| 00:01.41 | *** join/#asterisk `Sauron (n=sauron@dsl001-130-033.aus1.dsl.speakeasy.net) |
| 00:02.01 | *** join/#asterisk luke-jr (n=luke-jr@ip70-187-26-118.om.om.cox.net) |
| 00:02.08 | luke-jr | Any way to disable native bridging on Zap? |
| 00:04.13 | *** join/#asterisk KOCATEPE (n=admin@88.247.137.187) |
| 00:04.52 | KOCATEPE | hi all |
| 00:05.21 | KOCATEPE | is from Turkey , m 40 |
| 00:09.15 | *** join/#asterisk outtolunc (n=me@c-67-164-8-168.hsd1.ca.comcast.net) |
| 00:09.30 | C4colo | Qwell: I can't even get the "free" phone for less that $150 now |
| 00:09.42 | C4colo | gotta wait 18 more months or something |
| 00:10.15 | *** join/#asterisk Bad_Robot- (n=Bad_Robo@cpe-76-173-219-25.socal.res.rr.com) |
| 00:10.16 | KOCATEPE | :) |
| 00:12.03 | *** part/#asterisk KOCATEPE (n=admin@88.247.137.187) |
| 00:12.59 | *** join/#asterisk KOCATEPE (n=admin@88.247.137.187) |
| 00:13.08 | KOCATEPE | back again |
| 00:13.12 | KOCATEPE | connection problem |
| 00:13.23 | justdave | is there a way in Meetme for an admin to mute a user and allow that user to still unmute themselves? |
| 00:13.42 | justdave | if you use the normal meetme admin commands to mute a person, only the admin can unmute them, they can't unmute themselves |
| 00:14.27 | *** join/#asterisk russellb (n=russell@asterisk/digium-open-source-team-lead/russellb) |
| 00:14.27 | *** mode/#asterisk [+o russellb] by ChanServ |
| 00:17.10 | *** join/#asterisk jayyers (n=jayyers@c-71-59-10-252.hsd1.ga.comcast.net) |
| 00:17.39 | jayyers | jus installed asteriskNow 1.5 and was wanting some help setting up my sip trunk with nexvortex can anyone point me in the right direction? |
| 00:17.44 | KOCATEPE | i m using debian php5 combination |
| 00:18.29 | edibrac | are there any known problems with linux software raid? AFAIK what I read on the mailinglist are old postings ...that with current hardware, the fears over software raid in general are more about perception than reality |
| 00:18.33 | KOCATEPE | is there any php admin panel for settings |
| 00:19.04 | beek | <PROTECTED> |
| 00:19.44 | edibrac | yeah what I mean is linux software raid with asterisk |
| 00:19.51 | *** join/#asterisk yardB (n=oats@c-68-44-45-241.hsd1.nj.comcast.net) |
| 00:20.06 | drmessano^ | How would Asterisk know it's on a RAID? |
| 00:20.09 | edibrac | linux software raid itself is fine as fa rask I know |
| 00:20.26 | drmessano^ | How would it behave differently or have problems? |
| 00:20.27 | *** join/#asterisk Micc (n=Michael@c-76-121-255-52.hsd1.wa.comcast.net) |
| 00:20.38 | drmessano^ | O.o |
| 00:20.55 | C4colo | throughput |
| 00:20.58 | *** join/#asterisk russellb (n=russell@asterisk/digium-open-source-team-lead/russellb) |
| 00:20.58 | *** mode/#asterisk [+o russellb] by ChanServ |
| 00:20.58 | edibrac | I'm just going off past mailing list entries where people seem to point the finger at RAID |
| 00:21.00 | Micc | I need to find local termination in seattle asap. Anyone know of anyone that has local termination in seattle area? |
| 00:21.22 | C4colo | if you are trying to utilize software raid on a system with less than 1ghz processor then you probably shouldn't be running an application as needy as asterisk |
| 00:21.36 | C4colo | however, on modern hardware software raid is negligable |
| 00:22.00 | C4colo | make sure you are running it on a descent hard drive controller |
| 00:22.28 | C4colo | make sure you get at least 50MB/s on the drives before implementing software raid |
| 00:23.03 | C4colo | and, if you have a good harddrive controler you should see that go up to 75-100MB/s after raid has been implemented |
| 00:23.14 | *** join/#asterisk KOCATEPE (n=admin@88.247.137.187) |
| 00:23.26 | C4colo | man hdparam |
| 00:23.48 | *** join/#asterisk huisnah (n=nhuisman@aeko.ifa.hawaii.edu) |
| 00:23.50 | C4colo | there are lots of tests there to verify the configuration of your hard drives |
| 00:24.02 | huisnah | drmessano^, you around? |
| 00:24.14 | drmessano^ | yeah |
| 00:24.38 | C4colo | Micc: vitelity.net should have DIDs there |
| 00:25.10 | jayyers | jus installed asterisk 1.5 and was wanting some help setting up my sip trunk with nexvortex can anyone point me in the right direction? |
| 00:25.16 | jayyers | <PROTECTED> |
| 00:25.22 | C4colo | what is asterisk 1.5? |
| 00:25.54 | *** join/#asterisk mog (n=mog@c-68-62-172-152.hsd1.al.comcast.net) |
| 00:25.54 | *** mode/#asterisk [+o mog] by ChanServ |
| 00:25.54 | [TK]D-Fender | C4colo: Comes bundled with res_fluxcapacitor.so |
| 00:25.55 | jayyers | *asteriskNow 1.5b |
| 00:26.49 | C4colo | ah that one |
| 00:26.52 | jayyers | i was using switchvox and with no problems getting everything working but wanted to switch to a more customizable system (asteriskNow 1.5b) but am having trouble can anyone help? |
| 00:26.57 | C4colo | I've never touched asterisknow |
| 00:27.23 | edibrac | I'm getting dropped calls and HDLC abort errors, so that's why I'm looking into lower level,. physical or kernel layer stuff that might cause it. No asterisk doesn't "Know" it's no raid, but it makes sense to question it when I'm troubleshooting |
| 00:27.25 | [TK]D-Fender | has... and felt... dirty afterwards |
| 00:27.51 | jayyers | i used it cause it was easy to get asterisk installed and the gui all at once without having to mess with compiling/dependancy issues |
| 00:28.30 | C4colo | there are those of us that depend on the compilation/dependancies for our livelyhood |
| 00:28.46 | C4colo | and those GUIs are messing it up for us |
| 00:29.14 | C4colo | if this stuff becomes easy enough for anyone to set up then where are we? |
| 00:29.39 | C4colo | actually, setting it up shouldn't be too hard |
| 00:29.39 | jayyers | C4colo: lol, well if you want i can stick a distro of your choise in the computer and install it and enable ssh and give u the credentials and let you take the rest but i doubt u want to do that lol |
| 00:30.13 | yardB | when i sip provider give 3 trucks [name-trunk1] [name-trunk2] [name-trunk3] ..what is the significance .. how is applied .. i am only accustum to a single trunk |
| 00:30.17 | C4colo | is this the asterisknow with freepbx or the asterisk gui? |
| 00:30.35 | jayyers | freepbx |
| 00:30.35 | C4colo | yardB: do you have three DIDs with them? |
| 00:31.17 | C4colo | jayyers: did your provider give you a freepbx example? |
| 00:31.21 | yardB | yes |
| 00:31.21 | C4colo | or an asterisk example? |
| 00:31.27 | C4colo | try using the freepbx example |
| 00:31.37 | yardB | C$ yes |
| 00:31.41 | jayyers | they gave me an asterisk example |
| 00:32.10 | C4colo | I have set up freepbx trunks before but the guys in #freepbx would probably be more help |
| 00:32.22 | Micc | C4colo, I already have vitelity and DID's there. I need someone to provide local unimited pri dial out to that area. |
| 00:32.34 | jayyers | https://www.nexvortex.com/tempdocs/Setup%20Guide%20Asterisk.pdf |
| 00:32.35 | yardB | C4colo: yes, i am setting up the config |
| 00:32.37 | Micc | I suppose I'll have to get my own PRI there. |
| 00:32.37 | C4colo | ah, you need local termination |
| 00:33.01 | C4colo | bandwidth.com I think does PRIs in many major cities |
| 00:33.13 | C4colo | also look at the Public Utilities Commission for the area |
| 00:33.20 | C4colo | they should have a list of LECs and CLECS for the area |
| 00:33.32 | C4colo | call them all up |
| 00:33.43 | jayyers | i tried to put all info as specified by the doc into the gui but i dont realy know what goes into the peer part and what goes into the User/context part of the freepbx gui, i tried the best i could but it doesnt register the trunk |
| 00:33.51 | C4colo | bandwidth.com might even be able to provide you with a SIP trunk for local termination |
| 00:33.53 | edibrac | have you guys seen a particular card cause dropped calls, yet after swapping it with another one of the same model, it's fine? |
| 00:34.11 | C4colo | edibrac: what manufacturer? |
| 00:34.12 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 00:34.19 | edibrac | C4colo: Digium TE121 |
| 00:34.30 | edibrac | PCI-E card |
| 00:34.32 | C4colo | yea, probably hardware issue, is it under warranty? |
| 00:34.44 | yardB | C4: do i need DID if i only want them to terminate my calls? |
| 00:35.09 | edibrac | C4colo: well I ordered a 2nd one, in hopes it is that particular card .. I'm going to try it out tonight to see how it oges |
| 00:35.11 | C4colo | yardB, they probably have outbound only, I'm not sure of their offerings |
| 00:35.41 | C4colo | oh, you haven't tested the new card yet? |
| 00:36.10 | C4colo | it is possible that it is hardware, but it is more likely the configuration, unfortunatly I'm not too good with PRIs at this point |
| 00:36.11 | edibrac | C4colo: the odd part is that I've had the same dropped calls and HDLC errors on a known working asterisk server |
| 00:36.43 | yardB | C$ you are in great mand ;) |
| 00:36.50 | yardB | demand |
| 00:36.51 | edibrac | C4colo: the confusing part is that the PRI monitoring unit our telco droped off, says it's our box that is having problems. |
| 00:37.13 | C4colo | does this "pri monitoring unit" have the name Adtran on it? |
| 00:37.53 | beek | edibrac: Good luck with that. It took me over three weeks to get the telco to admit that they were having problems. |
| 00:38.10 | edibrac | C4colo: it is Westell-specific .. so actually "pri monitoring unit" is a term I made up. They say it "monitors the NIU", a westtell. |
| 00:38.17 | beek | edibrac: I'm assuming that what they gave you was a CSU |
| 00:38.42 | C4colo | monitors the NIU? |
| 00:38.48 | C4colo | what kind of crap is that? |
| 00:38.58 | beek | C4colo: I'd say NI |
| 00:39.07 | edibrac | well it sits between the NIU and CPE ..then logs problems to determine which side is broken |
| 00:39.21 | C4colo | the NIU should have internal monitoring, called RED/BLUE/YELLOW alarm, and some LEDs on it |
| 00:39.29 | *** join/#asterisk etfonhomey (n=chatzill@mobile-166-214-063-253.mycingular.net) |
| 00:39.43 | C4colo | interesting |
| 00:39.49 | C4colo | and what has this monitoring unit said? |
| 00:39.56 | C4colo | that the call is being dropped by your pbx? |
| 00:40.20 | edibrac | C4colo: the logs correspond to our HDLC asterisk log errors |
| 00:40.52 | edibrac | which correspond to dropped calls ..but we have had dropped calls when there isn't a HDLC log entry. |
| 00:41.02 | edibrac | and all that is intermittent |
| 00:41.25 | C4colo | ok, it is possible they are not providing you with an actual PRI |
| 00:41.34 | C4colo | the device that "monitors the NIU" could be an IAD |
| 00:41.56 | C4colo | using the bandwidth of the T1 for SIP termination and generates a PRI signalled line locally |
| 00:42.42 | C4colo | we had a similar problem with an ADTRAN IAD (since that's actually how we provide PRIs on our network) ... we fixed it by changing the SIP server address to it's actual IP address instead of the FQDN of the server |
| 00:42.58 | edibrac | if what you say is true, could it be that some/most calls work most of the time? Our situation is that -- that dropped calls happen with no pattern. |
| 00:43.05 | C4colo | it was a DNS lookup timeout, the call would take too long to set up so it would drop |
| 00:43.17 | C4colo | that's why it took us so long to figure it out |
| 00:43.39 | C4colo | when the DNS server was overloaded and took more than so many milliseconds to respond, and when the DNS cache had expired, it would drop the call |
| 00:43.46 | C4colo | but those two things only lined up every once in a while |
| 00:44.10 | C4colo | so it would be fine for hours then it would drop a call, then it would take another hour or two and drop two at the same time |
| 00:44.22 | edibrac | well I have my asterisk configs to match up with the signalling of a PRI -- if they weren't providing a PRI, nothing would work, right? Or..are you saying there's a way they can "simulate" a PRI? |
| 00:45.11 | C4colo | the way we do it is that we send an IP T1 to the customer premise and then use an Adtran IAD to generate a PRI locally |
| 00:45.38 | C4colo | then the IAD connects back to our network via a SIP trunk |
| 00:46.05 | C4colo | that allows us to bond two or three T1s and they get the full bandwidth minus whatever phone calls are going on |
| 00:46.18 | C4colo | or we can set up a partial PRI and give them the rest of the bandwidth |
| 00:46.28 | C4colo | this may not be the issue on your particular setup |
| 00:46.56 | C4colo | while most companies are going this route for PRIs because it is cheaper, the company you are using may still be providing true PRI signalled lines |
| 00:46.58 | edibrac | in your setup, would I setup my box the same way as if it were a PRI? |
| 00:47.05 | C4colo | yea |
| 00:47.08 | C4colo | it's a real PRI |
| 00:47.19 | C4colo | just that it's 3 feet long instead of thousands of feet long |
| 00:47.47 | C4colo | and it's using TCP/IP for the backhaul to the LEC |
| 00:48.00 | *** join/#asterisk grandpapadot (n=no@adsl-074-185-089-046.sip.bhm.bellsouth.net) |
| 00:48.06 | C4colo | instead of channelized ISDN |
| 00:49.02 | edibrac | if I'm setting up a asterisk box on that setup, should I have to know how the PRI gets to me? Those details should be "encapsulated" right? |
| 00:49.44 | edibrac | or i guess your describing it to point out how different it is from other setups |
| 00:50.10 | grandpapadot | edibrac: I missed the first part of what you said, can you restate your question regarding PRIs? |
| 00:50.29 | C4colo | it shouldn't matter to your configuration 'how' the PRI gets to you |
| 00:51.00 | C4colo | but, if it is a locally generated PRI by an IAD, then the configuration of that IAD could be causing the problem |
| 00:51.20 | C4colo | I'm just saying that we had that issue with some Adtran IADs on our network, sounds close enough to suggest it as a possibility |
| 00:51.48 | C4colo | has the "monitoring" box been there the entire time? or did they add it when you were having dropped calls? |
| 00:52.13 | edibrac | grandpapadot: our telco (XO) dropped off something that monitors problems between their NIU and the CPE (our asterisk box). The errors correspond to our asterisk HDLC abort errors in the asterisk logs. The HDLC errors match up to dropped calls on our side. The weird thing, is that we had a known working asterisk box do the same thing, connected to the same PRI. |
| 00:52.40 | edibrac | C4colo: they added it afterwards when we requested it |
| 00:52.53 | C4colo | ah, then that is not an IAD |
| 00:52.58 | grandpapadot | PCI bug in the mobo? |
| 00:53.33 | edibrac | i read a post in the mailing list that it could be the card ..or mobo. Just sounds like a hardware thing. |
| 00:53.43 | grandpapadot | What board are you terminating the PRI with in your box? |
| 00:53.43 | edibrac | I just don't understand why it would do it for both boxes. |
| 00:53.52 | edibrac | perhaps there are 2 problems. |
| 00:53.55 | C4colo | "The weird thing, is that we had a known working asterisk box do the same thing, connected to the same PRI." |
| 00:54.00 | grandpapadot | Are they both the same BIOS revision? |
| 00:54.02 | C4colo | so you have tested another pbx? |
| 00:54.07 | C4colo | and it acts the same way on this line |
| 00:54.15 | C4colo | while the "good" pbx doesn't have that problem on other PRI lines?L |
| 00:54.27 | grandpapadot | What card are you using to terminate the PRIs? |
| 00:54.55 | edibrac | the old one is TE110 new one is TE121 |
| 00:55.10 | edibrac | Digium |
| 00:55.35 | edibrac | BIOS.. i haven't check. They are both supermicro boxes |
| 00:55.35 | JT | have you checked zttest? |
| 00:55.37 | grandpapadot | Ok, well if everything else is the same, it's obviously the card since they are different. |
| 00:55.49 | edibrac | yeah zttest is cool. |
| 00:55.58 | JT | what results does it give? |
| 00:56.00 | grandpapadot | Check to see if the BIOS versions are the same, sometimes vendors fix PCI timing bugs with BIOS updates. |
| 00:56.20 | grandpapadot | Which card is problematic, the 110 or 121? |
| 00:56.51 | edibrac | one thing - i was reading about PCI latency. And on the current broken one TE121, only the video card has a higher priority. |
| 00:57.04 | JT | edibrac: what zttest results do you get? |
| 00:57.06 | edibrac | but i figure that shouldn't matter - it's just asterisk installed. |
| 00:57.13 | grandpapadot | Is your 121 by change sharing an IRQ? |
| 00:57.24 | JT | digium cards are extremely finicky |
| 00:57.39 | grandpapadot | s/change/chance |
| 00:57.40 | JT | pri bitslips are a common problem |
| 00:57.46 | grandpapadot | JT: Agreed. |
| 00:58.21 | edibrac | JT - IOW it's entirely reasonable that if I get antoher TE121 that may fix the issue? |
| 00:58.26 | JT | no |
| 00:58.33 | edibrac | i checked IRQs they look good |
| 00:58.41 | JT | they all have the same issues, they don't like some motherboards, etc |
| 00:58.44 | grandpapadot | Have you tried another slot? |
| 00:58.52 | JT | edibrac: third time, what zttest results do you get? |
| 00:59.20 | edibrac | JT: yeah sorry - it looks good 99.998734% |
| 00:59.36 | edibrac | er: --- Results after 104 passes --- |
| 00:59.36 | edibrac | Best: 100.000 -- Worst: 99.991 -- Average: 99.998434, Difference: 99.998434 |
| 00:59.47 | JT | edibrac: do you have zttest running during any of the times you've had a problem occur? |
| 01:00.07 | JT | sometimes zttest results will only go intermittently bad |
| 01:00.13 | JT | generally corresponding with a bitslip |
| 01:00.16 | edibrac | that would be nice, but I have no idea when it happens. I guess I could leave zttest running all the time --- that shouldn't hurt anything right? |
| 01:00.39 | JT | should be alright, hard to gather useful stats from zttest when it runs for ages though |
| 01:00.55 | edibrac | I take it you recommend Sangoma? |
| 01:01.19 | JT | either that or a pri to sip gateway, eliminates a whole raft of possible headaches |
| 01:03.32 | edibrac | i went down the "is this an IRQ miss problem" route but that doens't seem to be the case |
| 01:04.08 | edibrac | i definately can't "force" the HDLC error to happen myself |
| 01:06.29 | edibrac | how relevant is this 2005 posting about how this guy dealt with his HDLC problems: http://www.mail-archive.com/asterisk-users@lists.digium.com/msg119398.html |
| 01:08.38 | *** join/#asterisk lucky|aba (n=lucky@ip72-205-200-133.sb.sd.cox.net) |
| 01:08.56 | JT | edibrac: btw, did you try changing pci slots? sometimes that solves the issue |
| 01:09.45 | edibrac | only one PCI-E slot :( |
| 01:09.58 | edibrac | or maybe there's an adapter thing? |
| 01:10.58 | *** join/#asterisk zpinter (n=zpinter@206-124-6-30.denver.dsl.forethought.net) |
| 01:11.50 | jets | ya and also make sure nothing else is sharing the irq with the card. |
| 01:12.09 | jets | oh nevermind |
| 01:12.15 | jets | i didn't see your note above about an irq miss problem. |
| 01:12.17 | jets | ;) |
| 01:22.19 | *** join/#asterisk JoshuaP0x (n=Administ@unaffiliated/joshuap0x) |
| 01:23.01 | JoshuaP0x | how do I verify that /usr/src/ contains a symbolic link named linux-2.4 pointing to your kernel source? |
| 01:23.18 | JoshuaP0x | how do I verify that /usr/src/ contains a symbolic link named linux-2.4 pointing to my kernel source? |
| 01:24.12 | *** join/#asterisk grndslm (n=grndslm@24-116-87-97.cpe.cableone.net) |
| 01:25.48 | *** join/#asterisk jer (n=jtregunn@unaffiliated/jer) |
| 01:27.44 | *** join/#asterisk grndslm (n=grndslm@24-116-87-97.cpe.cableone.net) |
| 01:30.31 | edibrac | what's the most "generic" hardware/distro setup for asterisk? something like CentOS, latest asterisk/dahdi/libpri by source and .. a popular digium or Sangoma card? |
| 01:30.38 | edibrac | and a dell box? |
| 01:31.14 | drumkilla | sounds about right |
| 01:31.17 | drumkilla | except for the Sangoma part |
| 01:31.18 | drumkilla | :-p |
| 01:32.18 | [TK]D-Fender | drumkilla: You're right...the Digium card would be ...."generic". |
| 01:32.31 | [TK]D-Fender | edibrac: So.... you wan't an awesome system.....right? ;) |
| 01:32.36 | drumkilla | harsh |
| 01:32.49 | [TK]D-Fender | drumkilla: You said it ;) |
| 01:32.55 | [TK]D-Fender | (j/k overll) |
| 01:33.00 | [TK]D-Fender | overall* |
| 01:33.00 | JoshuaP0x | how do I verify that /usr/src/ contains a symbolic link named linux-2.4 pointing to my kernel source? |
| 01:33.04 | drumkilla | I was taking generic to mean most common, most likely to work properly, etc. :-p |
| 01:33.16 | drumkilla | linux-2.4? |
| 01:33.21 | JoshuaP0x | yes |
| 01:33.25 | drumkilla | dahdi doesn't even support linux 2.4 .. |
| 01:33.27 | JoshuaP0x | I don't know what that means |
| 01:33.30 | drumkilla | oic. |
| 01:33.32 | [TK]D-Fender | uses the dictionary when implementing words and deploying sentences. |
| 01:33.39 | *** join/#asterisk moy (n=moy@CPE001f3a8fd7bd-CM0011ae8a6af8.cpe.net.cable.rogers.com) |
| 01:33.40 | JoshuaP0x | the book says to do that |
| 01:33.48 | drumkilla | disables the pedantic flag on [TK]D-Fender |
| 01:33.52 | *** join/#asterisk luckyaba (n=lucky@ip72-205-200-133.sb.sd.cox.net) |
| 01:34.29 | *** join/#asterisk normsteel (n=nathank@206.173.193.34.ptr.us.xo.net) |
| 01:34.34 | [TK]D-Fender | ERROR: "pedantic=no" is no longer supported in this version, and good riddence! |
| 01:34.39 | *** join/#asterisk Bad_Robot- (n=Bad_Robo@cpe-76-173-219-25.socal.res.rr.com) |
| 01:34.59 | [TK]D-Fender | drumkilla: So like.. y0 'sup homie? ;) |
| 01:35.20 | drumkilla | holla back, yo |
| 01:35.21 | [TK]D-Fender | drumkilla: Sportin' the streen-nick an all... |
| 01:35.26 | [TK]D-Fender | street* |
| 01:35.57 | [TK]D-Fender | is STILL moving his company's offices around... right through midnight |
| 01:37.07 | normsteel | does raid really effect asterisks stability? |
| 01:37.20 | normsteel | ..raid1 |
| 01:38.06 | drumkilla | if disk I/O locks out interrupts for a long time, and you're using TDM hardware, then yes |
| 01:39.10 | hardwire | you guys eve seen a cat5e patch panel that has centronics connectors and a big bundled tie cable going to another similar patch? |
| 01:39.13 | hardwire | I'd.. love.. that |
| 01:39.29 | mmattice | yeah |
| 01:39.40 | hardwire | you know who makes it? |
| 01:39.53 | mmattice | you should be able to get it at any major telco supply |
| 01:40.03 | hardwire | is it cat5e rated? |
| 01:40.20 | mmattice | possibly |
| 01:40.25 | hardwire | I've only seen 8 way versions that don't really say cat5 anything on them |
| 01:40.37 | mmattice | why 5e? |
| 01:41.09 | hardwire | dunno.. will any ol twisted pair work? |
| 01:41.24 | hardwire | I don't want to use a funny awg and crappy connectors that don't pass muster |
| 01:41.30 | mmattice | for what? phones? |
| 01:42.01 | hardwire | gigabit network |
| 01:42.21 | mmattice | I wouldn't suggest doing that. |
| 01:42.42 | hardwire | if I can't find the product, I won't be able to anyways :) |
| 01:43.17 | edibrac | . |
| 01:43.30 | edibrac | normsteel: you suck |
| 01:52.24 | hardwire | I guess they took it outside. |
| 01:52.57 | hardwire | http://www.siemon.com/apps/Utilities/showImageDisplay.asp?showImage=/share/products05/mp_hd5-quick-patch-panel_big.jpg |
| 01:53.18 | hardwire | that's cute |
| 01:53.54 | JoshuaP0x | what does the make command do? |
| 02:03.23 | *** join/#asterisk joat (n=joat@ip70-174-79-200.hr.hr.cox.net) |
| 02:07.01 | eppigy | hello |
| 02:07.09 | eppigy | i am dave |
| 02:07.28 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 02:07.49 | *** join/#asterisk Deeewayne (n=dwayne@c-76-29-245-9.hsd1.al.comcast.net) |
| 02:07.49 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 02:08.03 | justdave | wonders if that's a bot that just does that at random |
| 02:08.46 | eppigy | negative |
| 02:32.41 | *** join/#asterisk joat (n=joat@ip70-174-79-200.hr.hr.cox.net) |
| 02:36.06 | [TK]D-Fender | justdave: It's just signaling the shift in persona... it is kinda crowded up there after all ;) |
| 02:38.45 | eppigy | [TK]D-Fender: hello i am trying to use chan_celliax to contact puerto rico |
| 02:38.51 | eppigy | and seem to be getting an error |
| 02:38.54 | eppigy | can you help me |
| 02:46.12 | *** join/#asterisk jicksta (n=jicksta@c-24-6-87-187.hsd1.ca.comcast.net) |
| 02:50.33 | *** join/#asterisk Carlos_PHX (n=Carlos@ip68-3-162-244.ph.ph.cox.net) |
| 02:53.32 | *** join/#asterisk joecool01 (n=chatzill@24.174.122.65) |
| 02:56.41 | *** join/#asterisk mog (n=mog@c-68-62-172-152.hsd1.al.comcast.net) |
| 02:56.41 | *** mode/#asterisk [+o mog] by ChanServ |
| 02:57.29 | *** join/#asterisk andresmujica (n=andresmu@190.27.0.198) |
| 03:01.17 | *** join/#asterisk davidR2008 (n=david@nc-71-48-8-214.dhcp.embarqhsd.net) |
| 03:01.56 | davidR2008 | hey all |
| 03:02.14 | davidR2008 | any festival experts? |
| 03:03.28 | *** join/#asterisk coppice (n=chatzill@33.155.17.210.dyn.pacific.net.hk) |
| 03:05.02 | TrentCreek | sure..i love festivals |
| 03:05.49 | eppigy | i see what you did there |
| 03:06.34 | davidR2008 | well I do too, but I doubt you'll love my festival ;-) |
| 03:08.05 | eppigy | davidR2008: I am dave as well |
| 03:15.24 | davidR2008 | I followed method one here: http://www.voip-info.org/wiki-Asterisk+Festival+installation and it's not working. I'm on CentOS 5.1 / Asterisk 1.4.20 |
| 03:16.29 | *** join/#asterisk lucasb (n=lucasb@s154-5-252-231.bc.hsia.telus.net) |
| 03:16.35 | davidR2008 | I was hoping to talk it through with someone more knowledgeable then me on this |
| 03:17.47 | *** join/#asterisk phix (n=threat@123-243-44-131.tpgi.com.au) |
| 03:17.49 | phix | hey |
| 03:18.18 | phix | No matter what number I dial on my Nokia E65, it always rings s on my asterisl dial plan |
| 03:18.22 | phix | any one had this issue? |
| 03:18.45 | [TK]D-Fender | phix: Show us the SIP debug of the call |
| 03:25.59 | ScribbleJ | Whoot, AGI rocks. I guess this probably exists a million times over but I just put together an AGI script to reject an incoming call, then automatically place a call to that same incoming number, if it meets a list of approved #s. |
| 03:26.09 | ScribbleJ | Asterisk is so fun. |
| 03:28.49 | [TK]D-Fender | ScribbleJ: Depending on how much logic is required to make that decision it could all be done in dialplan... |
| 03:29.08 | ScribbleJ | Really? I am new to this, but I couldn't fgure out how to get a dialplan to hang up on someone, basically, then call them back. |
| 03:29.16 | ScribbleJ | I thought the dialplan terminated with the call. |
| 03:31.28 | ScribbleJ | I did it by writing an AGI that takes the caller ID, tells asterisk to hangup, spawns another process and returns, the other process then writes a .call file with the appropriate details and moves it into the /spool/outbound dir to generate the call. |
| 03:31.34 | ScribbleJ | Works like a charm. |
| 03:31.59 | *** join/#asterisk axisys (n=axisys@ip68-98-177-71.dc.dc.cox.net) |
| 03:33.38 | [TK]D-Fender | ScribbleJ: use System() to call the script that will issue the call-file with the # as a parameter. In your call-file, the channel it uses will be a Local channel whose dialplan will start with a Wiat(). Your script will almost immediately return. You then simply hangup. |
| 03:34.30 | [TK]D-Fender | ScribbleJ: this is the "execution part". The lookup part depends on the level of complexity you require. *'s DB capabilities could possible handle this all direct, or AstDB for local minimal stuff |
| 03:34.39 | ScribbleJ | Yeah. |
| 03:34.52 | [TK]D-Fender | ScribbleJ: If nothing else, this is food for thought for you for future tasks |
| 03:35.55 | ScribbleJ | That all makes sense to me except the 'local channel' part, I guess that means instead of having the .call file place the outbound call directly, I have it call 'asterisk' and go into a dialplan that wait()s then makes the call out... I see, yeah. |
| 03:35.57 | ScribbleJ | Clever. |
| 03:36.17 | ScribbleJ | Well as my goal was to learn to use AGI I guess I did all right anyhow. |
| 03:36.17 | ScribbleJ | Heh |
| 03:36.25 | [TK]D-Fender | <- smarter than the average bear |
| 03:36.48 | ScribbleJ | That also solves anohter problem I was having - |
| 03:37.00 | ScribbleJ | Couldn't figure out how to specify ultiple providers to try in the .call file |
| 03:37.10 | ScribbleJ | Easy neough to do in a dialplan though. |
| 03:37.18 | [TK]D-Fender | ScribbleJ: You... Local channels make so many other problems disappear |
| 03:37.22 | [TK]D-Fender | yup* |
| 03:37.42 | [TK]D-Fender | ScribbleJ: jsut remember to be VERY careful about when channels get "answered" |
| 03:37.59 | ScribbleJ | That makes perfect sense, I will have to learn how to set up local channels. |
| 03:38.09 | *** join/#asterisk jer (n=jtregunn@unaffiliated/jer) |
| 03:38.57 | ScribbleJ | Oh? I'm still a bit fuzzy on the proper use of Answer() and Hangup(). In fact I didn't want to Answer() the incoming call at all, just let it ring once then reject it, but that I could not figure out how to do. |
| 03:39.34 | ScribbleJ | If I hangup() without having Answer()d my one provider goes nuts trying to retry me and fills all my channels, the other drops them to it's vm system. |
| 03:39.44 | ScribbleJ | Neither of which is what I'd hoped to get. |
| 03:46.39 | [TK]D-Fender | ScribbleJ: Play around a bit, I'm sure you'd jsut about on top of it. |
| 03:47.27 | ScribbleJ | Oh, I am. Playing with Asterisk has been awesome fun these last two days. I plan to screw around with it most of the weekend. |
| 03:47.48 | ScribbleJ | This is probably inadviable overall, but right now I plan to write some AGI to process a credit card payment over the phone. |
| 03:48.10 | ScribbleJ | Just for playing around... I'm sure I'll look into the details and learn my VoiP protocols are totally insecure and unsuitable. |
| 03:49.46 | [TK]D-Fender | ScribbleJ: AGI pays off if you have a variety of little things that are a PITA for * to do in dialplan as well. It jsut a question of evaluating a specific case. |
| 03:49.53 | phix | [TK]D-Fender: ok I will pastebin it for you mate |
| 03:49.54 | *** join/#asterisk emist (n=emist@unaffiliated/emist) |
| 03:50.47 | ScribbleJ | Yeah, I can see getting all the required data in a dialplan, but - |
| 03:51.28 | ScribbleJ | Oh, I see, I could accumulate the data in a dialplan and then uses System() to make a call out to process... eh, maybe, exposing the cc num on the command line like that sounds like a bad idea. |
| 03:52.22 | ScribbleJ | Eh, there's so many ways to do everything, I guess 'learning asterisk' is mostly the process of learning the good ways and the bad ways. |
| 03:53.49 | *** join/#asterisk sah-work (n=Bawbatos@adsl-76-203-173-209.dsl.pltn13.sbcglobal.net) |
| 03:54.48 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 03:55.48 | *** join/#asterisk nhuisman_work (n=to@lepo.ifa.hawaii.edu) |
| 03:58.54 | *** join/#asterisk elguero (n=elguero@ns1.nashuacs.com) |
| 04:01.04 | *** join/#asterisk andresmujica (n=andresmu@190.27.0.198) |
| 04:06.08 | *** join/#asterisk dacs (n=chatzill@unaffiliated/dacs) |
| 04:06.14 | dacs | howdy |
| 04:06.22 | rue_desk | hi |
| 04:10.16 | [TK]D-Fender | doody |
| 04:13.58 | phix | :D |
| 04:19.02 | *** join/#asterisk Subdolus (n=subby@subby.afraid.org) |
| 04:24.35 | dacs | ~book |
| 04:24.36 | jbot | [book] 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 |
| 04:25.14 | phix | [TK]D-Fender: Dec 20 15:24:53 WARNING[4324]: chan_sip.c:1229 retrans_pkt: Maximum retries exceeded on transmission A613Z5Y0oIc-OM4mjBjQhwdRZwYIcr for seqno 1437 (Critical Response) |
| 04:26.16 | [TK]D-Fender | phix: Its been over 1/2 hour since you said you were gtting me that pastebin, and ANOTHER 1/2 prior is when I asked you for it in the first place... |
| 04:26.24 | [TK]D-Fender | phix: and you paste one useless line? |
| 04:26.34 | [TK]D-Fender | phix: that doesn't tell anything |
| 04:26.42 | [TK]D-Fender | phix: Try again... |
| 04:27.05 | phix | haha yeah I was busy TK :) |
| 04:27.07 | phix | I will try again |
| 04:27.19 | phix | should I check the number before I try again? |
| 04:27.32 | phix | set debug 5? |
| 04:27.35 | phix | that dgood enough? |
| 04:30.15 | [TK]D-Fender | phix: SIP DEBUG. you know.. what I asked for over an hour ago... |
| 04:31.38 | eppigy | business as usual |
| 04:32.13 | thehar | hides |
| 04:34.36 | *** join/#asterisk Carlos_PHX (n=Carlos@ip68-3-162-244.ph.ph.cox.net) |
| 04:36.41 | *** join/#asterisk Carlos_PHX (n=Carlos@ip68-3-162-244.ph.ph.cox.net) |
| 04:37.10 | phix | [TK]D-Fender: hmmm, I think I will debug this later, I am over it :) thanx any way for being there :P |
| 04:39.05 | dacs | anyone here using Linksys PAP2 |
| 04:39.29 | dacs | i just picked one from walmart for $2 |
| 04:39.37 | *** join/#asterisk pcrane (n=pcrane@125-238-255-99.broadband-telecom.global-gateway.net.nz) |
| 04:39.39 | dacs | brand new in the box |
| 04:40.13 | dacs | i don't know if it was mislabeled or what |
| 04:40.34 | thehar | dacs i use them but not usING |
| 04:40.56 | dacs | ? |
| 04:41.07 | thehar | we use them for 2 line residential iads |
| 04:42.01 | dacs | do i have to hack it or is it open... i mean is it lock to a specific provider |
| 04:43.22 | thehar | uhm we use the pap2t |
| 04:44.26 | dacs | mine is just PAP2 |
| 04:44.54 | [TK]D-Fender | dacs: Did the box mention a provider? |
| 04:45.12 | [TK]D-Fender | dacs: Walmart generally never carries stuff like this that isn't locked to some company or another |
| 04:45.38 | dacs | [TK]D-Fender: i can't remember i throw the box |
| 04:45.50 | [TK]D-Fender | SMRT |
| 04:45.55 | dacs | :) |
| 04:46.04 | [TK]D-Fender | dacs: plug it in and try to admin it. |
| 04:46.24 | [TK]D-Fender | dacs: then if that fails wireshark whatever it treis to do and see where it's phoning home to |
| 04:48.16 | drmessano^ | Nice |
| 04:48.29 | drmessano^ | Now you have no clue what version it is to even unlock it |
| 04:48.33 | drmessano^ | SMRT |
| 04:50.36 | dacs | i just throw the box this morning in the dumpster ... let me go try my luck |
| 04:50.56 | dacs | i will go dig it out |
| 04:51.05 | dacs | i just hope they didn't take it |
| 04:51.07 | dacs | :( |
| 04:51.15 | [TK]D-Fender | Dumpster diving : The bottome of the hacking food-chain |
| 04:51.31 | *** join/#asterisk jeffgus (n=jeffgus@static-173-51-181-3.lsanca.ftas.verizon.net) |
| 04:51.39 | dacs | :) |
| 04:54.58 | *** join/#asterisk jeffgus (n=jeffgus@static-173-51-181-3.lsanca.ftas.verizon.net) |
| 04:59.44 | dacs | [TK]D-Fender: i got it...hahaha my neighbor was laughing his ass off... i said you wouldn't understand |
| 05:00.09 | drmessano^ | SO what color is the top of the box? |
| 05:00.14 | drmessano^ | Orange or Grey? |
| 05:00.28 | eppigy | and what coplor was it before you put it in the dumpster |
| 05:00.36 | [TK]D-Fender | :p |
| 05:00.41 | dacs | its Vonage |
| 05:00.43 | dacs | :( |
| 05:00.47 | drmessano^ | SO what color is the top of the box? |
| 05:00.48 | drmessano^ | Orange or Grey? |
| 05:00.59 | drmessano^ | They're all Vonage if you got a PAP2 from wal-mart |
| 05:01.00 | dacs | oh wait whats that bad smell |
| 05:01.01 | [TK]D-Fender | dacs: Good... they are better documented as far as unlocking goes |
| 05:01.02 | dacs | lol |
| 05:01.07 | drmessano^ | Orange or Grey? |
| 05:01.13 | dacs | its orange |
| 05:01.22 | drmessano^ | Ok, its a V1 then |
| 05:01.33 | drmessano^ | Easy to unlock |
| 05:02.09 | *** join/#asterisk MaliutaLap (i=biteme@S0106001a927737b1.fm.shawcable.net) |
| 05:02.59 | dacs | drmessano^: am ready to read |
| 05:03.08 | drmessano^ | So go do it |
| 05:04.23 | dacs | you have a document |
| 05:04.30 | thehar | google |
| 05:05.01 | drmessano^ | Theres a whole internet out there |
| 05:05.26 | drmessano^ | I have several, Google has dozens |
| 05:19.55 | [TK]D-Fender | ok, heading home. Back tomorrow some time... |
| 05:20.00 | thehar | byeeee |
| 05:20.22 | thehar | chatzilla? wow |
| 05:20.28 | thehar | i didn't know people actually used that. |
| 05:20.30 | thehar | irssi ftw |
| 05:23.49 | dacs | i keep getting bad password |
| 05:24.43 | *** join/#asterisk freakazoid0223 (n=mattc@pool-72-81-1-234.phlapa.east.verizon.net) |
| 05:25.18 | drmessano^ | Of course you do |
| 05:25.20 | drmessano^ | Its locked |
| 05:25.22 | drmessano^ | dug |
| 05:25.24 | drmessano^ | duh |
| 05:25.25 | drmessano^ | too |
| 05:25.29 | thehar | drink up drmessano^ |
| 05:25.30 | thehar | drink up |
| 05:26.23 | drmessano^ | I don't drink |
| 05:27.29 | coppice | you must get damned thirsty |
| 05:27.43 | thehar | PARCHED |
| 05:27.54 | drmessano | points to the IV fluid bag |
| 05:28.32 | drmessano | After I got over 700lbs, I stopped being able to open my jaw to drink |
| 05:28.40 | drmessano | :( |
| 05:29.10 | thehar | big big bendy perm-straw |
| 05:29.23 | drmessano | Can't open my mouth |
| 05:29.28 | drmessano | Wanna fight about it? |
| 05:29.29 | *** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose) |
| 05:29.31 | thehar | refrains |
| 05:30.21 | dacs | drmessano: can you help me unlock my PAP2 |
| 05:30.32 | coppice | I think anyone claiming to be 700lb on IRC is probably a hot young woman in real life |
| 05:30.43 | thehar | or hot young man |
| 05:31.28 | *** join/#asterisk Talkradio (i=talkradi@linuxgeneration.net) |
| 05:31.45 | drmessano | dacs: There's DOZENS of guides online |
| 05:31.46 | coppice | whatever lights your candle :-) |
| 05:32.18 | thehar | would you light my candle? |
| 05:32.28 | coppice | drmessano: and they all agree with one another? |
| 05:32.58 | drmessano | coppice: Pretty much the same method.. and only one method |
| 05:33.49 | drmessano | http://www.unlockmypap2.com/ |
| 05:33.54 | drmessano | Z O M GOOGLE |
| 05:33.57 | thehar | gasp |
| 05:36.22 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 05:38.29 | coppice | a somewhat narrowly focussed site. I wonder if they registered anything like sodvonage.com? |
| 05:41.43 | *** join/#asterisk glaz (i=strke@fibermage.org) |
| 05:46.15 | echinos | I see sample configs etc. with leading underscores in front of extensions like so:exten => _1NXXNXXXXXX |
| 05:46.21 | echinos | what's the underscore for? |
| 05:47.13 | thehar | pattern matching |
| 05:47.51 | echinos | ah, yes |
| 05:48.06 | echinos | thx |
| 05:48.16 | thehar | asterisk oreilly book. reads it |
| 05:48.21 | echinos | it's the "this is a pattern match extension" token |
| 05:48.44 | echinos | hads to buys it firsts |
| 05:48.53 | thehar | pdf free online |
| 05:49.00 | jasonpr | got a link? |
| 05:49.01 | echinos | wha!? |
| 05:49.10 | thehar | sec |
| 05:49.12 | echinos | link or it's not true ;) |
| 05:49.21 | thehar | SECOND |
| 05:49.22 | jasonpr | lol |
| 05:49.25 | echinos | KIDDING! |
| 05:49.46 | thehar | oreilly charges but it's everywhere.. let me link |
| 05:50.01 | echinos | I'd give you kudos if there was a bot to keep track of it |
| 05:50.02 | thehar | i think leif has it on his domains |
| 05:50.26 | thehar | jbot |
| 05:50.26 | jasonpr | 2nd edition? |
| 05:50.32 | thehar | jbot speak! |
| 05:50.33 | jbot | ARFFFFFFFFFFFF |
| 05:50.39 | thehar | jbot pdf me |
| 05:50.45 | echinos | thehar++ |
| 05:50.57 | echinos | jbot karma? |
| 05:50.58 | jbot | it has been said that karma is $1 has power karma |
| 05:51.13 | echinos | jbot thehar karma? |
| 05:51.22 | thehar | [5~ree downloadable PDF at http://www.asteriskdocs.org --- HTML at http://tfot.leifmadsen.com or see ~buybook |
| 05:51.49 | *** join/#asterisk DMMatt (n=mlager@srvr-1.lynxcom.net) |
| 05:51.51 | DMMatt | Hi |
| 05:52.05 | echinos | you da man |
| 05:52.26 | echinos | tyvm |
| 05:52.50 | jasonpr | bought book..... I keep hitting <Ctrl-F> nothing happens |
| 05:52.51 | thehar | :) |
| 05:52.59 | thehar | i have book in pdf and hard copy |
| 05:53.06 | thehar | i like highlighters and stickie tabs |
| 05:53.19 | DMMatt | When I get incoming calls, the caller ID is always "New User"... Have I mis configured something? |
| 05:53.26 | thehar | perhaps |
| 05:53.29 | jasonpr | has anyone done a fax server in linux.... |
| 05:53.37 | *** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194) |
| 05:53.41 | thehar | yes. hylafax |
| 05:54.00 | jasonpr | don't you have to have zap channels for that to work? |
| 05:54.15 | jasonpr | hardlines n-such? |
| 05:54.33 | thehar | probably. but i have many pri |
| 05:54.36 | thehar | or. |
| 05:54.46 | thehar | airquote hardlines |
| 05:55.24 | coppice | jasonpr: what kind of server are you looking for, if you don't want hard lines? |
| 05:55.47 | jasonpr | I was hoping I could get something running on t38.... |
| 05:56.00 | jasonpr | I really have no clue how t38 works. |
| 05:56.27 | coppice | callweaver, asterisk 1.6, or hylafax + t38modem are options for that |
| 05:57.03 | jasonpr | asterisk 1.6 + fax == smoke flames and plenty of crashes |
| 05:57.08 | jasonpr | atleast for me |
| 05:57.22 | jasonpr | every time I call sendFax |
| 05:57.26 | jasonpr | it kills asterisk |
| 05:58.24 | jasonpr | Honestly I'd really rather pay a decent rate to use an XML webservice |
| 05:58.37 | jasonpr | but the only decent service out here is like $0.20/page |
| 05:59.08 | jasonpr | asterfax is kind of a mess to get setup |
| 05:59.50 | jasonpr | I'll have to hylafax a shot |
| 06:00.32 | *** join/#asterisk joako (n=joako@adsl-074-170-252-213.sip.gnv.bellsouth.net) |
| 06:04.10 | coppice | if asterisk 1.6 + fax == smoke flames you probably have multiple copies of spandsp installed. asterisk built with one, and the runtime is trying to use the other |
| 06:11.31 | jasonpr | ya know I think that's probably it.... thanks coppice |
| 06:20.12 | d-tech | anyone recommend a economical dual fxs with failover and iax support? |
| 06:21.56 | orkid | effin betamax up to their scamming ways again, some of the calls that were marked FREE suddenly changed to FUP and got charged. wtf |
| 06:22.18 | phix | I keep pressing <CRTL+ALT+WORK> button sequence but it isn't working yet :) |
| 06:22.43 | orkid | i dont have a work button |
| 06:22.46 | orkid | :) |
| 06:22.51 | d-tech | you forgot the SHIFT key! |
| 06:22.58 | *** join/#asterisk implicit (n=bayan@unaffiliated/implicit) |
| 06:24.51 | phix | d-tech: ah! thanx, that was the problem :) |
| 06:24.55 | phix | it is working now :P |
| 06:25.05 | echinos | Ok, so I have my * box announcing the caller ID before it answers the phone, but can I make the caller hear ringing while the System function is running? |
| 06:25.22 | phix | yes |
| 06:25.28 | phix | there is an option to do it |
| 06:25.32 | echinos | Right now they hear 5 or 6 seconds of silence, then they hear the Playback message |
| 06:25.33 | phix | I don't know what it is though |
| 06:25.41 | echinos | k |
| 06:25.48 | echinos | I'll go spelunking |
| 06:28.20 | echinos | so maybe use the ringing() cmd first.... lesee what I get with that |
| 06:30.02 | echinos | well, it's _better_, but I only hear one ring, and then I hear more silence... It at least provides feedback that something is happening |
| 06:34.40 | phpboy | LOL :( |
| 06:34.54 | *** join/#asterisk xxoxx (n=xxoxx@tor/regular/xxoxx) |
| 06:42.33 | *** join/#asterisk Bad_Robot- (n=Bad_Robo@76.173.219.25) |
| 06:46.37 | phix | I has scotch + sushi |
| 06:46.41 | phix | for the win |
| 06:49.04 | phpboy | I'm reading up on the benefits of drinking green tea |
| 06:49.32 | *** join/#asterisk sah-work (n=Bawbatos@adsl-76-203-173-209.dsl.pltn13.sbcglobal.net) |
| 06:57.34 | drmessano | I just got details banged out for a power efficient PBX |
| 06:58.34 | coppice | watt? |
| 06:59.28 | drmessano | Calls come in and the unused modem on a workstation sees the line ring... sends a Wake-On-Lan to the PBX, turning it on.. If the line is still ringing in 2 or 3 minutes, asterisk takes the call, passes it thru. When the call ends, a cron job, which is running every 30 seconds, checked for active calls using AMI and if none, shuts the PBX back down. |
| 06:59.32 | drmessano | Its brilliant |
| 07:00.42 | drmessano | I'm hoping to chop that 2 or 3 minutes down to a minute and a half, which will greatly reduce dropped calls.. but that's for later down the line |
| 07:01.58 | phpboy | hmmmm, it seems asterisk stops transmitting voice all of a sudden from SIP to PTSN |
| 07:02.11 | phpboy | but only from the SIP side, can anybody think of what that could be? |
| 07:02.30 | *** part/#asterisk FlyboySR22 (n=rsears@hq.fw.americanis.net) |
| 07:03.28 | phix | phpboy: nice, I can tell you the benefits of drinking scotch :) |
| 07:04.40 | phpboy | phix :D |
| 07:04.50 | phpboy | I need to figure out why SIP stops transmitting voice |
| 07:05.09 | phpboy | the fucked up part of it is, I can here the voices from both sides of the recording |
| 07:05.11 | phix | it is probably firewall / NAT related |
| 07:05.17 | phpboy | LAN |
| 07:05.28 | phix | oh |
| 07:05.37 | phpboy | What are the odds of it being my PRI cable? |
| 07:05.42 | phpboy | perhaps electric interference? |
| 07:06.11 | phix | maybe |
| 07:06.24 | phpboy | what are the odds though, vs it being asterisk? |
| 07:07.01 | phix | possible |
| 07:07.10 | phix | are you testing it with softphones? |
| 07:07.15 | phix | or hardware phones? |
| 07:07.18 | phpboy | nope, snom 300's |
| 07:07.24 | phix | <3 |
| 07:07.27 | phix | how much are those? |
| 07:07.32 | phix | are they worther buying? |
| 07:07.37 | phpboy | definitely |
| 07:07.46 | *** join/#asterisk amaache (n=maache@41.221.16.35) |
| 07:07.47 | phpboy | a touch expensive, well at least in my country |
| 07:07.54 | phpboy | but worth every cent |
| 07:08.37 | phpboy | I lie |
| 07:08.45 | phpboy | it's actually the PTSN to SIP that's the problem |
| 07:08.47 | phix | where are you from? |
| 07:09.06 | phpboy | I can here the person from the PSTN side speaking but the person on the SIP phone can't hear him/her |
| 07:09.25 | phpboy | South Africa |
| 07:09.29 | drmessano | NAT |
| 07:10.26 | phix | ah |
| 07:10.32 | *** join/#asterisk UQlev (n=kvirc@91.184.220.73) |
| 07:10.35 | phix | sounds NAT related |
| 07:10.48 | phix | phpboy: what model do you recomend? |
| 07:11.40 | phpboy | drmessano, phix: Can't be NAT, I don't use nat |
| 07:11.42 | phpboy | it's a LAN |
| 07:11.43 | phpboy | :( |
| 07:11.58 | phpboy | perhaps network related? |
| 07:12.02 | phpboy | it's a HUGE network |
| 07:12.10 | phpboy | some 300 handsets/computers connected |
| 07:12.24 | phpboy | phix: for hard phone? |
| 07:14.02 | coppice | if you call 300 handsets a HUGE network, what adjective do you use for China Mobile's 450M handsets on a network? |
| 07:14.27 | phpboy | coppice: In South Africa |
| 07:14.30 | phpboy | this is huge |
| 07:14.31 | phpboy | :( |
| 07:14.38 | phpboy | ok, I'll rephrase |
| 07:14.44 | phpboy | it's a big network? |
| 07:14.46 | phpboy | :P |
| 07:14.59 | phpboy | anyhoo, do you guys think it may be network related? |
| 07:16.38 | phix | phpboy: yes |
| 07:16.40 | phix | snom |
| 07:16.57 | phpboy | phix: for a desk phone, definitely the Snom 300 |
| 07:17.04 | phpboy | Grandstream is the DEVIL |
| 07:17.17 | phix | I cant find it on ebay :( |
| 07:17.18 | phpboy | phix: How can I confirm that it's a network problem though? |
| 07:17.36 | phpboy | I'm guessing you're from the state? |
| 07:17.39 | phpboy | *states |
| 07:17.44 | *** join/#asterisk jeffspeff (n=jeffspef@c-98-211-62-9.hsd1.ky.comcast.net) |
| 07:22.26 | phix | phpboy: wrong guess :) I am closer than you think |
| 07:22.42 | phix | in the southern hemisphere |
| 07:24.34 | *** join/#asterisk uyhfd (n=chatzill@bgl93-3-82-230-208-124.fbx.proxad.net) |
| 07:25.19 | phpboy | Zimbabwe? |
| 07:25.23 | phpboy | :P |
| 07:25.39 | phpboy | ah |
| 07:25.40 | phpboy | .au |
| 07:25.42 | phpboy | nice |
| 07:25.42 | phix | :P |
| 07:25.44 | phix | yup |
| 07:26.00 | phpboy | anyhoo, is there any way I can prove my SIP issues are network related? |
| 07:26.19 | *** join/#asterisk CrazyTux (n=brandon@ip68-111-67-4.oc.oc.cox.net) |
| 07:26.20 | phix | hmmmmm |
| 07:26.39 | phpboy | also considering this is a layer 1 network |
| 07:26.43 | phpboy | nothing advanced :( |
| 07:26.47 | phpboy | which I know is a problem |
| 07:27.09 | phix | wtf, 300 nodes on a layer 1 network? as in just hubs? no switches? |
| 07:27.10 | phpboy | considering it's sizer |
| 07:27.17 | phpboy | switches |
| 07:27.22 | phpboy | but on layer1 level |
| 07:27.37 | phix | the physical layer |
| 07:27.37 | phix | ? |
| 07:27.42 | phpboy | it would be 600+ nodes |
| 07:27.46 | phpboy | yeah |
| 07:27.49 | uyhfd | hi |
| 07:27.50 | phix | hmmm, switched network == data link |
| 07:27.57 | uyhfd | to all |
| 07:28.13 | phpboy | Layer 1 switch, i.e. NOTHING special |
| 07:28.20 | phpboy | we're moving to layer 3 switches |
| 07:28.23 | phpboy | but not yet :/ |
| 07:28.48 | phpboy | phix: you think this would be a problem? |
| 07:28.58 | phix | Layer 1 isn;t a switch, it is a hub :) |
| 07:29.09 | phpboy | then what is a basic switch? |
| 07:29.20 | phpboy | considering layer 2 is managed? |
| 07:29.59 | phix | no |
| 07:30.05 | phix | a basic switch is layer 2 |
| 07:30.10 | phix | a basic hub is layer 1 |
| 07:30.12 | phpboy | hmmmm |
| 07:30.20 | phix | layer 2 doesn't mean it is managed |
| 07:30.38 | phpboy | hmmm |
| 07:30.55 | phpboy | Then why does cisco for instance call the 2924XL switch Layer 2? |
| 07:31.00 | phpboy | and it happens to be manager |
| 07:31.01 | phpboy | managed |
| 07:31.07 | drmessano | O.o |
| 07:31.11 | phix | you can get a managed hub, managed means you have more control over the workings of the hub / switch, you can set different thingies on each port, do other snazzy things too |
| 07:31.39 | phpboy | ok, let's take a step back |
| 07:31.45 | phix | hehe |
| 07:31.51 | phpboy | I'm not too worried about switch terminology now |
| 07:32.03 | phpboy | I need to figure out what's up with this system |
| 07:32.14 | phpboy | so you think it's safe to bet it's network related? |
| 07:33.09 | phix | lol |
| 07:33.20 | phpboy | :( |
| 07:33.29 | phpboy | phix, speak to me |
| 07:33.30 | phpboy | :/ |
| 07:33.31 | phix | I DONT KNOW! :) do some tests |
| 07:33.37 | phix | install wireshark |
| 07:33.43 | phpboy | hmmm |
| 07:33.48 | phix | see if there are any known network issues |
| 07:33.58 | phix | then try and connect to certain ports that SIP uses |
| 07:34.40 | phix | yay! MacGyver made a new friend! |
| 07:35.14 | drmessano | Sounds like a Layer 8 problem to me |
| 07:35.36 | phpboy | drmessano: Elaborate please? |
| 07:36.27 | drmessano | phpboy: How old are you? |
| 07:37.13 | phpboy | drmessano: Does it matter? |
| 07:37.39 | drmessano | Nevermind then |
| 07:38.02 | phpboy | Did I just cut my nose to spite my face? |
| 07:39.39 | phpboy | drmessano: I'm 23 |
| 07:39.51 | phpboy | drmessano: Now, do you care to elaborate, please |
| 07:42.59 | *** join/#asterisk h-idrisi (n=h-idrisi@212.100.196.195) |
| 07:43.14 | orkid | dont mind him |
| 07:46.24 | drmessano | Troll |
| 07:47.13 | *** join/#asterisk PanGoat (n=PanGoat@node2.sensoryresearch.net) |
| 07:48.16 | *** join/#asterisk r0d3nt (i=astrutt@pinky.ratman.org) |
| 07:54.16 | *** join/#asterisk thansen (n=thansen@7.247.sfcn.org) |
| 07:56.30 | phpboy | drmessano: Let's just assume because I'm South African, I'm not well educated and you being the giving person you are, you'll help me see the proverbial light? |
| 07:57.04 | phpboy | drmessano: I would really appreciate it if you could help me here, please. |
| 08:02.09 | drmessano | I can't testify to how well educated you are, but certainly you show no lacking of it based on your use of proper punctuation and a better-than-5th-grade-vocabulary. |
| 08:02.31 | drmessano | Does your asterisk box have multiple NICs? |
| 08:04.14 | phpboy | It does, 1 NIC for LAN and the other for a direct SIP connection to another network (this is hooked up directly to the router connected to the external network) |
| 08:04.38 | drmessano | Sounds like you're having problems with reinvites |
| 08:04.55 | drmessano | I would suggest setting up wireshark and watching whats going on |
| 08:05.21 | phpboy | What would this entail? |
| 08:05.44 | drmessano | Setting up wireshark and then doing some watching of what is going on |
| 08:05.46 | phpboy | ok, I'm guessing I'd have to put wireshark on the asterisk server? |
| 08:05.51 | drmessano | NO! |
| 08:06.02 | drmessano | Google is your friend |
| 08:06.03 | phpboy | I've never used wireshark before :( |
| 08:06.07 | drmessano | Set up wireshark |
| 08:06.10 | drmessano | Go read |
| 08:06.16 | phpboy | I've already installed it on my local pc, playing around with it. |
| 08:06.30 | phpboy | drmessano: Anything else I may want to have a look at? |
| 08:07.07 | phpboy | or at least bear in mind? |
| 08:07.15 | drmessano | You can check the asterisk log for errors |
| 08:08.03 | phpboy | I've been looking through /var/log/asterisk/debug, nothing looks strange in there |
| 08:11.29 | phpboy | [Dec 20 10:09:21] WARNING[31532] ast_expr2.fl: If you have questions, please refer to doc/channelvariables.txt in the asterisk source. <--- that doesn't help too much |
| 08:11.41 | phpboy | where would I look for that in the configs? |
| 08:16.32 | phpboy | woops, I'm being stupid |
| 08:18.25 | *** join/#asterisk drmessano (n=nonya@pdpc/supporter/active/drmessano) |
| 08:19.24 | phpboy | :( reloaded asterisk and not getting anymore typos, but still getting errors in /var/log/asterisk/messages :/ |
| 08:29.56 | *** join/#asterisk SparFux (n=raoul@e182024204.adsl.alicedsl.de) |
| 08:30.44 | *** join/#asterisk flaccid (n=flaccid@127.185.233.220.exetel.com.au) |
| 08:31.04 | flaccid | excuse me if i sound like a n00b, but is asterisk all sweet on freebsd atm? |
| 08:39.32 | SparFux | Why noob? It's a valid question. |
| 08:41.50 | *** join/#asterisk reneger (n=reneger@p3EE2D7AB.dip.t-dialin.net) |
| 08:48.12 | PanGoat | flaccid: I can't answer with any authority, but did succesfully install and currently run it on several Mac OS X boxes FWIW. |
| 08:49.03 | PanGoat | the only hitch I ran into was that you have to use the ztdummy driver (if you don't have hardware) and want to do conferencing. |
| 08:49.37 | PanGoat | there are instructions for compiling, and I think even a FreeBSD binary |
| 08:53.22 | Subdolus | i have "Set(TIMEOUT(absolute)=2400)" in my context, how can i execute another command only when it times out and hungs up? |
| 08:53.35 | Subdolus | I tried ,t,Command(here) but it didnt go to that |
| 08:58.44 | flaccid | thanks PanGoat |
| 08:59.00 | flaccid | i shall install the freebsd port and go from there. what is fwiw? |
| 08:59.56 | PanGoat | For What Its Worth |
| 09:01.06 | flaccid | ah cheers |
| 09:01.11 | PanGoat | meaning "just a casual piece of information that you can take at your leisure". I'm just as much a asterisk noob most likely. |
| 09:02.16 | flaccid | i aint used it yet. can i just have a working modem in my computer and use that for hooking up pstn service to the pbx/asterisk ? |
| 09:05.30 | SwK | flaccid, no |
| 09:06.00 | phpboy | flaccid: depends on what you wanna do with it |
| 09:06.06 | phpboy | basic SIP, works just fine |
| 09:06.12 | flaccid | ok. how can i achieve this? i just want a pbx for home |
| 09:06.13 | SwK | you need a very specific modem with a very specific chip that was discontinued a while back for hooking you your phoneline... |
| 09:06.25 | SwK | you can get sip service from a variety of places onthe cheap tho |
| 09:06.31 | flaccid | oh. i have about 30 modems in a box but |
| 09:06.32 | phpboy | flaccid: start by installing the asterisk port and asterisk-addons port |
| 09:06.47 | flaccid | thanks phpboy and SwK |
| 09:17.13 | TrentCreek | sure |
| 09:22.29 | Subdolus | guys, how do i put the equivelent of "g" in Dial, in my call file? |
| 09:22.56 | Subdolus | so when a call that i made with the call file will continueto go on after it is hungup |
| 09:25.59 | *** join/#asterisk joako (n=joako@adsl-074-170-252-213.sip.gnv.bellsouth.net) |
| 09:49.40 | *** join/#asterisk amaache (n=maache@41.221.17.95) |
| 09:49.50 | amaache | Hi |
| 09:50.48 | amaache | Plz have u a good support sites for my univ telecom study :-) |
| 09:55.12 | amaache | Plz have u a good support for asterisk & co sites for my univ telecom study :-) |
| 09:56.57 | stintel | ~book |
| 09:56.58 | jbot | i heard 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 |
| 09:57.32 | tzafrir_laptop | amaache, have you tried some basic research? |
| 09:58.14 | tzafrir_laptop | e.g. feed the query "asterisk support" to your favorite search engine? |
| 09:59.00 | tzafrir_laptop | yahoo gives some relevant results there |
| 10:09.16 | *** join/#asterisk TOrrIeri (n=torrieri@nelug/crew/torrieri) |
| 10:14.21 | *** join/#asterisk amaache (n=maache@41.221.16.75) |
| 10:37.35 | *** join/#asterisk dsp2877 (i=dsp28777@118.100.44.178) |
| 10:37.47 | dsp2877 | hello all |
| 10:41.56 | dsp2877 | anyone here have much experience with chan_alsa? |
| 11:04.30 | TrentCreek | never heard of it |
| 11:05.23 | dsp2877 | lol ok |
| 11:05.39 | dsp2877 | its the sound/console driver for asterisk |
| 11:07.25 | TrentCreek | try again in a about 3-4 hours when more people are on |
| 11:07.32 | dsp2877 | okay |
| 11:07.46 | TrentCreek | most people on here are in the US |
| 11:07.55 | dsp2877 | ic |
| 11:08.02 | dsp2877 | ok got that |
| 11:12.05 | *** join/#asterisk magronez (n=eusei@unaffiliated/magrao/x-2903) |
| 11:12.48 | *** join/#asterisk Thorn (n=thorn@unaffiliated/thorn) |
| 11:31.16 | *** part/#asterisk tmiw (i=mooneer@voldemort.lifeafterking.org) |
| 11:42.18 | *** join/#asterisk amaache (n=maache@41.221.17.60) |
| 11:50.15 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
| 11:50.15 | *** mode/#asterisk [+o denon] by ChanServ |
| 12:15.02 | *** join/#asterisk Segnale007 (n=Pietro@host199-252-dynamic.35-79-r.retail.telecomitalia.it) |
| 12:16.53 | *** join/#asterisk DarkRift (n=dark@bas1-sthubert21-1279505630.dsl.bell.ca) |
| 12:20.40 | amaache | Hi all; do you know any support site of asterisk or trixbox? :-) |
| 12:23.25 | *** join/#asterisk troubled (n=troubled@unaffiliated/troubled) |
| 12:25.01 | *** join/#asterisk onats (n=onats@unaffiliated/onats) |
| 12:25.08 | onats | hi, is there a gui based config tool for asterisk? |
| 12:25.33 | onats | which version should i use btw? I only have a single fxo card at home, and i want to be able to use it as a small pbx? |
| 12:28.06 | *** join/#asterisk toot (n=chris@93-97-255-18.zone5.bethere.co.uk) |
| 12:43.30 | TrentCreek | onats: FreePBX |
| 12:44.10 | onats | ok thanks |
| 12:44.12 | yang | ~asterisk-gui |
| 12:44.13 | jbot | [~asterisk-gui] The Asterisk-GUI is an open source project lead by Digium. It's client-driven, and its only dependency is Asterisk itself. Check out the new version from svn here: $ svn co http://svn.digium.com/svn/asterisk-gui/branches/2.0 asterisk-gui-2.0. For support go to #asterisk-gui |
| 12:45.10 | joat | any opinions on the LInksys SPA3102? (i.e., is it worth the $30 a friend is asking?) |
| 12:45.11 | onats | yang, thanks! |
| 12:45.33 | joat | err.. s/LInksys/Linksys |
| 12:45.42 | joat | can't type! |
| 12:46.09 | yang | joat: You can get SIPURA's on EBAY for less than 10 usd |
| 12:46.27 | TrentCreek | i would be careful of Linksys stuff..those are usually locked |
| 12:46.33 | joat | this has one FXS and one FXO |
| 12:46.38 | joat | supposedly open |
| 12:46.52 | joat | he got it from voipsupply |
| 12:47.03 | TrentCreek | okay..then prob good |
| 12:47.09 | yang | joat: 30 USD is the usual price for it in the stores |
| 12:47.52 | joat | for the one with one FXS and one FXO? they go for around $70 |
| 12:48.19 | TrentCreek | I got a Linksys PAP2T with 2 lines, and you can have it use different providers for eahc line..only $50 a year ago from voipsupply |
| 12:48.56 | TrentCreek | it's based on Sipura..been happy with it |
| 12:49.33 | joat | liked my PAP2, lost it in a lightning strike... |
| 12:49.59 | yang | I saw one like that yesterday from the company trust - brand new for 20 usd |
| 12:50.15 | yang | I ll get it next time |
| 12:50.41 | TrentCreek | ohh.....nice price..i am looking to get more of them |
| 12:52.25 | yang | What i was curious about, if its possible to make the really old analog phones, which didn't have a tone dial to plug them to asterisk |
| 12:52.33 | TrentCreek | darn..it's in Europe..by the time I pay shipping..loss on converting to Euro, etc..It would be the same price as here |
| 12:53.28 | TrentCreek | yang: you dont plug them into Asterisk..you plug them into a A/D converter |
| 12:53.59 | yang | yes, i mean over that |
| 12:54.19 | yang | probably i would need an FXO/FXS card |
| 12:54.45 | yang | but you know those phones had a 3-pin plug |
| 12:54.52 | TrentCreek | yeah.not sure you can plug a rotary into a VOIP device |
| 12:55.02 | *** join/#asterisk gambler1 (n=mene_sun@dragan.eunet.yu) |
| 12:55.13 | TrentCreek | but why even do that when Tone ones can be had for $5? |
| 12:55.28 | yang | Simply out of curiosity |
| 12:56.13 | yang | If you do the calculation a sipura is more expenssive than the low-end VOIP phones (ebay) |
| 12:56.51 | TrentCreek | depends on what you define less expensive. |
| 12:57.03 | TrentCreek | You get what you pay for |
| 12:57.54 | TrentCreek | I had a cheap DLink..nothing but trouble. I think it is in some landfill right now |
| 12:58.35 | TrentCreek | so now I am up to a larger cost because of junk..so much for "more" expensive |
| 13:02.37 | yang | I agree with you, low-cost phones make you more troubles when speaking through them |
| 13:02.49 | yang | bad sound quality |
| 13:06.33 | TrentCreek | yes, I have been very happy with the sound quality. Just like a land line |
| 13:06.45 | yang | so which one do you use now? |
| 13:08.34 | yang | I have been using Grandstream fro the last year (trouble), and latelly I have been trying out Snoms |
| 13:12.53 | SwK | hint: just buy a polycom or a snome or a cisco hard phone ... its worth the extra money |
| 13:13.11 | SwK | s/snome/snom/ |
| 13:16.51 | yang | SwK: About Cisco's are the Asterisk (SIP) capable ? |
| 13:17.53 | joat | some are, some aren't (need to read the specs) |
| 13:19.11 | yang | linksys are among the top phones I read |
| 13:19.12 | SwK | yang, if you get they have SIP images available... like joat said... depends on the phone... |
| 13:19.21 | SwK | SPA phones are crap imho |
| 13:19.40 | yang | SwK: have you tried linksys? |
| 13:19.49 | SwK | been doing this for over 5 years ;) |
| 13:20.02 | yang | SwK: so, snoms are better in your opinion? |
| 13:20.28 | SwK | I rank them as follows Polycom, Snom, Cisco, and then down hill from there |
| 13:20.41 | yang | Snoms are also like the dual price of linksys |
| 13:20.43 | SwK | (some people will debate that top 3 ordering, but they are the best) |
| 13:20.53 | yang | ok |
| 13:20.59 | yang | What about AAstra ? |
| 13:21.04 | SwK | i loath them |
| 13:21.16 | yang | loath ? |
| 13:21.21 | SwK | hate |
| 13:21.23 | yang | ok |
| 13:21.39 | SwK | loathing ;) good english word for you... look it up heh |
| 13:22.37 | SwK | but AAstras work ok on a LAN dont try using them if NAT seperates them (unless the finally fixed that issue in the past 5 or 6 months) |
| 13:23.39 | SwK | they are ok quality... thing is I prefer to use something like Snom or Polycom as your configs are all the same for them basically and they offer a variety of phones from the 100 usd mark and up (and if you look just a little but you can find them cheaper then that |
| 13:24.14 | yang | hehe, I was also curious on testing out this (for video) http://www.provu.co.uk/ipvideo_bvp8882.html |
| 13:24.35 | SwK | that might be interesting |
| 13:24.42 | yang | I can get it for 15 EUR |
| 13:24.54 | TrentCreek | yang: I use Linksys PAP2T |
| 13:25.07 | SwK | cant remember the website but I just found some korean sip video phones I think the us distributor is going to send me for testing |
| 13:25.10 | *** join/#asterisk amaache (n=maache@41.221.16.132) |
| 13:25.16 | SwK | PAP2Ts are the best ATAs |
| 13:25.20 | joat | looks weird... usb for wifi dongle? |
| 13:25.31 | TrentCreek | SwK: Seems most of the discussion on here is about problems with Polycoms |
| 13:25.31 | yang | SwK: however i am afraid of the low audio quality |
| 13:25.38 | SwK | you cant go wrong with a sipura or linksys ATA for an analog phone |
| 13:26.31 | SwK | TrentCreek, every phone has problems... they are software driven... I've been using polycoms w/ asterisk for over 5 years and where i've had the occation software issues, once you have them setup and working they just work |
| 13:26.45 | yang | the good quality video phones still range around 300 EUR + |
| 13:26.47 | SwK | same thing can be said of snom and cisco |
| 13:26.54 | TrentCreek | groovy |
| 13:27.21 | SwK | yeah thats any good quality video phone... thats why I wanna see what these korean things are like... they look pretty nice, but the proof is actually laying hands on one |
| 13:27.46 | SwK | TrentCreek, the problem with alotta this stuff is people have to fix whats not broken if you know what I mean |
| 13:27.58 | yang | SwK: you should write some reviews, if you tested so many phones :) |
| 13:28.15 | SwK | you have a nice stable system, and some new bell or whistle comes out and they go screwing with things and then all of a sudden problems all over the place |
| 13:28.18 | joat | yang: the provu comes with PTZ as an option? even weirder |
| 13:28.31 | SwK | yang, i'm not writer heh |
| 13:28.33 | yang | joat: I don't know what PTZ is |
| 13:28.43 | SwK | pan tilt zoom |
| 13:28.44 | joat | pan tilt zoom |
| 13:28.49 | joat | camera control |
| 13:29.06 | SwK | pan == left right, tilt = up down and well you know what zoom lenses are |
| 13:29.07 | yang | I tried video with the Ekiga soft-client so far so good |
| 13:29.17 | SwK | I use eyebeam on my mac all the time |
| 13:29.27 | SwK | that works with a number of video soft and hardphones |
| 13:29.35 | joat | tried ekiga to grandstream 2000 video... |
| 13:29.41 | SwK | Polycom makes a super nice video conf phone |
| 13:29.42 | joat | the grandstream worked nicely |
| 13:29.51 | joat | ekiga ate up a lot of processing power |
| 13:29.53 | yang | joat: you mean GXV-3000 ? |
| 13:29.59 | joat | ah... yeah |
| 13:30.17 | joat | had it on loan from work |
| 13:30.28 | yang | joat: I hate GS, This week 3 firmware upgrades went corrupt |
| 13:30.31 | SwK | hahaha |
| 13:30.40 | TrentCreek | SwK: yeah I knwo what you mean |
| 13:30.56 | joat | yikes |
| 13:30.58 | SwK | I have a handytone286 or whatever its called.. |
| 13:31.14 | yang | I think there is no way to restore the firmware back, only factory reset |
| 13:31.17 | SwK | this thing feels like it was made in china for the toy markets of walmart |
| 13:31.29 | yang | SwK: yep |
| 13:31.34 | joat | can't bitch about the one bt200 on my desk (it was free) |
| 13:31.41 | SwK | heh |
| 13:31.55 | SwK | i have a polycom 550 on my desk here at home and its a super nice phone |
| 13:32.04 | yang | joat: I would assume that BT-series are much better than the GXP-series |
| 13:32.13 | SwK | hahaha |
| 13:32.17 | SwK | the bt series are crap |
| 13:32.17 | joat | heh |
| 13:32.25 | joat | low end |
| 13:32.28 | SwK | super low end |
| 13:32.36 | joat | it's actually my alarm clock |
| 13:32.40 | SwK | like so low end walmart would say thats too low end for us |
| 13:32.43 | yang | I purchased a BT-100 , will see how it works |
| 13:33.14 | joat | allison reads the weather forecast every morning |
| 13:33.24 | joat | that's about all the use it sees |
| 13:33.29 | SwK | http://cgi.ebay.com/Polycom-Soundpoint-IP-550-Phone-Four-Lines-SIP-VoIP_W0QQitemZ290283276422QQcmdZViewItemQQptZCOMP_Telecom_IP_Telephony?hash=item290283276422&_trksid=p3286.c0.m14&_trkparms=72%3A1234|66%3A2|65%3A12|39%3A1|240%3A1318|301%3A1|293%3A1|294%3A50 |
| 13:33.36 | SwK | see thats apretty good price on that |
| 13:33.38 | yang | there is no market for voip phones in my country, only imports mailorders |
| 13:33.50 | SwK | yang, what country is that? |
| 13:34.31 | yang | SwK: uh, weird design :) |
| 13:34.37 | yang | Slovenia |
| 13:36.14 | yang | Heh, I was impressed by the SNOM 820 it has some sort of a magnet in the headset (no hook) |
| 13:38.00 | yang | joat: I was just thinking about the weather forecast station, which would give me the information about local weather, over SIP ... they usually do that on commercial numbers here |
| 13:38.28 | joat | yang: that provu really has a lot of odd options... looks like what'd you'd get if you tossed an x-10 camera, a grandsteam, and a video sender into a blender |
| 13:38.47 | joat | yang, i scrape the noaa feed for the local airport |
| 13:39.07 | joat | it's a horrible kluge but it works |
| 13:39.41 | yang | joat: could you do a sip trunk for a test ? |
| 13:39.42 | joat | s/grandsteam/grandstream/ |
| 13:41.35 | SwK | I dont thing they have NOAA feed for Slovenia tho heh |
| 13:41.54 | toot | ah the snom 820 is good? ain't heard a review yet - cool |
| 13:43.38 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 13:44.55 | *** join/#asterisk slima (i=slima@unaffiliated/slima) |
| 13:45.23 | yang | toot: I haven't done much testing on it yet |
| 13:45.41 | yang | toot: but I like it's design |
| 13:46.12 | yang | toot: However its a phone, that only a few could afford buying in my country |
| 13:46.17 | *** join/#asterisk error404notfound (n=shoaibi@115.186.130.39) |
| 13:46.40 | toot | yeah - while i love snoms most of our clients are loving grandstream ata's atm :) |
| 13:46.56 | yang | toot: Well my boss is all HOT on Grandstreams |
| 13:47.09 | error404notfound | I get "[Dec 20 18:44:25] NOTICE[35543] chan_sip.c: Registration from '"User1" <sip:user1@192.168.20.41>' failed for '192.168.20.204' - Wrong password" what could be reason? password is correct, I have checked |
| 13:47.12 | joat | noaa didn't have it but weather underground did |
| 13:47.23 | joat | can scrape that |
| 13:47.52 | joat | there appears to be an rss feed also (not sure how often that's updated though) |
| 13:47.56 | gambler1 | yang: since your country is close to mine, can you tell me the price for snom 820 (if you know) |
| 13:48.14 | yang | gambler1: Around 400-450 I think |
| 13:49.04 | gambler1 | yang: tnx, it's less expensive then I thought |
| 13:49.25 | yang | gambler1: you can have 12 extensions on it |
| 13:49.37 | yang | I doubt that you will need those |
| 13:49.53 | toot | although i am biased i loved the openvpn support in the snoms |
| 13:50.20 | gambler1 | yang: well, I need one phone for testing, I have now 4 SPA942 on my desk.. |
| 13:50.36 | yang | toot: Its also a phone that has the OCS (M$) support |
| 13:52.33 | toot | that would be bad for me business :) |
| 13:52.35 | error404notfound | anyone? |
| 13:53.09 | yang | error404notfound: maybe your password is in the wrong field, check also in your sip.conf if the passwords match |
| 13:53.25 | error404notfound | yang: it does... |
| 13:53.28 | *** join/#asterisk qdk_ (n=qdk@78.156.208.174.bredband.3.dk) |
| 13:53.36 | [TK]D-Fender | error404notfound: It isn't lying so whatever you this is right is either wrong, or not actually in effect |
| 13:54.16 | dacs | Morning all |
| 13:55.14 | dacs | got 2 hrs to read and implement stuff before kido wakes up! |
| 13:56.13 | error404notfound | password is correct :( |
| 13:58.12 | yang | error404notfound: then its in the wrong field of your phone GUI |
| 14:05.21 | error404notfound | I am checking using the snom's web ui... |
| 14:05.26 | error404notfound | so there is no way.. |
| 14:06.26 | error404notfound | even installed the unhide-passwords extension for firefox and re-re-verified |
| 14:06.37 | gambler1 | I just take a look at user manual for one sip provider, and it seems that they recommend to reregister every 180 sec? Is there any recommended value for reregistration? For now, I only have seen a default of 3600 sec |
| 14:07.37 | *** join/#asterisk mog (n=mog@c-68-62-172-152.hsd1.al.comcast.net) |
| 14:07.37 | *** mode/#asterisk [+o mog] by ChanServ |
| 14:07.45 | [TK]D-Fender | error404notfound: You aren't showing us anything useful to help you with. |
| 14:08.26 | error404notfound | [TK]D-Fender: what can I show you? I told you the log message |
| 14:08.33 | [TK]D-Fender | gambler1: Normally shouldn't matter. High frequency re-reg is for those with connection / NAT issues, etc |
| 14:08.50 | [TK]D-Fender | eoorboth sets fo configs, SIP debug of the failed attempts |
| 14:09.05 | [TK]D-Fender | error404notfound: Both sets of configs, SIP debug of the failed attempts |
| 14:09.22 | error404notfound | SIP debug? |
| 14:09.51 | error404notfound | can asterisk work with empty password? |
| 14:09.55 | [TK]D-Fender | error404notfound: yes, that magical thing at CLI that sho0ws you the whole SIP packet as is sent & received @ * |
| 14:10.02 | gambler1 | [TK]D-Fender: tnx |
| 14:10.10 | *** join/#asterisk russellb (n=russell@asterisk/digium-open-source-team-lead/russellb) |
| 14:10.10 | *** mode/#asterisk [+o russellb] by ChanServ |
| 14:10.18 | [TK]D-Fender | error404notfound: The rest of the world is able to configure their phones just fine. |
| 14:14.33 | error404notfound | I totally changed my sip and extension conf file, reassign some extensions, changed passwords, do I need some extra step to make those conf active except restarting asterisks? |
| 14:17.34 | *** join/#asterisk jicksta (n=jicksta@c-24-6-87-187.hsd1.ca.comcast.net) |
| 14:17.44 | [TK]D-Fender | error404notfound: You don't need to completely restart * for SIP changes to take effect |
| 14:18.08 | [TK]D-Fender | error404notfound: for whatever "reassign some extensions" means |
| 14:18.46 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 14:23.59 | *** part/#asterisk russellb (n=russell@asterisk/digium-open-source-team-lead/russellb) |
| 14:31.02 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.165) |
| 14:32.37 | *** join/#asterisk Subdolus (n=subby@subby.afraid.org) |
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| 14:42.27 | *** join/#asterisk fun330 (n=manning_@224.217.188.72.cfl.res.rr.com) |
| 14:43.56 | fun330 | i want to set up a voip proxy for billing purposes would asterisk be the best way or should i think about freeswitch? |
| 14:55.32 | *** join/#asterisk drmessano (n=nonya@pdpc/supporter/active/drmessano) |
| 14:57.45 | *** join/#asterisk dacs (n=chatzill@unaffiliated/dacs) |
| 14:58.27 | dacs | guy anyone with much luck unlocking linksys PAP2 firm 3.1.9(LSc) |
| 14:58.58 | dacs | it keep asking for admin password when i try to tftp upgrade |
| 15:03.38 | dacs | anyone? |
| 15:09.39 | *** join/#asterisk amaache (n=maache@41.221.16.120) |
| 15:16.56 | [TK]D-Fender | dacs: unlock PAP2 3.1.9 |
| 15:17.00 | [TK]D-Fender | dacs: JFGI |
| 15:24.45 | *** join/#asterisk amaache (n=maache@41.221.17.120) |
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| 16:09.51 | drmessano | dacs: You're doing it wrong |
| 16:10.07 | drmessano | I've unlocked dozens of 3.1.9 boxes |
| 16:10.16 | *** join/#asterisk ice_croft (n=ice_crof@81.26.135.117) |
| 16:10.26 | ice_croft | hi guys |
| 16:10.34 | *** join/#asterisk fiXXXerMet (n=kyle@cmu-24-35-53-185.mivlmd.cablespeed.com) |
| 16:11.44 | ice_croft | Fender, im gettin this on remote IAX2 peer while callin to it: http://pastebin.ca/1290212 |
| 16:11.55 | ice_croft | help plz. context is correct |
| 16:12.53 | ice_croft | TK]D-Fender> peers and regs r correct too |
| 16:13.51 | ice_croft | TK]D-Fender> but the call cant make it even to it's dialplan context |
| 16:16.03 | *** join/#asterisk drmessano^ (n=nonya@pdpc/supporter/active/drmessano) |
| 16:19.33 | ice_croft | oh nevermind |
| 16:19.53 | *** join/#asterisk Carlos_PHX (n=Carlos@ip68-3-162-244.ph.ph.cox.net) |
| 16:19.54 | ice_croft | i found that iax2 peer context is always "default". damn |
| 16:27.33 | eppigy | hello |
| 16:27.35 | eppigy | i am dave |
| 16:29.11 | ice_croft | Rejected connect attempt from xx.xx.xx.xx, requested/capability 0x4/0xe004 incompatible with our capability 0xe703. |
| 16:29.19 | ice_croft | what does it mean? |
| 16:29.23 | ice_croft | im stuck ( |
| 16:31.04 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 16:35.55 | *** join/#asterisk propellerhead (n=yogurt2u@host79.190-30-199.telecom.net.ar) |
| 16:46.47 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 16:47.43 | fiXXXerMet | Hi everyone. After updating my centos box (yum update), asterisk is no longer working. Trying to do a test with ekiga softphone, I can register my extension. But when I try to place a call to another extension, I am getting "Call from '5107' to extension '5001' rejected because extension not found." |
| 16:49.06 | TrentCreek | ice_croft: seems like in compatable codec |
| 16:49.22 | *** join/#asterisk codefreeze-lap (n=murf@72.21.67.40) |
| 16:50.58 | fiXXXerMet | Incomming calls (from the outside) seem to work |
| 16:51.06 | fiXXXerMet | But outgoing does not |
| 16:52.00 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 16:53.13 | fiXXXerMet | Any other information needed to help me figure this out? |
| 16:56.02 | Dovid | fiXXXerMet: it is something in your dial plan that changed. Do you have trixbox or freepbx ? |
| 16:56.33 | *** join/#asterisk d3wayne (n=dwayne@c-76-29-245-9.hsd1.al.comcast.net) |
| 16:56.33 | *** mode/#asterisk [+o d3wayne] by ChanServ |
| 16:58.08 | fiXXXerMet | Dovid: Yes, trixbox (that was updated as well) |
| 16:58.27 | Dovid | then you need to as in #trixbox |
| 16:58.47 | fiXXXerMet | That place is useless :( |
| 17:00.14 | jaytee | it may be useless but it's the preferred support channel for useless derivatives of Asterisk |
| 17:03.35 | *** join/#asterisk phix (n=threat@123-243-44-131.tpgi.com.au) |
| 17:04.07 | *** join/#asterisk Bad_Robot- (n=Bad_Robo@cpe-76-173-219-25.socal.res.rr.com) |
| 17:07.46 | *** join/#asterisk Daejeo (n=chatzill@118.221.248.67) |
| 17:13.33 | drmessano^ | Ouch |
| 17:13.39 | drmessano^ | Yum update on a trixbox |
| 17:13.44 | drmessano^ | Good luck with that |
| 17:14.20 | drmessano^ | Thats the easiest way to absolutely bomb your box out |
| 17:14.21 | jaytee | linux+asterisk+updates=high probability of breakage |
| 17:15.10 | jaytee | trixbox+updates=guaranteed breakage |
| 17:15.16 | eppigy | hi probability of recompile |
| 17:15.19 | eppigy | YA FEEL ME |
| 17:17.05 | drmessano^ | or complete reinstall |
| 17:17.19 | *** join/#asterisk amaache (n=maache@41.221.17.145) |
| 17:17.28 | *** join/#asterisk russellb (n=russell@asterisk/digium-open-source-team-lead/russellb) |
| 17:17.28 | *** mode/#asterisk [+o russellb] by ChanServ |
| 17:20.11 | Daejeo | does any have an experience of using a bluetooth device(cellphone) as FXO or FXS channel? |
| 17:21.24 | Daejeo | can any recommend a bluetooth adapter for using with asterisk ? |
| 17:21.47 | gambler1 | hmmm I have updated centos and * many times and still have no luck of breakage :) |
| 17:22.54 | jaytee | gambler1, when you updated centos were there kernel updates? did you have to do a recompile? because most of us do when there's a kernel update or major dependency updates. |
| 17:23.34 | jaytee | but once you update and recompile * is usually good to go because updates won't break your dialplan in asterisk like they seem to all too often in trixbox |
| 17:25.07 | gambler1 | jaytee: hmmmmm no, I have updated linux boxes with and without kernel updates and still have no problems with * (1.4 and 1.6) |
| 17:25.18 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 17:25.38 | jaytee | gambler1, using zaptel or dahdi? |
| 17:25.59 | gambler1 | jaytee: but, it does not mean that there will be no problems in the future or that I was lucky enough... |
| 17:26.15 | gambler1 | jaytee: no.. neither of them |
| 17:26.29 | drmessano^ | The issue with trixbox is not the CentOS updates |
| 17:27.02 | jaytee | there ya go. if you run a hybrid pbx that uses PRI or analog as well as SIP or uses ztdummy or dahdi_dummy for timing for MeetMe then a kernel update will break it. |
| 17:27.06 | drmessano^ | Its the updates pulled from the Fonality repos, like new Asterisk RPMs that are missing config files, overwrite config files, etc |
| 17:27.12 | drmessano^ | Nothing to do with the OS itself |
| 17:27.23 | jaytee | drmessano^, no, it's the crap from Trixbox that overwrites it's own configs |
| 17:27.58 | esaym | I have a question with priority jumping. I have this setup: http://pastebin.com/m4700452b I am using lookupblacklist to block calls. If a number is in the database, lookupblacklist will jump to prio 102. But also if the line is busy, doesn't voicemail jump to prio 102? How do I change that so they don't conflict? |
| 17:28.02 | jaytee | because it's too stupid to go, "oh! a preexisting install! I should backup the config and then restore it when I'm done updating" |
| 17:28.03 | uyhfd | kjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjj |
| 17:28.19 | jaytee | nkkkkjjjjjjjjjjjjjjjjjjjjjj |
| 17:28.27 | drmessano^ | Thats what I said |
| 17:28.34 | eppigy | thats what she said |
| 17:28.42 | jaytee | drmessano^, yes, that's what you said |
| 17:28.54 | jaytee | and I was agreeing with you |
| 17:28.59 | gambler1 | esaym: use gotoif instead |
| 17:29.44 | drmessano | I was confused by the "no" bit.. I was expecting a counter of what I said from there on. Interesting tactic.. I will have to remember that.. |
| 17:30.28 | drmessano | Kinda like when you get into a fight with someone.. what is the first thing you should do? |
| 17:30.30 | jaytee | drmessano, sorry my english not so good! :-) |
| 17:30.34 | drmessano | Punch yourself in the face.. |
| 17:30.49 | gambler1 | :)))) |
| 17:30.55 | drmessano | Because then the dude is thinking "Shit, if hes gonna do that to himself, WTF is he gonna do to me???" |
| 17:30.55 | esaym | gambler1: hmm never heard of that, ty, I will look into it |
| 17:31.41 | drmessano | Slamming your face on a nearby object would have the same effect |
| 17:31.46 | jaytee | esaym, might want to look at using gotoif and labeled priorities to get away from priority jumping. IIRC it's getting "phased out" of asterisk. |
| 17:32.18 | drmessano | "CRAZY?? CRAZY?? NOBODY HERE IS CRAZY, FRED FLINTSTONE BARNEY RUBBLE BOWOOOOBAH" |
| 17:32.44 | gambler1 | esaym: take a look at http://downloads.oreilly.com/books/9780596510480.pdf |
| 17:33.33 | jaytee | wow! there's a book on Asterisk????? who knew!!!! |
| 17:33.46 | jaytee | ~book |
| 17:33.47 | jbot | from memory, 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 |
| 17:34.29 | drmessano | Im gonna get my name legally changed to Z4QQQ Batman Symbol |
| 17:34.38 | jaytee | lol |
| 17:34.42 | Dovid | haha |
| 17:34.44 | gambler1 | jaytee: ok, ok... I am the newcomer here... |
| 17:34.49 | *** join/#asterisk russellb (n=russell@asterisk/digium-open-source-team-lead/russellb) |
| 17:34.49 | *** mode/#asterisk [+o russellb] by ChanServ |
| 17:35.17 | Z4QQQ_Batman_Sym | Hmmm |
| 17:35.18 | jaytee | I got a Merry Yule "Cthulu" card today from a friend. "Have a Merry Yule or Cthulu will eat you!" |
| 17:35.31 | jaytee | almost as good as a FSM card :-) |
| 17:38.07 | jaytee | looks like we'll need to post a RFC for a new Extended ASCII set that includes the Batman symbol. Or at least a new font like Wingdings but better |
| 17:38.41 | eppigy | true |
| 17:38.57 | drmessano | Well, I think its damn stupid that the batman symbol can't be used in an Asterisk dialplan |
| 17:40.11 | drmessano | I should write a patch giving Batman() the same function as Hangup() |
| 17:40.35 | ScribbleJ | Oh boy. |
| 17:41.32 | drmessano | Answer, do stuff, do stuff, do stuff, BATMAN |
| 17:42.17 | jaytee | LOL |
| 17:43.02 | x86 | exten => s,1,Answer() exten => s,2,??? exten => s,3,Profit() |
| 17:43.18 | drmessano | FTW |
| 17:43.25 | drmessano | Now that, is win |
| 17:43.30 | *** join/#asterisk interfaithquest (n=chatzill@ip67-88-184-130.z184-88-67.customer.algx.net) |
| 17:43.31 | x86 | yay :) |
| 17:43.38 | x86 | took me long enough to get a win :P |
| 17:44.02 | x86 | err, where's my karma increment? hehe |
| 17:44.25 | drmessano | You just moved from -75 to -73 |
| 17:44.27 | drmessano | So YAY |
| 17:44.59 | drmessano | It could be worse |
| 17:45.01 | jaytee | wait! you forgot the really important one! exten => s,1,Answer(), exten => s,2,DoSomethingCool(), exten => s,3,DoubleBill(), exten => s,4,Batman() |
| 17:45.24 | drmessano | [TK]D-Fender's karma is so low, I have to wrap his in an electric blanket to keep it from freezing |
| 17:45.25 | jaytee | ~karma jaytee |
| 17:45.25 | jbot | jaytee has neutral karma |
| 17:45.45 | drmessano | ~karma drmessano |
| 17:45.45 | jbot | drmessano has karma of 1 |
| 17:45.51 | jaytee | what? after all the time I've spent in here giving out bad advice I'm still neutral? |
| 17:45.55 | drmessano | How the hell did that happen? |
| 17:46.20 | interfaithquest | hello telco joe , can an ATA device transfer a call via asterisk ? using features.conf |
| 17:46.21 | x86 | ~karma x86 |
| 17:46.21 | jbot | x86 has karma of -2 |
| 17:46.22 | jaytee | ~karma [TK]D-Fender |
| 17:46.22 | jbot | [tk]d-fender has karma of 10 |
| 17:46.28 | x86 | jbot-- |
| 17:46.29 | drmessano | Either you suck and lost 1 from the default of 1, or someone screwed up and gave me 1 |
| 17:46.57 | drmessano | ~karma gambler1 |
| 17:46.57 | jbot | gambler1 has neutral karma |
| 17:46.59 | jaytee | no more botsnacks for jbot from this guy |
| 17:47.02 | drmessano | HA |
| 17:47.06 | drmessano | Default is neutral |
| 17:47.10 | drmessano | Someone GAVE me a 1 |
| 17:47.19 | drmessano | What sort of dumba... |
| 17:47.40 | jaytee | I don't even know how to "give" karma |
| 17:47.51 | drmessano | ~addkarma jaytee |
| 17:47.57 | drmessano | ~givekarma jaytee |
| 17:48.03 | drmessano | ~karma +1 jaytee |
| 17:48.03 | jbot | +1 jaytee has neutral karma |
| 17:48.07 | drmessano | LOL |
| 17:48.13 | drmessano | ~karma jaytee +1 |
| 17:48.13 | jbot | jaytee +1 has neutral karma |
| 17:48.21 | drmessano | ~karma drmessano3000 |
| 17:48.21 | jbot | drmessano3000 has neutral karma |
| 17:48.36 | jaytee | maybe only ops can give karma |
| 17:48.51 | jaytee | ~karma |
| 17:48.51 | jbot | jaytee has neutral karma |
| 17:49.06 | jaytee | jbot karma |
| 17:49.06 | jbot | jaytee has neutral karma |
| 17:49.10 | jaytee | jeez |
| 17:49.23 | jaytee | ~help karma |
| 17:49.33 | d00gster | gents how do I fix asterisk time? |
| 17:49.59 | jaytee | jbot> Karma is a community rating system. Use "X++" to increase the karma, or "X--" to decrease it. Ask for ratings using "karma for X?" |
| 17:49.59 | jbot | is a community rating system. use "x++" to increase the karma, or "x--" to decrease it. ask for ratings using "karma for x?" has neutral karma |
| 17:50.19 | jaytee | ~karma drmessano X++ |
| 17:50.19 | jbot | drmessano x++ has neutral karma |
| 17:50.25 | eppigy | lol |
| 17:50.58 | jaytee | ~karma drmessano++ |
| 17:50.58 | jbot | drmessano++ has neutral karma |
| 17:51.03 | jaytee | wtf |
| 17:51.11 | russellb | jbot: jaytee++ |
| 17:51.13 | gambler1 | so... if veryone is in the mood can I ask one question, that I can'f find a good answer? |
| 17:51.25 | drmessano | ~x++ jaytee |
| 17:51.28 | russellb | jbot: karma for jaytee ? |
| 17:51.28 | jbot | for jaytee ? has neutral karma |
| 17:51.32 | russellb | jbot: karma for jaytee |
| 17:51.32 | jbot | for jaytee has neutral karma |
| 17:51.32 | jaytee | ~karma jaytee |
| 17:51.32 | jbot | jaytee has karma of 1 |
| 17:51.36 | russellb | there. |
| 17:51.37 | gambler1 | not veryone but everyone |
| 17:51.49 | jaytee | ah, I see now |
| 17:51.51 | eppigy | gambler1: out with it bro |
| 17:51.51 | drmessano | ~x++ jaytee |
| 17:51.54 | jaytee | thanks russellb |
| 17:51.57 | russellb | np |
| 17:52.01 | drmessano | ~karma jaytee |
| 17:52.01 | jbot | jaytee has karma of 1 |
| 17:52.18 | jaytee | jbot: drmessano++ |
| 17:52.21 | russellb | ~karma russellb |
| 17:52.21 | jbot | russellb has neutral karma |
| 17:52.26 | russellb | ~karma drumkilla |
| 17:52.26 | jbot | drumkilla has karma of 8 |
| 17:52.29 | russellb | ooh |
| 17:52.41 | gambler1 | How the hell I can get a list of active (registered users) when I use dynamic realtime? |
| 17:52.43 | jaytee | they should be the same |
| 17:53.17 | russellb | gambler1: query your database i guess |
| 17:53.46 | jaytee | russell should have a karma of at least 50 since he actually creates new apps and functions and fixes broken ones. |
| 17:53.58 | gambler1 | russellb: I know you are a great dev but that not possible if the user does not say bye... :( |
| 17:54.31 | russellb | what do you mean by "say bye" ? |
| 17:54.31 | gambler1 | and using sip timers does not update the db as I was expected... |
| 17:54.44 | russellb | you were asking about registration status, right? |
| 17:54.58 | jaytee | bbiab, gotta run an errand |
| 17:55.13 | *** join/#asterisk pikachu2000 (n=pikachu2@196-209-197-110-rrdg-esr-2.dynamic.isadsl.co.za) |
| 17:55.15 | gambler1 | say.. I have ext 100 and register on my * box, then my adsl connection goes down.. how * knows that I am not online anymore? |
| 17:55.33 | russellb | your registration will time out |
| 17:55.42 | eppigy | in 3600 seconds it will |
| 17:55.56 | gambler1 | yes... but thats not online users page anymore... |
| 17:56.11 | eppigy | lower re-register time |
| 17:56.26 | eppigy | to a margin of error you are comfortable with |
| 17:57.29 | gambler1 | thats the option but I was expecting sip timers would do that, because I can't afford to call XXXX clients to change the sip settings... |
| 18:00.49 | gambler1 | eh, sorry if I am hmmmm a "worst kind of user"... still tnx for your time :) |
| 18:01.44 | russellb | isn't there an option in sip.conf to force a minimum registration time? |
| 18:02.08 | eppigy | gambler1: select * from agent_table where fullcontact = 'NULL' |
| 18:02.48 | gambler1 | russellb: not that I am awareoff, but since you say it must be there.. thank you. |
| 18:03.05 | russellb | well, i may be crazy, i can't remember everything that's there ... |
| 18:03.12 | russellb | but it sounds familiar |
| 18:03.16 | russellb | i may be thinking of something else |
| 18:03.24 | gambler1 | eppigy: yes, thats the query we are using |
| 18:03.24 | eppigy | heard it in a dream |
| 18:03.40 | eppigy | well whats the deal dog |
| 18:03.46 | eppigy | maybe i am not understanding you |
| 18:03.51 | gambler1 | :)))))) I will take a look |
| 18:03.55 | eppigy | YEAH SON |
| 18:04.54 | gambler1 | eppigy: maybe, but still if the minimum registration time exist in sip.conf and is used to force clients to reregister then... there is np at all :) |
| 18:05.06 | eppigy | awesome |
| 18:05.35 | russellb | maxexpiry=3600 ; Maximum allowed time of incoming registrations |
| 18:05.35 | russellb | <PROTECTED> |
| 18:05.49 | eppigy | maxexpirey=3600 |
| 18:05.49 | eppigy | defaultexpirey=3600 |
| 18:05.53 | eppigy | you bastard |
| 18:05.58 | russellb | i found it first, i win :-p |
| 18:06.25 | eppigy | 8[] |
| 18:06.28 | *** part/#asterisk russellb (n=russell@asterisk/digium-open-source-team-lead/russellb) |
| 18:06.31 | *** join/#asterisk russellb (n=russell@asterisk/digium-open-source-team-lead/russellb) |
| 18:06.31 | *** mode/#asterisk [+o russellb] by ChanServ |
| 18:06.36 | russellb | i hate this client, ugh |
| 18:07.46 | eppigy | what are you using? |
| 18:07.58 | d00gster | gents how do I fix asterisk time? |
| 18:07.59 | gambler1 | russellb: I was just looking at that at the same time :D ok, since you are the winner, all the money goes to your bank account :) |
| 18:08.09 | russellb | yay |
| 18:08.21 | russellb | d00gster: asterisk doesn't have time, it uses the system time. |
| 18:08.23 | eppigy | d00gster: ntpd |
| 18:08.26 | eppigy | BOOYA |
| 18:08.34 | d00gster | my system time is correct |
| 18:08.45 | d00gster | I'll check again |
| 18:08.49 | russellb | then your time zone configuration is probably wrong |
| 18:08.56 | gambler1 | d00gster: problem with time in cdr ? |
| 18:09.03 | d00gster | voicemail |
| 18:09.14 | russellb | see timezone configuraiton in voicemail.conf |
| 18:09.52 | drmessano | HappyClownPBX doesn't use timezones.. it's all based on Phangal Coordinated Time |
| 18:10.09 | eppigy | it just got real |
| 18:10.24 | drmessano | Speaking of which, it's almost 15Z65.. I have like 25F before I need to go to the store |
| 18:10.53 | drmessano | O.o |
| 18:11.01 | eppigy | approximately 14kg away? |
| 18:11.11 | drmessano | Asterisk handles PCT just fine, btw |
| 18:11.31 | drmessano | ~HappyClownPBX |
| 18:11.31 | jbot | [HappyClownPBX] is currently in closed beta, approaching 12GB in size, uses Asterisk for it's core, it pwns, is also now compatible with the Diahatsumashiniriki Keyotason 200LP-A11 SIP phone |
| 18:12.14 | d00gster | russellb, can you give me an example of how voicemail.conf sets time? |
| 18:12.38 | russellb | d00gster: did you look at the sample config? |
| 18:13.29 | russellb | I'm not here to hand hold :) |
| 18:13.32 | eppigy | ; tz=central ; Timezone from zonemessages above. Irrelevant if envelope=no. |
| 18:13.33 | d00gster | found one online. unfortunately I started off with freepbx so it's not standard |
| 18:14.19 | russellb | you can check the book, or check configs/voicemail.conf.sample in the asterisk source tree. |
| 18:15.26 | gambler1 | after first question, I feel a little (more than little) stupid, so anyone can help me for one more bug in * I guess? |
| 18:16.02 | eppigy | be careful calling it a bug |
| 18:16.12 | eppigy | russellb will come to where you sleep |
| 18:16.36 | gambler1 | with the gun? :))) |
| 18:16.37 | russellb | there are no bugs in asterisk. |
| 18:16.48 | russellb | shuts down apache on bugs.digium.com |
| 18:17.26 | gambler1 | hehehehe to late... codefreeze yesterday told me to check a newer version of * :)))) |
| 18:18.18 | codefreeze-lap | gambler1: find anything interesting? |
| 18:18.38 | gambler1 | never the less (i still do not know exact meaning of this sentence, as you might note that I am not native english speaker) |
| 18:18.56 | *** part/#asterisk JoshuaP0x (n=Administ@unaffiliated/joshuap0x) |
| 18:19.14 | gambler1 | codefreeze-lap: nope, but I did found that it happens only when the call is unanswered |
| 18:19.39 | drmessano | No bugs, just undocumented features |
| 18:19.53 | gambler1 | codefreeze-lap: I tried and 1.6.0.3-rc1 |
| 18:19.54 | codefreeze-lap | gambler1: now that is interesting.. I've been playing with disposition problems |
| 18:20.09 | *** join/#asterisk amaache (n=maache@41.221.16.138) |
| 18:20.17 | Daejeo | does any have an experience of using a bluetooth device(cellphone) as FXO or FXS channel? |
| 18:20.26 | codefreeze-lap | gambler1: just for fun, did you try 1.6.1 from svn? |
| 18:20.36 | Daejeo | can any recommend a bluetooth adapter for using with asterisk ? |
| 18:21.04 | drmessano | Daejeo: The $5 ones work |
| 18:21.20 | drmessano | EDR's |
| 18:21.28 | drmessano | Well, sort of |
| 18:21.37 | drmessano | Seems Bluetooth isn't working here right now |
| 18:21.42 | gambler1 | codefreeze-lap: no, I didnt try 1.6.1, I will take that tonight on one of test servers |
| 18:22.01 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 18:22.33 | Daejeo | drmessano: any brand name? |
| 18:22.43 | drmessano | No, no brand name |
| 18:22.49 | drmessano | Google for EDR Bluetooth |
| 18:23.06 | drmessano | You can get them on ebay |
| 18:23.54 | *** join/#asterisk freddyk (n=freddyk@host139-25-dynamic.11-79-r.retail.telecomitalia.it) |
| 18:25.53 | codefreeze-lap | gambler1: I have 7 bugs assigned to me with "answered" in the summary... some have patches attached, ready for testing; you might look thru the list and see if any of my patches might help/solve the problem. |
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| 18:30.21 | gambler1 | codefreeze-lap: I will take a look now, tnx for your support. I will let you know (if you want) if any of those solved the problem |
| 18:31.07 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 18:38.59 | gambler1 | codefreeze-lap: assigned to? I cant find you on the list? Doing something wrong? |
| 18:41.54 | *** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose) |
| 18:43.06 | codefreeze-lap | gambler1: I'm murf on the bug tracker; sorry... |
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| 19:15.33 | *** part/#asterisk interfaithquest (n=chatzill@ip67-88-184-130.z184-88-67.customer.algx.net) |
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| 19:25.35 | *** part/#asterisk KOCATEPE (n=admin@88.247.137.187) |
| 19:32.49 | *** join/#asterisk XnOSX (i=3e39f963@gateway/web/ajax/mibbit.com/x-6d0f625c1be3fa40) |
| 19:32.59 | XnOSX | hello friends |
| 19:33.09 | XnOSX | i have a problem with my asterisk |
| 19:33.19 | XnOSX | i have a asterisk with 2 pri lines |
| 19:33.31 | XnOSX | but i have these ports down |
| 19:33.47 | XnOSX | central*CLI> zap show status Description Alarms IRQ bpviol CRC4 T2XXP (PCI) Card 0 Span 1 RED 0 0 0 T2XXP (PCI) Card 0 Span 2 RED 0 0 0 |
| 19:34.24 | Dovid | XnOSX: Check ur configs and that the cables r in the correctly |
| 19:34.46 | XnOSX | i was check these and all its ok |
| 19:35.11 | XnOSX | the config is the conf that i always use in this server |
| 19:35.12 | Dovid | wut do u get for zttool ? |
| 19:35.20 | Dovid | then call ur telco company |
| 19:35.44 | Dovid | it happend all of a sudden ? |
| 19:35.45 | XnOSX | Dovid: my Telco Telefon is BT |
| 19:36.07 | Dovid | so ask them what they see |
| 19:36.10 | XnOSX | in the zttool have alarm in RED |
| 19:36.36 | Dovid | red alarm is either config or bad cable or it isnt plugged in |
| 19:37.18 | XnOSX | Dovid can i up or reset the spans? |
| 19:37.25 | XnOSX | take a look |
| 19:37.26 | XnOSX | central*CLI> pri show spans PRI span 1/0: Provisioned, In Alarm, Down, Active PRI span 2/0: Provisioned, In Alarm, Down, Active |
| 19:38.15 | XnOSX | when i make pri show spans asterisk say Provisioned, In Alarm, Down, Active |
| 19:38.42 | XnOSX | its posible that the BT primary are down connect? |
| 19:39.22 | Dovid | it can be. u need to ask them |
| 19:39.29 | XnOSX | sorry my abglis is not so good |
| 19:39.41 | XnOSX | ummm ok |
| 19:40.12 | XnOSX | Dovid: know you something for make test to a pri card or ports? |
| 19:42.15 | *** join/#asterisk ghento (n=ghento@CPE0014bfaf7d7a-CM00159a03312c.cpe.net.cable.rogers.com) |
| 19:43.27 | XnOSX | Dovid: if i have other PRI line from other telco company can i change the PRI cable in the card and restart zaptel and asterisk and this would take positive effect isnt it? |
| 19:51.57 | *** join/#asterisk UQlev (n=kvirc@91.184.220.73) |
| 19:57.09 | *** join/#asterisk LND (n=lee@fazer1-adsl.demon.co.uk) |
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| 19:59.41 | BezNalogov | Hello people. Can somebody tell me if there is a solution to implement a fax2mail application on asterisk 1.4.22? I found many examples on internet, but none of them seems to work. Is this goal possible with asterfax or is that only for mail2fax? |
| 20:05.09 | *** part/#asterisk LND (n=lee@fazer1-adsl.demon.co.uk) |
| 20:05.33 | XnOSX | i have a question: how i can to know if the digium pri card its OK? |
| 20:07.45 | Dovid | XnOSX: create a loop back |
| 20:08.47 | Dovid | XnOSX: http://kb.digium.com/?View=entry&EntryID=95 |
| 20:18.05 | *** join/#asterisk obnauticus (n=lol@about/windows/regular/obnauticus) |
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| 20:50.09 | d00gster | what's the most stable version of * |
| 20:51.10 | Dovid | 1.6.X |
| 20:52.23 | jezier | hi.. I have problem with chan_mobile + samsung sgh-x820.... incoming calls works... but not outgoing... |
| 20:52.56 | jezier | it looks like my phone doesn't understand ATD... but hangup AT+CHUP works... |
| 20:57.07 | *** join/#asterisk jicksta (n=jicksta@c-24-6-87-187.hsd1.ca.comcast.net) |
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| 21:49.18 | gambler1 | codefreeze-lap: sorry for the wrong info about the cdr. It happens to all calls not only unanswered. I overlooked master.csv :( |
| 21:49.35 | root52 | Hey everybody. Anyone have any priceing or suggestions for wholesale SIP termination? |
| 21:52.51 | *** join/#asterisk Thorn_ (n=thorn@unaffiliated/thorn) |
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| 22:00.24 | jeff | hrm. when using callfiles, i will get: |
| 22:00.25 | jeff | Auto fallthrough, channel 'OutgoingSpoolFailed' status is 'UNKNOWN' |
| 22:00.25 | jeff | [Dec 20 16:58:26] NOTICE[8421]: pbx_spool.c:355 attempt_thread: Call failed to go through, reason (3) Remote end Ringing |
| 22:00.29 | *** join/#asterisk jetlagmk2 (i=jetlag@pool-70-104-80-249.pskn.east.verizon.net) |
| 22:00.37 | jeff | how can i log that reason? |
| 22:00.52 | jeff | it shows up on the console, but only after the failed extension fires off. |
| 22:07.01 | *** join/#asterisk moy (n=moy@CPE001e6891a732-CM00080dab8485.cpe.net.cable.rogers.com) |
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| 23:57.03 | conathan | Greetings. I am having some problems configuring asterisk 1.4.21 to receive incoming calls from les.net |
| 23:57.21 | conathan | <PROTECTED> |
| 23:57.42 | glaz | conathan: in sip.conf , which context did you put your incoming calls? |
| 23:57.42 | conathan | exten => s, 1,DIAL(SIP/conathan&SIP/PAP2_1_L1,45,r) |
| 23:57.46 | conathan | default |
| 23:57.57 | glaz | you shouldnt use default for incoming calls |
| 23:58.01 | conathan | also tried incoming back when I was trying to make it more complicated |
| 23:58.44 | glaz | paste your extensions.conf and sip.conf to rafb.net/paste |
| 23:58.51 | glaz | hide the username and passwords if any |
| 23:59.15 | conathan | ok, one min |
