00:01.02 | aldoenviro | I guess this is a good first question... will asterisk work with a normal voice modem? |
00:03.57 | mvanbaak | normally, no |
00:04.25 | mvanbaak | if it's a X100P clone model, it will work |
00:04.37 | mvanbaak | but my experience is, most voice modems wont work |
00:06.12 | aldoenviro | well that defeats the whole purpose. I wanted to test it out using incoming calls on a voicemodem. Run the calls through a menu system, voicemail etc... |
00:08.19 | *** part/#asterisk mog (n=mog@c-71-207-200-130.hsd1.al.comcast.net) |
00:09.03 | *** join/#asterisk mroberto (n=dads@S0106001346face5f.ed.shawcable.net) |
00:09.37 | mroberto | I need some help with ceptral i have the module install just not sure how to use it ? I am using version for 2.4 |
00:10.25 | Voice2 | Uniqueid has a dot.. think i can force it not with a dot ? |
00:10.45 | Voice2 | that UNIXTIMESTAMP + random 0000-9999 ? |
00:12.19 | aldoenviro | anyone know any products that would allow incoming calls on a voice modem to be answered by an auto attendant with voicemail? |
00:15.06 | aldoenviro | So an X100P FXO PCI Card |
00:15.18 | aldoenviro | That would do the trick for asterisk? |
00:16.35 | mvanbaak | if you are talking analog lines, yeah |
00:17.06 | *** join/#asterisk anthonyl (n=fbffff@r74-193-79-81.pfvlcmta01.grtntx.tl.dh.suddenlink.net) |
00:17.09 | JT | if you didn't want a very good card, then maybe |
00:17.48 | aldoenviro | What is a decent "hobyist" card in that case? |
00:18.31 | JT | there's no such thing as a card that is both cheap and decent |
00:18.38 | JT | an ATA might be a better plan |
00:20.22 | aldoenviro | Cheap isn't necessarily a requirement... just a desire |
00:21.09 | aldoenviro | ATA? |
00:21.13 | Voice2 | ok wahts the alternatve to uniqueid ? i dont want a dot |
00:21.13 | Qwell | ~ata |
00:21.15 | jbot | ata is probably Analog Telephone Adapter which is used to put a normal analog phone onto ethernet, see http://www.voip-info.org/tiki-index.php?page=Analog%20Telephone%20Adapters for more info |
00:22.22 | `p4r14h | aldoenviro: get a tdm400p before you start buying x100p's |
00:23.30 | `p4r14h | that is if you plan on using it for more than a hobby |
00:24.15 | JT | i prefer to do my "hobby" stuff properly too :) |
00:24.36 | aldoenviro | in all actuality, I would like it to answer my phone... I might eventually like to have a wifi cordless... |
00:24.49 | `p4r14h | well, if you just want to learn it and not use it heavily the x100p works just fine if you only need one PTSN interface |
00:24.54 | JT | wifi ip phones, don't go there |
00:25.23 | aldoenviro | I played with a cisco phone a few months ago... it was pretty cool |
00:25.33 | ChrisHardie | JT/File/anyone: I'm still stuck with getting asterisk to build/use a chan_zap driver. |
00:25.58 | *** join/#asterisk Meaty` (n=meaty3@office.abi.ca) |
00:25.58 | ChrisHardie | I installed the zaptel driver from SVN and got it to recognize the card |
00:26.19 | ChrisHardie | make menuselect recognizes chan_zap as an option and has it selected |
00:26.28 | ChrisHardie | but still no chan_zap.so being built. |
00:28.02 | ChrisHardie | If anyone has any advice/pointers, that'd be great. |
00:28.12 | Strom_M | which versions of everything are you running? |
00:28.17 | Strom_M | ChrisHardie: |
00:28.27 | aldoenviro | Would this do the trick? http://www.newegg.com/Product/Product.aspx?Item=N82E16833203012 |
00:28.36 | ChrisHardie | Asterisk 1.4.2, and I was using zaptel 1.4.0 but was told to use the version from SVN instead. |
00:28.56 | *** join/#asterisk cr4z3d (n=cr4z3d@ip70-162-97-91.ph.ph.cox.net) |
00:28.57 | Strom_M | ChrisHardie: so you got zapte;-1.4 svn branch? |
00:29.00 | ChrisHardie | yes |
00:29.04 | Qwell | aldoenviro: very, very low rating. avoid |
00:29.07 | Strom_M | or did you get zaptel trunk? |
00:29.18 | ChrisHardie | I had problems with getting modprobe to find it so I had to copy the .ko files to the right kernel directory, |
00:29.35 | ChrisHardie | I used " svn co http://svn.digium.com/svn/zaptel/branches/1.4 zaptel" |
00:29.58 | tzafrir | ChrisHardie, this may be a bad sign. It may mean you were building with an incorrect configuration |
00:30.08 | christo | is it possible to pass a call through an AGI before it's answered? |
00:30.08 | Strom_M | ChrisHardie: ok...if you'll wait for a few minutes, i'm building the latest 1.4 brahcn of everything right now |
00:30.19 | Strom_M | er |
00:30.21 | Strom_M | branch |
00:30.25 | Strom_M | god, i'm going dyslexic :) |
00:30.30 | tzafrir | And thus the modules will refuse to load (if this is indeed the case) |
00:30.33 | ChrisHardie | I don't doubt that something is flubbed up. I just didn't see how doing "make / make install" in each one was going to steer me wrong. |
00:31.06 | cr4z3d | hey i'm completely new to asterisk and i've been trying all day to get my VoIP system up and running.. I'm having some trouble figuring out what some of the debug stuff means |
00:31.09 | cr4z3d | can anyone help me out with that? |
00:31.22 | aldoenviro | I am reading that the zaptel drivers will work with my modem... any success on this? |
00:31.35 | ChrisHardie | Strom/tzafrirI got the modules to load once they were in the right directory. ztcfg shows 4 channels configured for the 4 ports on the card. |
00:31.41 | tzafrir | which modem do you have? |
00:31.55 | tzafrir | ChrisHardie, does ztcfg give an error? |
00:32.12 | ChrisHardie | nope, and when run with -vvvv it shows 4 channels configured. |
00:34.06 | tzafrir | What is your kernel version? |
00:34.20 | ChrisHardie | 2.6.12-9-386 |
00:34.52 | tzafrir | Anyway, building asterisk actually has nothing to do with the loaded zaptel. It has to do with /usr/include/zaptel/zaptel.conf (in 1.4) |
00:34.57 | aldoenviro | Communication controller: Intel Corporation 536EP Data Fax Modem I thought it was a 537... |
00:35.16 | tzafrir | ChrisHardie, Do you have /etc/asterisk/zaptel/zaptel.conf ? Is it from today? |
00:35.23 | aldoenviro | Communication controller: Rockwell International HCF 56k Data/Fax/Voice/Spkp (w/Handset) Modem (rev 01) one of these too |
00:35.52 | ChrisHardie | tzafrir: it's /etc/zaptel.conf now, but yes it's from today |
00:35.57 | aldoenviro | el cheapo modems that have been hanging around in a box for ages |
00:36.55 | tzafrir | oops, my typo: /usr/include/zaptel/zaptel.h |
00:37.10 | ChrisHardie | tzafrir: yes, have that too |
00:37.12 | tzafrir | I managed to make a mess of three different files |
00:37.38 | *** join/#asterisk Dovid (n=Dovid@l192-117-114-1.broadband.actcom.net.il) |
00:37.49 | tzafrir | ChrisHardie, your problem is with building asterisk, right? How exactly do you see that something is wrong? |
00:37.49 | *** join/#asterisk MaartenB (n=Maarten@84-105-197-100.cable.quicknet.nl) |
00:38.10 | MaartenB | hello everyone, can somebody please help me, I can not access my asterisk anymore with -r |
00:38.19 | ChrisHardie | tzafrir: When I try to make a call incoming or outgoing using the digium card, I get an error about the zap channel being unavailable |
00:38.24 | Dovid | MaeertenB: please explain |
00:38.27 | MaartenB | I got asterisk running as user asterisk now, but it says it is not running, while it is :( |
00:38.33 | ChrisHardie | I looked in the asterisk build directory in /channels, and it's not building chan_zap |
00:38.39 | tzafrir | MaartenB, is there an asterisk process running (ps aux | grep asterisk # or whatever) |
00:38.45 | Dovid | MaeertenB: u may need to connect as that user. |
00:38.53 | ChrisHardie | In menuselect, the chan_zap option is available and selected. |
00:38.55 | tzafrir | run: make menuconfig |
00:39.00 | MaartenB | tzafrir, yes, there is, it is working too, only asterisk -r is failing |
00:39.18 | MaartenB | it says " Unable to connect to remote asterisk (does /var/run/asterisk.ctl exists?) |
00:39.27 | Dovid | MaeertenB: I personally start asterisk in a screen session. |
00:39.30 | tzafrir | ChrisHardie, look in channels . Is chan_zap X-ed out? deselected? |
00:39.40 | ChrisHardie | in menuconfig under "Channel Drivers", "chan_zap" is NOT X-ed out, and is selected. |
00:39.40 | MaartenB | it is started with the /etc/init.d/asterisk script |
00:39.57 | Dovid | MaeertenB: I had that issue when I was trying to connect to it as a diffrent user than the one that it is running on |
00:40.00 | Dovid | (non root) |
00:40.16 | MaartenB | how did you solve it? |
00:40.23 | tzafrir | MaartenB, Asterisk either failed writing to /var/run/asterisk.ctl or wrote to another file (/var/run/asterisk/asterisk.ctl ?) |
00:40.23 | Dovid | ran asterisk as root |
00:40.36 | Dovid | tzafrir knows more - follow him |
00:40.39 | tzafrir | do you have /var/run/asterisk ? |
00:40.56 | MaartenB | tzafrir, yes, it exists, is owned by asterisk |
00:41.01 | MaartenB | tzafrir, empty... :( |
00:41.05 | ChrisHardie | tzafrir: I'm considering just trying to go back to the 1.2 series for now in hopes it would work better. |
00:41.46 | tzafrir | ChrisHardie, I asked you a simple question. Please go back to the asterisk build dir, run 'make menuselect' and answer it |
00:42.12 | ChrisHardie | I did answer it: "in menuconfig under "Channel Drivers", "chan_zap" is NOT X-ed out, and is selected." |
00:42.15 | tzafrir | MaartenB, how exactly do you start asterisk? |
00:42.28 | MaartenB | tzafrir, service asterisk start |
00:43.16 | tzafrir | ChrisHardie, have you tried re-running 'make' ? |
00:43.27 | ChrisHardie | Yes, several times, after "make clean" |
00:44.01 | tzafrir | Are there any local modifications to channels/Makefile? |
00:44.02 | christo | Hi all - I am dialling out and trying to run the call through an agi script as soon as it's placed (ie before it is answered). Is this possible? My agi isn't completing until I pick up. |
00:44.16 | ChrisHardie | tzafrir: I haven't made any, no. |
00:44.22 | Strom_M | ChrisHardie: does the word "zap" auto tab complete at the asterisk CLI? |
00:44.54 | tzafrir | we're in the build phase now |
00:45.53 | ChrisHardie | If someone's available for hire to help address this, I would be happy to negotiate something quickly. 3 hours of having our system down hasn't been pleasant. :) |
00:46.52 | *** join/#asterisk bkuhn (n=bkuhn@fsf/member/bkuhn/bkuhn) |
00:47.13 | *** join/#asterisk mmartinn (n=martin@adsl-065-005-200-225.sip.gnv.bellsouth.net) |
00:51.09 | christo | line 4 in this script implies that it's possible to do what I'm trying - is this correct? http://www.oldskoolphreak.com/tfiles/voip/mysql_call_logger.agi |
00:51.55 | cr4z3d | the person you are trying to reach is currently unreachable, please try again later |
00:52.09 | cr4z3d | is that a message setup by default on asterisk? |
00:54.46 | cr4z3d | i have iax2 debug enabled and i keep getting a bunch of tx-fram retry and rx-fram retry |
00:54.55 | cr4z3d | does that mean it's not autheticating properly? |
00:56.05 | *** join/#asterisk putzz (n=me@CPE00155824933c-CM00111ae07ac2.cpe.net.cable.rogers.com) |
01:00.10 | christo | cr4z3d - I think it would be more obvious at the CLI if your auth was failing |
01:00.27 | cr4z3d | i was at the CLI |
01:01.05 | cr4z3d | this is the exact messages: http://forums.digium.com/viewtopic.php?t=14469 |
01:01.42 | cr4z3d | i'm using version 1.12 (ubuntu repository only has that one) |
01:03.58 | spanglesontoast | I don't think my sip is registering |
01:04.02 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
01:04.48 | *** join/#asterisk Dovid[Laptop] (n=Dovid@l192-117-114-1.broadband.actcom.net.il) |
01:12.24 | cr4z3d | *CLI> iax2 shoMar 21 18:12:00 WARNING[15991]: res_musiconhold.c:421 spawn_mp3: Found no files in '/usr/share/asterisk/mohmp3' |
01:12.24 | cr4z3d | Mar 21 18:12:00 WARNING[15991]: res_musiconhold.c:493 monmp3thread: Unable to spawn mp3player |
01:12.28 | cr4z3d | why does that randomly come up |
01:16.10 | spanglesontoast | missing the music on hold files |
01:16.45 | spanglesontoast | that's why |
01:16.54 | cr4z3d | yeah but why are they even trying to go |
01:16.58 | cr4z3d | i don't have that setup |
01:17.06 | spanglesontoast | no idea |
01:17.10 | spanglesontoast | just likes to miss them ;0 |
01:17.17 | cr4z3d | i can't even seem to get my asterisk to connect to nufone |
01:17.31 | spanglesontoast | I can't get asterisk to connect to anything lol |
01:17.36 | cr4z3d | yeah me either haha |
01:18.33 | spanglesontoast | and I've dmzed this sucker |
01:20.36 | *** join/#asterisk robl^ (n=robl@pdpc/supporter/monthlybyte/robl) |
01:24.28 | cr4z3d | yeah asterisk doesn't seem like the kind of program i can just jump right into and have it working in 5 minutes |
01:24.35 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
01:26.18 | [TK]D-Fender | ~osmosis~sipnat |
01:26.28 | [TK]D-Fender | ~sipnat |
01:26.31 | jbot | extra, extra, read all about it, sipnat is for for more information about configurtion of Asterisk with SIP behind NAT, see http://voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
01:26.31 | spanglesontoast | yea been messing with nat |
01:26.31 | spanglesontoast | ;) |
01:26.31 | spanglesontoast | aswell |
01:26.54 | bulle | just forward 5060 and rtp ports to the asterisk box behind nat, and enter the routers public ip in the sip.conf file, as the comments tell you to |
01:27.09 | bulle | and it will work, its not harder then that |
01:27.21 | spanglesontoast | hmm |
01:27.43 | bulle | you most likely want to narrow the default rtp port ranges a bit, as they are from 10 000 to 20 000 |
01:27.44 | cr4z3d | is it normal when i type reload chan_iax2.so in the CLI for it to just stop after Registered IAX2 to blah blah |
01:28.51 | bulle | start from say 16384 and add as many ports as you might need, if unsure, and for just personal use, add say 100 ports |
01:29.03 | spanglesontoast | but the thing is dmzed :| |
01:29.31 | spanglesontoast | theres no firewall between them |
01:29.47 | spanglesontoast | I was wondering where I could see if it's registering or not |
01:29.47 | cr4z3d | Tx-Frame Retry[-01] -- OSeqno: 002 ISeqno: 002 Type: IAX Subclass: ACK |
01:29.57 | cr4z3d | does that mean i've autheticated with the iax2 server |
01:30.04 | bulle | spanglesontoast: the asterisk box wont have your public ip, even if its dmzed |
01:30.20 | bulle | spanglesontoast: so you still need to set the public ip of the router, in the sip.conf file, as per documentation |
01:30.33 | bulle | spanglesontoast: ifconfig eth0 tells you what ip number ? |
01:31.07 | spanglesontoast | it's 192.168.1.3 I set them manually ;) |
01:31.15 | bulle | see |
01:31.19 | bulle | so it cant possibly work that way |
01:31.45 | spanglesontoast | bindaddr is bound to that and extern is to my outside ip |
01:32.40 | bulle | externip=real ip |
01:32.45 | bulle | and then a localnet definition ? |
01:33.11 | spanglesontoast | subnet you mean ? |
01:33.12 | lokkju_wrk | question: would anyone be interested in a script you could point at some text files (configurable) that would read off the per minute rate for the outbound number you are dialing, and then ask you to confirm that you would like to make the call - perhaps also allow you to set it so it only activates when the rate/min is over a specified amount? (or, does someone know of a script that already does this?) |
01:33.25 | bulle | spanglesontoast: no, localnet most likely, in your sip.conf, if memory servers me right |
01:33.37 | bulle | localnet=192.168.0.0/255.255.0.0; All RFC 1918 addresses are local networks |
01:33.48 | spanglesontoast | localnet = 192.168.1.0/255.255.255.255 |
01:34.15 | Qwell | 255.255.255.255? |
01:34.29 | spanglesontoast | yea |
01:34.30 | JT | hmm yeah |
01:34.34 | JT | spanglesontoast: that's wrong |
01:34.37 | Qwell | umm, why? |
01:34.40 | spanglesontoast | er why ? |
01:35.07 | JT | <PROTECTED> |
01:35.09 | bulle | spanglesontoast: because that netmask makes no sense ?, the it basicly says that all bits in the ip are for the net and no for the host |
01:35.33 | spanglesontoast | so I need to change it on the router :| |
01:35.35 | bulle | spanglesontoast: also, dont use spaces between = etc, it tends to make asterisk go nuts from time to time |
01:35.47 | bulle | spanglesontoast: on the router ? |
01:35.50 | spanglesontoast | yea |
01:35.57 | spanglesontoast | that's what the router is assigning |
01:36.12 | JT | 255.255.255.255 is the broadcast address |
01:36.16 | [TK]D-Fender | qwell : what are you talking about? thats a perfectly valid host entry! ;) |
01:36.30 | Qwell | I never said it wasn't :P |
01:36.38 | [TK]D-Fender | ;) |
01:36.51 | *** join/#asterisk Weems (n=frodo@unaffiliated/weems) |
01:37.41 | spanglesontoast | well should i change the subnet ? |
01:38.05 | cr4z3d | 255.255.255.0 |
01:38.08 | cr4z3d | should be your subnet |
01:38.09 | bulle | yes, most likely to 255.255.255.0 |
01:38.23 | bulle | if all your machines are on ips on 192.168.1. |
01:38.26 | cr4z3d | yeah |
01:38.55 | spanglesontoast | hang on that's screwed up |
01:39.01 | spanglesontoast | the drop down says 255.0 |
01:39.18 | spanglesontoast | but on the other it says 255.255 |
01:39.36 | JT | the drop down... yes |
01:39.39 | cr4z3d | quick question, how do i find out if i'm connecting to my iax2 server correctly in the CLI? |
01:39.56 | JT | ...we know what you're talking about.. |
01:40.03 | spanglesontoast | ooh I can switch off nat :) |
01:40.13 | cr4z3d | uh |
01:40.16 | cr4z3d | i wouldn't recommend it |
01:40.19 | JT | ~wglwat |
01:40.22 | jbot | i guess wglwat is well, good luck with all that |
01:40.40 | lokkju_wrk | cr4z3d, iax2 show peers, perhaps? |
01:40.43 | marc\cba | spanglesontoast what are you trying to do? |
01:40.44 | spanglesontoast | ah |
01:40.46 | spanglesontoast | 192.168.1.0255.255.255.00.0.0.0LAN & Wireless |
01:40.54 | spanglesontoast | it is that your quite right |
01:41.04 | spanglesontoast | that was the wan one |
01:41.35 | cr4z3d | lokkju_wrk, when i do that i get 0online 0 offline and 1 unmonitored |
01:42.14 | spanglesontoast | hmm still don't work grrr |
01:42.24 | cr4z3d | woah Mar 21 18:41:09 WARNING[17408]: chan_iax2.c:9680 load_module: Unable to open IAX timing interface: No such file or directory |
01:42.42 | cr4z3d | that could be a huge reason it's not working couldn't it |
01:45.44 | spanglesontoast | is there anything more verbose than -v |
01:45.45 | spanglesontoast | :) |
01:46.23 | cr4z3d | screw this i'm removing asterisk and building it from source with the newest one.. |
01:46.30 | JT | spanglesontoast: -vvvvvvvvvvvvvvvvvvvvvvvvvv |
01:46.40 | JT | really, only 5 to 10 vs are needed |
01:46.43 | cr4z3d | set verbose 10 |
01:46.48 | mmartinn | "set verbose 99999999999999999999999999" |
01:46.49 | JT | there is no more verbosity after that |
01:47.10 | mmartinn | I guess it's "core set verbose 99999999999999999999" now |
01:47.19 | cr4z3d | anyone here ever setup asterisk with NuFone.. i'm having such a hard time getting it working |
01:47.19 | spanglesontoast | lol |
01:47.20 | JT | yeah but who uses 1.4? :P |
01:47.26 | Strom_M | core set verbose 9999999999999999999999999999999999999999999999999999999999999999999999 :) |
01:47.27 | cr4z3d | i'm using 1.12 |
01:47.34 | JT | 1.12 |
01:47.35 | cr4z3d | since that's what ubuntu has with apt-get |
01:47.37 | spanglesontoast | hmm it showed me something I didn't see |
01:47.38 | JT | ? |
01:47.53 | spanglesontoast | yea I'm using ubuntu maybe why we have so many problems cr4z3d |
01:48.07 | JT | i doubt it |
01:48.14 | JT | distro is usually irrelevant |
01:48.21 | cr4z3d | yeah i doubt it has any effect either |
01:48.27 | spanglesontoast | which version of ubuntu you using ? |
01:48.30 | mmartinn | The * from ubuntu's multiverse is 1.12? |
01:48.31 | cr4z3d | edgy |
01:48.34 | spanglesontoast | ah |
01:48.37 | spanglesontoast | fiesty here :) |
01:48.42 | [TK]D-Fender | spanglesontoast, You have * running. once you've gotten that far you need to learn how to USE it |
01:48.45 | JT | cr4z3d: i doubt it's 1.12 |
01:48.51 | Qwell | No such version |
01:48.52 | JT | cr4z3d: show version |
01:48.59 | cr4z3d | hold on i just removed it |
01:49.00 | Qwell | 1.2.12 maybe |
01:49.07 | *** join/#asterisk oQPa (n=uawename@33.Red-83-34-60.dynamicIP.rima-tde.net) |
01:49.19 | spanglesontoast | well it's just a mission for the thing to connect to this sip server |
01:49.33 | cr4z3d | oh my bad |
01:49.35 | cr4z3d | 1.2.12 |
01:49.38 | Qwell | eww |
01:49.43 | Qwell | eww |
01:49.51 | Qwell | both of you need to upgrade |
01:49.58 | mmartinn | 1.2.12 is still better than 1.12, lol |
01:50.02 | cr4z3d | will that help out at all though |
01:50.08 | spanglesontoast | Asterisk 1.2.16 |
01:50.10 | JT | better than something that doesn't exist |
01:50.11 | cr4z3d | i can't even get it to connect to nufone |
01:50.12 | spanglesontoast | oh |
01:50.13 | spanglesontoast | lol |
01:50.16 | Qwell | spanglesontoast: You too. |
01:50.18 | Qwell | Upgrade |
01:50.20 | spanglesontoast | lol |
01:50.21 | *** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.itb.ac.id) |
01:50.22 | *** join/#asterisk oQPa (n=uawename@33.Red-83-34-60.dynamicIP.rima-tde.net) |
01:50.26 | spanglesontoast | this the reason then ? |
01:50.29 | spanglesontoast | :D |
01:50.59 | [TK]D-Fender | spanglesontoast, pastebin your [general] section of sip.conf masking only passwords |
01:51.01 | [TK]D-Fender | ~pb |
01:51.05 | jbot | somebody said pb was a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
01:51.55 | cr4z3d | i'm reinstalling and building from source.. this way i know it's nothing wrong with the one ubuntu gives you |
01:51.55 | mihinomenest | I have a serious problem with asterisk. |
01:52.19 | bulle | mihinomenest: who doesnt |
01:52.23 | mihinomenest | it goes down everytime the power goes out because of the thunderstorm. |
01:52.39 | sevard | get a UPS |
01:52.45 | [TK]D-Fender | mihinomenest, Your ire is misplaced |
01:52.46 | spanglesontoast | http://pastebin.ca/405972 |
01:53.10 | [TK]D-Fender | mihinomenest, And your lack of strategy apalling :) |
01:53.44 | spanglesontoast | I can't believe how much I've learn't over the past few hours |
01:53.53 | spanglesontoast | I know it's only the sip part |
01:53.55 | orlock | mihinomenest: you know that your phones will BLOW UP if a lightning bolt hits them? |
01:54.01 | orlock | in fact your whole house may catch on fire. |
01:54.03 | lokkju_wrk | [TK]D-Fender, did you see my idea for a script earlier? have you seen anything that already does it? |
01:54.03 | spanglesontoast | but with channels etc and conference rooms |
01:54.05 | mihinomenest | you actually banged your head on the desk because of me, didn't you [TK]D-Fender? |
01:54.08 | orlock | i would go and bitch to somebody about that too. |
01:54.15 | [TK]D-Fender | lokkju_wrk, No I didn't |
01:54.29 | [TK]D-Fender | mihinomenest, No.... you're not worth it :) |
01:54.32 | sevard | TK bangs his head on the table every 3 and a half minutes. |
01:54.42 | lokkju_wrk | [TK]D-Fender, a script you could point at some text files (configurable) that would read off the per minute rate for the outbound number you are dialing, and then ask you to confirm that you would like to make the call - perhaps also allow you to set it so it only activates when the rate/min is over a specified amount? (or, does someone know of a script that already does this?) |
01:54.48 | sevard | he's like old faithful |
01:54.50 | mihinomenest | good idea. |
01:55.01 | mihinomenest | forgoe the frustration and go right for the pain. |
01:55.21 | *** join/#asterisk billzybub (i=bill@68.82.147.79) |
01:55.21 | [TK]D-Fender | lokkju_wrk, Sounds easy enough, but nothing "canned" out there that I've heard of off-hand |
01:55.34 | billzybub | good evening fine people |
01:55.48 | sevard | billzybub: eat smack |
01:55.57 | [TK]D-Fender | harem* |
01:56.00 | lokkju_wrk | k, cool... I want it so I get warned if I am about to make a 10/min call, so I'll just have to write it myself :) |
01:56.04 | spanglesontoast | http://pastebin.ca/405972 .... |
01:56.05 | [TK]D-Fender | *bangs* |
01:56.06 | billzybub | smack? |
01:56.10 | [TK]D-Fender | geez... can't type tonight |
01:56.11 | sevard | lokkju_wrk: I could make such a script |
01:56.12 | billzybub | as in heroin? |
01:56.25 | lokkju_wrk | sevard, I'll have no problem writing it, thank you much :) |
01:56.28 | *** part/#asterisk oQPa (n=uawename@33.Red-83-34-60.dynamicIP.rima-tde.net) |
01:56.29 | sevard | lokkju_wrk: ah. |
01:57.18 | spanglesontoast | damn flies |
01:57.20 | spanglesontoast | :) |
01:57.46 | mmartinn | lokkju_wrk: I think I saw something about a rate list of exchanges and rates per carrier when searching about NANP once |
01:59.09 | lokkju_wrk | mmartinn, very likely - but I want a verbal "this call will be charged at XX per minute. Please dial 1 to continue" |
01:59.20 | lokkju_wrk | mmartinn, course, I *would* like to see if anyone has reasonable rates for calling iridium... most people are $8+/min - some are as high as $11 |
01:59.23 | billzybub | hey, can somone tell me what the best front end for asterisk is? |
01:59.38 | orlock | vi |
01:59.52 | mihinomenest | nano. |
02:00.01 | tzafrir | vim |
02:00.06 | billzybub | i like vi, but i dont think my employees will |
02:00.20 | cr4z3d | Mar 21 18:59:59 WARNING[18388]: chan_sip.c:12863 reload_config: Failed to bind to 0.0.0.0:5060: Address already in use |
02:00.30 | mihinomenest | employees don't configure asterisk. |
02:00.35 | lokkju_wrk | well, technically a soft or hard phone is a "front end" for asterisk... so the Polycoms! |
02:00.38 | [TK]D-Fender | spanglesontoast, http://pastebin.ca/405983 |
02:00.38 | billzybub | mine do |
02:00.47 | mihinomenest | Lord of the Network configures asterisk. |
02:00.51 | tzafrir | what do they need to configure? |
02:00.56 | orlock | billzybub: freepbx |
02:00.58 | [TK]D-Fender | spanglesontoast, What EXACTLY have you forwarded to your * server? |
02:01.02 | cr4z3d | how come i get Mar 21 18:59:59 WARNING[18388]: chan_iax2.c:9680 load_module: Unable to open IAX timing interface: No such file or directory |
02:01.12 | spanglesontoast | no that's outbound |
02:01.14 | cr4z3d | when i start up asterisk after a fresh install |
02:01.17 | lokkju_wrk | no, no, the BOFH configures it for you... and controls your life. and the life of everyone on his network.... |
02:01.18 | tzafrir | destar has nice user config |
02:01.25 | spanglesontoast | basically you dial the prefix and it dials a number on there |
02:01.45 | orlock | i am a bastard and i'm ok.. i work all night and i sleep all day! |
02:02.07 | cr4z3d | oh man asterisk now sounds so easy |
02:02.08 | spanglesontoast | the voipuser one |
02:02.27 | [TK]D-Fender | tzafrir : I keep reading that as Death Star ;) |
02:04.54 | *** join/#asterisk Hansin321 (i=Eric@c-71-56-216-97.hsd1.co.comcast.net) |
02:05.09 | mmartinn | the manager's manager_event is making me cry... |
02:05.45 | billzybub | orlock, thanks checking out the screenies now |
02:06.21 | JT | [TK]D-Fender: yeah, misreadings are funny like that |
02:06.28 | JT | [TK]D-Fender: i read deskstar as deathstar |
02:07.21 | *** join/#asterisk AJaymn (n=Me@66-188-80-40.dhcp.mdsn.wi.charter.com) |
02:07.26 | [TK]D-Fender | JT : They WERE nick-named that :) |
02:07.39 | [TK]D-Fender | JT Hitachi/IBM's IDE drive series that is... |
02:07.41 | *** part/#asterisk AJaymn (n=Me@66-188-80-40.dhcp.mdsn.wi.charter.com) |
02:07.42 | spanglesontoast | also what you mean forwarding ? |
02:07.57 | [TK]D-Fender | spanglesontoast, What ports are your forwarding from your router to *? |
02:08.04 | spanglesontoast | it's dmzed |
02:08.07 | JT | i know :) |
02:08.11 | JT | i had one :/ |
02:08.26 | [TK]D-Fender | JT, Worse still.. my frined had 2 ... in raid *0* |
02:08.49 | [TK]D-Fender | JT, "repetitive tragic failure" comes to mind... |
02:09.29 | mihinomenest | my dad calls those drives "deathstar" |
02:09.46 | [TK]D-Fender | spanglesontoast, Ok well * is set up for basic NAT work. Now its up to you to make sure you are registring properly (as you appear to be attempting), and that those entries you made match the auth you need to send to place calls |
02:11.15 | spanglesontoast | -- Executing Dial("SIP/edd-08195f40", "SIP/8009178765@voipuser|60") in new stack |
02:11.15 | spanglesontoast | <PROTECTED> |
02:11.20 | spanglesontoast | that's what it's doing |
02:11.37 | spanglesontoast | and just carries on |
02:12.51 | bulle | i bet there is no host named voipuser on the internet |
02:13.06 | spanglesontoast | www.voipuser.org |
02:13.19 | spanglesontoast | allows you to make free uk calls :) |
02:13.49 | spanglesontoast | and I know it works as a community demoed it on the radio with asterisk |
02:14.09 | spanglesontoast | I can use a normal soft client with it fine |
02:14.37 | spanglesontoast | and they have a setup aswell http://www.voipuser.org/forum_topic_330.html |
02:14.38 | cr4z3d | can someone help me figureout why i keep getting these two errors when loading asterisk? |
02:15.35 | bulle | spanglesontoast: so what is the name of the voipuser.org sip proxy then ? |
02:15.37 | *** join/#asterisk ooor4 (i=blah@69-163-163-195.atlsfl.adelphia.net) |
02:15.41 | sevard | http://www.tfproject.org/tfp/showthread.php?t=114784 |
02:15.57 | spanglesontoast | sip.voipuser.org |
02:16.02 | *** join/#asterisk stkn (n=stkn@gentoo/developer/pdpc.active.stkn) [NETSPLIT VICTIM] |
02:16.02 | *** join/#asterisk nitram (i=foo@superblob.com) [NETSPLIT VICTIM] |
02:16.02 | *** join/#asterisk kore (i=kore@mindwipe.org) [NETSPLIT VICTIM] |
02:16.05 | bulle | so place the call to that then |
02:16.10 | spanglesontoast | hmm |
02:16.14 | sevard | uh, wrong paste. |
02:16.18 | sevard | nsfw :\ |
02:17.32 | mmartinn | <awkward_silence/> |
02:17.35 | [TK]D-Fender | spanglesontoast, And why don't you have a peer entry for [voipuser] like that sample SHOWS you? |
02:17.51 | spanglesontoast | doesn't make any difference |
02:18.01 | spanglesontoast | plus I've been messing with it |
02:18.15 | sevard | hehehe |
02:18.37 | [TK]D-Fender | spanglesontoast, that dial statement is LOOKING FOR IT. Damn right its IMPORTANT |
02:19.43 | spanglesontoast | exten => _81.,1,Dial(SIP/${EXTEN:3}@voipuser,60) |
02:20.47 | spanglesontoast | oh |
02:20.54 | spanglesontoast | that did something error 503 on sjphone |
02:21.14 | spanglesontoast | says the circuit is busy |
02:21.19 | spanglesontoast | could that be true ? |
02:22.01 | [TK]D-Fender | spanglesontoast, you don't have an entry for [voipuser]. it is also not a valid hostname. that entire line is a DEAD END. |
02:22.16 | [TK]D-Fender | spanglesontoast, Go follow what the sample handed to you |
02:22.24 | spanglesontoast | well I changed it to [voipuser] |
02:22.25 | [TK]D-Fender | spanglesontoast, And then expand on it |
02:27.48 | *** join/#asterisk sbingner (n=sam@pdpc/supporter/sustaining/sbingner) |
02:28.57 | *** join/#asterisk ppyy (i=ppyy@218.93.70.78) |
02:29.19 | Dovid | spanglesontoas: u have ${EXTEN:3} |
02:29.29 | Dovid | dont u want ${EXTEN:2} ? |
02:29.50 | bulle | he firstly needs a proper entry for the peer |
02:30.02 | Dovid | bulle: he dosent have the peer ? |
02:30.10 | Dovid | rofl |
02:30.21 | bulle | nope |
02:30.26 | Dovid | hahahahahaha |
02:30.50 | Dovid | its like a client of mine who's wifi wasnt working. it wasnt plugged in (true story) |
02:31.01 | Qwell | grr, I need a second tv tuner.. missing the last 10-20 seconds of a show so it can start recording another is annoying |
02:31.33 | Dovid | hehe |
02:31.43 | Dovid | Qwel: what r u using to record ? |
02:31.46 | Qwell | mythtv |
02:31.51 | Dovid | there is an open source os... |
02:31.53 | Dovid | ah. u got it.. |
02:31.55 | Dovid | how is it ? |
02:31.58 | Qwell | love it |
02:32.01 | Dovid | hehe |
02:32.06 | Dovid | how many gigs per hour ? |
02:32.11 | Qwell | dunno |
02:32.28 | Dovid | when i get back to the US ia m getting one |
02:32.31 | Dovid | with multiple cards. |
02:32.36 | Qwell | probably just about a gig |
02:32.40 | Dovid | can u set it to remove commercials by default ? |
02:32.49 | Qwell | yeah |
02:32.52 | Dovid | like the old tivo's ? |
02:32.56 | Dovid | nice !!! |
02:32.57 | Qwell | it's pretty good at it too |
02:33.51 | carrar | woot |
02:34.04 | carrar | so tired of walking around all day! |
02:34.37 | *** join/#asterisk Ac1dcrawl (n=cow@64.31.169.118) |
02:34.46 | bulle | carrar: get a segway |
02:35.10 | carrar | man that would be perfect here |
02:35.16 | Qwell | "here"? |
02:35.18 | carrar | put a seat on it |
02:35.22 | Ac1dcrawl | I'm having a problem, I have all incoming calls set to ring to a single extension. When I call into the asterisk box, it rings once on the extension then goes busy |
02:35.23 | carrar | springvon |
02:35.27 | Qwell | ahh, figured |
02:35.29 | Ac1dcrawl | I see the following line in the logs: DIALSTATUS=CANCEL |
02:35.31 | Ac1dcrawl | any ideas? |
02:35.45 | spanglesontoast | hmm |
02:35.48 | Qwell | carrar: except for those damn escalators :p |
02:35.53 | carrar | heh |
02:35.53 | spanglesontoast | why is the network congested :| |
02:35.55 | carrar | elevator! |
02:35.58 | Qwell | those bugged the hell out of me, heh |
02:36.45 | Qwell | I actually wonder how a segway would work on an escalator |
02:37.23 | spanglesontoast | I'm sleepy now i'm going to bed ty for all those put up with my dumbness ;) |
02:40.34 | *** join/#asterisk anthonyl (n=fbffff@r74-193-79-81.pfvlcmta01.grtntx.tl.dh.suddenlink.net) |
02:42.38 | carrar | my obligratory photo with Mark http://www.osburn.com/IMG_5659.jpg |
02:42.41 | carrar | heh |
02:43.06 | Qwell | my picture with Mark rocks :D |
02:43.10 | carrar | heh |
02:44.34 | cr4z3d | hmm.. i keep getting these 3 error messages when starting up asterisk. can anyone help me? i posted the errors and config files on the forum: http://forums.digium.com/viewtopic.php?p=47177#47177 |
02:46.55 | *** join/#asterisk boch (n=fran@190.48.194.212) |
02:49.19 | Qwell | carrar: see msg ;) |
02:51.03 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
02:54.25 | billzybub | cr4zed, why dont you grab the 1.4x source, rebuild and see how that works for you? |
02:55.51 | [TK]D-Fender | cr4z3d, First you're running Ekiga on the same box as * aren't you? |
02:55.52 | billzybub | cr4zed, also, since your using ubuntu make sure you have all the development tools installed |
02:56.13 | [TK]D-Fender | billzybub, * is INSTALLED and running. that isn't his problem so far |
03:01.00 | billzybub | nice, a nuclear radioactive device has been stolen in philly |
03:01.02 | cr4z3d | [TK]D-Fender, yeah ekiga came with ubuntu default |
03:01.09 | cr4z3d | is that a problem? |
03:01.28 | billzybub | ekiga is just a client, that should keep you from binding to 5060 on your interface |
03:01.43 | cr4z3d | ooh alright so that's what's stopping that |
03:01.46 | [TK]D-Fender | cr4z3d, tahts the reason you have the "already bound" error. Ekiga has claimed ownership of the SIP port before *. |
03:02.02 | cr4z3d | ok so if i just turn of ekiga that won't happen anymore? |
03:02.03 | billzybub | crazed, is ekiga running ? |
03:02.05 | [TK]D-Fender | cr4z3d, You need to run your soft-phone on a differen port and set taht in * |
03:02.08 | cr4z3d | yeah it's running |
03:02.24 | billzybub | i dont run my xlite on a different port... |
03:02.44 | cr4z3d | i'm just going to run a windows soft phone instead |
03:02.54 | cr4z3d | to avoid any of that port stuff for now |
03:03.06 | cr4z3d | i just want to get it up and connected to my NuFone |
03:04.29 | *** join/#asterisk djs_2_6 (n=DJS@cpe-071-077-048-198.nc.res.rr.com) |
03:04.51 | cr4z3d | ok so now i disabled ekiga and damn |
03:04.58 | cr4z3d | i got like 3 pages of notices and warnings |
03:05.16 | cr4z3d | most seem to do with musiconhold.c |
03:06.05 | [TK]D-Fender | crazed, just make sure that "mode=files" in musiconhold.conf |
03:06.41 | billzybub | crazed, get into a shell and type this in: netstat -tap |
03:06.53 | [TK]D-Fender | cr4z3d, And that post you made did not actualyl show a SINGLE error related to your nufone setup |
03:06.58 | cr4z3d | default, mode=quietmp3 |
03:07.01 | billzybub | look for ekiga |
03:07.10 | [TK]D-Fender | cr4z3d, Change it and reload |
03:07.14 | billzybub | what port is it established tto |
03:07.47 | [TK]D-Fender | billzybub, 5060... thats th clrea reason for the error.... |
03:07.59 | *** join/#asterisk aptura (n=jondoe@S010600a0c93f6f7e.vs.shawcable.net) |
03:08.16 | cr4z3d | ok now i get notices about res_odbc.c |
03:08.41 | cr4z3d | and this warning |
03:08.42 | cr4z3d | Mar 21 20:07:50 WARNING[21264]: res_odbc.c:565 odbc_obj_connect: res_odbc: Error SQLConnect=-1 errno=0 [unixODBC][Driver Manager]Data source name not found, and no default driver specified |
03:09.31 | billzybub | crazed try this for your softphone http://www.xten.com/index.php?menu=download_xlite&platform=linux |
03:09.33 | cr4z3d | [TK]D-Fender, really? so how can i see if i'm connected to NuFone or not? iax2 show peers says 0 online |
03:10.15 | cr4z3d | alright billzybub i'll take a look at as soon as i can get these errors away |
03:10.31 | [TK]D-Fender | cr4z3d, You are not issuing a "qualify" to it so don't EXPECT to see anything. |
03:10.47 | [TK]D-Fender | cr4z3d, That is not in and of itself indicative of anything. |
03:10.53 | [TK]D-Fender | cr4z3d, lets move along... |
03:11.06 | *** join/#asterisk CunningPike (n=CunningP@S010600095b33697f.vc.shawcable.net) |
03:11.19 | [TK]D-Fender | cr4z3d, in modules.conf add "noload => res_odbc.so" |
03:11.25 | [TK]D-Fender | cr4z3d, and start it up again |
03:12.57 | cr4z3d | hm this time i got a really bad error and the program crashed |
03:13.10 | cr4z3d | Mar 21 20:12:31 WARNING[21446]: loader.c:325 __load_resource: /usr/lib/asterisk/modules/func_odbc.so: undefined symbol: odbc_obj_disconnect |
03:13.10 | cr4z3d | Mar 21 20:12:31 WARNING[21446]: loader.c:554 load_modules: Loading module func_odbc.so failed! |
03:13.43 | [TK]D-Fender | cr4z3d, in modules.conf add "noload => func_odbc.so" |
03:13.51 | Qwell | there will be more |
03:14.04 | Qwell | cdr_odbc et al |
03:14.05 | [TK]D-Fender | qwell : and they will in turn be stifled :D |
03:14.18 | billzybub | damn, see that chicago cop beeting up that lady bartender for flaggin him? |
03:14.39 | tene | billzybub: "flaggin"? |
03:14.39 | [TK]D-Fender | cr4z3d, How about you go into Synaptic and install unixODBC :) |
03:15.04 | cr4z3d | one error |
03:15.06 | cr4z3d | Mar 21 20:14:36 WARNING[21535]: chan_iax2.c:9680 load_module: Unable to open IAX timing interface: No such file or directory |
03:15.22 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
03:15.29 | *** join/#asterisk xtr-II (n=94752345@S0106000c41ed11e1.vf.shawcable.net) |
03:15.31 | cr4z3d | unixODBC? |
03:16.18 | Qwell | if res_odbc got built, it is installed |
03:16.24 | Qwell | on 1.4 at least? |
03:16.26 | *** join/#asterisk mroberto (n=dads@S0106001346face5f.ed.shawcable.net) |
03:16.43 | mroberto | I need some help using cepstral with asterisk, is anybody using it |
03:16.49 | cr4z3d | yeah it's already installed |
03:16.52 | cr4z3d | and newest version too |
03:17.57 | mroberto | i need some help using it |
03:18.09 | [TK]D-Fender | qwell :Ubuntu packaged <0 |
03:18.29 | *** join/#asterisk dmux (n=dmux@c906497d.virtua.com.br) |
03:18.41 | cr4z3d | it's version 1.2.12 if that helps at all |
03:18.42 | [TK]D-Fender | cr4z3d, Make sure to install Zaptel as well |
03:18.54 | billzybub | tene, its a term used by bartenders that pretty much means no more serving a patron because they are obviously thoroughly inebriated or disorderly |
03:19.10 | mroberto | <[TK]D-Fender>: Do youhave experience with cepstral |
03:19.11 | cr4z3d | it's install zaptel right now |
03:19.25 | cr4z3d | oh wait it ran into an error while installing |
03:19.29 | [TK]D-Fender | mroberto, Nope |
03:19.41 | billzybub | mroberto, thats that do? |
03:19.49 | cr4z3d | Zaptel telephony kernel driver: FATAL: Module ztdummy not found. |
03:20.12 | billzybub | modprobe ztdummy |
03:20.17 | billzybub | reload |
03:20.18 | mroberto | I have the module setup and when i use it it i dont get any sound |
03:20.47 | cr4z3d | FATAL: Module ztdummy not found. |
03:21.06 | billzybub | crazed type modprobe ztdummy |
03:21.14 | cr4z3d | yeah i did and i got that fatal message |
03:21.33 | billzybub | you need to make install that then |
03:21.37 | *** join/#asterisk kgx (n=kgx@60.234.20.178) |
03:22.33 | cr4z3d | hm how would i go about doing that and what exactly is ztdummy? it hink i read something about it during the setup guide |
03:23.57 | JT | you only need zaptel if you are using zap hardware, or meetme conferences, or iax2 trunking |
03:24.14 | JT | ztdummy if you have zaptel but no zap hardware |
03:24.43 | billzybub | ztdummy is a timer that runs off of some imbeded usb hardware i think |
03:25.07 | cr4z3d | hm i dont plan on doing any meetme confrences or using zap hardware |
03:25.16 | cr4z3d | and i'm pretty sure i'm not doing any iax2 trunking |
03:25.19 | JT | yes it uses usb |
03:25.20 | [TK]D-Fender | Of either USB or the 2.6 kernel timer |
03:25.33 | JT | either way |
03:25.34 | billzybub | are you just usinf a softfone or do you have any fxs setup? |
03:25.41 | JT | you don't need it unless you do :) |
03:25.53 | cr4z3d | my plan is to just use a softphone until i can get a sip enabled hardware phone |
03:26.39 | billzybub | so just build asterisk, its my understanding zaptel is a seperate package i think, just installed asterisk myself yesterday |
03:27.16 | JT | it is a seperate package |
03:27.18 | cr4z3d | i have it installed ad the only error left thanks to [TK]D-Fender is Mar 21 20:26:47 WARNING[22830]: chan_iax2.c:9680 load_module: Unable to open IAX timing interface: No such device or address |
03:27.38 | cr4z3d | oh timing that was the zaptel thing wasn't it |
03:27.38 | Juggie | 'modprobe ztdummy' |
03:27.43 | Qwell | cr4z3d: it's a warning, NOT an error |
03:27.54 | cr4z3d | oh so does it really mean anything for me? |
03:28.02 | Juggie | you wont be able to use iax trunking without it. |
03:28.05 | Qwell | Not if you aren't using iax trunking |
03:28.18 | Qwell | (and if you're connecting to nufone - you aren't) |
03:28.29 | cr4z3d | ok cool so everything should be working then? |
03:28.55 | [TK]D-Fender | cr4z3d, Well.. lets say nothing "pre-broken" :) |
03:29.16 | *** join/#asterisk BigCanOfTuna (n=arustad@dsl-mac-66-18-226-119-cgy.nucleus.com) |
03:29.38 | cr4z3d | well yeah now it's just error from my configurations if it's not working right? |
03:29.48 | billzybub | crazed why arent you using the 1.4 source? |
03:29.54 | [TK]D-Fender | cr4z3d, So far... |
03:30.05 | cr4z3d | because i didn't compile it from source i just apt-get install asterisk |
03:30.10 | cr4z3d | and used the one it came with |
03:30.16 | billzybub | oh no, dont do that |
03:30.19 | [TK]D-Fender | billzybub, Becuase he's on Ubuntu and working with lovely packages! |
03:30.29 | billzybub | im on ubuntu too |
03:30.44 | cr4z3d | packages are awesome i wish windows had something like that |
03:30.50 | billzybub | heh |
03:31.14 | cr4z3d | but anyway how do i check to see if i connected with nufone? |
03:31.29 | billzybub | is nufone a softfone? |
03:31.40 | JT | it's an itsp |
03:31.41 | cr4z3d | no it's my service provider |
03:31.45 | billzybub | sudo asterisk -r should put you in the console |
03:31.51 | cr4z3d | i'm in the console |
03:31.56 | cr4z3d | iax2 show peers |
03:32.03 | cr4z3d | i see it but it says status is unmonitored |
03:32.47 | billzybub | that may be a good thing :) |
03:33.04 | cr4z3d | really? |
03:33.04 | billzybub | what do they charge you over there? |
03:33.11 | cr4z3d | uh 2 cents a minute i think |
03:33.16 | cr4z3d | and $5 a month for a phone # |
03:33.20 | billzybub | on all calls? |
03:33.28 | cr4z3d | 5 cents international i believe |
03:33.32 | [TK]D-Fender | cr4z3d, Ok, new concept for you. you are not "connected" to anything. SIP/IAX do not maintin constant "connections". |
03:33.48 | cr4z3d | oh did not know that |
03:33.53 | cr4z3d | so how can i check if they can connect |
03:34.12 | [TK]D-Fender | cr4z3d, dial the # they gave you that you supposedly set up. Try dialing OUT. |
03:34.20 | [TK]D-Fender | cr4z3d, You know... USE THEM. |
03:34.34 | billzybub | crazed, how many out-bound connections does that give you? |
03:34.38 | cr4z3d | i can't test dialing out yet since i had to turn off ekiga for now but i'll see if ican dial in |
03:34.47 | *** join/#asterisk mog (n=mog@c-71-207-200-130.hsd1.al.comcast.net) |
03:34.47 | *** mode/#asterisk [+o mog] by ChanServ |
03:35.17 | billzybub | try xlite when you get things set up, i think its neet-er :D |
03:35.37 | billzybub | kinda looks like a phone instead of a window |
03:35.47 | cr4z3d | ok when dialing in 15 seconds later it goes "the person you are calling is currently unreachable" |
03:36.00 | *** join/#asterisk vader-- (n=me@71.226.201.15) |
03:36.02 | vader-- | hola |
03:36.03 | JT | are you registering? |
03:36.05 | cr4z3d | and the iax2 show peers status never changed |
03:36.22 | cr4z3d | nothing shows up in the CLI window |
03:36.25 | [TK]D-Fender | cr4z3d, you are not WATCHING the peer, nor do you need to |
03:36.29 | JT | you need to register |
03:36.33 | billzybub | hey crazed, when you get things rolling and i peer to you for my outbound domestic? |
03:36.34 | [TK]D-Fender | cr4z3d, Go set up a phone now. |
03:36.47 | [TK]D-Fender | JT, No, he DOESN'T. (not to place calls anyways) |
03:36.47 | vader-- | Do you guys know of any good voip providers that offer goodpricing that have the following, regular lines, fax and 800 with the ability for me to port a current fax line over and a 800 number over |
03:37.01 | JT | [TK]D-Fender: he said he's calling himself |
03:37.07 | JT | pay attention :) |
03:37.20 | JT | he "can't" call out because ekiga is off |
03:37.21 | vader-- | i don't think vonage will allow me to port numbers over to fax line or port a 800 number over |
03:37.24 | [TK]D-Fender | cr4z3d, "iax2 show registry" |
03:37.25 | cr4z3d | i have it set up with register => user:pass@switch-1.nufone.net if i remember the config correctly |
03:37.26 | billzybub | intervention is on A&E it a great show |
03:37.40 | cr4z3d | state registered |
03:37.41 | JT | vader--: umm, fax? |
03:37.42 | billzybub | oxycontin addict this week :D |
03:37.44 | vader-- | ya |
03:37.48 | [TK]D-Fender | vader--, FoIP = death |
03:37.52 | JT | vader--: very few people do fax properly yet |
03:37.56 | Qwell | FoIP == good |
03:37.57 | JT | FoIP is alright |
03:38.00 | Qwell | FoVoIP == dumb |
03:38.02 | JT | FoVoIP bad |
03:38.04 | vader-- | i have a pots line that is used for faxing |
03:38.15 | vader-- | vonage offers a fax line with a voice line for 50$ a month |
03:38.33 | JT | vader--: very few people do T.38 yet |
03:38.34 | cr4z3d | [TK]D-Fender, the state shows registered |
03:38.39 | JT | T.38 is what you need for faxing |
03:38.53 | [TK]D-Fender | cr4z3d, Yay. No on to the more than probable networking issues! |
03:39.00 | vader-- | i need to figure out if i can port a 800 number over |
03:39.08 | cr4z3d | haha alright |
03:39.18 | billzybub | hey dont you need a timer for IAX peering? |
03:39.19 | [TK]D-Fender | cr4z3d, So.. your * server behind NAT? |
03:39.31 | [TK]D-Fender | billzybub, No, only for TRUNKING. |
03:39.35 | cr4z3d | well i thought iax tunneled around nat? |
03:39.37 | cr4z3d | but yeah |
03:39.48 | vader-- | i need to figure out what provider owns this 800 number |
03:39.50 | [TK]D-Fender | cr4z3d, You need to forward 4569 UDP to your * box |
03:40.02 | cr4z3d | hm alright no problem |
03:40.15 | JT | [TK]D-Fender: not if he registers he shouldn't |
03:40.29 | JT | as long as there's no interfering firewall |
03:40.40 | cr4z3d | does ubuntu have a firewall by default |
03:40.46 | JT | no idea |
03:40.47 | cr4z3d | i know on my router i have the spi firewall disabled |
03:42.17 | [TK]D-Fender | JT, He's behind NAT. Don't think for a second that Nufone will be wasting packets keeping a UDP port forwarding towards his * |
03:42.32 | [TK]D-Fender | cr4z3d, Yes, you have to forward it. |
03:42.55 | JT | [TK]D-Fender: umm, i'm behind NAT too |
03:43.18 | [TK]D-Fender | JT, if his NAT doesn't get any UDP keep-alive it'll close down the exterior port. |
03:43.20 | JT | [TK]D-Fender: and have NEVER needed to port forward to use SIP or IAX if i'm connected to a provider (ie. clients don't connect to my asterisk) |
03:43.37 | [TK]D-Fender | JT only way it'd survive is if YOUR system was nagging the outside world constantly. |
03:43.49 | JT | [TK]D-Fender: that's what registration in a period of time less than the NAT timeout |
03:43.57 | [TK]D-Fender | so fer cryin' out loud, just forward the darned port! |
03:43.59 | [TK]D-Fender | :) |
03:44.02 | JT | [TK]D-Fender: that's how most VoIP nat-punching systems do it |
03:44.15 | [TK]D-Fender | JT you shouldn't be registering every other minute :) |
03:44.31 | [TK]D-Fender | JT thats not the way to run an * server for it... |
03:44.32 | JT | but you should :) the providers build their servers to handle the load |
03:44.49 | [TK]D-Fender | JT : thats amongst the "last ploys of the desperate". |
03:45.01 | [TK]D-Fender | JT Oh.. thats a LOAD alright! ;) |
03:45.07 | JT | well this is how softphones work by default |
03:45.15 | JT | you only need a REGISTER every minute or two |
03:45.30 | *** join/#asterisk michaelross (n=michael@203.59.123.167) |
03:45.59 | [TK]D-Fender | JT : thats one way, or the OPTIONS packet for QUALIFY. But again, this is what you do for more intermittant CLIENTS, not your SERVER |
03:46.23 | cr4z3d | ok ubuntu has no firewall by default but anyway i'll go ahead and forward that port ad see if that help |
03:47.34 | JT | [TK]D-Fender: if his IP is dynamic, he has to register anyway |
03:48.38 | cr4z3d | ok port is now forwarded |
03:48.47 | cr4z3d | should i just restart asterisk to be safe? |
03:49.10 | JT | it won't actually do anything |
03:49.13 | JT | restarting it |
03:49.21 | JT | since port forwarding is completely seperate |
03:49.33 | cr4z3d | true |
03:49.48 | cr4z3d | damn even the port forward i get the same error |
03:50.04 | cr4z3d | well not really an error but basically nufone saying nothing is connected |
03:50.17 | billzybub | hey is anyone fammiliar with TDMOE? |
03:50.25 | JT | billzybub: midly |
03:50.28 | JT | mildly |
03:50.48 | billzybub | time division multiplex over ethernet |
03:51.44 | billzybub | it kinda gives me the impression i could use an off the shell ethernet card as a t1 card and plug it straight from the dmarc? |
03:51.48 | *** join/#asterisk djs_2_6 (n=DJS@cpe-071-077-048-198.nc.res.rr.com) |
03:52.13 | JT | NO |
03:52.17 | JT | most definately not |
03:52.18 | *** join/#asterisk bmg505 (n=leon@196.209.249.12) |
03:52.27 | billzybub | doesnt it sound like it though? |
03:52.29 | JT | it allows TDM to be framed over ethernet |
03:52.37 | billzybub | i got real excited when i first read it |
03:52.42 | JT | it doesn't allow money to appear from nothing |
03:52.43 | JT | ;) |
03:52.50 | cr4z3d | haha |
03:53.11 | cr4z3d | so it does say registered |
03:53.28 | cr4z3d | what could other possible problems be that are stopping me from getting to my * server |
03:53.48 | billzybub | if you can frame doesnt it stand to reason that you should be able to send those frames without encapsulation? |
03:54.13 | JT | billzybub: what? |
03:54.27 | JT | billzybub: seriously, i have no idea what you're talking about now |
03:55.05 | billzybub | im kinda talking about an ethernet card taking the place of expensive t1 interfaces |
03:55.14 | *** join/#asterisk ManxPower (n=manxpowe@72.sub-70-196-33.myvzw.com) |
03:55.35 | billzybub | but i just smoked some crypler so maybe im not making much sense |
03:55.41 | JT | billzybub: i'm kind of suggesting your crazy |
03:55.44 | JT | yes |
03:55.53 | JT | totally different L1 and L2 |
03:56.03 | [TK]D-Fender | billzybub, YOU'RE CRAZY. |
03:56.24 | [TK]D-Fender | JT, there.. I SAID IT. You have real commitment iddues ;) |
03:56.28 | [TK]D-Fender | issues* |
03:56.50 | JT | har har |
03:57.09 | JT | hey, i have a winmodem..... |
03:57.13 | [TK]D-Fender | cr4z3d, ok, enable IAX2 debug and pastebun the failed inbound call attempt |
03:57.15 | [TK]D-Fender | ~pb |
03:57.27 | jbot | somebody said pb was a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
03:57.27 | JT | i need to terminate a t3 though |
03:57.30 | billzybub | i dare to dream |
03:58.02 | JT | that hurt :( |
03:58.04 | cr4z3d | nothing comes up when i call in |
03:58.09 | billzybub | seriously though, is it truly that inconceivable? |
03:58.16 | JT | yes |
03:58.18 | cr4z3d | oh wait i got some tx-frame retrys |
03:58.32 | JT | billzybub: is there a reason why you don't plug your keyboard into the power point? |
03:58.39 | JT | surely it's just a software issue |
03:58.46 | JT | keyboard over electricity |
03:59.01 | cr4z3d | ~pb |
03:59.12 | jbot | from memory, pb is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
03:59.48 | billzybub | why is that, no one ever thought for a second in 1980 that one day, not only would they be able to play pac-man on a home computer, but actually play the original game rom and every other game in their local arcade. |
04:00.19 | *** join/#asterisk rrrobert (n=rrobert@58-65-160-140.nayatel.pk) |
04:00.27 | JT | yes, but they also built computers to play pacman |
04:00.28 | cr4z3d | [TK]D-Fender, http://pastebin.ca/406120 looks like the same pattern keeps repeating every 30 seconds or so since i enabled iax2 debug |
04:00.50 | JT | they didn't just get the joystick of an arcade machine and hammer it together with a home tv set |
04:01.14 | billzybub | i could make a million analogies, mark my words, we will see it done |
04:01.51 | billzybub | i must get back to installing freepbx |
04:01.54 | JT | billzybub: you are saying one day we will be able to plug a T1 into a 10/100base-T ethernet card? |
04:02.07 | cr4z3d | haha that would be insane |
04:02.14 | JT | ~freepbx |
04:02.26 | jbot | [freepbx] unable to be supported here. It's a complex piece of dialplan and GUI, and can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
04:02.29 | billzybub | i am saying one day it will be possible, yes |
04:02.51 | JT | billzybub: you must be smoking something good to think it's possible, they're TOTALLY DIFFERENT |
04:02.57 | JT | you'd need a hybrid chipset |
04:03.04 | JT | it is not a software issue |
04:04.01 | billzybub | not so different, you just need to figure out how to hand the media off to the cpu |
04:04.10 | JT | no, the voltage levels are different |
04:04.13 | JT | the PINS are different |
04:04.13 | JT | the framing is different |
04:04.14 | *** join/#asterisk ManxPower (n=manxpowe@57.sub-70-196-73.myvzw.com) |
04:04.14 | JT | the clock is different |
04:04.18 | billzybub | most modern cards have auto detection for the pins, turn a cross over cable into straight through per see |
04:04.21 | cr4z3d | oh crap i gotta go.. [TK]D-Fender, if you can figure out what's wrog with that just pm it to me either way thanks for the help man |
04:04.30 | billzybub | clock can be handled by an external application |
04:04.41 | billzybub | framing can be handled by an external app |
04:04.50 | JT | it would be possible to build a card to do it |
04:04.55 | JT | not make a current one do it |
04:05.08 | billzybub | i bet you could hack the right ethernet card to do it |
04:05.13 | JT | and the reason it's unlikely anyone would built such a card is... WHERE ON EARTH IS THE DEMAND? |
04:05.17 | JT | no demand = no product |
04:05.33 | JT | i bet you've been smoking too much |
04:05.55 | billzybub | your right, as the world becomes more digital the demand would go down |
04:06.04 | JT | the demand was never there |
04:06.13 | billzybub | whats the point in voip really if your still using trunks |
04:06.18 | JT | people with T1s/E1s = not poor people |
04:06.34 | JT | what's wrong with trunks? |
04:06.41 | JT | and the point is obvious |
04:07.09 | *** join/#asterisk Belize (n=vasya@80.237.99.222) |
04:07.56 | billzybub | oh i disagree, spent the last 6 years of my life as a director in a marketing organizations, we had call centers ive provisioned many a t1 for use in NEC NEAX and 2000 series phone switches |
04:08.35 | billzybub | we used pri and t's (d4-ami) |
04:08.54 | billzybub | t1 cards would go bad on occasion |
04:09.06 | JT | sure, that's possible |
04:09.25 | billzybub | so why would you say the demand was never there ? |
04:10.09 | JT | it means the demand is for a replacement card or a failover solution, not a crackwhore solution of a mutant ethernet + pri card which solves no real problem |
04:10.34 | billzybub | its not a crackwhore solution, it would be fun to work on |
04:10.59 | JT | fun, that's about it, it doesn't solve a real issue |
04:11.01 | billzybub | do you have any idea what big iron t1/e1/pri interface cards cost?! |
04:11.10 | JT | yes i bet they cost a bit |
04:11.17 | JT | so why would someone make you a cheap one? |
04:11.20 | JT | defies logic |
04:11.28 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
04:11.29 | JT | things are cheaper when mass produced too |
04:11.41 | billzybub | it doesnt , i think it would be such an increadible boost for asterisk |
04:11.43 | JT | the demand is obviously not that high for pri cards |
04:12.03 | JT | only if someone made one for cheap, and there is no reason for any company to do so |
04:12.16 | billzybub | iwhat if you can hack one together for cheap |
04:12.27 | billzybub | thats the point i trying to make, dont be such a nay sayer |
04:12.28 | JT | then when it fails you get fired, good idea. |
04:12.34 | JT | i'm being realistic |
04:12.40 | JT | you're being fanciful :) |
04:12.41 | billzybub | your being boring |
04:12.45 | ManxPower | Uh, what are we talking about? |
04:13.00 | JT | T1s aren't meant to be a fun exciting party trick |
04:13.08 | ManxPower | sounds like billzybub is not an Asterisk True Believer |
04:13.18 | JT | ManxPower: billzybub things someone should make a driver to make an ethernet card work as a T1/pri card |
04:13.23 | JT | s/things/thinks/ |
04:13.28 | *** join/#asterisk Strom_M (n=strom@12-189-87-2.att-inc.com) |
04:13.30 | billzybub | im and talking about cost effectively hacking your off the shelf eathernet card to replace expensive t1/e1 interfaces |
04:13.35 | ManxPower | Fun and exciting T-1s get you fired. |
04:13.40 | JT | exactly |
04:13.47 | billzybub | jobt they've already done half the work with tdmoe |
04:14.01 | ManxPower | JT: Ah. How fanciful |
04:14.03 | JT | no, i keep telling you it's totally different |
04:14.20 | Strom_M | no, quite obviously it's exactly the same |
04:14.27 | Qwell | clearly |
04:14.29 | JT | :) |
04:14.29 | Strom_M | </throwing shit into the fire> |
04:14.32 | ManxPower | JT: For one thing T-1 voltages would blow an ethernet card |
04:14.32 | acidchild | whats a good asterisk howto? |
04:14.41 | JT | ManxPower: that's what i was trying to say |
04:14.47 | ManxPower | acidchild: The Book |
04:14.48 | ManxPower | ~book |
04:14.50 | jbot | extra, extra, read all about it, book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
04:14.50 | JT | apparently it's only a software issue |
04:14.58 | acidchild | thanks. |
04:15.17 | billzybub | manpower, ive plugged t1 from the dmarc into and ithernet card with no bad side effects |
04:15.27 | acidchild | pdf's blow ;/ |
04:15.30 | ManxPower | JT: Ever had Polycom phones lose their call forwarding settings? |
04:15.34 | JerJer | acidchild: i am slightly biased but here's one: http://tinyurl.com/2aal8h |
04:15.37 | JT | acidchild: free books don't though |
04:15.43 | JT | ManxPower: nup |
04:16.02 | Qwell | JerJer: You write docs now? :P |
04:16.28 | [TK]D-Fender | billzybub, Congratulations, no giant mushroom cloud. But will it do anything USEFUL? |
04:16.36 | ManxPower | Property Services got REALLY PISSED that their forwarding to the answering service failed 2 nights in a row. I had to write a damn dialplan based call forwarding for them |
04:16.49 | [TK]D-Fender | billzybub, If you think the CARDS are expensive now, who has a PRI at HOME!? |
04:17.06 | JerJer | Qwell: guess so |
04:17.11 | JT | i have a T1 at home |
04:17.12 | [TK]D-Fender | JerJer, Get me some crack while you're at it! |
04:17.16 | JT | but it's 3 metres long |
04:17.30 | JerJer | crack is a waste of good cocain yo |
04:17.37 | billzybub | im not talking about home use |
04:17.39 | [TK]D-Fender | JT and NO a channel bank does NOT count :p |
04:17.41 | ManxPower | The thing about DSP-less T-1/PRI cards these days is that they cost about about the same as 1 - 2 months of T-1/PRI service. |
04:17.45 | JT | [TK]D-Fender: :( |
04:17.48 | ManxPower | Anyone that can get a PRI can afford a card |
04:17.49 | billzybub | silly, who would want 24 lines in their house |
04:17.49 | *** join/#asterisk HaMYaI (i=HaMYaI@202.8.86.162) |
04:17.50 | [TK]D-Fender | billzybub, So whats the big deal of 50)$ for a PRI card? |
04:17.56 | [TK]D-Fender | $500 |
04:18.28 | [TK]D-Fender | billzybub, You're jsut here thinking that it should cost 25$ for the card. |
04:18.47 | [TK]D-Fender | billzybub, Like the dev time and sales volume would warrant such a price |
04:18.58 | Qwell | Any of you guys ever done RAID0+10? heh |
04:18.58 | billzybub | well, it does stand to reason that should somone come up with the solution it wouldnt be very good for Digium would it? |
04:19.11 | ManxPower | If DSPless T-1/E-1 cards had the VOLUME of ethernet cards they would cost the same as ethernet cards |
04:19.22 | billzybub | you think? |
04:19.36 | ManxPower | billzybub: MANY people have tried to make cards significantly less that Digium. They have all failed. |
04:19.38 | JT | it's obvious |
04:19.42 | JT | supply and demand |
04:19.58 | JT | Qwell: O+10? :o |
04:20.01 | Qwell | sure |
04:20.09 | ManxPower | Some people would say there are BETTER cards than Digiums -- but they still cost about the same |
04:20.11 | JT | don't you mean raid 10? |
04:20.21 | [TK]D-Fender | billzybub, If someone came up with such a think it would KILL Digium. Diguim is in the business of making cards. if someone undercut them that much they'd SINK. |
04:20.21 | Qwell | no, I mean RAID 0+10 |
04:20.32 | JT | Qwell: define 0+10 :) |
04:20.49 | Qwell | 0+1, with an extra 0 :p |
04:21.10 | ManxPower | [TK]D-Fender: that would suck for Asterisk. 8-| |
04:21.16 | JT | well RAID10 is a 2 * RAID0 |
04:21.29 | JT | 0+10 would be that plus another raid 0? |
04:21.44 | Qwell | yes |
04:21.53 | Qwell | but at the lowest level |
04:22.05 | [TK]D-Fender | ManxPower, Yes it would. Whicle the project could live on, the loss of paid full-time programmers would be a serious blow |
04:22.06 | *** part/#asterisk mtaht4 (n=m@c-71-198-23-124.hsd1.ca.comcast.net) |
04:22.06 | Qwell | so, 8 drives |
04:22.12 | Qwell | striped, redundant, and striped again |
04:22.25 | JT | seems pretty pointless |
04:22.30 | Qwell | not really :D |
04:22.39 | [TK]D-Fender | RAID 3 - For that "plaid" feeling :) |
04:22.42 | JT | but it is |
04:23.01 | JT | no real gain for money better spent on backup or clustering/failover |
04:23.19 | Qwell | but I want uber-fast :p |
04:23.30 | Qwell | ...except on writes |
04:23.44 | billzybub | do you really think it would crush asterisk? I dont, asterisk got very far before digium started making money, if theyve actually made any money yet. You dont think the development community is strong enough to support asterisk on it own - without asterisk? |
04:23.47 | DocHolliday | too bad asterisk cant do failover without dropping calls :| |
04:24.13 | JT | nothing this side of $20k can |
04:24.24 | [TK]D-Fender | billzybub, It would seriously hamper growth, yes. |
04:24.37 | billzybub | man theres some crazy thing on discovery hd right now |
04:24.49 | JT | Qwell: what you're refering to is RAID100 btw |
04:24.58 | Qwell | JT: nope, different levels of strip |
04:24.59 | Qwell | stripe* |
04:25.03 | [TK]D-Fender | Sounds more like HAL 9000 to me.... |
04:25.18 | Qwell | raid100 is a striped striped set of redundant disks |
04:25.19 | JT | Qwell: what actually supports this mode? |
04:25.22 | billzybub | id be interesting to see digiums numbers |
04:25.23 | Qwell | nothing |
04:25.28 | JT | nice! :D |
04:25.50 | Qwell | It's just software though :D |
04:25.57 | [TK]D-Fender | Qwell : Roughly how many drones do you guys keep on staff these days? |
04:26.02 | Qwell | 90+ |
04:26.07 | ManxPower | billzybub: Digium employs almost all of the Asterisk core programming team |
04:26.17 | [TK]D-Fender | billzybub, and thats just the DRONES. Digium is not a "tiny" company |
04:26.26 | Qwell | oh, "those" drones |
04:26.26 | Qwell | no |
04:26.36 | Qwell | significantly less than that |
04:26.48 | [TK]D-Fender | qwell : No, I'm not referring to your pre-patch chan_skinny botnet ;) |
04:26.50 | Qwell | that's the entire company |
04:27.06 | Qwell | don't quote me on that though |
04:27.17 | billzybub | anyone use beryl? |
04:27.19 | *** join/#asterisk viking78 (i=aherbert@66-168-102-94.dhcp.jcsn.tn.charter.com) |
04:27.36 | [TK]D-Fender | billzybub, "<Qwell> that's the entire company <Qwell> don't quote me on that though" |
04:27.57 | billzybub | eh? |
04:27.58 | [TK]D-Fender | billzybub, You are just ALL "bling" aren't you? |
04:28.08 | billzybub | what do you mean? |
04:28.10 | [TK]D-Fender | </sarcasm> |
04:28.11 | [TK]D-Fender | :D |
04:29.31 | [TK]D-Fender | qwell : c'mon, that was witty, and unexpected! And this medium robs of my Keystone Cops-style of hunour.. |
04:30.42 | Qwell | No, witty is r131 of the aadk repository :P |
04:31.12 | Qwell | Add a comment for clarification, to explain where this file is generated. |
04:31.12 | Qwell | It isn't where you'd expect. |
04:31.12 | Qwell | (perhaps unexpectedly, it is not the Spanish Inquisition that does it) |
04:31.42 | *** join/#asterisk djs_2_6 (n=DJS@cpe-071-077-048-198.nc.res.rr.com) |
04:32.03 | file | Qwell is so mean |
04:32.07 | Qwell | indeed |
04:32.24 | Qwell | I would try to push JerJer over too, but...yeah |
04:32.31 | JerJer | yeah |
04:32.35 | Qwell | I think we all know how well *THAT* would work |
04:32.43 | *** join/#asterisk ShadowHntr (i=sentinel@wikipedia/Shadowhntr) |
04:34.30 | [TK]D-Fender | JerJer, Imposing, are we? ;) |
04:34.56 | JerJer | more like huge |
04:35.01 | Qwell | [TK]D-Fender: the ratio is...very low |
04:35.30 | [TK]D-Fender | Qwell : ... I'm afraid to ask WHICH..... |
04:35.35 | Qwell | he's probably > a foot taller than me :P |
04:35.40 | [TK]D-Fender | qwell : Although a few come to mind.... |
04:36.52 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
04:36.54 | [TK]D-Fender | I'm nearing the break-point before pumping serious iron again.... floating at 190lbs currently @ 6'2-3" |
04:37.19 | JerJer | i'm 375 and 6'5" |
04:37.20 | DocHolliday | [TK]D-Fender, i got a personal trainer to gain weight :P |
04:37.36 | Qwell | > 1 foot, and > 3x weight... |
04:37.57 | [TK]D-Fender | JerJer, ouch |
04:38.18 | Qwell | 5'3, 115 - what now? :P |
04:38.26 | [TK]D-Fender | JerJer, You are at some serious risk there.... |
04:39.29 | JerJer | I was almost 400 |
04:39.42 | [TK]D-Fender | JerJer, I hit a point about 5 years ago when I said "enough" when I was just risen back to my high of 265. In 5 months I hit 195. |
04:40.24 | *** join/#asterisk kb1_kanobe (i=user@d207-216-143-5.bchsia.telus.net) |
04:40.30 | [TK]D-Fender | then 2 years later at the start of summer I freakish dropped from around 195-200 to 176 :| |
04:40.52 | [TK]D-Fender | out of thin air... never figured quite what the trigger was... |
04:41.19 | [TK]D-Fender | I'm doing decent where I'm at and jsut need to get off my ass and actually WORK for my goals. |
04:42.16 | *** join/#asterisk gammah (n=gammah@cpe-66-69-224-62.austin.res.rr.com) |
04:43.43 | billzybub | my wireless network card is giving me headaches |
04:44.47 | vader-- | tk thats kinda my goal |
04:44.54 | vader-- | last year i dropped from 250 down to 219 |
04:45.00 | vader-- | and now im back up to around 235 |
04:45.05 | billzybub | ive got one of those linksys wireless g cards that you need to run the firmware cutter on to extract the driver and its flakey at best |
04:45.08 | [TK]D-Fender | control* |
04:45.09 | [TK]D-Fender | ashdasasdf |
04:45.20 | vader-- | i started drinking coffee |
04:45.20 | [TK]D-Fender | aphasia setting in full.. its getting late... |
04:45.28 | vader-- | and my coffee is 800 calories |
04:45.30 | vader-- | it's killing me |
04:46.06 | billzybub | vader can you sleep after you drink coffee? |
04:46.09 | [TK]D-Fender | vader--, Mine is NEGATIVE. over the last 2 months I've switched to blak and only 1 artificail sweetener, and have gone pure-black on occasion |
04:46.10 | vader-- | ya |
04:46.18 | billzybub | wish i could |
04:46.24 | vader-- | i drink it in the morning |
04:46.25 | [TK]D-Fender | vader--, the trick is repeatability and portions... |
04:46.26 | billzybub | i cant sleep for sh1t |
04:46.28 | vader-- | one 24oz cup |
04:46.29 | vader-- | thats is |
04:46.35 | *** part/#asterisk kb1_kanobe (i=user@d207-216-143-5.bchsia.telus.net) |
04:46.44 | vader-- | tkd ya i lost all that weight before by cutting all soda and eating right |
04:46.45 | [TK]D-Fender | You need to knock off the EVIL shit though... |
04:46.46 | billzybub | sleep is a big part of weight loss though |
04:47.00 | vader-- | well the problem was i always felt fatigued |
04:47.08 | vader-- | with coffee i don't feel fatigued |
04:47.12 | vader-- | i feel how i should feel |
04:47.16 | vader-- | but im gaining soo much weight |
04:47.35 | Juggie | dont put sugar in it |
04:47.42 | vader-- | ya i put 4-5 packets |
04:47.46 | [TK]D-Fender | vader : meanwhile the insulin spikes you're getting hit with plus the caffeine are wreakiong havoc |
04:47.48 | vader-- | i have to learn to drink it black |
04:47.49 | Juggie | well... there you go, jeeze. |
04:47.55 | Juggie | use splenda. |
04:47.57 | billzybub | i got that virus thing from mono and never got rid of it the doctors tell me, causes all kinds of problems one of them being insomnia |
04:48.06 | [TK]D-Fender | Juggie, for 800 cal trust me its not the sugar HE's adding |
04:48.13 | Juggie | splenda rox. |
04:48.14 | vader-- | splenda gives me headaches |
04:48.14 | CrashHD | what makes asterisk iniate the voicemail box setup greeting? |
04:48.26 | CrashHD | *initiate |
04:48.31 | Juggie | you could just go to the gym and get in shape |
04:48.33 | [TK]D-Fender | vader--, Actually... I'm beginning to think I'm experiencing the splenda bit myself... |
04:48.35 | billzybub | crash the voicemail program |
04:48.36 | vader-- | im drinking dunkin donuts french vanilla |
04:48.37 | vader-- | XL |
04:48.38 | Juggie | that helps having more energy. |
04:48.43 | vader-- | extra cream, extra sugar |
04:48.45 | [TK]D-Fender | Juggie, I lost my 70lbs doing jack shit. |
04:49.01 | [TK]D-Fender | Juggie, Like the law of programming says. GIGO <- |
04:49.03 | Juggie | well, yah, if you are 70pounds overweight you can loose it by just not eating. |
04:49.14 | CrashHD | billzybub: it asked to record the directory name on a box, I had never heard that. what is the indicator it uses to actually determine that? |
04:49.15 | JerJer | not if you like to eat |
04:49.18 | vader-- | tkd i couldn't figure out why i was getting these headaches when all i was drinking was flavored water |
04:49.24 | vader-- | here the water was flavored with splenda |
04:49.28 | Juggie | myself, i go to the gym 4-5 times a week. |
04:49.29 | JerJer | i get crazy headaches if i don't eat |
04:49.32 | [TK]D-Fender | JerJer, You can. You just need to change WHAT you eat and when. |
04:49.36 | billzybub | crash, ask me that again but in a different way |
04:49.37 | vader-- | i cut that stuff out and headaches went away |
04:49.56 | CrashHD | well |
04:50.00 | CrashHD | I have a voicemail box |
04:50.05 | JerJer | [TK]D-Fender: oh yeah - i eat 4-5 small meals a day |
04:50.11 | [TK]D-Fender | vader--, I started getting them more when I cut the creme from my coffee |
04:50.15 | JerJer | and 30-45 minutes carido every day |
04:50.22 | CrashHD | that is constantly telling the user to setup the box, resetting password, setting directory recording |
04:50.23 | JT | losing weight by not eating is STUPID |
04:50.27 | [TK]D-Fender | JerJer, Sounds like you're on the right track. |
04:50.30 | JerJer | 5-6 glasses of water |
04:50.34 | billzybub | alot of people think thats the key to losing weight, many small meals so your never hungry and never binge |
04:50.34 | JT | you lose muscle mass as well as fat |
04:50.37 | Juggie | yah drink lots of liquid. |
04:50.43 | JT | which is bad for you |
04:50.45 | [TK]D-Fender | JT I never advocated NOT eating. just change what & when. |
04:50.55 | CrashHD | I'm trying to figure out why |
04:51.01 | CrashHD | and determine what I can do about it |
04:51.22 | Juggie | if water bores you, drink crystal light. |
04:51.29 | [TK]D-Fender | I lost 10 -15 since taking up eating 5 meals a day, bringing fruit to work, and having a smoothie at night. keeps things even through the day |
04:51.29 | JT | JerJer: what's in the meals though? |
04:51.57 | Juggie | another thing is to NEVER EVER EVER eat late |
04:52.01 | Juggie | dont ever eat later then like 6-7pm |
04:52.02 | [TK]D-Fender | JT : If you are in an unhealthy state, you can afford a little muscle loss for the larger gaint o be had. |
04:52.06 | billzybub | crash, i followed the tuorial here: http://www.asteriskguru.com/tutorials/asterisk_voicemail.html , works for me |
04:52.15 | JerJer | JT: chicken, fish some beef |
04:52.30 | JT | [TK]D-Fender: with the right diet, you can lose fat and gain muscle at the same time |
04:52.36 | JT | JerJer: good good |
04:52.44 | JerJer | minmial carbs, but i cannot cut them out all the way |
04:52.51 | Juggie | if you are doing exercise you have to loose 17 pounds to loose 15. |
04:52.58 | Juggie | thats what i've been told. |
04:53.02 | JT | JerJer: heard of carb cycling? |
04:53.15 | Juggie | because if you burn 17 pounds of fat, your going to add muscle. |
04:53.17 | ManxPower | [TK]D-Fender: I believe that eating many small meals helps. |
04:53.20 | billzybub | everyone i know who did the atkins diet lost alot of weight. But they also gained it all back :( |
04:53.28 | [TK]D-Fender | JT You can, but that does require a fair amount of work. Muscle mass takes a lot of work to gain. Its a qustion of whre you are in the phase of development. If you are simply coming DOWN a LOT then put your focus on the easy bit first |
04:53.34 | JT | Juggie: depends how you burn it |
04:53.38 | Juggie | billzybub, i did that diet, lost 25 pounds and gained back about half. |
04:53.46 | [TK]D-Fender | ManxPower, Indeed, I've subscribed to that for some time. |
04:54.01 | JT | [TK]D-Fender: you can reliably lose a lot of weight with carb cycling |
04:54.02 | Juggie | i am 210 now (down from 225 @ christmas) |
04:54.05 | JT | 2 days a week no carbs |
04:54.11 | Juggie | and my goal is 190. which is a long way off. |
04:54.13 | [TK]D-Fender | Atkins has a GREAT theory.. if it doesn't KILL you (like its founder) |
04:54.15 | billzybub | i think losing weight is a real lifestyle commitment and it takes an enormous amount of will power |
04:54.15 | JT | 2 days low |
04:54.26 | Juggie | [TK]D-Fender, he slipped and fell on ice. |
04:54.30 | JT | 2 days carbs ok |
04:54.33 | Juggie | but i truly do believe less carbs are better. |
04:54.36 | ManxPower | [TK]D-Fender: I have known VERY few overweight people that eat many small meals |
04:54.39 | Juggie | but you cant remove them all together. |
04:54.58 | JT | i'm talki8ng about for dieting |
04:55.07 | [TK]D-Fender | Lets say "no refined sugar". no "crap" fats. no junk like chips, canies, etc. |
04:55.13 | billzybub | when i was a kid i was always fat, in my twenties i starved myself and became anorexic for 5 years, now im fat again |
04:55.17 | [TK]D-Fender | Nobody ever got fat on fruits & veggies. |
04:55.21 | vader-- | tk do you have a 6 pack? |
04:55.28 | JT | some saturated fats are good for you |
04:55.33 | JT | just not too much |
04:55.40 | billzybub | chocolate covered strawberries |
04:55.44 | ManxPower | I suppose y'all will hate me if I mention that I eat mostly fat and sugar |
04:55.55 | billzybub | fat is good for you |
04:55.58 | JT | fat is not the enemy |
04:55.59 | Juggie | ManxPower, i dispise people like you, screw you and your high metabolism :) |
04:56.01 | [TK]D-Fender | Turkey, Salmon, porc are all great meats these days. Yogurt is a great add-on. Donane Silhoette is only 40cal per ..... |
04:56.01 | JT | carbs are :P |
04:56.01 | billzybub | :D |
04:56.08 | [TK]D-Fender | Danone* |
04:56.15 | ManxPower | Juggie: I eat many small meals. |
04:56.19 | [TK]D-Fender | vader--, Sure.. in the back of my fridge :) |
04:56.29 | JT | nobody ever got smart on fruits and vegies |
04:56.35 | [TK]D-Fender | vader--, Though technically I'm close to beginning the real work on it :) |
04:56.53 | JunK-Y | [TK]D-Fender: now get a gf to get good foods and for cooking ur meals :P |
04:56.54 | vader-- | just curious because im the same height |
04:57.00 | billzybub | you need fat, even small amounts in your stomach makes you feel satiated. |
04:57.06 | vader-- | and i was wondering what would be showing if i got down to your weight |
04:57.10 | JT | yeah, that and protein |
04:57.19 | [TK]D-Fender | JT Both are the enemy for most people, especially TOGETHER. Hece the origin of the De Montignac diet |
04:57.37 | [TK]D-Fender | JT and is a basis for Atkis as well. |
04:57.37 | ManxPower | If I don't eat I turn into a total asshole, get a headache and start to sweat. |
04:57.55 | [TK]D-Fender | vader--, a 6-pack is a LOT of hard work. thats typically the last to go as an endomorph. |
04:57.58 | Qwell | I subscribe to the one meal a day philosophy :D |
04:58.08 | billzybub | ive done the south beach diet, it works very well, kinda like a light version of atkins but the food prep is insanely time consuming |
04:58.12 | [TK]D-Fender | ManxPower, How would we tell? ;) |
04:58.14 | JT | [TK]D-Fender: fruits and vegies? |
04:58.16 | [TK]D-Fender | (J/K) |
04:58.26 | JunK-Y | better: eat an apple a day and keep the doctor away. |
04:58.33 | ManxPower | [TK]D-Fender: It is a good thing I like you. 8-) |
04:58.33 | [TK]D-Fender | sorry.... can't ...resist ... temptation ;) |
04:58.34 | JT | most diets with stupid names are retarded :P |
04:58.34 | vader-- | endomorph? |
04:58.46 | JT | nothing worse than chick diets |
04:58.56 | Juggie | the worst thing you can do is snack on bad food. |
04:58.58 | ManxPower | vader--: The Wiki is your friend |
04:59.06 | Qwell | JT: what about that nympho diet..? |
04:59.06 | Juggie | so one thing i do is keep good food accessible. |
04:59.11 | [TK]D-Fender | JT indeed.... thats why I apply a lot of common sense and the known bits around, and control the portions and time for the rest. |
04:59.23 | Juggie | eg, pieces of chicken pre cooked and cut up in the fridge. |
04:59.29 | [TK]D-Fender | vader--, those who put weight on around the middle. |
04:59.39 | JT | Juggie: freshly cooked is better for you |
04:59.56 | JT | when you cook food with fats and then leave it, the fats become more toxic to the body |
04:59.57 | Juggie | JT, how does a piece of cooked chicken change |
04:59.58 | [TK]D-Fender | Juggie, Core in my method is making sure that food is NEVER complicated or time consuming. |
05:00.01 | JT | hearder to break down |
05:00.14 | JT | the fats go "rancid" for want of a better word |
05:00.19 | Juggie | JT, but it hasnt changed. |
05:00.23 | Juggie | there is 0 fat in chicken |
05:00.24 | vader-- | see im always eating out im never home |
05:00.27 | JT | it does change |
05:00.29 | JT | lol 0 fat |
05:00.34 | vader-- | i live between my parents and my gf's parents house |
05:00.38 | [TK]D-Fender | Juggie, thats what kills so many diet plans for people. the effort. My foods are all so damned easy to wrok with and you don't have to eat every meal well-rounded so long as it evens out. |
05:00.49 | vader-- | so i need to find food that i can eat out that is healthy |
05:00.56 | billzybub | i like peanut chews and recees peanut butter cups at about 3 in the morning |
05:00.58 | JT | vader--: meat, lots of it :) |
05:01.00 | Juggie | [TK]D-Fender, my suppers in the evening, are usually wraps. |
05:01.15 | Juggie | like a wrap with some meat, lettuce, peppers, hot peppers, and mustard. |
05:01.16 | [TK]D-Fender | So for dinner home at 5pm I'll just doup some steak or salmon. when I come home from the gym I'll have a smoothie. |
05:01.19 | vader-- | jt by eat out i don't mean resturants |
05:01.29 | vader-- | i mean like fast food places or wawa, 7-11 joints |
05:01.30 | JT | vader--: i know, it could be anywhere |
05:01.44 | DocHolliday | i made steak and pasta tonight :) |
05:01.50 | JT | meat is good, |
05:01.59 | JT | bread rice and pasta conspire to make you fat :) |
05:02.15 | [TK]D-Fender | Juggie, Wraps can be good or bad depending. be careful about the wrap itself, but I have considered writing that off and super stuffing them with just ssalad greens, tomatoe, etc. Minimal blasamic dressing. |
05:02.32 | Juggie | its a spinach high fiber wrap. |
05:02.34 | JT | that's like a lump of carbs soaked in fats that are no longer fresh (most of the time) |
05:02.36 | [TK]D-Fender | JunK-Y, Tabarnac retourn-toi donc a ton maudite poutine! |
05:02.40 | JT | but i still eat chips every so often :P |
05:02.45 | billzybub | omg |
05:02.48 | Juggie | there are also some low carb wraps you can get, but last time i went to the supermarket they were out |
05:02.58 | JunK-Y | poutine hummmm |
05:03.06 | [TK]D-Fender | JT I HAVE completely cut out bread from my home.... |
05:03.24 | flenders | there's also low carb beer |
05:03.27 | [TK]D-Fender | Juggie, Best is to make it yourself.... |
05:03.43 | Juggie | [TK]D-Fender, the only bread i buy is low carb |
05:03.55 | Juggie | and even so i use it very infrequentally. |
05:04.04 | JT | isn't low carb bread like saying low-bread bread? :P |
05:04.09 | [TK]D-Fender | ;) |
05:04.27 | [TK]D-Fender | JT, I got a pouch of dehydrated water... now what do I do with it ;) |
05:04.33 | JT | ;) |
05:04.41 | JT | it's a very fine powder isn't it? |
05:04.42 | billzybub | this cafe near me make a mean cob wrap with romain, avacado, bacon, chicken and other goodies that escape me at the moment |
05:06.31 | [TK]D-Fender | JT, You're right..... I should mail it to the gov't for analysis... unmarked to IRS right? :) |
05:06.49 | JT | right |
05:07.08 | k-man_ | [TK]D-Fender, have you tried rehydrating it? |
05:09.18 | [TK]D-Fender | Another GREAT guilt-free filler/dessert : Jell-o fat free pudding. 30 cal, VERY filling, and a super protein/carb ratio |
05:10.00 | [TK]D-Fender | and I bought 7lbs of bananas today :) |
05:10.23 | DocHolliday | [TK]D-Fender, im just glad i am trying to increase my weight, the decreasing thing would kill me |
05:10.39 | [TK]D-Fender | CunningPike, Don't worry.. I'm still anal-retentive ;) |
05:10.46 | CunningPike | [TK]D-Fender: lol |
05:10.59 | k-man_ | jt, so how was the meeting last night? |
05:11.06 | [TK]D-Fender | load chan_selfdeprecation.so |
05:11.12 | JT | http://www.mindandmuscle.net/articles/jonathon_fass/carb_cycling |
05:11.14 | k-man_ | jt, also what is the mailing list i should join? |
05:11.24 | JT | k-man_: still have the url i sent? |
05:11.31 | JT | k-man_: that has a link to the list |
05:11.32 | k-man_ | no |
05:11.33 | [TK]D-Fender | DocHolliday, Want to lose weight? Eat 6 meals a day. Want to gain weight? six meals a day. |
05:11.44 | [TK]D-Fender | DocHolliday, Lessons learned from my personal trainers. |
05:12.12 | ManxPower | Well, walked across it at least from the car is parked to where I'm staying. |
05:12.19 | DocHolliday | [TK]D-Fender, haha |
05:12.38 | DocHolliday | [TK]D-Fender, i eat a lot of stuff.. its just not the right kind of food |
05:13.14 | [TK]D-Fender | DocHolliday, If you're that low, have they put you on Boost yet? |
05:13.32 | ManxPower | DocHolliday: No matter how many calories you consume, you still can't gain weight. I can understand that. |
05:13.39 | [TK]D-Fender | DocHolliday, And weight training sounds like a good idea... you'll need a base to build on... |
05:13.50 | ManxPower | I'll sometimes eat 2,000 calories at one meal. |
05:13.51 | JT | k-man_: http://lists.openvoip.org.au/cgi-bin/mailman/listinfo/openvoip |
05:14.06 | billzybub | heh |
05:14.15 | billzybub | i can do that with one cheese steak |
05:14.18 | toombaloomba | man thats the first time I scroll up and theres pages of non-asterisk talk |
05:14.30 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
05:14.31 | *** mode/#asterisk [+o russellb] by ChanServ |
05:14.42 | k-man_ | jt, thanks |
05:14.49 | JT | i have never had a cheese steak |
05:14.54 | JT | not something we see over here |
05:14.56 | billzybub | really? |
05:14.57 | k-man_ | jt, so how was it anyway? |
05:15.00 | billzybub | where ya from? |
05:15.05 | JT | australia |
05:15.06 | Qwell | it's a russellb |
05:15.11 | k-man_ | jt, no... does cheese and steak go well together do you think? |
05:15.20 | billzybub | welp, if you ever co to philadelphia, check out jims |
05:15.22 | JT | k-man_: not too bad, had a good chat at the pub afterwards |
05:15.23 | DocHolliday | [TK]D-Fender, yeah i see the trainer twice a week and i go once on my own |
05:15.38 | JT | k-man_: sounds like a good carb free source of protein ;) |
05:15.44 | [TK]D-Fender | DocHolliday, What are your weight & height currently? |
05:15.51 | billzybub | shit dide, you got them funnel web spiders near where you live? |
05:15.56 | JT | yes |
05:16.03 | billzybub | omg |
05:16.06 | k-man_ | jt, was the talk any good? |
05:16.10 | [TK]D-Fender | billzybub, We're talking Philly-chese steak here, right? |
05:16.10 | billzybub | i couldnt sleep |
05:16.12 | JT | i burn them with blowtorches |
05:16.17 | billzybub | yeah |
05:16.27 | JT | but they're not a big worry really |
05:16.31 | [TK]D-Fender | billzybub, Yeah... AWESOME. EVIL, but still awesome |
05:16.36 | JT | generally you have to aggrevate them to get bitten |
05:16.50 | billzybub | nah, theyll only turn your lungs into bloody jelly in about 5 minutes |
05:16.53 | JT | k-man_: yeah, interesting to see where voicetronix was going |
05:16.59 | JT | it was the ceo who did the talk |
05:17.04 | JT | talked to him at the pub too |
05:17.06 | JT | nice guy |
05:17.35 | k-man_ | ah cool |
05:17.36 | JT | billzybub: what turns your lungs into bloody jelly? |
05:17.47 | billzybub | the venom from the funnel web |
05:17.48 | k-man_ | so what are they planning to do with their products jt? |
05:20.02 | russellb | file: how are you, sir? |
05:20.25 | Qwell | russellb: got any good toys from VON yet? :D |
05:20.38 | file | russellb: sleepy... and headache-like... but meh |
05:20.41 | file | russellb: you?!? |
05:20.55 | russellb | Qwell: haven't collected any schwag, no ... |
05:20.58 | russellb | file: very tired |
05:21.02 | russellb | very sore |
05:21.19 | denon | you know, I think vendors should just load bags for attendees .. and give them to you at the door |
05:21.26 | denon | bags full of everyone's stuff |
05:21.31 | russellb | agreed |
05:21.33 | denon | so you dont have to do all the work |
05:21.49 | denon | they could just deliver them to your car for you |
05:22.05 | russellb | except it would be a lot more expensive |
05:22.05 | russellb | because people generally only want to give their stuff to people that are interested |
05:22.07 | denon | or fedex the junk right to your office |
05:22.21 | denon | russellb: yeah, I know -- but we all pretend to be interested at the vendors with truly cool stuff |
05:22.23 | denon | and it always works |
05:22.32 | [TK]D-Fender | JT : thanks for the link... good read |
05:22.44 | denon | Ive got a handful of free PC hardware to prove it |
05:22.47 | JT | np |
05:23.06 | russellb | yup |
05:23.07 | JT | http://www.mindandmuscle.net/articles/anssi_manninen/low_carb_diets |
05:23.07 | JT | http://www.mindandmuscle.net/articles/twin_peak/carbohydrate_cycling |
05:23.11 | JT | there's a couple more |
05:23.19 | denon | russellb: well, that, and the people with good food |
05:23.24 | Qwell | denon: oh so true... |
05:23.25 | denon | ice cream stuff always goes over well on a hot day |
05:23.35 | Qwell | the pretending to be interested part :D |
05:24.01 | denon | Qwell: and then you let em scan your badge, with the phony DID and PO box |
05:24.07 | Qwell | I don't ;) |
05:24.09 | denon | well, not a phony DID - just a DID that terminates to voicemail |
05:24.11 | denon | no? |
05:24.20 | denon | I usually set up an alias that I remove after words |
05:24.21 | denon | and a temp did |
05:24.33 | denon | and I hand out my biz card to anyone I really care about |
05:24.41 | denon | "they got my info wrong on the badge .. here. use my card.. " |
05:25.01 | Qwell | I somehow always end up with cards... |
05:25.16 | denon | they sneak em into the bags |
05:25.19 | russellb | i intentionally lose cards |
05:25.27 | Qwell | well, by people handing them to me, I mean |
05:25.30 | Qwell | not vendors either |
05:25.30 | DocHolliday | russellb, ouch |
05:25.31 | denon | salespeople always think that by giving you contact info, you're going to want to call them |
05:25.35 | russellb | :-p |
05:25.43 | denon | ah |
05:25.47 | russellb | not really. |
05:25.54 | DocHolliday | denon, i have seen someone find their own business card on the ground once, wasnt pretty |
05:26.02 | denon | haha |
05:26.07 | Qwell | russellb: wait until after the show to admit that ;) |
05:26.11 | russellb | oh, right |
05:26.18 | russellb | it was just a joke anyway :-p |
05:26.19 | denon | Ive only given my card to vendors I actually plan to do stuff with |
05:26.33 | file | have your people talk to my people |
05:26.40 | denon | file has people? |
05:26.43 | denon | oh, at the bakery |
05:26.44 | Qwell | brb |
05:26.45 | file | denon: do I ever! |
05:27.07 | DocHolliday | denon, last weekend someone asked me for a card and i didnt have any.. used infrared to send my mobile card :P |
05:27.07 | denon | file: I do wish you'd quit handing out irc.freenode.net/#asterisk as your "management team" |
05:27.08 | JT | DocHolliday: they had a public hissy fit? |
05:27.19 | file | denon: pfft |
05:27.20 | DocHolliday | JT, yeah |
05:27.24 | JT | wow |
05:27.30 | denon | DocHolliday: would have been more fun if you had used infrared to upload a virus, and wipe their pda ;) |
05:27.34 | DocHolliday | infrared is best, then they cant lose it |
05:27.35 | JT | you'd have thought they're pretend nothing happened |
05:27.57 | russellb | i'd like to code ... but i don't think my brain can handle it |
05:28.05 | DocHolliday | yeah i wasnt very concerned, i dont give out a lot of business cards, infrared is neat though |
05:28.12 | denon | my favorite is the people who dial your number into their sell phone . hit dial, then hang up .. |
05:28.16 | denon | and say "good, now its saved" |
05:28.21 | denon | and refer to their dialed calls logs later |
05:28.31 | denon | the sheer ignorance is comical |
05:28.54 | JT | what's wrong with that? :( |
05:28.57 | denon | they're usually the ones who ask for my number again a couple weeks later |
05:29.01 | denon | saying they werent sure which one I was |
05:29.03 | JT | well there's that |
05:29.18 | JT | i only do it when i'm going to need a number as a once off or something |
05:29.19 | denon | you occasionally get wrong number calls from those people |
05:29.23 | denon | which is funny, you know what they're doing |
05:29.39 | denon | JT: I spose.. but doesnt your phone let you hit save after you dialed the number? |
05:29.59 | denon | you could put 2 letters in, in like 2 seconds, and probably remember it a lot better |
05:30.00 | JT | i can, but i don't want to save randoms i am going to call from my mobile once |
05:30.13 | denon | spose |
05:30.20 | denon | didn't mean to insult my friends down under :) |
05:30.23 | file | russellb: go to sleep! |
05:30.29 | JT | you're right though |
05:30.31 | JT | there is a line |
05:30.32 | DocHolliday | denon, im a big fan of infrared.. most people dont even know they have it |
05:30.45 | DocHolliday | but once i give it to them, rarely have ever 'lost it' |
05:30.48 | denon | DocHolliday: unfortunately, lots of newer stuff doesnt |
05:30.55 | JT | if you plan to call a number more than once ever from your mobile, store it or memorise it |
05:30.58 | denon | darn bluetooth |
05:31.02 | JT | i memorise heaps of numbers |
05:31.21 | denon | I remember paying like $100 extra on my first tdma phone to get infrared |
05:31.37 | DocHolliday | denon, i dont know why it never took off |
05:31.39 | denon | and being sorely disappointed to find out my laptop's infrared chipset was screwy, and didnt communicate well |
05:31.45 | DocHolliday | haha |
05:31.48 | `p4r14h | winders mobile with exchange is good for work :) |
05:31.50 | denon | DocHolliday: too slow for anything useful :( |
05:32.02 | denon | that new linear optical deal is kinda neat |
05:32.04 | DocHolliday | i take the attachments i get on my smartphone and use infrared to upload them to my laptop on the go |
05:32.09 | denon | infrared idea, but much faster |
05:32.19 | `p4r14h | DocHolliday: what kind of smartphone? |
05:32.20 | denon | and recognized people/etc |
05:32.22 | *** join/#asterisk fbffff (n=fbffff@r74-193-79-81.pfvlcmta01.grtntx.tl.dh.suddenlink.net) |
05:32.23 | denon | without knowing what you're doing :) |
05:32.30 | denon | er recognizes |
05:32.43 | DocHolliday | `p4r14h Treo 650 |
05:32.44 | denon | DocHolliday: yeah, I do the same with bluetooth |
05:32.48 | [TK]D-Fender | ok, way late... I'm off... later all |
05:32.52 | denon | built into laptop, and phone, of course |
05:32.55 | `p4r14h | DocHolliday: that is palm isn't it? |
05:32.57 | denon | goes pretty quick |
05:33.59 | DocHolliday | `p4r14h correct, and btw why not pick a nick someone can pronounce? |
05:34.20 | `p4r14h | DocHolliday: `p TAB is all u need |
05:34.25 | denon | or at least something in binary |
05:34.29 | `p4r14h | :D |
05:34.41 | j3j3j3j3j | it's random nick day |
05:34.51 | j3j3j3j3j | eep! |
05:34.52 | russellb | wow, what random luck |
05:34.56 | DocHolliday | denon, yeah i didnt select the bluetooth option on my laptop, didnt think i'd need it |
05:35.07 | denon | i'm such a n00b |
05:35.33 | russellb | hook, line and sinker .. |
05:35.39 | russellb | :) |
05:35.45 | `p4r14h | where would i set the DTMF tone default? my phones numbers arn't exactly in sync with say the voicemail system |
05:35.47 | JT | haha |
05:36.02 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
05:36.02 | *** mode/#asterisk [+o russellb] by ChanServ |
05:36.10 | denon | sorry :) |
05:36.11 | denon | couldnt resist |
05:36.15 | russellb | hehe |
05:36.31 | denon | funny that the random characters ended up spelling your name |
05:36.36 | denon | the odds are astronomical |
05:36.43 | russellb | crazy |
05:37.41 | denon | if I must use your stupid service, at LEAST leave my default funding source as credit card |
05:47.16 | billzybub | POOP! |
05:47.22 | russellb | um. |
05:47.27 | russellb | no? |
05:48.02 | billzybub | i just spent an hour installing and configuring all the pre reqs for freePBX to have the installer script tell me it doesnt support asterisk 1.4 :( |
05:48.21 | JT | heh what did i say earlier :P |
05:48.24 | JT | ~freepbx |
05:48.36 | jbot | i guess freepbx is unable to be supported here. It's a complex piece of dialplan and GUI, and can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
05:48.39 | billzybub | i need a drink |
05:48.55 | DocHolliday | why use freepbx anyway? |
05:49.42 | JunK-Y | cause customers ask it? |
05:51.31 | denon | you actually have users asking for freepbx? |
05:51.40 | denon | sounds like a customer education issue |
05:52.00 | billzybub | yeah try telling a customer to use vi. |
05:52.10 | *** join/#asterisk wwq222 (n=chatzill@c-71-231-5-6.hsd1.or.comcast.net) |
05:52.10 | billzybub | than try living on bread and water. |
05:52.16 | denon | I wouldn't equate asterisk to vi |
05:52.24 | JT | customers shouldn't be making PBXes from scratch |
05:52.26 | denon | it's not that complicated, even for a novice |
05:52.33 | JT | that's what the telephony guy is for |
05:52.38 | denon | nod |
05:52.54 | JT | i agree an interface to do simple things would be useful |
05:52.57 | denon | dont expect to run your business on a system you dont understand, and don't have anyone to manage |
05:53.01 | billzybub | i would use it as a selling point |
05:53.02 | JT | like changing the name of a station |
05:53.06 | JT | and some voicemail stuff |
05:53.14 | wwq222 | Hi i'm trying to implement multiple calls (one call right after another) in Asterisk - is there some way I can do that aside from having some outside app do the scheduling? I tried the 'g' option in the Dial option (to try to call Dial multiple times), but it doesn't seem to get past the 1st Dial |
05:53.35 | billzybub | im wrecked |
05:54.13 | russellb | billzybub: try the new asterisk gui |
05:54.18 | russellb | it takes like 2 seconds to install |
05:54.23 | denon | you could use one of the few thousand guis out there |
05:54.31 | denon | yeah. the official one even |
05:54.34 | billzybub | russellb, whats it called? |
05:54.41 | billzybub | didnt know there was one |
05:54.53 | russellb | billzybub: yeah, you can see screenshots at asterisknow.org |
05:55.08 | russellb | AsteriskNOW is a full linux-distro that includes it, but you can just install it on a regular 1.4 install |
05:55.52 | russellb | <PROTECTED> |
05:55.52 | russellb | <PROTECTED> |
05:55.52 | billzybub | do i have to reinstall my OS for it or is it just a layover? |
05:55.53 | russellb | sorry, copied that from a topic in another channel ... |
05:55.53 | russellb | either one |
05:55.53 | CrashHD | anyone know how to do hints on parked calls when using app_valetparking? |
05:55.59 | `p4r14h | i use asterisk now, its alot better than trixbox+freepbx |
05:56.05 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
05:56.08 | russellb | `p4r14h: i am very glad to hear that |
05:56.23 | russellb | it's basically brand new |
05:56.31 | russellb | but we've got some folks working very hard on it |
05:56.42 | `p4r14h | its still just a complex dialplan, but its all developed by one company |
05:56.51 | JerJer | AsteriskNOW kicks ass |
05:57.07 | russellb | `p4r14h: the dialplan that the asterisknow gui generates is *nothing* compared to what freepbx does |
05:57.13 | *** join/#asterisk sbingner (n=sam@pdpc/supporter/sustaining/sbingner) |
05:57.17 | Qwell | ^^ and that's a good thin |
05:57.18 | Qwell | g |
05:57.25 | JerJer | more than a good thing |
05:57.35 | `p4r14h | russellb: you know anything about the voicemail sending out plain text emails, rather than one with an attachment? |
05:57.37 | denon | "what freepbx does" == generate more spaghetti than all of italy |
05:57.39 | billzybub | russellb, that is extremely helpful, i wish you were here 2 hours ago when i asked and got nothing but a bunch of smart ass answers |
05:57.55 | Qwell | `p4r14h: That should be fixed if you update asterisk from conary or whatever |
05:57.58 | russellb | `p4r14h: i fixed that a while ago, it will be fixed in the next beta |
05:58.05 | russellb | billzybub: i'm sorry :( |
05:58.11 | `p4r14h | how can i fix it right now? |
05:58.13 | billzybub | not your fault |
05:58.34 | russellb | `p4r14h: figure out how to update asterisk using conary ... |
05:58.38 | `p4r14h | please inform me :D |
05:58.48 | russellb | i don't know how to use the rpath tools :( |
05:59.06 | JerJer | russellb: yeah they are pretty crazy |
05:59.09 | `p4r14h | i updated using rpath.... |
06:00.56 | russellb | JerJer: they are supposed to come in and train all of us at some point ... |
06:00.56 | russellb | i just haven't spent the time to learn it on my own |
06:00.56 | JerJer | Real Soon Now(tm) |
06:00.56 | russellb | right |
06:02.56 | russellb | we have some "AsteriskNOW for dummies" books here to give out at the show, heh |
06:02.58 | russellb | pretty cool |
06:03.25 | *** part/#asterisk mog (n=mog@c-71-207-200-130.hsd1.al.comcast.net) |
06:03.50 | Juggie | russellb, i know you have an iaxy or x100p in your office you want to send me. |
06:04.01 | *** join/#asterisk zeeesh (i=zeeesh@202.38.55.125) |
06:04.03 | JT | they're actually For Dummies books? |
06:04.09 | russellb | yeah |
06:04.20 | russellb | it's not very big, though |
06:04.35 | denon | page 1, paragrph 1... download this file, quit your whinin' |
06:05.04 | Juggie | russ, whats the linux util to list the functions within a binary and the librarys they are linked against. |
06:05.07 | Qwell | russellb: who is that author? never seen the name |
06:05.19 | Juggie | or qwell, feel free to answer as well. |
06:05.19 | Qwell | Juggie: ldd? nm? |
06:05.22 | russellb | Qwell: i have absolutely no idea. |
06:06.56 | Juggie | Qwell, ldd shows the libs, but i'm looking for functions |
06:07.02 | Qwell | nm? |
06:07.11 | Juggie | something in this code is linking aginst GLIBC_PRIVATE |
06:07.15 | Juggie | and i'm trying to see what it is |
06:07.37 | wwq222 | Does anyone know how I can make a call, have it hang up after x seconds, and then automatically call someone else? |
06:08.19 | denon | hmm, that reminds me .. asterisk needs an user-exposed threadpool :) |
06:08.30 | russellb | denon: what? |
06:08.30 | denon | er a |
06:08.55 | denon | well .. think of spool items .. but launched from extensions.conf :) |
06:09.05 | denon | with the management properties of a thread pool |
06:09.09 | denon | priorities, delays, etc |
06:09.30 | russellb | System(echo "asdfadfads" > /var/spool/asterisk/outgoing/foo.call) |
06:09.37 | russellb | :-p |
06:09.43 | denon | you're so not getting this :) |
06:09.48 | russellb | nope |
06:09.50 | russellb | i am very tired |
06:09.54 | denon | ditto |
06:10.01 | Juggie | nmmm... Qwell, i see i see... now how to i know which one of those is GLIBC_PRIVATE hrmf. |
06:10.01 | denon | but it's still a good idea :) |
06:10.04 | denon | anywho, I'm outta here |
06:10.06 | russellb | try me again another day :-p |
06:10.08 | Qwell | bed |
06:10.08 | russellb | g'night |
06:10.10 | denon | sure |
06:10.10 | denon | catch ya later |
06:11.16 | *** join/#asterisk Exhar (n=Roy@213-73-139-87.cable.quicknet.nl) |
06:13.52 | *** join/#asterisk slinky (n=slinky@bent1.dsl.xmission.com) |
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06:16.49 | `p4r14h | sudo conray update asterisk almost worked, minus /var/lib/asterisk/sound/silence was owned by another package or something..... |
06:16.57 | `p4r14h | *conary |
06:19.12 | russellb | maybe an asterisk-sounds package? |
06:19.42 | CrashHD | anyone know about the app_valetparking module? |
06:20.26 | *** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
06:20.27 | billzybub | russel, after the make do i need to do a reload from the cli to bring the httpd up? |
06:20.32 | `p4r14h | it is, im trying conary erase asterisk-sounds, then ill update asterisk, then i will reinstall sounds |
06:22.11 | *** join/#asterisk codestr0m (n=asura@ns1.netsyncro.com) |
06:23.21 | russellb | `p4r14h: ah, cool |
06:23.34 | russellb | billzybub: yeah, assuming you made that necessary config changes |
06:23.50 | russellb | you can run "make checkconfig" in the gui directory to make sure you set up the config correctly |
06:24.08 | russellb | but I think "make install" of the gui does that automatically |
06:25.56 | billzybub | yeah, ran checkconfig 6 times to get it right ;) |
06:26.03 | billzybub | this is awesome |
06:26.23 | billzybub | umma put this on my laptop and go hit the strip |
06:26.35 | russellb | cool |
06:26.40 | codestr0m | can someone point me to a best practices for setting up a pbx.. I'm mostly interested in 1) which audio format that * supports will give the highest quality sound and small things like should I turn my sound files into a bigger file or just programmically string them together.. thanks |
06:26.48 | JT | hit the strip? |
06:27.03 | JT | codestr0m: g.711 |
06:27.28 | JT | stringing the files together.. that depends on usage scenarios really |
06:27.33 | billzybub | jt: umma sell the shit outta this |
06:27.45 | JT | i see |
06:28.31 | `p4r14h | man conary has to d/l the file everytime =\ |
06:28.32 | `p4r14h | vmplayer -> winxp -> pptp tunnell -> asterisk box, pretty lame i wish pptp was working on my box :( |
06:28.37 | billzybub | can anyone recomend a wifi sip fone? |
06:29.00 | JT | no |
06:29.06 | JT | they all suck to be honest |
06:29.16 | billzybub | :( |
06:29.35 | JT | use a normal digital cordless phone with an FXS port |
06:29.45 | billzybub | just like all the asterisk gui's i suppose? |
06:30.50 | JT | what? |
06:31.16 | russellb | billzybub: hey now, be nice :-p |
06:31.30 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
06:33.02 | billzybub | this is some slick webwork |
06:33.02 | JerJer | billzybub: find one that supports WMM |
06:33.04 | `p4r14h | for future reference, 'conary update asterisk' is the command to update, but if there are packages that it will conflict with uninstall using 'conary erase PACKAGE' then run 'conary update asterisk' again. after asterisk updates, reinstall the packages with 'conary emerge PACKAGE' |
06:33.30 | russellb | `p4r14h: cool, thanks |
06:33.36 | JerJer | put that on jbot |
06:33.38 | JerJer | perhaps |
06:34.19 | billzybub | hey russ, should the users iver defined in my extensions.conf show up on the web thingy? |
06:34.30 | russellb | probably not .... |
06:34.33 | `p4r14h | that was pretty straight forward, im impressed :D |
06:34.43 | CrashHD | anyone recommend a better parking solution for asterisk? |
06:34.49 | CrashHD | multi parking lots? |
06:35.13 | billzybub | hrm, found the advanced tab but its grey'd out |
06:35.23 | billzybub | i only have sip users defined |
06:35.37 | billzybub | but i cant click on the sip option |
06:36.48 | `p4r14h | repeat question: anyone know what config file to change the default DTMF tones? my hardphones number key tones aren |
06:36.57 | `p4r14h | 't working with voicemail system |
06:41.37 | `p4r14h | holy hell asterisk-sounds has a butt load of sound archives..... |
06:42.09 | JT | yep |
06:44.01 | *** join/#asterisk sbingner (n=sam@pdpc/supporter/sustaining/sbingner) |
06:46.14 | `p4r14h | STOP IT NOW!! I WANT TO SLEEP!!! =\ |
06:49.06 | tzafrir | tt-monkeys |
06:49.16 | tzafrir | tt-monkeys |
06:49.17 | JT | tt-weasels |
06:53.04 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
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06:58.18 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
06:59.55 | billzybub | anyone play with zfone? |
07:03.44 | *** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust826.leed.cable.ntl.com) |
07:04.39 | *** join/#asterisk nuonguy (n=john@c-24-6-175-26.hsd1.ca.comcast.net) |
07:08.35 | *** join/#asterisk tutt9876 (n=tut123@cvl92-2-82-228-144-230.fbx.proxad.net) |
07:09.36 | tutt9876 | hi, anyone has an experiecne with 1.4.1: my peers are not answering |
07:09.52 | billzybub | i installed it yesterday :D |
07:10.06 | tutt9876 | did you have any problem? |
07:10.06 | billzybub | as my first asterisk experiance |
07:10.25 | billzybub | a little bit but i worked wthrough it with some web tute |
07:10.49 | billzybub | im pure sip over here no fxs/fxo |
07:11.00 | tutt9876 | do you know if there are special config to use ? |
07:11.34 | tutt9876 | my peers are not answering ? |
07:12.11 | billzybub | what did you upgrade from? |
07:12.27 | tutt9876 | from 1.2 |
07:12.41 | tutt9876 | I just compile over the 1.2 |
07:12.50 | billzybub | i defined my sip users in sip.conf then defined them in extensions.conf |
07:13.25 | tutt9876 | I just copy my config files from 1.2 that are sworking with it |
07:14.13 | billzybub | i would assume that your peers are no longer working do to some changes in syntax |
07:14.27 | *** part/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
07:14.32 | tutt9876 | changes? |
07:15.33 | billzybub | are your peers sip? |
07:15.57 | tutt9876 | yes |
07:16.29 | billzybub | are they showing up in the CLI |
07:17.46 | tutt9876 | I found unmonitored: don't know is it's ok in fact |
07:17.53 | tutt9876 | if |
07:18.43 | tutt9876 | the command sip show peers doesn't not really intersting information |
07:18.54 | tutt9876 | not really give |
07:23.08 | *** join/#asterisk shodan (n=shodan@ip138.96-113-216.pppoe1.joliette.intermonde.net) |
07:25.21 | *** join/#asterisk ComaVN (n=blaargh@unaffiliated/comavn) |
07:25.45 | *** join/#asterisk sbingner (n=sam@pdpc/supporter/sustaining/sbingner) |
07:33.24 | *** join/#asterisk dj-fu (n=ajc@unaffiliated/dj-fu) |
07:33.26 | *** join/#asterisk antlers (n=antlers@ip68-224-230-141.lv.lv.cox.net) |
07:34.03 | *** join/#asterisk DrukenLPY (n=jdumais@CPE000854ddcdb1-CM00137189cb0c.cpe.net.cable.rogers.com) |
07:37.35 | *** join/#asterisk uwe (i=uwe@217.66.226.39) |
07:37.49 | uwe | good morning :) |
07:40.22 | uwe | im trying to chage the queue ring strategy, and apparently anything other than ring all causes trouble with delays, the caller stays in the queue and only one agent is ringing at a time, what can be done? or do i have to hack app_queue.c ? |
07:51.12 | *** join/#asterisk Turt|e (n=a@80.196.52.186) |
07:51.21 | *** join/#asterisk sashion (n=djbdsf@dsl-244-58-123.telkomadsl.co.za) |
07:54.07 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
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07:55.55 | *** join/#asterisk tengulre (n=tengulre@222.90.66.10) |
07:58.11 | tengulre | hi,all |
08:08.04 | SwK | uwe: what version of asterisk? there is a option you can set in later versions for that |
08:13.58 | *** join/#asterisk Powerkill (n=Powerkil@84.205.154.179) |
08:14.01 | Powerkill | hi |
08:14.06 | rward | hi |
08:14.56 | Powerkill | I upgrade to 1.2.17 and now I've plenty of error like this Mar 22 09:14:11 WARNING[17016]: chan_sip.c:1228 retrans_pkt: Maximum retries exceeded on transmission5898a4-c2010151-13c4-4a40ca-220d1704-3be3@sip.annatel.net for seqno 2 (Critical Response) |
08:14.56 | Powerkill | Mar 22 09:14:11 WARNING[17016]: chan_sip.c:1245 retrans_pkt: Hanging up call 5898a4-c2010151-13c4-4a40ca-220d1704-3be3@sip.annatel.net - no reply to our critica l packet. |
08:15.05 | Powerkill | any idea ? |
08:18.11 | *** join/#asterisk dlynes_laptop (n=dlynes@S01060016b6c052ee.vc.shawcable.net) |
08:19.43 | uwe | SwK: its 1.2.16 , i noticed there is autofill option is 1.4 , but i wish i can do it on 1.2.x |
08:20.32 | Powerkill | I found that this is the problem http://bugs.digium.com/view.php?id=5215 |
08:21.39 | SwK | uwe: the original patch for that was 1.2 and should be on the tracker from 'twisted' |
08:22.23 | sashion | does autofill actually work and what is it's purpose ? |
08:23.06 | SwK | http://bugs.digium.com/view.php?id=5577 |
08:23.25 | SwK | autofill is an app_queue patch that speeds up app_queue in delivering calls |
08:23.48 | SwK | basically if there are 3 calls in queue w/out autofill it only handles delivering 1 call at a time to agents |
08:24.22 | SwK | (so in something like roundrobbin it would only ring 1 call from the queue at any time... the other calls would have to wait until the first call is answered |
08:25.38 | SwK | autofill w/ combines w/ like round robbin will send calls to agents as long as there are avail agents w/out calls loosing their place in queue if an agent doesnt pick up the call |
08:26.00 | Powerkill | please someone know about these error ? Mar 22 09:14:11 WARNING[17016]: chan_sip.c:1228 retrans_pkt: Maximum retries exceeded on transmission 5898a4-c2010151-13c4-4a40ca-220d1704-3be3@sip.annatel.net for seqno 2 (Critical Response) |
08:26.00 | Powerkill | Mar 22 09:14:11 WARNING[17016]: chan_sip.c:1245 retrans_pkt: Hanging up call 5898a4-c2010151-13c4-4a40ca-220d1704-3be3@sip.annatel.net - no reply to our critical packet. |
08:26.39 | SwK | (ie: 5 calls in queue, 3 avail agents, round robbin ring strategy, calls 1,2,3 ring the agents, calls 4,5 just wait, the agent that got call 2 didnt answer it so it stays at the head of the line and goes to the next avail agent |
08:27.23 | SwK | powerkill the far end isnt answering thats what that means |
08:28.01 | SwK | and if you want to show people messages like that use the pastebin instead of spamming the channel |
08:28.04 | *** join/#asterisk yonahw-work (n=yonahw-w@genie03-173-74.inter.net.il) |
08:28.06 | SwK | ~pastebin |
08:28.17 | jbot | i guess pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste, or http://rafb.net/paste/, or http://pastebin.com is usually painfully too slow and unresponsive to use, use one of the other pastebin sites, or dpaste.com is a very nice pastebin as well |
08:28.17 | SwK | pastebin |
08:28.17 | SwK | !pastebin |
08:28.26 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
08:29.07 | Powerkill | ok |
08:29.08 | *** join/#asterisk letherglov (n=lethergl@cpe-24-25-218-137.san.res.rr.com) |
08:30.17 | *** join/#asterisk sashion (n=djbdsf@dsl-244-58-123.telkomadsl.co.za) |
08:32.14 | sashion | SwK: basically at a load point (where I have about 8 - 15) calls in queue, the system starts seeing logged in and available agents (agent show online) as busy (by doing a queue show queuename) and qpp_queue never passes a call until I do a reload on app_queue |
08:33.51 | codestr0m | I downloaded the ulaw sounds from (http://www.astlinux.org) and I'm getting this perm denied message (WARNING[6263]: file.c:804 ast_streamfile: Unable to open demo-echotest (format 0x4 (ulaw)): Permission denied) on the ulaw files, but not the sln files.. I double checked the perms.. and set modules autoload yes.. how can I trace this further.. I see ulaw under show file formats |
08:34.18 | gfraysse | <PROTECTED> |
08:34.52 | tene | codestr0m: check the permissions on the files |
08:35.10 | codestr0m | tene: yeah. I did that first and it's asterisk:asterisk.. it's not that |
08:35.22 | tene | Huh. |
08:35.54 | rward | codestr0m: try restart? |
08:35.55 | sashion | codestr0m: does ls /var/lib/asterisk/sounds/demo-echotest.* show anything ? |
08:36.17 | codestr0m | <PROTECTED> |
08:36.17 | codestr0m | <PROTECTED> |
08:36.40 | Powerkill | if I back revert to 1.2.13 everything is working again |
08:36.55 | codestr0m | 644 perms and asterisk:asterisk.. yeah. I checked all this. |
08:38.04 | codestr0m | could it be incorrect file extension? core show file formats shows == ulaw au au |
08:38.26 | sashion | codestr0m: rename that file to .wav |
08:40.13 | codestr0m | same stupid message. if I change it to .au or .wav.. it's not recognizing ulaw.. (wth?) |
08:40.16 | *** join/#asterisk kremoo (n=kremo@213.215.118.28) |
08:42.46 | *** join/#asterisk af_ (n=getsmart@ip-156-32.sn2.eutelia.it) |
08:43.21 | Turt|e | Hi, i just made an queue, and the youarenext is only played on timeouts, i use 1.4.1. What might i overlook ? |
08:45.46 | codestr0m | sashion: I started * as root and it worked.. what other file perms could be involved? |
08:48.48 | JT | umm |
08:48.52 | JT | codestr0m: |
08:49.05 | JT | ls -la <filename> |
08:49.11 | JT | either message me output |
08:49.14 | JT | or use pastebin |
08:49.33 | JT | i will tell you why it's not working :) |
08:49.52 | codestr0m | ls -lash demo-echotest.ulaw |
08:49.53 | codestr0m | 156K -rw-r--r-- 1 asterisk asterisk 153K Mar 22 10:25 demo-echotest.ulaw |
08:50.43 | codestr0m | does it have to be +x ? |
08:50.47 | sashion | codestr0m: easiest way is to log in as asterisk under a shell and then traverse /var/lib/asterisk/sounds and see where asteirsk can't go any furthuer |
08:50.47 | tzafrir | no |
08:50.54 | sashion | if you understand what I mean :) |
08:51.23 | tzafrir | can you play any other sound file? |
08:51.40 | codestr0m | sashion: I was one step ahead and yes. when I copied the dir over.. didn't put 755 perms on the dir |
08:51.48 | JT | codestr0m: now show ls -la on a file that works fine with asterisk running as user asterisk |
08:52.03 | JT | seems quite weird |
08:52.26 | *** join/#asterisk Avochelm (n=damien@144.136.166.42) |
08:52.33 | codestr0m | JT: yeah. my umask as root when I copied the dir over.. I'm not thinking. it's too early :P |
08:54.17 | JT | codestr0m: wait, you've fixed the prob? |
08:55.11 | *** join/#asterisk lorinc (n=ang@pool-5535.adsl.interware.hu) |
08:55.24 | codestr0m | JT: yeah.. sounds dir wasn't 0755. thanks.. now I just need to figure out the best way to convert my .wav to sln or ulaw with sox :) |
08:55.37 | JT | nice |
08:55.50 | JT | did you sample the wav at 16bits? |
08:56.28 | codestr0m | JT: I had it done in a professional studio.. I have no idea, but I think it's 16bit 44100 wav files.. |
08:57.02 | JT | ah ok |
08:57.11 | JT | you need 16bits at least for best quality |
08:57.27 | JT | as ulaw or alaw gives you more dynamic range than 8 bits |
08:57.29 | codestr0m | I may be wrong. (AUDIO: 48000 Hz, 2 ch, s16le, 768.0 kbit/50.00% (ratio: 96000->192000)) |
08:57.45 | JT | yeah that spec doesn't specify |
08:57.47 | JT | good enough! |
08:57.51 | JT | 48kHz, heh |
08:58.13 | *** join/#asterisk hermuli (n=Eladamri@a88-112-255-26.elisa-laajakaista.fi) |
08:58.19 | JT | they should've recorded in mono though |
08:58.20 | codestr0m | that good enough to start with you think? |
08:58.23 | JT | stereo is useless |
08:59.02 | codestr0m | I can always just drop a channel/track I suppose? I'm still trying to figure out audicity |
08:59.28 | hermuli | hello everyone. can anyone help with this: i got debian etch and get this from trying to make asterisk-addons configure: error: no acceptable C compiler found in $PATH |
08:59.31 | JT | hopefully the tracks are the same |
08:59.36 | hermuli | am not that good with linux anyway |
08:59.40 | JT | hermuli: install gcc |
09:00.04 | hermuli | aww... too obvious |
09:00.08 | JT | codestr0m: what country are you in? |
09:00.09 | codestr0m | JT: I was wrong. that was a format I exported as.. the orignal wav was (AUDIO: 44100 Hz, 2 ch, s16le, 1411.2 kbit/100.00% (ratio: 176400->176400)) |
09:00.19 | JT | hmm ok |
09:00.24 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
09:00.35 | codestr0m | JT: I'm currently in eastern europe, but was born in the US. family is swedish and I travel far too much |
09:00.54 | JT | ok, better question, what country will the asterisk box be serving? |
09:01.09 | hermuli | thanks JT |
09:01.16 | codestr0m | JT: I have 3 pops. 1 SJC 1 SEA 1 AMS |
09:01.16 | JT | np |
09:01.31 | JT | dude those don't mean much to me :) |
09:01.34 | JT | i can guess |
09:01.46 | JT | san jose california? |
09:01.58 | JT | amsterdam something something |
09:01.59 | codestr0m | san jose ca, seattle, wa and amsterdam |
09:02.23 | JT | so the first two are ulaw, the last one is alaw |
09:02.39 | codestr0m | nope. I'm not doing any zap to sip |
09:02.47 | *** join/#asterisk sashion (i=synergy@41.208.192.24) |
09:02.56 | JT | it will never ever touch the pstn? |
09:03.25 | codestr0m | JT: not unless I start putting a box in each country and use SS7 |
09:03.49 | puzzled | morning |
09:03.50 | JT | private voice network? |
09:05.13 | codestr0m | JT: sorta.. zap, sip is high quality, but there are providers (*mci*) which can deliver quality over IP |
09:05.45 | JT | heh |
09:05.56 | sashion | codestr0m: good luck with ss7 and asterisk :) |
09:06.16 | codestr0m | sashion: lol. been there. it's not ready imho. |
09:06.57 | sashion | codestr0m: likewise. Had an interlink to a Nokia switch, but had to drop to PRI as libss7 kept crashing... |
09:07.18 | *** join/#asterisk jm|laptop (n=jm@sentry.flags.co.uk) |
09:07.36 | codestr0m | sashion: well. it can work, but you have to patch it and really overbuild the machine which it runs on |
09:07.59 | codestr0m | It's been about a year since I fought with it at all |
09:08.41 | *** join/#asterisk stimpie (n=michiel@ip565faf27.direct-adsl.nl) |
09:09.48 | sashion | codestr0m: yeah I noticed :) had to do a couple patches for feature like remote retreival, as well as additional optional params... |
09:10.59 | codestr0m | well. it may be my lay impression, but the fact SS7 always requires there to be communication over the line seemed like it created much more load than pri.. not so say I know either protocol very well, but.. |
09:11.09 | codestr0m | 1000000000 little packets all the time == bad |
09:12.17 | sashion | codestr0m: very true... guess ss7 is just a little paranoid about loosing it's peer :) |
09:12.50 | sashion | starting to get irritated with asterisk's queuing.... |
09:12.50 | JT | cant imagine why |
09:12.54 | codestr0m | well. I wish the people who engineered bgp would have done ss7 :P |
09:13.25 | JT | the Internet is less reliable than most tdm networks, i know which i'd rather |
09:14.04 | codestr0m | the internet is only less reliable because the threshold set by ARIN and friends lets asshats route their own traffic |
09:14.28 | JT | i think there's a few more reasons than that... |
09:15.44 | codestr0m | JT: ok.. + the fact that most bandwidth is being used for porn and media.. :P |
09:16.05 | JT | telcos have just sliiightly better hardware, staff and processes |
09:17.05 | codestr0m | gotta run guys.. thank a lot! |
09:19.27 | *** join/#asterisk dngcy2k (n=dngcy2k@mail.netregy.com) |
09:19.52 | dngcy2k | anyone can help? |
09:20.49 | dngcy2k | can Windows AD as user directory services for asterisk |
09:23.10 | *** join/#asterisk RoyK (n=roy@217-175-152.100710.adsl.tele2.no) |
09:23.11 | stimpie | dngcy2k, asterisk can get the users from a ldap server |
09:23.33 | dngcy2k | what bout Windows Active directory? |
09:23.57 | sbingner | Windows AD == LDAP |
09:24.05 | dngcy2k | ic.. |
09:24.33 | stimpie | dngcy2k, google for 'asterisk realtime' and 'asterisk realtime ldap' |
09:25.04 | dngcy2k | can we use a single extension/user sign on on a distributed asterisk system |
09:25.10 | dngcy2k | thanks stimpie |
09:25.11 | *** part/#asterisk codestr0m (n=asura@ns1.netsyncro.com) |
09:26.02 | stimpie | dngcy2k, single sign on isnt that hard but with a distributed asterisk several other problems arise. |
09:27.07 | dngcy2k | meaning it's not recommended to have single signon on a distributed system |
09:28.32 | stimpie | all depends on what you try too accomplish |
09:29.30 | dngcy2k | basically we have 13 branches nationwide and we intend to implement asterisk solution over the branches |
09:29.57 | dngcy2k | we're in the midst of designing the network topology and toying with the idea whether to centralised or decentralised the system |
09:31.11 | stimpie | well one of the issues with single sing on with 13 branches, how do you know where an extension is? |
09:32.41 | dngcy2k | that's what bugging us now |
09:34.19 | *** join/#asterisk stoffell (n=stoffell@fw.catsanddogs.com) |
09:34.53 | DrukenLPY | how many users per office? |
09:36.11 | dngcy2k | apprx. 200 users per branch |
09:36.28 | DrukenLPY | shit, your talking one huge fucken system... |
09:36.38 | DrukenLPY | decentralized is the way to go |
09:36.39 | dngcy2k | yes |
09:38.11 | stimpie | you could use sip phones with a centralized sip server for mobile users |
09:39.57 | dngcy2k | mmmmm |
09:42.21 | dngcy2k | any idea on asterisk HA solution |
09:42.38 | DrukenLPY | HA ? |
09:42.44 | dngcy2k | High availability |
09:42.58 | rward | maybe a SER server infront of the asterisk? |
09:43.47 | DrukenLPY | when your talking 2000+ seats, and 13+ sites.. you really gotta do your homework |
09:44.10 | dngcy2k | yes that's what we're doing now |
09:44.22 | dngcy2k | information and solution gathering |
09:44.58 | DrukenLPY | do you require extensions or will be just incoming or outgoing calls only ie call center status |
09:46.39 | dngcy2k | Full functional PBX with incoming/outgoing and extension |
09:47.02 | DrukenLPY | sounds like loads of fun |
09:47.35 | DrukenLPY | 5 digit extensions and server exchanges all over the place... |
09:49.42 | *** join/#asterisk AlienPenguin (n=Miranda@ip-145-151.sn2.eutelia.it) |
09:50.33 | *** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox) |
09:50.50 | *** join/#asterisk ghenry (n=ghenry@suretecsystems.plus.com) |
09:50.57 | Turt|e | hi, does someone know a way to make an dialplan so a user could pull an single call from an queue? ex dialing *99 and the first caller in the queue would be pulled from the queue ? |
09:52.02 | uwe | SwK[Work]: thank you very very much |
09:53.00 | JT | DrukenLPY: i can't see why you'll need more than 4 digit extensions |
09:53.08 | JT | unless the growth rate is massive |
09:53.54 | AlienPenguin | can anyone give me a hint on why if i put in features.conf the blindtrasfer => #1 does not work while if i just put # it does? (it seems not to wait for the following digits) |
09:53.54 | *** join/#asterisk badcfe (n=cso@LNeuilly-152-22-86-193.w193-251.abo.wanadoo.fr) |
09:54.31 | X-Rob | AlienPenguin, probably because you're using asterisk 1.2.12 or 1.2.13? I think that bug was fixed in .14 |
09:55.17 | AlienPenguin | no i am on 1.4 right now |
09:55.25 | AlienPenguin | and i think it worked on 1.2.15 |
09:55.57 | *** join/#asterisk SwK (n=Silik0nJ@12-214-191-109.client.mchsi.com) |
09:57.47 | dngcy2k | anyone know what is flexible extension logic |
09:58.46 | DrukenLPY | JT: i was figuring 5 digit extensions... XX site and XXX extension |
09:59.49 | JT | i guess that's one way to do it |
09:59.52 | dngcy2k | for interbranch prefix? |
10:00.08 | JT | you could fit it in 4 digits with more complicated pattern matching |
10:01.07 | DrukenLPY | JT: yeah, but your users have to be able to figure out the pattern matching :) and well, i'm lazy hehehe |
10:03.33 | JT | well you could have little sheets howing that 2500-2700 is in City A, etc |
10:03.37 | JT | showing |
10:05.24 | *** join/#asterisk [shodan] (n=shodan@ip211.96-113-216.pppoe1.joliette.intermonde.net) |
10:09.00 | *** join/#asterisk Mavvie (n=edwin@ppp1-52.lns1.syd7.internode.on.net) |
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10:11.14 | DrukenLPY | JT: you could... |
10:11.25 | *** join/#asterisk Arno[Slack] (n=hellSOUN@master.infinityperl.org) |
10:12.04 | JT | if you had some CTI, they could select the person they're dialling from an address book on screen :) |
10:13.05 | Mavvie | funny logging: |
10:13.11 | Mavvie | <PROTECTED> |
10:13.23 | Mavvie | that should have been SIP/ccm-subscriber: |
10:13.31 | Mavvie | han_sip.c: Peer 'ccm-subscriber' is now UNREACHABLE. |
10:16.54 | DrukenLPY | JT: yeah... that would be nice eh? |
10:20.23 | JT | yep |
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10:32.14 | pif | hi, does 1.4.x include misdn ? |
10:33.26 | tzafrir | yes. And latest 1.2 as well |
10:33.40 | sashion | hi... can anyone tell me why there is a difference between agent show online and queue show? Attached pastebin: http://pastebin.ca/406343 |
10:33.44 | ParaNoir | Hey, Anybody succeeded with connecting Swyx to Asterisk via SIP? |
10:34.03 | AlienPenguin | pif: using 1.4.1 with misdn right now |
10:34.16 | pif | tzafrir: but I don't see it in the debian package |
10:34.48 | tzafrir | I'm having problems getting the misdn lib in place |
10:34.55 | ParaNoir | pif: don't you have asterisk-chan-misdn? |
10:35.08 | tzafrir | That package is currently rather broken |
10:35.10 | pif | that's an outdated package |
10:35.15 | ParaNoir | ok |
10:35.25 | ParaNoir | my server crashed with that one ;) |
10:35.39 | tzafrir | I never really needed misdn... |
10:35.51 | pif | tzafrir: in any case I appreciate your work with the * debs |
10:36.57 | JT | there's always bristuff :D |
10:37.05 | pif | yuck! |
10:37.13 | JT | err, excuse me? |
10:37.18 | JT | it's far better than misdn |
10:37.23 | pif | no way |
10:37.24 | ParaNoir | ;0 |
10:37.29 | JT | yes, it is actually |
10:37.37 | pif | I ran away from bristuff screaming |
10:37.41 | ParaNoir | lol |
10:37.42 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
10:37.45 | pif | after 1 year of various bugs |
10:37.48 | ParaNoir | i was trying 5 days to get it working.... |
10:37.54 | JT | and went to misdn? you're crazy |
10:37.57 | JT | misdn has way more bugs |
10:37.59 | ParaNoir | untill i noticed they cut the line ;) |
10:38.04 | pif | misdn has been a dream for me since then |
10:38.13 | JT | maybe you have a very basic setup |
10:39.30 | pif | pretty basic: making and receiving phone calls |
10:41.42 | JT | you've got to be more specific that, instead of being a smartarse |
10:41.47 | JT | like what modes are you using |
10:41.50 | JT | what cards |
10:41.54 | JT | how many ports |
10:41.59 | JT | etc |
10:42.21 | pif | okay, 5 different sites, using 4BRI junghanns cards |
10:42.55 | JT | connecting to only telco? |
10:43.00 | pif | various setups from 1 to 4 ports, some with idsn-dect phones attached |
10:43.32 | JT | misdn's NT mode support is a joke |
10:43.38 | JT | and current bristuff versions are very stable |
10:44.10 | pif | I haven't stressed NT mode too much |
10:44.24 | pif | basic use works fine |
10:44.29 | JT | misdn doesn't even support group dial on nt ports |
10:45.19 | pif | NT ports are meant to ring a specific phone, not a group |
10:45.30 | pif | use app_queue |
10:45.39 | JT | no, that's the stupid one-tracked thinking that misdn authors use |
10:45.49 | JT | no, i'd rather do it properly |
10:46.24 | JT | bristuff supports all zap grouping methods irrespective of TE or NT mode |
10:46.27 | pif | well, say hi to kapejod (if you can find him) |
10:46.38 | JT | are you a misdn author? |
10:46.41 | JT | you seem bitter |
10:46.44 | pif | nope |
10:47.02 | JT | i'm talking about what the situation is like these days, not years ago |
10:47.11 | JT | at the moment, bristuff is leading |
10:47.19 | JT | especially if you use NT mode in any real capacity |
10:47.26 | pif | I just barely lost my job thanks to bristuff |
10:48.15 | JT | that may be why you are bitter |
10:48.31 | JT | never rely on free software unless it's been tested ;) |
10:48.58 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
10:49.57 | ParaNoir | hey, is absolutely nobody familiar with Swyx? |
10:50.06 | uwe | is there any good reason why on ringall sometimes the queue simply doesnt take calls frequently, it acts as if its another strategy? |
10:53.51 | *** join/#asterisk stoffell (n=stoffell@fw.catsanddogs.com) |
10:54.12 | pif | tzafrir: what kind of problems you had with the misdn libs? |
10:54.16 | pif | what version you using? |
10:55.50 | JT | as far as i'm concerned, all BRI solutions are fairly beta |
10:56.00 | JT | but current bristuff is least beta |
10:56.06 | JT | especially when doing anything complicated |
10:57.00 | *** join/#asterisk k31th (n=keith@87.117.194.66) |
10:57.02 | k31th | Is there a way of having 1 phone book db with asterisk? so they is apears on the phones? (I have Linksys SPA942's) |
10:57.43 | *** join/#asterisk basilisk (n=jerry@192.18.43.225) |
11:00.11 | zeeesh | i made 2 peers ... like 100 and 200 registered at server ... 100 can make calls to 200 but 200 is not able to call at 100 .. both extensions are same ... like 100 extensions is . exten => 200,1,Dial(SIP200@200) AND .. 200 extensions is exten => 100,1,Dial(SIP100@100) ... then y 200 is not able to make call ???? |
11:00.39 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
11:01.53 | JT | zeeesh: i'm not terribly sure, but that dial command is wrong |
11:02.08 | JT | Dial(SIP/200/200) |
11:04.17 | zeeesh | is it also works .. somewhere its works .. i checked somewhere its not .. exten => 200,1,Dial(SIP/200) ? |
11:04.49 | *** join/#asterisk MaartenB_ (n=Maarten@h8441243087.dsl.speedlinq.nl) |
11:05.10 | JT | yes, that would work to dial sip entry 200 without specifying a number at that entry |
11:05.45 | *** join/#asterisk nutcase (n=nutcase@i59F556B5.versanet.de) |
11:05.48 | blitzrage | are you looking for the format Dial(SIP/200@proxy) ? |
11:06.20 | *** join/#asterisk donkilla (n=rogers@196.200.26.174) |
11:06.34 | donkilla | Hi |
11:06.59 | blitzrage | y0 |
11:07.28 | donkilla | I bought 2 Digum TDM04B and am having problem with the configuration |
11:08.29 | donkilla | I got an error "No channel type registered for 'zap' whenever i try to use a Zap Channel |
11:08.42 | sashion | donkilla: did you load the modules ? |
11:09.01 | donkilla | modprobe wctdm ? |
11:09.23 | sashion | donkilla: yep... hope you did a modprobe zaptel first ? |
11:09.44 | donkilla | Yes |
11:09.53 | blitzrage | lsmod | grep zaptel |
11:09.56 | sashion | did you load the chan_zap.so module in asterisk ? |
11:10.14 | donkilla | How do i load chan_zap.so? |
11:10.25 | sashion | load chan_zap.so in asterisk CLI |
11:10.31 | blitzrage | *CLI> load chan_zap.so |
11:10.41 | sashion | or if you using 1.4.x... simply type module load chan_zap.so |
11:11.02 | donkilla | I'm using 1.4 |
11:11.11 | blitzrage | you compiled and installed in this order right? libpri, zaptel, asterisk |
11:11.26 | blitzrage | chan_zap.so won't build without zaptel being installed before you compile Asterisk |
11:11.33 | donkilla | yes |
11:12.04 | sashion | donkilla: pastebin a ztcfg -vvvv |
11:12.10 | donkilla | No i did Zaptel Libpri then Asterisk |
11:12.58 | sashion | you don't really need libpri unless you have a TE4XXP card... |
11:13.02 | uwe | hmm ... ok ... is this the way ringall works ? scenario : callers A,B,C ... extensioins X,Y,Z , caller A calls, XYZ ring, B in que and so is C, A . X answers A, B rings on Y and Z, Y picks up ... C stays waiting and Z doesnt ring |
11:13.05 | sashion | or planning to use a PRI interface |
11:13.25 | JT | sashion: don't forget the TE110P and others |
11:13.30 | donkilla | Yes am planning to. I got a T1 card |
11:13.44 | JT | donkilla: libpri, then zaptel, then asterisk |
11:13.52 | tzafrir | pif, basically: not sure what to use |
11:14.07 | donkilla | Looks like i will have to follow that order |
11:14.20 | JT | donkilla: it's logically really ;_ |
11:14.22 | JT | ;) |
11:14.25 | tzafrir | pif, if you have any useful comments regarding misdn in the debs, I'd appreciate them |
11:14.28 | JT | zaptel depends on libpri |
11:14.33 | JT | asterisk on zaptel |
11:14.54 | sashion | and voip depends on asterisk :) |
11:15.03 | sashion | well a portion of it anyway :) |
11:15.08 | *** join/#asterisk cweiske (n=cweiske@dslb-088-074-128-101.pools.arcor-ip.net) |
11:16.21 | donkilla | When i do module load chan_zap.so i get "Error loading module 'chan_zap.so:usr/lib/asterisk/modules/chan_zap.so: cannot open shared object file: No such file or directory |
11:16.36 | cweiske | Hello. An misdn question: is someone using a sitecom dc-104? it seems hfcsusb does not recognize it when I plug it in |
11:17.03 | donkilla | I wanna reinstall and see if i'll be lucky |
11:17.06 | *** join/#asterisk tcseke (n=chatzill@22-36.adsl.etel.hu) |
11:17.25 | JT | donkilla: chan_zap did not build |
11:18.25 | donkilla | Thanks JT |
11:18.59 | blitzrage | donkilla: with 1.4, after compliing and installing zaptel, run in the asterisk dir: make distclean ; ./configure ; make menuselect |
11:19.08 | blitzrage | donkilla: then in menuselect you should see chan_zap in the channel drivers |
11:19.18 | blitzrage | [*] and not XXX |
11:20.38 | *** join/#asterisk MACscr (n=MACscr@adsl-75-23-64-115.dsl.peoril.sbcglobal.net) |
11:22.00 | donkilla | let me try it out blitzrage |
11:24.57 | donkilla | i've gonne to menuselect and i can't see chan_zap |
11:25.25 | MACscr | hmm, how do agents login to a call queue |
11:26.06 | donkilla | Sorry i've seen it |
11:26.40 | uwe | MACscr: heh, same problem here :) but basically using AddQueueMember |
11:27.02 | tzafrir | zaptel does not depend on libpri |
11:27.25 | MACscr | i had the system originally setup so that all users were always logged in, but after i switched my conf files to a new server, its not working anymore |
11:27.27 | uwe | MACscr: if you get further , please tell me |
11:27.56 | donkilla | I can'nt select the chan_zap |
11:28.05 | JT | tzafrir: it does if you intend to use it. |
11:28.12 | *** join/#asterisk __freedom__lover (n=eduardo@clipper.provale.com.br) |
11:28.21 | uwe | MACscr: do you have them as static members in the queue ? |
11:28.37 | donkilla | If i press 'y' nothing happens |
11:28.38 | tzafrir | zaptel does not use libpri . Only chan_zap does |
11:28.47 | donkilla | It has XXX |
11:28.59 | __freedom__lover | hi all, good morning |
11:29.09 | JT | tzafrir: fine, you win :P |
11:29.24 | donkilla | Good morning there and good evening here! |
11:29.29 | __freedom__lover | i have a trouble with sound in asterisk, can anyone help me about this? |
11:29.52 | tzafrir | __freedom__lover, hopefully |
11:29.58 | __freedom__lover | donkilla: where are you from? i'm in brazil |
11:30.06 | donkilla | Kenya |
11:30.17 | __freedom__lover | hum.. africa.. |
11:30.23 | tzafrir | __freedom__lover, what device do you use to hear that audio? What's on the other side? |
11:30.23 | MACscr | stupid linux question, but how do i simply copy all contents of a file to the clipboard so i can paste it in pastebin |
11:30.37 | donkilla | Yes |
11:30.59 | tzafrir | MACscr, which clipboard? normally you just mark it with the mouse |
11:31.00 | __freedom__lover | i have a ac97 |
11:31.12 | __freedom__lover | my OS is freebsd.. |
11:31.21 | MACscr | tzafrir: but what if i cant view the entire file at one time |
11:31.29 | MACscr | im talking about in terminal |
11:31.38 | MACscr | using something like nano to view a file |
11:31.52 | tzafrir | pressing the right mouse button should extend the selection. At least on xterm |
11:32.58 | __freedom__lover | tzafrir_laptop: when my dialplan plays a sound, some message or just playback(tt-monkeys), it stay mute |
11:33.33 | uwe | MACscr: mybe you can do "cat filename" then select the whole thing with mouse if the terminal has scroll bar |
11:34.03 | MACscr | hmm, thats an idea |
11:34.10 | __freedom__lover | tzafrir_laptop: in terminal, i've played a music, unsing mpg123, and i heard the sound, but in asterisk don't |
11:34.40 | MACscr | uwe: when you asked if they were static members, did you mean persistent? as in all the time? |
11:35.54 | uwe | yes, i ment that maybe since you had them all logged in by default, you might have had them added to the queue by default, and they didnt have to log into/added the queues |
11:36.18 | uwe | i have to run |
11:36.31 | tzafrir | __freedom__lover, as a last resort, you can run things in script(1) . But before that, send them to the logs and grab from there... |
11:38.05 | *** join/#asterisk reber (n=reber@cl-158.dub-01.ie.sixxs.net) |
11:38.16 | donkilla | I need little help. I need to select chan_zap module in menuselect but pressing 'y' or 'F8' does nothing. It's marked XXX |
11:38.21 | __freedom__lover | tzafrir: i1ll try this |
11:38.50 | tzafrir | Is there any script to paste text to a pastebin from a file? |
11:42.31 | __freedom__lover | tzafrir: are you talking to me? |
11:44.16 | tzafrir | no. Just asking the general channel population |
11:44.21 | blitzrage | donkilla: if it's XXX'd out then zaptel might not have compiled / installed properly |
11:44.40 | blitzrage | tzafrir: pastebin.ca lets you upload a post |
11:44.56 | *** part/#asterisk cweiske (n=cweiske@dslb-088-074-128-101.pools.arcor-ip.net) |
11:45.31 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
11:46.19 | donkilla | Let me try it out again |
11:47.05 | blitzrage | cd /usr/src/zaptel ; make distclean ; ./configure ; make menuselect ; make install ; cd /usr/src/asterisk ; make distclean ; ./configure ; make menuselect ; make install |
11:47.46 | blitzrage | you might also check out the config.log file for zaptel stuff to see any errors and such |
11:48.23 | Ng | Is there a way to check if a sip extension exists? I'm passing ddis to sip extensions with a pattern, but not all numbers will be used initially. Catching the unused ones and redirecting them would be ideal :-) |
11:48.55 | MACscr | what does:Timeout, but no rule 't' in context 'default' mean? what is rule 't'? |
11:50.18 | blitzrage | Ng: ChanIsAvail()? |
11:50.20 | *** join/#asterisk qdk (n=qdk@0x50c627be.bynxx11.adsl-dhcp.tele.dk) |
11:50.41 | blitzrage | MACscr: 't' is a built in extension (like 's') that you can use in that situation |
11:50.54 | blitzrage | i.e. exten => t,1,Verbose(1|Timeout rule was hit) |
11:51.05 | Ng | Sounds interesting :-) |
11:51.18 | blitzrage | Ng: show application ChanIsAvail |
11:51.25 | blitzrage | might do what you want |
11:51.37 | MACscr | blitzrage : i have no idea what s even is |
11:51.44 | MACscr | why use something so not descrip |
11:51.59 | blitzrage | MACscr: because I'd assumed you'd read some documentation like the O'Reilly book |
11:52.11 | blitzrage | all of this would be much more clear if you had done so |
11:52.17 | donkilla | blitzrage - what about zttool |
11:52.39 | donkilla | It has XXX and i see Depends on: libnewt |
11:52.43 | blitzrage | donkilla: what about it? it's just a utility -- you don't have ncurses-devel installed probably if it won't compile |
11:52.52 | MACscr | blitzrage : nah, im just playing with asterisknow. I had someone setup my other asterisk system |
11:53.06 | blitzrage | oh yah -- libnewt-devel -- you're still missing a package |
11:53.21 | *** join/#asterisk skirmisha (n=viki@87-126-55-7.btc-net.bg) |
11:53.53 | skirmisha | guys if i have sip trunk in asterisk and i have IAX traffic, can asterisk do conversion from iax to sip auto |
11:54.09 | blitzrage | skirmisha: yes |
11:54.29 | skirmisha | ok great |
11:54.50 | blitzrage | when you drop the IAX channel into the dialplan, then you just do Dial(SIP/my_trunk) and Asterisk will take care of the rest |
11:58.10 | *** join/#asterisk uwe (n=uwe1@dogbert.palnet.com) |
11:58.10 | *** join/#asterisk nasls_lsa (n=chatzill@athedsl-178333.otenet.gr) |
11:59.17 | donkilla | When i compile libnewt i get error "/usr/lib/gcc/i486-slackware-linux/3.4.6/../../../../i486-slackware-linux/bin/ld: cannot find -slang |
11:59.42 | *** join/#asterisk champster (n=asterisk@AH.tescogroup.com) |
12:00.13 | donkilla | I mean -lslang |
12:01.40 | blitzrage | donkilla: means you need the 'slang' package (dependency) |
12:02.21 | *** join/#asterisk zmef420 (n=zmef420@metarb3-pool2-40.mtco.com) |
12:02.37 | skirmisha | blitzrage what do u mean drop iax in dialplan |
12:02.46 | skirmisha | they just send iax traffic to me |
12:02.54 | blitzrage | skirmisha: right...... and you handle it via the dialplan |
12:02.57 | skirmisha | and asterisk should do routing transparently |
12:03.04 | blitzrage | i.e. context=incoming_iax_traffic |
12:03.19 | skirmisha | well i don;t have iax users on my asterisk |
12:03.28 | skirmisha | i just want to forward that traffic as sip |
12:03.29 | blitzrage | you have an iax trunk |
12:03.42 | skirmisha | i use trunk only as outgoing |
12:04.05 | blitzrage | then you're taking SIP users and sending them via IAX trunk>? |
12:04.11 | skirmisha | nope |
12:04.25 | skirmisha | iax user from another asterisk will send iax traffic to my asterisk |
12:04.31 | blitzrage | right |
12:04.32 | skirmisha | i just convert it and send it as sip |
12:05.02 | *** join/#asterisk msetim (n=marcos@200.195.161.164) |
12:05.03 | skirmisha | so for asterisk is just transparent traffic |
12:05.20 | skirmisha | only thing is to do conversion to sip |
12:05.22 | blitzrage | then you need an iax type=user in iax.conf, which has a context=incoming_iax_traffic, where you handle the call |
12:05.53 | blitzrage | When the call comes in, that context [incoming_iax_traffic] just basically would hit a Dial(SIP/my_trunk/${EXTEN}) or somethign like that |
12:06.14 | orlock | bwbwba, /. is down |
12:06.18 | skirmisha | yes i have this |
12:06.19 | blitzrage | good |
12:06.24 | blitzrage | skirmisha: well there you go |
12:06.29 | skirmisha | but i am sending it to sip context |
12:06.32 | skirmisha | not iax |
12:07.02 | skirmisha | basically i have set in sip.conf all unknown calls to go to that context |
12:07.06 | blitzrage | well whatever context doesn't matter... you get the call via IAX, put it into the dialplan to be handled, then you send the call back out using the Dial() application |
12:07.07 | skirmisha | and from there to sip trunk |
12:07.24 | skirmisha | ok hope it will wokr |
12:07.26 | skirmisha | work |
12:07.58 | blitzrage | iax user --> auth via iax.conf --> handled via context=foo directive --> dialplan --> application Dial() --> SIP trunk --> outgoing call |
12:08.08 | blitzrage | I can't make it much clearer than that... |
12:13.02 | giasai68 | how can i redirect a call if these fail? e.g. exten => 1265,1,Dial(Phone/phone0) if phone0 is unavailbale how can i redirect on phone1? |
12:14.56 | blitzrage | GotoIf($[${DIALSTATUS} = CHANUNAVAIL]?failover,1) |
12:15.05 | blitzrage | or some other method of parsing on ${DIALSTATUS} |
12:15.07 | *** join/#asterisk antoniobrandao (n=yytrttry@201-43-56-71.dsl.telesp.net.br) |
12:15.15 | giasai68 | ok thank you very much |
12:15.34 | blitzrage | show application dial will show the values DIALSTATUS may contain |
12:15.51 | antoniobrandao | hello, good morning. Does anyone knows wath means "Planning to masquerade channel" |
12:16.15 | antoniobrandao | Mar 21 10:25:53 DEBUG[15218] channel.c: Planning to masquerade channel SIP/voiplink-1ecf8cd0 into the structure of Local/554733637471@callingcard-6501,1 |
12:19.43 | blitzrage | antoniobrandao: it's a debug message that just tells you what Asterisk is doing -- turn off debugging |
12:19.46 | blitzrage | that's normal |
12:22.36 | zoa | blitz!!! |
12:22.39 | zoa | how did you like the phone |
12:22.40 | zoa | ? |
12:22.41 | *** join/#asterisk jeebusroxors (n=jeebusro@cpe-75-80-248-142.dc.res.rr.com) |
12:23.31 | MACscr | hmm, im having an issue with my main menu playing for a trunk, even though its setup to automatically be sent to directly to an extension |
12:24.08 | MACscr | exten = _X.,1,Goto(default|644|1) |
12:24.41 | MACscr | where should i be looking besides extensions.conf for this issue |
12:28.03 | *** join/#asterisk IPmonger (n=ipmonger@c-68-84-208-206.hsd1.pa.comcast.net) |
12:29.52 | rward | MACscr - do you have a specific entry for (default|644|1) |
12:30.00 | *** join/#asterisk Narkov- (n=Narkov@c58-108-246-199.kelvn1.qld.optusnet.com.au) |
12:30.19 | rward | and don't you maybe mean to do a (Dial,644) (check that syntax :p ) |
12:30.26 | *** join/#asterisk Dr-Linux (n=Nothing@202.125.139.198) |
12:30.34 | Dr-Linux | exten => 32345,1,DeadAGI,pager2sms.agi|03339615969\@ufone.com |
12:30.35 | Narkov- | when on an analog ZAP channel the blind transfer function will only accept one digit before immediately saying invalid extension...any ideas? |
12:30.46 | Dr-Linux | anything wrong with this dialplan line? |
12:31.00 | MACscr | default is the context name |
12:31.07 | Narkov- | my time out is set to 3000ms but it immediately says invalid extension |
12:31.27 | rward | MACscr - yes, but do you have an entry in that context for 644? |
12:32.05 | rward | Narkov - what function are you using? |
12:32.17 | Narkov- | blindxtfr |
12:32.22 | MACscr | rward: should that be in sip.conf or users.conf |
12:32.41 | rward | theres a timeout for the amout of time it will wait for the digits to be pressed and then theres one for the pause period between digit presses .. |
12:32.51 | Narkov- | i get the voice anouncement correctly "what extension" but as soon as I hit one number (which registers fine) it immediately "Accepts" the number as invalid |
12:33.13 | rward | MACscr - are you using a GUI at all? |
12:33.15 | blitzrage | Dr-Linux: yah -- you're calling DeadAGI() not from 'h' |
12:33.24 | Narkov- | i only see "featuredigittimeout" which i have set to 3000 |
12:33.32 | MACscr | rward: for the most part, but im having to fix things in the conf files |
12:33.39 | blitzrage | Dr-Linux: DeadAGI() is for being executed after the channel has been destroyed |
12:33.55 | Narkov- | my bad..."transferdigittimeout" is set to the default of 3 seconds |
12:34.21 | Dr-Linux | blitzrage: ok i understand |
12:34.37 | rward | MACscr - then in your extensions.conf , in the [default] section - add a line "exten => 644,1,Dial(SIP/AIX2, 644)" (check that dial command) |
12:34.42 | Dr-Linux | blitzrage: what about last part of the line? |
12:35.01 | rward | MACscr - cos I think its not finding an entry in [default] for 644 - so its dropping to you default.. |
12:35.30 | blitzrage | Dr-Linux: well, I prefer the DeadAGI() syntax (not using commas), but not really anything I see wrong -- what is the error? |
12:35.31 | rward | or change the "goto(default, 644,1)" to "dial(SIP/whatever, 644) |
12:35.51 | Narkov- | the config documentation says transferdigittimeout is in seconds but is it actually meant to be miliseconds? |
12:36.04 | *** join/#asterisk gambolputty (n=gambolpu@cblmdm72-240-185-4.buckeyecom.net) |
12:36.12 | blitzrage | rward: that dial line would dial for 644 seconds |
12:36.32 | rward | sorry - I know - can't remember the DIAL syntax - it needs to be fixed :p |
12:36.42 | blitzrage | what needs to be fixed? It works fine |
12:36.42 | rward | but basically - dial extension 644 .. |
12:36.52 | blitzrage | Dial(SIP/proxy/644) |
12:36.54 | rward | soryr - replying to MACScr |
12:37.10 | Dr-Linux | blitzrage: basically i'm trying to use an SMS application in perl AGI, so i need to put number@email.com in dialplan |
12:37.36 | Dr-Linux | blitzrage: but i'm not sure how can i type email address in dialplan bcoz of "@" |
12:37.41 | blitzrage | Dr-Linux: ok -- I don't see any error in the syntax -- it must be a script issue |
12:38.06 | blitzrage | you're escaping the @ -- what does the script see? |
12:38.42 | *** join/#asterisk myiagy (n=myiagy@200.175.61.250.static.gvt.net.br) |
12:39.44 | Dr-Linux | blitzrage: maybe you can see here the script: http://portal.mmasson.com/asterisk/files/pager2sms-0.2.agi |
12:40.30 | blitzrage | Dr-Linux: I don't have time right now |
12:40.46 | Dr-Linux | blitzrage: No Problem. Thanks |
12:40.53 | blitzrage | you not showing any of the output -- like what asterisk is seeing or doing, or what the script is seeing or doing |
12:43.24 | *** join/#asterisk tcseke (n=chatzill@22-36.adsl.etel.hu) |
12:48.17 | *** join/#asterisk donkilla (n=rogers@196.200.26.174) |
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12:53.16 | donkilla | blitzrage: I sorted out slang package and now when i go to chan_zap in menuselect it still has XXX. Depends on:zaptel_vldtmf(E),zaptel(E),tonezone(E) |
12:53.20 | *** part/#asterisk Narkov- (n=Narkov@c58-108-246-199.kelvn1.qld.optusnet.com.au) |
12:53.45 | blitzrage | donkilla: you need to checkout zaptel from SVN |
12:53.52 | blitzrage | donkilla: asterisk-1.4.2 right? |
12:54.01 | donkilla | yes |
12:54.13 | blitzrage | svn co http://svn.digium.com/svn/zaptel/branches/1.4 zaptel-1.4 |
12:54.36 | blitzrage | yah... that was a bit of a bug on Digiums part. I expect to see a Zaptel-1.4.1.1 come out to address that... |
12:55.41 | donkilla | thanks.Let me try it out |
13:01.06 | MACscr | rward : here is my extensions.conf |
13:01.06 | MACscr | http://pastebin.ca/406431 |
13:01.18 | MACscr | i still havent been able to resolve the issue |
13:01.25 | MACscr | i tried your different ideas |
13:01.44 | MACscr | whats odd is that it used to work on one trunk, but not the other, then both dont work |
13:01.49 | *** join/#asterisk friedrich| (n=friedric@e177249102.adsl.alicedsl.de) |
13:02.15 | MACscr | the only things i have changed for incoming calls was from a call queue to directly to an extension. Figured it was less complicated for testing purposes |
13:03.11 | *** join/#asterisk lilwookie (n=lilwooki@30-82-252-216-static.enter-net.com) |
13:06.44 | *** join/#asterisk galeras (n=root@200.118.211.115) |
13:10.49 | *** join/#asterisk heison (n=heison@ns.somanetworks.com) |
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13:15.46 | MACscr | wow, things died in here |
13:15.53 | galeras | Hello. Which tcp ports do i need to open to place a sip call through a firewall? |
13:16.33 | galeras | init 3 |
13:16.58 | sashion | galeras: UDP: 5060 and then range UDP 10000-20000 for RDP |
13:18.22 | galeras | thanks sashion, is 5060 used for signalling and 10000-20000 for voice? |
13:18.31 | zoa | yes |
13:19.12 | blitzrage | galeras: note -- he said UDP, not TCP (asterisk doesn't use TCP) |
13:19.20 | blitzrage | might solve you some pains later :) |
13:19.36 | galeras | thanks guys |
13:19.47 | *** part/#asterisk donkilla (n=rogers@196.200.26.174) |
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13:34.41 | *** mode/#asterisk [+o mog] by ChanServ |
13:34.56 | heison | anyone here using SPA-3102 ? |
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13:37.44 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
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13:39.11 | GaryH | Hi, is anyone here familiar with chan_misdn? |
13:39.45 | EtienneG | Hi folks ... looking for an IRC that would interface with * to print incoming call number and CID in an IRC channel |
13:39.52 | EtienneG | does anybody know of such a thing ? |
13:40.15 | EtienneG | (an IRC bot, even) |
13:40.30 | nasls_lsa | ?? |
13:40.48 | *** part/#asterisk galeras (n=root@200.118.211.115) |
13:40.49 | nasls_lsa | an idea I can give you |
13:41.04 | nasls_lsa | that find an IRC bot writen in PHP |
13:41.10 | zeedo | EtienneG: http://www.telephreak.org/code/astbot-1.0 |
13:41.14 | zeedo | you could use that as a basis |
13:41.14 | nasls_lsa | and then connect with AGI ( I think ) |
13:41.29 | zeedo | not sure exactly what functions it has, but it appears to interface with * |
13:41.39 | zeedo | never used it, just come across it |
13:43.44 | *** part/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
13:49.11 | EtienneG | thanks guy, I will look into astbot |
13:50.12 | *** join/#asterisk bkuhn (n=bkuhn@fsf/member/bkuhn/bkuhn) |
13:50.27 | bkuhn | sterday |
13:54.08 | JT | eww php |
13:54.14 | JT | irssi does perl scripting |
13:54.21 | JT | that'd be easy to interface with AGI |
13:54.27 | marc\cba | chaps |
13:54.33 | marc\cba | if my SIP trunk provider |
13:54.50 | marc\cba | specifies the DID number, what was dialed to reach my pbx |
13:55.01 | marc\cba | in the second packet, the ACK (as opposed to the INVITE) |
13:55.16 | marc\cba | how could i go about routing based on that in my dialplan? |
13:56.03 | marc\cba | i.e |
13:56.33 | marc\cba | instead of this: INVITE sip:08450000010@000.000.228.32 SIP/2.0 |
13:56.36 | marc\cba | the trunk passes |
13:56.48 | marc\cba | INVITE sip:000.000.228.32 SIP/2.0 |
13:56.56 | marc\cba | and it is not until the second packet |
13:56.57 | marc\cba | the ACK |
13:57.09 | marc\cba | that it mentions the DID |
13:57.23 | marc\cba | To: <sip:448450000010@n.e164.org.uk> |
13:59.22 | rward | uuuummm i give up :/ |
14:05.35 | *** part/#asterisk EtienneG (n=etienne@modemcable178.77-70-69.static.videotron.ca) |
14:05.35 | *** join/#asterisk stony (n=steinche@p57b38ac1.dip0.t-ipconnect.de) |
14:05.49 | stony | hi, did the Dial() function change in asterisk 1.4 ? |
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14:08.02 | *** mode/#asterisk [+o denon] by ChanServ |
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14:16.38 | *** mode/#asterisk [+o anthm] by ChanServ |
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14:21.29 | *** mode/#asterisk [+o mog] by ChanServ |
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14:24.33 | giasai68 | how can i accept call only from some ip on asterisk? i explain better now i can call trought asterisk using all ip (sip,h323) can i accept call only from some ip or can i set some users/IPs? |
14:24.53 | b11d | use a firewall? |
14:24.56 | *** join/#asterisk Ac1dcrawl (n=cow@64.31.169.118) |
14:25.10 | antoniobrandao | hellou, is there any kind of media-proxy for asterisk? i would like to relay rtp traffic out of asterisk machine, but reinvite is not always possible due to nat issues |
14:25.10 | b11d | or, dont you specify the IP of the host in sip.conf with the "host = x.x.x.x" line? |
14:25.24 | giasai68 | this is only mode (firewall) |
14:25.32 | Ac1dcrawl | I am running asterisk 1.2.13 and for the some reason when I am using the CLI when doing command-line completion the CLI locks up |
14:25.39 | Ac1dcrawl | any ideas what could be wrong? |
14:25.41 | b11d | then upgrade |
14:25.46 | b11d | 1.2.16 is out of the 1.2 branch |
14:28.15 | *** join/#asterisk jm|laptop (n=jm@sentry.flags.co.uk) |
14:28.53 | *** join/#asterisk af_ (n=getsmart@ip-156-32.sn2.eutelia.it) |
14:29.02 | *** join/#asterisk penguinFunk (n=penguin@87.224.86.46) |
14:29.31 | b11d | . |
14:29.54 | *** join/#asterisk hijacked (i=qib1@cerebus.clandestineresearch.com) |
14:31.53 | hphinc | hark, how is everyone? |
14:32.16 | hphinc | is anyone an expert in presence / parking? |
14:32.23 | *** join/#asterisk Mercestes (n=Merceste@cpe-24-175-82-3.houston.res.rr.com) |
14:32.35 | sashion | giasai68: are you using static hosts for your sip hosts, or are you using host=dynamic ? |
14:33.13 | b11d | mercestes |
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14:33.43 | Mercestes | b11d |
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14:36.25 | b11d | god i hate you |
14:36.49 | *** join/#asterisk CreepyCrawly (n=creepy@62.135.86.11) |
14:36.57 | cyanid3 | Is it possible with any phone / asterisk combination to make it so that when someone (a receptionist) presses a button on their phone for the ivr that's currently in use to be changed? |
14:37.02 | CreepyCrawly | good afternoon people, is there a port of freepbx available ? from freebsd ? |
14:37.16 | b11d | ask in #freepbx |
14:37.20 | b11d | i use asterisk on FreeBSD just fine though |
14:37.30 | Mercestes | I love you too, b11d. |
14:37.32 | b11d | :) |
14:37.39 | b11d | i retract my statement of hatred |
14:38.03 | Mercestes | yay |
14:38.09 | Mercestes | you are always off topic |
14:38.10 | CreepyCrawly | b11d, i got it running and all :) but i wanna try out freepbx |
14:38.16 | b11d | i know Mercestes.. it's what i do. :) |
14:38.27 | b11d | oh.. well then go to #freepbx and talk to them about it :) |
14:38.44 | CreepyCrawly | i am hehe :) |
14:38.45 | Mercestes | CreepyCrawly, You want to try out retardation? will it make you feel smarter when you give it up or are you seeking addiction to the ludicrous? |
14:39.02 | CreepyCrawly | Mercestes, none of those ;) |
14:39.04 | b11d | wtf is up with you today Mercestes.. what wrong? |
14:39.15 | CreepyCrawly | heh |
14:39.39 | Mercestes | CreepyCrawly, ;) Freepbx is a royal mess and a complete clusterfsck. I'd avoid it...unless you like pain. Then I suggest #bondage as an alternative to #freepbx if no one is answering there |
14:39.44 | stony | my asterisk installation isn't dialing .... it says all channels are busy, but they aren't ... any ideas ? |
14:39.54 | CreepyCrawly | haha |
14:40.29 | Mercestes | CreepyCrawly, lol. Just make backups and don't take it too seriously. It should prove educational tho. Good luck. |
14:40.39 | CreepyCrawly | hah fsck it |
14:40.57 | CreepyCrawly | i aint going through all that havoc ill just stay comfy with the cmd lime kthx |
14:41.07 | Mercestes | good man. :) If you need scripts or pointers or magick just ask in here, there is lots of help in here. |
14:41.18 | CreepyCrawly | :) |
14:41.21 | Mercestes | if you even have freepbx files on your install they give you the #freepbx boot. :( |
14:41.22 | hphinc | is there anyone who knows presence / parking in here? |
14:41.27 | Mercestes | we're kinda elitest liek that. |
14:41.36 | hphinc | I am looking to park a call and have the light light up on my Snom 360. |
14:41.40 | hphinc | need some guidance.... |
14:41.45 | Mercestes | b11d: I'm in a ...."mercestes mood" :D |
14:41.46 | b11d | there's a right place for the right things.. |
14:41.53 | b11d | i dont know what that means yet.. |
14:41.56 | CreepyCrawly | hah |
14:42.30 | Mercestes | b11d: Both halves of your sentance are subject to opinion independently tho. =/ |
14:42.39 | Mercestes | hphinc: Ok. What exactly is wrong then?? |
14:42.53 | *** join/#asterisk Opperior (n=chatzill@24.61.165.73) |
14:42.59 | hphinc | Mercestes: I don't understand how the function keys communicate with Asterisk. |
14:43.01 | *** join/#asterisk antlers (n=antlers@ip70-173-90-39.lv.lv.cox.net) |
14:43.01 | b11d | aye, all things are interpreted subjectively though. |
14:43.20 | hphinc | Park() and Parkand Announce() I can read docs on...but how does the phone tell asterisk to execute those commands? |
14:44.11 | Mercestes | via an extension. |
14:44.15 | Mercestes | or via features.conf |
14:44.24 | *** join/#asterisk flujan (n=flujan@200-160-115-020.static.spo.ctbc.com.br) |
14:44.25 | *** join/#asterisk antlers (n=antlers@ip70-173-90-39.lv.lv.cox.net) |
14:44.29 | flujan | hi guys... |
14:44.29 | hphinc | ok, let me re-read about features.conf, and I will get back to you. Thanks. |
14:47.02 | hphinc | Mercestes: does the phone use applicationmap to talk to asterisk in reference to parking calls? |
14:48.28 | *** part/#asterisk GaryH (n=chatzill@2001:618:42d:101:213:72ff:fecf:8262) |
14:48.58 | b11d | i wish i had an ipv6 address |
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14:54.29 | *** join/#asterisk chefrs (n=joe@c-24-8-226-145.hsd1.co.comcast.net) |
14:54.53 | chefrs | Any idea when I dial an outbound trunk, about 50% of the time it rings once and then just sits there? |
14:55.08 | hphinc | Mercestes: I have parking working, but I want to use the function keys to automatically get the parked call. (and light up like a key system to show that someone is on that extension, parked). Any thoughts? |
14:56.23 | Mercestes | hphinc: not that I am aware of. |
14:56.35 | hphinc | busted..... |
14:56.36 | Mercestes | Hphinc: Yea, use a key system. |
14:56.37 | hphinc | :-( |
14:56.50 | hphinc | I have seen it done, but I can't figure out how to do it. |
14:56.57 | Mercestes | hphinc: You could map a "function key" to a "key map" or you couild just program in *8 as call park, with park and announce... |
14:57.04 | hphinc | something about the metermaid patch, which I have done. |
14:57.08 | Mercestes | hphinc: But you would still have to dial the extension to pick up the call. |
14:57.20 | hphinc | how do I map a function key to a key map? |
14:57.25 | hphinc | in what conf file? |
14:57.33 | JT | antoniobrandao: canreinvite=no |
14:57.45 | Mercestes | hpinc: As far as "lighting up a light" like on a key system.....I'm not even sure that would work as the lighting of the lights is a buddy watch, not a "line watch" There are no lines in Asterisk, so therefore, there is no key system. |
14:57.47 | hphinc | Mercestes: I am able to park with the function key. Just not pickup or show the light. |
14:58.10 | chefrs | Any idea when I dial an outbound trunk, about 50% of the time it rings once and then just sits there? |
14:58.15 | hphinc | Mercestes, when I figure it out, you want me to post it back here? |
14:58.25 | Mercestes | hphinc: When you park a call the call belongs to asterisk, not the snom. The snom becomes oblivious. |
14:58.56 | hphinc | right. But if I set the snom function buttons to 701,702,703....etc... I can have those buttons light up when a call is placed on park. |
14:58.57 | antoniobrandao | but canreinvite=no can't "relay rtp traffic out of asterisk machine" |
14:59.12 | Mercestes | hphinc: Let me use an analogy...it's like a girlfriend.... On a key system, she shows up at a party, and wants you. On a key system, she comes to me and I bring her to you and I go "I have your girlfriend, here.." |
14:59.25 | Mercestes | In asterisk, you get a sticky note that says "I have your girlfriend, come get her when yoru ready." |
14:59.33 | chefrs | lol |
14:59.43 | JT | antoniobrandao: what are you quoting? yes, it can. |
14:59.46 | hphinc | What if I have two girlfriends? |
14:59.51 | hphinc | (were I so lucky). |
14:59.54 | hphinc | :-) |
15:00.10 | Mercestes | hphinc: in a key system, your girlfriends would conflict because you could only play with one at a time.... |
15:00.21 | Mercestes | hphinc: In asterisk you can play with 2 or even 100 girlfriends simultaneously. |
15:00.21 | hphinc | Mercestes: That's no fun. |
15:00.22 | kremoo | <PROTECTED> |
15:00.39 | hphinc | Mercestes: Asterisk: more chicks that you can handle. |
15:00.46 | Mercestes | hphinc: exactly. |
15:01.15 | Mercestes | hphinc: The point is....once you park the call, the snom is out of the picture entirely. It cannot monitor the state of that call. The call is released to the PBX and handled there and the snom is free to....set fire or something. |
15:01.39 | Mercestes | there is no line 1,2,3,4 Everything is dynamic, free flowing, and assigned as needed. |
15:02.07 | Mercestes | Flash Operator Panel can monitor the parked slots of 700, 701, 702, 703 and via the manager interface you can give an indicator, and maybe even HACK the Snom to do what you want. |
15:02.13 | hphinc | so, then, how do we tell the Snom to light up that light? |
15:02.17 | Mercestes | but manager API is about the only thing yo uhave to attempt what you want. |
15:02.18 | b11d | . |
15:02.22 | hphinc | while it is setting fire to something.... |
15:02.27 | hphinc | extension hints? |
15:02.46 | antoniobrandao | JT, quotting myself. With canreinvite=no all rtp traffic will go in asterisk machine. I would like to not use asterisk machine bandwith for rtp traffic. Like openser does with rtp_rpxy and media_proxy. |
15:03.07 | *** join/#asterisk tkowal (n=nospamto@74.93.82.14) |
15:03.27 | JT | antoniobrandao: sorry, make up your mind |
15:03.34 | hphinc | Mercestes: http://www.voip-info.org/wiki/view/Asterisk+phone+snom |
15:03.48 | JT | antoniobrandao: you either want rtp do go through the box or not? |
15:04.29 | tzafrir | kremoo, I figure you saw http://www.xorcom.com/pdfs/AB001_OpenDoors.pdf . |
15:04.55 | chefrs | Anyone have any ideas? |
15:05.22 | hphinc | I had an idea once. It was brief, fleeting, and I eventually lost it. |
15:05.29 | chefrs | Any idea when I dial an outbound trunk, about 50% of the time it rings once and then just sits there? |
15:05.50 | antoniobrandao | JT, i always made my mind long time ago. No rtp traffic in asterisk. Canreinvite=yes is great, but with nat doesn't work. So, what to do? |
15:06.21 | JT | antoniobrandao: nothing you can do except give them all public ips |
15:06.41 | JT | antoniobrandao: otherwise you can't reinvite |
15:07.07 | *** join/#asterisk frenzy_ (n=frenzy@unaffiliated/frenzy) |
15:07.30 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
15:07.55 | antoniobrandao | JT, ok, tks. That what i tought. Will have to put some ser in this scenario |
15:08.11 | JT | antoniobrandao: SER won't help. |
15:08.18 | JT | it's a protocol issue |
15:08.22 | JT | that SER does not solve |
15:10.30 | *** join/#asterisk heh_v_water (n=heh_v_wa@71-210-51-58.hlna.qwest.net) |
15:10.37 | antoniobrandao | JT, yes, it does. ser/openser with media_proxy solves by putting an external daemon. The natted clientes send rtp media to that daemon which relays to the another end. |
15:11.15 | JT | which is the same as running asterisk with canreinvite=no |
15:11.23 | JT | different implementation |
15:11.27 | JT | but same effect |
15:12.31 | antoniobrandao | JT, no, its not. With canreinvite=no one machine is responsible to handle all rtp traffic |
15:12.55 | *** join/#asterisk tkowal (n=nospamto@74.93.82.14) |
15:13.01 | antoniobrandao | with media_proxy you can use many machines |
15:13.21 | JT | antoniobrandao: that's up to you |
15:13.25 | JT | the implementation |
15:13.34 | JT | you can use multiple asterisk machines too |
15:13.37 | JT | so moot point |
15:13.52 | JT | i'm not saying asterisk is necessarily better |
15:13.55 | *** part/#asterisk frenzy_ (n=frenzy@unaffiliated/frenzy) |
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15:20.59 | ctooley | I'm having an issue with ExternalIVR. When a call comes in and is in the dialplan, they can press a button, the DTMF is received and action is taken, when ExternalIVR is called, no more audio, either way. |
15:21.13 | ctooley | It's like the sound conduits go away altogether. |
15:22.32 | *** join/#asterisk nasls_lsa (n=chatzill@athedsl-179959.otenet.gr) |
15:22.44 | vlt | Hello. I have defined two SIP phones A and B each in "callgroup=1". But when I try to pickup a ringing call on A from B I get "chan_sip.c:10458 handle_request_invite: Nothing to pick up". What id I miss? |
15:24.50 | *** join/#asterisk drachenfels (n=king@85.189.8.11) |
15:27.05 | m4rkl4r | i'm using the manager api to originate calls. If I say: Channel: SIP/9043067733@joinuneta.com, then the CLI says: |
15:27.05 | m4rkl4r | chan_sip.c:1980 create_addr: No such host: joinuneta.com |
15:27.05 | m4rkl4r | channel.c:2432 __ast_request_and_dial: Unable to request channel SIP/9043067733@joinuneta.com |
15:27.05 | m4rkl4r | However, joinuneta.com has an srv record, and using dig on the asterisk server shows it is properly defined. |
15:27.05 | m4rkl4r | Furthermore, I get precisely the same error if I use the direct hostname: SIP/9043067733@sip.joinuneta.com |
15:27.13 | pif | vlt: you forgot pickupgroup |
15:27.34 | vlt | pif: Do I need both? |
15:27.43 | pif | yep |
15:28.10 | JunK-Y | m4rkl4r: quick hack, uses its ip. |
15:28.11 | m4rkl4r | furthermore if I use the direct IP address, I get the same error. |
15:28.49 | Gido-E | m4rkl4r default gw problems? |
15:30.22 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
15:30.30 | JT | m4rkl4r: why don't you do it properly with a sip.conf entry for that host? |
15:30.31 | m4rkl4r | Gido-E: no.. i am able to ping yahoo.com, for example |
15:31.31 | drachenfels | does anyone know how to trobleshoot an fxo module? whenever we hook it up to a line, the line suddenly goes engaged - and we can't call in.. We've tried the same line, connected to a different port on the card and it works fine, which makes me think it's the module on the card, but I've no idea how to troubleshoot it.. |
15:31.49 | *** join/#asterisk nasls_lsa (n=chatzill@athedsl-179959.otenet.gr) |
15:32.04 | *** join/#asterisk pif (n=ldm@zenon.apartia.fr) |
15:32.16 | m4rkl4r | JT: That doesn't work either. |
15:32.32 | m4rkl4r | Doing sip reload, I see something interesting: |
15:32.58 | m4rkl4r | ar 22 11:32:39 WARNING[530]: chan_sip.c:12708 reload_config: Empty context specified at line 5 for domain 'uneta.com' |
15:32.58 | m4rkl4r | Mar 22 11:32:39 WARNING[530]: chan_sip.c:12708 reload_config: Empty context specified at line 6 for domain 'youneta.com' |
15:32.58 | m4rkl4r | Mar 22 11:32:39 WARNING[530]: chan_sip.c:12708 reload_config: Empty context specified at line 7 for domain 'sip.uneta.com' |
15:32.58 | m4rkl4r | Mar 22 11:32:39 WARNING[530]: chan_sip.c:12708 reload_config: Empty context specified at line 8 for domain '66.129.95.19' |
15:33.17 | puzzled | ~pb |
15:33.18 | jbot | pb is probably a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
15:33.23 | JT | maybe you should specify a valid context |
15:33.33 | m4rkl4r | i am specifying a valid context. |
15:33.41 | JT | with nothing in it? |
15:33.43 | b11d | lol |
15:33.46 | m4rkl4r | haha |
15:33.47 | m4rkl4r | no. |
15:33.58 | JT | that's exactly what the error suggests |
15:34.01 | puzzled | does anyone know if the sangoma patches for zaptel can be applied without interfering with the proper functioning of digium cards? |
15:34.12 | JT | send the whole of your extensions.conf and sip.conf to pastebin.ca |
15:34.20 | m4rkl4r | i built a new asterisk box, copied over the config files, with extensions based off of odbc. |
15:34.24 | m4rkl4r | ok. |
15:34.30 | coppice | puzzled: I mix cards, and never had a problem |
15:34.33 | JT | odbc |
15:34.37 | puzzled | coppice: thanks |
15:34.42 | JT | is the odbc connection working? |
15:35.41 | drachenfels | does anyone know how to trobleshoot an fxo module? whenever we hook it up to a line, the line suddenly goes engaged - and we can't call in.. We've tried the same line, connected to a different port on the card and it works fine, which makes me think it's the module on the card, but I've no idea how to troubleshoot it.. |
15:35.52 | *** join/#asterisk wunderkin (n=kev@dslstat-ppp-95.fastq.com) |
15:36.10 | JT | drachenfels: i swear i saw that exact same question not a couple of minutes ago.. |
15:36.25 | b11d | weird, me too. |
15:36.30 | m4rkl4r | jt: extensions.conf in http://pastebin.ca/406616. |
15:36.30 | m4rkl4r | and odbc is working. |
15:36.38 | puzzled | you guys need glasses. see everything double :) |
15:36.46 | b11d | ") |
15:36.48 | b11d | doh |
15:36.48 | b11d | :) |
15:37.03 | *** join/#asterisk nasls_lsa (n=chatzill@athedsl-179959.otenet.gr) |
15:37.27 | JT | m4rkl4r: ah like i thought after you mentioned realtime, can't really diagnose that bit since i don't know what is in your db |
15:37.54 | m4rkl4r | jt: sip.conf: http://pastebin.ca/406618 |
15:37.56 | m4rkl4r | yes. |
15:38.04 | m4rkl4r | the thing that is odd, jt, |
15:38.09 | b11d | you just need more marklar.. |
15:38.24 | b11d | marklar the marklar over the third marklar and then you'll have marklar working. |
15:38.34 | b11d | see? |
15:38.37 | m4rkl4r | if I take the command that I'm sending to the manager api, change all the Variable directives to Set, and put it in the spool, it works. |
15:38.41 | m4rkl4r | yes i see, marklar |
15:38.57 | JT | m4rkl4r: do not use spaces with = symbols in sip.conf |
15:40.09 | m4rkl4r | jt: ok.. i didn't know that.. but the error message still is there after fixing that |
15:40.20 | MrWup | Aastra 9133i VoIP Phone |
15:40.22 | MrWup | what do you think? |
15:40.27 | MrWup | ive just ordered 20 of em |
15:40.46 | pif | you shoulda asked before ordering |
15:40.49 | JT | a bit late to ask ;) |
15:40.53 | MrWup | i can always cancel |
15:40.59 | MrWup | i did ask beforehand too |
15:41.01 | pif | cancel and get polycoms |
15:41.07 | L|NUX | is there any bug in asterisk 1.2.17 |
15:41.07 | coppice | i guess this is a masochism thing :-) |
15:41.13 | m4rkl4r | They don't have stun, or encrypted channels for provisioning |
15:41.19 | m4rkl4r | the astraphones, that is |
15:41.21 | MrWup | polycoms are too expensive |
15:41.35 | pif | 430 aren't |
15:41.36 | *** join/#asterisk xpot (n=jim@c-71-195-241-115.hsd1.ma.comcast.net) |
15:41.38 | m4rkl4r | I've been pretty happy with the linsys spa 941 so far |
15:41.56 | JT | m4rkl4r: most people use sip phones on a lan, don't need stun, actually, you don't need stun if your sip server doesn't suck, even if you're behind nat |
15:42.00 | L|NUX | i am sending calls to my * no rule found for that number so it should go to s but its not working |
15:42.08 | L|NUX | any one have similar issue ? |
15:42.14 | JT | m4rkl4r: asterisk doesn't even support encrypted SRTP |
15:42.14 | drachenfels | surely posting sip.conf with usernames and passwords is a bad idea? |
15:42.27 | xpot | anyone know of existing issues with automon in 1.4? |
15:42.29 | m4rkl4r | yes, I suppose it is, drachenfels |
15:42.33 | MrWup | i thought aastra was good quality |
15:42.41 | Opperior | I've had to use stun with * when both the server and phone are behind a nat |
15:42.58 | JT | aastra is decent, but they're fairly low end phones |
15:43.03 | *** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
15:43.04 | JT | haven't used them personally |
15:43.11 | JT | surely 1000X better than grandstream |
15:43.24 | m4rkl4r | drachenfels: but those are old, invalid paswords |
15:43.43 | Strom_M | having a white hot railroad spike driven six feet into your skull is better than grandstream :) |
15:43.57 | L|NUX | any one have similar issue like i have |
15:44.00 | JT | Opperior: server behind nat :(, need to port forward then |
15:44.03 | L|NUX | can some one help me with that :) |
15:45.15 | JunK-Y | linux: you are sending it s@context1? |
15:45.29 | *** join/#asterisk marv[work] (n=timr@24.214.206.254) |
15:45.34 | Opperior | JT: I do. But with the phone behind a nat as well, RTP still won't work without stun. At least, not that I've found |
15:46.26 | pif | MrWup : how much do they cost? |
15:46.28 | Gido-E | L|NUX do you have s,1 ? |
15:46.40 | L|NUX | yes |
15:46.47 | JT | Opperior: must've been setup wrong |
15:46.51 | JT | STUN is really a hack |
15:46.55 | JT | for bad sip servers |
15:46.59 | L|NUX | JunK-Y : nope |
15:46.59 | JT | that are not nat smart |
15:47.08 | JunK-Y | linux: what does: dialplan show s@context1 says? |
15:47.20 | Opperior | well, I won't deny the possibility exists |
15:47.21 | JunK-Y | linux: ya just said yes to Gido-E. |
15:48.17 | L|NUX | wait |
15:48.38 | L|NUX | show dialplan ? |
15:48.43 | JunK-Y | if 1.2, yes |
15:48.53 | JunK-Y | show dialplan s@ur_context |
15:49.36 | L|NUX | ok |
15:49.46 | L|NUX | <PROTECTED> |
15:49.53 | *** join/#asterisk spanglesontoast (n=edd@eddland.plus.com) |
15:50.04 | spanglesontoast | hmm |
15:50.10 | L|NUX | <PROTECTED> |
15:50.21 | L|NUX | <PROTECTED> |
15:50.59 | JunK-Y | and when yo're calling that s,1, whats the output in the CLI? |
15:51.47 | L|NUX | wait |
15:52.06 | JunK-Y | and why are ya passing all these args? they are automatiquely passed to the agi anyways, ya can directly grab them in ur agi. |
15:52.16 | L|NUX | well because AGI need them :) |
15:52.17 | L|NUX | hehe |
15:52.30 | L|NUX | JunK-Y : AGI has been written in that way |
15:52.30 | JunK-Y | they are all passed by default to agi |
15:52.35 | L|NUX | yes |
15:52.42 | JunK-Y | so you dont need to pass them. |
15:52.56 | JunK-Y | do agi debug, ya will see rxing from * |
15:53.03 | L|NUX | one thing i want to know |
15:53.13 | L|NUX | hwo can i get log of sepcific number |
15:53.20 | L|NUX | like there are too many calls coming |
15:53.49 | JunK-Y | actually, ya cant directly in *. ya have to handle it urself. |
15:53.57 | L|NUX | okies |
15:53.59 | L|NUX | logs |
15:53.59 | L|NUX | :) |
15:54.02 | L|NUX | full will help me |
15:54.03 | L|NUX | :) |
15:54.04 | JunK-Y | bingo. |
15:54.08 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
15:54.24 | JunK-Y | but in the future, ya will be able to filter CLI output. |
15:54.40 | *** join/#asterisk ping2921 (n=marc3234@206-248-178-169.dsl.teksavvy.com) |
15:55.49 | L|NUX | hummm |
15:55.49 | L|NUX | ok |
15:56.09 | xpot | anyone know how to get automon to work properly? |
15:56.53 | JunK-Y | linux: so ur cli output? |
15:58.34 | *** join/#asterisk lukketto (n=lukketto@82.59.103.134) |
15:59.27 | L|NUX | wait |
16:00.27 | *** join/#asterisk stony (n=steinche@p57b38ac1.dip0.t-ipconnect.de) |
16:02.11 | *** join/#asterisk Hmmhesays (n=Neg@24-117-131-41.cpe.cableone.net) |
16:02.40 | Hmmhesays | is it possible in linux to check what init state was used when you booted it? |
16:03.18 | *** join/#asterisk astersip (i=53f08b07@gateway/web/cgi-irc/ircatwork.com/x-9390171b59d5b8a7) |
16:03.33 | Qwell[] | Hmmhesays: not really |
16:03.47 | Qwell[] | I mean, if you check inittab you can see the default, but that won't tell you if you used single or not |
16:03.49 | astersip | hi |
16:04.13 | astersip | if someone can help me....i have a E1 width 30 numbers |
16:04.16 | Hmmhesays | yeah thats what I thought |
16:05.38 | astersip | i would like to make that one extension when make a outbound call |
16:05.38 | astersip | that they got out by one of thouse 30 numbers |
16:06.18 | *** join/#asterisk zmef420 (n=zmef420@metarb3-pool2-40.mtco.com) |
16:06.51 | *** join/#asterisk kanelbullar (n=kanelbul@83.240.200.92) |
16:07.16 | *** join/#asterisk spanglesontoast (n=edd@eddland.plus.com) |
16:07.20 | spanglesontoast | anyone played with voipuser ? |
16:07.26 | [TK]D-Fender | astersip: Set(CALLERID(num)=1234567) , Set(CALLERID(name)=Just Me) |
16:07.39 | [TK]D-Fender | spanglesontoast: Added that peer entry like the sample said yet? :) |
16:08.02 | spanglesontoast | what the one that calls throug to the correct extension ? |
16:08.18 | [TK]D-Fender | spanglesontoast: the one that lets you dial out... |
16:08.24 | spanglesontoast | yea it dials up |
16:08.31 | spanglesontoast | but says congestion |
16:09.03 | spanglesontoast | also upgraded to 1.4.2 |
16:09.04 | [TK]D-Fender | spanglesontoast: We shouldn't even have to ASK for you to pastebin up the CLI output of that failed call and your matching configs w/o passwords. Just FYI.... |
16:09.27 | astersip | [TK]D-Fender: where do i put that ? i'm using trixbox and freepbx |
16:09.38 | [TK]D-Fender | ~trixbox |
16:09.46 | jbot | trixbox is probably junk - avoid. It is also unable to be supported here. Try joining #freepbx and asking there, or use the trixbox forums at http://www.trixbox.org/modules/newbb/ |
16:09.47 | [TK]D-Fender | ~freepbx |
16:09.50 | jbot | i heard freepbx is unable to be supported here. It's a complex piece of dialplan and GUI, and can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
16:10.05 | [TK]D-Fender | Qwell[]: You've been... buys ;) |
16:10.09 | [TK]D-Fender | busy* |
16:10.13 | Qwell[] | I don't know what you're talking about. |
16:10.31 | [TK]D-Fender | Qwell[]: Denial.... its not jsut a river in Egypt ;) |
16:11.09 | Mercestes | morning, society |
16:11.14 | gambolputty | No, its a state of the union. |
16:11.19 | gambolputty | onion |
16:11.20 | [TK]D-Fender | Qwell[]: Fonality has a direct business relationship with Fonality do they not? |
16:11.30 | Qwell[] | I would hope so? |
16:11.59 | Qwell[] | [TK]D-Fender: re-read and re-ask your question :) |
16:12.06 | SwK[Work] | hah |
16:12.06 | [TK]D-Fender | Qwell[]: Just wondering, because looking at their new Trixbox "appliance (read : full server in day-glo case), they seem to be using Sangoma cards... |
16:12.20 | Qwell[] | it made no sense at all, heh |
16:12.28 | SwK[Work] | qwell[] hey qwell has a direct business relationship with qwell right? |
16:12.31 | [TK]D-Fender | Qwell[]: : Yeah... cross-wire . Clearly meant to say Digium.... |
16:12.36 | Qwell[] | [TK]D-Fender: no |
16:12.42 | Qwell[] | I mean |
16:12.45 | Qwell[] | no |
16:12.49 | [TK]D-Fender | :) |
16:12.53 | Mercestes | lol |
16:12.59 | [TK]D-Fender | ok, explains a bit... |
16:13.02 | Qwell[] | I don't know :P |
16:13.05 | Mercestes | yay, we got english and spainish translations. |
16:13.13 | [TK]D-Fender | For some reason I just sort of figured they did... |
16:13.29 | *** join/#asterisk uwe (n=uwe1@dogbert.palnet.com) |
16:18.06 | spanglesontoast | hmm what version of the sip protocol does 1.4.2 use ? |
16:20.37 | [TK]D-Fender | spanglesontoast: Same as before with a few more RFC's for SLA |
16:20.49 | spanglesontoast | ok |
16:21.05 | SwK[Work] | any telemarketers around? PM me have a question for you |
16:22.55 | spanglesontoast | the annoying thing is theres lots of posts about the issue fender |
16:22.58 | Mercestes | Of course, all telemarketers hang out in freenode: #asterisk all of us being geeks and all. |
16:23.04 | spanglesontoast | but no one seems to resolve it |
16:23.10 | *** join/#asterisk Xen^ (n=linux@unaffiliated/lnux/x-10290) |
16:23.16 | Mercestes | if you want telemarketers just post your phone # and I'll hook you up with all you can handle. |
16:23.21 | [TK]D-Fender | spanglesontoast: What annoying is you keep complaining and don't "show us the money" :) |
16:23.35 | [TK]D-Fender | spanglesontoast: Stop whining and start pastebin-ing! |
16:24.00 | spanglesontoast | paste what lol I reckon it's voipusers end |
16:24.32 | SwK[Work] | mercestes you would be suprised how many hang out in here and use asterisk based dialers |
16:24.46 | SwK[Work] | mercestes I sell dialers to them all time |
16:24.46 | Mercestes | SwK[Work], Only in secret. |
16:24.55 | SwK[Work] | thats why I said PM me :P |
16:24.58 | Mercestes | asterisk is not pro+telemarketing bastards. |
16:25.08 | Mercestes | but thanks for identifying yourself. |
16:25.23 | SwK[Work] | maybe true, but they are a large market segment that spends a lotta money |
16:25.53 | Mercestes | ON annoying people in futile and ineffective marketing techniques that preys upon the retarded and senile. |
16:26.01 | SwK[Work] | sure |
16:26.02 | SwK[Work] | why not |
16:26.06 | [TK]D-Fender | spanglesontoast: Last we saw you didn't do it right, and you still are not showing us that you've corrected this or have any valid claim for it to be their fault. |
16:26.20 | [TK]D-Fender | spanglesontoast: You are wasting time with this approach. |
16:26.22 | SwK[Work] | only a moron would assume it doesnt work... cause obviously it does work... or they wouldnt be pouring the kind of money into it that they are |
16:26.29 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
16:26.55 | Mercestes | SwK[Work], Of course it works, there are lots of retarded and confused old people out there. I said it was immoral and ineffectual. |
16:26.59 | SwK[Work] | its like spam... why do people keep spamming? because idiots read that crap then buy that product |
16:27.09 | Mercestes | .... |
16:27.19 | Mercestes | Ppl spam just to cause internet problems. |
16:27.41 | Mercestes | but I guess if you hit a billion ppl and find 1000 retards with a dollar....you make a pretty good profit on yoru open source software. |
16:27.52 | SwK[Work] | well you yourself just admitted its not ineffectual... as far as morality, who are you to dictate your morality on anyone else... you ahve the DNC |
16:28.03 | Mercestes | but to me your the kind of person who'd sell abused raped females on the streets for cash if you could get away with it. |
16:28.28 | Mercestes | PM me, I have products to make it more effective to prey on idiots . |
16:28.32 | [TK]D-Fender | SwK[Work]: DNC ... yeah... in the words of Dr. Phil "How's that workin' out for ya?" |
16:28.33 | SwK[Work] | mercestes: i did work in a strip club for 2 years does that count? |
16:28.50 | SwK[Work] | heh |
16:28.53 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
16:28.54 | Mercestes | SwK[Work], It just makes me hpapy that I won't be dealing with you in the next life. Is your marketing campaign done? |
16:29.18 | [TK]D-Fender | Mercestes: Next life? You don't even have one NOW! ;) |
16:29.23 | SwK[Work] | actually I'm not marketing anything here... I was wanting a marketers oppinion on something |
16:29.27 | [TK]D-Fender | Mercestes: Get a life! |
16:29.41 | [TK]D-Fender | (and a haircut... and a real job... and .... |
16:29.47 | Mercestes | [TK]D-Fender, ...true. pwned...:( *cries* |
16:30.14 | [TK]D-Fender | Mercestes: You seem to be able to do that nearly on cue... you should consider day-time dramas ;) |
16:30.22 | *** part/#asterisk m4rkl4r (n=markp@outboundemail.uneta.com) |
16:30.32 | Mercestes | [TK]D-Fender, LOL. maybe you've seen me on the Spainish soaps, amigo. |
16:30.42 | *** join/#asterisk qdk (n=qdk@0x50c627be.bynxx11.adsl-dhcp.tele.dk) |
16:30.47 | errr | Im not sure what to call this feature Im looking for but what I want to do is be able to put someone on hold in my office, then head to our server room and pick put the phone, dial a number and be able to transfer the call I had on hold in my office to the phone Im on in the server room. Can this be done, and if so what is this called? |
16:31.14 | puzzled | parking |
16:31.18 | [TK]D-Fender | Mercestes: Nope.... I don't have cable/satelite/etc. I also haven't even plugged in the 'ol rabbit ears (to my *52" HDTV*) |
16:31.38 | errr | puzzled: ok thanks |
16:31.44 | [TK]D-Fender | errr: Call Parking. Look it up on the WIKI |
16:31.51 | puzzled | errr: configure it in one of the conf files |
16:31.53 | b11d | TK, same here man.. |
16:32.09 | Mercestes | [TK]D-Fender, Just for IRC huh? Nice. D-Fender, wireless keyboard and a 52" plasma mounted on the wall, on his ergonomic recliner...in his speedos...giving sound technical advice. |
16:32.13 | errr | [TK]D-Fender, puzzled thanks, having a name makes it eaiser to search for :) |
16:32.42 | [TK]D-Fender | Mercestes: No, I use a 22" for my main PC. the HDTV is for my * server :) |
16:33.01 | puzzled | [TK]D-Fender: for some reason ppl from North-America often mention the size of their TV. Why is that? |
16:33.18 | puzzled | I would not even know the size of my TV |
16:33.24 | Mercestes | lol. 52" of * cli baby! |
16:33.25 | [TK]D-Fender | puzzled: Dunno, I did mine for cdramatic contrast to the fact I don't watch TV per-se. |
16:33.51 | [TK]D-Fender | puzzled: Not knowing what you own just makes you an uninformed consumer :) |
16:33.59 | coppice | our TV is only 42", but its what you do with it that counts :-) |
16:34.49 | puzzled | funny because I have yet to see a decent quality American TV and the content sucks soo much that I'd rather have a tiny one hidden in the closet :) |
16:34.54 | [TK]D-Fender | coppice: But is is..... Plug & play? ;) |
16:35.56 | puzzled | ah, coppice must be off doing that special thing with his tv |
16:36.45 | puzzled | [TK]D-Fender: no deal. I'll never set a foot in Nevada again |
16:37.06 | pif | tzafrir: I just rebuilt your packages with misdn support without problems |
16:37.28 | puzzled | pif: but did it pass a call? |
16:37.57 | pif | not yet, next thing |
16:37.59 | *** join/#asterisk nortex (n=breeves@snapper.titanspecialties.com) |
16:39.20 | [TK]D-Fender | ok, heading out for a bit, back later |
16:40.09 | Mercestes | take care fender. |
16:41.11 | *** join/#asterisk FuriousGeorge (n=brian@ool-43536ea8.dyn.optonline.net) |
16:48.15 | spanglesontoast | wtf |
16:48.19 | spanglesontoast | I just rang japan |
16:48.27 | spanglesontoast | well I'm getting somewhere :) |
16:48.54 | pif | mushi mushi? |
16:48.55 | *** join/#asterisk __freedom__lover (n=eduardo@clipper.provale.com.br) |
16:49.11 | Mercestes | spanglesontoast, Are you ever going to post yoru error/sip.conf or are you just going to trol forever?? |
16:49.19 | __freedom__lover | hi all, good afternoon |
16:49.28 | aydiosmio | can someone point me to the Remote-Party-Identity configuration lines for *? |
16:49.28 | Mercestes | Afternoon, freedom. |
16:49.37 | b11d | anyone know of a good 4+ port KVM that supports both DVI & VGA ? |
16:49.38 | Mercestes | aydiosmio, Is that anything like callerID? |
16:49.41 | __freedom__lover | someone here uses asterisk on freebsd? |
16:49.45 | b11d | i do |
16:49.52 | Mercestes | b11d, netgear makes one. Really good |
16:49.56 | b11d | oh yeah? |
16:49.59 | aydiosmio | Mercestes: kind of, it's a proxy identifier like Asserted-Identity |
16:50.12 | Mercestes | aydiosmio, Is that an ss7 thing? |
16:50.15 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
16:50.18 | __freedom__lover | b11d: you have a asterisk running on freebsd? |
16:50.22 | aydiosmio | but I know asterisk doesn't support Asserted Identity natively |
16:50.29 | aydiosmio | have to use SIPAddHeader |
16:50.39 | b11d | i dont see any netgear kvm's |
16:50.46 | aydiosmio | Mercestes: no, sip thing |
16:50.48 | b11d | __freedom__lover.. yes.. |
16:50.57 | Mercestes | aydiosmio, I wouldn't know then. I would put in a feature request because I dont' think that's a feature. |
16:51.16 | aydiosmio | I know I've seen it somehwere |
16:51.19 | aydiosmio | ah! here it is |
16:51.28 | Mercestes | b11d, I'm pretty sure it's netgear. I'll look it up tonight. They make good 2 port and 4ports. You can find them at bestbuyand stuff, pretty cheap |
16:51.30 | aydiosmio | http://voip-info.org/wiki/view/P-Asserted-Identity+and+Remote-Party-ID+header |
16:51.42 | aydiosmio | sendrpid = yes |
16:51.48 | b11d | both DVI and VGA though eh |
16:52.07 | Mercestes | Ah cool, thanks. I added it to my favorites so I can read it later. |
16:52.20 | Mercestes | See folks?? Good example! he finds the answer...then posts it so I can educate myself. :) |
16:52.42 | Mercestes | b11d: Oh. try Telepathy. I honestly have no clue. ;) |
16:52.54 | Mercestes | I don't play with DVI KVMs. |
16:52.56 | *** join/#asterisk jm|laptop (n=jm@sentry.flags.co.uk) |
16:53.05 | b11d | Telepathy? Is that a VoIP company? |
16:53.13 | aydiosmio | Mercestes: Verizon requires either RPID or Asserted ID for wholesale termination |
16:53.22 | aydiosmio | that's why I'm setting it up |
16:53.56 | Mercestes | b11d: yea. |
16:54.08 | Mercestes | aydiosmio, Cool. Let us know how Verizon is. I'm watching that company. |
16:54.15 | b11d | neat. |
16:54.27 | Mercestes | They're bringing fiber to my area soon. They have some cool residential stuff. |
16:54.33 | *** join/#asterisk webman (n=adamg@52.87.233.220.exetel.com.au) |
16:55.07 | aydiosmio | yeah FIOS is leaking into NYC slowly |
16:55.20 | aydiosmio | hopefully Queens will get hooked up bfore I get tired of this city |
16:55.54 | *** join/#asterisk deeperror (n=deeperro@mail.banctel.com) |
16:55.54 | spanglesontoast | come on ;) |
16:56.18 | webman | anyone know how to get hylafax to route incoming faxes when received from iaxmodem?? I've seen to use CALLID4 and also tried to use CIDNAME but both are always blank?? |
16:56.39 | *** join/#asterisk cr4z3d (n=cr4z3d@ip70-162-117-14.ph.ph.cox.net) |
16:57.02 | *** join/#asterisk elpropagandista (n=elpropag@208-106-57-5.dsl.dynamic.sonic.net) |
17:00.32 | cr4z3d | can anyone help me get my VoIP up and running with NuFone? i've been trying for about a day now. iax show registry shows i'm registered and when i turn on iax2 debug all i see is tx-frame retry and rx-frame retry over and over |
17:01.09 | webman | cr4z3d: are you trying to get inbound or outbound working ? |
17:01.13 | cr4z3d | inbound |
17:01.27 | *** join/#asterisk dps (n=dps@129.64.30.213.rev.vodafone.pt) |
17:01.32 | dps | Hello |
17:01.37 | webman | cr4z3d: so when you call your number do you see any rx traffic ??? |
17:01.48 | cr4z3d | nop |
17:02.04 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.static.bezeqint.net) |
17:02.28 | *** join/#asterisk CunningPike (n=CunningP@dhcp-10-234.district.north-van.bc.ca) |
17:02.38 | dps | Can anyone give me a clue how do i make the rtp pass throu both phones without asterisk staying in the middle? |
17:03.07 | webman | cr4z3d: then either you are not registered correctly, or else your firewall/something is blocking the traffic |
17:03.13 | cr4z3d | webman, does that mean nufone is screwed up? |
17:03.20 | cr4z3d | or that they can't get it to me |
17:03.55 | webman | cr4z3d: I'd usually blame firewall/something.... check what IP asterisk says it has registered with?? |
17:04.21 | cr4z3d | last night someone told me to forward udp port 4569 on my router |
17:04.25 | cr4z3d | i disabled the router spi firewall |
17:04.34 | cr4z3d | and ubuntu has no default installed firewall/i never installed one |
17:05.04 | dps | cr4z3d: type iptables -L |
17:05.06 | webman | try and use "tcpdump port 4569" to see if you can see the incoming traffic |
17:05.21 | dps | cr4z3d: and see if there's any rule |
17:05.42 | cr4z3d | iptables -L has nothing |
17:05.57 | *** join/#asterisk darylvoip (n=darylvoi@c-71-224-42-97.hsd1.pa.comcast.net) |
17:06.08 | cr4z3d | webman, so call the number and do tcpdump? |
17:06.29 | cr4z3d | tcpdump gave me no suitable device found |
17:06.33 | webman | also try to get asterisk to register, and then ASAP call the number (maybe your firewall has a short timeout) |
17:06.50 | dps | For the luve of god and madonna and my cousin, i'm about to go get drunk and pissed with everything that as eletricity, i need to make an rtp stream pass by 2 phone that are registered on asterisk and the bastard stays in the middle doing rtp proxy |
17:06.56 | cr4z3d | oh would QoS settings mess any of this up? |
17:06.59 | darylvoip | Hey...anyone around that can help me figure out why asterisk is giving me a "Looking for s in default" followed by a 404 in my new config (replaced some openser boxes with a sansay sbc)? |
17:07.13 | dps | Nerf asterisk rtp proxy! |
17:07.46 | darylvoip | dps: sounds like you have one of my unsolved problems too |
17:07.57 | webman | dps: ensure no nat between the phones or asterisk (ie all public IP or all same LAN) |
17:08.05 | dps | Same switch |
17:08.09 | dps | same network |
17:08.13 | dps | same... same |
17:08.20 | dps | same codec |
17:08.23 | webman | dps: check sip.conf then |
17:08.35 | dps | yes.... keep going? |
17:08.51 | webman | dps: canreinvite=yes |
17:08.59 | webman | for each of the phones |
17:09.02 | dps | yes |
17:09.09 | Qwell[] | nat=no? |
17:09.10 | dps | it's on canreinvite=yes |
17:09.14 | dps | nat is never |
17:09.16 | bulle | perhaps the phones themselves doesnt reinvite ? |
17:09.32 | cr4z3d | hm.. still can't get any results |
17:09.32 | webman | dps: also check what you set for the localnet settings |
17:09.49 | cr4z3d | should i try setting the asterisk as a DMZ |
17:09.59 | cr4z3d | just to rule out any nat/routing issues? |
17:10.02 | *** join/#asterisk ownerge (n=sergo@217.147.230.35) |
17:10.09 | dps | webman: localnet in sip.conf is 10.0.0.0/24 |
17:10.09 | ownerge | hi guys |
17:10.10 | Qwell[] | cr4z3d: Making it DMZ doesn't give it a public IP |
17:10.12 | ownerge | need help |
17:10.17 | webman | cr4z3d: well, could be worth a try ... |
17:10.23 | ownerge | have problem connection SIP phone to my ASterisk server |
17:10.29 | cr4z3d | oh good point Qwell |
17:10.36 | dps | yes ownerge what's up? |
17:10.38 | ownerge | can any1 help me? |
17:10.47 | cr4z3d | but as far as QoS goes |
17:10.48 | webman | cr4z3d: basically your first step is to try and receive *something* .... |
17:10.59 | cr4z3d | yeah i know i can't get anything |
17:11.02 | cr4z3d | but it says i'm registered |
17:11.23 | ownerge | look |
17:11.32 | ownerge | I have Cisco 7906 phone |
17:11.37 | webman | cr4z3d: if it says you are registered, then you must have had *SOME* two way traffic.... |
17:11.39 | Qwell[] | 7906? |
17:11.41 | ownerge | i've created an XML file |
17:11.46 | ownerge | uploaded to phone |
17:11.51 | ownerge | trying to dial |
17:11.55 | ownerge | nothing happens |
17:12.03 | ownerge | phone says that it tries to register |
17:12.08 | dps | asterisk .-rvvvvvvvvvv |
17:12.12 | webman | anyone using iaxmodem and know how to set the CID info for faxdispatch to route properly?? |
17:12.15 | dps | sip show peers |
17:12.16 | cr4z3d | yeah definitely says registered under iax2 show registry |
17:12.17 | phearless | how can I do a (shell?) script that get the list of the "sip show channels" ? |
17:12.20 | dps | sip show register |
17:12.21 | ownerge | 1 sec |
17:12.24 | ownerge | will check now |
17:12.28 | dps | good |
17:12.28 | phearless | is there a command to launch CLI commands? |
17:12.35 | phearless | that I can use in a script |
17:12.36 | *** join/#asterisk s1gny|wrk (n=s1gny@p54916365.dip.t-dialin.net) |
17:12.40 | dps | asterisk -r |
17:12.42 | *** part/#asterisk s1gny|wrk (n=s1gny@p54916365.dip.t-dialin.net) |
17:13.01 | webman | phearless: asterisk -rx 'some command' |
17:13.38 | dps | god! oh GOD! doesn't the sip idea is to make the rtp port of each user agent pass in the sip header and let the damnn phone talk to each other? |
17:13.41 | cr4z3d | i do have QoS enabled on my router though you think that would do anything webman? |
17:13.48 | ownerge | 911/911 (Unspecified) D N 0 UNKNOWN |
17:13.52 | ownerge | that's what I got |
17:14.02 | ownerge | it's a my test extention :D |
17:14.02 | phearless | thanks webman |
17:14.03 | ownerge | 911 |
17:14.17 | dps | use qualify=yes |
17:14.30 | dps | to see if it's reachable |
17:14.39 | *** join/#asterisk Assid (n=assid@59.183.60.24) |
17:14.39 | webman | cr4z3d: I have no idea in relation to your config.... but at some point you need to look at the traffic / network level before trying to go further. I like tcpdump for that.... |
17:15.15 | cr4z3d | tcpdump gave me nothing when i tried it on 4569 |
17:15.21 | ownerge | it is set to yes |
17:15.26 | ownerge | also |
17:15.29 | ownerge | forgot to tell |
17:15.31 | bulle | cr4z3d: is surely must atleast show outgoing traffic |
17:15.51 | *** join/#asterisk xo8ox (n=pride_32@wsip-66-210-250-2.ph.ph.cox.net) |
17:15.57 | ownerge | I've tried this extentiuon from VIOP pc client |
17:15.58 | ownerge | it works |
17:16.12 | ownerge | can be there any problem regarding dialplans or something like that |
17:16.13 | ownerge | ? |
17:16.38 | cr4z3d | bulle, tcpdump 4569 gives me no suitable device found |
17:16.53 | bulle | uh ? |
17:17.38 | cr4z3d | same with tcpdump port 4569 |
17:17.46 | webman | cr4z3d: are you using linux?? what network devices do you have ?? |
17:17.55 | *** join/#asterisk phillipk (n=pkey@216.248.143.77) |
17:17.58 | cr4z3d | ooh shit |
17:18.05 | cr4z3d | wasn't running it with root |
17:18.06 | Assid | err.. can someone help me with something |
17:18.07 | cr4z3d | i thought i was |
17:18.15 | Assid | all of a sudden my polycom doesnt want to register anymore |
17:18.28 | cr4z3d | alright it's listening on it |
17:18.33 | cr4z3d | and i'm trying to call in |
17:18.43 | cr4z3d | nothing shows up |
17:18.46 | webman | do a iax2 reload |
17:18.57 | ownerge | I'm mad |
17:19.03 | ownerge | this thing still not working |
17:19.03 | cr4z3d | nothing |
17:19.03 | ownerge | :@ |
17:19.17 | webman | you should at least see the outbound register request |
17:19.20 | cr4z3d | -- Registered IAX2 to '66.225.202.72', who sees us as 70.162.117.14:4569 with no messages waiting |
17:19.25 | cr4z3d | is what i got in cli |
17:19.28 | webman | how many network devices do you have ?? |
17:19.31 | cr4z3d | and nothing from tcpdump |
17:19.34 | xo8ox | guys I setup a que in *now but when I call the que I only hear the music on hold ! |
17:19.40 | cr4z3d | a wireless and an ethernet |
17:19.41 | *** join/#asterisk l2cache (n=ghansen@64.128.254.98) |
17:19.44 | cr4z3d | it's on wireless right now |
17:19.52 | cr4z3d | which is eth1 |
17:19.54 | *** part/#asterisk elpropagandista (n=elpropag@208-106-57-5.dsl.dynamic.sonic.net) |
17:20.05 | webman | so you need to do tcpdump -i eth1 port 4569 |
17:20.08 | cr4z3d | oh of course tcpdump is on eth0 |
17:20.32 | Assid | i keep getting back SIP/2.0 401 Unauthorized |
17:20.54 | cr4z3d | ok now it captured some packets |
17:21.10 | webman | so now try to call in and you should get more packets |
17:21.29 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
17:21.43 | cr4z3d | yep i got some packets when i called in |
17:21.53 | webman | BTW: in my experience, some firewalls will drop the translation rules in under 30 seconds.... so you gotta be quick :) |
17:22.01 | ownerge | look |
17:22.06 | ownerge | i have only 1 question |
17:22.09 | webman | cr4z3d: ok, did you see anything from asterisk ? |
17:22.13 | ownerge | if phone is not registered on Asterisk |
17:22.20 | ownerge | he can't create package |
17:22.24 | ownerge | right? |
17:22.38 | cr4z3d | asterisk cli gave me nothing |
17:22.42 | webman | ownerge: no, not registered means can't receive calls, but the phone could make calls |
17:22.49 | webman | cr4z3d: iax2 debug |
17:22.51 | cr4z3d | and the tcpdump is coming up with stuff every few seconds |
17:23.03 | ownerge | mine can't do anything :D |
17:23.07 | ownerge | can't call |
17:23.09 | webman | cr4z3d: and make sure you started asterisk with asterisk -vvvvdc |
17:23.13 | ownerge | can't receive'em |
17:23.21 | ownerge | this is what i got in output |
17:23.22 | cr4z3d | webman, with that on i get a bunch of tx-frame retry results every couple of minutes |
17:23.24 | *** join/#asterisk Gunnar (n=gunnar@nat.sigmasoft.com) |
17:23.31 | webman | ownerge: then you have a config error |
17:23.38 | cr4z3d | webman, i have asterisk -vvvc and then did set debug 10 and set verbose 10 |
17:23.39 | ownerge | 911/911 (Unspecified) D N 0 UNKNOWN |
17:23.52 | ownerge | other phones work |
17:23.55 | ownerge | with same cfg |
17:24.12 | webman | cr4z3d: I don't think that is always enough, I think you also need to start with d.... also check /etc/asterisk/logger.conf |
17:25.12 | cr4z3d | ok i restarted it with -vvvvdc |
17:25.18 | cr4z3d | starts up fine |
17:25.39 | Dr-Linux | hi |
17:25.43 | Dr-Linux | what's wrong with this: |
17:25.44 | Dr-Linux | exten => 32345,1,Answer |
17:25.44 | Dr-Linux | exten => h,1,DeadAGI,pager2sms.agi|03339615969\@ufone.com |
17:26.04 | Assid | err.. can someone help me with this polycom registration? |
17:26.06 | *** join/#asterisk thinwires (n=thinwire@24-49-196-96.kntnny.adelphia.net) |
17:26.07 | b11d | shoudlnt that be like, 32345,2, DeadAGi ? |
17:26.13 | Assid | it just doesnt want to work anymore |
17:26.31 | cr4z3d | webman, i'm in logger.conf should i change anything from the default? |
17:26.32 | __freedom__lover | does someone use asterisk on freebsd |
17:26.35 | __freedom__lover | ? |
17:26.45 | Dr-Linux | b11d: DeadAGi only works with h |
17:26.47 | thinwires | hi, does anyone here use Cisco 7941G IP Phones? |
17:26.50 | webman | cr4z3d: just make sure the console => line has debug as well |
17:27.09 | b11d | oh |
17:27.17 | webman | Dr-Linux: looks ok to me.... why should there be something wrong with it?? |
17:27.23 | cr4z3d | webman, nope just says console => notice,warning,error |
17:27.31 | cr4z3d | so i should through debug a t the end? |
17:27.34 | webman | assid: try reboot the phone :) |
17:27.45 | Dr-Linux | webman: bcoz it's not working for me |
17:27.47 | webman | cr4z3d: yep add debug at the end |
17:27.49 | Assid | webman: tried that.. even rebooted the modem |
17:28.01 | Assid | webman: http://pastebin.ca/406719 |
17:28.07 | webman | Dr-Linux: in what way... you haven't provided sufficient information |
17:28.13 | Dr-Linux | i'm calling this extension and on CLI i can only see Answer execution |
17:28.31 | Dr-Linux | webman: |
17:28.32 | Dr-Linux | nmmm |
17:28.35 | *** join/#asterisk ToyMan (n=Stuart@12.23.30.130) |
17:28.39 | cr4z3d | webman, ok reloaded it with debug in the consol line |
17:29.00 | *** join/#asterisk Mnabil (n=Mnabil@196.202.44.224) |
17:29.15 | webman | cr4z3d: now try the iax2 reload and then the inbound call |
17:29.55 | Mnabil | hello, i'm installed asterisk , but i can get asterisk-core-sound working with me, ass their ext. is alaw ? who can i got them work or install them |
17:30.03 | Dr-Linux | webman: why the doc suggested DeadAGI? |
17:30.33 | thinwires | Ok, quick question, Cisco 7941? or PolycomIP601? |
17:30.35 | cr4z3d | webman, still nothing but now i get debug messages like Mar 22 10:29:09 DEBUG[16482]: chan_iax2.c:7752 iax2_do_register: Registration created on call 2 |
17:30.51 | ownerge | I found solution |
17:30.55 | webman | Dr-Linux: dunno, to me it looks like you need to call from one channel to another, and when one channel hangs up the other continues to the deadagi |
17:30.56 | *** join/#asterisk Assid (n=assid@203.212.204.107) |
17:30.56 | b11d | polycom 601 |
17:30.57 | ownerge | very gooooooood solution |
17:30.58 | b11d | no questiojn :) |
17:31.05 | ownerge | will user analog lines |
17:31.06 | ownerge | :D |
17:31.10 | ownerge | no VOIP |
17:31.14 | ownerge | NO VOIP PHONES |
17:31.20 | ownerge | NO PAIN IN **** |
17:31.21 | ownerge | :D |
17:31.24 | b11d | yeah voip's on it's way out anyways.. |
17:31.29 | Dr-Linux | hhm.. :S |
17:31.32 | cr4z3d | is it really? |
17:31.33 | b11d | :) |
17:31.34 | bulle | no pain in 4-letter-word |
17:31.36 | thinwires | b11d: do you know if the polycom IP601 supports XML? |
17:31.40 | b11d | yeah it does |
17:31.44 | bulle | ownerge: what four letter word were you thinking of ? |
17:31.44 | ownerge | :D |
17:31.45 | webman | cr4z3d: do you see any packets/messages when you call your did ?? |
17:31.51 | b11d | DICK ? |
17:31.55 | b11d | BUTT? |
17:32.01 | b11d | ASSS ? |
17:32.02 | b11d | :P |
17:32.04 | ownerge | actualy i wanted to type 3 letters |
17:32.05 | ownerge | :D |
17:32.12 | cr4z3d | webman, no i keep getting the same kind of packets it looks like |
17:32.15 | ownerge | but in my conditions |
17:32.18 | webman | PSTN is a four letter work :) |
17:32.25 | cr4z3d | webman, 10:32:10.119359 IP unknown.ord.scnet.net.iax > 192.168.1.109.iax: UDP, length 48 |
17:32.25 | cr4z3d | 10:32:10.119694 IP 192.168.1.109.iax > unknown.ord.scnet.net.iax: UDP, length 12 |
17:32.27 | webman | s/work/word |
17:32.33 | ownerge | after all this pain in a *** with this Cisco 7906 |
17:32.33 | b11d | well thinwires.. XML as in how? |
17:32.34 | *** join/#asterisk robin01 (n=robin@010.152.dsl.concepts.nl) |
17:32.38 | b11d | where do you want to "use" xml? |
17:32.39 | ownerge | mistakes can be forgiven :D |
17:32.40 | b11d | in relation to the 601 |
17:32.51 | robin01 | hello all |
17:32.56 | l2cache | hi |
17:33.01 | b11d | holland for life! |
17:33.01 | cr4z3d | webman, also iax2 debug keeps saying tx-frame retry and rx-frame retry messages |
17:33.03 | Assid | webman rebooted for the 6th time.. lets see |
17:33.10 | robin01 | I'm having trouble whit installing the D410P |
17:33.15 | webman | cr4z3d: ok, at least you know you have inbound traffic to the box.... |
17:33.31 | robin01 | can anybody help me |
17:33.43 | l2cache | what is your question robin01? |
17:34.02 | robin01 | well I installed the card and it is detected |
17:34.03 | Assid | finally worked.. yeay! |
17:34.19 | webman | cr4z3d: get a log of what happens, as I said, start with iax2 reload and then attempt the call. If you can, edit the log to show where you made the call, and post it to pastebin.ca or something |
17:34.21 | cr4z3d | webman, this wouldn't have anything to do with my iax.conf, sip.conf, or extensions.conf at this point correct? |
17:34.26 | thinwires | b11d: does it support XML phone directories? |
17:34.33 | b11d | yeah, it does. |
17:34.41 | b11d | all the polycom's use xml phone directories |
17:34.44 | robin01 | at compiliung asterisk 1.4.2 the misdn driver is not detected |
17:34.57 | cr4z3d | webman, alright i'll get that log for you |
17:34.59 | robin01 | but I can see that the card is operational |
17:35.11 | webman | cr4z3d: well, that is what we should see in the debug output, if it is auth error, then problem is in iax.conf, or it might be no extension / context error |
17:35.51 | robin01 | does anybody have any experiance with the B410P card |
17:35.52 | robin01 | ? |
17:35.56 | *** join/#asterisk qdk (n=qdk@0x50c627be.bynxx11.adsl-dhcp.tele.dk) |
17:35.58 | cr4z3d | webman, ok cuz yeah i'm not sure about my extension.conf setup/sip setup but at this point i'm just trying to get it connected correctly |
17:36.22 | thinwires | b11d: awesome, thanks, I have a cisco 7941 on my desk now and I want to throw it out the frackin window. Nothing but trouble from starting it up to getting tech support |
17:36.30 | zoa | robin01: yes |
17:36.39 | robin01 | ok great zoa |
17:37.11 | b11d | yeah |
17:37.15 | b11d | the cisco phones pretty much suck ass |
17:37.17 | robin01 | zoa: I followed the installation instruction in the cards manual, but no positive result |
17:37.18 | b11d | they look nice though :) |
17:37.24 | robin01 | i'm using opensuse 10.2 |
17:37.34 | *** join/#asterisk qdk (n=qdk@0x50c627be.bynxx11.adsl-dhcp.tele.dk) |
17:37.40 | thinwires | oh yeah, it looks sexy... but it's like dating a hot girl with a horrible lough |
17:38.28 | l2cache | polys are the best |
17:38.34 | bulle | nokias are the best |
17:38.51 | l2cache | i have a 501 on my desk right now...beautiful |
17:38.55 | webman | cr4z3d: hows that debug going?? I gotta go soon, it's almost 5am :( |
17:39.04 | b11d | yeah im all about my 501 too |
17:39.04 | robin01 | can anybody help ? |
17:39.09 | b11d | i |
17:39.19 | cr4z3d | webman, i'm pasting the stuff in a pastebin right now |
17:39.25 | b11d | i want an adjustable backrest, a backlit display, and a built-in ATA and i'd be in paradise |
17:39.28 | *** join/#asterisk sysreq (n=sysreq@modemcable171.134-81-70.mc.videotron.ca) |
17:39.31 | l2cache | we run call centers with over 150 poly 301's...they are awesome |
17:39.33 | b11d | yeah the 601's are a little to pretentious for me |
17:39.46 | b11d | i dislike the 301s personally.. the 430's are nice. |
17:40.07 | webman | b11d: yeah, but they are great when you don't have to pay for them yourself :) |
17:40.11 | b11d | aye :) |
17:40.44 | webman | I read there is a new phone 650 or something.... meant to have a better quality codec.... |
17:40.49 | b11d | yeah the HD audio |
17:40.53 | b11d | it's pretty nice |
17:41.02 | cr4z3d | webman, http://pastebin.ca/406746 and thanks for helping me out so much. i'm guessing you're in the UK since it's 5am |
17:41.12 | b11d | the soundstation ip 4000 conference room phone is nice too, just overpriced. |
17:41.38 | Assid | bulle: i got a e61 |
17:41.38 | thinwires | well for me price isn't a big deal, one time expense... any suggestions on which phone to look at? |
17:41.38 | b11d | i used to live up near Darwin for a short time.. AU rocks. |
17:43.26 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
17:43.29 | ownerge | ppl |
17:43.32 | ownerge | another problem |
17:43.36 | webman | cr4z3d: at what time did you attempt to call your number though?? |
17:43.42 | ownerge | I just deleted old extention |
17:43.45 | ownerge | created another one |
17:43.47 | cr4z3d | well the end of 10:37 |
17:43.49 | ownerge | but |
17:43.51 | b11d | thinwires.. get one of each then. get a 430/501/650 |
17:43.58 | ownerge | wher i use sip show user |
17:44.08 | ownerge | i can c that old one still exists |
17:44.09 | *** join/#asterisk qdk (n=qdk@0x50c627be.bynxx11.adsl-dhcp.tele.dk) |
17:44.15 | ownerge | but new one doesn't show up |
17:44.36 | thinwires | lol, I just need one bloody phone to pick here, i'm looking into this HD Voice thing now, looks sexy |
17:44.40 | ownerge | typed extentions reload |
17:44.47 | drako | zaptel is independent of asterisk right |
17:44.49 | cr4z3d | webman, called at 10:37 then it turned 10:38 so around 10:37:5something |
17:45.09 | drako | i can have zaptel 1.2.16 with asterisk 1.2.14 |
17:45.10 | drako | ? |
17:45.23 | webman | cr4z3d: you are calling your number from a PSTN or mobile or something ?? |
17:45.30 | cr4z3d | webman, from a mobile |
17:45.53 | webman | cr4z3d: do you get any signal/tone/rva/something on the mobile ? |
17:46.14 | cr4z3d | webman, nothing just silence followed by "the caller is currently unreachable" |
17:46.22 | xo8ox | guys when I do 'asterisk -r' i get this: Unable to connect to remote asterisk |
17:46.29 | webman | cr4z3d: is the number assigned by nufone or is it a ported number? |
17:46.40 | xo8ox | how can I run asterisk ? |
17:46.44 | webman | xo8ox: try asterisk -c instead :) |
17:46.50 | cr4z3d | webman, also if i login to nufone.net i can see all the calls, and it was assigned by nufone |
17:47.19 | *** join/#asterisk Vec (n=Vec@dsl-244-215-172.telkomadsl.co.za) |
17:47.33 | xo8ox | I did and I got bunch of warnings |
17:47.37 | webman | cr4z3d: well, there is no traffic at all generated in relation to the inbound call (that I can see).... so something is screwed |
17:48.00 | cr4z3d | webman, should i contact nufone? or would it be my router blocking |
17:48.09 | webman | xo8ox: warnings are not errors... fix the warnings when you are ready to learn more |
17:48.24 | webman | cr4z3d: what kind of router is it ? |
17:48.53 | cr4z3d | webman, linksys wrt54g w/ DD-WRT firmware |
17:49.03 | webman | cr4z3d: can you make outbound calls through nufone ? |
17:50.38 | cr4z3d | webman, i have no idea never tried.. i haven't setup any of the sip stuff yet |
17:50.38 | webman | cr4z3d: ok, simpler option... download and install diax and use that to register to nufone, and test..... that will rule out asterisk config |
17:50.38 | aydiosmio | so is Asseted-Identity supported in * yet or are we still adding out own headers? |
17:50.38 | aydiosmio | Asserted |
17:50.38 | aydiosmio | this is a pretty important feature now |
17:50.39 | webman | aydiosmio: important for who.... :) |
17:50.39 | *** join/#asterisk generalhan (n=asd@ip67-90-64-2.z64-90-67.customer.algx.net) |
17:50.44 | cr4z3d | webman, ok so if i use diax that means my asterisk config is good ? |
17:50.45 | aydiosmio | for people connecting to wholesale voip providers |
17:50.55 | Mercestes | for people connecting to Verizon you mean. |
17:51.06 | aydiosmio | Verizon Biz support Remote-Party-ID but it's deprecated |
17:51.27 | Mercestes | Use Telepathy. |
17:51.28 | cr4z3d | webman, ah it's for windows i'll have to get my laptop out |
17:51.32 | webman | cr4z3d: no, but it means you are testing the router/internet without all the complex asterisk/linux config :) just a simple windows box with diax which has very few settings |
17:51.36 | aydiosmio | nah there's a few wholesale voip providers requiring it now |
17:51.46 | Mercestes | aydiosmio, I would put in a feature request. |
17:51.50 | aydiosmio | me too |
17:52.02 | aydiosmio | where the heck do I do that? |
17:52.03 | cr4z3d | webman, oh alright so basically at that point i know it's not the router if that works |
17:52.24 | webman | cr4z3d: yep, then you know if it also not nufone problem :) |
17:52.55 | *** join/#asterisk [shodan] (n=shodan@ip213.99-113-216.pppoe4.joliette.intermonde.net) |
17:53.06 | webman | cr4z3d: I gotta get going now.... but good luck with it! hopefully someone else can help you out soon... |
17:53.20 | Mercestes | aydiosmio, bugs.something.com or something. google asterisk bugs |
17:53.31 | b11d | bugs.digium.com |
17:53.36 | *** join/#asterisk shinux__ (n=shinux@86.62.8.178) |
17:53.46 | chefrs | Any idea when I dial an outbound trunk, about 50% of the time it rings once and then just sits there? |
17:53.48 | cr4z3d | webman, thanks, enjoy your sleep haha |
17:54.03 | b11d | that's strange eh chefrs.. what does the console report? |
17:54.30 | Mercestes | chefrs: Because you have a random number generator that 50% of the time does a Dial and the other 50% of the time it does a Ringing(6) and then does an endless loop routine?? |
17:54.50 | chefrs | Might be it. |
17:54.55 | Mercestes | YES! |
17:54.57 | Mercestes | man I'm good |
17:55.00 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
17:55.06 | xo8ox | asterisk wont run |
17:55.08 | chefrs | b11d: Reports a dial out, says a zap channel has answered it, but then the phone just... sits. |
17:55.11 | Mercestes | All without logs of a failed call and a copy of the configuration. |
17:55.20 | chefrs | b11d: Looks just like a normal, successful outbound call. |
17:55.27 | b11d | and the phone rings once, and then stops. |
17:55.31 | chefrs | Yeah. |
17:55.32 | Mercestes | xo8ox: we know. we're sorry. We hope to have a usable product some day. |
17:55.36 | b11d | is this possibly a reinvite issue? |
17:55.43 | b11d | do you have canreinvite=yes or =no in sip.conf? |
17:55.50 | chefrs | Hmm, haven't touched on anything like that. Lemme check. |
17:56.07 | xo8ox | lol |
17:56.17 | b11d | i dont know enough about how that works to say for certain if it is or not. |
17:56.25 | chefrs | reinvite is no |
17:56.40 | chefrs | But it's in extensions, not sip |
17:56.43 | Mercestes | xo8ox Could yyou maybe type out an asterisk -cvvvvvvvvvvvvvvvvv and pastebin what that output says maybe>??? |
17:56.46 | xo8ox | guys seriously I wish I was linux guro but the person who set up our asterisk line quit and now we are stock |
17:56.56 | Mercestes | it's guru. |
17:57.02 | xo8ox | guru |
17:57.03 | xo8ox | hehe |
17:57.08 | xo8ox | whats the difference |
17:57.10 | b11d | oh |
17:57.11 | Mercestes | and stuck |
17:57.21 | Mercestes | and I have a resume. but I'm expensive. |
17:57.25 | l2cache | so your place is hiring? |
17:57.28 | Mercestes | and I need the output of asterisk -cvvvvvvvvvvvvvvvvvvv |
17:57.33 | l2cache | where |
17:57.36 | Mercestes | from your command line, please. |
17:57.38 | Mercestes | in pastebin. |
17:57.41 | Mercestes | ~pastebin |
17:57.51 | jbot | hmm... pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste, or http://rafb.net/paste/, or http://pastebin.com is usually painfully too slow and unresponsive to use, use one of the other pastebin sites, or dpaste.com is a very nice pastebin as well |
17:57.51 | b11d | umm.. reinvite stuff should be in sip.conf according to what I have.. |
17:58.13 | chefrs | b11d: What is your reinvite set to? |
17:58.16 | b11d | no |
17:58.18 | chefrs | b11d: Also, what does it do? |
17:58.31 | wunderkin | xo8ox, i'm local if you need someone to come out |
17:58.32 | b11d | it has to do with handling traffic from phone to phone, or from phone to asterisk, to phone. |
17:58.35 | b11d | if I remember correctly. |
17:59.25 | chefrs | Alright well I'll set it in there and see if it starts to work a bit better. |
17:59.35 | b11d | yeah im not even sure it's related to the problem.. |
17:59.44 | chefrs | Worth a shot. |
17:59.47 | b11d | sure |
17:59.54 | b11d | read voip-info.org's page on sip.conf too |
18:00.03 | spanglesontoast | hmm to use meetme do I need zaptel ? |
18:00.24 | chefrs | Also any idea why when I dial out, the phone company says to add a 1 to the #, but Asterisk does add a 1? |
18:00.27 | chefrs | Heh. |
18:00.39 | b11d | look at your extensions.conf |
18:00.48 | b11d | see if it's adding a 1 or not. |
18:00.52 | chefrs | It is. |
18:00.54 | chefrs | I can see it in CLI |
18:00.55 | b11d | ok.. so whats the prob |
18:01.04 | b11d | ohhh |
18:01.04 | chefrs | *shrugs* |
18:01.07 | chefrs | <PROTECTED> |
18:01.13 | b11d | it still asks for the 1 even though 1 is being passed to the telco? |
18:01.17 | chefrs | Yeah. |
18:01.29 | Mercestes | spanglesontoast, Yes, we covered this two days ago |
18:01.30 | b11d | weird, dunno on that one. |
18:01.33 | Mercestes | spanglesontoast, Are you a bot? |
18:01.37 | spanglesontoast | was yestaday P |
18:01.38 | spanglesontoast | ;) |
18:01.38 | chefrs | Righto |
18:01.55 | aydiosmio | the bug tracker no longer accepts feature requests |
18:01.59 | spanglesontoast | ~zaptel |
18:02.07 | jbot | well, zaptel is zapata telephony interface. A low level interface designed to abstract hardware access to a variety of devices for BRI, PRI or analogue access. |
18:02.07 | aydiosmio | HEY DEVS |
18:02.08 | spanglesontoast | :| |
18:02.12 | spanglesontoast | where can I get it |
18:02.13 | aydiosmio | add P-Asserted-Identity to the SIP confifuration |
18:02.15 | spanglesontoast | :( |
18:02.19 | b11d | aydiosmio.. #asterisk-dev :P |
18:02.21 | aydiosmio | you mother farkers. |
18:02.52 | Mercestes | spanglesontoast, uhh....from ftp.digium.com or emerge zaptel. or google zaptel downloads. |
18:03.29 | Mercestes | spanglesontoast, really. you should give up. You shouldn't be touching modern technology. Your brain is an empty void of worthlessness that simply sin't absorbing any of this or registering any level of higher conciousnesss. |
18:03.40 | spanglesontoast | lol |
18:04.07 | spanglesontoast | why not just give me a gun |
18:04.07 | spanglesontoast | ;) |
18:04.07 | Mercestes | spanglesontoast, You are an audible amoeba, asking for stimulus and blindly responding to stimulus with no abosorption or processing...nothing more. |
18:04.09 | Mercestes | spanglesontoast, Either evolve...or go away. |
18:04.14 | b11d | you're a dick Mercestes :) |
18:04.20 | b11d | i heart you |
18:04.27 | spanglesontoast | bah |
18:04.32 | Mercestes | spanglesontoast, Google is at http://www.google.com It's a very useful tool. reading is also a useful tool. The world is at your fingertips...stimulate yourself a little. |
18:04.45 | *** join/#asterisk friedrich| (n=friedric@e177242105.adsl.alicedsl.de) |
18:04.48 | Mercestes | spanglesontoast, If you get *STUCK* ...please....ask for help. I will be glad to help you in your evolution... |
18:04.59 | Mercestes | spanglesontoast, Don't ask me to just stimulate you for random reactions. I have pets for that. |
18:05.05 | *** join/#asterisk anonymouz666 (n=anonymou@re366.compuland.com.br) |
18:05.08 | spanglesontoast | lol your sick hehe |
18:05.09 | spanglesontoast | ;) |
18:05.10 | Mercestes | b11d, I agree. :) I'm a dick. |
18:05.13 | b11d | lol |
18:05.19 | Mercestes | spanglesontoast, human pets. Sicker than you believe. |
18:05.30 | spanglesontoast | hehe |
18:05.34 | spanglesontoast | git it done |
18:05.35 | spanglesontoast | ;) |
18:06.12 | b11d | trite. |
18:07.12 | Strom_M | possibly moronic question: is there a way to get a phone connected to an fxs port on a zaptel card to reset to dialtone rather than reorder? |
18:08.07 | spanglesontoast | yea Mercestes some of the stuff that asterisk has is kinda blowing my mind |
18:08.30 | spanglesontoast | I mean being able to dial to a phone line and make a script execute by a prefix that's awesome |
18:08.36 | Mercestes | Strom_M, by reset....you mean if it fails a call or you dial an invalid number?? |
18:08.59 | Mercestes | spanglesontoast, I can format a harddrive with an extension. |
18:09.05 | Mercestes | what are you talking about? |
18:09.08 | Strom_M | Mercestes: when the called party unsupervises |
18:09.14 | Ac1dcrawl | In my dialplan, how would I analyze only 2 digits in the incoming number to match an extension? |
18:09.49 | Mercestes | Strom_M, Man you telco ppl. Should give blankness for about 60 seconds or so then reorder. Kind of "the bells" standard. This in asterisk?? |
18:10.26 | Strom_M | yes, in asterisk |
18:10.29 | Strom_M | it's quite simple |
18:10.41 | Mercestes | Strom_M, Kind of a hack but you might be able to exten => h,1, it and Hangup() then Answer(). |
18:10.53 | Strom_M | when the called party unsupervises, have a battery drop on the local circuit followed by dial tone instead of reorder |
18:10.56 | Mercestes | But your looking at overriding a driver level function I dareasay |
18:11.01 | Strom_M | well, duh |
18:11.14 | Strom_M | is there a way to set that? :) |
18:11.20 | spanglesontoast | bummer I forgot to uncommet ztdummy |
18:11.28 | Mercestes | can't do a h,1,Hangup h,2,Answer? |
18:11.30 | b11d | totally bummer |
18:11.38 | Mercestes | yea, a h,1,1hangup would be retarded but.....you know what I mean. in a new context. |
18:11.41 | Strom_M | Mercestes: no, i want to do it channel-level |
18:11.45 | Mercestes | h1,1Goto(Do my freakystuff,1,1) |
18:11.48 | Mercestes | Oh. |
18:11.51 | Strom_M | doing it in extensions.conf screws the CDRs |
18:11.58 | Mercestes | Strom_M, Can't you just pactch zaptel? |
18:12.01 | b11d | isnt there some NoCDR command? |
18:12.20 | spanglesontoast | ah a make all will do it ;) |
18:12.22 | xo8ox | how do you stop and start asterisk .. |
18:12.25 | Mercestes | spanglesontoast, I covered that too like 2 days ago |
18:12.30 | b11d | hahaha |
18:12.36 | Mercestes | xo8ox: stop now. |
18:12.46 | xo8ox | in asterisk -r mode ? |
18:13.02 | spanglesontoast | stop now ;) |
18:13.04 | xo8ox | how do you do it from linux command line |
18:13.10 | spanglesontoast | please stop putting that thing in me :P |
18:13.24 | xo8ox | I can't get into the CLI |
18:13.25 | b11d | nice.. that's getting pulled out of context. |
18:14.04 | Mercestes | xo8ox, I would prefer in the channel but yes in the asterisk -r mod... |
18:14.37 | *** join/#asterisk cr4z3d (n=cr4z3d@ip70-162-97-91.ph.ph.cox.net) |
18:14.50 | Mercestes | xo8ox um....killall -9 asterisk |
18:14.57 | Mercestes | or /etc/init.d/asterisk stop for me. |
18:15.03 | spanglesontoast | oh crud sticks |
18:15.11 | Mercestes | but I embraced the gentoo tastyness and left the Centos behind me. |
18:15.16 | b11d | crud sticks.. that rocks. |
18:15.30 | spanglesontoast | lol |
18:15.41 | b11d | go the way of the BSDs.. they taste GREAT. |
18:15.55 | Mercestes | BSD tastes like nutsack. |
18:16.00 | spanglesontoast | hate bsd so sucky |
18:16.02 | b11d | you know what that tastes like? |
18:16.03 | Mercestes | yea, some ppl like it....but nobody normal. |
18:16.11 | b11d | eatith me! |
18:16.11 | aydiosmio | BSD is amazing |
18:16.15 | aydiosmio | shuddap |
18:16.19 | b11d | yes.. yes it is |
18:16.21 | spanglesontoast | bsd doesn't have support for flash :) |
18:16.24 | aydiosmio | it runs linux binaries better than linux does |
18:16.26 | Mercestes | BSD is .....yes...amazing is a good word for it. |
18:16.27 | b11d | yep.. lets turn this into an OS war :) |
18:16.28 | b11d | again.. |
18:16.31 | spanglesontoast | keep it for servers :) |
18:16.39 | Mercestes | it's amazing that it ever made it into public. |
18:16.39 | b11d | yeah I do run asterisk on a server.. |
18:16.57 | wunderkin | xo8ox, if you can't asterisk -r, then asterisk isn't running... what happens if you asterisk -vvvcng |
18:17.02 | b11d | Mercestes.. im coming over to beat you for that one. |
18:17.10 | Mercestes | b11d, promises, promises |
18:17.15 | spanglesontoast | yea mines running on ubuntu best desktop ever ;) |
18:17.17 | b11d | lol |
18:17.22 | *** join/#asterisk RoyK (n=roy@ti211310a080-5748.bb.online.no) |
18:17.32 | spanglesontoast | well my lappy which is downstairs lol |
18:17.35 | Ifaistos | greetings to all |
18:17.36 | Mercestes | Saybayon is the best desktop ever. |
18:17.44 | spanglesontoast | someone say fried eggs ? |
18:17.46 | spanglesontoast | :P |
18:17.53 | b11d | AIX has the best desktop.. maybe QNX |
18:17.54 | b11d | :P |
18:17.56 | Mercestes | Ubuntu is the best............whatever it is. |
18:18.01 | Ifaistos | what caching dns server would you propose for use with an asterisk system ? |
18:18.04 | Mercestes | b11d, ..... your a freak. |
18:18.08 | b11d | I use djbdns as an external cache |
18:18.08 | Strom_M | ubuntu is the best ubuntu |
18:18.09 | b11d | myself |
18:18.17 | b11d | Mercestes, thats why you like me so much. |
18:18.19 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
18:21.17 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
18:21.40 | spanglesontoast | hmm zaptel hates me |
18:21.51 | b11d | yeah they sure do |
18:22.06 | spanglesontoast | something is stopping it from compiling :| |
18:22.12 | b11d | i just got off the phone with zaptel and they are pissed because you never return the CD's you borrow |
18:22.14 | blitzrage | afternoon all! Anyone able to enlighten me as to why when you use one-touch recording with the 'W' flag the file is mixed BEFORE the 'h' extension executes, and if you use the 'w' flag, why it is mixed AFTER the 'h' extension executes? |
18:23.47 | *** join/#asterisk HaDAk (n=hans@152.160.16.90) |
18:24.06 | HaDAk | Has anyone had experience installing Asterisk on a WRT54G? |
18:24.18 | aydiosmio | the difference between W and w is who hangs up first |
18:24.22 | spanglesontoast | lol on a router lol give that up lol |
18:24.32 | HaDAk | spanglesontoast: don't be so cocky. |
18:24.34 | HaDAk | http://www.voip-info.org/wiki-Asterisk+Linksys+WRT54G |
18:24.58 | HaDAk | now. has anyone had experience installing asterisk on a wrt54g? |
18:25.15 | *** join/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
18:25.23 | *** part/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
18:25.56 | b11d | spanglesontoast.. you've exceeded the maximum number of permitted " |
18:26.03 | b11d | "lol"'s in a single sentance. |
18:26.04 | spanglesontoast | huh |
18:26.04 | b11d | doh. |
18:26.10 | *** part/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
18:26.42 | Ifaistos | HaDAk : Well major problem is limited memory |
18:26.58 | spanglesontoast | brb food |
18:27.54 | Ifaistos | HaDAk : also almost one of the compressing codecs can work (gsm is the exception) |
18:28.16 | HaDAk | Ifaistos: i have no problem using NAS |
18:28.41 | Ifaistos | HaDAk : memory=ram |
18:28.48 | HaDAk | right. |
18:28.53 | HaDAk | it has, what? 15 meg? |
18:29.05 | Ifaistos | HaDAk : 8 |
18:29.22 | HaDAk | hmm |
18:29.28 | HaDAk | and what's the overhead on asterisk? |
18:29.38 | Ifaistos | HaDAk : but depends on the model |
18:29.50 | HaDAk | lemme check and see what i've got. |
18:29.52 | Mercestes | HaDAk, no.....I only have experience installing Asterisk on a WRT54GL not a WRT54G. |
18:29.52 | *** join/#asterisk spatulamaan (n=ggilmore@207.188.8.252) |
18:30.55 | Assid | i guess if you use no transcoding.. there should be no issues at all |
18:30.56 | cr4z3d | softphone for windows, what's a good one |
18:30.57 | Mercestes | spanglesontoast, And yes, on a router, baby. We are uber-133t. You can't even compile zaptel. |
18:30.59 | Assid | ulaw-ulaw calls |
18:31.05 | Mercestes | cr4z3d Xlite. |
18:31.10 | cr4z3d | thanks Mercestes |
18:31.29 | HaDAk | Ifaistos: this is a WRT54G v2 |
18:31.30 | aydiosmio | x-lite is good? |
18:31.34 | HaDAk | back when they made em nice. |
18:31.38 | aydiosmio | I think sjphone is easier to work with |
18:31.49 | *** join/#asterisk anonymouz666 (n=anonymou@re366.compuland.com.br) |
18:31.51 | zoa | cr4z3d: idefisk! |
18:32.05 | zoa | version 2 prerelease available when you privmsg me |
18:32.06 | zoa | :) |
18:32.11 | Mercestes | cr4z3d: NP. You can get the C# source for it too. I really liked it for that. |
18:32.17 | b11d | no wayu |
18:32.33 | cr4z3d | i'm just looking for a simple one that i can connect to my asterisk server with |
18:32.39 | zoa | idefisk! |
18:32.55 | Strom_M | zoa: what does the name mean, anyway |
18:32.57 | zoa | Mercestes: xlite comes with sources ? |
18:33.00 | zoa | good question |
18:33.03 | zoa | nothing really |
18:33.04 | cr4z3d | do i need to configure sip for idefisk? i'm thinking of just using iax2 to connect if that's even possible |
18:33.13 | zoa | idefisk does that |
18:33.14 | Mercestes | zoa: yea, I got sources somewhere. |
18:33.14 | Strom_M | it sounds like....you're making a shish kebab out of an IDE drive |
18:33.30 | zoa | yeah i know |
18:33.35 | zoa | Mercestes: WTF |
18:33.36 | zoa | :) |
18:33.37 | aydiosmio | how can I do a SIPAddHeader from a .call file? |
18:33.42 | zoa | for xlite ??? |
18:33.50 | aydiosmio | need to put headers in the INVITE |
18:33.58 | Ifaistos | HaDAk : v2 has 4MB flash 16Ram |
18:34.24 | Assid | gimme gimme |
18:34.45 | Mercestes | zoa: uh..yea... |
18:34.55 | Ifaistos | HaDAk : http://wiki.openwrt.org/OpenWrtDocs/Hardware/Linksys/WRT54G?highlight=%28OpenWrtDocs/Hardware%29 |
18:34.57 | cr4z3d | HaDAk, try openwrt white russian |
18:35.11 | Mercestes | HaDAk, I agree with cr4z3d |
18:35.21 | cr4z3d | i was going to try that.. but i have a v5 |
18:35.24 | Mercestes | I hear freewrt is good too. Discussed that over beer last night |
18:35.26 | cr4z3d | not support by openwrt |
18:35.38 | Assid | cr4z3d: i think it does |
18:35.39 | cr4z3d | i'm using dd-wrt instead |
18:35.41 | *** join/#asterisk xtr-II (n=94752345@S0106000c41ed11e1.vf.shawcable.net) |
18:35.54 | Ifaistos | most of the wrt distro's are pretty stable |
18:36.06 | Assid | anyone got anything for a BEFW11S4 ? to make it repeater mode? |
18:36.11 | cr4z3d | yeah i've been happy with dd-wrt for now |
18:36.12 | HaDAk | Mercestes: there shouldn't be a huge difference between the wrt54g and the L |
18:36.22 | cr4z3d | there is |
18:36.24 | cr4z3d | the l |
18:36.27 | cr4z3d | is the same as v4 |
18:36.31 | cr4z3d | the last one to use linux |
18:36.35 | cr4z3d | natively |
18:37.22 | HaDAk | Ifaistos: sorry, plugging in this router lagged me a bit. white russian you recommend? |
18:37.25 | *** join/#asterisk qdk (n=qdk@80.243.125.204) |
18:37.34 | Mercestes | HaDAk, Well, there is one very important difference. There *IS* a wrt54g that you cannot flash with a new image. The "L" means "linux" which means you can image it. Otherwise, they are identical. |
18:38.00 | *** join/#asterisk Hmmhesays (n=Neg@24-117-131-41.cpe.cableone.net) |
18:38.05 | __freedom__lover | :D i've got it!! |
18:38.25 | Ifaistos | HaDAk : white-russian or dd-wrt both are good |
18:38.25 | Hmmhesays | anyone ever have any trouble with a telco not passing the name with number for callerid over a pri? |
18:38.27 | Assid | i gotta find a way to make my BEFW11s4 go into repeater mode |
18:38.43 | cr4z3d | Mercestes, as far as i've read a few months ago v5 and v6 can only use the micro versions of dd-wrt |
18:38.50 | b11d | Hmmhesays.. yeah and it was the telco's end. |
18:39.03 | b11d | it took me two weeks to convince them of that when I had the issue here with CP Telecom |
18:39.18 | Ifaistos | yeah recent versions are 2MB flash.... there isn't much you can fit in there |
18:39.26 | b11d | and it was like "yeah, the NAME is why i needed the PRI" |
18:39.37 | Mercestes | Hmmhesays, It's a telco thing. Some do. Some don't. some charge extra. |
18:39.44 | *** join/#asterisk Xen^ (n=linux@unaffiliated/lnux/x-10290) |
18:40.47 | Ifaistos | HaDAk : what ever you use make a cron script to restart asterisk at night |
18:40.47 | Hmmhesays | yeah telco's are a pain in the @$$ |
18:40.51 | b11d | yeah they can be.. |
18:42.14 | cr4z3d | Mercestes, when setting up xlite where it asks for authorization user name, what is that in the sip.conf? |
18:42.35 | cr4z3d | because i already entered a user name |
18:42.41 | Mercestes | cr4z3d, Pretty sure it's the username in sip.conf and the username in xlite is the [name] in sip.conf |
18:42.48 | Mercestes | cr4z3d, I make it easy and make them all the same thing. |
18:43.28 | cr4z3d | ok so [xlite] user=xlite secret=1234 |
18:43.31 | cr4z3d | should be enough |
18:43.37 | JunK-Y | some1 here is using asterisk-gui? where do we set to which ami user will it uses? |
18:44.30 | cr4z3d | oh yeah Mercestes what about the dtmfmode? rfc2833? |
18:44.36 | Mercestes | cr4z3d, yea. |
18:44.43 | Mercestes | cr4z3d, Dtmfmode=auto |
18:44.48 | cr4z3d | oh they have auto? |
18:44.49 | Mercestes | always dtmfmode=auto |
18:45.04 | Mercestes | yea, it does what dtmfmode=rfc2833 is *SUPPOSED* to do. |
18:45.32 | *** join/#asterisk RoyK (n=roy@ti211310a080-5748.bb.online.no) |
18:47.02 | cr4z3d | Mercestes, [xlite] type=friend user=xlite host=dynamic defaultip=ipofcompusingxlite dtmfmode=auto secret=1234, that looks good right? |
18:48.46 | Mercestes | Looks good to me. |
18:49.02 | *** join/#asterisk Meaty` (n=meaty3@office.abi.ca) |
18:49.03 | [TK]D-Fender | cr4z3d: context=thecontextthatsayswhatyoucandial <_ |
18:49.06 | Mercestes | HaDAk, Turn your away messages and your auto away stuff off. |
18:49.16 | [TK]D-Fender | cr4z3d: defaultip= worthless idea, remove... |
18:49.17 | Mercestes | thatnks, D. WB |
18:49.46 | [TK]D-Fender | Mercestes: y0 |
18:50.56 | cr4z3d | Mercestes, wow good call completely forgot the context |
18:51.09 | Mercestes | I didnt' call that, D-fender did. |
18:51.16 | Mercestes | thank him. :) |
18:51.43 | cr4z3d | oh wow |
18:51.48 | cr4z3d | i didn't even read the name |
18:51.52 | cr4z3d | thanks [TK]D-Fender |
18:52.08 | *** part/#asterisk deeperror (n=deeperro@mail.banctel.com) |
18:52.31 | *** join/#asterisk sysreq (n=sysreq@86-198-0-72-ppp.3menatwork.com) |
18:53.07 | [TK]D-Fender | cr4z3d: np, good to hear we didn't lose you from yesterday :) |
18:53.42 | cr4z3d | [TK]D-Fender, haha i'm still not being able to make calls in so i decided to see if i could call out |
18:54.21 | [TK]D-Fender | cr4z3d: .... and? |
18:55.18 | cr4z3d | [TK]D-Fender, still in the process of setting it up, exten => _1NXXNXXXXXX,1,Dial,IAX2/cr4z3d@NuFone/${EXTEN} does that look like a good ext? that's what nufone gave me |
18:56.24 | [TK]D-Fender | cr4z3d: Doesn' look quite right |
18:56.42 | [TK]D-Fender | cr4z3d: Pastebin your sip.conf for them minus PW's |
18:57.11 | cr4z3d | ~pb |
18:57.18 | jbot | it has been said that pb is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
18:57.23 | aydiosmio | [TK]D-Fender: any idea how to get SIPAddHeader to insert some headers inot the INVTE for an auto-dial (call file) |
18:57.57 | [TK]D-Fender | aydiosmio: You shoveit in your dialplan like anything else. Show us how YOU'RE trying it right now. |
18:59.34 | cr4z3d | [TK]D-Fender, http://pastebin.ca/406846 pasted my extensions and sip conf |
18:59.56 | aydiosmio | I have no idea where auto-dial runs from in the dialplan |
19:00.19 | *** join/#asterisk topping (n=topping@204.152.96.238) |
19:00.21 | aydiosmio | tried it in the call file context, but of course that won't work |
19:00.24 | [TK]D-Fender | cr4z3d: You need to set up a SIP peer for NuFone. You do not want that in yourdialplan the way you hve it. |
19:00.33 | [TK]D-Fender | aydiosmio: SHOW |
19:01.29 | JerJer | cr4z3d: Take a look at the tutorial in the Members Portal |
19:02.27 | cr4z3d | [TK]D-Fender, i have nufone peer in the iax.conf i need one in sip too? |
19:02.48 | [TK]D-Fender | cr4z3d: oops... yeah... iax2.conf would be kinda nice :) |
19:02.49 | cr4z3d | JerJer, that's where i got the extension from and how i setup the iax.conf |
19:02.57 | fbcit | Is IAXTel working OK? |
19:03.47 | fbcit | Regardless of allow=all in iax.conf I still receive " Call rejected by 69.73.19.178: Unable to negotiate codec" |
19:04.54 | [TK]D-Fender | fbcit: Odds are because it finds a match that includes a codec you CAN'T really do, yet * knows about like G.729 or G.723 |
19:04.57 | cr4z3d | [TK]D-Fender, here's iax.conf http://pastebin.ca/406854 |
19:05.41 | JerJer | cr4z3d: pastebin the CLI |
19:05.54 | *** join/#asterisk SomethingISODD (n=dan@NTL208H101-91-124.nt.net) |
19:05.55 | [TK]D-Fender | cr4z3d: exten => _1NXXNXXXXXX,1,Dial(IAX2/NuFone/${EXTEN}) |
19:05.55 | JerJer | ie show what happens when you actually make a call |
19:06.03 | [TK]D-Fender | cr4z3d: exten => _NXXNXXXXXX,1,Dial(IAX2/NuFone/1${EXTEN}) |
19:06.20 | [TK]D-Fender | cr4z3d: exten => _NXXXXXX,1,Dial(IAX2/NuFone/1555${EXTEN}) <- change accordingly |
19:06.20 | JerJer | and you don't really want/need type=friend for [NuFone] |
19:06.21 | SomethingISODD | hello all question do i need any special hardware to beable to send sms messages through asterisk? |
19:06.33 | Assid | [TK]D-Fender: hjow much do you like polycom ? |
19:06.43 | JerJer | and you do not register |
19:06.52 | JerJer | PLEASE read the tutorial |
19:06.57 | cr4z3d | JerJer, i had them seperate before as the tutorial said |
19:07.06 | Assid | my 301 just managed to piss me off TOTALLY.. it all off a sudden decides not to register anymore |
19:07.15 | cr4z3d | JerJer, i just put htem as one cuz i saw it somewhere online and it was less lines |
19:07.18 | [TK]D-Fender | fbcit: Disallow=all , allow=ulaw , allow=alaw, allow=gsm |
19:08.01 | JerJer | don't follow other examples - follow the one that NuFone provides |
19:08.24 | cr4z3d | JerJer, so the register => line is not needed? |
19:08.38 | [TK]D-Fender | Assid: Polycom is great. Aastra is very decent as well and I hope to have a better idea shortly. |
19:08.39 | JerJer | simply to place outbound calls, no |
19:09.10 | fbcit | [TK]D-Fender:I get the same error after making those changes. |
19:09.52 | *** join/#asterisk Assid (n=assid@203.212.204.107) |
19:10.06 | cr4z3d | JerJer, so if i remove register => from [general] it won't make any difference at all? and i'll just copy the rest from the tutorial? |
19:10.15 | [TK]D-Fender | fbcit: And did you apply them? perhaps you should enable debug for the protocol you are working with so as to see what they are offereing... |
19:10.28 | Assid | [TK]D-Fender: for some strange reason i cant get my polycom every now and then |
19:10.31 | Assid | softphone works perfect |
19:11.01 | *** join/#asterisk joe (n=nnjsauer@ip66-107-33-195.z33-107-66.customer.algx.net) |
19:11.04 | zoa | which one ? |
19:11.09 | zoa | idefisk ? |
19:11.22 | JerJer | cr4z3d: correct |
19:11.29 | fbcit | [TK]D-Fender:I did a 'iax2 reload' from the console. I'll enable debug and try it again. |
19:11.42 | cr4z3d | JerJer, ah wait nvm it says to have that for the incoming call config |
19:11.50 | Assid | didnt get a chance to use that yet.. but xten for now |
19:11.51 | JerJer | not to switch-1 |
19:11.59 | cr4z3d | oh wow |
19:12.06 | cr4z3d | i didn't see that tutorial the first time |
19:12.21 | cr4z3d | maybe this is why i'm having so many difficulties haha |
19:12.22 | *** join/#asterisk reber (n=reber@cl-158.dub-01.ie.sixxs.net) |
19:15.04 | fbcit | [TK]D-Fender:OK. I have some debug output. Shows perfered codec as ulaw. It does an AUTHREQ, an AUTHREP, and then a REJECT with the reason being the cause mentioned previously. |
19:15.14 | cr4z3d | JerJer, exten => 4022160528,2,Dial,<Something valid> <-- something valid meaning an extension setup to one of my sip phones? |
19:15.27 | fbcit | [TK]D-Fender:I can paste it somewhere if you want to see it. |
19:15.33 | *** part/#asterisk __freedom__lover (n=eduardo@clipper.provale.com.br) |
19:16.25 | fbcit | [TK]D-Fender:I am registered OK with the IAXTel server. |
19:16.49 | JerJer | cr4z3d: sure |
19:17.05 | *** join/#asterisk RoyK (n=roy@ti211310a080-5748.bb.online.no) |
19:18.45 | *** join/#asterisk ToyMan (n=Stuart@user-0cevdmv.cable.mindspring.com) |
19:19.39 | *** join/#asterisk ClydeGoffe (n=ClydeGof@base/student/clydegoffe) |
19:20.44 | [TK]D-Fender | fbcit: Yes, please pastebin that call + your iax2.conf minus passwords |
19:20.54 | anonymouz666 | Can I use a function inside Another ? |
19:21.38 | *** join/#asterisk mmgtrans (n=mmg@einstein.transtelco.net) |
19:21.59 | *** join/#asterisk oQPa (n=uawename@33.Red-83-34-60.dynamicIP.rima-tde.net) |
19:22.28 | mmgtrans | Somebody experiencinf DTMF problems with cisco phones, getting error New DTMF and can not find a free spot in dtmf array on console. Version 1.2.16 |
19:22.34 | *** part/#asterisk oQPa (n=uawename@33.Red-83-34-60.dynamicIP.rima-tde.net) |
19:22.48 | cr4z3d | holy shit i made a call out |
19:24.27 | [TK]D-Fender | whee! |
19:25.10 | [TK]D-Fender | cr4z3d: http://www.albinoblacksheep.com/flash/weeee.php |
19:25.18 | cr4z3d | haha |
19:25.26 | cr4z3d | but still getting nothing for callng in |
19:26.37 | cr4z3d | [TK]D-Fender, http://pastebin.ca/406873 is the cli output i get when trying to make a call in |
19:27.21 | [TK]D-Fender | cr4z3d: -- Executing Dial("IAX2/66.225.202.80:4569-6", "6000") in new stack |
19:27.22 | fbcit | [TK]D-Fender:Well, I am using the sample iax.conf and missed a later 'disallow all' which apparently negated my earlier allows. Seems to work now. Thanks for the help. |
19:27.34 | [TK]D-Fender | cr4z3d: That is not a valid way to dial anything |
19:27.44 | cr4z3d | oh right |
19:27.54 | cr4z3d | how would i make it dial that extension |
19:28.03 | cr4z3d | i'm so bad with these extension things |
19:28.33 | *** join/#asterisk IOscanner (n=IOscanne@cpe-76-187-194-128.tx.res.rr.com) |
19:28.39 | *** join/#asterisk toerkeium (i=oo@201.216.206.221) |
19:28.42 | [TK]D-Fender | cr4z3d: Mor appropriate might be : exten => 4022160528,2,Dial(SIP/xlite,20) |
19:29.10 | cr4z3d | yeah that's true but eventually i plan on having more than on extension that i can dial |
19:29.29 | cr4z3d | basically i want to dial in and enter some options |
19:29.31 | cr4z3d | eventually |
19:29.37 | cr4z3d | first things first haha |
19:29.52 | spanglesontoast | yea the question is why can't I comple zaptel |
19:30.52 | [TK]D-Fender | cr4z3d: Correct. For now, let it ring 1 phone and do "whatever". Then you can move on and learn how to make IVR's, etc. |
19:31.23 | cr4z3d | [TK]D-Fender, true i'll work on that stuff later then |
19:31.24 | [TK]D-Fender | spanglesontoast: Sorry, this channel hasn't gotten its copy of res_psychic.so ..... |
19:32.15 | IOscanner | is there a way to use precense from a cisco phone |
19:32.54 | [TK]D-Fender | IOscanner: Only is used with SCCP |
19:33.06 | [TK]D-Fender | IOscanner: one reason I don't suggest them |
19:33.32 | cr4z3d | oh man i'm so happy i can call and recieve calls |
19:33.53 | IOscanner | They have DND can't this not be used to set presence in the flash OP Panel |
19:35.16 | spanglesontoast | bah |
19:35.41 | *** join/#asterisk Evil_Lyra (n=Evil_Lyr@viper.pop-pr.rnp.br) |
19:35.48 | *** join/#asterisk kratzers (n=kratzers@martha.pa.net) |
19:36.12 | cr4z3d | does anyone know if all the sounds are installed by default using apt-get install asterisk on ubuntu? |
19:36.16 | [TK]D-Fender | IOscanner: * can't know about DND. It isn't a constant state. its the phone being told to reject calls and only comes up PER CALL. |
19:36.32 | HaDAk | i'm working on getting this wrt54g working with asterisk. i've got white russian on it, but what i need to know is what packages i need to put on it from: http://downloads.openwrt.org/whiterussian/newest/packages/ |
19:37.04 | Evil_Lyra | hi there |
19:37.21 | kratzers | is there a trick to getting ChanIsAvail to set ${AVAILSTATUS} to something other than 0 for a SIP peer? |
19:37.23 | spanglesontoast | Fender you love to talk down to me :P |
19:37.35 | spanglesontoast | you into bondage ;) |
19:37.41 | b11d | you have been asking the same questions over and over spanglesontoast.. |
19:37.45 | b11d | for days |
19:38.03 | [TK]D-Fender | spanglesontoast: I'd love to, but I'm all tied up right now.... |
19:38.18 | Evil_Lyra | I´m having a little troube trying to make asterisk transfer a call using "flash" |
19:38.44 | [TK]D-Fender | spanglesontoast: And that wasn't meant to be taken as "condescending", so much as "sarcastic and inspiring". |
19:38.54 | Evil_Lyra | my * is connected to a pstn extension of a traditional PBX |
19:38.57 | spanglesontoast | hmm |
19:39.08 | spanglesontoast | so you help with zaptel ? |
19:39.23 | tzafrir_laptop | we try |
19:39.43 | tzafrir_laptop | Evil_Lyra, and? |
19:39.57 | Evil_Lyra | I wan to transfer a call using Flash(), SendDTMF(<extension>), HangUp() |
19:40.00 | *** join/#asterisk RoyK (n=roy@ti211310a080-5748.bb.online.no) |
19:40.06 | [TK]D-Fender | spanglesontoast: Maybe once you actually show us whats WRONG :) You see you only said "it doesn't work, why?!?!". That does not help us to help you. |
19:40.12 | Evil_Lyra | but what´s happening is |
19:40.20 | spanglesontoast | http://www.pastebin.ca/406893 |
19:40.23 | spanglesontoast | ;) |
19:40.28 | Evil_Lyra | the extensions ring one time only |
19:40.53 | Evil_Lyra | and the asterisk hangup the connection with the callee |
19:41.01 | Evil_Lyra | ops, I mean the caller |
19:41.25 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
19:42.26 | Evil_Lyra | without asterisk, we only do pick up the phone, press flash, dial extension and put the phone on the hook |
19:42.37 | tzafrir_laptop | where exactly do you press Flash? On which phone? connected where? |
19:42.46 | [TK]D-Fender | spanglesontoast: Ok, now at least we have something to see. I unfotuantely don't have any advice for your current state. |
19:42.56 | Evil_Lyra | ok, from the beginning: |
19:43.02 | Evil_Lyra | forget asterisk |
19:43.14 | spanglesontoast | well I got the kernel headers can't think what's missing |
19:43.26 | [TK]D-Fender | Can someone spare a sec to see if the reason for spanglesontoast's problem compiling zaptel 1.4.0 here : http://www.pastebin.ca/406893 |
19:43.27 | Evil_Lyra | someone calls, pick the phone, press flash, dial extension, put the phone on the hook |
19:43.37 | Evil_Lyra | my pbx then trasnfer the call |
19:43.44 | Evil_Lyra | as it was supposed to do |
19:44.04 | Evil_Lyra | now, instead of having a phone, lets put a asterisk with a fxo card |
19:44.37 | [TK]D-Fender | Evil_Lyra: "show application flash" |
19:44.46 | Evil_Lyra | asterisk should Awnser, flash(), SendDTMF, and HangUp rigth? |
19:44.58 | [TK]D-Fender | Evil_Lyra: Sounds about right |
19:45.04 | tzafrir_laptop | spanglesontoast, why do you need zttranscode? |
19:45.06 | Evil_Lyra | but doesnt |
19:45.12 | tzafrir_laptop | Do you have a transcoder card? |
19:45.22 | Evil_Lyra | no |
19:45.23 | spanglesontoast | nope |
19:45.30 | spanglesontoast | I just want the ztdummy for meetme |
19:45.40 | tzafrir_laptop | consider just disable building zttranscode |
19:45.57 | tzafrir_laptop | grep for transcode in the Makefile and disable it |
19:46.24 | cr4z3d | how come i get a permission denied when trying to launch an AGI script? |
19:46.27 | *** join/#asterisk Idle (n=brian@S010600a024969312.ed.shawcable.net) |
19:46.32 | *** part/#asterisk acidchild (i=ash@unaffiliated/acidchild) |
19:46.37 | JunK-Y | cr4z3d: chmod +x script.agi |
19:46.41 | Evil_Lyra | [TK]D-Fender: |
19:46.41 | SomethingISODD | anyone know if i can SMS messages out through a VOIP line and recevie them also through a VOIP did? |
19:46.41 | Evil_Lyra | [Synopsis] |
19:46.42 | Evil_Lyra | Flashes a Zap Trunk |
19:46.45 | JunK-Y | or a chown. |
19:46.45 | Idle | is there a nice list of IAX providors? I am trying to find one in canada.. |
19:46.47 | Ng | anyone about who's on a UK ISDN line? I can dial UK numbers, but not special ones like 123 or 118118, and I can't dial internationally.... any suggestions? :) |
19:47.08 | zoa | www.voipcharges.com |
19:47.20 | zoa | the page is a bit ugly though |
19:47.35 | anonymouz666 | i already put saynumber() in a loop to say some numbers....100,200,300,400 - what do you suggest for me to do something like: press 1 for 100, press 2 for 200 - how can I mix the apps to make this possible? |
19:47.42 | spanglesontoast | ok ty tza |
19:47.48 | zoa | 11 there |
19:47.53 | zoa | but you could take one from any country |
19:47.54 | Evil_Lyra | [TK]D-Fender: flash seems to work, because I get one ring on the extensions that I´m transfering to |
19:47.54 | zoa | anywhere |
19:48.25 | Evil_Lyra | if I put a "Wait(30)" before hangup, I can hear the extension ringing across the room |
19:49.10 | Evil_Lyra | but as soon as asterisk hangup it drop the connection with the original caller |
19:50.24 | Idle | hm, a best of 3 / 5 rating... thats not good |
19:50.50 | Powerkill | someone find a solution for this bug : http://bugs.digium.com/view.php?id=8923 ? |
19:51.29 | zoa | idle, no need to take that rating into account |
19:51.36 | zoa | mostly competitors giving bad credits probably |
19:51.37 | zoa | :) |
19:51.50 | *** join/#asterisk friedrich| (n=friedric@e177243084.adsl.alicedsl.de) |
19:52.06 | b11d | Powerkill.. i've only seen that error come up when I've had a SIP peer "registred" but then isnt actually available. |
19:52.50 | Idle | :P |
19:52.53 | tclark | does anyone know of another co that makes a impedance matching device like this |
19:52.54 | Powerkill | b11d I have this error lot of time since I upgrade to 1.2.14 |
19:52.55 | tclark | hehe that looks interesting http://sandman.com/echo.html LINE IMPEDANCE MATCHER |
19:52.58 | Idle | I mainly need a recomendation |
19:53.01 | CunningPike | Hmm - trying to upgrade a 1.4.1 install to 1.4.2 and 'make menuselect' won't let me select chan_zap |
19:53.02 | Powerkill | so I've back revert to 1.2.13 |
19:53.20 | *** join/#asterisk chefrs (n=joe@c-24-8-226-145.hsd1.co.comcast.net) |
19:53.22 | Idle | someone stable.. dont specificly need anything more then number portability (is that possible in canda?) |
19:53.29 | Powerkill | But I need to get 1.2.17 for all the bug fix and I don't know how to correct the problem |
19:53.47 | chefrs | Any idea why my voicemail doesn't use the recorded message and instead uses the prerecorded gal that came with the system? |
19:54.02 | Idle | chefrs: did you set your flags correctly? |
19:54.10 | chefrs | What flags? Inbound? |
19:54.50 | Idle | well |
19:54.53 | Idle | in your extensions |
19:55.48 | Idle | http://www.voip-info.org/tiki-index.php?page=Asterisk+cmd+VoiceMail |
19:56.42 | spanglesontoast | hmm what would the build line be if I only wanted to compile zaptel with ztdummy support |
19:58.57 | Mercestes | spanglesontoast, cat makefile | sed s/#ztdummy/ztdummy/g > makefile && make clean && make && make install |
19:59.22 | spanglesontoast | nice |
19:59.26 | *** join/#asterisk steve___ (n=steve@kit-dhcp1.porchlight.ca) |
19:59.26 | *** part/#asterisk steve___ (n=steve@kit-dhcp1.porchlight.ca) |
19:59.29 | wunderkin | s/makefile/Makefile/ |
19:59.32 | wunderkin | g |
19:59.35 | spanglesontoast | yea |
19:59.37 | spanglesontoast | noticed ;) |
20:00.18 | *** join/#asterisk steve___ (n=steve@kit-dhcp1.porchlight.ca) |
20:00.18 | b11d | sure |
20:00.58 | Mercestes | wunderkin, Thanks. :) |
20:00.59 | *** join/#asterisk FuriousGeorge (n=brian@ool-43536ea8.dyn.optonline.net) |
20:04.03 | *** join/#asterisk m4rkl4r (n=markp@outboundemail.uneta.com) |
20:04.18 | Idle | hmmm |
20:04.22 | spanglesontoast | hmm that didn't work |
20:04.32 | Idle | does Vonage let you connect asterisk, or linksys desk phones, to their service? |
20:05.06 | Mercestes | Idle: I think Vonage only wants you to use their stuff. |
20:05.56 | brad_mssw | probably want the 'business plus' plans if you plan on using your own equipment with vonage |
20:06.01 | Idle | Mercestes: thats what I though |
20:06.08 | m4rkl4r | Are there techniques for making realtime extensions.conf faster? When the dial plan goes from, say context a, exten 1 to context a, exten 2, I'm seeing delays measured in seconds. |
20:06.14 | Idle | I just had a guy call and wanted to know that, cause he wants to use a linksys phone |
20:06.31 | Idle | sadly, theres no good IAX/SIP providors in canada here... maybe one of these will work tho |
20:06.32 | spanglesontoast | didn't understand what wunderkin did |
20:06.50 | m4rkl4r | the database is postgres in a machine on the same gigabit switch |
20:07.07 | brad_mssw | Idle: https://subscribe.vonage.com/smb-subscribe/index.htm?smb_id=acns .. you can use your own PBX with those plans |
20:07.43 | Idle | jesus |
20:08.00 | Mercestes | Idle: Teliax? |
20:08.08 | Idle | Mercestes: canada |
20:08.22 | wunderkin | s/spanglesontoast/spoogesontoast/ |
20:08.23 | Mercestes | spanglesontoast, Use a captial M instead of a lowercase M |
20:08.23 | wunderkin | what about that? |
20:08.28 | Idle | its kinda the trump card, sadly |
20:08.35 | Idle | anyhow, I've gotta get to class..... |
20:08.36 | Mercestes | Idle, they can't do Canada? |
20:08.37 | Idle | bbl |
20:08.54 | Mercestes | s/spanglesontoast/incurable_troll |
20:08.58 | spanglesontoast | s/#ztdummy/ztdummy/ was it mean't to be part of this |
20:09.09 | Mercestes | yes it was |
20:09.16 | wunderkin | s/Mercestes/mercestes/ :P |
20:09.24 | Mercestes | ahh, no! |
20:09.43 | spanglesontoast | yea but now make says no targets stop |
20:09.59 | spanglesontoast | cat Makefile | sed s/#ztdummy/ztdummy/g > Makefile && make clean && make && make install |
20:10.21 | cr4z3d | how do i make it wait a few seconds after answering? |
20:10.28 | cr4z3d | wait(x seconds)? |
20:10.29 | Mercestes | spanglesontoast, Are you even in /usr/src/zaptel<tab>/ ? |
20:10.38 | Mercestes | cr4z3d, Yes. |
20:10.46 | Mercestes | cr4z3d, Just use one to fix the voice cutoff. |
20:11.02 | spanglesontoast | erm I'm in /home/edd/zaptel-1.4.0 |
20:11.04 | spanglesontoast | ;) |
20:11.04 | cr4z3d | 1 is good for that? |
20:11.05 | cr4z3d | alright |
20:11.43 | *** part/#asterisk m4rkl4r (n=markp@outboundemail.uneta.com) |
20:11.53 | cr4z3d | oh and is there an easier way to reload the extensions.conf file |
20:11.58 | cr4z3d | besides stoping/starting asterisk |
20:12.03 | *** join/#asterisk TechJournalist (n=tk421@d141-129-92.home.cgocable.net) |
20:12.03 | b11d | just "reload" |
20:12.13 | aydiosmio | does asterisk dial out form a context when a call file is used? |
20:12.16 | aydiosmio | from |
20:12.16 | b11d | it wont interrupt anything, and isnt a shutdown&restart |
20:12.38 | aydiosmio | (not the Context=, but whereever Dial() is executed for the Channel=) |
20:13.01 | Mercestes | spanglesontoast, ..... |
20:13.10 | spanglesontoast | well that's the src |
20:13.36 | spanglesontoast | didn't know everyone moved it /usr/src |
20:13.37 | Mercestes | spanglesontoast, http://www.voip-info.org/tiki-index.php?page=Asterisk+consultants+USA |
20:14.09 | spanglesontoast | ? |
20:14.24 | b11d | lol |
20:14.33 | Mercestes | spanglesontoast, The only page you'll ever need. |
20:14.41 | Mercestes | the answer to all your problems. |
20:14.53 | spanglesontoast | don't have cash... |
20:15.09 | wunderkin | google and visa, the rest ;D |
20:15.10 | b11d | sell your body |
20:15.16 | Mercestes | Sorry, your screweed. |
20:15.26 | spanglesontoast | already sold myself... |
20:15.27 | Mercestes | I offer an exchange rate for virgins. |
20:15.31 | spanglesontoast | normally for ciggys |
20:15.47 | aydiosmio | So I guess I can't do SIPAddHeader for a call file dial-out eh? |
20:15.50 | aydiosmio | that sucks |
20:16.02 | Mercestes | I guess that disqualifies you as a virgin so....that means you had better start recruiting. |
20:16.31 | spanglesontoast | but using the command that you did just toasts the makefile |
20:16.43 | Mercestes | ... |
20:17.10 | spanglesontoast | any alternatives to ztdummy |
20:17.18 | Mercestes | spanglesontoast, a consultant. |
20:17.57 | spanglesontoast | hmm |
20:17.58 | [TK]D-Fender | aydiosmio: Sure you can... |
20:18.14 | [TK]D-Fender | aydiosmio: Show what you've done to dat |
20:18.35 | Mercestes | spanglesontoast, What distro are you on, first of all? |
20:18.37 | *** part/#asterisk MarkWD (n=MarkWD@rrcs-67-78-88-186.sw.biz.rr.com) |
20:18.42 | spanglesontoast | ubuntu |
20:19.39 | Mercestes | spanglesontoast, Pastebin your build error |
20:20.14 | spanglesontoast | er basically I can't change the makefile to the command you have sent |
20:20.38 | Mercestes | Dude, just replace #ztdummy with ztdummy zOMG |
20:20.51 | cr4z3d | oh man ztdummy.. had problems with that yesteray |
20:21.01 | aydiosmio | [TK]D-Fender: I'm sorry, I have no idea what to do, I have a stock trixbox install and I have no idea what context calls that originate from a call file use. I can't use the Context= specification because that happens after the INVITE |
20:21.02 | cr4z3d | dude i can't believe i finally got it working |
20:21.09 | spanglesontoast | oh is it commented out ? |
20:21.10 | Mercestes | and pastebin your error |
20:21.11 | *** join/#asterisk sbingner (n=sam@pdpc/supporter/sustaining/sbingner) |
20:22.03 | aydiosmio | is there a debug for what path the dialplan takes? |
20:22.18 | b11d | yeah "set debug" |
20:22.21 | b11d | and watch the console |
20:22.21 | b11d | :P |
20:22.29 | b11d | and "set verbose" |
20:22.42 | aydiosmio | I set them both to 30 and I don't see anything |
20:22.57 | b11d | nothing at all? nothing comes across your console? |
20:23.04 | aydiosmio | the sip debug |
20:23.08 | aydiosmio | which I have enabled |
20:23.14 | b11d | turn it off for a few mins |
20:23.22 | b11d | maybe your other debug info is being lost in that sea of information |
20:24.00 | *** join/#asterisk lukketto (n=lukketto@82.59.103.134) |
20:24.18 | aydiosmio | <PROTECTED> |
20:24.43 | aydiosmio | > Channel SIP/vs-outbound-08eb0218 was answered. |
20:24.58 | b11d | thats all you see? |
20:25.04 | Mercestes | spanglesontoast, YES. we covered that about 50 times. |
20:25.14 | b11d | /ignore spanglesontoast |
20:25.16 | b11d | :P |
20:25.18 | spanglesontoast | nope it ain't there |
20:25.23 | spanglesontoast | as commented |
20:25.32 | aydiosmio | b11d: that's all there is before the Context is connected |
20:25.54 | Mercestes | is it there uncommented? |
20:26.00 | b11d | and whats the problem then? |
20:26.10 | Mercestes | would that maybe be because I told you to uncomment it like...2 days ago??? |
20:26.21 | aydiosmio | I'm trying to figure out where asterisk Dial()s for call files |
20:26.28 | aydiosmio | I need to run SIPAddHeader for the INVITE |
20:26.33 | spanglesontoast | http://www.pastebin.ca/406951 |
20:26.35 | spanglesontoast | there we go |
20:28.00 | Mercestes | spanglesontoast, Did you even check the md5 of this src? |
20:28.57 | spanglesontoast | nah normally go by size |
20:28.57 | *** join/#asterisk sjaak_yen (n=chatzill@d5c53145.dsl.concepts.nl) |
20:28.57 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
20:28.57 | Mercestes | yes, I know you like the big ones. Cna you md5 your src please? |
20:32.26 | [TK]D-Fender | aydiosmio: Ok you clearly haven't figured out what you SHOULD be using as your originateing channel. You are now clearly direct-dialing an outside tech. You need to dial a LOCAL CHANNEL. and in there do your SIPAddHeader, then the actualy Dial to call the guy. |
20:32.34 | Mercestes | and I suggest you rdl regardless and start over. |
20:32.43 | Mercestes | That looks like a src error to me. |
20:32.54 | [TK]D-Fender | aydiosmio: And what you're describing for this is basically an automated page (wakeup call, etc...) |
20:33.19 | aydiosmio | basically |
20:33.40 | aydiosmio | ah! |
20:33.42 | aydiosmio | I see |
20:33.48 | SomethingISODD | anyone know if i can SMS messages out through a VOIP line and recevie them also through a VOIP did? |
20:33.49 | [TK]D-Fender | aydiosmio: so your originating channel should look like "Local/12345@myscriptcontext" |
20:34.09 | aydiosmio | So if I use SIP as the originating channel I have no control over it |
20:34.10 | spanglesontoast | erm theres no md5 checker |
20:34.11 | spanglesontoast | :| |
20:34.12 | aydiosmio | I see |
20:34.13 | [TK]D-Fender | aydiosmio: And use 1 entry for the guy to call, and 1 for the actions to take after answer |
20:34.24 | *** join/#asterisk [shodan] (n=shodan@ip181.99-113-216.pppoe4.joliette.intermonde.net) |
20:34.51 | *** join/#asterisk Waverly360 (n=irc@adsl-070-148-122-203.sip.bna.bellsouth.net) |
20:36.54 | *** join/#asterisk keescook (n=kees@ubuntu/member/keescook) |
20:37.05 | *** part/#asterisk Evil_Lyra (n=Evil_Lyr@viper.pop-pr.rnp.br) |
20:37.58 | keescook | can anyone point to the svn commit that fixed the vulnerability mentioned in the 1.2.17 release notes? |
20:39.12 | *** join/#asterisk docelmo (n=vircuser@c-68-45-140-42.hsd1.de.comcast.net) |
20:39.28 | docelmo | anyone know how to fix this: frame.c:214 __ast_smoother_feed: Dropping extra frame of G.729 since we already have a VAD frame at the end |
20:39.32 | docelmo | Im using Asterisk 1.4 |
20:40.11 | spanglesontoast | hang on a sec |
20:40.27 | spanglesontoast | why does it say 1.2.16 for zaptel and I'm using 1.4.0 :O |
20:40.40 | spanglesontoast | in the topic :| |
20:41.22 | spanglesontoast | oh |
20:41.24 | spanglesontoast | :| |
20:42.12 | [TK]D-Fender | spanglesontoast: Using a mismatched version are we? :) |
20:42.43 | spanglesontoast | well wondered where no one noticed it |
20:42.43 | spanglesontoast | lol |
20:43.29 | docelmo | [TK]D-Fender do you have any experience with that error? I have disabled VAD on my switch and its still doing it. |
20:43.45 | [TK]D-Fender | docelmo: No clue, never touched 1.4 (well ONCE..) |
20:44.23 | docelmo | I would prefer not to but not my end decision. |
20:44.57 | spanglesontoast | brb going for a smoke |
20:46.22 | Ifaistos | docelmo : what g729 is the other end using ? g729, g729a g729b ? |
20:46.23 | docelmo | I need something stronger.. |
20:46.49 | docelmo | yes.. I have g729, 729a and 729ab on one end and just asterisk's g729 on the pbx |
20:47.07 | docelmo | Well I have all 3 listed for use |
20:47.19 | Ifaistos | docelmo : try g729a |
20:47.21 | Strom_M | docelmo: turn off VAD on the other end |
20:47.27 | docelmo | I killed vad already |
20:47.31 | docelmo | I will do that.. |
20:47.45 | Mercestes | spanglesontoast, http://www.sing365.com/music/Lyric.nsf/The-Headless-Waltz-lyrics-Voltaire/554E5ADB2197F70D48256DAA00253BF2 |
20:50.52 | Waverly360 | I'm attempting to use a custom script to pipe audio from an external sound device for music on hold. The script is executable, and when I run it, garbage gets piped to my screen from the audio. So the script should be working fine. However, when asterisk attempts to start music on hold, it immediately says that music on hold has been stopped. |
20:51.25 | Waverly360 | I'm not getting any error messages or anything. |
20:51.33 | Waverly360 | I have verbosity and debug set to 99 |
20:51.47 | Waverly360 | Is there another place where errors might end up? |
20:51.56 | Ng | BT ISDN30e users - any hints for settings to make dialling international numbers possible? :) |
20:53.19 | *** join/#asterisk dj-fu (n=ajc@unaffiliated/dj-fu) |
20:54.21 | florz | Ng: Have you tried changing the pridialplan setting in the zapata.conf? |
20:54.38 | docelmo | the g729ab was causing the issue.. Thanks Ifaistos |
20:54.51 | Ng | florz: I tried setting it to local, but I'm wondering if unknown would have been a better setting |
20:56.11 | Ifaistos | docelmo : not all g729 are created equal :) |
20:58.43 | Ng | florz: do you know of any other things I can/should test? I have no net access at the site in question so I'm trying to stock up a bunch of things to try |
20:59.08 | florz | Ng: that ISDN line is not connected to the PSTN? |
20:59.36 | Ng | florz: no it's on the PSTN, it's a regular BT ISDN30e |
21:00.01 | florz | Ng: I mean, how come that you don't have net access then? |
21:00.03 | anonymouz666 | anyone know in what package i have the word "for" recorded |
21:00.04 | anonymouz666 | ? |
21:00.11 | Mercestes | spanglesontoast, Even better. http://www.tsrocks.com/g/gerhard_schoene_texts/spar_deinen_wein_nicht_auf_fur_morgen.html |
21:01.04 | Ng | florz: well I suppose I could get a dialup account and then figure out how to get pppd to talk to the E1 card, but that seems like a lot of faffing about and liable to interfere with taking the asterisk server up/down a lot to test various things |
21:01.45 | Ng | I have been promised the 100Mb LES would be live every day this week, so I was holding out my hopes for that ;) |
21:01.59 | florz | Ng: IC :-) |
21:02.39 | Ng | is getting a ppp connection over a zaptel device documented somewhere? |
21:03.11 | florz | Ng: Well, troubleshooting is a bit difficult without even some exact error message. But generally a wrong pridialplan setting would be a good candidate ... |
21:04.35 | Ng | florz: ok. If changing that doesn't fix it (and I have my doubts, pridialplan=local broke everything) I'll get some pri debugging logs |
21:06.55 | Waverly360 | Anyone ever had a MOH issue like mine before? |
21:09.15 | *** join/#asterisk sashion (n=djbdsf@dsl-241-213-43.telkomadsl.co.za) |
21:18.13 | *** join/#asterisk kgx (n=kgx@60.234.20.178) |
21:21.47 | *** join/#asterisk l3jj (n=chatzill@dsl093-001-038.det1.dsl.speakeasy.net) |
21:23.58 | *** part/#asterisk keescook (n=kees@ubuntu/member/keescook) |
21:27.26 | *** join/#asterisk fbffff (n=fbffff@r74-193-79-81.pfvlcmta01.grtntx.tl.dh.suddenlink.net) |
21:28.32 | errr | is it possible to make a call from the asterisk cli? |
21:28.38 | b11d|bbl | yes |
21:28.40 | b11d|bbl | use "Dial" |
21:28.45 | errr | ah a big D |
21:28.51 | errr | I was using a little one |
21:28.53 | b11d|bbl | no, its not case sensitive |
21:28.53 | errr | ty |
21:28.57 | errr | oh? |
21:29.01 | errr | maybe I did it wrong |
21:29.01 | sashion | errr: core dial tech/number |
21:29.16 | sashion | or core dial tech/num@context |
21:29.27 | *** join/#asterisk remmo (n=chatzill@smack.isp.net.au) |
21:29.41 | errr | hmm core is causing an error |
21:29.56 | dlynes_laptop | errr: core is only a valid keyword if you're using asterisk 1.4 i think |
21:30.00 | b11d|bbl | are you 1.4 or 1.2? |
21:30.03 | errr | 1.2 |
21:30.13 | sashion | then its dial tech/num |
21:30.15 | b11d|bbl | dial works fine for me on 1.2 |
21:30.18 | *** join/#asterisk friedrich| (n=friedric@e177243084.adsl.alicedsl.de) |
21:30.19 | b11d|bbl | i dunno what your issue is. |
21:30.34 | errr | No such command 'dial' (type 'help' for help) |
21:30.39 | dlynes_laptop | b11d|bbl: he probably doesn't have chan_alsa or chan_oss loaded |
21:30.40 | b11d|bbl | weird, i've got it. |
21:30.48 | b11d|bbl | yeah.. probably not. |
21:30.51 | errr | ah |
21:30.57 | dlynes_laptop | errr: do you have chan_alsa or chan_oss loaded? |
21:31.02 | errr | I doubt it. |
21:31.09 | dlynes_laptop | errr: if not, that would be your conundrum |
21:31.23 | dlynes_laptop | errr: that cli app isn't available unless you have one of those two channels loaded |
21:31.32 | dlynes_laptop | errr: it pipes chan_alsa/chan_oss through chan_local |
21:31.37 | errr | do I have to have a sound card to have them? |
21:31.53 | dlynes_laptop | errr: well, to have a sound card driver, it usually helps to have a soundcard, yes |
21:32.08 | errr | this is a rack mount box, no card |
21:32.31 | dlynes_laptop | errr: then how were you expecting to talk to the other end, or hear the other end? |
21:32.33 | *** join/#asterisk twisted[asteria] (n=twisted@pdpc/supporter/active/twisted) |
21:32.34 | *** mode/#asterisk [+o twisted[asteria]] by ChanServ |
21:32.39 | sashion | errr: best other way is to use a .call file |
21:32.40 | zoa | then why would you like to use chan_Alsa ? |
21:32.46 | errr | I dont want to hear or talk, just prove a point to my boss |
21:32.55 | dlynes_laptop | errr: use a sip phone, then |
21:33.16 | dlynes_laptop | errr: or a softphone |
21:33.24 | *** join/#asterisk tdi (n=tdi@reykin.pozman.pl) |
21:33.44 | tdi | hi. does anybody know a page maybe where all asterisk sounds are explained in text form ? |
21:34.06 | dlynes_laptop | tdi: you can't tell that by the filenames? |
21:34.14 | *** join/#asterisk pirast (n=martin@p508b2424.dip0.t-ipconnect.de) |
21:34.18 | tdi | not exactly |
21:34.19 | errr | thanks for the help everyone |
21:34.40 | tdi | i want to give ex. all vm sounds to record them by once more |
21:35.16 | pirast | hi, i have a question to http://bugs.digium.com/view.php?id=9203. in the last comment, it says "there were no *security* fixes in 1.2.17". is a dos vulnerability not considered as security fix? compare: http://www.asterisk.org/node/48339 ("his release incorporates a fix for the SIP DoS vulnerability recently discovered by INRIA") |
21:35.27 | dlynes_laptop | tdi: that sounds like a good use for an ex |
21:35.38 | tdi | ex? |
21:35.40 | dlynes_laptop | tdi: kinda like payback for them being your ex? |
21:35.52 | Mercestes | pirast: shh... we dont' talk about security issues. |
21:36.01 | tdi | dlynes_laptop: i do not understand |
21:36.04 | dlynes_laptop | tdi i want to give ex. all vm sounds to record them by once more |
21:36.10 | Mercestes | yes, it's fixed. K, thanks, bye. |
21:36.15 | Mercestes | Shhhh. |
21:36.19 | tdi | dlynes_laptop: yes and ? |
21:36.39 | dlynes_laptop | you want your ex to record all your voicemail sounds all over again? |
21:36.43 | *** join/#asterisk keescook (n=kees@ubuntu/member/keescook) |
21:36.50 | tdi | dlynes_laptop: aha nope |
21:36.58 | Mercestes | hmm... |
21:36.59 | tdi | ex. - example osrry |
21:37.07 | Mercestes | lol |
21:37.10 | dlynes_laptop | tdi: still makes no sense :) |
21:37.11 | tdi | i am giving it to the company that records such things |
21:37.13 | Mercestes | my pet ex. |
21:37.21 | dlynes_laptop | yeah...i like to pet my ex, too |
21:37.22 | *** join/#asterisk friedrich| (n=friedric@e177243084.adsl.alicedsl.de) |
21:37.31 | Mercestes | woohoo |
21:37.39 | *** join/#asterisk sahafeez (n=sahafeez@67.109.14.227.ptr.us.xo.net) |
21:37.45 | Mercestes | wait....wrong channel. >.> |
21:37.52 | tdi | look client want to record that the way he wants |
21:38.04 | tdi | why should i think about that :) |
21:38.08 | Mercestes | tdi: client no like comedian mail voice prompts? |
21:38.16 | tdi | yep |
21:38.21 | dlynes_laptop | tdi: write a script to automatically play them all back to you, so that you can write down what they are, then :) |
21:38.27 | Mercestes | client make new sound voicemail he does, yes? |
21:38.28 | dlynes_laptop | hehehehe |
21:38.30 | tdi | dlynes_laptop: lol |
21:38.42 | Mercestes | like the existing voicemail prompts he does not, no? |
21:38.53 | tdi | yes he wants new |
21:38.54 | *** join/#asterisk tuan_modulis (n=chatzill@3-82-252-216-static.enter-net.com) |
21:39.15 | tdi | the same voice the menu is |
21:39.15 | Mercestes | monitor you must do, with playback of sound, yes. |
21:39.17 | Mercestes | or with record would work, no? |
21:39.34 | Mercestes | asterisk cmd monitor, or asterisk cmd record, google you must. |
21:39.45 | Mercestes | may the force be wtih you. |
21:39.49 | tdi | Mercestes: why should i? |
21:40.00 | Mercestes | for recording of voicemail prompts. |
21:40.10 | spanglesontoast | bah |
21:40.13 | Mercestes | damnit! Now I can't stop talking like that. |
21:40.14 | tdi | Mercestes: i do not want to record them lol |
21:40.24 | tdi | i want the text transcripts |
21:40.24 | Mercestes | tdi: would you like to soxmix them maybe? |
21:40.24 | spanglesontoast | wheres that usb digital tv turner gone |
21:40.36 | Mercestes | tdi: Then I suggset you take a speed writing class. |
21:41.18 | Mercestes | spanglesontoast, You can barely work a keyboard and your linux skills came from "hooked on phonics." you don't have a USB digital TV tuner and even if you did you wouldn't know how to work it. |
21:41.23 | dlynes_laptop | spanglesontoast: they found out it was being used on Linux so Microsoft bought it, and hid it deep somewhere on their website |
21:41.24 | Mercestes | spanglesontoast, quit trying to pretend your a techie. |
21:41.42 | [TK]D-Fender | Talks does Mercestes often funny, hmmmmMMM!!!@!? |
21:41.43 | spanglesontoast | nah it's transcoded to my xbox 360 ;) |
21:42.01 | Mercestes | [TK]D-Fender, ROFL Multilingual, mercestes is, yes? |
21:42.11 | [TK]D-Fender | Mercestes: s/your/you're/ ;) |
21:42.41 | [TK]D-Fender | Mercestes: Clearly well versed in GIBBERISH.... and not always a comprehensible sub-dialect of which... |
21:42.54 | Mercestes | [TK]D-Fender, indeed. |
21:43.24 | dlynes_laptop | [TK]D-Fender: what gibberish doth thou speaketh of? |
21:43.25 | *** part/#asterisk keescook (n=kees@ubuntu/member/keescook) |
21:43.53 | *** join/#asterisk ToyMan (n=Stuart@ool-45784fde.dyn.optonline.net) |
21:44.02 | *** part/#asterisk ClydeGoffe (n=ClydeGof@base/student/clydegoffe) |
21:44.36 | [TK]D-Fender | Franglais :) |
21:44.47 | *** join/#asterisk tsurko (n=tsurko@vpn-pppoe-213-240-242-237.megalan.bg) |
21:45.01 | Waverly360 | I'm an idiot. |
21:45.12 | Waverly360 | :P |
21:45.23 | Waverly360 | stupid mispelled word caused me hours of grief |
21:45.28 | Waverly360 | *sigh* |
21:45.31 | dlynes_laptop | Waverly360: for moh? |
21:45.41 | Waverly360 | dylnes_laptop: Yeah. |
21:45.50 | dlynes_laptop | there's your problem |
21:45.55 | dlynes_laptop | you're even misspelling my name |
21:45.56 | dlynes_laptop | sheesh |
21:46.00 | Waverly360 | lmao |
21:46.01 | dlynes_laptop | i think you need more timmy's |
21:46.01 | Waverly360 | damnit |
21:46.08 | Waverly360 | what I need is a vacation |
21:46.15 | Waverly360 | and glasses that actually let me see better |
21:46.30 | Waverly360 | You stare at the screen long enough, all the letters just run together. |
21:46.46 | dlynes_laptop | that's the beer |
21:46.50 | dlynes_laptop | don't blame it on the glasses |
21:46.51 | tdi | http://www.voip-info.org/wiki/view/Asterisk+sound+files |
21:46.57 | [TK]D-Fender | Waverly360: Walleye-vision... I've seen it claim many a firend of mine... |
21:47.09 | [TK]D-Fender | Waverly360: When in doubt rely on auto-complete ;) |
21:47.32 | Waverly360 | [TK]D-Fender: Unfortunately, auto-complete doesn't work within a vim session. |
21:47.46 | Waverly360 | [TK]D-Fender: otherwise, that's what I use. |
21:47.47 | [TK]D-Fender | Waverly360: I meant in HERE.... |
21:48.14 | Waverly360 | [TK]D-Fender: ...I just now realized that auto-complete worked with my IRC client... |
21:48.22 | dlynes_laptop | [TK]D-Fender: obviously |
21:48.39 | [TK]D-Fender | "warning : people in mirror are dumber than they appear" |
21:48.52 | JT | everyone hello |
21:48.54 | Waverly360 | [TK]D-Fender: hah. *sigh* |
21:48.59 | dlynes_laptop | [TK]D-Fender: you've been living in montreal too long :) |
21:49.17 | [TK]D-Fender | dlynes_laptop: How so... besides "all of my life"? :) |
21:49.22 | *** join/#asterisk santiago (n=santiago@debian/developer/santiago) |
21:49.46 | [TK]D-Fender | dlynes_laptop: I exist perfectly between worlds. This is a GREAT place for perspective if you aren't overly activist about anything... |
21:50.13 | dlynes_laptop | [TK]D-Fender: i just remember when i was there, there seemed to be a hell of a lot of stupid french people in the bilingual area |
21:50.33 | dlynes_laptop | [TK]D-Fender: well, i love it there |
21:50.45 | dlynes_laptop | [TK]D-Fender: don't know if i'd wanna live there though, with the politics |
21:52.27 | [TK]D-Fender | dlynes_laptop: that falls under the category of "people are stupid. Inidiviuals are smart." and "beware the power of stupid people in large groups" |
21:52.27 | dlynes_laptop | heh |
21:52.27 | dlynes_laptop | sounds like an advertisement for a government office |
21:52.27 | [TK]D-Fender | dlynes_laptop: I healthily avoid politics. Anybody who tries to shove a "party line" my way gets their ass handed to them :) |
21:52.50 | dlynes_laptop | yeah...politics in bc is just as screwed up |
21:53.05 | dlynes_laptop | nobody here knows what the hell they're going to vote for half hte time |
21:53.05 | mcab | always has been |
21:53.29 | mcab | BC Politics is usually a complete gongshow |
21:53.35 | dlynes_laptop | and even then, half the time when they vote, it's a retaliation vote |
21:53.41 | dlynes_laptop | yeah, no doubt |
21:53.52 | dlynes_laptop | provincial politics here is absolutely stupid |
21:53.56 | JT | i have to vote tomorrow |
21:54.14 | dlynes_laptop | vote Rhinoceros Party! |
21:54.25 | JT | new south wales state election |
21:54.29 | *** part/#asterisk l2cache (n=ghansen@64.128.254.98) |
21:54.53 | [TK]D-Fender | dlynes_laptop: Not terribly different here. Typically is PQ, and when they screw up BAD (ie referrendum, etc) we follow the LEADER we like the most which is why the Liberals got smashed alst time and the PC's made it back in offic for the first time in ages. |
21:55.03 | mcab | dlynes_laptop: not the Natural Law Party? :-) |
21:55.10 | [TK]D-Fender | dlynes_laptop: Pot party! |
21:56.00 | dlynes_laptop | mcab: puuuuhleeaze...voting for a party that thinks they can levitate all the problems of the government away? |
21:56.08 | JT | come to Australia, where the conservative party is called the Liberaly Party |
21:56.20 | JT | Liberal Party |
21:56.29 | mcab | JT: welcome to BC, we have the same thing :-) |
21:56.30 | dlynes_laptop | [TK]D-Fender: yeah...Mark Emery :) |
21:56.55 | dlynes_laptop | mcab: yeah...the liberal party here is a bunch of doofuses...they're only liberal because the socreds got destroyed |
21:57.00 | *** part/#asterisk mog (n=mog@c-71-207-200-130.hsd1.al.comcast.net) |
21:57.05 | dlynes_laptop | and the pc party in bc is completely non-existent |
21:57.24 | [TK]D-Fender | dlynes_laptop: You guys actually HAVE NDP out there... damned socialists! |
21:57.32 | dlynes_laptop | what i wouldn't give for the pc's to get in, in bc |
21:57.35 | mcab | dlynes_laptop: c'mon, Natural Law's platform is as realistic as most of the main-stream platforms :-) |
21:57.38 | dlynes_laptop | i hate socialist parties |
21:57.42 | dlynes_laptop | especially ndps |
21:58.24 | dlynes_laptop | ndp is extremely bad for busines |
21:58.31 | dlynes_laptop | s/busines/business/ |
21:59.21 | [TK]D-Fender | ok, I'm off... back tonight... |
21:59.44 | *** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin) |
21:59.46 | PakiPenguin | hi |
22:01.19 | dlynes_laptop | cessia galee |
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22:06.18 | *** join/#asterisk RoyK (n=roy@ti211310a080-5748.bb.online.no) |
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22:09.01 | spanglesontoast | hmm |
22:11.45 | Sweeper | feelings on astlinux? |
22:13.46 | flenders | JT: that's one of the good things about not being a citizen here.... I dont need to vote tomorrow |
22:14.05 | JT | if you think that's a good thing :) |
22:15.36 | flenders | aren't you annoyed with the ads on the radio? |
22:15.36 | *** join/#asterisk sysreq (n=sysreq@modemcable171.134-81-70.mc.videotron.ca) |
22:16.24 | JT | yeah they're a bit annoying, they've stopped now as a ban on electronic election advertising is in force |
22:16.45 | JT | whether or not you vote won't have any impact upon if those ads will be on |
22:16.52 | flenders | good I'll start listening to radio again |
22:16.53 | flenders | :D |
22:17.25 | JT | and i think voluntary voting for able-bodied persons is stupid |
22:17.48 | JT | compulsory is where it's at |
22:18.08 | JT | people will still put invalid votes in anyway if they really don't want to do it |
22:19.54 | flenders | JT: but if it's voluntary, sometimes you would be on a bbq, or at the beach, and even though you would vote right, you would just "nah... can't be bothered now" |
22:21.03 | JT | yeah |
22:21.14 | JT | look at how corrupt the US's voluntary voting system is, also |
22:21.26 | *** join/#asterisk mmartinn (n=martin@adsl-065-005-200-225.sip.gnv.bellsouth.net) |
22:23.21 | spanglesontoast | why doesn't meetme work even though I've got zaptel installed :| |
22:23.37 | Strom_M | spanglesontoast: what do you mean by 'it doesnt work'? |
22:23.52 | spanglesontoast | load_resource: Module 'MeetMe' could not be loaded. |
22:24.10 | Strom_M | did you build and install zaptel before building asterisk? |
22:24.24 | orlock | JT: could you reccomend anywhere to buy some server-grade 2RU systems? |
22:24.37 | JT | systems... like whole systems? |
22:24.42 | orlock | yeah |
22:24.47 | *** join/#asterisk Iamnacho (i=Iamnacho@ip68-103-153-140.ks.ok.cox.net) |
22:24.49 | orlock | currently looking at supermicro |
22:25.00 | spanglesontoast | no I did it after so I'm guessing I've gotta recompile it ? |
22:25.12 | orlock | from Digicor its something like $6k for a dual socket, dual core system |
22:25.21 | orlock | hot swap PSU, etc |
22:26.11 | Strom_M | spanglesontoast: yes, you must compile zaptel FIRST |
22:26.16 | orlock | might look at Sun and IBM as well |
22:27.14 | *** join/#asterisk steveaj (n=steve@82-71-61-44.dsl.in-addr.zen.co.uk) |
22:28.40 | *** join/#asterisk qdk (n=qdk@0x50c627be.bynxx11.adsl-dhcp.tele.dk) |
22:29.30 | orlock | fark. |
22:29.47 | orlock | dual cpu sun netra, $13k! |
22:30.16 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
22:30.16 | *** mode/#asterisk [+o russellb] by ChanServ |
22:30.35 | JT | orlock: is that a niagra core? |
22:30.51 | *** join/#asterisk Voice2 (n=Voice2@145-27.mc.cite.net) |
22:30.59 | Voice2 | where is url to see patches in svn web ? |
22:32.58 | orlock | JT: yeah, think so |
22:33.25 | JT | orlock: well no wonder |
22:33.29 | JT | they're awesome units |
22:34.05 | JT | orlock: you know each cpu has 8 cores? |
22:35.19 | JT | oh wait, it might be 4 |
22:35.22 | JT | not sure |
22:39.28 | JT | orlock: i just checked |
22:39.47 | JT | helped jog my memory as to why the Niagra core is so good |
22:39.52 | JT | 8 cores |
22:39.59 | JT | 4 threads each |
22:40.01 | JT | 32 threads |
22:40.17 | spanglesontoast | can I recompile asterisk and it should work ? |
22:40.37 | hijacked | JT: you mean "pipelines" not threads, right? |
22:40.54 | JT | hijacked: each core can run 4 threads at once |
22:41.29 | mihinomenest | yeah, pipelines. |
22:42.12 | Qwell[] | I own a T2000 :D |
22:42.34 | Qwell[] | it's slick |
22:42.44 | file | yes... own... |
22:42.46 | russellb | Voice2: svn.digium.com/view |
22:42.47 | Qwell[] | indeed |
22:42.48 | JT | what do you use it for? |
22:42.54 | Qwell[] | JT: it sits in a rack taking up space |
22:43.00 | JT | send! |
22:43.05 | Qwell[] | nah :p |
22:43.08 | *** join/#asterisk MACscr (n=MACscr@adsl-75-23-64-115.dsl.peoril.sbcglobal.net) |
22:43.11 | Qwell[] | doesn't Sun do the try and buy in Oz? |
22:43.37 | JT | try before you buy? |
22:43.46 | Qwell[] | yeah, they'll ship you one for free for like 60 days |
22:43.57 | Qwell[] | and they'll pay for return shipping - at least in the US |
22:44.00 | JT | i'm guessing you can't just be anyone ;) |
22:44.07 | Qwell[] | I'm just anyone |
22:44.18 | Qwell[] | hell, I even told them that I didn't have a credit card or a company |
22:44.19 | JT | didn't get it under a company name? |
22:44.20 | *** join/#asterisk af_ (n=getsmart@ip-156-32.sn2.eutelia.it) |
22:44.22 | Qwell[] | I did |
22:44.24 | Qwell[] | ... |
22:44.28 | JT | heh |
22:44.34 | Qwell[] | and I was *very* clear that it was a fake company name |
22:44.41 | JT | hmm ok |
22:45.00 | JT | not sure if they're that liberal here but i'm sure they have a similar program |
22:45.09 | MACscr | can someone point me to the updated asterisk manual. I keep stumbling upon depreciated things |
22:45.26 | Qwell[] | MACscr: deprecated |
22:46.14 | MACscr | Qwell[]: correction noted |
22:46.17 | mmartinn | Both :P |
22:51.12 | orlock | JT: ahh, these are all dual core, reason it was $13k is its a carrier grade system.. DC ower, etc |
22:51.24 | data23 | tum te tum, 70 mins till ps3 is finally launched over ere, might try installing * on it at the w/e if i'm bored, you can get USB fxs adapters right? |
22:52.08 | *** join/#asterisk jm|home (n=jm@zen.jamiem.com) |
22:54.34 | JT | orlock: ah, not a T2000? |
22:54.52 | JT | usb = :( |
22:55.14 | Mercestes | any PRI geniuses wanna help me troubleshot a HDLC abort issue? |
22:55.17 | data23 | hmm i meant FXO anyway, always get them two mixed up :) |
22:56.49 | fetcher | Is there a way to specify *preferred* codec(s) in sip.conf, rather than just allowed/disallowed ? |
22:57.07 | Mercestes | fetcher, It goes in order of your speceification. |
22:57.09 | JT | data23: just use an ethernet connected ATA |
22:57.26 | Mercestes | fetcher: So if you allow ulaw, g729, gsm it will prefer ulaw |
22:57.48 | data23 | aye but not quite as fun :), You were right btw JT, i never did get to the bottom of my ECT/2BCT issues and the developer of the patch has disappeared :) |
22:58.03 | fetcher | Mercestes: or, allow=g729, then allow=ulaw on the next line would use g.729 preferentially, then fall through to ulaw if all licenses were in use? |
22:58.06 | *** join/#asterisk Cyon (n=cyon@216.179.31.170) |
22:58.18 | tuan_modulis | here's a business idea that was pitched to me... .a myspace.... on a phone network |
22:58.32 | tuan_modulis | do u think it might work at all? |
22:58.33 | Qwell[] | tuan_modulis: *gasp*, you could call it a "party line" |
22:58.36 | Mercestes | no...allow = g729,ulaw would .....probably give you an issue if all licenses were in use. |
22:58.36 | Cyon | Hey, anyone have a rough count of the number of asterisk installs in North America? |
22:58.48 | JT | data23: heh, it's not really well supported at all in asterisk, ECT |
22:59.00 | Qwell[] | :( |
22:59.00 | fetcher | tuan_modulis: the myspace that already exists is bad enough :) |
22:59.14 | Mercestes | fetcher, But theorhetically, you wuld use allow=g729,ulaw not seperate allow statements...but..I think both are valid syntaxes by my recollection |
22:59.32 | tuan_modulis | the only thing that has going for it is voicemail... and saydigits |
22:59.38 | Mercestes | fetcher, however, I think ifyou run out of licenses it gives you a hard "out of licenses" error and returns congestion without trying other codecs. I would try it and see. |
22:59.57 | *** join/#asterisk mosty (i=mostyn@60-241-198-194.static.tpgi.com.au) |
23:00.01 | fetcher | Mercestes: ah... will have to do some testing then. |
23:00.28 | JT | data23: usb is not fun anyway... |
23:00.29 | Mercestes | fetcher, Yea, give it a test. if you think it's hard erroring without trying any other codecs then bug report it. it *should* roll back to ulaw if g729 fails for whatever reason tho. |
23:01.29 | *** join/#asterisk `p4r14h`work (n=josh@24-119-48-78.cpe.cableone.net) |
23:02.40 | Mercestes | pretty please digum /pri expert support? |
23:03.00 | JT | Mercestes: what's the problem? |
23:03.39 | Mercestes | JT: HLDC aborts |
23:04.04 | JT | is your zttest scoring ok? |
23:04.04 | orlock | wow |
23:04.04 | JT | no interrupt sharing? |
23:04.14 | orlock | JT: you know those shitty USB phones? |
23:04.20 | JT | yes |
23:04.33 | Mercestes | JT: Zttest is awesome. I have 3 PRIs in a 4port card. PRI 1 works great. PRI2 works great... |
23:04.38 | Mercestes | I plug in PRI 3 and PRI 2 goes to hell |
23:04.49 | orlock | JT: i have one plugged into my desktop, and have been playing with vmware.. vmware is using it for audio! |
23:04.54 | JT | Mercestes: haven't we been through this before? |
23:04.59 | Mercestes | In fact...after I reset configs back to orig..PRI 2 was still giving me hell, I had to warm recycle zaptel while asterisk was running to stop the errors |
23:05.06 | JT | orlock: okay, and? :) |
23:05.07 | Mercestes | JT: yes |
23:05.16 | orlock | JT: nothing, its just cool :) |
23:05.28 | JT | orlock: they're just a sound card basically |
23:05.28 | orlock | i have forgotten how nice commercial software can be compared to open source stuff |
23:05.46 | JT | nice... highly debatable |
23:06.04 | fetcher | Mercestes: that sounds like a possibly a timing source problem |
23:06.09 | JT | Mercestes: speak to twtc at all? |
23:06.32 | Mercestes | JT: Yea. They test ok on their end. In fact. I unpluged PRI1 adn plugged TWTC into it without changing anything and everything was ok if I did that. |
23:07.09 | Mercestes | what gets me is PRI2 was giving errors under the original configs and the only way I could stop it was to recycle zaptel with asterisk humming |
23:07.09 | JT | okay |
23:07.15 | Mercestes | *THAT* bothers me. |
23:07.27 | JT | yeah sometimes zaptel does that |
23:07.50 | Mercestes | that makes troubleshooting very difficult. |
23:08.04 | Mercestes | starting zaptel after astierisk also gernates errors |
23:08.45 | mosty | is there any way to find out why a channel died? the logs show "spawn extension (context, number etc) exited non-zero on SIP/foo) |
23:12.16 | tuan_modulis | the wiki explains a method.... but... it's time consuming |
23:14.21 | Strom_M | that was fun |
23:14.25 | Strom_M | I just ordered ISDN |
23:14.46 | JT | Strom_M: is this what the silly buggers with a b410p is about? :) |
23:14.52 | Strom_M | yes |
23:15.06 | JT | guinnea pig |
23:15.10 | Strom_M | :) |
23:15.15 | Strom_M | hacktacular |
23:15.24 | JT | do you think it will actually work? |
23:15.26 | russellb | Strom_M: um. chan_misdn does not support bri in the us ....... |
23:15.43 | Strom_M | russellb: yeah, but the question is how different NI1 is from Euro |
23:15.47 | JT | who cares about chan_misdn anyway ;) |
23:15.56 | russellb | JT: people using the b410p. |
23:16.06 | russellb | or the various other bri cards |
23:16.09 | JT | haven't used one myself |
23:16.12 | JT | umm bristuff |
23:16.15 | mosty | tuan_modulis: a method for what? |
23:16.20 | JT | everything else definately works with it |
23:16.32 | Strom_M | russellb: because I've got a PRI circuit here which is set to NI2 on the network side and Euro on the CPE side, and the damn thing works just fine |
23:16.34 | russellb | umm not everyone wants some random unsupported patchset :-p |
23:16.35 | tuan_modulis | mosty: lemme find it... |
23:16.54 | JT | russellb: err, no more random than the unsuported patchset of misdn |
23:17.01 | JT | misdn is alpha software |
23:17.11 | *** join/#asterisk P4C0 (n=ash@200.124.22.34) |
23:17.13 | JT | and has really *bad* NT mode support |
23:17.14 | russellb | misdn is included in asterisk, so we accept bug reports on it |
23:17.33 | JT | it's also nice to be able to use all zap features |
23:18.12 | Strom_M | russellb: so yes, i know full well going into this that it may not work at all |
23:18.13 | P4C0 | hello guys, one question... I'm planning to buy a license for codec g.729, however in the page says that you need one license per channel (or that the license (10.00 usd) is only valid for one channel... is that the same as one call at a time? |
23:18.22 | Strom_M | P4C0: yes |
23:18.30 | *** join/#asterisk JT_ (n=jon@unaffiliated/jt) |
23:18.41 | JT | misdn debugging is horrible as well |
23:19.15 | tuan_modulis | mosty: actually, my bad, it's to debug a deadlock |
23:19.22 | tuan_modulis | http://www.voip-info.org/wiki/view/Asterisk+debugging |
23:20.07 | P4C0 | Strom_M, humm right now , I'm using alaw between my server provider and my asterisk box, and I can have multiple calls at the same time, if I buy this license and change that link to 729, I'll only be able to have one call? so if I'm calling from an internal terminal and the voip provider pass me a call what will happend? |
23:20.10 | mosty | thanks anyway |
23:20.28 | Strom_M | P4C0: you need the license to transcode |
23:20.29 | mosty | P4C0: only one g729 call |
23:20.40 | Strom_M | P4C0: if you do g729 passthrough, you dont need said license |
23:21.17 | JT_ | I run BRI on production systems in TE and NT mode, and i can't stand misdn |
23:21.18 | P4C0 | Strom_M, my internal phones doesn't support g729... so I need to transcode... |
23:21.24 | JT_ | too buggy and lacking in features |
23:22.10 | P4C0 | btw, since in the local net bandwidth is not a problem, what's the best codec where i can transcode the g729? (less cpu usage in the transcode?) |
23:22.20 | JT_ | g.711 |
23:23.03 | spanglesontoast | anyone know why asterisk doesn't compile the chan_zap.so module |
23:23.06 | JT_ | what sort of phones don't support g.729 anyway? |
23:23.19 | Strom_M | P4C0: if you don't have bandwidth problems, don't bother with g729 |
23:24.01 | P4C0 | Strom_M, I don't have internal (local net bandwith problems, but the link between my voip provider is of 700 kbps) |
23:24.14 | Strom_M | 700kbps is gobs |
23:24.40 | Strom_M | ive only got a 768kbps uplink at home, and it's ulaw all the way for me |
23:25.03 | *** join/#asterisk quidpro (n=quid@CPE00131078ba5d-CM000f9f7eff1e.cpe.net.cable.rogers.com) |
23:25.33 | russellb | spanglesontoast: get zaptel from svn |
23:29.20 | JT | hrm, asterisk using 98% cpu |
23:29.28 | JT | seems non-optimal |
23:29.32 | JT | 1.2.16 |
23:29.46 | JT | <PROTECTED> |
23:30.06 | data23 | :} |
23:30.24 | JT | the thing is |
23:30.37 | JT | it's been doing absolutely nothing for over 12 hours, that instance |
23:31.45 | P4C0 | steve___, 768 is my down... up is like 128 or so |
23:31.55 | P4C0 | steve___, ups sorry that was for Strom_M ... |
23:32.08 | mosty | i have a wctdm card which isnt plugged into my phone line yet, when i try to dial over the card, the card answers the call and just waits. is there a way to make it realise that it cant make the call and return CHANUNAVAIL or something? |
23:32.12 | P4C0 | Strom_M, actually let me check my real badnwitdth |
23:32.49 | JT_ | why do people say "ups" when they mean "oops", i don't get that |
23:34.28 | spanglesontoast | why svn I just wnat the module |
23:34.43 | JT_ | spanglesontoast: what version of asterisk? |
23:34.51 | spanglesontoast | 1.4.2 |
23:35.25 | JT_ | spanglesontoast: you need to download zaptel from 1.4 SVN |
23:35.33 | P4C0 | Strom_M, 733 kbps down, 249 kbps up... i think g729 will improve things... what do you think? |
23:35.42 | spanglesontoast | I've got the current zaptel |
23:35.43 | spanglesontoast | .............. |
23:35.45 | JT_ | as the release version of zaptel that works for 1.4.2 is not out i believe |
23:35.55 | Strom_M | P4C0: yeah, it'll help, but it sounds like ass |
23:36.00 | JT_ | spanglesontoast: the latest RELEASE of zaptel does NOT work with 1.4.2 |
23:36.08 | JT_ | you must download zaptel from 1.4 SVN |
23:36.14 | spanglesontoast | where that ? |
23:36.20 | JT_ | until the next release comes out |
23:36.35 | JT_ | use a svn client or svn.digium.com |
23:36.39 | *** join/#asterisk kink0 (n=k@161.pool62-37-205.static.orange.es) |
23:36.41 | kink0 | hello |
23:37.00 | spanglesontoast | also how do I remove the existing compiled one ? |
23:37.19 | kink0 | a stupid question, when you get over 99.98% with zttest, the zttest is running while many calls are in progress ? |
23:37.43 | JT_ | kink0: depends on the machine, but on some yes |
23:38.16 | fetcher | P4C0: are you talking to another Asterisk on the far end, or directly to another SIP phone? |
23:38.37 | kink0 | JT_ is because I get 100% worst if I run while no calls or just fews calls, but while 40 calls, i get a worst value of 99.65% |
23:38.50 | P4C0 | fetcher, another asterisk... my voip service provider asterisk (it may not be asterisk... thro) |
23:39.27 | JT_ | kink0: 99.65 is unacceptable usually, do you notice problems? |
23:39.47 | kink0 | JT_ yes, noise whith 99.65 |
23:39.48 | fetcher | P4C0: some providers support iLBC, which is similar in bandwidth & quality to G.729 but doesn't require licenses |
23:40.05 | JT_ | kink0: yeah that's no good |
23:40.08 | spanglesontoast | is there no other way of getting conferencing to work without timing |
23:40.09 | JT_ | kink0: what are the specs? |
23:40.10 | P4C0 | fetcher, mine only alaw and g729 |
23:40.21 | fetcher | P4C0: also, using IAX instead of SIP, if you aren't already, will save a bit of bandwidth |
23:40.24 | kink0 | JT_ but I tried to compile kernel again and again, dedicate one cpu to TE405, enable and dissable ec, enable and dissable HT, no way |
23:40.57 | kink0 | JT_ I try two boxes, one is Dual Xeon 2.8, Supermicro, the other is Dual Xeon 3.2/2MCache, Supermicro |
23:41.01 | fetcher | P4C0: may be worth throwing a few test calls at them using various codecs anyway, just to check for undocumented features ;) |
23:41.02 | P4C0 | fetcher, my voip provider doesn't support iax |
23:41.13 | kink0 | JT_ no other user software is running ( except sshd ) |
23:41.16 | P4C0 | fetcher, already did :p |
23:41.34 | fetcher | oh, well... |
23:41.38 | JT_ | kink0: hrm, how many PRIs? |
23:41.45 | kink0 | 4 PRI |
23:41.50 | JT_ | hmm |
23:42.08 | JT_ | not sure what to suggest |
23:42.15 | kink0 | JT_ I use g729 for most, but CPU idle still over 50% |
23:42.28 | JT_ | kink0: any particular options you set in the kernel? |
23:42.36 | JT_ | kink0: g.729 kills cpu |
23:42.50 | kink0 | JT_ no, nothing. I also try some boot options, but same. |
23:43.08 | P4C0 | I'll go now, fetcher Strom_M thanks for your tips, c u |
23:43.11 | kink0 | yes, but as I get a high idle CPU, I think g729 is not the cause of lose IRQ |
23:43.15 | JT_ | you should set kernel HZ to 1000 |
23:43.22 | JT_ | not sure if it will have a major impact |
23:43.23 | kink0 | yes, is setted to 1000 |
23:43.38 | kink0 | no impact, changing from 250 to 1000 |
23:43.44 | JT_ | is the bus shared with anything else? |
23:45.06 | spanglesontoast | JT_ which version are you using |
23:45.09 | kink0 | JT_, I think is not shared, as well as is ussing IRQ 48, and there no other device with that IRQ, but... If I do lspci -vb I see there others ussing IRQ 5 |
23:45.10 | *** join/#asterisk JT (n=jon@unaffiliated/jt) |
23:45.41 | JT_ | spanglesontoast: 1.2. branch |
23:45.44 | kink0 | I have try boot option append=noapi but that did nothing |
23:45.55 | JT_ | 1.4 is not stable enough for production without lots of testing |
23:46.07 | *** join/#asterisk nasls_lsa (n=chatzill@athedsl-148609.otenet.gr) |
23:46.07 | kink0 | I still having my IRQ APIC again |
23:46.25 | *** join/#asterisk Ryanw (n=cableguy@ge0-0-15-lns0.207alg.qx21.net) |
23:46.56 | rudholm | Strom_M: why did you order ISDN when DSL is so much faster? |
23:47.02 | spanglesontoast | which outta these http://ftp.digium.com/pub/zaptel/releases/ |
23:47.08 | JT_ | rudholm: for voice :P |
23:47.21 | Strom_M | rudholm: because ISDN makes my internets go TWICE AS FAST |
23:47.28 | rudholm | that's right! |
23:47.34 | rudholm | 'cause it's 2 |
23:47.35 | rudholm | ! |
23:47.36 | kink0 | JT_ do you know if FSB has any influence here ? |
23:47.50 | JT | kink0: not sure |
23:47.50 | rudholm | Front-Side Bus? |
23:48.05 | kink0 | I try 400 and 533Mhz FSB, no difference, but I have not try 800 or 1066 |
23:48.15 | JT | kink0: anyway, i asked if the bus was shared |
23:48.35 | rudholm | you people who live in !california really need to get on your PUCs. BRI is 25$/month here. |
23:48.51 | rudholm | and that includes 200 b-channel hours, iirc. |
23:48.56 | Ryanw | does the goto command do pattern matching lookups in the destination context or must the extension you go to exist exactly ? |
23:48.57 | kink0 | JT: APIC gives a unique IRQ, but I see if I do lspci -vb that there more devices ussing IRQ 5 |
23:49.03 | Strom_M | rudholm: 200!? really?? |
23:49.10 | JT_ | kink0: i did not ask about IRQs |
23:49.14 | JT_ | kink0: i asked about the bus |
23:49.18 | Strom_M | rudholm: is that for voice and data calls? |
23:49.19 | rudholm | Strom_M: yeah, on Residential BRI, not Business. |
23:49.26 | rudholm | it's for data |
23:49.36 | Strom_M | ah ok...what about for voice? |
23:49.59 | rudholm | voice is billed the same as POTS |
23:50.11 | rudholm | it's really quite a good deal. |
23:50.12 | kink0 | JT_, yes the PCI bus 2 is also SCSI and Ethernet |
23:50.23 | Strom_M | rudholm: ah ok, so the same $3 allowance? |
23:50.35 | rudholm | yeah, or possibly "unmeasured" |
23:50.36 | rudholm | either way |
23:50.37 | JT_ | kink0: try and give it a dedicated bus, some servers have 2 PCI buses |
23:50.39 | rudholm | it's good |
23:50.45 | mosty | i have a problem with my zaptel device, when it's not plugged into the line and i try to dial over it, it answers the call but just stops and waits there, when (i think) it should exit immediately and set DIALSTATUS to CHANUNAVAIL or something. what could be wrong? |
23:51.00 | rudholm | GTE was bad, all "data" calls were billed per minute and there was no allowance. |
23:51.06 | spanglesontoast | guess no one has a working conference module |
23:51.18 | JT_ | spanglesontoast: conferences work fine here. |
23:51.21 | kink0 | JT_ I see, but I have only one PCI slot available here, there just 2 PCI slots , one is for low profile card, and the other is where I place the TE405 card |
23:51.24 | rudholm | so I did a lot of data calls sent as voice calls. if you weren't calling very far away, it'd almost always work. |
23:51.26 | orlock | mosty: its not plugged in to the line maybe? |
23:51.35 | spanglesontoast | then why doesn't it compile the module with asterisk |
23:51.48 | JT_ | kink0: where is there room for scsi and ethernet then? |
23:52.19 | kink0 | JT_ are integrated on motherboard |
23:52.23 | orlock | rudholm: theres a similar thing here for ISDN.. making data calls flagged as voice |
23:52.25 | kink0 | sorry, SCSI is in bus 3 |
23:52.27 | mosty | orlock: if it's not plugged in, i dont want zap calls to just hang (i'm trying to setup a failover to dialling over iax) |
23:52.29 | JT_ | kink0: are you sure they're on the pci bus? |
23:52.33 | kink0 | only Eth and TE are both in bus 2 |
23:52.44 | JT_ | kink0: read the motherboard system block diagram |
23:53.03 | kink0 | JT_ I think so, at least I get listed with lspci |
23:53.04 | *** join/#asterisk shodan- (n=shodan@ip062.96-113-216.pppoe1.joliette.intermonde.net) |
23:53.06 | rudholm | orlock: yeah, I found with GTE's pricing for BRI, my ISDN bill went from 400$/month to 50$/month when I switched to data as voice. |
23:53.12 | JT_ | kink0: hrm ok, would be nice if they were on different buses, but i don't know if that's causing your problem or not |
23:53.15 | rudholm | orlock: this was back in the late 90s |
23:53.23 | *** join/#asterisk coppice (n=chatzill@90.203.17.210.dyn.pacific.net.hk) |
23:53.28 | orlock | rudholm: pci isdn cards? |
23:53.49 | kink0 | JT_ as I know, the bus is not selectable by software and is fixed at hardware, right ? |
23:54.06 | rudholm | orlock: I was using a Netopia "ISDN Router" at the time. |
23:54.09 | JT_ | kink0: yes |
23:55.01 | mosty | orlock: do you understand my problem? |
23:55.11 | orlock | mosty: yeah, but i dont know sorry |
23:55.24 | mosty | thanks anyway |
23:55.38 | JT_ | kink0: what are all your versions again? |
23:58.08 | *** join/#asterisk X-Gen (n=X-Gen@dsl-242-28-178.telkomadsl.co.za) |
23:58.47 | kink0 | JT_ I tried with 1.4.1 and 1.2.12 |
23:59.26 | kink0 | zaptel 1.4.0 and zaptel 1.2.9.1 |
23:59.41 | JT_ | hmm ok |