| 00:00.23 | grey | heh, after the first few weeks, you don't need the script anymore, it's burned into your mind :P |
| 00:00.25 | CrashSys | linagee: Did you tell her that not only can you do all that, but you can retrofit her call-center with ViciDial for a fraction of the price of their proprietary ACD! |
| 00:00.52 | linagee | CrashSys: i told her that any of my lines are voice or fax just depending on what settings i enter. :) she was like, ??? |
| 00:01.02 | CrashSys | And that you can also do call campaigns... and call her... and ask her if she would like to buy DID's! |
| 00:01.13 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 00:01.16 | linagee | CrashSys: "and i can have an unlimited number of voice mailboxes. and an unlimited number of call conference rooms" |
| 00:01.31 | CrashSys | That is not a selling point... |
| 00:01.36 | CrashSys | I dont want any voicemail boxes |
| 00:01.40 | CrashSys | it's like e-mail |
| 00:01.44 | linagee | CrashSys: ??? some companies would actually charge you for that though. lol |
| 00:01.46 | CrashSys | or SMS |
| 00:01.57 | CrashSys | it can die a slow horrible death... |
| 00:02.03 | CrashSys | which e-mail has been |
| 00:02.06 | linagee | CrashSys: like charge you based on the number of voicemail addresses you have. lol |
| 00:02.47 | linagee | CrashSys: she was threatening to send out an ATA and she didn't even know it was called an ATA! :( |
| 00:02.49 | CrashSys | If the terrorists want to deal a deadly blow to america... and really piss us off... all they need to do is destroy the myspace datacenters... |
| 00:02.52 | CrashSys | that would work... |
| 00:03.00 | linagee | CrashSys: "sir, can i send out a phone adapter for free for 30 days?" |
| 00:03.09 | linagee | CrashSys: i was like, "an ATA?" hehehe |
| 00:03.28 | linagee | CrashSys: "will that work with my existing hard phone?" lol. :) |
| 00:03.50 | linagee | CrashSys: ??? i think that would be great if myspace were zapped to /dev/null |
| 00:03.55 | CrashSys | See... I just go "Fuck-Off" and hang up.... |
| 00:04.00 | linagee | CrashSys: lol. |
| 00:04.08 | linagee | CrashSys: she actually called me back! lol |
| 00:04.27 | linagee | CrashSys: damned retentive SOB. :) |
| 00:04.40 | linagee | "SIR! you have to buy this NOW!!!" (just kidding) lol |
| 00:05.01 | CrashSys | Yup... I dont have time to waste with them doing outbound call campaigns... |
| 00:05.38 | iBuMp | so whens the asterisk-gui going to be ready to roll?? |
| 00:05.41 | CrashSys | The return is only like 1% anyways... |
| 00:05.43 | iBuMp | semi solid? |
| 00:05.51 | linagee | CrashSys: it's funny when someone comes to you with the argument of "we are cheaper" when they're not. ;) |
| 00:06.23 | CrashSys | I never get that far... I answer... they say "Can I talk to the person in charge of <script>"... I say "Fuck-Off" and hang up... |
| 00:06.27 | CrashSys | conversation over... |
| 00:07.56 | rob0 | I don't say anything, just hang up. But thatnks to * I rarely get telemarketers now. :) |
| 00:08.05 | linagee | CrashSys: lol. then you find out it's a client. hehehe |
| 00:08.24 | linagee | CrashSys: of course in my case i'd know by the wacky area code. :) |
| 00:08.34 | CrashSys | They learn quick not to talk like a telemarketer |
| 00:08.55 | linagee | CrashSys: sir! i'm not reading from a script! heh |
| 00:09.04 | linagee | CrashSys: i need to put that on youtube or something. :) |
| 00:10.38 | CrashSys | Isn't there like a spews.org for telemarketer CID's? |
| 00:10.48 | linagee | CrashSys: ? interesting |
| 00:10.55 | coppice | you'd think telemarketers would learn to start a conversation in a way that makes it less obvious they are telemarketers |
| 00:11.07 | linagee | CrashSys: here's the one that was sent: 605-299-4105 |
| 00:11.13 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 00:11.13 | linagee | hahaha. USA CID. riiight. heh |
| 00:11.15 | CrashSys | Write a module to load that into my dial-plan... if cid matches telemarketer blacklist, hangup... :) |
| 00:11.30 | linagee | yes |
| 00:12.04 | linagee | CrashSys: "hello. you've called using a blacklisted number. if this is a real phone call and not a sales call, please email spam-filtered@mydomain.com" |
| 00:12.19 | CrashSys | That is potentially a selling feature to my clients... "Comes with a pre-installed list of known telemarketer call centers"... |
| 00:12.27 | linagee | CrashSys: lol. :) |
| 00:12.46 | linagee | CrashSys: how will other companies sell them stuff? heh |
| 00:12.50 | CrashSys | Make it update monthly... |
| 00:13.01 | linagee | CrashSys: ??? |
| 00:13.11 | CrashSys | Not my problem... telemarketers are the spam of the phone... |
| 00:13.14 | linagee | CrashSys: make it update when someone comes out there and updates it. (for an update service charge. :) ) |
| 00:13.48 | CrashSys | Blah... I dont like service calls... |
| 00:13.53 | linagee | no? |
| 00:13.58 | CrashSys | My goal is to install the thing and then never see it again... |
| 00:14.02 | linagee | heheheh |
| 00:14.15 | linagee | CrashSys: outsource your service calls to me. :) |
| 00:14.23 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 00:16.38 | CrashSys | Maybe it's bad business... but I dont believe in selling and installing something that will require them to have me come out every month so that I can hit a "Magic Button" and charge them $50... |
| 00:16.46 | CrashSys | I'd rather charge them when things blow up... |
| 00:18.20 | CrashSys | So... A good install is one where I install it, and dont see it again for a year or two... |
| 00:18.37 | CrashSys | Unless they rapidly outgrow it or reconfigure every month... |
| 00:19.37 | CrashSys | Although it usually goes more like Install it, tune the dial-plan for them for a month, then dont see it again for 6-months to a year... |
| 00:25.05 | KDan | 01:12 linagee > CrashSys: "hello. you've called using a blacklisted number. if this is a real phone call and not a sales call, please email spam-filtered@mydomain.com" << isn't there an * feature which already allows you to filter most of them out by disconnecting the line for a fraction of a second? |
| 00:26.28 | CrashSys | if it works |
| 00:26.39 | linagee | or you just dropped your caller. lol |
| 00:28.10 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 00:29.47 | KDan | hmm, ok, so asterisk cannot follow sym links to find its config files? |
| 00:31.06 | *** part/#asterisk BSD_Tech[laptop] (n=BSDTech@adsl-69-230-169-41.dsl.irvnca.pacbell.net) |
| 00:34.38 | CrashSys | Hmm... compiled/installed zap/asterisk... still get these errors... http://www.pastebin.ca/562963 |
| 00:38.56 | *** join/#asterisk techie (n=gus@antibala.net) |
| 00:42.51 | KDan | aah, no, it just needs to have read access to its own config files |
| 00:42.54 | KDan | oh well, fair enough |
| 00:46.30 | CrashSys | hmmm? |
| 00:46.56 | KDan | re: my earlier question |
| 00:47.01 | CrashSys | ok |
| 00:50.51 | *** join/#asterisk _VoiceMeUp_COM (n=_VoiceMe@modemcable159.131-56-74.mc.videotron.ca) |
| 01:01.10 | *** join/#asterisk BZBW (n=wlwzhang@ip67-153-142-110.z142-153-67.customer.algx.net) |
| 01:02.16 | *** join/#asterisk TUplink_ (n=TUplink@c-24-126-39-22.hsd1.wv.comcast.net) |
| 01:02.25 | TUplink_ | i have a prob |
| 01:02.34 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 01:02.36 | TUplink_ | [Jun 12 21:00:57] WARNING[473]: chan_sip.c:8126 check_auth: username mismatch, have <20009>, digest has <20001> |
| 01:02.48 | _VoiceMeUp_COM | yeah found some problem here too |
| 01:02.57 | TUplink_ | really |
| 01:02.59 | _VoiceMeUp_COM | some times something coruppst the db i think |
| 01:03.00 | *** part/#asterisk BZBW (n=wlwzhang@ip67-153-142-110.z142-153-67.customer.algx.net) |
| 01:03.01 | TUplink_ | you fix it yet? |
| 01:03.11 | _VoiceMeUp_COM | i jsut got a pap2 doing a register |
| 01:03.19 | _VoiceMeUp_COM | and right afdter that never worked again |
| 01:03.23 | _VoiceMeUp_COM | had to flush it all |
| 01:03.26 | TUplink_ | http://bugs.digium.com/view.php?id=9678 |
| 01:03.49 | _VoiceMeUp_COM | ah hehe |
| 01:03.51 | _VoiceMeUp_COM | not same |
| 01:04.08 | TUplink_ | no.. thats just what i found |
| 01:04.26 | TUplink_ | has the same error in it... just not all of the stuff is exact |
| 01:05.07 | *** join/#asterisk BZBW (n=wlwzhang@ip67-153-142-110.z142-153-67.customer.algx.net) |
| 01:05.17 | TUplink_ | but you have the same prob? |
| 01:05.48 | _VoiceMeUp_COM | looks like same but not |
| 01:05.55 | _VoiceMeUp_COM | its another i guess no idea it fixed it self |
| 01:06.20 | TUplink_ | hum.... |
| 01:06.36 | TUplink_ | well im using an AT&T one that i hacked |
| 01:06.45 | TUplink_ | kindof... :P |
| 01:08.48 | TUplink_ | i think i fixed it |
| 01:08.57 | TUplink_ | added insecure=very is that a bad solution? |
| 01:09.05 | _VoiceMeUp_COM | nope |
| 01:18.25 | *** join/#asterisk bintut (n=bintut@203.125.63.150) |
| 01:20.56 | DrukenHME | anyone know what a plant test is? |
| 01:21.29 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 01:25.26 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 01:26.42 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 01:34.05 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 01:34.05 | *** mode/#asterisk [+o anthm] by ChanServ |
| 01:38.34 | mrdigital-work | hey _VoiceMeUp_COM? |
| 01:38.55 | DrukenHME | fucken nosy bell canada |
| 01:39.18 | DrukenHME | they broke into my voip line today and tried to figure out what number it was |
| 01:39.27 | mrdigital-work | who did? |
| 01:39.39 | DrukenHME | bell canada |
| 01:39.42 | mrdigital-work | why? |
| 01:39.50 | DrukenHME | while installing a line today |
| 01:40.20 | mrdigital-work | DrukenHME: you pretty good in linux? |
| 01:40.30 | DrukenHME | define pretty good |
| 01:41.13 | DrukenHME | er, how about... what are ya trying todo? or what do you need help with |
| 01:41.59 | mrdigital-work | its not really asterisk related |
| 01:42.06 | mrdigital-work | but its stressing me the hell out |
| 01:42.17 | mrdigital-work | im trying to configure FTP |
| 01:42.21 | *** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox) |
| 01:42.21 | mrdigital-work | but it will not connect |
| 01:42.38 | _VoiceMeUp_COM | yes digital im here |
| 01:42.51 | mrdigital-work | hey man long time no see! you got time for a PM Chat? |
| 01:44.33 | rob0 | In Europe, it's AM now. |
| 01:48.24 | *** join/#asterisk Tond (n=t@CPE0014bf30c190-CM0010954a5ffa.cpe.net.cable.rogers.com) |
| 01:50.15 | *** join/#asterisk fbffff (n=fbffff@adsl-68-20-22-160.dsl.chcgil.ameritech.net) |
| 01:56.00 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 02:06.19 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 02:12.23 | Mahmoud | why doesn't aterisk authenticate unregistering IAX2 peers? |
| 02:12.33 | *** join/#asterisk iBuMp (n=ReDNeQ@cpe-66-68-37-190.austin.res.rr.com) |
| 02:12.48 | Mahmoud | it just accepts it without sending regauth |
| 02:13.14 | *** join/#asterisk iBuMp (n=ReDNeQ@cpe-66-68-37-190.austin.res.rr.com) |
| 02:13.35 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 02:14.26 | Mahmoud | The IAX2 rfc draft says: A registrant MAY also force an expiration in the registrar by sending the REGREL message. This message may be challenged with REGAUTH or if sufficient credentials were included, it will be accepted with REGACK. In response to a REGAUTH, a REGREL message SHOULD be resent using the specified credentials. |
| 02:15.05 | Mahmoud | as I sinffed with wireshark, my IAX2 client sends a REGREL message, with my username only |
| 02:15.44 | Mahmoud | it seems it is possible to unregister massive peers, by simply knowing their username, causing DoS |
| 02:20.31 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 02:22.04 | *** join/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net) |
| 02:22.04 | *** mode/#asterisk [+o mog] by ChanServ |
| 02:28.28 | *** join/#asterisk johann8384_home (n=johann83@71-81-221-188.dhcp.stls.mo.charter.com) |
| 02:31.40 | *** join/#asterisk TUplink_ (n=TUplink@c-24-126-39-22.hsd1.wv.comcast.net) |
| 02:32.49 | flenders | hey, can I create a dialplan like this: exten => _001155XX9XXXXXXX,1,Dial(blah)? |
| 02:33.20 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 02:33.50 | TUplink_ | dont see why not Flenders |
| 02:35.11 | flenders | I never used a '9' in the middle of the 'template' |
| 02:36.32 | TUplink_ | [Jun 12 22:19:05] WARNING[5216]: chan_sip.c:8126 check_auth: username mismatch, have <20009>, digest has <20001> |
| 02:36.32 | TUplink_ | [Jun 12 22:19:05] NOTICE[5216]: chan_sip.c:13386 handle_request_invite: Failed to authenticate user "Tommy Huff"<sip:20001@24.126.39.22>;tag=187e2716-13c4-466f622c-3bf38c-3c9e |
| 02:36.59 | TUplink_ | the insecure=verry does not keep the corect Callerid |
| 02:37.25 | Nugget | if you're spelling it "verry" it's not doing anything. :) |
| 02:38.12 | *** join/#asterisk guillote_GNU (n=guillote@190.7.23.192) |
| 02:38.16 | TUplink_ | i know |
| 02:38.38 | TUplink_ | i spelt it rite. cause it let me call |
| 02:38.41 | TUplink_ | i guess |
| 02:39.13 | TUplink_ | insecure=very |
| 02:42.40 | *** join/#asterisk lee_is_me (n=chatzill@12-201-102-196.client.mchsi.com) |
| 02:43.13 | lee_is_me | when a product for asterisk is offered "per channel", that means two licenses per phone call, correct? |
| 02:45.32 | ManxPower | lee_is_me: yes |
| 02:45.46 | ReDNeQ | what |
| 02:45.54 | ReDNeQ | i was sold by digium 1 license per analog port |
| 02:46.01 | ReDNeQ | thats what they asked |
| 02:46.11 | ReDNeQ | and that is what they sold me |
| 02:46.23 | ManxPower | ReDNeQ: different kind of license |
| 02:46.30 | lee_is_me | I was looking at http://www.octasic.com/en/octware/applications/softecho_asterisk.php |
| 02:46.36 | ReDNeQ | ahhh |
| 02:46.37 | ReDNeQ | <PROTECTED> |
| 02:46.38 | ReDNeQ | '_ |
| 02:46.55 | lee_is_me | A sangoma card I got for my customer comes with a 6 channel license for that echo cancelling software |
| 02:47.02 | ManxPower | lee_is_me: That would be 2 licenses if both legs of the call were analog |
| 02:47.18 | lee_is_me | the sangoma is giving terrible echo problems |
| 02:47.21 | lee_is_me | LOL |
| 02:47.27 | lee_is_me | your right |
| 02:48.03 | *** join/#asterisk iBuMp (n=ReDNeQ@cpe-66-68-37-190.austin.res.rr.com) |
| 02:48.26 | lee_is_me | the Octware EC is supposed to recognized my sangoma card, but it doesn't so I'll be emailing sangoma in a few minutes here |
| 02:48.48 | lee_is_me | First time I've used it vs. TDM400's |
| 02:49.01 | lee_is_me | I'm very unhappy with it :( |
| 02:49.04 | lee_is_me | LOL |
| 02:49.31 | lee_is_me | Sorry, that should have been *not* very happy |
| 02:49.55 | lee_is_me | but we'll see when that Octware EC is running...maybe make a difference |
| 02:51.26 | lee_is_me | ManxPower: so if one leg is analog (from POTS) and the other is sip (internal device) then it only one license... |
| 02:52.05 | *** join/#asterisk aptura (n=sales@bdsl.66.15.70.176.gte.net) |
| 02:54.23 | *** part/#asterisk TUplink_ (n=TUplink@c-24-126-39-22.hsd1.wv.comcast.net) |
| 03:07.24 | *** part/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net) |
| 03:12.31 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 03:12.36 | *** join/#asterisk wallyhts (n=davidwal@74-129-108-204.dhcp.insightbb.com) |
| 03:13.32 | wallyhts | If any one is looking for a New Job as A Asterisk Dev PM me |
| 03:19.52 | *** join/#asterisk fbffff (n=fbffff@dsl081-143-082.chi1.dsl.speakeasy.net) |
| 03:21.20 | Nugget | I want a job picking on file. Does Digium have an opening in that department? |
| 03:21.27 | Qwell | yep |
| 03:21.33 | Nugget | Yay |
| 03:25.05 | aptura | :) |
| 03:26.28 | *** join/#asterisk bonderponder (n=test@201.199.68.150) |
| 03:27.58 | bonderponder | hello, anybody can help to configure a DIGIUM TDM2400p with hw echo cancel ? |
| 03:28.33 | *** join/#asterisk florz (i=nobody@2001:1a50:503c:0:0:0:0:1) |
| 03:31.11 | bonderponder | <PROTECTED> |
| 03:35.55 | *** join/#asterisk n00dle (n=ccraft@ip-249-27.springsips.com) |
| 03:35.58 | snuffy22 | not me ;( |
| 03:36.05 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 03:36.59 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 03:37.05 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 03:38.38 | *** join/#asterisk saftsack (n=oliver@p54A7E8E4.dip.t-dialin.net) |
| 03:46.41 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 03:50.11 | *** join/#asterisk Vorondil (n=vorondil@unaffiliated/vorondil) |
| 03:52.17 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 03:52.42 | *** join/#asterisk bmg505 (n=leon@196.209.183.67) |
| 03:56.42 | *** part/#asterisk wallyhts (n=davidwal@74-129-108-204.dhcp.insightbb.com) |
| 03:59.33 | *** join/#asterisk saftsack (n=oliver@p54A7C0AB.dip.t-dialin.net) |
| 04:01.16 | *** join/#asterisk nwhit (n=nwhit@67.90.83.14) |
| 04:01.38 | nwhit | I am having some issues with SLA .. anyone here can help? |
| 04:02.28 | nwhit | Outbound is working great, but I can't get the incoming to pickup the lines correctly. It rings in on the correct extension, but I can't pick it up without some trouble |
| 04:05.28 | *** join/#asterisk [hC] (n=hardcore@190.10.13.145) |
| 04:10.17 | *** join/#asterisk saftsack (n=oliver@p54A7CE67.dip.t-dialin.net) |
| 04:11.11 | *** join/#asterisk Winkie (i=sd@87-194-8-125.bethere.co.uk) |
| 04:24.13 | *** join/#asterisk Cabal_ (n=Cabal@unaffiliated/cabal) |
| 04:31.56 | *** join/#asterisk saftsack (n=oliver@p54A7F472.dip.t-dialin.net) |
| 04:40.22 | *** join/#asterisk Marshall- (n=Marshall@cpe-76-181-117-192.columbus.res.rr.com) |
| 04:41.25 | *** join/#asterisk bintut (n=bintut@203.125.63.150) |
| 04:41.29 | *** join/#asterisk john-eman0n (n=eman0n@cpe-76-181-117-192.columbus.res.rr.com) |
| 04:44.08 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 04:44.35 | *** join/#asterisk matsk (i=matsk@h110n2fls32o882.telia.com) |
| 04:46.44 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 05:04.08 | *** join/#asterisk florz (i=nobody@2001:1a50:503c:0:0:0:0:1) |
| 05:04.47 | *** part/#asterisk elg (n=fugalh@216.31.27.110) |
| 05:12.15 | *** join/#asterisk tr2x (n=alvar@80-218-185-55.dclient.hispeed.ch) |
| 05:14.56 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 05:30.56 | *** join/#asterisk [hC] (n=hardcore@190.10.13.145) |
| 05:31.07 | *** join/#asterisk grndslm (n=grndslm@24-116-87-97.cpe.cableone.net) |
| 05:39.44 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 05:42.51 | *** join/#asterisk [hC] (n=hardcore@190.10.13.145) |
| 05:45.16 | *** join/#asterisk mosty (i=mostyn@60-241-198-194.static.tpgi.com.au) |
| 05:55.04 | *** join/#asterisk reber (n=reber@cl-158.dub-01.ie.sixxs.net) |
| 05:57.44 | mosty | i am trying to configure an E1 card, when i try to dial it says there are no active d-channels, what could be wrong? the telco tech is able to make calls using the line, but my asterisk box cant |
| 05:58.10 | mosty | pri show span 1, shows this status line: Status: Provisioned, Down, Active |
| 05:58.52 | kaldemar | status should say it's up. |
| 05:59.39 | mosty | kaldemar, what could be wrong that would make it down? |
| 05:59.42 | *** join/#asterisk [[blah]asfd (n=ckwall@c-71-195-196-198.hsd1.ut.comcast.net) |
| 05:59.48 | kaldemar | check and double check your configuration. spans, channels and signaling in zaptel.conf and switchtype, signaling etc. in zapata.conf. |
| 05:59.58 | mosty | i have already |
| 06:00.05 | kaldemar | a zillion things could be wrong. |
| 06:00.16 | kaldemar | which card do you have? |
| 06:00.17 | [[blah]asfd | i had a request from a customer who wants me to take a streaming media link and use it for music on hold. its an mms:// url. is this possible to do? |
| 06:01.01 | mosty | kaldemar, digium te412p |
| 06:01.20 | kaldemar | mosty: do you have a green led for the port you're using? |
| 06:01.24 | mosty | yes |
| 06:01.49 | mosty | kaldemar, the d-channel is down when i try to make a call |
| 06:02.11 | kaldemar | then you obviously have something wrong with your configuration. |
| 06:02.47 | kaldemar | pastebin your zaptel.conf and that span's lines in zapata.conf |
| 06:03.38 | *** part/#asterisk [[blah]asfd (n=ckwall@c-71-195-196-198.hsd1.ut.comcast.net) |
| 06:04.30 | mosty | kaldemar, http://pastebin.ca/563557 |
| 06:04.55 | *** join/#asterisk steliosk (n=Stelios@62.169.217.209) |
| 06:05.22 | *** join/#asterisk Jabroni (n=Jabroni@red-corp-200.76.249.142.telnor.net) |
| 06:06.28 | *** join/#asterisk inv_arp[work] (n=junya@c-71-229-122-61.hsd1.fl.comcast.net) |
| 06:07.30 | mosty | kaldemar, and with output from the console: http://pastebin.ca/563562 |
| 06:08.17 | mosty | can anyone see anything wrong? |
| 06:09.22 | kaldemar | set some debug on the span with "pri intense debug span 1" and dial again. pastebin the output again. |
| 06:09.56 | mosty | ok, it will take me a minute to setup logging so i can capture all that |
| 06:12.01 | *** join/#asterisk stimpie (n=michiel@ip565faf27.direct-adsl.nl) |
| 06:13.24 | mosty | kaldemar, http://pastebin.ca/563571 |
| 06:13.35 | *** join/#asterisk webman (n=adamg@gw1.websitemanagers.com.au) |
| 06:15.09 | webman | Am I the only one experiencing constant lockups on the current 1.4 SVN branch of asterisk ??? Ever since updating last week, (and again today) I've had very frequent lockups on a very light system :( |
| 06:16.22 | *** join/#asterisk mirco (n=mirco@87.139.74.16) |
| 06:18.43 | mosty | kaldemar, it prints out loads more than that, before timing out, do you want me to paste more? |
| 06:20.45 | *** join/#asterisk Gabril (n=gabril@ns2.webix.de) |
| 06:22.31 | flenders | mosty: pri show spans |
| 06:22.38 | flenders | is it up? |
| 06:23.11 | mosty | flenders, no it's down, i can't figure out why: Status: Provisioned, Down, Active |
| 06:23.22 | flenders | what sort of card is it? |
| 06:23.25 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 06:23.32 | mosty | it's a digium te412p |
| 06:23.45 | flenders | who's the carrier? |
| 06:23.48 | flenders | optus? |
| 06:23.51 | mosty | AAPT |
| 06:24.11 | flenders | so, their tech was there and using a tester he could make calls? |
| 06:24.15 | mosty | yes |
| 06:24.15 | kaldemar | everything seems to be fine in your end... |
| 06:24.35 | flenders | mate, I had a similar issue a couple of weeks ago |
| 06:24.45 | mosty | flenders, yeah, what was the fix? |
| 06:24.46 | flenders | optus sent out a tech to test line, cables etc |
| 06:25.03 | flenders | and the problem went away when I moved the card to a different slot |
| 06:25.28 | mosty | eek, it's a 2 hour flight each way to get to the box |
| 06:26.07 | flenders | where's the box at? |
| 06:26.15 | *** part/#asterisk Gabril (n=gabril@ns2.webix.de) |
| 06:26.43 | mosty | it's in brisbane |
| 06:27.07 | flenders | and you're in syd? |
| 06:27.14 | webman | is anyone using a very recent 1.4 branch of asterisk ? |
| 06:27.37 | mosty | flenders, no, melbourne |
| 06:27.50 | flenders | oh yeah, I've spoken to you before, haven't I? |
| 06:28.06 | mosty | probably, about a month ago |
| 06:28.09 | flenders | before I went on holidays |
| 06:28.17 | flenders | april I think |
| 06:28.30 | flenders | let me check your files to make sure |
| 06:28.52 | mosty | sure, my most recent paste is at http://pastebin.ca/563571 |
| 06:30.22 | flenders | what does 'ztcfg -vv' show you? |
| 06:31.11 | mosty | do you want the whole thing or just the first span? |
| 06:31.20 | flenders | the whole thing |
| 06:31.32 | flenders | is it a full PRI? |
| 06:32.27 | mosty | http://pastebin.ca/563591 |
| 06:32.57 | mosty | it's 2 full pri's and a 10 channel pri |
| 06:33.02 | mosty | i get the same results on each span |
| 06:33.37 | mosty | btw i only pasted the span def for span 1 in zaptel.conf, similarly for zapata.conf |
| 06:35.16 | mosty | spans 1,2,3 are provisioned, down, active and span 4 is provisioned, in alarm, down, active. |
| 06:35.35 | mosty | actually sorry it's 3 full pri's now |
| 06:35.48 | mosty | except aapt is calling them PRA's |
| 06:36.11 | flenders | PRA?? |
| 06:37.21 | mosty | yeah. according to wikipedia it's the same thing http://en.wikipedia.org/wiki/Integrated_Services_Digital_Network |
| 06:39.05 | flenders | can you receive calls on that span? |
| 06:39.20 | mosty | not last time i tried, let me find the number to try again |
| 06:39.25 | *** join/#asterisk KpoH (n=AID@host-86-106-208-182.moldtelecom.md) |
| 06:40.02 | mosty | the AAPT tech said their system doesn't see any response from my end |
| 06:40.31 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 06:41.50 | mosty | i just get an engaged tone when i call in, nothing in the asterisk console that i can see |
| 06:41.52 | *** join/#asterisk syneus (n=syneus@syneus.aemcom.net) |
| 06:42.12 | flenders | and they tested your cables too? |
| 06:42.27 | flenders | this is my zapata.conf: http://pastebin.ca/563615 |
| 06:42.31 | mosty | yes, they plugged an isdn phone in and made calls |
| 06:43.02 | flenders | have you seen that card working before? |
| 06:43.18 | *** join/#asterisk ReDNeQ (n=ReDNeQ@cpe-66-68-37-190.austin.res.rr.com) |
| 06:43.36 | flenders | I was chatting to the optus tech last week and he said it's pretty common |
| 06:43.37 | *** join/#asterisk iBuMp- (n=ReDNeQ@cpe-66-68-37-190.austin.res.rr.com) |
| 06:44.05 | flenders | faulty hardware |
| 06:44.07 | mosty | no, the same card had the same problem on a telstra line last time |
| 06:44.43 | flenders | so it could be a faulty card |
| 06:45.09 | flenders | I was about to buy another card when I decided to try a different slot on the server |
| 06:45.55 | mosty | thanks for your help, i'll try to get someone to swap the cards for me, and if that fails i'll fly up there with a sangoma card and try that |
| 06:46.23 | flenders | good luck mate! |
| 06:46.33 | mosty | thanks |
| 06:47.01 | *** join/#asterisk iBuMp (n=ReDNeQ@cpe-66-68-37-190.austin.res.rr.com) |
| 06:51.44 | *** join/#asterisk mellen (n=thansen@85.235.250.110) |
| 06:53.20 | *** join/#asterisk matsk (n=mk@194.68.102.174) |
| 07:02.48 | *** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
| 07:04.50 | *** join/#asterisk op3r (n=core@125.212.50.108) |
| 07:05.48 | *** join/#asterisk blueneon (i=hfklows@dsl-146-28-220.telkomadsl.co.za) |
| 07:06.56 | blueneon | hi, i've had to get a new pc and chuck asterisk on it as my old asterisk box died, i've managed to get everything working, except that my music on hold (mp3's using mpg123) sounds terrible, it sounds almost distant and fuzzy... any ideas? |
| 07:07.13 | iBuMp | you can try boosting the gain |
| 07:07.18 | iBuMp | on tx |
| 07:07.44 | blueneon | but normal voice etc sounds fine? |
| 07:07.50 | iBuMp | sure |
| 07:07.50 | blueneon | its just the moh thats not |
| 07:08.06 | iBuMp | i had that problem and noticed i mad my txgain 10 |
| 07:08.13 | iBuMp | it didnt make hrt me |
| 07:08.18 | blueneon | my gain is 0 |
| 07:08.21 | blueneon | afaik |
| 07:08.25 | *** join/#asterisk ramindia (n=ramindia@202.63.96.133) |
| 07:08.29 | blueneon | for both tx and rx |
| 07:08.42 | blueneon | what should i change it to? |
| 07:08.45 | iBuMp | mine are set 20 for receive and tx 10 |
| 07:08.50 | iBuMp | you can play with it and see |
| 07:08.52 | blueneon | yeaks |
| 07:08.54 | blueneon | ok |
| 07:08.55 | *** join/#asterisk fRiSi (n=fRiSi@194.183.146.178) |
| 07:08.57 | iBuMp | i started at 5 |
| 07:09.01 | blueneon | i'll try 20 and 10 to start |
| 07:09.07 | fRiSi | good morning #asterisk |
| 07:11.05 | creativx | top of the morning to you sir |
| 07:11.13 | blueneon | well that didnt work lol, i cant even dial with those settings, it ignores my keypad hehe |
| 07:12.17 | blueneon | hmm that sounds much louder, tho there is still a fuzzy sound going on |
| 07:12.25 | blueneon | like hissing almost |
| 07:12.42 | blueneon | everytime the person sings or whatever |
| 07:12.49 | blueneon | hard to explain :S |
| 07:17.37 | flenders | blueneon: are you using zaptel? |
| 07:17.44 | flenders | fxo modules? |
| 07:17.46 | *** join/#asterisk andyd (n=andyd@213-228-240-161.dsl.prodigynet.co.uk) |
| 07:19.48 | flenders | once you start messing around with tx/rx gains, you start getting a lot of echo on fxo channels |
| 07:26.42 | *** join/#asterisk gardo (n=gardo@121.97.211.111) |
| 07:27.52 | blueneon | flenders: hi yes fxo's |
| 07:28.10 | blueneon | i had the gain's set to 0.0 for both |
| 07:28.27 | blueneon | the reason i changed was because iBuMp recommended i do so |
| 07:28.33 | blueneon | tho it hasnt changed the issue |
| 07:28.41 | iBuMp | i thought it made it louder |
| 07:28.46 | iBuMp | but your saying its fuzz? |
| 07:28.50 | blueneon | im still getting the wierd hissy sound... almost like the singer has a lisp |
| 07:28.52 | blueneon | lol |
| 07:28.56 | iBuMp | haha |
| 07:29.02 | iBuMp | comon man give me some credit |
| 07:29.04 | blueneon | ye it has made it louder, but that wasnt the issue |
| 07:29.06 | iBuMp | i fixed half the problem |
| 07:29.11 | blueneon | it sounds, distant and fuzzy |
| 07:29.11 | blueneon | :/ |
| 07:29.27 | iBuMp | distant ohh i thouth you meant quite far away like |
| 07:29.29 | blueneon | louder is good, :P well done for that |
| 07:29.30 | iBuMp | my bad |
| 07:30.57 | creativx | i still think that moh is the work of the devil |
| 07:31.01 | creativx | give the callers BEEPS instead. |
| 07:31.03 | creativx | :> |
| 07:31.40 | blueneon | nah, i use it to advertise products |
| 07:31.49 | blueneon | i have professional voice of music |
| 07:31.59 | flenders | was it alright on your previous install? |
| 07:32.03 | blueneon | which worked fine and dandy on my old box |
| 07:32.10 | blueneon | yup |
| 07:36.10 | *** join/#asterisk purplet (n=purplet@010.041.dsl.concepts.nl) |
| 07:36.13 | blueneon | bah now i have an appointment to go to |
| 07:36.34 | blueneon | i think i might actually convert the mp3's to wav later and let asterisk play them directly |
| 07:36.43 | blueneon | cause it seems to play the existing wav's fine |
| 07:36.45 | blueneon | bbl |
| 07:37.47 | *** join/#asterisk porche (n=porche@88.239.132.50) |
| 07:37.54 | porche | hi all |
| 07:41.48 | *** join/#asterisk reber (n=reber@cl-158.dub-01.ie.sixxs.net) |
| 07:43.07 | *** part/#asterisk ramindia (n=ramindia@202.63.96.133) |
| 07:52.41 | *** join/#asterisk yassaccan (n=yassacca@admin136.hgo.se) |
| 07:55.06 | *** join/#asterisk Dibbler_ (n=Dibbler@host217-45-198-229.in-addr.btopenworld.com) |
| 07:56.31 | mosty | are there any issues with using a sangoma E1 card in the same machine as one of those digium transcoding cards? |
| 07:57.46 | *** join/#asterisk Vorondi1 (n=vorondil@unaffiliated/vorondil) |
| 08:00.17 | fRiSi | are there any solutions for more detailed call-logging than /var/log/asterisk/cdr-csv/Master.csv? |
| 08:00.21 | fRiSi | they don't tell me which extensions the caller navigated through |
| 08:01.42 | porche | firisi, do you mean mysql logging? |
| 08:03.40 | mosty | fRiSi, you can use agi scripts |
| 08:04.05 | fRiSi | porche: mysql, csv does not matter at the moment |
| 08:04.31 | fRiSi | mosty: i already though about an agi script that logs the current ${ANSWEREDTIME} to a custom logfile each time an extension was called and evaluate these logfile using awstats or analog. |
| 08:05.05 | porche | hms, you can run manager interface |
| 08:05.11 | porche | for more detailed |
| 08:05.17 | *** join/#asterisk thekidrio (n=thekidri@24-205-65-193.dhcp.psdn.ca.charter.com) |
| 08:05.19 | fRiSi | mosty: but i thought there might be a less intrusive way (cause this way i need to call the agi-script on each extension |
| 08:05.28 | creativx | tracking calls via the manager interface is to say at least interesting |
| 08:05.38 | mosty | ami is crap |
| 08:05.42 | porche | no, mosty meant manager interface i think |
| 08:06.01 | mosty | i'm suggesting AGI. AMI is crap |
| 08:06.11 | creativx | I got AMI to play along. |
| 08:06.53 | porche | but on AGI, you forward the control to agi script i think, out of dial plan |
| 08:07.00 | creativx | but for a bit different things |
| 08:07.02 | fRiSi | if ami is this: http://www.voip-info.org/wiki/view/Asterisk+cdr+manager - there is the same info as in Master.csv (only total call time) |
| 08:07.21 | thekidrio | anyone know if there is a soundboard for the asterisk sounds? |
| 08:07.34 | *** join/#asterisk linagee_ (n=linagee@unaffiliated/linagee) |
| 08:07.54 | linagee_ | are there any at&t gods up? LOL |
| 08:07.59 | *** join/#asterisk bintut (n=bintut@203.125.63.150) |
| 08:08.13 | linagee_ | i think San Diego's AT&T are down |
| 08:08.13 | thekidrio | gods never sleep, they just refuse to listen |
| 08:08.39 | linagee_ | my cox interwebs are down too. :( |
| 08:08.47 | linagee_ | chatting off the pda. heh |
| 08:09.06 | creativx | those damn interwebs. |
| 08:09.11 | linagee_ | someone must have done something really dubious |
| 08:09.32 | linagee_ | creativx: want to know a funny jokef |
| 08:10.06 | linagee_ | creativx: i can't call into cox right now. (i guess they eat their own dogfood? LOL) |
| 08:10.26 | *** join/#asterisk andyd (n=andyd@host90-152-23-30.ipv4.regusnet.com) |
| 08:10.40 | linagee_ | i'd feel sorry for all those saps with cox ATAs right about now |
| 08:12.21 | creativx | linagee_: well |
| 08:12.27 | creativx | first, what is cox |
| 08:12.39 | linagee_ | did someone plug the internet back in? yay! |
| 08:13.42 | thekidrio | cox is an access provider |
| 08:14.28 | *** join/#asterisk linagee (n=linagee@unaffiliated/linagee) |
| 08:16.30 | *** join/#asterisk ltd (n=z@nox.amused.net) |
| 08:18.52 | creativx | ah |
| 08:19.19 | *** join/#asterisk matsk (n=mk@194.68.102.174) |
| 08:21.43 | *** join/#asterisk darkskiez (n=mbryars@195-11-205-216.suip.mezzonet.net) |
| 08:22.12 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 08:22.43 | KpoH | does anybody know why may not work rtsavesysname=yes in sip.conf? |
| 08:29.30 | creativx | jaipon |
| 08:31.02 | JT | creativx: ? |
| 08:34.26 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
| 08:35.02 | creativx | nufin |
| 08:35.02 | creativx | :) |
| 08:36.28 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 08:45.50 | *** join/#asterisk friedrich| (n=friedric@e177248083.adsl.alicedsl.de) |
| 08:48.27 | fRiSi | porche, mosty thanks for your input. i posted the question on asterisk-users too let's see if i'll get additinal input. otherwhise i'll stay with an agi script |
| 08:53.24 | *** part/#asterisk fRiSi (n=fRiSi@194.183.146.178) |
| 08:57.52 | Nugget | my brain keeps trying to insert the "s" in "porche" |
| 08:58.25 | porche | :) yeah, he's right, there must be an "s", but i think "s" makes it ugly |
| 08:58.48 | Nugget | Perhaps you'd be happier with a Ferari. ;) |
| 08:59.03 | porche | no, two r's make is carismatic |
| 08:59.08 | porche | is=it |
| 09:00.49 | *** join/#asterisk mbranca (n=matteo@host-210-mi.linuxserver.it) |
| 09:05.59 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 09:08.36 | *** join/#asterisk thoughtpolice (n=austin@c75-111-136-171.plaicmtc01.tx.dh.suddenlink.net) |
| 09:08.40 | *** part/#asterisk thoughtpolice (n=austin@c75-111-136-171.plaicmtc01.tx.dh.suddenlink.net) |
| 09:12.07 | *** join/#asterisk lehel (n=sfdsd@86.125.81.206) |
| 09:13.49 | creativx | Farari |
| 09:13.56 | creativx | porcini |
| 09:14.11 | *** join/#asterisk pj_ (n=bpr@nan92-4-82-229-211-60.fbx.proxad.net) |
| 09:14.37 | pj_ | Hello |
| 09:15.00 | porche | farari is bad |
| 09:15.15 | porche | porcini not bad, but looks like a spagetti brand |
| 09:15.59 | Nugget | heh |
| 09:19.20 | *** join/#asterisk KevinS (n=kevin@d54C71C8A.access.telenet.be) |
| 09:19.22 | creativx | hehe |
| 09:19.26 | creativx | funky porcini is an artist i think |
| 09:20.55 | porche | it looks like spagetti anyways |
| 09:21.09 | porche | may be i am hungry ;) |
| 09:21.46 | *** part/#asterisk KevinS (n=kevin@d54C71C8A.access.telenet.be) |
| 09:22.00 | *** join/#asterisk Dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 09:22.22 | Dandre | hello, |
| 09:22.45 | Dandre | is there any doc on users.conf file? |
| 09:23.18 | Dandre | For instance how should trunks be defined in this file? |
| 09:25.05 | Nugget | http://macnugget.org/photos/2007c2s/mispeled <-- there ya go, porche :) |
| 09:27.17 | porche | ah cool |
| 09:27.24 | porche | yes that's better, dont you think so? |
| 09:27.25 | porche | :) |
| 09:27.32 | Nugget | no, absolutely not. :) |
| 09:27.38 | porche | :) |
| 09:30.57 | *** join/#asterisk Vorondil (n=vorondil@unaffiliated/vorondil) |
| 09:31.35 | *** join/#asterisk Fulk (n=fulk@87-194-176-39.bethere.co.uk) |
| 09:31.58 | Fulk | morning |
| 09:32.10 | Nugget | moo |
| 09:32.21 | Fulk | I have so many questions, where to start :-) |
| 09:32.34 | *** join/#asterisk ghenry (n=ghenry@suretecsystems.plus.com) |
| 09:33.00 | sevard | ~thebook |
| 09:33.02 | jbot | extra, extra, read all about it, thebook is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 09:33.09 | Fulk | read it |
| 09:33.13 | Fulk | was a bit basic IMHO |
| 09:33.17 | creativx | heheh Nugget, plenty of spare time what |
| 09:33.20 | Fulk | nice it was free :-) |
| 09:33.22 | sevard | a bit too basic |
| 09:33.28 | Nugget | insomnia can be a tricky thing. |
| 09:33.51 | Fulk | but, looking in my Oreilly account there's a lot of Asterisk books in the works to be released later this year |
| 09:33.56 | Fulk | 2007/2008 could be the year of Asterisk :-) |
| 09:34.05 | Fulk | or if you're pedantic, 2 years |
| 09:34.18 | Fulk | anyway |
| 09:34.21 | sevard | Nugget: valerian root |
| 09:34.23 | Fulk | are there any asterisk appliance/box vendors who are currently shipping? |
| 09:34.38 | Fulk | I've come across the Rhino Ceros, which to be honest looks ugly as hell |
| 09:35.36 | *** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust826.leed.cable.ntl.com) |
| 09:35.36 | Fulk | I'm wary against rolling my own again, I used a Poweredge 2850 and it keeps leaking memory and locking up |
| 09:36.09 | Fulk | I think it's to do with the digium PRI card, cos two other boxes on far cheaper hardware have been rock solid, but they're pure SIP/IAX |
| 09:36.29 | *** join/#asterisk stoffell (n=stoffell@fw.catsanddogs.com) |
| 09:37.25 | *** join/#asterisk stoffell (n=stoffell@fw.catsanddogs.com) |
| 09:38.20 | sevard | Nugget: although, that's a bit too intense for mild insomnia. i suggest chamomile. it' smuch more gentle |
| 09:38.23 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 09:38.58 | Nugget | I prefer to waste time on irc and make fun of porche. :) |
| 09:39.01 | Fulk | anyone been able to interface a hardware fax device into Asterisk? |
| 09:39.43 | *** join/#asterisk Ringod (n=Ringo@d54C71C8A.access.telenet.be) |
| 09:39.45 | sevard | I'd just like to say, Mel Gibson is a flaming piece of crap. |
| 09:40.11 | sevard | apocalypto was another snuff film+waste of time |
| 09:40.11 | Ringod | ? |
| 09:40.23 | sevard | I can't believe I used to like this homo. |
| 09:41.29 | Ringod | Oops I must joined the wrong channel, seems a movie channel |
| 09:41.53 | sevard | Yeah, this is *, imdb's fanfiction channel. |
| 09:42.00 | sevard | what were you looking for? |
| 09:42.02 | Nugget | Coming this Summer from New Line Pictures -- Asterisk! |
| 09:42.29 | Ringod | I'm an Asterisk nuub and was hoping to find some movie freaks in here :p |
| 09:43.06 | creativx | ehh |
| 09:43.09 | creativx | damn ip10s!! |
| 09:43.20 | creativx | why is it that one of 15 phones decides not to obey answer-after: 0 in the sip header |
| 09:43.50 | sevard | probably because you keep refering to them as "slaves" and trying to make them "obey" |
| 09:43.58 | creativx | the 14 others dont mind. |
| 09:44.20 | sevard | alpha-dog syndrome. off one to show who is boss. |
| 09:44.33 | creativx | i am considering it sevard. |
| 09:45.44 | creativx | AHA |
| 09:45.45 | creativx | wtf |
| 09:46.18 | creativx | it tricked me into thinking it had actually upgraded its firmware |
| 09:46.22 | creativx | no wonder its ape shit |
| 09:47.25 | *** join/#asterisk tengulre (n=tengulre@222.90.66.10) |
| 09:47.48 | creativx | oh well.. it still hangs on boot after doing a web reboot |
| 09:47.52 | creativx | must be the french programmers. |
| 09:47.54 | *** part/#asterisk porche (n=porche@88.239.132.50) |
| 09:50.41 | *** join/#asterisk Vorondi1 (n=vorondil@unaffiliated/vorondil) |
| 09:51.12 | sevard | frenchmen, not worth a dime stacked up against us irishmen. |
| 09:51.48 | sevard | but then, of course, who isn't? |
| 09:53.24 | creativx | well |
| 09:53.24 | creativx | the norwegians stack up pretty good. |
| 09:55.36 | sevard | I live with plenty of those. they stink to highhell and don't like potatos. they're out. |
| 09:56.01 | e-ddie | hrm |
| 09:56.08 | e-ddie | sounds like some weird norwegians you live with |
| 09:56.15 | *** join/#asterisk darviria (n=darviria@194-105-181-29.ifb.co.uk) |
| 09:56.42 | sevard | bad sampling? |
| 09:58.12 | sevard | what were you looking for? |
| 09:58.15 | sevard | Till the pupil found his master |
| 09:58.36 | sevard | what the crap, my client just spit out random lines! |
| 09:59.32 | creativx | sevard: we keep the good norwegians inside norway |
| 10:00.33 | e-ddie | creativx: no you dont |
| 10:00.59 | e-ddie | but it sounds like he hasnt met any of them |
| 10:01.44 | creativx | ok let me correct myself. I only know of those from the upper shelf. |
| 10:01.48 | creativx | =) |
| 10:01.53 | e-ddie | hehe |
| 10:02.56 | *** join/#asterisk Nobbie (n=anony@fwb003.fw.is.co.za) |
| 10:02.58 | Nobbie | heya = ) |
| 10:05.22 | creativx | su |
| 10:05.23 | creativx | p |
| 10:06.44 | *** join/#asterisk A-Data (n=asd@196.219.183.199) |
| 10:06.56 | A-Data | Hello I have question are there any body alive to help? |
| 10:07.06 | creativx | dont ask to ask |
| 10:07.06 | creativx | ;) |
| 10:07.15 | creativx | rule number 1 of the interwebnet |
| 10:07.23 | A-Data | lol ok |
| 10:08.16 | A-Data | here is the question we have a PBX system Normal One we want to link it to Asterik we haven`t isntalled asterisk yet. but we are planning .. then we want our Support operators to use there computers as VOIP clients is this ok or we can`t do that? |
| 10:09.30 | creativx | no problem. get interface cards for the * box and SIP/iax2 softphones for the clients |
| 10:09.55 | A-Data | what interface cards ?? is it normal Fax/Voice Modem or somthing different |
| 10:10.06 | creativx | different |
| 10:10.17 | creativx | you need some way to get the phone signals from your normal pbx into the asterisk |
| 10:10.22 | creativx | they dont travel automagically via air. |
| 10:10.26 | creativx | unfortunately :( |
| 10:10.48 | creativx | but there are others here who know a lot more about iface cards than I do, since I am running all-IP from end to end. |
| 10:12.17 | A-Data | what i mean the interface card is a normal fax/voice modem or somthing else |
| 10:14.30 | A-Data | ? |
| 10:16.00 | kaldemar | something else. you need to know what interfaces your current PBX has and then you can start planning the integration. |
| 10:16.26 | A-Data | kaldemar my PBX use the normal telephone lines |
| 10:16.34 | Nobbie | who know how to fix the broken SIP NOTIFY's on subscribed extensions in 1.4.4 ? |
| 10:16.46 | kaldemar | i'd replace the old system with an asterisk box instead of using two systems. |
| 10:17.07 | pj_ | I will be setting up an * server behind a redfone box... I was wondering if anyone knew about how well vmwared * worked with TDMoE ? |
| 10:17.41 | A-Data | kaldemar but what interfaces need asterisk box required to recive the call normal voice/fax or somthing differenet? |
| 10:17.46 | kaldemar | A-Data: digium's choices for that are here: http://www.digium.com/en/products/hardware/analogcards.php |
| 10:18.38 | kaldemar | you need telephony interfaces. i don't know what "normal" is, but i'd guess FXS/FXO. |
| 10:19.12 | A-Data | kaldemar what i mean with normal is the voice/data cards like motrola and so on |
| 10:19.30 | A-Data | i am good in linux but bad in communications issue that why i am confused |
| 10:20.15 | kaldemar | you need to find out what technology the PBX uses. |
| 10:20.37 | pj_ | A-Data: You want to plug a regular phone line (POTS) or phones directly into your asterisk ? If so you need FXS (for phones) and FXO (for lines) interfaces |
| 10:20.42 | pj_ | like TDM400P for instance |
| 10:21.11 | A-Data | normal phone line from the PBX |
| 10:21.24 | A-Data | beacuse all my users will use VOIP behind the Asterisk |
| 10:22.28 | A-Data | so in this case is it fxs or fxo |
| 10:23.02 | *** join/#asterisk d00gster (n=doughant@bas1-toronto12-1128667111.dsl.bell.ca) |
| 10:23.23 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 10:24.19 | A-Data | also can we use motrola voice fax modem as a start then we buy interfaces like TDM as u said or we have to purchase TDM |
| 10:24.20 | pj_ | Alas, it's more complicated |
| 10:25.10 | pj_ | It really depends on what kind of signalling your motorola is providing, if it plays nice, etc... |
| 10:25.53 | pj_ | You will have to buy a card somewhere down the path though |
| 10:26.28 | A-Data | Plays nice or not pj_ what we care now is to test the functionabilty if it ok we can buy somthing better but what only we need now is test so morola 56 voice/data modem is ok or we can`t use it at all |
| 10:28.26 | kaldemar | afaik, you can't use a voice/data modem to interface with PBX's. |
| 10:29.16 | A-Data | kaldemar it`s ok we can remove the PBX at all if the Asterisk support our test |
| 10:29.58 | *** join/#asterisk Vorondil (n=vorondil@unaffiliated/vorondil) |
| 10:31.15 | A-Data | so can we use the motorla 56 voice/data without the PBX |
| 10:32.57 | kaldemar | you can't interface phone lines with the modem. you need telephony hardware. |
| 10:33.30 | A-Data | what u mean i can`t interface? |
| 10:35.07 | *** join/#asterisk dofear (n=arodef@ppp157-151.static.internode.on.net) |
| 10:35.20 | kaldemar | you can't make calls through it. |
| 10:35.56 | A-Data | kaldemar even can`t accept incoming calls |
| 10:35.58 | A-Data | ? |
| 10:36.16 | *** join/#asterisk Vorondi1 (n=vorondil@unaffiliated/vorondil) |
| 10:36.16 | kaldemar | you can't do anything with it. |
| 10:36.21 | A-Data | sorry i am newbie hre may be i am asking folish question |
| 10:38.20 | A-Data | so what is the best card we can use in Egypt to work with our telephone system |
| 10:41.29 | *** join/#asterisk tmcpr (n=tmcpr@85-189-92-116.btlnet.managedbroadband.co.uk) |
| 10:42.20 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 10:43.35 | penguinFunk | A-Data: you will need an fxo card that will do the analogue to digital conversion between your phone lines and asterisk |
| 10:43.41 | penguinFunk | see www.digium.com |
| 10:45.13 | *** join/#asterisk florz (i=nobody@2001:1a50:503c:0:0:0:0:1) |
| 10:46.01 | *** join/#asterisk reber (n=reber@cl-158.dub-01.ie.sixxs.net) |
| 10:50.23 | A-Data | any one know egyptian seller or distrubuter |
| 10:50.23 | A-Data | ? |
| 10:57.31 | A-Data | while reading in forum i see somthing that i can use normal modems liek zoltrix 56 k and motrola 56k to pass the PTSN line to the Asterisk using somthing called silent answer ... can we do that?? and can any one please explain to me what is silent answer |
| 11:02.08 | *** join/#asterisk Vorondil (n=vorondil@unaffiliated/vorondil) |
| 11:03.58 | creativx | wtf |
| 11:04.06 | creativx | suddenly the ip10s dropped inbound audio |
| 11:08.02 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 11:08.50 | *** join/#asterisk odity (n=odity@213.136.48.97) |
| 11:09.18 | *** join/#asterisk canberk (n=cn@teknopet.com) |
| 11:09.24 | canberk | hello |
| 11:09.43 | canberk | <PROTECTED> |
| 11:09.44 | canberk | <PROTECTED> |
| 11:09.44 | canberk | <PROTECTED> |
| 11:09.44 | canberk | [Jun 13 14:06:18] ERROR[23957]: rtp.c:2370 ast_rtcp_write_sr: RTCP SR transmission error to 192.168.2.50:16421, rtcp halted Host is down |
| 11:27.56 | *** join/#asterisk kl1and2 (n=kakita@82.158.136.145.static.user.ono.com) |
| 11:28.03 | kl1and2 | hello |
| 11:28.19 | kl1and2 | anyone using one quadBRI card with power feeding? |
| 11:28.34 | kl1and2 | I am not able to give it working with one BRI phone |
| 11:29.17 | kl1and2 | thnks |
| 11:37.13 | *** join/#asterisk ifnotwhynot (n=davidh@nas-mid.nashuamp.co.za) |
| 11:37.29 | nephfl | hello anyone know if meridian phones will work with asterisk? |
| 11:37.31 | ifnotwhynot | please help me i am going to loose my customer |
| 11:39.34 | cpm | nephfl, doubtful |
| 11:39.35 | ifnotwhynot | is there any way you cab excecute a linux command from the asterisk dialplan???????// |
| 11:39.47 | cpm | ifnotwhynot, sure. |
| 11:39.54 | ifnotwhynot | ho wplease? |
| 11:39.57 | stoffell | ifnotwhynot, sure you can , check voip-info.org form some examples |
| 11:40.15 | ifnotwhynot | what would i search for? |
| 11:40.31 | creativx | system call |
| 11:40.43 | *** join/#asterisk jer (n=jtregunn@unaffiliated/jer) |
| 11:40.51 | creativx | [Description] |
| 11:40.51 | creativx | <PROTECTED> |
| 11:40.51 | creativx | fails, the console should report a fallthrough. |
| 11:40.54 | stoffell | kl1and2, does your phone get now power? |
| 11:40.57 | creativx | *CLI> show application System |
| 11:40.57 | stoffell | now=no |
| 11:41.10 | cpm | ifnotwhynot, also look into agi-bin |
| 11:41.10 | creativx | ahah crap! |
| 11:41.18 | creativx | dangerous to press ctrl-c in the wrong putty window |
| 11:41.22 | creativx | suddenly i killed asterisk :O |
| 11:41.32 | ifnotwhynot | can you tell me if this will work i want to exten => h,1,system(email -s "Incomming" -a /var/spool/asterisk/fax/${CALLERID}.tif -b cbl@nashua.co.za) |
| 11:41.40 | cpm | ifnotwhynot, also look into agi-bin |
| 11:42.41 | ifnotwhynot | i can receive the fax no problem but can not mail it |
| 11:43.10 | ifnotwhynot | i can run the command email -s ect from my command line in linux it sends no problme |
| 11:43.52 | nephfl | i asked the company that sold me a digium tdm card if i could use my existing phones and gave them the model info, they put me on hold and "researched" the issue and said they would work...i think they lied...bastards |
| 11:44.06 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 11:44.25 | *** join/#asterisk Calios (n=david@60-241-242-156.static.tpgi.com.au) |
| 11:44.32 | stoffell | ifnotwhynot,try it out .. if it doesn't work you can always do system(youremailscript.sh) (and insert your commands in the youremailscript.sh) |
| 11:44.42 | stoffell | ifnotwhynot, but also check the documentation on the system command.. |
| 11:45.01 | *** join/#asterisk angryuser (n=Miranda@df01t2-62-34-180-141.d4.club-internet.fr) |
| 11:45.06 | angryuser | good day |
| 11:45.25 | kl1and2 | stoffell...I have it connected to one old NT |
| 11:46.00 | kl1and2 | stoffell...one crossed cable from NT to card and phone plugged in the other NT rj45 connector |
| 11:46.07 | cpm | nephfl, then call them back and ask them |
| 11:46.34 | kl1and2 | stoffell..phone goes up and I listen dialtone..but when I do one call to one extension I dont look any messages in cli and back me call error |
| 11:46.52 | kl1and2 | stoffell...I did same test with one port card using zaphfc and it worked |
| 11:46.53 | ifnotwhynot | where can i get a sample of a mail script: stoffel? |
| 11:47.11 | *** part/#asterisk Calios (n=david@60-241-242-156.static.tpgi.com.au) |
| 11:47.11 | kl1and2 | stoffell...I have junghanns power feeding too but results are same |
| 11:48.11 | stoffell | ifnotwhynot, that's something you should figure out.. |
| 11:48.59 | stoffell | kl1and2, so using an isdn phone connected to zaphfc works, connecting it to quadbri does not work, right ? |
| 11:49.36 | kl1and2 | stoffell..correct |
| 11:49.58 | kl1and2 | stoffell..phone need power...I supply it with one old NT |
| 11:50.41 | kl1and2 | stoffell..same if I use power feeding in card with jumpers set to give power to quadBRI ports |
| 11:52.36 | stoffell | hm, weird |
| 11:52.44 | kl1and2 | stoffell...I tested with bri_net and bri_net_ptmp |
| 11:54.06 | *** join/#asterisk Nobbie (n=anony@fwb003.fw.is.co.za) |
| 11:54.28 | Nobbie | who upgraded to * |
| 11:54.32 | Nobbie | who upgraded to * 1.4.4 and lives to tell the tale ? |
| 11:56.06 | stoffell | kl1and2, and no power, weird, no idea then, maybe you can test if it really gives out power... |
| 11:56.14 | *** join/#asterisk codefreeze (n=steve_mu@216.166.159.235) |
| 11:56.32 | kl1and2 | stoffell...how can I test if quadBRI ports are getting power? |
| 11:56.49 | kl1and2 | stoffell...I know I have jumpers on pines..but... |
| 11:56.53 | penguinFunk | stick a wet finger into them? |
| 11:56.58 | Nobbie | hehehe |
| 11:57.01 | kl1and2 | ;) |
| 11:57.03 | penguinFunk | :P |
| 11:57.25 | Nobbie | a wet tongue is more accurate for those kinds of test |
| 11:57.25 | Nobbie | s |
| 11:57.33 | kl1and2 | but...if I use power feeding...my phone doesnt get power |
| 11:57.52 | Nobbie | PoE ? power over ether ? |
| 11:58.02 | Nobbie | oph, BRI, nm |
| 11:58.05 | kl1and2 | really..correct scenario mus to be powerfeeding attached to card..jumpers activating power on ports..and phone connected to port |
| 11:58.08 | stoffell | kl1and2, yep, so it could be the power feeding module, maybe there's a way to see if it gets power.. (use a meter for that, tongue is useful for something else) |
| 11:58.19 | stoffell | correct |
| 11:58.32 | kl1and2 | powerfeeding have led green |
| 11:58.48 | stoffell | check if the manual doesn't state anything.. if all else fails, also give junghanns.net a call.. (or mail) |
| 11:59.09 | kl1and2 | I opened one ticket with them..but..I know how junghanns support is ;) |
| 11:59.21 | kl1and2 | if next month I have one answer...I will be happy ;) |
| 11:59.38 | angryuser | i have some problems with queues, i have a queue support, when agent is busy asteisk go to n+1 and dont respect the timeout, can somebody explain me why? |
| 12:00.00 | stoffell | kl1and2,yeah, that's why calling is maybe faster.. oh, and being persistent :p |
| 12:00.07 | kl1and2 | about zaphfc..it works perfectly with old NT giving power to phone |
| 12:00.28 | kl1and2 | problem I need 4 BRI ports for this client...else.. |
| 12:00.42 | *** join/#asterisk Xen^ (n=linux@unaffiliated/lnux/x-10290) |
| 12:00.46 | kl1and2 | it has to be connected to one PABX and I have to act like telco NT |
| 12:01.03 | kl1and2 | giving line and power to PABX BRI card |
| 12:01.10 | kl1and2 | resuming..act like S0 bus |
| 12:02.18 | *** join/#asterisk Polis_ttt (n=your@194-237-172-225-no48.business.telia.com) |
| 12:08.27 | drako | *CLI> misdn show stacks |
| 12:08.27 | drako | BEGIN STACK_LIST: |
| 12:08.27 | drako | <PROTECTED> |
| 12:08.27 | drako | <PROTECTED> |
| 12:08.28 | *** join/#asterisk Corydon76-home (i=four@pdpc/supporter/sustaining/Corydon76-home) |
| 12:08.28 | *** mode/#asterisk [+o Corydon76-home] by ChanServ |
| 12:08.40 | drako | wy my links are all down? |
| 12:09.08 | dwmw2 | because you've been naughty |
| 12:09.20 | drako | :/ |
| 12:09.55 | pj_ | I will be setting up an * server behind a redfone box... I was wondering if anyone knew about how well vmwared * worked with TDMoE ? |
| 12:13.08 | *** join/#asterisk perf3kt (i=perf3kt@149.166.33.59) |
| 12:13.29 | stoffell | kl1and2, your theory is okay, if your jumpers are good then it must be the power feeding module that's faulty |
| 12:14.08 | *** part/#asterisk Dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 12:14.26 | stoffell | kl1and2, configuring an s0 bus is not so difficult |
| 12:15.08 | pj_ | (no success / failure story on TDMoE and vmware ?) |
| 12:16.35 | *** join/#asterisk guillote_GNU (n=guillote@host176.201-252-205.telecom.net.ar) |
| 12:16.35 | *** join/#asterisk yannj_fr (n=yannj@APuteaux-152-1-92-202.w90-2.abo.wanadoo.fr) |
| 12:20.08 | kl1and2 | stoffell...I can give power using one old NT |
| 12:20.15 | kl1and2 | stoffell...and I have same results |
| 12:20.45 | Fulk | the echo is terrible! |
| 12:21.13 | kl1and2 | stoffell..so I tested to use 2 different modes of giving power to ports |
| 12:21.42 | kl1and2 | stoffell...and with old NT it has to give power because it worked with one billion card (zaphfc) |
| 12:21.55 | drako | Fulk, i regret buying a sipura-3000 |
| 12:22.00 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 12:22.38 | *** join/#asterisk CBU[^_^]M`` (n=love@210.213.138.105) |
| 12:23.15 | Fulk | drako, I had a spa3000 b4, it broke within the warranty period but Linksys never got back to me |
| 12:23.29 | Fulk | I never had an issue with it, pig to configure though |
| 12:23.35 | Fulk | but there's lots of tutorials on the internet |
| 12:24.40 | stoffell | kl1and2, then your zaptel / zapata.conf might be wrong when using the quadbri? |
| 12:24.44 | drako | Fulk, my problem is the volumen is to low and... it does not hang or free some calls |
| 12:25.04 | drako | Fulk, so make the line unavalaible |
| 12:25.31 | *** join/#asterisk mosty (i=mostyn@60-241-198-194.static.tpgi.com.au) |
| 12:26.35 | Fulk | drako, you'll need to change the settings to your locale |
| 12:26.42 | drako | Fulk, i did |
| 12:27.13 | drako | Fulk, it happens only "sometimes" and i cannot find the source of the problem, never happens to me but always with other users so i can't figure what they do.... |
| 12:27.23 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 12:27.37 | drako | Fulk, but then the spa sip channel becomes unusable and return busy status |
| 12:27.58 | mosty | when i do an attended transfer, what context does the new call start in? |
| 12:28.11 | drako | Fulk, i did some adjustment and suddenly in a middle of a talk it hangs up... |
| 12:28.14 | mosty | ie via features.conf |
| 12:28.31 | shay|work | hard |
| 12:28.43 | Fulk | I guess TDM is the way to go :-) |
| 12:29.20 | drako | Fulk, well, if you can handles the echo problems, ok but... I don't think is the best alternative tho... |
| 12:29.23 | shay|work | that's why working with BRI and ISDN cards rock |
| 12:29.54 | Fulk | the X100P isn't used for the main calls, only for emergency |
| 12:30.06 | Fulk | everything else is pure SIP/IAX |
| 12:30.16 | Fulk | the GSM router works nice though |
| 12:30.34 | drako | Fulk, are you in eu or us? |
| 12:30.38 | Fulk | UK |
| 12:30.49 | Fulk | have free intra-staff calling s |
| 12:31.01 | Fulk | o I set up dialplans to match staff mobiles |
| 12:31.16 | Fulk | staff can also dial in and get a dial tone so they can dial out on VOIP |
| 12:31.20 | Fulk | very cool |
| 12:31.24 | *** join/#asterisk shido6 (i=shido6@d221-68-200.commercial.cgocable.net) |
| 12:31.54 | *** join/#asterisk waKKu (n=eurulo@unaffiliated/wakku) |
| 12:31.59 | waKKu | morning folks ;) |
| 12:32.08 | Fulk | afternoon |
| 12:32.15 | perf3kt | mornin |
| 12:32.23 | pj_ | morning |
| 12:32.26 | waKKu | 3 x 1 |
| 12:32.30 | waKKu | is morning ;D |
| 12:32.32 | Fulk | hey, I'm on GMT |
| 12:32.39 | Fulk | centre of the world |
| 12:32.45 | Fulk | :P |
| 12:32.46 | pj_ | afternoon for me too, but it doesn't hurt to adapt a bit :P |
| 12:32.56 | waKKu | hehehe |
| 12:33.08 | *** join/#asterisk Vorondi1 (n=vorondil@unaffiliated/vorondil) |
| 12:33.24 | Fulk | fear the mighty british empire of, umm.. Gibralta and the Falkland islands |
| 12:34.31 | tzafrir | kl1and2, can you do a loopback from a NT to a TE port on the card itself? |
| 12:34.56 | *** join/#asterisk bertrand^ (n=bertrand@ATuileries-151-1-71-140.w86-212.abo.wanadoo.fr) |
| 12:35.36 | kl1and2 | tzafrir: I have every ports set to NT |
| 12:36.04 | kl1and2 | how I can to do loopback? need any special cable? |
| 12:38.13 | tzafrir | no. I meant from NT to TE |
| 12:38.55 | kl1and2 | interconnect one port in TE mode with other in NT mode? |
| 12:38.58 | kl1and2 | u meant this? |
| 12:44.08 | *** join/#asterisk zeeesh (i=zeeesh@14-237-154-202.wol.net.pk) |
| 12:44.10 | zeeesh | hi |
| 12:44.42 | *** join/#asterisk tanacsdavid (n=david@office.axpnet.com) |
| 12:44.51 | tanacsdavid | Hellello! |
| 12:44.57 | tanacsdavid | I'm here, again. |
| 12:45.25 | tanacsdavid | still having problems, but now other |
| 12:46.35 | tanacsdavid | I broke up with trixbox/asterisknow, and trying to use pure asterisk |
| 12:46.54 | angryuser | tanacsdavid: wise choice ;) |
| 12:47.07 | tanacsdavid | I hope so... |
| 12:47.16 | tanacsdavid | but I hava other problems... |
| 12:47.26 | perf3kt | tanacsdavid:I'm with you |
| 12:47.38 | perf3kt | tanacsdavid: but only on chapter 2 of the book |
| 12:47.49 | angryuser | tanacsdavid: go ahead |
| 12:47.58 | tanacsdavid | Using the http://www.the-asterisk-book.com, I cannot really start asterisk |
| 12:48.25 | angryuser | tanacsdavid: start asterisk with -vvvvv option and pastebin |
| 12:48.25 | tanacsdavid | according to nmap it does not listen on the defined port (5060) |
| 12:48.51 | *** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox) |
| 12:49.35 | angryuser | tanacsdavid: somethins is using port 5060? |
| 12:50.01 | [TK]D-Fender | tanacsdavid: aND WHAT ABOUT NETSTAT? |
| 12:50.49 | tanacsdavid | http://pastebin.ca/564061 |
| 12:50.53 | tanacsdavid | this is my asterisk |
| 12:51.44 | [TK]D-Fender | tanacsdavid: == SIP Listening on 0.0.0.0:5060 |
| 12:51.46 | angryuser | == SIP Listening on 0.0.0.0:5060 hm |
| 12:51.55 | angryuser | hehe fender ;) |
| 12:52.21 | [TK]D-Fender | == Using SIP TOS: none |
| 12:52.22 | [TK]D-Fender | <PROTECTED> |
| 12:52.24 | [TK]D-Fender | <PROTECTED> |
| 12:52.25 | [TK]D-Fender | <PROTECTED> |
| 12:52.27 | [TK]D-Fender | <PROTECTED> |
| 12:52.29 | [TK]D-Fender | <PROTECTED> |
| 12:52.30 | [TK]D-Fender | <PROTECTED> |
| 12:52.33 | [TK]D-Fender | <PROTECTED> |
| 12:52.33 | [TK]D-Fender | <PROTECTED> |
| 12:52.35 | [TK]D-Fender | <PROTECTED> |
| 12:52.36 | [TK]D-Fender | <PROTECTED> |
| 12:52.38 | [TK]D-Fender | chan_sip.so => (Session Initiation Protocol (SIP)) |
| 12:52.39 | [TK]D-Fender | Sure LOOKS fine to me. |
| 12:52.45 | s0ck | tanacsdavid: it's a udp port, nmap -sU |
| 12:52.52 | [TK]D-Fender | tanacsdavid: First lesson : SIP = UDP so forget nmap :) |
| 12:52.58 | zeeesh | getting problem "*CLI> Can't locate Asterisk/AGI.pm in @INC (@INC contains: /usr/lib/perl5/site_perl/5.8.8/i386-linux-thread-multi "? |
| 12:53.01 | tanacsdavid | [TK]D-Fender: 0.0.0.0 should mean listen on every address, am I right? |
| 12:53.10 | [TK]D-Fender | tanacsdavid: Correct. |
| 12:53.17 | [TK]D-Fender | tanacsdavid: "netstat -an" |
| 12:53.18 | thevoke | zeeesh: install Asterisk::AGI |
| 12:53.19 | [TK]D-Fender | ^^^^^^^^^^ |
| 12:54.02 | angryuser | fender, do you know something about misdn? |
| 12:54.11 | angryuser | or anybody ;) |
| 12:54.14 | tanacsdavid | but when I forget nmap, the client (sjphone) can't find asterisk |
| 12:54.19 | tanacsdavid | nmap: http://pastebin.ca/564066 |
| 12:54.31 | *** join/#asterisk ToyMan (n=Stuart@user-12lcqol.cable.mindspring.com) |
| 12:54.31 | [TK]D-Fender | angryuser: I hear it has something to do with ISDN, and it sucks :) |
| 12:55.12 | [TK]D-Fender | tanacsdavid: pastebin what I told you to look for at CLI, and then tell my why you'[re even LOOKING AT THIS in the first place. |
| 12:55.24 | angryuser | [TK]D-Fender: tell it my boss, i would be happy |
| 12:55.32 | shido6 | zeeesh, http://asterisk.gnuinter.net/ , and http://www.voip-info.org/wiki-Asterisk+AGI |
| 12:55.45 | zeeesh | <thevoke>: how to find Asterisk::AGI installed or not ? |
| 12:55.56 | shido6 | . |
| 12:56.08 | [TK]D-Fender | zeeesh: Dod YOU install it? It sure didn't come included with *. |
| 12:56.20 | zeeesh | ok |
| 12:56.34 | tanacsdavid | [TK]D-Fender: I'm disturbing You, because I'm stucked according to the books... |
| 12:56.54 | tanacsdavid | trying to make it on my own, but I'm too little for this... |
| 12:57.14 | [TK]D-Fender | tanacsdavid: why don't you show us what you're actually configuring in * instead of thinking that SIP itself isn't loaded. |
| 12:57.14 | zeeesh | i tried to find by "yum search Asterisk::AGI" but could not get anything |
| 12:57.25 | tanacsdavid | ok |
| 12:57.28 | tanacsdavid | moment |
| 12:57.29 | [TK]D-Fender | tanacsdavid: You are using the wrong tools, and the wrong approch. |
| 12:57.43 | shido6 | zeeesh, http://asterisk.gnuinter.net/ , and http://www.voip-info.org/wiki-Asterisk+AGI |
| 12:57.45 | *** join/#asterisk MedozasSVR (n=MedozasS@p549B98EE.dip0.t-ipconnect.de) |
| 12:58.16 | MedozasSVR | hi! I was wondering if somebody can help me with gxp 2000 and indications.... |
| 12:58.46 | MedozasSVR | how does it work out with indications, does the gxp 2000 play the tones defined as in indications.conf? |
| 12:59.21 | MedozasSVR | my problem is, the phone has these definitions in its web interface too, but I have no idea to change them |
| 12:59.24 | [TK]D-Fender | MedozasSVR: indications.conf is for Zaptel. Any regionalization on a SIP phone has to be done BY the phone. |
| 12:59.26 | tanacsdavid | sip.conf: http://pastebin.ca/564073 |
| 12:59.27 | tanacsdavid | extensions.conf: http://pastebin.ca/564075 |
| 12:59.39 | tanacsdavid | anything else is the sample config |
| 13:00.30 | MedozasSVR | ok ... well can somebody help me with the fixed indications for germany? |
| 13:00.49 | [TK]D-Fender | tanacsdavid: Ok, so turn on SIP debug at * CLI and pastebin the call attempt |
| 13:02.02 | MedozasSVR | I've been looking all through the web .... I have german indications.conf |
| 13:02.03 | MedozasSVR | for example: |
| 13:02.03 | MedozasSVR | dial = 425 |
| 13:02.03 | MedozasSVR | but in the phone: |
| 13:02.03 | MedozasSVR | f1=350,f2=440; |
| 13:02.08 | MedozasSVR | documentation for the phone: |
| 13:02.08 | MedozasSVR | <PROTECTED> |
| 13:02.08 | MedozasSVR | (Frequencies are in Hz and cadence on and off are in 10ms) |
| 13:02.16 | MedozasSVR | sorry for longer posting |
| 13:02.19 | zeeesh | <[TK]D-Fender>: if u r talking about agi file which placed normally "/var/lib/asterisk/agi-bin/astcc.agi" then its already there .. i mean i have a perl script which already placed there |
| 13:02.48 | tanacsdavid | [TK]D-Fender: I don't even get to a call attemp. The clien cannot access the * |
| 13:02.57 | [TK]D-Fender | zeeesh: No, your problem is the Perl AGI module. |
| 13:03.21 | [TK]D-Fender | tanacsdavid: What is this "client", and where it it located relative to *? |
| 13:03.32 | *** join/#asterisk Dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 13:03.53 | Dandre | hello, |
| 13:04.38 | *** join/#asterisk angryuser (n=Miranda@df01t2-62-34-180-141.d4.club-internet.fr) |
| 13:04.44 | tanacsdavid | [TK]D-Fender: client is a Linksys SPA921 IP phone and an SJphone running on my computer. The * is on public IP address, without firewall |
| 13:04.47 | angryuser | sw |
| 13:04.53 | Dandre | is there any way to get a config file from manager interface in raw format instead of Categoryxxx and Linexxx format? |
| 13:05.06 | tanacsdavid | SJphone sais: server is not responding |
| 13:05.33 | [TK]D-Fender | tanacsdavid: Where is SJPHONE relative to *? |
| 13:05.44 | [TK]D-Fender | tanacsdavid: and NEITHER of them are talking to *? |
| 13:05.44 | shido6 | a config file from the manager interface.. |
| 13:05.45 | shido6 | hrmm |
| 13:05.52 | shido6 | u can construct one from the cli |
| 13:06.02 | shido6 | show dialplan |
| 13:06.05 | tanacsdavid | sjphone is on nat. |
| 13:06.10 | shido6 | asterisk -rx 'show dialplan' |
| 13:06.34 | shido6 | what do you mean by config file? |
| 13:06.46 | shido6 | Dandre |
| 13:06.49 | tanacsdavid | sjphone (192.168.11.182) -> router 88.151.100.144 ->internet -> * 88.151.100.75 |
| 13:07.11 | *** join/#asterisk slmnhq (n=salmanh@denali.asti-usa.com) |
| 13:07.23 | tanacsdavid | neither linksys, nor sjphone cannot see it |
| 13:07.29 | *** join/#asterisk FliTTi (n=chatzill@212.218.65.254) |
| 13:07.32 | [TK]D-Fender | tanacsdavid: congratulations, time to learn that SIP + NAT do NOT play nice! |
| 13:07.46 | FliTTi | hey@ll |
| 13:08.01 | [TK]D-Fender | tanacsdavid: For that phone's entry in sip.conf, add "nat=yes" , and "qualify=yes" |
| 13:08.05 | zeeesh | <[TK]D-Fender>: yeah agreed its perl module problem .. which path its indicating like "/usr/lib/perl5/site_perl/5.8.8/i386-linux-thread-multi" .. i search there is something kind of figures "5.8.5 5.8.6 5.8.7 5.8.8 site_perl vendor_perl" ? |
| 13:08.21 | Dandre | I have a module in perl that connects to the manager using a modified version of Asterisk::Manager perl module |
| 13:08.23 | [TK]D-Fender | zeeesh: time to get WIKI-ing |
| 13:08.41 | zeeesh | ok |
| 13:09.14 | Dandre | so when I send GetConfig users.conf command, I get a lot of junk. |
| 13:09.28 | Dandre | shido6: |
| 13:09.36 | shido6 | ok. |
| 13:09.55 | tanacsdavid | [TK]D-Fender: Hah! That was the solution, I think. :) |
| 13:10.10 | tanacsdavid | dialing between the two client is good. |
| 13:10.13 | tanacsdavid | Thank You! |
| 13:10.23 | tanacsdavid | May I pay You a beer? :) |
| 13:10.32 | Dandre | so I wondered if I could get a clean version of if I had to write it in my module |
| 13:10.48 | tanacsdavid | sorry for beeing so hard.. |
| 13:10.53 | tanacsdavid | or stupid |
| 13:10.55 | [TK]D-Fender | tanacsdavid: Ok, now what you should do : Start playing with extensions.conf. Creat more dial patterns for your phones to be able to call. |
| 13:11.45 | tanacsdavid | [TK]D-Fender: I try it. :) |
| 13:11.46 | [TK]D-Fender | tanacsdavid: Make an echo test. Make another exten to record some audio from the phone and play it back. Make an IVR (small!). Add voicemail to the extens that dial your SIP devices. |
| 13:11.54 | *** join/#asterisk qdk (n=qdk@145.Red-81-44-152.dynamicIP.rima-tde.net) |
| 13:12.52 | [TK]D-Fender | tanacsdavid: After that, also play around with codecs a little bit. sjphone supports quite a few, while the only one you're likely to get working with the SPA (without paying for G.729) is G.711 (ulaw or alaw). |
| 13:12.53 | FliTTi | i need a little help xD. If I get a call and make a 3way transfer to my associate, he alway see my number if he caling with the caller. It would be better if he see in his display the number of the caller, instead mine (like it's on a Blind transfer) does anyone have an idear, how it works? thanks |
| 13:13.23 | *** join/#asterisk lirakis (n=etamme@65.200.191.253) |
| 13:13.28 | *** part/#asterisk lirakis (n=etamme@65.200.191.253) |
| 13:13.51 | tanacsdavid | [TK]D-Fender: G711 is perfekt for me, because of the provider. |
| 13:13.54 | [TK]D-Fender | FliTTi: Thats not how it works, sorry. YOU are calling HIM. the fact you are planning to bridge in anlother caller is something his phone won't know about. |
| 13:14.40 | [TK]D-Fender | tanacsdavid: I was suggesting playing with the codecs for EXPERIENCE, not because you should use something smaller (you may need a low bandwidth option at some point). |
| 13:15.46 | FliTTi | i know that. But it is not possible to change the number in the display of my associate when he takeover the caller? |
| 13:15.59 | creativx | change CID midway conversation ? |
| 13:16.06 | creativx | remember he is still talking to you |
| 13:16.17 | creativx | for what his phone cares |
| 13:16.18 | tanacsdavid | [TK]D-Fender: I'll do it. I'm starting to like asterisk. :) |
| 13:16.22 | *** join/#asterisk SuPrSluG (n=SuPrSluG@pool-71-241-175-183.buff.east.verizon.net) |
| 13:16.58 | FliTTi | creativx: you mean me? |
| 13:17.32 | creativx | yes FliTTi |
| 13:17.47 | [TK]D-Fender | tanacsdavid: Congratulations on going GUI free. In setting up 2 SIP devices that can call to * and through to each other with it you have done all you need to really test & learn *. Everything else is just additive |
| 13:18.05 | [TK]D-Fender | FliTTi: NO. |
| 13:19.28 | mocker | Ugh, any Avaya people in the house? ;0 |
| 13:19.50 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 13:20.18 | [TK]D-Fender | mocker: Try asking in #avaya |
| 13:20.28 | FliTTi | i get a call. hold it. call myassociate (he see my number). i transfer the call (and put done my handset). now the associate should the the number of the caller which has called me. (like it's in a blind transfer). Is there really no possibility to change the CID? |
| 13:21.21 | FliTTi | sry about my typing |
| 13:21.24 | [TK]D-Fender | FliTTi: If you blind transfer, yes he will see the CID that YOU saw |
| 13:21.29 | *** join/#asterisk nexilus (n=caligo@213.134.125.3) |
| 13:21.55 | blitzrage | unless your dialplan logic changes it |
| 13:21.55 | FliTTi | right. And i whant to schow him these number in a 3way transfer |
| 13:21.58 | creativx | you could perhaps set up a trick extension that modifies the cid |
| 13:22.01 | *** join/#asterisk canberk (n=cn@teknopet.com) |
| 13:22.26 | blitzrage | you might just need 'o' flag in Dial() |
| 13:22.31 | [TK]D-Fender | FliTTi: If you are are in a 3-way call and try to disconnect, leaving the other 2 parties talking to each other, this is NOT a transfer, and his display will NOT miraculously change. |
| 13:22.51 | [TK]D-Fender | FliTTi: there is no such thing as a 3-way transfer. |
| 13:23.07 | [TK]D-Fender | FliTTi: You go from a locally conferenced call to an ATTENDED transfer. |
| 13:23.13 | mocker | [TK]D-Fender: I don't think Avaya admins know about this IRC thing. |
| 13:23.15 | mocker | ;) |
| 13:23.42 | mocker | [TK]D-Fender: My analog adapter arrives tomorrow! |
| 13:23.49 | FliTTi | and put an o in Dial () wouldn't change anything? like blitzrage says |
| 13:24.01 | [TK]D-Fender | mocker: and TCP is This is Currently Profitable / Intellectual Property ;) |
| 13:24.29 | blitzrage | FliTTi: you might need to set: ${TRANSFER_CONTEXT} Context for transferred calls |
| 13:24.34 | creativx | mmmm profit. |
| 13:24.48 | blitzrage | that way you can have separate logic when you transfer, than what you would use normally for a standard phone to phone call |
| 13:25.08 | blitzrage | that way you can either set the CID to whatever you want -- or utilize the 'o' flag in Dial() to pass the 'original' CID to the transferee |
| 13:25.19 | file | if it is an attended transfer, you do not know it is a transfer until the other call takes the place of yours... and you can not (currently) update the caller ID from yours to theirs |
| 13:25.26 | file | move along... move along... |
| 13:25.32 | creativx | pauses for repetition. |
| 13:25.37 | [TK]D-Fender | file: Just to make it through! |
| 13:25.46 | *** join/#asterisk DarylVOIP (n=daryl@c-71-224-42-97.hsd1.pa.comcast.net) |
| 13:26.10 | FliTTi | and this acts in a 3way transfer? |
| 13:26.37 | JerJer | Mr. Wizard died :( |
| 13:26.47 | blitzrage | JerJer: that sucks!!! |
| 13:27.06 | FliTTi | blitzrage: only put an o in Dial() ? |
| 13:27.19 | blitzrage | FliTTi: see file's msg's above |
| 13:28.06 | *** join/#asterisk Ethon (i=arne@Oldman.steinkamm.com) |
| 13:28.50 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 13:28.58 | FliTTi | yes, but there are to options: put an o to Dial() or set: ${TRANSFER_CONTEXT}? right? sry about my comprehension |
| 13:28.59 | *** part/#asterisk MedozasSVR (n=MedozasS@p549B98EE.dip0.t-ipconnect.de) |
| 13:29.44 | blitzrage | FliTTi: if it is an attended transfer, you do not know it is a transfer until the other call takes the place of yours... and you can not (currently) update the caller ID from yours to theirs |
| 13:29.52 | blitzrage | (quoted from file) |
| 13:30.32 | FliTTi | sorry |
| 13:30.37 | blitzrage | FliTTi: which means I was wrong -- you will only be able to do it on a blind transfer apparently (by you calling the other party first, and then hitting transfer, you are still doing a sort of attended transfer) |
| 13:30.46 | file | oh no, I've been quoted :( |
| 13:30.47 | DrukenHME | isn't the whole point of attended transfer to announce the call?? so CID would technically be irrelivant? |
| 13:30.54 | blitzrage | DrukenHME: bingo |
| 13:31.14 | [TK]D-Fender | "file" <- ah AH!!!!! cought you! |
| 13:31.25 | [TK]D-Fender | caught* |
| 13:31.38 | file | DrukenHME: people still use attended transfers as blind transfers too, depending on the one it's an extra step to do blind that they may not want to do |
| 13:31.47 | file | the one? the phone |
| 13:32.06 | FliTTi | yes. but if me associate whant to call the caller back he must ask him, and he can not look at his display |
| 13:32.25 | creativx | i think people should stop transferring calls at all. |
| 13:32.31 | creativx | it would even be better if people stopped calling! |
| 13:32.45 | *** join/#asterisk SirThomas (n=tomc@mail.kendeco.com) |
| 13:32.47 | [TK]D-Fender | FliTTi: Then end the 3-way, and blind transfer him rather than just leave them connected. |
| 13:32.47 | FliTTi | ^^ thanks |
| 13:32.57 | FliTTi | nice idear |
| 13:33.25 | FliTTi | thanks@all |
| 13:34.16 | DrukenHME | uhg... my house is a mess... |
| 13:34.21 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 13:34.21 | *** mode/#asterisk [+o anthm] by ChanServ |
| 13:35.55 | creativx | damnit |
| 13:36.02 | creativx | why oh why did my ghetto parking have to break |
| 13:37.53 | creativx | is there any way i can see the macro depth a current extension/chan is at |
| 13:37.54 | *** join/#asterisk ifnotwhynot (n=davidh@nas-mid.nashuamp.co.za) |
| 13:37.55 | DrukenHME | ghetto parking? |
| 13:37.56 | DrukenHME | hehe |
| 13:38.05 | creativx | yeah.. i made an extension that just rings forever |
| 13:38.09 | creativx | to place callers on park |
| 13:38.39 | creativx | and use redirect to put them there.. and get them back |
| 13:38.40 | ifnotwhynot | any know how i can send a fax that i receive using asterisk to a emailaddress? |
| 13:38.58 | DrukenHME | creativx: ahh... i use moh for that :) |
| 13:39.12 | [TK]D-Fender | ifnotwhynot: "man sendmail" |
| 13:39.14 | blitzrage | creativx: just use GoSub() and not Macro().... |
| 13:39.16 | creativx | DrukenHME: well I could ofcourse have played them some music.. but nah |
| 13:39.24 | creativx | and yeah, im trying to avoid using macros |
| 13:39.32 | creativx | and gosub, return, 10 goto 20 |
| 13:39.54 | DrukenHME | my god... i just had basic flahbacks |
| 13:40.01 | [TK]D-Fender | 10 PRINT "I AM GOING CRAZY!" 20 GOTO 10 |
| 13:40.03 | FliTTi | bye and thanks |
| 13:40.16 | *** join/#asterisk mkl1525 (n=qwertz@pD953063D.dip0.t-ipconnect.de) |
| 13:40.28 | mkl1525 | Hi, is there a way to connect to skype using *? |
| 13:40.30 | creativx | haha [TK]D-Fender |
| 13:40.32 | creativx | something like that yes |
| 13:40.33 | blitzrage | w00t gwbasic, what?! |
| 13:40.38 | mocker | Woo, 170 people have downloaded my simple AGI overview and nobody complained about my crappy code! |
| 13:40.55 | creativx | mocker: thats because we all write crappy code... er what |
| 13:41.16 | [TK]D-Fender | creativx: What are you talking about???! My shit is WAY better than his CRAP! |
| 13:41.21 | *** join/#asterisk jer (n=jtregunn@unaffiliated/jer) |
| 13:42.07 | creativx | hmm |
| 13:42.10 | creativx | well mine smells the worst |
| 13:42.10 | creativx | ! |
| 13:43.01 | drako | Jun 13 15:42:02 WARNING[5537]: res_monitor.c:319 ast_monitor_change_fname: Cannot change monitor filename of channel Zap/1-1 to null |
| 13:44.51 | mocker | creativx: Heh, good to hear! |
| 13:48.44 | Dandre | does anyone knows how to define trunks in users.conf file? |
| 13:49.05 | perf3kt | ~book |
| 13:49.07 | jbot | well, book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 13:49.13 | *** part/#asterisk FlyboySR22 (n=rsears@hq.fw.americanis.net) |
| 13:50.25 | *** join/#asterisk FlyboySR22 (n=rsears@hq.fw.americanis.net) |
| 13:52.04 | *** join/#asterisk Zeeek (n=IceChat7@pdpc/supporter/active/Zeeek) |
| 13:52.08 | *** join/#asterisk vutamhoan (n=hoavq@222.252.156.130) |
| 13:53.00 | Zeeek | blah |
| 13:53.26 | vutamhoan | I've got problem with TE110P card. The sync source is internally clocked, not external as I configure |
| 13:53.26 | creativx | hm |
| 13:53.35 | vutamhoan | can anyone help me? |
| 13:53.39 | [TK]D-Fender | Dandre: Check upgrade.txt, the users.conf sample, and sacrifice a goat! |
| 13:53.48 | creativx | if i have an extension named _4000.,1,noop(blah), i can redirect to any number beginning with 4000<jattajaatta> |
| 13:53.55 | [TK]D-Fender | vutamhoan: pastebin your zaptel.conf |
| 13:53.57 | [TK]D-Fender | ~pb |
| 13:53.59 | jbot | i heard pb is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste, or at paste.lisp.org |
| 13:54.01 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^ |
| 13:54.12 | creativx | but what if i want an extension named _test.,1,noop(blah), and gosub(mycontext,test555,1) ? |
| 13:54.24 | Zeeek | what if we don't like any of the above pastebins? |
| 13:54.39 | grey | rafb.net/paste |
| 13:54.39 | grey | ? |
| 13:55.29 | blitzrage | Zeeek: shush you! |
| 13:55.47 | Zeeek | [TK]D-Fender will be happy to see I have a new link o' the day |
| 13:56.07 | [TK]D-Fender | Zeeek: Where? |
| 13:57.12 | [TK]D-Fender | vutamhoan: no DCC |
| 13:57.17 | [TK]D-Fender | vutamhoan: www.apstebin.ca |
| 13:57.24 | *** join/#asterisk _VoiceMeUp_COM (n=_VoiceMe@modemcable159.131-56-74.mc.videotron.ca) |
| 13:57.26 | [TK]D-Fender | www.pastebin.ca |
| 13:57.28 | *** join/#asterisk A-Data (n=asd@217.54.224.153) |
| 13:57.29 | [TK]D-Fender | rather |
| 13:57.37 | [TK]D-Fender | vutamhoan: Please pay attention. |
| 13:57.42 | Zeeek | it was hard to find: http://politicalvideo.org/ |
| 13:57.50 | vutamhoan | yes |
| 13:57.54 | Zeeek | The GW Bush video archive |
| 13:58.02 | Zeeek | Better than voip-info.org |
| 13:58.09 | blitzrage | heh |
| 13:58.15 | Zeeek | 500 hours of comedy |
| 13:58.25 | vutamhoan | sorry Fender, I don't know the meaning of DCC |
| 13:58.27 | A-Data | Hello All. i have a question can ASterisk Query a database and play results from MYSQL database so that if user enter his phone number for example he get results like you have 3 new emails as emails is stored in database |
| 13:58.39 | [TK]D-Fender | A-Data: Yes. |
| 13:58.55 | A-Data | [TK]D-Fender any doc about this ? |
| 13:59.02 | [TK]D-Fender | A-Data: "show application mysql" , " show function ODBC" |
| 13:59.09 | yannj_fr | playdigit |
| 13:59.26 | [TK]D-Fender | A-Data: And start WIKI-ing. Then go read about AGI as it may be a more appropriate means to doing this. |
| 13:59.42 | [TK]D-Fender | ~book |
| 13:59.43 | jbot | from memory, book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 13:59.50 | [TK]D-Fender | A-Data: ---^ |
| 13:59.52 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 13:59.55 | Zeeek | ~dasbuch |
| 14:00.06 | *** join/#asterisk waptaxi (n=cahe@45.151-224-87.telenet.ru) |
| 14:00.23 | Nugget | Hi Guys. I want to walk across the room to get more coffee but my legs aren't doing anything. What now?!?! |
| 14:00.54 | Zeeek | Nugget pastebin your network topology |
| 14:00.55 | yannj_fr | buy a usb coofee maker |
| 14:01.04 | [TK]D-Fender | vutamhoan: STOP TRYING TO SEND VIA DCC! |
| 14:01.16 | vutamhoan | sorry |
| 14:01.17 | Zeeek | [TK]D-Fender I thought you didn't know what DCC was? |
| 14:01.19 | [TK]D-Fender | vutamhoan: Go to www.pastebin.ca and paste it THERE and provide the link! |
| 14:01.32 | vutamhoan | understand now :) |
| 14:02.36 | Zeeek | due to first life encroachment, there will be no Asterisk Users Conference tomorrow |
| 14:03.21 | tzanger | [TK]D-Fender: is it newbie smell getting a little thick in here? |
| 14:03.42 | tzanger | I just changed the reel yesterday |
| 14:03.51 | Zeeek | check the hose |
| 14:04.10 | Zeeek | 'slike 3 and the danger level is 280 |
| 14:04.33 | Zeeek | sorry that was for the #prostate window |
| 14:04.54 | *** join/#asterisk andyd (n=andyd@host90-152-23-30.ipv4.regusnet.com) |
| 14:06.12 | blitzrage | A-Data: 2nd edition of TFoT (when it is released...) will show you how to put VM into a database via ODBC, or via IMAP storage -- so yes, it is possible |
| 14:07.12 | [TK]D-Fender | blitzrage: ... he wasn't asking about VM.... |
| 14:07.42 | blitzrage | oh -- he said email |
| 14:07.44 | blitzrage | it's still early |
| 14:07.53 | blitzrage | then you probably just want to use func_odbc |
| 14:08.25 | Corydon76-home | Heh |
| 14:08.47 | tzanger | prostate? |
| 14:08.58 | A-Data | blitzrage any EST about 2nd TFOT |
| 14:09.02 | ManxPower | HEritic! |
| 14:09.07 | blitzrage | A-Data: ummm... August I would guess |
| 14:09.16 | [TK]D-Fender | vutamhoan: You also can't PM people if you aren't registered to Freenode... |
| 14:09.18 | blitzrage | it'd be sooner if I'd get off my duff and finish reading this copyedit |
| 14:09.23 | [TK]D-Fender | vutamhoan: jsut say it in here. |
| 14:09.23 | Corydon76-home | blitzrage: I don't drink coffee, either. Gives me migraines. |
| 14:09.30 | *** join/#asterisk ramindia (n=ramindia@202.63.96.9) |
| 14:09.33 | blitzrage | Corydon76-home: oh ya... don't drink coffee then |
| 14:09.37 | perf3kt | tk:well you can't be in this channel if you're no registered |
| 14:09.38 | ManxPower | sickos, all of you non-coffee people. |
| 14:09.55 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 14:09.59 | [TK]D-Fender | vutamhoan: You are clearly "batting 1000" (x 10 ^-4) |
| 14:10.09 | *** join/#asterisk blueneon (i=hfklows@dsl-146-31-112.telkomadsl.co.za) |
| 14:10.16 | Corydon76-home | And it's not the caffeine, it's some other chemical in coffee... since I drink Mtn Dew fine |
| 14:10.25 | blueneon | whats better to use within a local network with asterisk, SIP or IAX? |
| 14:10.29 | blitzrage | Corydon76-home: ya... wonder what it would be then |
| 14:10.40 | vutamhoan | this's the firs time i use mIRC :( |
| 14:10.44 | ramindia | blueneon: both depends |
| 14:10.45 | [TK]D-Fender | blueneon: SIP |
| 14:10.47 | blitzrage | mirc sucks :) |
| 14:11.07 | blueneon | ok next q, whats the best freebie SIP client for XP? |
| 14:11.18 | [TK]D-Fender | blueneon: idefisk |
| 14:11.24 | ramindia | eybeam |
| 14:11.26 | blueneon | idefisk it is then, ta |
| 14:11.27 | blitzrage | idefisk for sure |
| 14:11.34 | *** join/#asterisk crichardson (n=crichard@38.113.5.185) |
| 14:11.39 | [TK]D-Fender | blueneon: www.asteriskguru.com |
| 14:11.49 | blueneon | already downloading :) |
| 14:11.51 | vutamhoan | [TK]D-Fender - I can't post the zaptel.conf to pastebin.ca |
| 14:12.01 | [TK]D-Fender | vutamhoan: and why not? |
| 14:12.06 | vutamhoan | but, in my zaptel.conf |
| 14:12.14 | blueneon | now i need to remember how to setup a SIP channel |
| 14:12.15 | blueneon | :S |
| 14:12.22 | [TK]D-Fender | blueneon: .... |
| 14:12.23 | vutamhoan | span=1,1,0,cas,hdb3,yellow |
| 14:12.23 | vutamhoan | cas=1-15:1101 |
| 14:12.23 | vutamhoan | dchan=16 |
| 14:12.23 | vutamhoan | cas=17-31:1101 |
| 14:12.24 | [TK]D-Fender | ~osmosis |
| 14:12.26 | jbot | i heard osmosis is the act of beating yourself on the head repeatedly with THE BOOK, until some measure of absorption has occured ... or at least until your unconsciousness restores peace to the channel ... |
| 14:12.28 | vutamhoan | That's all |
| 14:12.57 | crichardson | i got a question has any one had an issue with snom phones after hanging up it seems to keep the channel open? i have noticed this in 1.2 and 1.4 |
| 14:13.29 | vutamhoan | I asked TE110p use far end clocked, but zttool tell internally clocked |
| 14:13.37 | *** join/#asterisk basty (n=basty@212.218.65.232) |
| 14:13.38 | basty | Hi |
| 14:13.45 | blueneon | [TK]D-Fender ...? |
| 14:13.47 | _VoiceMeUp_COM | hi |
| 14:14.30 | [TK]D-Fender | vutamhoan: CAS doesn't use a dchan....only PRI |
| 14:14.50 | blueneon | [16:12] <[TK]D-Fender> blueneon: .... sup mate? |
| 14:14.53 | [TK]D-Fender | vutamhoan: And what's on the other side of your TE110P? |
| 14:15.01 | [TK]D-Fender | blueneon: ... |
| 14:15.04 | [TK]D-Fender | ~osmosis |
| 14:15.06 | jbot | extra, extra, read all about it, osmosis is the act of beating yourself on the head repeatedly with THE BOOK, until some measure of absorption has occured ... or at least until your unconsciousness restores peace to the channel ... |
| 14:15.10 | *** join/#asterisk mindCrime (n=chatzill@66.83.208.219.nw.nuvox.net) |
| 14:15.10 | [TK]D-Fender | blueneon: ---^ |
| 14:15.15 | *** join/#asterisk irule (n=irule@189.164.43.19) |
| 14:15.18 | vutamhoan | I connect to NEC exchange |
| 14:15.19 | blueneon | dont understand |
| 14:15.19 | _VoiceMeUp_COM | lol |
| 14:15.23 | _VoiceMeUp_COM | this is so zen |
| 14:15.32 | _VoiceMeUp_COM | beaating yourself on the head repeatedly with THE BOOK, until some measure of absorption has occured |
| 14:15.35 | basty | is it possible to display the callerid due using the atted transfer on asterisk? Example: Incoming Call -> Member 1 answers -> Member 1 transfers the call to Member2. In the display of Member2 is the callerid of Member1 even when he is allready speaking to the incoming call... |
| 14:15.43 | shay|work | editing modules.conf, I got: |
| 14:15.48 | *** part/#asterisk ramindia (n=ramindia@202.63.96.9) |
| 14:15.52 | shay|work | ; The 'modem' channel driver and its subdrivers are \ ; obsolete, don't load them. |
| 14:15.54 | *** join/#asterisk anonymouz666 (n=anonymou@re366.compuland.com.br) |
| 14:15.55 | blueneon | you want me to purify water for you Fender? |
| 14:15.57 | vutamhoan | I want to use R2/MFC. And I DID successfully with TE205P, TE4xxp |
| 14:16.03 | shay|work | shouldn't I load them for ISDN/BRI support? |
| 14:16.10 | _VoiceMeUp_COM | nah just turn water into wine |
| 14:16.16 | [TK]D-Fender | basty: Attended transfer = they see YOUR CID, not your caller's |
| 14:16.27 | [TK]D-Fender | basty: thats "the way it works", and TFB. |
| 14:16.47 | [TK]D-Fender | basty: Use a blind transfer. |
| 14:17.04 | Zeeek | Ok |
| 14:17.22 | Zeeek | [TK]D-Fender et al, tell me if you see anything odd here |
| 14:17.23 | basty | [TK]D-Fender: Well but i need to tell Member2 who is on the phone before i transfer the call... |
| 14:17.34 | Zeeek | http://pastebin.ca/564199 |
| 14:17.57 | [TK]D-Fender | Zeeek: externip = my.external.ip.adr <- BAD |
| 14:18.05 | [TK]D-Fender | Zeeek: That should be externhost <- |
| 14:18.16 | Zeeek | what? Since when? |
| 14:18.18 | *** join/#asterisk umay (n=chris@71-208-191-53.hlrn.qwest.net) |
| 14:18.25 | [TK]D-Fender | Zeeek: And in all likelyhood should have an externrefresh to match |
| 14:18.35 | [TK]D-Fender | Zeeek: Since FOREVER |
| 14:18.41 | Zeeek | That's from the original 1.4 config file |
| 14:18.54 | Zeeek | I must have filled out the wrong var |
| 14:18.58 | *** join/#asterisk Putzz (n=me@74.120.140.142) |
| 14:19.12 | *** join/#asterisk coppice (n=chatzill@10.198.17.210.dyn.pacific.net.hk) |
| 14:19.19 | [TK]D-Fender | Zeeek: Oh, you subbed it. |
| 14:19.31 | [TK]D-Fender | Zeeek: I read it as a host name. if that's an IP then you're fine. |
| 14:19.48 | Zeeek | I ain't no n00b |
| 14:19.50 | [TK]D-Fender | Zeeek: Next time fake a number USING a number! |
| 14:19.56 | Zeeek | ip.adr means.... |
| 14:20.06 | vutamhoan | [TK]D-Fender: Have you got any suggestion with my TE110P? |
| 14:20.18 | Zeeek | [TK]D-Fender is very busy right now |
| 14:20.32 | [TK]D-Fender | Zeeek: And what is the problem? |
| 14:20.35 | Zeeek | The only thi,ng I can figure is a router weirdness |
| 14:20.45 | Zeeek | qualif makes everyone UNREACHABLE |
| 14:21.01 | Zeeek | when commented out, they're just Unmonitored |
| 14:21.16 | Zeeek | every peer except one works on my 1.2 box |
| 14:21.19 | [TK]D-Fender | vutamhoan: dunno.... |
| 14:21.30 | Zeeek | I checked the config head to head |
| 14:22.22 | basty | [TK]D-Fender: Same with the pickup...when i use pickup() with "*8" i see "*8" in the display..not the incoming callerid. |
| 14:22.36 | [TK]D-Fender | Zeeek: [SipuraL2] should be fine with qualify.... |
| 14:22.46 | Zeeek | It is on the other box |
| 14:22.53 | [TK]D-Fender | basty: thats because you are dialing OUT. |
| 14:23.06 | [TK]D-Fender | Zeeek: Sounds like a chan_sip bug to me then. |
| 14:23.15 | Zeeek | and TK, your beloved Polycom acts the same way. |
| 14:23.17 | *** join/#asterisk voipnet-tech (n=voipnet-@rrcs-24-97-250-50.nys.biz.rr.com) |
| 14:23.21 | basty | [TK]D-Fender: Any workaround ? |
| 14:23.22 | [TK]D-Fender | basty: Realize the direction of your calls. |
| 14:23.30 | [TK]D-Fender | basty: Get coding. |
| 14:23.43 | Zeeek | [TK]D-Fender you think I should submit it? No one else seems to have the pb |
| 14:23.55 | [TK]D-Fender | Zeeek: Your call |
| 14:24.08 | Zeeek | I'll do a debug of the Poly and pasebin it |
| 14:24.13 | Zeeek | just for fun ;) |
| 14:24.59 | [TK]D-Fender | Zeeek: Is the remote phone port forwarded or have any NAT keep-alive of its own in effect? |
| 14:25.00 | *** join/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net) |
| 14:25.00 | *** mode/#asterisk [+o mog] by ChanServ |
| 14:25.02 | *** join/#asterisk CunningPike (n=CunningP@dhcp-10-234.district.north-van.bc.ca) |
| 14:25.10 | basty | [TK]D-Fender: Thanks for your info ;) |
| 14:25.54 | blueneon | hmm trying to connect my SIP client to Asterisk now and getting: WARNING[30964]: chan_sip.c:1754 __sip_xmit: sip_xmit of 0x966f2b8 (len 529) to 192.168.1.251:0 returned -1: Invalid argument |
| 14:25.57 | blueneon | in the asterisk console |
| 14:26.35 | *** join/#asterisk wallyhts (n=wallyhts@74-133-144-118.dhcp.insightbb.com) |
| 14:26.39 | *** part/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 14:27.33 | Zeeek | [TK]D-Fender taking the Poly as an example, line 1 is connected to ast1.2 box. Line 2 was also connected to same. Worked fine. Line 2 connected to new 1.4 server (only the server was changed) now will not qualify |
| 14:28.16 | wallyhts | I have several asterisk developer positions PM if interested |
| 14:29.24 | *** join/#asterisk ivanfm (n=ivanfm@c93481ec.virtua.com.br) |
| 14:31.37 | *** join/#asterisk stack_ (n=stack@198.30.100.203) |
| 14:31.57 | [TK]D-Fender | Zeeek: And the answer to my question... ? |
| 14:32.04 | Zeeek | [TK]D-Fender should the Poly port be set to something other than 5060 if the two lines are different servers? Maybe that's it? |
| 14:32.14 | Zeeek | yes, NAT and forwarded |
| 14:32.16 | Zeeek | but |
| 14:32.36 | [TK]D-Fender | Zeeek: No NOT forward to the phone while using qualify. |
| 14:32.37 | Zeeek | now that I see ast trying to send to 5060, maybe there's confusion there |
| 14:32.55 | stack_ | I have two asterisk boxes and I am trying to route calls from one box to the other via IAX2. When I attempt to route a call to the default context specified for the IAX account, it works fine, but when I try to route to something directly to a different context, it's rejected... what am I doing wrong? |
| 14:32.56 | Zeeek | I repeat this all worked before |
| 14:33.02 | Zeeek | but I'll look at router again |
| 14:33.36 | [TK]D-Fender | Zeeek: I had my IP 501 reg to my local * and remote to another jsut fine. |
| 14:34.18 | Zeeek | That's what I would have thought |
| 14:34.43 | Zeeek | in fact, line three is connected to a third server (why have three ines otherwise?) |
| 14:35.15 | mocker | Any recommendations on PCI FXO cards? |
| 14:35.26 | mocker | Digium still the way to go, or is another company doing them better now? |
| 14:35.45 | mocker | I notice Rhino has some.. |
| 14:36.03 | Zeeek | [TK]D-Fender take a look at this: http://pastebin.ca/564236 |
| 14:36.20 | [TK]D-Fender | mocker: Sangoma A200d FTW |
| 14:36.36 | pj_ | I will be setting up an * server behind a redfone box... I was wondering if anyone knew about how well vmwared * worked with TDMoE ? |
| 14:36.47 | mocker | [TK]D-Fender: Personal experience? |
| 14:37.38 | Zeeek | wait a minute I think I see something |
| 14:37.38 | [TK]D-Fender | Zeeek: # User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.4.1 <--------- ICk |
| 14:38.01 | Zeeek | works fine everywhere else |
| 14:38.19 | [TK]D-Fender | mocker: All but 1 installation of mine either started with, or was forced to switch to running one. |
| 14:38.59 | [TK]D-Fender | Zeeek: If you are forwarding to that phone, STOP |
| 14:39.28 | [TK]D-Fender | Zeeek: and I see it using an IP to contact, AND a domain |
| 14:39.57 | Zeeek | yes, where is the domain thing coming from (what setting) |
| 14:40.54 | *** join/#asterisk rmayorga (i=rmyorg@unaffiliated/rmayorga) |
| 14:42.24 | [TK]D-Fender | pj_: Oh god... that sounds like a nightmare waiting to happen..... |
| 14:42.31 | Zeeek | removing the port forward cut off the other server |
| 14:42.51 | Zeeek | IOW, without 5060 to the phone, nothing works |
| 14:42.53 | *** join/#asterisk wunderkin (i=wunderki@ip68-104-149-97.ph.ph.cox.net) |
| 14:42.56 | Zeeek | this is double NAT |
| 14:43.44 | [TK]D-Fender | Zeeek: draw the path |
| 14:44.12 | Zeeek | PolyL2 ->NAT->Internet->NAT -> asterisk |
| 14:44.16 | mocker | [TK]D-Fender: http://www.voipsupply.com/product_info.php?products_id=1336 |
| 14:44.22 | mocker | Does that look like the one you are talking about? |
| 14:44.28 | mocker | A200D pulled up no results. |
| 14:44.30 | [TK]D-Fender | Zeeek: not the problem. |
| 14:44.52 | Zeeek | [TK]D-Fender this config has worked from 1.0 to 1.whatever three years |
| 14:45.05 | Zeeek | dunno why |
| 14:45.08 | perf3kt | does NAT only prevent calls coming into the aterisk box? |
| 14:45.09 | Zeeek | the other server forward |
| 14:45.20 | Zeeek | the other server qualifies with no problem |
| 14:45.25 | [TK]D-Fender | mocker: http://www.voipsupply.com/index.php?cPath=99_359_360 <- the ones with HWEC |
| 14:45.35 | *** join/#asterisk [GuS] (n=gdnet@unaffiliated/gus/x-663402) |
| 14:45.42 | [GuS] | Hello People :) |
| 14:45.48 | [TK]D-Fender | Zeeek: No idea... |
| 14:45.58 | [GuS] | i have a problem, if someone could guide me, i will be glad: |
| 14:45.59 | [TK]D-Fender | perf3kt: NAT can usually be worked around just fine. |
| 14:46.10 | [GuS] | i've install asterisk, latest 1.4x, configured users, and so on... |
| 14:46.26 | [GuS] | when i connect from my local network, with any users, its connects very well |
| 14:46.47 | [GuS] | but when connect from outside my newwork i get this many times, and cost to much to login: [Jun 13 14:44:40] NOTICE[2075]: chan_sip.c:14606 handle_request_register: Registration from '<sip:nobody@gdnet.com.ar>' failed for '190.48.75.117' - No matching peer found |
| 14:47.22 | [GuS] | and indeed "nobody" user does not exist... but i wonder why appears nobody, cause i am login with my user |
| 14:47.41 | mocker | [TK]D-Fender: You always recommend the $$$ things. |
| 14:47.45 | mocker | ;) |
| 14:48.15 | [TK]D-Fender | mocker: I recommend things you won't regret later. |
| 14:49.01 | [TK]D-Fender | [GuS]: "190.48.75.117'" is NOT the [whateverentry] for your phone in sip.conf or wherever you configured it. |
| 14:49.53 | [GuS] | indeed i use my host, gdnet.com.ar |
| 14:50.06 | Zeeek | What causes this line: "Authorization: Digest username" in debug? |
| 14:50.14 | [GuS] | and is configured in the sip.conf |
| 14:50.15 | Zeeek | I have it on one server but not the other |
| 14:50.16 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 14:50.49 | [TK]D-Fender | [GuS]: You put an IP where you should put the ACCOUNT NAME. |
| 14:50.56 | [TK]D-Fender | [GuS]: fix your PHONE. |
| 14:51.11 | [GuS] | ? |
| 14:51.12 | blueneon | when a call comes in, i want asterisk to dial my zap channel and a sip channel, how would i do that? |
| 14:51.21 | [GuS] | i didnt used the IP on my account name [TK]D-Fender |
| 14:51.21 | [TK]D-Fender | blueneon: |
| 14:51.26 | [TK]D-Fender | blueneon: "show application dial" |
| 14:51.38 | blueneon | i know how to use dial :P |
| 14:51.49 | blueneon | just dunno how to make it dial two chans at the same time |
| 14:52.02 | [TK]D-Fender | [GuS]: NOTICE[2075]: chan_sip.c:14606 handle_request_register: Registration from '<sip:nobody@gdnet.com.ar>' failed for '190.48.75.117' - No matching peer found <- see the IP there? Thats the username * sees coming in from your phone. |
| 14:52.14 | [TK]D-Fender | blueneon: READ THE INSTRUCTIONS. |
| 14:52.21 | blueneon | okies |
| 14:52.22 | blueneon | :P |
| 14:52.57 | [GuS] | then why someties i could login and sometimes not? |
| 14:53.20 | blueneon | [&SIP/blah] ta |
| 14:54.04 | Zeeek | what does this mean and where is it generated? (from sip debug: Supported: replaces) |
| 14:54.31 | s0ck | looking at getting some english voices for * |
| 14:54.34 | s0ck | cept |
| 14:54.42 | s0ck | they are still calling the # a 'pound' key :s |
| 14:55.17 | e-ddie | yeah... the £ is the lame key |
| 14:55.23 | *** join/#asterisk EricL (n=eric@74.9.83.194) |
| 14:55.38 | s0ck | wonder if it's political correctness or something :P |
| 14:55.52 | EricL | Can someone help me with callerid issues. I posted a message on asterisk-users and got no response. |
| 14:55.59 | s0ck | trying to avoid the word 'hash' |
| 14:56.10 | EricL | I can't figure out why the CALLERID(name) gets passed to the phone and not the CALLERID(number). |
| 14:56.24 | EricL | But the CALLERID(number) makes it into the cdr. |
| 14:57.06 | [TK]D-Fender | s0ck: Sounds like you are referring to BRITTISH rather... |
| 15:00.51 | shay|work | If I have an ISDN (HiSax, ISDN4Linux) card, which would be the channel configuration file for it? modems.conf? |
| 15:00.56 | mocker | s0ck: I thought it was an octothorpe |
| 15:01.06 | mocker | :) |
| 15:01.12 | zeeesh | getting problem installing perl [root@localhost asterisk-perl-0.05]# perl Makefile.PL |
| 15:01.14 | zeeesh | Warning: prerequisite Net::Telnet 0 not found. |
| 15:01.14 | zeeesh | Writing Makefile for asterisk-perl |
| 15:01.25 | [GuS] | anyway. thanks [TK]D-Fender i will try to solve it myseft :D |
| 15:01.42 | [GuS] | another thing, when i enmable videosupport i get this when i try to call a user: |
| 15:01.44 | [GuS] | [Jun 13 14:59:36] WARNING[4567]: chan_sip.c:4828 process_sdp: Error in codec string 'eo 10702 RTP/A' |
| 15:03.55 | mocker | zeeesh: perl -MCPAN -e 'install Net::Telnet' |
| 15:04.05 | mocker | or something like that.. |
| 15:04.27 | s0ck | mocker: who?! :D |
| 15:04.45 | s0ck | british/english/etc |
| 15:04.54 | s0ck | 'pound' key is just blatantly yank tho |
| 15:05.11 | *** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust826.leed.cable.ntl.com) |
| 15:05.33 | [TK]D-Fender | s0ck: wE'RE USED TO ivr'S ON THIS SIDE OF THE WORLD... WE pound ON THAT KEY all the time! |
| 15:05.42 | [TK]D-Fender | Silly caps-lock... |
| 15:07.40 | *** join/#asterisk demlak (i=demlak@schwarz-pUnK.de) |
| 15:07.42 | demlak | hi |
| 15:08.09 | *** join/#asterisk blueneon` (i=hfklows@dsl-146-31-112.telkomadsl.co.za) |
| 15:09.53 | blueneon` | ok i've registered the sip chan (idefisk) with asterisk, and all is groovy, i can dial out and receive calls via idefisk, but one big prob.. i can hear the caller on the other end, but they cant hear me... and its not my pc/mic etc as skype etc from the same pc using the same headset works... in fact even idefisk shows the progress type thingy moving as i talk so its detecting my voice, yet nothing seems to be sent to asterisk |
| 15:09.56 | blueneon` | any ideas |
| 15:09.56 | blueneon` | > |
| 15:09.57 | blueneon` | ? |
| 15:12.03 | waKKu | r u using NAT ? |
| 15:12.20 | waKKu | did u allow ports 10000:20000 in your firewall ? |
| 15:12.40 | demlak | anyone know something like this, to get possible words matching a phone number? http://www.phonespell.org/ |
| 15:12.51 | demlak | this site has not much words |
| 15:12.58 | demlak | maybe there is a tool? |
| 15:13.01 | *** join/#asterisk bintut (n=bintut@cm131.gamma180.maxonline.com.sg) |
| 15:13.08 | demlak | and maybe it has german words? =) |
| 15:13.18 | blueneon` | hmm, let me check the ports |
| 15:13.20 | bintut | hello all.. |
| 15:14.04 | [TK]D-Fender | demlak: Get coding. This is realtively easy. |
| 15:14.34 | bintut | i'm having a problem with echo here. i followed the FAQs and HOWTOs but still, with no luck. i got here a Digium Dev Kit with 1 FXO and 1 FXS on my Debian Etch i386. info of my server is at http://paste.debian.net/30410 |
| 15:14.56 | [TK]D-Fender | waKKu: Port forwarding may not be necessary. you did not confirm his network scenario or which side if any is behind NAT. |
| 15:15.09 | *** join/#asterisk ifnotwhynot (n=davidh@c1-29-15.rrba.isadsl.co.za) |
| 15:15.21 | ifnotwhynot | hi there |
| 15:15.24 | demlak | [TK]D-Fender i don´t want to learn coding for such a thing.. i know there are tools.. got one years ago.. |
| 15:15.37 | [TK]D-Fender | demlak: Get googling then. |
| 15:15.45 | demlak | ghdfgkfk |
| 15:15.47 | demlak | i did... |
| 15:15.47 | ifnotwhynot | need some help finetuning the System() command please |
| 15:16.07 | [TK]D-Fender | bintut: Played around with fxotune, gains, recompiled zaptel with different ec, etc? |
| 15:16.08 | demlak | i am not such an fucking idiot who is asking before... i was searching a lot before i was asking here |
| 15:16.17 | Zeeek | oh oh |
| 15:16.38 | [TK]D-Fender | demlak: I've never actually heard anybody ask about something like this in my years of being here. |
| 15:16.59 | demlak | and thats the reason for sending me too google? |
| 15:17.04 | demlak | -o |
| 15:17.44 | *** part/#asterisk Zeeek (n=IceChat7@pdpc/supporter/active/Zeeek) |
| 15:17.57 | demlak | well.. forget it... im out |
| 15:17.58 | *** part/#asterisk demlak (i=demlak@schwarz-pUnK.de) |
| 15:18.09 | [TK]D-Fender | demlak: well if you've searched the WIKI, yeah... |
| 15:18.16 | jkiff | What a d-bag. |
| 15:18.31 | jkiff | Last time I had to do that, I sat down in front of a keypad and used my brain. |
| 15:19.16 | [TK]D-Fender | jkiff: Well he is not an idiot yet, just unjustifiably miffed about not being handed code for his exact problem that noone has discussed in here in my long stay... |
| 15:19.41 | *** join/#asterisk JerJer (n=PhatJ@pdpc/supporter/bronze/jerjer) |
| 15:19.53 | *** join/#asterisk A-Data (n=asd@217.54.224.153) |
| 15:20.03 | [TK]D-Fender | jkiff: "yes, clearly everyone has coded whatever YOU want for * in a ready-to-use-with-no-IQ capacity! Download here!" |
| 15:20.19 | A-Data | hello If we made a voice Menu .. how can we listen it internal ??? or we have to dial it from outside the system |
| 15:20.40 | [TK]D-Fender | A-Data: Dial an exten that leads to it |
| 15:20.54 | [TK]D-Fender | A-Data: to *, a call is a call is a call is a call... |
| 15:21.07 | [TK]D-Fender | A-Data: There is no concept of "outside" |
| 15:21.11 | bintut | [TK]D-Fender: can you point me to a good discussions about fxotune? |
| 15:21.42 | [TK]D-Fender | bintut: Not offhand... WIKI and mailing lists, and other misc articles are out there, but I don't buy non-EC cards. |
| 15:22.02 | bintut | [TK]D-Fender: what's a "non-EC" cards? |
| 15:22.07 | Qwell[] | echo can |
| 15:22.11 | ifnotwhynot | i receive a fax but need to send it via sendmail using the System() command I have no problem sending a sample file on my root directory(s,1,answer();s,2,System(email -s "test" -a /var/test.jpg -h dave@testserver.com) the problem is where i am receiving a fax using s,1,Answer();s,2,Set(FAXFILE=/var/spool/asterisk/fax/${UNIQUEID}.tif) I AM having problems attaching the tif file to my command in System()?? someone please point me in the rig |
| 15:22.14 | [TK]D-Fender | bintut: Cards without hardware echo cancellation. |
| 15:22.25 | Qwell[] | [TK]D-Fender: Have you tried the HPEC? |
| 15:23.05 | bintut | [TK]D-Fender: so you mean that you have an EC module on your cards? |
| 15:23.10 | [TK]D-Fender | Qwell[]: Nope, no need.... |
| 15:23.14 | [TK]D-Fender | bintut: Correct |
| 15:23.16 | Qwell[] | [TK]D-Fender: figured |
| 15:23.24 | odity | hi does anyone haw a link to a good realtime tutorial? |
| 15:23.33 | Qwell[] | ~wikis |
| 15:23.34 | jbot | somebody said wikis was http://www.voip-info.org |
| 15:23.35 | Qwell[] | ~book |
| 15:23.36 | jbot | [book] a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 15:23.37 | *** join/#asterisk hfb (n=hfb@pool-72-87-254-188.lsanca.dsl-w.verizon.net) |
| 15:23.38 | Qwell[] | odity: both of those |
| 15:23.50 | odity | oh thanks |
| 15:25.05 | A-Data | [TK]D-Fender can i pvt u? |
| 15:25.20 | bintut | i have to research more.. gtg for now.. |
| 15:25.24 | [TK]D-Fender | A-Data: just ask here unless you're looking to hire me as a consultant. |
| 15:26.10 | A-Data | ok [TK]D-Fender we want our client to dial a number then he get the voice menu how can we do it? |
| 15:27.16 | A-Data | i mean he dials the number from the PSTN ... then Astrisk recive the call and play the voice menu |
| 15:28.35 | [TK]D-Fender | A-Data: How have you arranged for * to receive calls from the PSTN? |
| 15:29.09 | *** join/#asterisk rmayorga (i=rmyorg@unaffiliated/rmayorga) |
| 15:29.41 | A-Data | you mean incoming call DID? |
| 15:30.25 | [TK]D-Fender | A-Data: I mean WHATEVER. How do you plan of connecting your * to the PSTN. there are a number of ways. a DID is not a "thing" you can connect to *. |
| 15:31.04 | ifnotwhynot | got it to work |
| 15:31.09 | ifnotwhynot | yes yes yes |
| 15:31.35 | [TK]D-Fender | A-Data: usual options : hardware card to which you connect an analog or digital line of some kind. Or use an ITSP (who has this equipment) and sends you the call via a VoIP protocol. |
| 15:31.39 | zeeesh | what is "ncftpget" and normally where it located ? getting problem "Warning: ncftpget not found in PATH"? |
| 15:32.26 | zeeesh | Where is your ncftpget program? []? |
| 15:32.34 | zeeesh | installing perl agi |
| 15:32.50 | [TK]D-Fender | zeeesh: http://www.ncftp.com/ncftp/doc/ncftpget.html |
| 15:32.56 | A-Data | [TK]D-Fender hardware card |
| 15:33.18 | [TK]D-Fender | zeeesh: I found that in 5 SECONDS of googling, and clearly this is just a program you should know how to acquire for your distro. |
| 15:33.30 | [TK]D-Fender | A-Data: Ok, have you bought & installed it already? |
| 15:33.42 | *** join/#asterisk adker (n=chatzill@74-33-221-202.br1.glv.ny.frontiernet.net) |
| 15:33.45 | A-Data | [TK]D-Fender no net yet |
| 15:33.55 | zeeesh | ok |
| 15:33.58 | A-Data | we are just trying things and planning |
| 15:34.04 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 15:34.10 | [TK]D-Fender | A-Data: Ok, well once you di then you can set it up and have * process incoming calls any which way you like. |
| 15:34.37 | [TK]D-Fender | A-Data: int he meantime you can jsut test with soft-phones. |
| 15:34.53 | *** join/#asterisk blueneon (n=blueneon@dsl-146-31-112.telkomadsl.co.za) |
| 15:35.07 | A-Data | ok [TK]D-Fender thank you . |
| 15:35.48 | *** join/#asterisk RoyK (n=roy@ti211310a080-5282.bb.online.no) |
| 15:35.48 | blueneon | i've managed to get SIP working with asterisk, but the caller on the other end says my volume is WAY to low, they can hardly hear me... let i've set everything on full... is there a setting in asterisk i can change for the vol of SIP chans? |
| 15:36.05 | blueneon | i tried rxgain and txgain and that seemed to just break it |
| 15:36.07 | blueneon | :/ |
| 15:36.19 | *** join/#asterisk keulin (n=cray@ifth-pdcr2.infotheme.fr) |
| 15:38.44 | *** part/#asterisk Dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 15:40.48 | *** join/#asterisk VJFROMGT (n=vjfromgt@user-387g9ui.cable.mindspring.com) |
| 15:41.21 | A-Data | [TK]D-Fender which soft-phone support DMTF during call i tried 3cx but don`t work |
| 15:42.31 | [TK]D-Fender | A-Data: Every one I've seen does. on Windows : X-Lite , idefisk . For *nix : kphone , ekiga |
| 15:42.41 | VJFROMGT | I want to run asterisk on a wrt54gs , how can i tell what the max channels i can get out of it? |
| 15:42.54 | [TK]D-Fender | blueneon: there is no rxgain/txgain for SIP. |
| 15:43.18 | [TK]D-Fender | blueneon: what are you talking on? |
| 15:43.34 | A-Data | are here skype to * bridge ? |
| 15:43.42 | [TK]D-Fender | A-Data: No. |
| 15:43.44 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 15:46.41 | blueneon | Fender i think i found the issue, IDEFisk doesnt have Mic Boost on be default |
| 15:49.41 | [TK]D-Fender | blueneon: Your mixer is your OS's job to handle. |
| 15:53.35 | *** join/#asterisk KnaveMan (n=asdf@mail.centralfloridatechnology.com) |
| 15:54.09 | russellb | Nugget: <3 |
| 15:56.47 | *** join/#asterisk angryuser (n=Miranda@df01t2-62-34-180-141.d4.club-internet.fr) |
| 15:56.57 | ifnotwhynot | i |
| 15:57.20 | angryuser | the module for cdr_mysql.conf names cdr_mysql.so? |
| 15:57.24 | angryuser | *name |
| 15:57.28 | KnaveMan | Hello everyone, im running into an interesting issue which I could use some help with.... I have 3 SIP hardphones in the office. I opened up my SIP, IAX, IAX2, RTP ports in my ISA firewall (making sure to select UDP for the appropriate ports). I am now trying to configure a softphone outside of the office to connect in. My problem is that when I do ngrep -d any -Wbyline -t port 5060, i see an error saying that im Unauthorized. Any sugg |
| 15:57.59 | Corydon76-home | angryuser: cdr_addon_mysql.so |
| 15:58.53 | angryuser | hm strange i have compiled addons, but dont have this one |
| 15:58.55 | *** join/#asterisk Nobbie (n=anony@fwb003.fw.is.co.za) |
| 15:58.56 | Nobbie | hi. |
| 16:00.43 | krdian_ | angryuser: do you have mysql devel ? |
| 16:01.44 | krdian_ | angryuser: if your system doesn't have devels then doesn't compile this module |
| 16:02.16 | angryuser | krdian_: hm i have errors now |
| 16:02.29 | angryuser | [mxml/libmxml.a] Erreur 2 |
| 16:02.51 | angryuser | xml lib ? |
| 16:03.19 | krdian_ | yes |
| 16:03.34 | krdian_ | devel too |
| 16:03.59 | krdian_ | in debian for ex. is called libmxml-dev |
| 16:04.28 | angryuser | krdian_: i cant find the paquage mysql-dev hm |
| 16:04.55 | angryuser | found mysql-client |
| 16:04.57 | Nobbie | how does * 1.4 genius introduction of call-limit fit into call waiting and line settings on a device ? |
| 16:05.48 | krdian_ | angryuser: libmysqlclient-dev ? |
| 16:05.57 | *** join/#asterisk achu (n=achu@61.17.217.185) |
| 16:06.14 | Qwell[] | krdian_: where were you yesterday when I was trying to find that package? |
| 16:06.26 | achu | When I call from outside I am getting callerid as "Unknown" , How can I solve it ? |
| 16:06.31 | *** join/#asterisk matdon (n=matdon@eagle.bsd.st) |
| 16:06.32 | matdon | hi |
| 16:07.07 | krdian_ | Qwell[]: probably in bed, i'm sick :( |
| 16:07.10 | achu | It was working good , suddenly this happens |
| 16:07.26 | tzafrir | libmysqlclient15-dev ? |
| 16:07.41 | Qwell[] | tzafrir: libmysqlclient-dev is a meta package |
| 16:08.07 | Qwell[] | or, virtual, or whatever you Debian folks call them... |
| 16:08.48 | angryuser | i have installed libmxml-dev but still have this on make menuselect http://www.pastebin.ca/564401 |
| 16:11.00 | angryuser | strange |
| 16:11.11 | *** join/#asterisk hmm-home (n=hmm-home@24-119-176-74.cpe.cableone.net) |
| 16:11.48 | angryuser | can someone help me on this one? |
| 16:12.02 | hmm-home | with? |
| 16:12.33 | angryuser | http://www.pastebin.ca/564401 make in asterisk-addons output libmxml-dev installed |
| 16:13.04 | krdian_ | angryuser: thats all output ? |
| 16:14.09 | russellb | for (next = ast_channeltype_list(); next; next = next->next) |
| 16:14.11 | russellb | oops. |
| 16:14.15 | angryuser | krdian_: http://www.pastebin.ca/564421 complete one |
| 16:14.19 | achu | please help me to solve the callerid as "Unknown" problem |
| 16:14.34 | achu | I checked zapata.conf |
| 16:14.38 | achu | all seems good |
| 16:14.48 | achu | any other options I have to check ? |
| 16:16.35 | angryuser | krdian_: i have installed libmxml1 and libmxml-dev |
| 16:16.36 | [TK]D-Fender | achu: You TELL us its "all good". Why should we believe that? And since you've shown us nothing, what do you expect us to tell you? |
| 16:16.44 | angryuser | krdian_: ./configure pass well |
| 16:17.07 | achu | [TK]D-Fender, What options I have to check for showing the caller id correctly ? |
| 16:17.39 | [TK]D-Fender | achu: Rather than have us waste time guessing where you went wrong how about PASTEBIN -ing your configs so we can see. |
| 16:17.41 | [TK]D-Fender | ~pb |
| 16:17.43 | jbot | i guess pb is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste, or at paste.lisp.org |
| 16:17.46 | [TK]D-Fender | achu: ---^ |
| 16:17.49 | *** join/#asterisk joe-f (n=joef@c-71-201-188-239.hsd1.il.comcast.net) |
| 16:17.58 | joe-f | Would anyone know the best way to calculate the time in seconds of overlap, here's my array: http://hashbin.com/68d.html I should get 2700 seconds of overlap.. |
| 16:18.15 | KnaveMan | Anyone have any ideas to my problem with this whole 401 Unauthorized error? |
| 16:18.23 | RoyK | http://callweaver.org/blog/9 |
| 16:18.24 | joe-f | I know its php, but i'm thinking someones had to do something similar |
| 16:18.29 | *** join/#asterisk shido6 (i=shido6@d221-68-200.commercial.cgocable.net) |
| 16:18.58 | krdian_ | angryuser: hmmm, "6. autoconf: no input file" |
| 16:19.57 | krdian_ | angryuser: i'll try to compile those addons |
| 16:21.41 | angryuser | krdian_: dont bother i have redownloaded the paskage, it is working |
| 16:22.18 | krdian_ | angryuser: :) |
| 16:22.26 | pj_ | I will be setting up an * server behind a redfone box... I was wondering if anyone knew about how well vmwared * worked with TDMoE ? Anyone ? :' |
| 16:22.27 | hmm-home | anyone use vmware in linux? |
| 16:22.36 | angryuser | krdian_: make clean does not clean everything in addons |
| 16:22.37 | jkiff | Hrm, so they don't make spa-2000's anymore? What would be a good substitute? |
| 16:26.32 | [TK]D-Fender | jkiff: SPA-2102 |
| 16:27.12 | *** join/#asterisk blueneon (n=blueneon@dsl-146-31-112.telkomadsl.co.za) |
| 16:27.31 | blueneon | other than idefisk what other good SIP softphones are avail. for free? |
| 16:27.47 | [TK]D-Fender | blueneon: X-Lite |
| 16:27.49 | blueneon | (got disconnected so i cant see the other option given to me earlier :() |
| 16:28.38 | blueneon | kk |
| 16:29.47 | coppice | is wengophone really a free softphone with G.729 and AMR inside? the blurbs seems to indicate it is |
| 16:29.50 | *** join/#asterisk Dovid (n=Dovid@bzq-88-155-87-253.red.bezeqint.net) |
| 16:30.32 | Dovid | anyone know of a good snmp server to install on a box for monitoring ? (the SNMP config file on centos has my head spinning). |
| 16:32.11 | *** join/#asterisk mutilator (n=WebChat@the.drinkproject.com) |
| 16:32.12 | bmd | coppice: IIRC wengophone uses the ipp g.729 |
| 16:33.11 | coppice | they must have some devious way to deal with the licencing through their owners |
| 16:34.16 | bmd | I believe the impetus is on the user to compile it in |
| 16:34.49 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-88-153-47-145.red.bezeqint.net) |
| 16:35.17 | *** part/#asterisk macli (n=macli@nmc.brc.ubc.ca) |
| 16:35.32 | coppice | what about AMR, though. I don't know of one people can just download and run. the 3GPP reference code isn't in that form |
| 16:36.26 | bmd | dunno about amr |
| 16:37.54 | *** join/#asterisk Op3r (n=op3r@125.212.50.108) |
| 16:38.46 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 16:40.33 | *** join/#asterisk penguinFunk (n=penguin@unaffiliated/penguinfunk) |
| 16:41.28 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-88-153-47-145.red.bezeqint.net) |
| 16:48.20 | rene- | achu: R2 caller id? |
| 16:48.34 | *** join/#asterisk Dr-Linux (n=Nothing@202.125.139.198) |
| 16:48.39 | Dr-Linux | http://phpfi.com/241500 |
| 16:48.45 | Dr-Linux | please someone have a look |
| 16:49.03 | Dr-Linux | i'm trying to compile zaptel 1.4.3 and getting this error |
| 16:49.29 | *** join/#asterisk ManxPower (n=manxpowe@185.sub-70-220-226.myvzw.com) |
| 16:50.17 | *** join/#asterisk grndslm (n=grndslm@24-116-87-97.cpe.cableone.net) |
| 16:51.27 | file | Dr-Linux: your copy of make is old, update it |
| 16:52.31 | Dr-Linux | file: my OS make is old or Zaptel package Makefile? :S |
| 16:52.46 | file | the version of GNU Make that is installed is too old |
| 16:54.06 | *** join/#asterisk logyati (n=suporte@201.29.73.49) |
| 16:54.18 | RoyK | file, Dr-Linux: did you see http://callweaver.org/blog/9 ? |
| 16:54.43 | Dr-Linux | RoyK: no yet dude, lemme see |
| 16:56.52 | Dr-Linux | ohh, it looks new pbx |
| 16:57.23 | RoyK | the fork was long back, to openpbx, but a lot of things have happened since then |
| 16:57.30 | logyati | hey guys, im reading a lot about extensions.conf |
| 16:57.32 | RoyK | from asstrix to openpbx |
| 16:58.10 | logyati | but i didnt understand how it works when i call to asterisk from a pstn phone |
| 16:58.28 | *** join/#asterisk DaveCanoe (n=Dave@H6.C30.B96.tor.eicat.ca) |
| 16:58.41 | russellb | RoyK: that's quite off topic. |
| 16:59.28 | [TK]D-Fender | logyati: There is clearly something between * and this "analog phone" that allows them to talk. What is it? |
| 16:59.30 | De_Mon | openpbx changed its name again? |
| 16:59.58 | logyati | i mean, i want asterisk to pick up the phone, than wait till i type a number, then it calls to s ip phone |
| 17:00.30 | De_Mon | retrospectiva looks suspeciously like trac |
| 17:00.31 | logyati | [tk]d-fender hmm what you mean? |
| 17:00.32 | *** join/#asterisk Cresl1n (i=matt@nat/digium/x-b34e15566812e1d1) |
| 17:00.32 | *** mode/#asterisk [+o Cresl1n] by ChanServ |
| 17:01.02 | *** join/#asterisk mindCrime_ (n=chatzill@66.83.208.219.nw.nuvox.net) |
| 17:01.16 | De_Mon | ahh its ruby, yeah I was looking at this a while ago... |
| 17:01.31 | logyati | i have a POT line plugged in a fxo port of my tdm400p, the number of this line is 4363 |
| 17:01.32 | [TK]D-Fender | logyati: An analog phone by itself has not way of talking to *. This requires hardware and protocols. What is it that you are using that would ALLOW * to talk to this "analog phone"? |
| 17:01.38 | RoyK | russellb: :) |
| 17:01.40 | [TK]D-Fender | logyati: TDM400P, thanks. |
| 17:01.53 | *** join/#asterisk Zion800 (n=None@cpe-76-167-156-224.socal.res.rr.com) |
| 17:02.38 | [TK]D-Fender | logyati: Well you pick a context in zapata.conf for your channel and configure that context in extensions.conf. in there you do "exten => s,1,....... and do more stuff" and do whatever you want from there. |
| 17:02.42 | logyati | i want to call to 4363 from any other pot line |
| 17:02.55 | zeeesh | <PROTECTED> |
| 17:02.56 | zeeesh | Going to read /usr/src/perll/asterisk-perl-0.05/yes/sources/authors/01mailrc.txt.gz |
| 17:02.56 | zeeesh | yes: invalid option -- - |
| 17:02.56 | zeeesh | Try `yes --help' for more information. |
| 17:02.56 | zeeesh | "? |
| 17:03.29 | zeeesh | more than 10 lines followed these |
| 17:03.33 | logyati | [tk]d-fender ah, the name of the context inside zapata.conf has to be the same inside extensions.conf? |
| 17:03.36 | shido6 | pastebin.ca |
| 17:04.14 | _VoiceMeUp_COM | TK d always around to help.. |
| 17:04.27 | logyati | [tk]d-fender ok its the same now... so, i tried many things to make it work, first line is exten => s,1,Answer right? |
| 17:05.17 | [TK]D-Fender | logyati: in zapata.conf you set "context=acallfrommytdm400" and in extensions.conf [acallfrommytdm400] |
| 17:06.40 | logyati | [tk]d-fender hmm, but i wanna make a call TO tdm400p and not from, understood? |
| 17:07.12 | logyati | [tk]d-fender i created a context=from-pstn |
| 17:07.14 | *** part/#asterisk RoyK (n=roy@ti211310a080-5282.bb.online.no) |
| 17:07.38 | [TK]D-Fender | logyati: doesn't work that way |
| 17:07.54 | [TK]D-Fender | logyati: to place a call you use a DEVICE NAME, not a "context". |
| 17:08.07 | [TK]D-Fender | logyati: eg. Dial(Zap/1/1234567) |
| 17:08.36 | logyati | [tk]d-fender hmmm yes... well, im gonna pastebin my zapata.conf ok? and my extensions.conf |
| 17:08.45 | [TK]D-Fender | logyati: sure |
| 17:13.40 | *** join/#asterisk Keltus (n=Keltus@about/cooking/nakedchef/beefstew/Keltus) |
| 17:14.13 | *** join/#asterisk iBuMp (n=ReDNeQ@cpe-66-68-37-190.austin.res.rr.com) |
| 17:15.26 | *** join/#asterisk guillote_GNU (n=guillote@190.7.23.192) |
| 17:16.02 | *** join/#asterisk ReDNeQ (n=ReDNeQ@cpe-66-68-37-190.austin.res.rr.com) |
| 17:16.50 | logyati | [tk]d-fender zapata.conf -> http://www.pastebin.ca/564537 and extensions.conf -> http://www.pastebin.ca/564550 |
| 17:18.56 | [TK]D-Fender | logyati: That should cause * to see the line when it rings, then answer, then immediately hangup due to autofallthrough. |
| 17:20.25 | EricL | Does anyone have any idea why * would only pass the CALLERID(name) and not the CALLERID(num) forward when it has both? |
| 17:21.24 | logyati | [tk]d-fender so what should i do? i didnt find help on this in my book :( can you update the paste? |
| 17:22.37 | [TK]D-Fender | logyati: you have not made a suitable IVR at all. Not set timeout, no prompts, autofallthrough will kill your calls because you have not disabled it or used WaitExten. the 1234 you have there only has priority *2* and will never be callable in that state. |
| 17:22.46 | *** join/#asterisk rvosper (n=rvosper@site.gtn.ca) |
| 17:22.48 | [TK]D-Fender | logyati: time to hit the books and learn the BASICS. |
| 17:22.50 | [TK]D-Fender | ~book |
| 17:22.52 | jbot | book is, like, a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 17:22.52 | [TK]D-Fender | ~wikis |
| 17:22.53 | jbot | methinks wikis is http://www.voip-info.org |
| 17:23.03 | [TK]D-Fender | logyati: Start your focus on chapter 5. |
| 17:23.06 | rvosper | http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 17:23.13 | rvosper | woops sry |
| 17:24.11 | logyati | [tk]d-fender chapter 5 of this book im reading? or that one "Asterisk: The Future of Telephony"? |
| 17:24.29 | [TK]D-Fender | logyati: I just linked it to you. Do the math |
| 17:24.39 | logyati | [tk]d-fender k, ty |
| 17:27.35 | *** join/#asterisk samy_b1 (n=baind@2001:49f0:1000:0:0:0:0:7) |
| 17:28.04 | *** join/#asterisk andyd (n=andyd@213-228-240-161.dsl.prodigynet.co.uk) |
| 17:28.07 | *** join/#asterisk Hmmhesays (n=Neg@24-119-176-74.cpe.cableone.net) |
| 17:28.09 | samy_b1 | hi all |
| 17:28.11 | *** part/#asterisk andyd (n=andyd@213-228-240-161.dsl.prodigynet.co.uk) |
| 17:28.33 | Hmmhesays | hello |
| 17:29.17 | *** join/#asterisk mirco (n=mirco@88.128.38.243) |
| 17:29.25 | Zion800 | Any coders out there that wanna help me adapt a very small script to work with Asterisk 1.4? I think its only 3 lines of code...but I don't have any programming skills... |
| 17:30.07 | samy_b1 | can some one tell me how can i repalce digits on a dial plan ?? |
| 17:30.13 | Qwell[] | samy_b1: what? |
| 17:30.40 | *** join/#asterisk DJ-Pyro_ (n=abrezins@gateway.brevient.net) |
| 17:30.40 | [TK]D-Fender | Zion800: Your request for help filled 2 lines on my screen, so the overhead is a little hig. JUST ASK! :p |
| 17:30.46 | [TK]D-Fender | big* |
| 17:31.21 | DJ-Pyro_ | if there's additional information in the From field, is there any way to access that via the dial plan |
| 17:31.22 | samy_b1 | Qwell[]: well if you have a DID form bulgaria and you like to make your dila plan like A USA |
| 17:31.29 | DJ-Pyro_ | such as From: <sip:+18017349600@209.247.16.1;isup-oli=23>;tag=VPSF50603522629633 |
| 17:31.32 | jkiff | Is there a standardized length of time between the beeps that a fax machine makes that NVFaxDetect() uses to detect faxes? Right now I'm having it listen for four seconds, but some of them seem to sneak past exten=>fax. |
| 17:31.33 | *** join/#asterisk tsurko (n=tsurko@150-190.go.evo.bg) |
| 17:31.35 | DJ-Pyro_ | getting the value of isup-oli |
| 17:31.42 | samy_b1 | for example you have to dial 011359 |
| 17:32.05 | Zion800 | :p Page.agi --> Its a script that allows asterisk to intercom only available channels. It uses the output of "Show Hints" to determine this. In Asterisk 1.4, they changed the output of "show hints" to a different format. Thats why the script is broken. |
| 17:32.41 | [TK]D-Fender | Zion800: Wow... a dirty text-parser... thats the sort of 2-bit hack *I* would do ;) |
| 17:32.45 | samy_b1 | Qwell[]: sorry i;ll start over |
| 17:32.51 | Zion800 | haha |
| 17:32.55 | Zion800 | Here it is: http://www.voip-info.org/wiki/view/Script+to+page+mixed+SIP+%252F+SCCP+system&view_comment_id=13969 |
| 17:33.03 | [TK]D-Fender | Zion800: pastebin it. |
| 17:33.21 | [TK]D-Fender | Zion800: (your script + the CLI output of "show hints" |
| 17:33.26 | samy_b1 | how can i replace 088 with 01135988 in my dial plane |
| 17:33.57 | samy_b1 | so i can dial 01135988 insted of 088 |
| 17:36.00 | Zion800 | [TK]D-Fender: http://www.pastebin.ca/564597 |
| 17:36.36 | [TK]D-Fender | samy_b1: You don't. What ever you dialed is WHAT YOU DIALED. What you choose to DO with it... thats up to you. |
| 17:36.43 | EricL | exten => *5XXX,2,Voicemail(${EXTEN:1}@voicemail,u) How come that line doesn't send all calls start with *5 and then the last 3 digits of a valid exten to the extensions vm? |
| 17:37.06 | EricL | ( Line 1 is just Answer() ) |
| 17:37.29 | samy_b1 | [TK]D-Fender: aham |
| 17:37.47 | samy_b1 | whell there was a way but i don't remember how was it |
| 17:38.19 | samy_b1 | it was somting like 011359|088+Z. |
| 17:38.26 | [TK]D-Fender | samy_b1: Your question is very poorly worded for its intent |
| 17:38.28 | samy_b1 | but i'm not sure |
| 17:38.53 | [TK]D-Fender | EricL: because that isn't a pattern match |
| 17:39.14 | Zion800 | [TK]D-Fender: Gosh!! You're one busy person around here! |
| 17:39.28 | EricL | [TK]D-Fender: It didn't work when I had the _ in front of it either though. |
| 17:39.40 | [TK]D-Fender | EricL: SHOW ME. |
| 17:39.49 | samy_b1 | [TK]D-Fender: i'm sorry but i don't know how can i describe it better. |
| 17:40.03 | EricL | exten => _*5XXX,2,Voicemail(${EXTEN:1}@voicemail,u) |
| 17:40.17 | jer | whenever i dial *67 i get a busy congested tone but nothing shows up in my asterisk console with verbosity set to 4... how can i diagnose what's going wrong? |
| 17:40.21 | *** join/#asterisk exoxe (i=exoxe@ip70-171-16-60.ga.at.cox.net) |
| 17:40.23 | [TK]D-Fender | samy_b1: then we won't really be able to help you much. Find a way to reword your problem/request. |
| 17:40.41 | samy_b1 | i'll try |
| 17:40.55 | [TK]D-Fender | EricL: No. Go change your dialplan back to the way it SHOULD be, aplly the change. test, then pastebin upon failure |
| 17:41.15 | [TK]D-Fender | Zion800: I am regex-challenged... |
| 17:41.22 | [TK]D-Fender | Zion800: But still looking at it... |
| 17:41.24 | Zion800 | hah |
| 17:41.33 | EricL | I am actually more concerned about getting the caller id to work, but I will go give it a try. |
| 17:42.08 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 17:42.17 | KnaveMan | Anyone have any ideas to my problem with this whole SIP/2.0 401 Unauthorized error? |
| 17:42.34 | KnaveMan | Or has it been too long since ive asked the question for anyone to know whta im talking about :) |
| 17:42.34 | [TK]D-Fender | KnaveMan: user / pass is bad, like the error says |
| 17:43.03 | mirco | could someone plz review my log, cause I'm having problems with incoming call's, but yesterday it worked like a charm http://pastebin.ca/564586 |
| 17:43.22 | Zion800 | [TK]D-Fender: Did you get a chance to see what I pastebined? |
| 17:43.26 | [TK]D-Fender | "the computer said 'invalid password' ... so that muct mean it doesn't work on WEDNESDAYS!" |
| 17:43.39 | [TK]D-Fender | Zion800: Indeed I did. |
| 17:43.56 | blitzrage | I DON'T WANT TO KNOW YOUR NAME! I JUST WANT... |
| 17:44.34 | KnaveMan | [TD]D-Fender: ive had no problem if i have a vpn connection.... |
| 17:44.46 | [TK]D-Fender | blitzrage: ! ! ! |
| 17:45.04 | [TK]D-Fender | KnaveMan: Something else has changed... |
| 17:45.52 | [TK]D-Fender | mirco: * seems to answer just fine. GUI's are not supported here. |
| 17:46.07 | *** join/#asterisk iBuMp (n=ReDNeQ@cpe-66-68-37-190.austin.res.rr.com) |
| 17:47.34 | mirco | [TK]D-Fender: I know it takes the call, but don't know why it refuses to put it to an extension. And yes I'm on EasyVoxBox with FreePBX 2.2.2 |
| 17:47.37 | *** join/#asterisk A-Data (n=asd@217.54.224.153) |
| 17:47.49 | A-Data | Hello All any one know Web Based SIP client |
| 17:47.53 | [TK]D-Fender | ~freepbx |
| 17:47.55 | jbot | rumour has it, freepbx is unable to be supported here. It's a complex piece of dialplan and GUI, and can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 17:47.56 | *** join/#asterisk ramindia (n=ramindia@202.63.96.9) |
| 17:47.59 | [TK]D-Fender | mirco: ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ |
| 17:48.08 | ramindia | does * act a class 5 switch ? |
| 17:48.13 | *** join/#asterisk CrashSys (n=kumba@bartleby.crashsys.com) |
| 17:48.15 | [TK]D-Fender | ramindia: No. |
| 17:48.23 | A-Data | Hello All any one know Web Based SIP client |
| 17:48.33 | [TK]D-Fender | ramindia: * has no class at all... somewhat rude that way ;) |
| 17:48.43 | ramindia | does any one know, linux based class 5 switch ? for free |
| 17:48.45 | CrashSys | Anyone have any problems with a Polycom 320/330 hanging on reboot at the "Updating Initial Settings" screen? |
| 17:48.49 | [TK]D-Fender | A-Data: Go check the WIKI's list |
| 17:49.01 | A-Data | [TK]D-Fender give me link for it please |
| 17:49.09 | [TK]D-Fender | ~wikis |
| 17:49.11 | jbot | wikis is probably http://www.voip-info.org |
| 17:49.11 | mirco | jbot: thnx, I did so, but could u tell me if it's possible that it has to do with a disturbed line |
| 17:49.22 | CrashSys | wikis... sounds like a mixed drink... |
| 17:49.26 | [TK]D-Fender | mirco: jbot is not a PERSON, and your hardware is FINE |
| 17:49.38 | ramindia | [TK]D-Fender: can you tell me , can class 5 switch available for linux for free |
| 17:49.44 | [TK]D-Fender | CrashSys: Onlyl reason I can think of is bad firware/config |
| 17:49.55 | [TK]D-Fender | ramindia: I seriously doubt one exists. |
| 17:50.03 | exoxe | is there a way for VoiceMailMain() to understand the mailbox based on the extension/SIP device called from? there's ${MACRO_EXTEN} which stores the extension being called, but is there something for the extension that called? what's the best way to accomplish this? |
| 17:50.12 | iBuMp | shouldnt * not need a firewall declaration for phones across the wan/vpn... |
| 17:50.16 | mirco | because we had very bad weather yesterday and my DSL isn't working since this event, and I also think yesterday the leading zero on my GSM number was transmitted but today it's missing |
| 17:50.18 | CrashSys | D-Fender: Hmmm... I hope it's not a sign of polycom's future... be pissed if I find out they are rebadged budgettones :D |
| 17:50.31 | CrashSys | Phone was working, now it wont... thinking it's the phone... |
| 17:50.43 | [TK]D-Fender | mirco: Your problemis your freepbx logic. We do not support that here. |
| 17:50.46 | ramindia | [TK]D-Fender: ok , any suggestion reasonable cost class5 softswitch |
| 17:50.52 | [TK]D-Fender | ramindia: No. |
| 17:51.02 | wunderkin | CrashSys, sometimes it stays there for a very long time |
| 17:51.21 | [TK]D-Fender | iBuMp: watch out for those double/triple negatives.... |
| 17:51.27 | iBuMp | hahheha |
| 17:51.32 | iBuMp | i think thats whats killin me |
| 17:51.43 | CrashSys | Maybe i'll throw a qualify in the sip.conf... see if that helps it stay connected... |
| 17:51.51 | [TK]D-Fender | exoxe: Just ask yourself what info * has about the CALLER to help ID him..... |
| 17:51.52 | [TK]D-Fender | *HINT* |
| 17:53.02 | CrashSys | well the phone worked for a day... reminds me of grandstreams :D |
| 17:53.20 | mirco | OK, but I only want to know if my ISDN line could cut of the leading zero mistakly |
| 17:53.27 | CrashSys | I guess I should have gotten the 430... stupid silver insert looks like crap... |
| 17:54.40 | [TK]D-Fender | mirco: You can already clearly see what callerID is coming in. |
| 17:55.01 | *** join/#asterisk A-data (n=asd@196.219.183.199) |
| 17:55.20 | A-data | [TK]D-Fender what is the ports i have to open to make my system Open for the internet ??? |
| 17:55.23 | [TK]D-Fender | mirco: "-- Accepting voice call from '1792615690' to 's' on channel 0/1, span 1" |
| 17:55.31 | [TK]D-Fender | A-data: ALL OF THEM |
| 17:55.45 | mirco | yeap, and I think yesterday it had a leading zero, which is missing today. And I want someone to assure that it was cut of by my carrier |
| 17:55.57 | CrashSys | D-Fender: How do I get my rum to mix with my coke while looking at an asterisk CLI? |
| 17:55.57 | [TK]D-Fender | A-data: then your system will float free in the open waters of the internet! |
| 17:56.26 | [TK]D-Fender | mirco: Either cut there or by the ISND driver that your system is using. |
| 17:56.37 | A-data | no [TK]D-Fender what i mean is that i need my support staff to join the Network So that they can recive calls anywhere in the wolrd |
| 17:56.42 | [TK]D-Fender | CrashSys: Just sit back and it will mix itself. |
| 17:56.47 | CrashSys | I'm on it |
| 17:56.48 | mirco | thnx for the info ... |
| 17:56.55 | [TK]D-Fender | A-data: Again, entirely too vague. |
| 17:57.03 | CrashSys | but you are on your own to compile |
| 17:57.06 | A-data | what u mean [TK]D-Fender |
| 17:57.39 | [TK]D-Fender | A-data: "calls" is not a self-contained term that tells me what ports "it" requires. |
| 17:59.01 | file | it's up to you... |
| 17:59.05 | file | I don't know what's going on |
| 17:59.14 | jer | anybody know how i can go about debugging why *67 is not functioning? i don't get any messages in the console about it, but i do get a busy/congested signal |
| 17:59.19 | Nugget | gah! I said "it" again! |
| 17:59.31 | CrashSys | Compile and enjoy |
| 17:59.35 | [TK]D-Fender | jer : GUILTY!!!! |
| 17:59.42 | jer | (and just to clarify, when i say *67, i of course mean the temporary hiding of callerid info) |
| 17:59.42 | zeeesh | <[TK]D-Fender>: i changed my error now its a kind of "*CLI> Can't locate LWP.pm in @INC (@INC contains: /usr/lib/perl5/site_perl/5.8.8/i386-linux-thread-multi"? |
| 17:59.45 | jer | [TK]D-Fender, =] |
| 17:59.52 | A-data | [TK]D-Fender i just need to call them from my laptop and they are distrubuted in all over the world .... i setuped a public IP address but when i try to connect from outside network (Inetrnet) xlite give me 408 request time out also from the internal lan it work |
| 18:00.05 | [TK]D-Fender | zeeesh: I don't do Perl or AGI. |
| 18:00.39 | A-data | zeedo this man LWP.pm not in ur include path... or even in the same folder as ur script |
| 18:00.46 | Nugget | "setuped"? holy cow. |
| 18:00.46 | [TK]D-Fender | A-data: calls need a TYPE & TECHNOLOGY. "calls" don't just happen. |
| 18:01.07 | A-data | xlite - * - xlite |
| 18:01.21 | [TK]D-Fender | A-data: then read up <- |
| 18:01.26 | [TK]D-Fender | ~sipnat |
| 18:01.28 | jbot | sipnat is probably for for more information about configurtion of Asterisk with SIP behind NAT, see http://voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 18:01.29 | [TK]D-Fender | ---^ |
| 18:02.06 | A-data | ty [TK]D-Fender i know i am asking folish questions but i am new in field.. trying to understand hard |
| 18:02.08 | A-data | :$ |
| 18:02.09 | rene- | zeesh: you are missing a perl library |
| 18:02.12 | rene- | use the CPAN to get ir |
| 18:02.16 | rene- | it |
| 18:02.19 | zeeesh | <A-data>: right now i check my script there is kind of this statement " use LWP 5.64; my $browser = LWP::UserAgent->new; "? |
| 18:02.30 | [TK]D-Fender | A-data: Not foolish question... uselessly VAGUE ones :) |
| 18:02.31 | rene- | i cant remember thr right sintax since i now do ruby |
| 18:03.15 | [TK]D-Fender | rene-: and now yer speling is atroshus! |
| 18:03.39 | [TK]D-Fender | </sarcasm> |
| 18:03.47 | rene- | hehe |
| 18:04.06 | rene- | sorry |
| 18:05.46 | rvosper | Can anyone point me in the right direction for tracking a call from start to finish with the cdr table? The unique id seems unique for each action not each call nessecarily |
| 18:09.05 | Nugget | rvosper: the main problem is that the cdr table doesn't get an INSERT until the call concludes. |
| 18:09.14 | rvosper | thnx |
| 18:09.37 | Nugget | http://macnugget.org/projects/asterisk/page13 |
| 18:09.42 | Nugget | that's how I solved it, ymmv. |
| 18:09.43 | exoxe | [TK]D-Fender: thank you.. would help if internal calls would work when callerid is set though :\ |
| 18:09.45 | Nugget | it's not very pretty |
| 18:09.54 | exoxe | not sure why this issue happens |
| 18:12.47 | rene- | hey |
| 18:12.54 | rene- | are there any other mitel users |
| 18:12.56 | rene- | ? |
| 18:13.15 | rene- | i wonder if anyone has gotten MWI working with firmware version 6 |
| 18:13.32 | rene- | it used to work with the older less reliable versions |
| 18:14.40 | exoxe | hah, figured it out |
| 18:17.28 | [TK]D-Fender | exoxe: Why WOULDN'T you set a callerid ofr your phone devices? |
| 18:17.42 | Trevor_b | anyone good with the goto command changing contexts? I think i am missing something when checking the application and attempting to do an IVR. |
| 18:18.01 | [TK]D-Fender | Trevor_b: "show application goto" |
| 18:18.06 | Trevor_b | yeah, i did ;) |
| 18:18.13 | Trevor_b | i think im not understanding it or something. |
| 18:18.28 | Trevor_b | trying to Goto(lts-mainmenu,,1) |
| 18:18.29 | [TK]D-Fender | Trevor_b: It takes 3 parms MAX ( 2 of which are OPTIONAL. This is not Raw Cat science. |
| 18:18.31 | Trevor_b | doesnt seem to do it. |
| 18:18.38 | [TK]D-Fender | Trevor_b: you NEED an exten in there. |
| 18:19.24 | Trevor_b | well the exten is set to s right now, i was trying Goto(lts-mainmenu,s) was I missing the last comma? |
| 18:19.45 | [TK]D-Fender | Trevor_b: SAD... now you're missing a PRIORITY |
| 18:19.51 | jer | urgh this is driving me nuts; this should be working or at least give me some way of diagnosing it |
| 18:19.56 | Trevor_b | oh DUH the context is optional |
| 18:20.17 | [TK]D-Fender | Trevor_b: So is the exten if all you're doing is skipping PRIORITY |
| 18:20.19 | Trevor_b | sorry TK, frazzled today by clients asking for things to be done before they call... @#$*& |
| 18:21.29 | Trevor_b | Well taking a call in context default, trying to route to context lts-mainmenu, and then let the background play greeting waiting for exten, just seems not to play and drops call when transfering. |
| 18:22.22 | [TK]D-Fender | Trevor_b: So show me your final attempted format |
| 18:22.28 | Trevor_b | k |
| 18:24.24 | Trevor_b | [TK]D-Fender: http://pastebin.ca/564735 |
| 18:26.02 | [TK]D-Fender | Trevor_b: looks good |
| 18:26.12 | Trevor_b | [TK]D-Fender: the Playback of lts_sales_greeting was JUST tossed into that so i could see if it was actually answering... |
| 18:26.21 | Trevor_b | the Background of it doesnt seem to ever play. |
| 18:26.41 | Trevor_b | ok So my Goto SHOULD actually get answered by lts-mainmenu and step into the background? |
| 18:27.02 | [TK]D-Fender | Trevor_b: perhaps you can so me CLI output of an ACTUAL CALL |
| 18:27.22 | Trevor_b | i can, but its SPAMMY, running vicidial on this system as well. |
| 18:27.42 | [TK]D-Fender | Trevor_b: I'm not PSYCHIC, so you're going to have to show somthing... |
| 18:27.56 | Trevor_b | no no, im getting it, just trying to limit the amount of console crap right now ;) |
| 18:28.34 | *** join/#asterisk daveburr (n=Miranda@66.7.124.15) |
| 18:28.41 | *** join/#asterisk asterisknerds2 (n=asterisk@64.71.152.211) |
| 18:28.41 | asterisknerds2 | <PROTECTED> |
| 18:29.36 | [TK]D-Fender | asterisknerds : .. you don't say! |
| 18:31.34 | Trevor_b | [TK]D-Fender: http://pastebin.ca/564754 |
| 18:33.13 | Trevor_b | HMM |
| 18:33.25 | Trevor_b | found and exit code, wonder if i carried something over from 1.2 to 1.4 |
| 18:34.06 | *** part/#asterisk daveburr (n=Miranda@66.7.124.15) |
| 18:34.55 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 18:34.57 | Trevor_b | changing and retesting right now |
| 18:35.37 | [TK]D-Fender | Trevor_b: Dies on #3.. -- Executing [s@lts-mainmenu:2] SetMusicOnHold("SIP/dmafinancial-0b9605e8", "default") in new stack |
| 18:35.50 | Trevor_b | Yeah DigitTimeout not registered application |
| 18:35.53 | Trevor_b | removing it |
| 18:36.02 | [TK]D-Fender | Trevor_b: And yeah... #3 & 4 are LONG gone |
| 18:36.14 | Trevor_b | Background? |
| 18:36.15 | [TK]D-Fender | Trevor_b: "show function TIMEOUT" |
| 18:36.16 | mocker | If dundi has advertised extensions for 1XXX on server A and 1001 on server B, will it always choose the more specific (like in the dialplan)? |
| 18:36.24 | Trevor_b | OH DUH |
| 18:36.28 | Trevor_b | reading from my new numbers ;) |
| 18:40.58 | exoxe | [TK]D-Fender: sorry for the delay... I wasn't planning on leaving callerid omitted, but I was having problems when it was set - turned out to be that it dislikes quotes around the value (e.g. callerid="Me <123>") |
| 18:41.19 | exoxe | it was my mistake putting quotes around it |
| 18:41.22 | [TK]D-Fender | exoxe: its sort of WRONG that way :) |
| 18:41.29 | exoxe | yeah yeah |
| 18:41.38 | exoxe | didn't catch it |
| 18:42.07 | [TK]D-Fender | exoxe: If you don't want to use callerid, set a channel var fromt he device : "setvar=vmbox=123" |
| 18:42.29 | exoxe | nah I do :) |
| 18:42.34 | exoxe | but thanks |
| 18:44.45 | *** join/#asterisk marthag (n=martha@NaTr-net1.ser.netvision.net.il) |
| 18:47.13 | *** join/#asterisk Nuitari (n=nuitari@melchior.nuitari.net) |
| 18:48.06 | Nuitari | I'm getting a bunch of compile errors trying to get func_devstate.c on asterisk 1.4.4 |
| 18:48.22 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 18:48.28 | Qwell[] | Nuitari: sure, it's made for trunk |
| 18:48.43 | blitzrage | Nuitari: probably because it wasn't for 1.4.4.... (ya... what Qwell[] said) |
| 18:49.29 | marthag | hi |
| 18:49.45 | marthag | has anyone here gotten app_ldap to work with 1.4? |
| 18:49.50 | Nuitari | ok, good thing this is for a test server |
| 18:51.25 | marthag | it crashes asterisk on the second ldap query for me :-( |
| 18:52.54 | *** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust826.leed.cable.ntl.com) |
| 18:52.56 | Nuitari | When would trunk become a release? |
| 18:53.22 | [TK]D-Fender | Nuitari: 1.6 <- |
| 18:53.35 | Qwell[] | trunk itself would never become a release |
| 18:53.37 | [TK]D-Fender | Nuitari: Devstate is NOT making itno 1.4 |
| 18:53.46 | Qwell[] | it'll be branches as 1.6, and releases will be made from that |
| 18:55.44 | Nuitari | Thanks |
| 18:55.52 | *** join/#asterisk jkiff (n=jkiffmey@unaffiliated/vorondil) |
| 18:56.54 | *** join/#asterisk boch (n=fran@190.48.219.252) |
| 18:56.55 | *** join/#asterisk tsurko (n=tsurko@150-190.go.evo.bg) |
| 19:03.04 | Trevor_b | [TK]D-Fender: thanks for the help, call center is so damn spammy until i pastebin'd it i never saw the exit status zero on that priority. |
| 19:03.30 | *** join/#asterisk flashnet (n=flashnet@207-237-23-19.c3-0.avec-ubr11.nyr-avec.ny.cable.rcn.com) |
| 19:03.57 | [TK]D-Fender | Trevor_b: text search in a web browser is a wonderful thing... lets you trace everything :) |
| 19:04.42 | Trevor_b | yeah, MUCH nicer then trying to watch console fly by (thank god i grepped out the DEBUG!) were about to change some source for call centers to stop debug from manager 5-50 times per second. |
| 19:05.45 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 19:07.41 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 19:08.02 | Nuitari | parallel make doesn't seem to work well with trunk |
| 19:08.12 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 19:08.25 | EricL | Can someone at least help point me in the right direction with my callerid issues. Googling isn't getting me anywhere and I don't know where else to start. |
| 19:08.35 | [TK]D-Fender | EricL: Pastebin stuff.... |
| 19:08.38 | EricL | Does anyone have any idea why * would only pass the CALLERID(name) and not the CALLERID(num) forward when it has both? |
| 19:08.49 | EricL | [TK]D-Fender:What should I pb? |
| 19:09.22 | [TK]D-Fender | EricL: Proof of what it hold, how you're calling the destination (tech entry), and all hardware involved. |
| 19:14.38 | EricL | http://pastebin.ca/564849 |
| 19:14.48 | EricL | I don't know what else is pertinent. It happens on incoming calls. |
| 19:15.07 | EricL | It makes it into the CDR correctly, but doesn't get forwarded to the phones (the name does, but not the number). |
| 19:15.34 | EricL | The number makes it to comedian mail, but the email comedian mail sends to the user telling them they have vm contains no phone number. |
| 19:15.56 | [TK]D-Fender | EricL: NoOp callerid in there, and FYI your hint is broken. you can't pattern-match it. |
| 19:16.58 | EricL | I don't know what any of that means. |
| 19:19.06 | [TK]D-Fender | EricL: Use NoOp to display in CLI the contents of the CID, and I want to SEE the full CLI output of a call attempt |
| 19:19.18 | [TK]D-Fender | EricL: and your hint priority is BROKEN. |
| 19:20.06 | VJFROMGT | I am new to asterisk ,, old timer in freepbx,, where do i define sip trunk in sip.conf? |
| 19:20.54 | *** join/#asterisk cr4z3d (n=cr4z3d@ip70-162-117-79.ph.ph.cox.net) |
| 19:25.14 | *** join/#asterisk dlynes_laptop (n=dlynes@d207-216-161-56.bchsia.telus.net) |
| 19:25.45 | blitzrage | VJFROMGT: at the bottom |
| 19:26.12 | *** join/#asterisk qdk (n=qdk@145.Red-81-44-152.dynamicIP.rima-tde.net) |
| 19:26.38 | EricL | exten => _5[1-2]XX,2,NoOp(${CALLERID}) <-- This didn't print anything on the console (accessed via 'asterisk -r') |
| 19:27.17 | [TK]D-Fender | EricL: because that variable no longer exists in 1.4 |
| 19:27.25 | *** join/#asterisk matsk (i=matsk@h110n2fls32o882.telia.com) |
| 19:27.34 | [TK]D-Fender | VJFROMGT: ... |
| 19:27.36 | [TK]D-Fender | ~book |
| 19:27.37 | jbot | well, book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 19:27.38 | [TK]D-Fender | ~wikis |
| 19:27.39 | jbot | somebody said wikis was http://www.voip-info.org |
| 19:27.55 | [TK]D-Fender | VJFROMGT: http://www.jeremy-mcnamara.com/index.php/2007/02/26/how-to-configure-asterisk-your-first-installation/ |
| 19:30.48 | *** join/#asterisk ToyMan (n=Stuart@74-32-41-219.dsl1.mdl.ny.frontiernet.net) |
| 19:33.25 | EricL | [TK]D-Fender: I changed it to $CALLERIDNUM and still nothing is getting printer. |
| 19:34.33 | *** join/#asterisk _charly_ (i=kroseneg@sunrise.schmidham.net) |
| 19:34.36 | purplet | hello everyone. I'am still having some problems connection problems with an IAX connection between two asterisk server over an internet connection. In one way, after 11 minutes, voice drops. However, the connection itself doesn't drop. I'am suspecting NAT, but isn't voice and signaling transfered in the same IAX packets? In IAX debug output I still see messagings flowing, so isn't that weird then? After putting the call on hold, and back, the v |
| 19:35.05 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 19:37.05 | Nuitari | is this something serious: chan_iax2.c:6537 socket_read: Out of idle IAX2 threads for I/O, pausing ? |
| 19:38.47 | *** join/#asterisk guillote_GNU (n=guillote@host176.201-252-205.telecom.net.ar) |
| 19:39.12 | *** join/#asterisk Siya (n=djerk@217-195-248-251.dsl.easynet.nl) |
| 19:39.39 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 19:39.43 | [TK]D-Fender | EricL: and THAT variable ALSO no longer exists in 1.4 |
| 19:39.58 | [TK]D-Fender | EricL: "show function CALLERID". thi was introduced in 1.2 |
| 19:41.21 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 19:43.04 | *** part/#asterisk ramindia (n=ramindia@202.63.96.9) |
| 19:43.27 | Siya | Anyone here who can tell me whether it's impossible to have two lines on an 7960 behing nat? |
| 19:43.41 | Siya | I can make calls using one line but not the other |
| 19:43.51 | Siya | (like 2 extensions on the same phone) |
| 19:44.17 | Siya | [Jun 13 21:43:59] WARNING[5047]: chan_sip.c:8112 check_auth: username mismatch, have <103>, digest has <102> |
| 19:44.20 | Siya | [Jun 13 21:43:59] NOTICE[5047]: chan_sip.c:13362 handle_request_invite: Failed to authenticate user "102" |
| 19:45.16 | Siya | The issue is prolly that they both peers have the same sip source port: |
| 19:45.18 | Siya | 103/103 217.195.229.9 D N 5062 OK (147 ms) |
| 19:45.20 | Siya | 103/103 217.195.229.9 D N 5062 OK (147 ms) 102/102 217.195.229.9 D N 5062 OK (192 ms) |
| 19:45.33 | Siya | anyone? |
| 19:45.54 | EricL | [TK]D-Fender: I found that variable in the channelvariables.txt in the 1.4.4 documentation. |
| 19:46.24 | rene- | hey |
| 19:47.03 | rene- | when a user has twenty something voice mails, the vm system is unable to say the number of messages |
| 19:47.14 | rene- | as in the file 23 cant be foun |
| 19:47.30 | rene- | shouldnt it be building the number using the file twenty and the file three |
| 19:47.45 | rene- | should this happen automatically or something needs to be configured? |
| 19:48.07 | *** join/#asterisk sualk (n=klaus@flittner.org) |
| 19:51.34 | EricL | [TK]D-Fender: I still can't get it to print on the console, but when I do a "sip show channels" it shows me the correct phone number in the user/ANR column. |
| 19:51.54 | *** join/#asterisk Mahmoud (n=fake@unaffiliated/mahmoud) |
| 19:51.56 | [TK]D-Fender | EricL: please pastebin the call with the proper use of the callerid function... |
| 19:54.10 | EricL | http://pastebin.ca/564919 <-- The two ways I tried it are at the bottom of the paste. |
| 19:54.15 | mutilator | http://www.japanprobe.com/?p=1980 |
| 19:55.03 | *** join/#asterisk MrTelephone (n=MrTeleph@h697179-171.picriverisp.net) |
| 19:55.05 | [TK]D-Fender | EricL: and my CLI output please? |
| 19:55.16 | [TK]D-Fender | exten => _5[1-2]XX,2,NoOp(${CALLERID(num)}) <- right |
| 19:55.36 | MrTelephone | calls outgoing on the pri sound muffled to the called party but everything sounds fine for the calling party.. Any ideas? |
| 19:55.37 | [TK]D-Fender | exten => _5[1-2]XX,2,NoOp(${CALLERID(num)}---${CALLERID(name)}) <- better |
| 19:55.40 | EricL | [TK]D-Fender: There is nothing getting output'd to the cli. |
| 19:55.52 | MrTelephone | There is distortion (not clipping) when the caller speaks. |
| 19:56.00 | [TK]D-Fender | EricL: add it, repastebin and show me. |
| 19:56.39 | EricL | a "dialplan reload" should put it into effect, no? |
| 19:57.39 | Siya | no-one who can tell me about a cisco 7960 phone and nat? |
| 19:57.56 | [TK]D-Fender | EricL: I'd do "reload" |
| 19:58.03 | *** join/#asterisk n00dle (n=ccraft@hillel.springsips.com) |
| 19:59.19 | n00dle | Hey, Russell. |
| 20:00.33 | EricL | I did a reload and still nothing came up. (http://pastebin.ca/564937) |
| 20:01.13 | *** join/#asterisk voipnet-tech (n=voipnet-@rrcs-24-97-250-50.nys.biz.rr.com) |
| 20:01.16 | voipnet-tech | morning |
| 20:02.04 | [TK]D-Fender | EricL: CLI OUTPUT PLEASE <----------- |
| 20:02.28 | EricL | [TK]D-Fender: What output? I am not getting any output when I make a call? |
| 20:02.52 | *** join/#asterisk keulin (n=cray@ifth-pdcr2.infotheme.fr) |
| 20:03.01 | CrashSys | Hmmm |
| 20:03.09 | [TK]D-Fender | EricL: I want to see the call in its entirety. If you have NOTHING to show then you've screwed up far beyond this. |
| 20:03.34 | CrashSys | D-Fender: I like the gentle soothing way you word things... :) |
| 20:03.44 | CrashSys | I wish I had those kind of patience... |
| 20:03.47 | voipnet-tech | any linphone users? |
| 20:04.05 | EricL | [TK]D-Fender: Then I probably screwed up far beyond this. Do you want the output of SIP debugging? |
| 20:04.07 | MrTelephone | can static around talking be caused from txgain issues on a pri? |
| 20:04.35 | [TK]D-Fender | EricL: might help..... if dialplan code is not even EXECUTING, then like WTF.... |
| 20:05.19 | EricL | [TK]D-Fender: Well, the hint seems to be working and everything else seems to be functioning properly. So if I screwed something up, it can't be that bad. |
| 20:06.05 | [TK]D-Fender | EricL: could just be dealing with an apple & oranges scenario |
| 20:06.30 | *** join/#asterisk waKKu (n=eurulo@unaffiliated/wakku) |
| 20:06.39 | CrashSys | Anyone know a good way to identify if a pair is Analog/DSL or T1? (it's all unmarked, no tags) |
| 20:06.50 | CrashSys | Voltages on the pairs different? |
| 20:07.06 | [TK]D-Fender | CrashSys: well T1 is *2* pair. DSL... well just plug in a DSL modem and see if it syncs |
| 20:07.16 | Qwell[] | if it's analog, you'll get dialtone |
| 20:07.24 | Qwell[] | well, in theory |
| 20:07.29 | CrashSys | D-Fender: The building has 200-pair... |
| 20:07.31 | MrTelephone | how do you reserve an irq in the bios? |
| 20:07.38 | Qwell[] | MrTelephone: depends on the bios |
| 20:07.50 | [TK]D-Fender | Qwell[]: beat me to it. |
| 20:07.53 | EricL | http://pastebin.ca/564950 <-- I added the SIP debugging for the call |
| 20:07.55 | CrashSys | So any DSL line should give me staticy-dial-tone? |
| 20:08.35 | CrashSys | I didn't know if they still sent dial-tone on a line that only has DSL |
| 20:08.39 | Qwell[] | no |
| 20:08.42 | Nuitari | CrashSys: unless it's a naked copper loop just for DSL |
| 20:08.53 | Qwell[] | dsl is quite high freq, you shouldn't hear it |
| 20:09.08 | Qwell[] | the filters are to protect the high signal, not the normal one |
| 20:09.16 | EricL | [TK]D-Fender: What shows up on my phone is "Unavailable" (which is correct) and then "5201" instead of the phone number (which * obviously gets since its in the CDR and the SIP debugging output). |
| 20:09.24 | CrashSys | Qwell: But typically, even a dedicated DSL circuit will have dial-tone? |
| 20:09.30 | Qwell[] | no |
| 20:09.42 | CrashSys | Hmmm... |
| 20:10.17 | MrTelephone | qwell in my bios all there you can select what irq is reserved but you can't tell it what slot will use what irq.. is that normal? |
| 20:10.24 | Strom_M | I don't even know if you get talk battery on a DSL circuit |
| 20:10.25 | Qwell[] | pretty much |
| 20:10.35 | Qwell[] | Strom_M: a dry pair? I would doubt it |
| 20:10.39 | [TK]D-Fender | EricL: I'm not sure if this is apparent to you, but I care absolutely NOTHING for descriptions of problems. I want to SEE THEM. With my own eyes. Real CLI output. If there's nothing to see I can only trust that you are looking at something that is completely not relevent to this problem. |
| 20:10.39 | Siya | CrashSys: only ADSL with line sharing should give dial tone (most common type) |
| 20:10.40 | Strom_M | yeah |
| 20:11.26 | Nuitari | MrTelephone: it depends on your motherboard. |
| 20:11.34 | CrashSys | Hmmm... One idea is to rewire the building, rip all the old-crap back off, then call verizon and go "I have no dial-tone" |
| 20:11.44 | CrashSys | since they wont help me with anything else :) |
| 20:11.55 | Siya | CrashSys: SDSL and ADSL without line sharing you won't be able to detect if you don;t have equipment to do so... |
| 20:12.13 | EricL | [TK]D-Fender: That's the thing. I don't know what is relevant to the problem which is why I am asking. I am doing exactly what you are telling/showing me to do and getting no results. |
| 20:12.40 | EricL | [TK]D-Fender: I am not trying to be a PITA (although it does seem to be turning out that way). I just don't know why what you are telling me to do isn't working. |
| 20:12.47 | [TK]D-Fender | EricL: well if you aren't getting any CLI output at verbose 10 welll..... youa ren't even receiving a CALL. |
| 20:12.51 | Siya | if you need to find a copper pair have a tone put onto the line and use a 'tone seeker' (little box with a prastic pin which you run across all coper pairs. you'll hear the tone if the line is there |
| 20:13.09 | EricL | [TK]D-Fender: How do I increase the verbosity without restarting the * server? |
| 20:13.19 | [TK]D-Fender | set verbose 10 |
| 20:13.24 | EricL | Ah wait, I'm an f'n idiot. |
| 20:14.12 | CrashSys | Siya: Yeah, i'm familiar with a fox and hound... Here is my scenario... I have a total of 200-pairs dropped off in 4 different boxes on a commercial building. |
| 20:14.22 | EricL | [TK]D-Fender: grosvenor*CLI> |
| 20:14.23 | EricL | <PROTECTED> |
| 20:14.23 | EricL | <PROTECTED> |
| 20:14.23 | EricL | <PROTECTED> |
| 20:14.23 | EricL | [Jun 13 16:13:45] DEBUG[27679]: cdr_addon_mysql.c:210 mysql_log: cdr_mysql: inserting a CDR record. |
| 20:14.23 | EricL | [Jun 13 16:13:45] DEBUG[27679]: cdr_addon_mysql.c:226 mysql_log: cdr_mysql: SQL command as follows: INSERT INTO cdr (calldate,clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,duration,billsec,disposition,amaflags,accountcode) VALUES ('2007-06-13 16:13:40','','','s','internal', 'SIP/5201-08238878','','','',5,0,'NO ANSWER',3,'') |
| 20:14.27 | EricL | <PROTECTED> |
| 20:14.29 | EricL | [Jun 13 16:13:45] DEBUG[27679]: cdr_addon_mysql.c:210 mysql_log: cdr_mysql: inserting a CDR record. |
| 20:14.31 | EricL | [Jun 13 16:13:45] DEBUG[27679]: cdr_addon_mysql.c:226 mysql_log: cdr_mysql: SQL command as follows: INSERT INTO cdr (calldate,clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,duration,billsec,disposition,amaflags,accountcode) VALUES ('2007-06-13 16:13:40','\"Unavailable\" <2012147154>','2012147154','5201','default', 'SIP/5060-b6905ba0','SIP/5201-08238878','Dial','SIP/5201|18',5,0,'NO ANSWER',3,'') |
| 20:14.32 | CrashSys | there has been about 20-years of tenants running their own wires... My job is to make a wiring-room... |
| 20:14.36 | EricL | Whoops, the URL didn't take. |
| 20:14.38 | EricL | http://pastebin.ca/564963 |
| 20:14.40 | Strom_M | idiot |
| 20:14.45 | EricL | Yes I am. |
| 20:14.50 | [TK]D-Fender | EricL: PASTEBIN. and you KNOW I want to see the call in its entirety |
| 20:15.00 | Siya | CrashSys: I know the problem, this is what tone generators are used for, suffice to say cheap providers skimp on these first in their dslams |
| 20:15.02 | EricL | [TK]D-Fender: Alright. |
| 20:15.06 | *** join/#asterisk NirS (i=Nir@87.68.76.20.cable.012.net.il) |
| 20:15.11 | NirS | hey all |
| 20:15.18 | NirS | how is everybody doing today ? |
| 20:15.28 | Strom_M | boneriffic |
| 20:15.46 | [TK]D-Fender | EricL: does 5201 in ****DEFAULT*** look like 5201 in ***[DID]*** to you? |
| 20:15.58 | Siya | CrashSys: good luck to you... |
| 20:16.04 | CrashSys | Yeah |
| 20:17.48 | MrTelephone | that sucks because I don't think too many boards support interrupt irq mapping |
| 20:17.48 | NirS | hey all, any idea why Set(TIMEOUT(absolute)=XX) won't work ? |
| 20:17.48 | EricL | [TK]D-Fender: Nope, is that what's affecting the callerid, that I am changing the context? |
| 20:17.48 | Nuitari | I'm having a weird issue with trunk and IAX qualify, I've put the cli out put at http://www.pastebin.ca/564965 |
| 20:17.48 | CrashSys | Server boards support IRQ mapping? |
| 20:17.48 | [TK]D-Fender | EricL: You aren't even executing the code you've been showing me. pastebin your entire bloody dialplan please. |
| 20:18.01 | Strom_M | Sadomasochistic Tech Support, brought to you by [TK]D-Fender |
| 20:18.23 | [TK]D-Fender | Strom_M: No, they bring it... I'm just the announcer |
| 20:18.39 | [TK]D-Fender | CrashSys: Sometimes |
| 20:18.48 | CrashSys | Watching Fender give tech support is like waiting for a train to derail... :) |
| 20:19.03 | CrashSys | You can see it coming... and all the time you're just thinking "Hmm, this will be cool" |
| 20:20.14 | Nuitari | For the IRQ Routing, some boards have basically IRQ Channels that you can a assign an IRQ to, then you have to check your board documentation to find which slot and or integrated device is on each. |
| 20:20.41 | Nuitari | I've only come across 1 Compaq board that you can say "put this irq on that slot" |
| 20:21.03 | Nuitari | MrTelephone: why do you want to reserve an irq anyways? |
| 20:21.19 | EricL | [TK]D-Fender: http://pastebin.ca/564975 |
| 20:22.14 | MrTelephone | im having a lot of interrupt issues with my raid card and dual pri card on the same bus |
| 20:22.30 | *** join/#asterisk saftsack (n=saftsack@pD9E06530.dip.t-dialin.net) |
| 20:22.43 | MrTelephone | such as static on one side of the calls.. some clicking sounds |
| 20:23.07 | MrTelephone | I can't move the cards because the riser card that I have is sh*t and waiting for the proper one to come in |
| 20:23.52 | NirS | never mind, found the problem |
| 20:24.41 | MrTelephone | nuitari, i remember seeing that function on a board somewhere in the past too |
| 20:26.39 | *** join/#asterisk THX2000 (n=bob@netblock-208-127-94-59.dslextreme.com) |
| 20:26.41 | pfn | how convenient, nufone is down... |
| 20:27.18 | THX2000 | Has anyone had a good experience w/ atacomm? |
| 20:27.29 | purplet | hello everyone. I'am having some connection problems with an IAX connection between two asterisk server over internet. Calls setup in one direction are getting voice drops, after exactly 11 minutes. However, the IAX connection itself doesn't drop. I'am suspecting NAT, but isn't voice and signaling transfered in the same IAX packets? In IAX debug output I still see messages flowing, so isn't that weird then? After putting the call on hold, and t |
| 20:28.05 | *** part/#asterisk sualk (n=klaus@flittner.org) |
| 20:28.11 | CrashSys | I've used atacomm a lot and have good experiences with them |
| 20:28.14 | [TK]D-Fender | EricL: Last thing : pastebin "show dialplan" |
| 20:28.33 | MrTelephone | a pri card with onboard processor is like $7k? |
| 20:28.34 | MrTelephone | heh |
| 20:28.58 | CrashSys | Plus there's a real phone number you can call and real people to talk to... none of that e-mail BS! |
| 20:29.08 | NirS | is it possible to supply MOH when a meetme user is in Talk only mode ? |
| 20:29.14 | THX2000 | CrashSys: Having all kinds of issues getting them to fill an order. Contacted them a dozen times and still haven't received my full order. Then they had the nerve to blame it on me. |
| 20:29.37 | MrTelephone | what is the sangoma a102d card going for? |
| 20:29.46 | CrashSys | I guess i've been lucky then |
| 20:29.50 | CrashSys | How big was the order? |
| 20:29.51 | EricL | [TK]D-Fender: http://pastebin.ca/564996 |
| 20:30.20 | THX2000 | Pretty small...just trying to pick up some parts for R&D ... Aastra 53i, 57i CT and 560M sidecar |
| 20:30.43 | CrashSys | I've only ever ordered polycom and sangoma stuff from them... |
| 20:30.46 | *** join/#asterisk sergee (i=kvirc@195.94.224.197) |
| 20:30.51 | CrashSys | well I did order some grandstreams before.. once before :) |
| 20:30.57 | *** join/#asterisk DaveCanoe (n=Dave@H6.C30.B96.tor.eicat.ca) |
| 20:31.18 | CrashSys | I seem to remember someone saying aastra was having supply issues... |
| 20:32.15 | [TK]D-Fender | EricL: ok, I'm stumped... not even voicemail is being called... |
| 20:32.20 | THX2000 | Yea, they've but the blame on Aastra as well |
| 20:32.31 | *** join/#asterisk CBU[^_^]M`` (n=love@210.213.139.67) |
| 20:32.33 | THX2000 | Which is really none of my concern...don't sell the part if you have no way to aquire it |
| 20:32.37 | CrashSys | Something about the Aastra phones not meeting RoHS and being held up |
| 20:32.42 | sergee | can anybody explain me what is the difference between "Publicly Available Telephone Service (PATS) suppliers" and "Electronic Communication Service (ECS) suppliers"? |
| 20:33.04 | EricL | [TK]D-Fender: What do you mean not even voicemail is being called? |
| 20:33.05 | [TK]D-Fender | THX2000: good... then you still have time to change to Polycom instead ;) |
| 20:33.13 | *** join/#asterisk UCFmethod (n=UCFmetho@office.eyestreet.com) |
| 20:33.22 | sergee | who are "ECS suppliers"? |
| 20:33.27 | [TK]D-Fender | EricL: You don't answer the call and it hangs up right away. its supposed to fail over to voicemail. |
| 20:33.33 | THX2000 | haha, i picked up a couple polycoms...wasn't really into em |
| 20:33.36 | MrTelephone | the wholesale cost on a sangoma 102d is like 1450.00 cdn |
| 20:33.56 | [TK]D-Fender | MrTelephone: ENTIRELY worth it, |
| 20:34.09 | MrTelephone | im having problems with mine but i think its the server and not the card |
| 20:34.27 | MrTelephone | 8 carrier drops in the last 2 weeks?? |
| 20:34.28 | EricL | Everything else works just fine, including voicemail failovers. The only thing that isn't working is the CALLERID stuff though. |
| 20:35.06 | [TK]D-Fender | EricL: Then why don't I see voicemail being called in the pastebin? |
| 20:35.14 | MrTelephone | fender i was trying to do some traceing but the utilities never compile.. did you have that problem? |
| 20:35.18 | CrashSys | I've used sangoma A200d's and A102d's and they have been flawless... |
| 20:35.27 | CrashSys | well they've been as good as the server they went into... |
| 20:35.36 | MrTelephone | what kind of computers are you putting them in, and whats your uptime like? |
| 20:35.42 | [TK]D-Fender | MrTelephone: Nope. Sangoma = 100% success every time for me. CentOS, FC, Slackware |
| 20:36.02 | EricL | [TK]D-Fender: I don't know. I pb'd my extensions.conf and did a 'show dialplan' |
| 20:36.15 | EricL | [TK]D-Fender: I also have a voicemail.conf. |
| 20:36.36 | MrTelephone | Im using the beta drivers now because of the RBS bug. but overall i think its working well. I just have some voice quality issues (one-way) that I have to figure out |
| 20:36.38 | CrashSys | My uptime is usually limited to florida power :) |
| 20:37.01 | MrTelephone | I notice when I ssh to the server it seems laggy but the console is super fast. IRQ problems? Network Interface Problems? |
| 20:37.08 | CrashSys | But 3-4 months before florida power struck... |
| 20:37.09 | MrTelephone | It's really hard to find the cause of these issues |
| 20:37.13 | [TK]D-Fender | EricL: http://pastebin.ca/564963 <- this is NOT valling voicemail. I have a very strong suspicion you have shown me something that is NOT applicable. |
| 20:37.21 | MrTelephone | one guy posted that he reboots once a week? |
| 20:37.32 | CrashSys | Also depends on how many modules you are using... |
| 20:37.44 | [TK]D-Fender | ok, heading home... later all |
| 20:37.44 | MrTelephone | have you guys seen any carrier errors when you do 'ifconfig'? |
| 20:37.52 | EricL | [tk: That's possible. But like I said, I don't know what's applicable, I just showed you what you asked me for. |
| 20:38.14 | CrashSys | I used to let them run till they crash... but have since changed my motto and let the boxes recycle themselves at like 1pm on sunday's... |
| 20:38.26 | MrTelephone | all the problems I'm having are intermitten so I can't track them down |
| 20:38.49 | CrashSys | The A200's dont really get carrier error's... |
| 20:38.56 | MrTelephone | A102d pri card |
| 20:39.04 | CrashSys | but the 102 would get some carrier error's... specially if it was raining... |
| 20:39.26 | MrTelephone | people are hearing static around my voice as if you have your speakers turned up too loud |
| 20:39.42 | MrTelephone | its fibre to my office :( |
| 20:40.26 | MrTelephone | carrier error being there was no way to transport that 1 frame along the physical network |
| 20:40.43 | *** join/#asterisk rmayorga (i=rmyorg@unaffiliated/rmayorga) |
| 20:40.47 | MrTelephone | so if you get one carrier error that is one packet? or do you loose everything until the network resyncs |
| 20:40.47 | CrashSys | Dont get those that often... |
| 20:41.00 | CrashSys | maybe 1-2 times a month... random times... typically a verizon issue... |
| 20:41.03 | MrTelephone | maybe 1.5 carrier errors/day |
| 20:41.16 | MrTelephone | do you get dropped calls from that? |
| 20:41.25 | *** part/#asterisk marthag (n=martha@NaTr-net1.ser.netvision.net.il) |
| 20:41.31 | CrashSys | the whole span dies |
| 20:41.40 | MrTelephone | yikes |
| 20:41.46 | Nuitari | THX2000: I also had many good experiences with atacomm |
| 20:41.51 | CrashSys | Goes REd, then comes back up... |
| 20:41.59 | Nuitari | THX2000: the only issue I had was one fedex screwed up |
| 20:42.15 | MrTelephone | i wonder how telcos deal with that.. auto failover hardware? |
| 20:42.19 | MrTelephone | $$$$ |
| 20:42.21 | CrashSys | THX: And until Atacomm knows that there is a supply issue, they will keep selling it... it's not like Aastra is every going to call people and go "Dont sell our product, we have supply issues" |
| 20:42.52 | CrashSys | Telephone: Verizon deals with it by saying "That's as good as it gets, sorry" and hanging up |
| 20:42.53 | CrashSys | :D |
| 20:42.59 | Nuitari | THX2000: though I only got Polycoms and 1 grandstream from them |
| 20:43.10 | CrashSys | Same here |
| 20:43.27 | MrTelephone | i know but how often do you get a dropped call on verizons aanlog service, probably never |
| 20:43.45 | MrTelephone | if you guys want to buy hardware you should ask me |
| 20:43.48 | Nuitari | I wish Polycom supported gsm, or some other small, free, codec |
| 20:43.48 | MrTelephone | i'll sell you some |
| 20:43.55 | CrashSys | No... on the analog POTS lines you get crackle, noise, no dial tone, etc... |
| 20:43.57 | CrashSys | so it's worse :) |
| 20:44.00 | MrTelephone | g729 is cheap |
| 20:44.14 | Nuitari | isn't it a binary package? |
| 20:44.20 | MrTelephone | bell canada here in ontario is like 99.9% reliable |
| 20:44.29 | MrTelephone | so its really hard to match the service |
| 20:44.35 | CrashSys | Well Verizon Tampa Bay area isn't... |
| 20:44.45 | MrTelephone | what about bellsouth? |
| 20:44.52 | CrashSys | I've seen lines read 5,000-ft longer when it rains :) |
| 20:44.58 | CrashSys | We dont have bellsouth |
| 20:45.01 | CrashSys | we have verizon |
| 20:45.06 | MrTelephone | thats the only one? |
| 20:45.10 | MrTelephone | wow |
| 20:45.12 | CrashSys | Yes, that's the lec |
| 20:45.26 | CrashSys | and everyone else rides their copper... |
| 20:45.31 | MrTelephone | Its not suprising then that they have a "live with it" attitude |
| 20:46.04 | MrTelephone | funny how these companies have big fancy commercials but their infrastructure is falling apart |
| 20:46.09 | CrashSys | Well, they told me they would pull a new aerial... I laughed and hung-up.. |
| 20:46.22 | Nuitari | sigh, I'm sorry if I missed the answer, but I'm having a weird issue with Trunk and IAX qualify, I've put the cli output at http://www.pastebin.ca/564965 . The issue doesn't happen with 1.4.4. The affected servers both run 1.2.x |
| 20:46.36 | CrashSys | Verizon here isn't laying any new copper... and they're pretty much stopped maintaining the current stuff... |
| 20:47.12 | CrashSys | What they are doing here is pushing Fios... cause with fios they dont have to make provisions for anyone else |
| 20:47.15 | EricL | [tk: No ideas? |
| 20:47.41 | MrTelephone | i just did a wanrouter restart but with more and more customers its harder to do a restart when noone is on the phone |
| 20:47.48 | EricL | Guess he left. |
| 20:47.52 | CrashSys | So verizon is pulling fiber all over the place, and letting the copper rot :) |
| 20:48.03 | MrTelephone | I don't know what fios is |
| 20:48.07 | MrTelephone | oh |
| 20:48.28 | CrashSys | Fios is where they bring fiber to your building, and then break out into copper from there... |
| 20:48.44 | MrTelephone | im pulling a few fibres to split our cable network in half because copper sucks |
| 20:49.05 | MrTelephone | so all your services are on one pair of fibres? |
| 20:49.13 | CrashSys | Yup... |
| 20:49.15 | MrTelephone | nice |
| 20:49.37 | CrashSys | You would think so |
| 20:49.51 | MrTelephone | i can't see fibre being run through the homes.. too delicate |
| 20:49.57 | MrTelephone | copper can take a lot of shit kickin |
| 20:50.21 | CrashSys | They run fiber to the house, mount a break-out box on the wall, then send copper through the house... |
| 20:50.32 | MrTelephone | that would work quite well |
| 20:50.33 | CrashSys | fiber only makes it to their IAD... it's all copper after that... |
| 20:50.44 | Nuitari | that sucks... |
| 20:50.55 | CrashSys | Yeah... except it's verizon and their service usually sucks... |
| 20:51.00 | Nuitari | much more crap for the customer to maintain too I guess |
| 20:51.16 | CrashSys | Ohh yeah... and you have 4-hours of phone when the power goes out |
| 20:51.27 | saftsack | tzafrir_laptop: hi, do you have contact to peter from rowetel? |
| 20:51.45 | Nuitari | when power goes out here, it's at least for 12h |
| 20:52.15 | CrashSys | Yeah... so if after 4 hours you needed help, you're screwed |
| 20:52.16 | Nuitari | thankfully it very rarely goes out |
| 20:52.18 | tzafrir_laptop | saftsack, david? |
| 20:52.29 | saftsack | yes david ;) |
| 20:52.46 | saftsack | because i wrote a mail to one of the PR1 guys but i didnt get any answer |
| 20:52.48 | *** join/#asterisk crichardson (n=crichard@38.113.5.185) |
| 20:52.53 | crichardson | hey guys i was woundering if any one could take a look at this i just started using 1.4.4 today and it seems i can make 1 call then after that it gets a congestions and i have to do a restart inorder to place another call here is the out put from the console along with debug on my phone http://pastebin.ca/565028 thanks |
| 20:52.56 | CrashSys | Plus there is a battery-pack that has to be plugged into it... I dont know how they plan to sell this thing to apartment buildings... |
| 20:53.30 | Nuitari | http://en.wikipedia.org/wiki/Verizon_FiOS <-- big box for a small thing |
| 20:55.54 | MrTelephone | well you got 30 seconds to phone on voip system before the power goes out if you need to call someone |
| 20:56.06 | MrTelephone | on mine I meant to say |
| 20:56.15 | MrTelephone | no backup power :-/ |
| 20:56.31 | *** join/#asterisk Igbothom_III (n=Hilton@office.quarkit.com.au) |
| 20:56.50 | MrTelephone | how can ssh be laggy on a local network? |
| 20:56.53 | *** join/#asterisk boch (n=fran@190.48.205.198) |
| 20:57.05 | Mahmoud | does SIP TLS encrypt the SIP header as well? |
| 20:57.10 | Igbothom_III | MrTelephone, too much network traffic |
| 20:57.19 | Igbothom_III | or a bogged down server |
| 20:58.03 | Nuitari | Does anyone else has issues with DID World Wide? |
| 20:58.15 | CrashSys | That cable install looks ugly on that box :( |
| 20:58.19 | MrTelephone | load average: 0.01, 0.00, 0.00 |
| 20:58.44 | MrTelephone | what about gigabit interface cards on a 10/100 switch? |
| 20:59.10 | blitzrage | network card in half-duplex mode? |
| 20:59.30 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 21:00.00 | Nuitari | [TK]D-Fender: welcome back |
| 21:00.35 | Nuitari | Is there any way in the Trunk version to link the hints of 2 pbx together? |
| 21:00.36 | MrTelephone | maybe |
| 21:00.49 | Nuitari | without having to write a script and do everything through the manager interface |
| 21:00.59 | *** join/#asterisk EvilGreen_ (n=Miranda@ppp85-140-136-133.pppoe.mtu-net.ru) |
| 21:02.03 | CrashSys | SLA = Evil! |
| 21:02.45 | mutilator | http://news.bbc.co.uk/2/hi/uk_news/england/merseyside/4253849.stm A woman who ripped off her ex-boyfriend's testicle with her bare hands has been sent to prison. |
| 21:03.01 | crichardson | hmm it seems like my server is dropping the reg after it does one call anyone seen that before? |
| 21:04.01 | MrTelephone | how can you find out what kernel module your netcard is using :( |
| 21:04.20 | crichardson | lspci |
| 21:04.24 | crichardson | lsmod |
| 21:08.26 | *** join/#asterisk Toerkeium (i=oo@201.216.206.221) |
| 21:11.21 | MrTelephone | should a person disable ipv6 in the kernel? |
| 21:11.29 | MrTelephone | frig that crap is really annoying |
| 21:11.37 | MrTelephone | :P |
| 21:12.32 | [TK]D-Fender | MrTelephone, you and Dr-Linux show combine forces... together you would be UNSTOPPABLE. |
| 21:13.09 | [TK]D-Fender | MrTelephone, not that the assault seems like anything but non-stop as it is ;) |
| 21:15.03 | MrTelephone | hahah |
| 21:15.04 | *** part/#asterisk sergee (i=kvirc@195.94.224.197) |
| 21:15.19 | MrTelephone | look at me I can't find the driver that loads my network cards :-/ |
| 21:15.20 | MrTelephone | eek |
| 21:15.38 | [TK]D-Fender | MrTelephone, "dmesg|more" |
| 21:15.39 | *** join/#asterisk kolian123 (n=kolian@124.107.63.223) |
| 21:15.51 | kolian123 | Hello |
| 21:16.09 | kolian123 | Hi [TK] |
| 21:20.22 | MrTelephone | thats crazy |
| 21:20.26 | MrTelephone | still can't pinpoint it |
| 21:20.27 | MrTelephone | Probing IDE interface ide0... |
| 21:20.27 | MrTelephone | eth1: Tigon3 [partno(BCM95705A50) rev 3003 PHY(5705)] (PCI:33MHz:32-bit) 10/100/1000BaseT Ethernet 00:17:31:1b:31:af |
| 21:20.35 | *** join/#asterisk CBU[^_^]M`` (n=love@210.213.139.67) |
| 21:20.39 | MrTelephone | ide0? |
| 21:20.40 | MrTelephone | heh |
| 21:20.58 | [TK]D-Fender | BCM95705A50 |
| 21:21.06 | [TK]D-Fender | Broadcom... whats the hard part? |
| 21:21.37 | [TK]D-Fender | BCM95705A50-D BROADCOM GIGABIT NIC PCI BCM95705A50-D |
| 21:21.54 | MrTelephone | theres no modules with any similar markings |
| 21:22.14 | MrTelephone | lsmod displays modules built into the kernel too that are loaded so it should be there |
| 21:23.08 | MrTelephone | tg3 is the module |
| 21:24.25 | *** join/#asterisk SuperID (n=gary@c-65-96-225-97.hsd1.ma.comcast.net) |
| 21:24.42 | waKKu | lsmod doesnt shows built-in modules afaik |
| 21:24.51 | *** join/#asterisk kolian123 (n=kolian@124.107.63.223) |
| 21:25.30 | kolian123 | Anybody here familiar with PRI configuration |
| 21:25.33 | kolian123 | :_ |
| 21:28.57 | *** join/#asterisk CunningPike_ (n=CunningP@204.239.12.183) |
| 21:33.17 | *** join/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
| 21:34.58 | *** join/#asterisk kolian123 (n=kvirc@124.107.63.223) |
| 21:36.10 | EricL | [TK]D-Fender: I figured out the problem. It turns out that the variable 'fromuser' in sip.conf was being set and that was forcing the callerid to always come up as the extension of fromuser. |
| 21:36.21 | EricL | [TK]D-Fender: Thanks for the help and sorry for being a PITA. |
| 21:38.50 | Maxxed | http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&item=130121981790&category=18839 |
| 21:41.04 | kolian123 | Hi TK |
| 21:41.59 | kolian123 | Read the IRC log that you were helping somebody with issues on PRI |
| 21:42.27 | *** join/#asterisk ToyMan (n=Stuart@user-12lcqol.cable.mindspring.com) |
| 21:43.41 | [TK]D-Fender | kolian123, ... |
| 21:43.41 | [TK]D-Fender | ~ask |
| 21:43.48 | jbot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there, just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 21:44.00 | kolian123 | Hi |
| 21:44.29 | kolian123 | Can't bring PRI up...can't find D-channel |
| 21:44.38 | kolian123 | hold on i will paste the errors |
| 21:44.44 | [TK]D-Fender | ~pb |
| 21:44.46 | jbot | hmm... pb is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste, or at paste.lisp.org |
| 21:44.47 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^ |
| 21:44.57 | kolian123 | yes |
| 21:45.05 | [TK]D-Fender | kolian123, And make sure you pastebin your configs too |
| 21:45.35 | kolian123 | Yes:) let me paste it there |
| 21:46.27 | *** part/#asterisk EvilGreen_ (n=Miranda@ppp85-140-136-133.pppoe.mtu-net.ru) |
| 21:48.20 | *** join/#asterisk mvanbaak (n=mafkees@vanbaak.xs4all.nl) |
| 21:48.51 | kolian123 | http://www.pastebin.ca/565136 |
| 21:49.23 | kolian123 | seems like a few people encountering this...but haven't seen a resolution on google |
| 21:49.42 | kolian123 | i'm running TE405P with echo cancellation module |
| 21:51.39 | johann8384 | I've got a customer with a Teleadvantage system that works just fine as long as no one is making an outbound call. If someone is on an outbound call the INVITES for the inbound call are never responded to |
| 21:51.47 | johann8384 | Anyone have a suggestion? |
| 21:52.27 | [TK]D-Fender | kolian123, pastebin "cat /proc/interrupts" |
| 21:53.08 | [TK]D-Fender | johann8384, pastebin some debug info . |
| 21:53.09 | [TK]D-Fender | ~pb |
| 21:53.10 | jbot | from memory, pb is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste, or at paste.lisp.org |
| 21:54.01 | kolian123 | http://www.pastebin.ca/565146 |
| 21:54.52 | [TK]D-Fender | kolian123, wat ver of */zaptel? And ned PB "dmesg" |
| 21:55.07 | kolian123 | 1.4.3 |
| 21:55.19 | kolian123 | also the same with 1.4.2.1 |
| 21:55.49 | kolian123 | i will get dmesg for you...one thing that i'm wondering is this message |
| 21:56.22 | kolian123 | [ 11.028812] timing source auto card 0! |
| 21:56.56 | crichardson | anyone here usings snom 3xx phones? |
| 21:58.33 | kolian123 | http://www.pastebin.ca/565154 |
| 21:59.14 | kolian123 | could be something in config saying that SPAN1 is a primary timing source, though we have 1 to get timing from PRI |
| 22:00.59 | *** join/#asterisk bbryant (i=brett@nat/digium/x-ef8fd8caa6ddf676) |
| 22:01.30 | johann8384 | [TK]D-Fender: sorry it took so long, http://www.pastebin.ca/565157 |
| 22:02.07 | [TK]D-Fender | kolian123, Ok,no idea.... |
| 22:02.36 | kolian123 | everything looks good in config? |
| 22:02.37 | [TK]D-Fender | johann8384, sip debug, not core. |
| 22:02.42 | [TK]D-Fender | kolian123, Seems fine |
| 22:03.05 | kolian123 | could be bad line? |
| 22:03.16 | johann8384 | [TK]D-Fender: I'm getting that now, that is an tethereal output |
| 22:03.21 | *** join/#asterisk javar (n=javar@69.79.134.24) |
| 22:03.25 | kolian123 | Are there a lot of people having the same problem? |
| 22:04.31 | *** join/#asterisk unspin (n=unspin@24.82.161.85) |
| 22:04.42 | [TK]D-Fender | kolian123, not that I know of. |
| 22:05.04 | *** join/#asterisk Igbothom_III (n=Hilton@office.quarkit.com.au) |
| 22:05.15 | kolian123 | Thanks for help |
| 22:05.40 | kolian123 | Would you guess it's probably bad line, dead card, wrong config on telco side? |
| 22:06.25 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 22:07.09 | Cresl1n | looks like wrong config |
| 22:07.23 | Cresl1n | or a bad line |
| 22:10.47 | *** join/#asterisk Igbothom_III (n=Hilton@office.quarkit.com.au) |
| 22:11.19 | *** join/#asterisk diclophis-work (n=jbardin@65.203.37.58) |
| 22:11.29 | diclophis-work | hello all |
| 22:11.36 | diclophis-work | how could i roll my own gsm gateway? |
| 22:12.01 | *** part/#asterisk EricL (n=eric@74.9.83.194) |
| 22:12.02 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 22:13.22 | johann8384 | [TK]D-Fender: doesn't say much :) http://www.pastebin.ca/565179 |
| 22:13.24 | [TK]D-Fender | diclophis-work, First start with a dremel tool and get rid of those 90 deg edges... |
| 22:13.38 | diclophis-work | haha |
| 22:13.48 | diclophis-work | yea |
| 22:13.56 | diclophis-work | uh, for reals though |
| 22:14.03 | diclophis-work | is there like, linux gsm cards? |
| 22:14.12 | diclophis-work | er gsm interfaces |
| 22:14.21 | diclophis-work | interface cards? |
| 22:14.39 | *** join/#asterisk _charly_ (i=kroseneg@sunrise.schmidham.net) |
| 22:15.13 | [TK]D-Fender | johann8384, Sure it does |
| 22:15.14 | [TK]D-Fender | Looking for s in default (domain 209.40.224.166) |
| 22:15.20 | [TK]D-Fender | SIP/2.0 404 Not Found |
| 22:15.29 | javar | diclophis-work, i guess GSM gateways |
| 22:15.56 | diclophis-work | yea, the gsm gateways probably run on top of linux or sonmething, but they are all "black boxes" |
| 22:16.11 | kolian123 | Cresl1n, could you take a look at the config? |
| 22:16.34 | kolian123 | http://www.pastebin.ca/565136 |
| 22:16.50 | *** join/#asterisk Taadow (n=super@70.70.0.33) |
| 22:17.17 | voipnet-tech | http://www.voipcoop.org/forums/viewtopic.php?t=39 |
| 22:20.37 | voipnet-tech | 25 Channel SIP Trunking for under $100 / month |
| 22:22.26 | diclophis-work | is it possible to get toll-free numbers terminated to a SIM / GSM point? |
| 22:22.44 | Taadow | Anyone ever seen cascading SUBSCRIBEs when viewing sip show channels, from a single extension? When this occurs no-one else can register w/ the pbx. http://www.pastebin.ca/565181 |
| 22:23.08 | Taadow | It happens when certain staff register w/ their softphones. |
| 22:23.50 | johann8384 | [TK]D-Fender: so it does |
| 22:24.41 | johann8384 | [TK]D-Fender: actually, that's a different call-id |
| 22:27.28 | [TK]D-Fender | johann8384, pastebin your sip.conf masking only passwords |
| 22:28.18 | *** join/#asterisk pagec (n=pagec@cpe-74-73-191-68.nyc.res.rr.com) |
| 22:28.24 | *** part/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net) |
| 22:28.58 | johann8384 | it's realtime with about 200 entries. This guy may have it figured out, I'm waiting for him to try some things with his firewall |
| 22:29.11 | pagec | i haven't been able to find it on google; Does anyone here know where i would find how to get Polycom Soundpoint phones call history to properly work with an asterisk server? |
| 22:29.32 | johann8384 | [TK]D-Fender: thanks for your help, i'll hit you up again if I need more help |
| 22:30.14 | Taadow | Can anyone help with this please? http://www.pastebin.ca/565181 Is this normal to see so many SUBSCRIBEs for a single extension? |
| 22:30.45 | mcab | pagec: what's going wrong? |
| 22:37.58 | pagec | mcab: The polycom shows under Features->Call List->Missed Calls "1015551212@192.168.1.20", when i press the dial softkey nothing happens. i have logging on the asterisk server turned up to 10 and i see nothing on the console and the phones log on the server also shows nothing |
| 22:43.18 | mcab | pagec: hmmm, works for me when I dial-by-IP. could be you have url-dialing disabled on your SoundPoints... |
| 22:43.24 | *** join/#asterisk PaulTech85 (i=PaulTech@72.29.76.254) |
| 22:44.00 | mcab | but seems odd that you have <number>@<IP> in your call list - usually I just see <number> |
| 22:44.04 | PaulTech85 | Any fun known way to get calls in progress or is using the asterisk Manager API it? Trying to make a application that shows calls in progress but only way is to hit up the manager API for a list.. alot of overheading in connecting every X Seconds |
| 22:44.39 | mosty | PaulTech85, listen for events via AMI? just connect once |
| 22:44.42 | pagec | mcab: yes i do have <number>@<IP>, how do i make it just <number>? I am using asterisk 1.4 |
| 22:45.12 | mcab | pagec: sadly, I don't know asterisk that well :-7 |
| 22:45.42 | PaulTech85 | mosty: So set Events to 'call' and just somehow parse the calls that come in? |
| 22:45.53 | PaulTech85 | I guess I could connect and see what kind of output it gives and see if I can parse it |
| 22:46.01 | *** join/#asterisk CrashSys (n=kumba@158-211.187-72.tampabay.res.rr.com) |
| 22:46.07 | PaulTech85 | Problem is I need to pass this to a pretty frontend so not sure how im going to do that yet |
| 22:46.17 | pagec | mcab: ok, how do i enable url dialing? |
| 22:46.36 | mosty | PaulTech85, something like that |
| 22:47.10 | mcab | pagec: it's a config parameter in the sip.cfg file for the Polycoms, let me check which one |
| 22:48.01 | pagec | mcab: and you would happen to know how to get asterisk to change what it says its IP address is for the <IP> part would you? it is reporting it's IP address on the non-phone network |
| 22:48.37 | mcab | pagec: feature.9.name="url-dialing" feature.9.enabled="1" (The .9. might be different in your config though) |
| 22:48.54 | mcab | pagec: no idea, sorry |
| 22:49.52 | Toerkeium | guys, which is the packet loss % which a comunication can have before the comunication hungs up? or acceptable maximun packet loss ? |
| 22:50.45 | pagec | mcab: hmm that is on and i put the one on the network that works with <IP>, but still the dial of the call log doens't work, any other ideas? |
| 22:50.50 | blitzrage | until the user hangs up the phone :) |
| 22:51.06 | blitzrage | in chan_sip it seems to be 6 retransmissions |
| 22:51.38 | Toerkeium | blitzrage: what does it means? |
| 22:51.50 | blitzrage | pretty much what I just said... |
| 22:52.07 | blitzrage | once the call is setup, chan_sip doesn't care about how many packets are dropped |
| 22:52.14 | blitzrage | since it's just RTP being sent |
| 22:52.26 | blitzrage | which is a different protocol |
| 22:52.46 | *** join/#asterisk Sycofant (n=Dylan@ip-58-28-130-88.ubs-dsl.xnet.co.nz) |
| 22:52.57 | *** join/#asterisk mirco (n=mirco@tmo-055-9.customers.d1-online.com) |
| 22:53.30 | Sycofant | Hi, I have an AsteriskNow problem, but there's not a whole lot of activity in that channel, anyone here alive and interested in helping? |
| 22:53.39 | Toerkeium | blitzrage: and to what would affect the packet loss ? |
| 22:53.57 | blitzrage | Toerkeium: a poor network... |
| 22:54.22 | blitzrage | routers being over-burdened with packets which fill up their queue and they decide to start dropping them.... |
| 22:54.26 | blitzrage | lots of things |
| 22:54.56 | blitzrage | I wouldn't want to have more than 1-2% packet loss |
| 22:55.04 | Toerkeium | blitzrage: I am trying to find how many packet loss would affect the communication between 2 users in terms of percentage |
| 22:55.10 | Toerkeium | oh.. nice |
| 22:55.32 | blitzrage | having 1-2% packet loss would be a fair amount... |
| 22:55.40 | blitzrage | but still usable |
| 22:56.13 | blitzrage | Toerkeium: to increase you english skills (just trying to be helpful) -- a better sentence would use 'much' instead of 'many' |
| 22:56.14 | Toerkeium | blitzrage: excellent, thank you |
| 22:56.32 | Sycofant | [Jun 14 10:55:38] NOTICE[2536] chan_sip.c: Registration from '<sip:201@192.168.1.112>' failed for '192.168.1.51' - Not a local domain |
| 22:56.37 | Sycofant | This is my error. |
| 22:56.41 | Toerkeium | your english suggestion is very appreciated |
| 22:56.42 | Toerkeium | :) |
| 22:56.49 | blitzrage | Sycofant: turn on autodomain |
| 22:56.59 | Sycofant | Okay, hand on. |
| 22:57.02 | Sycofant | hang on even. |
| 22:57.04 | blitzrage | Toerkeium: ...suggestion is very much appreciated :D |
| 22:57.25 | blitzrage | english must be confusing as all hell to learn |
| 22:57.40 | Sycofant | In the AsteriskNow GUI 'Auto Domain' is checked. |
| 22:57.44 | blitzrage | to, two, too... different meanings, sound the same |
| 22:57.59 | Toerkeium | blitzrage: lol |
| 22:58.35 | Toerkeium | blitzrage: is "very appreciated" wrong? or you just think I have to be more thankful ? |
| 22:58.44 | Sycofant | blitzrage: It's their, they're and there that seem to flummox most people. |
| 22:59.03 | Toerkeium | I am glad that someone correct my english, but you will have a hard work with me :P) |
| 22:59.07 | *** join/#asterisk thekidrio (n=thekidri@66.107.42.12) |
| 22:59.39 | blitzrage | Toerkeium: haha... well, saying "very much appreciated" would be better grammer -- and you could even just drop the 'very' and just say, "much appreciated" |
| 22:59.41 | thekidrio | could I get someone to call me? The only phone I have avail outside the system is my cell and I just realized I don't have any reception in the server room |
| 22:59.54 | Sycofant | blitzrage: autodomain is also set in the sip.conf |
| 22:59.54 | Corydon76-home | blitzrage: how goes it? |
| 22:59.58 | thekidrio | its grammar |
| 23:00.01 | thekidrio | not grammer :P |
| 23:00.06 | Toerkeium | heheh nice! |
| 23:00.09 | blitzrage | thekidrio: haha... yes... stupid typo :) |
| 23:00.17 | blitzrage | Corydon76-home: oh it goes! |
| 23:00.18 | thekidrio | hehe its ok, I do that one all the time |
| 23:00.32 | blitzrage | thekidrio: ya, until recently I was doing "rediculous" instead of "ridiculous" |
| 23:00.36 | thekidrio | anyone avail for a quick call test? |
| 23:00.41 | blitzrage | thekidrio: sure, what number |
| 23:00.42 | thekidrio | blitzrage, hehe spell check on irc helped me a ton |
| 23:00.53 | blitzrage | thekidrio: hehe... ya, I don't have a spell checker turned on here :) |
| 23:01.03 | mcab | pagec: nope, that's all from me. sorry :-7 |
| 23:02.03 | thekidrio | wahoo |
| 23:02.16 | blitzrage | thekidrio: where are you located? |
| 23:02.21 | thekidrio | i have asterisk running on my mini itx computer haha |
| 23:02.27 | thekidrio | I am in Brea california atm |
| 23:02.27 | blitzrage | fun stuff |
| 23:02.32 | blitzrage | gotcha |
| 23:02.35 | blitzrage | you sound scottish :) |
| 23:02.40 | thekidrio | yeah |
| 23:02.47 | thekidrio | I am losing the accent slowly |
| 23:02.56 | blitzrage | ya, I could hear it fade in an out |
| 23:03.00 | blitzrage | keep it -- chicks dig accents |
| 23:03.09 | thekidrio | haha, gets you free beer too |
| 23:03.09 | blitzrage | I have no accent until I leave Canada and go to the US :) |
| 23:03.20 | thekidrio | my canadian accent is great |
| 23:03.31 | Corydon76-home | eh? |
| 23:03.33 | thekidrio | lot of scots up there |
| 23:03.37 | blitzrage | indeed |
| 23:03.50 | *** join/#asterisk rmayorga_ (i=rmyorg@unaffiliated/rmayorga) |
| 23:03.50 | blitzrage | Corydon76-home: :D |
| 23:04.01 | *** part/#asterisk diclophis-work (n=jbardin@65.203.37.58) |
| 23:04.16 | Sycofant | blitzrage: Anything else I should be looking at with this 'not a local domain' thing? |
| 23:05.00 | blitzrage | Sycofant: hrmm... not sure.. I don't typically do domain stuff |
| 23:05.33 | Sycofant | That's cool. Thanks. |
| 23:05.43 | Sycofant | Frustrating... Aiiiieeee. |
| 23:11.26 | PaulTech85 | thekidrio what part of scotland? |
| 23:11.34 | thekidrio | dundee |
| 23:11.37 | thekidrio | and port glasgow |
| 23:11.40 | PaulTech85 | <- Dunoon |
| 23:11.46 | PaulTech85 | Moved here about 7 years ago now thou |
| 23:11.46 | thekidrio | sweet |
| 23:11.48 | thekidrio | nice |
| 23:11.53 | PaulTech85 | and yes -- at losing the accent |
| 23:11.56 | thekidrio | I moved here, yeah about that a little more |
| 23:12.03 | PaulTech85 | Cant hear it in my voice no more untill I talk to a scot then it comes flying back :-) |
| 23:12.06 | thekidrio | when I got married we moved here |
| 23:12.09 | thekidrio | haha yeah same here |
| 23:12.18 | thekidrio | or the more whisky i drink |
| 23:12.31 | thekidrio | well whisky/belhaven/macewans |
| 23:12.41 | thekidrio | I miss it so much haha |
| 23:12.51 | thekidrio | I came to work last week in my sport kilt |
| 23:13.03 | CrashSys | sport kilt? |
| 23:13.15 | thekidrio | yeah, a modern easier to wear kilt heh |
| 23:13.18 | Sycofant | CrashSys: I wouldn't. |
| 23:13.31 | thekidrio | I have a great kilt passed down too |
| 23:13.34 | CrashSys | Maybe I want to wear one to work... maybe people will leave me alone... |
| 23:13.35 | thekidrio | but I don't wear that |
| 23:13.37 | Sycofant | thekidrio: A skirt, come one, own up. |
| 23:13.44 | thekidrio | heh |
| 23:13.49 | thekidrio | well kilt actually means pleat |
| 23:14.04 | thekidrio | so calling it a kilt is calling it a pleated skirt ;P |
| 23:14.06 | Sycofant | But do you wear the sporan too? |
| 23:14.15 | thekidrio | yup, i made my own sporran |
| 23:14.26 | thekidrio | carved the clan badge into it heh |
| 23:14.31 | Sycofant | Complete with rabbit, is it? |
| 23:14.37 | thekidrio | horse hair |
| 23:14.46 | thekidrio | but yeah its got fur :) |
| 23:14.49 | Sycofant | Is it sometimes rabbit, or am I mental? |
| 23:14.55 | thekidrio | it is often rabbit |
| 23:14.56 | thekidrio | or fox |
| 23:15.11 | CrashSys | so a pleated skirt and a purse |
| 23:15.12 | thekidrio | i never got into my roots so much as the week after i left home |
| 23:15.17 | thekidrio | yup |
| 23:15.36 | thekidrio | although, i would not say that to my face in a bar :P |
| 23:15.38 | *** join/#asterisk mindCrime_ (n=chatzill@cpe-065-190-188-124.nc.res.rr.com) |
| 23:17.44 | CrashSys | thekid: I'm not irish enough to tell a scot that in a bar :) |
| 23:18.24 | CrashSys | or was it the irish and italians that didn't get along |
| 23:18.34 | Sycofant | The irish and everyone. |
| 23:18.37 | CrashSys | Ahhhh |
| 23:18.48 | Sycofant | And the italians and everyone. |
| 23:18.57 | Sycofant | Actually everyone and everyone really. |
| 23:19.17 | thekidrio | i love the irish |
| 23:19.28 | thekidrio | shit whiskey (notice the e) but good beer |
| 23:19.31 | thekidrio | and great brawlers |
| 23:19.43 | thekidrio | americans do not understand the idea of a friendly bar brawl |
| 23:20.06 | CrashSys | Too many sensitive types... |
| 23:20.06 | Corydon76-home | Friendly? |
| 23:20.31 | *** join/#asterisk kiscokid (n=ron@208.106.33.66) |
| 23:20.51 | thekidrio | yah, no anger afterwards |
| 23:20.56 | thekidrio | and no fuzz |
| 23:21.09 | thekidrio | 2 of my very best friends I met fighting them |
| 23:21.24 | CrashSys | Beer gets spilled, everyone takes a few knocks, then you buy more beer... a friendly bar fight... |
| 23:22.14 | thekidrio | nothing like a good scottish punk show |
| 23:22.23 | thekidrio | even the vegan punks could throw down every now and then |
| 23:26.04 | CrashSys | I didn't know vegans could fight... I figured the lack of red meat made them too anemic :) |
| 23:26.14 | thekidrio | haha |
| 23:26.23 | thekidrio | oi polloi is a crazy scottish punk band |
| 23:26.25 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 23:26.33 | thekidrio | they sing about soy ice cream though.. very odd |
| 23:26.40 | CrashSys | I'll pass |
| 23:26.43 | thekidrio | haha |
| 23:26.46 | CrashSys | I didn't care much for punk bands anyways |
| 23:26.56 | thekidrio | I did, not so much anymore |
| 23:27.06 | thekidrio | Punk died for me around 1985 |
| 23:27.14 | blitzrage | I barely remember 1985 :) |
| 23:27.21 | blitzrage | I was 4, heh |
| 23:27.23 | CrashSys | I'm more of an 80's metal fan... |
| 23:27.33 | blitzrage | the 80s sicken me :) |
| 23:27.35 | CrashSys | and not that glam-rock big-hair bullshit... |
| 23:27.39 | blitzrage | long live the 70s! |
| 23:27.39 | thekidrio | haha blitzrage |
| 23:27.40 | *** part/#asterisk kiscokid (n=ron@208.106.33.66) |
| 23:27.48 | thekidrio | now I listen to underground hip hop haha |
| 23:28.05 | thekidrio | maybe polka is next |
| 23:28.19 | thekidrio | speaking of polka, anyone know a good free radio stream for music on hold? |
| 23:28.54 | thekidrio | wahoo off for the day! |
| 23:29.05 | thekidrio | cya all tomorrows |
| 23:29.25 | CrashSys | shoutcast? |
| 23:29.46 | CrashSys | dunno if mpg321 will play shoutcast streams |
| 23:34.16 | *** join/#asterisk _charly_ (i=kroseneg@sunrise.schmidham.net) |
| 23:41.36 | *** join/#asterisk boch (n=fran@190.48.205.198) |
| 23:43.23 | *** join/#asterisk Snake-Eyes (n=blog@70.55.220.203.static.comindico.com.au) |
| 23:44.08 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |