| 00:01.57 | *** join/#asterisk zno (~chatzilla@user-0cdfece.cable.mindspring.com) |
| 00:04.10 | zno | what's the difference between a ISDN PRI and a T1 when using asterisk? |
| 00:04.26 | daork | not this again |
| 00:04.33 | daork | ISDN PRI runs OVER T1 |
| 00:04.58 | daork | someone needs to write a "Telephony 101" |
| 00:05.00 | zno | I'm learning |
| 00:05.06 | daork | maybe it should be me |
| 00:05.24 | zno | the world would thank you for your efforts |
| 00:05.42 | zno | I'm trying to order services |
| 00:05.51 | tainted- | zno PRI has 23 voice channels and 1 data channel |
| 00:06.02 | tainted- | T1 is 24 data channels |
| 00:06.28 | tzanger | tainted-: well either way data or voice or any combo thereof |
| 00:06.32 | zno | according to Verizon's website, the channels can be configured for either voice or data on both, unless I'm reading it wrong |
| 00:06.40 | tzanger | daork: it also runs over E1 and J1 |
| 00:06.45 | tainted- | zno that's correct |
| 00:06.48 | Tuplink | SHOULD I BE using fxs_ls indtead of fxs_ks if my hangup isnt working rite? |
| 00:07.03 | Tuplink | n.m |
| 00:07.06 | tainted- | zno u get 24 in any combo u want |
| 00:07.10 | ManxPower | ISDN is like IP, it can run over pretty much anything. |
| 00:07.27 | daork | tzanger: it sure does |
| 00:07.33 | tainted- | what is J1? |
| 00:07.36 | *** join/#asterisk kimc (~freenode@pcp09643046pcs.wbrmfd01.mi.comcast.net) |
| 00:07.37 | ManxPower | nobody supports it except for T/E/J-1, of course. |
| 00:07.51 | tzanger | tainted-: japanese "just enough to be different" TDM standard |
| 00:07.57 | tainted- | i know E1 is 2Mbps |
| 00:08.03 | tainted- | oh |
| 00:08.17 | tainted- | E1 is the euro standard right? |
| 00:08.21 | daork | j1 is japanese T1 |
| 00:08.23 | tzanger | yeah |
| 00:08.24 | daork | yea, e1 is euro |
| 00:08.29 | ManxPower | tzanger: Of course! They have to support Kanji! |
| 00:08.38 | daork | we have e1 in oceanea too |
| 00:08.42 | tzanger | wow |
| 00:08.44 | tainted- | and Hiragana! lol |
| 00:08.44 | daork | (ie, australia and NZ) |
| 00:08.50 | fugitivo | argentina uses e1 too |
| 00:08.53 | tzanger | Wonder brand hamburger buns WILL eventually go moldy |
| 00:08.55 | tainted- | E1 is pretty much everyeone except US |
| 00:08.58 | daork | yeah |
| 00:09.01 | daork | tainted-: and japan |
| 00:09.10 | daork | tainted-: as they, as we covered, use J1 ;) |
| 00:09.23 | daork | i think a few other places in asia are J1 as well |
| 00:09.28 | daork | dont ask me who though |
| 00:09.30 | tzanger | I *think* J1 was developed as part of some protectionism/isolationism scheme |
| 00:09.40 | tzanger | support the japanese industries because nothing works with it |
| 00:09.49 | tainted- | wouldn't be surprised |
| 00:09.55 | daork | you can get converters |
| 00:10.04 | tainted- | wouldn't be surprised if there was a C-1 for china |
| 00:10.10 | daork | its probably just XOR'd or something |
| 00:10.11 | daork | heh |
| 00:10.12 | zno | so correct me if I'm wrong, T1's are ordered in fractions, but ISDNs are ordered one channel at a time? |
| 00:10.18 | daork | tainted-: they use POTS ;) |
| 00:10.25 | tainted- | lol |
| 00:10.45 | daork | zno: yeah |
| 00:11.00 | daork | zno: if you order one ISDN channel over a t1, you're getting two t1 fractions |
| 00:11.11 | daork | (one is an ISDN B channel, and one D channel) |
| 00:11.20 | file | yay D channel |
| 00:11.25 | |Vulture| | lol |
| 00:11.40 | tainted- | how many B channels can a D channel support? |
| 00:11.54 | |Vulture| | 23 |
| 00:12.03 | daork | nope |
| 00:12.06 | daork | 31 |
| 00:12.16 | |Vulture| | a T1 pri only has 23 voice channels |
| 00:12.20 | tainted- | is that the physical limitation? |
| 00:12.26 | daork | |Vulture|: and E1 has 31. |
| 00:12.28 | ManxPower | tainted-: 23/31, but with NFAS it can support more. How MANY more depends on your call volume. |
| 00:12.37 | |Vulture| | oh didn't know we were talking e1 |
| 00:12.50 | daork | zno: if you order two ISDN channels, you're getting 3 T1 fractions (1D + 2B) |
| 00:13.00 | daork | zno: the D is your signalling channel. |
| 00:13.09 | zno | ok |
| 00:13.17 | daork | and you only need one per T1 |
| 00:14.15 | daork | right |
| 00:14.19 | daork | lunch time here |
| 00:14.21 | daork | biab |
| 00:14.26 | *** join/#asterisk dizzan (~dan@24.115.99.243.res-cmts.haz.ptd.net) |
| 00:14.40 | *** join/#asterisk rvhi (~rv@66.175.65.89) |
| 00:18.58 | rvhi | anyone knows if voicemail supports playing last message first? |
| 00:19.14 | rvhi | now it plays from oldest to the latest. |
| 00:19.40 | *** join/#asterisk Trepalium (~chadk@wnpgmb02dc1-57-202.dynamic.mts.net) |
| 00:19.41 | |Vulture| | Anyone ever mess with *-Stat? |
| 00:20.06 | |Vulture| | like the code... not installing it |
| 00:20.20 | *** join/#asterisk tessier (~treed@210.245.38.228) |
| 00:21.53 | file | I SHALL ATTACK THEE |
| 00:22.27 | |Vulture| | right.... |
| 00:31.34 | bkw_ | OBEY THE FIST!!! |
| 00:32.00 | bkw_ | file dear... you are in 64bitness also eh? |
| 00:32.05 | bkw_ | but mine's a G5 |
| 00:32.27 | blitzrage | quick, someone send me a 64bit machine! |
| 00:32.37 | blitzrage | don't think... just do |
| 00:32.48 | bkw_ | blitzrage, do you have a need for one? |
| 00:32.56 | blitzrage | bkw_: thats a retorical question |
| 00:33.05 | drumkilla | jealous, i know ... |
| 00:33.12 | blitzrage | drumkilla: you need a new machine before me... |
| 00:33.31 | bkw_ | drumkilla, might get one if he will respond to the peeps |
| 00:33.35 | bkw_ | I got drumkilla da hookup |
| 00:33.41 | drumkilla | I responded to everyone |
| 00:33.49 | bkw_ | then expect em ;) |
| 00:34.03 | blitzrage | bkw_ / drumkilla: I noticed that on the mailing list :) |
| 00:34.08 | blitzrage | drumkilla: you certainly need a new machine though |
| 00:34.18 | blitzrage | drumkilla: I'd send you one if I had one |
| 00:34.26 | drumkilla | I appreciate it :) |
| 00:34.32 | drumkilla | it does the trick, I guess |
| 00:34.36 | blitzrage | haha... their just words though |
| 00:34.41 | blitzrage | they're*( |
| 00:34.47 | drumkilla | handles my personal calls :) |
| 00:34.50 | *** join/#asterisk AgiNamu (~AgiNamu@216.230.151.230) |
| 00:34.52 | AgiNamu | Hello |
| 00:34.57 | blitzrage | drumkilla: oh, I have a link I want to show you |
| 00:35.01 | AgiNamu | can someone here say exactly WTF is a session border controller? |
| 00:35.04 | drumkilla | uh oh! |
| 00:35.10 | drumkilla | AgiNamu: a buzz word |
| 00:35.12 | blitzrage | drumkilla: its a good one :) |
| 00:35.16 | AgiNamu | i keep hearing people say "OH, sip issues, we use a session border controller!" |
| 00:35.18 | AgiNamu | and shit like that |
| 00:35.29 | AgiNamu | "oh, you can buy our softswitch and OSS, but we recommend adding in an SBC" |
| 00:35.29 | blitzrage | ummmm... aka proxy :) |
| 00:35.54 | AgiNamu | my somethign sense is telling me it's half bs |
| 00:36.11 | AgiNamu | cause how do you just slam a device down on your network and magically like add all sorts of shit |
| 00:36.15 | blitzrage | drumkilla: I came up with the idea and Olle went off and expanded the thought |
| 00:36.22 | bkw_ | AgiNamu, in some sense of the word asterisk is a SBC |
| 00:36.45 | AgiNamu | cause these guys at like, say, Juniper are apparently selling a ton of them |
| 00:36.55 | AgiNamu | but as far as i can tell, it doesn't actually do anything necesary. |
| 00:37.11 | blitzrage | I sell stuff all the time people don't need :) (lol, not really though) |
| 00:37.26 | bkw_ | AgiNamu, well it some cases it can take care of all the stupid ugly crap |
| 00:37.36 | bkw_ | sip and other incompatibilites |
| 00:37.41 | bkw_ | SBC is something that is a crutch |
| 00:37.46 | bkw_ | fix the shit.. and you don't need it |
| 00:38.01 | iq | Hi, can network delay cause SIP clients to go to voice mail instead of ringing? |
| 00:38.13 | bkw_ | http://www.newport-networks.com/whitepapers/scwpes2.html |
| 00:38.20 | bkw_ | iq yes |
| 00:38.30 | bkw_ | so can lost nat translation |
| 00:39.03 | iq | bkw_, lost nat translation, on client side or server side? |
| 00:39.25 | bkw_ | client side |
| 00:40.23 | rvhi | I am going to post a US100 bounty for two voicemail features |
| 00:40.33 | rvhi | open source this time |
| 00:40.37 | shido6 | like what? |
| 00:40.40 | rvhi | want to know if anyone interested |
| 00:40.46 | rvhi | feature 1, one new option to play message from latest to the oldest. Now it only supports playing from oldest to latest. I'd like this as an option for each mailbox. Therefore some users have latest first while others have oldest first. |
| 00:40.53 | rvhi | feature 2, simplified folder, only two folders -- new and old. Now it supports several folders. Some users are overwhelmed by this. Most voicemail system only supports two folders new and old. When they save, it goes to old folder. Again, I'd like this as an option for each mailbox. |
| 00:41.05 | AgiNamu | thanks bkw |
| 00:41.12 | shido6 | not a bad idea |
| 00:42.14 | iq | bkw_, the problem I'm facing is that: ATA clients connected to a server in CA do not always get ring. But if the same ATA client connects to another SIP server that is in South Asia it always work fine.... so I think this could be the network delay on the SIP server in CA ? |
| 00:44.49 | *** join/#asterisk phantasis (~phantasis@c68.112.204.241.eau.wi.charter.com) |
| 00:45.03 | phantasis | how is dundi utilized? |
| 00:47.00 | *** join/#asterisk Kumbang (~ecvs@167.205.24.4) |
| 00:47.00 | tainted- | with packets and stuff |
| 00:47.16 | phantasis | but how does it find peers? |
| 00:51.26 | *** join/#asterisk remmo (~rem@smack.isp.net.au) |
| 00:51.29 | remmo | weee |
| 00:51.49 | *** join/#asterisk twisted (~twisted@twisted-professional-pdpc.developer.and.friend.of.asterisk) |
| 00:51.49 | *** mode/#asterisk [+o twisted] by ChanServ |
| 00:52.17 | tainted- | i think u have to tell it which peers u trust |
| 00:52.37 | tainted- | there's no central ENUM |
| 00:52.46 | phantasis | oh |
| 00:52.52 | phantasis | so there has to be some arrangement |
| 00:52.58 | phantasis | it just doesn't go out searching |
| 00:53.07 | tainted- | no |
| 00:53.15 | tainted- | someone who has implemented DUNDi pipe up |
| 00:53.55 | tainted- | Because DUNDi does not advertise routes but instead simply answers |
| 00:53.55 | tainted- | queries, it is not necessary to explicitly publish changes in the peering |
| 00:53.55 | tainted- | system with respect to new peering agreements, additions, removals or |
| 00:53.55 | tainted- | relocation of answers within the system |
| 00:53.58 | *** join/#asterisk Entegrity (~Entegrity@c-65-96-119-254.hsd1.ma.comcast.net) |
| 00:54.08 | tainted- | http://www.dundi.info/dundi.pdf - page 6 of 8 |
| 00:55.01 | *** join/#asterisk xxlukkystarxx (phreakphra@adsl-68-72-89-42.dsl.chcgil.ameritech.net) |
| 00:55.11 | xxlukkystarxx | hey guys i need a little help |
| 00:55.28 | xxlukkystarxx | im trying to build an IVR menu but i dont know what im doing |
| 00:55.39 | xxlukkystarxx | read through the tutorials.... |
| 00:55.44 | bkw_ | xxlukkystarxx, how long have you used asterisk? |
| 00:55.46 | xxlukkystarxx | really confused |
| 00:55.53 | xxlukkystarxx | lol 2 days |
| 00:56.01 | bkw_ | ok you're still atleast 7 days away from understanding |
| 00:56.07 | tainted- | lol |
| 00:56.08 | bkw_ | so us telling you HOW is not helping |
| 00:56.18 | xxlukkystarxx | no just a quick questions |
| 00:56.22 | tainted- | just ask |
| 00:56.25 | xxlukkystarxx | i dont understand the structure |
| 00:56.25 | bkw_ | the extensions.conf that gets installed when you do make samples has an example in it |
| 00:56.27 | xxlukkystarxx | here |
| 00:56.31 | bkw_ | so help me |
| 00:56.34 | bkw_ | if you paste in here |
| 00:56.35 | bkw_ | i'll smack you |
| 00:56.38 | bkw_ | www.pastebin.ca |
| 00:56.45 | bkw_ | ;) |
| 00:56.52 | tainted- | xxlukkystarxx okay bkw_ like u to say 'hi' before u ask questions |
| 00:57.04 | bkw_ | he said hey |
| 00:57.06 | bkw_ | thats fine |
| 00:57.11 | xxlukkystarxx | *she |
| 00:57.13 | tainted- | ok |
| 00:57.15 | bkw_ | oh she |
| 00:57.20 | tainted- | just ask |
| 00:57.29 | bkw_ | xxlukkystarxx, rock on.. more female asterisk users |
| 00:57.43 | tainted- | bkw_ gets his mack suit on |
| 00:57.46 | bkw_ | like I really care.. i'm as queer as a three dollar bill. |
| 00:57.51 | xxlukkystarxx | Exten => 1,1,Playback(welcome) |
| 00:58.03 | bkw_ | exten => 1,2,Dosomethingelse |
| 00:58.11 | xxlukkystarxx | 1,1 does what? |
| 00:58.16 | xxlukkystarxx | extension 1 |
| 00:58.17 | bkw_ | exten 1, priority 1 |
| 00:58.21 | xxlukkystarxx | okay |
| 00:58.24 | bkw_ | exten 1, priority 2 |
| 00:58.26 | bkw_ | and so on |
| 00:58.29 | tainted- | matches extension one, is first executed |
| 00:58.30 | bkw_ | they must be in order. |
| 00:58.49 | bkw_ | or you can also use exten => 1,1, then exten 1,n, |
| 00:58.52 | bkw_ | n being next |
| 00:59.01 | xxlukkystarxx | but how do i do i make it listen for dtmf? |
| 00:59.07 | bkw_ | use Background |
| 00:59.09 | bkw_ | instead of playback |
| 00:59.17 | bkw_ | show application playback |
| 00:59.18 | tainted- | Background(fileuwanttoplay) |
| 00:59.22 | bkw_ | show application background |
| 00:59.31 | bkw_ | same apps different results |
| 00:59.32 | xxlukkystarxx | then how do i get the dtmf input? |
| 00:59.35 | bkw_ | ok |
| 00:59.38 | bkw_ | while its playing the file |
| 00:59.40 | bkw_ | someone dials |
| 00:59.44 | xxlukkystarxx | thats my real question |
| 00:59.45 | bkw_ | it will match what they dial in the dial plan |
| 00:59.48 | bkw_ | so if you have |
| 00:59.56 | bkw_ | exten 1,1,Background(uberleetmenu) |
| 01:00.00 | bkw_ | and while thats plaing someone dials 123 |
| 01:00.11 | bkw_ | and you have exten => 123,1,blah |
| 01:00.17 | bkw_ | in the same context or included context |
| 01:00.19 | bkw_ | it will go there |
| 01:00.23 | tainted- | OMG what if they dial 456!!!1! |
| 01:00.24 | xxlukkystarxx | oh okay |
| 01:00.30 | bkw_ | tainted-, *SMACK* |
| 01:00.33 | bkw_ | I have been helpful |
| 01:00.33 | xxlukkystarxx | so if i did |
| 01:00.43 | xxlukkystarxx | exten 1,1,Background(uberleetmenu) |
| 01:00.51 | anthm | oh yeah btw |
| 01:00.56 | bkw_ | anthm, yes? |
| 01:00.58 | anthm | re: that regex thing you sent |
| 01:01.00 | xxlukkystarxx | exten => 123,1, Background(menu2) |
| 01:01.01 | anthm | emailed me |
| 01:01.04 | bkw_ | anthm, yes? |
| 01:01.08 | bkw_ | I got that from oej |
| 01:01.10 | tainted- | i like to do Background(oneSecondSilence) after my files just in case |
| 01:01.21 | anthm | do they know you can do regex now already ? |
| 01:01.24 | xxlukkystarxx | exten, 2, 453 |
| 01:01.28 | bkw_ | anthm, apparently NOT |
| 01:01.29 | tainted- | or just add some silence at the end of the file so it pads some time |
| 01:01.29 | bkw_ | haha |
| 01:01.30 | anthm | with the functon thing |
| 01:01.37 | xxlukkystarxx | then it would play a menu and goto x453? |
| 01:01.49 | bkw_ | exten => 1,1,Background(menu) |
| 01:01.58 | tainted- | xxlukkystarxx that's rigt |
| 01:01.58 | bkw_ | exten => 1,2,Goto(something) |
| 01:02.10 | bkw_ | exten => 123,1,Dial(blah/bob) |
| 01:02.16 | *** join/#asterisk Sedorox (brandon@Neptune.client.wlmsprt.pa.sed6.net) |
| 01:02.17 | bkw_ | anthm, I recommended that |
| 01:02.22 | tainted- | xxlukkystarxx it's flexible so u can do it in a lot of ways |
| 01:02.24 | xxlukkystarxx | and the final question i will bother you guys with is: |
| 01:02.33 | bkw_ | anthm, he also agrees the _. warning is stupid |
| 01:02.41 | xxlukkystarxx | how do i make a static meetme conference an extention? |
| 01:02.41 | bkw_ | _.. can kill the idiot warning |
| 01:03.12 | bkw_ | anthm, while reading that I was in kram mode.. "This will slow down the dialplan" |
| 01:03.27 | bkw_ | "This would be a performance hit like crazy" |
| 01:03.30 | bkw_ | hahaha |
| 01:03.46 | xxlukkystarxx | hmm? |
| 01:03.53 | bkw_ | xxlukkystarxx, check meetme.conf |
| 01:04.01 | bkw_ | exten => 124,1,Meetme(100) |
| 01:04.06 | bkw_ | 100 being the 100 room in meetme.conf |
| 01:04.08 | bkw_ | NEXT!!! |
| 01:04.18 | bkw_ | also you might wanna come to cluecon |
| 01:04.25 | bkw_ | Aug 3rd thru 5th :P |
| 01:04.35 | bkw_ | never know you might learn a few things |
| 01:04.38 | bkw_ | ;) |
| 01:04.47 | xxlukkystarxx | lol thanks a ton guys |
| 01:04.52 | xxlukkystarxx | think i got it figured out |
| 01:04.56 | bkw_ | now don't ask and run |
| 01:05.11 | bkw_ | hang around... and help others.... if they happen to ask a question you know the answer to ;) |
| 01:05.19 | Tuplink | any one here use FWD for their 1800 calls? |
| 01:05.28 | bkw_ | Tuplink, set from user and use *1800 |
| 01:05.30 | bkw_ | it should work |
| 01:06.05 | Tuplink | i call call 1800wwwdell but 1822555121 wont work |
| 01:07.17 | Tuplink | exten => _91800NXXXXXX,1,Dial(IAX2/${FWDNUMBER}:${FWDPASSWORD}@iax2.fwdnet.net/*${EXTEN:1}) |
| 01:08.04 | *** join/#asterisk poli (~poli@200-168-30-125.dsl.telesp.net.br) |
| 01:08.41 | Veto | Tuplink, wouldn't you need _918XXNXXXXXX? |
| 01:08.48 | bkw_ | WRONG |
| 01:08.52 | Tuplink | no.. |
| 01:08.56 | bkw_ | that will match 1818 |
| 01:09.00 | Tuplink | cause not all 800 are toal free |
| 01:09.03 | Veto | ahh, yea it would |
| 01:09.05 | bkw_ | yes they are |
| 01:09.08 | bkw_ | over here they are |
| 01:09.11 | Tuplink | oh... really |
| 01:09.19 | Tuplink | didnt know that |
| 01:09.32 | Tuplink | thought it was only 800 866 877 |
| 01:09.39 | bkw_ | 888 |
| 01:09.41 | Tuplink | 888 |
| 01:09.43 | bkw_ | 877 |
| 01:09.45 | Tuplink | yea |
| 01:09.51 | bkw_ | ya ya |
| 01:10.05 | *** part/#asterisk xxlukkystarxx (phreakphra@adsl-68-72-89-42.dsl.chcgil.ameritech.net) |
| 01:10.31 | Tuplink | but 8005558355 dont work but works from a landline |
| 01:10.44 | bkw_ | pulver might block it |
| 01:10.46 | bkw_ | oh set callerid |
| 01:10.50 | bkw_ | to something valid? |
| 01:10.56 | Tuplink | in IAX? |
| 01:11.01 | tainted- | dialplan |
| 01:11.04 | tainted- | SetCallerID() |
| 01:11.04 | bkw_ | you use fwdiax? |
| 01:11.05 | bkw_ | eww |
| 01:11.09 | bkw_ | use SIP |
| 01:11.13 | tainted- | lol |
| 01:11.19 | Tuplink | haha i use IAX |
| 01:11.26 | Tuplink | all the other stuff works |
| 01:12.53 | *** join/#asterisk three55ml (~three55ml@cpe-24-28-65-205.austin.res.rr.com) |
| 01:14.46 | Borgon | fwd allows free ptns 1800 calls? |
| 01:15.16 | Tuplink | yup |
| 01:16.12 | *** join/#asterisk Carp1 (carp_xigon@ip-204-97-151-194.modem.logical.net) |
| 01:16.54 | timecop | opensores. |
| 01:17.28 | poli | Borgon, yes, it does. I real help if you are outside the US and just can't use the 1-800 numbers! |
| 01:17.57 | AgiNamu | I dont understand why some countries can't call USA 800 numbers |
| 01:18.00 | AgiNamu | sounds retarded to me. |
| 01:18.08 | AgiNamu | why doesnt the country just charge for their portion of the call |
| 01:18.12 | fugitivo | how can i detect if the call was hang up with AGI? my script is still running and the call was ended |
| 01:18.15 | AgiNamu | Even back-asswards Guatemala does that. |
| 01:18.28 | *** join/#asterisk yxa (~void@203.118.40.42) |
| 01:18.55 | poli | AgiNamu, welcome to the business game... |
| 01:19.12 | AgiNamu | Where's the business in NOT allowing a customer to use a serice that you make money on? |
| 01:19.36 | AgiNamu | sounds like welcome to the Business Game for Special People |
| 01:20.03 | poli | AgiNamu, not everything in business is rational. Or ir could be that configuration on the equipment to allow that could be just too expensive. I know I can't call 1-800. |
| 01:20.26 | AgiNamu | yea, i know a lot of people can't. I couldn't in canada |
| 01:20.32 | AgiNamu | im just saying, it's stupid. |
| 01:20.47 | AgiNamu | and guatemala is a class-A retarded country as far as business rationale goes. |
| 01:20.52 | AgiNamu | If THEY figured it out, i dont see who can't. |
| 01:21.08 | *** join/#asterisk outtolunc (~me@adsl-69-110-56-125.dsl.pltn13.pacbell.net) |
| 01:22.10 | fugitivo | anyone can help me with agi and perl? |
| 01:22.14 | blitzrage | drumkilla: you still around? |
| 01:23.23 | tainted- | fugitivo what's up |
| 01:23.41 | tainted- | AgiNamu like special olympics |
| 01:23.42 | fugitivo | tainted-: i made a perl script, that keeps running when the call is ended |
| 01:23.50 | tainted- | mkay |
| 01:23.55 | fugitivo | tainted-: i need to detect the hungup |
| 01:24.00 | AgiNamu | tainted-, bingo |
| 01:24.32 | tainted- | <fugitivo> ... that keeps running when the call is ended = call hangup |
| 01:24.59 | fugitivo | but.. |
| 01:25.36 | tainted- | type man type |
| 01:25.36 | fugitivo | how do i detect, that the call is ended? |
| 01:26.04 | tainted- | exten => exten,h,DeadAGI(mysuperduperscript.pl) |
| 01:26.04 | Entegrity | anyone got a working vonage conf? |
| 01:26.11 | Entegrity | vonage sphone that is... |
| 01:26.14 | Moonwick | hmmm, I wonder if the SPA-841 stores separate volume settings for the handset and speakerphone |
| 01:26.14 | fugitivo | shouldn't the script be ended when the call is ended? |
| 01:26.20 | Moonwick | doesn't seem like it does |
| 01:26.24 | tainted- | Entegrity google "site:lists.digium.com vonage conf" |
| 01:26.40 | tainted- | fugitivo that will execute your script when call is ended |
| 01:27.29 | *** join/#asterisk GustavoIPA (~gustavole@200.141.74.130) |
| 01:27.56 | fugitivo | tainted-: nono, i mean, i have an extension with an agi script running, i call the extension and the script works, but, when i end the call, the script keeps running, asterisk doesn't end the script |
| 01:29.03 | *** join/#asterisk kimc (~freenode@69.241.249.181) |
| 01:29.09 | tainted- | are u using asterisk perl module |
| 01:29.19 | fugitivo | asteriskDOTbz::AGI |
| 01:29.30 | tainted- | how is the script kept running |
| 01:29.36 | tainted- | some while(1) { mess? |
| 01:30.02 | fugitivo | i have a loop, but, shouldn't it stop when the call is ended? |
| 01:30.06 | tainted- | no |
| 01:30.10 | tainted- | it shouldn't damn it |
| 01:30.25 | tainted- | are u using AGI() or DeadAGI() in dialplan |
| 01:30.26 | fugitivo | then, do i have a way to detect when the call is ended? |
| 01:30.48 | fugitivo | exten => 1234,1,agi,script.pl |
| 01:31.32 | tainted- | that loop is preventing exit |
| 01:31.51 | tainted- | i think ur logic might be fubar |
| 01:31.58 | tainted- | what are u trying to do with the pl |
| 01:32.34 | fugitivo | it's a loop that uses cepstral to read lines, the loop is to read line by line |
| 01:32.44 | *** part/#asterisk Kumbang (~ecvs@167.205.24.4) |
| 01:33.17 | tainted- | handle the entire call in pl |
| 01:33.41 | tainted- | why u loop.. for taking user input? |
| 01:33.59 | *** join/#asterisk ThE_BoOt (~erter@200.195.76.206) |
| 01:34.10 | fugitivo | i store some stuff in an array |
| 01:34.26 | ThE_BoOt | hi all |
| 01:34.38 | tainted- | well u have to change the logic of your pl |
| 01:34.51 | fugitivo | so there's no way to check if the call was ended? |
| 01:34.53 | tainted- | so it doesn't just have infinite loop |
| 01:36.08 | ThE_BoOt | I´m searching for the service cost of a Asterisk solution (about 300 extensions sip) with a integration by one E1 with a lagacy pbx? |
| 01:36.40 | tainted- | not if you're not handling the call within the pl |
| 01:38.00 | tainted- | ThE_BoOt $$$; # $ = lots |
| 01:38.12 | tainted- | what type of legacy pbx |
| 01:38.47 | tainted- | what type of servicing.. install or including ongoing maint. |
| 01:41.07 | ThE_BoOt | install and configure the asterisk with 300 extensions (IAX or SIP) and one E1 card for interconnecting it to a legacy PBX and configure the 30 ZAP channels for this integration |
| 01:41.14 | fugitivo | the agi command doesn't have a timeout value? |
| 01:41.26 | fugitivo | hmm, forget it |
| 01:41.29 | ThE_BoOt | just install, the maint will be a separated contract |
| 01:41.32 | tainted- | fugitivo u can exit the script that way |
| 01:41.52 | tainted- | fugitivo but no -- if u are looping it, ur SOL |
| 01:42.00 | *** join/#asterisk PBXtech (PBXtech@70-58-41-173.slkc.qwest.net) |
| 01:42.58 | PBXtech | i got a couple 7960's that when you put the caller on hold then resume the call it takes like 10 seconds to resume the audio path.... why |
| 01:43.04 | tzanger | damn udp socket programming's a pain |
| 01:44.52 | *** join/#asterisk _SMP_ (~SMP@pandora.burned.net) |
| 01:45.08 | *** join/#asterisk Mavvie (edwin@edwin.adsl.barnet.com.au) |
| 01:51.23 | *** join/#asterisk ariel_ (Ariel@250.sub-166-153-35.myvzw.com) |
| 01:51.30 | ariel_ | Hello everyone |
| 01:53.11 | Moonwick | heh. anyone happen to know offhand where the spa-841 derives the time and timezone from? |
| 01:54.08 | *** join/#asterisk Nuxi (~nuxi@lin-dsl-static-206-222-194-115.inetnebr.com) |
| 01:59.45 | ariel_ | Moonwick, there is a setting on the unit for where your time server is at. |
| 01:59.56 | fugitivo | well, tainted is gone |
| 02:00.01 | fugitivo | but, for my problem |
| 02:00.21 | fugitivo | the solution is checking the channel status, if it's different than 6, then exit the script |
| 02:02.09 | Moonwick | ariel_: yeah, but if it's using NTP, it still needs to know the local timezone... I don't see a place to enter that in the admin site |
| 02:02.12 | *** join/#asterisk tainted- (~ta_i_nted@65-60-70-243-cust.telepacific.net) |
| 02:02.20 | *** join/#asterisk rjburkh (~rjburkh@dialup-4.233.113.47.Dial1.LosAngeles1.Level3.net) |
| 02:02.21 | Moonwick | although, I just realized I'm still running 0.9.1 |
| 02:02.28 | Moonwick | latest firmware is 3.x.x. heh. |
| 02:02.48 | fugitivo | tainted- |
| 02:03.06 | fugitivo | tainted-: checking the channel status, if it's != 6, then exit the script |
| 02:03.12 | ariel_ | Moonwick, There is a setting to tell it your time zone. I don't have one handy here but you should look for it. |
| 02:03.32 | rvhi | i have a T1 T100P card, can it support T1 signalling instead of PRI? |
| 02:03.40 | Moonwick | ariel_: I'm going to upgrade the FW and look again... it's quite possible it's just missing from this release |
| 02:03.43 | Moonwick | thanks tho |
| 02:03.47 | fugitivo | tainted-: if ($AGI->channel_status != 6) { exit; } |
| 02:03.54 | PBXtech | i got a couple 7960's that when you put the caller on hold then resume the call it takes like 10 seconds to resume the audio path.... why |
| 02:03.59 | zno | Moonwick: I set up a time server with dhcp and it worked right off the bat |
| 02:04.33 | ariel_ | PBXtech, do you have canreinvite=yes on the sip.conf sections for those phones? |
| 02:04.54 | PBXtech | doubt it |
| 02:04.56 | PBXtech | that my prob |
| 02:05.10 | *** join/#asterisk Entegrity (~Entegrity@c-65-96-119-254.hsd1.ma.comcast.net) |
| 02:05.43 | PBXtech | yea its no |
| 02:05.51 | *** join/#asterisk lyoungz (~lyoung@ool-182d73f5.dyn.optonline.net) |
| 02:06.59 | Entegrity | for some reason I see nothing when I debug my sip peer |
| 02:07.14 | Entegrity | I redid my sip conf to see if I could get my vonage sphone working.. |
| 02:07.25 | Entegrity | now, its status is unmonitored |
| 02:07.25 | PyroSteve | for those of you who dont know ... sipura is being bought by Linksys ! |
| 02:07.28 | Entegrity | and has no debug info.. |
| 02:08.01 | PyroSteve | opps.. its already on the wiki |
| 02:08.10 | ariel_ | Entegrity, put qualify=yes and you will have the monitor back. |
| 02:08.32 | ariel_ | PyroSteve, where did you see the sipura is being bought by linksys? |
| 02:08.38 | PyroSteve | the wiki |
| 02:08.46 | PyroSteve | Cisco aquired Sipura |
| 02:08.51 | PyroSteve | 68 Million |
| 02:08.55 | PyroSteve | by the end of July |
| 02:09.53 | onlyI | linksys is Cisco |
| 02:09.58 | PyroSteve | I love Sipura |
| 02:10.05 | ariel_ | PyroSteve, strange that this is happening. Sipura guys made the original ATA for cisco then sold it to them. Made the new co. Sipura. Now Sipura has been making the Linksys ata and now round robin back to cisco. |
| 02:10.07 | techie | http://newsroom.cisco.com/dlls/2005/corp_042605.html?DCMP=BAC-TS01 |
| 02:11.54 | ariel_ | Well there goes a good company down the drain. |
| 02:13.25 | Zaw | damn. |
| 02:13.29 | Sedorox | grrrr |
| 02:13.47 | *** join/#asterisk lyoungz_ (~lyoung@ool-182d73f5.dyn.optonline.net) |
| 02:17.26 | *** join/#asterisk lyoungz (~lyoung@ool-182d73f5.dyn.optonline.net) |
| 02:18.08 | lyoungz | I'm having trouble getting a FXS extension going. |
| 02:18.18 | lyoungz | <PROTECTED> |
| 02:18.34 | lyoungz | <PROTECTED> |
| 02:18.42 | lyoungz | It seems to hang up immediately |
| 02:19.18 | lyoungz | I configured it on a context of [from-internal], through the AMPORTAL |
| 02:19.24 | lyoungz | the AMP interface |
| 02:20.06 | lyoungz | as I read the AMP extensions.conf, that should allow the extension to be |
| 02:20.15 | lyoungz | useable. |
| 02:20.49 | *** join/#asterisk PBXtech (PBXtech@70-58-41-173.slkc.qwest.net) |
| 02:21.18 | lyoungz | hmm, awfully quiet around here. |
| 02:21.50 | lyoungz | anyone home? |
| 02:21.59 | onlyI | noe |
| 02:22.07 | lyoungz | dit dit dit ... dah dah dah ... dit dit dit |
| 02:22.13 | onlyI | heheheheh |
| 02:22.21 | onlyI | we are sitting at the bar |
| 02:22.42 | lyoungz | perhaps I should pour myself a glass of wine as well |
| 02:23.07 | lyoungz | configuration is much less frustrating with a glass of wine. |
| 02:23.25 | blitzrage | lol |
| 02:24.07 | lyoungz | ok, I've got a nice glass of chilean Merlot, 2001. |
| 02:24.10 | lyoungz | yum |
| 02:24.35 | fugitivo | you should try argentinian wines |
| 02:24.36 | Veto | or a few vodka tonics...then you just wonder if the headache in the morning if from * or the vodka. |
| 02:24.55 | lyoungz | I prefer a good bourbon |
| 02:25.05 | lyoungz | Knob Creek is my current favorite. |
| 02:25.16 | fugitivo | whisky is for asterisk nights |
| 02:25.21 | lyoungz | there you go. |
| 02:25.51 | lyoungz | unfortunately I have an early morning meeting, so I have to stick with wine. |
| 02:25.54 | fugitivo | and programming nights |
| 02:26.05 | *** join/#asterisk klasstek (~nunyobiz@c-24-9-148-246.hsd1.co.comcast.net) |
| 02:26.10 | fugitivo | and tv nights |
| 02:26.47 | *** join/#asterisk jf_ (~jeanfranc@modemcable077.187-80-70.mc.videotron.ca) |
| 02:27.41 | jf_ | I'd like to program a voip soft phone base on iax2, is there any java api or iax specification to use that protocol |
| 02:27.57 | klasstek | chan_iax.c |
| 02:28.05 | jf_ | what is that |
| 02:28.15 | jf_ | the source file |
| 02:28.24 | fugitivo | jf_: make it nice to the eyes please |
| 02:28.27 | dizzan | anyone around familiar with nufone? |
| 02:28.36 | Qwell | dizzan: yep, whats up? |
| 02:28.46 | jf_ | fugitivo: what do you mean |
| 02:29.12 | fugitivo | jf_: user friendly and with eye candy |
| 02:29.32 | jf_ | lol, i will but i need to write the low level first |
| 02:29.34 | dizzan | Qwell: hi, i havn't used my account for a while and im setting up a new * box and i totally forget what the proper basic config is for iax |
| 02:29.46 | jf_ | im looking for api or info to implement the protocol |
| 02:30.15 | klasstek | Use the source Luke |
| 02:30.19 | Qwell | dizzan: I just so happened to send somebody an example config the other day... Whats your email? |
| 02:30.19 | dizzan | i have a user and peer set up but im getting "Rejected connect attempt from 198.22.67.70" when i make a call in to my DID |
| 02:30.29 | *** join/#asterisk riquisim0 (~riquisimo@205.240.200.120) |
| 02:30.31 | file | your user isn't correct |
| 02:30.37 | dizzan | Qwell: dizzan@gmail.com |
| 02:30.40 | fugitivo | Come back to the source neo |
| 02:30.41 | riquisim0 | hello |
| 02:30.46 | Qwell | dizzan: one sec, I'll forward it |
| 02:30.51 | jf_ | Klasstek: i will |
| 02:31.03 | riquisim0 | what codec does Asterisk work with, I read G.729 codec is sold |
| 02:31.10 | lyoungz | <PROTECTED> |
| 02:31.18 | fugitivo | riquisim0: a lot |
| 02:31.19 | file | riquisim0: asterisk.org has all the codecs |
| 02:31.25 | klasstek | riquisim0, show translations |
| 02:31.29 | lyoungz | my config is hanging up on me as soon as I connect. |
| 02:31.40 | file | on the 'Features' page |
| 02:31.45 | lyoungz | how can I trace why that is happening? |
| 02:31.56 | klasstek | lyoungz, NoOp is useful |
| 02:32.06 | jf_ | Who suggest sip instead of iax |
| 02:32.12 | riquisim0 | how many concurrent calls does asterisk support? |
| 02:32.28 | riquisim0 | depends on hardware? |
| 02:32.30 | jf_ | riquisim0: depend on ur cpu |
| 02:32.35 | lyoungz | klasstek: I guess you mean I can insert Noops with unique trace labels into my config? |
| 02:32.39 | file | and codecs, and bandwidth |
| 02:32.44 | file | and protocol |
| 02:32.50 | file | and number of file descriptors allowed |
| 02:32.52 | klasstek | riquisim0, tough to answer. depends on config, hardware and what the box is configured to do |
| 02:32.52 | lyoungz | klasstek: I'll give it a shot. |
| 02:32.54 | file | need I go on? |
| 02:32.55 | *** join/#asterisk stilex (~wow@pc-24-151-108-034.newm2.ct.charter.com) |
| 02:33.04 | riquisim0 | ntel Xeon 2.8GHz Processor, 1MB Cache and 1GB DDR2 RAM |
| 02:33.19 | klasstek | riquisim0, still to many variables. |
| 02:33.31 | jf_ | riquisim0: a shit load |
| 02:33.42 | riquisim0 | of all codecs listed at asterisk.org, which do you guys recommend? |
| 02:33.53 | file | using G729 with transcoding you wouldn't get that many jf |
| 02:33.58 | file | riquisim0: it depends... |
| 02:34.12 | riquisim0 | file: depends on .. ? |
| 02:34.13 | Qwell | dizzan: on its way |
| 02:34.18 | jf_ | yeah that'S true but with g711 should be good |
| 02:34.18 | stilex | could someone point me in the direction of figuring out a problem, I have a sip cisco phone 7940 and I use an upstream provider - outgoing calls work perfect but incoming calls to the sip device are choppy as hell - I've checked the codecs and everything is the same.... |
| 02:34.26 | klasstek | I like g726. easy on cpu and saves a little bandwidth without affecting quality |
| 02:34.35 | file | riquisim0: well do you want compressed, what are you looking for - LAN, over net, what |
| 02:34.52 | riquisim0 | over net |
| 02:35.23 | daork | riquisim0: also depends on how fast your internet connection is, and what the latency between endpoints is |
| 02:35.26 | daork | and jitter probably |
| 02:35.30 | jf_ | does g726 require $$, is it under patent |
| 02:35.31 | file | see how many variables there are? :) |
| 02:35.39 | file | jf_: G726-32 isn't, that's why we have it |
| 02:35.39 | onlyI | ashit load |
| 02:35.41 | onlyI | :) |
| 02:35.50 | JerJer | no g.726 patents have never been enforced |
| 02:36.01 | klasstek | sup JerJer ! |
| 02:36.08 | jf_ | where can i find it |
| 02:36.19 | riquisim0 | file: yep |
| 02:36.20 | file | there's other reasons we don't have all the G726 bitrates... but bleh |
| 02:36.33 | jf_ | g711 vs g726, which is better bandwidth saving vs cpu |
| 02:36.54 | daork | well, theory is that g.711 doesnt use much cpu at all |
| 02:36.54 | klasstek | g726 ~= g711/2 |
| 02:36.59 | file | g726-32 is half the bandwidth of g711, with quality being around the same |
| 02:37.03 | klasstek | for bandwidth |
| 02:37.16 | JerJer | klasstek: SSDD |
| 02:37.30 | jf_ | daork: yeah but use bandwithd instead |
| 02:37.45 | jf_ | nice |
| 02:38.09 | jf_ | how do i implements new codec in * |
| 02:38.11 | file | yada yada yada silly news |
| 02:38.30 | file | jf_: you read existing code, see how it works, read, and write it :p |
| 02:38.37 | file | did I mention reading? |
| 02:39.08 | klasstek | oh yea and reading :) |
| 02:39.14 | daork | what codec do you want to implement? |
| 02:39.16 | file | exactly! |
| 02:39.27 | jf_ | no this is not what i've met, i mean i download a codec, i want to install it in *, how |
| 02:39.32 | jf_ | g726 |
| 02:39.49 | file | g726-32 is already implemented |
| 02:40.02 | file | it's there, use allow=g726 and voila... g726 |
| 02:40.02 | dizzan | Qwell thanks, got it |
| 02:40.19 | file | there's just 4 bitrates for G726: 16, 24, 32, and 40 |
| 02:40.28 | file | asterisk doesn't support 16, 24, and 40 |
| 02:41.02 | jf_ | but if it wasnt, how can i accomplish this task |
| 02:41.29 | file | there's no guide |
| 02:41.44 | file | you have to read existing code, learn how it works, learn how to interface your own |
| 02:41.47 | jf_ | shit :P |
| 02:41.53 | *** join/#asterisk tessier (~treed@222.253.79.234) |
| 02:42.02 | jf_ | anyway, gotta go sleep, good night |
| 02:42.16 | *** join/#asterisk iceyp (~icepick@firewall.unix.co.nz) |
| 02:42.49 | iceyp | does anyone have a simular prepaid system to nufone and voipjet? |
| 02:43.11 | file | iceyp: yes - but can you have it, no |
| 02:43.29 | *** join/#asterisk Silik0n (~krice@newrso.suspicious.org) |
| 02:43.36 | iceyp | file are you selling it? |
| 02:43.39 | file | no |
| 02:43.42 | iceyp | ok. |
| 02:43.52 | iceyp | do you where I can purchase one? |
| 02:43.58 | file | try voip-info.org |
| 02:44.52 | riquisim0 | is the testmyvoip.com page for real? |
| 02:45.07 | file | sure why not |
| 02:45.43 | riquisim0 | cool |
| 02:47.16 | cypromis | w 17 |
| 02:47.37 | *** join/#asterisk bkw_ (~brian@bkw.developer.and.friend.of.asterisk) |
| 02:47.37 | *** mode/#asterisk [+o bkw_] by ChanServ |
| 02:47.39 | bkw_ | ya know what |
| 02:47.45 | bkw_ | READ PEOPLE READ... damn |
| 02:47.51 | bkw_ | awoierfjaoweifj |
| 02:48.07 | bkw_ | ok </rant> |
| 02:48.15 | denon | bkw_: good luck .. that problem is as old as software itself |
| 02:48.18 | Silik0n | reading is so overrated |
| 02:48.28 | denon | if its not extremely intuitive, you'll have morons on irc begging for a handout |
| 02:48.30 | JerJer | where can i top post? |
| 02:48.35 | Silik0n | we can just ask questions on IRC |
| 02:48.38 | bkw_ | well shit someone posts to the dev list.. sounds like a request instead of actually READING the sample configs |
| 02:48.44 | Silik0n | JerJer wherever you want to |
| 02:48.51 | bkw_ | JerJer, EVERY SINGLE TIME |
| 02:48.52 | tzanger | 22:50 <@bkw_> awoierfjaoweifj |
| 02:48.54 | tzanger | I can't read that |
| 02:48.58 | bkw_ | I do just to piss that guy off that bitches all the time about it |
| 02:49.01 | bkw_ | and I do it on purpose |
| 02:49.10 | tzanger | bkw_: hahahaha I think that's me |
| 02:49.19 | bkw_ | are you steve? |
| 02:49.22 | tzanger | no |
| 02:49.29 | tzanger | I'm steve's listcop buddy |
| 02:49.39 | denon | you know, I cant figure out people's fetish with where you post .. |
| 02:49.49 | file | I feel the need to strangle people |
| 02:49.50 | tzanger | actually top posting doesn't bother me nearly as much as people who just don't fucking trim |
| 02:50.03 | bkw_ | Oh I trim |
| 02:50.05 | tzanger | denon: because MOST of the time it destroys the conversation |
| 02:50.11 | tzanger | bkw_: I didn't need to know that |
| 02:50.15 | bkw_ | I even comment inline |
| 02:50.18 | denon | yeah, I like trimming .. though sometimes people trim way too much and stuff is way out of context |
| 02:50.24 | tzanger | denon: agreed |
| 02:50.27 | bkw_ | denon, yep |
| 02:50.30 | tzanger | there's a happy middle |
| 02:50.32 | denon | nod |
| 02:50.34 | bkw_ | yep |
| 02:50.38 | bkw_ | unsubscribe from the list |
| 02:50.42 | bkw_ | thats the only happy middle |
| 02:50.44 | onlyI | heh |
| 02:50.53 | bkw_ | can anyone resolve www.bkw.org yet? |
| 02:50.54 | tzanger | top posting to say "ok great I got it" is fine... top posting to try and carry on a conversation is horrible |
| 02:51.08 | denon | bkw: nope, not here |
| 02:51.11 | tzanger | $ host www.bkw.org |
| 02:51.11 | tzanger | Host www.bkw.org not found: 3(NXDOMAIN) |
| 02:51.12 | onlyI | not here |
| 02:51.19 | bkw_ | i'm very pisssssssssssssed at zone edit |
| 02:51.19 | file | people annoy me oh so much... argh |
| 02:51.28 | bkw_ | did they have fucking 4 week ttls |
| 02:51.36 | onlyI | ahehahehah |
| 02:51.49 | tzanger | bkw_: that sucks ass, I always have my TTLs to like 1hr |
| 02:52.07 | PBXtech | i got a couple 7960's that when you put the caller on hold then resume the call it takes like 10 seconds to resume the audio path.... why? |
| 02:52.18 | file | because I hate people, that's why |
| 02:52.22 | file | and I've singled you out! |
| 02:52.37 | onlyI | tzanger any news on digi HW |
| 02:52.51 | *** join/#asterisk TheEmperor (~user@203.114.48.47) |
| 02:52.54 | denon | PBXtech: did you take my advice? |
| 02:53.03 | PBXtech | yup latest firmware |
| 02:53.06 | tzanger | onlyI: not yet, but I was not at work today at all |
| 02:53.16 | denon | PBXtech: and factory reset? |
| 02:53.24 | file | ACHOO |
| 02:53.29 | PBXtech | no |
| 02:53.30 | bkw_ | can ya lookup ns.bkw.org |
| 02:53.31 | onlyI | tzanger roger well keep me in the loop |
| 02:53.40 | denon | 24.249.7.48 |
| 02:54.06 | PBXtech | why factory reset? |
| 02:54.07 | bkw_ | see |
| 02:54.13 | bkw_ | can you do a lookup against that please? |
| 02:54.23 | denon | reverse? |
| 02:54.25 | onlyI | ;; ANSWER SECTION: |
| 02:54.25 | onlyI | ns.bkw.org. 75890 IN A 24.249.7.48 |
| 02:54.26 | bkw_ | no forward |
| 02:54.33 | denon | I did .. that was the IP |
| 02:54.43 | bkw_ | 65.38.28.146 was the IP |
| 02:54.43 | bkw_ | that IP is right |
| 02:54.44 | denon | 24.249.7.48 = h48.7.249.24.cable.mcls.cablerocket.net |
| 02:54.50 | bkw_ | but it seems all inbound DNS lookups are failing |
| 02:54.50 | denon | ns.bkw.org = 24.249.7.48 |
| 02:54.58 | Qwell | ;; ANSWER SECTION: |
| 02:54.58 | Qwell | bkw.org. 86400 IN A 127.0.0.1 |
| 02:55.04 | Qwell | dig @ip bkw.org |
| 02:55.04 | bkw_ | can you do a query for www.bkw.org against ns.bkw.org |
| 02:55.08 | denon | ah |
| 02:55.14 | bkw_ | see if it returns |
| 02:55.19 | Qwell | bkw_: ^^ |
| 02:55.21 | denon | Server: ns.bkw.org |
| 02:55.22 | denon | Address: 24.249.7.48 |
| 02:55.22 | denon | Name: www.bkw.org |
| 02:55.22 | denon | Address: 24.249.7.48 |
| 02:55.26 | bkw_ | dirty bastards |
| 02:55.33 | onlyI | use dig |
| 02:55.41 | denon | doesnt matter how I do it, it worked. |
| 02:55.47 | Silik0n | anyone compiled head on fbsd lately? |
| 02:55.50 | denon | do it yourself if you have a tool preference. :) |
| 02:55.52 | Qwell | bkw_: dig @24.249.7.48 bkw.orgm shows 127.0.0.1 |
| 02:56.09 | Qwell | minus an m or two |
| 02:56.27 | file | he wanted www.bkw.org not bkw.org |
| 02:56.30 | Silik0n | denon I am asshole I was wondering if anyone else has |
| 02:56.42 | Qwell | ahh, then it returns the IP |
| 02:56.44 | Silik0n | its erroring |
| 02:57.03 | bkw_ | hrm |
| 02:57.15 | denon | it returned fine for me |
| 02:57.16 | Silik0n | hopefully it was just something tupid |
| 02:57.21 | klasstek | ; <<>> DiG 9.2.3 <<>> @ns.bkw.org www.bkw.org |
| 02:57.21 | klasstek | [...] |
| 02:57.21 | klasstek | ;; ANSWER SECTION: |
| 02:57.21 | klasstek | www.bkw.org. 86400 IN A 24.249.7.48 |
| 02:57.28 | bkw_ | ok |
| 02:57.47 | Qwell | bkw.org isn't allowed to return localhost :p Fix that! :P |
| 02:57.52 | klasstek | kothing without the www though |
| 02:57.59 | file | lots of pr0n available @ bkw.org!!! |
| 02:58.03 | bkw_ | ok thats fixed now |
| 02:58.05 | Qwell | file: you read my mind |
| 02:58.07 | tzanger | hahahahaha |
| 02:58.09 | tzanger | Yanno, mentioning that the guts were propelled 3.2 feet into the air sounds kinda like someone had nothing better to do than to measure the size of the splats with more precision than most people would consider sane. Now, I've had my suspicions about the Germans, but this is worse than anything I've previously read. |
| 02:58.15 | file | Qwell: what, got porn on the mind? |
| 02:58.18 | PBXtech | denon: why factory reset? |
| 02:58.24 | denon | tzanger: the frog thing? |
| 02:58.35 | tzanger | denon: yeah |
| 02:58.41 | denon | PBXtech: because 7960s very rarely do weird stuff like that .. and its worth getting it back to its original state |
| 02:58.50 | tzanger | me being of German descent I have to agree with the comment |
| 02:59.14 | PBXtech | ok |
| 02:59.21 | bkw_ | they mess with me i'll make my host records be www.bkw.org and bkw.org |
| 02:59.21 | tzanger | ahahha |
| 02:59.23 | tzanger | A dyslexic man walks into a bra. |
| 02:59.28 | bkw_ | and then my DNS will be served from the root servers |
| 02:59.56 | denon | bkw: Im pretty sure most registrars block or frown on that now |
| 02:59.57 | klasstek | how about 7960s that only lock up or reboot when connected to a POE switch. |
| 02:59.58 | mishehu | do you believe in dog? |
| 03:00.00 | denon | that used to be a fun trick though |
| 03:00.08 | *** join/#asterisk qwerp (~abc@219.95.105.74) |
| 03:00.12 | denon | klasstek: probably the wrong POE standard |
| 03:00.13 | qwerp | harlo.. |
| 03:00.13 | Juggie | 7960 is a great phone |
| 03:00.18 | tzanger | bkw_: what are you trying to do? |
| 03:00.19 | Juggie | except the sip firmware blows |
| 03:00.20 | Nugget | klasstek: the old 7960s use a different POE spec, before it was ratified. |
| 03:00.22 | klasstek | denon, cisco switches. |
| 03:00.26 | file | bkw is touching something in a very bad way |
| 03:00.29 | Nugget | there are two different cable types to use for new and old 79x0s |
| 03:00.31 | qwerp | i have a weird problem.. |
| 03:00.42 | bkw_ | tzanger, figure out why the rest of the world can't resolve my domain name |
| 03:00.47 | PBXtech | stupid router doesnt have tftp settings in it, factory reset is a pain :) |
| 03:00.51 | Qwell | bkw_: Did you set your domain to use ns.bkw.org? :P |
| 03:00.52 | tzanger | bkw_: because it hasn't propagated to the root servers yet? |
| 03:00.54 | klasstek | even done wall warts for those phones too. |
| 03:00.54 | Qwell | whois isn't showing it |
| 03:00.55 | PBXtech | in dhcp |
| 03:01.05 | Qwell | nevermind, it is, heh |
| 03:01.08 | bkw_ | tzanger, maybe another 24-48 hours |
| 03:01.11 | bkw_ | its bee 72+ now |
| 03:01.13 | Ahewes | Nugget: I thought all Cisco phones used proprietary Cisco PoE spec. |
| 03:01.16 | Qwell | 72 is plenty |
| 03:01.16 | onlyI | way take up to 72 hours bkw |
| 03:01.18 | Juggie | bkw, because you need two name servers |
| 03:01.20 | Juggie | thats the minimum |
| 03:01.26 | Nugget | I understand the new ones (since the spec was ratified) use the standard. |
| 03:01.31 | bkw_ | no thats a rule |
| 03:01.32 | Nugget | but perhaps I've been misinformed |
| 03:01.37 | tzanger | bkw_: wowza |
| 03:01.37 | fugitivo | i set the language of my asterisk to es, so the sound files are played from the asterisk/sounds/es directory, buy using the transfer from asternic, plays the files from asterisk/sounds, any idea? |
| 03:01.40 | bkw_ | not a requirement |
| 03:01.46 | bkw_ | dirty bastards |
| 03:01.49 | Nugget | the old ones used a cisco standard because no official standard existed at the time |
| 03:01.50 | onlyI | ;) |
| 03:01.59 | bkw_ | it let me put it in and its in the thing like that |
| 03:02.02 | Ahewes | Nugget: I'll ask google. |
| 03:02.09 | bkw_ | I have flown with a single nameserver before |
| 03:02.12 | klasstek | at any rate... we use crisco phones 40s and 60s but the only real problem phones are using POW. |
| 03:02.15 | Juggie | bkw, your whos also shows no ip for your name server |
| 03:02.39 | Juggie | and since you are hosting your own nameserver, and well theres no way for ns.bkw.org to resolve without first doing u know, its kind of a loop |
| 03:02.49 | klasstek | Name Server:NS.BKW.ORG |
| 03:02.49 | klasstek | Name Server: |
| 03:02.52 | bkw_ | bullshit |
| 03:02.55 | bkw_ | ns.bkw.org resolves |
| 03:02.59 | bkw_ | PIR fucked up .org |
| 03:03.03 | bkw_ | we are about to fix this |
| 03:03.06 | bkw_ | and fix it up good |
| 03:03.32 | Silik0n | yep socket.h is either f'd or needs another define |
| 03:03.37 | tzanger | whoa that *is* weird |
| 03:03.46 | tzanger | ns.bkw.org resolves |
| 03:03.51 | bkw_ | TOLD YA |
| 03:03.53 | tzanger | and ns.bkw.org will resolve www.bkw.org |
| 03:03.58 | tzanger | but www.bkw.org doesn't resolve |
| 03:03.58 | bkw_ | see |
| 03:04.00 | *** join/#asterisk TheEmperor (~user@203.114.48.47) |
| 03:04.07 | bkw_ | I told ya something is fucked up |
| 03:04.28 | Ahewes | Nugget: Cisco IP Phones 7971G-GE and 7970G support IEEE 802.3af PoE |
| 03:04.36 | tzanger | bkw_: wait |
| 03:04.39 | tzanger | your DNS server is fucked up |
| 03:04.39 | *** join/#asterisk qwerp (~abc@219.93.57.58) |
| 03:04.44 | Qwell | doesn't even resolve @4.2.2.2 |
| 03:04.45 | Ahewes | From http://www.cisco.com/en/US/products/hw/phones/ps379/ |
| 03:04.56 | Ahewes | So the newer ones are geting 802.3af support |
| 03:05.10 | bkw_ | tzanger, what? |
| 03:05.22 | tzanger | http://pastebin.ca/10396 |
| 03:05.26 | tzanger | that is not a normal response |
| 03:05.27 | PBXtech | can * do callerID screen pops? anyone got an app for that? |
| 03:05.27 | onlyI | ERROR: The parent servers say that the domain bkw.org does not exist |
| 03:05.35 | tzanger | ferinstance |
| 03:05.44 | Qwell | ultradns has auth |
| 03:05.45 | bkw_ | DIRTY BASTARDS GOD DAMN IT |
| 03:05.46 | denon | PBXtech: look at FOP |
| 03:06.42 | PBXtech | yea thats a good system overview. just wanting more of an individual basis |
| 03:06.42 | bkw_ | tzanger, I don't have reverse |
| 03:06.54 | mishehu | as the old saying goes, "if there's time enough to do it, there's time enough to do it half-assed" |
| 03:06.57 | tzanger | you don't need reverse |
| 03:07.01 | klasstek | bkw_, nada at the .org tld name servers for your www |
| 03:07.19 | *** join/#asterisk wolfson (~hehe@bcp-68-187-183-234.man.nc.charter.com) |
| 03:07.20 | bkw_ | see PIR has this fucked up |
| 03:07.27 | Qwell | PIR? |
| 03:07.28 | Nugget | Ahewes: cool, thanks for looking that up. |
| 03:07.50 | bkw_ | the dumb ass company over .org now |
| 03:07.54 | Qwell | ahh |
| 03:08.08 | bkw_ | they totally fucked up my domain |
| 03:08.12 | bkw_ | in the past |
| 03:08.16 | Qwell | They haven't released auth, it looks like |
| 03:08.33 | onlyI | bkw whos's own ip 24.249.7.48 |
| 03:08.35 | Qwell | never seen that for forward |
| 03:08.48 | Qwell | onlyI: That only matters for reverse |
| 03:08.54 | bkw_ | onlyI, its my cable modem here |
| 03:08.58 | klasstek | bkw_, whois shows Status:INACTIVE |
| 03:09.04 | bkw_ | WTF |
| 03:09.07 | bkw_ | those dirty bastards |
| 03:09.17 | klasstek | exp in 2014 |
| 03:09.40 | bkw_ | ya it don't expire for SOME time |
| 03:09.49 | Juggie | yah inactive |
| 03:09.51 | Juggie | thats your prob |
| 03:09.52 | stilex | could someone point me in the direction of figuring out a problem, I have a sip cisco phone 7940 and I use an upstream provider - outgoing calls work perfect but incoming calls to the sip device are choppy as hell - I've checked the codecs and everything is the same.... |
| 03:09.54 | Nugget | my personal domain turned ten today. |
| 03:10.08 | Nugget | http://nugget.livejournal.com/91132.html |
| 03:10.12 | klasstek | one issue though is that your contact email is at the same domain and you can't lookup the mx record to deliver mail. |
| 03:10.22 | Qwell | klasstek: funny |
| 03:10.26 | Qwell | ...sorta |
| 03:10.40 | PBXtech | whats a good windowsXP dhcp/tftp server? |
| 03:10.52 | klasstek | paper weight |
| 03:11.05 | denon | PBXtech: solarwinds tftp |
| 03:11.08 | bkw_ | klasstek, how did you check the domain status? |
| 03:11.12 | denon | nfi on dhcp, I think ISC's is ported |
| 03:11.22 | Qwell | bkw_: whois |
| 03:11.36 | klasstek | whois bkw.org on Linux g2vaio 2.6.9-gentoo-r13 #1 Sat Jan 15 14:23:35 MST 2005 i686 Intel(R) Pentium(R) M processor 1.70GHz GenuineIntel GNU/Linux |
| 03:11.41 | PBXtech | thx |
| 03:12.05 | bkw_ | I don't see that in the whois records |
| 03:12.47 | Qwell | bkw_: I'm talking to a guy, lemme see if I can't get you an answer |
| 03:12.57 | stilex | can someone point me in the right direction google doesnt have anything that will correct the issue |
| 03:13.01 | stilex | nor do the docs |
| 03:13.21 | bkw_ | stilex, what is the problem |
| 03:14.06 | Qwell | Status:OK |
| 03:14.11 | stilex | im having a problem with incoming audio only not outgoing - im using a cisco 7940 phone with nufone as the upstream provider |
| 03:14.25 | stilex | outgoing calls sound fine so I really dont understand it |
| 03:14.30 | file | *YAWN* |
| 03:14.35 | file | why must I be so tired |
| 03:14.36 | stilex | im using ser as a proxy for the sip device |
| 03:14.39 | Qwell | bkw_: You're all set... |
| 03:14.56 | Qwell | bkw_: http://pastebin.ca/10398 |
| 03:14.59 | stilex | iax2 show channels and sip show channels reflect the same codec and the using the same codec from nufone |
| 03:15.06 | iq | Hi, anyone using SERWEB... can you check how many Tabs do you get on main page? |
| 03:16.04 | Qwell | bkw_: Its using a different NS now too |
| 03:17.21 | onlyI | parent |
| 03:18.35 | onlyI | ultra should glue IP to is NS |
| 03:18.46 | bkw_ | Ok godaddy can GO TO HELL |
| 03:18.46 | bkw_ | totally |
| 03:18.59 | Qwell | powerpipe or enom |
| 03:20.28 | *** join/#asterisk iceyp (~icepick@202.150.105.150) |
| 03:20.48 | iceyp | i've just done a brand new install of asterisk and when running it i get the following error: |
| 03:20.48 | iceyp | [ Booting...Apr 27 15:19:55 WARNING[5670]: loader.c:305 __load_resource: /usr/local/lib/asterisk/modules/chan_modem.so: Undefined symbol "ast_pthread_create" |
| 03:20.48 | iceyp | Apr 27 15:19:55 WARNING[5670]: loader.c:463 load_modules: Loading module chan_modem.so failed! |
| 03:21.05 | iceyp | any ideas? |
| 03:21.29 | xai | Is there a rule of thumb for pricing proprietary phone systems and or * systems? $500/phone for a proprietary voip system for example? |
| 03:22.13 | shido6 | wow |
| 03:22.18 | shido6 | 500/phone is a lot of cash |
| 03:22.40 | xai | shido6: im talking about the entire system, pbx, phones, installation, etc. |
| 03:22.50 | iceyp | shido6 any idea what causes my above error from a new install? |
| 03:23.25 | iq | iceyp, can you help with serweb :) ? |
| 03:23.44 | Ahewes | xai: it depends what the system is for. for a small office, an epygi or an allworx is probably a fraction of that. for a call center, the sky's the limit. just depends on the features you need. |
| 03:24.01 | iceyp | iq not right now, sorry |
| 03:24.17 | *** join/#asterisk jskcr|lappy (~jskcr@jskcr.user) |
| 03:24.49 | xai | Ahewes: normally for 100 person office with standard voice messaging, forwarding, features.. I thought cisco and nortel were going to be in that range. |
| 03:25.10 | bkw_ | FEEEEEEEEAR ME |
| 03:25.13 | bkw_ | TAKE THAT PIR!!! |
| 03:25.17 | bkw_ | NEXT!!! |
| 03:25.19 | bkw_ | moving along now |
| 03:25.27 | file | silly bkw |
| 03:25.29 | bkw_ | where was I before morons invaded my evening |
| 03:25.31 | xai | ahewes: and perhaps some programming ability. |
| 03:25.37 | file | bkw_: dealing with idiots |
| 03:25.37 | Silik0n | bkw_ so fix the bsd problems since they just broke it all to hell and back |
| 03:25.58 | bkw_ | fuck BSD |
| 03:26.02 | bkw_ | run a supported platform |
| 03:26.04 | bkw_ | :P |
| 03:26.43 | blitzrage | preach it brotha |
| 03:26.49 | xai | Ahewes: I really hate getting quotes from vendors just to see if * will be X times cheaper than cisco, nortel, meridian, etc... they don't seem to be too forthcoming about their prices either. |
| 03:27.06 | iceyp | bkw_ who you talking to about bsd? |
| 03:27.24 | iceyp | ohh right |
| 03:27.27 | bkw_ | I don't mind BSD at all.. I love it |
| 03:27.29 | iceyp | Silik0n you also run bsd? |
| 03:27.37 | bkw_ | but if you wanna use asterisk.... you must run linux to have full support |
| 03:27.38 | bkw_ | I hate that |
| 03:27.40 | bkw_ | but its the truth |
| 03:27.44 | bkw_ | I hate linux |
| 03:27.45 | Silik0n | patch |
| 03:27.46 | bkw_ | HATE |
| 03:27.48 | iceyp | same |
| 03:27.50 | bkw_ | with a big fucking H |
| 03:27.55 | bkw_ | but since we have to live with it |
| 03:27.59 | bkw_ | I made a BSD like linux distro |
| 03:28.01 | bkw_ | muhahahahah |
| 03:28.04 | bkw_ | the feel of BSD |
| 03:28.07 | bkw_ | with the brains of linux |
| 03:28.11 | iceyp | bkw_ fix bsd support :) |
| 03:28.16 | Silik0n | damnit |
| 03:28.17 | bkw_ | no |
| 03:28.21 | bkw_ | pay me a million bucks |
| 03:28.22 | bkw_ | and we will |
| 03:28.24 | Silik0n | acl.c busted |
| 03:28.37 | blitzrage | I have a theory that people will hate whatever is popular :) |
| 03:28.39 | bkw_ | blitzrage, na linux is cool an dall |
| 03:28.40 | file | why do I have a feeling I, as the new guy, would do it? |
| 03:28.42 | bkw_ | but OS X is better |
| 03:28.51 | file | bkw_: and don't you forget to push 'da build environment to the CVS server |
| 03:28.53 | bkw_ | file na |
| 03:29.02 | bkw_ | file let me do dat really fast |
| 03:29.10 | bkw_ | i'll have to help jew with it a bit |
| 03:29.38 | file | k |
| 03:29.43 | blitzrage | screw apple and their rediculous stores |
| 03:30.07 | bkw_ | haha |
| 03:30.10 | bkw_ | you know you want one |
| 03:30.15 | bkw_ | you'll LOVE IT |
| 03:30.15 | bkw_ | I know it |
| 03:30.27 | bkw_ | i'm gonna install a mac mini in my car |
| 03:30.30 | bkw_ | hook up GPRS |
| 03:30.31 | bkw_ | isight |
| 03:30.33 | bkw_ | GPS |
| 03:30.34 | klasstek | I like to go into the stores and fire off infinite loops and memory hogs in a background shell then watch people say this machine is slow. |
| 03:30.37 | bkw_ | and be mobile |
| 03:30.47 | Silik0n | damnit |
| 03:31.05 | Silik0n | cable is pissing me off tonight |
| 03:31.05 | bkw_ | Silik0n, you know what it takes to get my attention ;) |
| 03:31.05 | bkw_ | muahahha |
| 03:31.28 | bkw_ | my cable modem is a PCI card |
| 03:31.28 | bkw_ | I love it |
| 03:31.33 | bkw_ | shows up as a regular ethernet card |
| 03:31.34 | bkw_ | in linux |
| 03:31.39 | Silik0n | bkw_: no i will not show you my package again... i showed you once and you wouldnt even unzip it to check it out |
| 03:31.43 | Strom_C | hello everyone |
| 03:31.46 | bkw_ | haha |
| 03:31.46 | Qwell | bkw_: Do they have an sbus card available? :p |
| 03:31.53 | bkw_ | screw sparc |
| 03:31.58 | bkw_ | opteron is where its at baby |
| 03:32.08 | Silik0n | bkw_ no not the IP part... the video par is a ver shitty signal |
| 03:32.09 | Qwell | bkw_: yeah, you spend $4k for a router. :p |
| 03:32.15 | mishehu | my next server box is an opteron |
| 03:32.17 | file | bkw_: we neeeeeeed an opteron |
| 03:32.21 | Qwell | I'm quite happy with my $50 sparc router, thank you very much. ;] |
| 03:32.27 | bkw_ | file what do you have? |
| 03:32.31 | Silik0n | i'm quite happy with my 486 router |
| 03:32.42 | file | bkw_: I have a P4 3.4GHz with EM64T and Hyperthreading |
| 03:32.57 | bkw_ | is that x86 64bit like the opteron? |
| 03:32.59 | file | but an opteron would be so fun to play with |
| 03:33.01 | bkw_ | and will it run opteron code? |
| 03:33.06 | bkw_ | or are they the asme? |
| 03:33.12 | file | it'll run amd64 stuff sure |
| 03:33.26 | file | Intel used AMD's 64-bitness and called it EM64T |
| 03:34.07 | tzanger | ??echotune |
| 03:34.07 | tzanger | er |
| 03:34.07 | tzanger | echotune? |
| 03:34.07 | tzanger | echocancel? |
| 03:34.09 | tzanger | where the fuck's tha tbot |
| 03:34.26 | Qwell | ~echocancel |
| 03:34.41 | bkw_ | haha |
| 03:35.11 | tzanger | echotune? |
| 03:35.38 | tzanger | jbot's dead |
| 03:36.25 | tzanger | damn I blasted the wrong guy in -users |
| 03:36.34 | tzanger | hopefully my reply finds him a solution |
| 03:37.08 | *** join/#asterisk irv999 (~irv999@ool-457249eb.dyn.optonline.net) |
| 03:37.10 | irv999 | lo all |
| 03:37.25 | _Vile | tzang, be nice |
| 03:38.12 | irv999 | can someone recommend a 6 line SIP phone for asterisk under $150? w/ caller ID? Something descent.. I was looking at the polycom ones |
| 03:38.18 | irv999 | and sipura |
| 03:38.24 | file | bkw_: that's sexy! |
| 03:38.35 | _Vile | irv, yeah polycom |
| 03:38.35 | Silik0n | get the polycom ip600... its a few extra bucks but well worth it |
| 03:38.44 | xai | Ahewes: this pbx is for realtors that want lots of messaging services... do those systems you mentioned work for an 100 person environment like that? |
| 03:38.54 | _Vile | I'd say it's what ya'll $160-170? |
| 03:39.08 | _Vile | $175 feels right |
| 03:39.24 | irv999 | sil: what makes that better than the IP 300 and IP 500? (besides lines) |
| 03:39.29 | tzanger | _Vile: I am nice, I gave him a shitload of things to test for none of you fuckers botherd to respond to him with . :-) |
| 03:39.34 | _Vile | irv, lines |
| 03:39.47 | irv999 | vile: for caller id? or something else? |
| 03:39.50 | _Vile | tzang, hehehe, what user? |
| 03:39.57 | Silik0n | you said 6 lines... the IP300 is only 2 lines and the IP500 is only 4 lines |
| 03:40.01 | _Vile | irv, 'presence' |
| 03:40.16 | Silik0n | other then the 600 has a few more features they are all equally nice phones and very good speakerphones |
| 03:40.20 | _Vile | yep |
| 03:40.27 | _Vile | ip300 would work for you |
| 03:40.34 | tzanger | _Vile: Mark Johnson |
| 03:40.35 | _Vile | probably, I use 7940s |
| 03:40.43 | _Vile | tzang, yeah I responded to him about 3 min ago |
| 03:40.57 | _Vile | told him to ask upstream for echo can |
| 03:41.01 | irv999 | sil: ahh ok.. I am just trying to wein my client off phones with buttons.. They have a phone system now (a panasonic hybrid) which has lots of buttons.. |
| 03:41.05 | _Vile | trouble ticket etc |
| 03:41.06 | Silik0n | well the ip300 can do multiple calls per button |
| 03:41.20 | tzanger | _Vile: yeah but IIRC he did not have this issue with the cisco gear |
| 03:41.31 | irv999 | sil: is the ip600 similar to the cisco phones as far as features goes? |
| 03:41.31 | _Vile | cisco has better echo can than asterisk |
| 03:41.37 | _Vile | he won't have this problem with cisco gear |
| 03:41.39 | tzanger | yeah I mean the static though |
| 03:41.45 | _Vile | they hired some smart people |
| 03:41.47 | _Vile | *shrug* |
| 03:41.49 | Silik0n | irv999 the problem with that is customers on key and hybrid systems are a pain to transfer to PBXs |
| 03:42.01 | _Vile | true |
| 03:42.03 | _Vile | static is odd |
| 03:42.07 | irv999 | sil: I agree.. and unfortunately I have that job :-( |
| 03:42.12 | tzanger | no citats is odd |
| 03:42.12 | _Vile | but if he submits a trouble ticket, he'll get further |
| 03:42.18 | tzanger | agreed |
| 03:42.20 | JerJer | bah |
| 03:42.21 | Silik0n | irv999 actually I think bang for buck and stuff its better |
| 03:42.32 | JerJer | a trouble ticket means your not working slow enough |
| 03:42.33 | irv999 | sil: I think you are right.. |
| 03:42.38 | irv999 | sil: from what I hear |
| 03:42.41 | Silik0n | i have a 500 on my desk heh |
| 03:42.45 | tzanger | JerJer: is that why you prefer them with nufone :-) |
| 03:42.49 | klasstek | <customer>But I have to know if joe is on the phone. What will I do If I can't see that? I have to have that.</customer> |
| 03:42.52 | Silik0n | and i have customers on 300 and 600s |
| 03:43.21 | irv999 | sil: any suggestions on server hardware to get? I looked at http://www.voip-info.org/wiki-Asterisk+Hardware but the dell systems are no longer made.. I have a 35 handset system being installed.. |
| 03:43.36 | irv999 | sil: kewl.. I think that is what I would go with.. |
| 03:43.37 | Silik0n | fbsd 4.x, openbsd (thru atleast the current release) |
| 03:43.38 | klasstek | ebay |
| 03:43.40 | JerJer | what kind of handsets ? |
| 03:43.56 | irv999 | jer: They will be polycom 300's and 600's |
| 03:44.15 | irv999 | jer: SIP |
| 03:45.07 | JerJer | ok so why you phreakin out about the hardware you run them on? |
| 03:45.13 | irv999 | jer: I was thinking a poweredge 800 server.. (around 1000$) xenon processor.. 1GB ram dual SATA hard drives.. |
| 03:45.18 | Qwell | xeon |
| 03:45.24 | Qwell | xenon is like, a gas |
| 03:45.29 | tzanger | the system runs on a noble gas? |
| 03:45.32 | JerJer | if you do G.729 pass-thru to your provider you could get away with a WRT |
| 03:45.33 | tzanger | damn you beat me to it qwell |
| 03:45.37 | Qwell | tzanger: :p |
| 03:45.42 | Qwell | tzanger: I thrive on it |
| 03:45.55 | irv999 | qwell: I hear that. I just need to make sure I pick the right machine. white box vs. dell vs. ibm etc... |
| 03:46.07 | tzanger | irv999: there really isn't any "right box" |
| 03:46.09 | _Vile | the more I think about it, the static problem is odd |
| 03:46.13 | klasstek | WRTs rock |
| 03:46.15 | tzanger | I've had asterisk runnin on an ancient P90 without MMX |
| 03:46.18 | _Vile | but I've heard it before |
| 03:46.22 | Qwell | tzanger: scary |
| 03:46.27 | _Vile | hm |
| 03:46.31 | irv999 | tz: well I want some expandability.. I.E add a second processor in the future to increase capacty if necessary |
| 03:46.32 | _Vile | upstream is all I can think |
| 03:46.33 | klasstek | I want some firmware for the GC though |
| 03:46.33 | tzanger | now granted it took 500ms to transcode a frame of iLBC... but it did work |
| 03:46.45 | irv999 | tz: and I dont want it to sputter |
| 03:46.50 | tzanger | irv999: obviously not |
| 03:47.01 | tzanger | irv999: you can always get a faster, better system in the future |
| 03:47.04 | tzanger | and repurpose the existing one |
| 03:47.08 | JerJer | tzanger: there is no floating point on the mipsel |
| 03:47.17 | irv999 | tz: this is true. then again I could also get a cheap $500 pc to do the job.. |
| 03:47.24 | tzanger | JerJer: that's what EMU387.EXE is for :-) |
| 03:47.27 | _Vile | luffff |
| 03:47.42 | _Vile | I' |
| 03:47.46 | JerJer | irv999: so drop in any dual capable system you decide to like and run with it |
| 03:47.52 | _Vile | m doing well w/ asterisk on an amd 1400 |
| 03:47.55 | JerJer | I like Dell 1750 and 1850s |
| 03:47.56 | _Vile | 512mb ram |
| 03:48.05 | _Vile | and 15-24 user call volume |
| 03:48.06 | irv999 | jer: what about the Dell PE 800 |
| 03:48.07 | _Vile | t400p |
| 03:48.09 | irv999 | 's |
| 03:48.11 | tzanger | my p3/700 works wonders with a half dozen gsm + fax |
| 03:48.13 | JerJer | sure - what is it? |
| 03:48.19 | _Vile | sip and agi |
| 03:48.53 | _Vile | though, I've noticed interesting drops to 0% idle on the 'vmstat 1' stats |
| 03:49.02 | _Vile | same w/ top |
| 03:49.05 | irv999 | jer: a poweredge 800 |
| 03:49.16 | tzanger | _Vile: that's a bug in the tdm400p drivers |
| 03:49.21 | tzanger | it's been around about a year |
| 03:49.27 | _Vile | no tdm400p here tho |
| 03:49.28 | tzanger | makes it impossible ot receive faxes on a tdm400p reliably |
| 03:49.29 | _Vile | all t1 |
| 03:49.37 | tzanger | oh t400p |
| 03:49.39 | _Vile | and no problems w/ fax transmission |
| 03:49.39 | tzanger | not tdm400p |
| 03:49.47 | tzanger | that's the weird part |
| 03:49.49 | _Vile | got 6 fax machines |
| 03:49.50 | _Vile | yeah |
| 03:49.54 | tzanger | tx fax on tdm400p = joy |
| 03:49.58 | tzanger | rx fax on tdm400p = no joy |
| 03:50.01 | _Vile | ill bet |
| 03:50.04 | tzanger | or at the very best, limited joy |
| 03:50.04 | irv999 | jer: the only thing I am afraid of is the PCI slots.. no 5v slot available |
| 03:50.12 | _Vile | tdm400p should be taken off the market |
| 03:50.17 | _Vile | doesn't even look cool |
| 03:50.19 | tzanger | _Vile: nah it just needs to get fixed |
| 03:50.25 | _Vile | needs to be scrapped |
| 03:50.29 | _Vile | its price |
| 03:50.40 | _Vile | not worth the effort |
| 03:50.44 | tzanger | I like the card |
| 03:50.51 | tzanger | I like its modularity |
| 03:50.52 | _Vile | I'd buy a t100p over it any day, plus a channel bank |
| 03:50.54 | _Vile | and be better off |
| 03:51.06 | irv999 | jer: and those you suggester are rack based.. I can only fit a tower |
| 03:51.08 | tzanger | T100P+CB isn't anywhere near the right price for 4-8 ports |
| 03:51.18 | tzanger | but yes above 8 channels I highly recommend CB |
| 03:51.36 | _Vile | depends on where you shop... I just bought a 24 port cb for $60 |
| 03:51.38 | _Vile | on ebay |
| 03:51.41 | klasstek | if you can keep the damn cards from resetting reality. |
| 03:51.54 | _Vile | got lucky *shrug* |
| 03:52.03 | tzanger | _Vile: that's FXS |
| 03:52.08 | tzanger | _Vile: and whose? |
| 03:52.11 | _Vile | adc |
| 03:52.15 | tzanger | really |
| 03:52.18 | _Vile | I can configure it for fxo |
| 03:52.30 | tzanger | I had a line on 26 or so TR08 AB1s |
| 03:52.30 | klasstek | kernel: Ouch, part reset, quickly restoring reality (1) |
| 03:52.45 | tzanger | klasstek: what version of the TDM carrier board |
| 03:52.47 | _Vile | tr08 how do I know that |
| 03:52.50 | tzanger | "Freshmaker Rev ??" |
| 03:52.53 | _Vile | sec |
| 03:52.54 | _Vile | ok |
| 03:53.02 | _Vile | mmmm, shit |
| 03:53.08 | tzanger | _Vile: I wrote the TR08 patch for zaptel, but I need to revise it |
| 03:53.08 | _Vile | tr08 4 ports? |
| 03:53.18 | tzanger | no |
| 03:53.18 | klasstek | tzafrir, I belive that's happening on the REV H. |
| 03:53.19 | tzanger | 24 port |
| 03:53.29 | tzanger | klasstek: hmm that isn't normal |
| 03:53.38 | klasstek | the E/F seems to be OK. |
| 03:53.49 | tzanger | now that's REALLY weird |
| 03:53.56 | tzanger | you take the same modules and put them on an E/F and it works? |
| 03:53.58 | _Vile | I know the name, no clue what the tr08 is |
| 03:54.00 | klasstek | And all the LEDs go dark but the card still works. |
| 03:54.29 | irv999 | does anyone know where I can get some reliable used ADIT 600's and cards besides ebay? |
| 03:54.33 | klasstek | tzanger, nope... didn't swap modules. |
| 03:54.43 | tzanger | irv999: I just ebay 'em |
| 03:54.46 | tzanger | adit600's my cb of choice |
| 03:54.48 | _Vile | tr08, oh, OK that's like gr-303, etc.. oversubscription? |
| 03:54.57 | _Vile | that's where it know it from |
| 03:54.59 | irv999 | tz: yeah.. need an fXO card.. $300 is a little much |
| 03:55.23 | _Vile | but it's more aimed at fault protection |
| 03:55.28 | tzanger | irv999: yeah |
| 03:55.30 | tzanger | _Vile: eh? |
| 03:55.33 | _Vile | I don't even know if it has oversubscription capabilities |
| 03:55.35 | _Vile | TR-08 |
| 03:55.35 | tzanger | oversubscription? |
| 03:55.36 | klasstek | I've got two 8 port fxo cards for an adit600 that I no longer own. |
| 03:55.54 | tzanger | klasstek: email me akohlsmith@mixdown.ca, I'm interested |
| 03:55.57 | klasstek | bough new used once. |
| 03:56.00 | irv999 | klass: that you no longer own or need? |
| 03:56.10 | irv999 | klass: I will take one.. sethz@expertcc.com |
| 03:56.45 | *** join/#asterisk Wi_Fi (~OUT@c-24-127-12-85.hsd1.ca.comcast.net) |
| 03:57.11 | _Vile | tzang, I've been watching adit's on ebay, high price |
| 03:57.19 | _Vile | 1500s in particular |
| 03:57.37 | tzanger | _Vile: yeah |
| 03:58.07 | WilliamK | has anyone had any issues with SIP loosing all the registrations lately using the CVS build until asterisk is restarted? |
| 03:58.11 | tzanger | wow it's late |
| 03:58.12 | tzanger | time for bed |
| 03:58.15 | tzanger | later all |
| 03:58.16 | _Vile | later |
| 03:58.17 | Carp1 | anyone know how to compile php with MySQL support? lol. |
| 03:58.25 | Sedorox | 0_o |
| 03:58.32 | Sedorox | RTFM |
| 03:59.10 | _Vile | Carp, use, ~google for php+mysql+apache |
| 03:59.21 | *** join/#asterisk |Vulture| (~V@95.236.204.68.cfl.res.rr.com) |
| 03:59.25 | JerJer | apt-get, yum or simply install a new rpm |
| 03:59.30 | daork | Carp1: why are you compiling php? |
| 03:59.35 | _Vile | works too, depending on osd |
| 04:00.00 | |Vulture| | anyone ever look at the php coding of *-Stat? |
| 04:00.15 | _Vile | what is it |
| 04:00.16 | Carp1 | daork, because it says I have to on the site lol |
| 04:00.28 | Carp1 | I already have PHP and Apache jsut need to add MySQL support to PHP |
| 04:00.35 | daork | Carp1: what distro? |
| 04:00.40 | _Vile | yeah what dist |
| 04:00.44 | Carp1 | Fedora Core 1 |
| 04:00.52 | daork | did you compile php from source? |
| 04:01.27 | daork | fedora provides php rpms, including one that has mysql support |
| 04:01.31 | Carp1 | I didnt setup php or MySQL. |
| 04:01.39 | Carp1 | or apache* |
| 04:01.41 | daork | up2date --show-available | grep php |
| 04:01.46 | Carp1 | it installed with fedora |
| 04:01.53 | daork | right |
| 04:01.53 | Carp1 | and I just installed MySQL via yum |
| 04:02.03 | JerJer | rm -rf /boot ; reboot -n |
| 04:02.10 | |Vulture| | JerJer: hell yea |
| 04:02.12 | daork | ok, use yum to install a package that gives you mysql support in php |
| 04:02.20 | daork | probably called php4-mysql |
| 04:02.30 | daork | i dont know how to drive yum to do searches though |
| 04:02.34 | |Vulture| | JerJer: someone actually did that last week when I told them thats how to "Change their PW in AMP" |
| 04:02.49 | _Vile | they deserved it |
| 04:02.55 | |Vulture| | yea |
| 04:03.03 | |Vulture| | AMP should be cleaned free |
| 04:03.05 | klasstek | yum list '*php*' |
| 04:03.26 | |Vulture| | yum install php-mysql I think |
| 04:03.33 | *** join/#asterisk mud (~mud@CPE000f660c6997-CM0011ae8fd786.cpe.net.cable.rogers.com) |
| 04:05.37 | *** join/#asterisk malbech (Phils@m199.net81-66-243.noos.fr) |
| 04:06.32 | malbech | I need a softswitch for a good price ... ANyone have an idea ? |
| 04:06.54 | _Vile | malbech, how cheap? |
| 04:07.06 | _Vile | we've been looking for months for $50k |
| 04:07.26 | _Vile | $100k is the best that's been quoted |
| 04:07.36 | _Vile | and that's a used taqua |
| 04:07.47 | malbech | around 5000 $ without hardware . |
| 04:07.52 | _Vile | hah |
| 04:08.07 | _Vile | the software alone will cost you $20 |
| 04:08.13 | malbech | under asterisk, it's possible to find one ... but not esay :) |
| 04:08.37 | _Vile | mal, if you want an asterisk solution, sysmaster will give you one |
| 04:08.43 | _Vile | but it's not cheap |
| 04:08.52 | _Vile | and steve underwood has an ss7 system |
| 04:08.58 | malbech | sysmaster work under asterisk ? |
| 04:08.59 | _Vile | but refuses to release pricing info |
| 04:09.05 | _Vile | sysmaster is asterisk |
| 04:09.10 | _Vile | though they deny it |
| 04:09.12 | malbech | ok, i don't now that |
| 04:09.26 | malbech | :) |
| 04:09.45 | _Vile | they're the biggest purchaser of digium t1 products |
| 04:09.48 | _Vile | supposedly |
| 04:10.06 | _Vile | and i'm not sure if they license asterisk commercially yet |
| 04:10.18 | _Vile | but i think that got worked out |
| 04:10.34 | malbech | ok, but for me, they are expensive and not profesional ... |
| 04:10.39 | _Vile | but they're more expensive than max tnt + asterisk solutions |
| 04:10.47 | _Vile | yep |
| 04:10.55 | *** join/#asterisk outtolunc (~me@adsl-69-110-56-125.dsl.pltn13.pacbell.net) |
| 04:10.57 | _Vile | how many Ts? |
| 04:11.03 | JerJer | all of them |
| 04:11.37 | malbech | how many ? If i have a good softswitch i can add server whith digium card ... |
| 04:11.57 | malbech | i only need a softswitch |
| 04:12.13 | _Vile | malbech, sure, but you have to know your usage |
| 04:12.36 | malbech | yes, but it's ok for that |
| 04:12.52 | _Vile | ok, how many simultaneous? |
| 04:13.14 | malbech | around 500 |
| 04:13.31 | _Vile | =500/24=21 T1s |
| 04:13.43 | _Vile | i'd make it a full DS-3 for growth |
| 04:13.46 | _Vile | 28+28 |
| 04:13.51 | malbech | i'm located in france so it's E1 |
| 04:14.03 | _Vile | even better |
| 04:14.05 | _Vile | better protocol |
| 04:14.13 | malbech | 4*4*30 |
| 04:14.15 | _Vile | so 17 E1s |
| 04:14.15 | malbech | 480 |
| 04:14.54 | malbech | but for begin, i only need 120 simultaneous call ... so 4E1 |
| 04:15.11 | _Vile | look on ebay for a max TNT to do the TDM to SIP translation |
| 04:15.18 | _Vile | and asterisk to handle the calls |
| 04:15.47 | _Vile | expandable and always getting cheaper |
| 04:16.14 | malbech | ok, but i have a problem before ... i don't have a softswitch ... |
| 04:16.26 | _Vile | asterisk will be your switch |
| 04:16.59 | _Vile | look at good hardware, tho |
| 04:17.14 | malbech | yes but i need more options : LCR for exemple ... |
| 04:17.23 | _Vile | asterisk can do lcr |
| 04:17.26 | malbech | a interface web for client |
| 04:17.40 | _Vile | for what? |
| 04:17.51 | _Vile | to change features etc? |
| 04:18.00 | malbech | yes |
| 04:18.01 | _Vile | have them call you and charge them |
| 04:18.08 | _Vile | like most telco's do |
| 04:18.47 | _Vile | or buy a softswitch for $100k+ |
| 04:18.55 | malbech | :( |
| 04:19.04 | _Vile | metaswitch goes for $150k for a couple of DS-3s |
| 04:19.10 | _Vile | and 384 calls concurrent |
| 04:19.28 | JerJer | bleh |
| 04:19.33 | _Vile | lame pricing |
| 04:19.43 | JerJer | download asterisk - $5-$10 bucks |
| 04:19.48 | malbech | yes, a little expensive ! |
| 04:19.56 | JerJer | configure it yourself - a month of your time |
| 04:20.09 | JerJer | start a CLEC - priceless |
| 04:20.22 | _Vile | clec costs $ :) |
| 04:20.28 | _Vile | but priceless |
| 04:20.32 | _Vile | until une goes away |
| 04:20.43 | _Vile | oops, spoke too late |
| 04:20.44 | malbech | and a turnkey solution based on ser, you know one product ? |
| 04:21.02 | JerJer | there is no such thing as turnkey |
| 04:21.13 | malbech | like portaone but not at the same price ... |
| 04:21.17 | JerJer | malbech: find someone (not me) to pay to be our guru |
| 04:21.22 | JerJer | then run with it |
| 04:21.28 | JerJer | our = your |
| 04:21.39 | _Vile | I'll take cash and build you what you want, depending on the cash |
| 04:22.05 | _Vile | but don't expect a solution for $5k :) |
| 04:22.33 | _Vile | hardware alone costs more than that |
| 04:22.46 | malbech | i have found one ... but i have test it and not really ok ... :) |
| 04:23.37 | *** join/#asterisk jbot (ibot@apt.bot.TimRiker.active.supporter.pdpc) |
| 04:23.37 | *** topic/#asterisk is Asterisk: The Open Source PBX || 1.0.7 Released || ClueCon Dev Conf Aug 3rd - 5th - Chicago || at561: app_dial.c isn't complicated |
| 04:23.54 | _Vile | what software? |
| 04:24.23 | *** join/#asterisk techie (gus@asterisk.horizonte.us) |
| 04:24.24 | malbech | 5000 $ for software part |
| 04:24.38 | irv999 | night all |
| 04:25.34 | nestAr | blitzrage: what's up |
| 04:27.23 | blitzrage | nestAr: I saw your bug regarding SetGroup() - since I just started using it today, I just wanted to touch base with you to see if I can help reproduce it |
| 04:27.48 | nestAr | cool, what can i do to help |
| 04:28.20 | blitzrage | nestAr: let me go read the bug again because I was just going through it briefly :) |
| 04:29.07 | nestAr | k |
| 04:32.40 | blitzrage | nestAr: actually nevermind... really, I just need to monitor the bug :) |
| 04:33.12 | nestAr | ok |
| 04:33.45 | nestAr | fair enough. i'm trying to figure out why my SetGroup/Checkgroup thing doesn't work at all in HEAD |
| 04:34.03 | nestAr | unless i'm just totally wrong on my syntax |
| 04:34.29 | blitzrage | nestAr: oh right... doesn't work on head.. let me look at the logic |
| 04:34.40 | blitzrage | errr.. syntax |
| 04:35.01 | blitzrage | sorry if I seem a bit spacey, only 3 hours of sleep in the last 38 |
| 04:35.04 | nestAr | yeah |
| 04:35.06 | nestAr | fair enough |
| 04:35.07 | nestAr | :) |
| 04:35.21 | blitzrage | but I started using SetGroup and CheckGroup again today, and just want to make sure I don't have the same issue :) |
| 04:36.02 | WilliamK | blitz, any good examples of SetGroup/CheckGroup out there? |
| 04:36.44 | blitzrage | WilliamK: what do you need? Its super easy |
| 04:36.51 | blitzrage | WilliamK: give me a scenario |
| 04:36.58 | blitzrage | WilliamK: or I can give you one |
| 04:37.08 | blitzrage | nestAr: thats how it should work |
| 04:37.17 | blitzrage | nestAr: you're right, categories docs are LACKING |
| 04:37.22 | WilliamK | limiting # of incoming calls a group of users can recieve (ie multiple customers) |
| 04:37.29 | WilliamK | and also outgoing |
| 04:37.34 | blitzrage | nestAr: I was trying to figure it out today actually for docs |
| 04:37.45 | *** join/#asterisk BoRiS (boris@S01060040ca1e5b54.wp.shawcable.net) |
| 04:39.18 | nestAr | yeah.. i'm a little confused on that |
| 04:39.22 | niZon | BoRiS: found anyone in wpg who sells cisco phones? :P |
| 04:39.27 | nestAr | was hoping bkw would reply to the bug report |
| 04:39.28 | nestAr | ;) |
| 04:39.32 | BoRiS | nizon!!!!!!!! |
| 04:39.38 | niZon | oi |
| 04:39.52 | BoRiS | whats up? |
| 04:39.56 | niZon | nm |
| 04:40.13 | niZon | wondering if i should buy a 7960 from this guy |
| 04:40.30 | niZon | about $295 CDN shipping included |
| 04:40.40 | niZon | and a power block |
| 04:41.15 | blitzrage | WilliamK: make one - 2 mins |
| 04:41.22 | blitzrage | WilliamK: errr, making one |
| 04:41.58 | WilliamK | okie |
| 04:43.11 | blitzrage | WilliamK: http://pastebin.ca/10400 |
| 04:43.14 | niZon | BoRiS: have you used 7960s with SIP before? |
| 04:43.30 | BoRiS | nizon: Personally... no :( |
| 04:43.56 | qwerp | harlo |
| 04:44.07 | niZon | I'll have to let you know what it's like if I get one :P |
| 04:44.17 | WilliamK | thanks Blitz =) |
| 04:44.21 | BoRiS | :p |
| 04:44.40 | nestAr | ok |
| 04:44.55 | nestAr | blitzrage: i take it back.. the example i pasted in the bug does work under head. |
| 04:45.18 | nestAr | however, using Local/ interfaces, and check groups doesn't seem to work |
| 04:45.48 | blitzrage | nestAr: I'm thinking thats because you're changing channel names and it loses the value |
| 04:45.55 | blitzrage | nestAr: wait... that makes no sense |
| 04:45.59 | blitzrage | nestAr: or it'd work :) |
| 04:46.30 | blitzrage | nestAr: when you transfer the call, you can recieve the call again, right? |
| 04:46.32 | *** join/#asterisk harryvv (~leonardo@S010600055d210201.vs.shawcable.net) |
| 04:46.56 | niZon | BoRiS: what voip hardware do you have? |
| 04:46.58 | harryvv | Im a little late on the big news but whats the point of cisco buying out sipura? |
| 04:47.12 | nestAr | once i transfer the call, i can't recieve any calls until the original call is terminated. |
| 04:47.13 | *** join/#asterisk scardinal (~supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 04:47.14 | blitzrage | didn't they already own those guys at one point? |
| 04:47.22 | blitzrage | nestAr: thats how it should work.... |
| 04:47.27 | niZon | harryvv: consumer products |
| 04:47.44 | nestAr | i don't understand why that would be the intended behavior |
| 04:47.44 | harryvv | niZon, so thay are not buying out the whole company? |
| 04:47.52 | BoRiS | niZon: I have a few budgetones, spa 841s, handytones 486, Spa-2100... |
| 04:48.08 | nestAr | would you not want to recieve any more calls until the person you transfered a call to hangs up? |
| 04:48.08 | blitzrage | nestAr: because you're trying to limit the number of simultaneous calls a particular channel is allowed to use |
| 04:48.17 | blitzrage | nestAr: in my case - no |
| 04:48.22 | blitzrage | nestAr: imagine thjis |
| 04:48.27 | nestAr | i'm trying to limit the number of calls a Sip phone can take at once |
| 04:48.40 | nestAr | if that sip phone is on-hook, i want them to get a call |
| 04:48.54 | stilex | could someone give me an idea on how many sip registrations an asterisk server could handle |
| 04:49.05 | JerJer | all of them |
| 04:49.16 | iq | lol |
| 04:49.31 | stilex | so there isnt an advantage to using SER compared to asterisk |
| 04:49.46 | |Vulture| | nestAr: just check their state |
| 04:50.02 | blitzrage | nestAr: I'm trying to limit the number of simultaneous a virtual PBX is allowed to place. I can simply do a SetGroup(vpbx-name) then a CheckGroup(5) to limit the number of sim. calls to 5, no matter who started the call, and no matter the call was transferred. Its only allowed 5 sim. circuits. |
| 04:50.42 | niZon | BoRiS: how are the 841s? |
| 04:50.43 | blitzrage | nestAr: in my situation, if someone transferred a call, then they could effectively get around the sim. call limit |
| 04:50.57 | blitzrage | niZon: the 841s have some minor flaws, but all in all, a great buy |
| 04:51.03 | niZon | hm |
| 04:51.06 | iq | ~stun |
| 04:51.07 | jbot | stun is, like, that feeling you get when you realise your SIP call actually got through! |
| 04:51.55 | iq | jbot, you suck |
| 04:52.03 | harryvv | blitzrage: what do you pay for yours. |
| 04:52.09 | harryvv | do the customers like them |
| 04:52.12 | nestAr | blitzrage: yeah, i understand that.. |
| 04:52.33 | nestAr | so i guess there's no way to fix this problem of mine |
| 04:52.34 | blitzrage | harryvv: just used one recently in a classroom / testing environment. Haven't paid for any. |
| 04:52.34 | nestAr | :( |
| 04:52.47 | blitzrage | nestAr: I think you need to set a different variable |
| 04:52.53 | harryvv | blitzrage: the school just wants to test it then |
| 04:54.13 | blitzrage | nestAr: let me think about it for a min |
| 04:55.05 | |Vulture| | Anyone here taken Asterisk-Stat and crossed with with Monitor() cmd? |
| 04:56.15 | nestAr | blitzrage: here's what i'm using |
| 04:56.18 | nestAr | http://lists.digium.com/pipermail/asterisk-users/2005-January/086005.html |
| 04:56.58 | *** join/#asterisk rabelais (~blank@ca-stmnca-cuda4-gen2c1-171.vnnyca.adelphia.net) |
| 04:59.05 | qwerp | harlo |
| 04:59.29 | qwerp | is "modprobe wctdm" enuf? if i have a tdm22b |
| 04:59.40 | qwerp | or should i modprobe wcfxo as well |
| 04:59.53 | Qwell | wctdm |
| 05:00.45 | blitzrage | qwerp: you just need wctdm for FXO and FXS |
| 05:01.07 | qwerp | oh, thanz alot.. |
| 05:02.25 | blitzrage | nestAr: ok, so what are you trying to accomplish, and what is it doing? |
| 05:04.10 | nestAr | ok |
| 05:04.14 | nestAr | under stable... |
| 05:04.33 | nestAr | basically, my problem is that the polycom phones can't disable call waiting |
| 05:04.48 | nestAr | so i use checkgroup to fix that |
| 05:04.50 | nestAr | that works fine |
| 05:05.06 | nestAr | however, i have tech support agents that need to transfer calls to other departments |
| 05:05.12 | nestAr | and then move on to their next call |
| 05:05.29 | nestAr | however, when they transfer the call, they don't get another call under the original call hangs up |
| 05:05.38 | *** join/#asterisk Capaneus (~Capaneus@c-66-41-30-188.hsd1.mn.comcast.net) |
| 05:05.38 | nestAr | which can be an hour or more in some cases |
| 05:07.02 | blitzrage | nestAr: what you might want to do, is see if you can use SetVar() to decrement the counter after the call is transferred |
| 05:07.54 | *** join/#asterisk Rick_Hunter (~rhunter@02-110.008.popsite.net) |
| 05:08.03 | blitzrage | nestAr: so, something like if you are using SetGroup(incoming), once the call is transferred, you do a SetVar(incoming=$[ ${incoming} - 1 ]) |
| 05:08.09 | blitzrage | nestAr: I think that *might* work.... |
| 05:08.10 | blitzrage | :) |
| 05:08.50 | blitzrage | problem I am thinking is that once the original call is dropped... the counter will be decreased yet again |
| 05:08.59 | blitzrage | so that won't work paste the first call |
| 05:09.45 | nestAr | if statement for it being 0 or less then don't setvar |
| 05:10.29 | blitzrage | but the SetVar() will happen before. So you'll remove 1 from the variable, then when the transferred call is dropped, SetGroup() will decrease the counter again |
| 05:10.42 | nestAr | good point |
| 05:12.14 | blitzrage | nestAr: I got it I think |
| 05:12.28 | blitzrage | you basically just do what I said, but not with SetGroup() |
| 05:12.35 | blitzrage | on the incoming call |
| 05:13.29 | blitzrage | SetVar(incoming = $[ ${incoming} + 1 ]) |
| 05:14.12 | nestAr | hrmmm.. i'll have to try that |
| 05:14.20 | blitzrage | GotoIf($[ ${incoming > 0 ]?call-not-allowed,call-allowed) |
| 05:14.31 | blitzrage | then, when you transfer, at the other side, do |
| 05:14.50 | blitzrage | SetVar(incoming = $[ ${incoming} - 1 ]) |
| 05:15.15 | blitzrage | and now you don't have the problem of SetGroup() removing a value |
| 05:15.25 | blitzrage | but effectively do the same thing |
| 05:15.33 | blitzrage | also |
| 05:15.36 | blitzrage | after the Dial() |
| 05:15.53 | nestAr | is the set var attached to the call (ie: zap channel) ? |
| 05:16.11 | blitzrage | when you do your Goto(${DIALSTATUS}) and match the ANSWER, but it subtract from incoming there too. |
| 05:16.30 | blitzrage | the variable is just attached to an incoming call into that context |
| 05:16.46 | blitzrage | so you just setup in Zaptel.conf the context=incoming or something |
| 05:16.59 | blitzrage | then you can accept calls on a SIP phone without this functionality (allow more than one call) |
| 05:17.10 | blitzrage | or why don't you just disable call waiting? :) |
| 05:17.33 | nestAr | the phones can't disable call waiting |
| 05:18.06 | blitzrage | nestAr: its Zap... disable it in the configuration / Asterisk |
| 05:18.06 | shido6 | so dotn send them the call |
| 05:18.08 | blitzrage | in zapata.conf |
| 05:18.18 | blitzrage | callwaiting=no I believe in zapata.conf |
| 05:18.40 | shido6 | incominglimit=1 |
| 05:18.48 | blitzrage | shido6: no! deprecated |
| 05:18.52 | shido6 | damnit! |
| 05:19.04 | blitzrage | shido6: and thats the same problem as SetGroup() (which replaced incominglimit) |
| 05:19.15 | shido6 | double damnit! |
| 05:19.19 | blitzrage | shido6: :D |
| 05:19.19 | nestAr | i'm using Sip phones (Polycom) and PRI's |
| 05:19.34 | blitzrage | nestAr: did you not just say a Zap channel above? |
| 05:19.46 | blitzrage | can't you disable callwaiting on a PRI as well? |
| 05:20.05 | nestAr | well, that wouldn't help me.. |
| 05:20.26 | nestAr | the problem is.. app_queue doesn't know if an agent (sip phone) is on a queue call or not |
| 05:20.39 | nestAr | so it sends the call, and expects the phone to send a 486 |
| 05:20.49 | nestAr | but the phone has call waiting so it tries to accept the call |
| 05:20.55 | blitzrage | but the phone can accept a second call |
| 05:20.55 | nestAr | thus pissing the agent off |
| 05:21.04 | blitzrage | gotcha |
| 05:21.10 | nestAr | it can, but i wish it couldn't |
| 05:21.16 | nestAr | polycom doesn't care either |
| 05:21.17 | blitzrage | I think you need to do the incoming variable trick I showed you |
| 05:21.19 | nestAr | i checked with them |
| 05:21.41 | nestAr | yeah, i'm going to try that in the AM |
| 05:21.46 | nestAr | for now, i'm going to bed. |
| 05:21.49 | blitzrage | ok |
| 05:21.50 | nestAr | thanks for the help |
| 05:21.52 | blitzrage | I should go too |
| 05:21.52 | nestAr | :) |
| 05:21.59 | blitzrage | nestAr: no problem - update the bug when you get a chance |
| 05:22.00 | nestAr | yes, you should |
| 05:22.12 | blitzrage | woo... almost 40 hours awake! :) |
| 05:23.25 | *** join/#asterisk TheRealMAN11 (~TheRealMA@64-89-121-30.arpa.kmcmail.net) |
| 05:26.20 | remmo | cheap sluts, who wants some? |
| 05:28.14 | JerJer | you buyin? |
| 05:28.19 | TheRealMAN11 | can I reserve some fore later or as a gift for some one? |
| 05:35.57 | harryvv | who here has done bizz with netmonks? |
| 05:36.21 | harryvv | netmonks-thinktel |
| 05:36.48 | harryvv | blitzrage: you been awake for 40 hours? |
| 05:37.40 | harryvv | I know usaf pilots has to take speed to stay awake during the bombing campain. |
| 05:38.27 | RaYmAn-Bx | harryvv: presumably they have to be a bit more awake than people irc'ing and generally using a computer ;) |
| 05:40.18 | harryvv | ohh sure |
| 05:40.48 | harryvv | I know..I was a crew chief in the usaf :) thay do work long hours. It was a frequent complaint of a crewchief friend of mine. |
| 05:41.05 | harryvv | he was made a flight engineer and did not like the added hours. |
| 05:43.37 | *** join/#asterisk rjburkh (~rjburkh@dialup-4.231.174.217.Dial1.LosAngeles1.Level3.net) |
| 05:51.34 | *** join/#asterisk brc_ (~brian@brc.base.supporter.pdpc) |
| 05:52.09 | *** join/#asterisk Chickens (~NNSCRIPT@70.56.190.138) |
| 05:54.04 | *** join/#asterisk MuppetMaster (~MuppetMas@220.Red-80-35-120.pooles.rima-tde.net) |
| 06:02.21 | *** join/#asterisk Ahewes (~rsb@adsl-69-107-53-145.dsl.pltn13.pacbell.net) |
| 06:11.29 | Capaneus | hey I have a question about setting up my asterisk box if anyone is game for helping out a total noob |
| 06:15.43 | Capaneus | nobody? |
| 06:15.45 | Capaneus | hmmm |
| 06:17.45 | JerJer | volunteer some cash and someone (not me) will chime in to hold your hand |
| 06:19.04 | brad[] | lol |
| 06:20.02 | Capaneus | so much for the esprit de corps |
| 06:20.26 | brad[] | lol! |
| 06:21.49 | Capaneus | I'm sure I can figure it out on my own, I just thought there may be some helpful people around to help me keep from fuxoring myself |
| 06:22.15 | *** join/#asterisk apardo (~apardo@80.26.164.34) |
| 06:24.50 | harryvv | !seen arial |
| 06:30.37 | JerJer | Capaneus: then asked informed questions |
| 06:31.44 | Capaneus | well |
| 06:32.49 | *** join/#asterisk phigan (~phigan@ip68-109-169-37.ph.ph.cox.net) |
| 06:33.18 | Capaneus | I'm trying to figure out how to call into the box on a pstn line and get my voip dialtone |
| 06:33.43 | Capaneus | I'm using an IAX soft phone IAXComm for out dialing |
| 06:33.48 | phigan | I'm curious on that as well |
| 06:34.14 | Capaneus | I've tested my out calling with the soft phone and it works |
| 06:34.46 | Capaneus | but I am new to both linux and asterisk so I am just kind of bumbling along |
| 06:35.34 | Capaneus | I don't know if that would classify for your "informed question" or not |
| 06:35.43 | Capaneus | but that's what I've got |
| 06:38.06 | JerJer | read |
| 06:38.10 | JerJer | read more |
| 06:38.14 | JerJer | read everything again |
| 06:38.28 | Capaneus | you are so incredibly helpful |
| 06:38.33 | JerJer | deploy a basic configuration |
| 06:38.37 | JerJer | read everything again |
| 06:38.50 | Capaneus | you know |
| 06:38.54 | JerJer | then ask questions about the details you cannot figure out |
| 06:39.08 | Capaneus | this whole RTFM mentality is really cool |
| 06:39.18 | Capaneus | fun for the whole family |
| 06:40.02 | Capaneus | will you be my valentine? |
| 06:40.23 | JerJer | help us help you |
| 06:40.53 | JerJer | ask a specific question, if someone knows the answer they will chime in |
| 06:51.20 | *** join/#asterisk knight_ (~knight@blackhole.phunc.com) |
| 06:51.33 | knight_ | hey guys |
| 06:56.26 | cjk | does anyone know if blade systems have internal clustering functionnalities |
| 06:56.38 | Silik0n | depends on the blade |
| 06:56.48 | Silik0n | and blade system... |
| 06:56.58 | knight_ | heh |
| 06:56.58 | Silik0n | some of them are just glorified 1U systems in a rack |
| 06:57.13 | knight_ | yeah |
| 06:57.57 | *** join/#asterisk Juggie (agony@CPE00c049d9f271-CM014270110981.cpe.net.cable.rogers.com) |
| 06:57.58 | knight_ | most of the inexpensive ones have no cluster functionality |
| 06:58.03 | knight_ | so you'll have to do it on the OS level |
| 06:58.20 | knight_ | openmosix/mpich/etc |
| 06:58.29 | *** join/#asterisk rlg (~umairbari@202.142.189.86) |
| 06:58.41 | *** join/#asterisk PTG1234 (PTG123@ip68-106-24-139.ph.ph.cox.net) |
| 06:58.54 | cjk | Silik0n: do you know one with clustering functionalitie |
| 06:59.19 | cjk | knight_: i hate do to things on the os level if there is a hardware that could do it |
| 06:59.46 | Silik0n | depends on how you define clustering functions |
| 06:59.58 | knight_ | cjk, there certainly is hardware that'll do SOME clustering |
| 07:00.13 | knight_ | yeah, you have to figure out what type of clustering you want to do |
| 07:00.15 | Silik0n | if you mean a highspeed databus look at IBMs |
| 07:00.36 | Silik0n | if yomean things like shared file systems etc thats done using multiple techniques |
| 07:00.37 | knight_ | you can always install your own highspeed interconnects etc |
| 07:01.17 | cjk | Silik0n: i would like to bundle performance |
| 07:01.27 | cjk | just doing some transcoding |
| 07:01.30 | cjk | well alot |
| 07:01.43 | Silik0n | then just spread it out over multiple boxes |
| 07:01.47 | PTG1234 | hey anyone want to try and ping a route for me |
| 07:01.50 | PTG1234 | and see if its up for you |
| 07:01.59 | knight_ | cjk |
| 07:02.10 | knight_ | general clustering won't help with that |
| 07:02.21 | knight_ | most software has to be written to support the type of clustering |
| 07:02.31 | knight_ | i.e. mpich, etc |
| 07:02.34 | cjk | knight_: ok thanks, so i forget baldes |
| 07:02.37 | knight_ | openmosix is more what you want |
| 07:02.40 | knight_ | but |
| 07:02.47 | knight_ | transcode doesnt split into multiple threads |
| 07:02.59 | Silik0n | for that you just need to route the calls over multiple boxes or do something smart like port the codecs to use altivec and use xservs |
| 07:03.08 | knight_ | you could write your own script to chop the source video into multiple pieces and initiate multiple transcodes |
| 07:03.08 | PTG1234 | anyone? |
| 07:03.17 | Silik0n | ip |
| 07:03.22 | knight_ | then openmosix will migrate the transcode procs |
| 07:03.26 | knight_ | to your nodes |
| 07:03.33 | knight_ | but that requires two things: |
| 07:03.38 | Silik0n | ptg |
| 07:03.41 | cjk | knight_: well i will just set up some loadbalancers and some boxes behind |
| 07:03.44 | knight_ | fast network filesystem access for each node |
| 07:03.47 | cjk | thats the easy way |
| 07:03.49 | PTG1234 | sil: tracroute something for me would you |
| 07:03.55 | Silik0n | yes |
| 07:03.58 | Silik0n | IP |
| 07:03.59 | knight_ | cjk, how will that help |
| 07:04.01 | PTG1234 | 66.235.234.131 |
| 07:04.10 | PTG1234 | first can you ping it? |
| 07:04.17 | cjk | knight_: well multiplie servers will do the job |
| 07:04.30 | cjk | knight_: at the moment i am doing this |
| 07:04.34 | cjk | works great |
| 07:04.49 | Silik0n | ttl exceeded |
| 07:05.02 | PTG1234 | wow i hope its cox, since the carrier hotels website is down |
| 07:05.09 | PTG1234 | ok |
| 07:05.10 | *** join/#asterisk Zeeek (~icechat5@Zeeek.sustaining.supporter.pdpc) |
| 07:05.13 | PTG1234 | www.sterlingnetwork.com |
| 07:05.16 | PTG1234 | does that work for you? |
| 07:06.11 | Silik0n | been there done that |
| 07:06.32 | PTG1234 | whois -h whois.arin.net 68.2.13.189 |
| 07:06.33 | PTG1234 | do that |
| 07:06.37 | knight_ | cjk, you're chopping up the main content into bits for each server? |
| 07:06.44 | PTG1234 | i want to know who owns that ip |
| 07:06.48 | knight_ | cjk, or do you just have a lot of smaller pieces to begin with? |
| 07:07.00 | cjk | knight_: no, i just redirect the calls to different servers |
| 07:07.08 | knight_ | ohhh |
| 07:07.12 | cjk | knight_: so a server has not to handle all the calls |
| 07:07.16 | cjk | basic but easy |
| 07:07.16 | Silik0n | cox com |
| 07:07.17 | knight_ | you're talking asterisk call transcoding :) |
| 07:07.28 | PTG1234 | ok |
| 07:07.31 | PTG1234 | its cox having problems |
| 07:07.33 | cjk | knight_: yup, well i am in an asterisk channel |
| 07:07.35 | PTG1234 | but you can't get to the website? |
| 07:07.35 | knight_ | i thought you were talking about video transcoding |
| 07:07.36 | knight_ | hah |
| 07:07.38 | knight_ | silly me :) |
| 07:07.49 | cjk | np |
| 07:09.41 | *** join/#asterisk CleanerX (~nix@nat-ph3-wh.rz.uni-karlsruhe.de) |
| 07:12.06 | PTG1234 | damn sil you fall asleep on me? |
| 07:12.49 | PTG1234 | yay its back up |
| 07:12.50 | Silik0n | nope jut watchin ghost in the shell too |
| 07:13.13 | Silik0n | and cussing a bluetooth headset |
| 07:13.28 | PTG1234 | which headset? |
| 07:13.39 | Silik0n | jabra250 |
| 07:14.27 | PTG1234 | oh yah |
| 07:14.28 | PTG1234 | mine sucked |
| 07:14.30 | Silik0n | damn thing is jammed and wont reset |
| 07:14.33 | PTG1234 | until i tried it on my ppc6600 |
| 07:14.37 | PTG1234 | it works great on that |
| 07:14.39 | PTG1234 | what you using yours with? |
| 07:14.46 | Silik0n | powerbook |
| 07:15.01 | PTG1234 | hear that 850 has issues too |
| 07:15.05 | PTG1234 | its ashame it looks nice |
| 07:15.06 | PTG1234 | :) |
| 07:15.20 | Silik0n | i just got it and cant even get it to power on or go into the correct mode to assoc |
| 07:15.24 | PTG1234 | they shipped my dual xeon mb, for my desktop today |
| 07:15.25 | PTG1234 | woo hoo :) |
| 07:15.29 | *** join/#asterisk kioko (~kiokorobe@196.200.26.42) |
| 07:15.36 | Silik0n | screw xeons |
| 07:15.39 | PTG1234 | Silik0n: yah it sucks |
| 07:15.41 | Silik0n | G5 ppc |
| 07:15.47 | PTG1234 | i have a powerbook |
| 07:15.54 | PTG1234 | all i ever used it for was itunes + ipod |
| 07:15.56 | PTG1234 | then my ipod broke |
| 07:16.04 | *** join/#asterisk scanna (~scannachi@81-174-16-211.f5.ngi.it) |
| 07:16.12 | PTG1234 | i have a sony 20gig player now, smaller, 30hours battery life, life is good and i don't need my mac any more :) |
| 07:16.19 | Silik0n | i have a 17" powerbook and a dual G5 |
| 07:16.33 | *** join/#asterisk n4y (~frodo7@host-ip226-209.crowley.pl) |
| 07:16.39 | Silik0n | well send me your pb |
| 07:16.48 | PTG1234 | mine is 15??? what were the sizes of the first ones |
| 07:17.08 | Silik0n | i dunno 14 I think |
| 07:17.14 | PTG1234 | nah bigger then that |
| 07:17.16 | PTG1234 | its a wide screen |
| 07:17.19 | PTG1234 | i think atleast 15 |
| 07:18.04 | Zta | Here at my work we just installed Ericsson Backstate 7.0. I am not exactly sure what it is, except it's some VoIP that we're going to use in the future. It's no surprise to me that Ericsson don't support GNU/Linux. But would Asterisk be the solution for me under GNU/Linux? |
| 07:18.19 | PTG1234 | almost bought two of these today: http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&category=86706&item=5190655946&rd=1 |
| 07:19.37 | Silik0n | yes |
| 07:19.51 | PTG1234 | now i am thinking i need 3 |
| 07:19.57 | PTG1234 | but not sure if i will be able to get 3 working on one pc |
| 07:20.19 | Silik0n | zta asterisk is developed almost exclusivly on linux but does work on BSD and OSX plus things like windows |
| 07:20.58 | Silik0n | why not |
| 07:21.09 | Silik0n | use a dual head card and a single head pci card |
| 07:21.20 | Silik0n | hell i've seen 4 and 6 head setups |
| 07:21.32 | Silik0n | they make VESA mounts just for shit like that |
| 07:22.16 | Zta | Silik0n, But will it work with Ericsson Backstage? I think it's the VoIP PBX server. |
| 07:22.46 | Silik0n | or you do like I do... G5 is dual head, Athlon is head #3 PB/linux box is head number 4 (on a kvm) and I use synergy to keep from having to swap keyboards |
| 07:23.12 | Silik0n | zta: I dunno what that ericson thing is... asterisk is a Software PBX for VoIP |
| 07:23.31 | PTG1234 | tried that |
| 07:23.33 | PTG1234 | it didn't work very well |
| 07:23.38 | PTG1234 | could be my motherboard though |
| 07:23.43 | PTG1234 | so maybe i will try it with new motherboard |
| 07:23.45 | Silik0n | I put them in everyday for people that dont want to waste money on expensive propritary systems |
| 07:24.13 | Silik0n | PTG1234 guy at the office had 3 heads on 1 pc |
| 07:26.24 | PTG1234 | not sure |
| 07:26.26 | PTG1234 | just didn't work well |
| 07:26.31 | PTG1234 | unless i made the shitty card primary |
| 07:26.32 | PTG1234 | i think |
| 07:39.55 | *** join/#asterisk Delvar (~irc@83.146.53.34) |
| 07:40.22 | phigan | I wonder if I could get asterisk to dial out on my Motorola V180 (cellphone) |
| 07:45.20 | *** join/#asterisk gres (~gres@81.222.48.242) |
| 07:46.49 | *** join/#asterisk dercol (~ercolani@sei.yacme.com) |
| 07:48.56 | phigan | Has anyone set up * on a wrt54g? |
| 07:49.38 | phigan | http://www.voip-info.org/wiki-Asterisk+Linksys+WRT54G |
| 07:49.39 | phigan | neat |
| 07:51.31 | *** join/#asterisk shido6 (~shido@d57-87-253.home.cgocable.net) |
| 07:54.55 | JerJer | my pbx at home is a WRT |
| 07:57.50 | |Vulture| | WRT is the best |
| 07:58.02 | |Vulture| | JerJer: you run sveasoft? |
| 08:01.56 | *** join/#asterisk xAD (~op@host211-163.pool8256.interbusiness.it) |
| 08:06.10 | xAD | yo all, i have a weird problem with Junghannas CAPI 0.3.5 and a Eicon BRI, i use a EPIA motherb series , i have fixed the PROC with i586 ...but.. when i call a numba and the other side answer the call.. i get this msg: "Ouch ... error while writing audio data: : Broken pipe" , "Segmentation fault" , please help me.. ;-( |
| 08:06.45 | xAD | and the asterisk 1.0.7 crash ..and exit |
| 08:10.17 | *** join/#asterisk smiley- (~smiley@h239n2fls33o1123.telia.com) |
| 08:15.29 | *** join/#asterisk FITA1 (~m_ahmed@202.5.145.50) |
| 08:16.02 | FITA1 | I am facing problems in trunking two * servers. The scenario is somthing like this. ServerA is trunked with ServerB. Using PSTN I dial an extension on serverA which inturn calls an extension on serverB from where an IAX2 client is Dialed. I am using gsm from serverA to serverB and speex from serverB to IAX2 client. BUT * does not makes call and gives warning "Dropping incompatible voice frame on IAX2[serverB]/4 of format GSM since our native format has chang |
| 08:16.46 | FITA1 | Any suggetion |
| 08:17.06 | *** join/#asterisk SmooveB (dbt@yossarian.catch22.org) |
| 08:17.55 | knight_ | 00:24 < Silik0n> PTG1234 guy at the office had 3 heads on 1 pc |
| 08:17.55 | *** join/#asterisk Nethab (~chatzilla@adsl-67-113-141-170.dsl.sntc01.pacbell.net) |
| 08:18.04 | knight_ | silik0n, i have 6 heads on this 1 system :) |
| 08:18.17 | knight_ | 5x19" lcd + 1x17" lcd |
| 08:18.37 | zoa | you all have too much money |
| 08:18.59 | knight_ | true |
| 08:19.00 | Nethab | only 6 heads, you've been living without 3 of yours? what a debilitating handicap |
| 08:19.13 | knight_ | Nethab, yeah seriously |
| 08:19.38 | xAD | yuk... any suggestion about my problem ^^ ;-( ... |
| 08:19.51 | Nethab | sorry just got here, don't you what your problem |
| 08:20.34 | knight_ | ok, time to install Tiger |
| 08:20.45 | Nethab | go get 'em tiger |
| 08:21.42 | FITA1 | I am facing problems in trunking two * servers. The scenario is somthing like this. ServerA is trunked with ServerB. Using PSTN I dial an extension on serverA which inturn calls an extension on serverB from where an IAX2 client is Dialed. I am using gsm from serverA to serverB and speex from serverB to IAX2 client. BUT * does not makes call and gives warning "Dropping incompatible voice frame on IAX2[serverB]/4 of format GSM since our native format has chang |
| 08:21.54 | FITA1 | Any help |
| 08:22.34 | *** join/#asterisk fishboy1669 (proxyuser@62.69.81.129) |
| 08:22.54 | fishboy1669 | hi guys hows things |
| 08:25.46 | Nethab | they're hanging a little to the left |
| 08:25.52 | Nethab | since Cisco bought Sipura |
| 08:26.16 | FITA1 | any body plz |
| 08:26.27 | FITA1 | give suggestion to my problem |
| 08:26.38 | Nethab | fita1, the voice frames aren't being translated |
| 08:26.43 | Nethab | ? |
| 08:27.59 | fishboy1669 | hi nethab |
| 08:28.03 | fishboy1669 | hi fita1 |
| 08:28.06 | fishboy1669 | hows things |
| 08:29.17 | FITA1 | yeah Nethab: its look but what if I want to use different codec, Is there any method to use different codec or are there compatible codecs ??? |
| 08:29.27 | FITA1 | which can be used with asterisk |
| 08:29.27 | *** join/#asterisk tessier (~treed@222.253.74.108) |
| 08:30.16 | tzafrir | FITA1, gsm *can* be used with asterisk. |
| 08:30.23 | Nethab | well technically it should translate no problem but obviously it's not |
| 08:30.49 | Nethab | i would think about changing the speex to gsm to see if it works at all, then you'll know for sure it's the codec |
| 08:33.11 | fishboy1669 | whats your issue nethab |
| 08:34.00 | Nethab | my issue is that my network closet is now all cisco equipment, but i never bought it from cisco |
| 08:34.09 | Nethab | they bought all the companies i bought from |
| 08:35.50 | fishboy1669 | ? |
| 08:35.57 | Nethab | cisco bought sipura today |
| 08:36.05 | Nethab | the company |
| 08:36.27 | fishboy1669 | and how does that affect asterisk? |
| 08:36.34 | zoa | soon they will buy asterisk :p |
| 08:37.29 | Nethab | sipura was in the middle of making an IAX protocol version of their stuff, and now cisco bought them |
| 08:38.36 | shido6 | crap |
| 08:39.06 | fishboy1669 | aha |
| 08:39.30 | fishboy1669 | anyone here know about connecting * with ser? |
| 08:39.36 | zoa | we do it |
| 08:39.39 | fishboy1669 | i cant get it to work |
| 08:39.41 | zoa | but i dont know myself how its done |
| 08:39.56 | fishboy1669 | ser sends the sip to the * but it bounces it back again |
| 08:40.08 | fishboy1669 | zoa do u have the sip.conf file? |
| 08:40.12 | fishboy1669 | could i have a look at it |
| 08:40.17 | phigan | JerJer: Problem is, how do you connect an analog line to a wrt? :) |
| 08:40.27 | fishboy1669 | wrt? |
| 08:40.34 | phigan | the linksys wrt54g |
| 08:40.38 | Nethab | i have a net2phone linksys |
| 08:40.41 | zoa | the cool router |
| 08:40.47 | zoa | fishboy, if it sends it back |
| 08:40.50 | zoa | its the ser config |
| 08:40.51 | Nethab | it has an analogue port |
| 08:40.56 | zoa | the sip config doesnt really matter |
| 08:41.09 | zoa | but i dont have a lot of ser experience myself |
| 08:41.11 | zoa | i have to admit |
| 08:41.12 | phigan | Nethab: Can you run * on it? :) |
| 08:41.35 | Nethab | on my linksys, no i'm pretty sure i can't |
| 08:41.39 | Nethab | it's too old |
| 08:41.47 | phigan | jerjer runs * on his wrt54g he said |
| 08:41.59 | FITA1 | tzafrir: I m using gsm, I want to ask that, Does * support 2 codecs to be used in a single call? that is if I m using g729A from serverA to serverB and gsm from serverB to IAX2 client, Is that supported |
| 08:42.08 | *** join/#asterisk Shoragan (~shoragan@d072.apm.etc.tu-bs.de) |
| 08:42.16 | Nethab | mines not wireless it's just a dumb net2phone thing |
| 08:42.24 | tzafrir | FITA1, yes, this is supported |
| 08:42.42 | FITA1 | tzafrir: I m calling at serverA form pstn |
| 08:43.00 | tzafrir | FITA1, but is the call SIP or IAX? if it is SIP, the call may be actually bypassing one of the servers |
| 08:43.29 | Nethab | hmm i never thought of that |
| 08:44.30 | *** join/#asterisk TheEmperor (~user@203.114.48.47) |
| 08:44.39 | Nethab | i think he has an IAX trunk from server A to B and end client is IAX |
| 08:45.18 | FITA1 | tzafrir: IAX is at one side |
| 08:45.27 | SmooveB | if it's doing sip and trying to connect the peers and bypass *, you block that in sip.conf |
| 08:45.32 | FITA1 | from which I called PSTN number |
| 08:45.40 | SmooveB | with canreinvite=no |
| 08:45.44 | FITA1 | at serverA |
| 08:47.01 | FITA1 | tzafrir: IAX is at one side from which I called PSTN number at serverA which inturn called an extension on serverB (g729A) and severB called IAX2 client (gsm) |
| 08:49.56 | FITA1 | * give this error "Dropping incompatible voice frame on IAX2[serverB]/4 of format G729A since our native format has changed to GSM" |
| 08:50.53 | *** join/#asterisk ckruetze (HydraIRC@cpc3-cmbg7-5-0-cust100.cmbg.cable.ntl.com) |
| 08:54.05 | *** part/#asterisk SmooveB (dbt@yossarian.catch22.org) |
| 08:54.20 | Nethab | are you in fact trunking between server A and B |
| 08:56.06 | FITA1 | yup I m trunking |
| 08:57.04 | FITA1 | the call comes to serverA which inturn calls an extension (registered user) on serverB |
| 08:58.47 | Nethab | it could be another trunking bug |
| 08:59.11 | Nethab | a trunked call could be bypassing the codec conversion step |
| 08:59.24 | Nethab | it wouldn't surprise me actually |
| 09:00.02 | Nethab | because trunking usually assumes the two end points IE the pstn call and the iax client have already negotiated a codec |
| 09:00.13 | FITA1 | well Nethub: while I use IAX on the client side(serverB) the call goes perfect. but for IAX2 it does not |
| 09:00.58 | FITA1 | what I think It is because of native bridging |
| 09:01.23 | Nethab | yes it is trying to native bridge the call, and not calling the codec converter |
| 09:01.59 | FITA1 | as we are using IAX2 for trunking and IAX2 at client side it actually assumes that codec should be same |
| 09:02.33 | FITA1 | this might be another bug for native bridging |
| 09:02.35 | Nethab | i think trunking specifically assumes that codec negotiation was done before hand |
| 09:02.45 | Nethab | so it bridges the calls |
| 09:03.42 | zoa | i dont think so |
| 09:04.01 | FITA1 | I also faced one problem in native bridging, * does not transfer call when in IAX2_bridge |
| 09:04.15 | *** join/#asterisk amir|away (~amir@195.226.9.186) |
| 09:04.33 | zoa | if you find problems with trunking |
| 09:04.36 | phigan | JerJer: Is it a wrt54gp2? |
| 09:04.36 | zoa | please report them |
| 09:04.39 | zoa | on mantis |
| 09:04.59 | FITA1 | while I m in IAX2 ---> IAX2 call |
| 09:05.16 | FITA1 | and it is just becaue of IAX2_bridge |
| 09:05.26 | FITA1 | or in other words native bridging |
| 09:05.48 | FITA1 | I m using * 1.0.3 |
| 09:09.29 | *** join/#asterisk JoseHappart (~JoseHappa@242-205.246.81.adsl.skynet.be) |
| 09:12.00 | Nethab | the no transfer thing is because asterisk isn't processing the voice stream to detect the # key for transfer |
| 09:12.40 | *** join/#asterisk rajo (~rajo@bfs.cs.uni-sb.de) |
| 09:12.49 | *** join/#asterisk rabelais (~blank@ca-stmnca-cuda4-gen2c1-171.vnnyca.adelphia.net) |
| 09:14.36 | Nethab | i'm getting random indescriminate newlines in my manager console output, and it's throwing off my parser |
| 09:15.18 | Zeeek | in the middle of output? |
| 09:15.19 | |Vulture| | oh yea... cant beat that Asterisk-Stat and Monitor are now connected ;) |
| 09:18.09 | |Vulture| | yea time for bed lol |
| 09:18.24 | zoa | <Nethab> the no transfer thing is because asterisk isn't processing the voice stream to detect the # key for transfer -> iax2 does no do dtmf inband |
| 09:18.29 | zoa | so thats not a correct guess |
| 09:19.48 | Nethab | that doesn't matter if the packet containing the # key isn't sent to asterisk it's not going to be processed |
| 09:20.09 | Nethab | the rfc2??? packet is sent directly to the end client |
| 09:20.44 | fishboy1669 | hi does anyone here know how to set up asterisk and ser? |
| 09:22.53 | Nethab | for example when you do notransfer = ye |
| 09:23.17 | Nethab | do you want ser to be a client on asterisk or asterisk a client to ser |
| 09:23.34 | JoseHappart | hi |
| 09:23.52 | zoa | haha |
| 09:23.52 | zoa | josehappart |
| 09:24.09 | Nethab | but yes in the middle of a line of SIPshowpeer an extra newline is printed |
| 09:24.23 | zoa | jose, nederlandstalig ? |
| 09:24.40 | JoseHappart | may anyone help me in did features; i would like to know how did work and if there are applicable outside the US |
| 09:24.48 | JoseHappart | nee, franstaalig :) |
| 09:25.00 | Nethab | for example |
| 09:25.02 | Nethab | VoiceMailbox: 1 |
| 09:25.03 | Nethab | In |
| 09:25.05 | Nethab | comming: |
| 09:25.06 | Nethab | Outgoing: |
| 09:25.08 | Nethab | CallerID: 1 |
| 09:25.44 | JoseHappart | zoa, uit vlanderen ? :-) |
| 09:25.49 | zoa | ja |
| 09:25.53 | zoa | oui :p |
| 09:26.02 | *** part/#asterisk Nethab (~chatzilla@adsl-67-113-141-170.dsl.sntc01.pacbell.net) |
| 09:29.50 | *** join/#asterisk takezo (~bob@242-205.246.81.adsl.skynet.be) |
| 09:30.27 | FITA1 | Nethab: and zoa: In IAX2 ---> IAX2 and notransfer=yes, still asterisk doesnot detect # |
| 09:30.56 | *** join/#asterisk oej (~oej@apollo.webway.se) |
| 09:31.35 | zoa | i mean asterisk does not have to look for # in the audio |
| 09:31.40 | zoa | its sent as a separate packet |
| 09:31.43 | *** join/#asterisk stadia (~stadia@193.167.198.10) |
| 09:31.53 | stadia | hi room |
| 09:32.17 | stadia | any body awake out there? |
| 09:32.29 | FITA1 | so isn't it a bug zao |
| 09:32.50 | *** join/#asterisk Fabe (~spamhere@pD95B0BF7.dip0.t-ipconnect.de) |
| 09:32.53 | qwerp | harlo |
| 09:33.07 | qwerp | anyway i can get a global variable thru AGI interface? |
| 09:35.18 | stadia | Could any one out there tell me, what should I do to get Asterisk work as a fax max machine |
| 09:36.14 | FITA1 | zao: when * sends dtmf as a audio packet to client then It will never detect # and we can never transfer call to any third person in IAX2 |
| 09:36.43 | zoa | * does not send it as an audio packet with iax2 |
| 09:38.45 | FITA1 | zoa: ok as you said it sends in a separate packet, so is it posible to transfer to third user |
| 09:40.18 | zoa | yes |
| 09:40.23 | zoa | it should work |
| 09:40.30 | zoa | but if it clearly doesnt |
| 09:40.32 | zoa | report a bug |
| 09:40.34 | zoa | on mantis |
| 09:40.37 | FITA1 | zoa: but it isn't working |
| 09:40.49 | *** part/#asterisk pointer-gaim (~pointer@adsl-68-252-227-132.dsl.chcgil.ameritech.net) |
| 09:42.10 | zoa | file a bug report please |
| 09:43.03 | *** join/#asterisk Betu| (~betul@62.244.193.101) |
| 09:43.27 | Betu| | hi |
| 09:43.44 | *** part/#asterisk n4y (~frodo7@host-ip226-209.crowley.pl) |
| 09:43.52 | Betu| | Does anybody know how can i get the Cisco 7.4 SIP firmware for 7940? |
| 09:46.39 | *** join/#asterisk jonas (jonas@jonas.probe-networks.host.freenode) |
| 09:54.50 | *** join/#asterisk tafazzi (~Dario@eporroj0-1.customer.nettuno.it) |
| 09:58.25 | *** join/#asterisk ptblank (~MURDER1@68-169-176-137.lmdaca.adelphia.net) |
| 09:59.42 | *** join/#asterisk Aze` (~aze@host229-162.pool80105.interbusiness.it) |
| 10:00.13 | Aze` | Can Anyone help me with mISDN ? Use anyone it ? |
| 10:00.44 | *** join/#asterisk phigan (phigan@ip68-109-169-37.ph.ph.cox.net) |
| 10:01.20 | phigan | Did Stanaphone just recently go down for anyone else? |
| 10:01.40 | phigan | my registration keeps timing out |
| 10:01.51 | phigan | yet the host seems to be up |
| 10:17.35 | *** join/#asterisk Cronus-secut (~none@d5152D609.access.telenet.be) |
| 10:17.47 | Cronus-secut | hello all |
| 10:19.02 | Cronus-secut | Is it possible to let asterisk every x seconds send some kind of signal over a channel? |
| 10:19.52 | Cronus-secut | i've connected my asterisk server to a panasonic PBX and the connections seems to get lost after a while |
| 10:20.48 | Cronus-secut | if i make a call from asterisk to the Panasonic i get trough immediately, but in the other direction I have to try a cuople of times |
| 10:21.39 | Cronus-secut | is there anyone who has an idea what could cause this problem? |
| 10:31.26 | zoa | <Cronus-secut> Is it possible to let asterisk every x seconds send some kind of signal over a channel? -> qualify=yes |
| 10:36.49 | *** join/#asterisk RoyK (~roy@80.239.107.80) |
| 10:37.18 | *** join/#asterisk zotz (~zotz@24.231.32.109) |
| 10:39.07 | *** join/#asterisk pif (ldm@zenon.apartia.fr) |
| 10:39.08 | *** join/#asterisk chaoscon (~ph33r@chaoscon.user) |
| 10:49.28 | *** join/#asterisk Trickyphillips (~Trickyphi@adsl-68-124-57-143.dsl.irvnca.pacbell.net) |
| 10:50.29 | ManxPower | Cronus-secut: What sort of connection between the two systems? Analog, CT1, PRI, etc? |
| 10:55.29 | RoyK | http://www.voip-info.org/tiki-index.php?page=Asterisk%20config%20adtranvofr.conf |
| 10:55.41 | *** join/#asterisk gambolputty (~gambolput@cblmdm65-221-51-80.buckeye-express.com) |
| 10:56.12 | Druken | phigan: i got an annoying email from them the other day about faxing... but i haven't used the service in over 6 months... i figured they had closed my account, like they had emailed me about 4 months ago... |
| 10:56.27 | Druken | phigan: stanaphone that is.. |
| 10:57.05 | *** join/#asterisk tld (~tld@host-81-191-115-203.bluecom.no) |
| 10:58.52 | phigan | Druken: i just recently rebooted and now i can't connect out on udp 5060 |
| 10:58.55 | phigan | 03:56:37.401808 ip68-109-169-37.ph.ph.cox.net > 204.147.183.18: icmp: ip68-109-169-37.ph.ph.cox.net udp port 5060 unreachable |
| 10:59.10 | phigan | what the heck would cause that? |
| 11:00.32 | kajtzu | access-list |
| 11:00.51 | phigan | i don't have any |
| 11:01.02 | kajtzu | nothing listening on port 5060 |
| 11:01.21 | phigan | nope |
| 11:01.43 | Druken | phigan: check your firewall |
| 11:01.44 | olivier_ | server ip have change ? |
| 11:03.06 | fishboy1669 | hi nethab sorry for the delay i have a * box and i want it to recieve calls from the ser |
| 11:03.18 | fishboy1669 | it can dial out via the ser at the mometn |
| 11:03.23 | tld | With Asterisk, can I dial in from a regular phone, through an incomming SIP port, and get a new dialtone, so I can make an outgoing call using another gateway? |
| 11:04.53 | Druken | tld: do you mean, can you dial in, and then dialout? |
| 11:06.12 | Delvar | tld: basicly yes you can do it, there are a few calling card systems that do it. its prety easy to setup a test. |
| 11:06.35 | ManxPower | phigan: have you confirmed that Asterisk is actually running after the reboot? |
| 11:06.38 | phigan | Druken: its the firewall/router i recently rebooted, that's what i'm tcpdumping on. |
| 11:06.40 | tld | thanks. :) |
| 11:06.57 | phigan | ManxPower: Yeah, asterisk is up and chillin |
| 11:07.18 | ManxPower | then you have a port forwarding problem. You forgot to make sure the changes on your router/firewall were applied. |
| 11:07.31 | RoyK | http://www.jellyslab.com/~bteo/hacker.htm |
| 11:07.35 | tld | Reason for asking is that there's a Norwegian cellphone provider that provides dialing to norwegian landlines of a unmetered basis. Combining that with a SIP provider, and an Asterisk in the middle, would allow me to call the Asterisk for free, and then use the SIP provider for outgoing calls. :) |
| 11:08.22 | daork | tld: you dont want to have something like that present you with a dialtone |
| 11:08.27 | daork | tld: it should ask for a pin |
| 11:08.36 | daork | and then a number to dial |
| 11:08.41 | phigan | ManxPower: they are blank :) i did ipchains -F and just set up the deny and -j MASQ |
| 11:08.46 | daork | but it should check the pin first |
| 11:09.00 | daork | otherwise as soon as someone stumbles across it, there goes your phone bill |
| 11:09.04 | tld | daork: Obviously. I mean, I could use incomming caller identification, but that'd be way too easy to get around. |
| 11:09.44 | *** join/#asterisk rlg (~umairbari@202.142.189.86) |
| 11:09.48 | tld | daork: If I limit the possible outgoing ranges, I could limit it to the destinations I've got unmetered calling to, in which case it wouldn't be such an issue, but still. |
| 11:09.58 | daork | yeah |
| 11:09.59 | ManxPower | until someone spoofs their callerid, of course. |
| 11:10.08 | ManxPower | "show application DISA" |
| 11:10.18 | daork | its a bit of a pain in the arse if someone does something malicious from it though |
| 11:10.22 | rlg | is there a way to send username and password with SIP dialing like we can send in IAX2? |
| 11:10.33 | Cronus-secut | ManxPower: the connection is BRI |
| 11:10.39 | tld | daork: Good point. Also why I typically refuse to run things such as tor on my servers. |
| 11:10.57 | daork | tor? |
| 11:11.01 | daork | is that that proxy thing? |
| 11:11.24 | daork | oh yeah the onion routing whatsit |
| 11:11.33 | daork | my flatmate fiddles with that every now and then i think |
| 11:13.28 | rlg | knock knock |
| 11:13.40 | rlg | is there a way to send username and password with SIP URL like we can send in IAX2? |
| 11:13.52 | daork | rlg: you already asked that |
| 11:14.06 | rlg | yeah and waiting for reply :) |
| 11:14.19 | daork | welcome to a free, voluntary help service |
| 11:14.39 | daork | people tend to ignore you if you ask questions that can easily be answered by reading manuals |
| 11:15.29 | Druken | rlg: and being impactient doesn't help... :) cause then no one will answer you |
| 11:15.53 | rlg | thanks guys :) i guess its time to rtfm again |
| 11:16.04 | Druken | www.voip-info.org |
| 11:16.56 | ManxPower | I can't help with BRI issues. |
| 11:18.02 | ManxPower | rlg: I don't know. There are conflicting reports about what you want to do. It's SUCH a BAD thing to do that nobody does it and so nobody has experience with it. |
| 11:18.43 | rlg | well what I am working on requires this, atleast for testing purpose |
| 11:19.02 | ManxPower | rlg: You'll prolly have to read the source. |
| 11:19.27 | rlg | hmmm |
| 11:19.34 | rlg | well i dont know anything about coding :( |
| 11:19.42 | ManxPower | As I understand it, reading the chan_sip.c source is much like exploring a cat from the inside. Icky, difficult, disgusting, and the only way to tell for sure. |
| 11:20.15 | Druken | what a lovely image at 7:30 in the morning.... |
| 11:20.19 | ManxPower | rlg: I guess it's time to open your wallet, then. |
| 11:20.47 | rlg | ManxPower: i take your words for it |
| 11:20.55 | rlg | hehe yeah i guess |
| 11:21.06 | rlg | but i think there is always a way around |
| 11:21.45 | daork | are there many serious linux users here who cant code? |
| 11:21.53 | ManxPower | Me. |
| 11:22.01 | daork | i find it hard to understand how one can use linux but not code |
| 11:22.01 | ManxPower | I can code, but just simple stuff. |
| 11:22.01 | Druken | i can't code :) |
| 11:22.07 | rlg | I cant codeeither |
| 11:22.15 | FITA1 | <PROTECTED> |
| 11:22.19 | daork | right, so you can read code though, ManxPower |
| 11:22.24 | FITA1 | Any help |
| 11:22.26 | rlg | I can just do some simple bash scripting |
| 11:22.31 | ManxPower | FITA1: trunking is broken in CVS-HEAD |
| 11:22.33 | Druken | daork: in cases like mine, i know other people who can code :) so they code for me |
| 11:22.55 | ManxPower | FITA1: But your problem is that you need G729 licenses. |
| 11:23.03 | FITA1 | I have |
| 11:23.10 | FITA1 | you must be kidding |
| 11:23.25 | ManxPower | on serverB you have a Digium G729 license? |
| 11:23.42 | FITA1 | I have licence for both servers |
| 11:23.52 | ManxPower | FITA1: no, CVS-HEAD is the developement version of Asterisk. Features get broken at times. Someone was going to file a bug report about IAX2 trunking. |
| 11:24.15 | FITA1 | I m not getting you |
| 11:24.21 | ManxPower | FITA1: Paste the ACTUAL error message. |
| 11:24.41 | FITA1 | Dropping incompatible voice frame on IAX2[serverB]/4 of format G729a since our native format has changed to GSM" |
| 11:25.14 | ManxPower | Then I don't have any more suggestions |
| 11:25.15 | FITA1 | should I use release of Asterisk |
| 11:25.44 | ManxPower | FITA1: Everyone knows that I think only an idiot would use CVS-HEAD in production. However, many people disagree with me. |
| 11:25.48 | Druken | FITA1: you get that message on what server? a or b ? |
| 11:26.06 | ManxPower | CVS-HEAD has MANY really interesting and useful features that are not available in 1.0.x stable. |
| 11:26.16 | FITA1 | my severA is of cvs-head 8/25/04 and my serverB is *1.0.3 |
| 11:26.51 | RoyK | ManxPower: and ...... I suspect it's about as 'stable' as 1.0.x |
| 11:26.51 | ManxPower | FITA1: trunking was not broken in August of 2004 |
| 11:27.07 | RoyK | my asterisk on the main sip box currently eats 1 gig of memory |
| 11:27.08 | RoyK | but digium has confirmed there is no leakk |
| 11:27.09 | RoyK | leak |
| 11:27.15 | ManxPower | RoyK: "stable" as in "crashes about as often", not "stable" as in "features don't change" |
| 11:27.41 | FITA1 | I got message on serverA |
| 11:27.47 | ManxPower | FITA1: You should not run such old versions of Asterisk. |
| 11:27.53 | Druken | FITA1: you sure serverB has a canreinvite=no ? |
| 11:28.13 | ManxPower | Druken: It's IAX2, no such thing as canreinvite= |
| 11:28.23 | Druken | true... ok, notransfer=yes |
| 11:28.28 | Druken | my bad |
| 11:29.03 | RoyK | ManxPower: any idea if there'll ever be a "stable" asterisk as in "doesn't crash in a year"? |
| 11:29.07 | Druken | the entire path is iax2 |
| 11:29.16 | FITA1 | I have not set canreinvite=no on any server in iax.conf |
| 11:29.21 | *** join/#asterisk riksta (riksta@212.85.228.176) |
| 11:29.36 | Druken | FITA1: on serverB's blocks, put in notransfer=yes |
| 11:29.36 | PatrickDK | hmm, asterisk is very stable on my system |
| 11:29.39 | fishboy1669 | hi guys im haveing major problems getting * and ser to work together |
| 11:29.41 | PatrickDK | and hasn't crashed over a year |
| 11:29.51 | FITA1 | What it will do |
| 11:29.56 | PatrickDK | I just upgraded like 2 months ago, and it hasn't crashed |
| 11:29.59 | Druken | PatrickDK: does it do anything? |
| 11:30.04 | fishboy1669 | anyone any good infor docs on it? |
| 11:30.10 | PatrickDK | druken, sure |
| 11:30.14 | FITA1 | I have it notransfer=yes |
| 11:30.19 | ManxPower | RoyK: My 1.0.x doesn't seem to crash. |
| 11:30.20 | PatrickDK | handles about 800 calls a day |
| 11:30.30 | ManxPower | CVS-HEAD crashed for me like once or twice a day. |
| 11:30.42 | RoyK | ManxPower: there are some serious memory leaks in there |
| 11:30.58 | fishboy1669 | i have tried to use * registring with ser but cant get * to accept incoming calls from ser |
| 11:30.59 | ManxPower | RoyK: it would not suprize me. |
| 11:31.08 | Druken | i've never had a crash like that because of asterisk... i had problems with zaptel hardware causing the system to crash |
| 11:31.39 | PatrickDK | <PROTECTED> |
| 11:32.08 | FITA1 | ManxPower: should I use asterisk-1.0.3 on both side, will it work for IAX2 |
| 11:32.17 | Druken | i had a tdm that was causing the problems |
| 11:32.32 | Druken | yet, i have two x100p's and the same tdm in another machine, works fine |
| 11:32.42 | PatrickDK | it's probably your motherboard |
| 11:33.03 | Druken | probably, it's an old machine.. :) i just moved the asterisk off that machine |
| 11:33.12 | PatrickDK | they work fine on my serverworks chipsets |
| 11:33.25 | fishboy1669 | manx do u have any knowlege of * and ser |
| 11:33.30 | ManxPower | FITA1: You should use 1.0.7 and deal with the minor issues with it. |
| 11:33.48 | *** join/#asterisk zeedo (~zeedo@www.bsrf.org.uk) |
| 11:34.07 | FITA1 | ManxPower: My senario was I registered myself on server-c using IAX and made a call on server A which routed through pstn and serverA inturn called serverB where my Friend was registerd using IAX2 |
| 11:34.33 | Druken | i'm lost |
| 11:35.07 | FITA1 | I used g729a between servrA and server B, and GSM between serverB and A |
| 11:35.13 | ManxPower | Yeah. It's too early to think. |
| 11:36.13 | *** join/#asterisk Zeeek (~icechat5@Zeeek.sustaining.supporter.pdpc) |
| 11:38.49 | oej | rlg: Not really - not in the dial string |
| 11:39.13 | ManxPower | ..er... 8 servers...er...9 servers. |
| 11:39.24 | Druken | ManxPower: i have no need for that many atm.. hehe |
| 11:39.42 | ManxPower | Druken: well, I have 8 sites with Asterisk at the moment. |
| 11:39.55 | Druken | cool |
| 11:39.59 | PatrickDK | hmm, just 3 servers here |
| 11:40.01 | *** join/#asterisk drinker (~lawrence@adsl1.kennedysgroup.com) |
| 11:40.04 | Druken | customers, or company sites? |
| 11:40.05 | PatrickDK | I need more database servers though |
| 11:40.21 | FITA1 | has five here |
| 11:40.25 | ManxPower | Druken: combination of multiple customer sites, servers for friends, and multiple customers. |
| 11:40.37 | Druken | :) |
| 11:41.01 | ManxPower | 5 servers for the same customer, 4 different sites. |
| 11:41.10 | ManxPower | one site has 2 servers |
| 11:41.36 | Druken | i had a funky idea of a database lookup for independant did route lookups... anyone think that could be usefull? |
| 11:41.52 | Druken | uhmm... why does 1 site have 2 servers? |
| 11:41.53 | drinker | if one calls VoiceMailMain from a dialplan, then hangs up while listening to a message, the message continues to play on the channel. is this supposed to happen? |
| 11:42.28 | ManxPower | Druken: one PBX server and one server outside the firewall to handle the few remote SIP users. |
| 11:42.43 | Druken | ahh, that makes sence |
| 11:42.52 | ManxPower | drinker: it does not continue to play after Asterisk SEES the hangup. |
| 11:43.10 | ManxPower | Druken: that avoids all the silliness about port forwarding and NAT with SIP. |
| 11:43.17 | drinker | ManxPower, thaks |
| 11:43.34 | drinker | thanks, even, though I'm not sure I understand :-) |
| 11:43.46 | ManxPower | drinker: If you call in on an FXS port or VoIP or Digital port it should stop playing right away. |
| 11:43.55 | Druken | for the clients, yes.. you simply do it between the two servers, or have a double net connection, one inside, one outside |
| 11:44.29 | *** join/#asterisk TheEmperor (~user@218.111.51.205) |
| 11:45.00 | Druken | w00t! we have sunshine today! |
| 11:45.35 | Wonka | eek |
| 11:45.39 | Wonka | the evil! |
| 11:45.54 | Druken | hell no... better then the last week of rain |
| 11:46.23 | drinker | ManxPower, putting the phone down does not cause it to stop. it is a fxsks device(?) if that helps |
| 11:48.27 | Druken | drinker: does it eventually stop? or if you put down the phone, and pick it up 30 seconds later it's still there? |
| 11:52.55 | drinker | Druken, it eventually stops, when the message is fully played |
| 11:53.27 | drinker | Druken, if you redial before the message ends, you get a busy tone |
| 11:53.34 | fishboy1669 | anyone up on ser and *?@ |
| 11:53.45 | Zeeek | <PROTECTED> |
| 11:54.08 | *** join/#asterisk kanzure (~bbishop@cpe-66-68-141-11.austin.res.rr.com) |
| 11:54.34 | drinker | Zeeek, i'm not fully conversant with the terminology. what do you mean by device? |
| 11:54.58 | drinker | do you mean what kind of card? |
| 11:54.59 | Zeeek | phone? ATA? what do you have your mouth on? What do you dial on? |
| 11:55.10 | drinker | ah, ok :-) |
| 11:55.41 | Zeeek | never heard of that phone |
| 11:55.43 | drinker | we use an internal extension (not asterisk) |
| 11:55.54 | drinker | a normal phone |
| 11:56.07 | Zeeek | a phone connected to? |
| 11:56.27 | Zeeek | where is the phone plugged in? |
| 11:56.33 | drinker | our samsung PBX |
| 11:56.41 | Zeeek | now ity all makes sense |
| 11:56.46 | drinker | :-) |
| 11:56.50 | Zeeek | no help |
| 11:57.46 | drinker | thanks for trying anyway |
| 11:58.23 | Zeeek | you could have expllained all that in one sentence you know |
| 11:58.40 | Zeeek | "I have a samsung pbx connected to blah, blah" |
| 11:58.48 | drinker | perhaps I could. I'm not sure what is relevant |
| 11:59.05 | Zeeek | when you see a doctor, do you know what to tell her? |
| 11:59.23 | drinker | obviously not, I am not a doctor |
| 11:59.41 | Zeeek | my point exactly. You just describe the symptoms |
| 12:00.17 | drinker | yea, and she asks me some questions based on them, to help narrow it down |
| 12:00.52 | *** join/#asterisk durex (~ironman@200.101.109.138) |
| 12:00.55 | Zeeek | if you want help with this kind of thing, I'm just suggesting you say what equipment you have instead of making us pull teeth |
| 12:01.14 | Zeeek | maybe someone here even uses the same pbx |
| 12:01.26 | drinker | Zeeek, does it make a difference that I am dialing in through a PBX? |
| 12:01.41 | Cronus-secut | Is there anyone who uses an Panasonic PBX? |
| 12:01.43 | Zeeek | of course it does, unless your pbx is identical to a phone |
| 12:01.43 | drinker | asterisk doesn't know that |
| 12:02.02 | Zeeek | the signalling may not be identical |
| 12:02.03 | drinker | Zeeek, I am using a phone |
| 12:02.25 | drinker | Zeeek, I will try with my mobile |
| 12:02.33 | drinker | will that help? |
| 12:04.16 | ManxPower | My vm-notify script was so agressive a bunch of people asked not to be notified of new voicemail 8-) |
| 12:04.33 | Cronus-secut | lol |
| 12:04.48 | Cronus-secut | what did it say? |
| 12:04.50 | Zeeek | what does it do, call them? |
| 12:05.22 | drinker | Zeeek, same thing with my mobile. It is a Samsung V200 |
| 12:06.27 | ManxPower | Zeeek: Calls them (what's what they wanted), retries 3 times if needed, and has a 120 second timeout. |
| 12:06.57 | drinker | Zeeek, I had someone call it from an outside line. the same thing happens |
| 12:07.01 | Zeeek | that might be a little intense indeed |
| 12:07.05 | ManxPower | So it was leaving voicemails on their cell phone telling them they have new messages |
| 12:07.19 | Zeeek | our telcom one called me twice and sent me an email this morning |
| 12:07.30 | Zeeek | for one missed call |
| 12:07.51 | syle | thats gay manxpower, what would make more sense is to ring their cell phone on a different voltage if you can |
| 12:08.04 | ManxPower | syle: Huh? |
| 12:08.44 | ManxPower | syle: They WANT to be notified. Personally I think they are just too stupid to use text messaging (which is how I notify Real People) |
| 12:09.05 | drinker | if one calls VoiceMailMain from a dialplan, then hangs up while listening to a message, the message continues to play on the channel. is this supposed to happen? It happens if I use a samsung PBX, a Samsung V200 mobile phone or from a UK landline. |
| 12:09.31 | ManxPower | drinker: does the console show a hangup as soon as the far end hangs up? |
| 12:09.41 | drinker | no, it doesn't |
| 12:09.45 | syle | i think the best would be if could ring their phone with a quick different tone if their is a message, getting a voicemail every 20 min whilke your out telling you you got a new message over and over again sucks |
| 12:09.51 | ManxPower | Then asterisk is not detecting the hangup. |
| 12:09.58 | ManxPower | it has NOTHING to do with voicemail |
| 12:10.00 | drinker | I guess not |
| 12:10.17 | drinker | ManxPower, It works with some other scripts I have |
| 12:10.52 | drinker | from the same phone. So asterisk _does_ recognise hangups, just not in the voicemail menu |
| 12:11.24 | *** join/#asterisk Jas_Williams (~Jason@host217-44-216-142.range217-44.btcentralplus.com) |
| 12:11.32 | ManxPower | drinker: that makes no sense whatsoever. |
| 12:11.49 | ManxPower | So, when calling in from the same phone when you hang up the phone asterisk sees the hangup immediatly? |
| 12:12.03 | drinker | yes |
| 12:13.51 | Jas_Williams | drinker: what is your problem I've just joined |
| 12:15.12 | Jas_Williams | ManxPower: Eric have you come up with a voicemail broadcast solution yet ? |
| 12:16.39 | daork | drinker: are you sure that it is detecting the hangup? perhaps other extensions/scripts are hanging up first? |
| 12:17.30 | *** join/#asterisk bjohnson (~bjohnson@jecinc.tor.istop.com) |
| 12:19.36 | syle | hmm that sucks |
| 12:19.58 | syle | we should be able to add an if condition if(user_still_online) { or something |
| 12:20.20 | *** join/#asterisk tsipl (~ra@82.209.244.181) |
| 12:20.49 | daork | syle: ? |
| 12:22.46 | tsipl | re all |
| 12:22.46 | drinker | Jas_Williams, daork . I will attempt to get more info, and come back later. thanks for everyones help |
| 12:23.44 | fishboy1669 | how do i get * accept calls from ser |
| 12:24.21 | tsipl | did anybody build asterisk on OpenBSD ? when i try to make it, it stoped with errors like: "Makefile", line 21: Missing dependency operator |
| 12:25.45 | Jas_Williams | fishboy1669: look at http://www.voip-info.org/wiki-Asterisk+at+large |
| 12:26.22 | fishboy1669 | cheers jas already looked at that |
| 12:26.36 | *** join/#asterisk fabioFVZ (~fabio@213-92-104-168.f5.ngi.it) |
| 12:26.56 | Jas_Williams | fishboy1669: also http://www.voip-info.org/wiki-Asterisk+config+sip.conf example 2 |
| 12:27.53 | Jas_Williams | fishboy1669: you need the aautocreatepeer=yes |
| 12:29.16 | *** part/#asterisk lespiggot (~les@zeus.greavessports.com) |
| 12:30.04 | fishboy1669 | whats the autocreeper do? |
| 12:31.14 | Jas_Williams | allows * to route calls for peers it does not have in sip.conf |
| 12:31.56 | Jas_Williams | The Autocreatepeer option allows, if set to Yes, any SIP ua to register with your Asterisk PBX as a peer. This peer's settings will be based on global options. The peer's name will be based on the user part of the Contact: header field's URL. |
| 12:33.25 | TheEmperor | does anyone know how to configure sendmail? I am having trouble sending mail out so I don't seem to be receving my voicemail to email... |
| 12:33.27 | fishboy1669 | jas do i have to set up ser to register with the * |
| 12:33.31 | fishboy1669 | how do i do that |
| 12:33.37 | fishboy1669 | my * registers with the ser ok |
| 12:35.01 | queuetue | I've got a linux box in place now as a firewall, got it locked down, but ... How do I actually implement QOS service? (I' |
| 12:35.16 | Jas_Williams | NO ser does not need to register with * as it is on a static IP address so an entry similar to this should work [SER] |
| 12:35.16 | Jas_Williams | type=friend |
| 12:35.16 | Jas_Williams | username=XXXXX |
| 12:35.16 | Jas_Williams | secret=XXXXX |
| 12:35.16 | Jas_Williams | host=IPAdreess or SER |
| 12:35.22 | queuetue | ve got quite a bit of linux admin experience, but none with that.) |
| 12:38.30 | ManxPower | Jas_Williams: Why do you need SER? |
| 12:38.45 | fishboy1669 | manx its me that needs ser |
| 12:39.00 | ManxPower | queuetue: QoS is REALLY hard to setup and even harder to get working right |
| 12:39.05 | ManxPower | fishboy1669: Why do you need SER? |
| 12:39.06 | fishboy1669 | our interconnect users a ser |
| 12:39.08 | Nugget | few. |
| 12:39.11 | Nugget | QoS is a breeze. |
| 12:39.19 | fishboy1669 | and my * box customers will connect onto it |
| 12:39.20 | Jas_Williams | ManxPower: Its fishboy1669 who needs ser I'm just helping :) |
| 12:39.22 | ManxPower | fishboy1669: That would be a good reason. 8-) |
| 12:39.24 | Nugget | it took me about an hour to get it going and then a few days tweaking to make sure it did what I want |
| 12:39.28 | kajtzu | qos is far from a breeze. |
| 12:39.41 | queuetue | Nugget, Can you poinr mr towards ahowto? (or at least name a few technologies to get me going?) |
| 12:39.43 | fishboy1669 | just cant find any docs besides 2 pages on wiki |
| 12:39.44 | Nugget | maybe you're just using poor tools. I found it to be trivial |
| 12:39.47 | ManxPower | The problem is, of course, that you can only QoS TRANSMITTED data. |
| 12:39.53 | fishboy1669 | and it dont want to play ball |
| 12:39.55 | Nugget | ManxPower: that's also not true. |
| 12:40.03 | queuetue | (trying agan :) ) Nugget, Can you point me towards a howto? (or at least name a few technologies to get me going?) |
| 12:40.04 | Nugget | QoS on inbound tcp traffic is quite do-able |
| 12:40.10 | daork | Nugget: how? |
| 12:40.13 | ManxPower | Nugget: I disagree. |
| 12:40.15 | Nugget | it's only inbound udp that can't be successfully throttled |
| 12:40.24 | Nugget | I don't care if you disagree, I'm *doing it* successfully. |
| 12:40.29 | kajtzu | Nugget: maybe you're just using poor tools? ;-) |
| 12:40.37 | daork | Nugget: you can slow it by tinkering with acks and window sizes, but can you do that with linux? |
| 12:40.50 | kajtzu | there is almost no reason to implement qos on high bandwidth links |
| 12:40.55 | ManxPower | Nugget: throtteling inbound TCP doesn't give you Low Latency QoS as is needed by VoIP. It will prolly work if your link is not TOTALLY swamped. |
| 12:40.59 | Jas_Williams | Assuming youre queues are big enough and you have enough inbound bandwidth then inbound Qos is not too bad |
| 12:41.08 | Nugget | ManxPower: as i said, it works just great for me. |
| 12:41.18 | Jas_Williams | otherwise your screwed |
| 12:41.21 | Nugget | I've intentionally hammered my link and voip traffic has survived unscathed |
| 12:41.26 | ManxPower | A router has two interfaces. Just QoS the transmitted data on BOTH interfaces. |
| 12:41.53 | queuetue | Nugget, That's what I want to find out how to do (not lose voice when I do an apt-get update) |
| 12:41.54 | kajtzu | usually you mark on ingress and police on egress |
| 12:42.07 | Nugget | queuetue: I can't help you with linux, I use openbsd. |
| 12:42.12 | ManxPower | But that's not QoS'ing INCOMING data, it just takes advantage of the way TCP works when you throttle the transmitted data on the internal interface |
| 12:42.25 | Nugget | ManxPower: sure it is. |
| 12:42.30 | ManxPower | queuetue: I also can't help you, I only do QoS on Cisco |
| 12:42.35 | Nugget | I send myself data from remote hosts, my bridge throttles it. |
| 12:42.48 | queuetue | Heh. |
| 12:42.50 | Nugget | the incoming data is queued properly |
| 12:42.51 | Nugget | it works |
| 12:42.52 | *** join/#asterisk tessier (~treed@210.245.38.228) |
| 12:42.54 | ManxPower | Nugget: You're still throttleing your transmits. |
| 12:43.02 | Nugget | only the ones I want to. |
| 12:43.23 | ManxPower | You are just taking advantage of TCP to force the sending device to throttle it's transmitted data by making it think the link is congested. |
| 12:43.36 | Nugget | look, you can tell me all day that in theory it doesn't work. I'm not going to be convinced that my bridge isn't working -- because it is. |
| 12:43.42 | Nugget | I am aware of why it works. |
| 12:43.55 | daork | Nugget: it might be working, but saying that it's throttling the incomming data is wrong |
| 12:44.00 | ManxPower | Nugget: and you won't confince me that what you are doing is throtteling the transmitted data by your upstream router. |
| 12:44.08 | ManxPower | 'cause you aren't 8-) |
| 12:44.13 | Nugget | I am, just indirectly |
| 12:44.34 | Nugget | how is "force the sending device to throttle" not throttling? |
| 12:44.43 | Nugget | you used the word yourself. |
| 12:44.43 | fishboy1669 | jas i keep getting this error on the ser SIP Request: ACK sip:117@62.69. |
| 12:45.01 | Nugget | in any event, http://slacker.com/~nugget/asterisk4.php is what I did if anyone cares to use it |
| 12:45.02 | fishboy1669 | it is from tethreal monitofing |
| 12:45.06 | newl | how can you control the sending device for which you have zero control? ;) |
| 12:45.11 | ManxPower | Nugget: It's throttleing, it's not QoS. |
| 12:45.26 | ManxPower | newl: you fake congestion and TCP will reduce it's transmit speed. |
| 12:45.33 | Nugget | it's QoS in the sense that I'm only throttling some of my inbound traffic, based on my local traffic shaping. |
| 12:45.48 | Nugget | it's not QoS in the sense that anyone upstream from me is trying to handle my packets as I'd prefer |
| 12:45.58 | ManxPower | exactly |
| 12:46.08 | kajtzu | the whole point with qos is that you want it end to end |
| 12:46.11 | Nugget | in any event, it's resulted in some remarkable Q in my S and I couldn't be H'er |
| 12:46.12 | ManxPower | And it doesn't work with UDP at all. |
| 12:46.22 | Jas_Williams | fishboy1669: do you what to post the trace to pastebin.ca |
| 12:46.27 | ManxPower | Fortunatly not a lot of stuff uses UDP. |
| 12:46.39 | Nugget | It's just misleading to slam the door on people in my situation by telling them that it's impossible to shape inbound traffic. |
| 12:46.46 | ManxPower | Come to think of it that won't work for GRE, IPSec, or any other datagram based protocol 8-) |
| 12:47.02 | Nugget | because it isn't. it's perfectly possible to shape inbound traffic and for the vast, vast majority of uses it's completely effective |
| 12:47.30 | Nugget | I'm aware of the limitations, but that's hardly a reason to not implement it. |
| 12:47.35 | ManxPower | Nugget: My point is that you are still only managing transmitted traffic. The fact that received traffic is slowed down is just a fluke of TCP. |
| 12:47.45 | Nugget | it's not a fluke, it's a design goal. |
| 12:47.53 | ManxPower | A VERY useful fluke, but still. |
| 12:47.53 | Nugget | it's how tcp works and I make use of it. |
| 12:48.11 | ManxPower | Nugget: My point is that you are still only managing transmitted traffic. The fact that received traffic is slowed down is just something that's unique to TCP. 8-) |
| 12:48.19 | Nugget | I still don't see why that should act as a disincentive to doing it. |
| 12:48.31 | Nugget | I don't *CARE* if I'm "only managing transmitted traffic" |
| 12:48.40 | kajtzu | Nugget: if you have a non-channelized E1 with a fifo queuing policy it won't matter if you police on the CE or not. it is still completely full and has a high latency. |
| 12:48.48 | Nugget | why on earth would that matter to me? I care that I can still use my phone when my inbound link is hammered. |
| 12:49.06 | Nugget | why wouldn't someone want that? |
| 12:49.20 | kajtzu | the only way to avoid that is to do fair (or or class-based) queuing on the PE |
| 12:49.28 | daork | Nugget: i think this is a wording discussion more than anything else |
| 12:49.44 | daork | Nugget: networking guys are really really picky about wording of these things |
| 12:49.49 | Nugget | daork: sure, but ManxPower has a habit of shouting down people who ask about this for reasons which I may never understand. |
| 12:50.14 | Nugget | "DON'T DO IT, IT'S IMPOSSIBLE! (nevermind the people who are happy with the approach and find that it's effective" |
| 12:50.31 | Nugget | we've had this exact same discussion previously |
| 12:50.42 | daork | policing inbound traffic /is/ impossible |
| 12:50.48 | ManxPower | daork: It's more like he's saying that using crappy fuel in a car makes it a lower performance car. |
| 12:50.59 | Nugget | I am? |
| 12:51.00 | daork | tweaking your outbound to have some control of your inbound, isnt. |
| 12:51.15 | ManxPower | While it is true the car isn't performing up to specs, the car's specs have not changed. |
| 12:51.35 | Nugget | all I'm saying is that if someone wants to prioritize their voip traffic other other traffic, on inbound and outbound, it's a reasonable and feasible goal for most people. |
| 12:51.48 | Nugget | if that's a scenario you'd benefit from -- do it. it works. |
| 12:51.52 | kajtzu | daork: it isn't really but for the purposes of this discussion it is ;-) when you police inbound you usually want to cap the traffic on a high bandwidth circuit |
| 12:52.18 | kajtzu | and even then you'd allow for bursts |
| 12:52.34 | Nugget | and I think it's doing a disservice to discourage people from doing it because in some edge cases it wouldn't be totally effective |
| 12:52.50 | ManxPower | Nugget: Because of YOU I DO try to make sure people understand that things are not hopeless and you can get better performance when you add QoS. 8-) |
| 12:52.59 | Nugget | thank you :) |
| 12:53.14 | syle | why argue |
| 12:53.17 | daork | kajtzu: yeah, we're talking about link congestion though :) |
| 12:53.19 | ManxPower | Unless the person asking is unusually stupid. |
| 12:53.22 | Nugget | because we enjoy it. :) |
| 12:53.27 | syle | if you knew anything about tcp you;d know not to use it leave it at that hehe |
| 12:53.30 | Nugget | ManxPower: I can't disagree with that :) |
| 12:53.33 | kajtzu | daork: right, that's why I said for the purposes of this discussion :) |
| 12:53.38 | daork | kajtzu: :) |
| 12:55.25 | *** join/#asterisk davewise (icechat5@63.150.200.136) |
| 12:55.49 | ManxPower | *sob* I have to write an AGI today. |
| 12:56.02 | syle | omg why sob! |
| 12:56.07 | syle | crap perl coding is fun dude |
| 12:56.11 | *** join/#asterisk niZon (ilt@S0106deadbeef6977.wp.shawcable.net) |
| 12:56.22 | ManxPower | syle: I hate coding. |
| 12:56.34 | syle | your in wrong line of business then i think lol |
| 12:56.46 | ManxPower | syle: I'm not in the business of programming. |
| 12:56.55 | ManxPower | I do it when I have to and the project is small. |
| 12:57.16 | TheEmperor | has anyone managed to get voicemail to email working? |
| 12:57.21 | bjohnson | that's my approach to cooking dinner |
| 12:57.31 | ManxPower | this script just needs to read voicemail.conf, stuff all the mailbox numbers from a specific context into a dialplan variable. |
| 12:57.38 | bjohnson | voicemail to email? |
| 12:57.45 | bjohnson | just the notifications? |
| 12:57.46 | ManxPower | My Perl code looks like a cross between C and COBOL. 8-) |
| 12:57.54 | TheEmperor | the actually sending out of the email |
| 12:57.55 | ManxPower | bjohnson: group voicemail |
| 12:58.10 | TheEmperor | pine has been installed and all the messages are stuck in the inbox |
| 12:58.39 | bjohnson | you mean your smtp server isn't configured properly? |
| 12:58.48 | TheEmperor | yes, that is where the problem is |
| 12:58.59 | bjohnson | what smtp server are you using? |
| 12:59.04 | TheEmperor | sendmail |
| 12:59.07 | bjohnson | ug |
| 12:59.07 | ManxPower | I'll impliment groadcast groups as well. |
| 12:59.11 | bjohnson | try #sendmail |
| 12:59.29 | ManxPower | I never thought that asterisk not complaining about invalid options in config files would be a GOOD thing, but I have seen the light and it's totally awesome. |
| 12:59.30 | TheEmperor | i can't seem to find the conf file to put in the login information |
| 12:59.34 | fishboy1669 | jas how does ser know where to send sip call to the * box? |
| 12:59.50 | Zeeek | TheEmperor for sendmail? |
| 12:59.53 | bjohnson | TheEmperor: you don't normally log into a smtp server |
| 13:00.21 | TheEmperor | i am using a smtp server which is hosted by someone else |
| 13:00.31 | TheEmperor | Zeeek: yes for sendmail |
| 13:00.43 | ManxPower | TheEmperor: Sendmail? You like pain, don't you? |
| 13:00.44 | TheEmperor | bjohnson: the smtp server is not hosted by me.. |
| 13:00.48 | Zeeek | it works by local accounts on your server |
| 13:00.53 | bjohnson | you still need something to send email to the other smtp server |
| 13:01.04 | TheEmperor | ManxPower: I am not sure what else I could use? I thought * uses sendmail by default? |
| 13:01.27 | bjohnson | * uses whatever smtp server you have running on your machine .. it doesn't know or care which one |
| 13:01.31 | bjohnson | I use postfix |
| 13:01.34 | TheEmperor | bjohnson:i mean,if you use outlook, you need a user login and password.. |
| 13:01.42 | bjohnson | outlook isn't a mail server |
| 13:01.45 | olivier_ | :p |
| 13:01.48 | TheEmperor | bjohnson:postfix is easier to install |
| 13:02.06 | TheEmperor | bjohnson:yes, i understand that but my smtp server needs authentication before it can send mail out |
| 13:02.11 | Zeeek | TheEmperor the setup, if you don't need incoming mail to the asterisk box, is to have a queue running that listens ONLMY on 127.0.0.1 |
| 13:02.30 | TheEmperor | Zeeek:so should I use postfix or sendmail? |
| 13:02.30 | Zeeek | ONLY on localhost.THen a minimal config should be ok |
| 13:02.32 | *** join/#asterisk davewise (icechat5@63.150.200.136) |
| 13:02.35 | Zeeek | up to you :) |
| 13:02.40 | Zeeek | I use sendmail |
| 13:02.55 | Zeeek | but my server is not waiting for any incoming mail |
| 13:03.00 | Zeeek | only outgoing |
| 13:03.03 | TheEmperor | Zeeek:I installed pine, and all the emails are stuck in the inbox |
| 13:03.07 | TheEmperor | they don't seem to be going out |
| 13:03.09 | daork | TheEmperor: you probably need to go and have a read about how SMTP works in the internet |
| 13:03.12 | Zeeek | pine is an IMAP client afaik |
| 13:03.33 | ManxPower | Do gas stations in Europe charge money to put air in your tires? |
| 13:03.38 | TheEmperor | daork:i'm just confused... |
| 13:03.42 | daork | TheEmperor: i know |
| 13:03.45 | daork | TheEmperor: so go have a read |
| 13:03.47 | Zeeek | ManxPower no |
| 13:03.54 | Zeeek | in fact it's all self service |
| 13:03.55 | bjohnson | ManxPower: I don't think I understand what you're trying to do .. but I'm sure you'll do a great job of it |
| 13:04.09 | TheEmperor | Zeeek:how did you configure your sendmail to send outgoing mail? |
| 13:04.10 | Zeeek | further you MUST have a bankcard with a stripe or chip |
| 13:04.13 | daork | TheEmperor: learning this kind of stuff from IRC is a bad idea, because you'll get a basic idea of what you need to do to make it work, but you wont understand whats going on |
| 13:04.32 | Zeeek | at night you can't get gas anywhere without a PIN enabled card |
| 13:04.35 | ManxPower | Zeeek: Here self service putting air in your tires still costs 50 cents |
| 13:04.57 | TheEmperor | daork:i just want to get the voicemail emails... :) |
| 13:05.01 | ManxPower | "In America not even the air is free!" |
| 13:05.02 | Zeeek | hmmmm. Maybe it does cost? I never thought of the fact that there could be a coin slot or something? |
| 13:05.09 | daork | TheEmperor: i know |
| 13:05.10 | bjohnson | TheEmperor: you really need to google |
| 13:05.22 | daork | TheEmperor: however, i have a policy that i wont help anyone who just wants something to work |
| 13:05.26 | daork | TheEmperor: they have to understand it |
| 13:05.35 | Zeeek | heh |
| 13:05.44 | daork | TheEmperor: you're free to take advice from others here, but my advice is to go read how smtp works |
| 13:05.47 | Zeeek | well, sendmail is pretty complicated, that's fer sure |
| 13:05.47 | bjohnson | ManxPower: depends on the station .. you can sometimes still find a freebie |
| 13:05.50 | TheEmperor | ok then, i'll go google on smtp :) |
| 13:05.59 | ManxPower | "Give a man a fish and you feed him for a day. Teach a man to fish and you give up your monopoly on fishing." |
| 13:06.09 | daork | TheEmperor: and how smtp servers interact, and how mail gets from your outlook "create new message" window, to my email inbox |
| 13:06.25 | Zeeek | ManxPower of course air can be free here, because the gas costs 4x as much |
| 13:06.46 | bjohnson | TheEmperor: you want a forwarding smtp server system .. it accepts mail from localhost and forwards it to another smtp server |
| 13:06.48 | ManxPower | Zeeek: Yeah, but the high gas taxes pay for public transportation |
| 13:07.09 | Zeeek | ManxPower I'm not sure that's exactly true, but one sort of does compensate for the other :) |
| 13:07.11 | TheEmperor | bjohnson: see, like now, I have my smtp mailserver which is mail.x.com |
| 13:07.16 | bjohnson | TheEmperor: but why forward it, why not just send it directly to it's destination (default setup of most mail servers) |
| 13:07.31 | ManxPower | Zeeek: In the USA the single national passenger rail company almost goes bankrupt every few years and have to beg the govt for a handout. |
| 13:07.55 | TheEmperor | bjohnson: i just don't understand why, but sendmail doesn't seem to send out the email.. |
| 13:07.57 | Zeeek | ManxPower I know, it goes hand in hand with the MY SUV generation, fuck the rest of you |
| 13:08.05 | bjohnson | ManxPower: I begged the government for a handout but they just said to shutup and pay my taxes |
| 13:08.07 | ManxPower | Zeeek: Yes. |
| 13:08.16 | Zeeek | TheEmperor I told you days ago yuou need to get the queue running |
| 13:08.21 | ManxPower | bjohnson: They do that to Amtrak too, but eventually give in. |
| 13:08.26 | TheEmperor | Zeeek:how do I check? |
| 13:08.28 | Zeeek | tryt to look through http://sendmail.org |
| 13:08.36 | Zeeek | I told you ALL THIS DAYS AGO |
| 13:08.39 | bjohnson | ManxPower: maybe I just need to keep asking |
| 13:08.43 | Zeeek | including how to check |
| 13:08.59 | TheEmperor | Zeeek:getting old I guess :) |
| 13:09.07 | bjohnson | Zeeek: ut he just wants someone to do it for him .. can't you help him out? |
| 13:09.08 | ManxPower | Amtrak gets NO official govt support. |
| 13:09.16 | Zeeek | once sendmail is installed, there are files in /etc/mail called .cf |
| 13:09.25 | olivier_ | <TheEmperor> join #sendmail here the wrong chan |
| 13:09.27 | ManxPower | Any time they get money from the govt it's a special thing. |
| 13:09.37 | daork | Zeeek: isnt it frustrating when people try to do this stuff without understanding 'basic' concepts? :-) |
| 13:09.40 | Zeeek | submit.cf controls the sublmit queue |
| 13:09.51 | Zeeek | I know nothing about anything. |
| 13:09.52 | bjohnson | ManxPower: I'd be happy with "unofficial support too |
| 13:09.57 | ManxPower | I don't understand why people still use sendmail |
| 13:10.06 | Zeeek | cause it's there |
| 13:10.13 | TheEmperor | thanks :) |
| 13:10.36 | Zeeek | there is a O'Reilly book about snedmail, two in fact |
| 13:10.36 | ManxPower | There are prostitutes in New Orleans, but that doesn't mean I partake of their services. |
| 13:10.45 | Zeeek | oh? Really? |
| 13:10.56 | olivier_ | lol |
| 13:11.03 | daork | ManxPower: some do |
| 13:11.04 | bjohnson | ManxPower: still the default on some systems, easy to keep using what you're using, it's a good mail server in many ways |
| 13:11.16 | daork | ManxPower: because thats how the like to get sex |
| 13:11.23 | daork | ManxPower: or, its the only option available to them |
| 13:11.33 | davewise | TheEmperor: Try the Sendmail or SNMP how to at www.tldp.org (TheLinuxDDDocProject). If it is still there I think it teaches you how SMTP works by trying it with telnet.... |
| 13:11.34 | newl | Nugget: dude, $41.44 for a quart of oil? Were you high when you bought it? 8) |
| 13:11.36 | Zeeek | sex isn't all it's CRACKed up to be :) |
| 13:11.44 | TheEmperor | davewise: thanks :) |
| 13:12.01 | daork | ManxPower: because their not very good at convincing people that it shouldnt be |
| 13:12.24 | Zeeek | I love sendmail. Once it works |
| 13:12.59 | bjohnson | I swore off sendmail when they screwed up my config by splitting the server into 2 pieces |
| 13:13.05 | Nugget | newl: heh. welcome to the world of german cars |
| 13:13.11 | TheEmperor | Zeeek: if sendmail is running ok, do I need to touch postfix? |
| 13:13.17 | newl | I've never had sendmail _not_ work. Qmail OTOH is one piece of crap I'd rather not administer ever again. |
| 13:13.26 | daork | ManxPower: or they're too lazy to do anything else |
| 13:13.30 | newl | Nugget: :) |
| 13:13.31 | Makenshi | i hate m4 |
| 13:13.31 | Zeeek | TheEmperor nothing to do with postficx |
| 13:13.38 | Zeeek | m4 is cool |
| 13:13.44 | TheEmperor | Zeeek:ok |
| 13:13.49 | Zeeek | once you get all the stuff configured |
| 13:13.50 | daork | now now, dont start an MTA war |
| 13:13.59 | Zeeek | that's way too geeky |
| 13:14.09 | Zeeek | much better to argue about QoS |
| 13:14.14 | newl | daork: sssh, it'd be a good followup to the DB war :) |
| 13:14.20 | daork | haha |
| 13:14.26 | daork | newl: true |
| 13:14.28 | daork | newl: carry on |
| 13:15.04 | daork | tune in tomorrow on #asterisk for "VI vs VIM" and "emacs vs xemacs" and "pico vs nano" |
| 13:15.12 | Nugget | I thought we were going to argue about foreign and domestic cars now. I'm disappointed. |
| 13:15.14 | daork | tune in next week for the grand final! |
| 13:15.14 | Zeeek | emacs is poo |
| 13:15.20 | Makenshi | im a little dissappointed noone mentioned exim :< |
| 13:15.39 | bjohnson | Nugget: funny thing is .. there all foreign to someone |
| 13:15.43 | bjohnson | err they're |
| 13:15.44 | Zeeek | pico/nano rocks - but I only found one difference between the 2 ? |
| 13:15.45 | Nugget | hah, yup |
| 13:15.52 | Makenshi | i heard they're porting windows to emacs |
| 13:16.00 | bjohnson | finally!! |
| 13:16.12 | Wonka | erm |
| 13:16.28 | daork | Makenshi: i've never used exim, but after having a chat to philip hazel one day over beer I was rather inspired to do so |
| 13:16.29 | Wonka | why doesn't someone port emacs to xen? |
| 13:16.49 | daork | Makenshi: but then i had far too much other stuff to do |
| 13:17.24 | daork | Zeeek: did you remember to count licensing? |
| 13:17.28 | daork | Zeeek: :) |
| 13:17.56 | *** join/#asterisk jmacz (~jmacz@63.245.86.163) |
| 13:18.29 | daork | anyway, its something like 1.30am here |
| 13:18.33 | daork | so im going to bed |
| 13:19.03 | daork | let me know who wins "Survivor: Unix MTA" |
| 13:20.23 | newl | meh..high latency to .de..brb |
| 13:20.24 | *** part/#asterisk newl (~newlook@203-59-112-161.dyn.iinet.net.au) |
| 13:20.31 | ManxPower | daork: That would be Postfix |
| 13:20.33 | *** join/#asterisk newl (~newlook@203-59-112-161.dyn.iinet.net.au) |
| 13:20.43 | newl | much better |
| 13:22.46 | *** join/#asterisk VirtualUK (~newkineti@shell3.sea5.speakeasy.net) |
| 13:22.53 | Zeeek | between pico and nano? We have a license for pine AFAIK |
| 13:24.11 | tzafrir | nano actually fixed some of pico's bugs. It still doesn't make it a decent text editor |
| 13:24.16 | VirtualUK | hi folks...I'm trying to get a x100P clone card working with the asterisk@home install but so far no joy, should it have detected the card in some way or do I have to manually insmod the module? |
| 13:24.30 | zoa | virtual no support for that |
| 13:24.55 | mmlj4 | morning, joe |
| 13:24.56 | tzafrir | But I love Nano's homepage |
| 13:25.02 | VirtualUK | zoa: no support for what? |
| 13:25.09 | zoa | for clone cards |
| 13:25.39 | VirtualUK | zoa: ok, if it was a genuine x100p, what would the answer have been? :) |
| 13:25.55 | zoa | it would be buy a tdm card |
| 13:26.16 | VirtualUK | b.o.a....anyone else know? |
| 13:26.19 | zoa | or contact digium for you free hour of support |
| 13:26.26 | tzafrir | Detecting those cards is easy: by PCI IDs. |
| 13:26.43 | tzafrir | modprobe identifies them by that. |
| 13:27.15 | tzafrir | VirtualUK, lsmod |grep zaptel |
| 13:27.28 | tzafrir | should show the loaded zaptel modules |
| 13:27.34 | mmlj4 | lspci, as root |
| 13:27.41 | VirtualUK | I can figure out if the card is alive or not, my question is how much of this should I expect to see automatically done by asterisk@home and how much is supposed to have been done by hand |
| 13:28.17 | tzafrir | VirtualUK, I can't answer you to that. I know pretty much all of it can be automated |
| 13:28.18 | *** join/#asterisk iq (~iq@204-26-74-75.omah.qwest.net) |
| 13:28.41 | tzafrir | (it is on Rapid, and I read that it is on Asterisk@Home as well) |
| 13:29.18 | ManxPower | VirtualUK: use "lspci" and look for this: Tiger Jet Network Inc. Intel 537 |
| 13:29.40 | ManxPower | VirtualUK: if you see that then the motherboard sees the card. |
| 13:29.51 | tzafrir | ManxPower, simpler: modprobe wcfxo /wctdm ... |
| 13:29.58 | ManxPower | It MAY be called something else, depending on your version of lspci |
| 13:30.02 | VirtualUK | ManxPower: Yup, I see it |
| 13:30.10 | ManxPower | tzafrir: Well, that was the next step I was going to suggest. |
| 13:30.22 | ManxPower | VirtualUK: now modprobe wcfxo and see what happens. |
| 13:30.31 | tzafrir | Before that: is it already loaded? |
| 13:30.35 | tzafrir | VirtualUK, lsmod |grep zaptel |
| 13:32.06 | VirtualUK | tza: lsmod | grep zaptel shows "a" module is there "zaptel 178560 4 [ztdummy]" |
| 13:32.53 | VirtualUK | Manx: the modpobe yielded : modprobe wcfxo |
| 13:33.09 | VirtualUK | " /lib/modules/2.4.21-27.0.1.EL/misc/wcfxo.o: init_module: No such device" |
| 13:33.51 | VirtualUK | is this because it's got the dummy driver instead of the proper wcfxo zaptel driver installed? |
| 13:34.39 | ManxPower | ztdummy should not cause that error, but make it not load anyway |
| 13:34.55 | *** join/#asterisk dmccollum (~dmccollum@eycb01-00-cntnga-69-164-245-72.atlaga.adelphia.net) |
| 13:37.45 | VirtualUK | Manx: ok, I've unloaded ztdummy, when I try to do an insmod wcfxo I get the following blurb (any idea where I set the params for the card?) |
| 13:37.55 | VirtualUK | "Using /lib/modules/2.4.21-27.0.1.EL/misc/wcfxo.o |
| 13:38.06 | VirtualUK | "/lib/modules/2.4.21-27.0.1.EL/misc/wcfxo.o: init_module: No such device" |
| 13:38.16 | VirtualUK | "Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters." |
| 13:38.27 | tzafrir | I guess I'll pitch genzaptelconf again |
| 13:38.35 | *** join/#asterisk bkw_ (~bkw_@bkw.developer.and.friend.of.asterisk) |
| 13:38.35 | *** mode/#asterisk [+o bkw_] by ChanServ |
| 13:38.43 | tzafrir | grab it from http://tzafrir.org.il/genzaptelconf |
| 13:38.46 | VirtualUK | tza: k, I'll give that a whiz |
| 13:39.14 | VirtualUK | tza: 404 |
| 13:39.37 | *** part/#asterisk Zeeek (~icechat5@Zeeek.sustaining.supporter.pdpc) |
| 13:40.04 | dmccollum | Morning everyone :) |
| 13:40.04 | VirtualUK | (just did a straight cut'n'paste) |
| 13:40.10 | tzafrir | VirtualUK, however you should first get modprobe to succeed |
| 13:41.12 | VirtualUK | tza: the link isn't correct, 404 not found |
| 13:42.02 | tzafrir | it is OK now |
| 13:42.15 | tzafrir | VirtualUK, however you should first solve that modprobe issue |
| 13:42.29 | VirtualUK | tza: thanks |
| 13:43.03 | tzafrir | VirtualUK, don't copy&paste: wget |
| 13:44.58 | *** join/#asterisk flot (~flot@user191.hovrino.net) |
| 13:45.26 | *** join/#asterisk dimas (~ds@84.53.210.46) |
| 13:46.40 | *** join/#asterisk fedor (~fbond@office.tura.ru) |
| 13:46.55 | VirtualUK | tza: will do, just trying to figure out how to get the wcfxo loaded properly first :) wondering how to figuer out what/where the params should be for this card |
| 13:47.06 | *** join/#asterisk anthm (~anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 13:47.06 | *** mode/#asterisk [+o anthm] by ChanServ |
| 13:47.43 | fedor | hi peopls, can could help me? |
| 13:47.50 | fedor | i`d try to use * together with Cisco AS53000. |
| 13:47.57 | fedor | If i make changes in configs dtmfmode=inband, than * say fuck and report error "g.729 codec dont support inband mode try to use rfc3328 codec instead" |
| 13:48.05 | fedor | if i use rfc3328 and try send sample dtmf from * to as5300 i listen unfinishable sound. Whats wrong? |
| 13:48.39 | tzafrir | VirtualUK, no params are needed. Those params were needed for older ISA cards. PCI should be plug&play |
| 13:48.51 | bkw_ | that should read rfc2833 |
| 13:49.02 | bkw_ | g729 can't do inband |
| 13:49.12 | bkw_ | and if its sending more than one.. I can guess why.. get an rtp debug |
| 13:49.15 | VirtualUK | tza: that's what I thought, so I wonder wtf the modprobe is complaining about then :( |
| 13:50.55 | tzafrir | VirtualUK, modprobe asked the kernel to load a module. The kernel issued the init procedure of the module. It has failed. modprobe only knows that it has failed and gives you a text that it hopes would be useful. |
| 13:51.20 | *** join/#asterisk moy (~kvirc@201.129.134.130) |
| 13:51.20 | tzafrir | VirtualUK, any useful text in the kernel logs? Are you working in the console? |
| 13:51.49 | tzafrir | dmesg |tail will also show you the recent messages from the kernel (don't paste here) |
| 13:51.57 | tzafrir | !pastebin |
| 13:52.17 | tzafrir | ~pastebin |
| 13:52.18 | jbot | pastebin is probably a place to paste your stuff without flooding the channel - try http://pastebin.ca |
| 13:52.19 | VirtualUK | tza: :) yeah.....nada showing up |
| 13:52.36 | Aze` | I'm trying Zaphfc with 1 billion HFC in bri_cpe euroisdn, why i dont recive call but i can make call ? asterisk dont recive any from zaptel ... why ? any idea ? |
| 13:53.56 | tzafrir | Aze`, I'm not so familiar with that, but how exactly do you see that Asterisk gets nothing from Zaptel? |
| 13:54.29 | *** join/#asterisk leandro_pt (~leandro@bl6-114-199.dsl.telepac.pt) |
| 13:55.59 | *** join/#asterisk durex (~ironman@200.101.109.138) |
| 13:58.16 | Aze` | tail -f /var/log/asterisk/full |
| 13:58.23 | Aze` | and CLI console |
| 13:58.39 | fedor | bkw_: and if its sending more than one.. I can guess why. guess what? |
| 13:59.11 | Aze` | How solution with 3 hfc-s isdn billion card ? |
| 13:59.25 | VirtualUK | tza: I think I know what the problem is :) I'm trying to get this all up and running on a VIA EPIA board with a C3 proc, which has problems claiming to be a i686 when it needs things to be compiled as i586 due to a missing instruction....I'm currently recompiling to see if it fixes things :) |
| 14:00.01 | tzanger | VirtualUK: yeah dn't optimize :-) |
| 14:05.37 | *** join/#asterisk danh-tbcnet (danh-tbcne@66-225-33-8.static.tbc.net) |
| 14:06.16 | *** join/#asterisk Gand_DJ (~gandalf@ptr-207-54-104-24.ptr.terago.ca) |
| 14:08.07 | ManxPower | ~google site:lists.digium.com EPIA |
| 14:08.47 | *** part/#asterisk xai (~pasta@cpe-70-112-17-10.austin.res.rr.com) |
| 14:10.20 | *** join/#asterisk kb1_kanobe (~krisbouti@h24-207-80-55.cst.dccnet.com) |
| 14:10.23 | kb1_kanobe | g'day all. |
| 14:11.28 | *** join/#asterisk mutilator (~animenodv@65.111.201.79) |
| 14:11.58 | *** join/#asterisk loud (~ariel@test254.telconinjas.net) |
| 14:13.25 | kb1_kanobe | is anyone aware of problems with the new jitterbuffer in the current cvs head? I seem to get about 30 to 45 seconds of audio and then it drops out, but the line stays up. This seems to correspond with the timestamps going berzerk in the iax debug? |
| 14:14.04 | tzanger | kb1_kanobe: turn off trunking |
| 14:14.09 | tzanger | it's a known issue |
| 14:14.25 | kb1_kanobe | I thought the new JB with trunktimestamps fixed that? |
| 14:14.48 | tzanger | also there's a problem with having the new jb on on the origination side and sending dtmf... you lose received audio for a number of seconds |
| 14:14.54 | tzanger | kb1_kanobe: supposed to but it seems to have broken it |
| 14:14.55 | *** join/#asterisk ruied (~a@213.22.166.175) |
| 14:15.24 | kb1_kanobe | Ewww.... ok, thanks. |
| 14:16.16 | zoa | kb |
| 14:16.21 | zoa | we have a tool to calculate it |
| 14:16.25 | *** join/#asterisk litnimax (~chatzilla@195.138.118.214) |
| 14:16.34 | litnimax | AGI(dtl_tell_balance_english) ne rabotaet a vot AGI(dtl_tell_bal) work - do u beleive me? |
| 14:16.45 | RoyK | http://karlsbakk.net/rart.gif <- stare closely at this pic, and you'll see a giraffe |
| 14:16.52 | kb1_kanobe | zoa: can you poing me there? |
| 14:16.54 | litnimax | AGI(dtl_tell_balance_english)does not work but AGI(dtl_tell_bal) works - do u beleive me? |
| 14:16.57 | kb1_kanobe | s/poing/point |
| 14:17.02 | zoa | http://www.asteriskguru.com/tool2.php |
| 14:17.07 | kb1_kanobe | ta |
| 14:18.40 | *** join/#asterisk Mother_ (~m@53.Red-217-126-93.pooles.rima-tde.net) |
| 14:18.43 | Mother_ | greetings |
| 14:19.56 | *** join/#asterisk scardinal (~supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 14:20.23 | leandro_pt | hi.. i compiled fcpci under debian 2.4.27-1 numerous times but am having trouble compiling it with kernel 2-4-27-2.. has anyone done that sucessfully? |
| 14:22.06 | Cronus-secut | Is it possible to send a signal every x seconds over a ISDN line from either asterisk or a shell script? |
| 14:22.08 | *** join/#asterisk cmk (~cmk_@p54A3BD8F.dip.t-dialin.net) |
| 14:24.12 | kb1_kanobe | zoa: why does the calculator say only 50pps for both iax2 and iax2 trunked, regardless of the number of channels? shouldn't it be 50*numchannels for non trunked? |
| 14:24.21 | *** join/#asterisk chaoscon (~ph33r@chaoscon.user) |
| 14:25.10 | zoa | kb1_kanobe: there is a bug there, the guy will upload a fix in a few minutes |
| 14:25.12 | Mother_ | signal....what signal...? |
| 14:25.23 | zoa | i noticed that this morning too |
| 14:25.23 | kb1_kanobe | zoa: ok, thanks. :-) |
| 14:26.03 | leandro_pt | nevermind.. i found out i was missing -DMODVERSIONS in the makefile |
| 14:26.33 | *** join/#asterisk scardinal (~supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 14:26.40 | Gand_DJ | Hrm, if cisco is buying sipura, I wonder if they will then start selling non-locked ata boxes for end users |
| 14:27.02 | TheEmperor | is there a feature in extensions.conf that you can put in so that one can put the calling party on hold, call another extension, that extension picks up, and then you transfer the calling party to that extension? |
| 14:27.31 | kb1_kanobe | Any ideas how I can get ping to generate 1150pps? 'ping -f' seems to cap out around 300, presumably because it's waiting for the reply to the previous frame...? |
| 14:27.37 | ManxPower | TheEmperor: That's usually handled by the phone you are using, not Asterisk. |
| 14:27.55 | docelm0 | Grand, I spoke to my rep at linksys he said they have no intentions to sell these to end users |
| 14:28.00 | TheEmperor | ManxPower:does it matter if I am using SIP or IAX2 protocol? |
| 14:28.10 | ManxPower | TheEmperor: It matters what phone you are using. |
| 14:28.12 | docelm0 | You can only purchase to authorized distributor or reseller like me. |
| 14:28.24 | ManxPower | TheEmperor: What ARE you trying to do? Make two calls and bridge them togather? |
| 14:28.30 | ManxPower | or do a three-way call? |
| 14:28.31 | TheEmperor | ManxPower: but in extensions.conf, you put in a t to allow the called user to transfer the call? |
| 14:28.35 | ManxPower | or a conference call? |
| 14:28.55 | kb1_kanobe | The new jitterbuffer and PLC - is it still only applicable to the speex codec? |
| 14:28.57 | ManxPower | TheEmperor: t is an ugly hack for phones that are too stupid to have their own transfer feature. Don't use it unless you have to. |
| 14:29.24 | Gand_DJ | docelm0, end users can't "legally" purchase linksys ata's from resellers.. as they are sold for voip providers only. |
| 14:29.29 | syle | docelm0 : which basically means any joe bloe can say i run a voip service gimme an ata |
| 14:29.37 | TheEmperor | ManxPower: well, say someone calls and an operator picks up, the caller wants to speak to extension 10, the operator puts the calling party on hold, calls the extension and asks if they want to speak to x, then if they say ok, the call is transferred |
| 14:30.21 | ManxPower | TheEmperor: What damn phone are you using? |
| 14:30.42 | ManxPower | TheEmperor: The correct term for that is "supervised/consultative transfer" |
| 14:30.49 | TheEmperor | ManxPower: trying out the phones with the PA168S chip that can use IAX2 protocol |
| 14:30.56 | TheEmperor | MaxnPower: ah.. |
| 14:31.02 | docelm0 | Grand yes.. If you purchase a block of minutes from me you can. When I spoke to him he said if you sell them w/ minutes then you can sell them to end users "unlocked" |
| 14:31.04 | ManxPower | TheEmperor: does the phone support supervised transfer? |
| 14:31.28 | TheEmperor | ManxPower: I think only on SIP protocol not IAX2 protocol...although putting t in will work to transfer a call |
| 14:31.30 | Gand_DJ | docelm0, what voip service do you run that offers minutes? |
| 14:31.45 | Gand_DJ | or did you team up with a provider such as broadvoice? |
| 14:31.58 | ManxPower | TheEmperor: Asterisk 1.0.x does not support supervised transfer using t/T. CVS-HEAD supports supervised transfer using t/T (or is it w/W), but I keep hearing it doesn't work. |
| 14:32.05 | TheEmperor | ManxPower:but I can't get the supervised/consultative transfer as you say... |
| 14:32.07 | ManxPower | TheEmperor: Get phones that support supervised transfer. |
| 14:32.18 | TheEmperor | ManxPower:what phones do you recommend? |
| 14:32.31 | ManxPower | Polycom IP 500 or SIPura SPA-841. |
| 14:32.51 | *** join/#asterisk cjk (~cjk@80.92.64.103) |
| 14:32.51 | ManxPower | I've not done supervised transfers with the SIPura but I think the docs says they support it. |
| 14:32.51 | TheEmperor | ok thanks |
| 14:32.59 | omadon | Can anybody help me with zaphfc problem and i4l question |
| 14:33.00 | TheEmperor | i see.. |
| 14:33.09 | ManxPower | We do supervised transfers all the time with Polycom 300 and 500 phones |
| 14:33.12 | cjk | anyone here who know how to solve the ilbc grandstream problem in the newest firmware? |
| 14:34.05 | johnnyb | cjk: what's the problem? |
| 14:34.55 | tzanger | kb1_kanobe: that's just for generic PLC |
| 14:35.25 | *** join/#asterisk Rick_Hunter (~rhunter@03-048.008.popsite.net) |
| 14:35.59 | kb1_kanobe | tzanger: I assume newjb would make use of it for codecs like g726 though? |
| 14:36.16 | kb1_kanobe | put another way, my pops & clics just went away. :-) |
| 14:36.23 | flot | hi. why h323.conf in [general] section --- command allow=xxx is work, in [userxxx] section --- allow=xx is NOT work ? |
| 14:37.01 | *** join/#asterisk Zeeek (~Zeeek@Zeeek.sustaining.supporter.pdpc) |
| 14:37.10 | *** join/#asterisk riksta (riksta@212.85.228.176) |
| 14:37.54 | cjk | johnnyb: metallic tones.... cant hear any word |
| 14:38.05 | *** join/#asterisk jsolares (~jsolares@200.30.141.85) |
| 14:38.39 | tzanger | kb1_kanobe: yes |
| 14:39.04 | tzanger | it's a generic interpolating PLC algorithm |
| 14:39.20 | Derkommissar | bkw_, are you around ? |
| 14:39.22 | kb1_kanobe | tzanger: good, tks. |
| 14:39.53 | *** join/#asterisk Lee__ (~lee@ool-44c26e7d.dyn.optonline.net) |
| 14:41.39 | *** join/#asterisk ciphergroup (~ciphergro@216.90.111.178) |
| 14:42.17 | Mother_ | ergh |
| 14:42.22 | *** join/#asterisk Juxt (~Juxt@64.135.20.202) |
| 14:42.27 | Juxt | good morning |
| 14:42.29 | Juxt | i have this question |
| 14:42.32 | Juxt | i am using firefly |
| 14:42.42 | Juxt | and i need to have an ability to transfer calls |
| 14:43.21 | bjohnson | try Tt in the dial command |
| 14:43.42 | *** join/#asterisk tld (~tld@196.80-202-89.nextgentel.com) |
| 14:43.59 | tld | Anyone have a SIP firmware image for a Cisco 7960, or is that off-topic? |
| 14:44.38 | Juxt | bjohnson: huh? |
| 14:44.59 | *** join/#asterisk rjburkh (~rjburkh@dialup-4.233.124.194.Dial1.LosAngeles1.Level3.net) |
| 14:46.20 | ManxPower | tld: that's copyright violation, just like asking for a CD key for MS office. |
| 14:46.21 | jalsot | hi |
| 14:46.34 | riksta | tld go buy a support contract |
| 14:46.51 | ManxPower | riksta: I heard that Cisco is cracking down on that lookhole too. |
| 14:46.59 | tld | Sorry, didn't mean to ask for warez. I thought getting a firmware image would be legal as long as I have a license for it. |
| 14:47.03 | ManxPower | Soon you will have to spend the $120 for the firmware. |
| 14:47.08 | riksta | ManxPower... i wouldn't know how then can possibly tell though |
| 14:47.17 | riksta | unless you give it out to people |
| 14:47.19 | ManxPower | tld: Cisco does not provide SIP firmware for free. |
| 14:47.20 | tld | riksta: Don't you have to buy that with a new device? |
| 14:47.41 | riksta | tld the devices liscence is non-transferable |
| 14:47.50 | ManxPower | riksta: they could just disable software downloads for people with $8/year support contract. |
| 14:47.59 | jalsot | any idea how to handle bouncing iax clients between * servers? [once they are registered under 1st, next time 2nd and I would like to make it reacheable everytime from both] |
| 14:48.07 | riksta | ManxPower oh...i see what you mean now, wasn't reading properly |
| 14:48.11 | tld | riksta: Hmm, didn't know that. Thanks. |
| 14:48.12 | kb1_kanobe | bbl |
| 14:48.13 | *** part/#asterisk kb1_kanobe (~krisbouti@h24-207-80-55.cst.dccnet.com) |
| 14:48.29 | festr_ | anyone have done testing jitterbuffer with g729 codec? |
| 14:48.47 | ManxPower | tld: Cisco phones require significant additional purchases. Polycoms are just as good, IMHO, but have everything you need, except PoE adapter. |
| 14:48.52 | tld | riksta: Not sure how that would stant up in the Norwegian legal system, but that's definitively off topic. I'll just get a new license to be on the safe side. |
| 14:48.58 | festr_ | is jitter buffer in HEAD compatible with g729 codec? (intel) |
| 14:48.59 | Juxt | festr_: i have it working fine |
| 14:49.04 | festr_ | Juxt: with g729? |
| 14:49.07 | Juxt | yes |
| 14:49.11 | festr_ | Juxt: which g729? |
| 14:49.13 | riksta | tld there's more info on the wiki (afaik) |
| 14:49.22 | tld | ManxPower: Difference is, I already have the Cisco 7960. Polycom would be an additional purchase, and I'm well... between jobs. |
| 14:49.24 | festr_ | Juxt: digium or intel? |
| 14:49.28 | tld | riksta: thanks. |
| 14:49.31 | Juxt | right now using the "unlicensed" one for testing |
| 14:49.31 | tld | thanks everyone actually. |
| 14:49.42 | festr_ | Juxt: so intels one? |
| 14:49.46 | Juxt | yes |
| 14:49.56 | Juxt | i had to run the debug version |
| 14:49.57 | *** join/#asterisk lilwookie (~zoidmeste@modemcable215.87-81-70.mc.videotron.ca) |
| 14:50.00 | tld | ManxPower: You've mentioned Polycom before... Any recommended retailer that'll ship internationally? (or recommended one in general?) |
| 14:50.01 | Juxt | cause athlon version kept crashing |
| 14:50.03 | festr_ | Juxt: so no cahnges to source of this codec is needed with cooperating with jitter buffer? |
| 14:50.16 | Juxt | i didn't make any changes |
| 14:50.24 | Juxt | it seemed to just work right out of the box |
| 14:50.47 | Juxt | i didn't major jitter issues but those went away |
| 14:51.20 | festr_ | Juxt: do you see some changes to quality on laggy networks with new jitter compares to the STABLE asterisk? |
| 14:51.34 | tld | Hmm, and anyone know what spesifically I need to ask for to get a SIP firmware and license for the phone? The only thing I can find is the smartnet agreements, but I suppose those just cover the software upgrades, not the license to the software itself? |
| 14:51.44 | Juxt | i never ran stable asterisk, i needed realtime functionality so i ran head from the get-go |
| 14:52.25 | festr_ | Juxt: ok thanks for reply :) |
| 14:52.40 | festr_ | so anyone here made some compares with stable and head jitter ? |
| 14:52.41 | Juxt | i had some jitter issues with g729 and they went way when i enabled jitter buffer in iax.conf |
| 14:52.44 | *** join/#asterisk MuppetMaster (~MuppetMas@177.Red-213-98-135.pooles.rima-tde.net) |
| 14:52.59 | festr_ | Juxt: yes iax has own jitter in stable too |
| 14:53.12 | festr_ | Juxt: but what i'm affraid, if jitter buffer will work in sip |
| 14:53.23 | Juxt | hmm |
| 14:53.28 | festr_ | with g729 |
| 14:53.38 | festr_ | iax does its own RTP ? |
| 14:53.39 | nestAr | blitzrage: wouldn't i need to use a Global variable for the example you showed last night? otherwise, the variable is attached to the channel, the next channel that comes up has no knowledge of the other channels variables |
| 14:53.46 | Juxt | i have sip phones on g729 but i never had jitter issues there |
| 14:54.01 | Juxt | if i knew how to simulate jitter i'd test it for you |
| 14:54.11 | mutilator | put it on wireless |
| 14:54.12 | *** join/#asterisk IronHelix (~irc@ool-182c3fe9.dyn.optonline.net) |
| 14:54.17 | Juxt | i don't have wireless |
| 14:54.22 | Juxt | i have gig-e thru the office |
| 14:54.35 | *** part/#asterisk MuppetMaster (~MuppetMas@177.Red-213-98-135.pooles.rima-tde.net) |
| 14:54.46 | mutilator | *shrug* |
| 14:54.55 | mutilator | those $15 usb wifi things are handy |
| 14:55.01 | mutilator | might want one just ot have it |
| 14:55.02 | mutilator | :P |
| 14:55.09 | Juxt | link me |
| 14:55.31 | Wonka | might want one that works under linux, though |
| 14:55.47 | mutilator | most do, they run off prism |
| 14:55.50 | mutilator | least that i've seen |
| 14:56.05 | Wonka | cool |
| 14:56.18 | Wonka | had one plugged some days ago - no way |
| 14:56.31 | mutilator | few that i have do |
| 14:56.34 | Juxt | what are you people talking about, wha ti the wireless thing? |
| 14:56.34 | mutilator | zonet |
| 14:56.54 | mutilator | www.zonet.com |
| 14:56.59 | mutilator | maybe? |
| 14:57.00 | mutilator | heh |
| 14:57.09 | Wonka | i do have prism_usb.ko, orinoco_usb.ko - both didn't want it |
| 14:58.03 | mutilator | http://www.zonetusa.com/DispProduct.asp?ProductID=152 |
| 14:58.06 | festr_ | Juxt: on switched networks is rarely jitter, if you can simulate >90ms latency you get jitter |
| 14:58.06 | mutilator | i use those |
| 14:58.27 | festr_ | no one here test jitter for SIP? |
| 14:58.33 | festr_ | in HEAD? |
| 14:58.35 | festr_ | :) |
| 14:58.45 | Juxt | >90ms you need TCP/IP over barbed wire |
| 14:58.54 | festr_ | what is barbed |
| 14:59.17 | Wonka | "Stacheldraht" |
| 14:59.22 | festr_ | lol |
| 14:59.24 | festr_ | wtf |
| 14:59.31 | mutilator | or an old 10mbit hub with tons of stuff going on |
| 14:59.31 | Juxt | http://www.rushcounty.org/BarbedWireMuseum/BWhistory.htm |
| 14:59.33 | Wonka | meaning "really bad cable" here |
| 14:59.41 | festr_ | mutilator: off course :) |
| 14:59.46 | festr_ | Wonka: aha |
| 15:00.14 | festr_ | i'm thinikg about some generator of call which simulate some jitter |
| 15:00.28 | Juxt | maybe ping flood the box? |
| 15:00.29 | festr_ | ach bad english sorry :) |
| 15:00.34 | flot | hi. why h323.conf in [general] section --- command allow=xxx, dtmfmode=xxx is work, in [userxxx] section --- allow=xx, dtmfmode=xx is NOT work ? |
| 15:00.36 | festr_ | Juxt: too big jitter |
| 15:00.53 | festr_ | flot: because of implementation |
| 15:00.57 | festr_ | flot: rewrite it |
| 15:01.30 | festr_ | Juxt: you need generate jitter every second for example |
| 15:01.33 | JerJer | because everything in h.323 is considered global |
| 15:01.41 | JerJer | they don't give a shit about each user |
| 15:02.02 | festr_ | JerJer: cisco gw can do this per user |
| 15:03.43 | festr_ | btw, is in CVS HEAD jitter for sip? |
| 15:03.48 | festr_ | or only for iax |
| 15:03.53 | tld | I'm starting to get the hang of dialplans, but they seem limited for large or dynamic setups. Can I push this into a database, or possibly use programmable logic? |
| 15:04.06 | *** join/#asterisk Manipura (~chatzilla@dsl-ep-209-115-250-i114-cgy.nucleus.com) |
| 15:04.07 | ionix | macro |
| 15:04.32 | flot | festr_: What to rewrite? reload I gave |
| 15:04.45 | festr_ | flot: source code |
| 15:05.03 | flot | festr_: :) |
| 15:05.08 | festr_ | flot: :) |
| 15:05.23 | festr_ | flot: h.323 is too complex protocol |
| 15:05.30 | festr_ | flot: it is impossible to get in in few days |
| 15:07.55 | festr_ | is it thrue, that jitter for SIP implementation is in progress? (voip-info.org) |
| 15:08.50 | flot | festr_: I do not understand at what here the report, misoperation with h323.conf is available simply |
| 15:08.58 | oej | festr: The jitter buffer is only for IAX2 in head. There are patches for SIP in the bug tracker. |
| 15:09.50 | oej | tld: Depends on the version. In stable, check res_config that reads config from an SQL database. In cvs head, check realtime, that can read config from database as well as use the database in real time. |
| 15:09.50 | festr_ | oej: what stability is those patches? i'm searching bug tracker right now |
| 15:10.04 | oej | festr_: I don't recommend them for other than testing. |
| 15:10.18 | oej | Do *not* put CVS head on a production server, never, ever. |
| 15:10.52 | festr_ | oej: i'm in very serious trouble with commercial clients on sip with little jittery lines (50-150ms) |
| 15:11.08 | festr_ | oej: i'm using testing on production servers, but this feature is minor for us |
| 15:11.20 | festr_ | oej: eeee stable on prod.servers |
| 15:11.29 | festr_ | oej: :) |
| 15:12.04 | festr_ | oej: exist some proxy which can do jitter transparently? |
| 15:12.14 | festr_ | oej: i think no one exist |
| 15:12.15 | oej | festr_: I think a lot of us wait anxiously for the SIP jitter buffer! |
| 15:12.29 | oej | festr_: I do not know anything about such a product |
| 15:12.37 | festr_ | oej: ok thank you |
| 15:13.23 | flot | festr_: To correct source codes I love, a current it is necessary to spend week to understand ideology of construction of the program...: ( |
| 15:14.38 | *** join/#asterisk RoyK (~roy@host-81-191-147-248.bluecom.no) |
| 15:15.06 | *** join/#asterisk zeedo (~zeedo@www.bsrf.org.uk) |
| 15:16.06 | *** join/#asterisk PTG1234 (PTG123@ip68-106-24-139.ph.ph.cox.net) |
| 15:16.43 | festr_ | oej: one more think, you are talking about jitter buffer in HEAD. i though, that jitter for iax2 is in stable too? |
| 15:17.15 | oej | Yes, there's a jitterbuffer for IAX2 in stable (check iax.conf). The one in CVS head is a different one, very much improved. |
| 15:17.30 | festr_ | flot: well, ideology of h323 library is too complex i think :) this is not work for one week :) |
| 15:17.35 | tzanger | very much improved but it has issues |
| 15:17.44 | festr_ | oej: aha ok |
| 15:17.51 | festr_ | tzanger: what issues |
| 15:18.06 | tzanger | DTMF mostly |
| 15:18.06 | *** join/#asterisk Seyr (~Seyr@rrcs-24-227-133-226.sw.biz.rr.com) |
| 15:18.12 | Seyr | heyas people |
| 15:18.22 | oej | tzanger: Yes, cvs head is shaky... |
| 15:18.49 | Seyr | i have a new asterisk box at a datacenter behind a SonicWall firewall. I can connect to it just fine, but I do not hear any audio. Any clues? |
| 15:18.56 | tzanger | oej: well HEAD's fine on its own, but hte jb is a little shaky |
| 15:19.24 | festr_ | hmm.. it looks that jitter buffer is still long way :) |
| 15:19.31 | oej | Seyr: You have a good bullshit filter in your firewall :-) |
| 15:19.39 | Seyr | :-) |
| 15:19.42 | festr_ | haha |
| 15:19.59 | festr_ | Seyr: allow any communitation to this bo |
| 15:20.01 | tzanger | no it's not a long way |
| 15:20.01 | festr_ | Seyr: x |
| 15:20.07 | tzanger | there's just some little things that need to be worked on |
| 15:20.11 | flot | festr_: IMHO here a problem not in most h323, and in reading h323.conf |
| 15:20.21 | festr_ | flot: yes it is |
| 15:20.25 | Seyr | festr_: ?? |
| 15:20.39 | Seyr | i have 5060 and 10000:20000 being forwarded to it |
| 15:20.44 | festr_ | Seyr: allow any communitation on firewall to your asterisk |
| 15:21.00 | Seyr | thats a pretty big security hole there festr_ |
| 15:21.03 | Seyr | if you allow any |
| 15:21.03 | Gand_DJ | anyone using Debian? |
| 15:21.20 | Seyr | Does anyone know the exact ports? |
| 15:21.25 | devel | Gand_DJ, sure. |
| 15:21.28 | zoa | oej |
| 15:21.32 | zoa | the jitter buffer is almost there |
| 15:21.35 | zoa | euh not oej |
| 15:21.37 | Gand_DJ | devel, you using hte latest? (V3.0r5)? |
| 15:21.38 | zoa | festr |
| 15:21.44 | festr_ | zoa: no |
| 15:21.47 | festr_ | zoa: yes |
| 15:21.54 | oej | zoa: To be oej or not oej - that's the question |
| 15:21.55 | zoa | oej, did you read the nat tutorial ? |
| 15:21.55 | devel | Gand_DJ, using unstable |
| 15:22.00 | festr_ | Seyr: do that as start point |
| 15:22.07 | oej | zoa: Been occupied, but it's on my list |
| 15:22.11 | zoa | k |
| 15:22.16 | oej | Trying to understand app_sms |
| 15:22.16 | Gand_DJ | devel, ok... have you ever tested the uae program that comes with it (Amiga emulator) |
| 15:22.18 | zoa | you also use a telephone book as todo list ? |
| 15:22.26 | devel | Gand_DJ, i have not |
| 15:22.29 | Gand_DJ | ok |
| 15:23.05 | festr_ | Seyr: produce your firewall some logs? |
| 15:23.09 | jalsot | does anybody have any idea how to handle moving iax clients between * servers? [once they are registered under 1st, next time 2nd and I would like to make it reacheable everytime from both] |
| 15:23.13 | *** join/#asterisk Markie (Mark@208.247.106.165) |
| 15:23.15 | festr_ | Seyr: check your logs what is dropped |
| 15:23.20 | Markie | anyone here from teliax? |
| 15:23.31 | Markie | i moved over to them last night...pretty dang happy |
| 15:24.02 | Markie | in fact everyone i'd called today says it's the best quality calls they ever had with me since i've had broadvoice for over a year |
| 15:24.41 | flot | festr_: I have problem: * and cisco AS5300 in dtmfmode=rfc2833 is not work. At sending DTMF from * aside 5300 tone is accepted by duration of 30 seconds. |
| 15:25.20 | festr_ | flot: change it to string |
| 15:25.28 | festr_ | flot: or some h245 |
| 15:25.42 | Markie | besides the whoel nat-friendly thing, is iax really much better than sip????? |
| 15:25.44 | RoyK | anyone that knows some good predictive dialer algorithms? |
| 15:25.53 | Markie | i mean i actually like a speerate control and data channel of sip |
| 15:25.59 | RoyK | Markie: I beleive it's better on asterisk. |
| 15:26.15 | *** join/#asterisk carbon60 (~adam@gw.techsupport.ca) |
| 15:26.15 | RoyK | but sip has more stuff |
| 15:26.16 | Markie | RoyK: as in better supported? |
| 15:26.25 | jalsot | does anybody know how to identify incoming SIP call from cisco as5300 in asterisk? What I found on the web about the topic doesn't work :( |
| 15:26.30 | RoyK | confort noise, for instance, although that's not supported in iax |
| 15:26.38 | Markie | hrmm....good point didnt think of that sip == better protocol, worse code...interesting |
| 15:27.03 | flot | festr_: It is possible on more in detail? |
| 15:27.05 | RoyK | the chan_sip code isn't really that good |
| 15:27.07 | carbon60 | Has anyone used AstTAPI that can share some opinion/comments? |
| 15:27.08 | RoyK | or rtp.c |
| 15:27.16 | RoyK | rtp in asterisk sucks |
| 15:27.17 | Markie | cause i'm not behind a nat or firewall or anything..and i can do SIP or IAX with teliax |
| 15:27.24 | festr_ | flot: change protocol for dtmf on cisco and oh323.conf to h245 or string |
| 15:27.25 | Markie | ok...nice to know |
| 15:27.55 | RoyK | flot: rfc2833 should work like a dream |
| 15:28.00 | RoyK | try sip debug |
| 15:28.00 | PTG1234 | sip is better |
| 15:28.09 | festr_ | RoyK: this is h323 |
| 15:28.19 | festr_ | RoyK: not sip |
| 15:28.29 | RoyK | festr_: shit. h323 support in asterisk is - well - limited |
| 15:28.36 | RoyK | or - well - it sucks outright |
| 15:28.54 | festr_ | RoyK: limited with global options in h323.conf |
| 15:29.22 | festr_ | RoyK: rtp is much better because of openh323 which implement jitter buffer, RTCP |
| 15:29.28 | Markie | i guess i cna always swithc back and forth with teliax which is nice |
| 15:29.31 | RoyK | I know |
| 15:29.34 | Markie | so i can try out both and see for myself |
| 15:29.36 | RoyK | chan_h323 is no good |
| 15:29.56 | RoyK | and they can't implement openh323 in asterisk since it's GPL and asterisk is dual-licensed |
| 15:29.59 | flot | festr_: i use H323.conf (not oh323.conf), dtmfmode=inband is work, but inband is not supported in g729 |
| 15:30.09 | olivier_ | i've try chan_h323 and chan_oh323. According to me oh323 isbest with option |
| 15:30.20 | olivier_ | with more option |
| 15:30.22 | festr_ | flot: switch to h323 inaccess networks |
| 15:31.25 | flot | what is inaccess? |
| 15:33.34 | JerJer | flot: then use dtmfmode=rfc2833 |
| 15:33.50 | JerJer | olivier_: too bad asterisk-oh323 only deals with a very small amount of channels |
| 15:33.55 | JerJer | and burns the hell out of CPU usage |
| 15:34.24 | JerJer | the people that have problems with chan_h323 simply do not follow the README |
| 15:34.32 | flot | RoyK: I could not collect oh323 for * from CVS. I have collected h323 with the last pwlib and h323 |
| 15:34.45 | JerJer | 99.999% of the problems I have dealt with come from simple bullshit problems defined in the README |
| 15:34.55 | tzanger | JerJer: I'd say 80% |
| 15:35.00 | tzanger | it's the 80/20 rule |
| 15:35.02 | JerJer | oh yeah and asterisk-oh323 will not compile on cvs -head |
| 15:35.20 | *** join/#asterisk wooden (~wooden@pD9EC47AD.dip0.t-ipconnect.de) |
| 15:35.21 | wooden | hi |
| 15:35.49 | wooden | what ports is needed to forward by using asterisk after a router? |
| 15:36.00 | wooden | ive forwarded 5060 and 10000-11000 |
| 15:36.10 | wooden | i can receive calls but cant call out |
| 15:36.17 | flot | JerJer: dtmfmode=rfc283 is not work from * to cisco 5300 :(( |
| 15:36.20 | wooden | but the TEST-Service means all nice |
| 15:36.24 | wooden | from my provider |
| 15:36.43 | *** join/#asterisk miller7 (~none@adsl49-static-gw1.access.acn.gr) |
| 15:37.13 | wooden | the asterisk was working but now not :( |
| 15:38.23 | bjohnson | wooden: 1. check your type=user settings 2. check your codecs |
| 15:38.31 | bjohnson | 3. get rid of nat |
| 15:38.37 | bjohnson | 4. correct your port forwards |
| 15:38.46 | wooden | what ports must be forwarded? |
| 15:39.04 | *** join/#asterisk Uther_P (~uther_p@66.180.120.83) |
| 15:39.11 | bjohnson | 5. use iax instead of sip |
| 15:39.25 | bjohnson | one or more of those is your problem |
| 15:39.46 | Manipura | Using IAX will eliminate a lot of problems |
| 15:40.07 | bjohnson | look in the conf files for the ports .. or look in the wiki (it's likely there too) |
| 15:40.10 | bjohnson | ~docs |
| 15:40.13 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 15:42.50 | zoa | wooden |
| 15:43.04 | zoa | read thhis draft |
| 15:43.08 | zoa | it might help you: http://www.asteriskguru.com/natut.php |
| 15:43.24 | wooden | ok, thx |
| 15:43.45 | *** join/#asterisk MattH (~matth@noc-wireless.chilitech.net) |
| 15:44.06 | *** join/#asterisk pjz (~pj@place.org) |
| 15:44.19 | MattH | Hi... can someone clear up confusion I'm having about G729 and G723? When do I need a license, and when don't I? |
| 15:44.34 | mutilator | always |
| 15:44.34 | zoa | pass thru no license |
| 15:44.36 | mutilator | and always |
| 15:44.40 | zoa | otherwise a license |
| 15:44.47 | zoa | and g723 has no license |
| 15:44.56 | zoa | and that passthru no license thing doesnt seem to work for me |
| 15:45.02 | jalsot | does anybody know what does this mean? chan_iax2.c:7144 socket_read: Immediately destroying 16386, having received INVAL |
| 15:45.29 | jalsot | this is on a trunked IAX2 peer |
| 15:45.46 | zoa | never seen it |
| 15:46.05 | jalsot | if I turn trunking off, everything works fine |
| 15:46.23 | zoa | i do a lot of trunking |
| 15:46.51 | jalsot | the initiating * has ztdummy, the other end has T110E |
| 15:46.54 | zoa | never seen it |
| 15:47.12 | jalsot | if I call in opposite direction, call goes through |
| 15:49.02 | jalsot | zoa: where can I see if the call is going in trunk? |
| 15:49.17 | zoa | iax2 trunk debug |
| 15:49.58 | jalsot | hmm, that doesn't show anything |
| 15:50.10 | jalsot | I guess this means, it didn't run in trunk mode |
| 15:50.39 | jalsot | does anybody use iax2 trunk with ztdummy here? |
| 15:50.47 | jalsot | [linux-2.6] |
| 15:50.47 | lilwookie | can I force a codec for a particular extension/dial ? |
| 15:51.55 | sretooh | is there a faxing workaround with *? I have some customers that can't fax or fax only half a page... |
| 15:52.29 | *** part/#asterisk miller7 (~none@adsl49-static-gw1.access.acn.gr) |
| 15:53.44 | blitzrage | nestAr: actually yes... a global variable may be needed |
| 15:56.07 | *** join/#asterisk bah (048830696@ACAC44F7.ipt.aol.com) |
| 15:56.09 | festr_ | i'm getting some SIP messages which a dont understand. CVS stable: planet vip-152 sip phone. i can register with asterisk, but i'm getting this (ethereal) SIP/2.0 401 Unauthorized after REGISTER. in next REGISTER i get 200 OK. and this cycle, one OK one Unauthorized |
| 15:56.16 | festr_ | any idea? |
| 15:56.16 | nestAr | blitzrage: i messed with it some.. * gets mad that ${incoming} doesn't exist before i add 1 to it. |
| 15:56.52 | nestAr | i think i solved that with another GotoIf |
| 15:57.01 | blitzrage | nestAr: you need to initialize the variable first |
| 15:57.10 | *** join/#asterisk fafnir (~hello@tdds-gw.Moscow.gldn.net) |
| 15:57.14 | blitzrage | nestAr: so like you said, it probably needs to be a global variable |
| 15:57.32 | nestAr | of course, then i need a variable for each agent, i guess. |
| 15:57.47 | nestAr | because using the same variable for each agent is not going to work so well |
| 15:57.58 | blitzrage | nestAr: yah, that seems the case |
| 15:58.04 | nestAr | oi vey |
| 15:58.37 | blitzrage | whenever I run into a scaling problem like that, I always go back and see if there is a better way (and there *usually* is :)) |
| 15:59.06 | *** join/#asterisk fugitivo (~ajf@201.255.102.200) |
| 15:59.18 | nestAr | i just keep thinking that someone has run into this and solved it |
| 15:59.19 | *** join/#asterisk Blackvel (~blackvel@dsl-213-023-034-043.arcor-ip.net) |
| 15:59.26 | nestAr | but i can't find anything in the lists |
| 15:59.35 | Blackvel | how can i use the manager api over telnet? some feedback? :) |
| 16:00.01 | RoyK | Blackvel: http://www.voip-info.org/wiki-Asterisk+Manager+API |
| 16:01.55 | Blackvel | I try to read |
| 16:01.57 | Blackvel | but nothing works |
| 16:02.12 | blitzrage | nestAr: hrmmmm... |
| 16:02.16 | Blackvel | can I use standard telnet for it? what is ><CRLF>? |
| 16:02.22 | blitzrage | nestAr: oh... better idea! |
| 16:02.25 | jsolares | a \n, or an enter |
| 16:02.29 | nestAr | oooo |
| 16:02.33 | nestAr | i like better ideas |
| 16:02.34 | blitzrage | nestAr: set the status for the agent in the AstDB |
| 16:02.41 | *** join/#asterisk cpatry (~grepmoo@65.39.228.5) |
| 16:03.08 | blitzrage | nestAr: still need a "variable" for each agent, but it'll be a cleaner implementation in the dialplan as you can just do a DBGet and DBPut() to set and unset the statii |
| 16:03.11 | blitzrage | cpatry: morning! |
| 16:03.22 | cpatry | y0 |
| 16:03.24 | cpatry | whats up? |
| 16:03.38 | nestAr | blitzrage: yeah, that sounds like a good idea |
| 16:03.51 | nestAr | i'll read up on the DB stuff and give it a go |
| 16:03.58 | nestAr | thanks for the suggestions |
| 16:04.09 | bjohnson | still having occasional problems with other people's IVRs registering my dtmf. This call went out over my SPA 3000 to 18003405024 .. cn someone else try it? |
| 16:05.00 | davewise | I have been looking at using a database with * and noticed that in the Zapata.config pages on the wiki, it suggests that it can database configured. However, I don't see where ANY of the DB projects suport zapata.config.... Am I wrong? |
| 16:05.04 | *** join/#asterisk outtolunc (outtolunc@adsl-66-218-53-170.dslextreme.com) |
| 16:05.31 | blitzrage | nestAr: find me in a bit after I've got osme other stuff out of the way and I'll see if I can come up with something. I'm thinking the use of GotoIf()'s and.... maybe a loop... but not sure if the loop will actually be necessary |
| 16:05.44 | nestAr | k |
| 16:06.22 | blitzrage | cpatry: not too much, just woke up after 42 hours of not sleeping :) |
| 16:06.34 | Blackvel | royk: I try to login but nothing seems to move on |
| 16:07.08 | *** join/#asterisk evilrabbi (evilrabbi@64.123.157.107) |
| 16:07.40 | RoyK | Blackvel: tried Events: on? |
| 16:07.45 | blitzrage | OT: where do I go to set the "screen resolution" in linux for the console? I'm not using X - I think I need to edit some sort of framebuffer command in grub.conf on boot, yes? |
| 16:07.58 | lilwookie | can I set the codec before a the dial app? and will Asterisk use it as primary? |
| 16:09.33 | Seyr | Anyone have any experience with * behind a SonicWall firewall (or any firewall) with phone being outside of firewall? |
| 16:09.49 | evilrabbi | i'm looking for something like the WRT54GP2 linksys router but i was wonder if anyone knew if it was Vonage only and if so is there a router that is like that that will work with asterisk? |
| 16:09.49 | davewise | blitzrage: you can put it in the grub.conf or in lilo.conf, but if it is set in the kernel compile, it overrides the boot .conf files |
| 16:10.13 | Blackvel | royk: do I get any reply when the login was successful? I am unsure if the login worked (and the standard linux telnet newline works) |
| 16:10.14 | *** join/#asterisk riksta (riksta@212.85.228.176) |
| 16:10.18 | Blackvel | <- tries again |
| 16:11.17 | evilrabbi | i dunno what it is in grub, but i do in lilo =\ |
| 16:11.20 | Lee__ | is it necessary for context names to be all lowercase, or does it not matter? |
| 16:12.18 | Blackvel | nothing |
| 16:12.38 | Blackvel | can I check on the cli if the telnet manager connection is logged in/could authenticate successful? |
| 16:13.36 | *** join/#asterisk BrianR___ (~brianr@e113a3b01956d519.node.tor) |
| 16:13.40 | jake | blackvel - did you edit manager.conf? |
| 16:13.47 | *** join/#asterisk zip (~zip@adsl-66-136-35-17.dsl.snantx.swbell.net) |
| 16:13.57 | Lee__ | Blackvel: put asterisk in -vvvv mode. it'll report manger connections to the CLI |
| 16:14.07 | Blackvel | sure |
| 16:14.21 | Blackvel | I'll try lee's suggestion then |
| 16:14.28 | RoyK | Blackvel: Response: Success |
| 16:14.28 | RoyK | Message: Authentication accepted |
| 16:14.35 | BrianR___ | http://osiris.978.org/~brianr/voip911/ |
| 16:14.46 | facek_ | anybody use PAP2 adapter? by linksys |
| 16:15.09 | Blackvel | I may be stupid, but do I only have to press the enter key on my keyboard for a newline? :) |
| 16:15.21 | jake | only problem i saw so far in the Manager API is Redirect doesn't seem to be affected by "Priority:" |
| 16:15.29 | jake | just shows 0 |
| 16:15.44 | facek_ | is that good? |
| 16:15.45 | facek_ | <PROTECTED> |
| 16:15.45 | facek_ | <PROTECTED> |
| 16:15.52 | jake | anyone else having that problem? |
| 16:16.44 | Seyr | If I have ports 10000:10100 UDP being forwarded to my * server from the SonicWall firewall and I do not get audio, what might the problem be? |
| 16:16.56 | BrianR___ | http://osiris.978.org/~brianr/voip911/ (comments via email are appreciated...) |
| 16:16.59 | evilrabbi | facek_ i'm looking for one that will work with vonage |
| 16:17.02 | evilrabbi | err |
| 16:17.08 | evilrabbi | will work with asterisk not vonage |
| 16:17.44 | jake | roy - have you use Rediect in manager? |
| 16:17.47 | jake | used |
| 16:17.56 | jake | Redirect |
| 16:18.03 | *** join/#asterisk GustavoIPA (~gustavole@mvx-200-196-55-70.mundivox.com) |
| 16:18.08 | Blackvel | oh, there is show manager connected |
| 16:18.09 | alt | BrianR___: quick comment regarding "zip codes". Many jursidictions used mixed numbers/letters. Canada, Australia and the UK come to mind. |
| 16:19.01 | facek_ | evilrabbi what? |
| 16:20.31 | evilrabbi | facek_ i like the pap2 adapter by linksys to be used by a non vonage server |
| 16:21.42 | facek_ | i use that with asterisk |
| 16:23.02 | *** join/#asterisk stevek (~stevek@slim-eth0.horizonlive.net) |
| 16:23.18 | evilrabbi | facek_ really? |
| 16:23.24 | evilrabbi | facek_ can i have the link? |
| 16:23.45 | evilrabbi | i have been wanting to use one |
| 16:24.14 | FITA1 | I m using asterisk-1.0.7 (serverA) and asterisk-1.0.3(serverB) serverA ---> serverB codec-g729a, and serverB --->client codec-gsm and getting this error on server A "Dropping incompatible voice frame on IAX2/serverB/6 of format gsm since our native format has changed to g729" |
| 16:24.28 | facek_ | evilrabbi how link? what is the problem? |
| 16:25.07 | evilrabbi | facek_ what is the link to the linksys pap2 device you use, because i would like to look at it |
| 16:25.22 | FITA1 | I m using IAX2 client |
| 16:27.42 | Blackvel | uppsi |
| 16:27.51 | Blackvel | I have to restart asterisk to enable managerapi :) |
| 16:28.40 | Seyr | If I have ports 10000:10100 UDP being forwarded to my * server from the SonicWall firewall and I do not get audio, what might the problem be? |
| 16:28.47 | Blackvel | weird |
| 16:28.51 | Blackvel | authentication does not work |
| 16:29.00 | *** join/#asterisk Caede (~Caede@204.94.248.81) |
| 16:29.32 | RaYmAn-Bx | Seyr: asterisk telling remote parties to send the audio to the wrong ip? (internal?) |
| 16:29.44 | jake | blackvel - pastebin your calling code |
| 16:30.10 | Seyr | RaYmAn-Bx: what config option is that? extenalip? |
| 16:31.16 | Caede | Using a TE410P on E1s... anyone have any idea why only EVEN numbered channels might work? |
| 16:31.27 | Lee__ | Seyr: and you verified that UDP traffic is actually going across those ports? |
| 16:31.29 | Blackvel | Username IP Address |
| 16:31.29 | Blackvel | <PROTECTED> |
| 16:31.30 | Blackvel | weird |
| 16:31.33 | Uther_P | Seyr: rtp can use a broad range of ports |
| 16:31.34 | RaYmAn-Bx | Seyr: yes, (well..externip) as well as localnet |
| 16:31.39 | Blackvel | no username there |
| 16:31.39 | *** join/#asterisk zno (~chatzilla@ip-160-79-174-101.autorev.intellispace.net) |
| 16:32.03 | Seyr | Lee__: It works 100% fine if I go to NAT and I have verified that the SonicWall is sending the traffic |
| 16:32.07 | jake | Blackvel - can you pastebin the applicable code from your client app? |
| 16:32.23 | Uther_P | you need your router to be rtp and sip aware and redirect based on protocol, and not port |
| 16:32.25 | MattH | sorry got distracted on something that came up.. G723 has no license? I thought it did? |
| 16:32.43 | MattH | and what does 'passthrough' mean? as in... I assume I can't use g723 to link my asterisk box to a provider without me having the g723 license? |
| 16:32.46 | Blackvel | jake: only telnet which does not work (maybe commands are not sent from telnet to manager?) |
| 16:32.51 | Lee__ | Seyr: what does "go to NAT" mean? |
| 16:33.03 | Blackvel | Connected to localhost. |
| 16:33.03 | Blackvel | Escape character is '^]'. |
| 16:33.03 | Blackvel | Asterisk Call Manager/1.0 |
| 16:33.08 | jake | blackvel - i just got this to work yesterday |
| 16:33.10 | Blackvel | the connect seems to work |
| 16:33.14 | Uther_P | also, It needs to change the ip address inside the protocol itself to change the destination ip of incomming rtp packets |
| 16:33.14 | Seyr | Lee__: if I go to the NAT address over the VPN |
| 16:33.30 | Blackvel | oh, with telnet? i just wanted to try the easiest way ;) |
| 16:33.32 | Uther_P | at least, thats why my isp had to do with our router |
| 16:34.01 | Lee__ | if you are connected via a real VPN you shouldn't have to worrt about any NAT. it's a direct route, no? |
| 16:34.04 | FITA1 | <PROTECTED> |
| 16:34.45 | Uther_P | yea, if both source and destination are rfc1918 addresses transported over a vpn, it doesn't need to be nat'ed |
| 16:34.49 | Seyr | Lee__: If I go to the private address across the VPN, it works fine. If I go to the public address across the internet, I get no audio |
| 16:34.51 | Uther_P | .. er, what lee said |
| 16:35.10 | riksta | FITA1 i think that's ok |
| 16:35.33 | Lee__ | Seyr: that's cause it doesn't know where to go cause of the NAT. Doesn't SIP use dynamic UDP ports? |
| 16:35.54 | Uther_P | sayr: is this through one natd -> over the internet then back through another natd? |
| 16:36.01 | Lee__ | I don't work with SIP much yet |
| 16:36.16 | Lee__ | IAX2 handles this very well |
| 16:36.19 | Seyr | Uther_P: yeh |
| 16:36.36 | FITA1 | riksta: plz explain it, how it is ok |
| 16:36.41 | Uther_P | are they both setup for nat traversal? |
| 16:36.54 | FITA1 | can't I use two codecs? |
| 16:36.55 | Seyr | Lee__: Is there an IAX2 softphone? :-) |
| 16:37.09 | zno | hey guys, when I get a call from coming in via my origination provider, there are no sounds ie Background() doesn't produce sounds, however, internally it works |
| 16:37.44 | Lee__ | Seyr: probably. I'm not too keen on softphones so I haven't paid that much attention. |
| 16:37.58 | Uther_P | inside the protocol it gives its local address, regardless of where its comming from... so it has to be told that its through a nat or the receiver wont send it back to the ip address in the ip header, but the ip in the sip header |
| 16:38.09 | *** join/#asterisk JonM (~JonM@pc-68-118-196-199.will.ct.charter.com) |
| 16:38.21 | JonM | can someone call a FWD # to test for me? |
| 16:38.59 | Seyr | Uther_P: that handled on the server or the router? |
| 16:39.11 | FITA1 | I m using asterisk-1.0.7 (serverA) and asterisk-1.0.3(serverB) serverA ---> serverB codec-g729a, and serverB --->client codec-gsm and getting this error on server A "Dropping incompatible voice frame on IAX2/serverB/6 of format gsm since our native format has changed to g729" |
| 16:39.29 | *** join/#asterisk carlos-d-man (~carlos@201.135.87.178) |
| 16:40.58 | Lee__ | FITA1: dude, you don't have to keep repeating yourself |
| 16:41.11 | Uther_P | Seyr: on the server there is a setting to tell it what your external ip is, so the return goes back to the gateway's ip, I think its externip=x.x.x.x |
| 16:41.19 | JonM | FWD# 632777 can someone call and test see ifit works |
| 16:41.29 | Seyr | Uther_P: thanks |
| 16:41.31 | Uther_P | Seyr: and on the phone there should be an option to say that it is behind a nat |
| 16:41.38 | Seyr | yeh |
| 16:42.19 | Uther_P | but the sip port (5060) itself has to be redirected in that config, otherwize neither one of them would ever be able to make the initial connection to register |
| 16:42.56 | FITA1 | Lee__: what should I do then If no one is answering |
| 16:42.59 | Uther_P | from that point I'm pretty sure they keep in contact to keep the entry in the translation table fresh |
| 16:43.59 | zno | is it possible to filter out the Background() app ie gsm sounds? |
| 16:44.51 | Uther_P | filter out? |
| 16:46.03 | ManxPower | Note: DO NOT ENABLE MMX OPTIMIZATIONS |
| 16:46.38 | Lee__ | zno: delete that from your dial plan |
| 16:47.14 | BrianR___ | alt: The 9 data type only deals with nine-digit US zip codes. Other jurisdictions would need their own data type. |
| 16:47.30 | zno | well I'm thinking voicepulse just shut that down or something cos internally I can hear my IVR menu fine, but when I call in via my origination provider, I don't hear anything from my IVR. however I can talk (ulaw) |
| 16:47.31 | BrianR___ | alt: (The proposal also relies on the fact that a 9-digit zip code in the US narrows location down very tightly) |
| 16:47.41 | *** join/#asterisk Sedorox (brandon@Neptune.client.wlmsprt.pa.sed6.net) |
| 16:47.56 | *** join/#asterisk brad- (~brad@brad.developer.gentoo) |
| 16:48.14 | BrianR___ | alt: I do propose a method for sending alphabetic characters in the apartment code section, but that's probably too slow for more than a few characters. |
| 16:48.55 | BrianR___ | alt: Use of ABCD would help too, but I'm not sure if PSAP DTMF decoders support those. |
| 16:50.28 | *** join/#asterisk jwitte (~jwitte_su@firefly.alpha-lab.net) |
| 16:51.49 | Seyr | Ok, I set externip in SIP.conf and set NAT=yes in SIP.conf global. I also changed X-Lite and I still have no audio |
| 16:52.03 | Seyr | :( |
| 16:52.16 | davewise | Is anyone real familiar with Database Projectss for Asterisk |
| 16:52.40 | *** join/#asterisk freat (~freat@c-67-162-109-129.hsd1.il.comcast.net) |
| 16:52.54 | freat | hi everyone... |
| 16:54.45 | *** part/#asterisk AgiNamu (~AgiNamu@216.230.151.230) |
| 16:54.56 | *** join/#asterisk shido6 (~shido@d57-87-253.home.cgocable.net) |
| 16:56.42 | *** join/#asterisk MikeJ[Laptop] (~ircatjerr@mi.origenfinancial.com) |
| 16:57.03 | *** join/#asterisk sivana (~sivana@mixdown.ca) |
| 16:59.23 | iq | hi freat |
| 16:59.46 | facek_ | what aboue that problem |
| 16:59.47 | facek_ | Apr 27 18:59:23 [kernel] zaphfc: dropped audio (z1=4234, z2=4194, wanted 8 got 40, dropped 32). |
| 17:00.46 | shido6 | anyone with an xbox ? |
| 17:00.58 | tzanger | ManxPower: ENABLE MMX IF YOU GOT MMX CAPABILITIES |
| 17:02.28 | nestAr | MMX = new hotness |
| 17:02.54 | *** part/#asterisk FITA1 (~m_ahmed@202.5.145.50) |
| 17:04.35 | tzanger | actually I have to take a closer look at zaptel drivers but I think that the enable MMX #define only changes the algorithm used, it doesn't actually make the compiler use mmx-only instructions |
| 17:05.26 | iq | shido6, what to do with xbox - just curious |
| 17:06.13 | Sedorox | load linux on it |
| 17:06.29 | *** join/#asterisk nix000 (~dos000@66.11.191.156) |
| 17:06.31 | *** join/#asterisk tld (~tld@80.203.70.227) |
| 17:06.34 | ManxPower | tzanger: No. It makes Asterisk crash randomly. turn it off and asterisk stops crashing every few days. |
| 17:07.00 | *** join/#asterisk fugitivo (~ajf@201.255.102.200) |
| 17:07.46 | nix000 | can someone plz tell me what i am doing wrong ? i have configured /etc/asterisk/zapata with a context as well as /etc/asterisk/extensions but it just never falls into that context when a call comes in. |
| 17:08.10 | shido6 | nix000, dont worry |
| 17:08.53 | *** join/#asterisk jakepdev (~jakepdev@pool-68-236-58-19.phil.east.verizon.net) |
| 17:08.55 | ManxPower | nix000: options must come BEFORE the channel => entry |
| 17:09.21 | nix000 | shido6, ok ... pray tell whats wrong ? did i miss anything ? when you configure your trun/channel with a certain context is it not to supposed to execute whatever is configured in extensions ? |
| 17:09.39 | *** part/#asterisk cpatry (~grepmoo@65.39.228.5) |
| 17:10.10 | shido6 | nix000, as ManxPower said, your options should come before the channel => entry :) |
| 17:10.30 | nix000 | ok .. trying that now. |
| 17:12.39 | Sedorox | *thinks* |
| 17:15.56 | freat | any idea why outbound audio on calls coming in via our voip provider would just stop working? |
| 17:16.02 | freat | the calls are connecting fine |
| 17:16.10 | freat | no errors via iax2 debug |
| 17:16.34 | freat | if I connect in via another asterisk box all is fine, so it leads me to believe it's a problem upstream |
| 17:16.38 | *** join/#asterisk Grooby (~Grooby@12.22.232.212) |
| 17:16.53 | *** join/#asterisk A-Tuin (~a-tuin@steves.ip.v4.me.uk) |
| 17:17.32 | freat | this sucks |
| 17:20.47 | mmlj4 | freat: who is your provider? |
| 17:21.09 | *** join/#asterisk A-Tuin (~a-tuin@steves.ip.v4.me.uk) |
| 17:22.28 | Seyr | any other ideas about my no audio problem Uther_P? |
| 17:24.42 | *** join/#asterisk chetan (freetibet@cpe-24-193-188-21.nyc.res.rr.com) |
| 17:25.33 | zno | freat: voicepulse? |
| 17:25.37 | chetan | is it possible to get the remote ip via agi for a sip channel? |
| 17:25.37 | freat | mmlj4: teliax |
| 17:26.05 | zno | voicepulse has major issues today |
| 17:26.06 | zno | they suck |
| 17:26.19 | zno | no incoming calls since 9:00 EST |
| 17:26.25 | freat | wow |
| 17:26.42 | freat | the calls actually connect to us, but the audio doesn't make it out |
| 17:26.47 | zno | yep |
| 17:26.49 | zno | that's what I mean |
| 17:26.56 | zno | I just called their tech support |
| 17:27.06 | freat | your'e talking about voicepulse though right? |
| 17:27.10 | zno | yeah |
| 17:27.17 | zno | is that who you have? |
| 17:27.36 | *** join/#asterisk Cheng29 (~cheng29@d57-87-253.home.cgocable.net) |
| 17:27.45 | freat | teliax |
| 17:28.01 | freat | they have been having problems with XO who provides our PRI |
| 17:28.24 | zno | thank god I didn't port my numbers over to them either |
| 17:28.26 | Seyr | anyone ever have audio problems when putting Asterisk behind a firewall? |
| 17:28.29 | freat | since it is incoming calls that experience the problem, it leads me to believe that the switch of the PRI may be going through (they were moving it to Level 3) |
| 17:28.47 | freat | outbound calls have been fine, probably because they route out a different PRI |
| 17:29.00 | shido6 | run the firewall on your Asterisk box |
| 17:29.02 | shido6 | :) |
| 17:29.08 | fugitivo | Seyr: what kind of problems? |
| 17:29.26 | shido6 | packets not arriving fast enough? |
| 17:29.35 | freat | open up the packet valve |
| 17:29.48 | mmlj4 | freat: i haven't heard of them. However (which is why I asked), I have heard of providers messing with VoIP packets crossing their network. |
| 17:30.03 | mmlj4 | i hope that's not your problem, in this case |
| 17:30.15 | mmlj4 | work & |
| 17:30.19 | Seyr | heyas fugitivo! |
| 17:30.36 | freat | I've worked with the senior network engineer of our ISP to get our traffic prioritized, so I think we're ok there |
| 17:30.49 | freat | woah |
| 17:30.54 | freat | 2 calls in a row just made it through! |
| 17:30.58 | shido6 | maybe on that side of things |
| 17:31.07 | freat | 3 calls! |
| 17:31.07 | Seyr | fugitivo: my Asterisk box is behind a SonicWall firewall at a data center. My softphone (X-Lite) is behind a firewall here at the office. If I connect over the VPN, everything works fine. If I connect to the public IP of the * server, I get no audio |
| 17:31.09 | freat | uh oh |
| 17:31.14 | freat | I'm getting my hopes up |
| 17:31.15 | shido6 | now download some movies and make some calls |
| 17:31.38 | freat | dammit, 4th call bombed |
| 17:32.04 | Seyr | The network admin said he is forwarding UDP 5060 and UDP 10000:10100 UDP to * |
| 17:32.13 | shido6 | errr |
| 17:32.35 | Seyr | I have externip defined as the external IP of the * box in default of sip.conf |
| 17:32.40 | freat | don't you want to use 4569? |
| 17:32.44 | fugitivo | Seyr: you must have some ports open |
| 17:32.56 | syle | tell him to put you on the outside switch with the router :) |
| 17:32.58 | shido6 | voip_control_port: "5060" |
| 17:32.58 | shido6 | start_media_port: "16384" |
| 17:32.58 | shido6 | end_media_port: "32766" |
| 17:33.06 | freat | sorry you're doing SIP |
| 17:33.10 | freat | my bad |
| 17:33.26 | iq | hi, anyone using SERWEB ? |
| 17:33.47 | Seyr | shido6: yeh, all that is defined |
| 17:34.10 | Seyr | my RTP ports are set to 10000 - 10100 |
| 17:34.24 | *** part/#asterisk chetan (freetibet@cpe-24-193-188-21.nyc.res.rr.com) |
| 17:34.37 | Seyr | and those ports are being forwarded from the firewall |
| 17:35.45 | nix000 | is there a default context that this thing falls in ? i cant seem to get it to work for my x100p channel. it was very easy to set ip up for my xten sip extension. |
| 17:35.52 | *** join/#asterisk viLeR (1000@ip-47-252.telesat.com.co) |
| 17:36.40 | shido6 | u need to match the context in extensions.conf with zapata.conf |
| 17:36.50 | fugitivo | Seyr: i'm behind a home router, and i open ports UDP 5060-5070, 10000-20000 |
| 17:36.54 | nix000 | shido6, i did ! |
| 17:36.59 | fugitivo | Seyr: it works fine |
| 17:37.05 | shido6 | then the order of your settings in zapta.conf is wrong |
| 17:37.09 | *** part/#asterisk flot (~flot@user191.hovrino.net) |
| 17:37.21 | shido6 | what do you see when you rung ztcfg -vv |
| 17:38.06 | nix000 | shido6, Channel 01: FXS Kewlstart (Default) (Slaves: 01) !!!!!!!!! |
| 17:38.36 | nix000 | shido6, i know i just changed the options as you suggested earlier |
| 17:39.20 | RaYmAn-Bx | Seyr: you might want to make sure the problem is actually on the asterisk side and not on your side..(as well as find out whether outbound traffic is restricted on the server?) |
| 17:39.37 | shido6 | nix000, ok show me your zapta.conf |
| 17:40.13 | bjohnson | still having occasional problems with other people's IVRs registering my dtmf. This call went out over my SPA 3000 to 18003405024 .. cn someone else try it? |
| 17:41.00 | nix000 | shido6, here is zapata http://pastebin.ca/10436 |
| 17:41.10 | ManxPower | bjohnson: any Zap involved? |
| 17:41.37 | nix000 | shido6, and extensions http://pastebin.ca/10437 |
| 17:41.48 | bjohnson | ManxPower: no |
| 17:42.08 | bjohnson | ManxPower: however .. |
| 17:42.42 | *** join/#asterisk grolloj (~chatzilla@slim-eth0.horizonlive.net) |
| 17:42.42 | bjohnson | ManxPower: nortel handsets going through Nortel CICS to fxs and then out fxo (all fxs and fxo are Sipura SPA) |
| 17:42.50 | *** join/#asterisk kb1_kanobe (~krisbouti@h24-207-80-52.cst.dccnet.com) |
| 17:43.05 | bjohnson | ManxPower: sometimes it works, womestimes it doesn't .. even when testing against the same number |
| 17:44.39 | kb1_kanobe | I'm having a problem with intermittant loss of audio on IAX channels. It seems to correlate to appearances of 'channel.c: Dropping voice to exceptionally long queue on IAX2/....' on the console. This is yesterdays cvs-head. Can anyone enlighten me as to possible causes? |
| 17:44.48 | ManxPower | bjohnson: the SIPura has the ability to increase the DTMF tone length on the FXO port. I don't recall the option name. |
| 17:45.22 | kb1_kanobe | The cvs-head'ness might be a red herring - I think I've been having this problem for a while but it's so sporadic I've not been able to catch it in the act. |
| 17:47.24 | nix000 | anyone can explain what TRUNK=Zap/g2 (in extensions.conf) would do to an x100p channel ? |
| 17:47.51 | *** part/#asterisk Seyr (~Seyr@rrcs-24-227-133-226.sw.biz.rr.com) |
| 17:47.58 | nix000 | duh ! |
| 17:49.21 | *** join/#asterisk bannerman (~bannerman@209.216.176.43) |
| 17:52.08 | nix000 | anyone can tell if there is a way to see which context a chanel is assigned to ? |
| 17:52.48 | Lee__ | grep? |
| 17:53.10 | Ahewes | Hi guys. Can anyone point me to a working config for a Mediatrix 1204? |
| 17:53.11 | jakepdev | <nix000>: show channels? |
| 17:54.02 | ManxPower | someone do a "show application park" and tell me if you think that Park(1234) should park the call on extension 1234. |
| 17:54.28 | nix000 | jakepdev, thanks, that shows that my channel is in the right context ... it still frikin does not get answered in that context. |
| 17:54.54 | jakepdev | <nix000>: why not? |
| 17:55.00 | bjohnson | ManxPower: looks like it .. unless 1234 is another exten |
| 17:55.01 | jakepdev | <nix000>: where does it go? |
| 17:55.14 | nix000 | jakepdev, i think default |
| 17:55.27 | jakepdev | <nix000>: you did a reload? |
| 17:55.33 | ManxPower | bjohnson: try it. 8-) It will still read back the first available parking slot |
| 17:55.36 | nix000 | jakepdev, multiple times |
| 17:55.40 | Ahewes | Is anyone actually using the 1204 with Asterisk? |
| 17:55.45 | jakepdev | <nix000>: sip IAX, or zap? |
| 17:55.52 | nix000 | jakepdev, zap |
| 17:56.03 | nix000 | my sip ext is very fine |
| 17:56.04 | shido6 | nix000, find a cheap consultant |
| 17:56.26 | shido6 | use the wiki |
| 17:56.30 | shido6 | www.voip-info.org |
| 17:56.56 | jakepdev | <nix000>: pastebin zapata.conf |
| 17:57.20 | jakepdev | <nix000>: or find a cheap consultant :) |
| 17:57.28 | bjohnson | ManxPower: park() is not listed in the wiki at all |
| 17:57.35 | ManxPower | http://pastebin.ca/10440 |
| 17:57.45 | ManxPower | bjohnson: its internal, not an app_park.c |
| 17:58.00 | Blackvel | jakepdev: haha...there are any? |
| 17:58.24 | jakepdev | <nix000>: depends what you consider cheap - i know one for $85 per hour will help you |
| 17:58.43 | nix000 | jakepdev, consultant has the word con in it .. so by definition they will cost more than you think ! |
| 17:58.55 | Blackvel | hehe |
| 17:59.10 | Blackvel | what is a real IT hourly dumping price? |
| 17:59.16 | jakepdev | <nix000/blackvel>: i hear that |
| 17:59.27 | bjohnson | ManxPower: spa DTMF Playback Length default is .1 .. do you figure it's a duration problem? |
| 17:59.31 | *** join/#asterisk thdrtdrhdyjdtyj (~jon@38.136.42.2) |
| 17:59.32 | ManxPower | I charge $120/hr, but give discounts |
| 17:59.32 | jakepdev | <blackvel>: i say it depends upon the demand |
| 17:59.40 | ManxPower | bjohnson: make it .5 |
| 17:59.45 | nix000 | jakepdev, are you a consultant yourself ? |
| 17:59.52 | bjohnson | ManxPower: I still would have thought that park would be documented on the wiki |
| 18:00.00 | jakepdev | <nix000> - I am - but on for * stuff :) |
| 18:00.03 | nix000 | i mean .. rom nufone ? |
| 18:00.06 | jakepdev | <nix000> - not |
| 18:00.06 | ManxPower | bjohnson: one would think so |
| 18:00.12 | Blackvel | if I had charged for my little project prices like 85$-115$ I would be rich now |
| 18:00.14 | ManxPower | bjohnson: I don't use parking much. |
| 18:00.18 | Blackvel | ;) |
| 18:00.23 | jakepdev | <nix000> - i don't work for nufone |
| 18:00.36 | thdrtdrhdyjdtyj | if I set up voip in my office using asterisk how ddo I connect to the outside world of phones. Can someone point to some good docs on subject |
| 18:00.45 | bjohnson | ManxPower: I don't ue it at all .. I was just interested |
| 18:00.54 | bjohnson | ~docs |
| 18:00.55 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 18:00.59 | bjohnson | thdrtdrhdyjdtyj: ^^ |
| 18:01.04 | ManxPower | bjohnson: I'm TRYING to do roundrobin for the parking lot number. Our receptionsts finish the supervised transfer before the digits are finished being read back since it's always the same parking lot, so the CALLER hears the parking lot numbers read to them |
| 18:01.10 | jakepdev | <nix000> - but someone from nufone got me up to speed pretty quick - so i'd say it was money well spent |
| 18:01.19 | Blackvel | anybody tried manager api originate yet? (I checked docs) |
| 18:02.32 | sretooh | anyone got a good recommendation for provider to terminate my toll free calls to? |
| 18:02.46 | bjohnson | nix000: TRUNK=Zap/g2 assigns "Zap/g2" to a variable named TRUNK .. it doesn't do anything to any hardware |
| 18:02.49 | shido6 | how many simultanous toll free calls sretooh ? |
| 18:03.06 | sretooh | very low volume, not more than 5 |
| 18:03.09 | ManxPower | <PROTECTED> |
| 18:03.17 | nix000 | jakepdev, in due time i will. but not now. |
| 18:03.22 | sretooh | just need something better than sixtel... |
| 18:04.18 | ManxPower | I sure seem to want to do things nobody else has ever done. |
| 18:04.48 | nix000 | jakepdev, plz take a look at http://pastebin.ca/10442 its my ext.conf and http://pastebin.ca/10443 zapata.conf |
| 18:04.55 | jakepdev | ManxPower - the visionary |
| 18:05.21 | nix000 | jakepdev, i just need this thing to get answered |
| 18:05.28 | bjohnson | ManxPower: .5 won't allow me to dial out .. gives me a beep beep beep signal |
| 18:05.34 | ManxPower | jakepdev: I doubt it. I'm just much more twisted than most people. 8-) |
| 18:05.41 | ManxPower | bjohnson: THATS weird. |
| 18:06.55 | jakepdev | <nix000> - context = pstn_x100p |
| 18:07.13 | nix000 | jakepdev, whats wrong ? |
| 18:07.36 | jakepdev | spaces possibly? |
| 18:07.43 | jakepdev | don't know for sure |
| 18:07.49 | blitzrage | actually... documentation cannot be found at #asterisk-doc unless someone is writing it :) |
| 18:08.13 | jakepdev | <nix000> - try "context=pstn_x100p" |
| 18:08.59 | jakepdev | <nix000> - plus - you're configs look a mess :) |
| 18:09.31 | jakepdev | why not simplify it by putting in only what you need for now |
| 18:09.33 | jakepdev | ? |
| 18:10.28 | nix000 | jakepdev, i modified the make samples config files ! |
| 18:11.53 | nix000 | jakepdev, it finally works ! that was it ... stupid space ! it was reported wrongly with zap show channels tho. |
| 18:12.32 | bjohnson | ManxPower: .5 isn't giving me weird signal after reboot of device .. but still not working with problem ivr |
| 18:12.51 | bjohnson | ManxPower: remote ivr works through stright telco line |
| 18:13.01 | bjohnson | maybe I'll try not through the nortel to test |
| 18:13.05 | jakepdev | <nix000> - it's less confusing to keep the sample files as samples and only add what you need. just personal taste though |
| 18:13.54 | nix000 | jakepdev, is there a minimum config files that need to be present ? |
| 18:14.30 | ManxPower | There's a patch on mantis to add that feature to CVS-HEAD that applies to 1.0.x stable |
| 18:14.41 | jalsot | could anybody help me configuring * to receive SIP call from cisco as5300? [I'm getting: Found no matching peer or user for 'IP:52668'] |
| 18:14.43 | jakepdev | you need all of the config files, maybe not just all the sample lines |
| 18:14.51 | bjohnson | ManxPower: do you think DTMF Playback Level:is the volume out or the playback volume back to the phone on the fxs? |
| 18:15.33 | nix000 | jakepdev, as a new user i had no way of knowing what was the minimum. so i took it all in the hope that as soon as i have something working ill start the liposuction effort. |
| 18:16.05 | Ahewes | Google is failing me. Would anyone like to make a quick 20 bucks via paypal to send me docs for Asterisk 1204 configuration? |
| 18:16.43 | Ahewes | If someone already has a 1204 installed, this shouldn't be a biggie. |
| 18:17.18 | Blackvel | how can I call the application commands (AGI EXEC commands) inside the manager api? the Action: Command only lets me execute non-application commands? I am interested in playing music files for a longer time and interrupt the dailing at some time |
| 18:18.11 | PTG1234 | what the heck is a 1204 |
| 18:18.11 | Blackvel | e.g calling MusicOnHold |
| 18:19.28 | zip | is there a map of the main e164 context of dundi, bound by the gpa? |
| 18:19.29 | zip | like the dundi-test map? |
| 18:19.48 | kajtzu | hmm |
| 18:19.57 | kajtzu | a user hung in musiconhold |
| 18:20.23 | *** join/#asterisk FarrisG (~jrush@h-68-164-19-170.dllatx37.covad.net) |
| 18:20.47 | FarrisG | Can anyone help me figure out why I can't get in-call DTMF to work correctly from X-Lite? |
| 18:21.37 | kajtzu | FarrisG: set dtmfmode to rfc2833 on * for that user |
| 18:21.42 | kajtzu | works for my users |
| 18:21.50 | kajtzu | no additional settings on the xlite end |
| 18:22.36 | *** join/#asterisk three55ml (~three55ml@cpe-24-28-65-205.austin.res.rr.com) |
| 18:22.43 | FarrisG | I'll give it a shot |
| 18:23.02 | FarrisG | Hang on, what do you mean by "on *"? |
| 18:23.02 | Blackvel | kajtzu: what do you mean? |
| 18:23.12 | kajtzu | * = asterisk :) |
| 18:23.14 | three55ml | Hello everyone |
| 18:23.18 | FarrisG | oh |
| 18:23.19 | FarrisG | DUH |
| 18:23.21 | FarrisG | hehe |
| 18:23.24 | FarrisG | sip.conf, right? |
| 18:23.27 | kajtzu | yeah |
| 18:25.38 | FarrisG | kajtzu: Awesome, thanks! |
| 18:25.40 | jakepdev | blackvel - that's not advisable as the commands will block in manager |
| 18:25.43 | *** join/#asterisk wildgoose (~edward@xunil.mail.wildgooses.com) |
| 18:26.09 | kajtzu | FarrisG: np |
| 18:26.12 | Ahewes | PTG1234 it's a mediatrix box, and I'm under the gun to get it working, but I'm quickly figuring it out. |
| 18:27.09 | Ahewes | This is the golden link on the Mediatrix 1204: http://lists.digium.com/pipermail/asterisk-users/2004-June/050906.html |
| 18:27.18 | Blackvel | hm, but I want to use the manager API to get around the damn FastAGI blocking ;) ;) |
| 18:27.27 | jakepdev | blackvel - you can |
| 18:27.50 | jakepdev | blackvel - just use redirect through manager |
| 18:28.15 | Blackvel | anyways I can not call MusicOnHold from the api. is there a right way to play a music file? |
| 18:28.30 | jakepdev | blackvel - you can call MOH from dialplan or AGI |
| 18:28.31 | Blackvel | redirect to what? |
| 18:28.37 | Blackvel | hm ok |
| 18:28.40 | jakepdev | blackvel - to a different context |
| 18:28.56 | Blackvel | what is the command for the dialplan? moh? |
| 18:29.03 | Blackvel | MusicOnHold? |
| 18:29.05 | jakepdev | yes |
| 18:29.53 | Blackvel | but when does it stop? |
| 18:30.06 | jakepdev | when you call Redirect from the Manager API |
| 18:30.11 | Blackvel | and how could I interrupt playing music when I want to dail an agent? |
| 18:30.14 | Blackvel | hm |
| 18:30.28 | Blackvel | first I dail with the m api redirect to an MusicOnHold extension |
| 18:30.39 | Blackvel | and this does not block, right? |
| 18:31.01 | jakepdev | blackvel - Manager should redirect no matter what |
| 18:31.04 | Blackvel | Am I able to call redirect a 2nd time when I want to connect the AGI call with a telphone? |
| 18:31.23 | jakepdev | blackvel - redirect as much as you want |
| 18:31.31 | shido6 | brb |
| 18:33.14 | Blackvel | and a 2nd redirect interrupts the first (which is still playing music)? |
| 18:33.28 | jakepdev | blackvel - yes |
| 18:33.44 | Hogie | I have a tmd with 3fxo's on it, hooked up to the PSTN (no other wiring besides to the fxo ports). I have echocancel=yes, echotraining=yes, echocancelwhenbridged=yes and I have been trying to get rx & tx gains right, but they still hear echo on the office side (as in, our cisco 7960's) on incoming calls only, not outgoing over those circuits... I dont know what to do.... |
| 18:33.52 | Blackvel | so when the agent is busy, I can play MOH, and when my system marks it as free, I can send a 2nd redirect |
| 18:33.59 | jakepdev | blackvel - yes |
| 18:34.00 | jakepdev | :) |
| 18:34.08 | Blackvel | this is hard work for me jakepdev :) |
| 18:34.20 | Blackvel | thanks for your patience |
| 18:34.25 | jakepdev | blackvel - np |
| 18:35.04 | ManxPower | Customer: "*scream* Feature X isn't working! Fix it ASAP!" Me: "Do you want me to stop work on feature Y to make feature X work?" Customer: "No! Feature X is top priority!" |
| 18:35.09 | ManxPower | Why do they even waste my time |
| 18:35.42 | Blackvel | do you mean Feature Y is top? |
| 18:35.49 | Blackvel | makes no sense somehow :) |
| 18:35.51 | Hogie | oh, im using 1.0 stable from april 11 |
| 18:35.52 | syle | that is what i was thinking lol |
| 18:35.54 | alt | ManxPower: because "the customer is always right". The Corrollary to that is "The customer is also always an idiot" |
| 18:36.19 | Blackvel | and the 2nd rule, if he is not right, make him thinking he is right ;) |
| 18:38.21 | fugitivo | why i can't transfer calls with my stanaphone and fwd account, do i need to set a special flag to trasnfer those calls? |
| 18:38.31 | Blackvel | jakepdev: and if the 2nd REDIRECT is connected to the agent, it it still not blocking? I could send a HANGUP from the agent to terminate the caller? |
| 18:38.39 | Grooby | anyone here have experience with IP500 phones? |
| 18:38.40 | jakepdev | blackvel - yes |
| 18:38.47 | jakepdev | blackvel - redirect as much as you want |
| 18:39.00 | *** join/#asterisk C4-Telecom (~sales@212.145.163.120) |
| 18:39.03 | jakepdev | blackvel - call hangup if you choose |
| 18:39.35 | Blackvel | and how do I get notified about that there was a hangup (the caller hangup e.g)? manager api event? e.g the agi exec dial returns -1 |
| 18:40.10 | jakepdev | blackvel - yes - manager api event |
| 18:40.44 | Blackvel | I feel I am on the right track |
| 18:40.46 | Blackvel | ;) |
| 18:41.01 | jakepdev | blackvel - i agree - in no time you'll have this up and running |
| 18:41.06 | *** join/#asterisk odie_flocon_ (~Odie@ptr-64-201-182-211.ptr.terago.ca) |
| 18:41.21 | *** join/#asterisk santiago (~santiago@63.245.86.199) |
| 18:41.26 | *** join/#asterisk Village (~Village@www.liquidcore.net) |
| 18:41.50 | ManxPower | > Well... Per everyone's advice I changed the motherboard (still Intel for now. Ordering an AMD of some sort) and the static WENT AWAY!!! I am still finding it amazing that I could go to a slower, crappier different Intel board and the problems go away. |
| 18:42.08 | ManxPower | If the poor sod had listened to my advice to change the motherboard in the first place.... |
| 18:42.20 | C4-Telecom | how can capture a call returned "No one is available to answer at this time"??? |
| 18:42.25 | Village | Grandstream GXP-2000..spoke with a Grandstream rep the other day and she said that it would support intercom/paging features in a future firmware update..now I can't seem to get back in touch with her. Anyone have any ideas about an ETA for those features? |
| 18:42.29 | C4-Telecom | is busy that?? |
| 18:42.31 | *** join/#asterisk FuzzyCat (~ScaredyCa@i169173.upc-i.chello.nl) |
| 18:42.40 | FuzzyCat | bkw_: you active? |
| 18:42.45 | phigan | Am I understanding this correctly? SER can be used as an rtp proxy when forwarding rtp ports doesn't work? |
| 18:42.50 | Blackvel | sorry, I am not convinced about this ;) I have to do it from the Java side. well I'll check how to pass the manager events |
| 18:43.05 | iCEBrkr | Well this is odd. |
| 18:43.16 | iCEBrkr | inbound voicepulse calls just started being rejected. |
| 18:43.22 | iCEBrkr | Wonder what I screwed up. |
| 18:43.52 | Nuxi | Hmmm. given the Blackvel conversation, I think it should be possible to interrupt a call every 15 minutes for advertizements. I should pattent this and sell it to the cable companies. |
| 18:44.03 | iCEBrkr | haha |
| 18:44.12 | iCEBrkr | Sicko |
| 18:44.14 | C4-Telecom | "No one is available to answer at this time" will go to N +101 ??? |
| 18:44.37 | iCEBrkr | C4-Telecom: You mean a 'busy'? |
| 18:44.48 | C4-Telecom | don´t know if busy |
| 18:44.49 | *** join/#asterisk Rick_Hunter (~rhunter@07-184.008.popsite.net) |
| 18:44.49 | phigan | Can SER be used as an rtp proxy when forwarding rtp ports doesn't work? |
| 18:44.50 | C4-Telecom | asking |
| 18:45.07 | iCEBrkr | C4-Telecom: Ibelieve so. I have my voicemail setup that way. |
| 18:45.09 | iCEBrkr | lemme check |
| 18:45.17 | Nuxi | could call it "triple-A" for Asterisk Advertizing API |
| 18:45.26 | C4-Telecom | i got that when all my terminating channels are full |
| 18:45.40 | phigan | or, is there anything in particular anyone can recommend as an rtp proxy, since forwarding my rtp ports doesn't work? |
| 18:45.53 | *** join/#asterisk Skarmeth (~Skarmeth@201009023158.user.veloxzone.com.br) |
| 18:46.00 | bjohnson | phigan: yes .. but if you're going to put the SER on a public IP .. why not just put * there |
| 18:46.11 | phigan | bjohnson: its a sparc 5 |
| 18:46.13 | FuzzyCat | phigan: there's an rtp proxy for SER, check the website.. |
| 18:46.19 | phigan | bjohnson: with 64mb ram and 100megs free |
| 18:46.22 | bjohnson | phigan: typically we recommend iax |
| 18:46.37 | phigan | bjohnson: Yeah I have iax working well. But I do want to connect to stanaphone |
| 18:46.45 | phigan | FuzzyCat: so SER isn't the rtp proxy? |
| 18:46.47 | santiago | Hi, I have a problem. I'm tring to use a sip phone registered from Internet to my internal extensions. From the client, I can hear but I'm not heard |
| 18:46.51 | iCEBrkr | C4-Telecom: I'm going to say yes... |
| 18:47.05 | iCEBrkr | santiago: NAT?? |
| 18:47.09 | phigan | FuzzyCat: My mistake :) |
| 18:47.10 | santiago | iCEBrkr, not |
| 18:47.25 | santiago | i've tried with different connections |
| 18:47.38 | *** join/#asterisk Ali42 (~ali@proxy-4.epitech.net) |
| 18:47.56 | Ali42 | hi |
| 18:48.07 | C4-Telecom | thanks.... i´m doing load balance, will add n+101 to reloop the call |
| 18:48.11 | iCEBrkr | santiago: firewall? |
| 18:48.20 | Ali42 | is it possible to link a ericcsson md110 pbx to an asterisk plateform? |
| 18:48.33 | santiago | iCEBrkr, no, the machine even doesn't have the iptables modules |
| 18:48.46 | iCEBrkr | haha |
| 18:48.48 | iCEBrkr | ok |
| 18:48.59 | FuzzyCat | bbl |
| 18:49.03 | *** part/#asterisk FuzzyCat (~ScaredyCa@i169173.upc-i.chello.nl) |
| 18:49.04 | jakepdev | Ali42 - what type of interface does it have - PRI,IAX,Analog, SIP? |
| 18:49.18 | iCEBrkr | santiago: describe the setup more. |
| 18:49.21 | santiago | i've sniffed and there isn't rtp traffic from the client in the server |
| 18:49.36 | Ali42 | talking about the md110? |
| 18:49.37 | phigan | hm, I got ser cvs and it didn't come with rtpproxy. i'll check the release |
| 18:49.42 | jakepdev | Ali42 - yes |
| 18:49.52 | Ali42 | jakepdev: analog |
| 18:49.59 | santiago | iCEBrkr, what config do you need? |
| 18:50.00 | Ali42 | an E1 |
| 18:50.32 | iCEBrkr | santiago: Just your network layout.. like where the sip device and asterisk machine is at. etc? And which direction you can and cannot be heard. |
| 18:50.33 | jakepdev | Ali42 - then yes - for analog you need the TDM cards or for E1, you need the TE100P |
| 18:50.45 | jakepdev | Ali42 - or the TE400P for 4 ports |
| 18:51.04 | Ali42 | Thid card is plugable in the md110? |
| 18:51.28 | Ali42 | no sorry i meant sirectly to the pc |
| 18:51.37 | iCEBrkr | We got a NEAX phone system and it's got an ethernet port.. I was thinking about trying to make a hybrid system |
| 18:51.46 | jakepdev | Ali42 - don't know abou that switch specifically, but asterisk is very flexible in that it can work with T1, E1, analong, Sip, IAX, etc... |
| 18:51.50 | Ali42 | directly (excuse my bad english) |
| 18:51.55 | iCEBrkr | ..I think that's what it's labeld. :*Shrug* |
| 18:52.04 | *** join/#asterisk riksta (riksta@212.85.228.176) |
| 18:52.08 | santiago | iCEBrkr, the server has an external IP, the client uses a dial-up connection. there isn't audio from the client to the server |
| 18:52.31 | santiago | iCEBrkr, and there isn't firewall between them |
| 18:52.32 | jakepdev | Ali42 - what are you trying to do with asterisk and your switch? |
| 18:52.44 | Ali42 | jakepdev: In facts, you mean that it's possible to have the md110 works on top of an asterisk? |
| 18:53.13 | iCEBrkr | santiago: the client being the sip device? |
| 18:53.14 | jakepdev | Ali42 - ontop of it, underneath it, and beside it |
| 18:53.32 | iCEBrkr | Japan lol |
| 18:53.36 | santiago | iCEBrkr, yes |
| 18:53.41 | iCEBrkr | err jakepdev: haha |
| 18:53.43 | *** join/#asterisk BrianR___ (~brianr@45e596fcf7c48cda.node.tor) |
| 18:53.54 | Ali42 | Then it's possible to connect the md110 to asterisk :) |
| 18:53.57 | iCEBrkr | santiago: Ok, You see the SIP device regster and all that jazz? |
| 18:54.16 | santiago | iCEBrkr, yes, the sip device is registered |
| 18:54.18 | jakepdev | Ali42 - yes - but you need to make sure you have the proper cards for your asterisk box to make it work |
| 18:54.38 | iCEBrkr | santiago: OK. You have a simple echo test or voicemail extension setup? |
| 18:54.52 | iCEBrkr | I had this problem, I just forget what I did to fix it. |
| 18:54.53 | santiago | iCEBrkr, i've tried with an echo test |
| 18:54.54 | Ali42 | ok but how can i connect pbx to asterisks' card? |
| 18:55.03 | jakepdev | Ali42 - E1? |
| 18:55.05 | Hogie | anybody have any good echo cancel info besides the wiki, I have done all I could find for asterisk on the wiki (Using TDM 3FXO and Cisco handsets, and only getting echo on incoming calls to the TDM ports, when picked up by the 7960 handsets) |
| 18:55.12 | Ali42 | the e1 is enteering the md110 not the asterisk |
| 18:55.31 | santiago | iCEBrkr, it works if i call from the same LAN of the server but not from INET, using the same client |
| 18:55.31 | iCEBrkr | Hogie: Have you adjusted your TX/RX gains? |
| 18:55.34 | jakepdev | Ali42 - what outputs do you have? analog? |
| 18:55.42 | Ali42 | the line from our provider |
| 18:55.53 | Hogie | iCEBrkr: Ive tried several (20+ combinations) settings |
| 18:56.04 | Hogie | and it only gets better when rxgain is like -15 |
| 18:56.07 | jakepdev | Ali42 - the line side is connected to the telco, but are there other outs? |
| 18:56.11 | Hogie | and its not that it is better, it just takes less time to train |
| 18:56.20 | Ali42 | We have a E1 wich arrives to the md110 |
| 18:56.36 | Grooby | who here uses the polycom ip500? |
| 18:56.39 | Hogie | its taking 25+ seconds to echotrain, and I have echocanel=yes, echotraining=yes and echocancelwhenbridged=yes |
| 18:56.44 | iCEBrkr | santiago: That should work. I'm remote in Florida, while my asterisk box is up in Ohio. My sip device down here is behind a router+firewall+nat. |
| 18:57.05 | iCEBrkr | santiago: I'm also using a STUN server and nat=yes |
| 18:57.09 | santiago | iCEBrkr, i don't know what to do |
| 18:57.11 | Ali42 | jakepdev: we have many cards on the md110, each one is for a group of lines |
| 18:57.40 | Blackvel | iCEBrkr: stun with asterisk together? |
| 18:57.40 | iCEBrkr | santiago: Like I said, I had the same issue ebfore, but I can't recall what I did to fix it.. |
| 18:57.56 | jakepdev | Ali42 - which group do you want to use to connect to asterisk, E1, analog, etc? |
| 18:58.02 | iCEBrkr | santiago: I was hoping in your description it'd jog my memory. :) |
| 18:58.08 | iCEBrkr | Blackvel: What do you mean? |
| 18:58.24 | Blackvel | iCEBrkr: you use STUN? for what? Asterisk? |
| 18:58.27 | iCEBrkr | Blackvel: My two NAT'd devices are configured to use a STUN server. |
| 18:58.32 | Blackvel | ah |
| 18:58.34 | iCEBrkr | Blackvel: Yes. |
| 18:58.38 | iCEBrkr | I guess. |
| 18:58.57 | *** join/#asterisk L|NUX (~linux@202.5.145.58) |
| 18:59.18 | iCEBrkr | My asterisk box back home is on a public IP. |
| 18:59.22 | santiago | Hogie, I had an echo problem, and the only way to solve it was to adjust the rx/tx gains |
| 18:59.27 | Hogie | iCEBrkr: any ideas then? |
| 18:59.29 | iCEBrkr | It's my SIP devices down here that are NAT'd. |
| 18:59.41 | santiago | Hogie, i you wait a second, i'll give a good link about that |
| 18:59.42 | iCEBrkr | Hogie: Just the TX/RX gains. |
| 18:59.43 | sivana | santiago: what did you set it to? |
| 19:00.06 | *** join/#asterisk florz (nobody@2001:1a50:503c:0:0:0:0:1) |
| 19:00.09 | iCEBrkr | Hogie: I havent' had much problems with echo, then again, I'm not using a high-end linecard. |
| 19:00.21 | Hogie | I didn't have echo problems on x100's |
| 19:00.22 | Hogie | lol |
| 19:00.30 | Hogie | dont have echo when calling out |
| 19:00.34 | machinehd | Hogie, echotraining=800 ? |
| 19:00.35 | Hogie | just incoming calls |
| 19:00.35 | iCEBrkr | Look at Florz flaunt his ipv6 |
| 19:00.35 | *** topic/#asterisk by bkw_ -> Asterisk: The Open Source PBX || 1.0.7 Released || ClueCon Dev Conf Aug 3rd - 5th - Chicago |
| 19:00.42 | Hogie | machinehd: tried that |
| 19:00.47 | Silik0n | anyone know the command to tell a cisco router to tell you its serial number? |
| 19:00.54 | bkw_ | show ver? |
| 19:00.56 | sivana | ua |
| 19:00.58 | sretooh | depends on hardware |
| 19:00.58 | sivana | ya |
| 19:01.01 | santiago | sivana, the rx/tx gains? |
| 19:01.02 | sretooh | show ver may show |
| 19:01.10 | sivana | santiago: ya |
| 19:01.42 | bjohnson | Silik0n: #break_thumbs() ? |
| 19:01.42 | Silik0n | bkw_ cpuboard ID == serial number? |
| 19:01.48 | Blackvel | jakepdev: if I use non-breaking manager API redirect call, would I use two separate socket channels to the caller api? one for sending commands? one for listening to events? |
| 19:01.50 | docelm0 | Is there gonna be a astricon in atlanta again this year? |
| 19:01.59 | Hogie | if I get rxgain below -5, it takes shorter to cancel, but they are complaining about volume then |
| 19:02.23 | jakepdev | blackvel - just 1 socket for manager is all you need |
| 19:02.23 | bjohnson | I think reduce txgain and increase rxgain |
| 19:02.28 | santiago | sivana, I've used some asterisk and zaptel patches to establish the optimal values |
| 19:02.46 | iCEBrkr | Blackvel: Yea, one connection to the manager port swill do. |
| 19:03.08 | Blackvel | and if I do so, how can I make sure what event relates to what user connection (there would be multiple concurrent connections) |
| 19:03.23 | Blackvel | hm |
| 19:03.25 | Blackvel | weird |
| 19:03.31 | Blackvel | if I send the redirect |
| 19:03.46 | jakepdev | blackvel - use channel |
| 19:03.48 | machinehd | Hogie, have you tried the same setup on a different box? When going from a cheapie motherboard to a higher end server quality motherboard I noticed a large drop in echo. It's almost non existant now |
| 19:04.02 | jakepdev | blackvel - from your AGI send |
| 19:04.04 | iCEBrkr | Blackvel: It's all in the responses. It tells ya |
| 19:04.13 | Blackvel | wouldn't my readLine() in Java block the socket, waiting for a reply (event)? I could not send e.g hangup over the manager api in that case? |
| 19:04.19 | Hogie | its actually on an abit mobo, its a decent mobo |
| 19:04.31 | iCEBrkr | Blackvel: You'll have to programatically keep track of the 'device-channel' |
| 19:04.34 | Blackvel | iCEBrkr: the events contain the uniqueue channel idds? |
| 19:04.39 | iCEBrkr | Yeah |
| 19:04.53 | machinehd | I went from an abit to an abit |
| 19:04.56 | iCEBrkr | It's kind of a pain in the ass, but once you get used to it, it's ok. |
| 19:04.56 | Hogie | I really dont have hardware I can change out on it |
| 19:05.05 | santiago | iCEBrkr, didn't you remeber how to solve that issue? |
| 19:05.07 | Blackvel | oh ok (just saw multiple REGISTER events). Maybe they miss that informations then |
| 19:05.11 | santiago | :( |
| 19:05.15 | iCEBrkr | santiago: Nope.:( |
| 19:05.30 | Blackvel | so you would like me? one socket manager api channel per client (agi) connection + 1 connection for reading the events |
| 19:05.35 | Blackvel | would do |
| 19:05.36 | iCEBrkr | Blackvel: Yea, I had to keep clearning my screen to keep track of the events while developing. |
| 19:05.52 | *** part/#asterisk Village (~Village@www.liquidcore.net) |
| 19:05.52 | *** join/#asterisk shido6 (~greg@d57-87-253.home.cgocable.net) |
| 19:05.59 | *** part/#asterisk Gand_DJ (~gandalf@ptr-207-54-104-24.ptr.terago.ca) |
| 19:06.07 | iCEBrkr | Blackvel: I did that with my 'call manager'. |
| 19:06.12 | Hogie | what the weird thing is that it goes away after 25 - 27 seconds |
| 19:06.23 | iCEBrkr | Blackvel: The call manager required you to 'sign in' and it only watched events for your extension. |
| 19:06.27 | odie_flocon_ | Man do I ever love my Hitachi Wireless phones. |
| 19:06.53 | Blackvel | iCEBrkr: something like a filter |
| 19:07.21 | syle | advertising odie: *kick* *grin* |
| 19:07.34 | odie_flocon_ | I'm ont advertising. |
| 19:07.40 | odie_flocon_ | I don't work for hitachi. |
| 19:07.52 | Blackvel | bbl |
| 19:07.55 | odie_flocon_ | I am just stating the fact that they are very good. |
| 19:08.13 | odie_flocon_ | they are the wifi sip phones. |
| 19:08.52 | santiago | Hogie, sivana, http://lists.digium.com/pipermail/asterisk-users/2004-November/071301.html |
| 19:09.07 | santiago | Hogie, sivana, that's what I did |
| 19:09.43 | odie_flocon_ | l8r all |
| 19:10.30 | Hogie | santiago: is there anyway to get that test line number? |
| 19:10.38 | Hogie | like is it a default? |
| 19:11.05 | *** join/#asterisk darby_t (~tom@dnq92.neoplus.adsl.tpnet.pl) |
| 19:11.21 | santiago | iCEBrkr, I think I'll have to use IAX |
| 19:11.58 | santiago | Hogie, nop, I call from a zap chan to another |
| 19:12.25 | santiago | Hogie, I didn't have another option |
| 19:12.51 | Hogie | call from 1 zap to another and do the tone? |
| 19:12.57 | iCEBrkr | Blackvel: http://www.cyberdyne.org/~icebrkr/cpg132/thumbnails.php?album=53 |
| 19:13.22 | santiago | Hogie, yes, calling the milliwatt app |
| 19:13.24 | santiago | to the |
| 19:14.00 | *** join/#asterisk Nukemizer (~Nuke@67.137.28.167) |
| 19:14.40 | Hogie | should I set the txgain on the channel im calling to 0 before I do that? |
| 19:15.03 | santiago | Hogie, yes |
| 19:16.50 | C4-Telecom | exit |
| 19:16.55 | C4-Telecom | sorry |
| 19:16.57 | C4-Telecom | jajajaj |
| 19:17.00 | *** part/#asterisk C4-Telecom (~sales@212.145.163.120) |
| 19:17.05 | kb1_kanobe | Hogie: you can probably get the test line by calling your telco repair and asking. |
| 19:17.54 | Hogie | I think its 817 608 0010 |
| 19:18.42 | santiago | the telco of the city where that server is, is very bad |
| 19:18.44 | kb1_kanobe | Usually it's local to your serving exchange, that way the long-distance network doesn't interfere with the levels. However it'll probably work fine. |
| 19:18.52 | *** join/#asterisk evilrabbi (evilrabbi@64.123.157.107) |
| 19:19.07 | Grooby | bah |
| 19:19.37 | *** join/#asterisk rakanuj (rakanuj@64.207.60.100) |
| 19:19.43 | Grooby | don't understand IP500 =( |
| 19:19.52 | *** join/#asterisk dant (~dan@81-86-69-213.dsl.pipex.com) |
| 19:19.54 | *** join/#asterisk salviadud (~dude@201.133.209.245) |
| 19:20.26 | rakanuj | I have a really basic question. if I am running asterisk as root (via a /etc/init.d/asterisk startup script on redhat), why can't I connect via 'asterisk -r' as another user? is there something special I have to do? I've been hunting through the docs for a while now, and maybe I'm just stupid because I can't find it. |
| 19:20.59 | dant | nufone cvs now? |
| 19:21.02 | dant | argh |
| 19:21.05 | dant | can't type |
| 19:21.09 | dant | nufone cvs down? |
| 19:21.23 | Nugget | what does nufone use cvs for? |
| 19:21.42 | ManxPower | Nugget: production servers |
| 19:22.04 | ManxPower | Wether that's a good idea or not is a very personal thing. |
| 19:22.19 | Nugget | oh, he means "nufone servers running cvs head"? |
| 19:22.25 | dant | erm nope |
| 19:22.38 | dant | nufone host one of the two cvs.digium.com public cvs servers |
| 19:22.42 | Nugget | ah, ok. |
| 19:22.46 | Nugget | I didn't know that. thanks |
| 19:23.07 | Signuts | How do I pass variables into a macro with asterisk? |
| 19:23.18 | Hogie | santiago: that ztmonitor change is in my source code of ztmonitor, but its not doing that printf... |
| 19:23.20 | Signuts | specifically using Dial(...|..|M(mymacro)) |
| 19:24.14 | santiago | Hogie, are you using the cvs version of zaptel? |
| 19:24.25 | Hogie | stable as of april 11 |
| 19:25.44 | Hogie | nm, I just commented everything out |
| 19:25.46 | Hogie | and it works now |
| 19:26.33 | iCEBrkr | rakanuj: Check www.voip-info.org for how to run asterisk as another user. |
| 19:26.54 | santiago | Hogie, it was commented? |
| 19:27.47 | rakanuj | iCEBrkr - no I'm trying to run asterisk as one user and allow access to it from another user. whether it's running as, for example, a user 'asterisk', and I connect via a user 'rakanuj' |
| 19:29.18 | *** join/#asterisk UBiQUiTY (~mike@68.160.103.76) |
| 19:31.34 | iCEBrkr | rakanuj: Doesn't sound too secure or a good idea. |
| 19:31.49 | rakanuj | maybe as root I agree. |
| 19:31.57 | Signuts | What's wrong with Dial(<my channel>|20|M(mymacro^2)), it is treating mymacro^2 as the whole context, where 2 is my argument? Is this because it's a number? |
| 19:33.28 | Signuts | Does argument passing into macros not exist is asterisk 1.0.7? |
| 19:33.53 | iCEBrkr | Signuts: That'd be lame. :) |
| 19:34.02 | ManxPower | Signuts: "show application dial" should tell you |
| 19:35.03 | Signuts | ManxPower, Well all it says is "execute macro X upon connect" it doesn't say anything about using ^ as an arguemtn delimiter. The wiki does. But the wiki fails to state which version this was introduced in. Therefore I assume it's not implemented in 1.0.7. But I want to make sure! |
| 19:36.15 | Signuts | Yes, so yet another reason to run CVS |
| 19:36.22 | iCEBrkr | Signuts: So try , |
| 19:36.25 | iCEBrkr | or just a space :) |
| 19:36.30 | drumkilla | yeah, I don't think macro args are in 1.0.x |
| 19:36.31 | ManxPower | Signuts: It's prolly a CVS-HEAD option. Do a SetVar before the Dial, then reference the variable in your Macro |
| 19:36.47 | Signuts | It's empty inside my macro |
| 19:37.13 | ManxPower | drumkilla: look at "show application park", then look at http://pastebin.ca/10440 |
| 19:37.13 | Signuts | i'll keep trying things, Thanks |
| 19:37.28 | iCEBrkr | Hey. |
| 19:37.37 | ManxPower | Signuts: That should work. |
| 19:37.37 | iCEBrkr | You can pass arguments to Macro() |
| 19:37.47 | iCEBrkr | Why wouldn't you be able to pass arguments to M() in dial? |
| 19:37.55 | ManxPower | i.e. SetVar(FNORD=bob) then ${FNORD} should be available in your macro. |
| 19:38.05 | ManxPower | iCEBrkr: not in 1.0.x apparently. |
| 19:38.12 | iCEBrkr | ManxPower: I'm running 1.0.3 |
| 19:38.19 | ManxPower | iCEBrkr: then help Signuts |
| 19:38.22 | iCEBrkr | lol |
| 19:38.37 | iCEBrkr | chrome*CLI> show version |
| 19:38.38 | iCEBrkr | Asterisk 1.0.3 built by |
| 19:38.53 | iCEBrkr | Signuts: How are you accessing your parameter in your macro? |
| 19:38.55 | *** join/#asterisk wildgoose (~edward@xunil.mail.wildgooses.com) |
| 19:38.58 | iCEBrkr | ${ARG1} |
| 19:39.32 | ScythelX | does anyone know if tftp uses another port besides 69 |
| 19:39.33 | ScythelX | udp |
| 19:39.34 | *** join/#asterisk Dovid (~hirisk@pool-141-150-135-136.mad.east.verizon.net) |
| 19:39.43 | drumkilla | yep, definitely not in 1.0 |
| 19:39.52 | iCEBrkr | drumkilla: How's it working for me then? |
| 19:40.09 | ManxPower | iCEBrkr: you are using M(macroname^parm1) in your Dial() |
| 19:40.14 | iCEBrkr | exten => s,3,Macro(stamp-callerid,${CALLERIDNUM}) |
| 19:40.14 | Nuxi | ScythelX, tftp can run over any transport. But typically uses udp 69. |
| 19:40.25 | iCEBrkr | ManxPower: I don't use the macro in the dial() application |
| 19:40.26 | ManxPower | iCEBrkr: scroll back, tht's not what they are talking about. |
| 19:40.29 | iCEBrkr | so it's probably not in dial() |
| 19:40.34 | *** join/#asterisk zotz (~zotz@24.231.32.109) |
| 19:40.45 | ManxPower | So I repeat i.e. SetVar(FNORD=bob) then ${FNORD} should be available in your macro. |
| 19:41.10 | iCEBrkr | But the thing is, a lot of code is reused so you'd think if Macro() works, Dial(..M()) would work too. :-/ |
| 19:41.19 | iCEBrkr | Yeah, what ManxPower said!! |
| 19:41.32 | rakanuj | ok I solved my problem. permissions on /var/run/asterisk.ctl I can at least manage the permissions on this and locate it somewhere safe. |
| 19:41.46 | rakanuj | the error message (in 1.0.7) sucks though. doesn't indicate the problem |
| 19:41.54 | Signuts | iCEBrkr, I am trying ${MYVAR} which is set before calling dial, which doesn't work |
| 19:42.13 | Signuts | iCEBrkr, I also tried ${ARG1}, but the thing is asterisk treats "mymacro^2" as the whole context |
| 19:42.16 | Nuxi | ScythelX, http://www.faqs.org/rfcs/rfc1350.html |
| 19:43.05 | drumkilla | are you using chan_agent or chan_local or something? |
| 19:43.10 | Signuts | I suppose I could consult the source |
| 19:43.31 | Dovid | does anyone know where we can go to bid on asterisk/pbx contracts |
| 19:43.38 | Dovid | ? |
| 19:45.09 | tzanger | Dovid: I'd subscribe to asterisk-biz |
| 19:45.27 | Grooby | damn it to hell |
| 19:45.45 | Grooby | new FC3 + asterisk with firewall disabled |
| 19:45.52 | Grooby | i can't get xlite to login |
| 19:46.22 | Markie | i need to make macros |
| 19:46.26 | Markie | blah |
| 19:48.47 | Ali42 | can i cnnect a md110 digital card to to te410 ? |
| 19:49.35 | Dovid | where can i get AMP ? |
| 19:49.54 | Dovid | Tzanger: whats the link for asterisk biz ? |
| 19:50.02 | Dovid | lists.digium ? |
| 19:50.02 | tzanger | lists.digium.com |
| 19:50.06 | jontow | ~AMP |
| 19:50.09 | jbot | somebody said amp was an Audio MPEG Player. [non-free], or http://amp.coalescentsystems.ca/ |
| 19:50.09 | jontow | (?) |
| 19:50.15 | jontow | ah.. wrong |
| 19:50.21 | jontow | check the wiki |
| 19:51.01 | *** join/#asterisk PCadach (~paul@www.east.telecom.kz) |
| 19:51.30 | Ali42 | Is it possible to connect an ericsson digital card to a pci card? |
| 19:51.40 | Ali42 | no one can help with that? :) |
| 19:52.39 | shido6 | erf? |
| 19:52.43 | shido6 | what is that? |
| 19:53.21 | *** join/#asterisk nwhit (~nwhit@65.107.59.67.ptr.us.xo.net) |
| 19:54.34 | nwhit | I am looking at connecting asterisk to a voice term provider. They are asking what commercial product (or emulated commercial product) I am using for session border control. How should I respond? |
| 19:55.20 | iCEBrkr | nwhit: Tell'm it's none of their business. :P |
| 19:55.35 | nwhit | i could... but i would like to work with them |
| 19:55.40 | nwhit | :> |
| 19:56.27 | iCEBrkr | Oh. |
| 19:57.41 | nwhit | does asterisk support session border control? |
| 19:57.53 | opus_ | i don't think so |
| 19:58.21 | opus_ | thats BGP right? |
| 19:59.49 | *** join/#asterisk guugmember (~guugmembe@200.6.222.103) |
| 20:01.48 | freat | that's Border Gateway Protocol |
| 20:01.49 | *** join/#asterisk jtodd (~jtodd@adsl-67-120-209-21.dsl.sntc01.pacbell.net) |
| 20:02.03 | freat | BGP is not something you should need to worry about unless you're an ISP |
| 20:02.34 | nwhit | i am an isp |
| 20:03.49 | *** join/#asterisk odie_flocon_ (~Odie@ptr-64-201-182-211.ptr.terago.ca) |
| 20:04.07 | rakanuj | a session border controller is essentially a sip proxy or an h323 gatekeeper |
| 20:04.38 | rakanuj | it manages the calls and routes them for termination according to rules you provide. you can do session border control with asterisk, but you have to pass the voice through it |
| 20:04.59 | rakanuj | whereas others (nextone for example) don't require that. |
| 20:05.01 | iCEBrkr | Yea, but BGP doesn't have anything to do with VoIP. I mean, I guess if it's TCP-IP it would.. but blah. |
| 20:05.08 | iCEBrkr | You knwo what I mean. |
| 20:05.20 | rakanuj | BGP has to do with routing |
| 20:05.24 | nwhit | ok... that make sense |
| 20:05.32 | iCEBrkr | rakanuj: Not call routing. :P |
| 20:05.36 | rakanuj | correct |
| 20:05.38 | rakanuj | ip routing |
| 20:05.42 | iCEBrkr | hehe |
| 20:05.47 | rakanuj | sheesh ;P |
| 20:05.51 | iCEBrkr | Yea, I know waht you mean.. |
| 20:06.21 | mutilator | cvs dead? |
| 20:06.21 | rakanuj | oh dear |
| 20:06.27 | mutilator | or just me? |
| 20:06.28 | nwhit | sbc relates to cdr, correct? |
| 20:06.32 | rakanuj | is that the fifth time that's been asked? |
| 20:06.41 | nwhit | it follows the call so that it can be billed correctly |
| 20:06.45 | iCEBrkr | Isnt' that like DUNDi? |
| 20:06.48 | rakanuj | cdr = call detail record. session border controllers typically spit out call detail records for billing |
| 20:06.49 | nwhit | or am i wrong? |
| 20:06.56 | rakanuj | asterisk spits them out too |
| 20:07.23 | mutilator | cvs [checkout aborted]: connect to cvs.digium.com(66.225.202.81):2401 failed: Connection refused |
| 20:07.28 | rakanuj | a CDR is a record of where the call originated from, terminated to, how long it lasted, etc. |
| 20:07.42 | rakanuj | it's usually different depending on the vendor. kind of a pain actually |
| 20:07.51 | nwhit | i am familiar with the cdr that asterisk supports... |
| 20:08.15 | nwhit | is it reasonable to do sbc through asterisk or does it make more sense to buy a commercial product? |
| 20:08.28 | rakanuj | commercial products can cost as much as $30k |
| 20:08.30 | rakanuj | or more |
| 20:08.38 | rakanuj | so yes it's reasonable ;) |
| 20:08.46 | rakanuj | some people don't like it, though |
| 20:08.52 | iCEBrkr | SBC through Asterisk? |
| 20:09.06 | iCEBrkr | What's SBC besides a phone comapny? |
| 20:09.09 | nwhit | so for a small operation, it would make more sense to route the calls through asterisk and let it handle the cdr |
| 20:09.17 | rakanuj | SBC = session border control |
| 20:09.21 | iCEBrkr | Oh duh. |
| 20:09.23 | nwhit | ~sbc |
| 20:09.24 | rakanuj | yeah I do that here |
| 20:09.39 | nwhit | ok... |
| 20:10.25 | iCEBrkr | I'm going to have to figure out why my voicepulse calls are being rejected. (inbound) |
| 20:10.32 | iCEBrkr | Just noticed that last night. |
| 20:10.49 | iCEBrkr | It used to work, I musta hosed something. |
| 20:10.54 | shido6 | that sux, iCEBrkr |
| 20:11.22 | iCEBrkr | Yeah, cuz I don't get but maybe 2 calls a month on that line. |
| 20:11.45 | iCEBrkr | So I dunno when it broke. |
| 20:12.20 | shido6 | I didnt go anywhere near the voicepulse dells since last sunday |
| 20:12.25 | shido6 | so u cant blame me this time |
| 20:12.33 | iCEBrkr | I'm assuming it can only be in 2 places.. iax.conf and my extension.conf |
| 20:12.42 | iCEBrkr | shido6: hehe |
| 20:12.55 | iCEBrkr | outbound works fine. |
| 20:13.16 | *** join/#asterisk rderr (~chatzilla@rrcs-24-227-114-94.se.biz.rr.com) |
| 20:14.51 | ManxPower | Whoo! Whoo! I just left my first orgy^H^H^H^Hgroup voicemail! |
| 20:16.00 | rderr | Ok, I'm at the end of my rope. I want to update from 1.0.1 to 1.0.7 but for some reason the playback and background dial commands either don't make any audio or distorted audio for loud recordings (like tt-monkeys). Nothing else with the configuration has changed, the playback and background commands don't spit out any error, and calls work without any problems. Does anyone have an idea? |
| 20:16.21 | Uther_P | geg |
| 20:16.22 | Uther_P | h eh |
| 20:18.33 | rderr | nothing? Noone? |
| 20:18.43 | ManxPower | rderr: hold on a min |
| 20:18.47 | rderr | Thanks |
| 20:25.43 | *** join/#asterisk flot (CCCP@213.152.157.68) |
| 20:31.05 | *** join/#asterisk Goshen (~Goshen@70-57-80-147.slkc.qwest.net) |
| 20:31.56 | *** join/#asterisk RoyK (~roy@143.80-202-166.nextgentel.com) |
| 20:33.16 | *** join/#asterisk kn0x (~nunya@ppp-68-251-90-102.dsl.chcgil.ameritech.net) |
| 20:33.22 | kn0x | hey |
| 20:33.28 | Goshen | hi |
| 20:33.38 | kn0x | just a fast question |
| 20:33.44 | kn0x | in meetme.conf... |
| 20:33.52 | kn0x | say i want a conference with no pin |
| 20:33.56 | kn0x | but an admin pin |
| 20:34.32 | kn0x | would i use: conf => 1, , 1234 |
| 20:34.47 | kajtzu | try conf => 1234,,5678 but I don't know if that's a legal choice |
| 20:36.45 | kn0x | well thats not going to work at all |
| 20:36.58 | kn0x | because it going to prompt for a pin code either way |
| 20:37.41 | kajtzu | heh |
| 20:39.06 | kn0x | no i didnt mean it in a bad way i mean i was wrong too |
| 20:40.31 | *** join/#asterisk meppl (mephisto@pD9E68F34.dip.t-dialin.net) |
| 20:43.31 | dercol | hello, I've a problem with Zultys LIPZ4 Softphone and CVS version of asterisk, a problem that doesn't exist with stable 1.0.7 version, anyone can help me? |
| 20:44.03 | ManxPower | rderr: You still here? |
| 20:44.08 | rderr | Ya |
| 20:44.40 | ManxPower | rderr: One of the major changes between 1.0.1 and 1.0.7 was significant bugs were fixed in the codec ordering stuff. What codec do your phones use? |
| 20:44.53 | rderr | ulaw |
| 20:45.40 | ManxPower | Make sure you don't have a bandiwdth= line in sip.conf or iax.conf. Make sure you have disallow=all and allow=ulaw and no other allow= lines. Then stop and start Asterisk just to be SURE. |
| 20:46.24 | tzanger | that whole bandwidth= option should be trashed |
| 20:46.30 | tzanger | just use allow/disallow |
| 20:46.32 | ManxPower | tzanger: Yeah! |
| 20:46.32 | rderr | I had the bandwidth option...Let me try that |
| 20:46.33 | *** join/#asterisk file (~file@mctn1-3974.nb.aliant.net) |
| 20:46.55 | mutilator | abort halt = stop 'stop gracefully'? |
| 20:47.00 | ManxPower | rderr: /msg me to let me know if it worked and tell the channel too. Don't expect a response to the /msg, I'll respond on the channel if I can help more. |
| 20:47.11 | ManxPower | mutilator: no abort halt = stop now |
| 20:47.35 | ManxPower | stop when convienent will stop when there are no active calls. |
| 20:47.44 | mutilator | yea |
| 20:47.47 | mutilator | what if i want to cancel it? |
| 20:47.54 | mutilator | that not what about halt does? |
| 20:47.56 | ManxPower | mutilator: I don't think you can. |
| 20:48.12 | mutilator | mmk, just saw it in the list, figured i'de ask |
| 20:48.21 | mutilator | not in my current running version |
| 20:50.49 | tzanger | ManxPower: huh? |
| 20:50.59 | ManxPower | rderr: Well, it was worth a try. I (and MANY others) use 1.0.x with no audio problems. |
| 20:51.06 | tzanger | abort halt is not "stand down" for "stop gracefully" ?? |
| 20:51.18 | ManxPower | tzafrir: I could be wrong, I guess. |
| 20:51.20 | rderr | I know it has to be a simple setting somwhere... |
| 20:51.38 | rderr | I'll just have to rebuild the thing from scratch |
| 20:51.44 | ManxPower | rderr: copy your sip.conf (or your iax.conf) to pastebin.ca |
| 20:51.53 | ManxPower | sans passwords, of course. |
| 20:52.00 | rderr | Ok...One sec.. Of course :) |
| 20:53.55 | *** join/#asterisk MattB2 (~m@pcp01068561pcs.andrsn01.tn.comcast.net) |
| 20:55.38 | *** part/#asterisk fedor (~fbond@office.tura.ru) |
| 20:56.21 | MattB2 | hi all |
| 20:56.22 | PTG1234 | drumkilla: you awake? |
| 20:56.24 | MattB2 | bit quiet in here |
| 20:56.27 | PTG1234 | anyone here use sip from asterisk outbound? |
| 20:56.34 | MattB2 | yup |
| 20:56.38 | Signuts | surely a lot do! :) |
| 20:56.45 | MattB2 | they do |
| 20:56.49 | MattB2 | and stop calling me shirley |
| 20:56.50 | MattB2 | etc etc |
| 20:56.53 | MattB2 | far too much bad tv |
| 20:57.08 | kajtzu | very bad tv |
| 20:57.08 | kajtzu | :) |
| 20:57.10 | *** join/#asterisk shodan (~shodan@216.113.99.141) |
| 20:57.10 | Signuts | hehe |
| 20:57.15 | PTG1234 | Well who in here does |
| 20:57.22 | MattB2 | I does |
| 20:57.23 | Signuts | I do |
| 20:57.32 | PTG1234 | you do outgoing FROM asterisk via sip, not iax? |
| 20:57.44 | Signuts | Correct-o-mundo |
| 20:57.46 | MattB2 | sip phones -> asterisk -> zap -> outside world |
| 20:57.54 | PTG1234 | MattB2: that would not be it |
| 20:58.00 | MattB2 | ohh |
| 20:58.02 | PTG1234 | Signuts: who is your provider? |
| 20:58.02 | MattB2 | sorry gotcha |
| 20:58.06 | MattB2 | ast -> sip -> outside |
| 20:58.09 | MattB2 | ah not me then |
| 20:58.13 | three55ml | Anyone here use Broadvox for carrier level services? I'm wondering what their minimum commitment is. |
| 20:58.21 | Signuts | I do Sip => asterisk => SIP internal calling |
| 20:58.27 | PTG1234 | three55ml: alot better choices out there then them :) |
| 20:58.37 | PTG1234 | Signuts: did you apply the patch to get it to work? |
| 20:58.51 | three55ml | I was just curious |
| 20:59.06 | Signuts | PTG1234, I'm running CVS, what patch? |
| 20:59.31 | PTG1234 | Sig: you need a patch for it ti work, unless its been put into cvs |
| 20:59.38 | PTG1234 | i run stable, so it won't help either way :) |
| 20:59.38 | MattB2 | I've got a dull over-asked ZAP question |
| 21:00.08 | MattB2 | X100P doesn't hangup but I do have disconnect supervision on the lines (tested with trusty backlit phone) |
| 21:00.13 | MattB2 | hanuponpolarityreverse is set |
| 21:00.15 | *** join/#asterisk fabioFVZ (~fabio@213-92-104-168.f5.ngi.it) |
| 21:00.16 | MattB2 | but just doesn't work |
| 21:01.04 | Signuts | well * fork, i've gotta hit the road. |
| 21:01.40 | ManxPower | MattB2: Um, don't do that. set Kewlstart in zapata.conf and zaptel.conf and no other disconnect options. |
| 21:03.24 | *** join/#asterisk ngb (~ngb@200.49.156.89) |
| 21:03.36 | ngb | some one can helpme with x100p ? |
| 21:04.04 | ngb | when i wanna make a phone call... the asterisk return no channel type registred for zap |
| 21:04.06 | tzanger | ngb: did you buy it from digium or a digium reseller? |
| 21:04.20 | ManxPower | ngb: you have to install zaptel BEFORE you build/install Asterisk |
| 21:04.21 | ngb | digium reseller, in argentina |
| 21:04.29 | tzanger | ngb: cool, you win. :-) |
| 21:05.03 | ngb | first i have to install the zaptel and later the asterisk ? |
| 21:05.07 | tzanger | unfortunately I have to go... gotta pick up my kids and get some supper :-) |
| 21:05.12 | tzanger | ngb: yes |
| 21:05.18 | tzanger | otherwise asterisk doesn't know that zap exists |
| 21:05.26 | ngb | all right |
| 21:05.28 | ngb | thanks |
| 21:05.29 | ngb | :) |
| 21:05.55 | *** join/#asterisk tzafrir_laptop (~tzafrir@bzq-179-40-134.cust.bezeqint.net) |
| 21:06.01 | *** join/#asterisk Shido (~greg@d57-87-253.home.cgocable.net) |
| 21:06.03 | ngb | if i reinstall it |
| 21:06.18 | ngb | this run without problems ? |
| 21:06.34 | ngb | i have installed the asterisk. and later i install the x100p and zaptel |
| 21:06.46 | ngb | if now i reinstall without uninstall the asterisk |
| 21:06.54 | ManxPower | install zaptel, then just reinstall asterisk. |
| 21:06.55 | ngb | it run without problems ? |
| 21:06.59 | ngb | ok |
| 21:07.08 | ManxPower | do a "make clean; make install" in the asterisk source directory after zaptel is installed. |
| 21:07.09 | anthm | more like re compile it |
| 21:07.30 | MattB2 | ManxPower: fxsks set everywhere, disabled busydetect, hanuponpolarity and it still won't work :( |
| 21:07.39 | Shido | :( |
| 21:07.44 | Shido | pastebin.ca |
| 21:08.02 | MattB2 | busydetect at least works when the nasty clangs come along |
| 21:08.08 | MattB2 | but gettnig occasional false disconnects during calls |
| 21:08.33 | ManxPower | MattB2: in the USA battery is removed from the line. The USA doesn't (normally) reverse the battery |
| 21:09.10 | ManxPower | busydetect is just an alias for randomlydisconnectcalls |
| 21:09.11 | MattB2 | ok, I could see that when I had the analog phone plugged in - about 8 seconds after hangup the lights faded out for a few seconds and came back |
| 21:09.13 | ngb | if asterisk run good with the x100p when i make a lsmod this print wcfxs used = 1 ???' |
| 21:09.24 | MattB2 | does the X100P not detect battery drop then? |
| 21:09.33 | ManxPower | MattB2: it's supposed to. |
| 21:09.56 | ManxPower | All 4 X100Ps I have in 4 different systems detects hangups fine with just kewlstart |
| 21:10.18 | MattB2 | are you using original X100Ps or clones? |
| 21:10.20 | MattB2 | I've got clones |
| 21:10.23 | MattB2 | maybe the issue? |
| 21:11.17 | ManxPower | both, I think. |
| 21:11.21 | MattB2 | hmm |
| 21:11.27 | ManxPower | no, all the clones are out of service since we didn't need them anymore. |
| 21:11.29 | ngb | some one speack spanish ? |
| 21:11.39 | MattB2 | damn, I'm running out of ideas :( |
| 21:11.42 | dercol | please, can one take a look to sessions dump of my problem with Zultys Softphone and CVS version of asterisk? the link is ftp://popolo.yacme.com/ercolani/pub/asterisk_problem.tgz |
| 21:13.09 | pjz | if I've got an asterisk process running in the background, can I get a cli to it? |
| 21:13.16 | MattB2 | asterisk -r |
| 21:13.36 | pjz | thanks |
| 21:16.41 | ManxPower | MattB2: you can modprobe wcfxo debug=1 and I think it will log debug stuff to syslog |
| 21:17.10 | *** join/#asterisk meppl (mephisto@pD9E68F34.dip.t-dialin.net) |
| 21:17.33 | MattB2 | ManxPower: thx, was just trying to find out some more debug info for the zap drivers! |
| 21:18.46 | *** join/#asterisk tessier (~treed@210.245.97.237) |
| 21:18.54 | pjz | if some of my ports change their plugged-in state, how do I get asterisk to rescan them? |
| 21:19.11 | pjz | will a reload do it? or do I need to restart asterisk itself? |
| 21:19.39 | *** join/#asterisk Nightwriter (~james@209.11.127.13) |
| 21:19.39 | ManxPower | I love this coffee! It's like expresso, but you brew it normally. |
| 21:20.59 | Shido | does it dragon punch you to wake the frog up, ........RIGHT NOW!!! |
| 21:21.18 | *** join/#asterisk lilo (lilo@levin-pdpc.staff.freenode) |
| 21:21.52 | *** join/#asterisk Nethab (~chatzilla@adsl-67-113-141-170.dsl.sntc01.pacbell.net) |
| 21:22.23 | Nethab | yawn hello |
| 21:23.55 | *** join/#asterisk meppl (mephisto@pD9E68F34.dip.t-dialin.net) |
| 21:27.02 | ngb | <PROTECTED> |
| 21:27.02 | ngb | Apr 27 18:23:43 NOTICE[10012]: app_dial.c:746 dial_exec: Unable to create channel of type 'Zap' |
| 21:27.04 | ngb | :( |
| 21:27.28 | ngb | why i have this prboem ? |
| 21:27.41 | brad- | Can asterisk tell which number was called to reach it? And act on that? |
| 21:28.32 | brad- | (Single PSTN line, multiple numbers w/ distinctive ring - I want to avoid the hassle of ring detection) |
| 21:28.41 | ngb | someone can help me ? |
| 21:29.18 | *** join/#asterisk vpp (~noone@host-83-146-50-131.bulldogdsl.com) |
| 21:29.21 | vpp | hi |
| 21:30.50 | ngb | pleaseeeeeeeeee!!! |
| 21:30.54 | *** part/#asterisk Uther_P (~uther_p@66.180.120.83) |
| 21:30.56 | ngb | i need help |
| 21:30.57 | ngb | :'( |
| 21:31.24 | MattB2 | ngb: can you verify that channel 12 of ZAP exists? |
| 21:31.30 | MattB2 | ngb: PS I'm no zap expert ;) |
| 21:31.31 | ngb | [incoming] |
| 21:31.31 | ngb | exten => s,1,Dial(SIP/Phone1&SIP/Phone3,20,tr) |
| 21:31.31 | ngb | exten => s,2,Voicemail,u1000 |
| 21:31.31 | ngb | exten => s,102,Voicemail,b1000 |
| 21:31.38 | ngb | [outgoing] |
| 21:31.39 | ngb | exten => _0X.,1,Dial,Zap/1/${EXTEN:1} |
| 21:31.54 | MattB2 | bah ignore my first comment |
| 21:32.00 | MattB2 | my second comment is very true ;) |
| 21:32.04 | pjz | how do you determine which context gets hit when someone dials their extension? |
| 21:32.05 | Shido | pastebin.ca ngb :) |
| 21:32.18 | Shido | pastebin.ca ngb :) |
| 21:32.21 | ngb | ? |
| 21:32.35 | Shido | pjz you specify that in the dialplan |
| 21:32.41 | Shido | in extensions.conf, pjz |
| 21:32.57 | MattB2 | ngb: are all your ZAP drivers etc running properly? |
| 21:33.26 | MattB2 | ztcfg -vvd |
| 21:33.27 | ngb | MattB2 yes |
| 21:33.32 | MattB2 | (and don't psot the respone in here!) |
| 21:33.35 | MattB2 | *post |
| 21:33.55 | ngb | Channel map: |
| 21:33.55 | ngb | Channel 01: FXS Kewlstart (Default) (Slaves: 01) |
| 21:33.55 | ngb | 1 channels configured. |
| 21:34.02 | *** join/#asterisk felipeao (~felipeao@berg.viaip.com.br) |
| 21:34.11 | ngb | d is a invalid option |
| 21:34.19 | MattB2 | k |
| 21:34.30 | MattB2 | you must have an older version of the zap drivers |
| 21:34.36 | MattB2 | or maybe a different version at least ;) |
| 21:34.47 | ngb | i download it from asterisk.org |
| 21:34.48 | MattB2 | and you def have a line plugged into the zap card? can you receive incoming calls? |
| 21:34.50 | ngb | web site |
| 21:35.05 | felipeao | Im trying to connect 2 sip clients using asterisk as the server, but I always get the following error: |
| 21:35.10 | ngb | yes.. i have the line and mi phone |
| 21:35.14 | ngb | in x100p |
| 21:35.35 | ngb | the line is connected to a phone central |
| 21:35.38 | felipeao | Apr 27 17:13:46 WARNING[4250]: chan_sip.c:694 retrans_pkt Maximum retries exceeded on call 54b5697f6f1e9a5b3043a5dc2b9b67d7@192.168.3.1 for seqno 104 (Non-critical Request) ........ |
| 21:35.45 | felipeao | whats wrong? |
| 21:35.45 | ngb | sorry, my english is very bad |
| 21:35.53 | vpp | anyone use astrisk @ home with h232? |
| 21:35.57 | *** join/#asterisk tld (~tld@80.203.70.227) |
| 21:35.58 | MattB2 | mgb: can you receive incoming calls? |
| 21:36.03 | vpp | *h323 |
| 21:36.11 | *** join/#asterisk Nuxi (~nuxi@lin-dsl-static-206-222-194-115.inetnebr.com) |
| 21:36.48 | pjz | if an extension has a lot of echo, can I fix it? |
| 21:37.09 | Shido | is the call going through only YOUR gear? |
| 21:37.13 | Nethab | felipeao, does the call connect? or does it fail, it could be a nat problem |
| 21:37.18 | ngb | MattB2 i receive a phone call from the pstn in my internal... but it not is an intertal of asterisk |
| 21:37.47 | MattB2 | mgb: are you saying that your asterisk is plugged into another PBX, not an outside line? |
| 21:37.47 | ngb | i have a phone central with 4 pstn lines... an i have an extention in this central |
| 21:37.52 | tld | Anyone know if the Cisco SIP version 2-series firmware has issues with MD5 DIGEST authentication? |
| 21:37.56 | MattB2 | central? what's a central? |
| 21:38.00 | ngb | yes |
| 21:38.05 | ngb | central = pbx |
| 21:38.06 | ngb | sorry |
| 21:38.07 | pjz | Shido: yes. From a softphone to a Uniden UIP200 |
| 21:38.10 | MattB2 | k np :) |
| 21:38.16 | pjz | Shido: via my asterisk pbx |
| 21:38.30 | Shido | get a noise cancelling mic on the softphone |
| 21:38.32 | ngb | i have a pbx with 4 lines... pbx panassonic |
| 21:38.44 | pjz | well, it's the Uniden that has the echo |
| 21:38.46 | MattB2 | ngb: have you tried plugging asterisk direct to an outside line? |
| 21:38.51 | ngb | and i've an extention in this pbx where i have connected my asterisk |
| 21:38.52 | Shido | fxo or fxs or t1 or e1, ngb ? |
| 21:38.59 | tld | I can *highly* recomment Jabra BT250v as a headset for VoIP. |
| 21:39.11 | ngb | MattB2 not in an intertal extention of pbx analogic |
| 21:39.17 | syle | hopefully gmail will put aol, yahoo, and hotmail out of business, then they can stop making money off sueing people |
| 21:39.40 | syle | how low can you get lol |
| 21:40.00 | Nethab | who is suing who |
| 21:40.01 | Himeko | tld you would recommend one for cell phone too? |
| 21:40.06 | Himeko | i was thinking of getting one |
| 21:40.08 | ngb | lsmod |
| 21:40.08 | ngb | wcfxo 7744 0 (unused) |
| 21:40.09 | ngb | zaptel 215552 4 [wcfxo] |
| 21:40.11 | felipeao | this is not a nat problem since both clients and the server are on a lan... the clients are able to make the call, but when either sides accept the call, the clients either hang or just close |
| 21:40.19 | felipeao | Nethab, this is not a nat problem since both clients and the server are on a lan... the clients are able to make the call, but when either sides accept the call, the clients either hang or just close |
| 21:40.22 | tld | Himeko: Yeah, it's terrific. |
| 21:40.32 | syle | man you should checkout aol's website, their new money making scheme is sueing anyone who spams them for millions of dollars |
| 21:40.34 | syle | what a joke |
| 21:40.38 | ngb | i've this configuration http://users.pandora.be/Asterisk-PBX/InstallWildcard.htm |
| 21:40.51 | tld | Himeko: Only think is that the windows bluetooth drivers are horrible, so it might take you a bit of time to figure out. |
| 21:41.05 | Nethab | are they trying to re-invite? |
| 21:41.15 | tld | Himeko: Magic trick is to replace the Windows stock drivers with pretty much any WIDCOMM driver, if you've got a WIDCOMM controller. |
| 21:41.25 | Himeko | i have no probs with the new bluetooth stack that is included in winxp sp2 |
| 21:41.36 | syle | no worries aol, soon noone will use your email service, then who you gonna sue lol |
| 21:41.37 | MattB2 | ngb: I'm no zap guru, there's not much more I can suggest :( |
| 21:41.39 | tld | Himeko: XP SP2 provides it's own Bluetooth support, which will by default override anything else. Problem is, it doesn't support Bluetooth Headset. |
| 21:41.41 | Himeko | i have all sort of probs with widcomm software |
| 21:41.48 | tld | Himeko: Feel free to bug me if you do get one, and it doesn't work. |
| 21:41.49 | felipeao | Nethab, asterisk does that 1 - 5 retries by itself... I already tried to use canreinvite=no, but the calls are still failing |
| 21:41.50 | *** join/#asterisk Blissex (~Blissex@82-69-39-138.dsl.in-addr.zen.co.uk) |
| 21:41.56 | ngb | MattB2 ok |
| 21:42.13 | MattB2 | ngb: from me guessing sounds like a problem with ZAP drivers, X100P or your panasonic PBX not liking Asterisk |
| 21:42.16 | MattB2 | sorry! |
| 21:42.24 | Himeko | well, i haven't had wincomm probs with head sets |
| 21:42.29 | Himeko | just with nokia phones |
| 21:42.29 | Nethab | have you tried sip history or sip debug to see exactly which messages are being dropped |
| 21:42.41 | felipeao | Nethab, I fell so lost and disappointed... :/ |
| 21:42.50 | felipeao | fell=feel |
| 21:43.19 | Shido | felipeao, what?! |
| 21:43.25 | tld | Himeko: YMMV seems to be the Windows Bluetooth rule of thumb. :( |
| 21:43.38 | felipeao | Nethab, yes, I did the sip debug, and the only thing I saw were the 5 retry attempts |
| 21:43.43 | Nethab | next question, are the clients trying to use a STUN server |
| 21:43.45 | Himeko | what is YMMV? |
| 21:43.52 | Shido | stop using friends, felipeao |
| 21:43.53 | Shido | use user and peer |
| 21:43.57 | Shido | in sip.conf |
| 21:43.59 | tld | Your Mileage May/Might Vary |
| 21:44.04 | felipeao | Nethab, no they are not.. I dont even know whats a stun server |
| 21:44.19 | Himeko | ic |
| 21:44.25 | felipeao | Shido, Yes, im using them as FRIENDS... how did u guess? |
| 21:44.39 | Nethab | a stun server is what the soft phones use to determine what their real Nat'd IP is |
| 21:44.45 | Shido | because Ive been doing this for 3 yrs, felipeao ;) |
| 21:45.07 | Himeko | tld who might have them for the best price? |
| 21:45.14 | felipeao | Shido, hope you're right... lets give it a try >>> |
| 21:45.28 | tld | Himeko: I have no idea, sorry. :( |
| 21:45.34 | Nethab | I use friends no problem, it has to be more complicated than that |
| 21:45.37 | tld | Himeko: Got mine in a store here in Norway. |
| 21:45.58 | Himeko | ah, ic |
| 21:46.16 | *** join/#asterisk Stormy (~Stormy@Stormy.member.p3m) |
| 21:46.25 | Stormy | has anyone tryed to compile asterisk on alpha? |
| 21:46.54 | Shido | you CAN use friends, Nethab but they cause more problems then solutions |
| 21:47.11 | MattB2 | ManxPower: dunno if you're interested but I enabled debugging and I'm seeing something I don't understand! |
| 21:47.25 | MattB2 | ManxPower: when I make a call, it states "NO BATTERY" |
| 21:47.32 | Nethab | i agree a seperate peer/user for providers (origination/termination) but for internal phones i use friends |
| 21:47.47 | MattB2 | ManxPower: when I forcefuly hang up the call (by hanging up the SIP phone) I get "BATTERY" |
| 21:47.49 | MattB2 | nothing inbetween |
| 21:47.59 | ManxPower | MattB2: I've never actually seen the debug info. |
| 21:48.04 | MattB2 | ah lol |
| 21:48.04 | Nethab | A salt and battery |
| 21:48.05 | MattB2 | well |
| 21:48.19 | MattB2 | I'd expect to see NO BATTERY then BATTERY after about 8 s |
| 21:48.21 | MattB2 | but I don't |
| 21:48.24 | MattB2 | never mind, something to investigate |
| 21:48.47 | dercol | Found the problem with Zultys LIPZ4 and CVS version of asterisk: Zultys wasn't natted, nat=no in sip.conf have to be substituted with nat=never in the CVS version of sip.conf. Thankyou to ngrep! |
| 21:49.22 | *** join/#asterisk meppl (mephisto@pD9E68F34.dip.t-dialin.net) |
| 21:51.30 | felipeao | Nethab, Shido, I tried to use USER / PEER instead of just friends, but im getting the very same error |
| 21:51.50 | felipeao | <PROTECTED> |
| 21:52.25 | *** join/#asterisk pdracevich (~bob@smtp.aucklandtax.co.nz) |
| 21:52.38 | pdracevich | moring all |
| 21:53.13 | pjz | argh. How do I deal with a ton of echo on my uniden UIP200 when talking to a softphone? |
| 21:53.46 | pdracevich | I have a CDR billing question, I have two servers I want to be able to place a call and the CDR information, to be copyed over to the next cdr so i can bill/trace the call |
| 21:54.44 | felipeao | Nethab, anything else u might think of? |
| 21:55.02 | Nethab | i think the call is being sent to the wrong place |
| 21:55.17 | Nethab | either because one client is mis reporting their IP address |
| 21:55.19 | *** join/#asterisk Romik (~romik@217.16.239.37) |
| 21:55.43 | *** join/#asterisk jas_williams (~Jason@host217-44-216-142.range217-44.btcentralplus.com) |
| 21:56.03 | Nethab | and the best way to see is to watch with tcpdump or go through the sip debug messages |
| 21:57.18 | Nethab | what soft phones are you using |
| 21:57.21 | felipeao | Nethab, im using Kphone as the sip client |
| 21:57.31 | *** join/#asterisk likwid-- (likwid@nc-65-40-170-92.dyn.sprint-hsd.net) |
| 21:57.42 | Nethab | and they both regisrer correctly |
| 21:57.45 | felipeao | Nethab, I already tried other softphones with no success. Im a noob in the asterisk area |
| 21:58.05 | ManxPower | Whoo! Whoo! OrgyVoicemail is now in production! |
| 21:58.07 | felipeao | yes, they do register correctly.. I get the OK info from both the server and clients |
| 21:58.22 | Nethab | what does sip show peers say their IP addresses are |
| 21:58.30 | ManxPower | I think it's time for a Guinness |
| 21:59.22 | ManxPower | felipeao: All Softphones Suck. |
| 21:59.32 | ManxPower | (c)2005, ManxPower |
| 21:59.40 | felipeao | Clients -> 192.168.3.50 and 192.168.3.200 |
| 21:59.45 | felipeao | the * is 192.168.3.1 |
| 22:00.07 | Nethab | for giggles i would try to have to phones dial directly by IP address, can you try that, bypass the asterisk altogether |
| 22:00.34 | felipeao | good idea, let me see |
| 22:05.12 | pjz | how can I ask asterisk what codecs it supports? |
| 22:05.12 | Nethab | wait did someone say Orgy? |
| 22:05.23 | *** join/#asterisk lilo (lilo@levin-pdpc.staff.freenode) |
| 22:05.33 | Nethab | show modules |
| 22:05.37 | Nethab | and look for codec |
| 22:06.11 | felipeao | Nethab, they work fine if I dont use asterisk |
| 22:06.26 | felipeao | Nethab, just connected both of them without asterisk |
| 22:06.40 | Nethab | ok that's a good start |
| 22:07.00 | felipeao | Nethab, what else can I do? |
| 22:07.18 | Nethab | it must be one of your settings |
| 22:07.48 | felipeao | sip.conf ? extensions.conf ? |
| 22:08.00 | pjz | where can I find docs on all allowable settings in sip.conf? |
| 22:08.16 | felipeao | Nethab, sip.conf ? extensions.conf ? |
| 22:08.19 | Nethab | sip.conf for starters, where did you get the settings for your softphones |
| 22:08.32 | felipeao | voip-info.org |
| 22:08.47 | *** join/#asterisk bajanman (~william@cp66-203-194-32.cp.telus.net) |
| 22:09.25 | Nethab | http://www.voip-info.org/wiki-Asterisk+phone+Kphone did you use this? |
| 22:09.51 | felipeao | yes, that the exact location |
| 22:10.55 | *** join/#asterisk jsolares (~jsolares@200.30.141.85) |
| 22:11.40 | felipeao | Nethab, I did exactly as its said on the site. And it works as the site said (they go until the registration part). The problem is afterwards, after dialing the other end, when the clients hang or crash |
| 22:11.51 | Nethab | can you get either phone to call an internal extension like voicemail on asterisk |
| 22:12.15 | felipeao | Nethab, I did with sucess... Listened to the sample mp3s and stuff |
| 22:12.44 | Nethab | and you have canreinvite = no in both phones sip.conf settings |
| 22:13.13 | felipeao | Nethab, I tried with and without it... now its set to yes, but I will change it to no and retest |
| 22:13.22 | *** part/#asterisk ScaredyCat (~ScaredyCa@i169173.upc-i.chello.nl) |
| 22:13.40 | *** part/#asterisk pdracevich (~bob@smtp.aucklandtax.co.nz) |
| 22:14.27 | *** part/#asterisk Grooby (~Grooby@12.22.232.212) |
| 22:15.22 | pjz | which has better voice quality? ulaw or alaw? |
| 22:15.23 | Nethab | if that doesn't work try putting localnet = 192.168.3.0/24 in the [general] section of sip.conf |
| 22:15.24 | syle | crap i forgot what DID stands for |
| 22:15.30 | Nethab | ulaw and alaw are the same |
| 22:15.32 | pjz | Direct Inward Dial |
| 22:15.44 | syle | ty |
| 22:15.49 | Nethab | just different methods to the same format |
| 22:15.49 | jas_williams | pjz, there both the same quality |
| 22:16.07 | felipeao | Nethab, I did put localnet= 192.168.0.0/24 ....... Im still testing the canreinvite->>> |
| 22:16.09 | pjz | anyone got a uniden UIP200 ? |
| 22:16.17 | Nethab | with the 3? or with 0 |
| 22:16.27 | Nethab | cause your localnet is 3.0/24 |
| 22:16.31 | pjz | I'm having some echo cancellation problems and I can't figure out what to tweak |
| 22:16.49 | pjz | I can't tell if it's the phone or the pbx |
| 22:16.52 | felipeao | JESUS CRIST |
| 22:16.58 | felipeao | Nethab, IT WORKED |
| 22:17.01 | syle | so question: what extra options do you get exactly in an ip phone vs just using an analog phone with an adapter? |
| 22:17.06 | Nethab | congratulations |
| 22:17.17 | Nethab | I'm not Jesus, but I'll let him know you called |
| 22:17.25 | pjz | syle: things like message waiting indicator probably |
| 22:17.39 | syle | oww right fuck |
| 22:17.45 | pjz | syle: multiple call appearances with only one cble going into the back |
| 22:17.49 | *** join/#asterisk meppl (mephisto@pD9E68F34.dip.t-dialin.net) |
| 22:17.49 | sivana | hehe |
| 22:17.58 | Nethab | you get to use buttons to transfer and voicemail and stuff, as well as voicemail indicator |
| 22:18.00 | felipeao | Nethab, Thank you a lot man!!!!!!!!!!!!!!!!!!!!! |
| 22:18.16 | felipeao | Nethab, Thanks thanks thanks u saved my day =) |
| 22:18.21 | Nethab | with an adapter you have to memoraize funny dial strings like # for transfewr |
| 22:18.34 | Nethab | or *1 for voicemail |
| 22:18.43 | Nethab | no problem |
| 22:19.13 | *** join/#asterisk obsidian-studios (~obsidian-@c-66-177-188-197.hsd1.fl.comcast.net) |
| 22:19.36 | syle | true, well i guess they both have their pros and weaknesses, analog will still be the way to go for home use i think till they make ip cordless phones |
| 22:19.54 | obsidian-studios | is chan_zap part of asterisk, zaptel, or zapata? For some reason I do not have a chan_zap.so after a fresh install? |
| 22:20.22 | obsidian-studios | syle: there are ip cordless phones but they are pricey |
| 22:20.34 | syle | and at the 5.8ghz frequency none of this 2 ghz crap |
| 22:20.39 | Nethab | uniden UIP1868 is an VoIP coordless phone, supposed to come out soon |
| 22:20.49 | Nethab | it's 5.8 ghz |
| 22:20.58 | syle | both ways? |
| 22:21.29 | Nethab | hey pal, i only go one way... oh wait nevermind |
| 22:21.34 | syle | thats important to check alot of manufacturers advertise 5.8 ghz phones but only do it one way |
| 22:22.11 | Nethab | Vtech and Uniden are supposed to have one |
| 22:22.21 | Nethab | VoIP coordless phones |
| 22:22.38 | JerJer | Under the terms of the agreement, Cisco will pay approximately $68 million in cash and options for Sipura. |
| 22:23.27 | Nethab | yeah we know |
| 22:23.27 | syle | what about cell phones, will they have anything to do with voip? |
| 22:23.37 | Elshar | What's this? |
| 22:23.40 | Nethab | pocketpc has sip soft phones |
| 22:23.45 | Nethab | and cell phone capability |
| 22:24.08 | Elshar | JerJer: Do you have a link to that? I'd like to read it. |
| 22:24.25 | syle | i;d like to see that nethab |
| 22:26.44 | malbech | I'm locking for a Softswitch works under Asterisk / SER and not expensive ... Any idea ? |
| 22:26.51 | obsidian-studios | where is the setting to increase the number of rings before a zap channel is picked up? |
| 22:27.05 | jas_williams | malbech, Asterisk is the softswitch ???? |
| 22:27.57 | syle | obsidian-studios: i am guessing by increasing your timeout value |
| 22:27.57 | malbech | i need fonctionnality ... |
| 22:28.06 | syle | i don;t think you can set exact rings |
| 22:28.17 | obsidian-studios | style: yes I did it a while back on another * setup, just forgot where :) |
| 22:28.38 | syle | extensions.conf |
| 22:28.43 | jas_williams | malbech, what functionality can asterisk not provide |
| 22:28.56 | bajanman | can anyone point me to a very simple dial plan that I can look at? I need a starting point, and the "simple configs" that * creates is a bit to complicated. |
| 22:29.37 | malbech | jas_williams : i have no time to develop a user Web Interface interface for exemple ... |
| 22:29.44 | *** join/#asterisk bjohnson (~bjohnson@66.11.165.59) |
| 22:30.14 | jas_williams | malbech, @home/AMP ? |
| 22:30.47 | malbech | jas_williams : don't understand ... |
| 22:31.00 | syle | bajanman: http://users.pandora.be/Asterisk-PBX/IVR.htm |
| 22:31.01 | blitzrage | bajanman: well... it really depends what you need :) |
| 22:31.45 | bajanman | blitzrage: I'm used to scripting, but I'm having problems "grasping" how its all done: I can't see the "flow" of how you construct a plan. |
| 22:31.52 | obsidian-studios | bajanman: it's not simple but a good one is http://sprackett.com/asterisk/conf/extensions.conf or just http://sprackett.com/asterisk/ |
| 22:32.00 | jas_williams | malbech, "home or amp provide a web based administration for asterisk |
| 22:32.08 | bajanman | blitzrage: so If I had a real simple plan, that has a sip phone.... and that's it.. that would help |
| 22:32.27 | syle | hmmm |
| 22:32.32 | syle | i think the asterisk book had one |
| 22:32.38 | blitzrage | bajanman: basically, it flows on a line per line basis. Contexts are security boundaries. |
| 22:32.52 | blitzrage | so to start super simple... |
| 22:32.56 | jas_williams | bajanman, Why not get us to setup your first phone for you and explain away |
| 22:32.57 | blitzrage | [phones] |
| 22:33.02 | blitzrage | exten => 1,1,Answer() |
| 22:33.07 | blitzrage | exten => 1,2,Dial(SIP/1000) |
| 22:33.14 | blitzrage | exten => 1,3,Hangup() |
| 22:33.17 | malbech | jas_william : i don't know this ... thanks, i look immediately ! |
| 22:33.52 | bajanman | blitzrage: exactly I see this |
| 22:34.19 | bajanman | , but it confuses me on the "flow" of putting it all together: with an inbound, and outbound, etc. |
| 22:34.30 | bajanman | and I really DON'T want to use @home and the AMP: that's a disaster |
| 22:34.31 | *** join/#asterisk harryvv (~leonardo@S010600055d210201.vs.shawcable.net) |
| 22:35.01 | syle | i plan on using all programs |
| 22:35.04 | bajanman | jas_williams: and how much would that be? lol |
| 22:35.07 | jas_williams | bajanman: start simple on phone as blitzrage has shown than build on outbound etc |
| 22:35.14 | obsidian-studios | bajanman: it is a bit confusing at first, but once you grasp the basics concepts it gets a bit better |
| 22:35.23 | *** part/#asterisk santiago (~santiago@63.245.86.199) |
| 22:35.23 | syle | see what they do in the background, then dump them for text editor like vi hehe |
| 22:35.24 | bajanman | true true |
| 22:35.42 | bajanman | I use vi, and prefer to access the conf's directly: not via web... |
| 22:35.51 | blitzrage | bajanman: the first number is an extension number (you dial it), the second is a priority number (a unique line number, must increment sequentially), and the third is the Application which takes arguments between the () |
| 22:35.54 | *** join/#asterisk Exstatica (Exstatica@jumping.on.the.bed.are.not.umpteenmonkeys.com) |
| 22:35.55 | bajanman | I am a sys admin, and used to creating./using scripts |
| 22:36.09 | blitzrage | bajanman: here is a key point |
| 22:36.14 | obsidian-studios | blitzrage: first number is not always a # |
| 22:36.22 | jas_williams | bajanman: lets get your first phone working then what phone do you have and can we get it connected to asterisk |
| 22:36.22 | blitzrage | obsidian-studios: I know - I'm keeping it simple |
| 22:36.31 | obsidian-studios | in fact even simpler would not have extensions and just answer or ring phone directly |
| 22:36.34 | blitzrage | obsidian-studios: believe me... I"ve done a TON of dialplan scripting :) |
| 22:36.39 | *** join/#asterisk mistral (mistral@jstevenson.plus.com) |
| 22:36.40 | obsidian-studios | I believe you :) |
| 22:36.41 | *** join/#asterisk Exstatica (Exstatica@jumping.on.the.bed.are.not.umpteenmonkeys.com) |
| 22:36.41 | blitzrage | obsidian-studios: that's the 's' extension |
| 22:36.44 | obsidian-studios | yes |
| 22:36.49 | blitzrage | anyways... |
| 22:36.53 | blitzrage | bajanman: key point - |
| 22:36.56 | bajanman | the 's" is what always confuses me...lol |
| 22:37.01 | pjz | what does 'TDM' mean? |
| 22:37.16 | jas_williams | Tim Division Multiplexer |
| 22:37.16 | pjz | ; Generally, it is not necessary (and in fact undesirable) to echo cancel |
| 22:37.18 | pjz | ; when the circuit path is entirely TDM. |
| 22:37.22 | obsidian-studios | s is like start or processed before extensions and etc |
| 22:37.26 | syle | aren;t the s extension examples , don;t they just mean do this by default |
| 22:37.31 | pjz | okay |
| 22:37.37 | jas_williams | Time |
| 22:37.38 | obsidian-studios | no, do this first then other stuff |
| 22:37.42 | blitzrage | bajanman: in channel configuration files (sip.conf, zapata.conf, iax.conf, etc..) each user (thats an incoming call) has a context=foo line. foo is the name of the context in extensions.conf where the incoming call will be pointed to. |
| 22:37.46 | syle | oww shit, good to know |
| 22:37.54 | obsidian-studios | also t is good to know and use |
| 22:37.54 | blitzrage | bajanman: assuming of course the user is authenticated |
| 22:38.08 | blitzrage | bajanman: now, here is where the 's' extension comes into play |
| 22:38.13 | obsidian-studios | and i for invalid extension |
| 22:38.23 | bajanman | blitzrage: yes I remember ok I get that |
| 22:38.58 | blitzrage | bajanman: assume someone places a call to your type=user in sip.conf. The incoming call doesn't ask for an extension number, but you want to do something with that call. Asterisk will look to see if an 's' extension exists, and if so, starts at priority 1 of the 's' context. |
| 22:39.02 | bajanman | I have my * up and runing: with x-lite, and connected to a voip. (via sip) that works fine |
| 22:39.07 | harryvv | anyone experaince issues with comedian mail female voice being choppy? |
| 22:39.12 | blitzrage | sorry... switch that last word 'context' with 'extension' |
| 22:39.19 | Shido | yeah |
| 22:39.34 | bajanman | hmmm ok |
| 22:39.36 | harryvv | Shido, seen that before? |
| 22:39.46 | obsidian-studios | bajanman: where are you stuck at? |
| 22:40.11 | jas_williams | bajanman: Some devices such as an analogue phone line do not pass extension information so need the s, s means no extension requested |
| 22:40.16 | pjz | how much bandwidth does ulaw take? |
| 22:40.22 | blitzrage | s == start |
| 22:40.31 | obsidian-studios | jas_williams: like an inbound call |
| 22:40.37 | bajanman | obsidian: I think I'm stuck at not being able to see the "whole concept"... if I can do that, then I can grasp the little bits. |
| 22:40.38 | bajanman | lol |
| 22:40.41 | blitzrage | pjz: 64kbps plus IP header overhead (about 87kbps total) |
| 22:40.44 | harryvv | sucks that cisco wants to buy out sipura |
| 22:40.46 | syle | how could an incomming call not ask for an extension? you mean that extension it asked for wasn;t define in extensions.conf so it defaulted to 's' then? |
| 22:41.10 | obsidian-studios | bajanman: whole concept is deep, stick with asterisk extensions, contexts, and sip stuff and it should be a bit clearer |
| 22:41.11 | blitzrage | pjz: approximately 11KB/s |
| 22:41.14 | bajanman | I guess. what I'd like to know, is how do you start out with an [inbound] and an [outbound]? and rules... |
| 22:41.23 | obsidian-studios | an extension is a line or lines in a context |
| 22:41.33 | obsidian-studios | extension can point to or navigate contexts |
| 22:41.34 | blitzrage | bajanman: thats where type=user and type=peer comes into play |
| 22:41.39 | bajanman | obsidian: I think you're right.. I keep bogging myself down, cause I can't see the whole thing.. gotta stop doing that |
| 22:41.41 | pjz | blitzrage: so on an unloaded 100Mbps net it's unlikely that bandwidth is a problem, then? |
| 22:41.45 | obsidian-studios | channels and etc can default to a context or etc |
| 22:41.51 | blitzrage | pjz: LOL... hell no |
| 22:41.57 | jas_williams | obsidian-studios: Like an inbound call over non digital trunks |
| 22:42.12 | pjz | blitzrage: I'm getting nasty echo on this Uniden UIP200 phone |
| 22:42.17 | obsidian-studios | jas_williams: exactly like inbound over pots or analog |
| 22:42.28 | pjz | blitzrage: any hints on how to get rid of it? |
| 22:42.36 | jas_williams | syle: no as obsidian-studios states |
| 22:42.37 | blitzrage | pjz: turn on echo cancellation? :) |
| 22:42.44 | pjz | blitzrage: where? |
| 22:42.56 | jas_williams | pjz: what type of pstn connectivity ? |
| 22:42.57 | syle | how can an incomming call have no destination? |
| 22:42.58 | harryvv | I just talked to a shaw tech and he states shaw is going to put there voip service on a totally seperate chanell frequency. good idea. |
| 22:43.02 | obsidian-studios | bajanman: you can have different contexts for inbound and outbound stuff |
| 22:43.14 | pjz | jas_williams: I'm getting the echo just going ext to ext without going to pstn |
| 22:43.50 | jas_williams | syle: that is how a pots line works it is directed to your server but has no extension information passed with the call so asterisk delivers it to the s extension |
| 22:43.55 | bajanman | obsidian: so. I have a [4165551212] in my sip.conf. this is the number I use to dial out, and in with |
| 22:44.03 | bajanman | obsidian: what is that part called? |
| 22:44.04 | obsidian-studios | things like sip channels if that's the right term can be mapped to individual extensions or a global one for all |
| 22:44.09 | blitzrage | bajanman: a context |
| 22:44.20 | bajanman | blitzrage: good ok. |
| 22:44.22 | bajanman | I get that |
| 22:44.24 | bajanman | now, |
| 22:44.26 | jas_williams | pjz: That should not happen |
| 22:44.36 | blitzrage | bajanman: sorry... you said sip.conf :) I call it a "definition" |
| 22:44.36 | pjz | jas_williams: no kidding |
| 22:44.38 | syle | then how does it get to its destination |
| 22:44.39 | *** join/#asterisk sandnigg0r (~niggerplz@66-55-197-254.gwi.net) |
| 22:44.41 | bajanman | [unlimitel-in] |
| 22:44.41 | bajanman | type=peer |
| 22:44.41 | bajanman | host=sip.unlimitel.ca |
| 22:44.41 | bajanman | context=unlimitel-inbound |
| 22:44.41 | bajanman | canreinvite=no |
| 22:44.49 | pjz | jas_williams: any idea how to tune it out? |
| 22:44.55 | bajanman | this is my [inbound]??? |
| 22:44.55 | blitzrage | bajanman: that context= does nothing |
| 22:45.02 | blitzrage | bajanman: peers are outbound |
| 22:45.03 | Nethab | that's a channel |
| 22:45.06 | blitzrage | bajanman: think of it this way |
| 22:45.13 | obsidian-studios | ok so when you pick up that phone it goes to unlimitel-inbound context in extensions.conf |
| 22:45.20 | blitzrage | bajanman: Asterisk accepts calls from /users/ and places calls to /peers/ |
| 22:45.23 | bajanman | obsidian: yes |
| 22:45.38 | bajanman | blitzarge: I get concept |
| 22:45.45 | ManxPower | friend should be an alias for type=userandpeer |
| 22:45.50 | blitzrage | bajanman: right... but a peer is for placing outbound calls |
| 22:45.53 | obsidian-studios | that is your outbound extension |
| 22:46.12 | bajanman | [unlimitel-inbound] |
| 22:46.13 | bajanman | exten => 6477223545,1,SetAccount(6477223545) |
| 22:46.13 | bajanman | exten => 6477223545,2,Ringing() |
| 22:46.13 | bajanman | exten => 6477223545,3,Dial(SIP/200,50) ; This is an example to send the call to a SIP device on your network. |
| 22:46.13 | bajanman | exten => 6477223545,4,Hangup; |
| 22:46.14 | obsidian-studios | since you will be picking up sip phone to call out, or outbound extension |
| 22:46.30 | blitzrage | bajanman: you need a type=user for unlimitel to place calls to you |
| 22:46.32 | bajanman | sooooo this is how I start to handle incoming calls??? |
| 22:46.43 | blitzrage | bajanman: yes, but you have nothing in sip.conf to accept the inbound calls |
| 22:46.44 | obsidian-studios | ok how does the phone line get to * |
| 22:46.51 | obsidian-studios | pots, pri, T-1 ? |
| 22:46.51 | blitzrage | obsidian-studios: there is no phon eline |
| 22:46.56 | obsidian-studios | ah ha |
| 22:46.57 | obsidian-studios | ok |
| 22:46.59 | blitzrage | obsidian-studios: he is using VOIP |
| 22:47.08 | obsidian-studios | so in one sip out another sip |
| 22:47.20 | blitzrage | yes - this is why we need users and peers |
| 22:47.24 | obsidian-studios | got it |
| 22:47.30 | bajanman | obsidian: I only have voip in and out |
| 22:47.34 | obsidian-studios | did not know it was sip in out |
| 22:47.43 | bajanman | blitzrage: right |
| 22:47.44 | obsidian-studios | should be even more straight forward? |
| 22:47.46 | bajanman | completely.. no pot |
| 22:47.55 | blitzrage | bajanman: now, for unlimitel to know where you are, you will also need a register statement, but we'll come back to that after |
| 22:47.57 | jas_williams | pjz: are you using speaker phones or the handset ? |
| 22:48.03 | obsidian-studios | hey man a little pot is not bad, just do not let police catch you :) |
| 22:48.11 | bajanman | yes I have that: in my [general] under sip.conf |
| 22:48.21 | blitzrage | bajanman: ok cool |
| 22:48.23 | bajanman | obsidain: Im canadian.. got friends who grow...lol |
| 22:48.38 | blitzrage | bajanman: Toronto? |
| 22:48.43 | bajanman | cops don't care here..YEP |
| 22:48.44 | bajanman | lol |
| 22:48.49 | obsidian-studios | lucky you, have to be medical for legal in US unless you are in San Mateo County in CA |
| 22:48.54 | blitzrage | bajanman: I'm in Toronto too |
| 22:49.04 | bajanman | blitzrage: good to know.... |
| 22:49.12 | syle | i heard its a fine in CA for a certain amount of pot |
| 22:49.12 | obsidian-studios | major there legalized small amounts due to crime and etc |
| 22:49.13 | blitzrage | bajanman: you need to change the [definition] section in your sip.conf to your unlimitel username |
| 22:49.22 | syle | treat it like a speeding ticket |
| 22:49.26 | obsidian-studios | syle: misdemeanor |
| 22:49.28 | obsidian-studios | yes |
| 22:49.35 | obsidian-studios | where I am in FL not so much |
| 22:49.44 | blitzrage | bajanman: thats what chan_sip will match on (although this is becoming less true as I test more and more...) |
| 22:49.55 | obsidian-studios | but in CA 100+ plants, no house or possessions for up to 7 years |
| 22:49.57 | blitzrage | so, a simple INCOMING definition in sip.conf might look like this |
| 22:50.03 | blitzrage | [unlimitel-username] |
| 22:50.05 | blitzrage | type=user |
| 22:50.09 | syle | best thing you can do is keep your weed rolled(so they can;t nail you with trafficking) and in little amounts on you |
| 22:50.11 | blitzrage | secret=password |
| 22:50.24 | blitzrage | context=unlimitel-inbound |
| 22:50.27 | obsidian-studios | <PROTECTED> |
| 22:50.28 | blitzrage | thats it |
| 22:50.29 | jsolares | anyone know a way to set the callerid the iaxy phone uses in the provisioning file? |
| 22:50.41 | obsidian-studios | I love sip |
| 22:50.55 | blitzrage | bajanman: context=foo in the type=peer is NOT NEEDED (and you shouldn't have one for security reasons) |
| 22:51.04 | syle | its really not possible to get caught with weed if you don;t have alot |
| 22:51.11 | obsidian-studios | I made the mistake of using sccp phones and boy does it suck |
| 22:51.19 | bajanman | oh boy. I started the herb talk, now I can't keep up....lol |
| 22:51.20 | syle | if they come to your door , flush it, if your in your car, throw it out the window etc |
| 22:51.35 | syle | no evidence= crown is fucked |
| 22:51.45 | obsidian-studios | syle: nah I just toss in my friends lap and say it's theirs :) |
| 22:51.50 | syle | lol |
| 22:52.09 | obsidian-studios | or grab a gun and go out OJ style |
| 22:52.44 | syle | i;d rather not pull out a gun, i want to live in my country hehee |
| 22:53.02 | obsidian-studios | I was speaking OJ Simpson, the gun was to his head in a back of car ;) |
| 22:53.34 | syle | some countries i have been to have different laws, in dominion republic for example, if someone shot your brother, police give you 3 weeks to hunt down his killer and kill him |
| 22:53.52 | syle | with laws like that there would be alot less deaths hehe |
| 22:53.56 | obsidian-studios | anyway I think the timeout I used in past for rings was on a extension part of the dial command like Dial(SIP/#,30) which is like 4 or so rings but how do I do that for an inbound call via a pots/zap channel? |
| 22:54.17 | obsidian-studios | syle: hell yeah we need that in the US |
| 22:54.47 | jas_williams | obsidian-studios: place a wait(4) in your s extension |
| 22:54.57 | syle | regardless you know if someone killed your brother , you;d hunt the bitch down and kill him anyways, but be nice if it was legal |
| 22:54.59 | obsidian-studios | ok thought so, before answer right? |
| 22:55.11 | jas_williams | obsidian-studios: yes |
| 22:55.18 | obsidian-studios | syle: death is to nice, I prefer torture and keep alive |
| 22:55.28 | obsidian-studios | like if a man, castrate :) |
| 22:55.33 | syle | haha |
| 22:55.57 | syle | yeah that would be kewl |
| 22:56.01 | obsidian-studios | but really here, grow lots of weed and you will be in jail longer than if you killed or raped someone or both :) |
| 22:56.12 | syle | if he used a gun, chop of all his fingers and send him home |
| 22:56.29 | obsidian-studios | that or shoot him once or twice in the hand |
| 22:56.32 | obsidian-studios | each |
| 22:56.33 | Moc | blitzrage, you use unlimitel too ? |
| 22:56.38 | blitzrage | Moc: hell no :) |
| 22:56.46 | bajanman | obsidian: ok thanks. I'll look at it. and come back with questions I guess.: thanks for the help everyone. |
| 22:56.56 | blitzrage | Moc: I have no opinion on them, never used them :) |
| 22:57.01 | blitzrage | Moc: I use mixnetworks.com |
| 22:57.06 | obsidian-studios | bajanman: it's deep man take your time, and bit of small chunks |
| 22:57.18 | syle | notice how gov't always sends you to jail if they don;t get their tax money |
| 22:57.35 | blitzrage | syle: its always been like that since money was invented |
| 22:57.37 | syle | ripping gov;t off of money is worse than murder to them |
| 22:57.43 | obsidian-studios | nah, here they try to double bill first then jail |
| 22:57.55 | Moc | they seem expensive |
| 22:58.04 | syle | oww right, interest and back pay, pay up or do time |
| 22:58.12 | Moc | oh it US |
| 22:58.17 | Moc | cheaper over here ;) |
| 22:58.18 | obsidian-studios | if you do not pay inflated bill, but really our tax laws are not for employees but employers, you can just about right off everything unless you need to show a profit :) |
| 22:58.19 | Moc | bbl |
| 22:58.24 | blitzrage | Moc: peas |
| 22:59.02 | syle | more so when republicans are in power i would imagine |
| 22:59.37 | syle | i think way it works in US, correct me if i am wrong, you vote in republicans for tax breaks and tax loopholes, and the liberals to improve the deficit |
| 22:59.50 | obsidian-studios | well we get a very bad rap I am republican to the bone, but a conservative liberal one :) |
| 22:59.55 | syle | to put it simply |
| 23:00.12 | obsidian-studios | all my liberal democratic friends are wealthy very |
| 23:00.46 | obsidian-studios | all my republican friends typically own small businesses, or are moderately wealthy but off hard work |
| 23:01.06 | syle | very interesting |
| 23:01.11 | obsidian-studios | it is |
| 23:01.21 | obsidian-studios | but things change is you get inward |
| 23:01.26 | syle | maybe your liberal friends just know more about investing than your republican friends :) |
| 23:01.31 | obsidian-studios | no |
| 23:01.35 | obsidian-studios | look a Kerry |
| 23:01.40 | obsidian-studios | married into a rich family |
| 23:02.00 | obsidian-studios | people I knew in CA who were rich got that way from entertainment and etc |
| 23:03.08 | obsidian-studios | democrats like to talk about problems, republicans like to get stuff done, right or wrong |
| 23:03.23 | obsidian-studios | neither are perfect |
| 23:03.52 | obsidian-studios | to anyone interested my el cheapo fxs cards showed up and I got them working without an problems or hickups :) |
| 23:04.15 | obsidian-studios | got myself a glorified * answering machine at the moment till I get more pots lines and some sip phones :) |
| 23:04.31 | *** join/#asterisk CoderCR (~creyna@ip68-8-131-103.sd.sd.cox.net) |
| 23:04.41 | obsidian-studios | supposedly they suck on echo training and etc? How can I test that? Echo on my end or callers end? |
| 23:04.43 | jas_williams | obsidian-studios: fxs cards or fxo |
| 23:04.44 | CoderCR | Hello all |
| 23:04.49 | harryvv | ob, well yea you can also route calls if you have a iax provider |
| 23:05.03 | harryvv | obsidian-studios: its more then that :) |
| 23:05.04 | obsidian-studios | crap I always get confused, fxo I believe, what you connect to pots lines :) |
| 23:05.26 | jas_williams | Yes FXO |
| 23:05.29 | stoyan | when placing a call through asterisk, in what context will the called party be? |
| 23:05.32 | obsidian-studios | now I got to get a DJ buddy of mine to make a cool recordings for me |
| 23:05.51 | syle | lol |
| 23:05.57 | obsidian-studios | <PROTECTED> |
| 23:06.11 | obsidian-studios | I hate the sound of my voice on a answering machine :) |
| 23:06.26 | syle | one thing i have had trouble finding is the dynamic variables you can use in a dial plan |
| 23:06.35 | syle | like phone_number |
| 23:06.40 | obsidian-studios | what do you mean? |
| 23:06.41 | jas_williams | stoyan: One included in the context of the calling phone with a matchin extension |
| 23:06.46 | obsidian-studios | there a all kinds of variables |
| 23:07.01 | syle | i just want a list of them :) |
| 23:07.11 | obsidian-studios | you check the wiki |
| 23:07.13 | stoyan | obsidian-studios : i was wondering if I can trigger a call from asterisk to pstn. play message on connect and get some dtmf digits. |
| 23:07.16 | syle | $1 $2 $3 whatever they are |
| 23:07.21 | jakepdev | http://www.voip-info.org/wiki-Asterisk+variables |
| 23:07.52 | obsidian-studios | stoyan: pstn is pots right or? |
| 23:08.02 | jas_williams | stoyan: yes if you make dial drop to a macro (try show application dial) |
| 23:08.17 | stoyan | yes |
| 23:08.22 | stoyan | ok |
| 23:08.29 | obsidian-studios | so you just want a menu system? |
| 23:08.30 | syle | well damn |
| 23:08.34 | syle | thx jake |
| 23:08.37 | jakepdev | np |
| 23:08.45 | obsidian-studios | that wiki has all kinds of stuff |
| 23:08.59 | harryvv | pstn is in your wall obsidian-studios |
| 23:09.02 | obsidian-studios | when I first started I could not get a question answered without someone pointing me to wiki |
| 23:09.24 | obsidian-studios | that's what I thought, still getting used to all the terminology |
| 23:09.44 | *** join/#asterisk phantasis (phantasis@137.28.208.13) |
| 23:09.45 | harryvv | is it possible to rebuild comedian mail without affecting asterisk? its just gettign way to choppy. |
| 23:11.00 | opus_ | hey harryvv |
| 23:11.09 | opus_ | you use motion too? |
| 23:11.10 | obsidian-studios | what is chan_zap part of zaptel or zapata? I recently installed * with both, but never got a chan_zap.so? I had to grab from another * machine? |
| 23:11.27 | phantasis | how do I setup SIP extensions using AMP |
| 23:11.30 | obsidian-studios | which seems weird? |
| 23:12.41 | *** join/#asterisk adker (~adker@67-51-234-116.dsl1.glv.ny.frontiernet.net) |
| 23:13.04 | Druken | harryvv: where's the source for comedian ? |
| 23:14.06 | jakepdev | phantasis - might want to join amportal |
| 23:14.29 | phantasis | thx |
| 23:15.17 | Nethab | Is there only two status messages for comedian mail, ie unavailable, and on the phone |
| 23:15.52 | Druken | how many do you want? hehe |
| 23:16.07 | Nethab | well, i'm used to unavailable, on the phone, and does not answer |
| 23:16.36 | Nethab | but there's no does not answer in asterisk, it says i'm on the phone, even when i just don't pick up |
| 23:16.42 | *** join/#asterisk rjburkh (~rjburkh@dialup-4.233.118.12.Dial1.LosAngeles1.Level3.net) |
| 23:16.58 | Druken | well, wouldn't does not answer be unavailable? |
| 23:17.27 | Nethab | no unavailable only occurs in asterisk when your not registered or when dial app can't connect to yopu |
| 23:17.29 | *** join/#asterisk Carp1 (carp_xigon@ip-204-97-151-69.modem.logical.net) |
| 23:17.45 | Nethab | if it can connect and you don't pick up it assumes your on the phone |
| 23:17.56 | moy | Hi everybody. Does anybody knows how can i do that asterisk hangups the calls when somedoy is trying to call with Dial(ZAP/g1/123456) and the lines in group 1 are not connected to the FXO interfaces?? |
| 23:18.09 | Druken | hmm.. true... |
| 23:18.19 | file | room rather |
| 23:18.24 | Nethab | file wheres your laptop or mac |
| 23:18.25 | Druken | but.. how do you differ between the two? |
| 23:18.49 | Nethab | if there's an existing channel to the same peer then i'm on the phone, otherwise i just didn't answer |
| 23:18.54 | jakepdev | moy - DIALSTATUS? |
| 23:19.02 | Druken | asterisk doesn't have "extensions" like a pbx... |
| 23:19.17 | phantasis | if I am using SIP between two extensions, do I need a trunk? |
| 23:19.32 | Druken | Nethab: ok.. so if your on the phone making a call, you just "didn't answer" ? |
| 23:19.33 | moy | jakepdev: the problem is that the process stucks when reach Dial() so i never reach the nex priority |
| 23:19.35 | Nethab | my complaint is that even when i don't answer it says on the phone |
| 23:20.01 | Nethab | so my wife thinks i'm talking to some other woman ok? |
| 23:20.13 | phantasis | lol |
| 23:20.26 | obsidian-studios | are you :) |
| 23:20.37 | Nethab | no i just didn't want to talk to the witch |
| 23:20.44 | Druken | so change it to a personal greeting, sorry i'm unavailable to come to the phone, blah blah blah... oh and btw huney... i'm not having an affair on you... |
| 23:20.55 | obsidian-studios | make an extension just to tie her up for hours :) |
| 23:21.01 | phantasis | drunken: haha |
| 23:21.34 | Nethab | "if your call is an emergency press 1 *beep* if your his wife press 1" |
| 23:21.34 | obsidian-studios | press 1 to see if I am cheating on you with another woman, press 2 for a man, press 3 for both :) |
| 23:21.35 | Druken | Nethab: exten => yourext/herCID :) |
| 23:22.10 | Nethab | i'm just saying when calling VoiceMail your only choices are b and u |
| 23:22.22 | Nethab | where's the third option |
| 23:22.39 | Nethab | or do not disturb status |
| 23:23.06 | Druken | my point there, how do you differ from b and didn't answer, without knowing the specific status of the device your dialing |
| 23:23.44 | Nethab | why not just say didn't answer instead of assuming "On the Phone" |
| 23:24.01 | Druken | what i'd like to see if a * option when the message begins to play, so you can login by calling your own number |
| 23:24.03 | Nethab | just seems like a bad case of defaults |
| 23:24.24 | Druken | Nethab: so change it to a personal message |
| 23:24.35 | Druken | you can do that ya know... it's an option in your mailbox |
| 23:24.54 | Druken | you can record BOTH of those messages |
| 23:25.05 | Nethab | yeah but then it's in my voice, I want to tell her, hey the computer said i didn't asnwer not me |
| 23:25.33 | Druken | hahaha so pay allison for the special prompt :) |
| 23:25.51 | *** join/#asterisk bkw_ (~brian@bkw.developer.and.friend.of.asterisk) |
| 23:25.51 | *** mode/#asterisk [+o bkw_] by ChanServ |
| 23:25.52 | Druken | it's what? 50 bux? |
| 23:26.49 | Nethab | you know i'm being mostly facetious right? |
| 23:27.10 | Druken | ok... my bad.. 12 bux |
| 23:27.35 | Druken | your being a pain in the ass :) |
| 23:28.01 | Nethab | complacency is a poor excuse for bad decisions |
| 23:28.11 | Nethab | i don't have a wife |
| 23:28.13 | ManxPower | Yeah! |
| 23:28.30 | phantasis | do I need to setup a sip trunk to make sip calls between extensions?? |
| 23:28.51 | *** part/#asterisk obsidian-studios (~obsidian-@c-66-177-188-197.hsd1.fl.comcast.net) |
| 23:28.56 | Nethab | ? no sip trunking isn't necessary except between multiple asterisk servers |
| 23:28.57 | ManxPower | phantasis: there is no such thing as a "sip trunk" in asterisk, as asterisk users think of a trunk. |
| 23:29.27 | Druken | Nethab: even between servers you don't HAVE to trunk |
| 23:29.30 | phantasis | argh, i can't even get my sip softphone to connect |
| 23:29.51 | Nethab | i mean between clients it's not even possible |
| 23:29.51 | bkw_ | I need someone I can call via h323 |
| 23:29.57 | bkw_ | anyone? anyone? |
| 23:30.16 | Nethab | wow, h323 does netmeeting work |
| 23:30.16 | ManxPower | phantasis: Asterisk has inbound calls and outbound calls. It assumes you can send anything you want to any endpoint. |
| 23:30.24 | blitzrage | bkw_: sure, you just gotta set it up on my box |
| 23:30.25 | Carp1 | Whats a good analog business phone? |
| 23:30.26 | *** join/#asterisk Wi_Fi (~OUT@c-24-127-12-85.hsd1.ca.comcast.net) |
| 23:30.44 | blitzrage | Carp1: I like the Northern Telecom Vista series |
| 23:30.50 | phantasis | Manxpower: so where do I put sip information? extensions.conf sip.conf both? |
| 23:30.57 | ManxPower | phantasis: Asterisk think of "trunking" as a special type of optimizaion for IAX2 that drastically reduces UDP overhead/per call. |
| 23:31.04 | ManxPower | phantasis: both. |
| 23:31.09 | phantasis | same format? |
| 23:31.11 | Voicelynx | Is there a way of forcing an EXTEN to (say) lower case in extensions.conf? |
| 23:31.15 | ManxPower | You define the endpoint/device in sip.conf. |
| 23:31.36 | phantasis | e.g. the softphone? |
| 23:31.50 | ManxPower | you define call stuff that doesn't really care about device configuration in extensions.conf. |
| 23:31.51 | Nethab | the endpoint can be a peer or a user doesn't matter |
| 23:32.21 | Nethab | it's basically a udp stream from one sip url to another sip url |
| 23:32.27 | phantasis | oh |
| 23:32.46 | Nethab | one url can be a phone or a service provider or any mix in between |
| 23:32.54 | ManxPower | phantasis: the included sample config files suck. They try to show lots of options, where a newbie needs a simple sample config. Check the wiki or the World Famous Fnords Site archive at http://www.fnords.org/~eric/asterisk/wffs.tar.gz there's all sorts of samples in there. |
| 23:33.24 | phantasis | thanls |
| 23:33.40 | Carp1 | Whats the cheapest way to add 10 analog extensions to my asterisk box? |
| 23:33.41 | ManxPower | Since the config files are basically the documentation for Asterisk it's good they try to show EVERYTHING. I use them as a reference for options. |
| 23:34.06 | Shido | 3 4 port FXS cards, Carp1 |
| 23:34.18 | Shido | oops |
| 23:34.27 | Shido | yeah that works |
| 23:34.28 | ManxPower | Carp1: "cheapest" as in "purchase cost", or "cheapest" as in "total cost including both initial purchase price and onging stability and support requirements"? |
| 23:34.37 | Carp1 | I dont have 3 slots available lol |
| 23:34.46 | Shido | how many slots available? |
| 23:34.47 | Nethab | i completely strip out my sip.conf iax.conf and extensions.conf |
| 23:34.48 | *** join/#asterisk vpp (~noone@host-83-146-50-131.bulldogdsl.com) |
| 23:34.48 | Carp1 | Thats ok. |
| 23:35.01 | Carp1 | How mich are 4 port FXS cards going for? |
| 23:35.10 | Carp1 | much* |
| 23:35.18 | Nethab | you can always get an FXS gateway appliance but it'll cost you |
| 23:35.30 | ManxPower | Carp1: for "total cost including both initial purchase price and onging stability and support requirements" you want TDM400Ps with the correct modules, $309/each |
| 23:35.34 | ManxPower | eep[! |
| 23:35.43 | Shido | 296.95 |
| 23:35.49 | ManxPower | Carp1: for "total cost including both initial purchase price and onging stability and support requirements" you want a TE110P and a Adtran channel bank from eBay. |
| 23:35.50 | Shido | for 4 FXS |
| 23:36.10 | ManxPower | Carp1: for you want TDM400Ps with the correct modules, $309/each "cheapest" as in "purchase cost" |
| 23:36.21 | Carp1 | Ok. |
| 23:36.27 | Carp1 | Thanks. |
| 23:36.55 | ManxPower | Go with the channel bank and (unless you are quite unlucky) it will Just Work once you have it set up. |
| 23:37.54 | Carp1 | Yeah. I'm looking around. |
| 23:38.26 | tld | Anyone have two seconds to spare to help save a 7960? It's looping on a software upgrade. After rebooting, it starts over again, reboots, starts over etc. |
| 23:39.22 | opus_ | corrupted rom |
| 23:39.43 | Nethab | is there a hard hard reset? |
| 23:39.48 | Druken | ManxPower: you want to buy me a te110p don't you? |
| 23:39.50 | *** join/#asterisk amir|away (~amir@195.226.9.186) |
| 23:40.10 | Nethab | it reboots while downloading software upgrade? |
| 23:40.14 | ManxPower | Druken: Well a forward thinking person will get a TE$)xP |
| 23:40.15 | ManxPower | ..er.. |
| 23:40.24 | ManxPower | Druken: Well a forward thinking person will get a TE4xxP |
| 23:40.39 | ManxPower | and spend the extra money for echo canceling hardware. |
| 23:40.46 | Druken | i was thinking a te401p but i didn't think you'd go for it :) |
| 23:40.50 | tld | Nethab: After the upgrade, it's supposed to reboot to start using the new firmware. So far so good, but after the boot it starts the upgrade all over again. |
| 23:41.15 | Nethab | rename the file on the ftp server |
| 23:41.17 | ManxPower | And, honestly, anyone can save a LOT of money if they keep an eye on eBay for equipment that's needed and plan far enough ahead to not run in "crisis" mode. |
| 23:41.21 | Nethab | so it can't grab it |
| 23:41.45 | moy | Hi there again. Does anybody knows how can i do that asterisk hangups the calls when somedoy is trying to call with Dial(ZAP/g1/123456) and the lines in group 1 are not connected to the FXO interfaces?? |
| 23:42.13 | Druken | ManxPower: got the channelbank.. just not the t1 interface |
| 23:42.39 | phantasis | anyone ever work with a CAC Adit 600? |
| 23:45.29 | mishehu | moy: there will be a red alarm if the lines in g1 are not physically connected to the lines |
| 23:45.41 | Carp1 | I see T1 cards on eBay for under $100 |
| 23:45.42 | mishehu | it should return congestion |
| 23:45.56 | mishehu | Carp1: made by whom? |
| 23:46.20 | Carp1 | Cisco WIC 1DSU T1 interface card |
| 23:46.31 | ManxPower | Carp1: That's a data only card. |
| 23:46.38 | ManxPower | We have bunches and bunches of them. |
| 23:46.46 | mishehu | and it requires a cisco router I bet |
| 23:46.49 | Carp1 | That would be why |
| 23:46.55 | ManxPower | mishehu: Um, yes. |
| 23:47.18 | ManxPower | For T-1 data into a Cisco router, they are great. |
| 23:47.26 | mishehu | I've got a t110p waiting for line delivery |
| 23:47.48 | mishehu | I hope I didn't make a mistake in getting the t110p... it's a tigerjet 320 also |
| 23:47.53 | Druken | mishehu: lemme forward you the address :) |
| 23:48.03 | ManxPower | mishehu: you can always change out the motherboard if you have to. |
| 23:48.19 | ManxPower | Also, find out EXACT model and part numbers of people using the TE110P successfully |
| 23:48.47 | mishehu | ManxPower: well, I was more concerned about the problems I've had with tdm400 cards and the random lockups. If I recall, some people told me they had no problems at all wtih the t1xx series |
| 23:48.47 | ManxPower | of the motherboard, of course. |
| 23:49.20 | ManxPower | mishehu: contact Digium support, tell them the details of the lockups and find out of a newer version of the card will be helpful. |
| 23:49.37 | syle | http://richard.jones.name/google-hacks/gmail-filesystem/gmail-filesystem.html |
| 23:49.39 | ManxPower | of = if |
| 23:49.43 | mishehu | ManxPower: already got the replacement, haven't been able to go in after-hours to swap it yet. |
| 23:49.58 | ManxPower | make it a priority |
| 23:49.59 | mishehu | ManxPower: I'm commenting btw |
| 23:50.20 | mishehu | just so you know I'm not trying to nag you for help. ;-) |
| 23:50.46 | ManxPower | We had a TDM400P card that was locking up ocassionally, I opened up the case and saw it was the first version of the TDM400P, the one without the power connectors. |
| 23:52.14 | *** join/#asterisk Evanrude (~david@wsip-68-15-251-34.dl.dl.cox.net) |
| 23:53.52 | Carp1 | What does a T1 port look like? |
| 23:54.08 | Druken | rj45 jack |
| 23:54.18 | drumkilla | like the mouth of a snake! |
| 23:54.33 | Carp1 | oh |
| 23:54.37 | Carp1 | lol |
| 23:54.54 | syle | damn we can mount gmail like nfs when it gets released i think |
| 23:54.56 | Carp1 | If I could afford a T1 card, I would be on my way lol. |
| 23:54.58 | ManxPower | Your first time, the RJ45 looks like a nightmare. |
| 23:55.33 | Carp1 | lol |
| 23:56.55 | Moonwick | the speakerphone on this spa-841 fades in and out constantly |
| 23:57.09 | Moonwick | anyone know if that's a known bug, or did I get a lemon? |
| 23:57.15 | Moonwick | google doesn't tell much (that I've found) |
| 23:58.50 | Druken | i've always found speakerphone to fade in and out.. on every phone i've used in the last 10 years |
