| 00:00.18 | gushi | Hey there...Silly question, how can I route calls from a given number to a fast-busy, without having to do it for every extension? |
| 00:00.41 | Qwell | gushi: You could try doing something like: |
| 00:01.00 | Qwell | exten => _NXXNXXXXXX/3105551212,1,Busy() |
| 00:01.09 | Qwell | where 310... is the number you want to"block" |
| 00:01.26 | *** join/#asterisk dkirker (n=dkirker@pcp063419pcs.wireless.calpoly.edu) |
| 00:01.35 | Linuturk | bah |
| 00:01.38 | Linuturk | I forgot sip.conf |
| 00:01.41 | Linuturk | is silly |
| 00:02.39 | TJNII | Don't you hate it when you mis something obvious? I spent 5 hours debugging a program the other night only to find I had botched 1 (One !!) character in a register assignment. |
| 00:02.41 | *** join/#asterisk xuser (n=xuser@unaffiliated/xuser) |
| 00:03.39 | Destroy | use winmerge ;P |
| 00:04.18 | Destroy | but don't use if your coding something custom only from samples |
| 00:04.24 | Destroy | lol my bad ;P |
| 00:04.48 | Katty | oh man, that apple was amazing. |
| 00:05.29 | *** part/#asterisk dkirker-mobile (n=dkirker@pcp063419pcs.wireless.calpoly.edu) |
| 00:06.44 | Linuturk | TJNII: yeah, I've been sweating over vimdiff for a bit now |
| 00:06.45 | Linuturk | lol |
| 00:09.04 | *** join/#asterisk dkirker (n=dkirker@pcp063419pcs.wireless.calpoly.edu) |
| 00:11.24 | *** join/#asterisk niekie (i=quasselc@dreamworld.bergnetworks.com) |
| 00:12.41 | *** join/#asterisk jmworx___ (n=jeval@mail.octasic.com) |
| 00:15.11 | *** join/#asterisk manxpower (n=ewieling@24.42.221.26) |
| 00:16.10 | *** join/#asterisk ZPertee (n=ZPertee@rrcs-74-219-221-11.central.biz.rr.com) |
| 00:16.35 | ZPertee | FYI if there is anyone out there with a google wave invite |
| 00:16.46 | Jumpie | i'd like one too |
| 00:16.47 | Jumpie | :d |
| 00:16.59 | ZPertee | i'll trade for google voice invite |
| 00:17.25 | *** join/#asterisk supa_disko (n=bleh@secure27.lnk.telstra.net) |
| 00:17.30 | Jumpie | hheh |
| 00:17.34 | Jumpie | i dont have either..i fail |
| 00:18.06 | ZPertee | :-( I've been using google voice since before google owned it |
| 00:18.07 | *** join/#asterisk hardwire (n=spencers@216-67-98-253.static.acsalaska.net) |
| 00:21.36 | manxpower | ~answers |
| 00:21.37 | infobot | i heard answers is Asterisk Book: http://astbook.asteriskdocs.org & SIP w/ NAT: http://www.aocomputing.net/?p=3 & Wiki: http://www.voip-info.org & docs: /path/to/src/asterisk/doc & 1.4 .vs. 1.6.0 .vs. 1.6.1: : /path/to/src/asterisk/UPGRADE*.txt |
| 00:23.18 | manxpower | I'll stick to VitelityVoice |
| 00:36.12 | *** join/#asterisk [netman] (n=netman@160.Red-88-23-80.staticIP.rima-tde.net) |
| 00:38.09 | p3nguin | zpertee: Is there anything I can do with the phone number they give me other than forward it to another number? |
| 00:39.08 | ZPertee | p3nguin, using gizmo you can make and receive unlimited US calls through your * box for starters |
| 00:39.19 | ZPertee | p3nguin, also had free SMS |
| 00:40.49 | manxpower | ~free |
| 00:40.50 | infobot | somebody said free was stuff might take awhile to get done, or http://wiki.maemo.org/Why_the_closed_packages |
| 00:41.02 | manxpower | Ok. That is not what I expected |
| 00:41.08 | manxpower | ~ygwypf |
| 00:41.09 | infobot | ygwypf is, like, You Get What You Pay For. If the sole factor in your decision to purchase a product or service is that it's cheaper than everything else out there, don't be surprised if it's also worse in every other respect than everything else out there. |
| 00:41.20 | manxpower | That's what I was looking for |
| 00:41.46 | *** join/#asterisk nix8n82 (n=nate@63.162.27.14) |
| 00:42.40 | manxpower | *grumble* I really need to QoS my router. |
| 00:42.44 | p3nguin | zpertee: Without gizmo, I can't connect it with Asterisk? |
| 00:42.55 | nix8n82 | Does anyone have experience with asterisk in a cloud environment especially amazon EC2? |
| 00:44.05 | ZPertee | p3nguin, there may be other ways I'm not sure. What I do is insert the settings on gizmo's website and then create a sip trunk to gizmo. works well and painlessly |
| 00:44.20 | ZPertee | p3nguin, and free I might add |
| 00:44.34 | ZPertee | p3nguin, google has yet to allow direct sip access |
| 00:45.31 | p3nguin | That's what I _really_ want. |
| 00:45.34 | mchou | ZPertee: yeah, that's really a shame |
| 00:46.02 | p3nguin | Their crappy forward requirement doesn't work well with my IVR. |
| 00:46.30 | mchou | you can try sipgate.com |
| 00:46.41 | nix8n82 | my real question is do you have to have an public ip for each instance for you to make calls or is it possible for you to make calls from like 10 instances to one or two voip providers while only using one public ip? |
| 00:46.57 | mchou | I doubt it would work any better but YMMV |
| 00:47.34 | TJNII | nix8n82: That is going to wreak havoc with your sip registrations. |
| 00:47.46 | mchou | nix8n82: what do you mean? That's what asterisk is for |
| 00:48.00 | p3nguin | If you can control the NAT setting and you have private IP addresses, you might be able to get around it. |
| 00:48.01 | russellb | or use a proxy ... |
| 00:48.59 | mchou | get around what? |
| 00:49.03 | manxpower | GOOD nat routers will keep track of multiple phones behind the same nat all talking to the same asterisk server. |
| 00:49.08 | mchou | there is nothing to get around |
| 00:49.19 | manxpower | much like multiple web browsers behind the same nat all talking to the same web server. |
| 00:49.25 | mchou | asterisk allows to connect to mutiple trunks |
| 00:49.26 | TJNII | I think he means the server is going to be hopping IPs on some cloud hosting setup. |
| 00:49.52 | mchou | multiple* |
| 00:49.53 | manxpower | TJNII: Aye! That would complicate things. |
| 00:50.06 | TJNII | All depends on how frequently is hops, I guess. |
| 00:50.45 | russellb | a lot of people use Asterisk on EC2 and other virtualized environments, with great success |
| 00:51.29 | TJNII | As long as it re-registers when it hops and doesn't hop during a call it should be fine. |
| 00:51.36 | TJNII | (AFAIK) |
| 00:51.47 | Katty | peeks in |
| 00:51.54 | TJNII | goes back to reading NE2000 datasheets |
| 00:52.01 | mchou | I wouldnt use asterisk for that |
| 00:52.07 | mchou | I'd use yate |
| 00:52.17 | mchou | failover already supported |
| 00:52.24 | *** join/#asterisk chendy (n=chatzill@113.91.37.208) |
| 00:52.34 | mchou | that's a form of hopping |
| 00:53.38 | mchou | not to mention that would avoid the "last register" problem |
| 00:53.51 | nix8n82 | p3nguin, what kind of control would I need for nat settings? would I have manually allocate ports to each server? like port 5060 5061 5062 and split up the rtp ports between each server? |
| 00:54.37 | p3nguin | Well, if you aren't using private IP addresses like I originally thought you were indicating, that's not really what you need to use. |
| 00:55.19 | nix8n82 | russellb, what proxy software would you recommend? |
| 00:55.39 | russellb | Kamilio, or one of the other similar variants |
| 00:56.16 | mchou | kamAilio |
| 00:57.34 | mchou | nix8n82: yate suffors failover even when calls are in progress. verry cool |
| 00:57.53 | mchou | s/suffors/allows |
| 00:59.03 | nix8n82 | yeah every instance has a private address, and I'm not sure I quite get how it all works, but lets say I have 5 instances all with a private ips and one public ip they all can register to my provider on port 5060 and use rtp 10000-20000 with the possibility of make 5000 calls at any one time? if I have the bandwidth and cpu cycles |
| 00:59.31 | nix8n82 | coo mchou thanks |
| 01:00.16 | mchou | nix8n82: what's the use case for this? you goona phone spam? |
| 01:00.20 | nix8n82 | being 5000 calls for all servers together not each |
| 01:00.21 | mchou | gonna* |
| 01:00.55 | TJNII | "Good evening, sir. Are you satisfied with the size of your p3n1s?" |
| 01:01.05 | mchou | haha |
| 01:01.06 | nix8n82 | no I would like to do like a town hall meeting over the phone |
| 01:01.38 | mchou | umm, why would you need to call out for that? |
| 01:01.43 | manxpower | nix8n82: there is a company that does that already |
| 01:02.24 | nix8n82 | speaking of that does anyone know of any software that would allow me to control something like that? |
| 01:02.35 | nix8n82 | manxpower, what company? |
| 01:03.00 | manxpower | I can't remember. |
| 01:03.38 | nix8n82 | well I would like to do that for my local area. |
| 01:04.28 | jblack | You'll almost certainly have to write it yourself. |
| 01:04.33 | nix8n82 | that and if I get good try to couple it with sphinx and see if I can't make it do radio contest and such |
| 01:04.37 | Katty | hugs jblack |
| 01:04.45 | jblack | hi catty |
| 01:04.49 | jblack | Gah! Katty. |
| 01:05.02 | manxpower | jblack: these people do things like host 5,000 participant townhall meetmes. |
| 01:05.08 | *** join/#asterisk joako (n=ston3d@opensuse/member/joak0) |
| 01:05.28 | Katty | jblack: let's watch pride and prejudice |
| 01:05.45 | jblack | manxpower: And there are companies that do conference call meetings for companies. |
| 01:06.02 | joako | Anyone knows how I can set the hostname on a Polycom phone? |
| 01:06.08 | *** join/#asterisk SaiSoma (n=SaiSoma@adsl-074-167-136-030.sip.mob.bellsouth.net) |
| 01:06.12 | jblack | I don't see where "someone else has done that" is pertinant, other than to indicate to nix8n82 that it's possible? |
| 01:06.26 | manxpower | jblack: they are doing it with Asterisk |
| 01:06.39 | nix8n82 | thank you jblack |
| 01:07.29 | jblack | manxpower: Great, more good news for him. Until you can provide contrary evidence that there's software available to him, I'll maintain that he'll have to write something himself. |
| 01:07.41 | jblack | becuase I know better than to try and prove a negative. |
| 01:07.43 | manxpower | jblack: I'm going to check on the name of the company for him. |
| 01:08.23 | jblack | I'm sure he'd appreciate that. =) |
| 01:08.50 | nix8n82 | I would |
| 01:09.08 | nix8n82 | if they can do it so can I |
| 01:09.36 | jblack | getting to your real question (how do I implement, rather than "who do I pay to do it for me").... |
| 01:10.07 | jblack | I doubt you'll be able to find any free software conferencing software that works at that scale for asterisk, and you'll probably have to write your own handling. |
| 01:10.45 | jblack | The two frameworks within asterisk that you'd be most interested in is the Asterisk Gateway Interface (AGI), and Asterisk Management Interface (AMI). |
| 01:11.09 | nix8n82 | right, I thought so too, but I don't believe I have to have all people in a conference, I only need a couple people in a conference and a way to stream audio to a bunch of calls |
| 01:11.18 | jblack | There's also a conferencing app that comes with asterisk called meetme, but I don't think it's intended to run at the scale that you intend. But who knows, maybe you could adapt it. |
| 01:11.29 | manxpower | I'd at least look into writing your own app (or even channel driver) and see if it's worth that or not. |
| 01:11.36 | TJNII | What, you don't want 5000 people to be able to talk to each other all on the same call? |
| 01:11.41 | TJNII | Sounds like fun! |
| 01:11.51 | TJNII | s/fun/white noise |
| 01:12.09 | jblack | nix8n82: I have a hunch you'll be doing some interesting stuff with bridging. =) |
| 01:12.09 | nix8n82 | yeah that would be a trip TJNII |
| 01:12.19 | manxpower | nix8n82: There's three major parts that I can think of. Call setup/teardown, rtp, and admin control. |
| 01:12.28 | *** join/#asterisk chendy (n=chatzill@119.139.171.209) |
| 01:13.19 | manxpower | If you could find something that could mux a couple of thousand calls, I bet SER/OpenSER could be used for call setup/teardown, asterisk for admin control |
| 01:14.00 | jblack | I wonder if he could just dump into a stream that's read by MOH. |
| 01:14.34 | *** join/#asterisk Katty (n=Angela@adsl-70-253-164-104.dsl.stlsmo.swbell.net) |
| 01:14.34 | manxpower | jblack: use a stream from the talker as the MoH source. |
| 01:14.44 | TJNII | Or set up meetmes in a spoke and hub topology with only the users in the hub conference having voice. |
| 01:14.47 | jblack | people "listening to the call" would actually be on hold. they dtmf to ask a question, which breaks them out of moh. |
| 01:15.04 | jblack | manxpower: mostly, yeah. |
| 01:15.05 | TJNII | Though then they wouldn't be able to talk easily... |
| 01:15.05 | manxpower | jblack: that's brilliant |
| 01:15.14 | nix8n82 | that would be cool jblack, what I'm thinking is I"m going to have to use the agi stream file and keep track of the offset and try to keep it lagged behind the speaker for no more than two seconds |
| 01:15.15 | jblack | with some bridging magic that can playback asked questions |
| 01:15.21 | *** part/#asterisk ZPertee (n=ZPertee@rrcs-74-219-221-11.central.biz.rr.com) |
| 01:15.23 | *** join/#asterisk SaiSoma (n=SaiSoma@adsl-074-167-136-030.sip.mob.bellsouth.net) |
| 01:15.41 | manxpower | TJNII: a second meetme for talkers, |
| 01:15.56 | TJNII | jblack: That is smart. Record their questions and allow the admin to play them back. |
| 01:16.02 | nix8n82 | if I can record and read the file while streaming it to a few different servers |
| 01:16.26 | jblack | nix8n82: Recording is the easy part, with Monitor() |
| 01:16.50 | *** join/#asterisk moy (n=moy@CPE001cdfec4cee-CM00080dab8485.cpe.net.cable.rogers.com) |
| 01:16.56 | jblack | Spool monitors() from listeners, have an assistant filter the useful ones out to another spool... |
| 01:17.24 | jblack | moderator presses a button to playback the top spooled monitor.... |
| 01:18.14 | jblack | though by the time your'e at this scale, you probably have a computer system where people click on described spools in a webbrowser, which has the logic to inject the described recorded monitor's into the meetme. |
| 01:18.35 | *** join/#asterisk CcRnp (n=shishir@208.179.165.18) |
| 01:18.45 | *** join/#asterisk thuddwhirr (n=wolthuis@mimezine.com) |
| 01:18.51 | nix8n82 | yeah, it's not going to be easy |
| 01:19.00 | nix8n82 | fun, but definatly not easy |
| 01:19.02 | jblack | i.e. from a listern's point of view. press a button, says record your message, and turns on monitor. that goes into a pile of other spooled messages |
| 01:19.04 | *** join/#asterisk robl^laptop (n=robl@c-98-197-98-39.hsd1.tx.comcast.net) |
| 01:19.07 | thuddwhirr | anyone know if you can retrieve the $RTPAUDIOQOS channel var from a sip channel created by the Dial application? |
| 01:19.07 | nix8n82 | need to learn ajax |
| 01:19.18 | jblack | assitant listens to them, describes them on the web page, and approves them. |
| 01:19.39 | manxpower | thuddwhirr: chances are that was moved in the CHANNEL() function. see "core show application channel" |
| 01:19.40 | jblack | the moderator, in the meetme that's being piped into moh, clicks on them, and bang, they're injected into the conference. |
| 01:19.50 | jblack | You don't need to learn ajax for that. |
| 01:20.07 | jblack | hell, you'd barely need to know php. |
| 01:20.09 | nix8n82 | for a nice dynamic control interface I do |
| 01:20.22 | CcRnp | guyz how can i access MEMBERINTERFACE queue variable ?? |
| 01:20.27 | CcRnp | for asterisk 1.4.24 |
| 01:20.31 | CcRnp | please help me out |
| 01:21.01 | CcRnp | TK]D-Fender do you have a idea about queue variable MEMBERINTERFACE |
| 01:21.41 | jblack | whatever. ui is a detail. |
| 01:21.54 | TJNII | nix8n82: Make sure to program a kickban system for the jerks. |
| 01:22.04 | CcRnp | [TK]D-Fender Do you have a idea bout queue vairable MEMBERINTERFACE |
| 01:22.10 | jblack | disagrees with tjnii |
| 01:22.36 | TJNII | Or at least make it so they can't record. |
| 01:22.43 | nix8n82 | jblack, russellb p3nguin TJNII..thank you all for your input, I really appreciate it..I have to go eat dinner mrs will get mad. |
| 01:22.46 | jblack | I still disagree. |
| 01:22.53 | nix8n82 | I disagree too |
| 01:23.04 | TJNII | jblack: Five thousand people. You will get at least one who wants to record "f*ck" every 3 seconds and submit it. |
| 01:23.27 | jblack | tjnii: And you'll have to live with such kids. |
| 01:23.39 | *** join/#asterisk Kumbang (n=kumbang@167.205.24.69) |
| 01:23.39 | TJNII | Live with yes, but not listen to. |
| 01:23.49 | nix8n82 | thats why there are moderators they don't have to be played in the main conference |
| 01:24.05 | TJNII | True. But the moderators should be able to weed out the abusers. |
| 01:24.10 | jblack | Yes. Listen to. Because a large majority of people have callerid blocking, which means you can't distinguish by CID. And if you hang up on them, they can just call right back. |
| 01:24.25 | jblack | That's why you record the messages, and have an assistant filter them out of band, tjnii. |
| 01:24.49 | jblack | a large minority, I meant. |
| 01:24.58 | TJNII | Well, I still say the assistant should be able to say "You know what, no voice for you." |
| 01:25.06 | TJNII | Just like we do here on IRC. |
| 01:25.37 | TJNII | A minority can be a huge disruption, just look at the /b/tards. |
| 01:26.50 | jblack | I'm sure there's people that call radio stations 30 times an hour. ANd I bet they get filtered out in _exactly_ the same way I describe |
| 01:27.32 | jblack | Tell you what. He can add that as a feature after he has an actual working system. |
| 01:28.03 | TJNII | Oh, I'm not disagreeing with your method. I agree completely. I think it is a great idea. However, I also think an ignore feature would be good, too. |
| 01:28.22 | jblack | grits his teeth. |
| 01:28.36 | TJNII | drops it |
| 01:28.47 | manxpower | There are many times I wish Pidgin had a /igore feature |
| 01:28.56 | jblack | tjnii: The best time to worry about the bells and whistles is after the basic "how do you make it work at all" is figured out. |
| 01:29.25 | TJNII | jblack: It is also good to plan for the fratures so you have the ability to add them. |
| 01:29.32 | jblack | Going off into the weeds with "it could this, and it could that, and xxx would be fantastic too!" is a great way to derail things that don't exist at all. |
| 01:29.52 | jblack | Fine. Great idea tjnii |
| 01:31.17 | *** join/#asterisk n3hxs (n=HAMming@75-151-157-13-Philadelphia.hfc.comcastbusiness.net) |
| 01:33.56 | *** join/#asterisk thansen (n=thansen@76.27.110.194) |
| 01:34.40 | *** join/#asterisk wonderworld (n=ww@ip-62-143-22-226.unitymediagroup.de) |
| 01:36.44 | nix8n82 | Yeah great idea TJNII, but most of these are set up by calling out and not in, and anyway you cut it it will be easy to have a unique id for each channels that is open, so if some is a tool it will still be easy for moderators to ignore them, and If I allow people to call in and there caller id is unkown I will make them put in a call back number, to make it easier to distribute the load to a less used server |
| 01:38.09 | jblack | fine. I'll do you one better. |
| 01:38.27 | jblack | Mark callers as "bad". Pretend to record their messages, but actually throw them away. |
| 01:38.36 | jblack | Then, they don't know to hang up, forge caller id, etc. |
| 01:38.44 | TJNII | nix8n82: Yea, I agree with jblack (even if he doesn't agree wih me). Build in a uid now and that will give you the framework, if needed, later. |
| 01:40.58 | nix8n82 | jblack, yeah I really like your thinking |
| 01:42.56 | nix8n82 | TJNII, you should take notes with me |
| 01:45.31 | nix8n82 | jblack, how do you think I should connect the speaker monitor spool to the other server with the listeners? mount them with sshfs or do think there is something better? |
| 01:46.03 | jblack | besides. if I knew you were auto-dropping misbehaving numbers... I could prevent someone from participating by faking their callerid and misbehaving |
| 01:46.44 | nix8n82 | exactly jblack |
| 01:46.53 | jblack | if you hand filter calls out of band like I suggest, you stop misbehavers from being productive (which is making a mess of the show), but still keep a human in the loop. |
| 01:47.43 | jblack | if there's really _that_ many questions being asked, add a second or even third assistant for filter them |
| 01:47.47 | nix8n82 | you never want to kick a call out of the meeting, you are getting paid to have someone listen to a speaker, you just act like you que there question like the rest |
| 01:48.28 | jblack | yeah. The question is already recorded. The person that listens to the already recorded question just decides if it's good enough to continue on, and describes it for the page. |
| 01:48.45 | jblack | The person that asked the question is already long since gone. =) |
| 01:49.26 | jblack | in simpler words, every caller "Will have take my answer off the air" |
| 01:50.02 | nix8n82 | yeah it's not a debate |
| 01:50.25 | nix8n82 | that really clears up a lot of things |
| 01:50.28 | jblack | Yup. press 1 to record a question, and press 1 when done asking it. |
| 01:50.55 | jblack | that question goes into a spool of other questions, and the asker goes back to moh. |
| 01:51.03 | jblack | you understand now, nix? |
| 01:51.25 | nix8n82 | yeah |
| 01:51.43 | jblack | so you have that pile of audio files that assistants listen to, and decide if they're worth of making it on. ANd describing them for the interface, and the moderator can pick from the list which questions he wants. =) |
| 01:52.14 | jblack | wet dream for a politician. |
| 01:52.23 | nix8n82 | it really is |
| 01:54.01 | nix8n82 | although I would like to be able to record that persons position where they left off to ask there question and be able to put them back in where they left off. |
| 01:54.07 | nix8n82 | if that is even possible |
| 01:54.27 | jblack | I have no idea what you just said |
| 01:54.40 | TJNII | The moh method should handle that. |
| 01:54.44 | TJNII | I would think. |
| 01:54.54 | jblack | Ohhhh |
| 01:55.14 | jblack | Like a pause button for the town meeting. |
| 01:55.22 | nix8n82 | yeah |
| 01:55.32 | jblack | I don't think the moh approach would do that, because there's one stream that everyone is listening to. |
| 01:56.14 | TJNII | Yea, something would have to buffer the stream |
| 01:56.19 | jblack | to control streams like that, you'd have to give everyone their own stream, and the resources required are going to get high |
| 01:56.20 | TJNII | Shouldn't be too hard, though. |
| 01:56.29 | TJNII | True |
| 01:56.33 | jblack | playing 5,000 audio files at once? Not too hard? |
| 01:56.41 | TJNII | Yea, you're right |
| 01:57.25 | jblack | if people want to hear the part htey missed, remind 'em that you'll have a podcast of it available later. :) |
| 01:57.39 | jblack | hell. do a live podcast, since you're already making the moh stream. :) |
| 01:57.44 | nix8n82 | I hope thats where agi and the stream file command would help out. because they could push a button and it should return the offset of the played file and then hopefully return to that offset after they are done recording |
| 01:57.47 | *** join/#asterisk kfife (n=Miranda@kfife.com) |
| 01:58.39 | nix8n82 | yeah might have to go with the podcast idea |
| 01:58.49 | jblack | nix8n82: Not with musiconhold, you're not. |
| 01:58.56 | jblack | That's one stream that everyone's listening to. |
| 01:59.02 | jblack | Not 5,000 individual streams. |
| 01:59.42 | jblack | let me put it to you in a way that's a little bit more clear. |
| 02:00.01 | jblack | Can you imagine your computer playing back 5,000 mp3s at the same time? |
| 02:00.35 | kfife | Easy to imagine: but impossible for the computer :-) |
| 02:00.39 | kfife | Hey guys: want to run dahdi_monitor from within Asterisk triggered by a dialplan event: Using SYSTEM() asterisk dialplan waits for return. Any way to run SYSTEM() spawned as a separate process so dialplan continues immediately? |
| 02:01.18 | TJNII | Heh. It is doable if you really want to get down to it. However, I've been reading about DMA nd ring buffers for the last day, so the method I'm currently thinking of is way, way out there. :) |
| 02:01.36 | jblack | nix8n82: if you buffered 500k of each stream, you're looking at 2.5 gigs of ram. For the buffering alone |
| 02:02.16 | nix8n82 | no but I can imagine about 25 to 50 virtual servers playing 100 to 200 streams a piece |
| 02:03.03 | TJNII | That probably won't work. Those 25 virtual servers may be on only 10 real servers. The real servers will be overloaded. |
| 02:03.15 | jblack | Are you sure you know what you're doing, nix8n82 / |
| 02:03.23 | jblack | 5,000 of anything is a lot. |
| 02:03.58 | kfife | not dollars :-) |
| 02:04.02 | *** join/#asterisk outtolunc (n=me@c-98-248-96-110.hsd1.ca.comcast.net) |
| 02:04.06 | jblack | except dollars. :) |
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| 02:04.24 | kfife | or women in a naked girl avalanche |
| 02:04.35 | kfife | (my preferred way to go) |
| 02:05.09 | kfife | so any ideas on system() |
| 02:05.12 | kfife | ?? |
| 02:05.24 | jblack | if you were avelanched by 5,000 80 pound anorexic girls, you would be smothered under 200 tons of meat and bone. |
| 02:05.31 | TJNII | kfife: Throw a & on the command |
| 02:05.31 | jblack | system huh what? |
| 02:05.53 | kfife | Bring about 5000 hither! Yee haw! |
| 02:06.00 | jblack | kfife: Dude, just use agai. |
| 02:06.05 | jblack | agi, that is |
| 02:06.08 | TJNII | But forking background processes from the dialplan is a super bad ides. |
| 02:06.18 | kfife | TJNII: thanks! |
| 02:06.45 | kfife | I am triggering dahdi_monitor with a byte limit. Trying to debug somethign -- a temporary situaiton. |
| 02:06.58 | kfife | TJNII: You're right though :-) |
| 02:07.05 | nix8n82 | jblack, I'm not quite sure what I'm doing..if I did I wouldn't ask question |
| 02:07.35 | jblack | nix8n82: Ok. You're not going to be doing 5,000 individual streams. |
| 02:08.14 | nix8n82 | either way for 5000 calls you will at least eat up 70mb of bandwidth |
| 02:08.41 | jblack | Dude, you're getting annoying. |
| 02:09.07 | jblack | It was only about 5 years ago that the default linux kernel let you keep open more than 1000 files at a time. |
| 02:09.10 | kfife | TJNII: I'm sorry, but where does the & go in the syntax? System(dahdi_monitor 1 -l 500000 -f /var/spool/asterisk/monitor/stream${UNIQUEID}.raw) |
| 02:09.20 | jblack | more than that, and you had to hack the kernel. |
| 02:09.58 | TJNII | kfife: At the end. Bash 101 |
| 02:10.02 | nix8n82 | right but this won't be ran entirely from one server |
| 02:10.03 | jblack | and that was files and sockets together, due to the defined FD_MAX |
| 02:10.23 | kfife | Sorry, my introduction to Linux was Asterisk |
| 02:10.41 | jblack | So on top of all the other complexity, you want to add distributive processing on top of it. |
| 02:10.54 | nix8n82 | yep |
| 02:11.19 | jblack | when you were five, you liked to doggy paddle in the deep end of the pool, didn't you |
| 02:11.42 | nix8n82 | back float not a real good swimmer |
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| 02:13.00 | jblack | I just don't know what to say |
| 02:14.06 | nix8n82 | you think I can do 5000 calls from one server with MoH? |
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| 02:15.30 | jblack | yeah. |
| 02:16.15 | kfife | Tryiing to decode the dahd_monitor output using the syntax suggested at the voip wiki: sox -r 8000 -s -2 stream1257192619.393.raw out.wav |
| 02:16.27 | kfife | sox complains: sox: bad input format for file stream1257192619.393.raw: data size was not specified |
| 02:16.47 | kfife | Can anyone help me correct my syntax? |
| 02:16.52 | nix8n82 | I read a blog where they used asterisk 1.6 with Amazon EC2 cloud to make 2000 using 20 medium instances with 100 calls going out a piece |
| 02:17.10 | nix8n82 | bill wasn't cheap but they made it happen |
| 02:29.03 | nix8n82 | http://www.amoocon.de/talks/27 |
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| 02:30.22 | nix8n82 | my numbers were wrong, but close |
| 02:36.26 | jblack | I think I'm not being clear. |
| 02:36.38 | jblack | Sure, this sort of work can (and eventually should) be distributed. |
| 02:38.06 | jblack | I also think that distributing work is much harder than non-distributed work. Thus, until you can do the simpler type, attempting the more complex type is silly. |
| 02:39.29 | jblack | by way of metaphor, there's no way you can convince me that you can rebuild a transmission until after you've at least proven you can change transmission fluid |
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| 02:45.44 | nix8n82 | right, I've set up multiple vicidial servers, I have one quad core server that does at least 200 concurrent streaming the same audio file or a different one, each of them playing there own stream. I have about 75 to 100mb or ram left at that point, and the server is running the mysql server and web front end. with more clock cycles to spare. I've wrote agi and script that use ami..I think I can pull this off, maybe not entirely by myself |
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| 03:23.08 | Katty | peeks in |
| 03:24.18 | Katty | HELLO THAR |
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| 03:27.30 | johnakabean | asterisk keeps crashing when I reload it; the script that starts it is safe_asterisk which is correctly looking for /var/run/asterisk/asterisk.pid |
| 03:27.31 | johnakabean | 0 |
| 03:27.43 | johnakabean | Asterisk ended with exit status 137 |
| 03:27.43 | johnakabean | Asterisk exited on signal EXITSTATUS-128. |
| 03:27.44 | johnakabean | Automatically restarting Asterisk. |
| 03:27.44 | johnakabean | mpg123: no process killed |
| 03:28.09 | johnakabean | 1.4.26.2 |
| 03:28.27 | manxpower | johnakabean: there's documentation on the doc directory of the asterisk source code with information on getting a core dump and reporting the bug |
| 03:29.14 | johnakabean | it does this, except when NOT reloading, with 1.6.1 |
| 03:32.26 | johnakabean | execincludes=yes what does this mean in asterisk.conf? |
| 03:32.53 | johnakabean | does it execute instead of including linked files? |
| 03:34.41 | p3nguin | How weird. I have an ATS cordless SIP phone, and it seems to capture (or block) the dialing of *69 from the handset. It never reaches asterisk, yet I receive a fast busy when I dial it. With verbose up and sip debug on, no packets even reach asterisk when dialing *69. |
| 03:35.55 | johnakabean | that's common, in the regional tab, ERASE ALL except call return and specific others that asterisk doesn't support |
| 03:36.19 | johnakabean | if there is no tab to erase service codes, you're screwed |
| 03:36.36 | p3nguin | Other star codes such as *32 and *54 work just fine from that phone, and *69 works from other phones... |
| 03:36.47 | p3nguin | Anyone ever experience such a thing before? |
| 03:37.07 | johnakabean | yes but those codes are being executed by the ATA, not passed to asterisk |
| 03:37.09 | jblack | p3nguin: any time I've seen a phone eat dtmf, it had an internal dialplan that needed adjustment |
| 03:37.33 | p3nguin | Ah, good. I know this phone has that in it. |
| 03:38.48 | p3nguin | dial plan: [1-9]xxxxxxxxxxxxxxxxxxxx|xx+*|xx+#|*.#|*.T3 |
| 03:39.18 | p3nguin | Well, that doesn't really explain why other star codes do reach asterisk, but *69 doesn't. |
| 03:40.38 | jblack | I don't see anything in there either |
| 03:41.52 | p3nguin | The phone has a "call return" button, which dials *69. I do get the same result by pressing that button and by dialing it directly on the keypad. |
| 03:42.07 | jblack | you've already said that asterisk isn't getting it. |
| 03:42.21 | p3nguin | Just trying to be complete. |
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| 03:42.42 | jblack | that sticks you right into "device specific configuration", and leaves you with some combination of google and the manufacturer to find a solution. |
| 03:43.06 | jblack | though I'll be happy to sit here and comiserate with you. |
| 03:44.10 | p3nguin | I guess I can do like PSTN, and accept 1169 (for people who haven't heard of touch tone) the same as *69. It's a silly workaround, but it would be effective. |
| 03:44.35 | jblack | sure. |
| 03:44.53 | jblack | tjpigj O |
| 03:45.02 | jblack | though I'd consider just "69" |
| 03:46.07 | jblack | sometimes I wonder how 69 got through the telco's prude filters.... |
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| 03:53.05 | Fister | i think i tested dahdi/zaptel once and found that it actually registers 11 switchhook pulses as a star. |
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| 06:03.39 | nsgn | well, i forgot who it was in here the other day who has voip.ms and was helping me with my issue |
| 06:03.46 | nsgn | lost my irc log |
| 06:03.52 | nsgn | but i'm reporting back as they requested |
| 06:04.14 | nsgn | Whoever you were/are, if you're here, speak up and i'll explain what ended up happening. Somehow it works now. |
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| 06:15.51 | mchou | nsgn: so what happened? |
| 06:16.09 | mchou | nsgn: you had a mistake in your dial plan? |
| 06:17.06 | mchou | nsgn: or your firewall prematurely closed ports? |
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| 06:32.28 | nsgn | mchou: no, it was a wild situation |
| 06:32.39 | nsgn | calls to my DID couldnt come in most of the time |
| 06:32.50 | mchou | right, I remember that |
| 06:32.52 | nsgn | i had a helpful friend here actually put my DID into his working * setup |
| 06:33.01 | nsgn | and the same problem happened! |
| 06:33.11 | nsgn | so it was clearly on voipms' end |
| 06:33.14 | mchou | that's why I said your firewall might have closed ports prematurely |
| 06:33.25 | nsgn | contacted them multiple times over a week and they said they couldnt find an issue |
| 06:33.30 | nsgn | finally i got a guy convinced it wasnt me |
| 06:33.35 | nsgn | and he put an elevated ticket in |
| 06:33.52 | mchou | and? |
| 06:34.05 | nsgn | the ticket never showed up on my account, nor was i ever contacted...but one business day later it all suddenly starts working. i can call it repeatedly tonight. could never do that before |
| 06:34.13 | mchou | heh |
| 06:34.29 | nsgn | one of those "it's not broken!" *secretly fixes* kind of things |
| 06:34.44 | mchou | nsgn: you should ask them what happend:) |
| 06:35.15 | mchou | nsgn: write to them and say "it's still borked" :) |
| 06:35.36 | nsgn | noooooooooooo |
| 06:35.41 | nsgn | if it works i'm not gonna screw with it |
| 06:35.46 | nsgn | i've learned that lesson many many times |
| 06:37.00 | mchou | there's not that many things that could go wrong with inbound sip signalling..... |
| 06:37.23 | mchou | either then sent the packet or they didnt |
| 06:37.34 | mchou | s/then/they |
| 06:37.54 | nsgn | yeah |
| 06:37.59 | nsgn | my asterisk debug said nothing ever arrived |
| 06:38.12 | nsgn | and the fact that i tested it on someone else's setup where the firewall was known to be fine kindof took that out of the loop too |
| 06:38.25 | mchou | no, asterisk debug was the wrong tool to use |
| 06:38.45 | mchou | should have used wireshark |
| 06:38.57 | mchou | on your router |
| 06:39.00 | nsgn | yes |
| 06:39.08 | nsgn | however the alternate test was easier |
| 06:39.12 | mchou | see if packets were making it there |
| 06:39.17 | nsgn | considering my firewall is highly active right now |
| 06:39.37 | nsgn | constant traffic to/from many many sources, and limited ability to stop the stream of traffic at a whim, limits things |
| 06:39.48 | nsgn | it's running an office with a server that calls outbound 24/7 |
| 06:39.49 | mchou | what?? |
| 06:40.03 | nsgn | i'm saying i couldnt play much with the firewall |
| 06:40.04 | mchou | that's full of baloney |
| 06:40.36 | nsgn | perhaps it can be done without any interruption. i can't claim to know wireshark well, i'm just saying i'm not in a position to screw with/take down the firewall setup here |
| 06:40.51 | mchou | you could have filtered ip packeds using wireshark so you didnt have to look at captures a mile long |
| 06:40.52 | nsgn | however it was apparently on their end, cause i made no changes on this end in the past 24 hours |
| 06:41.16 | mchou | you dont need to take down the firewall to do any of that |
| 06:41.31 | nsgn | ok. good to know. again i'm really not experienced with wireshark |
| 06:41.49 | mchou | nsgn: you never got to the root cause |
| 06:41.52 | nsgn | though the firewall is conveniently monowall on freebsd, which probably expands my options a good bit |
| 06:42.11 | nsgn | mchou: if it wasnt on my end i probably won't |
| 06:42.19 | mchou | nsgn: it wouldnt surprise me if it starts happening again |
| 06:42.21 | nsgn | being that they entered no formal ticket for it it would be hard for me to get them to track |
| 06:42.58 | mchou | yeah whatever |
| 06:43.05 | mchou | I'm not convince |
| 06:43.11 | mchou | convinced* |
| 06:43.30 | nsgn | i am not going to put this DID into operation without about a week of calling it day and night at random times. i don't need it for about a week anyway. if the issue happens once within that time i'll abort putting it into service. additionally i have an alternate DID/route configured to operate seamlessly, so there is no harm to callers if this doesn't work in operation |
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| 06:43.41 | nsgn | it just gives me a cheaper calling route with more channels |
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| 06:49.05 | nsgn | mchou: say it were on my firewall end....the heck could it be? it should be really simple. it's monowall in a basic, pretty open setup. nothing weird going on with packet filtering or proxying. SIP ports specified by voipms are forwarded to the asterisk server. no firwall on the asterisk server. calls that come through (about a third of them would during the problem time) are clear and never drop |
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| 06:52.01 | mchou | If you configured asterisk correctly there would be no need to even mess with firewall rules (aside from fowarding appropriate RTP ports0 |
| 06:52.19 | nsgn | exactly. all i had forwarded was sip control and RTP |
| 06:52.28 | mchou | nop |
| 06:52.41 | mchou | you dont even need to forward sip ports |
| 06:52.45 | nsgn | i have zero issues with ALL other traffic on this firewall...and it's doing a lot |
| 06:52.52 | mchou | that's the whole point |
| 06:53.37 | nsgn | it's why i use monowall. it's simple and easy, but configurable where i need. it really just works though |
| 06:54.19 | mchou | the issue here isnt the firewall per se |
| 06:54.32 | mchou | it's that you never figured out the root cause |
| 06:54.54 | mchou | so the problem could still be there |
| 06:55.04 | mchou | just masked right now |
| 06:55.31 | nsgn | sure. i'm asking what there is to do at this point |
| 06:55.40 | mchou | I mean you also have zero evidence that voip.ms actually did anything |
| 06:56.01 | nsgn | other than the fact that i did nothing and it now works solid |
| 06:56.08 | mchou | lol |
| 06:56.19 | mchou | that's what people always say |
| 06:56.29 | mchou | "I didnt do anything" |
| 06:56.43 | nsgn | and the last guy put in detailed notes to someone (didn't go on a ticket to me though) agreeing with me that the issue was likely on their DID provider's end |
| 06:56.49 | mchou | later we find out they restarted asterisk (or whatever) |
| 06:56.51 | nsgn | they may have inquired with the carrier |
| 06:57.01 | nsgn | i actually didn't even do that |
| 06:57.16 | mchou | nsgn: no, I'm using that as an example |
| 06:58.00 | nsgn | asterisk is in a VM on a server holding several other crucial VMs |
| 06:58.10 | nsgn | so no reboot, no firewall changes. heck i wasnt even HERE |
| 06:58.19 | nsgn | power didnt go out (server is on batteries anyway) |
| 06:58.30 | nsgn | i was working for the past 14 hours |
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| 06:58.58 | mchou | nsgn: it's simple. The bottom line is you didnt figure out anything |
| 06:59.06 | nsgn | i sure didnt |
| 06:59.21 | nsgn | not denying that. i'm just asking what you suggest can be done now |
| 06:59.34 | mchou | I told you already |
| 06:59.46 | nsgn | was fixed after bugging carrier. they provided no details. i can't trace much on this end now cause it works |
| 06:59.47 | mchou | tcpdump, wireshark |
| 07:00.33 | nsgn | what will they find on a working setup? |
| 07:01.04 | nsgn | do you expect there are still signs of abnormality despite it's current functionality? |
| 07:01.17 | mchou | I just told you |
| 07:01.32 | mchou | I expect the problem will return |
| 07:02.25 | nsgn | so you're saying use those when it returns? if so...durr. i'm asking if you're trying to tell me to use them when it is working |
| 07:02.54 | mchou | nsgn: lol |
| 07:03.15 | mchou | if it's working what problem is there to find? |
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| 07:04.00 | nsgn | that's precisely what i've been trying to figure out |
| 07:04.04 | nsgn | why you're telling me this |
| 07:04.11 | nsgn | if it comes back i'll obviously need to look into it :D |
| 07:04.19 | nsgn | but until then what can i do other than enjoy it? |
| 07:04.26 | nsgn | it's not mission critical since i have the backup route |
| 07:04.36 | nsgn | just relieves some channel congestion |
| 07:05.12 | mchou | nsgn: good luck |
| 07:05.24 | nsgn | thanks. now to do IVRs |
| 07:05.25 | nsgn | wooooo |
| 07:23.24 | ChannelZ | Anyone know who does reasonably priced SSL certs *besides* GoDaddy? |
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| 07:55.10 | DND | guys can anyone help me with nvfax? |
| 07:55.34 | DND | its hard to be 10 hours ahead on timezone :( |
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| 08:02.21 | *** join/#asterisk tareKhoury (n=tarekhou@tony11-128-131.inter.net.il) |
| 08:03.10 | tareKhoury | hello mates, any idea why this error keeps apearing |
| 08:03.13 | tareKhoury | WARNING[3643]: ast_expr2.fl:440 ast_yyerror: ast_yyerror(): syntax error: syntax error, unexpected '=', expecting $end; Input: |
| 08:03.13 | tareKhoury | <PROTECTED> |
| 08:03.31 | ChannelZ | you have a syntax error |
| 08:03.40 | tareKhoury | the line is casing it is |
| 08:03.43 | tareKhoury | exten => s,n,GotoIf($[${blacklisted} = 1]?blocked,1) |
| 08:03.51 | tareKhoury | i don`t see no error here :) |
| 08:04.53 | kaldemar | i do |
| 08:05.05 | tareKhoury | what is it? |
| 08:05.35 | kaldemar | blocked,1 should be blocked:1 |
| 08:05.45 | kaldemar | or no, not necessarily. |
| 08:05.53 | tareKhoury | it`s asterisk 1.6 |
| 08:06.03 | kaldemar | if it's a two part label and there is no false |
| 08:06.11 | ChannelZ | I don't think you want the spaces around the = actually |
| 08:06.22 | tareKhoury | i tried that also |
| 08:06.24 | tareKhoury | no good |
| 08:06.26 | tareKhoury | i removed the space |
| 08:06.41 | kaldemar | show the real call and the contents of the var |
| 08:06.44 | ChannelZ | then what is ${blacklisted} ? |
| 08:06.59 | tareKhoury | a variable that i set through AGI script |
| 08:07.07 | tareKhoury | after checking with database .. if listed |
| 08:07.13 | kaldemar | can it contain other than numbers? |
| 08:07.18 | tareKhoury | 0 or 1 |
| 08:07.27 | tareKhoury | 1 is listed .. 0 is not |
| 08:07.32 | ChannelZ | yes but what is it actually containing? Do you get truly just a 0 or 1 if you NoOp it? |
| 08:07.53 | tareKhoury | i`ll try to NoOp it now |
| 08:09.07 | tareKhoury | NoOp("SIP/0555555555-089c0b10", "my var is = 1") in new stack |
| 08:09.13 | tareKhoury | it is 1 |
| 08:09.16 | ChannelZ | haha no it's not |
| 08:09.24 | ChannelZ | it's "my var is = 1" |
| 08:09.30 | tareKhoury | noo i added that text ;p |
| 08:09.48 | tareKhoury | my noop is NoOp(my var is = ${blacklisted}) |
| 08:10.35 | ChannelZ | well the only thing I can think of is to quote it |
| 08:10.56 | ChannelZ | GotoIf($["${blacklisted}="1"]?blocked,1) |
| 08:11.05 | ChannelZ | whoops |
| 08:11.06 | tareKhoury | it will solve it .. but is it right to do that |
| 08:11.09 | ChannelZ | GotoIf($["${blacklisted}"="1"]?blocked,1) |
| 08:11.10 | tareKhoury | it`s not a text var |
| 08:11.21 | kaldemar | spaces aroud = are not bad. |
| 08:11.32 | ChannelZ | well I don't think it knows that it's an integer |
| 08:12.09 | tareKhoury | i guess im gonna have to quote |
| 08:16.36 | tareKhoury | exit |
| 08:16.41 | *** part/#asterisk tareKhoury (n=tarekhou@tony11-128-131.inter.net.il) |
| 08:25.03 | *** join/#asterisk tamiel (n=tamiel@213.30.183.226) |
| 08:35.13 | *** join/#asterisk Chodorenko (n=chodoren@ext.one.by) |
| 08:45.42 | *** join/#asterisk RypPn (i=TuMbL@rosscom.co.uk) |
| 08:45.55 | *** join/#asterisk cuco (n=Diego@bzq-79-179-193-107.red.bezeqint.net) |
| 08:46.44 | cuco | hi all, free pizza for first one that calls iax2:guest@local.xorcom.com/276 |
| 08:48.12 | *** join/#asterisk QaDeS (n=mklaus@p4FC72A5C.dip0.t-ipconnect.de) |
| 08:48.49 | mchou | lol |
| 08:48.55 | mchou | do you deliver? |
| 08:48.57 | *** join/#asterisk gr0mit (n=tim@2001:8b0:3d9:0:44be:b966:abd6:135) |
| 08:49.04 | cuco | mchou: no, tzafrir does |
| 08:49.16 | *** join/#asterisk TSM (n=the_soft@87-194-32-212.bethere.co.uk) |
| 08:49.51 | mchou | cuco: I have no idea who he is |
| 08:50.35 | TJNII | I wonder how much it costs to get a OUI from the IEEE.... |
| 08:53.02 | *** join/#asterisk war9407 (i=war@liquidswords.org) |
| 08:53.25 | *** join/#asterisk UQlev (n=yuriy@nb11-125.static.cytanet.com.cy) |
| 08:53.29 | mchou | cuco, how come u dont answer? |
| 08:53.31 | cuco | is rofl |
| 08:53.41 | cuco | mchou: that's not my extension :) |
| 08:54.24 | *** join/#asterisk soman (n=somnath@118.102.130.6) |
| 08:55.10 | mchou | cuco: you owe me dude! |
| 08:55.27 | mchou | cuco: dont make promises u cant keep! |
| 08:55.45 | TJNII | Dude, this is the internet. |
| 08:55.52 | mchou | lol |
| 08:55.55 | TJNII | Making empty promises is all anyone does. |
| 08:56.27 | mchou | to his credit tzafrir did offer me a pizza |
| 08:56.40 | mchou | but I'm hungry now |
| 08:56.43 | kaldemar | free interrogations for the first one to call obama! |
| 08:57.02 | mchou | kaldemar: you have his extension? |
| 08:57.25 | mchou | kaldemar: I can use a new waterboard |
| 08:57.32 | kaldemar | hey, if you get interrogations instead of pizza, you can find out yourself. |
| 08:58.21 | *** join/#asterisk Ashura (n=ashura@89.119.206.194) |
| 08:58.24 | mchou | but that's pretty lame |
| 08:58.47 | kaldemar | no shit |
| 08:58.49 | mchou | just found out the free version of zoiper allows only two accounts |
| 08:59.53 | mchou | that's what happens when you call random extensions for pizza |
| 09:00.09 | TJNII | Okay, this is completely off topic but someone here might know the answer\. I have a NIC that doesn't have a mac address. Either its eeprom was wiped or it cleared itself from lack of use. Anyways, I'm bit banging it with a MCU and I need to give it a mac address. Is there a OUI block I can just pick one from for testing? Otherwise I'm just going to take the mac of one of my other NICs and add 1. |
| 09:00.14 | kaldemar | hmm. it was two per protocol some time ago. |
| 09:01.54 | mchou | TJNII: you can probably pic an obsolete org from IEE OUI |
| 09:02.00 | mchou | pick* |
| 09:02.09 | mchou | plent of those around |
| 09:02.14 | mchou | plenty* |
| 09:02.16 | TJNII | mchou: Good idea |
| 09:02.27 | *** join/#asterisk KermitTheFragger (n=KermitTh@118-197.bbned.dsl.internl.net) |
| 09:02.48 | mchou | TJNII: dont blame me if black helicopters descend on you though |
| 09:03.09 | Ashura | hallo! |
| 09:03.15 | TJNII | pfft. If the mac address is visible to the outside I've got bigger problems then mac spoofing. |
| 09:04.10 | *** join/#asterisk sulex (n=sulex@88-149-154-95.static.ngi.it) |
| 09:04.28 | *** join/#asterisk d00gster (n=doughant@94.98.239.32) |
| 09:04.55 | Amorsen | There are plenty of private (local-assigned) MAC addresses |
| 09:05.25 | Amorsen | Back in the day it was quite popular to do your own MAC assignments |
| 09:06.24 | Amorsen | Just set first bit zero and second bit one, then go wild |
| 09:07.04 | TJNII | Yea, I see a number of OUIs labled private. |
| 09:07.20 | TJNII | I'll go with 00:01:01 |
| 09:07.21 | TJNII | Thanks! |
| 09:07.24 | mchou | nah |
| 09:07.31 | mchou | 00000F |
| 09:07.35 | mchou | Next |
| 09:07.42 | mchou | go mess with Steve |
| 09:11.38 | *** join/#asterisk mumtazah1 (n=mumtazah@203.82.91.104) |
| 09:12.51 | TJNII | Dude, Cisco has 7,365,197,824 addresses allocated. |
| 09:12.54 | Amorsen | 000101 is ETI |
| 09:12.57 | TJNII | That's nuts. |
| 09:13.18 | Amorsen | You really should set second bit one |
| 09:13.40 | mchou | Amorsen: ETI? |
| 09:13.47 | Amorsen | Yes, the company |
| 09:14.27 | TJNII | Well, not according to the oui list I just downloaded from ieee.org, but okay. |
| 09:14.59 | mchou | IEEE says "private" for that one |
| 09:15.00 | Amorsen | Half the MAC addresses are private, there's no reason to pick one which might collide |
| 09:15.13 | Amorsen | Private just means they won't tell who they assigned it to, I think |
| 09:15.21 | TJNII | Aah |
| 09:15.48 | TJNII | So does the second bit signify private addressing, then? |
| 09:15.49 | mchou | bah |
| 09:16.00 | mchou | just use a pseudomac address |
| 09:16.10 | mchou | that ought to spice things up |
| 09:16.12 | Amorsen | 4xxxxx is private |
| 09:16.50 | Amorsen | Unless I'm misreading the standard |
| 09:17.52 | TJNII | The page I just found on ieee.org agrees, so I will go with that. |
| 09:21.54 | *** join/#asterisk cuco (n=Diego@local.xorcom.com) |
| 09:25.49 | *** join/#asterisk TiToyz (n=TiToyz@aut75-5-82-239-181-57.fbx.proxad.net) |
| 09:26.30 | *** join/#asterisk howie (n=howie@71-95-220-206.static.mtpk.ca.charter.com) |
| 09:27.36 | cuco | mchou: thanks for beeing the bunny test :) |
| 09:30.01 | *** join/#asterisk ruyo (n=sayo@195-23-253-223.net.novis.pt) |
| 09:30.23 | *** join/#asterisk TommyBotten (n=noosjent@services.nbi.freecode.no) |
| 09:31.26 | mchou | cuco: this bunny is hungry for pizza |
| 09:31.57 | Ashura | sorry but... the mac address never crosses a router...you can just pick one of your existing card and add a 1 (supposing you stay unique) |
| 09:32.40 | Ashura | where "unique" means "unique in your lan segment" |
| 09:33.00 | Ashura | si ccnp |
| 09:34.31 | mchou | Ashura: you're a bit late to the party |
| 09:35.02 | Ashura | sorry, i'm at work :) |
| 09:35.56 | Ashura | is anyone using chan_skype (digium) + country subscription? |
| 09:40.17 | tzafrir__laptop | Ashura, your method is not exactly a good one. There's a reasonable chance you'll get a new card of the same vendor you got the previous card |
| 09:40.54 | Ashura | tzafrir__laptop, about skype? |
| 09:41.47 | tzafrir__laptop | no. about the MAC address |
| 09:42.00 | tzafrir__laptop | mchou, bunnies eat pizzas? |
| 09:42.05 | Ashura | well, is a good workaround anyway... |
| 09:42.30 | *** join/#asterisk ccesario (n=ccesario@189-19-6-236.dsl.telesp.net.br) |
| 09:42.31 | Ashura | anyway, a cheap eth card is really cheap these days :) |
| 09:42.56 | mchou | tzafrir__laptop: this bunny will eat anything |
| 09:43.24 | mchou | tzafrir: I'd eat a cow right now if I came across one |
| 09:43.34 | mchou | that's how famished I am |
| 09:44.25 | mchou | tzafrir: where are you in .il? |
| 09:44.36 | tzafrir | Near Carmiel |
| 09:44.51 | mchou | cool |
| 09:55.32 | lftsy | Hello, could you confirm me that chan_sip.c will allow only symmetric codecs! Thanks |
| 09:56.18 | tzafrir | lftsy, confirmed |
| 09:56.20 | cuco | mchou: now we know that "external" folks can molest us by sip/iax directly. we don't no stinking telcos. |
| 09:56.57 | tzafrir | Though it would still be nice if you drop by and say "hi" |
| 09:57.00 | mchou | hmm?? |
| 09:57.06 | mchou | cuco: ^^^ |
| 09:57.24 | mchou | cuco: you still need phone 3s :) |
| 09:57.30 | mchou | #s* |
| 09:58.24 | mchou | cuco: rest of world not prepared to enter sip or iax uris into their phone :) |
| 09:58.27 | cuco | mchou: we were not sure if people are able to directly dial into our pbx. we tried testing it ourselved (by connection a windows machine+zoiper open+wirelss). I suggested trying irc. |
| 09:59.08 | mchou | cuco: I'm not sure why you ever doubted that :) |
| 09:59.24 | mchou | unless you has some uber firewall |
| 09:59.39 | mchou | s/has/have* |
| 10:00.36 | mchou | cuco: or you just meant you needed someone outside to test your dialplan (been there, done that) |
| 10:00.52 | |stefan| | when using dialplan command dial . shouldn't it dial the specified time before passing to voicemail ? |
| 10:01.24 | |stefan| | like this... s,3,Dial(SIP/ext100,60,r) s,4,VoiceMail(100@default) s,5,Hangup |
| 10:01.36 | |stefan| | and it throws me directly into voicemail |
| 10:01.55 | TJNII | Is SIP/ext100 reachable? |
| 10:02.05 | |stefan| | yes |
| 10:02.07 | cuco | mchou: bingo. always easier to dump sip urls on the irc then testing myself |
| 10:02.36 | mchou | |stefan|: that's cause you have mistake in your dial plan |
| 10:02.51 | |stefan| | mchou: explain =) |
| 10:03.05 | mchou | |stefan|: direct to voicemail with no rings generally means that |
| 10:03.17 | lftsy | thanks tzafrir ! |
| 10:03.39 | |stefan| | mchou: heh yea =) |
| 10:04.06 | |stefan| | but it shouldn't do it judging from those lines right ? |
| 10:04.13 | mchou | post the relevant lines in pastebin |
| 10:04.16 | |stefan| | the dial command should finish it's 60 seconds |
| 10:05.11 | mchou | if there is no device it could also go straight to voicemail |
| 10:05.13 | TJNII | I'd rather see a pastebin of the console output, with verbosity >3. |
| 10:05.57 | |stefan| | should be though. dunno why ext100 would've stopped working. |
| 10:06.04 | |stefan| | here's the relevant code |
| 10:06.06 | |stefan| | http://pastebin.ca/1654584 |
| 10:06.55 | mchou | lol |
| 10:08.24 | |stefan| | -- SIP/ext100-b7536398 is circuit-busy |
| 10:08.29 | |stefan| | ye well. not much to think about. |
| 10:08.53 | |stefan| | the ata is probably frozen |
| 10:09.43 | mchou | |stefan|: what brand ATA is this? |
| 10:10.09 | |stefan| | oh wth is it now. voodoo something ? |
| 10:10.23 | mchou | voodoo??? |
| 10:10.25 | |stefan| | i'll just restart it when i get home =) |
| 10:10.51 | mchou | I dont think I'd ever trust an ATA with a name like Voddoo |
| 10:10.55 | |stefan| | heh =) well i don't remember |
| 10:11.04 | mchou | umm, Voodoo |
| 10:11.29 | TJNII | |stefan|: core show channels doesn't show it having an open call, does it? |
| 10:11.51 | |stefan| | nop |
| 10:12.18 | mchou | these is one easy way to find out if ATA is still working |
| 10:12.25 | |stefan| | son might even have left the phone off the hook =) |
| 10:12.45 | mchou | if it's frozen it wont be registering in 30 minutes |
| 10:12.47 | mchou | :) |
| 10:12.52 | |stefan| | it's no problem guys =) i'll get it to work when i get home =) |
| 10:13.04 | TJNII | If it was frozen it wouldn't reply that is is busy. |
| 10:13.20 | |stefan| | i have some grandstream ata in ma drawer otherwise |
| 10:13.27 | mchou | lol |
| 10:13.39 | mchou | not much of an improvement |
| 10:13.53 | |stefan| | if one is broken and the other works it is =) |
| 10:15.33 | |stefan| | is there any fresh dialplan documentation for asterisk 1.6 ? |
| 10:15.59 | mchou | Voodoo. What a great name for an ATA though |
| 10:16.30 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 10:16.37 | mchou | Chant magical incantations to make sure it works :) |
| 10:16.39 | DND | guys. does a single call really consules 2 channels? |
| 10:16.48 | DND | *consumes |
| 10:17.27 | mchou | depends what you mean by channels |
| 10:17.28 | DND | oh sorry its for freepbx |
| 10:17.47 | tzafrir | "Any sufficiently advanced technology is indistinguishable from Voodoo" |
| 10:17.49 | DND | because in freepbx gui, it says one external call but there's 2 channels used |
| 10:18.19 | |stefan| | well call comes into ur pbx and then your pbx opens a new "channel" to wherever you're directing the call ? |
| 10:18.41 | mchou | yu, the 2nd leg |
| 10:18.42 | tzafrir | DND, 'core show channels' would probably claim the same |
| 10:18.51 | mchou | of the B2BUA |
| 10:20.21 | DND | here are the two lines in core show channels: DAHDI/1-1 (None) Up AppDial((Outgoing Line)) and the second line: SIP/3111-0e7361c0 s@macro-dialout-trun Up Dial(DAHDI/g0/050XXXXXXX|300|) |
| 10:21.02 | DND | is that normal? |
| 10:22.06 | DND | cause i thought when you say channels, its incoming and outgoing |
| 10:22.48 | |stefan| | mchou: just remembered the ata name =) i3Micro Vood |
| 10:25.42 | mchou | |stefan|: ohhm, they need to be spanked |
| 10:25.55 | mchou | |stefan|: GPL violators |
| 10:26.15 | |stefan| | yeh well =) it works |
| 10:26.25 | |stefan| | mostly =) |
| 10:26.40 | mchou | the company is apparently defunct too |
| 10:27.03 | mchou | sigh |
| 10:27.16 | *** join/#asterisk fofware (n=chatzill@190.7.25.160) |
| 10:27.27 | DND | low? |
| 10:27.47 | mchou | I just don't understad why all these voip companies are dying like flies |
| 10:28.02 | *** join/#asterisk destructure (n=mu@couchdb/user/destructure) |
| 10:28.43 | mchou | totally fly-by-night operations |
| 10:29.44 | |stefan| | so. what's a good simple cheap ata then ? (that works in eu) |
| 10:30.20 | mchou | linksys PAP2T? |
| 10:30.53 | mchou | several (localized) variants |
| 10:31.11 | gr0mit | yup. these are good |
| 10:31.15 | TSM2 | stefan: what do you want to do? |
| 10:31.19 | mchou | spa2xxx? |
| 10:31.24 | TSM2 | if its faxing then use SPA2102 |
| 10:31.32 | TSM2 | if just calls then PAP2T |
| 10:31.49 | TSM2 | i have all the docs on how to TFTP set them up |
| 10:32.53 | *** join/#asterisk fofware (n=chatzill@190.7.25.160) |
| 10:33.03 | |stefan| | TSM2: just calls =) |
| 10:33.41 | mchou | I'm surprised faxing even works |
| 10:35.23 | ppc | yo |
| 10:36.10 | |stefan| | ye actually seems like a good alternative |
| 10:36.16 | |stefan| | thanks for the tip |
| 10:38.15 | mchou | |stefan|: get it from a reputable place |
| 10:38.26 | mchou | i.e. not fleabay |
| 10:38.42 | |stefan| | i don't really buy used stuff |
| 10:38.55 | |stefan| | i have too much crap anyway |
| 10:38.57 | *** join/#asterisk baijum (n=baiju@122.166.46.113) |
| 10:39.29 | |stefan| | just gave away like 4 bags of old ide/power/lpt cables and old cd/dvd roms and floppys |
| 10:39.46 | mchou | just letting you know there are lot's of "unlocked" versions out there that are really unlocked |
| 10:40.04 | mchou | aren't* |
| 10:40.23 | |stefan| | well if i buy if from a store it should be unlocked ? |
| 10:40.32 | |stefan| | buy it* |
| 10:40.56 | *** join/#asterisk shinao1 (n=shinao1@41.219.197.48) |
| 10:41.16 | mchou | in the US we generally dont see this from stores |
| 10:41.29 | mchou | only online distributors |
| 10:41.43 | mchou | YMMV in Europe |
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| 11:10.13 | *** join/#asterisk AllGoodNiksTaken (n=sean@160.77-136-217.adsl-dyn.isp.belgacom.be) |
| 11:10.26 | AllGoodNiksTaken | Hello! |
| 11:10.45 | AllGoodNiksTaken | Are there any Asterisk gurus home? |
| 11:12.13 | AllGoodNiksTaken | I've everything up & running on Asterisk 1.6.0.15 however when I perform an 'Originate' the billsec is always 0 and disposition='NO ANSWER'. |
| 11:12.59 | AllGoodNiksTaken | I've checked asterisk issue 14844 (Asterisk call file has CDR always set to NO ANSWER) which was closed as 'unable to reproduce' |
| 11:13.17 | AllGoodNiksTaken | however I am having this same problem as reported in the (now closed) issue. |
| 11:13.40 | AllGoodNiksTaken | Is anyone available to lend a helping hand? |
| 11:13.51 | *** join/#asterisk Gido-E (n=gido@lounge.datux.nl) |
| 11:19.01 | angryuser | AllGoodNiksTaken, do you absolutely need 1.6.0.15 ? |
| 11:19.18 | AllGoodNiksTaken | Hi Angryuser.. no I dont :) |
| 11:19.44 | AllGoodNiksTaken | I installed it after looking at the buglist in mantis and thought it might work better thank 1.4 |
| 11:19.48 | AllGoodNiksTaken | how do you think? |
| 11:20.02 | angryuser | AllGoodNiksTaken, try latest |
| 11:20.30 | AllGoodNiksTaken | ah ok, i'm using the yum version so I'll uninstall that and install the rpm instead. |
| 11:21.05 | *** join/#asterisk KermitTheFragger (n=KermitTh@118-197.bbned.dsl.internl.net) |
| 11:21.09 | angryuser | AllGoodNiksTaken, dont use rpm compile by yourself |
| 11:21.36 | AllGoodNiksTaken | <angryuser> ok :) |
| 11:21.39 | *** join/#asterisk Sajam (n=chatzill@beta.intelligile.com) |
| 11:22.51 | AllGoodNiksTaken | angryuser, thanks for your help. |
| 11:22.55 | *** join/#asterisk Sajam (n=sajam@beta.intelligile.com) |
| 11:23.34 | angryuser | AllGoodNiksTaken, you knew the answers, all you needed is a little kick in your ass |
| 11:23.54 | angryuser | AllGoodNiksTaken, you are welcome |
| 11:23.58 | *** join/#asterisk razu (n=razu@195.222.1.150) |
| 11:24.16 | *** join/#asterisk nighty^ (n=nighty@x122091.ppp.asahi-net.or.jp) |
| 11:24.25 | razu | does anyone use Teles iSwitch here ? |
| 11:24.50 | AllGoodNiksTaken | angryuser.. lol, thats true. |
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| 11:32.15 | *** join/#asterisk Skavin (n=kevins@cpsak1-r6.tranzpeer.net) |
| 11:34.26 | Skavin | Is there any way to break up the mailbox dirs in /var/spool/asterisk/voicemail/default/ ? |
| 11:35.43 | Skavin | into sub dirs of default based on last 2 digits or somthing would make managing the boxes easyer in large installs |
| 11:41.57 | *** join/#asterisk chasing`Sol (n=rc4@217.54.176.183) |
| 11:43.37 | DND | Skavin, i need that also |
| 11:44.16 | DND | its a pain looking at all the wav files in one folder |
| 11:44.33 | Skavin | yea playing with that dir at over 4000 users |
| 11:45.20 | DND | i think it should be configurable i just dont know where it is |
| 11:45.29 | Skavin | think it may be slowing asterisk down as using ext3 even an ls takes forever |
| 11:45.59 | DND | hmm i didnt know the FS affects performance |
| 11:46.07 | DND | *FS = file system |
| 11:47.21 | florz | Skavin: possibly your FS doesn't have directory indexing active? |
| 11:47.25 | *** join/#asterisk kombi (n=kombi@port-92-198-15-96.static.qsc.de) |
| 11:47.32 | kombi | has anyone managed to provide an internal euro-isdn s0 bus over a digium b410p? |
| 11:49.18 | kombi | should be possible with dahdi, just wonder how.. |
| 11:51.02 | tzafrir | Skavin, what filesystem do you use there? |
| 11:51.14 | Skavin | ext3 |
| 11:51.30 | tzafrir | just use dir_index or whatever it is called |
| 11:52.47 | tzafrir | But to answer your question: if the code of asterisk still uses a 'mkdir -p' to generate that mailbox directory, maybe it's only a matter of a simple patch |
| 11:53.07 | tzafrir | check apps/app_voicemail.c |
| 11:53.28 | Skavin | yea I may look at trying to create a patch |
| 11:53.34 | florz | yeah, dir_index is the fs option name |
| 11:53.54 | Skavin | was reading it before .... trying to track it is interesting |
| 11:53.58 | tzafrir | kombi, what do you mean? What should be the problem? |
| 11:55.03 | Skavin | indexing discribed here http://wiki.archlinux.org/index.php/Ext3_Filesystem_Tips#Using_Directory_Indexing |
| 11:55.49 | *** join/#asterisk jbw (n=jbw@dsl-105-162.cust.imagine.ie) |
| 11:56.32 | kombi | got an asterisk box working here in germany which is hooked up to pstn via isdn over a b410. We now need an isdn s0 bus to connect some special hardware. Since there are free RJ45s in the b410, I was thinking maybe on could be configured to do that |
| 11:56.45 | kombi | sorry, that was for tzafrir.. |
| 11:57.33 | tzafrir | It should work |
| 11:57.38 | tzafrir | IIRC |
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| 12:06.39 | kombi | thanks tzafrir, I'll delve into the conf files then..;) |
| 12:07.22 | tzafrir | hmm... I guess there's a problem with being PtMP NT |
| 12:07.31 | tzafrir | PTP NT works, though |
| 12:11.04 | ruyo | I've used PTMP NT, it worked. |
| 12:11.09 | *** join/#asterisk arpu (n=arpu@chello062178159144.10.14.univie.teleweb.at) |
| 12:11.18 | ruyo | At least I connected an NT port to a Siemens DECT configured to be PTMP. |
| 12:17.48 | Skavin | DND you using realtime? |
| 12:18.02 | mchou | anyone have experience with LG-Nortel 6812 sip phone? |
| 12:19.26 | Skavin | DND I am going to bed but looking at the code it may allow a / in the mailbox |
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| 12:22.26 | Skavin | will try changing the mailbox field to AB/xxxxxxAB looking at the code it only looks for @ and , |
| 12:25.10 | Skavin | thanks all its 01:30 I have to be up in 4.5 hours :) |
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| 12:37.05 | BentLee | can anyone tell me where to start looking for the cause of my problem: I can successfully dial "normal" geographical numbers, mobile numbers and international numbers via DAHDI on a Yeastar TDM400 clone, but when i dial toll-free or non-geographic numbers Asterisk 1.6.1.6 gives no errors whatsoever but i just get a PSTN dial tone |
| 12:37.34 | BentLee | i am in South Africa, which I suspect affects things |
| 12:39.33 | *** join/#asterisk QaDeS (n=mklaus@213.157.13.70) |
| 12:39.48 | kaldemar | BentLee: look at CLI during a call and see what happens |
| 12:40.13 | BentLee | thanks kaldemar - i will try that |
| 12:41.06 | kaldemar | if you can't figure out what happens, pastebin the output and show it here. someone will most likely take a look at it. |
| 12:43.00 | BentLee | thanks, in the interest of my ongoing education i will try to make sense of it first, but might well be back |
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| 13:03.08 | Chainsaw | In Asterisk 1.6.1.8; can I check for the existence of a queue *without* getting a console warning if the checked queue is not valid? QUEUE_VARIABLES warns, as does the Queue application. |
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| 13:07.38 | *** join/#asterisk ruben23 (n=RPL@121.1.37.147) |
| 13:09.25 | ruben23 | hi can eyebeam softphones register to a dynamic Public IP asterisk, using ddns.. |
| 13:09.28 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 13:17.42 | kaldemar | ruben23: it doesn't matter whether your ip is static or dynamic. so yes. |
| 13:18.38 | ruben23 | <PROTECTED> |
| 13:19.14 | kaldemar | obviously |
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| 14:22.21 | ManxPower-work | ~answers |
| 14:22.30 | infobot | from memory, answers is Asterisk Book: http://astbook.asteriskdocs.org & SIP w/ NAT: http://www.aocomputing.net/?p=3 & Wiki: http://www.voip-info.org & docs: /path/to/src/asterisk/doc & 1.4 .vs. 1.6.0 .vs. 1.6.1: : /path/to/src/asterisk/UPGRADE*.txt |
| 14:24.55 | Katty | good morning |
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| 14:26.12 | [TK]D-Fender | Katty: Mew. |
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| 14:26.44 | [TK]D-Fender | Polycom IP 335 = Everything its predecessors should ahve been : http://www.ichromis.com/blog/?p=1474 |
| 14:27.47 | coppice | [TK]D-Fender: but what exactly are its predecessors? |
| 14:27.57 | Chesther | 330/331 |
| 14:28.17 | coppice | I don't think so. its a lot more expensive than those |
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| 14:28.37 | Katty | hugs coppice |
| 14:28.39 | Katty | hugs _ShrikE |
| 14:28.41 | [TK]D-Fender | coppice: Don't knw the final price, but it should be less than the 450 |
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| 14:30.23 | ManxPower-work | "The great thing about model numbers is there are do many of them." |
| 14:31.10 | coppice | [Tk]D-Fender: they say they are sropping the 330, and keeping the 331. the 335 seems to be 30-40% more expensive |
| 14:34.53 | [TK]D-Fender | coppice: I'll reserve final judgement until it hits retail... no street pricing yet |
| 14:35.12 | coppice | some sites have prices |
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| 14:35.30 | coppice | though its possible those are inflated introductory ones |
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| 14:39.23 | [TK]D-Fender | coppice: Quite likely. the 331 = $102.50. 450 = $176.50. I could accept a 20% hike from 331>335 for the added features |
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| 14:43.18 | Katty | name a vegetable that begins will D |
| 14:43.40 | ManxPower-work | Katty: Daekon |
| 14:43.52 | ManxPower-work | (I'm sure it's not spelled correctly) |
| 14:43.57 | Katty | Daikon? |
| 14:44.01 | ManxPower-work | yeah, that. |
| 14:44.06 | Katty | kind of a white looking carrot |
| 14:44.14 | ManxPower-work | Yes, that's it. |
| 14:44.38 | coppice | [TK]D-Fender: they seem to have hiked the 331. it should be the same as the 330, but maybe it will be when stocks of the 330 run out |
| 14:44.39 | netpro25_ | I recently updated asterisk and now sometimes when I answer the phone I get a buzzing sound. Anyone familiar with this? |
| 14:44.45 | Katty | hmm. |
| 14:44.54 | Katty | i don't think i've ever seen this before, but i will look for it at the grocery store. |
| 14:45.01 | netpro25_ | I am unable to hear anything but the buzzing sound |
| 14:45.12 | Katty | thank you manx. |
| 14:45.23 | ManxPower-work | netpro25_: do you have an analog card? |
| 14:45.31 | netpro25_ | ManxPower-work: nope |
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| 14:45.48 | [TK]D-Fender | ouch |
| 14:46.10 | netpro25_ | it's almost like modem tones |
| 14:46.27 | netpro25_ | but a constant tone |
| 14:47.02 | ManxPower-work | netpro25_: you need to provide more details of your setup. I'm not interested in helping, but no matter who helps you they will need that info |
| 14:47.12 | *** join/#asterisk Naikrovek (n=jjohnson@unaffiliated/naikrovek) |
| 14:47.39 | netpro25_ | Is there a listing of known bugs for certain versions online? |
| 14:47.48 | *** join/#asterisk jtodd (i=s6g5lbx7@asterisk/community-director-and-tie-dye-shirt-lover/jtodd) |
| 14:47.48 | netpro25_ | Btw I have Asterisk 1.4.21 |
| 14:47.48 | *** mode/#asterisk [+o jtodd] by ChanServ |
| 14:47.49 | ManxPower-work | bugs.digium.com |
| 14:48.14 | leifmadsen | issues.asterisk.org |
| 14:48.27 | netpro25_ | thanks |
| 14:48.42 | leifmadsen | bugs.digium.com is the old address, and redirects to issues.asterisk.org |
| 14:48.52 | leifmadsen | netpro25_: 1.4.21 is pretty old now |
| 14:48.59 | ManxPower-work | issues.asterisk.org sounds like a site for online therapy of Asterisk addicts |
| 14:49.19 | netpro25_ | leifmadsen: yes, it's the debian package version |
| 14:49.28 | leifmadsen | yuck |
| 14:49.36 | ManxPower-work | netpro25_: do not ask for support of packaged versions of asterisk |
| 14:49.43 | leifmadsen | well, any issue you report is pretty much going to be closed with, "Please test something more recent" |
| 14:49.51 | netpro25_ | lol |
| 14:50.10 | ManxPower-work | Well you can ask all you want, but the best response you can hope for is people pointing at you and laughing. |
| 14:50.58 | netpro25_ | yea... I will see about updating as this started when I did a package update last time |
| 14:52.24 | Katty | hugs Naikrovek |
| 14:52.30 | Katty | hi mister madsen. |
| 14:52.32 | Katty | hugs leifmadsen |
| 14:52.38 | leifmadsen | awww :) |
| 14:52.39 | leifmadsen | hi! |
| 14:52.42 | Naikrovek | :) |
| 14:53.29 | netpro25_ | Whats the best way to switch from deb to a compiled version. Should I first uninstall the package then compile from source? |
| 14:53.49 | leifmadsen | you can compile, then remove the package before running 'make install' |
| 14:53.57 | tamiel | netpro25_: make your own package =) |
| 14:54.00 | leifmadsen | I'd probably back up your configs too just in case the package decides to remove them |
| 14:54.19 | Katty | ahh, config backups. |
| 14:54.22 | Katty | i do love me some backups. |
| 14:54.23 | netpro25_ | tamiel: I would love to do that, I dunno how though |
| 14:54.58 | Katty | leifmadsen: vegetable that starts with D, and is not Daikon. |
| 14:55.05 | Katty | leifmadsen: what do you think? |
| 14:55.23 | leifmadsen | Katty: I wouldn't have even gotten Daikon :) |
| 14:55.40 | Katty | leifmadsen: me either, but Manx did. |
| 14:55.41 | leifmadsen | dill pickle? :) |
| 14:55.52 | leifmadsen | dragon fruit? or is that a fruit? :D |
| 14:56.05 | ManxPower-work | leifmadsen: I watch too much Iron Chef America |
| 14:56.09 | leifmadsen | apparently |
| 14:56.23 | leifmadsen | I think I want an olive now |
| 14:56.31 | Katty | hrmm. |
| 14:56.38 | Katty | googles dragonfruit |
| 14:56.43 | netpro25_ | What version is stable? |
| 14:56.56 | netpro25_ | 1.6.0? |
| 14:57.10 | ManxPower-work | netpro25_: stay with 1.4.x |
| 14:57.20 | coppice | dragon fruit is delicious |
| 14:57.25 | netpro25_ | k, thanks |
| 14:57.32 | Katty | ah. so it's like a Kiwi |
| 14:57.39 | Katty | except bigger. |
| 14:58.00 | coppice | its nothing like a kiwi. it looks like a flame |
| 14:58.16 | Katty | http://z.about.com/d/thaifood/1/0/l/D/dragronfruitstep8.jpg |
| 14:58.29 | coppice | the chinese name means fire dragon fruit, which is much more descriptive |
| 14:58.37 | Katty | coppice: yeah but the way you prepare it is like a kiwi. |
| 14:58.42 | Katty | coppice: can't eat the skin. |
| 14:58.57 | coppice | its nothing like a kiwi outside or in |
| 14:59.03 | Katty | no? |
| 14:59.15 | Katty | is the fruit skwishy or hard? |
| 14:59.17 | ManxPower-work | Aren't they orange, smooth, and spiky? |
| 14:59.54 | coppice | nope |
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| 15:00.28 | coppice | they are red, smooth, and shaped like a flame |
| 15:00.56 | Katty | k |
| 15:01.01 | diedo | Hello, Iḿ looking for a little help trying to compile OSLEC with DAHDI |
| 15:01.13 | diedo | anyone with experience? |
| 15:02.40 | Katty | ah bummer. dragon fruit is in season come June |
| 15:03.01 | Katty | odds are the store won't have it i bet :< |
| 15:03.21 | [TK]D-Fender | netpro25_: 1.6.0.15 <- |
| 15:03.42 | netpro25_ | [TK]D-Fender: okay |
| 15:03.47 | ruyo | Is there an advised cpu for Asterisk? |
| 15:03.47 | netpro25_ | guess 1.6 it is |
| 15:03.55 | ruyo | Like intel or amd or so. |
| 15:04.04 | [TK]D-Fender | ruyo: "whatever" |
| 15:04.05 | ManxPower-work | netpro25_: if you switch to 1.6 you should read all the UPGRADT*.txt files. |
| 15:04.20 | ruyo | Whatever is nice, thanks. : ) |
| 15:04.23 | coppice | Katty: we have dragon fruit in the kitchen, even as I type |
| 15:04.32 | netpro25_ | ManxPower-work: thanks |
| 15:04.52 | ManxPower-work | netpro25_: there are many incompatible changes between 1.4 and 1.6.x.x |
| 15:04.56 | TheDavidFactor | netpro25_, is this an upgrade or a new install? |
| 15:05.02 | netpro25_ | upgrade |
| 15:05.18 | Katty | coppice: oh? where do you get it? |
| 15:05.49 | netpro25_ | TheDavidFactor: it's a pretty simple install, single user |
| 15:05.52 | netpro25_ | setup |
| 15:06.00 | netpro25_ | two phones |
| 15:06.15 | coppice | Katty: The supermarket just to the right of http://www.coppice.org/DiscoveryBay.jpg |
| 15:06.16 | TheDavidFactor | then, yes you should read the upgrade documents as Manx said, but I'm running a mix of 1.4.x and 1.6.x and I like 1.6.x however there are differences in the dialplan apps and config |
| 15:06.42 | netpro25_ | okay |
| 15:06.51 | Katty | coppice: is it a chain? :> |
| 15:07.39 | coppice | quite a big chain. Park'n'Shop |
| 15:09.05 | Katty | hmmmmmmmmmmmk. |
| 15:09.10 | eppigy | ALLO |
| 15:09.39 | Katty | herroes. |
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| 15:12.06 | Katty | bummer. |
| 15:12.10 | Katty | none in the states. |
| 15:13.21 | Katty | oh well. i'll find some eventually (= |
| 15:13.37 | Katty | whole foods up in st. louis might carry it. |
| 15:14.30 | Katty | eppigy: i've decided to do stuffed acorn squash for thanksgiving this year. |
| 15:14.44 | Katty | eppigy: i'm gonna stuff it with turkey and dressing. and then pour some sort of a cranberry compote on the top. |
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| 15:15.08 | mbrevda | is anyone aware of an issue in 1.4.26.2 where asterisk doenst show the sip registrations? |
| 15:16.06 | Katty | eppigy: http://tinyurl.com/y9wfuwt |
| 15:16.17 | Katty | eppigy: it'll look something like that. |
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| 15:33.10 | ZPertee | is looking for ideas for asterisk/google wave integration |
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| 15:54.46 | docelmo | Say anyone know if AMI produces any good logging? I am trying to figure out why when I send a message to AMI it keeps telling me originate failed. |
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| 15:57.51 | diedo | tzafrir? |
| 15:58.42 | tzafrir | here |
| 15:59.10 | *** join/#asterisk |Cybex| (n=John@atwork-20.r-212.178.82.atwork.nl) |
| 16:06.58 | *** join/#asterisk jmworx__ (n=jeval@mail.octasic.com) |
| 16:08.35 | *** join/#asterisk |Cybex| (n=John@atwork-20.r-212.178.82.atwork.nl) |
| 16:09.08 | *** join/#asterisk moos3 (n=rgenthne@216.52.121.66) |
| 16:09.23 | moos3 | how can I tell how many calls are in place? |
| 16:09.44 | russellb | *CLI> core show channels |
| 16:09.48 | *** join/#asterisk aidinb (n=Aidin@24-182-32-138.static.lnbh.ca.charter.com) |
| 16:10.39 | *** join/#asterisk ZPertee (n=ZPertee@rrcs-74-219-221-11.central.biz.rr.com) |
| 16:11.31 | *** join/#asterisk ajohnson (n=ajohnson@65-122-4-130.dia.static.qwest.net) |
| 16:16.44 | *** join/#asterisk jtodd (i=wnhlajz7@asterisk/community-director-and-tie-dye-shirt-lover/jtodd) |
| 16:16.45 | *** mode/#asterisk [+o jtodd] by ChanServ |
| 16:17.44 | *** join/#asterisk sgimeno (n=sgimeno@226.Red-80-33-64.staticIP.rima-tde.net) |
| 16:18.10 | *** join/#asterisk ruyo (n=sayo@195-23-253-223.net.novis.pt) |
| 16:20.43 | *** join/#asterisk Caplain (i=shayne@2001:470:5:fb:20ee:7162:c8e4:6f63) |
| 16:21.05 | *** join/#asterisk Tim_Toady (n=moi@adsl139-47.kln.forthnet.gr) |
| 16:25.31 | moos3 | what does this mean 2009-11-03 11:23:51] WARNING[25537]: file.c:644 ast_openstream_full: File queue-minute does not exist in any format |
| 16:26.16 | chazzm | looks like that file 'queue-minute' doesn't exist |
| 16:26.31 | moos3 | where should I check for that file to begin iwht |
| 16:26.34 | Chainsaw | moos3: It's trying to play a file called 'queue-minute'. I do have a file called queue-minutes though. |
| 16:26.38 | ChannelZ | it's a sound you're trying to play I think |
| 16:26.43 | Chainsaw | moos3: Perhaps a typo in your config? |
| 16:26.56 | moos3 | I'll look |
| 16:27.18 | chazzm | '/var/lib/asterisk/sounds' is the default location |
| 16:27.52 | *** join/#asterisk Naikrovek (n=jjohnson@unaffiliated/naikrovek) |
| 16:28.53 | moos3 | yeah I found the correct file you said but I can't find it in my extensions.conf |
| 16:29.31 | Chainsaw | moos3: Hm, okay. Did you copy/paste that warning message in or did you type it over? |
| 16:30.01 | ChannelZ | is it being called from an AGI script or something? |
| 16:30.02 | moos3 | cut and paste |
| 16:30.09 | moos3 | [2009-11-03 11:23:51] WARNING[25537]: file.c:950 ast_streamfile: Unable to open queue-minute (format 0x4 (ulaw)): No such file or directory |
| 16:30.16 | moos3 | nothing I know of |
| 16:30.49 | *** join/#asterisk |Cybex| (n=John@atwork-20.r-212.178.82.atwork.nl) |
| 16:31.15 | ChannelZ | I see a queue-minutes (plural) in my install, not queue-minute |
| 16:31.20 | *** join/#asterisk niekie (i=quasselc@dreamworld.bergnetworks.com) |
| 16:31.42 | moos3 | yeah same here |
| 16:31.59 | moos3 | we moved from a old server to a brand new install and moved all the configs over |
| 16:32.04 | ChannelZ | that said it's part of call queueing which isn't in your extensions, it's handled by that module.. so unless you're playing that file specifically yourself somewhere... |
| 16:32.33 | ChannelZ | look at your queues.conf |
| 16:33.00 | ChannelZ | The sound files it uses are configured there |
| 16:33.37 | *** join/#asterisk diatonic1 (n=diatonic@mail.clearwater-research.com) |
| 16:34.42 | [TK]D-Fender | [11:31]<moos3>we moved from a old server to a brand new install and moved all the configs over <- configs, yes... sounds, apparently not |
| 16:35.00 | *** join/#asterisk TimToady_ (n=moi@adsl23-102.kln.forthnet.gr) |
| 16:35.53 | moos3 | i rsyncd the sounds |
| 16:36.03 | Chainsaw | moos3: Did you upgrade from Asterisk 1.2 to 1.6? |
| 16:36.14 | Chainsaw | moos3: If you did, the sound files are now in different places. |
| 16:36.20 | moos3 | on 1.6.0 to 1.6.0.15 |
| 16:36.47 | dlynes | moos3: they are? |
| 16:36.50 | dlynes | erm |
| 16:36.51 | Chainsaw | transfers the call back to Fender |
| 16:36.54 | dlynes | Chainsaw: they are? |
| 16:37.07 | *** join/#asterisk neurosys (n=neurosys@173-9-159-182-miami.txt.hfc.comcastbusiness.net) |
| 16:37.14 | Chainsaw | dlynes: Yes, 1.2 used multiple en_uk subdirs. |
| 16:37.24 | Chainsaw | dlynes: 1.6 has one en_uk directory with subdirs for the different type of sounds. |
| 16:37.27 | dlynes | Chainsaw: oh...you mean for language folders |
| 16:37.31 | Chainsaw | dlynes: Exactly. |
| 16:37.32 | dlynes | Chainsaw: thought you meant in general |
| 16:37.40 | Chainsaw | dlynes: No, just that. But it caught me out as well. |
| 16:38.02 | dlynes | Chainsaw: ah...never used the Set(LANGUAGE=...) command |
| 16:38.16 | moos3 | my queue.conf is completely commented out |
| 16:38.17 | Chainsaw | dlynes: It has a fallback advantage. |
| 16:38.32 | Chainsaw | dlynes: If a sound is missing in our en_uk set, it'll fall back to the american lady. |
| 16:38.50 | dlynes | Chainsaw: yeah...but then I'd have to phone up allison and ask her to start talking Canadian, eh? |
| 16:39.03 | dlynes | too much hassle |
| 16:39.09 | *** join/#asterisk ryduh (n=ryduh@204.16.143.186) |
| 16:39.12 | dlynes | easier just to leave it on the american english |
| 16:39.20 | Chainsaw | We didn't really like the accent. |
| 16:39.31 | dlynes | besides...Canadians are already used to American accents |
| 16:39.41 | Chainsaw | And in the US you say "pound key" where we say "hash key". |
| 16:39.47 | Chainsaw | Couple more issues like that. |
| 16:39.59 | dlynes | Chainsaw: and in Canada, we say hash key, pound key and number key |
| 16:40.54 | *** join/#asterisk Davedan (n=me@CBL217-132-64-6.bb.netvision.net.il) |
| 16:41.11 | dlynes | english is a wonderful language |
| 16:41.18 | dlynes | nice and confusing to outsiders :) |
| 16:43.25 | Chesther | "English doesn't borrow words from other languages. English follows other languages down dark alleys, knocks them unconscious, and rifles through their pockets for loose grammar." |
| 16:45.50 | superbeef1 | when I look at an IAX debug, what does RR_LOSS actually mean |
| 16:46.25 | superbeef1 | loss is a great guess, but what exactly did it loose |
| 16:48.25 | Katty | so. i have a problem. |
| 16:48.33 | Katty | and maybe one of you fine people can point me in the right direction. |
| 16:48.41 | [TK]D-Fender | Katty: #drphil |
| 16:48.46 | Katty | how did you know |
| 16:48.50 | Katty | :< |
| 16:49.03 | Katty | MAYBE I HAVE A REAL PROBLEM |
| 16:49.16 | Katty | ...or maybe it really is about refined table sugar. |
| 16:49.29 | Katty | shakes fist at [TK]D-Fender |
| 16:50.30 | *** join/#asterisk SirThomas (n=tomc@mail.kendeco.com) |
| 16:51.23 | superbeef | how many problems can refinded table sugar cause |
| 16:51.35 | superbeef | refined |
| 16:51.55 | *** join/#asterisk Defraz (n=Defraz@corp.fuzecore.com) |
| 16:53.02 | Chesther | It depends on how many gas tanks you pour it into. |
| 16:53.03 | ryduh | superbeef: you don't know!?! |
| 16:53.39 | ManxPower-work | *heart* sugar in all it's glorious forms. |
| 16:53.39 | superbeef | nope.. i just figured it was less entertaining since its in a white packet |
| 16:53.40 | Katty | i'm just curious to know if there's some sort of REAL sugar i can use to make cranberry sauce that was not created in a lab. |
| 16:53.59 | Katty | please do not suggest Splenda, Aspartame or any of that stuff. |
| 16:54.13 | superbeef | Katty: how abotu the bag that says pure cane sugar |
| 16:54.13 | Katty | or table refined table sugar...or high fructose corn syrup...they are both built in a lab. |
| 16:54.23 | [TK]D-Fender | Katty: Raw sugar |
| 16:54.33 | *** join/#asterisk Badrobot- (n=badrobot@cpe-76-173-229-89.socal.res.rr.com) |
| 16:54.33 | Chesther | Refined table sugar is not made in a lab, but it is refined in a factory. |
| 16:54.46 | Chesther | But, yeah, raw sugar is probably what you're looking for. |
| 16:55.02 | Katty | raw sugar = pure cane sugar? |
| 16:55.55 | Chesther | More or less, yes. |
| 16:55.58 | *** join/#asterisk abradley (n=Data@72.12.1.38) |
| 16:56.09 | Chesther | Short of processing your own sugarcane, it's probably the closest you're going to get. |
| 16:56.40 | Katty | so raw sugar is just raw sugar, not require to be from sugar cane. |
| 16:56.42 | moos3 | if I use the queue.conf will it override the sounds anywhere else? |
| 16:56.43 | Katty | it could be from beets... |
| 16:57.03 | Chesther | No, because beets pretty much require processing. |
| 16:57.25 | Chesther | Check the package, but generally raw sugar is cane sugar, AFAIK |
| 16:57.33 | Katty | mmkay. |
| 16:57.36 | Katty | good ot know. |
| 16:57.39 | superbeef | Katty: i think you're making this too complicated.. just go buy hte big dumb bag of cane sugar |
| 16:58.02 | abradley | new install of AN, it boots to "localhost login:". I login with root and after I'm not sure what to do. How do I access the nic settings so that I can give it a static ip for the purpose of logging into the webmin |
| 17:00.41 | *** join/#asterisk ruyo (n=sayo@195-23-253-223.net.novis.pt) |
| 17:00.42 | diatonic1 | abradley: Not sure what AN is, but maybe 'system-config-network' or 'netconfig' ... or find ifcfg-eth0 and edit that file directly |
| 17:00.45 | [TK]D-Fender | abradley: AN is a custom build off of CentOS which = RHEL. |
| 17:00.45 | ManxPower-work | Katty: Honey |
| 17:00.56 | [TK]D-Fender | abradley: Go read up the basic admin info for it |
| 17:00.57 | *** join/#asterisk e4 (n=e4@rrcs-76-79-48-214.west.biz.rr.com) |
| 17:00.58 | abradley | AN = asteriskNow |
| 17:01.10 | ManxPower-work | ~asterisknow |
| 17:01.11 | infobot | somebody said asterisknow was based on Asterisk, but is difficult to support in #asterisk for a number of reasons. Please seek support in #asterisknow instead. |
| 17:01.20 | [TK]D-Fender | ManxPower-work: No need to send him there. |
| 17:01.26 | GameGamer43 | abradley: /etc/sysconfig/network-scripts/ifcfg-eth0 |
| 17:02.44 | *** join/#asterisk VaGoNeTaS (n=nobody@200.111.138.170) |
| 17:02.45 | diatonic1 | abradley: make sure that file ^ has BOOTPROTO=none IPADDR=192.168.0.1 NETMASK=255.255.255.0 ONBOOY=yes then 'service network restart' |
| 17:02.53 | VaGoNeTaS | hello everybody |
| 17:02.55 | Katty | ManxPower-work: i'm not sure that cooking cranberries in honey is ...well... |
| 17:03.02 | Katty | ManxPower-work: i mean it /might/ work, but i don't think it'd really turn out the same. |
| 17:03.06 | ManxPower-work | GameGamer43: you assume someone using Asterisk Now knows how to edit a file. |
| 17:03.20 | diatonic1 | abradley: Substitute your actual values, maybe add GATEWAY=x.x.x.x if you neet to route |
| 17:03.21 | ManxPower-work | Katty: do a test batch |
| 17:03.27 | VaGoNeTaS | i gotta a question, what package do i need to install cdr_pgsql ???? |
| 17:03.27 | Katty | okay. i can do that. |
| 17:03.33 | VaGoNeTaS | when im compilying asterisk |
| 17:03.56 | ManxPower-work | Katty: A good idea when you are cooking anything new or making a change to an existing receipe |
| 17:03.58 | Carlos_PHX | So...anyone been involved with an Asterisk system that can do around 1 million calls in about an hour? |
| 17:04.01 | VaGoNeTaS | i've postgresql 8-3 already installed on the server |
| 17:04.17 | *** join/#asterisk AllGoodNiksTaken (n=sean@160.77-136-217.adsl-dyn.isp.belgacom.be) |
| 17:04.53 | VaGoNeTaS | anyone knows? |
| 17:04.57 | Katty | ManxPower-work: nah, i just make it ...if i don't like it, riddick eats it ;) |
| 17:05.07 | ryduh | Katty: http://www.sugarintheraw.com/ |
| 17:05.14 | VaGoNeTaS | Depends on: pgsql(E) |
| 17:05.41 | AllGoodNiksTaken | Hey y'all, I hope you are well. |
| 17:05.47 | Katty | ryduh: thank you dear. |
| 17:05.52 | russellb | VaGoNeTaS: install the devel package, as well. Also, re-run the configure script. |
| 17:06.04 | ryduh | Katty: I love that stuff and I even used it in my coffee today |
| 17:06.28 | AllGoodNiksTaken | I installed Asterisk from source (latest) and all is working except outbound CDR always shows billsec=0 and disposition = NO ANSWER. Any ideas why this is happening? Been battling this all day :( |
| 17:08.04 | Katty | oh boy, it's vegan. |
| 17:08.06 | Katty | gotta love that. |
| 17:08.54 | TSM2 | is there any good SIP client for OSX apart from Xlite, ie a built in one? |
| 17:09.25 | Chainsaw | Ekiga may have an OS X port these days. |
| 17:09.37 | ryduh | TSM2: I doubt a built in one. What about Voiper |
| 17:10.24 | *** join/#asterisk svm_invictvs (n=patrick@unaffiliated/svminvictvs/x-938456) |
| 17:10.29 | svm_invictvs | Man |
| 17:10.31 | VaGoNeTaS | russellb : installed libpq-dev, problem solved |
| 17:10.34 | VaGoNeTaS | thank you buddy |
| 17:10.37 | VaGoNeTaS | cya later guys |
| 17:10.49 | svm_invictvs | So I had an asterisk box set up and then it just quit working. |
| 17:11.00 | svm_invictvs | I can't, for the life of me, figure out why |
| 17:11.06 | svm_invictvs | I just try to restart the server and the whole thing doesn't start up. |
| 17:11.11 | p3nguin | tsm2: zoiper works on Mac OS X, but it will have to be downloaded and installed. |
| 17:11.15 | svm_invictvs | And the logs aren't very helpful :( |
| 17:11.17 | Chainsaw | svm_invictvs: Share its excuse with us in less then 3 lines. |
| 17:11.23 | Chainsaw | svm_invictvs: Or a pastebin. |
| 17:11.35 | svm_invictvs | Chainsaw: Gonna post a dump of the logs, sec. |
| 17:12.09 | *** join/#asterisk TiToyz (n=TiToyz@aut75-5-82-239-181-57.fbx.proxad.net) |
| 17:12.32 | svm_invictvs | http://www.pastebin.ca/1655049 |
| 17:12.52 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 17:13.41 | AllGoodNiksTaken | If you do a login & originate on a SIP channel from phpagi is it possible to have an ANSWERED value in the disposition column of the CDR? I haven't seen this work. |
| 17:14.18 | *** part/#asterisk TiToyz (n=TiToyz@aut75-5-82-239-181-57.fbx.proxad.net) |
| 17:14.49 | svm_invictvs | Chainsaw: :-/ I checked the permissions of the files. |
| 17:14.58 | svm_invictvs | Why doesnt' asterisk tell me where/what it wouldn't read? |
| 17:15.19 | *** join/#asterisk wonderworld (n=ww@ip-62-143-22-226.unitymediagroup.de) |
| 17:15.39 | p3nguin | svm_invictvs: core set verbose 10 |
| 17:15.43 | p3nguin | svm_invictvs: then try again. |
| 17:15.47 | *** join/#asterisk ecrane (n=ecrane@o1-69-19-166-10.static.o1.com) |
| 17:15.50 | ecrane | j #asterisk-dev |
| 17:15.55 | svm_invictvs | p3nguin: ah |
| 17:15.58 | ecrane | dang |
| 17:16.43 | svm_invictvs | p3nguin: What file do I add that to? |
| 17:16.59 | p3nguin | svm_invictvs: You have to run that in the Asterisk console. |
| 17:17.11 | svm_invictvs | p3nguin: Asterisk isn't even starting up. |
| 17:17.53 | p3nguin | svm_invictvs: Then fix all the files that are listed there and see if that helps. Normally, your files will be under /etc/asterisk/ |
| 17:18.09 | svm_invictvs | p3nguin: yeah, that would be great if asterisk would start up. |
| 17:18.22 | svm_invictvs | p3nguin: Can I just add a verbose option to the init script? |
| 17:18.26 | p3nguin | svm_invictvs: Then fix all the files that are listed there and see if that helps. Normally, your files will be under /etc/asterisk/ |
| 17:18.40 | svm_invictvs | p3nguin: yeah, I had a bunch of custom configurations hacked up. |
| 17:18.58 | kaldemar | svm_invictvs: start asterisk with asterisk -vvvvvvc and see where it dies |
| 17:19.23 | svm_invictvs | p3nguin: And they worked fine for a while, then just quit working. I dunno if I accidentally touched one during an update. *BUT* asterisk won't even start up so I dont' see how accessing hte console to turn on verbose logging will help if the process doesn't even fucking start. |
| 17:19.29 | svm_invictvs | kaldemar: Ah |
| 17:19.50 | moos3 | ok I think I fixed my sound issue but now any call in the queue for more then 2 mins goes to the vm, but I have it set as exten => 1,n,ExecIF($[${csmagents} != 0]?Queue(csmsupportqueue,tT,,,300)) ;Place caller in queue for 5 min (300s) until exiting |
| 17:19.54 | moos3 | ideas on that one |
| 17:20.17 | svm_invictvs | There we go, lots of logging |
| 17:20.30 | ManxPower-work | moos3: queues.conf |
| 17:20.53 | ManxPower-work | or you have too many/few , |
| 17:20.54 | *** join/#asterisk raden_work (n=jon@69-179-99-17.stat.centurytel.net) |
| 17:21.32 | *** join/#asterisk netpro25_ (n=mmanning@64-238-176-248.ksg.apt.gru.net) |
| 17:21.50 | svm_invictvs | Everythign looks good until it get to..."Unable to load config phone.conf" |
| 17:21.51 | moos3 | well there set in the asterisk queues database |
| 17:21.53 | TSM2 | Zopier, what rubish (the free one), xlite i think will be the one |
| 17:22.26 | *** join/#asterisk Circlefusion (n=circlefu@96-28-115-69.dhcp.insightbb.com) |
| 17:23.50 | TSM2 | has anyone use Bria (paid xlite)? |
| 17:24.21 | ManxPower-work | svm_invictvs: "asterisk -cvvv" should indicate where is ACTUALLY fails. |
| 17:24.35 | ManxPower-work | svm_invictvs: I suspect you have zaptel/DAHDI configured and your kernel was upgraded. |
| 17:25.18 | kaldemar | svm_invictvs: do you need chan_phone? |
| 17:25.22 | *** join/#asterisk Malkor (n=marco@hlle-d9ba026a.pool.mediaWays.net) |
| 17:25.24 | netpro25_ | Is there any change in 1.6 regarding SIP registration? |
| 17:25.40 | svm_invictvs | kaldemar: I don't think so, no |
| 17:25.58 | Chainsaw | netpro25_: Yes, insecure=very no longer exists. |
| 17:26.14 | svm_invictvs | kaldemar: No, this is a VoIP box running in a slice host, it has no physical hardware at all. |
| 17:26.22 | kaldemar | svm_invictvs: put "noload => chan_phone.so" in /etc/asterisk/modules.conf and try to start it again |
| 17:26.46 | netpro25_ | Chainsaw: how do you work around that? |
| 17:27.02 | ryduh | svm_invictvs: are you running it at slicehost.com ? |
| 17:27.06 | svm_invictvs | kaldemar: Yep, I did that and it's all gravy. |
| 17:27.12 | Chainsaw | netpro25_: You change insecure=very to insecure=port,invite |
| 17:27.31 | netpro25_ | to "insecure=port,invite"?? |
| 17:27.40 | Chainsaw | netpro25_: Indeed. |
| 17:27.45 | netpro25_ | Chainsaw: thanks |
| 17:28.01 | Chainsaw | netpro25_: There are more changes, but it will take the old values under protest and log what you should be using instead. |
| 17:28.22 | netpro25_ | great |
| 17:28.22 | netpro25_ | thanks |
| 17:28.25 | netpro25_ | it works now |
| 17:28.56 | netpro25_ | Chainsaw: thanks got it working now, will look at the other stuff later |
| 17:29.02 | svm_invictvs | ryduh: no |
| 17:29.14 | Chainsaw | netpro25_: Good luck. "core set verbose 10" is your friend if it fails to admit what's going on. |
| 17:30.00 | ManxPower-work | netpro25_: DUDE! I TOLD you to read UPGRADE*.txt |
| 17:30.53 | svm_invictvs | Chainsaw: It all works now. |
| 17:31.05 | svm_invictvs | Chainsaw: I can't call out, but it forwards to my cell phone, that's the important thing. |
| 17:31.55 | svm_invictvs | "core set verbose 10" |
| 17:32.01 | svm_invictvs | "No such command core"? |
| 17:32.24 | jblack | what verson of asterisk are you running? |
| 17:33.19 | carrar | 1.0 |
| 17:33.24 | svm_invictvs | jblack: oh boy don't laugh |
| 17:33.34 | svm_invictvs | 1.2 |
| 17:33.37 | svm_invictvs | FUCK |
| 17:33.47 | p3nguin | jblack: Looks like the phone itself is simply doing a call return on the last number that the phone received, so that's why it is not sent to asterisk. My *69 exten is to check the last call into asterisk. So now we have the explanation. |
| 17:33.56 | *** join/#asterisk errotan (n=errotan@a1711.adsl.pool.eol.hu) |
| 17:34.20 | [TK]D-Fender | svm_invictvs: Just drop "core" |
| 17:34.23 | jblack | so it has a hardcoded dialplan. |
| 17:36.00 | p3nguin | Yeah, in addition to the one that is user-changeable, there has to be something that is hidden. |
| 17:37.18 | jblack | I can't find it in me to be deeply surprised. |
| 17:39.39 | svm_invictvs | [TK]D-Fender: So that'll change the verbosity in the logs? |
| 17:40.04 | kaldemar | svm_invictvs: no, in the CLI you're looking at |
| 17:40.13 | svm_invictvs | ah |
| 17:40.28 | *** part/#asterisk ManxPower-work (n=EWieling@24.42.221.26) |
| 17:41.25 | svm_invictvs | Now it all works for some reason. |
| 17:41.26 | svm_invictvs | Awesome |
| 17:41.30 | *** join/#asterisk albertoandrade (n=albertoa@189.4.48.141) |
| 17:43.06 | *** join/#asterisk BillyCrook (n=BillyCro@mars.advancedclustering.com) |
| 17:43.43 | BillyCrook | A very long time ago, I was able to telnet to some port on an asterisk server, and get a streaming output of "events" like phones going off hook, calls coming in, etc. |
| 17:44.05 | [TK]D-Fender | ~AMI |
| 17:44.06 | infobot | it has been said that ami is the Asterisk Manager Interface, a way to control an Asterisk server via a TCP/IP socket. See http://voip-info.org/wiki/view/Asterisk+manager+API |
| 17:44.06 | beek | BillyCrook: AMI |
| 17:44.08 | [TK]D-Fender | BillyCrook: ^^^ |
| 17:44.12 | beek | Hi [TK]D-Fender |
| 17:44.27 | [TK]D-Fender | beek: 'lo |
| 17:44.36 | *** part/#asterisk moos3 (n=rgenthne@216.52.121.66) |
| 17:45.08 | beek | [TK]D-Fender: Is there ANYTHING you haven't put into infobot? |
| 17:45.21 | [TK]D-Fender | beek: Lots |
| 17:46.08 | Qwell | ~string theory |
| 17:46.13 | Qwell | pity |
| 17:46.22 | [TK]D-Fender | ~stringtheory |
| 17:46.30 | *** join/#asterisk torrancew (n=torrance@btr.macconsultinggroup.com) |
| 17:46.35 | Qwell | ~string cheese theory |
| 17:46.39 | Kobaz | ~theory |
| 17:46.40 | infobot | theory is, like, clear :) but explain me the segfault i got in the opengl libraries :) |
| 17:46.46 | Qwell | strangely, that would be more likely to exist. |
| 17:47.12 | torrancew | can anyone recommend a good software solution that will either mirror the behavior of phonevalet, or provide growl-style notifications for calls hitting an asterisk box? |
| 17:47.34 | Kobaz | growls at noone in particular |
| 17:48.04 | BillyCrook | What Asterisk Event type do you use for striping higgs bosons out of subatomic particles? |
| 17:48.20 | BillyCrook | ~higgsboson |
| 17:48.36 | ryduh | torrancew: I don't know of anything but you could always make one yourself: http://growl.info/documentation/developer/ |
| 17:49.15 | ryduh | BillyCrook: I use the LHC event type |
| 17:49.21 | torrancew | ryduh: thanks. i'm sure it'd require a custom asterisk app as well, though, right? |
| 17:50.14 | VooDooNOFX | Silly question, but how do I get * to bind to 2 addresses (one internal, one external)? |
| 17:50.31 | [TK]D-Fender | VooDooNOFX: All or one. |
| 17:50.36 | Kobaz | by default asterisk binds to all interfaces |
| 17:50.40 | Qwell | ryduh: Any type of collider will do |
| 17:50.54 | ryduh | torrancew: http://mezzo.net/asterisk/app_notify.html |
| 17:50.56 | Kobaz | VooDooNOFX: have asterisk bind to all... and firewall off the ips you don't want |
| 17:51.11 | VooDooNOFX | so bind=0.0.0.0 will get all available IP's over all available interfaces? |
| 17:51.28 | Kobaz | asterisk will bind to all by *default* |
| 17:51.39 | ryduh | torrancew: scroll down to the Mac OS X client |
| 17:51.53 | torrancew | ryduh: thanks |
| 17:52.16 | ryduh | torrancew: I found it by going here: http://www.google.com/search?q=growl+asterisk&ie=utf-8&oe=utf-8&aq=t&rls=org.mozilla:en-US:official&client=firefox-a :) |
| 17:52.30 | torrancew | is Notify() 1.6 friendly, do you know? |
| 17:52.33 | *** join/#asterisk icyValk77 (n=icyValk7@host81-153-115-41.range81-153.btcentralplus.com) |
| 17:54.15 | ryduh | torrancew: Doesn't look like it has been tested with 1.6 |
| 17:54.30 | VooDooNOFX | another silly q: should I only use the ael, or the conf files, but not both? |
| 17:55.40 | ryduh | torrancew: check this out http://www.voip-info.org/wiki/view/Asterisk+call+notification |
| 17:56.47 | [TK]D-Fender | VooDooNOFX: doesn't matter. |
| 17:57.55 | ryduh | torrancew: I might check this one out http://adm.hamnett.org/ |
| 18:03.34 | *** join/#asterisk mumtazah (n=mumtazah@203.82.91.103) |
| 18:11.24 | *** join/#asterisk robl^laptop (n=robl@c-98-197-98-39.hsd1.tx.comcast.net) |
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| 18:15.46 | momelod | greetings channel |
| 18:16.20 | ryduh | torrancew: Did you try anyof those? |
| 18:16.52 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 18:17.04 | momelod | im currently using cisco phones as sip clients, but if i do a conf call the first call is put on hold, i initiate the second call and join them. But after the calls have been joined i cannot add another number into the conf. how can i enable the option to keep adding in callers into the conf? |
| 18:18.30 | [TK]D-Fender | momelod: You can't |
| 18:18.45 | momelod | boo |
| 18:18.47 | [TK]D-Fender | momelod: this is your phone's built-in 3-way calling. |
| 18:19.03 | [TK]D-Fender | momelod: A cisco phone is nto a "conferencing solution" |
| 18:19.06 | momelod | the same phone could do it before when it was connected to the call manager backend |
| 18:19.07 | [TK]D-Fender | not* |
| 18:19.14 | *** join/#asterisk cheako (n=cheako@97-127-93-82.mpls.qwest.net) |
| 18:19.16 | [TK]D-Fender | momelod: Because CM was doing the work <- |
| 18:19.26 | momelod | so its not a limitation of the phone.. |
| 18:20.10 | Kobaz | <PROTECTED> |
| 18:20.18 | Kobaz | someone forgot to change their callerid |
| 18:20.20 | *** join/#asterisk sahafeez (n=sahafeez@65-119-47-100.dia.static.qwest.net) |
| 18:23.34 | momelod | i suppose i could just transfer them into a meetme room. |
| 18:28.23 | *** join/#asterisk jtodd (i=pj0oqwiq@asterisk/community-director-and-tie-dye-shirt-lover/jtodd) |
| 18:28.24 | *** mode/#asterisk [+o jtodd] by ChanServ |
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| 18:39.02 | *** join/#asterisk ariel_ (n=chatzill@63.214.236.169) |
| 18:40.16 | ariel_ | Afternoon everyone |
| 18:42.02 | grharry | Hi, having some problems with echo ... I have an ISDN HFC-4S card using dahdi .. some help please ?? |
| 18:44.15 | gr0mit | echo fromwhere to where? |
| 18:45.03 | gr0mit | who is hearing the Echo? |
| 18:45.18 | grharry | when I make a call to a public num I am ( from asterisk ) having the echo !! |
| 18:46.09 | gr0mit | Okay, what kind of phone are you using on your *system? |
| 18:46.55 | grharry | A siemens connected to an linksys ata + cisco 7940 ( sip ) |
| 18:47.51 | *** join/#asterisk wierdo (n=chatzill@77.78.3.197) |
| 18:48.16 | gr0mit | now I'm confused |
| 18:48.39 | grharry | Allow me ?? |
| 18:48.48 | grharry | where are you confused ?? |
| 18:49.29 | gr0mit | so, your *boxes connected to the public network via an ISDN line, right? |
| 18:49.37 | grharry | yep |
| 18:49.43 | gr0mit | okay |
| 18:49.54 | gr0mit | so, you make a call to the public network |
| 18:49.58 | grharry | yep |
| 18:50.04 | gr0mit | from which handset are you making the call? |
| 18:50.08 | grharry | ok |
| 18:50.16 | *** part/#asterisk mbrevda (n=mbrevda@unaffiliated/mbrevda) |
| 18:50.55 | grharry | siemens plain phone ( no voip stuff ) -> linksys PAP2 -> Asterisk |
| 18:51.19 | grharry | sorry I typed ata ... I meant PAP2 |
| 18:51.32 | grharry | tired :-( |
| 18:51.52 | *** join/#asterisk momelod (n=smelo@CPE001f3a8fe859-CM0012c91df0bc.cpe.net.cable.rogers.com) |
| 18:52.03 | gr0mit | Okay so 2-wire analogue phone -> PAP2 --> asterisk -> HFC card -> public ISDN ? |
| 18:52.12 | grharry | perfect ! |
| 18:52.13 | gr0mit | brb |
| 18:52.18 | gr0mit | doorbell... |
| 18:52.41 | *** join/#asterisk Peaceful (n=Peaceful@70.102.57.178) |
| 18:52.46 | grharry | np |
| 18:53.07 | Peaceful | what the @#$%#$% would automatically run dahdi_genconf at reboot time and stomp on my /etc/dahdi/system.conf???? |
| 18:53.23 | gr0mit | ok back |
| 18:53.34 | grharry | tnx |
| 18:53.46 | gr0mit | so, you speak into your analogue phone, and u hear your own voice coming back at you? |
| 18:53.55 | grharry | correct ! |
| 18:53.58 | gr0mit | ok. |
| 18:54.19 | gr0mit | so, the echo is coming from the distant handet on your PSTN, soooo |
| 18:54.44 | gr0mit | are you using bristuff or chan_msisdn? |
| 18:54.57 | grharry | dahdi |
| 18:55.37 | gr0mit | ok, can u pls pastebin your dhadi.conf |
| 18:55.44 | gr0mit | or wotever its called |
| 18:55.57 | grharry | chan_dahdi ?? |
| 18:56.10 | grharry | ok |
| 18:56.24 | tzafrir | Peaceful, theoretically it hsould work. In practice I can think of a number of cases wher eit won't |
| 18:57.11 | *** join/#asterisk sranil (n=sranil@122.175.76.14) |
| 18:57.56 | grharry | chan_dahdi.conf http://pastebin.com/m6cca972a |
| 18:58.10 | grharry | brb |
| 19:01.03 | *** join/#asterisk theHub (n=theHub@69.177.93.21) |
| 19:01.53 | gr0mit | ok looks reasonable. Here is my zapata.conf (using bristuff) on a similar card |
| 19:01.57 | gr0mit | http://pastebin.com/m1ea7fa7a |
| 19:02.17 | *** join/#asterisk Whitor (n=mcneany@rrcs-24-97-4-146.nys.biz.rr.com) |
| 19:02.19 | gr0mit | I wonder if the echo training is confusing it? |
| 19:02.27 | gr0mit | try removing that line |
| 19:02.52 | *** join/#asterisk CGMChris (n=chris@74.143.228.142) |
| 19:03.08 | BillyCrook | Is there a wireshark-like tool for asterisk events? (so I can interactively filter them as I learn what they correlate to?) |
| 19:03.25 | BillyCrook | e.g. I don't care about peerstatus events for now |
| 19:03.29 | [TK]D-Fender | BillyCrook: What "Asterisk events"? |
| 19:03.46 | BillyCrook | the ones you see when connected to the asterisl manager port 5038 |
| 19:03.56 | [TK]D-Fender | BillyCrook: Nothing I've heard of... |
| 19:05.24 | *** join/#asterisk lozarythmic (n=lpraties@e1-1.ns500-1.ts.milt.as9105.net) |
| 19:07.40 | *** join/#asterisk Ad-Hoc (n=nimbus@ppp85-137.adsl.forthnet.gr) |
| 19:08.01 | *** join/#asterisk donnib (n=donnib@0x555281d0.adsl.cybercity.dk) |
| 19:08.38 | donnib | hi |
| 19:08.59 | Peaceful | tzafrir: I don't understand your comment. Something's running dahdi_genconf at reboot time without me asking for it. |
| 19:09.00 | donnib | i have a adaptor which looses registration then i get 401 |
| 19:09.00 | *** join/#asterisk TSM (n=the_soft@87-194-32-212.bethere.co.uk) |
| 19:09.12 | donnib | if i reboot it it works then after a wile i get 401 again |
| 19:09.13 | donnib | http://pastebin.com/d27df68 |
| 19:09.44 | donnib | the adaptor is 192.168.1.1 and extention is 2337, server is 129.168.1.10 |
| 19:09.53 | Peaceful | Is there some part of dahdi or asterisk that will call dahdi_genconf for you without you wanting it to? |
| 19:09.59 | donnib | can anybody help me figuring out why i get the 401 ? |
| 19:10.50 | *** join/#asterisk cosmicwombat (n=cosmicwo@69.7.44.68) |
| 19:11.39 | *** join/#asterisk breeves (n=chatzill@pool-173-74-20-27.dllstx.fios.verizon.net) |
| 19:12.06 | donnib | anyone ? |
| 19:12.50 | breeves | Hello everyone. I have a question about a setting in 1.6 for sip out-of-dialog messages, anybody around tried this? |
| 19:13.45 | russellb | which setting? |
| 19:15.00 | breeves | russellb: The ones introduced with this feature : https://issues.asterisk.org/view.php?id=13028&nbn=23 |
| 19:15.50 | russellb | ah, that's not in a release yet |
| 19:15.55 | russellb | won't be until 1.8 |
| 19:15.55 | *** join/#asterisk citywok (n=chatzill@c-71-231-179-213.hsd1.wa.comcast.net) |
| 19:16.10 | citywok | i just managed to crash asterisk transferring a call. 1.6.1.6 [6078636.835271] asterisk[7523]: segfault at a4 ip f7c49fd4 sp f5cf9a64 error 4 in libpthread-2.7.so[f7c41000+15000] |
| 19:16.16 | breeves | russellb: Is it in trunk? |
| 19:16.20 | russellb | trunk, yes |
| 19:16.27 | leifmadsen | brent reeves? |
| 19:16.57 | breeves | russellb: Ok, I'll try that, this is all theory anyway :) |
| 19:17.27 | tzafrir | Peaceful, not in the default install. Though I've seen some settings in which people do that |
| 19:17.36 | breeves | leifmadsen: Nope, Bruce. |
| 19:17.57 | breeves | leifmadsen: You and I discussed Dundi several years ago in KC |
| 19:18.18 | Peaceful | tzafrir: okay, well I never want it to run anyway, so I replaced /usr/sbin/dahdi_genconf with a bash script that spits out it's parent's pid. |
| 19:18.22 | Peaceful | weird |
| 19:19.37 | leifmadsen | breeves: wow, crazy! |
| 19:19.45 | leifmadsen | all the people I meet, and I suck with names, so I never remember, heh |
| 19:20.01 | leifmadsen | I just happen to know a Brent Reeves, so I was just checking ;) |
| 19:20.29 | breeves | leifmadsen: No problem. |
| 19:22.56 | *** join/#asterisk puzzled (n=foobar@puzzled.xs4all.nl) |
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| 19:32.10 | Katty | untangles christmas garland |
| 19:32.45 | *** join/#asterisk fofware (n=chatzill@190.7.25.160) |
| 19:33.44 | leifmadsen | nice |
| 19:33.48 | *** join/#asterisk tlarsen (n=tlarsen@c-71-207-223-160.hsd1.al.comcast.net) |
| 19:35.04 | Katty | they're little santa booties |
| 19:35.04 | Katty | on silver string |
| 19:35.04 | *** join/#asterisk trebaum (n=trebaum@207-67-92-30.static.twtelecom.net) |
| 19:35.04 | trebaum | hello folks |
| 19:35.04 | Katty | i'm gonna bring my ole tree to work too, and set it up after christmas. |
| 19:35.04 | trebaum | i'm having an issue with echo in a tdm400 card. |
| 19:35.05 | trebaum | I have echocancel=yes |
| 19:35.14 | Katty | one of the locate salvation armys was having christmas stuff out.. |
| 19:35.24 | Katty | so i went to support them (= |
| 19:35.40 | trebaum | though when I do dahdi show channel 1, it says 128 taps... turned off |
| 19:35.54 | trebaum | how do i get echo cancelling turned on? |
| 19:35.55 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 19:37.50 | *** join/#asterisk korcan (n=korcan@ip65-44-169-66.z169-44-65.customer.algx.net) |
| 19:37.53 | trebaum | anyone there? |
| 19:38.12 | Katty | a whole room full. |
| 19:38.21 | trebaum | no body is talking. |
| 19:38.34 | Katty | that's because they're working |
| 19:38.43 | Katty | or untangling christmas garland... |
| 19:38.49 | *** join/#asterisk uqlev (n=yuriy@91.184.221.31) |
| 19:39.04 | *** join/#asterisk ccesario (n=ccesario@189-19-6-236.dsl.telesp.net.br) |
| 19:39.50 | breeves | trebaum: What does dahdi show say when there is an active call on the analog lines? |
| 19:40.10 | breeves | Katty: Well . . .no garland here :) |
| 19:40.19 | trebaum | breeves: just a sec. |
| 19:40.23 | Katty | i have plenty of santa booties. |
| 19:40.29 | Katty | so far 6 8ft strands. |
| 19:41.09 | trebaum | breeves: another thing I should tell you is that I actually have 2 tdm400 cards. |
| 19:41.24 | trebaum | breeves: one with 4 fxs modules and the other with 4 fxo modules |
| 19:41.31 | *** join/#asterisk angryuser_laptop (n=angryuse@90-156-167-83.reverse.alphalink.fr) |
| 19:42.01 | trebaum | breeves: one is on the east coast, and the other on the west |
| 19:42.10 | trebaum | with an iax2 trunk 'tween the two |
| 19:42.36 | trebaum | when making a call, the server that the phone (plain analog phone) is connected to gives me no errors. |
| 19:42.44 | trebaum | the other... gives me this. |
| 19:43.42 | trebaum | [Nov 3 10:20:25] WARNING[18845] chan_dahdi.c: Unable to enable echo cancellation on channel 2 (No such device) |
| 19:43.49 | trebaum | breeves: any idea? |
| 19:44.49 | Katty | yay, untangled. |
| 19:44.50 | Kobaz | trebaum: make sure dahdi is set up correctly, make sure the span is up, make sure asterisk chan_dahdi is configured correctly |
| 19:47.24 | trebaum | I can make phone calls no problem. |
| 19:47.41 | trebaum | and the echo is only on my side. |
| 19:48.04 | trebaum | i'm actually making phone lines out of pots lines at the far end. |
| 19:48.12 | trebaum | err phone calls out of the other end |
| 19:48.42 | *** join/#asterisk thegoat (n=jircii@c-71-224-180-83.hsd1.pa.comcast.net) |
| 19:48.57 | Kobaz | does your card support echo cancelation? |
| 19:49.09 | thegoat | hey all, i am trying to get the postgres cdr integration working, but when i go to load the module i get oad_dynamic_module: Error loading module 'cdr_pgsql': libpq.so.5: cannot open shared object file: No such file or directory |
| 19:49.43 | Pan3D | lol, thegoat |
| 19:49.52 | thegoat | the cdr_pgsql.so exists in the right place, and linpq.so.5 exists in /opt/postgres/lib |
| 19:50.05 | thegoat | i am not sure where it is looking for libpq.so.5 |
| 19:50.14 | thegoat | is there a way to tell where asterisk is looking for it? |
| 19:50.15 | Pan3D | did you check perms? |
| 19:50.47 | trebaum | kobaz: i'm trying to use the software echo cancelling with dahdi. |
| 19:51.01 | trebaum | kobaz: i have tried all of the ones that comes with the source. |
| 19:51.28 | thegoat | asterisk is running as root, so it should have access to it |
| 19:51.48 | Kobaz | it's not an access issue if it says no such device |
| 19:51.56 | trebaum | what worries me is this line from dahdi show channel 1 |
| 19:51.57 | trebaum | Echo Cancellation: 128 taps, currently OFF |
| 19:52.01 | Pan3D | k |
| 19:52.09 | trebaum | how to I get currently ON |
| 19:52.51 | Kobaz | it would be safe to say that echo cancelation isn't enabled |
| 19:53.07 | thegoat | when i specify the --with-postgres during the ./configure do i provide the base dir for postgres or /opt/postgres/lib? |
| 19:53.08 | Kobaz | you're missing something from the configs |
| 19:53.24 | Kobaz | trebaum: paste your /etc/dahdi/system.conf and /etc/asterisk/chan_dahdi.conf |
| 19:53.46 | Pan3D | postgress itself IIRC |
| 19:53.54 | trebaum | kobaz: pastebin? |
| 19:55.20 | thegoat | tha's what i thought. |
| 19:55.35 | trebaum | kobaz: http://pastebin.com/d445a1ed4 |
| 19:56.03 | Pan3D | (you of course have to select it in the menuconfig as well) |
| 19:56.12 | thegoat | yep did that |
| 19:56.35 | Pan3D | hmmm, oddddd |
| 19:57.20 | thegoat | i wonder if it could be lookign in /var/lib or some place like that |
| 19:57.41 | trebaum | kobaz: any idea? |
| 19:59.12 | *** join/#asterisk bn-7bc (n=bjarne-i@mac.lan.noare-1.holmedal.net) |
| 20:01.30 | Kobaz | looking |
| 20:02.08 | thegoat | it was looking for it in /usr/lib64 |
| 20:02.22 | Kobaz | trebaum: the mg2 echo canceler is the hardware echo canceler, i think |
| 20:02.27 | *** join/#asterisk Ad-Hoc (n=nimbus@ppp85-137.adsl.forthnet.gr) |
| 20:07.06 | *** join/#asterisk puzzled (n=foobar@puzzled.xs4all.nl) |
| 20:08.42 | *** join/#asterisk bbt (n=Sam@180.189.138.103) |
| 20:18.51 | *** join/#asterisk mbrevda (n=mbrevda@unaffiliated/mbrevda) |
| 20:19.16 | mbrevda | how can you set a variable as a variable? (set by reference) |
| 20:21.14 | Pan3D | thegoat: where is that path set? |
| 20:22.29 | Kobaz | mbrevda: yeah |
| 20:22.34 | mbrevda | hu? |
| 20:22.57 | Kobaz | Set(var=foo) Set(${var}=bar) |
| 20:23.15 | Kobaz | NoOp(${var}) |
| 20:23.34 | mbrevda | Kobaz: ah. now, can you do that as a global? |
| 20:23.38 | leifmadsen | Kobaz: I think you mean NoOp(${foo}) |
| 20:23.48 | leifmadsen | which would return 'bar' |
| 20:23.50 | Kobaz | er... yes |
| 20:23.51 | Kobaz | ${foo} |
| 20:23.54 | leifmadsen | ;) |
| 20:24.17 | Kobaz | mbrevda: you can do whatever.. just put the var where you want it to expand |
| 20:24.37 | leifmadsen | mbrevda: yes, same idea: Set(GLOBAL(${var})=bar) |
| 20:25.05 | Kobaz | and that would set global(foo) |
| 20:25.21 | mbrevda | hmm, I shoudl explain. Im trying to a gloabl to, say, epoch (or uniquid, or something dynamoc that is acceseable only as a varibel) |
| 20:25.37 | Kobaz | you don't need anything fancy for that then |
| 20:25.51 | Kobaz | Set(GLOBAL(time)=STRFTIME(...)); |
| 20:25.57 | mbrevda | The easiest way is to just put it in the globals section. so my quesiton is can I set a global under gloabl to a varibles? |
| 20:26.15 | mbrevda | Kobaz: under [globals] you can just do key=val without set |
| 20:26.21 | Kobaz | mbrevda: yeah |
| 20:26.26 | Kobaz | if you're using extensions.conf |
| 20:26.30 | Kobaz | i've been using ael these days |
| 20:26.48 | trebaum | kobaz: if that is the hardware echo canceler, what software one should I use? |
| 20:26.57 | Kobaz | trebaum: i've never used the software one... i dunno |
| 20:27.05 | trebaum | kobaz: good answer |
| 20:27.11 | Kobaz | hah |
| 20:27.18 | mbrevda | yeah=yes you can? how? I tried as followes: TOUCH_MONITOR=${UNIQUEID}, but that would just give a blan var |
| 20:27.23 | trebaum | Does anyone know what software echo canceller I should use with the TDM400? |
| 20:27.32 | *** join/#asterisk andres833 (n=andres83@190.144.75.22) |
| 20:27.38 | mbrevda | trebaum:any. its unrelated to the card |
| 20:27.53 | Kobaz | he's looking for the line to use in dahdi.conf |
| 20:27.58 | mbrevda | oh |
| 20:29.18 | trebaum | ok, so do all the echo cancelling modules get built by default? |
| 20:29.27 | trebaum | software ones, that is. |
| 20:29.53 | trebaum | I have the newest source for dahdi linux/tools |
| 20:30.11 | trebaum | but I installed the default build. |
| 20:30.56 | mbrevda | Qwell: ping |
| 20:30.59 | Qwell | ? |
| 20:31.18 | mbrevda | any plans for 1.6.2 rpms? |
| 20:31.44 | Qwell | It isn't released yet...so no. |
| 20:32.23 | mbrevda | oh, sorry. though it was |
| 20:32.29 | SuPrSLuG | trebaum: you can alter which on is built in the dahdi.c source file. Forgot the line, but you can do it |
| 20:32.51 | mbrevda | Qwell: how about 1.6.1? (not that I need it or anything) |
| 20:35.11 | tzafrir | There's no dahdi.c |
| 20:35.59 | tzafrir | And you set which ones are built on drivers/dahdi/Kbuild |
| 20:41.42 | *** join/#asterisk CcRnp (n=shishir@208.179.165.18) |
| 20:42.47 | CcRnp | Can anyone suggest me which is the best speech recognition engine for asterisk ? |
| 20:42.49 | SuPrSLuG | yeah, apparently that was in the way back machine of zaptel. lol. |
| 20:44.28 | SuPrSLuG | been that long since I played with software echo cancel |
| 20:46.02 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 20:46.34 | *** join/#asterisk jtodd (i=k5smdd8g@asterisk/community-director-and-tie-dye-shirt-lover/jtodd) |
| 20:46.34 | *** mode/#asterisk [+o jtodd] by ChanServ |
| 20:47.45 | TheDavidFactor | CcRnp, I can't personally recommend this, but is was mentioned in the voip users conference two weeks ago http://www.digium.com/en/products/software/vestec.php |
| 20:48.05 | CcRnp | tahnks |
| 20:48.30 | CcRnp | TheDavidFactor thanks |
| 20:49.02 | *** join/#asterisk Skeeter- (i=Skeeter-@190-141.cgocable.ca) |
| 20:49.26 | Skeeter- | is there anyway to put my asterisk(voicemail voice) in french |
| 20:50.13 | *** join/#asterisk CunningPike (n=CunningP@204.239.8.157) |
| 20:51.02 | mbrevda | yes |
| 20:51.14 | TheDavidFactor | yw |
| 20:51.25 | CunningPike | When a call is placed via a SIP trunk to SER to multiple end points, is there a way to update the asterisk CDR with which endpoint answered the call? |
| 20:52.18 | CcRnp | isnt there any open source project that is best for speech recognition ? |
| 20:52.26 | CunningPike | The dstchannel in the CDR shows the same value (SIP/[nameoftrunk]) regardless of which endpoint picked up |
| 20:55.50 | Skeeter- | could anyone point me how to switch the language of asterisk?? |
| 20:56.24 | CunningPike | Skeeter-: Of the voice prompts? Or what? |
| 20:56.38 | mbrevda | that what he wants |
| 20:56.49 | angryuser_laptop | Edit > Options > switch language |
| 20:57.28 | angryuser_laptop | could not hold myself |
| 20:57.39 | CunningPike | hands angryuser_laptop a tissue |
| 20:58.27 | *** join/#asterisk angryuser_laptop (n=angryuse@90-156-167-83.reverse.alphalink.fr) |
| 20:58.47 | *** join/#asterisk johnakabean (n=john@static-173-50-101-13.nrflva.east.verizon.net) |
| 20:59.33 | johnakabean | hey everyone......when you set a global variable, does it stick for that channel or are all future channels forced to use it?\ |
| 20:59.50 | Skeeter- | cant find it, i must be blind |
| 21:00.27 | [TK]D-Fender | johnakabean: Wouldn't be "global" if ti only applied to that call, now would it? |
| 21:01.19 | johnakabean | fender, I thought it meant global for that call throughout all contexts |
| 21:01.46 | [TK]D-Fender | johnakabean: There is no variable scope in the dialplan. |
| 21:01.57 | johnakabean | thanks for clarification |
| 21:02.01 | [TK]D-Fender | johnakabean: This is not a structured programming language |
| 21:02.34 | johnakabean | yeah fender, you know I come from programming C and other binary languages |
| 21:02.47 | johnakabean | so asterisk is ass backwards in some things and that's why i get confused |
| 21:04.00 | johnakabean | asterisk is more simple but has too much overhead |
| 21:04.10 | TSM | has anyone used xlite on a mac? |
| 21:06.28 | [TK]D-Fender | johnakabean: And how long have you been using * now? |
| 21:07.03 | johnakabean | so, [TK]D-Fender, there is no point in doing setmusiconhold more than once in the entire dialplan, if it is already set when the call comes in. I just don't understand why, when waiting for an admin is set and in a meetme, the musiconhold is always default and not the musiconhold set. Is there an option for meetme to specify musiconhold? |
| 21:07.19 | johnakabean | programming it by hand, about 3 months |
| 21:07.47 | johnakabean | I learned it not from the web but by analyzing freepbx's syntax |
| 21:07.53 | [TK]D-Fender | johnakabean: So the far greater number are just running GUI installs? |
| 21:08.04 | [TK]D-Fender | Learning via freePBX = BAD |
| 21:08.14 | johnakabean | no, I always compiled and installed by hand |
| 21:08.31 | citywok | i'm using chanspy(,w) to try and whisper to the spied on channel, i can hear it, but i get no whisper (on either end of the call actually) |
| 21:08.32 | [TK]D-Fender | johnakabean: You'll learn a lot of ways you should NOT do things and it won't explain the "whys" |
| 21:08.40 | johnakabean | I have heard this lol |
| 21:09.03 | [TK]D-Fender | johnakabean: For your MoH question show me something meaningful... |
| 21:09.05 | CunningPike | [TK]D-Fender: Any insight into my trunk CDR question above? Am I stuck with SER accounting? |
| 21:09.20 | CunningPike | And,hi, by the way - been a long time :) |
| 21:09.39 | [TK]D-Fender | CunningPike: And Ditto.. on both fronts |
| 21:09.40 | CunningPike | Still dispensing wisdom to the unwashed, I see |
| 21:10.06 | johnakabean | exten => 901,n(USER),Set(MEETME_OPTS=oTcIMsr) |
| 21:10.08 | Skeeter- | asterisk wont switch to french language, i give up |
| 21:10.25 | johnakabean | the M option is music waiting on the leader and it always plays default class |
| 21:10.29 | [TK]D-Fender | Skeeter-: Where did you tell it to? |
| 21:10.41 | [TK]D-Fender | johnakabean: that 1 liner tells me nothing of value |
| 21:10.41 | johnakabean | even though the class is set per inbound |
| 21:10.53 | johnakabean | well the others are just standard |
| 21:10.54 | [TK]D-Fender | johnakabean: Prove what is set and show configs and the failed call |
| 21:11.05 | johnakabean | one is answer and the other is startmeetme |
| 21:11.14 | p3nguin | citywok: ChanSpy(SIP/${SPYEXTEN},qw) ; where SPYEXTEN is the number you want to spy on. |
| 21:11.51 | citywok | i left it filtered out so i dont have to worry about it for testing, and i havent used Q to silence the beeps, but yea i hear the channel audio, but cant whisper back in to it |
| 21:12.05 | johnakabean | the music on hold is set on all incoming did's |
| 21:12.13 | johnakabean | to different classes |
| 21:12.24 | Katty | looks in |
| 21:12.33 | citywok | p3nguin: i'm just using this for testing right now: ChanSpy(,w) |
| 21:12.49 | Skeeter- | [TK]D-Fender: i used the SetLanguage cmd, but i will just overwrite with the french voices rgiht now |
| 21:12.50 | citywok | normally it will be ,qwg(${REQUESTED_GROUP_NUMBER}) |
| 21:13.04 | [TK]D-Fender | johnakabean: You are giving me a story, not CLI output and configs... |
| 21:13.05 | CunningPike | Katty!!!! |
| 21:13.09 | johnakabean | exten => 9155551234/9155524584,1,Set(__FROM_DID=${EXTEN}) |
| 21:13.09 | johnakabean | exten => 9155551234/9155524584,n,Gosub(app-blacklist-check,s,1) |
| 21:13.09 | johnakabean | exten => 9155551234/9155524584,n,ExecIf($[ "${CALLERID(name)}" = "" ] ,Set,CALLERID(name)=${CALLERID(num)}) |
| 21:13.09 | johnakabean | exten => 9155551234/9155524584,n,SetMusicOnHold(Hip-Hop) |
| 21:13.09 | johnakabean | exten => 9155551234/9155524584,n,Set(__MOHCLASS=Hip-Hop) |
| 21:13.10 | johnakabean | exten => 9155551234/9155524584,n,Set(__CALLINGPRES_SV=${CALLINGPRES_${CALLINGPRES}}) |
| 21:13.12 | johnakabean | exten => 9155551234/9155524584,n,SetCallerPres(allowed_not_screened) |
| 21:13.14 | johnakabean | exten => 9155551234/9155524584,n,Goto(ext-queues,301,1) |
| 21:13.15 | Katty | meep |
| 21:13.16 | johnakabean | sorry for flood |
| 21:13.17 | [TK]D-Fender | johnakabean: PASteBIN |
| 21:13.33 | [TK]D-Fender | johnakabean: And provide what I asked for |
| 21:14.21 | johnakabean | if that number, 915524584, calls the music on hold is set to hip-hop; but transferring them to a conference renders default classical music |
| 21:15.21 | TSM | how bad does echo get if you dont have a analog card with EC? |
| 21:15.27 | [TK]D-Fender | johnakabean: ..... pastebin actual configs, actual dialplan, and an actual call |
| 21:15.30 | johnakabean | my instinct was to add |
| 21:15.31 | johnakabean | exten => 901,n(USER),Set(MEETME_OPTS=oTcIMsr) |
| 21:15.31 | johnakabean | exten => 901,n(USER),SetMusicOnHold(${__MOHCLASS}) |
| 21:16.14 | Katty | johnakabean: i'm going to go out on a limb and say he doesn't really care. he wants to see the configs. |
| 21:16.17 | johnakabean | without the user, though |
| 21:16.22 | [TK]D-Fender | johnakabean: LAST TIME |
| 21:16.26 | Katty | johnakabean: and no matter what you say isn't really going to help until he sees it |
| 21:16.27 | [TK]D-Fender | [16:15]<[TK]D-Fender>johnakabean: ..... pastebin actual configs, actual dialplan, and an actual call |
| 21:16.43 | Katty | also, i think i will have an orange. |
| 21:16.49 | Katty | leafs |
| 21:17.02 | johnakabean | i just have one question.......if the musiconhold is specified, and not changed again, does it need to be specified again to be set correcty in another context? |
| 21:17.16 | johnakabean | straight forward |
| 21:17.44 | [TK]D-Fender | johnakabean: No scope <- I was very clear on this |
| 21:17.53 | johnakabean | ok you answered my question |
| 21:17.54 | johnakabean | thank you |
| 21:18.26 | johnakabean | i was just wondering why that didn't apply to a conference when another setmusiconhold was never executed down the line |
| 21:18.37 | [TK]D-Fender | johnakabean: ... |
| 21:19.03 | [TK]D-Fender | ... |
| 21:20.28 | [TK]D-Fender | nvm, not worth it.... |
| 21:22.42 | eppigy | HELLO |
| 21:23.06 | Katty | HELLO THAR |
| 21:23.47 | Katty | i has an orange. |
| 21:23.54 | TSM | does anyone know the average cost of a BRI line in the US (NY pref) |
| 21:24.20 | Katty | your isp would be happy to give you a quote. |
| 21:24.29 | *** join/#asterisk wam (i=wam@unaffiliated/wam) |
| 21:24.29 | TSM | im UK based |
| 21:24.41 | TSM | i need to get it for our NY office |
| 21:24.48 | TSM | just average cost |
| 21:25.03 | Katty | you don't have a providor for your NY Office? (= |
| 21:25.05 | Katty | call them. |
| 21:25.14 | TSM | yup, timewarner |
| 21:25.19 | TSM | naa |
| 21:25.21 | TSM | sorry |
| 21:25.23 | Katty | ;) |
| 21:25.24 | TSM | broadcom |
| 21:25.27 | TSM | naaa |
| 21:25.29 | TSM | sorry again |
| 21:25.34 | TSM | broadview :) |
| 21:25.41 | TSM | my mum knows more than them |
| 21:25.46 | Katty | this sounds like a game of monopoly |
| 21:26.01 | Katty | next up is boardwalk! |
| 21:26.05 | Katty | for $450 |
| 21:26.35 | *** join/#asterisk jtodd (i=hd5vtdf6@asterisk/community-director-and-tie-dye-shirt-lover/jtodd) |
| 21:26.35 | *** mode/#asterisk [+o jtodd] by ChanServ |
| 21:26.42 | beek | Katty: I own boardwalk and I have a hotel on it. |
| 21:26.48 | Katty | :< |
| 21:26.50 | Katty | i don't wanna visit. |
| 21:27.06 | beek | It's dog and ferret friendly. |
| 21:27.11 | Katty | :> |
| 21:27.23 | Katty | it's hard to find a hotel that will let you bring a 90lb dog and 5 ferrets. |
| 21:27.48 | angryuser_laptop | not in korea |
| 21:27.48 | beek | You have five ferrets? Wow! |
| 21:27.53 | *** join/#asterisk KavanS (n=KavanS@static-173-50-141-22.ptldor.fios.verizon.net) |
| 21:28.00 | Katty | Merry Pippin Sammy Shire and BB, short for Bilbo Baggins |
| 21:28.22 | beek | angryuser_laptop: not including the kitchen or dining room. |
| 21:28.37 | Katty | well i don't want to move to korea. |
| 21:28.59 | Katty | it smells funny |
| 21:29.50 | angryuser_laptop | beek: too late to give the rules, you can even get payd if you bring some dogs xD |
| 21:29.59 | Katty | what, WHAT |
| 21:30.02 | Katty | WHAT?!?! |
| 21:30.17 | angryuser_laptop | Katty: finally you understood ? |
| 21:30.18 | Katty | nevermind, i don't want to know. |
| 21:30.42 | eppigy | scoots closer to Katty |
| 21:31.30 | Katty | angryuser_laptop: not listening, la la la! |
| 21:31.52 | Katty | eppigy: people are scary. |
| 21:31.58 | Katty | eppigy: can i go hide in your closet? |
| 21:32.23 | *** join/#asterisk angryuser_laptop (n=angryuse@90-156-167-83.reverse.alphalink.fr) |
| 21:33.40 | *** join/#asterisk darkdrgn2k3 (n=darkdrgn@bas2-toronto44-1176438379.dsl.bell.ca) |
| 21:33.57 | darkdrgn2k3 | Hye guys, do VOIP lines have a CID NAME attached to them when they ring on PSTNs? |
| 21:34.13 | Katty | i don't understand the question. |
| 21:34.19 | Katty | could you be more specific. |
| 21:34.39 | beek | darkdrgn2k3: sometimes yes, sometimes no |
| 21:34.40 | darkdrgn2k3 | Katty: when i call out on a VOIP line terminating on a PSTN, will the PSTN CID show the name |
| 21:34.54 | darkdrgn2k3 | or just the # |
| 21:34.55 | Katty | darkdrgn2k3: directly to your provider, yes. |
| 21:35.00 | Qwell | darkdrgn2k3: If your name is in the terminating telcos database |
| 21:35.04 | Katty | darkdrgn2k3: once it leaves your provider, the name is stripped. |
| 21:35.05 | beek | darkdrgn2k3: I have had some calls via VoIP not show up. |
| 21:35.14 | Katty | darkdrgn2k3: the number is compared against whatever database the next carrier uses. |
| 21:35.37 | darkdrgn2k3 | katty: So the question is does teh carrier register with any databases? |
| 21:36.15 | Katty | darkdrgn2k3: i can send out my name, and our companies callerid all i want. when it hits my cellphone, it's going to show my company name not my personal name. |
| 21:36.36 | Katty | darkdrgn2k3: sprint doesn't care about the name that's in there, only the number. i hope this helps. |
| 21:36.48 | darkdrgn2k3 | Katty: do you know how VOIP.MS works? |
| 21:36.55 | Katty | no, why don't you call them and ask. |
| 21:37.20 | darkdrgn2k3 | i knwo that incomming names are striped and you need to refer against a database |
| 21:38.19 | *** part/#asterisk sfire (n=sfire@businessservers.info) |
| 21:39.14 | eppigy | Katty: yes you are welcome in my closet any time |
| 21:39.18 | angryuser_laptop | darkdrgn2k3: there is no world wide database of names for telcos if that what you want to know |
| 21:40.07 | darkdrgn2k3 | angryuser: no i know that, im curious how cids are dealth with on voip provider. i assume pstn providers use their own databases |
| 21:40.26 | angryuser_laptop | darkdrgn2k3: only numbers |
| 21:40.43 | *** join/#asterisk ricko73 (n=dhartman@wilug/newlug/ricko73) |
| 21:41.00 | *** join/#asterisk Witch_Doc (n=me@69.196.64.134) |
| 21:41.07 | Witch_Doc | can asterisk emulate a pri? |
| 21:41.33 | *** join/#asterisk chasing`Sol (n=rc4@It.Only.Hurts.When.I.am.Breathing.mm.am) |
| 21:41.49 | angryuser_laptop | darkdrgn2k3: the bigges i saw is trixnet, all users of trixbox pro has their names shown when they cann any other user of trixbox pro server |
| 21:41.50 | *** join/#asterisk QaDeS (n=mklaus@p4FC72A5C.dip0.t-ipconnect.de) |
| 21:42.22 | *** join/#asterisk Luch (n=Dwayne@64.42.227.97) |
| 21:42.44 | Luch | hi guys, is it necessary for iaxmodems to auth every 50 sec? |
| 21:42.46 | angryuser_laptop | Witch_Doc: why emulate when you can have real stuff ? |
| 21:43.14 | angryuser_laptop | Luch you got the param in iaxmodem config file |
| 21:43.52 | Luch | im just wondering, i have 6 setup and my log is diseased with auth logs, and each time it shows 12 im just wondering whats up with it |
| 21:43.55 | Witch_Doc | angryuser_laptop i have a panasonic pbx that i need to extend to another pbx in another building |
| 21:44.00 | Luch | i have 6 iaxmodems |
| 21:44.01 | diatonic | Witch_Doc:You should be able to have a T1 on an Asterisk server configured with signalling pri_net to have it act as a PRI from the telco |
| 21:44.26 | angryuser_laptop | Witch_Doc: so why do you need emulator ? |
| 21:44.39 | Witch_Doc | to link the two pbx's together |
| 21:44.41 | chasing`Sol | which is better asterisk-gui or freepbx? |
| 21:44.43 | angryuser_laptop | Witch_Doc: anyway asterisk can do it |
| 21:45.56 | Luch | like would i run into issues if i only configured it to auth once every 6 hrs or something? |
| 21:46.55 | angryuser_laptop | Witch_Doc: panasonic <> pri <> asterisk <> network <> asterisk <> pri if that what you need, but why dont you use a cable to extend pri from panasonic ? |
| 21:47.09 | Witch_Doc | thats what i was thinking to do as wel |
| 21:47.12 | Witch_Doc | well* |
| 21:47.14 | angryuser_laptop | or you can replace second asterisk by a geteway |
| 21:47.53 | Witch_Doc | angryuser_laptop the distance is too far to use a cable from panasnoic |
| 21:51.31 | angryuser_laptop | Witch_Doc: if you replace second * by a gateway it will cost you more $$ but in both ways it is a simple setup |
| 21:52.04 | Katty | eppigy: :> |
| 21:54.54 | *** join/#asterisk Godfather_ (n=godfathe@187.Red-83-38-229.dynamicIP.rima-tde.net) |
| 21:54.57 | Godfather_ | hi |
| 21:55.14 | Katty | herroes. |
| 21:55.24 | *** join/#asterisk lesouvage (n=lesouvag@82.73.69.76) |
| 21:56.35 | Godfather_ | its possible to use with asterisk the pstn ? i mean, a PC connected wired to the RJ11 of a house, and then some IP thelephones configured with the asterisk server? |
| 21:57.18 | *** join/#asterisk [TK]D-Fender (n=joeblow@161.216.150.25) |
| 21:57.26 | Godfather_ | its possible to call to this house and asterisk getting the line to pass the comunication to the softphones? |
| 21:57.52 | [TK]D-Fender | Godfather_:yes |
| 21:58.03 | *** join/#asterisk Peaceful (n=Peaceful@70.102.57.178) |
| 21:58.28 | Katty | has one at her home. |
| 21:58.52 | Godfather_ | [TK]D-Fender, its needed special hardware ? (PC, modem, and IP-phones) |
| 21:59.23 | Katty | you will need a card to handle phone lines, yes. but not a modem. a modem is a very different piece of hardware. |
| 21:59.26 | p3nguin | You'll need an FXO port for your computer. |
| 21:59.43 | [TK]D-Fender | Godfather_:you need a haware interface if you want to use physical lines |
| 22:00.10 | Godfather_ | It cant be possible with a tipical PCI-modem? |
| 22:00.13 | [TK]D-Fender | Godfather_:and you said you wanted to use softphones.n |
| 22:00.14 | *** join/#asterisk tzafrir (n=tzafrir@bzq-218-155-147.cablep.bezeqint.net) |
| 22:00.34 | [TK]D-Fender | Godfather_: no. modems aren't usable |
| 22:01.18 | Godfather_ | [TK]D-Fender, p3nguin , can you give me a search to google? i dont know what type of device i need |
| 22:01.48 | [TK]D-Fender | Godfather_:regular analog POTS line? |
| 22:02.14 | angryuser_laptop | Godfather_: google about "fxs fxo" |
| 22:03.00 | Godfather_ | ok, i google it, thx to all |
| 22:03.23 | Katty | Godfather_: http://www.telephonydepot.com/Contact-Us |
| 22:03.35 | Katty | Godfather_: one of the most reasonably priced places around. give them a call. |
| 22:03.41 | darkdrgn2k3 | can pstn lines accept longer CID numbers then 10 |
| 22:04.04 | *** join/#asterisk jeffgus (n=jeffgus@2002:ad33:b504:0:0:0:0:1) |
| 22:04.21 | Godfather_ | Katty, better to have a link to the specific hard |
| 22:04.24 | *** join/#asterisk breeves (n=chatzill@pool-173-74-20-27.dllstx.fios.verizon.net) |
| 22:04.27 | Godfather_ | but thx too |
| 22:04.36 | dlynes | Does anyone know if there's a 64-bit version of fax for asterisk forthcoming? |
| 22:04.43 | Katty | Godfather_: I like talking a real person about what i'm wanting to do. |
| 22:05.02 | *** join/#asterisk evil_gordita (n=evilgord@ip70-188-50-186.rn.hr.cox.net) |
| 22:05.24 | angryuser_laptop | dlynes: fax for asterisk ? which one ? |
| 22:05.52 | *** join/#asterisk pzn (n=pzn@189.79.203.227) |
| 22:05.58 | *** part/#asterisk Peaceful (n=Peaceful@70.102.57.178) |
| 22:06.29 | pzn | can anyone help me to do in extensions.conf a rule that will go to step 10 if today is november 4th? |
| 22:06.49 | pzn | I intend to put a playback() to play "today is holiday" message... |
| 22:07.33 | _ShrikE | pzn: core show application gotoiftime |
| 22:08.27 | *** join/#asterisk Caplain (i=shayne@2001:470:5:fb:20ee:7162:c8e4:6f63) |
| 22:09.31 | Godfather_ | [TK]D-Fender, what i need is a "FXO gateway"? |
| 22:09.48 | dlynes | angryuser_laptop: what do you mean which one? |
| 22:09.58 | dlynes | angryuser_laptop: the one sold by digium, of course |
| 22:10.26 | angryuser_laptop | dlynes: why dont you do a tour on their site then ? |
| 22:10.48 | dlynes | angryuser_laptop: perhaps you didn't understand the question....I asked if there was one ___forth___coming |
| 22:10.49 | *** join/#asterisk angryuser_laptop (n=angryuse@90-156-167-83.reverse.alphalink.fr) |
| 22:10.58 | [TK]D-Fender | Godfather_:Liinksys SPA-3102 |
| 22:11.00 | dlynes | angryuser_laptop: not one already in existence |
| 22:11.27 | TSM | whats the worth in the ec available on analog cards? i thought echo comes mostly from missatched line impedance |
| 22:11.45 | p3nguin | I don't really like my 3102. It seems kinda crappy. |
| 22:11.46 | *** join/#asterisk sier (n=sier@unaffiliated/sier) |
| 22:11.48 | sier | Hello all |
| 22:11.48 | *** join/#asterisk lost_soul (i=shawn@cpe-74-71-234-70.twcny.res.rr.com) |
| 22:12.08 | dlynes | TSM: it's also caused by long local loops |
| 22:12.19 | dlynes | TSM: such as is common in rural areas |
| 22:13.12 | p3nguin | godfather_: The gateway is a box that would connect between the wall jack and an ethernet port. You could use that, but I think an actual FXO/FXS card would be nicer. |
| 22:14.01 | p3nguin | godfather_: Let Asterisk do all the work instead of putting another point of configuration into the mix. |
| 22:14.33 | mchou | ~sipnat |
| 22:14.34 | infobot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 22:15.14 | Godfather_ | p3nguin, can you give me a specific example to a fxo fxs card like the linksys spa-3102 ? |
| 22:15.42 | dlynes | russellb: any idea when fax for asterisk will have a 64-bit edition? |
| 22:15.50 | p3nguin | The SPA-3102 is a VoIP gateway, not a card. http://www.digiumcards.com/digium_cards_combos.html |
| 22:15.51 | *** join/#asterisk denon (i=denon@sassinak.net) |
| 22:15.51 | *** mode/#asterisk [+o denon] by ChanServ |
| 22:16.35 | Godfather_ | p3nguin, yep, sorry, thx |
| 22:16.58 | p3nguin | godfather_: Will you be using only IP phones once you put Asterisk inline? |
| 22:17.24 | TSM | dlynes: well our office is somewhere in manhattan, so cant guess the loop is that long |
| 22:18.08 | Godfather_ | p3nguin, well, what i believe (not sure), if i use this system, it cant be possible to use Analog phones, no? |
| 22:18.40 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 22:18.42 | p3nguin | godfather_: You can use analog phones, but IP phones are nice. |
| 22:18.50 | [TK]D-Fender | Godfather_:yes, with adapters |
| 22:19.32 | p3nguin | godfather_: http://www.digiumcards.com/tdm31b.html |
| 22:19.36 | *** join/#asterisk brut- (n=brut-@h66-173-4-254.mntimn.dedicated.static.tds.net) |
| 22:19.37 | Godfather_ | aps, well, i explain my problem, maybe you can give me a better solution (this is just for testing and for learning, no profesional) |
| 22:19.51 | p3nguin | godfather_: One FXO port to connect to the wall jack, three FXS ports to connect three regular phones. |
| 22:20.58 | Godfather_ | huh? i dont understand that last |
| 22:21.05 | Godfather_ | mmm |
| 22:21.18 | Godfather_ | an FXO wall port, connected to.. ? |
| 22:21.25 | p3nguin | That's not what I said. |
| 22:21.49 | Godfather_ | Well, you tried to explain me how to use analog phones no? |
| 22:22.05 | p3nguin | Phones have FXO ports. The wall jack is considered FXS. |
| 22:22.10 | Godfather_ | [TK]D-Fender told thats possible to have analog phones with an adapters |
| 22:22.17 | p3nguin | You mate the two together to create a circuit. |
| 22:23.19 | p3nguin | YOu could use a single FXO port on your computer, connect it to the wall jack, then use ethernet for every phone. |
| 22:25.11 | Godfather_ | p3nguin, yes, this is what i commented, all the phones on ethernet (ip-phones). but [TK]D-Fender told something about adapters to use analog phones, but i cant understand how an analog phone can have an ip ? |
| 22:25.25 | Godfather_ | how the adapter give him an ip and connect to the network? |
| 22:25.45 | p3nguin | If you don't have FXS ports on your computer to attach analog phones, then you will run IP/Ethernet. |
| 22:26.21 | p3nguin | If you run only IP/Ethernet for phones, you have two choices: Use IP phones or softphones; or use ATAs and analog phones. |
| 22:26.36 | p3nguin | The ATA is the adapater to use an analog phone on Ethernet. |
| 22:26.48 | Godfather_ | p3nguin, yep, ok, connected to the FXS on the computer, thats the question, ok |
| 22:28.01 | p3nguin | I think I would rather just have the FXO port on the computer to connect to the wall jack, then use IP phones everywhere in the house. |
| 22:28.17 | p3nguin | That requires ethernet cables to be ran, though. |
| 22:28.42 | angryuser_laptop | dect pap2t |
| 22:29.08 | mchou | yeah, word |
| 22:29.14 | mchou | dect pap2t! |
| 22:29.58 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 22:30.03 | sier | Guys.. I'm a total newbie w/ asterisk. This is what I'm trying to achieve: Someone calls me, it redirects to an auto-attendant, person chooses from the menu, message is displayed (depending on persons choice).. What do I need to acomplish this? I have a computer, and asterisk is installed.. I'm really lost.. I downloaded a book, but it has 600 pages, and I was wondering if someone could give me a quick overall.. |
| 22:30.06 | mchou | actually spa-2102 is probably better |
| 22:30.09 | Godfather_ | p3nguin, well, another solution is tou use de ATA adapter wireless, if it exists. I mean, connect the ATA adapter wireless to the main netwrork, and connect to it the analog phones |
| 22:30.17 | Godfather_ | this could be possible no? |
| 22:30.46 | mchou | Godfather_: you're making this way to complicated |
| 22:31.24 | Godfather_ | mchou, i know the best solution that p3nguin give me to me is use just ip-phones |
| 22:31.31 | mchou | Godfather_: look into spa-2102 |
| 22:31.31 | p3nguin | You would end up using an ATA _and_ a wireless client bridge at each ATA to provide wireless IP connectivity to an analog phone. Sounds terrible. |
| 22:31.38 | Godfather_ | http://www.ciudadwireless.com/cisco_spa2102_-spa2000-_2-port_router_with_phone_ports-p-699.html |
| 22:31.41 | Godfather_ | i found that |
| 22:31.56 | Godfather_ | but its wired, need to be connected to the router :/ |
| 22:32.05 | mchou | Godfather_: that has 1 fxo and 1 fxs port, iirc |
| 22:32.40 | mchou | Godfather_: I don't get it |
| 22:32.55 | mchou | Godfather_: what's wrong with connecting to a router? |
| 22:34.31 | p3nguin | Routers are good for connecting two discontiguous networks together. |
| 22:34.41 | *** join/#asterisk Godfather_ (n=godfathe@187.Red-83-38-229.dynamicIP.rima-tde.net) |
| 22:34.44 | Godfather_ | mchou, sorry |
| 22:34.59 | Godfather_ | i seen since "Godfather_: that has 1 fxo and 1 fxs port, iirc" |
| 22:35.05 | mchou | router=>spa2102=>dect base station=>cordless phones |
| 22:35.12 | Godfather_ | but i see on the "Adaptador de telefono con 2 FXF y 2 puertos Ethernet (LAN+WAN)" |
| 22:35.19 | Godfather_ | 2 fxf and 2 wan |
| 22:35.25 | p3nguin | You could also use softphones on wirless computers. |
| 22:35.57 | mchou | Godfather_: what's the issue with what I described? |
| 22:36.01 | p3nguin | Personally, I do not like softphones are every day communications. |
| 22:36.07 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194) |
| 22:36.28 | [TK]D-Fender | Godfather_: Ok, I'm home.. lets try this again.... |
| 22:36.29 | Godfather_ | p3nguin, i prefer ip-phones too |
| 22:36.41 | [TK]D-Fender | Godfather_: For your analog line, the SPA-3102 will give you a FXO to take in the line and 1 FXS port |
| 22:36.48 | p3nguin | You're probably not going to want to use IP phones wirelessly, though. |
| 22:36.56 | [TK]D-Fender | Godfather_: if you want you can run your entire home's wiring on that ONE PORT |
| 22:37.11 | mchou | yup, agree |
| 22:37.39 | mchou | I think spa=xxx is rated for 3REN, iirc |
| 22:37.51 | [TK]D-Fender | mchou: 7 |
| 22:38.08 | Godfather_ | hum... |
| 22:38.11 | p3nguin | That's a fair amount of devices. |
| 22:38.14 | loather-work | that's a lot of phones. |
| 22:38.19 | mchou | 7? I didnt realize it went that high |
| 22:38.32 | [TK]D-Fender | mchou: To my recollection |
| 22:38.45 | loather-work | that's enough to ring a couple of those oldschool western electric rotary things |
| 22:39.06 | Godfather_ | Well, i think ill but the spa-3102 and test it |
| 22:39.15 | Godfather_ | *i will buy |
| 22:39.30 | mchou | spa-3102 does have one very annoying bug |
| 22:39.44 | mchou | double hook flash doesnt work right |
| 22:39.46 | p3nguin | loather-work: They probably don't go much over 1, do they? |
| 22:40.04 | p3nguin | 1.5 at the most, I guess. |
| 22:40.21 | Godfather_ | mchou, well, if you see other devices, are 3xpriced compared to the linksys |
| 22:40.22 | Godfather_ | http://www.ciudadwireless.com/sobre_gateways-c-128_190.html?sort=4a&page=1 |
| 22:40.26 | loather-work | p3nguin: some of the older ones do... the VIC-2FXS i have in one of these routers when set to 4REN wouldn't ring it audibly. |
| 22:40.42 | *** join/#asterisk ReDNeQ (i=ReDNeQ@70.114.229.58) |
| 22:40.44 | p3nguin | wow |
| 22:40.46 | Katty | ponders dinner. |
| 22:40.48 | pzn | _ShrikE: good hint :-) gotoiftime worked as I needed. thanks! |
| 22:40.52 | p3nguin | katty: Do it! |
| 22:41.02 | Katty | p3nguin: i'm pondering what to make for dinner. |
| 22:41.11 | TJNII | A rotary is aupposed to be 1REN |
| 22:41.14 | p3nguin | katty: Chicken pot pie? |
| 22:41.15 | dlynes | TSM: but you don't think you could have mismatched impedances, either? |
| 22:41.19 | loather-work | suppossed to be, yeah |
| 22:41.24 | Katty | p3nguin: hmm. |
| 22:41.29 | TJNII | So if a 4REN device couldn't ring one rotary, then it isn't putting out 4REN |
| 22:41.30 | loather-work | dlynes: that's likely the case. |
| 22:41.36 | Katty | p3nguin: possibility |
| 22:41.37 | dlynes | TSM: if downtown manhattan is anything like any other downtown in the world, it probably has a lot of old wiring kicking around |
| 22:41.55 | loather-work | TJNII: it's quite likely an impedance problem. |
| 22:42.10 | TJNII | Aah. |
| 22:42.52 | TJNII | notes that Katty is very food-centric |
| 22:43.07 | loather-work | it's the best way to a woman's heart. |
| 22:43.10 | mchou | http://www.voiplink.com/Linksys_SPA_3102_p/linksys-spa-3102.htm |
| 22:43.10 | Godfather_ | anyone tested IP-thelephones wireless? |
| 22:43.14 | loather-work | feed her something she loves! |
| 22:43.36 | Katty | TJNII: it's my hobby. |
| 22:43.37 | mchou | "Maximum Ringer Load: 3 REN" |
| 22:43.37 | p3nguin | My SPA-3102 seems very generic and cheap. |
| 22:43.38 | loather-work | avoid 802.11 phones. none of them seem to work properly. |
| 22:43.45 | Godfather_ | mchou, my not? |
| 22:43.46 | Godfather_ | 58 |
| 22:43.53 | Godfather_ | 52 sorry |
| 22:44.04 | Godfather_ | yours 78$ |
| 22:44.12 | *** join/#asterisk jmacz (n=jmacz@200.85.225.62) |
| 22:44.27 | mchou | you could probably get it for aroun $65 |
| 22:44.33 | mchou | (in US) |
| 22:44.45 | Katty | TJNII: it's what i enjoy doing to unwind from my day |
| 22:44.48 | Godfather_ | well, but im in spain, this could be the reason :P |
| 22:44.49 | *** join/#asterisk jtodd (i=elr4knf3@asterisk/community-director-and-tie-dye-shirt-lover/jtodd) |
| 22:44.49 | *** mode/#asterisk [+o jtodd] by ChanServ |
| 22:44.57 | mchou | but that requires careful shopping |
| 22:45.05 | Godfather_ | mchou, have you tested a wireless ip-phone? |
| 22:45.05 | TSM | dlynes: yeh true |
| 22:45.23 | mchou | do no buy wifi IP phones |
| 22:45.27 | mchou | not* |
| 22:45.32 | Godfather_ | mchou, see http://www.ciudadwireless.com/cisco_wip310-g2_wireless-g_phone_-europe--p-2439.html |
| 22:45.42 | p3nguin | I have a cordless SIP phone that works pretty good. |
| 22:45.58 | Godfather_ | cordless = wireless ? |
| 22:46.03 | p3nguin | not really |
| 22:46.09 | loather-work | the only wireless ones that work at all are DECT phones. 802.11 phones suck. |
| 22:46.16 | mchou | Godfather_: cordless != wireless |
| 22:46.19 | p3nguin | cordless does mean without a wire, but it's not Wi-Fi. |
| 22:46.24 | mchou | wirless == wifi |
| 22:46.33 | [TK]D-Fender | O>o |
| 22:46.34 | Godfather_ | mchou, what problems have wifi phones? |
| 22:46.34 | mchou | cordless = DECT |
| 22:46.59 | p3nguin | It's a regular 2.4GHz cordless phone, but does SIP over Ethernet instead of being analog. |
| 22:47.00 | [TK]D-Fender | ~wifivoip |
| 22:47.01 | infobot | [~wifivoip] Wi-Fi (802.11a/b/g) VoIP phones on the market may possess any of the following drawbacks : poor battery life, limited range, lack of call features (many lack transfer, etc), poor or no NAT traversal, shoddy construction, etc. Because of this they are usually not recommended. Some alternatives are : VoIP ATA + cordless, DECT w/ VoIP base, etc |
| 22:47.03 | [TK]D-Fender | ^^^^^^ |
| 22:47.06 | mchou | Godfather_: most wifi sip phones outdate very quickly |
| 22:47.12 | mchou | lousy battery life |
| 22:47.41 | loather-work | you can't really do any meaningful QoS on 802.11 either |
| 22:47.56 | Godfather_ | mchou, p3nguin, then need a base station connected wired to the router no? |
| 22:47.56 | loather-work | so someone's bittorrent download in the other room will slam your phone call |
| 22:48.05 | *** join/#asterisk e4 (n=e4@rrcs-76-79-48-214.west.biz.rr.com) |
| 22:48.19 | p3nguin | godfather_: yes |
| 22:48.26 | mchou | Godfather_: [14:35:04] <mchou> router=>spa2102=>dect base station=>cordless phones |
| 22:48.51 | mchou | Godfather_: you're a bit slow on the uptake :) |
| 22:48.54 | TSM | if you want wifi sip in an office, then split networks |
| 22:49.14 | TSM | but best solution is dect, but expensive for offices |
| 22:49.34 | mchou | TSM: depends on his office :) |
| 22:49.59 | TSM | we looked at it but did not go for it in the ext |
| 22:50.00 | TSM | end |
| 22:50.22 | mchou | TSM: what did you guys use instead? |
| 22:50.33 | Godfather_ | mchou, well, first i need a spa3102 as i understand, then the config you told |
| 22:50.37 | *** join/#asterisk aidinb (n=Aidin@24-182-32-138.static.lnbh.ca.charter.com) |
| 22:50.51 | TSM | determined that the need was not as great as it was made out by the directors |
| 22:50.58 | mchou | lol |
| 22:51.06 | *** join/#asterisk tzafrir (n=tzafrir@212.179.75.202) |
| 22:51.07 | mchou | you went without |
| 22:51.07 | dlynes | TSM: isn't that always the case? |
| 22:51.17 | TSM | i was looking at the spectralink stuff, expeeeeennnnsive |
| 22:51.20 | Godfather_ | mchou, its to me? |
| 22:51.24 | thuddwhirr | is anyone familiar with how the "h" extension works when you have a Dial applicaiton in your dialplan? |
| 22:51.33 | mchou | no, that was to TSM |
| 22:51.37 | Godfather_ | ok |
| 22:51.45 | [TK]D-Fender | thuddwhirr: You don't dial from "h". EVER |
| 22:51.48 | johnakabean | Gotoif($["${__MOHCLASS}" != ""]:somecontext) would this check if the variable is set or not? |
| 22:51.50 | TSM | thuddwhirr: i think its what to do when the user has hung up |
| 22:51.51 | dlynes | thuddwhirr: it automatically gets called, if you have autofallthrough=yes |
| 22:52.01 | [TK]D-Fender | thuddwhirr: Aside from that, perhaps you should ask a more specific question |
| 22:52.17 | [TK]D-Fender | johnakabean: Yes |
| 22:52.17 | thuddwhirr | sorry, wasn't expecting the answeres so quickly :) |
| 22:52.24 | Godfather_ | mchou, then i need to buy spa3102 and the 2102, first to convert analog line to voip, and the second to convert the voip to analog phones no? |
| 22:52.26 | p3nguin | thuddwhirr: Sure. After your dial exits, it will finish the dialplan for that exten. Then, when that has expired, h runs. |
| 22:52.33 | mchou | Godfather_: no |
| 22:52.37 | Godfather_ | :-( |
| 22:52.47 | thuddwhirr | i'm trying to capture CHANNEL(rtpqos, audio, all) for the two channels involved in the dial application |
| 22:52.49 | mchou | Godfather_: only SPA-3102 will be required |
| 22:52.51 | [TK]D-Fender | Godfather_: Yes |
| 22:52.56 | thuddwhirr | i was hoping to do so in the "h" extension |
| 22:53.05 | dlynes | Godfather_: the 3102 will convert analog to voip and voip to analog |
| 22:53.14 | thuddwhirr | but it seems to get called *once* when any party hans up |
| 22:53.18 | [TK]D-Fender | thuddwhirr: If you are in "h" you aren't **IN** a call |
| 22:53.19 | dlynes | Godfather_: it has both an fxs port and an fxo port |
| 22:53.33 | johnakabean | h is the hangup context |
| 22:53.33 | p3nguin | thuddwhirr: That's what it should do. h runs when the call is over. |
| 22:53.43 | thuddwhirr | well, the call doesnt seem to be over yet |
| 22:53.46 | thuddwhirr | the dial leg is over |
| 22:53.46 | p3nguin | hangup exten, not context. |
| 22:53.55 | thuddwhirr | er, yes |
| 22:53.55 | [TK]D-Fender | thuddwhirr: "h" = HANGUP. Yes.. its dead. |
| 22:54.00 | *** join/#asterisk slinksh0t (n=slinksh0@adsl-9-177-182.mia.bellsouth.net) |
| 22:54.04 | drmessano | "he's dead, jim" |
| 22:54.05 | johnakabean | either way, h is executed when the channel is about to die |
| 22:54.12 | Godfather_ | hehe, i think now i understand... at leas!! jeje |
| 22:54.18 | Godfather_ | *at least |
| 22:54.27 | thuddwhirr | right. the dial channel, this is expected, the channel is dead. |
| 22:54.34 | [TK]D-Fender | Godfather_: You only need the SPA-2102 on top of that if you want the phones to act independently |
| 22:54.35 | johnakabean | s means special but can you clarify that for me and them, drmessano |
| 22:54.53 | [TK]D-Fender | Godfather_: otherwise the SPA-3102 might be enough for your full needs |
| 22:54.55 | thuddwhirr | the channel that called the dial continues, but when its over, i dont se "h" getting called |
| 22:55.00 | p3nguin | The s exten matches when there is no called number. |
| 22:55.11 | p3nguin | With SIP, you always have a called number. |
| 22:55.15 | Godfather_ | [TK]D-Fender, i think the spa-2102 to have more than one fxo port, and can connect more analog lines |
| 22:55.30 | dlynes | p3nguin: not if it's a macro :) |
| 22:55.32 | mchou | Godfather_: yup |
| 22:55.35 | Godfather_ | this because the 3102 have just one line |
| 22:55.37 | [TK]D-Fender | Godfather_: No, the SPA-2102 is for connecting PHONES, not LINES |
| 22:55.45 | Godfather_ | yes, analog phones sorry |
| 22:55.47 | drmessano | S means "special"? news to me |
| 22:55.48 | mchou | Godfather_: correct |
| 22:55.51 | [TK]D-Fender | Godfather_: an that is a 2-port unit |
| 22:55.52 | p3nguin | dlynes: There was still a called number, or the macro would not execute. |
| 22:55.59 | johnakabean | penguin, i use s in a lot of my dialplans as a starting point; i point to it from another context that catches the number dialed |
| 22:56.07 | dlynes | p3nguin: yes, but not a called 'exten'sion |
| 22:56.11 | [TK]D-Fender | drmessano: I'm sensing a whole lot of "special" around here.... |
| 22:56.20 | [TK]D-Fender | drmessano: In an Olympic sort of way... |
| 22:56.21 | mchou | Godfather_: linksys make many FXS port units |
| 22:56.28 | Godfather_ | with a 3102 and a 2102 i well be able to connect 3 analog phones |
| 22:56.31 | darkdrgn2k3 | how can i prepand a number to cid? |
| 22:56.32 | johnakabean | Gotoif($["${__MOHCLASS}" != ""]:somecontext) would this check if the variable is set or not? |
| 22:56.36 | [TK]D-Fender | johnakabean: No.... |
| 22:56.53 | [TK]D-Fender | [17:55]<johnakabean>penguin, i use s in a lot of my dialplans as a starting point; i point to it from another context that catches the number dialed <-- no was for this |
| 22:56.55 | mchou | Godfather_: figure out how many phones you need to hook up first |
| 22:57.00 | dlynes | darkdrgn2k3: Set(CALLERID(num)=${prepend}${CALLERID(num)}) |
| 22:57.08 | [TK]D-Fender | [17:56]<johnakabean>Gotoif($["${__MOHCLASS}" != ""]:somecontext) would this check if the variable is set or not? <- sure, why not... |
| 22:57.08 | johnakabean | just conversating, fender on that one |
| 22:57.29 | mchou | Godfather_: http://www.cisco.com/en/US/products/ps10025/index.html (just an example) |
| 22:57.31 | TSM | Godfather_: linksys make a 8 port FXS unit too |
| 22:57.45 | Godfather_ | TSM, price? |
| 22:57.49 | dlynes | TSM: that would be a huge waste of money |
| 22:57.50 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 22:57.56 | TSM | UK £200 UK |
| 22:58.00 | Godfather_ | i can buy for learning purpose a 50 device, not a 200 |
| 22:58.08 | mchou | dlynes: why? |
| 22:58.09 | johnakabean | what do you mean why not? would that successfully check to see if the variable is set; I don't want it to catch every channel |
| 22:58.11 | TSM | its also fully T38 unit |
| 22:58.29 | dlynes | mchou: just the quality that linksys is famous for |
| 22:58.34 | dlynes | shudders. |
| 22:59.03 | mchou | umm, if you know ATA that is better than linksys, speak up |
| 22:59.10 | mchou | dlynes: ^^^ |
| 22:59.13 | dlynes | I can just see 6 out of the 8 fxs ports going dead within a year |
| 22:59.26 | dlynes | mchou: Mediatrix? Audiocodes? Quintum? |
| 22:59.51 | TSM | http://www.cisco.com/en/US/prod/collateral/voicesw/ps6790/gatecont/ps10024/ps10025/data_sheet_c78-504120.html |
| 22:59.55 | TSM | SPA-8000 |
| 23:00.10 | [TK]D-Fender | [17:58]<johnakabean>what do you mean why not? would that successfully check to see if the variable is set; I don't want it to catch every channel <- catch every channel? PARDON?! |
| 23:00.32 | johnakabean | I don't want every channel to be sent to that extension |
| 23:00.43 | [TK]D-Fender | Godfather_: How many phones do you have? |
| 23:01.27 | drmessano | dlynes: I suppose a Grandstream ATA is better? |
| 23:01.29 | [TK]D-Fender | johnakabean: ... ok/fine/sure |
| 23:01.41 | Godfather_ | [TK]D-Fender, i repeat, its for learning purpose, i'll buy a 3102 |
| 23:01.50 | Godfather_ | and test it with my analog phone |
| 23:01.55 | [TK]D-Fender | Godfather_: So you have 1 analog line and a few analog phones? |
| 23:02.06 | mchou | Godfather_: if you are just learning 3102 will suffice |
| 23:02.13 | Godfather_ | [TK]D-Fender, now yes, but i will try a ip-phone too. |
| 23:02.16 | *** part/#asterisk ruben23 (n=RPL@122.55.48.243) |
| 23:02.31 | [TK]D-Fender | Godfather_: No real need for most people. |
| 23:02.46 | mchou | I dunno |
| 23:02.47 | [TK]D-Fender | Godfather_: You can get one if you want, but for home use there really isn't much of a point |
| 23:02.54 | mchou | ip phones are nice |
| 23:03.07 | Godfather_ | [TK]D-Fender, i know, but then i'll buy a ip-phone |
| 23:03.13 | [TK]D-Fender | mchou: Nice, yes... needed? Lets face it... little more than a toy |
| 23:03.20 | Godfather_ | and will be able to set up an office, i think. |
| 23:03.20 | mchou | nah |
| 23:03.30 | Godfather_ | just learning. |
| 23:03.39 | mchou | full duplex speakerphone I'd say is de rigeur |
| 23:03.47 | [TK]D-Fender | Godfather_: Ok, as long as you realize that you aren't going to be transferring calls at home all day you won't realize the real value in your situation. |
| 23:03.51 | TSM | i like it that i can easly give a phone to a user and he/she can take it home and presto its fully working like at the office with remote config |
| 23:04.09 | *** join/#asterisk Badrobot- (n=badrobot@cpe-76-173-229-89.socal.res.rr.com) |
| 23:04.10 | mchou | not too many consumer dect speakerphones are full duplex |
| 23:04.18 | Godfather_ | [TK]D-Fender, last question |
| 23:04.53 | johnakabean | dlynes, why would the fxs ports go dead? |
| 23:05.24 | TSM | because it looses SIP registration |
| 23:05.26 | Godfather_ | if a converstaion is stablished between analog phone for example, with my eschema, other IP-phones on the network when try to make a phone call throght the pstn what happens? |
| 23:05.45 | Godfather_ | i dont know if i explained it well, sorry for my english |
| 23:05.47 | TSM | depends on your setuup |
| 23:06.30 | Godfather_ | I mean, now in my house, without any PBX, if i hang up the phone i heard the conversation on the other analog phone |
| 23:06.35 | [TK]D-Fender | Godfather_: If your line is in use, the SPA will report that it is "busy" and Asterisk can choose another resource to call out if you configure it to, or indicate that the line is busy, etc. |
| 23:06.49 | Godfather_ | [TK]D-Fender, ah, ok |
| 23:06.52 | Godfather_ | ty |
| 23:06.59 | [TK]D-Fender | Godfather_: Or you could do what I advised earlier |
| 23:07.13 | Godfather_ | what you advised? |
| 23:07.14 | [TK]D-Fender | Godfather_: and run your enire home off of ONE FXS port. |
| 23:07.38 | [TK]D-Fender | godAnd then all of your analog phones act like they did before where anyone could pick up and join that call |
| 23:08.11 | Godfather_ | [TK]D-Fender, well, maybe this will be the setup with the SPA3102 that i will choose |
| 23:08.33 | Godfather_ | thx again |
| 23:08.34 | [TK]D-Fender | Godfather_: Perhaps it will. |
| 23:08.46 | [TK]D-Fender | Godfather_: it is a good purchase regarless |
| 23:09.22 | *** join/#asterisk denon (i=denon@sassinak.net) |
| 23:09.23 | *** mode/#asterisk [+o denon] by ChanServ |
| 23:09.30 | Godfather_ | [TK]D-Fender, what means regarless? |
| 23:09.46 | [TK]D-Fender | Godfather_: "regardless". Meaning "either way" |
| 23:09.56 | Godfather_ | ah ok |
| 23:10.01 | Godfather_ | to carry on |
| 23:10.50 | [TK]D-Fender | Godfather_: So if you don't need the FXO port it's FXS port could still be useful to you... it is a very versatile little box |
| 23:12.43 | TSM | does anyone have the SPA compiler tool? |
| 23:13.58 | Godfather_ | [TK]D-Fender, huh? |
| 23:14.43 | dlynes | johnakabean: no idea...why do their network ports go dead? |
| 23:14.51 | Godfather_ | i need the FXO port to connect the wall jack to my pc |
| 23:15.25 | dlynes | drmessano: Mediatrix, Audiocodes and Quintum are comparable to Grandsucks in your mind? |
| 23:15.30 | [TK]D-Fender | Godfather_: it has 1 FXS and 1 FXO interface. Meaning if you don't need one half, the4 other might still be useful. Also good if you have a line somewhere remote that you want to be able to use with *. This is advantage over using PCI cards |
| 23:17.37 | Godfather_ | [TK]D-Fender, "Meaning if you don't need one half"? what means here one half? |
| 23:17.48 | TSM | mehhhhh |
| 23:18.01 | [TK]D-Fender | Godfather_: the SPA-3102 lets you use a LINE... AND a PHONE. |
| 23:18.09 | *** join/#asterisk benklop (n=ben@174-16-222-132.hlrn.qwest.net) |
| 23:18.16 | [TK]D-Fender | Godfather_: both independent of each other |
| 23:18.21 | benklop | hello everyone |
| 23:18.49 | Godfather_ | [TK]D-Fender, i understand that: 3102 have a FXO to connect to the wall (the line pstn), and a FXS to connect the analog phone |
| 23:18.53 | Godfather_ | this ir right no? |
| 23:18.58 | [TK]D-Fender | Godfather_: Yes it is. |
| 23:19.05 | benklop | is there a way to reduce the latency when two calls are bridged together? |
| 23:19.23 | Godfather_ | [TK]D-Fender, well, then i dont understand what you say in that line |
| 23:19.34 | benklop | as in connecting one call to another call which is parked |
| 23:19.39 | [TK]D-Fender | Godfather_: Now lets say you decide to get rid of your analog line and use a VOIP Telephony service instead. You wouldn't need the FXO anymore. However the FXS port on it would still be useful for you |
| 23:19.47 | johnakabean | native briding benklop |
| 23:20.10 | johnakabean | you have to have canreinvite=always |
| 23:20.58 | Godfather_ | [TK]D-Fender, ahhh ok, i can be able to use the fxo port as a fxs port |
| 23:21.05 | [TK]D-Fender | Godfather_: No. |
| 23:21.09 | Godfather_ | arg |
| 23:21.17 | [TK]D-Fender | Godfather_: the SPA-3102 has 2 ports. 1 is FXS. 1 is FXO |
| 23:21.31 | benklop | johnakabean: ah, ok. that makes sense now... i'm pretty new to asterisk and voip and i forgot that sip endpoints can directly connect |
| 23:21.36 | [TK]D-Fender | Godfather_: they function independently |
| 23:21.44 | johnakabean | with native bridging, the endpoints of the calls are connected together. Say you have two providers, provider a and provider b. |
| 23:21.51 | johnakabean | :) |
| 23:22.17 | *** join/#asterisk e4 (n=e4@rrcs-76-79-48-214.west.biz.rr.com) |
| 23:22.45 | johnakabean | the bad part - your pbx loses complete control |
| 23:23.00 | johnakabean | until one endpoint hangs up |
| 23:23.12 | *** join/#asterisk denon (i=denon@sassinak.net) |
| 23:23.12 | *** mode/#asterisk [+o denon] by ChanServ |
| 23:23.59 | benklop | johnakabean: that's true.. is native bridging even possible when one endpoint is on the other side of a NAT |
| 23:24.36 | benklop | ? |
| 23:24.37 | Godfather_ | [TK]D-Fender, okok, i understand that if for any reason i dont use the fxo port, i will be able to connect the 3102 to the router and have a analog phone connected to it remotely |
| 23:25.37 | [TK]D-Fender | Godfather_: You will ALWAYS have the SPA conencted to your network.... |
| 23:25.47 | Godfather_ | yes, thats true. |
| 23:27.28 | *** join/#asterisk lucasb (n=lucasb@S0106000e08cda7cc.ok.shawcable.net) |
| 23:28.48 | *** join/#asterisk jmacz (n=jmacz@200.85.225.62) |
| 23:30.01 | Godfather_ | [TK]D-Fender, i see the 3102 have lan and wan ports, just need to connect the lan port no? |
| 23:30.31 | [TK]D-Fender | Godfather_: jsut saying that Godfather_ You don't need to use it as a router. |
| 23:30.48 | thuddwhirr | let me take another shot at this question :) Does anyone know how you would go about getting channel(rtpqos,audio) for both the channels after the Dial application complets? |
| 23:31.18 | Godfather_ | [TK]D-Fender, yep, i dont need to act as a router |
| 23:31.26 | Godfather_ | i'll connect to my wrt54gl |
| 23:31.28 | [TK]D-Fender | thuddwhirr: You can't. |
| 23:31.35 | [TK]D-Fender | Godfather_: Sure |
| 23:31.52 | Godfather_ | [TK]D-Fender, then, the wan port will be empty |
| 23:32.18 | thuddwhirr | seriously? that seems to limit the usefulness of that metric. . . |
| 23:33.36 | [TK]D-Fender | thuddwhirr: Because you seem to have issues understanding dialplan flow |
| 23:33.59 | thuddwhirr | i do. which is why i'm here asking questions |
| 23:35.04 | sier | What's cheaper? Internal Gateway Cards or External USB Cards? |
| 23:35.28 | [TK]D-Fender | seanbright: What "USB cards"? |
| 23:35.42 | sier | gateway* |
| 23:35.43 | Qwell | [TK]D-Fender: probably means Xorcom |
| 23:36.03 | [TK]D-Fender | Qwell: I try to avoid guessing what crazy people may be thinking |
| 23:36.19 | [TK]D-Fender | Qwell: And I like to hand out rope :) |
| 23:41.16 | Kobaz | [Nov 3 18:41:02] WARNING[26105]: chan_iax2.c:1219 __send_lagrq: I was supposed to send a LAGRQ with callno 1055, but no such call exists (and I cannot remove lagid, either). |
| 23:41.23 | Kobaz | i always get a boatload of those |
| 23:41.33 | Kobaz | is there some sort of setting to fix that? |
| 23:41.55 | sier | are my questions really stupid to the point where people ignore me? :P |
| 23:42.09 | Kobaz | could be |
| 23:42.26 | *** join/#asterisk coppice (n=chatzill@171.168.17.210.dyn.pacific.net.hk) |
| 23:42.42 | [TK]D-Fender | sier: No, this is the point where you answer my question with a stupid answer and THEN we start to ignore you :) |
| 23:43.22 | Kobaz | does the new calltoken stuff have anything to do with LAGRQ's |
| 23:43.26 | sier | but I didn't see any questions.. hmm |
| 23:43.41 | [TK]D-Fender | [18:35]<[TK]D-Fender>seanbright: What "USB cards"? |
| 23:43.44 | [TK]D-Fender | seiBad aim |
| 23:43.50 | [TK]D-Fender | sier: Bad aim |
| 23:43.52 | [TK]D-Fender | gah. |
| 23:44.11 | [TK]D-Fender | sier: Ok, so what "USB cards"? |
| 23:45.05 | sier | I see.. well I was just talking about it in general, I have a server.. that will eventually run asterisk.. I wanted to know ("in general"), what is cheaper.. Low density internal analog cards or external cards.. I don't have any models in mind.. |
| 23:45.34 | [TK]D-Fender | sier: Give us your expected usage and we'll give you our suggestions |
| 23:46.25 | sier | I need something with 1-3 ports (max), usage will be minimum.. I'm probably going to receive/make 5 calls a day |
| 23:46.42 | [TK]D-Fender | sier: thats a 300% margin there. |
| 23:46.47 | Kobaz | you'll need some serious connectivity to handle such a load |
| 23:46.47 | TSM | just come across the message 'wheezles have eaten our phonesystem' haaa |
| 23:46.51 | *** join/#asterisk ardnat (i=18e152f7@gateway/web/freenode/x-d30312a3a5b7f991) |
| 23:46.51 | [TK]D-Fender | sier: and what KIND of "ports"? |
| 23:47.32 | sier | RJ-11 |
| 23:47.49 | [TK]D-Fender | sier: that is a JACK. |
| 23:47.54 | [TK]D-Fender | sier: What SIGNALLING ON IT?> |
| 23:47.56 | Kobaz | haha |
| 23:48.20 | ardnat | Question:How can dial() and branchoff & transfer to a conference room, |
| 23:48.30 | Kobaz | TSM: you mean weasel? |
| 23:48.50 | *** join/#asterisk denon (n=denon@synapse.subneural.net) |
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| 23:48.52 | ardnat | basicly, enter a number, and it will dial that number and send it to the conference room |
| 23:49.38 | ardnat | i currently know how to make it call up a number by entering it and getting the numbers entered |
| 23:49.46 | *** join/#asterisk denon (n=denon@synapse.subneural.net) |
| 23:49.46 | *** mode/#asterisk [+o denon] by ChanServ |
| 23:49.48 | sier | [TK]D-Fender analog? |
| 23:50.11 | [TK]D-Fender | sier: WTF are you planning on plugging INTO this device? |
| 23:50.22 | Godfather_ | [TK]D-Fender, i go to sleep, thx a lot (and also mchou p3nguin and others) |
| 23:50.37 | Godfather_ | bye, i will test it in 2 day more or less |
| 23:50.39 | Godfather_ | :D |
| 23:50.59 | sier | [TK]D-Fender that's a good question.. |
| 23:51.45 | [TK]D-Fender | sier: Perhaps you should reconsider your asking advice for needs you can't even define. |
| 23:51.50 | ardnat | can you give me an example useage of transfer() |
| 23:52.00 | Kobaz | core show application transfer |
| 23:52.09 | [TK]D-Fender | ardnat: NOT APPLICABLE |
| 23:52.32 | TSM | Kobaz: yup, i just C/P a script for mass rebooting my polys, used it and then heard it |
| 23:52.47 | ardnat | thanks kob |
| 23:52.58 | Kobaz | rebooting polycoms are fun |
| 23:53.09 | Kobaz | when are they going to support reconfiguring without rebooting |
| 23:53.23 | TSM | beep beep beep beep beep beep beep beep beep :) |
| 23:53.28 | TSM | yup i wish that too |
| 23:53.29 | sier | [TK]D-Fender yes.. my knowledge is minimum.. what I know is.. I have a server running linux, I plan on installing asterisk on it, and I want to plug a analog phone @ it, using a RJ-11 jack.. and I want it to be able to receive calls, and have an auto-attendant.. |
| 23:53.32 | [TK]D-Fender | Kobaz: Why bother? Do it right the first time and you never have to reboot them :p |
| 23:53.45 | Kobaz | [TK]D-Fender: but what happens when you need to change something |
| 23:53.53 | [TK]D-Fender | sier: Good... you want to plug a PHONE into it. |
| 23:54.15 | [TK]D-Fender | sier: That was only a tad short of being harder than squeezing blood from a stone... |
| 23:54.27 | TSM | [TK]D-Fender: was nearly correct, just some teathing issues, the ones that nobody says would be nice to have when you had the meeting several weeks back about the new system |
| 23:54.29 | [TK]D-Fender | sier: So you want to use an analog phone with *. What else? |
| 23:55.06 | Kobaz | sier: how are you recieving calls... do you have phone lines in your wall? do you want to get calls over the interwebs? |
| 23:55.10 | [TK]D-Fender | TSM ? |
| 23:55.39 | Kobaz | sier: oh... and check out the book... it's free online |
| 23:55.39 | sier | I want to get calls over the interwebs.. I have a 1-800 number, and I wanted to link to this.. I also have google centralvoice , not sure if this matter, or if it can be used.. |
| 23:55.44 | Kobaz | ~book |
| 23:55.45 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 23:56.25 | [TK]D-Fender | sier: then if all you want is to use analog phones with *, depending on the volume, go with Linksys SPA adapters like the PAP2T-NA |
| 23:56.26 | sier | I started reading it, but It has 600 pages, it will take me a while.. i just wanted to implement something, and start making tweaks as I go.. |
| 23:56.31 | [TK]D-Fender | sier: $50 for 2 ports |
| 23:56.38 | sier | hm.. I see |
| 23:56.52 | Kobaz | linksys spa is nice |
| 23:56.59 | Kobaz | linksys fxo would be nice if callerid worked right |
| 23:56.59 | tzafrir | ~fxsfxo |
| 23:57.00 | infobot | [~fxsfxo] An FXO port expects to receive dialtone and receive ring voltage. You can connect it to a PSTN line from the telco. An FXS port expects to provide dialtone and provide ring voltage. You can connect a phone or a fax to it. |
| 23:57.08 | ardnat | kob whats the cmd for getting the entered digits again? |
| 23:57.13 | Kobaz | ardnat: Read() |
| 23:57.17 | ardnat | ty |
| 23:57.28 | tzafrir | sier, that's the "type" TK was asking about earlier |
| 23:57.35 | TSM | [TK]D-Fender: you replied to Kobas, his message was in relation to me having to mass reboot my pollys |
| 23:57.45 | [TK]D-Fender | TSM: AH. |
| 23:58.14 | Kobaz | busily continues converting more ael to perl |
| 23:58.22 | tzafrir | That PAP2 is a 2-ports FXS, for instance |
| 23:58.53 | tzafrir | anybody her eactually uses res_lua? |
| 23:58.54 | sier | I see, sorry guys.. So, how does it work? internet > pap2t-na > phone ? Where does asterisk come in place in this? I need to set-up auto-attendant and a voicemail.. |
| 23:59.08 | *** join/#asterisk wonderworld (n=ww@ip-62-143-22-226.unitymediagroup.de) |
| 23:59.43 | Kobaz | tzafrir: i tried lua... it's kinda a pain |
| 23:59.48 | [TK]D-Fender | sier: SPA plugs on your LAN. Internet > your router or whatever > * server > SPA > analog phone |
| 23:59.50 | tzafrir | sier, if you can manage all the logic you want with just the PAP2 device, that's fine. If not, you'll need a smarter system such as Asterisk |
