00:03.13 | *** join/#asterisk jksM (jks@193.189.93.254) |
00:15.27 | jaytee | wow, it's really jumping in here |
00:17.30 | *** join/#asterisk SaiSoma (~SaiSoma@adsl-074-167-136-030.sip.mob.bellsouth.net) |
00:18.18 | beek | Hi jaytee |
00:18.26 | jaytee | hi beek |
00:19.41 | beek | jaytee: I'm just sitting here watching the snow pile up. |
00:20.22 | jaytee | snowmageddon |
00:20.53 | beek | It's a PITA, for sure. |
00:21.25 | jaytee | where are you at? |
00:22.07 | beek | South of State College, PA |
00:22.21 | jaytee | oh, yeah! you're getting dumped on royally |
00:22.59 | beek | It's a nice addition to our 14" from last Saturday. The southern end of our county got 30" |
00:24.14 | *** part/#asterisk etfonhomey (~etfonhome@74-131-159-160.dhcp.insightbb.com) |
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00:32.41 | *** join/#asterisk staffmember (~singulari@static-71-183-79-24.nycmny.fios.verizon.net) |
00:32.53 | staffmember | is there any reason why i wouldnt be able to do "sip show peers" from CLI |
00:33.06 | staffmember | sips not even an option when i do ? |
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00:37.18 | *** join/#asterisk Tech_Travis (~Administr@cpe-76-168-191-127.socal.res.rr.com) |
00:38.10 | *** join/#asterisk werdan7 (~w7@freenode/staff/wikimedia.werdan7) |
00:39.56 | p3nguin | staffmember: Sounds like chan_sip isn't loaded. |
00:42.15 | *** part/#asterisk Cresl1n (~matt@asterisk/libpri-and-libss7-expert/Cresl1n) |
00:45.42 | *** join/#asterisk bobnormal (~bobnormal@94-195-193-13.zone9.bethere.co.uk) |
00:45.48 | bobnormal | how to show registered channel types? |
00:46.47 | *** join/#asterisk ReDNeQ (~ReDNeQ@70.114.229.58) |
00:50.14 | *** join/#asterisk spenguin[work] (~penguin@122.182.0.38) |
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00:52.28 | jaskew | bobnormal: try core show channeltypes |
00:52.41 | bobnormal | <PROTECTED> |
00:52.48 | jaskew | not sure if that's what u are looking for |
00:56.03 | staffmember | p3nguin: whats not loading if i cant do "reload" from cli ? |
00:57.09 | *** part/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
01:04.51 | *** join/#asterisk MatBoy (~MatBoy@wiljewelwetenhe.xs4all.nl) |
01:12.42 | bobnormal | after upgrade from 1.4 to 1.6 on 'module load chan_dahdi.so' i get 'chan_dahdi.c:2121 dahdi_open: Unable to specify channel 1: No such device or address'. cause? wanrouter status confirms connected. |
01:27.12 | *** join/#asterisk Faithful (~Faithful@ns.linuxterminal.com) |
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01:54.07 | spenguin[work] | anyone here got any comments on ipkall? |
02:05.16 | *** join/#asterisk voipmonk (~shido6@67.111.52.130.ptr.us.xo.net) |
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02:07.32 | *** join/#asterisk iq (~chatzilla@unaffiliated/iq) |
02:07.33 | iq | Hi |
02:07.36 | sebbl | moin |
02:08.01 | sebbl | what isdn card can i use for 2 isdn lines? |
02:15.19 | *** join/#asterisk Wgg (~hyena@75-119-229-164.dsl.teksavvy.com) |
02:16.05 | Wgg | Hi folks, I'm trying to get PlayDTMF to work and having trouble |
02:19.34 | p3nguin | Were you planning to provide any details and get some help, or were you just letting everyone know that you're having a problem? |
02:19.59 | Wgg | I was waiting to see if anyone here was alive ;) |
02:20.10 | bobnormal | mmm |
02:20.23 | Wgg | Ok, I get back the response "DTMF successfully queued" but never hear the tone |
02:20.48 | Wgg | I found a few other places where people mention this problem, but no solutions |
02:20.49 | *** part/#asterisk sebbl (~Momofu@HSI-KBW-109-192-163-191.hsi6.kabel-badenwuerttemberg.de) |
02:22.08 | Wgg | Are there any other details I should provide? |
02:23.30 | ChannelZ | Your credit card number |
02:23.56 | Wgg | Certainly! It's 88 |
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02:41.35 | *** join/#asterisk thansen (~thansen@c-76-27-110-194.hsd1.ut.comcast.net) |
02:43.54 | thansen | according to this.. http://www.voip-info.org/wiki/view/say+digits I can define escape digits. It seems to be escaping no matter the digit pressed. Is there any way to force all the digits be heard? |
02:44.11 | *** join/#asterisk nix8n82 (~AndChat@63.162.27.14) |
02:45.36 | jmcdowell | What do you mean escaping? |
02:45.48 | jmcdowell | Sorry, I just got here. |
02:47.32 | thansen | jmcdowell: that's all I've said so you haven't missed anything :) It's not completing the readback of all digits |
02:47.39 | jmcdowell | Anyone know how to prevent an extension from dialing it's self? |
02:48.04 | thansen | fun..infinite dialplan loop |
02:48.04 | jmcdowell | So... Where are you seeing this? |
02:48.21 | thansen | http://www.voip-info.org/wiki/view/say+digits |
02:48.25 | jmcdowell | in the asterisk debug output? |
02:48.51 | thansen | I'm calling and hearing that not all the digits are read and it just continues with the script |
02:49.03 | thansen | (when I hit a number) |
02:49.22 | jmcdowell | huh |
02:49.47 | thansen | what don't you understand? |
02:50.04 | jmcdowell | I have never used "say" digits |
02:50.09 | jmcdowell | wasn't aware it was supported |
02:50.32 | thansen | what do you use? |
02:50.37 | jmcdowell | The keys |
02:50.56 | thansen | how do the keys talk to you? |
02:51.08 | jmcdowell | They keys dont talk to me |
02:51.12 | thansen | lol |
02:51.13 | jmcdowell | ohhh |
02:51.19 | thansen | :D |
02:51.22 | jmcdowell | this is "playback" from asterisk ? |
02:51.34 | thansen | yes |
02:52.04 | thansen | it's playing back a set of digits...I want to force the person listening to hear them *all* |
02:52.15 | *** join/#asterisk KC9NVQ (trelane@funtoo/staff/trelane) |
02:52.20 | jmcdowell | Ahhh.. |
02:52.26 | jmcdowell | I know it can be done, but don't know how |
02:52.30 | thansen | right now any dtmf is 'escaping' from playback |
02:52.46 | jmcdowell | right |
02:54.53 | KC9NVQ | looking for something similiar to asteriskstat that can rate calls and produce phone bills. A2billing is ridiculously complicated, and seems to have to run on the server. I'd prefer to just hook up to cdr_mysql's database |
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02:59.33 | *** join/#asterisk Rapier1024 (~Rapier102@d-69-161-89-227.cpe.metrocast.net) |
02:59.53 | Rapier1024 | Hi folks |
03:01.46 | Rapier1024 | can anyone tell me if they have seen something like this before? I'm having intermittent dropout of audio on 480i's over a VPN connection, but only on the incoming stream to the server |
03:02.18 | Rapier1024 | It's random, and can last from 2 to 10 seconds |
03:03.21 | jmcdowell | Setup QoS |
03:04.01 | *** join/#asterisk Kumbang (~kumbang@rusnas.paume.itb.ac.id) |
03:04.40 | Rapier1024 | The LAN on both sides is dedicated to the phones with no other ip devices on either of them. This is only from the remote office to the home office |
03:05.16 | Rapier1024 | Everything going out on the port at the remote is "real time" QoS. |
03:05.41 | jmcdowell | Why are you pushing them through a VPN? |
03:06.01 | jmcdowell | Is this a real VPN or some bullshit MS vpn server ? |
03:06.24 | Rapier1024 | We had an issue a while back when I forwarded a port to my server. No this is a sonicwall hardware firewall with point ot point VPN |
03:06.40 | jmcdowell | Hmmm.. |
03:06.48 | jmcdowell | I would look at how you are doing VPN on the one side |
03:06.57 | jmcdowell | it too could cause issues since it's not balanced on the other end. |
03:06.58 | Rapier1024 | Have a TZ150 on both sides |
03:07.09 | Rapier1024 | same config on both ends |
03:07.20 | jmcdowell | Run IPTRAF |
03:07.24 | jmcdowell | iptraf |
03:07.28 | jmcdowell | and watch the traffic |
03:07.40 | jmcdowell | see if there is something overwhelming your vpn. |
03:08.00 | Rapier1024 | Yeah, the problem is this only happens once a day and it's pissing off my customer |
03:08.09 | jmcdowell | What time of day? |
03:08.13 | jmcdowell | Let me guess, LUNCH |
03:08.35 | Rapier1024 | It changes. LOL sometimes in the moring, and other in the afternoon |
03:08.41 | *** join/#asterisk styelz (~yoohoo@m0o0.mooo.com) |
03:08.52 | Rapier1024 | nobody is talking at lunch so I don't know about that. |
03:09.32 | Rapier1024 | This one is driving me nuts though. it's only on the incoming stream. The remote office can hear fine |
03:09.40 | jmcdowell | What time of day? |
03:10.09 | Rapier1024 | it's not a specific time. sometimes early, sometimes in the afternoon. |
03:10.29 | jmcdowell | So, I would run iptraf on both sides.. or at least the server side |
03:10.57 | Rapier1024 | the phones are registering fine, and the problem shows up whether it's a call to the outside, or to one of the internal phones |
03:11.28 | jmcdowell | What kind of phones? |
03:11.39 | Rapier1024 | aastra 480i's |
03:12.04 | jmcdowell | I don't know how to monitor your link, as it could be that too. |
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03:12.44 | Rapier1024 | iptraf rolls over doesn't it? in other words, I can leave it running for some time and have them let me know the second it happens, yes? |
03:12.55 | jmcdowell | You can log it |
03:13.01 | jmcdowell | I don't if you can have it notify you |
03:13.12 | jmcdowell | but I am sure they would let you know and then you could look at the logs. |
03:13.24 | Rapier1024 | I'll have them user let me know the minute it happens. |
03:13.39 | jmcdowell | Is it ONE or all lines affected? |
03:13.45 | Rapier1024 | all phones |
03:13.55 | jmcdowell | My polycoms were doing that and I had to adjust their codecs and everthing was fine. |
03:14.08 | Rapier1024 | hmmm |
03:14.19 | jmcdowell | I would almost bet that something is saturating the network link though because it doesn't like the VPN. |
03:14.41 | jmcdowell | Make sure you are watching ALL type of traffic in IP traf. IP, NON-IP and unknown.. EVERYTHING |
03:14.57 | Rapier1024 | okay that makes sense |
03:15.06 | jmcdowell | If you are experiencing some sort of an arp problem that is storming things, you want to be able to see it. |
03:15.22 | jmcdowell | I would also crank up all the debug values in asterisk |
03:15.36 | jmcdowell | so you can look there if IPTRAF doesn't show anything. |
03:15.48 | Rapier1024 | We have a 5mb ethernet over copper at the customer's main site, and comcast business at the remote. |
03:16.02 | jmcdowell | I would also use "sar" or HTOP to watch the CPU and memory usage on your server. |
03:16.03 | Rapier1024 | I don't have this problem with any of the phones at the main site. |
03:16.53 | jmcdowell | Right, but if you have a device that doesn't understand the fact that you have a VPN and starts spitting out information that causes confusion. It can cause problems. |
03:17.28 | jmcdowell | I would bet that if you setup QoS, you would be surprised. |
03:17.57 | Rapier1024 | The computer and the phone lans share the connection at the remote, and I have limited the computer port on the main switch to 512kb leaving 1.5mb for thephones but it didn't solve it |
03:18.17 | Rapier1024 | I need to make sure QOS is in place on the main side |
03:18.24 | jmcdowell | again crank up the logging |
03:18.36 | jmcdowell | use iptraf and watch the cpu memory utilization |
03:18.57 | jmcdowell | it could be that on the main site side, that QoS isn't allowing them what they need. |
03:19.53 | Rapier1024 | can you tell me the best way to lock down the asterisk server? When I opened theport the last time, someone got in and was screwing with it. |
03:20.19 | jmcdowell | Well, you just threw a WHOLE different set of metrics in there. |
03:20.27 | jmcdowell | Did you re-format and re-install ? |
03:20.43 | jmcdowell | Because without doing so, someone could still be screwing with it. |
03:21.10 | Rapier1024 | It's actually run from a module, not a disk. so if they did anything, resetting the thing eliminates it. |
03:21.25 | Rapier1024 | its set to run from a ram disk |
03:21.36 | jmcdowell | Hmmm.. |
03:21.41 | jmcdowell | Locking it down is fairly easy |
03:21.54 | jmcdowell | Dont set stupid passwords and lockout the root account |
03:22.02 | jmcdowell | chroot jail ftp |
03:22.15 | jmcdowell | and I can't see how they could get in. |
03:22.38 | Rapier1024 | Can you require certs in order for the phones to access the server? |
03:22.51 | jmcdowell | I think there is something like that, I have never done it |
03:22.54 | jmcdowell | I just use FTP |
03:23.02 | jmcdowell | with the password, they aren't getting in |
03:23.12 | jmcdowell | no root ssh logins are allowed |
03:23.32 | Rapier1024 | I thought so too, but haven't tried it |
03:23.40 | jmcdowell | No they are by default |
03:23.46 | jmcdowell | I am saying they shouldn't be |
03:24.00 | jmcdowell | only sudo su - access and even that is ridiculous hard to acheive |
03:24.01 | p3nguin | "sudo su" is retarrrrrrrded. |
03:24.13 | Rapier1024 | I meant I thought certs wew supposed to work |
03:24.15 | jmcdowell | Better than root logins allowed |
03:24.30 | Rapier1024 | no, root logins aren't allowed |
03:24.54 | jmcdowell | My default install allowed them, I disabled them yours is obviously different. |
03:25.12 | jmcdowell | I tried to "hack" my box once, even knowing everything but the password, I was never able to get in. |
03:26.03 | Rapier1024 | I don't know how they got in, and it pissed me off. The saving grace was in order to write the config back to the card, you have to know it's there |
03:28.14 | Rapier1024 | Well, I'll try what you suggested here. Thanks a great deal for your help, I really appreciate it. |
03:28.47 | jmcdowell | I don't know that I helped all that much.. |
03:29.09 | Rapier1024 | well you gave me places to look, and that's a start :) |
03:29.47 | Rapier1024 | I manage this system, and have some experience with Asterisk and linux, but I'm certainly no guru |
03:30.46 | jmcdowell | I would find a way to look at Link saturation |
03:31.06 | jmcdowell | and you should be able to do that throught the modem or router provided by your link provider. |
03:31.55 | Rapier1024 | I can see that on the router, and the logs will corellate the timing so that should tell me something |
03:32.35 | Rapier1024 | Asterisk is a challenge for me. Ask me about XenServer or Virtual servers and I'm fine |
03:33.09 | jmcdowell | I love asterisk |
03:33.14 | jmcdowell | I have only been using it for a month. |
03:33.14 | jmcdowell | :D |
03:33.26 | leifmadsen | it's good stuff |
03:33.36 | jmcdowell | indeed it is |
03:33.51 | jmcdowell | I am working on a "Provisioning center" plugin module for freepbx./ |
03:34.04 | Rapier1024 | I haven't had too many issues, but it's the wierd stuff that makes me want to pull my hair out |
03:34.20 | jmcdowell | yep, just follow the KISS theory and it usually works out |
03:34.26 | Rapier1024 | usually if you don't have audio, it's on bot directions |
03:34.37 | Rapier1024 | in both rather |
03:35.22 | Rapier1024 | Never had this issue with the SNOM phones, but I don't think there are any in the remote office |
03:36.30 | *** join/#asterisk maxagaz (~maxagaz@222.128.36.151) |
03:36.32 | jmcdowell | Once you collect all that info, I think you will be able to really zero in on something |
03:36.38 | jmcdowell | or at least say it's none of these issue. |
03:36.51 | Rapier1024 | yeah...that makes sense. |
03:37.07 | jmcdowell | at which point I would really look at the codec tweaks on the phones |
03:38.21 | Rapier1024 | I fought for some time with one phone that woul drop registration periodically. Discovered with the SonicWall TZ150 as the gateway, it was maxing out the connections. Didn't realize I could set the asterisk box as the gateway on the phones at that time |
03:38.42 | Rapier1024 | 10 user limit on the sonicwall |
03:39.05 | Rapier1024 | They don't have limits anymore thank goodness |
03:40.15 | jmcdowell | I would have just built linux based routers.. |
03:40.16 | jmcdowell | :D |
03:40.35 | Rapier1024 | hmmm....something just occurred to me. there are five phones in the remote office. there is one of them that is doing the same thing just recently. This could be the issue |
03:41.00 | Rapier1024 | something else must be plugged into that lan up there. |
03:41.17 | Rapier1024 | five phones wouldn't do that. |
03:41.59 | jmcdowell | What do you mean it's doing the same thing? |
03:42.11 | Rapier1024 | intermittently showing NR |
03:42.19 | Rapier1024 | losing it's regitration |
03:42.25 | Rapier1024 | registration rather |
03:43.05 | Rapier1024 | it happened before because the sonicwall would send an error instead of forwarding the packets |
03:44.01 | Rapier1024 | retturn an error I meant |
03:44.13 | Rapier1024 | boy I can't type tonight |
03:44.19 | jmcdowell | Could the sonic wall be "Storming" the link with bs packets? |
03:45.48 | Rapier1024 | no, I've looked at the data stream before. It would more likely be that someone has plugged a computer or two into that lan. That would cause the error. Several connections from multiple devices could exceed the 10 user limit |
03:46.15 | Rapier1024 | The thing is, it wouldn't always cause the NR |
03:46.21 | jmcdowell | lock it down by mac or install an IPCOP firewall and throw that sonicwall in the trash. |
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03:46.58 | Rapier1024 | Trying to manage a lan that is 4 hours away is starting to get on my nerves! |
03:47.07 | jmcdowell | lol |
03:47.28 | Rapier1024 | they have 10 times as many issues than the main office does |
03:48.45 | Rapier1024 | I have no idea what they plug in where, and it kills a whole day to go up there. GRRRRRR |
03:49.10 | Rapier1024 | I could tell these folks all day long what should go where, but it just doesn't sink in |
03:50.20 | Rapier1024 | the sonicwall is actually a pretty good device, unless you start opening ports on it |
03:51.59 | Rapier1024 | with that VPN in place, we have had no issue with folks getting into the lan |
03:53.16 | Rapier1024 | well thanks again, I think it's time for bed |
03:53.23 | jaskew | raiper1024 |
03:53.36 | Rapier1024 | ?? |
03:53.43 | jaskew | Just catching up on the thread - I had similar issues at my office |
03:54.02 | Rapier1024 | what solved it? or is it solved? |
03:54.02 | jaskew | I was getting interruptions in the RTP stream |
03:54.09 | Rapier1024 | from where? |
03:54.19 | jaskew | It is - QoS is the answer, but for me was only part of the answer |
03:54.44 | Rapier1024 | I need to make sure that's enabled on the main office end |
03:54.57 | jaskew | I'm not familiar with the SonicWall - I'm using a linux router here |
03:55.38 | Rapier1024 | Rather than setting each device, I set everything going out over that one port to be "real time" on the main switch |
03:55.45 | jaskew | THe problem was that my DSL modem has a pretty large queue. That effectively un-did any QoS stuff I had in my router |
03:56.08 | Rapier1024 | hmmm |
03:56.14 | Rapier1024 | how did you solve that? |
03:56.23 | jaskew | I solved it by shaping my traffic to my DSL speed at the router. That kept the queu from building up in my modem |
03:56.24 | Rapier1024 | I was reading on buffer size |
03:56.55 | jaskew | Then I was able to control the prioritization properly. |
03:57.29 | jaskew | Otherwise, I was prioritizing everything at my router and then just sticking all of the traffic into the (dumb - fifo) queu in the DSL modem |
03:57.43 | Rapier1024 | I had planned on doing something similar tomorrow. limiting the total bandwidth at the switch to 2mb |
03:58.01 | jmcdowell | I don't know if that's the answer |
03:58.01 | Rapier1024 | and then throttling the computer lan to only use half that |
03:58.18 | jaskew | So the VPN complicates it a little bit - do you know if the VPN passes through the QoS? |
03:58.20 | jmcdowell | I think I would do some serious logging, again Iptraf can be very very revealing of rouge devices |
03:59.04 | Rapier1024 | I think so, it just wraps the packets in UDP and then delivers them on the other end exactly as they originated |
03:59.26 | jaskew | so does the "envelope" UDP packet preserve the QoS (on the outside where it can be seen)? |
03:59.31 | Rapier1024 | I plan on doing the logging and iptraf to see that. |
04:00.05 | Rapier1024 | I can set the QoS both inside the packet, and then outside. Inside at the firewall, and then tag the UDP at the switch |
04:01.01 | jaskew | That's good. Here's a discussion of what I was talking about |
04:01.02 | jaskew | http://mailman.ds9a.nl/pipermail/lartc/2007q3/021607.html |
04:01.14 | jaskew | Read that & follow the link to lartc.org. |
04:01.45 | jaskew | If you can do something similar, it might help celar things up. Of course, you'll have to fact ro the VPN stuiff in |
04:01.59 | Rapier1024 | I think throtling the bandwidth (outbound) to the total for the line, and then setting the QoS on inside and outside should help this. |
04:02.12 | jaskew | worked for me :) |
04:02.43 | Rapier1024 | also making sure that the sonicwall isn't kicking the packets because of device user limit (ie...some idiot plugged in a computer to the lan) |
04:03.07 | jaskew | you probably will need to tune both ends to make sure that the RTP packets get put on the line ASAP when they come up |
04:03.48 | Rapier1024 | that's the frustrating part. If the sonicwall is kicking the packets, it would do EXACTLY what it's doing. It wouldn't be long enough to kill the call, but to just block outgoing traffic for 2 to 10 seconds |
04:03.56 | jaskew | putting them ASAP into a queu that is 1000ms deep doesn;t help ;) |
04:04.04 | Rapier1024 | I agree |
04:04.40 | jaskew | Good luck w/ it - hope that helps! |
04:04.41 | Rapier1024 | I just found out that there is a new router in place that belongs to comcast between our switch and the cable modem |
04:04.58 | Rapier1024 | have no idea who put that in place, or why |
04:05.16 | Rapier1024 | Thanks to both of you for your help! I'm off to bed |
04:05.19 | jaskew | well, if you limit the sonic wall output to the upstream speed, it should still prevent a queue from forming. |
04:05.31 | Rapier1024 | that's what I thought |
04:05.35 | jaskew | upstream speed - say 5% |
04:05.44 | jaskew | I men "minus" 5% |
04:05.47 | Rapier1024 | it could never overcome the queu |
04:06.51 | Rapier1024 | the sonicwall allows me to type a specific value in. The switch I have options I can pick, 2mb is one of them |
04:06.55 | jaskew | oh - you need to be in control of ALL traffic. That's a prerequisite! |
04:07.41 | Rapier1024 | yeah, there are two sonicwalls on this line out. one for computer lan, and one for phone so I have a good amount of control over what goes in or out |
04:08.06 | jaskew | that could present a challenge - best of luck with it |
04:08.34 | Rapier1024 | fortunately they allow seperate limits for both in and out so I won't be killing download speed |
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04:08.50 | Rapier1024 | thanks again, good night all |
04:08.55 | jaskew | you bet |
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04:31.55 | CyberCod | anyone familiar with the SpoTel TDM410's? are they comparable to the digium cards? |
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04:56.28 | sevv | i can't figure this out - are all of the pap2 firmwares compatible with the spa-2002 hardware? |
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05:12.59 | aruntomar | i'm getting this error " Unable to forward voice or dtmf", i googled a lot, but can't find the solution |
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05:48.10 | kuku | I get You do not appear to have the sources for the 2.6.18-164.6.1.el5.centos.plusPAE kernel installed - but kernel-PAE-devel is installed. |
05:55.36 | voipmonk | what version, kuku? did you make symlinks? |
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07:55.13 | benngard | ~phones |
07:55.14 | infobot | While personal preference will dictate which phone works best for you, general consensus on a rough order of quality and suggestibility is as follows: Polycom (any), Aastra 480i, Aastra 5i Series, Linksys SPA-9XX, Snom, Cisco 79XX, everything else. Do not consider Grandstream phones. Ever. |
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07:58.54 | basty | Hi |
07:59.04 | *** join/#asterisk ktwilight[m] (~ktwilight@27.170-65-87.adsl-dyn.isp.belgacom.be) |
08:00.02 | basty | I have a strange problem. I just switched from Asterisk 1.2 to 1.4.29. I am using AgentCallbackLogin(${CALLERID(NUM)}||${CALLERID(NUM)}@intern) for my agents to login. As soon as the queue gets called - the Agent is ringing. But as soon as the Agent tries to answer - nothing happens - it keeps ringing... |
08:00.53 | *** join/#asterisk florz (nobody@2001:1a50:503c::1) |
08:02.12 | TommyBotten | basty: Try to do a 'core set verbose 6' and paste the information to pastebin.ca and provide the link |
08:02.37 | basty | TommyBotten: the strange thing is, that in the cli debug everything seems normal...but lemme try again |
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08:07.55 | TommyBotten | in that case, enable sip debugging, and see what the phone is doing at pickup time |
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08:10.43 | basty | TommyBotten: ah okay...it seems that I have esthablished the call - but I cant hear anything |
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08:15.05 | basty | TommyBotten: the state of the agent is still |
08:15.09 | basty | "ringing" i mean |
08:15.28 | *** part/#asterisk Tech_Travis (~Administr@cpe-76-168-191-127.socal.res.rr.com) |
08:15.38 | TommyBotten | Hmm.. Did you look at the SIP traffic? |
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08:22.25 | basty | TommyBotten: nope...i dont know how...i am kinda newbie ;) |
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08:24.47 | basty | TommyBotten: but it seems that the local channel is making the problem |
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08:26.31 | TommyBotten | Do you get any error messages on that? |
08:26.45 | TommyBotten | for sip debugging, do a 'core set debug' and then TAB to see your options |
08:29.09 | basty | TommyBotten: http://pastebin.com/m336c06bc |
08:29.17 | basty | thats all i see...with full debug on asterisk cli |
08:29.36 | basty | TommyBotten: the agent is trying to answer the channel..but I keep listen to the "music on hold". |
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08:31.11 | basty | TommyBotten: If I set static agent with "SIP/87" into the queue.conf - it works...so it has to do something with the local channels |
08:36.12 | TommyBotten | Aha.. so the agent is not static, I take it? |
08:36.25 | TommyBotten | How is it configured? |
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08:40.43 | basty | TommyBotten: the agent is using the agentcallback like AgentCallbackLogin(300${CALLERID(NUM)}||${CALLERID(NUM)}@intern) in the agents.conf I have the agent => 300100 for example |
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08:53.53 | mbp | What do people prefer for queue statistics? Asternic, QueueMetrics, queue_stats or something else? |
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08:55.24 | kaldemar | TommyBotten: SIP debug is enabled with sip set debug, not with core set debug. |
09:06.52 | TommyBotten | good point |
09:06.55 | TommyBotten | Sorry about that |
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09:12.40 | basty | TommyBotten: if i change the stuff to "add queue member" everything works |
09:12.48 | basty | so it has to be something weirdo with the "local" channel it seems |
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09:14.15 | TommyBotten | Sounds reasonable |
09:14.25 | basty | but i would like to fix that |
09:14.25 | basty | ;) |
09:14.31 | TommyBotten | I haven't used the config with AgentCallBack |
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09:33.21 | *** join/#asterisk _omer (~omer@119.152.220.105) |
09:33.24 | _omer | hello |
09:34.04 | _omer | Is there a way to get AGENT or PEER logged in time ? for example: 3 minutes 24 seconds since logged in ... or something like that? |
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09:46.11 | krion | hi guys |
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09:55.00 | MrM4xXx | hello i'm european and i configured a patton as isdn-voip gateway.., when i call the signal results the typical american signal, not the european..; any idea? |
09:55.21 | MrM4xXx | is it an asterisk conf? |
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10:29.32 | kaldemar | MrM4xXx: depends on the whole scenario. in asterisk, you need to set your country in indications.conf. but your phone may also give you wrong tones if you're using a voip one. |
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10:34.35 | MrM4xXx | kaldemar: thx, I try |
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10:47.36 | MrM4xXx | kaldemar: thx, it's ok, now it's working properly. ;) |
10:56.28 | krion | anyone with a png who explain what's expected when 407 happened ? |
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11:12.50 | kaldemar | krion: what 407? |
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11:46.36 | *** join/#asterisk Zap-W (~marceloa@95.211.21.34) |
11:46.49 | Zap-W | does asterisk behave like a RTPProxy by default? |
11:47.57 | kaldemar | Zap-W: no |
11:48.52 | kaldemar | Zap-W: asterisk doesn't force itself on the media path by default. |
11:49.31 | Zap-W | how do I turn it on, then it will modify some SIP headers and some arguments in SDP payload? |
11:49.39 | kaldemar | what version are you using? |
11:49.43 | *** join/#asterisk UQlev (~yuriy@nb11-125.static.cytanet.com.cy) |
11:50.10 | Zap-W | 1.4.26.1 |
11:50.46 | kaldemar | put canreinvite=no into sip.conf |
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12:03.15 | Zap-W | interesting http://www.voip-info.org/wiki/view/Asterisk+sip+canreinvite |
12:07.32 | arj__ | is there more to enabling "blind transfer" than placing it into features.conf? I enabled it, but #1 (or ## or #) don't seem to be doing anything |
12:09.44 | kaldemar | Zap-W: what is? |
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12:19.15 | lbarth | hello |
12:20.12 | lbarth | Can anyone here help me with this ao2_unlock issue? (15915) |
12:21.11 | lbarth | i have no clue how to get asterisk 1.6 stable on debian |
12:21.35 | lbarth | it can't be true that i am the only one running asterisk 1.6 in a heavy load envireonmen |
12:23.25 | *** join/#asterisk viq (~viq@unaffiliated/viq) |
12:23.51 | kaldemar | lbarth: can you use another timing module? |
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12:24.37 | Dovid | oej: ping |
12:26.29 | lbarth | kaldemar: currently i have no interface cards in the maschine so i can currently only se pthread |
12:27.12 | lbarth | bu i just disovered that there is a packege timerfd |
12:27.15 | kaldemar | lbarth: there's timerfd in 1.6.2 and always dahdi_dummy |
12:27.55 | lbarth | is dadi_dummy a good solution for production? |
12:27.59 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
12:28.00 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
12:28.19 | kaldemar | people do use it, but you better test it for yourself first. |
12:30.29 | *** join/#asterisk smooth_penguin (~smoove@59.95.4.11) |
12:31.49 | lbarth | great compiling dahdi fails... i hate those days |
12:33.05 | Dovid | lbarth: You will live |
12:38.07 | *** join/#asterisk lftsy (~lftsy@leonhart.leurent.eu) |
12:39.36 | lftsy | Hello everybody, since my server is currently in a CPU 100% mode, and flooding SIP OPTIONS, I was able to provide many logs for ticket https://issues.asterisk.org/view.php?id=16382 Is there a developper that can say if another dump would be useful? Thanks |
12:41.16 | Dovid | lftsy: Try asterisk-dev in a few hours when they wake up in Huntsville |
12:42.34 | lftsy | Dovid: Thanks, it's just that I cannot keep this server blacklisted many time since all calls are now running on only one server... I will try it anyway. Thank you |
12:43.15 | *** join/#asterisk proute (~AnthonyCB@mail.sysun-technologies.com) |
12:43.33 | proute | hi everybody |
12:43.34 | kaldemar | lftsy: how about changing rtautoclear and ignoreregexpire? |
12:43.35 | *** join/#asterisk mbrevda (~mbrevda@unaffiliated/mbrevda) |
12:43.56 | mbrevda | looking for waywas to covert a pdf to tif for FFA - without using GhostScript |
12:44.06 | proute | Does Dahdi works fine with B410P and asterisk 1.6.0.x? Or is it better to works with asterisk 1.6.1.x? |
12:44.21 | mbrevda | I thought imagemagick could do it, but I cant get the correct commands |
12:46.18 | *** join/#asterisk basty (~basty@212.218.65.183) |
12:46.19 | basty | Hi |
12:46.24 | lftsy | kaldemar: I have tried to play with rtautoclear and ignoreregexpire but there is always peers with Expire -1 that apperas |
12:47.07 | basty | I just updated my asterisk pbx from 1.2 to 1.4.29. I have a lot of snom phones, that uses the blf function. It worked fine in 1.2.X but after the upgrade - the blf doesnt work anymore. Anyone knows why ? |
12:47.26 | kaldemar | mbrevda: is this chennel a phreaking google proxy to you? |
12:47.42 | kaldemar | mbrevda: imagemagick can do it, the command is convert. |
12:48.36 | mbrevda | kaldemar: convert itself doesnt work without the right commands... |
12:48.44 | mbrevda | hmm s/commands/arguments ? |
12:48.52 | kaldemar | mbrevda: convert your.pdf yout.tif |
12:49.06 | mbrevda | did that - ffa wont send the fax then |
12:49.15 | *** join/#asterisk razu (~razu@razu.data.ee) |
12:49.16 | mbrevda | 'cannot queue file' or somethign |
12:49.29 | kaldemar | ask digium about the format then. |
12:50.01 | mbrevda | kaldemar: why? are they a chennel for phreaking google proxy to you? |
12:50.22 | mbrevda | (sic) |
12:50.57 | kaldemar | mbrevda: fax for asterisk is a commercial product that they provide support for... |
12:55.43 | ManxPower-work | basty: the UPGRADE*.txt files should mention the call-limit changes that effect BLF. Always read the UPGRADE*.txt files when upgrading. |
12:56.32 | basty | ManxPower-work: mh - so that means, i will have to set a call-limit on every sip account ? When I try to check "sip show subscription" everything is on "idle". |
12:57.04 | ManxPower-work | basty: It means you should read the UPGRADE*.txt file. |
12:57.17 | ManxPower-work | I guess I could go read it for you, but I'm not that nice. |
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12:58.18 | basty | ManxPower-work: hehe - but maybe you know it without reading it.. ;-) |
12:58.25 | ManxPower-work | basty: I don't. |
12:58.34 | basty | mhh well |
12:58.37 | basty | thanks! |
12:58.45 | ManxPower-work | The call limit stuff is constantly changing in Asterisk. |
12:59.25 | ManxPower-work | basty: You should know that it does not matter WHAT the limit option is set to, just as long as it it set. |
12:59.34 | ManxPower-work | as it is set. |
12:59.43 | basty | okay thanks |
13:00.20 | basty | ManxPower-work: one more question...do I need to set the call-limit for every single sip account, or could I put something into the global section of sip.conf ? |
13:00.45 | *** part/#asterisk mbrevda (~mbrevda@unaffiliated/mbrevda) |
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13:06.49 | ManxPower-work | basty: I don't know, but sip.conf.sample should indicate where you can put the option. |
13:07.03 | basty | okay |
13:07.20 | basty | well..i just set the call-limit for every peer..but it is still not working..strange thing that. |
13:09.20 | ManxPower-work | basty: did you read the ;--------------------------- STATUS NOTIFICATIONS (SUBSCRIPTIONS) ---------------------------- section of sip.conf.sample? |
13:09.28 | basty | yep |
13:09.43 | ManxPower-work | so you have limitonpeers as the other settings set? |
13:09.48 | basty | yup |
13:10.17 | Naikrovek | yawns. omg tired |
13:10.54 | ManxPower-work | basty: put your sip.conf on pastebin.ca masking only passwords |
13:11.20 | basty | okay sec |
13:12.41 | basty | AH...I guess I found out, what the problem is |
13:13.00 | basty | i had to restart the phone.. |
13:13.09 | basty | old firmware....6.2.2 installed there.. |
13:13.48 | leifmadsen | basty: what version of asterisk? |
13:14.41 | leifmadsen | ah, 1.4.29 |
13:14.47 | *** join/#asterisk [TK]D-Fender (~chatzilla@216.191.106.163) |
13:14.58 | leifmadsen | and I didn't read far enough in to see you fixed it |
13:15.00 | leifmadsen | moves along now |
13:19.03 | basty | ;-) |
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13:28.46 | basty | bye |
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13:33.33 | *** join/#asterisk e-jones (~jkastner@nat/redhat/x-rizbxvljttcgmugt) |
13:33.43 | Zap-W | hi, in SIP RFC it says, server uses the port in the SIP header message for responding to requests "SIP operates in this manner so that a server can listen for all messages, both requests and responses, on a single IP address and port." I don't understand why it wouldn't be able to use a single IP address and port if it didn't look the port in the header message and instead used the UDP port the request came from |
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13:43.20 | ManxPower-work | Zap-W: nat=yes tells asterisk to look at the packet headers, not the data portion of the SIP packet. |
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13:44.39 | florz | Zap-W: which section would that be? I can't find any such text ... |
13:44.54 | Zap-W | florz, http://www.ietf.org/rfc/rfc3581.txt |
13:45.39 | ManxPower-work | Zap-W: By default all SIP devices look at the contents of the packet to find the address/port. This is the way SIP works. nat=yes tells Asterisk to stop doing that and try to infer the port/address from the packet header, not the SIP header. |
13:45.49 | florz | Zap-W: that's not exactly the SIP RFC =:-) |
13:45.59 | ManxPower-work | Zap-W: that's not the SIP RFC. |
13:47.10 | smooth_penguin | IAX is sooo nice |
13:47.22 | smooth_penguin | 1 port and all |
13:47.24 | Zap-W | an extension |
13:47.32 | Zap-W | I tell a white lie |
13:48.02 | Zap-W | thanks |
13:51.27 | *** join/#asterisk rhp (~c3f0bccd@gateway/web/freenode/x-onrvkimhpohbfrav) |
13:52.53 | rhp | Hi all. I would like to setup a system where a number of users can talk to eachother using head-sets. I was looking whether asterisk with SIP is an answer for this. Is it possible to talk with more than one person at a time with asterisk? |
13:53.11 | *** join/#asterisk |Cybex| (~John@atwork-21.r-212.178.82.atwork.nl) |
13:55.00 | [TK]D-Fender | rhp: Yes. MeetMe conferencing. |
13:56.04 | rhp | thanks. I've tried looking for some tutorials for setting this up, but I could not find much. Could you direct me to a tutorial, if it exists? |
13:57.05 | [TK]D-Fender | rhp: Install *. Read the book. make a basic dialplan to call MeetMe |
13:57.08 | [TK]D-Fender | ~book |
13:57.08 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
13:57.42 | rhp | [TK]D-Fender: thanks a lot. |
13:58.12 | *** join/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
13:58.40 | *** join/#asterisk chuckf (~chuckf@ubuntu/member/chuckf) |
14:02.41 | ManxPower-work | rhp: Asterisk is a toolkit that allows you to build a PBX. It's not designed to be functional "out of the box" |
14:03.09 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
14:03.10 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
14:05.58 | *** join/#asterisk af_ (~getsmart@88-149-230-64.dynamic.ngi.it) |
14:06.13 | *** join/#asterisk e-jones (~jkastner@nat/redhat/x-lrooragmwjztebjf) |
14:07.14 | Katty | hi |
14:07.39 | [TK]D-Fender | rhp: think of it like a box of artist's pencils. They may have instructions on how to properly open and close the box and how to use the sharpener it comes with, but don't expect to get instrucion on how to draw some specific thing that comes to mind. |
14:08.03 | sun28 | moin \o/ |
14:08.15 | [TK]D-Fender | rhp: Everyone does things their own way. Dialplan is programming. You say what you want * to do with every step |
14:08.22 | Katty | good morning. |
14:08.43 | smooth_penguin | hi Katty :> |
14:08.49 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
14:09.26 | *** join/#asterisk bsdmail (~dig@67.228.177.47) |
14:10.07 | bsdmail | can someone help-me with this issue: http://pastebin.com/d49e55da3 ? |
14:11.47 | Katty | wonders where all the critters are |
14:12.02 | ManxPower-work | bsdmail: you have a DTMF mode issue with your provider |
14:12.19 | ManxPower-work | bsdmail: you do not have one-way audio issues? |
14:12.25 | *** join/#asterisk Warp4 (~Robert@firewall-a.buf.ny.i-evolve.net) |
14:13.57 | *** join/#asterisk slashtom (~tom@k-rad.co.uk) |
14:14.13 | *** join/#asterisk ttwhy (~tekkno@p4FECFC04.dip.t-dialin.net) |
14:14.19 | bsdmail | ManxPower-work no, i can hear and he can hear. |
14:15.12 | bsdmail | i don't unterstand why this is happening, because i have to type the number of my friend, so before he answer my dtmf is being recognized |
14:16.53 | ManxPower-work | bsdmail: it's being recognized on the INBOUND CALL. There is also the OUTBOUND call. |
14:18.51 | Katty | hugs smooth_penguin |
14:19.21 | Katty | smooth_penguin: what did you make for dinner? |
14:19.50 | smooth_penguin | Katty, well nothing as yet just had some tea :> |
14:21.05 | *** join/#asterisk Akiraa (~Akiraaaa@79.112.39.45) |
14:22.13 | *** join/#asterisk jaytee (~jforde@unaffiliated/jaytee) |
14:22.59 | bsdmail | ManxPower-work both calls are from same trunk/provider |
14:23.12 | ManxPower-work | bsdmail: That does not matter. |
14:23.45 | ManxPower-work | DTMF issues are some of the most complex and hardest things to fix. |
14:24.43 | bsdmail | ok i think i'm gonna give up |
14:24.51 | bsdmail | thanks |
14:25.35 | smooth_penguin | Katty, whats cooking ? |
14:25.54 | [TK]D-Fender | bsdmail: What mode do you have set? |
14:26.37 | bsdmail | i dont know |
14:27.24 | [TK]D-Fender | bsdmail: This is YOUR configuration. How do you NOT know what you set? |
14:28.38 | krion | kaldemar: i wasn't calling the right sip URI... |
14:28.48 | bsdmail | because i'm new and i didn't set this kind of config, so i'm using the default? |
14:29.35 | bsdmail | DTMF is when you press a key and the pbx recognize the sound and assume a number to this sound right? |
14:29.49 | krion | yes |
14:30.19 | krion | you got the cmd Read who's fun with that |
14:30.35 | krion | Read(digito||1) |
14:30.38 | krion | SayDigits(${digito}) |
14:30.53 | krion | s/Read/SayDigits |
14:30.58 | [TK]D-Fender | bsdmail: Go look at your configs. |
14:31.18 | krion | hi [TK]D-Fender |
14:31.18 | bsdmail | [TK]D-Fender can u tell where i should look? |
14:31.26 | bsdmail | section.... |
14:31.33 | [TK]D-Fender | bsdmail: SIP.CONF. |
14:32.03 | kuku | I get You do not appear to have the sources for the 2.6.18-164.6.1.el5.centos.plusPAE kernel installed - but kernel-PAE-devel is installed. |
14:33.22 | [TK]D-Fender | kuku: Also need the headers and a bunch of other stuff. Read the insttructions in the tarball |
14:36.33 | *** join/#asterisk chilicuil (~sistemas@unaffiliated/chilicuil) |
14:37.13 | *** join/#asterisk andres833 (~andres833@190.144.75.22) |
14:38.45 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
14:38.47 | kuku | [TK]D-Fender: ok |
14:42.16 | *** join/#asterisk mrprozac (~mrprozac@132-82-ftth.onsneteindhoven.nl) |
14:46.29 | *** join/#asterisk drako (~luisjose@nelug/coreteam/luisjose) |
14:46.51 | drako | What's a good TextToSpeech ? |
14:46.57 | *** join/#asterisk jakent (~john@soleil.johnkent.mooo.com) |
14:47.20 | *** join/#asterisk crochat (~crochat@80.83.52.178) |
14:47.39 | jaytee | Cepstral |
14:48.08 | Faustov | isn't that more the other way, speech recognition? |
14:48.28 | jaytee | Faustov, no, you're probably thinking of Lumenvox |
14:48.48 | kuku | [TK]D-Fender: I read the README's for dahdi-tools and dahdi-linux, and I don't anything on headers |
14:51.20 | Katty | herroes. |
14:51.21 | Katty | again |
14:51.51 | drako | jaytee: looks pretty nice |
14:52.26 | leifmadsen | huh, well recording video prompts with H263 looks ok, but not so much with H264 :) |
14:53.54 | ManxPower-work | kuku: do "rpm -qa | grep kernel" and pastebin the result. |
14:56.47 | kuku | http://pastebin.com/d4ed9b79 |
14:57.43 | *** join/#asterisk chilicuil (~sistemas@unaffiliated/chilicuil) |
14:57.56 | ManxPower-work | kernel-headers-2.6.18-164.11.1.el5.centos.plus |
14:58.08 | ManxPower-work | that does not match your running kernel. |
14:58.20 | kuku | remove ? |
14:58.34 | ManxPower-work | I would remove the old RPMs and make sure you have the same version of kernel, kernel-dev and kernel-headers |
14:58.53 | ManxPower-work | then reboot to make sure your running kernel is the same as your current kernel |
14:59.32 | ManxPower-work | you should be able to leave the actual kernel RPMs, but remove any of the kernel-dev and kernel-headers and reinstall the same version of those RPMs as your kernel. If they don't all match DAHDI won't compile. |
14:59.46 | kuku | makes sense |
15:00.11 | kuku | yum wants to remove gcc as part of a dependency of kernel-headers-2.6.18-164.11.1.el5.c entos.plus |
15:01.34 | kuku | I'm looking for the flag so it doesnt remove gcc |
15:02.29 | leifmadsen | if you install the newer versions first, then it might not do that |
15:03.34 | leifmadsen | kuku: rpm -e --nodeps packagename -or- |
15:03.34 | leifmadsen | yum remove --nodeps packagename |
15:03.57 | kuku | http://pastebin.com/d37172bb8 |
15:04.04 | kuku | ok |
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15:13.40 | *** part/#asterisk benngard (~benngard@213.88.138.230) |
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15:14.53 | *** join/#asterisk mnick86 (~mnick86@188.195.88.246) |
15:14.55 | kuku | leifmadsen: thank you sir ! I removed the old devel rpm's and installed an older PAE devel ( since I had the latest kernel, but since I didn't reboot, it wasn't loaded ) |
15:15.13 | kuku | leifmadsen: dahdi is now compiling - thank you once again. |
15:15.28 | mnick86 | does somebody know if US-analog-phones use the same standard than german analog phones ? |
15:15.41 | leifmadsen | kuku: thank ManxPower-work, he did most of the suggesting :) |
15:16.55 | *** join/#asterisk Zambezi (Zulu@unaffiliated/zambezi) |
15:16.58 | kuku | spoke too soon: http://pastebin.com/d5006db0f |
15:19.52 | mnick86 | some german asterisk user online ?! |
15:20.29 | lbarth | yes |
15:21.07 | *** join/#asterisk mike8 (~mike@c14.audioline.ba.cust.gts.sk) |
15:21.20 | lbarth | mnick86 how can i hel you |
15:21.56 | *** join/#asterisk styelz (~yoohoo@m0o0.mooo.com) |
15:22.09 | *** join/#asterisk sysreq (~sysreq@unaffiliated/sysreq) |
15:24.51 | mnick86 | lbarth: have you ever tried a german analog phone on a Digium analog card ? |
15:26.36 | jaytee | be back later, going home early |
15:27.35 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
15:27.35 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
15:32.40 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
15:35.55 | angryuser | Have someone found the origins of : WARNING[22868] channel.c: Exceptionally long voice queue length queuing to Local/1010@from-internal-2277,2 (asterisk 1.4.27) ? |
15:35.58 | tzafrir_laptop | mnick86, is there such a thing as a german analog phone? |
15:36.04 | *** join/#asterisk KavanS (~KavanS@static-173-50-141-22.ptldor.fios.verizon.net) |
15:37.03 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
15:37.03 | *** mode/#asterisk [+o putnopvut] by ChanServ |
15:38.32 | *** join/#asterisk ELSEGO (~Juankar@unaffiliated/elsego) |
15:38.56 | ELSEGO | hi people |
15:39.00 | mnick86 | I can plug an anlog phone into "FXO Kewlstart" right ?! |
15:39.08 | *** part/#asterisk ELSEGO (~Juankar@unaffiliated/elsego) |
15:39.53 | Chainsaw | mnick86: An FXS port running FXO kewlstart signalling is suitable for that, yes. |
15:40.15 | *** join/#asterisk szasz (~szasz.sza@89.238.223.70) |
15:41.54 | szasz | hi all |
15:41.55 | Naikrovek | kewlstart? |
15:42.18 | szasz | in some calls I found that my Asterisk doesn't answer the BYE message |
15:42.22 | mnick86 | ok thanks ... damned phones |
15:43.09 | tzafrir_laptop | Naikrovek, though the phone couldn't care less if it is Kelstart or Loopstart |
15:43.16 | Chainsaw | Naikrovek: It's a fancy word for disconnect supervision. |
15:43.19 | Naikrovek | i was asking wtf kewlstart was |
15:43.27 | Naikrovek | kewlstart doesn't seem like a real name is all |
15:43.32 | Naikrovek | but if it is that's kewl |
15:43.40 | szasz | how can I solve this issue? |
15:44.03 | Naikrovek | szasz: you'll need to provide some SIP debugs of the behavior |
15:44.26 | *** join/#asterisk Skeeter- (~Skeeter@190-141.cgocable.ca) |
15:44.27 | Naikrovek | so we can see the reason (or at least what Asterisk sees as the reason) for ignoring the BYE |
15:45.07 | JonaY | Hi, I'm having a problem with asterisk - as soon as I connect our ISDN line I can no longer make internal calls, I've only got one outbound route (9|.) anyone got any ideas? |
15:45.36 | Chainsaw | JonaY: It depends on where you're connecting this ISDN line really. |
15:46.25 | mnick86 | how many wires does an american analog phone has in their connection cable ? |
15:46.38 | JonaY | It's into a Fritz PCI card, using the mISDN driver |
15:47.05 | szasz | it's hard to reproduce, but I will try |
15:47.11 | [TK]D-Fender | JonaY: FreePBX is NOT supported here... |
15:47.12 | [TK]D-Fender | ~freepbx |
15:47.13 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there, or fpbx |
15:47.16 | [TK]D-Fender | ^^^^ |
15:47.30 | *** join/#asterisk KavanS (~KavanS@static-173-50-141-22.ptldor.fios.verizon.net) |
15:49.14 | JonaY | ok thanks I'll try there |
15:50.03 | szasz | Naikrovek : are you talking about sip debugs from Asterisk or is enough the sip trace from my wireshark? |
15:51.52 | krion | how do you get the jitter from different call ? and what's an acceptable average jitter |
15:52.59 | *** join/#asterisk Tagor (~none@s55928c6d.adsl.wanadoo.nl) |
15:54.09 | Tagor | It seems my Asterisk is using random ports. I have set port 5060 as SIP port in sip.conf and RTP 10000_20000. But in my debug I see this: |
15:54.13 | Tagor | ast_rtcp_write_sr: RTCP SR transmission error to xxx.xxx.xxx.xxx:5005, rtcp halted Operation not permitted |
15:54.49 | Tagor | Anyone who knows how to force Asterisk to use port 5060? |
15:54.56 | [TK]D-Fender | Tagor: TRANSMISSION. the port you set are what you RECEIVE ON |
15:54.56 | Tagor | Or does it need to use more ports? |
15:55.37 | Tagor | [TK]D-Fender: Ah I see, 5005 is the port of the other party? |
15:55.47 | [TK]D-Fender | Tagor: Clearly |
15:58.06 | Naikrovek | szasz: sip debugs from asterisk. |
15:58.27 | Tagor | Another thing, in my debug output I see this: Peer audio RTP is at port 194.120.0.166:41880. This is also the other party right? |
15:59.51 | *** join/#asterisk jpeeler (~jpeeler@asterisk/digium-software-dev/jpeeler) |
15:59.55 | ManxPower-work | Tagor: that looks NATish |
16:00.17 | *** join/#asterisk cweagans (~cweagans@71-33-110-201.bois.qwest.net) |
16:00.20 | ManxPower-work | Tagor: we don't know since you masked all your IP addresses. |
16:00.35 | Tagor | ManxPower-work: that's the ip of my sip provider |
16:01.11 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
16:01.18 | ManxPower-work | remember you can only control the ASTERISK side of the connection. |
16:01.33 | Tagor | ManxPower-work: I'm trying to find out why I don't hear anything when my firewall is on. I opened 5060 and 10000_20000. As far as I can see in the debug (Audio is at 85.92.137.169 port 14758) it should work |
16:01.53 | Tagor | ManxPower-work: does that mean that if the provider uses other ports I need to open them too? |
16:02.50 | ManxPower-work | Tagor: do you have a FIREWALL or just a NAT ROUTER? |
16:03.19 | *** join/#asterisk myrthful (~emondpd@mercury3.Physics.McMaster.CA) |
16:03.24 | Tagor | ManxPower-work: just a firewall (APF), the server is connected to the internet directly |
16:03.42 | ManxPower-work | then you should see which packets are being blocked in your firewall logs. |
16:04.04 | Tagor | ManxPower-work: any idea how I can do this with APF? |
16:04.08 | ManxPower-work | I suepect you set your firewall to expect the far end to use the same ports as you configured in Asterisk. That is not the case. |
16:04.19 | ManxPower-work | Tagor: I can't help support your firewall. |
16:04.25 | myrthful | Would someone mind helping me figure out why my SendDTMF calls don't seem to be working? |
16:04.44 | ManxPower-work | myrthful: You don't hear the DTMF when calling that extension? |
16:04.59 | myrthful | Exactly |
16:05.13 | Tagor | ManxPower-work: does that mean it uses other ports besides the ones I configured in Asterisk (5060 / 10000_20000)? |
16:05.14 | myrthful | I've read about other people having this problem, but can't find a solution |
16:05.21 | ManxPower-work | myrthful: pastebin the output of a failed call including the SendDTMF. |
16:05.32 | ManxPower-work | Tagor: the FAR END will, yes. |
16:05.52 | ManxPower-work | ~pb |
16:05.53 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , or apt-get install pastebinit |
16:05.56 | myrthful | ManxPower-work: Ok, one minute |
16:06.02 | myrthful | ManxPower-work: and thanks for your help |
16:06.48 | Tagor | ManxPower-work: sorry, could be my English, but what do you mean with 'the far end will'? |
16:06.58 | ManxPower-work | Tagor: This is networking 101, you should know that the source port of a connection is usually chosen by the OS. |
16:07.25 | ManxPower-work | Tagor: each connection has two endpoints. In your case, one endpoint is Asterisk, the other endpoint is your provider. |
16:08.12 | ManxPower-work | You can only control the ports Asterisk uses. Your firewall rules should not specify the ports or the IP the provider will use. |
16:09.06 | *** join/#asterisk GGD (~GGD@ip72-196-241-104.dc.dc.cox.net) |
16:09.16 | ManxPower-work | The provider could easily send audio from a totally different set of IP addresses and ports. |
16:09.38 | *** join/#asterisk styelz (~yoohoo@m0o0.mooo.com) |
16:10.04 | Tagor | ManxPower-work: but if Asterisk is only listening on the ports I configured then it wouldn't work either if I had turned my firewall off? |
16:10.20 | ManxPower-work | Tagor: That is wrong. |
16:10.34 | ManxPower-work | The port numbers are NEGOTIATED for each call. |
16:10.37 | *** join/#asterisk Deeewayne (~dwayne@75.76.254.162) |
16:10.38 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
16:10.56 | bmoraca_work | Tagor: the RTP ports you specify are the ports that asterisk will use to initiate a connection with an endpoint. the actual ports used will be, as ManxPower-work states, are negotiated as part of the call setup (INVITE) |
16:11.42 | ManxPower-work | Tagor: So it doesn't matter what ports the far end uses, the far end tells Asterisk what ports it will use during the call setup. |
16:11.46 | Tagor | ManxPower-work: ok so they are assigned by my Asterisk box? So if I say the traffic should go over 10000_20000, it WILL go over these ports? |
16:12.21 | ManxPower-work | Tagor: Do you understand that all IP connections have a source IP/source port and destination IP/destination port? |
16:12.33 | Tagor | ManxPower-work: yes I do understand that part |
16:12.54 | *** join/#asterisk Skeeter- (~Skeeter@190-141.cgocable.ca) |
16:12.56 | ManxPower-work | Tagor: then your question makes no sense. |
16:13.03 | ManxPower-work | (11:11:46 AM) Tagor: ManxPower-work: ok so they are assigned by my Asterisk box? So if I say the traffic should go over 10000_20000, it WILL go over these ports? |
16:13.14 | ManxPower-work | That makes no sense since you do not talk about the other end of the connection. |
16:13.45 | *** join/#asterisk jakent (~john@68-247-204-36.pools.spcsdns.net) |
16:13.45 | ManxPower-work | Asterisk will SEND AUDIO FROM the ports you configure in rtp.con. the DESTINATION PORT could be ANY PORT. |
16:13.58 | Tagor | ManxPower-work: I thought it negotiates over port 5060 (standard sip port) and then it will establish a connection on one of the RTP ports I allow? |
16:14.23 | ManxPower-work | Tagor: Each call has two audio connections. Asterisk -> provider and provider-> asterisk. |
16:15.09 | Tagor | ManxPower-work: Ok, so the provider's ports do matter? |
16:15.21 | Tagor | ManxPower-work: so I should find out which ports my provider uses? |
16:15.22 | ManxPower-work | when the call gets set up, asterisk says, this the range of audio ports on my side that I will talk on. In your case that is 10,000 - 20,000. The provider then says "these are the ports I will accept audio on". |
16:15.33 | *** join/#asterisk Heretic (~fallen@dsl-246-127-184.telkomadsl.co.za) |
16:15.40 | ManxPower-work | Tagor: no! You should not limit what ports you allow in your firewall for packets coming from or going to your provider. |
16:15.54 | Heretic | lo all |
16:15.54 | *** join/#asterisk atis_work (~atis_work@193.238.212.171) |
16:16.00 | ManxPower-work | You are doing that. Stop doing that. |
16:17.12 | Tagor | ManxPower-work: but what if my provider uses several ips? They use a domain and the ips are different every time |
16:17.25 | ManxPower-work | Tagor: Nobody ever said this was going to be easy. |
16:18.07 | *** join/#asterisk Linuturk (~linuturk@unaffiliated/linuturk) |
16:18.12 | ManxPower-work | I wish you the BEST of luck and cannot help you further. |
16:18.21 | *** join/#asterisk darkskiez_ (~dz@62-50-207-156.client.stsn.net) |
16:18.49 | Tagor | ManxPower-work: well the thing is this worked with my previous provider. I just openend 5060 and 10000_20000 and it worked. Besides that if I want to connect to my Asterisk box from another location I will never be able to if I need to enter the ips first |
16:19.36 | *** join/#asterisk Victor_Yure (~victor@unaffiliated/victoryure/x-837844) |
16:19.57 | ManxPower-work | It's not that difficult to set your firewall to ignore the far end address/port |
16:20.18 | p3nguin | Why would you want to ignore your RTP stream, anyway? |
16:20.39 | p3nguin | Forward 5060 and 10000:20000. Done. Enjoy. |
16:20.42 | *** join/#asterisk cobolfoo (~cobolfoo@bas7-quebec14-1096763909.dsl.bell.ca) |
16:20.57 | cobolfoo | Hello, I need some support with DAHDI developement in C, there is the good channel to ask questions? |
16:20.58 | fenrus | allow established tcp sessions <3 |
16:21.07 | fenrus | coffe. |
16:21.17 | p3nguin | RTP is UDP, not TCP. |
16:21.24 | ManxPower-work | fenrus: too bad SIP and RTP is UDP, not TCP. |
16:21.41 | ManxPower-work | cobolfoo: #asterisk-dev |
16:21.45 | cobolfoo | thank you |
16:21.46 | p3nguin | SIP _can_ be TCP, but not many people use it there. |
16:21.46 | fenrus | ManxPower-work, indeed. |
16:21.52 | Tagor | p3nguin: I have allowed 5060 10000_20000 UDP in my firewall. But I still have no audio |
16:22.16 | *** join/#asterisk joesuffceren (~chatzilla@ip68-104-167-226.ph.ph.cox.net) |
16:22.20 | p3nguin | tagor: Are there other problems that I don't know about? Keep in mind I just got here. |
16:22.22 | joesuffceren | anyone know if it's possible to setup a call reject/ignore button or softkey on the cisco 7940s with SIP firmware? |
16:22.38 | ManxPower-work | p3nguin: his problem is that he neither understands SIP nor his firewall. |
16:23.34 | ManxPower-work | Maybe he has SIP AGL enabled on his firewall. It's a firewall issue, not an Asterisk issue. |
16:24.39 | *** join/#asterisk moy (~moy@74.12.129.100) |
16:25.03 | Tagor | p3nguin: the problem is that my Asterisk box doesn't work (no audio) when my firewall is enabled. I just switched my outgoing sip provider. It worked fine with the same firewall rules with my previous provider |
16:25.31 | *** join/#asterisk CunningPike (~CunningPi@204.239.8.157) |
16:26.01 | Tagor | ManxPower-work: I know it's a firewall issue. But I don't understand what's wrong. And it looks like I don't understand you, at least it's confusing me |
16:27.24 | Skeeter- | which OS fits best Asterisk? |
16:27.40 | Tagor | ManxPower-work: I don't understand why you can't tell me what is wrong or how I can solve it. I mean I don't have to know how to make a car to be able to drive it. |
16:27.52 | ManxPower-work | Skeeter-: Linux |
16:28.01 | Skeeter- | i meant distro, sorry |
16:28.08 | ManxPower-work | Skeeter-: the one you prefer. |
16:28.09 | Qwell | Skeeter-: doesn't matter |
16:28.21 | Skeeter- | Debian is giving me a hard time |
16:28.21 | *** join/#asterisk defswork (~andy@mx2.3gcomms.co.uk) |
16:28.32 | ManxPower-work | Tagor: you are not trying to drive a car. You are trying to rebuild part of your car. |
16:28.39 | Skeeter- | I would like to switch to another one, and i need suggestion |
16:29.09 | *** join/#asterisk jmacz (~jmacz@190.144.75.22) |
16:29.19 | Tagor | ManxPower-work: what's the problem with telling me what I'm doing wrong? |
16:29.43 | *** join/#asterisk eppigy (~Dave@216-139-241-102.aus.us.siteprotect.com) |
16:29.51 | eppigy | hello i am dave |
16:31.05 | Tagor | ManxPower-work: you're trying to force me to make a study of this instead of telling me what I'm doing wrong. I first need to do a knowledge test before you tell me what I'm doing wrong... I don't see the point of this |
16:31.06 | Skeeter- | ManxPower-work, Qwell : what distro are you guys using |
16:31.06 | ManxPower-work | Tagor: because I am not interested in spending 6 hours working with you to learn your firewall, your network, your setup, and your provider. Specifically I'm not interested in learning, and then teaching you how to debug your firewall. |
16:31.13 | ManxPower-work | Skeeter-: I use CentOS |
16:31.40 | *** join/#asterisk hluesea (~hulusikah@88.247.127.66) |
16:32.00 | ManxPower-work | Skeeter-: You'll find people here that run Asterisk just peachy on Debian |
16:32.02 | Tagor | ManxPower-work: probably this is some small thing I did wrong. For example I allowed the wrong ports |
16:32.44 | Tagor | ManxPower-work: I have never asked you to teach me how a firewall works etc. I just asked what could cause that there's no audio |
16:33.11 | *** join/#asterisk kruemeltee (~Maddin@port-92-198-62-82.static.qsc.de) |
16:33.18 | kruemeltee | hello again ;-) |
16:33.23 | ManxPower-work | And I told you what could cause no audio. |
16:33.49 | *** join/#asterisk JT (~j@unaffiliated/jt) |
16:34.04 | ManxPower-work | Log what the firewall is blocking, then go from there. Exactly like anyone else would debug the same issue. |
16:34.06 | Tagor | ManxPower-work: I already knew, my firewall is blocking traffic. But the question is WHY is it blocking the traffic? And don't tell me know 'cause I configured it like this', cause I already know that |
16:34.16 | Skeeter- | ManxPower-work, asterisk part is very fine, i cant get those Sangoma A200 to work properly |
16:34.22 | ManxPower-work | Tagor: you are missing the "WHAT is being blocked". |
16:34.43 | ManxPower-work | Skeeter-: did you follow the Asterisk install instructions from Sangoma's Wiki site? |
16:35.13 | ManxPower-work | Tagor: when you can put a pastebin of your firewall blocking packets then someone may be able to help your further. |
16:35.23 | ManxPower-work | But *I* have a job and cannot spend any more time with you. |
16:37.01 | Skeeter- | ManxPower-work, yes i did, somehow it try to go out via ZAP/g0 instead of DAHDI/g0 |
16:37.55 | ManxPower-work | Skeeter-: We can't support a GUI Asterisk here. |
16:37.57 | Skeeter- | Ive done that kinda of setup before and got everything working just fine.. I would like to fix it instead of formatting and reinstalling everything |
16:38.46 | [TK]D-Fender | Skeeter-: If you don't like the channel type its dialing, then fix your config, fix your GUI, or ditch both and do it yourself |
16:39.49 | Skeeter- | ManxPower-work, the GUI is not installed yet, i always try to make the Sangoma work then i install the GUI |
16:40.23 | *** join/#asterisk ttl- (~patrick@d5153A420.access.telenet.be) |
16:40.27 | [TK]D-Fender | Skeeter-: well "somehow it try to go out via ZAP/g0 instead of DAHDI/g0" = dialplan. Go fix it |
16:40.37 | ManxPower-work | Skeeter-: Dude, YOU CONFIGURED it. Just fix the damn channel name. |
16:41.22 | Skeeter- | [TK]D-Fender, ManxPower-work : making a call |
16:41.45 | Skeeter- | aight, instead of CHANUNAVAILABLE, i get CONGESTION |
16:41.57 | p3nguin | tagor: "iptables -L -nv" and put the output into a pastebin. If you're NATing, show "iptables -t nat -L PREROUTING -nv" too. |
16:43.12 | myrthful | ManxPower-work: I've got a pastebin for the PlayDTMF problem I mentioned earlier: http://pastebin.com/d633b1478 |
16:43.15 | Skeeter- | i used ./Setup dahdi, with that command, there is not much more during the configuration process that you can do |
16:43.49 | p3nguin | skeeter-: Distro choice is like choosing what to have for supper. Some people use Debian, some CentOS, I use ArchLinux, I've seen people using Gentoo... Pick one, use it, like it. If if you don't like it, change it. |
16:43.56 | Chainsaw | leifmadsen: Ready for testing now, I've backported for 1.6.1 & 1.6.2: https://issues.asterisk.org/view.php?id=16470 |
16:44.41 | Skeeter- | p3nguin, i like Debian, i got it for my home server and works just as espected for what I wanna do, i just think that it aint the best choice for Asterisk |
16:44.48 | leifmadsen | Chainsaw: okie, will update status |
16:45.13 | leifmadsen | Skeeter-: works fine -- just don't use the built in packages which tend to be woefully out of date |
16:45.57 | Skeeter- | leifmadsen, what do you mean by built in packages?? |
16:46.04 | p3nguin | skeeter-: I agree... I wouldn't use Debian for Asterisk, but I also wouldn't use it for anything. |
16:46.09 | leifmadsen | Skeeter-: I mean don't install asterisk via apt-get |
16:46.15 | leifmadsen | p3nguin: amen :) |
16:46.19 | Skeeter- | leifmadsen, i dont use apt-get for asterisk |
16:46.31 | leifmadsen | Skeeter-: I never said you did -- I just said don't use the packages via apt-get |
16:46.33 | Skeeter- | leifmadsen, i download the files from asterisk and Compile |
16:46.44 | leifmadsen | I like Ubuntu and CentOS |
16:46.50 | Skeeter- | leifmadsen, no problem, just wanted to specified |
16:46.55 | leifmadsen | but whatever works is what you should use |
16:47.00 | p3nguin | Ubuntu falls into the Debian category. |
16:47.20 | Skeeter- | Ubuntu has no restriction for non-free package |
16:49.05 | Skeeter- | the Linux-guy here always uses Gentoo |
16:49.27 | Tagor | ManxPower-work: here's the answer you asked for: PROTO=UDP SPT=19842 DPT=24872 |
16:49.57 | Skeeter- | btw, is there any Timecard addon/app for Asterisk? |
16:50.06 | p3nguin | tagor: Is that on an outbound packet? |
16:50.15 | Tagor | p3nguin: yes |
16:50.29 | Tagor | p3nguin: ** OUT_UDP DROP ** IN= OUT=eth0 SRC=85.92.137.169 DST=77.72.168.144 LEN=200 TOS=0x00 PREC=0x00 TTL=64 ID=0 DF PROTO=UDP SPT=19842 DPT=24872 LEN=180 |
16:50.51 | thehar | Anyone in Utah needing a job. We're hiring a Director of NetOps @ XMission http://saltlakecity.craigslist.org/sad/1596128717.html |
16:50.54 | *** join/#asterisk JT (~j@unaffiliated/jt) |
16:51.05 | bmoraca_work | Skeeter-: you could build one. there are so many ways that you could do it that no one else's method is probably going to work. |
16:51.06 | Skeeter- | 2nd, Cisco guy came to my work this week to show me how Cisco works, Cisco got a little "Wizard" when a user use the Voicemail for the 1st time, that would be great to be included in Asterisk |
16:51.20 | p3nguin | tagor: It's dropping the audio packets? That seems like a problem. |
16:51.46 | Tagor | p3nguin: thanks I will look into that |
16:51.48 | Qwell | thehar: How does it feel to host the best website on the planet? |
16:52.15 | thehar | Qwell: haha.. |
16:52.20 | Qwell | err, universe |
16:52.24 | thehar | Qwell: he is the only customer that has a subdomain to us as well |
16:52.29 | Qwell | seriously? |
16:52.31 | thehar | yup |
16:52.38 | Qwell | how'd he manage that? |
16:52.45 | thehar | because maddox is awesome |
16:52.51 | Qwell | That he is.. |
16:53.19 | *** join/#asterisk dandate2 (~dan@112.206.130.149) |
16:53.35 | thehar | nods |
16:54.02 | p3nguin | tagor: The iptables output I asked for would most likely reveal the problem. |
16:54.19 | dandate2 | when using asterisk 1.4: queue set to auto-fill: when there are a lot of people waiting on hold the cpu use of trying to ring multiple agents becomes so astronomical it damages the sound quality. <> Is this fixed in asterisk 1.6 or do I just need to get a faster machine? |
16:54.53 | p3nguin | dandate2: What speed is your CPU? |
16:55.11 | dandate2 | 2.2 ghz dual core intel |
16:55.40 | dandate2 | with 10 people waiting on hold the cpu use jumps to 90%, if i disable auto-fill it dallies between 2% and 25% |
16:58.16 | dandate2 | but sucks without auto-fill because there are 30 people ready to take calls, and it only rings one at a time |
16:58.43 | *** join/#asterisk Zambezi (Zulu@unaffiliated/zambezi) |
16:58.46 | p3nguin | even using ringall strategy? |
16:58.56 | dandate2 | yes ringall strategy takes a lot of cpu also |
16:59.25 | bmoraca_work | dandate2: just use roundrobin with a lower timeout. |
16:59.41 | Tagor | p3nguin: according the the output of iptables -L I think it's set correctly: |
16:59.44 | Tagor | Chain OUTPUT (policy ACCEPT) |
16:59.44 | Tagor | ACCEPT udp -- anywhere anywhere udp dpts:10000:20000 |
16:59.57 | p3nguin | rrmemory, that is... since roundrobin is deprecated |
17:00.12 | bmoraca_work | well, that's a given. the concept is the same, though |
17:00.14 | *** join/#asterisk paulc (~paulc@unaffiliated/paulc) |
17:00.22 | fenrus | anyone have a rrmemory configuration guide?, cant seem to get it working quite right |
17:00.29 | dandate2 | i just need to get a quad-core u-server huh |
17:00.47 | bmoraca_work | no... |
17:00.55 | p3nguin | fenrus: It's simple. Have strategy set to rrmemory and have multiple agents waiting for calls. |
17:01.05 | sun28 | strategy = rrmemory |
17:01.08 | sun28 | servicelevel = 10 |
17:01.13 | sun28 | timeout = 3 |
17:01.17 | sun28 | retry = 2 |
17:01.21 | sun28 | weight = 10 |
17:01.24 | fenrus | p3nguin, well - asterisk wont send a call past the first phone.. |
17:01.25 | sun28 | ringinuse = no |
17:01.25 | p3nguin | fenrus: What doesn't work? There's really nothing more to it. |
17:01.29 | sun28 | timeoutrestart = yes |
17:01.31 | sun28 | -_- |
17:01.39 | ManxPower-work | fenrus: IT SHOULD |
17:01.43 | *** join/#asterisk timeshell (~chatzilla@gw.lusi.on.ca) |
17:01.57 | fenrus | ManxPower-work, no matter how i'ts configured..? |
17:02.16 | ManxPower-work | fenrus: Obviously you screwed up your config somewhere. |
17:02.27 | fenrus | ManxPower-work, that's why i asked for some input ;) |
17:02.39 | ManxPower-work | For example maybe you don't have a call-limit set, or maybe yo don't have ringinuse=no |
17:03.02 | *** join/#asterisk came0 (~came0@rrcs-71-42-53-182.se.biz.rr.com) |
17:03.36 | bmoraca_work | indeed...queues don't work right without call-limit set...even if it's set to an absurdly high number |
17:03.44 | *** join/#asterisk ecrane (~ecrane@o1-69-19-166-10.static.o1.com) |
17:04.11 | *** join/#asterisk etnos (~chatzilla@190.98.20.221) |
17:04.38 | etnos | hi you all. |
17:04.42 | fenrus | hm, interesting.. |
17:06.24 | fenrus | setting the call-limit of the member of the group? |
17:09.56 | ManxPower-work | bmoraca_work: you'd think that info would be in queues.conf.sample or sip.conf.sample or UPGRADE*.txt |
17:10.34 | p3nguin | If it isn't, maybe someone will include it soon. |
17:12.22 | ManxPower-work | not if he's running 1.4 |
17:13.48 | *** join/#asterisk KavanS (~KavanS@static-173-50-141-22.ptldor.fios.verizon.net) |
17:15.51 | *** join/#asterisk mort_gib (~mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
17:19.29 | *** join/#asterisk JT (~j@unaffiliated/jt) |
17:20.03 | *** join/#asterisk benngard (~benngard@90-230-92-67-no148.tbcn.telia.com) |
17:21.54 | bmoraca_work | ManxPower-work: it's been an issue since I started using Asterisk...i'd be surprised if it isn't in almost every single config file associated with queues and endpoints |
17:24.24 | bmoraca_work | hrm...at&t's increasing the cost of my measured rate line...$15/mo instead of $12/mo |
17:24.34 | p3nguin | There's nothing about call-limit in queue.conf. |
17:25.46 | bmoraca_work | p3nguin: all i know is that it's one of the first issues I ran up against. anything that requires the status of a peer doesn't work if call-limit isn't set. i suspect that hints require it as well. |
17:26.08 | p3nguin | Hints were the only thing I knew about. |
17:26.23 | *** join/#asterisk albertoandrade (~albertoan@187.59.25.126) |
17:33.36 | *** join/#asterisk xperia (~chatzilla@zux182-249.adsl.green.ch) |
17:35.03 | xperia | hello to all. i have a quick small question about asterisk. is it possible to provide telebilling with asterisk. i need the possibility to offer telebilling and search for information how to do this from the hardware and software side. |
17:36.19 | *** join/#asterisk xperia (~chatzilla@zux182-249.adsl.green.ch) |
17:37.15 | xperia | ooops connection problem. was loged off. my question again. is it possible to make telebilling with asterisk ? if not what for possibility does exist ? |
17:39.07 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
17:39.28 | *** join/#asterisk avb (~avb@94stb14.codetel.net.do) |
17:39.34 | avb | he all |
17:39.35 | avb | hey |
17:39.58 | avb | guys, is there is any easy way to initiate outgoing calls and bridge them after? |
17:40.19 | avb | calls will be done via the sip |
17:40.20 | p3nguin | avb: originate from CLI |
17:40.29 | avb | from the extensions |
17:40.50 | p3nguin | okay |
17:41.22 | avb | what i figured out is only to create a meetme conference, and invite both of them |
17:41.49 | avb | but that doesnt seems clean and easy solution |
17:42.06 | avb | as sip can bridge 2 channels by itself |
17:42.25 | p3nguin | originate SIP/yourphone extension 5551212@outgoing |
17:42.29 | bmoraca_work | xperia: it is. you can use AGI or FUNC_ODBC depending on what other software you need to communicate with. |
17:43.31 | avb | p3nguin: sorry, havent got your, say i need to bridge 123123@myvoipprov and 242324@myvoipprov |
17:43.37 | avb | its kinda a callback |
17:43.58 | crochat | Hi everybody ! Got a big problem here with Dahdi 2.2.1 and multiple BRI in DDI (EuroISDN) |
17:44.17 | crochat | My hardware interface is a Junghanns QuadBRI (PCI) |
17:44.43 | xperia | bmoraca_work: woooww that is very impressive ! having the possibility to provide telebilling with asterisk is exactly what i need ! if it is really possible then big compliments to the programmers ! |
17:45.12 | crochat | Only the first BRI is "Provisioned, Up, Active". All the others are "Provisioned, In Alarm, Down, Active" |
17:45.15 | bmoraca_work | xperia: asterisk is simply a platform. what you implement with it is your own deal. |
17:46.32 | xperia | bmorca_work: do you have any experience with telebilling ? i am asking how exactly i can get the money from the caller after he called my number normally. i am totally new to this stuff. need a litlle more info about this. |
17:46.45 | crochat | All the BRI are connected ok. I tried to swap BRI (e. g. put BRI 2 on span 1 and it worked. Span1 was then up) |
17:47.04 | KavanS | thoughts....dahdi vs. zaptel...real reasons not just because "omg dahdi is new" |
17:47.13 | KavanS | for t400p hardware |
17:47.23 | KavanS | right now, I use asterisk 1.4.x and zaptel and it "just works! TM" |
17:47.30 | KavanS | I'm apprehensive about switching... |
17:48.05 | bmoraca_work | KavanS: if it ain't broke, don't fix it. that said, dahdi is very easy (easier than zaptel) to set up |
17:48.10 | p3nguin | If you're using an older 1.4 version and zaptel works, why are we even discussing it? |
17:48.28 | KavanS | well, I just wanted to see if there was any reasons to upgrade to zaptel I was unaware |
17:48.29 | KavanS | of... |
17:48.41 | KavanS | I suppose reading the changelog would be a step in the right direction, but sometimes just asking works... |
17:48.41 | bmoraca_work | there probably is not |
17:48.48 | KavanS | ok, roger that, thanks for the pro tips |
17:49.07 | bmoraca_work | on a new install, though, go dahdi |
17:49.13 | drako | im about to install a new system with a AEX800 and im wondering if i should go zaptel or dahdi |
17:50.01 | KavanS | bmoraca_work, yep it's new hardware |
17:50.19 | crochat | KavanS: We had sooooo many crashes with BRIstuff... so Dahdi was the only solution for us |
17:50.20 | KavanS | I also have hylafax using the t400p for fax, that also "just works" |
17:50.31 | bmoraca_work | drako & KavanS: new install, go dahdi |
17:50.38 | KavanS | roger that |
17:51.01 | bmoraca_work | KavanS: nothing wrong with hylafax...i use it, too. though i've heard that the fax functions in 1.6 work pretty well, I have not used them. |
17:51.35 | KavanS | ahh right on....well my EU counterparts are using 1.6, I'm just not ready to take the dive yet I suppose |
17:51.36 | crochat | But we had 3 BRI working with BRIstuff... with kernel crashes twice a month. And now, we have only one BRI working with Dahdi !!?? |
17:51.45 | drako | bmoraca_work: should i go 1.4 or 1.6 ? |
17:51.50 | KavanS | 1.4 works, and I have some "custom" dialplans....I just don't know how it'll play out in 1.6/1.8 |
17:51.58 | KavanS | should read some more I suppose |
17:52.07 | bmoraca_work | drako: brand new install, no existing configs, go 1.6 |
17:52.18 | bmoraca_work | KavanS: UPGRADE-*.txt will tell you any incompatibilities |
17:52.21 | drako | ok |
17:52.22 | p3nguin | 1.4 is considered long-time support, where 1.6 won't always be around. |
17:52.50 | drako | we wont be always around here |
17:54.49 | KavanS | drako, heh what? |
17:57.08 | *** join/#asterisk Badrobot- (~badrobot@cpe-76-173-229-89.socal.res.rr.com) |
17:59.46 | p3nguin | Anyone here use TurboTax Home and Business? I'm wondering if I need to purchase the 2009 version, or can I update the 2008 version and continue to use it this year. Anyone know? |
17:59.57 | *** part/#asterisk xperia (~chatzilla@zux182-249.adsl.green.ch) |
18:00.22 | *** join/#asterisk QubeZ (~qube@64.128.254.34) |
18:00.26 | QubeZ | hello all |
18:01.13 | Qwell | p3nguin: you buy the new version.. if you haven't tried the online version, you might consider doing so. it's pretty great |
18:02.54 | p3nguin | I'll take a look at it. If the online one is as easy as the installed one, there won't be any trouble with it. |
18:03.07 | Qwell | it is. and it's online, so...heh |
18:03.34 | p3nguin | Yeah, that'll keep me from having to use someone's Windows computer to do my taxes. |
18:03.38 | Qwell | not having to worry about storing the data files is great |
18:03.58 | Qwell | they store it all for you, and you can import previous years returns and such. pretty useful |
18:05.13 | tzafrir_laptop | crochat, basically Asterisk and libpri need to know to ignore that alarm and actively start the channel |
18:05.47 | *** join/#asterisk drako (~luisjose@nelug/coreteam/luisjose) |
18:06.14 | *** join/#asterisk Maximo (~maximo@CPE001217b1920e-CM00111ade9528.cpe.net.cable.rogers.com) |
18:08.15 | spenguin[work] | test |
18:08.36 | bmoraca_work | spenguin[work]: fail. |
18:09.42 | spenguin[work] | ohnoes |
18:11.18 | *** join/#asterisk jakent (~john@2001:470:8:1fc:226:8ff:fedd:93f6) |
18:12.02 | *** join/#asterisk abatista (~chatzilla@63.214.236.169) |
18:12.25 | myrthful | ManxPower-work: I've got a pastebin for the PlayDTMF problem I mentioned earlier: http://pastebin.com/d633b1478 |
18:13.07 | myrthful | Or, if anyone else can help me, I'm having trouble with PlayDTMF, it sends ok but I never hear the tones |
18:13.18 | myrthful | Any help is appreciated |
18:17.12 | abatista | hello folks |
18:18.03 | [TK]D-Fender | myrthful: What is that device? |
18:20.02 | myrthful | [TK]D-Fender: I'm not sure what you mean (warning, I'm an Asterisk newb) |
18:20.25 | drako | I dont get why are 2 branch 1.6.1 and 1.6.2 , whats the diference? |
18:20.29 | [TK]D-Fender | myrthful: Channel: SIP/emondpd-081bfd90 <------ what is this device? |
18:20.41 | [TK]D-Fender | drako: 0.0.1 |
18:21.17 | myrthful | [TK]D-Fender: I'm connecting from linphone, is that what you mean? |
18:21.18 | drako | erm |
18:21.38 | [TK]D-Fender | myrthful: Could be that the phone doesn't generate a tone so you won't hear it. |
18:22.38 | myrthful | [TK]D-Fender: Hmm, I can send tones from the linphone command line. I don't suppose that matters? |
18:22.58 | [TK]D-Fender | myrthful: Correct. it doesn't |
18:23.20 | myrthful | [TK]D-Fender: Ok, thanks |
18:25.18 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
18:25.23 | avb | ok |
18:25.28 | avb | meetme seems working fine |
18:26.30 | *** join/#asterisk voipmonk (~shido6@68.65.134.123) |
18:26.32 | *** join/#asterisk slima (slima@unaffiliated/slima) |
18:29.36 | benngard | shouldnt it? |
18:30.07 | avb | benngard: u missed my question :) |
18:30.19 | avb | i was looking a way to make a callback |
18:30.50 | avb | thought that meetme will not be a good solution, but in the end its not that bad :) |
18:31.43 | benngard | to like originate from meetme to setup a call? |
18:31.58 | benngard | or rather 2 calls, it will ofc work |
18:32.10 | Katty | stretches |
18:32.25 | Katty | goes in search of lunch |
18:32.32 | avb | benngard: i made it easy. im generating to originate calls can sending them into an extension which invites them into the conference |
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18:32.51 | benngard | goes and search fro a beer! |
18:32.51 | avb | generating spool files* |
18:33.10 | benngard | easy solutions are often the best ones |
18:33.40 | avb | yeh :) |
18:33.40 | avb | now i need to figure out whats wrong with a signaling on the freaking nokia internet calls app |
18:33.47 | avb | asterisk ignoring hangups from it |
18:35.26 | *** join/#asterisk Pimmetje (~Pimmetje@83.119.156.35) |
18:35.36 | voipmonk | avb whats the sip debug say? |
18:35.54 | avb | voipmonk: havent worked that out yet |
18:36.28 | *** join/#asterisk sier (~sier@unaffiliated/sier) |
18:36.37 | avb | preparing a manual for a client :) |
18:37.48 | Katty | returns with shrimp |
18:38.05 | *** join/#asterisk uqlev (~yuriy@91.184.221.31) |
18:38.25 | voipmonk | ooh technical writing.... fun |
18:38.27 | voipmonk | snorez |
18:41.17 | *** join/#asterisk hluesea (~hulusikah@88.247.127.66) |
18:41.48 | spenguin[work] | shrimp! |
18:42.48 | *** join/#asterisk Poincare (~jefffnode@amp89.ampersant.be) |
18:44.55 | voipmonk | wakes up.... |
18:44.57 | voipmonk | where? |
18:46.20 | spenguin[work] | voipmonk: I smell it |
18:48.36 | carrar | pours alfredo sauce all over Katty |
18:50.55 | Naikrovek | voipmonk: technical writing is interesting for me because, as i write, i think of what the reader would ask and often i wind up learning a ton about things i thought i knew |
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18:58.31 | Katty | carrar: ^_- |
19:00.16 | benngard | "Lanab Design AB (fax)" <038075855> <- is that a correct clid, i am thinking about the "(" and the ")"? |
19:00.30 | [TK]D-Fender | Naikrovek: If you end up learning a lot of stuff.. then you clearly don't know what you're writing about :p Go find someone qualified! |
19:00.38 | Naikrovek | i say feh |
19:00.50 | Naikrovek | i write documentation for software I write and I wind up learning things |
19:01.06 | Naikrovek | I wind up finding features I didn't know existed, etc |
19:01.19 | Naikrovek | and lots of bugs |
19:01.45 | Naikrovek | especially when i write APIs |
19:03.07 | *** join/#asterisk Wildy (~simba@178.176.93.29) |
19:04.16 | Skeeter- | anyone got access to work with Asterisk CRD, CSV files?? |
19:04.47 | Naikrovek | access? as in MS Access? |
19:05.02 | Skeeter- | ya |
19:05.09 | Naikrovek | can't say i have |
19:05.22 | Skeeter- | im open to any other program |
19:05.47 | *** join/#asterisk oej (~olle@eduroam-193-157-113-13.uio.no) |
19:05.47 | Skeeter- | i just wanna print out some stats |
19:06.27 | *** join/#asterisk oej_ (~olle@1x-193-157-196-250.uio.no) |
19:06.44 | Naikrovek | i've always put that kind of thing into mysql but that probably isn't the best solution for everyone |
19:07.16 | Skeeter- | Naikrovek, I would love to do that but i dont know how mysql works |
19:07.23 | Naikrovek | yeah that's fine |
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19:07.34 | Naikrovek | you wanna do total call time or something like that |
19:07.39 | Godfather_ | o/ |
19:07.43 | Naikrovek | access may be the best tool |
19:07.47 | ManxPower-work | Skeeter-: MS Access should support importing CSV files. I would recommend a spreadsheet instead. |
19:07.50 | Naikrovek | but i don't have access so i can't really help |
19:08.08 | Skeeter- | ManxPower-work, i tried excel but Access is much faster for reports |
19:08.25 | Skeeter- | ManxPower-work, i get an import error while importing it |
19:08.27 | Naikrovek | if he wants to query that data access would be better, i'd think |
19:08.31 | ManxPower-work | Skeeter-: I avoid using MS products when I can so I really can't help you. |
19:08.43 | Skeeter- | ManxPower-work, its ok |
19:08.54 | ManxPower-work | I would not even us Windows if I didn't have work applications that require it. |
19:08.55 | Naikrovek | you may ask in ##windows |
19:09.09 | Naikrovek | ManxPower-work: that's the whole basis of Windows' adoption |
19:09.15 | Skeeter- | I just want to get how many calls where made to XXX destination |
19:09.33 | Naikrovek | Skeeter-: access is probably best |
19:09.50 | Skeeter- | I dont use Windows a lot, its on my Virtualbox, some dude knows access well, but it doesnt import anything |
19:09.52 | ManxPower-work | grep "XXX" /var/log/asterisk/cdr-csv/Master.csv | wc |
19:10.17 | Naikrovek | that's not a bad idea |
19:10.29 | *** join/#asterisk lbarth (~lbarth@62.216.165.71) |
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19:12.24 | Skeeter- | <PROTECTED> |
19:12.38 | Naikrovek | wc -l will show you the number of lines (results) |
19:13.03 | Skeeter- | ok |
19:13.14 | Skeeter- | so that person received 2k calls right? |
19:13.17 | Naikrovek | first, just grep "XXX" Master.csv | less , and eyeball the output to make sure your grep is getting only what you want |
19:13.39 | Naikrovek | if it looks good, pipe it to wc -l rather than less |
19:13.56 | *** join/#asterisk rare1980_ (~sniper_ja@203.175.76.219) |
19:14.10 | rare1980_ | hello |
19:14.19 | Skeeter- | Naikrovek, works pretty well i guess |
19:14.31 | rare1980_ | any one expert in asterisk dialplan |
19:14.34 | Naikrovek | it's not ideal probably, but for quick things grep is nice |
19:14.34 | rare1980_ | ? |
19:14.41 | Naikrovek | rare1980_: just ask your question, the experts will pipe up |
19:14.49 | rare1980_ | sure |
19:17.40 | hluesea | how i can voip termination in asterisk (installation packets and extensions)? and are anyone prefer a free billing solution for sip termination ? I probably want to make comming calls forward another provider and make billings on my side. |
19:20.11 | rare1980_ | Naikrovek---- i need some help in asterisk dial plan... i hve miltple sip account has 3 channls and each account has limit to 300 mins call per week. i want 1st astiersk will use 3 sip chanls from one account and as soon as these 3 sip chanls are bz on call. then on 4th call astierks will use 2nd sip account for dialing. and it will use its 3 chanls and it will go like this.. |
19:20.50 | Qwell | rare1980_: are these 300 minute per week accounts free outgoing "test" accounts, or something? |
19:20.56 | *** join/#asterisk [TK]D-Fender (~chatzilla@216.191.106.163) |
19:21.05 | rare1980_ | no |
19:21.07 | ManxPower-work | rare1980_: you will end up spending weeks writing something that will do what you want. |
19:21.36 | [TK]D-Fender | ManxPower-work: How bad? |
19:21.48 | Qwell | [TK]D-Fender: sip accounts limited to 300 minutes per week |
19:21.50 | rare1980_ | is it diffcult? |
19:22.09 | [TK]D-Fender | Qwell: Not THAt bad.... |
19:22.23 | Qwell | ...switching between the accounts once the limit is reached |
19:22.44 | [TK]D-Fender | Qwell: You mean outbound load balancing? |
19:22.45 | rare1980_ | 2nd requirment= as soon as asterisk will use 300 mins from any account . then asterisk will not use that account for rest of the week |
19:22.59 | Qwell | [TK]D-Fender: if by "load" you mean "limited number of outbound minutes per week" |
19:23.18 | [TK]D-Fender | Qwell: Yeah I mean rotating trough carriers not to get screwed on overages. |
19:23.32 | ManxPower-work | [TK]D-Fender: keeping track of how mins have been used per peer along with rollover between the peers when either the number of calls exceeds 3 for that peer or if the mins used that month exceed 300 mins for that peer. |
19:24.05 | ManxPower-work | sounds practically impossible for a newbie to setup in even a day. |
19:24.10 | [TK]D-Fender | ManxPower-work: Yeah, a few hours to tractk the calls, etc |
19:24.37 | ManxPower-work | Someone like you or me or anyone else good with dialplan would not find it too hard to do. |
19:24.42 | *** part/#asterisk beek (~klinebl@pdpc/supporter/bronze/beek) |
19:24.59 | rare1980_ | soo |
19:25.00 | ManxPower-work | But someone like you or me does not count every single penny spent on calls. |
19:25.10 | rare1980_ | Manxpower_ can u do this? |
19:25.40 | Naikrovek | why not just get a provider that lets you make unlimited calls |
19:25.44 | ManxPower-work | rare1980_: Assuming you set up billing with the billing department and convince my boss to loan me out to you, then yes, I could do it. |
19:26.02 | rare1980_ | ok |
19:26.09 | ManxPower-work | I don't know what they would set my hourly rate at. |
19:26.11 | rare1980_ | and how much that billing would be? |
19:26.22 | ManxPower-work | rare1980_: I have no idea. We are a CLEC not a consulting company. |
19:26.42 | ManxPower-work | I doubt you'd get my boss to agree to loaning me out. |
19:26.44 | Naikrovek | typical rate is $125/hr for this kind of thing, I think |
19:26.50 | Naikrovek | ManxPower-work: moonlight |
19:27.05 | ManxPower-work | Naikrovek: I moonlight at my current job. |
19:27.07 | rare1980_ | and how many hours would it take? |
19:27.12 | rare1980_ | to do this work? |
19:27.16 | Qwell | wonders how many tenths of a cent would be saved by doing this |
19:27.18 | Naikrovek | wow you really need this |
19:27.19 | rare1980_ | ? |
19:27.26 | ManxPower-work | rare1980_: I do not know. I suggest you look at other options. |
19:27.33 | Naikrovek | rare1980_: contact voipmonk |
19:27.37 | Naikrovek | he can help you |
19:27.48 | Naikrovek | maybe it's "they" now |
19:27.55 | rare1980_ | ok |
19:28.00 | rare1980_ | thanks guys |
19:28.00 | Naikrovek | i've worked with voipmonk, he's good |
19:28.03 | Naikrovek | welcome |
19:28.07 | rare1980_ | for ur help |
19:28.24 | Naikrovek | thank voipmonk when he solves this for ya |
19:28.33 | rare1980_ | :) |
19:30.38 | Naikrovek | rare1980_: he'll be back online soon; he just signed off half an hour ago |
19:30.41 | Naikrovek | he'll be back |
19:30.45 | carrar | rare1980_, get writting, thats a simple agi script |
19:31.06 | carrar | You will learn good stuff in the process |
19:31.32 | [TK]D-Fender | carrar: Depending on how he intends on hoandling concurrency, timeouts, etc |
19:31.54 | carrar | well a simple roll over based on time used with that link is pretty simple |
19:31.55 | [TK]D-Fender | carrar: otherwise that brings in a live monitoring script and AMI all over the place |
19:33.10 | carrar | I have somthing similure for load balancing multiple sip peers based on usage |
19:33.34 | [TK]D-Fender | carrar: concurrency adds a lot of mess |
19:33.57 | carrar | Messy is where the fun begins :) |
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19:36.58 | Skeeter- | ManxPower-work, just found that Access give you a SQL table, but in the Master.csv there is no field |
19:37.16 | Naikrovek | the fields are comman separated |
19:37.21 | Naikrovek | s/comman/comma/ |
19:37.50 | carrar | What is Access |
19:37.56 | Naikrovek | MS Access |
19:37.59 | carrar | Whats MS? |
19:38.01 | Naikrovek | eh |
19:38.04 | Skeeter- | rofl |
19:38.08 | Skeeter- | i luv haters |
19:38.11 | carrar | :) |
19:38.18 | Skeeter- | M$ |
19:38.34 | carrar | I live about 3 bocks from the MS Bing building |
19:38.34 | Naikrovek | "M$" is the trademark of a douche |
19:38.47 | Naikrovek | s/douche/blowhard/ |
19:39.23 | Skeeter- | But Its Not Google |
19:39.42 | Skeeter- | wish i could make a graffiti of that on the building |
19:39.51 | [TK]D-Fender | Skeeter-: Either way, its a simple CSV |
19:40.06 | benngard | got a feelingt that u guys dont like ms ;) |
19:40.15 | Skeeter- | [TK]D-Fender, im aware of that, and it is very accurate |
19:40.23 | benngard | joins the party! |
19:42.15 | Naikrovek | i like MS software, the company is getting better but still evil |
19:42.51 | carrar | MS Software has gone downhill |
19:42.58 | carrar | I don't miss it |
19:43.00 | Naikrovek | wrong |
19:43.01 | Naikrovek | well |
19:43.03 | Naikrovek | you may miss it |
19:43.07 | Naikrovek | but it's gotten WAY better |
19:43.15 | Naikrovek | WAY WAY better |
19:43.18 | fenrus | Win7 semms rather nice |
19:43.26 | Naikrovek | i'm talking server stuff |
19:43.29 | Naikrovek | but yes win7 is nice |
19:44.01 | ManxPower-work | I guess that depends on what you call "way better". XP does not run nearly as fast as Win2k on my laptop. |
19:44.02 | *** join/#asterisk Poincare (~jefffnode@213.219.184.23) |
19:44.10 | ManxPower-work | I would call that "not better" |
19:44.16 | Naikrovek | XP doesn't run as fast as windows 7 on my laptop |
19:44.27 | Naikrovek | that's what I would call better |
19:44.36 | Naikrovek | obviously YMMV but overall they're far, far better |
19:44.38 | ManxPower-work | Just remember "better" does not mean "changed the tool bar so it looks different" |
19:44.55 | Naikrovek | that's not what i'm talking about |
19:45.07 | ManxPower-work | Skeeter-: Why don't you try OpenOffice. |
19:45.15 | Naikrovek | FAR easier to administrate now, I can script just about everything |
19:45.23 | Naikrovek | openoffice is great on paper, useless in practice |
19:45.39 | ManxPower-work | I find each version of Microsoft products to be even more confusing than the previous version. |
19:45.59 | [TK]D-Fender | MS Office 2007 burned a lot of users |
19:46.17 | Naikrovek | who can't learn new things, yes |
19:46.20 | ManxPower-work | I use Office 2003 on my work machine. |
19:46.36 | ManxPower-work | Naikrovek: you mean RElearn. |
19:46.43 | Naikrovek | i can do more with O2007 than I ever did with O2003 |
19:46.53 | Naikrovek | and I can do it all without the mouse |
19:47.04 | ManxPower-work | Other then Outlook randomly wanting me to reauthenticate, it doesn't seem to bog down my machine too much. |
19:47.40 | bmoraca_work | i freakin love Office 2007. it's so much faster and easier to use than 2003. |
19:47.46 | Naikrovek | agreed |
19:48.06 | Naikrovek | the keyboard shortcut mechanism introduced in 2007 is freaking stellar |
19:48.06 | carrar | If you can't do it with VI you shouldn't be doing it |
19:48.29 | ManxPower-work | I wonder how well 2007 would run on my 1.3 Ghz laptop (single proc, no HT) in my work VMWare virtual machine. |
19:48.40 | ManxPower-work | carrar: finally some sense around here. |
19:48.47 | bmoraca_work | Naikrovek: i like the ribbons, tbh |
19:48.48 | *** join/#asterisk arpu (~arpu@chello062178159144.10.14.univie.teleweb.at) |
19:48.56 | ManxPower-work | bmoraca_work: pervert |
19:49.04 | Naikrovek | hah |
19:49.10 | bmoraca_work | ManxPower-work: my wife runs it on a 1ghz Athlon XP with 768mb RAM without issue |
19:49.23 | bmoraca_work | ManxPower-work: don't be a hater. we all have our own tastes. :P |
19:49.54 | carrar | and some have poor taste |
19:49.55 | Naikrovek | that's his whole schtick |
19:49.56 | ManxPower-work | MS has done more than any other person or company to lower end user expectations of reliability on a PC. |
19:50.07 | Naikrovek | ManxPower-work: then you haven't used anything recent |
19:50.23 | ManxPower-work | Naikrovek: why would I. If they can't get right after 10 years..... |
19:50.49 | ManxPower-work | Much like Grandstream, they simply have too much bad history. |
19:50.59 | carrar | yup |
19:51.14 | carrar | They have too much cruft |
19:51.27 | Naikrovek | no they don't |
19:51.28 | Naikrovek | listen |
19:51.32 | Naikrovek | i'm no fan of microsoft |
19:51.37 | Naikrovek | but their recent software is really good |
19:51.46 | Naikrovek | and i say that because it genuinely is |
19:51.50 | Naikrovek | not because i want to argue |
19:51.54 | Naikrovek | or because i know a guy who works there |
19:52.04 | Naikrovek | i know because i use their stuff extensively |
19:52.07 | carrar | <Naikrovek> listen |
19:52.07 | carrar | <Naikrovek> i'm no fan of microsoft |
19:52.08 | Naikrovek | and have for 15 years |
19:52.09 | carrar | heh |
19:52.10 | ManxPower-work | Starting with Win2k, I consider their OS to be "reasonably stable" |
19:52.36 | Naikrovek | ManxPower-work: well win7 is worth a look for you i think |
19:52.50 | ManxPower-work | Naikrovek: How much is it going to cost me? |
19:52.57 | Naikrovek | in fact, the only bluescreens i've had in XP, Vista, or 7 have been related to hardware |
19:53.03 | Naikrovek | ManxPower-work: nothing to try it for 120 days |
19:53.15 | ManxPower-work | no, how much is it going to COST me? |
19:53.23 | Naikrovek | well i'm talking money |
19:53.28 | Naikrovek | what are you talking about |
19:53.28 | ManxPower-work | So am I. |
19:53.34 | Naikrovek | $0 for 120 days |
19:53.41 | Naikrovek | past that you have to reinstall or buy |
19:53.45 | ManxPower-work | The 120 days is just a hook. That is not the cost of the OS. |
19:53.46 | bmoraca_work | less dollars out right than man hours trying to make linux behave like windows in an active directory scheme |
19:53.58 | Naikrovek | linux is cheaper only if your time is worth $0 |
19:54.04 | carrar | ManxPower-wor, you will also need to buy 4GB more memory and bigger/faster processors |
19:54.14 | carrar | there is some real costs |
19:54.21 | carrar | and of course the license |
19:54.22 | Naikrovek | you won't need 4gb of ram |
19:54.24 | ManxPower-work | Naikrovek: *nod* So with Linux, I have to learn the OS. With MS I have to learn the OS AND pay money. I don't see the advantage to me. |
19:54.35 | Naikrovek | ManxPower-work: okay |
19:54.36 | carrar | and all the other licenses if you want to have remote users |
19:54.39 | ManxPower-work | Is it true you can't disable all the Win7 eye candy? |
19:54.43 | Naikrovek | you stay there while i experience progress. |
19:54.49 | ManxPower-work | Naikrovek: so how much does Win7 COST> |
19:54.52 | Naikrovek | ManxPower-work: not true |
19:54.54 | carrar | I expirence process with UNIX |
19:54.57 | carrar | it's faster |
19:54.57 | Naikrovek | ManxPower-work: for me: very little in time or money |
19:55.02 | carrar | progress |
19:55.13 | bmoraca_work | ManxPower-work: i paid $29 for my Windows 7 Professional upgrade license :P |
19:55.21 | ManxPower-work | I'm not spending $200 in a fscking OS. |
19:55.28 | ManxPower-work | bmoraca_work: upgrade from what? |
19:55.31 | Naikrovek | ManxPower-work: i spend less administrating 200 years on windows server 2008 with active directory and exchange, than I did on 10 users on linux |
19:55.35 | bmoraca_work | XP |
19:55.39 | carrar | How much was XP? |
19:55.46 | bmoraca_work | free |
19:55.50 | ManxPower-work | Naikrovek: and in my experience it is exactly the opposite. |
19:55.55 | carrar | XP is not free |
19:55.59 | bmoraca_work | it was for me |
19:56.04 | bmoraca_work | academic license ftw |
19:56.10 | Naikrovek | only fools pay MSRP |
19:56.25 | carrar | s/MSRP/MicroSfot/ |
19:56.29 | ManxPower-work | bmoraca_work: you must be an MSDN subscriber or you are not counting the cost of the OS being built into the cost of the PC. |
19:56.37 | bmoraca_work | ManxPower-work: there are a LOT of applications that do not work on linux that a lot of businesses need. |
19:56.45 | bmoraca_work | ManxPower-work: nope. both were academic discounts |
19:57.13 | Naikrovek | my employer would be out of business with linux. how much would linux cost us? everything, that's how much. |
19:57.15 | carrar | like? |
19:57.24 | ManxPower-work | bmoraca_work: That happens when you are a convicted monopolist. |
19:57.32 | bmoraca_work | ManxPower-work: for example, dental offices uses practice management and digital radiography packages that do not work and cannot work on Linux |
19:58.03 | bmoraca_work | ManxPower-work: i'm not talking about potential alternatives like openoffice (which is a piece of garbage). i'm talking about specialized software that DOES NOT RUN on anything but Windows. |
19:58.06 | ManxPower-work | bmoraca_work: you mean because they accepted the MS lockin when they went with their software. |
19:58.15 | carrar | he |
19:58.16 | carrar | h |
19:58.18 | bmoraca_work | ManxPower-work: no, because there is no alternative |
19:58.34 | ManxPower-work | bmoraca_work: and in those cases the choice to move away from MS is more expensive than the end user is willing to pay. |
19:58.37 | Naikrovek | and it's not because MS has a lock, it's because there's no one else who can make anything better |
19:58.40 | carrar | Could have been just as easy to write it on unix |
19:58.45 | ManxPower-work | It sucks, but they made their bed, they can sleep in it. |
19:58.46 | Naikrovek | carrar: lol |
19:58.47 | carrar | if not easier |
19:58.51 | Naikrovek | easier? wtf |
19:58.52 | Naikrovek | hahaah |
19:58.57 | carrar | open standards |
19:59.01 | bmoraca_work | ManxPower-work: no, the choice to move away from MS does not exist because there are no alternative software packages. |
19:59.21 | Naikrovek | open standards don't make it easier. |
19:59.24 | Naikrovek | or better |
19:59.27 | ManxPower-work | bmoraca_work: they can hire someone to write one. As I said, it would be too expensive for the customer. |
19:59.42 | ManxPower-work | But there is ALWAYS a choice. |
19:59.55 | Naikrovek | yes always a choice |
19:59.59 | Naikrovek | go out of business or use windows |
20:00.03 | bmoraca_work | not a rational or logical choice, but I suppose. |
20:00.32 | ManxPower-work | I have a choice of quitting my job and going to work for a company that does not use Windows. That choice is too much problem and effort for me to choose that option. I still have a choice. |
20:00.39 | Naikrovek | and let's not even get started on how shitty linux desktops are |
20:00.45 | Naikrovek | omg what pile those things |
20:00.47 | Naikrovek | are |
20:00.51 | bmoraca_work | point being: linux is absolutely NOT an option in every case. particularly at the workstation level. my god, I'd hate to have to support an office full of morons using linux workstations. i can't even begin to imagine that bullshit. |
20:01.03 | Naikrovek | any system that has to read from disk when i start typing in a notepad application is FLAWED |
20:01.18 | ManxPower-work | bmoraca_work: we are looking at moving some of our sales people to Linux workstations to cut down on support issues. |
20:01.34 | ManxPower-work | If you don't have permission to install crap support issues drop. |
20:01.37 | bmoraca_work | ManxPower-work: that's ass-backwards logic if i've ever heard of it. |
20:01.51 | carrar | nope |
20:01.59 | Naikrovek | yup |
20:02.02 | carrar | MS is ass-backwards logic |
20:02.03 | ManxPower-work | Once we get over the INITIAL training costs, of course. |
20:02.30 | ManxPower-work | No more removing spyware, no more removing viruses no more messing with the registry |
20:02.31 | Naikrovek | "howcome my fonts are unreadable" |
20:02.35 | bmoraca_work | ManxPower-work: get system administrators worth their salt, and you won't have to worry about permissions issues and people installing bullshit apps on their systems |
20:02.52 | Naikrovek | "howcome my tab key doesn't switch input fields?" |
20:02.53 | ManxPower-work | bmoraca_work: it is cheaper to retrain the existing admin. |
20:03.03 | ManxPower-work | Naikrovek: I've never had either of those issues. |
20:03.23 | Naikrovek | NEVER experienced a font issue? |
20:03.34 | Naikrovek | i'm almost convinced that you don't use a computer :) |
20:03.39 | ManxPower-work | Oh! Yes, I did have an issue with crappy fonts for about a day until I ran yum update |
20:03.42 | bmoraca_work | ManxPower-work: i'm not going to argue that linux has no uses, because I know it does. it just doesn't have very MANY uses where the majority of computer users are concerned. |
20:04.02 | carrar | haha |
20:04.07 | ManxPower-work | bmoraca_work: Of course Linux has issues. Some pretty significant ones on the desktop. So does Microsoft. |
20:04.09 | carrar | ignorance is bliss |
20:04.27 | [TK]D-Fender | bmoraca_work: As a home user who doesn't need special tax software, doesn't do PC gaming, Linux suits me jsut fine. |
20:04.32 | Naikrovek | just take a look at Germany's governmental rollout of linux for exery reason in the world and some new ones you've never heard of, of why linux on the desktop is a terrible idea |
20:04.44 | Naikrovek | and that's not anecdotal, that's emprical evidence |
20:05.15 | bmoraca_work | carrar: i'm still waiting for your linux practice management software. until you can point me to it, you're absolutely full of shit, because i have litterally thousands of customers who ABSOLUTELY MUST use this software which ONLY exists for Windows. |
20:05.42 | Naikrovek | technically, linux could be awesome |
20:05.46 | *** join/#asterisk ELSEGO (~Juankar@unaffiliated/elsego) |
20:05.51 | Naikrovek | i used to run several linux desktops |
20:05.51 | ecrane | Naikrovek: That's not fair. I know nothing about the German government, but deciding something is bad because a government tried to use it and failed is not a fair comparison.. governments fail at a lot of things they try, sometimes even good things... |
20:05.54 | Naikrovek | but i got tired of compiling |
20:06.06 | carrar | What is practice management software |
20:06.07 | bmoraca_work | anyway, i'm going to lunch. ideological arguments set my teeth on edge. |
20:06.08 | Naikrovek | ecrane: they didn't fail, they're still rolling out |
20:06.16 | bmoraca_work | carrar: now who's ignorant? |
20:06.18 | Naikrovek | ecrane: just read up on all the trouble they're having with user training |
20:06.21 | carrar | please explain |
20:06.33 | carrar | I'm not familure with "practice management software" |
20:06.36 | Naikrovek | ecrane: and system deployment issues. |
20:06.44 | Naikrovek | carrar: medical practice management software. |
20:06.48 | bmoraca_work | carrar: you have no idea about what you are talking, so you cannot begin to claim that "linux works for everyone". |
20:07.11 | carrar | You're right, I've wasted the last 20 years of my life making money with Linux |
20:07.15 | ManxPower-work | carrar: claimed that :linux works for everyone: ? |
20:07.22 | carrar | stupid me |
20:07.32 | carrar | hahah |
20:07.35 | carrar | You kids are funny |
20:07.49 | ManxPower-work | carrar: about as stupid as saying the same thing about Microsoft. |
20:07.50 | Naikrovek | it's always the old guys that are anti microsoft i'm noticing |
20:08.15 | carrar | with age comes wisdom my child |
20:08.21 | benngard | is pretty old |
20:08.29 | Naikrovek | well youth has the ability to see one's errors |
20:08.33 | Naikrovek | old = stodgy |
20:08.57 | Naikrovek | and it's provably easier to learn new stuff as a youth |
20:08.58 | carrar | and recklessness comes with youth |
20:09.11 | Naikrovek | wisdom comes with youth. it's just ignored until you can't rely on anything else |
20:09.17 | carrar | hahah |
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20:10.49 | benngard | but, can people stop sending fax through MY *, i wanna restart it :( |
20:10.54 | carrar | but alas I digress, I'm not a hater, I just have some time to kill because all my Linux boxes run themselves |
20:11.05 | Naikrovek | well that's fine |
20:11.13 | Naikrovek | my windows boxes run themselves, too |
20:11.27 | Naikrovek | it was a pointless discussion anyway |
20:11.27 | carrar | SECURITY UPDATE TODAY |
20:11.38 | Naikrovek | not a single one of us had our opinion changed |
20:11.40 | spenguin[w0rk] | ? |
20:11.47 | carrar | must reboot all windows boxes |
20:11.53 | carrar | heh |
20:11.57 | ecrane | Naikrovek: I searched google and bing but I am having trouble finding info about the problems the german government has. Do you have any links? |
20:12.03 | Naikrovek | carrar: tuesdays. and they reboot themselves. and we don't have to recompile a damned kernel to implement the fix |
20:12.08 | Naikrovek | nor any kernel modules |
20:12.27 | carrar | cause you couldn't if you wanted too |
20:12.40 | Naikrovek | carrar: don't *ever* need to |
20:12.43 | spenguin[w0rk] | recompiling kernel is fun and 3l33t |
20:12.47 | spenguin[w0rk] | unlink |
20:12.54 | spenguin[w0rk] | unlike*, click click click |
20:12.56 | spenguin[w0rk] | :p |
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20:13.18 | Naikrovek | ecrane: nothing that isn't in German and provided by a german citizen friend. let me dig something up though |
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20:14.24 | ecrane | Naikrovek: Thanks! I have no doubt you are correct; IT projects rarely go smoothly no matter what OS is involved.. I'm just interested in the details. |
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20:19.26 | Naikrovek | you linux guys should read the Linux Hater's blog for ideas about things to fix in Linux. he makes some points that no educated linux user can honestly deny |
20:20.04 | ManxPower-work | Naikrovek: And us linux users don't have to reboot every tuesday. I finally stopped automatically installing updates when I finally got tired that. |
20:20.17 | Naikrovek | ManxPower-work: windows users don't HAVE to either |
20:20.26 | Naikrovek | most of the fixes are ignorable |
20:20.49 | Naikrovek | and windows sites of any size have a local windows update services server so we can control the updates as we please |
20:20.52 | Kobaz | how hard is it to replace a piston in a 2 cycle short block engine |
20:21.03 | ManxPower-work | I have it notify me when new updates are available. |
20:21.33 | Naikrovek | Kobaz: that's a random question. i did it when i was a kid, don't remember it being too hard |
20:22.14 | Kobaz | Naikrovek: k |
20:22.31 | Kobaz | my snow blower sucked in a screw from the throttle plate in the carbutator\ |
20:22.38 | Naikrovek | ooh |
20:22.48 | Kobaz | i just finished rebuilding the carb and it was running great for about 5 minutes |
20:22.53 | Naikrovek | whatever it was I worked on had removable sleeves |
20:22.55 | Kobaz | the cylinder looks okay |
20:23.04 | Kobaz | the piston has several nicks in it |
20:23.19 | Naikrovek | so i pulled the head off, pulled the cylinder off the piston and replaced ... something. it's been a long time |
20:23.48 | Kobaz | it's like 160 for a new engine.... 30 bucks for a new piston |
20:24.08 | Kobaz | although 160 is pretty cheap compared to the 300 i spent on a new table saw motor |
20:24.33 | Kobaz | compression is at 60psi |
20:25.44 | Kobaz | i wonder if a new piston will put it back at 90 |
20:26.32 | Naikrovek | other than dents in the surface is the piston okay? |
20:26.43 | Kobaz | it seems to be |
20:26.54 | Naikrovek | i would imagine there are probably some grooves in the wall, yes? |
20:27.00 | Kobaz | i didn't see any |
20:27.08 | Naikrovek | why you replacing the piston then |
20:27.30 | Kobaz | well the piston looks guaged in some spots |
20:27.45 | Naikrovek | gouged ? |
20:27.53 | Kobaz | yeap |
20:27.54 | Kobaz | that too |
20:27.55 | Naikrovek | i suppose it might burn through any thin spots |
20:27.59 | Naikrovek | that would be bad |
20:28.09 | Naikrovek | piston only or connecting rod too |
20:28.17 | Kobaz | haven't looked at the rod |
20:28.27 | Naikrovek | well if you replace only the piston you gotta remove it from the rod |
20:28.30 | Kobaz | i think the screw bounced around in the cylinder and got thrown out the exhaust |
20:28.31 | Naikrovek | those bearings are press-fit |
20:28.38 | Naikrovek | ooh |
20:28.44 | Naikrovek | you may have a bent valve |
20:28.53 | Naikrovek | but if you can hold 60psi probably not |
20:29.31 | ecrane | This linux hater blog guy has a lot of hate in him. |
20:29.36 | Kobaz | good thing for a small intake... it tried to suck in the throttle plate too |
20:29.43 | Naikrovek | it's not difficult to take those small engines apart |
20:29.47 | Kobaz | yeah |
20:29.49 | Naikrovek | ecrane: and a lot of really good points |
20:29.52 | Kobaz | 6 screws and i had the head off |
20:30.35 | Kobaz | i'll have to do more inspection to see if there's any other damange |
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20:39.21 | Naikrovek | ecrane: the guy clearly uses linux daily, and has been holding back criticism for some time |
20:40.11 | Naikrovek | does linux have any Windows Deployement Services-like functionality? |
20:40.20 | Naikrovek | We had it on FreeBSD at Yahoo! |
20:40.39 | Naikrovek | plug a machine in, netboot, enter a command and 5 minutes later the server is ready to be deployed, all required software installed |
20:40.44 | ManxPower-work | Most distros support zeroconf as well as automated installation |
20:40.49 | Naikrovek | nice |
20:41.11 | ManxPower-work | All our installs are from a custom CentOS DVD. |
20:41.26 | ManxPower-work | (actually CD, not DVD) |
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20:41.43 | ManxPower-work | Naikrovek: Redhat based distros call it "kickstart" |
20:42.08 | Naikrovek | my windows installs are all custom, done from a distribution server. netboot, install, shutdown, carry to users desk, turn on, done. |
20:42.17 | Naikrovek | it's very handy; glad to see that it's in linux as well |
20:42.22 | Naikrovek | it was the bees knees when i used it at yahoo |
20:42.43 | Naikrovek | machine downtime due to hardware failure was less than 10 minutes even if i had to open a new server from the box it was shipped in |
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20:45.19 | ecrane | How is licensing for windows managed? Can/do all the images made from the distribution server have the same product keys? |
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20:48.08 | bmoraca_work | ecrane: depends. windows volume licensing is almost all honor-system. |
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20:53.35 | ecrane | I guess that's one argument against windows.... I'm making assumptions here, but I suspect volume licensing is out-of-reach for the average person... and I suspect that it is a hurdle in computer imaging. I remember at one job the admin had to do some kind of windows registry change on each machine afterwards, making up a unique id...maybe it's gotten better since then... |
20:54.53 | Naikrovek | ecrane: anyone can get a VLK, even if they buy a single license |
20:55.08 | Naikrovek | but yes, the images distributed are predefined with their key |
20:55.13 | Naikrovek | and preactivated |
20:55.22 | Naikrovek | ... maybe not preactivated |
20:55.23 | Naikrovek | actually |
20:55.29 | Naikrovek | been a while since i deployed a desktop |
20:55.33 | Naikrovek | oh that's right |
20:55.39 | bmoraca_work | ecrane: he was doing it wrong, then. |
20:55.40 | Naikrovek | not preactivated, but they activate on boot |
20:55.53 | Naikrovek | have a local activation "server" here |
20:55.59 | Naikrovek | what's it called ... mlk? |
20:56.03 | Naikrovek | forget |
20:56.04 | Naikrovek | dang |
20:56.30 | Naikrovek | ecrane: he was using ghost or something if he had to change crap in the registry |
20:56.32 | bmoraca_work | ecrane: a desktop volume license key can be used on as many systems as you want. however, each system will need its own system ID, etc. the easiest way to do that is to "sysprep" the image before you take it. it's not hard and it works really well. |
20:57.23 | bmoraca_work | ecrane: volume licensing is more expensive than regular licensing because the restrictions on it are far less. volume licensing really only makes sense on a large scale, as do most of these deployment and update services. |
20:57.26 | Naikrovek | oh man what is that server process called that activates your machines for you |
20:57.42 | Naikrovek | bmoraca_work: yes |
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20:57.58 | bmoraca_work | Naikrovek: that's new in 2008. i don't remember what it's called, but 2003/XP VLK did not need to be activated |
20:58.11 | Naikrovek | but we're a small office, just a few servers and maybe 25 desktops, but i have WSUS, WDS and whatever that activation thing is now |
20:58.33 | Naikrovek | i don't remember because i didn't have to install it. if you activate a single windows server with that key, it BECOMES a local activation server |
20:58.53 | Naikrovek | but from there on you don't have to know what your product keys are, they're all activated whether you're licensed or not |
20:58.59 | Naikrovek | i'm licensed, of course |
20:59.28 | hardwire | ok.. using asterisk 1.2 I have 4 DID registered with another provider and 4 SIP accounts per DID however whenever a call comes in they always appear to be the last mentioned DID as the SRC channel. |
20:59.30 | bmoraca_work | Naikrovek: yes, that's one of the ways they're doing VLK for 2008. It's KMS, I believe. you can also convert your KMS keys into MAK keys that allow you to activate to Microsoft instead of a local activation server |
20:59.38 | Naikrovek | that' |
20:59.39 | hardwire | even though the pcap says differently. |
20:59.41 | Naikrovek | that's it, KMS |
20:59.42 | hardwire | any idea how to get around that? |
21:01.56 | funtoo_nbu | anyone know why there might be delays when you transfer a call? |
21:02.50 | bmoraca_work | funtoo_nbu: almost exclusively an issue with the phone's configuration. |
21:03.27 | funtoo_nbu | the phone's config itself or its conf inside extensions.conf or sip.conf ? |
21:04.01 | bmoraca_work | the phone's configuration itself. if i'd meant extensions.conf, I'd have said "dialplan" or sip.conf would ahve been "peer" |
21:04.09 | funtoo_nbu | roger that |
21:04.12 | funtoo_nbu | any hints? |
21:04.17 | funtoo_nbu | they are polycom 450s |
21:04.33 | Naikrovek | what kind of delay are you talking about |
21:04.39 | Naikrovek | 3 seconds? |
21:04.43 | bmoraca_work | sip debugs and dialplan logs that corroborate your delay would be useful |
21:04.45 | funtoo_nbu | it seems random |
21:04.54 | funtoo_nbu | sometimes none sometimes > 20s |
21:04.57 | Naikrovek | wow |
21:05.03 | Naikrovek | not a digitmap timeout issue then |
21:05.08 | funtoo_nbu | like, caller calls in |
21:05.33 | funtoo_nbu | phone a answers, says ok ill transfer, hits transfer button (hold music plays) |
21:05.54 | funtoo_nbu | phone a is calling phone b, phone b answers, hold music stops playin for a bit then phone b answers |
21:06.08 | spenguin[w0rk] | whats in the console? |
21:06.08 | Naikrovek | where is the lag in that scenario |
21:06.47 | funtoo_nbu | not on console yet, im gona call in and try to replicate |
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21:09.06 | Naikrovek | my receptionist had a similar problem when she transferred |
21:09.30 | Naikrovek | she'd be on a call, press transfer, then dial the extension, and then the digitmap timeout would make her wait 5sec before it would dial |
21:09.52 | Naikrovek | i taught her to push # after the extension number and it dials when she pushes it now |
21:09.59 | Naikrovek | it would have before if she'd pushed it |
21:10.09 | Naikrovek | but yea that was the problem i had here |
21:10.12 | Naikrovek | was user problem |
21:10.15 | Naikrovek | not system problem |
21:11.06 | Naikrovek | if the extensions started with something other than a 1 i would be able to configure the digitmap entry to not timeout but it does, so i have to allow someone to dial 1 area code etc |
21:11.38 | funtoo_nbu | my delay is after the other transferring phone picks up |
21:11.51 | Naikrovek | really |
21:11.52 | funtoo_nbu | im trying to get them to pick up my call so i can get some log on it :D |
21:12.04 | funtoo_nbu | o shit, its lunch :/ |
21:12.14 | Naikrovek | eeeeeat |
21:12.20 | funtoo_nbu | maybe ill eat too |
21:15.40 | ManxPower-work | Naikrovek: can you have them never dial a 1 for outside line? That way first digit = 1 = extension, first digit != 1 = outside call. Seems backwards but it might work. |
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21:16.08 | Naikrovek | well my intention is to switch to 4 digit extensions that don't start with a 1 soon so the problem will go away then |
21:16.32 | funtoo_nbu | hey thats a good idea |
21:16.34 | KavanS | trying to compile 1.4.29 from source and it complains about iLBC codec....1.4.28 compiles fine |
21:17.50 | bmoraca_work | KavanS: iLBC source was removed from SVN. you need to provide it separately or deselect iLBC from menuselect |
21:19.17 | funtoo_nbu | if the patten matching is done right can you make it so if you dial 101 it knows its an extension and dial it right away (without having to click send)? |
21:19.34 | Naikrovek | yes |
21:20.08 | Naikrovek | not all my extensions start with 1 |
21:20.25 | KavanS | bmoraca_work, roger that, thanks for the tip |
21:21.11 | ManxPower-work | Can anyone think of a reason that 'asterisk -rvvv' works, but 'asterisk -rx "sip show peers' gives me: Unable to connect to remote asterisk (does /var/run/asterisk.ctl exist?) |
21:22.14 | kuku | spoke too soon: http://pastebin.com/d5006db0f |
21:23.57 | funtoo_nbu | mm my patten for outgoing is _1NXXNXXXXXX |
21:26.11 | Naikrovek | _? |
21:27.08 | funtoo_nbu | an extension name is a pattern if it starts with the underscore symbol (_). |
21:29.03 | carrar | ManxPower-work, Might i suggest you install win7 to fix that issue |
21:29.35 | [TK]D-Fender | checkout time, bbiab |
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21:34.01 | *** join/#asterisk etfonhomey (~etfonhome@74-143-192-74.static.insightbb.com) |
21:34.36 | etfonhomey | Is this the error message you get if you don't have the g729 transcoding license? [Feb 11 16:32:12] WARNING[3139]: chan_sip.c:3724 sip_write: Asked to transmit frame type 4, while native formats is 0x100 (g729)(256) read/write = 0x100 (g729)(256)/0x100 (g729)(256) |
21:34.42 | etfonhomey | Frame type 4 is g711u |
21:36.15 | *** join/#asterisk nny (~scott@64.203.239.83) |
21:36.23 | nny | hmmph |
21:36.35 | Kobaz | Naikrovek: okay so... upon further inspection. the cylinder is perfect as far as i can tell |
21:36.48 | nny | so these cisco 504g phones have the ability for programmable softkeys! Hooray.. ermm wait |
21:36.56 | Kobaz | Naikrovek: there is a 1mm gouge on the side of the piston where it hit the screw |
21:37.10 | Kobaz | Naikrovek: and i found the screw.. it was in the crank arm housing |
21:37.17 | Naikrovek | Kobaz: think it leaks compression? |
21:37.18 | Naikrovek | whoa what |
21:37.21 | Naikrovek | in the crank case? |
21:37.29 | Kobaz | it got below the piston somehow |
21:37.30 | Naikrovek | how the F did it get in there |
21:37.40 | nny | you can set the PSK for say, connected to be different (in this case, I am setting it up as a park button, *9). Sadly the stupid phone tries to dial *9 as a fresh dial, instead of overlayed onto the existing call :\ |
21:38.00 | Kobaz | Naikrovek: i think it most certainly leaks compression... it's quite a bit of a nick |
21:38.02 | nny | er rather you can set the PSKs based on different call states |
21:38.46 | nny | gonna ping the cisco engineers (you'd think it's be trivial) wonder if I can fashion a workaround through * though |
21:38.51 | Kobaz | that's the only thing i can find that would affect the compression... i think replacing the piston would do it...but there's a very very strange bolt holding the arm to the drive shaft.. it's like inverse star-drive |
21:38.52 | Naikrovek | well disconnect the connecting rod from the crankshaft, and take the connecting rod and piston to a local small engine place. they'll pull off the piston and order you a new one |
21:38.52 | nny | it'd be* |
21:40.14 | Kobaz | i think i found the piston i need online for like 30 bucks |
21:40.24 | nny | hmm a 1mm gouge shouldn't leak compression though, the oil should fill that gap no? |
21:40.32 | Kobaz | the problem is getting the piston out.. i'll take a picture of the bolt |
21:40.45 | nny | guess the only way to know is to put the head on and test the compression |
21:40.50 | Kobaz | it's at 60psi |
21:40.54 | nny | ahh |
21:41.00 | nny | yeah, that's low lol |
21:41.04 | Kobaz | a bit |
21:41.08 | nny | :D |
21:41.33 | Kobaz | maybe it's bigger than 1mm. it's quite a gouge, it's not big, but it's not small |
21:41.51 | nny | yeah I imagine so, someone drop a bit of some love down the intake |
21:41.52 | nny | ? |
21:41.59 | Kobaz | that didn't work |
21:42.20 | nny | er rather, my question was did someone drop something in the intake to cause the damage? |
21:42.26 | Kobaz | oh |
21:42.27 | Kobaz | yeah |
21:42.38 | Kobaz | the screw from the throttle plate got sucked in while running |
21:42.57 | Kobaz | and the throttle plate has a big chunk missing from it |
21:43.01 | nny | been there before, but it was the wingnut for the aftermarket air filter on my merc |
21:43.22 | Kobaz | a friend of mine got a brand new nissan sentra, and it sucked in a screw |
21:43.36 | nny | ouch |
21:43.39 | Kobaz | completely wrecked the engine.. so they warrentyd it |
21:43.45 | nny | nice |
21:44.06 | Kobaz | the guy at the shop was like... did you change the oil? and my friend said uhh... it's got 1500 miles on it.. no... that's not why the engine died |
21:44.30 | *** part/#asterisk Tobarja (~chatzilla@user-0c8h5rb.cable.mindspring.com) |
21:44.50 | Kobaz | anyways... so i mean like... with a smoother than a baby cylinder, and nicks in the piston |
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21:45.22 | momelod | greetings channel |
21:45.24 | Kobaz | well, the missing chunk of the throttle plate would probably affect compression too |
21:45.29 | nny | rephrasing my issue/ quandry here. These phones have the ability to dial *something* during a call. I tried some feature codes, but the dumb things dial the extension as a fresh line, or rather don't seem to try and treat it like you would a DTMF style entry. Any clever advice? |
21:45.38 | Kobaz | that's like a 3mm missing chunk |
21:45.49 | nny | the phones have the ability to speed dial something during a call* |
21:45.54 | nny | if I use the handset it works fine |
21:45.56 | momelod | I don't know a better place to ask so sorry if this is the wrong place. What is the best sip conference phone for use w/ asterisk? |
21:46.01 | nny | gah handset||keypad |
21:46.14 | nny | momelod had a lot of luck with the polycomms |
21:46.30 | nny | ip9000 or something rather one sec |
21:46.35 | momelod | nny any particular model u liked? |
21:46.36 | Kobaz | momelod: sip phone? huh? this is #engine_repair |
21:46.40 | nny | lol |
21:46.48 | momelod | :P |
21:47.16 | nny | momelod: http://www.polycom.com/products/voice/conferencing_solutions/conference_phones/soundstation/soundstation_ip7000.html |
21:47.36 | Naikrovek | yeah ip7000 is good |
21:47.46 | nny | momelod: seen that phone rebranded and sold by other vendors (mitel for example) |
21:48.31 | Naikrovek | cisco rebrands a couple of polycom phones |
21:48.38 | momelod | awesome thanks |
21:48.41 | Naikrovek | and yes i confirmed that they are rebranded |
21:49.12 | nny | yeah heh |
21:53.47 | momelod | ouch, not cheap |
21:56.39 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
21:57.13 | bmoraca_work | IP6000 is good, too...no real reason to go to IP7000 if you don't need all the extra mics and stuff |
21:57.20 | *** join/#asterisk Failrar (~Failrar@2001:470:1f15:316:2a0:d1ff:fe4e:e802) |
21:58.06 | *** join/#asterisk Wildy (~simba@89.222.134.42) |
22:00.04 | *** join/#asterisk gelpg (~chatzilla@qf.invitel.hu) |
22:05.03 | p3nguin | naikrovek: Which models? I know they used to rebrand some devices, but it was not something hidden or even attempted to be kept secret. |
22:07.39 | *** join/#asterisk mediaprodigy (~chatzilla@72.20.157.179) |
22:07.43 | mediaprodigy | hello |
22:07.56 | *** join/#asterisk TSM2 (~the_softw@87-194-32-212.bethere.co.uk) |
22:08.44 | bmoraca_work | p3nguin: Cisco 79x0s and pretty much everything older than that were rebranded from another company (don't remember the name right now) |
22:09.45 | jaskew | Can anyone recommend a 'favorite' appliance-style box for a small (say 5 SIP phones and two analog trunks) install. I've heard of PIKA Warp. Any thoughts? |
22:09.50 | mediaprodigy | Question: Is there an open source project already available that is similar to OCS cisco cups. |
22:09.59 | mediaprodigy | Question that has these features |
22:10.03 | mediaprodigy | ⢠Complete desktop call control |
22:10.05 | mediaprodigy | ⢠Incoming and outbound interactive call windows with name and number |
22:10.06 | mediaprodigy | ⢠Multiple call notification |
22:10.08 | mediaprodigy | ⢠Database directory for system and personal numbers |
22:10.10 | mediaprodigy | ⢠Dynamic redial list with most frequently called or received numbers |
22:10.11 | mediaprodigy | ⢠Speed dial for one click dialing |
22:10.13 | mediaprodigy | ⢠Busy lamp field indicators for internal users |
22:10.14 | mediaprodigy | ⢠Peek-a-Boo |
22:10.16 | mediaprodigy | ⢠Silent Message |
22:10.17 | mediaprodigy | ⢠Absence Message |
22:10.19 | mediaprodigy | ⢠Call Reminders |
22:10.20 | mediaprodigy | ⢠Email Page |
22:10.22 | mediaprodigy | ⢠Call Log with dial capability |
22:10.23 | mediaprodigy | ⢠PIM integration to popular databases with optional Power Dialer module |
22:10.25 | mediaprodigy | sorry |
22:10.31 | mediaprodigy | thought i had put in comma's |
22:10.43 | ManxPower-work | You must now do 300 hours of community service. |
22:10.54 | nny | wouldn't fop2 handle some of that? |
22:10.57 | jaskew | or Peek-a-Boo with inner city youth |
22:11.05 | etfonhomey | jaskew, I used this with a Sangoma B600DE: http://www.newegg.com/Product/Product.aspx?Item=N82E16816101262&cm_re=SuperMicro_atom-_-16-101-262-_-Product |
22:11.20 | mediaprodigy | i do not know what peeka-boo is |
22:11.23 | mediaprodigy | but you get the idea.. |
22:11.33 | mediaprodigy | is there a project out there that is already doing this. |
22:12.10 | bmoraca_work | mediaprodigy: i don't know of an opensource one, but iSymphony is very good for providing those types of features, and if you have 5 or fewer users, it's free. |
22:12.12 | carrar | mediaprodigy, are you on a windows box? |
22:12.28 | bmoraca_work | mediaprodigy: additionally, iSymphony does not require windows. |
22:12.44 | *** join/#asterisk andres833 (~andres833@190.144.75.22) |
22:13.16 | jaskew | etphonehomey: Thanks. any others I can/should look at. I assume Flash based stuff is out if it has Voicemail. |
22:13.37 | etfonhomey | ManxPower-work, Is this the message you get if you don't have the g729 license? [Feb 11 16:32:12] WARNING[3139]: chan_sip.c:3724 sip_write: Asked to transmit frame type 4, while native formats is 0x100 (g729)(256) read/write = 0x100 (g729)(256)/0x100 (g729)(256) |
22:13.58 | [TK]D-Fender | mediaprodigy: http://pastebin.com/m5e107a20 |
22:14.40 | carrar | heh |
22:14.51 | etfonhomey | jaskew, that's the only small install one I've tried other than the circa 2002 ThinkPad I run my person Asterisk install on. Builtin battery backup! If the battery is still good... |
22:18.14 | bmoraca_work | jaskew: http://www.newegg.com/Product/Product.aspx?Item=N82E16816101262&cm_re=intel_atom-_-16-101-262-_-Product |
22:18.24 | *** join/#asterisk mediaprodigy (~chatzilla@72.20.157.179) |
22:18.38 | mediaprodigy | Firebox crashed |
22:18.49 | mediaprodigy | so someone mentioned iSymphony |
22:18.49 | carrar | that red thing? |
22:19.02 | bmoraca_work | mediaprodigy: i don't know of an opensource one, but iSymphony is very good for providing those types of features, and if you have 5 or fewer users, it's free. |
22:19.13 | mediaprodigy | the reason why i ask |
22:19.37 | mediaprodigy | well first off would this be something that the telephony asterisk group would even want |
22:19.42 | mediaprodigy | or find interst in |
22:19.51 | KavanS | hrm, can asterisk 1.4.29 use zaptel? |
22:19.56 | KavanS | looks like it's only dahdi... |
22:20.23 | bmoraca_work | mediaprodigy: i'm not sure what you're asking. if you have a specific question, let's hear it. |
22:20.27 | mediaprodigy | i am trying to convince someone to release their company project to open source.. |
22:20.28 | [TK]D-Fender | mediaprodigy: "the telephony asterisk group" <- Pardon? Who/what is this? |
22:20.31 | bmoraca_work | KavanS: i believe that's correct, yes. |
22:20.37 | carrar | TAG |
22:20.39 | carrar | your it |
22:20.47 | mediaprodigy | the people in this channel |
22:20.48 | KavanS | ok roger that, thank you |
22:21.07 | carrar | [TK]D-Fender, you don't have the jacket? |
22:21.09 | [TK]D-Fender | mediaprodigy: You mean would WE want a 3rd party program that offers this functionality to * released as OSS? |
22:21.13 | bmoraca_work | mediaprodigy: i doubt that someone is going to take a useful, profitable app and give it away for free just because you asked |
22:21.29 | carrar | I've got the "Telephony Asterisk Group" patch |
22:21.55 | [TK]D-Fender | carrar: I tried that patch... but the cravings returned |
22:21.59 | carrar | haha |
22:22.47 | mediaprodigy | bmoraca_work: there are several major companies releasing software that will over take this software.. instead of letting it die out over the next few years.. what if that project was taken the open source route |
22:23.31 | mediaprodigy | I understand your perspective.. i am just here to get a feel for the waters. |
22:23.34 | jaskew | bmoraca_work & etfonhomey: thank yo u - two votes for that one - must be the ticket! |
22:24.07 | carrar | mediaprodigy, when do you start working on it/ |
22:24.08 | carrar | ? |
22:24.13 | *** join/#asterisk brettnem (~brett@user-0vvd88f.cable.mindspring.com) |
22:24.16 | bmoraca_work | mediaprodigy: it's not up to us whether a developer takes their product opensource or closed source...it's up to the developer. what we want is completely and totally irrelevant |
22:24.22 | brettnem | hello all |
22:24.45 | mediaprodigy | bmoraca_work: i do not know you but you are obviously missing my point |
22:24.50 | *** join/#asterisk fofware (~chatzilla@190.7.25.160) |
22:24.57 | nny | personally I am still trying to find *soemthing* to offer my clients that compares to HUD or similar. I use FOP2 right now + openfire and spark (rebranded yay!) however I have found some aspects of the system (spark asterisk plugin) aren't maintained. Short answer: yes it would be good to find a solution, i am going to ttry isymphony again |
22:25.01 | bmoraca_work | mediaprodigy: if you want a developer to open the source of a successful product, you'd better be prepared to pay for it |
22:25.01 | brettnem | hey, I have a fresh (today) 1.4 svn. Polycom attended transfers fail.. if I roll back to the latest 1.4 stable, they work.. anyone seen this? |
22:25.55 | bmoraca_work | mediaprodigy: perhaps. then again, perhaps i'm not. you're not clearly expressing what it is you're looking for. what it sounds like is you want an application to be released for free. i'm simply providing you with the pragmatic viewpoint that it's not going to happen. |
22:25.55 | brettnem | on the attended transfer, the attended part of it works fine. .however when you hit transfer the second time to bridge the parties together, the call drops |
22:26.17 | mediaprodigy | carrar: i have just brought it up a few times.. i think there is potential there.. |
22:26.31 | mediaprodigy | bmoraca_work: pessimism will get you nowhere. |
22:26.46 | *** join/#asterisk rossand (~aross@dhcp-233-179.tb-classrooms.carleton.ca) |
22:26.46 | bmoraca_work | mediaprodigy: pragmatism != pessimism. |
22:27.02 | etfonhomey | jaskew, my only beefs with that device are that you have to buy the PCI express riser board separate. |
22:27.03 | mediaprodigy | bmoraca_work: is there a need for call control like microsoft ocs |
22:27.09 | nny | so back to my quandry earlier, anyone use the SPA504g cisco phones? They offer PSk for in call use, but the dumb thing tries to start a new channel to perform the dial, I am just looking to add feature codes from features.conf to the phone interface |
22:27.15 | mediaprodigy | or ciscos version with Asterisk |
22:27.33 | *** join/#asterisk cvnet (~cvnet@dsl-69-172-67-161.acanac.net) |
22:27.36 | cvnet | hi all |
22:27.42 | etfonhomey | jaskew, and if you want RAID 1 or 0 with 2 x 2.5" drives, you have to buy the drive cage separately as well. |
22:27.44 | bmoraca_work | mediaprodigy: of course there is. and there exist a number of applications which provide it. they are, as I understand it, fairly successful. |
22:27.55 | carrar | cringes at the s ound of smeone calling a SPA phone a cisco :( |
22:28.02 | mediaprodigy | where you have software control, with instant messaging and potentially webcam support, active directory |
22:28.06 | carrar | I call it like it is, LinkSys |
22:28.16 | nny | carrar: hmm? |
22:28.28 | nny | carrar: they are cisco small business now, as branded |
22:28.47 | carrar | I know |
22:28.57 | nny | er rather small business pro* |
22:28.59 | ManxPower-work | They are still Linksys design, not Cisco design |
22:29.04 | nny | they work better with asterisk than the cisco phones |
22:29.12 | mediaprodigy | bmoraca_work: if there is already something out there then great.. I am just asking |
22:29.28 | cvnet | I answer a call and then dial plan looks like this exten => _X.,n,VoiceMailMain(6000,u) <== instead of saying please leave a voice message, it asks for password, what am doing wrong here? |
22:29.39 | nny | I dunno, cisco, the company and cisco's engineers are the people I deal with when I have an issue, so i call them cisco |
22:29.40 | bmoraca_work | mediaprodigy: iSymphony. to a lesser extent, hudlite. to an even lesser extent, flash operator panel. |
22:29.56 | nny | the cisco support office is in the same state as me |
22:30.00 | ManxPower-work | cvnet: do a "core show VoiceMail" and "core show VoiceMailMain" and notice the diffences |
22:30.05 | nny | and the support techs aren't outsourced |
22:30.27 | mediaprodigy | bmoraca_work: thanks for your help |
22:30.27 | cvnet | thanks |
22:30.42 | p3nguin | etfonhomey: As far as I know, lacking the license will not produce any failure whatsoever. |
22:30.44 | jaskew | etfonhomey: good - a few options. It's a charity project for a non-profit, so I get to do as musch as I can for free :) |
22:31.18 | nny | matter of fact the guy I am dealing with on a now fixed sidetone issue is listed as Cisco SBSC Network Engineer. |
22:31.38 | etfonhomey | p3nguin, other than no transcoding |
22:31.50 | p3nguin | etfonhomey: No. No failure of any sort. |
22:32.11 | p3nguin | etfonhomey: The license is a LEGAL issue, not a functional issue ... to my knowledge. |
22:32.12 | etfonhomey | p3nguin, before I just added the license it would not transcode from g711 to g729 |
22:32.32 | mediaprodigy | bmoraca_work: it is a shame.. but thanks for the information |
22:33.06 | nny | cvnet: Voicemail vs Voicemailmain |
22:33.06 | p3nguin | etfonhomey: I have known some people to not have license but also do have the codec and it does transcode both to and from g.729. |
22:34.12 | p3nguin | etfonhomey: If the lack of a license prevented the codec from working, their systems would not have been transcoding. |
22:34.12 | etfonhomey | p3nguin, Actually, I just didn't have the codec. Had to load it. |
22:34.12 | bmoraca_work | mediaprodigy: i'm not sure why it's a shame. i understand the want for an opensource software of this type, but as long as there is demand willing to pay for it, it's unlikely that existing products will be released as open source. if you want to start an opensource version, you're more than welcome, and i would embrace it...but it's a big, big undertaking |
22:34.13 | p3nguin | etfonhomey: precisely my point. |
22:34.14 | nny | honestly I'd rather pay for it |
22:34.26 | nny | in regards to bmoraca_work and mediaprodigy discussion |
22:34.30 | etfonhomey | p3nguin, but, I'm legal now anyway. It's only $8 and only needed one channel |
22:34.42 | p3nguin | etfonhomey: Sounds good to me. |
22:34.48 | nny | all the other attempts that are free always seem to get abandoned etc |
22:35.14 | nny | i'd pay 200 bucks or so on top just for a full featured non trixbox/freebx/fonality call center heads up solution |
22:35.46 | bmoraca_work | nny: i built one myself a while back...i could drag&drop transfer calls and had BLF capabilities as well as see who they were talking to...but i felt like I was reinventing the wheel and just ended up using iSymphony |
22:36.11 | etfonhomey | p3nguin, I only noticed the issue because my ITSP negotiated g711 with * while my phones negotiated g729 with *. So, when * bridged the calls, it couldn't transcode. I had g729 first in both lists, but my ITSP must have left out g729 on its side that time. |
22:36.39 | nny | bmoraca_work: yeah gonna look at that again, fop2 is pretty damn sweet and the guy only wants 50 bucks for a 15+ extension license, just missing anything other than the panel aspects. |
22:36.39 | bmoraca_work | nny: iSymphony is ~$650 for the server and ~$40 for each client...but it's well worth it if you need that functionality |
22:36.53 | p3nguin | etfonhomey: If you have g.729 on all your devices and g.729 on your "trunk," I don't think you even need to have the codec at all. |
22:37.09 | mediaprodigy | bmoraca_work: I worked on this software, microsoft and cisco have released OCS and whatever cisco's version is called.. to me i think that the software this company makes will not live many years longer because of the budgets allocated by these bigger entities.. i had already approached and asked once, shut down.. the second approach seemed to generate some interest.. but it is a big... |
22:37.11 | mediaprodigy | ...undertaking.. my idea is that if we release the code, who knows what developers might create.. the software already works.. |
22:37.15 | nny | bmoraca_work: price all depends on funcitonality. i would also like to rebrand it if possible |
22:37.42 | etfonhomey | p3nguin, exactly, but my ITSP did not negotiate g729 for some reason like it has been doing for weeks. |
22:37.46 | bmoraca_work | nny: the guys at i9 Technology will allow you to rebrand it. for instance, Intuitive Voice rebrands it as "iView" |
22:37.55 | nny | bmoraca_work: that's good |
22:38.14 | *** join/#asterisk nightrid3r (kvirc@41.214.236.77) |
22:38.25 | carrar | but then Apple will sue you |
22:38.28 | mediaprodigy | bmoraca_work: well i did not mean to get all into a discussion.. just had a question and it has been answered. |
22:38.30 | mediaprodigy | thank you to all |
22:38.35 | p3nguin | etfonhomey: disallow=all allow=g729 |
22:38.40 | p3nguin | etfonhomey: Then it wouldn't have a choice. |
22:38.44 | *** join/#asterisk nightrid3r (kvirc@41.214.236.77) |
22:38.46 | etfonhomey | p3nguin, I had never had to deal with the codec. So, it was good to see the process through once in case I had a client who needed it. |
22:38.59 | nny | bmoraca_work: pricing is a little wonky, per queue, per client, etc. Feel like it's nortel lol |
22:39.02 | etfonhomey | p3nguin, it wouldn't have choice but to drop the call if it for whatever reason couldn't negotiate g729. |
22:39.14 | nny | wish they would stop trying to dip into larger profits and just offer flat rates |
22:39.17 | p3nguin | etfonhomey: Yeah, I guess that's true. |
22:39.24 | etfonhomey | p3nguin, which is what is why I left g711u as the 2nd option after disallow=all, |
22:39.31 | nny | not like it requires more effort on their end to manage larger setups |
22:39.47 | p3nguin | etfonhomey: Sounds reasonable. |
22:39.48 | nny | unless they are gonna install it for me, train the client, and handle support lol |
22:40.40 | etfonhomey | p3nguin, of course, it must have been temporary because after I bought the codec and installed it, I did a sip show channels on a test call and, of course, both legs were g729... |
22:41.27 | p3nguin | etfonhomey: For the $8, it's worth it to know how to configure it. |
22:41.45 | p3nguin | etfonhomey: Even if you never use it. |
22:42.28 | *** part/#asterisk mediaprodigy (~chatzilla@72.20.157.179) |
22:44.55 | etfonhomey | p3nguin, this is true |
22:47.44 | bmoraca_work | nny: meh. i've found that, honestly, there's very little reason to provide that to anyone but receptionists and managers. it's easy enough to make a queue status application using AMI. |
22:49.08 | nny | bmoraca_work: true although we pride our setups on not having per user licensing |
22:49.15 | nny | er rather take pride |
22:49.19 | nny | man i suck at typing what I am thinking |
22:49.55 | bmoraca_work | nny: we do as well...but the fact remains that there's always going to be a "per-seat" examination of total cost. and I treat this type of software as "extra" |
22:52.38 | nny | bmoraca_work: indeed, i am going to evaluate it |
22:53.52 | [TK]D-Fender | nny: I've been evaluating WinZip for about a decade now apparently ;) |
22:56.07 | carrar | hahah [TK]D-Fender, how do you like it? |
22:56.30 | carrar | Not sure if it's really what you're looking for? |
22:56.32 | Katty | drags in |
22:56.54 | Katty | plops down |
22:56.54 | carrar | harro katty |
22:56.59 | Katty | stares blankly |
22:57.15 | Katty | harro carrar |
22:57.36 | nny | [TK]D-Fender: lol |
22:59.49 | Katty | ugah, what a day |
23:02.55 | Katty | so quiet. |
23:03.04 | carrar | Katty, tell us about your day! |
23:03.13 | Katty | it was boring. that's all there is to it. |
23:03.13 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
23:03.19 | Katty | how was your day? |
23:03.42 | carrar | So far not bad, wife made chocolate truffle cupcakes |
23:03.50 | Katty | yum. |
23:04.01 | carrar | and said she made them because I use UNIX and not windows |
23:04.06 | Katty | :> |
23:04.09 | Katty | <3 |
23:04.41 | carrar | About to go for a walk and attempt to burn that off |
23:04.47 | KavanS | ok, so if I'm switching from zaptel to dahdi, what would I have to replace in my extensions.conf ? I'm googling around but I'm finding mixed results on what to do for an upgrade |
23:04.50 | Katty | k |
23:05.07 | [TK]D-Fender | KavanS: s/zap/dahdi/ |
23:05.20 | KavanS | k, roger that |
23:10.21 | Katty | omnomnomnoms strawberry and banana slices |
23:10.29 | raden_work | :) |
23:10.44 | Katty | hugs raden |
23:11.10 | Katty | raden_work: haven't noticed any changes yet, but i've only been taking the omega 3 for about a week now...maybe less |
23:11.37 | raden_work | hugs Katty |
23:12.14 | raden_work | Katty, you will notice changes over a longer time period things will balance out |
23:13.36 | Katty | raden_work: lucky for me strawberries produce an immediate change |
23:13.41 | Katty | raden_work: smile inducing! |
23:14.16 | *** join/#asterisk beta2k (1000@d24-36-68-97.home1.cgocable.net) |
23:14.30 | beta2k | Anyone know if asterisk can handle distinctive ring on a FXO port? |
23:14.42 | beta2k | Or will all rings be answered the same? |
23:15.34 | nny | beta2k: try http://lists.digium.com/pipermail/asterisk-dev/2003-November/002196.html |
23:16.02 | nny | or http://www.voip-info.org/wiki/view/Asterisk+ZAP+channels#DetectingDistinctiveRingonIncomingCalls |
23:17.09 | raden_work | Katty, :) lol |
23:20.02 | pfn | hmm, what's wrong with teliax |
23:20.04 | pfn | did they shut down |
23:23.52 | raden_work | pfn, why u say that ? |
23:24.10 | pfn | their website is busted, err500 and no response from their proxies |
23:24.17 | pfn | and damnit, my voipjet account is busted, too |
23:25.15 | raden_work | no problem here |
23:25.18 | raden_work | check your provider |
23:25.32 | raden_work | jon@Server200:~> ping teliax.com |
23:25.32 | raden_work | PING teliax.com (63.211.239.26) 56(84) bytes of data. |
23:25.32 | raden_work | 64 bytes from www.teliax.com (63.211.239.26): icmp_seq=1 ttl=48 time=57.2 ms |
23:25.32 | raden_work | 64 bytes from www.teliax.com (63.211.239.26): icmp_seq=2 ttl=48 time=55.1 ms |
23:25.40 | pfn | it was an error 500 page |
23:25.45 | pfn | but that appears to have been resolved |
23:25.48 | *** join/#asterisk puzzled (~foobar@puzzled.xs4all.nl) |
23:25.50 | pfn | now waiting for their proxy to come back |
23:25.50 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
23:25.59 | *** join/#asterisk styelz (~yoohoo@m0o0.mooo.com) |
23:26.00 | *** join/#asterisk mnt_real (~sinan@bas1-montreal43-2925257034.dsl.bell.ca) |
23:26.00 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
23:26.00 | *** join/#asterisk ttwhy (~tekkno@p4FECFC04.dip.t-dialin.net) |
23:26.00 | *** join/#asterisk e4 (~e4@rrcs-76-79-59-194.west.biz.rr.com) |
23:26.06 | pfn | [Feb 11 15:21:36] NOTICE[486]: chan_sip.c:13048 handle_response_peerpoke: Peer 'teliax' is now Reachable. (916ms / 2000ms) |
23:26.08 | pfn | ah, finally |
23:26.23 | *** join/#asterisk slima (slima@unaffiliated/slima) |
23:26.30 | raden_work | there website tells you nothing about your registration |
23:26.40 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
23:26.40 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
23:26.42 | pfn | raden_work, both were down |
23:26.44 | raden_work | I would ponder to guess it was your ISP |
23:26.50 | raden_work | pfn, if u say so |
23:27.01 | pfn | no, it wasn't |
23:27.08 | pfn | http://status.teliax.com/ |
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23:27.36 | raden_work | wow get a real provider |
23:28.13 | pfn | yeah, I should switch away to someone else instead of teliax |
23:28.15 | pfn | shrugs |
23:28.42 | raden_work | Vitelity |
23:28.51 | raden_work | 185 days no connection interuption |
23:30.17 | pfn | I tried signing up for vitelity a while back, signing up with a credit card was a pita |
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23:32.22 | raden_work | pfn, cause of the verification ? |
23:32.51 | pfn | yeah, the verification was funky or something other, I just remember it being a pain and didn't bother following through |
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23:36.53 | raden_work | pfn, yeah its a pain was for me as well |
23:37.11 | pfn | but it's a good point, I should add another provider to my list |
23:37.31 | pfn | I have voipjet also, but I don't really use them much and it kinda got unconfigured |
23:38.32 | raden_work | pfn well you ever need serive I have $3/mo wi 1.55 cents per min |
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23:45.32 | raden_work | dadhi Install >>> You do not appear to have the sources for the 2.6.31.8-0.1-default kernel installed. |
23:45.33 | raden_work | make[1]: *** [modules] Error 1 |
23:45.41 | raden_work | installed kernel-source |
23:45.42 | raden_work | same thing |
23:45.54 | spenguin[w0rk] | is it the same kernel source? |
23:46.04 | *** join/#asterisk tzafrir (~tzafrir@212.179.75.202) |
23:46.18 | spenguin[w0rk] | same as 2.6.31.8-0.1 |
23:46.21 | [TK]D-Fender | because you need the matching HEADERS |
23:46.30 | raden_work | i will double check |
23:47.44 | raden_work | .12 instead of 8 thanks :) |