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| 00:33.19 | dan__t | Yeah, I saw Lumenvox, also read about in the Asterisk book. |
| 00:33.52 | dan__t | Thanks, jaytee. I'll look in to that. |
| 00:33.58 | dan__t | When they license per port, what does that mean? |
| 00:34.28 | beek | dan__t: That's the number of concurrent connections to the speech recognition engine. |
| 00:36.03 | dan__t | got it |
| 00:37.30 | *** join/#asterisk telnettech (n=telnette@206.48.21.148) |
| 00:40.57 | telnettech | have a question....is there a way in asterisk to make the calls to connect faster than they are now |
| 00:41.06 | telnettech | there is a 2 second delay it seems |
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| 00:41.58 | [TK]D-Fender | telnettech: what "calls"? |
| 00:41.58 | Nugget | telnet is eeeeeeevil! |
| 00:43.04 | carrar | faster box!! :) |
| 00:43.35 | telnettech | both internal and outbound calls |
| 00:43.45 | [TK]D-Fender | telnettech: .... WTF |
| 00:43.51 | [TK]D-Fender | telnettech: what "calls"? <---------- |
| 00:45.25 | carrar | change calldelay=2 to calldelay=9 |
| 00:45.27 | carrar | err 0 |
| 00:45.56 | [TK]D-Fender | carrar: wasn't it "slow=no"? |
| 00:46.06 | carrar | hahah oh yeah that changed that in 1.4 |
| 00:46.15 | telnettech | TK: the internal calls and the outbound calls |
| 00:46.16 | [TK]D-Fender | carrar: Gotta keep up... |
| 00:46.28 | carrar | hard too with so many calls |
| 00:46.29 | [TK]D-Fender | telnettech: useless description. |
| 00:46.40 | telnettech | what else do you need |
| 00:46.49 | telnettech | it is the extension to extension calls |
| 00:46.57 | [TK]D-Fender | telnettech: what DEVICE you're friggen talking about. |
| 00:46.59 | telnettech | and any calls going out of the system to the telco |
| 00:47.24 | telnettech | well it would be my asterisk box? and the outbound is thru a fxo gateway |
| 00:47.27 | [TK]D-Fender | telnettech: calls don't magically go to a telco. |
| 00:47.53 | telnettech | it is not just outbound calls to Telco |
| 00:47.55 | [TK]D-Fender | telnettech: WHAT ^&%$ING MODEL OF DEVICE!? |
| 00:48.23 | telnettech | it is a Mediatrix 1204 FXO gateway. |
| 00:48.31 | [TK]D-Fender | telnettech: Don't ask your mechanic how to fix your transmission and then at the end of the conversation say "Oh, I thought you knew I had a MANULA transmission!" |
| 00:48.50 | telnettech | well the mechanic would check that out!!!!! |
| 00:49.15 | [TK]D-Fender | telnettech: We're not psychic so wake up and realize you need to tell us these things |
| 00:49.20 | telnettech | im tell ing you any internal calls, which would mean ext to ext and outbound calls.... |
| 00:49.35 | [TK]D-Fender | external? How the F are we supposed to know what YOU use to go out? |
| 00:49.49 | [TK]D-Fender | telnettech: A TDM card? VoIP provider? direct GSM interface? |
| 00:49.56 | [TK]D-Fender | telnettech: F-ING SMOKE SIGNALS... |
| 00:50.08 | [TK]D-Fender | telnettech: PRI? Analog? |
| 00:50.20 | carrar | I use the SMOKE-X GW 2HI |
| 00:50.29 | telnettech | nevermind have a nice night....i dont need all the cursing.....all you have to do is ask questions |
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| 00:50.46 | [TK]D-Fender | telnettech: How do I know your PHONE isn't to blame, and not whatever magically equipement and/or service lets you reach the PSTN? |
| 00:51.12 | [TK]D-Fender | telnettech: When you want help, don't make people chase after you because you can't think to provide details. |
| 00:51.15 | telnettech | TK: if it is happening system wide, i would say it is a system problem not a phone problem |
| 00:51.28 | [TK]D-Fender | telnettech: Go show us in your CLI output at what point delay is introduced |
| 00:51.41 | telnettech | it is before the call is connected |
| 00:51.51 | telnettech | i cant show you on the CLI cause you cant see it |
| 00:51.55 | [TK]D-Fender | telnettech: And don't forget, on inbound ANALOG your interface will probably WAIT for CALLERID <- |
| 00:52.13 | [TK]D-Fender | telnettech: For outbound... well it takes time for the interface to DIAL the digits and the telco to take them and process. |
| 00:52.33 | justmehere | yeah, heck, I have worse with my cell phone |
| 00:52.36 | carrar | could be the phones own dialplan waiting for more digitals before it dials |
| 00:52.38 | [TK]D-Fender | telnettech: how fast do YOU dial DTMF? |
| 00:52.53 | [TK]D-Fender | carrar: Yup, compounding delay |
| 00:52.56 | carrar | err digits |
| 00:52.59 | telnettech | it seems to be too long of a delay though....that leads me back to my original question!!!! where in asterisk can you see what may be causing the delay |
| 00:53.12 | justmehere | console, |
| 00:53.21 | justmehere | set your debug level and get in there and log the goodies, |
| 00:53.32 | telnettech | CAR: i can see that part of the delay from the phone....im talking about when it should be calling |
| 00:53.34 | path_ | is it possible to set up a pbx and connect to a voip provider without using a PCI card ? |
| 00:53.38 | justmehere | or in my case when I had some delay's using a GSM to SIP gateway device |
| 00:53.41 | [TK]D-Fender | telnettech: your PHONE can be wasting time before sending the call to * due to its dialplan, your gateway needs a few sec to pass the # to the PSTN and for them to start ringingt. |
| 00:53.45 | [TK]D-Fender | telnettech: Take your pick./ |
| 00:53.49 | carrar | path, yes |
| 00:54.03 | justmehere | I wrote a small syslog application and used the syslog output to check timings and stuff |
| 00:54.14 | justmehere | so if the device has syslog option use that too |
| 00:54.17 | telnettech | TK: so what you are saying is that there is no where in the asterisk to be able to "adjust" delay |
| 00:54.18 | justmehere | to get some time stamping |
| 00:54.27 | [TK]D-Fender | telnettech: * does not introduce delay <- |
| 00:54.43 | justmehere | yeah, it's all the other interconnects, sip provider, upstream telco, etc |
| 00:54.50 | [TK]D-Fender | telnettech: So look at the precise point that SATARTS the call, then look from the moment you see * passing the call to your gateway |
| 00:54.56 | justmehere | * still has to wait for acks and stuiff before it can do what it does |
| 00:54.56 | telnettech | ok thats what i wanted...how hard was that to say that asterisk doesnt have delay built into it as a function |
| 00:55.16 | [TK]D-Fender | telnettech: that'd be retarded. * does what YOU tell it. |
| 00:55.27 | [TK]D-Fender | telnettech: So go look at each end of the call. |
| 00:55.33 | justmehere | it's just doing things as you ask, unless you put waits in the dialplan |
| 00:55.59 | telnettech | and i dont know all the options that the asterisk is capable of doing.....thats why i ask questions |
| 00:56.17 | justmehere | but other than that, it's just doing things as fast as it's allowed to with other systems having to do their things and other various network connectivity issues that ALL add up to latency in the execution of a dialplan |
| 00:56.37 | justmehere | it's just how it is |
| 00:57.18 | justmehere | but for sure if you watch * console output, the stuff that it is doing internally is going prettty good and fast |
| 00:57.45 | telnettech | just: thanks.....i dont have any timers for waiting or anything....i will check my phones and network |
| 00:57.54 | justmehere | the lags invariably happen outside of the box |
| 00:58.09 | Defraz | Does anyone have any experience with asterisk and using a Cisco 1760 and a pri card as a voip gateway. |
| 00:58.34 | justmehere | sorry no, all SIP/IAX here |
| 00:58.52 | Defraz | Specifically the dial-peers and dial-pattern syntax |
| 01:00.53 | carrar | google is your friend |
| 01:01.18 | justmehere | indeed, I agree, spent many the long cold night with my friend Google |
| 01:02.44 | telnettech | TK: relax dude. not all of us in here are asterisk gurus since the day it was thought of. I have less than 6 months. Before August 2008, i was working on Avaya and Ericsson legacy pbx. I have no clue about script writing, networking and computer programming. Im not stupid and I am learning as I go about all of this at one time |
| 01:03.35 | telnettech | i ask generic questions cause i dont fully understand how all this interfaces. But im willing to be patient and learn as I go |
| 01:03.52 | justmehere | yeah, that's what I've done |
| 01:04.03 | justmehere | been at it for about 3 years now |
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| 01:04.41 | telnettech | I do appreciate the stuff you have taught me over the time i have been coming in here. I dont know you from adam but you are a very intelligent person from what i have seen on your answers with me and other people |
| 01:05.10 | telnettech | but the rude tone is realy uncalled for. Please dont take this the qrong way. I would like for you to continue to answer my questions as I have them |
| 01:06.19 | telnettech | i feel i could learn alot. Im just asking for a little patience as I go thru this growth of learning my way thru this world of asterisk |
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| 01:07.20 | carrar | it's good to be back on efnet |
| 01:08.36 | [TK]D-Fender | *b00m* |
| 01:11.17 | telnettech | TK: do you understand where im coming from? |
| 01:14.04 | [TK]D-Fender | telnettech: I could never understand Virginians... |
| 01:14.15 | [TK]D-Fender | Crazy-folk, I say... |
| 01:14.51 | telnettech | Im not from Virginia...sorry...born and raised in Ohio |
| 01:15.01 | [TK]D-Fender | telnettech: And during the netsplit if you said anything useful, I missed it |
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| 01:15.17 | [TK]D-Fender | telnettech: Just what your IP says... anyways, that was in jest |
| 01:15.31 | [TK]D-Fender | telnettech: So what brilliant new insight are you bringing to the table now? |
| 01:15.35 | telnettech | but please be a little patient with me...I am trying to learn alot as I go |
| 01:15.51 | telnettech | and I will get better and ask the right questions the first time in the near future |
| 01:18.32 | justmehere | ;-) don't put too much text down in the room, he may not get it because of a *cough* netsplit *cough* again |
| 01:19.15 | telnettech | I was just saying to be patient casue not everyone is as intelligent as he is with asterisk |
| 01:19.34 | telnettech | i have less than 6 months on this system. |
| 01:20.01 | justmehere | yeah, I don't think anyone is as smart as he is |
| 01:20.01 | telnettech | I am trying to learn Linux, Asteisk, networking, and about all these different devices it takes to make this work |
| 01:20.19 | justmehere | yeah, it's a wild forest at the beginning, |
| 01:20.24 | justmehere | mostly all of the jargon |
| 01:20.43 | telnettech | I have gone to the Digium training and I learn quite alot about the software but it doesnt teach you evry option the software can do |
| 01:21.08 | [TK]D-Fender | telnettech: Knowing that your equipement is suspect has nothing to do with Asterisk experience. I've gone into #ubuntu with plenty of "newb" questions, but I went in with DETAILS. I'm using app X, version Y from Ubuntu 8.10. I did congi parms 1,2,3 and I get error message 456. can anyone hint where to go? |
| 01:21.44 | justmehere | yeah, that is true, details are important, I'll certainly give you that |
| 01:21.45 | jaytee | precise questions yield better answers |
| 01:21.46 | [TK]D-Fender | telnettech: Making getting the model of gateway out of you like getting blood from s stone shows that the light may be on, and the wheel still spinning, but the hamster is DEAD |
| 01:22.04 | telnettech | if I new what error 456 was I would tell you. but the rudeness is not necessary...i dont know how this all works together |
| 01:22.10 | justmehere | it can be frustrating having someone say "well it isn't working...can someone help me?" |
| 01:22.11 | jaytee | ok, ok, let's tone it down a bit. |
| 01:22.45 | telnettech | that is why i came into this chat cause jaytee told me in training that I would get alot out of it |
| 01:23.04 | justmehere | yeah, it's been very useful to me over the years |
| 01:23.10 | [TK]D-Fender | justmehere: "it". My favourite non-description pronoun. "it" doesn't work, "why?!?" |
| 01:23.11 | telnettech | I have telecom experience not computer and networking...im trying to find my way thru this |
| 01:23.12 | jaytee | you will, just have your information laid out in advance with particulars. |
| 01:23.30 | [TK]D-Fender | telnettech: Well did you come up with anything new to debug your situation? |
| 01:24.13 | telnettech | it looks like i need to look at the phones and the netowrk |
| 01:24.25 | telnettech | since asterisk doesnt have a delay option anywhere |
| 01:24.43 | justmehere | I would also, if your phones and devices have the option available to setup a syslog server to pipe debug output to |
| 01:24.54 | telnettech | ok thanks just |
| 01:25.11 | jaytee | do a test call from an internal phone WHILE watching the CLI. if the CLI doesn't output call info during the delay then it's most likely in the phone's dialplan. |
| 01:25.33 | jaytee | what kind of phones? |
| 01:25.35 | telnettech | ok thanks jaytee |
| 01:25.48 | telnettech | i have grandstream gxp2000 |
| 01:26.05 | [TK]D-Fender | telnettech: Look in CLI for the call coming from your phone. Any perceptible delay thre? Dialplan likely isn't adding any. Then look at the call out to your gateway. How long is the number? How long would it take YOU to dial it with an analog phone? Any real difference? now add up the cumulative delays between Phone > *, and * > Gateway |
| 01:26.33 | [TK]D-Fender | telnettech: Your phone can be adding delay between when you are finished dialing and it DECIDES that its finished taking input from you |
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| 01:28.02 | telnettech | TK: Im aware that the phone is doing that. I can accept that but from the time that I see the numbers on the phone change to signify that it called the pbx and then the connection, it seemd to be too long...like 1 to 2 seconds...it is noticeable to the customer |
| 01:28.29 | [TK]D-Fender | telnettech: Yes, well did you go and TIME it from * CLI? |
| 01:28.57 | telnettech | no i havent yet....but i will |
| 01:29.16 | [TK]D-Fender | telnettech: Todays magic word is "thorough" :) |
| 01:29.38 | justmehere | haha |
| 01:29.39 | [TK]D-Fender | telnettech: Go look and tell us... no, SHOW US what you see... |
| 01:30.17 | [TK]D-Fender | telnettech: Wouldn't want to excessively limit the interpretation of whats happening there by restricting to a single pair of eyes... |
| 01:30.17 | justmehere | [TK]D-Fender: Today's magic acronym is RTFM |
| 01:30.24 | justmehere | :-) |
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| 01:30.47 | [TK]D-Fender | justNo, I wouldn't through a "RTFM" out there jsut like that... there isn't a manual for "What do you mean you didn't even look?" :) |
| 01:30.59 | justmehere | haha |
| 01:31.55 | [TK]D-Fender | * CLI <- its what's "going on" |
| 01:32.20 | [TK]D-Fender | If you're wondering whats happening and not looking intently there you aren't even trying |
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| 01:32.54 | jaytee | NoOp is your friend! |
| 01:33.16 | [TK]D-Fender | jaytee: ANY ouptup would be "something". I'm not even getting app-picky yet! |
| 01:33.29 | jaytee | lol |
| 01:34.42 | jaytee | doing this frigging IVR rewrite is like taking a refresher course in Spanish I |
| 01:34.58 | justmehere | telnettech: on your phone, what is the No Key Entry Timeout setting? |
| 01:35.17 | telnettech | i will check |
| 01:35.25 | justmehere | jaytee: LOL, Si, Dende este el queso? |
| 01:35.28 | [TK]D-Fender | justmehere: Lets wait until he reporst back on the FIRST leg of the call before wasting time asking for specifics |
| 01:35.42 | telnettech | in the meantime....here is a pastebin with extensions to extension and an outbound call |
| 01:35.52 | [TK]D-Fender | justmehere: Or you'll be hand feeding him 1000 things he DOESN'T need to look at. |
| 01:36.08 | [TK]D-Fender | justmehere: See if the phone is any noticable part. Then pick apart why |
| 01:36.29 | telnettech | http://pastebin.com/d463c7fb5 |
| 01:36.31 | telnettech | sorry |
| 01:36.33 | jaytee | justmehere, the cheese is over there >>>>> but it's moldy |
| 01:36.41 | justmehere | [TK]D-Fender: Just asked because it seems that is a common complaint with that particular Grandstream model |
| 01:36.52 | justmehere | jaytee: LOL Classic |
| 01:37.22 | [TK]D-Fender | justmehere: But he hasn't confirmed anything to make the phone suspect yet. |
| 01:37.33 | [TK]D-Fender | justmehere: Get a little, give a little |
| 01:37.47 | [TK]D-Fender | justmehere: Otherwise you become a 100% surrogate brain. |
| 01:37.51 | telnettech | just...it is 4 seconds |
| 01:37.55 | justmehere | naw, I get paid too much, I give a lot and take a little ;-) |
| 01:38.00 | telnettech | default |
| 01:38.16 | [TK]D-Fender | telnettech: Go time the call between the phone & * |
| 01:38.27 | [TK]D-Fender | telnettech: End of last digit to start of activity |
| 01:38.41 | jaytee | IIRC, the default digit timeout on the Grandsuck 2000 is 4 seconds |
| 01:38.42 | telnettech | it is about 4 seconds |
| 01:38.45 | justmehere | telnettech: yeah, that is the common complaint, because what is going to happen is the phone is waiting 4 seconds from the last number pressed to send the call to * |
| 01:38.54 | jaytee | which you can change via the web interface |
| 01:38.59 | justmehere | drop it to 2 |
| 01:39.00 | [TK]D-Fender | telnettech: Then yoru phone dialplan needs to be tweaked |
| 01:39.12 | justmehere | maybe 3 |
| 01:39.22 | [TK]D-Fender | telnettech: depending on the patterns you allow it could be brought to pretty much 0 |
| 01:39.29 | [TK]D-Fender | (not a specific parm) |
| 01:39.30 | jaytee | if they're complaining I'd do 2 but never 1 |
| 01:39.37 | justmehere | yeah, Jaytee is right, more accurate dialplan would be best |
| 01:39.56 | [TK]D-Fender | jaytee: strict dialplans can drop that to 0 for 100% matches |
| 01:40.22 | telnettech | my dial plan has a few patern matches i need to make so that i dont have to have a bunch of different extensions....they have about 300 extensions |
| 01:40.36 | [TK]D-Fender | telnettech: its PATTERNS that count |
| 01:40.53 | justmehere | well, he could set it up to ignore the phone's dialplan and send the punches right to asterisk? Would get rid of all phone delay |
| 01:41.09 | telnettech | when you are talking....how does include statements affect the delay |
| 01:41.20 | jaytee | not noticible |
| 01:41.30 | [TK]D-Fender | telnettech: not ASTERISK's DIALPLAN <- |
| 01:41.36 | *** part/#asterisk justmehere (n=justmehe@24-176-157-165.dhcp.kgpt.tn.charter.com) |
| 01:41.38 | [TK]D-Fender | telnettech: and the answer is ZERO <- |
| 01:41.51 | telnettech | what diaplan are you talking about then? |
| 01:42.01 | [TK]D-Fender | telnettech: Your PHONE has an internal dialplan that tells it if it should wait for more digits bfore passing the END RESULT to * |
| 01:42.26 | telnettech | isnt that the no key entry that just was talking about? |
| 01:42.31 | [TK]D-Fender | telnettech: Your phone has a brain and makes its own decisions (can't believe I jsut said that of a GS device... *shudder*) |
| 01:42.50 | [TK]D-Fender | telnettech: Partially, probably that value. I don't know GS's exact naming. |
| 01:43.16 | [TK]D-Fender | telnettech: You should already have the admin guide open and be reading up on this as we speak.... |
| 01:44.45 | telnettech | im looking |
| 01:46.33 | [TK]D-Fender | telnettech: http://forum.voxilla.com/grandstream-support-forum/grandstream-gxp-2000-dial-plan-15174.html |
| 01:46.44 | telnettech | i have it thanks |
| 01:47.00 | [TK]D-Fender | telnettech: This forum post seems to confirm these devices don't even HAVE a dialplan, just that singular timeout. |
| 01:47.13 | [TK]D-Fender | telnettech: Which means, that is the only parm to tweak.... |
| 01:47.20 | [TK]D-Fender | which only goes to reinforce... |
| 01:47.22 | [TK]D-Fender | ~gs |
| 01:47.22 | jbot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
| 01:47.24 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^ |
| 01:47.45 | [TK]D-Fender | No f-ing RFC dialplan support! RETARDED! |
| 01:48.00 | [TK]D-Fender | kicks Grandstream in the nads.... |
| 01:48.06 | telnettech | yeah that is what im seeing also...i dont see anything that stands out about any dialplan |
| 01:48.13 | [TK]D-Fender | remembers the Grandstream HAS no balls.... |
| 01:49.28 | *** join/#asterisk Avelino (n=Avelino@189-46-231-195.dsl.telesp.net.br) |
| 01:53.46 | telnettech | ok the customer is happier with the quicker response from the phone....i guess i need to work on the other end......i am going to go to my room for the night and do some reading it looks like |
| 01:53.57 | telnettech | thanks for all your help again tonight |
| 01:54.20 | telnettech | jaytee: by the way, Im still in Aruba |
| 01:54.26 | jaytee | lucky bastard |
| 01:54.45 | telnettech | and it is carnival time |
| 01:54.53 | jaytee | it's 28F here and looking to be a pissa day tomorrow too |
| 01:54.58 | telnettech | woohoo |
| 01:55.06 | telnettech | guys thanks again |
| 01:55.10 | telnettech | good nite |
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| 01:58.25 | jaytee | took forever to bring up a grandsuck screen over my vpn, it's called No Key Entry Timeout on the Advanced Settings tab. |
| 01:59.01 | drmessano | 24!!!! |
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| 02:20.41 | Olobola | is this possible: externnotify=/test/convertWAV2MP3.php ? |
| 02:20.53 | Olobola | under general in voicemail.conf |
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| 03:09.46 | dandate2 | I am setting up an asterisk vPBX machine for the first time. If my employees do not have an internet softphone, how do I route the call to their hard phone? |
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| 03:14.33 | drmessano | Do they have a SIP hardphone? |
| 03:14.40 | dandate2 | no |
| 03:14.42 | dandate2 | not yet |
| 03:14.50 | dandate2 | but my SIP trunks are going up tommarow or tuesday |
| 03:14.53 | drmessano | What kind of phone do they have? |
| 03:15.06 | dandate2 | one has a hard phone, the others have skype phones, one uses a cell phone |
| 03:15.23 | drmessano | Skype wont work here |
| 03:15.28 | dandate2 | evenutally i'm going to get them all internet softphones, but until that transition is made isn't there a way to forward to a hard phone? |
| 03:15.35 | drmessano | The cell phone, I guess you could forward all those calls |
| 03:15.42 | NovceGuru | "hard phone" |
| 03:15.46 | drmessano | But you'll pay 3x for each call |
| 03:15.47 | drmessano | Yeah |
| 03:15.50 | drmessano | And whats the HARD PHONE? |
| 03:15.50 | NovceGuru | drmessano: what did you end up doing with your fax thingy? |
| 03:15.59 | dandate2 | a hard phone is like something in your hall way right |
| 03:16.05 | dandate2 | hooks up to the regular phone jack |
| 03:16.20 | dandate2 | what we've all been using for the past 100 years |
| 03:16.24 | drmessano | NovceGuru: Didnt go with a software solution. Nothing elegant I could put in place |
| 03:16.41 | drmessano | dandate2: You need an ATA or FXS card to connect it to the PBX |
| 03:16.49 | NovceGuru | dandate2: so you need sort of way to interface with that old fashioned hardphone, have you considered a SIP hardphone? |
| 03:16.55 | drmessano | Youre ordering SIP trunks and have no idea how to connect the phones? |
| 03:16.58 | NovceGuru | drmessano: yeah only thing I found was ugly h4x |
| 03:17.12 | *** join/#asterisk zerko (n=anthony@cpe-24-27-121-165.tx.res.rr.com) |
| 03:17.13 | drmessano | Is the asterisk even set up? |
| 03:17.19 | dandate2 | well i know that once they get linksys PAPII-na they can hook up a softphone and use that |
| 03:17.21 | drmessano | Do you know how to set it up? |
| 03:17.26 | drmessano | No |
| 03:17.29 | zerko | Ok, I just lost all of my config files and need to rebuild asterisk |
| 03:17.30 | dandate2 | its my first time i have a few guides |
| 03:17.32 | drmessano | You dont hook a softphone to a PAP2 |
| 03:17.39 | drmessano | Its an ATA |
| 03:17.44 | zerko | I forgot everything about configuring asterisk |
| 03:17.45 | NovceGuru | physically lol :( |
| 03:17.49 | drmessano | A softphone connects to the PBX |
| 03:17.52 | dandate2 | oh ok i understand now |
| 03:17.59 | NovceGuru | zerko: better get reading then! |
| 03:18.00 | zerko | Does the SIP providers information going into the sip.conf file? |
| 03:18.10 | *** join/#asterisk florz (i=nobody@2001:1a50:503c:0:0:0:0:1) |
| 03:18.13 | zerko | im pissed, heh. |
| 03:18.20 | dandate2 | well i haven't got that far yet i'm still in the process of the credit check and paying for the SIP trunks |
| 03:18.22 | drmessano | dandate2: You need to do some reading.. LOTS of it |
| 03:18.31 | drmessano | and fast |
| 03:18.32 | NovceGuru | sounds like you had asterisk running on an (unanswered) prayer |
| 03:18.39 | zerko | are there any conf generators online that would generate my configs for me if i put in the information? |
| 03:18.48 | NovceGuru | dandate2: how many sip trunks? |
| 03:18.49 | drmessano | No |
| 03:19.00 | drmessano | NovceGuru: Virtual PRI, I am guessing |
| 03:19.03 | dandate2 | i'm getting 5 lines with infinite inbound and outbound from bandwidth.com |
| 03:19.18 | dandate2 | that way I can have 5 sales reps on, or 2 sales reps with 3 in queue |
| 03:19.20 | NovceGuru | He needs to google some cluepons |
| 03:19.26 | drmessano | ROFL |
| 03:19.28 | drmessano | Indeed |
| 03:19.34 | zerko | I have a DID with a company |
| 03:19.37 | NovceGuru | you paying $30/month for the bandwidth.com lines? |
| 03:19.42 | drmessano | .... |
| 03:19.43 | dandate2 | yes is that a rip off? |
| 03:19.45 | zerko | I remember having to input my login/host/password etc |
| 03:19.50 | zerko | Which file does that usually go into ? |
| 03:20.04 | NovceGuru | eh, it's kinda high, they tout the "unlimited" bit |
| 03:20.14 | zerko | i def dont want to install trixbox o_O |
| 03:20.19 | NovceGuru | but google uses them, so that has to go for something |
| 03:20.39 | dandate2 | so my employees can get linksys ATA-na and connect that to my vpbx, that will then allow them to use their regular phones |
| 03:20.50 | NovceGuru | what is your budget for this project? |
| 03:20.56 | drmessano | $14.76 |
| 03:21.01 | zerko | heh |
| 03:21.04 | dandate2 | what is a short term solution that I could use to forward the calls to them until they get those routers? |
| 03:21.04 | drmessano | Give or take $2 |
| 03:21.13 | NovceGuru | drmessano: no shit! |
| 03:21.15 | drmessano | dandate2: Pretty much nothing |
| 03:21.23 | drmessano | Skype is out of the picture |
| 03:21.28 | NovceGuru | what do you want to do? |
| 03:21.31 | drmessano | The cell phone user.. youre gonna pay for every call |
| 03:21.34 | NovceGuru | foward the sip trunks to them somehow? |
| 03:21.49 | drmessano | Call in >> Forwared out (2 channels), then the incoming to the cell phone |
| 03:21.52 | NovceGuru | do the sip trunks have numbers on them you need asap or someting? |
| 03:22.00 | drmessano | AT&T is gonna buy a school girl outfit for xmas |
| 03:22.28 | dandate2 | well its because I run an inbound telemarketing center, all my employees work from home so their standards fvary |
| 03:22.42 | drmessano | Did you do ANY research on this like we told you? |
| 03:22.52 | drmessano | Instead youre ordering trunks, have NO phone plans |
| 03:22.58 | drmessano | and the Skype? Useless |
| 03:23.02 | dandate2 | well yes I read through the guides for configuring freepbx and what not |
| 03:23.03 | drmessano | You cant route the calls to skype |
| 03:23.05 | NovceGuru | get them $10 headsets from walmart and x-lite |
| 03:23.15 | NovceGuru | freepbx!! |
| 03:23.26 | NovceGuru | OH! |
| 03:23.40 | NovceGuru | doesn't really apply since we havent had any specific questions for either, haha |
| 03:23.52 | NovceGuru | this is like phone system: step 1 |
| 03:24.07 | drmessano | 1. Order a Virtual PRI |
| 03:24.11 | drmessano | 2. Install Trixbox CD |
| 03:24.14 | drmessano | 3. ?????? |
| 03:24.17 | drmessano | 4. Callcenter |
| 03:24.19 | NovceGuru | 4. PROFIT |
| 03:25.23 | dandate2 | doesn't freebpx and pbxinaflash come with PRI? |
| 03:25.26 | drmessano | I get the impression this is no more serious than one of the "Businesses" i had when I was a kid.. Like that one time all my friends and I started a detective agency. Then there was the time we were all pilots for hire. |
| 03:25.37 | drmessano | No, they dont come with a PRI |
| 03:25.40 | drmessano | Good god |
| 03:26.00 | NovceGuru | yeah they give them away with the software download |
| 03:26.12 | dandate2 | oh, well we have solid call volume, i just don't want there to be any downtime transitioning my folks |
| 03:26.22 | drmessano | There will be with this plan |
| 03:26.30 | drmessano | It sounds like youre not even half assing it |
| 03:26.31 | dandate2 | heh |
| 03:26.37 | dandate2 | well its my first time hey |
| 03:26.40 | NovceGuru | rofl |
| 03:26.41 | drmessano | No |
| 03:26.45 | drmessano | Thats a bullshit excuse |
| 03:26.55 | drmessano | We told you last week to do some reading |
| 03:27.01 | NovceGuru | Chewing capacity has been exceeded |
| 03:27.06 | drmessano | You just asked if FreePBX came with a PRI and what to do with the skype phones |
| 03:27.20 | drmessano | ~asterisk101 |
| 03:27.32 | drmessano | ~101 |
| 03:27.33 | jbot | from memory, 101 is Telephony 101, which is a good read if you're unfamiliar with traditional TDM telephony. You can download it at http://www.stromcarlson.com/docs/basics/NTtelephony101.pdf |
| 03:27.35 | dandate2 | well i thought there was some way I could route my SIP trunk to a forwarding service that will get the calls to my employees regardless of whether they have ATA adapters |
| 03:28.02 | NovceGuru | if they have standard phones, you can use another channel and foward back out to their landlines |
| 03:28.09 | dandate2 | so that way they could still use their skype until they transitioned into connecting with the main server |
| 03:28.15 | drmessano | So you have basically 2 1/2 phone lines now |
| 03:28.28 | drmessano | If they're single channel |
| 03:28.34 | drmessano | One in, one out, per call |
| 03:28.35 | NovceGuru | they are |
| 03:28.35 | dandate2 | whew thats a tough one |
| 03:28.38 | NovceGuru | fucking hate bandwidth.com |
| 03:28.42 | *** join/#asterisk s0lid (n=s0lid@60.53.250.155) |
| 03:28.49 | dandate2 | what do you recommenct guru? |
| 03:28.57 | drmessano | Learning Asterisk |
| 03:29.04 | drmessano | Oh, you meant him |
| 03:29.15 | drmessano | ~101 |
| 03:29.16 | jbot | [101] Telephony 101, which is a good read if you're unfamiliar with traditional TDM telephony. You can download it at http://www.stromcarlson.com/docs/basics/NTtelephony101.pdf |
| 03:29.16 | dandate2 | so if I have 5 phone lines I can use those channels for outbound and inbound calling |
| 03:29.21 | drmessano | ^^^^^^^^^^^^^^^^^^ READ |
| 03:29.24 | NovceGuru | I'm still very mich the novice |
| 03:29.31 | drmessano | dandate2: ONE DIRECTION AT A TIME |
| 03:29.37 | NovceGuru | much* |
| 03:29.37 | dandate2 | so if someone calls in and is forwarded to my employee, that would be tieing up 2 phone lines, but its doable? |
| 03:29.39 | drmessano | So ONE IN and ONE OUT per call |
| 03:29.46 | drmessano | So you dont have 5 lines now |
| 03:29.48 | drmessano | You have 2.5 |
| 03:30.00 | drmessano | Sure it is |
| 03:30.01 | drmessano | Go for it |
| 03:30.02 | dandate2 | i see, so I could do that as a temporary solution then |
| 03:30.10 | drmessano | Yep, indeed |
| 03:30.23 | NovceGuru | dandate2: http://office.microsoft.com/en-us/communicationsserver/default.aspx |
| 03:30.25 | dandate2 | now can anyone recommend a cheaper service than bandwidth.com that mabye won't put me through all the credit checks? |
| 03:30.37 | drmessano | dandate2: Skype.com |
| 03:30.49 | NovceGuru | ~itsp-us |
| 03:30.53 | drmessano | You'll actually be able to manage Skype |
| 03:30.56 | NovceGuru | w00h! |
| 03:30.59 | drmessano | I would recommend sticking with it |
| 03:31.20 | zerko | No matching peer found |
| 03:31.23 | zerko | What does this mean? |
| 03:31.37 | drmessano | That a peer that would match has not been found |
| 03:32.58 | dandate2 | really i don't appreciate the quality of skype tho |
| 03:33.08 | dandate2 | would by vpbx be better? |
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| 03:33.33 | dandate2 | so this microsoft office communicator is a Vpbx for windows? |
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| 03:35.24 | drmessano | Oh yes |
| 03:35.43 | drmessano | What is a VPBX by the way? |
| 03:35.46 | drmessano | I am not familiar |
| 03:35.48 | zerko | Ok |
| 03:36.48 | dandate2 | virtual pbx? |
| 03:37.02 | *** join/#asterisk kerx_ (n=pre_prog@38.118.193.10) |
| 03:37.41 | kerx_ | hi, i've programmed my DID through my provider, and I've set it to route to my Host IP:Port, but my Asterisk box is sending back a 407 Proxy Auth Required. Anyone know why this could be happening? |
| 03:38.26 | drmessano | What is virtual about it? |
| 03:38.40 | drmessano | Its a PBX.. That youre setting up physically |
| 03:38.42 | NovceGuru | dandate2: if you're on a time crunch and don't want to read, do the bandwidth.com phonebooth |
| 03:38.47 | drmessano | NOT virtual |
| 03:38.52 | NovceGuru | that would be my definition of vPBX |
| 03:39.01 | drmessano | Thats exactly what that is |
| 03:39.09 | NovceGuru | I consider that hosted pbx though |
| 03:40.16 | NovceGuru | your central point of failure is your probably not so glamerous internet connection so hosted wouldn't be a disadvantage |
| 03:40.28 | NovceGuru | say that 10x fast! |
| 03:40.34 | kerx_ | Anyone if can help me I would be appreciative |
| 03:40.36 | kerx_ | Thanks |
| 03:41.47 | *** join/#asterisk CrazyTux (n=brandon@ip68-111-67-4.oc.oc.cox.net) |
| 03:41.52 | NovceGuru | kerx_: http://www.google.com/search?hl=en&safe=off&client=firefox-a&rls=org.mozilla:en-US:official&hs=bGt&pwst=1&sa=X&oi=spell&resnum=0&ct=result&cd=1&q=auth+proxy+required+asterisk&spell=1 you try any of those? |
| 03:43.06 | dandate2 | i don't mind reading guru, but isn't phonebooth just a software my employees can use to handle the calls? |
| 03:43.28 | kerx_ | I see |
| 03:43.29 | kerx_ | Thanks |
| 03:43.29 | NovceGuru | no, it's a hosted PBX solution |
| 03:43.53 | *** join/#asterisk sah-work (n=Bawbatos@65-119-47-100.dia.static.qwest.net) |
| 03:45.38 | dandate2 | oh no phonebooth only comes with box set bundles, the SIP trunking is different |
| 03:46.34 | NovceGuru | still a hosted PBX solution |
| 03:46.43 | NovceGuru | and I'm sure SIP is somehow invloved |
| 03:46.59 | dandate2 | i talked to someone for a while about it he said all i would need is pbx in a flash, freepbx, and some SIP trunks from bandwidth.com |
| 03:47.13 | NovceGuru | then get to work |
| 03:47.16 | NovceGuru | #freepbx |
| 03:47.35 | NovceGuru | I was just offering a less headache solution where all the work was done for you |
| 03:47.47 | NovceGuru | since your deadline seems to be tomorrow |
| 03:47.50 | NovceGuru | or tuesday |
| 03:48.16 | dandate2 | oh i see, i was confused by their website cuz it says phonebooth comes with boxsets, but when I click on order the box-sets seem to be like T1 lines and such |
| 03:48.33 | NovceGuru | you can get it without an internet service |
| 03:48.41 | NovceGuru | the "unplugged" set |
| 03:48.58 | dandate2 | k |
| 03:49.14 | NovceGuru | what kind of internet connection do you have where this would be hosted? |
| 03:50.04 | dandate2 | comcast cable modem |
| 03:50.11 | dandate2 | decent enough right? |
| 03:50.27 | NovceGuru | will you be doing any sort of QoS? |
| 03:50.27 | dandate2 | will phonebooth allow me to sustain people in queue? |
| 03:50.44 | drmessano | Users are remote, NovceGuru |
| 03:50.55 | dandate2 | right i would like to beable to monitor my employees |
| 03:51.03 | dandate2 | or use that to train new ones |
| 03:51.22 | drmessano | How is the location of the PBX gonna change that? |
| 03:51.24 | drmessano | Its not |
| 03:51.28 | NovceGuru | drmessano: yeah, not sur what you mean |
| 03:51.46 | drmessano | NovceGuru: His Queued users are offsite.. teleworkers |
| 03:52.07 | NovceGuru | Thats what i'm saying, remote workers using a pbx hosted on a comkrap connection |
| 03:52.11 | drmessano | So the bandwidth will only apply to the onsite usage, which I assume will be minimal |
| 03:52.45 | drmessano | He should have no problem with 10 channels of audio if hes got a decent plan |
| 03:52.46 | NovceGuru | he probably has a torrent account he needs to keep his ratio sustained at |
| 03:52.47 | drmessano | 8/2 or so |
| 03:53.07 | SlicerDicer | transfer call in cli, 'console transfer' correct? |
| 03:53.09 | NovceGuru | nah wouldn't be a problem, minus uptime |
| 03:53.09 | drmessano | I do think the Hosted PBX makes more sense |
| 03:53.24 | SlicerDicer | cause I have no 'console' command in my cli? wtf? |
| 03:54.53 | NovceGuru | help console |
| 03:55.14 | dandate2 | the cable modem would be dedicated exclusively to the pbx machine |
| 03:55.18 | NovceGuru | dandate2: as far as your queues, I have no idea, I've never used phonebooth |
| 03:55.19 | dandate2 | i use a DSL on my personal computer |
| 03:55.47 | NovceGuru | I'd probably deffintely go hosted them, if you're paying for a dedicated internet connection |
| 03:55.50 | dandate2 | well what happens is if i have 2 sales reps on and 5 people are calling in, 3 of those people need to sit on hold listening to music until its their turn |
| 03:56.00 | NovceGuru | or get a box in a datacenter, but let's not go down that path |
| 03:56.19 | NovceGuru | you'll have to call and ask |
| 03:56.53 | dandate2 | k |
| 03:57.03 | NovceGuru | there are other hosted providers out there |
| 03:57.04 | dandate2 | well tommarows a big day, i find out if i have to pay a deposit for shitty credit heh |
| 03:57.21 | NovceGuru | why can't they just do month to month credit card? |
| 03:57.58 | dandate2 | i looked into em, like voicenation.com wants basically $100 a month for 2500 minutes at a rate of .2 cents per minute after. but my issue is that I am receiving 200-300 phone calls a day and will be using massive minutes with people sitting on hold |
| 03:58.28 | NovceGuru | get a plan with unlimited inbound but read the fine print |
| 03:58.40 | NovceGuru | not sure how you're planning 200-300 calls on 5 lines |
| 03:58.48 | NovceGuru | may be possible, never dealt with that capacity |
| 03:58.55 | dandate2 | i looked into em, they all said if i use their lines for telemarketing i'll get deeply fined |
| 03:59.12 | NovceGuru | well, don't call me ;) |
| 03:59.14 | dandate2 | well right now i just have 2 comcast residential lines that i advertise |
| 03:59.20 | dandate2 | no we are inbound call center |
| 03:59.22 | dandate2 | everyone calls us |
| 03:59.48 | drmessano | Are you gonna use comcast residential for the PBX? |
| 03:59.58 | NovceGuru | I think he sells it |
| 04:00.01 | NovceGuru | I don't know |
| 04:00.02 | drmessano | Are you gonna use comcast residential internet for the PBX? |
| 04:00.03 | NovceGuru | i'm so confused |
| 04:00.06 | dandate2 | cuz we hold for you an enormous list of pre-foreclosure properties ready for you now, willing to be taken over. Credit isn't an issue, call today 415-682-4237 |
| 04:00.09 | drmessano | Are you gonna use comcast residential internet for the PBX? |
| 04:00.20 | dandate2 | yes messano |
| 04:00.24 | NovceGuru | 23:00 < drmessano> Are you gonna use comcast residential internet for the PBX? |
| 04:00.24 | drmessano | What is wrong with you? |
| 04:00.28 | NovceGuru | FAIL |
| 04:00.28 | drmessano | Thats a HORRIBLE idea |
| 04:00.33 | dandate2 | why is that?? |
| 04:00.50 | drmessano | Because they give you 1/2 to 1/4 the UPSTREAM bandwidth |
| 04:01.02 | drmessano | and 3 days response time vs FOUR HOURS |
| 04:01.07 | NovceGuru | "my internets down!" comcast: we don't care! |
| 04:01.18 | drmessano | Business class Comcast is the ONLY way to go here |
| 04:01.22 | NovceGuru | you get that kind of SLA with their business class? |
| 04:01.22 | drmessano | Dont even CONSIDER residential |
| 04:01.30 | drmessano | You will be FUCKED |
| 04:01.37 | drmessano | Yes |
| 04:01.39 | drmessano | 4 hours |
| 04:01.43 | NovceGuru | we get 4hr SLA on our $600/month 5x5mbit fiber |
| 04:01.44 | dandate2 | oh but it costs so much more! |
| 04:01.47 | NovceGuru | lame! |
| 04:01.50 | *** join/#asterisk CrazyTux (n=brandon@ip68-111-67-4.oc.oc.cox.net) |
| 04:01.55 | NovceGuru | of course we are in BFE |
| 04:02.01 | drmessano | Yes it costs MORE |
| 04:02.09 | drmessano | and you get what you pay for |
| 04:02.17 | drmessano | Its like $59 for 6/2 service, WTF |
| 04:02.22 | drmessano | Err 6/1 |
| 04:02.27 | drmessano | I get about 2 tho on the up |
| 04:02.34 | NovceGuru | dandate2: for the love of god use a hosted service or get some better data services |
| 04:02.50 | NovceGuru | if youre serious about helping people get into foreclosed housing(tm) |
| 04:02.51 | drmessano | ~cluebat |
| 04:02.52 | jbot | *WHACK* *WHACK* *WHACK* |
| 04:02.56 | drmessano | WTF |
| 04:02.58 | NovceGuru | ~cluepon |
| 04:02.58 | jbot | cluepon is, like, a coupon for a clue. Get them while they're hot. |
| 04:03.08 | drmessano | Comcast residential? |
| 04:03.11 | drmessano | Come on |
| 04:03.12 | NovceGuru | ah hell, somewhere used to paste an awesome ASCII cluepon |
| 04:03.36 | drmessano | I have Comcast business at my HOUSE |
| 04:03.36 | dandate2 | lol |
| 04:03.46 | drmessano | and youre gonna run a business of comcast residential |
| 04:04.12 | dandate2 | well i never really have downtime, do u think the quality of sound would just be shitty? |
| 04:04.21 | NovceGuru | you get 200-300 calls a day? do you get 1% of the closing of a deal? |
| 04:04.29 | dandate2 | i'm in the city of san francisco so if anything goes wrong with comcast their lines will blow up with 7 million callers |
| 04:04.34 | drmessano | The bandwidth is gonna be a problem, and so is the SLA |
| 04:04.36 | drmessano | If you go down |
| 04:04.42 | drmessano | 3 DAYS till you see a tech |
| 04:04.51 | NovceGuru | atleast |
| 04:05.00 | NovceGuru | thats 600-900 missed calls! |
| 04:05.02 | NovceGuru | ATLEAST! |
| 04:05.12 | dandate2 | well its a $40 deal for one month subscription, i pay my teleworkers $18-20, then it rebills for next months subscription, i give my teleworker $14 for that |
| 04:05.29 | jaytee | I've got residential service with Comcast and I had to wait a week for service |
| 04:05.36 | drmessano | Exactly |
| 04:05.48 | drmessano | Its not that expensive for business |
| 04:05.52 | NovceGuru | dsl failover! |
| 04:06.03 | drmessano | Satellite failover |
| 04:06.03 | dandate2 | well i do have 2 internet connections here |
| 04:06.12 | NovceGuru | head --> desk |
| 04:06.15 | drmessano | I bet hes got a DirecTV dish he can use with hughesnet |
| 04:06.30 | dandate2 | but if anything, if my vpbx goes down i'll just have my employees take calls from the comcast residential lines again, i would just forward it to their skypes and land lines |
| 04:06.31 | NovceGuru | increase your MTU with the flux capacitor inductor |
| 04:06.45 | drmessano | danalien |
| 04:06.48 | drmessano | dandate2 |
| 04:06.54 | drmessano | If your Comcast goes down |
| 04:06.57 | drmessano | ITS DOWN |
| 04:07.01 | drmessano | You wont have phone lines |
| 04:07.09 | drmessano | The Comcast voice will be down too |
| 04:07.12 | dandate2 | oh you mean because i got shut down? |
| 04:07.16 | drmessano | and since youre a residential customer |
| 04:07.18 | drmessano | God damnit |
| 04:07.30 | NovceGuru | im sure they're cool with telemarketing on a residential line also LOL |
| 04:07.33 | drmessano | When did I say anything about you being shut down? |
| 04:07.44 | dandate2 | well i dno if its a breach of terms and conditions |
| 04:07.44 | drmessano | IF A FUCKING TREE FALLS ON THE NODE DOWN THE STREET |
| 04:07.46 | dandate2 | with comcast |
| 04:07.46 | NovceGuru | well, you made a good point :P |
| 04:07.50 | drmessano | and your shit goes DOWN |
| 04:07.53 | drmessano | You are DOWN |
| 04:08.02 | drmessano | For DAYS |
| 04:08.04 | dandate2 | ohh i undestand |
| 04:08.07 | drmessano | and no guaranteee |
| 04:08.08 | NovceGuru | or when the construction workers |
| 04:08.16 | NovceGuru | cut through 1000's lines of copper |
| 04:08.16 | drmessano | Because its FUCKING RESIDENTIAL |
| 04:08.26 | drmessano | Are you clueless? |
| 04:08.33 | NovceGuru | and your t1 is down for a week, and you get a free month of service! |
| 04:08.44 | drmessano | Seriously.. if you are mentally handicapped I promise I will stop.. |
| 04:08.51 | drmessano | I dont go there |
| 04:08.53 | NovceGuru | that $800 < $ you make selling foreclosed houses |
| 04:09.21 | *** join/#asterisk propellerhead (n=yogurt2u@host195.190-30-211.telecom.net.ar) |
| 04:09.26 | zerko | Who owns a T1? |
| 04:09.29 | dandate2 | alright i am understanding your wisdom drmess |
| 04:09.43 | [TK]D-Fender | zerko: Noone. |
| 04:09.48 | zerko | I do :) |
| 04:09.49 | dandate2 | i'm personally a hands on learner so i usually don't get it until i've failed every possible way heh |
| 04:10.08 | drmessano | Youre doing well here then |
| 04:10.13 | [TK]D-Fender | zerko: T1 is a line signalling protocol. How do you own a protocol that is open? |
| 04:10.15 | drmessano | Because ive seen trixbox users with more sense |
| 04:10.24 | drmessano | Actually, most of them |
| 04:10.24 | NovceGuru | http://fun.irq.dk/cluepon.jpg |
| 04:10.35 | zerko | actually, i own multiple GiGe lines |
| 04:10.45 | NovceGuru | drmessano: well, they usually get a little further then he has, then give up and turn to trixbox |
| 04:10.48 | zerko | Anyone here from Dallas? |
| 04:10.54 | zerko | or know about the Dallas Infomart? |
| 04:11.11 | NovceGuru | zerko: you own multiple gigE lines and can't figure out a proxy auth issue? |
| 04:11.13 | NovceGuru | 0_o |
| 04:11.42 | drmessano | NovceGuru: He hasnt even STARTED on Asterisk yet. He read the name somewhere, found the IRC channel, and based on the questions hes asked, like if a FreePBX comes with PRI, hes done ZERO reading.. Dont give him undue credit |
| 04:11.43 | NovceGuru | Which is fine, if those gigE lines aren't utilized > 0.1% for phones |
| 04:11.51 | drmessano | Hes probably never used Linux before |
| 04:12.05 | NovceGuru | drmessano: :| TRUTH! |
| 04:12.29 | drmessano | dandate2: Have you ever used Linux? |
| 04:12.32 | zerko | heh |
| 04:12.34 | NovceGuru | dandate2: do you, and us all a favor, and either go read for a week solid or go with a hosted solution by Tuesday |
| 04:12.35 | dandate2 | i found a trixbox tutorial on youtube, i'll check it out |
| 04:12.47 | drmessano | dandate2: Have you ever used Linux? |
| 04:12.47 | dandate2 | i used linux 8 years ago to setup a network server so i could hack everquest |
| 04:12.52 | drmessano | ... |
| 04:12.55 | dandate2 | the hardest thing i ever did in my life |
| 04:13.00 | drmessano | You win.. |
| 04:13.01 | zerko | I dont usually configure all of this. |
| 04:13.02 | dandate2 | heh |
| 04:13.05 | drmessano | You seriously, seriously win |
| 04:13.17 | drmessano | Wheres the camera and peter funt? |
| 04:13.18 | NovceGuru | zerko: sorry, I'm just on edge for some reason :) |
| 04:13.29 | jaytee | hardest thing I ever did in my life was invent time-travel but then I got stuck in this backwards century and can't get home |
| 04:13.39 | drmessano | jaytee: Are you Peter Funt? |
| 04:13.48 | zerko | its fine, i just never had time to learn all of this |
| 04:13.51 | jaytee | I thought it was Alan Funt |
| 04:13.53 | NovceGuru | no he's gary busey! |
| 04:13.56 | drmessano | Thats his son |
| 04:13.57 | jaytee | oh, that was his dad |
| 04:13.57 | zerko | so busy with other things, its so time consuming. |
| 04:14.02 | drmessano | Peter is Alan's son |
| 04:14.10 | drmessano | He's the new host |
| 04:14.18 | drmessano | Whenever they crank out one, like every 2 years |
| 04:14.19 | zerko | but yes, i have a suite in the dallas infomart |
| 04:14.23 | jaytee | Alan Funt was on the original Candid Camera with Derwood Kirby |
| 04:14.29 | drmessano | But yeah, wheres the camera |
| 04:14.39 | NovceGuru | quit typing i'm trying to copy! |
| 04:14.45 | drmessano | checks the mailbox and under that _Raptor_ guy |
| 04:15.15 | NovceGuru | secures up everquest server cluster |
| 04:15.48 | dandate2 | yeah using linux allowed you to read the packets from the everquest server to see where the rare monsters were and such |
| 04:15.57 | dandate2 | i was in highschool so that was a lot of fun |
| 04:16.09 | drmessano | Youre not now? |
| 04:16.22 | dandate2 | nah i'm 25 |
| 04:16.36 | drmessano | ...... |
| 04:17.18 | dandate2 | i started my business with nothing but a comcast residential and a website |
| 04:17.18 | drmessano | AND YOU CAN TOO!!! |
| 04:17.18 | dandate2 | haha |
| 04:17.18 | *** join/#asterisk florz (i=nobody@2001:1a50:503c:0:0:0:0:1) |
| 04:17.23 | drmessano | Hi everyone.. two months ago. I was a unemployed painter crack addict living in a shoebox |
| 04:17.31 | drmessano | Now, I make $100,000 A WEEK |
| 04:17.36 | dandate2 | ahh |
| 04:17.49 | dandate2 | nah check it out www.newlineequity.com |
| 04:18.05 | drmessano | Using my system.. Dan Date 2: Electric Boogaloo, you too can be a ZILLIONAIRE |
| 04:18.05 | dandate2 | its amateur, but we sign em up over the phone before they see the website anyway |
| 04:18.41 | drmessano | With Dan Date 2: Electric Boogaloo, all you need to remember is "Look good, pay the Comcast bill" |
| 04:18.47 | drmessano | and THATS ALL |
| 04:18.51 | dandate2 | haha thats it |
| 04:19.02 | dandate2 | and u have to pay indians for outsourced ad posting |
| 04:19.13 | drmessano | wonders if jaytee gets the reference |
| 04:19.17 | NovceGuru | http://www.newlineequity.com/images/main_image.gif |
| 04:19.18 | dandate2 | these guys will work hella hard for $2 an hour |
| 04:19.20 | NovceGuru | is that supposed to look 8 bit? |
| 04:19.32 | dandate2 | yeah its supposed to load fast for our 56k users |
| 04:20.19 | drmessano | Eat your heart out, Amway |
| 04:20.20 | NovceGuru | it's 65kb |
| 04:20.29 | NovceGuru | and looks like total shit! |
| 04:20.33 | drmessano | No no |
| 04:20.34 | NovceGuru | it could be 30kb and look 5x better |
| 04:20.34 | drmessano | Its doesnt |
| 04:20.40 | drmessano | It looks "Residential" |
| 04:20.45 | dandate2 | lol |
| 04:20.45 | jaytee | drmessano, I'm not sure what the reference was about but now I've got "Electric Avenue" stuck in my head! Wanker! |
| 04:20.45 | drmessano | Use the right term |
| 04:20.48 | NovceGuru | hhahaa |
| 04:20.52 | drmessano | Breakin' |
| 04:20.57 | NovceGuru | sigh |
| 04:21.03 | drmessano | Breakin' 2: Electric Boogaloo |
| 04:21.03 | dandate2 | ok u guys are cracking me up |
| 04:21.17 | drmessano | dandate2: We're actually cracking ourselves up |
| 04:21.24 | dandate2 | it doesn't matter tho cuz theres no log in to the website |
| 04:21.29 | drmessano | dandate2: You are absolutely adorable |
| 04:21.48 | dandate2 | the product comes by email, so noone cancels |
| 04:22.08 | dandate2 | no phone based customer support |
| 04:23.06 | dandate2 | that was the mistake of the last business i worked for www.firststepequity.com |
| 04:23.26 | dandate2 | they were also $40/mo but they offered phone customer support, so their cancellation rate was 90% |
| 04:24.45 | dandate2 | and mabye that wouldn't be the case if they weren't paying the service reps $10/hr. i always told them they should pay em for converting cancellation calls |
| 04:24.53 | dandate2 | noone in my business makes anything hourly |
| 04:25.12 | dandate2 | but i do pay 50% commission, if anyones interested in some phone sales heh |
| 04:26.49 | drmessano | Sorry, I dont have Skype |
| 04:27.08 | dandate2 | lol |
| 04:27.15 | dandate2 | well right now all u need is a landline |
| 04:27.23 | drmessano | What about a Magicjack? |
| 04:27.28 | dandate2 | hey why do u guys recommend trixbox over freepbx? |
| 04:27.30 | drmessano | Can I be in the club with one of those? |
| 04:27.37 | dandate2 | not sure what that is |
| 04:27.38 | drmessano | Who the fuck said that |
| 04:27.41 | drmessano | Trixbox sucks |
| 04:27.54 | dandate2 | lol |
| 04:28.20 | drmessano | Trixbox is like Comcast Residential of PBX's |
| 04:28.23 | drmessano | Oh sorry |
| 04:28.34 | [TK]D-Fender | dandate2: Whats better, a car, or wheels? |
| 04:28.38 | jaytee | I'd rather run asteriskwin32 |
| 04:28.49 | drmessano | AsteriskWin64 |
| 04:28.58 | jaytee | is that out? |
| 04:29.08 | drmessano | God, I wish |
| 04:29.12 | verywiseman | i need any technical resources about "channel signalling type" to understand it more , regardless using it in Asterisk |
| 04:29.14 | dandate2 | wheels are more versatile and have a broad variety of uses, but the car is obviously the more robust demonstration of the wheel |
| 04:29.15 | drmessano | I would be all over that crap |
| 04:30.38 | [TK]D-Fender | dandate2: Well you just compared a full car to a PART of a car. |
| 04:32.14 | dandate2 | alright well a cars better than wheels |
| 04:32.27 | dandate2 | unless they mean the same shit heh |
| 04:32.46 | [TK]D-Fender | dandate2: Then by that comparison Trixbox would be better than FreePBX. |
| 04:33.05 | dandate2 | is that your opinion also? |
| 04:33.48 | [TK]D-Fender | dandate2: No, I differentiate between a tool, and a toolbox that happens to come bundled with that specific tool |
| 04:33.52 | dandate2 | now if i I had 5 lines and someone calls in and then has to be channeled to my employees land lines, why does that leave me with 2.5 lines instead of 3? |
| 04:34.29 | dandate2 | i see are you indicating that trixbox also comes with free pbx? |
| 04:34.51 | [TK]D-Fender | dandate2: It does. |
| 04:35.02 | dandate2 | i see, and pbx in a flash comes with it as well right |
| 04:35.13 | jaytee | a forked version of freepbx |
| 04:35.13 | [TK]D-Fender | dandate2: And I see that you have mastered Verizon's New Math |
| 04:35.25 | [TK]D-Fender | dandate2: Yes |
| 04:35.34 | dandate2 | verizons new math? |
| 04:35.49 | [TK]D-Fender | dandate2: Your line calc |
| 04:36.19 | dandate2 | so it takes half a line to conference the forwarding of the inbound call to termination? |
| 04:36.29 | [TK]D-Fender | dandate2: Just my .02 CENTS worth..... |
| 04:36.39 | [TK]D-Fender | dandate2: Half a line? WTF is that? |
| 04:37.06 | dandate2 | meaning if I have 2 people using land lines on my vPBX that holds 5 lines, I can only have 2 callers and then everyone else gets a busy signal? |
| 04:37.09 | [TK]D-Fender | jaytee: We have a winner here... |
| 04:37.19 | jaytee | hehee |
| 04:37.20 | dandate2 | well dr mess was saying that a caller would use up 2.5 lines |
| 04:37.34 | dandate2 | i was just wondering where the .5 came from |
| 04:37.52 | [TK]D-Fender | dandate2: Did you know the word "gullible" isn't in the dictionary? |
| 04:38.07 | dandate2 | heh well |
| 04:38.13 | dandate2 | what could i say man i came here to trust u guys |
| 04:38.19 | NovceGuru | you have 5 lines, and want to have a call come in, and forward back to anothe rline |
| 04:38.43 | jaytee | Ding!!! "Tell him what he's won Don!" "Well, Monty. dandate2 gets a all expense paid vacation to Lapland where he'll get to dine on yak meat and yak butter, sleep in a goatskin yurt and freeze his ass off!" |
| 04:38.49 | zerko | BAH |
| 04:38.53 | zerko | night guys |
| 04:38.54 | dandate2 | right while my emps are still using skype phones and land lines, they have to order these ATA's through ebay or amazon |
| 04:38.56 | [TK]D-Fender | dandate2: Every call is a CHANNEL. PLease do not abuse the term "lines". Do the math. |
| 04:39.30 | dandate2 | k |
| 04:39.56 | dandate2 | so each $30/mo phone line through bandwidth.com is 2 channels right? |
| 04:40.07 | [TK]D-Fender | dandate2: what do THEY say? |
| 04:40.36 | jaytee | hahaahaaa, http://dilbert.com/strips/comic/2008-12-10/ |
| 04:43.14 | dandate2 | ahh i haven't had the insight to ask em, still a noob |
| 04:44.43 | drmessano | OMG |
| 04:44.50 | drmessano | I walk away.... |
| 04:44.58 | *** join/#asterisk CunningPike (n=arodgers@S01060014bf81366b.vc.shawcable.net) |
| 04:45.03 | [TK]D-Fender | dandate2: noob? No.. newb is not knowing some technical detail of a software package or the like. What you have just encountered is a "I didn't actually think to LOOK at the service they were offereing and want someone to hand me a complete solution." |
| 04:45.03 | drmessano | dandate2: Its ONE channel |
| 04:45.06 | drmessano | Already been stated |
| 04:45.26 | [TK]D-Fender | dandate2: Lets be honest here... |
| 04:45.27 | dandate2 | lol |
| 04:45.53 | drmessano | Youve googled and found Trixbox and FreePBX obviously |
| 04:45.53 | NovceGuru | I want to go to bed but I want to see where this goes |
| 04:45.54 | NovceGuru | torn |
| 04:46.11 | drmessano | You somehow found where FreePBX comes with a PRI |
| 04:46.12 | drmessano | I NEVER GOT MINE |
| 04:46.19 | drmessano | Umm |
| 04:46.32 | drmessano | Skype, Skype, Comcast Residential.. Its all a big blur |
| 04:47.03 | dandate2 | well i've come a long way, I was trying to figure out a way to get my callers listening to a pre-recorded message, and all these hosted PBX things were charging up the ass. Then I saw a hosted PBX with an "As Seen On TV" sign, and i knew then that hosted PBX was a scam I could do myself |
| 04:48.02 | [TK]D-Fender | dandate2: hosted PBX takes a host, PBX software and a termination provider. Noone said you needed ANY of this |
| 04:48.03 | dandate2 | so I found a howto on asterisk and came here to offer homage to the nerds that have come before me |
| 04:48.16 | drmessano | Youre no nerd |
| 04:49.01 | dandate2 | really? |
| 04:49.26 | drmessano | Not sure why you would think you have come far enough in any way to even sit next to the nerd table |
| 04:49.49 | dandate2 | fender: so theres a way I can get people to listen to the message and sit in queue without a pbx? |
| 04:50.09 | drmessano | You're like that weird kid that likes knives and tic tacs |
| 04:50.38 | [TK]D-Fender | dandate2: Certainly don't need a host. Don't need an ITSP..... |
| 04:50.38 | drmessano | But flattered you came here |
| 04:50.57 | [TK]D-Fender | dandate2: Lots of different options out there. |
| 04:51.24 | drmessano | I hear Skype is pretty cool |
| 04:52.43 | dandate2 | skype sounds like crap tho |
| 04:53.19 | dandate2 | and without a landline phone you can't have an amplifier |
| 04:53.24 | dandate2 | as far as i know |
| 04:54.31 | [TK]D-Fender | dandate2: What kind of amplifier? Who says that only works on a land-line phone? What do you NEED an "amplifier" for? |
| 04:55.00 | dan__t | Seems that there are a fair amount of RHEL/CentOS RPMs of Asterisk available. Anyone have a favorite packager? |
| 04:55.25 | [TK]D-Fender | dandate2: Perhaps you should describe this complete solution you are envisioning because the piecemeal version has more holes than a #9 sponge |
| 04:55.36 | dandate2 | we use a plantronics amplifier to sound louder and commanding |
| 04:55.38 | [TK]D-Fender | dan__t: TAR <- |
| 04:55.43 | dan__t | heh |
| 04:56.14 | [TK]D-Fender | dandate2: What model? |
| 04:56.27 | dandate2 | s12 |
| 04:56.50 | dandate2 | i know if my emps use a ATA they can use the plantronics |
| 04:57.26 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 04:57.26 | [TK]D-Fender | dandate2: RJ9 handset jacks on both sides to plug inline witha handset on an analog phone? |
| 04:57.33 | *** join/#asterisk Maliuta (n=scooby@kiev.lusan.id.au) |
| 04:58.24 | [TK]D-Fender | dandate2: Yup, looks like... |
| 04:58.38 | [TK]D-Fender | dandate2: Many SIP phones you can use that with as well |
| 04:59.27 | dandate2 | i'm not sure what RJ9 means heh |
| 04:59.59 | dandate2 | oh the modem cable |
| 05:00.12 | [TK]D-Fender | dandate2: " RJ9 handset jacks" Can't read up 4 lines either it seems |
| 05:00.18 | [TK]D-Fender | MODEM!? |
| 05:00.21 | *** join/#asterisk samay (n=samay@121.246.78.142) |
| 05:00.29 | samay | hi all |
| 05:00.39 | drmessano | He doesnt read, he skims |
| 05:00.59 | [TK]D-Fender | drmessano: 0% all right.... |
| 05:02.29 | NovceGuru | motim |
| 05:04.05 | dandate2 | yes that is correct, the s12 hooks up to the analog phone from the phone line, it can also plug into the ata adapter |
| 05:04.37 | dandate2 | an old call center i worked for was using VOIP like that, I was trying to do the same thing but didn't realize how cheap it would be |
| 05:05.20 | dandate2 | they had shoretel phones, but i don't want to have to buy very much hard ware so i was hoping they could just use their existing devices |
| 05:05.44 | dandate2 | be it a land line or a headset plugged into the computer |
| 05:06.01 | dandate2 | is that where having my own vPBX would be beneficial? |
| 05:06.25 | drmessano | PBX |
| 05:06.28 | drmessano | Not vPBX |
| 05:06.33 | drmessano | Its not Virtual |
| 05:06.39 | drmessano | Its a PBX |
| 05:06.43 | *** join/#asterisk Daejeo (n=chatzill@116.126.121.31) |
| 05:07.18 | [TK]D-Fender | dandate2: Stop talking like a PBX is necessarily something you'll plug that AMP into |
| 05:07.27 | Daejeo | after factory recent. what is default tftp address in cisco phone? |
| 05:07.29 | drmessano | ~vpbx |
| 05:07.30 | jbot | doubtful |
| 05:07.49 | [TK]D-Fender | dandate2: an ATA will let you use a regulat phone as a SIP device and it won't CARE where the PBX is, hosted or not. |
| 05:07.59 | dandate2 | well i have my own linksys PAP device, i would just plug that into the router that is also hosting the pbx machine right? |
| 05:08.07 | Daejeo | any idea? |
| 05:08.21 | [TK]D-Fender | dandate2: A router now "hosts" a PBX? huh? |
| 05:08.34 | drmessano | [TK]D-Fender: Sure |
| 05:08.37 | drmessano | [TK]D-Fender: or not |
| 05:08.37 | dandate2 | well provides the connection to the machine |
| 05:09.10 | [TK]D-Fender | dandate2: * is a piece of software on a server. Doesn't matter where it is |
| 05:09.44 | *** join/#asterisk jicksta (n=jicksta@c-24-6-87-187.hsd1.ca.comcast.net) |
| 05:10.04 | dandate2 | ok yes i understand that |
| 05:10.19 | dandate2 | the ATA will log into my * box with just the IP address and such right |
| 05:10.49 | dandate2 | if my employee does not have an ATA box but has a headset hooked up to a computer with internet, could he use a software to log into my * box? |
| 05:10.53 | dandate2 | and take calls? |
| 05:11.56 | [TK]D-Fender | dandate2: Yes |
| 05:12.11 | dandate2 | any recommendations on freeware? =) |
| 05:12.19 | [TK]D-Fender | ~zoiper |
| 05:12.20 | jbot | [~zoiper] Zoiper (Formerly known as Idefisk) is a free SIP and IAX soft-phone for Windows, MacOSX, and Linux that can be found at http://www.zoiper.com |
| 05:12.22 | [TK]D-Fender | ~xlite |
| 05:12.23 | jbot | [~xlite] X-Lite is a free SIP soft-phone for Windows, MacOSX, and Linux that can be downloaded from http://www.counterpath.com/ |
| 05:12.26 | [TK]D-Fender | ~ekiga |
| 05:12.27 | jbot | [~ekiga] Ekiga is an OSS SIP soft-phone for Windows, MacOSX, and Linux (Requires GTK+ which is included with the binary releases) that can be found at http://www.ekiga.org |
| 05:12.39 | dandate2 | alright finally some logworthy material |
| 05:13.08 | dandate2 | joking all of this has been logworhty |
| 05:13.59 | jaytee | nite all |
| 05:14.30 | dandate2 | thanks fender |
| 05:14.57 | samay | Hi. I want to setup a customer care helpline number, where i can save the number of the caller and the duration along with some tabs, can anyone give me a brief idea about how will it be possible >? |
| 05:15.29 | [TK]D-Fender | samay: CDR <- |
| 05:15.40 | [TK]D-Fender | samay: There is a nice chapter on this in THE BOOK |
| 05:15.42 | [TK]D-Fender | ~book |
| 05:15.43 | jbot | rumour has it, book is probably 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://tfot.leifmadsen.com or see ~buybook |
| 05:15.52 | [TK]D-Fender | samay: and what are "tabs"? |
| 05:16.05 | samay | i am sorry... its tags |
| 05:16.33 | samay | tags refer to words related to the query which the customer made |
| 05:16.48 | dandate2 | sorry whats CDR stand for? |
| 05:16.56 | samay | what is CDR ? |
| 05:17.15 | dandate2 | call data recording? |
| 05:17.53 | [TK]D-Fender | ~cdr |
| 05:17.54 | jbot | [cdr] Call Detail Record, a log of what happens to the call at each step through its traversal of the PBX, details like from, to, time, duration, number dialled etc, useful for billing also - it could also be Compact Disc Recordable, see cdrw |
| 05:18.15 | dandate2 | right |
| 05:18.41 | samay | Thanks jbot... but how can i make a number which anyone can dial from anywhere |
| 05:18.41 | jbot | no worries, samay |
| 05:19.12 | dandate2 | how can I program CDR to read information sent to it automatically by email? I would use my email responder to gather members information and put it into the VPBX, that way my employees can know if the person calling is a member already or not |
| 05:19.21 | samay | i mean a global helpline which any one can dial using VoIP or PSTN |
| 05:19.34 | dandate2 | samay you need to get yourself a 1-800 number |
| 05:19.51 | Daejeo | [TK]D-Fender: does cisco phone automatically pickup tftp address from dhcp? |
| 05:19.54 | dandate2 | and you need to get yourself some kind of hosting for your auto-attendant |
| 05:20.06 | Daejeo | after factory reset |
| 05:20.14 | samay | ok.. i think i can get more help abt CDR by reading.. but can u tell me what is 1-800 number |
| 05:20.19 | samay | and from where can i get it |
| 05:20.36 | dandate2 | try www.broadvoice.com |
| 05:20.44 | dandate2 | will host a pbx for you |
| 05:20.45 | [TK]D-Fender | Daejeo: I don't do Cisco... |
| 05:20.52 | *** join/#asterisk fogo (n=Paul@69.169.132.35.provo.static.broadweave.net) |
| 05:21.02 | dandate2 | but if you want to save on minuets you need to set up your own pbx machine and just pay for SIP trunking |
| 05:21.43 | [TK]D-Fender | samay: pay for lines with a telco, or sign up with an ITSP |
| 05:21.45 | [TK]D-Fender | ~itsp |
| 05:21.45 | jbot | [~itsp] An ITSP is an Internet telephony Service Provider or "VoIP Telephone company". The allow you to either PLACE calls to the PSTN (called Termination), RECEIVE calls from the PSTN (called Origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs |
| 05:22.24 | dandate2 | ~itsplist-us |
| 05:22.25 | jbot | [~itsplist-us] Here are some popular ITSPs (USA) starting with the more respected ones : http://www.teliax.com , http://connect.voicepulse.com , http://www.nufone.net , http://www.broadvoice.com, http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net |
| 05:22.46 | [TK]D-Fender | dandate2: He's in India... who said he cared about a North American based #? |
| 05:23.00 | samay | cool.. so once i have a number i can direct it to my server having asterisk and do all the programming to store the number, duration and other info from CDR |
| 05:23.07 | samay | am I right ?? |
| 05:23.16 | [TK]D-Fender | samay: Yes |
| 05:24.07 | [TK]D-Fender | samay: But please pay attention to what kind of #, local to where, etc |
| 05:25.12 | samay | Sorry Fender.. didnt get you... what kind of # ?? |
| 05:25.37 | Sargun | Can someone add flowroute to that list |
| 05:26.51 | [TK]D-Fender | samay: You say you want to have a number that people can call you on. WHERE? Do you want an INDIAN phone #? So that people there can call you "locally"? One in GERMANY? USA? WHERE? |
| 05:27.43 | samay | Well.. the helpline should be global. which means people can call from anywhere... |
| 05:28.05 | [TK]D-Fender | samay: Well anyone can call anywhere in the world that they want... its a question of who it COSTS <- |
| 05:28.52 | samay | ok... what abt a toll free number ? is it possible to have a toll free number around the world |
| 05:29.16 | samay | i will pay the cost |
| 05:29.29 | samay | most of my customers will be in India and US |
| 05:29.35 | [TK]D-Fender | samay: You can certainly do toll-free within a larger region. Is it always toll-free for someone anywhere in the world to call a USA based "toll-free" number? |
| 05:30.47 | samay | Fender.. do you mean if I have a Toll-free number for USA.. i can call from india without paying anything... am i right |
| 05:31.35 | [TK]D-Fender | samay: No, I'm asking YOU if you can call a USA toll-free without getting charged. |
| 05:31.59 | samay | i dont know.. i havent tried |
| 05:32.09 | [TK]D-Fender | samay: Go get your own answer then |
| 05:32.36 | samay | ok. will have to find out... |
| 05:32.41 | [TK]D-Fender | samay: As for the rest there are ITSP's that can serve just about any major center. Go shop around for service |
| 05:33.00 | samay | sure... thanks |
| 05:36.22 | *** join/#asterisk MaliutaLap (n=biteme@kiev.lusan.id.au) |
| 05:37.29 | dandate2 | I was looking through some of those itsp's jbot provided, i've basically narrowed the cheapest for high volume down to bandwidth.com and http://vitelity.net , can anyone make a distinguishment between the 2? |
| 05:39.38 | *** join/#asterisk xuser (n=xuser@unaffiliated/xuser) |
| 05:42.40 | [TK]D-Fender | dandate2: Both are pretty decent |
| 05:43.47 | *** join/#asterisk xuser (n=xuser@unaffiliated/xuser) |
| 05:46.34 | *** join/#asterisk xuser (n=xuser@unaffiliated/xuser) |
| 05:46.51 | dandate2 | i am a little confused though mabye someone can help me clarify. I am running an inbound call center so outbound calling makes up for less than 1% of our activity. Would I save money purchasing unlimmited inbound local numbers for only $8/month rather than purchasing bandwidth.com 's $30/mo unlimmited inbound/outbound? |
| 05:47.25 | dandate2 | or is there some other tricks and catches in vitelity's service with these prices for channels |
| 05:48.08 | dandate2 | and when i said inbound local numbers for $8/mo i was referring to vitelity |
| 05:49.14 | [TK]D-Fender | dandate2: People would be calling you at a number local to a place of your choosing? |
| 05:50.20 | dandate2 | people would call me long distance but my sales line is the local number to san francisco not a 1800 # |
| 05:51.05 | dandate2 | see i only advertise in california as of now |
| 05:51.21 | dandate2 | so people call my 415 # knowing they are calling san francisco, its local enough to em but stil long distance |
| 05:51.44 | [TK]D-Fender | dandate2: So a local number is fine? |
| 05:52.07 | *** join/#asterisk Bad_Robot- (n=Bad_Robo@cpe-76-173-219-25.socal.res.rr.com) |
| 05:55.28 | *** join/#asterisk ecret (n=ecret@CPE001e9002348e-CM001225d8ab30.cpe.net.cable.rogers.com) |
| 05:56.20 | *** join/#asterisk jicksta_ (n=jicksta@c-24-6-87-187.hsd1.ca.comcast.net) |
| 05:56.33 | dandate2 | yes a local number will do |
| 05:57.08 | dandate2 | i may want to advance to 1-800 # in the future and expand to other states, but for now just using local numbers brings in the calls |
| 05:58.15 | [TK]D-Fender | dandate2: well go do the math |
| 05:58.48 | dandate2 | well i am finding this hard to believe so i think i might be doing my math wrong heh |
| 05:59.28 | dandate2 | i could just buy 5 local lines for $40/mo from vitelity, or I could pay $150/mo to bandwidth.com and get unlimmited outbound which i don't need... |
| 05:59.43 | *** join/#asterisk jicksta (n=jicksta@c-24-6-87-187.hsd1.ca.comcast.net) |
| 05:59.45 | dandate2 | someone slap me if i'm being silly |
| 06:00.02 | drmessano | You dont need 5 lines either |
| 06:00.10 | drmessano | Get one line and make sure you have 5 channels |
| 06:00.19 | dandate2 | ok that sounds interesting |
| 06:02.39 | [TK]D-Fender | get 1 *DID* from a provider who'll offer you 5 simultaneous channels |
| 06:03.45 | dandate2 | ~did |
| 06:03.46 | jbot | i heard did is Direct Inward Dialing, or just a phone number |
| 06:03.56 | dandate2 | lol |
| 06:04.19 | dandate2 | thats interesting i wonder why noone has mentioned that before |
| 06:05.48 | *** join/#asterisk farkus_ (i=chatzill@cpe-98-14-94-76.nyc.res.rr.com) |
| 06:06.31 | drmessano | Then you have 1 DID to mess with |
| 06:07.04 | dandate2 | yeah thats genius drmess |
| 06:07.22 | dandate2 | u think that'l be even cheaper than bandwidths 5 sip trunks for $150/mo ? |
| 06:07.48 | [TK]D-Fender | dandate2: Do. The. Math. |
| 06:08.14 | [TK]D-Fender | dandate2: and you aren't breaking down the service you are now attempting to compare. |
| 06:08.43 | [TK]D-Fender | dandate2: We've already established that you didn't even read the details. |
| 06:09.14 | dandate2 | i'm looking at this www.didforsale.com now |
| 06:09.21 | dandate2 | sorry i'm just excited |
| 06:12.22 | dandate2 | i think that this didforsale.com might not run my credit |
| 06:12.25 | dandate2 | =) |
| 06:16.22 | dan__t | I'm asking for a sharp stick in the eye if I try to dev * in VMware, huh |
| 06:18.04 | hardwire | no |
| 06:18.09 | hardwire | should work fine there big guy |
| 06:18.29 | hardwire | I dev * in qemu/virtualbox and even other things like openvz and vserver |
| 06:18.38 | dan__t | Neat. |
| 06:19.02 | dan__t | I'm a BOFH for package management, but you know what, I don't want to have to wade through 219843784238423423 fscking packages to get * up and running rightnow. |
| 06:19.11 | hardwire | lots of virtual pbx companies use vmwareish stuff |
| 06:19.16 | dan__t | Just going to make a BS VM and hack on it in there. |
| 06:19.34 | hardwire | eh? |
| 06:19.38 | dan__t | used/new, dandate2? |
| 06:20.13 | hardwire | dan__t: install debian, install asterisk, install a lunch |
| 06:20.13 | dandate2 | used / new? |
| 06:20.20 | dan__t | I'd quicker stab myself in the d... arm. |
| 06:20.28 | dan__t | bad joke, dandate2, sorry. |
| 06:20.42 | dandate2 | hey does anyone knwo if I can find DID availability by zipcode? |
| 06:20.57 | hardwire | dandate2: check out teliax.com and didx |
| 06:21.00 | dan__t | zipcode, eh? Area code yes, zip code, not so sure. |
| 06:21.04 | dan__t | I was just going to say Teliax. |
| 06:21.16 | dan__t | I've used them in the past. Pleasant service, I just needed something more friendly at the time. |
| 06:21.29 | dan__t | Switched to Junction Networks' OnSIP |
| 06:21.35 | hardwire | otherwise.. look up charts on area codes per state and zip code somewhere |
| 06:21.41 | hardwire | they are bound to exist |
| 06:21.56 | hardwire | zip codes aren't really part of telephone systems |
| 06:22.02 | hardwire | area codes are where it's at |
| 06:22.16 | hardwire | ordering dids from sprint doesn't result in them ever asking about a zip code :) |
| 06:22.31 | dan__t | I haven't hacked on Asterisk for a very, very, very long time... an old friend of mine, an infomercial extraordinaire asked me to come up with a project, so you guys might see me hanging around a lot more heh. |
| 06:22.55 | dandate2 | ok well i was confused because i was at didforsale.com and their rate centers list my area code multiple times for different cities/districts but i didn't see mine |
| 06:23.06 | dandate2 | didn't see my specific district listed |
| 06:23.10 | hardwire | dan__t: ok.. if you need extra support just ask here.. I'm always available via privmsg as well |
| 06:23.21 | dan__t | I certainly appreciate it. |
| 06:23.34 | dan__t | I bought the ASterisk book today. Glad to see the 2nd Edition out. |
| 06:23.43 | dan__t | The first was a good crash course but didn't explain things very well. |
| 06:23.49 | dan__t | The 2nd Ed. is absolutely fantastic. |
| 06:24.00 | dan__t | Don't suppose the author chills in here by chance, eh? |
| 06:24.36 | [TK]D-Fender | dan__t: regularly |
| 06:24.40 | dan__t | Very nice. |
| 06:24.47 | dan__t | I'd like to thank him personally. |
| 06:26.16 | [TK]D-Fender | dan__t: "them" |
| 06:26.22 | dan__t | Yep, all three. |
| 06:26.28 | dan__t | Sorry. |
| 06:27.55 | dandate2 | quick question, i am about to sign up for didforsale.com . If I pick up 1 DID line with 5 channels, would I beable to forward calls from my * box to landline analog phone users? |
| 06:28.27 | dan__t | I wouldn't think so, seems like they're just a peer. |
| 06:28.29 | [TK]D-Fender | dandate2: look at the service they are offering... |
| 06:28.36 | dan__t | DID == exclusively inbound, no? |
| 06:29.33 | dan__t | ducks. |
| 06:29.55 | dandate2 | if hes confused, i am completely lost |
| 06:30.03 | dan__t | haha |
| 06:31.09 | *** join/#asterisk jks (i=jks@193.189.93.254) |
| 06:34.01 | *** join/#asterisk brian (n=brian@unaffiliated/brian) |
| 06:36.47 | *** join/#asterisk brian (n=brian@unaffiliated/brian) |
| 06:40.37 | *** join/#asterisk hrmphh (i=patrick@notchill.com) |
| 06:41.05 | hrmphh | for some reason my incoming faxes are just going unaswered (using hylafax). could someone please take a peek at http://pastebin.com/m4706fb8a? |
| 06:41.31 | hrmphh | keep getting FaxGetty[12901]: ANSWER: Ring detected without successful handshake |
| 06:42.14 | [TK]D-Fender | hrmphh: pastebin the * CLI output for the complete call |
| 06:43.13 | hrmphh | it gets sent to iaxmodem fine |
| 06:43.37 | hrmphh | iaxmodem answers and a while later returns w/non-zero exit |
| 06:44.18 | [TK]D-Fender | hrmphh: pastebin the * CLI output for the complete call |
| 06:45.39 | *** join/#asterisk drmessano (n=nonya@pdpc/supporter/active/drmessano) |
| 06:47.02 | *** join/#asterisk kerx (n=kerx@adsl-69-104-64-66.dsl.irvnca.pacbell.net) |
| 06:52.02 | *** join/#asterisk fogo (n=Paul@69.169.132.35.provo.static.broadweave.net) |
| 06:52.08 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 06:54.01 | *** join/#asterisk brian (n=brian@unaffiliated/brian) |
| 06:54.25 | hrmphh | k |
| 06:57.47 | hrmphh | hrm think i may have fixed |
| 06:57.49 | hrmphh | nfs-common wasnt installed |
| 06:57.56 | hrmphh | and this bad boy has /var/spool/hylafax nfs mounted |
| 06:58.55 | hrmphh | yup |
| 07:00.18 | hardwire | ew |
| 07:00.58 | hardwire | I'd probably end up using unison and a post-fax script. |
| 07:01.14 | hrmphh | ? |
| 07:01.22 | hrmphh | elaborate? |
| 07:01.31 | hardwire | well, I have a similar setup |
| 07:02.03 | hardwire | if I wanted all my hylafax /var/spool/ data to go somewhere else, I'd probably store locally (cause faxes are more important than stale nfs mounts) then push to the storage server |
| 07:02.22 | hrmphh | got it |
| 07:02.32 | hardwire | I don't need all of it to go to the storage server |
| 07:02.34 | hrmphh | not a bad idea but im trying to move to a hdd free asterisk box |
| 07:02.37 | hrmphh | cd-rom booted |
| 07:02.39 | hrmphh | and auto failover |
| 07:02.43 | [TK]D-Fender | ok, checkout time. Later all |
| 07:02.44 | hardwire | ah ok |
| 07:02.45 | hrmphh | to a backup box |
| 07:03.04 | hardwire | hrmphh: cool, hope it works out well for you |
| 07:03.10 | hrmphh | thx |
| 07:03.12 | hrmphh | what is unison btw? |
| 07:03.26 | hardwire | 2 way sync using librsync |
| 07:03.44 | hrmphh | ah |
| 07:03.45 | hardwire | so it can sync two repositories of data bi-directionally |
| 07:04.02 | hardwire | nfs booting a hylafax server is kinda interesting |
| 07:04.04 | hrmphh | why not just rsync? |
| 07:04.31 | hardwire | hrmphh: because rsync is one way? I supposed you would make changes to the stored data just as easily as the original data |
| 07:04.35 | hardwire | like deleting files |
| 07:04.46 | hrmphh | got it |
| 07:05.09 | hardwire | it's like a slow, cached, network file system. |
| 07:05.11 | hardwire | :) |
| 07:08.19 | SwK | nothing wrong with nfs heh... i've had whole clusters of machines pxe booting to run hylafax and asterisk |
| 07:08.40 | SwK | they had HDDs in them but only mounted for temp and log files |
| 07:14.11 | hardwire | SwK: I'd probably end up using nfs with nbd's for swap/log for diskless |
| 07:14.19 | hardwire | kinda like how ltsp works anyways |
| 07:21.01 | *** join/#asterisk IPGHOST (n=IPGHOST@116.71.170.25) |
| 07:21.56 | hrmphh | swk; you have t1 cards in those? |
| 07:22.31 | IPGHOST | hi |
| 07:22.38 | hrmphh | http://www.red-fone.com/Products/fonebridge2/ |
| 07:22.45 | hrmphh | looking at using one of those |
| 07:23.21 | IPGHOST | whats up buddy |
| 07:26.09 | dandate2 | how do i tell how many channels a DID line comes with from didforsale.com? |
| 07:28.29 | dandate2 | anyone know anything about callcentric.com? |
| 07:28.39 | drmessano | You just jumping back and forth and pasting? |
| 07:28.48 | SwK | I dont use T1 Cards unless its just for 1 or 2 T1s |
| 07:29.08 | SwK | most of the stuff I do is for 2 to 10 DS3s (sometimes more) |
| 07:31.50 | dandate2 | haha sorry i am hunting! |
| 07:42.29 | *** join/#asterisk aurax (n=kvirc@DSL212-235-111-122.bb.netvision.net.il) |
| 07:42.35 | aurax | morning |
| 07:46.55 | *** join/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 07:46.56 | aurax | I'm trying to setup a trunk to GSX9000 (cisco) that has no authentication enabled. how do i do that? any idea? |
| 07:51.09 | *** join/#asterisk zeeesh (n=zeeesh@203.215.179.43) |
| 07:59.07 | *** join/#asterisk fogo (n=Paul@69.169.132.35.provo.static.broadweave.net) |
| 08:03.41 | *** join/#asterisk KOCATEPE (n=admin@88.247.133.123) |
| 08:04.31 | aurax | anyone ? |
| 08:05.18 | *** join/#asterisk reneger (n=reneger@dslb-088-078-115-034.pools.arcor-ip.net) |
| 08:10.44 | *** join/#asterisk h-idrisi (n=h-idrisi@212.100.196.195) |
| 08:14.19 | SwK | aurax, just set up a sip trunk with no authentication on it... just make sure you have that cisco on a private network or filter the SIP traffic to it heh |
| 08:15.00 | *** join/#asterisk Deeewayne (n=dwayne@c-76-29-245-9.hsd1.al.comcast.net) |
| 08:15.00 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 08:20.29 | tzafrir_laptop | aurax, pastbin your existing configuration, error messages, sip debug? |
| 08:20.39 | *** join/#asterisk aiksa[LV] (n=aiks@mx.fiveplus.lv) |
| 08:20.57 | aurax | ok |
| 08:22.10 | aurax | http://pastebin.com/d73729616 <- users.conf |
| 08:29.06 | tzafrir_laptop | aurax, and what happens when you try to call it? |
| 08:29.18 | tzafrir_laptop | How are they supposed to identofy you? |
| 08:29.40 | *** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose) |
| 08:29.41 | aurax | by IP |
| 08:30.02 | *** part/#asterisk tessier (n=treed@kernel-panic/sex-machines) |
| 08:30.07 | aurax | when i remove user = ... and secret = they saying it stop's from trying to reconnect |
| 08:31.39 | tzafrir_laptop | also: do you only send calls to them or als get calls from them? |
| 08:31.53 | tzafrir_laptop | are you supposed to register with them? |
| 08:32.11 | aurax | both |
| 08:32.17 | aurax | i think i suppose to, yes... |
| 08:32.19 | aurax | no? |
| 08:32.58 | aurax | i mean, you must register, no ? |
| 08:33.11 | aurax | should i add insecure= very to trunk_1? |
| 08:37.01 | aiksa[LV] | Hi tzafrir_laptop , sorry I was wrong module reload chan_dahdi.so works without destroying active connections over E1 |
| 08:40.06 | tzafrir_laptop | aurax, for starters, don't guess . What do you see on: sip show peers ? |
| 08:40.10 | aiksa[LV] | I have another question regarding zapata.conf - If i wnated to created two groups of channels using the same channels, but having different txgain/rgain values how should I proceed? |
| 08:42.11 | tzafrir_laptop | group = 1,2,4 |
| 08:42.35 | tzafrir_laptop | The groups are technically saved as bitmasks for the channels |
| 08:42.39 | aiksa[LV] | tzafrir_laptop: and when how do i allocate group specific rxgains txgains? |
| 08:42.55 | tzafrir_laptop | you can't |
| 08:43.22 | aiksa[LV] | ok. Any option to manipulate this through dialplan before dial? |
| 08:43.24 | tzafrir_laptop | as I mentioned, the group does not exist as a seperate entity |
| 08:43.46 | aiksa[LV] | tzafrir_laptop: - ok. i got it now. |
| 08:44.06 | aiksa[LV] | and the feature listing before has an influence on the channels directive not the group |
| 08:44.24 | tzafrir_laptop | hmm... why do you need to set a different gain, BTW? |
| 08:46.02 | aiksa[LV] | different SIP user agents provide different loudness for outgoing audio. |
| 08:46.14 | *** join/#asterisk Deeewayne (n=dwayne@c-76-29-245-9.hsd1.al.comcast.net) |
| 08:46.14 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 08:49.18 | aiksa[LV] | tzafrir_laptop: and now I am at the point where I have to have a txgain on -3 something to battle echo for outgoing calls from Snom, but this is way to low for the outgoing calls for Zoiper |
| 08:52.27 | *** join/#asterisk Chris-NB (n=chris@85-126-61-10.work.xdsl-line.inode.at) |
| 08:55.37 | *** join/#asterisk Avelino (n=Avelino@mail.paterno.com.br) |
| 09:02.45 | *** join/#asterisk botox93 (n=botox93@213.221.82.242) |
| 09:05.25 | dan__t | Well, this is embarrassing. I keep getting: [[Jan 12 02:03:41] WARNING[29971]: file.c:582 ast_openstream_full: File test does not exist in any format. I see /var/lib/asterisk/sounds/test.wav, and it is referenced in the application by exten => 55,1,Playback(test) |
| 09:06.29 | *** join/#asterisk dandate (n=dandate2@c-71-202-125-220.hsd1.ca.comcast.net) |
| 09:06.51 | dandate | hey guys i am unable to log into my * box from outside the network, does this mean i need to enable port forwarding on my router? |
| 09:07.02 | dan__t | Define "log in". |
| 09:07.04 | dandate | the connection times out when i try to go to the ip |
| 09:07.11 | dandate | as in http://71.202.124.1 |
| 09:07.14 | dan__t | SSH, HTTP, SIP, etc etc |
| 09:07.23 | dan__t | I'd say yes, or 1:1 NAT |
| 09:07.27 | dan__t | (dmz the host) |
| 09:07.38 | dan__t | IN fact, if you're using SIP, I'd almost guarantee you'll need to DMZ the host. |
| 09:07.47 | dandate | ok |
| 09:07.49 | dan__t | More likely, though, your ISP is blocking inbound port 80. |
| 09:07.57 | aiksa[LV] | dan__t: i might be by mile of - but i am not sure that Playback can playback plain wav files |
| 09:07.58 | dan__t | That's assuming a lot, so take that with a grain of salt. |
| 09:08.08 | dan__t | I've even tried with known gsm files, aiksa[LV]. |
| 09:08.08 | aiksa[LV] | try putting ulaw alaw or gsm in that folder |
| 09:08.19 | dan__t | I'm concerned maybe I don't have a gsm codec...? |
| 09:08.20 | dandate | so i should demilartarized the IP of the * box or the router? |
| 09:08.30 | dan__t | Yes, dandate. |
| 09:08.37 | dan__t | I'd also test to see if your ISP blocks port 80. |
| 09:08.43 | dandate | how do i do that? |
| 09:09.08 | dan__t | That's not asterisk related, but for whatever reaosn you're trying to hit the Asterisk box via HTTP |
| 09:09.08 | aiksa[LV] | dan__t: another option is that Playback looks for file in other location |
| 09:09.14 | dan__t | Well. Port forward, try to hit it, and if it doesn't work, they're blocking it. |
| 09:09.32 | aiksa[LV] | please try Playback(/var/lib/asterisk/sounds/test) instead |
| 09:09.37 | dan__t | aiksa[LV], can I query Asterisk and find out where it thinks its sounds sound be, if not the default /var/lib/asterisk? |
| 09:09.52 | dandate | ok so before i try demilartirizing i should try enabling port 80 and then try logging in again? |
| 09:10.04 | aiksa[LV] | dan__t: I guess /etc/asterisk/asterisk.conf had a refernce to those locations |
| 09:10.37 | dan__t | I didn't see any. |
| 09:10.49 | dan__t | You can if you want, doesn't matter to me. |
| 09:11.00 | dan__t | I'm saying DMZ the host because it will then play better with external SIP clients trying to connect to it. |
| 09:12.29 | aiksa[LV] | dan__t: did Playback(/var/lib/asterisk/sounds/test) work? |
| 09:12.35 | dan__t | Bah. That goes against everything I've ever read. |
| 09:12.36 | dan__t | Yeah it did. |
| 09:13.21 | dandate | hmm my shady ass router does not allow me to enable ports on the firewall, i can set a DMZ zone though , will that fix the problem anyway? |
| 09:13.24 | dan__t | Inherently yes. |
| 09:13.38 | dan__t | What kind of router? |
| 09:13.53 | dan__t | Sometimes called "Applications and Gaming" or something |
| 09:13.57 | dandate | gigafast discontinued not for sale no manual avaial |
| 09:14.11 | dan__t | Very nice. |
| 09:14.24 | dandate | i even tried calling for customer service and they couldn't help me till i bought a linksys |
| 09:14.31 | aiksa[LV] | gigafast sounds like an appropraite name |
| 09:14.35 | dandate | but i am running 2 diff internet connections |
| 09:14.35 | dan__t | haha |
| 09:14.40 | dan__t | gigafasttogetofftheshelf |
| 09:15.22 | aiksa[LV] | it could as well be GigaGlobal telecomunications and consulting services LLC |
| 09:15.42 | aiksa[LV] | registered in some god forgotten island |
| 09:15.48 | dan__t | Thanks, aiksa[LV]. |
| 09:16.03 | dan__t | Although, like I've said, all sources just say to put a relative filename without an extension |
| 09:16.46 | dandate | ok i have demilitarized, can anyone tell me what happens if they go to http://71.202.125.220 for me its just asking me to log into my router |
| 09:17.11 | dan__t | waiting.. |
| 09:17.35 | dan__t | It comes up. |
| 09:17.53 | dan__t | Its waiting on something though. |
| 09:18.17 | dan__t | There, it goes. |
| 09:18.19 | dan__t | brb, smoke |
| 09:18.28 | *** join/#asterisk Avelino (n=Avelino@mail.paterno.com.br) |
| 09:18.47 | dandate | ok so you were able to access my pbx server? |
| 09:18.50 | dan__t | yes |
| 09:18.56 | dandate | cuz for me it just accesses my router, i wonder why |
| 09:18.58 | dan__t | smoke. brb. for real this time. |
| 09:19.09 | dan__t | Because the router is a piece of shit. |
| 09:19.13 | dandate | lol |
| 09:19.13 | dan__t | I don't know. brb. |
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| 09:21.48 | aiksa[LV] | dan__t: if Playback(/var/lib/asterisk/sounds/test) works it means asterisk is looking for files in some other lcoation |
| 09:34.04 | dan__t | I understand, that's why I was wondering how to query for that. |
| 09:34.47 | dan__t | I didn't find anything in the conf dir. |
| 09:35.23 | dandate2 | shit how do i check what my admin password is |
| 09:36.13 | dandate2 | i had it right before |
| 09:39.33 | dandate2 | sorry it was just maint |
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| 09:42.35 | dan__t | I don't know, its your router. |
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| 09:48.29 | *** join/#asterisk scruz (n=scruz@41.220.73.170) |
| 09:48.47 | scruz | good day |
| 09:51.33 | scruz | how may i use one asterisk setup as a client to another? i have asterisk-win32 on my own desktop, and i'd like to use it as a client to another asterisk server |
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| 09:57.22 | *** join/#asterisk ArtVandalae (n=SuperUnk@122.111.176.127) |
| 09:59.56 | ArtVandalae | Hi all, I'm looking to set up Asterisk (with a web GUI). I'm looking for a dedicated distribution that's small. I've had a look at both trixbox and elastic and both seem to be ~600MB. This seems really over-the-top (I mean what do these distributions ship to make it so large?). Are there any alternatives? |
| 10:00.48 | *** join/#asterisk dandate (n=dandate2@adsl-76-254-45-239.dsl.pltn13.sbcglobal.net) |
| 10:00.59 | dandate | my DID provider is instructing me to edit my sip.conf file, can i just do this from the freepbx gui? |
| 10:02.14 | *** join/#asterisk pikachu2000 (n=pikachu2@pikachu.cyberdesigns.co.za) |
| 10:06.59 | dandate | how do i find sip.conf anyway/ |
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| 10:15.24 | scruz | dandate: it should be in /etc/asterisk/ |
| 10:16.23 | scruz | ArtVandale: you get - a complete OS with asterisk and addons in 600MB |
| 10:16.45 | scruz | basically, it's a dedicated asterisk machine waiting for install |
| 10:18.02 | scruz | the alternative is to roll your own. but then it won't likely be a pbx-in-a-box like elastix or trixbox |
| 10:20.06 | scruz | artvanadalae my bad. |
| 10:20.40 | tzafrir_laptop | ArtVandalae, #astlinux |
| 10:21.12 | tzafrir_laptop | You can also build the same from Debian packages with much less |
| 10:21.59 | ArtVandalae | Thanks |
| 10:22.38 | vi390 | is there anything known, that when using Agi scripts, and complex class structures it simply does not work to pick up the call (no error messages, nothing. And scripts in standalone passes all tests, just when bound into Agi will not work...) |
| 10:22.51 | vi390 | used language: python |
| 10:23.05 | scruz | how can i use one asterisk install as a client to another? |
| 10:23.22 | vi390 | maybe the response time of the imported modules is to long ? |
| 10:23.53 | scruz | as a SIP client, that is |
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| 10:33.50 | scruz | sorry for asking what now appears as a dumb question |
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| 11:15.18 | scruz | i finally figured out the workaround for my automated calling system, even though someone already solved the problem |
| 11:16.04 | scruz | use *two* asterisk servers. one acts as a SIP client to the other, and picks the calls that come in to the channel. the other is for routing the calls to the outbound extension |
| 11:17.53 | scruz | so when the call file is processed, the user asterisk install answers the incoming call, and then the dialled extension can ring. when the dialled extension is answered, i can play the message to t :) |
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| 11:44.50 | exothermc | Where do you chose to compile pbx_functions.so ? |
| 11:53.30 | *** join/#asterisk ipguy2 (n=ipguy2@124-170-236-163.dyn.iinet.net.au) |
| 11:53.40 | ipguy2 | hi all |
| 11:54.35 | ipguy2 | in * is it possible to setup a extention that will require a user to press and number that will then user a particluar voip account ? |
| 11:55.06 | ipguy2 | like dialing 1, then the number to use voip provider #1 and 2 then the number for voip provider #2 ? |
| 11:55.15 | ipguy2 | ? |
| 11:56.24 | ipguy2 | so if i have two voip accounts registered i can use one account or another as i like ? |
| 11:57.01 | ipguy2 | am i makeing any sense ? |
| 12:13.45 | *** join/#asterisk farkus_ (i=chatzill@cpe-98-14-94-76.nyc.res.rr.com) |
| 12:21.43 | ipguy2 | anyone ? |
| 12:23.32 | ipguy2 | is anyone awake ? |
| 12:26.25 | yang | ~ãsk |
| 12:26.32 | yang | ~ask |
| 12:26.33 | jbot | extra, extra, read all about it, ask is Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 12:26.33 | ipguy2 | ask ? |
| 12:27.36 | yang | ipguy2: yes, what you are asking is that you need to modify the dialplan, its possible |
| 12:27.46 | ipguy2 | i have two voip registration, i want to be able to dial 9 then a number to use one provider and 8 and then a number to use the other provider. |
| 12:28.56 | yang | ipguy2: you need to create 2 sip peers (each to a VOIP uplink) then add the prefix for both in the dialplan and strip that prefix afterwards inside the dialplan |
| 12:29.30 | ipguy2 | i've tried ;exten => 9|_X.,1,Dial(SIP/${EXTEN}@pennytel-out,60,t) |
| 12:29.30 | ipguy2 | i've tried... "exten => 9|X.,1,Dial(SIP/${EXTEN}@pennytel-out" but that does n;t work |
| 12:29.38 | ipguy2 | ooops sorry |
| 12:31.00 | exothermc | What do I need to get this working: set_format: Unable to find a codec translation path from 0x4 (ulaw) to 0x2 (gsm) ? |
| 12:31.06 | ipguy2 | i'm a little lost |
| 12:32.48 | ipguy2 | yang, i'm a little lost. |
| 12:33.12 | yang | ipguy2: http://pastebin.com/m5adc5b01 |
| 12:34.06 | ipguy2 | i see... thanks !! will give it a try |
| 12:35.28 | yang | ipguy2: {EXTEN:1} will strip one digit on start, you can add {EXTEN:2} is you will want to strip two digits |
| 12:41.47 | ipguy2 | yang: it worked, thanks man !!! |
| 12:42.00 | ipguy2 | yang, appreciate your help ! |
| 12:44.22 | yang | ipguy2: you are welcome |
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| 12:46.54 | yang | exothermc: maybe try to add allow=gsm allow=ulaw into sip account |
| 12:47.23 | dandate2 | i seem to be getting direct SIP phone calls but i cannot receive incoming calls through my DID provider |
| 12:47.26 | *** join/#asterisk leeky (n=leeky@yaffle.leeky.org.uk) |
| 12:47.48 | yang | dandate2: have you configured incoming extensions ? |
| 12:47.53 | dandate2 | yes |
| 12:48.02 | yang | paste the error |
| 12:48.09 | dandate2 | ha none created! |
| 12:48.09 | yang | ~tell dandate2 about pb |
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| 13:18.31 | eppigy | hello |
| 13:18.40 | eppigy | i am dave |
| 13:19.46 | *** join/#asterisk russellb (n=russellb@asterisk/digium-open-source-team-lead/russellb) |
| 13:19.46 | *** mode/#asterisk [+o russellb] by ChanServ |
| 13:20.29 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 13:24.13 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.70.151) |
| 13:35.31 | *** join/#asterisk major_rus (n=major@thsaimaa.ru) |
| 13:36.33 | major_rus | can you say diff between 1.6 and 1.4 |
| 13:36.36 | major_rus | ? |
| 13:37.11 | russellb | yes |
| 13:37.17 | russellb | which version of 1.6? |
| 13:37.19 | russellb | or just trunk? |
| 13:37.27 | russellb | the diff is going to be quite large ... |
| 13:37.46 | [TK]D-Fender | major_rus: 0.2 |
| 13:38.23 | *** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee) |
| 13:38.37 | major_rus | version |
| 13:38.52 | russellb | oic. |
| 13:39.21 | russellb | svn diff http://svn.digium.com/svn/asterisk/branches/1.4 http://svn.digium.com/svn/asterisk/trunk | diffstat |
| 13:39.27 | russellb | <PROTECTED> |
| 13:39.35 | major_rus | :) |
| 13:39.49 | major_rus | fnks |
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| 13:52.24 | [TK]D-Fender | russellb: Would love to know how that kind of answer is of any value... except to a statistician :) |
| 13:53.17 | *** join/#asterisk waverly360 (n=waverly@ns2.dalcon.com) |
| 13:53.25 | aiksa[LV] | quit |
| 13:54.04 | russellb | [TK]D-Fender: *shrugs* |
| 13:54.19 | [TK]D-Fender | russellb: "I see crazy people..." |
| 13:54.28 | anonymouz666 | haha |
| 13:54.31 | anonymouz666 | this is [TK]D-Fender |
| 13:55.17 | eppigy | hello |
| 13:55.26 | [TK]D-Fender | eppigy: you are dave |
| 13:55.32 | eppigy | yes |
| 13:55.38 | eppigy | that is a factual statement |
| 13:55.39 | [TK]D-Fender | anonymouz666: No... this is SPINAL TAP! |
| 13:55.50 | eppigy | \o/ |
| 13:57.15 | *** join/#asterisk nn (i=A5E649A3@white.powder.nn2.us) |
| 13:57.55 | nn | any suggestions on codec for a fairly high latency, high speed link? |
| 13:58.23 | waverly360 | Anyone here had problems with the parking functionality of Asterisk 1.6. Anytime I attempt to transfer a call to the parking extension, asterisk crashes. I'm using Asterisk 1.6.0.3 |
| 13:59.31 | *** part/#asterisk nn (i=A5E649A3@white.powder.nn2.us) |
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| 14:11.58 | pgarcia | Hi everybody, in Asterisk 1.4 is it normal to get a autofallthrough message with status 'UNKNOWN' if I have a System() cmd in my dialplan without another command following it? |
| 14:14.34 | *** join/#asterisk nighty^ (n=nighty@x122091.ppp.asahi-net.or.jp) |
| 14:15.42 | [TK]D-Fender | pgarcia: Yes |
| 14:16.15 | [TK]D-Fender | pgarcia: Run out of priorities without setting "autofallthrough=no" and your call will end |
| 14:17.31 | pgarcia | [TK]D-Fender: ah, ok.. I thought it should be shown only if the System commands itself fails. So it makes sense. Thanks a lot! |
| 14:20.22 | kannan | hello, on a newly provisioned TE122 Digium E1 card, we sometimes get a yellow alarm for a few seconds and then it becomes OK. Any connected calls get disconnected at that time.I am seeing on zttool. The Telecom provider asked us to monitor the link for fluctuation for 24 hours. how to monitor like this , please advise |
| 14:25.39 | beek | kannan: That information should appear in the logs |
| 14:25.57 | kannan | beek , thanks , /var/log/asterisk |
| 14:26.01 | *** join/#asterisk Defraz (n=T0tal@gump.fuzecore.com) |
| 14:26.04 | kannan | ? |
| 14:26.28 | *** join/#asterisk fskrotzki (n=fskrotzk@host198.textwise.com) |
| 14:26.42 | beek | kannan: yes |
| 14:26.54 | kannan | beek , thanks |
| 14:27.16 | beek | kannan: look in there now and you should be able to locate the yellow alarm |
| 14:27.26 | kannan | beek , i will see |
| 14:28.23 | *** join/#asterisk mschangmii (n=besnard@LMontsouris-152-61-20-55.w80-13.abo.wanadoo.fr) |
| 14:28.27 | mschangmii | Hi |
| 14:28.41 | mschangmii | I am looking for a SIP softphone that supports TLS/SRTP |
| 14:29.01 | mschangmii | I remember using one for windows but I can't recall the name of the soft |
| 14:29.28 | [TK]D-Fender | mschangmii: Good odds eyeBeam supports these |
| 14:29.37 | mschangmii | is it free ? |
| 14:30.03 | *** join/#asterisk nicoAMG (i=asgalt@201.203.96.42) |
| 14:30.07 | beek | morning [TK]D-Fender |
| 14:30.16 | [TK]D-Fender | mschangmii: No |
| 14:30.22 | kannan | beek, i am not able to open the file, eben in vim it is too large. What should I grep for? |
| 14:30.36 | beek | kannan: 'Yellow' |
| 14:31.04 | beek | kannan: or 'alarm' |
| 14:31.26 | kannan | beek : thanks again |
| 14:31.48 | waverly360 | So no one has had asterisk 1.6.0 or 1.6.0.3 core dump on them when trying to park a call? |
| 14:32.18 | [TK]D-Fender | waverly360: How are you transferring the call? |
| 14:33.20 | waverly360 | [TK]D-Fender: I've tried using the attended transfer and blind xfer soft keys on a polycom 430, and I've tried using *5 to transfer within asterisk itself. |
| 14:33.31 | waverly360 | I get slightly different results, but both end with asterisk core-dumping |
| 14:33.41 | kannan | beek : yep its right there. Yellow alarm means we can see the pri_net side, but they cannot see us? is that right. Would it likely be an asterisk side issue or a telcom issue? |
| 14:34.21 | waverly360 | [TK]D-Fender: I've been digging through config files, because it may just be a misconfiguration on my part..but it seems odd that even a misconfiguration would cause a crash like that. |
| 14:34.49 | [TK]D-Fender | waverly360: ok, so SIP & features.conf transfers both core... sounds bug-report worthy. Check mantis to see if its already posted. If not, go for it and include your core file |
| 14:35.21 | waverly360 | [TK]D-Fender: k. I'll do that now. Thanks. |
| 14:35.47 | *** join/#asterisk dsp2877 (i=dsp28777@28.227.48.60.cbj02-home.tm.net.my) |
| 14:35.56 | dsp2877 | hi all |
| 14:36.20 | dsp2877 | does anyone know how to get the AlsaMonitor app working |
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| 14:38.46 | beek | kannan: http://www.ciscosystems.com/en/US/tech/tk713/tk628/technologies_tech_note09186a00800f2fa1.shtml#topic2 |
| 14:39.01 | kannan | beek thanks |
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| 14:39.38 | dsp2877 | anyone... |
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| 15:43.07 | Katty | good morning |
| 15:45.23 | Katty | pokes about |
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| 15:48.55 | *** join/#asterisk iEatChildren (n=WaffleMu@asa.redglaze.com) |
| 15:49.47 | iEatChildren | i have a couple phone numbers comming in off a t1. one of them works 100% but the other number just rings and rings and rings and nothing shows up in the CLI. nothing has changed over the weekend and nobody has been in the office. any ideas what i can check? |
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| 15:52.17 | [TK]D-Fender | iEatChildren: What signaling? |
| 15:52.38 | iEatChildren | honestly...im not sure what that means. normally someone else handles our asterisk stuff but hes out of the office for 2 weeks |
| 15:53.53 | iEatChildren | i do see "[Jan 12 09:47:11] WARNING[5321] chan_zap.c: Ignoring signalling" in the log. |
| 15:53.59 | iEatChildren | if that means anything to you |
| 15:54.17 | [TK]D-Fender | iEatChildren: No, pastebin your zapata.conf |
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| 15:54.54 | iEatChildren | ok, one second please |
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| 15:58.10 | iEatChildren | [TK]D-Fender: http://pastebin.ca/1306574 |
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| 16:01.19 | [TK]D-Fender | iEatChildren: There is nothing configured in there. What ver of *, and what interface is this T1 coming in on? |
| 16:02.15 | iEatChildren | do you know why one number would work and not another if nothing is configured in there? |
| 16:02.51 | *** join/#asterisk Sargun (n=Sargun@66.151.148.225) |
| 16:03.38 | [TK]D-Fender | iEatChildren: Perhaps you're set up what DAHDI. We'll see. Please answer the hardware question as to what interface your T1 is connected to |
| 16:04.25 | iEatChildren | can you tell me how to find out? |
| 16:07.36 | iEatChildren | i know its an analog card |
| 16:07.49 | iEatChildren | im not really sure how all of it is hooked together, its off in another office about 3 hours away |
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| 16:10.23 | [TK]D-Fender | iEatChildren: T1 is digital. Why are you telling me its an analog card now? What model? |
| 16:10.43 | [TK]D-Fender | iEatChildren: What ver of *? |
| 16:11.22 | iEatChildren | Asterisk 1.4.21.2 |
| 16:11.43 | iEatChildren | wctdm24xxp is what i have under "module name" in the gui |
| 16:12.41 | [TK]D-Fender | iEatChildren: Oh God... *-GUI? |
| 16:12.50 | khronos | <PROTECTED> |
| 16:12.53 | iEatChildren | lol. sorry |
| 16:13.08 | [TK]D-Fender | iEatChildren: crapTASTIC |
| 16:13.19 | iEatChildren | lol, why do yo usay that? |
| 16:13.22 | jtodd | Does your enterprise (>250 seats) use Asterisk? Let me know. Doing an interview today, and want to get some new names into my discussion. |
| 16:13.23 | [TK]D-Fender | iEatChildren: Ok, its all configured through users.conf |
| 16:13.26 | [TK]D-Fender | ~users.conf |
| 16:13.26 | jbot | users.conf is a flaming pile of sh1t that takes the fine control of several perfectly usable * config files and reduces them to the lowest common denominator and makes your system behave like a "toaster grade" PBX system. |
| 16:13.33 | iEatChildren | i see |
| 16:13.43 | iEatChildren | yeah, he did this weird, used gui for some parts and configs for others |
| 16:13.53 | *** join/#asterisk codefreeze-lap (n=murf@72.21.67.40) |
| 16:13.57 | [TK]D-Fender | iEatChildren: So... go to * CLI and set verbose 10, and PB up a good call, and then a bad one |
| 16:14.07 | iEatChildren | will do |
| 16:14.34 | *** join/#asterisk rue_mohr (n=rue@24.207.122.10) |
| 16:14.42 | rue_mohr | morning all |
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| 16:15.24 | iEatChildren | where do i set verbose at ? |
| 16:15.35 | iEatChildren | i see normally its in logging.conf....does the gui change that too? |
| 16:15.39 | rue_mohr | iEatChildren, you run asterisk -r |
| 16:15.46 | iEatChildren | ok |
| 16:15.53 | rue_mohr | then at the prompt say core set verbose 9999 |
| 16:15.57 | rue_mohr | or whatever |
| 16:16.08 | iEatChildren | ty |
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| 16:18.24 | iEatChildren | whats really strange is i call in....see nothing on the CLI and if another user starts to dial out it connects us |
| 16:18.24 | iEatChildren | otherwise it just rings forever |
| 16:18.24 | iEatChildren | for the bad line that is |
| 16:18.59 | [TK]D-Fender | iEatChildren: please PB what I requested and include confirmation of your verbose level. Additionally do "core set debug 10" |
| 16:19.10 | iEatChildren | its at 10 currentl |
| 16:19.11 | iEatChildren | y |
| 16:19.24 | iEatChildren | localhost*CLI> core set verbose 10 |
| 16:19.25 | iEatChildren | Verbosity was 3 and is now 10 |
| 16:19.31 | [TK]D-Fender | iEatChildren: verbose AND core |
| 16:19.42 | iEatChildren | oh..i see |
| 16:20.31 | rue_mohr | [TK]D-Fender, before I blindly follow these instructions, do you know what sip.ld IS? |
| 16:21.10 | [TK]D-Fender | rue_mohr: combined SIP application for Polycom phones |
| 16:21.13 | *** join/#asterisk neurosys (n=vinix@173.9.159.182) |
| 16:21.19 | rue_mohr | k, then it IS the firmware |
| 16:21.25 | [TK]D-Fender | rue_mohr: Half. |
| 16:21.38 | [TK]D-Fender | rue_mohr: 2 parts, BootROM, and Application. |
| 16:21.40 | rue_mohr | I'm nervous to update this phone |
| 16:21.44 | [TK]D-Fender | rue_mohr: this would be the latter |
| 16:21.48 | rue_mohr | fair enough |
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| 16:22.44 | iEatChildren | [TK]D-Fender: http://pastebin.ca/1306588 - this is the good line that works. nothing shows up for the bad line |
| 16:23.47 | Katty | wibbles |
| 16:24.45 | eppigy | observes |
| 16:27.47 | rue_mohr | [TK]D-Fender, do you know if I can back up what sip.ld is going to clobber? |
| 16:28.48 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 16:33.16 | [TK]D-Fender | iEatChildren: If nothing shows up I'd check the wiring |
| 16:33.31 | iEatChildren | yeah, thats what i was starting to think |
| 16:33.35 | [TK]D-Fender | rue_mohr: You cannot |
| 16:33.42 | iEatChildren | i dont know much but i know something should show up |
| 16:33.59 | [TK]D-Fender | rue_mohr: Make sure you have a proper set of configs to go with the firmware you are about to install |
| 16:34.30 | eppigy | i dont know much |
| 16:34.34 | eppigy | but i know i love you |
| 16:34.42 | eppigy | that may be |
| 16:34.45 | eppigy | all i need to know |
| 16:34.46 | [TK]D-Fender | eppigy: You had us... at the former ;) |
| 16:34.53 | eppigy | 8[] |
| 16:35.52 | rue_mohr | [TK]D-Fender, same archive |
| 16:38.35 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
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| 16:46.30 | [TK]D-Fender | rue_mohr: Oh? What version were you on before? For what model phone? What ver are you now putting on? |
| 16:46.31 | *** part/#asterisk Lunks (i=sbnc@pedro.nascimento.co.uk) |
| 16:46.44 | rue_mohr | ip601 I have no idea what version |
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| 16:48.17 | *** join/#asterisk FinboySlick (n=FinboySl@207.134.8.4) |
| 16:48.29 | [TK]D-Fender | rue_mohr: Amazing how you can assure "the same" |
| 16:48.46 | [TK]D-Fender | rue_mohr: You are advised to figure out what the heck you're doing |
| 16:49.02 | rue_mohr | what did you think I been doing? |
| 16:49.04 | FinboySlick | Normally, I should just put "wctdm24xxp" in my /etc/dahdi/modules file to have that loaded when the service starts, right? |
| 16:49.19 | rue_mohr | not like you can just go to school for this |
| 16:49.51 | [TK]D-Fender | rue_mohr: Not knowing what firmware you have on your phone or what you're about to install ISN'T bright, and it IS all documented in the admin & user guides |
| 16:49.52 | FinboySlick | rue_mohr: Hehe, with [TK]D-Fender you can get shcooled allright ;) |
| 16:50.21 | [TK]D-Fender | rue_mohr: Oh.. and you CAN go to "school" on this. |
| 16:50.25 | rue_mohr | I'm only updating the firmware cause I was told to by you lot |
| 16:50.57 | rue_mohr | [TK]D-Fender, polycom school? |
| 16:51.40 | [TK]D-Fender | rue_mohr: Well what are you putting in? What does your phone say it has on it now? |
| 16:52.01 | [TK]D-Fender | rue_mohr: Are you starting from scratch from this firmware pack? Completely fresh folder? |
| 16:52.19 | rue_mohr | I'm just going over the ftp log, seems the sip.ld was too big for the tftp server, so I might still be ok |
| 16:52.38 | [TK]D-Fender | rue_mohr: TFTP = ass. FTP = easier and more predictable |
| 16:52.42 | rue_mohr | [TK]D-Fender, yea, as instructed I tossed all the config files I'd made so far |
| 16:52.48 | *** join/#asterisk af_ (n=getsmart@88-149-240-27.dynamic.ngi.it) |
| 16:53.01 | rue_mohr | and am using the ones from polycom |
| 16:53.17 | [TK]D-Fender | rue_mohr: thats a start. So what VERSION did you go get? |
| 16:53.25 | rue_mohr | the newest... |
| 16:53.46 | [TK]D-Fender | rue_mohr: Try answering with a number... its lets us know you have at least a slight idea what you're doing... |
| 16:53.49 | rue_mohr | spip_ssip_3_1_1_release_sig.zip |
| 16:53.56 | [TK]D-Fender | rue_mohr: These days we'll settle for minor illusions |
| 16:54.19 | WHYS | thinks her ass is predictable |
| 16:54.38 | [TK]D-Fender | rue_mohr: Get 3.1.1.B |
| 16:54.54 | rue_mohr | B = beta? |
| 16:55.08 | rue_mohr | and that was the newest I was able to find on their page |
| 16:55.09 | [TK]D-Fender | rue_mohr: No, minor rev |
| 16:55.18 | [TK]D-Fender | rue_mohr: And they have newer |
| 16:55.23 | rue_mohr | iirc it was actaully the one I was pointed at |
| 16:55.24 | [TK]D-Fender | rue_mohr: http://downloads.polycom.com/voice/voip/sip_sw_releases_matrix.html |
| 16:56.14 | rue_mohr | "The release content for phones other than the IP 7000 is identical to that contained in release SIP 3.1.1" |
| 16:56.27 | rue_mohr | I'm using the 601, is that a test? |
| 16:57.28 | rue_mohr | every half hour someone walks into this room and asks me if the new phone system is working yet, and you know what they ask after I say no? |
| 16:57.35 | rue_mohr | -WHY- |
| 16:57.49 | rue_mohr | anyhow |
| 17:00.49 | [TK]D-Fender | rue_mohr: My advise is on the level, always has been. This version may not mack much of a differnce, or heck any, but I like knowing what goes into my setups. |
| 17:00.57 | [TK]D-Fender | rue_mohr: You can go ahead with what you've got now. |
| 17:01.04 | [TK]D-Fender | rue_mohr: So wahts on the phone NOW? |
| 17:01.34 | rue_mohr | just a sec I'll go interrigate it |
| 17:04.06 | *** join/#asterisk jameswf (n=james@unaffiliated/jameswf-home) |
| 17:04.36 | *** join/#asterisk erth64net (n=erth64ne@69-30-67-191.dq1sn.easystreet.com) |
| 17:07.28 | rue_mohr | sorry for the delay, interrogated again |
| 17:07.58 | rue_mohr | k, sip V 2.1.2.0078 polyDSP V 1.3.7.0007 bootrom 3.2.3.0002 |
| 17:08.19 | rue_mohr | so this will be a big sip upgrade for it |
| 17:09.11 | rue_mohr | k, I need to check that the aastra phone is ok with ftp before I redo all that |
| 17:09.32 | *** join/#asterisk vgster (n=vgster@94-194-190-189.zone8.bethere.co.uk) |
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| 17:14.42 | FinboySlick | Anybody here uses dahdi? I'm trying to figure out the 'proper' way to have its modules loaded at boot. My understanding is that 'dahdi_cfg' takes care of loading what's in /etc/dahdi/modules. Am I wrong? |
| 17:14.53 | *** join/#asterisk wastrel (n=wastrel@nylug/member/wastrel) |
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| 17:18.07 | rue_mohr | I do |
| 17:18.23 | rue_mohr | uh how does mine work |
| 17:19.33 | rue_mohr | mine is loaded by the etc/init.d/dahdi that was installed by the sources 'build install' |
| 17:19.46 | rue_mohr | might have been part of dahdi-tools |
| 17:19.47 | *** join/#asterisk riksta (n=rick@office.encompassmedia.co.uk) |
| 17:19.58 | rue_mohr | or utils, whatever its called |
| 17:20.09 | rue_mohr | FinboySlick, got that? |
| 17:21.36 | riksta | Is there a way to play a message onto the channel, and then continue to dial another party, but have the CDR not flag as answered or increment the duration until the call has got through to the channel dialled after the playback? |
| 17:21.42 | *** join/#asterisk Supaplex (n=supaplex@166.70.62.193) |
| 17:22.04 | rue_mohr | can the message be hold music? |
| 17:23.39 | Supaplex | how do I tell gdb to not pagenate? set pagenation off has nil effect for a --batch -ex "set pagenation off" -ex "thread apply all bt full" |
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| 17:26.09 | wastrel | hi, question about IVR prompt playback, we've got long delays between files - 5 to 10 even 15 seconds users are reporting |
| 17:26.26 | wastrel | any pointers what to look at, look for? |
| 17:26.38 | wastrel | i don't think it's a dialplan configuration issue... maybe system load? |
| 17:28.21 | dsp2877 | what files are those |
| 17:28.27 | dsp2877 | maybe the files have silence in them or something? |
| 17:28.33 | rue_mohr | well I didn't want to use an ftp server for having to deal with accounts and all but a quick google looks like atftpd is known to have these problems |
| 17:29.01 | [TK]D-Fender | rue_mohr: what problems? |
| 17:29.04 | dsp2877 | maybe paste the dialplan or something it will help |
| 17:29.34 | rue_mohr | [TK]D-Fender, atftpd apparently has block size limits that cause problems with ip phones, I see a few instances of it |
| 17:29.43 | [TK]D-Fender | rue_mohr: vsftpd <- |
| 17:30.03 | rue_mohr | I wanted to stay away from having to deal with security |
| 17:30.07 | rue_mohr | oh well |
| 17:30.09 | *** part/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 17:30.45 | wastrel | dsp2877: they're the stock ivr sound files |
| 17:31.08 | rue_mohr | wastrel, if you play them on you pc in succession are there delays? |
| 17:31.26 | fogo | rue_mohr: iirc, I ran into that once too - atftpd exibited the problem, but tftp worked fine (debian packages) |
| 17:31.32 | *** join/#asterisk matt_ (n=matt@mattspc.ipv6.mattstone.net) |
| 17:32.02 | kannan | hello, we set the configs for cisco 7960 phones on one tftp server. Now the files reside on the tftpserver in * box. The old settimngs are fine, the phones register, but i am not able to modify the SIP/name on any phone now. Any ideas , greatly appreciated |
| 17:32.23 | *** join/#asterisk Micc (n=Michael@c-76-121-255-52.hsd1.wa.comcast.net) |
| 17:32.35 | Micc | why do I keep getting sox: invalid parameter -m ? |
| 17:33.05 | rue_mohr | fogo, I had a problem with tftp |
| 17:33.40 | fogo | rue_mohr: I could be wrong. I just remember one worked, one didn't (atftpd and tftpd) |
| 17:34.44 | *** join/#asterisk theHub (n=theHub@69.177.93.21) |
| 17:36.29 | wastrel | rue_mohr: i haven't tried, they're the standard /var/lib/asterisk/sounds soundfiles |
| 17:36.42 | wastrel | rue_mohr: the problem comes and goes so i don't think it's intrinsic to the actual files |
| 17:37.08 | wastrel | ls |
| 17:37.12 | [TK]D-Fender | wastrel: pastebint he CLI output of a flakey call. |
| 17:37.13 | wastrel | wrong window :] |
| 17:37.14 | [TK]D-Fender | ~pb |
| 17:37.15 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste, or , http://bin.cakephp.org/ |
| 17:37.17 | [TK]D-Fender | ^^^^^^ |
| 17:38.18 | FinboySlick | rue_mohr: Yeah, I have the /etc/init.d/dahdi script. It just doesn't load any modules when it starts. |
| 17:39.10 | wastrel | [TK]D-Fender: from /var/log/asterisk/messages ? |
| 17:39.22 | rue_mohr | FinboySlick, what runlevel you in? |
| 17:39.23 | rue_mohr | 2? |
| 17:39.54 | wastrel | ah in the asterisk console ok |
| 17:40.30 | rue_mohr | fogo, may I ask what you ended up with? |
| 17:40.43 | *** join/#asterisk CunningPike (n=arodgers@204.239.10.119) |
| 17:41.05 | [TK]D-Fender | wastrel: No, CLI <- |
| 17:41.11 | fogo | rue_mohr: I don't remember. I think it was tftpd though |
| 17:41.48 | wastrel | http://rafb.net/p/Sgq0Fw32.html ? |
| 17:43.01 | [TK]D-Fender | wastrel: And you say there are several seconds between each file? |
| 17:43.23 | [TK]D-Fender | wastrel: What is that device? Describe the networking between it and * |
| 17:45.51 | wastrel | we're voip so all the clients are talking over the switched LAN here |
| 17:46.10 | wastrel | gigabit upstairs but we're 10/100 down here. |
| 17:46.21 | *** join/#asterisk pikachu2000 (n=pikachu2@196-209-199-207-rrdg-esr-2.dynamic.isadsl.co.za) |
| 17:46.29 | *** part/#asterisk pikachu2000 (n=pikachu2@196-209-199-207-rrdg-esr-2.dynamic.isadsl.co.za) |
| 17:46.48 | FinboySlick | rue_mohr: According to my distro, it's 'default'. |
| 17:47.09 | FinboySlick | rue_mohr: Doesn't work when I start it manually either. |
| 17:47.42 | wastrel | it's on a dell poweredge 1950 8gb RAM, xenon processor i think 2ghz |
| 17:47.42 | FinboySlick | rue_mohr: Says dahdi starts okay, but it doesn't load up any modules. |
| 17:47.53 | *** part/#asterisk Supaplex (n=supaplex@166.70.62.193) |
| 17:47.57 | dsp2877 | wastrel: i had that same issue before using a dell poweredge , is this the first time you are using it? |
| 17:48.01 | lowtek | FinboySlick: Can you pastebin your lspci output? |
| 17:48.15 | wastrel | dsp2877: weve been on this about 6 months now |
| 17:48.18 | FinboySlick | lowtek: Sure. |
| 17:48.35 | dsp2877 | wastrel: so its been like this for last 6 months? |
| 17:48.39 | wastrel | dsp2877: i've had reports about this prob for a month or so but am just now sitting down to look at it. |
| 17:48.47 | [TK]D-Fender | wastrel: what is that DEVICE? |
| 17:48.49 | FinboySlick | lowtek: Need it detailed, or do you just want to see: 01:07.0 Ethernet controller: Digium, Inc. Wildcard TDM800P 8-port analog card (rev 11) ? |
| 17:48.54 | wastrel | dsp2877: it used to be better, and it comes and goes. |
| 17:49.01 | lowtek | yea, that's what I wanted to see. |
| 17:49.01 | dsp2877 | ok |
| 17:49.08 | wastrel | [TK]D-Fender: device you mean the terminal? |
| 17:49.31 | lowtek | FinboySlick: I'm still using Zaptel, but can you load ztdummy? |
| 17:49.34 | dsp2877 | wastrel : i had similar/exact issue to this, spent months on it, only to find out that it was due to the Raid configuration of the dell poweredge |
| 17:49.41 | [TK]D-Fender | wastrel: yes |
| 17:49.42 | lowtek | FinboySlick: Or whatever the dahdi equivilent is. |
| 17:50.13 | wastrel | [TK]D-Fender: cisco 7960 for me but we have 2 line models i forget the serial 7940 maybe for most people |
| 17:50.31 | FinboySlick | lowtek: I can load all my modules just fine, it's just not done automatically by /etc/init.d/dahdi. I could have them loaded in the standard way for my distro... I just wanted to do it the dahdi way. |
| 17:50.36 | wastrel | dsp2877: i'll look into that thanks |
| 17:51.03 | [TK]D-Fender | wastrel: Ok, should almost certainly be a server-sire load issue of some kind against the HD... |
| 17:51.11 | dsp2877 | wastrel: i changed something in the Bios of the 1950 or reconfigured the raid mode etc and it worked fine |
| 17:51.19 | lowtek | FinboySlick: Ahh. Dunno, I would lean towards the OS way. What distro? |
| 17:51.30 | FinboySlick | lowtek: Gentoo. |
| 17:51.49 | lowtek | FinboySlick: Good luck! ;) |
| 17:52.16 | FinboySlick | lowtek: Hah, I'm pretty comfy with it, really. |
| 17:52.56 | FinboySlick | lowtek: I was just wondering if /etc/init.d/dahdi *should* load the modules itself or if I was misreading stuff. |
| 17:53.59 | lowtek | FinboySlick: I don't know enough about Gentoo to help, sorry. Use Debian or Redhat! :) |
| 17:54.26 | lowtek | FinboySlick: There could be some prerequisites not being loaded before hand, first though. |
| 17:54.30 | lowtek | s/though/thought |
| 17:54.47 | FinboySlick | I was 'brought up' on Debian, then wickedly forced to ruse RedHat... But my first distro was FreeBSD. |
| 17:55.14 | FinboySlick | lowtek: /etc/init.d/dahdi would complain about it then. |
| 17:55.27 | lowtek | FinboySlick: FreeBSD rocks. We use debian 4.x, rock solid. I'm sure Red Hat is equally rock solid. |
| 17:55.42 | lowtek | FinboySlick: FreeBSD and * however, not so much. |
| 17:57.15 | FinboySlick | lowtek: Yeah. The BSDs are great if your stuff is supported on it. But it's not as much fun as linux when it comes to funky new stuff. Gentoo would be entirely masochistic if I maintained more than a handful of boxes. But it keeps your skills sharp when you break things all the time ;) |
| 17:58.15 | *** join/#asterisk nexu (n=nexu@86.85.169.45) |
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| 17:59.47 | *** mode/#asterisk [+o angler] by irc.freenode.net |
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| 18:00.45 | *** mode/#asterisk [+o angler] by ChanServ |
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| 18:08.09 | *** join/#asterisk bbryant (n=Brett_Br@adsl-068-016-200-248.sip.chs.bellsouth.net) |
| 18:08.16 | *** join/#asterisk pulu (i=pulu@wsip-72-215-205-146.ph.ph.cox.net) |
| 18:10.40 | *** join/#asterisk Pazzo (n=ugelt@reserved-225136.rol.raiffeisen.net) |
| 18:16.47 | kannan | For cisco 7960 phones , regarding the tftp files, i had edited the SIP<mac>.cnf files with vi. do we need to use only put command. Also chkconfig --list show "off" how to start tftp server ? i am on centos. |
| 18:18.27 | *** join/#asterisk joshaidan (n=joshaida@S01060090f8009fa6.tb.shawcable.net) |
| 18:19.42 | carrar | If you installed out of a RPM chkconfig --level 2345 tftpd on |
| 18:19.56 | carrar | then service tftpd start |
| 18:22.08 | carrar | or you can start it manually in /etc/rc.local |
| 18:22.20 | carrar | using /usr/sbin/in.tftpd -vv -l -s -u ftp /tftpboot |
| 18:23.06 | carrar | best consult your unix admin for how it should be done |
| 18:23.22 | kannan | carrar , thanks |
| 18:23.59 | kannan | i am supposed to be the admin, but i havent done this ever before , hehe |
| 18:27.04 | JayTee52 | damn, downloading from the Lumenvox repos is like watching paint dry |
| 18:30.16 | *** join/#asterisk dieguito84 (n=diego@host248-192-dynamic.10-87-r.retail.telecomitalia.it) |
| 18:38.39 | kannan | carrar, do we need to back up * sources and configs ? does tftp have anything to do with that? |
| 18:38.50 | kannan | before i try to start the tftp server |
| 18:44.24 | *** join/#asterisk mikkel (n=mikkel@84-238-113-66.u.parknet.dk) |
| 18:47.46 | carrar | kannan, yes, you will need to backup your own configs using a whatever you want to back them up with |
| 18:48.08 | carrar | and no, tftp has nothing to do with backups |
| 18:48.51 | carrar | Lots of backup software out there |
| 18:51.02 | kannan | carrar, thanks |
| 18:57.37 | *** join/#asterisk Avelino (n=Avelino@mail.paterno.com.br) |
| 19:01.32 | Kobaz | <PROTECTED> |
| 19:01.32 | Kobaz | <PROTECTED> |
| 19:01.40 | Kobaz | can i disable that showing up in the console? |
| 19:01.53 | *** join/#asterisk csjp (n=csjp@wnpgmb1307w-ad02-39-179.dynamic.mts.net) |
| 19:01.55 | Kobaz | i changed manager.conf displayconnects=no |
| 19:01.58 | Kobaz | but that didn't do it |
| 19:02.31 | csjp | is it possible to connect a SIP client to a asterisk sip server on a different layer3 network? when I try it looks like the asterisk server is ignoring the requests |
| 19:02.45 | csjp | but sip client is on the same layer3 network there are no problems |
| 19:02.51 | csjp | there is no firewall in place |
| 19:04.18 | *** join/#asterisk Avelino (n=Avelino@mail.paterno.com.br) |
| 19:05.50 | jaytee | Kobaz, you might need to do a full restart after editing manager.conf |
| 19:06.35 | *** join/#asterisk Chris-NB (n=chris@home.fuerstaller.com) |
| 19:08.27 | Kobaz | yeah, i did |
| 19:09.51 | citats | Kobaz: i dont think there is a way to disable it, but it'd be pretty easy to modify the source to do that |
| 19:10.35 | Kobaz | mm |
| 19:10.36 | Kobaz | yeah |
| 19:10.56 | Kobaz | i have some scripts that connect pretty frequently, so it's kinda now an unhelpful message |
| 19:11.03 | justdave | my main asterisk box just kernel panicked |
| 19:11.09 | justdave | the traceback is in dahdi code |
| 19:11.13 | Kobaz | fun |
| 19:11.13 | *** join/#asterisk Defraz (n=T0tal@fw-poky.fuzecore.com) |
| 19:11.28 | Kobaz | don't run the bleeding edge on production boxes, is what i always say |
| 19:11.38 | justdave | I'm not |
| 19:11.43 | justdave | this is a stable release |
| 19:11.47 | Kobaz | dandi is brand new |
| 19:11.51 | justdave | (so called anyway) |
| 19:12.03 | justdave | it's been out for a while |
| 19:12.08 | Kobaz | 'a while' |
| 19:12.09 | justdave | I stalled switching to it for months |
| 19:12.10 | Kobaz | it's still new |
| 19:13.46 | justdave | unfortunately it killed things enough that it didn't log the traceback, only showed it on console |
| 19:13.54 | justdave | and the important part was scrolled off the top already |
| 19:15.02 | *** join/#asterisk DigitalIrony (n=eric@nat/digium/x-8be61a95d1787162) |
| 19:16.33 | *** join/#asterisk Remenic (n=Richard@cc1222307-a.frane1.fr.home.nl) |
| 19:16.34 | Remenic | hi |
| 19:16.46 | Remenic | anyone here know anything about getting astribank pri devices working using dahdi? |
| 19:17.22 | jaytee | file help? |
| 19:17.52 | *** join/#asterisk dandate (n=dandate2@adsl-99-55-169-103.dsl.pltn13.sbcglobal.net) |
| 19:18.00 | dandate | hey guys i'm having the toughest time, i can only connect to my * box web GUI by demilitiring the * box, port forwarding is not fuctioning correctly |
| 19:18.04 | file | jaytee: moo? |
| 19:18.25 | jaytee | getting this on my new server: WARNING[5814]: loader.c:363 load_dynamic_module: Error loading module 'res_speech_lumenvox.so': liblv_lvspeechport.so: cannot open shared object file: No such file or directory |
| 19:19.04 | file | jaytee: the lumenvox libraries are not being found by the loader... ld.so.conf |
| 19:19.13 | jaytee | res_speech_lumenvox.so is in modules.conf right after res_speech.so and liblv_speechport.so is in /usr/lib |
| 19:19.24 | file | are you sure? |
| 19:19.38 | file | cause there are tons of libraries that lumenvox has |
| 19:20.00 | file | ldd /usr/lib/asterisk/modules/res_speech_lumenvox.so |
| 19:20.04 | file | will tell you what is not found |
| 19:20.12 | *** join/#asterisk profXavier (n=jezus@unaffiliated/neverblue) |
| 19:20.39 | *** join/#asterisk vgster (n=vgster@94-194-190-189.zone8.bethere.co.uk) |
| 19:20.49 | profXavier | I have the following issue showing up in *: 'Detected alarm on channel 9: Red Alarm', could this be resultant of bad hardware? or is it a known issue with *? |
| 19:21.10 | carrar | means your T1 is down |
| 19:21.19 | carrar | possibly |
| 19:21.50 | jaytee | file, ldd /usr/lib/asterisk/modules/res_speech_lumenvox.so |
| 19:21.50 | jaytee | <PROTECTED> |
| 19:21.50 | jaytee | <PROTECTED> |
| 19:21.50 | jaytee | <PROTECTED> |
| 19:21.50 | jaytee | <PROTECTED> |
| 19:21.57 | jaytee | oops, sorry for the flood |
| 19:22.05 | jaytee | thought it would only be 2 lines |
| 19:22.17 | file | right... can't find that library |
| 19:22.21 | dandate | well u jsut made a fat flood |
| 19:22.25 | dandate | i';m trying to read!! |
| 19:23.05 | csjp | is it possible to connect a SIP client to a asterisk sip server on a different layer3 network? when I try it looks like the asterisk server is ignoring the requests |
| 19:23.25 | carrar | csjp, make sure you have a route to it |
| 19:23.32 | carrar | can you ping it? |
| 19:23.46 | csjp | carrar: yes, I can ssh into the asterisk server from the client |
| 19:23.58 | carrar | is it behind nat? |
| 19:24.02 | csjp | nope |
| 19:24.02 | jaytee | file, I just double checked and you're right, the liblv_lvspeechport.so isn't there. there are a bunch of other liblv_lv* files but that one is missing. |
| 19:24.26 | csjp | carrar: when I run tcpdump on the SIP server, I see the requests come in but nothing else |
| 19:24.37 | carrar | any IP filtering? |
| 19:24.43 | *** join/#asterisk dandate2 (n=dandate2@c-71-202-125-220.hsd1.ca.comcast.net) |
| 19:24.50 | csjp | nope |
| 19:25.01 | carrar | anything in asterisk logs? |
| 19:25.04 | carrar | or consol |
| 19:25.21 | csjp | nope, is there a way I can crank up the verbosity outside of -v ? |
| 19:25.32 | carrar | yeah |
| 19:26.03 | carrar | set verbose |
| 19:26.17 | carrar | core set verbose |
| 19:26.46 | carrar | would double check your sip.conf settings and client settings |
| 19:27.46 | dandate2 | i'm trying to configure my tcp port to 80 so I can access the web gui without demilitarizing, am i supposed to enter a trigger port of 80 and a public port of 80 tcp? |
| 19:28.45 | sah-work | hello, |
| 19:29.36 | sah-work | anyone have any issues with pcom phones and tftp blksize? i am upgrading to 3.1.1 and keep getting the application is not present error (sip.ld) and the logs show that it is complaining about blksize however i did increase it on the server. |
| 19:30.48 | profXavier | carrar, was those two lines, T1 being down, directed at me ? |
| 19:31.15 | carrar | Well if your T1 is down you will get that same error for all the T1 channels |
| 19:31.36 | profXavier | so its possibly on the providers end ? |
| 19:31.41 | ajohnson | Anyone here have much experience using the manager interface? |
| 19:32.16 | ajohnson | I'm trying to find a way to get a list of channels that doesn't end up truncating the channel name |
| 19:32.28 | carrar | profXavier, can have them test the T1 |
| 19:32.30 | ajohnson | Show channels shows channels, but quite frequently truncates part of the channel name |
| 19:32.32 | carrar | rule them out |
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| 19:33.39 | dandate2 | ahh i didn't set port mapping |
| 19:34.33 | *** join/#asterisk dan__t (n=dant@ip70-162-189-250.ph.ph.cox.net) |
| 19:34.34 | dan__t | Hello. |
| 19:35.25 | ajohnson | nm, status does what I need |
| 19:35.38 | csjp | carrar: ok |
| 19:35.44 | csjp | carrar: something strange is going on here |
| 19:35.59 | csjp | carrar: basically asterisk is responding but it's responding to some other IP address |
| 19:36.04 | dan__t | I haven't been able to find a function or an argument to use for this, although I'm still kind of new to Asterisk.... what would I be looking for if I wanted to have an active channel, and when the remote party disconnected, * came back on with a trigger of some sort |
| 19:36.14 | dan__t | I'd like to interject a post-call dialplan extension |
| 19:36.19 | *** join/#asterisk tobias (n=tobias@cpe-069-134-127-101.nc.res.rr.com) |
| 19:36.23 | dan__t | I don't know how to catch that, though. |
| 19:36.44 | carrar | csjp, yeah having two devices with the same registration is not going to work well |
| 19:37.25 | carrar | whatever last device to register will get the calls |
| 19:37.34 | [TK]D-Fender | dan__t: "h" Asterisk Standard Extension, or the "g" Dial option. |
| 19:37.37 | csjp | carrar: no devices are registered right now |
| 19:37.37 | [TK]D-Fender | dan__t: read up |
| 19:37.45 | dan__t | Will do, thanks, just needed a starting point :) |
| 19:38.19 | dan__t | I can't say enough about this Asterisk book. Second edition, of course. |
| 19:38.24 | dan__t | (just wanted to plug it one more time) |
| 19:39.38 | [TK]D-Fender | dan__t: Yeah, it has this nasty anal leakage issue.... |
| 19:39.39 | *** join/#asterisk voxter (n=voxter@76.77.95.2) |
| 19:39.44 | [TK]D-Fender | :D |
| 19:39.54 | dan__t | Maybe the 1st edition did. |
| 19:39.55 | dan__t | heh |
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| 19:42.29 | *** join/#asterisk mascool (n=george@71.238.125.243) |
| 19:42.57 | mascool | is there any way to do a dialplan show macro-blahblah and pass the parameters? |
| 19:43.12 | mascool | like dialplan show macro-blahblah(par1,par2) ? |
| 19:43.32 | mascool | and trace its behavior ? |
| 19:45.33 | [TK]D-Fender | mascool: there is no way to trace behavior short of placing an actual call to it |
| 19:45.46 | [TK]D-Fender | mascool: No "debug", no "test". Doing is testing |
| 19:46.35 | mascool | [TK]D-Fender, well it works when you do dialplan show extension@context |
| 19:46.45 | [TK]D-Fender | mascool: taht doesn't simulate anything |
| 19:46.46 | mascool | it's showing you which path it;s going to take |
| 19:47.00 | [TK]D-Fender | mascool: so the concept of testing parms doesn't apply there either |
| 19:47.01 | mascool | which is what i want to see but in a macro |
| 19:47.09 | mascool | well the number@ is the parameter |
| 19:47.13 | [TK]D-Fender | mascool: Macro = dialplan. |
| 19:47.20 | [TK]D-Fender | mascool: And you can dump that like any other context |
| 19:47.46 | [TK]D-Fender | mascool: But it won't emulate anything |
| 19:48.08 | mascool | i don't know if emulate is the right wording but if i do dialplan show number@context |
| 19:48.14 | [TK]D-Fender | mascool: And nothing you can't see just by starting at your raw dialplan anyways unless you had some kind of syntax error that rpevented a line from loading |
| 19:48.18 | dan__t | Awesome, so I can have the call grab the 'h' extension right when the remote party hangs up, and start all over again with a new dialplan huh |
| 19:48.26 | mascool | it will show me what exten it will match |
| 19:48.31 | [TK]D-Fender | mascool: "dialplan show macro-blah" |
| 19:48.54 | [TK]D-Fender | dan__t: Something like that. Read up & test... dead channel have many limitations including access to vars, etc |
| 19:48.55 | mascool | [TK]D-Fender, yes, that works, but the macro also has params which i would like to pass to dialplan show macro |
| 19:49.20 | [TK]D-Fender | mascool: those are interpreted in runtime jsut like any other dialplan. there is nothing for it to interpret |
| 19:49.30 | dan__t | Understood. |
| 19:49.53 | dan__t | I can't find an actual definition for it from an official source, just reading mail lists and such. DO you have a source for such references? |
| 19:50.20 | mascool | hmm ok |
| 19:50.28 | mascool | i understand what you're saying |
| 19:52.13 | [TK]D-Fender | dan__t: book should give some details, for the rest, jsut WIKI that up as I worded it |
| 19:53.01 | dan__t | Didn't find either in the book. Its entirely possible I'm looking for the wrong thing heh |
| 19:53.19 | [TK]D-Fender | dan__t: and the book is missing plenty of stuff... |
| 19:53.32 | dan__t | Its a good start, regardless. |
| 19:53.35 | [TK]D-Fender | dan__t: If I was contracted for it I'd re-write large chunks of it |
| 19:54.20 | dan__t | haha. |
| 19:54.48 | dan__t | So, you said vars are destroyed. That leads me to another question. Is there a function that will debug and dump all vars currently present at position X, Y, Z, etc etc? |
| 19:55.26 | [TK]D-Fender | dan__t: I can't comment too much on the particulars. you'll have to read up and test for that |
| 19:56.09 | dan__t | Yes, that's why I'm asking if there's a way for me to dump all vars to compare that against a normal session, to see what's missing. |
| 19:56.16 | dan__t | I'm going in to this blind, i don't know which vars I'll want. |
| 19:58.18 | dan__t | So, in general, is there a function to dump to debug, any and all variables that are available at any given point in the dialplan? |
| 19:59.01 | *** join/#asterisk dandate (n=dandate2@adsl-99-55-169-103.dsl.pltn13.sbcglobal.net) |
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| 20:00.39 | *** join/#asterisk jicksta (n=jicksta@c-24-6-87-187.hsd1.ca.comcast.net) |
| 20:04.59 | profXavier | carrar I did have them test the lines, and they said 'garbage' was coming from our end |
| 20:05.06 | rgsteele||work | Any recommendations for a good zaptel card? Building a new asterisk box, which is currently using a TDM400P |
| 20:05.07 | profXavier | but then our service improved |
| 20:05.18 | profXavier | still having drops on calls though |
| 20:07.08 | carrar | profXavier, plug in a loopback plug and have them test again |
| 20:07.18 | carrar | plug it in at the NIU |
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| 20:16.42 | dandate2 | <PROTECTED> |
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| 20:27.28 | dan__t | hrm, is teliax still any good as an all-purpose IAX provider? |
| 20:28.17 | *** join/#asterisk DarylVOIP (n=daryl@75.147.121.177) |
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| 20:28.21 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
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| 20:32.14 | rgsteele||work | Hum... doesn't even look like Digium markets the TDM400P's anymore on their site. |
| 20:32.29 | [TK]D-Fender | rgsteele||work: Replaces by the TDM410P |
| 20:33.56 | eppigy | TRABAJO |
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| 20:35.01 | *** join/#asterisk jgoo (n=jgoo@194.219.217.192) |
| 20:35.22 | jgoo | Anyone have any review of OpenPhone 52 / 71 / 73 and 75's ? |
| 20:35.37 | jgoo | Versus polycom |
| 20:36.33 | *** join/#asterisk farkus_ (i=chatzill@cpe-98-14-94-76.nyc.res.rr.com) |
| 20:39.57 | profXavier | anyone know where I can get a te120p, to be delivered in Canada? |
| 20:41.41 | *** join/#asterisk Dibbler (n=Dibbler@87-194-103-72.bethere.co.uk) |
| 20:41.42 | jaytee | profXavier, www.telephonydepot.com |
| 20:43.59 | rgsteele||work | [TK]D-Fender: Ah, okay. I think the TDM404E is all I need (or it's PCIe counterpart) |
| 20:46.55 | profXavier | thanks JT |
| 20:55.28 | *** join/#asterisk mcargile (n=mikec@rrcs-24-173-156-170.se.biz.rr.com) |
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| 20:59.09 | dandate2 | can i only use .wav files for my IVR announcement? |
| 21:04.05 | *** part/#asterisk mcargile (n=mikec@rrcs-24-173-156-170.se.biz.rr.com) |
| 21:04.55 | [TK]D-Fender | dandate2: Any format that * supports |
| 21:05.05 | [TK]D-Fender | dandate2: "core show modules like format" |
| 21:05.39 | dandate2 | ok from the CLI ? |
| 21:05.46 | *** join/#asterisk cesar_CR (n=cesar@200.91.75.67) |
| 21:06.04 | [TK]D-Fender | dandate2: Clearly |
| 21:06.38 | dandate2 | ok how do i disable debugging first |
| 21:06.59 | dandate2 | i did a sip set debug and i can't turn it off |
| 21:07.55 | [TK]D-Fender | dandate2: "sip set debug off |
| 21:09.03 | dandate2 | oh ok |
| 21:10.41 | denon | has anyone seen cisco 7960 hangup buttons act up? kinda have to mess with the button to get it to pick up |
| 21:10.56 | denon | (and of course, inevitably it'll also hang up again if you mess with it for an instant too long) |
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| 21:13.04 | dandate2 | could anyone tell me if there is a problem with my incoming and outgoing trunk settings? http://www.pastebucket.net/0nvr1t |
| 21:13.28 | dandate2 | i am simply unable to receive DID calls |
| 21:17.11 | dandate2 | its configured for didforsale.com their recommended settings are found here http://www.didforsale.com/blog/ |
| 21:17.58 | dandate2 | and i spoke with their tech rep about being behind a router, what he told me was this |
| 21:17.58 | dandate2 | You can use Free PBX GUI, that should work just fine. Only thing is if you are behind NAT/Router you will need to define enternip in sip.conf file. Other than that you are all good. |
| 21:19.34 | dandate2 | they asked me later to ad this code |
| 21:19.36 | dandate2 | http://www.pastebucket.net/vj2oxv |
| 21:25.54 | dandate2 | my trunk set up is a bunch of sphagetti code provided to me through IRC and the provider |
| 21:30.09 | [TK]D-Fender | danMany spelling mistakes, and you put stuff in peer entries that belong only under [general] |
| 21:30.12 | [TK]D-Fender | dandate2: read the guide : |
| 21:30.14 | [TK]D-Fender | ~sipt |
| 21:30.17 | [TK]D-Fender | ~sipnat |
| 21:30.18 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 21:30.19 | [TK]D-Fender | ^^^^^^^^^^ |
| 21:30.20 | [TK]D-Fender | BBIAb |
| 21:30.35 | *** join/#asterisk reneger (n=reneger@dslb-088-078-115-034.pools.arcor-ip.net) |
| 21:37.11 | phix | I finally got me a TDM400p |
| 21:37.26 | phix | I can now throw this linksys sipura in the bin |
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| 21:38.45 | dan__t | Hrm.... can I have the same extension under two different contexts? |
| 21:38.47 | dan__t | Does it work that way? |
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| 21:53.02 | EI5GTB | hi guys, about to build oslec into asterisk 1.4.22 + dahdi. There are no known incompatibilities? it works just the same as zap? |
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| 22:17.42 | jshriver | greetings |
| 22:17.53 | jshriver | anyone here use astlinux? which should I use |
| 22:18.08 | jshriver | 0.6.2 has geni net4801 net5501 via, etc.. but no readme file to explain them |
| 22:18.12 | dan__t | So when using the 's' |
| 22:18.14 | dan__t | uh. |
| 22:18.36 | dan__t | So when using the 's' extension... can't there only be one? Or is that one per context? I don't think I quite follow. |
| 22:19.16 | dan__t | How can you have multiple starting points under the same context? |
| 22:23.58 | *** part/#asterisk profXavier (n=jezus@unaffiliated/neverblue) |
| 22:25.22 | fogo | ~itsp |
| 22:25.23 | jbot | [~itsp] An ITSP is an Internet telephony Service Provider or "VoIP Telephone company". The allow you to either PLACE calls to the PSTN (called Termination), RECEIVE calls from the PSTN (called Origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs |
| 22:25.32 | fogo | ~itsplist-us |
| 22:25.32 | jbot | [~itsplist-us] Here are some popular ITSPs (USA) starting with the more respected ones : http://www.teliax.com , http://connect.voicepulse.com , http://www.nufone.net , http://www.broadvoice.com, http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net |
| 22:25.57 | dan__t | Cool, thanks, I signed up for Vitelity on a recommendation from a friend. |
| 22:25.58 | [TK]D-Fender | dan__t: it is an exten like any other... can only be one per context. Its meaning is horribly mangled by many docs |
| 22:26.03 | dan__t | Already have it set up with a DID heheh |
| 22:26.09 | dan__t | Sounds like it, [TK]D-Fender. |
| 22:26.27 | dan__t | Wouldn't that imply that the same extension was used more than once in a single context? Isn't that bad? |
| 22:26.29 | dan__t | That's my point. |
| 22:26.32 | [TK]D-Fender | ~stdextens |
| 22:26.33 | jbot | [~stdextens] The "s" Standard Extension : Where a call goes to when * does not know the destination of the call. Ex : Calls coming in on FXO ports (no DID), or from an ITSP that doesn't specify or where it was not set in the REGISTER line, or FXS port goes off-hook and "immediate=yes" in zapata.conf. "s" is also used to make IVRs & macros. |
| 22:27.04 | dan__t | Ok, that makes sense, I can take unknown destinations and roue them with that. Just wondering what the deal is when its used more than once in the same context. |
| 22:27.22 | dan__t | Well, unknown extensions, rather... |
| 22:27.56 | manxpower | well unknown DESTINATIONS |
| 22:28.10 | dan__t | Ok, an extension being a destination. I see it. |
| 22:28.34 | dan__t | Is there an asterisk.conf directive for a default sounds directory or something? |
| 22:28.51 | [TK]D-Fender | dan__t: libdir |
| 22:29.11 | dan__t | I see a astvarlibdir, is that the same? |
| 22:29.12 | [TK]D-Fender | dan__t: It expects a sounds/ folder under the one listed |
| 22:29.16 | [TK]D-Fender | dan_tyes, that one |
| 22:29.42 | dan__t | Ok, that's set properly, and there is a 'sounds/' dir in there with stuff in it, yet I can't call sounds with a relative pathname. Would I call it 'sounds/soundname'? |
| 22:30.02 | dan__t | Docs say reference just just as 'soundname', but that doesn't seem to be an option without more path specification. |
| 22:30.29 | [TK]D-Fender | dan__t: where is the intended file exactly? |
| 22:30.41 | dan__t | <PROTECTED> |
| 22:30.52 | dan__t | astvarlibdir is set to /var/lib/asterisk |
| 22:31.04 | [TK]D-Fender | dan__t: the base sounds folder doesn't need anything "relative" to be implied. that is already the starting point |
| 22:31.09 | dan__t | So, does * EXPECT a sounds/ dir to be there, or do I need to specify a relative location to that? |
| 22:31.18 | [TK]D-Fender | dan__t: just Playback(soundfile) theen |
| 22:31.22 | dan__t | Yeah, that's not the behavior I'm experiencing, that's why I'm asking :) |
| 22:31.42 | dan__t | Says that the file does not exit. |
| 22:31.44 | dan__t | exist, too. |
| 22:31.44 | [TK]D-Fender | dan__t: pastebin is your friend... |
| 22:31.49 | dan__t | Indeed it is, hold on. |
| 22:39.24 | *** part/#asterisk russellb (n=russellb@asterisk/digium-open-source-team-lead/russellb) |
| 22:40.34 | dan__t | http://pastebin.com/m2f977c98 |
| 22:41.05 | *** join/#asterisk egypcio (n=egypcio@unaffiliated/egypcio) |
| 22:41.39 | dan__t | I know the file is there, -rw-r--r-- 1 asterisk asterisk 1881 2009-01-12 01:48 /var/lib/asterisk/sounds/vm-isunavail.gsm |
| 22:44.31 | dan__t | Yea, even a relative path - sounds/vm-isunavail, does not work. |
| 22:45.07 | [TK]D-Fender | dan__t: try an absolute path |
| 22:45.18 | dan__t | I did, and like I said, that works fine. |
| 22:45.25 | dan__t | Its just time consuming and error prone :) |
| 22:46.15 | [TK]D-Fender | dan__t: have you restarted * recently? |
| 22:46.28 | dan__t | No. |
| 22:46.37 | [TK]D-Fender | dan__t: pastebin your asterisk.conf and a full "ls" of your counds folder including the call. |
| 22:46.39 | dan__t | However, I've made no changes that would reflect the path of sounds... |
| 22:47.44 | *** join/#asterisk path_ (n=path@pc-15-190-86-200.cm.vtr.net) |
| 22:49.16 | dan__t | Going to take a few minutes.... sorry |
| 22:54.12 | lowtek | I love it, from a customer -> "The polygon phone has no features and is very un-user firendly" |
| 22:54.15 | dan__t | I'll have to push that one aside, for the sake of learning I'll just use absolute paths. |
| 22:54.38 | dan__t | I happen to love my Polygon. |
| 22:54.51 | lowtek | I'm sending them a bunch of grandscreams. |
| 22:55.12 | dan__t | heh |
| 22:56.15 | EI5GTB | hi guys. Oslec works with dahdi ok? |
| 22:56.34 | lowtek | heh, and this "We don't have time to read manuals, tell us how to use these phones?" |
| 22:56.39 | drmessano | nope |
| 22:57.05 | drmessano | or Polycom |
| 22:57.07 | EI5GTB | drmessano, nope @ my comment? |
| 22:57.11 | drmessano | But who is counting |
| 22:57.11 | EI5GTB | oh, ok, sorry |
| 22:57.24 | drmessano | Oslec does not yet work with Dahdi |
| 22:57.46 | EI5GTB | hmm, dang.. when can we expect it to come online? |
| 23:01.57 | *** part/#asterisk korihor (n=korihor@201.210.239.172) |
| 23:08.04 | *** join/#asterisk aptura (n=lork@S010600a0c93f6f7e.vs.shawcable.net) |
| 23:10.42 | aptura | Well my * crashed and for some reason some files were no where to be found and did not have the time to troubelshoot it so went to recompile asterisk again. Compiling 1.4.22 on a old pentium. The compile failed at the very end of make install and have some kind of incompatability with modules. Never seen this before. I have it on pastebin at http://www.pastebin.ca/1306871. |
| 23:10.50 | *** join/#asterisk Bad_Robot- (n=Bad_Robo@cpe-76-173-219-25.socal.res.rr.com) |
| 23:11.28 | aptura | So need to figure why the compatability exist with the list of modules that I pasted in pastebin. |
| 23:11.36 | aptura | and get around it. |
| 23:12.27 | aptura | 29 module names are not compatible. |
| 23:13.27 | beek | aptura: Asterisk probably crashed because you have mixed modules. |
| 23:13.41 | [TK]D-Fender | aptura: Looks like you're trying to roll back from 1.6 |
| 23:13.46 | aptura | yea im just wondering how that could have happened. |
| 23:13.55 | beek | aptura: Delete all of the modules from that directory and recompile again. That will ensure that you have only the appropriate modules. |
| 23:14.02 | [TK]D-Fender | aptura: And indeed no a good thing. Whenever you change vers like that, trash your modules folder first |
| 23:14.19 | aptura | I was thinking that is probebly the thing i needed to do |
| 23:14.27 | beek | aptura: What [TK]D-Fender said. ;-) |
| 23:15.37 | mmlj4 | um, i seem to remember priority-jumping "+ 101" in extensions.conf went away, didn't it? |
| 23:16.01 | *** join/#asterisk path_ (n=path@pc-15-190-86-200.cm.vtr.net) |
| 23:16.12 | [TK]D-Fender | mmlj4: Yes... in *1.2* |
| 23:17.12 | [TK]D-Fender | is happy his line-in is functioning on his sound card for some unknown reason. |
| 23:17.12 | mmlj4 | fair enough |
| 23:23.43 | Micc | why do I keep getting sox: invalid option -- m? |
| 23:30.21 | NovceGuru | herro |
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| 23:40.02 | beek | I need to purchase a roll of the blue/blue-white wire used for telco connections. What is the stuff called so I can search for it? |
| 23:40.58 | [TK]D-Fender | beek: 24 AWG cross-connect wire |
| 23:41.06 | beek | [TK]D-Fender: Thanks. |
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| 23:46.59 | Micc | Can a ShoreTel pbx interface with asterisk? |
| 23:47.08 | Micc | Are shoretel phones SIP? |
| 23:56.07 | Qwell | That's the question we should be asking you. |
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