25.37 | [TK]D-Fender | rasterix: No, I wasn't restricting to that. Anyways hopefully you see that coding OSS is like herding cats and that these flaws we deal with work in concert with each other. Fix something and it'll break something else. real progress takes a lot of synchronized work. |
| 01:25.58 | [TK]D-Fender | rasterix: And while you are coding that the ground is moving out from underneath you as people "fix" other stuff. |
| 01:26.11 | rasterix | fender: point taken |
| 01:27.14 | [TK]D-Fender | rasterix: So gradb a drink, pull up to the bar and ask the barmaid why the univers won't just stop for you anymore... |
| 01:27.26 | [TK]D-Fender | *hic* |
| 01:27.29 | jer | ManxPower, just for clarification, bison is not a parser, it's a parser generator |
| 01:27.40 | jer | and it's not thread safe either |
| 01:27.42 | rasterix | but a substr function could be implemented... with the ${variable:3,4} to be deprecated sometime about version 5.4 |
| 01:27.47 | [TK]D-Fender | jer: Nothing about * is safe :) |
| 01:27.54 | jer | [TK]D-Fender, touché |
| 01:27.55 | jer | =] |
| 01:28.11 | jblack | just for clarification, touche', not touch |
| 01:28.24 | jer | jblack, that was an e with an accent... enable utf-8 |
| 01:28.25 | [TK]D-Fender | rasterix: sure, why not. Now the code has to be accepted into the tree and people will debate all the points I just made and more. |
| 01:28.34 | rasterix | or touchy... like most of the people in here |
| 01:28.40 | [TK]D-Fender | rasterix: that is reinventing the wheel and not even ending up and rounder. |
| 01:29.14 | ManxPower | rasterix: Feel free to submit code. Heck, even I have like 4 lines of code in the asterisk code. Smaller patches are more often accepted. |
| 01:29.33 | ManxPower | INVAL_EXTEN |
| 01:29.53 | [TK]D-Fender | rasterix: and as I stated doign substr as a function is so easy *I* could do it and I don't even know C |
| 01:30.09 | jblack | I've been thinking about patching i to work with Goto. Any thoughts? |
| 01:30.17 | [TK]D-Fender | rasterix: CUT is 99% of the work done already. |
| 01:30.31 | [TK]D-Fender | jblack: explain |
| 01:30.41 | [TK]D-Fender | jblack: Failed goto's go to "i"? |
| 01:30.41 | rasterix | fender: i guess the same arguments could hold... heck lets all go back to programming in basic with line numbers |
| 01:30.52 | jblack | [TK]D-Fender: In the destination context, yeah. |
| 01:31.01 | [TK]D-Fender | rasterix: We already are, jsut in * they are called "Priorities" :) |
| 01:31.06 | jer | rasterix, i refuse! i will stick to intercal thank you very much! |
| 01:31.12 | jblack | Goto(context,doesntexist,1) would go to context,i,1 |
| 01:31.13 | ManxPower | jblack: I think in the calling context would be better, what if your destination context is not valid. |
| 01:31.15 | rasterix | fender: my point exactly |
| 01:31.33 | [TK]D-Fender | jblack: sounds easy enough. |
| 01:31.42 | jblack | I didn't consider missing dest contexts. That's a good point. |
| 01:31.43 | *** join/#asterisk LeddyHM (i=leddy@da-club.with.my.beerandcondoms.com) |
| 01:31.50 | [TK]D-Fender | rasterix: See there are advantage to this approach as well.. |
| 01:32.00 | [TK]D-Fender | rasterix: For instance DB support for dialplan <- |
| 01:32.00 | ManxPower | missing context, extension or priority (or priority label) |
| 01:32.22 | jblack | Yeah. semantically, I think your suggestion is cleaner. |
| 01:33.08 | [TK]D-Fender | jblack: bad combo = "i"i in target context. Failing that "i" in CURRENT context, failing that next priority |
| 01:33.35 | [TK]D-Fender | jblack: 3-part. |
| 01:34.09 | [TK]D-Fender | jblack: personally there are cases I would want th "i" OPTIONAL. |
| 01:34.34 | rasterix | well i have to go sleep its beein interesting... go to #asterisk... make a suggestion and end up the channel's no 1 villain |
| 01:34.54 | jblack | you're not even in today's top 3. |
| 01:34.54 | ManxPower | rasterix: only for this week. Next week someone else will get their turn. |
| 01:34.56 | *** join/#asterisk russellb (n=russellb@asterisk/developer-and-stable-maintainer/drumkilla) |
| 01:34.56 | *** mode/#asterisk [+o russellb] by ChanServ |
| 01:35.10 | jeev | RUSSELLLLLLLLLLLLL |
| 01:35.15 | ManxPower | Now that russellb, he's a real rabble rouser. |
| 01:35.20 | ManxPower | oops! |
| 01:35.25 | jeev | russell is a bussel |
| 01:35.30 | [TK]D-Fender | rasterix: Don't think so highly of yourself, there are plenty of people I'd axe-muder LONG befor getting to you ;) |
| 01:35.31 | jblack | [TK]D-Fender: Hmmmmm. Might confuse new people, no? |
| 01:35.50 | [TK]D-Fender | jblack: this deserves a little extra thought. |
| 01:36.05 | jblack | Yeah. |
| 01:36.05 | russellb | o.O |
| 01:36.07 | russellb | you peoplez are teh silly |
| 01:36.15 | jblack | O RLY? |
| 01:37.05 | RecycleBin | I hate to asking agian ive been asking stuff all day but.. if i had 2 or more voip lines, is it a nic for each line ? |
| 01:37.21 | jblack | Nope. |
| 01:37.25 | RecycleBin | Cool, |
| 01:37.28 | jblack | what protocol, sip? |
| 01:37.30 | Strom_C | if you go to two or more websites, do you need a nic for each website? |
| 01:37.30 | [TK]D-Fender | RecycleBin: No |
| 01:37.41 | [TK]D-Fender | RecycleBin: don't ever think of them as "lines" |
| 01:37.49 | rasterix | recyclebin: yep if u have 100 voip lines u buy a pc with 100 pci slots :) |
| 01:38.08 | [TK]D-Fender | RecycleBin: Ever call is jsut another channel. How many your provide will accept depends on the service you buy from them. |
| 01:38.11 | RecycleBin | Hahaha, well when you put it that rasterix it becomes obvious |
| 01:38.20 | RecycleBin | Ty |
| 01:39.31 | jblack | is almost bored enough to watch disney pr0n |
| 01:39.45 | jeev | jblack, did you see joe biden's hot ass daughter? |
| 01:39.52 | jblack | nah. Didn't watch. |
| 01:39.56 | rasterix | < hides snow white from jblack |
| 01:40.04 | jeev | http://www.youtube.com/watch?v=ZX4mwaDlBmg 8 minutes, all the way to the right, WOW |
| 01:41.03 | jeev | http://news.yahoo.com/nphotos/slideshow/photo//080828/ids_photos_ts/r2015802311.jpg/ |
| 01:41.04 | jeev | i think that's her |
| 01:41.51 | *** join/#asterisk hatoon (n=musis@189.71.102.56) |
| 01:42.34 | jblack | heh. images.google.com, search for "ugliest girl. ever" |
| 01:42.36 | hatoon | boa noite a todos ! |
| 01:42.52 | *** join/#asterisk Levonk (n=lk@adsl-76-238-248-27.dsl.lsan03.sbcglobal.net) |
| 01:43.05 | jblack | "fugly" is even better |
| 01:44.02 | jblack | omg omg omg http://humorvice.blogspot.com/2007/12/nsfw-fugly-pictures-1.html |
| 01:45.02 | jeev | heh |
| 01:45.07 | jeev | images.google.com hottest girl. ever |
| 01:45.28 | jblack | you gotta check out that humorvice site. I can't breathe |
| 01:47.31 | jblack | [TK]D-Fender: I found a picture of r0land: http://bp2.blogger.com/_GGAmzDRA_BY/R2rCjnq0GRI/AAAAAAAAAic/4nAK-0GVU08/s1600-h/humor_vice_impossible465.jpg |
| 01:50.07 | [TK]D-Fender | jblack: I've officially dropped that case. |
| 01:50.31 | jblack | I beatcha by a 1/4 second |
| 01:51.10 | [TK]D-Fender | jblack: that you did... |
| 01:51.43 | jblack | Ohhh, look! I found the new new Orleans housing! http://bp0.blogger.com/_GGAmzDRA_BY/R2rAxHq0GJI/AAAAAAAAAhc/NIL2Ax9WTTE/s1600-h/humor_vice_redneck4479.jpg |
| 01:52.16 | jblack | jeev: <whistle> |
| 01:53.59 | RecycleBin | Sweet, i can set my own caller ID |
| 01:54.05 | RecycleBin | Nice! |
| 01:54.34 | *** join/#asterisk russellb (n=russellb@asterisk/developer-and-stable-maintainer/drumkilla) |
| 01:54.34 | *** mode/#asterisk [+o russellb] by ChanServ |
| 01:56.39 | jeev | lol |
| 02:15.51 | jblack | I'm in the conf if anyone wants to chat. |
| 02:17.12 | *** join/#asterisk korihor (n=korihor@201.211.168.130) |
| 02:22.09 | *** join/#asterisk Sargun (n=xbmodder@atarack/staff/sargun) |
| 02:23.06 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 02:25.40 | Sargun | YAY |
| 02:25.42 | Sargun | fuck. |
| 02:26.13 | jblack | Sargun: Your emotional control is lacking. :) |
| 02:27.48 | russellb | perfect for IRC |
| 02:28.03 | jblack | Life is great! But it sucks! |
| 02:29.28 | Sargun | hehe |
| 02:29.30 | *** join/#asterisk mosty (n=mosty@ppp191-34.static.internode.on.net) |
| 02:31.16 | *** join/#asterisk aliver (n=aliver@c-67-190-161-26.hsd1.co.comcast.net) |
| 02:31.45 | mchou | In the United States, does the callerID include the '1' before the area code? or is 10 digits enough? |
| 02:31.57 | aliver | Once I'm inside the "i" extension, what variable holds the invalid extension they just tried? |
| 02:31.59 | jblack | I ain't got nobody. I've got some money cause I just got paid. How I wish I had someone to talk to. |
| 02:32.03 | jblack | i'm in an awful. |
| 02:32.06 | jblack | way, that is. |
| 02:32.19 | aliver | that's a horrible song |
| 02:32.21 | jblack | mchou: I believe 10 digits is correct. |
| 02:32.55 | aliver | Isn't it that "drove the chevy to the levy" song? |
| 02:33.18 | jblack | That's don mclean. i'm quoting jimmy a buffet song. |
| 02:33.36 | aliver | Oh, well, shows you what I know about that era. |
| 02:33.55 | aliver | I guess I have heard that jimmy buffet song come to think of it. |
| 02:34.08 | jblack | http://www.actionext.com/names_j/jimmy_buffett_lyrics/another_saturday_night.html |
| 02:34.55 | aliver | http://www.actionext.com/names_a/amon_amarth_lyrics/friends_of_the_suncross.html |
| 02:35.01 | jblack | I should make that my hold music. |
| 02:35.14 | jblack | Then the next time my ex calls, leave her on hold for 12 hours |
| 02:35.35 | aliver | It's not like I can't get any jimmy buffett song lyrics out of my brain due to infinite exposure in every resturant across the damn country. |
| 02:35.36 | Qwell | jblack: then call her back with MoH |
| 02:36.11 | jblack | Oh, if I were gonna prank her, I'd do better than that. |
| 02:36.16 | aliver | Background("The-person-you-are-trying-to-reach-hates-your-guts") && Hangup() |
| 02:36.48 | aliver | Once I'm inside the "i" extension, what variable holds the invalid extension they just tried? Anyone know off hand? |
| 02:36.59 | jblack | She called me today. "My windows computer is all screwed up. Install linux on it for me" |
| 02:37.02 | aliver | wouldn't it just be the ${EXTEN} var? |
| 02:37.15 | *** join/#asterisk outtolunc (n=me@c-67-170-211-86.hsd1.ca.comcast.net) |
| 02:37.17 | Qwell | I imagine ${EXTEN} would be i at that point |
| 02:37.18 | aliver | jblack explain to her the meaning of "Ex" |
| 02:37.29 | jblack | That's why I almost did it |
| 02:37.57 | aliver | Qwell is there any way to get the thing they dialed to get thrown into "i" ? |
| 02:38.32 | Qwell | probably the most common way to get there would be after a WaitExten |
| 02:38.37 | jblack | aliver: Try a higher level priority backup goto |
| 02:38.38 | aliver | Hmm, maybe I'm doing this in one of my typical harebrained ways. I'll try using a real pattern besides relying on 'I" |
| 02:38.47 | Qwell | yeah, never rely on i |
| 02:38.48 | jblack | Yeah, goto doesn't use i. |
| 02:38.56 | outtolunc | looks at the 5 out of 10 caps that are blown on my primary desktop video card |
| 02:39.09 | jblack | We were just talking about that earlier in fact, on what the semantics should be on i w/ goto |
| 02:39.20 | aliver | heh, figures I was doing something igmonic |
| 02:39.28 | jblack | igmonic? |
| 02:39.35 | aliver | I just made it up |
| 02:39.42 | aliver | ignorant and moronic |
| 02:39.49 | jblack | Nah. That's reasonable. |
| 02:40.14 | aliver | Ah, well that's good then ;-) |
| 02:40.21 | jblack | Unreasonable is "${EXTEN:3:5} is insane. You need to change it to substr()!!!" |
| 02:41.08 | aliver | I hate that stupid syntax. Thank god for substr |
| 02:41.19 | aliver | much more readable |
| 02:41.19 | [TK]D-Fender | jblack: ruh-roh! |
| 02:41.32 | jblack | suspects he just fell into a bear trap |
| 02:42.17 | jblack | I wonder if calling a papa johns with it's own phone number for CID would work |
| 02:42.46 | Andre101 | Hello people.. When configuring a sip client that is behind NAT, but the server is on a public IP, do i need to use the NAT=yes statement? |
| 02:42.58 | jblack | Andre101: Yes. You'll need to do more stuff too |
| 02:43.01 | jblack | ~sipnat |
| 02:43.01 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 02:43.07 | [TK]D-Fender | Andre101: for the peer yes, for [general] no |
| 02:43.09 | jblack | Did that change? |
| 02:43.16 | *** part/#asterisk ManxPower (n=manxpowe@adsl-156-119-179.msy.bellsouth.net) |
| 02:43.49 | Andre101 | Thanks :) |
| 02:44.39 | jblack | considers his asterisk+festival+phone sex project some more |
| 02:44.50 | aliver | has anyone tried that free G729 module, is that some kind of piracy thing? |
| 02:44.58 | jblack | it's some kind of piracy thing. |
| 02:45.18 | aliver | I take it that it takes bread from the mouths of digium programmers? |
| 02:45.28 | jblack | as far as I know. i haven't heard of any legal way to use g729 without paying for the patent. |
| 02:45.39 | aliver | Sigh. Okay, I'll just order it. |
| 02:45.47 | jblack | can't you do without it? |
| 02:45.53 | jblack | surely you can do ulaw or gsm? |
| 02:46.21 | mosty | you can use patents for testing without a licence, i believe |
| 02:46.33 | mosty | but don't quote me on that |
| 02:46.48 | jblack | I think patents apply to the sale of a thing only? |
| 02:47.12 | Qwell | no |
| 02:47.14 | jblack | checks |
| 02:47.24 | aliver | jblack my call audio quality sucks rocks and the damn upstream provider only uses G711 or G729 |
| 02:47.28 | jblack | yeah, it's use based, not production based. |
| 02:47.30 | aliver | so, it's a tough situation. |
| 02:47.49 | jblack | G711U is ulaw, and that's as good as it gets for quality. |
| 02:47.58 | Qwell | no it's not |
| 02:48.11 | aliver | not if you can't get the network folks to QoS your UDP RTP traffic up above HTTP. |
| 02:48.21 | aliver | then it just sucks. |
| 02:48.27 | Qwell | G.722 <3 |
| 02:48.29 | aliver | because the jitter correction kills me. |
| 02:48.32 | jblack | I don't see how any lossy codec can outdo a nonlossy one? |
| 02:48.33 | aliver | or lack thereof |
| 02:48.41 | Qwell | != 8KHz |
| 02:48.42 | aliver | lower bandwdith, that's how. |
| 02:48.51 | jblack | Fair enough. |
| 02:49.06 | aliver | If I was dealing with a sane situation you'd be right, though. |
| 02:49.08 | *** join/#asterisk ManxPower (n=manxpowe@adsl-156-119-179.msy.bellsouth.net) |
| 02:49.12 | *** join/#asterisk jameswf-home (n=james@ip72-223-0-183.ph.ph.cox.net) |
| 02:49.16 | jblack | What I'm reading, g.722 ranges from 48 to 64. ulaw is 64 as well, so shouldn't they tie at best? |
| 02:49.18 | aliver | but it's "corporate" and "production" |
| 02:49.50 | Qwell | jblack: no |
| 02:49.59 | jblack | what am I missing? |
| 02:50.03 | Qwell | != 8KHz |
| 02:50.37 | Qwell | You're sampling twice as often, so you're losing less range |
| 02:50.40 | ManxPower | jblack: It stuffs a much higher dynamic range in 64 |
| 02:50.44 | mosty | ok, so there are apparently some situations where you can use a patent for testing without a license, but that is not always the case |
| 02:51.05 | ManxPower | mosty: I've been saying that for years |
| 02:51.25 | jblack | Ahh, ok. Gotcha. |
| 02:51.30 | jblack | thanks |
| 02:51.58 | ManxPower | It'll be cool when G722 is standard on phones. |
| 02:51.59 | Qwell | and ulaw/alaw sort of are lossy |
| 02:52.06 | florz | jblack/aliver: well, it's rather simple: do it somewhere where there are no laws enforcing (software) patent s... |
| 02:52.10 | Qwell | transcode back and forth between them a few times, and you'll see what I mean |
| 02:52.31 | jblack | sure, but any analog -> digital conversion is lossy. =) |
| 02:52.44 | ManxPower | I just wish I could pick the codec for my cell phone. I'd pay more for clear audio |
| 02:53.03 | ManxPower | jblack: he's talking about digital -> digital conversion |
| 02:53.03 | Qwell | ManxPower: would one be able to choose LPC10? |
| 02:53.05 | florz | Qwell: no, that's only if you do the transcoding incorrectly |
| 02:53.16 | ManxPower | Qwell: That's what my cell phone uses right now! |
| 02:53.28 | WimpMan | Modern gsm codecs would be nice, yes. |
| 02:53.41 | jblack | realizes he's way, way off base |
| 02:55.01 | jblack | wikipedia claims 722 patents have expired. |
| 02:55.05 | *** join/#asterisk kamanashisroy (n=kamanash@119.30.34.6) |
| 02:55.45 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 02:55.50 | florz | (which doesn't mean, G.711 wasn't lossy - it's lossy the same way linear PCM is, or at least close to it) |
| 03:02.25 | *** join/#asterisk Genji1981 (n=w@203.152.122.50) |
| 03:04.07 | Genji1981 | hello! okay, managed to getmy asterisk connecting to a voip companies sip peer. right.... now, trying to connect sjphone to asterisk. Problem. it says ACL error (permit/deny)... and ive got sip.conf and extensions.conf with the correct details. i think... |
| 03:04.16 | Genji1981 | so, what could be wrong? |
| 03:04.16 | *** join/#asterisk salzh (n=chatzill@58.247.193.245) |
| 03:04.22 | salzh | #join #freepbx |
| 03:04.51 | florz | Genji1981: permit/deny are the settings you want to look at |
| 03:05.03 | Genji1981 | okay.. where is permit/deny found? |
| 03:05.39 | florz | sip.conf |
| 03:06.21 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 03:07.06 | Genji1981 | no mention of permit in sip.conf... should there be? |
| 03:07.32 | *** join/#asterisk Levonk (n=lk@adsl-76-227-117-216.dsl.lsan03.sbcglobal.net) |
| 03:07.46 | mosty | Genji1981, you probably have put your details in to sjphone incorrectly, ie not in the correct SIP URI format (my guess anyway) |
| 03:08.10 | Genji1981 | hmm.. followed some asterisk sjphone instructions......hrm.. |
| 03:11.16 | Alton2 | core set verbose 10 |
| 03:11.20 | Alton2 | core set debug 10 |
| 03:11.25 | Alton2 | might help you catch more information. |
| 03:11.57 | mosty | Genji1981, perhaps you can pastebin the details you enter into sjphone? |
| 03:12.14 | Genji1981 | btw.. this is on failing to register, btw. |
| 03:12.15 | Andre101 | Another question guys, how do i display what asterisk is doing on the console (eg, call being placed).. I reinstalled (not using FreePBX anymore) and now asterisk doesn't say much on the console. |
| 03:13.24 | Genji1981 | Andre101: see above info from Alton2.. hes psychic. |
| 03:14.16 | ManxPower | Andre101: /etc/asterisk/logger.conf also "logger reload" and "asterisk -rvvv" and CLI> set verbose 3 |
| 03:14.47 | ManxPower | Andre101: FreePBX has incredible amounts of stuff spewed to the CLI, much more than most systems -- that's why we can't support it. |
| 03:14.51 | *** join/#asterisk CunningPike (n=arodgers@S010600095b33697f.vc.shawcable.net) |
| 03:17.23 | Alton2 | True about the logger.conf settings, that took me some time to find when I was a newbie. |
| 03:22.54 | *** join/#asterisk _-Jon-_ (n=jonmiron@CPE000d8861e8f7-CM00080d290642.cpe.net.cable.rogers.com) |
| 03:22.59 | _-Jon-_ | Hello all |
| 03:24.23 | Genji1981 | okay, haha... sip show users.. has infinite value. hrm.. seems, account was supposed to be the extension name. |
| 03:24.45 | _-Jon-_ | I'm wondering if it's possible to remotely set call forwarding. From home I can do *72 then dial the number I want forwarded to, but I will need to activate it while out of the country next month. I tried using a softphone on my computer but *72 didn't work on it |
| 03:25.26 | WimpMan | _-Jon-_: disa? |
| 03:26.12 | _-Jon-_ | WimpMan, can I use disa and pretend to be from the extension I want forwarded? |
| 03:26.40 | Genji1981 | okay, now ive got both voip company and softphone connected to asterisk.. and i can type my extension and get it to ring... how do i dial out? |
| 03:26.57 | WimpMan | You could do the authentication yourself and set the callerid accordingly. |
| 03:27.26 | WimpMan | Actually. there was an extra option wehn using a password file. Was that even the callerid? |
| 03:28.03 | _-Jon-_ | Hmmm, I'm not sure actually. I've only ever used disa to call in and then dial out as a context |
| 03:29.02 | WimpMan | I don't like the fact that you can't seem to place a second call. |
| 03:29.14 | WimpMan | So maybe it's better to DIY. |
| 03:42.29 | *** join/#asterisk Bonix (n=Bonix@196-lo1.rt2.isimples.com.br) |
| 03:47.34 | mchou | wow, midnite bike ride |
| 03:47.45 | mchou | cool beans |
| 03:47.46 | *** join/#asterisk codefreeze-lap (n=murf@216.166.159.235) |
| 03:56.26 | Genji1981 | anyone familiar with world xchange new zealand? |
| 03:56.35 | Genji1981 | is this correct? |
| 03:56.43 | Genji1981 | exten => _X!,1,Dial(SIP/${EXTEN}@as.wxcnz.net,30,r) |
| 03:57.19 | Genji1981 | keep getting 604 errors when dialing out... ya.. im almost there. |
| 04:03.10 | _-Jon-_ | Are the * codes a feature of Asterisk, or do both of my IP phones happen to have a feature built into them that I wasn't aware of |
| 04:07.04 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 04:07.06 | mosty | jon: which * codes? asterisk can be setup to handle * codes, some phones also use star codes internally |
| 04:09.12 | _-Jon-_ | *72 and *73 for example for call forwarding, and there are 2 others I can't remember do enable and disable DND on an extension. It's weird how my IP phones understand these but 2 softphones I've tried won't work |
| 04:10.36 | mosty | those are most probably a feature of those specific phones |
| 04:11.49 | _-Jon-_ | I had thought that too, but I've even gone as far as unplugging the handsets and tried calling and it still forwards the call. Asterisk even makes note of the forwarding call, as if I'm dialing it from that extension |
| 04:12.07 | _-Jon-_ | It's bizarre. |
| 04:15.07 | mosty | what kind of phone is this? |
| 04:15.20 | mosty | what are you unplugging the handset from? |
| 04:16.11 | _-Jon-_ | This is a Sipura ATA, and a Grandstream IP phone (older one). I unplugged the power cored from both |
| 04:17.07 | _-Jon-_ | *cord rather |
| 04:17.57 | mosty | so the ata is still plugged into asterisk but the analogue phone is not? |
| 04:19.08 | _-Jon-_ | No, I've unplugged power from the ATA itself, and also tried unplugging the power on the IP phone to see as well |
| 04:19.16 | Genji1981 | yay! i can ring out! |
| 04:19.22 | _-Jon-_ | Clearly, asterisk has some built in functionality |
| 04:19.28 | Genji1981 | but... calls coming in, don't reach me. |
| 04:19.33 | *** join/#asterisk Levonk (n=lk@75.62.136.60) |
| 04:28.37 | mosty | jon: is this a custom asterisk setup, or a gui job? |
| 04:28.53 | Genji1981 | my voip sip is 'registered' .... is that all i need, in order for asterisk to receive calls? |
| 04:29.40 | *** join/#asterisk luckyaba (n=lucky@ip68-6-98-146.sb.sd.cox.net) |
| 04:30.00 | mosty | Genji1981, have you read the book? |
| 04:30.03 | mosty | ~thebook |
| 04:30.03 | 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 |
| 04:30.21 | *** join/#asterisk hijacked (n=argh@c-71-205-135-177.hsd1.mi.comcast.net) |
| 04:32.15 | *** join/#asterisk codefreeze-lap (n=murf@216.166.159.235) |
| 04:36.30 | [TK]D-Fender | Genji1981: You usually need a peer entry in your sip.conf configured to auth their call. |
| 04:37.49 | *** join/#asterisk jeffspeff (i=jeff@c-98-240-113-135.hsd1.ky.comcast.net) |
| 04:41.06 | Genji1981 | [TK]D-Fender: you mean this? http://www.geekzone.co.nz/forums.asp?ForumId=65&TopicId=17174 |
| 04:41.20 | Genji1981 | ive set it up exactly like that. |
| 04:41.34 | Genji1981 | with my password and userid etc in the proper place. |
| 04:41.41 | [TK]D-Fender | Genji1981: that is a sample, yes |
| 04:41.47 | Genji1981 | and sip show registry, shows im registered. |
| 04:41.55 | Genji1981 | yet... no calls come through. |
| 04:42.50 | [TK]D-Fender | gengDescribe how your * server is connected to the internet. |
| 04:43.10 | *** join/#asterisk mitanef (n=mitanef@189.132.228.88) |
| 04:44.23 | Genji1981 | natted. |
| 04:44.53 | [TK]D-Fender | ~sipnat |
| 04:44.54 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 04:44.55 | [TK]D-Fender | ^^^^ |
| 04:44.58 | [TK]D-Fender | read up |
| 04:45.04 | jeev | stabs fender in the forehead |
| 04:45.14 | [TK]D-Fender | ~cluebat jeev |
| 04:45.15 | jbot | ACTION pulls out a ClueBat (tm) and thwaps jeev. |
| 04:45.20 | [TK]D-Fender | ClueBat NEVER misses! |
| 04:45.25 | jeev | was wearing a clubat protective condom. |
| 04:45.46 | jeev | you're unable to order any hits from jbot anyway, since you've been stabbed in the face. |
| 04:45.51 | jeev | admit it fender, you've lost. |
| 04:45.53 | jblack | Condoms break. Trust me. |
| 04:46.06 | jeev | when i lost my virginity, it broke |
| 04:47.23 | jeev | i'm gonna partner with my friend since he doesn't realy have that much money but he's an expert at building computers and repairing.. like me.. and he could stay there |
| 04:47.24 | jeev | minimal costs. |
| 04:47.34 | jeev | the location is questionable.. i need to figure out marketing tactics. |
| 04:49.44 | Genji1981 | so, registering isn't enough.. i have to open a few ports for the voip company to ring on? |
| 04:50.37 | jblack | yup. for sip, the sip port, and rtp |
| 04:54.13 | *** part/#asterisk mitanef (n=mitanef@189.132.228.88) |
| 05:03.15 | drmessano | yep |
| 05:03.58 | *** join/#asterisk Levonk (n=lk@adsl-75-62-140-186.dsl.lsan03.sbcglobal.net) |
| 05:06.37 | Andre101 | ~book |
| 05:06.38 | 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 |
| 05:08.19 | *** join/#asterisk dmz (n=dmz@64.203.203.232.dyn-cm-pool-64.hargray.net) |
| 05:12.08 | drmessano | ~koob |
| 05:12.08 | jbot | koobyub~ ees ro ,moc.nesdamfiel.toft//:ptth ta LMTH --- fdp.0840156950879/skoob/moc.ylliero.sdaolnwod//:ptth FDP elbadaolnwod eerF --- /0840156950879/golatac/moc.ylliero.www//:ptth ta sruoy redrO --- )9-84015-695-0 NBSI( noitidE dn2 ynohpeleT fo erutuF ehT :ksiretsA |
| 05:12.23 | drmessano | Sweet, it worked |
| 05:12.57 | jaytee | lol |
| 05:13.16 | drmessano | ?KOOB eht daer uoy evaH |
| 05:13.33 | jaytee | you're incorrigible |
| 05:13.35 | drmessano | ^^^^^^ KOOB |
| 05:14.03 | drmessano | Koob is my new troll |
| 05:16.37 | kaldemar | ~koobeht |
| 05:17.00 | kaldemar | depressing. |
| 05:17.51 | oilinki | good morning |
| 05:21.47 | *** join/#asterisk steliosk (n=Stelios@91.140.115.22) |
| 05:26.02 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-014ddb6aadcad014) |
| 05:33.52 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 05:36.01 | *** join/#asterisk Levonk (n=lk@75.62.140.99) |
| 05:37.08 | *** join/#asterisk imcdona (i=imcdona@c-24-19-84-150.hsd1.wa.comcast.net) |
| 05:41.41 | *** join/#asterisk jblack (n=jblack@pool-71-181-244-67.sctnpa.east.verizon.net) |
| 05:42.00 | jblack | Wow. Ten minute without internet is very uncomfortable |
| 05:44.26 | *** join/#asterisk _gm (n=gmustafa@202.133.78.60) |
| 05:44.40 | Nugget | jblack: http://xkcd.com/466/ |
| 05:44.43 | kaldemar | try a few weeks sometime. |
| 05:45.01 | jblack | No way. I'd die |
| 05:45.38 | jblack | Last winter, I went without power for 3 hours. |
| 05:45.51 | jblack | I had to go shelter at a friend's house. |
| 05:46.04 | kaldemar | doubt that. unless you use internet to take care of your hydration. *g* |
| 05:46.29 | oilinki | jblack: in which country? |
| 05:46.36 | jblack | The US. |
| 05:47.45 | oilinki | jblack: ok. here in thailand the 3 hour power cut would cause a bit of sweat, but nothing else :) |
| 05:47.54 | jblack | it was a disaster. I had to leave all my stuff behind. When I came back to survey the damage, the servers were off, the clocks were all wrong. Not even the tivo had survived unscathed. |
| 05:47.59 | drmessano | I remember losing power for a day |
| 05:48.03 | drmessano | During the ice storm |
| 05:48.19 | drmessano | No shower.. horrible shit |
| 05:48.30 | *** join/#asterisk kamanashisroy (n=kamanash@119.30.34.6) |
| 05:48.36 | drmessano | It was 18 outside, 42 inside the house |
| 05:48.37 | jblack | oilinki: Oh, there was no fear of freezing. It was in the 40's (think about 4 or 5 C) |
| 05:49.00 | jblack | Lemme guess. No generator? |
| 05:49.14 | Juggie | drmessano, sucks |
| 05:49.22 | Juggie | i dont loose hot water during a shower |
| 05:49.25 | Juggie | its gas |
| 05:49.42 | drmessano | The next year, I was prepared |
| 05:49.57 | jblack | last winter, my heating system failed (it's basically a radiator system) for several hours, during a really nasty cold snap. |
| 05:50.01 | oilinki | jblack: yes. it pretty steady 30C day and night here. |
| 05:50.14 | jblack | But I had power, I had internet. Everything would be ok. |
| 05:50.22 | drmessano | I grabbed all my shit in my go bag, flew over to my parents house.. got all unpacked and expected to have to work all during the storm |
| 05:50.38 | drmessano | My phone beeps when my servers come back up at home.. Lost power for like 90 mins total lol |
| 05:51.06 | drmessano | Oh |
| 05:51.09 | jblack | Heh. I'm only good without mains for about 45 minutes |
| 05:51.11 | drmessano | I almost forgot |
| 05:51.14 | drmessano | I go home |
| 05:51.23 | jblack | yeah? |
| 05:51.23 | drmessano | Get my shit unpacked |
| 05:51.32 | drmessano | 45 mins later my parents lost power |
| 05:51.35 | jblack | lol. |
| 05:51.38 | drmessano | They were without it for a day and a half |
| 05:52.10 | jblack | I want to get solar and an array of batteries. |
| 05:52.32 | drmessano | Obama is going get us all clean nuclear power for our individual homes |
| 05:52.39 | drmessano | and free corndogs for life |
| 05:52.40 | jblack | Sure he is. |
| 05:52.46 | jblack | That fuckhead voted for FISA> |
| 05:52.50 | drmessano | DO NOT DOUBT THE OBAMA |
| 05:52.54 | oilinki | what is fisa? |
| 05:53.02 | jblack | I dare not. He would know. |
| 05:53.04 | drmessano | I was watching the DNC tonight.. |
| 05:53.07 | drmessano | and I was all like |
| 05:53.11 | drmessano | "OMG, he's not white" |
| 05:53.16 | oilinki | hihi.. http://thinkprogress.org/2008/08/28/mccain-iraq-peaceful/ |
| 05:53.19 | drmessano | Man I feel out of touch |
| 05:53.19 | jblack | oilinki: A law that allows the feds to wiretap americans without a warrant. |
| 05:53.35 | oilinki | jblack: ah. ok. |
| 05:53.36 | jblack | are you serious? You didn't know he wasn't white? |
| 05:54.01 | jblack | Well, just so you know, McCain is old. Really Old. Mid 70s old. |
| 05:54.45 | jblack | As in "was alive during world war II" old. |
| 05:54.51 | drmessano | Dude.. Barack Obama... I was like "I bet he's from Alabama" |
| 05:55.00 | drmessano | No way I saw him as being Black |
| 05:55.14 | jblack | Well, he's only half black. |
| 05:55.40 | drmessano | No, I am kidding |
| 05:55.47 | drmessano | I actually like the guy |
| 05:55.54 | jblack | I did too, until FISA |
| 05:55.54 | drmessano | I think he's a little inexperienced |
| 05:56.02 | jblack | Now I'm going to vote for John Stewart. |
| 05:56.34 | drmessano | I was shocked that John McCain picked Joe Biden for his running mate too.. crazy old fucker |
| 05:56.46 | jblack | uhhhh? |
| 05:56.49 | drmessano | Doesn't he read the newspaper |
| 05:56.52 | jblack | what did you just say? |
| 06:02.34 | *** join/#asterisk fogo (n=Paul@rs-69-169-132-200-0003.broadweave.net) |
| 06:03.13 | fogo | am I going to get bitten by anything running asterisk on a 64 bit box? |
| 06:07.39 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.static.bezeqint.net) |
| 06:07.47 | jblack | fogo: i haven't seen any problems at all. |
| 06:08.18 | jblack | wonders what Flight of the Living dead is like. |
| 06:08.45 | *** join/#asterisk obnauticus (n=obnautic@about/windows/regular/obnauticus) |
| 06:10.17 | *** join/#asterisk sucituanbo (n=free@c-24-21-121-148.hsd1.wa.comcast.net) |
| 06:12.55 | fogo | jblack: thanks.. just looking at rebuilding a box and thinking of going 64 bit - looks like I'm going to head that direction |
| 06:18.42 | *** join/#asterisk ElSonico (n=tav@gprs-prointernet-ff31c200-18.dhcp.inet.fi) |
| 06:31.54 | *** join/#asterisk mintos (n=mvaliyav@nat/redhat-in/x-7ced9c9a20e10bd8) |
| 06:45.30 | *** join/#asterisk nighty^ (n=nighty@host217-37-109-17.in-addr.btopenworld.com) |
| 06:54.58 | *** join/#asterisk gerphimum (n=trekkie@cpe-24-174-138-171.satx.res.rr.com) |
| 06:57.33 | *** join/#asterisk miloux (n=miloux@213.88.194.120) |
| 06:58.44 | oilinki | btw. which asterisk version is best for running realtime? |
| 06:59.19 | oilinki | I got the realtime working pretty well with 1.4. but I wonder if the 1.6 is better for it? |
| 07:04.35 | *** part/#asterisk oej (n=olle@ns.webway.se) |
| 07:13.00 | *** join/#asterisk Levonk (n=lk@75.62.131.32) |
| 07:17.05 | *** join/#asterisk mikkel (n=mikkel@84-238-113-66.u.parknet.dk) |
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| 07:33.30 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 07:34.24 | *** join/#asterisk tengulre (n=tengulre@125.71.208.16) |
| 07:35.33 | *** join/#asterisk Genji1981 (n=w@203.152.122.50) |
| 07:35.49 | *** join/#asterisk irisht (n=irisht@cpe-70-122-11-142.austin.res.rr.com) |
| 07:36.46 | Genji1981 | yay! im fully connected. okay..... how do i get a voicemail status indicator, to go from my voip companies sip proxy, to asterisk, so i can see if i need to ring the voip companies voicemail? (yes, i do turn off my computer.) |
| 07:37.12 | *** join/#asterisk gr0mit (n=tim@81.187.32.146) |
| 07:50.18 | *** join/#asterisk ToTo (n=ToTo@209.8.41.110) |
| 07:55.19 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 07:57.02 | *** join/#asterisk felipex (n=felipex@213-140-21-233.fastres.net) |
| 07:57.44 | felipex | hi at all |
| 07:58.33 | felipex | in asterisk 1.4 is it possible to have the sip/channel of the member that answered a call in a queue? |
| 08:00.42 | *** join/#asterisk clive- (n=pirch@dsl-243-81-38.telkomadsl.co.za) |
| 08:02.17 | clive- | Hi. Does anyone have any ideas why an incomming sip call defined as a type=user does not get directed to the defined context=fromUser1 ? |
| 08:02.52 | clive- | it only wants to go to the global context defined in the general section of SIP.conf |
| 08:06.37 | felipex | is there a var like ${CALLERIDNUM} for the member that answered a call in a queue ? |
| 08:10.36 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 08:11.00 | *** join/#asterisk EI5GTB (n=EI5GTB@193.120.192.61) |
| 08:12.07 | Genji1981 | hrm... okay. is there any method of telling a linksys pap2 ata, to forward the call via sip, to a asterisk server? |
| 08:16.07 | C4away | get rid of the PAP2 and just register the asterisk server to the provider directly, then register the pap2 to asterisk |
| 08:16.53 | Genji1981 | C4away: hrm.. nice, if my asterisk server was online 24/7. |
| 08:17.06 | Genji1981 | what about connecting to the Pap2 via a softphone? |
| 08:17.14 | C4away | hmm |
| 08:17.41 | C4away | I haven't worked on a PBX in my life that wasn't intended to run 24/7 |
| 08:17.42 | Genji1981 | chain sip-ping... hmmm. |
| 08:18.21 | C4away | I'll need more information to make any suggestions |
| 08:18.36 | C4away | I actually can't imagine what you are trying to do |
| 08:19.44 | *** join/#asterisk gr0mit (n=tim@dhcp3.zuk40.mot-tools.co.uk) |
| 08:20.15 | Genji1981 | well.. im a person who likes to use his computer, and not have to go away from it, when the phone rings. like playing with asterisk. of course, the woman of the house likes to have a 'real' phone, so ill have to plug a Pap2 into the net. Whick defeats the need for asterisk... and makes answering the phone at the computer impossible. |
| 08:21.20 | C4away | unless you run an asterisk server |
| 08:21.21 | Genji1981 | so.. id like a way to keep Pap2 in the loop, when my comps online.. and when its not, go straight to the voip companies sip. |
| 08:21.41 | *** join/#asterisk linuxstb (n=linuxstb@rockbox/developer/linuxstb) |
| 08:21.47 | C4away | just put it in the basement or something |
| 08:22.04 | C4away | asterisk will run on a 500Mhz with 256MB of ram |
| 08:22.23 | C4away | I'm sure you could find something like that at a thrift store, if you don't already have a number of them laying around |
| 08:22.28 | gr0mit | will run on a wrt54GS too |
| 08:22.40 | C4away | yea |
| 08:22.57 | C4away | will run on just about anything that linux will run on |
| 08:23.19 | *** join/#asterisk KermitTheFragger (n=KermitTh@118-197.bbned.dsl.internl.net) |
| 08:23.27 | felipex | gr0mit can you help me? |
| 08:23.38 | gr0mit | depends how, felipex! |
| 08:23.51 | felipex | gr0mit i have a problem with a queue |
| 08:23.56 | gr0mit | mkay |
| 08:24.13 | gr0mit | well, i am british, so we know about queues ;-) |
| 08:24.32 | felipex | exten => 500,3,Queue(prm-queue||||60|respallarm.agi) |
| 08:24.38 | felipex | i have this queue |
| 08:24.46 | gr0mit | ok |
| 08:25.15 | felipex | in the agi script i need the number/channel of the member who answered the call |
| 08:25.52 | felipex | i don't find how |
| 08:26.03 | gr0mit | never used agi scripts |
| 08:26.05 | gr0mit | sorry! |
| 08:26.46 | felipex | anybody can help me? |
| 08:27.41 | gr0mit | reads the book on this subkject |
| 08:28.10 | Andre101 | when doing a dialplan, will asterisk match the most specific match? or the first match? |
| 08:28.58 | WimpMan | most specific |
| 08:29.10 | Genji1981 | hmm.. okay.. .is there a fast way to switch the PAP2 from the voip company, to the asterisk box, and back again? |
| 08:29.17 | Genji1981 | or is it a long process each time? |
| 08:29.34 | WimpMan | Actually they're sorted, but as X is at the end, it happens to work :-) |
| 08:33.18 | Genji1981 | oh! another idea. what about a openwrt box with mini-asterisk on it, it can serve calls to both my asterisk server and the PAP2 at the same time yes? |
| 08:34.05 | WimpMan | Genji1981: yes |
| 08:34.51 | WimpMan | However I'd go for a simple soft phone on the PC. Havong an * just as client doesn't make much sense. |
| 08:35.11 | Genji1981 | okay, lets try this.... if no one answers the PAP2.. and my asterisk server isn't online, and there is no voicemail on the mini-asterisk.. what happens to the call? does it go to the voip companies voicemail? |
| 08:35.31 | *** join/#asterisk angryuser (n=aster@88.140.126.251) |
| 08:35.36 | Genji1981 | or die/ |
| 08:35.36 | angryuser | hello |
| 08:35.37 | Genji1981 | ? |
| 08:36.41 | WimpMan | VM unless you do something else before that happens. |
| 08:37.27 | angryuser | i have 2 queue installed, they are working fine, but if the agent is called from internal, it is not reported as 'in use' for queue, is it normal ? |
| 08:37.49 | angryuser | i mean agent are not reported in use |
| 08:39.24 | drmessano | The agent can still get queue calls |
| 08:39.43 | Genji1981 | k.. how would i tell mini-asterisk to send it back to the voip company, or would the voip company not detect an answer, from anything in the chain? ... even though the mini-asterisk received the call and rang on pap2, it wouldn't count as an answer? |
| 08:40.41 | drmessano | Genji1981: WTF are you trying to do? |
| 08:41.14 | *** join/#asterisk linuxstb_ (n=linuxstb@rockbox/developer/linuxstb) |
| 08:43.22 | Genji1981 | drmessano: to have softphones and pap2 ring, and if no one answers, have it go to the voip companies voicemail. |
| 08:43.36 | Genji1981 | not my non-existant one. |
| 08:43.54 | angryuser | drmessano, yea he is getting the calls, the thing is that sometimes when he is speaking from internal, the queue call his and get busy, which is normal ;) i am not sure how to manually report agent 'in use' for queue system |
| 08:44.17 | angryuser | call him* |
| 08:46.49 | *** join/#asterisk Levonk (n=lk@adsl-75-62-134-33.dsl.lsan03.sbcglobal.net) |
| 08:47.32 | drmessano | Genji1981: If the VoIP company has voicemail, don't enable voicemail in Asterisk |
| 08:47.33 | drmessano | Simple |
| 08:47.38 | WimpMan | Genji1981: If noone answers, noone answers. |
| 08:48.37 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 08:49.04 | angryuser | Is there any way to manupulate the state of agents in asterisk ? |
| 08:54.13 | Genji1981 | anyone familiar with asterisk for openwrt? |
| 08:56.17 | *** join/#asterisk MrNaz (n=naz@ppp118-208-169-32.lns10.mel4.internode.on.net) |
| 08:59.11 | *** join/#asterisk erwinpogz (n=meow@67.159.178.21) |
| 09:02.05 | drmessano | Genji1981: What is your question? Asterisk is Asterisk |
| 09:02.21 | erwinpogz | hi, can someone help me with my dialplan http://www.nomorepasting.com/getpaste.php?pasteid=19676 |
| 09:02.39 | erwinpogz | i cant get the extension 2000 and 3000 connect |
| 09:03.45 | WimpMan | Jepp. Look at the priorities. |
| 09:04.48 | *** join/#asterisk EI5GTB (n=EI5GTB@193.120.192.61) |
| 09:07.24 | *** join/#asterisk Maliuta (n=nikolai@kiev.lusan.id.au) |
| 09:07.36 | Genji1981 | drmessano: what does asterisk-mini have, that asterisk does not? |
| 09:15.47 | *** join/#asterisk Illarane (n=heifer@pdpc/supporter/student/Veratien) |
| 09:17.18 | Illarane | Hmm... One of my colleagues thinks he's found a fairly major bug in Asterisk 1.4.21. |
| 09:18.10 | Illarane | When a user agent makes a call and then hangs up before the dialled number picks up, Asterisk doesn't seem to kill the connection, and the target handset continues ringing. |
| 09:18.32 | Illarane | I'm hoping we've just managed to misconfigure it somehow. :) Anyone got any ideas what could be causing this? |
| 09:19.05 | Illarane | Apparently it didn't happen in 1.4.19.1. |
| 09:20.32 | C4away | Illarane: that is a bug with asterisk x.x.xx |
| 09:20.40 | C4away | has been the case as long as I have been using asterisk |
| 09:20.51 | *** join/#asterisk kamanashisroy (n=kamanash@119.30.34.5) |
| 09:21.22 | Illarane | C4away: Ah. |
| 09:21.30 | Illarane | Wonderful. :) |
| 09:21.34 | C4away | yes |
| 09:21.44 | C4away | look for a fix in asterisk 1.8 maybe |
| 09:22.09 | C4away | the good news is that all you have to do to get rid of them is restart asterisk |
| 09:22.19 | Illarane | That's.... Kinda major, though, isn't it? :) |
| 09:22.37 | C4away | only if you handle about 15,000 calls per day |
| 09:22.40 | C4away | like we do |
| 09:23.05 | Illarane | Hmm... from SIP to PSTN? |
| 09:23.14 | Illarane | has a look at the calls log. |
| 09:23.38 | C4away | we handle about 15,000 calls per day sip to pstn |
| 09:23.44 | C4away | via an ss7 gateway |
| 09:23.51 | C4away | so sip->sip->sip->T1 |
| 09:24.06 | C4away | the second SIP in that chain is where they get hung |
| 09:25.41 | Genji1981 | asterisk modules (.so files) are the same throughout any architecture? |
| 09:26.31 | Illarane | C4away: There's 184 concurrent calls at this exact moment in time. :p |
| 09:26.55 | Illarane | Not sure about the total statistics of calls made per day, but I'd guess it's in the region of Manylots. |
| 09:27.21 | erwinpogz | WimpMan, have you replied to me while ago? |
| 09:29.57 | C4away | manylots? |
| 09:29.58 | C4away | lol |
| 09:30.16 | C4away | Illarane: what time is it there? |
| 09:30.28 | C4away | we get 80-100 calls consistantly throughout the day |
| 09:30.44 | C4away | average 2 calls per second |
| 09:31.36 | C4away | but it tapers off pretty quickly after 5pm local time |
| 09:33.07 | *** join/#asterisk MrNaz (n=naz@ppp118-208-154-148.lns10.mel4.internode.on.net) |
| 09:34.52 | clive- | grr, the EXTEN variable seems to dissappear , any ideas? |
| 09:35.48 | *** join/#asterisk felipex (n=felipex@213-140-21-233.fastres.net) |
| 09:41.32 | Illarane | C4away: 10:41. |
| 09:41.44 | *** join/#asterisk benneton (n=DELL@89.111.209.66) |
| 09:41.53 | Illarane | We peak at about 11:00 and 16:00, IIRC. |
| 09:41.54 | *** part/#asterisk benneton (n=DELL@89.111.209.66) |
| 09:43.30 | C4away | ours peaks at about 9:00 am and then again at about 13:00 |
| 09:43.36 | C4away | first morning calls and after lunch |
| 09:43.47 | C4away | we have about 90% business customers |
| 09:46.40 | Illarane | Probably about the same here. |
| 09:46.50 | angryuser | clive-, what do you mean ? |
| 09:47.53 | NoxIn- | C4away: you say "the good news is that all you have to do to get rid of them is restart asterisk" <-- is it not possible to "soft hangup" the faulty calls ? |
| 09:48.55 | Illarane | I don't think we can detect it easily. |
| 09:49.27 | Illarane | Also, restarting Asterisk is definitely not an option. ;) |
| 09:49.43 | *** join/#asterisk s0lid (n=s0lid@60.52.253.84) |
| 09:49.47 | C4away | NoxIn- no |
| 09:49.50 | C4away | tried everything |
| 09:50.01 | C4away | <sarcasm> "the good news" </sarcasm> |
| 09:50.17 | C4away | that's my nice way of saying "this is total bullshit" |
| 09:50.25 | NoxIn- | Illarane: usually I kill calls which are in down state since a few hours |
| 09:50.28 | Illarane | Yes, I detected that bit. :) |
| 09:50.35 | Illarane | NoxIn-: Our maximum call time is 3 hours. |
| 09:52.35 | Illarane | C4away: Our Support Wench (she's aware of the nickname. :D) mooted that it might be a Snom and Grandstream problem, since they're not exactly world renowned for standards compliance. :) |
| 09:53.29 | C4away | uh |
| 09:53.38 | C4away | sip show history shows the last message as "BYE" |
| 09:53.43 | C4away | I'd say that's an asterisk issue |
| 09:53.47 | Illarane | Ah. |
| 09:53.54 | C4away | and the calls don't drop after the 3600 minutes with no RTP audio |
| 09:53.58 | C4away | they are hung |
| 09:54.02 | C4away | in all senses of the word |
| 09:54.15 | C4away | 3600 seconds rather |
| 09:59.35 | *** join/#asterisk [netman] (n=netman@20.Red-88-25-137.staticIP.rima-tde.net) |
| 10:01.43 | Illarane | C4away: http://rafb.net/p/AjWvsS39.html <--- a 'slightly' modified Asterisk log for a dead call. |
| 10:02.36 | Genji1981 | ah cool. a mini asterisk box with low power usage. can't do voicemail, ill let the voip company handle unanswered calls. but for simultanious rings on softphones and PAP2, it'll work well. |
| 10:05.20 | *** join/#asterisk ElSonico (n=tav@nat/ibm/x-38c58ee63aa207d1) |
| 10:06.37 | C4away | by the time "answer" is executed in line 7 the call has already hung up? |
| 10:09.39 | *** join/#asterisk oilinki3 (n=oil@ppp-124-120-13-199.revip2.asianet.co.th) |
| 10:14.17 | *** join/#asterisk grybelfix (n=ply2@dslb-092-073-219-085.pools.arcor-ip.net) |
| 10:20.13 | *** join/#asterisk oilinki7 (n=oil@ppp-124-121-245-251.revip2.asianet.co.th) |
| 10:21.59 | *** join/#asterisk Great_Anta_Baka (i=c4219f6b@gateway/web/ajax/mibbit.com/x-729b33823b0b888e) |
| 10:23.17 | Genji1981 | hmm... how exactly does the PAP2 provisioning url work? any method of retrieving the config file from that url on a normal pc? |
| 10:24.17 | *** join/#asterisk Levonk (n=lk@adsl-75-62-135-21.dsl.lsan03.sbcglobal.net) |
| 10:34.35 | *** join/#asterisk ElSonico (n=tav@nat/ibm/x-6eeb3a78652bed3f) |
| 10:40.10 | *** join/#asterisk unenough (n=unenough@CBL217-132-93-165.bb.netvision.net.il) |
| 10:42.19 | Illarane | C4away: The customer hangs up when the DEBUG error appears, and the PBX forwards the request on to our carrier. :p |
| 10:48.23 | C4away | yep |
| 10:48.27 | C4away | sounds like asterisk |
| 10:49.19 | *** join/#asterisk Levonk (n=lk@adsl-75-62-133-195.dsl.lsan03.sbcglobal.net) |
| 10:49.39 | Illarane | Oh well, I don't think it's affecting our live servers since we can't replicate the problem on 1.4.19.1, which is what we've got on them. |
| 10:49.50 | Illarane | Looks like an upgrade to 1.4.21 is out of the question. :) |
| 10:54.08 | *** join/#asterisk quaqo (n=quaqo@83-103-40-166.ip.fastwebnet.it) |
| 10:54.48 | C4away | the problem comes and goes |
| 10:54.56 | C4away | one version will have it, the next not |
| 10:55.04 | C4away | has been on and off since 1.2 when I started using asterisk |
| 10:59.58 | Illarane | Great. |
| 11:00.01 | *** join/#asterisk gones (n=gones@203.193.37.251) |
| 11:02.46 | *** join/#asterisk codefreeze-lap (n=murf@216.166.159.235) |
| 11:07.51 | linuxstb | What's the recommended method for interfacing Asterisk (1.4.21-2) with a website? I want to build a small site to enable users to be able to control aspects of the dialplan, plus also the ability to initiate calls via a web-based phone directory. My first attempt has been to use the AMI via the AsteriskManager.php library, but this appears to be extremely slow (about 5-6 seconds to make a connection to asterisk, get some information, a |
| 11:07.51 | linuxstb | nd then close the connection). |
| 11:09.41 | kaldemar | the library must be quite special then. those functions don't take 5-6 seconds. |
| 11:10.14 | linuxstb | That's what I'm guessing - the lib is broken... |
| 11:11.14 | linuxstb | I'm happy to use either perl or php for the site, so am open to suggestions for ways to use the AMI from those languages... |
| 11:11.15 | kaldemar | using AMI is really quite simple, it won't probably take much effort to DIY. |
| 11:12.17 | linuxstb | That's what I'm finding - most people seem to DIY, rather than there being "standard" libs. So if that's the case, I'll do the same. |
| 11:12.53 | kaldemar | i've done a <50 line perl cgi that shows me all sorts of information via AMI. it was easier to do it myself than start looking for a library. |
| 11:13.41 | linuxstb | Also, would an AMI proxy be a sensible idea? The docs I've read suggest that Asterisk itself doesn't handle multiple simultaneous connections well - is that applicable to 1.4.21 ? We only have about 10 users, so I doubt there will ever be 2 simultaneous connections to this website |
| 11:13.57 | grybelfix | linuxstb: i use the manager api here smoothly with php ($socket = fsockopen("127.0.0.1","5038", ....) |
| 11:14.26 | grybelfix | depends on what you want to control, anyway.. |
| 11:16.16 | linuxstb | grybelfix: You mean whether i use a proxy or not? |
| 11:17.54 | grybelfix | you said you want to interface asterisk with a website |
| 11:20.14 | linuxstb | Yes. |
| 11:20.47 | grybelfix | i have no idea what that php library does, but for me the manager api speed is like "instant results" |
| 11:22.39 | linuxstb | Yes, reading the php library, it seems to be using sockets wrongly, which is introducing delays... I qiote like the simple API it provides though, so will probably try and fix it. |
| 11:23.09 | grybelfix | hmm if a big context-menu is enough "control" for you, have a look at http://astqueueicon.sourceforge.net/ |
| 11:24.27 | *** join/#asterisk gones (n=gones@203.193.37.251) |
| 11:24.31 | grybelfix | another advance of that solution against "web based" is that realtime infos can be displayed in the system tray too.. |
| 11:25.31 | linuxstb | It's a Windows app though, and most of the users don't use Windows... (either OS X or Linux) |
| 11:26.37 | grybelfix | the app is a few lines of code anyway, you should be able to do this for linux yopurself ;) |
| 11:26.41 | linuxstb | And this is also a learning experience for me, as we want to integrate asterisk into other web-based systems we develop. |
| 11:27.25 | grybelfix | however the socket is used correctly by the php script since it runs very stable |
| 11:28.20 | linuxstb | Thanks for your help - now that I know that it _should_ work fine, I'll continue with what I'm doing. |
| 11:29.27 | grybelfix | good luck! with correcy usage you can php-use the manager api once a second all day long without problems. ;) |
| 11:29.40 | grybelfix | correcy=correct |
| 11:41.07 | *** join/#asterisk coppice (n=chatzill@177.162.17.210.dyn.pacific.net.hk) |
| 11:45.19 | *** join/#asterisk ToTo (n=ToTo@209.8.41.202) |
| 11:47.30 | *** join/#asterisk ElSonico (n=tav@nat/ibm/x-005e9c0b4c532b59) |
| 11:51.47 | *** join/#asterisk Levonk (n=lk@adsl-75-62-132-99.dsl.lsan03.sbcglobal.net) |
| 12:00.11 | *** join/#asterisk juniowww (n=juniowww@189.58.28.91.adsl.gvt.net.br) |
| 12:01.52 | *** join/#asterisk DagMoller (n=aguirre@unaffiliated/dagmoller) |
| 12:15.42 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 12:16.50 | *** join/#asterisk Gat0rvean (n=gredish@64.191.128.145) |
| 12:17.58 | *** join/#asterisk defswork (n=andy@mx2.3gcomms.co.uk) |
| 12:19.54 | felipex | if i have a diaplan like this |
| 12:20.13 | felipex | exten => 500,1,Answer |
| 12:20.18 | felipex | exten => 500,2,AGI(insertallarm.agi) |
| 12:20.24 | felipex | exten => 500,3,Queue(prm-queue||||60) |
| 12:20.29 | felipex | exten => 500,4,AGI(closeallarm.agi) |
| 12:20.59 | felipex | why after hangup at 3 it doesn't go to 4 |
| 12:21.37 | [TK]D-Fender | felipex: because that's what the "h" Asterisk Standard Extension is for. |
| 12:22.02 | felipex | so i have to change 4 with h |
| 12:26.30 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 12:27.19 | *** join/#asterisk linuxer_igor (n=igor@mlsrj200152096p202.mls.com.br) |
| 12:27.47 | linuxer_igor | hi. anybody use version 1.6 b9 ? |
| 12:28.10 | clive- | linux not that brave |
| 12:28.10 | mchou | ~book |
| 12:28.11 | 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 |
| 12:28.19 | linuxer_igor | hi, Anyware use version 1.6 b9 ? |
| 12:30.04 | [TK]D-Fender | linuxer_igor: Go and ask your actual question. |
| 12:33.19 | linuxer_igor | the command the AgentCallbackLogin is entire function |
| 12:33.47 | [TK]D-Fender | linuxer_igor: That app was deprecated in 1.4 It should no longer exist in 1.6 |
| 12:37.13 | linuxer_igor | ok but how to make queue to send the linking for agent X that he is in the branch yyyy? |
| 12:37.51 | linuxer_igor | ex: if I add Agent/008 on queue .. |
| 12:38.31 | linuxer_igor | I need to say where extension it is |
| 12:38.36 | [TK]D-Fender | linuxer_igor: "core show applications like queue" |
| 12:39.30 | [TK]D-Fender | linuxer_igor: Go read the instructions for each app. |
| 12:40.58 | linuxer_igor | I see... .. I know add agent in queue but before I need linking the agent to extension |
| 12:42.53 | [TK]D-Fender | linuxer_igor: Read the apps instructions, your answer is in there. |
| 12:42.53 | linuxer_igor | I used the AddQueueMember to add to one agent queue and later the AgentCallbackLogin command to say where extension this agent this |
| 12:43.08 | linuxer_igor | ok |
| 12:43.11 | linuxer_igor | thanks |
| 12:47.02 | *** join/#asterisk Nate187 (n=Nate187@gw.bigrivertel.net) |
| 12:47.20 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 12:47.20 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 12:47.27 | *** join/#asterisk penguinFunk (n=penguin@unaffiliated/penguinfunk) |
| 12:48.12 | *** join/#asterisk _zoomy_ (n=zoomy@dahlin.csbnet.se) |
| 12:49.04 | *** join/#asterisk awk (n=awk@security.web.za) |
| 12:49.05 | awk | f using wireless SIP phones is there any restrantes on the network as to how many |
| 12:49.08 | awk | <PROTECTED> |
| 12:49.11 | awk | err if |
| 12:49.29 | [TK]D-Fender | linuxer_igor: Its all in awk No more than any other WiFi device |
| 12:50.17 | [TK]D-Fender | awk: rather |
| 12:50.17 | awk | I am getting told contradicting stories.. I need to get 30 concurrent calls on 802 11b network |
| 12:50.17 | awk | possible? |
| 12:51.00 | *** join/#asterisk sCOTTo (n=scottnai@124-170-243-111.dyn.iinet.net.au) |
| 12:51.20 | sCOTTo | hey guys - any mac users in here? |
| 12:51.55 | lmadsen | awk: possibly not... if each end point is separate from each other... if a single end point (another asterisk box), should be possible |
| 12:52.11 | *** join/#asterisk fantasticmrfox (n=fantasti@212.57.232.254) |
| 12:52.14 | lmadsen | sCOTTo: you might just want to ask a question :) |
| 12:52.21 | fantasticmrfox | Does anyone know if Skype allows you to dial SIP URI's ? |
| 12:52.40 | awk | lmadsen but where is the strain? |
| 12:52.41 | sCOTTo | fantasticmrfox, gimme your sip and I will try it :) lol |
| 12:52.46 | awk | on the asterisk box or the network? |
| 12:53.11 | lmadsen | awk: you realize that every time you attach a separate device to a wireless network that you drop the bandwidth available by half right? |
| 12:53.24 | awk | lmadsen no? |
| 12:53.24 | postel | you do? |
| 12:53.37 | postel | damn.. all those years.. |
| 12:53.42 | fantasticmrfox | sCOTTo, :) Sent in PM :P |
| 12:55.08 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 12:56.03 | *** join/#asterisk linuxer_igor (n=igor@mlsrj200152096p202.mls.com.br) |
| 12:56.16 | linuxer_igor | hi. .. look this ... |
| 12:56.17 | linuxer_igor | The difference between the AgentLogin application and the AgentCallbackLogin application is that with the first one you have to keep the phone receiver open. If you hang up the phone, the agent will be logged off from the queue. With the second application, you are allowed to hang up the phone after the login process is finished. |
| 12:56.42 | linuxer_igor | I need to hang up the phone ! |
| 12:56.54 | linuxer_igor | before login |
| 12:58.38 | *** join/#asterisk moy (n=moy@CPE000d9d59afef-CM00140454d33e.cpe.net.cable.rogers.com) |
| 12:59.38 | *** part/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 13:00.26 | *** join/#asterisk l2trace99 (n=jr@static-71-251-65-16.tampfl.fios.verizon.net) |
| 13:02.49 | _zoomy_ | Hello everyone. I'm trying to follow the "Asterisk - The Future of Telephony"-book (awesome book by the way) and set up my first call between the server and an x-lite softphone on another machine behind a firewall |
| 13:02.59 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
| 13:03.53 | _zoomy_ | I have tried to port forwad correctly but I get this error in the asterisk CLI: [Aug 29 15:02:01] NOTICE[13177]: chan_sip.c:14262 handle_request_invite: Unable to create/find SIP channel for this INVITE |
| 13:04.49 | _zoomy_ | I am losing packages, right? I have played around alot but cant seem to fix it. Input anyone? |
| 13:05.12 | grybelfix | hmm |
| 13:05.50 | *** join/#asterisk Innocent_Devil (n=phonetal@203.99.184.239) |
| 13:05.59 | _zoomy_ | When the timeout has expired I get this message in asterisk CLI: [Aug 29 15:02:19] WARNING[13177]: chan_sip.c:1950 retrans_pkt: Maximum retries exceeded on transmission 38E52454-1606-7F00-F969-73E34547C962@193.11.244.90 for seqno 47218 (Critical Response) |
| 13:06.01 | Innocent_Devil | Hello all |
| 13:06.13 | Innocent_Devil | i m new to asterisk |
| 13:06.19 | Innocent_Devil | and i m having compilation error |
| 13:07.45 | grybelfix | is that a "well known" bug that bridging zoiper<-iax->asterisk<-sip->provider is broken? :( |
| 13:08.26 | _zoomy_ | I dont know, you think so? |
| 13:08.27 | grybelfix | i am soo disappointed. last time i tried this with asterisk 1.0.9 and an early version of idefisk, but the problem still persists... :-( |
| 13:08.33 | Innocent_Devil | anyone ? |
| 13:08.43 | *** join/#asterisk ajohnson (n=ajohnson@63.147.46.186) |
| 13:08.58 | grybelfix | Innocent_Devil: any details about it? |
| 13:09.45 | Innocent_Devil | grybelfix: http://www.pastebin.ca/1188476 |
| 13:11.12 | _zoomy_ | which ports do I have to forward to x-lite? I have tried 5060 and 8000 to 18000. |
| 13:11.54 | grybelfix | Innocent_Devil: sorry pastebin.ca is unreachable from here :( |
| 13:11.56 | *** join/#asterisk codestr0m (n=codestr0@unaffiliated/codestr0m) |
| 13:12.15 | _zoomy_ | x-lite is connected to asterisk but is "unmonitored" using "sip show peers", is this alright? |
| 13:13.41 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:14.17 | brodiem | _zoomy_: turn qualify on |
| 13:14.50 | [TK]D-Fender | _zoomy_: READ UP : |
| 13:14.52 | [TK]D-Fender | ~sipnat |
| 13:14.53 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 13:14.54 | [TK]D-Fender | ^^^^^^^^^^^ |
| 13:15.24 | brodiem | Innocent_Devil: install libstdc++ packages |
| 13:15.41 | _zoomy_ | sorry... |
| 13:16.58 | *** join/#asterisk mfdavid (n=user@189.100.103.252) |
| 13:17.27 | _zoomy_ | i have really tried searching around the net and have tried to follow "asterisk - the future of telephony" but i am lost so i just thought i could ask here... |
| 13:17.48 | oilinki7 | stupid question. what does mean gsm codec in asterisk? |
| 13:17.59 | [TK]D-Fender | _zoomy_: I just linked you to a speicif guide. Read it. |
| 13:18.14 | [TK]D-Fender | oilinki7: Its a codec. What more is there to know? |
| 13:18.19 | [TK]D-Fender | specific* |
| 13:18.31 | _zoomy_ | thanks alot! |
| 13:18.39 | Alton2 | There are different standards used to compress and decompress voice data. These standards are implemented as codecs. gsm is one of them. |
| 13:18.42 | oilinki7 | I suppose it's not the amr which is used with mobile phones |
| 13:19.02 | [TK]D-Fender | oilinki7: GSM-610 |
| 13:20.26 | oilinki7 | ok .thanks. how widely that is used? |
| 13:20.43 | oilinki7 | I'm currently using g729 and pretty happy for it. |
| 13:20.59 | Alton2 | g729 is probably the one to stay with if you can tolerate the licensing thing. |
| 13:21.18 | Alton2 | gsm is also popular but I can't say what is most prevalent. |
| 13:21.31 | oilinki7 | but then thinking if there would be even better solutions. g711a/u might not be good as I'm far away from everywhere. |
| 13:21.57 | oilinki7 | do you know which codecs the nokia voip-platform supports? |
| 13:22.01 | Alton2 | stick with g729 if you can |
| 13:22.09 | [TK]D-Fender | oilinki7: If you already have G.729, stick with it. Its lighter than GSM-610 |
| 13:22.11 | Alton2 | I don't know. Maybe someone else here does. |
| 13:22.23 | *** join/#asterisk Levonk (n=lk@adsl-75-62-136-214.dsl.lsan03.sbcglobal.net) |
| 13:22.26 | [TK]D-Fender | oilinki7: Debatebly better as well |
| 13:22.30 | mfdavid | hello. I cant make sip phone calls from the console. console dial 11123 where 11123 is the phone number I want to call. But I get an error saying No such extension '11123 ' in context 'default'. How/where I configure that?! The sip is configured, I can re |
| 13:22.54 | Alton2 | ok, off to work |
| 13:23.03 | oilinki7 | I understood that when I'm just passing the treffic, I do not have to worry about the license? |
| 13:23.09 | oilinki7 | for g729 |
| 13:23.28 | Alton2 | right |
| 13:23.35 | *** join/#asterisk anonymouz666 (n=anonymou@200.218.193.6) |
| 13:23.48 | oilinki7 | but then again. if I use the voicemail, should I use some other codec for it? |
| 13:24.22 | *** join/#asterisk juniowww (n=juniowww@189.58.28.91.adsl.gvt.net.br) |
| 13:24.49 | brodiem | oilinki7: I do know nokia's stack supports g729 |
| 13:25.47 | oilinki7 | brodiem: yes they do, but I was wondering if the gsm was somehow related to amr (which I understood is quite expensive). |
| 13:26.18 | oilinki7 | brodiem: and I was wondering of the old nokia network voip-server |
| 13:26.41 | brodiem | ah, k |
| 13:27.12 | [TK]D-Fender | mfdavid: You need to set up an extension in your dialplan to match what was dialed. |
| 13:27.24 | [TK]D-Fender | oilinki7: Only when * needs to transcode. |
| 13:27.46 | [TK]D-Fender | oilinki7: AKA prompts not natively available in G.729, MeetMet, etc. |
| 13:28.20 | oilinki7 | [TK]D-Fender: then. what is the best codec for pstn-voipserver-asterisk-voipserver-pstn codec? |
| 13:28.45 | [TK]D-Fender | oilinki7: G.711 clearly. |
| 13:29.08 | oilinki7 | I suppose there would be some transcoding in both of the voipservers if using g729 |
| 13:29.19 | oilinki7 | [TK]D-Fender: what if the bandwith is issue? |
| 13:30.10 | [TK]D-Fender | oilinki7: GSM-610 is decent, as is G.729. 729 is better if transcoding isn't an issue. |
| 13:30.17 | *** join/#asterisk tobias (n=tobias@user-0c2hj2f.cable.mindspring.com) |
| 13:30.37 | [TK]D-Fender | oilinki7: G.726 is they happen to support is a good compromise |
| 13:30.46 | oilinki7 | [TK]D-Fender: I |
| 13:31.00 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 13:31.01 | oilinki7 | [TK]D-Fender: I'm not sure for that yet. Need to check this one out |
| 13:31.36 | [TK]D-Fender | oilinki7: think of it as "half G.711" |
| 13:31.53 | oilinki7 | at some point, when I tried to make calls from my home (in thailand) to singapore with g729 the sound was quite ok. when using g711a, all I got was a blurry voice |
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| 13:34.11 | oilinki7 | [TK]D-Fender, alton2 and brodiem: thanks for the information. this was helpfull |
| 13:35.43 | *** join/#asterisk hatoon (n=musis@189.12.212.204) |
| 13:36.38 | [TK]D-Fender | oilinki7: You're welcome. |
| 13:36.40 | hatoon | algum Brasileiro |
| 13:38.59 | juniowww | hatoon eu |
| 13:42.42 | *** part/#asterisk jameswf-home (n=james@ip72-223-0-183.ph.ph.cox.net) |
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| 13:53.29 | ph8 | ~book |
| 13:53.29 | 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 |
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| 14:00.04 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:08.45 | mfdavid | hatoon: sim |
| 14:13.10 | *** join/#asterisk tkbeat (n=tk@80.64.182.204) |
| 14:17.45 | *** join/#asterisk ChrisHardie (n=ChrisHar@frigga.summersault.com) |
| 14:17.45 | *** join/#asterisk jmacz (n=jmacz@190.158.236.60) |
| 14:19.11 | ChrisHardie | You'd say I was just dreaming if I asked about adding a VOIP DID number into a PSTN hunt group managed by the local telco, right? |
| 14:19.38 | ManxPower | ChrisHardie: not at all. |
| 14:19.42 | ManxPower | it's up to your telco. |
| 14:20.01 | ManxPower | Its fairly common for people to have their PSTN lines roll over to a VoIP DID. |
| 14:20.15 | ChrisHardie | ManxPower: does there tend to be requirements about where the DID is provisioned from, etc? |
| 14:20.49 | mfdavid | I get this error when I try to make a call from console dial: handle_response_invite: Received response: "Forbidden" from '"asterisk" <sip:asterisk@192.168.15.102>;tag=as06091877'. Any tips? |
| 14:21.31 | [TK]D-Fender | ChrisHardie: I have done this personally. |
| 14:21.49 | [TK]D-Fender | ChrisHardie: Common anme of what to ask for is "busy / no-answer transfer" |
| 14:22.29 | [TK]D-Fender | mfdavid: Depends what it is in response TO. Pastebin the full SIP debug for the communication that generates that error |
| 14:22.29 | ChrisHardie | I assume the rollover DID number needs to be in the local exchange, though, to avoid LD or other fees/complications? |
| 14:22.42 | [TK]D-Fender | ChrisHardie: Its done by your telco. |
| 14:22.44 | jmacz | Hi everyone, does anyone know where I cand find the defult values that Asterisk sets for t.301 and t.303 timers in libpri 1.4.X? (pri_timers.h says that there are not configurable but doesn't show their default values) |
| 14:26.00 | *** join/#asterisk xenonex (n=xenonex@89.218.233.198) |
| 14:26.30 | *** part/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 14:26.52 | *** join/#asterisk _Krieger_ (n=krieger@195.114.133.197) |
| 14:27.05 | *** join/#asterisk spokra (n=spokra@host093-179-187.sea0.speakeasy.net) |
| 14:29.49 | *** join/#asterisk _zoomy_ (n=_zoomy_@dahlin.csbnet.se) |
| 14:30.06 | _zoomy_ | [TK]D-Fender: Thanks so much! With that push in the right direction I managed to make my first call!!! Turned out I had not configured iptables correctly... God the words "Hello world" must be one of the scentances throughout history which have created the most joy ;-) |
| 14:30.22 | *** join/#asterisk The-Bat (n=The-Bat@203.199.114.33) |
| 14:32.54 | [TK]D-Fender | _zoomy_: You're welcome. |
| 14:34.04 | *** join/#asterisk ElSonico (n=tav@cs181238069.pp.htv.fi) |
| 14:35.50 | codestr0m | I'm used to gentoo (which is configured correctly) and now on fedora.. I'm having an issue.. /var/lib/asterisk/sounds/ has the sounds in it, but I'm getting ast_streamfile: Unable to open please_hold_while_I_try_that_extension (format 0x4 (ulaw)): No such file or directory |
| 14:36.06 | codestr0m | where the hell are the sound files supposed to go then? |
| 14:37.57 | *** join/#asterisk seanmh (i=seanmh@216.31.101.103) |
| 14:40.43 | *** join/#asterisk linuxstb (n=linuxstb@rockbox/developer/linuxstb) |
| 14:46.10 | km- | is there a way to request the remote sip peer use a certain port range for rtp? |
| 14:46.53 | km- | ah, rtp.conf |
| 14:47.03 | km- | returns to lurking |
| 14:49.17 | ManxPower | codestr0m: the location changed between 1.2 and 1.4. It should have been mentioned in UPGRADE.txt |
| 14:49.54 | codestr0m | ManxPower: thanks. I'll take a look.. |
| 14:50.41 | _Krieger_ | please can somebody give me any sip traces of srtp`ed calls (the calls between phones are preferred)? |
| 14:52.43 | jeev | wow, mccain picked a female? |
| 14:53.36 | *** join/#asterisk tkbeat (n=tk@80.64.182.204) |
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| 14:54.46 | *** join/#asterisk linuxer_igor (n=igor@mlsrj200152096p202.mls.com.br) |
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| 14:55.45 | linuxer_igor | help. I need the AgentCallbackLogin functionality and in asterisk-1.6 is retired how to substitute ? |
| 14:55.55 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
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| 14:56.55 | *** join/#asterisk tkbeat (n=tk@80.64.182.204) |
| 14:56.56 | ManxPower | linuxer_igor: there should be examples either in the upgrade.txt or in extensions.conf.sample. You did not see information there? |
| 15:00.01 | linuxer_igor | I see this samples . .but don't substitute the functionality AgentCallbacklogin |
| 15:00.06 | linuxer_igor | look |
| 15:00.32 | linuxer_igor | I have agent statics logged in queue |
| 15:01.24 | linuxer_igor | and I need to link the agent to extension where he is |
| 15:05.50 | ManxPower | linuxer_igor: Why in the world did you install 1.6BETA anyway? |
| 15:06.16 | ManxPower | Or is this not a production system? |
| 15:09.48 | seanbright | linuxer_igor: also, if it's possible, could you take any longer to respond to simple questions? kthx. |
| 15:10.09 | linuxer_igor | sorry |
| 15:10.11 | seanbright | "i have urgent problem." |
| 15:10.15 | seanbright | "what is it?" |
| 15:10.19 | seanbright | ... 10 minutes later ... |
| 15:10.23 | seanbright | "i have urgent problem" |
| 15:10.46 | seanbright | needs to start drinking earlier in the day |
| 15:11.28 | Qwell | seanbright: coffee or whiskey? |
| 15:11.41 | seanbright | Qwell: hemlock |
| 15:11.49 | linuxer_igor | the 1,6 are very better. many new features. and I like to be beta test |
| 15:11.51 | jeev | drinking is for _losers_ |
| 15:11.53 | linuxer_igor | hehehee |
| 15:15.44 | *** join/#asterisk daniev (n=ganbarim@190.144.60.154) |
| 15:16.41 | _Krieger_ | please can somebody give me any sip traces of srtp`ed calls (the calls between phones are preferred)? |
| 15:17.01 | ManxPower | _Krieger_: I doubt anyone here uses SRTP |
| 15:17.45 | ManxPower | Asking a question and then going AFK is one the most rude things you can do here. |
| 15:18.11 | Qwell | ManxPower: why is that? |
| 15:18.18 | Qwell | hmm, that doesn't work on IRC...nevermind. |
| 15:18.36 | Qwell | ManxPower: If this was jabber, my question would have been immediately followed by "Qwell is now away." |
| 15:19.40 | _Krieger_ | ManxPower, i suggest your last message was not for me? |
| 15:21.27 | *** join/#asterisk buzzyd (n=psp-man@86.54.239.113) |
| 15:22.40 | *** join/#asterisk Sargun (n=xbmodder@atarack/staff/sargun) |
| 15:22.42 | *** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler) |
| 15:22.48 | buzzyd | hi guys anyone know how I can restrict calls coming into my pbx to only those that either I've registered with or registered with me as I appear to be able to dial say 1234@domain and ring in still |
| 15:23.14 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 15:24.31 | ManxPower | _Krieger_: it is for anyone that asks a question and then goes AFK. |
| 15:24.56 | ManxPower | buzzyd: set context=INVALID in [general] then set the correct context= line for each of your peers. |
| 15:25.12 | buzzyd | thanks |
| 15:26.07 | *** join/#asterisk F00JIN (n=F00JIN@lns-bzn-24-82-64-133-40.adsl.proxad.net) |
| 15:26.12 | F00JIN | hi ! |
| 15:29.27 | Deeewayne | O.O |
| 15:29.36 | Deeewayne | hello |
| 15:30.11 | *** join/#asterisk fogo (n=Paul@69.169.132.200) |
| 15:30.39 | F00JIN | I'm voip noob and i'd like to have some advices |
| 15:32.22 | *** join/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 15:32.36 | F00JIN | i'd like to test voip with asterisk but i don't know what is the best distro to use |
| 15:32.56 | ManxPower | F00JIN: W |
| 15:33.02 | *** join/#asterisk aliver (n=aliver@ip-216-17-149-97.rev.frii.com) |
| 15:33.05 | ManxPower | F00JIN: Whatever distro you are most comfortable with. |
| 15:33.27 | F00JIN | asterisknow, trixbox, switchvox, elastix |
| 15:33.46 | RecycleBin | asteriskNow seems decent you get some nice GUI tools from what i read |
| 15:34.01 | aliver | I must be misreading the "show application gotoif" instructions. My gotoif() isn't working. Anyone see a problem with this: |
| 15:34.02 | aliver | exten => _7xx,2,Gotoif(${CALLERID(dnid)} == ${EXTEN}:7:3 |
| 15:34.13 | aliver | besides the missing ) |
| 15:34.19 | aliver | That just got cut off |
| 15:34.53 | aliver | I basically want to say "If the guy calling the extention is actually calling FROM the same extension then go to priority 7" |
| 15:35.34 | aliver | Or maybe using "dnid" is the wrong thing? |
| 15:36.07 | F00JIN | i'm gonna try this |
| 15:38.56 | Nate187 | try: GotoIf($["${CALLERID(dnid)}" = "${EXTEN}:7:3"]?1001) |
| 15:39.03 | Nate187 | where 1001 is the exten to goto |
| 15:39.04 | [TK]D-Fender | aliver: Go read about "asterisk expressions" on the WIKI |
| 15:39.24 | [TK]D-Fender | aliver: And "channelvariables.txt froom your source docs folder |
| 15:43.27 | *** join/#asterisk grandpapadot (n=no@adsl-074-185-089-046.sip.bhm.bellsouth.net) |
| 15:45.33 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 15:45.39 | buzzyd | ManxPower: doing what you suggested has stopped all inbound calls including peers any idea's am I missing something in my extensions.conf should it have the [peer] as a context instead of [general]? |
| 15:46.28 | *** part/#asterisk RandalSchwartz (i=merlyn@p3m/member/merlyn) |
| 15:46.29 | ManxPower | buzzyd: In each peer do you have a context=whaevercontextyouneed? |
| 15:46.47 | ManxPower | If there is no context= line in the peer, then the context= in [general] will be used. |
| 15:46.47 | buzzyd | yes |
| 15:47.03 | ManxPower | buzzyd: then there is either something you are not telling me or you did it wrong. |
| 15:47.20 | ManxPower | I suspect the incoming calls do not match the peers you think it's matching. |
| 15:49.51 | *** join/#asterisk af_ (n=getsmart@88-149-241-182.dynamic.ngi.it) |
| 15:50.39 | *** join/#asterisk CunningPike (n=arodgers@204.239.8.157) |
| 15:52.25 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 15:53.04 | ManxPower | buzzyd: you can confirm this by commenting out all your peers, put the correct context= in [general] and then try. Even without your peers defined, I bet it will accept the calls. |
| 15:53.13 | buzzyd | ManxPower: can I send you a txt file? |
| 15:54.44 | [TK]D-Fender | ~pb |
| 15:54.45 | 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 |
| 15:54.46 | [TK]D-Fender | ^^^^^^^^^^^^^ |
| 15:54.53 | ManxPower | buzzyd: No. I am not paid to be here. |
| 15:54.56 | bbhenry | ls |
| 15:56.30 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 15:56.30 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 15:56.31 | buzzyd | ManxPower: Perhaps you could have a look at http://paste.lisp.org/display/66067. |
| 15:56.38 | buzzyd | I appreciate that |
| 15:56.40 | aliver | Is it possible to have DHCPd hand out the FTP server for the phones to get firmware and upload logs to, rather than TFTP? |
| 15:56.56 | lmadsen | Corydon76-dig: question... is there anything that does res_odbc failover of DBs in Asterisk which I haven't seen yet? |
| 15:57.13 | *** join/#asterisk jonathanr (n=jonathan@87-194-164-154.bethere.co.uk) |
| 15:57.19 | Corydon76-dig | Only the insert failover for updates |
| 15:57.21 | *** join/#asterisk theHub (n=theHub@69.177.93.21) |
| 15:57.26 | lmadsen | hrmmmm |
| 15:57.27 | lmadsen | darn |
| 15:57.30 | [TK]D-Fender | aliver: For polycom the server address has nothing to do with the PROTOCOL used to access it. That is chosen in your BootROM |
| 15:57.51 | Corydon76-dig | M13083 |
| 15:58.05 | lmadsen | Corydon76-dig: which means I need to probably write a script to track the status of a remote database, then failover to the local DB should it become unreachable by changing my res_odbc.conf file then doing a 'module reload res_odbc.so' |
| 15:58.57 | aliver | [TK]D-Fender in other words you have to program the phone manually to change it from the default (TFTP) to use FTP instead? |
| 15:58.59 | Corydon76-dig | Yeah, res_odbc doesn't do failover, but func_odbc does |
| 15:59.04 | ManxPower | buzzyd: I stand by my statement even after looking at your pastebin. |
| 15:59.05 | lmadsen | Corydon76-dig: ya, unfortunately that is useful for func_odbc, but I need to do failover for my realtime peers as well, darn, guess I need to figure out an out of band solution |
| 15:59.19 | Corydon76-dig | Interesting |
| 15:59.20 | lmadsen | Corydon76-dig: thanks for the update |
| 15:59.38 | Corydon76-dig | I suppose that could be done, but I'd do it in extconf |
| 15:59.49 | lmadsen | Corydon76-dig: ya, for clustering, have a client that is going to have a centralized DB for all peers, but if the network connection should go down, I want to failover to a local DB |
| 16:00.08 | buzzyd | ManxPower: You are spot on it does still accept my calls why is that? |
| 16:00.09 | Corydon76-dig | add a third parameter for each which is the priority of selection |
| 16:00.12 | lmadsen | Corydon76-dig: I agree, I think it'd be nice to specify multiple DBs for failover in extconfig.conf |
| 16:00.20 | lmadsen | makes sense |
| 16:00.38 | Corydon76-dig | The issue I have with it is timeout |
| 16:00.39 | lmadsen | sipusers => odbc,my_database,sipfriends,1 |
| 16:00.40 | *** join/#asterisk beek (n=klinebl@65.211.106.242) |
| 16:00.49 | lmadsen | sipusers => odbc,my_database_failover,sipfriends,2 |
| 16:00.51 | Corydon76-dig | Right |
| 16:01.26 | lmadsen | Corydon76-dig: what about setting a global option in res_odbc.conf or something to specify the timeout in seconds? |
| 16:01.37 | lmadsen | or maybe that would also go in extconfig.conf |
| 16:01.42 | *** join/#asterisk Levonk (n=lk@76.237.13.3) |
| 16:01.48 | lmadsen | but if it was in res_odbc.conf then you could use it for func_odbc and extconfig.conf |
| 16:02.06 | Corydon76-dig | I'd basically need to do a ping function in a thread within res_odbc |
| 16:02.18 | lmadsen | aye |
| 16:02.20 | lmadsen | SELECT 1? |
| 16:02.36 | lmadsen | that won't work with all DBs though right? (I think I remember you saying that) |
| 16:02.41 | Corydon76-dig | whatever sanitysql is. We already use that. |
| 16:02.46 | lmadsen | ah gotcha |
| 16:03.19 | lmadsen | not entirely sure why I haven't thought of this feature before :) |
| 16:03.31 | buzzyd | ManxPower: is the the SIP provider that is sending it to general? |
| 16:03.47 | *** join/#asterisk grandpapadot (n=no@adsl-074-185-089-046.sip.bhm.bellsouth.net) |
| 16:04.38 | codestr0m | I've read the UPGRADE 1.4 doc and while it mentions a language prefix... my sound files are all english.. I've tried sounds/ and sounds/en I've changed languageprefix=yes and still.. can't find my sounds.. help? |
| 16:04.49 | codestr0m | this just worked on gentoo and fedora obviously was compiled differently |
| 16:05.46 | ManxPower | buzzyd: no. The sip provider is not sending as user "peer1-in". |
| 16:05.58 | grandpapadot | codestr0m: I mention only because you mention compiliing, there are a couple of known issues with GCC 4.2+ |
| 16:06.05 | ManxPower | It is sending some OTHER username and you do not have anything that matches that username in sip.conf |
| 16:06.30 | codestr0m | grandpapadot: I'm using the yum package for asterisk.. this should be dead simple.. I copied over my configs.. my sounds.. and the shit isn't working |
| 16:06.43 | codestr0m | same version of asterisk I double checked |
| 16:06.46 | ManxPower | whatever in sip.con that is inside [here] is what Asterisk expects the incoming username to be. |
| 16:06.51 | tzafrir_laptop | grandpapadot, a number? |
| 16:06.51 | *** join/#asterisk Deeewayne (n=Deeewayn@nat/digium/x-89b25ede741d2819) |
| 16:06.51 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 16:07.19 | grandpapadot | codestr0m: What format are your sound files in? |
| 16:07.22 | wasim | i'm going to send you some more funds this coming month and hopefully we'll be able to do it monthly as the customer will commit some support to us |
| 16:07.24 | tzafrir_laptop | gcc > 4.2 is the default compiler on all new distros |
| 16:07.27 | grandpapadot | tzafrir_laptop: eh? |
| 16:07.48 | grandpapadot | tzafrir_laptop: I'm referring to the GSM codec issue... |
| 16:08.00 | grandpapadot | tzafrir_laptop: Just throwing it out there if his files are gsm |
| 16:08.21 | codestr0m | grandpapadot: file /var/lib/asterisk/sounds/en/netsyncro_com_welcomes_you_please_hold_while_I_try_that_extension.sln |
| 16:08.22 | codestr0m | /var/lib/asterisk/sounds/en/netsyncro_com_welcomes_you_please_hold_while_I_try_that_extension.sln: data |
| 16:08.29 | codestr0m | file /var/lib/asterisk/sounds/en/netsyncro_com_welcomes_you_please_hold_while_I_try_that_extension.wav |
| 16:08.30 | codestr0m | /var/lib/asterisk/sounds/en/netsyncro_com_welcomes_you_please_hold_while_I_try_that_extension.wav: RIFF (little-endian) data, WAVE audio, ITU G.711 mu-law, mono 44100 Hz |
| 16:08.38 | coppice | you'd think someone would fix the codec after all this time :-) |
| 16:08.44 | grandpapadot | What does the console show? (try pastebin) |
| 16:08.47 | buzzyd | ManxPower: Thanks for you help please let me know where to send beer tokens :) |
| 16:08.56 | ManxPower | Paypal eric@fnords.org |
| 16:09.08 | ManxPower | or just do a ~manxpower |
| 16:09.29 | *** part/#asterisk bbhenry (n=oper@wsip-68-105-250-226.sd.sd.cox.net) |
| 16:09.45 | buzzyd | got it expect some from psp-man |
| 16:09.47 | ManxPower | codestr0m: your're not doing something stupid like specifying the file extension are you? |
| 16:10.30 | codestr0m | ManxPower: I'm copying over working configs so I'd hope stupid things would be generally ruled out |
| 16:10.31 | grandpapadot | codestr0m: You could try longer filenames... You may have to upgrade to a 64-bit kernel for the OS to understand them but it might help |
| 16:11.16 | x86 | Aug 29 11:10:36 rpc-pbx-peo-02 asterisk[1712]: NOTICE[1740]: chan_iax2.c:6599 in socket_read: Out of idle IAX2 threads for I/O, pausing! |
| 16:11.20 | x86 | how do i fix that? |
| 16:11.21 | codestr0m | I can shorten the file names. that's not an issue.. (which I haven't tried) http://rafb.net/p/YBxL3N42.html |
| 16:11.26 | ManxPower | Playback(jump-thru-hoops.wav) <--- WRONG. Playback(jump-thru-hoops) <-- CORRECT |
| 16:11.26 | x86 | I just restarted asterisk completely! |
| 16:11.37 | *** join/#asterisk neurosys (n=neurosys@adsl-067-032-132-060.sip.bct.bellsouth.net) |
| 16:12.07 | ManxPower | codestr0m: Um, Asterisk expects 8000Hz, not 44100 Hz |
| 16:12.12 | codestr0m | ManxPower: I appreciate you trying to help, but I think I got it.. exten => xxxxx,3,Playback(netsyncro_com_welcomes_you_please_hold_while_I_try_that_extension) |
| 16:12.26 | codestr0m | I'm also providing the sln |
| 16:12.33 | x86 | gah, i said stop now in CLI, it dropped me back to a command prompt, but failed to actually stop asterisk |
| 16:12.34 | grandpapadot | codestr0m: check the path in /etc/asterisk/asterisk.conf just to make sure and then just for the heck of it shorten one |
| 16:12.36 | x86 | fun stuff |
| 16:12.48 | ManxPower | x86: you are in the middle of a crash. |
| 16:13.15 | x86 | was, until I kill -9'd asterisk ;) |
| 16:13.28 | x86 | seems I have to do that about once a week or so |
| 16:13.58 | ManxPower | Joy. Our mail server has almost 600,000 messages to backup to our DRC |
| 16:14.16 | x86 | I should setup a cron job to kill -9 asterisk every night around 10pm or so (inittab will automagically start it back up) |
| 16:14.16 | ManxPower | x86: what version of Asterisk are you running anyway? |
| 16:14.26 | x86 | ManxPower: heh, it's old man.. 1.4.12.1 :p |
| 16:14.51 | ManxPower | x86: afraid to upgrade, I assume?" |
| 16:15.10 | x86 | well it seems to work fine except tiny quirks like that |
| 16:15.22 | x86 | I'm waiting to upgrade until a stable 1.6 comes out |
| 16:17.18 | codestr0m | grandpapadot: [Aug 29 16:16:42] WARNING[9314]: file.c:602 ast_openstream_full: File shorter does not exist in any format.. same thing. I've also checked the perms as user asterisk... |
| 16:17.29 | codestr0m | maybe mpg123 or whatever it was didn't get installed? |
| 16:18.24 | *** join/#asterisk jameswf-home (n=james@dsl093-157-131.phx1.dsl.speakeasy.net) |
| 16:18.43 | x86 | gah what's the default password on the web interface of polycom phones? isn't it 456? |
| 16:19.02 | x86 | ah, figured it out... username has to be filled in as Polycom too |
| 16:19.42 | x86 | what the hell, you can't tell a polycom phone the TFTP server from the web interface? that's crap |
| 16:19.58 | ManxPower | codestr0m: mpg123 is not ever used for playing files. in 1.4 mpg123 is no longer needed to play MoH files. |
| 16:20.20 | ManxPower | x86: no. You do it via the pre-boot menu or via provisioning files. |
| 16:20.34 | ManxPower | or best of all do it in DHCP like everyone else does. |
| 16:21.14 | ManxPower | come to think of only recent SIP firmwares allow you to set that in provisioning files, of course it's pretty pointless. |
| 16:22.28 | codestr0m | is it standard to install the asterisk-oss/alsa rpm's? ManxPower grandpapadot.. could that be it? |
| 16:22.36 | ManxPower | if it can get to the provisioning files then the boot method is alreadt set. |
| 16:23.15 | ManxPower | codestr0m: What is your fetish for 3rd party software? You don't even need a fricking sound card in the Asterisk server, and certinally don't need OSS or ALSA unless you want to run a console phone. |
| 16:23.41 | codestr0m | ManxPower: my fetish with 3rd party software? |
| 16:24.00 | ManxPower | codestr0m: you keep wanting to install software that has nothing to do with Asterisk. |
| 16:24.29 | ManxPower | mpg123 was not used since the days of 1.2. You never ever needed a sound card or OSS or ALSA. |
| 16:24.50 | *** join/#asterisk hubguruJR (n=hubguruJ@mail.ntegratedsolutions.com) |
| 16:25.03 | ManxPower | come to think of it, mpg123 has not been used since the days of 1.0. 1.2 introduced the ability to do MoH without mpg123 |
| 16:25.07 | codestr0m | ManxPower: ok. so.. what's missing or was it just simply not compiled with the support I need? |
| 16:25.25 | ManxPower | codestr0m: I have no idea what is misssing, but it's not software |
| 16:25.29 | grandpapadot | codestr0m: Is your asterisk from an RPM or did you compile from source? |
| 16:25.40 | codestr0m | grandpapadot: rpm |
| 16:25.51 | ManxPower | grandpapadot: If he answers "RPM" then we get to knife him in the kidneys. |
| 16:25.55 | grandpapadot | codestr0m: that whole situation sound messy ... |
| 16:26.00 | ManxPower | codestr0m: nobody can help you with an RPM install. |
| 16:26.02 | x86 | ManxPower: well I don't want other crap trying to boot from my TFTP server (PXE desktops, thin clients, routers, switches, etc) |
| 16:26.13 | ManxPower | x86: then put the phones on their own vlan |
| 16:26.23 | x86 | ManxPower: so I usually setup the TFTP server setting on the phone manually, but I thought it might be possible (and sane) to do it from the website |
| 16:26.31 | grandpapadot | codestr0m: Dump Fedora as a production OS (use CentOS, Redhat, debian) and then compile from source and you'll be in great shape. |
| 16:26.31 | codestr0m | ManxPower: fair enough.. this is what I get for trying to save time.. *grumbles* |
| 16:26.41 | x86 | ManxPower: can't... some of the switches we use don't support 802.1Q |
| 16:26.47 | codestr0m | I'll just grab the sources and the same options gentoo used |
| 16:26.47 | x86 | ManxPower: it's a mess over here |
| 16:26.50 | ManxPower | codestr0m: Installing Asterisk from a package NEVER EVER saves time. |
| 16:27.05 | x86 | ManxPower: hell, most of our switches don't even support STP |
| 16:27.08 | codestr0m | ManxPower: yeah. well.. I thought this was pretty dead simple to get /right/ |
| 16:27.11 | grandpapadot | ManxPower++ |
| 16:27.18 | ManxPower | In fact you have wasted not only your own time, but wasted everyone else's time too. |
| 16:27.46 | grandpapadot | codestr0m: Asterisk is quite easy to compile from source ... |
| 16:27.54 | x86 | ManxPower: I'm almost certain that asterisk crash was related to a broadcast storm, due to one of our older switches' lack of STP support |
| 16:28.00 | codestr0m | k. thanks guys. I'll manage on my own from here |
| 16:28.01 | *** join/#asterisk hfb (n=hfb@pool-96-247-108-198.lsanca.dsl-w.verizon.net) |
| 16:28.33 | *** join/#asterisk eric256 (n=eric256@208.42.253.116) |
| 16:30.09 | grandpapadot | codestr0m: Google said this *might* be helpful if you want to continue using fedora and compile Asterisk from source: http://www.asteriskguru.com/tutorials/asterisk_installation_compilation_fedora.html |
| 16:32.00 | ph8 | If i want to use 5 traditional phone handsets with an asterisk install can i somehow get an analog to ethernet plug? I'd need to them to register with SIP as well - is that just impossible? |
| 16:32.36 | bkw_ | ph8: an ATA |
| 16:33.24 | grandpapadot | ph8: It's possible but a Polycom 330 can be had for $100 and will be a much better choice, imho. |
| 16:33.26 | *** part/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 16:33.52 | ph8 | grandpapadot: I have a grandstream :-) someone who fancies a cheap voip is asking me |
| 16:34.03 | grandpapadot | ph8: grandscream is crap |
| 16:34.48 | grandpapadot | ph8: By the time you deal with the hassle, additional ATA equipment, etc, I'm willing to bet you could have bought the 330's |
| 16:34.57 | ph8 | you're probably right |
| 16:35.03 | ph8 | grandpapadot: Polycom advocate? ;) |
| 16:35.08 | ph8 | my grandstream's doing me very nicely |
| 16:35.16 | Maliuta | my cisco is nice |
| 16:35.27 | grandpapadot | grandpapadot: Not necessarily but they are *very* good matches for any asterisk deployment and are very dependable phones. |
| 16:35.33 | grandpapadot | ph8: to you |
| 16:35.39 | Maliuta | I also have a TMD400P that handles some FXO/FXS for me |
| 16:36.15 | grandpapadot | Cisco's work great with SIP as long as you're not using G.729 or NAT in most cases |
| 16:36.19 | Maliuta | but grandpapadot is right with the cost+hassle of 5 ATA devices you'd be better off getting digi handsets |
| 16:36.33 | Maliuta | grandpapadot: and that I am :) |
| 16:36.35 | grandpapadot | The 7940/7960's seem to work great with NAT but the G.729 issues are still there |
| 16:37.06 | Maliuta | grandpapadot: I bought some digium licences, I don't have any issues at all |
| 16:37.08 | ManxPower | what G729 issue? |
| 16:37.39 | Maliuta | ManxPower: was assuming he was talking about licence issues |
| 16:37.41 | grandpapadot | Maliuta: With NAT and G.729 you can't "confernce in" a second party from the 7940/7960's |
| 16:38.04 | ManxPower | grandpapadot: that has nothing to do with NAT |
| 16:38.10 | Maliuta | grandpapadot: unless the confernce is on the same * box |
| 16:38.34 | ManxPower | it has to do with 1) cisco has only one license for G729 per phone and I believe the phones simply don't have enough CPU to do two G729 calls. |
| 16:38.51 | ManxPower | I'm suprized anyone even uses Cisco phones anymore -- there are so many good phones out there now. |
| 16:39.12 | ManxPower | Polycom and Linksys are some of the most common ones, as well as Aastra and Snom |
| 16:39.24 | grandpapadot | What ManxPower said, a much better authority to be sure |
| 16:40.03 | Maliuta | I wouldn't do a conference on the phone anyhow, that's what I have a server for |
| 16:40.35 | grandpapadot | Considering the 7940 was introduced in 1999 and still around it gets at very least my respect |
| 16:40.43 | *** part/#asterisk eric256 (n=eric256@208.42.253.116) |
| 16:40.44 | ManxPower | Why deal with a company that doesn't even WANT you to buy their products? |
| 16:40.53 | grandpapadot | ManxPower: true, true |
| 16:41.12 | ManxPower | grandpapadot: Cisco does not treat their products like mayflys. They treat them like business products. |
| 16:42.57 | ManxPower | That is one of the things I like about Cisco (at least for their non-phone products). Generally you can buy the same models you currently have. |
| 16:43.08 | grandpapadot | ManxPower: my point really was that it's a great piece of engineer and has held it's own in. I personally think it's a great little platform that with some attention from Cisco for non-cisco adoption (beyond just releasing SIP 8.2 to the public) might even keep it around a quite a bit longer. |
| 16:43.10 | ManxPower | unlike many companies where they discontinue a product after sometimes only a few weeks. |
| 16:43.22 | ManxPower | grandpapadot: it's a fine phone. |
| 16:43.39 | ManxPower | But I don't really want to buy phones from a company that doesn't even want to sell me phones. |
| 16:43.51 | ph8 | grandpapadot: What problems have you had with grandstream? |
| 16:43.52 | grandpapadot | ManxPower: I agree with you completely on that. |
| 16:44.00 | grandpapadot | ph8: grandscream is crap |
| 16:44.07 | grandpapadot | ph8: junk, junk, junk, junk |
| 16:44.08 | ph8 | i'm not experienced, but the one i've had worked fine - i was about name them 'supplier of choice' in my mind |
| 16:44.09 | ph8 | but why? |
| 16:44.12 | grandpapadot | ph8: not suitable for anything |
| 16:44.20 | grandpapadot | ph8: Not even good for makeshift hockey pucks |
| 16:44.26 | grandpapadot | ph8: Not good for doorstops |
| 16:44.34 | grandpapadot | ph8: not good for dog chew toys |
| 16:44.42 | grandpapadot | ph8: not good for free |
| 16:44.54 | grandpapadot | ph8: not good for <insert anything uselful here> |
| 16:44.56 | ph8 | great justification |
| 16:45.08 | grandpapadot | lol |
| 16:45.47 | *** part/#asterisk hubguruJR (n=hubguruJ@mail.ntegratedsolutions.com) |
| 16:53.01 | grandpapadot | ph8: seriously, grandstream is considered the bottom of the barrel choice in voip end-points |
| 16:53.22 | ph8 | but you haven't told me why |
| 16:54.51 | *** join/#asterisk LeddyHM (i=leddy@da-club.with.my.beerandcondoms.com) |
| 16:54.52 | grandpapadot | ph8: I don't have time to go into it but google is your friend |
| 16:55.32 | ph8 | i might learn my own lessons with them |
| 16:55.36 | ph8 | as i say it works perfectly atm |
| 17:01.32 | aliver | How do I get the phone to upload it's current configuration to TFTP? Or do I need to have a template already there? (PolyCom IP320) |
| 17:01.53 | ManxPower | aliver: it will do so automagically |
| 17:02.03 | aliver | Hmm, okay. |
| 17:02.19 | ManxPower | That is how I created a template config file. |
| 17:02.43 | ManxPower | Set up the phone like I want, let the phone upload it's config, use the uploaded config as a template for all the other 100+ phones. |
| 17:03.24 | aliver | Sounds reasonable. Thanks. |
| 17:04.12 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 17:06.58 | *** join/#asterisk sah-work (n=Bawbatos@65-119-47-100.dia.static.qwest.net) |
| 17:09.22 | *** join/#asterisk SuPrSluG (n=SuPrSluG@24.75.47.130) |
| 17:09.52 | *** join/#asterisk gr0mit (n=tim@81.187.32.146) |
| 17:10.55 | *** join/#asterisk zeeesh (i=zeeesh@203.215.179.43) |
| 17:12.19 | ph8 | does anyone know of any software for ubuntu / linux generally that does things like popup a box when i get a call etc? |
| 17:12.27 | zeeesh | at normal asterisk console we can verify how many sip peers are registered with the server. using REALTIME asterisk ... how to check which peers are registered or not ? |
| 17:12.29 | *** join/#asterisk F00JIN (n=F00JIN@lns-bzn-24-82-64-133-40.adsl.proxad.net) |
| 17:14.48 | ph8 | sip show peers |
| 17:19.54 | *** part/#asterisk ChrisHardie (n=ChrisHar@frigga.summersault.com) |
| 17:22.03 | zeeesh | <ph8>: i know its work with normal asterisk .. i m asking if u r working on realtime asterisk ... then which command u will give at console? |
| 17:22.27 | ph8 | ah sorry i have no idea what that is |
| 17:23.03 | ManxPower | zeeesh: Have you TRIED it? |
| 17:23.33 | ManxPower | zeeesh: have you read the realtime docs in the doc directory? |
| 17:25.08 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 17:26.10 | *** join/#asterisk makkksimal (n=makkksim@e177210248.adsl.alicedsl.de) |
| 17:26.16 | zeeesh | <ManxPower>: sip show peers.. it does'nt show those peers status which are saved in database ... it only shows which peer i hv staticly define in sip.conf.. so thats y asking ? |
| 17:28.50 | ManxPower | zeeesh: You answered only the first of my two questions. |
| 17:31.31 | brodiem | zeeesh: realtime peers will be available in your show peers list once they become registered |
| 17:33.09 | brodiem | zeeesh: and rtcachefriends=yes |
| 17:40.02 | [TK]D-Fender | aliver: Polycom is FTP by default |
| 17:40.53 | ManxPower | brodiem: I assume that's documented in /path/to/src/asterisk/doc |
| 17:48.20 | *** join/#asterisk lirakis (n=lirakis@65.200.191.241) |
| 17:48.37 | lirakis | is there anyway to disable the meetme notification for when users enter/exit a conference?? |
| 17:49.14 | *** join/#asterisk tobias (n=tobias@user-0c2hj2f.cable.mindspring.com) |
| 17:50.42 | kaldemar | lirakis: parameter q. core show application meetme |
| 17:51.22 | lirakis | kaldemar: ah great! thanks .. i was looking at the app ref in ATOFT .. must have missed it. |
| 17:51.23 | lirakis | thanks |
| 17:52.23 | *** join/#asterisk rcahilig (n=root@202.78.75.254) |
| 17:52.57 | rcahilig | hello |
| 17:53.42 | beek | hello |
| 17:55.38 | *** part/#asterisk rcahilig (n=root@202.78.75.254) |
| 17:55.51 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 17:58.18 | ManxPower | The only official source of application docs is "core show application X" All other sources can be out of date. |
| 18:03.28 | *** join/#asterisk macli (n=macli@nmc.brc.ubc.ca) |
| 18:07.31 | *** join/#asterisk nn (n=nn@unaffiliated/nn) |
| 18:09.12 | *** join/#asterisk Firass-z0r (n=asadf@ead224-222.housing.wwu.edu) |
| 18:14.20 | *** join/#asterisk linuxstb (n=linuxstb@rockbox/developer/linuxstb) |
| 18:18.59 | lmadsen | ManxPower: and probably are |
| 18:20.55 | *** join/#asterisk ohayden (i=ohayden@adhd.irule.net) |
| 18:21.08 | *** join/#asterisk mattx86_ (n=root@static20247.sftncc.ken-tennwireless.com) |
| 18:24.29 | mattx86_ | hey guys, any ideas as to why dialing any two digits on my Analog Phones/Channel Bank/PSTN Asterisk setup, it goes straight to busy and Asterisk shows the phone has hungup. |
| 18:24.37 | *** part/#asterisk lirakis (n=lirakis@65.200.191.241) |
| 18:24.48 | *** join/#asterisk neurosys (n=neurosys@c-66-229-91-186.hsd1.fl.comcast.net) |
| 18:25.07 | mattx86_ | I've tried different phones and different channels, different channel banks, different settings and I still can't figure it out |
| 18:25.40 | *** join/#asterisk Ei5GTB (n=Paul@A-78-200.cust.iol.ie) |
| 18:26.09 | lmadsen | ~nei |
| 18:26.17 | lmadsen | hrmmm |
| 18:27.36 | Ei5GTB | so...recomendations on an ip phoine for the desk here... needs to able to grab asterisk by the balls and tell it what to do.. |
| 18:27.44 | lmadsen | nei is not enough information -- please provide any information relevant in answering your question. Asking if someone has an idea about what is wrong with your vaguely referenced situation is going to be nearly impossible to help move your issue forward. Please provide relevant dialplan examples, console output, or anything relevant. Also see ~pastebin. |
| 18:28.01 | lmadsen | jbot: nei is not enough information -- please provide any information relevant in answering your question. Asking if someone has an idea about what is wrong with your vaguely referenced situation is going to be nearly impossible to help move your issue forward. Please provide relevant dialplan examples, console output, or anything relevant. Also see ~pastebin. |
| 18:28.01 | jbot | okay, lmadsen |
| 18:28.51 | mattx86_ | you pick up the handset: Asterisk - starting simple switch on zap/xx; Handset - Dialtone. Dial, eg, 12: Asterisk - hungup zap/xx; Handset - busy signal |
| 18:29.04 | [TK]D-Fender | mattx86pastebin your dialplan and zapata.conf |
| 18:29.15 | lmadsen | Ei5GTB: I like Polycom phones, others will like Aastra, and still others Linksys. Nearly no one will recommend Grandstream. |
| 18:29.43 | Ei5GTB | good good.. im sweeping through ebay,. see what i cant come across..ill look up polycom |
| 18:30.05 | mattx86_ | [TK]D-Fender: hm.. let me switch computers. one moment. |
| 18:30.33 | grandpapadot | Ei5GTB: The Polycom 330 is the best "bang for the buck" imho |
| 18:31.16 | Ei5GTB | grandpapadot, pls dont message me to advertise...why would i pay someone to do somthing i am doing 80% for fun and 20% for necesity |
| 18:31.43 | Ei5GTB | ill look up the polycom 330 tho, tnx |
| 18:31.49 | grandpapadot | Ei5GTB: It was a suggestion not advertising ... |
| 18:32.09 | Ei5GTB | i already have a working pbx...asterisk is a bit more fun to play with |
| 18:32.12 | Ei5GTB | its just a hobby |
| 18:32.14 | Ei5GTB | for the house |
| 18:32.19 | [TK]D-Fender | Ei5GTB: What do you really intend for your choice of phone? |
| 18:32.41 | [TK]D-Fender | Ei5GTB: SIP phones are fun and all but for basic use ATA's are just fine and far cheaper |
| 18:32.41 | Ei5GTB | well...i would like to be able to......mind you...well.. |
| 18:32.47 | Ei5GTB | gets his thoughts straight |
| 18:33.09 | Ei5GTB | a phonebook would be nice... and being able to easly dial extensions (programmable buttons) |
| 18:33.26 | Ei5GTB | transfer and hold calls |
| 18:33.31 | Ei5GTB | connect calls |
| 18:33.36 | Ei5GTB | together |
| 18:33.58 | *** join/#asterisk denon (n=denon@tooth.decay.org) |
| 18:33.58 | *** mode/#asterisk [+o denon] by ChanServ |
| 18:33.59 | Ei5GTB | etc... i know all that could all be done from asterisks dialplan anyway.. but i like loadsa buttons |
| 18:34.10 | *** join/#asterisk jjshoe (n=jjshoe@72.37.252.50) |
| 18:34.11 | Ei5GTB | also being able to intergrate it into a headset... |
| 18:34.35 | Ei5GTB | i want one for my radio bench, so when im on the radio, i can pres a button and have the radio in one ear and the phonecall in the other |
| 18:34.37 | Ei5GTB | etc.. |
| 18:34.43 | jjshoe | anyone know how to get a grandstream to off hook on a call from a call file? |
| 18:35.09 | [TK]D-Fender | Ei5GTB: for a "lot of buttons", the 3XX comes up short. You'd be looking at a 5XX or 6XX series phone for that and the price goes up quite a bit |
| 18:35.17 | grandpapadot | jjshoe: Step 1 - Place grandscream on the floow, Step 2 - smash with shoe, Step 3 - buy a Polycom |
| 18:35.17 | Ei5GTB | i see |
| 18:35.25 | grandpapadot | s/floow/floor |
| 18:35.29 | [TK]D-Fender | jjshoe: Go lookup "paging" on the WIKI |
| 18:35.29 | Ei5GTB | http://cgi.ebay.ie/Polycom-Soundpoint-IP-601-SIP-VOIP-Handset_W0QQitemZ260257578739QQihZ016QQcategoryZ61841QQrdZ1QQssPageNameZWD1VQQcmdZViewItemQQ_trksidZp1638Q2em118Q2el1247 |
| 18:35.31 | Ei5GTB | not to bad |
| 18:35.37 | Ei5GTB | oops, long url |
| 18:36.05 | Ei5GTB | if i went down the ATA route....whats the cheapest ata? |
| 18:36.10 | Ei5GTB | pci preferable |
| 18:36.17 | [TK]D-Fender | Ei5GTB: UK Polycom pricing is usually harsh. I REALLY hate to say it but look at Snom as well. |
| 18:36.31 | [TK]D-Fender | Ei5GTB: ATA's are little ethernet boxes, not "cards" |
| 18:37.03 | jjshoe | I know it's Call-Info: answer-after=0, but would that turn into set Call-Info=answer-after=o ? |
| 18:37.08 | jjshoe | err 0, not o |
| 18:37.10 | Ei5GTB | [TK]D-Fender, oic |
| 18:37.11 | Ei5GTB | http://cgi.ebay.ie/Zapmicro-ZMA400P-4-Port-FXO-FXS-for-Asterisk-modules_W0QQitemZ180283156758QQihZ008QQcategoryZ61841QQssPageNameZWDVWQQrdZ1QQcmdZViewItem |
| 18:37.14 | Ei5GTB | so thats a? |
| 18:37.29 | [TK]D-Fender | jjshoe: "core show application sipaddheader" |
| 18:37.42 | aliver | I switched to G729 and now the voicemail sounds (vm-*) sound like utter crap. Any tips? |
| 18:37.48 | aliver | They are all staticy |
| 18:37.53 | [TK]D-Fender | Ei5GTB: that is a Chinese knock-off of the Digium TDM400P |
| 18:37.58 | Ei5GTB | i see |
| 18:38.05 | Ei5GTB | but what os the device called? |
| 18:38.07 | [TK]D-Fender | Ei5GTB: Linksys PAP2T-NA <-- THAT is an ATA |
| 18:38.10 | ManxPower | aliver: how many G729 licenses did you buy? |
| 18:38.10 | jjshoe | [TK]D-Fender can you add to the sipheader from a call file? |
| 18:38.14 | grandpapadot | aliver: what format are they in? |
| 18:38.21 | aliver | grandpapadot gsm |
| 18:38.22 | Ei5GTB | [TK]D-Fender, k |
| 18:38.29 | aliver | ManxPower a bunch (30) |
| 18:38.32 | [TK]D-Fender | aliver: well go get them in G.729 |
| 18:38.40 | grandpapadot | aliver: what ver of GCC did you use to compile asterisk? |
| 18:38.46 | aliver | grandpapadot 4.1 |
| 18:38.48 | ManxPower | maybe you are seeing the GSM bug. |
| 18:38.50 | grandpapadot | aliver: Or what TK said for no transcoding |
| 18:39.05 | aliver | [TK]D-Fender Are there native G729 files I can fetch somewhere? |
| 18:39.10 | grandpapadot | aliver: It's likely not the GSM bug if you're using GCC < 4.2 |
| 18:39.11 | ManxPower | Transcoding to/from G729/GSM should not cause significant audio quality issues. |
| 18:39.24 | aliver | Also, what extension do they need to have? .g729 ? |
| 18:39.28 | [TK]D-Fender | aliver: Clearly as I just told you to go GET THEM. |
| 18:39.33 | Ei5GTB | [TK]D-Fender, does that so both FXO and FXS? |
| 18:39.35 | ManxPower | aliver: what version of GCC are you using? |
| 18:39.53 | [TK]D-Fender | Ei5GTB: the SPA-3102 does both, All the others are FXS only |
| 18:40.02 | aliver | ManxPower gcc version 4.2.3 |
| 18:40.04 | Ei5GTB | i see |
| 18:40.10 | ManxPower | There you go! |
| 18:40.21 | grandpapadot | aliver: I thought I just asked that !? |
| 18:40.22 | Ei5GTB | well, i only need 2 possibly 3 fxo ports |
| 18:40.32 | aliver | grandpapadot I was wrong |
| 18:40.34 | ManxPower | Whats the jbot word to have it spew the info? |
| 18:40.37 | aliver | I just checked |
| 18:40.44 | grandpapadot | <aliver> grandpapadot 4.1 |
| 18:40.53 | aliver | grandpapadot I was wrong (again) |
| 18:41.09 | *** join/#asterisk dmz (n=dmz@64.203.203.232.dyn-cm-pool-64.hargray.net) |
| 18:41.14 | jjshoe | anyone know how to get a grandstream to off hook on a call from a call file moved into the spool? |
| 18:41.23 | ManxPower | aliver: you realize that a single mistake in setting up Asterisk could open up your PBX to callers to make international calls and have them billed to you. |
| 18:41.37 | ManxPower | Be CAREFUL and be ACCURATE. |
| 18:42.08 | ManxPower | jjshoe: the .call file would have to dial a Local/ channel, then you can do what you need to do in the dialplan |
| 18:42.30 | jjshoe | ManxPower ah, stink, not nearly as much fun as aastra or poly |
| 18:42.44 | ManxPower | jjshoe: for ANY phone. |
| 18:42.46 | aliver | ManxPower Umm. Yeah. |
| 18:42.53 | Ei5GTB | probably a dumb Q.. but can you use a modem as an FXO port? |
| 18:42.56 | ManxPower | ~gcc-bug |
| 18:43.04 | aliver | But what does that have to do with gsm vs g729 audio codec issues? |
| 18:43.15 | jjshoe | ManxPower um, no, for aastra and poly you don't need to send them to a specific context that does it for you. |
| 18:43.27 | ManxPower | aliver: there is a bug somewhere that causes massive corruption of the decoded GSM audio. |
| 18:43.42 | ManxPower | jjshoe: then you can do the same with whatever phone you are using. |
| 18:43.50 | ManxPower | there is no difference between models |
| 18:43.53 | *** join/#asterisk tkbeat (n=tk@p54B96FD6.dip.t-dialin.net) |
| 18:43.53 | jjshoe | um, |
| 18:43.55 | jjshoe | wow |
| 18:43.59 | jjshoe | yes, yes there is. |
| 18:44.28 | jjshoe | both poly and aastra look for the var alert_info, but take different values to auto-offhook |
| 18:44.33 | ManxPower | (1:41:16 PM) jjshoe: anyone know how to get a grandstream to off hook on a call from a call file moved into the spool? |
| 18:44.43 | ManxPower | I assumed you knew how to do it without using a .call file. |
| 18:44.53 | ManxPower | I guess I was wrong. Go look it up in the Wiki. |
| 18:45.04 | jjshoe | I looked in the wiki, but I can see you're not clued :) |
| 18:45.09 | *** join/#asterisk bradleyprice86 (n=bradleyp@fw.datafax.net) |
| 18:45.13 | ManxPower | *sigh* |
| 18:45.20 | ManxPower | what grandstream model? |
| 18:46.17 | bradleyprice86 | Having trouble trying to dial out making long distance calls. exten => 4700,n,Dial(ZAP/g2/${NumToDial}) |
| 18:46.44 | bradleyprice86 | Does that seem like it should work when a have put a 1 and area code? |
| 18:46.58 | lmadsen | if ${NumToDial} is actually populated, then yes |
| 18:47.35 | bradleyprice86 | It keeps returning -- PROGRESS with cause code 127 received |
| 18:48.00 | ManxPower | For example: http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+page |
| 18:48.20 | ManxPower | All SIP phones that I am aware of, if they support audo answer, do so via a SIP header. |
| 18:48.56 | bradleyprice86 | I am trying to dial 18004664411 as a test. I can dial it directly and it works fine, just not when I use the dial method. |
| 18:51.12 | *** part/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 18:51.53 | *** join/#asterisk ElSonico (n=tav@xdsl-179-7.nblnetworks.fi) |
| 18:52.29 | ManxPower | jjshoe: This I was not aware of: http://lists.digium.com/pipermail/asterisk-users/2006-September/165377.html |
| 18:52.36 | [TK]D-Fender | bradleyprice86: PASTEBIn <--- |
| 18:53.23 | ManxPower | jjshoe: Or even this one: http://www.voip-info.org/wiki-Asterisk+Paging+and+Intercom |
| 18:53.45 | ManxPower | I'm sure I could find more in just a few mins if you would like? |
| 18:54.23 | [TK]D-Fender | ManxPower: and its not like I handed him the command either... |
| 18:54.33 | bradleyprice86 | [TK]D-Fender: http://pastebin.com/m6e438683 |
| 18:55.06 | [TK]D-Fender | ManxPower: Or told him to look up "paging" on the wiki which couldn't POSSIBLY have landed him on those very pages... |
| 18:56.47 | ManxPower | [TK]D-Fender: I wonder if s/he would like a foot massage and a complimentary beverage too. I mean if I'm going to be his google proxy and all. |
| 18:57.17 | [TK]D-Fender | bradleyprice86: If you are using a PRI you'd better be setting your CALLER ID f*before* you dial out. |
| 18:58.03 | bradleyprice86 | ok |
| 18:59.04 | ManxPower | bradleyprice86: at first glance I'd say it is either a telco problem (is this a new install or move?) or you have problems with the setting of pridialplan/prilocaldialplan/and priindication |
| 18:59.53 | ManxPower | Ah yes. I imagine many carriers would get upset if your callerid was not 10 digits, no - or . or quotes |
| 19:02.07 | bradleyprice86 | [TK]D-Fender & ManxPower: That was the problems. Works like a charm. Thanks guys. |
| 19:04.22 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 19:04.54 | ManxPower | which was? |
| 19:05.34 | _ShrikE | ManxPower: still in nola? |
| 19:05.40 | ManxPower | _ShrikE: yup. |
| 19:05.51 | ManxPower | Chances are I'll drive home tomorrow |
| 19:06.43 | _ShrikE | That may end up being a nasty drive. |
| 19:07.08 | ManxPower | We did pretty good when leaving for Katrina, not using the interstate, etc. |
| 19:07.48 | bradleyprice86 | ManxPower: not specifying cidnumber |
| 19:07.58 | ManxPower | bradleyprice86: Ah. |
| 19:08.14 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 19:09.28 | ManxPower | I'm monitoring the sync of the mail and web servers in covington to the secondary NOC in Jackson MS. about a half a million e-mail messages. |
| 19:10.29 | ManxPower | If they keep their house as clean as their e-mail boxes their houses would be demolished by the city. |
| 19:11.26 | *** join/#asterisk molkmin (i=d17c33d4@gateway/web/ajax/mibbit.com/x-f8df17ec3845cc68) |
| 19:11.53 | molkmin | hello |
| 19:12.04 | molkmin | my question is how to handle telemarketers |
| 19:14.02 | molkmin | or, does anyone know where to obtain an asterisk-compatible disconnect tone? |
| 19:17.47 | *** join/#asterisk bradleyprice86 (n=bradleyp@fw.datafax.net) |
| 19:17.54 | *** join/#asterisk bbryant (n=brett@c-68-59-20-153.hsd1.sc.comcast.net) |
| 19:19.54 | [TK]D-Fender | molkmin: "core show application playtones" |
| 19:21.23 | *** part/#asterisk codestr0m (n=codestr0@unaffiliated/codestr0m) |
| 19:24.09 | *** join/#asterisk Alanonzander (n=azander@baghdad.netonecom.net) |
| 19:25.24 | *** join/#asterisk mtaht4 (n=m@WAN59-246.cablenet.com.ni) |
| 19:26.16 | *** join/#asterisk Pimpachu (n=Pimpachu@c-71-200-221-159.hsd1.fl.comcast.net) |
| 19:26.56 | RecycleBin | Where can i find telephone prefix's for provinces. Locally i got all the prefixs out of my phone book. But no luck on the net for other provinces |
| 19:27.24 | molkmin | [TK]D-Fender: must be I have an older version of asterisk..command "core" not found |
| 19:28.55 | *** join/#asterisk devhen_ (n=devhen@107.235.sfcn.org) |
| 19:28.59 | ManxPower | molkmin: you like ZapataTeller? |
| 19:29.28 | molkmin | I'm not familiar with it, ManxPower |
| 19:30.47 | ManxPower | The magical command "show applications" and "show application ZapataTeller" are your friends |
| 19:31.07 | molkmin | looks like that's what I am looking for, based on my google search for it |
| 19:31.38 | *** join/#asterisk rwaite (n=richard@rrcs-74-218-125-86.central.biz.rr.com) |
| 19:32.26 | molkmin | thanks, ManxPower, looks like that should do the trick |
| 19:32.40 | RecycleBin | Anyone ? |
| 19:34.36 | Alanonzander | RecycleBin, you looking for 'area codes' ? |
| 19:34.45 | RecycleBin | No, area codes i can find |
| 19:34.58 | RecycleBin | its the 3 digits after the area codes for a province im after |
| 19:35.01 | Alanonzander | then I don't know. Sorry |
| 19:35.24 | RecycleBin | I mite have to buy phone books for provinces i want to call |
| 19:36.24 | [TK]D-Fender | RecycleBin: Why? |
| 19:36.41 | *** join/#asterisk lanning (n=lanning@66.151.128.195) |
| 19:36.57 | RecycleBin | Because i need the prefixe's being used so i can generate the phone numbers |
| 19:37.11 | *** part/#asterisk mtaht4 (n=m@WAN59-246.cablenet.com.ni) |
| 19:37.17 | ManxPower | it might be helpful if you actually specified what country you are interested in. |
| 19:37.22 | *** join/#asterisk rasterix (n=IceChat7@host81-155-2-150.range81-155.btcentralplus.com) |
| 19:37.26 | RecycleBin | Canada, |
| 19:37.28 | [TK]D-Fender | RecycleBin: Oh crap, you're jsut going to spam the entire bloody phone-book? |
| 19:37.38 | jblack | Diamondcard is thinking of selling a did fax service. |
| 19:37.41 | ManxPower | [TK]D-Fender: I new he was a bad kid. |
| 19:37.52 | RecycleBin | Well how else is it done ? |
| 19:38.04 | RecycleBin | isnt that what all theese outbound places are doing ? |
| 19:38.23 | rasterix | evening people |
| 19:38.26 | [TK]D-Fender | RecycleBin: We consider your kind "bad" around here. Unwanted mass callouts = BAD |
| 19:38.30 | ManxPower | RecycleBin: good thing you are not in the USA. We have a do-not-call list that a telemarketer must purchase and if you call a number on that list you get fined $10,000. |
| 19:38.44 | ManxPower | for each call |
| 19:38.54 | rasterix | we have that in the uk manxpower |
| 19:38.59 | rasterix | its called the tps |
| 19:39.05 | RecycleBin | We have do not call lists here aswell. You need to give the respondent a chance to say they do not want to be called and build a list |
| 19:39.07 | km- | canada has even more stringent rules than the US for telemarkting, I thought? |
| 19:39.23 | ManxPower | RecycleBin: in the usa you just register your phone number with the list for free. |
| 19:39.27 | jblack | ManxPower: Did you evacuate yet? |
| 19:39.47 | RecycleBin | I should do some research then |
| 19:39.48 | ManxPower | jblack: the storm is 5 days out. |
| 19:39.52 | RecycleBin | maybe i need to purchase said list |
| 19:40.00 | ManxPower | I'll see what happens tomorrow. |
| 19:40.02 | km- | recyclebin: and to answer your larger question, no, people don't mass-spam the phonebook in telemarketing because it's inefficient. you buy targeted number lists from companies |
| 19:40.23 | ManxPower | don't help him, for dawg's sake! |
| 19:40.30 | rasterix | i can sell u a cloned list recyclebin... although i accept no liability if u call someone on the real list |
| 19:40.41 | rasterix | cloned lists come with no support |
| 19:40.51 | RecycleBin | if its for canada i mite be interested |
| 19:40.57 | km- | manx: hey, if I save one random number the hassle of being annoyed with his shpeel, I have helped, havent I? |
| 19:40.58 | rasterix | yah it is |
| 19:41.06 | RecycleBin | ill keep you in mind |
| 19:41.07 | RecycleBin | Ty |
| 19:41.09 | rasterix | ill make it now with a random number generator |
| 19:41.36 | RecycleBin | Your going to make a joke and say every number available |
| 19:42.08 | RecycleBin | how is buying a list from someone any better then generating your own list |
| 19:42.12 | km- | well, lets analyze. you're in an irc channel full of people who get calls at dinner time how to mass-spam an entire country |
| 19:42.32 | km- | if you don't know that, then you really need to rethink whatever business plan you're coming up with |
| 19:43.20 | RecycleBin | .... |
| 19:43.28 | rasterix | if u get unlimited day time calls i guess u can spam call for free |
| 19:43.44 | *** join/#asterisk ctooley (n=ctooley@209.33.108.195) |
| 19:43.49 | jblack | I'm confused. Which one of them is the jackass worthy of having a coke bottle shoved up their ass? |
| 19:43.50 | [TK]D-Fender | I would personally take down all their info and inform them that if I ever got another call from them, or if my FRIENDS get the same call that I'd be paying a "personal" visit.... |
| 19:43.58 | rasterix | the only real number going on the list i sell recyclebin will be mine :) |
| 19:44.21 | RecycleBin | Cigar time |
| 19:44.31 | RecycleBin | I do appreciate the help thus far. |
| 19:44.33 | ctooley | I've got an Asterisk 1.4.0 system that I "acquired" the other day and after upgrading to 1.4.21.2 all of the calls that AMD is checking seem to be getting detected exactly backwards. |
| 19:44.46 | km- | ponder your business plan over that cigar, sir. And ponder learning more about the business you're trying to enter. |
| 19:45.03 | RecycleBin | Thats a good plan |
| 19:45.06 | RecycleBin | Thankyou for the advice |
| 19:45.28 | jblack | Consider that if some people can find you in real life, they'll physically harm you. |
| 19:46.10 | RecycleBin | Hmm. |
| 19:46.27 | RecycleBin | i hope work has security gaurds cause ive been using my real name |
| 19:46.31 | km- | I setup an ani match in my asterisk box, if the ani comes in as _8XXXXXXXXX it goes directly into a prompt that tells them to hit 1 for a voicemail since they'll never hear from me |
| 19:46.49 | jblack | do those guards follow you home? |
| 19:46.56 | km- | recycle: what company do you work for? |
| 19:47.12 | Alanonzander | Total noobie here... I came in and saw the last bit ogf the help given to molkmin. I guess I need the same help, but I haven't a clue as to how to add it to my config that I inherited :( |
| 19:47.15 | jblack | There's some real nut cases out there. I'd know. |
| 19:47.42 | RecycleBin | i do surveys at work |
| 19:47.43 | km- | jblack: heh, member of that group? :) |
| 19:47.52 | seanbright | jblack: everything you are saying is coming off as veiled threats |
| 19:48.01 | RecycleBin | we call random numbers and try to do surveys for companys |
| 19:48.04 | seanbright | not cool. |
| 19:48.13 | km- | snicker. |
| 19:48.27 | rasterix | jblack: likes to fight... let him have his fun |
| 19:48.57 | RecycleBin | Anyhow later, i hope your bickering has been therapeutic |
| 19:49.03 | Alanonzander | How to I set up this zapatateller ? |
| 19:49.17 | Alanonzander | Google only confused me more :( |
| 19:50.39 | [TK]D-Fender | Alanonzander: You don't. You simply call it from your dialplan |
| 19:52.25 | Alanonzander | how? |
| 19:52.37 | [TK]D-Fender | Alanonzander: vi extensions.conf |
| 19:54.19 | Alanonzander | k |
| 19:55.29 | km- | every time someone asks a question in here I really think that opening up a generic asterisk support company would be a mint |
| 19:55.53 | [TK]D-Fender | km-: Except everyone expects it to be free |
| 19:56.27 | km- | it's a two way transaction; if more people charged for answering questions the expectation of free would depreciate |
| 19:57.26 | _ShrikE | The support is here is excellent as long as you aren't an idiot. I would hate to support the paying customer base. |
| 19:57.50 | km- | the idiots are the ones more likely to pay, though, which would make me happier to help |
| 19:58.15 | km- | alanonzander: if I told you I'd make it work for you for $10, would you do it? |
| 19:59.00 | jblack | ManxPower: I don't think ou have 5 days. I think you have about 3.5 days. |
| 19:59.24 | km- | tech support is like mowing the lawn, sure it sucks, but it's not like you really have to use your brain for it |
| 20:00.36 | *** join/#asterisk nhuisman_work (n=nhuisman@dhcp42.ifa.hawaii.edu) |
| 20:00.47 | nhuisman_work | Does anyone think the new asterisk 2.0 gui is worth playing with? |
| 20:02.04 | aliver | Is ${CALLERID(num)} the person who originated the call? |
| 20:02.19 | jblack | It is until you change it. |
| 20:02.27 | ctooley | nhuisman_work, apparently someone did, they released it |
| 20:02.40 | km- | asterisk has a gui? |
| 20:02.43 | RecycleBin | It would appear i dont have to buy any list, but rather generate my own do not call list |
| 20:02.45 | km- | wait. stupid phrasing |
| 20:02.54 | jblack | km-: Don't you have a lawn to mow? |
| 20:02.55 | RecycleBin | hides |
| 20:03.02 | km- | jblack: haha |
| 20:03.03 | *** join/#asterisk Arck-FR (n=Arck-FR@cvl92-2-82-228-145-232.fbx.proxad.net) |
| 20:03.14 | km- | asterisk has a digium-sanctioned gui? |
| 20:03.31 | Corydon76-dig | km-: two, in fact |
| 20:03.35 | km- | huh. |
| 20:04.02 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 20:04.06 | ctooley | Corydon76-dig, 2? You mean the one in Switchvox? |
| 20:04.13 | Corydon76-dig | ctooley: bingo |
| 20:04.28 | km- | Corydon: haven't chatted with you in a while, how's life? |
| 20:04.36 | nhuisman_work | km-: well i know asterisk be has a gui |
| 20:04.42 | Corydon76-dig | km-: not bad |
| 20:04.43 | nhuisman_work | km-: it's not all that great though |
| 20:04.45 | ctooley | No offense to the asterisk-gui developers, but I like the Switchvox one a lot better. |
| 20:04.51 | ctooley | It lacks some things though |
| 20:05.00 | Corydon76-dig | ctooley: developer |
| 20:05.16 | nhuisman_work | I need something for the dumb people in my building to use to edit the phones |
| 20:05.19 | km- | once you've learned how to do it all in the conf files, the idea of a gui seems a lil kludgey |
| 20:05.22 | nhuisman_work | but I don't want to break my whole config with a gui unless it's good. |
| 20:05.23 | seanbright | i should work on the gui |
| 20:05.40 | km- | but I suppose it would help with greater adoption |
| 20:05.46 | Corydon76-dig | km-: the gui is a direct configuration engine to the config files |
| 20:05.59 | nhuisman_work | Corydon76-dig: direct except for all the features it can't configure. |
| 20:06.01 | *** join/#asterisk elguero (n=elguero@ns1.nashuacs.com) |
| 20:06.16 | Corydon76-dig | nhuisman_work: true enough |
| 20:06.16 | Alanonzander | km- I don't cnotrol the $$$ |
| 20:06.28 | [TK]D-Fender | ctooley: Keep in mind Digium bouth out Switchvox, so they didn't develop it from scratch. It was commercial from the start and not a sample framework like *-GUI. |
| 20:06.43 | Alanonzander | My problem is tha tI don't understand any of this, yet |
| 20:06.43 | Corydon76-dig | At some point, the GUI may harness func_odbc, although that's a little ways off yet |
| 20:06.52 | km- | wait, I'm confused, wasn't switchvox a voicexml ivr platform? |
| 20:06.52 | ctooley | [TK]D-Fender, yeah, I know, was just pointing out the preference |
| 20:07.21 | [TK]D-Fender | Alanonzander: ... |
| 20:07.23 | [TK]D-Fender | ~book |
| 20:07.23 | 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 |
| 20:07.25 | [TK]D-Fender | ^^^^^^^^^^ |
| 20:07.27 | Corydon76-dig | km-: switchvox is generally a PBX administration interface |
| 20:07.39 | [TK]D-Fender | Alanonzander: If you don't like the free download I'll give you DOUBLE your money back! |
| 20:08.02 | Alanonzander | [TK]D-Fender: hehehe |
| 20:08.10 | Alanonzander | [TK]D-Fender: loading now, thanks |
| 20:08.30 | ctooley | Corydon76-dig, I thought the idea was that it was a packaged deal |
| 20:08.35 | Alanonzander | going from 0 to "it has to work" in a day is a pain :/ |
| 20:08.38 | km- | corydon: hmm. I swore some dude came in here a while ago asking for help with getting their IVR platform, which I swore he said was switchvox, to terminate properly to asterisk. |
| 20:08.51 | nhuisman_work | For this time only, get your free The Fututre of Telephony and a verbal thapping from [TK]D-Fender (valued at $20). Also free shipping. |
| 20:08.55 | ctooley | Hardware that is certified, a nice pretty "working" software package, and a user interface on top of it all. |
| 20:09.08 | nhuisman_work | :) |
| 20:09.09 | Corydon76-dig | ctooley: yeah, pretty much |
| 20:09.16 | Alanonzander | Possibly Trixbox ? |
| 20:09.25 | km- | corydon: has bkw been around lately? I haven't talked to him in ages |
| 20:09.36 | km- | ahh, freeswitch. |
| 20:09.54 | Corydon76-dig | km-: he's around, still, but he's left the community and is still rather hostile about it |
| 20:09.59 | km- | I remember him saying "come to freeswitch! we're cooler!" |
| 20:12.20 | km- | I remember the conversation falling short when I asked what freeswitch offered that asterisk didn't |
| 20:12.57 | km- | I'm not a fanboy by any means but when you've got commercial telephony, you dont just opt to swap out backends because you think the name is cooler. |
| 20:13.56 | ctooley | km-, yeah, but Asterisk obviously can't scale very well. :) |
| 20:14.08 | km- | hehe. that's why they invented SER ;) |
| 20:14.27 | ctooley | km-, it was a joke, we've... uh... scaled it quite nicely. |
| 20:14.43 | ctooley | OpenSER has actually been a bigger pain in the butt for us than it is worth. |
| 20:15.00 | ctooley | but... at this point... it's in the production call flow, so it's not coming out. |
| 20:15.02 | km- | has asterisk broken the "can handle more than 500 calls without croaking" barrier? |
| 20:16.03 | km- | when I see a DS3 card i'll truly become a believer |
| 20:16.31 | denon | nothin wrong with jacking in a ds3 via a standalone gateway |
| 20:16.40 | bkruse | oh multiple t1 cards? |
| 20:16.43 | bkruse | or* |
| 20:16.47 | bkruse | I have done it before |
| 20:16.50 | km- | there are 4 port t1 cards |
| 20:16.57 | bkruse | yes... |
| 20:16.58 | km- | but there's a difference between 92 channels and 672 |
| 20:17.01 | denon | nah, I mean ds3 into a sip gateway |
| 20:17.12 | errr | km-: he was just pitching freeswitch to the freepbx devs 10 mins ago :) |
| 20:17.27 | denon | asterisk is perfectly happy to take a ds3 via sip :) |
| 20:17.28 | km- | denon: I've heard that asterisk has issues operating with more than 400 sip calls at once |
| 20:17.36 | bkruse | km-: I had 7 quad span t1's in a machine before |
| 20:17.40 | bkruse | 644 channels, very close. |
| 20:18.22 | km- | denon: when did this change? |
| 20:18.32 | km- | denon: for the longest time it was said chan_sip just couldn't handle that much. |
| 20:18.38 | denon | I didn't say the call numbers changed .. |
| 20:18.39 | [TK]D-Fender | Alanonzander: If it has to go from 0 to production in a day with no learning you shouldn't be doing the job |
| 20:18.44 | denon | just that you can drop any circuit you want into asterisk |
| 20:18.48 | Alanonzander | I inherited it |
| 20:18.57 | Alanonzander | It wasn't part of my job to begin with |
| 20:19.02 | km- | denon: ok, so simultaneous calls are still a problem |
| 20:19.04 | file | km-: have to be specific... chan_sip is handling call setups, not actual audio |
| 20:19.19 | [TK]D-Fender | Alanonzander: My rates are very accessible :) |
| 20:19.20 | km- | file: ah. ok. |
| 20:19.23 | denon | km-: I don't know what the current limitations are .. most of it is dealing with lots of simul call setups I think |
| 20:19.29 | Alanonzander | Very good book you refered me to |
| 20:19.35 | bkruse | [TK]D-Fender: Time for the book, or hide [TK]D-Fender :) |
| 20:19.39 | bkruse | hire* |
| 20:19.42 | denon | but everyone seems to know a different tweak to make their scenario handle an insane number of calls |
| 20:19.50 | DarylVOIP | denon: it is? |
| 20:20.06 | Alanonzander | If the book doesn't help, I'll pester the bean-counters for funds to hiher out :) |
| 20:20.08 | DarylVOIP | I'm lucky to get 250 calls per box if I'm doing anything more than just passing traffic. |
| 20:20.08 | Alanonzander | err hire |
| 20:20.11 | nhuisman_work | how big has anyone gotten a conference call? |
| 20:20.17 | DarylVOIP | forgest it if your transcoding or playing sound files |
| 20:20.21 | *** join/#asterisk legis (i=estar@unaffiliated/legis) |
| 20:20.21 | nhuisman_work | My machine have dual core xeons 3.0ghz |
| 20:20.53 | denon | well, if you're talking about handling DS3s of calls, I hope you're not also planning to transcode them all to g729 or something |
| 20:20.59 | nhuisman_work | kind of curious how many my max calls in the conference would be |
| 20:21.02 | bkruse | denon: exactly |
| 20:21.14 | legis | How can I tell if my asterisk box is registering to my ITSP? |
| 20:21.26 | DarylVOIP | I use IMGs for that mostly. |
| 20:21.28 | Maliuta | legis: sip show registry |
| 20:21.28 | bkruse | unless you have... (counts in head) 16 PCI slots and a hell of a box with a new and revolutionary pci bus :D |
| 20:21.31 | DarylVOIP | (the transcoding) |
| 20:21.45 | bkruse | legis: sip show registry (if it's sip) |
| 20:21.51 | nhuisman_work | bkruse: I think at that point you start using an external box |
| 20:21.53 | legis | Maliuta: thanks |
| 20:22.01 | legis | bkruse: yeah sip thx |
| 20:22.32 | bkruse | nhuisman_work: Indeed. |
| 20:22.34 | bkruse | legis: np |
| 20:22.51 | bkruse | Maliuta: ahh you beat me |
| 20:22.58 | bkruse | jbot: Maliuta++ |
| 20:23.17 | *** join/#asterisk Gat0rvean (n=gredish@64.191.128.145) |
| 20:23.46 | Maliuta | bkruse: I just happened to be looking at the channel at that moment |
| 20:23.59 | Maliuta | bkruse: and it helps that I can type fairly fast :) |
| 20:24.02 | [TK]D-Fender | ~karma |
| 20:24.02 | jbot | [tk]d-fender has karma of 10 |
| 20:24.10 | [TK]D-Fender | ~karmakarma |
| 20:24.11 | jbot | Karma Chameleon! |
| 20:24.11 | bkw_ | looks around |
| 20:24.13 | [TK]D-Fender | :D |
| 20:24.20 | legis | calls work but i don't see anything in sip show registry |
| 20:24.30 | [TK]D-Fender | bkw_: My karma ran over your dogma :p |
| 20:24.40 | [TK]D-Fender | legis: Calls IN? |
| 20:24.44 | DarylVOIP | are you actually registering? That's not actually a requirement to send a call, depnding on how the far end is set up. |
| 20:24.56 | [TK]D-Fender | Its almost NEVER required to send |
| 20:25.02 | bkw_ | [TK]D-Fender: thats ok... I won't get mad.. |
| 20:25.37 | *** join/#asterisk steliosk (n=Stelios@79.107.52.223) |
| 20:25.42 | legis | [TK]D-Fender: calls out, the ITSP is doing the termination |
| 20:25.45 | DarylVOIP | depends on who you are send to :) It is on my network. |
| 20:25.58 | [TK]D-Fender | legis: That does not normally depend on being registered |
| 20:26.00 | [TK]D-Fender | ~sipregister |
| 20:26.01 | jbot | [~sipregister] SIP Registration is to tell your provider as to what IP address & EXTEN to send INCOMING calls to. Some ITSP's let you use a fixed IP or host rather that register. Registration is NOT normally needed to PLACE a call, as those are typically auth'd independently. Others accept UN-AUTH'D calls once you are registered (saves on negotiation BW) |
| 20:26.02 | [TK]D-Fender | ^^^^^^^^^^ |
| 20:26.25 | baliktad | I narrowed down a memory leak on my system to *, if I keep * running the system loses about 1MB per hour (until it hard hangs and I have to reboot) |
| 20:26.32 | baliktad | How can I figure out what's causing the leak? |
| 20:26.49 | legis | [TK]D-Fender: oh ok |
| 20:26.50 | DarylVOIP | Sounds like its working just fine ;) |
| 20:27.14 | jblack | baliktad: Look at valgrind. |
| 20:27.15 | [TK]D-Fender | ok, checkout time. Off to the gym. BBIAB |
| 20:27.33 | legis | can I make calls from asterisk CLI? |
| 20:27.45 | bkw_ | legis: I think their is an originate cli in 1.4 |
| 20:27.47 | Qwell | people in the press REALLY need to stop putting words in my mouth |
| 20:28.10 | jblack | USA Today; Qwell says press needs to STFU |
| 20:28.17 | denon | Qwell: "No, I really don't want a newspaper subscription!" |
| 20:28.17 | nhuisman_work | um does anyone know how to reset the password for the asterisk be gui? |
| 20:28.22 | legis | bkw_: thx |
| 20:28.29 | nhuisman_work | I thought it would be the same as my user account |
| 20:28.52 | bkw_ | Qwell: did someone misquote you? |
| 20:29.05 | Qwell | I didn't *say* anything. :) |
| 20:29.15 | Qwell | like I said - words are being put in my mouth |
| 20:29.52 | Qwell | misinterpreting my motives, I guess |
| 20:30.13 | x86 | what's the channel variable for inbound channel? for example, how could i determine the calling channel? |
| 20:30.42 | jblack | Qwell says the press a pile of slandering, misquoting bigots. |
| 20:30.43 | bkw_ | Qwell: examples? |
| 20:30.49 | *** part/#asterisk my007ms (i=master@217.139.17.150) |
| 20:31.43 | x86 | would ${CHANNEL} contain the inbound channel? |
| 20:31.56 | jblack | Woot. It was bound to happen. "Ohio Jury convicts mom in microwaved baby case" |
| 20:32.15 | jblack | "Prosecutors said Arnold intentionally put her baby in the microwave oven and cooked the child to death after a fight with her boyfriend" |
| 20:32.30 | ctooley | jblack, that's just disturbing. |
| 20:32.46 | *** join/#asterisk LoOoD (n=gman@64.201.247.2) |
| 20:32.50 | jblack | For the record, cooking babies in microwaves is just plain wrong. They end up mushy. Next time, use the broiler. |
| 20:34.06 | *** join/#asterisk MikeJ (n=MikeJ@freeswitch/developer/mikej) |
| 20:34.18 | km- | now that I have a newborn that story is even more disturbing for me |
| 20:34.35 | nhuisman_work | um does anyone know how to reset the password for the asterisk gui? |
| 20:35.15 | DarylVOIP | I don't think anyone knows what GUI you are talking about |
| 20:35.40 | *** join/#asterisk jtodd (i=_squid@c-76-27-193-118.hsd1.or.comcast.net) |
| 20:35.43 | nhuisman_work | DarylVOIP: probably not, the one with asterisk be is what I have, but I assume the asterisknow one is the same thing so if anyone knows for either one that would work. |
| 20:35.48 | DarylVOIP | What GUI are you using? |
| 20:36.07 | nhuisman_work | the one that comes with asterisk business edition, which is the same one from asterisknow. |
| 20:36.33 | nhuisman_work | i'm not actually using it, just wanted to take a look at it and see if what features were missing. |
| 20:36.51 | nhuisman_work | oh well i'll just call digium |
| 20:37.07 | DarylVOIP | I haven't use that one....so I don't know. |
| 20:40.19 | *** join/#asterisk AlexTO (n=alex@201.228.24.214) |
| 20:43.37 | *** join/#asterisk Levonk (n=lk@adsl-75-62-133-224.dsl.lsan03.sbcglobal.net) |
| 20:45.37 | *** join/#asterisk UnixDawg (n=unixdawg@181.128.204.68.cfl.res.rr.com) |
| 21:03.13 | *** join/#asterisk Levonk (n=lk@76.243.64.113) |
| 21:04.37 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 21:06.48 | x86 | wow sure is quiet in here this afternoon |
| 21:06.55 | x86 | must be smooth sailing :) |
| 21:07.10 | x86 | either that or everyone is taking off early for the long weekend |
| 21:08.33 | jblack | x86: Well, I'm watching a movie. |
| 21:10.25 | Maliuta | it's already the weekend, its 7am saturday |
| 21:10.27 | MikeJ | everyone is .... off sailing :D |
| 21:10.44 | MikeJ | its a 3 day weekend in the us too... |
| 21:11.22 | jblack | wonders if the zombie will eat the nun |
| 21:21.07 | *** join/#asterisk moetop (n=moetop@66-227-253-88.dhcp.bycy.mi.charter.com) |
| 21:21.10 | [intra]lanman | jblack: isn't it a sin to eat a nun? |
| 21:23.45 | _ShrikE | mmmm nuns... |
| 21:26.15 | moetop | I am getting errors when I try and start asterisk. |
| 21:26.31 | moetop | <PROTECTED> |
| 21:26.42 | moetop | About 5 of those.. it looks like it is parsing comment lines, or error text. and then this on |
| 21:26.54 | moetop | WARNING[13650]: manager.c:3117 init_manager: Unable to bind socket: Address already in use |
| 21:26.55 | *** join/#asterisk nicoAMG (i=asgalt@201.203.96.42) |
| 21:27.50 | moetop | I do a netstat and I dont see anything on the ports it should be using... |
| 21:28.39 | *** join/#asterisk Maliuta (n=nikolai@kiev.lusan.id.au) |
| 21:38.39 | x86 | moetop: well what protocol are you looking at in netstat? |
| 21:39.07 | x86 | moetop: netstat -anp --inet6 | grep 5038 |
| 21:39.12 | x86 | that should show it |
| 21:39.21 | x86 | as well as the PID of the process bound to that port |
| 21:39.40 | x86 | hmm, actually, it is an ipv4 socket |
| 21:39.49 | x86 | netstat -anp --inet | grep 5038 |
| 21:39.59 | moetop | Thanks.. I found it.. one of my .config files was messed up |
| 21:40.28 | moetop | Now I cant get to the GUI, but that's for another channel.. :) |
| 21:41.45 | rasterix | if i have an application like record(${mypath}myfilenamestart,myfilenamend,5,60) how do i escape the first comma if it is part of the filename? |
| 21:44.37 | x86 | rasterix: dude, dont use commas in filenames |
| 21:44.44 | x86 | rasterix: use hyphens if anything |
| 21:44.47 | x86 | or underscores |
| 21:44.50 | x86 | but never commas |
| 21:44.54 | rasterix | x86: its just an example |
| 21:45.09 | rasterix | how do i escape the comma? |
| 21:45.21 | rasterix | perhaps filenames was a bad choice of example |
| 21:45.25 | jblack | doctor: It hurts when i stick my fingers in electrical sockets. |
| 21:45.27 | Qwell | the same way you escape any other char in any other language |
| 21:45.40 | rasterix | i just \ |
| 21:45.40 | x86 | the point is, you should choose a character that you dont have to escape :) |
| 21:45.42 | rasterix | ? |
| 21:45.57 | Qwell | yes |
| 21:46.05 | rasterix | qwell: do you go out of your way to be unhelpful sometimes? |
| 21:46.36 | jblack | rasterix: He seemed helpful to me. How do you escape characters in bash, sed, grep, and C? |
| 21:47.43 | *** part/#asterisk Deeewayne (n=Deeewayn@nat/digium/x-89b25ede741d2819) |
| 21:47.46 | jblack | rasterix: With a backslash. |
| 21:47.53 | jblack | goes back to his zombie movie |
| 21:48.41 | Pimpachu | Anyone see why this would not add the Diversion header to my *72 calls: |
| 21:48.42 | Pimpachu | http://pastebin.ca/1188894 |
| 21:53.39 | *** join/#asterisk nix8n82 (n=nate@63.162.28.92) |
| 21:54.49 | kaldemar | Pimpachu: definitely not. is that a freepbx macro? |
| 21:55.50 | Pimpachu | kaldemar, Everything except the sipAddheader() lines |
| 21:56.23 | kaldemar | freepbx is not supported here, they have their own channel, #freepbx. |
| 21:56.57 | Pimpachu | Right |
| 21:57.40 | Pimpachu | But the macro should work like that to add the header |
| 21:57.53 | Pimpachu | Perhaps I am putting it on the wrong line |
| 21:58.05 | kaldemar | if it doesn't, then it obviously shouldn't. |
| 21:58.43 | *** join/#asterisk outtolunc (n=me@c-67-170-211-86.hsd1.ca.comcast.net) |
| 21:59.12 | *** join/#asterisk sah-work (n=Bawbatos@65-119-47-100.dia.static.qwest.net) |
| 22:00.17 | Pimpachu | >_< |
| 22:02.13 | *** join/#asterisk Alanonzander (n=azander@baghdad.netonecom.net) |
| 22:02.14 | kaldemar | i don't want to debug your problem any further, but what are you trying to achieve with the header as your not dialing anywhere with that code? |
| 22:04.25 | Alanonzander | came back to say thank you. I got it working, and now I can take the time to do some learning. |
| 22:04.31 | Alanonzander | Thanks again! |
| 22:05.25 | Pimpachu | My voip provider requires I add a diversion header to my sip messages when doing a call forward |
| 22:05.46 | Pimpachu | So what that attemps to do is when a person does *72 it will include the diversion header |
| 22:06.59 | kaldemar | that macro is just setting a value to asterisk's database, you can't use it to add sip headers anywhere. |
| 22:07.17 | *** part/#asterisk theHub (n=theHub@69.177.93.21) |
| 22:08.12 | kaldemar | go ask freepbx people how to do it. you'll get more help there. |
| 22:14.17 | *** join/#asterisk mvanbaak_ (n=michiel@asterisk/contributor-and-bug-marshal/mvanbaak) |
| 22:14.47 | *** join/#asterisk AlexTO (n=alex@201.228.24.214) |
| 22:18.17 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 22:18.27 | *** join/#asterisk VJFROMGT (n=vjfromgt@pool-96-232-5-84.nycmny.east.verizon.net) |
| 22:18.37 | VJFROMGT | can someone tell me what the following means |
| 22:18.57 | VJFROMGT | [WARNING] Unknown 192.168.20.91 ( this trunk works ) |
| 22:19.39 | VJFROMGT | sorry it should eb |
| 22:19.41 | VJFROMGT | [WARNING] Unknown host: '192.168.20.91' |
| 22:21.10 | ManxPower | VJFROMGT: Is that a copy/paste or did you retype the message? |
| 22:21.11 | [netman] | does 192.168.20.91 mean something for you? |
| 22:21.49 | VJFROMGT | it s a trunk which works fine |
| 22:21.52 | aliver | Can someone get jbot to tell me about the GCC > 4.1 + GSM bug? |
| 22:22.07 | ManxPower | VJFROMGT: then ignore the message |
| 22:22.08 | VJFROMGT | its a copy paste,, let me paste enire line |
| 22:22.21 | VJFROMGT | Fri Aug 29 22:19:29 GMT 2008: [WARNING] Unknown host: '192.168.20.91 |
| 22:22.49 | Qwell | that warning appears nowhere in 1.4 |
| 22:23.07 | VJFROMGT | 20.91 is a trunk, that trunk is working |
| 22:23.14 | VJFROMGT | sip |
| 22:23.20 | ManxPower | VJFROMGT: Then ignore the message. |
| 22:23.29 | Qwell | nor does it appear in 1.6 |
| 22:23.55 | VJFROMGT | i have been, just curios why its there |
| 22:24.03 | Qwell | VJFROMGT: where exactly are you seeing it? |
| 22:24.14 | VJFROMGT | set verbose 9 |
| 22:24.22 | Qwell | on what? |
| 22:24.27 | ManxPower | VJFROMGT: You must be running 1.2.x |
| 22:24.43 | Qwell | doesn't exist in 1.2 either |
| 22:24.45 | ManxPower | or maybe you are running an Asterisk GUI |
| 22:24.55 | VJFROMGT | 1.2 |
| 22:25.09 | ManxPower | VJFROMGT: there is no 1.2. Is that 1.2.0, 1.2.1, 1.2.2? |
| 22:25.09 | *** join/#asterisk Levonk (n=lk@adsl-75-62-138-233.dsl.lsan03.sbcglobal.net) |
| 22:25.37 | ManxPower | Are you really that secret about your version number? |
| 22:25.48 | ManxPower | "show version" in the CLI will give you the version number. |
| 22:25.50 | VJFROMGT | 1.2.0 |
| 22:25.55 | Qwell | ... |
| 22:26.00 | ManxPower | VJFROMGT: We can't help you then. |
| 22:26.09 | ManxPower | You need to be running at least something recent. |
| 22:26.29 | VJFROMGT | i know,, just wondering ,,, |
| 22:26.39 | ManxPower | VJFROMGT: the thing is, as Qwell said, that message does not exist in the Asterisk source code. |
| 22:26.56 | ManxPower | So either you are on drugs or you are not running a stock compiled from source Asterisk |
| 22:27.00 | Qwell | Doesn't exist in the 1.2.0 source either. |
| 22:27.36 | ManxPower | My be is non-stock Asterisk |
| 22:27.40 | ManxPower | bet that is |
| 22:27.41 | *** part/#asterisk VJFROMGT (n=vjfromgt@pool-96-232-5-84.nycmny.east.verizon.net) |
| 22:27.48 | RecycleBin | . |
| 22:28.17 | Qwell | google says he's a trixbox user |
| 22:29.00 | *** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk) |
| 22:29.02 | ManxPower | What an evil horrid person |
| 22:29.22 | RecycleBin | Whats wrong with trixbox |
| 22:29.45 | ManxPower | ~trixbox |
| 22:29.45 | jbot | hmm... trixbox is a full linux distro that includes FreePBX and other 3rd party add-ons. It is all this extra stuff which makes trixbox VERY difficult to support, and is not supported in #asterisk. Try asking in #trixbox or on their forums & wiki at http://www.trixbox.org/. We do not recommend using it. |
| 22:29.57 | RecycleBin | Oh |
| 22:30.08 | ManxPower | The horrid part is because he never said he was using Trixbox |
| 22:30.14 | RecycleBin | Is asteriskNow supported ? |
| 22:30.26 | ManxPower | RecycleBin: Same thing. |
| 22:30.29 | RecycleBin | Oh |
| 22:30.43 | RecycleBin | Are they setting up asterisk in a non standard way ? |
| 22:30.43 | *** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194) |
| 22:30.58 | ManxPower | yes!!! |
| 22:31.03 | ManxPower | That is the PROBLEM |
| 22:31.03 | RecycleBin | That sux |
| 22:31.20 | [netman] | ~elastix |
| 22:31.21 | jbot | i guess elastix is a full linux distro that includes FreePBX and other 3rd party add-ons. It is all this extra stuff which makes Elastix VERY difficult to support, and is not supported in #asterisk. Try asking in #Elastix or on their forums at http://www.elastix.org |
| 22:32.14 | RecycleBin | It's unfortunate they are doing it that way, in theory a asterisk distro isnt such a bad idea if they were setting up asterisk in a standard way |
| 22:32.57 | [TK]D-Fender | There is no such thing as "setting Asterisk up in a standard way". |
| 22:37.02 | ManxPower | The Asterisk GUIs (with the possible exception of AsteriskNOW/Asterisk GUI) have such complex dialplans and AGIs that cause hundreds of lines of CLI output just for 1 call. |
| 22:38.32 | *** join/#asterisk angryuser (n=angryuse@54.244.146.195.dynamic.adsl.abo.nordnet.fr) |
| 22:39.50 | RecycleBin | Sounds like a waste of good CLI real estate |
| 22:45.22 | [netman] | somebody should make a tool to make easier the debugging of such dialplans |
| 22:46.16 | ManxPower | [netman]: That would be deleting the config files and creating them from scratch |
| 22:46.26 | ManxPower | The thing is, that would not solve the problem. |
| 22:46.42 | grandpapadot | [netman] There is one -> rm /etc/asterisk/extensions.conf |
| 22:49.44 | [netman] | grandpapadot: lol |
| 22:50.22 | *** join/#asterisk [hC] (n=hardcore@S01060016b6b53c0c.vc.shawcable.net) |
| 23:12.50 | *** join/#asterisk MrNaz (n=naz@ppp118-208-174-70.lns10.mel4.internode.on.net) |
| 23:16.12 | *** join/#asterisk grantm (n=grant@68.142.138.4) |
| 23:17.51 | nhuisman_work | is users.conf only for the asterisk gui users? |
| 23:17.55 | nhuisman_work | seems like a copy of sip.conf |
| 23:18.22 | Qwell | anybody can use it |
| 23:18.26 | fakhir | users.conf is used by the GUI but anyone can use it |
| 23:18.39 | Qwell | it replaces sip.conf, iax.conf, etc |
| 23:18.42 | nhuisman_work | ah |
| 23:18.54 | Qwell | "replaces".. obviously the others are still available |
| 23:18.58 | nhuisman_work | so what if you have stuff in more then one file? is one higher |
| 23:19.04 | nhuisman_work | say you have the same # |
| 23:19.15 | Qwell | it would be like listing it twice in sip.conf |
| 23:19.23 | nhuisman_work | I forget, what does that do? |
| 23:19.27 | Qwell | not sure :p |
| 23:19.29 | nhuisman_work | hehe |
| 23:19.44 | nhuisman_work | I'm pondering switching to the asterisk gui because i'm leaving the company soon and I doubt they will figure out the conf files |
| 23:19.45 | rasterix | lol |
| 23:20.30 | angryuser | good night @all |
| 23:21.00 | rasterix | nhuisman: leave them the conf files and charge through the nose when they need u back? |
| 23:21.23 | nhuisman_work | rasterix: heh, they would probably just hire digium or something at that point |
| 23:21.50 | nhuisman_work | rasterix: the problem is it is a university, I'm not sure it's very easy for them to hire someone for that kind of thing. |
| 23:22.00 | rasterix | ahhh ok |
| 23:22.06 | nhuisman_work | rasterix: mmm leave myself a backdoor, rape the config and then get paid |
| 23:22.07 | nhuisman_work | good idea ;P |
| 23:22.32 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 23:22.34 | nhuisman_work | "do no evil!" |
| 23:22.36 | rasterix | hey universities have a good budget... why not :) |
| 23:23.40 | rasterix | dont feel you are ripping them off... remember how much money they are saving by using asterisk in the first place! |
| 23:23.44 | *** join/#asterisk deeperror (n=deeperro@d149-67-49-94.try.wideopenwest.com) |
| 23:23.47 | nhuisman_work | laugh |
| 23:23.51 | nhuisman_work | yeah i think I saved them 20-30k |
| 23:23.58 | rasterix | exactly |
| 23:24.11 | rasterix | so basically they owe u 20k in contracting fees |
| 23:24.16 | nhuisman_work | laugh |
| 23:24.24 | nhuisman_work | boy remind me never to hire you :p |
| 23:24.28 | nhuisman_work | or work for you |
| 23:24.31 | nhuisman_work | evil evil evil |
| 23:24.33 | nhuisman_work | ;) |
| 23:25.33 | rasterix | if its a university they should have someone that can get to grips with asterisk fairly quickly |
| 23:25.43 | nhuisman_work | yeah maybe |
| 23:25.49 | rasterix | if you want to be nice |
| 23:26.02 | rasterix | personally id leave them with a card and my rates |
| 23:26.04 | rasterix | :) |
| 23:26.05 | nhuisman_work | depends on how much I convolute the config before I leave |
| 23:26.39 | *** join/#asterisk Levonk (n=lk@adsl-76-237-14-84.dsl.lsan03.sbcglobal.net) |
| 23:27.00 | kd8ikt | sounds more like extorsion now |
| 23:27.16 | nhuisman_work | heh |
| 23:27.18 | rasterix | why not just agree a support contract should they need you after you leave? |
| 23:27.26 | rasterix | nothing dishonest in that |
| 23:27.28 | nhuisman_work | yeah I wouldn't do anything like that |
| 23:27.38 | nhuisman_work | I want a good job reference |
| 23:28.08 | *** join/#asterisk juniowww (i=juniowww@189.4.59.201) |
| 23:28.23 | kd8ikt | i'd like to boast high uptimes without ever really having to mess with it again |
| 23:28.26 | rasterix | leaving them a support option IS good |
| 23:28.34 | kd8ikt | that should be reference enough |
| 23:28.47 | nhuisman_work | rasterix: I think the option i'm leaving is digiums contracting number |
| 23:28.59 | rasterix | that will cost them more |
| 23:29.07 | rasterix | you should feel bad |
| 23:29.14 | rasterix | peoples education will suffer |
| 23:29.17 | nhuisman_work | laugh |
| 23:29.21 | rasterix | because you dont want to offer support |
| 23:29.23 | rasterix | EVIL |
| 23:30.08 | rasterix | im thinking the correct thing to do is leave them your number for support at $200 per hour |
| 23:30.22 | rasterix | you know the systems so its probably still cheaper than getting digium in |
| 23:30.38 | rasterix | remote support only of course |
| 23:30.44 | nhuisman_work | I think I mentioned to them that I will be available for remote support contracting |
| 23:30.46 | rasterix | your not a charity after all |
| 23:30.47 | jblack | rasterix: Can you predict how many more days it will be before you get tired of this channel and go elsewhere? |
| 23:31.05 | rasterix | jblack: i think ill stay all the time im annoying you |
| 23:31.13 | *** join/#asterisk cmantito (n=gphreak@pool-72-73-228-3.cmdnnj.fios.verizon.net) |
| 23:32.00 | jblack | annoy me? You're entertaining me. |
| 23:32.08 | rasterix | you entertain me too |
| 23:32.12 | rasterix | so its all good |
| 23:32.36 | rasterix | your like a small child defending their patch... its so cute |
| 23:33.59 | jblack | my what is like a small child? |
| 23:34.19 | rasterix | im tempted to say intellect... but ill refrain |
| 23:34.20 | *** part/#asterisk beek (n=klinebl@65.211.106.242) |
| 23:34.28 | jblack | You can if you wish. |
| 23:34.44 | jblack | You have my personal permission to call me a moron. |
| 23:35.10 | Qwell | jblack: moron |
| 23:35.16 | jblack | You don't. :) |
| 23:35.23 | Qwell | lame! |
| 23:35.30 | jblack | That's life. unfair all around |
| 23:35.50 | Qwell | figured it was a global permission ;) |
| 23:35.59 | jblack | I run linux, not windows! |
| 23:36.51 | bkruse | Qwell: Power in Use 2178W 2384W :X |
| 23:36.53 | jblack | By the way, Flight of the living dead is great. It's Zombies.. on planes! |
| 23:36.57 | Qwell | !!! |
| 23:36.58 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 23:37.09 | Qwell | bkruse: that's insane |
| 23:37.19 | Qwell | all 14? |
| 23:37.21 | bkruse | Qwell: Totally, and when I restarted all from the management interface _ wow _ |
| 23:37.22 | bkruse | yes |
| 23:37.31 | Qwell | UPS dying yet? |
| 23:37.37 | bkruse | no UPS, couldn't handle it |
| 23:37.40 | Qwell | haha |
| 23:37.41 | bkruse | straight circuit lol |
| 23:37.41 | jblack | bkruse: That's enough power for a small home. |
| 23:37.46 | Qwell | that does not surprise me |
| 23:38.00 | bkruse | jblack: Seriously, I had to hook a server up right under it, and my face was burning, literally, it was red lol |
| 23:38.07 | denon | man, how disappointing |
| 23:38.13 | denon | this motorola sip phone requires avaya |
| 23:38.13 | bkruse | kpfleming laughed at me |
| 23:38.19 | denon | they claim it'll only work with avaya |
| 23:38.35 | Qwell | bkruse: how loud is it in there now? |
| 23:38.41 | denon | (and it's not sip yet, apparently, in a couple'a firmwares it will be . .for now, just h323 .. but even the sip will only be avaya) |
| 23:38.46 | bkruse | Qwell: ehh, fans are down to 40%, so it's not bad |
| 23:38.46 | Qwell | I'm gonna have to make a quick stop on my way out |
| 23:38.49 | bkruse | it got very warm though |
| 23:38.49 | Qwell | ahh |
| 23:38.55 | Qwell | the room did? |
| 23:38.58 | bkruse | lol yes |
| 23:39.00 | Qwell | wow |
| 23:39.04 | bkruse | bladecenter VS AC |
| 23:39.10 | Qwell | WTB more AC |
| 23:39.54 | *** join/#asterisk jpcansa (n=jpbenavi@190.10.2.87) |
| 23:39.58 | nhuisman_work | Qwell: me too |
| 23:40.13 | nhuisman_work | Qwell: we just got a new 16 node cluster with 2x quad core per node, had to turn off other computers |
| 23:40.20 | nhuisman_work | stupid slow ass ac vendors |
| 23:40.27 | bkruse | Qwell: totally |
| 23:40.33 | bkruse | jeffery is going to _flip_ lol |
| 23:41.00 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
| 23:42.30 | jpcansa | HI, i dont know if i´m missing something, i just installed AsteriskNOW, i´m on the AsteriskNOW Console Menu, and it is pointing me to an IP Address to configure the system, i put that address on my browser and i get nothing. any ideas?? |
| 23:42.54 | nhuisman_work | jpcansa: ip:8088 |
| 23:43.04 | nhuisman_work | it uses a nonstandard http port. |
| 23:43.08 | nhuisman_work | at least on asterisk be it does. |
| 23:43.34 | bkruse | jpcansa: asterisk -rx 'http show status' |
| 23:45.27 | jpcansa | nu such command 'asterisk -rx' |
| 23:45.49 | jpcansa | *no |
| 23:46.39 | ManxPower | Have you considered asking on the AsteriskNOW channel |
| 23:46.48 | jpcansa | ip:8088 not working either |
| 23:48.12 | nhuisman_work | jpcansa: check manager.conf and see what port it's bound to, and make sure it's enabled. |
| 23:48.17 | *** part/#asterisk macli (n=macli@nmc.brc.ubc.ca) |
| 23:51.41 | *** join/#asterisk InHisName (n=rich@c-71-225-221-149.hsd1.pa.comcast.net) |
| 23:52.24 | InHisName | I got a wayward process showing in "sip show channels" how do I end/kill it ? |
| 23:52.36 | jpcansa | nhuisman: can i access manager.conf from the * console?? |
| 23:54.05 | InHisName | jpcansa: I access from browser or ssh. * console==CLI> prompt I assume. |
| 23:55.28 | jpcansa | Inhisname: i got asteriskNOW, it give me access only to * console |
| 23:56.13 | InHisName | jpcansa: NO gui interfaces? I thought that was lotta gui interfaces. I am just plain vanilla asterisk. |
