| 00:00.36 | buZz | gpx1000: btw , i dont think what you want is possible |
| 00:00.47 | buZz | you cant use audiogw other way around .. afaik |
| 00:00.50 | mmlj4 | Jizzbug: thanks |
| 00:01.01 | buZz | but if you figure it out |
| 00:01.03 | buZz | let me know :P |
| 00:01.05 | buZz | or |
| 00:01.11 | buZz | post on voip-info frontpage ^_^ |
| 00:01.24 | [av]bani | anyone wanna duplicate a MOH bug? |
| 00:01.25 | *** join/#asterisk darkskiez (n=darkskie@bb-87-81-62-203.ukonline.co.uk) |
| 00:01.38 | buZz | [av]bani: like what? |
| 00:01.48 | [av]bani | you have MOH working? |
| 00:01.53 | buZz | yep |
| 00:02.01 | [av]bani | force an extension to use gsm, and try MOH |
| 00:02.14 | [av]bani | ulaw and g729 work fine, gsm borks |
| 00:02.23 | _Sam-- | [av]bani: can i try! |
| 00:02.25 | buZz | try MOH? |
| 00:02.28 | [av]bani | yes |
| 00:02.37 | gpx1000 | buZz, thanks. |
| 00:02.37 | buZz | how would i 'try moh' on an extensions |
| 00:02.47 | buZz | just by calling in to my asterisk from gsm? |
| 00:02.55 | buZz | or call TO gsm and enable moh? |
| 00:03.03 | [av]bani | call into asterisk using gsm codec |
| 00:03.13 | buZz | i've done that |
| 00:03.15 | [av]bani | setup a test extension which just plays moh |
| 00:03.16 | buZz | works fine |
| 00:03.21 | buZz | allready have it |
| 00:03.27 | [av]bani | forced to gsm? |
| 00:03.32 | *** join/#asterisk sonicGB- (n=Miranda@138.25.71.101) |
| 00:03.33 | _Sam-- | disallow = all |
| 00:03.35 | _Sam-- | allow = gsm |
| 00:03.37 | _Sam-- | sip.conf |
| 00:03.49 | _Sam-- | or iax.conf if yo're using iax |
| 00:03.54 | [av]bani | sip show channels should show gsm for the channel |
| 00:03.56 | [av]bani | not ulaw |
| 00:04.41 | buZz | Feb 15 01:04:25 NOTICE[3503]: chan_sip.c:3593 process_sdp: No compatible codecs! |
| 00:04.44 | buZz | :O |
| 00:04.57 | [av]bani | ha, your phone refusing gsm |
| 00:05.06 | buZz | well no |
| 00:05.11 | buZz | my inbound SIP |
| 00:05.18 | [av]bani | or you have borked settings somewhere |
| 00:05.26 | [av]bani | maybe you have only ulaw in [global] |
| 00:05.28 | buZz | but |
| 00:05.29 | buZz | <PROTECTED> |
| 00:05.36 | *** join/#asterisk franck (n=franck@tikiwiki/franck) |
| 00:05.40 | *** part/#asterisk w32 (n=123@adsl-70-224-74-204.dsl.sbndin.ameritech.net) |
| 00:05.46 | franck | Hi all |
| 00:05.53 | franck | How can I test ENUM with * |
| 00:06.03 | [av]bani | sounds like your phone is refusing to use gsm |
| 00:06.12 | [av]bani | or something |
| 00:06.15 | buZz | i have no physical phone |
| 00:06.24 | _Sam-- | [av]brainz: who else beside myself has duplicated that bug |
| 00:06.27 | buZz | i have sip clients , all using GSM allready |
| 00:06.30 | _Sam-- | desides |
| 00:06.34 | _Sam-- | damn i cant type today |
| 00:06.36 | [av]bani | dunno |
| 00:06.44 | _Sam-- | the gsm moh bug i mean |
| 00:06.45 | buZz | only my 'in/outbound' sip account didnt run on gsm |
| 00:06.48 | _Sam-- | im the only one you found to test it? |
| 00:06.48 | buZz | but it says its able |
| 00:07.07 | [Latre] | Jizzbug: do you have any problems with this phone? |
| 00:07.15 | [av]bani | _Sam--: so far |
| 00:07.28 | [av]bani | _Sam--: what timing source you have, ztdummy? |
| 00:08.24 | [Latre] | Jizzbug: i have a problem, the other side listenme but, i dont........... |
| 00:08.24 | _Sam-- | nazaptel 221344 51 wcte11xp |
| 00:08.24 | *** part/#asterisk frenzy (n=frenzy@196.45.144.40) |
| 00:08.24 | [av]bani | k, you have "real" timer then, so they cant blame ztdummy eithe r:) |
| 00:08.38 | *** join/#asterisk Kumbang (n=kumbang@167.205.24.5) |
| 00:08.42 | _Sam-- | i could try it on my other asterisk machine at my clients place with ztdummy |
| 00:08.50 | _Sam-- | but i have to wait for them to leave before i go messing around |
| 00:09.12 | _Sam-- | then i can just register this gxp their * and see |
| 00:09.20 | Idle_ | hmmm... where can I find some good docs about modules, like Goto, or Dial? |
| 00:09.34 | [av]bani | _Sam--: if you have non gxp it would be a good test too |
| 00:09.56 | _Sam-- | i have software clients i should be able to try |
| 00:09.57 | *** join/#asterisk w32 (n=123@adsl-70-224-74-204.dsl.sbndin.ameritech.net) |
| 00:10.01 | _Sam-- | but no other ip phones |
| 00:11.00 | *** join/#asterisk emergion (n=pauly@84.133.233.220.exetel.com.au) |
| 00:11.26 | emergion | Hello could someone tell me where I can find some information on simple call forwarding ? IE put someone through to extension #xxx |
| 00:13.59 | sonicGB- | Hi folks. Question about asterisk capabilities if I may: I have a hardware ip (SIP only) phone. My 'grand plan' is to install ipphone(1) in my office and have it register itself with asterisk(1) in |
| 00:14.07 | sonicGB- | office. Have asterisk(1) automagically pass *all* outbound calls from ipphone(1) to asterisk(2) at home, then have asterisk(2) take care of establishing a SIP or IAX2 call with a destination ipphone(2). |
| 00:14.14 | sonicGB- | Is this within the realms of possibility? (I wasn to get my facts straight before I get started!) :-) |
| 00:14.47 | sonicGB- | s/wasn/want/; |
| 00:15.14 | *** join/#asterisk rayvd (i=rayvd@arthur.bludgeon.org) |
| 00:15.50 | *** join/#asterisk sack (n=sack@196.Red-83-50-146.dynamicIP.rima-tde.net) |
| 00:15.55 | rayvd | How can I control how often NOTIFY's are sent to client ATA's (SIP)? |
| 00:16.02 | *** join/#asterisk rogier (n=rogier@16-65-dsl.ipact.nl) |
| 00:16.13 | rayvd | When a customer has voicemail, Asterisk seems to be sending a NOTIFY anywhere from every 10 seconds to every 30 seconds--seems completely random. |
| 00:16.42 | rayvd | Causes the ATA to "ring" as it's supposed to... I can disable that, but wondered where it is I can tell Asterisk to not notify quite so often.. |
| 00:18.40 | [Latre] | Jizzbug: |
| 00:23.20 | *** part/#asterisk pbd (n=plancomm@200.168.1.84) |
| 00:23.55 | *** join/#asterisk areski (n=areski@172.Red-83-34-12.dynamicIP.rima-tde.net) |
| 00:24.39 | *** join/#asterisk trelane (i=trelane@2001:4830:150c:0:20d:61ff:fe31:a6c) |
| 00:25.29 | *** join/#asterisk CpuID (n=nathan@dsl-202-173-176-82.qld.westnet.com.au) |
| 00:27.22 | CpuID | hey ppls, anyone here ever had an issue where an fxs interface starts acting up every 12-24hrs? ive got a setup thats using 2 fxo modules and 1 fxs module on a tdm400p, 2 landlines into the 2 fxo's, and a cordless handset on the fxs...seems every morning when you first goto pickup the cordless handset, you get a weird dialtone, which doesnt allow making outbound calls, and inbound calls which ring the fxs module, dont cause the cor |
| 00:27.23 | CpuID | dless to ring, ideas anyone? |
| 00:27.55 | CpuID | theres no mailbox specified at all on the fxs channel, so its not a MWI dialtone :) |
| 00:28.34 | *** join/#asterisk st3v (n=spj@netblock-66-218-41-231.dslextreme.com) |
| 00:33.39 | *** join/#asterisk silly_ (n=silly@cpe-24-174-162-34.satx.res.rr.com) |
| 00:36.24 | rayvd | HOMESTEAD! |
| 00:39.58 | *** join/#asterisk WasPhantom (n=neil@203-86-192-98.tasman.net) |
| 00:42.59 | *** join/#asterisk mrdigital (n=Mrdgitia@pool-68-236-15-51.phil.east.verizon.net) |
| 00:43.32 | mrdigital | anyone know of a Auto-attendent free trial with toll free # that does not require a credit card to use the trial |
| 00:46.38 | *** join/#asterisk file (n=joshnet@mctnnbsa24w-142167033017.pppoe-dynamic.nb.aliant.net) |
| 00:46.58 | *** join/#asterisk bugz (n=bugz@cpe-24-27-67-66.houston.res.rr.com) |
| 00:47.04 | *** join/#asterisk brodiem (i=1000@cpe-66-69-222-36.austin.res.rr.com) |
| 00:47.16 | bugz | anyone know how to change the softbutton appearance on a cisco 7940 |
| 00:48.05 | bugz | like as in the display on the phone looks messed up where the buttons are |
| 00:48.07 | brodiem | I'm wondering how SIP phones can display a queue status (i.e. number of callers waiting, hold time). Is this something that Asterisks-compatible phones will automatically be able to retrieve, or does Asterisk need to be configured to send some type of SIP text to the phones, or?? |
| 00:48.32 | bugz | brodiem: there is almost a way to do that with polycoms and their config files |
| 00:48.46 | bugz | the display on those phones is a lightweight html parser |
| 00:48.51 | bugz | like the cisco phones |
| 00:49.01 | bugz | the only difference is the cisco phones dont have ANY documentation |
| 00:49.05 | bugz | and the polycoms have a little |
| 00:49.20 | brodiem | bugz, I know I've seen it done before, it was an * PBX and Aastra 480i phones |
| 00:49.37 | brodiem | but I don't seem to be able to find any docs relating to it |
| 00:49.39 | *** join/#asterisk bjohnson (n=bjohnson@i216-58-67-128.cybersurf.com) |
| 00:51.34 | [av]bani | brodiem: its phone-vendor-specific |
| 00:51.45 | *** join/#asterisk kFuQ (n=somedude@c-67-185-114-199.hsd1.wa.comcast.net) |
| 00:51.51 | *** join/#asterisk jorgito (n=knoppix@gw-u2-1.cd-t.cz) |
| 00:51.52 | jorgito | hi |
| 00:51.57 | [av]bani | brodiem: http://www.o2m8.com/modules.php?name=News&file=article&sid=25 |
| 00:52.10 | bugz | jesus i cant wait for digium to make a phone FOR asterisk... |
| 00:52.17 | jorgito | my dtmf is not working in asterisk LOL, |
| 00:52.21 | _Sam-- | [av]bani: if we had the damn minibrowser like ive been asking for our gxps could do it! |
| 00:53.07 | [av]bani | _Sam--: or even sendtext |
| 00:53.24 | [av]bani | hmm |
| 00:53.38 | [av]bani | you could possibly fake it with an extension with an empty ring |
| 00:53.46 | [av]bani | and then just hack up callerid to send messages over |
| 00:53.58 | [av]bani | yucky though |
| 00:54.23 | brodiem | [av]bani ahh so basically I would need to write an XML/PHP script to talk to a web interface or some other real time source it could talk to? |
| 00:54.34 | [av]bani | brodiem: that is one option |
| 00:55.20 | [av]bani | _Sam--: you need to replace the bulb i think |
| 00:55.24 | *** part/#asterisk bkw_ (n=bkw_@adsl-70-142-51-127.dsl.tul2ok.sbcglobal.net) |
| 00:55.28 | brodiem | [av]bani, keep going :) |
| 00:55.37 | mrdigital | [av]bani: do yuo know of a company giving auto-attendant free trials? |
| 00:55.46 | [av]bani | brodiem: it totally depends on the phone you have. every vendor has different ways of doing it. |
| 00:56.06 | [av]bani | mrdigital: no |
| 00:56.15 | _Sam-- | its a low wattage bulb |
| 00:56.26 | _Sam-- | i need an LED |
| 00:56.56 | brodiem | [av]bani, I didn't buy any phones yet, I've been searching for a decent phone. I need about 25 of them..any recommendations? Would like it to cost less than the 480i though |
| 00:57.19 | [av]bani | if you want decent app support like that, youre going to spend about $200 a phone |
| 00:57.20 | bugz | brodiem: what do the 480i cost? |
| 00:57.30 | _Sam-- | [av]bani: can you put a feature request in: you shoiuldnt have to reboot the damn phone to add account 2/3/4 sip info |
| 00:57.31 | [av]bani | nothing sub-$200 can do it yet |
| 00:57.35 | brodiem | I thought they were about $250 a piece |
| 00:57.47 | [av]bani | ~phones |
| 00:57.48 | jbot | i heard phones is at http://bani.anime.net/phones/ |
| 00:57.48 | bugz | you can get an ip501 for about that |
| 00:57.53 | [av]bani | brodiem: ip501 wont do it |
| 00:57.55 | bugz | probably less |
| 00:58.06 | [av]bani | ip501 has no support |
| 00:58.08 | [av]bani | for that |
| 00:59.22 | [av]bani | the low end aastras will do xml, but lollerskates trying to do anything useful with 3 text lines |
| 00:59.31 | [av]bani | i think only 2 are usable |
| 00:59.37 | *** join/#asterisk essaredee (n=srd@cpc1-lich2-0-0-cust257.brhm.cable.ntl.com) |
| 00:59.48 | _Sam-- | [av]bani: chekcing client's * now for gsm/moh prob |
| 00:59.53 | brodiem | hmm |
| 00:59.54 | _Sam-- | first im just listening to it on Ulaw |
| 00:59.59 | _Sam-- | sounds fine, mpg123 on this one |
| 01:00.05 | [av]bani | k |
| 01:00.10 | [av]bani | gsm... dun dun duuuuuuunnnnnnn |
| 01:00.15 | _Sam-- | will switch it to gsm now |
| 01:00.59 | _Sam-- | sure enough....breaking up until mic input |
| 01:01.00 | brodiem | [av]bani The snom 360? |
| 01:01.08 | [av]bani | yay, so its independent of native or mpg123 |
| 01:01.22 | _Sam-- | yep. |
| 01:01.26 | _Sam-- | BUT |
| 01:01.28 | _Sam-- | gxp phone |
| 01:01.28 | essaredee | I've got asterisk setup on a colo'd machine and I've tried changing sip.conf/extensions.conf around abit from looking at examples on the web, I can place calls, but cannot receive them (fully) |
| 01:01.33 | _Sam-- | have to try another client in a minute |
| 01:01.43 | [av]bani | brodiem: yes, snom 360 does it... but i would not recommend a snom 360 |
| 01:02.05 | essaredee | If I put in a non-existant context it'll complain when I dial the voip number up, nothing happens if the context= line is removed |
| 01:02.24 | Qwell | essaredee: it goes to default |
| 01:02.28 | brodiem | [av]bani which would you recommend? |
| 01:02.32 | essaredee | right |
| 01:02.38 | [av]bani | brodiem: what are your requirements? |
| 01:03.10 | essaredee | if I use a context which exists, nothing comes up in the asterisk terminal |
| 01:03.23 | *** join/#asterisk _deg_ (n=deg@201.22.46.190.adsl.gvt.net.br) |
| 01:03.38 | Qwell | essaredee: run a sip debug |
| 01:04.03 | brodiem | [av]bani Basically it needs to support PoE and be able to display some queue stats. I'm not sure of what the extent of what you can do with programmable soft keys are, but adding soft keys for say agent logins, DND (pause/unpause from queue) would be a bonus |
| 01:05.02 | [av]bani | well, if you can tolerate the bugs, snom 360 is mostly usable |
| 01:05.18 | [av]bani | ip601 is more solid, though it is more expensive |
| 01:06.02 | [av]bani | you get a larger LCD though, and its higher rez and greyscale |
| 01:06.25 | *** join/#asterisk angom_w (n=angom@red-corp-200.38.16.10.telnor.net) |
| 01:06.33 | brodiem | [av]bani, thanks |
| 01:06.40 | [av]bani | havent used the aastra, but it looks like it would do what you want also |
| 01:06.50 | _Sam-- | [av]bani: most phones have to reboot when you change account specific info? |
| 01:07.02 | [av]bani | one gotcha with the 480i though -- it is PoE _only_ -- it doesnt even have a wallwart :D |
| 01:07.04 | jorgito | what is relaxdtmf in sip.conf? |
| 01:07.27 | [av]bani | _Sam--: the snom is the only phone so far that doesnt seem to require a reboot every time you move the mouse |
| 01:07.54 | [av]bani | _Sam--: polycoms are even worse than the gxp2000. you can at least change a few things on the gxp2k without rebooting. you sneeze at a ip601 and you have to reboot |
| 01:09.39 | essaredee | Nothing is catching my eye in the debug |
| 01:10.11 | essaredee | except for a SIP/2.0 404 not found |
| 01:10.17 | [av]bani | _Sam--: and the ip601 takes 3+ minutes to boot... |
| 01:12.04 | bugz | jesus... |
| 01:12.12 | bugz | because the softkeys are corrupt on this whole batch of phones |
| 01:12.19 | bugz | i have to design a whole new button schema |
| 01:12.26 | bugz | just to fix some stupid shit with this phone |
| 01:12.44 | [av]bani | what phone? |
| 01:12.50 | bugz | 7940 |
| 01:12.52 | [av]bani | ha |
| 01:13.05 | bugz | i despise cisco |
| 01:13.12 | Qwell | corrupt? |
| 01:13.14 | Qwell | I'll buy them cheap |
| 01:13.18 | Qwell | $100 each |
| 01:13.23 | bugz | heh |
| 01:13.47 | bugz | i just want to know how to fix this without having to do what i think im having to do |
| 01:13.54 | Qwell | upgrade the firmware |
| 01:14.00 | Qwell | if you can't do that...then that |
| 01:14.09 | bugz | when the phone boots the softkeys say "<p> </p>" |
| 01:14.11 | Qwell | s your problem. Should have done the research before buying |
| 01:14.12 | bugz | instead of "Dial" |
| 01:14.30 | bugz | Qwell: hey im just an admin |
| 01:14.40 | bugz | nobody asks me, but i can bet the will from now on |
| 01:14.47 | Qwell | how many phones are we talking? |
| 01:14.49 | bugz | this is costing us too much effort |
| 01:14.49 | bugz | 30 |
| 01:14.57 | Qwell | I can fix them, if you give me 2 :p |
| 01:15.27 | bugz | why dont you just let me in on your little secret |
| 01:15.32 | Qwell | I already did.. |
| 01:15.39 | bugz | the firmware... |
| 01:15.42 | Qwell | indeed |
| 01:15.58 | bugz | you believe this to be the problem? |
| 01:16.05 | Qwell | wouldn't have said it otherwise |
| 01:16.15 | bugz | i hate these phones |
| 01:16.21 | bugz | what kind of company puts out a phone |
| 01:16.28 | bugz | that you have to upgrade firmware on right out of the box |
| 01:16.31 | Qwell | That works very well, if you know what you're doing? Cisco |
| 01:16.43 | bugz | thus paying twice for the phone |
| 01:16.55 | Qwell | a smartnet contract is like $8... |
| 01:17.00 | Qwell | That's hardly 2x the price |
| 01:17.04 | bugz | i'll put ip601's everywhere before this deal is done |
| 01:17.14 | bugz | its still rediculous |
| 01:17.17 | Qwell | well, like I said...I'll buy them from you, for $100 each ;] |
| 01:18.19 | bugz | voipsupply.com has a retarded sales staff |
| 01:18.22 | bugz | for the record |
| 01:18.24 | *** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox) |
| 01:18.27 | Qwell | yes, they do |
| 01:19.26 | [av]bani | bugz: thats what the intarweb is for |
| 01:19.50 | bugz | hmm.. i could swear that when the phones first booted up they had buttons displaying correctly |
| 01:19.56 | bugz | a "factory reset" didnt do the trick |
| 01:20.08 | bugz | and i havent attempted anything with the firmware |
| 01:20.23 | bugz | i cant believe nobody else has had this problem to speak of |
| 01:20.46 | [av]bani | everyone else has smartnet |
| 01:21.16 | bugz | we should all move to call manager too i guess |
| 01:21.20 | [av]bani | there you go |
| 01:21.30 | bugz | and then we can all install windows millenium edition on our pbx's |
| 01:21.50 | [av]bani | i like your positive attitude |
| 01:21.55 | [av]bani | very refreshing |
| 01:22.09 | bugz | well, i had a refreshing day |
| 01:22.13 | bugz | ;) |
| 01:22.35 | [av]bani | sounds like it, the kind of refreshing you get after unloading a 12ga in the office? |
| 01:23.19 | *** join/#asterisk justinu (n=Justin@cpe-72-129-86-208.socal.res.rr.com) |
| 01:23.28 | *** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox) |
| 01:25.12 | bugz | i dont know about that but i know id enjoy a bottle of whiskey and a couple boxes of 7940's for target practice |
| 01:25.23 | bugz | i wonder if theyd fit in a skeet shooter |
| 01:25.39 | bugz | haha man that would be fun |
| 01:25.51 | bugz | i feel better just thinking about watching one fly apart in a million pieces |
| 01:26.08 | *** join/#asterisk s34n (n=s34n@ip-206-159-190-125.mvdsl.com) |
| 01:29.54 | Qwell | bugz: well...if you had a smartnet contract... |
| 01:29.59 | Qwell | bugz: You'd get a free replacement |
| 01:30.00 | Qwell | :P |
| 01:30.05 | bugz | for 1 location |
| 01:30.13 | bugz | we will never put in another cisco phone as long as i work here |
| 01:30.23 | Qwell | because you can't figure them out? |
| 01:30.29 | bugz | i know whats up with them |
| 01:30.43 | Qwell | like I said...send me two, and I'll have them fixed |
| 01:31.00 | bugz | they ring dont they... messed up firmware is different |
| 01:31.17 | bugz | i know what we will do... we will pitch it to the customer |
| 01:31.18 | bugz | and say |
| 01:31.40 | bugz | "since you like cisco so much you can buy the smartnet support you need to make the phone behave as advertised ;D " |
| 01:32.03 | Qwell | $8 X 30 = $240... |
| 01:32.06 | Qwell | hardly a large investment |
| 01:32.12 | bugz | $8? |
| 01:32.16 | justinu | can anyone just pay the 8 bucks for that account? |
| 01:32.17 | bugz | wait... |
| 01:32.18 | Qwell | For the second time, yes |
| 01:32.27 | bugz | i missed that one |
| 01:32.32 | bugz | i must have been to busy complaining |
| 01:32.35 | justinu | lol |
| 01:32.40 | bugz | i was told it was $85 |
| 01:32.42 | justinu | that happens a lot around here |
| 01:32.56 | Qwell | you were told wrong, or you asked the wrong question |
| 01:32.56 | bugz | this changes things considerably |
| 01:33.14 | *** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox) |
| 01:33.37 | bugz | ok i feel even better |
| 01:33.46 | bugz | so what is the best bang for buck when it comes to an ip phone? |
| 01:33.49 | Qwell | Cisco |
| 01:33.53 | justinu | polycom |
| 01:33.56 | Qwell | They work, and they work well |
| 01:34.30 | WasPhantom | I like my polycom |
| 01:34.33 | s34n | I wrote a macro that included this: |
| 01:34.37 | s34n | exten => s,2,Dial(SIP/${ARG1}@switch,20) |
| 01:34.55 | robin_z | I wrote a macro that included this: |
| 01:35.06 | s34n | when executed, * is putting a space between the / and the ARG1 |
| 01:35.20 | Qwell | s34n: ${ARG1} probably contains a space |
| 01:35.36 | robin_z | exten => dial/sip/$myExten if BUSY |
| 01:35.45 | robin_z | it dont work, because its gibberish :) |
| 01:36.06 | s34n | Qwell: that would be too obvious :( |
| 01:39.09 | bugz | hmm |
| 01:39.28 | bugz | i wonder what it would be worth to my employer for me to learn XML services |
| 01:39.43 | Qwell | about a buck |
| 01:39.58 | justinu | 50 cents |
| 01:40.06 | bugz | we'll see about that |
| 01:40.16 | [av]bani | bugz: gxp2000 is a lot of phone for $80 |
| 01:40.24 | Qwell | I'd just hire somebody who already knew about it. ;) |
| 01:40.57 | justinu | so my customer is just not happy with the gxp... they've decided to order polycoms |
| 01:41.08 | [av]bani | heh, i bet theyll be suprised |
| 01:41.13 | [av]bani | "3 minutes to boot? wtf" |
| 01:41.22 | Qwell | "omg, a phone that works?!" |
| 01:41.26 | justinu | heh |
| 01:41.31 | Qwell | I'm sure that'll trump the 3 min boot time |
| 01:41.36 | [av]bani | 'why does it reboot when i sneeze' |
| 01:41.53 | FuriousGeorge | show of hands, how many people got their valentine an snom 360 this year? |
| 01:42.08 | Qwell | FuriousGeorge: I got mine a cisco 7985. That's how I roll |
| 01:42.08 | Qwell | :P |
| 01:42.16 | FuriousGeorge | keep'em up if you did that because you grilfriend is a rig running asterisk |
| 01:42.33 | FuriousGeorge | Qwell: you a pimp, yo |
| 01:42.35 | FuriousGeorge | play on |
| 01:42.41 | justinu | word |
| 01:43.23 | Qwell | I need to get me a 7985 or two somehow... |
| 01:45.15 | [av]bani | ha |
| 01:47.34 | justinu | if I really wanted one, i would just buy it... |
| 01:47.35 | bugz | i will probably install a thousands of these phones this year... |
| 01:47.58 | Qwell | justinu: They're quite expensive, heh |
| 01:48.10 | Qwell | not worth the price |
| 01:48.21 | justinu | i've spent countless thousands on other hobbies |
| 01:48.27 | justinu | tens of thousands |
| 01:49.05 | justinu | quite foolish |
| 01:49.10 | s34n | bugz: how much do you pay for a standard office phone? |
| 01:49.17 | justinu | 170 |
| 01:49.34 | bugz | standard? our standard is ip501's |
| 01:49.46 | bugz | and we do literally sell those for $1000 a seat |
| 01:49.46 | justinu | that's my standard too |
| 01:49.55 | s34n | standard non-ip |
| 01:49.59 | bugz | and we pay about 170 |
| 01:50.07 | bugz | s34n: no such thing in my office |
| 01:50.19 | bugz | we fill boxes full of nortel and avaya junk |
| 01:50.26 | bsdfreak | heh |
| 01:50.47 | s34n | bugz: That junk was sold by me, for $200 ea |
| 01:50.58 | bugz | haha... nice |
| 01:51.07 | bugz | then we put it on ebay and you buy it back |
| 01:51.09 | bugz | works for me |
| 01:51.16 | s34n | I now sell 501's for $200 ea |
| 01:51.36 | bugz | im no fan of 501's |
| 01:51.54 | bugz | or the ip line of polycoms in general only for the fact that you have to mess with them for 10 minutes out of the box |
| 01:52.00 | s34n | So I don't think ip phones are really expensive in comparison to the alternatives. |
| 01:52.11 | bugz | i can script everything on the server but you STILL have to put IP info in |
| 01:52.16 | bugz | passwords, blah |
| 01:52.43 | bugz | id think there should be a way to put a phone's FTP config on it automatically |
| 01:52.45 | bugz | but i cant find it |
| 01:52.47 | justinu | what's wrong with the 501? |
| 01:53.05 | Qwell | I can have a Cisco configured in about 30 seconds |
| 01:53.14 | Qwell | dhcp, option 66, tftp, bam, done |
| 01:53.22 | bugz | Qwell: write me a how-to and i will send you a cisco POE cable |
| 01:53.46 | bugz | ok how bout * |
| 01:53.51 | Qwell | I can already do POE |
| 01:54.01 | Qwell | I've already stated my offer :P |
| 01:54.02 | Qwell | ] |
| 01:54.43 | [hC] | Anyone here able to give me a hand with a 7970? |
| 01:54.51 | [hC] | I dont suppose you're near one of yours eh qwell? :) |
| 01:54.56 | *** join/#asterisk SibRphrek (n=SibRphre@user-12lccke.cable.mindspring.com) |
| 01:54.57 | Qwell | [hC]: The thing I saw on the users list? |
| 01:54.57 | Qwell | nope... |
| 01:55.06 | [hC] | Qwell: yeah. |
| 01:55.41 | Qwell | I remember seeing a fix for that in the archives... |
| 01:56.57 | bugz | proprietary #$^$^ phone configs.. @$!#$^!# |
| 01:57.13 | bugz | if it werent for open source none of this would be possible |
| 01:57.24 | bugz | yet people continue to enable companies to hinder themselves |
| 01:58.17 | Qwell | voip wouldn't be possible without open source? |
| 01:58.38 | bugz | Qwell: ok for ATT |
| 01:58.42 | bugz | and Cisco |
| 01:58.53 | *** join/#asterisk Kumbang (n=kumbang@167.205.24.5) |
| 01:58.55 | Qwell | voip wouldn't be possible for Cisco without open source? |
| 01:59.35 | bugz | no, they would, along with microsoft, have perverted so many standards by now without open source that the state of this 'union' would really be trite |
| 01:59.47 | *** join/#asterisk tengulre11 (n=tengulre@61.185.224.66) |
| 01:59.58 | bugz | i think the product market is healthy because of open standards |
| 02:00.11 | bugz | but at some level its always a hush hush secret you have to pay money for |
| 02:00.13 | Qwell | So, you think that phones with different featuresets...should all have the same config? |
| 02:00.30 | bugz | only because some huge company would walk all over them, steal the code under some stupid IP law |
| 02:01.01 | bugz | Do you think that all phones should have secret configuration options that nobody can use but the manufacturer unless you pay heavy licensing fees? |
| 02:01.22 | Qwell | right...because it's impossible to search cisco.com for the sample configs |
| 02:01.24 | *** join/#asterisk moprilo (n=jjohn@201.192.107.58) |
| 02:01.37 | tengulre11 | HI,ALL! i m backing!! say good to everyone! |
| 02:01.46 | bugz | I see no indication of the nature of my problem with these phones other than from you. |
| 02:01.49 | *** join/#asterisk SibRphrek (n=SibRphre@user-12lccke.cable.mindspring.com) |
| 02:02.03 | bugz | Which, in itself, is a 'give me some shit and I might help' approach. |
| 02:02.19 | moprilo | i have a latency ranging between 160ms to 230ms. is there i way to make asterisk always wait 250ms for the packets? is high but not too high. i'm having jitter problems. :S |
| 02:02.22 | Qwell | If you have an asterisk problem...sure, I'll help you |
| 02:02.29 | Qwell | buy things not related to asterisk...I'm a consultant |
| 02:02.33 | *** join/#asterisk austinnichols10 (n=austinni@dsl-10-169.cofs.net) |
| 02:02.33 | moprilo | and the jitter option on the iax is not work'n |
| 02:02.35 | Qwell | s/buy/but/ |
| 02:03.03 | Qwell | I already gave you a suggestion on how to fix it. If you want somebody to fix it for you...it's gonna cost |
| 02:03.05 | bugz | Fair enough for you to defend your livelihood. |
| 02:03.22 | bugz | Its gonna cost anyway since Id have to pay for an upgrade to fix a bug. |
| 02:03.29 | Qwell | it's not a bug |
| 02:03.42 | bugz | User error? |
| 02:03.47 | bugz | Plug it in and it breaks? |
| 02:03.49 | bugz | My bad. |
| 02:03.52 | Qwell | You bought these phones used |
| 02:03.59 | bugz | Hahaha nope. |
| 02:04.07 | Qwell | These phones have been connected to CCM |
| 02:04.22 | bugz | You can tell this much without a doubt? |
| 02:04.35 | bugz | On what grounds, may I ask? |
| 02:04.36 | Qwell | indeed |
| 02:04.40 | Qwell | experience? |
| 02:04.55 | bugz | I'll need a little more than that when I RMA these. |
| 02:05.13 | bugz | They are to be new, in fact, they were supposed to come with a SIP image on them. |
| 02:05.32 | bugz | I know Cisco doesn't do this but partners might. |
| 02:05.34 | Qwell | Then they most certainly aren't new |
| 02:05.45 | Qwell | What firmware IS on them? |
| 02:05.46 | mrdigital | qwell whats a good company that provides toll free #'s and auto-attendants and has a free trial |
| 02:05.58 | bugz | 7 something... i dont remember |
| 02:06.06 | Qwell | of what? sip? |
| 02:06.13 | bugz | Skinny. |
| 02:06.16 | Qwell | "They were supposed to come with SIP", implies they didn't |
| 02:06.17 | Qwell | indeed |
| 02:06.27 | Qwell | Who did the upgrade? |
| 02:06.39 | Qwell | obviously they did |
| 02:06.53 | bugz | Not me. I wasnt very familiar with Cisco phones until today. |
| 02:07.14 | bugz | So now I have grounds to RMA this stuff up someones ass. |
| 02:07.24 | Qwell | What locale are they using? |
| 02:07.31 | bugz | Thats all I need. |
| 02:07.53 | bugz | Tell you the truth Qwell I had enough time to get it dialing and duck out. |
| 02:08.08 | bugz | I can probably telnet into though... |
| 02:08.15 | *** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox) |
| 02:09.06 | Qwell | You can't telnet to sccp phones |
| 02:09.08 | essaredee | how do you delete a voicemail greeting? |
| 02:09.28 | essaredee | if you can't do it from the phone, where is the file stored? |
| 02:09.45 | Qwell | essaredee: /var/spool/asterisk/context/mailbox/greet.* |
| 02:09.50 | bugz | well that sucks |
| 02:09.54 | essaredee | cheers |
| 02:09.59 | Qwell | I probably missed a dir |
| 02:11.49 | justinu | http://justinu.smugmug.com/photos/56381841-O.jpg |
| 02:12.09 | bugz | Version 3.1(MF.G2) |
| 02:12.12 | bugz | there we go |
| 02:12.27 | Qwell | bugz: pastebin that whole page |
| 02:12.34 | Qwell | That's from the http, I assume? |
| 02:12.41 | bugz | lynx, yeah |
| 02:12.56 | Qwell | yeah, pastebin the whole thing |
| 02:13.10 | bugz | http://pastebin.com/555330 |
| 02:13.30 | brodiem | [av]bani, you still around? |
| 02:13.48 | Qwell | yeah, that's old |
| 02:14.00 | Qwell | That's the stock firmware |
| 02:14.55 | Abydos313 | you guys see this? a 10 dollar packet8 adapter and service for 1 month required |
| 02:15.02 | Abydos313 | can i post a link? |
| 02:15.06 | Qwell | Abydos313: yeah |
| 02:15.08 | Qwell | but it's locked I bet |
| 02:15.10 | Abydos313 | http://www.tigerdirect.com/applications/searchtools/item-details.asp?EdpNo=1786796&Tab=5 |
| 02:15.30 | Abydos313 | maybe it can be unlocked. other for packet8 where |
| 02:15.54 | [av]bani | moo |
| 02:15.59 | Qwell | Abydos313: more trouble than it's worth |
| 02:17.12 | *** part/#asterisk Paco-Paco (n=elb@12-208-106-139.client.insightBB.com) |
| 02:17.15 | *** join/#asterisk lodeon (n=not4u@h119n5c1o1023.bredband.skanova.com) |
| 02:17.20 | Abydos313 | probably right |
| 02:17.22 | Qwell | eww, it's Ethan! |
| 02:17.30 | Qwell | I hate that guy...heh |
| 02:17.38 | ManxPower | Packet8 devices have not been unlockable for at least a year. |
| 02:17.46 | Qwell | okay, maybe not hate |
| 02:17.49 | Qwell | but he sure is a dick |
| 02:18.19 | Abydos313 | ManxPower thx for info |
| 02:23.32 | _Sam-- | justinu: i need your gxps |
| 02:23.39 | _Sam-- | can i buy the used ones? |
| 02:23.40 | justinu | heh, ok i'll tell the customer that |
| 02:23.45 | justinu | probably |
| 02:24.21 | kFuQ | http://tinyurl.com/7es9t <-- LOL wtf? |
| 02:25.35 | [av]bani | _Sam--: looks like the gsm bug is gxp only! |
| 02:25.41 | Abydos313 | that's beyond funny |
| 02:25.54 | [av]bani | dunno why i thought the snom had the problem earlier, i could swear i duped it last night |
| 02:26.04 | _Sam-- | justinu: aside from just 'not liking' the phone...is there any constructive criticism they gave that we could be used to make it better? |
| 02:26.16 | _Sam-- | i didnt try with any other clients yet |
| 02:26.25 | _Sam-- | let me try with this iax client on my desktop |
| 02:26.40 | Nivex | kFuQ: that is sooooo wrong |
| 02:26.57 | kFuQ | yah.. |
| 02:26.59 | [av]bani | nikon d2h... |
| 02:27.05 | kFuQ | YEEEEEHHHAAAAAAAAWWWWWWWWWWWW!!!!!!!!!! |
| 02:27.37 | kFuQ | dukes of hazzard meets jackass |
| 02:27.38 | kFuQ | aha |
| 02:28.41 | _Sam-- | [av]bani: moh is fine in my iax client :) |
| 02:28.46 | _Sam-- | using gsm |
| 02:28.55 | _Sam-- | actual format = gsm, |
| 02:29.09 | _Sam-- | sounds mint |
| 02:29.51 | _Sam-- | typical! |
| 02:30.01 | essaredee | I'm trying to use playback on a file which exists silence/1 but it's saying it doesnt' (format ulaw) |
| 02:30.11 | Qwell | typical of a $20 phone, with a $100 pricetag |
| 02:30.31 | _Sam-- | told you they were P'sOS :) |
| 02:31.23 | justinu | _Sam--: no, they had no comments about the phone features, etc. |
| 02:32.17 | _Sam-- | Qwell: when you use beta software i guess you expect the unexpected. |
| 02:32.19 | *** join/#asterisk relyuhcs (n=Bosco@c-69-247-188-78.hsd1.al.comcast.net) |
| 02:32.49 | _Sam-- | have the option to stay with crappy firmware, thats broken...or upgrade to beta firmware...thats crappy and broken too :) |
| 02:32.50 | _Sam-- | your choice! |
| 02:33.08 | Qwell | another phone is my choice |
| 02:33.10 | *** join/#asterisk citats (n=james@mrplow.gnuinternet.com) |
| 02:34.31 | *** join/#asterisk warthawg (n=warthawg@cpe-66-68-176-215.austin.res.rr.com) |
| 02:34.59 | warthawg | anyone familiar with AGI that can answer a simple question, please? |
| 02:35.12 | a1fa | _Sam-- |
| 02:35.17 | warthawg | well, maybe 2 questions |
| 02:35.29 | a1fa | dang dude u are still here |
| 02:35.36 | justinu | sam is hardcore |
| 02:35.44 | warthawg | does asterisk pass the same data to every agi? |
| 02:35.47 | _Sam-- | im just trying to learn :) |
| 02:35.55 | justinu | hardcore o g |
| 02:36.00 | [av]bani | _Sam--: thats what i wanted to confirm |
| 02:36.13 | JunK-Y | shit, that stream kicks ass. |
| 02:36.17 | _Sam-- | whats up a1f |
| 02:36.37 | justinu | bani: you know the snom 360 freezes when you try and make a TLS connection to a SER box w/ a self signed cert? |
| 02:36.42 | a1fa | not much dude |
| 02:36.50 | a1fa | my girl is forcing me to watch gilmore girls |
| 02:36.54 | justinu | lol |
| 02:36.59 | a1fa | i feel like i've lost my soverignity |
| 02:37.10 | _Sam-- | yeah wait til you're married :) |
| 02:37.13 | warthawg | heh |
| 02:37.14 | [av]bani | justinu: havent tried srtp/sips yet. wouldnt suprise me though, they arent hard to crash :) |
| 02:37.15 | _Sam-- | i just got done watching one with mine |
| 02:37.25 | [av]bani | justinu: you can crash them with attended transfers |
| 02:37.37 | justinu | bani: crash w/ reboot, or freeze? |
| 02:37.41 | [av]bani | justinu: freeze |
| 02:37.44 | justinu | jeeze |
| 02:37.49 | [av]bani | like, reach around and unplug |
| 02:37.53 | a1fa | god bless wifi and laptops |
| 02:37.54 | justinu | yep, i know the drill |
| 02:38.29 | justinu | well, i know they work with a real cert |
| 02:38.31 | a1fa | _Sam-- : man.. i had a good ride.. redlining in first, second and third gear through the streets ;P |
| 02:38.38 | justinu | so i'm trying to get a 3rd party signed cert |
| 02:38.39 | a1fa | 14k RPM |
| 02:38.40 | a1fa | hehehe |
| 02:38.40 | [av]bani | justinu: i dont think they work at all with self signed carts |
| 02:38.42 | justinu | i just don't wanna pay with one |
| 02:38.50 | justinu | they don't |
| 02:38.50 | [av]bani | justinu: one of the complaints about it |
| 02:38.57 | justinu | so it just freezes? |
| 02:38.58 | a1fa | the bike just doenst sound as good below 12k RPM |
| 02:38.59 | justinu | wtf is that? |
| 02:39.04 | a1fa | wtf is what |
| 02:39.06 | austinnichols10 | a1fa: been registered for 24 hours now! |
| 02:39.10 | [av]bani | its called normal behavior |
| 02:39.13 | justinu | ok |
| 02:39.22 | a1fa | austin |
| 02:39.25 | a1fa | what did you change? |
| 02:39.27 | [av]bani | Welcome to snom, where regression testing is a completely foreign concept |
| 02:39.39 | austinnichols10 | 45 second re-registers for now |
| 02:39.41 | justinu | bani: you have any links re: self signed certs? |
| 02:39.53 | a1fa | not bad |
| 02:40.07 | austinnichols10 | I've kind of arrived at the conclusion that I should only use phones that support keepalive at the remote site (bye-bye 7960s) |
| 02:40.17 | a1fa | hahaha |
| 02:40.22 | a1fa | maybe firmware upgrade buddy? |
| 02:40.53 | austinnichols10 | that's what I'm thinking. My brother works for Cisco and he's checking on what's in the next release |
| 02:40.56 | _Sam-- | austinnichols101: these are the ones behind your wrt? |
| 02:41.01 | austinnichols10 | yes |
| 02:41.04 | _Sam-- | i think maybe you should look at firmware for that thing |
| 02:41.07 | _Sam-- | what are you running on it? |
| 02:41.20 | austinnichols10 | currently I'm running openwrt |
| 02:41.28 | a1fa | you can open a TAC case and have them add that feautre |
| 02:41.50 | _Sam-- | mine is running the linksys regular stuff , maybe the dumber the better? |
| 02:41.52 | Qwell | umm |
| 02:41.54 | austinnichols10 | so I went from linksys to dd-wrt and then to openwrt. The ser implementation in dd-wrt is very broken |
| 02:42.03 | a1fa | _Sam-- : same here.. no problems |
| 02:42.04 | Qwell | austinnichols10: Why not just use qualify=yes? |
| 02:42.09 | austinnichols10 | sam: that's where I really want to be |
| 02:42.10 | w32 | hey I'm getting kinda irritated with aah- Does any one have any how to resources on configuring asterisk on a regular linux box |
| 02:42.16 | a1fa | Qwell : its not working out for him |
| 02:42.34 | austinnichols10 | qwell: with the openwrt setup the qualify wouldn't keep the connections open - they would fail after 60 seconds |
| 02:42.35 | [hC] | soooo anyone here set up a cisco 7970? :) |
| 02:42.44 | Qwell | austinnichols10: Then it' |
| 02:42.45 | [hC] | Im just gonna keep asking until someone bites.. :) |
| 02:42.49 | Qwell | s a problem with the router, not the phone |
| 02:42.51 | austinnichols10 | my next step is to drop back to the linksys original and start retesting |
| 02:42.52 | a1fa | i had the same problem with BT100 and Checkpoint FW1 |
| 02:42.59 | austinnichols10 | qwell: I think so too |
| 02:43.00 | a1fa | i figured out what it was |
| 02:43.09 | [hC] | NAT Timeout? :) |
| 02:43.19 | a1fa | [hC] : 86000s |
| 02:43.31 | a1fa | NAT timeout == connection state timeout |
| 02:43.34 | _Sam-- | it seems like it something to do with stateful packet |
| 02:43.43 | [av]bani | justinu: if you'll notice, snom has no idea how to design a phone UI |
| 02:43.45 | austinnichols10 | I've been looking at those new linksys 942s thinking that they may be a nice option. |
| 02:44.03 | a1fa | austinnichols101: check our WIP300 |
| 02:44.11 | a1fa | check out WIP300 |
| 02:44.14 | Abydos313 | i use freeman basic 1.04 on my linksys |
| 02:44.15 | austinnichols10 | url... |
| 02:44.16 | [hC] | I quite like the 941's ive been using. this last batch i received seems to have some sort of major bug that ive got opened with sipura |
| 02:44.21 | Abydos313 | it's pretty damn nice |
| 02:44.31 | [av]bani | [hC]: dont count on it being fixed anytime soon |
| 02:44.35 | Abydos313 | i can email a copy |
| 02:44.42 | [hC] | a1fa: so it was the state timeout then. If your timeout was set to 86000 seconds, that seems plenty high to sustain? |
| 02:44.58 | [hC] | [av]bani: h eh. they already responded to my email i sent this morning once, escalating it... |
| 02:44.59 | moprilo | can i make asterisk work with g723? |
| 02:45.14 | a1fa | [hC] : i send out about 30gb a day :P |
| 02:45.14 | moprilo | maybe a package to download :?? jej |
| 02:45.20 | w32 | no one ? |
| 02:45.25 | a1fa | http://us.gizmodo.com/gadgets/gadgets/linksys-wip300-voip-handset-154011.php |
| 02:45.46 | austinnichols10 | a1fa: I love the comments on the wip300: "the question is if it will work with Skype". |
| 02:45.51 | a1fa | lol |
| 02:45.55 | a1fa | some idiot wrote that |
| 02:45.59 | austinnichols10 | big time |
| 02:46.02 | a1fa | n0b |
| 02:46.07 | [av]bani | [hC]: sipura got bought out by cisco and the sipura software development seems to be in hiatus since then. |
| 02:46.20 | [av]bani | well, it was sipura->linksys->cisco |
| 02:46.22 | Qwell | no, sipura got bought by linksys |
| 02:46.27 | a1fa | yeah |
| 02:46.38 | [av]bani | since then, support has dropped to zero |
| 02:46.38 | a1fa | linksys pwns u |
| 02:46.43 | [hC] | [av]bani: well, this is pretty major. out of 10 ive purchased, te last batch of 4 like to reboot at random times, for seemingly no reason. :) |
| 02:46.47 | a1fa | linksys wants your money |
| 02:46.55 | [hC] | seems like a defect, since none of the others do it |
| 02:47.01 | austinnichols10 | abydos313: what does freeman basic do for you? I read that it was just a slight bit of difference from sveasoft standard |
| 02:47.10 | [av]bani | theres quality control for you |
| 02:47.21 | *** join/#asterisk bjohnson (n=bjohnson@i216-58-67-128.cybersurf.com) |
| 02:47.34 | a1fa | _Sam-- : have you used that voicechanger app? |
| 02:47.51 | _Sam-- | hopefully you'll go there again so i can afford more phones :)nope i never even heard of it |
| 02:47.55 | _Sam-- | er |
| 02:48.00 | austinnichols10 | sam: I'm shooting for as simple as possible. The idea is to be able to set up a remote site with a minimum of hardware that's all available at the local best buy, etc. |
| 02:48.05 | _Sam-- | hah, id ont know how that came back out! |
| 02:48.05 | a1fa | _Sam-- : voip-info voice changer |
| 02:48.13 | austinnichols10 | not there yet but getting closer... |
| 02:48.26 | [av]bani | a1fa likes to prank call |
| 02:48.32 | a1fa | i love to prank call |
| 02:48.32 | _Sam-- | austinnichols101: sounds reasonable enough |
| 02:48.35 | a1fa | people get freaked out |
| 02:48.57 | a1fa | they want to call 5-0! |
| 02:48.57 | *** part/#asterisk w32 (n=123@adsl-70-224-74-204.dsl.sbndin.ameritech.net) |
| 02:49.07 | a1fa | so i try not to prank with the voice changer |
| 02:49.07 | _Sam-- | people still 'phreak'? |
| 02:49.12 | a1fa | only people that i know |
| 02:49.31 | a1fa | _Sam-- : 26000 |
| 02:49.38 | _Sam-- | i remember in the 80s like 82...making conference calls and LD stuff, dialers, etc |
| 02:49.47 | _Sam-- | i didnt know people still do it |
| 02:49.55 | _Sam-- | i forget what we used to get on back then |
| 02:50.00 | [av]bani | they bruteforce peoples pbxes these days |
| 02:50.04 | [av]bani | and steal credit cards |
| 02:50.07 | a1fa | yeah dude |
| 02:50.08 | _Sam-- | hell yeah |
| 02:50.08 | austinnichols10 | here's a good one - my PRI started giving fast-busy on toll calls this afternoon. I did a few test calls and then captured a pri debug and sent it along to the telco once they started finger pointing. The freaked out at the amount of info I had available... |
| 02:50.12 | _Sam-- | the good ol days |
| 02:50.16 | a1fa | someone was trying to bruteforce my pbx |
| 02:50.31 | a1fa | they were putting all these usernames :P |
| 02:50.31 | _Sam-- | i forget what we use to call the conference rooms/calls |
| 02:50.32 | [av]bani | thats about the extent of it these days... most real phone phreaking is long dead |
| 02:50.34 | a1fa | and they are idiots |
| 02:50.40 | a1fa | i use md5sum for my usernames |
| 02:50.42 | austinnichols10 | a1fa: you need an intruShield |
| 02:50.45 | [av]bani | everyone else has either moved on or gone to prison |
| 02:50.49 | a1fa | austinnichols101: i got ids |
| 02:50.51 | *** part/#asterisk relyuhcs (n=Bosco@c-69-247-188-78.hsd1.al.comcast.net) |
| 02:50.59 | a1fa | austinnichols101: i also got firewalls.. |
| 02:51.21 | _Sam-- | but savvy phreakers could still be listening on calls and getting CC info and whatnots? this goes on to this day? |
| 02:51.21 | austinnichols10 | a1fa: just a quick call to your isp and the authorities and no more bruteforcce |
| 02:51.27 | a1fa | nah |
| 02:51.29 | [av]bani | _Sam--: not likely |
| 02:51.46 | a1fa | nothing would happen |
| 02:51.48 | [av]bani | phreakers dont really exist anymore, not like they did in teh 80s |
| 02:51.54 | _Sam-- | i see |
| 02:52.05 | austinnichols10 | a1fa: although ips is a much better solution than ips |
| 02:52.07 | _Sam-- | calls are too cheap to risk it anyway :) |
| 02:52.10 | austinnichols10 | sorry ids |
| 02:52.11 | [av]bani | like i said they either moved on to other things or they're currently in prison |
| 02:52.34 | *** join/#asterisk brookshire[home] (n=matt@68.62.235.16) |
| 02:52.44 | _Sam-- | [av]bani: how old are ya if you dont mind me asking? |
| 02:52.55 | JunK-Y | brookshire[home]: i owe u one pitcher. |
| 02:53.01 | [av]bani | too old! |
| 02:53.07 | brookshire[home] | only one? |
| 02:53.11 | [av]bani | i was just bitching about that to cow orkers the other day |
| 02:53.26 | a1fa | authorities suck |
| 02:53.27 | _Sam-- | lol...i know the feeling. ok how about this..how old were you in 1982? :) |
| 02:53.30 | JunK-Y | oky, 3, but u owe me 2! |
| 02:53.39 | JunK-Y | that live stream fucking rocks! |
| 02:53.39 | a1fa | i like pre-emptive defence |
| 02:53.45 | a1fa | or rather, pro-active defence |
| 02:53.55 | Qwell | defence? |
| 02:54.00 | a1fa | defense |
| 02:54.14 | a1fa | pizza hut sucks |
| 02:54.16 | a1fa | 1hour |
| 02:54.39 | De_Mon | free pizza doesn't suck |
| 02:54.40 | _Sam-- | why didnt you get some slices when you were out riding |
| 02:54.41 | a1fa | time to prank call them |
| 02:54.49 | [av]bani | _Sam--: not old enough! |
| 02:54.55 | a1fa | _Sam-- : too busy reving that bitch up to 14k |
| 02:55.03 | _Sam-- | all 250cc's of it? :P |
| 02:55.07 | a1fa | _Sam-- : yes! |
| 02:55.11 | _Sam-- | my snowblower makes almost as much HP :) |
| 02:55.15 | _Sam-- | ok...my lawn tractor does |
| 02:55.28 | a1fa | right... |
| 02:55.37 | _Sam-- | that thing maybe puts out 30-35hp |
| 02:55.40 | a1fa | wait till i get my 10R |
| 02:55.45 | a1fa | 35hp |
| 02:55.54 | a1fa | i want to get Aprilia RS250 |
| 02:56.01 | _Sam-- | "Kaw, ex250, stock, all stock, ~3k miles . ~ 25 - 28 True HP. Kaw, " |
| 02:56.02 | a1fa | tu-stroke 250 with 75hp |
| 02:56.16 | _Sam-- | i raced a honda rs250 for a while |
| 02:56.20 | _Sam-- | and a yam. tz250 |
| 02:56.25 | _Sam-- | i never liked the priller myself |
| 02:56.31 | _Sam-- | wasnt a "real" two stroke gp bike |
| 02:56.40 | a1fa | i just want to ride that biznatch on the street |
| 02:56.42 | a1fa | ;P |
| 02:56.44 | _Sam-- | i hear that |
| 02:56.44 | a1fa | no tags |
| 02:57.10 | _Sam-- | if you want to see a dyon for your bike... |
| 02:57.13 | _Sam-- | its only about 25-28hp |
| 02:57.13 | austinnichols10 | sam: what was the two stroke kenny roberts model called? |
| 02:57.22 | moprilo | i try to work with the jitterbuffer, but it feels like nothing.. is the jitterbuffer work'n in the asterisk 1.2? |
| 02:57.26 | _Sam-- | rz350 |
| 02:57.30 | _Sam-- | and rz500 |
| 02:57.36 | [av]bani | again #asterisk mutates to #bikes |
| 02:57.40 | a1fa | hey.. you got a 3.5mm headphones @ your store? |
| 02:57.50 | austinnichols10 | yeah - I remember that one. It was all ting-ting-ting at idle. very nice |
| 02:57.57 | a1fa | it is |
| 02:58.01 | a1fa | ting ting ting |
| 02:58.07 | a1fa | kenny roberts is trash :P |
| 02:58.13 | a1fa | his dad was ok |
| 02:58.19 | _Sam-- | austinnichols101: they make a street bike version of that that is the baddest ass thing. |
| 02:58.20 | a1fa | nicky hayden is a bit better |
| 02:58.28 | _Sam-- | its from the 80s |
| 02:58.33 | _Sam-- | street legal still |
| 02:58.41 | _Sam-- | rz350 AND an rz500 in canada |
| 02:58.43 | austinnichols10 | sam: that'll work - so am I.. |
| 02:58.53 | a1fa | RS250 is legal in EU |
| 02:59.02 | a1fa | that bastard redlines at 19k |
| 02:59.11 | _Sam-- | its legal in the US, or was a few years ago at least |
| 02:59.17 | a1fa | sounds like a pissed off lawn mower |
| 02:59.25 | a1fa | _Sam-- : so hard to find a street legit version |
| 02:59.32 | a1fa | i looked for months |
| 02:59.35 | a1fa | i found RS50 |
| 02:59.35 | austinnichols10 | next bike has to be a torque monster - tired of ringing ears |
| 02:59.37 | a1fa | almost bought it |
| 02:59.47 | SkramX | any suggestions for a *simple* billing app for Asterisk? |
| 03:00.04 | a1fa | SkramX : blink blink |
| 03:00.23 | SkramX | ?? |
| 03:00.39 | [hC] | SkramX: the closest thing i found that isnt even complete is asterisk-stat |
| 03:00.47 | [hC] | youll have to hack it quite a bit |
| 03:01.06 | SkramX | Oy. |
| 03:01.09 | _Sam-- | SkramX : how many bills do you need to generate? |
| 03:01.22 | SkramX | _Sam--: not too many.. shouldnt matter. |
| 03:01.36 | _Sam-- | you log to sql? |
| 03:01.47 | [av]bani | that gxp2k page is getting mighty large |
| 03:02.11 | SkramX | _Sam--: that will be preferrable. |
| 03:02.22 | _Sam-- | it was a question...do you log to sql |
| 03:02.24 | SkramX | a flat file would be /OK/... we can just insert it into mysql |
| 03:02.27 | SkramX | Oh. |
| 03:02.45 | SkramX | As of now, no. |
| 03:02.57 | *** join/#asterisk websae (n=icechat5@adsl-64-149-206-121.dsl.milwwi.sbcglobal.net) |
| 03:04.27 | xachen | I think SkramX wants just to have a simple prepaid system that can be liked into SIP/IAX accounts |
| 03:04.31 | a1fa | _Sam-- : i need a single 3.5mm headset for my helmet |
| 03:04.32 | xachen | linked* |
| 03:04.33 | _Sam-- | i have a cdr analyser but im not sure what its called ....its i know its one of the peices of AMP but i forget where i got it....it works well. |
| 03:04.47 | _Sam-- | you could generated detailed cdr reports from sql using it |
| 03:05.02 | _Sam-- | and somehow try to tie to billing or something |
| 03:05.13 | _Sam-- | sorry to be of so much info |
| 03:05.21 | _Sam-- | i just dont know what its called, and it doesnt say either. |
| 03:05.45 | _Sam-- | ah...this is it: |
| 03:05.49 | _Sam-- | http://areski.net/asterisk-stat-v2/about.php |
| 03:06.09 | _Sam-- | i like it |
| 03:06.27 | _Sam-- | a1fa: if you got a bluetooth phone i could help you out :) |
| 03:06.32 | _Sam-- | or a bluetooth pda even :) |
| 03:07.12 | a1fa | nah.. this is for radio |
| 03:07.21 | a1fa | my girl and i need it to communicate while riding |
| 03:07.30 | _Sam-- | so get a communicator :) |
| 03:07.35 | a1fa | we bought those radios at walmart but i cant find |
| 03:07.40 | a1fa | a single 3.5mm |
| 03:07.44 | a1fa | headphone |
| 03:07.47 | _Sam-- | dont be talking to me anymore about walmart! |
| 03:07.52 | a1fa | :P |
| 03:08.10 | _Sam-- | i like the place, its fine and dandy, got nothing against it... |
| 03:09.02 | a1fa | Radio Crap has them |
| 03:09.14 | *** join/#asterisk colinm_ (n=colin@VDSL-130-13-10-116.PHNX.QWEST.NET) |
| 03:09.15 | _Sam-- | you have the motorola talkabouts? |
| 03:10.15 | *** join/#asterisk Carp1 (i=Carp1@ip-204-97-151-100.modem.logical.net) |
| 03:10.29 | a1fa | something similar |
| 03:10.36 | _Sam-- | i have headsets for em |
| 03:10.41 | _Sam-- | use them for our race team |
| 03:10.51 | a1fa | can i race for you? |
| 03:10.54 | a1fa | i can knee drag |
| 03:10.57 | a1fa | i need a sponsor |
| 03:11.10 | _Sam-- | sure after you win some regional championships, send me your resume |
| 03:11.17 | _Sam-- | i'll see if we can fit some room for you in the semi |
| 03:11.31 | a1fa | you guys pay for bikes? |
| 03:11.34 | a1fa | what bike? |
| 03:11.52 | _Sam-- | we've gotten free bikes every year from yamaha since we started racing at a high level in 2002 |
| 03:13.31 | _Sam-- | this will be the first year since 2002 that we are not committed to racing a full year in the AMA. |
| 03:13.35 | *** join/#asterisk mwright1 (n=matt@203-217-29-237.perm.iinet.net.au) |
| 03:13.36 | a1fa | lucky dog |
| 03:13.37 | mwright1 | hey |
| 03:13.40 | a1fa | let me race for you |
| 03:13.46 | mwright1 | Just wandering if someone could give me some product guidance |
| 03:13.59 | _Sam-- | sorry, there is no class in the AMA for ex250s :P |
| 03:14.11 | mwright1 | WHat's the difference between asterisk and asterisk@home, does asterisk@Home have limitations (ie as it is bundled with everything and on centos) |
| 03:14.12 | a1fa | no |
| 03:14.18 | *** join/#asterisk pauldy (n=pauldy@24-155-86-154.ip.grandenetworks.net) |
| 03:14.18 | a1fa | there is for RS250 |
| 03:14.23 | _Sam-- | wrong |
| 03:14.32 | _Sam-- | all the two stroke classes were eliminated in 04 |
| 03:14.37 | _Sam-- | you CAN ride it |
| 03:14.40 | _Sam-- | it just wont be competitive |
| 03:14.45 | _Sam-- | you'd race in formula extreme |
| 03:14.50 | a1fa | damn |
| 03:14.52 | _Sam-- | and be the only one on a two stroke |
| 03:15.08 | a1fa | yeah.. i can still smoke a diesel when i see him :P |
| 03:15.35 | a1fa | not really |
| 03:15.38 | _Sam-- | im gonna go smoke some of my own diesel...and hang out with my wife. |
| 03:15.46 | *** join/#asterisk ahattar (n=kjsd@ool-43551487.dyn.optonline.net) |
| 03:15.47 | a1fa | nice dude |
| 03:15.52 | _Sam-- | have a good night |
| 03:15.56 | ahattar | hi all |
| 03:16.05 | a1fa | _Sam-- : i will talk to you later, remember me |
| 03:16.10 | _Sam-- | how could i forget! |
| 03:16.11 | _Sam-- | later. |
| 03:16.15 | a1fa | later |
| 03:16.48 | ahattar | question, how to add a sip user in Asterisk? |
| 03:16.49 | [TK]D-Fender | mwright1 : The real culpri with A@H = AMP. |
| 03:18.11 | [av]bani | :P |
| 03:18.18 | *** join/#asterisk bkw_ (n=bkw_@adsl-70-142-51-127.dsl.tul2ok.sbcglobal.net) |
| 03:19.08 | De_Mon | ahattar edit sip.conf |
| 03:19.36 | De_Mon | a1fa did you get your discount? |
| 03:22.21 | ahattar | thnx de_mon |
| 03:27.44 | a1fa | nah |
| 03:27.44 | a1fa | :P |
| 03:27.53 | a1fa | De_Mon : he has a race team! |
| 03:28.01 | a1fa | thats bad ass.. i am going to send him my resume |
| 03:28.50 | xachen | Where can I get real cheap 800 origination that allows Canada? |
| 03:29.34 | [av]bani | people live in canada? |
| 03:29.59 | xachen | you'd never guess ;) |
| 03:31.15 | xachen | yes? |
| 03:31.41 | essaredee | that's all you've got to say to me?! |
| 03:31.45 | xachen | oh hey there Sean :) |
| 03:31.53 | essaredee | lol |
| 03:32.08 | essaredee | hey :) |
| 03:32.15 | [TK]D-Fender | xachen : major metro primarily, or all throughout? |
| 03:32.29 | xachen | pretty much everywhere |
| 03:32.36 | xachen | I don't care about the territories |
| 03:33.17 | xachen | Link2voip offers it for 4.5c/min as cheap as I can see |
| 03:35.07 | [av]bani | [TK]D-Fender: http://lists.digium.com/pipermail/asterisk-users/2006-February/146983.html <- more polycom excellence |
| 03:35.31 | essaredee | anyone know if the 7970's work with asterisk yet without problems? |
| 03:35.39 | Qwell | essaredee: with chan_sccp, they work fine |
| 03:36.07 | essaredee | I heard sometime back that the working between the two weren't that great |
| 03:36.15 | essaredee | albeit it was bit over a year ago |
| 03:36.20 | xachen | I crashed * with my SIP softphone the other day dialing into a conference... but nobody prob. needed to know that |
| 03:37.10 | essaredee | I'm happy I finally got * to compile onto tornado |
| 03:37.17 | [av]bani | 7970, thats an expensive phone |
| 03:37.24 | essaredee | aye |
| 03:37.34 | essaredee | but it has a pretty colour display :) |
| 03:37.34 | Qwell | ~$425 now |
| 03:37.39 | Qwell | definitely worth it |
| 03:37.56 | essaredee | I'll wait until I can get it on ebay for less than £200 |
| 03:38.05 | xachen | tornado huh? |
| 03:38.06 | [TK]D-Fender | [av]bani : Yeah, I've only been talking abot that for a few months now :) |
| 03:38.13 | xachen | what OS is that? BSD? |
| 03:38.20 | essaredee | yeah, it wouldn't compile on it |
| 03:38.23 | essaredee | yeah, 5.2.1 |
| 03:38.40 | [TK]D-Fender | and supporting BLA (SLA?) is a GOOD thing. It'll add to the SPA-9xx's value as well... |
| 03:39.16 | xachen | i had no probs on 5.3 |
| 03:39.18 | [TK]D-Fender | I'm patient. So far I love the REST of what it offers, and at least have a better DATE to work with for whats to come. |
| 03:39.21 | essaredee | xachen: you using * for your business? |
| 03:39.48 | xachen | its more a pet project but more or less yes |
| 03:40.05 | essaredee | cool |
| 03:40.08 | xachen | i crashed my bsd box unloading the kern module for the bsd port for ztdummy |
| 03:40.18 | essaredee | hehe |
| 03:40.23 | [av]bani | Qwell: $425? where? |
| 03:40.30 | Qwell | [av]bani: ebay |
| 03:40.41 | Qwell | and if you were on asterisk-biz, you'd know where you could get some for $385 |
| 03:40.53 | [av]bani | :P |
| 03:41.50 | *** join/#asterisk ahattar (n=kjsd@ool-43551487.dyn.optonline.net) |
| 03:41.55 | essaredee | hrm, that's about £245, still out my desired price range |
| 03:42.20 | essaredee | not bad either, 10 available from this one person in texas |
| 03:42.35 | lucasjb | Hiyas, question: I have a SIP peer that I want to use for inbound and outbound calls. I have an inbound context [voice.myprovider.com] and an outbound [myprovider-outbound]. Each of them work in isolation, but when I enable both, my inbound sip calls always find the peer myprovider-outbound rather than voice.myprovider.com. How can I use the same host= in both contexts but make sure inbound calls are using the right context? |
| 03:43.20 | essaredee | fromhost ? |
| 03:43.37 | lucasjb | essaredee, ooer, lemme try... |
| 03:43.47 | essaredee | sorry, that might not be right, I'm just making a guess and seeing if anyone corrects me :) |
| 03:44.12 | lucasjb | essaredee, Oh well I have fromdomain, but that doesn't seem to work without host |
| 03:44.29 | essaredee | fromdomain=xxx.xxx |
| 03:44.31 | essaredee | i mean |
| 03:44.43 | essaredee | in your sip.conf file |
| 03:44.48 | essaredee | for [peer] |
| 03:46.01 | essaredee | aha ok |
| 03:46.16 | [av]bani | 7970 is sccp only... |
| 03:46.30 | essaredee | if i read this right, you want all outbound calls to go through [context1] and inbound through [context2]? |
| 03:46.48 | essaredee | for that one peer |
| 03:47.42 | lucasjb | essaredee, yes. |
| 03:48.28 | essaredee | right.. under [peer] in sip.conf use context=inbound-context-in-extensions.conf |
| 03:48.53 | essaredee | that will handle the incoming bit |
| 03:49.06 | lucasjb | essaredee, Yep... I have context=default and it does work... |
| 03:49.21 | lucasjb | essaredee, until I uncomment the [outbound] context... |
| 03:49.51 | essaredee | then |
| 03:50.11 | essaredee | change the context names in extensions.conf so |
| 03:50.26 | essaredee | [general] is [outbound] and [outbound] is [general] |
| 03:50.29 | *** join/#asterisk mzo (i=user@ool-435193b3.dyn.optonline.net) |
| 03:50.34 | essaredee | in other words flip them around |
| 03:51.37 | ahattar | Feb 14 22:51:04 NOTICE[5195]: chan_sip.c:10933 handle_request_register: Registration from 'abe <sip:abe@10.0.0.11>' failed for '10.0.0.101' - Username/auth name mismatch |
| 03:51.45 | ahattar | any idea why? |
| 03:52.06 | [TK]D-Fender | exactly what it says.... |
| 03:52.10 | essaredee | im assuming abe is a phone on your network? |
| 03:52.15 | [TK]D-Fender | Username/auth name mismatch |
| 03:52.15 | lucasjb | essaredee, Hmm... this confuses me. |
| 03:52.26 | *** part/#asterisk mwright1 (n=matt@203-217-29-237.perm.iinet.net.au) |
| 03:53.26 | essaredee | under [general] in sip.conf what does context=? |
| 03:53.33 | a1fa | ss |
| 03:53.33 | a1fa | s |
| 03:53.39 | lucasjb | essaredee, default |
| 03:53.58 | essaredee | do you have a context= in [yourphone]? |
| 03:54.12 | ahattar | context=default |
| 03:54.27 | essaredee | what contexts do you have in extensions.conf? |
| 03:54.45 | ahattar | context= in abe |
| 03:54.48 | lucasjb | essaredee, I have context=default, host=voice.myprovider.com in [voice.myprovider.com] my incoming SIP calls context |
| 03:55.33 | essaredee | no... when you look in extensions.conf what all [sections] have you got? |
| 03:56.30 | lucasjb | essaredee, I have several that I've created... umm, [general], [global], [default], [local], [std], [international]... |
| 03:56.44 | essaredee | okeydoke |
| 03:57.11 | essaredee | which one are you wanting to use for outbound and which one for incoming? |
| 03:57.56 | [av]bani | hmm those 7970g's are tempting, but sccp is a turnoff |
| 03:58.06 | lucasjb | essaredee, well outbound calls start in [default] and then go to [local] if ${EXTEN} looks like a local number... |
| 03:58.07 | essaredee | how stable is sccp? |
| 03:58.07 | Qwell | [av]bani: sccp is far better than sip, imo |
| 03:58.12 | Qwell | essaredee: it works very well |
| 03:58.20 | essaredee | ok |
| 03:58.49 | De_Mon | ahattar make sure [peer] is the same as username= |
| 03:59.01 | De_Mon | !sccp |
| 03:59.02 | De_Mon | ~sccp |
| 03:59.04 | jbot | it has been said that sccp is Proprietary protocol used between Cisco Call Manager and Cisco VOIP phones. Also supported by some other vendors. Also Signaling Connection Control Part (SCCP), a routing protocol in SS7 protocol suite in layer 4, provides end-to-end routing for TCAP messages to their proper database. |
| 03:59.05 | essaredee | lucasjb: and for inbound? |
| 03:59.28 | De_Mon | Cisco Call Control Protocol? |
| 03:59.39 | lucasjb | essaredee, well inbound calls start in default also, then are sent to some SBR rules based on ${EXTEN} again |
| 03:59.42 | Qwell | skinny client control protocol |
| 03:59.52 | De_Mon | Qwell propriatary > opensource? |
| 04:00.05 | Qwell | De_Mon: I use an open source sccp implementation |
| 04:00.22 | essaredee | hm ok |
| 04:00.39 | essaredee | I think I'm lost now lol sorry, don't know exaclty what you're after |
| 04:00.45 | lucasjb | essaredee, I think I may have used the wrong terminology when I first asked my question. I think the issue I'm having is with /channels/ in sip.conf |
| 04:00.49 | De_Mon | Qwell ahso the both of best worlds |
| 04:01.00 | De_Mon | s/both of best/best of both/ |
| 04:01.03 | Qwell | De_Mon: indeed, I love chan_sccp |
| 04:01.05 | De_Mon | haha |
| 04:01.45 | lucasjb | essaredee, Thanks for your help, let me try to rephrase one more time... |
| 04:01.57 | De_Mon | Qwell do you love app_queue too? |
| 04:02.07 | Qwell | no, app_queue gives me nightmares |
| 04:02.15 | Qwell | he and app_voicemail gang up on me in a dark alley |
| 04:02.15 | essaredee | ok |
| 04:02.50 | De_Mon | what's an alternative solution to app_queue? |
| 04:02.53 | essaredee | wish I could get 1471 workin |
| 04:03.26 | lucasjb | essaredee, I have sip.conf with many 'register => 12345678@voice.myprovider.com...' lines to register with my provider. Then I have one channel definition in sip.conf called [voice.myprovider.com] which sets dtmfmode=inband and context=default and a couple of other things. This works for dialling in to my system through the regular phone network... |
| 04:03.44 | *** join/#asterisk MstlyHrmls (n=mh@melbourne.mostly-harmless.ca) |
| 04:04.04 | lucasjb | essaredee, [voice.myprovider.com] contains host=voice.myprovider.com as well of course... |
| 04:04.29 | brookshire[home] | qwell: so what do you like? |
| 04:04.39 | Qwell | brookshire[home]: chan_sccp :) |
| 04:04.45 | Qwell | and...cake |
| 04:04.47 | Qwell | I like cake |
| 04:04.49 | lucasjb | essaredee, Now, I want to make outbound calls using the same provider, so I create a sip.conf channel called [myprovider-outbound] and I use auth details from one of my 'register => ' lines... |
| 04:04.53 | brookshire[home] | did you guys actually get it working tonight? |
| 04:05.54 | lucasjb | essaredee, This also works, /except/ now inbound calls are broken because when the call first comes in, it tries to use the channel [myprovider-outbound] instead of [voice.myprovider.com] as it was doing previously |
| 04:06.09 | lucasjb | essaredee, Does that make sense? |
| 04:06.30 | essaredee | i think so |
| 04:08.00 | essaredee | does [myprovider-outbound] use the same username/password? |
| 04:08.10 | *** join/#asterisk mogorman (n=mogorman@user-24-236-84-48.knology.net) |
| 04:10.26 | brookshire[home] | wow.. so like.. every file changed |
| 04:10.27 | brookshire[home] | since yesterday |
| 04:10.27 | brookshire[home] | lol |
| 04:10.27 | brookshire[home] | in head |
| 04:10.35 | *** join/#asterisk Mavvie (n=edwin@252-131-222-203.rev.techex.net.au) |
| 04:10.46 | lucasjb | essaredee, it uses one of the username/password combinations from my 20 'registration => ' lines |
| 04:10.46 | Mavvie | "zap show channels" should have a span option. |
| 04:11.02 | essaredee | aha |
| 04:11.35 | essaredee | then that should only affect one of the incoming lines |
| 04:11.42 | essaredee | in any case |
| 04:12.06 | essaredee | set the context= for [myprovider-outbound] to general |
| 04:12.14 | essaredee | as that only affects incoming calls |
| 04:12.23 | lucasjb | essaredee, But what happens is that all incoming calls find the [myprovider-outbound] channel first, rather than [voice.myprovider.com] and I don't get it |
| 04:12.37 | lucasjb | essaredee, Lemme try that... |
| 04:12.39 | De_Mon | essaredee what type of peer did you make your sip.conf provider? |
| 04:12.48 | De_Mon | type= ? |
| 04:12.51 | mogorman | hey brookshire sorry i was pissy today |
| 04:14.10 | lucasjb | De_Mon, are you talking to me? |
| 04:14.20 | De_Mon | lucasjb maybe :) |
| 04:14.50 | lucasjb | De_Mon, heh, inbound and outbound channels are both type=peer |
| 04:15.37 | De_Mon | I'd change 1 to type=friend and nuke the 2nd. Then set context=[inbound context] |
| 04:16.29 | lucasjb | De_Mon, OK, so for the inbound SIP channel you think I should use type=friend and context=myprovider-inbound... |
| 04:16.44 | De_Mon | yah |
| 04:17.18 | lucasjb | OK, Let me spend some time fiddling with my extensions.conf |
| 04:17.23 | De_Mon | inbound should work then. |
| 04:17.32 | lucasjb | essaredee, De_Mon Thanks for listening #_^ |
| 04:17.43 | essaredee | np :) |
| 04:18.59 | De_Mon | as of 1.2 peer and friend are essentially the same thing, yes? |
| 04:19.35 | De_Mon | no, nm I misread |
| 04:20.23 | essaredee | someone said something about sccp and closed source.. my response: if someone made a voip phone with 7970 quality then I'd go for that instead :P |
| 04:20.32 | [av]bani | hmm. so what is the thing to use for 7970g... chan_skinny, chan_sccp or chan_sccp2 ? |
| 04:20.36 | Qwell | essaredee: won't happen for a while |
| 04:20.42 | Qwell | [av]bani: chan-sccp.berlios.de |
| 04:20.47 | De_Mon | essaredee open source? |
| 04:20.55 | [av]bani | Qwell: you use that with your 7970g? |
| 04:20.59 | brookshire[home] | essaredee: what about polycom phones? |
| 04:21.02 | Qwell | my boss does, but yes |
| 04:21.12 | essaredee | polycom makes a voip phone with a colour display? |
| 04:21.15 | Qwell | brookshire[home]: polycoms may be good, but you've gotta admit, they're no 7970 |
| 04:21.28 | Qwell | color+touchscreen = <3<3<3<3 |
| 04:21.34 | De_Mon | lol |
| 04:21.34 | essaredee | exactly :) |
| 04:21.45 | De_Mon | I could care less about color on my phone |
| 04:21.45 | Qwell | they're very hot |
| 04:22.09 | essaredee | I have colour everything, even my analogue cordless phone has a colour display |
| 04:22.26 | De_Mon | what does it colorize? |
| 04:22.31 | [av]bani | De_Mon: pr0n |
| 04:22.37 | Qwell | 7970 can do pr0n |
| 04:22.42 | De_Mon | 7970 pr0n |
| 04:22.42 | essaredee | lol |
| 04:23.00 | De_Mon | at what resolution? |
| 04:23.03 | Qwell | meta refresh goodness |
| 04:23.08 | [av]bani | ~phones |
| 04:23.10 | jbot | methinks phones is at http://bani.anime.net/phones/ |
| 04:23.18 | De_Mon | its not the phone thats hawt is the nakied girlies on in you like! |
| 04:23.18 | brookshire[home] | dial-a-nude ;) |
| 04:23.27 | Qwell | brookshire[home]: omg, you're a genius |
| 04:23.35 | mzo | i haw, naked phonz |
| 04:23.50 | mzo | 7970 porn? whoa |
| 04:24.18 | brookshire[home] | qwell: next we need to make an app for asterisk that fetches us a drink and gets us beer from the fridge |
| 04:24.25 | mzo | that's coming |
| 04:24.39 | mzo | expect it in 2.0 |
| 04:24.43 | De_Mon | I have that, I dial 1 for it |
| 04:24.45 | brookshire[home] | fetch us a sandwitch.. i mean |
| 04:24.53 | [TK]D-Fender | brookshire : Chan_x10 :D |
| 04:24.54 | essaredee | phone base in the shape of a womans bust and the handset like a mans p*nis with vibration functionality |
| 04:25.03 | brookshire[home] | mzo: but 2.0 runs on the .net platform |
| 04:25.16 | [TK]D-Fender | I can make coffe already! Fething a beer can't be far off now! |
| 04:25.18 | mzo | my 2.0 does it, but it requires .marriage compatibility libs. |
| 04:25.35 | [TK]D-Fender | And I can't type for beans tonight! |
| 04:25.38 | mzo | and it's not multiprocessor aware, it just stops running if another processor is detected. |
| 04:25.40 | brookshire[home] | app_beer |
| 04:26.01 | brookshire[home] | mzo: that's hot |
| 04:26.09 | mzo | it IS |
| 04:26.16 | mzo | worth the fact that it runs in ring 0 =( |
| 04:26.22 | De_Mon | I hope you like flat beer because it'll be in beta for a while |
| 04:26.48 | *** part/#asterisk ahattar (n=kjsd@ool-43551487.dyn.optonline.net) |
| 04:27.08 | brookshire[home] | haha |
| 04:27.15 | brookshire[home] | asterisk has a beta? |
| 04:27.26 | mzo | asterisk is like google software |
| 04:27.28 | mzo | it's always in beta |
| 04:27.29 | mzo | ;) |
| 04:27.33 | brookshire[home] | app_beotch |
| 04:27.35 | De_Mon | word |
| 04:27.50 | [av]bani | just wait till Microsoft PBX gets released |
| 04:27.52 | De_Mon | when it's outa beta they will start charging for it |
| 04:28.16 | brookshire[home] | doesn't microsoft already have one? |
| 04:28.22 | De_Mon | ? |
| 04:28.22 | [av]bani | i havent seen one |
| 04:28.37 | De_Mon | microsoft@home? |
| 04:28.53 | brookshire[home] | http://www.microsoft.com/office/livecomm/prodinfo/default.mspx |
| 04:28.57 | brookshire[home] | but that's mainly for im |
| 04:35.02 | *** join/#asterisk tuppa (n=tuppa@eclipse.tuppa.org) |
| 04:36.00 | [av]bani | looks like its only im, no voip at all |
| 04:38.22 | *** join/#asterisk glm2k (n=GLM@rrcs-24-199-11-46.west.biz.rr.com) |
| 04:41.15 | mogorman | just wait till we get im.... |
| 04:41.19 | mogorman | ^_^ |
| 04:44.00 | brookshire[home] | actually.. it does more than just im |
| 04:44.02 | brookshire[home] | The architecture of Live Communications Server uses the industry-standard protocols Session Initiation Protocol (SIP) and SIP for Instant Messaging and Presence Leveraging Extensions (SIMPLE), and includes a rich set of application programming interfaces (APIs). Microsoft supports these standards because of their broad potential in future communications, a potential that goes far beyond instant messaging. |
| 04:44.26 | mogorman | all you just said was im... |
| 04:44.34 | mogorman | simple or msn messaging is just im |
| 04:44.58 | mogorman | now writing in a jabber client into openoffice might not be a a bad idea brookshire but its not our problem |
| 04:45.00 | brookshire[home] | so it just detects presence |
| 04:45.21 | [av]bani | no RTP |
| 04:45.22 | mogorman | well they have hooks in office |
| 04:45.30 | mogorman | for messaging documents etc |
| 04:45.41 | [av]bani | doesnt sound like a pbx to me |
| 04:45.51 | mogorman | well they do some voip switching |
| 04:45.52 | brookshire[home] | it's the start of one |
| 04:45.55 | mogorman | but not any real stuff |
| 04:46.13 | brookshire[home] | but i will agree, they don't focus on telephony yet |
| 04:46.18 | brookshire[home] | :) |
| 04:46.21 | brookshire[home] | and probably never will |
| 04:46.28 | [av]bani | microsoft certified solitaire expert |
| 04:46.32 | Qwell | If they do, it's the death of us all! |
| 04:46.34 | mogorman | after seeing them at von |
| 04:46.43 | Qwell | mogorman: They'll be at VON this year too |
| 04:46.48 | mogorman | yeah |
| 04:46.52 | mogorman | they are doing more and more with sip |
| 04:46.58 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
| 04:47.12 | brookshire[home] | i guess sip 4 life |
| 04:47.14 | brookshire[home] | :( |
| 04:47.51 | [av]bani | just watch tehm integrate it into xbox 360 or something |
| 04:47.58 | mogorman | lol |
| 04:47.59 | [av]bani | xbox pbx |
| 04:48.03 | [av]bani | pbxbox! |
| 04:48.06 | [av]bani | (tm) |
| 04:48.07 | mogorman | they do speex for the audion on the xbox |
| 04:48.10 | mogorman | i just find that so funny |
| 04:48.34 | [av]bani | seems they'd rather use wmv or something |
| 04:48.36 | mogorman | that they wont put up money for g729 and support the evil open source |
| 04:48.46 | [av]bani | wouldnt that be embarassing to use speex? |
| 04:48.56 | mogorman | its what it uses |
| 04:49.11 | [av]bani | because open source is all commies and stuff |
| 04:49.14 | jontow | man, i think i hacked the crap out of our CRM..heheh |
| 04:49.15 | jontow | myb ad |
| 04:49.20 | mogorman | yeah its hard to believe |
| 04:49.45 | [av]bani | well youd think microsoft spent billions on developing wmv codecs |
| 04:49.49 | [av]bani | and they go and use speex... |
| 04:49.53 | [av]bani | ? |
| 04:49.59 | mogorman | yeah |
| 04:50.07 | mogorman | speex is a great audio codec |
| 04:50.12 | mogorman | if you have the cpu |
| 04:50.17 | brookshire[home] | [av]bani: the pbxbox is sooo 2000 |
| 04:50.22 | mogorman | and seeing as how they only do 1 channel |
| 04:50.25 | [av]bani | funny, people always seem to whinge and bash speex here |
| 04:50.27 | mogorman | its nothing |
| 04:50.31 | [av]bani | OMG SPEEX SUX |
| 04:50.43 | mogorman | lol |
| 04:50.49 | mogorman | no its just hard to work with |
| 04:50.53 | lucasjb | essaredee, De_Mon in case you're interested, reversing the order of the channel definitions in sip.conf has solved the problem. It appears that the last channel that is configured is the first to be looked at when a call is inbound. I tried playing with seperate contexts which didn't help. |
| 04:51.00 | mogorman | and its easier to just buy g729 most of the time |
| 04:51.04 | mogorman | or just use gsm |
| 04:52.09 | brookshire[home] | jontow: sugar? |
| 04:53.31 | De_Mon | lucasjb I thought you were removing all but 1 |
| 04:55.24 | jontow | nah |
| 04:55.30 | jontow | internal custom deal |
| 04:55.33 | brookshire[home] | oh |
| 04:55.37 | jontow | quite nice, but lacking on the linux side imo |
| 04:55.49 | jontow | so i've been doing a bit of 11pm-midnight work tonight on making it more usable for me:) |
| 04:55.52 | brookshire[home] | sugar is pretty rad |
| 04:56.09 | brookshire[home] | i hacked the hell out of it for work.. |
| 04:56.10 | brookshire[home] | lol |
| 04:56.30 | Abydos313 | post patches for us :)) |
| 04:56.41 | jontow | :D |
| 04:57.03 | jontow | i like this one, but it needs some work imo.. like a rewrite using sane languages front and back |
| 04:57.09 | jontow | (coldfusion in the back, flash in the front..) |
| 04:57.44 | jontow | concepts are good though.. |
| 04:57.57 | jontow | hell, its even a solid implementation, considering the tools it is working with |
| 04:58.35 | brookshire[home] | patches for sugar? |
| 04:58.54 | Abydos313 | i was joking |
| 04:59.04 | brookshire[home] | hehe.. well i made modules |
| 04:59.06 | brookshire[home] | anyways |
| 04:59.11 | brookshire[home] | so no patches :) |
| 04:59.52 | Abydos313 | i was just speaking for those of us that are programming challenged..hahah |
| 05:00.04 | brookshire[home] | hah.. ok |
| 05:00.19 | brookshire[home] | well... i made a module that lets you do quotes and orders in sugar |
| 05:00.25 | brookshire[home] | so you don't have to buy the enterprise version |
| 05:00.28 | brookshire[home] | lol |
| 05:00.33 | Abydos313 | nice |
| 05:00.44 | Abydos313 | i actually haven't even checked that out yet |
| 05:00.45 | brookshire[home] | but it's kinda ghetto and i didn't use their api |
| 05:01.05 | brookshire[home] | i mean.. it looks rad.. but i didn't use their hooks for data handling |
| 05:01.21 | Abydos313 | every good project starts somewhere |
| 05:02.07 | brookshire[home] | well.. i might post them somewhere someday |
| 05:02.15 | brookshire[home] | we're also writting an rma system for it |
| 05:02.52 | Abydos313 | are you on a team doing it ? |
| 05:03.28 | brookshire[home] | yeah.. |
| 05:03.41 | Abydos313 | sweet |
| 05:04.01 | Abydos313 | i barely do some bullshit c++ heh |
| 05:04.07 | brookshire[home] | well this is php |
| 05:04.13 | brookshire[home] | php is easy |
| 05:04.34 | Abydos313 | how much functionality is the goal? |
| 05:04.48 | brookshire[home] | we're putting everything into it |
| 05:04.50 | brookshire[home] | all of our systems |
| 05:05.01 | brookshire[home] | only problems with sugar |
| 05:05.14 | brookshire[home] | is it doesn't really support group level permissions |
| 05:05.31 | brookshire[home] | which would be helpful |
| 05:05.42 | brookshire[home] | because you don't want rma looking at projects for sales |
| 05:06.14 | Abydos313 | you'll figure something out |
| 05:06.44 | brookshire[home] | well.. i try to stay within sugar's framework |
| 05:06.53 | brookshire[home] | because i don't want to have to update it everytime they update sugar |
| 05:07.00 | brookshire[home] | it's quite a task |
| 05:07.08 | Abydos313 | good idea |
| 05:07.30 | brookshire[home] | but they switched over to subversion recently.. so it might make things easier :) |
| 05:07.53 | brookshire[home] | so i might just pull their tree.. then make my mods |
| 05:07.56 | brookshire[home] | in a branch |
| 05:08.08 | Abydos313 | way over my head |
| 05:08.44 | brookshire[home] | subversion lets you keep track of your changes versus theirs.. |
| 05:08.52 | brookshire[home] | rather than having to download a tar everytime |
| 05:09.01 | brookshire[home] | i just do an update and get the latest release |
| 05:09.03 | Abydos313 | i know what it is, just the thought of it is way over my head |
| 05:09.08 | brookshire[home] | then i can merge my code into their's |
| 05:09.31 | brookshire[home] | nah.. subversion is really easy |
| 05:09.48 | brookshire[home] | it just takes a little bit to force yourself to do things the svn way |
| 05:13.55 | tronix | only thing I wish svn had was relative-to-a-project version number like cvs does, but makes sense why they have to do it on a global basis |
| 05:14.11 | tronix | I can live with that minor thing, when it's got tons of other benefits |
| 05:15.07 | tronix | I just wonder how others like perforce stacks up to svn |
| 05:17.16 | brookshire[home] | tronix: that's the beauty of viewsvn! |
| 05:17.17 | *** join/#asterisk bozo_the_clown (n=icechat5@bishop.msi.net.au) |
| 05:17.35 | brookshire[home] | actually.. there is a command to show all the changes and the version number |
| 05:17.49 | brookshire[home] | on a per file base |
| 05:19.50 | tronix | hmm nice. |
| 05:22.22 | blitzrage | brookshire[home]: !!! |
| 05:22.34 | brookshire[home] | blitzzzzzzzzzzzz! |
| 05:22.46 | brookshire[home] | how's my 3rd favorite canadian? |
| 05:24.04 | konfuzed | slePP: are you here? |
| 05:26.05 | *** join/#asterisk niZon (n=ilt@S010600080db4ab60.wp.shawcable.net) |
| 05:29.07 | bozo_the_clown | Hey guys, looking for a tad of advice. We are bridging e1 to a pbx and we are trying to work out how to record a call without actually answering it first. I heard a suggestion that it is possible in some of the newer versions but not sure how. |
| 05:29.31 | *** join/#asterisk `Sauron (i=sauron@h-69-3-12-50.hstqtx02.covad.net) |
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| 05:43.38 | brookshire[home] | bozo: chanspy :) |
| 05:50.35 | bozo_the_clown | brookshire[home]: Thanks, that may well be what I'm looking for :) |
| 05:51.22 | litage | what needs to be configured in Asterisk if SER is going to forward/redirect calls to Asterisk? |
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| 06:36.36 | jsaunders | Is there some sort of auto volume leveling asterisk does by default with moh (or calls for that matter)? |
| 06:37.36 | jsaunders | When I listen to mp3's by dialing an extension for moh, it seems to volume down at quiet parts of the music. |
| 06:40.07 | *** join/#asterisk IOscanner (n=IOscanne@c-24-0-183-125.hsd1.tx.comcast.net) |
| 06:43.59 | IOscanner | has cdr_mysql changed in asterisk 1.2? |
| 06:44.23 | IOscanner | I do cdr status and I only see csv I don't see mysql anymore |
| 06:44.36 | IOscanner | is it not part of asterisk-addons-1.2 |
| 06:46.08 | *** join/#asterisk xtr (n=01928375@S0106000c41ed11e1.vf.shawcable.net) |
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| 06:50.34 | tronix | it is. |
| 06:50.45 | tronix | *CLI> show modules like sql |
| 06:50.49 | IOscanner | where I don't see it in cdr/ |
| 06:50.56 | tronix | one sec |
| 06:52.07 | IOscanner | it has cdr_csv cdr_manager cdr_pgsql cdr_tds cdr_custom cdr_odbc cdr sqlite |
| 06:52.09 | IOscanner | no mysql |
| 06:52.11 | tronix | it's in the main directory |
| 06:52.18 | tronix | of asterisk-addons tarball when extracted |
| 06:52.45 | tronix | for example: |
| 06:52.49 | tronix | wget http://ftp.digium.com/pub/asterisk/asterisk-addons-1.2.1.tar.gz |
| 06:52.59 | *** join/#asterisk oej (n=oej@72.80-202-179.nextgentel.com) |
| 06:52.59 | tronix | tar zxf asterisk-addons-1.2.1.tar.gz && cd asterisk-addons-1.2.1 |
| 06:53.01 | tronix | it's right there. |
| 06:53.04 | tronix | (in the main dir) |
| 06:53.18 | IOscanner | I don't see it |
| 06:53.24 | IOscanner | I have 1.2 |
| 06:53.54 | tronix | it's there in 1.2.0 too |
| 06:54.01 | tronix | sounds like you've got a defective tarball. |
| 06:54.09 | tronix | or |
| 06:54.12 | tronix | if this is package-based |
| 06:54.17 | tronix | your distro may have repackaged it differently. |
| 06:54.35 | Assid | can this snd_timer cause an issue with the RTC? |
| 06:54.41 | IOscanner | I just pulled a new version down from svn |
| 06:54.54 | IOscanner | brounches/1.2 asterisk-addons-1.2 |
| 06:55.04 | IOscanner | just a sec |
| 06:55.06 | IOscanner | damn |
| 06:55.19 | *** join/#asterisk Psykick (n=anon@phoenixone.co.nz) |
| 06:55.22 | Psykick | hi guys |
| 06:55.29 | Abydos313 | hi |
| 06:55.34 | tronix | morning. |
| 06:55.37 | tronix | Assid: don't know. |
| 06:55.40 | Psykick | people in my call queues are getting disconnected after a few mins of being connected |
| 06:55.46 | Psykick | can anyone suggest some places to google |
| 06:55.55 | Psykick | or some things to google for? |
| 06:56.15 | holmeh | Don't you get any notifications from asterisk? |
| 06:56.16 | tronix | IOscanner: all I can tell you is that the release tarballs from Digium has it. :) |
| 06:56.16 | IOscanner | if you have asterisk and not asterisk-addons in the path before branches it will pull the asterisk-1.2 code instead |
| 06:56.17 | holmeh | In the CLI. |
| 06:56.22 | tronix | IOscanner: ahhh. |
| 06:56.23 | IOscanner | doh |
| 06:56.35 | Psykick | holmeh: such as ? |
| 06:56.53 | IOscanner | I thought svn would error if I put a bad path I guess it just passes the data from that dir |
| 06:56.53 | holmeh | I dunno, I am just asking if you have seen there at all. =) |
| 06:56.56 | lucasjb | This is a really basic question but I can't figure it out. I've got zapata.conf: musiconhold=default and musiconhold.conf: mode => quietmp3, directory = /var/lib/asterisk/mohmp3, application => /usr/local/bin/mpg123 -q -r 8000 -f 8192 -b 2048 --mono -s --- Asterisk says: -- Started music on hold, class 'default', on channel 'SIP/2130-159f' then immediately -- 'Stopped music on hold on SIP/2130-159f' --- no music, anyone? |
| 06:57.05 | Assid | okay i rmmod that.. but i still get rtc: lost some interrupts at 1024Hz. when i load up ztdummy |
| 06:57.23 | tronix | lucasjb: just for future reference -- pastebin is great. ;) |
| 06:57.25 | tronix | ~pb |
| 06:57.26 | jbot | i guess pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
| 06:57.41 | IOscanner | got it sorry. |
| 06:57.44 | IOscanner | thanks |
| 06:57.44 | tronix | lucasjb: with that said, not sure. |
| 06:57.47 | tronix | np |
| 06:57.49 | Assid | tronix: any clue on that? |
| 06:58.02 | lucasjb | Thanks tronix |
| 06:58.15 | RaYmAn-Bx | lucasjb: as always, check that you have the correct version of mpg123 |
| 06:58.50 | tronix | Assid: I'm not too familiar with snd_timer specifically. Are you seeing error messages? |
| 06:59.11 | tronix | Assid: nevermind -- just read scrollback, sorry. |
| 07:00.15 | tronix | Assid: what is output of: $ cat /proc/sys/dev/rtc/max-user-freq |
| 07:00.26 | lucasjb | RaYmAn-Bx, right-oh, pretty sure I do, but will check. |
| 07:00.45 | RaYmAn-Bx | IOscanner: the last argument in a checkout command in svn is just the name of the local directory to use..it's only the first path that matters wrt what you check out |
| 07:00.47 | lucasjb | Version 0.59r (1999/Jun/15). Written and copyrights by Michael Hipp. |
| 07:01.09 | Psykick | the other weird thing that is happening with my queues is that if someone sits in there long enough that get to queue position 1 then get moved to queue position 4 if there are other people waiting in the queue |
| 07:01.10 | Assid | 64 |
| 07:01.19 | tronix | Assid: ooh, thought so. let's see |
| 07:01.33 | RaYmAn-Bx | lucasjb: it says mpg123 somewhere in the info as well, right? :) |
| 07:01.51 | holmeh | Psykick: Sounds like my ISP. |
| 07:01.58 | tronix | Assid: try: # echo 1024 > /proc/sys/dev/rtc/max-user-freq |
| 07:02.30 | lucasjb | RaYmAn-Bx, yeah somewhere in there... |
| 07:02.31 | bsdfreak | heh |
| 07:02.41 | tronix | Assid: the app is trying to ask for timer updates often; kernel is delivering it 64 times each second instead of 1024 times each second |
| 07:02.54 | Psykick | holmeh: any ideas on what could be causing the issue? |
| 07:02.58 | tronix | Assid: so the application is losing timer interrupts that it needs to work correctly. |
| 07:03.01 | Assid | no.. still same issue |
| 07:03.05 | tronix | Hmm. |
| 07:03.09 | Psykick | I'm currently using asterisk 1.2.0 |
| 07:03.25 | Assid | its running SMP.. (HT) |
| 07:03.46 | Psykick | Assid: I've heard of quite a few issues with SMP and HT |
| 07:03.56 | Psykick | http://www.asteriskguru.com/tutorials/pci_irq_apic_tdm_ticks_te410p_te405p_noise.html |
| 07:03.57 | holmeh | Psykick: I am sorry, no clue, my best shot would be to run asterisk with 'asterisk -vvvvc' and try and look at the debug output. |
| 07:04.29 | Psykick | ahh well |
| 07:04.36 | Psykick | I'll investigate further |
| 07:04.40 | Assid | so is there a way to fix this? |
| 07:04.47 | Psykick | thanks for the chat anyways guys |
| 07:04.48 | Assid | or do i have to lose SMP.. |
| 07:04.56 | Psykick | losing SMP ain't a bad thing |
| 07:05.12 | Psykick | I started to wonder the same thing |
| 07:05.17 | Psykick | but it works just as well |
| 07:05.28 | Psykick | and I'm operating a call center with asterisk |
| 07:06.01 | holmeh | Large? |
| 07:06.14 | Psykick | not overly huge but we are currently at 40 seats |
| 07:06.24 | holmeh | I am thinking about converting from "some branded product from our telecom supplier" to asterisk. |
| 07:06.54 | Assid | Feb 15 02:03:39 localhost kernel: Registered tone zone 0 (United States / North America) |
| 07:06.54 | Assid | Feb 15 02:03:39 localhost kernel: rtc: lost some interrupts at 1024Hz. |
| 07:06.59 | Psykick | best suggestion I can make is to use decent quality headsets (this was our biggest issue with voice quality) |
| 07:07.00 | Assid | thats where im losing out |
| 07:07.16 | Psykick | that and IRQ |
| 07:07.21 | holmeh | I am just getting into using asterisk now, though, Psykick. :-) |
| 07:07.31 | holmeh | Started to play with it before last weekend. |
| 07:07.46 | Psykick | I'm in the process at the moment of developing a better AMP |
| 07:07.55 | holmeh | So I am idling in here to catch some hints and pointers. |
| 07:07.57 | Psykick | using hash tables for the family's and keys |
| 07:08.07 | Assid | Psykick: so what do you suggest i do with this? |
| 07:08.32 | Assid | it only does that when i load up ztdummy |
| 07:08.36 | holmeh | What is interesting is to get my friends and family to hook up to asterisk, and use IAX2 - then distribute extentions to have free calls. :-P |
| 07:08.42 | trixter | wow looks like a new service I launched (sorta) 2 days ago is gonna push 1.5M minutes this month.. who'd a thought that free tollfrees where you can spoof ani and no registration would be so popular |
| 07:08.44 | Psykick | Assid: ok try assigning an IRQ to your card (in bios) .... turn off smp ... turn off apic |
| 07:08.58 | Assid | no card |
| 07:09.09 | Psykick | Assid: and read the tips on this site http://www.asteriskguru.com/tutorials/pci_irq_apic_tdm_ticks_te410p_te405p_noise.html |
| 07:09.18 | holmeh | Do any of you run high-traffic asterisk servers? |
| 07:09.20 | Psykick | Assid: ok what's the rest of the problem |
| 07:09.43 | Psykick | holmeh: not at the moment but using SIP and SER should take care of things nicely |
| 07:10.05 | holmeh | How is this going in the means of traffic? |
| 07:10.08 | holmeh | (Network traffic) |
| 07:10.17 | Assid | well..i need ztdummy so i can use meetme.. but when i load it up, play() dies out... |
| 07:10.40 | Assid | so i got it figured that if i clean up zaptel issue.. everything SHOULD fall in place |
| 07:10.42 | Psykick | holmeh: well we're using g729 codec so it uses bugger all network traffic |
| 07:11.36 | Psykick | g729 = 8 kbps |
| 07:11.42 | Psykick | awesome voice quality as well |
| 07:11.43 | holmeh | Okay. Nice. |
| 07:11.52 | holmeh | 8/8 symmetric? |
| 07:11.57 | lucasjb | Does the context in extensions.conf have anything to do with music on hold? |
| 07:12.02 | holmeh | per client, right. |
| 07:12.23 | Psykick | holmeh: http://www.asteriskguru.com/tutorials/pci_irq_apic_tdm_ticks_te410p_te405p_noise.html |
| 07:12.26 | Psykick | oops |
| 07:12.27 | Psykick | sorry |
| 07:12.31 | holmeh | :-) |
| 07:12.37 | Psykick | holmeh: http://www.voip-info.org/wiki-Bandwidth+consumption |
| 07:13.05 | holmeh | So, lets say you get 600 clients with the g729 codec, you will use 5mbps if everyone call at once? It's not much. :-) |
| 07:13.08 | holmeh | Thanks. |
| 07:13.22 | Psykick | Assid/Holmeh: I've gotta get home .... will come back on when I get there |
| 07:13.31 | holmeh | I gotta get to work. |
| 07:13.35 | Psykick | lol |
| 07:13.39 | Psykick | talk to you guys soon |
| 07:13.59 | Assid | tronix: you still around |
| 07:15.32 | Assid | damn.. im stuck |
| 07:18.08 | *** join/#asterisk xeet2 (n=xeet3@72.1.157.34) |
| 07:18.25 | holmeh | I try to get out of the house, but I am stuck by the computer. :-P |
| 07:18.47 | holmeh | It's warm and nice here... :) |
| 07:19.24 | xeet2 | what would be the easiest way to search through a text file with about 6k lines, find a phone number, and call that number? I'm assuming I should just use mysql, but are there any easy ways to just do this with a text file? |
| 07:32.32 | *** join/#asterisk EriSan (n=erisan@151.8.109.84) |
| 07:35.04 | *** join/#asterisk los415 (i=los415@los.race.com) |
| 07:40.21 | *** join/#asterisk ramtha (n=ramtha@195.14.234.162) |
| 07:40.38 | ramtha | hi, i have two quad4 cars in one mashine |
| 07:41.11 | ramtha | zaptel for one card was easy, know i added span 5 - 8 and added the channels |
| 07:41.24 | ramtha | but /proc/zaptel only shows 5 devices instead of 8 |
| 07:41.35 | ramtha | ztcfg say noch sich device 6 |
| 07:41.47 | ramtha | where is the magic behind two cards? |
| 07:43.43 | *** join/#asterisk kmilitzer (n=km@office-gw.westend.com) |
| 07:43.54 | kmilitzer | Good morning everyone ... |
| 07:45.12 | kmilitzer | Does anybody have an idea how to use different SIP peers in a round robin fashion when dialing out? |
| 07:45.35 | Qwell | use a queue to dial out? |
| 07:45.47 | Qwell | probably not worth the effort, heh |
| 07:46.03 | brookshire[home] | HAHAH |
| 07:46.04 | trixter | its not that hard |
| 07:46.11 | brookshire[home] | qwell: you're brilliant :) |
| 07:46.15 | kmilitzer | Well, I tried that, but I didnt't find a way how to give the extension to the agent ... |
| 07:46.17 | Qwell | I could see a macro with a bunch of AddQueueMember()'s |
| 07:46.21 | Qwell | That'd be hot |
| 07:46.35 | Qwell | kmilitzer: ^ |
| 07:46.44 | brookshire[home] | that would be the best use of asterisk ever! |
| 07:47.08 | Qwell | strategy=ringall |
| 07:47.08 | kmilitzer | My problem is, that I have two or more PSTN-Gateways, that I want to load balance the calls for ... like 1st call 1st GW, 2nd call 2nd GW, 3rd Call 1st GW and so on ... |
| 07:47.14 | Qwell | add like...200 random people |
| 07:47.20 | Qwell | every call you get > 200 people |
| 07:47.21 | brookshire[home] | only 200? |
| 07:47.28 | Qwell | as an e.g. |
| 07:47.45 | brookshire[home] | actually.. i think that's the point of queues... so it's probably not that far fetched, lol |
| 07:47.58 | Qwell | it'd be like...reverse queue |
| 07:48.02 | trixter | you could even use variables and a for loop each variable contains the provider specific info in it |
| 07:48.21 | trixter | if you want it to be round robbin for all calls think global variables |
| 07:48.57 | Qwell | oh, man... |
| 07:49.02 | ManxPower | um, generate a random number between 1 and max providers, set the values of variables you pass to Dial to the required info. |
| 07:49.04 | Qwell | I could use queues to call a radio station |
| 07:49.08 | kmilitzer | So how would it look exactley? |
| 07:49.17 | Qwell | Why didn't I think of that before/ |
| 07:49.22 | brookshire[home] | why use queues when you can use /var/spool/asterisk/outgoing ?? |
| 07:49.31 | Qwell | because...then I can use ringall :D |
| 07:49.40 | brookshire[home] | & ! |
| 07:49.44 | Qwell | have it call the station like...spawn 50 times |
| 07:49.57 | Qwell | then it'll constantly retry, and as soon as one answers...bam, they all stop |
| 07:50.00 | brookshire[home] | you should write a patch for that purpose |
| 07:50.01 | trixter | Qwell: that is not that different from what someone posted to asterisk users and I commented on month ago or so on the list :) |
| 07:50.12 | Qwell | trixter: meh, -users is lame |
| 07:50.12 | brookshire[home] | call it.. app_american_idol |
| 07:50.17 | Qwell | brookshire[home]: haha |
| 07:50.23 | Qwell | I wonder if it costs money to vote? |
| 07:50.27 | Qwell | we could all like... |
| 07:50.30 | Qwell | pool our resources |
| 07:50.32 | ManxPower | kmilitzer, ask if you see me active in the morning and I'll write up an example when I'm not tired. |
| 07:50.36 | brookshire[home] | nah.. i think they make you text this year |
| 07:50.36 | Assid | Qwell! hows you |
| 07:50.37 | Qwell | make the spread like...massive |
| 07:50.38 | trixter | I have a ds3 for american idol, I will ask someone who actually watches it to tell me whom to vote for (the worst candidate) 3 hours of calling ds3 1 minute calls |
| 07:50.39 | brookshire[home] | i don't keep up |
| 07:50.40 | trixter | you do the math |
| 07:50.40 | outtolunc | obviously noone has heard of mitnick |
| 07:50.44 | brookshire[home] | BUT STILL! |
| 07:50.45 | trixter | I can make anyone a winner! |
| 07:51.01 | kmilitzer | Right now my queue.conf looks like this for the agents: member => SIP/${EXTEN}@nc-out |
| 07:51.01 | kmilitzer | member => SIP/${EXTEN}@bt-out |
| 07:51.02 | ramtha | no one installed 2 TE4XXP in one mashine? |
| 07:51.08 | trixter | mitnick didnt do the radio station stuff that was agent steel (kevin pulson) |
| 07:51.11 | kmilitzer | but this does not work |
| 07:51.18 | trixter | and he owned the pbx to rig the contest to win the porche |
| 07:51.42 | los415 | didnt they start charging for american idol |
| 07:51.44 | *** join/#asterisk nagl (n=nagl@137.208.4.162) |
| 07:51.50 | Qwell | los415: for the sms, yeah, it costs |
| 07:51.59 | Qwell | I don't know about calling though |
| 07:52.18 | los415 | hrmmm dunno |
| 07:52.25 | los415 | i really dont pay attention to that show |
| 07:52.44 | trixter | I heard its tollfrees |
| 07:52.53 | Qwell | used to be. probably still is |
| 07:52.53 | brookshire[home] | lol |
| 07:53.00 | *** join/#asterisk A-Tuin|work (n=A-Tuin@212.41.185.81) [NETSPLIT VICTIM] |
| 07:53.00 | *** join/#asterisk riksta (n=rick@213.121.151.210) [NETSPLIT VICTIM] |
| 07:53.00 | *** join/#asterisk kll (i=kll@insomnia.juniks.net) [NETSPLIT VICTIM] |
| 07:53.01 | *** join/#asterisk dant (n=dan@host-84-9-188-2.bulldogdsl.com) [NETSPLIT VICTIM] |
| 07:53.02 | trixter | so they legally cant charge for the call, they would have to accept a credit card or something that and that ugly |
| 07:53.04 | Qwell | which means you need to change ANI also |
| 07:53.08 | trixter | no one would do it so it would be free |
| 07:53.15 | trixter | I thought it was 900s but was told tonight its 800s |
| 07:53.22 | Qwell | it's like 866 or 877 |
| 07:53.23 | trixter | Qwell: that is trivial |
| 07:53.33 | los415 | heheh |
| 07:53.36 | trixter | 800 is like kleenex or coke |
| 07:53.39 | trixter | generic term now :P |
| 07:53.39 | los415 | flood there lines with a couple ds3's |
| 07:53.45 | brookshire[home] | qwell: you need to make it distributed too |
| 07:53.51 | Qwell | brookshire[home]: indeed |
| 07:53.52 | trixter | I only have one *and* I get $$ for each call I place :P |
| 07:53.56 | Assid | Qwell: could you help me with a zaptel issue? |
| 07:53.59 | Qwell | los415: 1 DS3 won't hurt them at all... |
| 07:54.07 | Qwell | los415: I'm sure they have tons of capacity |
| 07:54.08 | trixter | well to tollfrees anyway, why I give em free and let people specify arbitrary ani |
| 07:54.32 | trixter | 1 ds3 all voting for the worst person for 3 hours when they are 1 minute calls will skew the results |
| 07:54.40 | Qwell | that it will |
| 07:54.41 | trixter | the idea is not to take em out its to make the worst person win |
| 07:55.07 | los415 | heh |
| 07:55.12 | Assid | whenever i load up ztdummy i get rtc errors |
| 07:55.17 | los415 | a couple ds3's would really screw with results |
| 07:55.18 | *** join/#asterisk oej (n=oej@82.147.33.178) |
| 07:55.36 | Assid | <PROTECTED> |
| 07:55.44 | *** join/#asterisk X-Gen (n=claude@dsl-146-97-08.telkomadsl.co.za) |
| 07:55.49 | Qwell | actually...it won't skew it that much |
| 07:55.57 | Qwell | That's only 117,000 votes |
| 07:56.04 | brookshire[home] | lol |
| 07:56.09 | Qwell | they get some 500k+ |
| 07:56.10 | brookshire[home] | that's why we got to work together |
| 07:56.18 | Qwell | (500k+ per person) |
| 07:56.33 | trixter | um an *additional* 117k votes is > 20% |
| 07:56.33 | Qwell | though...the range may be enough |
| 07:56.36 | trixter | that is a fairly major skew |
| 07:56.37 | Qwell | yeah |
| 07:56.50 | Qwell | If they sucked THAT bad, they'd get 0 votes though :P |
| 07:56.59 | brookshire[home] | <PROTECTED> |
| 07:57.05 | trixter | I have no plans of watching it though |
| 07:57.06 | Qwell | brookshire[home]: and ANI, no doubt |
| 07:57.13 | trixter | and 117k votes would yield me some quick cash to boot |
| 07:57.17 | brookshire[home] | so.. i doubt it will count all that much |
| 07:57.21 | trixter | since I get $$ per minute I am connectedto a tollfree |
| 07:57.22 | trixter | :D |
| 07:57.27 | Qwell | trixter: eh? |
| 07:57.32 | brookshire[home] | it would be fun to chanspy all of those channels, lol |
| 07:58.32 | Qwell | put them into one big ass meetme |
| 07:58.43 | trixter | at any rate I plan to offer free termination to tollfrees via sip and iax (there appears to be a problem with sip in asterisk 1.2.4 in fbsd 6, once resolved it will be opened) and lets people specify arbitrary ani for the calls |
| 07:58.51 | trixter | if theydont specify something valid it wil assign a random one |
| 07:59.01 | Qwell | trixter: How do you manage to get money for calling tollfree? |
| 07:59.47 | los415 | become a clec |
| 08:00.41 | *** join/#asterisk w32 (n=123@adsl-70-224-65-121.dsl.sbndin.ameritech.net) |
| 08:01.23 | los415 | i have a question have been looking in voip-info at all the calling card / pre paid apps out there which would u guys say is the best and best keeped up |
| 08:02.06 | trixter | Qwell: skill? |
| 08:03.03 | trixter | if anyone can push large volume tollfree traffic to US/CA I can work out some split on the revenue, it all depends on volume |
| 08:03.09 | trixter | low volume you gotta use the free site :P |
| 08:04.19 | trixter | its midnight and I have 96 active channels.. well at least that is better than yesterday with only 30-40 ... not bad for only doing this for 2-3 days now and still in testing :D |
| 08:06.15 | w32 | hi |
| 08:06.15 | w32 | not much going on tonite eh? |
| 08:07.06 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 08:10.07 | trixter | if anyone sets up asterisk boxes for customers this is one way to add revenue to those setups :) |
| 08:12.41 | *** join/#asterisk TonyM (n=TonyM@adsl-solo-80-168-226-4.claranet.co.uk) |
| 08:14.14 | kmilitzer | I still need a way to distribute outgoing calls in a round robin fashin over different GWs ... and i still don't know how to make that work with queues ... so anybody knowing how to do it please let me know ;) |
| 08:15.14 | Skumling | trixter: which way? :-) |
| 08:17.07 | brookshire[home] | kmilitzer: call group? |
| 08:17.15 | trixter | Skumling: if you call tollfrees I can split the money I get for terminating them to the PSTN but only if there is volume, if there isnt volume then you gotta rely on the free service :P |
| 08:18.11 | Skumling | trixter: humm okay. actually I've never really understood the term "tollfree calls"... |
| 08:18.44 | Skumling | trixter: is it just numbers you can call for free? |
| 08:18.50 | *** join/#asterisk astar` (n=astar@ANantes-154-1-22-24.w81-53.abo.wanadoo.fr) |
| 08:19.05 | *** join/#asterisk droops (n=droops@adsl-065-005-212-128.sip.jan.bellsouth.net) |
| 08:19.12 | kmilitzer | brookshire[home]: Does call groups work with SIP? I thought that wasn't possible |
| 08:19.21 | trixter | Skumling: you call free the receiver pays for the call |
| 08:19.31 | Skumling | trixter: ah, okay. |
| 08:20.03 | Skumling | trixter: do you run a PSTN gateway? |
| 08:22.13 | *** join/#asterisk Koshatul (n=evangeli@ip157-65-132.cust.bit.net.au) |
| 08:22.30 | trixter | yes |
| 08:25.19 | Skumling | trixter: cewl. |
| 08:25.28 | Skumling | trixter: then you are kind of telco ;-) |
| 08:26.27 | trixter | kinda |
| 08:26.39 | trixter | I am trying to do national DIDs as well give those away free for inbound usage |
| 08:26.53 | trixter | but that may take a little more time than outbound does |
| 08:27.29 | X-Rob | trixter, advertise 1-800 through e164.org |
| 08:27.44 | trixter | you are supposed to own the number you publish |
| 08:27.53 | X-Rob | no, not .arpa, .org |
| 08:27.54 | trixter | they are supposed to verify that you own it as well |
| 08:28.04 | trixter | thge ietf runs that one too right? |
| 08:28.17 | X-Rob | nope |
| 08:28.22 | *** join/#asterisk adelas (n=booger@rrcs-24-199-21-141.west.biz.rr.com) |
| 08:29.14 | X-Rob | they currently advertise voipmich for all 1-800 terminations |
| 08:29.41 | trixter | I know freenum does that |
| 08:29.41 | X-Rob | e164.org NAPTR query results (1 records) :: |
| 08:29.42 | X-Rob | sip:18005551212@tf.voipmich.com |
| 08:29.45 | trixter | voipmitch I mean |
| 08:30.19 | X-Rob | so speak to 'em - the person to speak to is 'evilbuny' on here. |
| 08:30.46 | *** join/#asterisk frenzy (n=frenzy@196.45.144.40) |
| 08:30.52 | X-Rob | (not currently online tho 8) |
| 08:31.14 | holmeh | I wonder why musiconhold() is screwed =) |
| 08:31.21 | holmeh | I never got it to play music |
| 08:31.23 | X-Rob | holmeh, you don't have a timing source, that's why. |
| 08:31.25 | X-Rob | oh |
| 08:31.31 | X-Rob | many reasons for that. |
| 08:32.53 | holmeh | aha. |
| 08:33.14 | holmeh | <PROTECTED> |
| 08:33.19 | holmeh | <PROTECTED> |
| 08:33.20 | holmeh | :-P |
| 08:33.23 | *** join/#asterisk cuco (n=diego@local.xorcom.com) |
| 08:33.25 | holmeh | It says it's playing then stopping |
| 08:33.35 | holmeh | But, I dunno - it's weird. |
| 08:34.00 | holmeh | I should give more RAM to this box. =) |
| 08:37.04 | X-Gen | that sounds like, oh its not working, lets reboot the box |
| 08:37.16 | holmeh | I just saw it sucked up all the ram |
| 08:37.24 | holmeh | it's virtual, so I gave it 512 more megs |
| 08:38.06 | *** join/#asterisk BugKham (n=lamer@202.8.86.170) |
| 08:38.28 | Assid | Qwell: you around? |
| 08:38.44 | Qwell | Assid: barely |
| 08:38.52 | BugKham | anyone using the E1 + FXOs? |
| 08:38.57 | Assid | could you help me with this zaptel issue? |
| 08:39.00 | Assid | please |
| 08:39.04 | Qwell | Assid: probably not |
| 08:39.09 | Assid | :| |
| 08:40.00 | Assid | <PROTECTED> |
| 08:40.05 | Assid | just doesntmake sense |
| 08:40.25 | BugKham | how is this zaptel config? span=1,1,0,ccs,hdb3,crc4 bchan=1-15 dchan=16 bchan=17-31 fxsks=32-33 |
| 08:41.05 | brookshire[home] | assid: just buy an x100p for timing.. and ditch ztdummy |
| 08:41.08 | Qwell | isn't an E1 32 channels? I can't think right now, but 31 seems wrong |
| 08:41.18 | X-Rob | BugKham, looks like an Onramp30 to me. |
| 08:41.25 | X-Rob | Qwell, no, that's correct. |
| 08:41.29 | holmeh | X-Rob: lacking mpg123 =) |
| 08:41.32 | holmeh | fixed it |
| 08:41.36 | Qwell | it is 31 channels? |
| 08:41.40 | X-Rob | 30 B channels + 1 D channel == 31 |
| 08:41.44 | BugKham | QWell: yeah one for framing and the other for signalling |
| 08:41.46 | Qwell | odd...okay |
| 08:41.57 | brookshire[home] | assid: optionally you can try svn zaptel |
| 08:42.11 | X-Rob | BugKham, you in .au? |
| 08:42.18 | Assid | brookshire: i had a FA82537EP modem..but same issues |
| 08:42.32 | BugKham | X-Rob: have you got TDM running together with E1s? |
| 08:42.37 | *** join/#asterisk pif (n=ldm@zenon.apartia.fr) |
| 08:42.44 | *** join/#asterisk spunz_ (n=spunz@h081217096096.dyn.cm.kabsi.at) |
| 08:42.46 | X-Rob | BugKham, 'TDM' means 'Time Division Multiplexing' |
| 08:42.51 | BugKham | X-Rob: no, in .th |
| 08:42.53 | X-Rob | What exactly do -you- mean by TDM? |
| 08:42.54 | Assid | brookshire[home]: if i use SVN.. apache+php doesnt wanna recomile if i need it |
| 08:43.02 | *** join/#asterisk sulex (n=sulex@pdpc/supporter/active/sulex) |
| 08:43.12 | brookshire[home] | assid: what does svn have to do with apache+php |
| 08:43.21 | Assid | something with libpri |
| 08:43.26 | Assid | or rather.. libapr |
| 08:43.32 | brookshire[home] | is this gentoo? |
| 08:43.35 | Assid | debian |
| 08:43.53 | Assid | i moved my home box to gentoo.. but didnt see if it causes that problem |
| 08:44.09 | brookshire[home] | why are you compiling apache+php anyways? |
| 08:44.22 | Qwell | time for bed...I'm so tired |
| 08:44.33 | Assid | need it there.. for web interface |
| 08:44.47 | Assid | i dotn like debs |
| 08:44.52 | Assid | atleast for those |
| 08:44.57 | brookshire[home] | heh |
| 08:45.02 | Qwell | If you're going to compile something with deps... |
| 08:45.02 | Assid | certain things.. i prefer compiling |
| 08:45.03 | brookshire[home] | then compile subversion |
| 08:45.04 | brookshire[home] | :) |
| 08:45.06 | Qwell | you often need to compile the deps |
| 08:45.10 | Qwell | What he said |
| 08:45.16 | Qwell | or, at least, apr |
| 08:45.26 | Qwell | apr = Apache Portable Runtime |
| 08:45.39 | brookshire[home] | qwell: i never needed to :( |
| 08:45.48 | brookshire[home] | well there was this one time |
| 08:45.55 | brookshire[home] | 'back in the day' |
| 08:46.00 | Qwell | brookshire[home]: if you compile the latest and greatest, but you use stable...you'll have issues |
| 08:46.00 | Assid | okay what if i download the svn of zaptel on windows.. and copy it over to that box |
| 08:46.07 | brookshire[home] | i ran out of files in linux.. |
| 08:46.16 | brookshire[home] | and well.. tried to fix it with apache |
| 08:46.16 | Qwell | stable packages, that is |
| 08:46.40 | brookshire[home] | so i had to recode apache |
| 08:47.14 | Assid | so you suggest i get zaptel(svn) ? |
| 08:47.25 | brookshire[home] | worth a shot |
| 08:47.26 | brookshire[home] | :) |
| 08:47.32 | Qwell | mv Qwell /dev/bed |
| 08:47.40 | Qwell | chattr +comfy Qwell |
| 08:47.45 | Assid | damn.. this is gonna get ugly for me |
| 08:48.02 | brookshire[home] | or! |
| 08:48.04 | brookshire[home] | you can try this |
| 08:48.05 | brookshire[home] | http://www.backports.org/debian/pool/main/z/zaptel/ |
| 08:48.17 | X-Rob | I use zaptel-trunk and libpri-trunk with asterisk/branches/1.2 everywhere. |
| 08:48.27 | X-Rob | the echo can in zaptel trunk is _far_ superior to 1.2 |
| 08:49.03 | Assid | 1.2.1 ? |
| 08:49.18 | brookshire[home] | you can mix versions |
| 08:49.27 | brookshire[home] | but that's a deb! |
| 08:49.28 | brookshire[home] | :D |
| 08:49.37 | brookshire[home] | so probably tested and works |
| 08:49.38 | Assid | i got 1.2.3 and still giving me issues |
| 08:49.42 | Assid | but i got other debian boxes running this perfectly fine.. |
| 08:50.03 | brookshire[home] | 1.2.3 is radically different from 1.2.2 |
| 08:50.09 | brookshire[home] | from what i understand |
| 08:50.25 | brookshire[home] | and same with 1.2.4 from 1.2.3 |
| 08:51.02 | brookshire[home] | oh yeah.. 1.2.4 isn't out yet |
| 08:51.03 | brookshire[home] | oops |
| 08:51.07 | brookshire[home] | lol.. it will be tomorrow |
| 08:51.09 | Assid | no way to figure out why the RTC just wants to die |
| 08:51.43 | brookshire[home] | bascially.. i have no idea |
| 08:51.50 | brookshire[home] | but upgrading might fix it :) |
| 08:51.55 | brookshire[home] | or downgrading |
| 08:52.14 | Assid | the system has.. SMP (intel HT) |
| 08:52.20 | Assid | you think its cause of that? |
| 08:52.33 | brookshire[home] | hmm.. maybe |
| 08:52.44 | brookshire[home] | i know they have had SMP issues before with ztdummy |
| 08:52.47 | kmilitzer | Queues suck ;) |
| 08:53.01 | Assid | okay time to get rid of SMP then acpi ? |
| 08:53.14 | *** join/#asterisk BugKham (n=lamer@202.8.86.170) |
| 08:53.18 | Assid | http://pastebin.com/554746 <-- thats how the system was before i got rid of the modem |
| 08:53.20 | kmilitzer | Everybody seems to think queues are the answer to everything, but that's wrong!!! |
| 08:53.40 | brookshire[home] | is this SMP as in hyperthreading? |
| 08:53.48 | Assid | yeah (HT) |
| 08:53.48 | brookshire[home] | or SMP as in dual processor |
| 08:53.57 | brookshire[home] | i don't think it ever affected that |
| 08:54.04 | brookshire[home] | but i could be wrong, yet again! |
| 08:54.09 | trixter | queues are great |
| 08:54.17 | holmeh | Does anyone have an idea why the music on hold is just noise? =) |
| 08:54.18 | brookshire[home] | (when they work) |
| 08:54.20 | brookshire[home] | :D |
| 08:54.20 | trixter | why when I was living in scotland I stood in a queue all the time |
| 08:54.25 | trixter | goto mcdonalds there is a queue |
| 08:54.29 | trixter | goto the train station a queue |
| 08:54.33 | holmeh | haha. |
| 08:54.34 | trixter | they dont suck |
| 08:54.44 | Assid | dude.. i live in india.. with 2nd highest population in the world.. |
| 08:54.56 | Assid | queues are like the livelihood we live by |
| 08:55.00 | kmilitzer | trixter: OK, that's right, queues IRL are great ... only in asterisk they aren't |
| 08:55.02 | Assid | the code of ethics |
| 08:55.08 | holmeh | I live in Norway, we got 4 million for the whole country. :) |
| 08:55.09 | trixter | hehe |
| 08:55.16 | Assid | lucky SOB.. |
| 08:55.20 | trixter | ireland (free) only has 4M |
| 08:55.31 | trixter | wonder what the land mass difference is |
| 08:55.38 | trixter | cause most of ireland is quite rural, 50% live in dublin county |
| 08:55.51 | trixter | cork county is pretty high up on the list of population too |
| 08:55.52 | brookshire[home] | hei! |
| 08:56.09 | holmeh | we got about 10-15% living here in Oslo :-) |
| 08:56.11 | holmeh | the capital |
| 08:56.26 | brookshire[home] | that's the only word i know in norwegian :( |
| 08:56.28 | trixter | well the city of dublin is quite small so it wouldnt be that many people there, but the county is where they all live |
| 08:56.39 | holmeh | brookshire[home]: The opposite is 'hadet' :-) |
| 08:56.48 | holmeh | as in 'bye' |
| 08:56.54 | trixter | is that like the hadets at starfleet academy? |
| 08:56.58 | Assid | okay so what all do i disable .. SMP and acpi ? |
| 08:57.02 | trixter | they go in as hadets come out as officers |
| 08:57.16 | astar` | hello |
| 08:57.17 | trixter | but with holosuites and replicators you gotta wonder why anyone actually works |
| 08:57.20 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 08:57.25 | Assid | do i need HPET timer support? |
| 08:57.29 | brookshire[home] | Ja! |
| 08:57.37 | brookshire[home] | i know that too |
| 08:57.45 | astar` | in incoming calls via pstn i have a bad quality voice but incoming calls via internet are very good |
| 08:57.48 | Assid | and SMT? |
| 08:57.56 | brookshire[home] | Mitt navn er matt ;) |
| 08:58.00 | astar` | is this a zapata.conf setting ? |
| 08:58.11 | Assid | "Local APIC support on uniprocessors (NEW)" do i need that? |
| 08:58.23 | holmeh | brookshire[home]: ;-) |
| 08:58.38 | brookshire[home] | oh... and |
| 08:58.45 | brookshire[home] | Hvor er toalettet? |
| 08:58.47 | holmeh | haha. |
| 08:58.52 | Assid | brookshire[home]: do i need those? |
| 08:59.10 | BugKham | X-Rob: hi, I missed your last message |
| 08:59.11 | brookshire[home] | assid: i have no idea :) |
| 08:59.23 | X-Rob | BugKham, yes you did. No idea what you want to do. |
| 08:59.59 | holmeh | I still wonder why musiconhold is just noise, I play mp3 with madplay or mpg123 |
| 09:00.06 | BugKham | X-Rob: I need to config and E1 card with 2 FXOs |
| 09:00.11 | holmeh | both returns *noise* |
| 09:00.29 | X-Rob | BugKham, your E1 card has 10 20 or 30 lines. Not 2. |
| 09:00.32 | X-Rob | Uh |
| 09:00.37 | X-Rob | E1 _line_ from your telco |
| 09:00.43 | BugKham | X-Rob: but not sure if my config. is right as I haven't had an actual link to my card |
| 09:00.43 | brookshire[home] | holmeh: good noise, or no noise? |
| 09:01.21 | BugKham | X-Rob: yeah, I'll ask for an ISDN PRI |
| 09:01.32 | holmeh | brookshire[home]: just noise |
| 09:01.41 | holmeh | like when the TV falls out |
| 09:01.49 | X-Rob | Yes. That gives you 30 FXS ports. |
| 09:01.54 | X-Rob | (or 20 or 10) |
| 09:02.05 | X-Rob | your zaptel.conf config, as posted, is correct |
| 09:02.11 | holmeh | I am dealing a PRI tomorrow. |
| 09:02.15 | holmeh | Actually three PRI lines |
| 09:02.15 | brookshire[home] | sounds like mpg123 is messed up :( |
| 09:02.21 | BugKham | X-Rob: here ther's no partial ISDN PRI. I will have 30 |
| 09:02.21 | holmeh | madplay does the same |
| 09:02.44 | holmeh | I tried with two different mp3s at a static bitrate |
| 09:02.59 | BugKham | X-Rob: what do you think of this zaptel.conf? span=1,1,0,ccs,hdb3,crc4 bchan=1-15 dchan=16 bchan=17-31 fxsks=32-33 |
| 09:03.11 | X-Rob | I told you before, that's correct for an E1. |
| 09:03.22 | brookshire[home] | do the mp3s play from mpg123 on another box? |
| 09:03.23 | X-Rob | Depending on your telco, you may or may not need the crc4 on the end. |
| 09:03.31 | holmeh | I will try, brookshire[home], just a second. |
| 09:03.47 | BugKham | X-Rob: ok, what about the rest? |
| 09:04.03 | X-Rob | BugKham, THAT IS CORRECT |
| 09:04.04 | holmeh | Yes, brookshire[home] |
| 09:04.08 | X-Rob | FOR FUCKS SAKE READ WHAT I SAY |
| 09:04.46 | BugKham | X-Rob: oh, right. sorry I missread it |
| 09:05.08 | brookshire[home] | it's the correct version right? |
| 09:05.22 | holmeh | of mpg123? |
| 09:05.26 | brookshire[home] | yeah |
| 09:05.33 | holmeh | That might be the problem. |
| 09:05.34 | Assid | okay set the box to recompile |
| 09:05.50 | brookshire[home] | you must have a 0.59r one |
| 09:05.52 | brookshire[home] | i believe |
| 09:05.56 | BugKham | X-Rob: what about in zapata.conf? signalling=pri_cpe, signalling=fxsks? |
| 09:05.58 | brookshire[home] | mpg123 -v |
| 09:06.08 | holmeh | Yeah, I just apt-get it |
| 09:06.14 | holmeh | So, it's probably the wrong version |
| 09:06.41 | BugKham | X-Rob: and switchtype=euroisdn |
| 09:07.09 | BugKham | X-Rob: I don't know where to place those signalling stuffs |
| 09:10.05 | *** join/#asterisk pif (n=ldm@zenon.apartia.fr) |
| 09:10.19 | X-Rob | BugKham, please call digium for support. |
| 09:10.33 | X-Rob | You need handholding. You've bought a digium card, they will hold your hand. |
| 09:11.03 | brookshire[home] | email works too |
| 09:11.11 | brookshire[home] | support@digium.com |
| 09:11.12 | brookshire[home] | :D |
| 09:11.53 | pif | should I get HW EC on my digium card? |
| 09:12.03 | pif | or can zaptel EC take care of most cases? |
| 09:12.52 | brookshire[home] | pif: which card? |
| 09:13.16 | BugKham | X-Rob: okay |
| 09:13.34 | brookshire[home] | tdm2400 or te4xx ? |
| 09:13.55 | astar` | in incoming calls via pstn i have a bad quality voice but incoming calls via internet are very good , is this a zapata.conf setting ? |
| 09:13.57 | pif | brookshire: a TE411P |
| 09:14.26 | pif | or a TE406P |
| 09:15.00 | brookshire[home] | well.. if cpu is really not an issue... then either hw or sw echo can should be enough |
| 09:15.04 | pif | er, I mean I'm hesitating between a 411 and a 410 |
| 09:15.56 | pif | ok, os basically digium HW EC is the same as zaptel but in software? |
| 09:16.07 | brookshire[home] | and apparently the software echo can in zaptel head "kicks ass" |
| 09:16.14 | brookshire[home] | no no no.. |
| 09:16.18 | *** join/#asterisk bigjb_ (n=bigjb@195.60.10.114) |
| 09:16.19 | brookshire[home] | completely different |
| 09:16.58 | pif | as digium writes zaptel _and_ conceives the cards, one could infer a proximity in implementations |
| 09:17.50 | pif | or maybe they give a "poor man's" implementation in zaptel to funnel clients to the high-end cards? |
| 09:18.04 | brookshire[home] | no.. not the case |
| 09:18.27 | brookshire[home] | hardware echo can is really just easier to correctly setup |
| 09:18.31 | brookshire[home] | and uses less resources |
| 09:19.02 | pif | because of the pci bus latency? |
| 09:19.14 | brookshire[home] | huh? |
| 09:19.49 | pif | cpu not being an issue, pci bus latency makes it harder to EC in software? |
| 09:20.08 | brookshire[home] | probably not |
| 09:20.15 | *** join/#asterisk gr0mit (n=w10277@206.41.25.138) |
| 09:21.05 | brookshire[home] | anyways.. ni ni |
| 09:21.11 | brookshire[home] | need to sleep |
| 09:21.12 | brookshire[home] | :) |
| 09:22.03 | pif | did you brush your teeth? |
| 09:23.40 | Assid | damn its still compiling the kernel |
| 09:30.10 | *** join/#asterisk welles (n=welles@222.90.170.64) |
| 09:31.00 | welles | hi all |
| 09:32.56 | *** join/#asterisk Pegger (n=peg@pool-68-163-155-106.bos.east.verizon.net) |
| 09:33.14 | Pegger | I am having toruble starting a newly compiled asterisk |
| 09:33.24 | Pegger | homer:/usr/src/asterisk# asterisk -vvvvvvvvvvvvvvr |
| 09:33.25 | Pegger | Unable to connect to remote asterisk (does /var/run/asterisk.ctl exist?) |
| 09:33.28 | astar` | someone know what to do when incoming pstn calls have poor quality but not incoming internet calls |
| 09:33.35 | *** join/#asterisk Abbas (i=Abbas@203.81.200.119) |
| 09:33.49 | Pegger | astar`, ditch the pstn provider |
| 09:34.03 | trixter | Pegger: did you first start asterisk? |
| 09:34.19 | Pegger | yes root 10440 0.0 5.5 15424 7000 ? Ss 04:19 0:00 asterisk |
| 09:35.16 | astar` | if i put a phone directly on the pstn no problem |
| 09:35.18 | trixter | when you run asterisk -r are you running it with enough perms to read the ctl file? |
| 09:35.25 | trixter | which is likely to be root in your case |
| 09:36.03 | Pegger | well i just checked and I dont seam to have /var/run/asterisk.ctl |
| 09:36.06 | *** join/#asterisk wellng (n=welles@222.90.170.64) |
| 09:36.12 | Pegger | when does it get created |
| 09:36.55 | trixter | when asterisk starts |
| 09:37.23 | Pegger | oah cool it started woring |
| 09:37.56 | wellng | trixter, are u talking to me? |
| 09:39.18 | trixter | no |
| 09:40.08 | wellng | trixter, do u use mp3player on asterisk |
| 09:40.23 | trixter | sometimes |
| 09:40.56 | wellng | what 's difference mp3player and waitmuscionhold? |
| 09:41.28 | wellng | waitmusiconhold also can play mp3 files |
| 09:41.34 | trixter | I have called mp3player directly from the dialplan to various things, mostly streams |
| 09:41.39 | trixter | I dont usei t as MoH |
| 09:43.01 | wellng | when i use mp3player the asterisk will hangup after several seconds and cause code is 16. the mp3 file has not finished .why? |
| 09:44.11 | Pegger | has anyone seen this type of problem? Rejected connect attempt from 69.25.143.141, who was trying to reach '16178303190@' |
| 09:44.53 | trixter | I dont know why your asterisk crashes.. perhaps its your mp3 file |
| 09:45.11 | wellng | maybe |
| 09:45.17 | trixter | Pegger: yeah someone who wasnt authorized from that IP tried to call someone on your box with that extension |
| 09:45.37 | trixter | its not a problem, if you want guests to be able to do that you should configure it so they can |
| 09:45.43 | *** part/#asterisk taec (n=phil@eventhorizon.hosting365.ie) |
| 09:46.45 | Pegger | trixter, well that is my did trying to call me, so when I call the DID from a regular phone I get that message |
| 09:47.05 | trixter | maybe you need to configure your box a little different |
| 09:47.10 | Pegger | trixter, I do have it in my extensions.conf http://pastebin.com/555660 |
| 09:47.27 | Pegger | but I dont understand why it is being rejected |
| 09:47.28 | trixter | some providers send stuff odd ways perhaps you need to adjust the type for the provider perhaps you need to add an insecure=... line |
| 09:47.30 | trixter | hard to say |
| 09:47.48 | trixter | it wouldnt reject from extensions.conf |
| 09:47.58 | trixter | if anything you would see a message that it cant find a given extension in a given context |
| 09:48.06 | trixter | it would instead be sip.conf or iax.conf or ... |
| 09:48.16 | Pegger | ok i wll check iax.conf |
| 09:51.44 | *** join/#asterisk welles (n=welles@222.90.170.64) |
| 09:52.05 | *** join/#asterisk Bambr (n=Bambr@213-35-237-83-dsl.end.estpak.ee) |
| 09:56.04 | Pegger | hum i am really confused could soem people please take a look at my pastebin http://pastebin.com/555666 |
| 09:58.39 | *** join/#asterisk basty (n=basty@212.218.65.233) |
| 09:58.40 | basty | Hi |
| 09:58.42 | Pegger | any ideas trixter |
| 09:58.55 | *** join/#asterisk NassoR (n=root@step.funcitec.rct-sc.br) |
| 09:59.20 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
| 10:00.37 | basty | If I use a G.711 Codec (64Kbit/s) how much Bandwidth is that for Up and Downstream? Each 87,2Kbit? or -> 43,6Kbit UP and 43,6Kbit down? |
| 10:03.30 | *** join/#asterisk welles (n=welles@222.90.170.64) |
| 10:04.42 | Assid | 64kbit up/down |
| 10:05.01 | Assid | its actually around 84kbit including the headers or so |
| 10:06.16 | basty | Ah so it is 64Kbit for incomming and 64kbit for outgoing...so 64kbit each! |
| 10:07.07 | *** join/#asterisk wellng (n=welles@222.90.170.64) |
| 10:07.23 | Assid | yep |
| 10:07.37 | Assid | catch around 8-9KB/sec |
| 10:07.56 | *** join/#asterisk florz (n=florz@2001:1a50:503c:0:0:0:0:1) |
| 10:08.11 | *** join/#asterisk boddy (n=e@212.58.24.138) |
| 10:08.18 | Assid | you may want to use 729 |
| 10:08.30 | boddy | hii Can I use asterix as sip server ? |
| 10:08.35 | Assid | which cuts it around 4.5KB/sec |
| 10:08.40 | Assid | boddy: yes.. |
| 10:08.56 | Assid | but if you are just using sip and no termination .. you could use SER |
| 10:09.08 | Assid | which would eat less resources |
| 10:12.36 | basty | Assisd: Okay cool - thanks. You may have an URL to check all different codecs ? |
| 10:13.29 | Assid | erer.. just google for it |
| 10:14.26 | *** join/#asterisk fishboy1669 (i=proxyuse@62.69.81.129) |
| 10:15.33 | basty | okay thanks |
| 10:15.39 | trixter | asteriskguro.org has a bandwidth calculator |
| 10:15.45 | *** join/#asterisk __chris (n=chris@unaffiliated/redlined) |
| 10:15.47 | basty | ah cool..thanks again, trixter |
| 10:15.47 | trixter | er guru |
| 10:16.55 | Assid | http://www.voip-info.org/wiki-Bandwidth+consumption |
| 10:17.39 | Assid | look at the NEB |
| 10:17.45 | Assid | thats the actual bandwith used |
| 10:17.54 | Assid | the BR is the bandwith required for the codec itself |
| 10:18.21 | Assid | bitrate even |
| 10:20.14 | *** join/#asterisk hertell (n=Rene@jumbo52.adsl.netsonic.fi) |
| 10:20.14 | __chris | extensions_additional.conf:OUTCID_5 = 0870******* - does this syntax look ok? 1899 seems to be passing my real 01 phone number via CLI, not sure if they restrict sending a different cid ? |
| 10:20.19 | hertell | Hi all! |
| 10:21.22 | droops | hey, im trying to get cdr_mysql to work |
| 10:21.29 | droops | do i need to compile asterisk again |
| 10:21.59 | droops | as i cant seem to find my cdr_addon_mysql.so |
| 10:22.35 | hertell | I wanted to check that is source-version from asterisk.org installed into /usr/local/sbin or just /usr/sbin? |
| 10:22.36 | Assid | hrmm i should move these guys into iax2 trunking |
| 10:23.02 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
| 10:23.24 | webmind | morning, I'm trying to make a call from kphone to a grandstream sip phone.. but I get the following warnings (and calls fail (forbidden)): |
| 10:23.27 | webmind | <PROTECTED> |
| 10:23.32 | webmind | can anyone help me out with this ? |
| 10:23.54 | fishboy1669 | hi RoyK |
| 10:23.55 | RoyK | it's a usual codec problem |
| 10:24.11 | webmind | RoyK, good.. how do I fix it.. or where / |
| 10:24.12 | webmind | ? |
| 10:24.26 | RoyK | webmind: for instance, one part tries to talk g.729 with another which only supports g.711 |
| 10:24.39 | webmind | uhuh, but asterisk is able to translate this right ? |
| 10:24.39 | RoyK | webmind: start with 'disallow=all, allow=alaw' |
| 10:24.55 | RoyK | only between supported codecs, obviously |
| 10:25.06 | RoyK | do a sip debug, pastebin it and ask again |
| 10:25.09 | RoyK | ~pb |
| 10:25.11 | jbot | rumour has it, pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
| 10:25.14 | RoyK | also add your sip config |
| 10:25.33 | webmind | ok |
| 10:25.40 | webmind | I'll try the allow first |
| 10:25.58 | pif | where is that website again with full g729 sources? (from a country that doesn't apply software patents) |
| 10:26.23 | boddy | Assid I have a project.I will install asterix on central with real ip and make connection santral via E1 and client reach this asterix via adsl behind the nat |
| 10:26.39 | boddy | is this possible ? |
| 10:26.41 | webmind | allow=alaw doesn't work..how do you go to debug ? (currently running with -vvvc) |
| 10:28.08 | webmind | err |
| 10:28.17 | webmind | how to you do a sip debug |
| 10:29.27 | webmind | nm :) |
| 10:29.37 | hertell | can asterisk be installed in parallell with eg. the default debian version? |
| 10:30.02 | Assid | boddy: shouldnt have a problem |
| 10:31.10 | boddy | cliet adsl modems has 2 port fxs I am planing this modems register itselft to asterix that is on central via sip |
| 10:32.10 | trixter | hertell: yes, infact if in debian you do 'apt-get install asterisk' it will be installed |
| 10:32.21 | trixter | however its a slightly older than current version of asterisk, but it works well |
| 10:32.23 | hertell | i'm asking this because I want to test the wengo-patch that patches chan_sip.c in * 1.0.9 |
| 10:33.02 | trixter | you can get 1.0.9 and install that if you want |
| 10:33.07 | trixter | I think that is what comes with etch |
| 10:33.20 | trixter | sarge iirc is 1.0.7 but it may have been upgraded by now |
| 10:33.49 | trixter | if you want to compile any parts of asterisk I do not recommend using the debian binaries |
| 10:33.58 | hertell | trixer: i'm running the debian-version of asterisk (1.0.7), but what i wanted to do is test is tne 1.0.9 without having to remove the debian-version |
| 10:34.28 | trixter | becuase they expect stuff to be in different places than the default version expects them and there is a little more wokr to configure it, its easier to just have a lcean box and get either maintained packages or source but not both |
| 10:34.47 | trixter | ahh that way, yeah you can put asterisk anywhere you want, even chroot it |
| 10:34.57 | hertell | usually source-softwares are installed into /usr/local, but it looks to me that asterisk is installed into /usr |
| 10:35.24 | *** join/#asterisk Gunnar (n=gunnar@bkkb-gw.bitcon.no) |
| 10:35.28 | trixter | chroot may be the option that you are looking for although if you dont care about the differences in /etc/asterisk between the two it wont matter much.. modules are the only thing that you should be careful of |
| 10:35.48 | hertell | is it ok to set the install_prefix variable to /usr/local/asterix? |
| 10:35.51 | trixter | since /etc/asterisk/asterisk.conf tells it where to go to load the modules, and you may not want to mix stock with any patched things, then again you might |
| 10:35.59 | trixter | if you want |
| 10:38.15 | hertell | well, that should not be any problem, if i can just start asterisk with eg a parameter that would use /etc/asterisk/asterisk_1.0.9.conf, then it's even easier :-) |
| 10:38.54 | hertell | the other config-files do not probably need any tweaking..? |
| 10:39.40 | Assid | WARNING: /lib/modules/2.6.15.2Feb2006/misc/zaptel.ko needs unknown symbol _read_unlock |
| 10:39.54 | Assid | i just finished re-compiling the kernel.. and thats what i get |
| 10:40.35 | webmind | http://pastebin.com/555708 #sip debug log |
| 10:40.45 | webmind | http://pastebin.com/555710 #sip.conf |
| 10:40.55 | webmind | RoyK, there you go |
| 10:42.13 | webmind | or anyone else that can help me with this error |
| 10:42.16 | tzafrir | hertell, try our debs from http://rapid.dotsrc.org/ |
| 10:42.29 | tzafrir | or get the ISO image from http://xorcom.com/rapid |
| 10:42.51 | hertell | tzafir: do they have the "wengo-patch"? |
| 10:43.26 | tzafrir | hertell, did you get wengophone to build on Sarge? what is that patch? |
| 10:43.36 | tzafrir | no, they don't have that patch |
| 10:44.05 | hertell | tzafir: i run just aserisk on sarge, and on my desktop i run Ubuntu |
| 10:44.16 | viperdude | hi guys |
| 10:44.28 | hertell | i use a sipura spa-3000 for placing calls |
| 10:45.50 | tzafrir | hertell, anyway, you'll find there source debs. THough I'd recommend that you get 1.0.10 if you want to stick with the 1.0 branch for now |
| 10:46.01 | tzafrir | Have those patches been applied into 1.2? |
| 10:48.23 | webmind | problem fixed :) |
| 10:49.44 | RoyK | webmind: ding |
| 10:50.44 | hertell | tzafir: i don't know.. If i get this to work, i'll try to make a patch for the 1.2 version.. |
| 10:51.26 | hertell | the problem is that i have no clue about french, is i can't tell what the guys on the wengo.fr forum are talking about ;-) |
| 10:57.25 | *** join/#asterisk HamYaI (i=HamYai@125.24.9.145) |
| 10:57.53 | HamYaI | has anyone seen coppice today? |
| 10:58.42 | *** join/#asterisk h4mm3r` (n=h4mm3r@85-18-14-10.fastres.net) |
| 11:03.12 | *** join/#asterisk ravenpi (n=chatzill@londonderry-cuda1-68-234-68-160.lndnnh.adelphia.net) |
| 11:03.51 | tzafrir | ~seen coppice |
| 11:03.53 | jbot | coppice <n=chatzill@199.193.17.210.dyn.pacific.net.hk> was last seen on IRC in channel #asterisk, 3d 2h 32m 20s ago, saying: 'but it is the one people complain about the most :-)'. |
| 11:04.21 | hertell | is anyone hooked up to fwd? |
| 11:05.20 | hertell | i just wanted to doublecheck if it's me who has problems with connectiong to the echo-test (613) or is it a fwd-problem |
| 11:08.27 | *** join/#asterisk ful|work (n=fulgas@209.8.233.207) |
| 11:21.19 | Assid | aaargh |
| 11:21.24 | Assid | i still cant get this working |
| 11:21.45 | Assid | am still getting this error kernel: rtc: lost some interrupts at 1024Hz. |
| 11:21.55 | Assid | i even recompiled the kernel WITHOUT smp |
| 11:22.48 | boddy | Assid is there any software to access asterix without using telephone ? I mean ip phone software like skype |
| 11:24.31 | Assid | yes |
| 11:24.42 | Assid | get eyebeam/xlite |
| 11:24.44 | Assid | or something |
| 11:25.51 | Assid | tzafrir: you around? |
| 11:26.06 | *** join/#asterisk benquartier (n=benq@adsl-84-227-165-227.adslplus.ch) |
| 11:31.21 | *** join/#asterisk bozo_the_clown (n=bozo_the@dsl-210-15-201-42.QLD.netspace.net.au) |
| 11:33.24 | Assid | brookshire: yo u there? |
| 11:35.01 | BugKham | does anyone know if it's currently a holiday in the US? |
| 11:35.31 | Assid | why whats there today? |
| 11:35.42 | BugKham | I won a bid in ebay, paid for it for 2 days |
| 11:35.58 | BugKham | but the guy never contacted me back |
| 11:36.14 | Assid | give him a day or so more.. else you try and get in touch |
| 11:36.19 | BugKham | it's also happing to a friend of mine |
| 11:36.51 | *** join/#asterisk NassoR (n=root@step.funcitec.rct-sc.br) |
| 11:36.51 | BugKham | Assid: or Valentine's day is considered a holiday |
| 11:37.00 | Assid | nah |
| 11:37.08 | BugKham | s/happing/happening |
| 11:40.14 | *** join/#asterisk Koshatul (n=evangeli@ip157-65-132.cust.bit.net.au) |
| 11:46.01 | *** join/#asterisk cuco (n=diego@local.xorcom.com) |
| 11:46.07 | znoG | hi. I have a Linksys/Sipura ATA and I find that they all do this little "chirp" (or short ring) after they hang up, and sometimes when the phone is idle. Anyone experienced this? |
| 11:49.08 | *** join/#asterisk X-Gen (n=x-gen@dsl-146-97-08.telkomadsl.co.za) |
| 11:54.15 | *** join/#asterisk bigjb (n=bigjb@195.60.10.114) |
| 12:05.35 | *** join/#asterisk welles (n=welles@222.91.77.14) |
| 12:08.22 | *** part/#asterisk xeet2 (n=xeet3@72.1.157.34) |
| 12:11.02 | *** join/#asterisk wellng (n=welles@222.91.77.14) |
| 12:19.52 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/sustaining/cpm) |
| 12:20.30 | cpm | hrmm |
| 12:20.31 | cpm | http://www.theregister.co.uk/2006/02/14/uma_voip_analysis/ |
| 12:21.33 | *** join/#asterisk Eitch (n=hugo@unaffiliated/eitch) |
| 12:22.00 | *** join/#asterisk kmilitzer (n=km@office-gw.westend.com) |
| 12:26.16 | viperdude | cpm: http://www.theregister.co.uk/2006/02/15/skype_3/ |
| 12:27.17 | dezent | does anyone know a good howto to configure precense and im with asterisk ? |
| 12:28.51 | *** join/#asterisk _deg_ (n=deg@200.163.193.247) |
| 12:29.37 | *** part/#asterisk Bushrat (n=bozo_the@dsl-210-15-201-42.QLD.netspace.net.au) |
| 12:30.49 | *** join/#asterisk hypnox (n=dan@cornelyn.force9.co.uk) |
| 12:31.20 | hypnox | ive just checked out asterisk-addons, but there's no RealTime modules in there anymore, where did they go? |
| 12:35.59 | *** join/#asterisk j3g (n=j3g@200.130.8.1) |
| 12:37.03 | j3g | I am configuring asterisk for use with " iconnecthere" network, i have followed some examples... but when I try to dial with "kphone" or xlite i get 404 not found for numbers outside my network. are there any hints? |
| 12:42.17 | cpm | viperdude: interesting. |
| 12:50.26 | j3g | could anyone help me with this setup ? |
| 12:50.29 | *** join/#asterisk lorinc (n=ang@caracas-3107.adsl.interware.hu) |
| 12:51.45 | *** join/#asterisk pengyong (n=lala@222.185.16.126) |
| 12:52.15 | *** join/#asterisk _deg_ (n=deg@200.163.193.247) |
| 13:01.43 | *** join/#asterisk AlexCTI (n=alex@weston-69.65.86.231.myacc.net) |
| 13:01.45 | *** join/#asterisk DarKnesS_WolF (n=sherif@212.103.170.135) |
| 13:07.34 | De_Mon | j3g don't ask to ask just ask |
| 13:08.01 | j3g | De_Mon, i posted my problem :) |
| 13:08.19 | j3g | i get 404 not found error when trying asterisk with a sip provider |
| 13:09.59 | *** join/#asterisk bweschke (n=bweschke@c-68-36-51-213.hsd1.nj.comcast.net) |
| 13:10.50 | benquartier | Hi. somone use webmeetme with asterisk 1.2.4? |
| 13:10.53 | *** join/#asterisk tdonahue (n=tdonahue@208.51.101.201) |
| 13:13.01 | tronix | j3g: http://www.voip-info.org/wiki/view/Asterisk+How+to+connect+to+IConnectHere |
| 13:13.39 | tronix | j3g: if you followed the steps there but still stuck, please pastebin.com your extensions.conf contents |
| 13:13.51 | j3g | it seems to be trying to call xxxxxxxxxxx@172.27.2.33 (xxx being the phone number) instead of calling out |
| 13:13.55 | *** join/#asterisk SplasPood (n=jwb@pool-68-237-52-176.ny325.east.verizon.net) |
| 13:13.58 | j3g | and my asterisk just returns that it doesn't exist |
| 13:14.10 | j3g | 172.27.2.33 = my asterisk ip |
| 13:14.24 | De_Mon | j3g pastebin your extensions.conf |
| 13:14.29 | j3g | okie |
| 13:15.45 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 13:16.41 | boddy | is there any gui to configure asterix |
| 13:17.26 | tronix | yes. it's Asterisk Management Portal, which is an add-on |
| 13:18.42 | boddy | could I make everything with this tool on asterix |
| 13:18.53 | j3g | De_Mon, the pastebin is at : http://pastebin.com/555854 |
| 13:19.05 | tronix | boddy: the Asterisk setup? pretty much, yes. |
| 13:19.58 | j3g | de_mon: it seems to consider the phone I dialed as local extensions instead of outside number (xxxxxxx@172.27.5.3) |
| 13:20.03 | tronix | j3g: what country are you in? |
| 13:20.08 | j3g | brazil |
| 13:20.14 | j3g | (+55) |
| 13:20.25 | tronix | j3g: ahh ok. you have a dial plan for outbound iconnecthere calls that looks like north american. |
| 13:20.32 | tronix | j3g: you could add this entry: |
| 13:21.02 | j3g | exten => _5561XXXXXXXX,1,SetCallerID(556121063608) 55-61-XXXX are numbers in brazil |
| 13:21.13 | tronix | ah, you're right. hmm. |
| 13:21.27 | j3g | first question, what does the leading _ do ? |
| 13:21.31 | j3g | i couldn't find that in the docs |
| 13:21.50 | tronix | iconnecthere might need 011 prefixed to deliver the brazil calls properly. |
| 13:21.55 | tronix | i'm checking. |
| 13:22.00 | tronix | _ means start of matching a pattern |
| 13:22.14 | j3g | hmm ok.. but it isn't even trying to get to iconnecthere |
| 13:22.29 | j3g | it tries 5561xxxxxxxxx@172.27.2.33 |
| 13:22.39 | tronix | you have a sip.conf entry for iconnecthere? |
| 13:22.44 | j3g | yes |
| 13:22.48 | j3g | sip show peers shows its ok |
| 13:22.54 | j3g | iconnecthere/54 213.137.73.140 255.255.255.255 5060 OK (172 ms) |
| 13:23.01 | tronix | ok good |
| 13:24.08 | *** join/#asterisk CrummyGummy (n=wayne@dsl-145-83-110.telkomadsl.co.za) |
| 13:24.14 | *** join/#asterisk moreece (n=m@196.46.142.23) |
| 13:24.21 | moreece | woot woot |
| 13:24.30 | moreece | ok p33ps |
| 13:24.34 | moreece | having a problem with my new SIP phones |
| 13:24.39 | moreece | strange thing |
| 13:24.51 | moreece | been using Xlite SIP for testing |
| 13:25.14 | tronix | j3g: so you are dialing a 11 digit number? starts with 5561? |
| 13:25.15 | moreece | recently purchased a couple hardphones, they register on the asterisk box but I have one way audio problems |
| 13:25.55 | tronix | j3g: exactly 12 digits, right? |
| 13:26.01 | moreece | ??? |
| 13:26.18 | j3g | no i am trying to just dial 556121063608 for example |
| 13:26.19 | moreece | now I am aware that SIP has issues with NAT ect, however this is just on our local lan |
| 13:26.23 | moreece | any ideas? |
| 13:26.29 | j3g | oh 11 digit :) |
| 13:26.38 | tronix | j3g: no, my typo. :) meant 12. sorry. thinking |
| 13:26.39 | j3g | 12 digit number |
| 13:26.40 | j3g | yes |
| 13:26.47 | j3g | that's what I am trying |
| 13:27.05 | tronix | j3g: on * console, do this: set verbose 3 |
| 13:27.14 | tronix | then pb output from console that appears |
| 13:27.18 | tronix | when you dial the 12 digit 5561 number |
| 13:27.24 | moreece | help? |
| 13:27.39 | tronix | moreece: not familiar with audio issues. sorry. :( |
| 13:27.53 | j3g | tronix i set verbose 9 :) |
| 13:28.00 | moreece | think it has something to do with the RTP or something |
| 13:28.01 | j3g | this is what I get on sip debug |
| 13:28.06 | j3g | ACK sip:556121063608@172.27.2.33 SIP/2.0 |
| 13:29.11 | tronix | j3g: ah i think I see problem |
| 13:29.21 | j3g | Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) |
| 13:29.21 | j3g | Looking for 556121063608 in default |
| 13:29.21 | j3g | Reliably Transmitting (no NAT): |
| 13:29.21 | j3g | SIP/2.0 404 Not Found |
| 13:29.42 | tronix | j3g: in [default] context in extensions.conf, add this line: |
| 13:29.51 | tronix | include => iconnecthere |
| 13:30.01 | tronix | then on * console, do 'reload' |
| 13:30.25 | j3g | i did the oposite, i added " default" to iconnecthere |
| 13:30.29 | tronix | saw. |
| 13:30.30 | j3g | i have no [default] line |
| 13:30.35 | j3g | i think it's default heheh |
| 13:30.41 | j3g | i'll add the include |
| 13:30.45 | tronix | ok |
| 13:33.40 | j3g | tronix, trying |
| 13:33.52 | j3g | says it's ringing |
| 13:34.11 | j3g | it think it worked ;) |
| 13:34.15 | tronix | great |
| 13:34.39 | j3g | Thank you tronix :) |
| 13:34.43 | tronix | you're welcome |
| 13:35.15 | moreece | this is driving me mad? |
| 13:35.21 | moreece | why why why |
| 13:36.50 | tronix | moreece: sounds like you need more info on call setup / progress -- suggest you do 'set verbose 10' on console, etc |
| 13:37.09 | tronix | maybe run tethereal 'host <ip #1> && host <ip #2>' |
| 13:37.20 | tronix | you're basically trying to look for unusual errors |
| 13:37.34 | *** join/#asterisk linville (n=linville@nat-pool-rdu.redhat.com) |
| 13:37.40 | moreece | yeah ur right busy doing that at the moe |
| 13:38.14 | tronix | and maybe also 'sip debug' -- i think xten's sip based |
| 13:38.33 | tronix | since I can't hear, I've never become good at audio-related debugging. ;) |
| 13:38.56 | moreece | lol |
| 13:39.08 | tronix | :-) |
| 13:41.32 | *** join/#asterisk edy2 (n=no@212.247.4.149) |
| 13:41.35 | edy2 | hi |
| 13:41.48 | edy2 | please recommend a provisioning system for IP phones |
| 13:42.16 | *** join/#asterisk jimmy_deanPB (n=jhodapp@indianalifesciences.com) |
| 13:43.49 | tronix | edy2: there's AgileVoice but I think that's commercial and $$$ |
| 13:44.05 | tronix | edy2: for freeware, you probably have to hack together some scripts and related stuff |
| 13:49.48 | *** join/#asterisk Meaty (n=cp_simbu@office.abi.ca) |
| 13:52.23 | edy2 | tronix:Thanks, I am planning to make one |
| 13:52.34 | edy2 | I want to what sort of a graphical user interface would be suitable |
| 13:53.06 | moreece | stoked, got it working. firstly I'm an idiot |
| 13:53.19 | moreece | I had the pclink and the networklink plugged in back 2 front |
| 13:53.22 | moreece | :( |
| 13:53.25 | moreece | lol |
| 13:53.57 | *** join/#asterisk saftsack (n=oliver@p54A7F277.dip.t-dialin.net) |
| 13:55.28 | *** join/#asterisk MattH (n=MattH@63.174.244.174) |
| 13:55.44 | MattH | Hi.. when I'm doing a Dial(SIP/blahblah) is there anyway to force a codec to use? |
| 13:56.10 | dezent | anyone know what could be wrong here.. i can call user@domain.com and other people can call me at anders@anykey.se but i cant call johan@anykey.se for some reason.. is there anything speciall i need to do for calling a user in my own domain ? |
| 13:56.30 | iCEBrkr | yo yo yo |
| 13:56.30 | edy2 | tronix:agilebill seems to be a billing software, i need a provisioning one |
| 13:57.22 | *** join/#asterisk fugitivo (n=ajf@201.255.179.7) |
| 13:58.36 | fugitivo | hello |
| 13:58.43 | iCEBrkr | fugitivo: hi |
| 13:59.57 | *** join/#asterisk gr0mit (n=w10277@206.41.25.138) |
| 14:00.11 | *** join/#asterisk spunz_ (n=spunz@h081217096096.dyn.cm.kabsi.at) |
| 14:02.46 | kmilitzer | Does anyone know the pin assignement of the RJ-45 interface on digium E1 cards? |
| 14:03.33 | *** join/#asterisk freat (n=freat@h-72-244-84-43.chcgilgm.covad.net) |
| 14:04.36 | AlexCTI | kmilitzer, it use pins 1,2 and 4,5 |
| 14:05.04 | *** join/#asterisk jimmy_deanPB (n=jhodapp@indianalifesciences.com) |
| 14:06.37 | AlexCTI | Someone can tell me what means it: Feb 15 03:44:50 WARNING[4287]: pbx.c:2405 __ast_pbx_run: Timeout, but no rule 't' in context 'from-zap' |
| 14:06.45 | iCEBrkr | AlexCTI: 't' = Timeout |
| 14:06.50 | iCEBrkr | errduh |
| 14:06.51 | iCEBrkr | LOL |
| 14:06.59 | *** join/#asterisk SplasPood (n=jwb@pool-68-237-52-176.ny325.east.verizon.net) |
| 14:07.20 | iCEBrkr | AlexCTI: It seems that your IVR menu is timingout and you don't have a 't' extension defined |
| 14:07.21 | fugitivo | AlexCTI: well, i think that the message is enough clear to understand what you need to do |
| 14:07.25 | *** join/#asterisk Dr-Linux (n=Nothing@202.125.141.8) |
| 14:07.34 | Dr-Linux | i have a question |
| 14:07.35 | Dr-Linux | <PROTECTED> |
| 14:07.35 | Dr-Linux | <PROTECTED> |
| 14:07.35 | Dr-Linux | <PROTECTED> |
| 14:07.35 | Dr-Linux | <PROTECTED> |
| 14:07.35 | Dr-Linux | <PROTECTED> |
| 14:07.37 | Dr-Linux | sorry |
| 14:07.45 | iCEBrkr | Dr-Linux: Surrrrrrrrrrrre you are. |
| 14:07.46 | Dr-Linux | i'm really sorry for pasting that |
| 14:07.52 | iCEBrkr | Yeah. Right. :P |
| 14:07.56 | AlexCTI | Ok, thanks i was confuse woth the rule 't' |
| 14:07.59 | Dr-Linux | http://pastebin.com/555925 |
| 14:08.11 | Dr-Linux | i was pasting this pastebin |
| 14:08.14 | iCEBrkr | haha |
| 14:08.14 | fugitivo | does the te110p need a rj45 or rj48? |
| 14:08.49 | iCEBrkr | Comfort noise support incomplete?? |
| 14:08.58 | Dr-Linux | iCEBrkr: yes, |
| 14:09.02 | iCEBrkr | WTF |
| 14:09.02 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 14:09.14 | Katty | hmm. |
| 14:09.28 | Dr-Linux | iCEBrkr: my problem is that when i someone calls on our main asterisk number .. that takes too long |
| 14:10.19 | iCEBrkr | <PROTECTED> |
| 14:10.19 | iCEBrkr | <PROTECTED> |
| 14:10.19 | iCEBrkr | <PROTECTED> |
| 14:10.20 | iCEBrkr | <PROTECTED> |
| 14:10.34 | iCEBrkr | Hrrmm. |
| 14:10.44 | *** join/#asterisk mozartsghost (n=ice@dyn.isogo.co.za) |
| 14:10.59 | Dr-Linux | iCEBrkr: i have 3 number from VOIP provider and all these number pointed to one number .. |
| 14:11.02 | iCEBrkr | Dr-Linux: I'm just as confused as you |
| 14:11.08 | iCEBrkr | :) |
| 14:11.28 | Dr-Linux | so when i dial any of number out of 3 that works very fine and doesn't take long |
| 14:11.43 | Dr-Linux | but if i dial that main number .. that takes too long |
| 14:11.52 | Dr-Linux | all i want is to not take that much long .. |
| 14:11.58 | *** join/#asterisk TheCops (n=henri@206-248-136-187.dsl.teksavvy.com) |
| 14:12.00 | TheCops | Hi |
| 14:12.07 | *** join/#asterisk Meaty (n=cp_simbu@office.abi.ca) |
| 14:12.22 | iCEBrkr | else if (rtpPT.code == AST_RTP_CN) { |
| 14:12.22 | iCEBrkr | <PROTECTED> |
| 14:12.26 | TheCops | When an agent is logged in, there's a way to hang up the phone and not stay on the music on hold during wait call |
| 14:12.30 | *** join/#asterisk [TK]D-Fender (n=joe@toronto-HSE-ppp4122655.sympatico.ca) |
| 14:12.42 | iCEBrkr | Dr-Linux: Seems like some sort of RTP codec selection thing |
| 14:12.58 | iCEBrkr | Dr-Linux: I could be totally off base here-- I'm just weeding through the code |
| 14:13.01 | *** join/#asterisk SupZ (n=icechat5@200-161-148-83.dsl.telesp.net.br) |
| 14:13.02 | *** join/#asterisk rLg (n=restless@202.61.49.31) |
| 14:13.02 | Dr-Linux | iCEBrkr: as you seen in my pastebin. all this happend when call progress is at provider end .. so can't i do anything to reduce the time |
| 14:13.03 | Abydos313 | morning everone |
| 14:13.11 | iCEBrkr | [TK]D-Fender: 'morning |
| 14:13.15 | kmilitzer | AlexCTI: Thanks, but which pin does what (RX, TX, etc) |
| 14:13.16 | iCEBrkr | Abydos313: hey |
| 14:13.42 | Abydos313 | this channel is active most while i sleep..heh |
| 14:13.48 | [TK]D-Fender | iCEBrkr : blarg.... |
| 14:14.04 | Dr-Linux | iCEBrkr: i googled it as well .. but it says asterisk doesn't yet work with comfort noise |
| 14:14.04 | iCEBrkr | [TK]D-Fender: agreed. |
| 14:14.16 | iCEBrkr | Dr-Linux: I don't even know what Comfort Noise is |
| 14:14.23 | mozartsghost | howsit guys. quick thing. I got * running with 2 tdm cards, 8 x FXS. all dialing up out over a sip provider. sip provider only support g729 and g723. when I hook up a sip phone to the * box, and dial out it works 100%, however when I try phone out from on of the tdm channels. i just get the, bleep bleep bleep. |
| 14:14.44 | [TK]D-Fender | iCEBrkr : Comfort noise is what SO's do while faking it :) |
| 14:14.49 | iCEBrkr | LOL |
| 14:14.55 | *** join/#asterisk wellng (n=welles@61.150.12.230) |
| 14:14.57 | iCEBrkr | [TK]D-Fender: In that case, I don't support it either! |
| 14:14.58 | mozartsghost | btw, got 8 g729 liscence working on the box. |
| 14:15.16 | Dr-Linux | [TK]D-Fender: hi |
| 14:15.25 | Katty | mister fender. |
| 14:15.29 | *** join/#asterisk welles (n=welles@61.150.12.230) |
| 14:15.32 | iCEBrkr | mozartsghost: Your PRI's configured? |
| 14:15.39 | Dr-Linux | [TK]D-Fender: how can i reduce the time after call answered |
| 14:15.40 | [TK]D-Fender | Dr-Linux : hi |
| 14:15.44 | Dr-Linux | http://pastebin.com/555925 |
| 14:15.55 | mozartsghost | err, its running analog dude |
| 14:15.59 | mozartsghost | analogue. |
| 14:16.09 | iCEBrkr | mozartsghost: It's morning here.. I've just had my first sip of coffee :P |
| 14:16.15 | mozartsghost | lolol, ok |
| 14:16.18 | mozartsghost | :) |
| 14:16.25 | [TK]D-Fender | Dr-Linux: Reduce WHAT time? |
| 14:16.31 | mozartsghost | shucks, really struggling with this friggin box now. |
| 14:16.37 | mozartsghost | *pulls out some more hair* |
| 14:16.40 | iCEBrkr | mozartsghost: So you're dialing outbound via ZAP, correct? |
| 14:16.45 | iCEBrkr | mozartsghost: And that's when you get the fast busy? |
| 14:17.13 | Dr-Linux | [TK]D-Fender: when someone call it takes to long to start IVR pormpt |
| 14:17.22 | iCEBrkr | [TK]D-Fender: Dr-Linux is claiming that inbound calls are taking a long time to connect when he gets that Comfort noise notice. |
| 14:17.33 | Dr-Linux | iCEBrkr: i have 3 number from VOIP provider and all these number pointed to one number .. |
| 14:17.48 | mozartsghost | well, lets make it simple. ok. I got a normal pstn phone connected to the tdm card, fxs port 1. default context is to dial out over SIP/xxxPINxx#number@my.sip.provider.com |
| 14:17.51 | Dr-Linux | but if i dial that main number .. that takes too long |
| 14:18.03 | [TK]D-Fender | Dr-Linux : pastebin your extensions.con |
| 14:18.04 | Dr-Linux | iCEBrkr: as you seen in my pastebin. all this happend when call progress is at provider end .. so can't i do anything to reduce the time |
| 14:18.05 | iCEBrkr | mozartsghost: ok sounds good |
| 14:18.20 | Dr-Linux | ok |
| 14:18.22 | mozartsghost | however when I dial from sip phone setup in sip.conf over that sip extension to my provider. it works 100% |
| 14:18.24 | *** join/#asterisk zaf (n=tfournet@wsip-68-228-9-79.br.br.cox.net) |
| 14:18.31 | iCEBrkr | mozartsghost: good, good... |
| 14:18.58 | [TK]D-Fender | Dr-Linux : Wait.. you mean it rings to long on Zap before answering? |
| 14:19.05 | mozartsghost | I'm thinking that its not using the correct codec. dunno how I would change the codec that the tdm card uses to dial over sip. I thaught it would use the same codecs configured in sip.conf |
| 14:19.26 | Katty | [TK]D-Fender: way to not say hi ;) |
| 14:19.29 | iCEBrkr | mozartsghost: When you're trying to dial out over your TDM card, you need to use a different technology. |
| 14:19.30 | SplasPood | heh, i wonder how bad running asterisk within VirtualPC would be |
| 14:19.44 | iCEBrkr | SplasPood: they have a Asterisk+VMWare install :P |
| 14:19.47 | Abydos313 | runs great in vmware |
| 14:19.51 | Nivex | SplasPood: don't do it. please. for the love of whatever deity you worship, don't. |
| 14:20.00 | SplasPood | Well just for development purposes |
| 14:20.01 | SplasPood | obviously |
| 14:20.02 | Nivex | iCEBrkr: who is "they" that I might eviscerate them? |
| 14:20.12 | iCEBrkr | Nivex: I dunno, haha, I saw it on the Wiki news |
| 14:20.34 | iCEBrkr | mozartsghost: When you dial via your TDM out of your PSTN line, you have to use ZAP/ instead of SIP/ |
| 14:20.36 | SplasPood | ice: too bad I'm on a mac, so no vmware |
| 14:20.46 | iCEBrkr | SplasPood: No VMWare for mac? really? |
| 14:21.00 | mozartsghost | no no, its an fxs card. I'm dialing over sip. not pstn |
| 14:21.04 | Katty | iDunno: :> |
| 14:21.05 | SplasPood | ice: well it historically being something other than X86 prolly has something to do with it :P |
| 14:21.13 | iCEBrkr | mozartsghost: DUDE.. WTF are you talking about? |
| 14:21.22 | benquartier | Does anybody user app_cbmysql.so with asterisk 1.2.4? |
| 14:21.38 | iCEBrkr | mozartsghost: If you're making PSTN calls, you're dialing via ZAP |
| 14:21.56 | iCEBrkr | iDunno/Katty, you two are making me sick.. |
| 14:22.13 | *** join/#asterisk moprilo (n=jjohn@201.192.107.58) |
| 14:22.16 | iCEBrkr | Oh, here we go. |
| 14:22.21 | [TK]D-Fender | Katty: Soryy I missed that there... had a shitty night... |
| 14:22.27 | iCEBrkr | * Katty bats her eyelashes innocently. |
| 14:22.47 | Dr-Linux | [TK]D-Fender: pastebin is in your pvt |
| 14:23.02 | moprilo | anyone here worked with jitter on a iax trunk?.. i can't seem to make it work |
| 14:25.42 | *** part/#asterisk zaf (n=tfournet@wsip-68-228-9-79.br.br.cox.net) |
| 14:26.02 | mozartsghost | iCEBrkr: {TDM40B} 4 FXS ports, normal telephones plug into that, those are configured to dial out to a SIP provider on my * pabx. |
| 14:26.17 | *** join/#asterisk ful|work (n=fulgas@82.102.2.254) |
| 14:26.23 | mozartsghost | my sip provider supports only g729 & g723 |
| 14:26.46 | mozartsghost | when I connect with a SIP softphone capable of g729 to that same pabx, and dial to the sip provider. it works fine. |
| 14:26.54 | iCEBrkr | mozartsghost: But you keep saying you're dialing out over PSTN |
| 14:27.26 | hensema | is there a simple application to record some audio and then play it back to the user, to easily test incoming channels? |
| 14:27.38 | mozartsghost | :/ sorry, thats what I acctually meant. |
| 14:28.13 | mozartsghost | so any idea why the digium card is not working over the sip provider ? |
| 14:28.21 | *** join/#asterisk zaf (n=tfournet@wsip-68-228-9-79.br.br.cox.net) |
| 14:28.29 | iCEBrkr | mozartsghost: You just said it was working via SIP just fine. |
| 14:28.30 | RoyK | SIP over ISDN? |
| 14:28.31 | RoyK | :) |
| 14:28.54 | mozartsghost | iCEBrkr: sip to sip works fine. when I phone ZAP to SIP it doesn't work |
| 14:29.00 | mozartsghost | I see the call gets initiated fine |
| 14:29.04 | mozartsghost | but the provider just drops the call |
| 14:29.04 | iCEBrkr | mozartsghost: So inbound calls aren't working? |
| 14:29.10 | mozartsghost | when I run sip show channels |
| 14:29.11 | RoyK | methinks sip over rfc1149 is a good idea |
| 14:29.21 | mozartsghost | it shows the codec is unknown... |
| 14:29.28 | iCEBrkr | mozartsghost: inbound calls aren't working? |
| 14:29.28 | mozartsghost | instead of g729 like when I run sip to sip. |
| 14:29.36 | mozartsghost | I don't care about inbound calls. |
| 14:29.39 | iCEBrkr | ok |
| 14:30.04 | iCEBrkr | mozartsghost: Ok, so outbound for now. |
| 14:30.11 | mozartsghost | yehp |
| 14:30.16 | iCEBrkr | mozartsghost: So you pick up your SIP phone and dial a phone number.. |
| 14:30.32 | iCEBrkr | mozartsghost: and you're trying to dial via VoIP or PSTN? |
| 14:30.39 | mozartsghost | VoIP |
| 14:30.50 | iCEBrkr | mozartsghost: ok, I'm starting to understand. |
| 14:30.55 | mozartsghost | :) |
| 14:31.15 | iCEBrkr | mozartsghost: You have your providers register line in sip.conf, I'm assuming? |
| 14:31.53 | *** join/#asterisk argos73 (n=mike@adsl-70-228-108-76.dsl.akrnoh.ameritech.net) |
| 14:33.05 | mozartsghost | well, I'm using this in extensions to dial out on |
| 14:33.09 | mozartsghost | exten => _X.,1,Dial(SIP/xxPINxx#27${EXTEN:1}@196.25.173.91) |
| 14:33.21 | Katty | [TK]D-Fender: what happened? |
| 14:33.33 | mozartsghost | not using username/password. so i dunno how I would use a register command to register to the provider |
| 14:33.59 | mozartsghost | that exact same exten works fine, when I'm dialing from a sip softphone. |
| 14:34.02 | iCEBrkr | mozartsghost: Mind me asking who your provider is? |
| 14:34.30 | mozartsghost | its south african company, called budgetcalls |
| 14:34.51 | iCEBrkr | mozartsghost: never used/heard of them. :( |
| 14:35.04 | iCEBrkr | mozartsghost: Ok, so you're trying to dial a number like say your cellphone? |
| 14:35.13 | mozartsghost | yehp, exactly that |
| 14:35.25 | iCEBrkr | mozartsghost: that Dial() line you pasted won't work. |
| 14:35.39 | mozartsghost | howcome ? |
| 14:35.55 | iCEBrkr | mozartsghost: You need something like _NXXNXXXXXXX,Dial(SIP/ |
| 14:36.21 | mozartsghost | ok, but it recognizes the number fine. |
| 14:36.23 | iCEBrkr | or whatever the pattern is for your area of dialing |
| 14:36.30 | mozartsghost | and it acctaully starts dialing out |
| 14:36.36 | [TK]D-Fender | Katty: Well still "living" with the GF as we are splitting up and well... being Valentines she felt all the pressures of everyone else getting attention and we've been silent. She was hoping I'd be able to be more "available" physically but I found that what I thought I could offer her I really can't.... |
| 14:36.37 | mozartsghost | but as soon as my provider sees the call, it just hangs up |
| 14:36.49 | mozartsghost | because I think its trying to use the wrong codec. |
| 14:36.56 | iCEBrkr | mozartsghost: Well, I still think you need a register line in your sip.conf |
| 14:37.16 | iCEBrkr | mozartsghost: and if it's a problem with codecs, configure the allowed codecs in your sip.conf for each extension |
| 14:37.22 | iCEBrkr | or in the [general] section |
| 14:37.28 | Katty | [TK]D-Fender: meep :< |
| 14:37.28 | [TK]D-Fender | Katty: So its seperate bedrooms for us as of this weekend and things just keep getting more awkward... |
| 14:37.56 | znoG | guys, slightly OT question: I have a Linksys/Sipura ATA and I find that they all do this little "chirp" (or short ring) sometimes after a person hangs up, and sometimes even when the phone is idle. Anyone experienced this? |
| 14:38.20 | mozartsghost | I've done that. [general] disallow=all allow=g729 |
| 14:38.21 | iCEBrkr | [TK]D-Fender: You could be more 'available' physically with the back of your hand telling her to STFU, it's just another day.. |
| 14:38.27 | iCEBrkr | [TK]D-Fender: I'm kidding of course. :-/ |
| 14:38.41 | Katty | [TK]D-Fender: meep :< |
| 14:38.47 | iCEBrkr | [TK]D-Fender: That totally sucks ass. |
| 14:39.27 | iCEBrkr | mozartsghost: do your phones support g729? |
| 14:39.55 | iCEBrkr | [TK]D-Fender: I'm not sure what's worse? I think I got a girlfriend without my knowledge. |
| 14:39.57 | [TK]D-Fender | yeah don't I know it.... I can't be looking for someone else and still be with her. I don't work that way..... |
| 14:40.11 | Dr-Linux | iCEBrkr: [TK]D-Fender helped me as ever and problem is sloved. actually was confused with some zap things |
| 14:40.14 | mozartsghost | its a normal touchtone fone. it doesn't even know what g729 is. the dialplan works, with any media, sip.iax. whateve.r its just when I dial from an fxs port on digium card |
| 14:40.23 | iCEBrkr | Dr-Linux: cool stuff |
| 14:40.31 | mozartsghost | its not using the correct codec |
| 14:40.31 | [TK]D-Fender | ...yay |
| 14:40.33 | iCEBrkr | mozartsghost: oh yeah. duh, forgot |
| 14:40.37 | [TK]D-Fender | kjlskldjfhljkdsjhfkldsgfgfdsfgds |
| 14:40.38 | mozartsghost | ;) |
| 14:40.45 | [TK]D-Fender | At least I'm appreciated somewhere.... |
| 14:40.49 | *** join/#asterisk NassoR (n=root@step.funcitec.rct-sc.br) |
| 14:40.52 | Abydos313 | always |
| 14:40.56 | mozartsghost | I think I need to buy iCEBrkr a 6pack. |
| 14:41.05 | iCEBrkr | [TK]D-Fender: Whatever man, don't let a girl get ya down.. Things didn't work out, it's not your fault. |
| 14:41.06 | mozartsghost | hehe |
| 14:41.12 | iCEBrkr | mozartsghost: That might help! |
| 14:41.28 | *** join/#asterisk _deg_ (n=deg@200.163.193.247) |
| 14:41.40 | iCEBrkr | mozartsghost: I wish they'd let me drink on the job.. It's not like I'd get wasted, I just need a little buzz. |
| 14:41.41 | [TK]D-Fender | iCEBrkr : Well there is lingering finacial requirement for shared apt, and not wanting to strip out my stuff from the place and lave here there cold... |
| 14:41.49 | mozartsghost | hehe |
| 14:41.56 | iCEBrkr | [TK]D-Fender: Ok, THAT sucks. |
| 14:42.21 | Katty | [TK]D-Fender: hope things get better for you... |
| 14:42.25 | iCEBrkr | [TK]D-Fender: Ya live and ya learn. I totally avoid having a live-in GF> |
| 14:42.29 | Dr-Linux | <[TK]D-Fender> kjlskldjfhljkdsjhfkldsgfgfdsfgds ? difficult english :) |
| 14:42.38 | [TK]D-Fender | iCEBrkr: I was going to cut&run 2 weeks ago, but she arrived home early while I was packing and I lost it. We had a big talk and all but the fact is living like this is killing me a little more every day and I'm already empty inside.... |
| 14:42.41 | iCEBrkr | Dr-Linux: I think that was his head bashing the keybaord. |
| 14:43.08 | Dr-Linux | iCEBrkr: yeah i made him temper with my stupid questions |
| 14:43.14 | iCEBrkr | [TK]D-Fender: Hey, at least you can talk and not have a huge argument that the whole neighborhood knows about. ;) |
| 14:43.15 | mozartsghost | eish, well. If I find the problem, i'll let you know what it was. I need to get this box running tho, Cause nextweek i'm doing a dual * setup, with 17 24ports digium cards. and 4 PRI cards. for a hospital. 400 analog extensions, 100 isdn extensions. |
| 14:43.16 | mozartsghost | lol |
| 14:43.17 | mozartsghost | fun fun fun. |
| 14:43.39 | iCEBrkr | mozartsghost: Sounds like you're in over your head. :P |
| 14:43.46 | mozartsghost | lol, yeahp |
| 14:43.57 | iCEBrkr | Nothing like learning on that $40k contract job. |
| 14:44.24 | iDunno | [TK]D-Fender: eep - that sucks :/ |
| 14:44.50 | buZz | gets* |
| 14:44.59 | buZz | becomes* |
| 14:44.59 | iCEBrkr | ha |
| 14:45.08 | iCEBrkr | buZz: Smoke more dude. |
| 14:45.10 | iCEBrkr | :) |
| 14:45.17 | buZz | ;) |
| 14:45.31 | Abydos313 | wake and bake |
| 14:45.43 | *** join/#asterisk brif8 (n=Techno@lazyjtrainingcenter.com) |
| 14:46.21 | *** join/#asterisk heka (n=Horror@82.114.68.124) |
| 14:46.37 | heka | Hello, how are phones connected to a TDM2400P card? |
| 14:47.17 | Dr-Linux | iCEBrkr: one of my user has somehow greetings deleted from voicemail .. where can i verify at server end? |
| 14:47.30 | Nugget | if the tdm400p has an FXS interface, phones plug into it. |
| 14:47.39 | Nugget | if it doesn't have an FXS, then phones are not connected to it |
| 14:47.53 | iCEBrkr | Dr-Linux: /var/spool/asterisk/voicemail/ is a good start |
| 14:48.04 | fugitivo | do i need a rj48 or rj45 for the te110p? |
| 14:48.14 | heka | Im talking about TDM2400, Im not clear about it yet! |
| 14:48.35 | Dr-Linux | iCEBrkr: yeah i know that but thats voicemail messages. i want to find/see HIS greeting messages ? |
| 14:48.46 | iCEBrkr | Dr-Linux: it's in there too |
| 14:49.00 | heka | is there a tdm400p needed to connect with TDM2400 or what? |
| 14:49.16 | iCEBrkr | Dr-Linux: greet.wav |
| 14:49.51 | pb__ | fugitivo: rj48 |
| 14:50.22 | Dr-Linux | iCEBrkr: oo yeah i see that ... but how can go through with this file :S |
| 14:50.29 | fugitivo | pb__: thanks |
| 14:50.30 | Dr-Linux | should i give him only file :S |
| 14:50.30 | iCEBrkr | ?? |
| 14:50.40 | Dr-Linux | or whats decent way to revert back his setting |
| 14:50.56 | iCEBrkr | Dr-Linux: If the greeting is deleted or missing, he'll have to re-record it |
| 14:51.12 | brif8 | Can the older Dev Kit with the TDM400P, does the card handle T1 or just a single phone line? my TDM400P currently has the FXS module, Can I purchase a FXO module that will handle T1 best would be 2 x FXO modules for 2 x T1? |
| 14:51.48 | Dr-Linux | iCEBrkr: yeah mm..... but how it possible he lost the greetings? :S |
| 14:52.44 | iCEBrkr | Dr-Linux: Not sure |
| 14:52.54 | Dr-Linux | iCEBrkr: thats his request |
| 14:53.09 | iCEBrkr | Dr-Linux: I'd have him re-record it.. |
| 14:53.59 | pb__ | brif8: you can't do T1 with a TDM400P; you would need something like a TE110P for that. The TDM400P is POTS only. |
| 14:54.17 | brif8 | pb__: :( thanks anyways |
| 14:54.40 | Dr-Linux | iCEBrkr: fine. but for the next time should i have manage some backup way on daily basis? :) |
| 14:54.44 | tronix | pb__: or T1 -> channel bank -> plug in to TDM400P cards |
| 14:55.21 | iCEBrkr | Dr-Linux: Wouldn't hurt to back it up.. |
| 14:55.29 | iCEBrkr | Dr-Linux: You should be backing up voicemails anyhow :P |
| 14:55.53 | Dr-Linux | iCEBrkr: yeah entire dir |
| 14:56.01 | *** join/#asterisk kpettit (n=keith@69.15.174.114) |
| 15:00.04 | *** join/#asterisk ast_freak (n=ast_frea@68-112-130-237.dhcp.stcl.mn.charter.com) |
| 15:02.44 | Hmmhesays | god I love gmail |
| 15:04.17 | Hmmhesays | hows pakistan today Dr-Linux |
| 15:04.27 | Abydos313 | violent! |
| 15:04.40 | *** join/#asterisk GoRK (n=GoRK@amarillo.energynet.com) |
| 15:04.41 | Abydos313 | 8 more died today from protests |
| 15:04.52 | Hmmhesays | sounds like a fun place |
| 15:04.52 | *** join/#asterisk crich1999 (n=crich@p54BF86E0.dip0.t-ipconnect.de) |
| 15:05.04 | Abydos313 | i wouldn't want to live there |
| 15:05.23 | Hmmhesays | iw ant to live anywhere except here |
| 15:05.28 | Hmmhesays | i hear panama is nice |
| 15:05.42 | Abydos313 | Hmmhesays if you mean the good ol' US i agree! |
| 15:05.43 | *** join/#asterisk shnarff (n=whois@216.190.144.90) |
| 15:05.53 | Hmmhesays | no, nothing wrong with the US |
| 15:06.04 | Hmmhesays | just this frozen hell I call home |
| 15:06.06 | shnarff | hey guys -- does ast realtime come with 1.2.4 or do i need to check out HEAD |
| 15:06.07 | Abydos313 | Hmmhesays where do you live |
| 15:06.07 | shnarff | ? |
| 15:06.13 | GoRK | hello; can anyone help me out with some polycom sip firmware files? I need bootrom 3.1.2 (or newer) and firmware 6.1.3 (or newer) -- my Soundstation ip 4000 is a brick now w/o them .. newest i have is 3.1.0/1.6.2 which is one version back |
| 15:06.20 | Hmmhesays | Fargo ND |
| 15:06.59 | Abydos313 | lots of snow there recently |
| 15:07.59 | shnarff | lots of snow here today |
| 15:08.10 | shnarff | i wanna move to sunny ca |
| 15:08.17 | Abydos313 | it's nice here |
| 15:08.23 | Hmmhesays | there is literally nothing here for me but trouble with the law, debt and heartache |
| 15:08.24 | shnarff | where? |
| 15:08.28 | Abydos313 | california |
| 15:08.33 | Hmmhesays | if it wasn't for booze i'd be dead |
| 15:08.34 | shnarff | wher in ca? |
| 15:08.38 | shnarff | LOL |
| 15:08.38 | Abydos313 | did you mean canada? |
| 15:08.46 | shnarff | no california |
| 15:08.48 | shnarff | <PROTECTED> |
| 15:08.52 | Abydos313 | los angeles |
| 15:08.54 | shnarff | i grew up in la |
| 15:08.59 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 15:09.01 | Hmmhesays | i'm contemplating moving there |
| 15:09.01 | Abydos313 | right new universal studios |
| 15:09.05 | Abydos313 | near |
| 15:09.14 | shnarff | nice |
| 15:09.25 | Abydos313 | crazy traffic though. |
| 15:09.27 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 15:09.27 | *** mode/#asterisk [+o anthm] by ChanServ |
| 15:09.31 | shnarff | i took a bunch of cisco classes somehwere near there |
| 15:09.37 | shnarff | in north hollywood |
| 15:09.44 | Hmmhesays | i need to leave this place |
| 15:09.47 | Abydos313 | 2 miles away from me |
| 15:09.50 | SupZ | I'm having a lot of echo in voip calls, does anybody know any special conf to exterminate it? |
| 15:09.57 | *** join/#asterisk HamYaI (i=HamYai@125.24.0.65) |
| 15:10.01 | saftsack | hi so visdn is working here :) |
| 15:10.02 | shnarff | i would concur crazy traffic --- atleast for this idaho boy |
| 15:10.08 | saftsack | does someone else has visdn here? |
| 15:10.20 | Hmmhesays | i need to find a shack in the middle of the woods |
| 15:10.31 | Hmmhesays | and a large generator to run my guitar amp |
| 15:10.34 | Abydos313 | i work 27 miles from home and i currently drive about 2.5-3hrs a day in traffic :( |
| 15:10.47 | shnarff | lol yeah that i dont miss.... |
| 15:10.54 | shnarff | i live a bloclk from work |
| 15:10.58 | shnarff | and drive less than 5 a day |
| 15:11.04 | Hmmhesays | 7 miles of highway here |
| 15:11.13 | Abydos313 | makes a 40hr week basically the same as 60 since you lose the time anyways |
| 15:11.31 | shnarff | lol y3ea i would look at it the same way |
| 15:11.50 | shnarff | anyoen ever do the realtime tutorial off asteriskguru.com? |
| 15:11.53 | Abydos313 | all that free time in the car got me into talk radio |
| 15:12.05 | shnarff | OMG no! who do you listen to? |
| 15:12.06 | GoRK | one of our guys drives 110 miles (each way) to and from work every day |
| 15:12.23 | shnarff | GoRK: slap him for me will ya? |
| 15:12.27 | Abydos313 | medved,john and ken, savage, elder i like a few diff shows |
| 15:12.28 | shnarff | that crazy |
| 15:12.42 | GoRK | he likes to live out in the middle of nowhere.. he lives in an underground house too |
| 15:12.43 | Abydos313 | that is crazy |
| 15:12.53 | Hmmhesays | Hey katty |
| 15:12.54 | Abydos313 | unibomber? heh |
| 15:12.54 | shnarff | i couldnt imagine |
| 15:12.58 | Katty | (= |
| 15:13.10 | GoRK | haha at least the drive is not long .. 75mph, no traffic |
| 15:13.18 | shnarff | nice! |
| 15:13.21 | *** join/#asterisk rogier (n=rogier@16-65-dsl.ipact.nl) |
| 15:13.40 | Katty | Hmmhesays: ready for tomorrow? |
| 15:13.42 | GoRK | his reasoning is that it's less than half the time a lot of people living in suburbs around a city spend on their daily commute, so why not? still gas sucks |
| 15:14.25 | Hmmhesays | http://maps.google.com/maps?f=q&hl=en&q=+37th+ave+n+fargo+nd+to+1905+hwy+10+E+moorhead+mn&ll=46.90548,-96.765862&spn=0.11165,0.234146 |
| 15:14.29 | Hmmhesays | theres my drive to work |
| 15:14.32 | a1fa | lol |
| 15:14.33 | shnarff | yeah i bet -- hope he doesnt drive a suburban or somethin lol |
| 15:14.35 | Hmmhesays | Katty , nothing I have to do |
| 15:14.43 | Hmmhesays | except wait for the call from the lawyer |
| 15:15.18 | Hmmhesays | we have e85 up here |
| 15:15.22 | synthetiq | awww our first #asterisk couple |
| 15:15.28 | a1fa | where? |
| 15:15.32 | Hmmhesays | i think my next vehicle will be an e85 vehicle |
| 15:15.33 | a1fa | where where where? |
| 15:15.56 | GoRK | why not biodiesel |
| 15:16.02 | Hmmhesays | don't be stupid |
| 15:16.05 | gr0mit | hi - anyone using SS7 on Asterisk? |
| 15:16.21 | Katty | Hmmhesays: oh, you don't have to be there? |
| 15:16.33 | Hmmhesays | Katty: nope, thats I paid a grand for the lawyer |
| 15:16.36 | synthetiq | wahts an e85 mercedes |
| 15:16.37 | Hmmhesays | *thats why |
| 15:17.20 | a1fa | Hmmhesays : why did you need a layer? |
| 15:17.22 | Katty | Hmmhesays: oh. |
| 15:17.25 | a1fa | err.. liar |
| 15:17.39 | Hmmhesays | a1fa: because I got myself into trouble |
| 15:17.40 | Hmmhesays | duh |
| 15:17.50 | a1fa | what did you do, negro? |
| 15:17.59 | a1fa | hehe |
| 15:18.01 | Katty | a1fa: let's refrain from that sort of language. |
| 15:18.02 | a1fa | btw, i is black |
| 15:18.04 | Katty | a1fa: kthx. |
| 15:18.06 | a1fa | :P |
| 15:18.20 | shnarff | LOL |
| 15:18.28 | a1fa | i need a lawyer too |
| 15:18.33 | Katty | a1fa: leave the ghetto in the ghetto (= |
| 15:18.33 | Hmmhesays | shouldn't you be selling a cop some dope? |
| 15:18.47 | a1fa | s0re |
| 15:19.00 | shnarff | ahh |
| 15:19.04 | a1fa | damn dude |
| 15:19.05 | a1fa | its payday |
| 15:19.10 | a1fa | and I allready spent a grand |
| 15:19.22 | a1fa | car payments + furniture payments |
| 15:19.25 | Katty | Hmmhesays: and when does all this Stuff start? |
| 15:19.32 | Katty | Hmmhesays: or will you not know until tomorrow. |
| 15:19.33 | Hmmhesays | black people can buy furniture? |
| 15:19.42 | _Sam-- | they rent it! |
| 15:19.42 | shnarff | lol |
| 15:19.48 | _Sam-- | <PROTECTED> |
| 15:19.51 | Hmmhesays | Katty: i'll know more tomorrow |
| 15:19.54 | a1fa | _Sam-- : hey man! |
| 15:19.58 | _Sam-- | sorry a1f, you know im just playing around. |
| 15:20.07 | Katty | Hmmhesays: okies. i shall bug you tomorrow. |
| 15:20.07 | a1fa | i know man! and i respect you |
| 15:20.10 | _Sam-- | whats up jigga |
| 15:20.12 | a1fa | dont diss me like that :P |
| 15:20.14 | a1fa | not much |
| 15:20.15 | a1fa | working |
| 15:20.15 | Hmmhesays | thats the beauty of teh internets... who the fark cars |
| 15:20.20 | Hmmhesays | *cares even |
| 15:20.30 | Hmmhesays | I will let you know Katty |
| 15:20.37 | a1fa | _Sam-- : i got in scissors today riding my bike to work |
| 15:20.44 | Hmmhesays | i hope it goes well i'm not looking forward to jail |
| 15:20.47 | a1fa | i was passing 50 people |
| 15:20.59 | Katty | Hmmhesays: yeah, and i'm not looking forward to surgery :< |
| 15:21.10 | buZz | i hate the intarweb |
| 15:21.11 | Hmmhesays | You'll be fine, i had mine out a couple years ago |
| 15:21.11 | Katty | Hmmhesays: $500 worth of drugs for a weekend. *sigh* |
| 15:21.12 | a1fa | heheh.. and redlined half way there |
| 15:21.35 | Katty | Hmmhesays: that won't keep me from worrying...and hallucinating, as strong drugs make me do. |
| 15:21.42 | Hmmhesays | that could be fun |
| 15:21.47 | Katty | it's never fun. |
| 15:21.51 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 15:21.54 | Katty | it's always something Bad like spiders and needles |
| 15:21.58 | Hmmhesays | that sucks |
| 15:22.16 | Hmmhesays | i need a new guitar |
| 15:22.18 | Katty | i don't think Sucks(tm) quite covers it. |
| 15:22.25 | Hmmhesays | yeah |
| 15:22.27 | buZz | yeah, trips will always be bad if you are not at easy / comfortable |
| 15:22.30 | Katty | especially when your mind is seeing thousands of spiders crawling down the walls. |
| 15:22.34 | a1fa | _Sam-- : i am gonna go check your inventory for flushmount blinkers and ZX mirrors |
| 15:22.38 | Hmmhesays | unless you're into that |
| 15:22.39 | a1fa | super sport mirrors |
| 15:22.47 | _Sam-- | you wont find either |
| 15:22.54 | a1fa | dang dude |
| 15:23.08 | Hmmhesays | I think we're going to try play "crazy bitch" at jam night |
| 15:23.13 | _Sam-- | but check out our racing gallery from last year: http://www.kneedraggers.com/gallery/ |
| 15:23.19 | Hmmhesays | now that we have a singer that can pull it off |
| 15:23.20 | shnarff | Katty: wisdom tooth pull? |
| 15:23.36 | Katty | shnarff: i wish. |
| 15:23.48 | a1fa | _Sam-- : i really want to be sponsored ;( |
| 15:23.50 | _Sam-- | htats actually just from daytona i think |
| 15:23.50 | Katty | shnarff: they're not even showing yet, much. |
| 15:23.57 | Katty | shnarff: the surgeon is going in after them. |
| 15:24.03 | Hmmhesays | who here has heard the new buckcherry album? |
| 15:24.07 | a1fa | _Sam-- : is you the guy there? |
| 15:24.07 | Katty | shnarff: one is impacting against another tooth. |
| 15:24.16 | _Sam-- | no, thats the main michelin guy in the USA...im in some pics |
| 15:24.20 | Hmmhesays | yeah one of mine was impacted too |
| 15:24.21 | synthetiq | katty finally getting the stomach stapled huh |
| 15:24.25 | _Sam-- | the older, fatter, balder guy = the michelin guy |
| 15:24.25 | synthetiq | heh heh heh |
| 15:24.41 | shnarff | Katty: oh they are growing sideways i bet,, i hear that sucks just had mine out roots gre around jaw and they had to break |
| 15:24.44 | a1fa | so you are not in any of the pictures? |
| 15:24.52 | _Sam-- | i am in some someplace. |
| 15:24.56 | Katty | shnarff: they're not growing sideways. i've already had xrays. |
| 15:25.02 | a1fa | are these r6s? or r1s? |
| 15:25.05 | Katty | synthetiq: i'm never having children, deary. |
| 15:25.08 | Hmmhesays | mine was coming in at about a 45 degree angle |
| 15:25.12 | _Sam-- | i think most of those pictures are R6 |
| 15:25.25 | Hmmhesays | i want a yamaha r1 |
| 15:25.30 | Hmmhesays | rolling death machine |
| 15:25.35 | Hmmhesays | but i'd go out with a bang |
| 15:25.45 | a1fa | damn dude |
| 15:25.48 | Katty | so how does one take oxy-contin without having bad hallucinations? |
| 15:25.48 | a1fa | that guy is low |
| 15:25.59 | a1fa | Katty : stay of pot+alcohol |
| 15:26.02 | _Sam-- | there's a reason he had two number 1 plates from the year before :) |
| 15:26.25 | shnarff | Katty: you do really? from that? |
| 15:26.27 | *** join/#asterisk bronze (n=clark@c-24-91-157-212.hsd1.ma.comcast.net) |
| 15:26.51 | a1fa | yeah man |
| 15:26.53 | *** join/#asterisk adminguru (n=atze@dslb-084-060-168-199.pools.arcor-ip.net) |
| 15:26.54 | Katty | shnarff: anything more than the equivilent of 600mg of ibuprofen. |
| 15:26.57 | a1fa | i am looking that picture |
| 15:26.58 | bronze | Hi all, anyone familiar with as asterisk issue of opening too many files? |
| 15:26.59 | Katty | shnarff: and nyquill too |
| 15:27.04 | a1fa | first picture he and 4 other guys in a turn |
| 15:27.05 | a1fa | 2nd picture |
| 15:27.07 | Hmmhesays | i need a new vice |
| 15:27.09 | buZz | Katty: you trip on ibuprofen? :O |
| 15:27.09 | Hmmhesays | not meth though |
| 15:27.13 | a1fa | he is out of the curve, all str8 |
| 15:27.20 | Katty | buZz: any 'strong' drug makes me hallucinate. |
| 15:27.28 | *** join/#asterisk MikeJ[Laptop] (n=vircuser@d14-69-8-30.try.wideopenwest.com) |
| 15:27.28 | a1fa | while the other 4 are still dragging behind |
| 15:27.38 | _Sam-- | that guy is fast as shit |
| 15:27.41 | buZz | Katty: thats not my question |
| 15:27.46 | _Sam-- | we got on the podium in the AMA last year |
| 15:27.47 | Katty | buZz: then what's your question. |
| 15:27.58 | buZz | the one with the questionmark :P |
| 15:27.59 | buZz | 16:27 < buZz> Katty: you trip on ibuprofen? :O |
| 15:28.14 | Katty | buZz: i think ibuprofen falls under 'any' |
| 15:28.17 | buZz | as, tripping on ibuprofen points to serious mental issues |
| 15:28.20 | a1fa | what gallery software is this.. its very nice |
| 15:28.29 | a1fa | ImmageVau? |
| 15:28.31 | Katty | buZz: you're a doctor eh? |
| 15:28.41 | _Sam-- | one of our web guys loaded it up, i have no idea |
| 15:28.53 | buZz | no , but i know that taking 2800mg of ibuprofen is 'still ok to drive' |
| 15:28.53 | *** join/#asterisk austinnichols101 (n=austinni@70.46.69.130) |
| 15:29.07 | buZz | so if you trip on ~600 |
| 15:29.14 | Katty | buZz: if you're not a doctor, then shut up. not everyone is the same as you. |
| 15:29.18 | tronix | buZz: some people's brains are wired differently. |
| 15:29.18 | buZz | you must have neurological problems with the taking |
| 15:29.28 | buZz | tronix: thats what i am saying :S |
| 15:29.33 | Hmmhesays | Katty: i'm going to send you a song, since it has been awhile |
| 15:29.39 | Katty | Hmmhesays: okies. |
| 15:29.39 | buZz | Katty: if you dont want to talk medical , dont talk medical |
| 15:30.00 | Katty | buZz: hence me telling you to shut up (= |
| 15:30.03 | fugitivo | wtf? |
| 15:30.06 | buZz | i actually did a small neurological course |
| 15:30.06 | Katty | i just love how some males don't listen. |
| 15:30.10 | bronze | I'm sorry, I thought this was the asterisk channel.. ;-) clearly I made a mistake... :) |
| 15:30.10 | shnarff | I dont trip from meds but i get seriously paranoid so i dont take them |
| 15:30.11 | buZz | i'm not male |
| 15:30.19 | Katty | oh, well then that explains it ;) |
| 15:30.31 | buZz | i'm omnisexual |
| 15:30.37 | Katty | k |
| 15:30.47 | Hmmhesays | i love how you kiss, i love how you sound and baby the way you make my world go round |
| 15:31.00 | shnarff | So...... about that realtime architecture... anyone know anything abou tit? |
| 15:31.00 | fugitivo | buZz: neural networks for AI? :) |
| 15:31.01 | Katty | Hmmhesays: which email address are you sending to? |
| 15:31.06 | buZz | fugitivo: no ^_^ |
| 15:31.08 | Hmmhesays | gmail |
| 15:31.09 | *** join/#asterisk slak- (i=slak@rewted.biz) |
| 15:31.09 | Katty | k |
| 15:31.12 | slak- | HEY PLAYAZ |
| 15:31.20 | slak- | k |
| 15:31.24 | Hmmhesays | this guys' voice is seriously kickass |
| 15:31.31 | fugitivo | buZz: male is a gender, omnisexual is a sexual orientation |
| 15:31.41 | fugitivo | or is that bisexual? |
| 15:31.46 | Hmmhesays | being from northdakota i'm into buffalo |
| 15:31.48 | slak- | boss comes up to me and asks if its possible for asterisk to place a call/page based on email |
| 15:31.56 | Hmmhesays | yes it is |
| 15:32.04 | slak- | like we have a buggy webapp system, when theres a problem is shoots us an email |
| 15:32.08 | slak- | and we want asterisk to let us know |
| 15:32.11 | slak- | if its like 3am |
| 15:32.16 | buZz | then i'm omnigender (Y) |
| 15:32.20 | slak- | how Hmmhesays |
| 15:32.41 | shnarff | OMG,, iwas just saying to someone the other day,, you know i have never heard anything going on in ND,, and seriously started to questions its existence.... |
| 15:32.45 | Hmmhesays | have your pos webapp trigger a shell script that makes a call file |
| 15:32.49 | a1fa | _Sam-- |
| 15:32.53 | _Sam-- | ? |
| 15:32.53 | fugitivo | buZz: are you hermaphrodite? |
| 15:32.57 | buZz | nope |
| 15:32.59 | a1fa | just saying hi |
| 15:33.00 | fugitivo | so? |
| 15:33.01 | slak- | Hmmhesays: well it runs remotely mang |
| 15:33.11 | buZz | fugitivo: hence omni, not bi |
| 15:33.12 | buZz | :) |
| 15:33.15 | Hmmhesays | so? |
| 15:33.20 | sevard | What's a good way to make .gsm? dial *77, grab the .wav, use an application to convert it? |
| 15:33.20 | fugitivo | so? |
| 15:33.25 | fugitivo | are you a male or a woman? |
| 15:33.27 | slak- | Hmmhesays: this needs to be done via EMAIL Hmmhesays |
| 15:33.34 | buZz | sevard: i actually record .gsm straight |
| 15:33.36 | slak- | and Hmmhesays , whats a "callfile" |
| 15:33.40 | sevard | slak-: using |
| 15:33.47 | buZz | fugitivo: in this world , male |
| 15:33.51 | Hmmhesays | so you want asterisk to receive and email and trigger a call? |
| 15:33.51 | slak- | sevard: what |
| 15:34.00 | sevard | slak-: what do you use to record straight to gsm |
| 15:34.01 | fugitivo | buZz: great, it's good when you accept what you are |
| 15:34.05 | Hmmhesays | if your web app can send an email it can trigger a script |
| 15:34.11 | *** join/#asterisk mkl1525 (n=daniel@212.80.239.153) |
| 15:34.21 | slak- | im sure it can |
| 15:34.27 | buZz | sevard: ,Record(/tmp/asterisk-recording:gsm) |
| 15:34.38 | slak- | Hmmhesays: what would the script need to do |
| 15:34.38 | austinnichols101 | is there a way to specify the order in which individual trunks are selected. In my case I have 7 trunks on a PRI that are selected in order from 1-7 for outbound calls and would like to change it so that it's 7-1. |
| 15:34.55 | Hmmhesays | either originate the call via the manager or with a callfile |
| 15:35.04 | shnarff | slak-: set your maildir up as the asterisk call dir? ;p |
| 15:35.06 | slak- | what does a callfile look like |
| 15:35.08 | Hmmhesays | google voip-info asterisk callfile |
| 15:35.09 | shnarff | j/k |
| 15:35.10 | buZz | fugitivo: i accept that there is no way i can a 'normal' relationship with any person, animal or vegatable |
| 15:35.11 | slak- | sneak: HAHAHA |
| 15:35.16 | slak- | er shnarff LOL |
| 15:35.17 | buZz | not even platonic |
| 15:35.25 | Hmmhesays | i sure as hell can't have a normal relationship |
| 15:35.27 | shnarff | :D |
| 15:35.35 | Katty | normal's boring anyway, Hmmhesays |
| 15:35.37 | sevard | buZz: so, like exten => 444,1,Answer exten => 444,2,Record(/tmp/asterisk-recording:gsm) exten => 444,3,Hangup |
| 15:35.56 | fugitivo | buZz: none of us can have a normal relationship |
| 15:35.57 | Hmmhesays | lets see my last string of gf's has consisted of a crazy chick who's new bf tried to kill me, a psycho native (yet hot) and an engaged girl |
| 15:36.40 | Hmmhesays | oh what I wouldn't give to be sitting out at niki's ranch right now |
| 15:36.43 | fugitivo | Hmmhesays: i like that type, do you still have her number? |
| 15:36.52 | Hmmhesays | 306-2692 |
| 15:36.57 | fugitivo | where is that? |
| 15:37.06 | Katty | i'm guessing north dakota |
| 15:37.12 | Hmmhesays | that would take all the fun out of it |
| 15:37.24 | synthetiq | Hmmhesays has a bit of an ego problem no |
| 15:37.34 | *** join/#asterisk ManxPower (n=ewieling@dpc6745150107.direcpc.com) |
| 15:37.35 | Hmmhesays | why do you say that synthetiq? |
| 15:37.42 | Katty | synthetiq: you just have to slap him a few times and he's fine. |
| 15:37.53 | Hmmhesays | heh, no I don't have an ego problem |
| 15:38.03 | buZz | sevard: yes |
| 15:38.06 | Hmmhesays | i have chronic "crazy bi@tch" syndrome |
| 15:38.08 | synthetiq | it happens after laying your first few females |
| 15:38.09 | fugitivo | (256) 306-2692 ? |
| 15:38.16 | Katty | Hmmhesays: but you have a lack of hugs problem. |
| 15:38.21 | synthetiq | because the fact is all females are crazy |
| 15:38.26 | Hmmhesays | sure fugitivo, call it up |
| 15:38.34 | Hmmhesays | sent that song Katty |
| 15:38.36 | *** join/#asterisk DarkFlib (n=DarkFlib@cpc4-nfds9-6-0-cust148.leic.cable.ntl.com) |
| 15:38.43 | *** join/#asterisk gmoney_____X (n=ggggg@c-66-176-86-40.hsd1.fl.comcast.net) |
| 15:38.50 | Hmmhesays | i don't have an ego problem, although I feel like a god when i'm playing guitar |
| 15:39.01 | gmoney_____X | whats going on |
| 15:39.05 | shnarff | what kind of guitar? |
| 15:39.15 | synthetiq | why dont we all brag in here |
| 15:39.23 | Hmmhesays | les paul is my stage guitar, i have a couple of strat knock off's |
| 15:39.26 | fugitivo | Hmmhesays: do you know how to play "for the love of god" from steve vai? |
| 15:39.33 | Hmmhesays | haha, yeah .. right |
| 15:39.35 | Katty | Hmmhesays: i'm listening to it. |
| 15:39.45 | fugitivo | Hmmhesays: if not, you can't feel like god |
| 15:39.49 | Katty | Hmmhesays: it's ok, but definately not tilo wolff. |
| 15:40.06 | Hmmhesays | josh todd has an incredible voice, and he has "chaos" tattoo'd across his belly |
| 15:40.10 | shnarff | Weezer > Steve Vai |
| 15:40.20 | gmoney_____X | could someone help me with a quick question? |
| 15:40.21 | Hmmhesays | guns n roses |
| 15:40.43 | Katty | stanley jordan. |
| 15:40.46 | Katty | is..........GOD |
| 15:40.53 | fugitivo | Weezer?? |
| 15:41.03 | Katty | no, stanley jordan. |
| 15:41.03 | shnarff | steve vai -- isnt he a property of the $buttRock object? |
| 15:41.15 | shnarff | lol j/k |
| 15:41.18 | Hmmhesays | fugitivo if you're good enough to rock the crowd, then yes you can feel like a god |
| 15:41.26 | bronze | gmoney_____X: Sorry, No asterisk questions are being answered until the kids get their sex and music chats out of the ay... |
| 15:41.30 | fugitivo | Hmmhesays: that depends on the crowd... |
| 15:41.38 | Hmmhesays | this is a college town |
| 15:41.39 | Katty | http://video.google.com/videoplay?docid=8094685026660127371&q=stanley+jordan <- him. |
| 15:41.39 | fugitivo | i mean |
| 15:41.40 | synthetiq | lol bronze |
| 15:41.45 | fugitivo | what kind of crowd |
| 15:41.50 | synthetiq | this channel needs active moderation =/ |
| 15:41.55 | _Sam-- | that guy is pretty good on a geetar |
| 15:41.59 | Katty | Hmmhesays: learn to play like stanley jordan. i'd swoon. |
| 15:42.01 | _Sam-- | seen him a few times |
| 15:42.07 | Hmmhesays | theres your moderation |
| 15:42.07 | gmoney_____X | thanks les claypool is god |
| 15:42.10 | shnarff | bronze,, not fair i have atleast asked 2 questions and tried to answer one..... just have to roll with the punches... |
| 15:42.16 | Hmmhesays | Katty i'll have to check it out |
| 15:42.24 | _Sam-- | i was always a garcia / trey anastasio fan myself for guitar :) |
| 15:42.32 | Katty | Hmmhesays: you'll like it. |
| 15:42.35 | *** join/#asterisk mhnoyes (n=mhnoyes@user-2ivfli9.dialup.mindspring.com) |
| 15:43.05 | fugitivo | flamenco guitar players are "real" guitar players |
| 15:43.12 | fugitivo | that's hard to play |
| 15:43.17 | shnarff | listen,, strange woman laying in ponds, distributing swords is no basis for a system of government..... |
| 15:43.18 | Hmmhesays | i'm not a musician, i just want to get laid |
| 15:43.41 | _Sam-- | i like that guy ottmar leibert for flamenco style |
| 15:43.48 | Katty | Hmmhesays: are you being a typical male? |
| 15:43.56 | synthetiq | hey |
| 15:43.58 | bronze | Katty: he is. |
| 15:44.00 | synthetiq | go to the off topic chan |
| 15:44.00 | Hmmhesays | Katty: since when have I been anything but? |
| 15:44.23 | *** join/#asterisk lathos42 (n=lathos42@65-42-27-66.dowdingindustries.com) |
| 15:44.28 | Katty | Hmmhesays: well you helped me afterall (= |
| 15:44.39 | Katty | Hmmhesays: and i'm pretty sure it wasn't to get laid. heh |
| 15:44.40 | synthetiq | #asterisk-groupies-OT |
| 15:44.47 | buZz | yeah |
| 15:44.56 | shnarff | so...... about that realtime tutorial on asteriskguru,, anyone know if it wors? |
| 15:44.56 | Hmmhesays | hey synthetiq do you have question? |
| 15:44.58 | buZz | #thisisnotasteriskrelatedatall |
| 15:44.59 | Hmmhesays | Katty: no |
| 15:45.01 | shnarff | works* |
| 15:45.14 | Katty | Hmmhesays: no it wasn't, or no i was wrong? |
| 15:45.21 | Hmmhesays | Katty: no it wasn't |
| 15:45.23 | fugitivo | shnarff: this isn't an asterisk related channel |
| 15:45.27 | Katty | Hmmhesays: see, that's not typical maleish! |
| 15:45.31 | fugitivo | shnarff: we talk about sex and guitars |
| 15:45.34 | fugitivo | shnarff: and monkeys |
| 15:45.36 | Hmmhesays | well.... |
| 15:45.43 | Hmmhesays | don't forget midgets and donkey's |
| 15:45.49 | Katty | and hugs. |
| 15:45.55 | fugitivo | and bikes |
| 15:46.01 | Katty | i like bikes. |
| 15:46.01 | fugitivo | and trains! don't forget the trains |
| 15:46.01 | Hmmhesays | synthetiq: do you have a question? if you don't then seriously get off your high horse |
| 15:46.03 | bronze | anything but asterisk |
| 15:46.13 | Katty | bronze: precisely. |
| 15:46.26 | synthetiq | i dont ask the questions i provide the answers |
| 15:46.27 | fugitivo | Katty: what kind of bikes? |
| 15:46.40 | Katty | fugitivo: shiny ones that purr. |
| 15:46.43 | Hmmhesays | synthetiq: why are you complaining? |
| 15:46.48 | fugitivo | synthetiq: who is better, vai or satriani? |
| 15:46.58 | synthetiq | i am fugitivo |
| 15:47.06 | Hmmhesays | neither |
| 15:47.11 | Hmmhesays | slash |
| 15:47.18 | Hmmhesays | joe perry, stevie ray |
| 15:47.21 | fugitivo | yeah! slash |
| 15:47.24 | sevard | Does everyone record their menus on the phone? Or is it really dumb to do that and I should get in a recording studio, have them record stuff to wav and find out a way to convert it to gsm |
| 15:47.26 | fugitivo | but that's another style |
| 15:47.31 | shnarff | ok so.... hypothetically if a monkey were to put down his guitar and grab 1.2.4 build of asterisk (after hugging an african swallow) and sit down to try and make his realtime config work -- would he find that at the end of the tutorial on asteriskguru it would work? even though he point iaxpeers at the sip table? |
| 15:47.40 | Hmmhesays | sevard: i just grab the voice files off the wiki |
| 15:48.02 | Hmmhesays | shnarff: sounds like something yo ushould try |
| 15:48.25 | fugitivo | shnarff: is he on a train? |
| 15:48.31 | shnarff | yes |
| 15:48.36 | sevard | Hmmhesays: I need custom prompts but alison is expensive plus I have a friend who does books on tapes for a living that will record prompts for me |
| 15:48.40 | shnarff | his bike is 3 cars down |
| 15:48.51 | Hmmhesays | sevard: so what is the problem? |
| 15:48.53 | fugitivo | shnarff: he'll do a mess using realtime |
| 15:48.54 | synthetiq | maye to stay on topic ish we need a trivia bot |
| 15:49.06 | sevard | Hmmhesays: I'm asking if recording all the prompts on the phone is stupid |
| 15:49.07 | fugitivo | shnarff: i'd tell the monkey not to use realtime, but that's my preference |
| 15:49.13 | Hmmhesays | the question is why are you so obsessed with staying on topic? |
| 15:49.16 | fugitivo | sevard: hire a professional |
| 15:49.22 | Hmmhesays | none of us are geting paid to help anyone here |
| 15:49.26 | shnarff | alison is cheap |
| 15:49.42 | Hmmhesays | sevard: not stupid, but it sounds like crap <scottish accent> |
| 15:49.44 | synthetiq | i want a picture of alison |
| 15:49.47 | Katty | never call a woman cheap, shnarff |
| 15:49.48 | a1fa | detach |
| 15:49.49 | a1fa | :P |
| 15:49.50 | shnarff | um i only meant that the voice price is cheap |
| 15:49.52 | shnarff | oi |
| 15:49.56 | Hmmhesays | she's kind of naughty too |
| 15:49.59 | sevard | Hmmhesays: so really the best way to go is find a recording studio |
| 15:50.01 | fugitivo | synthetiq: www.theivrvoice.com, she's not beautiful... |
| 15:50.05 | shnarff | <dies> |
| 15:50.23 | Hmmhesays | sevard: do it on your pc and clean up the audio with soundforge or somehting |
| 15:50.27 | Hmmhesays | *something |
| 15:50.35 | fugitivo | his own pc will add a lot of noise |
| 15:50.37 | Hmmhesays | you don't need to waste your money on a studio |
| 15:50.39 | gmoney_____X | quick question i can dial an ext. with my sip phones. but the server will not listen to any button commands after the voicemail starts |
| 15:50.46 | fugitivo | one hour of recording studio is cheap |
| 15:50.51 | Hmmhesays | youc an clean up audio pretty well with soundforge |
| 15:51.02 | fugitivo | hmm |
| 15:51.04 | Katty | fugitivo: i think she's pretty (= |
| 15:51.22 | sevard | She's not that bad looking, she has a double chin but that's about it |
| 15:51.25 | iDunno | today -> pot. |
| 15:51.40 | Hmmhesays | ok, i think my drunk from last night is subsiding |
| 15:51.48 | *** join/#asterisk Utah_Dave (n=boucha@0-2pool130-130.nas28.salt-lake-city1.ut.us.da.qwest.net) |
| 15:51.49 | Hmmhesays | dollar taps are EVIL |
| 15:51.50 | shnarff | fugitivo: why would you councel the monkey not to use real time --- he really thinks it would make a project he is working on go much faster if he didnt have to parse the file(s) evertime he wanted to make a change |
| 15:51.50 | bronze | sevard: If you can get one, use a USB based microphone. This eliminates a lot of the RFI you get with a direct plug in mike. |
| 15:51.53 | *** join/#asterisk SplasPood (n=jwb@206.252.198.100) |
| 15:51.56 | synthetiq | She looks decent |
| 15:52.00 | Katty | sevard: better than seeing ribs. |
| 15:52.11 | Hmmhesays | what if its like one rib |
| 15:52.17 | sevard | Katty: seeing ribs turns me off :( |
| 15:52.22 | shnarff | ok i want to kow who the girl on the digium site is <3 |
| 15:52.23 | Katty | sevard: yep. |
| 15:52.24 | fugitivo | shnarff: because communications are priority and they can't depend on a database |
| 15:52.26 | Hmmhesays | "wow babe you have a freakishly large rib" |
| 15:52.35 | fugitivo | lol |
| 15:52.51 | shnarff | LOL |
| 15:52.52 | bronze | sevard: We use them for Speech recognition systems |
| 15:52.55 | Hmmhesays | shnarff: i've been wondering htat for year |
| 15:52.56 | synthetiq | eys the girl on digium site |
| 15:52.56 | Hmmhesays | s |
| 15:53.01 | synthetiq | i pictures her as the alison |
| 15:53.16 | Hmmhesays | I picture her singing veruca salt "volcano girls" |
| 15:53.26 | shnarff | if that is the case im in love with alison |
| 15:54.00 | shnarff | well as much as one can be by seeing a hot girl on the top of the digium website in passing |
| 15:54.04 | fugitivo | shnarff: she's not alison |
| 15:54.09 | sevard | bronze: what mics do you use? |
| 15:54.15 | shnarff | <synthetiq> i pictures her as the alison |
| 15:54.19 | sevard | Katty: sorry, I have an awesome girlfriend, :) |
| 15:54.23 | shnarff | <shnarff> if that is the case im in love with alison |
| 15:54.29 | Hmmhesays | i had one of those once |
| 15:54.43 | Hmmhesays | now I play guitar a lot more, and have more money |
| 15:55.03 | Katty | sevard: good for you (= |
| 15:55.08 | bronze | I'm using a cheapo headset from dragon system plugged into a modded USB hub I had made up long time ago. works wuitye well. |
| 15:55.09 | shnarff | you know i used to tell myself that being single wouldnt be so bad..... it is |
| 15:55.09 | fugitivo | Hmmhesays: do you earn money playing guitar? |
| 15:55.11 | shnarff | ;p |
| 15:55.16 | Hmmhesays | some |
| 15:55.16 | sevard | Katty: just the grins, and the v-day, you know. :) |
| 15:55.22 | Hmmhesays | its about 400 a weekend |
| 15:55.30 | *** join/#asterisk FlyboySR22 (n=Richard@searsair-linksys.adnc.com) |
| 15:55.31 | bronze | wuitye*quite |
| 15:55.38 | Hmmhesays | ends up being about 10 hours of work |
| 15:55.47 | synthetiq | he plays in downtown where many wealthy visitors walk buy throwing change into his can |
| 15:55.49 | fugitivo | Hmmhesays: well, that's a nice extra |
| 15:56.02 | bronze | Katty, we noticed. |
| 15:56.05 | Hmmhesays | i don't know how much in free liqour |
| 15:56.05 | sevard | Are there any free tools for linux that convert wav to gsm? I think i might be able to pull some strings and get a recording studio for free. |
| 15:56.08 | Hmmhesays | probably a lot |
| 15:56.15 | fugitivo | sevard: sox |
| 15:56.16 | xachen | yeah |
| 15:56.17 | xachen | there is lots |
| 15:56.21 | xachen | sox is the most popular |
| 15:56.28 | xachen | and convert to gsm, raw is the way these days :p |
| 15:56.32 | fugitivo | sevard: search the wiki for converting audio files for asterisk |
| 15:57.05 | Katty | bronze: (= |
| 15:57.21 | Katty | bronze: believe it or not, i do know a little about asterisk :P |
| 15:57.32 | Katty | bronze: i simply don't ask questions about it in here. |
| 15:57.33 | _MartinCabrera_ | Solved:! How to get answer here |
| 15:57.50 | bronze | Katty: i believe |
| 15:57.52 | shnarff | Katty: where do you ask questions? do they get answered? |
| 15:57.53 | slak- | down with audix! |
| 15:57.53 | shnarff | :D |
| 15:57.58 | sevard | I do have a question, what's so great about GSM |
| 15:58.03 | Hmmhesays | wow a punk song with a waltz beat |
| 15:58.13 | Katty | shnarff: yup, i usually ask people directly though. ones i know that have the answer (= |
| 15:58.15 | slak- | who here likes audix |
| 15:58.27 | Katty | shnarff: like anthm and bkw and twisted and Hmmhesays and file. |
| 15:58.27 | synthetiq | if they posted katty's picture on digium website..... would #asterisk exist? http://jessilane.typepad.com/my_weblog/images/100_1191_1.JPG |
| 15:58.28 | shnarff | Hmmhesays: listening to dead milkmen? |
| 15:58.30 | fugitivo | sevard: size of the file only |
| 15:58.33 | Katty | shnarff: asking a question in here is mostly useless. |
| 15:58.39 | bronze | fugitivo: I found asterisk.org, but can you tell me where the wiki is? |
| 15:58.48 | Hmmhesays | simple plan "addicted" is in a 3/4 beat |
| 15:58.49 | slak- | <PROTECTED> |
| 15:58.50 | ursuspacificus | Hi, All... Dial Plan... CUT() vs. Cut()... What's the diff? |
| 15:58.53 | slak- | is that Katty? |
| 15:58.54 | fugitivo | bronze: www.voip-info.org |
| 15:58.55 | Katty | slak-: no |
| 15:58.58 | synthetiq | yes |
| 15:59.03 | Hmmhesays | one is deprecated i think? |
| 15:59.08 | sevard | fugitivo: there are usually tradeoffs between compression/size |
| 15:59.13 | bronze | fugitivo: tx. |
| 15:59.14 | fugitivo | Katty: i have answers too, who helped you with FOP? |
| 15:59.15 | slak- | Katty: A-S-L pls thx! |
| 15:59.28 | Katty | slak-: heh. |
| 15:59.31 | slak- | what |
| 15:59.35 | shnarff | LOL |
| 15:59.36 | Hmmhesays | any hot transgender people type 23423432413241243 for chat |
| 15:59.36 | Katty | slak-: i'm all asled out, sorry ;) |
| 15:59.46 | *** join/#asterisk bigb (n=bigb@static-70-21-248-201.nwrk.east.verizon.net) |
| 15:59.48 | Katty | slak-: and i certainly don't weigh 200 lbs. |
| 15:59.51 | bigb | Question for you guys |
| 15:59.56 | slak- | Katty: over or undee |
| 15:59.56 | Hmmhesays | say hello |
| 15:59.57 | slak- | r |
| 16:00.03 | Beirdo | jeez |
| 16:00.06 | Katty | slak-: under, by /a lot/ |
| 16:00.08 | slak- | I bet 90% of the guys here are 300+ pounders |
| 16:00.13 | bronze | synthetiq: Thats mean! |
| 16:00.14 | Hmmhesays | I for one am |
| 16:00.16 | Beirdo | asking a woman her weight?! bad. |
| 16:00.27 | Katty | Hmmhesays: you're a scrawny little thing. |
| 16:00.28 | shnarff | <-- not me |
| 16:00.32 | bigb | When joining a meetme, if a user hits "#" after the conference, it hangs up |
| 16:00.35 | fugitivo | slak-: i'm not |
| 16:00.41 | bigb | anyway to stop that from happening? |
| 16:00.41 | Katty | Beirdo: i know :< |
| 16:00.44 | ursuspacificus | Hmmhesays: Error I get says "Cut is deprecated, use CUT"... but... can't find any indication of any syntax difference... is there any? |
| 16:00.46 | Katty | Beirdo: also! yay :> |
| 16:00.46 | Hmmhesays | don't forget incredibly seXAY |
| 16:00.47 | fugitivo | bigb: upgrade |
| 16:00.51 | Beirdo | Morning, Katty :) |
| 16:00.51 | *** part/#asterisk mhnoyes (n=mhnoyes@user-2ivfli9.dialup.mindspring.com) |
| 16:00.51 | synthetiq | the truth is mean? |
| 16:00.52 | sevard | i'm 5'11" 160lbs |
| 16:00.54 | slak- | Katty you're a sexy thing I wanna rub on your titties |
| 16:00.55 | Katty | Beirdo: ewwo (= |
| 16:00.57 | bigb | upgrade what? |
| 16:01.00 | bigb | I'm on *1.2 |
| 16:01.01 | Katty | slak-: good luck. |
| 16:01.08 | fugitivo | bigb: 1.2.? |
| 16:01.09 | Katty | slak-: you'd be more likely to get slapped for that. |
| 16:01.10 | Hmmhesays | pffffftt: slak- is looking for a clawing |
| 16:01.11 | bigb | yup |
| 16:01.17 | Katty | Hmmhesays: indeed. |
| 16:01.18 | iDunno | (apparently 216lbs, though :/) |
| 16:01.19 | fugitivo | bigb: what release? |
| 16:01.24 | synthetiq | i dont sk a woman her weight, rather want pant size she is |
| 16:01.24 | fugitivo | bigb: 1.2.4 ? |
| 16:01.28 | jontow | hmm, save i've got a cisco 7960 on an external IP, and an asterisk machine behind a NAT'd firewall, and i've punched through 5060/udp .. does RTP also need to be redirected inbound? or if i set nat: 1 on the 7960, does that take care of it? |
| 16:01.32 | bigb | Asterisk 1.2.4 built by root |
| 16:01.36 | slak- | pants and bra size pls kthx |
| 16:01.38 | Beirdo | slak-: give up :) |
| 16:01.40 | iDunno | (that can't be right... will recheck that :) |
| 16:01.45 | Hmmhesays | jontow depends on the router |
| 16:01.50 | Katty | Hmmhesays: you? sexy? |
| 16:01.52 | synthetiq | if she says anythign greater than 6, i walk away |
| 16:01.53 | jontow | freebsd 5.4, ipfilter+natd |
| 16:01.55 | Beirdo | you are venturing into a boot to the groin territory |
| 16:01.57 | Hmmhesays | Katty: LOL |
| 16:02.08 | Hmmhesays | I thought you'd get a kick out of that |
| 16:02.08 | Katty | Hmmhesays: now let's be reasonable ;) |
| 16:02.15 | slak- | jontow: its not that difficult |
| 16:02.17 | iDunno | (oh, maybe it is :/) |
| 16:02.18 | Katty | Hmmhesays: you are cute though. |
| 16:02.27 | jontow | slak; i know, i've had it working before -- just don't see what i'm missing this time. |
| 16:02.29 | Katty | Hmmhesays: little too scrawny for my tastes. |
| 16:02.36 | *** join/#asterisk RoyK (n=roy@a217-118-45-74.bluecom.no) |
| 16:02.37 | jontow | i've had it working 20+ times inf act, which is why this is bothering me :/ |
| 16:02.37 | *** join/#asterisk Rhizome (n=Rhizome@tor/session/x-15b7c8d23c260a5f) |
| 16:02.40 | Katty | Hmmhesays: put on some weight before you blow away, boy! |
| 16:02.41 | Hmmhesays | i've actually put on about 10lbs |
| 16:02.47 | Katty | put on some more. |
| 16:02.51 | Hmmhesays | i'm a buck fiddy now |
| 16:03.05 | Beirdo | Hmmhesays: you can have 50lbs of mine if you want... cheap |
| 16:03.05 | bigb | Anything I try to get this to strip the "#", or just ignore it doesn't seem to work |
| 16:03.08 | fugitivo | hell, why you talk about feet and lbs, talk about kg and meters :) |
| 16:03.18 | bigb | tried an ignorepat=>#, did nothing. |
| 16:03.22 | Katty | Hmmhesays: if i can get both my hands around your bicep, you need more muscle. |
| 16:03.30 | jontow | 68kg, roughly, fugitivo :) |
| 16:03.46 | Hmmhesays | I could be bigger yeah, but ohwell |
| 16:03.52 | fugitivo | now i understand what you'r talking about :) |
| 16:04.01 | Katty | Hmmhesays: just get some free weights. |
| 16:04.10 | Hmmhesays | i don't put on any weight easily |
| 16:04.20 | fugitivo | Katty: the size of the muscle depends on ... errr well |
| 16:04.21 | slak- | I want to kiss you where you pooo |
| 16:04.24 | Katty | that's probably because you smoke and have the metabolism of a ferret :P |
| 16:04.28 | ursuspacificus | Is it just me or has this gotten a little off topic? |
| 16:04.38 | Hmmhesays | wtf is with you people and being on topic |
| 16:04.56 | Hmmhesays | are you insane? seriously? this isn't paid support it is an IRC channel |
| 16:04.57 | slak- | man smoking is such shiat! |
| 16:04.58 | Beirdo | topics are such a waste of time :) |
| 16:05.02 | bigb | No ideas? |
| 16:05.08 | fugitivo | ursuspacificus: if you have something ontopic to say, go ahead, if not, shut up |
| 16:05.17 | synthetiq | Intelligence of the room decreases as Hmmhesays and katty speak |
| 16:05.19 | slak- | everyone scroll goatse ascii and smoke blunts! |
| 16:05.26 | Katty | synthetiq: if you don't like it, get out ;) |
| 16:05.42 | ursuspacificus | fugitivo: Hi, All... Dial Plan... CUT() vs. Cut()... What's the diff? |
| 16:05.44 | jontow | ursuspacificus: we've been talking on-topic for months, DON'T STIFLE OUR CREATIVE NATURE!#)*& |
| 16:05.50 | Katty | synthetiq: or you could ask an op to interviene |
| 16:05.57 | Katty | synthetiq: perhaps twisted[asteria] is around! |
| 16:06.07 | fugitivo | ursuspacificus: none, NEXT! |
| 16:06.08 | Hmmhesays | life is way to short to be that serious about an irc channel |
| 16:06.11 | Rhizome | Anyone know why cdr_addon_mysq doesn't compile when I do a make in asterisk-addon-1.2.1? |
| 16:06.21 | jontow | Rhizome: try cdr_odbc |
| 16:06.23 | fugitivo | Rhizome: because mysql is evil |
| 16:06.24 | slak- | ops are night time creatures |
| 16:06.28 | fugitivo | Rhizome: try postgresql |
| 16:06.29 | Rhizome | hehe |
| 16:06.32 | Hmmhesays | Rhizome: missing libmysqlclient? |
| 16:06.36 | shnarff | you never have a right to get mad about something you are not paying for :D |
| 16:06.54 | synthetiq | maybe i should ask an ircop to kline hmmhe and katty isntead |
| 16:07.10 | bronze | shnarff: so if someone rapes you for free.... |
| 16:07.10 | ursuspacificus | fugitivo: Thanks. You're a gem. Hope you manage to get laid soon. |
| 16:07.19 | jontow | i spent 20mins and got cdr_odbc working, having never touched it.. and that was with many mistakes |
| 16:07.21 | Hmmhesays | better throw beirdo and fugitivo in there too |
| 16:07.21 | shnarff | you paid for it... trust me |
| 16:07.26 | twisted[asteria] | synthetiq, if you don't like the channel, don't use it. |
| 16:07.27 | twisted[asteria] | end of story. |
| 16:07.29 | twisted[asteria] | thank you, good night. |
| 16:07.34 | jontow | awful lot of documentation about the matter, in fact. |
| 16:07.34 | bronze | ha ha :-) |
| 16:07.46 | Beirdo | synthetiq: good luck. |
| 16:08.03 | bigb | hmm, please? :) |
| 16:08.13 | Beirdo | hehe |
| 16:08.14 | Rhizome | right, so, use odbc with postgresql instead? lol glad I don't care what people think.. hm :P |
| 16:08.14 | shnarff | LOL |
| 16:08.25 | jontow | Rhizome: use cdr_odbc and WHATEVER YOU WANT ;) |
| 16:08.27 | jontow | thats the point of it |
| 16:08.28 | fugitivo | ursuspacificus: thank you, same to you |
| 16:08.30 | Beirdo | Rhizome: you can use ODBC -> MySQL |
| 16:08.34 | Beirdo | that's how I did it |
| 16:08.37 | Hmmhesays | Rhizome: do you have the mysql client libraries? |
| 16:08.38 | iDunno | use a ms access database ;) |
| 16:09.33 | jontow | if you're more comfortable with mysql, then use that.. if postgresql, .... |
| 16:09.33 | shnarff | Rhizome: no reason not to use postgres |
| 16:09.33 | iDunno | (go on, you know you want to ;) |
| 16:09.33 | jontow | hell, if your company has an MS-SQL server, use the FreeTDS crap and use that! |
| 16:09.33 | Beirdo | or Oracle if you have a license :) |
| 16:09.33 | iDunno | *grin* |
| 16:09.33 | shnarff | ewwe |
| 16:09.33 | jontow | literally, your options become wide open. |
| 16:09.33 | Rhizome | Hmmhesays: thanks, that was the problem :) |
| 16:09.34 | bigb | Alright, how about getting hint working on parked extensions? |
| 16:09.34 | jontow | shnarff: didn't say it was a GOOD idea ;) just said it was possible, given cdr_odbc :D |
| 16:09.34 | slak- | hey you guys are getting on topic again |
| 16:09.34 | shnarff | yeah i know hehe |
| 16:09.34 | Hmmhesays | Rhizome: cool, now paypal me a six pack |
| 16:09.34 | slak- | lets bother Katty for nudes |
| 16:09.34 | fugitivo | no need for oracle |
| 16:09.34 | Hmmhesays | i need to get rid of this hangover |
| 16:09.34 | fugitivo | postgresql alone is ok |
| 16:09.36 | bronze | jontow: is postgresql and ODBC a bad combination? |
| 16:09.36 | Katty | slak-: you won't get them. |
| 16:09.36 | synthetiq | do you want nudes of this? http://jessilane.typepad.com/my_weblog/images/100_1191_1.JPG |
| 16:09.40 | fugitivo | if not, you can get mssql, it's cool too |
| 16:09.47 | *** join/#asterisk Gunnar (n=gunnar@bkkb-gw.bitcon.no) |
| 16:09.51 | *** join/#asterisk angler (n=angler@pdpc/sponsor/digium/angler) |
| 16:09.52 | slak- | Katty: you are so uptight |
| 16:09.53 | Hmmhesays | www.johndenvernude.com |
| 16:10.01 | Rhizome | Hmmhesays: What happend to charity? :D |
| 16:10.02 | jontow | bronze: i've heard postgres+odbc is great..haven't used it, myself. |
| 16:10.02 | twisted[asteria] | slak-, i suggest learning tact |
| 16:10.15 | *** join/#asterisk hikenboot (n=hikenboo@c-24-218-84-234.hsd1.ma.comcast.net) |
| 16:10.16 | slak- | synthetiq: yes shes not bad |
| 16:10.23 | Hmmhesays | Rhizome: lol it went out with my slowly draining checking account |
| 16:10.41 | *** join/#asterisk insomni (n=insomni@x1-6-00-e0-18-6f-34-ad.k455.webspeed.dk) |
| 16:10.47 | slak- | thats what i picture a female in a channel like #asterisk on freenode to look like |
| 16:10.47 | Hmmhesays | gretchen wilson is seriously hot |
| 16:10.52 | bronze | jontow: K, just wondered if there were speed issues. |
| 16:10.52 | fugitivo | jontow: yes, it's true, it's great, wonderful, the best of the best |
| 16:11.01 | jontow | :) |
| 16:11.13 | Beirdo | slak-: go wank on your own time, leave us out of it please. |
| 16:11.24 | slak- | heh |
| 16:11.29 | slak- | im just messin around |
| 16:11.31 | slak- | later dudes |
| 16:13.43 | Beirdo | ahhhh. |
| 16:13.44 | Hmmhesays | so um, who wants to buy me a new EQ? |
| 16:13.52 | Hmmhesays | :D |
| 16:14.24 | *** join/#asterisk Gunnar (n=gunnar@bkkb-gw.bitcon.no) |
| 16:14.58 | Hmmhesays | I still can't decide between a digital EQ or an analog |
| 16:15.11 | *** join/#asterisk frenzy (n=frenzy@196.45.144.40) |
| 16:15.44 | kmilitzer | Am I too dumb or is there really no way to give back the HANGUPCAUSE if a Dial Command fails with e.g. USER_BUSY or UNALLOCATED, etc.? |
| 16:16.00 | [TK]D-Fender | Analog.... Otherwise you'll pile on too many A>D, D<A conversions in your path.... |
| 16:16.13 | Hmmhesays | that would be the only one |
| 16:16.24 | Hmmhesays | A->D->A |
| 16:16.26 | Katty | Hmmhesays: eq? |
| 16:16.29 | Katty | Hmmhesays: an equalizer? |
| 16:16.36 | shnarff | what exactly does make template generate? |
| 16:16.37 | Hmmhesays | buying a new equalizer for the band |
| 16:16.41 | [TK]D-Fender | Hmmhesays : And then tack on your effects loop.... whats that do? |
| 16:17.17 | *** join/#asterisk wunderkin (i=kev@ip68-226-113-228.ph.ph.cox.net) |
| 16:17.20 | Hmmhesays | [TK]D-Fender: yeah you're right even though I rarely use anything but the amps 2 distorted and 1 clean channel |
| 16:17.35 | *** join/#asterisk snowolfe (n=snowolfe@firewall.bayou.com) |
| 16:17.48 | Hmmhesays | but its a solid state amp so theres A->D there |
| 16:17.54 | *** part/#asterisk kmilitzer (n=km@office-gw.westend.com) |
| 16:18.30 | [TK]D-Fender | What kind of amp/speaker setup? |
| 16:18.38 | Juggie | if anyones bored take a look @ http://bugs.digium.com/view.php?id=6491 |
| 16:18.43 | snowolfe | helo... anyone around? |
| 16:18.48 | *** join/#asterisk crash3m (i=crash3m@unaffiliated/crash3m) |
| 16:19.20 | *** join/#asterisk Egonis (n=chultay@CPE000255fa0fde-CM00407b87dc7b.cpe.net.cable.rogers.com) |
| 16:19.50 | snowolfe | anyone here played any with dundi? |
| 16:19.54 | Hmmhesays | [TK]D-Fender: 100watt head plugged into a peavey 4x12 cab, preampout going to the mixer which goes to a 1200watt poweramp that drives two peavey 215's |
| 16:20.05 | Egonis | I am emerging (gentoo) asterisk, zaptel, and zapata -- I get a series of compile errors relating to 'class_device_create' when emerging zaptel, what could be the cause? |
| 16:20.18 | *** join/#asterisk kristinG (n=kristin@gentoo/user/kristinG) |
| 16:20.25 | kristinG | hi! |
| 16:20.25 | [TK]D-Fender | Hmmm... all hard-rock setup.... no effects on top? |
| 16:20.54 | Hmmhesays | sometimes I'll throw a chorus in there |
| 16:21.04 | *** join/#asterisk jsharp (n=jsharp@65.88.255.245) |
| 16:21.04 | kristinG | i have a question for anyone that is lucent tnt savvy :) |
| 16:21.17 | Hmmhesays | i have the effects loop plugged in but rarely use it |
| 16:21.18 | snowolfe | on the sip side or just the tnt side? |
| 16:21.26 | FlyboySR22 | hey everyone |
| 16:21.31 | [TK]D-Fender | Hmmhesays : What kind of stuff do you normally play? |
| 16:21.39 | kristinG | tnt -> asterisk zip t1 card |
| 16:21.51 | kristinG | err |
| 16:21.53 | hikenboot | I have six to be connected for a small business how much would the voice over ip cost with astisk to do this (six seperate lines)???? |
| 16:21.55 | kristinG | via t1 card |
| 16:22.02 | Hmmhesays | lit, greenday, cheap trick, tom petty |
| 16:22.18 | Hmmhesays | some country stuff |
| 16:22.45 | Hmmhesays | I think we're going to try "crazy bitch" by buckcherry at jam night on sunday |
| 16:22.49 | hikenboot | how do i find out this information? |
| 16:22.52 | snowolfe | kristin, hmmm... the only thing i really have in my tnt is a couple of 10 chan t1 cards and about 400 modems... oh and a ds3... trying to think |
| 16:22.53 | kristinG | here is the call path: PSTN -> TNT -> Asterisk (via DS1) |
| 16:23.02 | jsharp | hikenboot: Cost depends on what provider you pick. |
| 16:23.10 | Hmmhesays | it can vary a lot hikenboot |
| 16:23.15 | [TK]D-Fender | Cool.. Lit... I still play "I'm my Worst Enemy" from them..... that goes back... |
| 16:23.16 | hikenboot | is there a list of providers somewhere? |
| 16:23.18 | hikenboot | and prices? |
| 16:23.34 | bronze | hikenboot: voip-info.org |
| 16:23.38 | kristinG | i can route calls out the ds1 via cross over if i send calls to trunk-group 11 |
| 16:23.38 | snowolfe | kristin... look at sharkdata.net and give that guy a call... he can probably help you... he does nothing but lucent and used to be an engineer for them... and thats all he does now is use tnt's for sip |
| 16:23.43 | hikenboot | and is there associate qos? and a list of the quality of service? |
| 16:23.47 | *** join/#asterisk steve___ (n=steve@store-fw.porchlight.ca) |
| 16:23.50 | shnarff | make templates make the asteri odbc templates file? |
| 16:23.56 | Hmmhesays | [TK]D-Fender: yeah that is just a fun song |
| 16:24.10 | kristinG | but i am thinking that i need a call-route to point calls to XXX-XXXX to trunk-group 11 |
| 16:24.35 | Hmmhesays | i'd like to play "basketcase" by greenday but its hard to find a drummer that can actually do it |
| 16:24.50 | Hmmhesays | trey cool is a freakishly good drummer |
| 16:24.51 | *** join/#asterisk supjigatr (n=syslod@152.53.16.10) |
| 16:25.05 | [TK]D-Fender | Hmmhesays : Oh yeah? Try finding one who'll do "Hot For Teacher" :) I miss my days with that band..... |
| 16:25.32 | snowolfe | kritin... what you got on the call-routing so far, what's it not doing? |
| 16:25.47 | kristinG | i have so far: |
| 16:26.38 | Hmmhesays | heh, no doubt, even alex van halen can't play it |
| 16:26.45 | Hmmhesays | there were 3 drummers in that song |
| 16:27.11 | kristinG | index* = { { { shelf-1 slot-16 0 } 0 } 0 } |
| 16:27.11 | kristinG | active = yes |
| 16:27.11 | kristinG | trunk-group = 11 |
| 16:27.11 | kristinG | phone-number = 2754 |
| 16:27.11 | kristinG | preferred-source = { { any-shelf any-slot 0 } 0 } |
| 16:27.12 | kristinG | call-route-type = trunk-call-type |
| 16:27.55 | supjigatr | kristinG: MaxTNT? |
| 16:28.00 | snowolfe | kritin gimme a sec... im pulling mine up |
| 16:28.20 | kristinG | yes |
| 16:28.24 | kristinG | 11.0.3 |
| 16:28.30 | kristinG | supershelf |
| 16:28.42 | Hmmhesays | i think i'd go insane these days without the band |
| 16:28.46 | kristinG | ds3 + madd cards + 8 port t1 |
| 16:28.52 | Hmmhesays | between my women and legal issues.. |
| 16:29.19 | supjigatr | kristinG: If you don't get help lemme know. |
| 16:30.08 | kristinG | i need to proof oc concept this so i can then plug it isinto my keysystem |
| 16:30.55 | kristinG | one ds1 trunk will go to my televantage system, the second trunk to my asterisk via the ds1 caes |
| 16:31.03 | kristinG | card |
| 16:31.06 | kristinG | no sip |
| 16:31.06 | snowolfe | kristin... so basically you have pri in... and you want to route back out another to the ast? or am i confusing myself |
| 16:31.15 | kristinG | yes |
| 16:31.21 | kristinG | i have a ds3 in |
| 16:31.24 | snowolfe | ah ok |
| 16:31.27 | kristinG | with 400 did's |
| 16:31.30 | snowolfe | gotcha |
| 16:31.59 | [TK]D-Fender | Hmmhesays : Yeah... I think going single again I need to pick things up myself..... |
| 16:32.01 | *** join/#asterisk RoyK (n=roy@a217-118-45-74.bluecom.no) |
| 16:32.11 | kristinG | and out of those 400, i have 5 did's that i want routed to the phone system, the rest will be sent to asterisk and openser |
| 16:32.17 | snowolfe | so call comes in... but if they call xxx-xxxx you want to send out the ds1 pri that is cross-connected to the asterisk (digium, sanbgoma, whatever) card |
| 16:32.36 | kristinG | snowolfe, precisely! |
| 16:32.52 | *** join/#asterisk flashnet (i=flashnet@Darkstar.AceShells.com) |
| 16:33.07 | kristinG | i have it working via sip |
| 16:33.10 | snowolfe | yeah ... i played a bit with this trying to re-route some of my isdn stuff... gimme a min... gonna have to run through my ssystem and find where i was testing that at |
| 16:33.32 | kristinG | sip is easy, i just poin it to trunk-group 11 |
| 16:33.58 | Hmmhesays | quintum has a pretty kickass gateway for doing things like that |
| 16:34.00 | kristinG | where i have my t1 configured as network side |
| 16:34.06 | *** join/#asterisk dca[laptop] (n=dca[lapt@sta-208-139-193-162.rockynet.com) |
| 16:34.11 | snowolfe | yeah but theres a way to do something like a "filter" in the call-routing that says "if it matches this" send it somewhere else |
| 16:34.25 | kristinG | i know there is. i forogt where it is though |
| 16:34.49 | kristinG | i did this about 9 months ago and i lost my notes in a coffee mishap :p |
| 16:34.58 | Hmmhesays | i have gateways that accept the a call on the t1, look for a route in whatever sip server i'm using and if it can't find one it passes through to terminate on another t1 |
| 16:35.00 | kristinG | speaking of, one sec, refill |
| 16:35.42 | snowolfe | ah ok-.. i think i remember now |
| 16:35.53 | snowolfe | under call-route... you gho by the shelf-slot |
| 16:36.59 | snowolfe | if you want a certaing shelf to respond to a certain number exclusively (if I remember right) you put the phone number ... or a part of the number in under that shelfs profile under the call-route dir |
| 16:37.59 | snowolfe | lemme see where i put the silly manuals |
| 16:38.40 | *** join/#asterisk trelane` (n=trelane@209.43.90.13) |
| 16:38.55 | *** join/#asterisk zaf (n=tfournet@wsip-68-228-9-79.br.br.cox.net) |
| 16:40.41 | kristinG | like what i already have? |
| 16:41.38 | snowolfe | <PROTECTED> |
| 16:41.43 | kristinG | index* = { { { shelf-1 slot-16 0 } 0 } 0 } |
| 16:41.43 | kristinG | active = yes |
| 16:41.43 | kristinG | trunk-group = 11 |
| 16:41.44 | kristinG | phone-number = 2754 |
| 16:41.44 | kristinG | preferred-source = { { any-shelf any-slot 0 } 0 } |
| 16:41.44 | kristinG | call-route-type = trunk-call-type |
| 16:41.46 | kristinG | cost = 0 |
| 16:43.23 | snowolfe | im wondering if answer-defaults would be overriding it it any way |
| 16:43.41 | kristinG | since my MTC only sends 7 digits for inbount calls, i guess i need to do : phone-number = 3272754 |
| 16:44.28 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 16:45.05 | snowolfe | yeah... if i remember right i think it goes from left to right... so putting just 2574 would be looking for 2574*** instead of ***2574 |
| 16:45.23 | snowolfe | err 2754 |
| 16:45.32 | kristinG | correct |
| 16:45.59 | snowolfe | so like if you wanted the shelf or trunk to catch anything 327.. you would just put 327 |
| 16:46.11 | kristinG | though the asterisk wiki says last for of your did |
| 16:46.18 | kristinG | four |
| 16:46.40 | buZz | are those 'cheapass' grandstream phones any good? |
| 16:46.47 | buZz | that 101 model .. is only ~40USD |
| 16:47.08 | snowolfe | try it with the entire number and see if it works |
| 16:47.24 | kristinG | the trick to getting to work is: set use-trunk-groups = yes |
| 16:47.30 | shnarff | buZz: there was alot of chat here yesterday that seemed to indicate they are not |
| 16:47.31 | kristinG | under the system profile |
| 16:47.35 | nestar | buZz: they're ok, they'd work around the house |
| 16:47.40 | _Sam-- | file : where art thee |
| 16:47.40 | snowolfe | i may have it backwards... been a few months since i tried that with the dsl too |
| 16:47.46 | buZz | hmm k |
| 16:47.46 | nestar | i wouldn't use them in a business enviroment |
| 16:47.53 | buZz | the specs gave me a mixed feeling |
| 16:48.09 | snowolfe | for home we send em linsys pap2's or g-nets |
| 16:48.09 | buZz | do you recommend a grandstream cheap over a SPA-1000 ? |
| 16:48.11 | nestar | they're like the $15 phones you buy at wal-mart.. except this one does SIP |
| 16:48.29 | nestar | probably not. a nice analog phone in a SPA would be nice.. |
| 16:48.33 | kristinG | i like the sipura myself |
| 16:48.37 | nestar | or even a cordless phone |
| 16:48.37 | buZz | i was thinking of grabbing a SPA-1000 on ebay |
| 16:48.45 | buZz | for about the same price as the grandstream |
| 16:48.51 | shnarff | Thomson rules |
| 16:48.54 | buZz | and just add some dirtcheap DECT phone to it |
| 16:48.55 | kristinG | i have 2 at home and they work great |
| 16:49.03 | buZz | kristinG: grandstream? |
| 16:49.04 | snowolfe | yeah we use sipuras for connecting to pre-existing pbx systems... linsys, sipura... same firmware |
| 16:49.09 | kristinG | ni spa1000 |
| 16:49.11 | nestar | i have a couple spa-2000 and a spa-3000, i also have 3 budgetones and a bunch of IP300's and 500's |
| 16:49.44 | znoG | snowolfe: ever experienced a problem with your sipuras/linksys where they send a little "chirp" (or short ring) to the phones attached to it only sometimes after a person hangs up, and sometimes even when the phone is idle? |
| 16:49.55 | snowolfe | yep |
| 16:50.00 | znoG | ahh so it's not just me |
| 16:50.02 | kristinG | only problem with the spa-1000 is that the last firmware still has issues and they have not bothered to fix |
| 16:50.13 | znoG | snowolfe: did you find out how to fix it? |
| 16:50.20 | snowolfe | turn off call waiting if not done already and ... hold on while i pull one up... theres another setting under line |
| 16:50.27 | znoG | ohh you rock |
| 16:50.29 | znoG | call waiting is already off |
| 16:51.21 | snowolfe | under user look for your splash len settings... set them to 0 |
| 16:51.39 | snowolfe | be down at the bottom under ring settings |
| 16:51.40 | supjigatr | Anyone have a good source for getting openser/ser working with * and a maxtnt? |
| 16:51.50 | snowolfe | i think they default to a .5 or something |
| 16:52.28 | znoG | snowolfe: VMWI ring splash length? |
| 16:52.33 | snowolfe | yep |
| 16:52.49 | znoG | i wish sipura documented what each and every setting in the web admin means |
| 16:53.07 | znoG | so would it be doing that because of voicemail waiting? |
| 16:53.15 | snowolfe | i had three users complaining about chirps and even quik rings ... same as your describing... setting splash to 0 fixed for me |
| 16:53.18 | znoG | MWI = message waiting indicator... and the "v" ? |
| 16:53.26 | Qwell | voicemail |
| 16:53.30 | snowolfe | yep... but it seems to be erronous on the voicemail |
| 16:53.39 | znoG | heh, pretty obvious eh Qwell :) |
| 16:54.02 | snowolfe | ive actually been testing one that would rinng like there was voicemail when there wasnt even a voicemail box setup for the account |
| 16:54.04 | znoG | i figured message waiting indicator was a good enough tag which has to be for voicemail |
| 16:54.30 | buZz | so ok |
| 16:54.34 | buZz | i'll go for SPA1000 then |
| 16:54.39 | buZz | or something simular |
| 16:54.39 | *** join/#asterisk }btorch{ (n=kvirc@208.63.19.172) |
| 16:54.41 | znoG | snowolfe: strange |
| 16:54.48 | buZz | i just want the cheapest possible sip hardware |
| 16:54.52 | znoG | snowolfe: ok i'll just set all the splash settings to 0 |
| 16:55.23 | snowolfe | znoG: do that... i'm willing to bet that will fix it... i haven't heard complaints after doing that |
| 16:55.29 | znoG | awesome |
| 16:55.36 | znoG | i've had complaints from each and every user :) |
| 16:55.37 | }btorch{ | hey guys I'm setting up a box now with a siemens PBX ... I just a second PRI card installed and I got some guy over here that gives us support to the SIEMENS box |
| 16:55.44 | znoG | (some 20 extensions on 10 sipuras) |
| 16:55.55 | snowolfe | well i gotta go for now... i'll drop by again later |
| 16:56.01 | znoG | thanks heaps snowolfe |
| 16:56.02 | znoG | :) |
| 16:56.06 | snowolfe | no prob |
| 16:56.12 | kristinG | i would try the spa 1001 |
| 16:56.30 | kristinG | they are not making firmware for the 1000's anymore it would seem |
| 16:56.30 | *** join/#asterisk salviadud (n=ralfalfa@201.137.161.198) |
| 16:56.39 | [TK]D-Fender | SPA1001 = waste... spend the extra 10$ and getthe 2002.... |
| 16:56.41 | }btorch{ | I tried asterisk with my digium card using a T1 crossover with the first PRI card we had in the siemens box and it worked fine as long as I had the zapatel signal = pri_net |
| 16:57.06 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 16:57.07 | }btorch{ | what should I use now ? p |
| 16:57.23 | salviadud | hey, if i want to call another sip phone, but i don't get an answer, should i follow the dial extension with a hangup? |
| 16:57.39 | salviadud | something like |
| 16:57.41 | Pegger | could somewone please take a took at my iax2 debug / extensiosn and help me figure out what is going on http://pastebin.com/556180 |
| 16:57.42 | salviadud | exten => 101,1,Dial(SIP/101) |
| 16:57.42 | salviadud | exten => 101,n,Hangup(SIP/101) |
| 16:57.45 | salviadud | ? |
| 16:58.10 | *** join/#asterisk jcwunder (n=chris@ppp-82-135-64-171.mnet-online.de) |
| 16:58.11 | salviadud | i keep calling it, and its still busy... |
| 16:58.24 | salviadud | freakin' weird |
| 17:00.10 | salviadud | i know its early... |
| 17:00.20 | _Sam-- | justinu : you up yet? |
| 17:00.25 | _Sam-- | i need them gxps asaps |
| 17:00.50 | salviadud | sam, could you answer my question buddy? im a redneck using asterisk, give a boy a chance |
| 17:01.07 | Pegger | any ideas http://pastebin.com/556180 ??? |
| 17:01.29 | gambolputty | After the Dial command, use the DIALSTATUS variable. |
| 17:01.30 | _Sam-- | salviadud: in that case, i dont really think it matters...because that example will just ring extension 101 until the caller hangs up |
| 17:01.40 | _Sam-- | so the hangup there is redundant |
| 17:02.15 | gambolputty | dialstatus is better than 101 |
| 17:02.20 | salviadud | thank you! |
| 17:02.55 | _Sam-- | one step at a time, gamboler. |
| 17:02.59 | _Sam-- | first, he should work a on timeout |
| 17:03.05 | _Sam-- | then worry about dialstatus. |
| 17:03.11 | _Sam-- | in my opinion, anyway |
| 17:03.33 | *** join/#asterisk mut (n=animenod@65.111.201.79) |
| 17:03.41 | gambolputty | timeout within dial command |
| 17:03.42 | *** part/#asterisk gr0mit (n=w10277@206.41.25.138) |
| 17:03.51 | gambolputty | dialstatus for result of dial command |
| 17:04.27 | jsharp | Glorp |
| 17:04.29 | Dr-Linux | question, when caller calls my asterisk via ZAP he listens 2 rings before getting IVR prompt. how can i remove these 2 rings |
| 17:04.38 | Dr-Linux | i want caller direct listen IVR prompt |
| 17:04.59 | mut | why would anyone want a fxs port on their system? |
| 17:05.11 | jsharp | Its going to ring at least once for asterisk to detect it. |
| 17:05.34 | _Sam-- | my pri's used to answer on 0 rings |
| 17:05.42 | _Sam-- | no ringing heard to calling party |
| 17:06.04 | jsharp | Well, yeah, on a digital circuit. |
| 17:06.16 | jsharp | But on an analog circuit... |
| 17:06.16 | _Sam-- | ive had my VOIP calls come in with 0 rings as well |
| 17:06.20 | [TK]D-Fender | Dr-Linux : I told you... its because its waiting to get the CallerID info from the line... you'd have to disable it in Zapata.... |
| 17:06.23 | _Sam-- | he didnt say he had an analog? |
| 17:06.26 | _Sam-- | or did he? |
| 17:07.05 | jsharp | If you've got an analog line, you can set usecallerid=no in zapata.conf and it'll answer on the first ring. |
| 17:08.23 | *** join/#asterisk aaaa (n=lovecoff@client-82-199-203-13.speedy.sellinet.net) |
| 17:08.25 | [TK]D-Fender | _Sam-- : "ZAP" |
| 17:08.28 | aaaa | what should I do when my asterisk doesn't understand when the remote side hangs up? |
| 17:08.31 | Pegger | could some one please help me out with this error http://pastebin.com/556180 |
| 17:08.38 | [TK]D-Fender | PRI's don't "ring" per-se |
| 17:08.39 | _Sam-- | i had zap channels on my pri |
| 17:09.00 | sevard | Say, Can anyone suggest a free sound editing appliction? perhaps windows and linux |
| 17:09.19 | [TK]D-Fender | Pegger : The error explains itself.. that exten does not exist in that context. |
| 17:09.28 | [TK]D-Fender | sevard : Audacity |
| 17:09.57 | Pegger | [TK]D-Fender, I saw that but as you can see right above it I have the extension |
| 17:10.05 | aaaa | ok, what about me? |
| 17:10.08 | jsharp | Whats the @ at the end of the extension? |
| 17:11.09 | Pegger | jsharp, not sure about the @ |
| 17:11.23 | jsharp | pastebin the actual contents of your extensions.conf? |
| 17:11.24 | [TK]D-Fender | Pegger : pastebin your extensions.conf |
| 17:11.25 | [TK]D-Fender | ~pb |
| 17:11.27 | jbot | rumour has it, pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
| 17:11.33 | sevard | [TK]D-Fender: Thank you. |
| 17:11.41 | [TK]D-Fender | I'm pretty sure there's an invalid "@" in there tooo |
| 17:12.02 | Pegger | http://pastebin.com/556194 |
| 17:13.12 | jsharp | remove the @ from your extension. |
| 17:13.15 | jsharp | There's your problem. |
| 17:13.35 | [TK]D-Fender | YUP.. I knew it :) |
| 17:13.55 | *** join/#asterisk bweschke (n=bweschke@c-68-36-51-213.hsd1.nj.comcast.net) |
| 17:14.00 | Pegger | oha the @ I put in for testing purposes it makes no diffrence |
| 17:14.55 | [TK]D-Fender | YES it makes a difference..... |
| 17:15.33 | tzafrir | I'm trying to figure this with dialplan varibles expansion: I have the following in extensions.conf: |
| 17:15.47 | tzafrir | exten,1,System(for i in `seq ${EXTEN:5}` ; do ${AST_CMD}; done) |
| 17:16.20 | tzafrir | in 'show dialplan' I see: System(for i in `seq ${EXTEN:5}`(System(for i in `seq ${EXTEN:5}`) |
| 17:16.34 | Pegger | still same error http://pastebin.com/556204 |
| 17:16.56 | tzafrir | I replaced "System" with "NoOp" and got a similar error (with s/System/NoOp/) |
| 17:17.01 | jsharp | Did you reload after making extensions.conf changes? |
| 17:17.10 | Pegger | jaiger, yup' |
| 17:17.15 | aaaa | what should I do when my asterisk doesn't understand when the remote side hangs up? |
| 17:17.16 | [TK]D-Fender | No authority found <----- |
| 17:17.27 | [TK]D-Fender | Bad ID setup |
| 17:17.34 | jsharp | aaaa: Analog lines? |
| 17:17.37 | tzafrir | jsharp, yes |
| 17:17.38 | Pegger | [TK]D-Fender, yaha what does that mean, it is not in extensions |
| 17:17.43 | tzafrir | BTW: this is asterisk 1.0.10 |
| 17:17.55 | [TK]D-Fender | Pegger : You user/pass setup is bad in iax.conf |
| 17:18.05 | aaaa | jsharp yes |
| 17:18.34 | jsharp | aaaa: You have signalling set to fxs_ks or fxs_ls? |
| 17:18.37 | Pegger | [TK]D-Fender, humm if my user/pass is wrong they why would I be able to connect to my voip provider |
| 17:18.51 | Pegger | [TK]D-Fender, this is my DID tryign to call me |
| 17:18.55 | aaaa | i am not sure but i think that it is not kewlstart |
| 17:19.02 | [TK]D-Fender | Pegger : your INBOUND settings are bad... |
| 17:19.12 | jsharp | Gotta have kewlstart to use disconnect supervision. |
| 17:19.30 | jsharp | Pegger: Do you have "default" as the context in your iax.conf for your provider entry? |
| 17:19.36 | aaaa | well i know, but i don't have it, if i have it i would ask here;) |
| 17:19.41 | aaaa | what should i do in this case? |
| 17:19.55 | jsharp | Set your signalling to fxs_ks in zapata.conf and zaptel.conf |
| 17:19.57 | *** join/#asterisk Trevor_b (i=[EtudKbM@bolt.sonic.net) |
| 17:20.04 | jsharp | That *should* fix it, but it may not. |
| 17:20.41 | Pegger | oha wow the default in iax.conf might be it |
| 17:20.55 | tzafrir | hmmm, the problem seems to lie with ";" |
| 17:21.06 | aaaa | well it certainly doesn't fix it |
| 17:21.36 | tzafrir | Removing it makes the expansion issue go away. But I can't just ditch it here. Any idea for a smart for loop without ; ? |
| 17:21.47 | jsharp | Your telco line may not be sending disconnect supervision. |
| 17:21.59 | tzafrir | (any loop, actually) |
| 17:26.16 | *** join/#asterisk _Thor (i=CS@user-vc8fl7l.biz.mindspring.com) |
| 17:26.26 | _Thor | hello everyone |
| 17:27.33 | FlyboySR22 | Hey |
| 17:28.00 | _Thor | I have a question for gurus |
| 17:29.36 | jsharp | Shoot. |
| 17:30.01 | tdonahue | anyone know where to set the DTMF mode on the polycom 501's? |
| 17:31.12 | tdonahue | i currently have it set to rfc2833, but asterisk does not seem to be receiving the dtmf presses |
| 17:31.17 | _Thor | I have a customer who always has all kind of problems with his calls, calls are all IP based, meaning with the server off-site. How can I analize his internet to find out the reason the quality of his calls is so bad? |
| 17:31.49 | *** join/#asterisk generalhan (i=general_@ip67-90-64-2.z64-90-67.customer.algx.net) |
| 17:32.22 | _Thor | meaning, the calls always drop, etc, etc. Any suggestions on tools I can use to point at the reason the call drops, for example |
| 17:32.50 | tdonahue | _Thor: Ethereal has some good statistical analysis for SIP and RTP, as well as finding common problems like packets arriving out of sequence. |
| 17:33.44 | _Thor | how can I point to the actual reason that makes the call drop? |
| 17:34.23 | jsharp | Lots of debugging on Asterisk. |
| 17:34.56 | _Thor | how can I for example, say the call drop for this reason, and in fact, correct it?. This is of course provided that the reason is within the network equipment of the customer |
| 17:35.16 | jsharp | See why Asterisk drops the call. See if its getting a call disconnect message from the far end or if Asterisk is dropping the call because of packet timeouts. |
| 17:36.07 | _Thor | as far as I can see, I only see hangup on the cli, how can I find out the reason that asterisk drops a call? |
| 17:36.24 | jsharp | Depends on what protocol you're running. |
| 17:36.28 | _Thor | sip |
| 17:36.36 | jsharp | sip debug peer <peername> |
| 17:37.03 | jsharp | You'll get a dump of the SIP session for that peer and you can analyze it & see why the call dropped. |
| 17:38.13 | _Thor | for example, if it was dropped because of packet timeouts, what will be the message on the sip debug, packet timeout? |
| 17:38.54 | justinu | you talking about dropped signalling packets, or RTP? |
| 17:39.12 | jsharp | I'm not sure of the actual error message. |
| 17:39.13 | justinu | if the dropped packet is a provisional response, there is no retry with asterisk |
| 17:39.20 | _Thor | justinu: that's what I want to find out |
| 17:39.32 | justinu | otherwise you'll see a message that indicates something like "maximum number of retries exceeded" |
| 17:40.19 | _Sam-- | justinu: can i buy dem gxps? |
| 17:40.21 | _Sam-- | i have a need right away |
| 17:40.30 | _Sam-- | if not, its all good, but i really need some |
| 17:40.43 | _Thor | suppose it is because of packet timeouts, then what will solve it for the client?, is it a matter of increasing the bandwitch? |
| 17:41.22 | justinu | _Sam--: i'll have an answer for you today, hopefully |
| 17:41.26 | justinu | what are you paying? |
| 17:41.55 | _Sam-- | if you still have the boxes, id pay 70ish each? |
| 17:42.04 | justinu | k, i still have the boxes |
| 17:42.06 | _Sam-- | but i mean, i dont have time to wait around, i need some shipping today. |
| 17:42.17 | justinu | understood |
| 17:42.19 | *** join/#asterisk harryvv (n=none@S010600a0c93f6f7e.vs.shawcable.net) |
| 17:42.57 | _Sam-- | thanks, hope i didnt sound like a jerk, but i got new employees that need phones :) |
| 17:43.15 | justinu | nope |
| 17:43.59 | *** join/#asterisk nagl (n=nagl@vie-086-059-104-148.dsl.sil.at) |
| 17:44.07 | [av]bani | _Sam-- you jerk! |
| 17:44.31 | _Sam-- | lol , i cant help it, it comes naturally :) |
| 17:44.53 | *** join/#asterisk littlejohn (n=little@213-140-22-71.fastres.net) |
| 17:45.21 | _Sam-- | [av]bani : did you know manxpower was an ardent teliax supporter? |
| 17:45.28 | _Sam-- | i had no idea |
| 17:45.54 | _Sam-- | [av]bani: you still using junction at all? |
| 17:52.46 | ManxPower | All ITSPs suck. Teliax seems to (usually) suck less than most. |
| 17:53.04 | [av]bani | _Sam--: yes |
| 17:53.11 | _Sam-- | im not sure if they suck less than most, though |
| 17:53.23 | [av]bani | ManxPower: i have unsolvable stutter issues with teliax that i don't have with junction, though junction are 2x farther |
| 17:53.23 | ManxPower | Nufone has good service, but does not have all the services I need. |
| 17:53.25 | _Sam-- | [av]bani: you got junction for orignation? |
| 17:53.32 | [av]bani | sevard: nope,termination only |
| 17:53.38 | [av]bani | _Sam--: nope, termination only |
| 17:53.41 | ManxPower | [av]bani, the Teliax stutter started for me only recently. |
| 17:53.43 | [av]bani | damn irssi autocomplete |
| 17:53.45 | _Sam-- | you should try asterlink for termination |
| 17:53.47 | [av]bani | ManxPower: me too |
| 17:53.56 | _Sam-- | they are cheaper than junction, and quality has been good for me |
| 17:53.56 | [av]bani | _Sam--: junction is great for termination |
| 17:54.03 | [av]bani | _Sam--: they also let me abuse CID |
| 17:54.11 | _Sam-- | same at *link |
| 17:54.12 | ManxPower | [av]bani, talk to Darwin25 (or is it Darwin35) when he's online. He works for Teliax. |
| 17:54.28 | _Sam-- | im only 10ms to asterlink, so i guess im partial |
| 17:54.35 | [av]bani | ManxPower: i've already talked with teliax guys, no solution (and my billing is still screwed too) |
| 17:54.36 | _Sam-- | but its been working really good for us here. |
| 17:55.01 | *** join/#asterisk chops (n=moise@207-172-221-143.c3-0.smr-ubr2.sbo-smr.ma.cable.rcn.com) |
| 17:55.30 | _Sam-- | i have a friend connect to my * from colorado (same place as teliax is)....he is 120ms away (my friend), and his calls sounded better than my calls through teliax at 60ms |
| 17:55.32 | ManxPower | [av]bani, I get stutter on PSTN<->Teliax<->PSTN calls, i.e. my server is not registered to Teliax so the call goes to my failover number. |
| 17:55.40 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 17:55.43 | salviadud | what do i need to dial to get his extension exten => _*314XX.,1,GoTo(fwddialout,${EXTEN:4},1) ??? |
| 17:55.57 | salviadud | i meant, this, not his |
| 17:56.36 | salviadud | i dial 314 and nothing happens... i got this extension under default context |
| 17:56.42 | ManxPower | salviadud, Dial *314 + any number of any digit, then wait for DigitTimeout (because you have a . in your pattern) |
| 17:56.46 | _Sam-- | i LIKE teliax...they are good guys, they treated me brilliantly.... |
| 17:56.56 | _Sam-- | but the service isnt as good for my location as some others. |
| 17:57.34 | Skumling | anyone experienced that HP OfficeJet faxes can't send to spandsp's faxreceieve? spandsp pickups up but the officejet never connects |
| 17:57.49 | salviadud | interesting... |
| 17:58.03 | Skumling | when spandsp picks up, it starts with sending a short time of ugly noise out to the fax |
| 17:58.18 | salviadud | damn sipura!!! |
| 17:58.27 | salviadud | i can't dial an * |
| 17:58.38 | salviadud | if i take it out? does it matter? |
| 17:58.39 | _Sam-- | file where are you |
| 17:58.47 | Nivex | salviadud: I think you have to turn off IP dialling and/or modify your dialplan on the sipura. |
| 17:59.03 | salviadud | thank you nivex... i will check that out now |
| 18:05.41 | *** join/#asterisk hertell (n=Rene@jumbo52.adsl.netsonic.fi) |
| 18:05.58 | hertell | good evening everyone :-) |
| 18:10.24 | *** join/#asterisk synthetiq (n=roger@64.201.13.50) |
| 18:10.40 | ManxPower | [root@fs-1 asterisk]# ftp ftp.digium.com |
| 18:10.41 | ManxPower | Connected to ftp.digium.com. |
| 18:10.41 | ManxPower | 421 Service not available, remote server has closed connection |
| 18:10.41 | ManxPower | ftp> |
| 18:11.11 | [TK]D-Fender | :O |
| 18:11.35 | Assid | heya |
| 18:11.47 | Abydos313 | hey guys, i'm opening 5060 udp on firewall and remote softclient won't connect. |
| 18:12.06 | Abydos313 | using sip connection |
| 18:12.08 | ManxPower | Abydos313, try also opening up the ports for audio |
| 18:12.25 | puppet | abydos313: and be sure to configure the softclient |
| 18:12.46 | Abydos313 | i only open 5060 on firewall and machine and it connects fine when i vpn in |
| 18:12.55 | *** join/#asterisk pifiu (n=myassisb@208.205.181.170) |
| 18:13.03 | puppet | you need more then 5060... |
| 18:13.08 | Assid | can sommeone please help me with this: kernel: rtc: lost some interrupts at 1024Hz. |
| 18:13.10 | *** join/#asterisk docelm0 (n=docelmo@66.239.192.34.ptr.us.xo.net) |
| 18:13.11 | Abydos313 | kinda going thru two firewalls.. a sonicwall and an xp one cause server is running in vmware |
| 18:13.23 | puppet | >< |
| 18:13.24 | salviadud | does somebody have a spa-841 |
| 18:13.26 | salviadud | ? |
| 18:13.28 | Abydos313 | hehe |
| 18:13.29 | docelm0 | Say can anyone reccomend a POE injector for the GXP-2000's? |
| 18:13.42 | salviadud | im going nutz with this thing |
| 18:13.47 | docelm0 | Not looking for switch.. Just onesy or twosie.. |
| 18:13.54 | Abydos313 | can you tell me what ports exactly? |
| 18:14.11 | *** part/#asterisk chops (n=moise@207-172-221-143.c3-0.smr-ubr2.sbo-smr.ma.cable.rcn.com) |
| 18:14.12 | ManxPower | docelm0, voipsupply.com has some |
| 18:14.23 | Assid | ManxPower: any clue on that issue |
| 18:14.28 | *** join/#asterisk jijgeh (i=jijgeh@0-2pool130-130.nas28.salt-lake-city1.ut.us.da.qwest.net) |
| 18:14.42 | ManxPower | Assid, If I could, I would have said something |
| 18:15.04 | Assid | im starting to lose it.. i dont know which side is up anymor |
| 18:15.06 | puppet | abydos313: check voip-info says in the wiki there somewhere dont remember wher ei found it |
| 18:15.12 | Abydos313 | ok |
| 18:15.14 | Abydos313 | thx |
| 18:18.57 | *** join/#asterisk kuku5 (i=kuku@c-67-175-218-223.hsd1.il.comcast.net) |
| 18:19.18 | kuku5 | Looking for a company that will do quality origination in the us - 5-10k minutes |
| 18:19.58 | justinu | _Sam--: i'm afraid I can't get ahold of the customer, so you may want to order phones now |
| 18:20.48 | *** join/#asterisk shido6 (n=shido6@d221-68-216.commercial.cgocable.net) |
| 18:24.45 | hertell | does anyone know why my sound does not transfer out from my *? I can hear the incomming voice, but my voice is not transferred anywhere outside asterisk... |
| 18:25.29 | puppet | hertell: network issues? |
| 18:25.41 | hertell | puppet: should not be... |
| 18:25.53 | puppet | hertell: are you behind nat? |
| 18:25.56 | puppet | hertell: or firewall? |
| 18:26.00 | hertell | puppet: yep |
| 18:26.06 | puppet | hertell: then i say network issues |
| 18:26.08 | puppet | to 99% |
| 18:26.22 | hertell | puppet: i have specified nat=yes |
| 18:26.27 | puppet | ... |
| 18:26.30 | hertell | also the outbound ip |
| 18:26.32 | puppet | it doesnt work cause you change one line |
| 18:26.32 | hertell | etc |
| 18:26.55 | hertell | puppet: i used this howto:_ http://www.voip-info.org/tiki-index.php?page=Asterisk+FWD+NAT+Config+Example |
| 18:27.16 | Juggie | Corydon, are you alive? |
| 18:27.16 | puppet | im behind nat, and i dont use nat=yes ;P |
| 18:27.20 | puppet | it all depends how you configure the net |
| 18:27.34 | Juggie | haha |
| 18:27.51 | Juggie | i just saw an incomming email pop up |
| 18:27.51 | Juggie | and without thinking i hit reply and wrote something quick |
| 18:27.51 | Dr-Linux | [TK]D-Fender: around? |
| 18:27.58 | Juggie | turns out i relpied to the mantis bug tracker |
| 18:27.58 | Juggie | nice |
| 18:28.09 | puppet | juggie: gj :X |
| 18:28.35 | Juggie | can anyone think of a dialplan application that uses dialplan variables |
| 18:28.36 | hertell | puppet: :-) |
| 18:28.38 | Juggie | eg it reads them for settings |
| 18:28.42 | Juggie | something thats simple. |
| 18:28.46 | Juggie | trying to get a code example |
| 18:29.15 | puppet | juggie: redialer, redial last number that called u |
| 18:29.25 | puppet | juggie: redialer, redials last number u called |
| 18:29.59 | hertell | puppet: i get this error: Had to drop call because I couldn't make SIP/625662-c477 compatible with SIP/phone1-e6a3 |
| 18:30.11 | hertell | i guess it is something with wrong codecs..? |
| 18:30.20 | hertell | but how do i check that? |
| 18:30.46 | puppet | http://lists.digium.com/pipermail/asterisk-users/2004-August/058287.html < second hit on google |
| 18:30.53 | puppet | http://www.asteriskguru.com/tutorials/xlite_softphone.html < 4th hit on google |
| 18:31.08 | puppet | etc etc etc |
| 18:31.18 | Assid | somethings wrong.. zaptel doesnt like me |
| 18:31.56 | puppet | hertell: i may be "cranky" but please, do some searching before you ask after help |
| 18:33.17 | [av]bani | it seems all ITSPs suck |
| 18:33.44 | iCEBrkr | [av]bani: I blame the internet. :P |
| 18:33.50 | [av]bani | i blame the intarweb |
| 18:33.55 | *** join/#asterisk newmember[laptop (n=newmembe@S0106000d88b06ac4.cg.shawcable.net) |
| 18:34.00 | [av]bani | and those computarmachines |
| 18:34.03 | iCEBrkr | [av]bani: you can't guarentee something when traffic traverses 4 different networks. |
| 18:34.16 | HamYaI | what version of * is the most stable? |
| 18:34.17 | [av]bani | well, if ITSPs would sell private channels |
| 18:34.24 | [av]bani | of course, i dont think that would fix anything with some ITSPs |
| 18:34.29 | [av]bani | who just suck period |
| 18:34.39 | iCEBrkr | [av]bani: and in the professional VoIP world, they have a 2hop rule.. |
| 18:36.10 | *** join/#asterisk gongoputch (n=gongoput@c-68-82-194-31.hsd1.de.comcast.net) |
| 18:37.30 | [TK]D-Fender | here |
| 18:38.17 | *** join/#asterisk jpablo (n=jpablo@200.94.130.194) |
| 18:38.20 | *** join/#asterisk newmember[laptop (n=newmembe@S0106000d88b06ac4.cg.shawcable.net) |
| 18:38.50 | [av]bani | iCEBrkr: sounds rubbish to me |
| 18:39.43 | iCEBrkr | [av]bani: I'm only repeating what I've heard from friends in the biz. |
| 18:39.45 | *** join/#asterisk techie (i=gus@antibala.com) |
| 18:40.17 | hertell | puppet: it was a codec problm.. i removed a few allow = g723 rows :-) |
| 18:41.33 | *** join/#asterisk Russ598 (n=Spook@213-162-110-13.russel759.adsl.metronet.co.uk) |
| 18:42.05 | *** part/#asterisk Russ598 (n=Spook@213-162-110-13.russel759.adsl.metronet.co.uk) |
| 18:43.13 | [av]bani | iCEBrkr: then i can only conclude the biz is braindamaged |
| 18:43.36 | iCEBrkr | [av]bani: You're the one bitching about how VoIP providers suck. |
| 18:43.40 | iCEBrkr | and I"m telling you why they suck |
| 18:43.44 | _Sam-- | [av]bani: wanna start one? |
| 18:43.55 | _Sam-- | i have some good co-lo ideas for where we could go |
| 18:43.56 | _Sam-- | :) |
| 18:44.53 | _Sam-- | however i dont like the future prospects of trying to make a living at 1.5c/min competing aginst the rbocs/clecs/cable co.s sometime soon |
| 18:45.45 | salviadud | how do i make a spa 841 wait for a freakin' digit timeout? |
| 18:45.52 | salviadud | i think its near impossible... |
| 18:46.05 | Dr-Linux | [TK]D-Fender: if i disbale the callerid option from zapata.conf then callerid facility won't work anymore? |
| 18:46.33 | Dr-Linux | it will not work with my own phone lines or not even with provider's line? |
| 18:46.39 | salviadud | i really can't complain though... im at work and chatting on irc, sure beats the call center... |
| 18:48.28 | harryvv | Sam, I agree. plus the fact that whosale voip is never as reliable as a local telco |
| 18:49.46 | _Sam-- | MikeJ[Laptop] : you around? |
| 18:50.02 | _Sam-- | i have an asterlink question, and no file to help |
| 18:50.46 | Skumling | is there other fax reception programs for use with spandsp than the 'shipping' rxfax? |
| 18:51.03 | salviadud | ok, here's a real question, what the floobangs is this? |
| 18:51.05 | salviadud | Feb 14 12:12:44 NOTICE[11450]: app_dial.c:1011 dial_exec_full: Unable to create channel of type 'IAX2' (cause 3 - No route to destination) |
| 18:51.05 | salviadud | <PROTECTED> |
| 18:51.05 | salviadud | <PROTECTED> |
| 18:51.06 | salviadud | <PROTECTED> |
| 18:51.57 | hikenboot | by using astrisk can I get rid of vonage? |
| 18:52.27 | Hmmhesays | http://www.aculab.com/products/prosody_x.htm <-- anyone seen/used one of those before? |
| 18:53.07 | Himeko | hikenboot sure, but you still have to pay someone for service |
| 18:53.08 | *** join/#asterisk Tagor (n=Tagor@s55928c6d.adsl.wanadoo.nl) |
| 18:53.13 | Tagor | Hi |
| 18:53.40 | hikenboot | ok ...can expect to get better qos for same or cheaper price? |
| 18:54.16 | Hmmhesays | depends on what you are paying vonage |
| 18:54.21 | hikenboot | im in massachusetts |
| 18:54.30 | Hmmhesays | good for you |
| 18:54.31 | justinu | a masshole? |
| 18:54.36 | Hmmhesays | depends on what you are paying vonage |
| 18:54.46 | hikenboot | 29.99 a month right now whenever i download somthing my voip phone drops off |
| 18:54.59 | Hmmhesays | thats not vonages problem |
| 18:55.06 | justinu | that's a qos issue on your side |
| 18:55.06 | Hmmhesays | your modem buffer is probably filling up |
| 18:55.21 | hikenboot | i cant make qos adjustements on the qos for the phone which sucks..beyound the single adjustement it allows from the service |
| 18:55.45 | Hmmhesays | if you're using p2p to download, yes that is going to kill any voip connection |
| 18:56.12 | hikenboot | yes i was using it to download astrisk for example and the connection dropped off |
| 18:56.16 | _Sam-- | hikenboot : they make simple VOIP prioritizers |
| 18:56.22 | _Sam-- | they are like 75 bucks, and will work for you |
| 18:56.27 | _Sam-- | for residential type setups |
| 18:56.38 | _Sam-- | i tested a dlink one at my house, it worked perfectly |
| 18:56.38 | Tagor | I still have a problem with asterisk. If I try to make a call I don't hear anything. This is also internally. When I look in x-lite I only see the microphone volume bar moving. The speaker volume bar seems to do nothing. I also tried IAX. But that also doesn't work. Anyone an idea how to fix this? |
| 18:56.42 | *** join/#asterisk Skarmeth (n=Skarmeth@201009039176.user.veloxzone.com.br) |
| 18:56.52 | *** join/#asterisk Russ598 (n=Spook@213-162-110-13.russel759.adsl.metronet.co.uk) |
| 18:56.56 | Hmmhesays | i don't like x-lite |
| 18:57.15 | _Sam-- | Hmmhesays : if you find an SIP client you like better for free, let me know..im looking. |
| 18:57.21 | Skarmeth | hi all |
| 18:57.28 | hikenboot | thanks sam....also i was thinking of selling voip install services to my networking customers...small to medium sized businesses but i worry about qos for them |
| 18:57.46 | _Sam-- | hikenboot : the linksys wrt54g router does QoS for small biz fine |
| 18:57.51 | _Sam-- | i use it for a real estate office |
| 18:58.04 | Tagor | Anyone an idea? |
| 18:58.20 | hikenboot | how many people use it at once and what internet connection you using and speed of connection? |
| 18:58.35 | _Sam-- | i have 15 people at a real estate office using commercial cable modem service |
| 18:58.41 | Russ598 | Hi, I wonder if anyone can help me. I have an issue with Asterisk. On incoming calls the system seems to strip the leading 0 off on the caller ID, not a problem, but on outgoing calls it seems to add a 0 in automatically somwhere, so if I want to dial a normal UK number I need to omit the leading 0 |
| 18:58.43 | _Sam-- | they browse web, make calls, etc etc etc |
| 18:58.44 | hikenboot | _Sam you running the linux software on that router? |
| 18:58.47 | Skarmeth | I'm about to buy 30 Polycom SoundPoint IP 301 SIP phones... does someone here use this hardware? what's your comments about it? |
| 18:58.58 | _Sam-- | no...i run off the rack linksys new firmware from last month on it. |
| 18:59.49 | hikenboot | are you running that with vonage or other provider? if so which one? |
| 19:00.19 | Hmmhesays | its not really the provider most are pretty good |
| 19:00.27 | _Sam-- | they use asterlink mostly, though i have 3 or 4 backup providers they've also used (teliax, sixtel, broadvoice and one other) |
| 19:00.32 | Hmmhesays | i use voipjet and sixtel for outgoing |
| 19:00.45 | Hmmhesays | sixtel for incoming is cheap |
| 19:01.03 | _Sam-- | some people here warned about the sixtel sound qual |
| 19:01.05 | harryvv | im pissed...I need a RELIABLE voip carrier. this is the second time in two days it has dropped my connection. |
| 19:01.07 | hikenboot | ah so you have different inbound and outbound? |
| 19:01.10 | Hmmhesays | i don't have any problems |
| 19:01.25 | _Sam-- | interesting..thanks. |
| 19:01.25 | Hmmhesays | called least cost routing hiken |
| 19:01.25 | _Sam-- | hikenboot: mostly, no. |
| 19:01.25 | _Sam-- | but you could. |
| 19:01.27 | harryvv | need a good reliable voip carrier in the states. any one recomend? |
| 19:01.28 | Hmmhesays | I do |
| 19:01.34 | Hmmhesays | asterlink |
| 19:01.36 | Katty | you do? |
| 19:01.52 | Hmmhesays | use different carriers for incoming and outgoing |
| 19:01.56 | hikenboot | I wonder about the possibility of using multiple outbound internet connections lets say one cable one dsl??? |
| 19:02.11 | Hmmhesays | depending on where the customers calls most and such |
| 19:02.14 | hikenboot | I imagine i would need to run fatpipe or somthing |
| 19:02.19 | harryvv | Hmmhesays i have asterlink but dont use them as my current outgoing carrier. |
| 19:02.19 | Katty | i think that went over some people's heads. |
| 19:02.30 | [av]bani | Skarmeth: ip301... shame about the lcd? |
| 19:02.40 | harryvv | Hmm, how many outgoing calls have you made though asterlink? |
| 19:02.43 | Hmmhesays | ugh multiple net connections = nightmare |
| 19:02.53 | [av]bani | http://www.anandtech.com/IT/showdoc.aspx?i=2694 <- $1.3 *billion* in 2005 sales O_O |
| 19:03.03 | Hmmhesays | its not like you could have failover during a call with it |
| 19:03.03 | _Sam-- | ive made thousands of minutes through asterlink, minimum |
| 19:03.04 | [av]bani | thats like the economy of a small country |
| 19:03.11 | [av]bani | People's Republic of Newegg |
| 19:03.19 | harryvv | sam, any dropped calls ? |
| 19:03.34 | _Sam-- | they have had a cuople problems that were quickly resolved. |
| 19:03.42 | _Sam-- | most of the problems have been inet / my end. |
| 19:03.49 | _Sam-- | though they did have one or two on their end as well. |
| 19:03.50 | harryvv | okay but the issues are rather infrequent then? |
| 19:03.55 | hikenboot | fatpipe takes care of internet connection but i wonder if it would have service level failover for voip |
| 19:04.03 | hikenboot | not call level but service |
| 19:04.14 | [av]bani | whats weird is teliax is fine for origination... no stutter at all |
| 19:04.15 | _Sam-- | if you get a t1 you have SLA |
| 19:04.17 | [av]bani | only termination stutters |
| 19:04.38 | [av]bani | hmm.. might use JN for termination and teliax for origination |
| 19:04.45 | [av]bani | since JN doesnt have local DIDs |
| 19:04.48 | _Sam-- | i still have a couple number originate from teliax |
| 19:04.53 | _Sam-- | for just taht reason... |
| 19:05.00 | _Sam-- | cant find another local DID originator |
| 19:05.18 | _Sam-- | i guess i could try sixtel |
| 19:05.41 | hikenboot | t1's are expensive..been 3 years since i purchased one though...then there is dedicated vs switching |
| 19:05.51 | harryvv | teliax is reliable ? |
| 19:06.22 | _Sam-- | i dont think its matter of one being more reliable than another really...the critical factor is the path / route your data takes to get to each one. |
| 19:06.31 | _Sam-- | buecase most problems are not the itsp problem |
| 19:06.37 | _Sam-- | but rather a routing problem between you and them |
| 19:06.49 | harryvv | sam, well you can forget that if thay drop your calls to somone your talking to. |
| 19:06.51 | [av]bani | http://www.anandtech.com/IT/showdoc.aspx?i=2694&p=5 <- ahhh the $2.99 shipping explained :D |
| 19:07.17 | _Sam-- | i dont have many complaints about dropped calls from any of the providers ive used |
| 19:07.25 | *** join/#asterisk Utah_Dave (n=boucha@0-2pool130-130.nas28.salt-lake-city1.ut.us.da.qwest.net) |
| 19:07.32 | _Sam-- | 90% of your problems will be routing related |
| 19:07.38 | _Sam-- | a router along the way is assing up |
| 19:07.42 | _Sam-- | dropping packets, etc |
| 19:08.03 | _Sam-- | maybe even 95% of the problems or more |
| 19:08.05 | hikenboot | you mean the originating router or somewhere in the cloud |
| 19:08.09 | harryvv | yea need someone who has a better rep. I may get a hold of somone though asterlink and supstitute my outgoing from sixtel to asterlink. |
| 19:08.14 | _Sam-- | anyr router between you and your ITSP |
| 19:08.15 | Hmmhesays | somewhere in the cloud |
| 19:08.18 | _Sam-- | in either direction |
| 19:08.34 | Hmmhesays | asterlink doesn't do international terminations do they? |
| 19:08.40 | [av]bani | _Sam--: well, why would termination stutter and origination not... same path |
| 19:08.44 | hikenboot | yeah thats what i fear most thats what you can do the least about |
| 19:08.47 | harryvv | problem is, asterlink does not sell DIDs in canada. |
| 19:08.53 | _Sam-- | [av]bani: i said MOST...95% of the problems |
| 19:08.54 | _Sam-- | not all |
| 19:08.58 | [av]bani | lies |
| 19:09.01 | _Sam-- | i said most for a reason :) |
| 19:09.23 | _Sam-- | based on my remote gateway experiences here, which has been plenty... |
| 19:09.29 | _Sam-- | my problems almost always are routing related. |
| 19:09.34 | _Sam-- | "almost always" |
| 19:09.36 | Russ598 | Hi, is anyone able to help me with an Asterisk configuration problem? |
| 19:09.47 | Hmmhesays | routing related or modem buffer related |
| 19:09.51 | docelm0 | spit it out and we will see |
| 19:09.53 | hikenboot | and those are the ones you can do the least about...right? |
| 19:10.04 | _Sam-- | depends on where the issues are happeneing. |
| 19:10.15 | Russ598 | On incoming calls my system seems to strip the leading 0 off on the caller ID, not a problem, but on outgoing calls it seems to add a 0 in automatically somwhere, so if I want to dial a normal UK number I need to omit the leading 0 |
| 19:10.25 | _Sam-- | last week i emailed a few isps about their router's packet loss, and actually got them to do things. |
| 19:10.31 | _Sam-- | so its not futile. |
| 19:10.34 | Hmmhesays | its kind of hard to dictate the route when you don't own the hardware |
| 19:10.37 | _Sam-- | but its a pain when it happens. |
| 19:10.53 | docelm0 | Russ598, so whats your question? |
| 19:11.09 | harryvv | manx how much do your pris cost? |
| 19:11.16 | Katty | harryvv: twenty five cents. |
| 19:11.22 | _Sam-- | heh |
| 19:11.28 | hikenboot | not being an internet guy how would you actually know where the packets are being dropped..for instance i keep having to switch my mtu size on my cable modem between 1500 and 1492 and back on different weeks |
| 19:11.38 | Hmmhesays | um traceroute |
| 19:11.40 | harryvv | katty, do you work in china? |
| 19:11.42 | [av]bani | so _Sam-- how much would you pay for a gxp autoprovisioner :D |
| 19:11.43 | _Sam-- | hikenboot : if you are not an internet guy, do not use a remote gw |
| 19:11.49 | _Sam-- | you NEED to be an internet guy |
| 19:11.53 | Russ598 | the issue above, I don't know where it's adding the leading 0 in and I don't want it to, it means we can't dial phone numbers that don't start with a 0, for example the talking clock over here is 123 but if you dial that our phone system dials 0123 |
| 19:11.56 | Katty | harryvv: hmmm, no. |
| 19:12.00 | harryvv | :) |
| 19:12.13 | Hmmhesays | pastebin.ca your dialplan |
| 19:12.20 | _Sam-- | [av]bani: considering ive lived without this long...0? |
| 19:12.20 | hikenboot | when i say not an internet guy i mean im no cisco ccie or anything |
| 19:12.35 | _Sam-- | if you dont know how to determine packet loss and where its happening... |
| 19:12.38 | hikenboot | i have built a frame-relay before and stuff like that but thats as far as i go |
| 19:12.40 | _Sam-- | you will never be able to live with a remote gw |
| 19:12.46 | [av]bani | _Sam-- ok no autoprovisioner for YUO! |
| 19:12.51 | [av]bani | :P |
| 19:12.58 | [av]bani | SUFFAR |
| 19:13.14 | Hmmhesays | hikenboot: TRACE ROUTE |
| 19:13.17 | Hmmhesays | google it |
| 19:13.22 | _Sam-- | traceroute doesnt show packet loss so good |
| 19:13.26 | _Sam-- | something like mTR is a good tool |
| 19:13.30 | _Sam-- | mtr |
| 19:13.38 | salviadud | yeah, i got mtr on suse 9.3 |
| 19:13.50 | salviadud | yet, i prefer slackware... |
| 19:14.01 | hikenboot | will it tell you if its an mtu size problem or give hints to the cause? |
| 19:14.09 | Dandan | lack rullz! |
| 19:14.10 | salviadud | still, i love chameleons |
| 19:14.14 | Dandan | *slack rullz! |
| 19:14.28 | salviadud | yeah, slack is the best |
| 19:14.29 | _Sam-- | i dont think many problems these days are MTU related |
| 19:14.31 | Hmmhesays | _Sam-- you are right, its good for narrowing the issue quickly though |
| 19:14.31 | hikenboot | <---hikenboot does apt-get install mtr |
| 19:14.32 | _Sam-- | but i could be wrong. |
| 19:14.38 | salviadud | the init scripts ere shiznits |
| 19:15.14 | _Sam-- | [av]bani: ok i will give you the lofty sum of forty seven cents to use the autobanimaker |
| 19:15.14 | salviadud | debian is so new school |
| 19:15.22 | Assid | anyone here good with zaptel |
| 19:15.26 | salviadud | takes out the fun of compiling it yourself |
| 19:15.31 | Hmmhesays | i like debian, it keeps me warm at night |
| 19:15.31 | *** join/#asterisk bbrdrgz (n=alex@p54B00C06.dip0.t-ipconnect.de) |
| 19:15.36 | _Sam-- | <--debianite |
| 19:15.40 | Assid | i cant get someone to help me with this.. and its really getting to me |
| 19:15.49 | Assid | kernel: rtc: lost some interrupts at 1024Hz. |
| 19:15.49 | Russ598 | any idea docelm0? |
| 19:15.49 | Hmmhesays | with what? |
| 19:15.52 | *** join/#asterisk adminguru (n=atze@u6-177.dsl.vianetworks.de) |
| 19:16.20 | Hmmhesays | ~pastebin |
| 19:16.21 | jbot | methinks pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca, or http://channels.debian.net/paste |
| 19:17.16 | Assid | Hmmhesays: thats the error.. |
| 19:17.18 | hikenboot | interesting my router is loosing about 2 % of packets |
| 19:17.26 | Assid | i tthink all my errors are cause of that one thing |
| 19:17.37 | }btorch{ | has anyone here setup zap with E&M wink ? |
| 19:17.42 | Hmmhesays | kernels real time clock is freaking out? |
| 19:17.56 | malverian[work] | Anyone know of a good load/stress testing suite for Linux? |
| 19:17.57 | [av]bani | _Sam--: do not mock happy fun bani |
| 19:18.06 | hikenboot | and even worse when i ping cisco there loosing 60% of my packets |
| 19:18.06 | docelm0 | is 123 a PSTN number? |
| 19:18.18 | Hmmhesays | um |
| 19:18.23 | docelm0 | 1st.. What is your asterisk config? |
| 19:18.29 | Hmmhesays | docelm0: you talk in cryptic language |
| 19:18.35 | _Sam-- | [av]bani: i would pay for it, but how do i know its not broked like all the other GS stuff? :P |
| 19:18.40 | Hmmhesays | 1234 is a coolio song |
| 19:18.43 | docelm0 | A@H or AMP or something? |
| 19:18.45 | fugitivo | malverian[work]: just tell us where we must pingdeath |
| 19:18.48 | Russ598 | 123 is the number for the talking clock in the UK |
| 19:18.49 | docelm0 | Hmmhesays, I know this |
| 19:18.56 | Russ598 | yeah, a pstn number |
| 19:19.01 | Hmmhesays | 1-2-3-4 get your woman on the floor |
| 19:19.04 | Hmmhesays | everybody get up and get down |
| 19:19.15 | docelm0 | Russ598, ok.. What did you use to configure asterisk to dial out? |
| 19:19.39 | docelm0 | And are you using ZAP? |
| 19:19.55 | Hmmhesays | hello everybody so glad you're here, coolio put the flow back in your ear |
| 19:20.11 | Russ598 | The box was built using the latest AAH iso, and most of it is configured through the AMP |
| 19:20.16 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 19:20.18 | Hmmhesays | lovely |
| 19:20.31 | Russ598 | yeah I'm pretty sure it uses ZAP |
| 19:20.32 | docelm0 | Russ598, um, to fix your problem FORMAT! |
| 19:20.46 | Russ598 | you're gonna say it's an AAH problem yeah? |
| 19:20.58 | docelm0 | A@H isnt something we support in this channel.. Quite frankly it sucks.. |
| 19:21.08 | docelm0 | No.. Im just saying A@H SUCKS |
| 19:21.11 | Hmmhesays | docelm0 you're only partially riht |
| 19:21.15 | Hmmhesays | *right |
| 19:21.24 | Hmmhesays | I like a@h for somethings |
| 19:21.28 | Russ598 | I'm a Windows man tho, frankly linux scares the bejesus out of me |
| 19:21.28 | docelm0 | ok FOP is ok.. but the rest blows |
| 19:21.34 | docelm0 | sigh.. |
| 19:21.36 | Hmmhesays | naw |
| 19:21.40 | Hmmhesays | i kind of like amp too |
| 19:21.41 | docelm0 | Then asterisk isnt the way to go then bub |
| 19:21.46 | *** join/#asterisk exstatica (i=exstatic@redline.mednor.net) |
| 19:21.52 | Russ598 | I'm tryin my hardest tho |
| 19:21.59 | Hmmhesays | Russ598 i'll fix your shit for pay |
| 19:22.05 | Hmmhesays | i need some booze money |
| 19:22.06 | docelm0 | I will too.. :) |
| 19:22.07 | *** join/#asterisk jozjan2 (n=jozjan@msi.cnl.tuke.sk) |
| 19:22.13 | docelm0 | I have booze.. |
| 19:22.16 | docelm0 | I just need money |
| 19:22.17 | docelm0 | haha |
| 19:22.23 | Russ598 | Hehe, I don't think my boss will like that |
| 19:22.27 | Dandan | . |
| 19:22.31 | Dandan | <PROTECTED> |
| 19:22.32 | Hmmhesays | actually i have to buy some more band gear |
| 19:22.32 | Dandan | <PROTECTED> |
| 19:22.32 | docelm0 | Russ598, in your config you are inserting a 0 somewhere |
| 19:22.41 | Dandan | srry, lag... |
| 19:23.00 | Hmmhesays | its not rocket science to find in aah |
| 19:23.02 | harryvv | Russ, really? |
| 19:23.03 | docelm0 | You need to check your dialplan for something like exten => 1XX.,1,Dial(ZAP/g1/0${EXTEN}) or something |
| 19:23.12 | harryvv | Russ, how does linux scare you? |
| 19:23.17 | Hmmhesays | i know exactly where to look Russ598 |
| 19:23.24 | jozjan2 | i have a registered SIP phone to asterisk but when i trying to call it from outside then i've got a msg SIP/2.0 403 Forbidden, any idea? |
| 19:23.27 | Russ598 | the actual number that's being passed to the trunk is the number that's entered without the 0 |
| 19:23.42 | *** join/#asterisk Dandan (i=dandan@ellie.pacanka.com) |
| 19:23.47 | Dandan | re :) |
| 19:23.59 | Russ598 | one of the guys I work with thinks it might be something to do with the zaptel or zapata configs |
| 19:24.05 | Russ598 | but I'm not sure :o) |
| 19:24.08 | Hmmhesays | doubtful |
| 19:24.21 | *** part/#asterisk Utah_Dave (n=boucha@0-2pool130-130.nas28.salt-lake-city1.ut.us.da.qwest.net) |
| 19:24.27 | Hmmhesays | so give me access to your box and i'll think about fixing it |
| 19:24.33 | Russ598 | harryvv, just don't know how to use it yet as well as I can use dos etc, but I'm learning gradually |
| 19:25.04 | Russ598 | can't do that, it's on the internal company network :-P |
| 19:25.07 | asterboy | anyone know if the IP500 headset can be used in conjunction with the handset? |
| 19:25.09 | harryvv | Russ, it takes time. But it has a much longer history of being stable then windows has ever been. |
| 19:25.11 | Hmmhesays | i accept hookers blow and beer as payment |
| 19:25.36 | Russ598 | harryvv, I'm starting to like the idea of it for server apps etc |
| 19:25.41 | justinu | nothing like doing lines off a dead hooker's ass |
| 19:25.42 | Hmmhesays | haha Russ598: i call bullshit on that one |
| 19:25.46 | asterboy | Hmmhesays, male of female? |
| 19:25.53 | Hmmhesays | you know it justinu |
| 19:26.00 | Hmmhesays | asterboy female |
| 19:26.13 | asterboy | no Brokeback Mountain for you! |
| 19:26.25 | [av]bani | _Sam--: you know its not broked because i wrote it |
| 19:26.27 | Hmmhesays | no i'm not into the gay cowboy thing |
| 19:26.33 | asterboy | lol |
| 19:26.39 | Russ598 | docelm0 where abouts is that part of the dialplan that you mentioned? |
| 19:27.01 | Hmmhesays | look at your outgoing trunk settings in AMP |
| 19:27.18 | Assid | aargh.. zaptel pissing me off |
| 19:27.29 | Hmmhesays | recompile your kernel |
| 19:27.36 | Hmmhesays | make sure rtc is part of it |
| 19:27.36 | asterboy | Can the Polycom IP 500 handset and HEADset work at the same time? |
| 19:27.57 | asterboy | Anyone here have a Polycom IP 500 with headset? |
| 19:28.18 | [TK]D-Fender | asterboy : No |
| 19:28.23 | asterboy | Plantronics makes the T100 Headset I see on eBay all the time. |
| 19:28.27 | Assid | i think its cause i have another timing device |
| 19:28.34 | Russ598 | the trunks have no dialling rules set, the oubound routing has the 9|. rule set but that's it |
| 19:28.34 | Hmmhesays | Assid: why? |
| 19:28.40 | malverian[work] | Dang... "stress" is pretty cool. |
| 19:28.43 | Assid | apparently its finding localhost kernel: usbcore: registered new driver wcusb |
| 19:28.44 | Assid | localhost kernel: Wildcard USB FXS Interface driver registered |
| 19:28.46 | asterboy | thx TKD |
| 19:28.52 | hikenboot | one last question is the astrisk image found for vmware a recommendation for use with businesses or should one custome install? |
| 19:29.00 | malverian[work] | Lets you specify how many threads to do cpu bound operations, how many for IO bound, how many memory bound, etc. |
| 19:29.14 | Hmmhesays | asterisk on vmware huh? sounds like a lot of late night tech calls to me |
| 19:29.20 | malverian[work] | Then it just forks that number of worker threads and runs them infinitely. |
| 19:29.34 | Hmmhesays | normally i'm drunk at night so that wouldn't be cool |
| 19:29.46 | hikenboot | lol |
| 19:29.54 | GerbilWrk | has anyone found a way to not show missed calls on a phone if the call goes to a queue with multiple agents and someone else picks it up? |
| 19:29.56 | asterboy | [TK]D-Fender: asterisk can do live monitoring of a SIP conversation right? |
| 19:30.05 | hikenboot | I run vmware esx in production ...works like a charm |
| 19:30.10 | malverian[work] | 14:28:30 up 1:08, 2 users, load average: 22.16, 16.88, 8.55 |
| 19:30.22 | [TK]D-Fender | asterboy : I believe you'd be thinking of "ChanSpy" |
| 19:30.26 | _Sam-- | malverian[work] : i bet calls sound good on that box :) |
| 19:30.36 | Hmmhesays | hikenboot: send me a copy |
| 19:30.38 | malverian[work] | _Sam--, Heh.. it's not a production box ;) |
| 19:30.39 | Hmmhesays | i'll try it |
| 19:30.48 | Assid | Hmmhesays: would that qualify it as another timing device? |
| 19:31.05 | _Sam-- | at what load number do calls start being impact, malverian[work] ? |
| 19:31.13 | _Sam-- | when the load hits 5 calls are impacted? |
| 19:31.14 | *** join/#asterisk chops (n=moise@207-172-221-143.c3-0.smr-ubr2.sbo-smr.ma.cable.rcn.com) |
| 19:31.16 | _Sam-- | or? <just curious> |
| 19:31.29 | docelm0 | Well the problem with AMP is its in AMP.. |
| 19:31.43 | docelm0 | Unless you know where to find it in the GUI you cant change it.. |
| 19:31.44 | Hmmhesays | your kernels rtc module should not effect your zaptel stuff |
| 19:31.46 | malverian[work] | _Sam--, That's a good question, something worth looking into. |
| 19:31.48 | hikenboot | for test purposes hmhesays...you should download vmware server beta...from vmware and run it on top of ubuntu with xfce...thats what i use at home. |
| 19:31.55 | Hmmhesays | sure you can I make changes to amps DP all the time |
| 19:32.04 | *** join/#asterisk Utah_Dave (n=boucha@0-2pool130-130.nas28.salt-lake-city1.ut.us.da.qwest.net) |
| 19:32.11 | Hmmhesays | hikenboot, i'll check it out |
| 19:32.23 | malverian[work] | _Sam--, I'd have to find the right combination of options to slowly increment the load average on a machine. |
| 19:32.32 | hikenboot | then download the vmware image..i would be interested in knowing what someone with experience has to say about astrisk vmware image version |
| 19:32.34 | asterboy | [TK]D-Fender: thanks! |
| 19:32.38 | Hmmhesays | you trying to cook an egg on your processor or something? |
| 19:32.43 | Russ598 | I'm browsing through the config files at the moment |
| 19:32.44 | _Sam-- | malverian[work] : what is creating the load on that box? sip? |
| 19:32.48 | _Sam-- | er sipp |
| 19:32.54 | asterboy | ~chanspy |
| 19:32.56 | jbot | extra, extra, read all about it, chanspy is an application that adds the ability to spy on any bridged call, this includes VoIP only calls where ZapScan/ZapBarge couldn't this can. As of october 19 2004, ChanSpy is not included in the standard Asterisk distribution or the development CVS tree. |
| 19:33.02 | malverian[work] | _Sam--, No no.. I'm using "stress" it's a stress testing utility. |
| 19:33.16 | hikenboot | Hmmhesays: one thing to note is that you need to be running a good amount of ram vmware is very ram hungry |
| 19:33.25 | Hmmhesays | Russ598 you should be able to find it in amp Russ598 |
| 19:33.34 | malverian[work] | _Sam--, I'm using it to do some quick tests of system stability on this new machine. |
| 19:33.39 | Hmmhesays | hikenboot: i used to run whatever the workstation version is |
| 19:33.56 | hikenboot | workstation is slow especially on top of windows |
| 19:34.07 | hikenboot | 38% overhead |
| 19:34.16 | hikenboot | 25% on linux |
| 19:34.23 | hikenboot | same with vmware server |
| 19:34.26 | Hmmhesays | whats the point then? |
| 19:34.28 | hikenboot | esx has only 3% |
| 19:34.33 | Hmmhesays | interesting |
| 19:34.34 | hikenboot | oops 8% |
| 19:34.40 | Hmmhesays | can I install windows on it? |
| 19:34.45 | hikenboot | yes |
| 19:35.02 | Assid | Hmmhesays: what do you suggest? |
| 19:35.04 | Hmmhesays | interesting, i could bring my laptop back to life |
| 19:35.23 | Hmmhesays | Assid: you could try unloading rtc and watch if you system freaks |
| 19:35.25 | hikenboot | xen which is the open source alternative will run windows using pacifica and vt enabled chips in the future it only has 3% overhead |
| 19:35.26 | Russ598 | Is it the extensions.conf I should be looking through? I'm pretty sure there's nowhere we've added that config so unless it's there by default then it's a different issue |
| 19:35.49 | Assid | unload rtc? |
| 19:35.52 | Assid | its a remote box |
| 19:36.05 | hikenboot | <---VMs are hikenboots thing! |
| 19:36.15 | malverian[work] | Load average up to 52,30,16 now ;) |
| 19:36.21 | iCEBrkr | oh WTF |
| 19:36.22 | iCEBrkr | Study Finds Linux Less Expensive Than Windows |
| 19:36.25 | iCEBrkr | oops wrong window |
| 19:36.30 | Pegger | http://pastebin.com/556415 does that mean that the DID does not know what context to enter into??? |
| 19:36.33 | trelane` | where do we submit requests for general-purpose prompts that aren't in asterisk-sounds-extras |
| 19:37.04 | iCEBrkr | trelane`: you pay for them and commit them yourself :P |
| 19:37.04 | Assid | Hmmhesays: should i just load up bristuff and try zaprtc? |
| 19:37.10 | hikenboot | anyways i will download the astrisk image and figure out what hardware i need for testing |
| 19:37.18 | trelane` | iCEBrkr, that's doable, elaborate a bit on this "submit" yuo speak of? |
| 19:37.20 | Hmmhesays | Assid: honestly man I have no idea I don't use zaptel hardware for anything |
| 19:37.39 | Assid | i just want meetme :( |
| 19:37.54 | trelane` | iCEBrkr, err commit I believe was the word you used |
| 19:38.24 | iCEBrkr | trelane`: Basically, you have Alison record the prompts you want and then you commit them to the asterisk-sounds-extra package. |
| 19:38.48 | Assid | whois alison? |
| 19:39.06 | Hmmhesays | what error do you get when you run meetme? |
| 19:39.32 | malverian[work] | woohoo.. 92 |
| 19:39.51 | _Sam-- | the load is at 92? |
| 19:39.52 | iCEBrkr | Assid: The voice of asterisk :P |
| 19:39.54 | trelane` | Assid, a woman with perhaps the world's most phenominal voice |
| 19:40.05 | Pegger | read: Rejected connect attempt from 69.25.143.141, who was trying to reach '16178303190@' |
| 19:40.05 | Pegger | <PROTECTED> |
| 19:40.10 | trelane` | iCEBrkr, right and what I'm trying to figure out is how to go about doing that |
| 19:40.39 | iCEBrkr | trelane`: I believe she has a link on the Wiki somewhere.. |
| 19:40.51 | Hmmhesays | Pegger: you using aah? |
| 19:41.19 | Hmmhesays | I ran into some weird stuff with aah the other day it said "ignoring invite" |
| 19:42.02 | Pegger | Hmmhesays what is aah? |
| 19:42.09 | _Sam-- | <PROTECTED> |
| 19:42.09 | Hmmhesays | asterisk at home |
| 19:42.12 | malverian[work] | <PROTECTED> |
| 19:42.15 | malverian[work] | BWAHAHAH... |
| 19:42.23 | _Sam-- | malverian[work]: you may end up with data corruption |
| 19:42.49 | iCEBrkr | malverian[work]: I've done that without a stress tester :P |
| 19:42.51 | malverian[work] | _Sam--, Like i said, this is a non-production machine that I'm just stress testing the hardware. |
| 19:42.55 | Hmmhesays | malverian[work] you should write a script to test disk writing speed... run it over night |
| 19:43.12 | _Sam-- | im not sure what running the load to 700 proves |
| 19:43.17 | malverian[work] | Hmmhesays, "stress -d 10" |
| 19:43.20 | _Sam-- | the box surely isnt receptive to input |
| 19:43.21 | mog_work | hey malverian[work] any luck on sphinx? |
| 19:43.29 | Pegger | Hmmhesays any ideas why it can not find the extension |
| 19:43.30 | *** join/#asterisk stoffell (n=stoffell@d5153FC1B.access.telenet.be) |
| 19:43.36 | Hmmhesays | then in the morning you can cook some pancakes on the same drive! |
| 19:43.38 | chops | So I'm working on a voicemail-related bounty, and I'm winding up with a lot of general refactoring of app_voicemail.c |
| 19:43.43 | stoffell | 'evening |
| 19:43.47 | malverian[work] | _Sam--, It's only took about 2 minutes to execute that uptime command ;) |
| 19:43.53 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-82-81-86-5.red.bezeqint.net) |
| 19:44.03 | malverian[work] | _Sam--, I was just seeing what combinations yield different load averages. |
| 19:44.17 | chops | Should I be submitting each refactor to bugs.digium.com as a separate "general cleanup" patch? |
| 19:44.20 | malverian[work] | mog_work, Pff.. I should do something with that again, eh? |
| 19:44.40 | mog_work | yes please |
| 19:44.50 | iCEBrkr | haha |
| 19:44.54 | mog_work | so i can have my own dial a therapist |
| 19:45.09 | *** join/#asterisk chapeaurouge (n=chap@user-85-201-82-146.tvcablenet.be) |
| 19:45.15 | iCEBrkr | "How long have you felt like this, preas 1 for 1 to 2 yrs, press 2 for 3 to 4 yrs..." |
| 19:45.15 | Assid | is there a way to stop wcusb from loading because of compatible hardware |
| 19:45.20 | malverian[work] | Maybe I should work on i ttoday. |
| 19:45.21 | iCEBrkr | "Enough about me, lets talk about you" |
| 19:45.41 | Hmmhesays | i don't think i could have a hot therapist |
| 19:45.48 | malverian[work] | mog_work, No, the reason for app_sphinx is that you don't have to press buttons ;) |
| 19:45.59 | malverian[work] | Oops, meant to send that to iCEBrkr |
| 19:46.08 | Hmmhesays | she would probably notice me mentally undressing her the entire time |
| 19:46.12 | iCEBrkr | malverian[work]: oh, I wasn't paying attention to that part. :P |
| 19:46.39 | iCEBrkr | So someone is actually working on a app_sphinx? |
| 19:46.42 | Hmmhesays | Assid: what error do you get when you fire up meetme? |
| 19:46.43 | mog_work | heh true but i would have app_eliza |
| 19:46.49 | mog_work | i would just use your stuff |
| 19:47.40 | *** join/#asterisk angom_w (n=angom@red-corp-200.38.16.10.telnor.net) |
| 19:47.49 | Hmmhesays | i'm trying to forget that i'm addicted to you |
| 19:49.54 | *** join/#asterisk ToTo (n=ToTo@host136-208.pool872.interbusiness.it) |
| 19:50.41 | *** join/#asterisk TonyM (n=TonyM@adsl-solo-80-168-226-4.claranet.co.uk) |
| 19:52.48 | *** join/#asterisk austinnichols101 (n=austinni@70.46.69.130) |
| 19:53.29 | salviadud | does FWD ever work??? |
| 19:53.31 | salviadud | hugh! |
| 19:53.35 | fugitivo | yes |
| 19:53.38 | fugitivo | just keep trying |
| 19:53.49 | salviadud | any other way i can do it? |
| 19:53.54 | salviadud | besides restart now |
| 19:54.00 | Hmmhesays | fwd works for me all the time |
| 19:54.00 | fugitivo | restart? |
| 19:54.05 | salviadud | i have FWD configured via iax.conf |
| 19:54.08 | salviadud | would that be it? |
| 19:54.26 | Hmmhesays | why would you do that? isn't fwd sip? |
| 19:54.39 | salviadud | yes it is |
| 19:54.47 | salviadud | yet... there is a way to bridge it |
| 19:55.15 | salviadud | seems like it sux big time though |
| 19:55.27 | salviadud | i'll try configuring it through sip tomorrow |
| 19:55.41 | salviadud | im hungry |
| 19:55.55 | salviadud | cya guys later, i love you, this is an awesome community of hackers |
| 19:56.25 | GerbilWrk | someone mind looking at this and help me figure out why i get hung up on when i call an ext. with DND on instead of going to voicemail. http://pastebin.com/556441 |
| 19:57.35 | fugitivo | GerbilWrk: where is your DIALSTATUS that goes to voicemail when the phone is busy? |
| 19:58.09 | GerbilWrk | where would that go? |
| 19:58.23 | Assid | Hmmhesays: WARNING[5318]: chan_zap.c:915 zt_open: Unable to open '/dev/zap/pseudo': No such device or address |
| 19:58.29 | Assid | ERROR[5318]: chan_zap.c:7396 chandup: Unable to dup channel: No such device or address |
| 19:58.31 | Assid | those |
| 19:58.36 | fugitivo | GerbilWrk: look at the sample files that comes with asterisk |
| 19:58.48 | fugitivo | GerbilWrk: there's a macro called stdexten that will fit your needs |
| 19:59.18 | GerbilWrk | i was using stdexten, and it wasn't doing it |
| 19:59.44 | fugitivo | it should |
| 19:59.47 | GerbilWrk | according to my book, if the phone is busy or congested, Dial sends you to priority n+101, so priority 102 should kick in |
| 19:59.55 | Hmmhesays | Assid: what are you using for a timing source? |
| 20:00.12 | *** join/#asterisk AlexCTI (n=alex@weston-69.65.86.231.myacc.net) |
| 20:00.16 | fugitivo | GerbilWrk: you can use DIALSTATUS now |
| 20:00.28 | Assid | trying to use ztdummy |
| 20:00.35 | GerbilWrk | ok, i'll look into that then, thanks |
| 20:00.36 | Assid | but if i do that.. i get that RTC error |
| 20:00.49 | Assid | so while i was playin around.. i noticed wcusb is recognised as well |
| 20:01.03 | Assid | so i tried that.. and i get the same error as above |
| 20:01.12 | Hmmhesays | so.. did you load ztdummy? |
| 20:01.21 | Assid | when i got that error.. NO |
| 20:01.27 | Hmmhesays | so load it |
| 20:01.40 | Assid | if i load ztdummy.. play() and everything else dies out |
| 20:01.51 | Hmmhesays | define "dies out" |
| 20:02.03 | Assid | it shows in console Playing file.. but i cant hear nothing |
| 20:02.14 | Assid | voicemailmain doesnt work. nothing with play does |
| 20:02.41 | Assid | play, background.. nothing |
| 20:03.41 | *** join/#asterisk HamYaI (i=HamYai@125.24.6.73) |
| 20:04.25 | Skarmeth | it ( Polycom IP 301) has only two lines, but all my users will be in front of a computer |
| 20:05.03 | Skarmeth | and a model with more lines will cost to high to my here in Brazil |
| 20:06.04 | buZz | :) |
| 20:07.37 | *** join/#asterisk IOscanner (n=IOscanne@c-24-0-183-125.hsd1.tx.comcast.net) |
| 20:08.06 | [av]bani | anyone here with a snom 360? |
| 20:08.17 | *** join/#asterisk fulgas (n=fulgas@a81-84-117-84.cpe.netcabo.pt) |
| 20:09.15 | Hmmhesays | assid that is way odd |
| 20:09.24 | Hmmhesays | you don't have any zaptel hardware right? |
| 20:09.27 | *** join/#asterisk moprilo (n=jjohn@200.122.157.92) |
| 20:10.38 | GerbilWrk | ok guys, about the DND issue, does this look any better? http://pastebin.com/556466 |
| 20:10.38 | moprilo | how do you call, when you here another phone ring, and you want to pull the call to your phone? |
| 20:10.38 | GerbilWrk | because it still isn't working |
| 20:10.38 | Katty | hihi. |
| 20:10.39 | moprilo | i mean, what-s the name for that |
| 20:10.39 | moprilo | when you 'hear' |
| 20:10.39 | robin_z | meep? |
| 20:10.39 | stoffell | moprilo, pick-up? |
| 20:10.50 | moprilo | but i mean, like if i hear the boss phone ringing, i could answer the call from my phone, on my desk |
| 20:11.26 | moprilo | that has a name, doesn't it? |
| 20:11.53 | Assid | Hmmhesays: not that i am aware of.. but wcusb does get registered |
| 20:12.00 | *** join/#asterisk KranZ (n=user@sme.bestline.net) |
| 20:12.00 | *** part/#asterisk KranZ (n=user@sme.bestline.net) |
| 20:12.02 | *** join/#asterisk KranZ (n=user@sme.bestline.net) |
| 20:12.02 | Assid | localhost kernel: usbcore: registered new driver wcusb |
| 20:12.02 | Assid | localhost kernel: Wildcard USB FXS Interface driver registered |
| 20:12.03 | stoffell | moprilo, read the answer.. |
| 20:12.08 | Hmmhesays | GerbilWrk ring groups? |
| 20:12.13 | Assid | as i said above.. and that doesnt work for me |
| 20:12.23 | Hmmhesays | Assid yank that out |
| 20:12.29 | Assid | onboard?!? |
| 20:12.32 | AlexCTI | Hi someone can help me http://pastebin.com/556473 |
| 20:12.33 | moprilo | ok |
| 20:12.45 | Hmmhesays | you have an onboard FXS internface? |
| 20:12.57 | GerbilWrk | Hmmhesays, what do you mean ring groups? |
| 20:13.00 | Assid | nope |
| 20:13.09 | Assid | onboard usb |
| 20:13.19 | Hmmhesays | remove the wildcard usb fxs interface |
| 20:13.21 | Assid | which is registered as a fxs (wildcard usb ) |
| 20:13.27 | Hmmhesays | hmmm |
| 20:13.29 | Hmmhesays | odd |
| 20:13.32 | Hmmhesays | or not I dunno |
| 20:13.53 | robin_z | still need a mirror to use my GXP2000 :( |
| 20:14.31 | stoffell | robin_z, you have the bug from latest firmware? |
| 20:14.43 | robin_z | indeed |
| 20:15.08 | stoffell | hm, sad it is, daily reboot helps, we experience it on 1 phone.. (other 8 phones no problem) |
| 20:15.15 | stoffell | wait for firmware update.. |
| 20:15.17 | robin_z | daily? |
| 20:15.20 | robin_z | hourly |
| 20:15.32 | stoffell | oh, it's so bad? oops.. |
| 20:15.43 | robin_z | phone is now useless |
| 20:15.59 | _Sam-- | someone said they had a 1.0.2.9 gxp version already |
| 20:16.02 | _Sam-- | but i havaent seen it myself |
| 20:16.08 | _Sam-- | maybe its in germany |
| 20:16.13 | Hmmhesays | my bad GerbilWrk: pickupgroup |
| 20:16.17 | stoffell | hm, _Sam--, maybe an internal build?(not public) |
| 20:16.30 | _Sam-- | stoffell : i think maybe |
| 20:17.50 | GerbilWrk | Hmmhesays, think you meant that for moprilo |
| 20:18.29 | Hmmhesays | you are right |
| 20:19.44 | *** join/#asterisk synthetiq (n=roger@64.201.13.50) |
| 20:22.22 | *** join/#asterisk airdog (n=kvirc@S01060007e9584bcd.vs.shawcable.net) |
| 20:23.54 | *** join/#asterisk FuriousGeorge (n=Brian@ool-44c5a9b8.dyn.optonline.net) |
| 20:24.23 | FuriousGeorge | hey all,m anyone have an issue with their snom360 where it rings for the first incomming call but not for the second |
| 20:24.48 | [av]bani | FuriousGeorge: did oyu repro the snom redial bug? |
| 20:25.10 | FuriousGeorge | [av]bani: i dont believe i have |
| 20:25.17 | [av]bani | FuriousGeorge: what firmware ver? |
| 20:25.19 | FuriousGeorge | i got the latest firmware |
| 20:25.23 | FuriousGeorge | 5.3b i believe |
| 20:25.38 | [av]bani | does redial missed call work for you? |
| 20:25.54 | [av]bani | ring the snom from an extension, then on the snom try to redial last missed call |
| 20:25.55 | hensema | hi, is there a way to reduce the number of interrupts/sec generated by hfc isdn cards, or should be just use some bigger iron in order to keep up with the cards? |
| 20:26.00 | FuriousGeorge | gonna see |
| 20:26.20 | [av]bani | mine just sits there, says "dialing" and lights the led, but does nothing |
| 20:26.22 | Katty | Hmmhesays: what does this "Services" button on my polycom 500 supposed to be for? |
| 20:26.30 | Katty | Hmmhesays: I poke it and nothing happens. |
| 20:26.40 | AlexCTI | Hi someona can help ti fix this problem: Unable to find a codec translation path from g729 to ulaw |
| 20:26.40 | [av]bani | Katty: hit it with a hammer |
| 20:26.47 | _Sam-- | Katty: give it some oxycontin |
| 20:26.47 | Katty | [av]bani: k |
| 20:26.58 | Katty | _Sam--: that's for me, thankyouVERYmuch. |
| 20:27.01 | _Sam-- | hahah |
| 20:27.09 | [av]bani | OMG SHARE DA WEALTH |
| 20:27.16 | Katty | [av]bani: shan't. |
| 20:27.18 | Hmmhesays | Not a clue, i don't have a polycomm phone |
| 20:27.23 | Katty | Hmmhesays: okies. |
| 20:27.25 | [av]bani | Katty: shall! |
| 20:27.31 | _Sam-- | Katty: when do the teeth come out? |
| 20:27.41 | Katty | [av]bani: that oxycontin is to keep me from screaming in pain (= |
| 20:27.47 | Katty | _Sam--: friday. |
| 20:28.01 | _Sam-- | [av]bani: did you hear, katty donated 11 inches to needing women |
| 20:28.12 | [av]bani | o_O |
| 20:28.17 | *** join/#asterisk Teeli (i=Tili@202-133-67-129-dialup.sat.net.pk) |
| 20:28.19 | _Sam-- | ya its true |
| 20:28.32 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 20:28.40 | FuriousGeorge | [av]bani: forget redialing last missed call, as of right now i get "no route to destination when i call the snom360" but it can call out just fine |
| 20:28.45 | _Sam-- | guess thats one of the benefits of having long hair |
| 20:29.05 | Katty | [av]bani: quite true. |
| 20:29.13 | Katty | [av]bani: it went to little girls who have cancer. |
| 20:29.23 | ManxPower | FuriousGeorge, sounds like the phone isn't registered |
| 20:29.23 | FuriousGeorge | [av]bani: but i have a missed call from myself, and i was able to call it back, thats not the same as last missed call though right? |
| 20:29.27 | pb_ | FuriousGeorge: "no route to destination" generally means that the phone has dropped its registration. use "sip show peers" to see whether it is in fact registered. |
| 20:29.42 | _Sam-- | how long was hair, past mid-back it sounds? |
| 20:29.50 | _Sam-- | s/was/was your/ |
| 20:30.30 | [av]bani | FuriousGeorge: use the redial function to do it |
| 20:30.36 | FuriousGeorge | pb_, [av]bani: u guys are right its dropping registration after the first incomming call |
| 20:31.04 | [av]bani | FuriousGeorge: have someone call your snom, but dont answer. then on your snom, hit redial and go to missed calls, then hit ok on the missed call and see if the snom can auto-redial the missed call |
| 20:31.12 | FuriousGeorge | done |
| 20:31.21 | FlyboySR22 | Hey Everyone |
| 20:31.24 | ManxPower | AlexCTI, The fix is to either disable G729 or purchase the G729 licenses |
| 20:31.44 | FuriousGeorge | and it can |
| 20:31.49 | [av]bani | hmm |
| 20:31.58 | FuriousGeorge | wait |
| 20:32.07 | [av]bani | is it actually ringing or is it just sitting there |
| 20:33.18 | FuriousGeorge | now its working |
| 20:33.21 | [av]bani | ? |
| 20:33.34 | batphone | i cant seem to match the vendor-class-identifier for a cisco 7940 running sip 5.3 |
| 20:34.00 | batphone | it reports back as "Cisco IP Phone 7490", but when I put that in the dhcpcd.conf it wont match |
| 20:34.19 | _Sam-- | [av]bani: i pipe the output of your banimaker forever to /dev/null! |
| 20:34.20 | batphone | i can get it to match on other VC strings but in this case I dont even SEE the VC string actually coming across |
| 20:34.35 | Pegger | <PROTECTED> |
| 20:35.02 | *** join/#asterisk shnarff (n=whois@216.190.144.90) |
| 20:36.16 | bcnl | 12:27 < AlexCTI> Hi someona can help ti fix this problem: Unable to find a codec translation path from g729 to ulaw |
| 20:36.31 | bcnl | AlexCTI: buy some g729's it's worth it for the less bandwidth it takes |
| 20:36.53 | GerbilWrk | ok guys, about the DND issue, does this look any better? http://pastebin.com/556466 |
| 20:38.36 | GerbilWrk | because i'm still just getting a dropped call |
| 20:39.19 | Juggie | Pegger, in iax.conf you have given that peer access to context=* probally |
| 20:39.30 | Juggie | and in your dial you are donig IAX2/phonenumber |
| 20:39.45 | Juggie | either restrict that peer to a context eg, context=iax in your iaxconf |
| 20:39.46 | Juggie | OR |
| 20:39.54 | Juggie | do IAX2/phonenumber@iax |
| 20:40.01 | FuriousGeorge | [av]bani: it seems to be working and it seems to be redialing fine |
| 20:40.01 | Juggie | or whatever the proper context is for you |
| 20:40.09 | *** join/#asterisk Whisk (i=whisk@whisk.gotadsl.co.uk) |
| 20:40.17 | FuriousGeorge | i dunno what was happening before but there was nat |
| 20:40.18 | FuriousGeorge | is nat |
| 20:40.19 | ManxPower | GerbilWrk, you forgot to do a reload |
| 20:40.21 | FuriousGeorge | so... |
| 20:40.48 | GerbilWrk | i did a reload, and a stop now, restart, reload |
| 20:41.01 | GerbilWrk | still getting the same thing, i'm very confused |
| 20:41.18 | Pegger | Juggie, i belive so http://pastebin.com/556508 with user/pass commented out |
| 20:41.44 | [av]bani | FuriousGeorge: hrm |
| 20:41.45 | ManxPower | GerbilWrk, Well I'm not seeing the Goto in the CLI output on pastebin.ca |
| 20:41.46 | Juggie | Pegger, you have no context=0 |
| 20:41.50 | Juggie | er, context= |
| 20:42.18 | Pegger | Juggie, please explain what you mean |
| 20:42.19 | AlexCTI | Someone can tell me if I can move my licence g729 to another server? |
| 20:42.33 | Pegger | Juggie, feel free to edit the pastebin |
| 20:42.40 | bcnl | AlexCTI: kinda, depends on digiums mood :P |
| 20:42.44 | Juggie | pegger, the problem is the lack of a context |
| 20:42.56 | Pegger | Juggie, please show me what I need to fix |
| 20:43.00 | Juggie | the other end does not know which context to use to handle the call so it rejects it |
| 20:43.06 | Juggie | well that totally depends on your provider |
| 20:43.18 | AlexCTI | bncl, so do I need directly with them.. right? |
| 20:43.27 | Pegger | Juggie, what part of the config is the context |
| 20:43.36 | bcnl | AlexCTI: yea, I just got some this weekend... user response has been really positive |
| 20:43.38 | _Sam-- | hey bani, hints can only hint 7 extensions at once? |
| 20:43.39 | Juggie | Pegger, show me your dial(iax2/ string |
| 20:43.39 | Pegger | Juggie, could you edit the pastebin |
| 20:43.49 | Juggie | i can, but the problem isnt really in here. |
| 20:43.57 | *** join/#asterisk ke4qqq (n=chatzill@68-115-212-155.static.spbg.sc.charter.com) |
| 20:43.57 | bcnl | AlexCTI: you just oder via the web $10usd per channel and they email you a license code to activate them with |
| 20:44.13 | GerbilWrk | ManxPower, reload it, i pasted the most recent error, may or may not be different, with the bottom three lines of the reload |
| 20:44.18 | [av]bani | _Sam--: ? |
| 20:44.19 | Pegger | Juggie, well the thing is taht when i call the did from my cell phone i get the Rejected connect attempt from 69.25.143.141, who was trying to reach ' |
| 20:44.43 | FuriousGeorge | [av]bani: there goes registration again. dropped |
| 20:44.44 | [av]bani | _Sam--: no, thats a polycom limitation |
| 20:44.48 | _Sam-- | i read something on the ast-users about polycom only being able to show 7 |
| 20:44.49 | [av]bani | FuriousGeorge: yay! |
| 20:44.52 | _Sam-- | but a digium guy responded |
| 20:44.58 | _Sam-- | and said it was an asterisk thing |
| 20:45.05 | [av]bani | _Sam--: its because polycom has a 7 buddy watch limit, not an asterisk limit |
| 20:45.28 | Juggie | pegger, call the did with your cell and paste all of the asterisk output into pastebin |
| 20:45.29 | [av]bani | _Sam--: polycom _can_ monitor more than 7 lines, but not via buddy watch. you need a different sip mechanism to do that, which * does not yet support |
| 20:45.31 | Juggie | and send me the link |
| 20:45.43 | [av]bani | _Sam--: but snom and other phones have no such limit |
| 20:45.46 | _Sam-- | i see...buddy watch is not like 'hints' ? |
| 20:45.57 | [av]bani | _Sam--: hints is what polycom calls buddy watch |
| 20:46.17 | _Sam-- | i thought maybe there was a reason the GXP only had 7 buttons for that |
| 20:46.20 | [av]bani | _Sam--: its a software limit, and polycom stated they have no intention to increase the limit |
| 20:46.20 | _Sam-- | but i guess not |
| 20:46.36 | FuriousGeorge | actually, its still registered and the clie says Calling RemoteSteve" but the phone doesnt respond |
| 20:46.44 | FuriousGeorge | of course i can reboot the thing and it will ring again |
| 20:46.45 | }btorch{ | can someone give me hand ? I'm trying to setup a TE110P to talk to a T1 card that is using E&M wink signalling ... the zaptel modules are loaded fine but I keep on getting the RED alarm |
| 20:46.49 | MstlyHrmls | [av]bani: I don't think that's the case; that's rather stupid... |
| 20:46.50 | FuriousGeorge | for the first few calls |
| 20:47.03 | }btorch{ | the red light on the back keeps blinking |
| 20:47.14 | Juggie | }btorch{, did you run 'ztcfg -vvv' |
| 20:47.15 | _Sam-- | MstlyHrmls : they said it in writing in an email, or it seemed that way at least. |
| 20:47.30 | MstlyHrmls | _Sam--: hmmmm |
| 20:47.31 | AlexCTI | bcnl: Can you tell me how many lic aleady have set up on my server? |
| 20:47.40 | }btorch{ | I have been googling around but everything I tried didn't work |
| 20:47.41 | [av]bani | MstlyHrmls: well, that's what digium officially stated |
| 20:47.42 | _Sam-- | i am out of the loop on that one, so i dont know the real truth, mstly. |
| 20:47.54 | [av]bani | MstlyHrmls: on the asterisk-users ml |
| 20:47.54 | }btorch{ | Juggie: yes I did |
| 20:48.00 | }btorch{ | Juggie: it works fine |
| 20:48.23 | [av]bani | MstlyHrmls: i guess it's possible digium is lying! |
| 20:48.29 | MstlyHrmls | [av]bani: interesting, I'll have a look |
| 20:48.35 | MstlyHrmls | [av]bani: heh, I doubt that :-) |
| 20:49.00 | bcnl | AlexCTI: show g729 |
| 20:49.06 | bcnl | if you get a error, then you have none |
| 20:49.25 | ke4qqq | anyone ever have astapi add 1's to the dial string? |
| 20:49.46 | [av]bani | MstlyHrmls: http://lists.digium.com/pipermail/asterisk-users/2006-February/146983.html |
| 20:50.02 | [av]bani | MstlyHrmls: OMG DIGIUM LIES |
| 20:50.05 | MstlyHrmls | [av]bani: ta |
| 20:50.14 | MstlyHrmls | [av]bani: dude, I didn't say that |
| 20:50.14 | }btorch{ | I have both my zaptel.con f and zapata.conf on pastebin .. http://pastebin.com/556523 |
| 20:50.17 | [av]bani | MstlyHrmls: :D |
| 20:51.09 | }btorch{ | I don't think it matters if asterisk is running or not since just by loading the modules I already get a RED light (although its a solid red) |
| 20:51.12 | Hmmhesays | FuriousGeorge you have the phone behind nat? |
| 20:51.39 | MstlyHrmls | [av]bani: *grumble* that's @#$#@ retarded... |
| 20:51.42 | MstlyHrmls | :-) |
| 20:51.57 | [av]bani | MstlyHrmls: thats polycom |
| 20:52.01 | _Sam-- | yeah like whats the point of the expansion module thing |
| 20:53.43 | ManxPower | _Sam--, Not much, since it can only monitor up to 8 extensions |
| 20:54.15 | ManxPower | That was talked about on the mailing lists a couple of days ago. Too bad you miss out on all that useful information by not being subscribed to the mailinglists. |
| 20:54.30 | [av]bani | you're supposed to use polycoms with Polycom(tm) PBX(c)(r) System(p.pend)(tm) |
| 20:55.14 | [av]bani | not that commie asterisk opensores crapola |
| 20:55.15 | *** join/#asterisk bbrdrg1 (n=alex@p54B000F5.dip0.t-ipconnect.de) |
| 20:55.21 | ManxPower | I should be authorized to download Polycom firmware in less than a week 8-) |
| 20:55.21 | bbrdrg1 | Hi everyone. a simple question - i need to dial pstn calling card with IVR from asterisk, sipUA -> asterisk -> FXO -> PSTN calling card, any ideas ? |
| 20:55.43 | [av]bani | MstlyHrmls: cool, you're getting your top secret national security clearance then |
| 20:55.51 | *** join/#asterisk denon (i=denon@tor/session/x-0a9152d04a2265ee) |
| 20:55.51 | *** mode/#asterisk [+o denon] by ChanServ |
| 20:55.59 | [av]bani | MstlyHrmls: just passed the background check and lie detector? |
| 20:56.01 | Hmmhesays | i've got 501's in the field |
| 20:56.12 | [av]bani | damn autocomplete |
| 20:56.21 | MstlyHrmls | [av]bani: it's ManxPower that's getting the secret polycom decoder ring |
| 20:56.22 | MstlyHrmls | :-) |
| 20:56.28 | MstlyHrmls | heh |
| 20:56.33 | [av]bani | yeah, irssi doesnt seem to think so though |
| 20:56.49 | MstlyHrmls | I guess irssi just likes me better |
| 20:57.13 | Nugget | It's all part of ManxPower's master plan to put the polycom firmware on grandstream phones. ;) |
| 20:57.16 | *** join/#asterisk Cybertoy (n=maxim@dsl254-123-241.nyc1.dsl.speakeasy.net) |
| 20:58.56 | }btorch{ | is there anything special that really needs to be changed from pri_net setup to a e&m ? |
| 20:59.02 | [av]bani | FuriousGeorge still about? |
| 20:59.07 | FuriousGeorge | ya |
| 20:59.12 | FuriousGeorge | was just about to ask you something |
| 20:59.17 | *** join/#asterisk Cyon (n=cyon@216.179.31.166) |
| 20:59.19 | Cyon | Hey all |
| 20:59.29 | }btorch{ | the card worked fine with a pri setup but now as a e&m it gives me a headache |
| 20:59.32 | FuriousGeorge | you think this mysterious behavior warrents a support ticket |
| 20:59.38 | [av]bani | FuriousGeorge: ? |
| 20:59.43 | [av]bani | oh the reg loss? |
| 20:59.45 | Cyon | Quick question; had a box die, rebuilding it now; what is the optimal version of linux to run when I need to use the digium cards (zaptel)? |
| 20:59.50 | Cyon | redhat? slack? |
| 20:59.51 | malverian[work] | I'm thinking about writing OSS firmware for SNOM 320 :-P |
| 21:00.00 | _Sam-- | ManxPower: i wont miss anything...since i subscribed last night, thank you for checking :) |
| 21:00.01 | FuriousGeorge | [av]bani: ive since discovered that sometimes * doesnt even know the reg is lost |
| 21:00.02 | malverian[work] | They provide all of their build tools and the patched kernel sources and such they use. |
| 21:00.10 | [av]bani | FuriousGeorge: nat? |
| 21:00.30 | FuriousGeorge | [av]bani: yeah, on both sides |
| 21:00.41 | [av]bani | malverian[work]: http://www.openhardphone.org/ |
| 21:00.41 | FuriousGeorge | but i got the ports forewarded server side |
| 21:00.45 | FuriousGeorge | and eyeveam always workds |
| 21:00.48 | FuriousGeorge | worksa |
| 21:00.53 | [av]bani | FuriousGeorge: snom + nat = badness |
| 21:00.56 | denon | worksa? |
| 21:01.00 | Cyon | Sorry to be annoying, but any recommendations? Want to grab a distro asap. |
| 21:01.02 | *** join/#asterisk WasPhantom (n=neil@203-86-192-98.tasman.net) |
| 21:01.02 | denon | what're you, norwegian? :) |
| 21:01.30 | FuriousGeorge | [av]bani: that really stinks, why can eyebeam get it right but not the 360? |
| 21:01.31 | [av]bani | FuriousGeorge: yep, just like polycom cant get it right either |
| 21:02.20 | bbrdrg1 | Any ideas on how to connect a sip call leg to pstn which has IVR in such a way, that the calling party would not hear the ivr on pstn ? |
| 21:02.25 | [av]bani | FuriousGeorge: out of all the phones ive used, gxp2000 does best with nat |
| 21:02.25 | FuriousGeorge | [av]bani: you think its nat on the phone side, server side, or a combination |
| 21:02.35 | [av]bani | FuriousGeorge: phone side |
| 21:03.01 | FuriousGeorge | [av]bani: any idea why it would consistently work at first, then consistently stop working? |
| 21:03.06 | Hmmhesays | you need to set your re-reg period to something insanely low |
| 21:03.09 | Hmmhesays | like 30 seconds |
| 21:03.19 | FuriousGeorge | Hmmhesays: hmmmm |
| 21:03.22 | ManxPower | bbrdrg1, NOW you are asking a decent question. Are you using analog ports or digital ports? |
| 21:03.25 | [av]bani | FuriousGeorge: because the nat connection times out |
| 21:03.34 | ManxPower | Hmmhesays, or just use qualify=yes |
| 21:03.34 | Hmmhesays | your dynamic port map is closing |
| 21:03.46 | [av]bani | FuriousGeorge: try setting up a local stun server and pointing the snom at it |
| 21:03.50 | bbrdrg1 | ManxPower: analog FXO ports |
| 21:03.52 | Hmmhesays | try not using stun |
| 21:03.54 | _Sam-- | qualify = yes is a good start to try |
| 21:04.01 | [av]bani | qualify=yes would keep it alive too |
| 21:04.11 | Hmmhesays | should, i've had some problems with that though |
| 21:04.15 | ManxPower | Dial(Zap/1/5551212wwww4444444ww666666) |
| 21:04.20 | ManxPower | where "w" means "wait .5 second" |
| 21:04.25 | Hmmhesays | god i'ms till burping pizza from last night, ugh |
| 21:04.27 | Cyon | Grrr, so slack is easily solid for zap? |
| 21:04.57 | _Sam-- | Cyon: i think the opinion is that one distro vs. another doesnt make much difference |
| 21:05.04 | _Sam-- | at least, thast my opinion. |
| 21:05.17 | Hmmhesays | my *nix is better than your *nix _Sam-- |
| 21:05.21 | FuriousGeorge | trying w nat=yes and checking stun settings |
| 21:05.26 | FuriousGeorge | i mean qualify=yes |
| 21:05.27 | Hmmhesays | stun sucks |
| 21:05.44 | ManxPower | If you use nat=yes in asterisk you should NOT enable NAT settings on the device |
| 21:06.17 | bbrdrg1 | ManxPower: will the same "w" work with SIP/2333wwww333w22w22345 ? |
| 21:06.25 | Cyon | _Sam--: Works for me, just didn't want to go with one that had known bugs with asterisk.zap |
| 21:06.31 | Cyon | s/\./\// |
| 21:06.32 | FuriousGeorge | ManxPower: stun and ice are both off in the device |
| 21:06.42 | ManxPower | bbrdrg1, no, because w only works for analog FXO |
| 21:06.48 | FuriousGeorge | and i see there it has a keep alive field which is blank so ill set that if qualify dont work |
| 21:06.51 | justinu | is there a working free/opensource x server for windows? |
| 21:06.55 | ManxPower | That's why I asked what type of port you are using. |
| 21:07.08 | [av]bani | FuriousGeorge: turn on both ice and point it at a stun |
| 21:07.11 | [av]bani | FuriousGeorge: and use qualify |
| 21:07.17 | ManxPower | For digital ports you need to "show application dial" and use whatever it says about sending DTMF after answer. |
| 21:07.19 | [av]bani | FuriousGeorge: my guess is qualify will work best |
| 21:07.39 | [TK]D-Fender | Cybertoy : Slackware works very well for all my servers. |
| 21:07.41 | FuriousGeorge | im trying qualify and ill turn on ice since i got it in eyebeam, and eyebeam works ok |
| 21:07.58 | [av]bani | FuriousGeorge: have you noticed the snom 360 dialtone is weird? and the busy tone too |
| 21:08.12 | justinu | mine sounds ok... what's up with it? |
| 21:08.20 | [TK]D-Fender | Cyon rather..... |
| 21:08.32 | [av]bani | justinu: setup a * extension to Playtones(dial), then compare the snom 360 dialtone to what you get from * |
| 21:08.39 | justinu | k |
| 21:08.50 | [av]bani | justinu: the snom 360 dialtone sounds harsh and aliased |
| 21:09.03 | [av]bani | playtones is smooth, and sounds exactly like i get from the PSTN |
| 21:09.07 | justinu | i'm not surprised... i wasn't that impressed with the audio quality on the 360 |
| 21:09.25 | [av]bani | justinu: no no, playtones(dial) _through_ the 360 from * sounds fine |
| 21:09.35 | sevard | Can * create bill sheets for long distance calls based on extension? |
| 21:09.38 | [av]bani | justinu: nothing wrong with 360 audio quality, its just the hardcoded indications |
| 21:09.50 | ManxPower | "w" works the same way with the option |
| 21:10.03 | justinu | i'm still not that impressed with the audio |
| 21:10.09 | justinu | it's adequate |
| 21:10.22 | [av]bani | justinu: and then compare Playtones(busy) with what you get from Congestion() |
| 21:10.22 | justinu | i don't think they have any PLC or jitter buffer tho |
| 21:10.30 | *** join/#asterisk MikRoB (n=nevesh@80.91.116.41) |
| 21:10.53 | [av]bani | justinu: congestion() on the snom360 sounds very weird, while playtones(busy) sounds the same as what i get from the PSTN and from any other voip phone |
| 21:11.34 | *** join/#asterisk Lathos42 (n=Lathos42@adsl-68-76-48-105.dsl.lgtpmi.ameritech.net) |
| 21:11.52 | *** part/#asterisk MikRoB (n=nevesh@80.91.116.41) |
| 21:12.01 | ManxPower | congestion or busy would send the correct message to the device, where playtones just sends the audio and the phone doesn't know that it's a special indication, it things it's voice. |
| 21:12.25 | Pegger | if I have odbc installed shoudl asteisk default to the text files if it can not connect to the database |
| 21:12.28 | [av]bani | ManxPower: what the snom360 plays for congestion is bizarre, it doesnt sound like what i get from PSTN _at all_ |
| 21:12.44 | [av]bani | ManxPower: while every other phone I have, plays the proper PSTN congestion sound |
| 21:12.50 | ManxPower | [av]bani, that would be a problem in the config of the phone. It's prolly a german tone, since the phones are german. |
| 21:12.53 | [av]bani | ManxPower: polycom, even the cheapy grandstream sounds correct |
| 21:13.01 | [av]bani | ManxPower: it's set for US |
| 21:13.08 | ManxPower | [av]bani, complain to SNOM |
| 21:13.14 | [av]bani | i did, they said "what problem?" |
| 21:13.24 | Katty | hmm. |
| 21:13.31 | [av]bani | i'm trying to get some corroboration here so i can tell them theyre full of shit |
| 21:14.11 | justinu | heh |
| 21:14.19 | ManxPower | [av]bani, ask other SNOM users. |
| 21:14.19 | justinu | i'll do the comparison next time I'm near that phone |
| 21:14.24 | [av]bani | ManxPower: even barring that, dialtone sounds bogus too. its a weird approximation of dialtone |
| 21:14.31 | sevard | Can Asterisk create bill sheets for long distance calls based on extension? |
| 21:14.34 | [av]bani | ManxPower: like they used square waves or something |
| 21:14.45 | Cyon | [TK]D-Fender: Thanks. :) |
| 21:14.50 | ManxPower | sevard, no, but it will log the information and you can feed it into your custom billing application |
| 21:14.56 | jsharp | sevard: Not directly. You'd have to pull it out of the CDR records and massage the data yourself. |
| 21:15.25 | [av]bani | ManxPower: that's what i'm doing right now.... eg justinu and furiousgeorge, before you butted in :) |
| 21:15.30 | sevard | Does anyone have an example of how this works? That's a bit fuzzy. |
| 21:15.55 | ManxPower | sevard, /var/log/asterisk/cdr-csv |
| 21:16.05 | justinu | bani: probably will be friday |
| 21:16.23 | [av]bani | justinu: ?! |
| 21:16.27 | sevard | ManxPower: So Master.csv is a record of all calls |
| 21:16.33 | ManxPower | sevard, should be |
| 21:16.53 | jsharp | Or you can set up logging of CDR to an SQL database if that makes you feel more warm & fuzzy. |
| 21:17.05 | justinu | bani: the phone is at the office... i'm telecommuting until friday |
| 21:17.08 | sevard | So, if I create a perl script to parse that information, that'd be good |
| 21:17.23 | ManxPower | sevard, there really are not many billing applications for Asterisk because 1) everyone's billing needs are different and 2) people who write their own billing apps don't want to give the code to their competition |
| 21:17.23 | sevard | jsharp: that sounds better, do you have a document explaining how to go about that? |
| 21:17.35 | justinu | astbill.com |
| 21:17.36 | sevard | ahh. |
| 21:17.43 | [av]bani | justinu: :< |
| 21:17.59 | justinu | bani: sorry :( |
| 21:18.10 | [av]bani | justinu: go into work right now dammit! |
| 21:18.15 | sevard | I was looking at how one might keep track of long distance calls from extensions, such as a hotel might do |
| 21:18.15 | [av]bani | just for me! |
| 21:18.15 | justinu | lol |
| 21:18.23 | jsharp | http://www.voip-info.org/wiki-Asterisk+cdr+mysql |
| 21:20.56 | jsharp | hmm. |
| 21:21.33 | *** join/#asterisk darkskiez (n=darkskie@bb-87-81-62-203.ukonline.co.uk) |
| 21:22.49 | [av]bani | http://www.aviransplace.com/index.php/archives/2006/02/15/microsoft-upgraded-motherboard-new-licence/ |
| 21:22.53 | [av]bani | yay! |
| 21:22.53 | austinnichols101 | what's the correct codec to download for g729 on a dual p3 - 700 box? |
| 21:22.55 | [av]bani | its about time |
| 21:24.27 | Assid | are they freaking crazy |
| 21:24.38 | harryvv | wow |
| 21:24.48 | harryvv | new motherboard requires a new windows licence? |
| 21:25.08 | harryvv | that sucks. |
| 21:25.39 | Curus | The whole concept of non-transferrable licenses is invalid in several jurisdictions |
| 21:25.41 | Nugget | nutty |
| 21:26.04 | Nugget | I guess it sort of makes sense for OEM license, though |
| 21:26.09 | GoRK | does anyone have the release notes pdf for polycom sip firmware 1.6.5 |
| 21:26.11 | Nugget | they've always been handled differently |
| 21:26.11 | Assid | i aint gollowing it |
| 21:26.19 | Assid | i bought my license that s it |
| 21:26.23 | Assid | not following shit |
| 21:29.19 | [av]bani | Curus: do you think microsoft cares? |
| 21:29.33 | [av]bani | Curus: they dont even care about federal law, why should they care about local laws :D |
| 21:30.01 | [av]bani | Assid: three words: "windows activation code" |
| 21:30.39 | Assid | ta hell with them |
| 21:30.48 | Assid | i'll tell them my motherboard blew up |
| 21:31.08 | iCEBrkr | Pir8 W1nd0z3 |
| 21:31.39 | [av]bani | Assid: "we dont believe you. pay up sucker" |
| 21:32.05 | iCEBrkr | <Assid> K/THX/BYE, I'll just use Linux.. *flips off M$* |
| 21:32.09 | Skumling | hrm... damn spandsp |
| 21:32.13 | Nugget | or you could just buy a retail license. |
| 21:32.18 | Skumling | iCEBrkr: are you using spandsp? |
| 21:32.25 | iCEBrkr | Skumling: nope, should I be? |
| 21:32.30 | Assid | i cant cause of "certain" other sw |
| 21:32.30 | shnarff | no just call them ive dont it a bunch they will transfer it |
| 21:32.36 | iCEBrkr | Skumling: I'm 100% VoIP, faxing won't work for me. :) |
| 21:32.46 | [av]bani | faxing works for me |
| 21:32.50 | [av]bani | and im 100% voip |
| 21:32.54 | Skumling | iCEBrkr: wish that I could route fax to /dev/null |
| 21:33.03 | Skumling | [av]bani: okay, are you using spandsp? |
| 21:33.08 | iCEBrkr | [av]bani: ulaw? |
| 21:33.14 | [av]bani | Skumling: yes |
| 21:33.19 | *** join/#asterisk doughecka_ (n=Miranda@unaffiliated/doughecka) |
| 21:33.21 | [av]bani | Skumling: but passthrough works also |
| 21:33.26 | [av]bani | iCEBrkr: of course |
| 21:33.26 | bbrdrg1 | any idea why dial(SIP/peer,D(${EXTEN}) would not work ? |
| 21:33.31 | iCEBrkr | [av]bani: and that works? |
| 21:33.37 | [av]bani | iCEBrkr: yes |
| 21:33.49 | iCEBrkr | bbrdrg1: don't you have the D in the wrong spot? or am I smoking? |
| 21:34.06 | iCEBrkr | wait. WTF is D()? |
| 21:34.12 | Skumling | [av]bani: when you receive a fax using rxfax, does your asterisk/rxfax then send noise out right away when picking up... lasts for about ½ to 1 sec |
| 21:34.20 | bbrdrg1 | iCEBrkr: right spot for D, according voip-info |
| 21:34.36 | bbrdrg1 | D(digits): After the called party answers, send digits as a DTMF stream, then connect the call to the originating channel. |
| 21:34.51 | iCEBrkr | Yea, I still think it goes after it. |
| 21:34.53 | [av]bani | Skumling: no |
| 21:35.14 | iCEBrkr | bbrdrg1: Dial(SIP/1123,D(1234)) |
| 21:35.14 | *** part/#asterisk Vyeperman (n=Vye@ip68-8-174-154.sd.sd.cox.net) |
| 21:35.21 | iCEBrkr | errr |
| 21:35.29 | iCEBrkr | bbrdrg1: Dial(SIP/1123,45,trD(1234)) |
| 21:35.31 | iCEBrkr | or something |
| 21:35.32 | Skumling | [av]bani: seems like most transmitting faxes don't care about it, but some faxmachines get confused and won't handshake after the noise |
| 21:35.33 | [av]bani | Skumling: i use nvbackgrounddetect, and then fax,1,rxfax(blabla) |
| 21:35.50 | [av]bani | Skumling: and it sounds fine to me |
| 21:36.05 | iCEBrkr | [av]bani: Hrrm, I haven't tinkered with Fax with VoIP due to codec compression issues. I never new ulaw would work. |
| 21:36.17 | [av]bani | iCEBrkr: ulaw _must_ work |
| 21:36.18 | Skumling | [av]bani: okay, I've got DID's for the faxing, and when rxfax(yadda) is executed, it starts out with some ugly noise |
| 21:36.27 | [av]bani | Skumling: i dont get any noise |
| 21:36.32 | fafnir | i'll shake your hand |
| 21:36.33 | iCEBrkr | bbrdrg1: Oh and the line you pasted is missing a ) |
| 21:36.41 | Skumling | [av]bani: hrm okay. which versions of asterisk and spandsp? |
| 21:36.41 | [av]bani | Skumling: try answer() or playing some short blank sound before rxfax |
| 21:36.57 | Skumling | [av]bani: oooh yes, that is tried in a bunch of combinations ;) |
| 21:36.58 | [av]bani | Skumling: maybe routing the did directly into rxfax causes issues |
| 21:37.15 | harryvv | how does asterisk work with incomming faxes? does it reroute it to another server or store the fax image? |
| 21:37.27 | iCEBrkr | harryvv: they land in the 'fax' extension |
| 21:37.32 | [av]bani | Skumling: spandsp 0.0.2pre25, asterisk 1.2.4 |
| 21:37.40 | iCEBrkr | well, I can't confirm that as I never got it working right |
| 21:37.46 | bbrdrg1 | iCEBrkr: do i have to put r with D ? the line Dial(SIP/peer-name,D(12345)) |
| 21:37.50 | Skumling | [av]bani: same stuff here... are you using the "shipping" rxfax and txfax apps? |
| 21:37.56 | *** join/#asterisk r0d3nt (i=nobody@wsip-24-234-241-145.lv.lv.cox.net) |
| 21:37.59 | [av]bani | Skumling: yes |
| 21:37.59 | harryvv | ice, so the caller needs to enter in a extention that is for incomming faxes/ |
| 21:38.13 | Skumling | [av]bani: not using bristuff, I guess? |
| 21:38.22 | iCEBrkr | bbrdrg1: Dial(SIP/peer-name,25,D(12345)) |
| 21:38.23 | [av]bani | ab6983b51c412883545b36993d704999 app_rxfax.c |
| 21:38.26 | [av]bani | 8c8fcb263b76897022b4c28052a7b439 app_txfax.c |
| 21:38.33 | iCEBrkr | harryvv: no |
| 21:38.33 | [av]bani | Skumling: nope |
| 21:38.54 | iCEBrkr | harryvv: I believe that Asterisk will detect it's a fax and land in exten => fax |
| 21:39.04 | [av]bani | Skumling: but for now, i'm using fax passthrough since i dont trust rxfax yet |
| 21:39.19 | *** join/#asterisk ctooley (n=ctooley@rrcs-24-227-212-162.sw.biz.rr.com) |
| 21:39.25 | Skumling | [av]bani: humm... my rxfax is eb4f29e0264c9464398a9dd2ede8ef65 app_rxfax.c |
| 21:39.34 | Skumling | [av]bani: could I try your rxfax.c ? |
| 21:40.01 | Skumling | [av]bani: what are you passing through to? normal faxmachine? |
| 21:40.01 | AlexCTI | Someone knows how can I see how many concurrence lic for g729 i'm using on my sever? |
| 21:40.16 | *** join/#asterisk brif8 (n=Techno@lazyjtrainingcenter.com) |
| 21:40.31 | *** join/#asterisk saftsack (n=saftsack@p54A7F5D2.dip.t-dialin.net) |
| 21:40.32 | saftsack | hi |
| 21:40.34 | ctooley | What happens if I don't "Logoff" a manager connection? |
| 21:40.47 | [av]bani | Skumling: yes |
| 21:41.03 | [av]bani | fax,1,dial(SIP/FXS5) |
| 21:41.03 | *** join/#asterisk ToTo (n=ToTo@host136-208.pool872.interbusiness.it) |
| 21:41.08 | KranZ | and angel looses its wing |
| 21:41.11 | KranZ | er an |
| 21:41.27 | saftsack | how is the speech quality of the budge tel 101 phone? |
| 21:41.28 | KranZ | god, spelling sux *loses |
| 21:41.51 | ctooley | I need to originate calls but don't want to block on waiting for the response. |
| 21:41.57 | De_Mon | WARNING[1980]: app_meetme.c:281 careful_write: Failed to write audio data to conference: Bad address |
| 21:42.11 | Skumling | [av]bani: thank you... the files where identical though, besides of some code being indented |
| 21:42.12 | De_Mon | I get spammed with those when using app_meetme in 1.2.4 |
| 21:42.24 | synthetiq | whats the big deal with eyepea winnign the project in belgium? i have 2 installs with 400+ phones |
| 21:42.29 | synthetiq | :-P |
| 21:42.45 | iCEBrkr | saftsack: My BT100 isn't bad |
| 21:42.49 | De_Mon | synthetiq did you enter the competition? |
| 21:42.59 | *** join/#asterisk Jizzbug (n=derekm@199.227.154.26) |
| 21:43.14 | brif8 | Can anyone suggest a highly dependable, high quality VoIP provider: I want to route all my calls (LD and Local) VoIP +/- 40 concurrent calls. but quality is critical ? |
| 21:43.29 | iCEBrkr | brif8: No such thing LOL |
| 21:43.51 | tuxinator_linux | brif8: when you find one, let us know |
| 21:43.55 | Hmmhesays | hey, hey I wanna be a rockstar |
| 21:43.57 | brif8 | iCEBrkr: care to explain more ? |
| 21:44.09 | ctooley | Aha! Async is the correct answer here. |
| 21:44.10 | iCEBrkr | brif8: Nothing is guarenteed. |
| 21:44.13 | Skumling | [av]bani: do you know of any other spandsp-complaint apps for fax-reception? |
| 21:44.24 | *** join/#asterisk Flauto (n=zhao@71.194.194.48) |
| 21:44.40 | *** join/#asterisk Halshair (i=Halshair@ool-457c7be0.dyn.optonline.net) |
| 21:44.44 | [av]bani | Skumling: nope |
| 21:44.50 | [av]bani | Skumling: have oyu tried passthrough? |
| 21:44.51 | saftsack | iCEBrkr, my bt 101 sounds horrible :( |
| 21:44.53 | GerbilWrk | can someone clear this up for me, Feb 15 15:44:19 WARNING[24374]: pbx.c:1893 ast_pbx_run: Channel 'SIP/420-ce9e' sent into invalid extension '407-CONGESTION' in context 'usawide', but no invalid handler |
| 21:44.57 | Skumling | [av]bani: damnit. maybe I have to think of HylaFAX then |
| 21:45.00 | iCEBrkr | saftsack: Does it? What's wrong with it? |
| 21:45.43 | Skumling | [av]bani: yeah, and it works... but I would really like just getting e-mails with attached pdf's... |
| 21:46.03 | _Sam-- | Skumling : i am really out of the loop,. have not been watching your chat with bani... |
| 21:46.07 | _Sam-- | but you could use iaxmodem? |
| 21:46.11 | *** join/#asterisk justinu (n=Justin@cpe-72-129-86-208.socal.res.rr.com) |
| 21:46.25 | _Sam-- | i heard yesterday someone saying iaxmodem with hyla was working 100% for incoming |
| 21:46.45 | Skumling | _Sam--: okay? don't know iaxmodem... I want to receive faxes from my zaphfc devices primarily |
| 21:46.57 | Skumling | _Sam--: i don't really trust faxing over IP yet |
| 21:47.10 | Skumling | _Sam--: mmm, 100% sounds sweet++ |
| 21:47.12 | _Sam-- | for incoming, this person claimed 100% success rate. |
| 21:47.18 | Skumling | mm, 100% |
| 21:47.20 | saftsack | iCEBrkr, its not as good as my isdn telephone :( |
| 21:47.24 | _Sam-- | i still use hylafax with analog modems/POTs myself |
| 21:47.30 | hensema | hmmmm, would a preemptable kernel improve irq latencies so zaphfc will perform better? |
| 21:47.33 | saftsack | i use it too |
| 21:47.37 | Skumling | _Sam--: and that is just running smoothly? |
| 21:47.48 | _Sam-- | Skumling: ive been running it that way for over 10 years now |
| 21:47.51 | _Sam-- | runs fine. |
| 21:48.09 | Skumling | _Sam--: hrm... I need someone who has done more "real testing"... ;-D |
| 21:48.14 | FuriousGeorge | [av]bani: either its working or i set a record for registration uptime |
| 21:48.18 | FuriousGeorge | thnaks |
| 21:48.23 | FuriousGeorge | ice and qualify |
| 21:48.26 | bbrdrg1 | does anyone have any ideas why Dial(SIP/peer-name,30,D(12345)) is not working ? it dials the peer, byut never sends the DTMF according to D() option. Anyone ? please |
| 21:48.29 | Skumling | well, 10 years seems fine to me... hopefully I don't touch fax-thingys anymore at that time :) |
| 21:48.35 | FuriousGeorge | i have as feeling it was qualify that helped though, i thought i had that in there |
| 21:48.51 | *** join/#asterisk Vyeperman (n=Vye@ip68-8-174-154.sd.sd.cox.net) |
| 21:49.31 | _Sam-- | it is qualify. |
| 21:49.51 | *** join/#asterisk stkn_ (i=nobody@gentoo/developer/pdpc.active.stkn) |
| 21:50.36 | brif8 | iCEBrkr: what about quality ? |
| 21:51.03 | iCEBrkr | brif8: How can you guarentee quality when calls run across the wild-wild-internet? |
| 21:51.15 | Halshair | Is this the right place to ask an Asterisk@home question? |
| 21:51.20 | iCEBrkr | ~amp |
| 21:51.22 | jbot | methinks amp is NOT supported here! people using it should join #amportal |
| 21:52.34 | Nivex | bah! There's nothing like hand coding your dialplan :) |
| 21:52.40 | *** join/#asterisk }btorch{ (n=kvirc@208.63.19.184) |
| 21:53.00 | *** join/#asterisk MoutaPT (i=MoutaPT@85.139.183.206) |
| 21:53.02 | Halshair | The checkgroup command seems to send me to hangup instead of n+101. Is that the way it is supposed to work? |
| 21:53.17 | }btorch{ | is there a channel for zaptel stuff ? |
| 21:53.36 | MoutaPT | Hello all, does any one ever tried VoIP voice Usb phone (usually sold for skype) with Sjphone? |
| 21:53.46 | FuriousGeorge | i found some setting once to test the ringer in this snom and now i cant locate it again |
| 21:53.47 | MoutaPT | i got audio ok, but no dialpad |
| 21:53.49 | *** part/#asterisk Halshair (i=Halshair@ool-457c7be0.dyn.optonline.net) |
| 21:53.59 | saftsack | gn8 |
| 21:54.00 | FuriousGeorge | in fact, the default ringer section i did find doesnt seem to do anything at all |
| 21:54.49 | [av]bani | http://video.google.com/videoplay?docid=6305488934079810351 |
| 21:55.21 | bbrdrg1 | Any ideas why Dial(SIP/peer-name,30,D(12345)) is not working ? it dials the peer, but never sends the DTMF according to D() option. Anyone ? please ? |
| 21:56.35 | *** join/#asterisk Halshair (i=Halshair@ool-457c7be0.dyn.optonline.net) |
| 21:57.18 | }btorch{ | is there a way to debug a digium card to see what is going on ? |
| 21:58.23 | *** join/#asterisk Jizzbug (n=derekm@199.227.154.26) |
| 22:00.22 | hensema | right, both isdn cards are generating around 8000 interrupts/sec, which is a bit much for an interface capable of just 36 KB/sec, isn't it? :/ |
| 22:00.27 | *** part/#asterisk austinnichols101 (n=austinni@70.46.69.130) |
| 22:02.22 | GoRK | bbrdrg: why dont you use SendDTMF after doing the dial? |
| 22:02.34 | GerbilWrk | ok, i've redone the macro with the default macro, and i'm still getting a dropped call when DND is pressed on a Snom 360, instead of getting voicemail, http://pastebin.com/556669 for examples |
| 22:03.04 | *** join/#asterisk Gunnar (n=gunnar@34.80-203-209.nextgentel.com) |
| 22:03.12 | X-Rob | GoRK, I know the reason why. Why don't you try to figure it out? Here's a hint: When does 'Dial' exit? |
| 22:03.33 | GoRK | hmm good point |
| 22:04.42 | X-Rob | [av]bani, fwor. |
| 22:04.49 | X-Rob | like, fwor. |
| 22:04.56 | bbrdrg1 | GoRK: i get a dialtone on remote channelbank and need to dial the actual number |
| 22:05.16 | GoRK | i agree it should work as documented |
| 22:05.28 | GoRK | it probalby only works on zap channels or something |
| 22:06.05 | GoRK | are you calling *from* a sip channel to this other sip channel? |
| 22:06.38 | bbrdrg1 | <PROTECTED> |
| 22:07.13 | GoRK | if so you might try canreinvite=no on the calling channel to make sure * the sip endpoints aren't just getting connected together |
| 22:08.08 | X-Rob | GerbilWrk, do a 'NoOp(Dialstatus is ${DIALSTATUS})' after the dial. That may give youa hint. |
| 22:09.40 | bbrdrg1 | GoRK: the end point ua doesn't accept 1234@peer-name, when it picks up it simply provides dialtone, nothing more |
| 22:14.06 | GerbilWrk | X-Rob, would that go in the macro? |
| 22:14.30 | X-Rob | GerbilWrk, after the Dial, put that. |
| 22:14.34 | X-Rob | It doesn't matter where it is. |
| 22:14.44 | *** join/#asterisk Eitch (n=hugo@unaffiliated/eitch) |
| 22:15.10 | X-Rob | (eg, I think your problem is dialstatus == CHANUNAVAIL, but you're not checking for that) |
| 22:17.38 | GerbilWrk | ok, i added it at priority 2, and got nothing different at the CLI |
| 22:17.51 | GerbilWrk | i also added, exten => s-CHANUNAVAIL,1,Voicemail(b${ARG1}) to the macro, and the same thing |
| 22:17.55 | GerbilWrk | happened |
| 22:20.29 | X-Rob | GerbilWrk, are you using asterisk 1.2? |
| 22:21.25 | GerbilWrk | well here's the interesting thing, i installed 1.2.4 i believe it was, and the CLI shows 1.0.10 when i reconnect to the server |
| 22:21.41 | X-Rob | you didn't install it properly then. Try again |
| 22:22.24 | FuriousGeorge | [av]bani: you still around |
| 22:22.45 | *** join/#asterisk terrapen (n=cjs@166.70.183.108) |
| 22:22.50 | terrapen | howdy |
| 22:22.51 | GerbilWrk | X-Rob, is there a way to completely uninstall it? |
| 22:22.53 | FuriousGeorge | or anyone that has played with the LEDs on the snom |
| 22:22.59 | FuriousGeorge | 320 or 360 |
| 22:23.06 | *** join/#asterisk X-Rob_ (n=rob-x@dsl-202-173-151-24.qld.westnet.com.au) |
| 22:23.17 | GerbilWrk | FuriousGeorge, i have them mostly working on a 360 |
| 22:23.29 | *** join/#asterisk Utah_Dav1 (n=boucha@0-2pool130-130.nas28.salt-lake-city1.ut.us.da.qwest.net) |
| 22:23.41 | terrapen | is there a good Asterisk jobs page? |
| 22:23.48 | terrapen | where I could post a job? |
| 22:24.18 | FuriousGeorge | GerbilWrk: i gotem working as destinations, but the lights just stay lit, they wont indicate any kind of status w/o bristuff, right? |
| 22:24.25 | *** join/#asterisk Zodiacal (i=1232321@bdsl.66.14.242.199.gte.net) |
| 22:25.11 | Zodiacal | anyone know why i keep getting "no load specified" when i try to load a sccp firmware to my cisco 7960 phone? i think i setup the tftp server files correctly.. but it just doesn't want to load... |
| 22:25.11 | GerbilWrk | i haven't used bristuff as far as I know, and i get status, but they occasionally turn on or off randomly |
| 22:25.22 | Zodiacal | do i have to load old versions or something before i load the latest version? |
| 22:25.26 | Zodiacal | right now it has sip |
| 22:25.32 | Zodiacal | if that maters |
| 22:25.34 | FuriousGeorge | GerbilWrk: you have them set as destinations with hints and thats it? |
| 22:26.09 | X-Rob | Gerbilwrk - http://pastebin.com/556714 is a 'better' 1.2 macro. |
| 22:26.12 | GerbilWrk | yeah |
| 22:27.14 | *** join/#asterisk JASON99 (n=jason@jason.unitz.ca) |
| 22:27.21 | JASON99 | <PROTECTED> |
| 22:27.42 | GerbilWrk | X-Rob, thanks, i'll be up here around midnight to reload Asterisk and will try the new macro |
| 22:27.58 | GerbilWrk | do you know of an easy way to completely uninstall Asterisk from a slackware box? |
| 22:31.30 | _Sam-- | JASON99 : there is a setting called dtmfmode on asterisk's confs... |
| 22:31.34 | _Sam-- | check it, learn it. |
| 22:31.44 | JASON99 | Thanks.. Tips are what I'm looking for ;) |
| 22:32.01 | FuriousGeorge | GerbilWrk: ok hint,${PEEREXTEN},SIP/Peername right? when does the status change do the have to answer |
| 22:32.03 | _Sam-- | if you are using sip clients, you can find it in sip.conf |
| 22:32.24 | JASON99 | Its a problem when the call is coming from the PSTN |
| 22:32.31 | FuriousGeorge | *, do they have to answer? shouldnt it indicate ringing? |
| 22:32.33 | JASON99 | and I have Asterisk connected to a PRI |
| 22:32.42 | GoRK | does anyone have the release notes to Polycom sip firmware 1.6.5? |
| 22:33.05 | GerbilWrk | FuriousGeorge, this is what i have, and they do have to answer, exten => 401,hint,SIP/401 |
| 22:33.14 | IronHelix | i wonder if they'll ever get a clue and stop making their firmware hard for their customers to get... |
| 22:34.01 | *** part/#asterisk ctooley (n=ctooley@rrcs-24-227-212-162.sw.biz.rr.com) |
| 22:34.24 | _Sam-- | JASON99: i dont know if there are any dtmf setting for zapata.conf |
| 22:34.26 | [av]bani | _Sam--: so your gxps are all happy with qualify=no ? |
| 22:34.28 | _Sam-- | im not a zap guy |
| 22:34.38 | _Sam-- | [av]bani : yep! |
| 22:34.41 | [av]bani | yay |
| 22:34.43 | *** join/#asterisk KranZ (n=user@imail.bestline.net) |
| 22:34.45 | _Sam-- | i left one gxp with qualify = yes |
| 22:34.47 | _Sam-- | and all the others are fine |
| 22:35.10 | X-Rob | FuriousGeorge, you can't use variables in a hint |
| 22:35.21 | FuriousGeorge | X-Rob: i know |
| 22:35.43 | X-Rob | <PROTECTED> |
| 22:35.46 | _Sam-- | JASON99 : i think are dtmf comments here about your zap: http://www.google.com/search?sourceid=navclient&ie=UTF-8&rls=GGLD,GGLD:2005-08,GGLD:en&q=zapata%2Econf+dtmf |
| 22:35.51 | _Sam-- | er |
| 22:35.54 | _Sam-- | here |
| 22:35.55 | _Sam-- | http://www.voip-info.org/wiki/index.php?page=Asterisk+config+zapata.conf |
| 22:35.57 | [av]bani | FuriousGeorge: so redial missed on snom 360 is ok for you? |
| 22:36.04 | *** join/#asterisk bkw_ (n=bkw_@adsl-70-142-51-127.dsl.tul2ok.sbcglobal.net) |
| 22:36.10 | FuriousGeorge | X-Rob: i wasa using the variable in the sense that that is what im gonna put there |
| 22:36.14 | *** part/#asterisk Cybertoy (n=maxim@dsl254-123-241.nyc1.dsl.speakeasy.net) |
| 22:36.43 | _Sam-- | JASON99 : note the part about "relaxdtmf" |
| 22:36.52 | _Sam-- | dont know if that will or will not be of any help. |
| 22:36.55 | _Sam-- | sounds like it |
| 22:38.07 | Hmmhesays | omg i'm laughing so hard i'm crying |
| 22:40.03 | X-Rob | Hmmhesays? |
| 22:40.45 | *** join/#asterisk drumkilla (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 22:40.45 | *** mode/#asterisk [+o drumkilla] by ChanServ |
| 22:41.02 | *** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239) |
| 22:41.04 | _Sam-- | grrr someone deleted my feature request from the GXP page! |
| 22:41.16 | *** join/#asterisk rogier (n=rogier@16-65-dsl.ipact.nl) |
| 22:41.16 | Hmmhesays | http://forums.fark.com/cgi/fark/comments.pl?IDLink=1913593 |
| 22:41.16 | *** topic/#asterisk by drumkilla -> Zaptel 1.2.4 Released ... More information available on http://www.asterisk.org |
| 22:41.23 | Hmmhesays | single best thread today |
| 22:41.24 | [av]bani | _Sam--: ?! |
| 22:41.33 | _Sam-- | they just f'd up the request above mine |
| 22:41.37 | _Sam-- | and made 1 paragraph |
| 22:41.40 | _Sam-- | out of like 3 things |
| 22:41.49 | _Sam-- | grrr |
| 22:41.52 | [av]bani | ?? |
| 22:42.05 | _Sam-- | search for this on the page : |
| 22:42.09 | _Sam-- | bklang |
| 22:42.12 | *** join/#asterisk FlyboySR22 (n=Richard@searsair-linksys.adnc.com) |
| 22:42.14 | *** part/#asterisk drumkilla (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 22:42.22 | _Sam-- | and you will see what that idiot did to my request :) |
| 22:42.30 | _Sam-- | i would fix it but i have to go home :) |
| 22:42.34 | JASON99 | Thanks Sam |
| 22:42.47 | _Sam-- | JASON99: sure, hope it helps |
| 22:42.48 | [av]bani | good lord, he borked thewhole page |
| 22:42.56 | justinu | lol |
| 22:43.41 | *** join/#asterisk Alric (n=nbowyer@masq.hyperusa.com) |
| 22:43.49 | [TK]D-Fender | :O |
| 22:43.59 | JASON99 | will a simple reload make asterisk reload the zapata.conf |
| 22:44.00 | JASON99 | ? |
| 22:44.31 | De_Mon | how do I increase asterisk log verbosity? |
| 22:44.38 | bcnl | -vvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvv |
| 22:44.43 | brif8 | when using NAT an * box behind a router/gateway which is better SIP or IAX ? |
| 22:44.44 | bcnl | :> |
| 22:44.50 | _Sam-- | heh how many is the max v's before it stops making a difference? |
| 22:44.53 | De_Mon | bcnl it doesn't fork if I do that |
| 22:44.57 | darkskiez | De_Mon: check logger.conf |
| 22:45.09 | De_Mon | _Sam-- I heard 10, but 3 seems more likely |
| 22:45.31 | X-Rob | brif8, use IAX. It'll still be painful, but it will be slightly less than using SIP. |
| 22:45.39 | [TK]D-Fender | brif8 : Both can work fine.... |
| 22:45.43 | _Sam-- | if you're just behind one single firewall SIP is fine |
| 22:45.54 | bcnl | why the hell wasn't SIP written with nat in mind |
| 22:46.08 | brif8 | ok thanks, SIP uses port 5060 and IAX 4569 what other ports need opened on the firewall ? |
| 22:46.17 | brif8 | or routed to the internal machine ? |
| 22:46.19 | [av]bani | fixed the page. you'll like the history comment |
| 22:46.22 | *** join/#asterisk Tuttle_ (n=Tuttle@kelinat210.keli.cz) |
| 22:46.28 | bcnl | brif8: if you have a static ip things will be easier as well |
| 22:46.37 | brif8 | bcnl: yes I do |
| 22:46.39 | bcnl | you can tell * the external IP and it'll use that in its outbound messages |
| 22:47.58 | Tuttle_ | Is there some advantage from having Asterisk (run on my internet server) connected somehow with my VoIP service provider (which I will use to reach PSTN/mobile networks anyway)? |
| 22:48.04 | brif8 | bcnl: care to explain more, or give me a page to read ? |
| 22:49.54 | *** part/#asterisk Alric (n=nbowyer@masq.hyperusa.com) |
| 22:49.59 | brif8 | bcnl: ooops yes I have the externip set in sip.conf |
| 22:50.16 | brif8 | but what ports do I need open on the firewall 5060 and what else ? |
| 22:50.28 | bcnl | brif8: http://www.voip-info.org/wiki/index.php?page=Asterisk+config+SIP.conf |
| 22:50.51 | bcnl | brif8: also look in rtp.conf |
| 22:51.00 | bcnl | those are the incomming audio ports (me thinks) |
| 22:51.18 | [TK]D-Fender | brif8 : typically 10000-20000 UDP for RTP and you're set |
| 22:52.05 | kuku5 | I need good origination - Nobody is answering :( |
| 22:52.21 | justinu | what have you tried? |
| 22:53.09 | *** join/#asterisk enemy^x (n=eqwrweqr@morpheus.dataguard.no) |
| 22:53.33 | JASON99 | Sam: That appeared to have helped a lot |
| 22:55.07 | De_Mon | Tuttle_ huh? an advantage over what? |
| 22:57.23 | FuriousGeorge | cool thats working now |
| 22:57.34 | FuriousGeorge | the only thing that appears borked at the moment is my ringtopnes |
| 22:57.47 | FuriousGeorge | changing the default ringtone doesnt do squat |
| 22:58.51 | FuriousGeorge | but it worked out of the box |
| 23:02.21 | *** join/#asterisk RoyK (n=roy@10.80-203-106.nextgentel.com) |
| 23:02.31 | }btorch{ | anyone knows how to use a manager E crap from siemens ? |
| 23:02.45 | *** join/#asterisk BladeRunner05 (n=feelme@adsl-ull-233-71.44-151.net24.it) |
| 23:02.53 | *** part/#asterisk techie (i=gus@antibala.com) |
| 23:02.54 | Abydos313 | just got the asterisk book by o'reilly sweeeet! |
| 23:03.39 | *** join/#asterisk elliot (n=elliot@rdu-nat.rpath.com) |
| 23:03.55 | *** join/#asterisk bjohnson (n=bjohnson@i216-58-67-128.cybersurf.com) |
| 23:04.37 | elliot | I'm running asterisk 1.0.9 and occationaly see floods of empty voicemails for a user. |
| 23:05.01 | shnarff | so how do you enter register and auth in realtime? i dont see cloumns that match these |
| 23:05.03 | elliot | It seems like they will get a call where the caller hangs up at the last minute and then they end up with ~25 voicemails |
| 23:05.31 | elliot | this has only happened three times, so I haven't had a change to debug |
| 23:05.37 | }btorch{ | ok I found out that my T1 card on the siemens box is setup using e&m signalling and it gets its timing from the PSTN .. also is setup to be T1 analog and it has 24 trunks created |
| 23:06.03 | Tuttle_ | De_Mon: I'm trying to gather an information about what can I do with Asterisk on my private internet server (I have some users - my friends - there). We can of course establish our own phone network. But is there some possible advantage in having * used by the group of friends in conjuction with out VoIP service providers? Me and my friends will in future all have our own VoIP-SP's... |
| 23:06.07 | }btorch{ | does that mean I need to create FXO within zapata.conf ? |
| 23:06.59 | Tuttle_ | De_Mon: advantage over - the case we just use our differenct VoIP-SP and make our calls via them. |
| 23:13.10 | De_Mon | Tuttle_ what happens if someone calls you from the VoIP-SP? |
| 23:13.39 | De_Mon | Tuttle_ if you use asterisk you can setup menus and extensions for everyone, conference rooms etc |
| 23:14.00 | De_Mon | If I'm running asterisk 1.0.7 what version of zaptel should I run? |
| 23:14.33 | *** join/#asterisk techie (i=gus@antibala.com) |
| 23:14.38 | Tuttle_ | de_mon: okay, me an my friends will have one proxy set as our * and the second as our SP's that will connect us to the rest of the world. right? |
| 23:15.00 | De_Mon | Tuttle_ the second what? |
| 23:15.16 | Tuttle_ | the second SIP proxy in our sw/hw phones. |
| 23:15.19 | De_Mon | you setup * and tell it who your SP is and tell * to dial out through them for outbound calls |
| 23:15.20 | Tuttle_ | (sorry) |
| 23:15.34 | *** join/#asterisk MGSsancho (n=user@adsl-67-121-107-88.dsl.irvnca.pacbell.net) |
| 23:16.09 | X-Rob | So has aonyone _used_ this astribank-8? |
| 23:16.40 | Tuttle_ | De_Mon: and the * also passes calls coming from the SP to my home phone (behind NAT possibly) when I have set this * as my only SIP proxy? |
| 23:16.59 | De_Mon | yep or goes to voicemail if your phone is off |
| 23:17.00 | shnarff | Where do Register and Auth fit into the sip_conf tables? Is it even possible to register with a providers service with ARA? |
| 23:17.24 | Tuttle_ | in other words, * registers at a given SP as being able to route calls to my own world-wide telephone number? |
| 23:17.31 | hertell | can anyone point me to some kind of howto for the sipura spa-3000 in how to config an incoming PSTN-call to asterisk? |
| 23:17.41 | De_Mon | Tuttle_ yep, just like any other SIP phone would |
| 23:17.52 | De_Mon | hertell did you look on voip-info? |
| 23:18.58 | Tuttle_ | De_Mon: without the SP ever knows or block this feature. some SP may restrict clients to be hw/sw phones to connect to them, not some noname proxies run by unknown administrator? |
| 23:19.00 | shnarff | Does and complete documentation exist for ARA? trust me id read it if i could find it -- the stuff on voip-info and asteriskguru just do not cover it |
| 23:19.04 | hertell | De_Mon: i have checked this http://www.voip-info.org/wiki-Sipura+3000 but for some reason the call is not received by asterisk.. |
| 23:19.18 | shnarff | any* |
| 23:19.38 | Abydos313 | hertell i personally got the pdf..it's got a full setup on spa3k and single line use |
| 23:20.16 | FuriousGeorge | can anyone with an snom try going into their firmare, setting the ringer to something esle and seeing if it actually changes |
| 23:20.22 | hertell | Abydos313: do you mean the pdf from sipura? |
| 23:20.22 | De_Mon | Tuttle_ asterisk is a sip device, your hw/sw phone is a sip device. there is no black magic involved with being an ASTERISK SERVER its just a sip device |
| 23:20.25 | FuriousGeorge | im wondering if this firmware i got borked it |
| 23:20.31 | Abydos313 | no |
| 23:20.33 | Abydos313 | http://members.optusnet.com.au/~bsharif/asterisk/ |
| 23:20.50 | De_Mon | hertell shrug, I use softphones |
| 23:21.11 | Abydos313 | De_Mon same here until i get my spa3k |
| 23:21.14 | justinu | ~seen jc |
| 23:21.18 | jbot | jc <n=jcw@adsl-065-006-151-062.sip.asm.bellsouth.net> was last seen on IRC in channel #classiccmp, 13d 23h 35m 14s ago, saying: 'OK, wife says it's dinner time. Back in a bit.'. |
| 23:21.26 | Tuttle_ | De_Mon: So the SP never distinguish between clients types - * proxying multiple phones / actual phone? |
| 23:21.28 | justinu | back in a bit... heh |
| 23:21.31 | brif8 | -- Executing Playback("IAX2/teliax-2", "pls-hold-while-try.gsm") in new stack |
| 23:21.33 | hertell | De_Mon: yeah, but i can hook my dect to this :-) |
| 23:21.37 | brif8 | Feb 15 18:20:33 WARNING[7796]: file.c:509 ast_openstream_full: File pls-hold-while-try.gsm does not exist in any format |
| 23:21.42 | brif8 | Why it exists ? |
| 23:22.15 | Juggie | remove .gsm |
| 23:22.26 | *** join/#asterisk Qwell[] (i=north@unaffiliated/qwell) |
| 23:22.37 | Abydos313 | Qwell! |
| 23:22.38 | *** part/#asterisk shnarff (n=whois@216.190.144.90) |
| 23:22.46 | Qwell[] | Abydos313 ? |
| 23:22.53 | Abydos313 | just saying hi |
| 23:23.07 | Abydos313 | got that book you suggested.. asterisk :) |
| 23:23.14 | Qwell[] | good :p |
| 23:23.28 | X-Rob | FuriousGeorge, it works. |
| 23:23.29 | Abydos313 | had to buy it, it's a pain to read on pc |
| 23:23.34 | X-Rob | I use snom's everywhere. |
| 23:23.39 | FuriousGeorge | X-Rob: right when you hit save you hear the ring right |
| 23:23.39 | FuriousGeorge | ? |
| 23:23.48 | FuriousGeorge | thats what mine used to do |
| 23:24.05 | X-Rob | FuriousGeorge, reset to factor and start again. They do get confused sometimes. |
| 23:25.03 | FuriousGeorge | X-Rob: sigh, i just got it set up so nice ;) |
| 23:25.18 | X-Rob | FuriousGeorge, well, wget http://ip.of.phone/settings.htm |
| 23:25.31 | X-Rob | then use that as a template to rebuild it. |
| 23:25.47 | FuriousGeorge | X-Rob: you da man, woot! |
| 23:25.49 | FuriousGeorge | ;) |
| 23:27.59 | *** join/#asterisk stkn (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 23:28.00 | }btorch{ | how come everytime I change from signalling type fxo_ks to fxs_ks ... zap show channel <num> tells me offhook/onhook ? |
| 23:28.20 | Qwell[] | }btorch{: Are you restarting * after changing signalling? |
| 23:28.33 | }btorch{ | yes |
| 23:29.37 | *** part/#asterisk darkskiez (n=darkskie@bb-87-81-62-203.ukonline.co.uk) |
| 23:30.01 | *** join/#asterisk techie (i=gus@antibala.com) |
| 23:31.01 | wunderkin | brif8: remove the .gsm extension from the playback options |
| 23:31.23 | Tuttle_ | Having * doing a SIP proxy for me and my friends, our SP does not distinguish * from our home VoIP phones (if i got this correctly). Is there any possibility and advantage if the SP's software communicate with our private * in some advanced manner than just SIP? (Let's say the SP uses * too.) |
| 23:32.01 | brif8 | thanks it works I also had to have answer |
| 23:32.42 | FlyboySR22 | Tuttle_, If your SP uses *, talk IAX |
| 23:32.57 | Qwell[] | FlyboySR22: Why? |
| 23:33.21 | Tuttle_ | FlyboySR22: I'm asking for advantages and whether the SP are often offer this. |
| 23:33.22 | FlyboySR22 | Qwell, mostly quality and less issues with * behind firewalls. |
| 23:33.26 | }btorch{ | this sucks the damn te110 xp light won't turn green |
| 23:33.28 | Qwell[] | quality? |
| 23:33.44 | *** join/#asterisk doug (i=doug@zaxxon.telerama.com) |
| 23:33.47 | Qwell[] | FlyboySR22: You're gonna have to explain that one to me |
| 23:33.50 | FlyboySR22 | Tuttle_, I use several SPs and they use (and prefer) IAX to SIP |
| 23:34.39 | doug | hey |
| 23:34.41 | doug | Feb 15 17:30:40 NOTICE[756293648]: 66.45.41.57 tried to authenticate with non-existant user 'user' |
| 23:34.45 | doug | i get that every 10 seconds |
| 23:34.49 | doug | any clue on how to track that down? |
| 23:35.00 | Qwell[] | doug: Tell 66.45.41.57 to stop trying to auth as 'user' |
| 23:35.01 | X-Rob | }btorch{, ring or email digium. They give support. |
| 23:35.04 | Qwell[] | put in the real username |
| 23:35.45 | Tuttle_ | Are all SPs successful in delivering phone calls to SIP clients behind NAT? |
| 23:35.57 | }btorch{ | X-Rob: I have talked to them already ... they told me I had to configure my span timing which I thought I did but nothing |
| 23:36.27 | X-Rob | }btorch{, what country? |
| 23:36.39 | }btorch{ | US |
| 23:36.50 | X-Rob | ring 'em back. Tell 'em it's not working. |
| 23:37.08 | X-Rob | they can definately help you 8) |
| 23:37.14 | doug | um |
| 23:37.15 | doug | well |
| 23:37.23 | doug | 66.45.41.57 is actually the asterisk server |
| 23:37.31 | doug | something is attempting to log in every 10 seconds as "user" |
| 23:37.35 | }btorch{ | I'll try that again i guess |
| 23:37.36 | doug | my guess is it's something i did |
| 23:37.50 | X-Rob | doug, grep user /etc/asterisk/sip.conf |
| 23:38.07 | doug | there's no sip user "user" |
| 23:38.09 | Abydos313 | i have nat=yes in sip_additional.conf and when i query * it still shows and Nat=N |
| 23:38.18 | doug | and nothing in the crontab triggering every 10 seconds |
| 23:38.30 | X-Rob | Abydos313, you're using AMP. You shouldn't be fucking with the sip_additional.conf. |
| 23:38.32 | doug | and i can't remember setting up anything to do that |
| 23:38.35 | X-Rob | use the web interface. |
| 23:39.18 | Abydos313 | X-Rob yes i checked them after i put settings in the changes exist in file, but when i bring up * info it show Nat=N |
| 23:39.30 | Abydos313 | i used amp to change the settings |
| 23:39.40 | FlyboySR22 | Qwell, Sorry - phone call. We run around 15 or 20 Asterisk systems all over the country with a variety of providers behind a variety of firewalls. We had many, many problems getting SIP based coonnections to work as reliabily as the IAX trunks, so we always default to IAX no matter what, I no longer have the issues I was having with SIP trunks. |
| 23:39.43 | X-Rob | Abydos313, you're not listening to me. Don't edit the file. Use the web interface. Click on the extension, then change 'nat' to 'yes' |
| 23:39.49 | X-Rob | Then click 'apply changes' |
| 23:39.51 | X-Rob | then it'll work. |
| 23:40.12 | Abydos313 | exactly what i did |
| 23:40.13 | doug | is there a nice asterisk web interface? |
| 23:40.17 | X-Rob | If it doesn't work THEN, then ask on #amportal |
| 23:40.21 | X-Rob | doug, AMP. |
| 23:40.52 | X-Rob | Doug: http://www.coalescentsystems.ca/index.php?option=com_content&task=view&id=31&Itemid=57 |
| 23:40.54 | doug | got a url to it? |
| 23:40.55 | doug | tanks |
| 23:40.57 | Abydos313 | i looked at file after i made changes.. the settings took, i even rebooted but the * info window still shows them as nat=n |
| 23:41.06 | X-Rob | '* info window'? |
| 23:41.28 | X-Rob | you're typing 'sip show peers' right? |
| 23:41.31 | *** join/#asterisk Derkommissar (n=Alberto@adsl-153-235-135.mia.bellsouth.net) |
| 23:41.34 | Derkommissar | Hello |
| 23:41.35 | Abydos313 | on the maintenace page |
| 23:41.39 | _Sam-- | hey justinu: all the sangoma devices for fxo are configured exactly the same as digium cards? |
| 23:41.49 | Abydos313 | no i wasn't doing it from cli |
| 23:42.02 | hertell | darn... |
| 23:42.10 | Derkommissar | sometimes afther a couple of hours of working,,,, the agi scripts stop working and i get this error |
| 23:42.12 | Derkommissar | Feb 15 16:10:33 WARNING[16925]: res_agi.c:259 launch_script: Failed to fork(): Cannot allocate memory |
| 23:42.18 | justinu | _Sam--: yeah, it's all done thru zaptel.conf and zapata.conf still |
| 23:42.22 | Derkommissar | what is this suposed to mean ? |
| 23:42.25 | X-Rob | Derkommissar, upgrade to asterisk 1.2.4 |
| 23:42.33 | Derkommissar | what kind of memory is it talking about ? |
| 23:42.40 | hertell | how on earth is the spa-3000 firmware upgraded...? I don't run windows!! ;-( |
| 23:43.02 | X-Rob | It's talking about memory memory. You're running out of it. |
| 23:43.05 | doug | wow, AMP has a lot of prereqs |
| 23:43.10 | Derkommissar | Ram ? |
| 23:43.14 | Qwell[] | doug: Such as not having a brain... |
| 23:43.16 | doug | is AMP worth it? |
| 23:43.19 | X-Rob | Doug, download 'Asterisk@Home', it does it all for you. |
| 23:43.20 | Qwell[] | ~amp |
| 23:43.24 | jbot | hmm... amp is NOT supported here! people using it should join #amportal |
| 23:43.24 | _Sam-- | justinu: you've used the A2000*s w/ ec? |
| 23:43.33 | Derkommissar | x-rob Ram ? |
| 23:43.47 | Abydos313 | under column Nat it still shows N.. i ran sip show peers |
| 23:43.48 | Derkommissar | why upgrade ? was this a bug ? |
| 23:43.49 | justinu | _Sam--: nope, just the digital boards |
| 23:43.58 | X-Rob | Derkommissar, what other sort of memory is there? (*puzzled*) |
| 23:44.04 | justinu | A2000 brings back some old Amiga memories tho |
| 23:44.07 | doug | which is better, amp or asterisk@home? |
| 23:44.09 | Abydos313 | 202/202 70.237.101.231 D N 5060 OK (25 ms) |
| 23:44.10 | Abydos313 | 201/201 192.168.2.134 D N 5060 OK (5 ms) |
| 23:44.17 | Abydos313 | sorry |
| 23:44.26 | X-Rob | doug, a@h _uses_ amp. It just does all the prerequisites for you. |
| 23:44.27 | doug | asterisk@home is distributed as an iso? that's a little worrisome |
| 23:44.30 | _Sam-- | are those card avail? |
| 23:44.31 | doug | ah |
| 23:44.38 | _Sam-- | says at voipsupply they are taking pre-orders |
| 23:44.39 | justinu | _Sam--: i don't think they're shipping yet |
| 23:44.54 | X-Rob | Doug, A@H is CentOS 4.2 + AMP + asterisk + other bits. |
| 23:44.57 | doug | that kind of implies that a@h takes over the entire machine |
| 23:45.02 | Qwell[] | doug: It also blows away anything on your drive...without asking. :) |
| 23:45.16 | doug | ah. |
| 23:45.19 | doug | not sure that's what i want. |
| 23:45.32 | X-Rob | Qwell[], well it does assume you're not retarded. |
| 23:45.38 | doug | sounds like it has a lot of goodies, otherwise. |
| 23:45.40 | X-Rob | which is, possibly, a foolish assumption to make. |
| 23:45.50 | *** part/#asterisk zaf (n=tfournet@wsip-68-228-9-79.br.br.cox.net) |
| 23:46.14 | doug | never heard of centos |
| 23:46.26 | Qwell[] | ~centos |
| 23:46.28 | jbot | i heard centos is better than Fedora Core |
| 23:46.29 | X-Rob | doug, if you're competent in linux, youc an install amp and all it's prerequisites in about 45 mins with a reasonable speed net connection |
| 23:46.43 | X-Rob | CentOS is Red Hat Enterprise Linux, basically, without the redhat bit. |
| 23:46.51 | doug | centos kinda looks like an uppity linux distro |
| 23:47.07 | Abydos313 | centos is redhat enterprise edition basically |
| 23:47.09 | doug | well, net i've got. |
| 23:47.34 | KranZ | amp's a good start if you're looking to design a web frontend for users |
| 23:47.38 | doug | i'm not as clued with linux as some, being a freebsd lifer |
| 23:47.43 | tuxinator_linux | Abydos313: not basically, more exactly, minus Redhat lable |
| 23:47.59 | doug | i do have a couple of OC-3's |
| 23:48.01 | Abydos313 | yeah you're right |
| 23:48.13 | Abydos313 | we run oracle on centos at work, runs fine |
| 23:48.22 | doug | what's the quickest way to install, say, libxml2 on linux? |
| 23:48.26 | tuxinator_linux | I use is also, only with Debian |
| 23:48.31 | doug | not compile from source, i trust. |
| 23:48.37 | tuxinator_linux | s/only/onlong |
| 23:48.43 | [av]bani | a200 not a2000 |
| 23:48.47 | Qwell[] | doug: Using gentoo? source, yes :p |
| 23:48.49 | doug | there must be some semi-automated way for finding the rpm and installing it |
| 23:48.50 | KranZ | who's a gentoo user? |
| 23:48.57 | malverian[work] | Hmmmmm... |
| 23:48.58 | doug | no no |
| 23:48.58 | KranZ | emerge libxml2 |
| 23:48.58 | doug | Linux asterisk.aaronsen.com 2.4.18-14rlx2 #1 Sat Nov 2 02:00:23 CST 2002 i686 i686 i386 GNU/Linux |
| 23:49.01 | doug | not gentoo |
| 23:49.06 | X-Rob | doug, depends on your net and cpu speed. centos 'yum install libxml2' installs the package, debian uses apt-get, mandrake uses 'urpmi libxml2' |
| 23:49.15 | X-Rob | various distros have various installation procedures |
| 23:49.27 | doug | RH8.0 for me |
| 23:49.32 | doug | which i never use |
| 23:49.33 | malverian[work] | If I use the M() option to Dial to call a macro, why can't I see the ${DIALEDNUMBER} channel var? |
| 23:49.45 | doug | i do have apt-get |
| 23:50.05 | X-Rob | doug, serously, download A@home. it'll save you hours of work, and you'll have a working, reasonably up to date, base system |
| 23:50.07 | doug | apt-get install libxml2 ? |
| 23:50.19 | KranZ | doug try it |
| 23:50.43 | doug | x-rob, the install i've got is a couple of years going now, with alot of local configs |
| 23:50.51 | doug | i'd like to keep what we've done |
| 23:51.01 | X-Rob | ok, so spend $200 and get a new machine. |
| 23:51.05 | doug | a@h looks pretty committing, leaving everything behind. |
| 23:51.16 | doug | well |
| 23:51.19 | malverian[work] | Here's a better question... |
| 23:51.30 | doug | i'll give myself a couple of hours and reconsider a@h if i suck at this |
| 23:51.34 | KranZ | doug, if you got a spare box, try it |
| 23:51.51 | KranZ | take a look behind the scenes and you'll learn some stuff you probably didnt know |
| 23:52.00 | malverian[work] | If I use Dial to call multiple channels, eg. "Dial(SIP/101&SIP/102,,M(somemacro))" is there any way for me to see which channel actually picked up the call? |
| 23:52.03 | doug | i'm remote, though... |
| 23:52.07 | *** join/#asterisk tbs_ (n=ingen@fw.sg12.dk) |
| 23:52.10 | doug | 2000 miles from my box(es) |
| 23:52.20 | *** join/#asterisk Cresl1n (n=matt@gateway.digium.com) |
| 23:52.21 | malverian[work] | From [macro-somemacro] that is. |
| 23:52.22 | doug | and a@h looks like it would be easier if i were on site |
| 23:52.26 | KranZ | you're not at home? |
| 23:52.31 | KranZ | no pun |
| 23:52.37 | Abydos313 | heh |
| 23:52.38 | Qwell[] | asterisk@datacenter |
| 23:52.49 | tbs_ | Hi there, everyone... I'm sitting here with Skumling, struggling with fax-issues... anybody up for the task of trying to send a fax or two to a Danish landline? |
| 23:53.26 | tbs_ | The main issue is, well, that it works fine for the 3-4 persons who's tried, but not the 5th (who |
| 23:53.39 | tbs_ | (who's using an HP OfficeJet G95) |
| 23:53.44 | doug | maybe i'll just delegate this |
| 23:53.49 | KranZ | tbs_: that's just how stable faxing is |
| 23:53.52 | doug | anyone wanna earn an easy $50? |
| 23:54.11 | Qwell[] | $50...to install *@~? |
| 23:54.14 | *** join/#asterisk danzig (n=chatzill@130.226.169.177) |
| 23:54.28 | doug | i assume it'll take about 20 minutes of actual attention for someone who's experienced. |
| 23:54.31 | KranZ | doug: we're not your whores |
| 23:54.42 | danzig | EHLO * * gurus |
| 23:54.43 | Abydos313 | comedy hour in here |
| 23:54.55 | KranZ | next show at 7pm cst |
| 23:54.56 | justinu | lol |
| 23:54.59 | doug | h, not sure you speak for everyone, kranz... |
| 23:55.04 | tbs_ | KranZ: if it wasn't for the fact that we've tried it a brazillion times with the HP G95, I would agree |
| 23:55.28 | KranZ | tbs_: some new faxes are able to negociate at rates higher than 9600 |
| 23:55.35 | *** part/#asterisk elliot (n=elliot@rdu-nat.rpath.com) |
| 23:55.40 | KranZ | try setting the baud to 9600 on that machine |
| 23:55.42 | KranZ | or less |
| 23:56.03 | X-Rob | doug, US$60/hour and I'll happily install AMP on a resonably current machine. (EG, something that uses 2.6 kernel and udev) |
| 23:56.05 | tbs_ | sorry, nocando... |
| 23:56.26 | tbs_ | KranZ: erhm but why can't asterisk handle more than 9600 bps? |
| 23:56.37 | doug | if it only takes 45 minutes, i'm offering more than $60/hour... |
| 23:56.52 | doug | unfortunately, that 2.4 box is what i've got.. |
| 23:57.02 | doug | i'll come back if the first hit doesn't pan out, thanks. |
| 23:57.04 | KranZ | tbs_: asterisk can, but your latency probably cant |
| 23:57.11 | X-Rob | You're offering $50/hour for about 6 hours work. |
| 23:57.20 | KranZ | and the fax standard is 9600 |
| 23:57.34 | doug | > youc an install amp and all it's prerequisites in about 45 mins witha reasonable speed net connection |
| 23:57.37 | doug | didn't someone say that? |
| 23:57.41 | X-Rob | And I'm not really interested in wasting a day beating an ancient rh8 machine up to date enough to run php5 |
| 23:57.48 | KranZ | anything higher pushes the limits of g711 |
| 23:58.13 | X-Rob | doug, I did. With a current machine. Not when I'd have to upgrade _everything_, reverse engineer all your changes, upgrade them to the current stuff, etc etc. |
| 23:58.19 | Himeko | tbs_ you are using g711 right? |
| 23:58.40 | *** join/#asterisk stkn (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 23:58.42 | tbs_ | Himeko: it does not pass through SIP... |
| 23:58.54 | tbs_ | Himeko: it's ISDN/zaptel |
| 23:59.00 | tbs_ | all the way |
| 23:59.03 | KranZ | no voip? |
| 23:59.07 | tbs_ | nope |
| 23:59.10 | KranZ | well then |
| 23:59.13 | KranZ | diff story |
| 23:59.14 | tbs_ | (or FOIP) |
| 23:59.51 | tbs_ | KranZ: okay? Tell me what's wrong, then ;-) |
