| 00:00.04 | *** part/#asterisk zedkatuf (n=knoppix@82-33-115-145.cable.ubr08.azte.blueyonder.co.uk) |
| 00:01.22 | *** join/#asterisk psiud (n=not@S01060004e2c9d879.wp.shawcable.net) |
| 00:02.02 | psiud | question about the following |
| 00:02.11 | psiud | exten => faxout,1,Dial((Zap/1/XXXXXX) |
| 00:02.11 | psiud | exten => faxout,2,txfax(/var/spool/asterisk/faxout/sample.tif|caller) |
| 00:02.14 | *** join/#asterisk [hC] (n=hardcore@8.10.2.40) |
| 00:02.16 | *** join/#asterisk oplog2 (n=oplog2@206.222.29.50) |
| 00:02.31 | psiud | sorry should have been this |
| 00:02.39 | psiud | exten => faxout,1,Dial(Zap/1/XXXXXX) |
| 00:02.39 | psiud | exten => faxout,2,txfax(/var/spool/asterisk/faxout/sample.tif|caller) |
| 00:03.44 | psiud | I excute this buy *26 ... it dials out fine but I am unable to get it go to the faxout,2 when the line is pickuped at the other end |
| 00:03.49 | [hC] | Has anyone come across an issue with CDR storage where over a PRI ( Maybe more, i only have pri and voip to test with) sometimes the callerID that gets saved in a cdr record isnt correct, its a previous caller's number (even though asterisk's console and ${CALLERIDNUM} are correct? |
| 00:03.59 | psiud | what am I missing |
| 00:05.07 | bkw_ | psiud, use a call file |
| 00:05.09 | bkw_ | not the dialplan |
| 00:05.23 | bkw_ | Zap/1/blah = faxout@context |
| 00:05.29 | bkw_ | er s/=/=>/ |
| 00:05.35 | bkw_ | go look at sample.call |
| 00:06.48 | psiud | I looked at that .. except I would like to make the call have a variable number that is dialed ... ie. always faxing the same sheet except to different numbers |
| 00:07.11 | *** join/#asterisk MRH2 (n=Mr_happy@fcirc-adsl.demon.co.uk) |
| 00:07.23 | *** part/#asterisk twisted[work] (n=twisted@asterisk/friend-and-developer/pdpc.professional.twisted) |
| 00:07.35 | *** part/#asterisk zedkatuf2 (n=audela@82-33-115-145.cable.ubr08.azte.blueyonder.co.uk) |
| 00:07.40 | *** join/#asterisk Berkey (n=berkey@las-cust-208.57.3.251.mpowercom.net) |
| 00:08.33 | bkw_ | then drop call files that differ |
| 00:08.38 | bkw_ | on the Zap/1/blah side |
| 00:08.41 | bkw_ | its not too hard to do |
| 00:09.08 | psiud | hmm |
| 00:09.09 | tsume | ring god damn it, ring! |
| 00:10.31 | tsume | sob, when I get home I'm going to maje the best fucking admin program in the world for asterisk |
| 00:10.38 | tsume | I'm not goin gto deal with any more days like this |
| 00:10.49 | psiud | lol |
| 00:10.54 | bkw_ | tsume, what rm -rf ? |
| 00:11.09 | tsume | bkw_: snom trouble |
| 00:11.15 | bkw_ | ah |
| 00:11.16 | bkw_ | IC |
| 00:11.20 | tsume | *ARRGGGG |
| 00:11.24 | tsume | *stabs snom* |
| 00:11.47 | *** join/#asterisk CoolAcid (n=jason@216.99.98.39) |
| 00:12.05 | tsume | is there a way to verify what phone queues are getting routered to? |
| 00:12.16 | tsume | basically I mean the groups the calls some in for a queue |
| 00:12.32 | tsume | I didn't set this system up, but I'm going to make it easy for me for now on |
| 00:13.08 | tsume | maybe I need to reboot asterisk, it seems to have gone stupid |
| 00:13.28 | Vco | yea..the computer has gone stupid... |
| 00:13.56 | tsume | was working yesterday :) |
| 00:13.58 | *** join/#asterisk cgcorea (n=cgcorea@63.245.14.194) |
| 00:14.04 | tsume | it magically stoped |
| 00:14.12 | tsume | like some body entered the room and reboot the server |
| 00:15.22 | *** join/#asterisk R3DB0x (i=nobody@66.142.28.36) |
| 00:18.07 | *** join/#asterisk zotz (n=zotz@24.231.36.100) |
| 00:18.20 | *** join/#asterisk shido6 (n=shido6@d57-87-253.home.cgocable.net) |
| 00:18.50 | *** join/#asterisk gordonb (i=gordon@206.138.224.74) |
| 00:19.52 | *** join/#asterisk bowman (n=bowman@195.46.47.202.static.cablesurf.de) |
| 00:20.09 | bowman | heya. does gnophone still work with asterisk after the switch to IAX2? |
| 00:21.30 | psiud | bkw_ |
| 00:21.30 | bkw_ | the question should be does gnophone even work at all |
| 00:21.31 | psiud | exten=> *26,1,Dial(Zap/1/9873000) = faxout@dialoutfax |
| 00:21.31 | psiud | [dialoutfax] |
| 00:21.31 | psiud | exten => faxout,1,txfax(/var/spool/asterisk/faxout/test.tif|caller) |
| 00:21.40 | bkw_ | take the callerpart off txfax |
| 00:21.46 | bkw_ | if you're sending a fax |
| 00:23.09 | bkw_ | adn NO |
| 00:23.12 | bkw_ | ninny boi |
| 00:23.23 | bkw_ | exten=> *26,1,Dial(Zap/1/9873000) = faxout@dialoutfax <== NOT VALID |
| 00:23.27 | bkw_ | you MUST use a call file |
| 00:23.29 | bowman | bkw_: right. I have never seen gnophone working. :) |
| 00:23.47 | bowman | by the way I still have trouble with kiax, anyone in here using it? |
| 00:24.02 | Darwin35 | kiax ? |
| 00:24.03 | shido6 | dont use startup disk in osx intel |
| 00:24.10 | shido6 | unless you have mad grub skills |
| 00:24.12 | *** join/#asterisk da_monumental_1 (n=da_monum@rrcs-24-172-102-198.midsouth.biz.rr.com) |
| 00:24.25 | bowman | Darwin35: an IAX softphone for linux |
| 00:24.37 | bkw_ | psiud, |
| 00:24.38 | bkw_ | here |
| 00:24.39 | bkw_ | Channel: Zap/1/4000 |
| 00:24.39 | bkw_ | Context: dialoutfax |
| 00:24.39 | bkw_ | Extension: faxout |
| 00:24.42 | bkw_ | Priority: 1 |
| 00:24.46 | bkw_ | those 4 lines are all you need in your call file |
| 00:24.57 | bkw_ | cp it /var/spool/asterisk/outgoing |
| 00:25.01 | bkw_ | change the zap to be what you want |
| 00:25.12 | bkw_ | you can also do |
| 00:25.37 | bkw_ | Application: TxFax |
| 00:25.44 | bkw_ | Data: /var/spool/asterisk/faxout/test.tif |
| 00:25.53 | bkw_ | instead of context,exten,pri |
| 00:27.01 | *** join/#asterisk stkn (i=nobody@gentoo/developer/pdpc.active.stkn) |
| 00:27.37 | psiud | would not the .call file be deleted though ? having to place it into /spool/outgoing all the time ? |
| 00:28.02 | Darwin35 | wow |
| 00:28.07 | Darwin35 | it works on fbsd |
| 00:28.14 | Darwin35 | I just compiled it |
| 00:28.19 | Darwin35 | no issue |
| 00:28.27 | Darwin35 | ok I have get this into ports |
| 00:29.37 | *** join/#asterisk Netgeeks_ (n=Chris@68-185-24-2.static.mdfd.or.charter.com) |
| 00:30.02 | FuzzyCat | *blink* |
| 00:31.41 | Netgeeks_ | ... |
| 00:32.05 | Netgeeks_ | every single time I have to work with MySQL I have to take exedrin |
| 00:32.53 | FuzzyCat | use pgsql then :) |
| 00:33.00 | Netgeeks_ | That won't help |
| 00:33.19 | Netgeeks_ | Foreign keys, and cascading updates and deletes scramble my brain |
| 00:33.34 | Netgeeks_ | so I should say every time I have to work with a db |
| 00:34.02 | bkw_ | psiud, ues |
| 00:34.12 | bkw_ | psiud, thats the only way you're going to link a call to the app |
| 00:34.19 | bkw_ | unless you use action originate from the manager |
| 00:34.30 | Netgeeks_ | okay |
| 00:34.33 | psiud | hmm |
| 00:34.42 | FuzzyCat | ehehe..I remember doing an emergency provisioning system using perl and mysql at 4am one day, they were shipping the phones out the door as they were done... |
| 00:35.56 | Rowter | 505 (505) available at '503' (musiconhold is 'default') |
| 00:36.03 | Rowter | what does aviable at means_ |
| 00:36.10 | Netgeeks_ | I never ever can remember where the cascade constraint goes, and I invariably get it wrong |
| 00:36.53 | *** join/#asterisk shido6 (n=shido6@d57-87-253.home.cgocable.net) |
| 00:37.11 | psiud | bkw_ ... can that be done with in the extensions ? |
| 00:37.40 | *** join/#asterisk snewpy (n=markl@203-217-78-139.dyn.iinet.net.au) |
| 00:39.17 | psiud | s,1,system('asterisk -rv ..... ? |
| 00:39.54 | *** join/#asterisk shmaltz (n=chatzill@69.28.255.210) |
| 00:41.06 | shmaltz | helo every1 |
| 00:42.20 | pygrammer | helo |
| 00:43.06 | shmaltz | I just filled up my tank, $44.01 a week ago I paid $33.00 what a jump |
| 00:43.22 | wunderkin | next week it will be 55.00 |
| 00:43.40 | shmaltz | who knows |
| 00:43.46 | shmaltz | wunderkin, where do you live? |
| 00:43.52 | wunderkin | phx |
| 00:45.02 | litage | since asterisk 1.2.0-beta1 has just been released, is there any idea when 1.2.0 (stable) will be released? by 2006 maybe? |
| 00:45.33 | shmaltz | litage, nah much earlier I believe |
| 00:45.51 | shmaltz | wunderkin, how much is a gallon in phx now? |
| 00:46.02 | *** join/#asterisk kFuQ (n=somedude@c-67-185-114-199.hsd1.wa.comcast.net) |
| 00:46.38 | SwK | gas was ~2.70 to 2.80 on my way home in HUntsville |
| 00:46.50 | SwK | per gal that is |
| 00:46.51 | wunderkin | i havent been out for a couple of days, was 2.63 |
| 00:47.08 | SwK | it was $2.40 this morning |
| 00:47.42 | tsume | okay 500 is log on yes? |
| 00:47.43 | Vco | "haven't been out for a couple of days".....? |
| 00:47.46 | shmaltz | $3.01 a few minutes ago, but I use 93, 87 was 2.83 |
| 00:47.51 | tsume | does the password go after it, or what? |
| 00:48.19 | tsume | does anyone know why a phone would repeatedly beep as well :) |
| 00:48.39 | gek | meep |
| 00:48.46 | shmaltz | tsume, what phone? |
| 00:48.54 | tsume | snom 220 |
| 00:49.16 | tsume | this pos is pissing me off |
| 00:49.20 | tsume | wont register to agents |
| 00:49.42 | *** join/#asterisk Sanjana (n=sbhogale@staff.starnetusa.net) |
| 00:50.12 | tsume | 500*101 means login with password, yes/ |
| 00:50.19 | tsume | I'm just giving it a guess here |
| 00:52.19 | *** part/#asterisk Sanjana (n=sbhogale@staff.starnetusa.net) |
| 00:53.33 | *** join/#asterisk mark_mia (n=mark__mi@24-55-227-126.miamfl.adelphia.net) |
| 00:55.38 | *** join/#asterisk twitchmax (n=twitch@c-24-98-56-147.hsd1.ga.comcast.net) |
| 00:56.04 | twitchmax | greetings all |
| 00:56.17 | mark_mia | greetings |
| 00:57.13 | twitchmax | i was wondering if anyone could help me build my outbound routes to my h323 gatekeeper |
| 00:58.10 | mark_mia | what seems to be the problem? |
| 00:58.52 | twitchmax | lack of docs, everything i can find related to h323 is for using like an ata or something... not as the route to the ptsn |
| 00:59.04 | *** join/#asterisk Reltsuh (n=Takeshi@S010600095b10aecd.ed.shawcable.net) |
| 00:59.17 | mark_mia | have u researched the voip wiki for reference? |
| 00:59.45 | mark_mia | I believe a lot of people are using h.323 to trunk to Cisco equipment |
| 00:59.56 | twitchmax | wish i could find a howto.... |
| 00:59.58 | *** join/#asterisk ManxPower (i=eric@1Cust5758.an1.dfw28.da.uu.net) |
| 01:00.19 | mark_mia | what is it that u exactly want to achieve? |
| 01:02.17 | twitchmax | i have an asterisk box at home, an as580 in ny, and an as5850 and as5300 in santiago, using a sysmaster gatekeeper, i want to be able to send test calls from my home, thru the gk, and out the corrosponding gw |
| 01:02.41 | *** join/#asterisk Rowters (n=SilverDr@dsl-201-129-89-96.prod-infinitum.com.mx) |
| 01:03.38 | mark_mia | so, you want to create h.323 trunks to each of your as5xx in NY and SCL? |
| 01:03.39 | Katty | hi. |
| 01:03.58 | *** join/#asterisk kiwnix (n=egarcia@82.158.154.119) |
| 01:04.02 | twitchmax | or just the as5300 |
| 01:04.26 | mark_mia | I'm pretty sure there is a sample of how to do that on the voip wiki |
| 01:04.28 | twitchmax | once i get one built, the rest should be pretty much the same, correct? |
| 01:04.50 | *** join/#asterisk Borgon (n=L3orgon@67-50-80-66.br1.tbr.ga.frontiernet.net) |
| 01:04.52 | mark_mia | I would think so |
| 01:04.59 | Borgon | Hello is anyone familiar with iaxcomm? |
| 01:05.07 | twitchmax | a bit, whats up? |
| 01:05.20 | Borgon | When making a call, the operator ask me to push numbers.. i try to put the numbers in but its not being accepted |
| 01:05.37 | bowman | iaxcomm doesnt run on my box :) |
| 01:05.56 | Borgon | i would click on the active line, then in the exentsion side i type in the numbers i want.. but its not being recognized |
| 01:06.08 | twitchmax | strange |
| 01:06.22 | Borgon | Recommend another good iax phone? |
| 01:06.33 | Borgon | softphone |
| 01:06.34 | bowman | anyone ever checked out iaxcomm from cvs? |
| 01:06.41 | mark_mia | twitchmax try the following link... |
| 01:06.43 | mark_mia | http://ertw.com/blog/archives/asterisk_and_an_as5350_sip_peer-190405.html |
| 01:06.48 | bowman | Borgon: try kiax, I hate it, but it seems to work |
| 01:07.19 | mark_mia | Firefox has an IAX client plugin that though very Spartan works well |
| 01:07.43 | hugo-v6 | puh. 3am and im still awake :/ |
| 01:07.55 | hugo-v6 | but to much damn good doc. sources about asterisk to read |
| 01:08.53 | mark_mia | 3am? Europe...Spain? |
| 01:08.54 | twitchmax | yeah, i've seen lots of stuff building sip trunks, the problem is that it has to hit the as5350 h323, or the gk will ignore it |
| 01:09.15 | *** join/#asterisk Snake-Eyes (n=blog@203.201.97.169) |
| 01:09.15 | hugo-v6 | mark_mia: europe/de |
| 01:09.39 | bowman | mark_mia: ffx plugin? where? |
| 01:09.41 | *** join/#asterisk litage (n=nick@203.201.97.169) |
| 01:09.46 | twitchmax | is there a way i could make it go into the 5350 sip and it send it to the gk h323? |
| 01:09.51 | *** join/#asterisk ManxPwr (n=eric@1Cust2433.an7.dfw28.da.uu.net) |
| 01:10.06 | hugo-v6 | hiho ManxPwr |
| 01:10.07 | Borgon | This KIAX softphone can work only with Linux OS |
| 01:10.15 | mark_mia | MozIAX I believe it is called....check out the Firefox site |
| 01:10.25 | Borgon | bowman: any windows iax softphone? |
| 01:10.46 | bowman | Borgon: no idea, check voip-info.org, there is a wiki page about voip phones :) |
| 01:10.50 | hugo-v6 | Borgon: iirc yes. look at the softphone section on voip-info.org |
| 01:11.01 | Borgon | k thx |
| 01:11.01 | mark_mia | I use the Firefox plugin on both WinXP and Ubuntu..works very nice |
| 01:11.30 | mark_mia | twitchmax...does your AS53xx handle SIP also? |
| 01:11.37 | *** join/#asterisk j0sephi (i=j0e@2001:6f8:99e:243:0:0:0:32) |
| 01:11.42 | twitchmax | it can |
| 01:12.04 | twitchmax | but all my other incoming stuff is h323 |
| 01:12.10 | mark_mia | that may be easier...but I'm certain ive seen some examples of how to connect Asterisk to a AS53XX via h.323 |
| 01:13.15 | *** join/#asterisk kiwnix (n=egarcia@82.158.154.119) |
| 01:14.02 | *** join/#asterisk r4t0n (n=polariso@dsl-201-144-16-187.prod-infinitum.com.mx) |
| 01:14.26 | *** join/#asterisk mog_home (n=mogorman@user-24-236-84-48.knology.net) |
| 01:15.40 | *** part/#asterisk r4t0n (n=polariso@dsl-201-144-16-187.prod-infinitum.com.mx) |
| 01:15.47 | mark_mia | For those of you interested in the Firefox IAX client... |
| 01:15.48 | mark_mia | http://moziax.mozdev.org/ |
| 01:15.52 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
| 01:16.02 | pygrammer | mark_mia: thanks! will take a look |
| 01:17.33 | hugo-v6 | someone posted that a while ago. still no new version but looks ok. |
| 01:17.47 | hugo-v6 | and it got a small "manager" also |
| 01:17.55 | *** join/#asterisk krisguy (n=krisguy@h216-170-039-057.adsl.navix.net) |
| 01:18.19 | twitchmax | think i might have found something, thanx for the help mark |
| 01:18.32 | hugo-v6 | well... going to look after my bed now. gd n8 pals. |
| 01:21.06 | Snake-Eyes | is there any recommened 2.6 kernel version for asterisk on debian? Is the stable debian kernel good enough or does some of the latest kernels offer better driver configs or higher loads? |
| 01:21.27 | JunK-Y | Your operating system is not currently supported by Windows Media Player. |
| 01:21.29 | JunK-Y | heheh |
| 01:21.31 | *** join/#asterisk inetws1 (n=Snak@vt-williston-cuda1a3-216.sbtnvt.adelphia.net) |
| 01:21.43 | *** part/#asterisk inetws1 (n=Snak@vt-williston-cuda1a3-216.sbtnvt.adelphia.net) |
| 01:21.59 | *** join/#asterisk inetws1 (n=Snak@vt-williston-cuda1a3-216.sbtnvt.adelphia.net) |
| 01:22.03 | JunK-Y | Snake-Eyes: the latest debian kernel isnt 2.6.8 ? |
| 01:23.11 | inetws1 | . |
| 01:23.53 | *** join/#asterisk njclark (n=njclark@mx4.ibcjapan.co.jp) |
| 01:24.12 | Snake-Eyes | JunK-Y: the lastest debian kernel is 2.6.8, I'm just wondering if installing 2.6.12 or 2.6.13 would be any better? |
| 01:24.30 | njclark | hi. |
| 01:24.43 | njclark | has someone played around with the latest IAXy module from digium? |
| 01:25.02 | [hC] | this is odd. I have 5 g729 licenses on this box, but i hear no audio on the 5th channel (it was a single license i added afterwards) |
| 01:25.04 | [hC] | this is the output |
| 01:25.05 | [hC] | jrc*CLI> show g729 |
| 01:25.05 | [hC] | 5/4 encoders/decoders of 5 licensed channels are currently in use |
| 01:25.09 | njclark | I'd like to try using different codecs on it. |
| 01:25.11 | [hC] | the 5th channel is dead air |
| 01:25.48 | mog_home | and its hott |
| 01:27.39 | JunK-Y | Snake-Eyes: dunno, ive 2.4.27 and im fine with that. |
| 01:28.54 | bkw_ | someone called my name? |
| 01:29.19 | JunK-Y | ur mother. |
| 01:29.33 | [hC] | what the heck. |
| 01:30.03 | tsume | ugh* |
| 01:30.07 | tsume | question |
| 01:30.16 | kiwnix | hi, does ulaw codec gives the best audio quality? |
| 01:30.18 | tsume | I don't know queues very well, and this was working yesterday |
| 01:30.27 | tsume | I have agent => 101,101,Reception 1 |
| 01:30.31 | tsume | in agents.conf |
| 01:30.44 | tsume | and the agent is registered on the system |
| 01:30.52 | tsume | the console is sending 500*101 |
| 01:31.15 | tsume | and then in the asterisk -r, it says it fails with nonzeroo (which obviously maens failed) |
| 01:31.20 | tsume | any idea why this would happen? |
| 01:31.44 | tsume | unless I'm not understanding my friends setup correctly |
| 01:31.47 | tsume | this should workl |
| 01:32.28 | *** join/#asterisk jcims (n=jcims@cpe-24-210-60-100.columbus.res.rr.com) |
| 01:33.13 | tsume | <PROTECTED> |
| 01:33.17 | tsume | <PROTECTED> |
| 01:33.29 | tsume | show agents => 101 (Reception 1) not logged in (musiconhold is 'default') |
| 01:34.26 | tsume | wait... |
| 01:34.34 | tsume | <PROTECTED> |
| 01:34.36 | tsume | <PROTECTED> |
| 01:34.44 | tsume | this means It was authenticated, right? |
| 01:34.59 | tsume | then why does exten => s,7,Hangup() mess it up? |
| 01:35.17 | [hC] | I should be able to install two separate g729 licenses in * just fine right? I have one for 4 and one for 1, and for some reason my 5th call always has no audio, like it ran out of codecs. |
| 01:36.19 | tsume | oh there is really something wrong :( |
| 01:39.12 | *** join/#asterisk tbone__ (n=bdixon@adsl-33-135-21.asm.bellsouth.net) |
| 01:42.36 | *** join/#asterisk Hmmhesays (n=Neg@24-117-213-113.cpe.cableone.net) |
| 01:43.32 | *** part/#asterisk tbone__ (n=bdixon@adsl-33-135-21.asm.bellsouth.net) |
| 01:43.47 | *** join/#asterisk Kumbang (n=unknown@167.205.24.5) |
| 01:44.56 | Hmmhesays | what up |
| 01:45.53 | *** part/#asterisk bowman (n=bowman@195.46.47.202.static.cablesurf.de) |
| 01:45.58 | tsume | holy hell this sucks |
| 01:46.51 | tsume | I've been sitting here for 6 hours trying to get a phone to log in |
| 01:47.14 | mark_mia | humm tsume....u got problems |
| 01:47.30 | tsume | mark_mia: no, the friggen phone has problems |
| 01:47.46 | tsume | it was working yester day then poof. It can make calls but not log in as an agent |
| 01:47.49 | mark_mia | but your asking about 5 different things |
| 01:48.00 | mark_mia | let's break it down and maybe we can help |
| 01:48.13 | Borgon | Anyone recommend a good prepaid IAX termination provider? |
| 01:48.23 | tsume | why can I do everything, and then always get nailed by stupid crap like this.. |
| 01:48.29 | Hmmhesays | tsume |
| 01:48.38 | Ariel_ | Borgon, nufone,voipjet,asterlink |
| 01:48.40 | Hmmhesays | what is your issue |
| 01:48.51 | Ariel_ | tsume, is that phone already login |
| 01:49.07 | tsume | Ariel_: its registered, but it wont log in as an agent :( |
| 01:49.28 | tsume | I'm at a medical facility and they get no calls in for 6 hours |
| 01:49.41 | Hmmhesays | tsume: what is the issue |
| 01:49.45 | *** join/#asterisk file (n=root@66.199.241.90) |
| 01:49.54 | Hmmhesays | what type of phone, post your sip debug, sip.conf |
| 01:50.37 | Hmmhesays | medical facility? got take a hit of vicodin and come back |
| 01:50.46 | tsume | snom 220 |
| 01:50.50 | *** join/#asterisk rking (n=rking@ip68-105-231-56.lu.dl.cox.net) |
| 01:50.58 | Hmmhesays | and its not registering? |
| 01:51.28 | JunK-Y | move ur feet! ohhhhhhhh yeah! |
| 01:51.40 | Hmmhesays | hey hey |
| 01:51.40 | JunK-Y | dont stop the beat, gO! |
| 01:51.43 | JunK-Y | sup? |
| 01:51.49 | Hmmhesays | nm, drinking a cold one |
| 01:51.54 | Hmmhesays | trying to help out tsume |
| 01:52.01 | JunK-Y | i just finished my orange shnaps from chicago. |
| 01:52.10 | Hmmhesays | sweet |
| 01:52.15 | Hmmhesays | cheers JunK-Y |
| 01:52.22 | JunK-Y | and received my damn cell bill, 92$ fucking shit. |
| 01:52.29 | Hmmhesays | yeah mine was 162 |
| 01:52.31 | JunK-Y | cheers! |
| 01:52.37 | rking | would it be safe to: cd /etc/asterisk; mkdir bak; mv *.* bak; and start from nothing? i'm pretty confused by all this default config, and all i want is to set up a place where people can dial in via iaxcomm to conference |
| 01:52.39 | *** join/#asterisk Cybertank (n=cybertan@CPE000dbd0f269c-CM00111ae6ff9c.cpe.net.cable.rogers.com) |
| 01:53.08 | Hmmhesays | safe to move it there? yes safe to build your own config? prolly not |
| 01:53.19 | tsume | http://rafb.net/paste/results/PM4K1t79.html |
| 01:53.24 | tsume | theres what happens |
| 01:53.27 | *** join/#asterisk ManxPower (n=eric@1Cust2218.an3.dfw28.da.uu.net) |
| 01:53.50 | rking | Hmmhesays: hehe, ok |
| 01:53.53 | Hmmhesays | ok tsume were is the problem, that looked pretty good |
| 01:54.09 | tsume | Hmmhesays: show agents say ... |
| 01:54.15 | tsume | 101 (Reception 1) not logged in (musiconhold is 'default') |
| 01:54.26 | tsume | I must really be missing something simple |
| 01:54.38 | *** join/#asterisk Qwell (n=north@24-50-66-194.vnnyca.adelphia.net) |
| 01:54.42 | Hmmhesays | rking the iax conference is pretty easy, just add an iax user in iax.conf in extensions.conf make a meetme conference in the same context of your iax user |
| 01:54.47 | rking | Hmmhesays: well, i'm liking the looks of this: http://www.voip-info.org/tiki-index.php?page=Asterisk+IAX+clients if i replaced my iax.conf with something like that, and left all the rest as default, would i probably be ok? |
| 01:54.51 | Vco | ahh...fix for the 220 http://gallery.tigert.com/gallery/skeet/aal?set_albumName=skeet&id=aal |
| 01:55.28 | Hmmhesays | or you could leave your iax.conf there and just add a user entry at the bottom |
| 01:55.31 | rking | Hmmhesays: ok, i have made some attempts that are sort of like that, to no avail, but it is a new day, so maybe i will have new success. |
| 01:57.38 | tsume | I can't believe this.. |
| 01:57.51 | Hmmhesays | rking you send me a postcard and i'll tell you exactly what to do |
| 01:57.57 | Hmmhesays | unless you are from the USA then I don't care |
| 01:58.17 | Borgon | Hello i am shopping around for prepaid iax termination providers... I see junction networks offering VOIP Termination for $.029/minute, but other sites are 2.0cents per minute? is this for international or is that a real rate? |
| 01:58.34 | Hmmhesays | tsume: were you at? |
| 01:58.44 | rking | Hmmhesays: i could launder my boring US postcard through a friend in Austria, if that would help |
| 01:58.52 | Hmmhesays | haha no thats ok |
| 01:59.00 | Hmmhesays | pm me, i'll give you what you need to set in iax.conf |
| 01:59.17 | *** join/#asterisk greekman (n=alex@host254.209.113.199.conversent.net) |
| 01:59.24 | tsume | Hmmhesays: this doesn't mean anything does it? |
| 01:59.25 | tsume | <PROTECTED> |
| 01:59.25 | tsume | <PROTECTED> |
| 01:59.35 | tsume | Hmmhesays: still on the phone |
| 01:59.35 | ManxPower | Once I confirm my house is destroyed, I'l start the "Manx Power Katrina Recovery Fund" |
| 01:59.44 | njclark | Borgon: why don't you try out Voipjet ? -- http://www.voipjet.com |
| 01:59.44 | tsume | That message repeats when I 500*101 |
| 01:59.48 | Katty | beep |
| 01:59.52 | tsume | not that message |
| 01:59.55 | tsume | I meant the authentication |
| 01:59.55 | Hmmhesays | tsume: are you using asterisk at home? |
| 02:00.00 | tsume | Hmmhesays: no. |
| 02:00.15 | Hmmhesays | what the hell is trying to connect to the manager interface then? |
| 02:00.23 | tsume | asterisk -r I guess |
| 02:00.30 | tsume | my friend set this server up |
| 02:00.30 | Katty | Hmmhesays: hi |
| 02:00.31 | Hmmhesays | um no |
| 02:00.32 | DarthClue | ManxPower: glad to see you made it out. |
| 02:00.35 | Hmmhesays | Katty: hi |
| 02:00.40 | tsume | Hmmhesays: I don't know. I didn't set this server up |
| 02:00.45 | Hmmhesays | ouch |
| 02:00.48 | ManxPower | DarthClue, Yeah. |
| 02:00.57 | rking | Hmmhesays: i sure will pm you - at the moment i'm having a bit of trouble - for the longest time i had my password automatic, and then when i lost my config, i lost the ability to register. |
| 02:01.00 | rking | =) |
| 02:01.23 | ManxPower | Have you seen on the news about Waveland MS? |
| 02:01.24 | Hmmhesays | I don't want, anybody else, when i think about you I touch myself whooaaaa |
| 02:01.30 | ManxPower | I live(d?) in Waveland MS |
| 02:01.32 | DarthClue | ManxPower: I'm pretty sure the place is gone, and I don't even live there. then again, maybe your former residence has already been featured on cnn and the likes. |
| 02:01.34 | Hmmhesays | haha Katty |
| 02:01.46 | tsume | Hmmhesays: could I not be starting something? |
| 02:01.53 | Hmmhesays | tsume I'm not sure |
| 02:01.56 | ManxPower | DarthClue, all the coverage I've seen is for places SOUTH of the railroad embankment. |
| 02:01.57 | DarthClue | ManxPower: haven't seen wakeland mentioned. |
| 02:01.58 | Hmmhesays | let me look at your pastebin again |
| 02:01.59 | tsume | Hmmhesays: safe_asterisk I use |
| 02:02.06 | Hmmhesays | tsume why? |
| 02:02.08 | Katty | Hmmhesays: i say we blow this taco stand and go to the strip club. |
| 02:02.16 | DarthClue | amateur night? |
| 02:02.25 | Hmmhesays | Katty: cheers to that |
| 02:02.27 | tsume | Hmmhesays: its what my friend had in the scripts |
| 02:02.30 | Hmmhesays | we'll bring darth |
| 02:02.34 | tsume | Hmmhesays: I'm afraid to run anything else |
| 02:02.40 | Hmmhesays | tsume why? |
| 02:02.44 | Hmmhesays | ain't working anyway |
| 02:02.44 | tsume | Hmmhesays: he has the phones logging in as root on ftp too |
| 02:02.46 | Katty | DarthClue: oh you're right. it's wednesday |
| 02:02.48 | DarthClue | i can commandeer a bus |
| 02:02.54 | Hmmhesays | tsume: is your friend stupid? |
| 02:02.56 | Katty | Hmmhesays: nevermind. i don't want to go |
| 02:02.57 | Hmmhesays | no offense |
| 02:03.05 | tsume | Hmmhesays: yes |
| 02:03.23 | DarthClue | Katty: we won't make you dance, you can just watch while Hmmhesays gets his freak on with the wall again |
| 02:03.41 | tsume | Hmmhesays: He completely hacked around this system |
| 02:03.46 | tsume | Hmmhesays: its a HORRID mess |
| 02:03.49 | tsume | I'm surprised it works |
| 02:03.51 | tsume | *worked |
| 02:03.57 | Hmmhesays | w00t to getting your freak on with the wall! |
| 02:04.30 | DarthClue | ManxPower: from what i've heard Bay St Louis is gone, is wakeland on much higher ground or about the same? if the same, it is probably gone as well. |
| 02:04.54 | Hmmhesays | tsume are the other agent phones working? |
| 02:05.30 | tsume | Hmmhesays: hmm |
| 02:05.33 | DarthClue | ManxPower: doesn't look good ... http://news.google.com/news?q=waveland%2C%20ms&hl=en&hs=yQy&lr=&safe=off&client=firefox-a&rls=org.mozilla:en-US:official&sa=N&tab=wn |
| 02:05.39 | ManxPower | DarthClue, yes, the part of Waveland I am/was in is much higher ground than bay st. louis |
| 02:05.39 | tsume | Hmmhesays: there are supposedly more, but I haven't checked |
| 02:05.42 | *** join/#asterisk MRH2 (n=Mr_happy@fcirc-adsl.demon.co.uk) |
| 02:05.45 | Hmmhesays | check them |
| 02:05.59 | Hmmhesays | you need to find out if the problem is global, or unique the that snom |
| 02:06.49 | ManxPower | DarthClue, Hence my usage of the term "confirm" |
| 02:07.21 | DarthClue | yeah, your house may have been leveled by the winds as many in that area were |
| 02:08.09 | DarthClue | quote from one of those articles ... "They had not heard anything about Waveland on the news, and they said they thought that meant it had been spared. They were wrong." |
| 02:08.11 | MRH2 | can anyone forsee a problem putting externip address from sip.conf as a name in hosts file pointing to 127.0.0.1 |
| 02:08.32 | *** join/#asterisk Sedorox (i=brandon@smartserv/cna/Sedorox) |
| 02:08.43 | *** part/#asterisk twitchmax (n=twitch@c-24-98-56-147.hsd1.ga.comcast.net) |
| 02:08.44 | kshumard_home | wow, I lived in Bay St Louis for a summer.... hard to imagine it's just gone |
| 02:08.44 | tsume | oh dear god |
| 02:08.53 | tsume | Hmm-Beerme1: yes |
| 02:08.56 | tsume | oops |
| 02:09.02 | tsume | oh |
| 02:09.07 | tsume | Hmm-Beerme1: heh, :( Yes |
| 02:09.10 | Hmm-Beerme1 | the rest are working? |
| 02:09.17 | Hmm-Beerme1 | so its just this snom thats farking up |
| 02:09.25 | tsume | all I did was restart asterisk, it used to work |
| 02:09.32 | tsume | Hmm-Beerme1: nope. |
| 02:09.34 | tsume | shit. |
| 02:09.51 | tsume | what would prevent agent login? |
| 02:09.51 | Hmm-Beerme1 | if hes using a bunch of scripts you need to reboot the whole system |
| 02:10.00 | tsume | okay, rebooting |
| 02:10.10 | *** join/#asterisk The_Ball (n=alex@static-227.35.240.220.dsl.comindico.com.au) |
| 02:10.18 | Hmm-Beerme1 | i'm listening to sublime, drinking beer, life is good |
| 02:10.28 | tsume | lets see if it works |
| 02:10.35 | tsume | and if it does. I'm going to be really angry |
| 02:10.47 | Hmm-Beerme1 | haha, I'll probably laugh and you'll owe me a postcard |
| 02:11.04 | tsume | Hmm-Beerme1: I'll send you some pictures. I live in Alaska |
| 02:11.08 | *** join/#asterisk Tabo2 (n=acura0@S010600119503044b.vc.shawcable.net) |
| 02:11.09 | Hmm-Beerme1 | rock |
| 02:11.15 | Hmm-Beerme1 | hey file[laptop] what up? |
| 02:11.25 | file[laptop] | oh not much, you? |
| 02:11.38 | Hmm-Beerme2 | not much, listening to sublime and enjoying some beer |
| 02:11.52 | Tabo2 | does anyone have documention on how to read debug screens? |
| 02:11.53 | Qwell | file[laptop]: y0 |
| 02:12.03 | Qwell | Tabo2: right to left, top to bottom |
| 02:12.08 | Qwell | ...left to right |
| 02:12.14 | Tabo2 | SIP/2.0 200 OK |
| 02:12.21 | Tabo2 | does that mean extension 200 is working |
| 02:12.41 | file[laptop] | no, it means that a 200 OK message came from somewhere |
| 02:12.41 | Qwell | 200 is the code |
| 02:12.49 | Qwell | rather, "200 OK" is the code |
| 02:12.50 | file[laptop] | like... OVER THERE!1 |
| 02:12.55 | bkw_ | file dear |
| 02:12.56 | file[laptop] | 200 is the reply code |
| 02:12.58 | bkw_ | go test the rpid patch again |
| 02:13.05 | Qwell | bkw_: Where were you?! |
| 02:13.10 | bkw_ | I updated it to be pointers on the pvt so we don't malloc the memor when not needed |
| 02:13.17 | bkw_ | Qwell, as in? |
| 02:13.19 | bkw_ | i'm at home |
| 02:13.19 | rkingle | Hmm-Beerme2: here's my current iax.conf - http://rafb.net/paste/results/mWkdaz16.txt |
| 02:13.21 | Qwell | I was totally in the gayest store in Vegas on Monday |
| 02:13.23 | file[laptop] | arf |
| 02:13.23 | bkw_ | where is home? Oklahoma |
| 02:13.33 | file[laptop] | I shall test it in a bit. |
| 02:13.36 | bkw_ | Qwell, and? |
| 02:13.40 | Qwell | I needed help... |
| 02:13.53 | tsume | Hmm-Beerme2: awww |
| 02:13.56 | tsume | Hmm-Beerme2: nope :( |
| 02:14.05 | Qwell | guys were like...staring at me... |
| 02:14.09 | *** join/#asterisk Cybertoy (n=Cybertoy@dsl254-123-241.nyc1.dsl.speakeasy.net) |
| 02:14.11 | Hmm-Beerme2 | tsume: post your extensions.conf where you have the agents login |
| 02:14.27 | tsume | I hate my friend |
| 02:14.29 | Hmm-Beerme2 | I dated a girl from oklahoma once |
| 02:14.32 | Hmm-Beerme2 | but you've all heard about that |
| 02:14.51 | Hmm-Beerme2 | it was actually bkw_ in drag |
| 02:14.56 | Hmm-Beerme2 | er.. j/k LOL |
| 02:15.00 | bkw_ | No I don't do drag |
| 02:15.05 | Vco | heard of it? everyone tapped that.. |
| 02:15.22 | *** join/#asterisk vuvie (n=vuvie@bb219-74-47-90.singnet.com.sg) |
| 02:15.27 | *** part/#asterisk vuvie (n=vuvie@bb219-74-47-90.singnet.com.sg) |
| 02:15.31 | tsume | http://rafb.net/paste/results/NOJ2sF91.html |
| 02:15.37 | Qwell | bkw_: anyhow, I had nobody to help me fend them off :p |
| 02:15.51 | DarthClue | bkw_ doesn't do drag, not even when he has had way too many. although, he does shake hands alot. |
| 02:16.26 | Hmm-Beerme2 | should I have another beer before I go and enjoy 50 cent tap night? |
| 02:16.38 | Qwell | Hmm-Beerme2: nope, not if its 50 cents |
| 02:16.40 | tsume | Hmm-Beerme2: should the above link not be corect? |
| 02:16.42 | *** join/#asterisk TheCops (n=mdb@206-248-136-187.dsl.teksavvy.com) |
| 02:16.42 | Hmm-Beerme2 | tsume, post the line that calls that macro |
| 02:16.47 | tsume | oh |
| 02:16.49 | bkw_ | Qwell, I can tell you how to get them to go away |
| 02:16.56 | Qwell | bkw_: please do |
| 02:17.00 | bkw_ | Qwell, pick your nose |
| 02:17.01 | hmodes | brah! |
| 02:17.01 | tsume | exten => 500*101,1,Macro(agent-add,101,) ; |
| 02:17.05 | Qwell | heh |
| 02:17.10 | bkw_ | thats so unattractive |
| 02:17.11 | Vco | pick AND flick |
| 02:17.17 | Qwell | is it normal for gay guys to totally stare and grin at you? |
| 02:17.17 | bkw_ | sure sure |
| 02:17.18 | Vco | you need to do to followthrough |
| 02:17.26 | bkw_ | Qwell, the bad ones sure.. |
| 02:17.33 | Qwell | bad as in? oh god |
| 02:17.41 | *** join/#asterisk MikeJ[Laptop] (n=ircatjer@d14-69-8-30.try.wideopenwest.com) |
| 02:17.42 | bkw_ | bad as in don't get drunk around them |
| 02:17.44 | Ariel_ | Qwell, are you having issues? |
| 02:17.46 | Vco | but what do you do if they start picking their nose? |
| 02:17.46 | Qwell | yeah... |
| 02:17.47 | tsume | Qwell: 9 ince monsters ;) |
| 02:17.50 | Qwell | Ariel_: I was monday : |
| 02:17.51 | Qwell | p |
| 02:17.51 | Hmm-Beerme2 | you are authenticating "agent-add"? |
| 02:17.52 | bkw_ | because your ass will be sore afterwords |
| 02:17.54 | Hmm-Beerme2 | wtf |
| 02:18.04 | tsume | Hmm-Beerme2: my friend set it up |
| 02:18.08 | tsume | Hmm-Beerme2: I don't understand his work |
| 02:18.20 | Hmm-Beerme2 | it looks like he ripped this from aah |
| 02:18.21 | Qwell | bkw_: TMI...thanks for the heads up though :p |
| 02:18.32 | tsume | Hmm-Beerme2: what is it supposed to look lik? |
| 02:18.48 | file[laptop] | all my friends are going away, how sad |
| 02:18.48 | MikeJ[Laptop] | hmmm |
| 02:18.50 | file[laptop] | all I have left is you guys |
| 02:18.51 | file[laptop] | furtehr sad |
| 02:18.53 | file[laptop] | er further |
| 02:18.57 | file[laptop] | GROUP HUG! |
| 02:19.20 | *** join/#asterisk cybyc (n=cyb@CPE00062919e286-CM0012250233aa.cpe.net.cable.rogers.com) |
| 02:19.28 | tsume | Hmm-Beerme2: I almost have to go.. |
| 02:19.34 | tsume | Hmm-Beerme2: I don't want to come back tomarrow.. |
| 02:19.42 | Hmm-Beerme2 | tsume: when you log in the snom does it prompt you for authentication? |
| 02:19.43 | tsume | Hmm-Beerme2: do you have an example I can check? |
| 02:19.58 | tsume | Hmm-Beerme2: no, I set it in the web browser |
| 02:20.02 | tsume | Hmm-Beerme2: it says its 1234 |
| 02:20.04 | tsume | the password |
| 02:20.07 | Hmm-Beerme2 | omg you are using amp |
| 02:20.25 | Hmm-Beerme2 | that complicates things considerably |
| 02:20.26 | tsume | thats how it gets on asterisk registration |
| 02:20.35 | *** join/#asterisk mark_mia (n=mark__mi@24-55-227-126.miamfl.adelphia.net) |
| 02:20.51 | rkingle | if, from iaxcomm, i put this in the Extension box: foo@localhost/s - the breakdown is this, yes?: foo = user to connect as, localhost = host, /s = extension s? |
| 02:21.03 | Ariel_ | hello mark_mia just another person from my home town. |
| 02:21.13 | Qwell | rkingle: correct |
| 02:21.22 | mark_mia | yep |
| 02:22.20 | rkingle | Qwell: hrmm, so when i do something just like that, why would i get this from "asterisk -c": chan_iax2.c:3910 register_verify: No registration for peer 'rking' (from 127.0.0.1) ... socket_read: Rejected connect attempt from 127.0.0.1 - what i mean is, where does "rking" even come into play, if i'm connecting as user foo ? |
| 02:22.26 | *** join/#asterisk santiago (n=santiago@63.245.86.163) |
| 02:23.01 | *** join/#asterisk Gronker__ (n=Gronker2@adsl-220-93-76.ags.bellsouth.net) |
| 02:23.31 | mark_mia | 'rking' must be a context, user or peer in your iax.conf |
| 02:23.49 | Qwell | user with a static IP perhaps |
| 02:23.50 | mark_mia | or it could be somethign who is trying to authenticate against your * from the outside |
| 02:23.55 | rkingle | mark_mia: "must be" as in "should be", or "apparently is"? |
| 02:24.00 | Hmm-Beerme2 | tsume I dunno whats up and I have to run |
| 02:24.26 | rkingle | grepping "rking" in /etc/asterisk/* yields nothing, btw |
| 02:25.19 | mark_mia | very interesting |
| 02:25.24 | cybyc | Hi! Does anyone know if there is a way to match for exten _%23XX? My SIP phone insists on sending the # symbol as '%23' in SIP its requests. |
| 02:25.53 | Qwell | does _%23XX not work? |
| 02:26.02 | mark_mia | and originating on your localhost...very interesting |
| 02:26.11 | cybyc | it doesn't seem to... neither does _#XX |
| 02:26.24 | Ariel_ | cybyc, your matching # from what phone or device. |
| 02:26.36 | tsume | dear god this sucks |
| 02:26.42 | Hmm-Beerme2 | what does show agents |
| 02:26.44 | Hmm-Beerme2 | say |
| 02:26.48 | Ariel_ | tsume, set the user as static |
| 02:26.56 | *** part/#asterisk Gronker__ (n=Gronker2@adsl-220-93-76.ags.bellsouth.net) |
| 02:26.58 | tsume | 101 (Reception 1) not logged in (musiconhold is 'default') |
| 02:26.59 | cybyc | Aastra 9133i or 480i |
| 02:27.08 | tsume | I feel sick |
| 02:27.24 | Ariel_ | # is used to send the call out in most sip phones |
| 02:27.25 | tsume | Ariel_: I've been at a screen for 6 hours, I'm braindead |
| 02:27.53 | tsume | Ariel_: how, in dear god, would I do that? |
| 02:28.13 | tsume | *mark* 7 hours |
| 02:28.13 | Ariel_ | tsume, through the amp gui setup for the queues |
| 02:28.22 | tsume | what? |
| 02:28.24 | tsume | I'm not using amp |
| 02:28.35 | tsume | I used the web administration only on the snom |
| 02:28.46 | tsume | I like command line |
| 02:28.59 | tsume | I just didn't want to run back and forth through the building :) |
| 02:29.02 | cybyc | Ariel_: yes -- on the Aastra phone, they seem to allow # anywhere in the dialplan... The SIP requests ends up being sent to * with a header like this: To: "#63" <sip:%2363@192.168.1.2:5060>; |
| 02:29.09 | Ariel_ | tsume, does your asterisk box have amp setup? |
| 02:29.16 | Qwell | Don't use #... |
| 02:29.19 | tsume | Ariel_: don't think so |
| 02:29.30 | tsume | Ariel_: my friend barely got the box up I think :) |
| 02:29.41 | tsume | Ariel_: wait |
| 02:29.51 | tsume | would I add AgentLogin(101) in my extensions.conf? |
| 02:30.00 | Qwell | cybyc: why not make it _*XX ? |
| 02:30.15 | tsume | Ariel_: thank god its not my box, I'd shoot myself |
| 02:30.30 | cybyc | Qwell: I might have to change the #XX codes - pitty. |
| 02:30.32 | bkw_ | cybyc, actually those should be URL decoded by default as per the RFC in asterisk |
| 02:30.42 | cybyc | Qwell: The *XX codes are already taken |
| 02:30.44 | bkw_ | but asterisk doesn't do what the RFC says in that case because it might take 1 nano second longer |
| 02:30.45 | Ariel_ | cybyc, #is not a very good thing to use with sip devices |
| 02:31.10 | bkw_ | blitzrage, what was the note on this one |
| 02:31.10 | *** join/#asterisk nick125 (n=nick125@unaffiliated/nick125) |
| 02:31.58 | tsume | Ariel_: what line do I need to add to my scripts for a static agent? |
| 02:32.44 | Ariel_ | tsume, it's setup in queues.conf |
| 02:32.52 | tsume | erm |
| 02:32.53 | tsume | minute |
| 02:32.56 | tsume | let me look |
| 02:33.22 | cybyc | @bkw_, I'm ok with asterisk not decoding the %23 - I am just surprised that I can't match on it |
| 02:33.53 | tsume | Ariel_: okay. |
| 02:34.02 | tsume | Ariel_: I guess I need to make many static =] |
| 02:35.01 | bkw_ | if (pedanticsipchecking) |
| 02:35.01 | bkw_ | <PROTECTED> |
| 02:35.04 | tsume | Ariel_: might you be able to tell me? :) |
| 02:35.07 | bkw_ | it onlly does it when pedantic is on |
| 02:35.10 | bkw_ | which is just lame |
| 02:35.28 | bkw_ | it should either be an option flag.. or on my default as per the RFC |
| 02:35.44 | Borgon | Any good prepaid voip iax providers that start minimum 10 bucks with minutes less than 2.0cents/min ? I am searching and junction networks does .029/min unbelievable and i dont see how |
| 02:36.20 | *** part/#asterisk Cybertoy (n=Cybertoy@dsl254-123-241.nyc1.dsl.speakeasy.net) |
| 02:36.55 | Qwell | Borgon: millions of minutes/month |
| 02:37.23 | tsume | Ariel_: please tell me you're at least still there |
| 02:37.28 | cybyc | @bkw_, thanks! I'll try with pedantic on - even if it is evil ;) |
| 02:37.31 | tsume | I'm tired hungry, and want to go home |
| 02:37.34 | Borgon | Qwell: hehe i just got 10 dollars and am trying to find a good iax termination provider |
| 02:37.50 | Qwell | asterlink; 2c/min |
| 02:37.52 | bkw_ | cybyc, you'll be lucky if your device will even register with pedantic on |
| 02:38.01 | bkw_ | its very very not right |
| 02:38.50 | Qwell | bkw_: make it an option |
| 02:39.00 | Borgon | ill check asterlink out |
| 02:39.00 | bkw_ | nick125, are you using sip now btw? |
| 02:39.02 | rkingle | is iaxComm supposed to not be remembering anything i put in the Extension box? seems like it would be handy for it to remember - if this is known behavior, then that's fine, but if i've got a wonked version of iaxComm, i'd like to know, too. =) |
| 02:39.03 | Qwell | if (pedanticsipchecking || option_on) |
| 02:39.15 | bkw_ | Qwell, well it shoudl really be on by default |
| 02:39.19 | nick125 | bkw_: no, whats going on? |
| 02:39.28 | bkw_ | nick125, iax2 has audio issues at times |
| 02:39.31 | Borgon | How can it be .029/min for junction and 2cents/min for asterlink |
| 02:39.32 | bkw_ | just letting ya know ;) |
| 02:39.38 | bkw_ | I switched all my stuff to use SIP |
| 02:39.52 | bkw_ | nick125, if you have the rpid patch I helped with off the tracker you can send callerid properly |
| 02:39.53 | bkw_ | via sip |
| 02:40.21 | nick125 | including name? or is that the same? |
| 02:40.37 | bkw_ | ast_uri_decode |
| 02:40.38 | bkw_ | doh |
| 02:40.42 | bkw_ | http://bugs.digium.com/view.php?id=2471 |
| 02:40.44 | bkw_ | just number |
| 02:40.51 | nick125 | when i call it, it seems to be same quality |
| 02:41.08 | bkw_ | nick125, inbound sure.. but the person your calling will hear audio clips and misses |
| 02:41.23 | tsume | bkw_: do you have a page which explains static agent? |
| 02:41.28 | bkw_ | tsume, no |
| 02:41.34 | bkw_ | but a static agent is one that is there at all times |
| 02:41.38 | Qwell | bkw_: even head-head? |
| 02:41.40 | bkw_ | ie agent => Agent/1001 |
| 02:41.43 | Qwell | with the iax issues |
| 02:41.44 | bkw_ | or |
| 02:41.48 | bkw_ | agent => SIP/10 |
| 02:41.56 | bkw_ | Qwell, yes cvs-head has these issues under about a 20 call load |
| 02:42.00 | tsume | my friend has member => Agent/101 |
| 02:42.06 | bkw_ | ya ya |
| 02:42.06 | tsume | *has, |
| 02:42.07 | bkw_ | thats it |
| 02:42.07 | nick125 | bkw_: it seems to be real poor quality, not in skips and such, it just sounds real weird |
| 02:42.09 | Qwell | ahh |
| 02:42.13 | tsume | I'm so screwed |
| 02:42.17 | bkw_ | nick125, muttled right? |
| 02:42.23 | nick125 | kinda |
| 02:42.24 | *** join/#asterisk ZX81 (n=ZX81@222-153-80-230.jetstream.xtra.co.nz) |
| 02:42.28 | bkw_ | switch to sip it won't do that |
| 02:42.30 | bkw_ | ;) |
| 02:42.37 | nick125 | ok like to assist? =P |
| 02:42.50 | bkw_ | the config is exactly the same as iax.conf's |
| 02:42.59 | bkw_ | replace auth=md5 with fromuser=user_id |
| 02:43.01 | *** part/#asterisk ZX81 (n=ZX81@222-153-80-230.jetstream.xtra.co.nz) |
| 02:43.02 | nick125 | so i just cut and paste? |
| 02:43.03 | tsume | I'm really hating my friend now. |
| 02:43.04 | bkw_ | yep |
| 02:43.05 | mark_mia | tsume why don't u cut and paste your extensions.conf |
| 02:43.11 | tsume | mark_mia: I didn't |
| 02:43.16 | tsume | mark_mia: I'm not a dumb ass, I know |
| 02:43.33 | tsume | I sure as hell would never had made spaghetti script |
| 02:43.40 | bkw_ | brb |
| 02:43.46 | nick125 | bkw_: ok, im migrating now |
| 02:43.58 | mark_mia | ok, cut and paste the important stuff |
| 02:44.12 | mark_mia | w/o reveling confidential info |
| 02:44.18 | Borgon | Qwell: when i look at the asterlink signup.. they require Optional Expedited Toll-Free Number Porting (extra $45) |
| 02:44.50 | tsume | screw it |
| 02:45.03 | tsume | I'll leave a email and voicemail for my friend |
| 02:45.10 | tsume | I'm tired of his linux newbieness |
| 02:45.15 | tsume | I don't have the time to waste on this. |
| 02:45.28 | mark_mia | give it a reboot for good measure and go home |
| 02:45.32 | greekman | hi all |
| 02:45.46 | greekman | 1 |
| 02:46.04 | *** part/#asterisk greekman (n=alex@host254.209.113.199.conversent.net) |
| 02:46.05 | opus_ | I have asterisk behind nat. I have a polycom behind nat. I have sip.conf general nat=yes. I have externalip/internal IP. I have sip user nat=yes. I have quailify =yes. When I type rtp debug, however, it is sending the packet to the internal address of the IP 500 phone. |
| 02:46.07 | *** join/#asterisk bkw__ (n=brian@adsl-69-155-21-122.dsl.tulsok.swbell.net) |
| 02:46.07 | *** mode/#asterisk [+o bkw__] by ChanServ |
| 02:46.16 | opus_ | is this a bug? |
| 02:46.23 | opus_ | Sent RTP packet to 192.168.1.102:2262 (type 0, seq 35327, ts 22560, len 160) |
| 02:46.51 | opus_ | under, 'sip show peers' i have NAT, there is a N. |
| 02:46.54 | opus_ | does N stand for no? :) |
| 02:46.58 | Qwell | Borgon: They require an OPTIONAL feature? read it again |
| 02:47.22 | Borgon | Qwell: but 2cents a min.. how can jcnt do .029? |
| 02:47.34 | Qwell | Borgon: use 10 million minutes per month |
| 02:48.08 | Borgon | Qwell: huh? |
| 02:48.30 | Qwell | when you use a fuckload of minutes, the price from the volume providers will go down |
| 02:48.56 | MikeJ[Laptop] | I think he is saying 2.9cents |
| 02:48.56 | Qwell | if you don't have a million dollars per month to spend, don't worry about what other providers get |
| 02:48.58 | Borgon | but wouldnt i have to pay 2cents/min for the million mins ? hehe |
| 02:49.03 | Qwell | MikeJ[Laptop]: no, he means .029 |
| 02:49.09 | *** join/#asterisk greekman (n=alex@host254.209.113.199.conversent.net) |
| 02:49.16 | MikeJ[Laptop] | who does .029? |
| 02:49.29 | Qwell | "jcnt", according to him |
| 02:49.35 | Borgon | MikeJ[Laptop]: VOIP Termination for $.029/minute |
| 02:49.43 | Borgon | http://junctionnetworks.com/ |
| 02:49.51 | MikeJ[Laptop] | ok.. that is 2.9cents |
| 02:49.58 | Borgon | aah i feel dumb now |
| 02:49.58 | MikeJ[Laptop] | not .029 cents |
| 02:49.59 | Qwell | heh |
| 02:50.00 | Borgon | thanks |
| 02:50.03 | opus_ | the internet community so needs a not for profit VOIP provider |
| 02:50.05 | Borgon | hehe i was like wtf how can it be possible |
| 02:50.26 | Qwell | I imagine .029c/min is still possible with enough volume |
| 02:50.29 | MikeJ[Laptop] | level 3 dose not even get below 1/2 cent and 7million min\mo |
| 02:50.41 | gek | opus_: that is probably what most providers are anyway. |
| 02:50.41 | *** join/#asterisk asterisk_newbie (n=root@12-203-248-191.client.insightBB.com) |
| 02:50.47 | Qwell | oh |
| 02:50.57 | MikeJ[Laptop] | so getting .029 would take you being a celec w/ a lot of minutes |
| 02:50.59 | opus_ | gek - yeah, but, like atleast the non-profit would stay around |
| 02:51.44 | rkingle | if someone will help me get iaxcomm conferencing set up tonight, i'd gladly paypal $50 to you. |
| 02:51.53 | rkingle | i'm really hating skype these days |
| 02:51.55 | Qwell | rkingle: What do you need setup with it? |
| 02:51.57 | opus_ | hmmm. make zaptel in CVS HEAD doesn't comiple for me (centos 4) |
| 02:52.02 | Qwell | oh, conferencing... |
| 02:52.07 | Qwell | Do you have zap hardware? |
| 02:52.14 | asterisk_newbie | i am getting the following error any ideas whats wrong chan_sip.c:4053 sip_reg_timeout: Registration for '914xxx@sip.xxxx.net' timed out, trying again |
| 02:52.29 | MikeJ[Laptop] | rkingle, iaxcomm conferencing? |
| 02:52.30 | rkingle | Qwell: i want the ability to host an iax conference with <10 people, and to be able to send/recieve 1-on-1 calls |
| 02:52.47 | rkingle | Qwell: no hardware - is hardware really necessary? |
| 02:52.48 | Qwell | rkingle: setup a MeetMe in asterisk |
| 02:53.00 | Qwell | for meetme it is...or you COULD use ztdummy, but...its not great |
| 02:53.02 | *** join/#asterisk shido6 (n=shido6@d57-87-253.home.cgocable.net) |
| 02:53.52 | Borgon | Qwell: am in the asterlink 2nd page signup.. it says Choose your Toll-Free Number, i dont see an option in the drop down menu to skip or not choose a phone for incoming.. Also how fast was the setup instant and do they call back? |
| 02:54.09 | opus_ | Why is asterisk ignoring my NAT=yes??? |
| 02:54.15 | Qwell | Borgon: join #asterlink, they can help you out |
| 02:54.15 | asterisk_newbie | i am getting the following error any ideas whats wrong chan_sip.c:4053 sip_reg_timeout: Registration for '914xxx@sip.xxxx.net' timed out, trying again |
| 02:54.18 | *** join/#asterisk jaike (n=jaike@203.131.137.76) |
| 02:54.25 | opus_ | its sending RTP packets to the internal address, but SIP packets are going to the external address |
| 02:54.32 | opus_ | (i am using realtime) |
| 02:54.47 | Ariel_ | asterisk_newbie, it can't find 914XXX@sip.XXXX.net |
| 02:54.58 | asterisk_newbie | so it cant find my proxie |
| 02:55.06 | *** part/#asterisk cybyc (n=cyb@CPE00062919e286-CM0012250233aa.cpe.net.cable.rogers.com) |
| 02:55.08 | asterisk_newbie | so i need to call them |
| 02:55.08 | Borgon | Qwell: thanks |
| 02:55.21 | Ariel_ | opus_, you have externip= and localnet= in the sip.conf |
| 02:55.22 | asterisk_newbie | just wanted to make sure |
| 02:55.35 | jaike | hmmm...asterisk is featured at tomshardware.com |
| 02:55.47 | opus_ | localnet? oh . was that changed ? |
| 02:55.59 | opus_ | http://pastebin.ca/21797 |
| 02:56.10 | Ariel_ | localnet=192.168.XXX.XXX/255.255.255.0 |
| 02:56.58 | opus_ | do I really use XXX? |
| 02:58.42 | opus_ | http://pastebin.ca/21798 |
| 02:58.50 | opus_ | i just upgraded to CVS HEAd. that is my 'sip show settings' |
| 03:00.14 | Ariel_ | but you don't have in the sip.conf the externip= |
| 03:00.22 | opus_ | I do... |
| 03:00.32 | opus_ | in my realtime database |
| 03:00.49 | Ariel_ | hummm I don't use realtime. |
| 03:01.07 | opus_ | sip.conf | general | externip | xxx.xxx.xxx.xxx |
| 03:02.39 | Ariel_ | well I have to go see you all later. |
| 03:04.14 | rkingle | anyone have an FXO they want to unload before i get this?: http://cgi.ebay.com/Digium-Wildcard-X100P-OEM-FXO-PCI-Card-Asterisk-PBX_W0QQitemZ5803556889QQcategoryZ61841QQrdZ1QQcmdZViewItem |
| 03:05.27 | *** join/#asterisk incognegro (n=chronocr@Toronto-HSE-ppp3699400.sympatico.ca) |
| 03:05.33 | *** join/#asterisk Defraz_ (n=t0tal@24-119-12-238.cpe.cableone.net) |
| 03:06.33 | opus_ | If i'm doubled natted, should I forward RTP ports as well? |
| 03:06.47 | opus_ | i mean, asterisk is nat and polycom is nated |
| 03:07.48 | *** join/#asterisk Corndawg_ (i=whoisit@c-66-176-249-51.hsd1.fl.comcast.net) |
| 03:08.25 | *** part/#asterisk incognegro (n=chronocr@Toronto-HSE-ppp3699400.sympatico.ca) |
| 03:09.06 | *** join/#asterisk AcAp (n=polarisx@131.93.16.5) |
| 03:09.20 | *** join/#asterisk loud (n=ariel@cypher.punk.net) |
| 03:10.13 | *** join/#asterisk skeffling (n=Andrew_H@andrew.1ec.aaisp.net.uk) |
| 03:11.39 | opus_ | Weird |
| 03:11.40 | opus_ | Ok |
| 03:11.47 | opus_ | It works one out of 20 times |
| 03:12.00 | *** join/#asterisk greekman (n=alex@host254.209.113.199.conversent.net) |
| 03:15.12 | opus_ | no it doesn't work at all |
| 03:16.55 | *** part/#asterisk jcims (n=jcims@cpe-24-210-60-100.columbus.res.rr.com) |
| 03:17.48 | opus_ | this is defintely a big bug |
| 03:18.44 | opus_ | is there a way in gdb that I can show variables .. fairly easy? |
| 03:20.56 | *** join/#asterisk Twister (i=Twister@24-179-91-088.dhcp.chtn.wv.charter.com) |
| 03:21.23 | Twister | is the ability to do sip routing based on distinctive ring in the newest -HEAD? |
| 03:22.01 | *** join/#asterisk rbd (n=robbyd@24-181-194-064.dhcp.hckr.nc.charter.com) |
| 03:23.30 | *** join/#asterisk spootnick (n=julio@50.118.233.220.exetel.com.au) |
| 03:24.05 | spootnick | how do i get rid of "zaptel: Unknown symbol crc_ccitt_table" on dmesg? |
| 03:24.06 | rbd | hey guys, I have a SIP related question somewhat related to askerisk. in a call center setup, where the SIP endpoints are agent phones with their own extensions, is their "email" identifier usually the phone extension number? (e.g. 6505@10.1.14.55:5060 for ext 6505) |
| 03:24.21 | *** join/#asterisk oe1 (n=oej@digium.com) |
| 03:24.46 | rbd | if anyone has any SIP gateway to SIP UA dumps of a phone in this kind of environment that they could point me to, it would be great. I've seen some online but I can't find much |
| 03:25.48 | *** join/#asterisk newsmafia (n=newsmafi@wsip-68-15-19-142.sd.sd.cox.net) |
| 03:26.27 | *** join/#asterisk marrandy (n=marrandy@209.216.76.1) |
| 03:26.42 | marrandy | hello everyone |
| 03:26.52 | marrandy | what's cooking ? |
| 03:27.00 | *** join/#asterisk visible (n=cbb@84.240.52.103) |
| 03:27.01 | visible | <visible> #diablo2 go all |
| 03:27.02 | *** part/#asterisk visible (n=cbb@84.240.52.103) |
| 03:27.46 | marrandy | What does this mean ? |
| 03:27.49 | marrandy | VERBOSE[2025]: == Registered application 'ZapScan' |
| 03:27.49 | marrandy | VERBOSE[2025]: [app_rxfax.so]Aug 10 12:05:13 WARNING[2025]: libspandsp.so.0: cannot open shared object file: No such file or directory |
| 03:27.49 | marrandy | WARNING[2025]: Loading module app_rxfax.so failed! |
| 03:28.39 | tsume | heh, I rebooted the server 5 times, I don't think anything is going to change |
| 03:28.53 | tsume | brought work home with me anyway |
| 03:29.05 | tsume | I can look over the files carefully |
| 03:29.25 | tsume | maybe I'll document it somewhat so I can make some nice libraries |
| 03:29.27 | JunK-Y | cause u need spandsp. |
| 03:30.48 | bkw__ | marrandy, add /usr/local/lib to /etc/ld.so.conf |
| 03:30.51 | marrandy | just found it - had to do a ldconfig |
| 03:30.51 | bkw__ | re rund ldconfig |
| 03:30.55 | bkw__ | er run ldconfig |
| 03:31.05 | file[laptop] | bkw_: do you know how to manually trigger an event on 'da platform? |
| 03:31.05 | marrandy | tks |
| 03:31.09 | *** join/#asterisk brookshire (n=matt@digium.com) |
| 03:31.09 | tsume | is there an asterisk online forum I can post my problem? |
| 03:31.16 | bkw__ | file waht kind of event? |
| 03:31.21 | file[laptop] | bkw_: welcome letter |
| 03:31.24 | Qwell | tsume: asterisk-users mailing list |
| 03:31.28 | bkw__ | file for? |
| 03:31.34 | tsume | Qwell: alright, thanks |
| 03:31.34 | bkw__ | the events are just like cron |
| 03:31.38 | file[laptop] | bkw__: an Asterlink customer |
| 03:31.43 | tsume | hopefully I can get this solved |
| 03:31.43 | bkw__ | you can do a custom * * * * * just like cron |
| 03:31.46 | bkw__ | on any event |
| 03:31.53 | bkw__ | ;) |
| 03:32.03 | file[laptop] | well I don't wanna do a custom anything, it's just this one guy's welcome letter didn't trigger and I don't want to do it manually |
| 03:32.14 | bkw__ | is his account pending? |
| 03:32.19 | file[laptop] | no, active |
| 03:32.26 | SwK | just do it manually and STFU |
| 03:32.26 | bkw__ | thats very odd |
| 03:32.34 | bkw__ | you can't really |
| 03:32.35 | file[laptop] | yeah he broke stage5 |
| 03:32.40 | bkw__ | haha |
| 03:32.44 | file[laptop] | apparently you can according to Teddy |
| 03:32.51 | bkw__ | I have never done that |
| 03:32.55 | bkw__ | ask teddy how ;) |
| 03:33.03 | file[laptop] | he's ignoring me :P |
| 03:33.05 | file[laptop] | SO |
| 03:33.06 | file[laptop] | manual it is |
| 03:33.07 | bkw__ | he's at home |
| 03:33.17 | bkw__ | well lets get the info for that and document it |
| 03:33.25 | file[laptop] | I was gonna write up a script |
| 03:33.38 | SwK | file[laptop]: set a cron job with sipsak to call him every 5 minutes |
| 03:33.44 | bkw__ | hahaha |
| 03:33.45 | file[laptop] | haha |
| 03:33.47 | file[laptop] | you're mean |
| 03:33.56 | SwK | and? |
| 03:34.10 | file[laptop] | well, you're mean |
| 03:34.13 | file[laptop] | I thought I got my point across |
| 03:34.16 | SwK | its still funny |
| 03:36.16 | SwK | anyone know an old school tracker program that lets you enter note sheet music style? Difficulty: OSX |
| 03:36.30 | bkw__ | garageband? |
| 03:36.32 | bkw__ | duh |
| 03:37.18 | *** mode/#asterisk [+o kram] by ChanServ |
| 03:38.24 | SwK | does garageband do that? |
| 03:38.34 | pauldy | hrm I thought sound tracker let you do that too |
| 03:38.39 | file[laptop] | I had a double chocolate muffin today, it was sooo delicious |
| 03:38.42 | SwK | hell I dunno |
| 03:41.55 | bkw__ | ya know |
| 03:41.58 | bkw__ | svn is really nice |
| 03:42.06 | Qwell | bkw_: better then cvs |
| 03:42.34 | bkw__ | than |
| 03:42.35 | bkw__ | yes |
| 03:42.59 | *** join/#asterisk twisted (n=twisted@asterisk/friend-and-developer/pdpc.professional.twisted) |
| 03:42.59 | *** mode/#asterisk [+o twisted] by ChanServ |
| 03:44.47 | *** join/#asterisk Kumbang (n=unknown@167.205.24.5) |
| 03:45.35 | bkw__ | Kumbang, your nick always makes me laugh |
| 03:45.48 | file[laptop] | same here |
| 03:46.04 | bkw__ | I think Kumbag |
| 03:47.01 | *** join/#asterisk znoG (n=gs@200.115.218.81) |
| 03:47.57 | Kumbang | bkw_: pretty funny huh |
| 03:48.02 | bkw__ | yes |
| 03:48.37 | *** join/#asterisk apardo (n=apardo@90.Red-83-46-197.pooles.rima-tde.net) |
| 03:49.09 | bkw__ | lets see if we can make some interesting sentences out of everyone's nicks |
| 03:49.22 | Qwell | bkw_ kumbang file |
| 03:49.25 | Qwell | :D |
| 03:49.29 | file[laptop] | O.O |
| 03:49.55 | bkw__ | twisted kram Kumbang dudes |
| 03:50.12 | *** join/#asterisk skeffling (n=Andrew_H@andrew.1ec.aaisp.net.uk) |
| 03:50.12 | bkw__ | ok thats stupid |
| 03:50.14 | bkw__ | NEXT!!! |
| 03:50.19 | Qwell | 10 points if you can use Beave and blop |
| 03:50.27 | bkw__ | ya really |
| 03:50.39 | bkw__ | Beave, twisted kram's blop |
| 03:50.43 | rkingle | i'm feeling that surge of happiness that comes from replacing a commercial solution with an open-source one. |
| 03:50.59 | bkw__ | rkingle, they have a cream for that |
| 03:51.34 | rkingle | bkw_: sugarcrm? yeah, it's open-source, but it has one of those "crippled until you pay money" kinks, so i wouldn't call it real open-source. |
| 03:52.07 | *** join/#asterisk bmg505 (n=leon@rndf-146-5-135.telkomadsl.co.za) |
| 03:52.15 | *** join/#asterisk chendy (n=Alex_Dot@web1.ningo.net) |
| 03:52.42 | tsume | mmmm |
| 03:53.10 | tsume | just tired as hell since I've only had one hour of sleep in the past 48 hours |
| 03:53.54 | *** join/#asterisk MikeJ[Laptop] (n=ircatjer@d14-69-8-30.try.wideopenwest.com) |
| 03:55.11 | file[laptop] | is there a Rick Alm in here? |
| 03:55.39 | brc_ | no, but I'm bobikiw |
| 03:55.43 | brc_ | pleased to meet you |
| 03:55.46 | file[laptop] | I bet you are |
| 03:56.08 | MikeJ[Laptop] | you bet he is pleased? |
| 03:56.17 | file[laptop] | probably not much, but maybe so |
| 03:56.35 | MikeJ[Laptop] | well.. |
| 03:57.07 | brc_ | *sigh* |
| 03:57.08 | brc_ | you caught me, I couldn't be more horrified to make your acquaintance |
| 03:58.04 | brc_ | note to self: when eating microwaved potato, allow center to cool before attempting to swallow |
| 03:58.15 | nick125 | good idea! |
| 03:58.27 | brc_ | er, did I say that out loud? |
| 03:58.31 | Qwell | ...I should have listened to brc_ |
| 03:58.53 | brc_ | having your wisdom teeth out sucks |
| 03:59.01 | brc_ | apparently it takes a year for the bone to heal |
| 03:59.04 | Qwell | not having them yet sucks more |
| 03:59.19 | Qwell | I have getting them removed to look forward to |
| 03:59.24 | brc_ | weeee |
| 03:59.43 | brc_ | something went wrong with one of mine |
| 03:59.46 | *** join/#asterisk Mango (n=Miranda@Toronto-HSE-ppp3663951.sympatico.ca) |
| 03:59.46 | brc_ | hurt for two weeks |
| 03:59.49 | brc_ | still does actually |
| 03:59.58 | twisted | yay |
| 04:00.05 | brc_ | hello josh |
| 04:00.06 | twisted | perhaps this safari update will fix my safari crashing issues |
| 04:00.11 | Qwell | twisted: never |
| 04:00.22 | twisted | i only sent them a note every single time it happened :) |
| 04:00.33 | twisted | which has probably been over 500 in the past 2 weeks |
| 04:00.49 | Qwell | "Hi, Josh again...still crashing. Thanks." |
| 04:00.53 | *** join/#asterisk ManxPower (i=eric@1Cust663.an1.dfw28.da.uu.net) |
| 04:00.55 | twisted | Qwell, no |
| 04:00.58 | twisted | the internal reporting |
| 04:01.03 | Qwell | oh |
| 04:01.05 | twisted | Reason for crash: "i don't know" |
| 04:01.18 | twisted | what you were doing when the crash happened: "surfing the web...duh" |
| 04:01.45 | twisted | anyway, brb, gotta restart |
| 04:02.03 | *** part/#asterisk twisted (n=twisted@asterisk/friend-and-developer/pdpc.professional.twisted) |
| 04:03.41 | *** join/#asterisk twisted (n=twisted@asterisk/friend-and-developer/pdpc.professional.twisted) |
| 04:03.41 | *** mode/#asterisk [+o twisted] by ChanServ |
| 04:04.58 | JerJer | ~manxpower |
| 04:05.00 | jbot | manxpower is probably your God. |
| 04:05.07 | JerJer | ~seen manxpower |
| 04:05.12 | jbot | manxpower is currently on #asterisk (4m 19s) |
| 04:05.23 | JerJer | ManxPower: you survive? |
| 04:05.27 | MikeJ[Laptop] | ManxPower is currently in mississippi |
| 04:05.34 | JerJer | ok cool |
| 04:05.39 | MikeJ[Laptop] | err.. #mississippi? |
| 04:05.54 | JerJer | not enough issi's |
| 04:06.03 | Qwell | he's using irssi |
| 04:06.11 | Qwell | so its okay |
| 04:06.11 | JerJer | missississississippi |
| 04:06.33 | MikeJ[Laptop] | I just heard from a friend today who is in New Orleans still |
| 04:06.40 | twisted | MikeJ[Laptop], STILL?\ |
| 04:06.44 | MikeJ[Laptop] | yep |
| 04:06.52 | MikeJ[Laptop] | in french 1/4 |
| 04:07.07 | MikeJ[Laptop] | she rode the storm out in a hotel her boyfriend works at |
| 04:07.15 | MikeJ[Laptop] | said that the building took it pretty good |
| 04:07.16 | twisted | whoa |
| 04:07.20 | twisted | still |
| 04:07.24 | twisted | they're ordering everyone out |
| 04:07.37 | MikeJ[Laptop] | there is some water on the first floor.. but the upper levels of the hotel are fine |
| 04:07.46 | MikeJ[Laptop] | yeah.. they have the dog and everything there.. |
| 04:08.00 | MikeJ[Laptop] | they are waiting for the military vehicles to come get them out. |
| 04:08.20 | MikeJ[Laptop] | becuase there is no way to get out otherwise |
| 04:08.40 | MikeJ[Laptop] | I think they are coming up to detroit for a bit |
| 04:08.43 | Qwell | its expected to be 15,000-20,000 per day |
| 04:08.54 | MikeJ[Laptop] | ? |
| 04:08.54 | Qwell | sorry, just saw that, had to throw in my 2c/min :p |
| 04:09.05 | Qwell | people out per day via military vehicles |
| 04:09.16 | MikeJ[Laptop] | yeah |
| 04:09.25 | SwK | MikeJ[Laptop]: Detroit has plenty of vacant realestate for them doesnt it? |
| 04:09.26 | twisted | this so sucks. |
| 04:09.39 | MikeJ[Laptop] | the good news is the french 1/4 faired pretty good. |
| 04:09.55 | MikeJ[Laptop] | the rest of New Orleans is pretty much no longer there |
| 04:10.05 | MikeJ[Laptop] | did you see the pictures of downtown |
| 04:10.08 | SwK | so Mardi Gras is saved! |
| 04:10.16 | MikeJ[Laptop] | the freeways are under 15 feet of water |
| 04:10.33 | MikeJ[Laptop] | yes, mardi gras is saved.. maybe not next year... |
| 04:11.18 | MikeJ[Laptop] | last I heard, 3-6 months before the city will be dried out and starting to have basic services again |
| 04:11.21 | JerJer | or when Hurricane Lee blows through next week |
| 04:11.59 | twisted | lee is headed towards new brunswick |
| 04:13.36 | *** join/#asterisk ManxPwr (i=eric@1Cust855.an1.dfw28.da.uu.net) |
| 04:13.47 | *** join/#asterisk newmember (n=newmembe@S010600a0c93dce87.cg.shawcable.net) |
| 04:14.04 | MikeJ[Laptop] | ManxPwr, you lost some vowels |
| 04:14.33 | Qwell | ManxPwr: you'll appreciate this. I saw a lady named Maxine Power on TV the other day |
| 04:14.47 | twisted | ManxPwr, how you faring? |
| 04:15.45 | ManxPwr | I could be better |
| 04:16.05 | twisted | ManxPwr, any way we can help? |
| 04:16.41 | ManxPwr | twisted, once I confirm that my house no longer exists, I'll start begging for donations to the ManxPower Hurricane Recovery Fund. |
| 04:17.08 | MikeJ[Laptop] | ManxPower, I herd from a friend who is sill in the french quarter... |
| 04:17.09 | ManxPwr | I may also start begging for consulting work. The customer I get 90% of my income from is in New Orleans |
| 04:17.23 | twisted | ManxPwr, i'll be glad to help you out in that aspect... may not be a whole lot, but i'll definitely help you out with what I can |
| 04:17.25 | MikeJ[Laptop] | aparently it faired ok |
| 04:18.10 | ManxPwr | MikeJ[Laptop], Last I heard the french quarter was burning |
| 04:18.19 | MikeJ[Laptop] | when? |
| 04:18.33 | ManxPwr | sometime today. |
| 04:18.38 | ManxPwr | Who knows if that's accurate. |
| 04:18.42 | Har|ock | time to build a new city further inland |
| 04:19.43 | *** join/#asterisk spootnick (n=julio@50.118.233.220.exetel.com.au) |
| 04:19.44 | ManxPwr | Both of my lovers work for the tourism office of New Orleans |
| 04:20.11 | *** join/#asterisk santiago (n=santiago@63.245.86.163) |
| 04:21.09 | ManxPwr | Overall I'm pretty depressed. |
| 04:21.20 | MikeJ[Laptop] | where did you live? |
| 04:21.54 | ManxPwr | All reports I get about where I live (Waveland, MS -- 20 miles west of Gulfport MS) is that the city doesn't relly exist anymore. |
| 04:22.13 | MikeJ[Laptop] | there are videos of it.. |
| 04:22.18 | MikeJ[Laptop] | how far from the water? |
| 04:22.39 | ManxPwr | 1 miles, but there's a railway embankment that acts like a levie. |
| 04:22.43 | ManxPwr | between me and the water |
| 04:22.44 | MikeJ[Laptop] | are here train tracks about a mile up from the .. |
| 04:22.45 | MikeJ[Laptop] | yeah |
| 04:22.51 | MikeJ[Laptop] | there is a video |
| 04:22.58 | MikeJ[Laptop] | which side of the tracks |
| 04:23.03 | MikeJ[Laptop] | let me find it |
| 04:23.06 | ManxPwr | where I am, the train tracks are about 1/4 mile from the water |
| 04:24.23 | ManxPwr | At this point we are a wandering tribe. Me, Sam, Matthew, a dog and 2 cats. |
| 04:24.53 | ManxPwr | Stayed in Jackson MS for 2 days, arrived in Bivins TX (near Texarcana TX) last night. |
| 04:25.10 | MikeJ[Laptop] | it's on msnbc.com |
| 04:25.13 | ManxPwr | Anyway, I don't want to talk about it anymore or i'll get too upset. |
| 04:25.15 | MikeJ[Laptop] | still looking for it. |
| 04:25.22 | MikeJ[Laptop] | it was 9min long |
| 04:25.45 | *** join/#asterisk r0m (n=SysOp@a81-84-68-89.cpe.netcabo.pt) |
| 04:26.08 | ManxPwr | anything interesting with 1.2 in the past 4 days? |
| 04:26.31 | twisted | ManxPwr, 1.2 beta 1 was released |
| 04:27.49 | ManxPwr | nod, saw that. |
| 04:29.49 | MikeJ[Laptop] | ManxPwr, ok... foun the video |
| 04:30.09 | ManxPwr | BTW, if anyone from Teliax is online, now would be a good time for them to /msg me. |
| 04:30.24 | MikeJ[Laptop] | go to msnbc, click onthe video.. on the video search bar go to secials-->hurricane katrina.. then select gulf coast devasation |
| 04:31.11 | ManxPwr | no thanks. |
| 04:31.34 | ManxPwr | I'm having enough trouble not totally freaking out. |
| 04:32.37 | ManxPwr | I just wish godaddy.com would load (my domain registrar) |
| 04:33.43 | pauldy | a nomad with irc access how appropriate |
| 04:33.45 | *** join/#asterisk cgcorea (n=cgcorea@205.240.200.116) |
| 04:33.56 | ManxPower | pauldy, I signed up with PeoplePC today. |
| 04:34.17 | pauldy | you have to be able to at least admit the irony |
| 04:34.27 | pauldy | even if the wounds are still fresh |
| 04:34.51 | Rowters | anyone has HT496 firmware? I cant seem to get it from gs webpage |
| 04:35.02 | Rowters | and I have an old version 1.0.0.8 |
| 04:35.23 | pauldy | looks like they have done some updates without updating their firmware page |
| 04:35.47 | Katty | beep |
| 04:35.51 | Rowters | yeah.. I will call tomorrow.. |
| 04:36.32 | pauldy | http://www.grandstream.com/DOWNLOAD/FIRMWARE/HT386_488_496/ |
| 04:36.36 | pauldy | hrm a nice mepty dir |
| 04:37.21 | pauldy | http://www.grandstream.com/BETATEST/HT488_496_386/ |
| 04:37.23 | *** join/#asterisk dolson (n=dana@toronto-HSE-ppp4301949.sympatico.ca) |
| 04:37.25 | pauldy | might have what you want |
| 04:37.30 | *** join/#asterisk MikeJ__ (n=ircatjer@d14-69-8-30.try.wideopenwest.com) |
| 04:37.55 | Rowters | pauldy, beta ahh |
| 04:38.11 | dolson | does anyone have any good links or info for ARA? I've read the Wiki pages and the stuff in doc/ but I can't figure out what I'm doing wrong |
| 04:38.15 | pauldy | well it is labeled as release |
| 04:38.54 | Rowters | pauldy, I have one with 1.0.2.3 and has bridge mode. what I need.. |
| 04:39.12 | pauldy | I can't even begin to pretend I know |
| 04:39.27 | pauldy | but I can point you around the grandstream site a bit |
| 04:39.30 | r0m | good evening |
| 04:39.32 | Rowters | pauldy, you know all the tftp vars.. mmh let me try to get some info from the http of beta. |
| 04:39.38 | Rowters | pauldy, thanks |
| 04:39.42 | pauldy | np |
| 04:40.04 | ManxPower | PeoplePC is much more AOL like than I expected. |
| 04:40.07 | pauldy | I have a gxp-2000 so I've spent quite a bit of time pounding their site waiting for speakerphone |
| 04:40.16 | ManxPower | Y'all can help. |
| 04:40.24 | ManxPower | I need to get an unpublished number. |
| 04:40.32 | bkw__ | ManxPower, did you make it out with anything? |
| 04:40.38 | dolson | I am getting messages like this on my console: Sep 1 00:39:29 ERROR[14334]: res_config_mysql.c:605 mysql_reconnect: MySQL RealTime: Failed to connect database server on . Check debug for more info. |
| 04:40.50 | ManxPower | bkw_, My laptop and a weeks worth of clothes and my cats. |
| 04:40.58 | pauldy | dolson do you have mysql running? |
| 04:41.00 | bkw__ | ManxPower, well hope your place is ok |
| 04:41.06 | pauldy | and have you checked debug for more info |
| 04:41.15 | ManxPower | bkw_, It prolly isn't. |
| 04:41.24 | bkw__ | oh fun |
| 04:41.45 | dolson | pauldy: yes, I have it running, and I don't see debug anywhere. I have debug level 9, and I don't get any more info than that |
| 04:41.52 | bkw__ | I see no server name |
| 04:41.55 | pauldy | shit happens last night my wife and I setup 4 trashbags worht o fclothes to send to people who didn't have any plus some cash |
| 04:41.56 | ManxPower | bkw_, I live in Waveland MS |
| 04:42.05 | bkw__ | but using res_config_mysql.c is a shot in the foot in the first place |
| 04:42.07 | bkw__ | its evil |
| 04:42.10 | ManxPower | bkw_, all my voip stuff is offline, all my domains are off line |
| 04:42.23 | bkw__ | ManxPower, man that sucks |
| 04:42.34 | ManxPower | fortunatly my domains are on a server north of new orleans |
| 04:42.37 | pauldy | dolson, if you run asterisk with asterisk -c -vvvv -dddd you will get some more console output |
| 04:43.06 | ManxPower | Can anyone here help me with finding an unpublished number? |
| 04:43.12 | Qwell | ManxPower: for peoplepc? |
| 04:43.16 | *** join/#asterisk MikeJ[Laptop] (n=ircatjer@d14-69-8-30.try.wideopenwest.com) |
| 04:43.16 | Qwell | erm, nm |
| 04:43.18 | Qwell | unrelated. |
| 04:43.40 | ManxPower | for family of missing friends |
| 04:43.42 | pauldy | ManxPower, you want to find someones unpublished number or you want to find yourself a number that is unpublished |
| 04:43.43 | bkw__ | ManxPower, who you lookin for? |
| 04:43.52 | dolson | bkw__: I'm sorry. I just installed the 1.2 beta, and I've tried to follow the info in the wiki and docs... (I have been running 1.0.x for months, so I'm not really a newbie, in case that matters) |
| 04:44.05 | ManxPower | The listing would be for last name "Stark" in "Splendora TX" |
| 04:44.08 | bkw__ | dolson, res_config_mysql is just lame |
| 04:44.17 | pauldy | best thing I could think of doing is contacting th red cross |
| 04:44.24 | bkw__ | ManxPower, when you call info its unlisted? |
| 04:44.29 | pauldy | I can only imagine they are maintaining databases right now of people |
| 04:44.41 | Qwell | ManxPower: not Bob? |
| 04:45.00 | *** join/#asterisk websae (i=websae@207-118-143-134.dyn.centurytel.net) |
| 04:45.00 | ManxPower | bkw_, correct |
| 04:45.27 | bkw__ | ManxPower, and the lady on the phone don't get that its an emergency? the nerve of people at a telco actually doing their jobs |
| 04:45.33 | bkw__ | the ONE time you need them to not do that.. they refuse |
| 04:45.37 | websae | criptos: you in here? |
| 04:45.44 | Qwell | ManxPower: is the first name Bob? |
| 04:45.55 | ManxPower | we have no idea. |
| 04:45.56 | pauldy | can they not patch you through? even if the number is unlisted |
| 04:45.57 | bkw__ | ManxPower, how big is the town? and do you have an address? |
| 04:46.11 | ManxPower | no. they are the parents of the missing people. |
| 04:46.19 | pauldy | are they religious? |
| 04:46.20 | dolson | pauldy: I restarted * with that command, and when the other server tries to register, it gives the same exact thing, this time it says Asterisk Urgent Handler at the end though. |
| 04:46.23 | ManxPower | wanted to find out of they made it to their family. |
| 04:46.32 | bkw__ | I would call the police dept. in that town |
| 04:46.34 | Qwell | ManxPower: see msg. thats one hit |
| 04:46.40 | bkw__ | and tell them that you need to get in touch with them |
| 04:46.44 | bkw__ | ASAP and give them the name and such |
| 04:46.47 | bkw__ | see if they can assist |
| 04:46.59 | ManxPower | Qwell, thanks |
| 04:47.06 | pauldy | dolson, I don't know nearest I would think is you are missing a database or relevant table info |
| 04:47.38 | *** join/#asterisk Rowter (n=SilverDr@dsl-201-129-89-96.prod-infinitum.com.mx) |
| 04:47.41 | Qwell | ManxPower: the population of Splendora is only about 1500 |
| 04:47.52 | Qwell | 1275 in 2000 |
| 04:48.02 | pauldy | never heard of splendora and I've lived in texa sall my life |
| 04:48.04 | ManxPower | Qwell, Yeah. It's a small town. |
| 04:48.21 | Qwell | so, chances are, even if it isn't the right number... |
| 04:48.33 | pauldy | they know who youa re looking for |
| 04:48.33 | Qwell | "Do you know the number for any other Stark's in Splendora?" |
| 04:48.54 | dolson | pauldy: do you know where to specify the mysql db user/pass and whatnot in the config files? from reading the wiki, I am under the impression to put the info into iax.conf, sip.conf, etc. but does it go under [general] or somewhere else? |
| 04:49.29 | Delta34 | hi manxpower, can you help me with this warning i am getting from * |
| 04:49.38 | MRH2 | is this likeky to cause a problem: putting externip=address from sip.conf as a name in hosts file pointing to 127.0.0.1 |
| 04:49.40 | ManxPower | Delta34, no. |
| 04:49.43 | pauldy | dolson, I think there is an sql file |
| 04:49.46 | pauldy | one second |
| 04:49.46 | ManxPower | I'll be afk for a bit |
| 04:50.23 | pauldy | cdr_mysql.conf |
| 04:51.00 | dolson | pauldy: isn't that only for call data records? |
| 04:51.14 | *** join/#asterisk shido6 (n=shido6@d57-87-253.home.cgocable.net) |
| 04:51.22 | pauldy | apparently is works for real time configuration data also |
| 04:53.30 | MRH2 | it seems asterisk really hates starting up sip when the externip is unavailable |
| 04:56.11 | dolson | ok, that didn't work either. same thing happens. I also tried adding noload => res_mysql_config.so |
| 04:56.27 | Delta34 | anybody using cisco phones here? |
| 05:00.15 | dolson | bkw__: if I am supposed to not use res_config_mysql, then what do you recommend? and why not mysql? |
| 05:01.13 | Qwell | kram: evening |
| 05:01.45 | kram | evening qwell |
| 05:01.48 | pygrammer | Let's talk VoIP. |
| 05:05.55 | pygrammer | Let's talk VoIP. Let's talk VoIP. Let's talk VoIP. Let's talk VoIP. Let's talk VoIP. |
| 05:06.11 | *** join/#asterisk tugalone (n=tugalone@pcp0010318441pcs.avenel01.nj.comcast.net) |
| 05:08.02 | Qwell | pygrammer: ok, so these two IP phones walk into a bar, right |
| 05:08.30 | dolson | oh well.. I need sleep. gotta work in 6 hours. thanks for your help, pauldy and bkw__ |
| 05:08.33 | *** part/#asterisk dolson (n=dana@toronto-HSE-ppp4301949.sympatico.ca) |
| 05:09.01 | pygrammer | Qwell: yeah, what do they say? |
| 05:09.08 | pygrammer | err |
| 05:09.09 | pygrammer | the bartender* |
| 05:09.13 | Qwell | pygrammer: nothing, they just walked in |
| 05:09.21 | pygrammer | OMG HAHAHAHAHHAAHAHHAHHAh |
| 05:09.22 | Qwell | then they sat down for a drink |
| 05:09.22 | pygrammer | AHAHHAHAHAHHAH |
| 05:09.24 | pygrammer | ROFLMFAO |
| 05:09.49 | pygrammer | wow |
| 05:10.03 | pygrammer | callpacket.com is offering free voip service like stanaphone, but you get 100 free outbound minutes, too |
| 05:10.11 | pygrammer | i've been hunting down free outbound minutes :) |
| 05:10.18 | pygrammer | plus, they have really good prices on the 7960 |
| 05:10.20 | pygrammer | $309 |
| 05:10.26 | pygrammer | + $10 for fedex ground |
| 05:10.27 | pygrammer | pretty good |
| 05:10.38 | pygrammer | or the 7905 for $139 |
| 05:10.56 | *** join/#asterisk dasuberdavid (n=egg@pcp01534754pcs.huntsv01.al.comcast.net) |
| 05:11.41 | rabelais | hod do I fix the "found no matching user" problem for an incoming sip call? |
| 05:11.55 | rabelais | no matching peer or user, sorry |
| 05:12.23 | pygrammer | "Main screen after logining:" << on their proprietary softphone details page |
| 05:12.30 | pygrammer | hahaha, stupid Indian outsourcing |
| 05:13.10 | *** join/#asterisk elwin (n=yellowsn@dsl-213-134-245-123.solcon.nl) |
| 05:14.59 | pygrammer | BlastMe SM |
| 05:15.00 | pygrammer | BlastMe SM is a customer favorite. BlastMe SM rings all the numbers in your Ring List simultaneously. The call will connect to the first phone that's picked up. |
| 05:15.08 | pygrammer | BlastMe sounds so fucking hilarious |
| 05:16.39 | pygrammer | I bet a lot of these voip providers use Asterisk for all their features |
| 05:16.43 | pygrammer | they are just loaded with features |
| 05:16.51 | pygrammer | many of them are exactly what asterisk provides out of the box |
| 05:16.56 | Delta34 | can someone mind telling me why I am getting this warning, I been trying to figure this out for days now |
| 05:16.58 | pygrammer | they probably use SER and * |
| 05:17.09 | Delta34 | WARNING[3924]: chan_sip.c:701 retrans_pkt: Maximum retries exceeded on call |
| 05:17.31 | Delta34 | it only occurs from a cisco phone |
| 05:18.54 | pygrammer | hahaha |
| 05:19.12 | pygrammer | i called callpacket's 24/7 phone support and i got a lame TTS female voice |
| 05:19.19 | pygrammer | how cheap -- they couldn't even hire a real voice |
| 05:19.22 | pygrammer | it sounds terrible |
| 05:19.41 | pygrammer | so robotic |
| 05:19.52 | pygrammer | Delta34: i heard about that problem on another phone |
| 05:19.55 | pygrammer | i think it's related |
| 05:20.37 | pygrammer | it's a feature that sends sip messages to Asterisk *on each digit* to see if Asterisk recognizes them, but asterisk simply sends back 484 Address Incomplete messages if it doesn't, and then the phone gets confused and then Asterisk gives that warning |
| 05:20.55 | Delta34 | i'm just doing a basic sip call using the dial(sip/4140,20) |
| 05:21.02 | pygrammer | so turn off that functionality on your phone |
| 05:21.06 | pygrammer | hm |
| 05:21.12 | pygrammer | what model? |
| 05:21.15 | pygrammer | 7940/60? |
| 05:21.17 | Delta34 | 7960 |
| 05:21.30 | pygrammer | (btw, how are you liking that phone?) |
| 05:21.35 | *** join/#asterisk bluesky (n=maxgluck@200.109.166.172) |
| 05:23.12 | Delta34 | when i do a sip debug, after the 102 CANCEL, i get this Retransmitting #1 (no NAT): |
| 05:23.12 | Delta34 | SIP/2.0 487 Request Terminated |
| 05:23.35 | Delta34 | it retransimit to #5 |
| 05:23.52 | Delta34 | then the maximum retry warning |
| 05:25.12 | bluesky | has anyone that uses lcdial encountered problems with last version from cvs? i'm getting duplicate dial string on second attempts to dial! |
| 05:26.11 | Piranha- | is the Beta worth using? |
| 05:26.50 | bluesky | asterisk? |
| 05:26.54 | pygrammer | I don't know, Delta34; i haven't seen that problem with a cisco |
| 05:26.58 | pygrammer | bluesky: what is lcdial? |
| 05:27.03 | Piranha- | yea |
| 05:27.19 | Piranha- | as in stability wise |
| 05:27.26 | bluesky | if your going to use realtime i believe it is required |
| 05:27.39 | Piranha- | realtime? |
| 05:27.44 | bluesky | lcdial? least cost dialing |
| 05:27.51 | Piranha- | ahh |
| 05:28.08 | Piranha- | err... |
| 05:28.14 | pygrammer | ahh |
| 05:28.20 | bluesky | realtime, database configuration and dialplans |
| 05:28.23 | *** join/#asterisk apardo (n=apardo@90.Red-83-46-197.pooles.rima-tde.net) |
| 05:29.30 | *** join/#asterisk dwC` (n=dwc@S0106002078e03b3f.vs.shawcable.net) |
| 05:29.37 | *** part/#asterisk litage (n=nick@203.201.97.169) |
| 05:31.55 | Piranha- | will dialplans and configs from 1.0.9 work in 1.2 |
| 05:32.43 | bluesky | can't tell you, never used stable |
| 05:33.11 | Piranha- | I guess ill find out in the next 2 minutes after its done compiling |
| 05:36.12 | *** join/#asterisk tugalone (n=tugalone@pcp0010318441pcs.avenel01.nj.comcast.net) |
| 05:36.38 | bluesky | probably you'll get deprecated funtions messages |
| 05:37.57 | *** join/#asterisk Romeo (i=TRBILgir@84.18.70.211) |
| 05:40.16 | *** join/#asterisk elwin (n=yellowsn@dsl-213-134-245-123.solcon.nl) |
| 05:49.44 | pygrammer | Anyone used thet Sipura-841? |
| 05:50.48 | Piranha- | seems to work :) |
| 05:51.00 | pygrammer | That's it? :) |
| 05:51.04 | *** join/#asterisk LilConor (n=dovi5988@pool-141-153-233-195.mad.east.verizon.net) |
| 05:51.10 | Piranha- | haha thats to bluesky |
| 05:51.17 | Piranha- | but im sure the sipura-841 works too |
| 05:51.38 | pygrammer | oh |
| 05:51.42 | pygrammer | lol |
| 05:51.46 | pygrammer | you've never tried it? |
| 05:51.49 | Piranha- | nah |
| 05:51.54 | Piranha- | I have the 1001 and 2002 |
| 05:52.22 | Vco | it's an ok phone... |
| 05:52.35 | pygrammer | Piranha-: what are those? sipuras? |
| 05:52.38 | Vco | get strange echo from time to time... |
| 05:52.38 | pygrammer | Vco: have you used it? |
| 05:52.42 | Vco | bu totherwise ok |
| 05:52.48 | Vco | yea..it's my kitchen phone... |
| 05:52.56 | pygrammer | ahh :) |
| 05:52.56 | Vco | what with the mile long cord and all...... |
| 05:52.59 | pygrammer | lol |
| 05:53.07 | Piranha- | yea they're sipuras |
| 05:53.07 | pygrammer | so your whole house is voip? |
| 05:53.13 | Vco | ya |
| 05:54.05 | Vco | 2 fxs ports in the tdm card and 2 ports on the ata386 |
| 05:54.35 | pygrammer | Vco: why would you need fxs ports if you're using voip phones? |
| 05:54.47 | Vco | cordless |
| 05:55.02 | Vco | and some other cheap handsets.. |
| 05:55.15 | Vco | don't need $100+ phones everywhere.. |
| 05:55.29 | pygrammer | ah |
| 05:55.44 | pygrammer | cordless pots phones are still $50 |
| 05:55.55 | Vco | yup.. |
| 05:56.04 | Vco | 5.8 motorola was $50 |
| 05:56.10 | *** join/#asterisk JohnJacob (n=JohnJaco@pcp0011543387pcs.mainf01.in.comcast.net) |
| 05:56.29 | Piranha- | my 5.8 panasonic was about 40-50 |
| 05:56.31 | Piranha- | really good |
| 05:56.35 | Vco | find the quality running off the fxs directly from the server far more stable than sip phone |
| 05:56.36 | *** join/#asterisk criptos (n=criptos@201.135.121.4) |
| 05:56.46 | Piranha- | tga500cf |
| 05:57.08 | Vco | so.... |
| 05:57.13 | Vco | i'm sitting here... |
| 05:57.34 | Vco | starting at the "confirm order" button for this UTStarCom F1000 WIFI VOIP Phone |
| 05:57.40 | Vco | for like... 15 minutes |
| 05:57.57 | Vco | "do i really need this thing..........?" |
| 05:58.05 | Vco | anyone had any luck with them? |
| 05:58.10 | pygrammer | Vco: omfg, that's *exactly* what i was doing two nights ago |
| 05:58.15 | pygrammer | for a polycom ip501 |
| 05:58.30 | pygrammer | and you can bet i didn't get it -- yet |
| 05:58.32 | pygrammer | "_ |
| 05:58.34 | pygrammer | :)* |
| 05:58.42 | Vco | well...there some wireless kit i was wanting as well... |
| 05:58.44 | pygrammer | i'm happy i didn't, because i'm still thinking about which to get |
| 05:58.53 | Vco | which would also make the phone more usefull |
| 05:59.02 | Piranha- | how much is taht voip phone vco |
| 05:59.12 | Vco | $169 |
| 05:59.26 | Piranha- | kinda steep |
| 05:59.38 | Piranha- | you in canada too? |
| 05:59.39 | Vco | i'd like to test it out a bit... |
| 05:59.42 | Vco | ya |
| 05:59.46 | Supaplex | 0.99 $ |
| 05:59.56 | Piranha- | sc? wheres that |
| 05:59.58 | Vco | on a good day |
| 06:01.14 | Vco | sc.... |
| 06:01.16 | Vco | hmm... |
| 06:01.20 | Vco | good question.. |
| 06:01.30 | Piranha- | haha im wonderingwhat the initals are for |
| 06:01.41 | Vco | in in wpg |
| 06:01.45 | Piranha- | ah |
| 06:01.46 | Vco | i'm in...rather |
| 06:02.03 | Piranha- | ours is fm |
| 06:02.21 | Piranha- | well my 2 neighbours connections do :p |
| 06:05.09 | criptos | hi everyone... |
| 06:05.15 | *** join/#asterisk flankk (n=flankk@S010600e029837f7e.ok.shawcable.net) |
| 06:05.28 | Rowter | its posible to make a hardreset on a gs100 to tftp download the firmware again? |
| 06:05.30 | criptos | Which one do you prefer: snom or grandstream? :) |
| 06:07.12 | *** join/#asterisk razu (n=razu@fw.voicenet.ee) |
| 06:11.13 | *** join/#asterisk gdsm (n=gdsm@e1-1.ns500-1.ts.milt.as9105.net) |
| 06:17.37 | *** join/#asterisk point (n=point@80.80.102.2) |
| 06:18.16 | *** join/#asterisk tzafrir_ (n=tzafrir@bzq-179-75-202.cust.bezeqint.net) |
| 06:22.23 | *** join/#asterisk darkskiez (n=darkskie@host86-133-147-187.range86-133.btcentralplus.com) |
| 06:23.06 | Uberbot | generalhan: sorry I missed you. |
| 06:23.42 | Delta34 | anybody using cisco 7.5 on 7960 phone? I think there's a bug |
| 06:27.27 | *** join/#asterisk vlrk (n=vlrk@59.93.64.128) |
| 06:31.01 | *** part/#asterisk flankk (n=flankk@S010600e029837f7e.ok.shawcable.net) |
| 06:33.25 | *** join/#asterisk Beccara (n=Beccara@219.89.208.80) |
| 06:35.18 | *** join/#asterisk sanleng (n=david_la@222.66.56.50) |
| 06:35.41 | sanleng | hi,everyone |
| 06:36.22 | mog_home | hi sanleng |
| 06:36.57 | *** part/#asterisk criptos (n=criptos@201.135.121.4) |
| 06:38.41 | *** join/#asterisk nvrs (i=RUR@69.156.154.49) |
| 06:40.39 | opus_ | hello |
| 06:41.30 | mog_home | hello |
| 06:43.47 | *** join/#asterisk jiro5281 (n=jiro5281@203.131.137.76) |
| 06:43.51 | drray | Delta - I'm using 7.5 sip on my 7960 |
| 06:44.09 | sanleng | hehe |
| 06:44.43 | sanleng | how can i use asterisk with ser? |
| 06:45.00 | mog_home | point people to asterisk from ser |
| 06:45.14 | point | who said point |
| 06:45.58 | *** join/#asterisk Bonzai090 (n=pirch@wbs-146-190-212.telkomadsl.co.za) |
| 06:46.05 | Bonzai090 | hi all |
| 06:46.15 | sanleng | i don't understand you, |
| 06:46.25 | Bonzai090 | why cant i transfer calls from a sip phone or even x-lite ? |
| 06:46.37 | mog_home | i did |
| 06:46.44 | Bonzai090 | its a fresh install |
| 06:47.00 | shido6 | how do you like 7.5 |
| 06:47.00 | point | ;) |
| 06:47.03 | shido6 | I still have 7.4 |
| 06:47.08 | shido6 | did they fix the clock ? |
| 06:47.19 | *** join/#asterisk bendy24 (n=slb@50.tender.1meg.golden.net) |
| 06:47.30 | *** part/#asterisk point (n=point@80.80.102.2) |
| 06:47.35 | drray | 7.5 seems fine |
| 06:47.53 | sanleng | asterisk must be registered as an agent with ser? |
| 06:48.04 | shido6 | do you ahve the 7.5 firmware? |
| 06:48.19 | drray | I downloaded it from cisco |
| 06:48.27 | shido6 | good answer :) |
| 06:49.23 | Bonzai090 | ...... |
| 06:49.40 | Bonzai090 | any answers or should i can asterisk and dust off my 3com pbx?.. |
| 06:49.46 | sanleng | mog_home, can you answer me ? |
| 06:49.57 | mog_home | huh |
| 06:49.58 | mog_home | sorry |
| 06:50.01 | mog_home | i was elsewhere what |
| 06:50.27 | shido6 | Bonzai090 ? |
| 06:50.51 | shido6 | if you bring that damn 3com pbx online Im gonna send you a nuke packet ... put that thing on ebay for some other unfortunate bastard :) |
| 06:50.57 | shido6 | whats the problem, Bonzai090 ? :) |
| 06:52.05 | Bonzai090 | shido6 i installed a nice new asterisk pbx but i cant transfer calls |
| 06:52.17 | Bonzai090 | not on a sip phone not on a x-lite soft phone |
| 06:52.28 | shido6 | config issue |
| 06:52.33 | shido6 | pastebin ur sip.conf and extensions.conf |
| 06:52.36 | shido6 | pastebin.ca |
| 06:52.38 | shido6 | ~pastebin |
| 06:52.39 | jbot | pastebin is, like, a place to paste your stuff without flooding the channel - try http://pastebin.ca/ |
| 06:52.40 | *** join/#asterisk djin_ib (n=djin_ib@gridfox.xs4all.nl) |
| 06:53.20 | Bonzai090 | shido6 ok shibo lemme paste the sip and extentions.conf files. |
| 06:54.10 | *** join/#asterisk stkn_ (i=nobody@gentoo/developer/pdpc.active.stkn) |
| 06:56.13 | Bonzai090 | ok asterisk fired up lets try to paste that stuff shido6 |
| 06:56.58 | Uberbot | shido6: Do you have an ETA for the pap2-na? (H) |
| 06:57.14 | *** join/#asterisk TheCops (n=mdb@206-248-136-187.dsl.teksavvy.com) |
| 06:57.35 | shido6 | Uberbot - yeah - Im going to need to issue a refund unless you can wait a couple more days |
| 06:57.39 | *** join/#asterisk Borgon (n=L3orgon@67-50-80-66.br1.tbr.ga.frontiernet.net) |
| 06:57.41 | Borgon | hello |
| 06:58.04 | shido6 | I would rather not get into the story - |
| 06:58.07 | Borgon | What are the minimum requirements to run asterisk.. as far as for bandwith/ram? |
| 06:58.14 | shido6 | what do you want to do Borgon |
| 06:58.17 | Uberbot | Days I can wait. |
| 06:58.19 | shido6 | how many calls do you want to use? |
| 06:58.22 | Borgon | shido6: just 1 |
| 06:58.26 | shido6 | ha |
| 06:58.29 | Borgon | shido6: personal use |
| 06:58.31 | shido6 | you can run it on a watch in java then |
| 06:58.34 | shido6 | or a router |
| 06:58.44 | shido6 | jj has asterisk on his wrt linksys router :) |
| 06:58.56 | Borgon | shido6: am looking to get a vps server for 6.99/mo to use it under |
| 06:59.05 | shido6 | v pee esss |
| 06:59.08 | shido6 | for 1 call? |
| 06:59.13 | Borgon | the stats are 128mb/256Kpbs |
| 06:59.24 | Borgon | yeah, so i wasnt sure if it was going to be enough or not |
| 06:59.26 | shido6 | you may want to wait for members.nufone.net :) |
| 06:59.43 | shido6 | jj is almost finished with it |
| 06:59.51 | shido6 | no more colo's needed |
| 06:59.56 | Delta34 | drray are u having any issues with it? |
| 07:00.01 | Borgon | jj? |
| 07:00.05 | Uberbot | shido6: are you thinking under a week? |
| 07:00.07 | shido6 | JerJer |
| 07:00.12 | shido6 | YES uberbot |
| 07:00.25 | Uberbot | No problem, then. Thanx. |
| 07:00.52 | Delta34 | i found out so far that I get this warning message on 7.5 and not 6.3 |
| 07:00.55 | Uberbot | Not trying to be a pest. :-D |
| 07:01.03 | shido6 | I know |
| 07:01.17 | shido6 | I bitched at my supplier very loudly |
| 07:01.41 | Delta34 | WARNING[3924]: Maximum retries exceeded on call, only occurs from a 7.5 cisco phone |
| 07:03.00 | *** join/#asterisk jeffgus (n=jeffgus@2002:d856:c704:0:0:0:0:1) |
| 07:04.03 | Rowter | on show agents what does available at stand for? |
| 07:05.13 | shido6 | waiting on that pastebin.... |
| 07:05.27 | Bonzai090 | shido6 sorry its taking to long |
| 07:05.30 | *** join/#asterisk bjohnson (n=bjohnson@i216-58-59-175.cybersurf.com) |
| 07:05.33 | shido6 | :) |
| 07:05.52 | Bonzai090 | copy and paste via ssh is a bit hard lol |
| 07:05.52 | shido6 | I had a bottle of lipobitan so Im awake |
| 07:06.30 | shido6 | hehehe |
| 07:06.33 | shido6 | sz |
| 07:06.35 | shido6 | scp |
| 07:10.26 | *** join/#asterisk meppl (n=mephisto@p54AAF5F3.dip.t-dialin.net) |
| 07:10.54 | Borgon | So how much ram/bandwith is sufficient for a personal pbx server for 1 or 2 calls? |
| 07:15.34 | pygrammer | Borgon: i'mrunning a 400 mhz IPII with 128 megs of ram, though it's consistently using 122 megs of that 128 |
| 07:15.42 | shido6 | not much |
| 07:15.44 | shido6 | less than 80k |
| 07:15.44 | pygrammer | i could use a bit more, but it's not required |
| 07:15.55 | shido6 | about 64kish |
| 07:16.06 | pygrammer | depends on your codec ... |
| 07:16.15 | shido6 | make it 128k ish to be safe |
| 07:16.27 | shido6 | gsm, g729 with email and chat downloads :) |
| 07:16.30 | Borgon | 64MB of ram can handle it? |
| 07:16.38 | Borgon | its a vps |
| 07:16.47 | opus_ | pygrammer how does that run you |
| 07:17.20 | pygrammer | opus_: it's ok; i wouldn't notice any problems as this is the only thing i've had |
| 07:17.24 | pygrammer | there is occasionally some stuttering |
| 07:17.31 | pygrammer | which may be a RAM deficiency |
| 07:17.44 | pygrammer | 256 megs is often recommended |
| 07:17.53 | pygrammer | i don't know though |
| 07:18.10 | Borgon | because its going to be the main process running when am using it |
| 07:18.25 | pygrammer | it is here ... |
| 07:19.00 | opus_ | i run asterisk on a pentium prescott and a pentium II 300mhz |
| 07:19.11 | pygrammer | dual procs? |
| 07:19.15 | opus_ | nope. |
| 07:19.18 | pygrammer | oh, each |
| 07:19.22 | pygrammer | the prescott is a p4? |
| 07:19.30 | opus_ | i think the pentium II is actually a celeron |
| 07:19.35 | opus_ | prescott yeah its 3.0ghz |
| 07:19.41 | pygrammer | that's a nice proc :) |
| 07:19.46 | pygrammer | how much ram is in that PII? |
| 07:19.52 | opus_ | 512mb |
| 07:19.56 | pygrammer | oh, heh |
| 07:20.03 | pygrammer | problem is, this is an old deskpro |
| 07:20.15 | pygrammer | and i don't know how much RAM its mobo can handle and what type |
| 07:20.15 | opus_ | .wav files, .mp3 files slow as hell. text to speech is like, 1 channel maximum and very very lagged |
| 07:20.45 | opus_ | why not get a cheap athlon if your on a budget. or sempron |
| 07:20.54 | pygrammer | yea |
| 07:20.54 | pygrammer | i could |
| 07:20.57 | opus_ | fry's electronics has $179 semptron 2200 |
| 07:21.00 | pygrammer | anything would be better than this |
| 07:21.03 | Borgon | asterisk will be the main process.. just voip with iax and 1 person talking at a time.. on 64ram with 128mbswap hold? |
| 07:21.19 | pygrammer | 64 may not cut it; it might |
| 07:21.20 | Borgon | i want to click order, but i dont want to get it and then it will suck |
| 07:21.21 | pygrammer | but it's a stretdch |
| 07:21.24 | pygrammer | *stretch |
| 07:21.25 | Borgon | k |
| 07:21.35 | pygrammer | where are you ordering it, and for how muc? |
| 07:21.42 | opus_ | borgon you'll need fast swap or something |
| 07:21.53 | pygrammer | opus_: yeah, it's just a bummer to waste this system (and spend more $$$, as i need a hardphone, too!) |
| 07:22.07 | opus_ | what are you trying to do? |
| 07:22.57 | pygrammer | opus_: this is just a test/fun/experimentation/personal pbx server |
| 07:23.01 | Borgon | opus_: just have a personal pbx.. on a vps |
| 07:23.05 | pygrammer | it's not for business |
| 07:23.10 | pygrammer | oh, sorry |
| 07:23.12 | Borgon | opus_: yeah its not in productional use |
| 07:23.21 | opus_ | Borgon on what hardware ? |
| 07:23.59 | Borgon | opus_: am not sure.. am just shopping around for cheap vps to hold it.. I could probably run it on my own computer, but i would have ot use vmware to run linux and it will slow my pc |
| 07:24.04 | *** join/#asterisk Gronker__ (n=Gronker2@adsl-220-93-76.ags.bellsouth.net) |
| 07:24.10 | *** join/#asterisk Tabo2 (n=acura0@S010600119503044b.vc.shawcable.net) |
| 07:24.17 | tsume | okay |
| 07:24.24 | tsume | its what the fuck time :) |
| 07:24.26 | tsume | http://zanshin.tsumelabs.com/ |
| 07:24.58 | tsume | I posted all my configs there. Trouble is any calls which come in, and go to queue1, don't ring at the receptionist desk |
| 07:24.59 | *** join/#asterisk sigmounte (n=sigmount@213.251.156.186) |
| 07:25.40 | *** join/#asterisk Snake-Eyes (n=blog@203.201.98.225) |
| 07:26.52 | *** join/#asterisk cavelx (n=aaron@222-152-160-73.jetstream.xtra.co.nz) |
| 07:27.04 | Bonzai090 | shido6 and finaly i got the whole thing pasted. |
| 07:27.07 | Bonzai090 | can yo uplease check |
| 07:27.15 | shido6 | back |
| 07:27.17 | shido6 | gimme the url |
| 07:27.17 | *** part/#asterisk newsmafia (n=newsmafi@wsip-68-15-19-142.sd.sd.cox.net) |
| 07:27.26 | shido6 | lucky i had speech on in this thing, I forgot all about this window |
| 07:27.28 | shido6 | :) |
| 07:27.44 | shido6 | what url did pastebin spit back at ya? |
| 07:28.09 | Bonzai090 | shido6 waiting for it to upload |
| 07:28.12 | Bonzai090 | damn adsl :P |
| 07:28.23 | shido6 | heh |
| 07:29.18 | *** part/#asterisk Gronker__ (n=Gronker2@adsl-220-93-76.ags.bellsouth.net) |
| 07:31.08 | opus_ | is there a SIP stress program that also does RTP ? |
| 07:32.46 | *** join/#asterisk grimse (n=grimse@p5481EFF7.dip.t-dialin.net) |
| 07:33.09 | Bonzai090 | shido6 and i'm still waiting for this dman thing to upload :( |
| 07:33.23 | shido6 | wow |
| 07:33.25 | shido6 | :) |
| 07:33.28 | Bonzai090 | shido6 and finaly |
| 07:33.32 | Bonzai090 | shido6 http://pastebin.ca/21811 |
| 07:34.03 | Bonzai090 | shido6 i pasted the sip.conf and extentions.conf there |
| 07:34.09 | shido6 | ok |
| 07:34.15 | shido6 | its takin a sec to download :) |
| 07:34.23 | shido6 | like its stuck... :) |
| 07:34.29 | gek | quit |
| 07:34.47 | shido6 | there we go |
| 07:34.59 | shido6 | oh no |
| 07:35.05 | shido6 | an asterisk@home victum |
| 07:35.07 | shido6 | u = i |
| 07:35.20 | *** join/#asterisk DrJolo (n=chatzill@host-ip58-113.crowley.pl) |
| 07:35.23 | shido6 | ok what phones do u need to xfer |
| 07:35.24 | shido6 | ? :) |
| 07:35.47 | Bonzai090 | shido6 all the extentions |
| 07:36.05 | Bonzai090 | i need to be able to do form 200 -> 204 or where ever it will end |
| 07:36.21 | Bonzai090 | as long as i can transfer incomming calls and between extentions |
| 07:36.45 | shido6 | whats in extensions_custom.conf |
| 07:36.45 | shido6 | ? |
| 07:36.47 | tsume | okay, I've posted to asterisk-users |
| 07:36.59 | Bonzai090 | shido6 lemme look |
| 07:37.31 | Bonzai090 | shido6 i oly have a extensions_custom.conf.sample |
| 07:37.39 | Bonzai090 | so no custom as yet i presume |
| 07:38.02 | *** join/#asterisk Delvar (n=irc@host-83-146-53-34.bulldogdsl.com) |
| 07:39.28 | hugo-v6 | gd morning * |
| 07:39.53 | Bonzai090 | shido6 also my caller ID dont work all in comming calls tells me is a unknown number |
| 07:40.01 | shido6 | what about extensions_additional.conf? |
| 07:40.29 | Bonzai090 | shido6 lemme look |
| 07:41.16 | Bonzai090 | Bonzai090 uploading it now |
| 07:42.01 | Bonzai090 | err shido6 uploading it now to pastebin |
| 07:42.12 | opus_ | i nee dto figure out a good way to debug this sip problem |
| 07:43.33 | *** join/#asterisk ManagedGI (i=user@p549236D2.dip0.t-ipconnect.de) |
| 07:44.23 | hugo-v6 | opus_: start asterisk with a lot of -v's and a few -d's? |
| 07:44.26 | Bonzai090 | shido6 see anything that needs changing?.. |
| 07:44.39 | opus_ | well, yeah. thats how i notice the bug |
| 07:44.42 | opus_ | that and 'rtp debug on' |
| 07:45.36 | hugo-v6 | well then... |
| 07:46.39 | opus_ | the problem is, when asterisk is behind nat and the client is behind nat, and with a database config, asterisk will send the packets to the phone's inside nat address 1 out of 20 times |
| 07:47.04 | Bonzai090 | shido6 : http://pastebin.ca/21812 for my extensions_additional.conf |
| 07:47.50 | pygrammer | damn this fucking $15 POS headset hurts my ear |
| 07:47.56 | pygrammer | it's so unfuckingcomfortable |
| 07:48.53 | Bonzai090 | pygrammer lol now u know why i give call centre ppl the cheapest headsets hehe |
| 07:48.59 | *** join/#asterisk weazul (n=weazul@82-169-62-42-mx.xdsl.tiscali.nl) |
| 07:49.14 | glm2k | pygrammer: what i do is set the volume to max and let it rest just in front of my ear, above my cheek |
| 07:49.34 | *** join/#asterisk Kumbang_ (n=unknown@167.205.24.5) |
| 07:49.45 | tsume | pygrammer: I have configs to show for my problem =] |
| 07:50.03 | tsume | pygrammer: http://zanshin.tsumelabs.com/ |
| 07:50.06 | Bonzai090 | shido6 can you private me with a solution when you done please. i need to go to the server room for a bit |
| 07:50.11 | *** part/#asterisk ManagedGI (i=user@p549236D2.dip0.t-ipconnect.de) |
| 07:50.24 | shido6 | dont use asterisk@home :) |
| 07:50.30 | tsume | yes, I'm still trying to solve the _same_ problem |
| 07:50.49 | Bonzai090 | shido6 i stay away from asterisk@home lol |
| 07:50.58 | Bonzai090 | i just install AMP for ease of use thats about it |
| 07:51.41 | *** join/#asterisk Krazylegz (n=Krazyleg@pcp0010326949pcs.coatsv01.pa.comcast.net) |
| 07:51.46 | *** part/#asterisk Krazylegz (n=Krazyleg@pcp0010326949pcs.coatsv01.pa.comcast.net) |
| 07:54.16 | *** join/#asterisk greekman (n=alex@host254.209.113.199.conversent.net) |
| 07:56.19 | greekman | hello |
| 07:56.49 | *** join/#asterisk Ogun (n=johangri@h236n2fls34o865.telia.com) |
| 07:56.54 | Bonzai090 | shido6 dont want to rush you any luck?.. :) |
| 07:57.12 | shido6 | too much crap to look through :) |
| 07:57.40 | Bonzai090 | shido6 awww so i wasted time copy and paste?.. |
| 07:57.41 | *** join/#asterisk greekman (n=alex@host254.209.113.199.conversent.net) |
| 07:57.54 | shido6 | no, you could pay someone and still use the url |
| 07:58.06 | Bonzai090 | ok lemme go look for that 3com switch then |
| 07:58.10 | Bonzai090 | thanks for nothing guys |
| 07:58.34 | Bonzai090 | put the 3com pbx back to its old glory least it could transfer calls lol |
| 07:58.36 | shido6 | write your own configs |
| 07:58.56 | Bonzai090 | shido6 i tried thus i use amp cause i dont know the configs fully yet |
| 07:59.06 | shido6 | oh I see |
| 07:59.20 | shido6 | take amp and throw it out of the window till you do :) |
| 07:59.21 | Bonzai090 | ohh well shows how shocking support is on asterisk and how bad the asterisk community is |
| 07:59.38 | shido6 | yeah its pretty horrible, isnt it :) |
| 07:59.49 | Bonzai090 | yip waste of time really |
| 07:59.51 | shido6 | its only 4am |
| 07:59.55 | Bonzai090 | why people bother with it i wont know |
| 08:00.01 | hugo-v6 | Bonzai090: nope. |
| 08:00.09 | Bonzai090 | yeah for you its 10:00 am here |
| 08:00.16 | Bonzai090 | hugo-v6 what u mean nope?... |
| 08:00.42 | hugo-v6 | Bonzai090: if u read the documentation (which is damn good) then its no problem to get it working. but if u want a idiot-safe solution and tried amp, thats the wrong way |
| 08:00.49 | Bonzai090 | a simple problem like call transfer and no one can help jees i dont want to know how bad it will be if i had a bigger problem |
| 08:00.55 | hugo-v6 | Bonzai090: the support and the community. |
| 08:01.10 | *** join/#asterisk meppl (n=mephisto@p54AADA40.dip.t-dialin.net) |
| 08:01.12 | Bonzai090 | hugo-v6 the doc's i got did not help alot |
| 08:01.28 | hugo-v6 | Bonzai090: u are the problem, not the asterisk nor the community o the support here |
| 08:01.33 | Bonzai090 | just showed me the basics so i downloaded amp anbd went with amp |
| 08:01.33 | shido6 | "/usr/src/asterisk/configs/* " |
| 08:01.46 | hugo-v6 | Bonzai090: then u cant read or havent looked at: |
| 08:01.47 | hugo-v6 | ~docs |
| 08:01.49 | jbot | docs is, like, Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk |
| 08:02.19 | hugo-v6 | the 3rd and 4th got me working asterisk without problems |
| 08:02.23 | Bonzai090 | hugo-v6 how hard canit be to just fox a call transfer issue. really |
| 08:03.09 | Bonzai090 | oohh well when i have time one day. ill just put the 3com pbx back and continue with a proper working one for now |
| 08:03.15 | hugo-v6 | Bonzai090: well then. pay someone who makes everything for u if u didnt want to set it up for your own. |
| 08:03.42 | Bonzai090 | hugo-v6 yeah right pay some one that cant even proof he can fix a simple transfer issue |
| 08:04.03 | Bonzai090 | man this is funny stuff maybe i should put it on bash.org lol |
| 08:04.14 | Bonzai090 | anycase enough time wasted lemme go put a proper working pbx back |
| 08:04.34 | hugo-v6 | Bonzai090: u dont get it. go away. |
| 08:04.46 | *** part/#asterisk Uberbot (n=Uberbot@pcp01880954pcs.sandia01.nm.comcast.net) |
| 08:05.44 | pygrammer | Bonzai090: You're retarded. |
| 08:05.47 | hugo-v6 | Bonzai090: but dont blame the support here nor the community which cant do anything about your laziness |
| 08:06.21 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 08:07.05 | puzzled | morning all |
| 08:09.33 | weazul | Bonzai090: yes you sure dont get the picture .... damn go play with avaya or something don't bother us with your complains |
| 08:13.05 | weazul | someone who's awake and familiar with radius and asterisk CVS head? |
| 08:13.10 | *** join/#asterisk djin (n=djin@213-132-172-4.multikabel.nl) |
| 08:13.58 | hugo-v6 | os is at least a document on setup asterisk with radius? would be from itnrest for me also |
| 08:14.06 | hugo-v6 | s/os/or/ |
| 08:14.30 | weazul | voip-info.org says on 24-12-2004: patches have been tested only against asterisk 1.0.2, we're working on making them available for cvs version |
| 08:14.42 | hugo-v6 | uh damn old. |
| 08:14.51 | weazul | yes check this link |
| 08:15.01 | weazul | http://www.voip-info.org/tiki-index.php?page=PortaOne+Radius+auth |
| 08:15.20 | weazul | i'm applying/testing this while we speak |
| 08:15.21 | *** join/#asterisk darkskiez (n=darkskie@194.247.78.146) |
| 08:15.47 | pygrammer | damn voip-info is so slow |
| 08:16.19 | hugo-v6 | weazul: its only from intrest if its smarter than getting billing-infos from csv and configuring with my/pqsql |
| 08:16.39 | *** join/#asterisk soundguy (n=soundguy@watchguard.blendtek.com.au) |
| 08:17.06 | Borgon | Is there a problem running asterisk on your local pc and getting a softphone to connect to it? |
| 08:17.11 | Borgon | locally? |
| 08:17.40 | pygrammer | Borgon: if that's your main pc, yes, it's going to get slow |
| 08:17.50 | hugo-v6 | Borgon: if u set it up right, i dont see a problem |
| 08:18.17 | pygrammer | wow i think voip-info is down or being flooded or something |
| 08:18.20 | Borgon | pygrammer: I was running vmware with a bootable linux distro and asterisk ran fine.. Now that i put my local ip number or localhost/127.0.0.1 doesnt connect to asterisk |
| 08:19.06 | pygrammer | it may run, but that doesn't mean it will run correctly, efficiently, and without noticeable problems |
| 08:19.10 | Borgon | i got iax2 on debug mode and it doesnt detect a connection |
| 08:19.38 | pygrammer | bah stupid voip-info!!! |
| 08:19.39 | pygrammer | it's down |
| 08:20.28 | shido6 | heh |
| 08:20.36 | pygrammer | just when i needed it -- great |
| 08:20.59 | pygrammer | shido6: do you know off the top of your head whether asterisk supports mp3s or wavs or something in Background or Playback? |
| 08:21.02 | pygrammer | wavs are fine |
| 08:21.04 | pygrammer | does it? |
| 08:21.10 | shido6 | wavs |
| 08:21.13 | pygrammer | ok |
| 08:21.15 | pygrammer | great |
| 08:21.16 | shido6 | dunno about mp3s |
| 08:21.51 | pygrammer | yea, i think MP3Player() would work, but i don't know if it acts more like background or playback |
| 08:22.23 | shido6 | playback i believe |
| 08:22.29 | pygrammer | hmm |
| 08:22.30 | pygrammer | ok |
| 08:23.27 | soundguy | Hey. Is it possible with asterisk to answer another extensions phone that is ringing from another extension? Ie: I have incoming calls from one provider going directly to one phone, but sometimes I am not in that room and would like to be able to answer it. IS this possible (using Grandstream BudgeTone phones) |
| 08:23.48 | pygrammer | yay voip-info is back up |
| 08:25.40 | *** join/#asterisk _m_ (n=m@fbta199.fbta.uni-karlsruhe.de) |
| 08:27.55 | *** join/#asterisk mithro (n=tim@c213-100-42-188.swipnet.se) |
| 08:28.31 | hugo-v6 | pygrammer: i was on it as u told it is down. cant approve that |
| 08:30.02 | hugo-v6 | Borgon: btw, why didnt u give that machine a private ip address? like 192.186.191.191? |
| 08:30.23 | hugo-v6 | i think thats a far better idea than using localhost |
| 08:31.26 | Borgon | hugo-v6: am testing different things.. my real ip/192.168.0.101 which si the router ip |
| 08:32.15 | hugo-v6 | well use that |
| 08:34.18 | *** join/#asterisk nagl (n=nagl@rih.zid-nw.wu-wien.ac.at) |
| 08:35.03 | *** join/#asterisk audela (n=audela@82-33-115-145.cable.ubr08.azte.blueyonder.co.uk) |
| 08:36.13 | *** join/#asterisk nesys (n=nesys@81-174-12-111.f5.ngi.it) |
| 08:38.11 | Borgon | hugo-v6: am trying all my local ips and nothing |
| 08:46.46 | *** join/#asterisk yaboo (n=jsirucka@220.245.131.131) |
| 08:47.00 | *** join/#asterisk fenlander (n=fenlande@82.152.81.57) |
| 08:48.27 | pygrammer | omfg |
| 08:48.32 | pygrammer | Audacity has a fucking bug |
| 08:48.57 | pygrammer | when it says illegal directory name if you're creating a directory when you're saving a file, it just closes the whole fucking application without letting you specify a different location |
| 08:49.03 | pygrammer | so i just lost an hour of work |
| 08:49.06 | pygrammer | great -- fuck |
| 08:53.47 | *** join/#asterisk weazul (n=weazul@82-169-62-42-mx.xdsl.tiscali.nl) |
| 08:55.26 | *** join/#asterisk hellagony (n=egutierr@200.121.129.178) |
| 08:58.48 | *** join/#asterisk mbranca (n=matteo@81.208.92.210) |
| 09:00.26 | *** join/#asterisk Tili (n=Tili@202-133-65-176-dialup.sat.net.pk) |
| 09:01.21 | hugo-v6 | Borgon: well u can define a ip on which asterisk should listen on at a few places. |
| 09:02.04 | *** join/#asterisk PupenoL_ (n=pupeno@200.123.183.89) |
| 09:02.30 | *** join/#asterisk Henguei (n=smoka@196.203.249.73) |
| 09:02.40 | Henguei | hi |
| 09:02.51 | *** join/#asterisk fulgas (n=fulgas@213.58.130.46) |
| 09:06.08 | weazul | guys, being busy with radius icw asterisk CVS head i noticed that the patches arent applyed yet and still needed to be applyed manually .... for those who care ;-) |
| 09:07.27 | *** join/#asterisk TheCops (n=mdb@206-248-136-187.dsl.teksavvy.com) |
| 09:10.10 | *** join/#asterisk jollyr (i=user@202.124.138.88) |
| 09:11.00 | thalunil | hmm, is mpg321 version 0.59q fine for asterisk? |
| 09:11.32 | thalunil | i need at least r, right? |
| 09:12.13 | *** join/#asterisk Shoragan (n=shoragan@d072.apm.etc.tu-bs.de) |
| 09:12.16 | *** join/#asterisk apardo (n=apardo@90.Red-83-46-197.pooles.rima-tde.net) |
| 09:14.41 | *** join/#asterisk djin_ib (n=djin_ib@81.58.48.104) |
| 09:15.40 | jollyr | anyone experience in Vierling GSM |
| 09:18.26 | jollyr | i am peering the gsm to my asterisk |
| 09:18.42 | cypromis | ecotel ? |
| 09:19.01 | jollyr | yes |
| 09:19.20 | cypromis | problems with it ? |
| 09:20.13 | jollyr | i used the PRI port connected to my asterisk with sangoma card... |
| 09:20.22 | puzzled | thalunil: you need exactly "r" and nothing else |
| 09:20.27 | jollyr | and its seems both PRI are not talking |
| 09:20.39 | thalunil | fn~puzzled: ok |
| 09:20.51 | thalunil | fn~puzzled: still get this: app_mp3.c:91 timed_read: Poll timed out/errored out with 0 |
| 09:22.01 | hugo-v6 | thalunil: this psybnc prefix sucks ;) btw gruss an die #linux.de gilde. |
| 09:22.01 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 09:22.11 | jollyr | cypromis: Do you think this is ok ? |
| 09:22.19 | thalunil | fn~hugo-v6: sucks big time, i know |
| 09:22.37 | *** join/#asterisk Gunnar (n=gunnar@34.80-203-209.nextgentel.com) |
| 09:22.44 | thalunil | fn~hugo-v6: this damn bitch x translates my text into this prefix shit |
| 09:22.47 | *** join/#asterisk mastix (n=mastix@gw.2n.cz) |
| 09:23.04 | jollyr | . |
| 09:23.08 | blitzrage | evening all |
| 09:23.19 | gordonjcp | hello |
| 09:23.27 | mastix | hi does anyone know about any TDM hardware for asterisk which has hardware codecs on board |
| 09:24.20 | mastix | i need to build a box which will route 30 PRI channels to 30 g.729 or g.723 voip |
| 09:25.12 | cypromis | for the moment you can only use dialogic or aculab |
| 09:25.25 | cypromis | \dialogic driversw are available for a fee from digium |
| 09:25.32 | cypromis | aculab you can get from aculab |
| 09:26.03 | *** join/#asterisk CleanerX (n=nix@nat-ph3-wh.rz.uni-karlsruhe.de) |
| 09:26.15 | thalunil | this error msg sucks: Poll timed out/blablabla.... |
| 09:26.22 | thalunil | the damn mp3 file is just *NOT* present |
| 09:26.44 | mastix | cypromis: was it for me? |
| 09:28.05 | *** join/#asterisk elwin (n=yellowsn@dsl-213-134-245-123.solcon.nl) |
| 09:30.41 | hugo-v6 | thalunil: does every client, since it looks to him as this is the real usernick |
| 09:30.57 | hugo-v6 | uh bad english- |
| 09:31.07 | *** join/#asterisk glLoadIdentity (n=tuyan@81.214.255.57) |
| 09:31.58 | *** join/#asterisk |cleric| (n=dacleric@p548281AF.dip0.t-ipconnect.de) |
| 09:34.13 | *** join/#asterisk atmel (n=vlad@ip68-4-101-199.oc.oc.cox.net) |
| 09:38.22 | *** join/#asterisk RoyK (n=roy@80.239.107.80) |
| 09:39.15 | h3x | damn |
| 09:41.19 | h3x | bkw |
| 09:47.34 | *** join/#asterisk gaggaman (n=leo@host-82-135-28-39.customer.m-online.net) |
| 09:48.45 | *** join/#asterisk azumi (n=xxx@gwcl.ralsko.net) |
| 09:49.30 | *** join/#asterisk Ogun (n=ogun@h236n2fls34o865.telia.com) |
| 09:57.36 | mastix | hi |
| 09:58.15 | mastix | does anybody know wheter i need to buy g.729 codec from digium when i want just try it or is there any evaluation period for that? |
| 09:58.20 | *** join/#asterisk Wayne_ (n=wayne@81.56.250.173) |
| 09:58.49 | *** join/#asterisk zurab (n=zurab@ws2.sitel.com.ua) |
| 10:00.50 | gaggaman | is there any docu about the sccp / skinny protocoll on the net? I'd like to play with the cisco 7970 video functions. |
| 10:02.12 | *** join/#asterisk Mother_ (n=Mother@53.Red-217-126-93.pooles.rima-tde.net) |
| 10:02.39 | Mother_ | hi all |
| 10:04.46 | *** join/#asterisk krypt0n (n=Hunter@84.235.7.60) |
| 10:07.21 | zurab | hello, everybody!!! Could someone explain me how disable VAD at all? I have Asterisk 1.2.0-beta1 and ATA186 (SIP 3.1.2: AudioMode 0x00140014) |
| 10:07.45 | zurab | for alaw |
| 10:09.40 | *** join/#asterisk djin_ib (n=djin_ib@81.58.48.104) |
| 10:10.24 | Borgon | Am running asterisk on my localhost.. when i try to setup a softphone its not connecting with iax... IS there a bindport or some config i need to set ? |
| 10:11.19 | *** join/#asterisk MobileLoic (n=MobileLo@per92-7-82-236-197-96.fbx.proxad.net) |
| 10:17.12 | *** join/#asterisk jollyr_ (i=user@202.124.138.88) |
| 10:22.05 | gaggaman | Borgon did you set up your iax.conf correctly? See http://www.voip-info.org/tiki-index.php?page=Asterisk+IAX+clients |
| 10:22.10 | jack | can I display messages on a cisco 7960 like 'phone x is ringing' ? |
| 10:22.30 | Bonzai090 | ahhh got my 3com pbx up and running |
| 10:23.26 | Bonzai090 | now i can delete asterisk and go to a normal working pbx |
| 10:23.57 | Bonzai090 | damn asterisk cant even transfer calls and the support is worse than our local telecoms guys |
| 10:26.43 | jack | you have to buy the pro edition to transfer calls |
| 10:26.51 | pr0m | asterisk can transfer calls. i do it all the time. |
| 10:27.00 | soundguy | Hey. Is it possible with asterisk to answer another extensions phone that is ringing from another extension? Ie: I have incoming calls from one provider going directly to one phone, but sometimes I am not in that room and would like to be able to answer it. IS this possible (using Grandstream BudgeTone phones) |
| 10:27.28 | gaggaman | soundguy that's call Pickup |
| 10:27.52 | soundguy | uhuh....expand?? |
| 10:28.02 | Bonzai090 | pr0m then why the F cant mine spent like 2 days with little sleep and still no damn luck |
| 10:28.25 | soundguy | need to add t to the extensions line for each phone |
| 10:28.26 | pr0m | what phone are you using? |
| 10:28.30 | Bonzai090 | leaves me with a bad taste in my mouth bad enough to warn all my clients not to bother with it |
| 10:28.39 | Bonzai090 | pr0m x-lite and snom |
| 10:28.42 | gaggaman | sound see features.conf |
| 10:28.44 | Bonzai090 | both wont transfer any calls |
| 10:28.51 | puzzled | Bonzai090: off course asterisk can transfer calls. the problem is prolly that *you* can't properly configure asterisk to do that |
| 10:29.15 | gaggaman | sound: ;pickupexten = *8 ; Configure the pickup extension. Default is *8 |
| 10:29.25 | soundguy | ta gaggaman :) |
| 10:29.47 | Bonzai090 | puzzled : did i rattle your cage?.... |
| 10:30.13 | pr0m | well. it took me more than 2 days to figure out asterisk... and i'm still learning. |
| 10:30.15 | Bonzai090 | i even tried AMP to see if i cant get a extension to transfer |
| 10:30.23 | Mother_ | Bonzai090: I agree with you that some transfer features have been really lacking for way too long, such as attended transfers (that was born with PBXs methinks), but you certainly *can* transfer calls |
| 10:30.23 | gaggaman | Bonzi go and play somewhere else. |
| 10:30.30 | *** join/#asterisk shadebob (n=shadebob@ll81-144-114-192-81.ll81.iam.net.ma) |
| 10:30.33 | shadebob | hi |
| 10:30.35 | puzzled | Bonzai090: off course not. the world is full of whiners that have no clue what they are talking about. I could care less |
| 10:31.02 | Mother_ | Reading Makes You Smarter (tm) |
| 10:31.03 | puzzled | hi shadebob |
| 10:31.16 | Bonzai090 | Mother_ well i spend over a week i got everything working accept transfers |
| 10:31.22 | shadebob | Anyone can help me with FOP and TDM400 zap channels? |
| 10:31.25 | puzzled | my transfer, attended & blind, work fine |
| 10:31.44 | puzzled | s/transfer/transfers |
| 10:31.57 | Mother_ | puzzled: what version are you using? |
| 10:32.13 | puzzled | cvs HEAD never older than 24 hours |
| 10:32.16 | gaggaman | If you haven't even been able to get call transfer working with asterisk, you better go and play somewhere else. You won't get far with asterisk. |
| 10:32.19 | pr0m | i transfered a call just a couple of days ago to my wife's cellphone from an analog phone using a digium card. |
| 10:32.26 | Mother_ | my attended transfers with Cisco phones are still edgy, this is a march HEAD however |
| 10:32.43 | Mother_ | it works though so I'm not touching it too much ;) |
| 10:32.52 | pr0m | heck. everytime someone calls my house they choose an extension and the call is "transferred". |
| 10:33.02 | Mother_ | all hail pr0m, all hail pr0m!!! |
| 10:33.07 | puzzled | indeed |
| 10:33.08 | Mother_ | j/k |
| 10:33.11 | hugo-v6 | puh. kiwano is a strange fruit |
| 10:33.16 | pr0m | Mother_: i didn't ask for that. |
| 10:33.36 | pr0m | it seems rather simple once the extensions are setup. |
| 10:33.42 | gaggaman | sound: pickupexten will work with asterisk >= 1.0 |
| 10:33.50 | pr0m | of course i'm not using any phancy phones or anything. |
| 10:34.03 | pr0m | gotta get ready for work. |
| 10:34.16 | *** join/#asterisk Mimmus (n=viggiani@ext.pitagora.it) |
| 10:34.29 | *** part/#asterisk Mother_ (n=Mother@53.Red-217-126-93.pooles.rima-tde.net) |
| 10:34.30 | Mimmus | good morning |
| 10:34.59 | Mimmus | any idea why I'm not able to hear music ina conference even if I have 'M' in MeetMe? |
| 10:35.23 | puzzled | got mpg123-0.59r installed? |
| 10:35.55 | Mimmus | I got mpg123 with AsteriskAtHome |
| 10:36.11 | Mimmus | it is Version 0.59r |
| 10:36.29 | Mimmus | MOH duirng calls works well |
| 10:36.45 | *** part/#asterisk jollyr_ (i=user@202.124.138.88) |
| 10:37.31 | Bonzai090 | well this chan as been as much help as a fart in a wind. first some idjit asks me to paste my extentions config and some other configs. took me like 20 min to get it off the server then he tells me he can fix it but he wants to be paid.. am i wrong to asume linux was intended to be free and this chan is here for support?... |
| 10:37.32 | gaggaman | Mother I have two 7970 with chan_sccp (chan-sccp.berlios.de) - never had probs with transfers. |
| 10:38.24 | *** join/#asterisk apardo (n=apardo@62-15-238-102.inversas.jazztel.es) |
| 10:40.23 | Delvar | Bonzai090: most people here are just normal users, and some have jobs supporting asterisk etc.. if you want free support then you have to find a nice person to do it for you... |
| 10:40.31 | *** part/#asterisk dwC` (n=dwc@S0106002078e03b3f.vs.shawcable.net) |
| 10:41.07 | Delvar | Bonzai090: or you can get a support contract from digium or asterisk bussness edition |
| 10:42.12 | *** join/#asterisk w0w0_ (n=apardo@90.Red-83-46-197.pooles.rima-tde.net) |
| 10:42.13 | Bonzai090 | Delvar i asked nicely about 6 hours ago if some one can just point out were i have gone wrong just a pointer i did not ask to have my whole dialpan rewritten and all i got was i had to pay some one to look at it |
| 10:42.46 | Delvar | Bonzai090: they the maling lists then. |
| 10:42.54 | Delvar | try* |
| 10:42.58 | Bonzai090 | def asterisk is becomming to look worse and worse |
| 10:43.06 | Bonzai090 | Delvar been there done that still waiting |
| 10:43.26 | Delvar | have you tried voip-info.org? |
| 10:44.41 | puzzled | Bonzai090: yes you are wrong. you don't have a *right* to support. you *may* get support. asterisk is complex with quite a learning curve as you have found out. if you want your hand held than it's going to cost you. otherwise do what the rest do: figure it out yourself and learn along the way |
| 10:44.48 | Bonzai090 | Delvar i tried about 6 sites so far all they show me is how to setup a extention |
| 10:45.39 | Bonzai090 | puzzled its a simple freaking issue not a problem with a dialplan or queue just a simple press a button and transfer jees man |
| 10:46.23 | Bonzai090 | setting up a extention is easy i can do that in my sleep but to get it to transfer is the issue |
| 10:47.14 | shadebob | How can transfert SIP call? |
| 10:47.21 | shadebob | between call? |
| 10:48.14 | Mimmus | shadebob: do you want to know what button you have to press to transfer call sto another ext? |
| 10:48.41 | Delvar | Bonzai090: your using a snom and x-lite? |
| 10:49.17 | Bonzai090 | Delvar yes there is 3 on x-lite and 3 snom phones |
| 10:49.22 | shadebob | Mimmus : button or built in key (#* etc..) |
| 10:49.27 | Bonzai090 | simple small pbx for the office |
| 10:49.54 | Delvar | Bonzai090: what is the process that you are using to transfer the calls and what problems do you get? |
| 10:49.57 | Mimmus | shadebob: press '#' or 'R' on som ephones |
| 10:50.19 | Delvar | R = Hold? |
| 10:50.35 | Delvar | nm |
| 10:50.40 | *** join/#asterisk surfdue (n=surfdue@user-0c6t1g9.cable.mindspring.com) |
| 10:50.41 | Mimmus | yes, on som eanalog phones too |
| 10:51.34 | Bonzai090 | Delvar i just try to use the transfer button on x-lite ohh and i got a elcheapo EQTEL phone and even that transfer buttons does not work |
| 10:51.38 | *** join/#asterisk skeffling (n=Andrew_H@andrew.1ec.aaisp.net.uk) |
| 10:52.00 | Bonzai090 | so i donno if its asterisk or if its the phones certainly it cant be the x-lite and snom and EQTEL |
| 10:52.10 | Mimmus | actually on my phone I have a 'FLASH' button! |
| 10:52.53 | Bonzai090 | mine says TRANSF on the eqtel right next to me |
| 10:53.17 | Delvar | Bonzai090: im trying to help... for xlite i think you have to 1. call one online, 2. click line 2 button, 3. dial extension, 4. press transfer, is that what you have bene doing? |
| 10:54.02 | Mimmus | Delvar: true? is this the procedure to do attended call transfer with X-Lite? |
| 10:54.23 | Delvar | Mimmus: i think so, im jsut setting it up now to try, its been a while since i used it |
| 10:54.54 | Mimmus | Delvar: I tried normally, pressing '#', but I get only an unattended (blind) transfer |
| 10:55.23 | Mimmus | Delvar: your method can be last hope! |
| 10:55.30 | Delvar | Mimmus: yes asterisk # transfer should work. |
| 10:55.45 | Delvar | but thats blind |
| 10:56.04 | *** join/#asterisk xpasha (n=pavel@217.30.252.68) |
| 10:59.12 | *** join/#asterisk atmos4 (n=sunset@213-182-118-30.teleos-web.de) |
| 10:59.19 | atmos4 | hi |
| 10:59.19 | dlu | i have a question about an invite |
| 11:00.32 | atmos4 | anyone has experience with AstTAPI and inbound calls, I can't seem to make inbound call announce working, I'm using TAPICall 1.2.33 for announce and have tried various settings for inbound chan, but no luck, my asterisk version is 1.07 |
| 11:00.37 | dlu | if i call dial(SIP/username) the INVITE hat username@1.2.3.4:1234 and in the to: field username@1.2.3.4:1234 that is totally ok i guess |
| 11:00.55 | Bonzai090 | Delvar yip even read there help page |
| 11:01.29 | Delvar | Bonzai090: have you got a firewall or anything liek that running? |
| 11:01.30 | atmos4 | I tried eg. Zap/g1/9914 which works for the User Channel, but no luck |
| 11:01.40 | atmos4 | the manager login is ok |
| 11:01.45 | dlu | but if i call dial(SIP/1234@username) the invite have 1234@1.2.3.4:1234 and the to: has 1234@1.2.3.4:1234 |
| 11:01.57 | Delvar | Bonzai090: if at teh asterisk CLI you have 'sip debug' on do you see the transfer request come in? |
| 11:02.41 | dlu | i think the invite should be username@1.2.3.4:1234 and the to header can be 1234@1.2.3.4:1234 or i am totally wrong???? |
| 11:02.44 | zurab | hello, everybody!!! Could someone explain me how disable VAD at all? I have Asterisk 1.2.0-beta1 and ATA186 (SIP 3.1.2: AudioMode 0x00140014) for alaw |
| 11:02.48 | Delvar | Bonzai090: in the sip entity try setting 'canreinvite=no' in the extensions entities and try again, i have a problem liek this before bt a long time ago. |
| 11:02.59 | Ogun | Problems with dialplans on my cisco 7905g SIP phone. I can not get it to allow me to dial *#21# |
| 11:03.08 | dlu | can anyone help me out? |
| 11:03.09 | puzzled | zurab: you need to disable it on the ATA186 |
| 11:03.11 | *** join/#asterisk Thumann (n=Thumann@217.157.30.66) |
| 11:03.14 | Thumann | hi all ;> |
| 11:03.22 | Bonzai090 | Delvar not tried to debug but even with sample config even with asterisk@home i cant transfer |
| 11:03.59 | Delvar | Bonzai090: look at the debug in CLI, see if it shows any errors, also 'set verbose 1000000000' |
| 11:04.08 | zurab | puzzled: I have AudioMode 0x00140014 there |
| 11:04.26 | Delvar | Bonzai090: transfer is one of the most simplest things so it should cause any problem... |
| 11:04.35 | puzzled | zurab: dunno, don't have a ATA186. check the manual, voip-info.org, google, etc |
| 11:05.14 | Thumann | well.. wierd support visit of the day.. 'we had to restart our server.. and we accidentially knocked it over.. something fell out.. please come' |
| 11:05.26 | Bonzai090 | Delvar i know thats why i dont understand surely transfer should be a easy thing |
| 11:05.53 | Bonzai090 | Delvar i'd say it should be enabled by default |
| 11:06.57 | Delvar | Bonzai090: its not a matter of it being disabled... its not. id say it was a problem with your asterisk server... do you see any error at the cli? |
| 11:07.05 | puzzled | Thumann: nice one |
| 11:07.08 | *** join/#asterisk Mother_ (n=Mother@53.Red-217-126-93.pooles.rima-tde.net) |
| 11:07.24 | Mother_ | did I miss much |
| 11:07.37 | puzzled | Bonzai090: have you checked that you are using the proper dtmfmode? |
| 11:07.45 | mover | can anyone help me with my invite problem? |
| 11:07.46 | Delvar | yeah asterisk became easy to use |
| 11:07.49 | puzzled | Mother_: nope |
| 11:08.08 | *** join/#asterisk TheCops (n=mdb@206-248-136-187.dsl.teksavvy.com) |
| 11:08.11 | Delvar | oh no wait... i was dreaming |
| 11:08.37 | Delvar | puzzled: bugger i totaly for got that |
| 11:08.48 | Thumann | puzzled: yeh.. side panel had fallen off.. a fan and 2 drive bays came loose.. |
| 11:08.51 | Mother_ | good :) |
| 11:08.59 | Thumann | puzzled: had to reseat the ram aswell.. :D works fine now.. |
| 11:08.59 | Delvar | Bonzai090: for xlite and snoms the default dtmfmode = rfc2833 |
| 11:09.53 | puzzled | hehe, sturdy little puter |
| 11:11.47 | Thumann | ibm ftw! :> |
| 11:12.45 | Thumann | i told it to a friend on msn.. he replied that it was nothing.. he had gotten this 'our internet connection is down.. there's fire in TDC's Central (isp), we can see the flames from here.. ' |
| 11:14.11 | h3x | %spam.gsm%Thank you for calling Special Price Analysis and Marketing. For cheap |
| 11:14.11 | h3x | <PROTECTED> |
| 11:14.11 | h3x | press 2. To enlarge your penis or breasts, please press 3. To speak with the M |
| 11:14.11 | h3x | inister of Finance of Nigera about a 50 million dollar account, press 4. |
| 11:14.15 | h3x | That rules |
| 11:14.29 | puzzled | yeah, that one is fun |
| 11:14.47 | Bonzai090 | Delvar i use rfc2833 by default on all my sip phones |
| 11:15.07 | Thumann | h3x: lol |
| 11:15.08 | puzzled | Bonzai090: make sure that in sip.conf dtmfmode is set to the same |
| 11:16.02 | *** join/#asterisk mithro (n=tim@itwan251-93.wan.it.kth.se) |
| 11:16.05 | Bonzai090 | puzzled i checked through all configs |
| 11:16.06 | sivana | frig.. now MS is getting into VoIP... that's Google, AOL, Yahoo, Skype, Vonage, and now MS |
| 11:16.12 | Delvar | Bonzai090: ok tell me the results of the CLI |
| 11:16.18 | Bonzai090 | so ocnfused i am reloading centos now and starting from scratch |
| 11:16.34 | Bonzai090 | Delvar i killed it and reloading now |
| 11:17.05 | Delvar | Bonzai090: ok once you have a basic ocnfig setup try it again, and get tehoutput from cli with sipdebug and verbos set high, hopefully it will spew out an error we can chase down |
| 11:17.16 | Bonzai090 | Delvar what can i use for call logging want to check who called who and how much time was spent? |
| 11:17.40 | Delvar | Bonzai090: asterisk automaticly stores CDRS in Masters.csv file |
| 11:18.30 | Delvar | Bonzai090: you can also get asterisk-addons and get teh cdr_mysql addon to log to mysql database |
| 11:18.32 | Bonzai090 | Delvar any app i can use to show people the results you know accountants they want to see were the money is going |
| 11:18.53 | Delvar | Bonzai090: not tha ti know of, everything we use here was built inhouse |
| 11:18.55 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
| 11:19.25 | Delvar | Bonzai090: have a look on voip-info for billing systems im sure theres some prebuilt ones |
| 11:21.26 | Bonzai090 | Delvar i'll look at the voip-info all i want something to show who called who and how long they were on the phone. |
| 11:22.44 | Bonzai090 | thats basically all i want to do is get transfer's going and detailed billing so i can get the boss off my damn case |
| 11:23.01 | puzzled | Bonzai090: there is a nice webinterface of CDR's at areski.net |
| 11:24.22 | Thumann | Bonzai090: and if you're willing to pay for it.. i know a dude who has made a really nice web interface for it aswell.. we use it here.. |
| 11:24.52 | Bonzai090 | Thumann thats the problem i believe linux is intended to be free so why pay?... |
| 11:25.08 | konrads | Hello, I am having troubles with mISDN. I have the module loaded in the correct order, but misdn show channels says (nil) |
| 11:25.24 | konrads | also if i try to fiddle with misdn port (*) segfaults |
| 11:25.54 | Thumann | Bonzai090: yes.. linux is free.. if your time has no value... |
| 11:26.05 | Mother_ | er....who said linux is meant to be free? |
| 11:26.11 | Mother_ | Thumann: agreed |
| 11:26.21 | konrads | THe question is: Is that normal? And how do mISDN channels differ from capi channels |
| 11:26.41 | konrads | and how do I specify that channel? |
| 11:26.42 | Bonzai090 | then linux is becomming anothe rmicrosoft we can just as well then port asterisk over to microsoft lol |
| 11:26.53 | Mother_ | I really hate all these people that just because it runs under linux then by their definition it should be free |
| 11:27.06 | Mother_ | well there *is* a win32 port... |
| 11:27.17 | *** join/#asterisk SwK_ (i=hulgyj@12-219-144-126.client.mchsi.com) |
| 11:27.40 | puzzled | Mother_: yeah, rather sicko project |
| 11:28.19 | Mother_ | there is nothing wrong in making a living making software than runs under linux, just as with any other OS on the planet - if people are then so damn generous as to give away their work then YOU should be thankful |
| 11:28.23 | *** join/#asterisk [ASK]ithrynn (n=nick@host217-34-132-179.in-addr.btopenworld.com) |
| 11:28.29 | Mother_ | puzzled: agreed :) |
| 11:28.35 | Thumann | Mother_: agreed |
| 11:28.39 | gordonjcp | Bonzai090: because using commercial stuff without paying for it is stealing, basically |
| 11:28.42 | gordonjcp | that's why you pay |
| 11:29.17 | Henguei | sorry for my bad english but i need help please |
| 11:29.20 | Bonzai090 | heh also true |
| 11:29.23 | Thumann | when i go to my boss and say.. dude.. we need this [insert random application] for our linux servers.. |
| 11:29.32 | *** join/#asterisk MikeJ[Laptop] (n=ircatjer@d14-69-8-30.try.wideopenwest.com) |
| 11:29.32 | gordonjcp | Bonzai090: free != gratis |
| 11:29.32 | Thumann | he goes.. : isn't that linux stuff free? |
| 11:29.43 | Mother_ | and I think it's really generous of digium to be putting all this work into *, what if you had to do the same thing with cisco? |
| 11:29.46 | Thumann | i just give him the: free if you time has no value one.. and he shuts up.. :D |
| 11:29.48 | Mother_ | lol |
| 11:29.50 | Henguei | i have always the same problem i want to install the french sound of asterisk |
| 11:30.36 | Thumann | I just made an asterisk box for a company in finland.. lol.. their IVR sounds.. weird.. |
| 11:30.40 | Henguei | i download it but i have not it clear |
| 11:31.03 | Thumann | one of the words has 18 letters... :D and it's all h,o,ö and i's.. |
| 11:31.15 | puzzled | Thumann: if you think finish sounds weird, wait till you meet them in person :) |
| 11:31.30 | Bonzai090 | lol puzzled they r weird ppl |
| 11:31.59 | Mother_ | hahaha |
| 11:32.12 | Bonzai090 | we have like 11 home languages here and one of our clients want about 5 languages loaded |
| 11:32.19 | Thumann | puzzled: i have.. i called them to test the box.. and then he had to give a coworker a longer message.. lol.. he could have flamed me from here to eternity.. i didn't understand a single word.. and i live in Denmark.. (right next to it.. or.. kinda) |
| 11:32.27 | Bonzai090 | press 1 for english 2 for afrikaans and so forth |
| 11:32.40 | Mother_ | Henguei: what is the problem? - you have to extract the sounds to the /sounds/xx dir where xx is the language |
| 11:32.51 | Mother_ | then set the language in the conf files or for each call etc |
| 11:33.42 | [ASK]ithrynn | hi dee ho |
| 11:33.59 | wiizard | Having a slight mare with Incoming calls thru a SIP Trunk |
| 11:35.01 | *** join/#asterisk XtremeGuest55438 (n=X@85.101.1.102) |
| 11:35.20 | wiizard | call comes thru from the gateway, and hits the pbx call comes in but i get a message saying "The service cannot be connected" |
| 11:35.46 | Thumann | how many digits do your pbx recieve? |
| 11:35.51 | wiizard | all |
| 11:36.01 | Thumann | and you have 'all' in your extensions.conf? |
| 11:36.13 | wiizard | well heres the thing |
| 11:36.18 | wiizard | this trunk is DID capable |
| 11:36.21 | Henguei | Mother_, i do it |
| 11:36.41 | Mother_ | Henguei: then what is the problem?? |
| 11:36.44 | *** join/#asterisk ruski (n=ruski@adsl-160-122.swiftdsl.com.au) |
| 11:37.55 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/sustaining/cpm) |
| 11:38.49 | Thumann | wiizard: yeh.. most larger trunks are.. |
| 11:38.51 | Thumann | ? |
| 11:39.03 | wiizard | number isnt in my extensions |
| 11:39.07 | wiizard | im using amp tho |
| 11:39.41 | wiizard | but what you saying that i need to add the number being called into my extensions? |
| 11:40.12 | Henguei | i can't listen to the sounds |
| 11:40.17 | Thumann | well.. if your pbx recieves 11223344 or smt. like that.. and you dont have it in your extensions.conf.. i've seen it many times that you get bounced back to the phone compay and get a recorded message like.. 'the number you have called has been tampered with by monkeys' |
| 11:40.28 | mover | i need help about an invite question |
| 11:40.39 | *** join/#asterisk zoo (i=nobody@ip-176-16.travedsl.de) |
| 11:40.40 | Henguei | and it's not clear |
| 11:40.55 | wiizard | ok that would make sense |
| 11:41.04 | wiizard | and the voice im hearing is english not yank |
| 11:41.06 | Thumann | wiizard: |
| 11:41.06 | Thumann | exten => 666,1,Answer |
| 11:41.07 | Thumann | exten => 666,2,MusicOnHold |
| 11:41.13 | Thumann | try adding that one.. change the number ofc... |
| 11:41.19 | Bonzai090 | Delvar how can i get the caller ID to work i know my lines provide me the caller ip but i also get unknown number |
| 11:41.20 | Thumann | you should get endless musiconhold |
| 11:41.24 | wiizard | what context? |
| 11:41.46 | puzzled | Bonzai090: add it in sip.conf |
| 11:41.49 | Thumann | wiizard: default |
| 11:41.55 | wiizard | rgr |
| 11:42.36 | Bonzai090 | puzzled add what to sip.conf |
| 11:42.50 | Mother_ | Henguei: define "not clear" |
| 11:43.06 | *** part/#asterisk zoo (i=nobody@ip-176-16.travedsl.de) |
| 11:43.06 | Mother_ | I'm sorry but you are being a bit vague |
| 11:43.57 | Henguei | i can't inderstand the voice |
| 11:44.35 | Thumann | Henguei: poor sound quality? or foreign language? :) |
| 11:44.52 | Henguei | poor sound quality |
| 11:45.03 | konrads | If I have CAPI channel configured (it says Contr1: 2 B channels total, 2 B channels free. |
| 11:45.10 | konrads | how do I test it? |
| 11:45.33 | konrads | (*) is plugged into alcatel PBX |
| 11:45.44 | konrads | supposedly it has 61 assigned to that port |
| 11:45.50 | Henguei | i said may be there is some problem with the package |
| 11:46.01 | konrads | however, when I dial 61, asterisk shows no info |
| 11:46.28 | ruski | hey .. is anyone a cisco expert ? i have a 2811, i also have some g703 E1 wics, im wishing to get a onramp circuit from the telco and route calls in over the E1 - cisco into my asterisk server .. is this possible with the g703 wic E1 ? |
| 11:46.37 | puzzled | Bonzai090: read through sip.conf and look at the examples |
| 11:46.49 | puzzled | Bonzai090: look for callerid |
| 11:47.24 | Bonzai090 | puzzled ok kewl. |
| 11:47.47 | puzzled | brb |
| 11:49.52 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 11:51.32 | Mother_ | Henguei: I don't know, have you tried opening a file with a sound editor and listening to it? |
| 11:52.32 | Henguei | but i can't do it .ogg or .gsm with xmms or with sox |
| 11:52.47 | Thumann | Mother_: could you be related to Daddy- ? |
| 11:53.09 | Mother_ | Thumann: to *your* daddy |
| 11:53.12 | Mother_ | :) |
| 11:53.15 | Thumann | hehe ;> |
| 11:53.27 | Thumann | Mother Underscore & Daddy Dash.. |
| 11:53.30 | Thumann | sounds cool |
| 11:53.32 | Thumann | hehe |
| 11:53.37 | Mother_ | Henguei: convert the gsm to pcm with sox and open it wherever |
| 11:53.42 | Mother_ | LOL |
| 11:54.35 | Henguei | ok i'll try it thks Mother_ |
| 11:54.50 | Henguei | :) |
| 11:55.03 | greekman | mother are you a digium tech? |
| 11:55.13 | Mother_ | greekman: nope |
| 11:55.22 | greekman | darn |
| 11:55.31 | greekman | any digium techs on right now? |
| 11:55.38 | konrads | greekman: you wish :) |
| 11:55.54 | Mother_ | if you pay me enough I will load the a-la-Matrix-style digium tech cartridge |
| 11:55.57 | greekman | I had one on until 4 this morning, poor bastard finally passed out on me |
| 11:57.02 | konrads | Mother_: and what are you rates? |
| 11:57.39 | Thumann | Mother_: does tapping in to The Matrix work with asterisk? and if so.. what codec is used? |
| 11:58.09 | Mother_ | I dunno, it's all them squiggles dropping downwards |
| 11:58.48 | Mother_ | what I still don't know is how didn't Cisco manage to get product placement in those movies |
| 11:58.54 | *** join/#asterisk r0m (n=SysOp@a81-84-68-45.cpe.netcabo.pt) |
| 11:58.56 | Thumann | well.. it's just.. it isn't mentioned in www.voip-info.org and.. i'd hate to enter the matrix with my pants on my head cuz the signal got icky... |
| 11:59.28 | Mother_ | they seem to put their 7960s in *everything* nowadays, I even saw a movie the other day where the screens where blank!!! |
| 11:59.33 | Mother_ | cheapo sods |
| 11:59.56 | Thumann | Mother_: lol.. same thing here.. a danish crime show thing.. blank screens.. also the 7960's... |
| 12:00.09 | puzzled | Mother_: yeah they don't even bother to hook them up |
| 12:00.12 | Bonzai090 | whats a good free software switchboard?.. |
| 12:00.19 | puzzled | FOP |
| 12:00.53 | r0m | good morning |
| 12:01.32 | puzzled | morning |
| 12:01.40 | Thumann | good afternoon |
| 12:02.07 | *** join/#asterisk lathos42 (n=lathos42@65-42-27-66.dowdingindustries.com) |
| 12:03.44 | *** join/#asterisk BhaalWK (i=bhaal@freenode/staff/bhaal) |
| 12:04.19 | puzzled | Thumann: what did you get? |
| 12:04.38 | Thumann | http://www.viewsonic.com/products/desktopdisplays/lcddisplays/xseries/vx924/ |
| 12:04.43 | Thumann | 4 ms refresh rate :> |
| 12:04.57 | puzzled | up & down? |
| 12:05.16 | Thumann | grey 2 grey |
| 12:05.45 | Thumann | but i've allready sold it and bought another.. |
| 12:06.09 | wiizard | Thumann |
| 12:06.13 | wiizard | that trick did it |
| 12:06.17 | wiizard | i now have MOH |
| 12:06.17 | Thumann | bought this: http://product.samsung.com/cgi-bin/nabc/product/b2c_product_detail.jsp?eUser=&prod_id=LS19BIDKBV%2fXAA |
| 12:06.25 | Thumann | wiizard: thought so :) |
| 12:06.28 | wiizard | hehe |
| 12:06.42 | wiizard | so do i have to do that for every number that wants to ring me lol |
| 12:06.47 | wiizard | I could be sittin here a long time |
| 12:06.53 | wiizard | with a very large context |
| 12:07.09 | Thumann | my extensions.conf is 26 pages long.. hehe.. tho' mine's a mess.. it can be simplified a lot.. |
| 12:07.16 | wiizard | lol 26pages |
| 12:07.20 | Thumann | :-X |
| 12:07.26 | wiizard | ive just seen why the call was being binned |
| 12:07.34 | atmos4 | hmm anyone got callerid detection via manager interface working with Zap-channels? |
| 12:07.34 | wiizard | exten => _.,1,AbsoluteTimeout(15) |
| 12:07.34 | wiizard | exten => _.,2,Congestion |
| 12:07.34 | wiizard | exten => _.,3,Hangup |
| 12:07.39 | atmos4 | I have no problems using it with sip |
| 12:07.53 | atmos4 | but can't seem to get it to work with Zap-channels |
| 12:07.55 | wiizard | i guess '_.' means any number |
| 12:07.56 | Delvar | lol |
| 12:08.00 | Thumann | puzzled: the viewsonic sux compared 2 the samsung one.. same refreshrate... |
| 12:08.04 | Delvar | any number or LETTER |
| 12:08.08 | Delvar | including h |
| 12:08.10 | Delvar | bad |
| 12:08.19 | Delvar | use _X. instead |
| 12:08.25 | wiizard | this is a default config |
| 12:08.49 | Delvar | bad |
| 12:08.54 | wiizard | ive just been chewing my arm trying to figure out why my calls were being rejected faster than a spotty kids by paris hilton |
| 12:09.08 | Bonzai090 | hmmm so if i got the caller id of some one who i dont wish to talk to i can give them an engaged tone as well?.. |
| 12:09.09 | wiizard | - a |
| 12:09.21 | Delvar | Bonzai090: yes |
| 12:09.24 | puzzled | Thumann: will check them out |
| 12:09.32 | Bonzai090 | Delvar hmmmm now this sounds fun |
| 12:09.34 | wiizard | whats to make it dial an extension |
| 12:09.40 | wiizard | number,1,EXT ? |
| 12:09.41 | Delvar | Bonzai090: you can give out any tone... or forward them to anywhere |
| 12:10.09 | Thumann | wiizard: i use a macro for that.. |
| 12:10.24 | wiizard | a macro? |
| 12:10.26 | Delvar | wiizard: exten => num,1,Dial(SIP/EXTENSION) |
| 12:10.29 | Bonzai090 | hmmm |
| 12:10.32 | wiizard | ta del |
| 12:10.45 | wiizard | Thumann what does ur macro do |
| 12:10.52 | Bonzai090 | Delvar so our yellow pages here is 1023 so i can direct them to 1023?.. |
| 12:10.53 | wiizard | allows extens to be dialed? |
| 12:11.03 | h3x | hahahahaaaahha |
| 12:11.16 | Delvar | wiizard: but using a macro will be better so you can do thing slike voaimail timeouts etc for all your extensions without having gobs of repeating codes |
| 12:11.17 | h3x | theres all these gas stations that are charging 2.99 per gallon coz theres no way to set a 3 for the 1st digit |
| 12:11.24 | Bonzai090 | or put them on hold with music in a endless loop hehee |
| 12:11.28 | sivana | tzanger: ping |
| 12:11.31 | wiizard | sounds like a hot idea |
| 12:11.32 | wiizard | however |
| 12:11.34 | Delvar | Bonzai090: yes... but the call will cost you |
| 12:11.38 | h3x | thats a nice little bug |
| 12:11.41 | wiizard | ive not got the know how to start making bombshells for myself |
| 12:11.54 | Bonzai090 | Delvar not as much as to listen to there whining hehe |
| 12:12.06 | Delvar | Bonzai090: did you get transfers working? |
| 12:12.25 | Bonzai090 | Delvar still installing from scratch doing updates now |
| 12:12.46 | weazul | hi all, i patched channel.c, channel.h and chan_sip.c from 1.02 (from voip-info.org) to asterisk CVS head. This concerns Radius. How can i share these files with others, so that they save patching time? |
| 12:12.54 | hugo-v6 | is there a documentation about macros for extensions conf? and is there a thing for sip.conf too? |
| 12:12.55 | wiizard | i thought with qualify= i could set the timeout |
| 12:12.59 | Thumann | h3x: lolz.. here in Denmark the added an extra digit to all the gaz stations.. 9.99 was the limit.. now 99.99 is! :S |
| 12:13.06 | Thumann | *they added |
| 12:13.17 | h3x | nice |
| 12:13.33 | h3x | its the new y2k bug |
| 12:13.37 | h3x | we didnt even see it coming |
| 12:13.45 | Thumann | h3x: so all the shitty looking gaz stations around the country now have shiny new signs.. :D |
| 12:14.37 | hugo-v6 | anyone? |
| 12:15.08 | Thumann | hugo-v6: www.voip-info.org |
| 12:15.53 | hugo-v6 | Thumann: well... nvm |
| 12:16.33 | Thumann | heh? |
| 12:16.54 | *** join/#asterisk Ganlron (n=Ganlron@omega.csolve.net) |
| 12:18.44 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 12:21.41 | *** join/#asterisk docelm0 (n=docelm0@66.237.242.41.ptr.us.xo.net) |
| 12:23.05 | Bonzai090 | Delvar were can i get a simple tutorial to just get the basics going on asterisk just incomming calls to a extention and adding extentions?. |
| 12:23.08 | lathos42 | Anyone got a favorite Polycom Reseller that I could contact for 70 phones? |
| 12:24.42 | *** join/#asterisk mithro_ (n=tim@itwan251-93.wan.it.kth.se) |
| 12:25.46 | Delvar | Bonzai090: your using asterisk @ home? - http://chayden.net/Asterisk/SeUpAsteriskAtHome.htm |
| 12:27.25 | tzanger | sivana: here |
| 12:27.38 | Bonzai090 | Delvar nope asterisk 1.0.9 on centos |
| 12:28.23 | sivana | tzanger: got your email |
| 12:29.52 | tzanger | cool |
| 12:30.18 | Delvar | Bonzai090: http://www.voip-info.org/tiki-index.php?page=Asterisk |
| 12:30.41 | Delvar | Bonzai090: http://www.voip-info.org/tiki-index.php?page=Asterisk+installation+tips |
| 12:31.04 | *** join/#asterisk DrWho (n=MIKE@mike-new.tc3net.com) |
| 12:31.21 | Delvar | Bonzai090: http://www.asteriskguru.com/tutorials/asterisk_installation.html |
| 12:31.40 | Delvar | Bonzai090: just look around on voip-info... lots of good stuff |
| 12:33.27 | *** join/#asterisk azrishahril (n=azrishah@60.50.198.59) |
| 12:34.08 | azrishahril | Asterisk can only provide pass-thru functionality for g723.1 ? anybody care to explain what does pass-thru mean ? |
| 12:37.36 | Bonzai090 | Delvar lemme look i just need to get the basics up and running for this evening |
| 12:38.15 | konrads | how do I tell if a call reaches my ISDN line |
| 12:38.15 | konrads | ? |
| 12:38.18 | Thumann | azrishahril: http://www.voip-info.org/tiki-index.php?page=Asterisk+G.723+pass-thru |
| 12:38.40 | *** join/#asterisk gr0mit (n=w10277@fw.mot-tools.co.uk) |
| 12:39.46 | *** join/#asterisk azrishahril (n=azrishah@60.50.198.59) |
| 12:41.31 | *** mode/#asterisk [+o drumkilla] by ChanServ |
| 12:43.50 | hugo-v6 | is there a nice frontend for asterisk+mysql? |
| 12:46.05 | tzanger | I can't believe it's september already |
| 12:46.13 | *** join/#asterisk MikeJ[Laptop] (n=ircatjer@mi.origenfinancial.com) |
| 12:47.23 | *** join/#asterisk Druken (i=Druken@67.69.139.226) |
| 12:47.53 | hugo-v6 | .o(whake me up when september ends) |
| 12:47.54 | gaggaman | is there any docu about the sccp / skinny protocoll on the net? I'd like to play with the cisco 7970 video functions. |
| 12:47.58 | Thumann | tzanger: crap.. then i'm late for something.. |
| 12:48.17 | hugo-v6 | s/wh/w/ |
| 12:48.33 | *** join/#asterisk jiro5281 (n=jiro5281@203.131.137.76) |
| 12:50.07 | atmos4 | can I somehow create a dummy extension, just for callerid pickup? |
| 12:50.17 | *** join/#asterisk Bobby_Ewing (n=stationa@82.152.102.44) |
| 12:50.31 | Druken | at, meaning ? |
| 12:50.31 | atmos4 | the problem with chan_capi is, that it doesn't immediately set callerid when creating the channel |
| 12:50.46 | atmos4 | instead it does that later using a Newcallerid event |
| 12:51.02 | atmos4 | so eg. asttapi or other cid anouncers don't work |
| 12:51.31 | atmos4 | but if I monitor eg. a sip channel, I have no problem getting the callerid |
| 12:51.53 | Bobby_Ewing | Hi all - is there a guide for queues online anywhere? I haven't touched my * config for ages and I want to adjust a queue time, i.e. so someone only stays in the queue for say 10 minutes before being directed elsewhere... ? |
| 12:52.07 | jack | how do you push information to a cisco 7960 running sip ? |
| 12:52.24 | Druken | Bobby_Ewing: the wiki if your friend |
| 12:52.36 | Druken | is even |
| 12:52.36 | gaggaman | jack: that's an easy one: |
| 12:52.41 | gaggaman | not at all. |
| 12:52.46 | jack | gaggaman: damn :( |
| 12:52.55 | atmos4 | e.g. when I check using yaacid, I see the incoming call on capi with callerid=unknown, then few moment later I see a manager event Newcallerid, which sets the correct callerid |
| 12:52.56 | gaggaman | xml push is only possible with SCCP |
| 12:53.15 | Thumann | jack: use tftpd |
| 12:53.24 | gaggaman | you could use the sccp images with chan-sccp |
| 12:53.48 | gaggaman | Thu: tftp? for what? |
| 12:54.00 | jack | Thumann: and reboot the phone every time? :) |
| 12:54.08 | Thumann | jack: sure :) |
| 12:54.08 | gaggaman | LOL |
| 12:54.28 | Thumann | we have a web page where you can make the changes.. and click reboot.. and the phone reboots.. |
| 12:55.43 | hugo-v6 | hmmm my gf got a phone-call and went away so i cant hear it. maybe she has a new lover? *think* |
| 12:55.58 | Thumann | jack: http://wow.thumann.dk/images/phone.JPG |
| 12:56.08 | Druken | hugo-v6: or maybe she |
| 12:56.19 | Druken | is ordering you a stripper and doesn't want you to know |
| 12:56.28 | gaggaman | hugo: tiki for "call recording" :-) |
| 12:56.30 | hugo-v6 | Druken: possible ;)) |
| 12:56.42 | hugo-v6 | gaggaman: thats a damn good idea : |
| 12:56.44 | hugo-v6 | :> |
| 12:57.34 | Druken | it's am option for all incoming dids in my dialplan... |
| 12:57.46 | Druken | god damn i can't type today |
| 12:58.06 | *** join/#asterisk ymorin (n=ymorin@savoirfairelinux.net) |
| 12:58.33 | gaggaman | Thu nice but that has nothing to do with xml push. it is used to display dynamic info, like stock prices or whatever |
| 12:59.03 | Druken | hehe a ticker on a phone? |
| 12:59.08 | Druken | that'd be kinda kewl |
| 12:59.19 | jack | Thumann: I want to display on all phones if the 'out to lunch' voicemail is activated on the 'general' phone number etc ;) |
| 13:01.05 | Thumann | jack & gaggaman: ahh! :-# |
| 13:01.14 | *** join/#asterisk lehel (n=asd@82.79.20.17) |
| 13:01.31 | lehel | hello |
| 13:01.49 | *** join/#asterisk Akelavlk (n=jansun@82.119.239.141) |
| 13:02.05 | *** join/#asterisk xpasha (n=pavel@217.30.252.68) |
| 13:02.13 | *** join/#asterisk rob314 (n=rob314@207.58.194.2) |
| 13:02.26 | *** join/#asterisk greekman (n=alex@host254.209.113.199.conversent.net) |
| 13:03.27 | Akelavlk | Hello, how can I play music while dialing? I tried PlayOnHold and also BackGround command. But system start dialing after music ends. |
| 13:04.07 | wiizard | New question for you dudes |
| 13:04.07 | hugo-v6 | no frontend for mysql with asterisk? found only this phpmyedit. |
| 13:04.10 | wiizard | If i want to direct my incoming number to a macro |
| 13:04.22 | *** join/#asterisk scanna (n=scannach@81-174-16-211.f5.ngi.it) |
| 13:05.01 | *** join/#asterisk zotz (n=zotz@24.231.36.100) |
| 13:05.07 | xpasha | hello guys |
| 13:05.21 | Thumann | wiizard: |
| 13:05.21 | Thumann | exten => 6001,1,Macro(direct) |
| 13:05.21 | Thumann | exten => 6002,1,Macro(direct) |
| 13:05.23 | Thumann | etc.. |
| 13:05.32 | wiizard | rgr |
| 13:05.36 | xpasha | who could tell me how it is possible to detect # symbol as a end of dialing number? |
| 13:05.55 | xpasha | if the user dials international number and press # after that |
| 13:05.59 | Druken | Akelavlk: RTFM :) it's an option in the dial string |
| 13:06.17 | *** join/#asterisk coppice (n=chatzill@167.203.17.210.dyn.pacific.net.hk) |
| 13:06.53 | Akelavlk | Druken, Aha.. Thanks.. Empty heand today.. |
| 13:10.07 | *** join/#asterisk BlueSky_ (n=BlueSky@prg1-v-4-213.static.adsl.vol.cz) |
| 13:10.26 | *** join/#asterisk _fenlander (n=neils@82.152.81.57) |
| 13:11.10 | Bobby_Ewing | Drunken: i swear i had searched the wiki but couldn't find what i was after.. yet I found it first time now.. and then.. to top it off, i had an old context in my config file with the command being used too..! |
| 13:12.29 | mutilator | whats a good place to buy rack servers? |
| 13:12.42 | mutilator | bare bone kits or even just parts |
| 13:12.51 | *** join/#asterisk RoyK (n=roy@213.160.242.93) |
| 13:13.20 | wiizard | Thumann dude i dont get where this macro grabs the extension ID from? |
| 13:13.48 | wiizard | wait hold on ignore that |
| 13:13.57 | wiizard | im splicing different ideas together here |
| 13:14.28 | Thumann | 'Operation *' Magicmarker ftw! |
| 13:14.45 | wiizard | lol @ operation magicmarker |
| 13:16.44 | *** part/#asterisk BlueSky (n=BlueSky@prg1-v-4-213.static.adsl.vol.cz) |
| 13:18.56 | Katty | morning |
| 13:19.36 | sivana | morning |
| 13:20.07 | Thumann | afternoon |
| 13:20.10 | *** join/#asterisk Henguei (n=smoka@196.203.53.151) |
| 13:21.15 | BrianR___ | Has anyone made a knockoff of the stupid java app that compiles grandstream GXP-2000 config files? |
| 13:21.53 | *** join/#asterisk Craziman2 (n=Craziman@boromir.apid.com) |
| 13:22.12 | *** join/#asterisk Hmmhesays (n=Hmmm@66.173.103.107) |
| 13:22.15 | *** part/#asterisk Akelavlk (n=jansun@82.119.239.141) |
| 13:22.23 | Hmmhesays | its morning here |
| 13:23.36 | Hmmhesays | wtf is isdn30 |
| 13:23.50 | Thumann | Hmmhesays: isdn with 30 lines.. hehe |
| 13:24.45 | Thumann | Hmmhesays: same as an E1 line |
| 13:24.57 | Hmmhesays | british telcom said "its an isdn 30 line" |
| 13:25.00 | Hmmhesays | heh |
| 13:25.04 | Thumann | hehe |
| 13:25.26 | coppice | Hmmhesays: its something bad |
| 13:25.41 | Thumann | works fine here! :> |
| 13:25.47 | *** join/#asterisk bhima^ (n=gf2e@i13pc168.ilkd.uni-karlsruhe.de) |
| 13:26.09 | Hmmhesays | so what exactly did they mean? I asked them for isdn protocol type |
| 13:26.11 | coppice | its something totally non-standard, that nobody outside the UK uses |
| 13:26.16 | *** join/#asterisk gambolputty (n=gambolpu@cblmdm72-240-241-108.buckeyecom.net) |
| 13:26.25 | Thumann | i'm in denmark.. we use it.. |
| 13:26.28 | Delvar | bugger xlite doesnt seem to allow transfering calls, |
| 13:26.32 | wiizard | ISDN30 |
| 13:26.37 | Delvar | the transfer button does nothing |
| 13:26.45 | wiizard | e |
| 13:26.49 | coppice | Thumann: denmark uses euroisdn |
| 13:27.04 | bhima^ | asterisk has recently started instantly segfaulting when it tried to dial out via my Fritz card. Incoming works just fine. Haven't changed anything that I'm aware of... |
| 13:27.18 | Thumann | sure.. but the phonecompanys call them what they want.. we have a 'isdn30' they say.. :) |
| 13:27.36 | coppice | well, BT's ISDN30 is not EuroISDN |
| 13:27.46 | *** join/#asterisk insomni (n=insomni@x1-6-00-e0-18-6f-34-ad.k455.webspeed.dk) |
| 13:27.53 | bhima^ | capi debug doesn't show anything. ANy suggestions about where to start debugging? |
| 13:28.19 | Thumann | tomato tomato |
| 13:29.23 | Thumann | coppice: well.. then it's a isdn configuration with 30 b chans.. and one d chan.. |
| 13:29.39 | Hmmhesays | thats great I don't think this e1 gateway is going to work |
| 13:30.15 | Thumann | Hmmhesays: why not? |
| 13:30.47 | Thumann | Hmmhesays: |
| 13:30.47 | Thumann | span=1,1,0,ccs,hdb3,crc4 |
| 13:30.48 | Thumann | bchan=1-15,17-31 |
| 13:30.48 | Thumann | dchan=16 |
| 13:30.55 | Thumann | for our isdn30 ;P |
| 13:31.11 | coppice | I don't think * supports BT ISDN30 |
| 13:31.16 | Hmmhesays | what signaling type? |
| 13:31.18 | *** join/#asterisk Akelavlk (n=jansun@82.119.239.141) |
| 13:32.06 | *** join/#asterisk ericm_06 (n=ericm_06@c-24-9-147-88.hsd1.co.comcast.net) |
| 13:32.08 | Akelavlk | Has anybody idea how to use ZapBarge on AIX phones? |
| 13:33.00 | Akelavlk | Or how to use ZapBarge. I made a call between two AIX phones and when It's not possible call ZapBarge |
| 13:33.31 | Thumann | Hmmhesays: well.. framing type is cas or ccs |
| 13:33.35 | *** join/#asterisk Dave^ (n=Miranda@dlg89.neoplus.adsl.tpnet.pl) |
| 13:33.39 | *** part/#asterisk cpm (n=Chip@pdpc/supporter/sustaining/cpm) |
| 13:34.02 | Thumann | Hmmhesays: and coding is ami or hdb3 |
| 13:34.05 | Hmmhesays | well in this gateway you have to set the isdn signaling type, etsi, 5ess etc... |
| 13:34.25 | Dave^ | Hi, can anyone here help me with realtime and sip ? |
| 13:35.17 | Dave^ | <PROTECTED> |
| 13:35.26 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:36.19 | Katty | Hmmhesays: mew |
| 13:36.29 | Katty | Hmmhesays: i have issue. |
| 13:36.37 | Ahrimanes | s |
| 13:36.42 | Katty | Hmmhesays: lack of mp3 player with vorbis ogg encoding support stuffs. |
| 13:36.46 | *** join/#asterisk Ariel_ (n=Ariel@adsl-068-157-125-248.sip.mia.bellsouth.net) |
| 13:36.48 | Katty | Hmmhesays: and winamp is too bloated. |
| 13:36.53 | Katty | Hmmhesays: fix pls |
| 13:37.21 | Thumann | Hmmhesays: well.. afaik the signaling consists of framing and coding.. |
| 13:37.29 | Thumann | Katty: winamp is bloated? |
| 13:37.42 | tzanger | Thumann: uh, yes |
| 13:37.53 | Katty | Thumann: very |
| 13:38.06 | Hmmhesays | easy fix, don't use vorbis |
| 13:38.12 | Katty | Hmmhesays: :-o |
| 13:38.16 | Akelavlk | Has anybody idea how to use ZapBarge with AIX phones? |
| 13:38.18 | Katty | Hmmhesays: blasphamey |
| 13:38.19 | Hmmhesays | just re-encode your stuff into mp3 |
| 13:38.31 | Katty | Hmmhesays: but i /like/ oggs |
| 13:38.32 | Hmmhesays | unless you are talking about a software mp3 player |
| 13:38.32 | tzanger | Akelavlk: um. IAX != Zap, how do you expect to use it? |
| 13:38.35 | *** part/#asterisk bhima^ (n=gf2e@i13pc168.ilkd.uni-karlsruhe.de) |
| 13:38.35 | Thumann | Katty: in what way is winamp bloated? ;> |
| 13:38.37 | Katty | Hmmhesays: not to mention have thousands of them |
| 13:38.45 | Katty | Thumann: ... |
| 13:38.53 | Katty | Thumann: have you not ran winamp? ever? |
| 13:38.57 | Dave^ | sip, realtime, caching anyone ? |
| 13:39.06 | Hmmhesays | I like winamp, with the dfx plugins it just rocks |
| 13:39.10 | Katty | Hmmhesays: yes, exactly. |
| 13:39.11 | Thumann | Katty: myes.. ? |
| 13:39.18 | Hmmhesays | use the ogg plugin |
| 13:39.29 | Katty | Hmmhesays: but winamp is resource hoggy |
| 13:39.34 | Hmmhesays | what? |
| 13:39.38 | Katty | Hmmhesays: and has all this fluffy stuff with it now |
| 13:39.50 | Hmmhesays | oh, I use winamp classic |
| 13:40.00 | *** join/#asterisk zoo (i=nobody@ip-176-16.travedsl.de) |
| 13:40.03 | Akelavlk | tzanger. Two AIX users will calling and one of them call ZapBarge and call to next user. Then next user can listen call.. |
| 13:40.08 | Katty | Hmmhesays: did i mention windows? |
| 13:40.08 | Thumann | Katty: classic is cool.. |
| 13:40.18 | Hmmhesays | i use winamp classic in windows |
| 13:40.20 | tzanger | Akelavlk: again, it doesn't sound like htere's a Zap channel involved. |
| 13:40.21 | Akelavlk | tzanger, you know two IAX and one Zap user. |
| 13:40.24 | Katty | i shall investigate |
| 13:40.34 | Hmmhesays | i use xmms in linux |
| 13:40.36 | tzanger | IAX is a channel technolgoy. Zap is a channel technology. You can't use ZapBarge on an IAX channel. |
| 13:41.02 | *** part/#asterisk ericm_06 (n=ericm_06@c-24-9-147-88.hsd1.co.comcast.net) |
| 13:41.03 | Katty | Hmmhesays: so do i |
| 13:41.11 | Akelavlk | tzanger is there any function which allows just listen call to another users? |
| 13:41.13 | *** join/#asterisk zoo (i=nobody@ip-176-16.travedsl.de) |
| 13:41.24 | Hmmhesays | Katty: winamp 5 will play ogg by default and you can set the skin to classic skin |
| 13:41.25 | tzanger | Akelavlk: Monitor() ? |
| 13:41.39 | Akelavlk | Monitor just store call into file.. |
| 13:41.43 | tzanger | Hmmhesays: Winamp is leaner than the alternatives but that doesn't make it unbloated |
| 13:42.14 | *** join/#asterisk Connor_ (n=billy@198-144-174-5.knx.tn.nxs.net) |
| 13:42.16 | Hmmhesays | <shrug> these aren't the days of having 16 meg of system memory anymore |
| 13:42.22 | Hmmhesays | I don't care if winamp takes up 5 meg |
| 13:42.46 | Hmmhesays | point me at a better mp3 player with better plugins in windows |
| 13:43.27 | Thumann | Hmmhesays: agreed.. i've got 1Gb ram in my windows box.. who cares about 5mb is used for winamp.. |
| 13:43.33 | Thumann | who cares if it uses 20... |
| 13:43.41 | Hmmhesays | i use winamp5 at work, you don't notice the difference between that and classic if you have the classic skin off |
| 13:44.31 | Hmmhesays | er... classic skin ON even |
| 13:44.37 | Thumann | hmm.. what does a Hummer cost in US? the car... |
| 13:44.46 | Hmmhesays | not enough |
| 13:44.57 | spackle | 60 K USD |
| 13:44.58 | tzanger | Thumann: winamp still takes several seconds to load on my 3GHz P4 |
| 13:45.00 | tzanger | that's bloat |
| 13:45.00 | spackle | ballpark |
| 13:45.01 | Thumann | omg... |
| 13:45.09 | Katty | omgwtfbbq? |
| 13:45.09 | *** part/#asterisk zoo (i=nobody@ip-176-16.travedsl.de) |
| 13:45.12 | Katty | lolzkthxbi |
| 13:45.18 | Hmmhesays | several seconds? your system is b0rken |
| 13:45.23 | *** join/#asterisk basty (n=basty@212.218.65.251) |
| 13:45.23 | spackle | outrageous isn't it? |
| 13:45.24 | basty | Hi |
| 13:45.25 | Thumann | spackle: 130$ in denmark.. |
| 13:45.49 | Hmmhesays | people should be shot in the leg if they try and buy hummers |
| 13:45.51 | Thumann | tzanger: mine loads ~instantly on my p4 3ghz... with a playlist of ~2k... |
| 13:46.06 | tzanger | Thumann: I bet you have it preloaded |
| 13:46.14 | Thumann | tzanger: negative |
| 13:46.16 | tzanger | it's the same way that MS Office loads "instantly" |
| 13:46.19 | spackle | Thuman: people who live next door to me have more money invested in what's sitting on the driveway than in their house. |
| 13:46.46 | basty | I have setup Asterisk for internal and external phone-calls. The internal works well. The external calls hang up after approx 20 secs. I am using a VoIP Gateway for external calls. For right now I test it with X-Lite (softphone). Anyone has an idea ? |
| 13:46.51 | Hmmhesays | here is the logic. high gas prices 'lets buy a gas guzzling vehicle','lets bring consumption to an all time high','lets give the oil companies reasons not to keep the price high' |
| 13:47.20 | Katty | i was all :<<< this morning when gas was 2.79 a gal. |
| 13:47.24 | Hmmhesays | and I can't type this morning |
| 13:47.35 | Hmmhesays | $2.99 here, it costs 10 bucks to fill my motorcycle |
| 13:47.35 | jpayne | basty: do you have a firewall between you and the voip gateway? |
| 13:47.40 | Thumann | here they added an extra digit so it could pass the 9.99 max.. :S |
| 13:47.42 | basty | jpayne: yep. |
| 13:47.45 | Thumann | 10.75 dkr now.. |
| 13:47.47 | Thumann | pr. litre.. |
| 13:47.53 | `Sauron | 08:43 <Moonwick> omg, gas is eleventy billion dollars for a thimbleful |
| 13:47.53 | spackle | anybody from Hawaii? It's always 5ish (USD) there. |
| 13:47.54 | *** join/#asterisk RoyKa (n=roy@213.160.242.93) |
| 13:47.56 | Hmmhesays | god i love this skid row video |
| 13:48.05 | Katty | `Sauron: :> |
| 13:48.08 | basty | jpayne: a cisco pix 520..... |
| 13:48.08 | jpayne | look to see if the firewall has "fixup" for SIP or whatever you're using |
| 13:48.11 | jpayne | ah |
| 13:48.19 | Thumann | `Sauron: lol |
| 13:48.23 | newl | Hmmhesays: which? |
| 13:48.28 | jpayne | fixup protocol sip |
| 13:48.34 | jpayne | either turn it on, or turn it off |
| 13:48.34 | Hmmhesays | newl: 'little wing' |
| 13:48.43 | jpayne | depending on what it curently is :) |
| 13:48.45 | spackle | 'Sauron: how did you ever make out with the gumstix computers? |
| 13:48.57 | tzanger | Hmmhesays: that's a wicked song |
| 13:48.58 | coppice | you'll soon be buying gas in litre bottles |
| 13:49.01 | basty | jpayne: I thought about a problem with the extension or sip.conf. Is there any variable to set the timeout ? |
| 13:49.14 | tzanger | although I like the pink floyd version |
| 13:49.16 | `Sauron | Well, the score for now is Me: 0.5 - Gumstix: eleventy billion |
| 13:49.18 | `Sauron | Nah |
| 13:49.25 | tzanger | `Sauron: why |
| 13:49.28 | jpayne | basty: look at the firewall first |
| 13:49.32 | Hmmhesays | tzanger: yeah no matter who is playing it |
| 13:49.35 | basty | okay thanks, jpayne. |
| 13:49.40 | `Sauron | I had some struggles with them, getting some ofthe stuff I wanted to use, to work |
| 13:49.44 | tzanger | no I really am not keen on Hendrix playing it himself |
| 13:49.53 | Hmmhesays | we have e85 in north dakota/minnesota its about 60 cents less a gallon |
| 13:49.57 | jpayne | np |
| 13:50.00 | `Sauron | Took me a week to debug the soldering job for the serially-connected GPS receiver I have |
| 13:50.06 | Hmmhesays | next car I buy will be an e85 vehicle |
| 13:50.24 | `Sauron | Until it turned out that I had to set some really screwy settings in stty for the serial port |
| 13:50.41 | *** join/#asterisk diego_br (n=liebert@200.208.241.178) |
| 13:50.41 | `Sauron | Now me and gpsd are having an argument^Wdisagreement |
| 13:50.54 | *** join/#asterisk toddf (n=toddf@net-66-210-104-107.theshop.net) |
| 13:51.17 | spackle | 'Sauron: still after it then? |
| 13:51.21 | `Sauron | yup |
| 13:51.21 | *** join/#asterisk kupy (n=kupy@81-203-10-19.user.ono.com) |
| 13:51.25 | `Sauron | don't give up that easily |
| 13:51.29 | kupy | hi all |
| 13:51.32 | drray | :) |
| 13:51.33 | *** join/#asterisk zoo (i=nobody@ip-176-16.travedsl.de) |
| 13:51.52 | `Sauron | I think if I go pick up the higher-flash-capacity version, I'll play around with making one of them a sip/iax2-ish wifi phone thingy |
| 13:51.55 | *** part/#asterisk zoo (i=nobody@ip-176-16.travedsl.de) |
| 13:51.57 | `Sauron | hook up a serial keypad |
| 13:52.00 | *** join/#asterisk ^_leo_^ (n=leo@213.171.250.130) |
| 13:52.05 | `Sauron | and use the audiostix for the sound portion |
| 13:52.11 | ^_leo_^ | hi |
| 13:52.29 | `Sauron | well, I'd use the 16MB flash one for my current project |
| 13:52.38 | `Sauron | and the current one for the iaxcomm thing |
| 13:52.56 | spackle | 'Sauron, that was what you were considering when you first netioned you had one. or along those lines |
| 13:53.01 | ^_leo_^ | has anyone experienced problems receiving sound throug ISDN4Linux in new 1.2.0_b1 release? |
| 13:53.03 | `Sauron | ya |
| 13:53.12 | `Sauron | That's on hold while I play with my original idea |
| 13:53.16 | `Sauron | which is all hush hush |
| 13:53.21 | `Sauron | ;) |
| 13:55.24 | spackle | 'Sauron: so what is your opinion of gumstix then? Are they useful or just not there yet? |
| 13:55.46 | ^_leo_^ | I was running 1.0.1 for some time, but now I have compiled 1.2.0b1, and made the necessary changes in conf files and all is working except the reception of the sound throug i4l (i.e. the caller could hear my voice but I hear such a noisy background that I can't undestand anything from the caller) |
| 13:55.54 | Katty | Hmmhesays: winamp lite doesn't have the built in radio?! :<<<<< |
| 13:56.39 | Hmmhesays | just use winamp5 regular |
| 13:56.51 | Thumann | lol.. nice music on hold @ telia.. and for 200$/hour.. i guess i can listen to some music |
| 13:56.55 | ^_leo_^ | I there any substancia change made to the i4l manager that would be generating this sound? |
| 13:57.05 | ^_leo_^ | I -> is |
| 13:57.08 | ^_leo_^ | lol |
| 13:57.09 | Dave^ | can anyone here explain caching and sip realtime ? |
| 13:57.14 | Katty | Hmmhesays: k |
| 13:57.30 | brad_mssw | anyone in here have the UTStarcom F1000 WIFI phone ? |
| 13:57.32 | *** join/#asterisk MikaEL` (n=mikael@193.230.183.230) |
| 13:57.39 | Hmmhesays | with the classic skin you'll never notice its not winamp classic |
| 13:57.49 | yaboo | brad_mssw, waiting for my samples |
| 13:57.49 | ^_leo_^ | Is there any substancial change made to the i4l manager that would be generating this sound? |
| 13:58.13 | brad_mssw | yaboo: hmm, i have an issue with attended call transfer with mine ... other than that, it works great ... |
| 13:58.31 | brad_mssw | yaboo: just the attended transfer not working is a real PITA ... blind transfer works like a champ though |
| 13:58.44 | *** join/#asterisk skeffling (n=Andrew_H@andrew.1ec.aaisp.net.uk) |
| 13:58.54 | Katty | Hmmhesays: hmm :< |
| 13:59.05 | yaboo | brad_mssw, my company is australia is becoming the distributor for them, and awaiting my free ten samples to play with |
| 13:59.12 | `Sauron | spackle: they're very useful. Really easy to develop for, since it's just a linux box |
| 13:59.26 | brad_mssw | yaboo: well, great, maybe you'd have some pull to get them to fix the problem! |
| 13:59.31 | `Sauron | The space on the 4MB flash ones is a bit cramped, but that's what the 16MB flash one is for |
| 13:59.43 | brad_mssw | yaboo: I can't find any firmware updates or anything for them |
| 13:59.53 | yaboo | brad_mssw, what problem |
| 13:59.59 | *** join/#asterisk Henguei (n=smoka@196.203.249.98) |
| 14:00.52 | brad_mssw | yaboo: basically when you do an attended transfer, it contacts the third party just fine, but when you hit transfer again to complete the transfer, it tries to establish another connection to the third party, to transfer the call to, instead of utilizing the current connection ... so you end up transferring someone to the third-parties voicemail |
| 14:01.21 | brad_mssw | other sip phones we have work fine though :/ |
| 14:01.34 | *** join/#asterisk drewskie (n=drew@yankee.proximitygroup.com) |
| 14:02.02 | yaboo | brad_mssw, since I haven't recieved them yet, all I can do is copy your problem and email them |
| 14:02.22 | yaboo | and I cannot replicate the problem to state more to them |
| 14:02.27 | drewskie | hey guys, im attempting to migrate my current Toshiba digital phones to a system that would work with Asterisk, is it possible to just purchase iaxy ata's for each of the toshiba digital phones, or would i have to purchase new phones entirely? |
| 14:02.35 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 14:02.40 | yaboo | brad_mssw, we do more business selling there cdma and gsm phones |
| 14:02.46 | Hmmhesays | you can't use digital phones with ata's |
| 14:02.49 | yaboo | so the f1000 is something new to us |
| 14:02.54 | drewskie | i was afraid of that =/ |
| 14:03.07 | brad_mssw | yaboo: that's cool ... just remember the issue when you go to test them ... |
| 14:03.20 | drewskie | i had read that the toshiba's were compat with asterisk, i was hoping the ata's would solve my problems......so it looks like i have to get all new phones |
| 14:03.24 | brad_mssw | yaboo: I've got sip debug logs for a good sip transfer, and their bad sip transfer if you need them |
| 14:03.31 | brad_mssw | yaboo: but it's easy to replicate ... |
| 14:03.41 | Hmmhesays | most pbx's use some type of proprietary signaling to do all the fancy stuff |
| 14:03.44 | Katty | Hmmhesays: :> |
| 14:03.52 | *** join/#asterisk Thumann (i=Thumann@80.62.78.194) |
| 14:03.55 | Hmmhesays | Katty: whats the smirk for? |
| 14:04.06 | coppice | Hmmhesays: quite a few use Q.SIG |
| 14:04.14 | drewskie | yah got a toshiba strata that i want to faze out |
| 14:04.14 | Katty | Hmmhesays: (radio)++ |
| 14:04.20 | Hmmhesays | cool |
| 14:04.32 | Katty | Hmmhesays: (swiss internet radio - classical opera symphony)+++++++++ |
| 14:04.41 | Katty | Hmmhesays: you can't get that in missouri ;) |
| 14:04.58 | Hmmhesays | Katty: sure you can, over the intarwebs |
| 14:05.06 | Katty | Hmmhesays: yay, intarweb |
| 14:06.01 | Hmmhesays | those cialis commercials with "Bob" are great |
| 14:06.43 | Katty | bob and tom? |
| 14:07.06 | Hmmhesays | no, the ones for "natural male enhancement" bob is the smiling guy |
| 14:08.05 | *** join/#asterisk mkrufky (n=mk@68.160.103.77) |
| 14:08.20 | Katty | i....................see. |
| 14:09.29 | Ariel_ | morning everyone |
| 14:09.44 | Hmmhesays | is there a decent visio type program in linux? |
| 14:10.10 | crash3m | Hmmhesays: dia is out there, but it kinda lacks IMHO |
| 14:10.44 | crash3m | no arbitrary rotation, think it lacks templates, etc.. |
| 14:10.47 | bjohnson | Hmmhesays: for diagrams, etc? |
| 14:10.54 | Ariel_ | Hmmhesays, there are a few things that are better in windows...argh did I just say that. |
| 14:10.54 | Hmmhesays | yeah |
| 14:11.05 | bjohnson | Hmmhesays: haven't used it but openoffice is supposed to have something |
| 14:11.06 | Hmmhesays | Ariel_ yeah smart draw in windows just rocks |
| 14:11.17 | Hmmhesays | bjohnson: yeah OO blows for diagrams |
| 14:11.53 | *** join/#asterisk grimse (n=grimse@p5481C6CD.dip.t-dialin.net) |
| 14:12.55 | hugo-v6 | hmmm fresh selfmade bloody marry *sip* |
| 14:13.03 | hugo-v6 | skøll |
| 14:13.06 | tzanger | damn |
| 14:14.28 | basty | jpayne: After taking a non firewalled ip - still the same problem :-\ |
| 14:15.03 | *** join/#asterisk netnameus (n=netnameu@pcp05000344pcs.shrpsr01.tn.comcast.net) |
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| 14:15.40 | *** join/#asterisk RoyK (n=roy@216-99-212.0506.adsl.tele2.no) |
| 14:15.46 | Ariel_ | Ok so why would people in New Orleans fire on army helicopters trying to evacuate them? |
| 14:16.01 | netnameus | hi, i can get my xlite phone to connect to my asterisk box, but not my polycom 501... I've double checked all settings, but it just says "unregistered". Are there logs I can check to see why it's not connecting? |
| 14:16.07 | mutilator | cause people in the south are nuts? |
| 14:16.33 | Ariel_ | netnameus, how did you setup the polycom with an ftp server or via the phone menu or web? |
| 14:16.44 | netnameus | phone menu |
| 14:17.00 | Ariel_ | it's sick that you shoot the people that are trying to help you. |
| 14:17.04 | *** join/#asterisk file[laptop] (n=jcolp@mctnnbsa30w-156034035160.nb.aliant.net) |
| 14:17.35 | Ariel_ | netnameus, can I recommend you set it up via the ftp server. |
| 14:18.21 | *** join/#asterisk DI_PU (n=hamadjim@200.101.83.12) |
| 14:18.44 | netnameus | sure you can, but i may not do it ;) Just kidding. Yeah, do I just use regular FTP and manually edit the files or somethign? |
| 14:19.41 | Ariel_ | netnameus, yes I have only set mine up via the ftp. I tried the menu once and I kept missing one setting or another |
| 14:19.43 | *** join/#asterisk Gronker__ (n=Gronker2@adsl-220-90-11.ags.bellsouth.net) |
| 14:20.02 | Ariel_ | netnameus, you just need to point it to the ftp server you have and put the files there in the root. |
| 14:20.46 | tzanger | bkw_: around? |
| 14:20.50 | *** join/#asterisk Goshen (n=Goshen@c-67-172-238-57.hsd1.ut.comcast.net) |
| 14:20.56 | bkw_ | yes |
| 14:21.02 | netnameus | by "ftp server" do you mean polycom, or my asterisk box? |
| 14:21.10 | netnameus | (i've never tried it this way) |
| 14:21.20 | Ariel_ | you can put an ftp server on the asterisk box |
| 14:21.42 | Ariel_ | the settings are in the network setup on the menu you just point the ip address and user name and password on the phone there. |
| 14:21.58 | Ariel_ | unless you use an dhcp server to point it to your ftp for you. |
| 14:22.01 | netnameus | ok, let me experiment |
| 14:22.18 | Ariel_ | netnameus, http://www.voip-info.org/tiki-index.php?page=Polycom+Phones |
| 14:22.29 | *** join/#asterisk tekati (n=captain@cpe-66-75-215-63.bak.res.rr.com) |
| 14:22.38 | *** join/#asterisk apardo (n=apardo@90.Red-83-46-197.pooles.rima-tde.net) |
| 14:22.44 | bkw_ | tzanger, yes? |
| 14:22.45 | netnameus | ty |
| 14:23.01 | Ariel_ | also if you don't have upgrade the sip to 1.5.2 it's really great. Leave the firmware at 2.6.x don't upgrade the firmware 3.x |
| 14:23.40 | netnameus | i need to run out for a few, but i will be back to try this |
| 14:24.06 | tzanger | bkw_: just asking you a question regarding cisco phones in /msg |
| 14:24.17 | bkw_ | I didn't get a msg from ya |
| 14:24.20 | *** join/#asterisk rg1_ (n=rg1@mail.airlinksystems.com) |
| 14:24.55 | *** join/#asterisk cianhughes (n=cian@cian.ws) |
| 14:25.48 | Nivex | hail kram! |
| 14:26.05 | kram | greetings nivex |
| 14:26.16 | tzanger | bkw_: odd I can't send messages to you |
| 14:26.19 | tzanger | I get them from you |
| 14:26.20 | tzanger | heh |
| 14:26.21 | tzanger | anyway |
| 14:26.31 | bkw_ | learn to use IRC boi.. :P j/k |
| 14:26.33 | tzanger | I was just intending on asking some questions regarding the cisco sip phones you know and love |
| 14:26.34 | bkw_ | just ask |
| 14:26.38 | rg1_ | hello |
| 14:26.48 | cypromis | shitco |
| 14:27.01 | tzanger | regarding mulitple "line" appearances, the ability to be on a call and see another incoming anad decide whether to take it or not |
| 14:27.22 | bkw_ | yes you can do that |
| 14:27.23 | rg1_ | question: I am using XLite Sip softphone and I get an echo back of myself talking - what config file/parameter might I look at to try to adjust this? |
| 14:27.29 | bkw_ | the current call is put on hold if you answer it |
| 14:27.59 | rg1_ | hey there bkw_, got up and running - thanks for your help the other day on the XO/PRI |
| 14:28.11 | bkw_ | rg1_, np |
| 14:28.28 | rg1_ | can you (or anyone else for that matter) help me on the echo problem I just put up? |
| 14:28.37 | tzanger | bkw_: right I understand that part. what can you all do with the screen and soft buttons? are there any incompatibilities or weirdnesses with * and it with SIP? |
| 14:29.12 | `Sauron | Hum, are there any iaxclient devs around here? |
| 14:29.40 | tzanger | I'm kind of torn between the cisoc and the polycom |
| 14:29.45 | coppice | no. none at all |
| 14:30.28 | bkw_ | tzanger, the phone just works |
| 14:30.34 | tzanger | bkw_: that is good to know :-) |
| 14:31.08 | *** join/#asterisk rob314 (n=rob314@207.58.194.2) |
| 14:31.13 | tzanger | bkw_: any real reason to go 7960 instead of 7940 or 20? |
| 14:31.15 | rg1_ | anyone here know how to reduce/eliminate the echo of my own voice when i'm talking to someone else (I'm using a soft phone) |
| 14:31.37 | tzanger | bkw_: the speakerphone is good on it? |
| 14:31.38 | bkw_ | 7920 is a wifi sccp phone |
| 14:31.47 | tzanger | oh not 7920 then |
| 14:31.48 | bkw_ | I never had any issues with the speaker phone |
| 14:31.50 | spackle | rg1: with a softphone, you probably need a better headset. Do you have USB headset? Softphones suck. |
| 14:32.00 | rg1_ | yes USB headset |
| 14:32.07 | rg1_ | yes softphones suck |
| 14:32.12 | rg1_ | but they are cheap |
| 14:32.22 | Nugget | the only difference between the 7960 and thr 7940 is the number of lines. Unless you think you'll need all six lines on a 7960, the 7940 will probably do just fine for you |
| 14:32.25 | spackle | rg1: and you get what you pay for. |
| 14:32.30 | tzanger | bkw_: difference between 7960 and G? |
| 14:32.39 | bkw_ | G has pictures on the buttons |
| 14:32.41 | Nugget | the G has pictures instead of words on the buttons |
| 14:32.45 | bkw_ | the non-g has words |
| 14:32.46 | tzanger | that's it?? |
| 14:32.48 | Nugget | yep |
| 14:32.51 | bkw_ | G is Global |
| 14:32.52 | tzanger | g = global? |
| 14:32.53 | tzanger | yeah |
| 14:32.54 | tzanger | ok |
| 14:32.57 | bkw_ | ok people |
| 14:33.01 | bkw_ | OFF MY BRAIN WAVE |
| 14:33.08 | bkw_ | hehe |
| 14:33.09 | Nugget | but the bkw_ wave is the fun wave. |
| 14:33.15 | bkw_ | hehe |
| 14:33.18 | bkw_ | true true |
| 14:33.35 | *** join/#asterisk BladeRunner99 (n=feelme@adsl-198-213.37-151.net24.it) |
| 14:33.40 | coppice | G is for string |
| 14:33.48 | tzanger | haha |
| 14:34.01 | tzanger | so the 7940 is two lines, 7960 is 6 |
| 14:34.05 | Nugget | right |
| 14:34.28 | jack | or speeddials :) |
| 14:34.44 | bkw_ | tzanger, ya |
| 14:35.19 | BladeRunner99 | Hi all, wich voiptel u suggest budgetone 101, 102 o snom ? |
| 14:35.22 | tzanger | ok good good |
| 14:36.00 | Vco | buy one of each |
| 14:36.01 | rg1_ | spackle - do you know if there is anyway to adjust that feedback/echo? |
| 14:36.21 | BladeRunner99 | vco: why ? |
| 14:36.35 | Vco | why not? |
| 14:36.46 | Ariel_ | BladeRunner99, polycom |
| 14:37.09 | *** join/#asterisk DanielArndt (n=DanielAr@reverse-82-141-48-249.dialin.kamp-dsl.de) |
| 14:37.39 | BladeRunner99 | Ariel_: polycom at www.polycom.com ? |
| 14:37.45 | spackle | rg1: not very experienced with it, as I gave up on softphones, maybe reduce the input volume on the microphone? Are you hearing a definite echo or is it just loud sidetone? |
| 14:38.27 | *** join/#asterisk Borgon (n=L3orgon@70-100-53-251.dsl1.tbr.ga.frontiernet.net) |
| 14:38.30 | Ariel_ | BladeRunner99, polycom IP-300/1 for low end IP-500/1 for normal everyday use and IP-600 for poe and really top of the line phones |
| 14:38.32 | Borgon | ok guys, you all wish me good luck |
| 14:38.47 | Ariel_ | Borgon, why? in any good luck |
| 14:38.49 | Borgon | AFter much fighting am about to get asterisk working on linux |
| 14:38.53 | Borgon | i mean windows |
| 14:38.56 | coppice | polycom don't echo cancel their handsets |
| 14:39.21 | tzanger | coppice: really? I would have thought polycom would have had that figured out to a T, they're renowned for their kickass speakerphones |
| 14:39.22 | DanielArndt | hello everybody, |
| 14:39.26 | Borgon | Ariel_: ived tried asteriskwin32 and astwind and both suck... So i just installed colinux indepently and am gonna install asterisk no it from scratch |
| 14:39.38 | Borgon | so far colinux is just like a linux box |
| 14:40.00 | Ariel_ | Borgon, ok good luck. |
| 14:40.13 | Borgon | I havent slept since yesterday =( |
| 14:40.31 | coppice | tzanger: if you turn up the handset volume people complain a lot |
| 14:40.46 | DanielArndt | is it possible to "tunnel" an isdn connection with AIX through a Wireless Lan ? |
| 14:40.46 | Ariel_ | coppice, funny thing is I have a customer that put a filter (foam from an airfilter) over the mic's and they really help allot |
| 14:41.16 | drewskie | I have another quick newbie question, i have seven analog lines coming in, if i understand this correctly, i need an FXO port for each line? so a td card with 4 fxo modules and another one with 3? |
| 14:41.18 | tzanger | coppice: hmm... that really steers me toward cisco then |
| 14:41.37 | rg1_ | spackle - its definately an echo - when i talk into the microphone without being on a call - i hear myself, but in real-time/no-echo |
| 14:41.37 | MikeJ[Laptop] | tzanger, it's a price issue.. polycomm knows how to make a propper echo can |
| 14:41.46 | `Sauron | Ariel: Also known as a "pop shield" |
| 14:41.47 | rg1_ | when i'm on a call, i'm hearing myself delayed |
| 14:41.54 | rg1_ | only on an outside call though |
| 14:41.59 | MikeJ[Laptop] | rg1_, echo |
| 14:41.59 | rg1_ | on ext->ext inside, no echo |
| 14:42.06 | coppice | MikeJ: rubbish |
| 14:42.11 | rg1_ | MikeJ - yes? |
| 14:42.22 | MikeJ[Laptop] | coppice, which rubbish? |
| 14:42.34 | MikeJ[Laptop] | that polycom knows how to make a proper echo can? |
| 14:42.35 | coppice | you are talking rubbish |
| 14:42.42 | Ariel_ | drewskie, yes but there are known issue when running more then one tdm board on a box. |
| 14:42.47 | coppice | no. that it is a price issue |
| 14:42.55 | drewskie | ariel: irq issues right? i read about those on the wiki |
| 14:42.56 | tzanger | MikeJ[Laptop]: yes but if their ip phones haven't got this... |
| 14:42.57 | MikeJ[Laptop] | well, how else would you explain it? |
| 14:43.16 | *** join/#asterisk dsfr_ (n=dsfr@pdpc/sponsor/digium/dsfr) |
| 14:43.23 | drewskie | im pretty sure i can navigate those issues if that is all im facing.... |
| 14:43.24 | MikeJ[Laptop] | I have a polycom vortex.. can't get a much cooler echocan than that |
| 14:43.25 | Ariel_ | drewskie, yes but that is just part of it. You also have echo problems. |
| 14:43.28 | coppice | the phone has an EC which is used in speakerphone mode. the design is just a cock-up |
| 14:43.39 | drewskie | an echo problem? |
| 14:43.40 | MikeJ[Laptop] | ok, that I'd buy |
| 14:43.44 | MikeJ[Laptop] | echo |
| 14:43.47 | tzanger | Ariel_: a pair of TDM cards should run just fine in a box so long as they're not sharing IRQs with each other or anything else and the box works well with the TJ320 PCI bridge they use |
| 14:43.55 | spackle | rg1_: yep, is the external call digital over a PRI or through analog line? |
| 14:44.02 | MikeJ[Laptop] | echo |
| 14:44.12 | tzanger | MikeJ[Laptop]: the joke's getting old. :-) |
| 14:44.25 | coppice | MikeJ[Laptop]: the joke's getting old. :-) |
| 14:44.30 | Ariel_ | tzanger, yes I know I have about 3 systems with more then 1 in it and some with just 1. But echo on all of them is still a problem. |
| 14:44.50 | Borgon | OMG i love colinux the only true linux distro on windows without virtualization |
| 14:45.17 | drewskie | ariel_: like a big problem? like a little slight echo or to the point where you cant have a normal conversation? |
| 14:45.18 | tzanger | Ariel_: have you tried KB1? |
| 14:45.25 | bkw_ | OK |
| 14:45.26 | Vco | why don't you just install linux? |
| 14:45.29 | rg1_ | spackle - PRI |
| 14:45.29 | Ariel_ | tzanger, what is KB1 |
| 14:45.30 | *** join/#asterisk macTijn (i=martijn@linda.net.insecure.nl) |
| 14:45.35 | Borgon | Vco: i dont have another pc |
| 14:45.36 | tzanger | Ariel_: the KB1 echo canceller in CVS HEAD |
| 14:45.44 | bkw_ | YO macTijn |
| 14:45.44 | tzanger | basically a "fixed"/refactored MARK2 |
| 14:45.48 | Borgon | Vco: and this will be for personal use.. but colinux really looks promising |
| 14:45.48 | bkw_ | long time no see |
| 14:45.49 | Borgon | so fast |
| 14:46.00 | macTijn | eek! |
| 14:46.06 | macTijn | bkw: I had some connectivity problems :/ |
| 14:46.14 | bkw_ | really? |
| 14:46.20 | bkw_ | is it something super glue can fix? |
| 14:46.22 | Ariel_ | tzanger, is this part of asterisk build or zaptel |
| 14:46.22 | macTijn | this machine was being ddos'ed |
| 14:46.29 | tzanger | Ariel_: zaptel |
| 14:46.37 | macTijn | and our provider cut us off from any transit |
| 14:46.41 | spackle | rg1_: Have you adjusted your PRI settings? Such as gain and the echo cancelling settings? |
| 14:46.47 | Ariel_ | and what changes do we make on the makefile to get it? |
| 14:46.55 | rg1_ | spackel - nope |
| 14:47.01 | rg1_ | is that in zapata.conf? |
| 14:47.03 | tzanger | Ariel_: none. you edit zconfig.h and choose KB1 instead of MARK2 |
| 14:47.13 | macTijn | ah well |
| 14:47.16 | macTijn | working again now |
| 14:47.31 | macTijn | hmm |
| 14:47.33 | macTijn | 1.2.0 |
| 14:47.35 | macTijn | \o/ |
| 14:47.41 | Ariel_ | tzanger, do you know you can use the zaptel from head with stable |
| 14:48.02 | tzanger | I make a few other chances too which seem to help ecohcan in general... enable MMX, enable XLAW (optimize for few channels), and add -march=pentium4 (or whatever your CPU is) to CFLAGS and KFLAGS just under the comments about zconfig.h in the zaptel Makefile |
| 14:48.09 | tzanger | Ariel_: I am not sure |
| 14:48.13 | spackle | rg1_, my friend, it is time for you to read the wiki. Search on echo cancellation, echo avoidance, and echo. |
| 14:48.18 | Ariel_ | if it's better then mark2 then why is it not already set up for it. |
| 14:48.24 | *** part/#asterisk greekman (n=alex@host254.209.113.199.conversent.net) |
| 14:48.29 | tzanger | Ariel_: it's brand new in HEAD (about a week and a half old) |
| 14:48.30 | rg1_ | spackle - will do. thanks much |
| 14:48.33 | Ariel_ | tzanger, I am telling you can. I use the zaptel for head for stable |
| 14:48.40 | tzanger | oh |
| 14:48.42 | tzanger | I dind't know that |
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| 14:52.09 | *** part/#asterisk Akelavlk (n=jansun@82.119.239.141) |
| 14:52.30 | cpatry | is there any place that i could find whats the main topic of astricon? there's a list of names, but isnt complete. |
| 14:52.50 | greekman | anyone using snom 360 phones? |
| 14:55.05 | *** join/#asterisk file (n=jcolp@mctnnbsa30w-156034035160.nb.aliant.net) |
| 14:55.45 | Ariel_ | tzanger, I see the KB1 on the file now how do I tell it to use it? |
| 14:55.52 | *** join/#asterisk rob314 (n=rob314@207.58.194.2) |
| 14:55.54 | file | woot dependencies |
| 14:56.03 | tzanger | Ariel_: /* #define ECHO_CAN_MARK2 */ |
| 14:56.12 | tzanger | Ariel_: #define ECHO_CAN_KB1 |
| 14:56.18 | tzanger | instead of the opposite |
| 14:56.48 | *** join/#asterisk mhnoyes (n=mhnoyes@user-38lc118.dialup.mindspring.com) |
| 14:56.57 | file | how does KB1 work? |
| 14:57.12 | tzanger | file: very well IMO |
| 14:57.20 | file | neat |
| 14:57.24 | *** join/#asterisk wunderkin (i=kev@12-219-162-233.client.mchsi.com) |
| 14:58.23 | bkw_ | file dear |
| 14:58.39 | bkw_ | :P |
| 14:58.53 | file | yesssss? |
| 14:59.16 | Ariel_ | tzanger, I am sorry to ask but can you see the http://pastebin.ca/21832 and let me know how you change it. Looks like there off turn off. |
| 15:00.01 | tzanger | Ariel_: no not there |
| 15:00.08 | tzanger | zconfig.h there are a buch of #defines |
| 15:00.11 | tzanger | for ecoh cancel |
| 15:00.22 | Katty | hmm. |
| 15:00.35 | Ariel_ | tzanger, that is all the defines I found |
| 15:00.36 | netnameus | Ariel_: will the ftp method work with a dynamic ip with my phone (polycom) |
| 15:00.39 | *** join/#asterisk Twister (i=Twister@24-179-91-088.dhcp.chtn.wv.charter.com) |
| 15:00.43 | tzanger | ? |
| 15:00.46 | Ariel_ | netnameus, yes |
| 15:01.19 | Borgon | yes |
| 15:01.19 | netnameus | what files do i need in the dir, just ipmid.cfg, phone1.cfg, and sip.cfg? |
| 15:01.21 | tzanger | Ariel_: http://pastebin.ca/21834 |
| 15:01.22 | netnameus | and it will just use those? |
| 15:01.24 | Katty | hmmmm |
| 15:01.42 | Twister | ayone in here gotten the distinctive ring routing to work with * |
| 15:02.21 | Twister | from broadvoice |
| 15:02.36 | *** join/#asterisk punk_kaos (n=kaos@dsl001-133-196.pdx1.dsl.speakeasy.net) |
| 15:02.39 | Ariel_ | tzanger, strange I can't fine that in the file. |
| 15:02.49 | tzanger | ... zaptel's zconfig.h |
| 15:02.53 | tzanger | not asterisk |
| 15:03.23 | Ariel_ | tzanger, I had the wrong file my bad. I was looking at zaptel.h |
| 15:03.42 | tzanger | :-) |
| 15:04.17 | BrianR___ | aparently there is an expansion key module for the grandstream gxp-2000? Has anyone seen it? |
| 15:05.51 | *** part/#asterisk Gronker__ (n=Gronker2@adsl-220-90-11.ags.bellsouth.net) |
| 15:06.08 | bkw_ | BrianR___, is it out yet |
| 15:06.26 | bkw_ | I know it was a future item |
| 15:06.48 | BrianR___ | bkw_: No idea. Some sites selling the GXP-2000 refer to it as currently available.... |
| 15:07.00 | bkw_ | I doubt it |
| 15:07.01 | BrianR___ | Of course there were lots of other things about the GXP-2000 in that class until recently... |
| 15:07.30 | *** join/#asterisk sams2100 (n=sams@pcp424683pcs.naugus01.ga.comcast.net) |
| 15:07.39 | bkw_ | I had the pleasure of being totally insulted by the GrandStream CEO at VON |
| 15:07.42 | bkw_ | it was quite funny |
| 15:07.43 | BrianR___ | If it had BLF and SIP MESSAGE support (epemeral and read-only would be just fine with me) it'd be pretty much feature complete.. |
| 15:08.03 | bkw_ | needless to say him and I have talked about this very key expansion module |
| 15:08.07 | bkw_ | they haven't delivered |
| 15:08.08 | Hmmhesays | did he call you 'teh ghey'? |
| 15:08.30 | bkw_ | no he goes "Oh you know that brian guy on the mailing list he can be so abrasive and mean at times" |
| 15:08.33 | BrianR___ | bkw_: It seems the management / marketing guys there are making promises before engineering can deliver... |
| 15:08.35 | *** join/#asterisk nain (n=nain@137.101.144.131) |
| 15:08.37 | bkw_ | I hold up my name tag and go "You mean me?" |
| 15:08.46 | bkw_ | he fell over |
| 15:08.54 | Hmmhesays | ROFL, classic bkw_ |
| 15:09.01 | bkw_ | shook my hand and offered to buy me a beer |
| 15:09.18 | bkw_ | the best part is I know i'm a bitch.... but i'm not in person |
| 15:09.21 | BrianR___ | Heh.. Take advantage of that audience with the embarassed CEO to get features you want accelerated :) |
| 15:09.29 | bkw_ | hehe |
| 15:09.54 | *** join/#asterisk DanielArndt (n=DanielAr@reverse-82-141-48-249.dialin.kamp-dsl.de) |
| 15:09.55 | Hmmhesays | someone should explain to him the concept of free tech support |
| 15:10.13 | *** join/#asterisk CoffeeIV (n=CoffeeIV@mail.airlinksystems.com) |
| 15:10.20 | DanielArndt | hi, a newbie question, what means: error while writing audio data: : Broken pipe |
| 15:10.22 | Hmmhesays | you don't have to go through all of the "be nice as you can" bs, you can just get straight to the point |
| 15:10.37 | DanielArndt | how can i fix this? |
| 15:10.45 | bkw_ | that is not an error |
| 15:10.47 | nain | Hi every body |
| 15:10.50 | bkw_ | thats from mpg123 |
| 15:10.56 | DanielArndt | oh |
| 15:11.01 | bkw_ | you can ignore it |
| 15:11.02 | bkw_ | NEXT!!! |
| 15:11.20 | DanielArndt | but asterisk kill after that... |
| 15:11.30 | Hmmhesays | <PROTECTED> |
| 15:11.31 | Hmmhesays | whats up |
| 15:11.35 | bkw_ | DanielArndt, something else is up |
| 15:11.39 | *** join/#asterisk elusive (n=elusive@elusive.net0.feitoza.com.br) |
| 15:11.42 | nain | bkw_: i also got the same kind of error and call hangup "Unable to write to fd 86 (32, Broken pipe)." |
| 15:11.57 | nain | Hi Hmmhesays |
| 15:11.58 | netnameus | Ariel_: is there a way to reboot my polycom w/o unplugging and plugging back in? |
| 15:12.05 | elusive | can anyone help me? |
| 15:12.08 | elusive | I got this message: |
| 15:12.08 | DanielArndt | hmm just get damn warning before that.. will play around a litte thanks |
| 15:12.13 | elusive | ZT_CHANCONFIG failed on channel 1: Invalid argument (22) |
| 15:12.14 | elusive | Did you forget that FXS interfaces are configured with FXO signalling |
| 15:12.14 | elusive | and that FXO interfaces use FXS signalling? |
| 15:12.16 | Ariel_ | netnameus, yes look on the menu on the phone |
| 15:12.28 | BrianR___ | netnameus: hold down +/- hold and messages |
| 15:12.32 | Ariel_ | netnameus, also there is a script you can put in the asterisk box to do this. |
| 15:12.33 | nain | Hmmhesays: i am again back with error "Unable to write to fd 86 (32, Broken pipe)." |
| 15:12.37 | netnameus | ahh, found it |
| 15:12.55 | netnameus | i'm still trying to get it to connect to my asterisk box ;) |
| 15:12.56 | Hmmhesays | nain: what is that causing asterisk to do? |
| 15:12.57 | sivana | if I have an * module that uses a utility app, do I need to give up source to the utility? |
| 15:13.06 | *** join/#asterisk mtaq (n=chatzill@198.111.183.2) |
| 15:13.33 | CoffeeIV | I want to have someone selecting a certain option on a voice menu ring several phones one at a time until someone picks up. I thought agents and call queues would be the way to do this. However, I don't want the agent to have to log in -- I want this to be a perment setup. Is there a different way to do this ? |
| 15:13.49 | Twister | i cant seem to get it working right..the question i have is is the patch for this in the newest -HEAD or do i need to apply it manually, and if i need to apply it, why does most of it fail |
| 15:13.55 | nain | Hmmhesays: I don't know what's it causing but it's an h323 call while i m using asterisk oh323 0.6.6 driver. |
| 15:14.05 | Ariel_ | CoffeeIV, yes create a ring group |
| 15:14.13 | *** join/#asterisk drbrown (n=chatzill@63.238.118.90) |
| 15:14.27 | CoffeeIV | Ariel: I'll look it up on voip-info, thanks |
| 15:15.05 | Ariel_ | CoffeeIV, you can ring more then one phone like dial(sip1&Sip2&Sip3) if you create a macro to do it with you can control it. |
| 15:15.16 | CoffeeIV | ok |
| 15:15.17 | *** join/#asterisk [1]JohnJacob (n=JohnJaco@pcp0011543387pcs.mainf01.in.comcast.net) |
| 15:15.19 | *** join/#asterisk twisted[work] (n=twisted@asterisk/friend-and-developer/pdpc.professional.twisted) |
| 15:15.20 | *** mode/#asterisk [+o twisted[work]] by ChanServ |
| 15:15.32 | nain | Can any one help me to solved this error: Call dropped after this error "Unable to write to fd 86 (32, Broken pipe)" |
| 15:17.37 | nain | bkw_: can u plz help |
| 15:17.37 | coppice | does anyone know anywhere in * that a read can return more than one frame? |
| 15:18.13 | nain | Hmmhesays: did u got any idea about it ? |
| 15:18.22 | Hmmhesays | nain: sorry I have no idea what it is |
| 15:18.57 | twisted[work] | nain, you're out of file descriptors. go to compusa and ask for more. |
| 15:19.32 | hugo-v6 | nain: u can buy them also from me, 1$ for each |
| 15:19.54 | *** join/#asterisk Ferrari_ (n=Ferrari@66.64.128.142.nw.nuvox.net) |
| 15:19.56 | twisted[work] | lol |
| 15:19.58 | *** join/#asterisk Punk_Kao1 (n=root@dsl001-133-196.pdx1.dsl.speakeasy.net) |
| 15:20.08 | coppice | you mean file descriptors like "that ***** file has so many ****** errors it makes you want to ******"? |
| 15:20.08 | hugo-v6 | ideed ure ;) |
| 15:20.39 | *** join/#asterisk crich1999 (n=crich@p54BFA998.dip0.t-ipconnect.de) |
| 15:20.46 | twisted[work] | coppice, lol |
| 15:20.51 | nain | twisted[work]: hugo-v6: file descriptors ? what is it and how it is relate to error "Unable to write to fd 86 (32, Broken pipe)." |
| 15:21.04 | twisted[work] | fd == file descriptor |
| 15:21.18 | Katty | twisted[work]: hi. |
| 15:21.23 | twisted[work] | hi Katty :) |
| 15:21.24 | hugo-v6 | nain: u can change them with ulimit/pam/kernel |
| 15:21.35 | *** join/#asterisk eKo1 (n=bernd@metrored-gw.tropicohn.com) |
| 15:21.36 | Katty | twisted[work]: i need a shopping co-conspirator |
| 15:21.49 | hugo-v6 | nain: iirc the asterisk README tells about how to change them |
| 15:21.52 | twisted[work] | uh oh |
| 15:21.52 | Uther_P | how could it be that he is "out" of file descriptors... if it didn't have one to allocate in the first place, wouldn't the call have never established in the first place? he said that the call drops... implying that his call was connected, *then* was disconnected |
| 15:21.59 | Katty | twisted[work]: you're under 300 miles away. |
| 15:22.06 | Katty | twisted[work]: whichs make you acceptable for kidnapping |
| 15:22.13 | twisted[work] | Uther_P, ir you read above, I was joking. |
| 15:22.17 | twisted[work] | s/ir/if |
| 15:22.19 | nain | hugo-v6: hmmmmmm, thanks let me check |
| 15:22.21 | *** part/#asterisk Punk_Kao1 (n=root@dsl001-133-196.pdx1.dsl.speakeasy.net) |
| 15:22.48 | *** join/#asterisk punk_kaos (n=kaos@dsl001-133-196.pdx1.dsl.speakeasy.net) |
| 15:22.48 | hugo-v6 | nain: but maybe it isnt like uther_p said. never got that problem ;) |
| 15:23.13 | twisted[work] | more than likely he's not out of file descriptors, but something happened to the one that was in use |
| 15:23.31 | nain | hugo-v6: is it related to security limit of kernal files? |
| 15:23.41 | twisted[work] | Katty, okay, come get me |
| 15:23.42 | twisted[work] | ;) |
| 15:23.43 | *** join/#asterisk Assid (n=assid@203.115.64.59) |
| 15:23.44 | Ariel_ | tzanger, the new KB1 seems to help allot |
| 15:23.46 | Katty | twisted[work]: :< |
| 15:23.54 | hugo-v6 | nain: well, dont hink so. |
| 15:23.57 | hugo-v6 | +t |
| 15:24.14 | twisted[work] | mmm doughnuts. |
| 15:24.15 | twisted[work] | brb. |
| 15:24.19 | Hmmhesays | dcc me some |
| 15:26.45 | Thumann | mmm coke from yesterday... |
| 15:26.46 | Thumann | :/ |
| 15:26.47 | Uther_P | you can't send donuts through dcc silly |
| 15:26.48 | groogs | woohoo, new battery for my laptop.. 95% = 2:10h instead of 40 minutes |
| 15:26.50 | Uther_P | you gotta fax them |
| 15:26.56 | nain | hugo-v6: but how can i find that my machine is PAM? |
| 15:27.03 | Katty | Hmmhesays: come visit |
| 15:27.07 | Katty | Hmmhesays: for shopping porpoises. |
| 15:27.30 | Uther_P | heh, I'd like to see the mall that shopping porposes go to |
| 15:27.51 | Uther_P | nothing but fish resteraunts |
| 15:27.55 | Hmmhesays | i'm all full up on porpoises though |
| 15:27.58 | tzanger | Ariel_: yup I know :-) |
| 15:28.02 | tzanger | Ariel_: try the new fxotune too |
| 15:28.18 | tzanger | Uther_P: you can, but your cdrom needs serous cleaning afterward |
| 15:28.41 | nain | hugo-v6: after making changes in /etc/security/limits.conf do i need to restart the system or just aterisk ? |
| 15:29.22 | Ariel_ | tzanger, ok it's in the zaptel ./fxotune |
| 15:29.24 | Katty | Hmmhesays: k |
| 15:29.25 | Ferrari_ | good day all.... anyone know what i should set TOS= inorder to work with a dlink managed switch it only supports dscp values 1 - 64 |
| 15:29.34 | Uther_P | tzanger: meh, just access it directly via address xf00d |
| 15:30.00 | tzanger | Uther_P: :-) |
| 15:30.08 | *** join/#asterisk Henguei (n=smoka@196.203.249.98) |
| 15:30.09 | Hmmhesays | i'd point that sh1t straight at the sky and shoot heaven on down for you |
| 15:30.18 | netnameus | now my polycom keeps saying "error loading config file phonedefault.cfg" and then resetting |
| 15:30.31 | hugo-v6 | nain: stop asterisk log out log in restart asterisk |
| 15:30.53 | *** join/#asterisk stkn (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 15:31.02 | Ariel_ | netnameus, you need to make the right phone.cfg and point via the mac.cfg |
| 15:31.12 | twisted[work] | Katty, am I really under 300 miles away? |
| 15:31.19 | nain | hugo-v6: but as Katty said is it realy the problem of file descriptor ? |
| 15:31.23 | Katty | twisted[work]: yes. |
| 15:31.27 | twisted[work] | Katty, sweet |
| 15:31.37 | twisted[work] | i didn't realize that |
| 15:31.47 | Katty | twisted[work]: 249, actually |
| 15:31.53 | netnameus | Ariel_, would you mind looking at my ftp folder at the files and telling me what i need to fix? I'm kinda lost |
| 15:31.56 | twisted[work] | hehe, that's one tank of gas for me |
| 15:32.14 | Hmmhesays | whats that about 75 bucks right now? |
| 15:32.15 | Katty | twisted[work]: actually, you probably shouldn't. |
| 15:32.18 | Ariel_ | netnameus, yes but after I get back from a service call I need to go too. |
| 15:32.22 | twisted[work] | Hmmhesays, hahaha, roughly. |
| 15:32.24 | Katty | twisted[work]: being a whole FOOT TALLER THAN ME would surely make me implode. |
| 15:32.25 | *** join/#asterisk Phoebers (n=phoenix_@82-168-146-4-bbxl.xdsl.tiscali.nl) |
| 15:32.32 | twisted[work] | Katty, :( |
| 15:32.36 | Phoebers | hello :) |
| 15:32.46 | Hmmhesays | I need to ride my motorcycle more |
| 15:32.52 | Ferrari_ | twisted.... you familiar with TOS settings in astersik |
| 15:33.16 | twisted[work] | Ferrari_, in what way? |
| 15:33.44 | Ferrari_ | trying to figure out how to translate tos=??? in sip.conf to match dscp values in my dlink switch |
| 15:33.50 | Katty | i'm nearly almost starting to look into a pilot's license. |
| 15:34.00 | Hmmhesays | Katty: me too |
| 15:34.04 | Thumann | Katty: Nearly almost? |
| 15:34.12 | Katty | Thumann: that's what i said, bunny brad. |
| 15:34.14 | Katty | i mean bread. |
| 15:34.15 | Thumann | Katty: :> you're almost there then... |
| 15:34.19 | Hmmhesays | actually i'm into the I'm seriously looking into it phase |
| 15:34.26 | Katty | Hmmhesays: neato, post urls. |
| 15:34.40 | Hmmhesays | i'm going to go through the xplane training first I think to see |
| 15:35.20 | Thumann | a friend of mine.. took the license.. spend his live savings on a plane... crash landed on a field.. broke a leg.. now he reads books.. |
| 15:35.39 | Hmmhesays | xplane is the ultimate flight simulator |
| 15:35.45 | Hmmhesays | even the airforce uses it |
| 15:35.53 | *** join/#asterisk klasstek (n=nunyobiz@sta-206-168-218-206.rockynet.com) |
| 15:36.11 | Thumann | unlike the terrorists who use google earth and MS flight sim hehe |
| 15:36.26 | Katty | zomgterrorists |
| 15:36.28 | Uther_P | eh? a friend of mine is a developer for the software that the air force uses |
| 15:36.31 | Hmmhesays | xplane is more accurate |
| 15:36.46 | twisted[work] | Ferrari_, uhm, TOS= tos bits. |
| 15:36.54 | Hmmhesays | Uther_P: the airforce uses xplane for certain flight training |
| 15:37.05 | Hmmhesays | I don't remember what exactly, but I could research |
| 15:37.23 | Ferrari_ | twisted correct and as i understand it the dscp is a translation of tos bits |
| 15:37.24 | Thumann | the real question is, to quote the muppets, WHO CARES! |
| 15:37.25 | Thumann | :D |
| 15:37.45 | netnameus | thanks Ariel_, i PMed you |
| 15:38.31 | Hmmhesays | ms flight sim uses a huge database of different statistics to dictate flight characteristics, xplane calculates them on the fly |
| 15:38.50 | *** join/#asterisk rbd (n=robbyd@24-181-194-064.dhcp.hckr.nc.charter.com) |
| 15:38.52 | wiizard | I have an unrelated mission for you all.. |
| 15:39.01 | wiizard | Can anyone find out which company owns 65.18.203.2 |
| 15:39.10 | Hmmhesays | www.blocklist.org |
| 15:39.12 | tzanger | wiizard: whois -h whois.arin.net |
| 15:39.38 | *** join/#asterisk djin_ib (n=djin_ib@gridfox.xs4all.nl) |
| 15:39.43 | mutilator | http://ws.arin.net/whois/?queryinput=!%20NET-65-18-200-0-1 |
| 15:40.03 | Ferrari_ | so is anyone in here using TOS by chance with a dlink managed switch |
| 15:40.49 | Hmmhesays | http://www.x-plane.com/about.html |
| 15:41.34 | *** join/#asterisk dolson (n=dana@toronto-HSE-ppp4301949.sympatico.ca) |
| 15:41.45 | *** join/#asterisk asterisk_newbie (n=root@12-203-248-191.client.insightBB.com) |
| 15:42.31 | Hmmhesays | haha they have a map of mars too |
| 15:42.33 | tzanger | Ferrari_: just use the damn lowlatency tos bit |
| 15:42.39 | asterisk_newbie | can someone help me i am having trouble getting my asterisk box to connect to my sip proxie and was wondering if someone could look at my sip.conf and see what im missing |
| 15:43.04 | Uther_P | gheh, funny.. xplane's site says it runs on "PC, Mac and Linux systems" <-- pc and mac are architectures, linux and mac are OS's, but linux isn't and arch and PC isn't an os |
| 15:43.50 | Thumann | i run a pc system and a linux system on the same box!!! |
| 15:43.53 | Thumann | hehe |
| 15:44.02 | Ferrari_ | tzanger thanks.... should the managed switch work out of the box then |
| 15:44.02 | Uther_P | no, you run a linux os on a PC |
| 15:44.12 | Thumann | Uther_P: no kidding |
| 15:44.15 | Uther_P | linux *based* os, heh |
| 15:44.37 | asterisk_newbie | can someone help me i am having trouble getting my asterisk box to connect to my sip proxie and was wondering if someone could look at my sip.conf and see what im missing |
| 15:44.56 | vlrk | hello friends "cctai" like this is there any tool to know the current channel capacity ? |
| 15:45.02 | vlrk | in asteirsk |
| 15:45.41 | Thumann | sec.. i'll just boot into pc |
| 15:45.42 | Thumann | hehe |
| 15:45.58 | dolson | can you use the n priority in ARA extensions? |
| 15:46.42 | *** join/#asterisk dsfr_ (n=dsfr@pdpc/sponsor/digium/dsfr) |
| 15:46.44 | tzanger | Ferrari_: should, IIRC the TOS field is only 6 bits wide anyway |
| 15:47.53 | Hmmhesays | oooohhh xplane has a shuttle module, rock |
| 15:49.59 | *** join/#asterisk _cleric_ (n=dacleric@p548281AF.dip0.t-ipconnect.de) |
| 15:54.54 | pauldy | hrm nerd porn |
| 15:54.57 | vlrk | any tools kind of thing for asterisk channels monitoring |
| 15:54.59 | malverian[work] | OH MY!!! |
| 15:55.06 | Assid | vlrk : sip show channels |
| 15:55.06 | Hmmhesays | where? |
| 15:55.09 | malverian[work] | I have no idea how to set up multiple cards in my zaptel config :-P |
| 15:55.15 | pauldy | your xplane shuttle gewber |
| 15:55.15 | Hmmhesays | i wish you could drink at work |
| 15:55.30 | vlrk | no no .. |
| 15:55.35 | Uther_P | I can |
| 15:55.37 | Uther_P | :D |
| 15:55.44 | Hmmhesays | they frown upon it here though |
| 15:55.46 | twisted[work] | Hmmhesays, who says you can't? |
| 15:55.50 | twisted[work] | oh |
| 15:55.57 | Uther_P | just get a pocket flask |
| 15:55.57 | vlrk | Assid i think one tool that starts like ctai.. something like this |
| 15:56.22 | vlrk | yesterdya brc_ or some body was suggesting me i did not remember properly |
| 15:56.26 | Hmmhesays | making stupid sales network diagrams sucks |
| 15:56.38 | pauldy | it would be kewl if someone would write a module like meetme where you dial a number then dial th ext and now you have listen only |
| 15:57.02 | *** join/#asterisk `Sauron (i=sauron@h-69-3-12-50.hstqtx02.covad.net) |
| 15:57.02 | Hmmhesays | um, start meetme muted? |
| 15:57.18 | pauldy | hmmhesays then you would have to conference all outbound calls in |
| 15:57.27 | cpatry | Monitor ur conf. |
| 15:57.45 | malverian[work] | Hmm.. |
| 15:58.15 | *** join/#asterisk `Sauron (i=sauron@h-69-3-12-50.hstqtx02.covad.net) |
| 15:58.15 | malverian[work] | What if I have a FXO/FXS card, and also a T1 card.. can I configure them both in the same zaptel.conf ? |
| 15:58.25 | *** join/#asterisk redder86 (n=lee@gateway.howardsilvan.com) |
| 15:58.41 | malverian[work] | Ohhh.. I see.. |
| 15:58.42 | malverian[work] | Nevermind :) |
| 15:58.45 | JerJer | malverian[work]: sure |
| 16:00.08 | Assid | umm.. if i have a entry with n .. and then i use n+101 |
| 16:00.20 | Assid | does the next one have to be n+102 or just n again? |
| 16:00.31 | *** join/#asterisk `Sauron (i=sauron@h-69-3-12-50.hstqtx02.covad.net) |
| 16:01.13 | Uther_P | assid: you mean the dialplan order for an extension? |
| 16:01.14 | pauldy | Assid, I guess it depends on how you want it to act |
| 16:01.39 | Assid | well.. im trying to set it such if it doesnt work through one Dial() it should try another route |
| 16:01.54 | *** join/#asterisk Ralcala (n=operador@148.223.28.169) |
| 16:02.15 | Assid | but when i tried.. i encountered n,Congestion which is officially my last entry |
| 16:02.26 | redder86 | JerJer: I sent an e-mail to support about a week ago and haven't had a response. Is there a problem? |
| 16:02.29 | Assid | my 2nd Dial() is n+101 |
| 16:03.03 | pauldy | Assid, chane your congestion to attempt another dial |
| 16:03.07 | Ralcala | i just finished to upgrade my fc4 box with a brand new 2.6.13 fresh kernel with the new SIP_CONNTRACK module.... |
| 16:03.28 | Assid | so i just stick to n ? no need off n+101 ? |
| 16:03.48 | Uther_P | if you jump to n+101, then the next after it would be n+102, i fyou dont jump, then its n+1 |
| 16:03.57 | Ralcala | now i gonna make a template in fwbuilder.... |
| 16:04.25 | pauldy | there should be no jump if I remeber how the dial plans of most ext.confs I've seen |
| 16:04.31 | pauldy | you just dial dial dial congestion |
| 16:04.45 | pauldy | no n+101 crap cause there are no branches |
| 16:04.49 | Ralcala | stay tuned if i got succed on this...voip vs. nat its gonna be history fellas.... |
| 16:05.20 | Uther_P | ah true, because it waits on a dial unti lthe call is terminated... and when that happens it exits the dialplan anyway |
| 16:05.22 | pauldy | man I've been running voip with sip for about 6 months now |
| 16:05.26 | redder86 | I'm looking at writing a simple IAX client program. Looking at libiax... it seems that it hasn't been touched in nearly 4 years. Does that mean that libiax is rock-solid or does it mean that it's abandoned and rotting? |
| 16:05.34 | pauldy | without the lame arse ext ip etc al. |
| 16:05.54 | *** join/#asterisk glm2k (n=GLM@rrcs-24-199-11-45.west.biz.rr.com) |
| 16:06.03 | Ralcala | whit fw behind... pauldy |
| 16:06.12 | pauldy | linksys |
| 16:06.19 | Ralcala | router |
| 16:06.34 | pauldy | linux based router |
| 16:06.38 | Ralcala | linksys got a tiny fw pal.. |
| 16:06.59 | Ralcala | iptables? |
| 16:07.02 | pauldy | yup |
| 16:07.13 | Ralcala | version |
| 16:07.26 | pauldy | 2.4 something ruther |
| 16:07.46 | Ralcala | 2.4? |
| 16:07.47 | Hmmhesays | i need a new domain |
| 16:08.03 | Ralcala | maybe you're talking about kernel |
| 16:08.17 | Ralcala | iptables its on 1.3.3 now... |
| 16:08.37 | Ralcala | do you have hardphnoes or softphones? |
| 16:08.53 | pauldy | sorry kernel is 2.4 iptables v1.2.11 |
| 16:09.10 | pauldy | Linux sgc 2.4.20 |
| 16:09.35 | Tabo | guys, do you have to setup port forwarding on routers for all the voip lines in asterisk? |
| 16:09.45 | Tabo | I currently have two voip lines |
| 16:09.57 | *** part/#asterisk redder86 (n=lee@gateway.howardsilvan.com) |
| 16:11.36 | pauldy | Tabo I can't imagine why you wold as long as they are directed to the machine that will accept all inboud sip requests |
| 16:12.08 | vlrk | do we have any tools where we can get the active channels history in asterisk with graphs kind of thing |
| 16:12.11 | Ralcala | this is my scenerio pauldy, grandstream BT100+callserver+firewall ------ firewall+grandstream BT100.. |
| 16:12.54 | *** join/#asterisk Sedorox (i=brandon@smartserv/cna/Sedorox) |
| 16:13.05 | BladeRunner99 | C'e' qualche italiano qui? |
| 16:13.17 | pauldy | Ralcala in that situation I would do a vpn between the machines |
| 16:13.35 | pauldy | then you don't have to worry about NAT in the least if it is causing you issue |
| 16:13.57 | pauldy | what kind of firewall/routers do you have |
| 16:14.05 | Ralcala | i can make signal pass thru fw's but no audio for both sides |
| 16:14.08 | Tabo | pauldy: what I did was I dmz the asterisk box |
| 16:14.17 | Tabo | I have dlink router, no unix firewall |
| 16:14.24 | *** join/#asterisk FuriousGeorge (n=ads@pool-151-198-123-165.nwrk.east.verizon.net) |
| 16:14.26 | Uther_P | thats how I do mine too... just setup an ipsec tunnel between the networks |
| 16:14.30 | Ralcala | fedora's |
| 16:14.54 | *** join/#asterisk shmaltz (n=chatzill@69.28.255.210) |
| 16:15.04 | *** join/#asterisk Borgon (n=L3orgon@70-100-53-251.dsl1.tbr.ga.frontiernet.net) |
| 16:15.05 | pauldy | Ralcala, the diff between your setup and mine is my other end is broadvoice and they have public ips for their machines |
| 16:15.20 | Borgon | ep 1 12:12:19 NOTICE[236]: res_musiconhold.c:493 monmp3thread: Request to sche |
| 16:15.29 | Borgon | How can i cancel that message from displaying? |
| 16:15.31 | pauldy | Tabo, probably get your asterisk box out of the DMZ asap |
| 16:15.36 | Ralcala | no fw behind other end? |
| 16:15.51 | pauldy | next you should be fine with providers as you are registering witht hem |
| 16:16.20 | pauldy | Ralcala, on my side just my firewall on theirs while I'm sure they have them they are abstracted |
| 16:16.35 | pauldy | from the equation |
| 16:16.35 | Uther_P | my t1 provider is also my voip provider, and they setup all their sip and media servers with rfc1918 addresses, which screwed me up cause I had to rework my firewalls to allow their ip's in from the outside network :/ |
| 16:16.46 | *** join/#asterisk ManxPower (i=eric@1Cust5320.an8.dfw28.da.uu.net) |
| 16:17.46 | pauldy | I still don't know why they don't use at least a 40 bit mppe connection for a lot of these voip providers |
| 16:17.50 | Tabo | paul, but do I need to setup port forwarding for ports 5060 with my asterisk box |
| 16:18.11 | pauldy | Tabo I can tell you I don't have any ports forwarded to my asterisk box |
| 16:18.37 | pauldy | I use broadvoice and it works fine, registering to another provider or line should not cause any issue |
| 16:18.38 | Tabo | ok |
| 16:18.45 | Uther_P | tabo: you should only have to have the nat setting turned on for the clients that are behind a nat |
| 16:18.50 | pauldy | it all has to do with port traversal |
| 16:18.55 | Tabo | yes my nat setting is off |
| 16:19.03 | pauldy | port 5060 should only be connected to by you to your provider |
| 16:19.18 | Tabo | Thanks Paul and Uther |
| 16:19.20 | pauldy | they will respond ont he UDP port your machine picked when sending the request |
| 16:19.38 | pauldy | the router should remember this in a related table and pass the packet to your asterisk box |
| 16:19.53 | *** join/#asterisk adrenaline (n=mike@h-68-164-41-130.sttnwaho.covad.net) |
| 16:20.14 | Uther_P | out of curiosity... when sip establishes a call, which end decides what ports to use for the rtp media stream? |
| 16:20.33 | wiizard | device |
| 16:20.36 | Tabo | hey pauldy |
| 16:20.41 | Hmmhesays | look at the sip debug, you'll have your answer |
| 16:20.44 | pauldy | yea Tabo |
| 16:20.45 | Tabo | one of my voip lines isn't setup properly |
| 16:20.50 | Tabo | what happens is when I call the number |
| 16:20.59 | Tabo | it just goes straight to voicemail |
| 16:21.02 | Tabo | do you have any idea the cause? |
| 16:21.04 | Hmmhesays | ivegotmail.org is available |
| 16:21.08 | pauldy | ok two things, |
| 16:21.11 | Tabo | I see the attempts in teh deub |
| 16:21.12 | Tabo | debug |
| 16:21.16 | pauldy | #1 do you have the line setup properly in steerisk |
| 16:21.22 | Tabo | not sure |
| 16:21.24 | spackle | Tabo, you have some things working now? |
| 16:21.25 | Tabo | debug shows it |
| 16:21.27 | Tabo | yes |
| 16:21.28 | pauldy | is it registering |
| 16:21.36 | Tabo | spackle: thanks to your tips, my phone works |
| 16:21.38 | Uther_P | seems it would make more sense to have standardized the port(s) that the rtp media streams use... would have been simpler on the nat traversal process |
| 16:21.41 | pauldy | do you get an ok for that account when registering |
| 16:21.46 | Tabo | but now I have problems with the voip line |
| 16:21.57 | Tabo | ok hold |
| 16:22.11 | *** join/#asterisk newmember (n=newmembe@dsl-lkbn-66-18-211-34-cgy.nucleus.com) |
| 16:22.13 | pauldy | the next thing to check is if you have some sort of autodeny for unidentified incomming sip calls |
| 16:22.41 | pauldy | are you using anything like asterisk at home? |
| 16:22.52 | Tabo | paul: yes I'm using asterisk @ home |
| 16:23.28 | Uther_P | someday I'll have to look into asterisk@home, just to see why so many people say it sucks |
| 16:23.36 | pauldy | ok did you add a did for the number you are trying to dial |
| 16:23.45 | faa_ | Can someone help me with using astxs |
| 16:23.57 | Tabo | asterisk1*CLI> sip show peers |
| 16:23.58 | Tabo | Name/username Host Dyn Nat ACL Mask Port Status |
| 16:23.58 | Tabo | phone1/phone1 192.168.0.156 D 255.255.255.255 5060 Unmonitored |
| 16:23.58 | Tabo | DV/6046280000 209.17.160.129 N 255.255.255.255 5060 Unmonitored |
| 16:24.01 | cpatry | faa: what? |
| 16:24.12 | pauldy | wow use pastbin or something |
| 16:24.15 | Tabo | Uther: i'm new to unix, asterisk@home is a dream come true to me |
| 16:24.15 | Tabo | haha |
| 16:24.21 | spackle | Uther_P, it only sucks if you are used to doing it the hard way. It's actually pretty slick. |
| 16:24.23 | *** join/#asterisk yogurt2ungue (n=yogurt2u@44-170-114-200.fibertel.com.ar) |
| 16:24.23 | wiizard | uther the proxy usually defines the RTP port range |
| 16:24.35 | Tabo | and I love the GUI interface it rocks |
| 16:24.39 | faa_ | cpatry I wrote an apllication for asterisk, and want now to install it, but that apps must include some additional libs when compiling. |
| 16:24.50 | wiizard | but the EP can also define its own |
| 16:25.06 | [Airwolf] | good evening :) |
| 16:25.24 | [Airwolf] | I have a little question. |
| 16:25.28 | pauldy | Tabo, I'm betting you just need an inbound route |
| 16:25.40 | Ralcala | many on these problems gonna be history if this new module on new kernel works... |
| 16:25.41 | cpatry | faa: like which lib? |
| 16:25.46 | Tabo | hold |
| 16:25.48 | Tabo | let me check |
| 16:26.01 | pauldy | go to did routes |
| 16:26.04 | Ralcala | you now sip,rtp,rtps ports |
| 16:26.10 | pauldy | do you have both of your numbers defined? |
| 16:26.14 | faa_ | cpatry #include <loudmouth/loudmouth.h> |
| 16:26.25 | Ralcala | no vpn, no stun, no ipsec, no iptunneling.... |
| 16:26.33 | [Airwolf] | For every extention i have entry's like this in mijn extentions.conf. It works fine, but I think I'm doing alot of unneeded work. |
| 16:26.35 | pauldy | sip 5060, all transports 10000-20000 |
| 16:26.35 | [Airwolf] | exten => 566,1,Dial(SIP/566,20,r) |
| 16:26.35 | [Airwolf] | exten => 566,2,Voicemail(us566@default) |
| 16:26.35 | [Airwolf] | exten => 566,3,PlayBack(vm-goodbye) |
| 16:26.35 | [Airwolf] | exten => 566,4,HangUp() |
| 16:26.40 | faa_ | cpatry standard.. in makefile i use @$(CC) -fPIC $(CFLAGS) `pkg-config --cflags loudmouth-1.0` -c -o $@ $<, but dont know how to get that same effect in astxs |
| 16:27.04 | cpatry | thats not a lib, thats an header. just add it at the beginning of ur app? |
| 16:27.22 | wiizard | that line there SIP/566,20,r does that tell it to ring for 20 then goto 2? |
| 16:27.38 | [Airwolf] | wiizard, yes |
| 16:27.44 | wiizard | thought so |
| 16:27.45 | cpatry | -append=EXTOBJ:"-lpq" adds libpq if that could help. |
| 16:28.00 | pauldy | Tabo, so whats the verdict |
| 16:28.20 | faa_ | cpatry I did that, |
| 16:28.22 | Tabo | paul... I don't see any inbound routes |
| 16:28.26 | Tabo | in the asterisk@home GUI |
| 16:28.30 | Tabo | I only see outbound routes |
| 16:28.40 | Tabo | I see incoming calls thou, its pointing to digital receptionist |
| 16:28.45 | pauldy | did you click did routes on the lower left |
| 16:28.56 | wiizard | the incoming calls thing is pap |
| 16:29.05 | *** part/#asterisk Ralcala (n=operador@148.223.28.169) |
| 16:29.05 | wiizard | cant change the context from-pstn |
| 16:29.09 | Tabo | both goes directly to voicemail now |
| 16:29.27 | pauldy | Tabo, did you click did routes on the lower left |
| 16:29.33 | Tabo | did routes are set to digital receptionist |
| 16:29.38 | Tabo | so I should be hearing a voice |
| 16:29.44 | Tabo | yup |
| 16:29.52 | pauldy | ok so they are both set |
| 16:29.53 | Uther_P | I was just curious how a server being one nat, and a phone behind another nat would establish the media connections without having some firewall module loaded to monitor it... I think you would have to redirect port 5060 to the server at least, so the phones can register... but it seems it would be best if, after that, when the media path is being established, if the two sides specified the rtp port and ext ip they are going to use, then each side to sen |
| 16:30.07 | pauldy | ok do you have shell access for asterisk |
| 16:30.11 | Tabo | yes |
| 16:30.14 | Tabo | i'll private msg u |
| 16:30.23 | pauldy | ok asterisk -r -x "stop now" |
| 16:30.33 | pauldy | then run asterisk -c -vvvv -dddd |
| 16:30.49 | pauldy | once it comes up dial in and check what you see |
| 16:32.00 | spackle | Tabo: still like your phone? |
| 16:32.01 | faa_ | cpatry thanks, thats it |
| 16:32.16 | cpatry | faa: you owe me 1 beer now :) |
| 16:32.39 | *** join/#asterisk mithro (n=tim@c213-100-42-188.swipnet.se) |
| 16:32.45 | Tabo | spackle I love it, thou I really like the nortels more |
| 16:32.52 | Tabo | but too bad they are analog |
| 16:33.16 | Uther_P | nortel makes analog phones? |
| 16:33.20 | Uther_P | heh |
| 16:33.24 | pauldy | hehe |
| 16:33.24 | Tabo | paul: dang doesn't work |
| 16:33.26 | Uther_P | never seen one |
| 16:33.35 | *** join/#asterisk sigmounte (n=sigmount@www.sighq.net) |
| 16:33.53 | pauldy | ? when you say it doesn't work it doesn't give me a very concise view of what you mean |
| 16:33.55 | Tabo | sorry |
| 16:33.58 | Tabo | I meant Vista 350 |
| 16:34.25 | Tabo | paul: when I call it, it goes straight to voicemail |
| 16:34.29 | Tabo | but numbers I have |
| 16:34.30 | Tabo | both |
| 16:34.47 | pauldy | what do you see in the console? |
| 16:34.48 | Tabo | I got 1 zap channel |
| 16:35.03 | pauldy | anything interesting |
| 16:35.29 | pauldy | if you want to send it all use http://paste.linux-noob.com/ |
| 16:35.30 | *** join/#asterisk PiruchoOSX (n=mundurra@adsl-69-233-188-132.dsl.pltn13.pacbell.net) |
| 16:35.50 | Tabo | what is that |
| 16:36.01 | pauldy | a terrorist tool for communications |
| 16:36.08 | pauldy | sssh dont tell |
| 16:36.16 | Tabo | they should make a webpage |
| 16:36.21 | Tabo | that translates the asterisk coding |
| 16:36.28 | Tabo | asterisk debug |
| 16:36.29 | pauldy | just keeps you from flooding the channel when you have a lot of lines to share |
| 16:36.41 | Tabo | and explains what it it means |
| 16:36.44 | Tabo | u think that is possible? |
| 16:36.48 | pauldy | paste it on that page then send back the link |
| 16:36.49 | Tabo | I can help out on the web design |
| 16:37.48 | cpatry | tabo: that would be cool, but really low priority :) |
| 16:37.50 | pauldy | not very practical right now |
| 16:38.18 | pauldy | by the time you got it implimented it woul dbe out of date as 1.2 would be here |
| 16:38.25 | Tabo | why does the debug display every 5 seconds |
| 16:38.39 | Tabo | does this take alot of bandwidth |
| 16:38.42 | pauldy | cause you have sip debug set |
| 16:38.55 | pauldy | it is just text |
| 16:38.55 | MikeJ[Laptop] | Tabo, then turn off debug! |
| 16:38.57 | Tabo | i know but if i sip no debug it |
| 16:39.13 | Tabo | i'm sure the asterisk box connects to my provider every 5 secs I think |
| 16:39.22 | pauldy | it may be |
| 16:39.34 | pauldy | copy everything you see in the debug output right now |
| 16:39.39 | Tabo | ok hold |
| 16:39.44 | pauldy | paste it into that link I gave you and send me the url |
| 16:39.55 | cpatry | sip debug isnt just connection, its all patckets |
| 16:40.18 | MikeJ[Laptop] | oohh.. and paste me the ones with the passwords in it.. I have some calls to make tonight ;) |
| 16:40.24 | *** join/#asterisk myiagy (n=myiagy@200.138.215.78) |
| 16:40.38 | *** join/#asterisk Kai (n=jmwoodga@45.e6.d12c.cidr.airmail.net) |
| 16:41.02 | pauldy | does it show the md5pass hash |
| 16:41.23 | *** join/#asterisk Morakai (n=jmwoodga@45.e6.d12c.cidr.airmail.net) |
| 16:42.04 | Tabo | http://paste.linux-noob.com/index.php?query=665 |
| 16:42.07 | *** join/#asterisk marv[work] (n=timr@border0hsv.asterisksgi.com) |
| 16:42.10 | Tabo | Paul |
| 16:42.11 | Tabo | thats the link |
| 16:42.31 | Tabo | I see my workplace's caller id name too |
| 16:42.52 | pauldy | SIP/2.0 403 Forbidden |
| 16:43.21 | Tabo | there pw incorrect? |
| 16:44.11 | pauldy | I don't know what I do know is that your not registering to broadvoice |
| 16:44.21 | *** join/#asterisk myiagy (n=myiagy@200.138.215.78) |
| 16:44.35 | pauldy | you should be able to go into your account portal and get the pass straight from there and paste it into your config |
| 16:44.41 | *** part/#asterisk gr0mit (n=w10277@fw.mot-tools.co.uk) |
| 16:45.19 | *** join/#asterisk apardo (n=apardo@90.Red-83-46-197.pooles.rima-tde.net) |
| 16:45.38 | Tabo | which file? |
| 16:45.40 | Tabo | sip.conf? |
| 16:46.04 | pauldy | should be in amp portal trunks and the sip trunk you defined |
| 16:46.37 | pauldy | is this a new account like created in the past 24 hours? |
| 16:47.13 | Tabo | paul its right |
| 16:47.15 | Tabo | I see it |
| 16:47.23 | Tabo | in peer details |
| 16:47.26 | Tabo | secret=xxxx |
| 16:47.32 | Uther_P | you are calling *into* your asterisk box? |
| 16:47.39 | Tabo | yes Uther |
| 16:47.41 | Uther_P | from an outside phone? |
| 16:47.41 | pauldy | he isn't even registering right now |
| 16:47.45 | Tabo | exactyl |
| 16:47.59 | Tabo | context=from-pstn |
| 16:48.01 | Tabo | is that wrong? |
| 16:48.04 | Tabo | its voip |
| 16:48.15 | Uther_P | it looks like the asterisk box is saying that the extension isn't found |
| 16:48.19 | pauldy | that shoudl be fine |
| 16:48.20 | faa_ | cpatry ? |
| 16:48.25 | *** join/#asterisk fugitivo (n=ajf@201.255.102.212) |
| 16:48.25 | Tabo | uther its going to digital receptionist |
| 16:48.27 | fugitivo | hello |
| 16:48.55 | Uther_P | Tabo: digital receptionist? |
| 16:49.00 | Tabo | yup |
| 16:49.02 | Tabo | it should go there |
| 16:49.04 | Tabo | than ask for extensions |
| 16:49.05 | BladeRunner99 | Do u know a place to record freely a voice ? |
| 16:49.06 | pauldy | it would be easier to read if he had run asterisk -c -vvvv -dddd |
| 16:49.26 | cpatry | BladeRunner99: show application record :) |
| 16:49.44 | Uther_P | by digital receptionist, you mean an IVR? |
| 16:49.48 | BladeRunner99 | I need a robotly voice..... |
| 16:49.55 | fugitivo | BladeRunner99: try www.cepstral.com |
| 16:49.59 | Uther_P | ivr running on asterisk or somewhere else? |
| 16:50.03 | fugitivo | BladeRunner99: they have online demos with effects |
| 16:50.43 | *** join/#asterisk nrg8 (n=bclark@64.171.107.5) |
| 16:50.53 | *** join/#asterisk generalhan (i=general_@63.133.146.82) |
| 16:51.11 | pauldy | <PROTECTED> |
| 16:51.50 | Tabo | pauldy: the incoming should go straight to ivr |
| 16:51.54 | wiizard | what would cause my AMP to drop off from my proxy |
| 16:51.57 | Tabo | before allow extensions |
| 16:52.21 | *** join/#asterisk thal (n=thalunil@walledcity.de) |
| 16:52.30 | pauldy | Tabo, I understand what youa re trying to do but what I saw was a registration that was forbiden that must be rectified first |
| 16:52.33 | BladeRunner99 | fugitivo: I try but the italian voice is not nice.... other suggest ? |
| 16:52.48 | pauldy | otherwise the only way you will ge tto that tree is to get your phone and dial 7777 |
| 16:53.10 | gaggaman | is there any docu about the sccp / skinny protocoll on the net? I'd like to play with the cisco 7970 video functions. |
| 16:54.20 | brettnem | ugh.. asterisk voicemail sucks |
| 16:54.39 | Hmmhesays | its a decent voicemail system |
| 16:54.47 | brettnem | blah |
| 16:54.57 | file | hot-n-sexy Hmmhesays!!! |
| 16:54.57 | bkw_ | grrr |
| 16:55.13 | bkw_ | spandsp is leaking memory like mad |
| 16:55.22 | Hmmhesays | the way I figure it.... it works, its free |
| 16:55.24 | brettnem | bkw_: sometimes I do that too |
| 16:55.25 | Hmmhesays | hey file howsit? |
| 16:55.26 | file | gaggaman: it's proprietary so there's not an awful lot |
| 16:55.32 | file | Hmmhesays: good good, you? |
| 16:55.40 | generalhan | hey brettnem you got a few minutes to try and help me out ? |
| 16:55.53 | Hmmhesays | t'ain't bad |
| 16:55.58 | brettnem | ugh.. not really |
| 16:56.02 | generalhan | crap |
| 16:56.06 | brettnem | I just wanted to complain for a moment |
| 16:56.17 | Hmmhesays | this is irc, do it somewhere else |
| 16:56.24 | brettnem | go read the wiki |
| 16:56.25 | generalhan | ok, well let me know if you do im having some issues with my 7960. i have it flashed to SIP but it wont connect to the asterisk server |
| 16:56.31 | Hmmhesays | lol j/k |
| 16:56.32 | gaggaman | file yes i know it's proprietary. |
| 16:56.36 | Tabo | paul 7777 doesn't owrk |
| 16:56.40 | brettnem | ;) |
| 16:56.44 | gaggaman | that's the problem :-) |
| 16:56.47 | Tabo | nor any of the *98 etc.. |
| 16:57.06 | fugitivo | i'm looking for a voip provider with low rates to spain |
| 16:57.08 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
| 16:57.11 | fugitivo | anyone? |
| 16:57.13 | brettnem | generalhan: do a tethereal trace and see if it's even trying |
| 16:57.16 | *** join/#asterisk Borgon (n=L3orgon@70-100-53-251.dsl1.tbr.ga.frontiernet.net) |
| 16:57.58 | Borgon | How can i apply the RPID patches to my current install? http://bugs.digium.com/view.php?id=2471 |
| 16:58.19 | cpatry | Borgon: patch -p0 < file.diff |
| 16:58.22 | bkw_ | yep |
| 16:58.24 | *** join/#asterisk Samoied (n=Samoied@popeye.opens.com.br) |
| 16:58.59 | Borgon | thanks |
| 16:59.28 | *** join/#asterisk jeffgus (n=jeffgus@2002:d856:c704:0:0:0:0:1) |
| 17:00.00 | *** join/#asterisk [hC] (n=hardcore@8.10.2.40) |
| 17:03.22 | *** join/#asterisk mAsH` (n=mash@ppp-217-133-150-46.cust-adsl.tiscali.it) |
| 17:03.27 | *** join/#asterisk brc_ (n=brc@pdpc/supporter/basic/brc) |
| 17:03.47 | mAsH` | hi all |
| 17:03.56 | mAsH` | anyone use * on VIA C3 ? |
| 17:03.57 | Tabo | I'm from canada, can I use broadvoice services? |
| 17:04.24 | CoffeeIV | I have set up a ring group, but when I call it, it doesn't ring the extension -- dialparties.agi returns 0 (or something) in the macro-rg-group and it falls out to the default of the group (voicemail). Does anyone know why this might happen ? |
| 17:04.28 | mutilator | you could use it from mars if you had an internet connection there |
| 17:05.09 | generalhan | brettnem: i did a sip debgug on that line and its trying to connect and i keep getting the same error |
| 17:05.35 | *** join/#asterisk Assid (n=assid@203.115.64.59) |
| 17:06.43 | *** join/#asterisk Samoied (n=Samoied@popeye.opens.com.br) |
| 17:08.05 | mAsH` | anyone use * on VIA C3 processor ? |
| 17:08.34 | *** join/#asterisk tangel (i=tangel@208.239.77.234) |
| 17:09.25 | JerJer | mAsH`: sure |
| 17:09.27 | *** join/#asterisk __unnamed__ (n=pape@modemcable224.217-130-66.mc.videotron.ca) |
| 17:09.34 | JerJer | just don't expect much |
| 17:11.46 | Tabo | is broadvoice host sip.broadvoice.com 147.135.20.128? |
| 17:12.10 | CoffeeIV | What is the purpose of dialparties.agi, and why does macro-rg-group use it but no other part of my * configs use it ? Is it supposed to dial multiple extensions at the same time in some kind of fancy way ? |
| 17:12.33 | Borgon | cpatry: download the 3 .cs to /usr/src/asterisk , then do patch -p0 <file.diff right? |
| 17:15.54 | malverian[work] | Sep 1 13:14:44 ERROR[7612]: chan_zap.c:6666 mkintf: Unable to open channel 26: No such device |
| 17:15.54 | malverian[work] | here = 0, tmp->channel = 26, channel = 26 |
| 17:16.05 | malverian[work] | Yet "/dev/zap/26" exists and is readable/writable... any ideas? |
| 17:16.46 | *** join/#asterisk SplasPood (i=jwb@ludicrous.paravolve.net) |
| 17:16.56 | eKo1 | what does zttool say? |
| 17:16.59 | malverian[work] | /proc/zaptel/2 shows the card usable with those channels. |
| 17:17.05 | malverian[work] | eKo1, zttool says "OK" |
| 17:17.18 | malverian[work] | channel 25 is fxs and 26-28 are fxo |
| 17:17.55 | *** join/#asterisk |cleric| (n=dacleric@p54828807.dip0.t-ipconnect.de) |
| 17:19.55 | *** join/#asterisk Sedorox (i=brandon@smartserv/cna/Sedorox) |
| 17:19.59 | adrenaline | What port does asterisk use? |
| 17:20.11 | Katty | 5060 |
| 17:20.39 | eKo1 | asterisk uses many ports. |
| 17:20.56 | mutilator | or none |
| 17:21.00 | mutilator | depends on what you're doing |
| 17:21.02 | mutilator | :O |
| 17:21.09 | Hmmhesays | sip default is 5060 |
| 17:22.17 | *** join/#asterisk drumkilla_laptop (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 17:22.17 | *** mode/#asterisk [+o drumkilla_laptop] by ChanServ |
| 17:22.41 | adrenaline | Katty, should that show up in netstat? |
| 17:22.59 | malverian[work] | eKo1, Any idea? |
| 17:23.08 | Hmmhesays | sweet |
| 17:23.12 | Hmmhesays | oh crap |
| 17:23.16 | adrenaline | haha |
| 17:23.19 | Hmmhesays | i thought you meant some real adrenaline |
| 17:23.30 | Katty | Hmmhesays: i can't fit all teh gear in the trunk, sorry |
| 17:23.31 | Hmmhesays | it was 50 cent tap night last night i'm dragging ass right now |
| 17:23.33 | adrenaline | I am the next best thing |
| 17:23.42 | eKo1 | malverian[work]: try rebooting the machine. |
| 17:24.09 | [hC] | anyone here applied multiple digium g729 licenses to one * box? |
| 17:24.12 | Hmmhesays | what do you need adrenaline |
| 17:24.31 | adrenaline | ok my question is if asterisk is running should i see ports open in netstat -pant |
| 17:24.33 | [hC] | I applied one 4 channel license, and one 1 channel license. for some reason the 5th call is always dead air like the license isnt there, even though asterisk says it has a total of 5 |
| 17:24.38 | *** join/#asterisk cool4ever2 (n=craeck@mail.innovate-it.ch) |
| 17:25.33 | Hmmhesays | no |
| 17:25.42 | adrenaline | good thanks |
| 17:25.53 | *** join/#asterisk PupenoL (n=pupeno@200.123.183.89) |
| 17:26.03 | adrenaline | thanks |
| 17:27.36 | cool4ever2 | we are testing beta and we have 3 Problems. 1. restrictcid does not work anymore / 2. cdrs contain sometimes wrong callerid / 3. call transfer from a pstn call between 2 snoms is beeing terminated after 180 seconds. Has anyone similair problems? |
| 17:28.41 | Katty | adrenaline: (= |
| 17:32.14 | BrianR___ | Is there any way to send a flash on a zap channel with Dial()? |
| 17:33.12 | *** join/#asterisk odie_flocon (n=Odie@ptr-64-201-182-211.ptr.terago.ca) |
| 17:34.03 | Uther_P | BrianR___: you mean... as part of the dial string? |
| 17:34.33 | BrianR___ | Uther_P: yes. |
| 17:34.38 | *** join/#asterisk Exstatica (i=exstatic@static-71-116-196-11.lsanca.dsl-w.verizon.net) |
| 17:34.59 | BrianR___ | Essentially I want a certain extension to pick up a zap channel, dial some digits, flash, dial some more digits, and then connect the calling channel to the zap channel. |
| 17:35.27 | Hmmhesays | talkin' on my cisco, on my cisco thats where i'll be just talkin' on my cisco to meeee |
| 17:35.29 | BrianR___ | The Flash() application can only flash the calling channel, which is not useful. |
| 17:36.15 | generalhan | wish i could talk on my cisco |
| 17:36.20 | generalhan | i cant get the freaking thing to work |
| 17:36.21 | generalhan | lol |
| 17:36.27 | Uther_P | BrianR___: I don't think so... but I used to use asterisk as a voicemail server for an external pbx via analog lines that required hook flashes and such in order to make it xfer and such |
| 17:36.59 | BrianR___ | Uther_P: I've been able to do it with .call files, but... |
| 17:37.05 | gaggaman | general whats the prob w th cisco? |
| 17:37.11 | generalhan | who knows |
| 17:37.12 | generalhan | lol |
| 17:37.20 | generalhan | it wont register |
| 17:37.28 | generalhan | <PROTECTED> |
| 17:37.55 | generalhan | and i can log in with that name and pass with my X-Lite phone so i have no clue whats going on |
| 17:38.28 | Uther_P | BrianR___: yea, I was xfering the same channels, so I was flashing the calling channel... hrm... the only thing I can think of is to loop the call back around as the incomming, then flash it... but that probably wouldn't do you ay good |
| 17:38.52 | Uther_P | whacha needing to do by a hook flash? |
| 17:39.09 | BrianR___ | Uther_P: paging on my norstar requires me to dial [FLASH] *60 1 1 |
| 17:39.14 | gaggaman | oh, SIP. I don't know to much about the SIP fw on ciscos. Anyway, I had a 7960 SIP which reg'd without a problem |
| 17:39.36 | generalhan | well i wish mine would causea im going crazy |
| 17:39.42 | BrianR___ | I have one zap channel on the asterisk that's connected to the norstar, I want to make an extension that does a page on the norstar... |
| 17:40.01 | shido6 | its just the sip.conf , man |
| 17:40.16 | generalhan | i know you keep saying that, but i cant see one thing wrong with it |
| 17:40.23 | gaggaman | did you try with sip debug in * ? |
| 17:40.27 | generalhan | yep |
| 17:40.42 | generalhan | i get PAGES of stuff thats all greek to me |
| 17:41.50 | Uther_P | BrianR___: you have to pick up the line, then hook, then dial *60 1 1? |
| 17:41.57 | BrianR___ | Uther_P: yes. |
| 17:42.19 | generalhan | shido6: i just dont understand why i can register with that user/pass with X-Lite but not with the phone |
| 17:42.25 | Uther_P | BrianR___: strange... agi is the only thing I can thin of :/ |
| 17:42.30 | Morakai | is it a good idea to leave the debugging on? |
| 17:42.58 | BrianR___ | Uther_P: There's another place I do it to light voicemail MWI - I use a .call file |
| 17:43.14 | shido6 | 2 different things |
| 17:43.28 | BrianR___ | I suppose I could do it that way. Then the zap channel would be the calling channel and I could use the flash() application. |
| 17:43.30 | Morakai | generalhan, have you set up the registration with the sip phone/box? |
| 17:43.44 | generalhan | well it must be REALLY different, cause i can log in with X-Lite and with my Aastra phone. |
| 17:43.53 | generalhan | yea |
| 17:44.00 | *** join/#asterisk Chilango (n=Chilango@dsl-201-129-245-34.prod-infinitum.com.mx) |
| 17:44.02 | generalhan | i did it in the cisco config files AND on the phone itself |
| 17:44.19 | Morakai | can you web browse to the phone? |
| 17:44.43 | generalhan | no |
| 17:44.48 | Morakai | cuz I use sipura boxes, and had this problem, just wondering if it might be a similar problem =) |
| 17:45.04 | Morakai | ah, ok then, sorry =/ |
| 17:45.09 | gaggaman | no, the CIscos dont have a web server |
| 17:45.15 | gaggaman | but a telnet.- |
| 17:45.50 | gaggaman | did u try with a very simple pw? xxx or so? no special chars or anything? |
| 17:46.05 | drray | what firmware version? |
| 17:46.11 | *** part/#asterisk netnameus (n=netnameu@pcp05000344pcs.shrpsr01.tn.comcast.net) |
| 17:46.30 | odie_flocon | exit |
| 17:46.57 | *** join/#asterisk firestrm (n=no@S01060013105c1c69.gv.shawcable.net) |
| 17:47.16 | Uther_P | BrianR___: maybe with the G(context^exten^pri) option in dial... "if the call is answered transer both parties to the specified extension"... you could use the dial to call the norstar and specify G(someexten) so when the norstar picks up it will be xfered to another extension, at that extension it would be the calling line, so you could hook flash it there |
| 17:47.22 | PiruchoOSX | i am playing with FOP for * it is asking me for a sucurity code ?? i already put all the password and nothing happend any help is welcome |
| 17:48.00 | firestrm | iax2 should traverse NAT without port forwarding right?? |
| 17:48.52 | PiruchoOSX | even their web has that do i need to pay for it? |
| 17:49.28 | Uther_P | BrianR___: or you may be able to simulate a hook flash if you specify the L options in dial to limit the call to so many miliseconds, and the g options to continue in context when the dial app finishes.. then right after, pickup the line again |
| 17:49.48 | CoffeeIV | what possible reasons might cause dialparties.agi to return no extensions to dial in my ringgroups ? |
| 17:50.40 | *** join/#asterisk mhnoyes (n=mhnoyes@user-38lc0nh.dialup.mindspring.com) |
| 17:50.42 | *** part/#asterisk tsume (n=tsume@unaffiliated/tsume) |
| 17:51.22 | *** part/#asterisk Ferrari_ (n=Ferrari@66.64.128.142.nw.nuvox.net) |
| 17:52.37 | generalhan | yea i made it WAY simple ... the username/pass are the same 5 chars nice and simple |
| 17:52.54 | generalhan | and im using firmware 07-3-00 |
| 17:52.55 | Chilango | hello? |
| 17:53.36 | Chilango | Sorry people. I am kind new to the PC-PBX scene and have a basic question here. |
| 17:53.50 | *** join/#asterisk podzap (n=podzap@roswell.pp.saunalahti.fi) |
| 17:54.02 | Uther_P | then you probably want... |
| 17:54.03 | Uther_P | ~doc |
| 17:54.05 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk |
| 17:54.08 | Uther_P | :P |
| 17:54.51 | Chilango | I have seen the cards to connect a PC to a T1 or E1, but the conector that most of the cards shown is something similar to an RJ-45, when I would be expecting more like a BNC or SMC conector. |
| 17:55.03 | *** join/#asterisk toddf (n=toddf@adsl-65-70-118-15.dsl.okcyok.swbell.net) |
| 17:55.43 | Uther_P | Chilango: the t1 has to go through a t1 circuit first |
| 17:55.48 | firestrm | im trying to get an iax connection working and ive got 1 way audio... nat problem? do i have to port forward? |
| 17:56.09 | *** join/#asterisk meppl (n=mephisto@p54AACAC5.dip.t-dialin.net) |
| 17:57.28 | Uther_P | firestrm: is the phone/server defined as being behind a nat? |
| 17:57.39 | Chilango | Uther, how do you do that? In the past I usually just hook my PBX to the point using, well, coaxial cable. |
| 17:57.53 | Hmmhesays | hmm these guys want me to run asterisk as non root |
| 17:57.56 | Hmmhesays | anyone else doing that? |
| 17:58.07 | Uther_P | Chilango I've never seen a t1 conencted via a bnc |
| 17:58.19 | firestrm | yes, i had it working before, but i moved, and ip address changed.. but i dont think that is it.. also changed the router, and i may have ad a port forwarding entry in the onld one but i cant remember |
| 17:58.45 | malverian[work] | Helllllp :-P |
| 17:58.49 | ender | how can I completely disable musiconhold ? |
| 17:59.01 | malverian[work] | I'm trying to put an FXO/FXS card in the same computer I have a T1 card in. |
| 17:59.04 | Ariel_ | ender, remove mpg123 |
| 17:59.15 | malverian[work] | Sep 1 13:48:20 WARNING[7956]: chan_zap.c:894 zt_open: Unable to specify channel 26: No such device |
| 17:59.15 | malverian[work] | Sep 1 13:48:20 ERROR[7956]: chan_zap.c:6666 mkintf: Unable to open channel 26: No such device |
| 17:59.26 | Ariel_ | driver not loaded |
| 17:59.48 | firestrm | malverian[work], sounds like zaptel.conf problem |
| 17:59.50 | malverian[work] | It's loaded.. the device exists... |
| 17:59.54 | Rowter | to make a group of channels for dialing is in /etc/asterisk/zapata.conf group=1 channels=> 1-12? |
| 18:00.01 | Uther_P | Chilango: usually your t1 provider bring the line into a t1 circuit, and out of that comes a cat5 via an rj45 connector |
| 18:00.03 | malverian[work] | firestrm, hmm.. |
| 18:00.13 | PupenoL | Is anybody mantaining a darcs version of Asterisk's repository ? |
| 18:00.20 | malverian[work] | pbx ~ # cat /proc/zaptel/2 |
| 18:00.20 | malverian[work] | Span 2: WCTDM/0 "Wildcard TDM400P REV I Board 1" |
| 18:00.20 | malverian[work] | <PROTECTED> |
| 18:00.20 | malverian[work] | <PROTECTED> |
| 18:00.21 | Chilango | Uther Ah, that is the way phone people deliver t1/e1 in Mexico. They give you a port which consist in a pair of BNC or SMC conectors. That is why I am so surprised to see the connectors on, say, the digium cards. |
| 18:00.27 | ender | Ariel_: ah ok. |
| 18:00.32 | malverian[work] | firestrm, See above. |
| 18:00.47 | ender | anybody familiar w/ the IP501s and the Contact Directory stuff? |
| 18:01.13 | Hmmhesays | I have 4 days to learn everything I possibly can about SER |
| 18:01.15 | Hmmhesays | fsck |
| 18:02.05 | Uther_P | Chilango: a pair of bnc... just a guess, but probably each bnc is the equiv. of the send and recv pair... but I dunno... maybe ask the telco? |
| 18:02.47 | *** join/#asterisk pbxbart__ (n=pbxbart@p54B010C2.dip0.t-ipconnect.de) |
| 18:02.51 | malverian[work] | firestrm, It looks that zaptel is configured correctly, no? Maybe my zapata.conf is screwed? /dev/zap/26 exists. |
| 18:02.58 | pbxbart__ | hi |
| 18:03.15 | *** join/#asterisk trig (n=jb@xob.neospire.net) |
| 18:03.35 | pbxbart__ | can anyone explain me or point me to re right information in the qwiki what exact DIALSTATUS=NOWANSER mean? |
| 18:04.02 | Hmmhesays | banana phone - ring ring, ring ring, ring ring, ring ring, timeout |
| 18:04.58 | Hmmhesays | as opposed to banana phone - ring ring, ring ring, ring ring, ring ring, "HEWWWOOOO chinese take out, we serve you rearl fast" |
| 18:04.58 | Assid | how do you autologin to voicemailmain ? |
| 18:04.59 | *** join/#asterisk [1]JohnJacob (n=JohnJaco@pcp0011543387pcs.mainf01.in.comcast.net) |
| 18:05.11 | Assid | or just ask for the password |
| 18:05.24 | Assid | and use the mailbox number its calling from |
| 18:05.24 | Chilango | Uther, I have done, they were as mystified as myself. :) . And certainly, each conector is either the send or the receive. I been looking for impedance converters (for I am almost certain impedance is different in eahc medium) or media converters with no luck. |
| 18:05.37 | Hmmhesays | voicemailmain(<my mailbox number>) |
| 18:05.48 | *** part/#asterisk FuzzyCat (n=ScaredyC@84.119.131.232) |
| 18:05.51 | firestrm | malverian[work], i looks like zapata.conf is configured, but zaptel.conf may be a different story.. |
| 18:05.55 | Uther_P | Hmmhesays: heh dude, what the hell are you talking about |
| 18:06.12 | malverian[work] | firestrm, Wouldn't my card not show configured properly in /proc/zaptel/2 if zaptel.conf was incorrect? |
| 18:06.15 | Hmmhesays | Uther_P, with the banana phone stuff? |
| 18:06.18 | Uther_P | yea |
| 18:06.32 | pbxbart__ | sorry.. NOANSWER is what i am talking about |
| 18:06.48 | Hmmhesays | I was explaining what DIALSTATUS=NOANSWER is |
| 18:06.51 | malverian[work] | firestrm, Here is my zaptel: http://pastebin.ca/21861 |
| 18:06.57 | Uther_P | oh, heheh |
| 18:07.04 | firestrm | malverian[work], yes, but you may not have the channels defined properly. it sees the car, but does it know what to do with it. |
| 18:07.09 | malverian[work] | firestrm, I have 1 te110p card, and one tdm31b |
| 18:07.23 | malverian[work] | firestrm, Loaded modules in that order. |
| 18:07.25 | Ariel_ | pbxbart__, just means your device timeout and did not pickup in the allot time. |
| 18:07.35 | *** join/#asterisk SpamapS (n=clint@c-064-186-245-122.sd2.redwire.net) |
| 18:07.35 | pbxbart__ | @hmmhesays: but i i recive nowanswer right after the begin of call |
| 18:07.44 | Hmmhesays | odd |
| 18:07.57 | Ariel_ | pbxbart__, between what devices/ |
| 18:08.11 | *** part/#asterisk pbxbart__ (n=pbxbart@p54B010C2.dip0.t-ipconnect.de) |
| 18:08.11 | Netgeeks_ | twisted & drumkilla either or both of you around? |
| 18:08.17 | Hmmhesays | why do people where baseball caps crooked on their head |
| 18:08.19 | Assid | Hmmhesays: use callerid ? |
| 18:08.26 | *** join/#asterisk pbxbart__ (n=pbxbart@p54B010C2.dip0.t-ipconnect.de) |
| 18:08.35 | Uther_P | pbxbart__: you aren't by chance using the L option in the dial command? |
| 18:08.35 | Ariel_ | Hmmhesays, it's a style |
| 18:08.42 | Hmmhesays | Assid: if ${CALLERIDNUM} matches your voicemailbox |
| 18:08.48 | Hmmhesays | Ariel_ silly kids |
| 18:08.54 | pbxbart__ | sorry.. between me and my provider |
| 18:09.03 | pbxbart__ | no, no L option |
| 18:09.07 | SpamapS | Hrm.. I want to find some sort of vendor that is doing IAX based services in my area.. so I can just give all my people SIP phones and point them at my Asterisk box, but let somebody else deal with PRI lines and stuff... |
| 18:09.13 | Ariel_ | pbxbart__, iax2 or sip? |
| 18:09.15 | Uther_P | Hmmhesays: because they feel that their ear is more important to shield from the sun than their eyes |
| 18:09.20 | pbxbart__ | iax2 |
| 18:09.28 | SpamapS | Anybody know of a directory of such providers? I know they must be out there. |
| 18:09.29 | Hmmhesays | no, not completely sideways |
| 18:09.38 | Hmmhesays | ~docs |
| 18:09.39 | jbot | docs is, like, Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk |
| 18:09.43 | Ariel_ | pbxbart__, do you have in the iax.conf notransfer=yes |
| 18:09.46 | Hmmhesays | voip-info |
| 18:09.56 | pbxbart__ | yes |
| 18:10.16 | gaggaman | is there any docu about the sccp / skinny protocoll on the net? I'd like to play with the cisco 7970 video functions. |
| 18:10.26 | Hmmhesays | is there boobies on the net? |
| 18:10.32 | malverian[work] | firestrm, It's only about 8 lines long, do you see something wrong with it? |
| 18:11.20 | *** join/#asterisk apardo (n=apardo@90.Red-83-46-197.pooles.rima-tde.net) |
| 18:12.19 | Uther_P | Hmmhesays: I don't think people would allow that kind of filth on the internet |
| 18:13.07 | Assid | Hmmhesays: ${CALLERIDNUM} shows my sip name |
| 18:13.57 | *** join/#asterisk _chris (n=chris@12-216-145-145.client.mchsi.com) |
| 18:14.39 | malverian[work] | Why won't this card work? |
| 18:14.51 | _chris | i've been struggling with a newbie problem that is likely common. |
| 18:15.04 | *** join/#asterisk MooingLemur (n=troy@shells200.pinchaser.com) |
| 18:15.20 | _chris | i can place outgoing calls but incoming calls get the "goodbye" message |
| 18:15.34 | *** join/#asterisk roulduke_ (i=oqd8luwz@p508D2DD7.dip0.t-ipconnect.de) |
| 18:15.43 | _chris | any ideas? |
| 18:15.45 | Assid | bbiab |
| 18:16.08 | *** join/#asterisk Ethon (i=arne@Oldman.steinkamm.com) |
| 18:16.33 | *** join/#asterisk Romik (n=romik_@212.143.5.146) |
| 18:17.51 | malverian[work] | Hahahaha |
| 18:17.58 | malverian[work] | I forgot to put in the molex connector to my tdm400 :-P |
| 18:18.03 | malverian[work] | Ugh.. sorry firestrm :) |
| 18:18.22 | drray | it fell out |
| 18:18.24 | drray | :) |
| 18:20.26 | *** join/#asterisk Henguei (n=smoka@196.203.249.98) |
| 18:20.42 | _chris | is the failure to receive incoming calls related to a problem in zapata.conf. Outgoing calls work fine. |
| 18:21.42 | *** part/#asterisk cool4ever2 (n=craeck@mail.innovate-it.ch) |
| 18:22.41 | *** join/#asterisk Rowters (n=SilverDr@dsl-201-129-89-96.prod-infinitum.com.mx) |
| 18:22.51 | Ariel_ | _chris, it could be your context. Incoming calls are from the PSTN? |
| 18:24.18 | podzap | hi, are there any FreeBSD users in the channel??? |
| 18:24.30 | _chris | ariel, i just saw that on a forum post. it was context=incoming |
| 18:24.45 | fugitivo | should i get a xeon for 100 concurrent calls? g729 codec |
| 18:24.45 | _chris | i am going to change to context=from-pstn |
| 18:25.49 | mutilator | is it reencoding at all fugitivo? |
| 18:26.14 | *** join/#asterisk |Vulutre| (n=Vulutre@13.220.204.68.cfl.res.rr.com) |
| 18:26.35 | fugitivo | mutilator: yes |
| 18:26.47 | mutilator | to what |
| 18:26.53 | fugitivo | ulaw |
| 18:27.22 | Ariel_ | _chris, you have amp setup? |
| 18:27.56 | mutilator | yea |
| 18:28.03 | mutilator | dual 2.8's or something |
| 18:28.05 | |Vulutre| | Anyone know of a program for linux that will run as a daemon to allow Windows PCs to backup to it with incremental backups with a retention period, so you could roll back say 10 days? |
| 18:28.18 | fugitivo | thanks |
| 18:28.21 | *** join/#asterisk mastix (n=mastix@ip-85-160-10-94.eurotel.cz) |
| 18:28.25 | mastix | hi |
| 18:28.27 | *** join/#asterisk spackle (n=spackle@209.234.83.19) |
| 18:29.42 | mastix | does anybody know how to send 183 session progress instead of ring when i dial through my external gw? |
| 18:30.21 | Chilango | Vulutre, not sure if is the righ channel to ask, but then you can either do it using scripting in the linux (as long as you share the disk to back up in the Windows PCs) or acquiere some nice software such as winbackup. |
| 18:31.06 | |Vulutre| | Chilango: I will look into that I have been using rsync but it doesn't seem to do everything i need it to |
| 18:31.19 | *** part/#asterisk podzap (n=podzap@roswell.pp.saunalahti.fi) |
| 18:32.29 | *** join/#asterisk djin_ib (n=djin_ib@gridfox.xs4all.nl) |
| 18:32.35 | *** join/#asterisk netnameus (n=netnameu@pcp05000344pcs.shrpsr01.tn.comcast.net) |
| 18:33.07 | netnameus | My polycom phone will not register with asterisk, but I can get my xlite phone to. Are there connection log files I can look at to debug the problem? |
| 18:33.07 | _chris | ariel, changing context=from-pstn caused several stack messages in the console. auto attendent never picked up |
| 18:33.28 | *** join/#asterisk stkn (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 18:33.29 | _chris | ariel, yes I have amp setup |
| 18:35.07 | _chris | Starting Zap/1-1 at incoming,s,1 failed so falling back to exten 's' |
| 18:35.14 | *** join/#asterisk silug (n=steve@206.80.72.34) |
| 18:35.15 | _chris | when context is set to incoming i get: |
| 18:35.56 | *** join/#asterisk jsaunders (i=js@70.70.38.160) |
| 18:39.01 | jsaunders | Haha |
| 18:39.21 | jsaunders | Question... has anyone ever played w/ gnugk virtual queues? |
| 18:39.32 | firestrm | nope |
| 18:39.37 | *** part/#asterisk pbxbart__ (n=pbxbart@p54B010C2.dip0.t-ipconnect.de) |
| 18:39.46 | jsaunders | I know this is #asterisk, no #gnugk chan though so I figured this was my best shot. |
| 18:40.27 | *** join/#asterisk dolson_ (n=dana@toronto-HSE-ppp4305634.sympatico.ca) |
| 18:40.54 | Ariel_ | _chris, you need to setup your incoming setup |
| 18:41.00 | Ariel_ | _chris, via amp |
| 18:41.02 | *** part/#asterisk Nuxi (n=nuxi@cust-sdsl-204-250-82-202.bzn-co-i1000-01.bridgeband.net) |
| 18:41.09 | firestrm | do you need to port forward iax? |
| 18:41.54 | firestrm | anyuone? |
| 18:42.02 | jontow | yes, 4569 |
| 18:42.08 | firestrm | tcp? |
| 18:42.09 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 18:42.09 | *** mode/#asterisk [+o anthm] by ChanServ |
| 18:42.18 | firestrm | or both tcp/udp? |
| 18:42.19 | jontow | oughttadoublecheck |
| 18:42.26 | *** join/#asterisk hadi57 (n=al_moghr@83.136.8.206) |
| 18:42.28 | jontow | i've done both in the past when i was in a hurry |
| 18:43.13 | _chris | ariel, i believe i did set this up in amp under incoming calls. during biz hours it goes to the dig. reception |
| 18:43.18 | firestrm | jontow, i had the connection working great for over a year, then switched routers, and to make matters worse i forgot the password to the old router so i cant even go check there.. |
| 18:43.23 | jontow | your answer: |
| 18:43.25 | jontow | ^:jontow@bsdgate:~% netstat -a | grep 4569 |
| 18:43.25 | jontow | udp 0 0 *.4569 *.* |
| 18:43.27 | jontow | :) |
| 18:43.53 | firestrm | jontow, thanks.. my hero :) |
| 18:43.56 | jontow | np |
| 18:44.27 | jontow | really wish there was a clear path of upgrade for this laptop |
| 18:44.40 | jontow | instead of blow-it-away-and-boot-from-floppy or swap the drive.. |
| 18:48.01 | *** join/#asterisk Bentley (n=Bentley@S01060080c8135e6a.cg.shawcable.net) |
| 18:48.34 | *** join/#asterisk jdbecker1968 (n=jbecker@S0106000f3d016dd2.cg.shawcable.net) |
| 18:49.51 | jdbecker1968 | hello, anyone able to give a hand re: Sangoma configuration |
| 18:50.12 | jdbecker1968 | Sangoma has zero doc re: TDM config of their cards (A102) |
| 18:50.29 | jdbecker1968 | wiki not much help either re: wancfg utility (wanpipe1.conf) |
| 18:50.49 | jdbecker1968 | "wanrouter start" locks the box up solid |
| 18:51.23 | spackle | jdbecker1968: download the sangoma drivers, there is a crapload of info in them about configuring them |
| 18:51.32 | spackle | several readme files. |
| 18:51.53 | jdbecker1968 | there is not a single /etc/wanpipe/samples file re: TDM config |
| 18:51.58 | jdbecker1968 | using latest beta drivers |
| 18:53.26 | spackle | try one prior to the beta? I think most of it is on their site too. Failing that, call them, they seem very friendly. |
| 18:53.34 | jdbecker1968 | with latest beta drivers selected the option presented for TDM install for Asterisk (that option not in stable Setup script) |
| 18:53.59 | jdbecker1968 | have a call into them spackle but looks like Tech Support is a 1-man operation? |
| 18:54.10 | spackle | could be. |
| 18:54.53 | jdbecker1968 | i guess we'll wait for their call, thx anyways spackle |
| 18:55.00 | *** part/#asterisk jdbecker1968 (n=jbecker@S0106000f3d016dd2.cg.shawcable.net) |
| 18:55.03 | spackle | jdbecker1968: The driver package I downloaded from their site - not the beta - for the 102 card, had a lot of text files about asterisk and wanpipe |
| 18:55.11 | spackle | anyways... |
| 18:55.32 | *** join/#asterisk stepcut (n=user@207.67.194.2) |
| 18:56.18 | *** part/#asterisk Bentley (n=Bentley@S01060080c8135e6a.cg.shawcable.net) |
| 18:57.10 | *** join/#asterisk used (n=used@c-24-22-125-179.hsd1.or.comcast.net) |
| 18:57.18 | mastix | hello ...anyone ho knows how to get real tones from PSTN when i dial via external gateway |
| 18:57.26 | mastix | and call is in progress |
| 18:57.46 | *** join/#asterisk brookshire (n=matt@digium.com) |
| 18:57.49 | mastix | i dont want to get ring generated by asterisk |
| 18:59.04 | *** join/#asterisk Bentley (n=Bentley@S01060080c8135e6a.cg.shawcable.net) |
| 18:59.29 | *** part/#asterisk Bentley (n=Bentley@S01060080c8135e6a.cg.shawcable.net) |
| 18:59.32 | lathos42 | oops |
| 18:59.40 | *** join/#asterisk jalsot (n=tamas@abacus.eworldcom.hu) |
| 18:59.46 | mastix | anybody |
| 18:59.52 | mastix | guys |
| 18:59.57 | Chilango | Thanks everyone, see you later. |
| 19:00.04 | *** part/#asterisk Chilango (n=Chilango@dsl-201-129-245-34.prod-infinitum.com.mx) |
| 19:00.27 | mastix | does anyone know wheter it is possible at all |
| 19:00.38 | spackle | mastix: you want inline dtmf - what kind of phone are you using |
| 19:01.01 | mastix | spackle: i dont want to have inline dtmf |
| 19:01.03 | bjohnson | mastix: don't use the r option in your dial string |
| 19:01.20 | mastix | bjohnson :yes |
| 19:01.25 | bjohnson | no |
| 19:01.26 | mastix | what should be there |
| 19:01.30 | bjohnson | no 'r' |
| 19:01.37 | mastix | for seesion progress |
| 19:01.37 | spackle | ;-) |
| 19:01.40 | *** join/#asterisk bipolar (n=bipolar@146.145.26.90) |
| 19:01.54 | bjohnson | learn what the option do, and use the ones you want |
| 19:02.05 | bjohnson | or start with no options and go from there |
| 19:02.10 | mastix | ok |
| 19:02.22 | *** join/#asterisk jdbecker1968 (n=jbecker@S0106000f3d016dd2.cg.shawcable.net) |
| 19:02.22 | mastix | i just wanted to know if there is possibility |
| 19:02.33 | *** join/#asterisk Bentley (n=Bentley@S01060080c8135e6a.cg.shawcable.net) |
| 19:02.36 | mastix | thx |
| 19:02.38 | bjohnson | there is. most people use it |
| 19:02.48 | bipolar | Does anyone know if a PCTel chipset softmodem can do full duplex? I've been googleing for a while and found nothing. It's the only hardware I have in my parts bin. |
| 19:02.51 | *** join/#asterisk [1]JohnJacob (n=JohnJaco@pcp0011543387pcs.mainf01.in.comcast.net) |
| 19:03.10 | jdbecker1968 | spackle: i see that wanpipe-beta13m-2.3.3.tgz has a sample file (wanpip1.tdmv_a104) |
| 19:03.12 | jdbecker1968 | thx! |
| 19:03.16 | mastix | is it also possible to transfers progress tones via digium pri boards? |
| 19:03.43 | distortio | progress tones? |
| 19:04.09 | mastix | tones which comes from PSTN when call is in progress |
| 19:04.31 | distortio | thats done in signaling |
| 19:04.38 | *** part/#asterisk jdbecker1968 (n=jbecker@S0106000f3d016dd2.cg.shawcable.net) |
| 19:05.02 | distortio | with pri its a specific message called progress indicator |
| 19:05.58 | mastix | i see |
| 19:06.25 | *** part/#asterisk Bentley (n=Bentley@S01060080c8135e6a.cg.shawcable.net) |
| 19:06.37 | distortio | with h323 its mapped and transmitted over signaling, in h245 with sip its also transmitted in signaling |
| 19:06.44 | distortio | but its not tones |
| 19:06.57 | mastix | it isnt true |
| 19:07.03 | mastix | when you dial to PSTN |
| 19:07.25 | mastix | you get message of course that remote phone is ringing |
| 19:07.45 | distortio | thats different, thats an inband ringtone |
| 19:07.45 | mastix | but in phone you can hear ring tone which should be generated in PSTN |
| 19:07.59 | mastix | that is what i ask about :-) |
| 19:09.02 | mastix | my question is wheter i can pass this tone through PRIcard when i dial to PSTN |
| 19:09.29 | distortio | the answer is yes, although i dont know the config specifics |
| 19:09.40 | mastix | it is ok |
| 19:09.51 | *** join/#asterisk apardo (n=apardo@23.Red-83-44-179.pooles.rima-tde.net) |
| 19:10.18 | mastix | i ask only because someone told me that it is possible just with cards from aculab because of DSPs |
| 19:10.54 | *** join/#asterisk netnameus (n=netnameu@pcp05000344pcs.shrpsr01.tn.comcast.net) |
| 19:11.36 | mastix | so thank you |
| 19:12.08 | mastix | i have more question for all you guys :-) |
| 19:12.41 | mastix | can you tell me how many guys here is selling asterisk solutions? |
| 19:13.41 | adrenaline | I am working on it for small businesses like mortgage companies |
| 19:13.48 | *** part/#asterisk greekman (n=alex@host254.209.113.199.conversent.net) |
| 19:14.23 | mastix | ok thx ...where are you from |
| 19:14.26 | adrenaline | I a actually selling prebuilt boxes but we are testing building our own solutions |
| 19:14.30 | adrenaline | WA |
| 19:14.41 | *** join/#asterisk JohnJacob (n=JohnJaco@pcp0011543387pcs.mainf01.in.comcast.net) |
| 19:14.43 | mastix | WA ??? sorry :-) |
| 19:14.55 | adrenaline | Washington State USA |
| 19:15.02 | mastix | i thought |
| 19:15.08 | mastix | :-) |
| 19:15.39 | mastix | i thing that CZ also doesnt tell you so much :-) |
| 19:16.00 | mastix | anyway how many units can you sell per month |
| 19:16.15 | mastix | i am just wondering what is the penetration of asterisks |
| 19:16.27 | mastix | or what could be :-) |
| 19:17.21 | *** join/#asterisk Defraz (n=t0tal@tim.ibccom.net) |
| 19:20.44 | adrenaline | We just started but we have sold 4 so far in a month time |
| 19:20.56 | cypromis | cz?? |
| 19:21.03 | cypromis | hmm just one border away |
| 19:21.23 | mastix | :-) you are from? |
| 19:21.31 | Hmmhesays | monkkey business by skid row is such a great song |
| 19:21.39 | cypromis | pl |
| 19:22.05 | Hmmhesays | t'ain't bad either |
| 19:22.55 | mastix | so hi brother :-) |
| 19:23.34 | Ariel_ | adrenaline, how are you advertising them? |
| 19:27.55 | adrenaline | We are a Network Consulting Business |
| 19:28.13 | Ariel_ | adrenaline, so your selling them to your exciting base of customers |
| 19:28.19 | *** join/#asterisk cool4ever2 (n=craeck@80-218-106-233.dclient.hispeed.ch) |
| 19:28.37 | adrenaline | Yes and offering them to new clients as well |
| 19:29.09 | Ariel_ | adrenaline, so you have a sales department that is handling that part. |
| 19:29.30 | adrenaline | somewhat yes |
| 19:29.47 | Ariel_ | adrenaline, where in Washington state are you located? |
| 19:30.12 | adrenaline | about 40 north of seattle |
| 19:30.28 | adrenaline | miles |
| 19:31.13 | malverian[work] | Any idea why when I try to modprobe wctdm, it says "can't find module wcfxs" ? |
| 19:31.30 | cpatry | cause wctdm isnt made? |
| 19:31.59 | malverian[work] | ....? |
| 19:32.07 | malverian[work] | No.. wctdm.ko exists. |
| 19:32.18 | malverian[work] | But in cvs there is no wcfxs to build. |
| 19:32.33 | *** part/#asterisk Beave (n=beave@vistech.org) |
| 19:32.35 | *** join/#asterisk Tabo (n=acura0@s142-179-67-118.bc.hsia.telus.net) |
| 19:32.36 | malverian[work] | for some reason if I use modprobe to install wctdm.ko into the kernel I get that error.. but if I use insmod directly it works... |
| 19:33.04 | Tabo | what .conf file is where you enter into your voip companies settings? ie. server address, username, and file |
| 19:33.07 | Tabo | for incoming line |
| 19:33.26 | Ariel_ | Tabo, if your using amp you do it from the trunk menu |
| 19:33.43 | Ariel_ | if not the if there sip on sip.conf if there iax on iax.conf |
| 19:33.50 | Tabo | ariel_: my manager said the source has wrong informatoin |
| 19:34.08 | Ariel_ | Tabo, really |
| 19:34.11 | Tabo | everything is setup properly, the problem is asterisk connectiong to the voip company |
| 19:34.12 | Tabo | yup |
| 19:34.19 | cpatry | wcfxs is deprecated by wctdm iirc. |
| 19:34.22 | ender | any Polycom IP501 users? |
| 19:34.24 | Tabo | do u have fax ariel? |
| 19:34.27 | Tabo | I can fax u the documents |
| 19:34.43 | Ariel_ | 3055740212 |
| 19:34.48 | Tabo | wah |
| 19:34.51 | Tabo | private msg me next time |
| 19:35.00 | *** join/#asterisk santiago (n=santiago@63.245.87.180) |
| 19:35.25 | Tabo | ariel I setup a broadvoice account as well |
| 19:35.38 | Ariel_ | argh I have to reboot this stupid xp pc.... |
| 19:35.43 | adrenaline | ender, I use Polycoms501s |
| 19:36.12 | ender | adrenaline: are you able to put names/numbers into the contacts list on the phone? |
| 19:36.18 | *** join/#asterisk MattB2 (n=MattB2@mail.tricycleinc.com) |
| 19:36.24 | Tabo | arield faxed |
| 19:36.40 | adrenaline | I haven't tried hold on and I will give it a whirl |
| 19:36.58 | MattB2 | hi all |
| 19:37.44 | Tabo | why don't they make a trunks.conf? |
| 19:37.54 | *** join/#asterisk L|NUX (n=linux@202.5.146.13) |
| 19:38.14 | pussfeller | i wanna forward my did to my cricket phone... possible? |
| 19:38.18 | MattB2 | anyone know (and/or recommend) of a VoIP provider in USA that will provide a number (PSTN -> IAX/SIP) at a one-off fee and not charge a monthly fee for outgoing calls? |
| 19:38.34 | *** join/#asterisk used (n=used@c-24-22-125-179.hsd1.or.comcast.net) |
| 19:38.35 | MattB2 | there's hundreds in the UK but i'm struggling to find any in US |
| 19:39.10 | Uther_P | when an ip packet is being sent across a gateway, does the header contain the gateway address somewhere, or is the transport of the packet to the gateway handled completely by the Link Layer frame (e.g. arp address) |
| 19:41.24 | *** join/#asterisk rephorm (n=brian@67.95.13.46) |
| 19:41.46 | rephorm | anyone here ever used a TE110P with a VIA mini-itx board? |
| 19:42.46 | *** join/#asterisk dudes (n=dudes@12-215-34-84.client.mchsi.com) |
| 19:43.30 | ender | adrenaline: any luck? |
| 19:44.24 | *** join/#asterisk snewpy_ (n=markl@203.214.131.5) |
| 19:44.28 | stepcut | MattB2: What about iax.cc ? I think they are canadian, but they offer US numbers -- no monthly/setup fees, 1.7c/min |
| 19:44.49 | adrenaline | I had to put one together |
| 19:44.55 | MattB2 | thanks stepcut, i'll have a look |
| 19:45.02 | ender | oh |
| 19:45.35 | bjohnson | iax.cc isn't canadian |
| 19:45.43 | stepcut | oh wait, I take it back, they do charge per month |
| 19:45.56 | adrenaline | I only have 301 setup right now |
| 19:46.32 | *** join/#asterisk netnameus (n=netnameu@pcp05000344pcs.shrpsr01.tn.comcast.net) |
| 19:46.43 | MattB2 | they do, but it is very cheap |
| 19:46.54 | stepcut | I have an 1800 through iax, which is only 0.10/month (now 0.20 I think), but it is 2.0/min instead of 1.7/c |
| 19:47.20 | netnameus | why would my x-lite be able to connect to my * box, but neither my spa-1001 nor my polycom can |
| 19:47.26 | stepcut | another option is to use ipkall or something for free incoming, and iax (or something) for paid outgoing |
| 19:47.42 | adrenaline | ender what exactly are you trying to do? |
| 19:48.10 | stepcut | netnameus: probably because of something really minor, but nearly impossible to figure out, unfortunately |
| 19:48.31 | *** join/#asterisk Assid (n=assid@203.115.64.59) |
| 19:48.38 | netnameus | stepcut: not the answer i was wanting :D |
| 19:48.50 | netnameus | i've been at this for hours |
| 19:49.35 | MattB2 | stepcut: at http://iax.cc/show.php?go=local, are those prices outgoing or incoming? |
| 19:49.56 | ender | adrenaline: Menu -> Features -> Contact Directory try to add a contact |
| 19:50.15 | stepcut | MattB2: I believe both |
| 19:51.05 | stepcut | MattB2: http://iax.cc/show.php?go=services |
| 19:51.23 | adrenaline | ya I am erroring out too |
| 19:52.11 | ender | adrenaline: damn, ok. Looks like a more global issue. What bootrom and sip ware are you using? |
| 19:52.29 | adrenaline | wait I actually entered a number and it saved |
| 19:52.33 | *** part/#asterisk used (n=used@c-24-22-125-179.hsd1.or.comcast.net) |
| 19:52.38 | ender | adrenaline: I have bootrom 2.6.20032 |
| 19:52.41 | ender | adrenaline: oh?! |
| 19:52.43 | stepcut | MattB2: if you don't care about area code, you can also look into these folks for free incoming: http://www.ipkall.com/ |
| 19:53.28 | adrenaline | I filled in first last and phone number and saved |
| 19:53.41 | adrenaline | contact == phone number |
| 19:54.23 | ender | let me try and do that... |
| 19:54.52 | ender | nope, I still just get 'No records' |
| 19:55.01 | Delta34 | hi all, anybody using 7.5 on cisco phones with *? |
| 19:55.18 | Delta34 | i'm seeing an issue and was wondering if anyone else has the same issue |
| 19:56.23 | adrenaline | my bootrom is 2.6.0003 |
| 19:56.38 | adrenaline | 2.6.1.003 |
| 19:57.03 | adrenaline | 2.6.1.0003 <----that one lol |
| 19:57.07 | Delta34 | i am getting this warning on all sip generated calls from cisco 7.5, WARNING[3924]: Maximum retries exceeded on call |
| 19:57.15 | Delta34 | warning disappears on 7.4 |
| 19:57.17 | *** join/#asterisk drumkilla_laptop (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 19:57.17 | *** mode/#asterisk [+o drumkilla_laptop] by ChanServ |
| 19:57.18 | ender | adrenaline: ah, you have th eolder firmware. |
| 19:57.22 | MattB2 | thx again stepcut |
| 19:57.24 | ender | adrenaline: it may be an issue w/ the new firmware. |
| 19:57.46 | MattB2 | in a perfect world i'd have a local number (don't mind paying one-off or even yearly), free incoming to IAX or SIP and outgoing at < 2c per minute |
| 19:58.27 | shido6 | how do you make a folder readable and writeable by ALL |
| 19:58.30 | shido6 | a+rw |
| 19:58.31 | shido6 | ? |
| 19:58.48 | *** join/#asterisk cgcorea (n=cgcorea@63.245.14.194) |
| 19:58.50 | MattB2 | chmod a+rw /path/to/folder -R |
| 19:58.58 | MattB2 | (the -R recurses and sets perms on subfolders and files) |
| 19:59.02 | shido6 | thanks |
| 19:59.02 | pygrammer | Hmm. |
| 19:59.07 | *** join/#asterisk dasuberdavid (n=dasuberd@digium.com) |
| 19:59.37 | pygrammer | I suppose this could be due to a multitude of things, but would anyone be able to speculate on why people on the other line seem to skip every five or so seconds a bit? |
| 20:00.02 | pygrammer | I'm using ulaw to Voxee.com and my * server is a PII 400/128 |
| 20:00.14 | pygrammer | with a headset and X-lite |
| 20:01.07 | adrenaline | ender, sorry I wish I could be more help then |
| 20:01.24 | ender | adrenaline: I've posted to the mailing list and I'm trying ot get ahold of polycom. We'll see. |
| 20:01.55 | *** join/#asterisk asterisk_newbie (n=root@12-203-248-191.client.insightBB.com) |
| 20:02.17 | ender | adrenaline: seems to effect the 301s as well. |
| 20:02.36 | asterisk_newbie | my asterisks is failing to connec to my sip proxie provider could it be a problem on my end or theirs |
| 20:02.38 | adrenaline | Polycom are jerks I have gotten a hold of them but they say they don't provide support and they divert you to your retailer who in turn tells you they just sell the phone and don't support it either |
| 20:02.52 | adrenaline | I will try my 301 too |
| 20:03.01 | Insanity5 | Is it easy to set up asterisk to record all incoming and outgoing calls? |
| 20:03.12 | pygrammer | So, any suggestions anyone? |
| 20:03.18 | pygrammer | Insanity5: Yes, it is. |
| 20:03.21 | pygrammer | Use Monitor() |
| 20:04.41 | hugo-v6 | got someone a minimum extensions.conf for me? |
| 20:05.05 | drumkilla_laptop | the minimum would be blank :) |
| 20:05.06 | adrenaline | ender, it worked on my 301 too |
| 20:05.15 | ender | adrenaline: must be a firmware issue. |
| 20:05.23 | hugo-v6 | drumkilla: maybe with the most used features/funktions ;) |
| 20:05.33 | hugo-v6 | not the contexts |
| 20:06.00 | hugo-v6 | there is a [general]. |
| 20:06.32 | hugo-v6 | i still use example. but well... its better to get it for myself. nvm |
| 20:07.01 | pygrammer | Can someone advise me as to why this skipping/lag might be? |
| 20:07.08 | pygrammer | I'm using ulaw to Voxee.com and my * server is a PII 400/128 |
| 20:07.12 | hugo-v6 | a) first think then say, b) not too much beer *burps* |
| 20:07.18 | pygrammer | with a headset and X-lite |
| 20:07.44 | pygrammer | when i call someone, i get quick skipping every five or so seconds |
| 20:07.48 | *** join/#asterisk Cresl1n (n=Cresl1n@digium.com) |
| 20:11.23 | Tabo | asterisk_newbie: i got the same problem!! |
| 20:11.27 | hadi57 | hello all, can i use this software as a pbx using land line, or just voip only? and if cards are needed are they available any where? |
| 20:11.29 | Tabo | who is your provider |
| 20:11.49 | Tabo | hadi: yes, you need to buy a digium card with 100m modudles |
| 20:11.50 | Tabo | modules |
| 20:11.51 | Hmmhesays | ~rking |
| 20:11.54 | Hmmhesays | err |
| 20:11.56 | Hmmhesays | ~seen rking |
| 20:12.01 | jbot | rking <n=rking@ip68-105-231-56.lu.dl.cox.net> was last seen on IRC in channel #debian, 15h 12m 10s ago, saying: 'cheal: hehe - nm, it's not a big deal'. |
| 20:12.01 | Tabo | if you want to user a landline |
| 20:12.18 | Tabo | you can use voip lines which is great if you do alot of long distant calling |
| 20:12.18 | wunderkin | pygrammer: narrow the problem down, if the * server is local, try calling extensions |
| 20:12.20 | lathos42 | ender: I just scrolled back and saw your messages about the IP501.. I just checked and the same thing happens to me |
| 20:13.11 | malverian[work] | Is it a big deal for a TE110P to be on the same interrupt as an ethernet card? |
| 20:13.21 | stepcut | malverian[work]: interrupt 11 ? |
| 20:13.27 | Hmmhesays | yes |
| 20:13.36 | malverian[work] | <PROTECTED> |
| 20:13.37 | malverian[work] | <PROTECTED> |
| 20:13.43 | Hmmhesays | unless you like strange things happening |
| 20:14.03 | malverian[work] | Hmmhesays, I've never had IRQ problems as much as I've had them with Digium cards :-P |
| 20:14.12 | bkw_ | oh thats a nice combo |
| 20:14.13 | Hmmhesays | so buy sangoma |
| 20:14.27 | *** join/#asterisk MattH (n=MattH@63.174.244.174) |
| 20:14.39 | Hmmhesays | what up bkw_ |
| 20:14.47 | malverian[work] | I had to buy a new sound card to resolve an irq problem that was causing crashes (had a single span tdm card) |
| 20:14.52 | MattH | Has anyone here gotten "accountcode" to work with the current CVS-HEAD ? My config seems to ignore it.. even when it worked in 1.0.9 |
| 20:15.08 | ender | lathos42: are you using the latest firmware? |
| 20:15.11 | Hmmhesays | anyone using ldapget? i'm curious if it is stable |
| 20:15.12 | pygrammer | wunderkin: ok |
| 20:15.17 | hadi57 | well i am sharing satellite internet on 4 different locations using linux with proxy server and 90% of my clients use pc2phone to call their overseas home, so this will be helpful then |
| 20:15.18 | Hmmhesays | well, stable enough for production |
| 20:15.21 | fugitivo | how many bandwith do i need for 100 simultaneous g729 calls? |
| 20:15.28 | Hmmhesays | 2 meg |
| 20:15.38 | malverian[work] | If having the same IRQ causes problems, why does the BIOS insist on assigning the same IRQ to several hardware? |
| 20:15.43 | pygrammer | wunderkin: I hear it even on calls to "Comedienne mail" |
| 20:15.48 | lathos42 | ender: yeah, 2.6.2, with SIP 1.5.2 |
| 20:15.50 | pygrammer | wunderkin: she says "Ma-aaa-aailbox" |
| 20:15.57 | pygrammer | it skips |
| 20:16.03 | lathos42 | ender: I'm going to try downgrading the firmware |
| 20:16.18 | ender | lathos42: yeah. Downgraded firmware doesn't seem tow ork as well for other things ): |
| 20:16.44 | ender | lathos42: think it's the sip release? |
| 20:16.50 | lathos42 | ender: What problems do you see with the older Bootrom? |
| 20:17.26 | ender | lathos42: I can't remember now, and i tmay have been older than the one previous to 1.5.2 |
| 20:17.46 | Assid | umm.. when i use voicemailmain(${calleridnum}) .. i am getting my sip username |
| 20:17.52 | lathos42 | ender: Oh, i'm downgrading to Bootrom 2.6.1, but keeping SIP 1.5.2, and i'll let you know if I have the contact list problem |
| 20:17.53 | Assid | is there a way to get the number itself? |
| 20:18.03 | ender | lathos42: ok. |
| 20:18.04 | pygrammer | wunderkin: any advice? (see above) |
| 20:18.11 | Hmmhesays | because calleridnum is not a variable |
| 20:18.33 | ender | Assid: I ended up using extensiosn as the sip username |
| 20:18.33 | Hmmhesays | well it can be, but not the one you're looking for |
| 20:18.53 | Hmmhesays | cheers to that lathos42 |
| 20:19.02 | Assid | ender: already set up everything :| |
| 20:19.06 | *** join/#asterisk zeedo (n=zeedo@www.reboot-robot.net) |
| 20:19.12 | ender | Assid: guess you didn't test that first |
| 20:19.36 | Assid | nope :( |
| 20:19.41 | Assid | came and bit me in the ass |
| 20:20.10 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 20:20.58 | Hmmhesays | why is it eleventy billion degrees in this office |
| 20:21.16 | pygrammer | Hmmhesays: Turn up the AC? :) |
| 20:21.16 | ender | Hmmhesays: because obviously your thermomiter is broken. |
| 20:21.18 | Cresl1n | malverian[work]: The problem is that you're dealing with a realtime application |
| 20:21.36 | Hmmhesays | I don't have access to the environmental controls here |
| 20:21.51 | Cresl1n | malverian[work]: so if you loose data (miss interrupts) it can cause problems |
| 20:21.56 | ender | lathos42: how might I force my phone to downgrade? |
| 20:22.00 | JerJer | throw a chair out of the window - gotta let ppl know you are serious |
| 20:22.11 | cpatry | Hmmhesays: access only to ur bar in fargo? :) |
| 20:22.11 | JerJer | seriously insane |
| 20:22.34 | Cresl1n | malverian[work]: zaptel tries to make the PC look as much like a real TDM switch as possible |
| 20:22.35 | lathos42 | ender: I just copied the older bootrom.ld to my tftp root, and the phone did the rest |
| 20:22.38 | Hmmhesays | cpatry: have i have idea what you just said |
| 20:22.45 | ender | oh ok. |
| 20:22.55 | Hmmhesays | ldapget anyone? |
| 20:23.04 | ender | lathos42: yeah, it' sprobably sip |
| 20:23.22 | ender | damn, mine just skipped right by. |
| 20:23.24 | Hmmhesays | ugh i'm going to have to open a web browser and go to the mailing list |
| 20:23.35 | ender | went straight to loading application. |
| 20:24.02 | tzanger | bkw_: I figred out why the private msg stuff to you wasn't working |
| 20:24.10 | bkw_ | why? |
| 20:24.12 | tzanger | they turned off privmsg from unregistered accounts and my connection must have bounced |
| 20:24.15 | tzanger | I'm registered now |
| 20:24.24 | ender | interesting |
| 20:24.29 | bkw_ | thats intresting |
| 20:24.29 | Assid | eeeks.. all of a sudden.. my softphone causing issues : Sep 1 16:23:45 NOTICE[3096]: chan_sip.c:5617 check_auth: stale nonce received from |
| 20:24.37 | Hmmhesays | yes they did, freenode screams at you if you try it |
| 20:24.38 | tzanger | 10:21 [freenode] -!- Private messages from unregistered users are currently blocked due to spam problems, but you can |
| 20:24.42 | tzanger | <PROTECTED> |
| 20:25.10 | tzanger | I hate stale nonces |
| 20:25.11 | Hmmhesays | yo file |
| 20:25.14 | tzanger | the fresh ones are so much tastier |
| 20:25.18 | Hmmhesays | tell me you've used ldapget |
| 20:25.31 | file | I haven't |
| 20:25.37 | file | poor you :P |
| 20:25.40 | Hmmhesays | no one has, it looks sweet, but I dunno if it works |
| 20:26.04 | *** join/#asterisk Reltsuh (n=Takeshi@S010600095b10aecd.ed.shawcable.net) |
| 20:26.08 | BrianR___ | Hmm.. The asterisk developer conference bridge now reads your callerid with a speech synthesizer... |
| 20:26.09 | Reltsuh | Hello? |
| 20:26.13 | ender | lathos42: d'oh, I was pointing at the wrong tftp server. Trying again to get older sip. |
| 20:26.22 | Reltsuh | Hey dudes I have a quick question. |
| 20:26.23 | Hmmhesays | it would be nice to grab an extension and associated user with ldap |
| 20:26.33 | lathos42 | ender: Which SIP version are you trying? |
| 20:26.33 | pygrammer | http://www.amazon.com/exec/obidos/tg/detail/-/B0009PVXD4/ref=olp_product_details/002-9753406-7514458?v=glance&s=electronics |
| 20:26.34 | BrianR___ | that's very neato.. |
| 20:26.39 | ender | lathos42: 1.4.1 |
| 20:26.49 | ender | lathos42: looks like the one released just before 1.5.2 |
| 20:27.01 | Reltsuh | Does anyone know if theres a way to check if your CPU is working? |
| 20:27.10 | Reltsuh | Besides putting it in another box. |
| 20:27.19 | *** join/#asterisk Ariel_ (n=Ariel@adsl-068-157-125-248.sip.mia.bellsouth.net) |
| 20:27.23 | Hmmhesays | listen to your post beeps? |
| 20:27.37 | Reltsuh | I think theres just 1 solid beep like normal dude. |
| 20:27.55 | Reltsuh | I don't know what the problem is.. I have a new PSU, a new mobo, and new RAM.. |
| 20:27.57 | Hmmhesays | you think |
| 20:28.09 | Reltsuh | and my old CPU but its all compatible. |
| 20:28.14 | Reltsuh | so.. |
| 20:28.16 | Hmmhesays | people used to think the sun revolved around the earth |
| 20:28.19 | Reltsuh | i dont know man. |
| 20:28.23 | Assid | whats a stale none ? |
| 20:28.27 | Assid | nonce eve? |
| 20:29.03 | Reltsuh | Can someone help me with my hardware question? |
| 20:29.11 | Ariel_ | Tabo, I got your fax. There are a few settings you need to add to your sip.conf file. |
| 20:29.22 | lathos42 | Anyone remember me complaining about my IP 501 rebooting after being gone for the weekend? |
| 20:29.31 | tzanger | lathos42: yes |
| 20:29.51 | Ariel_ | lathos42, yes I remember |
| 20:30.01 | Ariel_ | so is it still rebooting |
| 20:30.18 | ender | lathos42: older sip release looks fugly |
| 20:30.19 | Assid | tzanger: wjhats that issue supposed to be about? |
| 20:30.22 | Assid | any clue? |
| 20:30.24 | ender | menu structure that is. |
| 20:30.26 | tzanger | Assid: no idea |
| 20:30.28 | *** part/#asterisk Reltsuh (n=Takeshi@S010600095b10aecd.ed.shawcable.net) |
| 20:30.42 | lathos42 | Turns out that the switch on my desk is a piece of crap that keeps disconnecting the devices plugged into it.. I needed to look at my dmesg today, and all it displayed were "eth0: failed to detect cable link", and the corresponding reconnects |
| 20:31.19 | Ariel_ | lathos42, I knew it was not the IP-501. There good phones. |
| 20:31.33 | Tabo | ariel |
| 20:31.37 | Tabo | i'm trying broadvoice now |
| 20:31.44 | Ariel_ | Tabo, ok |
| 20:31.55 | ender | lathos42: ok, yeah, older sip release worked. |
| 20:32.01 | Tabo | ariel.. can u resetup the sip? |
| 20:32.08 | Tabo | like add it again |
| 20:32.10 | lathos42 | Ariel_: I had a feeling it wasnt either.. and I had a further feeling about it when my SPA-841 randomly rebooted yesterday.. but I couldnt figure out the reason exactly, until today |
| 20:32.15 | Tabo | id on't think broadvoice is instant |
| 20:32.18 | Tabo | so may have to wait 24 hours |
| 20:32.34 | rephorm | Ariel_: i've had a few 301's and one 500 crap out on me (network port just stopped working. but, overall they're really nice phones |
| 20:32.44 | Ariel_ | Tabo, see the PM |
| 20:32.49 | lathos42 | ender: so it looks like old = fugly but works, new = pretty but doesnt let you add contacts |
| 20:32.55 | *** join/#asterisk atmel (n=vlad@ip68-4-101-199.oc.oc.cox.net) |
| 20:33.12 | Ariel_ | rephorm, I have only had one with a problem. But I have about 200 now at customers working fine. |
| 20:33.17 | rephorm | lathos42: menu > features > contact list > add |
| 20:33.20 | ender | lathos42: thats about the size of it. |
| 20:33.37 | ender | rephorm: doesn't save them w/ Sip 1.5.2 release |
| 20:33.43 | rephorm | Ariel_: cool. i've installed about 200 total and had about 4 or 5 crap out. |
| 20:33.50 | file | Asterlink customers watch out, I'm emailing you :P |
| 20:34.35 | ender | lathos42: the contact save is the only issue I had w/ the 501s |
| 20:35.03 | ender | I was having a hell of a time w/ GMT stuff until I realized that my DHCP server was sending out an offset setting that was for somewhere else. |
| 20:36.21 | rephorm | ender: hmm. saves on this 301 with 1.5.2 dont' have any 501's around to test atm. |
| 20:37.29 | Uther_P | can someone tell me where make gets its Global variables from? (I updated perl, and make;s Globals are still pointing to the old version) |
| 20:37.59 | malverian[work] | Hmmhesays, Okay.. my bad I had APIC support turned off in the kernel ;) |
| 20:38.05 | malverian[work] | I don't understand why that's not enabled by default... hah |
| 20:38.16 | ender | rephorm: strange. I've had 2x 501s personally, and lathos42's 501 not work. My one 301 doesn't work either. |
| 20:38.31 | tzanger | from "sip debug" output, does "Looking for 5194894902 in default |
| 20:38.33 | Hmmhesays | were you actually saying that to me or did you tab complete? |
| 20:38.39 | tzanger | mean that it's looking for that exten in the default context? |
| 20:38.39 | Cresl1n | malverian[work]: some boards have buggy apics |
| 20:38.47 | tzanger | my user entry says to use 'dids' not 'default' |
| 20:38.48 | Cresl1n | malverian[work]: it can cause problems with some kernels |
| 20:38.59 | Ariel_ | are you just trying to save a contact on the IP-500/1 via the menus on the phones? |
| 20:38.59 | Cresl1n | file!!! |
| 20:39.00 | Cresl1n | :-) |
| 20:39.03 | file | Cresl1n!!! |
| 20:39.12 | malverian[work] | Cresl1n, Hmm.. it's a pretty new Intel board.. hopefully there won't be any issues. |
| 20:39.14 | ender | Ariel_: yes |
| 20:39.32 | Ariel_ | let me go see if I can save one on the phones here. |
| 20:39.33 | brettnem | hey all |
| 20:39.38 | ender | Ariel_: k |
| 20:39.52 | *** join/#asterisk naturalblue (n=Adam@195.26.12.229) |
| 20:39.57 | brettnem | any anyone know why ulaw inband dtmf from asterisk to asterisk wouldn't work?? |
| 20:40.25 | tzanger | oh for fuck sakes |
| 20:40.41 | Cresl1n | brettnem yeah |
| 20:40.45 | Cresl1n | brettnem: codecs |
| 20:40.50 | brettnem | ULAW |
| 20:41.03 | Cresl1n | brettnem: maybe some bad packet loss |
| 20:41.07 | brettnem | nope |
| 20:41.18 | brettnem | low jitter.. no packet loss |
| 20:41.21 | gtigene | My new Asterisk server won't play any sounds. Like when I execute VoicemailMain, the console says "playing vm-password" but it doesn't. What could I have overlooked? |
| 20:41.37 | brettnem | any other ideas? |
| 20:41.41 | Cresl1n | nope :-( |
| 20:41.45 | brettnem | crud |
| 20:42.23 | ender | lathos42: see! |
| 20:42.36 | Ariel_ | well it works fine in the directory on my phones here in the office. I did 3 of them all save and are able to use the directory. |
| 20:42.41 | *** join/#asterisk crich1999 (n=crich@port-212-202-0-21.dynamic.qsc.de) |
| 20:43.39 | ender | Ariel_: sip.ld version? |
| 20:44.26 | *** join/#asterisk cgcorea (n=cgcorea@63.245.14.194) |
| 20:44.51 | ender | lathos42: 1.4.1 doesn't use the same concept for line registrations |
| 20:44.51 | Tabo | Ariel: any luck>? |
| 20:44.57 | lathos42 | ahh |
| 20:45.53 | rephorm | ender: is it not saving at all, or just not saving when you reboot? |
| 20:46.30 | *** part/#asterisk naturalblue (n=Adam@195.26.12.229) |
| 20:46.45 | Ariel_ | Tabo, I am not able to connect to your system right now. In about an hour or so I can. |
| 20:47.15 | *** join/#asterisk r0d3nt (i=anonymou@tinfoilhat.net) |
| 20:47.30 | ender | rephorm: not saving at all. Usually get a 'Busy! Please try again later' message. |
| 20:48.14 | rephorm | ender: hmm. strange... |
| 20:48.23 | gtigene | Is there any known problem with CVS Head not playing the sound gsm files correctly? |
| 20:49.10 | Hmmhesays | file |
| 20:49.10 | lathos42 | I wonder if the phone tried to update the directory on the tftp server |
| 20:49.15 | brc_ | BRANE SURGERY SUCKS! |
| 20:49.17 | lathos42 | err tries |
| 20:49.20 | Hmmhesays | LOL |
| 20:49.20 | brc_ | isn't that right Hmmhesays |
| 20:49.23 | ender | lathos42: could be. |
| 20:49.24 | *** join/#asterisk derek_1234 (n=derek@203.167.203.10) |
| 20:49.26 | Hmmhesays | that is right *twitch* |
| 20:49.34 | brc_ | I tried it once |
| 20:49.40 | brc_ | the patient died |
| 20:49.49 | Hmmhesays | file I have 4 days to learn a lot about SER, is there any good reading you can recommend besides the administrators guide? |
| 20:49.53 | *** join/#asterisk j4m3s (n=debbie@digium.com) |
| 20:49.55 | file | Damin__: cheeky bugger |
| 20:49.57 | brc_ | ha. |
| 20:50.02 | j4m3s | hello |
| 20:50.05 | Ariel_ | Hmmhesays, no |
| 20:50.05 | file | Hmmhesays: I really don't learn using that stuff, I just dive into it |
| 20:50.20 | Hmmhesays | If I had more time I would |
| 20:50.21 | file | I set out to do something, and do it |
| 20:50.24 | Hmmhesays | but I need something more concise |
| 20:50.50 | rephorm | gtigene: sometimes sound gets screwy if you have a zap card with buggered drivers |
| 20:50.53 | Hmmhesays | cause 4 days is not very long |
| 20:51.00 | j4m3s | brc_ kevin told me to talk to you about jira |
| 20:51.05 | brc_ | how'd you get roped into that? |
| 20:51.11 | brc_ | j4m3s, hrm? |
| 20:51.19 | gtigene | rephorm: thanks |
| 20:51.50 | derek_1234 | What is this shit on the mailing list about doxygen comments on functions. |
| 20:52.06 | derek_1234 | Surely, it is required to have doxygen comments on functions. |
| 20:52.10 | ender | ah! |
| 20:52.13 | Hmmhesays | for SER Ariel_? |
| 20:52.21 | Ariel_ | Hmmhesays, yes |
| 20:52.24 | ender | there is no ipmid.cfg file in the 1.5.2 release of firmware. |
| 20:52.33 | derek_1234 | I think the developers have been lazy in not enforcing standard rules on coding styles. |
| 20:52.53 | derek_1234 | Real projects (alsa for example, openh323) all have doxygen comments. |
| 20:53.08 | derek_1234 | it is a point of shame (in my view) that there is no such comments. |
| 20:53.14 | brettnem | "real projects" |
| 20:53.28 | derek_1234 | However, given that you have 1600 line long functions, how long would the comments have to be ? |
| 20:53.44 | brettnem | "real voip servers support voiceXML" |
| 20:53.49 | ender | or rather, the config files may not have a <directory> entry |
| 20:54.00 | brettnem | "real T1 cards support 15 minute performance statistics" |
| 20:54.13 | brettnem | "real carriers provide redundancy" |
| 20:54.22 | Ariel_ | rephorm, ender 1.5.2 |
| 20:54.49 | rephorm | Ariel_: yep |
| 20:54.54 | lathos42 | ender: I found out the problem |
| 20:54.57 | ender | Ariel_: yeah, I think I found whats going on. |
| 20:55.03 | *** join/#asterisk hellagony (n=egutierr@200.121.129.178) |
| 20:55.12 | ender | lathos42: phone not set to use volit memory for contacts? |
| 20:56.00 | lathos42 | ender: I'm not sure about that, but I created a <mac-addr>-directory.xml for the phone, and it saved properly.. when I checked the file on the server, it had added the entry to the xml file |
| 20:56.02 | *** join/#asterisk bluesky (n=maxgluck@200.109.166.172) |
| 20:56.33 | ender | lathos42: yeah, tftp? |
| 20:56.44 | lathos42 | ender: Yeah |
| 20:57.04 | bluesky | hello, Asterisk doesn't start as a service, I'm getting a permission denied to execvp. Any advise on how to set permission please? |
| 20:57.06 | lathos42 | ender: I hadnt gotten around to creating a directory yet since i'm still messing around with settings |
| 20:57.27 | ender | lathos42: so you just touched the file, or did you put in the xml headers and stuff? |
| 20:58.01 | *** join/#asterisk oden (n=oden@194-237-146-22.customer.telia.com) |
| 20:58.02 | lathos42 | ender: I copied the example one that had been put there.. So it had the sample entries in it |
| 20:58.05 | ender | n/m, I have an example to copy over. |
| 20:58.07 | ender | nod |
| 20:58.37 | *** join/#asterisk mithro (n=tim@c213-100-42-188.swipnet.se) |
| 20:59.27 | malverian[work] | Is there any way to set up a pseudo-console that just outputs incoming audio packets to /dev/null or something? |
| 21:01.08 | *** part/#asterisk oden (n=oden@194-237-146-22.customer.telia.com) |
| 21:02.48 | *** join/#asterisk DrGamut (n=DrGamut@c-67-183-224-69.hsd1.wa.comcast.net) |
| 21:02.52 | DrGamut | Can anyone in here read Japanese? |
| 21:03.15 | Nugget | I can read "entrance", "exit", and "coffee sold here". |
| 21:03.23 | Nugget | beyond that I haven't really needed to learn much. :) |
| 21:03.46 | lathos42 | ender: It looks like maybe it just needs to have its directory started before it'll let you save.. I removed the phone specific directory, and its still got the entry I added saved |
| 21:04.22 | *** part/#asterisk Samoied (n=Samoied@popeye.opens.com.br) |
| 21:05.17 | lathos42 | ender: I'll have to work on this some more tomorrow, its time for me to get out of here |
| 21:06.10 | adrenaline | Does anybody know how to set up a stun server in the Polycom configs? |
| 21:06.13 | ender | lathos42: yeah, could be saved in volit or something. |
| 21:06.40 | *** part/#asterisk DrGamut (n=DrGamut@c-67-183-224-69.hsd1.wa.comcast.net) |
| 21:09.25 | *** join/#asterisk sigmounte (n=sigmount@www.sighq.net) |
| 21:11.09 | *** join/#asterisk ttyp0 (n=ttyp0@112.Red-83-53-241.pooles.rima-tde.net) |
| 21:14.06 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 21:14.57 | *** join/#asterisk Equinox (n=secret@27-74.202-68.tampabay.res.rr.com) |
| 21:15.00 | *** join/#asterisk Gronker__ (n=Gronker2@adsl-220-90-11.ags.bellsouth.net) |
| 21:19.49 | hugo-v6 | hrmz. i think i dont like chan_misdn. bad documentation. |
| 21:20.42 | *** part/#asterisk Gronker__ (n=Gronker2@adsl-220-90-11.ags.bellsouth.net) |
| 21:21.09 | Cresl1n | me too :-) |
| 21:21.52 | *** join/#asterisk kFuQ (n=somedude@c-67-185-114-199.hsd1.wa.comcast.net) |
| 21:22.18 | hugo-v6 | well... i think i dont like mISDN also :( i loaded every needed module (at least i think so) but i dont get it to work :/ |
| 21:24.16 | groogs | building zaptel HEAD .. is there any difference between "make" and "make linux26" ? i ran just make, and got .ko files.. |
| 21:24.37 | Cresl1n | that should work fine |
| 21:24.50 | *** join/#asterisk zotz (n=zotz@24.231.36.100) |
| 21:28.21 | many | hugo: haha. |
| 21:30.38 | Hmmhesays | hieroglyphics I wanna be pacific I wanna get down in your south seas |
| 21:32.07 | *** join/#asterisk bobbyd (n=bobby@195.137.31.171) |
| 21:32.15 | bobbyd | hi |
| 21:32.17 | hugo-v6 | many: dont laugh, thats sad... a little bit more and i start to cry like a baby |
| 21:32.28 | bobbyd | would I use asterix to do conference calling with voip? |
| 21:32.34 | hugo-v6 | capiinfo wont give me output only no such device |
| 21:32.40 | Hmmhesays | so anyone know if tsume fixed his problem last night? I felt kind of bad for the guy |
| 21:33.06 | many | hugo: qry me over there tomorrow, ill help ya out ;) |
| 21:33.10 | bluesky | <PROTECTED> |
| 21:33.13 | pygrammer | So, has anybody had skipping/jitter/whatever it's called, and what did you do to fix it? I have it even internally to internal extensions, not over the Net |
| 21:33.27 | *** join/#asterisk ttyp0 (n=ttyp0@112.Red-83-53-241.pooles.rima-tde.net) |
| 21:33.30 | pygrammer | i'm using uLaw in all directions |
| 21:33.31 | hugo-v6 | many: thats nice. thank you. on which time? |
| 21:33.38 | hugo-v6 | at |
| 21:33.39 | *** join/#asterisk __unnamed__ (n=pape@modemcable224.217-130-66.mc.videotron.ca) |
| 21:33.43 | pygrammer | with a headset and x-lite softphone |
| 21:33.54 | many | 10-14 uhr. danach is meeting und danach bin ich unterwegs |
| 21:34.19 | hugo-v6 | ja prima :) thank you dude. |
| 21:34.35 | Hmmhesays | ~seen Ariel_ |
| 21:34.38 | jbot | ariel_ is currently on #asterisk (1h 7m 20s). Has said a total of 14 messages. Is idling for 40m 16s |
| 21:34.48 | Hmmhesays | ~seen Hmmhesays |
| 21:34.50 | jbot | hmmhesays is currently on #asterisk (8h 12m 38s). Has said a total of 195 messages. Is idling for 2s |
| 21:34.53 | hugo-v6 | at 10 am should the postman bring me my snom :) good time to start with you :) |
| 21:34.59 | pygrammer | jesus christ |
| 21:35.01 | pygrammer | anyone? |
| 21:35.02 | many | hehe |
| 21:35.05 | Hmmhesays | hugo-v6 snoms are cool |
| 21:35.06 | pygrammer | i've asked for advice twice |
| 21:35.12 | pygrammer | can anyone give me advice? |
| 21:35.16 | Hmmhesays | pygrammer what is your major malfunction |
| 21:35.20 | pygrammer | this is really fucking frustrating |
| 21:35.24 | hugo-v6 | Hmmhesays: ill hope so. looks nice and got blinking leds :> |
| 21:35.26 | Ariel_ | Hmmhesays, what's up? |
| 21:35.39 | pygrammer | Hmmhesays: i already told you ... i'm getting skipping/jitter/whatever it's called, even internally |
| 21:35.43 | Hmmhesays | did tsume ever fix his problem last night? I felt bad for the guy |
| 21:35.54 | pygrammer | using uLaw in all directions, but i have a relatively fast connection here, and obviously a 100 mbps connection to my asterisk server |
| 21:35.57 | *** join/#asterisk Sedorox (i=brandon@smartserv/cna/Sedorox) |
| 21:35.59 | Hmmhesays | pygrammer: are you swapping rtp? |
| 21:36.11 | glm2k | pygrammer: hmmm, tricky |
| 21:36.13 | pygrammer | maybe it's a RAM problem on my * server as it only has 128 megs and is a PII |
| 21:36.16 | Ariel_ | Hmmhesays, don't know I had to go. |
| 21:36.19 | pygrammer | Hmmhesays: ? |
| 21:36.20 | many | das setup laeuft hier mit snom und misdn (und einer beronat quad isdn) |
| 21:36.24 | hugo-v6 | many: ok will talk to you tomorrow and go to get some sleep now. this beer makes be dizzy |
| 21:36.30 | pygrammer | what do you mean? i'm using either sip or iax to *, and then sip to Voxee |
| 21:36.31 | many | yea, hf :) |
| 21:36.35 | Hmmhesays | pygrammer: good call use top while you make a call |
| 21:36.37 | hugo-v6 | many: mein mann :) |
| 21:36.46 | glm2k | pygrammer: what Hmmhesays said |
| 21:36.56 | hugo-v6 | well then, bis morgen :) |
| 21:37.01 | hugo-v6 | gd n8 pals. |
| 21:37.03 | Hmmhesays | if you see swap usage like mad that is your problem |
| 21:37.22 | Ariel_ | pygrammer, are you using any zap phones? |
| 21:37.29 | pygrammer | Hmmhesays: ok, but someone told me people run * on 16 and 32 megs of RAM |
| 21:37.33 | pygrammer | Ariel_: none |
| 21:37.35 | pygrammer | no external phones |
| 21:37.37 | Hmmhesays | they were drunk |
| 21:37.40 | pygrammer | just one softphone |
| 21:37.57 | pygrammer | Hmmhesays: btw, my cpu usage is almost always pretty idle |
| 21:37.59 | Ariel_ | 16 to 32 mgs |
| 21:38.15 | pygrammer | but ram is consistently 123 Megs in use |
| 21:38.18 | Hmmhesays | pygrammer: do what I said |
| 21:38.19 | Ariel_ | wow you sure need to make sure you trim down asterisk with lots of noload on what you dont need. |
| 21:38.22 | pygrammer | even when not making a call |
| 21:38.40 | Ariel_ | pygrammer, do you have mpg123 running |
| 21:38.42 | glm2k | 2nd to compiling your own kernel |
| 21:38.54 | pygrammer | playing on-hold music, 97.0% idle (call in progress to my own extension) |
| 21:39.01 | pygrammer | 95.4% idle |
| 21:39.08 | Hmmhesays | i said swap space boi |
| 21:39.08 | pygrammer | still 123 megs of RAM in use |
| 21:39.15 | pygrammer | Ariel_: yes, on-hold music is playing now |
| 21:39.16 | pygrammer | sounds fine |
| 21:39.32 | pygrammer | then it goes back to 99.0% idle |
| 21:39.42 | Ariel_ | pygrammer, you only have 16 to 32 mgs of ram? |
| 21:39.47 | pygrammer | ram doesn't change much, but it's almost always 122-124 megs in use |
| 21:39.50 | pygrammer | Ariel_: no... i have 128 |
| 21:39.58 | pygrammer | but someone told me that it isn't a ram problem, as people run * with 16-32 |
| 21:40.00 | Ariel_ | ahh |
| 21:40.04 | *** join/#asterisk mogorman (n=mogorman@digium.com) |
| 21:40.12 | pygrammer | so, based on what i said Hmmhesays what do you think? |
| 21:40.17 | pygrammer | i can try an external call |
| 21:40.23 | Ariel_ | pygrammer, yes the do but scale down on moh and other stuff that they don't load |
| 21:40.25 | Hmmhesays | hold |
| 21:41.09 | pygrammer | like right now listening to my messages, the quality of Allison's voice just doesn't sound righty |
| 21:41.10 | pygrammer | *right |
| 21:41.20 | pygrammer | it sort of sounds "off |
| 21:41.22 | pygrammer | "* |
| 21:41.40 | pygrammer | not continuous, but sort of "jittery" -- really can't explain |
| 21:42.10 | pygrammer | btw, i have 377 megs of swap, 2.9 megs used |
| 21:42.57 | adelas | someone recall the keys to unlock cisco 7960? |
| 21:43.03 | adelas | ##* or something |
| 21:43.30 | pygrammer | Hmmhesays: like, i just called my bank's automated system to listen to the automated voice, and it sounds sort of like Allison (the voice of Asterisk) sounds internally -- it's not skipping, but it just doesn't sound continuous and smiith |
| 21:43.40 | pygrammer | it's very hard to put into words |
| 21:43.42 | pygrammer | *ack |
| 21:43.43 | pygrammer | *smooth |
| 21:43.49 | Hmmhesays | pygrammer |
| 21:44.00 | pygrammer | ye |
| 21:44.01 | pygrammer | *yes |
| 21:45.15 | Phoebers | is this a good channel to spam newbie questions about Asterisk? :) |
| 21:45.22 | Cresl1n | yep |
| 21:45.30 | Phoebers | great :) |
| 21:45.35 | pygrammer | Hmmhesays: yes? |
| 21:45.38 | Ariel_ | pygrammer, your not running this on a network hub. Your using a switch correct. |
| 21:45.43 | Hmmhesays | hold on dude i'm on the phone |
| 21:45.45 | pygrammer | Ariel_: everything's on a router |
| 21:46.10 | pygrammer | behind a NAT, but i managed to get sip -> NAT to work |
| 21:46.20 | pygrammer | but even within the network, it is skippy |
| 21:46.33 | opus_ | pygrammer what problems did you run into? |
| 21:46.42 | Phoebers | ill spam some questions in one of the upcoming days ;). good night all. |
| 21:46.42 | *** join/#asterisk ttyp0 (n=ttyp0@143.Red-83-56-192.pooles.rima-tde.net) |
| 21:46.48 | pygrammer | gotta explain my problem like 5 times :-p |
| 21:46.52 | opus_ | I am having problems when, based on random behavior, asterisk will send the packet to the wrong destination |
| 21:47.00 | opus_ | hold on i'll read your log |
| 21:47.05 | pygrammer | ok |
| 21:47.18 | opus_ | can't find it |
| 21:47.23 | opus_ | i'm having problems as well |
| 21:47.30 | pygrammer | just scroll up ... |
| 21:47.37 | pygrammer | it's too much to type/paste again |
| 21:47.47 | opus_ | I have asterisk behind nat. I have a polycom behind nat. I have sip.conf general nat=yes. I have externalip/internal IP. I have sip user nat=yes. I have quailify =yes. When I type rtp debug, however, it is sending the packet to the internal address of the IP 500 phone. |
| 21:48.05 | opus_ | sorry, i'm running irc in screen and can't scroll up |
| 21:48.32 | *** join/#asterisk Nukemizer (n=Nuke@67.137.28.165) |
| 21:48.48 | pygrammer | opus_: basically, just getting poor voice quality and skipping when i call anything/anybody -- even internally within the network |
| 21:49.31 | *** join/#asterisk Dabba (n=d@ipv6.mfnx.ip6net.net) |
| 21:50.21 | opus_ | pygrammer do you have ToS turned on in asterisk, and do you have QoS switching hardware? (also check with tcpdump to see if it is ACTUALLY setting the ToS bit.) |
| 21:50.47 | pygrammer | opus_: what is ToS? |
| 21:50.57 | opus_ | ~ToS |
| 21:50.59 | jbot | well, tos is type of service, or the operating system for the Atari ST.. Terms of Service |
| 21:51.01 | opus_ | hmm |
| 21:51.10 | glm2k | Atari, lol |
| 21:51.19 | opus_ | pygrammer look at sip.conf.sample |
| 21:52.19 | pygrammer | opus_: i don't have that |
| 21:52.29 | pygrammer | i think i have it on my other machine |
| 21:52.32 | pygrammer | h/o |
| 21:53.42 | Dabba | if i do exten => s,1,Background(askfordate_and_time) then i want the caller to input a day being 0-6 and time being 0-23 then stick both those into varibles is there a simple way rather than having to _X. for each thing and putting $EXTEN into each var ? |
| 21:54.11 | pygrammer | umm...... |
| 21:54.14 | pygrammer | this isn't good |
| 21:54.20 | pygrammer | i enabled tos=184 and tos=lowdelay |
| 21:54.27 | pygrammer | and then i get this weird buzzing thing and then it's disconnected |
| 21:54.48 | *** join/#asterisk buddah (n=djbrianc@67.110.253.129) |
| 21:54.58 | buddah | anyone able to help me out with an IVR issue? |
| 21:55.22 | pygrammer | umm |
| 21:55.27 | pygrammer | anyone? this is really weird! |
| 21:55.49 | pygrammer | i'm using asterisk 1.0.7 |
| 21:55.58 | pygrammer | maybe there are bugs related to this |
| 21:56.17 | buddah | can't get it to accept an extension that gets keyed in |
| 21:56.23 | buddah | after it plays a message |
| 21:58.01 | buddah | http://pastebin.ca/21878 |
| 21:58.05 | buddah | thats the setup |
| 22:00.48 | Dabba | buddah looks ok to me, u using a sip device to get to the ivr ? |
| 22:01.22 | buddah | no |
| 22:01.43 | Dabba | pstn ? iax ? |
| 22:01.47 | buddah | thats another thing, i can't call a sip device from another sip device on the system |
| 22:01.48 | buddah | pstn |
| 22:01.54 | buddah | it plays the message |
| 22:01.56 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 22:02.05 | buddah | i key in 101 |
| 22:02.06 | buddah | nothing |
| 22:02.07 | buddah | 102 |
| 22:02.08 | buddah | nothing |
| 22:02.14 | buddah | then after the 10 sec, it goes to the timeout |
| 22:02.22 | buddah | and dials them all |
| 22:02.53 | Dabba | i have had that but with sip and it was the dtmf type, dunno re pstn is it isdn ? |
| 22:02.55 | *** join/#asterisk PyroSteve (n=pyrostev@24-159-79-219.dhcp.jcsn.tn.charter.com) |
| 22:03.38 | PyroSteve | helllo my friends |
| 22:03.58 | PyroSteve | finnally got an wireless connection after fleeying from new orleans |
| 22:04.25 | pygrammer | PyroSteve: i don't want to insult you, but that's hard to believe |
| 22:04.26 | buddah | gratz on getting out |
| 22:04.32 | PyroSteve | why ? |
| 22:04.45 | pygrammer | i don't know -- a lot of people say that to attract attention |
| 22:04.53 | PyroSteve | I live in metairie, 10 miles outside of new orleans |
| 22:04.57 | PyroSteve | Jefferson Parish |
| 22:04.59 | buddah | a lot of people say that because they did |
| 22:05.01 | buddah | get out |
| 22:05.15 | pygrammer | PyroSteve: well, congrats |
| 22:05.31 | PyroSteve | i dont my area got too much damage |
| 22:05.36 | PyroSteve | but i really dont know yet |
| 22:05.45 | PyroSteve | im in jackson, tenn. |
| 22:05.51 | PyroSteve | 6.5 hour drive |
| 22:06.02 | PyroSteve | found a nice Days Inn hotel with a access points |
| 22:06.14 | PyroSteve | I left my * box behind |
| 22:06.19 | PyroSteve | and I have a nufone account |
| 22:06.20 | pygrammer | ohnoes |
| 22:06.30 | PyroSteve | I noticed that nufone supports SIP |
| 22:06.38 | PyroSteve | I looked on thier site |
| 22:06.51 | PyroSteve | and cant find what ip or hostname to register with |
| 22:06.58 | PyroSteve | ive always used IAX |
| 22:07.12 | PyroSteve | but only have my laptop with winblows |
| 22:07.22 | PyroSteve | and Ive got Xlite installed |
| 22:07.57 | PyroSteve | what is the hostname for nufone that I can register with |
| 22:08.01 | *** join/#asterisk spackle (n=spackle@209.234.83.19) |
| 22:08.02 | PyroSteve | via SIP |
| 22:08.41 | PyroSteve | and do I need have nufone setup my account for SIP instead of IAX ? |
| 22:09.43 | *** join/#asterisk kFuQ (n=somedude@c-67-185-114-199.hsd1.wa.comcast.net) |
| 22:13.14 | *** join/#asterisk kFuQ (n=somedude@c-67-185-114-199.hsd1.wa.comcast.net) |
| 22:13.33 | fugitivo | isn't 2mb too much for 100 calls using g729? |
| 22:15.13 | *** join/#asterisk kFuQ (n=somedude@c-67-185-114-199.hsd1.wa.comcast.net) |
| 22:16.03 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 22:16.07 | PyroSteve | Oh WOW |
| 22:16.27 | PyroSteve | Seven kids |
| 22:16.41 | PyroSteve | under care of voluenteers |
| 22:16.48 | PyroSteve | all 7 kids |
| 22:16.53 | *** join/#asterisk kFuQ (n=somedude@c-67-185-114-199.hsd1.wa.comcast.net) |
| 22:16.55 | PyroSteve | are brothers/sisters |
| 22:17.02 | PyroSteve | they lost thier mother |
| 22:17.09 | PyroSteve | doesn't know where she is |
| 22:17.20 | PyroSteve | according to one of the kids |
| 22:17.30 | PyroSteve | that is older enough to communicate |
| 22:17.47 | PyroSteve | told people that her mothers name is ...... |
| 22:17.58 | PyroSteve | Katrina Williams !! |
| 22:17.59 | PyroSteve | wow |
| 22:18.02 | PyroSteve | shocking |
| 22:18.36 | *** join/#asterisk MikeJ[Laptop] (n=ircatjer@d14-69-8-30.try.wideopenwest.com) |
| 22:19.28 | buddah | grr |
| 22:19.45 | *** join/#asterisk MikeJ__ (n=ircatjer@d14-69-8-30.try.wideopenwest.com) |
| 22:21.36 | gaggaman | I am searching for info about the cisco skinny protocol, particular the video functions for the 7970. Has anybody ever seen anything about this? |
| 22:21.42 | generalhan | can some one tell me the command to turn off the sip debug ? |
| 22:22.11 | buddah | on the CLI? |
| 22:22.13 | gaggaman | sip no debug |
| 22:22.14 | buddah | sip no debug |
| 22:22.17 | generalhan | thanks guys |
| 22:22.40 | gaggaman | general got you cisco running? |
| 22:22.58 | *** join/#asterisk MikeJ[Laptop] (n=ircatjer@d14-69-8-30.try.wideopenwest.com) |
| 22:23.25 | Tabo | is 4 gigs enought for a asterisk server |
| 22:23.29 | Tabo | voicemail |
| 22:23.41 | Tabo | expecting like 30 voicemessages per day |
| 22:23.53 | *** join/#asterisk eKo1 (n=kino@metrored-gw.tropicohn.com) |
| 22:24.19 | gaggaman | tabo easily if you are not collecting them. |
| 22:24.36 | buddah | grr |
| 22:24.45 | gaggaman | one second is about 7k at a/u-law |
| 22:24.58 | eKo1 | isn't 0017324609000 supposed to match _00N. |
| 22:25.00 | buddah | wish this ivr worked |
| 22:25.04 | eKo1 | for some reason it won't work |
| 22:25.08 | glm2k | buddah: still the same problem? |
| 22:25.15 | buddah | yeah |
| 22:25.22 | buddah | cant put in the extension |
| 22:25.26 | glm2k | hmmm |
| 22:25.43 | gaggaman | general is your cisco sip working now? |
| 22:25.49 | *** join/#asterisk MikeJ__ (n=ircatjer@d14-69-8-30.try.wideopenwest.com) |
| 22:27.14 | pygrammer | so, no one has had skipping problems? |
| 22:27.21 | pygrammer | i.e. bad voice quality, skipping, that kind of stuff |
| 22:27.39 | glm2k | pygrammer: your guess is as good as mine, i'd blame latency first |
| 22:27.43 | fugitivo | pygrammer: with what? |
| 22:27.48 | pygrammer | glm2k: latency to my * server? |
| 22:27.51 | pygrammer | internally? |
| 22:27.53 | pygrammer | via 100 mbps lan? |
| 22:27.54 | *** join/#asterisk jsaunders (i=js@70.70.38.160) |
| 22:27.56 | pygrammer | 10 feet away? |
| 22:28.13 | fugitivo | pygrammer: what protocol, codec, phone and hardware? |
| 22:28.13 | glm2k | pygrammer: my point exactly, nothng else to blame. |
| 22:28.25 | jsaunders | How my bytes on average is consumed per typical h323 audio call, signalling only, no rtp. Anyone? |
| 22:28.47 | pygrammer | fugitivo: sip OR iax, uLaw all directions, headset/IAXComm/xlite, asterisk server is a PII w/ 128 megs |
| 22:28.47 | buddah | glm: any ideas? |
| 22:29.09 | pygrammer | glm2k: would using an older version of asterisk -- 1.0.7 -- have any relation? |
| 22:29.13 | PyroSteve | i cant register to nufone via sip ... with my account that i normally use via IAX |
| 22:29.20 | fugitivo | pygrammer: probably it's the softphone |
| 22:29.36 | fugitivo | pygrammer: try with a hardware phone |
| 22:29.38 | pygrammer | fugitivo: why would it be the softphone? |
| 22:29.41 | pygrammer | i don't have a hardphone |
| 22:29.45 | *** join/#asterisk MikeJ[Laptop] (n=ircatjer@d14-69-8-30.try.wideopenwest.com) |
| 22:29.46 | pygrammer | iaxcomm isn't reported to have those bugs |
| 22:29.49 | pygrammer | neither is xlite |
| 22:29.50 | PyroSteve | what ip/hostname do i register to |
| 22:30.00 | fugitivo | pygrammer: it depends on the machine running the softphone |
| 22:30.06 | pygrammer | this is *unacceptable* skipping and a lot of people use these apps, so people wouldn't use them if these problems were inherent in them |
| 22:30.18 | pygrammer | this is a p4 1.6 w/ 256 megs of RAM |
| 22:30.20 | PyroSteve | why am i being ignored ? |
| 22:30.28 | fugitivo | pygrammer: search for the problem in your pc |
| 22:30.29 | buddah | maybe nobody knows? |
| 22:30.36 | pygrammer | fugitivo: ? |
| 22:30.52 | fugitivo | pygrammer: maybe irq conflict between your video card and sound card? |
| 22:31.12 | PyroSteve | does anyone here use nufone via sip ? |
| 22:31.29 | pygrammer | fugitivo: heh, i suppose it could be anything. :( |
| 22:31.33 | pygrammer | so hard to pinpoint these things |
| 22:31.48 | glm2k | buddah: try inserting a _X,1,NoOp( only this got typed ${EXTEN} ) right after line 5 |
| 22:31.53 | fugitivo | pygrammer: i have a test machine, p2 400, 256mb of ram, it works perfectly with my hardphones, a bit choppy with xlite |
| 22:32.11 | *** join/#asterisk MikeJ[Laptop] (n=ircatjer@d14-69-8-30.try.wideopenwest.com) |
| 22:32.14 | pygrammer | fugitivo: yea |
| 22:32.26 | buddah | ok |
| 22:32.26 | pygrammer | fugitivo: what about iaxcomm? have you tried that? |
| 22:32.42 | glm2k | pygrammer: are you running anything graphical or virtual frame buffer perhaps? |
| 22:32.56 | glm2k | pygrammer: silly question, but i had to ask |
| 22:33.06 | fugitivo | pygrammer: not really, i use hardphones usually |
| 22:33.20 | pygrammer | glm2k: huh? what do you mean "anything graphical or virtual frame buffer?" |
| 22:33.25 | pygrammer | what is a virtual frame buffer? |
| 22:33.31 | *** join/#asterisk bjohnson (n=bjohnson@i216-58-16-130.cybersurf.com) |
| 22:33.33 | *** join/#asterisk io_ (n=io@65.39.69.38) |
| 22:33.41 | glm2k | it's a display mode |
| 22:34.01 | pygrammer | a display mode in what? |
| 22:34.28 | *** part/#asterisk eKo1 (n=kino@metrored-gw.tropicohn.com) |
| 22:34.56 | io_ | kernel 2.6, ztdummy loads with no problems, but ztcfg -vvv pukes, what should zaptel.conf look like? |
| 22:35.28 | glm2k | pygrammer: http://www.voip-info.org/tiki-index.php?page=Asterisk+disable+frame+buffer |
| 22:35.49 | pygrammer | it's funny that not many people report this problem -- it would seem to me a very common problem, that people just aren't satisfied with the voice quality -- the most noticeable thing in a new setup -- and choppiness of phone calls |
| 22:36.34 | glm2k | i think echo is a more common problem |
| 22:36.54 | glm2k | bad cell or landlines can be jittery, so i believe people are more forgiving |
| 22:37.04 | *** part/#asterisk yaboo (n=jsirucka@220.245.131.131) |
| 22:37.12 | pygrammer | well i'm not -- our landlines and cell phones are good |
| 22:37.24 | pygrammer | and i'd like this to be as good (or better sip-to-sip) |
| 22:37.35 | pygrammer | glm2k: do you think that's what it is (the frame buffer thing)? |
| 22:37.51 | buddah | g1m2k: still not working |
| 22:38.18 | glm2k | i wasn't saying your landlines or cell are to blame, just a note as to why people might complain less about jitter then echo. |
| 22:38.32 | pygrammer | glm2k: interestingly, music-on-hold sounds perfect |
| 22:38.37 | pygrammer | so i don't think it's a latency thing |
| 22:38.38 | glm2k | pygrammer: and no, i can't be sure either, just following tips and tricks for * on voip-info |
| 22:39.00 | glm2k | buddah: what does it say in the cli? |
| 22:39.13 | buddah | it shows it up through playing the message |
| 22:39.18 | buddah | then after 10 sec, it does a blast |
| 22:39.23 | glm2k | pygrammer: interesting alright. mp3 can be a bit of a load |
| 22:39.56 | *** join/#asterisk darkskiez (n=darkskie@host86-133-147-187.range86-133.btcentralplus.com) |
| 22:39.56 | buddah | http://pastebin.ca/21885 |
| 22:39.59 | buddah | thats what it looks like now |
| 22:40.09 | pygrammer | glm2k: is what i'm talking about "jitter"? |
| 22:40.11 | buddah | the brf context has all the extensions |
| 22:40.12 | pygrammer | it's more like "chop," isn't it? |
| 22:41.36 | *** join/#asterisk Samoied (n=Samoied@201.14.188.138) |
| 22:41.37 | *** join/#asterisk Mulvane (n=mulvane@c-24-129-34-230.hsd1.fl.comcast.net) |
| 22:42.28 | Mulvane | I'm looking for something to act as a sip phone proxy. When a number rings, it will ring every software/hardware phone attached to the proxy and act like a pots. What can do this? |
| 22:42.37 | Mulvane | asterisk or ser a better option? |
| 22:44.02 | *** join/#asterisk CoolAcid (n=jason@216.99.98.39) |
| 22:44.28 | pygrammer | glm2k: do you recommend that I just buy a hardphone and try that? |
| 22:44.41 | pygrammer | i just hope that works -- i'm looking for one anyway, but don't want to buy one and have the *exact* same problems |
| 22:44.46 | glm2k | true |
| 22:45.00 | glm2k | wait,sso yo're using softphones? |
| 22:46.00 | *** join/#asterisk ManxPower (i=eric@1Cust6600.an1.dfw28.da.uu.net) |
| 22:46.52 | *** join/#asterisk smcmahon (n=admin@digitaldatabits.net) |
| 22:46.56 | smcmahon | Anyone know anything about alarmreciever.c? |
| 22:47.07 | pygrammer | glm2k: yeah |
| 22:47.12 | pygrammer | iaxcomm right now |
| 22:47.28 | smcmahon | huh? |
| 22:48.08 | pygrammer | glm2k: using ulaw, but when i playback messages that i recorded on the voicemail to test it, it sounds bad -- not jittery really, just bad quality |
| 22:48.21 | bobbyd | sorry can someone just tell me if asterix is the software I'd use to do conference calling with voip clients? |
| 22:48.35 | smcmahon | It would work that way |
| 22:48.38 | smcmahon | bobbyd |
| 22:49.05 | bobbyd | smcmahon, thanks |
| 22:49.22 | smcmahon | I use Flash Operator Panel and Asterisk to do confrence calling ... FOP Allows you to drag callers and place calls into a meetme room. Rather neat |
| 22:50.02 | io_ | could someone please point me to some doc's about setting up /etc/zaptel.conf for use with ztdummy, not real Digium hardware? |
| 22:50.36 | smcmahon | Umm what kind of hardware you using? |
| 22:50.42 | PyroSteve | come on guys |
| 22:50.56 | PyroSteve | does anyone here use Nufone.net via SIP ? |
| 22:50.59 | smcmahon | io_ hello... |
| 22:50.59 | pygrammer | glm2k: yes, i am using a softphone |
| 22:51.03 | pygrammer | iaxcomm |
| 22:51.12 | *** join/#asterisk ManxPower (i=eric@1Cust6600.an1.dfw28.da.uu.net) |
| 22:51.13 | io_ | smcmahon : no hardware, i'm using the ztdummy module |
| 22:51.14 | smcmahon | iaxcomm sucks |
| 22:51.17 | pygrammer | and now, it doesn't skip as much, but it sounds sort of "staticky" |
| 22:51.30 | pygrammer | some noise when the voice starts talking |
| 22:51.30 | PyroSteve | i checked the wiki for the info i need but none is posted |
| 22:51.32 | smcmahon | so just load the drivers and dont specify a device?? |
| 22:51.37 | ManxPower | What is the name of the recent virus/worm that infected win2k and xp machines? The one that got all the press? |
| 22:51.40 | PyroSteve | i even checked the list archives |
| 22:51.45 | pygrammer | smcmahon: would you please explain why you think iaxcomm sucks? |
| 22:51.49 | smcmahon | I dont know but I blocked the virus |
| 22:52.01 | lesouvage | I have a problem with a speedtouch190 ata and a beacom 6000 decset connected to an asterisk. Everything is working fine except de dtmf signals for the voicemenu. Any suggestion? |
| 22:52.17 | pygrammer | smcmahon: i don't take things people say at face value and immediately heed their outbursts |
| 22:52.29 | io_ | smcmahon: "ztdummy is a Linux kernel module that will provide your Asterisk with a Zaptel timer even if you haven't got any Digium hardware installed in your Linux server." |
| 22:52.30 | smcmahon | Umm IAX sucks in general because of the sound quality when you have alot of people using it. Thats why my VOIP Provider has had us switch from IAX2 to SIP. |
| 22:52.36 | io_ | smcmahon : so no, no hardware |
| 22:52.55 | *** join/#asterisk funjon (i=jdisher@fawkes.parad.net) |
| 22:52.58 | pygrammer | smcmahon: can't speak for large volumes of calls at one time |
| 22:53.04 | pygrammer | but it seems nice for one or several clals |
| 22:53.05 | pygrammer | *calls |
| 22:53.29 | smcmahon | I really dont have time to answer in detail. |
| 22:53.43 | smcmahon | I like the IAX Feature of Trunking and that's pretty much it |
| 22:54.00 | glm2k | pygrammer: if the quality changes, like that "staticky" sound comes and goes, try looking for a condition on the server, a cron job perhaps? |
| 22:54.25 | *** join/#asterisk NirS (n=Nir@84.94.52.50.cable.012.net.il) |
| 22:54.33 | NirS | hey all |
| 22:54.40 | NirS | how is everybody feeling tonight ? |
| 22:55.38 | NirS | hello ? |
| 22:55.40 | NirS | anybody home ? |
| 22:55.43 | NirS | everybody's asleep ? |
| 22:55.53 | buddah | yup |
| 22:55.55 | buddah | all sleeping |
| 22:55.56 | pygrammer | glm2k: well, i shut down festival since i'm not using it now, and that freed up some memory (now using only 116 megs) |
| 22:56.01 | pygrammer | glm2k: didn't see any cron jobs doing |
| 22:56.02 | pygrammer | *going |
| 22:56.12 | pygrammer | i suppose i can shut down exim, too, but i might want to email voicemail |
| 22:56.15 | NirS | buddah, surely looks like it |
| 22:56.18 | buddah | hehe |
| 22:56.21 | NirS | or the network is extremely lagged |
| 22:56.28 | buddah | i'm ircing in my sleep |
| 22:56.34 | NirS | :) |
| 22:56.49 | NirS | I'm having some severe zaptel issues for some reason |
| 22:57.16 | bobbyd | smcmahon, would it be completely insane to try and run asterix on a pda? |
| 22:57.24 | funjon | bobbyd: uh, yes. |
| 22:57.32 | buddah | lol |
| 22:57.35 | bobbyd | smcmahon, I'm trying to work out how to conference call using voip and pda's |
| 22:57.50 | pygrammer | could shut down apache too |
| 22:58.11 | funjon | can someone give me a hand with a strange configuration issue? |
| 22:58.18 | NirS | funjon, shot |
| 22:59.25 | *** join/#asterisk Pkunk (n=Pkunkage@mbbs.munnabhai.info) |
| 22:59.40 | funjon | i'm still in the process of getting asterisk initially set up. i migrated my asterisk config from one machine to another, same asterisk ver, same OS ver, but for some reason, when i dial voicemail (8500), my phone immediately goes to "call ended", like it cant find that extension, but if i dial my phone's extension (1001), i get my extension's vm |
| 22:59.49 | funjon | i also cant dial 1000 for the demo |
| 22:59.53 | *** join/#asterisk wasim (n=wasim@pdpc/supporter/active/wasim) |
| 22:59.55 | funjon | this config worked fine on the old box |
| 23:00.14 | smcmahon | the demo sucks |
| 23:00.43 | funjon | yeah, but its how i verify my phones are working |
| 23:01.00 | NirS | hmmmm |
| 23:01.44 | *** join/#asterisk Tupau (i=Tupau@212.170.78.73) |
| 23:01.47 | NirS | funjon, did you try debugging the console messages |
| 23:01.54 | NirS | to find out what happens when you call a number ? |
| 23:02.12 | *** join/#asterisk Dabba (n=d@ipv6.mfnx.ip6net.net) |
| 23:02.27 | NirS | anyone has an idea why would zaptel generating the following ": Invalid module format" on kernel 2.6 (Mandriva) ? |
| 23:02.38 | *** part/#asterisk bobbyd (n=bobby@195.137.31.171) |
| 23:02.44 | *** part/#asterisk Tupau (i=Tupau@212.170.78.73) |
| 23:02.46 | funjon | NirS: how would I do so? |
| 23:03.00 | io_ | NirS : not that i can help you any, but are you using ztdummy ? |
| 23:03.16 | NirS | io, I'm trying to use ztdummy |
| 23:03.24 | io_ | me to |
| 23:03.30 | glm2k | NirS: how did you install zaptel? rpms? |
| 23:03.38 | io_ | what your zaptel.conf look like? |
| 23:03.40 | NirS | glm2k, tried both compiling and RPMS |
| 23:03.48 | Dabba | can anyone tell me why if $EXTEN is 019 both my variables are 19 |
| 23:03.50 | NirS | zaptel.conf is empty |
| 23:03.54 | Dabba | http://pastebin.ca/21894 |
| 23:03.58 | *** join/#asterisk kshumard (n=root@digium.com) |
| 23:04.05 | io_ | your using 2.6 right? did you do 'make linux26' instead of just 'make' ? |
| 23:04.14 | funjon | invalid module format sounds vaguely like its trying to use 2.4 modules in a 2.6 kernel |
| 23:04.18 | NirS | funjo, you need to add the DEBUG directive to /etc/asterisk/logger.con |
| 23:04.26 | NirS | then issue a logger reload in the CLI |
| 23:04.35 | NirS | once you make a call, you'll see more information |
| 23:04.52 | NirS | funjon, already been to that |
| 23:05.43 | NirS | io, yes, i'm using 2.6 |
| 23:05.48 | NirS | and I did compile with linux26 |
| 23:06.09 | NirS | funny thing is that even the Mandriva zaptel package doesn't modprobe, and ends with the same error |
| 23:06.40 | funjon | Sep 1 09:33:29 NOTICE[9108]: Unable to create channel of type 'SIP' |
| 23:06.43 | funjon | what the heck? |
| 23:06.53 | Dabba | us |
| 23:07.06 | Dabba | sorry |
| 23:07.18 | NirS | that means that your missing a sip.conf file |
| 23:07.30 | NirS | or that the chan_sip module is not loaded on your system |
| 23:07.47 | file | or the peer isn't registered |
| 23:07.57 | Dabba | http://pastebin.ca/21894 why oh why are both my variablse 19 when $EXTEN is 019 |
| 23:07.58 | NirS | that is an option too |
| 23:08.13 | JerJer | or sip cannot bind an IP |
| 23:08.20 | JerJer | thus not loaded |
| 23:08.30 | *** join/#asterisk cmurray (n=cmurray@zoidberg.digital-storm.net) |
| 23:08.35 | NirS | hey JJ |
| 23:08.43 | NirS | yes, that is a valid option too |
| 23:09.37 | funjon | yeah, but if i dial my own extension, i get the vm system |
| 23:09.40 | funjon | so its opening a sip connection |
| 23:10.03 | NirS | JJ, any experience with ZTdummy on 2.6 ? |
| 23:10.39 | *** join/#asterisk santiago (n=santiago@63.245.87.180) |
| 23:12.12 | lesouvage | What payload value for RFC2833 should I use configuring an ata to make it work. The current value is 96. |
| 23:12.25 | file | 101 |
| 23:12.34 | lesouvage | file: thanks again |
| 23:12.41 | file | yup yup |
| 23:13.31 | *** join/#asterisk Pussfeller_the_s (n=todd@12.150.129.171) |
| 23:14.23 | funjon | which is the better codec again, g729a or g723? |
| 23:15.26 | lesouvage | file: should this fix the dtmf problem I have with an ata and a connected phone? |
| 23:15.35 | file | maybe |
| 23:15.39 | file | try and see |
| 23:16.39 | funjon | http://pastebin.com/352354 is what i get when i try to make a call |
| 23:16.53 | CoffeeIV | can asterisk page someone when they have a new voicemail ? |
| 23:19.08 | Sedorox | yes |
| 23:19.12 | Nugget | sure. have you even looked at the voicemail.conf? |
| 23:19.24 | CoffeeIV | yes reading it now |
| 23:19.50 | file | bleh where's Hmmhesays... |
| 23:20.14 | pygrammer | i'm on the phone listening to on-hold music for newegg tech support |
| 23:20.26 | pygrammer | the music keeps skipping -- it's not terrible, but it's annoying |
| 23:20.28 | file | lol |
| 23:20.34 | pygrammer | i'd just assume use a land-line at this point |
| 23:20.45 | pygrammer | and i just can't figure out why the hell it's skipping |
| 23:21.02 | *** join/#asterisk bkw__ (n=brian@adsl-69-155-21-122.dsl.tulsok.swbell.net) |
| 23:21.02 | *** mode/#asterisk [+o bkw__] by ChanServ |
| 23:21.15 | pygrammer | maybe a hardphone would be better ... ? |
| 23:21.29 | pygrammer | but my uncle at SAIC uses a softphone (nortel) and he says it works great |
| 23:21.40 | Vco | that would be a safe wager |
| 23:21.50 | Vco | re: hardphone |
| 23:23.25 | Vco | think of hardware mpg decoder vs software decode back in the day....or winmodem vs hardware modem... |
| 23:23.54 | *** join/#asterisk meriad (n=parsec@snoid.gfed.net) |
| 23:23.54 | Vco | sure..one of the 2 would work fine...the other invariably would work better however... |
| 23:24.09 | pygrammer | Vco: yes, but why does my uncle's softphone work so well? |
| 23:24.12 | pygrammer | of course, maybe it's windows |
| 23:24.14 | pygrammer | i don't know |
| 23:24.40 | Vco | "he say's it works great.." |
| 23:24.47 | Vco | did you try it? |
| 23:24.50 | pygrammer | yes, and i talk to him on it |
| 23:24.52 | meriad | what is PBX? |
| 23:25.03 | pygrammer | he sounds like he's on a regular phone |
| 23:25.06 | pygrammer | i don't know how i sound on the other end |
| 23:25.17 | pygrammer | but i know the person on the other end sounds like we're both on cell phones |
| 23:25.30 | pygrammer | definitely not up to land-line quality -- at least now |
| 23:25.33 | glm2k | buddah: you still around? |
| 23:25.40 | Vco | are you usign the same client as yoru uncle? |
| 23:25.43 | Vco | your |
| 23:25.48 | pygrammer | Vco: no -- i'm on linux and using OSS stuff |
| 23:25.53 | glm2k | heh. |
| 23:25.53 | Vco | well.. |
| 23:26.05 | Vco | lets take this apple and this orange............ |
| 23:26.07 | glm2k | you should get better reception! lol |
| 23:26.13 | pygrammer | then why the hell would anyone use softphones?!?! |
| 23:26.15 | pygrammer | they fucking suck |
| 23:26.19 | pygrammer | they sound terrible |
| 23:26.21 | pygrammer | or at least the oss ones |
| 23:26.32 | Vco | no.. |
| 23:26.39 | Vco | you had it right the first time.. |
| 23:26.39 | glm2k | your milage may vary as they say |
| 23:26.56 | pygrammer | Vco: no what? |
| 23:26.59 | pygrammer | had what right? |
| 23:27.01 | Vco | they suck |
| 23:27.03 | pygrammer | oh |
| 23:27.06 | pygrammer | heh |
| 23:27.14 | Vco | "generally speaking" |
| 23:27.16 | funjon | because if you're travelling, it's sometimes better than carrying a full hardphone |
| 23:27.25 | Vco | sure... |
| 23:27.31 | pygrammer | glm2k: ymmv? it's probably more a difference of opinions -- some people could care less about quality, but i for one don't like hearing crackling and distortion while i'm conversing |
| 23:27.37 | Vco | but don't expect a great quality phone.. |
| 23:27.46 | funjon | of course not |
| 23:27.53 | Vco | i mean shit...look at the soap you use in the hotel..... |
| 23:27.56 | funjon | but if im in the datacenter in Frankfurt, and I need to call someone here in SF in the office... |
| 23:27.59 | Vco | ;) |
| 23:28.01 | glm2k | pygrammer: true, true |
| 23:28.05 | funjon | a softphone and a headset is better than dragging a hardphne with me |
| 23:28.09 | pygrammer | i don't know if you all have the same experience, or if it's my config, or what... i'd like to try someone else's system to see |
| 23:28.14 | glm2k | pygrammer: hence my earlier statement. |
| 23:28.17 | darkskiez | get one of em there sip wifi phones! |
| 23:28.24 | funjon | hah |
| 23:28.28 | glm2k | lol |
| 23:28.29 | funjon | i wish |
| 23:28.35 | Vco | $169 isn't bad |
| 23:28.38 | funjon | the two hardphones i have now are sipura spa841's |
| 23:28.48 | pygrammer | so -- i'm guessing *none* of you primarily use softhpones? |
| 23:28.49 | funjon | cheapest i could find because people didnt want to spring for spendy stuff |
| 23:28.50 | pygrammer | *softphones |
| 23:28.57 | Katty | beep |
| 23:29.04 | darkskiez | Is there any hardphones that have a built-in ATA, so you can attach a modem ? |
| 23:29.08 | glm2k | not primarily here. but i did use them for testing |
| 23:29.33 | pygrammer | glm2k: how can you test with them? you don't get an accurate "picture" of your system, since the quality and shit is so bad |
| 23:29.34 | glm2k | heck, when i was still using gnugk and h.323, i used a command line softphone. worked well. |
| 23:29.42 | funjon | i'm thinking using a modem on an ip pbx would perhaps be troublesome |
| 23:29.43 | Faithful | darkskiez: what are you saying? |
| 23:29.59 | Vco | modem works fine here |
| 23:30.06 | Faithful | some/most IP phones support PPPoE |
| 23:30.09 | io_ | any recomendations on a softphone for testing? |
| 23:30.16 | shido6 | xlite |
| 23:30.17 | shido6 | iaxcom |
| 23:30.24 | Vco | SKYPE! |
| 23:30.27 | lesouvage | io: gizmo works fine |
| 23:30.30 | glm2k | pygrammer: my stand was: if i could get pretty good (ie, bad cellphone/landline) quality with a softphone, i could do better with a hardphone. |
| 23:30.36 | darkskiez | Faithful: I need a hardphone that has an ata built in, so you could attach an analogue phone or modem into it |
| 23:30.41 | glm2k | or in my case, i bought a ton of adapters then. |
| 23:30.54 | funjon | can someone give me some sample configs that have, say, a couple working sip phones and working zaptel channels for dialout? |
| 23:30.56 | pygrammer | Vco: actually, skype is the exception for softphones, it seems |
| 23:31.01 | pygrammer | the quality is amazing for a softphone |
| 23:31.06 | funjon | skype is pretty damn good |
| 23:31.12 | pygrammer | all this other shit sucks |
| 23:31.15 | glm2k | because it's proprietary |
| 23:31.17 | io_ | thanks! |
| 23:31.17 | darkskiez | Faithful: think hotel ip pbx deployment, with guests who want to dial into their work |
| 23:31.21 | Faithful | darkskiez: an ATA converts a normal phone to an IP phone |
| 23:31.27 | Vco | i dont' give a rat ass....is still some propriety crap...free or not |
| 23:31.37 | pygrammer | glm2k: so how do your hardphones sound? as good as/better than a land-line most of the time? |
| 23:31.43 | lesouvage | darskiez: I have a snom 200 with an extra ethernet port you can use to attach an ata and then attach a analog phone. |
| 23:32.00 | glm2k | better. i'm using g.711 so i technically have landline quality. |
| 23:32.07 | Faithful | darkskiez: some have failover lines |
| 23:32.08 | lesouvage | darskiez: or just connect a pc, where it is normally used for. |
| 23:32.11 | pygrammer | isn't 711 better than land-line? |
| 23:32.24 | glm2k | quality varies with latency. |
| 23:32.35 | Vco | OSS OSS! Proprietary is evil! Proprietary is evil! oh...it's free...thats ok... |
| 23:32.38 | glm2k | landlines use g.711, or at least i've read the original phone lines did |
| 23:32.43 | darkskiez | faithful: you're not getting my requirements, similar to what lesouvage was suggesting is what I want, but all in one unit. |
| 23:32.48 | pygrammer | pj |
| 23:32.49 | pygrammer | *oh |
| 23:33.03 | pygrammer | glm2k: btw, my latency is great to voxee -- average 23.5 ms |
| 23:33.09 | NirS | well, time for bed now |
| 23:33.11 | pygrammer | so it isn't latency that's causing this distortion |
| 23:33.13 | NirS | c'ya all tomorrow |
| 23:33.16 | glm2k | yep |
| 23:33.24 | pygrammer | NirS: night. |
| 23:33.27 | lesouvage | darskiez: the difium ata looks very smll on picuters |
| 23:33.31 | Vco | what spec system are you running the client on? |
| 23:33.32 | *** join/#asterisk Ariel_ (n=Ariel@dsl-20-177.cofs.net) |
| 23:33.33 | pygrammer | glm2k: could be RAM -- don't know |
| 23:33.35 | lesouvage | difium is digium |
| 23:33.49 | pygrammer | Vco: p4 1.6/256 megs of RAM (an additional stick is coming tuesday) |
| 23:34.02 | Faithful | darkskiez: I don't think I have ever seen such a gadget especially for the fact that basically no VoIP solution except for ILBC is going to function well over a dial up modem. |
| 23:34.12 | pygrammer | glm2k: that is, ram on * server |
| 23:34.13 | glm2k | heh, better than my pIII 750 |
| 23:34.16 | Faithful | darkskiez: so no one provides it. |
| 23:34.22 | glm2k | and i've been using this for testing for ages. |
| 23:34.25 | lesouvage | I'm going to bed I'm loosing control over my keyboard |
| 23:34.30 | pygrammer | are there any softphones that support 729? |
| 23:34.32 | darkskiez | Faithful: i dodnt want to put voip over a dialup! |
| 23:34.34 | pygrammer | iaxcomm doesn't |
| 23:34.43 | pygrammer | xlite doesnt |
| 23:34.44 | darkskiez | Faithful: I want to put dialup over a voip |
| 23:34.58 | glm2k | hmmm, g.711 should be the best way to go since it has the least cpu overhead. |
| 23:34.59 | Vco | i think you want to do that the other way around |
| 23:35.08 | pygrammer | glm2k: oh |
| 23:35.09 | Faithful | darkskiez: then any old ATA does that just about |
| 23:35.18 | *** part/#asterisk CoffeeIV (n=CoffeeIV@mail.airlinksystems.com) |
| 23:35.41 | darkskiez | Faithful: yes, I know, thus my original question: Is there any hard-phones with a built in ATA ? |
| 23:36.10 | glm2k | pygrammer: it costs you in bandwidth though because it isn't compressed. |
| 23:36.13 | glm2k | darkskiez: yes |
| 23:36.24 | pygrammer | glm2k: yeah |
| 23:36.25 | glm2k | voip-info should have a few documented |
| 23:36.29 | pygrammer | hence why i just want to konw |
| 23:36.31 | pygrammer | *know |
| 23:36.43 | pygrammer | if there are any 729 softphones just to test if it is bandwith, though i know it 99.9999% most likely isn't |
| 23:36.51 | pygrammer | just to confirm |
| 23:37.02 | Faithful | darkskiez: I haven't seen any |
| 23:39.19 | *** join/#asterisk darkskie1 (n=darkskie@host86-132-169-66.range86-132.btcentralplus.com) |
| 23:39.20 | Rowters | whats the var for the uniqid? ${uniqid}? |
| 23:39.37 | darkskie1 | Faithful: yes, I know, thus my original question: Is there any hard-phones with a built in ATA ? |
| 23:39.40 | darkskie1 | having an ip phone and a bit of cable and an ata and a power adapter for each is just not good |
| 23:39.46 | darkskie1 | requirements: 1 IP phone, POE, with analogue port to allow a modem to dial up over the IP infrastructure |
| 23:39.52 | puzzled | Rowters: it's spelled uniqueid so try that |
| 23:40.03 | Rowters | puzzled, thanks |
| 23:40.34 | puzzled | Rowters: realize that it must be activated in the asterisk-addons Makefile |
| 23:41.34 | *** join/#asterisk MikeJ[Laptop] (n=ircatjer@adsl-69-209-167-100.dsl.sfldmi.ameritech.net) |
| 23:41.46 | pygrammer | btw, glm2k, i'm curious: when you record a conversation via Monitor() in asterisk, should the quality be different from the softphone, i.e. what you would receive on a hardphone? |
| 23:42.22 | pygrammer | or would there be the same quality that's on the softphone, since it's recording the softphone's input, too |
| 23:42.29 | io_ | can you call sip:1000@domain.tld addresses with skype? |
| 23:42.31 | Faithful | darkskie1: well I would get one of the little ATAs like the grandstream 286 or 486 and buy a little corded handset ... I had one that was only about 4x3 cm with a flip lid on the buttons. |
| 23:42.38 | pygrammer | probably the same as the softphone, at least from the softphone to * |
| 23:42.51 | pygrammer | since it's still the softphone sending asterisk the input audio |
| 23:42.55 | pygrammer | i don't know though |
| 23:43.05 | Rowters | puzzled, know where on voip-info is the list of vars? |
| 23:43.14 | pygrammer | i don't know how asterisk's monitor function works |
| 23:43.19 | Rowters | puzzled, am searching.. |
| 23:43.29 | darkskie1 | faithful: this is for a hotel deployment, that isnt suitable. |
| 23:43.45 | puzzled | Rowters: in the src look in doc/README.variables |
| 23:43.58 | Rowters | ok tkans |
| 23:43.59 | Faithful | darkskie1: not for putting in you suitcase for travel. |
| 23:44.09 | Katty | hi. |
| 23:44.21 | DarthClue | evening Katty! |
| 23:44.30 | Katty | i've not been very sociable of late. |
| 23:44.42 | DarthClue | haha. i've not been very here as of late. |
| 23:44.45 | darkskie1 | Faithful: for installing in 800 rooms to provide guests with an telephone and datapoint that they can attach their laptops modem and dial into their work. |
| 23:44.58 | puzzled | Rowters: add this in the asterisk-addons Makefile after CFLAGS+=-D_GNU_SOURCE: CFLAGS+=-DMYSQL_LOGUNIQUEID |
| 23:45.09 | Vco | poly ip 301? |
| 23:45.13 | Katty | DarthClue: how's teh bus? |
| 23:45.29 | Rowters | puzzled, excelent thanks |
| 23:45.39 | darkskie1 | Faithful: ideally, a single ip phone, with power over ethernet, and an ata built in, is what i want. |
| 23:45.44 | DarthClue | Katty: compared to the world of computers, it's a vacation. |
| 23:45.46 | io_ | thanks for the help everyone |
| 23:45.57 | Vco | get grandstream to make one.. |
| 23:45.58 | DarthClue | a bit on the hot side, but that will change as soon as it cools off |
| 23:46.00 | Katty | DarthClue: :> |
| 23:46.05 | Vco | incidentaly get someone else to make the firmware |
| 23:46.40 | Faithful | darkskie1: you cannot dial a modem over an ata??? and why would you want to? If you are providing ethernet to each room... let them plug into that??? |
| 23:47.11 | darkskie1 | Faithful: because 80% of remote workers still use dial up rather than VPN. |
| 23:47.21 | darkskie1 | Source: Hilton Hotels |
| 23:47.28 | *** part/#asterisk puowvip (i=ircuser@thegrey.diamond.org) |
| 23:47.42 | Katty | DarthClue: i have lettuceses. |
| 23:47.54 | darkskie1 | Faithful: why cant you dial a modem over an ata? |
| 23:48.07 | Nugget | what? how are dialup and vpn exclusive? |
| 23:48.17 | DarthClue | lettuceses? as in multiple lettuce? um, k. |
| 23:48.39 | Faithful | darkskie1: because an ATA turns a standard socket into an IP phone |
| 23:48.43 | darkskie1 | Nugget: they arent, but dial up is HEAVILY used for remote-working. |
| 23:48.53 | Nugget | sure, and so is vpn. often on top of dial up. |
| 23:48.56 | Nugget | that's what confuses me. |
| 23:49.15 | Katty | DarthClue: ! |
| 23:49.16 | darkskie1 | Faithful: No, it turns and IP socket into a standard phoneline. |
| 23:49.22 | *** part/#asterisk meriad (n=parsec@snoid.gfed.net) |
| 23:49.25 | Katty | DarthClue: actually, that's what i had for lunch |
| 23:49.30 | Faithful | darkskie1: not so |
| 23:49.32 | Nugget | are you using "vpn" to mean "broadband or non-dialup connectivity"? |
| 23:49.42 | *** part/#asterisk cmurray (n=cmurray@zoidberg.digital-storm.net) |
| 23:49.48 | Faithful | darkskie1: not so at all |
| 23:49.58 | DarthClue | i might have it tomorrow. it's just around the corner from the bus lot so it makes it quick and cheap. |
| 23:50.12 | darkskie1 | Faithful: yes so, yes so indeed. |
| 23:50.19 | DarthClue | i wonder if i could get my hands on a limo or a bus for cluecon... |
| 23:50.57 | Faithful | darkskie1: no an ATA is a SIP phone adapter for a standard PSTN phone |
| 23:51.10 | darkskie1 | which is not what you said |
| 23:51.13 | Katty | DarthClue: i got my new laptop |
| 23:51.19 | Katty | DarthClue: tis a dual boot now |
| 23:51.33 | Katty | DarthClue: and none of all that crappy hp trial stuffs that clog it all up |
| 23:51.39 | Faithful | darkskie1: which is what I said... and what you think an ATA is is wrong. |
| 23:52.02 | darkskie1 | how is it wrong, ethernet in one side, phoneline out other |
| 23:52.17 | Nugget | I'm still waiting to hear how you can't use a vpn over dialup. |
| 23:52.31 | Faithful | darkskie1: it is not "ethernet on one side" |
| 23:52.51 | Faithful | darkskie1: it is "PSTN one side SIP the otherside" |
| 23:53.10 | glm2k | pygrammer: it should be as good. i admit, i haven't played around with monitor so i'm not familiar with it. |
| 23:53.20 | darkskie1 | Faithful: and whats to stop you plugging a modem into one ? |
| 23:53.33 | Faithful | darkskie1: or perhaps it is "PSTN one side IAX the otherside" |
| 23:53.43 | DarthClue | Katty: sweet. i'm looking around for a new one. i think mine finally gave up. |
| 23:53.58 | Katty | DarthClue: aww :< |
| 23:54.39 | DarthClue | Katty: oh well. it happens. |
| 23:54.46 | darkskie1 | Faithful: i know exactly what an ATA is and does, that is not the issue. |
| 23:55.05 | Katty | DarthClue: i'm going to probably need help to get my wireless working in debian |
| 23:55.15 | darkskie1 | Faithful: why cant you plug a modem into one ? |
| 23:55.16 | Katty | DarthClue: and some sort of dvd burning software stuffs. |
| 23:55.28 | Faithful | darkskie1: have you tried? |
| 23:55.30 | Katty | DarthClue: but i am on 3 irc servers...all of which are geeky channels. |
| 23:55.36 | Katty | DarthClue: i imagine i'll do fine (= |
| 23:55.43 | darkskie1 | Faithful: yes, and it works. |
| 23:55.48 | pygrammer | glm2k: but i mean -- even if i'm using a softphone? because i heard skipping and general lack of quality on the played-back conversation, even on the other side, so is that how it will sound should I buy a hardphone? |
| 23:55.57 | DarthClue | Katty: i was a firsthand witness to your true geekiness. I'm sure you'll do fine. |
| 23:56.09 | Faithful | darkskie1: so you are only doing VoIP internally? |
| 23:56.24 | Katty | DarthClue: indeed, i can put the proper shoe on teh proper foot. |
| 23:56.32 | Katty | DarthClue: i'm movin up in the world :> |
| 23:56.44 | asterisk_newbie | my asterisks is failing to connect to my sip proxie provider could it be a problem on my end or theirs |
| 23:56.49 | Katty | DarthClue: oh! i bought yoga pants today! |
| 23:56.51 | darkskie1 | Faithful: yes, but that is not of the concern. |
| 23:57.01 | DarthClue | Katty: um, i'm pretty sure you can even tie the shoe too. |
| 23:57.06 | Katty | DarthClue: and two incredibly soft shirts (((((= |
| 23:57.14 | Katty | soft++ |
| 23:57.50 | glm2k | pygrammer: that's the thing. if the degradation is server side, then a hardphone won't improve things much. |
| 23:57.58 | darkskie1 | Faithful: I need a phone with POE, and a built in ATA to allow modems to dial up over the IP infrastructure. |
| 23:58.36 | darkskie1 | Faithful: Justification: Many hotel guests need this facility, and providing a phone and an ata and a powersupply for each room is clumsy and messy and expensive. |
| 23:59.10 | DarthClue | darkskie1: just get a polycom and an ata, split the network as it comes into the room, one for the polycom and one for the ata. |
| 23:59.45 | Ariel_ | good evening everyone |
| 23:59.54 | DarthClue | evening Ariel_ |
| 23:59.58 | Nugget | I'd guess that 99.9999% of these hypothetical guests would prefer a phone and an ethernet jack. |
