| 00:00.10 | sonic2wb | hackeron: wheres my beer |
| 00:00.14 | _Sam-- | what is your main point anonymouz666 i would like to hear |
| 00:00.36 | anonymouz666 | Everyday we have another new PBX company offering Asterisk... |
| 00:00.41 | sivana | tzanger: ping |
| 00:00.53 | _Sam-- | why is that bad? |
| 00:00.55 | tzafrir_laptop | anonymouz666, and Mark is making money from the work of Jim Dixon. |
| 00:01.19 | tzafrir_laptop | As long as both are making money, things are fine. |
| 00:01.22 | anonymouz666 | I think he helped Jim Dixon writing the loadables kernel modules? |
| 00:01.28 | hackeron | sonic2wb: hey, sorry, sec, just setting up apache for the firmware upgrade of these shitty Granstream GXP-2000 - havent forgotten about you :) |
| 00:01.34 | mog_work | he did anonymouz666 |
| 00:01.37 | _Sam-- | hack be careful |
| 00:01.40 | _Sam-- | fried one today |
| 00:01.42 | _Sam-- | gxp2000 |
| 00:01.43 | [hC] | anonymouz666: people are making money with the work of linus torvalds and community too. its their choice |
| 00:01.44 | mog_work | but that is the point of oss we all get rich ! |
| 00:01.46 | _Sam-- | updating to 13 |
| 00:01.50 | [av]bani | :o |
| 00:01.52 | mog_work | exactly |
| 00:01.53 | hackeron | _Sam--: oh? |
| 00:01.53 | _Sam-- | make sure there are no incoming calls |
| 00:01.57 | mog_work | we make money off of linus |
| 00:01.58 | [hC] | digium makes money |
| 00:02.00 | anonymouz666 | Linus has 1% kernel copyright |
| 00:02.05 | _Sam-- | i upgraded about 15 gxp2000s fine |
| 00:02.06 | mog_work | if it wasnt for kernel wed be sunk |
| 00:02.08 | [hC] | its not like theyre going broke |
| 00:02.12 | _Sam-- | but one died because a call came to it during upgrade |
| 00:02.15 | anonymouz666 | Mark Spencer wrote more than 70% of asterisk ? |
| 00:02.16 | _Sam-- | i talked to grandstream today |
| 00:02.20 | _Sam-- | its toasted |
| 00:02.25 | sonic2wb | lol |
| 00:02.30 | sonic2wb | that sucks |
| 00:02.33 | [hC] | mark knew what he was offering when he wrote it |
| 00:02.34 | [av]bani | lol |
| 00:02.34 | [hC] | he's not upset |
| 00:02.40 | [av]bani | sam -> warranty? |
| 00:02.52 | *** join/#asterisk mhnoyes_ (n=mhnoyes@user-2ivfnjj.dialup.mindspring.com) |
| 00:02.59 | _Sam-- | yeah they are going to hook me up, but i had to dump the phone first via ethereal so they could see what its sending |
| 00:03.03 | _Sam-- | which was nothing |
| 00:03.06 | [hC] | its great that people make money off of it and various people contribute |
| 00:03.12 | [hC] | that means that everyone benefits from everyone elses work |
| 00:03.16 | _Sam-- | as of now i dont have an RMA for it but they said they would take care of it |
| 00:03.18 | [hC] | and can each make more money based on the things that other people put into it |
| 00:03.30 | [hC] | im sure jim dixon makes money selling asterisk and its related components too that he DIDNT write |
| 00:03.33 | [hC] | the world is a happy equal place |
| 00:03.47 | *** join/#asterisk bkw__ (n=brian@adsl-69-148-35-86.dsl.tulsok.swbell.net) |
| 00:04.18 | anonymouz666 | this world is not fair |
| 00:04.34 | anonymouz666 | hehe |
| 00:05.03 | [av]bani | it isn't!?@? |
| 00:05.56 | tzafrir_laptop | riddlebox, can you start asterisk using the init.d script? |
| 00:06.39 | tzafrir_laptop | If so, trace that script using 'sh -x /etc.init.d/asterisk start' and see what command was used to start asterisk . |
| 00:07.03 | tzafrir_laptop | Probably something with start-stop-daemon |
| 00:07.33 | riddlebox | tzafrir_laptop, how would I try to start it with the init.d script, I reboot and asterisk does not start |
| 00:07.53 | tzafrir_laptop | su - # to get a root shell |
| 00:08.00 | tzafrir_laptop | /etc/init.d/asterisk start |
| 00:08.00 | riddlebox | I am there already |
| 00:08.15 | hackeron | _Sam--: hmm, thanks for the warning |
| 00:08.30 | _Sam-- | sure thing |
| 00:08.35 | _Sam-- | .13 works good for me too |
| 00:08.44 | _Sam-- | better sound quality in the earpiece |
| 00:08.48 | hackeron | _Sam--: just forced voicemail for the ugprade :) |
| 00:09.02 | riddlebox | tzafrir_laptop, I found the problem, /etc/default/asterisk needs to be edited |
| 00:09.03 | _Sam-- | good idea...i like those gxp2000s really |
| 00:09.05 | hackeron | _Sam--: sound is better than .12? |
| 00:09.09 | _Sam-- | for what they are |
| 00:09.23 | hackeron | $80 phones with enterprise features :) |
| 00:09.28 | _Sam-- | i had .09...never went to 12 |
| 00:09.56 | hackeron | we're on .09 too |
| 00:10.01 | _Sam-- | and i was really surprised how easily i got someone knowledgable on the phone |
| 00:10.07 | _Sam-- | and that they actually took care of me |
| 00:10.12 | _Sam-- | well we'll see about that still |
| 00:10.16 | _Sam-- | but it was a pleasant surprise |
| 00:10.18 | justinu | hc: what made you decide to swap out the kernel? |
| 00:11.42 | riddlebox | tzafrir_laptop, it is now telling me Starting Asterisk PBX: Unable to open pid file '/var/run/asterisk.pid': Permission denied |
| 00:11.42 | riddlebox | asterisk |
| 00:12.34 | tzafrir_laptop | if you want to run asterisk as auser, edit /etc/asterisk/asterisk.conf and set the rundir to /var/run/asterisk |
| 00:12.59 | tzafrir_laptop | and then mkdir /var/run/asterisk; chown asterisk /var/run/asterisk |
| 00:15.41 | riddlebox | tzafrir_laptop, now it says unable to setuid to 108 (asterisk) |
| 00:15.49 | *** join/#asterisk Koshatul (n=evangeli@ip157-65-132.cust.bit.net.au) |
| 00:16.20 | hackeron | _Sam--: how do you upgrade from phone directly. I see an upgrade after x days -- I tried setting it to 1, I tried setting it to 0, nothing |
| 00:16.30 | tzafrir_laptop | Is start-stop-daemon doing the setuid on its own? |
| 00:16.46 | *** join/#asterisk alephcom (n=alephcom@host75.net14.mcsnet.ca) |
| 00:16.49 | riddlebox | I dont know, this is a default install |
| 00:17.01 | tzafrir_laptop | Could you try removing the group setting? |
| 00:17.14 | riddlebox | I am not sure what you mean by that |
| 00:17.32 | hardwire | its like white.. but not.. and I am back in it. |
| 00:17.42 | tzafrir_laptop | Who exactly gives that error message? |
| 00:17.48 | hardwire | some. |
| 00:18.22 | riddlebox | tzafrir_laptop, when I type /etc/init.d/asterisk start I get that error |
| 00:18.44 | hardwire | except for a few misc issues |
| 00:18.47 | sonic2wb | riddlebox what error |
| 00:19.10 | tzafrir_laptop | can you trace the script and paste here the command start-stop-daemon is running? |
| 00:19.18 | riddlebox | starting Asterisk PBX: Unable to set high priority, unable to setuid to 108 (asterisk) |
| 00:19.22 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 00:19.26 | hardwire | just put "echo" before start-stop-daemon |
| 00:19.28 | hardwire | and run it |
| 00:21.37 | hackeron | _Sam--: oh, it just rebooted on its own, shows time and volume, no connection and cant connect to it from web interface, is that normal? |
| 00:22.29 | _Sam-- | i had to power down many of them...they got hung. all of them came back with the new version when they rebooted....i would just be patient, make sure you give it enough time to confirm that its really not doing anything |
| 00:22.50 | hackeron | _Sam--: hey, so did mine, beautiful :) |
| 00:23.52 | *** part/#asterisk frenzy (n=frenzy@196.45.144.41) |
| 00:24.23 | *** join/#asterisk apardo (n=apardo@174.Red-83-33-49.dynamicIP.rima-tde.net) |
| 00:24.38 | _Sam-- | something to remember...the up and down arrows on the phone wont change the ringer volume anymore |
| 00:24.45 | _Sam-- | you need to go to the menu and change it |
| 00:25.00 | _Sam-- | my sales guys were ssaying it rings like 3x as loud as before if you dont change it |
| 00:25.06 | *** join/#asterisk alephcom_ (n=alephcom@host75.net14.mcsnet.ca) |
| 00:25.21 | *** join/#asterisk mjmtaiwan (n=michael@ip68-100-65-52.dc.dc.cox.net) |
| 00:25.31 | *** join/#asterisk Defraz_ (n=t0tal@72.24.26.215) |
| 00:25.35 | [av]bani | sam -> so on a scale of 1 to 10, how badly do the gxp2000 suck |
| 00:25.59 | justinu | the gxp2000 is a fine phone if you can live with the lack of features and can sort out the echo |
| 00:26.11 | justinu | i'd give it a 6 |
| 00:26.28 | _Sam-- | id give them at least a 7 |
| 00:26.32 | justinu | mostly because the UI sucks |
| 00:26.42 | justinu | and some of the problems with sound quality |
| 00:26.43 | _Sam-- | i have 15 people that talk on them all day doing mail order sales |
| 00:27.03 | justinu | cool |
| 00:27.07 | justinu | you never had problems with echo? |
| 00:27.26 | _Sam-- | justinu: alot of the echo and sound quality problems from what i read from the wiki were when the phones were being used with zap interfaces...im 100% voip |
| 00:27.29 | alephcom_ | In my opinion, it all depends what you are comparing them with.... I have one on my desk and it's fine for what I expect. I've never had echo problems with it....I'd give it a 7 |
| 00:27.38 | _Sam-- | and i dont have any echo related to the phone that ive noticed |
| 00:27.39 | justinu | yeah, i'm 100% voip too, and they complain about echo :( |
| 00:28.14 | _Sam-- | its funny...i gave up my gxp20000 to the guy who's phone died... |
| 00:28.19 | _Sam-- | and i fired up a soft client.. |
| 00:28.24 | _Sam-- | and the first person i called complained about the echo |
| 00:28.25 | alephcom_ | Is it echoing in their heads? .:-) |
| 00:28.41 | justinu | softphones are the worst |
| 00:29.02 | justinu | they induce a lot of latency into the call because of the shitty sound driver architecture, i guess |
| 00:30.12 | _Sam-- | but with regards to those phones...again i think its a great value for the money...i dont know what you would compare it to at a similar price point |
| 00:30.19 | justinu | there's nothing |
| 00:30.28 | justinu | except a spa841 or budgetone |
| 00:30.49 | _Sam-- | justinu: what firmware version are your gxps? |
| 00:31.08 | justinu | 9 |
| 00:31.25 | justinu | what's your recommendation there? |
| 00:31.36 | _Sam-- | i have been on 13 long enough...i just upgraded 10 out 15 people yesterday |
| 00:31.41 | _Sam-- | will see what the comments are next week |
| 00:31.45 | _Sam-- | but so far so good |
| 00:31.58 | _Sam-- | that were crying for it because they couldnt stand the old annoying call waiting tones |
| 00:32.04 | justinu | ah |
| 00:32.30 | _Sam-- | ive got alot of positive feedback that the calls on both ends sound better |
| 00:32.34 | mjmtaiwan | Hi... can I interupt and ask a question? |
| 00:32.38 | justinu | no! |
| 00:32.41 | justinu | (yes) |
| 00:32.58 | sonic2wb | lol |
| 00:33.03 | mjmtaiwan | Thanks... Its about setting up to recive calls via broadvoice. |
| 00:33.19 | _Sam-- | shouldnt be too hard |
| 00:33.32 | mjmtaiwan | By the volume of posts and google hits ... its a common problem. |
| 00:33.50 | _Sam-- | you are using your asterisk to connect to broadvoice? |
| 00:33.54 | mjmtaiwan | yes |
| 00:33.59 | _Sam-- | behind any firewalls? |
| 00:34.14 | justinu | sam, where you from? |
| 00:34.15 | mjmtaiwan | hmm... just my belin wireless router at home |
| 00:34.19 | _Sam-- | when you do sip show registry does it show you registered? |
| 00:34.28 | _Sam-- | im near philadelphia, how about yourself? |
| 00:34.34 | justinu | los angeles |
| 00:34.35 | mjmtaiwan | yes |
| 00:34.38 | mjmtaiwan | I do register |
| 00:34.41 | justinu | trying to find someone who lives around here |
| 00:35.06 | _Sam-- | when you do 'sip show peers' do you see broadvoice? |
| 00:35.12 | _Sam-- | from the console |
| 00:35.30 | mjmtaiwan | yes |
| 00:35.39 | _Sam-- | and you can make an outgoing call ok? |
| 00:35.49 | mjmtaiwan | I've opend up port 5060 in the firewall and pointed to the server. |
| 00:35.54 | mjmtaiwan | Yes... can make outgoing calls. |
| 00:36.10 | _Sam-- | when you try an incoming call , you see nothing at all on your console? |
| 00:36.31 | mjmtaiwan | yes... I can post it here if you like |
| 00:36.38 | justinu | what's sip show registry say? |
| 00:36.47 | _Sam-- | paste it to pastebin maybe |
| 00:36.51 | justinu | yeah |
| 00:36.53 | _Sam-- | or here if its not too big? |
| 00:37.03 | justinu | one or 2 lines is ok |
| 00:37.05 | justinu | no more |
| 00:37.14 | mjmtaiwan | it shows the host the user name refresh =23 state = registered |
| 00:37.19 | justinu | ok |
| 00:37.40 | mjmtaiwan | what is the pastebin? |
| 00:37.45 | _Sam-- | heh i have a broadvoice test account that i cant register to :) |
| 00:37.48 | justinu | did you put the inbound number in your context? |
| 00:38.16 | mjmtaiwan | well... I've ripped out the entire sip.conf and extentions.conf and am trying to start over |
| 00:38.17 | justinu | whatever context you set for broadvoice |
| 00:38.26 | mjmtaiwan | I've got just the bare min in both now |
| 00:38.27 | justinu | paste both those files |
| 00:38.32 | justinu | obfuscate both passwords |
| 00:38.38 | *** join/#asterisk tessier (n=treed@rrcs-67-53-110-66.west.biz.rr.com) |
| 00:38.38 | justinu | or all passwords |
| 00:38.54 | justinu | pastebin.com |
| 00:39.13 | mjmtaiwan | lol.... pastebin.com... I'll look it up |
| 00:39.27 | justinu | ~pb |
| 00:39.29 | jbot | from memory, pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
| 00:39.51 | mjmtaiwan | ok... do I just pop it in or register or something |
| 00:39.54 | justinu | mjm, you in taiwan? |
| 00:39.59 | justinu | just paste it in |
| 00:40.00 | hackeron | I have the Grandstream GXP-2000 and the manual says supports ilbc but ilbc isnt in the list even with the latest firmware. Anyone has ilbc working with this phone? |
| 00:40.02 | justinu | no registration |
| 00:40.06 | mjmtaiwan | na... lived there for 6 years. |
| 00:40.26 | _Sam-- | for my external stuff i use all IAX...but sip doesnt even work through one firewall / wireless router type device? |
| 00:40.31 | justinu | mjm: crap, i was hoping you were gonna set me up with a taiwanese DID |
| 00:40.34 | mjmtaiwan | posted the extensions.conf |
| 00:40.42 | justinu | sam: you can make SIP work |
| 00:40.47 | justinu | mjm: what's the link? |
| 00:40.52 | justinu | paste the link here |
| 00:40.56 | hackeron | _Sam--: no, it works, just set nat=yes |
| 00:41.03 | _Sam-- | ok just checking |
| 00:41.09 | hackeron | _Sam--: I have a phone behind a NAT in london connecting to an asterisk server in new york |
| 00:41.27 | justinu | gotta love the internet |
| 00:42.05 | hackeron | justinu: yeah, they just dial my extension and my phone rings in london :) |
| 00:42.05 | mjmtaiwan | sorry justinu |
| 00:42.10 | _Sam-- | hack what does the asterisk server in new york connect? PRI or aanother VOIP? |
| 00:42.25 | hackeron | _Sam--: IAX to teliax |
| 00:42.30 | mjmtaiwan | http://pastebin.com/494275 |
| 00:42.38 | *** join/#asterisk kernoman (n=kernoman@82-33-9-8.cable.ubr14.newt.blueyonder.co.uk) |
| 00:42.58 | _Sam-- | and when you call from london-->new york--and out teliax, the delay isnt too long to degrade the call quality? what is the ping time from london to teliax? |
| 00:43.25 | hackeron | _Sam--: sounds so good its scary |
| 00:43.34 | _Sam-- | when you call the out from the new york asterisk server to teliax...does that server in new york get out of the way and bridge the call between teliax and you? |
| 00:43.36 | hackeron | _Sam--: and its a business dsl on the asterisk, not even a t1 |
| 00:43.37 | justinu | i talk to pakistan over public internet |
| 00:43.40 | justinu | it works great |
| 00:43.50 | hackeron | _Sam--: yes, it does if nat=yes |
| 00:44.01 | kernoman | i need to set up a dial rule that takes a number like 801503....... and converts it to just 1503... but I dont know the syntax |
| 00:44.10 | hackeron | _Sam--: no wait, other way around |
| 00:44.20 | hackeron | _Sam--: it all goes through asterisk when nat=yes |
| 00:44.30 | justinu | mkmtaiwan: you don't need line 10 |
| 00:44.34 | *** join/#asterisk secret_ (n=secret@pool-71-251-78-124.tampfl.fios.verizon.net) |
| 00:44.38 | justinu | or line 5 |
| 00:45.10 | _Sam-- | i see ...im just trying to figure out how some hosted PBX stuff would work out...if my asterisk box would get out of the way |
| 00:45.17 | hackeron | _Sam--: hmm, can it do that? |
| 00:45.21 | justinu | yes |
| 00:45.30 | justinu | it's called a sip reinvite |
| 00:45.37 | _Sam-- | if i had people connect to my asterisk via SIP...and then my asterisk calls out through teliax...would the caller be connected directly to teliax |
| 00:45.50 | _Sam-- | i have some learnin to do |
| 00:45.55 | justinu | teliax is iax |
| 00:45.59 | justinu | they don't support reinvite |
| 00:46.04 | _Sam-- | they do SIP as well |
| 00:46.08 | mjmtaiwan | ok... 5 is out for the moment as is 10 |
| 00:46.10 | *** join/#asterisk Jzalae (n=sk@dsl-66-63-110-48.gwi.net) |
| 00:46.11 | justinu | ok |
| 00:46.21 | *** join/#asterisk kio (n=kio@ool-4577adba.dyn.optonline.net) |
| 00:46.26 | mjmtaiwan | ready for the sip.conf |
| 00:46.27 | justinu | mjm taiwan, and sip.conf? |
| 00:46.29 | _Sam-- | so that is a function of canreinvite=yes in the sip.conf of the client? |
| 00:46.31 | justinu | yes, put it up |
| 00:46.35 | mjmtaiwan | fast |
| 00:46.36 | justinu | sam: exactly. |
| 00:46.48 | _Sam-- | is there anything need to be set on the client? |
| 00:46.52 | *** join/#asterisk Sedorox (i=brandon@smartserv/cna/Sedorox) |
| 00:46.52 | justinu | no |
| 00:47.15 | justinu | sip devices are intelligent about this stuff |
| 00:47.16 | mjmtaiwan | http://pastebin.com/494286 |
| 00:47.18 | _Sam-- | and my asterisk would be totally 100% out of the loop? |
| 00:47.25 | justinu | sam: for media, yes. |
| 00:47.29 | justinu | for signalling? no. |
| 00:47.39 | hackeron | _Sam--: hmm, so all thats requires is to make asterisk use sip instead of iax and it should do that? |
| 00:48.04 | _Sam-- | you are asking me? i just am learning the same thing as you! |
| 00:48.04 | justinu | i'm not sure if canreinvite=yes works with nat=yes |
| 00:48.07 | justinu | listen to this guys |
| 00:48.15 | *** join/#asterisk tessier (n=treed@rrcs-67-53-110-66.west.biz.rr.com) |
| 00:48.20 | justinu | there's two different ways of hooking up phones behind nat |
| 00:48.28 | justinu | one, is to set nat=yes on asterisk side |
| 00:48.40 | kernoman | i need to set up a dial rule that takes a number like 801503....... and converts it to just 1503... but I dont know the syntax |
| 00:48.50 | justinu | what that does, is tells *: "the phone is lying about it's IP address (it thinks it's on an internal non-routable network" |
| 00:48.52 | hackeron | _Sam--: well, the phones get bridged if I remove nat=yes so I dont see why not. Very interesting. |
| 00:49.00 | mjmtaiwan | this sip.conf is a mix of many different sources... books ... posts, how toos... that's why I striped most out and tried just to get in and out calling over broadvoice to work first. |
| 00:49.33 | justinu | when nat=yes, * takes the IP address the phone is connecting from, and stuffs it in place of the IP the phone told * to contact it on. |
| 00:49.54 | justinu | that puts the burden of supporting nat on the * server itself |
| 00:50.10 | hackeron | justinu: externalip=x |
| 00:50.11 | justinu | follow that? |
| 00:50.29 | _Sam-- | yes |
| 00:50.33 | hackeron | justinu: oh, wait, misread |
| 00:50.38 | justinu | the 2nd way to make a phone behind NAT work is by setting nat=no |
| 00:50.45 | justinu | and turning on the STUN support on the phone itself |
| 00:50.58 | justinu | this makes the PHONE intelligent enough to tell asterisk about it's EXTERNAL IP |
| 00:51.02 | hackeron | justinu: where do you get a stun server you can trust though? |
| 00:51.08 | justinu | you put up your own |
| 00:51.21 | justinu | or federate with someone |
| 00:51.26 | hackeron | are there any stable stun servers for linux? |
| 00:51.30 | justinu | stun takes nothing to setup, it's super simple |
| 00:51.47 | _Sam-- | in order for that to work, the stun server has to be on a public internet ip right? |
| 00:51.55 | justinu | yeah |
| 00:52.04 | _Sam-- | obviously...just making sure |
| 00:52.04 | justinu | i just use xten's stun :P |
| 00:52.23 | justinu | anyways, my whole point was that IF you use method #2, you can do reinvites |
| 00:52.34 | _Sam-- | for every call you make you are relying on their server being running and that you can connect to it? |
| 00:52.38 | _Sam-- | their stun server |
| 00:52.43 | hackeron | justinu: there are no stun servers for linux I can find that arent pre-alpha |
| 00:53.03 | justinu | but, only if UA2 is on a public IP also |
| 00:53.21 | justinu | stun is pretty simple, i have a java class that implements it |
| 00:53.25 | _Sam-- | what happens if a route is down between you and the stun server |
| 00:53.29 | _Sam-- | does the call work or not |
| 00:53.43 | justinu | it works as long as the clients IP didn't change since the last STUN query, i guess |
| 00:54.21 | _Sam-- | so just to help me understand...the phone sends a request to the stun server for its own ip address? then when it has it, it uses that as the ip for the call? |
| 00:54.33 | justinu | yeah, and it also determines the type of NAT it's behind |
| 00:54.37 | justinu | xlite does this |
| 00:54.53 | hackeron | justinu: huh? -- the phone routes UDP traffic through STUN, if STUN is down, phone is down. Knowing the client IP doesnt help if you cannot connect to the IP |
| 00:55.00 | justinu | mjmtaiwan: still there? |
| 00:55.05 | mjmtaiwan | Yo |
| 00:55.10 | justinu | hackeron: that's not how stun works :) |
| 00:55.24 | hackeron | justinu: heh, fair enough, not a STUN expert :) -- how does it work? |
| 00:55.42 | justinu | all it does is find out what external IP and what external PORT you're getting, and sends it back to you |
| 00:55.54 | justinu | you then fill in the SIP headers with that info, instead of what comes from the OS |
| 00:56.18 | justinu | mjm: looks ok |
| 00:56.24 | hackeron | justinu: what external port? - there is no external port for the phone |
| 00:56.35 | justinu | mjm: sip debug peer sip.broadvoice.com |
| 00:56.42 | justinu | make an inbound call |
| 00:56.46 | justinu | paste the entire sip dialog |
| 00:56.48 | mjmtaiwan | yea... that is what I thought. I can't figure why. |
| 00:56.58 | mjmtaiwan | here gos |
| 00:57.00 | justinu | hackeron: the external UDP port |
| 00:57.01 | mjmtaiwan | goes |
| 00:57.12 | justinu | you do understand how NAT works, right? |
| 00:57.27 | hackeron | justinu: the gateway opens an external UDP port? |
| 00:57.46 | justinu | most cheap NAT devices pass UDP from LAN->WAN |
| 00:58.03 | _Sam-- | what is the address for mjm's pastebin? |
| 00:58.04 | hackeron | justinu: ah, wow, thats amazing. Why isnt that used for p2p? |
| 00:58.09 | justinu | hackeron: it is! |
| 00:58.14 | hackeron | justinu: which p2p!? |
| 00:58.19 | mjmtaiwan | http://pastebin.com/494286 |
| 00:58.34 | mjmtaiwan | just getting a busy signal and nothing in the CLI |
| 00:58.45 | *** join/#asterisk sigmounte (n=sigmount@www.sighq.net) |
| 00:58.54 | kernoman | i need to set up a dial rule that takes a number like 801503....... and converts it to just 1503... any help appreciated |
| 00:59.21 | kernoman | i.e chops of the leading 8 and 0 |
| 00:59.21 | justinu | hackeron: the problem starts when you have two devices behind port restricted cone NATs |
| 00:59.36 | justinu | that want to talk to each other |
| 00:59.39 | wunderkin | kernoman, README.variables |
| 00:59.41 | justinu | mjmtaiwan: did you turn on sip debug like I said? |
| 01:00.04 | _Sam-- | justinu: mjm doesnt have an extension definred for the phone number in extensions.conf |
| 01:00.06 | kernoman | thanks, dumb question but where do i find that? |
| 01:00.07 | sonic2wb | O_O time for STARGATES |
| 01:00.13 | justinu | sam: oh, you're right ;) |
| 01:00.23 | *** join/#asterisk alephcom (n=alephcom@host75.net14.mcsnet.ca) |
| 01:00.27 | mjmtaiwan | yep... I'll try again but its hard to get useful output without geting everything |
| 01:00.43 | _Sam-- | hold on i'll add what you need to the pastebin |
| 01:00.47 | justinu | sam is right |
| 01:00.53 | mjmtaiwan | Thanks Sam |
| 01:00.55 | wunderkin | kernoman, doc directory inside your asterisk source directory |
| 01:00.56 | hackeron | justinu: hmm, I see. So what p2p uses STUN. bittorrent doesnt, right? |
| 01:01.01 | kernoman | thanks |
| 01:01.03 | justinu | hackeron: i think they all do |
| 01:01.12 | justinu | because you can talk to stuff with public IPs |
| 01:01.17 | justinu | you wouldn't even be able to do that without STUN |
| 01:01.36 | justinu | you can't talk to other people behind port restricted cone NATs tho |
| 01:01.58 | justinu | because each client needs to make an outbound connection to open up the NAT binding, but they can't do it |
| 01:02.06 | mjmtaiwan | THis pastebin has the output of sip debug = http://pastebin.com/494305 |
| 01:02.23 | justinu | google talk is using STUN in a way that's better than most people tho |
| 01:02.37 | _Sam-- | mjmtaiwan: look at the pastbin 494306 |
| 01:02.38 | justinu | it's pretty ingenious, but it still requires a server |
| 01:02.47 | _Sam-- | as long as you have SIP/200 set up right that will ring it |
| 01:03.05 | justinu | hackeron: you might be able to tell, i've worked a lot on the issue of UDP behind NAT :) |
| 01:03.05 | _Sam-- | add the part that says add this part to your extensions.conf |
| 01:03.18 | hackeron | justinu: no, all p2p I tried says if I'm behind a nat and other person is behind a nat, you're shit out of luck. At least 1 party has to have a listening port and my nat has no blocked ports, its just no ports are accessible externally unless I do port forwarding because the router goes online |
| 01:03.40 | justinu | hackeron; tht's right |
| 01:04.00 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
| 01:04.03 | hackeron | justinu: so where does STUN come in? |
| 01:04.19 | justinu | they real key to make stun work (like google talk does) |
| 01:04.34 | justinu | is to take both clients, have them do STUN to figure out their NAT type and external IPs |
| 01:04.42 | justinu | report them to a man in the middle |
| 01:04.49 | justinu | the man in the middle exchanges that information with each client |
| 01:04.53 | _Sam-- | alright good luck mjm...i gotta get off this idiot box |
| 01:05.01 | justinu | and they connect to the right ports, and voila, it works! |
| 01:05.04 | mjmtaiwan | sam - didn't work |
| 01:05.08 | mjmtaiwan | still getting busy |
| 01:05.15 | _Sam-- | it only didnt work because you dont have sip/200 extension setup right |
| 01:05.15 | justinu | paste the sip debug |
| 01:05.30 | mjmtaiwan | sorry... to extentions.conf./// lets try that again! |
| 01:05.33 | hackeron | justinu: ah, I get it, its the person in the middle thats the problem - the p2p I'm talking about tries to be decentralised |
| 01:05.37 | _Sam-- | yes extensions.conf with that |
| 01:05.44 | _Sam-- | then extensions reload on the console |
| 01:05.48 | _Sam-- | "extensions reload" |
| 01:05.56 | _Sam-- | type that on the console after you add it |
| 01:05.58 | justinu | hackeron: yeah, but they already do it |
| 01:06.04 | justinu | with things like "ultrapeers" |
| 01:06.12 | justinu | ultrapeers need public IPs tho :P |
| 01:06.13 | _Sam-- | does your SIP extension 200 work right? |
| 01:06.24 | hackeron | justinu: ah, thats what they do. I thought they actually route the data through them. |
| 01:06.34 | justinu | nope, that's a big waste |
| 01:06.46 | justinu | that's primitive (the way asterisk does it with nat=yes) |
| 01:07.04 | mjmtaiwan | reloaded both... |
| 01:07.19 | _Sam-- | are you sure that extension 200 is registered on your asterisk box |
| 01:07.23 | mjmtaiwan | still busy :< |
| 01:08.05 | hackeron | justinu: wow, thanks for that, that really kicks ass then! |
| 01:08.11 | justinu | np |
| 01:08.24 | justinu | keep in mind, there are certain kinds of NAT that won't work with |
| 01:08.34 | justinu | but most cheapies aren't that kind |
| 01:08.44 | mjmtaiwan | .... YES... thanks Sam... when I removed the [from-broadvoice] that I had and used yours it worked. |
| 01:08.51 | mjmtaiwan | Thanks!! First incoming in days. |
| 01:09.02 | _Sam-- | enjoy |
| 01:09.05 | justinu | hakceron: i highly recommend you read this http://www.faqs.org/rfcs/rfc3489.html |
| 01:09.17 | justinu | it's quite informative |
| 01:09.17 | hackeron | justinu: well, even the expensive stuff will work unless its configured to block specific signatures, no? |
| 01:09.36 | justinu | they describe the different kinds of NATs |
| 01:09.40 | _Sam-- | if you want it ring on a different phone than extension 200 then change it |
| 01:09.41 | justinu | why they behave the way they do |
| 01:09.42 | _Sam-- | to SIP/whatever |
| 01:09.59 | _Sam-- | of make it ring a few at once... (SIP/200&SIP/202) |
| 01:10.15 | _Sam-- | then type 'extensions reload' again |
| 01:10.17 | mjmtaiwan | now for outbound and the rest of the dial plan... I'll be back! (time to take out my wife for night at the library - Taiwanes Girls... what can you do?) |
| 01:10.18 | hackeron | justinu: thanks, reading now. |
| 01:10.23 | justinu | like this: |
| 01:10.46 | justinu | <PROTECTED> |
| 01:10.46 | justinu | <PROTECTED> |
| 01:10.46 | justinu | <PROTECTED> |
| 01:10.46 | justinu | <PROTECTED> |
| 01:10.46 | justinu | <PROTECTED> |
| 01:10.47 | justinu | <PROTECTED> |
| 01:10.57 | justinu | that probably got truncated |
| 01:11.07 | justinu | i'm not using an irc client, so I can't tell |
| 01:11.13 | mjmtaiwan | THanks Sam! I'll see you soon - justinu... thanks! |
| 01:11.15 | Mother | greetings |
| 01:11.22 | justinu | taiwan: enjoy |
| 01:11.26 | _Sam-- | sure thing ...good nite |
| 01:11.33 | brc_ | can you use both AEL & extensions.conf? |
| 01:11.45 | hackeron | justinu: wow, that sucks |
| 01:11.59 | Mother | there must be some howto that explains why I keep getting this "you do not seem to have the sources for xyx kernel installed" when I make zaptel |
| 01:12.08 | hackeron | justinu: who would use such a thing :( |
| 01:12.10 | saftsack | someone here who sends faxes with txfax? |
| 01:12.20 | Mother | even though I *do* have the sources, and they are linked to /usr/src/linux |
| 01:12.21 | justinu | the port restricted ones are OK |
| 01:12.28 | justinu | it's the symmetric NATs that bite you in the ass |
| 01:13.46 | hackeron | justinu: are there any advantages to port restricted cones and symetric nats? |
| 01:14.26 | justinu | hackeron: i'm not sure, you'd probably need to talk to a network security expert about that |
| 01:14.47 | justinu | i bet that certain types of NAT are less complex to code |
| 01:15.01 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 01:15.12 | hackeron | justinu: err, cd /usr/src/linux; make menuconfig -- what could be so hard? :) |
| 01:15.21 | justinu | heh |
| 01:15.36 | justinu | that's if you don't mind using GPL code |
| 01:15.55 | *** join/#asterisk Defraz_ (n=t0tal@72.24.26.215) |
| 01:15.57 | hackeron | err, cd /usr/src/freebsd; make config? :) |
| 01:16.03 | justinu | i guess some people want to own their stuff |
| 01:16.08 | justinu | who the hell knows why people write things like that |
| 01:16.27 | justinu | who do you think wrote the cisco pix NAT code? |
| 01:16.34 | justinu | (i have no idea, that's why I'm asking) |
| 01:16.50 | justinu | and why did that person/team write it? |
| 01:16.52 | hackeron | retards on a very tight schedule that had to make look like they were doing something not to get fired? |
| 01:17.04 | *** join/#asterisk dorphalsig (n=dorphals@200.106.223.5) |
| 01:17.10 | dorphalsig | Hi |
| 01:17.34 | dorphalsig | I have a question... I have a T400P card |
| 01:17.54 | dorphalsig | and I have a E1 - PRI connected to span 1 |
| 01:18.15 | dorphalsig | but I also have two channelbanks connected to slots 3 and 4 |
| 01:18.38 | dorphalsig | *spans |
| 01:18.48 | _Sam-- | sounds nice |
| 01:18.55 | justinu | werent you in here earlier as someone else? |
| 01:19.09 | dorphalsig | yeah, as chiardon (my bosses nick =) ) |
| 01:19.11 | dorphalsig | now I need to know if I need to have two clocking |
| 01:19.13 | justinu | i thought so |
| 01:19.16 | dorphalsig | sources |
| 01:19.19 | justinu | no |
| 01:19.21 | dorphalsig | or just 1 |
| 01:19.27 | _Sam-- | justinu never forgets |
| 01:19.32 | justinu | just one timing source, telco PRI |
| 01:19.58 | dorphalsig | justinu --> You should probably remember me being here yesterday and the day before :P |
| 01:20.25 | justinu | yes |
| 01:20.58 | xachen | ddd |
| 01:21.08 | dorphalsig | heh |
| 01:22.45 | justinu | dorphalsig: all other ports on your card should use the timing from the source port |
| 01:23.11 | justinu | and make sure that your channel banks are set up to sync from the signal coming off the asterisk E1 card |
| 01:23.13 | *** join/#asterisk _DAW-LAPTOP (n=DAW-LAPT@71-81-63-86.dhcp.slid.la.charter.com) |
| 01:24.47 | *** join/#asterisk timeh (n=vye@ip68-7-245-143.sd.sd.cox.net) |
| 01:25.58 | dorphalsig | thats my problem I have NO idea on how to make the channelbanks sync from that signal |
| 01:26.23 | justinu | well, they have an option to either sync from an internal oscillator |
| 01:26.29 | justinu | or sync from one of their E1 ports |
| 01:27.05 | dorphalsig | now that I come and think of it ... that could not have been the cause of my lost D-chan errors right? |
| 01:27.13 | justinu | not really |
| 01:27.25 | justinu | not unless the asterisk box was trying to derive timing from one of the channel banks |
| 01:27.48 | riddlebox | tzafrir_laptop, you still around |
| 01:27.58 | tzafrir_laptop | yes |
| 01:28.04 | justinu | dorphalsig: what country you from? |
| 01:28.11 | dorphalsig | Colombia, South America |
| 01:28.14 | timeh | I'm probably going to have about 10 ip phones, I need to be able to use intercom so I think I need to get a pci card, I don't know much about them so does anyone have anything to say about this one: Digium Wildcard E100P? |
| 01:28.19 | justinu | cool, what city? |
| 01:28.32 | dorphalsig | Bogota |
| 01:28.38 | dorphalsig | have you been here? |
| 01:28.49 | sonic2wb | da |
| 01:28.52 | justinu | no, of course not |
| 01:29.03 | riddlebox | tzafrir_laptop, I got it working now but I get this error now: cdr_custom.c:127 custom_log: Unable to re-open master file /var/log/asterisk/cdr-custom/Master.csv : Permission denied |
| 01:29.10 | justinu | we americans don't go to places like that because they tell us we'll get kidnapped |
| 01:29.26 | *** join/#asterisk zotz (n=zotz@24.231.47.175) |
| 01:29.35 | tzafrir_laptop | and the permissions there? |
| 01:29.37 | *** join/#asterisk Dr-Linux (n=Nothing@202.125.141.6) |
| 01:30.09 | dorphalsig | lol |
| 01:30.14 | dorphalsig | its not that bad |
| 01:30.30 | tzafrir_laptop | ls -l /var/log/asterisk/cdr-custom/Master.csv |
| 01:30.41 | dorphalsig | some time ago it was reaaaally bad |
| 01:30.50 | Dr-Linux | question, i want if caller is waiting in queue. he should have option to stay in queue. or he wanna left the queue by pressing any key? is it possible |
| 01:30.51 | dorphalsig | but lately things have improved dramatically |
| 01:30.53 | *** join/#asterisk tessier (n=treed@rrcs-67-53-110-66.west.biz.rr.com) |
| 01:31.02 | riddlebox | tzafrir_laptop, -rw-r--r-- |
| 01:31.08 | sonic2wb | dorphalsig: WHATS UP |
| 01:31.21 | justinu | dorphalsig: what do you attribute that to? |
| 01:31.29 | dorphalsig | sonic2wb --> WAZZUP! MAN |
| 01:31.32 | tzafrir_laptop | riddlebox, and the owner is asterisk? |
| 01:31.42 | riddlebox | tzafrir_laptop, it is root |
| 01:31.59 | tzafrir_laptop | chown -R asterisk /var/log/asterisk |
| 01:32.05 | dorphalsig | justinu --> I guess the armed forces are doing their job quite well |
| 01:32.20 | justinu | glad to hear that |
| 01:32.26 | riddlebox | tzafrir_laptop, Thanks for the help |
| 01:32.27 | dorphalsig | they have been training hard to fight back and prevent thise things happening |
| 01:33.19 | OloBola | can anyone see any reason why my bash script runs fine from CL but not from a cron job? http://pastebin.ca/36065 |
| 01:34.38 | Dr-Linux | any clue about my question? |
| 01:35.33 | shido6 | Dr-Linux: yes you can do that |
| 01:36.25 | justinu | dr-linux: what's up? |
| 01:36.50 | Dr-Linux | shido6: thanks, should i need to define that in the queues.conf or it needs to be done in extensions.conf .. what WIKI says "context = queue-out ; Here we go when the caller presses a single digit, while in the queue " |
| 01:37.00 | Dr-Linux | but they didn't mentions where to define this KEY |
| 01:37.11 | Dr-Linux | justinu: hi friend whatssup? |
| 01:37.29 | *** join/#asterisk DrukenHME (i=Druken@CPE00121716da99-CM000e5cde4ca2.cpe.net.cable.rogers.com) |
| 01:40.02 | DrukenHME | evening everyone |
| 01:41.10 | justinu | druken: tu es quebeqois? |
| 01:41.14 | *** part/#asterisk kenrstone (n=krstone@ool-4573f3dc.dyn.optonline.net) |
| 01:41.47 | DrukenHME | je ne pais francais |
| 01:41.52 | justinu | ah |
| 01:42.01 | DrukenHME | parque anglias? :) |
| 01:42.05 | justinu | haha yah |
| 01:42.16 | justinu | i dunno why, but i thought you were one of the frenchies |
| 01:42.17 | justinu | :) |
| 01:42.32 | DrukenHME | last name is french, that's where it stops |
| 01:43.32 | DrukenHME | oh, and i eat alot of french fries.... hehe |
| 01:45.11 | DrukenHME | ok, is anyone working on the source of asterisk anymore? |
| 01:47.48 | rob0 | There were some recent CVS commits, but I think those were the last ones ever. ;) |
| 01:48.01 | *** join/#asterisk harryvv (n=none@S010600a0c93f6f7e.vs.shawcable.net) |
| 01:48.28 | Soul | greetz |
| 01:48.36 | Soul | quick disrty question: |
| 01:49.00 | DrukenHME | just thinking about the multipul registeration thing... how hard would that be to impliment.... |
| 01:49.06 | DrukenHME | can't be that hard can it? |
| 01:49.10 | Soul | i've managed to do unattended transfer on our asterisk server: just pressed # and then the extension number |
| 01:49.34 | Soul | the question is: can i transfer to a sip url or just to an extension number ? |
| 01:49.43 | _Sam-- | using the # sign just an extension |
| 01:50.05 | _Sam-- | i could be wrong |
| 01:50.07 | Soul | _Sam--, is there any other way than # ? |
| 01:50.09 | *** join/#asterisk kenrstone (n=krstone@ool-4573f3dc.dyn.optonline.net) |
| 01:50.16 | pfn | considering everything in asterisk is extension based, how could you ever transfer to a SIP URL? |
| 01:50.22 | DrukenHME | would would you enter in a sip url with only the dialpad? |
| 01:50.35 | harryvv | I came across a voip vendors client today stating that there older digital phones work with there new voip gateway made in china. I asked if there older pbx was still installed and thay said no. So how many voip gateways will integrate with older legacy digital phones and are not propriatory equipment? |
| 01:51.03 | Soul | good question, but we use recent phones, which have alpha keys. with sw phones would work too |
| 01:51.16 | harryvv | alpha keys? |
| 01:51.24 | Soul | hackeron, abc |
| 01:51.35 | _Sam-- | maybe it will work for external calls using # |
| 01:51.42 | DrukenHME | must be one big fucken phone |
| 01:52.10 | Soul | Druken, no, the usual number+alpha keys, just like a cisco 79xx |
| 01:52.30 | pfn | anyway, no, you can't transfer to a SIP url |
| 01:52.38 | pfn | the process is just much too clunky to do |
| 01:52.44 | Dr-Linux | hhm.. |
| 01:52.52 | Dr-Linux | justinu: do you know something about my question? |
| 01:53.09 | Soul | ok, thanks. but can i transfer to any telephone number, even that's not an extension ? an outside number ? |
| 01:53.13 | *** join/#asterisk welles (n=welles@222.90.198.102) |
| 01:53.23 | pfn | soul you can only transfer to what is defined in your dialplan |
| 01:53.28 | *** join/#asterisk wellng (n=welles@222.90.198.102) |
| 01:53.33 | pfn | if your diaplan allows external dialing, then you can transfer to external numbers |
| 01:54.20 | *** part/#asterisk wellng (n=welles@222.90.198.102) |
| 01:54.31 | Soul | pfn, thanks, thats what i wanted to know. we're using mcc for the billing stuff, and transferring to external numbers does not work. i was pretty convinced that it was an mcc problem |
| 01:54.36 | welles | hi al |
| 01:55.09 | Soul | what about attended transfer ? can i have that ? |
| 01:55.28 | Soul | it works fine with our cisco 79xx, but thats cisco magic, not asterisk's |
| 01:55.38 | Soul | does not work with the other phones |
| 01:57.46 | *** join/#asterisk Zodiacal (i=hehehe@bdsl.66.14.242.199.gte.net) |
| 01:58.11 | Zodiacal | anyone know why the voice attendent is kinda crackly? |
| 01:58.20 | hackeron | Soul: ?? |
| 01:58.22 | Zodiacal | it seems like it pauses for a split second then continues , type of noise |
| 01:58.30 | Zodiacal | is that normal? |
| 01:59.06 | *** join/#asterisk asterboy (n=Snake@S01060204ee2b6007.ed.shawcable.net) |
| 01:59.17 | Soul | hackeron, call transfer with our cisco 79xx 's is attended, ie, you can call with the person you want to transfer to, before passing the call |
| 01:59.46 | *** join/#asterisk linlin (i=linlin@c-67-184-231-233.hsd1.il.comcast.net) |
| 02:00.02 | Soul | but any other phones connected to our asterisk cant do attended transfer, junst unattended (you just press # and the extension number, and the call is immediatly transferred) |
| 02:00.27 | Zodiacal | im using x-lite if that matters |
| 02:00.30 | Soul | i think that's because the cisco call transfer is done by the cisco phone, and not by asterisk |
| 02:00.56 | Soul | i'm searching voip-info now, looking for attended transfer in asterisk |
| 02:01.13 | harryvv | soal, asterisk show do a attended transfer. Ive done it |
| 02:01.16 | wunderkin | Zodiacal, is set transmit silence set to yes |
| 02:01.31 | Zodiacal | wunderkin where is that option? |
| 02:01.32 | wunderkin | not a real attended transfer though |
| 02:01.59 | hardwire | I hate. |
| 02:02.06 | hackeron | Soul: errr, on my cheap ass $80 GXP-2000 phones I press line2, dial the number, tell them hey jim, bob wants to talk to you, go back to line 1 where I have bob, say transferring now, hit transfer and then 2 and presto |
| 02:02.11 | hackeron | Soul: thats with asterisk |
| 02:02.12 | Soul | harryvv, you just press # and the extension number ? |
| 02:02.18 | harryvv | I would press the transfer button push the extention talk to the other party if thay want the call then transfer. |
| 02:02.21 | harryvv | i have a ip500 |
| 02:03.01 | harryvv | something like what hackeron has said but in better detail. |
| 02:03.47 | *** join/#asterisk ManxPower (n=ewieling@24-179-48-91.static.slid.la.charter.com) |
| 02:04.02 | Soul | gonna try it with x-lite, now |
| 02:04.57 | _Sam-- | i dont think the transfer button works in xlite |
| 02:05.01 | Zodiacal | wunderkin any ideas? |
| 02:05.03 | _Sam-- | its a teaser isnt it |
| 02:05.13 | Qwell | _Sam--: yep, I believe so |
| 02:05.33 | *** join/#asterisk Flauto (n=zhao@c-24-13-6-136.hsd1.il.comcast.net) |
| 02:05.54 | Dr-Linux | http://pastebin.com/494373 << here is my exntension and queue.conf, i want the caller should have an option, if he wanna stay in queue, or he can left queue by pressing a key? |
| 02:06.25 | Soul | _Sam--, yes, in fact, but have x-pro, also |
| 02:06.25 | justinu | sam: it is a teaser |
| 02:06.46 | Qwell | Dr-Linux: Why the hell are you putting T in your Queue()? |
| 02:07.02 | _Sam-- | in case they want to transfer to an outside extension! |
| 02:07.11 | Qwell | Dr-Linux: and look at the context= line in queues.conf |
| 02:07.15 | Dr-Linux | Qwell: sorry :( |
| 02:07.32 | Dr-Linux | Qwell: how can i do that i want? |
| 02:07.42 | Qwell | by using context= in queues.conf |
| 02:08.35 | Dr-Linux | Qwell: yeah, i put there context=exntesions [extensions] is a queue, but what key users can press to got there, and where to define that key? |
| 02:08.38 | _Sam-- | drilinux: do you understand why qwell was upset with the T? |
| 02:08.55 | Qwell | Dr-Linux: in [exntesions] |
| 02:09.27 | Dr-Linux | _Sam--: no? |
| 02:09.31 | _Sam-- | qwell...what would the T have allowed..would be hard to transfer still |
| 02:09.58 | Dr-Linux | Qwell: can you give me an example : ididn't understand how to put that key in [extensions] ? |
| 02:10.04 | Zodiacal | wunderkin i tried togging it both ways |
| 02:10.09 | Zodiacal | same thing |
| 02:10.17 | Qwell | Dr-Linux: exten => 1,1,DoSomething() |
| 02:10.20 | _Sam-- | someone in your queue i believe with the T could have hit the # sign and tried to transfer themselves? |
| 02:10.24 | Qwell | Dr-Linux: exten => 2,1,DoSomethingElse() |
| 02:10.32 | Qwell | _Sam--: exactly |
| 02:10.51 | _Sam-- | they could transfer themselves? |
| 02:10.56 | _Sam-- | or they would be trying to tranfer someone else |
| 02:10.56 | Qwell | yes |
| 02:11.01 | Qwell | themselves |
| 02:11.20 | _Sam-- | so if there was an outgoing dial rule in place, they could transfer themselves, to their buddy in say karachi pakistan |
| 02:11.26 | Qwell | yes |
| 02:11.35 | _Sam-- | trying to explain it to drlin |
| 02:11.49 | Dr-Linux | _Sam--: what about pakistan? :S |
| 02:11.52 | _Sam-- | i thought normally the # sign transferred the OTHER party |
| 02:11.56 | Qwell | _Sam--: He's from .pk...use another example :P |
| 02:12.01 | _Sam-- | like if ii am talking to you... |
| 02:12.04 | _Sam-- | and i hit the # sign |
| 02:12.07 | _Sam-- | i am going to transfer YOU |
| 02:12.08 | _Sam-- | not me |
| 02:12.30 | Qwell | _Sam--: With Queue, T will let the caller transfer himself |
| 02:12.35 | Dr-Linux | :S |
| 02:12.36 | _Sam-- | ok just making sure |
| 02:12.58 | Qwell | BUT |
| 02:13.01 | Qwell | even with Dial... |
| 02:13.16 | Qwell | T would let me transfer the call to a 900 number I control, which costs $600/minute |
| 02:13.30 | _Sam-- | drlinux: that was why qwell was upset with the T |
| 02:13.33 | Qwell | So the call only lasts 10 seconds...that's a cool hundred |
| 02:13.41 | _Sam-- | because callers could transfer themselves to bad places |
| 02:13.41 | Dr-Linux | Qwell: [extensions] context has too many things already, the lines you told me, are need to define in the same dialplan for queue? |
| 02:14.05 | harryvv | Qwell, so there is a market for 900 voip minites? |
| 02:14.13 | _Sam-- | qwell you should write an autodialer that looks for bad call queues! |
| 02:14.15 | Qwell | harryvv: I've asked |
| 02:14.17 | _Sam-- | and then have it call home on your 900 |
| 02:14.32 | Qwell | I honestly haven't found a single 900 provider |
| 02:14.38 | harryvv | That would be interesting but to me, I associate a 900 number with a sex line. |
| 02:15.51 | justinu | 900's are out |
| 02:16.07 | justinu | when the lec's started blocking them by default, no one could call them anymore |
| 02:17.55 | wunderkin | so no more calling 1-900-sex-toys? |
| 02:18.41 | *** join/#asterisk _DAW-LAPTOP (n=DAW-LAPT@adsl-156-94-42.msy.bellsouth.net) |
| 02:18.41 | justinu | yeah, now you call 1-800-sex-toys and put in your cc# or they hang up on you |
| 02:18.49 | ManxPower | ~docs |
| 02:18.53 | jbot | [docs] probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 02:18.54 | ManxPower | ~mailinglist |
| 02:18.55 | jbot | well, mailinglist is Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search. Browse the mailing list archive at http://lists.digium.com/ or search through it at http://www.asteriskguru.com/archives/search.php, or and there is also the Macintosh Asterisk mailing list at http://www.astmasters.net/maml.html |
| 02:19.22 | Dr-Linux | Qwell: i changed the context=ivr in queues.conf [ivr] context has all press 1, 2, 3 and so .... but it doesn't go there? what i'm missing? |
| 02:19.30 | Dr-Linux | even i remoeve that T option :S |
| 02:20.17 | *** join/#asterisk hardwire (n=hardwire@209-112-136-72-cdsl-rb1.nwc.acsalaska.net) |
| 02:20.44 | timeh | ManxPower: I want an intercom feature on my voip network so I believe I need a PCI card, which one do you sugguest for say... 10 phones? or where could I go to get some more information on which card would work best for me? |
| 02:21.14 | ManxPower | Define "intercom" |
| 02:21.21 | _Sam-- | im not sure you need a timing card for that |
| 02:21.33 | ManxPower | Would that be "overhead paging", "all phone page", or "station to station page"? |
| 02:22.12 | hardwire | atm I am doing all |
| 02:22.13 | hardwire | :) |
| 02:22.15 | timeh | ManxPower: all phone page+station to station? |
| 02:22.32 | ManxPower | timeh, What make/model of phone? |
| 02:22.45 | Qwell | station to station page == a phone call |
| 02:22.45 | Qwell | heh |
| 02:22.57 | *** join/#asterisk xheliox (n=jeff@user-0c6s3h2.cable.mindspring.com) |
| 02:23.06 | timeh | ManxPower: Polycom 501 |
| 02:23.09 | ManxPower | Station to Station Page == phone call where destination auto answers on it's speakerphone. |
| 02:23.42 | Qwell | ManxPowe: sure, if you want to get technical |
| 02:23.43 | ManxPower | timeh, Pretty much all you need to know is on the Wiki (see auto answer) or the mailing list archives. Also see "Show application page" in 1.2+ |
| 02:23.43 | Qwell | :p |
| 02:23.44 | timeh | ManxPower: I don't know what overhead paging is.. I thought that was all phone page |
| 02:23.56 | *** join/#asterisk calennert (n=calenner@66-191-55-096.dhcp.gnvl.sc.charter.com) |
| 02:24.06 | Qwell | timeh: overhead paging is a speaker in the ceiling |
| 02:24.13 | hardwire | timeh: overhead == its in the ceiling :) |
| 02:24.19 | timeh | oh... yeah don't need that |
| 02:24.25 | hardwire | its over your head.. don't worry about it. |
| 02:24.32 | ManxPower | Overhead Paging, like used in a warehouse. dial an extension and blast the dulcent tones of your voice to all the poor sods that have to listen to it. |
| 02:24.34 | timeh | har har |
| 02:24.39 | hardwire | timeh: :) |
| 02:24.44 | hardwire | who loves you baby. |
| 02:24.45 | hardwire | I do. |
| 02:24.48 | timeh | heh |
| 02:25.16 | justinu | tim-EH! |
| 02:25.22 | hardwire | heh |
| 02:25.26 | hardwire | timeh: what phones? |
| 02:25.28 | *** join/#asterisk Flauto (n=zhao@c-24-13-6-136.hsd1.il.comcast.net) |
| 02:25.35 | timeh | lol... it's for you southpark watchers. |
| 02:25.41 | Dr-Linux | Qwell: even i added nT options as well, but sill no luck :S |
| 02:25.41 | timeh | hardwire: polycom 501 |
| 02:25.45 | hardwire | timeh: gotcha |
| 02:25.57 | ManxPower | I didn't know that T/t worked in queues. |
| 02:26.26 | Dr-Linux | ManxPower: WIKI says it work |
| 02:26.43 | Qwell | ManxPower: indeed. T is different though I think |
| 02:26.46 | Dr-Linux | ManxPower: did you see this problem? http://pastebin.com/494373 |
| 02:26.59 | Qwell | (only slightly) (and only to cost you more money if enabled) |
| 02:27.11 | justinu | cripple fight! |
| 02:27.27 | wunderkin | timmay timmay timmay |
| 02:27.49 | ManxPower | Dr-Linux, Well if the Wiki says it, it MUST be true. What does the mailing list archive say? |
| 02:28.08 | timeh | jimmay |
| 02:28.24 | De_Mon | Qwell it says to read the wiki! |
| 02:28.32 | justinu | lmao |
| 02:28.45 | Qwell | for Queue options? |
| 02:28.46 | Dr-Linux | ManxPower: i don't know about that, ManxPower, all i want my caller should have an option if they stay in the queue or they should have an option to leave queue by pressing some key :S |
| 02:28.52 | Qwell | how about 'show application Queue'? |
| 02:28.53 | timeh | ManxPower: when you said "(see auto answer)" did you mean search voip-info.org/wiki/ for that? |
| 02:29.01 | Qwell | Dr-Linux: context= |
| 02:29.07 | ManxPower | Dr-Linux, "show application queue" |
| 02:29.07 | Qwell | read the part of the config file where it talks about it |
| 02:29.29 | Qwell | timeh: yes |
| 02:31.45 | timeh | k so this means I don't need a pci card, I just need to configure asterisk. |
| 02:31.49 | Dr-Linux | Qwell: i tried that as well :S i don't know what i'm missing :S |
| 02:32.11 | *** join/#asterisk _Soul_ (n=Soul@87-196-15-155.net.novis.pt) |
| 02:32.19 | Qwell | a clue |
| 02:33.14 | Assid | anyone here using voipjet? |
| 02:33.40 | Qwell | ManxPower: You know what amazes and confounds me? |
| 02:34.01 | Qwell | ManxPower: people actually PAY some of these people, to do this... |
| 02:34.17 | ManxPower | Qwell, people are 1) stupid 2) lazy |
| 02:34.27 | Qwell | 3) well paid |
| 02:34.46 | _DAW-LAPTOP | can anyone here tell me if native moh is more reliable than mpg123? |
| 02:35.03 | Qwell | _DAW-LAPTOP: it's native. mpg123 wasn't even that reliable |
| 02:35.06 | *** join/#asterisk DannyF (n=dannyf@c-25bbe455.24-0099-74657210.cust.bredbandsbolaget.se) |
| 02:35.30 | _DAW-LAPTOP | mpg123 has cause me several problems. I am hoping the native works out better. |
| 02:35.35 | Qwell | try it |
| 02:35.49 | Dr-Linux | Qwell: i checked that as well, but there :S nothing like that i want :S |
| 02:38.04 | *** join/#asterisk Lurr (n=pr0ph3t@65.162.120.25) |
| 02:38.04 | Dr-Linux | justinu: today i'm goinggg homeeeeeeeee, ;) montaionssss hills, family, grrrrrrrrrrr ;) |
| 02:38.37 | *** part/#asterisk Lurr (n=pr0ph3t@65.162.120.25) |
| 02:39.24 | kuku5 | whats the cheapest option for cisco 79XX phones? |
| 02:40.37 | Qwell | kuku5: steal them |
| 02:40.45 | ManxPower | kuku5, It's called "polycom" |
| 02:40.50 | *** part/#asterisk Cybertoy (n=maxim@dsl254-123-241.nyc1.dsl.speakeasy.net) |
| 02:41.44 | Dr-Linux | justinu: where are youuuuuu? |
| 02:42.57 | *** join/#asterisk Seyr (n=kaladin@cpe-67-10-139-141.houston.res.rr.com) |
| 02:43.36 | *** join/#asterisk MasterObiWanK (n=master@201.199.76.194) |
| 02:44.43 | MasterObiWanK | Qwell , hello |
| 02:45.03 | Qwell | Hi |
| 02:46.33 | wunderkin | master obi wank? |
| 02:47.50 | Dr-Linux | wunderkin: |
| 02:50.51 | *** join/#asterisk SwK (n=SwK@12-219-151-128.client.mchsi.com) |
| 02:52.43 | *** join/#asterisk J4k3 (n=j4k3@wls-a044.intrastar.net) |
| 02:52.49 | *** join/#asterisk ElMo2 (n=chatzill@202.63.43.50) |
| 02:52.50 | objRobMitch | need advice |
| 02:54.11 | wunderkin | don't eat yellow snow |
| 02:54.12 | objRobMitch | setup service with telasip, now testing with softphone X-lite after config of outbound |
| 02:54.50 | objRobMitch | what about red, brown or green snow? |
| 02:55.40 | objRobMitch | no voice/audio on my end after calling home PSTN line |
| 02:55.42 | ElMo2 | Question with: PHPAGI, how do i dial a queue line as the extention. (eg. ZAP/3 or SIP/100 works) but how would i get it to connect to an internal queue? |
| 02:56.27 | objRobMitch | they can hear me though... |
| 02:56.33 | Qwell | ElMo2: huh? |
| 02:56.46 | ManxPower | ElMo2, See the AGI EXEC think about AGI EXEC QUEUE |
| 02:57.56 | ElMo2 | thanks for the heads up |
| 02:59.33 | objRobMitch | * and x-lite softphone behind router |
| 02:59.43 | objRobMitch | forwarding ports.... |
| 03:02.34 | objRobMitch | forwarding ports 10000:20000, UDP for ip address of * box (happens to be 192.168.1.101) |
| 03:02.44 | *** join/#asterisk dcoulson (n=dcoulson@wilbur.geekcolony.net) |
| 03:02.57 | objRobMitch | also tried 8000 : 8005 UDP for same ip |
| 03:03.42 | objRobMitch | and 5060 : 5082 UDP, again for same ip |
| 03:04.09 | objRobMitch | tried all these concurrently, by the way |
| 03:04.09 | *** join/#asterisk DruidFyr (i=DruidFyr@99-26.119-70.tampabay.res.rr.com) |
| 03:04.11 | Dr-Linux | Qwell: its working nowwwww ;) |
| 03:04.28 | Dr-Linux | harryvv: helped me and its working the way i want |
| 03:05.17 | *** join/#asterisk ajd5723 (n=kvirc@ppp-70-230-114-21.dsl.chcgil.ameritech.net) |
| 03:05.25 | *** join/#asterisk jef_ (i=fischer@p54844AE3.dip.t-dialin.net) |
| 03:05.37 | *** join/#asterisk wellng (n=welles@222.90.198.102) |
| 03:05.57 | hardwire | I need a good wireless phone for an operator. |
| 03:06.11 | hardwire | above and beyond a uh.. battery powered ethernet bridge and polycom |
| 03:06.18 | hardwire | operator backpack. |
| 03:06.30 | *** join/#asterisk coppice (n=chatzill@166.168.17.210.dyn.pacific.net.hk) |
| 03:06.32 | *** join/#asterisk gnosys (n=gnosys@ip68-9-201-250.ri.ri.cox.net) |
| 03:13.26 | *** join/#asterisk dcoulson_ (n=dcoulson@wilbur.geekcolony.net) |
| 03:18.16 | *** join/#asterisk kenrstone (n=krstone@ool-4573f3dc.dyn.optonline.net) |
| 03:18.39 | *** join/#asterisk Cyon (n=cyon@cyons.net) |
| 03:18.43 | Cyon | asterisk |
| 03:18.45 | Cyon | Doh |
| 03:20.54 | Dr-Linux | Cyon: Doh? what does this fuckin word mean? |
| 03:21.08 | wunderkin | this function stuff is a pain |
| 03:21.50 | wunderkin | is there any way to get more ( and { in there |
| 03:21.53 | Cyon | Dr-Linux: http://www.urbandictionary.com/define.php?term=doh |
| 03:21.55 | Dr-Linux | ooooo ic |
| 03:22.10 | Dr-Linux | Cyon: so you have pain? :S |
| 03:22.55 | objRobMitch | man id like to defunk her.... |
| 03:22.55 | Cyon | Dr-Linux: Umm, yes, exactly :-P |
| 03:23.17 | Dr-Linux | Cyon: where you feel that? in head? ass ? |
| 03:23.32 | Cyon | Dr-Linux: my heart |
| 03:23.37 | Dr-Linux | Opss |
| 03:23.44 | Cyon | Probably fatal... |
| 03:23.52 | justinu | lol |
| 03:23.58 | Dr-Linux | Cyon: its look like ... love problem :S or wife problem, girl problem :S |
| 03:24.24 | Dr-Linux | but i'm just various US guys should not have this problem :S |
| 03:24.30 | Cyon | Dr-Linux: Oh how easily you deduced it. |
| 03:24.46 | *** join/#asterisk welles (n=welles@222.90.198.102) |
| 03:25.04 | Dr-Linux | justinu: deducced? :S what he said? |
| 03:25.12 | justinu | figured it out |
| 03:25.25 | Cyon | justinu: :-P |
| 03:25.41 | justinu | dr-linux is english challenged |
| 03:25.41 | Dr-Linux | justinu: did he abuse me? :@ |
| 03:25.55 | justinu | no |
| 03:26.08 | Dr-Linux | Cyon: wtf did you just said? :S |
| 03:26.14 | Cyon | lol |
| 03:26.21 | justinu | deduced = figured it out |
| 03:26.34 | Cyon | == ;-) |
| 03:26.36 | Dr-Linux | oo ic |
| 03:26.44 | justinu | cyon = dork; |
| 03:26.47 | justinu | that's an assignment, yo |
| 03:26.54 | Dr-Linux | Cyon: sorry i can't do that with you, bcoz i don't need AIDS :S |
| 03:27.11 | Cyon | Dr-Linux: I feel so alone in this world now. ;-) |
| 03:27.12 | justinu | ouch |
| 03:27.20 | Dr-Linux | awww |
| 03:27.26 | Dr-Linux | Cyon: are you serious? :S |
| 03:27.32 | Cyon | Dr-Linux: No. ;-) |
| 03:27.37 | Dr-Linux | if yes, that i'm sorry |
| 03:27.40 | objRobMitch | the process of deductive reasoning is heavily used in the scientific process and the formulation of a hypothesis |
| 03:27.41 | justinu | lol |
| 03:27.42 | Dr-Linux | oooooooo ok ok |
| 03:27.58 | Dr-Linux | then go to Bush ,, he is a Gay , he will help you |
| 03:28.07 | harryvv | hehehe |
| 03:28.38 | Cyon | :) |
| 03:28.46 | Dr-Linux | he also helped Tony blayer |
| 03:29.18 | justinu | i bet him and tony used to do coke together in the 70s |
| 03:29.38 | Qwell | used to? |
| 03:29.41 | Cyon | Seriously |
| 03:29.42 | Dr-Linux | coke |
| 03:29.43 | justinu | lol |
| 03:29.45 | Cyon | And still do |
| 03:29.57 | harryvv | coke=cocain |
| 03:30.00 | Cyon | Dr-Linux: a drug |
| 03:30.02 | Qwell | e |
| 03:30.13 | justinu | an expensive drug |
| 03:30.26 | Cyon | But you get what you pay for ;-) |
| 03:30.32 | Dr-Linux | Justin: i'm goingggggggg home today, i'll come back after 10 days |
| 03:30.33 | justinu | lol, wouldn't know :P |
| 03:30.38 | justinu | dr-linux: awesome!! |
| 03:30.42 | Dr-Linux | we don't have internet in tribals :( |
| 03:30.44 | justinu | have a great time |
| 03:30.51 | justinu | say hi to your mom for me! |
| 03:30.51 | objRobMitch | ..can we not talk about, gaze, any militia, someone's BUSH, or POLITICS |
| 03:31.08 | Dr-Linux | justinu: okey i'll :) |
| 03:31.08 | Cyon | objRobMitch: drugs are ok though? |
| 03:31.13 | justinu | heh |
| 03:31.16 | Sedorox | lool |
| 03:31.28 | justinu | dr-linux: stay out of trouble! |
| 03:31.36 | Dr-Linux | objRobMitch? who is monica? |
| 03:31.48 | harryvv | lewinski? |
| 03:31.53 | objRobMitch | drugs are ok if thats what floatz ur boat |
| 03:32.06 | justinu | ~asterboy |
| 03:32.10 | jbot | asterboy is, like, a weed smoker |
| 03:32.10 | Dr-Linux | Drugs are very bad thing.. it makes you TUN |
| 03:32.21 | harryvv | im outa here |
| 03:32.23 | justinu | bye |
| 03:32.26 | Cyon | objRobMitch: No, but was wondering if it was allowed. lol |
| 03:32.39 | *** part/#asterisk harryvv (n=none@S010600a0c93f6f7e.vs.shawcable.net) |
| 03:33.15 | Dr-Linux | justinu: my queue problem is sovled, i was missing very small thing |
| 03:33.34 | *** join/#asterisk Cybertoy (n=maxim@dsl254-123-241.nyc1.dsl.speakeasy.net) |
| 03:33.48 | Dr-Linux | brb |
| 03:34.41 | justinu | dr-linux is going to vist his parents in the northwest frontier province of pakistan |
| 03:35.09 | justinu | where al qaeda lives :P |
| 03:35.32 | *** join/#asterisk ms345 (n=mike_sim@dsl027-163-193.atl1.dsl.speakeasy.net) |
| 03:38.35 | wunderkin | it would be cool to chop off a certain number of characters at the end of a string without using len |
| 03:40.02 | *** join/#asterisk malakkara (n=none@pcp0011525589pcs.midltn01.nj.comcast.net) |
| 03:40.13 | wunderkin | NoOp(${LEN(${CHANNEL})}) whats the magic to subtract 2 from that? i guess you need brackets in there somehow |
| 03:40.29 | Qwell | ~striplastdigit |
| 03:40.30 | jbot | striplastdigit is probably ${EXTEN:0:$[${LEN(${EXTEN})} - 1]} , will remove the last digit from EXTEN, making 5551212 become 555121. Change the "1" to remove more digits. |
| 03:40.34 | *** join/#asterisk joelsolanki (i=joelsola@202.160.161.93) |
| 03:40.35 | Qwell | :D |
| 03:40.59 | wunderkin | ah thx this is a mess.. and its late.. heh.. |
| 03:45.41 | OloBola | is it possible to specify a delimter for bash arrays |
| 03:45.55 | OloBola | instead of space |
| 03:46.04 | OloBola | 1.2.3.4.5 periods |
| 03:47.06 | OloBola | I'd like to split a string into an array |
| 03:47.34 | OloBola | explode and grab shrapnel |
| 03:47.48 | OloBola | earn a purple heart |
| 03:47.50 | wunderkin | havent messed with arrays yet |
| 03:47.50 | trixter | you mean the IFS variable? |
| 03:47.57 | OloBola | I guess |
| 03:48.04 | OloBola | is that what that is? |
| 03:48.10 | trixter | I dont think he means arrays I think he means the field seperator that bash uses by default |
| 03:48.15 | trixter | which is normally the IFS variable |
| 03:48.19 | trixter | as in IFS=. |
| 03:48.26 | OloBola | thank you |
| 03:48.48 | trixter | however if you just want to split a string wouldnt cut or something be more effective? |
| 03:49.05 | trixter | such as echo 1.2.3.4.5 | cut -d. -f1 |
| 03:49.17 | trixter | or perhaps uae staples from days gone by.. sed and awk |
| 03:49.41 | trixter | echo 1.2.3.4.5 | sed -e 's/\\./ /g' | awk '{printf $1}' |
| 03:49.59 | trixter | or maybe something else, but IFS can also work depending on how you do it |
| 03:50.22 | OloBola | cut looks good to me, thank you |
| 03:51.43 | *** join/#asterisk bmg505_ (n=leon@dsl-146-44-72.telkomadsl.co.za) |
| 03:53.27 | trixter | wow ael2 |
| 03:53.33 | trixter | I might have to plop that on a test machine |
| 03:58.56 | ManxPower | Why not just use res_js or res_perl? |
| 03:59.22 | trixter | maybe I dont want to use that, the better syntax checking that ael2 advertises can make it better than ael1 |
| 03:59.48 | ManxPower | I am just not confident AEL is stable enough for real world use. |
| 04:00.32 | Luke-Jr | trixter: cut spawns a new process, and would be less efficient |
| 04:00.42 | Luke-Jr | trig: IFS=. read a b c d e |
| 04:00.52 | *** join/#asterisk Cybertoy (n=maxim@dsl254-123-241.nyc1.dsl.speakeasy.net) |
| 04:01.27 | *** join/#asterisk Shakh (i=Shakhruz@83.221.168.212) |
| 04:01.38 | ElMo2 | ManxPower: I have done some research on what you told me, but I am still coming up short? And I can’t find correct documentation on connecting external numbers to queues only zap or sip. can you provide me with any more assistance? |
| 04:02.51 | ManxPower | ElMo2, No, since I don't use PGPAGI |
| 04:02.58 | trixter | Luke-Jr: yes, however you have to set it back when done and if you do it in a function it spawns a new process anyway |
| 04:03.45 | Luke-Jr | trixter: no need to set it back the way I put it |
| 04:03.53 | ElMo2 | ManxPower: Can you suggest any other method of connecting it, through a Perl script or any other means? |
| 04:03.55 | Luke-Jr | it's only IFS=. for the read |
| 04:04.05 | *** join/#asterisk TheCops (i=nobody@got.securebinary.com) |
| 04:04.09 | TheCops | Hi |
| 04:04.29 | *** join/#asterisk asterboy (n=Snake@S01060204ee2b6007.ed.shawcable.net) |
| 04:04.42 | TheCops | How you can do a return key on the UserEvent command for dialplan? |
| 04:04.47 | ManxPower | ElMo2, I'm sure I could spend several hours writing up a hacked togather tutorial for you, but then I would have to demand large piles of cash and things could get ugly. Perhaps you could ask on the mailing lists? |
| 04:05.42 | welles | hi |
| 04:06.05 | welles | how to define a userevent?TheCops |
| 04:06.47 | TheCops | welles, UserEvent(name|body) |
| 04:06.48 | trixter | Luke-Jr: as you did it however its harder to take an arbitrary variable that is already set to read that |
| 04:07.06 | TheCops | I want a return key |
| 04:07.26 | trixter | to make it more useful you will have to set IFS to . (as I suggested earlier) and then reset it when done.. becuase you may not have stdin as your input source |
| 04:07.26 | Luke-Jr | trixter: IFS=. read a b c d e <<<"$arbitraryvariable" |
| 04:07.32 | TheCops | but all caracthere like /n or stuff like that not working |
| 04:07.39 | trixter | yeah now make that useful |
| 04:07.45 | trixter | more than reading a single value from stdin |
| 04:08.13 | trixter | I mean an arbitrary variable could contain the data to be parsed .. ya know when I was talking about other than from stdin |
| 04:08.22 | ElMo2 | ManxPower: thank you for you suggestions, I appreciate your time. |
| 04:08.25 | Luke-Jr | IFS=. read a b c d e <<<"$arbitraryvariable" |
| 04:08.28 | Luke-Jr | that doesn't use stdin |
| 04:08.34 | trixter | what does read work off of? |
| 04:08.44 | *** join/#asterisk brc_ (n=brian@pdpc/supporter/basic/brc) |
| 04:08.51 | trixter | ahh the <<< |
| 04:08.58 | trixter | thought you were doing irc grfx not shell stuff |
| 04:09.06 | welles | hi trixter |
| 04:09.06 | Qwell | 3 <'s? |
| 04:09.17 | Luke-Jr | Qwell: yep |
| 04:09.26 | Qwell | Why 3? |
| 04:09.32 | Luke-Jr | puts the stuff on stdin |
| 04:09.40 | Qwell | one would do that |
| 04:09.43 | Luke-Jr | no |
| 04:09.48 | Luke-Jr | one would connect stdin to a file |
| 04:09.50 | Qwell | okay, 2 |
| 04:09.56 | *** join/#asterisk Shakhruz (i=Shakhruz@83.221.168.212) |
| 04:09.56 | ManxPower | ElMo2, The mailing lists would be your best resouce. |
| 04:10.00 | Luke-Jr | not sure what 2 would do |
| 04:10.03 | trixter | that doesnt work |
| 04:10.15 | trixter | it puts everything into the first var |
| 04:10.20 | trixter | two would be like |
| 04:10.23 | trixter | read blah << EOF |
| 04:10.24 | trixter | moo |
| 04:10.24 | trixter | ojnk |
| 04:10.28 | trixter | EOF |
| 04:10.36 | Luke-Jr | trixter: works for me |
| 04:10.45 | Luke-Jr | luke-jr@tsurukikun ~ $ arbitraryvariable=2.4.3.2.1 |
| 04:10.45 | Luke-Jr | luke-jr@tsurukikun ~ $ IFS=. read a b c d e <<<"$arbitraryvariable" |
| 04:10.45 | Luke-Jr | luke-jr@tsurukikun ~ $ echo $a $b $c $d $e |
| 04:10.45 | Luke-Jr | 2 4 3 2 1 |
| 04:10.56 | welles | hi trixter do u use meetme? |
| 04:10.57 | trixter | put em on new lines |
| 04:11.08 | trixter | or mark the vars |
| 04:11.11 | Luke-Jr | ... |
| 04:11.12 | trixter | rather than all on the same line like that |
| 04:11.21 | asterboy | skype = ebay |
| 04:11.21 | trixter | echo A: $a B: $B ... |
| 04:11.22 | Cyon | using bash? You need to be... |
| 04:11.30 | trixter | welles: yes sometimes |
| 04:11.32 | trixter | depends on what I need |
| 04:11.33 | asterboy | ~skype |
| 04:11.35 | jbot | it has been said that skype is um programa de bate-papo via voz, proprietário e fechado, que usa padrões proprietários e fechados, dos mesmos autores do (spyware) kazaa; procure usar alternativas (pelo menos) com padrões abertos/livres, como os do projeto openh323 <http://www.openh323.org/>, speakfreely <http://www.speakfreely.org/> ... |
| 04:11.45 | trixter | if this is about your codec issue again I dont think you cna do it the way you described |
| 04:11.50 | welles | trixter ,how do u use it? |
| 04:12.05 | trixter | are you asking about forcing a codec to it? |
| 04:12.09 | trixter | or is this a new question? |
| 04:12.30 | welles | can i use meetme to invite my friends to join in |
| 04:12.54 | trixter | anyone who joins the same meetme conference are bridged together |
| 04:13.08 | Luke-Jr | yea, some of my shell code requires BASH 3+ |
| 04:13.20 | trixter | trixter@trix:~$ bash --version |
| 04:13.20 | trixter | GNU bash, version 3.00.16(1)-release (i486-pc-linux-gnu) |
| 04:13.29 | trixter | trixter@trix:~$ ./blah.sh |
| 04:13.29 | trixter | A: 1 2 3 4 B: C: D: |
| 04:13.35 | trixter | it puts it all in the first var not the other ones the way you said |
| 04:14.12 | Luke-Jr | um |
| 04:14.22 | Luke-Jr | note your input should be delimited by the IFS= char |
| 04:14.27 | Luke-Jr | not spaces |
| 04:14.32 | trixter | blah=1.2.3.4 |
| 04:14.34 | trixter | um it is |
| 04:14.40 | trixter | and your code doesnt work as you said it would |
| 04:14.47 | trixter | why you spent all this time to tell me I am wrong I dont know |
| 04:14.49 | trixter | try it |
| 04:14.55 | trixter | especially the mod I suggested of marking your variables |
| 04:15.50 | asterboy | ~ebay |
| 04:15.51 | jbot | ebay is, like, a7r's new friend when it comes to upgrading a7r's network. ;> |
| 04:15.51 | Luke-Jr | luke-jr@tsurukikun ~ $ x=1.2.3.4.5 |
| 04:15.51 | Luke-Jr | luke-jr@tsurukikun ~ $ IFS=. read a b c d e <<<"$x" |
| 04:15.51 | Luke-Jr | luke-jr@tsurukikun ~ $ echo a=$a b=$b c=$c d=$d e=$e |
| 04:15.51 | Luke-Jr | a=1 b=2 c=3 d=4 e=5 |
| 04:16.08 | asterboy | a7r? |
| 04:16.10 | trixter | well it doesnt like bash 3.00 so its more than a 3+ issue |
| 04:16.14 | *** join/#asterisk Evanrude (n=david@ip68-111-117-158.lu.dl.cox.net) |
| 04:16.17 | trixter | what specific version are you running? |
| 04:16.27 | Luke-Jr | GNU bash, version 3.00.16(1)-release (x86_64-pc-linux-gnu) |
| 04:16.33 | asterboy | jbot, ebay is skype. |
| 04:16.34 | jbot | ...but ebay is already something else... |
| 04:16.34 | Luke-Jr | Note your output makes no sense at all |
| 04:16.39 | trixter | ahh same version.. intersting |
| 04:16.39 | Luke-Jr | how are you replacing the . with space? |
| 04:16.44 | asterboy | jobt, ebay is also skype |
| 04:16.57 | asterboy | jbot, ebay is also skype |
| 04:16.58 | jbot | asterboy: okay |
| 04:16.58 | trixter | blah=1.2.3.4 |
| 04:16.58 | trixter | IFS=. read a b c d <<<"$blah" |
| 04:16.58 | trixter | echo A: $a B: $b C: $c D: $d |
| 04:17.05 | asterboy | ~ebay |
| 04:17.06 | jbot | i guess ebay is a7r's new friend when it comes to upgrading a7r's network. ;>, or skype |
| 04:17.24 | Qwell | jbot: ebay is shit |
| 04:17.25 | jbot | Qwell: please, watch your language. |
| 04:17.25 | Qwell | :D |
| 04:17.28 | asterboy | lol |
| 04:17.39 | Luke-Jr | A: 1 B: 2 C: 3 D: 4 |
| 04:17.47 | trixter | ~lart qwell |
| 04:18.11 | asterboy | like on Austin Powers when he sprays the midget |
| 04:18.43 | trixter | ~lart qwell |
| 04:18.49 | trixter | haha |
| 04:18.53 | Qwell | ~kill trixter |
| 04:18.58 | jbot | ACTION shoots a inverse pseudoelectron gun at trixter |
| 04:19.04 | Luke-Jr | trixter: A: 1 B: 2 C: 3 D: 4 |
| 04:19.15 | trixter | yes I saw the first time |
| 04:19.23 | trixter | why it doesnt work on the same version elsewhere is interesting to say hte least |
| 04:19.31 | trixter | but that also implies some compatibility problems |
| 04:19.32 | Luke-Jr | trixter: your build must be buggy |
| 04:19.41 | trixter | especially if you are on more of a stock FBSD machine |
| 04:19.46 | trixter | its hte debian build |
| 04:19.51 | asterboy | ~smell asterboy's finger |
| 04:19.52 | trixter | I doubt that its buggy and no one else noticed |
| 04:19.59 | Luke-Jr | or you typo'd |
| 04:20.06 | trixter | I cut and pasted from the script |
| 04:20.11 | trixter | typos dont magically get corrected on a paste |
| 04:20.21 | Luke-Jr | you never know ;p |
| 04:20.27 | Luke-Jr | X-Chat can be weird sometimes |
| 04:20.41 | Luke-Jr | it won't let me type the |
| 04:20.42 | Luke-Jr | err |
| 04:20.43 | Luke-Jr | t e h |
| 04:20.53 | trixter | the |
| 04:21.03 | trixter | yeah it does that I have stock words which is like 5 or so |
| 04:21.08 | trixter | none are in that script |
| 04:21.15 | trixter | plus what was displayed lookedl ike my script |
| 04:21.21 | *** join/#asterisk joat (n=joat@ip70-160-150-20.hr.hr.cox.net) |
| 04:21.44 | Luke-Jr | I really don't see how your results are possible tho |
| 04:21.50 | Luke-Jr | it would need to replace all '.' with ' ' |
| 04:22.38 | trixter | yes I know what IFS does that is why I was the original person to suggest it |
| 04:22.43 | Cyon | Luke-Jr: http://www.faqs.org/docs/bashman/bashref_33.html |
| 04:23.12 | ManxPower | http://www.gizmodo.com/ See the 2nd item |
| 04:23.26 | Luke-Jr | my results are the correct ones when it is run |
| 04:24.04 | trixter | according to that it would be the expansion part of it |
| 04:24.16 | trixter | that may be however it appears that its not as portable as it should be |
| 04:24.45 | trixter | and that also presumes that you run a system that comes with bash, something I dont like depending on becuase I do have accounts that dont use bash, like freebsd which has a real bourne shell for sh and bash is extra |
| 04:25.09 | Luke-Jr | BASH > Bourne =p |
| 04:25.34 | justinu | bash is cool, but slow as fuck |
| 04:26.09 | trixter | bourne is standard though, course my experience goes back to the 80s with unix and bash wasnt common on commercial installs back then, it still isnt stock on fbsd (well it might be in 5.x I havent checked yet) |
| 04:26.16 | trixter | bash does a lot internally too |
| 04:26.18 | trixter | like pwd |
| 04:26.26 | Cyon | trixter: Nope; still tcsh iirc |
| 04:26.28 | trixter | you can get into trouble with bash by relying on the internals for stuff like that |
| 04:27.07 | Luke-Jr | just depend on BASH :) |
| 04:27.22 | wasim | did you guys read the latest in teh seriers from robert ludlum, "the bourne shell" |
| 04:27.27 | *** join/#asterisk loud (n=ariel@cypher.punk.net) |
| 04:27.44 | trixter | few months ago at the local linux users group that was discovered, someone reported problems with 'sh' acting weird between solaris and linux not knowing that solaris /bin/sh isnt bash. and that when you do pwd in real bourne shell it executes pwd not an internal function. bash was keeping track of the CWD differently and so results were different |
| 04:27.56 | wasim | ok, sorry, too early in the morning ... but this cough medicine does rock |
| 04:28.09 | trixter | wasim: nope, saw the bourne identity though |
| 04:28.13 | *** part/#asterisk joat (n=joat@ip70-160-150-20.hr.hr.cox.net) |
| 04:28.13 | justinu | wasim: dxm? |
| 04:28.14 | trixter | that was a good movie |
| 04:28.35 | trixter | bourne supremacy made up for its lack of action by hiring an epelictic cameraman |
| 04:28.40 | trixter | it wasnt so good |
| 04:28.55 | justinu | i thought war games was better the bourne movies |
| 04:29.02 | justinu | not wargames |
| 04:29.04 | justinu | i mean spy game |
| 04:29.39 | wasim | "bourne shell" is all about the burial of david webb |
| 04:30.18 | trixter | spy game was pretty good |
| 04:30.19 | wasim | oak casket or willow is the major dilemma, and as usual the cia was throwing spanners in the works by insisting on a metal casket |
| 04:30.20 | trixter | so is sneakers |
| 04:30.25 | trixter | ronin is one of my top 5 movies though |
| 04:30.37 | trixter | the saint and the professional also rank pretty high in my book |
| 04:30.41 | justinu | i liked ronin also too |
| 04:30.47 | trixter | the saint with val kilmer not the roger moore tv series |
| 04:30.49 | justinu | trixter: did you see "leon"? |
| 04:30.56 | trixter | no only the shorter version |
| 04:30.58 | justinu | wow |
| 04:31.01 | justinu | you need to see it |
| 04:31.01 | trixter | why I didnt say 'leon the professional' :P |
| 04:31.04 | trixter | oh wait |
| 04:31.21 | trixter | I did see it somewhere.. where the hell did I see it cause I dont own it, netflix doesnt have it, it was never on TV in the US |
| 04:31.21 | justinu | there's one in french |
| 04:31.23 | *** part/#asterisk miketaht (n=mtaht@67-127-179-114.ded.pacbell.net) |
| 04:31.34 | justinu | i thought netflix did have it |
| 04:31.37 | trixter | well this was in english, leon: the professional was the international version it has a lot of deleted scenes |
| 04:31.44 | trixter | maybe they are on the professinoal dvd which I do have |
| 04:31.52 | trixter | hrm ... |
| 04:31.53 | justinu | ohh, sorry, the one i was thinking was la femme nikita |
| 04:31.56 | Cyon | I love Leon... |
| 04:31.58 | justinu | the french movie, not the TV show |
| 04:32.00 | trixter | I do remember those scenes though, the extra 30 minutes or whatever |
| 04:32.09 | *** join/#asterisk Brixius (n=Brixius@c-69-180-132-70.hsd1.mn.comcast.net) |
| 04:32.12 | trixter | ahh no I havent seen the movie |
| 04:32.15 | justinu | if you like leon, watch la femme nikita |
| 04:32.15 | trixter | not in full, saw parts |
| 04:32.43 | trixter | le femme nikita is the basis for um ... bridget fonda ... um |
| 04:32.47 | trixter | irish guy was herboss |
| 04:32.55 | trixter | gah what is that movie |
| 04:33.06 | trixter | and bridget is really cute |
| 04:33.18 | justinu | this is what I'm talking about: http://www.imdb.com/title/tt0100263/ |
| 04:33.22 | justinu | not the bridget fonda one |
| 04:33.30 | trixter | I know you werent why I said its the basis for that movie |
| 04:33.42 | trixter | which I still cant think of the name |
| 04:33.48 | justinu | oh |
| 04:33.58 | trixter | its the same basic story line |
| 04:34.07 | justinu | i must not have seen it |
| 04:34.10 | trixter | there is a 3rd that predates both of those two |
| 04:34.27 | trixter | I forget that name too :/ |
| 04:35.08 | justinu | point of no return? |
| 04:35.11 | justinu | that movie sucked |
| 04:35.26 | trixter | yeah |
| 04:35.29 | trixter | same story line |
| 04:35.38 | justinu | nikita was out first |
| 04:35.40 | justinu | 1990 |
| 04:35.47 | trixter | bridget is really cute in that though so the story line and acting doesnt really matter :P |
| 04:35.55 | justinu | bridget is always cute |
| 04:35.56 | trixter | I know why I said it was the basis for point of no return |
| 04:36.05 | trixter | however there is a 3rd that predates both of those |
| 04:36.10 | justinu | oh |
| 04:36.20 | justinu | i don't speak irc well tonight, so don't mind me |
| 04:36.26 | justinu | no way out is a pretty good spy flick |
| 04:36.31 | trixter | I went through this with someone like a year ago, part of why I am having such a hard time remembering the 3rd becuase it was discovered when talking about those movies |
| 04:37.12 | justinu | and then there was "gotcha" |
| 04:37.18 | trixter | oh yeah |
| 04:37.19 | trixter | 1984 iirc |
| 04:37.22 | trixter | that was a good movie |
| 04:37.25 | justinu | http://www.imdb.com/title/tt0093640/ |
| 04:37.26 | Brixius | hello |
| 04:37.27 | trixter | I was 13 or so when that came out |
| 04:37.41 | justinu | http://www.imdb.com/title/tt0089222/ |
| 04:37.43 | justinu | damn, you're old |
| 04:37.44 | justinu | i was 8 |
| 04:37.49 | trixter | heh |
| 04:37.52 | trixter | I am only 33 :P |
| 04:37.56 | trixter | ahh 85 |
| 04:37.59 | justinu | you're in your thirtys, lol |
| 04:38.01 | trixter | I was close :) |
| 04:38.14 | justinu | anyways, dinner time |
| 04:38.19 | justinu | :P |
| 04:39.28 | *** join/#asterisk pigpen2 (n=mark@fw.seamans.cc) |
| 04:40.25 | pigpen2 | hey, what to ya'll use for asterisk reporting?? Web based preferably. |
| 04:40.27 | *** part/#asterisk stkn (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 04:41.28 | trixter | depends on what you want reported I guess |
| 04:41.38 | *** join/#asterisk int19h (n=tirath@dyn-130-194-137-18.eng.monash.edu.au) |
| 04:41.48 | pigpen2 | call useage, general stats, etc... |
| 04:42.02 | pigpen2 | you know...the stuff "managers" want to know... |
| 04:42.11 | pigpen2 | mainly to fire people.... |
| 04:42.36 | xachen | and get their productivity bonsues ? :P |
| 04:42.45 | `Sauron | Rubber Duckie to pig pen, come in, over |
| 04:42.45 | pigpen2 | yeah...that's it... |
| 04:42.47 | Qwell | I always just make up numbers |
| 04:42.53 | trixter | You can also use the "aelparse" program to check your extensions.ael2 file before feeding it to asterisk. Wouldn't it be nice to eliminate most errors before giving the file to asterisk? |
| 04:42.57 | trixter | what a great addition to ael |
| 04:44.00 | pigpen2 | Well, I guess just dumping the file down to excel will have to do. |
| 04:44.47 | Qwell | trixter: test it out...I bet murf would love to see comments |
| 04:45.39 | int19h | omg 1.2.1 is out?!?! heheh the last time I used asterisk was... 1.0.something I'm pretty sure |
| 04:46.30 | trixter | you do realize they are using the linux kernel version schema right? X.odd is dev X.even is stable |
| 04:46.55 | trixter | so it went 1.0.9 -> 1.2.0 (from what I have heard there was no 'official' 1.0.10 but people have talked about a 1.0.10 so ...) |
| 04:46.58 | int19h | oh rite rite... I forgot about that... thx trixter |
| 04:47.14 | pigpen2 | int19h, so belive him...he is "trix'ing" you... |
| 04:47.32 | trixter | oh man finally |
| 04:47.36 | trixter | ael2 can include other files |
| 04:47.49 | trixter | that was a severe limitation on ael1 in my opinion |
| 04:48.23 | *** join/#asterisk welles (n=welles@222.90.27.131) |
| 05:02.24 | *** join/#asterisk mcquaid (i=mcquaid@toronto-hs-216-138-233-79.s-ip.magma.ca) |
| 05:02.53 | mcquaid | hello, i was trying to get asterisk working with the linux sip client, twinklephone |
| 05:03.10 | trixter | congrats |
| 05:03.14 | mcquaid | it will reg, and send out calls ok, but asterisk can't call it |
| 05:03.22 | justinu | what's up, mcquaid |
| 05:03.30 | mcquaid | hey how's it going, some progress |
| 05:03.39 | trixter | what type is the acct |
| 05:03.41 | trixter | on asterisk |
| 05:03.54 | justinu | mcquaid: what's the problem now? |
| 05:03.58 | mcquaid | well hear me out, i tried xten and it worked fine |
| 05:04.12 | mcquaid | just want to confirm if this is a bug or user error, probably the later |
| 05:04.38 | mcquaid | when i reg with twinklephone and do sip show peer home i see: Reg. Contact : sip:home.10.1.0.12:6000@10.1.0.12 |
| 05:04.45 | *** join/#asterisk TaSo (i=licucude@ool-44c784a0.dyn.optonline.net) |
| 05:04.55 | mcquaid | but with xten it's sip:home@10.1.0.12:6001 which works fine |
| 05:04.57 | justinu | yeah, i remember |
| 05:05.13 | justinu | paste screen shots of twinklephone's configuration |
| 05:05.14 | justinu | :P |
| 05:05.24 | justinu | cuz I don't have a box with KDE or whatever |
| 05:05.37 | mcquaid | i emailed the author of twinkle and he said that he changed the behaviour in the latest version for multiple accts |
| 05:05.52 | mcquaid | i'll quote him it's easier instead of me explaining |
| 05:06.08 | mcquaid | in theory one can have an an account username@domainA and |
| 05:06.08 | mcquaid | another account |
| 05:06.08 | mcquaid | at username@domainB |
| 05:06.18 | mcquaid | When you would activate both of these accounts then the contact for both accounts |
| 05:06.30 | mcquaid | would become username@IPaddr. So username@IPaddr is not unique. So in 0.5 I concatenated the username with the domain value to form |
| 05:06.35 | *** join/#asterisk asterboy (n=Snake@S01060204ee2b6007.ed.shawcable.net) |
| 05:06.47 | asterboy | FWD is down |
| 05:06.49 | mcquaid | username.domain@IPaddr as contact. This is allowed according to the SIP specs RFC 3261. So if asterisk cannot handle this, it looks like a bug in asterisk to me. |
| 05:07.14 | mcquaid | so he's saying he changed this by design and it's according to spec |
| 05:07.15 | trixter | fwd is ringing |
| 05:07.18 | trixter | fwd is working |
| 05:07.29 | trixter | just called 613 (echo test) to verify |
| 05:08.40 | mcquaid | justinu, do you if he is right that this is according to spec and should be acceptable by asterisk? |
| 05:08.55 | asterboy | no the web site |
| 05:09.07 | asterboy | they'll bbl |
| 05:09.23 | trixter | well many people dont read RFCs correctly, that may be the issue here since there is opposition on something that alledgly appears within |
| 05:09.33 | trixter | there is 'should <not>' and 'must <not>' |
| 05:09.54 | trixter | its only against hte RFC if it violates a 'must' clause, where its simply not recommended if it violates a 'should' clause |
| 05:10.18 | asterboy | FWDin FWDout and Store are all "comeing soon" |
| 05:10.30 | trixter | further there is the general rule with internet protocols (although not an RFC afaik) that states that you should be VERY liberal with what you accept and VERY strict with what you send |
| 05:10.35 | trixter | that ensures maximum compatiblity |
| 05:10.48 | asterboy | trixter, how is your FWD configured? Through asterisk? |
| 05:11.07 | trixter | yes via asterisk |
| 05:11.10 | trixter | trying fwdout now |
| 05:11.14 | trixter | got congested |
| 05:11.33 | trixter | another number worked |
| 05:11.34 | asterboy | thought fwdout was not available yet. |
| 05:11.44 | asterboy | unless you mean on your * |
| 05:11.44 | trixter | its been available at least in part for a while |
| 05:11.50 | asterboy | interesting |
| 05:11.52 | trixter | no I mean routing via fwd |
| 05:12.45 | trixter | so if its not working for you its not FWD, its either your box or the network between you and them |
| 05:12.53 | asterboy | ok |
| 05:12.54 | trixter | either of which can be part of the problem |
| 05:13.05 | trixter | oh yeah I am not using iax with fwd sometimes their iax box blows chunks and dies |
| 05:13.09 | trixter | even though the rest of fwd is working |
| 05:13.10 | asterboy | I was talkin about the main web page being down. |
| 05:13.23 | trixter | and that is the reason why I dont use their iax box :) |
| 05:13.27 | trixter | ohhhhh |
| 05:13.33 | asterboy | its up again. |
| 05:13.33 | trixter | thought you meant the service : |
| 05:13.36 | trixter | :) |
| 05:13.58 | asterboy | Can't test the service yet...I'm still trying to get it working direct on my Polycom |
| 05:14.13 | asterboy | getting 403 forbidden in my logs. |
| 05:14.49 | mcquaid | the other thing i'm trying to get working is basic voicemail, i noticed this error in * |
| 05:14.59 | mcquaid | No entry in voicemail config file for '666' |
| 05:16.05 | *** join/#asterisk praet (n=praet@wsip-68-15-32-50.ri.ri.cox.net) |
| 05:16.34 | trixter | look at voicemail.conf for that context |
| 05:16.55 | trixter | see if you really do have an entry, and if you added it recently did you reload the voicemail stuff? |
| 05:17.00 | asterboy | For FWD username=FWD# , password=whatever |
| 05:17.32 | asterboy | FWD should word with any SIP device no? |
| 05:19.10 | asterboy | The expiration has expired on the current version of pulver.Communicator. Please update to the newest version of FWD.Communicator. Sorry for any inconvenience. |
| 05:19.22 | asterboy | "expiration has expired" ??? |
| 05:19.39 | asterboy | didn't know expired could expire |
| 05:21.58 | justinu | haha |
| 05:22.10 | mcquaid | ya i see what i was doing wrong, cool basic voicemail |
| 05:22.25 | justinu | FWD.communicator will only connect to FWD |
| 05:22.29 | justinu | but it'll dial any sip URI |
| 05:22.34 | justinu | so it's useful for that |
| 05:23.34 | distortion | justinu: you the man |
| 05:24.40 | distortion | I see you in here all the time, answering newbie questions just takin care of bidniz |
| 05:24.49 | justinu | wow, thanks man :) |
| 05:25.09 | distortion | just thought id throw some "props" your way :P |
| 05:25.21 | distortion | and some thanks too |
| 05:25.29 | justinu | appreciate that |
| 05:25.45 | justinu | mcquaid: the guy who wrote tinklephone is smoking some good weed |
| 05:25.59 | mcquaid | heh |
| 05:26.21 | mcquaid | is there a way to override the value Reg. Contact? |
| 05:26.26 | justinu | i guess i'll have to open up the rfc |
| 05:28.53 | justinu | Its |
| 05:28.53 | justinu | <PROTECTED> |
| 05:28.53 | justinu | <PROTECTED> |
| 05:29.03 | justinu | secition 19.1.1 of rfc 3261 |
| 05:29.06 | justinu | http://www.ietf.org/rfc/rfc3261.txt |
| 05:30.22 | justinu | you have the source code, i bet you could fix it |
| 05:30.30 | justinu | if you know c |
| 05:31.07 | *** join/#asterisk bkw__ (n=brian@adsl-69-148-35-86.dsl.tulsok.swbell.net) |
| 05:34.31 | justinu | The Contact header field values of the request typically consist of |
| 05:34.31 | justinu | <PROTECTED> |
| 05:34.31 | justinu | <PROTECTED> |
| 05:35.58 | mcquaid | i can get voicemail working just dialing in from my softphone, but if i try to dial in from a landline i get silence |
| 05:36.09 | mcquaid | * reports this error: No audio available on SIP/mcquaid-35e2?? |
| 05:36.14 | trixter | sounds like a context issue |
| 05:36.30 | trixter | well that error doesnt |
| 05:36.35 | justinu | never seen that error |
| 05:38.23 | mcquaid | hmm |
| 05:38.39 | mcquaid | this works fine when i dial in locally: exten => 666,1,VoiceMail(666@mcquaid_mailbox) |
| 05:39.09 | mcquaid | but i was trying this for incoming calls: exten => _4.,1,Dial(SIP/home,3) |
| 05:39.14 | mcquaid | exten => _4.,1,Dial(SIP/home,3) |
| 05:39.16 | mcquaid | ops |
| 05:39.19 | mcquaid | exten => _4.,2,VoiceMail(666@mcquaid_mailbox) |
| 05:39.33 | justinu | does it ring your phone at all? |
| 05:39.45 | mcquaid | i commented out the dial and changed 2 to 1 for mailbox just to see if it would go immediately but no go |
| 05:39.48 | mcquaid | yes it rings |
| 05:39.56 | justinu | 666 is the number of the beast, mcquaid :P |
| 05:39.57 | mcquaid | i made it short for testing but ya it receives the call |
| 05:40.18 | mcquaid | well, it's gotta have some evil inside ;) |
| 05:40.55 | justinu | so it rings the phone, then hangs up on you, and the CLI prints that error? |
| 05:41.21 | mcquaid | it seems like it's receiving it prints: Playing 'vm-intro' (language 'en') |
| 05:41.28 | mcquaid | Playing 'beep' (language 'en') |
| 05:41.30 | justinu | ok |
| 05:41.32 | mcquaid | Recording the message |
| 05:41.40 | mcquaid | x=0, open writing: /var/spool/asterisk/voicemail/mcquaid_mailbox/666/INBOX/msg0010 format: wav, 0x8179890 |
| 05:41.43 | justinu | ohh |
| 05:41.44 | justinu | der |
| 05:41.45 | mcquaid | x=0, open writing: /var/spool/asterisk/voicemail/mcquaid_mailbox/666/INBOX/msg0010 format: wav, 0x8179890 |
| 05:41.50 | mcquaid | damnit |
| 05:41.51 | Qwell | yes, we get it |
| 05:41.56 | justinu | add an answer command between the Dial/Voicemail command |
| 05:41.56 | mcquaid | Jan 7 00:35:17 WARNING[987]: app.c:653 ast_play_and_record: No audio available on SIP/mcquaid-35e2?? |
| 05:42.14 | Qwell | justinu: no need |
| 05:42.23 | justinu | oh, voicemail will answer? |
| 05:42.35 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
| 05:42.57 | mcquaid | so no need for the answer right? |
| 05:43.03 | justinu | guess not |
| 05:44.01 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
| 05:44.28 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
| 05:45.07 | *** join/#asterisk konfuzed (n=KonfuzeD@H135.C72.B0.tor.eicat.ca) |
| 05:48.09 | trixter | TelecomReport Future of open Internet in question? http://www.marketwatch.com/news/story.asp?guid=%7BDEC1D52A-7D0D-4F50-AEAC-1D1698E93A07%7D&siteid=google |
| 05:48.16 | trixter | interesting read |
| 05:49.10 | coppice | unless they stop the drift back to telecoms monopolies in the US, i can't see how the opern internet can survive. |
| 05:49.41 | coppice | i doubt regulating the monopolies as common carriers will entirely work |
| 05:49.58 | trixter | Still, phone and cable operators complain that their networks cost billions of dollars to construct and maintain and that Internet companies should pay a share of the cost. They have broached the idea of charging Internet content providers a fee to guarantee quick and reliable access for customers trying to access their Web sites |
| 05:50.09 | trixter | well the problem is they already are paying the monthly fees for service ... |
| 05:50.42 | trixter | companies are just looking for a legislative means by which to charge additional on top of that and that will force people to leave ... but this applies to more than web sites - any server even itsps could fall into this.. |
| 05:52.03 | justinu | that whole debate is a load of crap |
| 05:52.19 | justinu | i pay for the pipe, i should be able to run anything I want over it |
| 05:53.13 | trixter | right |
| 05:53.15 | *** join/#asterisk KaBewM (n=DA-MAN@66-215-7-106.dhcp.psdn.ca.charter.com) |
| 05:53.18 | *** join/#asterisk PupenoL (n=pupeno@200.123.183.89) |
| 05:53.21 | trixter | the new claim by executives is that you arent really paying for it |
| 05:53.39 | justinu | then what am I paying for? |
| 05:53.44 | trixter | the monthly fee you pay doesnt cover usage !!! how dare you think that! its a ploy to just jack rates on a metered basis |
| 05:53.49 | justinu | both sides can't be expected to pay for the same bandwidth twice |
| 05:54.10 | trixter | which they can do anyway without going to legislators to try to get a new law passed that is stupid |
| 05:54.18 | justinu | the bandwidth itsp's use costs them money too |
| 05:54.41 | trixter | Still, phone and cable operators complain that their networks cost billions of dollars to construct and maintain and that Internet companies should pay a share of the cost |
| 05:54.44 | trixter | read that again |
| 05:54.51 | trixter | they want to go after the businesses ie those that run servers |
| 05:55.00 | trixter | to pay twice effectively, not the user side of it |
| 05:55.14 | justinu | i understand that |
| 05:55.23 | trixter | the user is effectively going to get 'free' bandwidth becuase aparently they claim they dont pay for the bandwidth |
| 05:55.24 | justinu | their argument only works if my pipe was free |
| 05:55.26 | trixter | which is totally silly |
| 05:55.37 | justinu | i pay 50 some odd real dollars for my pipe |
| 05:56.00 | trixter | they got into a voluntary price war and cant provide the service with the management bloat and advertising bloat they inflicted upon themselves so they want the means to bill someone else |
| 05:56.33 | trixter | which means that people will either switch providers or they will have to charge more for access, neither is a good option for this |
| 05:56.46 | trixter | and that means freenode would potentially have ot pay as well or be cut off.. that too is silly |
| 05:57.23 | trixter | becuase that will just make a lot of free stuff go away or go paid and who will pay? as fewer people pay those that do have less reason to pay which causes attrition |
| 05:57.29 | Qwell | it will never happen |
| 05:57.35 | justinu | if the net becomes like that, i'd leave the country |
| 05:57.39 | Qwell | What are ISPs gonna do, charge non US customers? |
| 05:58.21 | linagee | Qwell: sure. USA will tax everyone! :) |
| 05:58.30 | Qwell | s/customers/companies/ |
| 05:58.38 | Qwell | wow |
| 05:58.58 | trixter | the thing is pay up or be blacklisted ... |
| 05:59.06 | linagee | Qwell: if you buy something online, where do you pay tax? the state you bought it in or the state where the company is located? :) |
| 05:59.17 | Qwell | linagee: neither |
| 05:59.18 | trixter | that is not a good thing in general |
| 05:59.28 | trixter | and I wouldnt say never happen because stranger things have already happened |
| 05:59.42 | trixter | you are suipposed to pay to the state you live in |
| 05:59.52 | trixter | however no one does, its overlooked tax evasion technically |
| 05:59.59 | justinu | so, since when did the bot understand s//? |
| 06:01.03 | asterboy | ~astriskguru |
| 06:01.14 | mcquaid | hmm, i can receive call routed to my local sip phone fine, but i can't get a peep out of asterisk when it answers an outside call |
| 06:01.36 | Qwell | justinu: got me |
| 06:01.38 | justinu | mcquaid: codec problem? |
| 06:01.43 | mcquaid | i thought that |
| 06:01.49 | mcquaid | i made sure gsm was allowed |
| 06:01.51 | mcquaid | it wasn't before |
| 06:01.57 | mcquaid | thought i solved it for a sec |
| 06:02.05 | asterboy | ~xlite |
| 06:02.08 | jbot | i guess xlite is at download xlite at: http://snipurl.com/5tgi | and see sample configs at http://snipurl.com/5tgj, or xlite is a free SoftPhone (software phone, requires no hardware) from xten inc, |
| 06:02.08 | justinu | no, what is your ITSP sending you calls on? |
| 06:02.08 | justinu | what codec? |
| 06:02.45 | mcquaid | hmm, not sure |
| 06:02.48 | justinu | but usually with a codec mismatch, * will just drop the call |
| 06:03.27 | mcquaid | since uncommenting gsm i no longer get that error i posted previously |
| 06:03.45 | justinu | the silence error? |
| 06:03.59 | justinu | that was just the voicemail recording app saying it couldn't hear anything |
| 06:04.06 | mcquaid | right now i'm trying |
| 06:04.09 | mcquaid | exten => _4.,1,Answer |
| 06:04.09 | mcquaid | exten => _4.,2,PlayBack(vm-goodbye) |
| 06:04.09 | mcquaid | exten => _4.,3,HangUp() |
| 06:04.16 | mcquaid | and asterisk looks like it's doing fine |
| 06:04.22 | mcquaid | ya i guess 'the silence error' |
| 06:04.34 | *** join/#asterisk DeadAGI (n=lin@oldflame.stephye.net) |
| 06:04.39 | justinu | so that works, but you hear nothing? |
| 06:04.48 | mcquaid | yes i hear nothing |
| 06:04.54 | mcquaid | nothing stands out in asterisk output |
| 06:05.02 | mcquaid | -- Executing Answer("SIP/mcquaid-c732", "") in new stack |
| 06:05.02 | mcquaid | <PROTECTED> |
| 06:05.02 | mcquaid | <PROTECTED> |
| 06:05.02 | mcquaid | <PROTECTED> |
| 06:05.10 | asterboy | jbot, astriskguru has some nice tools, tutorials and download links here: http://www.asteriskguru.com/ |
| 06:05.14 | mcquaid | looks like it's doing it, but i hear nothing |
| 06:05.18 | DeadAGI | anyone used A2Billing? |
| 06:05.18 | justinu | do you have canreinvite on for your softphone? |
| 06:05.33 | mcquaid | i was just playing with that |
| 06:05.35 | asterboy | jbot, astriskguru is a site that has some nice tools, tutorials and download links here: http://www.asteriskguru.com/ |
| 06:05.36 | jbot | asterboy: okay |
| 06:06.20 | mcquaid | i just tried it with canreinvite yes/no same thing |
| 06:06.39 | DeadAGI | anyone? |
| 06:06.46 | justinu | if you turn canreinvite=no, does your phone still work? |
| 06:06.57 | asterboy | jbot, voipcharges is a great site for review of voip termination services here: http://www.voipcharges.com/ |
| 06:06.59 | jbot | asterboy: okay |
| 06:07.55 | mcquaid | trying |
| 06:09.30 | justinu | mcquaid: after that, turn on sip debug, and rtp debug |
| 06:09.38 | justinu | call your * box, paste the output |
| 06:09.40 | justinu | it'll be a lot |
| 06:09.49 | mcquaid | ok |
| 06:11.17 | justinu | hey, none of that :P |
| 06:11.22 | mcquaid | hmm, not sure what i've done, can't get voice working in a call now either |
| 06:11.33 | justinu | ok, confirms my suspicions |
| 06:11.43 | justinu | mcquaid: it was the canreinvite=no that broke it |
| 06:12.06 | justinu | mcquaid: make sure that externip= is set |
| 06:12.10 | justinu | and localnet |
| 06:12.31 | Dr-Linux | justinu: what is canreinvite ? i have this in sip.conf but i don't know what does it mean? |
| 06:12.34 | mcquaid | i just called out to my landline and that still works |
| 06:12.41 | mcquaid | but not incoming from pstn |
| 06:12.46 | justinu | dr-linux: it's a long story |
| 06:13.07 | Dr-Linux | justinu: make it short and tell me |
| 06:13.18 | Dr-Linux | your egnlish is nice |
| 06:13.21 | justinu | too hard to make it short |
| 06:13.32 | Dr-Linux | :S |
| 06:14.20 | justinu | it has to do with removing asterisk from the RTP stream |
| 06:14.27 | justinu | allowing the two sip peers to talk directly to each other |
| 06:15.37 | Dr-Linux | ooo ic :P |
| 06:17.29 | Dr-Linux | Justin, i was told, i will move to other dept, after one month |
| 06:17.37 | justinu | is that good? |
| 06:17.42 | Dr-Linux | justinu: bcoz they are unable to find my replacment |
| 06:17.44 | Dr-Linux | yeah |
| 06:17.49 | justinu | more $? |
| 06:18.16 | Dr-Linux | justinu: they double my salary, and given me, Network and Software engineer Designation |
| 06:18.55 | justinu | dr-linux: excellent!! |
| 06:18.58 | justinu | hey mcquaid |
| 06:19.01 | justinu | you there? |
| 06:19.03 | Dr-Linux | Thanks! |
| 06:19.10 | justinu | dr-linux: congratulations |
| 06:19.11 | mcquaid | ya still trying to get it back the way it was |
| 06:19.18 | asterboy | congrats! |
| 06:19.31 | justinu | mcquaid: show the guy who wrote tinkerphone this |
| 06:19.38 | mcquaid | i don't get why i could make and receive calls but now only make outgoing to pstn successfully |
| 06:20.13 | justinu | http://pastebin.com/494542 |
| 06:20.29 | justinu | mcquaid: just do what I said with the sip/rtp debug |
| 06:20.36 | justinu | that'll tell the whole story |
| 06:21.48 | Dr-Linux | justinu: please give me link for asterisk good book, i'll study it while i'm at home |
| 06:21.55 | justinu | ~thebook |
| 06:21.56 | jbot | methinks thebook is Asterisk: The Future of Telephony, released under the Creative Commons license and available at http://www.asteriskdocs.org << Read the book online! |
| 06:22.20 | Dr-Linux | free ? |
| 06:22.24 | justinu | yes |
| 06:22.33 | justinu | PDF file |
| 06:22.52 | *** join/#asterisk welles (n=welles@222.90.27.131) |
| 06:23.20 | Dr-Linux | okey |
| 06:23.41 | justinu | Dr-Linux: when does the bus leave? |
| 06:24.46 | Dr-Linux | i'll go at night |
| 06:25.00 | justinu | 10 hour journey? |
| 06:25.03 | Dr-Linux | justinu: now i have to do some shoping for my family |
| 06:25.08 | justinu | what are you buying? |
| 06:25.23 | *** join/#asterisk redman (i=redman@dhcp-0-2-b3-9a-4a-5b.cpe.quickclic.net) |
| 06:25.23 | Dr-Linux | yes but with changings buses i reach to 12 hours |
| 06:25.31 | Dr-Linux | from here |
| 06:25.41 | justinu | where do you change busses? |
| 06:25.43 | Dr-Linux | i gonna buy a Cell phone Nokia 6600 :P |
| 06:26.59 | mcquaid | justinu, i notcied this Adding non-codec 0x1 (telephone-event) to SDP |
| 06:27.00 | asterboy | interesting...pulver.com has a link to http://www.freeworlddialup.com/index.html that says "FWD is currently getting a facelift right now. We'll be back up shortly." |
| 06:27.19 | Flauto | my fwd is still working |
| 06:27.21 | Dr-Linux | justinu: please friend give me a book specific link, that you think is good for me |
| 06:27.22 | asterboy | but if you go direct to the site: http://www.freeworlddialup.com/ all is good. |
| 06:27.27 | Flauto | yours is not? asterboy? |
| 06:27.37 | asterboy | just testing now...downloaded x-lite |
| 06:27.43 | justinu | mcquaid: that's not a problem |
| 06:27.43 | *** join/#asterisk wellng (n=welles@222.90.27.131) |
| 06:27.53 | mcquaid | ok |
| 06:27.54 | asterboy | been trying to get it to work direct without and * sip proxy to my Polycom Phone. |
| 06:28.06 | justinu | dr-linux: just read what interests you |
| 06:28.31 | Dr-Linux | i need to know about dialplans nothing else |
| 06:28.32 | mcquaid | justinu, pasted output here: |
| 06:28.34 | mcquaid | http://pastebin.ca/36079 |
| 06:28.35 | Dr-Linux | all other things are easy |
| 06:28.43 | justinu | Dr-Linux: lol |
| 06:28.54 | Dr-Linux | but SIP :@ |
| 06:29.08 | justinu | mcquaid: you had rtp debug on? |
| 06:29.10 | Dr-Linux | justinu: you have with me, so i never care about SIP stuff :P |
| 06:29.13 | asterboy | hmmmm xlite is timeing out on my login. |
| 06:29.16 | mcquaid | damn no |
| 06:29.18 | mcquaid | one sec |
| 06:30.12 | mcquaid | with rtp debug on i see a bunch lines of from to's |
| 06:30.20 | mcquaid | so it looks like it's sending/receiving |
| 06:30.21 | justinu | ok, that's good |
| 06:30.25 | justinu | you can turn it off |
| 06:30.28 | Dr-Linux | justinu: please give me book pdf link |
| 06:30.36 | justinu | ~thebook |
| 06:30.37 | jbot | somebody said thebook was Asterisk: The Future of Telephony, released under the Creative Commons license and available at http://www.asteriskdocs.org << Read the book online! |
| 06:30.38 | Dr-Linux | i'm in hurry so i can't find any for dialplan |
| 06:30.40 | justinu | get it from that site |
| 06:30.47 | Dr-Linux | i'm there |
| 06:31.17 | justinu | http://www.nufone.net/downloads/asteriskdocs/AsteriskTFOT.zip |
| 06:31.21 | *** join/#asterisk dudes (n=dudes@12-215-33-205.client.mchsi.com) |
| 06:31.42 | Dr-Linux | Thanks |
| 06:31.53 | Dr-Linux | justinu: do you need anything from Tribals? |
| 06:32.02 | justinu | yeah, bring me a stinger missile |
| 06:32.41 | Dr-Linux | give me your address, i'll send you for free |
| 06:32.49 | justinu | lol |
| 06:34.14 | mcquaid | i don't get that now even restoring my sip.conf and extensions.conf that i know were working only works one way now |
| 06:35.05 | *** join/#asterisk DeadAGI (n=lin@oldflame.stephye.net) |
| 06:35.08 | justinu | i see the problem mcquaid |
| 06:35.24 | mcquaid | really? cool |
| 06:35.32 | justinu | line 23 of your paste |
| 06:35.34 | justinu | see it? |
| 06:35.54 | mcquaid | yes |
| 06:35.58 | justinu | you need to set localnet, and externip in your sip.conf |
| 06:36.11 | justinu | your * box is telling your ITSP that it needs to send RTP to 10.1.0.12 |
| 06:36.15 | justinu | a non-routeable IP |
| 06:36.56 | asterboy | Interesting peering page: http://www.freeworlddialup.com/learnmore/?p=features&s=accessnumbers#peering |
| 06:36.56 | mcquaid | ok but why was it working earlier with these same configs? |
| 06:37.10 | justinu | i dunno, fix that problem first |
| 06:37.13 | mcquaid | k |
| 06:37.28 | justinu | asterboy: i found most of those to not be working |
| 06:37.35 | justinu | FWD seems like it's always in a semibroken state |
| 06:37.37 | asterboy | hmmm |
| 06:37.47 | *** join/#asterisk alephcom_ (n=alephcom@host75.net14.mcsnet.ca) |
| 06:38.00 | asterboy | gotta agree...even the web site is kindof...well...useless |
| 06:38.09 | justinu | it's useful, sure |
| 06:38.18 | mcquaid | justinu, so i put my external ip right? |
| 06:38.23 | asterboy | everything is "comeing soon" |
| 06:38.25 | justinu | mcquaid: yes |
| 06:38.44 | asterboy | search for "x-lite" shows nothing. |
| 06:38.53 | justinu | you can use xlite with fwd |
| 06:39.16 | asterboy | no mention of it on the main page...gotta dig |
| 06:39.56 | justinu | fwd is still open |
| 06:39.58 | justinu | for now at least |
| 06:40.06 | *** join/#asterisk goh (n=goh@202.133.101.88) |
| 06:40.38 | *** join/#asterisk zoid99_ (n=chris@24.96.150.105) |
| 06:41.23 | *** join/#asterisk tdscott2 (n=chatzill@24.176.241.71) |
| 06:42.30 | mcquaid | hmm, still not working |
| 06:43.02 | justinu | hmm |
| 06:43.12 | justinu | is the sdp correct now? |
| 06:45.02 | *** join/#asterisk aNaSTaCia_geBeri (n=Ceren_in@85.102.152.221) |
| 06:45.09 | mcquaid | sdp? |
| 06:45.22 | justinu | that part of the sip message |
| 06:45.36 | *** join/#asterisk harryvv (n=none@S010600a0c93f6f7e.vs.shawcable.net) |
| 06:45.37 | mcquaid | now it looks like this o=root 1699 1701 IN IP4 10.1.0.12 |
| 06:45.41 | justinu | still wrong |
| 06:45.47 | justinu | paste your sip.conf |
| 06:45.51 | mcquaid | k |
| 06:47.19 | *** join/#asterisk Flauto (n=zhao@c-24-13-6-136.hsd1.il.comcast.net) |
| 06:47.30 | mcquaid | http://pastebin.ca/36080 |
| 06:47.37 | *** join/#asterisk zimdog (n=zimdog@c-24-9-24-165.hsd1.co.comcast.net) |
| 06:48.49 | *** join/#asterisk Luke-Jr (n=luke-jr@user-0c938qu.cable.mindspring.com) |
| 06:49.37 | *** join/#asterisk CleanerX (n=nix@nat-ph3-wh.rz.uni-karlsruhe.de) |
| 06:49.40 | asterboy | FWD.Communicator just dies when I try to run the download: http://www.freeworlddialup.com/products/count.php?filename=pC_953_050602.exe |
| 06:50.16 | justinu | my fwd communicator is working ok |
| 06:50.31 | asterboy | no, the file to INSTALL communicator. |
| 06:50.37 | justinu | oh |
| 06:50.49 | asterboy | man, can't catch a break. |
| 06:51.07 | justinu | welcome to telcom |
| 06:51.11 | justinu | only the strong survive :P |
| 06:51.11 | asterboy | :P |
| 06:51.30 | asterboy | doh |
| 06:53.13 | mcquaid | justinu, did u take a look? |
| 06:53.24 | justinu | no... spaced out |
| 06:53.32 | mcquaid | heh |
| 06:54.17 | justinu | back at ya |
| 06:54.18 | justinu | http://pastebin.ca/36082 |
| 06:56.22 | mcquaid | now the localnet should be my internal ip? |
| 06:56.48 | justinu | the internal network |
| 06:56.51 | harryvv | anyone know of any voip gateway boxes that work with legacy digital phones? |
| 06:58.48 | harryvv | Guess not |
| 06:59.03 | justinu | yeah, what kind of legagy digital phone? |
| 06:59.46 | *** join/#asterisk ptiggerdine (n=ptiggerd@c220-237-93-88.rochd1.qld.optusnet.com.au) |
| 07:00.07 | justinu | mcquaid: waiting... :P |
| 07:00.16 | mcquaid | still trying |
| 07:00.25 | harryvv | justinu, I did not look at the brand at this office today but thay had there phone system upgraded and had some chinese made voip gateway interfaced to it. |
| 07:00.33 | harryvv | I was a little suprised. |
| 07:00.42 | justinu | yeah, they have such devices |
| 07:01.05 | harryvv | really like what gateway models do you know of? |
| 07:01.31 | mcquaid | ok just to clarify, my internal ip is 10.1.0.12 |
| 07:01.33 | *** join/#asterisk Qwell (n=north@unaffiliated/qwell) |
| 07:01.33 | mcquaid | should i be using that? |
| 07:01.38 | mcquaid | for the localnet? |
| 07:01.46 | justinu | no |
| 07:01.51 | justinu | use 10.0.0.0/255.0.0.0 |
| 07:02.01 | justinu | you're describing a NETWORK, not a HOST |
| 07:02.45 | mcquaid | ok tried that still silence |
| 07:02.54 | justinu | SDP is correct now? |
| 07:03.01 | mcquaid | checking |
| 07:04.27 | mcquaid | now i see this: o=home 2746019974 2746022367 IN IP4 10.1.0.12 |
| 07:04.31 | drray | Prestige 2000W VoIP Wi-Fi phone is $160 at http://store.wifi-parts.com/zypr20.html |
| 07:04.35 | mcquaid | not sure if i'm looking in the wrong area |
| 07:04.36 | justinu | damn, it's still wrong. |
| 07:04.44 | justinu | did you do a sip reload? |
| 07:04.50 | mcquaid | yes |
| 07:04.56 | mcquaid | well i jsut do reload |
| 07:05.11 | justinu | mcquaid, can you paste the 200 OK message? |
| 07:05.36 | justinu | btw, the reason you're having all these problems is because SIP and NAT don't behave well |
| 07:05.43 | mcquaid | Transmitting (no NAT) to 10.1.0.12:6001: |
| 07:05.43 | mcquaid | SIP/2.0 200 OK |
| 07:05.51 | *** join/#asterisk dokhench (n=no@adsl-065-080-180-134.sip.bna.bellsouth.net) |
| 07:05.53 | mcquaid | that? |
| 07:06.05 | justinu | um, actually we want to see the one going out to the sip provider |
| 07:06.11 | mcquaid | yeah, i've read people have problems, but with clients directly connected i had no need for nat nor stun |
| 07:06.13 | mcquaid | it just worked |
| 07:06.20 | mcquaid | and so did * a few hours ago |
| 07:06.37 | justinu | that's because the phone is smart enough to tell your provider about it's external IP |
| 07:06.52 | mcquaid | ah |
| 07:07.10 | justinu | tinklephone apparently does that part right :) |
| 07:07.18 | mcquaid | heh |
| 07:08.43 | mcquaid | anything else i can check/do? |
| 07:09.05 | justinu | hmm, i dunno why it's not picking up the settings from sip.conf |
| 07:09.37 | justinu | i've run * like this before |
| 07:11.36 | mcquaid | i just wanted to make sure and connected directly with twinkle and i can still make/receive calls |
| 07:12.42 | justinu | yeah, maybe restart * completely? |
| 07:12.52 | mcquaid | i've done that a couple of times |
| 07:12.57 | justinu | o |
| 07:12.58 | justinu | ok |
| 07:13.02 | justinu | um... trying to think |
| 07:13.09 | mcquaid | do i have to enable nat when specifiying externip/local? |
| 07:13.14 | justinu | oh |
| 07:13.24 | mcquaid | i hat nat commented out |
| 07:13.28 | mcquaid | i assume default it off? |
| 07:13.31 | justinu | i can't remember |
| 07:13.34 | justinu | play with the settings |
| 07:13.38 | justinu | you might find a combo that works |
| 07:15.01 | mcquaid | i still dont get using 10.0.0.0 when no box here has that |
| 07:15.24 | mcquaid | and to be honest i've never really understood what a subnet mask does |
| 07:15.44 | justinu | yeah, you gotta clear up on this stuff |
| 07:15.52 | justinu | it means something to you now :) |
| 07:15.57 | mcquaid | heh ya |
| 07:16.22 | justinu | 10.0.0.0 is what you call a network address |
| 07:16.55 | justinu | you couldn't assign that to a PC, using the subnet mask 255.0.0.0 |
| 07:17.57 | justinu | read up one some basic TCP/IP stuff |
| 07:20.20 | trixter | Cell Phone Includes Internet Telephony Capability http://abcnews.go.com/Technology/PCWorld/story?id=1480807 |
| 07:20.28 | mcquaid | ya'll ill do that |
| 07:21.10 | drray | trixter - the holy grail |
| 07:21.23 | drray | probably a mirage though |
| 07:21.32 | trixter | prolly |
| 07:21.38 | trixter | havent read it yet just came in |
| 07:22.02 | drray | but I'd pay $300 for the right phone |
| 07:22.08 | trixter | it might just be a smartphone with a softphone |
| 07:22.24 | trixter | I have a friend that owns a cell store, I might see if he can get a demo |
| 07:23.12 | trixter | The GF200 looks like just another cell phone, and that's the point.You get both types of functionality from the one phone that you'll be carrying around anyways. |
| 07:23.30 | trixter | ok, not a traditional smartphone (basically a phone+pda) ... that should be good at least.. cause most of those are too large |
| 07:24.05 | trixter | Note: The phone can use WEP authentication but can't yet connect to hotspots that require Web authentication. |
| 07:24.30 | trixter | big drawback, but without a browser ... there needs to be some standard to let these types of devices auth without a web browser |
| 07:24.40 | drray | http://www.tomsnetworking.com/Sections-article138-page5.php |
| 07:24.49 | trixter | Hmm.. here is an idea a voice server (asterisk perhaps) at every hotspot to let such phones connect |
| 07:28.45 | *** join/#asterisk kusznir (n=kusznir@pool-70-110-34-187.sea.dsl-w.verizon.net) |
| 07:29.05 | drray | http://www.voipsupply.com/product_info.php?cPath=95_115&products_id=1067&osCsid=88036fabb68f07533d230eb0230fa669 |
| 07:29.52 | *** join/#asterisk DA-MAN (n=DA-MAN@66-215-7-106.dhcp.psdn.ca.charter.com) |
| 07:30.44 | trixter | that last one isnt a cell phone too is it? |
| 07:30.56 | drray | well, since I believe it's fake, sure |
| 07:31.00 | drray | er |
| 07:31.02 | drray | no |
| 07:31.05 | drray | :) |
| 07:31.19 | trixter | heh |
| 07:31.37 | trixter | I should start hitting up people about getting the dualmode phone donated to the sacaug :) |
| 07:32.43 | drray | although I don't know what I would do with a voip wifi phone |
| 07:32.52 | drray | other than play with it |
| 07:32.58 | trixter | sacaug got like $650 worth of stuff (not incl shipping charges which were also donated) for prizes in a contest which has yet to be anounced (janurary we lay the framework feb we finalize the rules, may or so we vote) from thevoipconnection.com and digium |
| 07:33.08 | justinu | heading off to bed for the night.... bye |
| 07:33.11 | trixter | tdm410p, gxp2000s and some bugettone ATAs |
| 07:33.15 | *** part/#asterisk justinu (n=justinu@207.181.0.86) |
| 07:33.26 | Qwell | trixter: well, the tdm's are good, but...damn |
| 07:33.28 | trixter | drray: well the one I posted was a gsm phone that also does sip via wifi |
| 07:33.35 | trixter | so if you arent in the office or home you are on gsm |
| 07:33.43 | drray | yeah |
| 07:34.00 | trixter | which is good I think becuase as the article points out most office workers place calls from the office or home ... |
| 07:34.23 | drray | we use walkie talkies right now |
| 07:34.33 | drray | I'd love to switch us |
| 07:34.35 | kusznir | trixter: Could you re-post the URL? I joined the chat after you posted that. |
| 07:34.48 | drray | so everyone would not have to hear about the toilet that overflowed |
| 07:35.00 | drray | plus you could patch to outside |
| 07:35.03 | drray | etc |
| 07:37.02 | *** join/#asterisk arbius (n=root@c-67-173-45-34.hsd1.il.comcast.net) |
| 07:38.12 | drray | I also think push to talk, walkie talkie mode is something that would rule for asterisk |
| 07:38.41 | trixter | Cell Phone Includes Internet Telephony Capability http://abcnews.go.com/Technology/PCWorld/story?id=1480807 |
| 07:38.55 | kusznir | thanks! |
| 07:39.15 | trixter | all PTT is is a on demand half duplex channel ... think paging on an individual phone |
| 07:39.23 | drray | yeah |
| 07:39.33 | trixter | so you have the framework to do that |
| 07:39.45 | drray | just need a phone that would support it |
| 07:39.54 | trixter | because its PTT a channel isnt allocated unless there is (alledgly) traffic to send |
| 07:40.09 | drray | autoanswer based on callerid |
| 07:40.17 | trixter | so it utilizes better its a built in feature of iDEN (motorolas mobile technology that nextel uses) not an add on |
| 07:40.42 | trixter | well it shouldnt be callerid per se, it should be configurable though |
| 07:40.52 | trixter | because you dont want all calls to be that way |
| 07:40.59 | trixter | imagine someone calling and screaming at 3am while you sleep |
| 07:41.17 | drray | sure, but you set which caller ID's you'll take messages from |
| 07:41.33 | trixter | which means you have to know about everyone that can page you |
| 07:41.37 | trixter | a configuration nightmare |
| 07:41.40 | drray | or hell, which asterisk servers |
| 07:41.53 | drray | you have to know that now with teh PTT nextels? |
| 07:42.04 | drray | don't they have to be in the same group? |
| 07:42.13 | trixter | not anymore with nextel |
| 07:42.15 | trixter | its national |
| 07:42.20 | trixter | well group maybe but before it was group and region |
| 07:42.32 | trixter | I dont know if that is the case though, you just have to dial more digits |
| 07:43.05 | kusznir | you don't have to be in the same group. If you're not, you have to dial the group number * the phone number. |
| 07:43.11 | trixter | its like a region code, city code and subscriber number. region code is like your group, city code is geographical based and subscriber is obvious. |
| 07:43.35 | trixter | I had nextel when you had to not only be in the same group but the same region as well |
| 07:43.38 | drray | kusznir - but if I don't tell you my group number, you can't call |
| 07:43.45 | drray | unless you do it at random |
| 07:44.05 | trixter | I did get calls from people that scanned to find random people |
| 07:44.30 | drray | are mobile to mobile calls free under that system? |
| 07:45.05 | kusznir | that's right. But your PTT number is different from your phone number, so for you to talk to me at all, you have to give me your number. So for example, my direct connect might be 112*5732. Normally you give your number out like that to everyone. If you are in the same group and type the entire thing in anyway, there's no penalty. |
| 07:45.25 | *** join/#asterisk xp__ (n=knoppix@c-24-5-52-183.hsd1.ca.comcast.net) |
| 07:45.50 | kusznir | drray: last I looked at Nextel, ptt was unlimited within the state, but metered accross state boundaries. |
| 07:47.26 | xp__ | ok I am a newbie wanting to know some things about asterisk, anyone in a good mood for newbieness? |
| 07:47.36 | kusznir | I' |
| 07:47.45 | kusznir | lll help where I can. |
| 07:48.04 | Johnnie | xp__: Go ahead. |
| 07:48.48 | joelsolanki | anybody knows the asterisk + voice master sync configuration? |
| 07:48.50 | xp__ | well I just had a party with this tool called powersmb that setup and migrated an nt pdc like so fast I cried inside (not outside cuz I'm a man) |
| 07:49.06 | xp__ | is there a commercial config tool to make an asterisk tool so damn quick its scary? |
| 07:49.33 | Johnnie | Yeah, it's called Digium. haha |
| 07:49.48 | xp__ | is it a software configuratoin tool that sets it all up hella fast? |
| 07:49.51 | xp__ | how much is it? |
| 07:50.03 | kusznir | There are a few frontends; I haven't used any of them so I don't know. Asterisk@home is one that comes to mind |
| 07:50.17 | kusznir | Digium is the company that started and still does the majority of the development for asterisk. |
| 07:50.22 | kusznir | They consult for a fee. |
| 07:50.26 | Johnnie | That was a partial joke, xp__... Digium is the company that basically produces Asterisk. |
| 07:50.48 | Johnnie | There are a lot of freelancers here too, xp__. |
| 07:50.49 | mcquaid | i'm dealing with incoming calls to * from outside line are silent, but calling * with sip is fine |
| 07:50.55 | xp__ | so there is no real kind of turnkey little config software tool I could run to set it all up? |
| 07:50.57 | Johnnie | Asterisk is actually pretty easy. |
| 07:51.12 | trixter | I thought open source developers create asterisk and digium just sells it via their business edition |
| 07:51.38 | xp__ | well I am in a small predicament, where i live I have a cell phone but no signal, I want to have an internet phone, what are you recommendations and how much would it cost to setup/have? |
| 07:51.39 | Johnnie | Well, there are tools to aid in the process, like AMP (Asterisk Management Portal), it's basically a web based interface which will write the configuration files for you. |
| 07:51.51 | kusznir | xp__ asterisk@home is its own distrabution...its supposed to have an easy-to-use interface to config everything. It uses AMP, IIRC. |
| 07:52.00 | xp__ | well that would be cool |
| 07:52.07 | Qwell | xp__: vi |
| 07:52.17 | Qwell | vi is the best configuration tool for Asterisk |
| 07:52.20 | Johnnie | I should have clarified... AMP is part of Asterisk@Home. |
| 07:52.22 | kusznir | the downside is it is designed to run on a dedicated box (e.g., dedicate a system to running asterisk) |
| 07:52.30 | dudes | Qwell - next to nano |
| 07:52.42 | Qwell | dudes: I prefer nano actually :p |
| 07:52.52 | Qwell | but, nano users are newbs. heh |
| 07:52.54 | xp__ | what about my at home internet phone? |
| 07:53.04 | dudes | I can use vi but I figure why bother |
| 07:53.09 | Qwell | dudes: exactl |
| 07:53.09 | Qwell | y |
| 07:53.12 | xp__ | I take it I should use asterisk? |
| 07:53.17 | Qwell | xp__: sure |
| 07:53.23 | kusznir | Most precompiled packages come with a sample config that more-or-less works out of the box (you have to config a phone to use it with); from the sample files you can config up basic configs pretty easy. |
| 07:53.24 | Johnnie | xp__: You can do that, yes. |
| 07:54.18 | mcquaid | anyone good with potential nat issues? i can dial out to my landline without issue, but calls from pstn to * are silence |
| 07:54.18 | xp__ | should I use vonage? |
| 07:54.23 | Qwell | xp__: no |
| 07:54.25 | kusznir | not if you want to use asterisk |
| 07:54.45 | xp__ | how much will it cost for me to use asterisk so I can call a land line if I have no land line? |
| 07:55.18 | kusznir | Vonage won't release your SIP userid and password |
| 07:55.23 | Qwell | xp__: how many minutes do you use per month? |
| 07:55.24 | kusznir | asterlink is one... |
| 07:55.43 | xp__ | Qwell alot |
| 07:55.46 | kusznir | $0.02/min, toll free number, no monthly charges. |
| 07:55.50 | Qwell | xp__: how many? |
| 07:55.58 | Qwell | 500, 1500, 5000, 50000? |
| 07:55.59 | xp__ | 3000+ |
| 07:56.15 | Qwell | any "unlimited" providers will cut you off at that rate |
| 07:56.21 | Qwell | so, $60+ |
| 07:57.24 | xp__ | so how does that work, I get an account then point asterisk to their server to convert to a landline my calls? |
| 07:57.38 | kusznir | xp__ check www.voip-info.org for a list of providers. |
| 07:57.48 | xp__ | cool |
| 07:58.15 | trixter | BV tends to let you do 5000/mo on a residental line without complaints really |
| 07:58.26 | Qwell | trixter: good luck sustaining > 3k |
| 07:58.46 | drray | I think you just can't be in their top 5% offenders list |
| 07:58.55 | trixter | but if they fell that based on calling patterns that you are a business they can change your rate plan and back bill you for the prior 6 months (assuming you had service for 6 months) at the business rate which can include per minute charges depending on where you call |
| 07:59.09 | trixter | Qwell: I did 5k/mo for a long time on multiple accts |
| 07:59.14 | trixter | we had over 13 doing that each month |
| 07:59.28 | drray | if you are reselling, or multiple users.. I can see their point |
| 07:59.31 | trixter | and all the calls were to the same number (all 13) and all were during the same hours |
| 07:59.34 | kusznir | You set asterisk up to route calls to a particular provider based on the number (or all numbers, if you desire). It will place calls through that provider. In order to recieve calls, your provider must set up a DID to point to you. If you want to keep your existing number, you have to make sure your provider supports porting. |
| 07:59.43 | trixter | thye questioned it but I talked my way out of it and they let it go (security called me) |
| 07:59.52 | kusznir | Many providers have numbers in select areas, but few have it everywhere. |
| 08:00.02 | trixter | well I told em that I put everything on my credit card so I get frequent flier miles but they were friends accounts |
| 08:00.07 | xp__ | lets say I had a friend who had a box at a site that hooked to a box for outgoing calls, I could hook into that could I not? |
| 08:00.11 | trixter | why there were so many all to/from the same place :P |
| 08:00.15 | drray | xp - sure |
| 08:00.28 | trixter | speaking of which I should get my final $1700 payment for doing that |
| 08:00.36 | trixter | arbitrage :D |
| 08:00.56 | drray | trixter - they should give you a commision as well |
| 08:01.13 | trixter | you have to be with em for 3 months for that to kick in |
| 08:01.19 | trixter | they were all built upon each other for that reason |
| 08:01.23 | trixter | but its not instant |
| 08:01.33 | *** join/#asterisk SERGEUS (n=s@195.112.98.13) |
| 08:01.34 | trixter | and the $1700 isnt from BV its from using BV circuits to make calls for which we profited |
| 08:01.39 | trixter | thus the arbitrage comment |
| 08:01.42 | Qwell | trixter: Why were they all to the same place? |
| 08:01.48 | drray | trunking |
| 08:02.11 | trixter | typically arbitrage is used in the stock market but it applies to others.. lets say you see the *same* stock on 2 different markets at the same time, and there is a price difference. you know (as a broker) you can buy on one and sell on the other instantly and profit |
| 08:02.37 | trixter | telephone service largely is a commodity too, if capacity isnt used its lost forever |
| 08:02.38 | drray | market makers |
| 08:03.06 | drray | at some point I believe bandwidth will be sold that way |
| 08:03.55 | trixter | it already is |
| 08:03.57 | kusznir | Relating to that, I've been hearing that many telcos that provide backbone internet circuts are thinking about setting up QoS -- to degrade VoIP connections. |
| 08:04.04 | trixter | and providers were talknig about adding another tier to service a few weeks ago |
| 08:04.17 | drray | a 3 tier system would be ideal |
| 08:04.24 | trixter | if you want priority and all you have to pay extra, basically cheaper accounts would be surplus bandwidth no guarantees |
| 08:04.30 | drray | free - regular - uuber |
| 08:04.47 | drray | free would be for web/public service |
| 08:04.48 | trixter | there is a telecom market that lets you buy excess capacity, no guarantee your call will go through, wont be cut off, but you can get it damn near cost |
| 08:05.04 | *** part/#asterisk kusznir (n=kusznir@pool-70-110-34-187.sea.dsl-w.verizon.net) |
| 08:05.11 | trixter | becuase the people that own the circuits know that if they sell it for just over cost they get at least something if they dont sell it that capacity is lost foever |
| 08:05.22 | *** join/#asterisk postel (n=jk@unaffiliated/postel) |
| 08:05.25 | trixter | it will never be free |
| 08:05.30 | trixter | not in the truest sense of the word |
| 08:05.37 | trixter | the cost can approach free but it will never be free |
| 08:05.59 | trixter | 'free' stuff that has advertising on it isnt 'free' in the truest sense of the word but it is paid for by the advertiser |
| 08:06.25 | trixter | someone somewhere has to pay, people will close their doors and do something else 99.99999% of the time than give it away free indefinately |
| 08:07.41 | xp__ | I want to hear some stories of interesting uses with asterisk at home, did you call russia or something cool for free and stuff? |
| 08:08.06 | drray | well, free in the sense that it's a public service |
| 08:08.34 | drray | I've never understood why the cable companies did not wire everyone for free cable (6 channels) |
| 08:08.49 | drray | back in the analog days |
| 08:09.31 | mcquaid | anyone good with potential nat issues? |
| 08:09.51 | *** join/#asterisk argos73 (i=1000@jason.argos.org) |
| 08:10.10 | mcquaid | i can call out to my land line, but incoming calls are just silence |
| 08:12.22 | *** join/#asterisk himalrana (n=himal@59.95.201.172) |
| 08:12.39 | *** join/#asterisk spunz_ (n=spunz@h081217096096.dyn.cm.kabsi.at) |
| 08:12.47 | himalrana | hello |
| 08:12.48 | himalrana | all |
| 08:13.19 | himalrana | asterisk is good |
| 08:13.25 | himalrana | but not for video |
| 08:13.48 | himalrana | i m currntely working to asterisk do this |
| 08:14.20 | himalrana | using gstreamer to generate h263 files required by asterisk |
| 08:14.40 | himalrana | but dont know how to integrate those files to asteriks |
| 08:15.57 | *** part/#asterisk arbius (n=root@c-67-173-45-34.hsd1.il.comcast.net) |
| 08:17.20 | himalrana | does any one has any idea |
| 08:18.08 | *** part/#asterisk himalrana (n=himal@59.95.201.172) |
| 08:18.26 | *** join/#asterisk himalrana (n=himal@59.95.201.172) |
| 08:22.16 | himalrana | hi all |
| 08:26.05 | *** join/#asterisk MatsK (i=MatsK@c83-253-29-22.bredband.comhem.se) |
| 08:27.01 | *** part/#asterisk himalrana (n=himal@59.95.201.172) |
| 08:44.13 | *** join/#asterisk TonyM (n=TonyM@adsl-solo-80-168-224-3.claranet.co.uk) |
| 08:45.24 | *** join/#asterisk zhao_ (n=zhao@c-24-13-6-136.hsd1.il.comcast.net) |
| 08:47.22 | *** join/#asterisk welles (n=welles@222.90.27.131) |
| 08:51.31 | *** join/#asterisk riksta (n=rick@213.121.151.210) |
| 08:57.06 | *** join/#asterisk tengulre (n=tengulre@219.145.57.9) |
| 08:58.01 | *** join/#asterisk drray_ (i=drray@dsl254-011-243.sea1.dsl.speakeasy.net) |
| 09:03.10 | SplasPood | Does anyone know what happened to res_data? |
| 09:07.29 | *** join/#asterisk [Airwolf] (n=airwolf@82-171-75-4.dsl.ip.tiscali.nl) |
| 09:08.13 | brc_ | HAHAHAHAHA |
| 09:08.27 | brc_ | "where can i find all .C files" |
| 09:11.03 | *** join/#asterisk mogorman (n=mogorman@user-24-236-84-48.knology.net) |
| 09:11.12 | Qwell | mogorman: ! |
| 09:11.36 | mogorman | qwell! |
| 09:12.14 | mogorman | whats up man? |
| 09:12.44 | Qwell | not a lot |
| 09:13.02 | Qwell | kernel compile...moving to udev...emerge -u world... |
| 09:13.03 | Qwell | fun |
| 09:22.38 | sancho|laptop | lol |
| 09:28.42 | argos73 | anyone know what the weather is like in Las Vegas in early March? |
| 09:29.31 | Qwell | about 400 degrees |
| 09:29.41 | argos73 | hehe |
| 09:29.51 | MatsK | warm and sunny ? |
| 09:30.01 | Qwell | bed time |
| 09:30.19 | argos73 | would be nice... thinking of surprising my with with a birthday trip |
| 09:30.33 | Qwell | your with, eh? |
| 09:30.40 | argos73 | oops |
| 09:30.41 | argos73 | wife |
| 09:30.54 | *** join/#asterisk Simon- (i=byte@2001:4bd0:1000:1:2e0:4cff:feed:1cfb) |
| 09:31.12 | argos73 | damned trained fingers |
| 09:35.55 | argos73 | 69/47, per weathe.com |
| 09:36.06 | argos73 | not bad |
| 09:37.17 | welles | hi mogorman |
| 09:39.16 | welles | are u there mogorman? |
| 09:50.20 | *** join/#asterisk mkl1525 (n=daniel@84.19.203.193) |
| 10:00.26 | *** join/#asterisk af_ (n=af@ip-142-84.sn1.eutelia.it) |
| 10:02.50 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 10:05.08 | *** join/#asterisk UlbabraB (n=caplaz@host241-43.pool8172.interbusiness.it) |
| 10:16.17 | *** join/#asterisk welles (n=welles@222.90.27.131) |
| 10:20.51 | *** join/#asterisk [Airwolf] (n=airwolf@82-171-75-4.dsl.ip.tiscali.nl) |
| 10:25.34 | *** join/#asterisk zhao__ (n=zhao@c-24-13-6-136.hsd1.il.comcast.net) |
| 10:25.46 | zoa | hey ho airwolf |
| 10:31.35 | [Airwolf] | hey zoa |
| 10:34.41 | *** join/#asterisk Shoragan (n=shoragan@d072.apm.etc.tu-bs.de) |
| 10:35.56 | wasim | kudos zoa for the new jb |
| 10:36.14 | zoa | did you give it a try wasim ? |
| 10:37.08 | wasim | not yet, sir, but from what i've heard it rocks |
| 10:37.21 | wasim | we intend to after the weekend |
| 10:37.53 | zoa | great |
| 10:37.54 | zoa | let us know |
| 10:44.38 | *** join/#asterisk svenna_ (n=svenna@p548D3217.dip0.t-ipconnect.de) |
| 10:51.09 | *** join/#asterisk MatsK (i=MatsK@c83-253-29-22.bredband.comhem.se) |
| 10:56.40 | *** join/#asterisk chapeaurouge (n=chap@85.201.81.201) |
| 11:29.52 | *** join/#asterisk loud (n=ariel@cypher.punk.net) |
| 11:42.54 | *** join/#asterisk dcoulson (n=dcoulson@wilbur.geekcolony.net) |
| 11:56.50 | *** join/#asterisk backblue_ (n=moo@87-196-6-187.net.novis.pt) |
| 11:57.12 | *** join/#asterisk OloBola (n=not@adsl-69-110-121-26.dsl.pltn13.pacbell.net) |
| 11:58.28 | OloBola | can I pass a variable from dial plan to php? |
| 11:59.52 | *** join/#asterisk zotz (n=zotz@24.231.47.175) |
| 12:01.34 | wasim | OloBola: only if you pay 2c first everytime |
| 12:01.55 | wasim | or 60 for a dollar |
| 12:04.31 | *** join/#asterisk ToTo (n=ToTo@host242-83.pool8260.interbusiness.it) |
| 12:08.13 | mkl1525 | Hi, haven't found app_rxfax and app_txfax in spandsp-0.0.2pre21.tar.gz or in my bristuff asterisk version - so could anybody please give me a hint where to get them? |
| 12:09.01 | OloBola | ok, thanks! |
| 12:13.00 | *** join/#asterisk agx (n=agx@ip-37-53.sn1.eutelia.it) |
| 12:13.25 | agx | morning, wich IAX softclient allow to specify a different port? i need to set it to 8080 to bypass firewall :) |
| 12:13.26 | OloBola | dudes, not gonna cost me a dime: exten => 1,2,agi,test.py|toll-free-var |
| 12:14.04 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 12:14.35 | OloBola | xten? |
| 12:14.43 | *** join/#asterisk endre (i=endre@endre.be) |
| 12:14.47 | endre | +hi |
| 12:15.05 | OloBola | http://www.xten.com/ |
| 12:18.10 | agx | let's give xten a try... i didn't liked it too much when using SIP :) |
| 12:21.20 | agx | dude Xten seems SIP only... i need IAX |
| 12:23.08 | endre | +mw3: greetings |
| 12:23.34 | Mw3 | hello endre |
| 12:24.11 | *** join/#asterisk spunz_ (n=spunz@h081217096096.dyn.cm.kabsi.at) |
| 12:24.53 | *** join/#asterisk [Airwolf] (n=airwolf@82-171-75-4.dsl.ip.tiscali.nl) |
| 12:25.04 | *** join/#asterisk azrishahril (n=azrishah@60.50.193.185) |
| 12:26.41 | OloBola | how do I retreive my variables from php script sent from my dialplan? script.agi|arg1|arg2 |
| 12:30.57 | *** join/#asterisk CPC (n=aifds@201.29.181.236) |
| 12:33.06 | agx | bye, asking on forum, this seems a bot only channell :-P |
| 12:33.08 | *** part/#asterisk agx (n=agx@ip-37-53.sn1.eutelia.it) |
| 12:36.09 | *** join/#asterisk fugitivo (n=ajf@201.255.179.24) |
| 12:36.09 | fugitivo | morning |
| 12:39.48 | zoa | morning |
| 12:40.14 | endre | morning |
| 12:40.33 | endre | it is 1.40pm here in the 'morning' |
| 12:43.53 | OloBola | how can I pass an argument from dialplan to an agi script? |
| 12:43.54 | fugitivo | :) 9:40am here |
| 12:44.18 | zoa | 2:44 here |
| 12:44.19 | zoa | :p |
| 12:44.23 | zoa | pm |
| 12:44.34 | fugitivo | OloBola: agi(command|args) |
| 12:44.37 | zoa | all your clocks are belong to me |
| 12:44.47 | tuxinator_linux | almost 5 AM here in SoCal |
| 12:44.56 | tuxinator_linux | zoa is silly |
| 12:45.00 | zoa | what is socal ? |
| 12:45.02 | fugitivo | OloBola: show application agi will answer your question next time |
| 12:45.12 | OloBola | fugitivo: what is the variable name from within my agi script? |
| 12:45.16 | tuxinator_linux | Sothern California, that state where you met me |
| 12:45.21 | *** join/#asterisk shawn (n=welles@222.90.27.131) |
| 12:45.26 | CPC | who could help me in pvt to fix some problens that I have to link 2 * servers?? |
| 12:45.38 | fugitivo | OloBola: that depends on your programming language |
| 12:45.40 | tuxinator_linux | CPC, people don't do private here |
| 12:45.59 | drray_ | CPC, ask in public.. waht problem are you having? |
| 12:46.01 | fugitivo | OloBola: it's the same as passing arguments on the command line |
| 12:46.15 | zoa | ask in public, or in private if you are willing to pay :p |
| 12:46.27 | fugitivo | we accept paypal |
| 12:46.29 | tuxinator_linux | zoa, going to make it to VON in San Jose this year? |
| 12:46.39 | zoa | cpc -> cleyverson@terra.com.br |
| 12:46.48 | CPC | :) |
| 12:46.53 | zoa | why do you cross post between irc and the mailinglist |
| 12:46.57 | zoa | right away ? |
| 12:47.07 | zoa | give people some time to react |
| 12:47.07 | fugitivo | CPC: so, tell us your problems |
| 12:47.13 | zoa | I'm traying to link 2 * servers using SIP and the following errors was show: |
| 12:47.13 | zoa | <PROTECTED> |
| 12:47.13 | zoa | "SIP/AsteriskA:AsteriskA@10.0.0.121/100") in new stack |
| 12:47.13 | zoa | Dec 13 22:46:57 WARNING[8767]: chan_sip.c:1398 create_addr: No such |
| 12:47.13 | zoa | host: 10.0.0.121/100 |
| 12:47.16 | zoa | this is his problem |
| 12:47.26 | fugitivo | why sip? |
| 12:47.28 | fugitivo | use iax for that |
| 12:47.30 | zoa | yeah why sip |
| 12:47.32 | zoa | use iax |
| 12:47.55 | zoa | SIP/AsteriskA:AsteriskA@10.0.0.121/100 -> this dial line is wrong |
| 12:48.02 | zoa | this is the format for a iax2 call i think |
| 12:48.03 | tuxinator_linux | I third it, use IAX |
| 12:48.15 | fugitivo | just replace SIP/ with IAX2/ and it'll work :p |
| 12:49.08 | fugitivo | i love the "switch" |
| 12:49.19 | zoa | i never tried the switch |
| 12:49.22 | zoa | is it any good ? |
| 12:49.23 | fugitivo | it's cool |
| 12:49.36 | zoa | how does it work exactly ? |
| 12:49.51 | zoa | it will use a secret iax2 call ? |
| 12:49.53 | fugitivo | if you dial an extension that doesn't belong to your dialplan, * will find the extension on the server specified by the switch |
| 12:50.07 | zoa | yeah that i know |
| 12:50.11 | zoa | but how do they do that ? |
| 12:50.13 | fugitivo | yes, it's an iax2 call |
| 12:50.35 | fugitivo | you'll see in your cli something like "Accepting call from xxx.xxx.xxx.xxx" |
| 12:50.51 | zoa | so its kinda similar to a match-all extension that would go to the other server i guess |
| 12:51.17 | fugitivo | yes, but you don't need to specify nothing on the other server |
| 12:51.28 | fugitivo | only the user:secret |
| 12:51.41 | zoa | and no context either ? |
| 12:51.50 | zoa | i dont see the difference :) |
| 12:52.30 | *** join/#asterisk kio (n=kio@ool-4577adba.dyn.optonline.net) |
| 12:52.46 | fugitivo | well, i'd never used a match-all extension |
| 12:52.57 | fugitivo | with switch you only need to setup an iax2 user |
| 12:53.01 | fugitivo | on the other server |
| 12:53.11 | fugitivo | and add a switch line to your own server |
| 12:53.41 | zoa | sounds the same as functionality you could do with some extension magic |
| 12:54.03 | zoa | hmm |
| 12:54.08 | fugitivo | well, i'm sure you can do it using extensions |
| 12:54.14 | fugitivo | but this is only a command |
| 12:54.17 | fugitivo | switch => |
| 12:54.18 | zoa | yeah |
| 12:54.18 | fugitivo | that's all |
| 12:54.19 | zoa | true |
| 12:54.29 | *** join/#asterisk DrukenHME (i=Druken@CPE00121716da99-CM000e5cde4ca2.cpe.net.cable.rogers.com) |
| 12:55.01 | fugitivo | i don't know if it works with sip |
| 12:55.09 | fugitivo | i think it does |
| 12:55.14 | zoa | probably it does |
| 12:55.18 | fugitivo | because you specify switch => IAX2/ |
| 12:55.22 | zoa | yeah |
| 12:57.08 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
| 12:58.08 | *** join/#asterisk dingo (n=dingo@58.69.74.218) |
| 12:58.31 | dingo | ok ok whats happened to sip show peers and sip show registry in 1.2.1 |
| 12:59.20 | *** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239) |
| 13:02.09 | DrukenHME | dingo: you using realtime ? |
| 13:02.22 | dingo | dont think so |
| 13:02.32 | CPC | [fugitivo] do u know y this error happends when we use SIP?? |
| 13:03.07 | dingo | DrukenHME, whats the easiest way to tell |
| 13:03.27 | dingo | Asterisk 1.2.1, Copyright (C) 1999 - 2005 Digium. is all it says |
| 13:03.35 | DrukenHME | you'd know if you were.... |
| 13:03.39 | *** join/#asterisk zhao__ (n=zhao@c-24-13-6-136.hsd1.il.comcast.net) |
| 13:03.47 | DrukenHME | realtime allows you to take from a database |
| 13:04.09 | dingo | wel yes with AMP its configured for MySQL |
| 13:04.23 | DrukenHME | your using amp? |
| 13:04.27 | dingo | yes |
| 13:04.39 | dingo | but i still used to be able to do sip show peers |
| 13:04.40 | DrukenHME | then i can't help ya... :) |
| 13:04.49 | DrukenHME | goto #AMportal |
| 13:04.59 | *** join/#asterisk Kidaz (n=kidaz@thedowds.demon.co.uk) |
| 13:05.03 | dingo | wait does the db need to be live for a valid response ? |
| 13:05.05 | Kidaz | Hi all :) |
| 13:05.36 | DrukenHME | well wtf do you think? |
| 13:06.02 | dingo | Druken, let me explain it this way i took only the amp configs from a linux box and put them on a bsd box so the amp db isnt installed just the configs |
| 13:06.05 | *** join/#asterisk TheCops (i=nobody@got.securebinary.com) |
| 13:06.19 | dingo | Druken, hrmm you might have a valid point |
| 13:06.22 | TheCops | hi |
| 13:06.27 | DrukenHME | might? |
| 13:06.51 | DrukenHME | oh shit!, the cops are here! |
| 13:07.03 | TheCops | someone is using UserEvent command for the dialplan and can make more than one line as body? |
| 13:08.55 | dingo | well ok i see the issue, untouched configs work fine, AMP old linux based sql configs dont |
| 13:09.00 | dingo | so much for that idea |
| 13:09.15 | dingo | any good php based guis for asterisk thatll run on FreeBSD :) |
| 13:10.06 | DrukenHME | yeah, the one you create from scratch :) |
| 13:10.07 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
| 13:10.14 | [TK]D-Fender | You failed at "any good php based guis for asterisk" :) |
| 13:10.21 | [TK]D-Fender | GUI = evil |
| 13:10.58 | DrukenHME | [TK]D-Fender: gui CAN be good, but not for certain configs of asterisk |
| 13:11.00 | zoa | hey royk |
| 13:11.16 | zoa | i saw on mantis that you got the jb to work ? |
| 13:11.22 | fugitivo | i don't want to see the letters "AMP" anymore |
| 13:11.24 | TheCops | hi fender |
| 13:11.29 | TheCops | tu est matinal |
| 13:11.29 | TheCops | :) |
| 13:11.30 | zoa | :) |
| 13:11.32 | zoa | ela |
| 13:11.35 | DrukenHME | fugitivo: i'm with you... |
| 13:11.43 | fugitivo | #amportal exists for that |
| 13:12.06 | dingo | AMP desnt run on FreeBSD and its a simple broadvoice config |
| 13:12.13 | DrukenHME | ~amp |
| 13:12.16 | jbot | methinks amp is a web based interface for configuring Asterisk. See http://amp.coalescentsystems.ca/ |
| 13:12.40 | fugitivo | how to add "join channel #amportal" to that? |
| 13:12.41 | dingo | again people AMP desnt run on FreeBSD its written for linux |
| 13:12.56 | fugitivo | dingo: then don't say "AMP" again |
| 13:12.59 | DrukenHME | jbot amp isn't supported here, people using amp should join channel #amportal |
| 13:13.13 | fugitivo | DrukenHME: yes! |
| 13:13.18 | fugitivo | ~amp |
| 13:13.20 | jbot | i heard amp is a web based interface for configuring Asterisk. See http://amp.coalescentsystems.ca/ |
| 13:13.24 | fugitivo | :/ |
| 13:13.25 | DrukenHME | bah! |
| 13:13.26 | dingo | so then back to the question simple php app simple config any gui ??? |
| 13:13.34 | DrukenHME | jbot: amp isn't supported here, people using amp should join channel #amportal |
| 13:13.44 | fugitivo | dingo: we don't support gui here |
| 13:13.49 | dingo | ok |
| 13:13.53 | DrukenHME | jbot, amp isn't supported here, people using amp should join channel #amportal |
| 13:13.55 | dingo | ill keep digging |
| 13:14.04 | fugitivo | dingo: we can help you with plain text config files |
| 13:14.04 | CPC | someone here has a simple sample about how to link 2 asterisk servers using IAX2?? |
| 13:14.07 | DrukenHME | jbot is pissing me off |
| 13:14.21 | drray_ | hey Druken |
| 13:14.24 | fugitivo | DrukenHME: i think you need to use the word "also" |
| 13:14.49 | fugitivo | i saw the intructions of that bot in some place... |
| 13:14.52 | DrukenHME | morning drray_ |
| 13:15.14 | DrukenHME | jbot forget amp |
| 13:15.15 | jbot | DrukenHME: i forgot amp |
| 13:15.20 | DrukenHME | jbot, amp isn't supported here, people using amp should join channel #amportal |
| 13:15.27 | dingo | jbot forget dingo |
| 13:15.27 | jbot | dingo: i forgot dingo |
| 13:15.36 | DrukenHME | ~amp |
| 13:15.49 | fugitivo | jbot forget your mamma |
| 13:15.49 | jbot | i didn't have anything called 'your mamma' to forget, fugitivo |
| 13:15.58 | DrukenHME | jbot amp isn't supported here, people using amp should join channel #amportal |
| 13:16.18 | DrukenHME | i think it's ignoring me :) |
| 13:16.45 | DrukenHME | hehehe |
| 13:16.49 | DrukenHME | god i'm dumb... |
| 13:16.59 | [TK]D-Fender | jbot amp is also isn't supported here, people using amp should join channel #amportal |
| 13:17.01 | jbot | okay, [TK]D-Fender |
| 13:17.05 | [TK]D-Fender | ~amp |
| 13:17.06 | jbot | rumour has it, amp is isn't supported here, people using amp should join channel #amportal |
| 13:17.07 | DrukenHME | nah... |
| 13:17.13 | fugitivo | is isn't |
| 13:17.14 | [TK]D-Fender | :p |
| 13:17.15 | DrukenHME | ya see... you used IS |
| 13:17.24 | DrukenHME | i was using ISN'T |
| 13:17.26 | DrukenHME | :) |
| 13:17.29 | [TK]D-Fender | jbot amp is also n't supported here, people using amp should join channel #amportal |
| 13:17.31 | jbot | okay, [TK]D-Fender |
| 13:17.31 | fugitivo | jbot forget amp |
| 13:17.34 | [TK]D-Fender | ~amp |
| 13:17.36 | jbot | well, amp is isn't supported here, people using amp should join channel #amportal, or n't supported here, people using amp should join channel #amportal |
| 13:17.43 | DrukenHME | jbot forget amp |
| 13:17.54 | Kidaz | Hey all, I am thinking of setting up an asterisk system at home here. I have a question. If I use one service provider to buy a PSTN number for incoming calls, can I use another (cheaper) service for outgoing PSTN calls so that the same number comes up on the receivers CID?? Like spoofing emails on SMPT servers. |
| 13:17.54 | [TK]D-Fender | jbot amp is n't supported here, people using amp should join channel #amportal |
| 13:17.56 | jbot | ...but amp is already something else... |
| 13:17.58 | fugitivo | hell |
| 13:18.05 | [TK]D-Fender | jbot amp is now n't supported here, people using amp should join channel #amportal |
| 13:18.06 | jbot | ...but amp is already something else... |
| 13:18.14 | fugitivo | jbot forget amp |
| 13:18.19 | fugitivo | damn :) |
| 13:18.21 | fugitivo | what a mess |
| 13:18.28 | DrukenHME | good going.... |
| 13:18.35 | DrukenHME | :) |
| 13:19.22 | Kidaz | I mean, if I have 2 voip services one for outgoing and one for incoming... can they both use the same PSTN number?? |
| 13:19.23 | fugitivo | Kidaz: not really, "i think" your service provider will replace your callerid info |
| 13:19.45 | *** join/#asterisk kenrstone (n=krstone@ool-4573f3dc.dyn.optonline.net) |
| 13:19.51 | [TK]D-Fender | Kidaz : Technically yes no reason it can't work like you want, but you'd have to find a provider that would allow that kind of setting. |
| 13:19.58 | wasim | Kidaz: in short, yes |
| 13:20.37 | Kidaz | Im sure Im not the only person to ask this question.... have you guys done stuff like this to save money? If so any recommended service providers? |
| 13:20.54 | [TK]D-Fender | Kidaz : I do taht with my work connection. |
| 13:21.29 | *** join/#asterisk [Airwolf] (n=airwolf@82-171-75-4.dsl.ip.tiscali.nl) |
| 13:21.52 | [TK]D-Fender | have a PRI at work with a DID that points to my home. On busy/no-anser at home it forwards to that DID which heads right back here over SIP. When my home line is busy it trunks through the office faking my home CallerID. |
| 13:22.11 | [TK]D-Fender | Kidaz : but no suggestions on who is known to provide that service. |
| 13:22.21 | [TK]D-Fender | just start calling them up and ask. |
| 13:22.54 | CPC | what happends if i configure all files to try to outbound a call and do not have a FXS board?? |
| 13:23.13 | [TK]D-Fender | ?! |
| 13:23.13 | fugitivo | CPC: what files?? |
| 13:23.34 | Kidaz | interesting approach... i have an assumption that it would be hard to get hold of knowledgeable people to answer such questions on the phone. |
| 13:23.59 | CPC | zapata.conf and others |
| 13:24.22 | fugitivo | CPC: your question is nonsense |
| 13:24.37 | [TK]D-Fender | Kidaz : Hardly. Its a really simple question : "I have another phone number which I use for incoming calls and would like you to use that as my CallerID. Is that doable?" |
| 13:25.03 | [TK]D-Fender | FXS is for PHONES, not LINES. |
| 13:26.00 | Kidaz | [TK]D-Fender, yup... i guess i should be more positive about companies |
| 13:26.01 | DrukenHME | stupidity is only allowed in this channel monday to friday, 11am to 1pm |
| 13:26.20 | CPC | i made a mistake ... |
| 13:26.24 | fugitivo | ~docs |
| 13:26.29 | jbot | it has been said that docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 13:26.29 | *** join/#asterisk coppice (n=chatzill@166.168.17.210.dyn.pacific.net.hk) |
| 13:26.46 | [TK]D-Fender | Kidaz : I've learned that service companies like an informed customer who know what they want and can express it very clearly. If you don't like what they're offering, keep looking. |
| 13:27.05 | *** join/#asterisk znoG (n=gs@33-138-114-200.fibertel.com.ar) |
| 13:27.08 | *** join/#asterisk Kernel_co (n=I@107.230.dial-up.xter.net) |
| 13:27.17 | [TK]D-Fender | CPC : Start over again... |
| 13:28.08 | Kidaz | [TK]D-Fender, I have found, what I believe to be a good UK number company eSMS, but I am looking for a company which can provide local USA numbers, any reccommendations? |
| 13:28.42 | *** join/#asterisk Cybertoy (n=maxim@dsl254-123-241.nyc1.dsl.speakeasy.net) |
| 13:29.07 | [TK]D-Fender | Kidaz : I've never had to shop for VoIP termination so I don't know the players that well. |
| 13:29.08 | CPC | huh....I need connect a PC cliente to a PSTN (just a fake)...i need to simulate that situation but I do not have the hardware to complete the transaction...i would like just to configure the files and see what the asterisk will show... |
| 13:29.16 | [TK]D-Fender | And I'm not American :) |
| 13:29.22 | Kidaz | [TK]D-Fender, ahhh ;) |
| 13:29.38 | fugitivo | CPC: start with the documentation |
| 13:29.40 | fugitivo | ~docs |
| 13:29.46 | jbot | hmm... docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 13:29.58 | [TK]D-Fender | CPC : What it will show? Not much to say... What is the point of "faking it"? |
| 13:30.08 | [TK]D-Fender | fugitivo : Man Of The ~docs ..... |
| 13:30.11 | *** join/#asterisk apardo (n=apardo@174.Red-83-33-49.dynamicIP.rima-tde.net) |
| 13:31.19 | CPC | ok i'll try to research something else...thx |
| 13:31.44 | [TK]D-Fender | CPC : But really, what does trying to simulate it do for you> |
| 13:31.54 | DrukenHME | ~amp |
| 13:32.03 | jbot | from memory, amp is NOT supported here! people using it should join #amportal |
| 13:32.11 | DrukenHME | bout fucken time! |
| 13:32.12 | *** join/#asterisk fulgas (n=fulgas@a81-84-117-84.cpe.netcabo.pt) |
| 13:32.37 | coppice | ~volt |
| 13:32.49 | jbot | well, volt is a unit of potential equal to the potential difference between two points on a conductor carrying a current of 1 ampere when the power dissipated between the two points is 1 watt; equivalent to the potential difference across through a resistance of 1 ohm when 1 ampere of current flows it. |
| 13:32.49 | [TK]D-Fender | CPC : If you're just starting, lets say that * works, and can probably do whatever you're looking for it to do. Is it that you want to see if your configuration would route calls the way you would like before making a purchase of equipment? |
| 13:33.23 | CPC | sure |
| 13:34.12 | CPC | its just a college taks, but its important to us :) |
| 13:34.41 | [TK]D-Fender | CPC : well as everyone here will say : "it works". Just set up your dialplan to do things like "Saydigits(${EXTEN:1}) in place of the actual Dial commands so you know what its really dialing. And NoOp's to show messages in the CLI and just watch the call's progress. |
| 13:34.41 | DrukenHME | [TK]D-Fender: your not a yank? what are you? |
| 13:34.58 | [TK]D-Fender | Canadian :) |
| 13:35.01 | CPC | me and my friend r novice in this subject..and do not have enough time ..we guess :) |
| 13:35.09 | DrukenHME | piss off... where? |
| 13:35.15 | [TK]D-Fender | Montreal |
| 13:35.21 | DrukenHME | aww... |
| 13:35.31 | DrukenHME | Barrie :) |
| 13:35.42 | [TK]D-Fender | Forget where that it in ON... |
| 13:35.46 | [TK]D-Fender | is* |
| 13:35.55 | DrukenHME | an hour north of toronto |
| 13:36.03 | DrukenHME | up 400 |
| 13:36.03 | fugitivo | [TK]D-Fender: he was using AMP... |
| 13:36.17 | DrukenHME | ~amp |
| 13:36.30 | jbot | extra, extra, read all about it, amp is NOT supported here! people using it should join #amportal |
| 13:36.31 | DrukenHME | :) |
| 13:36.46 | [TK]D-Fender | fugitivo : Must have missed that. Is this where one would say "sod off?" ;) Nahh.. I try to be nice about it and get them to understand the error of their ways... |
| 13:36.48 | fugitivo | amp logic is not real asterisk logic |
| 13:37.02 | fugitivo | [TK]D-Fender: that's why i was the "~docs guy" ;) |
| 13:37.37 | DrukenHME | since when do canadians say sod off? i've never said that in my life.... |
| 13:37.40 | fugitivo | the problem with amp is that it doesn't teach people how asterisk works, so when trying to touch plain text files, they'll frustrated |
| 13:37.54 | [TK]D-Fender | AMP logic IS Asterisk logic. Just made to their buggy cookie-cutter specification which is implemented in a nearly unreadable manner :) |
| 13:38.22 | fugitivo | i think i don't know how to use AMP, it doesn't look like asterisk logic |
| 13:38.37 | [TK]D-Fender | Druken : I've picked up bits of every culture, that was Brit... |
| 13:39.11 | [TK]D-Fender | fugitivo : It's written messy, and poorly documented in the dialplan. It does sorta work though. |
| 13:39.13 | *** join/#asterisk Drukenlpy (i=Druken@CPE00121716da99-CM000e5cde4ca2.cpe.net.cable.rogers.com) |
| 13:39.35 | fugitivo | too many includes, contexts and macros, impossible to read :) |
| 13:40.27 | *** join/#asterisk backblue (n=moo@87-196-6-187.net.novis.pt) |
| 13:40.34 | Drukenlpy | fugitivo: don't ever try to read my configs then :) hehehe |
| 13:40.48 | fugitivo | too complex? |
| 13:40.53 | fugitivo | or messy? |
| 13:40.58 | Drukenlpy | even for my brain sometimes |
| 13:41.19 | Drukenlpy | gives me headaches when i have to change things |
| 13:41.52 | fugitivo | lol |
| 13:42.07 | Drukenlpy | fugitivo: complex and messy, lil of both |
| 13:42.36 | Drukenlpy | and absolutly no commenting :) |
| 13:44.33 | *** join/#asterisk RoyK (n=roy@ti200720a080-0856.bb.online.no) |
| 13:44.40 | shawn | hi all |
| 13:44.57 | [TK]D-Fender | Mine doesn't have a lot of comments, but its written rather cleanly and spacing is key. |
| 13:45.01 | Drukenlpy | so [TK]D-Fender: you want to give me a did?? :) |
| 13:45.09 | fugitivo | when i reach that point, i delete everything and do it again :) |
| 13:45.28 | fugitivo | yes, spacing is the key |
| 13:45.42 | Drukenlpy | fugitivo: oh god... no... i don't think i'd be able to rewrite mine... i'd be lost.. hehehe |
| 13:45.48 | fugitivo | but i comment almost everything |
| 13:45.59 | *** join/#asterisk pigpen2 (n=mark@fw.seamans.cc) |
| 13:46.20 | [TK]D-Fender | Druken : I could.... but its an exposure question. |
| 13:46.21 | RoyK | zoa: ping |
| 13:46.54 | Drukenlpy | [TK]D-Fender: exposure question ? |
| 13:47.11 | *** join/#asterisk thymian (n=guest@dslb-084-056-160-159.pools.arcor-ip.net) |
| 13:47.22 | shawn | when i load res_php.so ,the asterisk give warning :Warning, flexibel rate not heavily tested! and then hint: error while writing audio data::broken pipe. what 's wrong |
| 13:47.24 | thymian | hi / moin |
| 13:48.14 | shawn | wasim , can u help me? |
| 13:48.35 | tzanger | res_php? oh lord, it's started |
| 13:48.52 | shawn | tzanger ,yes |
| 13:48.53 | *** join/#asterisk Tili (i=Tili@202-133-67-30-dialup.sat.net.pk) |
| 13:49.06 | shawn | i try to use php_asterisk |
| 13:49.07 | Drukenlpy | wtf is res_php ? |
| 13:49.18 | tzanger | shawn: it *sounds* like one end hung up, so the latter part of that message is just informational |
| 13:49.25 | tzanger | and variable rate, are you playing with some kind of new codec? |
| 13:49.53 | shawn | no . do you know php_asterisk? |
| 13:49.59 | Drukenlpy | the variable rate is USUALLY a mp3 thing... isn't it? |
| 13:50.09 | tzanger | yeah this is true |
| 13:50.35 | shawn | tzanger ,but there is nothing with mp3 i just want to use php application |
| 13:50.53 | fugitivo | "Asterisk PHP allows you to control the dial-plan and write applications for Asterisk in PHP. This is faster and more flexible than phpAGI." |
| 13:50.57 | *** join/#asterisk RoyKa (n=roy@c51008425.inet.catch.no) |
| 13:51.05 | tzanger | fugitivo: yeah, but it's php, man... <shudders> |
| 13:51.10 | fugitivo | yes |
| 13:51.12 | tzanger | what next, voice fragments inside the PHP Script? :-) |
| 13:51.14 | thymian | hi i read a lot of documentation, faqs and howtos but can't figure out which approach is the recommended one for running asterisk with an avm fritz card (a1 v2.0) on debian sarge. either the old 2.4.27-2 kernel with selfcompiled avm fcpci drivers and chan_capi, but the capiutils are missing on my apt-deb server. or the modular approach with 2.6 and the mISDN drivers via chan_misdn. Any voice will be heard and part of my decision |
| 13:51.14 | thymian | <PROTECTED> |
| 13:51.19 | shawn | yes ,fugitivo you are rihgt |
| 13:51.38 | fugitivo | i don't know why people is using a programming language designed for web |
| 13:51.40 | tzanger | thymian: unfortunately I have no idea -- I have never used an AVM card. |
| 13:52.00 | tzanger | fugitivo: well as much as I do dislike it, php has become more and more general |
| 13:52.05 | tzanger | but still.. <shudders> :-) |
| 13:52.10 | fugitivo | i like perl for that kind of tasks |
| 13:52.13 | RoyK | ~seen zoa |
| 13:52.24 | jbot | zoa is currently on #asterisk (22h 8m 41s). Has said a total of 57 messages. Is idling for 40m 52s, last said: 'ela'. |
| 13:52.24 | {zombie} | thymian: I've had success using the fcpci drivers, but I've been told chan_misdn is much better |
| 13:52.27 | thymian | tzanger, is there any difference in features between chan_capi and chan_msidn ? |
| 13:52.32 | {zombie} | you gain ptp support etc |
| 13:52.35 | coppice | some people have an unreasonable bias against php |
| 13:52.45 | Drukenlpy | fugitivo: uhmm... isn't perl for web as well? |
| 13:52.45 | tzanger | thymian: again, I have never used ISDN BRI. Lots of PRI but that's a different beast :-) |
| 13:52.59 | shawn | fugitivo: do you use php_asterisk? |
| 13:53.01 | thymian | tzanger, ack |
| 13:53.08 | fugitivo | Drukenlpy: not really, it wasn't designed for web, but you can use it as cgi |
| 13:53.13 | tzanger | coppice: I am one of those people. :-) My bias stems from every single piece of PHP code I"ve ever seen or tried to modify being messier than obfuscated Perl. |
| 13:53.16 | fugitivo | shawn: no |
| 13:53.26 | thymian | zombie, which kernel are you running ? |
| 13:53.40 | shawn | how do u think of it? |
| 13:53.42 | tzanger | coppice: mind you, I am no perl bigot either; I can code in it but it's very difficult for me to retain my perl knowlege, I find myself relearning what I did months later. |
| 13:53.43 | {zombie} | I think that box was running 2.4.25 |
| 13:53.59 | {zombie} | no wait, i have it on 2.4.31 as well |
| 13:53.59 | fugitivo | tzanger: that's true |
| 13:54.02 | shawn | fugitivo : how do u think of it? |
| 13:54.03 | thymian | zombie, thx |
| 13:54.04 | fugitivo | tzanger: it happens to me too |
| 13:54.18 | fugitivo | shawn: i'm talking about general php |
| 13:54.36 | thymian | anyone has looked at the php_asterisk code ? |
| 13:54.42 | tzanger | fugitivo: :-) I mean I like it, but I just can't seem to keep it in my head. ANd Perl6 has just turned me completely off of Perl. My current study is Python, but I dispise its forced indentation |
| 13:54.52 | thymian | *g* |
| 13:55.05 | tzanger | however it seems very solid |
| 13:55.35 | fugitivo | tzanger: i didn't learn python "yet" :) i want to try it, but no time to play with it |
| 13:55.57 | fugitivo | tzanger: meanwhile I stick with perl and php for web |
| 13:56.13 | tzanger | fugitivo: well we run an exchange4linux server, and that's written entirely in python. hell the outlook connector is too. I need to write a better IMAP server for it :-) |
| 13:56.39 | Kernel_Core | hi all |
| 13:56.44 | fugitivo | tzanger: exchange4linux? is that like exchange from microsoft? |
| 13:56.47 | tzanger | I was good at PHP and learnign Perl and wondered why I was trying to keep up with two very similar languages... so I dropped PHP because Perl was much more rounded and general. Now I'm kind of doing the same with Python and Perl |
| 13:56.56 | tzanger | fugitivo: www.exchange4linux.org |
| 13:56.58 | tzanger | very very good |
| 13:57.06 | tzanger | best I've runa cross, and I've tried damn near everything |
| 13:57.17 | shawn | fugitivio: you mean you are using phpagi |
| 13:57.27 | tzanger | there is something a little eerie about running arbitrary SQL queries on your email, contacts, todos and such :-) |
| 13:57.27 | {zombie} | e4l is good when it works, but when it doesn't there's just no debug info to help find out what's wrong |
| 13:57.35 | fugitivo | shawn: no, I DON'T use php, only for WEB |
| 13:57.57 | shawn | ok i c fugitivo |
| 13:58.07 | tzanger | {zombie}: true, but with full source to the server it's not terrible to add it in |
| 13:58.15 | fugitivo | tzanger: does it have all ms exchange functionalities? |
| 13:58.20 | tzanger | {zombie}: we've got a 34G postgres database with all our crap in it |
| 13:58.28 | {zombie} | yeah but there's no debug info from the client |
| 13:58.32 | tzanger | fugitivo: seems to; there is a newer version we haven't moved to yet |
| 13:58.41 | fugitivo | GPL? |
| 13:58.41 | Kernel_Core | I have x100p clone , when I call to asterisk it answers but when I dial desired extension , I get this error : -- Starting simple switch on 'Zap/1-1' |
| 13:58.41 | Kernel_Core | <PROTECTED> |
| 13:58.41 | Kernel_Core | <PROTECTED> |
| 13:58.41 | Kernel_Core | <PROTECTED> |
| 13:58.41 | Kernel_Core | <PROTECTED> |
| 13:58.42 | Kernel_Core | <PROTECTED> |
| 13:58.45 | tzanger | {zombie}: the e4l people seem very receptive to that kind of input |
| 13:58.52 | fugitivo | ~pastebin |
| 13:58.56 | jbot | extra, extra, read all about it, pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
| 13:59.11 | {zombie} | I have one site where the client just won't connect, and they just weren't very helpful on the mailing list |
| 13:59.15 | tzanger | {zombie}: hell I suggested a new e4l taskbar icon and they adopted it, although I think it looks stupider now, I need to try again :-) |
| 13:59.32 | fugitivo | tzanger: do you use outlook clients only? |
| 13:59.34 | tzanger | {zombie}: no the mailing list is really bad, I just contact Helmuth directly and CC the list, that seems to work a lot better |
| 13:59.39 | fugitivo | tzanger: does it work with kontact, evolution? |
| 13:59.41 | tzanger | fugitivo: except for me, which is KOntact :-) |
| 13:59.44 | Kernel_Core | here is my conf files http://pastebin.com/494877 |
| 13:59.53 | tzanger | fugitivo: well the IMAP server needs a lot of work, which is what I'm learning Python for |
| 13:59.55 | fugitivo | tzanger: yeah, kontact is cool |
| 14:00.03 | {zombie} | oh, it works with kontact now? |
| 14:00.27 | tzanger | {zombie}: no, not really |
| 14:00.43 | tzanger | it crashed out the IMAP server |
| 14:00.55 | tzanger | but I mean meeting requests and whatnot that come in work just fine |
| 14:01.02 | tzanger | but that's more an outlook and kontact thing than an e4l thing |
| 14:01.10 | fugitivo | is it compatible with exchange or does it use only "standard" services? |
| 14:01.35 | shawn | Drukenlpy: have you any ideas for my question? |
| 14:02.16 | Kernel_Core | what is wrong with my conf ? |
| 14:02.29 | tzanger | fugitivo: no outlook has no idea it's nto talking to Exchange Server |
| 14:02.58 | tzanger | its architecture is a little cracked out but I can't deny that it works |
| 14:03.07 | fugitivo | that's great |
| 14:03.12 | {zombie} | tzanger: you're using the complete distro then? not as an addon to your existing linux server? |
| 14:03.16 | fugitivo | can you share calendars? |
| 14:03.23 | fugitivo | and contacts to other users? |
| 14:03.28 | tzanger | basically you pull your email in through POP/IMAP and send through SMTP. The E4L connector then synchronizes your OUtlook data with the E4L server in the background. Full archiving and everythign works |
| 14:03.41 | tzanger | and then your global stores and shared folders are all done in E4L though the connector |
| 14:03.47 | fugitivo | great |
| 14:03.50 | tzanger | delegation, resource allocation, etc. is all handled |
| 14:03.57 | fugitivo | i have to try it |
| 14:04.01 | fugitivo | hmm, i'll need another server |
| 14:04.09 | coppice | tzanger: "its architecture is a little cracked out " sounds like exchange. but "I can't deny that it works" appears to mean you are talking about the other software |
| 14:04.21 | tzanger | yes, it's not bad for price either, and having your SERVER 100% open source and your data in an open format database is great :-) |
| 14:04.22 | fugitivo | tzanger: and you use postgresql to store data? |
| 14:04.48 | {zombie} | yeah that is rather funky |
| 14:05.03 | tzanger | coppice: hahaha well Exchange is a traditional server/client model. E4L is more of a standalone client that syncs with a central server model |
| 14:05.08 | tzanger | fugitivo: correct |
| 14:05.14 | {zombie} | I wrote a query that uses the e4l contact database to do a reverse number lookup on incoming calls on asterisk :) |
| 14:05.23 | {zombie} | (which just brought this conversation back on topic ;) |
| 14:05.28 | tzanger | 4000+ outlook contacts, unbelievable amoutns of email including video attachments |
| 14:05.37 | fugitivo | nice |
| 14:05.47 | tzanger | {zombie}: :-) We do something simialr with our after-hours pager that looks in a shared calendar to find out who is on call |
| 14:06.37 | tzanger | the queries get big fast because there is a lot of table linking to do to get what you want :-) |
| 14:06.44 | {zombie} | yeah |
| 14:08.15 | *** join/#asterisk A-Tuin|work (n=A-Tuin@nat.office.legend.net.uk) |
| 14:08.26 | tzanger | fugitivo: E4L works "more" than the other solutions because it doesn't try to interpret the Outlook MAPI data; it just stores it. THere are a lot of well defined and described MAPI tags, but I think they are outnumbered by the undocumented "MAPI_______0x325252" type MAPI tags. That is where the other solutions fail, they try to either ignore them or interpret them. E4L just stores them and gives them back to Outlook as requested. |
| 14:09.25 | fugitivo | no complains from outlook users? |
| 14:09.45 | tzanger | there are a couple of interesting MAPI explorer plugins for outlook that let you see the MAPI ojects and try to make sense of them, but it's harsh :-) |
| 14:10.18 | tzanger | fugitivo: our server dies the odd time (out of memory) but that's not E4L's fault, and we're using an older version too which has known memory leaks. We have a new server ready but just ahven't had time to move it |
| 14:10.58 | tzanger | you want to give it a LOT of memory. For us (50 users, enormous contacts and shared folders) we're speccing 2G and a Xeon 3.02GHz, but we'd likely give it 4G before we're done |
| 14:11.06 | tzanger | I want to see what the new version is like on memory first |
| 14:12.22 | fugitivo | hmmm, that's a lot of machine compared to the requirements of a ms exchange |
| 14:12.41 | tzanger | yes |
| 14:12.50 | tzanger | our old Exchange Server was a P90 with 64M of memory |
| 14:13.11 | fugitivo | why is that? because it's written in python? |
| 14:13.17 | tzanger | now it didn't work perfectly either but then again it wasn't running compiled Python code, Apache and a big Postgres database |
| 14:13.57 | tzanger | I think the postgres is the big hog on memory, and E4L itself the CPU |
| 14:14.10 | tzanger | but again we're running an older version which has a lot of cruft for supporting two distinct E4L versions |
| 14:14.29 | fugitivo | yes, postgresql takes a lot of memory |
| 14:15.12 | tzanger | and no matter what you give it it likes more :-) |
| 14:15.31 | fugitivo | must it work with a database or you can use the hard drive to store data? |
| 14:16.42 | tzanger | nope it is reliant on postgres |
| 14:17.14 | tzanger | it'd be nice if it broke out attachments, md5'd them and stored only one copy rather than putting them in the DB as LOs but that's a future rev |
| 14:20.24 | *** join/#asterisk MikeJ[Laptop] (n=jayden@d14-69-8-30.try.wideopenwest.com) |
| 14:24.08 | *** join/#asterisk javar (n=javar@Dynamic-IP-cr20011865115.cable.net.co) |
| 14:24.55 | *** join/#asterisk javar (n=javar@Dynamic-IP-cr20011865115.cable.net.co) |
| 14:27.53 | Drukenlpy | wuts reliant on postgres? |
| 14:28.03 | *** join/#asterisk tsetane (n=tsetane@pppoecl69000.minlos.no) |
| 14:36.19 | tzanger | Drukenlpy: exchange4linux |
| 14:39.33 | *** join/#asterisk jahani (n=k@adsl196-206-241-217-196.adsl196-16.iam.net.ma) |
| 14:41.44 | tzafrir_laptop | Corydon76, I tend to update my bug report just as you add a comment... |
| 14:45.21 | *** join/#asterisk kashmish_ (n=kashmish@m1.ince.net) |
| 14:46.54 | tzafrir_laptop | and it's using a proprietary plug-in on the clients. |
| 14:47.01 | tzafrir_laptop | Not fun |
| 14:50.42 | backblue | anyone knows something about attended tranfers from *? |
| 14:53.45 | ManxPower | backblue, it's easiest if you use the PHONE' |
| 14:54.23 | ManxPower | backblue, it's easiest if you use the PHONE'S transfer features. If your phone is too stupid to do that, then you might consider the Asterisk transfer features (a hack IMNSHO) |
| 14:54.43 | ManxPower | Usually "stupid phones" are an indication of a "cheap ass user" |
| 14:55.07 | backblue | ManxPower: i sayed * tranfers, not phone transfer :P |
| 14:55.55 | Drukenlpy | stupid user? |
| 14:56.03 | ManxPower | backblue, that's only available on 1.2, see features.conf and the wiki. |
| 14:56.04 | backblue | ManxPower: stupid facilitys makes big softwares, from stupid, or just one more software. |
| 14:56.08 | *** join/#asterisk Kidazzy (n=kidaz@thedowds.demon.co.uk) |
| 14:56.20 | ManxPower | The mailing list archves will have additional information |
| 14:56.46 | backblue | ManxPower: i have un-attended transfer, but attended i dont. i will take a look again tks. |
| 14:56.52 | *** join/#asterisk dorphalsig (n=dorphals@200.106.223.5) |
| 14:56.55 | dorphalsig | hello |
| 14:57.02 | dorphalsig | Me Again =) |
| 14:57.21 | *** join/#asterisk Kidaz|Real (n=kidaz@thedowds.demon.co.uk) |
| 14:57.50 | dorphalsig | Once again, for those of you who havent seen me bugging everyone here lately... |
| 14:58.25 | dorphalsig | I am running a * 1.2 on a T410P with one PRI E1 on span 1 |
| 14:58.32 | dorphalsig | and 2 channelbanks on spans 2 and 3 |
| 14:58.34 | *** join/#asterisk N9URK (n=icechat5@user-0ce2dhc.cable.mindspring.com) |
| 14:58.42 | dorphalsig | its been running for a couple of days now |
| 14:58.44 | N9URK | happy saturday everyone |
| 14:58.56 | dorphalsig | but every couple of hours the lines go dead |
| 14:59.02 | fugitivo | dorphalsig: what channelbanks? |
| 14:59.20 | dorphalsig | hey fugitivo ... (its me the same guy that sometimes comes on as Chiardon) |
| 14:59.28 | fugitivo | hi chiardon :) |
| 14:59.41 | fugitivo | i see you're making progress with your asterisk |
| 14:59.42 | backblue | ManxPower: i found it. tks |
| 14:59.48 | dorphalsig | indeed |
| 14:59.55 | *** join/#asterisk mtgh (n=chatzill@dsl093-001-038.det1.dsl.speakeasy.net) |
| 14:59.56 | backblue | one more thing, i cant make a transfer for a sip url? only extensions? |
| 14:59.57 | dorphalsig | yet, it keeps going down for some reason |
| 15:00.07 | *** join/#asterisk Seyr (n=kaladin@cpe-67-10-139-141.houston.res.rr.com) |
| 15:00.09 | dorphalsig | I look in asterisk/messages |
| 15:00.13 | fugitivo | dorphalsig: any message on the CLI? |
| 15:00.23 | dorphalsig | and I get these errors allways before it falls down |
| 15:00.24 | dorphalsig | Jan 6 20:09:35 WARNING[2328] chan_zap.c: zt hook failed: Device or resource busy |
| 15:00.24 | dorphalsig | Jan 7 04:02:31 WARNING[2327] chan_zap.c: No D-channels available! Using PrimarJan 7 el 16 as D-channel anyway! |
| 15:00.40 | fugitivo | dorphalsig: pastebin your zaptel.conf |
| 15:00.57 | N9URK | HI all, I have had asterisk running for a week or so and then today it gave me this error "Unable to connect to remote asterisk (does /var/run/asterisk.ctl exist?)" when I was trying to reload. Did I screw something up? |
| 15:00.59 | fugitivo | it seems you don't have a dchannel specified |
| 15:01.10 | Seyr | I have a Wildcard TE411P (nothing is plugged in it) and I just compiled Zaptel for it and recompiled Asterisk, now I have no audio. Anyone have any ideas? |
| 15:01.41 | N9URK | It won't let me start asterisk, I rebooted the machine |
| 15:01.51 | *** join/#asterisk ReD-MaN (i=redman@dhcp-0-2-b3-9a-4a-5b.cpe.quickclic.net) |
| 15:01.53 | N9URK | and nothing shows in ps -aux for * |
| 15:02.01 | tzafrir_laptop | N9URK, ps aux | grep asterisk |
| 15:02.51 | tzafrir_laptop | N9URK, so your asterisk fails to start. Do you run asterisk as a user or as root? |
| 15:02.58 | N9URK | root |
| 15:03.13 | tzafrir_laptop | asterisk -vvvvvvvvvvv |
| 15:03.23 | tzafrir_laptop | what are its last words? |
| 15:03.24 | fugitivo | when is asterisk going to run as user by default? |
| 15:03.37 | dorphalsig | fugitivo --> http://pastebin.ca/36099 |
| 15:03.43 | tzafrir_laptop | fugitivo, when you install the Debian packages |
| 15:03.58 | fugitivo | tzafrir_laptop: i mean "default" |
| 15:04.03 | fugitivo | tzafrir_laptop: no packages |
| 15:04.15 | N9URK | <PROTECTED> |
| 15:04.15 | N9URK | Yuck! Error in buffer handling...: Connection reset by peer |
| 15:04.15 | N9URK | Warning, flexible rate not heavily tested! |
| 15:04.15 | N9URK | Yuck! Error in buffer handling...: Connection reset by peer |
| 15:04.16 | dorphalsig | there should be asterisk rpms :P |
| 15:04.19 | dorphalsig | lol |
| 15:04.36 | tzafrir_laptop | fugitivo, it takes some settings, that are not exactly trivial |
| 15:04.57 | fugitivo | tzafrir_laptop: i know what it takes, i run it as user, i mean "default" like any other serious service |
| 15:05.18 | *** join/#asterisk anonymouz666 (n=lynx@200.218.193.6) |
| 15:05.24 | tzafrir_laptop | fugitivo, what's the bug number for that? |
| 15:05.51 | fugitivo | tzafrir_laptop: is that a bug or a feature? :) |
| 15:06.06 | tzafrir_laptop | feature request, I believe |
| 15:06.07 | Seyr | Anyone know why I would lose sound afetr installing Zaptel? |
| 15:06.40 | tzafrir_laptop | no audio where? did you use zaptel before that? |
| 15:06.46 | Seyr | No |
| 15:06.59 | N9URK | thanks for the help there tzafrir |
| 15:07.00 | tzafrir_laptop | Seyr, does Milliwatt() still play a sound? |
| 15:07.09 | Seyr | new server, had inbound number going to MeetMe, which played sound but gave errors (no zaptel) |
| 15:07.12 | N9URK | I guess it doesn't like one of my music on hold dirs |
| 15:07.19 | Seyr | just got around to installing zaptel and now lost sound |
| 15:07.23 | N9URK | strange |
| 15:07.59 | tzafrir_laptop | N9URK, do you use native MOH? |
| 15:08.13 | N9URK | you mean the MOH with mpg123? |
| 15:08.17 | tzafrir_laptop | N9URK, what version of Asterisk? |
| 15:08.28 | N9URK | 1.2.1 |
| 15:10.04 | Seyr | tzafrir: Milliwatt is silence as well |
| 15:11.46 | N9URK | tzafrir->does that mean that one of the mp3's is corrupted or something? |
| 15:12.26 | *** join/#asterisk Assid (n=assid@203.115.64.14) |
| 15:16.13 | *** join/#asterisk mhnoyes (n=mhnoyes@user-2ivfk3s.dialup.mindspring.com) |
| 15:17.28 | *** join/#asterisk mhnoyes_ (n=mhnoyes@user-2ivfnjo.dialup.mindspring.com) |
| 15:19.25 | *** join/#asterisk Seyr (n=Seyr@cpe-67-10-139-141.houston.res.rr.com) |
| 15:20.09 | Seyr | sorry tzafrir, lost connection. Milliwatt was silence as well |
| 15:20.16 | Seyr | nothing, not even playback works |
| 15:20.28 | *** join/#asterisk kenrstone (n=krstone@ool-4573f3dc.dyn.optonline.net) |
| 15:20.40 | Seyr | Everything was playing sound until I installed Zaptel |
| 15:22.16 | Seyr | Ok, if I "chkcofnig zaptel off" and reboot, sound works |
| 15:22.23 | Seyr | but if I enable Zaptel, no sound |
| 15:22.45 | Seyr | I'm only using wct4xxp |
| 15:24.54 | *** join/#asterisk RoyK (n=roy@ti200720a080-0856.bb.online.no) |
| 15:25.39 | tzafrir_laptop | what type of sound? through what channel do you play? |
| 15:25.58 | Seyr | just using SIP right now |
| 15:26.02 | Seyr | playback of any sound file |
| 15:26.55 | Seyr | Before I did zaptel, i call in a 800 number that goes to the box via SIP and I have it going straight into meetme, and it played fine, just get the /dev/zap errors |
| 15:26.57 | Seyr | no biggie |
| 15:27.10 | *** part/#asterisk thazza (n=thazza@229.9.233.220.exetel.com.au) |
| 15:27.24 | tzafrir_laptop | strange. If milliwatt would have played something, I would have suspected a timing source issue. But anyway, try running 'zttest -v' when zaptel is used |
| 15:27.24 | Seyr | So today I compiled Zaptel and loaded the module for the PRI card I have (although nothing is plugged in it), just for timing and i lost all audio |
| 15:28.01 | Seyr | I disable Zaptel and sound plays fine |
| 15:28.40 | tzafrir_laptop | is "zttest -v" stuck? or does it give numbers that are 100% or close? |
| 15:28.43 | Seyr | if this were a 2.4 kernel, i'd just use ztdummy, but they wanted 2.6 on that box... *sigh* |
| 15:28.52 | *** join/#asterisk thazza (n=thazza@229.9.233.220.exetel.com.au) |
| 15:29.02 | tzafrir_laptop | kernel 2.6 has an even better ztdummy |
| 15:29.04 | *** join/#asterisk FastJack_ (i=fastjack@p5091C47C.dip.t-dialin.net) |
| 15:29.51 | Seyr | i've had no luck with ztdummy on 2.6 |
| 15:29.58 | Seyr | always echos or sounds like crap |
| 15:30.17 | Seyr | i've moved all 5 of my * boxes back to 2.4 |
| 15:30.32 | tzafrir_laptop | Seyr, the output of zttest, please? |
| 15:30.44 | Seyr | getting there, had to restart zaptel |
| 15:31.19 | tzafrir_laptop | if you have a card you don't need ztdummy, anyway. With a T1 connection you probably need to be syncd from the remote connection. |
| 15:31.34 | Seyr | theres a card, but its not being used |
| 15:31.55 | Seyr | there were gonna get a PRI, but they are just connecting to another system |
| 15:32.49 | tzafrir_laptop | It should be a timing source just as well, I believe. Is its span set as "0" in the timing source in zaptel.conf? |
| 15:34.07 | Seyr | everything is commented out, except location |
| 15:36.09 | Seyr | zttest is just sitting there |
| 15:36.14 | Seyr | [root@astrx zaptel-1.2.1]# ./zttest -v |
| 15:36.14 | Seyr | Opened pseudo zap interface, measuring accuracy... |
| 15:36.42 | Seyr | I've never configured a card in a box before, so I could be missing something. i've always used ztdummy |
| 15:37.29 | Seyr | I RTFM: http://www.voip-info.org/wiki-Asterisk+Zaptel+Installation |
| 15:37.54 | tzafrir_laptop | Seyr, grep ^span /etc/zaptel.conf |
| 15:38.27 | Seyr | all lines are commented out |
| 15:38.28 | tzafrir_laptop | zttest doesn't read anything from the timing source |
| 15:38.31 | *** join/#asterisk gnosys (n=gnosys@ip68-9-201-250.ri.ri.cox.net) |
| 15:38.33 | [TK]D-Fender | tzafrir_laptop : span 1,1,0 |
| 15:38.49 | *** part/#asterisk mhnoyes_ (n=mhnoyes@user-2ivfnjo.dialup.mindspring.com) |
| 15:38.52 | [TK]D-Fender | To use as timing coming from telco |
| 15:39.06 | Seyr | I have no lines plugged into the card |
| 15:39.17 | tzafrir_laptop | Actually: |
| 15:39.20 | Seyr | and will have no lines plugged in it |
| 15:39.36 | *** join/#asterisk brockj49464 (n=brockj49@63.87.56.159) |
| 15:39.38 | *** join/#asterisk imran (n=codentes@cpe-68-206-53-16.houston.res.rr.com) |
| 15:39.46 | imran | Hi everyone! |
| 15:40.03 | *** part/#asterisk imran (n=codentes@cpe-68-206-53-16.houston.res.rr.com) |
| 15:40.06 | tzafrir_laptop | span=1,0,0,<framing>,<coding> |
| 15:40.21 | [TK]D-Fender | first 0 in there should be a "1" to take timing from telco... |
| 15:40.43 | brockj49464 | is there a way in * to lookup a PSTN number and see if it is on an open sip provider? |
| 15:40.46 | tzafrir_laptop | span=1,0,0,esf,ami |
| 15:41.35 | Seyr | So if I am not using the telco for timing and all I have is one wct4xxp card, what would I put? |
| 15:41.35 | tzafrir_laptop | you don't connect to anybody on the other side, so you're the timing source. Hence the second value is "0" and not "1" |
| 15:41.49 | Seyr | ah |
| 15:42.22 | tzafrir_laptop | After editing zaptel.conf , run ztcfg to apply it |
| 15:42.28 | Seyr | I'll try that, thanks. Just took the wct4xxp out and put ztdummy in and rebooting, to see if I still lose sound with ztdummy |
| 15:42.36 | Seyr | thanks |
| 15:44.27 | gnosys | what causes the "boop, boop" tones that sound in my ear when I get an incoming call from my * server while I'm already on a line? Is that callwaiting=yes in zapata.conf? |
| 15:45.47 | fugitivo | yes |
| 15:46.04 | fugitivo | that's call waiting |
| 15:46.12 | fugitivo | you can get the other call pressing hook or flash |
| 15:47.52 | gnosys | Is there a way to revise that behavior slightly, and have it only happen under certain circumstances? Say, when I'm on a certain phone or on a certain line on a certain phone. I have visual indications on some phones of an incoming call, and don't want or need that boop boop tone with them, but would like to keep in on other phones which have no visual cues. |
| 15:48.05 | Seyr | tzafrir_laptop: ZT_SPANCONFIG failed on span 1: Invalid argument (22) |
| 15:48.20 | Seyr | tzafrir_laptop: using: span=1,0,0,esf,ami |
| 15:48.32 | *** join/#asterisk MatsK (i=enforcer@c83-253-29-22.bredband.comhem.se) |
| 15:48.39 | tzanger | gnosys: I don't know of any way to do that at this pint |
| 15:48.52 | tzanger | you could have callwaiting turned off and then just rely on the visual hints |
| 15:49.01 | gnosys | thanks tzanger |
| 15:49.03 | tzanger | and then just put the party you're with on hold |
| 15:49.09 | tzanger | or park them and pick up the line |
| 15:49.18 | fugitivo | gnosys: are those fxs or ip phones? |
| 15:49.29 | tzanger | Seyr: esf,ami? that's unusual |
| 15:49.45 | *** join/#asterisk redman (i=redman@207.210.20.97) |
| 15:49.51 | gnosys | Well, the phones themselves are IP phones, but the lines are PSTN lines (fxo lines). |
| 15:49.55 | tzafrir_laptop | tzanger, what's the usual? |
| 15:50.07 | tzanger | ami coding with esf |
| 15:50.12 | tzafrir_laptop | (for a T1 span) |
| 15:50.20 | gnosys | why do you ask fugitivo? |
| 15:50.21 | tzanger | I didn't say invalid, just unusual |
| 15:50.34 | tzanger | almost every T1 these days is b8zs |
| 15:50.45 | tzanger | since AMI doesn't work well with long strings of 0s |
| 15:50.53 | *** join/#asterisk MatsK (n=mk@c83-253-29-22.bredband.comhem.se) |
| 15:51.33 | tzanger | AMI is usually associated with *old* T1s or T1 equipment which also uses D4 or SF |
| 15:51.40 | tzanger | ESF links are almost always b8zs |
| 15:51.46 | *** join/#asterisk imran (n=codentes@cpe-68-206-53-16.houston.res.rr.com) |
| 15:52.33 | *** join/#asterisk af_ (n=af@ip-142-84.sn1.eutelia.it) |
| 15:55.09 | Seyr | span=1,0,0,esf,b8zs gives the same error, ZT_SPANCONFIG failed on span 1: Invalid argument (22) |
| 15:55.43 | RoyK | zoa: ding? |
| 15:56.04 | RoyK | span=9,1,1,wtf,stfu |
| 15:56.45 | coppice | RoyK: does stfu suppress error reporting? |
| 15:57.10 | RoyK | i beleive so, yes |
| 15:57.26 | tzanger | Seyr: you're just guessing |
| 15:57.34 | tzanger | Seyr: what do you have, and what is it expecting |
| 15:58.04 | Seyr | tzanger: Well, kinda guessing... i'm reading the stuff above where span goes |
| 15:58.19 | Seyr | "The framing is one of "d4" or "esf" for T1 or "cas" or "ccs" for E1", "The coding is one of "ami" or "b8zs" for T1 or "ami" or "hdb3" for E1" |
| 15:58.32 | tzanger | Seyr: what card do you have, and what are you connecting it to? |
| 15:59.08 | Seyr | tzanger: TE411P, and its connecting to nothing |
| 15:59.20 | Seyr | was just going to use it for timing, instead of ztdummy |
| 15:59.31 | Seyr | but when I have zaptel load the module, i lose sound |
| 15:59.41 | tzanger | damn, you've got a $2100 card lying around just for timing? |
| 15:59.56 | Seyr | using: wct4xxp |
| 16:00.27 | *** part/#asterisk kenrstone (n=krstone@ool-4573f3dc.dyn.optonline.net) |
| 16:00.32 | Seyr | If I unload wct4xxp and use ztdummy, I get my sound back |
| 16:01.10 | Seyr | tzanger: they may drop a PRI to the server at a later date, which is why they got the card |
| 16:01.57 | tzanger | Seyr: grep -v '^;' /etc/zaptel.conf and paste the output to pastebin.ca |
| 16:01.59 | tzanger | and give us the link |
| 16:03.00 | gnosys | that question about call waiting... I just tried it with callwaiting=no in zapata.conf and I still get the call waiting pips. These are SIP handsets. Is there a separate config for them than for fxs handsets? It seems that callwaiting= in zapata.conf only affects fxs handsets. |
| 16:03.16 | Seyr | tzanger: span=1,0,0,esf,ami |
| 16:03.16 | Seyr | loadzone = us |
| 16:03.16 | Seyr | defaultzone=us |
| 16:03.43 | tzanger | Seyr: that wasn't pasted to pastebin, but you're lucky it was only small. :-) |
| 16:03.54 | Seyr | tzanger: thats why I didnt pastebin :-) |
| 16:04.12 | Seyr | pastebin for 3 lines is kinda like using a 2100 card just for timing.... a waste :-) |
| 16:04.36 | tzanger | Seyr: hahah |
| 16:04.58 | tzanger | ok, repeat that span line 3 more times and make it so it's span=1... span=2... span=3... span=4... |
| 16:05.18 | tzanger | I kind of doubt that's the issue but we'll see |
| 16:05.28 | tzanger | and ztcfg -v output, I want to see what it's saying |
| 16:06.26 | Seyr | Same error as before, but with 4 new lines: "SPAN 1: ESF/ AMI Build-out: 0 db (CSU)/0-133 feet (DSX-1)" but replace SPAN 1 with 2,3 & 4 |
| 16:06.45 | tzanger | is the kernel module loading? |
| 16:07.05 | *** join/#asterisk MatsK (n=mk@c83-253-29-22.bredband.comhem.se) |
| 16:07.10 | tzanger | do an rmmod wct4xxp zaptel ; dmesg -c ; modprobe wct4xxp and dmesg again and paste the LAST dmesg output (not hte first) |
| 16:07.12 | tzafrir_laptop | have a look at /proc/zaptel |
| 16:07.20 | tzafrir_laptop | no need to rmmod |
| 16:07.23 | gnosys | anyone know how to turn off the call waiting pips ("boop boop") with a sip handset? Is the setting in the phone config files or * config files? I see no mention of such a setting in sip.conf on the wiki. |
| 16:07.37 | tzanger | gnosys: oh I thought you were talking about zap before... |
| 16:07.42 | tzanger | I'm not familliar with SIP |
| 16:07.46 | tzafrir_laptop | and what's wrong with dmesg | tail |
| 16:07.47 | tzafrir_laptop | ? |
| 16:08.06 | fugitivo | gnosys: i don't think you can disable that |
| 16:08.27 | tzafrir_laptop | gnosys, it should be in the configuration of the handset |
| 16:08.59 | Seyr | ugh, sorry.. when I try to restart zaptel with those 4 lines for SPAN, it errors |
| 16:09.00 | tzanger | tzafrir_laptop: because I like seeing the entire output and it's easier to get clean output with what I said than to tell them to search for the start of the modprobe wct4xxp outptu :-) |
| 16:09.23 | tzafrir_laptop | /proc/zaptel/1 |
| 16:11.39 | gnosys | tzafrir_laptop: do you happen to know what the setting is for Cisco 79XX phones? I don't see anything promising-looking in my config files. |
| 16:11.50 | *** join/#asterisk darkskiez (n=mhb@bb-195-172-50-6.ukonline.co.uk) |
| 16:12.34 | tzafrir_laptop | gnosys, you can always drop the mailbox= from sip.conf... |
| 16:12.42 | *** join/#asterisk imran (n=codentes@cpe-68-206-53-16.houston.res.rr.com) |
| 16:13.05 | gnosys | I don't have any mailbox= in my sip.conf |
| 16:14.30 | Seyr | ok, |
| 16:14.41 | Seyr | tzanger, tzafrir: http://pastebin.ca/36106 |
| 16:15.53 | *** join/#asterisk buzzyd (n=buzzyd@82-45-247-173.cable.ubr01.enfi.blueyonder.co.uk) |
| 16:16.44 | *** part/#asterisk darkskiez (n=mhb@bb-195-172-50-6.ukonline.co.uk) |
| 16:17.00 | buzzyd | hi all, I have a CDR question looking through the CDR database I see under DST letters like t and s |
| 16:17.03 | *** join/#asterisk imran (n=codentes@cpe-68-206-53-16.houston.res.rr.com) |
| 16:17.11 | buzzyd | t assume is timeout but what is s? |
| 16:17.55 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
| 16:18.23 | *** join/#asterisk jsmith (n=jsmith@smithfam.dsl.xmission.com) |
| 16:20.38 | gnosys | I think I found the setting for Cisco 79XX. It's call_waiting in SIPDefault.cnf. Discovered by telnetting into the phone and experimenting with the commands there. "show config" gave me a bunch of settings, and I found this one. I'll try it now. |
| 16:22.02 | *** join/#asterisk darkskiez (n=mhb@bb-195-172-50-6.ukonline.co.uk) |
| 16:22.42 | *** join/#asterisk Ariel_ (n=Ariel@dsl-20-177.cofs.net) |
| 16:23.06 | Ariel_ | hello everyone |
| 16:23.47 | buzzyd | anyone one know what the letters mean in the CDR database under dst field |
| 16:24.17 | *** join/#asterisk viLeR (i=1000@66.128.47.232) |
| 16:26.43 | Seyr | tzanger, tzafrir: I need to take the kids to Wal-Mart, will check back with everyone on the silence problem later. thansk for the help so far |
| 16:26.49 | *** part/#asterisk Seyr (n=Seyr@cpe-67-10-139-141.houston.res.rr.com) |
| 16:26.54 | jsmith | buzzyd: Destination? |
| 16:29.52 | *** join/#asterisk jahani (n=k@adsl196-206-241-217-196.adsl196-16.iam.net.ma) |
| 16:30.07 | buzzyd | jsmith: I know DST means Destination but asterisk is inputting fields with letters t and s instead of say an extension just wondering what that means t i assume is timeout but s I can workout |
| 16:30.36 | [TK]D-Fender | "s" as in the default extension..... |
| 16:31.25 | buzzyd | yeah I suppose would that mean the caller dialed it went to s then they hungup? |
| 16:32.22 | [TK]D-Fender | buzzyd : should be the last exten dialed. if thats a hangup, oh well |
| 16:33.30 | *** join/#asterisk kareena (n=k@adsl196-104-23-206-196.adsl196-1.iam.net.ma) |
| 16:34.40 | *** join/#asterisk TheCops (i=nobody@got.securebinary.com) |
| 16:36.14 | *** join/#asterisk Barby[talkative] (n=BeaT@85.108.150.190) |
| 16:38.08 | SplasPood | Hrm, my CDR rows always seem to have a disposition of FAILED, anyone know what might be causing that |
| 16:41.11 | tzanger | yeah there's a bug for that already |
| 16:41.13 | tzanger | I get it too |
| 16:41.16 | tzanger | on one of my boxes |
| 16:41.18 | *** join/#asterisk anonymouz666 (n=lynx@200.218.193.6) |
| 16:41.34 | tzanger | I have office* - iax2 - colo* - pstn |
| 16:41.40 | tzanger | and home* - iax2 - colo* - pstn |
| 16:41.49 | tzanger | office* uses PRI to a KSU |
| 16:41.59 | tzanger | home* uses TDM420 to regular phones |
| 16:42.06 | tzanger | home* has FAILED for every CDR |
| 16:42.09 | tzanger | office* is fine |
| 16:42.11 | tzanger | colo* is fine too |
| 16:42.24 | tzanger | all running the same code |
| 16:43.38 | tzanger | oh this is interesting |
| 16:43.40 | tzanger | not all of them are failed |
| 16:43.42 | tzanger | wtf |
| 16:46.22 | *** part/#asterisk imran (n=codentes@cpe-68-206-53-16.houston.res.rr.com) |
| 16:46.37 | *** join/#asterisk imran (n=codentes@cpe-68-206-53-16.houston.res.rr.com) |
| 16:47.32 | *** join/#asterisk ManxPower (i=ewieling@101.sub-70-197-187.myvzw.com) |
| 16:48.58 | tzanger | hmm since 8:30pm yesterday they're showing answered instead of failed |
| 16:49.06 | *** join/#asterisk schuylerdigium (n=Bosco@pcp03052091pcs.huntsv01.al.comcast.net) |
| 16:49.36 | tzanger | SplasPood: try updateing to lasted svn trunk |
| 16:50.14 | *** join/#asterisk wunderkin (i=kev@ip68-226-113-228.ph.ph.cox.net) |
| 16:51.10 | *** join/#asterisk RoyKa (n=roy@ti200720a080-0856.bb.online.no) |
| 16:51.13 | gnosys | with call_waiting set to 0 in SIPDefault.cnf for Cisco79XX, I still get the call waiting pips (boop boop). Does anyone know how to turn these off? Is there an * setting for it? |
| 16:52.28 | tzanger | gnosys: I am guessing it's being inserted into the audio stream in asterisk |
| 16:52.28 | tzanger | you'd have to dig around int he chan_sip code |
| 16:52.28 | SplasPood | tzanger: ok |
| 16:52.37 | tzanger | careful though, I don't think anyone who's done that has come out alive |
| 16:52.48 | gnosys | thanks for the suggestion, tzanger |
| 16:53.01 | tzanger | In fact, I think chan_sip was designed specifically to provide blood sacrifices to digium. :-) |
| 16:53.44 | gnosys | Well, if you guys don't hear from me in a couple of hours or so, you'll know what happened... send emergency services for me (kidding!) |
| 16:55.21 | tzanger | :-) |
| 16:58.02 | SplasPood | tzanger: question, would the SVN trunk be different than the current CVS trunk? |
| 16:59.20 | fugitivo | gnosys: as i said, i don't think you can disable it |
| 16:59.39 | gnosys | :-( |
| 17:00.13 | gnosys | it's very annoying... I miss things that the caller is telling me when the pips come in... I'm constantly asking them to repeat themselves... |
| 17:00.30 | fugitivo | gnosys: what do you get if you disable callwaiting from zapata.conf? |
| 17:00.47 | gnosys | it is already disabled. I'm still getting the pips. |
| 17:01.19 | fugitivo | is the call generated on the same line? or another line? |
| 17:01.51 | fugitivo | when you hear the beeps |
| 17:02.29 | SplasPood | that'd be virus.. |
| 17:03.05 | tzanger | fugitivo: this isn't zapata it's sip |
| 17:03.49 | fugitivo | tzanger: isn't he receiving the call from pstn/zaptel? |
| 17:03.52 | gnosys | the way it happens is that regardless of which line I'm on (on the Cisco 79XX), if any line receives an incoming call while I'm there, I get the pips. Even if the line I'm on is not being rung by the * server. |
| 17:03.59 | *** join/#asterisk Chuji (i=Chuji@pcp09180602pcs.nash01.tn.comcast.net) |
| 17:04.10 | TheCops | How can you make a return key in body of the UserEvent dialplan command ? |
| 17:04.31 | fugitivo | gnosys: is that a pstn/zaptel line? |
| 17:05.19 | fugitivo | gnosys: if it's, ask your telco to disable call waiting |
| 17:05.27 | dorphalsig | bye |
| 17:05.29 | dorphalsig | thnx |
| 17:05.34 | gnosys | the incoming lines to * are pstn/zaptel lines, the lines from the * server to all internal phones are SIP lines. The incoming call comes in over pstn, and there are two pstn lines. When the first is already in use, the telco automatically rolls it over to the second. |
| 17:05.54 | tzanger | gnosys: so the call waiting tones are definitely coming from * |
| 17:05.59 | gnosys | yep. |
| 17:06.24 | tzanger | out of curiousity, enable sip debug and see if you get a sip message when it's sending that |
| 17:06.24 | fugitivo | he has callwaiting disable on zapata.conf |
| 17:06.27 | fugitivo | d |
| 17:06.28 | gnosys | and i think they're being generated strictly on the sip channel |
| 17:06.35 | tzanger | that'll tell you whether it is done in audio or not |
| 17:06.45 | tzanger | fugitivo: again, callwaiting=yes is for zapata, not incomign zapata |
| 17:07.15 | fugitivo | right |
| 17:07.19 | fugitivo | zapata handsets |
| 17:07.22 | tzanger | yes |
| 17:07.27 | fugitivo | sorry, my mistake |
| 17:07.36 | tzanger | gnosys: i.e. establisth a call |
| 17:07.40 | tzanger | sip debug |
| 17:07.48 | tzanger | call again and get the call waiting beeps |
| 17:07.54 | tzanger | see if you got SIP messages during that time |
| 17:08.19 | fugitivo | shouldn't be that a handset feature? |
| 17:08.30 | *** join/#asterisk anonymouz666 (n=lynx@200.218.193.6) |
| 17:08.50 | gnosys | ok. i'll try it, tzanger. thanks. if i do get SIP messages, what does that tell me? perhaps a patch to the chan_sip code is needed? |
| 17:09.19 | tzanger | gnosys: it means it'll be EASIER to get rid of :-) |
| 17:09.29 | gnosys | ok |
| 17:09.37 | tzanger | otherwise you'll have to dig aroudn some more and see if you can kill the generator that's being invoked to make those tones |
| 17:09.50 | file[laptop] | chan_sip won't generate call waiting tones |
| 17:09.51 | tzanger | file[laptop]: my brother does that |
| 17:10.07 | fugitivo | i think it should be deactivated on the handset |
| 17:10.17 | file[laptop] | a call is a call, it doesn't interrupt it with call waiting tones... it simply sends the call to the device and the device does whatever |
| 17:10.17 | tzanger | file[laptop]: 2 zap lines, no call waiting on them. telco does rollover between them. |
| 17:10.25 | tzanger | call comes in, rings SIP/100 |
| 17:10.29 | tzanger | SIP/100 answers. |
| 17:10.32 | tzanger | conversation ensues |
| 17:10.33 | gnosys | fugitivo: there is a setting for it in the handset (Cisco 79XX), but it doesn't work as advertised. |
| 17:10.45 | tzanger | call comes in, rings SIP/100 again btu SIP/100's on the phone (Cisco 7960 IIRC) |
| 17:10.50 | tzanger | "beep beep" |
| 17:10.52 | *** join/#asterisk Astar (n=astar@ANantes-154-1-63-208.w81-53.abo.wanadoo.fr) |
| 17:10.53 | hackeron | can anyone recommend iax/sip software to connect to the asterisk server from the linux desktops? |
| 17:10.56 | Astar | hello |
| 17:11.07 | file[laptop] | then the Cisco should send back a sip reply of temporarily unavailable, busy here, whatever |
| 17:11.09 | tzanger | Asterisk is either sending a SIP message or is injecting audio |
| 17:11.11 | fugitivo | gnosys: i see that setting on my pap2 too, and i get the beeps, so i think it's a handset feature, asterisk just sends the call |
| 17:11.19 | tzanger | file[laptop]: that's why I have asked for a sip debug |
| 17:11.29 | [TK]D-Fender | kphone, linphone, GnomeMeeting (sson) |
| 17:11.43 | tzanger | hackeron: iaxclient, xten for linux, kphone, gnophone...? |
| 17:11.55 | hackeron | tzanger: [TK]D-Fender: hmm, which one do you prefer and why? |
| 17:11.59 | fugitivo | kiax |
| 17:12.06 | tzanger | hackeron: I don't use one yet |
| 17:12.13 | fugitivo | iaxcomm |
| 17:12.16 | tzanger | but I'd likely use either kphone or a qt/kde frontend to iaxclient |
| 17:12.25 | hackeron | fugitivo: kiax looks nice |
| 17:12.27 | fugitivo | tzanger: kphone is buggy |
| 17:12.33 | fugitivo | tzanger: use kiax instead |
| 17:12.36 | tzanger | fugitivo: they're *all buggy |
| 17:12.44 | fugitivo | but kphone is terribly buggy |
| 17:12.46 | [TK]D-Fender | hackeron : Wouldn't know, never used. They're jsut the popular ones. |
| 17:12.47 | fugitivo | bad code |
| 17:12.49 | tzanger | I need better BT headset support in there |
| 17:13.16 | tzanger | I was able to, however, play mp3s through my (mono, 8khz) headset... took a lot of CPU :-) |
| 17:13.39 | Astar | i'm trying to receive call with my asterisk@home with an sip account & i cant but i can call .... |
| 17:14.48 | hackeron | tzanger: wow, speex :) -- fancy. Does asterisk transcode it to ulaw? |
| 17:15.11 | tzanger | no this wasn't through asterisk |
| 17:15.15 | tzanger | it was mplayer out to bt-alsa |
| 17:18.25 | *** join/#asterisk roulduke_ (i=u0ru5dja@p508D37F5.dip0.t-ipconnect.de) |
| 17:20.04 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 17:22.31 | gnosys | tzanger: the sip debug output is at http://pastebin.com/495088 |
| 17:23.25 | *** part/#asterisk schuylerdigium (n=Bosco@pcp03052091pcs.huntsv01.al.comcast.net) |
| 17:24.10 | *** join/#asterisk gambolputty (n=gambolpu@cblmdm72-240-240-242.buckeyecom.net) |
| 17:24.31 | gnosys | the boop sound seemed to happen exactly simultaneously with the first SIP debug messages. As soon as I saw output, I disconnected the call, so when the call destroy messages come in, the boop sound has already come and gone. |
| 17:24.34 | Cyberchen | is someone here using a snom190 with the 1.2.1 asterisk? |
| 17:24.39 | gambolputty | yes |
| 17:24.40 | gambolputty | I am |
| 17:25.02 | Cyberchen | hmm |
| 17:25.05 | Cyberchen | does it work? |
| 17:25.08 | gambolputty | yes |
| 17:25.13 | gambolputty | 3.60x firmware |
| 17:25.16 | gambolputty | upgrade if need be |
| 17:25.18 | Cyberchen | strange |
| 17:25.27 | Cyberchen | i have upgraded to the latest fw now |
| 17:25.33 | Cyberchen | and i use the 1.2.1 * |
| 17:25.37 | Cyberchen | all phones here are working |
| 17:25.45 | Cyberchen | but with the snom 190 i can only dial out |
| 17:25.54 | gambolputty | but you cant get calls? |
| 17:25.58 | Cyberchen | right |
| 17:26.04 | gambolputty | then it is an * problem |
| 17:26.08 | gambolputty | not a phone problem |
| 17:26.28 | Cyberchen | hmm |
| 17:26.30 | gnosys | tzanger: I'm thinking that lines 22 through 31 are relevant (in http://pastebin.com/495088 ). what do you think? |
| 17:29.04 | *** join/#asterisk sonic2wb (i=sonic2wb@user-1120vtv.dsl.mindspring.com) |
| 17:29.41 | sonic2wb | Good Morning |
| 17:31.12 | sonic2wb | whats the command for Setting the callerid |
| 17:31.13 | sonic2wb | ? |
| 17:31.16 | sonic2wb | Setcid? |
| 17:31.42 | gnosys | which version of *, sonic? |
| 17:32.06 | sonic2wb | 1.2.1 |
| 17:32.06 | sonic2wb | 1 |
| 17:32.09 | sonic2wb | grr |
| 17:32.13 | sonic2wb | 1.2.1 |
| 17:32.30 | gnosys | Set(CALLERID(num)=XXXXXXXXXX) |
| 17:33.12 | *** part/#asterisk RoyKa (n=roy@ti200720a080-0856.bb.online.no) |
| 17:33.18 | tzanger | gnosys: no |
| 17:33.26 | tzanger | gnosys: but you see Asterisk sending an INVITE |
| 17:33.36 | gnosys | yep |
| 17:33.39 | tzanger | and you're not getting a message back from the Cisco saying to piss off |
| 17:33.47 | *** join/#asterisk RoyK (n=roy@ti200720a080-0856.bb.online.no) |
| 17:33.48 | tzanger | so it looks like your cisco config needs to be changed |
| 17:33.50 | gnosys | right |
| 17:34.21 | tzanger | PolycomSoundPointIP-SPIP_501-UA/1.6.2.0041 |
| 17:34.28 | tzanger | that's also not a Cisco, it's a Polycom IP501 |
| 17:34.53 | iDunno | heh. |
| 17:35.11 | gnosys | Right. This is a test setup at home, different from my production setup. I thought I was looking for a message from * to the phone that was causing the tone to be generated. |
| 17:35.23 | tzanger | no, the phone's doing it all itself |
| 17:35.38 | tzanger | it's getting an INVITE and it's configured to generate call waiting tones apparently |
| 17:35.39 | *** join/#asterisk alephcom (n=alephcom@host75.net14.mcsnet.ca) |
| 17:35.51 | gnosys | the phone is? |
| 17:35.55 | tzanger | yes |
| 17:36.32 | tzanger | configure the IP501 so that it rejects calls or that it doesn't do the tones... I don't know those phones well enough, maybe polycom has some kb or whatnot you can consult |
| 17:36.45 | sonic2wb | how do u set name Set(CALLERID(name?)=XXXXXXXXXX) |
| 17:36.53 | dpryo | Ok, I just made an ircbot accepting "messages", and a call-command that allow me to call a number, and have a text-to-speech engine say my text! |
| 17:36.58 | gnosys | Well, I'm sure there's a parallel setting in the polycom phones for turning off call waiting. I'll hunt around and see what I can find. So, you're thinking that the phone SHOULD be telling * to piss off in response to the INVITE command? And if it does so, then I won't get the pips? |
| 17:37.42 | gnosys | sonic2wb: probably can't be done. You can try with that command, but if you're using telco lines, all they do is pass the name assigned to the number you set. |
| 17:37.44 | *** join/#asterisk Gunnar (n=gunnar@34.80-203-209.nextgentel.com) |
| 17:38.00 | *** join/#asterisk NDT (n=me@cpe-24-195-216-41.nycap.res.rr.com) |
| 17:38.09 | tzanger | gnosys: no, the phone is configured to do this, if you configure the phone not to do that, you shouldn't get the tones |
| 17:38.56 | gnosys | well that's what I've done with the production system. I can telnet into the Cisco 7960 and do a show config and see that call_waiting is 0, but I still get the pips. |
| 17:39.22 | tzanger | gnosys: set the "calls per button" to 1 |
| 17:39.30 | tzanger | on the IP501 |
| 17:40.00 | gnosys | "calls per button" that's a polycom 501 setting? I don't recall seeing a similar setting in the cisco, but I'll look again. |
| 17:40.28 | Cybertoy | Dial(some.sip.provider/<phonenumer>,,t) ... shouldn't that make the person called be able to transfer the call with # ?? |
| 17:40.49 | tzanger | gnosys: try "calls per DN" (directory number) on Cisco |
| 17:40.53 | Cybertoy | or does that only work if the sip extension is locally connected to asterisk? |
| 17:41.06 | tzanger | I found this information just with google and "polycom disable call waiting" and "cisco disable call waiting" |
| 17:41.11 | tzanger | within the first 5 hits of each |
| 17:41.43 | tzanger | on cisco you can also try disabling "ring setting (phone active)" |
| 17:42.03 | tzanger | https://puck.nether.net/pipermail/cisco-voip/2004-November/002139.html |
| 17:43.09 | NDT | Anyone run into this? Cause no one seems to have a distinct answer to this: error while writing audio data: : Broken pipe |
| 17:43.12 | tzanger | not sure if that's callmanager-specific or not |
| 17:43.32 | *** join/#asterisk kiwnix (n=egarcia@229.red-82-158-154.user.auna.net) |
| 17:43.37 | tzanger | NDT: it means the call went away jsut before * wrote the next frame. it's not a problem. |
| 17:44.06 | NDT | ahh ok |
| 17:44.32 | NDT | seems asterisk isn't running though |
| 17:44.36 | NDT | now |
| 17:44.54 | NDT | <PROTECTED> |
| 17:44.54 | NDT | Ouch ... error while writing audio data: : Broken pipe |
| 17:45.02 | NDT | how it all comes out |
| 17:45.10 | tzanger | gnosys: http://www.upenn.edu/computing/voice/voip/help/phones.html |
| 17:45.24 | NDT | Then drops me back to command prompt |
| 17:45.26 | Astar | i'm trying to receive call with my asterisk@home with an sip account & i cant but i can call .... |
| 17:45.40 | tzanger | NDT: again it looks like you're trying to use VBR MP3s, I told you this about 2 or 3 hours ago |
| 17:45.58 | tzanger | Astar: you may want to try the asterisk@home lists and irc channel, |
| 17:46.16 | NDT | didn't tell me heh...I never asked this question...but...I have no sounds..other then the ones by default |
| 17:46.23 | gnosys | thanks, tzanger. I just posted the output of show config on the cisco phones to http://pastebin.com/495118 I'll read up on your suggestions. |
| 17:46.25 | Astar | ok |
| 17:46.47 | gnosys | I didn't find anything like calls per line or calls per DN, but thought I'd post anyway. |
| 17:47.03 | NDT | The VBR is what I found searching the web too...but seeing as how I never added any sounds...makes no sense |
| 17:47.04 | *** join/#asterisk Defraz (n=t0tal@001-742-648.area1.spcsdns.net) |
| 17:47.51 | SplasPood | tzanger: updating to latest svn trunk doesn't seem to have changed my disposition cdr problem |
| 17:48.11 | sonic2wb | i got a question: i get choppy sound when use a sip softphone outside connection from asterisk, on the local net sound is fine |
| 17:48.14 | tzanger | hmm, I used to have that problem but it appears to have corrected itself; I'm not sure what to tell you |
| 17:48.35 | tzanger | gnosys: wait |
| 17:48.49 | tzanger | gnosys: it looks like the call manager looks after this |
| 17:48.55 | tzanger | so you need to do it in asterisk |
| 17:49.05 | tzanger | if the channel is in use, don't ring the phone |
| 17:49.26 | tzanger | but you may be able to hint it or otherwise notifiy the phone without ringing it specifically |
| 17:49.46 | SplasPood | tzanger: hrm, it doesn't seem to be all my calls... calls between exten show the proper disposition, but calls from the PSTN in (via an IAX link) show FAILED |
| 17:50.04 | Qwell | SplasPood: There is a bug on the tracker |
| 17:50.17 | *** join/#asterisk [hC] (n=lisa@65.182.25.47) |
| 17:50.19 | SplasPood | yeA? /me goes to look |
| 17:50.30 | SplasPood | [hC]: hey lisa :P |
| 17:50.34 | Qwell | 5918 |
| 17:50.37 | gnosys | tzanger: I get the impression from https://puck.nether.net/pipermail/cisco-voip/2004-November/002139.html that this is a setting that one would activate by using the buttons on the phone and navigating through the menus. I'm guessing that those setttings should also be adjustable by telnet commands or SIPDefault.cnf or SIPXXX.cnf commands. |
| 17:50.41 | [hC] | :P I'm on my girlfriends laptop |
| 17:50.45 | Qwell | [hC]: Your name is Lisa? :P |
| 17:50.48 | Qwell | oh, nm |
| 17:50.53 | [hC] | Haha |
| 17:51.00 | [hC] | Yeah, its really awkward at times. |
| 17:51.05 | Qwell | so, wait |
| 17:51.21 | [hC] | So qwell |
| 17:51.21 | [hC] | That issue with the x100p iw as having, i fixed it. |
| 17:51.22 | [hC] | Youll never fucking guess what did it. |
| 17:51.26 | Qwell | umm |
| 17:51.29 | Qwell | solar flares? |
| 17:51.32 | sonic2wb | lol |
| 17:51.35 | [hC] | I installed the -686 kernel in debian, rather than using the -386. |
| 17:51.39 | Qwell | wtf |
| 17:51.44 | Qwell | you shitting me? |
| 17:51.47 | [hC] | yeah |
| 17:51.53 | [hC] | did that, recompiled zaptel |
| 17:51.56 | [hC] | 100% better. |
| 17:52.00 | Qwell | wow |
| 17:52.14 | [hC] | the only other possibility is that i hadnt done a make clean in zaptel, and was just doing make a lot of the time previously |
| 17:52.23 | Qwell | maybe.. |
| 17:52.39 | gnosys | tzanger: following up your comment: "if the channel is in use, don't ring the phone" are you talking about some conditional logic in extensions.conf here? |
| 17:52.41 | Qwell | upgraded gcc recently? |
| 17:52.42 | [hC] | Anyways, completely mental, but it works fine now. |
| 17:52.47 | [hC] | Nope. |
| 17:52.51 | [hC] | Fresh install |
| 17:52.51 | Qwell | funky |
| 17:53.05 | [hC] | also, i was having /dev/rtc - no such device errors on the sc420 i was using first |
| 17:53.10 | [hC] | after upgrading to --686, all gone. |
| 17:53.13 | tzanger | http://www.voip-info.org/wiki/index.php?page=Asterisk+phone+cisco+79xx |
| 17:53.19 | tzanger | jsut past halfway down |
| 17:53.20 | Qwell | vmware is complaining about no /dev/rtc |
| 17:53.22 | [hC] | so there was some sort of timing thing enabled in the -686 one i think. |
| 17:53.38 | Qwell | I set my rate to 1000Hz last night...need to reboot to fix it |
| 17:53.42 | tzafrir_laptop | [hC], why not use zaptel debs, then? |
| 17:53.55 | tzanger | gnosys: if that can't help, then you will need to do the conditional logic in extensions.conf yes |
| 17:53.57 | [hC] | Now i am just dealing with echocan on these x100p (clones) |
| 17:53.59 | gnosys | tzanger: I read that, but it didn't quite seem applicable to me... I'll read again. Are you talking about the section titled "Call Waiting"? |
| 17:54.03 | gnosys | or similar? |
| 17:54.06 | tzanger | yeah |
| 17:54.11 | [hC] | tzafrir_laptop: I hadnt seen any 1.2.1 debs out yet |
| 17:54.15 | tzanger | do you have multiple appearances with the same sip login? |
| 17:54.34 | tzafrir_laptop | [hC], what version of Debian? |
| 17:54.41 | [hC] | tzafrir_laptop: testing |
| 17:54.57 | [hC] | oddly enough the mark2 echo can seems to work better than mg1. |
| 17:55.30 | [hC] | ag, back in a minute.. |
| 17:55.39 | tzafrir_laptop | [hC], hmmm, I'm sure it got into Sid. Not sure about Etch |
| 17:55.41 | Qwell | brb, reboot |
| 17:56.29 | *** join/#asterisk vandien (i=sted@aditu.dahltronics.de) |
| 17:56.38 | NDT | 1.2.1 error is gone now |
| 17:57.40 | vandien | hi all... i need a preferebly large image of the asterisk with that bubble around it... does anybody have something like that? |
| 17:58.31 | gnosys | tzanger: I have each of the six lines configured as a separate sip account on * and registered independent of one-another. I considered the multiple appearances with the same sip login option, but thought that for my use, my choice was the better of the two. That's why I didn't see that section as very applicable to me. |
| 17:58.59 | vandien | http://www.asterisk.org/images/logo_rev.gif <-- that one's a little small.. |
| 17:59.28 | NDT | make it again in photoshop 8) |
| 18:00.20 | vandien | i wish i could |
| 18:00.28 | tzanger | hmm |
| 18:00.29 | tzanger | so why's it accepting calls |
| 18:00.37 | tzanger | what happens if youturn call waiting on on the 7960? |
| 18:00.39 | *** join/#asterisk Qwell (n=north@unaffiliated/qwell) |
| 18:00.40 | NDT | how big? |
| 18:01.14 | gnosys | the call_waiting setting in SIPDefault.cnf seems to do nothing. On or off, I get the pips. |
| 18:01.29 | tzanger | well |
| 18:01.36 | tzanger | I'm starting to think this is a cisco SIP bug |
| 18:01.43 | tzanger | you've got a license, ask them |
| 18:02.38 | gnosys | When I get an incoming call on one of the Zap lines, I have * ring the last two lines on each of the cisco phones. But regardless of whether I'm on one of those two lines, or on one of the other four lines, I hear the pips. |
| 18:02.53 | gnosys | ok. I'll try cisco too... |
| 18:03.04 | *** join/#asterisk ToTo (n=ToTo@host242-83.pool8260.interbusiness.it) |
| 18:03.23 | tzanger | yeah see I think that's a bug but I don't know. |
| 18:03.57 | gnosys | I'll experiment some more with the polycom phones. if they work as expected on this note, then I'm inclined to agree with you. |
| 18:04.13 | [hC] | back |
| 18:04.56 | [hC] | man I'm like a giddy little schoolgirl waiting for the announcements this tuesday at moscone |
| 18:05.10 | [hC] | how sad |
| 18:05.41 | tzanger | ? |
| 18:05.54 | [hC] | the new apple announcements |
| 18:06.10 | [hC] | I believe its the macworld expo, i forget which conference exactly |
| 18:06.15 | vandien | the apple expo? |
| 18:06.28 | [hC] | but its either new intel ibooks, or ive read rumors of some large plasma displays with osx86 integrated into the display |
| 18:06.30 | vandien | i thought it's at "the end og january" |
| 18:07.12 | [hC] | I heard tueday. I could be mistaken |
| 18:08.02 | vandien | me too :) |
| 18:08.39 | [hC] | Jan 9-13 |
| 18:08.54 | [hC] | lots of rumors |
| 18:08.57 | vandien | yeah |
| 18:09.10 | [hC] | intel powerbooks, ibooks, mac minis, new ilife, new ipod shuffles, preview of leopard, etc |
| 18:09.18 | sonic2wb | what would cause incomming sound from asterisk to be choppy |
| 18:09.31 | [hC] | from what device? |
| 18:09.31 | vandien | i personnaly hope for better / cheaper cinema displays... i want one :D |
| 18:09.44 | [hC] | dell just released apple's 30 inch display for 1999 |
| 18:10.18 | *** join/#asterisk A-Tuin|work (n=A-Tuin@nat.office.legend.net.uk) |
| 18:10.18 | sonic2wb | HC: iax connection to voip |
| 18:10.20 | vandien | dell... i won't put a dell display next to my powerbook :p |
| 18:10.38 | vandien | although it does not look that bad... |
| 18:12.23 | vandien | uh, the 30" needs dual link dvi... my book doesn't have that. well, 23" should be enough... |
| 18:13.15 | *** join/#asterisk pigpen2 (n=mark@fw.seamans.cc) |
| 18:13.34 | [hC] | vandien: you know the dell displays ARE the apple displays, same parts 100%, just different case. |
| 18:13.38 | gnosys | tzanger, thanks for the help. though i haven't yet solved the problem, i think i've gotten alot closer thanks to your assistance. |
| 18:13.41 | gnosys | :-) |
| 18:13.50 | tzanger | gnosys: :-) well I try :-) |
| 18:13.51 | [hC] | vandien: and yeah you wont have dual link DVI unless you have the last generation laptops |
| 18:14.15 | [hC] | sonic2wb: and what happens to the audio? its jittery? |
| 18:14.42 | [hC] | sonic2wb: you may have latency spikes to the iax provider, or you may want to play with your jitterbuffer settings in iax.conf |
| 18:15.47 | *** join/#asterisk Qwell (n=north@unaffiliated/qwell) |
| 18:17.03 | vandien | hehe, but the case makes it an apple ;) |
| 18:18.54 | [hC] | interesting. full airport signal here, yet 3 feet away i can barely get anything. |
| 18:18.56 | [hC] | brb, reboot time |
| 18:23.01 | [TK]D-Fender | [avDell displays are NOT Apple Displays... check the TGH comparisons.... |
| 18:25.19 | *** join/#asterisk Qwell (n=north@unaffiliated/qwell) |
| 18:25.24 | *** join/#asterisk pigpen2 (n=mark@fw.seamans.cc) |
| 18:26.05 | vandien | TGH? you mean THG? |
| 18:26.25 | vandien | [TK]D-Fender: toms hardware guide? |
| 18:30.06 | [TK]D-Fender | THG |
| 18:30.08 | [TK]D-Fender | yes |
| 18:30.25 | [TK]D-Fender | The did all sorts of reviews of 20" disps, and Dell won out. |
| 18:30.38 | fugitivo | thg is crap |
| 18:30.48 | fugitivo | not true all the reviews |
| 18:30.49 | [TK]D-Fender | I myself have switched over to 16:10 (Acer AL1916W 19" 1440x900) |
| 18:31.13 | [TK]D-Fender | Dunno... these kind of tests I don't knock.... |
| 18:31.20 | fugitivo | my 15" is 1400x1050 |
| 18:31.30 | [TK]D-Fender | Dell Laptop? |
| 18:31.42 | fugitivo | no, hypersonic custom laptop |
| 18:31.48 | [TK]D-Fender | I seem to recall those on my D610's |
| 18:32.00 | [TK]D-Fender | AH.... haven't seen that res aound much elsewhere. |
| 18:32.11 | fugitivo | the brand is AU |
| 18:32.23 | fugitivo | Au Optronics |
| 18:32.27 | fugitivo | model B150PG01 |
| 18:32.34 | fugitivo | (lcd brand) |
| 18:32.54 | fugitivo | maybe dell laptops use the same brand |
| 18:33.30 | [TK]D-Fender | Or maybe just a mroe common size now.... I find it awkward with native aspect ratio. |
| 18:33.48 | fugitivo | it's not new my laptop |
| 18:33.51 | fugitivo | 2 years old |
| 18:34.07 | [TK]D-Fender | My Dell's seem to rescale things so it feels more like 1024x768 in windows (font's & icons) |
| 18:34.54 | *** join/#asterisk ast_freak|Laptop (n=jesse@12.28.106.2) |
| 18:35.16 | fugitivo | i'm using linux right now with dual head |
| 18:35.35 | fugitivo | my laptop with 1400x1050 and a samsung lcd 17" 1280x1024 |
| 18:36.08 | tzanger | nice |
| 18:36.33 | fugitivo | my laptop needs more bright |
| 18:37.29 | [TK]D-Fender | I'd bet they're the same.. my D610's are kinda wimpy too.... |
| 18:37.46 | fugitivo | remove the lcd, and you'll see the brand :) |
| 18:37.54 | fugitivo | it's quite easy to do that |
| 18:38.21 | [TK]D-Fender | no thanks :) I'd rather not hear ANYTHING from my salesmen.... |
| 18:38.27 | fugitivo | :) |
| 18:39.03 | fugitivo | anyone with a nokia 770? |
| 18:39.45 | dingo | ok whats Cannot connect to Asterisk Manager with user/pass mean ? whats broken ? |
| 18:39.58 | fugitivo | that's self explained |
| 18:40.29 | NDT | There a free webmin module anywhere for asterisk? |
| 18:40.41 | fugitivo | something like that |
| 18:40.43 | fugitivo | ~amp |
| 18:40.47 | jbot | methinks amp is NOT supported here! people using it should join #amportal |
| 18:40.47 | *** join/#asterisk troyb (n=central@CPE00907f17e478-CM014300013422.cpe.net.cable.rogers.com) |
| 18:40.59 | [TK]D-Fender | No Webin for *. There are other GUI's that can wreck it for you though ;) |
| 18:41.06 | NDT | hehe |
| 18:41.46 | *** join/#asterisk gnosys_ (n=gnosys@ip68-9-201-250.ri.ri.cox.net) |
| 18:41.51 | *** join/#asterisk codword (n=doc@cpe-71-64-192-28.cinci.res.rr.com) |
| 18:42.27 | troyb | i have a cisco 7940 IP Phone that i picked up and its complaining about a file auth failure on 7-4 + i've been told i need to use version 5-1 for SIP first however im not sure if this is true |
| 18:42.32 | NDT | saw the webmin folder in pub on asterisk.org...maybe they are making one heh |
| 18:43.34 | dingo | well shoot all someone had to say was look in manager.conf |
| 18:43.36 | codword | hey guys.. new to asterisk, but not new to linux, networking, or Voip stuff.. question for ya, whats the cheapest (within reason) FXS port solution I can add to my server? preferably something i can pickup locally at like, radio shack/compusa/whatever.. ? |
| 18:44.03 | fugitivo | codword: linksys pap2-na, 2fxs |
| 18:44.16 | drray_ | I don't know of a local pickup fxs port |
| 18:44.26 | codword | fugitivo: sweet, thanks.. just wanted confirmation of something that is known to work well with asterisk before I head out. |
| 18:44.36 | fugitivo | codword: be sure it's -na |
| 18:44.42 | fugitivo | codword: that means it's not locked |
| 18:44.51 | [TK]D-Fender | codword : Don't bet that anything yuo can pick up in retail isn't locked and going to cause you trouble. Best deal these days is Sipura/Linksys SPA-2002 for $70 at www.voipsupply.com |
| 18:44.54 | codword | right.. configurable to register to my H323 of choice ? |
| 18:45.02 | drray_ | which means he won't be able to find it |
| 18:45.07 | fugitivo | codword: SIP |
| 18:45.27 | codword | hmm.. fat chance of finding a -na in stock at any 'regular' store huh ? |
| 18:45.37 | codword | I wanted to mess with this stuff today ;-( |
| 18:45.43 | fugitivo | really? it's preatty easy to find it here |
| 18:45.49 | drray_ | where is here? |
| 18:45.52 | fugitivo | argentina |
| 18:45.57 | codword | Cincinnati Ohio USA |
| 18:45.58 | fugitivo | :) |
| 18:46.00 | drray_ | :) |
| 18:46.04 | codword | we have all the regular stuff like compusa, micro center, etc.. |
| 18:46.09 | codword | bestbuy,circuitcity, etc etc etc |
| 18:46.19 | drray_ | will that take an openwrt firmware? |
| 18:46.47 | NDT | Yeah but why would those places carry anything for asterisk...they just carry normal things that won't gather dust heh |
| 18:47.45 | codword | these 'linksys' products with FXS ports, are hard-coded to work with 'vonage' |
| 18:47.48 | codword | unless they have -na ? |
| 18:47.59 | *** join/#asterisk gnosys_ (n=gnosys@ip68-9-201-250.ri.ri.cox.net) |
| 18:48.07 | *** join/#asterisk [Airwolf] (n=airwolf@82-171-75-4.dsl.ip.tiscali.nl) |
| 18:48.15 | codword | ie,, this is an example: http://www.compusa.com/products/product_info.asp?product_code=328892&pfp=BROWSE |
| 18:48.16 | *** join/#asterisk file (n=joshnet@mctnnbsa24w-142167048221.nb.aliant.net) |
| 18:48.24 | fugitivo | i don't know for what company are locked, i know you have to find the -NA |
| 18:48.59 | NDT | Well if it's not sold in a vonage box or something I would assume they aren't locked? |
| 18:49.08 | fugitivo | no |
| 18:49.09 | codword | Compusa has the PAP2-CU for free :) |
| 18:49.10 | drray_ | if it says -na |
| 18:49.15 | drray_ | it's not locked |
| 18:49.17 | codword | I bet someone has a way to ahck those :) |
| 18:49.18 | fugitivo | what is -CU? |
| 18:49.32 | codword | I dunno.. this is what I'm looking at: http://www.compusa.com/products/product_info.asp?product_code=314460&pfp=BROWSE |
| 18:49.59 | fugitivo | that's for vonage |
| 18:50.04 | fugitivo | it's locked |
| 18:50.05 | drray_ | that's going to be locked to vonage |
| 18:50.10 | troyb | i have a cisco 7940 IP Phone that i picked up and its complaining about a file auth failure on 7-4 + i've been told i need to use version 5-1 for SIP first however im not sure if this is true |
| 18:50.10 | fugitivo | you need that one, but -NA |
| 18:50.15 | codword | It's gotta be hackable :) |
| 18:50.32 | drray_ | why not get a dev kit? |
| 18:50.47 | fugitivo | if it's free, then they'll ask you to get a vonage account |
| 18:51.06 | drray_ | and it's not free, it's after rebate |
| 18:51.11 | *** join/#asterisk maite (n=Administ@200.75.29.98) |
| 18:51.14 | NDT | It $100 with $100 mail in rebate heh |
| 18:51.21 | codword | well, thats 'free enough' heh |
| 18:51.22 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
| 18:51.23 | codword | hmmm |
| 18:51.23 | maite | buenas |
| 18:51.23 | codword | :( |
| 18:51.26 | maite | como estan todos |
| 18:51.28 | troyb | i never mail in the rebate :( |
| 18:51.32 | maite | alguien que hable español |
| 18:51.35 | maite | por favor |
| 18:51.36 | fugitivo | hola |
| 18:51.38 | drray_ | get a tdm wildcard |
| 18:51.43 | drray_ | or a softphone |
| 18:51.44 | troyb | i've counted up my rebates before and it totaled a few hundred bucks |
| 18:51.48 | *** join/#asterisk [hC] (n=lisa@65.182.25.47) |
| 18:51.59 | NDT | Funny...thing says was $59.99 now FREE... $100 with $100 mail in rebate...wtf LOL |
| 18:52.00 | codword | drray_: I use cisco Call Manager and cisco softphones now |
| 18:52.10 | fugitivo | $100 is expensive |
| 18:52.11 | codword | I wnated to set something up though that is more wife friendly..and I wanted to learn asterisk :) |
| 18:52.16 | drray_ | the cisco softphone will work with asterisk |
| 18:52.23 | codword | I realize that... |
| 18:52.29 | troyb | damn you file auth fail what do you want from me :( |
| 18:52.35 | drray_ | :) |
| 18:52.35 | codword | but I wanna use regular phones.. so she wont yell at me about it being complicated |
| 18:52.39 | codword | need a cheap FXS |
| 18:52.45 | maite | somebody can help me please with one e1 |
| 18:52.55 | codword | I cant believe someone doesnt have a hack for the PAP2-CU :) |
| 18:53.17 | maite | when asterisk me devuel look to publishes it as I can quiet it |
| 18:53.24 | drray_ | I can't believe someone does not say an Iax equiv |
| 18:53.29 | codword | I wonder if you could 'nat' the PAP2-CU to redirect to asterisk :) |
| 18:53.51 | troyb | anyone here experienced with loading firmware on cisco ip phones? |
| 18:53.59 | drray_ | on the 7960 |
| 18:54.01 | drray_ | herhe |
| 18:54.04 | drray_ | er, here |
| 18:54.06 | *** join/#asterisk theblue (n=theblue@pcp04402293pcs.nrockv01.md.comcast.net) |
| 18:54.08 | theblue | Hi all. |
| 18:54.24 | codword | troyb: I've upgraded 7940's before |
| 18:54.28 | codword | and 7960's |
| 18:54.30 | codword | via tftp |
| 18:54.36 | drray_ | once you get UAL on, you are done |
| 18:54.38 | codword | but using call manager :) |
| 18:54.43 | maite | alguien que hable español |
| 18:54.45 | maite | pleace |
| 18:54.46 | troyb | codword yup yup my firmware isnt upgrading for some reason |
| 18:54.48 | fugitivo | maite: yo |
| 18:54.59 | codword | troyb: tftp ? |
| 18:55.00 | drray_ | troyb which firmware? |
| 18:55.02 | troyb | im using TFTP, it requests the OS(whatever).txt just fine |
| 18:55.07 | troyb | it takes my .sbn file fine |
| 18:55.13 | troyb | then it bitches about a file auth fail |
| 18:55.24 | maite | fugitivo: me puedes ayudar con un asterisk y una e1 por favor |
| 18:55.26 | codword | hmmm |
| 18:55.32 | troyb | drray_ i've tried 7.4 and 7.5 |
| 18:55.35 | codword | try re-getting the images just for kicks? |
| 18:55.41 | fugitivo | maite: conmigo hablaste ayer, pero no me respondiste mas |
| 18:55.51 | troyb | the issue is i dont have a cisco CCO account because i bought the phone from a friend and was only given these 2 images |
| 18:56.15 | troyb | so i have no real way of knowing if its my image or i just need to go to 5-1 first |
| 18:56.19 | codword | ah crap if I were at work I could login using my company's CCo account and help ya out :( |
| 18:56.27 | codword | sounds like your images are bad |
| 18:56.44 | troyb | could be |
| 18:56.51 | maite | tienes msn para que hablemos por msn por favor |
| 18:57.08 | hypa7ia | i'm pretty sure you just need to upgrade to a UAL fw first troyb |
| 18:57.19 | *** join/#asterisk alephcom (n=alephcom@host75.net14.mcsnet.ca) |
| 18:57.20 | theblue | Can anyone tell me what the IAX and SIP ports that I'd need to map on a firewall? |
| 18:57.22 | troyb | oh look who it tis :P |
| 18:57.26 | *** join/#asterisk ikey (i=ikey@220.226.17.30) |
| 18:57.49 | fugitivo | theblue: udp 5060, udp 10000-20000 for sip |
| 18:57.55 | hypa7ia | troyb: what, did you think i /wouldn't/ be in here? :) |
| 18:58.11 | troyb | hypa7ia you never know.. i have to be more careful in the future :) |
| 18:58.30 | fugitivo | theblue: udp 4569 for iax2 |
| 18:59.24 | hypa7ia | lol |
| 18:59.41 | troyb | codword you never know who your going to find on here =P |
| 19:00.44 | theblue | fugitivo: Thanks. |
| 19:02.28 | *** join/#asterisk TaSo (i=licucude@ool-44c784a0.dyn.optonline.net) |
| 19:02.36 | *** join/#asterisk linlin (n=linlin@c-67-184-231-233.hsd1.il.comcast.net) |
| 19:02.39 | *** join/#asterisk Voicelynx (n=rda@8.8.197.77) |
| 19:02.56 | *** join/#asterisk TaSo (i=licucude@ool-44c784a0.dyn.optonline.net) |
| 19:03.05 | *** join/#asterisk enmaca (n=enmaca@201.145.73.99) |
| 19:03.18 | *** join/#asterisk maite (n=Administ@200.75.29.98) |
| 19:03.45 | *** join/#asterisk TaSo (i=licucude@ool-44c784a0.dyn.optonline.net) |
| 19:03.57 | marcus2 | hrm |
| 19:04.01 | marcus2 | anyone here use call parking with * ? |
| 19:04.14 | codword | What do you guys think of this ?? http://www.compusa.com/products/product_info.asp?product_code=334130&pfp=BROWSE |
| 19:04.32 | codword | it doesnt appear to be 'hard coded' to any particular service.... |
| 19:06.29 | *** join/#asterisk [hC] (n=lisa@65.182.25.47) |
| 19:13.29 | NDT | same one I am running with vonage |
| 19:14.11 | codword | is it locked to vonage? |
| 19:14.18 | codword | I wanna use it with asterisk |
| 19:14.23 | NDT | Mine is...but I know they sell them non locked to |
| 19:14.33 | codword | I doubt compusa does :( |
| 19:14.40 | troyb | hahah yes yes yes universal application loader booting up |
| 19:14.44 | NDT | If you have a router put it behind it cause none of its router type features work heh |
| 19:14.50 | codword | I'm lookin for an FXS solution that I can put together today |
| 19:14.53 | [TK]D-Fender | codword : thats the same price as the SPA-2002 and a lot less trustworthy... |
| 19:14.59 | *** join/#asterisk mad_jack (i=krzwasil@nat-mia.aster.pl) |
| 19:15.05 | codword | yeah.. my router is a hacked xbox running linux =P |
| 19:15.10 | codword | I love it =P |
| 19:15.22 | sivana | pretty expensive router |
| 19:15.38 | codword | how do ya figure? it cost me $40 bucks used |
| 19:15.44 | sivana | ok..that's not bad :) |
| 19:15.57 | codword | and now that the 360 is out they are a dime a dozen :) |
| 19:16.04 | codword | they make great front-ends for mythtv |
| 19:16.07 | codword | anyway I digress... |
| 19:16.34 | codword | I heard there was a generic PCI/modem that could be used as an FXS port for asterisk? as long as it was a particular kind of chipset? |
| 19:16.51 | codword | I just really wanted to fool with asterisk today but looks like I'll have to order an FXS solution from online and wait :( |
| 19:17.14 | NDT | heh that motorola says in the description: Can be used as a router...only thing it has that does work is dhcp |
| 19:17.44 | NDT | dmz/port forwarding...etc...non of that shit works on it |
| 19:18.09 | codword | I wouldnt use an off-the-shelf-anything for a router anyhow |
| 19:18.22 | codword | bout the best you can do is a WRT54GS running openWRT whiterussian4 :) |
| 19:19.39 | troyb | codword i got the system to load the universal application loader but then it says protocol application invalid |
| 19:19.44 | dingo | w00t SUCCESS AMP working on FreeBSD 6.0 w/ asterisk 1.2.1 |
| 19:19.54 | *** part/#asterisk mad_jack (i=krzwasil@nat-mia.aster.pl) |
| 19:20.50 | dingo | thanks to those of you who helped |
| 19:21.51 | *** join/#asterisk alexhopper (n=a27386@CPE000103d29ae2-CM001225dfdfe0.cpe.net.cable.rogers.com) |
| 19:22.02 | [TK]D-Fender | AMP...... *shudder* |
| 19:22.34 | codword | ok, the vt1005 is hackable |
| 19:22.37 | codword | but it looks like a pain in the ass |
| 19:22.40 | codword | http://www.0xdecafbad.com/Unlocking-Motorola-VT1000.html |
| 19:23.35 | codword | :( |
| 19:24.08 | SplasPood | can you use dialplan variables in a switch => Realtime/ line/ |
| 19:24.22 | Qwell | SplasPood: doubtful |
| 19:24.52 | SplasPood | oh lame, so you really do need to reload and create a dummy context for each context you add to the db |
| 19:24.58 | Qwell | yes |
| 19:25.08 | *** join/#asterisk mad_jack (i=krzwasil@nat-mia.aster.pl) |
| 19:25.18 | SplasPood | Is there a good reason for it being that way? |
| 19:25.32 | *** part/#asterisk mad_jack (i=krzwasil@nat-mia.aster.pl) |
| 19:25.33 | *** join/#asterisk mad_jack (i=krzwasil@nat-mia.aster.pl) |
| 19:25.36 | *** part/#asterisk mad_jack (i=krzwasil@nat-mia.aster.pl) |
| 19:27.49 | NDT | codword: call em and askem if that is vinage only or open...doesn't say anything about vonage |
| 19:28.15 | gnosys_ | ok folks: disabling call waiting on SIP lines. I get the impression this is done in the handset config files, but for the life of me, I cannot figure out how to do it with Polycom 501 or with Cisco 79XX. I've googled and wiki'd and experimented, all to no avail. Has anyone done this on either of these types of phone? Care to share clues on how you did it? |
| 19:28.17 | troyb | codword any reason why it would say protocol application invalid? |
| 19:30.18 | dogtanian | gnosys_: I dial *71 from my sip line and it makes the approfriate database entry |
| 19:30.22 | *** join/#asterisk r0d3nt (i=nobody@wsip-24-234-241-145.lv.lv.cox.net) |
| 19:30.35 | gnosys_ | is this a per-call thing? |
| 19:30.53 | dogtanian | per line i think |
| 19:31.38 | alephcom | dingo: Are you making a port? |
| 19:31.45 | dogtanian | gnosys_: is it working? |
| 19:31.50 | codword | troyb: i dont think I've ran across that one before :( |
| 19:32.04 | dogtanian | gnosys_: *70 reactivates |
| 19:32.15 | gnosys_ | not sure yet, dogtanian. I'll try it. |
| 19:32.22 | dogtanian | :) |
| 19:33.48 | gnosys_ | *71 from the sip phone? Is that configured in your dial plan or hard-coded into the binaries? *71 on my SIP phone (Polycom 501) gives me a fast busy signal. Where are you dialing *71? On a zap phone or a sip phone? |
| 19:34.20 | dogtanian | it works on my cisco 7960 and also using x-lite |
| 19:34.24 | dogtanian | i'm running AAH tho |
| 19:34.30 | gnosys_ | ok |
| 19:36.11 | *** join/#asterisk syle2 (n=blah@unaffiliated/syle) |
| 19:40.27 | *** part/#asterisk codword (n=doc@cpe-71-64-192-28.cinci.res.rr.com) |
| 19:42.43 | troyb | haha i got sip loaded |
| 19:43.27 | troyb | thanks y'all :P |
| 19:43.38 | NDT | carrier is sending the 10 didgit string that was dialed to us...how do I get asterisk to read that 10digit string as the extension number? |
| 19:47.16 | *** join/#asterisk morale (i=russell@S010600111155e117.cg.shawcable.net) |
| 19:48.34 | *** join/#asterisk linlin (n=linlin@c-67-184-231-233.hsd1.il.comcast.net) |
| 19:49.37 | *** join/#asterisk TaSo (i=licucude@ool-44c784a0.dyn.optonline.net) |
| 19:53.13 | *** join/#asterisk Naturalblue (n=Kay@195.26.12.229) |
| 19:54.10 | *** part/#asterisk Naturalblue (n=Kay@195.26.12.229) |
| 19:54.17 | fugitivo | "Signalling requested on channel 1 is PRI Signalling but line is in R2 Signalling signalling" |
| 19:54.35 | fugitivo | that really means that the line is R2, or it's a wrong setting that makes asterisk believe it's R2? |
| 19:54.37 | troyb | hey guys whenim in the configuration password screen what do i enter as the default password? |
| 19:54.48 | TheCops | How can you make a return key in body of the UserEvent dialplan command ? |
| 19:54.49 | fugitivo | does asterisk actually detects the signalling on the line? |
| 20:06.31 | Qwell | troyb: of what? |
| 20:06.44 | *** join/#asterisk mogorman (n=mogorman@user-24-236-84-48.knology.net) |
| 20:07.19 | Qwell | mogorman: :) |
| 20:08.05 | *** join/#asterisk rhousand (n=rhousand@rrcs-24-199-246-10.midsouth.biz.rr.com) |
| 20:09.19 | mogorman | qwell: :) |
| 20:10.19 | *** join/#asterisk Jick (n=Jick@209-83-240-53-static.dsl.oplink.net) |
| 20:10.35 | rhousand | I am new to Asterisk and I am trying to setup asterisk at home to use iax incomming trunk. When I call the number it is bussy. |
| 20:10.37 | Qwell | I think iptables hates me |
| 20:10.38 | Jick | hey guys |
| 20:10.45 | gnosys_ | dogtanian, do me a fav and peek in your extensions.conf file on your AAH box (or better yet, do a "show dialplan" on your AAH console. Is *71 mapped to anything there? |
| 20:11.49 | mjmtaiwan | rhousand: the guys helped me out a while ago with a busy signal. In my case I didn't have the right configuration in my extentions.conf file. |
| 20:13.21 | mjmtaiwan | I hate to admit it but I need more help on that exten.conf file... its at >>http://pastebin.com/495365 |
| 20:13.33 | mjmtaiwan | this time... no calls out |
| 20:13.48 | Qwell | What might cause this after a reboot, during ztcfg or modprobe wctdm? ZT_CHANCONFIG failed on channel 1: No such device or address (6) |
| 20:15.15 | *** join/#asterisk jsmith (n=jsmith@smithfam.dsl.xmission.com) |
| 20:15.16 | kram | doesn't see your card? |
| 20:15.42 | Qwell | kram: it would appear that way |
| 20:16.11 | Qwell | lspci sees it fine |
| 20:16.37 | Jick | can Asterisk allow a dialplan to be set up such that an incoming call rings multiple SIP phones on the network, giving the call to the first person who answers? |
| 20:16.47 | Qwell | Jick: sure |
| 20:16.52 | Qwell | Dial(SIP/101&SIP/102) |
| 20:16.59 | Jick | ahh, great |
| 20:17.07 | Jick | thanks Qwell :) |
| 20:17.10 | Qwell | np |
| 20:19.51 | kram | qwell: check subvendorid |
| 20:20.04 | Qwell | kram: I think it's because I recompiled my kernel, and didn't redo zap |
| 20:20.07 | kram | qwell: might have to compile the driver with the matchall stuff set in the short term |
| 20:20.57 | mogorman | omg kram is in #asterisk |
| 20:20.59 | JunK-Y | ghey Qwell u comin to etel ? |
| 20:21.05 | Qwell | JunK-Y: I am! |
| 20:21.08 | Qwell | JunK-Y: mind a msg? |
| 20:21.13 | Chuji | kram is always here |
| 20:21.16 | denon | mogorman: he's often in asterisk, he just avoids you .. |
| 20:21.20 | Chuji | just a lurker |
| 20:21.25 | Chuji | like denon |
| 20:21.25 | *** join/#asterisk kiwnix (n=egarcia@229.red-82-158-154.user.auna.net) |
| 20:21.25 | mogorman | i know he does |
| 20:21.32 | denon | hmm? im no lurker |
| 20:21.33 | mogorman | because he is afraid |
| 20:21.39 | mogorman | afreaid of the threat i represent |
| 20:21.43 | mogorman | he feels the power |
| 20:21.44 | mogorman | ^_^ |
| 20:21.52 | rob0 | afraid he will have to give you a pay raise!! |
| 20:21.52 | denon | what threat? the threat of misspelling things? |
| 20:21.54 | denon | :) |
| 20:22.03 | mogorman | yes |
| 20:22.05 | rhousand | mjmtaiwan: I thank i found it thanks |
| 20:22.08 | mogorman | well its from some movie |
| 20:22.12 | mogorman | cant remeber which |
| 20:22.14 | denon | Digiam Asturisk ruulez |
| 20:22.31 | mogorman | ward yo! |
| 20:23.04 | mjmtaiwan | :) |
| 20:23.16 | mogorman | lol rob0 |
| 20:24.16 | denon | woah, ogorman gets paid? |
| 20:24.20 | denon | when did this happen? |
| 20:24.26 | mogorman | well its like this denon |
| 20:24.30 | mogorman | one day mark came to me |
| 20:24.34 | mogorman | and trippled my salary |
| 20:24.38 | denon | 0 * 3 |
| 20:24.44 | mogorman | exactly |
| 20:24.48 | mogorman | sshh now |
| 20:24.50 | mogorman | ^_^ |
| 20:24.52 | kram | lol |
| 20:24.55 | kram | mogorman why ain't u here |
| 20:24.58 | Qwell | yep, that did it |
| 20:25.03 | mogorman | you at work? |
| 20:25.03 | kram | whatdya think this is, the weekend? |
| 20:25.05 | Qwell | silly kernels |
| 20:25.06 | mogorman | yeah |
| 20:25.11 | denon | kram: I thought we agreed that his salary should be water and muffins |
| 20:25.12 | mogorman | let me go become mog_work |
| 20:25.18 | mogorman | whats this water |
| 20:25.23 | mogorman | i only know redbull |
| 20:25.28 | mogorman | and red delight |
| 20:25.39 | mogorman | sunny delight + redbull = red delight |
| 20:25.44 | denon | ew |
| 20:25.45 | mogorman | mmmm mmmm good |
| 20:25.46 | denon | sounds uh .. wrong |
| 20:25.48 | iDunno | redbull == EVIL |
| 20:25.54 | denon | redbull does weird things to your system .. |
| 20:25.56 | iDunno | Coffee is much better for your health. |
| 20:25.57 | msw | redbull is the nectar of life |
| 20:26.07 | iDunno | no, redbull is the nectar of death. |
| 20:26.08 | denon | yeah, if life is intended to be lived in hell .. |
| 20:26.11 | msw | heh |
| 20:26.12 | iDunno | and it smells of baby puke. |
| 20:26.13 | *** join/#asterisk mishehu (n=mishehu@crosscreek.cartissolutions.net) |
| 20:26.17 | iDunno | and it tastes crap. |
| 20:26.21 | denon | nah, more like cought syrup |
| 20:26.22 | mogorman | speaking of im going to where there is free redbull |
| 20:26.23 | mogorman | bye |
| 20:26.24 | *** part/#asterisk mogorman (n=mogorman@user-24-236-84-48.knology.net) |
| 20:26.50 | denon | hmm .. he's going to work at Microsoft? |
| 20:27.28 | iDunno | I thought microsoft provided free crack?! |
| 20:28.23 | denon | nah, free beverages |
| 20:28.37 | denon | though, it is redmond/seattle .. |
| 20:30.37 | *** join/#asterisk jtodd (n=jtodd@ti.fox-den.com) |
| 20:33.32 | mjmtaiwan | Has anyone had problems with ingnorepat? |
| 20:36.48 | Qwell | mjmtaiwan: What types of problems? |
| 20:37.51 | mjmtaiwan | Well.. it doesn't seem to give me dial tone when pressing 9 for an outside line |
| 20:43.31 | jsmith | mjmtaiwan: Paste your extensions.conf at asterisk.pastebin.ca, and I'll look at it |
| 20:43.46 | mjmtaiwan | thanks |
| 20:44.21 | mjmtaiwan | Its at >> http://pastebin.com/495365 |
| 20:44.44 | *** join/#asterisk bruce_rmz (n=Bruce@201.135.32.68) |
| 20:45.05 | bruce_rmz | hi... |
| 20:45.09 | jsmith | mjmtaiwan: You do realize that you have ignorepat set to 8 instead of 9, right? |
| 20:45.13 | jsmith | Howdy bruce_rmz |
| 20:45.14 | bruce_rmz | i'm new to asterisk |
| 20:45.29 | bruce_rmz | hi smith |
| 20:45.34 | mjmtaiwan | yes. tried 9 too |
| 20:45.51 | Qwell | What type of phone? |
| 20:45.55 | bruce_rmz | i've a question for you all, hoping you can help me out |
| 20:45.55 | jsmith | mjmtaiwan: What type of phone are you dialing from... |
| 20:46.01 | jsmith | bruce_rmz: Go ahead and ask... |
| 20:46.19 | mjmtaiwan | budgetone 101 , aastra 911, Hello Kitty analog |
| 20:46.34 | jsmith | mjmtaiwan: ignorepat only works for analog phones |
| 20:46.39 | mjmtaiwan | ah... |
| 20:46.48 | jsmith | mjmtaiwan: With SIP phones, each vendor handles it differently |
| 20:46.57 | bruce_rmz | i've been reading and I want to set up voicemail boxes... theres a lot of info, using addmailbox cmd... |
| 20:47.14 | jsmith | bruce_rmz: Just edit voicemail.conf ... no need to run addmailbox, etc. |
| 20:47.16 | bruce_rmz | but I can not find that particular script in the system |
| 20:47.18 | mjmtaiwan | OK... I know where to start... Did you see anything in the extentions.conf that would indicate why I can't call out local |
| 20:47.26 | bruce_rmz | ok... |
| 20:47.27 | mjmtaiwan | I can call international... but local... no |
| 20:47.32 | jsmith | mjmtaiwan: I'll take a closer look. |
| 20:48.00 | bruce_rmz | so , jus using voicemail.conf, the boxes can be created? |
| 20:48.36 | mjmtaiwan | bruce - remember you'll need to set up sendmail to have that nice unified messaging function. |
| 20:48.45 | jsmith | bruce_rmz: Yes... if you define it in voicemail.conf, and a call comes in, the mailbox will get created |
| 20:49.07 | jsmith | mjmtaiwan: Explain to me which context the calls from your phone come into. |
| 20:49.15 | bruce_rmz | I thoutgh it was neccesary to use the addmailbox... so it can create the folders once used... nice |
| 20:49.23 | mjmtaiwan | yea... its kindofa mess |
| 20:49.28 | bruce_rmz | That's better I think... :) |
| 20:49.31 | jsmith | bruce_rmz: Nope, that hasn't been needed in several years... |
| 20:49.36 | mjmtaiwan | the broadvoice goes to from-broadvoice |
| 20:49.44 | bruce_rmz | ok... |
| 20:49.52 | mjmtaiwan | which won't take s (I guess) |
| 20:50.04 | jsmith | mjmtaiwan: The way I see it, you're letting anyone who dials in via broadvoice make long distance calls for free... |
| 20:50.19 | bruce_rmz | Thanks a lot Jsmith... I Appretiate your helpful Advice... Ill be rounding here... |
| 20:51.25 | mjmtaiwan | hmm... (hears phone ring) |
| 20:51.25 | jsmith | mjmtaiwan: That's a "bad thing" :-) |
| 20:51.25 | bruce_rmz | I must unplug for a while... |
| 20:51.25 | bruce_rmz | but I'll keep around... |
| 20:51.26 | jsmith | bruce_rmz: If you don't know about it yet, check out the Asterisk book at www.asteriskdocs.org -- free downloads! |
| 20:51.26 | bruce_rmz | Thx Jsmith... |
| 20:51.26 | jsmith | no problem |
| 20:51.26 | bruce_rmz | Ok |
| 20:51.26 | gnosys_ | hey folks, i'm trying to figure out how to deactivate the call-waiting pips that I hear when I'm on the phone and I get an incoming call. I'm using a Cisco7960 and a Polycom 501 SIP handset. dogtanian here mentioned that *71 from his Cisco7960 (connected to an AAH box) works for him. I'm reviewing the configurations for an AAH box, and I see a statement: DBdel(CW/${CALLERIDNUM}) that seems to be the action statement. How would I map thi |
| 20:51.26 | bruce_rmz | I will |
| 20:51.54 | mjmtaiwan | jsmith: I need to divide the extentions conf properly .. I'm in the process of trying to figure out how ... where to put what. |
| 20:51.59 | jsmith | gnosys_: I'll be AAH looks in the DB to see if that key is set, and if it's not set, then it won't send the second call to the phone. |
| 20:52.06 | jsmith | s/be/bet/ |
| 20:52.39 | *** join/#asterisk AlexCTI (n=alex@weston-69.65.87.240.myacc.net) |
| 20:52.43 | gnosys_ | Hmmm... |
| 20:53.00 | bruce_rmz | Thx... It's nice to get fast response from you... Surely, i'll come back... :) (Haven't use the IRC for years i think) |
| 20:54.18 | *** part/#asterisk AlexCTI (n=alex@weston-69.65.87.240.myacc.net) |
| 20:54.21 | mjmtaiwan | jsmith: If I remove include => from-broadvoice from [inside] that would take care of the security problem? |
| 20:54.26 | gnosys_ | jsmith: if that's the case (seems like a reasonable guess to me), then it seems like an awful kludge... isn't there an elegant way to turn off this feature that these SIP handsets have for call-waiting? |
| 20:54.30 | *** join/#asterisk AlexCTI (n=alex@weston-69.65.87.240.myacc.net) |
| 20:54.54 | jsmith | mjmtaiwan: No. |
| 20:55.07 | mjmtaiwan | hm. |
| 20:55.13 | dogtanian | gnosys_ / jsmith: it will send a second call to teh phone if its for a different line |
| 20:55.21 | dogtanian | ie. different line on teh same phone |
| 20:55.26 | AlexCTI | Hi,somedobe can help me to set zaptel on debian? |
| 20:55.44 | jsmith | mjmtaiwan: You'd need to remove the exten => _1NXXNXXXXXX lines from the [from-broadvoice] context |
| 20:56.02 | gnosys_ | dogtanian: when that happens, do you still hear the "boop" tones (call waiting pips)? |
| 20:56.38 | jsmith | dogtanian: OK, that's good to know. Never really had a need to use AAH, so it was just a guess on my part |
| 20:56.46 | dogtanian | gnosys_: yes i do... but i think its the phone generating the boop's rather than the pbx |
| 20:56.55 | gnosys_ | Aha! |
| 20:57.21 | gnosys_ | So the fix of *71 in AAH doesn't really fix the problem... |
| 20:57.24 | mjmtaiwan | ok.. then how do I make long distance calls w/BV... different context? |
| 20:57.26 | dogtanian | jsmith: heh. yeah - i only found it out last night once i'd applied the bug-fix for ring strategy = hunt |
| 20:57.31 | tzafrir_laptop | AlexCTI, what's the problem? what Debian exactly? |
| 20:58.08 | AlexCTI | My problem is that i did the install of zaptel but my system is not taken it |
| 20:58.15 | dogtanian | gnosys_: the *71 will mean you won't get two sets of pip's (one set from teh pbx and another from teh cisco) |
| 20:58.27 | mjmtaiwan | AlexCTI: did you do the install in the correct order? |
| 20:58.27 | *** join/#asterisk dudes (n=dudes@12-215-32-62.client.mchsi.com) |
| 20:58.38 | AlexCTI | I think so, |
| 20:59.04 | AlexCTI | maybe that could be the poblem |
| 20:59.22 | AlexCTI | How can I confirm it? |
| 20:59.26 | gnosys_ | dogtanian: not sure I follow... you mean that you think the *71 DOES do something to the PBX? (ie, not route another call to that line?) |
| 21:00.01 | mjmtaiwan | checking my book |
| 21:00.08 | gnosys_ | dogtanian: would you try something for me? |
| 21:00.27 | AlexCTI | where is it? |
| 21:00.49 | gnosys_ | try changing your SIPDefault.cnf for your Cisco phones to include this line: |
| 21:00.50 | gnosys_ | call_waiting: 0 |
| 21:00.56 | dogtanian | gnosys_ in AAH the *71 and *70 sets/unsets a database flag for call waiting on eaxk sip extension |
| 21:00.59 | gnosys_ | and then reboot your 7960 |
| 21:00.59 | dogtanian | ...however... |
| 21:01.22 | gnosys_ | eaxk? |
| 21:01.39 | *** join/#asterisk ryan__ (n=rhousand@rrcs-24-199-246-10.midsouth.biz.rr.com) |
| 21:01.43 | mjmtaiwan | AlexCTI: did you have any issues with linux26? |
| 21:01.56 | dogtanian | on teh cisco 79xx, if you are on an active sip call, and another sip line on the same phone is ringing... the firmware [i suspect its the firmware] on teh phone generates a pip on teh active line |
| 21:02.07 | AlexCTI | at this moment not yet |
| 21:02.19 | jsmith | AlexCTI: Try running "ztcfg -vvv", and let me know what the output is. |
| 21:02.37 | AlexCTI | Command not found |
| 21:02.54 | gnosys_ | dogtanian: what's eaxk? each? |
| 21:02.59 | AlexCTI | sorry, wrong type |
| 21:03.02 | dogtanian | gnosys_: yeah - i've just confirmed what i said above |
| 21:03.04 | jsmith | AlexCTI: Did you type "make install" after compiling? |
| 21:03.11 | dogtanian | *each :) |
| 21:03.20 | dogtanian | sorry :) |
| 21:03.28 | gnosys_ | did you try my suggestion with SIPDefault.cnf? |
| 21:03.38 | AlexCTI | line 0: Unable to open master devive /dev/zap/ctl |
| 21:03.47 | mjmtaiwan | AlexCTI: No, I ment when you installed Asterisk you had to compile Zaptel |
| 21:03.52 | jsmith | AlexCTI: Have you loaded the kernel modules? |
| 21:04.00 | Qwell | and are you using udev? |
| 21:04.09 | jsmith | AlexCTI: type "lsmod" and look for "zaptel" |
| 21:04.15 | AlexCTI | ok |
| 21:05.18 | tzafrir_laptop | AlexCTI, what's the rest of the error message? "no such file" or "no such device"? |
| 21:05.29 | tzafrir_laptop | lsmod | grep ^zaptel |
| 21:05.37 | tzafrir_laptop | (pastable in the channel) |
| 21:06.39 | tzafrir_laptop | AlexCTI, if you don't have the ztcfg user-space, apt-get install zaptel. |
| 21:06.49 | tzafrir_laptop | IIRC it has hardly changed |
| 21:06.50 | dogtanian | gnosys_: i've just had a look in the cisco phone config section in AMP... and there IS an option to turn call-waiting onn/off |
| 21:07.12 | AlexCTI | lsmod |grep ^zaptel don't show me anything |
| 21:07.24 | tzafrir_laptop | so the module zaptel is not loaded. |
| 21:07.25 | dogtanian | gnosys_: although the config is a php page so i'm not sure which file/key the php toggles |
| 21:07.39 | AlexCTI | the whole error on is |
| 21:07.49 | dogtanian | gnosys_: if you're really desperate i could prolly work it out somehow |
| 21:07.50 | tzafrir_laptop | AlexCTI, what version of Debian? |
| 21:08.09 | AlexCTI | let me confrm |
| 21:08.21 | kFuQ | http://www.linuxtoday.com/ are a bunch of fucking sellouts -- look at all the linux bashing shit on the front page courtesy of M$ |
| 21:08.32 | kFuQ | what a load of shit |
| 21:08.38 | AlexCTI | 2.4.27-2-386 |
| 21:08.50 | tzafrir_laptop | So it is Sarge, I figure. |
| 21:09.08 | Qwell | kFuQ: a real news site will post news, good or bad |
| 21:09.17 | Qwell | kFuQ: If you want only pro-Linux news, go read /. |
| 21:09.20 | gnosys_ | I'm guessing that the AMP setting does the same thing as the *71. Communicates to *. I would very much like to find out exactly what those settings are telling * to do. But would you try my SIPDefault.cnf suggestion first? If there are two sets of pips (one from *, one from the Cisco), then I have turned one off, and just need to figure out how to turn the other off. It seems that I have the Cisco pips turned off (call_waiting: 0) and |
| 21:09.24 | Qwell | If you want to know what's REALLY happening...find a real news site |
| 21:09.45 | AlexCTI | it wan on the debian webside |
| 21:10.10 | dogtanian | gnosys_: yup you're right... it will be in SIPdefault.cnf |
| 21:10.15 | *** join/#asterisk Cybertoy (n=maxim@dsl254-123-241.nyc1.dsl.speakeasy.net) |
| 21:11.14 | dogtanian | gnosys_: and SPdefault.cnf is included in the file SIP[00mac_address_of]phone.cnf |
| 21:11.15 | tzafrir_laptop | AlexCTI, the latest build of zaptel for your kernel is of zaptel 1.2.0 ... I hope to upload 1.2.1 in a matter of hours |
| 21:11.16 | Qwell | kFuQ: but, think of it this way...even microsoft.com will sometimes put high severity patches on its front page |
| 21:11.30 | tzafrir_laptop | http://rapid.sunsite.dk/experimental/ |
| 21:11.48 | tzafrir_laptop | deb http://rapid.sunsite.dk/ experimental/ #it's also a deb-src |
| 21:11.56 | *** join/#asterisk [BaByFaCe] (n=Connecti@62.162.14.82) |
| 21:12.20 | dogtanian | gnosys_: so after making the change in SIPdefault.cnf.. i suspect you'd have to do a superficial edit/save cycle on each SIP[mac].cnf file for it to be included |
| 21:12.26 | AlexCTI | so, is better if i wait? |
| 21:12.31 | alephcom | Is anybody here using FreeBSD who would like to try out a couple of ports. I made a port for asterisk-perl and the POE asterisk stuff. |
| 21:12.53 | gnosys_ | you'll need to reboot the phone (or there may be a less drastic way to reload the file, but rebooting works) |
| 21:12.53 | kFuQ | but shit like "RadioShack Saves Millions of Dollars by Choosing Windows Over Linux" on a linux site is a bit fucking lame |
| 21:12.59 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
| 21:13.01 | Qwell | kFuQ: not really |
| 21:13.12 | gnosys_ | dogtanian: did you get my private message? |
| 21:13.37 | dogtanian | yup - i'm there now |
| 21:14.00 | kFuQ | that's like ford bashing the shit outta chevy on chevrolet.com |
| 21:14.03 | AlexCTI | tzafri_laptop, do i need add these sources on my machine? |
| 21:14.28 | Qwell | kFuQ: No, it's like fordfans.com reporting that Chevy beat Ford in their earnings |
| 21:14.44 | Qwell | (no, that isn't a real domain) |
| 21:15.12 | tzafrir_laptop | AlexCTI, for one deb you can simply wget it and install it with dpkg -i |
| 21:15.37 | tzafrir_laptop | If you starte t obe beaten by dependencies, apt-get is the smart choice |
| 21:16.03 | AlexCTI | i'm new on it... i'll make the easy way.. |
| 21:17.10 | mjmtaiwan | Thanks for the help ya'll - I'm going now to watch the Redskins beat the Bucks! Back in 4 hours to beg for more help. |
| 21:17.51 | *** join/#asterisk [hC] (i=turnerd@66.199.130.40) |
| 21:18.01 | *** join/#asterisk angler- (n=opera@pcp01540308pcs.huntsv01.al.comcast.net) |
| 21:18.46 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
| 21:20.04 | AlexCTI | tzafrir_laptop: which one is the one to download there? |
| 21:20.29 | dudes | AlexCTI - I'll take care of that later |
| 21:20.34 | *** join/#asterisk brockj49464 (n=brockj49@63.87.56.159) |
| 21:20.45 | *** join/#asterisk MatsK (i=enforcer@c83-253-29-22.bredband.comhem.se) |
| 21:21.06 | AlexCTI | oki.. |
| 21:21.17 | tzafrir_laptop | http://rapid.sunsite.dk/experimental/zaptel-modules-2.4.27-2-386_1.2.0-1+2.4.27-10_i386.changes http://rapid.sunsite.dk/experimental/zaptel_1.2.0-1_i386.deb |
| 21:21.37 | tzafrir_laptop | The first one is kernel modules, the second one is userspace |
| 21:26.06 | *** join/#asterisk MeTaL (i=Windows@62.162.14.93) |
| 21:28.00 | *** join/#asterisk lorinc (n=ang@caracas-4068.adsl.interware.hu) |
| 21:28.50 | *** join/#asterisk Chiardon (n=yo@200.71.58.39) |
| 21:29.28 | Chiardon | I'm trying to compile MFC R2 |
| 21:29.45 | Chiardon | specifically speaking spandsp |
| 21:29.50 | Chiardon | can anybody give me a hand? |
| 21:30.37 | Chiardon | its asking me for libtiff but I have it installed |
| 21:30.46 | [hC] | debian? |
| 21:31.03 | [hC] | apt-get install libtiff-dev |
| 21:31.04 | Chiardon | fc2 |
| 21:31.17 | [hC] | or the equivalent in fedora core.. i dont use it so im not sure what the package is called |
| 21:31.32 | [hC] | but basically you can install the libtiff libraries, as well as the development headers package separately |
| 21:31.39 | [hC] | you need the development package as well, to compile with it. |
| 21:32.07 | Chiardon | ohhh thnx |
| 21:32.19 | Chiardon | lets see if I can find it over at rpm.pbone =) |
| 21:33.19 | [hC] | np :) |
| 21:34.19 | *** join/#asterisk kenrstone (n=krstone@ool-4573f3dc.dyn.optonline.net) |
| 21:35.53 | *** join/#asterisk MatsK (i=enforcer@c83-253-29-22.bredband.comhem.se) |
| 21:39.51 | *** join/#asterisk xtr (n=01928375@S0106000c41ed11e1.vf.shawcable.net) |
| 21:42.13 | tzafrir_laptop | [hC], debs for Sarge are availble at http://rapid.dotsrc.org/unstable/ |
| 21:42.52 | [hC] | tzafrir_laptop: ah, thanks :) |
| 21:43.04 | tzafrir_laptop | oops, misreading... |
| 21:43.41 | [hC] | ah no 1.2.1 |
| 21:44.28 | [hC] | odd, they have 1.2.1 extras etc, just no core. |
| 21:47.28 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 21:47.38 | *** join/#asterisk mog_work (n=mogorman@gateway.digium.com) |
| 21:49.26 | Chiardon | thnx |
| 21:51.14 | Qwell | omgmog |
| 21:51.26 | mog_work | omg mogog !!! |
| 21:51.37 | Qwell | ommog! |
| 21:53.20 | mog_work | oh my mog? |
| 21:53.26 | fugitivo | iuashdfiuoui1!! |
| 21:53.37 | Qwell | mog_work: yes |
| 21:53.44 | Qwell | too far? |
| 21:53.59 | mog_work | too far from waht? |
| 21:54.02 | Qwell | dunno |
| 21:54.08 | Qwell | reality |
| 21:54.29 | *** join/#asterisk ManxPower (i=ewieling@24.sub-70-210-168.myvzw.com) |
| 21:54.29 | mog_work | i havent started in on the redbull yet |
| 21:54.32 | mog_work | so im fine |
| 21:56.48 | sivana | how do I tell which type of filesystem is a partition? |
| 21:56.55 | Qwell | sivana: Is it mounted? |
| 21:56.58 | sivana | ya |
| 21:57.01 | Qwell | if so, df -T |
| 21:57.08 | Qwell | (or check /etc/mtab) |
| 21:57.11 | sivana | ext3 ? |
| 21:57.15 | Qwell | sure |
| 21:57.18 | sivana | ah |
| 21:57.25 | sivana | can I have one ext3 and one xfs? |
| 21:57.28 | Qwell | sure |
| 21:57.34 | Chiardon | hi |
| 21:57.34 | sivana | does it really matter? |
| 21:57.36 | Qwell | in fact, I do just that |
| 21:57.38 | Qwell | sivana: yes |
| 21:57.56 | Chiardon | I am now tring toinstall libmfrc2 |
| 21:57.57 | sivana | I see... so one is better than the other I presume |
| 21:58.02 | Qwell | xfs > reiser > xfs > ext3 > reiser > ext3 > ext2 |
| 21:58.07 | Chiardon | but I get an error while making |
| 21:58.08 | mog_work | ext3 over all |
| 21:58.25 | mog_work | it just always worsk |
| 21:58.31 | Qwell | I always do ext3, except my latest / partition is xfs |
| 21:58.33 | *** join/#asterisk wundaboy (n=asdf@c-24-21-100-201.hsd1.or.comcast.net) |
| 21:58.39 | mog_work | i did xfs back in 99 |
| 21:58.42 | mog_work | lost all my files once |
| 21:58.46 | mog_work | havent gone back since |
| 21:58.51 | Chiardon | I get libtool: link: only absolute run-paths allowed |
| 21:58.51 | fugitivo | how does xfs work this days? |
| 21:58.54 | sivana | ok... my / is ext3 and I have another one for "data", trying to figure out what I should make it |
| 21:59.09 | mog_work | ext3 is tried and true |
| 21:59.52 | Qwell | sivana: when in doubt, ext3 |
| 21:59.58 | sivana | ok |
| 22:00.01 | sivana | my RSG is not available :) |
| 22:00.10 | sivana | remote smart guy hehe |
| 22:00.59 | denon | mog_work: you were pretty easily intimidated into coming to work, huh |
| 22:01.15 | Qwell | he was offered redbull |
| 22:01.21 | mog_work | redbull at desk |
| 22:01.24 | denon | Qwell: and he came anyway |
| 22:01.24 | mog_work | mmm mmm good |
| 22:01.30 | mog_work | and getting to work on my project |
| 22:01.33 | mog_work | on clock ^_^ |
| 22:01.42 | denon | hehe |
| 22:01.51 | Qwell | I need to start getting paid on the weekends... |
| 22:02.22 | mog_work | well i took tuesday off too |
| 22:02.28 | Qwell | slacker! |
| 22:02.30 | mog_work | because it was while i was moving into dev |
| 22:03.04 | denon | I'm just still in awe that mog is getting paid |
| 22:03.18 | Qwell | wait, Digium pays you guys? |
| 22:03.27 | Qwell | :p |
| 22:03.30 | denon | yeah.. thats what he said |
| 22:03.33 | mog_work | should i feel insulted denon ? |
| 22:03.44 | denon | mog_work: naaah |
| 22:03.50 | denon | we just assumed that digium was truly open source <G> |
| 22:03.54 | Qwell | I just like the fact that I get to work on * at work |
| 22:04.02 | mog_work | yeah it is denon |
| 22:04.18 | mog_work | i mean any bum could come into office right now and work |
| 22:04.19 | denon | open source labor, under MIT license |
| 22:04.25 | Qwell | mog_work: brt |
| 22:04.29 | mog_work | heh |
| 22:04.34 | mog_work | ill leave the door open for ya |
| 22:05.02 | mog_work | heh |
| 22:05.05 | mog_work | i have one on my desk |
| 22:05.12 | mog_work | its my "stable" stick |
| 22:05.15 | denon | here.. print this fedex label.. |
| 22:05.22 | Qwell | denon: You're a Digium guy, right? What do you do there? |
| 22:05.29 | denon | no, im not |
| 22:05.31 | Qwell | oh |
| 22:05.56 | denon | no free redbull at my office, either |
| 22:06.21 | denon | 'course, I guess since I do a lot of the policy, I could do free redbull as a new policy.. |
| 22:06.22 | mog_work | thats obviously why you arent there |
| 22:06.35 | *** join/#asterisk darkskiez (n=mhb@bb-195-172-50-6.ukonline.co.uk) |
| 22:08.32 | Chiardon | has anybody here installed libmfcr2 ? |
| 22:08.47 | denon | mog_work: ah, but I am |
| 22:08.53 | mog_work | oh lame..... |
| 22:09.06 | mog_work | once or twice Chiardon |
| 22:09.18 | denon | mog_work: I'm just too lazy to re-nick |
| 22:09.18 | Qwell | Chiardon: What distro? |
| 22:09.29 | Chiardon | mog_work, ... I am getting a funny error message "libtool: link: only absolute runpaths are allowed" |
| 22:09.33 | Chiardon | Qwell, DC2 |
| 22:09.37 | Qwell | DC? |
| 22:09.40 | Chiardon | Qwell, FC2 |
| 22:09.40 | Qwell | FC? |
| 22:09.47 | Qwell | yum install it |
| 22:09.57 | mog_work | yeah i havent seen that error before |
| 22:10.05 | mog_work | google say anything about Chiardon |
| 22:10.16 | denon | Chiardon: fix -r |
| 22:10.20 | denon | er -R |
| 22:10.39 | denon | (look in the configure script for -R or --runpath or something) |
| 22:11.54 | Chiardon | fix -R ? |
| 22:12.38 | Chiardon | Google says its a small problem with paths |
| 22:13.02 | Chiardon | says to use ./configure --prefix=/usr/local |
| 22:13.10 | Chiardon | will try that and keep you posted =) |
| 22:14.14 | Chiardon | Ok everythings done |
| 22:14.20 | Chiardon | that worked fine =)=)=)=)=)=)=) |
| 22:14.23 | mog_work | yay! |
| 22:14.23 | Chiardon | Google r00lz |
| 22:14.24 | Chiardon | :P |
| 22:14.28 | mog_work | that it does |
| 22:14.58 | trixter | yeah soon you can get your google mail on your google pc fed by your google internet while talking on your google voice |
| 22:15.12 | trixter | they signed a deal with walmart for their own systems running googleOS |
| 22:15.21 | Qwell | no they didn't |
| 22:15.27 | trixter | they didnt? |
| 22:15.30 | trixter | ok newspapers lied |
| 22:15.34 | Qwell | nope |
| 22:15.51 | mog_work | yes |
| 22:16.09 | Qwell | trixter: surely you aren't surprised by that? |
| 22:16.49 | *** join/#asterisk Cybertoy (n=maxim@dsl254-123-241.nyc1.dsl.speakeasy.net) |
| 22:19.36 | mog_work | if you cant believe everything you read whats the point |
| 22:19.55 | msw | mog_work: I don't believe you |
| 22:20.05 | mog_work | then dont read |
| 22:20.07 | [hC] | im pretty sure a googleOS-ish something is in the works though, i have a friend working at goog who is developing option roms for some google-only thing |
| 22:20.09 | mog_work | go be illeterate |
| 22:20.13 | [hC] | either that or some special embedded device |
| 22:20.16 | trixter | yeah I know I cant believe everything I read |
| 22:20.22 | trixter | I read stuff from this channel all the time |
| 22:20.33 | mog_work | i imagine google and any other large company builds and shelves stuff all the time |
| 22:20.53 | [hC] | oh yeah for sure |
| 22:20.55 | [hC] | look at apple |
| 22:20.56 | mog_work | like apple has probably built quite a few pdas that never made it off the prototype area |
| 22:21.09 | Qwell | banks... |
| 22:21.11 | [hC] | theyve had tons of stuff thats been shelved for years |
| 22:21.14 | [hC] | look at osx86 |
| 22:21.14 | Qwell | we do that ALLL the time |
| 22:21.34 | Qwell | we spend several million dollars on a project... |
| 22:21.36 | Qwell | go over budget... |
| 22:21.38 | Qwell | and just drop it |
| 22:21.43 | mog_work | lol |
| 22:21.50 | mog_work | what kind of projects does wells fargo do? |
| 22:22.00 | mog_work | i thought they just did money and packages |
| 22:22.00 | Qwell | well, alright, here's an example |
| 22:22.02 | [hC] | banking stuff, of course. |
| 22:22.04 | mog_work | how hard is that |
| 22:22.07 | mog_work | ^_^ |
| 22:22.10 | Qwell | We have these two completely different mainframe systems for mortgage |
| 22:22.17 | Qwell | they had this project called "Project One" |
| 22:22.24 | Qwell | probably spent a good 10 milion on it |
| 22:22.29 | Qwell | completely abandoned |
| 22:22.33 | mog_work | heh |
| 22:22.41 | trixter | wells fargo hired the company I worked for to do a rather large project then after getting everything sold the division off to buy some bank |
| 22:22.42 | Qwell | they were trying to merge the two systems |
| 22:23.00 | Qwell | trixter: sounds like WF :P |
| 22:23.11 | Qwell | in fact, we bought a bank just to get one of the two aforementioned mainframes |
| 22:23.32 | trixter | we did the neikki investment advisors stuff - convert em from their prime mainframe to solaris and hpux -- although we were under budget and under time |
| 22:23.33 | [hC] | cheaper to buy the bank than to buy the equipment and set it up again? |
| 22:23.34 | [hC] | :P |
| 22:23.37 | Qwell | [hC]: yes |
| 22:23.40 | Qwell | :p |
| 22:23.52 | trixter | they sold that and bought some bank in the southwest |
| 22:23.53 | Qwell | trixter: under budget? BLASPHEMY |
| 22:24.05 | Qwell | trixter: how long ago? |
| 22:24.08 | trixter | it doesnt take much to make an accurate time and cost estimate |
| 22:24.12 | [hC] | ive been approached to 'buy something' to fill in budget gaps |
| 22:24.15 | [hC] | sweet, new toys! |
| 22:24.19 | trixter | mid 90s iirc |
| 22:24.21 | Qwell | heh |
| 22:24.24 | Qwell | Norwest |
| 22:24.34 | Qwell | trixter: interestingly, we're talking about the same thing |
| 22:24.56 | Qwell | probably anyhow |
| 22:25.00 | trixter | there are advantages to having all the code trnasfered rather than working part with and part without a company |
| 22:25.07 | *** join/#asterisk mrtwister (n=andrius@cable-10-68.cgates.lt) |
| 22:25.41 | trixter | plus that wasnt the first prime->sun migration we did, that was what we mainly did so we had tools and everything to convert prime fortran to C (the gpl one blew chunks at that time dunno about now) and cpl to csh |
| 22:26.35 | mrtwister | hello. it is possible to initiate asterisk call manager commands form extension (without AGI)? |
| 22:27.30 | mog_work | system |
| 22:30.41 | mrtwister | mog_work: how? i have to telnet to localhost and send commands... at least i think, have to make .sh script... i do not want to do disk operations |
| 22:30.50 | mrtwister | mog_work: then fastagi may be better |
| 22:31.11 | Qwell | What do you need to do through manager, that you can't do from an active call? |
| 22:31.34 | mrtwister | Qwell: i want to initiate two more calls (callback) |
| 22:31.39 | mrtwister | active call is trigger |
| 22:31.57 | mrtwister | call came from outside or another asterisk and it only initiating call |
| 22:32.16 | mrtwister | now i can make .call file and copy it to /var/spool/asterisk/outgoing/ |
| 22:32.19 | mrtwister | it is not stable |
| 22:32.27 | mrtwister | manager is much better |
| 22:33.02 | mrtwister | so, i need 1) execute manager commands through extension or 2) use fastagi or 3) just agi, i looking for 1st way |
| 22:34.05 | *** join/#asterisk Igbothom (n=HiltonT@office.quarkit.com.au) |
| 22:34.47 | alephcom | You could write an agi script that created the calls through the manager interface. They would not have to write call files |
| 22:37.06 | trixter | does the te110p do atm? |
| 22:38.36 | *** join/#asterisk zhao__ (n=zhao@c-24-13-6-136.hsd1.il.comcast.net) |
| 22:45.08 | trixter | alephcom: you still have reads on loading the agi unless you put everything that would be read in a ramdisk |
| 22:45.31 | Chiardon | I have a question ... |
| 22:45.48 | Chiardon | the patch that softwitch has for * with R2 |
| 22:45.55 | Chiardon | is for * 1.1 |
| 22:46.13 | Chiardon | will I break anything if I install that on * 1.2¿ |
| 22:46.19 | Qwell | likely |
| 22:46.27 | fugitivo | no, it works for 1.2 |
| 23:01.09 | *** join/#asterisk jbot_ (i=ibot@pdpc/supporter/active/TimRiker/bot/apt) |
| 23:01.09 | *** topic/#asterisk is Asterisk 1.2.1 has been released! -//- http://www.asterisk.org |
| 23:01.20 | *** part/#asterisk kareena (n=k@adsl196-104-23-206-196.adsl196-1.iam.net.ma) |
| 23:01.25 | *** join/#asterisk wundaboy (n=asdf@c-24-21-100-201.hsd1.or.comcast.net) |
| 23:01.38 | *** join/#asterisk jahani (n=k@adsl196-104-23-206-196.adsl196-1.iam.net.ma) |
| 23:01.56 | qwertz | Hi, could anybody explain what the difference between quietmp3, mp3, mp3nb, quietmp3nb in musiconhold.conf is? |
| 23:02.30 | *** part/#asterisk viLeR (i=1000@66.128.47.232) |
| 23:03.19 | *** part/#asterisk mrtwister (n=andrius@cable-10-68.cgates.lt) |
| 23:03.20 | *** join/#asterisk mcquaid (i=mcquaid@toronto-hs-216-138-233-79.s-ip.magma.ca) |
| 23:54.08 | *** join/#asterisk jbot (i=ibot@pdpc/supporter/active/TimRiker/bot/apt) |
| 23:54.08 | *** topic/#asterisk is Asterisk 1.2.1 has been released! -//- http://www.asterisk.org |
| 23:54.40 | [hC] | is it common for the rxgain/txgain settings you set at the top of zapata.conf to not apply to everything below it? I have 3 channels, the first is tuned appropriately, the 2 channels that follow it seem to be cut in half each time |
| 23:57.37 | *** join/#asterisk jbot (i=ibot@pdpc/supporter/active/TimRiker/bot/apt) |
| 23:57.37 | *** topic/#asterisk is Asterisk 1.2.1 has been released! -//- http://www.asterisk.org |
| 23:57.41 | _Sam-- | you are a wanker, and only hurting yourself by not using one. |
| 23:59.40 | *** join/#asterisk jbot (i=ibot@pdpc/supporter/active/TimRiker/bot/apt) |
| 23:59.40 | *** topic/#asterisk is Asterisk 1.2.1 has been released! -//- http://www.asterisk.org |
| 23:59.44 | trixter | I think he hurts the community at large by being behind a movement that restricts choice and mocks those that advocate choice |
