| 00:00.02 | muntz | hmm |
| 00:00.17 | muntz | I should maybe move the conf aside then |
| 00:00.49 | *** join/#asterisk Barmal (~1@adsl-34-13-135.asm.bellsouth.net) |
| 00:00.54 | muntz | thanks |
| 00:01.38 | Barmal | how can If do in asterisk something like this: repeat next three lines 5 times? can I make it with gotoif? |
| 00:06.43 | [hC] | hm. okay maybe i have to reword that. Is it possible (or advisable) to try to combine in+out over a single iax peer definition? It doesnt seem intuitive, or maybr im missing something |
| 00:08.51 | bkw_ | [hC], thats called a "friend" |
| 00:08.57 | bkw_ | type=friend |
| 00:09.01 | bkw_ | it can make and receive calls |
| 00:09.09 | bkw_ | in larger setups its not recommended |
| 00:09.13 | bkw_ | like provider setups |
| 00:09.22 | bkw_ | but most home/small office stuff can get away with it |
| 00:09.31 | [hC] | Right, but what im running into, is i want to place calls out and also receive calls, (me linking to my main provider) so obviously the context name, username and password has to match in the iax.conf on both sides |
| 00:09.38 | [hC] | but the host= line wouldnt match |
| 00:09.55 | [hC] | cause you have to specify what the opposing server's ip address is, in order to send traffic over |
| 00:10.05 | [hC] | but they have DIDs terminating to us, and we also use them for outgong calls |
| 00:10.21 | bkw_ | no you fail to understand it |
| 00:10.25 | muntz | I figured it out |
| 00:10.28 | bkw_ | a user will register |
| 00:10.30 | bkw_ | a peer does not |
| 00:10.31 | [hC] | I suppose i do fail |
| 00:10.50 | bkw_ | we receive calls from a user |
| 00:10.50 | bkw_ | and send calls to a peer |
| 00:10.53 | [hC] | okay, and how about a friend? :) |
| 00:10.54 | muntz | in sip.conf I had bindaddr:My.External.IP.ADDY where I needed |
| 00:11.03 | muntz | bindaddr:0.0.0.0 |
| 00:11.10 | bkw_ | it can make and receive |
| 00:11.13 | muntz | bindaddr=0.0.0.0 |
| 00:11.21 | muntz | W00T! |
| 00:11.48 | [hC] | Right, so lets say i have an iax entry called [myprovider] that is type=friend, with a username and password, and i need to specify host=<their ip> so that i can Dial(IAX2/myprovider/extension) |
| 00:11.51 | *** join/#asterisk djflux (~djflux@cpe-24-165-117-88.cinci.res.rr.com) |
| 00:12.25 | bkw_ | for one |
| 00:12.35 | bkw_ | well see in this case you can't |
| 00:12.38 | [hC] | and calls will go to them. in this case, they ALSO have to have an entry in iax.conf called [myprovider], no? |
| 00:12.40 | bkw_ | you want fine control over it |
| 00:12.43 | bkw_ | you can't do it with one entry |
| 00:12.47 | [hC] | ok |
| 00:12.50 | [hC] | fair enough |
| 00:13.13 | [hC] | im just confused, because friend seems to imply that you can send calls back and forth, but only if one side has a register line |
| 00:13.14 | bkw_ | you could with one friend and one user |
| 00:13.21 | bkw_ | or one friend and one peer |
| 00:13.36 | [hC] | What would be the best way to do this then? They need to dial us for our DID, and we need to dial them for outgoing local calls |
| 00:13.39 | bkw_ | you don't have to register |
| 00:13.44 | bkw_ | you could do user@ip |
| 00:13.52 | bkw_ | but you would have to do user:pass@ip |
| 00:13.57 | bkw_ | but I don't recommend that at all |
| 00:14.00 | [hC] | Right |
| 00:14.12 | djflux | I'm having a little issue with a DTA310 and asterisk ... I can call from a SIP soft client to the DTA and the phone attached to the DTA rings however I can't call the soft cleint from the phone on the DTA ... any ideas? |
| 00:14.14 | [hC] | I could just do two contexts on both sides I suppose |
| 00:14.20 | [hC] | er |
| 00:14.21 | bkw_ | brb |
| 00:14.49 | *** join/#asterisk Hydr0p0nX (~Hydr0p0nX@482-brhm1.adsl.wwisp.net) |
| 00:15.03 | file | djflux: I can't call my soft client isn't exactly descriptive, do you get any error messages? a congestion tone? have you done a sip debug to see what the sip messages say? |
| 00:15.50 | *** join/#asterisk ariel_ (~Ariel@dsl-20-177.cofs.net) |
| 00:16.00 | ariel_ | hello everyone. |
| 00:16.09 | djflux | file: sorry ... if I call a number that's not in the asterisk dialplan I get a fast busy on the phone. if I call the number of the SIP soft client it appears that phone call goes through however I get no notification on the soft client and no ringing on the phone |
| 00:16.35 | file | what does the CLI show? |
| 00:16.44 | file | hi ariel |
| 00:17.05 | ariel_ | hello file hope all is well. |
| 00:17.17 | file | meh |
| 00:17.49 | *** join/#asterisk bkw_ (~brian@bkw.developer.and.friend.of.asterisk) |
| 00:17.49 | *** mode/#asterisk [+o bkw_] by ChanServ |
| 00:18.35 | rvhi | any ast_data users? |
| 00:18.45 | rvhi | how do i include a context? |
| 00:19.28 | [hC] | hey bkw.. so what i'll need to do then, is have a client-in and client-out on the provider's side, and then provider-in and provider-out on the client side |
| 00:19.57 | ariel_ | sounds like lots of in and out's |
| 00:20.03 | [hC] | i guess client-in will be type=user and client-out will be friend |
| 00:20.14 | ariel_ | user - peer |
| 00:20.22 | [hC] | client-in on the server side will be for us dialing into them, and client-out will be used for sending the DID to us |
| 00:20.28 | file | ariel_: thanks for correcting him while I was eating |
| 00:20.37 | djflux | file: http://pastebin.ca/11552 |
| 00:20.41 | ariel_ | eating you get to eat...? |
| 00:20.57 | [hC] | ok ill use user and peer then |
| 00:20.58 | file | yesssss |
| 00:21.04 | file | djflux: subscribe has nothing to do with a call |
| 00:21.51 | djflux | file: I just did a sip debug ip [IP] and called |
| 00:22.00 | djflux | that's what appeared on the console :) |
| 00:22.30 | file | then your ATA is SERIOUSLY weird |
| 00:23.14 | *** join/#asterisk tld (~tld@80.203.70.227) |
| 00:23.32 | ariel_ | djflux, what sip device do you have? |
| 00:23.52 | djflux | Windows Messenger ... probably not the best client to test with :) |
| 00:23.54 | file | subscribe... with SDP... that's weird, really weird |
| 00:23.57 | ariel_ | bkw_, did you ever go to the Philippines for your installation there? |
| 00:24.03 | ariel_ | argh |
| 00:24.10 | ariel_ | get xlite, get xlite |
| 00:24.19 | djflux | Windows Messenger 5.0.0468 |
| 00:24.29 | ariel_ | djflux, only 4.7 worked correctly. |
| 00:24.29 | file | that's the softphone |
| 00:24.36 | file | but his actual device is something else... I forget |
| 00:24.39 | djflux | I can call the DTA extension fine |
| 00:24.45 | file | it's sending a SUBSCRIBE instead of an INVITE |
| 00:24.55 | djflux | DTA310 |
| 00:25.17 | ariel_ | dta310 sip express I have not seen it in over 2 years. |
| 00:25.43 | bkw_ | ariel_, not yet.. I was going over there to redo some stuff |
| 00:25.52 | bkw_ | and to crack da whip type thing |
| 00:25.52 | bkw_ | hehe |
| 00:25.54 | file | bkw_: wanna see something weird? |
| 00:26.00 | djflux | from Messenger to DTA rings and works |
| 00:26.01 | bkw_ | file sure send me your picture |
| 00:26.04 | ariel_ | djflux, use this for testing. http://www.xten.com/index.php?menu=products&smenu=download |
| 00:26.08 | file | bkw_: you already have one sexy! |
| 00:26.12 | bkw_ | haha |
| 00:26.15 | file | bkw_: anyway, look at http://pastebin.ca/11552 |
| 00:26.18 | djflux | ariel_, k ... downloading now |
| 00:26.24 | file | it's a subscribe... with SDP! |
| 00:26.27 | ariel_ | bkw_, that was over 4 months ago you were going. |
| 00:27.00 | bkw_ | file and? |
| 00:27.07 | *** join/#asterisk Moc (~Moc@modemcable165.109-70-69.mc.videotron.ca) |
| 00:27.08 | file | bkw_: he's actually trying to place a call |
| 00:27.09 | bkw_ | ariel_, ya |
| 00:27.37 | Moc | hail |
| 00:27.41 | ariel_ | Windows Messager version 5 and up had problems with asterisk 4.7 work just fine. |
| 00:27.41 | file | hi Moc |
| 00:28.25 | ariel_ | I hate getting a summer cold. I feel like shit today. Maybe I should go to bed early today. |
| 00:28.28 | bkw_ | 5 had sip support removed |
| 00:28.34 | bkw_ | and you have to use insecure=yes |
| 00:29.56 | djflux | ariel_, no dice with xlite |
| 00:30.00 | djflux | same problem |
| 00:30.11 | file | it's the DTA301 |
| 00:30.12 | djflux | can call the hard phone with xlite, but not the other way around |
| 00:30.18 | CoaxD | SMOKIN THE GANJ |
| 00:30.22 | djflux | hooptie! |
| 00:30.23 | file | er 310 |
| 00:30.31 | djflux | hooptie ass 310 |
| 00:30.36 | *** join/#asterisk zilas (~1@adsl-211-229-248.asm.bellsouth.net) |
| 00:32.32 | ariel_ | djflux, set the dta 310 settings to inscure=very in the sip.conf |
| 00:32.47 | bkw_ | yes would accomplish the same thing |
| 00:32.50 | bkw_ | you don't need a secret |
| 00:32.53 | djflux | ariel_, already done |
| 00:33.02 | bkw_ | insecure=very would still make you need a secret |
| 00:33.08 | djflux | that's the only way I could get the stupid thing to connect to asterisk :) |
| 00:33.08 | rvhi | anyone knows how to include a context in ast_data? |
| 00:33.13 | ariel_ | djflux, get a gun and shoot it then get yourself an sipura. |
| 00:33.17 | *** join/#asterisk Inv_arp (junya@adsl-3-244-124.mia.bellsouth.net) |
| 00:33.17 | djflux | LOL |
| 00:33.25 | file | the SIP stack on the DTA310 is crap, 'nuff said |
| 00:34.06 | djflux | would having both insecure=yes and insecure=very in the same extenstion in sip.conf cause issues? |
| 00:34.23 | file | the world would explode |
| 00:34.25 | ariel_ | djflux, depending on which one it is. the black ones you could upgrade the firmware. you will need to check with there distributor adp I think. |
| 00:34.38 | *** join/#asterisk newmedian (~crowlther@Quebec-HSE-ppp230300.qc.sympatico.ca) |
| 00:34.42 | djflux | I have a beige one from packet8 |
| 00:34.53 | ariel_ | djflux, remember the gun. |
| 00:34.57 | djflux | lol |
| 00:35.19 | djflux | dang |
| 00:35.25 | djflux | hooptie ass DTA |
| 00:35.36 | file | This answer brought to you by Asterlink and Cluecon. Have a nice day, and remember to attend Cluecon! ^^^ |
| 00:36.22 | file | wrong answer! |
| 00:36.40 | file | although I never asked a question... |
| 00:37.51 | bkw_ | the answer would be 42 |
| 00:37.58 | [hC] | so... does the username i send to the other host for an iax peer.. do they have to have that username set as the iax context name in iax.conf? |
| 00:38.08 | file | bkw_: you should do all the stuff for me you said you'd do today |
| 00:38.14 | zip | bkw_, ha hhgttg :) |
| 00:38.16 | bkw_ | IAX2/remoteusername@localpeer/exten |
| 00:38.23 | *** join/#asterisk newbien (~e@147.241.33.65.cfl.res.rr.com) |
| 00:38.27 | [hC] | ie if i am doing a Dial(IAX/provider/extension) they have to have an entry in iax.conf called [provider] ? |
| 00:38.35 | [hC] | it seems to be that way |
| 00:38.35 | bkw_ | DO NOT DIAL LIKE THAT |
| 00:38.36 | bkw_ | damn |
| 00:38.40 | bkw_ | who tells you to do that |
| 00:38.42 | bkw_ | do this |
| 00:39.08 | bkw_ | IAX2/remote_username_in_remote_iax_conf_file@local_peer_name_in_local_iax_conf/remote_exten |
| 00:39.21 | bkw_ | you can't change the username if you do IAX2/peer/exten |
| 00:39.23 | *** join/#asterisk MikeJ[Laptop] (~jayden@pcp02795302pcs.roylok01.mi.comcast.net) |
| 00:39.45 | [hC] | Okay.. i was just following a voip-info example.. :/ |
| 00:40.10 | bkw_ | and 70% of that is bullshit |
| 00:40.13 | bkw_ | NEXT |
| 00:40.13 | bkw_ | brb |
| 00:40.18 | sean | then change it. |
| 00:40.20 | sean | (-; |
| 00:40.26 | MikeJ[Laptop] | hey hey hey |
| 00:40.38 | *** join/#asterisk jets (~brian@guardian.pmt.org) |
| 00:40.40 | MikeJ[Laptop] | I'm the other 30% |
| 00:40.50 | MikeJ[Laptop] | or was I the 70%? |
| 00:41.01 | jets | shouldn't a pri debug go to my /var/log/asterisk/full or whatever i have debug specified in logger.conf? |
| 00:41.51 | syle | so whats difference between iax and iax2? |
| 00:42.14 | MikeJ[Laptop] | syle, iax is older |
| 00:42.14 | file | run a diff on the source to find out |
| 00:42.20 | ariel_ | syle, lots of things. iax was replaced long ago. |
| 00:42.34 | MikeJ[Laptop] | iax is also commonly used as a name for what is now iax 2 |
| 00:42.36 | syle | so is iax.conf in /etc/asterisk iax2? |
| 00:42.41 | file | yes |
| 00:42.45 | syle | ok |
| 00:43.02 | MikeJ[Laptop] | there is no old iax in asterisk |
| 00:43.15 | ariel_ | I wish they would have just renamed the iax2 back to iax. |
| 00:43.49 | ariel_ | MikeJ[Laptop], yes it's still there in stable last I checked. Or at least on version 1.0.5 |
| 00:44.12 | MikeJ[Laptop] | really? |
| 00:44.40 | MikeJ[Laptop] | maybe I have just never paid any attention to it |
| 00:44.55 | ariel_ | yes I found a asterisk box still using it between two computers on monday. I changed them.... |
| 00:45.02 | MikeJ[Laptop] | pretend it does not exist, I do :p |
| 00:45.51 | ariel_ | Those boxes have so many older file setup and patches by there ex employee I am not looking at upgrading them. |
| 00:46.17 | *** join/#asterisk sandnigg0r (~niggerplz@66-55-197-254.gwi.net) |
| 00:46.37 | MikeJ[Laptop] | :) |
| 00:49.40 | *** join/#asterisk JunK-Y (~junky@modemcable174.107-81-70.mc.videotron.ca) |
| 00:51.20 | nwhit | with call parking, how do I return a parked call back to the extension that parked it after the timeout? |
| 00:56.38 | newbien | nwhit: maybe transfer to previous extension ? |
| 00:57.06 | JunK-Y | nwhit: it will return automatiquely to the exten which parked it after the timeout. |
| 00:58.05 | *** join/#asterisk Nethab (~chatzilla@adsl-67-113-141-170.dsl.sntc01.pacbell.net) |
| 00:58.57 | Nethab | anyone know the proper way to use the new ${DB()} functions? |
| 01:01.11 | RickTick | hello ALL: How can I append a "1" to my callerID? Is the prefix command obsolete? |
| 01:01.35 | ManxPower | RickTick: SetCIDNum(${CALLERIDNUM}1) |
| 01:02.06 | RickTick | ManxPower ..thanks |
| 01:03.32 | ManxPower | Whoo! Whoo! The USD is slightly stronger today! |
| 01:03.44 | ariel_ | RickTick, you need to added it to the end or to the beginning? 1800 or 80012345671 |
| 01:05.15 | Sedorox | ManxPower: compared to....? |
| 01:05.25 | Nethab | compared to his monkey |
| 01:05.33 | *** join/#asterisk kimo_sabe (nick@zappa.azrackspace.net) |
| 01:05.49 | Nethab | who uses DBget and DBput |
| 01:06.30 | ManxPower | Sedorox: compared to a few days ago. |
| 01:06.40 | ManxPower | It's only a few cents, but every little bit helps |
| 01:07.16 | newmedian | Looking to enjoy those Euros ManxPower? |
| 01:09.15 | RickTick | ManxPower ... I actually want to put a " 1 " infront of the 10 digit callerID. |
| 01:09.35 | RickTick | Areil to the front |
| 01:10.17 | ariel_ | SetCIDNum(1${CALLERIDNUM}) |
| 01:12.01 | RickTick | Ariel: I tried that and it gives ... 1"xxxxxxxxxx" .I want 1xxxxxxxxxx |
| 01:12.57 | Nethab | are you sure you didn't accidently use CALLERID, instead of CALLERIDNUM |
| 01:13.17 | ariel_ | RickTick, what is your string? |
| 01:14.31 | RickTick | Thanks guys... I mistaken was using CALLERID ... instead of CALLERIDNUM ... I thought they were the same. |
| 01:14.53 | Nethab | no problem |
| 01:14.57 | Nethab | happens all the time |
| 01:15.05 | *** join/#asterisk juice (~juice@mo-65-173-76-11.dyn.sprint-hsd.net) |
| 01:15.11 | mmlj4 | hey ManxPower: i got * up and running here, 4 softphones, they all work locally, and voicemail works |
| 01:15.23 | Nethab | i hear a but.. in there |
| 01:15.47 | ariel_ | mmlj4, great to hear it. |
| 01:15.47 | mmlj4 | Nethab: but out :-P |
| 01:15.47 | drbrown | anyone had any experience with fax detection? |
| 01:16.08 | ariel_ | drbrown, as incoming via zap ports? |
| 01:16.17 | drbrown | ariel: yes |
| 01:16.58 | drbrown | ariel_: it works just fine, but my phones ring once before it sends the call to the fax |
| 01:17.23 | ariel_ | ad a wait |
| 01:17.39 | Nethab | the call has to be Answer() then wait(1) before trying to Dial your phone |
| 01:17.46 | drbrown | ariel_: I added a wait(2), should I increase this? |
| 01:17.59 | ariel_ | try 3 |
| 01:18.13 | Nethab | i guess it depends on the length of the fax tone |
| 01:18.22 | drbrown | ariel_: is it in seconds? |
| 01:18.37 | JunK-Y | yes |
| 01:19.22 | ariel_ | JunK-Y, hello how are you tonight? |
| 01:19.25 | JunK-Y | huh? its in ms apparently, that's change? |
| 01:19.36 | JunK-Y | im fine |
| 01:19.44 | JunK-Y | im watching the simpsons now :) |
| 01:20.03 | Nethab | they changed the DBput and DBget stuff too |
| 01:20.07 | Nethab | thanks for the warning |
| 01:20.42 | ariel_ | I am watching for the 1 million'th time a pooh movie with my baby. |
| 01:21.17 | Nethab | watching and typing, that's multiasking if i ever heard it |
| 01:23.31 | *** join/#asterisk roamer323 (~sing@HSE-MTL-ppp64171.qc.sympatico.ca) |
| 01:31.58 | ariel_ | just wondering, why do some people say Many happy returns of the day instead of just wishing the person a happy birthday. |
| 01:32.49 | *** join/#asterisk iq|laptop (~iq@63-230-45-16.omah.qwest.net) |
| 01:32.52 | Mavvie | ariel_: sounds like they're wishing you a long life instead of a wishing you a happy day. |
| 01:33.41 | ariel_ | no I am watching Winnie the Pooh with my kid. That is what they say.. just sounds strange to me. |
| 01:34.29 | newmedian | But the talking bear isn't. |
| 01:35.25 | *** join/#asterisk ChrisHodgetts (~chris@topanga.archnetnz.com) |
| 01:35.34 | ChrisHodgetts | hello |
| 01:36.02 | *** join/#asterisk znoG (gs@200.115.216.109) |
| 01:36.17 | iq|laptop | hello ChrisHodgetts |
| 01:36.32 | [hC] | is there any way to determine a better error when trying to dial over an iax peer than 'no authority found' - like can i figure out if it was a bad user, bad password, or something? |
| 01:37.32 | file | you use your head to figure it out |
| 01:38.14 | [hC] | is it possible to get that error if the context i am being dropped into has a problem? |
| 01:38.25 | [hC] | cause from what i can see everything else is fine |
| 01:38.41 | file | nope, it would say a different error |
| 01:38.44 | file | it's your username/pass that is wrong |
| 01:41.44 | file | [hC]: you should come to Cluecon, www.cluecon.com, you'd learn lots! |
| 01:41.50 | [hC] | Har! |
| 01:42.02 | Qwell | file: get me a pass :p |
| 01:42.25 | file | you get a free pass by paying for it |
| 01:42.40 | Qwell | ahh, ok |
| 01:42.48 | file | isn't that logical? |
| 01:42.51 | ChrisHodgetts | does anyone use Linphone as a softphone to talk to asterisk here? |
| 01:42.58 | Qwell | file: somewhat |
| 01:43.17 | file | but really, $650 for access and that includes hotel for 3 days 2 nights, and 3 lunches |
| 01:43.31 | Qwell | oh, thats not bad |
| 01:43.37 | file | plus you get to meet the minds behind asterisk, and watch us give presentations |
| 01:43.41 | file | plus there's a Q&A session! |
| 01:43.48 | Qwell | umm |
| 01:43.52 | Qwell | no thanks :P |
| 01:43.57 | file | c'mon |
| 01:44.00 | file | you know you wanna come |
| 01:44.14 | Qwell | hmm, that domain isn't even loading |
| 01:44.24 | file | but in real life I'll actually start rambling on and on |
| 01:44.25 | file | http://www.cluecon.com/ |
| 01:44.31 | Qwell | no such animal |
| 01:44.34 | file | August 3rd to 5th! |
| 01:44.39 | Qwell | Where is it? |
| 01:44.42 | file | in Chicago |
| 01:44.51 | newbien | ChrisHodgetts: tried many times to get linphone to be registerd in ast*; segfaults every time |
| 01:44.53 | Qwell | maybe my boss will send me |
| 01:45.01 | file | try and see, we'd love to have you attend |
| 01:45.11 | newmedian | ChrisHodgetts: you're aware of http://www.sipserve.co.nz/ ? |
| 01:45.12 | ChrisHodgetts | newbien I have it registering, I got the latest |
| 01:45.19 | Qwell | You're having them filter my IP on cluecon.com, aren't you? :P |
| 01:45.19 | ChrisHodgetts | newmedian yeah I signed up |
| 01:45.23 | ChrisHodgetts | thats really why I cam in |
| 01:45.23 | Qwell | You don't want me there at all! |
| 01:45.26 | file | cant' say I am! |
| 01:45.32 | newbien | ChrisHodgetts: linphone 1.01? |
| 01:45.54 | ChrisHodgetts | I got it working on 0.12.2 |
| 01:46.08 | Qwell | man, slow...must be ipv6 |
| 01:46.15 | ChrisHodgetts | well, say *working* I got it so I could call local extentions |
| 01:46.24 | Qwell | ahh, it is |
| 01:46.28 | ChrisHodgetts | newmedian do you use sipserve.net.nz? |
| 01:46.44 | newbien | ChrisHodgetts: k, segfaults for linphone 1.0x using the linphone 0.12.2 tiki setups |
| 01:47.04 | newmedian | ChrisHodgetts: I've got a (test) account, yes. |
| 01:47.12 | file | Qwell: see? we don't stop anyone from coming! |
| 01:47.13 | ChrisHodgetts | could you help me with something then please newmedian |
| 01:47.14 | Qwell | file: Do you know whoever is hosting the domain? They should really add an ipv4.cluecon.com or something |
| 01:47.40 | ChrisHodgetts | newbien hmmmm -- I have never tried linphone 1.0x |
| 01:47.43 | newmedian | ChrisHodgetts: ask away and let's see what happens |
| 01:47.44 | Qwell | everyone should... |
| 01:47.55 | iq|laptop | file, how many seats are available? |
| 01:47.56 | ChrisHodgetts | newmedian I have it set up to the point where it attemtps to bridge the call |
| 01:47.59 | ChrisHodgetts | but I hear nothing |
| 01:48.01 | file | iq|laptop: tons |
| 01:48.06 | ChrisHodgetts | I have a friend who called my sipserve.net.nz number |
| 01:48.09 | ChrisHodgetts | heard my talking clock |
| 01:48.27 | iq|laptop | file, great - I'll try to talk my boss into paying for all this ;) |
| 01:48.29 | file | the more the merrier, and you will definately learn some valuable stuff |
| 01:48.31 | ChrisHodgetts | but when I attempt to make a call out, linphone tells me the call is connected |
| 01:48.40 | ChrisHodgetts | and I see asterisk bridging |
| 01:48.42 | ChrisHodgetts | but I get no audio |
| 01:49.18 | ChrisHodgetts | calling internal extentions work, calling out via my zap device works |
| 01:49.26 | newmedian | ChrisHodgetts: This sounds like one of those NAT/public IP problems that everyone here is so good at debugging. :) |
| 01:49.42 | ChrisHodgetts | is newmedian being sarcastic |
| 01:49.46 | ChrisHodgetts | ;) |
| 01:49.59 | kryme | Speaking of ... is there some great FAQ to make SIP and NAT work happily together? |
| 01:50.08 | newmedian | No, not really. I see that kind of question come up a lot if you lurk around this channel. |
| 01:50.31 | ChrisHodgetts | I belive I have nat working, by virtue that inbound calls to my external register can hear things |
| 01:50.45 | *** part/#asterisk Nuxi (~nuxi@cust-sdsl-204-250-82-202.bzn-co-i1000-01.bridgeband.net) |
| 01:50.52 | ChrisHodgetts | but I guess not nat from laptop > pabx > gateway > outsideworld |
| 01:51.47 | ariel_ | ChrisHodgetts, start with canreinvite=no |
| 01:51.59 | file | ariel_: are you going to Cluecon? |
| 01:52.11 | ChrisHodgetts | tried that |
| 01:52.18 | *** join/#asterisk TheEmperor (~user@203.114.48.47) |
| 01:52.25 | ChrisHodgetts | do I do this on the extention or on the sip proxy |
| 01:52.30 | TheEmperor | hello |
| 01:52.32 | file | hello |
| 01:52.38 | ariel_ | file, at this present time I can't say. I don't have the cash for it right now. But maybe alittle close I might. |
| 01:52.39 | TheEmperor | can someone please tell me what this means: -- Saved useragent "PHONE" for peer 2001 |
| 01:52.40 | ChrisHodgetts | only done on sip proxy at present |
| 01:52.40 | Qwell | file: get the owner of cluecon.com to add an ipv4 only dns entry :p |
| 01:52.58 | file | Qwell: fyi only mail has IPv6 :P |
| 01:53.07 | Qwell | no, they all do |
| 01:53.20 | file | nope |
| 01:53.21 | Qwell | cluecon.com. 1137 IN AAAA 2002:42fa:4403:: |
| 01:53.22 | Qwell | yep |
| 01:53.34 | file | cluecon.com. 1800 IN A 66.250.68.3 |
| 01:53.36 | file | that's the only one I have |
| 01:53.37 | Qwell | dig aaaa |
| 01:53.52 | ChrisHodgetts | ariel_ sorry, is that done in the sip proxy or in the extention? |
| 01:54.02 | *** join/#asterisk Juggie (~agony@CPE00c049d9f271-CM00137186c8d8.cpe.net.cable.rogers.com) |
| 01:54.10 | file | I even had anthm make sure |
| 01:54.30 | Qwell | file: you need to do `dig aaaa domain.com` to get the ipv6 hosts (generally) |
| 01:54.53 | *** join/#asterisk Pkunk (~Pkunkage@mbbs.munnabhai.info) |
| 01:55.10 | CyberKnet | cd 4 |
| 01:55.13 | CyberKnet | heh |
| 01:55.16 | file | dear me you're right |
| 01:55.19 | Pkunk | is there any such thing as a "stable" broadvoice server ? |
| 01:55.20 | Qwell | file: I'm always right. :P |
| 01:55.25 | file | Qwell: HA |
| 01:55.37 | file | you should! |
| 01:55.43 | file | Qwell: if you go to Cluecon I'll buy you a drink |
| 01:55.44 | Pkunk | currently i've to keep changing servers nearly everyday |
| 01:56.02 | Qwell | file: I think I owe drumkilla like 8 drinks. ManxPower 1 or 2. |
| 01:56.12 | file | how many do you owe me? lol |
| 01:56.21 | Qwell | umm |
| 01:56.24 | newbien | Pkunk: why change? server wont authenticate a caller? |
| 01:56.25 | Qwell | alot :p |
| 01:56.35 | nwhit | with call parking, how do I return a parked call back to the extension that parked it after the timeout? it is currently returning to s,1 of the default context |
| 01:56.41 | ChrisHodgetts | newmedian I dont belive so |
| 01:57.57 | JunK-Y | nwhit: which version? with latest head, it returns to the extension which parked the call. |
| 01:58.11 | nwhit | JunK-Y, I am running 1.0.7 |
| 01:58.29 | JunK-Y | maybe that's why. |
| 01:58.30 | nwhit | JunK-Y, I was thinking about upgrading to the new cvs head |
| 01:58.34 | JunK-Y | im not running stable. |
| 01:58.40 | JunK-Y | ya should. |
| 01:58.47 | nwhit | JunK-Y, how is the current head? |
| 01:59.05 | nwhit | i had problems a couple of weeks ago and went back |
| 01:59.18 | JunK-Y | whatcha mean how? im running it on all my prod machines. |
| 01:59.30 | nwhit | JunK-Y, stable enough? |
| 02:00.03 | file | JunK-Y: The cluecon registration is up btw |
| 02:00.11 | file | nwhit: and you should go to Cluecon dude! |
| 02:00.34 | JunK-Y | file: yeah i know, i'll tell my boss to register me. |
| 02:00.50 | JunK-Y | he's paying the conf, and im paying the rest. |
| 02:00.57 | nwhit | file, oh... why? |
| 02:01.00 | file | so you're paying airfare and spending? |
| 02:01.09 | JunK-Y | do you have the list of speakers? |
| 02:01.12 | file | nwhit: it's a great learning experience, it'll help you with asterisk |
| 02:01.15 | file | JunK-Y: not yet |
| 02:01.15 | JunK-Y | yes |
| 02:01.34 | file | plus you get to meet the minds behind it, meet he people who help in here (like me) |
| 02:01.34 | nwhit | when is it and where? |
| 02:01.42 | file | Chicago, IL August 3rd to 5th |
| 02:01.57 | file | $650 for it, but that includes 3 days 2 nights hotel and 3 lunches |
| 02:02.04 | nwhit | what am I missing here... ldconfig is complaining about "annot find -lidn" |
| 02:02.21 | nwhit | when is the last day to register? |
| 02:02.23 | newmedian | Perhaps you should just make an autogreet that pimps Cluecon? |
| 02:02.24 | file | and, according to my research, the hotel does have high speed internet |
| 02:02.28 | Pkunk | newbien: i dunno . it just works fine for a while |
| 02:02.30 | file | newmedian: that would be a great idea |
| 02:02.38 | file | nwhit: a long ways away from now :) |
| 02:02.41 | Pkunk | newbien: and then after a day you get DIALSTATUS=CONGESTION |
| 02:02.56 | nwhit | ok... i am definitely interested |
| 02:03.20 | nwhit | what library is idn? |
| 02:03.24 | file | http://www.cluecon.com/ has a schedule for the days |
| 02:03.31 | Pkunk | sometimes , i need to try 3-4 diff. servers until i get one that works |
| 02:03.48 | ChrisHodgetts | ok :) |
| 02:04.19 | Pkunk | is it an asterisk issue ? or does Broadvoice just suck ? |
| 02:04.56 | *** join/#asterisk mtgh (~chatzilla@dsl093-001-038.det1.dsl.speakeasy.net) |
| 02:06.58 | nwhit | ~idn |
| 02:07.08 | file | Pkunk: Broadvoice has been having problems lately |
| 02:07.19 | nwhit | what am I missing here... ldconfig is complaining about "cannot find -lidn" on compiling cvs head |
| 02:12.26 | TheEmperor | this call doesn't go through when i try and dial out on a zap channel, any ideas? |
| 02:12.28 | TheEmperor | -- Executing Answer("SIP/2005-9ff1", "") in new stack |
| 02:12.29 | TheEmperor | <PROTECTED> |
| 02:12.29 | TheEmperor | <PROTECTED> |
| 02:12.29 | TheEmperor | <PROTECTED> |
| 02:14.57 | *** part/#asterisk wwalker (~wwalker@wwalker.sustaining.supporter.pdpc) |
| 02:15.49 | newbien | Pkunk: DIALSTATUS=CONGESTION more than 50% of the time? |
| 02:16.31 | bkw_ | and what is up in here |
| 02:16.44 | nwhit | WOW!!! my intercom now works on my snom phone... woohooo!!!! |
| 02:17.05 | file | bkw_: building kernel for PBlX |
| 02:17.11 | bkw_ | w00t |
| 02:17.15 | file | 2.6.9 hated me |
| 02:17.19 | file | it was b0rken |
| 02:18.26 | *** join/#asterisk Carp1 (carp_xigon@ip-204-97-151-139.modem.logical.net) |
| 02:18.27 | file | bkw_: can I just do a make iso afterwards? |
| 02:18.53 | Carp1 | hey asterisklings |
| 02:19.20 | niZon | asterlings sounds better |
| 02:19.28 | Carp1 | true |
| 02:19.32 | niZon | :P |
| 02:19.33 | file | watch out, your asterisk is showing! |
| 02:19.47 | niZon | uh oh! |
| 02:19.52 | rabelais | does nufone support number portability? |
| 02:20.12 | rabelais | like, can I transfer my number to nufone? |
| 02:20.20 | file | unless you live in Michigan, no |
| 02:20.46 | Carp1 | THey only allow DIDs from Mich |
| 02:20.55 | Carp1 | and of course toll free numbers |
| 02:21.33 | rabelais | oh |
| 02:24.55 | Carp1 | hmm lol |
| 02:25.15 | Carp1 | so file, you never got that app working? |
| 02:25.51 | *** join/#asterisk Weezey (WeezeyD@206.210.109.233) |
| 02:25.59 | Weezey | jbot, have you seen my baseball? |
| 02:26.46 | iq|laptop | ~seen baseball |
| 02:26.47 | jbot | i haven't seen 'baseball', iq|laptop |
| 02:26.57 | Weezey | heh |
| 02:27.08 | iq|laptop | ~iq |
| 02:27.09 | jbot | [iq] that apts IQ is lower than 1 |
| 02:27.31 | Weezey | ~three point one four |
| 02:27.32 | jbot | three point one four is, like, a great song about finding a new vagina by The Bloodhoung Gang. |
| 02:29.03 | L|NUX | ~iq L|NUX |
| 02:29.07 | L|NUX | ~L|NUX |
| 02:30.01 | Weezey | ~L|NUX |
| 02:30.04 | jbot | it has been said that l|nux is struggling to reunite his parents via asterisk |
| 02:30.40 | *** join/#asterisk ChulJin (~chuljin@adsl-68-121-94-237.dsl.irvnca.pacbell.net) |
| 02:30.51 | ChulJin | Good evening Gentlemen! |
| 02:31.19 | *** part/#asterisk darwin35 (~darwin35@24.3.226.147) |
| 02:31.31 | nwhit | bye |
| 02:31.34 | newmedian | All our base? |
| 02:32.22 | Weezey | no your |
| 02:32.32 | *** join/#asterisk outtolunc (~me@ppp-69-237-32-168.dsl.pltn13.pacbell.net) |
| 02:32.40 | Weezey | quiet tonight |
| 02:32.56 | jskcr|lappy | yuo |
| 02:33.00 | jskcr|lappy | oops yup |
| 02:33.20 | ChulJin | a long shot, but I might as well try: newmedian, are you Justin Newman of Newman Telecom? |
| 02:33.23 | ChulJin | :) |
| 02:33.47 | rabelais | I'm looking for a reliable service provider to transfer my broadvoice number over to, any suggestions? |
| 02:33.55 | Weezey | Do you live across the hall from Jerry Seinfeld? |
| 02:34.10 | rabelais | area code 310 |
| 02:34.26 | bkw_ | rhm |
| 02:34.29 | bkw_ | 310 what area is that? |
| 02:34.33 | ChulJin | WEST SIDE! |
| 02:34.34 | ChulJin | :) |
| 02:34.36 | file | Santa Monica, California |
| 02:34.44 | bkw_ | we could actually do that ya know |
| 02:34.53 | bkw_ | we do cover most of cali |
| 02:34.56 | bkw_ | for did's |
| 02:34.56 | ChulJin | rabelais: I don't know if they do LNP, but I rather like VoicePulse for DIDs |
| 02:35.01 | ChulJin | outgoing is rather pricey tho |
| 02:35.13 | file | bkw_: oh rightttttt I forgot about that |
| 02:35.29 | bkw_ | did you kill the bot? |
| 02:35.31 | *** join/#asterisk jeffik (jefik@69.158.19.117) |
| 02:35.36 | file | bkw_: they've done it again |
| 02:35.51 | jeffik | question about sipura 1001 |
| 02:35.56 | Weezey | damn. h323 just is not working for me. |
| 02:36.14 | rabelais | Weezey: is aix.cc even a voip service provider? |
| 02:36.20 | Weezey | sure |
| 02:36.33 | TheEmperor | hi guys, what does this mean? Asterisk ended with exit status 1 |
| 02:36.33 | bkw_ | iax.cc you mean? |
| 02:36.33 | TheEmperor | Asterisk died with code 1. |
| 02:36.40 | Weezey | bkw; yes |
| 02:36.59 | *** join/#asterisk tsp (~tyler@S01060080c825173c.vc.shawcable.net) |
| 02:37.03 | jeffik | Weezey: may i aks you? |
| 02:37.12 | tsp | why is asterisk and alsa making all these clicking sound throughout my incoming audio? |
| 02:37.14 | rabelais | ah, iax.cc yes...he told me aix.cc |
| 02:37.14 | Weezey | aks or axe? |
| 02:37.22 | bkw_ | eeeks |
| 02:37.27 | jeffik | Weezey: axe |
| 02:37.29 | jeffik | please |
| 02:37.38 | rabelais | voicepulse, ok...I'll look at that, chanks ChulJin |
| 02:37.39 | bkw_ | no iax is pronounced eeeks |
| 02:37.50 | *** part/#asterisk Gand_DJ (~gandalf@ptr-207-54-104-24.ptr.terago.ca) |
| 02:37.51 | bkw_ | like "eeeks a mouse" |
| 02:37.55 | file | eeks! |
| 02:37.59 | jeffik | jsut set up a 1001 works ok outbound, when i call the extension says user is unavailablre |
| 02:38.03 | ChulJin | rabelais: sorry, specifically VoicePulse Connect (http://connect.voicepulse.com) |
| 02:38.05 | file | or as the swedish call it, eye axe! |
| 02:38.07 | ChrisHodgetts | wouldnt that bee eek a mouse? |
| 02:38.17 | bkw_ | well ya |
| 02:38.20 | bkw_ | but you get the picture |
| 02:38.22 | bkw_ | smart ass |
| 02:38.23 | bkw_ | :P |
| 02:38.25 | ChulJin | Does anyone, by any chance, use any of the NV*Detect[s]? |
| 02:38.26 | Weezey | jeffik; did you make it available for incoming? |
| 02:38.34 | bkw_ | haha |
| 02:38.46 | bkw_ | ChrisHodgetts, na don't do that |
| 02:38.48 | jeffik | Weezey: do i set that in the 1001 config? |
| 02:39.00 | file | yay 1001, my extension! |
| 02:39.08 | jeffik | ok |
| 02:39.18 | rabelais | Weezey: is iax.cc reliable? |
| 02:39.26 | Weezey | I haven't had any trouble yet. |
| 02:39.51 | rabelais | Weezey: do you know if they support number portability? |
| 02:39.58 | jeffik | rabeblais: i use them some times |
| 02:40.00 | Weezey | most places do. |
| 02:40.08 | jeffik | can't get them for support though |
| 02:40.19 | jeffik | i have a bette provider using a SER |
| 02:40.21 | rabelais | cause 1.7c a min and only $1.49 for a local hnumber is amazing ;) |
| 02:40.23 | Weezey | oh, no, you gotta know what you're doing. |
| 02:40.34 | ChulJin | rabelais: is that incoming? |
| 02:40.38 | tsp | whats up with alsa/ |
| 02:40.40 | Weezey | in or out |
| 02:40.46 | *** join/#asterisk Grooby (~Grooby@ip24-250-126-171.dc.dc.cox.net) |
| 02:40.54 | rabelais | well, I know what I'm doing, and I can't even get through to broadvoice support to complain anyhow |
| 02:41.00 | jeffik | File: can you give mre a hint whre to enablre it? |
| 02:41.01 | rabelais | an, so incoming isn't free? |
| 02:41.22 | Grooby | I am with BV |
| 02:41.33 | Grooby | and i can't get international call out propery..it's pissing me off |
| 02:41.36 | file | jeffik: what happens on your asterisk CLI? |
| 02:41.36 | jeffik | rabrelais: no it's per minutes, but they promised me a chicago did for 6 weeks and nevre delivererfd |
| 02:41.39 | ChulJin | rabelais: do you expect to have more than 6h22m21s of incoming traffic? |
| 02:41.45 | ChulJin | (per month) |
| 02:42.04 | jeffik | All: i'm using a local Tronnto service, us,ca, intl |
| 02:42.35 | *** part/#asterisk tsp (~tyler@S01060080c825173c.vc.shawcable.net) |
| 02:42.39 | rabelais | ChulJin: no, not really...but often times I do call into my number and then bounce a call out, so it'd use twice the line charge |
| 02:43.00 | ChulJin | rabelais: iax.cc=$1.49+$.017 ... voicepulse=$7.99+$.00 ... voicepulse is cheaper after 382 minutes... |
| 02:43.14 | file | TOXIC! |
| 02:43.27 | file | Voicepulse raised their rates ya know |
| 02:43.33 | ChulJin | *blink* |
| 02:43.37 | file | $11/mth now |
| 02:43.44 | ChulJin | oh, that's right |
| 02:43.50 | file | and up to 2.4 cents/min outbound |
| 02:44.00 | file | now, recalculate in 5 milliseconds |
| 02:44.03 | file | okay times up, you lose |
| 02:44.06 | rabelais | well, I'm around 600 minutes total a month |
| 02:44.21 | jeffik | file: can you give me a hint as to whrere to look for allow calls on my 1001 |
| 02:44.23 | rabelais | assuming my bloody service _works_ |
| 02:44.43 | ChulJin | 'bloody'? you are in Santa Monica, aren't you? :) |
| 02:44.55 | file | jeffik: your question is very generic and vague, authentication problem? dialplan problem? codec problem? |
| 02:45.00 | file | jeffik: what happens when you try to call out |
| 02:45.04 | *** join/#asterisk HeppyCat (~unknown@cpe-24-164-217-41.jam.res.rr.com) |
| 02:45.08 | ChulJin | OK, make that 9h19m25s |
| 02:45.19 | HeppyCat | good evening |
| 02:45.19 | rabelais | ya, I am in los angeles |
| 02:45.39 | jeffik | file: oh my calls complete |
| 02:45.51 | *** join/#asterisk PatrickDK (patrickdk@dyn-19-218.myactv.net) |
| 02:45.52 | file | you just can't do a Dial(SIP/1001) ? |
| 02:46.26 | rabelais | uh, the cheapest plan I see at voicepulse is $15 |
| 02:46.39 | file | rabelais: http://connect.voicepulse.com/ |
| 02:46.39 | rabelais | am I missing something? |
| 02:46.41 | file | two separate entities |
| 02:46.44 | file | kinda sorta |
| 02:46.49 | rabelais | ah |
| 02:46.50 | ChulJin | http://connect.voicepulse.com/rates.aspx |
| 02:46.55 | *** join/#asterisk _SMP_ (~SMP@pandora.burned.net) |
| 02:47.14 | jeffik | file: i can call out, just when i dial the sipura 1001 i get recording the user is unavailable |
| 02:47.30 | rabelais | ah, it's like the backdoor |
| 02:47.32 | rabelais | hehe |
| 02:47.50 | *** join/#asterisk mechtn (~Aerbrax@198.164.78.66.aeneasdsl.com) |
| 02:47.53 | file | jeffik: well is it registering to your asterisk? do you have the dialplan setup correctly? what does your asterisk CLI say? |
| 02:47.56 | *** part/#asterisk JunK-Y (~junky@modemcable174.107-81-70.mc.videotron.ca) |
| 02:47.57 | ChulJin | telasip is supposed to be good, I guess...I just signed up with them a few days ago... $14.95/month with 2 DIDs, unlimited incoming and outgoing; sadly they happily ignore my outgoing CID (Option o or not on Dial) |
| 02:48.33 | file | darn, it's not tomorrow yet |
| 02:49.23 | rabelais | what constitutes "long distance" for voicepulse? cause it says 2.4c for long distance |
| 02:49.29 | file | anywhere |
| 02:49.37 | file | well, US/Canada I believe? |
| 02:49.52 | rabelais | ah, so it's $11 fora number and then 2.4c for outgoing calls? |
| 02:49.56 | file | yes |
| 02:50.32 | rabelais | asterlink doesn't seem to be geared towards the consumer |
| 02:50.45 | ChulJin | rabelais: anywhere |
| 02:50.48 | file | define consumer |
| 02:50.53 | ChulJin | er...well...yeah, US |
| 02:51.01 | file | if you want a place where you can prepay money and make calls, that's both asterlink and voicepulse connect |
| 02:51.09 | file | and tons of other companies |
| 02:51.21 | ChulJin | rabelais: the only 'free' outgoing calls are NANPA toll-free, I believe. |
| 02:51.27 | ChulJin | file: it never will be... |
| 02:51.46 | ChulJin | because as soon as you just reach tomorrow, it becomes today, and tomorrow once again jumps out of your grasp |
| 02:51.50 | ChrisHodgetts | why cant people accecpt that sometimes you actually have to *PAY* for shit |
| 02:51.57 | file | ChrisHodgetts: welcome to #asterisk |
| 02:52.03 | ChrisHodgetts | cheers |
| 02:52.15 | ChrisHodgetts | my neck got sore from hanging it before |
| 02:52.20 | file | I give free help everyday, but when someone asks me to essentially write them an entire dialplan for free... then I say no :) |
| 02:52.39 | ChrisHodgetts | hence my , Sometimes you have to pay :) |
| 02:52.45 | file | exactly! |
| 02:52.47 | ChrisHodgetts | file needs to eat |
| 02:52.51 | ChrisHodgetts | pay his net bill |
| 02:52.59 | ChrisHodgetts | drive |
| 02:53.05 | ChrisHodgetts | (I assume) |
| 02:53.07 | *** join/#asterisk daork (~daork@jade.daork.net) |
| 02:53.17 | file | actually not really, but still - I have other stuff |
| 02:53.54 | rabelais | iax.cc seems to make sense, looks like the cheapest given my situation |
| 02:54.22 | ChrisHodgetts | hope not to offend people here, but I get angry when people ask in user forums how to do something that you can tell is part of their job, and expect the user community to help them out |
| 02:54.25 | jeffik | <file> jeffik: well is it registering to your asterisk? do you have the dialplan setup correctly? what does your asterisk CLI say? |
| 02:54.41 | jeffik | file: i'll have to look at it |
| 02:54.45 | jeffik | i'm running *@home |
| 02:54.52 | file | oh dear god |
| 02:54.53 | jeffik | soon to e *@soho |
| 02:55.01 | jeffik | file: it's great i love it |
| 02:55.21 | jeffik | no no it's great for me |
| 02:55.28 | rabelais | I had a question now at the asterisk level, lets say that I connect to iax.cc via the iax protocol, but my phones are all sip, assuming they're all working on the same codec, g.711 or something, will my server be punished heavily trying to transcode between iax and sip? |
| 02:55.42 | jeffik | file: i'a a 20 year nortel admin so this is really good |
| 02:55.46 | file | rabelais: no. |
| 02:55.47 | *** join/#asterisk santiago (~santiago@63.245.86.227) |
| 02:55.53 | file | rabelais: it's not transcoding between different codecs |
| 02:56.17 | rabelais | file: ok, so it's just fiddling with packets, and there's no problem going from iax to sip? |
| 02:56.22 | file | nope, not a problem |
| 02:56.25 | file | in the core it's all the same |
| 02:56.29 | TheEmperor | can someone tell me if my zapata.conf is correct? |
| 02:56.30 | rabelais | wonderful |
| 02:56.42 | file | br |
| 02:56.43 | file | er brb |
| 02:56.47 | |Vulture| | Anyone know if there is a list of the prefixes that are in your local area calling? |
| 02:57.13 | TheEmperor | http://pastebin.ca/11562 |
| 02:57.17 | ChulJin | nah, I want an ITSP that gives me all that and a bag of chips. |
| 02:57.27 | rabelais | would there be any reason for me to use sip to connect to iax.cc if whey support iax? |
| 02:57.36 | |Vulture| | say for instance if 201-222-XXXX is in your area but 201-200-XXXX isn't.. is there a database that shows that? |
| 02:57.49 | ChulJin | ...and, of course, pays me to use their service...in exchange for the valuable feedback I will give them. |
| 02:58.03 | Qwell | |Vulture|: it often changes by LEC, doesn't it? |
| 02:58.05 | ChulJin | vulture: such DBs exists, but from everything I've seen, they are not freely available. |
| 02:58.19 | Qwell | and sometimes, a call may be local to me, but them to me isn't |
| 02:58.21 | file | TheEmperor: you have the same channel range specified twice, with two different signalling types - can't do that |
| 02:58.33 | jeffik | file: our @home has been up for over 90 days witout failure |
| 02:58.36 | |Vulture| | yea I can't find it on Bellsouth's website |
| 02:58.41 | TheEmperor | file:thank you. what should be the correct way? |
| 02:59.07 | TheEmperor | file:i have a 4 port fxo card used to receive and make calls using pstn |
| 02:59.15 | jeffik | file: so may i ask where to look to enalre the 1001? |
| 03:00.04 | ChrisHodgetts | what .. |
| 03:00.10 | ChrisHodgetts | regulation on who can use it! |
| 03:00.22 | newmedian | Pricing and competition, that sort of thing. |
| 03:00.23 | ChrisHodgetts | dat shit is whack! |
| 03:00.30 | *** part/#asterisk santiago (~santiago@63.245.86.227) |
| 03:00.38 | ChrisHodgetts | crappy |
| 03:00.42 | newmedian | Major providers battling it out, and small upstart companies who want to get into the VOIP game paying the price. |
| 03:01.40 | newmedian | From what I recall, pure VOIP i.e. SIP/IAX2 wasn't the issue, it was once you cross to/from the PSTN where they wanted to get into pricing regulation. |
| 03:02.11 | newmedian | (And I seem to recall heading down the mandatory-911 emerg services path as well) |
| 03:04.19 | CoaxD | actually, that shit is NOT whack |
| 03:04.30 | CoaxD | for voip TELCOS, it might not be a bad idea |
| 03:04.47 | CoaxD | right now, voip telcos can get away with *anything*, including going offline tomorrow - and there ain't squat that any customer could do about it |
| 03:05.13 | CoaxD | at least you could have 'regulated' telcos and 'non-regulated' telcos and let the customer make the choice, and assume all risks if going unregulated |
| 03:05.47 | CoaxD | in the USA, if you get a number through a celphone or landline telco, YOU OWN IT |
| 03:06.11 | NewSole2 | but one good thing tho |
| 03:06.13 | CoaxD | if you get a DID from a voip telco, however, THEY own it. Aint no way you can snag that number away from them either without their permission |
| 03:06.52 | CoaxD | which basically means.. ...if your voip telco goes down tomorrow, and you happen to have a business DID through them, you are FUCKED |
| 03:07.00 | NewSole2 | if you are on bell and you want to go to voip with new laws bell has to sell number before they did not |
| 03:07.15 | CoaxD | newsole2: You got it |
| 03:07.23 | CoaxD | newsole2: But unregulated telcos don't have to sell said number |
| 03:07.36 | CoaxD | because the voip telco OWNS the number. |
| 03:08.29 | NewSole2 | yes but I tried to get my normal phone number bell put to my PSTN and they would not release it... now I can go to them next week and demand it |
| 03:08.29 | CoaxD | the only thing you could do would be to get the voip telco to agree (in writing) that YOU OWN THE NUMBER before the transfer is made |
| 03:08.36 | Grooby | so can anyone recommand any good IAX voip provider that has good international rates? |
| 03:08.45 | CoaxD | newsole2: In the USA, they have to allow you to do it regardless |
| 03:08.53 | CoaxD | newsole2: Thats a new law this year |
| 03:08.56 | NewSole2 | in canada they dont |
| 03:08.58 | CoaxD | like. beginning of the year |
| 03:09.05 | CoaxD | er. no, it was last year |
| 03:09.17 | CoaxD | yeah, see, that sucks. but, tis the way it was for years and years here, too |
| 03:09.31 | CoaxD | the pstn had to be heavily modified to get that shit to work, tho |
| 03:09.45 | NewSole2 | this new rules that are passing tomorrow enfocres that they have to sell it |
| 03:09.53 | CoaxD | (The whole POINT of pstn's areacode/exchange system was that you'd never have a phone number that was "owned" outside its native switch) |
| 03:10.24 | *** join/#asterisk Juggie (~agony@CPE00c049d9f271-CM00137186c8d8.cpe.net.cable.rogers.com) |
| 03:10.29 | NewSole2 | we have alot of biz in the area that want to switch over to voip but dont want to change numbers |
| 03:10.43 | CoaxD | duh :) |
| 03:11.01 | NewSole2 | come next week that wont be a problem now |
| 03:11.09 | CoaxD | heh coolio :) |
| 03:12.28 | *** join/#asterisk FuriousGeorge (~brian@ool-43516aa2.dyn.optonline.net) |
| 03:12.28 | NewSole2 | I been following this info for new rules being set for voip in canada |
| 03:12.31 | ChulJin | Approximately six times the timeout having passed: #2. Does anyone, by any chance, use any of the NV*Detect[s]? |
| 03:14.41 | NewSole2 | http://www.crtc.gc.ca/eng/NEWS/RELEASES/2005/r050404.htm |
| 03:14.57 | *** join/#asterisk budi_ (~budi@210.11.72.49) |
| 03:15.06 | FuriousGeorge | should i look for anything special in a switch for a network with an * server? |
| 03:15.50 | newbien | is voip still tax free? broadvoice and packet8 are adding a fcc tax to their bills |
| 03:16.27 | ChrisHodgetts | $39.95 per month, for 5gig |
| 03:21.09 | meshuga | hahah <G> |
| 03:21.19 | meshuga | someone spent a little time on bbses :) |
| 03:21.35 | outtolunc | just a few years |
| 03:21.52 | meshuga | ah i spent my whole childhood of them |
| 03:21.58 | meshuga | woulda went longer but the internet killed them all |
| 03:22.14 | outtolunc | i was joking, i used to run one in the old days |
| 03:22.39 | meshuga | me too |
| 03:23.08 | outtolunc | cool |
| 03:23.10 | meshuga | what software? |
| 03:23.15 | outtolunc | DLX |
| 03:23.21 | meshuga | ah, hehe |
| 03:23.26 | meshuga | wasnt a fan |
| 03:23.28 | outtolunc | multiline chat |
| 03:23.32 | meshuga | i used the telix clone tho :P |
| 03:23.50 | outtolunc | galaticomm was good too had that for awhile |
| 03:23.55 | meshuga | i co-op'd a few major ones |
| 03:23.58 | meshuga | indeed |
| 03:24.11 | meshuga | i still got a ton of mbbs/wg software |
| 03:24.16 | meshuga | my friends gunna put one up |
| 03:24.22 | meshuga | you still own a legit copy? |
| 03:24.37 | meshuga | i lost all mine in a fire :( |
| 03:24.43 | meshuga | and gcomm was sold off already |
| 03:24.52 | meshuga | and the majormud folks told me to go screw myself |
| 03:25.00 | meshuga | cuz i couldnt prove anything |
| 03:25.19 | meshuga | i had up to mod4 too. |
| 03:25.22 | meshuga | and tele-arena |
| 03:25.24 | outtolunc | probably somewhere 'on floppy' <G> |
| 03:27.27 | meshuga | haha i hear ya. i only need the reg code, really. :P |
| 03:27.53 | ChrisHodgetts | quit |
| 03:28.02 | meshuga | if you ever find that, i'd give ya some money/hardware/asterisk help/something cool for it |
| 03:28.03 | meshuga | :P |
| 03:28.05 | newmedian | it takes willpower to quit |
| 03:29.28 | meshuga | my personal favorite was synchronet |
| 03:29.33 | meshuga | it was a late bloomer tho, sadly to say |
| 03:29.35 | *** join/#asterisk pakapole (~pakapole@nusnet-214-56.dynip.nus.edu.sg) |
| 03:29.37 | meshuga | supported like 24 door types |
| 03:29.40 | meshuga | multi line chat |
| 03:29.45 | meshuga | message baord with 3 different formats |
| 03:29.48 | *** part/#asterisk pakapole (~pakapole@nusnet-214-56.dynip.nus.edu.sg) |
| 03:29.55 | *** join/#asterisk pakapole (~pakapole@nusnet-214-56.dynip.nus.edu.sg) |
| 03:31.22 | newmedian | meshuga, outtolunc, you're making me feel old, ya know. |
| 03:31.35 | NewSole2 | lol |
| 03:31.41 | NewSole2 | i know |
| 03:31.54 | meshuga | newmedian: i try. :D |
| 03:32.02 | Juggie | meshuga, everyone knows iniquity and renegade were the best :) |
| 03:32.04 | meshuga | i ran a bbs list in 94 |
| 03:32.04 | m0f0x | Hi... does anyone got problems when compiling asterisk-h323 channel driver on Linux? |
| 03:32.09 | meshuga | hahah dude juggie, iniquity sucked |
| 03:32.16 | ChulJin | ah, BBSes...a 300-baud modem the size of an 8-track plugged into the side of my CoCo |
| 03:32.19 | meshuga | Juggie: i talked to fiend everyday until he disappeared :) |
| 03:32.25 | Juggie | meshuga, iniquity rocked :) |
| 03:32.27 | meshuga | renegade was a horrible hack of WWIV |
| 03:32.29 | meshuga | naw |
| 03:32.37 | meshuga | fiend would be the first to tell you that, too. |
| 03:32.40 | Juggie | oh yah, i partially took over developement of iniquity |
| 03:32.42 | meshuga | and iniquity was oooold. |
| 03:32.44 | meshuga | er neew |
| 03:32.45 | meshuga | i mean |
| 03:32.49 | Juggie | never wrote any code, just did support and such |
| 03:32.51 | meshuga | like, the last bbs software developed |
| 03:32.54 | Juggie | well it was pascal |
| 03:32.57 | Juggie | but it was good i liked it |
| 03:33.02 | outtolunc | aw come on, no one mentioned fossil drivers yet <G> |
| 03:33.04 | meshuga | Juggie : indeed. so was renegade and wwiv. |
| 03:33.13 | Juggie | renegade had alot of hard coded strings |
| 03:33.18 | meshuga | outtolunc : i used synchronet and mbbs, who needs fossils :P |
| 03:33.19 | Juggie | people hex edited renegade to mod |
| 03:33.28 | Juggie | iniquity there was no need |
| 03:33.38 | meshuga | iniquity was pretty, dont get me wrong |
| 03:33.41 | Juggie | plus it had IPL (iniquity programming language) |
| 03:33.45 | meshuga | but functionality lacked extremely. |
| 03:33.46 | Juggie | which was pretty cool |
| 03:33.50 | meshuga | no, IPL was next to worthless |
| 03:34.04 | meshuga | Juggie : you never saw PCL, or baja then |
| 03:34.07 | Juggie | meshugga, it worked for me for 3-4 years i ran a bbs |
| 03:34.12 | Juggie | meshuga, admittedly no... |
| 03:34.30 | meshuga | Juggie : pcl rocked, baja was a 2nd. IPL was like, qbasic or something.. |
| 03:34.38 | meshuga | i miss bbses. |
| 03:34.38 | Juggie | ipl was pascal |
| 03:34.45 | meshuga | no i mean in functionality |
| 03:34.49 | meshuga | not in syntax/structure/etc |
| 03:34.51 | Juggie | complete with =: for assignement |
| 03:34.57 | meshuga | yea i know |
| 03:35.06 | Juggie | well, it just offered you stuff you needed |
| 03:35.06 | meshuga | Juggie : remember the editor bug? |
| 03:35.07 | meshuga | baahahha |
| 03:35.18 | Juggie | meshuga, i probally knew at the time, what was it |
| 03:35.39 | meshuga | Juggie : in 1.25 you could gain access to the admin menu or something |
| 03:35.41 | meshuga | by hitting some key |
| 03:35.45 | *** join/#asterisk t-mobile (~mirc@c-24-91-31-152.hsd1.ma.comcast.net) |
| 03:35.45 | meshuga | i think it was 1.25 |
| 03:35.49 | Juggie | ahh... i dunno |
| 03:35.55 | Juggie | i helped release iniquity 2 |
| 03:35.58 | meshuga | heh |
| 03:36.09 | meshuga | 2 was almost 2 years after fiend left |
| 03:36.15 | meshuga | a year after bbses were dying |
| 03:36.15 | Juggie | yah |
| 03:36.16 | meshuga | like what, 99? |
| 03:36.21 | Juggie | bbsing was pretty much dead |
| 03:36.24 | meshuga | maybe 97 |
| 03:36.25 | Juggie | but we did anyway |
| 03:36.34 | meshuga | Juggie : did you know nivenh? |
| 03:36.38 | meshuga | he took over for awhile i think |
| 03:36.42 | meshuga | who did the coding? |
| 03:36.50 | Juggie | meshuga, someone from demonic or some mod group |
| 03:36.53 | Juggie | did some patches |
| 03:36.57 | meshuga | i hung out with all of them |
| 03:37.04 | meshuga | ah demonic was a joke |
| 03:37.09 | meshuga | fiend made fun of them all the time :P |
| 03:37.14 | Juggie | i forget but some guy did some patches |
| 03:37.17 | meshuga | thats all i really remember of them |
| 03:37.21 | *** join/#asterisk implicit (~implicit@ip68-5-148-1.oc.oc.cox.net) |
| 03:37.21 | Juggie | and he was with some mod group but i dont remember who |
| 03:37.28 | meshuga | good times. |
| 03:37.32 | Juggie | www.iniquitybbs.com |
| 03:37.34 | Juggie | still up :) |
| 03:37.40 | meshuga | www.syncro.net |
| 03:37.42 | meshuga | er |
| 03:37.44 | meshuga | synchro.net |
| 03:37.51 | Juggie | right |
| 03:37.53 | Juggie | comatose |
| 03:37.56 | Juggie | thats who i worked with |
| 03:37.59 | Juggie | i did docs and that shit |
| 03:38.07 | meshuga | hahahha, that guy knew howto code? |
| 03:38.07 | Juggie | i didnt know pascal and didnt care to learn |
| 03:38.08 | meshuga | since when |
| 03:38.19 | Juggie | he didnt know much |
| 03:38.37 | Juggie | but he knew more then me, all we really did was bug fixes, added some minor features |
| 03:39.17 | meshuga | damn no history button |
| 03:39.19 | meshuga | i miss fiend |
| 03:39.22 | hardwire | fieeeeeend |
| 03:39.28 | meshuga | i wonder how nova soctica's treating him these days |
| 03:39.49 | meshuga | Juggie : did you ever use 1.25? his complaints about being a 286 with a CGA were truth, haha |
| 03:40.09 | Juggie | http://www.iniquitybbs.com/idt.html |
| 03:40.12 | Juggie | good letter there. |
| 03:40.22 | meshuga | read it already |
| 03:40.33 | meshuga | years ago |
| 03:41.29 | meshuga | <PROTECTED> |
| 03:41.32 | meshuga | hahah i bet thats fiend. |
| 03:41.35 | meshuga | fiend was nuts like that |
| 03:41.52 | Juggie | i remember iniq had poor fido mail support |
| 03:41.56 | Juggie | it was a hack to get it in threre |
| 03:41.59 | Juggie | but it got better later on |
| 03:42.02 | meshuga | very much so |
| 03:42.08 | meshuga | iniquity was designed for glitz |
| 03:42.11 | meshuga | stuff like tetris |
| 03:42.12 | Juggie | it was so customizable tho |
| 03:42.14 | meshuga | was totally just glitz. |
| 03:42.16 | Juggie | thats why i liked it |
| 03:42.17 | meshuga | not really. |
| 03:42.22 | meshuga | it just looked rad out of the box |
| 03:42.28 | meshuga | and it was alot better then obv/2 |
| 03:42.32 | Juggie | anything and everything the user saw was customizible |
| 03:42.40 | Juggie | yeah, but i didnt run my bbs out of the box |
| 03:42.53 | Juggie | i had a few ansi artists and shit mine was totally customized. |
| 03:42.59 | meshuga | Iniquity is copyright 1994-1995 by Mike Fricker. |
| 03:43.01 | meshuga | heh |
| 03:43.05 | meshuga | its been 10 years? |
| 03:43.09 | Juggie | yeah |
| 03:43.09 | meshuga | goddamn. |
| 03:43.11 | Juggie | amazing eh |
| 03:43.17 | meshuga | apparently i've been on irc for 10 years. |
| 03:43.26 | Juggie | diddo |
| 03:43.30 | Juggie | i ran iniqnet too |
| 03:43.34 | Juggie | i was hub for that |
| 03:43.38 | Juggie | the iniquity fido support network |
| 03:44.11 | meshuga | yea i rememeber iniqnet |
| 03:44.16 | meshuga | never used it |
| 03:44.20 | meshuga | but i do recall |
| 03:44.31 | meshuga | Subject: <q:2.02> - What is the latest version of Iniquity? |
| 03:44.31 | meshuga | Date: 20 Mar 1996 00:00:00 CDT |
| 03:44.31 | meshuga | <PROTECTED> |
| 03:44.31 | meshuga | <PROTECTED> |
| 03:44.33 | meshuga | thats what i mean |
| 03:44.37 | meshuga | 1.00 alpha 25 |
| 03:44.47 | Juggie | yah, there were a few patches after that |
| 03:44.52 | Juggie | for bug fixes |
| 03:44.53 | Juggie | then 2 |
| 03:44.57 | meshuga | i dont think by him |
| 03:44.59 | Juggie | no |
| 03:45.04 | meshuga | he gave up |
| 03:45.06 | Juggie | first they were by some dude in a mod group |
| 03:45.11 | Juggie | then us |
| 03:46.12 | Juggie | http://bbslist.textfiles.com/709/ |
| 03:46.15 | Juggie | i'm on the list |
| 03:47.01 | TheEmperor | anyone know how I can get rid of this? |
| 03:47.04 | TheEmperor | WARNING[2957]: chan_sip.c:694 retrans_pkt: Maximum retries exceeded on call 55efc7fd28d5cb3f0f1886562b185209@203.114.48.140 for seqno 102 (Non-critical Request) |
| 03:47.18 | meshuga | yea dude, youre sip isnt authenticating proper |
| 03:47.37 | hardwire | pcponline man |
| 03:49.01 | Juggie | TheEmperor, it means something is going on with your sip packets not going to the right address |
| 03:49.07 | Juggie | probally do nat=yes? |
| 03:49.17 | TheEmperor | Juggie: thanks, will have a look at that |
| 03:49.25 | meshuga | look at that call line |
| 03:49.30 | meshuga | i got that problem today when it didnt auth |
| 03:49.52 | TheEmperor | all nat=yes in sip.conf for all users.. |
| 03:50.20 | TheEmperor | and reinvite=no |
| 03:50.35 | TheEmperor | also qualify=no |
| 03:53.29 | Juggie | no |
| 03:53.32 | Juggie | qualify=yes |
| 03:53.34 | Juggie | for nat |
| 03:54.05 | TheEmperor | ah... |
| 03:54.08 | TheEmperor | thank you Juggie |
| 03:54.51 | TheEmperor | and i also have jitterbffer=yes |
| 03:56.11 | rabelais | with iax.cc, what would be my reason to not get the 1.55c plan if I intend on using it? I mean, 65 minimum isn't that terribly much, what would be my reason to prepay only $10 and get the 1.7c? |
| 03:57.10 | Juggie | there are cheaper places then iax.cc |
| 03:58.01 | rabelais | Juggie: that include a local did in los angeles? |
| 03:58.07 | Juggie | i believe so |
| 03:58.15 | TheEmperor | when i try to call out, the call never goes through, any ideas? : Executing Answer("SIP/2005-a453", "") in new stack |
| 03:58.15 | TheEmperor | <PROTECTED> |
| 03:58.15 | TheEmperor | <PROTECTED> |
| 03:58.15 | TheEmperor | <PROTECTED> |
| 03:58.23 | Juggie | it was like 15$ for unlimited... and it had good reviews on voip-info |
| 03:59.11 | *** join/#asterisk FuriousGeorge (~brian@ool-43516aa2.dyn.optonline.net) |
| 04:00.12 | Juggie | http://www.voip-info.org/wiki-TelaSIP |
| 04:01.38 | meshuga | aint telasip brand new? |
| 04:02.43 | *** join/#asterisk t-mobile (~mirc@c-24-91-31-152.hsd1.ma.comcast.net) |
| 04:02.50 | Juggie | brand new to resedential |
| 04:02.57 | Juggie | but they have been doing business services for over a year |
| 04:04.08 | meshuga | hrm |
| 04:04.10 | meshuga | interesting |
| 04:04.11 | Juggie | dns created in 2003 |
| 04:05.04 | TheEmperor | can someone tell me what this means? WARNING[3106]: pbx.c:1889 ast_pbx_run: Channel 'Zap/4-1' sent into invalid extension 's' in context 'default', but no invalid handler |
| 04:06.05 | Juggie | well, its looking for |
| 04:06.11 | Juggie | exten => s,1,..... |
| 04:06.12 | Juggie | not finding |
| 04:06.16 | Juggie | then it looks for |
| 04:06.23 | Juggie | exten => i,1.... also not finding |
| 04:07.28 | TheEmperor | i put that under [incoming], would it help if i put it under default? |
| 04:08.18 | Juggie | you clearly have no idea what you are doing :) |
| 04:08.23 | Juggie | www.pastebin.ca |
| 04:08.28 | Juggie | put in your extensions.conf |
| 04:08.32 | Juggie | lets see what your up to |
| 04:08.34 | TheEmperor | ok :) |
| 04:09.56 | TheEmperor | Juggie: got it to work now |
| 04:10.11 | Juggie | ok |
| 04:10.15 | Juggie | each context is seperate |
| 04:10.27 | Juggie | when you receive a call it goes into the incomming context |
| 04:10.27 | TheEmperor | figured that out, thank you :) |
| 04:10.32 | Juggie | as defined by your conf |
| 04:10.38 | TheEmperor | what does this mean? 12:12:16 NOTICE[3648]: chan_sip.c:6638 handle_response: Peer '2002' is now REACHABLE! |
| 04:10.39 | Juggie | it can vary per device and per user |
| 04:10.48 | Juggie | it means that that sip phone was unreachable for a moment |
| 04:10.50 | Juggie | and now its fine |
| 04:10.52 | TheEmperor | i see |
| 04:10.55 | Juggie | if you do sip show peers |
| 04:11.01 | Juggie | it should show you your phones, and the latency |
| 04:11.10 | Juggie | it will prboally say like OK (125ms) |
| 04:11.10 | Juggie | etc |
| 04:11.35 | TheEmperor | how about this? ICE[3648]: chan_sip.c:6644 handle_response: Peer '2002' is now TOO LAGGED! |
| 04:11.40 | TheEmperor | is that a phone setting |
| 04:11.41 | Juggie | well... |
| 04:11.44 | Juggie | thats obvious |
| 04:11.46 | Juggie | where is your phone |
| 04:11.49 | Juggie | on the same network? |
| 04:11.53 | TheEmperor | yes |
| 04:12.07 | Juggie | you shoudnt be getting those messages then |
| 04:12.10 | Juggie | but sometimes it happens |
| 04:12.14 | TheEmperor | yeah.. |
| 04:13.47 | newmedian | You should probably only be worried if you get a message like NOTICE[1337]: cha_sip.c:b34r handle_response: Finish Him! |
| 04:14.10 | Juggie | hah |
| 04:14.15 | FuriousGeorge | is there a defualt button sequence for call x-fer? i notice eyebeam does it out of the box, and i didnt set anything up |
| 04:14.45 | FuriousGeorge | its madness |
| 04:14.54 | newmedian | cats and dogs, living together |
| 04:15.07 | FuriousGeorge | newmedian: what? asterisk and eyebeam? |
| 04:15.11 | Juggie | its called the sip protocol |
| 04:15.14 | Juggie | it includes transfer |
| 04:15.21 | Juggie | fantastic eh |
| 04:15.33 | FuriousGeorge | fantabulous indeed |
| 04:15.43 | FuriousGeorge | but if i wanted to do it from a zap channel |
| 04:15.51 | Juggie | depending on the channel |
| 04:15.54 | Juggie | you would do dif things |
| 04:15.55 | FuriousGeorge | do i gotta manually set that in my dial plan |
| 04:16.11 | FuriousGeorge | it goes out via sip |
| 04:16.16 | Juggie | if you are attaching analog devices i think a bunch of *codes are supported |
| 04:16.19 | Juggie | unless you overwrite them |
| 04:16.35 | Juggie | if you are using a sip ata |
| 04:16.37 | FuriousGeorge | Juggie: thats what i figured, *72 or something |
| 04:16.39 | Juggie | then it depends on what the ata does |
| 04:16.46 | Juggie | all depends on how you hook it up |
| 04:16.56 | FuriousGeorge | im using a tdm400 for my two fxs |
| 04:16.58 | newmedian | For example, I can dial a 9*67w####### to block outbound caller ID |
| 04:17.22 | FuriousGeorge | where can i find a list of supported * commands |
| 04:17.25 | Juggie | is fxs providing service? or receiving |
| 04:17.26 | Juggie | i forget |
| 04:17.28 | FuriousGeorge | or whatever theyre called |
| 04:17.45 | FuriousGeorge | Juggie: it makes a dialtone and talk bat for my phones |
| 04:18.01 | FuriousGeorge | the green one, right? |
| 04:18.06 | Juggie | right so it provides dialtone |
| 04:18.06 | Juggie | ok |
| 04:18.08 | Juggie | look here |
| 04:18.09 | Juggie | http://www.voip-info.org/wiki-Asterisk+zap+channels |
| 04:18.14 | Juggie | its 1/2 way down |
| 04:18.40 | Juggie | or a little more |
| 04:18.42 | Juggie | you'll find it |
| 04:18.47 | Juggie | theres like 10 listed |
| 04:19.10 | TheEmperor | Juggie: why doesn't this call dial out? Executing Answer("SIP/2005-f047", "") in new stack |
| 04:19.11 | TheEmperor | <PROTECTED> |
| 04:19.11 | TheEmperor | <PROTECTED> |
| 04:19.11 | TheEmperor | <PROTECTED> |
| 04:19.16 | TheEmperor | it just hangs there... |
| 04:19.39 | Mavvie | it is dialing out, your logging shows it. |
| 04:19.40 | Juggie | TheEmperor, pastebin.ca your extensions.conf |
| 04:19.56 | *** join/#asterisk t0p (t0p@tech-mgr.chatri.com) |
| 04:20.19 | TheEmperor | k |
| 04:20.20 | Juggie | it is dialing, have you confirmed dialing is working? |
| 04:20.24 | FuriousGeorge | it does not appear zap channels do the sip snaziness |
| 04:20.25 | Juggie | from zap |
| 04:20.40 | Juggie | FuriousGeorge? |
| 04:20.45 | Juggie | sip snazziness? |
| 04:20.47 | FuriousGeorge | or at least, i guess that would work if i had a landline |
| 04:20.55 | t0p | anyone running Asterisk on FC3 here? |
| 04:21.00 | Juggie | me |
| 04:21.10 | budi_ | me |
| 04:21.13 | FuriousGeorge | Juggie: was refering to auto foreward, etc. |
| 04:21.14 | Juggie | FuriousGeorge, what are you trying to do thats not working |
| 04:21.20 | TheEmperor | Juggie:how do i check it if it is dialling? |
| 04:21.23 | FuriousGeorge | the *xx commands |
| 04:21.29 | Juggie | did you see the supported list? |
| 04:21.33 | FuriousGeorge | but i dont have a landline |
| 04:21.34 | Juggie | whats missing? |
| 04:21.38 | FuriousGeorge | ya, looking at it |
| 04:21.40 | Juggie | you dont need a land line |
| 04:21.48 | Juggie | asterisk supports those on a zap channel |
| 04:21.56 | Juggie | when you are providing analog service to phones |
| 04:22.04 | t0p | Juggie, budi_ : can you tell me how i am supposed to start zaptel on FC3? |
| 04:22.32 | Juggie | modprobe zaptel; Sleep(5000); ztcfg -vvvv |
| 04:22.55 | t0p | there's a problem when put 'modprobe zapte' in /etc/rc.modules |
| 04:23.07 | Juggie | yes because of udev |
| 04:23.07 | FuriousGeorge | Juggie: i dunno, i pick up the phone and dial *0 which should send hook flash and i hear a busy tone and asterisk doesnt really say anything except "picked up" "hung up" |
| 04:23.30 | Juggie | it takes a few seconds for the /dev/ whatever to create |
| 04:23.49 | FuriousGeorge | ooooh ooh i know, README.udev in zaptel source |
| 04:23.51 | Juggie | FuriousGeorge, this is on a analog phone connected stright into an digium board irght |
| 04:23.57 | FuriousGeorge | Si |
| 04:24.53 | FuriousGeorge | everything else works (si=yes, in case for some reason you didnt know) |
| 04:24.56 | Juggie | so you pick up and do *69 or something |
| 04:25.02 | Juggie | asterisk shows what |
| 04:25.07 | FuriousGeorge | Juggie: i got that far all by myself |
| 04:25.09 | FuriousGeorge | nothing |
| 04:25.19 | FuriousGeorge | "pciked up zap/1" "hungup" |
| 04:25.28 | *** join/#asterisk JonR800 (jr@pcp05013027pcs.plyntv01.mi.comcast.net) |
| 04:26.07 | Juggie | hrmmm |
| 04:26.09 | Juggie | does dialing work |
| 04:26.11 | Juggie | can u dial a number? |
| 04:26.29 | FuriousGeorge | it also says "starting a simple switch" between pickup and hangup. dialing worked a second ago, lemme check again |
| 04:26.44 | FuriousGeorge | still works |
| 04:26.55 | Juggie | hrmmm |
| 04:27.00 | Juggie | i've never used this functionality |
| 04:27.05 | FuriousGeorge | i called an 888 which fwded me to a phone in switzerland, i was amazed |
| 04:27.32 | Juggie | but it seems like its all enabled by default |
| 04:27.34 | *** join/#asterisk Cresl1n (~matt@216.207.245.23) |
| 04:27.41 | FuriousGeorge | Juggie: I think maybe SIPPhone.com (my provider) just doesnt support that stuff |
| 04:27.48 | Juggie | no |
| 04:27.49 | Juggie | i know |
| 04:27.54 | Juggie | your dialplan is conflicting |
| 04:27.55 | FuriousGeorge | but then again, the same thing happens when i do ** |
| 04:28.05 | Juggie | pastebin.ca your extensions.conf |
| 04:28.05 | FuriousGeorge | which should work, so it may not be that |
| 04:28.11 | Juggie | you have overlap |
| 04:28.12 | Juggie | thats why |
| 04:28.14 | Juggie | lemme see |
| 04:28.20 | FuriousGeorge | one sec |
| 04:29.35 | TheEmperor | Juggie: http://pastebin.ca/11566 |
| 04:29.55 | FuriousGeorge | http://pastebin.ca/11567 |
| 04:29.57 | Juggie | ahh |
| 04:29.57 | Juggie | well for one |
| 04:29.58 | Juggie | bad bad |
| 04:30.32 | Juggie | you shoudnt include everything into everything like that |
| 04:31.30 | TheEmperor | talking to me? :) |
| 04:31.50 | Juggie | yes |
| 04:31.56 | TheEmperor | what did I do wrong? |
| 04:31.56 | Juggie | you have everything included everywhere |
| 04:32.00 | Juggie | not good |
| 04:32.04 | TheEmperor | oh.. |
| 04:32.09 | Juggie | lets logically look at whats going on |
| 04:32.14 | TheEmperor | ok |
| 04:32.15 | Juggie | you have to think in terms of your inputs and outputs to the system |
| 04:32.48 | Juggie | do you have pstn service from a phone company? |
| 04:32.51 | TheEmperor | yes |
| 04:33.03 | TheEmperor | that is why i have the [local-out] |
| 04:33.09 | Juggie | ok, so you get pstn calls in |
| 04:33.12 | Juggie | you get sip calls in |
| 04:33.17 | TheEmperor | yes and we use those same calls to call out as well |
| 04:33.18 | TheEmperor | yes |
| 04:33.20 | Juggie | you send pstn calls out |
| 04:33.25 | Juggie | and you send sip calls out |
| 04:33.27 | TheEmperor | as well, yes |
| 04:33.37 | TheEmperor | we send pstn calls out using sip phones |
| 04:33.47 | TheEmperor | going through the digium 4 port fxo card |
| 04:33.52 | Juggie | you dont make any local calls through the pstn? |
| 04:34.05 | TheEmperor | all pstn lines are connected to the * box |
| 04:34.09 | Juggie | right |
| 04:34.11 | TheEmperor | no analog phones |
| 04:34.17 | Juggie | ohhhhhh |
| 04:34.17 | Juggie | ok |
| 04:34.21 | Juggie | i misunderstood your setup |
| 04:34.29 | Juggie | you told me you were providing dialtone to some phones |
| 04:34.33 | TheEmperor | so the sip phones would make local calls using the pstn lines through the * box |
| 04:34.38 | TheEmperor | no.. |
| 04:34.40 | Juggie | ok, only voip phones |
| 04:34.42 | TheEmperor | yes |
| 04:34.47 | Juggie | ok well thats why *?? dont work |
| 04:34.51 | Juggie | but your dialplan is still ass |
| 04:34.57 | Juggie | i have a few spare moments to lets go over it |
| 04:35.00 | TheEmperor | i see.. |
| 04:35.01 | TheEmperor | ok :) |
| 04:35.05 | Juggie | you have local incomming |
| 04:35.10 | Juggie | and sip incomming |
| 04:35.12 | Juggie | and sip outgoing |
| 04:35.18 | Juggie | you send no calls OUT over PSTN? |
| 04:35.32 | TheEmperor | no, we use the sip phones to call out over pstn |
| 04:35.39 | TheEmperor | [local-out] |
| 04:35.47 | Juggie | i think your misunderstanding my question |
| 04:35.53 | TheEmperor | dialling 9 as a prefix |
| 04:35.58 | Juggie | you have telephony service from two places |
| 04:36.03 | Juggie | sip, and phone company |
| 04:36.08 | TheEmperor | correct |
| 04:36.10 | Juggie | you are using sip phones |
| 04:36.12 | Juggie | ignore that fact |
| 04:36.13 | TheEmperor | yes |
| 04:36.16 | Juggie | forget that |
| 04:36.19 | TheEmperor | k |
| 04:36.20 | Juggie | when you make out bound calls |
| 04:36.27 | Juggie | do you wish to use ONLY sip for out bound |
| 04:36.33 | TheEmperor | yes |
| 04:36.41 | Juggie | so pstn is ONLY used for incomming calls |
| 04:36.47 | Juggie | and perhaps in an emergency situation |
| 04:36.50 | TheEmperor | erm |
| 04:37.00 | TheEmperor | we still want to use pstn for outgoing calls |
| 04:37.04 | Juggie | ok |
| 04:37.08 | TheEmperor | but using sip phones |
| 04:37.16 | Juggie | forget about your sip phones |
| 04:37.25 | Juggie | think in terms of service to the pbx |
| 04:37.28 | newmedian | There is a certain pythonequeness to this conversation. |
| 04:37.34 | Juggie | indeed. |
| 04:37.45 | TheEmperor | then yes, we do want to use pstn for outgoing calls |
| 04:37.47 | Juggie | ok, well through relentless questioning, i know some information finally :) |
| 04:38.01 | Juggie | i understand there is likely a language barrier so its ok. |
| 04:38.14 | newmedian | :) |
| 04:38.26 | Juggie | ok, so you need local-in/local-out/sip-in/sip-out |
| 04:38.26 | TheEmperor | :D |
| 04:38.29 | Juggie | definitally |
| 04:38.31 | *** join/#asterisk t0pCop (t0p@tech-mgr.chatri.com) |
| 04:38.33 | TheEmperor | yes |
| 04:38.37 | HeppyCat | sure is interesting that my phone gives a 404 when i try to make a call |
| 04:38.54 | Juggie | stop using default... |
| 04:39.06 | Juggie | let me open notepad here i'll start doing something up |
| 04:39.13 | TheEmperor | so i should have a local in context? |
| 04:39.13 | TheEmperor | ok |
| 04:39.59 | Juggie | yes |
| 04:40.05 | Juggie | because you need local dialing rules |
| 04:40.45 | TheEmperor | i see |
| 04:41.14 | Juggie | see, its all over the place... what you do is assign incomming calls to feed into local-in |
| 04:41.24 | Juggie | incomming sip calls feed into sip-in |
| 04:41.25 | TheEmperor | so something like [default] and then local in in there? |
| 04:41.32 | Juggie | default is no more |
| 04:41.35 | Juggie | there is no default |
| 04:41.43 | TheEmperor | i see |
| 04:41.51 | Juggie | let me mock something up here |
| 04:41.54 | TheEmperor | ok |
| 04:41.57 | TheEmperor | :) |
| 04:42.31 | FuriousGeorge | Juggie: i was thinking about just that the other day, what if i want the incomming calls to be treated the same regardless of whether or not they are sip |
| 04:42.58 | Juggie | its possible that you might |
| 04:43.08 | Juggie | but you cant depend on sip caller id so much as you can from the pstn |
| 04:43.26 | Juggie | i guess you could :) |
| 04:43.33 | Sedorox | holy shit.... |
| 04:43.35 | FuriousGeorge | makes sense, but pardon my ignorance, why does it matter about the caller id |
| 04:43.48 | Sedorox | my did I ordered back in Feb from Link2voip STILL IS NOT setup... |
| 04:44.14 | FuriousGeorge | i mean if i run a business for instance i want everyone to hear: Welcome To Spacely's Sprockets |
| 04:44.17 | Juggie | its just organization, callerid, well you could sort calls per 1-800 calls, long distance, local calls etc.. i dunno... let me do this guys dialplan :) |
| 04:44.26 | FuriousGeorge | and all the rest when they call in |
| 04:44.36 | FuriousGeorge | sorry proceed |
| 04:44.45 | Juggie | you could argue either way yes |
| 04:44.53 | Juggie | if you wish to treat both inputs as the same |
| 04:45.16 | Juggie | but sometimes the pstn needs Wait() commands to get caller id for example |
| 04:45.19 | Juggie | something that sip wont need |
| 04:45.33 | FuriousGeorge | i get it, using caller id to determine where the call goes! lol, im new to this |
| 04:45.33 | *** join/#asterisk scardinal (~supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 04:45.55 | FuriousGeorge | i see |
| 04:46.04 | FuriousGeorge | makes perfect sense |
| 04:46.12 | Juggie | TheEmperor, why do you have extensions that give you direct dialtone on a card |
| 04:46.16 | Juggie | not a great idea :) |
| 04:46.24 | Juggie | FuriousGeorge, its called the ex-girlfriend logic |
| 04:46.27 | Juggie | its on the wiki, check it out |
| 04:46.27 | TheEmperor | DISA? |
| 04:46.29 | TheEmperor | no good? |
| 04:46.34 | Juggie | its not disa |
| 04:46.42 | Juggie | you are giving people dialtone on the local pstn line |
| 04:46.46 | Juggie | they can now do anything |
| 04:46.49 | Juggie | and you have no record of it |
| 04:46.57 | TheEmperor | which part is that? |
| 04:47.02 | Juggie | the part with 998 999 |
| 04:47.12 | Juggie | in local-out |
| 04:47.12 | TheEmperor | oh ok, i can erase that :) |
| 04:47.15 | Juggie | yes |
| 04:47.20 | Juggie | for your safety, please do :) |
| 04:47.24 | TheEmperor | ok :) |
| 04:47.27 | *** join/#asterisk kb1_kanobe (~krisbouti@h24-207-80-55.cst.dccnet.com) |
| 04:47.31 | kb1_kanobe | evening all. |
| 04:47.45 | TheEmperor | so i should delete default? |
| 04:47.53 | TheEmperor | [default] i mean |
| 04:48.07 | Juggie | let me show you what i'm going to do |
| 04:48.11 | TheEmperor | ok.. |
| 04:48.20 | Juggie | TheEmperor, how many local lines do you have? |
| 04:48.33 | TheEmperor | pstn? |
| 04:49.29 | Juggie | yes |
| 04:49.31 | TheEmperor | 4 |
| 04:49.32 | Juggie | pstn=local lines |
| 04:49.55 | Juggie | ok, and how about sip lines |
| 04:50.03 | Juggie | how many numbers did you get over sip |
| 04:50.04 | TheEmperor | 5 |
| 04:50.14 | Juggie | do you want to answer the same way for all lines? |
| 04:50.20 | TheEmperor | yes |
| 04:50.25 | Juggie | ok |
| 04:50.46 | Juggie | why are you doing an exten=> t in incomming |
| 04:50.51 | Juggie | there should be no timeout here |
| 04:51.04 | TheEmperor | if there was a menu there, i should delete that one actually.. |
| 04:51.13 | Juggie | also, why are you dialing all phones |
| 04:51.16 | Juggie | that was just testing? |
| 04:51.22 | TheEmperor | yes |
| 04:51.26 | Juggie | ok |
| 04:51.34 | iq|laptop | Hi, can I use my own sip stack with asterisk? Will it be too hard to do? I already have a working sip stack. |
| 04:51.34 | TheEmperor | also, so that if other people are not around, someone will pick up the cal |
| 04:51.52 | Juggie | so you intend to write an ivr which asks for an extension and such? |
| 04:52.08 | TheEmperor | at this point no.. |
| 04:52.25 | Juggie | so then when any of the 9 numbers are dialed you want to ring every phone? |
| 04:52.32 | TheEmperor | now just internal calls and also using pstn to call out |
| 04:52.41 | TheEmperor | no.. |
| 04:53.04 | *** join/#asterisk MajestiK (~MajestiK@S01060800208687ec.ed.shawcable.net) |
| 04:53.25 | Juggie | so when someone calls this company, what is supposed to happen |
| 04:53.39 | Juggie | or is this just for outgoing calls |
| 04:53.49 | TheEmperor | a few phones ring and then whoever picks up the calls gets that call |
| 04:54.16 | Juggie | ok |
| 04:54.31 | Juggie | why does 0 go to Zap/1 |
| 04:54.34 | TheEmperor | and everyone can call everyone internally |
| 04:54.42 | Juggie | i thought all the zap lines were pstn lines |
| 04:54.49 | TheEmperor | yes |
| 04:54.56 | TheEmperor | so it should then be s,1,Answer |
| 04:54.59 | TheEmperor | no () |
| 04:55.00 | TheEmperor | ? |
| 04:56.27 | Juggie | no, let me show you |
| 04:56.46 | TheEmperor | ok.. |
| 04:57.25 | Juggie | luckly this is cut n paste |
| 04:57.35 | Juggie | like this macro |
| 04:57.45 | Juggie | which country are you? |
| 04:57.55 | TheEmperor | Malaysia |
| 04:58.04 | Juggie | what format are your phone numbers in? |
| 04:58.24 | TheEmperor | 8 digit numbers |
| 04:58.34 | TheEmperor | 10 digit for mobiles |
| 04:59.42 | Juggie | do they start a certain way |
| 04:59.48 | Juggie | so you have any idea if you can expect 8 or 10 digits? |
| 04:59.53 | *** join/#asterisk Synapse-_ (~pnats@c211-30-74-249.belrs2.nsw.optusnet.com.au) |
| 05:00.10 | TheEmperor | if you dial out on pstn, dialling a land number starts from 3 to 9 |
| 05:00.12 | budi_ | /SIP/Registry//budi : 192.168.1.162:5060:120:budi:sip:budi@192.168.1.162:5060 |
| 05:00.19 | TheEmperor | dialling a mobile starts with 01 |
| 05:00.20 | budi_ | anyone know how to delete that entry from database? |
| 05:00.23 | Juggie | ahh |
| 05:00.24 | budi_ | from cli? |
| 05:00.24 | Juggie | great |
| 05:06.28 | *** join/#asterisk HuangDi (~user@203.114.48.47) |
| 05:06.39 | HuangDi | Juggie: which macro is that? |
| 05:06.47 | *** join/#asterisk iswm (iswm@iswm.user) |
| 05:07.13 | Juggie | HuangDi, just one for creating sip extensions |
| 05:07.21 | Juggie | TheEmperor, does your sip dialing work |
| 05:07.41 | HuangDi | Juggie: I can't seem to log back in as TheEmperor, so I logged in as HuangDi |
| 05:07.47 | HuangDi | ;) |
| 05:08.04 | HuangDi | I missed out what you posted just now.. |
| 05:09.57 | Juggie | http://pastebin.ca/11569 |
| 05:10.00 | Juggie | check it out |
| 05:10.02 | Juggie | see if it makes more sense |
| 05:10.04 | HuangDi | thanks :) |
| 05:10.08 | Juggie | i added notes to the top for how to set it up |
| 05:10.19 | Juggie | i dont guarantee it will work without any changes but its alot easier to see whats going on |
| 05:11.24 | HuangDi | i see |
| 05:11.26 | Juggie | there could be some hangups added in to clean stuff up |
| 05:11.30 | Juggie | make much sense? |
| 05:12.19 | HuangDi | yes, it looks clearer |
| 05:12.19 | HuangDi | yes |
| 05:12.19 | HuangDi | :) |
| 05:12.19 | HuangDi | thank you |
| 05:12.19 | Juggie | woops |
| 05:12.19 | Juggie | in [internals] |
| 05:12.19 | Juggie | when i cut and pasted the line |
| 05:12.19 | Juggie | and added everyones extensions |
| 05:12.19 | Juggie | i forgot to change the number |
| 05:12.19 | Juggie | see how they are all 102? |
| 05:12.25 | HuangDi | yeah |
| 05:12.33 | Juggie | do you undestand how the macro works? |
| 05:12.54 | HuangDi | yes |
| 05:12.57 | HuangDi | what is ARG1? |
| 05:13.12 | Juggie | macro is a stackable function |
| 05:13.21 | Juggie | you can do macro(sip,something,somethingelse,etc) |
| 05:13.24 | Juggie | and they get put in the macro as |
| 05:13.37 | Juggie | ${ARG1} ${ARG2} and so on |
| 05:13.47 | Juggie | in this case i am just passing their sip phone nma |
| 05:13.50 | Juggie | *name |
| 05:13.55 | Juggie | which is arg1 |
| 05:14.00 | HuangDi | i see |
| 05:14.07 | HuangDi | let me put all this in and see how it works :) |
| 05:14.30 | budi_ | hi anyone know how to set up call forward on busy and direct diversion on SIP ? |
| 05:14.45 | Juggie | when you do |
| 05:14.47 | Juggie | exten => s,2,Dial(SIP/2001&SIP/2002&SIP/2003&SIP/2004$SIP/2005|120) |
| 05:14.56 | Juggie | you should use the extensions you created in [internals] |
| 05:15.01 | Juggie | 101,102,103 etc |
| 05:15.18 | Juggie | if you want to do that, add an include => internals into your sip-in local-in |
| 05:15.23 | HuangDi | zapata.conf, is this correct? autodetect=on |
| 05:15.23 | HuangDi | context=local-in |
| 05:15.23 | HuangDi | signalling=fxs_ks |
| 05:15.24 | HuangDi | channel => 1-4 |
| 05:15.32 | Juggie | looks good |
| 05:15.51 | HuangDi | sweet |
| 05:15.59 | Juggie | you have to make two changes in sip.conf |
| 05:16.03 | Juggie | one for sipphone.com |
| 05:16.11 | Juggie | and one for all your sip phones around the office |
| 05:17.38 | HuangDi | sipphone.com |
| 05:17.39 | HuangDi | ? |
| 05:17.45 | HuangDi | i thought only sip.conf? |
| 05:17.51 | Juggie | yes |
| 05:17.57 | Juggie | but your sip provider is sipphone.com |
| 05:18.02 | Juggie | according to the example you gave me |
| 05:18.16 | Juggie | they should have a user in sip.conf if you have it setup properly |
| 05:18.23 | Juggie | so that you can receive calls via sip |
| 05:18.43 | HuangDi | erm, i define the users in sip.conf, i don't need sippphone.com i think.. |
| 05:18.44 | Juggie | how this user is setup will control what context sip calls get sent to when they are received |
| 05:18.54 | Juggie | if you dont put it there, you wont get calls over sip |
| 05:18.57 | Juggie | only send them |
| 05:19.00 | *** join/#asterisk CleanerX (~nix@nat-ph3-wh.rz.uni-karlsruhe.de) |
| 05:19.20 | HuangDi | ok |
| 05:20.03 | Juggie | i guess you never tried that |
| 05:20.42 | *** join/#asterisk bkw_ (~brian@bkw.developer.and.friend.of.asterisk) |
| 05:20.42 | *** mode/#asterisk [+o bkw_] by ChanServ |
| 05:21.20 | HuangDi | no..i don't really understand that part.. |
| 05:21.34 | HuangDi | what do you mean? |
| 05:21.48 | Juggie | you have sip service from a sip provider |
| 05:22.01 | Juggie | when they receiver a call on a number they provided for you |
| 05:22.08 | Juggie | they send it to your server, via sip |
| 05:22.24 | HuangDi | oh |
| 05:22.28 | Juggie | to receive the call, they need to have an account in your sip.conf so that your pbx knows who they are |
| 05:22.30 | HuangDi | yes, i understand now :) |
| 05:22.45 | HuangDi | no, no need for that yet :) |
| 05:22.51 | Juggie | ok then forget sip-in for now |
| 05:23.00 | Juggie | leave it in the extensions but it wont be used |
| 05:23.06 | Juggie | regardless, you need to configure your sip phones |
| 05:23.08 | Juggie | in sip.conf |
| 05:23.13 | Juggie | such that the context for your phones |
| 05:23.15 | Juggie | is sip-phone |
| 05:23.54 | HuangDi | ok |
| 05:24.22 | Juggie | you made me miss the simpsons |
| 05:24.38 | Juggie | ohh |
| 05:24.40 | Juggie | in sip-out |
| 05:24.50 | Juggie | change ${EXTEN} |
| 05:24.51 | Juggie | to |
| 05:24.57 | Juggie | ${EXTEN:1} |
| 05:25.01 | Juggie | as i added an 8 |
| 05:25.04 | Juggie | for dialing out via sip |
| 05:25.09 | *** join/#asterisk bkw_ (~bkw_@bkw.developer.and.friend.of.asterisk) |
| 05:25.09 | *** mode/#asterisk [+o bkw_] by ChanServ |
| 05:25.11 | Juggie | but it has to trim off the 8 |
| 05:26.58 | HuangDi | for Macro, does sip have to be SIP or it doesn't matter? |
| 05:28.19 | Juggie | well, it can be |
| 05:28.24 | Juggie | [macro-????] |
| 05:28.28 | Juggie | and when you call it you do |
| 05:28.34 | Juggie | Macro(????,variables.....) |
| 05:28.39 | Juggie | so whatever you want |
| 05:28.53 | HuangDi | i see |
| 05:30.56 | t0p | Juggie: so, have you ever used X100P on *+FC3? |
| 05:31.49 | Juggie | no |
| 05:31.55 | Juggie | just a 405P |
| 05:32.57 | HuangDi | Juggie: can I do a macro-iax or iax extensions? |
| 05:33.29 | Juggie | you can yes |
| 05:33.34 | Juggie | if u want to do iax that way |
| 05:34.18 | *** join/#asterisk PTG1234 (~PTG123@ip68-106-24-139.ph.ph.cox.net) |
| 05:40.12 | *** join/#asterisk shidan (~shidan@CPE000625dbadc2-CM014280007905.cpe.net.cable.rogers.com) |
| 05:40.47 | shidan | hey blitz whats happenin |
| 05:42.55 | *** join/#asterisk brettnem (~brettnem@user-0ccsr10.cable.mindspring.com) |
| 05:49.09 | HuangDi | Juggie: VoiceMailMain is this correct? |
| 05:57.08 | HuangDi | Juggie: i still can't make outgoing calls from the pstn.. |
| 05:59.27 | *** join/#asterisk ClayReiche123 (~creiche@73-117.35-65.tampabay.res.rr.com) |
| 06:00.26 | *** join/#asterisk showtime031 (~badwolf@ool-18b94f16.dyn.optonline.net) |
| 06:00.38 | HuangDi | anyone around still :) |
| 06:00.43 | showtime031 | hi all |
| 06:01.18 | showtime031 | hi huangdi |
| 06:01.28 | HuangDi | hello showtime031 |
| 06:02.13 | showtime031 | are you running asterisk yet? |
| 06:02.59 | HuangDi | i'm having trouble calling out on my pstn |
| 06:03.17 | HuangDi | do you have any idea on zapata.conf configuration? i think i did something wrong there |
| 06:04.18 | showtime031 | sorry i am new to this asterisk stuff |
| 06:04.38 | ClayReiche123 | Can someone tell me why I'm getting "From: "asterisk" <sip:asterisk@65.45.14.7>" in my INVITE header? |
| 06:04.42 | HuangDi | ok :) |
| 06:07.26 | showtime031 | i want to be able to offer voip service, is asterisk the right solution? |
| 06:07.51 | ClayReiche123 | can be.... |
| 06:08.40 | ClayReiche123 | How many customers do you want? |
| 06:10.18 | showtime031 | well it will grow every week to about 5 to 10 customers a week but really speaking i have about 1000 or more customers |
| 06:10.31 | t0p | anyone knows what this error means "app_dial.c:968 dial_exec_full: Unable to create channel of type 'Zap' (cause 0)" |
| 06:11.00 | *** join/#asterisk pakapole (~pakapole@nusnet-214-56.dynip.nus.edu.sg) |
| 06:11.27 | t0p | I just finished setting up my * with 1 X100P |
| 06:11.44 | ClayReiche123 | I'm using it right now for voip and have hit some challenges. We are getting ready to roll out service to 1000+ customers and I'm having doubts that it will handle it. |
| 06:12.01 | t0p | Zaptel seems to be installed correctly |
| 06:12.25 | ClayReiche123 | I have about 100 testers on it right now and getting some stability issues. |
| 06:12.36 | t0p | I'm using only simple settings |
| 06:12.50 | ClayReiche123 | I'm starting to think that SER is the way to go, but I have no experience with it. |
| 06:15.12 | ClayReiche123 | Can someone tell me why I'm getting "From: "asterisk" <sip:asterisk@65.45.14.7>" in my INVITE header? |
| 06:16.15 | Juggie | ClayReiche123, ser would only help if it was doing load balencing across more then one asterisk server |
| 06:16.16 | RaYmAn-Bx | I'm not sure what exactly you're confused about, but if it's the callerid, then it's because no callerid was available on the asterisk sending that message |
| 06:16.38 | Juggie | anyways b ack to simpsons, sleep |
| 06:17.26 | RaYmAn-Bx | presumably SER would also help if a lot of the calls are IP-only |
| 06:18.07 | RaYmAn-Bx | isn't that was most bigger asterisk-using providers do? Use SER on client side and let that handle everything except calls to PSTN |
| 06:18.21 | Juggie | yes |
| 06:18.30 | Juggie | that may help for free world dialup |
| 06:18.42 | Juggie | but for a provider whos business is connecting users to the pstn |
| 06:18.50 | Juggie | ser isnt going to help with media path |
| 06:19.09 | Juggie | it will however help be a registration server, as well as load balance across outbound asterisk servers |
| 06:19.19 | HuangDi | Juggie: everything is working now, except the pstn calling out, but when i change it to Zap/4 it works |
| 06:19.39 | HuangDi | how do i configure it so that * makes the call out on whichever port is available? |
| 06:19.39 | Juggie | huang, your group isnt setup proper |
| 06:19.48 | HuangDi | zap/g1 doesn't work.. |
| 06:19.50 | HuangDi | oh.. |
| 06:19.50 | Juggie | paste me that part of your zapata.conf again |
| 06:20.07 | RaYmAn-Bx | the provider I primarily use do quite well with one SER and one asterisk for pstngw..with around 8000 users (although, I have no idea about simultaneous calls...Haven't ever tried not being able to call though) |
| 06:20.17 | HuangDi | autodetect=on |
| 06:20.17 | HuangDi | context=local-out |
| 06:20.17 | HuangDi | signalling=fxs_ks |
| 06:20.17 | HuangDi | channel => 1-4 |
| 06:20.25 | *** join/#asterisk gres (~serg@81.222.48.242) |
| 06:21.15 | Juggie | before channel => 1-4 |
| 06:21.16 | Juggie | add |
| 06:21.18 | Juggie | group=1 |
| 06:21.22 | RaYmAn-Bx | actually nevermind, they might actually be receiving PSTN stuff over IP anyways. |
| 06:21.51 | HuangDi | autodetect=on |
| 06:21.52 | HuangDi | context=local-out |
| 06:21.52 | HuangDi | signalling=fxs_ks |
| 06:21.52 | HuangDi | group=1 |
| 06:21.52 | HuangDi | channel => 1-4 |
| 06:22.01 | Juggie | yes that should do it |
| 06:22.07 | Juggie | restart and try |
| 06:22.19 | HuangDi | also, on the top, is this correct? |
| 06:22.27 | HuangDi | ; Default context |
| 06:22.27 | HuangDi | ; |
| 06:22.27 | HuangDi | context=local-in |
| 06:22.47 | Juggie | yes that seems fine just try it |
| 06:22.49 | Juggie | i'm tired :) |
| 06:22.58 | HuangDi | ok :) |
| 06:23.02 | HuangDi | thanks man!!! |
| 06:23.27 | *** join/#asterisk K9DI_BSD_WrkStn (~k9bsd@207-246-185-168.EastVillage.ResNet.wiu.edu) |
| 06:24.20 | Juggie | did it work |
| 06:26.54 | Juggie | hhhhhmmm |
| 06:29.17 | ClayReiche123 | Juggie: I'm having some issues with asterisk that I suspect SER might handle better. Like codec negotiating... I wish (oh how I wish...) that asterisk would "learn" or pass through my endpoint codec list.... |
| 06:29.51 | pakapole | hi! how do i post a question? |
| 06:29.53 | Juggie | ClayReiche123, alot of the codec stuff got fixed in head |
| 06:30.07 | Juggie | some of it may have been backported to cvs-stable |
| 06:30.12 | Juggie | talk to drumkilla |
| 06:30.27 | ClayReiche123 | Juggie: also, I posted this question a minute ago with no response. I suspect SER might work for this too... Can someone tell me why I'm getting "From: "asterisk" <sip:asterisk@65.45.14.7>" in my INVITE header? |
| 06:30.46 | Juggie | because reinvite=no? |
| 06:31.14 | RaYmAn-Bx | ClayReiche123: which part do you not expect? |
| 06:31.33 | RaYmAn-Bx | the "asterisk" name and CID? Or something else? |
| 06:31.38 | Juggie | looks to me like asterisk is stayinig in the media path |
| 06:31.43 | ClayReiche123 | Juggie: 'From: "asterisk"' |
| 06:31.44 | pakapole | can someone help me with Realtime and extensions? |
| 06:31.47 | Juggie | which if you are going voip<-> pstn |
| 06:31.53 | RaYmAn-Bx | afaik it also does that when no callerid is available |
| 06:32.06 | ClayReiche123 | I'm having trouble with re-invites.... |
| 06:32.21 | Juggie | ClayReiche123, why are reinvites even a concern |
| 06:32.30 | Juggie | all your calls are sip<->pstn |
| 06:32.48 | ClayReiche123 | juggie: I'm getting 1 way audio when re-invites=yes. |
| 06:33.10 | Juggie | do you have your sip clients set for nat=yes? |
| 06:33.19 | tainted- | anyone good with AGI? |
| 06:33.36 | tainted- | i need asterisk to playback a string like "FOO123" |
| 06:33.36 | ClayReiche123 | Juggie:The ta manufacturer says that reinvites aren't working properly with asterisk. |
| 06:33.47 | tainted- | and say "F-O-O-1-2-3" |
| 06:33.48 | ClayReiche123 | Juggie: nat=yes yes |
| 06:34.11 | Juggie | ClayReiche123, then post a bug report |
| 06:34.21 | Juggie | bugs.digium.com include a sip debug |
| 06:34.28 | ClayReiche123 | Juggie: I'm not sure I believe him |
| 06:34.55 | Juggie | reinvite as i undestand is for changing the media path |
| 06:35.04 | Juggie | such that the rtp can go from phone to phone |
| 06:35.09 | Juggie | rather then through asterisk |
| 06:35.29 | Juggie | but since you are going out allways does it matter? |
| 06:35.33 | ClayReiche123 | Juggie: He says he sees 4 invites... the 3rd one is correct and all would be well if it stopped there, but the 4th one hoses everything... |
| 06:36.05 | Juggie | something tells me i've seen this on bugs.digium before |
| 06:36.09 | Juggie | did you post something |
| 06:36.48 | ClayReiche123 | Juggie: no... but I will investigate further. I hesitate to post a bug report when I don't fully understand. |
| 06:37.35 | Juggie | with canreinvite=no it works? |
| 06:38.05 | RaYmAn-Bx | tainted-: I'm haven't exactly used AGI, but can't you do the same as you would do from the dialplan? i.e. use SayAlpha and similar |
| 06:38.35 | ClayReiche123 | yes, the call works perfect. But apparently I get the 'From: "asterisk"' in the from header. |
| 06:39.08 | Juggie | thats because asterisk is passing the media |
| 06:39.11 | Juggie | the rtp |
| 06:39.20 | Juggie | so it is, from asterisk |
| 06:39.28 | ClayReiche123 | I would love to stop doing that.... |
| 06:39.54 | Juggie | what does it affect |
| 06:40.26 | ClayReiche123 | does it not put load on the server? |
| 06:40.45 | Juggie | but your calls are not sip to sip |
| 06:40.58 | Juggie | arnt you selling did's going sip->pstn |
| 06:41.15 | ClayReiche123 | yes they are. Going to a sip gateway provider. |
| 06:41.51 | ClayReiche123 | I have no digium hardware in these machines |
| 06:41.51 | Juggie | so your phones are on a private network |
| 06:42.08 | Juggie | asterisk is on private as well? or does it have two network cards |
| 06:42.15 | ClayReiche123 | all over the place. We are attempting a carrier scenario. |
| 06:42.30 | ClayReiche123 | asterisk is public |
| 06:42.41 | Juggie | it has a public & private ip? |
| 06:42.49 | Juggie | or just public |
| 06:42.50 | ClayReiche123 | no just public |
| 06:42.58 | Juggie | and phones are mostly private then |
| 06:43.08 | Juggie | behind nat |
| 06:43.15 | ClayReiche123 | most of our customers are at home behind a broadband router. |
| 06:43.19 | ClayReiche123 | yes |
| 06:43.19 | Juggie | right |
| 06:43.30 | Juggie | well.... |
| 06:43.48 | Juggie | when there was one way audio |
| 06:43.50 | Juggie | which way did it work |
| 06:43.52 | ClayReiche123 | and it would be nice to be able to down the server without dropping calls as well... |
| 06:44.17 | RaYmAn-Bx | I would suspect we're back to SER with asterisk providing extra services...(like voicemail etc).. |
| 06:44.27 | Juggie | which way did did the audio work |
| 06:44.41 | ClayReiche123 | voip phone could not hear pstn |
| 06:44.58 | Juggie | not suprising |
| 06:44.59 | RaYmAn-Bx | that's a typical NAT issue |
| 06:45.06 | Juggie | heres the scenario |
| 06:45.18 | Juggie | your phone has an ip of whatever on the private network |
| 06:45.25 | Juggie | if you say nat=yes in asterisk |
| 06:45.31 | Juggie | it ignores the from: ip |
| 06:45.50 | Juggie | and sends udp back to the address the packet came from |
| 06:46.07 | Juggie | i suspect your provider was not aware of that... |
| 06:46.19 | Juggie | some devices may work |
| 06:46.28 | Juggie | those that are smart enough to know their internet ip |
| 06:46.50 | Juggie | those that report their local lan ip will fail unless * is in the path |
| 06:46.52 | ClayReiche123 | I have some stun settings in my device, but that didn't seem to help. |
| 06:47.04 | Juggie | its not a good solution |
| 06:47.10 | Juggie | you cant count on the user device |
| 06:47.56 | Juggie | not all devices have stun etc.... but regardless i suspect your voip provider didnt know the proper address to send packets |
| 06:48.01 | ClayReiche123 | The provider claimed that asterisk was sending the private ip in the re-invite... |
| 06:48.11 | ClayReiche123 | ...asterisk, or the ta device.... |
| 06:48.27 | Juggie | asterisk would only send the private ip if the device provided it |
| 06:48.31 | ClayReiche123 | everybody was pointing fingers at somebody else....it has been a nightmare. |
| 06:48.46 | Juggie | well you can see whats going on by doing a sip debug |
| 06:48.59 | Juggie | if the device provides a local lan ip |
| 06:49.02 | Juggie | asterisk can deal |
| 06:49.16 | ClayReiche123 | I've suspected the provider all along. I can point traffic out of our Cisco 5350 and re-invites work all day long.... |
| 06:49.38 | Juggie | it may not be the provider.... look at the sip debug |
| 06:49.45 | Juggie | watch what asterisk tells them is the from: |
| 06:49.50 | *** join/#asterisk harryvv (~leonardo@S010600a0c93f6f7e.vs.shawcable.net) |
| 06:49.58 | Juggie | if the from is a locallan ip thats why |
| 06:50.01 | ClayReiche123 | in the 2nd invite? |
| 06:50.14 | ClayReiche123 | would that then be the device? |
| 06:50.14 | Juggie | anywhere, just look for private ip adrresses in the debug |
| 06:50.34 | Juggie | the device provides it |
| 06:50.39 | Juggie | when you do a sip show peers |
| 06:50.41 | harryvv | Is there a way in the dialplan to listen to a series of dtmf numbers then echo back another set of dtmf numbers? |
| 06:50.43 | ClayReiche123 | gonna try that right now. |
| 06:50.49 | Juggie | does it show the public ip |
| 06:50.51 | Juggie | or private? |
| 06:50.59 | ClayReiche123 | trying |
| 06:51.03 | Juggie | does it say N under Nat? |
| 06:51.59 | *** join/#asterisk BerndR (~konversat@mich2-145-8.utaonline.at) |
| 06:52.10 | Juggie | harryvv, Read |
| 06:52.25 | RaYmAn-Bx | harryvv: for sending there is SendDTMF..for receiving digit-timeout might work, but it goes to an extensions as opposed to retrieve the numbers |
| 06:52.40 | Juggie | 'show application Read' |
| 06:52.51 | RaYmAn-Bx | or that :) |
| 06:54.47 | harryvv | Ray, good :) want to see what it takes to energize the front door solinoid with a dtmf echo responce :) |
| 06:56.10 | *** join/#asterisk j_c (~jc@c-24-245-47-12.hsd1.mn.comcast.net) |
| 06:56.12 | RaYmAn-Bx | lol |
| 06:56.17 | harryvv | :) |
| 06:56.56 | harryvv | Its usefull for the idiot who forgets the keys alot but not for me. Just want to see how well it works. |
| 07:00.10 | implicit | hi harryvv |
| 07:00.17 | harryvv | hello |
| 07:00.22 | harryvv | whats you doing up this late |
| 07:00.29 | implicit | going insane |
| 07:00.32 | harryvv | hehe |
| 07:00.36 | implicit | lol |
| 07:00.45 | implicit | doing way too much work |
| 07:00.48 | implicit | :) |
| 07:00.50 | harryvv | i bet |
| 07:00.59 | implicit | SIP-ninjaish stuff |
| 07:01.36 | ClayReiche123 | juggie:the sip conversation is not clear to me from debug.... |
| 07:01.50 | implicit | ClayReiche123, don't worry about it |
| 07:02.04 | implicit | Juggie is a nice guy |
| 07:03.09 | ClayReiche123 | Juggie has been extremely helpful |
| 07:03.17 | Juggie | i'm tired.... but i can tell you its a nat issue.... with canreinvite=yes asterisk but nat=yes asterisk should probally present the ip you connected from (rather then the ip in the header) to your sip provider |
| 07:03.19 | implicit | i know |
| 07:03.21 | Juggie | however it likely does not |
| 07:03.23 | *** join/#asterisk techie (gus@asterisk.horizonte.us) |
| 07:03.31 | implicit | he is a nice buffer |
| 07:03.43 | Juggie | buffer? |
| 07:03.46 | Juggie | you mean patience? :P |
| 07:04.03 | implicit | he deals with all the BS that many of us are just can't handle any more |
| 07:04.14 | implicit | ;) |
| 07:04.18 | Juggie | its called living with female roomates only for the last 3 years |
| 07:04.24 | implicit | how many |
| 07:04.29 | Juggie | i've got a high tolorence |
| 07:04.30 | implicit | it must suck |
| 07:04.36 | Juggie | 2 for a while, now 1 |
| 07:04.39 | Juggie | no not really... its ok |
| 07:04.46 | ClayReiche123 | I'm used to looking at ethereal traces... I have a hard time figuring out source and destination with debug... |
| 07:04.46 | Juggie | me and vicki have been roomates for like 3 years |
| 07:04.49 | *** join/#asterisk ezhdeha (~Plork@60-240-44-231.tpgi.com.au) |
| 07:04.54 | Juggie | we are like sisters |
| 07:05.04 | Juggie | ClayReiche123, if you feel comfortable with me sshing into your server |
| 07:05.08 | Juggie | email me @ donnyk@gmail.com |
| 07:05.11 | Juggie | and we can talk tomorow |
| 07:05.30 | ClayReiche123 | ok. thank you very much! |
| 07:05.50 | harryvv | night all |
| 07:05.56 | implicit | ClayReiche123, it would be very nice if you could also give Juggie a bit of monetary compensation if juggie can help you get what you need done |
| 07:06.10 | Juggie | but like i said, with nat=yes, canreinvite=yes asterisk should present the ipv4 packet header ip to the server |
| 07:06.15 | Juggie | when it does the reinvite |
| 07:06.24 | Juggie | but i'd bet it does not |
| 07:06.33 | ClayReiche123 | You're talking about the nat'd ip? |
| 07:06.44 | Juggie | it instead presents what your sip phone presented in its sip header |
| 07:06.45 | implicit | Juggie, i deal with all my nat-fixing in SER |
| 07:06.48 | Juggie | which is likely an internal ip |
| 07:07.01 | ClayReiche123 | hehehe... we've come full circle.... |
| 07:07.13 | Juggie | it all comes down to the useragent |
| 07:07.14 | *** join/#asterisk firestrm (firestrm@S010600047577bccd.gv.shawcable.net) |
| 07:07.18 | Juggie | if it can detect its real ip |
| 07:07.18 | ClayReiche123 | ....I started out commenting how I think SER would help me.... |
| 07:07.23 | Juggie | your going to have alot less problem |
| 07:07.25 | implicit | SER is excellent |
| 07:07.34 | implicit | i center my network around SER |
| 07:07.39 | implicit | i even do all my accounting/billing on SER |
| 07:07.45 | implicit | routing, normalization, everything |
| 07:07.45 | Juggie | ser could potentially do re-writing of the sip packets |
| 07:07.53 | Juggie | to help with nat issues |
| 07:07.59 | implicit | then when i need to actually play with media i forward to sems or asterisk |
| 07:08.03 | RaYmAn-Bx | ClayReiche123: it only wasn't going to help you if you only did SIP2PSTN calls with your own hardware |
| 07:08.04 | implicit | or if i need voicemail, etc |
| 07:08.08 | ClayReiche123 | man... that sounds nice.... I've been afraid of it for so long now.... |
| 07:08.22 | Juggie | RaYmAn-Bx, hes scamming :) |
| 07:08.29 | Juggie | hes reselling someone elses voip |
| 07:08.40 | RaYmAn-Bx | Juggie: so? |
| 07:08.42 | implicit | RaYmAn-Bx, you are full of shit |
| 07:09.01 | Juggie | so when he does rtp, he wants to cheat and go direct from the user agent to the sip reseller he is buying from |
| 07:09.03 | implicit | ClayReiche123, how familiar are you with SIP? |
| 07:09.11 | implicit | at the protocol level |
| 07:09.19 | Juggie | essentially making only the SIP session use his bandwidth |
| 07:09.25 | implicit | Juggie, exactly |
| 07:09.26 | Juggie | and the RTP does direct to who he buys voip from |
| 07:09.32 | RaYmAn-Bx | Juggie: yeah, I know...but there are wholesale providers |
| 07:09.37 | t0p | Juggie: do you have any idea of how to cope with the error "app_dial.c:968 dial_exec_full: Unable to create channel of type 'Zap' (cause 0)"? |
| 07:09.49 | Juggie | t0p, your zap driver not loaded? |
| 07:09.50 | RaYmAn-Bx | implicit: obviously I'm just guessing...can't be right everytime then :P |
| 07:09.50 | ClayReiche123 | Well.... we actually have a Cisco 5350. We send some calls out our own tdm t-1 but most out of an ip provider. |
| 07:09.57 | implicit | RaYmAn-Bx, :) |
| 07:10.16 | ClayReiche123 | I familiar with sip... certainly no expert. |
| 07:10.19 | implicit | ClayReiche123, similar setup to a few couple of my clients |
| 07:10.21 | Juggie | implicit, thats actually a good idea tho i just said |
| 07:10.27 | t0p | it's loaded correctly that's what I'm guessing |
| 07:10.28 | Juggie | when nat=yes canreinvite=yes |
| 07:10.36 | implicit | ClayReiche123, although on a slightly larger scale than 1 t1 |
| 07:10.45 | implicit | :) |
| 07:10.46 | Juggie | send ipv4 header ip in packet to whom ever |
| 07:10.53 | Juggie | that way they have the proper return ip |
| 07:11.01 | Juggie | as likely it just uses the ip in the sip header |
| 07:11.02 | Juggie | hmmm |
| 07:11.08 | Juggie | i'll have to look into that further |
| 07:11.10 | firestrm | anyone here use nufone? How long does it take to get a payment processed? it been 3 days for me now.. |
| 07:11.21 | Juggie | firestrm, bug jerjer |
| 07:11.21 | implicit | firestrm, man, get a real provider |
| 07:11.34 | ClayReiche123 | We have 2-4 port t-1 cards... just not enough customers yet. |
| 07:11.35 | firestrm | implicit, like...? |
| 07:11.45 | implicit | depends on your traffic |
| 07:11.57 | implicit | ClayReiche123, yeah |
| 07:12.04 | firestrm | implicit, not much, single company right now.. |
| 07:12.07 | Juggie | ClayReiche123, when you understand the protocol, its easy |
| 07:12.14 | implicit | firestrm, oh:-\ |
| 07:12.21 | Juggie | i have my asterisk box on a private dmz network behind a firewall |
| 07:12.27 | Juggie | my phones are on different private networks |
| 07:12.27 | BerndR | hello, when just playing .wav files is then codec ulaw the right one? |
| 07:12.32 | Juggie | and i make it work no problem |
| 07:12.37 | Juggie | just have to understand how sip works |
| 07:12.46 | t0p | Juggie: I do "modprobe wcfxo,modprobe zaptel,ztcfg -vv" |
| 07:13.02 | Juggie | does ztcfg show any lines configured |
| 07:13.08 | Juggie | or spew an errror |
| 07:13.18 | t0p | Juggie: yes, Channel 01: FXS Kewlstart (Default) (Slaves: 01),1 channels configured. |
| 07:13.36 | firestrm | implicit,i take it you have had bad experience with nufone? |
| 07:13.50 | Juggie | and are you doing Dial(Zap/1/${EXTEN}) or whatever |
| 07:14.47 | Juggie | whats your dial line |
| 07:14.59 | t0p | Juggie: there's no error, it's used to show an error when I specify wrong "signalling=" in /etc/asterisk/zapata.conf |
| 07:15.03 | implicit | firestrm, well first i like SER, and i wouldn't send SIP calls over to * when they only need to be terminated, also i have reliable carriers that i deal with, so i don't need hacked-termination-providers like nufone |
| 07:16.12 | t0p | Juggie: yeah, this is what I put "exten => 3333,1,Answer,exten => 3333,2,Dial(Zap/1/026554820,20,tr)" |
| 07:16.36 | firestrm | implicit, ive had nothing but trouble with sip, thats why nufone was appealing with IAX and all.. I just want to get away from Iconnect.. they are drving me insane! |
| 07:16.43 | t0p | Juggie: then I used my SIP phone to dial the "3333" extension |
| 07:16.45 | Juggie | thats not all on one line is it top |
| 07:16.46 | implicit | firestrm, chan_sip is not SIP |
| 07:16.53 | implicit | firestrm, it is a very poor implementatio |
| 07:17.28 | implicit | anyone familiar with the SIP protocol knows how flexible and powerful it is |
| 07:17.44 | t0p | Juggie: No, they are not on one line. I just don't want to flood this this channel |
| 07:17.48 | Juggie | yah but its not without its issues |
| 07:18.19 | Juggie | it doesnt deal well with nat |
| 07:18.23 | Juggie | not without header mangling |
| 07:18.26 | budi_ | hi anyone know whether it is possible to redirect information like redirected reason (on busy), redirected number, etc when we do call forwarding? |
| 07:18.49 | Juggie | umm t0p, the dial loks fine... i'm not too sure... google for it, i have to sleep... i have to work in 5hrs |
| 07:19.04 | firestrm | implicit, good point, however, i wish some time was spent on fixing things like chan_sip rather than adding new features.. |
| 07:19.19 | t0p | Juggie: Ok, will talk to you later |
| 07:19.52 | Juggie | if ztcfg shows a channel configured that seems ok |
| 07:19.57 | t0p | Juggie: I've serached though google before coming to ask here, but still have no luck |
| 07:20.16 | Juggie | go through one of the X100 walk throughs |
| 07:20.20 | Juggie | and see if your missing something |
| 07:20.30 | *** join/#asterisk Inv_arp (junya@adsl-3-244-124.mia.bellsouth.net) |
| 07:20.30 | Juggie | my brain is poop |
| 07:20.47 | t0p | Juggie: Ok |
| 07:29.31 | *** part/#asterisk ClayReiche123 (~creiche@73-117.35-65.tampabay.res.rr.com) |
| 07:31.07 | *** join/#asterisk makkia (~pippo@nat.xsec.it) |
| 07:31.09 | makkia | hello |
| 07:31.37 | makkia | i can use a ISDN TA USB with asterisk ? |
| 07:31.46 | makkia | is chan_modem ? |
| 07:33.17 | *** join/#asterisk |Vulture| (~V@95.236.204.68.cfl.res.rr.com) |
| 07:34.04 | |Vulture| | hmm this is strange trying to do a /outgoing/ callback file, it calls the # but says the context does not exist, yet I know it does and works |
| 07:38.18 | *** join/#asterisk Jas_Williams (~jas_willi@host217-43-100-176.range217-43.btcentralplus.com) |
| 07:39.23 | t0p | under [channels] |
| 07:39.27 | t0p | is it "channel=1 |
| 07:39.44 | t0p | or channel=>1 |
| 07:39.59 | t0p | which is correct? |
| 07:42.48 | Jas_Williams | channel => 1 |
| 07:43.00 | Jas_Williams | in zapata.conf ? |
| 07:43.28 | t0p | yes,in /etc/asterisk/zapata.conf |
| 07:43.39 | Jas_Williams | then channel => is correct |
| 07:43.51 | t0p | but I saw manny people put channel=1 |
| 07:43.55 | *** join/#asterisk r0d3nt (~RatMan@wsip-24-234-241-84.lv.lv.cox.net) |
| 07:45.10 | t0p | Jas_Williams: are you using X100P? |
| 07:45.17 | Jas_Williams | Yes |
| 07:47.13 | t0p | Jas_Williams: any special setting to get the X100P to dialout? |
| 07:48.28 | t0p | Jas_Williams: I got the channel configured correctly but can't receive/send any call |
| 07:48.59 | Jas_Williams | t0p, can you try zap show channels in the cli |
| 07:51.01 | *** join/#asterisk cjk (~cjk@80.92.64.103) |
| 07:51.53 | t0p | <PROTECTED> |
| 07:51.53 | t0p | <PROTECTED> |
| 07:51.53 | t0p | <PROTECTED> |
| 07:51.57 | cjk | hi, what h323 channel driver do you recommend me for interconnecting a nice asterisk and an evil h323 |
| 07:52.10 | cjk | hi, what h323 channel driver do you recommend me for interconnecting a nice asterisk and an evil callmanager |
| 07:52.11 | cjk | sorry |
| 07:52.17 | *** join/#asterisk truescot (~truescot2@213.201.171.186) |
| 07:52.20 | Jas_Williams | chan_h323 works for me |
| 07:52.35 | Jas_Williams | or callmanager v4 and sip trunks :) |
| 07:53.09 | Jas_Williams | t0p, also ztcfg -vvv |
| 07:53.14 | t0p | Jas_Williams: as you see I set it to go to the "from-analog" context |
| 07:53.18 | truescot | hello people am looking for help if anyone can, i cannot start up asterisk because i get the message "error while writing audio data: : broken pipe" anyone got any clues to what i should be looking for? |
| 07:53.21 | Jas_Williams | from caommand line :) |
| 07:53.50 | t0p | Channel map: |
| 07:53.51 | t0p | Channel 01: FXS Kewlstart (Default) (Slaves: 01) |
| 07:53.51 | t0p | 1 channels configured. |
| 07:53.59 | t0p | Channel map: |
| 07:53.59 | t0p | Channel 01: FXS Kewlstart (Default) (Slaves: 01) |
| 07:53.59 | t0p | 1 channels configured. |
| 07:54.02 | t0p | sorry |
| 07:54.32 | t0p | Jas_Williams: I plug only one X100P in there |
| 07:55.44 | Jas_Williams | t0p, looks good |
| 07:56.05 | t0p | and the /var/log/messages shows |
| 07:56.11 | t0p | May 12 14:51:20 tas-pbx kernel: Found a Wildcard FXO: Generic Clone |
| 07:56.12 | t0p | May 12 14:51:20 tas-pbx kernel: Wildcard USB FXS Interface driver registered |
| 07:56.21 | Jas_Williams | what do you have in the [from-analog] context |
| 07:56.30 | Jas_Williams | use pastebin.ca |
| 07:57.11 | t0p | exten => s,1,Dial(SIP/1333&SIP/2333,20) |
| 07:57.11 | t0p | exten => s,2,Voicemail2(u1333) |
| 07:57.11 | t0p | exten => s,3,Hangup |
| 07:57.11 | t0p | exten => s,102,Voicemail2(b1333) |
| 07:57.11 | t0p | exten => s,103,Hangup |
| 07:57.20 | t0p | sorry |
| 07:58.46 | Jas_Williams | t0p, cat /proc/zaptel/1 |
| 07:59.17 | t0p | it's at http://pastebin.ca/11572 |
| 07:59.49 | r0d3nt | brain init.d # cat /proc/zaptel/1 |
| 07:59.49 | r0d3nt | Span 1: WCFXO/0 "Wildcard X101P Board 1" |
| 07:59.49 | r0d3nt | <PROTECTED> |
| 08:01.52 | Jas_Williams | t0p, was that you ? r0d3nt ? |
| 08:02.15 | Jas_Williams | I don't think so |
| 08:02.19 | Jas_Williams | t0p, cat /proc/zaptel/1 |
| 08:02.32 | t0p | no, thats not me |
| 08:04.01 | t0p | I put it as a comment on http://pastebin.ca/11572 |
| 08:04.08 | Jas_Williams | k |
| 08:04.17 | *** join/#asterisk tld (~tld@80.203.70.227) |
| 08:04.36 | cjk | Jas_Williams: thanks |
| 08:05.26 | *** join/#asterisk naif (~User@host250-27.pool62110.interbusiness.it) |
| 08:05.30 | Jas_Williams | t0p, you currently have a RED alarm Span 1: WCFXO/0 "Generic Clone Board 1" RED |
| 08:05.30 | Jas_Williams | <PROTECTED> |
| 08:05.53 | t0p | Jas_Williams: what does that usually mean? |
| 08:06.12 | t0p | Jas_Williams: IRQ conflict? |
| 08:06.35 | Jas_Williams | t0p, Your cable pin outs are incorrect or not plugged into a working phone line |
| 08:07.31 | Jas_Williams | to test for irq conflict to a cat /proc/interrupts and pate to pastebin |
| 08:08.44 | t0p | done |
| 08:09.02 | naif | hi |
| 08:09.12 | naif | any news related to asterisk encryption? |
| 08:10.35 | Jas_Williams | t0p, you do have an interrupt conflict as well 9 is shared 9: 998435 XT-PIC acpi, uhci_hcd, wcfxo |
| 08:10.55 | Jas_Williams | with acpi Uhci and zaptel card (wcfxo) |
| 08:12.26 | t0p | Jas_Williams: is it the same issue as the 'RED alarm Span 1: WCFXO/0 "Generic Clone Board 1" RED'? |
| 08:12.47 | t0p | Jas_Williams: because my phone line is working properly |
| 08:13.12 | t0p | Jas_Williams: might just try to swap the tip/ring pair |
| 08:13.49 | Jas_Williams | t0p, I have not seen an int conflict cause non detection of the phone line, Where are you UK, Europe ? |
| 08:14.18 | t0p | Jas_Williams: Thailand |
| 08:14.28 | Jas_Williams | t0p, are you sure you are plugged into the correct port on the card |
| 08:14.59 | Jas_Williams | t0p, it needs to be the one with a plug picture rather than the phone picture |
| 08:15.00 | t0p | Jas_Williams: yeah, there are only two ports |
| 08:15.31 | t0p | Jas_Williams: let me check again anyway |
| 08:16.44 | t0p | Jas_Williams: it's plugged into the correct port |
| 08:17.30 | t0p | Jas_Williams: any other way to test? |
| 08:17.30 | Jas_Williams | t0p, In that case I would check your Tip and Ring cables as you are seeing a RED alarm |
| 08:17.59 | *** join/#asterisk TheEmperor (~user@203.114.48.47) |
| 08:18.43 | tzafrir_laptop | any ppc folks around here? any idea why zaptel is built with -msoft-float on ppc? (not using hardware floating point?) |
| 08:19.38 | t0p | Jas_Williams: I just took a measurement of the voltage across tip/ring pair, and it's 63.9 Volts DC |
| 08:20.08 | t0p | Jas_Williams: it seems very high, should this cause a worry? |
| 08:20.33 | Jas_Williams | t0p, which pair is it presented to on the X100P it should be 2&3 |
| 08:21.01 | t0p | Jas_Williams: yeah, the two middle ones |
| 08:21.37 | jskcr|lappy | in the us its normally around 48 volts |
| 08:21.48 | Jas_Williams | t0p, I haven't measured the voltage I would expect 48volts |
| 08:22.52 | t0p | Jas_Williams: that's what I thought |
| 08:23.20 | t0p | from my pabx it's ~48 |
| 08:23.37 | tld | Any FreeBSD users around? |
| 08:24.15 | syle | any examples of a 4 port (2 fxo, 2fxs) digium card out there? |
| 08:24.20 | syle | for extensions.conf |
| 08:24.39 | Jas_Williams | t0p, do you have a message waiting light as I beleve this uses upto 90v to light the light |
| 08:25.26 | t0p | Jas_Williams: no, just an ordinary POTS |
| 08:26.05 | t0p | Jas_Williams: I will check the cable and change to 2 core cable |
| 08:26.18 | Jas_Williams | k |
| 08:30.03 | PTG1234 | what do you need tld |
| 08:30.32 | tld | PTG1234: I'm starting to look at MeetMe and IAX trunks, so I need to get timing properly up and running. |
| 08:30.38 | tld | PTG1234: Was curious how others are doing it. |
| 08:30.55 | tld | PTG1234: Kernel with HZ=1000 and ztdummy kernel module enough? |
| 08:31.04 | PTG1234 | no idea, iax is pointless to use :) and i don't use meet me |
| 08:31.18 | PTG1234 | sip is much smarter, you using your own pri |
| 08:31.23 | PTG1234 | or a voip provider |
| 08:31.23 | tld | oki. |
| 08:31.46 | tld | IAX has some nicities, mostly the trunking, and easy NAT handling. |
| 08:31.52 | tld | Other than that, I'm running pure SIP. |
| 08:32.05 | PTG1234 | do you really need to save that 2k of overhead doing a trunk |
| 08:32.11 | PTG1234 | and nat is no better on iax thats a rumor |
| 08:32.23 | PTG1234 | its just b/c iax updates more often, you can do the same in sip |
| 08:32.33 | tld | ahh, I see. |
| 08:32.34 | PTG1234 | with sip your box gets out of the loop |
| 08:32.37 | PTG1234 | which is the important part |
| 08:32.41 | PTG1234 | decreases latency |
| 08:32.44 | tld | I thought perhaps it had better handling of RTP streams. |
| 08:32.59 | tld | I prefer SIP for most things. |
| 08:33.07 | PTG1234 | sip is a better way to go for sure :) |
| 08:33.19 | tld | And have only used SIP so far, but I'm adding a new DID in Australia, which wanted IAX. |
| 08:33.33 | PTG1234 | any provider that uses iax you should run away from |
| 08:33.37 | budi_ | # D(digits): After the called party answers, send digits as a DTMF stream, then connect the call to the originating channel. |
| 08:33.38 | PTG1234 | they don't know what they are doing |
| 08:33.39 | PTG1234 | :) |
| 08:33.44 | tld | I'm a really big fan of standards, which is pretty much enough to make me want SIP over IAX. |
| 08:33.46 | budi_ | where does asterisk send dtmf to? |
| 08:34.20 | tld | PTG1234: Yeah, probably, but this is only for family convenience. |
| 08:34.32 | tld | PTG1234: If I start doing anything serious, I'll take another look at the providers there. |
| 08:34.32 | budi_ | sorry |
| 08:34.42 | budi_ | i mean in one of the options of Dial() command |
| 08:34.42 | PTG1234 | heh |
| 08:34.51 | budi_ | there is option D(digits) |
| 08:34.59 | PTG1234 | but anyhow |
| 08:35.12 | PTG1234 | i do use iax and i dont worry about timing or anything |
| 08:35.14 | PTG1234 | it just works |
| 08:35.19 | PTG1234 | i use it for testing |
| 08:35.25 | RaYmAn-Bx | budi_: presumably the called party since it would be saner to just send it using SendDTMF to the caller |
| 08:35.40 | budi_ | but when i received the call caller or recipient doesn't hear any dtmfs |
| 08:35.42 | tld | The thing with Asterisk is that if you use either IAX trunks, or MeetMe, you need a timing source. |
| 08:35.57 | PTG1234 | budi: sounds like you need inband dtmfs, or oppisite |
| 08:35.57 | *** join/#asterisk shido6 (~greg@d57-87-253.home.cgocable.net) |
| 08:36.25 | budi_ | i am using inband |
| 08:36.41 | PTG1234 | ok then don't :) |
| 08:36.44 | PTG1234 | what codec you using? |
| 08:36.48 | t0p | Jas_Williams: does the /proc/zaptel/1 update itself when I plug the line in or I will have to restart the PBX |
| 08:36.56 | budi_ | g711 |
| 08:37.00 | shido6 | when u plug a line in |
| 08:37.02 | shido6 | it knows |
| 08:37.53 | PTG1234 | try the oppisite of inband :) |
| 08:37.56 | PTG1234 | inband=no |
| 08:37.58 | PTG1234 | or whatever |
| 08:37.59 | shido6 | rfc2833 |
| 08:38.22 | shido6 | dtmfmode=rfc2833 |
| 08:38.22 | tld | note to self: Do *not* pull the zaptel module from a running kernel. Bad Things Will Happen. |
| 08:38.38 | PTG1234 | yah do what shido says |
| 08:39.01 | PTG1234 | sometimes i wonder how people have such problems, b/c if you use defaults it just works |
| 08:39.11 | budi_ | sorry it solved |
| 08:39.36 | PTG1234 | how did you solve it? |
| 08:39.39 | budi_ | my conf is wrong |
| 08:39.41 | budi_ | :) |
| 08:39.50 | PTG1234 | well we figured as much :) |
| 08:39.52 | PTG1234 | how was it wrong |
| 08:40.03 | budi_ | it used rfc2833 |
| 08:40.04 | budi_ | before |
| 08:40.11 | budi_ | then i change it to inband |
| 08:40.38 | shido6 | pastebin.ca your conf |
| 08:40.45 | shido6 | respond with the url it gives you |
| 08:42.48 | gres | Hi all. I have * with te100p. I have extention for reception, where people, ringing from pstn phone, input local number. Most pstn phone work well. But when i call from pstn phone panasonic 2363 and iput local nuber, * do not see anything. Can anybody help me? |
| 08:43.01 | gres | How can i debug dtmf? |
| 08:43.49 | gres | Sorry for my english. ^) |
| 08:46.57 | *** join/#asterisk ChrisHodgetts (~chris@topanga.archnetnz.com) |
| 08:50.45 | *** join/#asterisk prh (~paul@wacka.mjr.org) |
| 08:52.11 | BerndR | gres, i have a similar problem like you. |
| 08:52.45 | BerndR | gres, dtmf works just with snom (voip-hardphone) |
| 08:52.57 | BerndR | gres, im my case |
| 08:53.42 | BerndR | gres, if i make a call from a classic pstn-phone or cell-phone dtmf does not work |
| 08:54.20 | gres | Mmm... |
| 08:54.45 | BerndR | gres, what codec are you using? |
| 08:56.05 | gres | In my case. DTMF work well from any phone ( cellular, pstn, fax and so on), besides panasonic 2326 and old pulse phone. |
| 08:56.15 | shido6 | ok i found my wallet |
| 08:56.47 | gres | I use mainly two codecs, for faxing 711, for talks 729. |
| 08:57.17 | shido6 | ok sleepy time |
| 08:57.34 | tld | Any Australians here? |
| 08:58.38 | *** join/#asterisk fantomax1 (~fanto@81.208.114.250) |
| 09:00.08 | ChrisHodgetts | New Zealand here |
| 09:00.13 | ChrisHodgetts | sorry tld |
| 09:02.09 | tld | I'm trying to understand Australian number plans. |
| 09:02.28 | tld | I got the following DID information: DID/Centrex Extension: 0892821007 - 9143 |
| 09:02.37 | fantomax1 | hi all |
| 09:02.41 | tld | So I'm trying to find out which number I should dial to get to it. |
| 09:02.57 | ChrisHodgetts | tld I am having a hard time to figgure NZ and it's simple here :( |
| 09:03.16 | fantomax1 | is there anyone that experienced the prob with too many files opened, cannot allocate SIP/RTP channels and so on ... ? |
| 09:05.57 | *** join/#asterisk Mc_Tr (~Mc_Tr@bacterio.knet.es) |
| 09:06.03 | Mc_Tr | HI! |
| 09:06.40 | Mc_Tr | i have a TE110P T1/E1 card |
| 09:06.55 | Mc_Tr | how i configure as E1? |
| 09:07.06 | Mc_Tr | or it's automatic. |
| 09:07.09 | BerndR | gres, still here? |
| 09:07.59 | Jas_Williams | t0p, I'm Back |
| 09:08.09 | gres | BerndR ye |
| 09:09.09 | gres | BerndR: do you solve some problem? |
| 09:09.25 | BerndR | gres, i get a 'Unable to find a path from gsm to g729' when i config sip.conf for using 729, gsm and ulaw |
| 09:10.36 | gres | Use CLI show translation |
| 09:11.03 | *** join/#asterisk TheEmperor (~user@203.114.48.47) |
| 09:11.20 | TheEmperor | anyone know why when i play voicemail it sounds really scratchy? |
| 09:11.23 | BerndR | gres, thanks |
| 09:11.28 | gres | then in sip.conf allow codecs you need... |
| 09:11.43 | gres | Everything mast work well. ^) |
| 09:11.48 | gres | must |
| 09:13.15 | BerndR | gres, in my sip.conf i have a 'allow = g729' but no translation for g729 |
| 09:13.15 | tzafrir_laptop | hmmm, what exactly is ztcfg-dude? |
| 09:13.25 | tzafrir_laptop | ztcfg-dude.c in the zaptel tarball? |
| 09:13.50 | BerndR | gres, 'show translations' shows 'g729 - - - - - - - - - - -' |
| 09:14.02 | gres | BerndR install codec_g729.so first. |
| 09:14.39 | gres | then in cli loadmodule codec_g729 |
| 09:16.43 | TheEmperor | anyone know why my voicemail sounds really choppy? |
| 09:22.33 | BerndR | gres, is there a open version of g729 |
| 09:23.30 | BerndR | gres, 'Found total of 0 G.729 licenses' grrr.. |
| 09:24.36 | fantomax1 | Bern |
| 09:25.35 | BerndR | fantomax, do you mean me? |
| 09:25.40 | fantomax1 | yes |
| 09:26.52 | fantomax1 | gres do u know anything about the prob .. too many files open , or cannot allocate channel |
| 09:27.10 | fantomax1 | unable to allocate socket .. and so on? |
| 09:27.23 | gres | fantomax1: no |
| 09:27.25 | fantomax1 | does anyone know anything about it ? |
| 09:27.37 | fantomax1 | thanks anyway gres |
| 09:28.51 | RaYmAn-Bx | BerndR: there is sort of an open version of G729, but it's obviously illegal to use without proper licenses (depending on country) |
| 09:29.32 | *** join/#asterisk RoyK (~roy@80.239.107.80) |
| 09:30.28 | *** join/#asterisk Fabe (~spamhere@pD95B0BF7.dip0.t-ipconnect.de) |
| 09:32.56 | Mc_Tr | know anyone how to configure TE110P T1/E1 for e1 line |
| 09:37.21 | RoyK | change the strapping? |
| 09:37.28 | RoyK | or is it without jumpers? |
| 09:37.39 | RoyK | insmod whatever t1e1override=1 |
| 09:37.40 | RoyK | perhaps |
| 09:39.54 | *** join/#asterisk my007ms (~arkuser@217.139.240.35) |
| 09:42.23 | naif | anyone know if from E1 PRI interface |
| 09:42.30 | naif | in europe it's possible to do spoofing of caller ID? |
| 09:43.19 | zoa | yes its possible |
| 09:43.44 | t0p | Jas_Williams: thought you've gone to bed |
| 09:43.51 | *** join/#asterisk r0d3nt (~RatMan@wsip-24-234-241-84.lv.lv.cox.net) |
| 09:44.33 | Jas_Williams | t0p, No I'm in the UK its only 10:44 AM here :) I had some real work to do |
| 09:46.15 | t0p | Jas_Williams: It shows 1 WCFXO/0/0 FXSKS (In use) only if asterisk has started, right? |
| 09:47.05 | Jas_Williams | (In use) when zaptel drivers loaded and asterisk started |
| 09:47.33 | Jas_Williams | Span 1: WCFXO/0 "Generic Clone Board 1" |
| 09:47.33 | Jas_Williams | <PROTECTED> |
| 09:48.28 | Jas_Williams | Span 1: WCFXO/0 "Generic Clone Board 1" without any error menas good |
| 09:48.34 | t0p | Jas_Williams: I changed to my internal line now (with 51.1) Volts |
| 09:48.43 | Jas_Williams | k |
| 09:48.54 | t0p | Jas_Williams: but it's still the same |
| 09:49.09 | Jas_Williams | Could be a card fault ? |
| 09:49.45 | t0p | Jas_Williams: I thought the RED was referring to the "RED" module |
| 09:50.18 | Jas_Williams | no RED means RED alarm ie no VOLTS/Dialtone from the phone line |
| 09:50.19 | t0p | Jas_Williams: could be, but it's a brand new one I got from ebay last week |
| 09:50.58 | t0p | Jas_Williams: I will test with another X100P tomorrow |
| 09:52.06 | Jas_Williams | k |
| 09:53.16 | *** join/#asterisk christo (~chris@office.enovi.com) |
| 09:58.13 | *** join/#asterisk jonas (jonas@jonas.probe-networks.host.freenode) |
| 09:58.30 | naif | but spoofing of CLI |
| 09:58.35 | naif | on PRI E1 in europe |
| 09:58.44 | naif | depends on the telco? |
| 09:58.54 | naif | does someone know some telco that allow such operations trough asterisk? |
| 09:59.35 | cjk | do i reall need to loac chan_modem? |
| 09:59.51 | Mc_Tr | naif |
| 10:00.08 | naif | Mc_Tr: yes? |
| 10:00.11 | Mc_Tr | do you know how configure E1 in asterisk, in spain |
| 10:00.36 | Mc_Tr | i know this on my E1 line: Framing: CRC4 Encoding: HDB3 |
| 10:01.29 | Mc_Tr | but zaptel don't know Framing CRC4 |
| 10:01.39 | Jas_Williams | Mc_Tr, What protocol EuroISDN ? |
| 10:02.34 | ChrisHodgetts | I had issue this morning, and the problem is still there :( |
| 10:02.44 | ChrisHodgetts | when I make a call out via SIP to a Sip proxy |
| 10:02.52 | ChrisHodgetts | I get no audio on the softphone end - |
| 10:02.56 | ChrisHodgetts | the call setup is all correct |
| 10:03.16 | ChrisHodgetts | but when I ethereal I see the rtp packets but coming back from the machine with the softphone, I am seeing |
| 10:03.25 | ChrisHodgetts | a Destination Port unreachable |
| 10:03.34 | Jas_Williams | ChrisHodgetts, Sounds like a NAT problem |
| 10:03.38 | ChrisHodgetts | 7079 |
| 10:03.48 | ChrisHodgetts | it's an internal box, talking to an internal box |
| 10:04.20 | Mc_Tr | Jason357, i dont know, but it posible |
| 10:04.44 | Mc_Tr | Jas_W,i dont know, but it posible |
| 10:05.51 | Jas_Williams | Mc_Tr, can you post your zapata.conf and zaptel.conf |
| 10:05.59 | Jas_Williams | use pastebin.ca |
| 10:06.04 | *** join/#asterisk Zaw (zaw@zaw.subneural.net) |
| 10:06.05 | RoyK | pastebin them |
| 10:06.20 | RoyK | ~pastebin |
| 10:06.21 | jbot | well, pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca |
| 10:06.23 | RoyK | :) |
| 10:06.55 | Mc_Tr | zaptel.conf: span=1,0,0,ccs,hdb3 |
| 10:06.55 | Mc_Tr | bchan=1-15,17-31 |
| 10:06.55 | Mc_Tr | dchan=16 |
| 10:06.55 | Mc_Tr | loadzone = us |
| 10:06.55 | Mc_Tr | defaultzone = us |
| 10:06.57 | *** join/#asterisk nemisus (~nem@203-217-80-42.dyn.iinet.net.au) |
| 10:07.19 | Jas_Williams | Mc_Tr, in zaptel.conf |
| 10:07.29 | Mc_Tr | yes! |
| 10:07.44 | Jas_Williams | span=1,1,0,ccs,hdb3,crc4 |
| 10:08.10 | Mc_Tr | ok |
| 10:08.14 | Mc_Tr | donE! |
| 10:08.25 | *** join/#asterisk tessier (~treed@222.253.82.154) |
| 10:08.28 | tessier | Hello all! |
| 10:08.35 | Jas_Williams | Mc_Tr, Now reboot |
| 10:08.38 | tessier | Didn't realize I had failed to join this channel until just now when I had a question. |
| 10:08.40 | tessier | http://fr.pastebin.ca/10058 |
| 10:08.49 | tessier | Anyone know what causes this? Been trying to solve this problem for nearly a month. |
| 10:09.06 | tessier | Found various other people with the exact same problem via google but as usual, nobody replied to them with answers! |
| 10:10.52 | syle | how do i say dial on zap 1-2? |
| 10:11.26 | syle | exten => s,2,Dial(Zap/g1/2|20,t) |
| 10:11.38 | syle | i thought this was right but dials zap/1-1 still |
| 10:11.44 | tessier | syle: Zap/g1 means group 1 |
| 10:11.55 | tessier | In your zaptel.conf (I think) you have defined a group. |
| 10:12.02 | tessier | If you dial by group it will just pick any free line |
| 10:12.09 | tessier | If you want to dial on a particular line don't use the group |
| 10:12.27 | syle | problem is i don;t have a phone even connected to one of the ports but it uses it anyways |
| 10:12.39 | syle | so i want to be specific |
| 10:12.58 | tessier | right |
| 10:13.08 | tessier | Problem is you have a line with nothing plugged into it defined as part of your group |
| 10:13.08 | syle | g1 includes both fxo ports |
| 10:13.55 | syle | well i am just testing right , how do i say dial second zap port in group 1? |
| 10:14.50 | tessier | Soo....anyone know what auto-fall through means? |
| 10:14.57 | Jas_Williams | syle, just dial the port directly exten => s,2,Dial(Zap/2,20,t) |
| 10:15.02 | tessier | I swear the behavior of auto-attendants has changed since stable asterisk |
| 10:15.21 | tessier | As soon as it finishes playing the message it drops the call. |
| 10:15.31 | tessier | Do I have to put something in the dialplan now that tells it to wait for input? |
| 10:15.44 | syle | that works, anyway to specify zap/2 with a group to? |
| 10:16.05 | syle | i guess no point nm |
| 10:18.34 | syle | well what i want to do |
| 10:18.42 | syle | is put my homelines on zap/1 |
| 10:18.48 | syle | and my fax machine on zap/2 |
| 10:19.12 | syle | i am guessing i can just pickup a part for zap/1 so i can plug multiple lines into it |
| 10:19.26 | syle | cordless, non-cordless etc |
| 10:20.56 | syle | what option can change ring tones? |
| 10:22.03 | syle | distinctive rings i guess |
| 10:22.47 | syle | be nice if i could ring the phone a different way to know if it was long distance or not was what i was thinking |
| 10:25.29 | *** join/#asterisk nine76 (~t00r@cpe-69-135-184-24.woh.res.rr.com) |
| 10:25.38 | Jas_Williams | syle, Typically the card will run upto 4 phones, |
| 10:35.03 | *** join/#asterisk Sato1 (~rauleli@sato1.wizardteam.com) |
| 10:36.19 | cjk | cant i do a noload => all in modules.conf and then load what is necessary |
| 10:37.16 | shido6 | something like that |
| 10:37.43 | shido6 | I do an autoload yes |
| 10:37.49 | shido6 | and choose to do a noload |
| 10:39.37 | ManxPower | I hate mornings |
| 10:40.04 | RoyK | ManxPower: sleep longer |
| 10:40.37 | shido6 | heheh |
| 10:40.51 | shido6 | after 4 am hit I figured, what the hell Im not sleeping today |
| 10:42.20 | *** join/#asterisk _pat_msg (pat@r00tworld.com) |
| 10:42.26 | _pat_msg | hi |
| 10:43.30 | ManxPower | I went to bed early, but had trouble sleeping. |
| 10:43.53 | ManxPower | 7 days until I leave |
| 10:44.58 | RoyK | ManxPower: leave where? |
| 10:44.59 | RoyK | VON? |
| 10:45.17 | RoyK | shit. need to book hotel........ |
| 10:45.30 | ManxPower | RoyK: The ManxPower 2005 European Tour. |
| 10:46.00 | ManxPower | (which starts out in Stockholm) |
| 10:48.36 | ManxPower | RoyK: I'm staying at a small B&B just outside of city center. |
| 10:50.02 | *** join/#asterisk cmk (~cmk_@p54A3CA40.dip.t-dialin.net) |
| 10:56.57 | *** join/#asterisk Dovid (~hirisk@pool-138-89-178-170.mad.east.verizon.net) |
| 10:57.49 | Mc_Tr | Hi Manx, |
| 11:00.51 | tzanger | hahaha |
| 11:00.54 | tzanger | Your certificate is set to expire in approximately -23.86 days time, you can renew this by going to the following URL: |
| 11:02.30 | Mavvie | yes, that's cacert.org for you. |
| 11:02.44 | Mavvie | they replied to me with: |
| 11:02.50 | Mavvie | We've only just started sending out notifications of certificates about |
| 11:02.50 | Mavvie | to expire, and wasn't sure who has renewed their certificates or not. |
| 11:03.33 | ManxPower | Out CA started sending us renewal e-mails 6 months before our certificate expired |
| 11:04.16 | Mavvie | but you would get 3 extra months if you immediatelly renewed it. |
| 11:07.08 | tzanger | heh |
| 11:16.14 | *** join/#asterisk gambolputty (~gambolput@cblmdm69-45-216-83.buckeye-express.com) |
| 11:19.26 | *** part/#asterisk RoyK (~roy@80.239.107.80) |
| 11:20.23 | *** join/#asterisk RoyK (~roy@80.239.107.80) |
| 11:25.58 | *** join/#asterisk Newbie___ (~me@60.48.45.107) |
| 11:26.08 | Newbie___ | hi, please help me |
| 11:26.24 | Mavvie | sure, rate is AU$ 80 per hour |
| 11:26.32 | Newbie___ | -- Extension '9500' in context 'default' from '0164229929' does not exist. Rejecting call on channel 0/21, span 1 |
| 11:26.39 | Newbie___ | caller cant call in |
| 11:26.47 | Mavvie | the solution is there! |
| 11:28.12 | Newbie___ | paypay dont take my cc, i am not in the 45 countries listed to accept cc |
| 11:29.44 | cjk | how can i disable iaxprov? |
| 11:29.49 | MikeJ[Laptop] | do you have an extension 9500 in your default context |
| 11:30.04 | Newbie___ | no, 9500 is my PRI number |
| 11:30.19 | Newbie___ | fuck, i am dead |
| 11:30.26 | MikeJ[Laptop] | ok, well there is your problem |
| 11:30.37 | MikeJ[Laptop] | your are trying to make a call to 9500 and it's not there |
| 11:30.39 | *** join/#asterisk __a (~a@212.154.32.104) |
| 11:30.51 | __a | does anyone know what's wrong with digium's CVS server? |
| 11:30.53 | MikeJ[Laptop] | why? |
| 11:30.55 | Newbie___ | my PRI number is 2199500 |
| 11:31.03 | __a | I can't checkout HEAD |
| 11:31.06 | ManxPower | __a: rerun cvs a couple of times. |
| 11:31.25 | ManxPower | __a: Digium's cvs server is actually serveral servers. |
| 11:31.26 | __a | I did around 10 times, it just gets stuck after a few files |
| 11:31.41 | ManxPower | __a: Oh well. Wait a few hours and try it again. |
| 11:31.44 | Dovid | eh |
| 11:31.45 | ManxPower | or get 1.0.x |
| 11:31.48 | Dovid | what are you looking for ? |
| 11:32.24 | __a | zaptel drivers |
| 11:32.24 | Dovid | hmm |
| 11:32.24 | Dovid | i got that on my server |
| 11:32.25 | Dovid | just not the latest drivers |
| 11:32.25 | __a | but I'm running * HEAD, so I believe I need zaptel HEAD as well |
| 11:32.25 | Dovid | http://www.h6315.com/pub |
| 11:32.25 | Dovid | don have the HEAD |
| 11:32.25 | MikeJ[Laptop] | ManxPower, digiums cvs servre is not several servers anymore.. it's only 1 |
| 11:32.30 | RoyK | __a: yes, you do |
| 11:33.09 | __a | cvs server: Updating zaptel |
| 11:33.09 | __a | U zaptel/.cvsignore |
| 11:33.09 | __a | U zaptel/ChangeLog |
| 11:33.12 | __a | stuck forEVER |
| 11:33.14 | __a | pretty lame |
| 11:33.16 | *** join/#asterisk FITA1 (~m_ahmed@202.5.145.50) |
| 11:33.36 | FITA1 | hi all |
| 11:33.38 | BerndR | does anyone know how to make speex codes running in asterisk? |
| 11:33.45 | ManxPower | MikeJ[Laptop]: When did that happen? |
| 11:34.02 | MikeJ[Laptop] | __a, try checking out into a new directory |
| 11:34.27 | MikeJ[Laptop] | a week ago, NuFone no longer supplies a CVS mirror |
| 11:34.51 | MikeJ[Laptop] | asterlink now supplies the only mirror |
| 11:34.51 | ManxPower | MikeJ[Laptop]: Ah. not suprizing. |
| 11:35.12 | ManxPower | One would think that Digium could afford to provide their own CVS server. |
| 11:36.19 | MikeJ[Laptop] | but why bother |
| 11:36.22 | FITA1 | I m in a conference(app_meetme2) with my fried after few minutes we thought that we should invite our other friend. Can we do this by dial his phone number while we are in conference?????? |
| 11:36.30 | Newbie___ | calling out from that span is fine, but no incoming |
| 11:36.41 | MikeJ[Laptop] | FITA1, there is a bug in mantis |
| 11:36.56 | MikeJ[Laptop] | Newbie___, I already told you what the problem is |
| 11:37.40 | __a | yea, they could put it on sourceforge |
| 11:37.45 | FITA1 | MikeJ[Laptop]: what is that bug |
| 11:37.54 | Newbie___ | MikeJ[Laptop]: but i never define any 9500 previously |
| 11:38.08 | Newbie___ | where can i look ? extensions.conf ? |
| 11:38.14 | MikeJ[Laptop] | you need an extension with the DNIS that is getting delivered down the line (9500 in this case) and in the context for that line (default in this case_ |
| 11:38.21 | MikeJ[Laptop] | FITA1, look it up |
| 11:38.44 | MikeJ[Laptop] | Newbie___, yes, that's in extensions.conf |
| 11:38.45 | *** join/#asterisk newbien (~e@147.241.33.65.cfl.res.rr.com) |
| 11:39.20 | Newbie___ | MikeJ[Laptop]: ok |
| 11:39.23 | FITA1 | MikeJ[Laptop]: I m unable to catch you |
| 11:39.33 | MikeJ[Laptop] | __a, can you put dual licensed commercial stuff on sf? |
| 11:39.44 | MikeJ[Laptop] | FITA1, go to bugs.digium.com |
| 11:39.56 | MikeJ[Laptop] | FITA1, look for the meetme bugs |
| 11:40.11 | Newbie___ | MikeJ[Laptop]: can i trouble you to look at http://pastebin.ca/11575 |
| 11:40.12 | MikeJ[Laptop] | FITA1, one of them is for outdialing from meetme |
| 11:40.16 | Newbie___ | please |
| 11:40.28 | FITA1 | MikeJ[Laptop]:ok |
| 11:40.49 | MikeJ[Laptop] | Newbie___, if you don't know how to make an extension, then you are in big troubls |
| 11:41.17 | MikeJ[Laptop] | Newbie___ do you want it to do that if it comes in as dnis 9500 |
| 11:41.26 | Newbie___ | i am already in big trouble |
| 11:41.52 | Newbie___ | MikeJ[Laptop]: yes, |
| 11:41.54 | MikeJ[Laptop] | exten ==> 9500,1,goto(default,s,1) |
| 11:44.37 | Newbie___ | MikeJ[Laptop]: before or after [default] |
| 11:45.56 | MikeJ[Laptop] | in the default context |
| 11:46.22 | Newbie___ | MikeJ[Laptop]: thanks you SOOOO much |
| 11:46.23 | MikeJ[Laptop] | Newbie___, goto to voip info and read the stuff on configuration |
| 11:46.27 | MikeJ[Laptop] | ~docs |
| 11:46.41 | 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 |
| 11:46.41 | Newbie___ | it worked |
| 11:46.54 | Newbie___ | but, i never add 9500 in [default] before, why now ? any idea ? |
| 11:47.15 | MikeJ[Laptop] | you never took calls on that dnis before? |
| 11:47.39 | Newbie___ | it has been working as it was for the last 9 mths, same 9500 number |
| 11:47.44 | MikeJ[Laptop] | if you didn't know how to add that, you REALLY need to spend some time reading |
| 11:48.00 | Newbie___ | no, i meant. i never add that and it was working |
| 11:48.35 | MikeJ[Laptop] | I don't know... that's a PRI config... if it never came in on that dnis before and now it is, and you changed nothing then your pri provider did |
| 11:49.49 | *** join/#asterisk makhtar (~ageller@mail.bulletinnews.com) |
| 11:50.59 | *** join/#asterisk Networker (~icechat5@cpe-069-132-048-043.carolina.res.rr.com) |
| 11:53.17 | *** join/#asterisk darwin35 (~darwin35@24.3.226.147) |
| 11:54.03 | BerndR | does any know the problem of interrupting calls when playing audio files? |
| 11:54.21 | *** part/#asterisk Networker (~icechat5@cpe-069-132-048-043.carolina.res.rr.com) |
| 11:54.30 | *** join/#asterisk TheEmperor (user@218.111.50.44) |
| 11:54.31 | BerndR | it's no matter if the files are wav or mp3 |
| 11:54.46 | BerndR | the codec also does not matter |
| 11:55.28 | BerndR | and the amount of parallel call also does not matter |
| 11:56.34 | BerndR | every few seconds a short interrupt could be heard |
| 11:58.39 | BerndR | the hardware is a dual xeon with 1GB RAM |
| 11:59.25 | BerndR | CPU load is just about 1-2% |
| 12:00.07 | Mc_Tr | when y call to an one zap channel, and this zap channel are forward to a sip extension, why it takes 3 seconds (more or less) in ring sip extension? |
| 12:01.52 | BerndR | i only have sip calls (no zap) |
| 12:03.03 | BerndR | the duration of the interrupt is about 100-200ms |
| 12:03.10 | ManxPower | Mc_Tr: Perhaps Zap is waiting for callerid information? |
| 12:04.36 | darwin35 | anyone here have call forwarding working |
| 12:04.48 | ManxPower | darwin35: yes |
| 12:05.00 | darwin35 | manX hey |
| 12:05.04 | darwin35 | mines stopped |
| 12:05.09 | darwin35 | dont know why |
| 12:05.14 | ManxPower | darwin35: Just set it in your polycom phone. |
| 12:05.22 | darwin35 | did they change phrasing |
| 12:06.06 | darwin35 | I need it for the grandstreams |
| 12:06.36 | ManxPower | darwin35: Sorry, I would not even give my ex-wife a grandstream device. |
| 12:06.51 | ManxPower | I'd give her a bomb, but not a grandstream. |
| 12:07.01 | darwin35 | well it was the first set of phones I bought |
| 12:07.09 | darwin35 | before I knew they where shit |
| 12:07.32 | darwin35 | but I still keep them around |
| 12:07.39 | darwin35 | they are paid for |
| 12:09.59 | darwin35 | lol |
| 12:10.14 | darwin35 | but back to cf |
| 12:11.42 | *** join/#asterisk durex (~ironman@weber.anpa.org.br) |
| 12:11.43 | darwin35 | http://pastebin.ca/11561 there is what I have |
| 12:12.09 | syle | lets say you had exten => _1NXXNXXXXXX first in config file, isn;t a exten => _1800NXXXXXX after never going to get executed since 1800 matches the first instance of it? or should you put toll free numbers before long distance? |
| 12:12.13 | darwin35 | its not suppost to dial the nmbr just store it but it dials it and does not store it |
| 12:12.27 | *** join/#asterisk tld (~tld@196.80-202-89.nextgentel.com) |
| 12:12.42 | ManxPower | syle: patterns in the SAME context are matched most specific first, regardless of order. |
| 12:13.02 | ManxPower | This may or may not apply to include =>'d context stuff |
| 12:13.50 | darwin35 | ? |
| 12:15.16 | *** join/#asterisk Zeeek (~Zeeek@Zeeek.sustaining.supporter.pdpc) |
| 12:15.38 | darwin35 | it use to work on 1.0.7 but on head it stopped |
| 12:15.50 | ManxPower | well do back to 1.0.x |
| 12:15.58 | Zeeek | I have a ManxPower emergency |
| 12:16.14 | ManxPower | But as everyone knows CVS-HEAD is production quality and has no major bugs. |
| 12:16.34 | darwin35 | hahah |
| 12:16.36 | ManxPower | Zeeek: Is there any significant reason I should stop in Paris during my trip? |
| 12:16.39 | Zeeek | I locked up my ip500 so tight 468* doesn't unlock it |
| 12:16.46 | Zeeek | any siggestions? |
| 12:16.51 | Zeeek | or sug? |
| 12:16.51 | ManxPower | Zeeek: I've never had that happen. Did you powercycle it? |
| 12:16.54 | darwin35 | shooot it |
| 12:17.06 | Zeeek | hundreds of desperate times |
| 12:17.19 | darwin35 | call the manufacture |
| 12:17.25 | ManxPower | Zeeek: I don't have any suggestions then, other to RMA it. |
| 12:17.28 | Zeeek | not easy |
| 12:17.30 | ManxPower | darwin35: Polycom does not support Asterisk users. |
| 12:17.50 | Zeeek | well, since I just brought it in my suitcase, I may be screwed |
| 12:17.53 | darwin35 | that bites |
| 12:18.02 | ManxPower | Zeeek: Try powering o the phone without it being plugged into the network. |
| 12:18.07 | Zeeek | gotta stop playing with yoys when jetlagged |
| 12:19.09 | ManxPower | Zeeek: Is there any significant reason I should stop in Paris during my trip? |
| 12:19.13 | Zeeek | I've done that, but I'll try it again later. Right now I am on the same hub. It show athe logo on powerup and get an ip from DHCP |
| 12:19.27 | ManxPower | Zeeek: Well, that's a start. 8-) |
| 12:19.36 | Zeeek | ManxPower other than it's prolly one of the better cities, I dunno |
| 12:19.55 | Zeeek | brb |
| 12:20.05 | ManxPower | Zeeek: I meant from a "talking to people about a job" perspective 8-) |
| 12:21.32 | *** join/#asterisk eeek (~Zeeek@80.125.80.38) |
| 12:22.39 | *** join/#asterisk mbranca (~matteo@81.208.92.210) |
| 12:24.04 | eeek | shit, that vreally looked like a good phone :( |
| 12:26.53 | BerndR | is there a possibility to configure asterisk to use smaller udp packages for data transport? i think default is 172 |
| 12:27.52 | *** join/#asterisk roamer323 (~sing@HSE-MTL-ppp64388.qc.sympatico.ca) |
| 12:28.32 | *** join/#asterisk ilium007 (~brantwint@220-253-92-177.QLD.netspace.net.au) |
| 12:28.47 | FITA1 | <PROTECTED> |
| 12:29.06 | FITA1 | should I use that patch |
| 12:30.40 | *** join/#asterisk Zeeek (~Zeeek@Zeeek.sustaining.supporter.pdpc) |
| 12:31.03 | *** join/#asterisk Romik (~romik@1.fix.netvision.net.il) |
| 12:31.08 | FITA1 | can any body help me, i m new to bugs.digium.com. Is it safe to use patch avaiable on this link http://bugs.digium.com/view.php?id=3405 |
| 12:31.47 | *** join/#asterisk Aze` (~aze@host229-162.pool80105.interbusiness.it) |
| 12:32.25 | ilium007 | hi all |
| 12:32.41 | ilium007 | can anyone help me with an asterisk install |
| 12:32.52 | ilium007 | i found a doc last night on the net about installing on gentoo linux |
| 12:32.55 | ilium007 | cant find it now |
| 12:32.59 | *** join/#asterisk bjohnson (~bjohnson@jecinc.tor.istop.com) |
| 12:33.16 | ilium007 | just wondering if there is anyone that can help |
| 12:33.23 | BerndR | ilium007, emerge asterisk |
| 12:33.44 | ilium007 | i gather emerge is the gento package insatller ? |
| 12:33.46 | Aze` | Anyone know gsm box products ? |
| 12:34.10 | BerndR | ilium, yes |
| 12:34.13 | _pat_msg | there is problem with gentoo because zaptatel don't use devfs |
| 12:34.19 | _pat_msg | for u information |
| 12:34.45 | *** join/#asterisk CdtDelta (~CdtDelta@dsl081-225-161.chi1.dsl.speakeasy.net) |
| 12:34.47 | BerndR | ilium, use version 1.0.7 |
| 12:35.26 | BerndR | ilium: it's masked but ok |
| 12:36.10 | ilium007 | ok |
| 12:36.18 | ilium007 | masked ?? |
| 12:37.10 | BerndR | ilium, the currrent relese in portage tree is 0.9 as far as i know |
| 12:37.22 | ilium007 | oh ok |
| 12:37.35 | ilium007 | i have never installed gentoo |
| 12:37.50 | ilium007 | i have downloaded the minimal install cd iso |
| 12:37.59 | ilium007 | not as easy as freebsd !! |
| 12:38.24 | flickerfly | apparently, there is a patch for zaptel for devfs, but it's ugly |
| 12:38.26 | BerndR | ilium, i never installed freebsd :) |
| 12:38.39 | darwin35 | dude asterisk works fine on fbsd |
| 12:38.53 | ilium007 | yeah ???? |
| 12:38.57 | darwin35 | yes |
| 12:39.03 | darwin35 | its int he ports |
| 12:39.05 | ilium007 | i though id give gento a go |
| 12:39.10 | ilium007 | never seen it before |
| 12:39.11 | darwin35 | ok have fun |
| 12:39.22 | ilium007 | what is the recommended option ? |
| 12:39.27 | darwin35 | ./usr/ports/net/asterisk |
| 12:39.38 | darwin35 | ./usr/ports/misc/libpri |
| 12:39.39 | BerndR | ilium, /usr/portage/net-misc/asterisk |
| 12:39.50 | durex | yes |
| 12:39.55 | flickerfly | or use udev and not devfs |
| 12:39.55 | darwin35 | ./usr/ports/misc/zaptel |
| 12:39.56 | durex | I'm running all my * on FBSD |
| 12:40.02 | ilium007 | ok |
| 12:40.13 | ilium007 | makybe time to tak out gentoo cd and star with freebsd again |
| 12:40.43 | ilium007 | i dont have any digium cards yet, but i read you can connect * box to VoiP provider |
| 12:40.50 | ilium007 | <PROTECTED> |
| 12:40.57 | darwin35 | ye you can |
| 12:40.57 | ilium007 | is this tru |
| 12:41.04 | darwin35 | sip and iax2 |
| 12:41.08 | ilium007 | then for testing i can just use softphoens |
| 12:41.10 | ilium007 | cool |
| 12:41.14 | darwin35 | I have mine connected to 3 providers |
| 12:41.25 | ilium007 | this is so damm cool |
| 12:41.27 | BerndR | ilium, if you like conferences you need udev too |
| 12:41.40 | darwin35 | I have a iax nmbr a sip nmbr a 888 nmbr abd a local nmbr |
| 12:42.02 | ilium007 | i am interested in it all as I am just about to spend $100k on 3 systems for work, Alcatel boxes |
| 12:42.06 | ManxPower | ~docs |
| 12:42.07 | 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 |
| 12:42.12 | ilium007 | in my searched i came acros asterisk |
| 12:42.17 | *** join/#asterisk MikeJ[Laptop] (~ircatjerr@mi.origenfinancial.com) |
| 12:42.48 | Aze` | Anyone use xfer trasfert on * ? |
| 12:43.10 | ilium007 | i also read that without the digium cards i need to set up timing |
| 12:43.13 | ManxPower | Aze`: All the time. Press the transfer button your polycom phone. |
| 12:43.29 | ilium007 | will this be on top of the free bsd software that is in the ports ? |
| 12:43.35 | ManxPower | Is nobody able to spell this morning, or is my display screwed up? |
| 12:43.51 | ilium007 | heheheh i can never spell in chat rooms ! |
| 12:43.57 | Aze` | ManxPower i want use atxfer function of asterisk. |
| 12:44.39 | ManxPower | Aze`: Yes. Press the transfer button on your phone. |
| 12:44.48 | darwin35 | ztdummy is part of zaptel in pports |
| 12:44.48 | ManxPower | Does your phone NOT have a transfer button? |
| 12:44.57 | darwin35 | you add 1 line to your kernel |
| 12:45.08 | BerndR | ilium, for conferences you'll need a timer |
| 12:45.09 | darwin35 | options HZ=1000 |
| 12:45.25 | Aze` | ManxPower i've proble with use zap channel... i listen double ring .. 1th simulated by * and 2th real.. why ? |
| 12:45.29 | darwin35 | recompile and loadup ztdummy |
| 12:45.29 | Aze` | problem |
| 12:45.42 | Druken | 100,000? god i wish i had that much money to blow on a phone system |
| 12:45.54 | darwin35 | same here |
| 12:45.58 | ManxPower | Aze`: I don't know, but if you want to transfer using a zap channel then enable trasnfers in zapata.conf and press FLASH/RECALL to use it. |
| 12:46.27 | ilium007 | we are a hospital |
| 12:46.32 | ilium007 | multiple sites |
| 12:46.47 | darwin35 | going to add wifi phones for the DR ? |
| 12:47.00 | ilium007 | DR ? |
| 12:47.06 | darwin35 | Doctors |
| 12:47.09 | ilium007 | oh docs...DECT |
| 12:47.34 | ilium007 | staying away from wifi phones, need to spend more time on radius / 802.1x security solution |
| 12:47.53 | Druken | radius? wtf for? |
| 12:48.14 | ilium007 | WPA authentication for a wireless data network |
| 12:48.19 | darwin35 | they have wep in them |
| 12:48.27 | darwin35 | yeah |
| 12:48.33 | ilium007 | wep |
| 12:48.36 | ilium007 | wep aint wpa |
| 12:48.50 | darwin35 | the have 128 bit encryption |
| 12:49.02 | darwin35 | I use them in nursing homes |
| 12:49.07 | ManxPower | I wish I knew why so many people have a fetish for t/T transfers with Asterisk. |
| 12:49.08 | ilium007 | yeah - and a static key |
| 12:49.18 | darwin35 | yep |
| 12:49.19 | ilium007 | it dont take too long to bust a 128 but WEP key |
| 12:49.41 | darwin35 | I use very stange keys |
| 12:49.50 | ilium007 | anyway i like the dect phones and i dont have decent access points |
| 12:49.59 | Aze` | ManxPower but isnt annunced trasfer but blind:) |
| 12:50.03 | jskcr|lappy | it only takes 1 hour to break a 128k wep key |
| 12:50.04 | ilium007 | i dont think it matters how strange you make your keys |
| 12:50.10 | ilium007 | they are always the same - static |
| 12:50.21 | ManxPower | Aze`: No it is NOT. |
| 12:50.28 | ilium007 | WPA changes its encryption keys too quick to be able to crack them |
| 12:50.35 | ManxPower | Aze`: the FLASH supports announced transfers |
| 12:50.46 | ManxPower | FLASH+DIAL+talk+FLASH+hangup |
| 12:50.52 | ManxPower | or maybe even |
| 12:50.55 | ManxPower | FLASH+DIAL+talk+hangup |
| 12:51.45 | Grooby | FLASH didn't work too well for me with an ATA |
| 12:51.51 | darwin35 | well I had to have wireless for the nurses and no one has messed with the systems thus fare |
| 12:51.54 | Grooby | i can never be sure when it's transfered through |
| 12:51.57 | ilium007 | i spose |
| 12:52.04 | Druken | Grooby: works good for me |
| 12:52.05 | ilium007 | i would just feel better with WPA |
| 12:52.15 | Grooby | that's good for you |
| 12:52.16 | Grooby | :) |
| 12:52.24 | ManxPower | <PROTECTED> |
| 12:52.30 | Romik | manxpower: what is this r t/T transfers ? |
| 12:52.31 | Pkunk | it's the usual ASCII charset , with those wierd |^_^| like higher chars inside |
| 12:52.33 | ilium007 | he, can someone take a look at this site and let me know if this install guide for * on gentoo is acurate: http://thinkhole.org/projects/pbx/ |
| 12:52.39 | Pkunk | whoops wrong chan |
| 12:52.42 | ManxPower | Romik: no, this is with flash transfers |
| 12:52.43 | Aze` | flash is #.. ? |
| 12:52.55 | ManxPower | no, flash is the flash or recall button on your analog phone. |
| 12:53.13 | ManxPower | T/t # transfers were never designed to be the primary way to transfer in Asterisk. |
| 12:53.16 | Aze` | on my sipura spa-841 ... i havent it |
| 12:53.42 | ManxPower | Aze`: Make up your mind. Are you using a SIP phone or a Zap channel? |
| 12:53.49 | Druken | ManxPower: FLASH or LINK |
| 12:54.02 | darwin35 | if I where you I would use a debian minimal install and then install the needed packages |
| 12:54.05 | ManxPower | Aze`: The read the damn docs for the SIPura phone. It will tell you how to do superviserd transfers. |
| 12:54.07 | Romik | manxpower: we use Flash to transfer call with flash on zap phones... receive call - flash dial speak....flash (3 parties on line) hangup. |
| 12:54.11 | darwin35 | then build and install * |
| 12:54.21 | darwin35 | but thats me for linux boxes |
| 12:54.23 | ManxPower | Romik: exactly! no T/t crap needed. |
| 12:54.25 | darwin35 | or slackware |
| 12:55.02 | Aze` | manxPower .. i know how use superviserd trasfers on SIPura phone.... but it's complicated to use by stupid receptionist.. |
| 12:55.07 | Romik | manxpower: what is T/t ? it some special button? |
| 12:55.13 | ilium007 | not freebsd ??? i got the disks here thats all !!!!!! |
| 12:55.24 | darwin35 | fbsd is easy also |
| 12:55.35 | ManxPower | Romik: no it's a special option to the dial command to allow you to use # to transfer if you are using devices that are too stupid to support transfers. |
| 12:55.38 | Aze` | i want use special button to do supervised trasfers |
| 12:55.46 | darwin35 | and the ports ver installs the bri stuff |
| 12:55.57 | darwin35 | isdn support |
| 12:55.59 | Romik | manxpower: thanks! i undestand! |
| 12:56.16 | darwin35 | what ver of fbsd |
| 12:56.22 | darwin35 | I am on 5.4-s |
| 12:56.23 | ManxPower | Aze`: The last I heard supervized transfers using t/T in CVS-HEAD do not work. |
| 12:56.36 | ilium007 | 5.4 |
| 12:56.40 | darwin35 | ok |
| 12:56.48 | ManxPower | I don't have even 1 person using t/T transfers. |
| 12:56.54 | ManxPower | Since it makes IVRs not work. |
| 12:57.27 | Druken | ManxPower: how does it not make ivr's work? |
| 12:57.54 | ManxPower | Druken: Because most IVRs want you to press # at some point, at which time Asterisk asks you for the extension to transfer to. |
| 12:58.16 | ilium007 | what is the zaptel driver - is this the timing thing |
| 12:58.29 | *** join/#asterisk Fabe (~spamhere@pD95B0BF7.dip0.t-ipconnect.de) |
| 12:58.32 | darwin35 | ztdummy is timiing |
| 12:58.53 | ilium007 | oh and thats the one that depends on the type of usb chipset you have ???? |
| 12:58.53 | *** join/#asterisk ariel_ (~Ariel@adsl-068-157-125-248.sip.mia.bellsouth.net) |
| 12:59.01 | darwin35 | its part of the zap drivers only neede for meetme |
| 12:59.01 | ariel_ | hello everyone |
| 12:59.20 | darwin35 | not on fbsd it uses any usb chip |
| 12:59.26 | syle | any examples on setting up block lists? |
| 12:59.41 | darwin35 | youput 1 line in your kernel |
| 13:00.07 | darwin35 | and then loadup ztdummy and you have timing |
| 13:01.21 | ilium007 | ok cool |
| 13:01.22 | darwin35 | nnice to know all my porting work is being used |
| 13:01.31 | ilium007 | what do you mean by 1 line in kernal |
| 13:01.53 | ilium007 | hmmm i am getting lost quick here ! |
| 13:01.53 | darwin35 | when you build your kernel you add the line options HZ=1000 |
| 13:02.32 | darwin35 | have you never built a kernel on fbsd |
| 13:03.17 | ilium007 | nope :( |
| 13:03.24 | darwin35 | wow |
| 13:03.31 | ilium007 | :| |
| 13:03.50 | ilium007 | this is just hobby stuff at the moment for me :) |
| 13:04.02 | ilium007 | i am willing to learn though ! |
| 13:04.22 | darwin35 | heheh but you willing to pay to be trained |
| 13:04.25 | darwin35 | hhehehe |
| 13:04.30 | darwin35 | lol |
| 13:04.42 | darwin35 | or are you office broken |
| 13:04.44 | ilium007 | its a hobby :) thats part of it - find help for free :) |
| 13:04.58 | BerndR | does anyone have a solution for this jittery sound while playing audio files? |
| 13:05.12 | darwin35 | yes its callled timing |
| 13:05.28 | darwin35 | and dropping mpg123 and get madplay |
| 13:05.36 | BerndR | timing? |
| 13:06.09 | BerndR | i'm using mono wav with 8000 |
| 13:06.13 | darwin35 | you have to have a timing device for MOH |
| 13:06.25 | darwin35 | ook |
| 13:06.25 | bjohnson | not that I've heard |
| 13:06.30 | darwin35 | not sure then |
| 13:06.39 | bjohnson | timing is for iax trunks and meetme |
| 13:06.53 | darwin35 | are you using mpg123 |
| 13:07.10 | *** join/#asterisk eye69 (magnus@upcore.net) |
| 13:07.16 | BerndR | mpg123 is running |
| 13:07.37 | darwin35 | did you install the one that comes with * |
| 13:07.46 | BerndR | yes |
| 13:07.51 | darwin35 | with make mpg123 in the * dir |
| 13:07.56 | eye69 | Hey. Can anybody give a tip on a workable free SIP client for Windows? I just want my friends to be able to use my Asterisk server. |
| 13:08.03 | *** part/#asterisk Grooby (~Grooby@ip24-250-126-171.dc.dc.cox.net) |
| 13:08.03 | ilium007 | darwin35: can you give me an idea on what is required to add that line to the frbsd kernel ? |
| 13:08.12 | RaYmAn-Bx | eye69: x-lite or sjphone |
| 13:08.13 | BerndR | installed by emerge |
| 13:08.13 | MikeJ[Laptop] | BerndR, head or stable? |
| 13:08.17 | ilium007 | x-lite ?? |
| 13:08.28 | eye69 | RaYmAn-Bx: Ok, thanks. |
| 13:08.28 | BerndR | 1.0.7 |
| 13:08.32 | darwin35 | ill pvt me |
| 13:09.37 | BerndR | i use mpg123 Version 0.59s-r9 (2000/Oct/27) |
| 13:09.46 | syle | how do i say if incomming calling number is 999-9999...do this...? |
| 13:10.33 | MikeJ[Laptop] | syle, use 2 lines XXXX and 999 |
| 13:10.33 | ManxPower | exten => 9999999,1,Dial(SIP/666) |
| 13:10.46 | ManxPower | syle: Is English not your native language. |
| 13:10.47 | ManxPower | ? |
| 13:11.02 | MikeJ[Laptop] | wait, was that a range of numbers or 1 number? |
| 13:11.10 | MikeJ[Laptop] | hehe |
| 13:11.21 | MikeJ[Laptop] | it's one of our answers anyway |
| 13:11.28 | ManxPower | MikeJ[Laptop]: His lack of English language skills does make it hard to help him. |
| 13:11.41 | MikeJ[Laptop] | be nicer |
| 13:11.47 | *** join/#asterisk jterrero (~jt@66.28.34.162) |
| 13:11.56 | BerndR | mike, i'm -20 |
| 13:12.03 | MikeJ[Laptop] | ? |
| 13:12.24 | syle | maxpower i mean for calls comming into me |
| 13:12.55 | syle | not going out |
| 13:12.55 | BerndR | mike, the priority of asterisk |
| 13:13.05 | ManxPower | syle: That IS for calls coming into you. All calls into Asterisk are treated the same. Asterisk tries to match the dialed number with an exten => line. |
| 13:13.08 | MikeJ[Laptop] | BerndR, sorry not following |
| 13:13.22 | ManxPower | Unless you are using analog, in which case it will try to match exten => s,1,whatever |
| 13:13.27 | MikeJ[Laptop] | syle, replace the dial line with whatever you want |
| 13:13.50 | MikeJ[Laptop] | but the part after extern => is the number matching part |
| 13:14.00 | MikeJ[Laptop] | exten |
| 13:14.21 | Sato1 | do i really need to compile zaptel to make work asterisk if i m not going to use any digium in this box? |
| 13:15.20 | syle | hmm i tried that and it default to s , so i guess i can;t match on the analog fx port at all? |
| 13:15.38 | ManxPower | Sato1: You do not need Zaptel if you are not using any Digium or Sangoma hardware and do not need MeetMe or IAX2 TRUNKING |
| 13:15.39 | darwin35 | if you want trunking and meetme yes |
| 13:15.54 | ManxPower | syle: FXwhat? FXS or FXO? |
| 13:16.03 | *** join/#asterisk |Vulture| (~V@95.236.204.68.cfl.res.rr.com) |
| 13:16.09 | ManxPower | We can't help you if you are being lazy about your typing. |
| 13:16.19 | syle | fxo |
| 13:16.25 | ManxPower | You cannot match on dialed number on analog FXO ports. |
| 13:16.25 | |Vulture| | is it required to patch HEAD with g729 or is it on there now? |
| 13:16.31 | Sato1 | ManxPower, i need iax2 actually, this box will be connected to another asterisk via iax2 |
| 13:16.37 | ManxPower | syle: Since the telco doesn't even send you the dialed number. |
| 13:16.43 | *** join/#asterisk Patrick^ (~patrickm@birch4.mountaincable.net) |
| 13:16.51 | syle | it does i get call display |
| 13:17.14 | ManxPower | Sato1: IAX2 works without zaptel. IAX2 TRUNKING does not. TRUNKING is something you can enable to save bandwidth when you have 3 or more calls going between the same two asterisk servers. |
| 13:17.28 | Sato1 | got it |
| 13:17.30 | Sato1 | thanks |
| 13:17.33 | ManxPower | syle: NO! The telco sends you the CALLING number (CallerID), not the DIALED number. |
| 13:18.01 | *** join/#asterisk trig (~jb@xob.neospire.net) |
| 13:18.20 | syle | so is there a way around this? environment variable for callerid |
| 13:18.41 | ManxPower | syle: You cannot get the dialed number. You can get the calling number. What do you want to do? |
| 13:19.10 | syle | i want if 9999999 is matched to send this bastard telemarketer to a busy signal |
| 13:19.12 | BerndR | what version of asterisk-oh323 is better? 0.6.5 or 0.5.10 |
| 13:19.38 | ManxPower | syle: So you want to send the call to a specific location if the CALLERID matches something? |
| 13:19.41 | *** join/#asterisk Gand_DJ (fabsced@ptr-207-54-104-24.ptr.terago.ca) |
| 13:19.47 | syle | yes |
| 13:19.53 | ManxPower | syle: You are going to have a lot of trouble if you have trouble expressing yourself. |
| 13:20.00 | ManxPower | syle: look up "ex-girlfriend" in the Wiki |
| 13:20.16 | ManxPower | ~google site:lists.digium.com ex-girlfriend |
| 13:22.06 | syle | exten => 6153248305/_931NXXXXXXX,1,Queue(queue1); |
| 13:22.27 | syle | CID i take it is the second part |
| 13:22.29 | ManxPower | syle: You cannot know the dialed number on calls from an FXO port. |
| 13:22.39 | ManxPower | exten => s/_931NXXXXXXX,1,Queue(queue1); |
| 13:23.07 | *** part/#asterisk eye69 (magnus@upcore.net) |
| 13:24.26 | *** part/#asterisk Zeeek (~Zeeek@Zeeek.sustaining.supporter.pdpc) |
| 13:24.37 | MikeJ[Laptop] | syle read the wiki. |
| 13:26.07 | *** join/#asterisk airwolf (~airwolf_1@adsl241.dyn106.pacific.net.sg) |
| 13:26.16 | airwolf | hi everyone |
| 13:26.19 | MikeJ[Laptop] | ~google site:www.voip-info.com ex-girlfriend |
| 13:26.36 | MikeJ[Laptop] | well that wasn't helpful |
| 13:26.45 | *** join/#asterisk iq (~iq@65-103-164-141.omah.qwest.net) |
| 13:26.59 | airwolf | anyone know how to call inphonex account from asterisk? |
| 13:27.15 | airwolf | i had register a free service from inphonex |
| 13:27.41 | airwolf | how can i call another inphonex user? |
| 13:29.38 | *** join/#asterisk h3x0r (Justino@64.192.116.29) |
| 13:35.24 | *** join/#asterisk anthm (~anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 13:35.24 | *** mode/#asterisk [+o anthm] by ChanServ |
| 13:36.14 | *** join/#asterisk boch (~as24@200.59.172.98) |
| 13:39.15 | tzafrir_laptop | I'm trying to figure out something in the zaptel makefile |
| 13:40.27 | tzafrir_laptop | INSTALL_PREFIX serves like a prefix under which the package should be installed, right? |
| 13:40.39 | tzafrir_laptop | Because there are some places where it is ommited. |
| 13:41.03 | *** join/#asterisk likwid-- (likwid@nc-65-41-168-104.dyn.sprint-hsd.net) |
| 13:41.17 | tzafrir_laptop | And I'm trying to figure out why in the current debian package they have replicated this functionality with DESTDIR |
| 13:45.02 | *** join/#asterisk kryme (~PK@66-211-192-4.velocity.net) |
| 13:45.39 | CyberKnet | Those Teliax guys are pretty nice =) |
| 13:45.49 | *** join/#asterisk Betu| (~betul@62.244.193.101) |
| 13:45.51 | ManxPower | CyberKnet: Yes they are |
| 13:46.28 | CyberKnet | Hopefully they'll get 60/6 billing soon with no connection fee. |
| 13:46.50 | ManxPower | I don't really mind the connection fee TOO much. |
| 13:46.55 | kryme | Is there a FAQ somewhere that tells how to setup SIP through a NAT device? I haven't been able to find any real good information on it. |
| 13:47.15 | ManxPower | kryme: Is the SIP device behind NAT and Asterisk NOT behind NAT? |
| 13:47.16 | Sato1 | voip-info.org? |
| 13:47.26 | kryme | ManxPower, correct. |
| 13:47.47 | Sato1 | nat=yes |
| 13:47.51 | ManxPower | kryme: if you use nat=yes in the [sipdevice] section of sip.conf then there is no special setup on the SIP device. |
| 13:48.11 | kryme | Wow. Too easy. Thanks. :) |
| 13:48.12 | ManxPower | In fact, if you do do special setup on the SIP device or the NAT router then it will usually break if you use nat=yes |
| 13:48.35 | kryme | That's awesome. Then things will work out amazingly for me. :D |
| 13:48.40 | ManxPower | kryme: I'm assuming you do not have a totally stipud SIP device and not a totally stupid NAT router. |
| 13:49.41 | kryme | Well, right now I'm just using a softphone. I'll be getting a ZyXEL Prestige 2000W shortly, tho. |
| 13:49.51 | ManxPower | kryme: That's a very bad idea. |
| 13:49.56 | kryme | Oh? |
| 13:50.09 | bkw_ | "Because its his opinion" |
| 13:50.16 | ManxPower | ~google site:lists.digium.com Zyxel problem |
| 13:50.36 | ManxPower | Results 1 - 10 of about 147 from lists.digium.com for Zyxel problem |
| 13:50.44 | bkw_ | for the "most" part the phone works from what I hear |
| 13:51.06 | Sato1 | is it cheap? |
| 13:51.09 | bkw_ | the early firmware for th ephone sucked |
| 13:51.09 | ManxPower | kryme: FULLY research the phone first before buying it. Some people have had issues. Make sure you are aware of the issues first. |
| 13:51.32 | bkw_ | Later firmware seems to have addressed a lot of the problems |
| 13:51.33 | h3x0r | all voip phones suck |
| 13:51.39 | bkw_ | no they don't |
| 13:51.39 | h3x0r | NEXT! |
| 13:51.40 | ManxPower | kryme: Then if you still want to, go buy it. |
| 13:51.50 | ManxPower | All softphone suck |
| 13:51.53 | bkw_ | ManxPower, has me on his ignore. |
| 13:51.54 | ManxPower | NEXT! |
| 13:51.57 | Sato1 | i just got a soyo g668 to play around for a little while |
| 13:51.59 | bkw_ | and he's speaking out his ass righ tnow |
| 13:52.07 | bkw_ | If softphone sucks.. asterisk sucks |
| 13:52.13 | syle | i did search for ex-girlfriend on wiki pages don;t think i got one, would someone be nice enough to provide the full url? |
| 13:52.15 | bkw_ | because asterisk is nothing more than a softphone |
| 13:52.19 | bkw_ | on steroids |
| 13:52.35 | ManxPower | syle: Try searching for "exgirlfriend" |
| 13:52.35 | kryme | OK. I appreciate it. I'll do some more research. |
| 13:52.36 | h3x0r | yeah well |
| 13:52.39 | ManxPower | It's a BASIC feature of Asterisk |
| 13:52.42 | Betu| | Hi; is there any doc. about incompatible codec setting ? |
| 13:52.46 | h3x0r | a soft phone is like a girl without a pussy |
| 13:52.50 | h3x0r | whats the point |
| 13:52.55 | bkw_ | h3x0r, no |
| 13:52.58 | h3x0r | haha |
| 13:53.10 | Sato1 | i wouldnt say that much |
| 13:53.12 | CyberKnet | bkw_: so I'm curious... do you just have a number in a different state given the lack of porting by asterlink for oklahoma, or do you have a 405 area code, or just an 800 number, or... (how else can I get more invasive? =P) |
| 13:53.14 | bkw_ | softphones with dedicated audio hardware like the DA-60's I think.. work great. |
| 13:53.17 | syle | no matches |
| 13:53.27 | ManxPower | syle: then search the mailing list. |
| 13:53.29 | bkw_ | CyberKnet, we have nothing but tollfree right now. |
| 13:53.37 | bkw_ | CyberKnet, but that is about to change. |
| 13:53.41 | ManxPower | If you want me to hold your hand then I require dinner, drinks, and cash. |
| 13:53.43 | CyberKnet | bkw_: Aaaaah. |
| 13:53.51 | CyberKnet | bkw_: about to change in what regards? |
| 13:53.57 | h3x0r | broadwing's generator is so fucking loud%#!^#!$^$#! |
| 13:54.14 | bkw_ | CyberKnet, can't say yet |
| 13:54.18 | bkw_ | doing interop testing soon |
| 13:54.22 | *** join/#asterisk chipach (~chip@chocolate.chip.net) |
| 13:54.40 | CyberKnet | does doing number porting require having hardware in that area code, or is it just software based? |
| 13:55.05 | bkw_ | it depends |
| 13:55.11 | CyberKnet | like so many things =) |
| 13:55.29 | FITA1 | I m in a conference(app_meetme2) with my fried after few minutes we thought that we should invite our other friend. Can we do this by dial his phone number while we are in conference?????? |
| 13:55.46 | ManxPower | kryme: Also be sure to research WiFi issues with VoIP. Expecially "turbo" mode. I'm not sure if the turbo mode issues still apply or not, but it's better to do the research before you drop the money on a phone |
| 13:55.54 | CyberKnet | Well, please let me know when you get Oklahoma. I really want to use asterlink, but I couldn't convince my wife to get an 800 number instead of porting our existing number. |
| 13:56.04 | bkw_ | CyberKnet, will do.. I want it also |
| 13:56.20 | ManxPower | CyberKnet: no other providers have numbers in OK? |
| 13:56.33 | *** part/#asterisk Betu| (~betul@62.244.193.101) |
| 13:56.37 | CyberKnet | bkw_: I can imagine so. Is it realistic to expect you might get it, or are there many more area codes in line before it?> |
| 13:56.41 | CyberKnet | ManxPower: Not many. |
| 13:56.50 | CyberKnet | ManxPower: Vonage does. =P" |
| 13:56.54 | bkw_ | kryme, you can ignore ManxPower .. he's an opinionated prick that thinks he knows everything and has an opinion on just about everything, most of which is based on hearsay. |
| 13:57.17 | ManxPower | CyberKnet: looked at Teliax? |
| 13:57.27 | bkw_ | kryme, he hitachi wifi phones are very nice |
| 13:57.29 | ManxPower | CyberKnet: there are a little more expensive, but I've been happy with them. |
| 13:57.30 | bkw_ | s/he/the/ |
| 13:57.33 | Beirdo | bkw_: I think every channel collects a few of em :) |
| 13:57.35 | CyberKnet | ManxPower: I was in negotiations with them earlier today. |
| 13:57.49 | bkw_ | Beirdo, i'm sure |
| 13:58.01 | CyberKnet | heh |
| 13:58.12 | ManxPower | CyberKnet: If you have very high usage someone else might be better, but for low to medium usage I think they are a good choice. |
| 13:58.16 | bkw_ | CyberKnet, i'm in oklahoma.. so you know I am gonna get that one first |
| 13:58.19 | FITA1 | bkw_:I m in a conference(app_meetme2) with my fried after few minutes we thought that we should invite our other friend. Can we do this by dialing his phone number while we are in conference?????? |
| 13:58.32 | bkw_ | FITA1, I dont use meetme or meetme2 |
| 13:58.33 | CyberKnet | bkw_: heh. Wasn't sure of your position in the ocmpany heirachy chart =) |
| 13:58.35 | bkw_ | so I dont know |
| 13:58.50 | bkw_ | CyberKnet, hehe |
| 13:58.56 | FITA1 | any suggestion |
| 13:59.13 | CyberKnet | bkw_: Well then, I will wait with bated breath. |
| 13:59.13 | FITA1 | bkw_: what do you use |
| 13:59.16 | bkw_ | go search for outdial on the wiki |
| 13:59.35 | bkw_ | we wrote our own conf app |
| 13:59.50 | CyberKnet | ManxPower: I make about 200 calls a month, average about 1000 minutes on those calls. teh connect fee is only totalling about $2.00 a month, but it's $2.00 that counts. |
| 13:59.55 | FITA1 | I have looked at it, but it doesn't suit or not the approprait solution |
| 14:00.19 | ManxPower | CyberKnet: if $2 makes or breaks a deal you have problems I cannot help with. |
| 14:00.35 | FITA1 | bkw_: ur own conf app is not and open source |
| 14:00.49 | bkw_ | FITA1, nope |
| 14:00.50 | CyberKnet | ManxPower: there isn't a person in the world who should *not* be concerned about $2.00 |
| 14:00.51 | bkw_ | its internal only |
| 14:01.01 | FITA1 | so sad |
| 14:01.07 | h3x0r | two bucks? |
| 14:01.13 | h3x0r | jesus |
| 14:01.14 | Beirdo | CyberKnet: sure... Bill Gates |
| 14:01.18 | bkw_ | CyberKnet, its good to be froogle |
| 14:01.27 | FITA1 | do somthing for others toooooo |
| 14:01.30 | bkw_ | froogle people aren't usually in debt |
| 14:01.31 | FITA1 | bkw_: |
| 14:01.36 | Beirdo | s/froogle/a cheep bastard/ :) |
| 14:01.39 | CyberKnet | bkw_: Yes sir. |
| 14:01.39 | Beirdo | heh |
| 14:01.46 | bkw_ | i'm semi froogle |
| 14:01.46 | CyberKnet | Beirdo: You already know I'm a scotsman =P" |
| 14:01.51 | Beirdo | although there is something to be said for that |
| 14:01.51 | bkw_ | but i'm also semi in debt |
| 14:02.02 | CyberKnet | bkw_: $2 for connect charges. $5 for access period. It all adds up. |
| 14:02.04 | FITA1 | well |
| 14:02.09 | Beirdo | I spend lots, and am currently not in debt |
| 14:02.09 | bkw_ | CyberKnet, yep |
| 14:02.26 | CyberKnet | Beirdo: Thats because your pay check is the size of the empire state building =) |
| 14:02.30 | CyberKnet | *grin* |
| 14:02.32 | bkw_ | Beirdo, send some to me |
| 14:02.34 | bkw_ | i'll help you |
| 14:02.35 | Beirdo | hehe. |
| 14:02.52 | CyberKnet | I have a PAP2 courtesy of Beirdo =) |
| 14:03.13 | Beirdo | too bad you now want to ditch Vonage, it won't be as useful now |
| 14:03.15 | CyberKnet | although the bastard is locked... but I knew that when I acquired it. |
| 14:03.33 | CyberKnet | Beirdo: It has gained me 7 digit dialing for 3 months. well worth the 25 bucks I paid. |
| 14:03.37 | CyberKnet | ;) |
| 14:03.39 | Beirdo | :) |
| 14:03.52 | Beirdo | and it got a useless box out of my apt |
| 14:03.58 | CyberKnet | plus it is WAY more reliable than that motorola piece of crap I had. |
| 14:04.09 | CyberKnet | I'll definitely be returning the motorola over the linksys =) |
| 14:04.23 | CyberKnet | (have to return one to avoid a $30 charge) |
| 14:04.29 | Beirdo | yeah |
| 14:04.38 | FITA1 | bkw_ I want an application boss |
| 14:04.47 | CyberKnet | and there's a *tiny* chance that I could unlock the PAP2 eventually. Not the Motorola though. |
| 14:05.05 | CyberKnet | Is unlocking ATA's a banned subject in here? I hadn't considered that it may be. |
| 14:05.08 | bkw_ | FITA1, get da checkbook out |
| 14:05.16 | chipach | Anyone here have any experience using rxfax with *? |
| 14:05.23 | FITA1 | i have :) |
| 14:05.26 | bkw_ | chipach, yes it works |
| 14:05.31 | bkw_ | FITA1, how much you got? |
| 14:05.32 | chipach | I'm getting "Poor Line Quality" |
| 14:05.41 | FITA1 | how much you want |
| 14:05.53 | chipach | It doesn't seem to be training correctly, yet two fax machines over the same physical lines DO work. |
| 14:05.59 | ManxPower | CyberKnet: bkw_ will disagree with me, but I think that trying to unlock a PAP2 is rather pointless unless you just want to do it for the hack value. |
| 14:06.03 | boch | is this: "SIP/user:pwd@host/ext" right, for an outgoing call ? |
| 14:06.19 | CyberKnet | ManxPower: for what particular reason? |
| 14:06.31 | Beirdo | ManxPower: that makes no sense |
| 14:06.39 | CyberKnet | ManxPower: an FXS costs $50 minimum. |
| 14:06.42 | ManxPower | boch: bkw will disagree with me, but I think I saw something on the -dev list about that format not working. |
| 14:06.43 | Beirdo | he has a 2-line ATA that he could use if it were unlocked |
| 14:06.52 | CyberKnet | ManxPower: Are you now going to tell me I have problems for wanting to save $50? |
| 14:07.04 | bkw_ | ManxPower, can fuck off because I don't disagree with him.... It's pretty pointless. |
| 14:07.17 | boch | ManxPower deam |
| 14:07.28 | bkw_ | boch, no |
| 14:07.36 | bkw_ | you setup peers/friends |
| 14:07.47 | bkw_ | and its SIP/exten@peer |
| 14:07.57 | bkw_ | or SIP/ip/exten |
| 14:08.01 | bkw_ | or SIP/exten@ip |
| 14:08.07 | boch | im making a dynamic routing with my agi |
| 14:08.09 | bkw_ | take your pick |
| 14:08.44 | ManxPower | CyberKnet: no, I'm saying that chances of you being able to unlock the PAP2 are pretty small. |
| 14:08.45 | CyberKnet | I would like to hear why it is pointless please, if someone wouldn't mind explaining. |
| 14:08.52 | Mc_Tr | Himeko, i come back |
| 14:08.53 | boch | damn, my work is useless |
| 14:09.11 | CyberKnet | ManxPower: aaaah. Yes. I'm not going to invest much time in it.... more like "keep an eye open and try if something gets thrown my way" |
| 14:09.23 | ManxPower | CyberKnet: now THAT is a good idea. |
| 14:09.42 | CyberKnet | ManxPower: The likelihood of it being able to be unlocked are astronomically low. |
| 14:09.54 | Sato1 | todays HEAD-CVS does not compile |
| 14:09.59 | Beirdo | which is why I sold it off to a Vonage subscriber |
| 14:09.59 | h3x0r | speaking of cheap ass mother fuckers |
| 14:10.00 | Beirdo | heh |
| 14:10.02 | bkw_ | Sato1, what is the error? |
| 14:10.07 | bkw_ | because I just compiled it |
| 14:10.12 | h3x0r | i was moving and needed to get rid of some steelcase furniture |
| 14:10.20 | h3x0r | i couldnt sell it for $20 if my life depended on it |
| 14:10.24 | Sato1 | chan_zap.c:61:2: #error "You need newer libpri" |
| 14:10.29 | h3x0r | but i put it on craigslist for $0 the day i had to move out |
| 14:10.31 | *** join/#asterisk bprice20 (~brandon@Dynamic-216.120.224.151.hrnoc.net) |
| 14:10.33 | h3x0r | and i had 25 calls about it |
| 14:10.42 | CyberKnet | Beirdo: And why I will probably do the same and use the cash towards either a digium FXO+FXS card, or an SPA-3000 |
| 14:10.48 | bkw_ | Sato1, update your libpri |
| 14:10.49 | Beirdo | :) |
| 14:10.49 | bkw_ | duh |
| 14:10.53 | bkw_ | its telling you exactly what to do |
| 14:10.54 | Sato1 | i did |
| 14:10.56 | Beirdo | third-hand PAP2 |
| 14:11.14 | Sato1 | still giving the same error |
| 14:11.17 | bkw_ | no you didn't |
| 14:11.25 | bkw_ | make clean |
| 14:11.26 | bkw_ | make again |
| 14:11.28 | CyberKnet | Beirdo: we have to share the "love" around. heh. |
| 14:11.33 | Beirdo | yup |
| 14:11.47 | Sato1 | Updating from CVS |
| 14:11.48 | Sato1 | cvs server: Updating . |
| 14:11.48 | Sato1 | [root@master libpri]# |
| 14:11.49 | CyberKnet | strange kind of vonage love. |
| 14:12.00 | CyberKnet | Sato1: make clean in libpri and make install |
| 14:12.01 | bkw_ | haha |
| 14:12.05 | Sato1 | ok, lets do it again |
| 14:12.11 | bkw_ | make update libpri too |
| 14:12.13 | bkw_ | make clean |
| 14:12.16 | bkw_ | make update clean install |
| 14:12.17 | bkw_ | there |
| 14:12.22 | CyberKnet | yes |
| 14:12.26 | CyberKnet | sorry, that's what I meant. |
| 14:12.30 | bkw_ | hehe |
| 14:12.37 | CyberKnet | <-- new clubie |
| 14:12.42 | boch | is it possible to read the var ${HANGUPCAUSE} in agi script, after the $agi->exec(dial ? |
| 14:12.44 | Sato1 | i did it, lets do it again |
| 14:13.40 | Sato1 | done with libpri, lets go back with asterisk |
| 14:13.47 | ManxPower | boch: Unless they fixed that, you cannot get access to automatically set extensions.conf variables within an AGI without doing something like SetVar(MY_HANGUPCAUSE=${HANGUPCAUSE}) before calling the AGI |
| 14:14.14 | ManxPower | boch: Ah, after execing a Dial. Gads, I have no idea. |
| 14:14.34 | ManxPower | boch: Does you AGI script even still have control after Dial exits? |
| 14:14.47 | Mc_Tr | ManxPower, i solved my problem |
| 14:14.50 | Sato1 | so.. lets see, it will take some time, its an AMD-k6-II 500mhz |
| 14:14.51 | Mc_Tr | Thanks |
| 14:15.16 | Mc_Tr | in zapata.conf i change usecallerid=yes to usecallerid=no |
| 14:17.26 | ManxPower | Mc_Tr: You're welcome |
| 14:17.36 | boch | ManxPower yes |
| 14:18.17 | boch | i need that for the accounting, or if the call failed, pick another route |
| 14:19.59 | boch | maybe a i could convine DeadAGI and uniqueid.. |
| 14:20.06 | bprice20 | I have 2 sip questions. |
| 14:20.06 | *** join/#asterisk grolloj (~chatzilla@slim-eth0.horizonlive.net) |
| 14:20.34 | bprice20 | <PROTECTED> |
| 14:20.34 | bprice20 | port number there. Currently is = Contact: |
| 14:20.35 | bprice20 | <sip:+18666775910@67.15.74.73> |
| 14:20.35 | *** join/#asterisk mhnoyes (~mhnoyes@user-2ivfmq5.dialup.mindspring.com) |
| 14:20.39 | ManxPower | boch: see this |
| 14:20.50 | ManxPower | ~google site:lists.digium.com HANGUPCAUSE AGI |
| 14:21.01 | bprice20 | and in the SDP body of the PSTN - SIP OK message could you send a |
| 14:21.01 | bprice20 | ptime:20. |
| 14:21.05 | *** join/#asterisk dalabera (~Dalabera@mail.pmrtechnologies.com) |
| 14:21.42 | Juggie | ManxPower, i think they may have fixed that because i've had no problems |
| 14:21.57 | boch | thanks ManxPower |
| 14:22.50 | ManxPower | Juggie: There was talk of it. I guess it would depend on what version of Asterisk you are using. |
| 14:24.11 | *** join/#asterisk DrWho17 (~MIKE@mike-new.tc3net.com) |
| 14:24.39 | *** part/#asterisk christo (~chris@office.enovi.com) |
| 14:25.03 | boch | Once you run Dial from an AGI script, you lose control of the call via |
| 14:25.03 | boch | the AGI script. |
| 14:25.10 | boch | thats the answer :( |
| 14:25.58 | m0f0x | Hey, does anyone got a successful build of channels/h323, on CVS version? |
| 14:26.10 | ManxPower | boch: I thought so. Want to know how I dealt with the problem? |
| 14:26.41 | Sato1 | m0f0x, just read the README file, do what it says, it should compile with no problems |
| 14:27.01 | m0f0x | Sato1, I did, can I paste the error on a pastebin? |
| 14:27.05 | boch | ManxPower yes, tellme |
| 14:27.19 | bkw_ | Never dial in an AGI |
| 14:27.24 | bkw_ | only set Vars and exit out of the AGI |
| 14:27.33 | ManxPower | boch: This is how I did it in the past. Not doing it anymore. |
| 14:27.51 | Sato1 | m0f0x, well, may be some one else may help, i just did it with the mentioned versions of pwlib and openh323 |
| 14:27.52 | Sato1 | so lets see |
| 14:28.03 | ManxPower | boch: I run the AGI to set the required variables, in the next priority I run the Dial, then in the following priority I run an AGI script to process the results. |
| 14:28.18 | Juggie | that is the best thing to do |
| 14:28.34 | bkw_ | Cluecon is a good place to go to learn all about this stuff |
| 14:28.38 | m0f0x | Sato1, Which Linux distribution you're using? |
| 14:28.41 | bkw_ | www.cluecon.com... registration is open! |
| 14:28.45 | Juggie | you can use the same agi if you want |
| 14:28.47 | boch | ManxPower i see |
| 14:28.49 | Juggie | just be sure to pass in flags.... |
| 14:28.53 | Juggie | so you know your state |
| 14:28.59 | Jason357 | m00 |
| 14:29.25 | ManxPower | boch: it may seem more complicated than required, but it's what needs to be done. Also investigate the "g" option to Dial |
| 14:29.27 | Sato1 | m0f0x, an old one, rh9 |
| 14:29.37 | boch | ManxPower but, why dont you do that any more? |
| 14:29.43 | m0f0x | Sato1, take a look: http://pastebin.ca/11586 |
| 14:29.52 | m0f0x | Sato1, are you using Asterisk stable or CVS? |
| 14:29.55 | ManxPower | boch: I don't have need dial from within an AGI anymore. |
| 14:30.08 | ManxPower | boch: It was my first attempt as a Super dial script. |
| 14:30.08 | boch | oh, okey |
| 14:30.20 | boch | lol |
| 14:30.23 | Sato1 | m0f0x, i m using the one before todays |
| 14:30.34 | m0f0x | Sato1, I see |
| 14:30.37 | *** part/#asterisk ilium007 (~brantwint@220-253-92-177.QLD.netspace.net.au) |
| 14:31.10 | m0f0x | Sato1, You build pwlib & openh323 just by running make opt, right? |
| 14:31.11 | ManxPower | boch: I decided it was easier to have a billion line extensions.conf macro 8-) |
| 14:31.22 | boch | heh |
| 14:31.24 | Sato1 | m0f0x yup |
| 14:31.54 | Sato1 | m0f0x but... once again, make sure you download the right versions of pwlib and openh323 mentioned in the readme file |
| 14:32.10 | m0f0x | Sato1, I did that... :( |
| 14:32.18 | *** join/#asterisk rcam (~rcammobil@adsl-218-151-77.jax.bellsouth.net) |
| 14:32.23 | bjohnson | ManxPower: you should check out the superdial macro on the wiki |
| 14:32.24 | boch | ManxPower: i have all the routes in a sql db, so i order by price, and dial it |
| 14:32.36 | boch | if it fails, pick the next one |
| 14:33.17 | Sato1 | bkw_, still, getting the same error while compiling todays version, it says that i need a new version of libpri, and i just made sure for 3th time it was the latest from cvs |
| 14:33.25 | ManxPower | bjohnson: that has not even close to the functionality of my script |
| 14:33.40 | rcam | Would you guys recommend installing Asterisk on Debian or Red Hat Linux Enterprize? |
| 14:33.54 | ManxPower | bjohnson: let me put up a copy of my script on pastebin |
| 14:34.17 | rcam | Enterprise rather* |
| 14:34.26 | m0f0x | Sato1, Thanks anyway, I'll recheck everything again |
| 14:34.35 | Moc_ | Anyone have a DLink DES-1526 switch É |
| 14:34.58 | bjohnson | anyone know how to get 'find' to exclude a directory? |
| 14:35.22 | *** join/#asterisk juice (~juice@mo-69-68-105-244.dyn.sprint-hsd.net) |
| 14:36.02 | ManxPower | bjohnson: http://pastebin.ca/11587 |
| 14:40.19 | boch | nice dialplan |
| 14:40.30 | jterrero | can someone please help me out? i get the following message when trying to modprobe wct4xxp |
| 14:40.33 | jterrero | zaptel: Unknown symbol crc_ccitt_table |
| 14:41.01 | ManxPower | "The variables can be set using SetVar, DBGet, AGI, web based CGI script, manager interface, government mind control rays, etc. How they are set is up to you." |
| 14:41.24 | ManxPower | ~google site:lists.digium.com crc_ccitt_table |
| 14:41.29 | rcam | In the lastest build of * is some sort of timing source still required for MOH? |
| 14:41.36 | ManxPower | jterrero: now don't you feel silly? |
| 14:41.47 | ManxPower | rcam: no, Not since 0.70 |
| 14:41.52 | ManxPower | or maybe 0.90 |
| 14:41.54 | rcam | ManxPower Nice. |
| 14:42.07 | *** join/#asterisk JonR800 (jr@pcp05013027pcs.plyntv01.mi.comcast.net) |
| 14:42.15 | jterrero | ManxPower: ??? what do you mean |
| 14:42.42 | Juggie | jterrero, for asking the question without doing research first. |
| 14:42.58 | jterrero | ive looked at the digium mail list and postings from other users |
| 14:43.02 | jterrero | none of them fix my issue |
| 14:43.06 | jterrero | ive been doing this since yesterday |
| 14:43.24 | rcam | ManxPower Are there any isssues with running Asterisk on Debian 3.0 Stable? |
| 14:43.24 | jterrero | i just recompiled latest kernel for gentoo, did everything from scratch |
| 14:43.27 | jterrero | same shit |
| 14:43.30 | jterrero | sorry, language |
| 14:43.41 | Juggie | its fuckin ok |
| 14:43.42 | Juggie | :P |
| 14:43.45 | rcam | ;) |
| 14:43.56 | ManxPower | rcam: no idea. |
| 14:44.14 | Beirdo | or like our highschool bus driver once screamed at us: "HEY! Watch your fucking language!" |
| 14:44.17 | Beirdo | heh |
| 14:44.26 | ManxPower | jterrero: you mean none of them talk about making sure to enable that feature in the kernel build? |
| 14:44.39 | rcam | What do you guys think of Asterisk@Home? |
| 14:45.12 | ManxPower | rcam: I think it could be good for people with simple needs for a home asterisk install |
| 14:45.23 | Juggie | jterrero, http://lists.digium.com/pipermail/asterisk-users/2004-October/066548.html |
| 14:45.25 | bjohnson | ManxPower: that doesn't work the same as the superdial macro |
| 14:45.47 | bjohnson | ManxPower: that is only suitable for incoming calls |
| 14:45.57 | *** join/#asterisk Grooby (~Grooby@66.160.105.186) |
| 14:46.15 | Juggie | oh oh |
| 14:46.17 | ManxPower | bjohnson: see the dial-result macro near the end of that pastebin |
| 14:46.20 | Juggie | libpri is fuxored |
| 14:46.22 | rcam | ManxPower Would it not suffice for a company? |
| 14:46.27 | bjohnson | eg s-BUSY only goes to voicemail or hangs up .. doesn't return to the dialplan to try another extension |
| 14:46.36 | Juggie | like 3-4 bug reports about libpri in the last 1-2 days |
| 14:47.08 | ManxPower | bjohnson: Um, s-BUSY will try another extension. |
| 14:47.18 | ManxPower | set by the CFBL_DEST variable |
| 14:47.43 | ManxPower | bjohnson: unless I screwed it up yesterday when I worked on updating the macro. 8-) |
| 14:47.55 | Juggie | ahh |
| 14:47.59 | Juggie | someone finally pinpointed it |
| 14:48.01 | Juggie | manx, http://bugs.digium.com/bug_view_page.php?bug_id=4247 |
| 14:48.12 | Juggie | april 22nd, libpri got screwed |
| 14:48.16 | bjohnson | it only tries one alternate number .. not an unlimited list |
| 14:48.34 | *** join/#asterisk El^Diablo (~konversat@216.52.67.200) |
| 14:48.42 | ManxPower | bjohnson: that is correct. |
| 14:49.05 | ManxPower | bjohnson: I don't have an unlimited number of paths to a destination 8-) |
| 14:49.16 | bjohnson | but you might have more than 2 |
| 14:49.26 | pussfeller | rcam, a company shouldnt be using gui tools to set up their communications, they need to understand how to do it themselves, by hand |
| 14:49.26 | ManxPower | bjohnson: I feel that if your providers are THAT unreliable then you have problems no macro can solve. |
| 14:50.05 | ManxPower | bjohnson: I have considered adding support to macro-dial-result for more than one backup destination |
| 14:50.05 | bjohnson | I'm also thinking about it's application in a followme call system or ringing multiple extensions within an office in sequence |
| 14:50.40 | bjohnson | ManxPower: just noop() it back to the dialplan that called it .. the admin can then recall the macro with a different destination |
| 14:51.09 | bjohnson | or dump it to a different voicemail, or hangup, whatever |
| 14:51.25 | bprice20 | How do i force asterisk to use a=ptime:20 as opposed to a=ptime:30 in the sdp stream |
| 14:51.37 | *** join/#asterisk ChkDigit (~mike@static65-87-228-18.regina.accesscomm.ca) |
| 14:51.49 | *** join/#asterisk vooduhal (~cmcbee@64-18-104-139.adsl.catt.com) |
| 14:51.55 | bjohnson | like a subroutine call in a program usually returns back to the line from which it was called |
| 14:52.47 | ChkDigit | That would be a nice way to hang a program... |
| 14:55.07 | vooduhal | Could someone help me with a swig/perl issue. I'm trying to port ast_readstring over for use with res_perl and I'm having a bit of trouble writing the .xs file to run xsubpp to produce what I need to add to res perl. The prototype for ast_readstring is int ast_readstring(struct ast_channel *chan, char *s, int len, int timeout, int ftimeout, char *enders); |
| 15:00.44 | bprice20 | How do i force asterisk to use a=ptime:20 as opposed to a=ptime:30 in the sdp stream? |
| 15:01.43 | boch | extension 's' will be always executed for an incoming call on a context? |
| 15:01.54 | *** join/#asterisk Aze` (~aze@host229-162.pool80105.interbusiness.it) |
| 15:02.18 | ChkDigit | boch: Not necessarily |
| 15:02.19 | sudhir492 | boch: not really |
| 15:02.32 | boch | i cant figure out how it works |
| 15:03.17 | sudhir492 | the best strategy is to use some variant of Goto(context,s,1) |
| 15:03.27 | ChkDigit | If there are other extensions that match the incoming call (given extension via SIP or H323, or pattern matching on CallerID), then it can run another extension. |
| 15:04.59 | Meaty | I want show ib a web page, how channel is presently actived on asterisk, whats is the best way to do that ? |
| 15:05.19 | Meaty | Can i do a pipe on asterisk ? |
| 15:05.25 | ChkDigit | www.asternic.org |
| 15:05.45 | ChkDigit | or the Asterisk Manager interface. |
| 15:06.10 | Meaty | kk |
| 15:06.22 | *** join/#asterisk loud (~ariel@blaqhat.com) |
| 15:09.11 | *** join/#asterisk Koshatul (~evangelio@ip157.net65.ipnetworks.net.au) |
| 15:09.59 | El^Diablo | When I get static on my line, I seem to get this message in my logs (several times a second) May 12 11:09:02 NOTICE[5037]: PRI got event: HDLC Abort (6) on Primary D-channel of span 1. Any ideas? |
| 15:10.28 | FITA1 | Does any one use app_meetme2(coferencing), and dial out to invite a friend |
| 15:10.53 | Juggie | app_meetme2? |
| 15:10.56 | Juggie | i've never heard of that |
| 15:11.02 | Meaty | .. |
| 15:11.14 | Juggie | not in code form anyway |
| 15:11.18 | Juggie | i heard it talked about |
| 15:11.58 | Juggie | as for inviting a friend |
| 15:12.00 | Juggie | thats easy |
| 15:12.18 | Juggie | you just need to use agi or a .call file to generate a call which throws you into the context which has code to join a converence |
| 15:12.41 | Juggie | if you want to make it dynamic, set a variable in your call generation that tells asterisk what conf room to join. |
| 15:13.16 | *** join/#asterisk A-Tuin|work (~A-Tuin@nat.office.legend.net.uk) |
| 15:13.52 | *** join/#asterisk Maveric (maveric@ip68-3-248-136.ph.ph.cox.net) |
| 15:15.16 | *** join/#asterisk |dennis| (~dennis@200.32.197.2) |
| 15:16.11 | |dennis| | question: i just downloaded mpg123-0.59r, compiled it and installed it. I then enabled musiconhold in my zaptel.conf. When i put a call on hold..all i hear are static ounds..... any help??? |
| 15:16.42 | *** join/#asterisk jmacz (~jmacz@63.245.86.229) |
| 15:16.45 | ManxPower | |dennis|: have you confirmed that your removed and previous version of mpg123? |
| 15:17.12 | El^Diablo | nobody has any ideas on HDLC Aborts? |
| 15:17.17 | ManxPower | El^Diablo: Yes. |
| 15:17.41 | |dennis| | i never had any previous install of mpg.....it was a fresh install of sarge..i installe dmpg321 first..but same static sound..so i removed mpg321 and installed mpg123... |
| 15:17.50 | rcam | Are there any issues with Debian Woody and the lastest version of *? |
| 15:17.52 | ManxPower | El^Diablo: It means "we got corrupted data from the card". The most common reason is some device locking interrupts for too long. You read the mailing list archives for the common fixes? |
| 15:18.56 | El^Diablo | Yeah. They talked about shared interrupts, but it it is on it's own interrups and it isn't showing any interrupt errors. |
| 15:19.16 | rcam | Something is seriously wrong with http://www.asternic.org/ |
| 15:19.19 | El^Diablo | It also talked about clock slip (presumably due to the interrupts), but I didn't see any way of testing that directly |
| 15:19.21 | *** join/#asterisk ilium007 (~brantwint@220-253-92-177.QLD.netspace.net.au) |
| 15:19.33 | ManxPower | ~google site:lists.digium.com hdparm HDLC |
| 15:19.43 | ManxPower | ~google site:lists.digium.com hdparm abort |
| 15:19.50 | ManxPower | ~google site:lists.digium.com hdparm |
| 15:20.13 | Meaty | :o |
| 15:20.20 | ManxPower | El^Diablo: IDE, RAID, graphics all commonly cause this problem as well. |
| 15:20.45 | ManxPower | So do just crappy chipsets. I've run into motherboards that just won't work without massive problems with HDLC abots |
| 15:21.06 | El^Diablo | It could be a crappy chipset, I'll have to see which one it is using |
| 15:21.10 | |dennis| | ManxPower: i never had any previous install of mpg.....it was a fresh install of sarge..i installed mpg321 first..but same static sound..so i removed mpg321 and installed mpg123... |
| 15:22.01 | ManxPower | |dennis|: what is the output of "ps -ax | grep mpg123"? |
| 15:22.13 | Sato1 | now the error changes |
| 15:22.14 | Sato1 | chan_zap.c: In function `zt_handle_event': |
| 15:22.14 | Sato1 | chan_zap.c:3084: `ZT_EVENT_DTMFDIGIT' undeclared (first use in this function) |
| 15:22.39 | ManxPower | El^Diablo: modern server chipsets are commonly pretty crappy when it comes to interrupt latency |
| 15:22.46 | |dennis| | ManxPower root 17246 0.0 0.3 3764 1576 ? S 08:58 0:00 mpg123 -q -s --mono -r 8000 -b 2048 -f 4096 fpm-calm-river.mp3 fpm-sunshine.mp3 fpm-world-mix.mp3 |
| 15:22.47 | Sato1 | it also said: (Each undeclared identifier is reported only once for each function it appears in.) |
| 15:23.07 | |dennis| | ManxPower: and there is no call in progress.. |
| 15:23.31 | ManxPower | |dennis|: mpg123 is always running |
| 15:23.46 | |dennis| | oh okiee...just wondering...but yeah..thats the output...:) |
| 15:24.08 | ManxPower | |dennis|: do a "find / -name mpg123 -print" moh sound rpoblems are almost ALWAYS issues with some oddball version of mpg123 laying around somewhere. |
| 15:24.53 | |dennis| | /usr/src/mpg123-0.59r/mpg123 |
| 15:24.53 | |dennis| | /usr/local/bin/mpg123 |
| 15:25.14 | ManxPower | |dennis|: Weird. No idea what the problem is. It should work. |
| 15:26.08 | |dennis| | weird....do i need any sound drivers or anything on the server installed? by any chance? or do i even need a sound card on the server? i would not think so but you never know...??? |
| 15:26.22 | ManxPower | |dennis|: nope |
| 15:26.42 | |dennis| | thanks...shall see what else i can do....thanks anyways...:) |
| 15:27.17 | *** join/#asterisk jief- (~jief@modemcable196.182-80-70.mc.videotron.ca) |
| 15:28.09 | jief- | hello guys, i just received my TDM400P with 4 FXO ports. i modprobe'd zaptel, and now i try to modprobe wcfxo. but i get unresolved symbols. is there another module i should load also? |
| 15:28.43 | tzanger | jief-: wctdm |
| 15:28.53 | tzanger | wcfxo is for the X100/X101 cards IIRC |
| 15:29.09 | jief- | tzanger: i dont have that module. its not part of CVS |
| 15:29.15 | ManxPower | tzanger: it's still wcfxs in 1.0.x |
| 15:29.24 | ManxPower | 1.0.x MAY have an alias for wctdm |
| 15:29.33 | tzanger | oh okay I didn't realize it was 1.0.x |
| 15:29.43 | ManxPower | tzafrir: I don't know what verison he's running. |
| 15:29.53 | jief- | tzanger: wouldn't i load wcfxo if i only have FXO ports? |
| 15:30.07 | cpatry | jief-: no FXS |
| 15:30.11 | jief- | im running 1.0.7 |
| 15:30.12 | ManxPower | jief-: no. you would still load wcfxs or wxtdm. Yes, the module name is confusing. |
| 15:30.18 | cpatry | wcfxs is for FXO ports. |
| 15:30.28 | ManxPower | jief-: if you had read the README in the zaptel directory you would know this. |
| 15:30.29 | jief- | crack is bad for developers ... |
| 15:30.29 | cpatry | its always the opposite. |
| 15:30.55 | cpatry | jief-: crack is bad for which arent reading all the infos here and there too huh? |
| 15:30.58 | cpatry | :) |
| 15:30.59 | ManxPower | jief-: you want the REAL answer? read the README in the zaptel source directory? |
| 15:31.05 | jief- | ManxPower: our developer built a RPM for the drivers, and i dont have the README with me |
| 15:31.22 | jief- | alright, ill get the source |
| 15:31.25 | cpatry | get the src from cvs, not package. |
| 15:31.33 | ManxPower | jief-: go to your developer and smack him upside the head |
| 15:31.44 | cpatry | and tell him to STOP CRACK too! |
| 15:32.10 | ManxPower | jief-: and say to him "next time give me the docs, asshole!" |
| 15:32.25 | jief- | hehe |
| 15:32.47 | cpatry | tell him ya had fun with his mother yesterday night too. :) |
| 15:33.04 | jief- | so, now that i read the README and know i need to load wcfxs. what could prevent it from loading? does it require any other modules? |
| 15:33.36 | Juggie | if i add someone to a meetme conference as listen only, i cant unmute them is that correct? |
| 15:34.06 | BerndR | has anyone patched rtp.c for improving the problem of jittering while doing sip calls? |
| 15:34.12 | BerndR | http://lists.digium.com/pipermail/asterisk-dev/2004-January/002902.html |
| 15:34.24 | BerndR | does this help? |
| 15:34.31 | cpatry | Juggie: guess so. |
| 15:34.33 | *** join/#asterisk malcolmd (~malcolmd@malcolmd.digium.sponsor.pdpc) |
| 15:34.37 | malcolmd | bleh, network card died |
| 15:35.10 | *** join/#asterisk Seyr (~Seyr@rrcs-24-227-133-226.sw.biz.rr.com) |
| 15:35.53 | *** join/#asterisk jsharp (~jsharp@65.90.64.82) |
| 15:36.30 | jsharp | Can asterisk, with chan_skinny or chan_sccp pretend to be a skinny phone or is it a call manager only? |
| 15:37.13 | ManxPower | "There is some sort of wow factor to the 35-pound cat in your Manhattan apartment." |
| 15:37.14 | ManxPower | oops |
| 15:37.22 | jsharp | MEOW |
| 15:38.18 | airwolf | how to make call from Asterisk to a Inphonex user? |
| 15:38.51 | airwolf | sip.inphonex.com |
| 15:42.08 | *** join/#asterisk malcolmd (~malcolmd@malcolmd.digium.sponsor.pdpc) |
| 15:45.12 | *** part/#asterisk loud (~ariel@blaqhat.com) |
| 15:48.00 | *** join/#asterisk |dennis| (~dennis@200.32.197.2) |
| 15:48.29 | *** join/#asterisk loutux (oooooooo@200.124.234.228) |
| 15:48.41 | loutux | hi everybody! |
| 15:49.07 | |dennis| | ManxPower: could it be because I dont have any zaptel ahrdware. all i did was compile and install asterisk. I have not installed ztdummy. Do i have to or is it part of asterisk? |
| 15:49.26 | Seyr | Whhat would cause audio to crackle(??) a little from an * server? Im connecting with a phone or softphone to it and every now and then it crackles a bit |
| 15:49.33 | Seyr | using GSM |
| 15:50.17 | Juggie | Seyr, sharing irq's |
| 15:50.31 | Juggie | if you are using a zap board make sure its not sharing an irq with anything |
| 15:50.34 | *** join/#asterisk Cresl1n (~matt@216.207.245.23) |
| 15:50.38 | Seyr | not using one |
| 15:50.52 | Seyr | using IAX |
| 15:51.50 | *** join/#asterisk angler_ (~angler@suid.digium.com) |
| 15:52.19 | Seyr | brand new dell 1850 server with FC3 and * installed, no other apps except pre-reqs and base FC3 install. box is not behind a firewall. audio crackles just a bit. enough to notice |
| 15:52.36 | Juggie | Seyr, then check for irq sharing |
| 15:52.41 | Juggie | something sharing irq with network c ard |
| 15:52.41 | Juggie | etc |
| 15:53.17 | Seyr | thanks, checking |
| 15:53.32 | Seyr | its got 3 NICS in it. 2 onboard and one PCI |
| 15:53.44 | Seyr | i didnt order it.... |
| 15:54.02 | Juggie | hah |
| 15:54.05 | Juggie | disable any you arnt using |
| 15:54.16 | Juggie | via bios or whatever |
| 15:54.18 | Juggie | remove the pci one even |
| 15:55.02 | Seyr | thanks |
| 15:55.22 | Seyr | i needed a reason to get out of the office. going to the datacenter to do that is the perfect excuse :-) |
| 15:55.51 | Seyr | almost ready to start ordering sip phones for it |
| 15:56.58 | Seyr | i have the menus all done and have it talking via SOAP to our windows boxes (for db, etc). just starting to work on using phones with it |
| 15:59.03 | *** join/#asterisk NewSole (dave@i216-58-44-245.avalonworks.net) |
| 16:00.45 | denon | so http://www.wendys.com/w-1-0.shtml |
| 16:02.52 | darwin35 | BV is down again |
| 16:03.05 | loutux | hi! |
| 16:03.15 | loutux | what are you talking about? |
| 16:03.30 | |dennis| | ManxPower..it works now..just needed a reboot....now..how do i increase the volume? |
| 16:03.39 | |dennis| | on music on hold? |
| 16:05.35 | *** join/#asterisk fefede (~fna@r200-125-63-176-dialup.adsl.anteldata.net.uy) |
| 16:05.46 | fefede | hi all |
| 16:06.31 | Grooby | Darwin, i am looking for new providers now |
| 16:06.38 | Grooby | BV is just unacceptable |
| 16:07.36 | Seyr | Grooby: For your * box? |
| 16:07.49 | Grooby | yeah |
| 16:07.57 | Seyr | Asterlink is who I am using |
| 16:08.00 | Seyr | so far 0 problems |
| 16:08.04 | Seyr | IAX |
| 16:08.06 | Grooby | www.asterlink.com? |
| 16:08.10 | Seyr | yeh |
| 16:08.23 | Stereo | hi folks |
| 16:08.49 | Grooby | i don't see residential |
| 16:08.56 | Grooby | you using it for business? |
| 16:09.17 | Stereo | I'm looking for a company that can provide me with a number a POTS user can call. Except this is in Luxembourg. Any ideas of where to start looking? |
| 16:09.18 | Romik | somebody uses via c3 processors? on which freq. possible to use it without fan? |
| 16:09.45 | Grooby | 800mhz is the highest w/ out fan |
| 16:09.54 | Grooby | else you have to get those special heatsink case |
| 16:10.11 | Romik | grooby: i have 798000 kHz (100 %) it very hot without fan! |
| 16:10.28 | Seyr | Grooby: yeh |
| 16:10.31 | Romik | grooby: when i stop the fan |
| 16:11.21 | Seyr | Grooby: should just call and tell em what you need. |
| 16:11.48 | fefede | i am try to get working some ipphone's... I have some problems in the codec negotiation... How can I do, to make asterisk don't interfere in the negotiation?? |
| 16:12.07 | fefede | I like that the end point resolve de codec to use....! |
| 16:12.28 | Grooby | ok... |
| 16:12.40 | Grooby | i am looking for something cheap for local in US |
| 16:12.50 | Grooby | i already sign up for Teliax pay as I go plan |
| 16:12.57 | Grooby | where it's only for outbound international calls |
| 16:13.23 | Sato1 | www.soyo.com |
| 16:13.40 | Sato1 | they have now voip-plans |
| 16:14.04 | Sato1 | connect.voicepulse.com also pay as you go |
| 16:14.28 | Grooby | i'll have to look into that |
| 16:14.56 | Seyr | Asterlink is flat 2 cents a minute, I believe. no fee and a toll-free # |
| 16:15.26 | Sato1 | fwd has toll-free |
| 16:15.55 | Sato1 | free accounts, and you may dial to many other providers for free |
| 16:18.21 | *** join/#asterisk _omer (~dfsdf@202.147.174.178) |
| 16:18.28 | _omer | hi |
| 16:18.55 | *** join/#asterisk BoRiS (boris@S01060040ca1e5b54.wp.shawcable.net) |
| 16:19.13 | *** join/#asterisk juice (~juice@mo-69-68-105-244.dyn.sprint-hsd.net) |
| 16:21.55 | _omer | how to take the backups of Asterisk configuration to protect is from accidental crash... |
| 16:22.04 | _omer | how to take the backups of Asterisk configuration to protect it from accidental crash... |
| 16:22.42 | Romik | omer: burn it on CD |
| 16:22.48 | cpatry | _omer: i would say isnt an asterisk question, its a *nix question. |
| 16:23.29 | _omer | Romik: which folder? |
| 16:23.33 | *** join/#asterisk outtolunc (~me@adsl-66-218-53-170.dslextreme.com) |
| 16:23.39 | Romik | -> /etc/asterisk/ |
| 16:23.47 | _omer | alright.....thanks :) |
| 16:24.07 | tzanger | tar -czvf /tmp/asteriskconfig-20050512.tgz /etc/asterisk /etc/zapata.conf |
| 16:24.20 | tzanger | and if you have custom sounds or voicemail /var/spool/asterisk |
| 16:24.26 | *** join/#asterisk bprice20 (~brandon@Dynamic-216.120.224.151.hrnoc.net) |
| 16:25.03 | _omer | thanks tzanger. |
| 16:26.00 | *** join/#asterisk R3DB0x (nobody@66.142.28.36) |
| 16:26.29 | pepzi | I get empty sip reads every 10 seconds.. Sip read:\n\n0 headers, 0 lines .. why? i have two eyebeam-softphones connected |
| 16:29.22 | ManxPower | ~google site:lists.digium.com eyebeam |
| 16:32.14 | *** join/#asterisk HeadachesAbound (HeadachesA@wsip-68-99-73-32.tu.ok.cox.net) |
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| 16:34.44 | *** mode/#asterisk [+o drumkilla] by ChanServ |
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| 16:35.33 | *** part/#asterisk Moc_ (~mochouina@h66-201-214-109.gtconnect.net) |
| 16:35.51 | ksmloh | !list |
| 16:37.35 | *** part/#asterisk ksmloh (~asd@bb220-255-174-59.singnet.com.sg) |
| 16:38.03 | *** join/#asterisk moy (~kvirc@dsl-200-95-105-36.prod-infinitum.com.mx) |
| 16:41.00 | jsharp | launch |
| 16:41.07 | *** join/#asterisk cmk (~cmk_@p54A3BB6C.dip.t-dialin.net) |
| 16:43.05 | outtolunc | so, anyone know why if you use Background(file) and Read() right behind it.. if while Background() is playing you hit digits.. it *think* isn't an extension and tries to dial <G> when all you want is Read() to get the digits <G> |
| 16:44.13 | ManxPower | outtolunc: because you should not use Background if you don't want to accept DTMF during playback. Use Playback |
| 16:44.51 | outtolunc | what i wanted was the 'exit on dtmf' |
| 16:45.04 | outtolunc | so that it stopped playing |
| 16:45.35 | outtolunc | <G> |
| 16:45.53 | bkw_ | Dev meeting today 1pm CST IAX2/guest@switch-3.asterlink.com/996 |
| 16:46.01 | outtolunc | oh shit that's right |
| 16:46.09 | *** join/#asterisk wasabi_ (~wasabi@207.55.180.100) |
| 16:46.15 | outtolunc | man this week has gone by fast |
| 16:46.19 | wasabi_ | Anybody able to recommend a voip phone that works perfectly with asterisk? |
| 16:46.21 | jontow | i've got a loop caused by a transfer.. :o and all sorts of semi-cosmetic problems with my PRI on freebsd |
| 16:46.41 | jontow | oh i need to fix that |
| 16:47.34 | *** topic/#asterisk by bkw_ -> Asterisk: The Open Source PBX || Astricon Europe -- Madrid, Spain -- June 15-17 || ClueCon Dev Conf Aug 3rd - 5th - http://www.cluecon.com (Registration Open Now) || Dev meeting today 1pm CST IAX2/guest@switch-3.asterlink.com/996 |
| 16:48.01 | jontow | what the hell.. |
| 16:48.18 | jontow | << [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/132-d98e] |
| 16:49.32 | *** part/#asterisk Seyr (~Seyr@rrcs-24-227-133-226.sw.biz.rr.com) |
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| 17:05.54 | *** join/#asterisk hypa7ia (~leigh@8067a4e99f1c1022.node.tor) |
| 17:05.56 | *** join/#asterisk zotz (~zotz@24.231.32.109) |
| 17:07.46 | h3x0r | jonas: wheres the freebsd zaptel |
| 17:08.01 | h3x0r | jontow |
| 17:08.03 | h3x0r | i mean |
| 17:08.03 | h3x0r | heh |
| 17:08.25 | *** join/#asterisk terrapen (~cjs@fw-01.satx.bikeworld.net) |
| 17:08.27 | terrapen | heh |
| 17:08.29 | terrapen | http://j-walk.com/other/googlecb/index.htm |
| 17:10.34 | _omer | anybody who have used ASTERISKWIN32? |
| 17:10.50 | *** join/#asterisk ChulJin (~chuljin@adsl-68-121-94-237.dsl.irvnca.pacbell.net) |
| 17:10.56 | ChulJin | Good morning Gentlemen! |
| 17:11.03 | *** join/#asterisk rcam (~rcammobil@adsl-218-151-77.jax.bellsouth.net) |
| 17:11.05 | ChulJin | W00T! on several levels... |
| 17:11.16 | *** join/#asterisk Xander77 (~Alex@exten-halls-243.soton.ac.uk) |
| 17:11.22 | Romik | somebody can help to with expression that replace / to - in the string in asterisk language |
| 17:11.37 | bkw_ | show functions |
| 17:11.39 | bkw_ | regexp? |
| 17:12.09 | rcam | Where does lynx save downloads by default? |
| 17:12.10 | *** join/#asterisk fidsap (~fidsap@213.199.2.66) |
| 17:12.57 | ChulJin | PWD before launching Lynx, I believe |
| 17:13.20 | rcam | ChulJin PWD? |
| 17:13.28 | ChulJin | your working directory |
| 17:13.31 | rcam | I see. |
| 17:13.35 | rcam | Thanks. |
| 17:13.49 | *** part/#asterisk cpatry (~grepmoo@65.39.228.5) |
| 17:14.38 | rcam | I tried to find a guide to do a remote install of Asterisk@Home or any distro for that matter... Any ideas of where to go? |
| 17:14.49 | Romik | bkw: i can't find way to use regexp on this? ther regexp to extract not to replace :( |
| 17:16.04 | jontow | it is a T100P running freebsd-zaptel-0.9 from latest ports |
| 17:18.29 | *** join/#asterisk harryvv (~leonardo@S010600a0c93f6f7e.vs.shawcable.net) |
| 17:18.54 | hypa7ia | rcam, onto what OS? |
| 17:19.50 | *** join/#asterisk shmaltz (~chatzilla@ool-43551098.dyn.optonline.net) |
| 17:20.02 | bjohnson | rcam: yes. you need something to connect to first .. ie a ssh daemon |
| 17:20.14 | shmaltz | does callpickup (features.conf) work for queues? |
| 17:22.03 | *** join/#asterisk durex (~ironman@weber.anpa.org.br) |
| 17:22.06 | durex | asterisks.... |
| 17:22.09 | rcam | hypa7ia bjohnson I have a Debian box running now. |
| 17:22.33 | rcam | hypa7ia bjohnson I am considering leaving it as is an just installing all of the software... But I like the ease of A@H. |
| 17:22.42 | rcam | and* |
| 17:22.42 | durex | I'm trying to authenticate to a VoIP provider via ISP. I have to specify realm to this authentication. Where do I specify realm on registry line on sip.conf? |
| 17:23.00 | durex | sorry, via SIP |
| 17:23.08 | harryvv | * ports are all udp right? |
| 17:23.17 | hypa7ia | well, rcam you can just apt-get install asterisk |
| 17:23.26 | hypa7ia | that won't give yuo the latest version tho |
| 17:23.26 | rcam | hypa7ia I did. |
| 17:23.29 | shmaltz | durex, I beleive asterisk should be smart enough to do this alone |
| 17:23.31 | rcam | hypa7ia I know. |
| 17:23.41 | rcam | hypa7ia I need AMP installed as well. |
| 17:23.47 | durex | shmaltz I have to change the realm to other one than the usual... |
| 17:23.49 | *** join/#asterisk PCadach (~paul@www.east.telecom.kz) |
| 17:23.51 | hypa7ia | rcam, so you wantt he latest version, is that the issue? or is it AMP that you're looking to install |
| 17:23.52 | rcam | hypa7ia No package for that. |
| 17:24.02 | shmaltz | durex, then check the wiki |
| 17:24.03 | hypa7ia | well then either make one or install it from scratch |
| 17:24.08 | rcam | hypa7ia AMP. |
| 17:24.22 | rcam | hypa7ia I only know my way around... Doing all of that takes me a lot of time. |
| 17:24.40 | rcam | hypa7ia Which is why I was considering A@H. |
| 17:24.59 | shmaltz | durex, http://www.voip-info.org/wiki-Asterisk+config+sip.conf |
| 17:25.13 | terrapen | are there any EEs around? |
| 17:25.15 | hypa7ia | those are your only options as far as i know... you could always do a fresh install ofA@H |
| 17:25.17 | BoRiS | harryvv: yes |
| 17:25.47 | shmaltz | what a schmuck, askes a questions, I try to help him, no thanks and when I want to give him the answer he is gone. |
| 17:26.04 | shmaltz | durex, don't ask *me* again |
| 17:26.44 | rcam | hypa7ia Which is what I was thinking... But it's a remote machine... I have never done a remote linux install. |
| 17:26.51 | terrapen | every CRT monitor in my building is wobbling very severely |
| 17:27.11 | shmaltz | terrapen, are they watching some porn on it? |
| 17:27.15 | harryvv | Boris never mind :) I was doing some firewall test and got a report that iax2 was blocked but just made a two way iax call so obviosly that was not the case |
| 17:27.16 | rcam | hypa7ia I downloaded the ISO with Lynx. |
| 17:27.23 | terrapen | uhh..no |
| 17:27.25 | outtolunc | probably because the nuclear reactor near you is leaking <G> |
| 17:27.37 | terrapen | i think its the power company |
| 17:27.40 | terrapen | but im not sure |
| 17:27.46 | shmaltz | so call the power company |
| 17:27.59 | Himeko | i get the wobble in my current house sometimes |
| 17:28.00 | shmaltz | if it's a power issue, they will get ruined |
| 17:28.07 | harryvv | wobble? |
| 17:28.20 | shmaltz | gtg guys |
| 17:28.22 | shmaltz | ce ya |
| 17:28.24 | harryvv | you mean a power drop |
| 17:28.25 | Himeko | it's even plugged into a apc smart-ups |
| 17:28.51 | *** join/#asterisk netofsickcoder (~netofsick@200.121.129.178) |
| 17:28.52 | outtolunc | apc smartups does not have isolation (at least any worth a damn) |
| 17:29.29 | Himeko | well, if i unplug it and it is runnign off battery and it still wobbles it is certaily isolated |
| 17:29.29 | ChulJin | NB to all (cf. my complaints yesterday): I can now recommend telasip. |
| 17:29.37 | ChulJin | fromuser= will always git ya |
| 17:29.39 | hypa7ia | rcam, at that point you're talking about a chrooted install... i don't know how to do that. i'd go with putting in AMP manually. |
| 17:29.43 | harryvv | I worked at a company that had a Fiery print server hooked to a power strip. Some kind of power surge or blackout was not enough for that strip to stop it and wipped out the OS. Company presidents prior network manager obviosly should have thought more of that print server. |
| 17:29.57 | outtolunc | isolation and isolated aren't really the same thing <G> |
| 17:30.08 | hypa7ia | does anyone have experience with whitebox vs. CentOS as a RedHat clone platfor for asterisk? |
| 17:30.11 | ChulJin | (success 1) |
| 17:30.15 | outtolunc | and if it's happening on batt power, it's either the monitor itself or the ups <G> |
| 17:30.22 | ChulJin | success 2: I finally got RxFax working |
| 17:30.27 | harryvv | hypa, it works great in fedora core 3 |
| 17:30.34 | Himeko | or weird em fields |
| 17:30.52 | outtolunc | that or a magnetic field like i mentioned before <G> |
| 17:31.09 | outtolunc | remember the nuclear plant <G> |
| 17:32.16 | *** join/#asterisk durex (~ironman@weber.anpa.org.br) |
| 17:32.53 | hypa7ia | harryvv, i'm eventually going to be putting it on real supported redhat, so i was looking for something closer.. the two big ones seem to be centOS (that's what A@H runs on) and Whitebox |
| 17:33.15 | harryvv | fedora core 3 is close. It has the red had embelem near the top. |
| 17:34.31 | *** join/#asterisk jjhall (~chatzilla@24-119-114-94.cpe.cableone.net) |
| 17:35.22 | hypa7ia | lol, harryvv, i know that, it's less binary-compatible though as i understand |
| 17:35.31 | jjhall | anyone know why an entry in hosts.conf wouldn't be picked up? |
| 17:35.32 | *** join/#asterisk bannerman (~bannerman@209.216.176.43) |
| 17:35.40 | *** join/#asterisk tandrews (~tandrews@mail.grok.co.za) |
| 17:35.46 | jjhall | Thinking of 2 things at once. :-) |
| 17:35.56 | tandrews | hello :) |
| 17:35.57 | jjhall | I mean /etc/hosts |
| 17:36.32 | harryvv | hypa, dont know but..if you have a althlon 64 do not use it as a work station. You miss out on alot of x86_64 aps |
| 17:37.45 | Romik | "Zap/23" : "/Zap/huyap/" this expression will replace Zap to huyap in the asterisk? |
| 17:38.17 | sivana | ariel_: ping |
| 17:38.37 | ariel_ | hello sivana |
| 17:38.41 | sivana | hey :) |
| 17:39.12 | hypa7ia | harryvv, that's good to know. thanks! |
| 17:40.11 | tandrews | um |
| 17:41.25 | tandrews | what exactly is a "wink" ? As in "rxwink" time in zapata.conf ? |
| 17:41.50 | durex | folks, |
| 17:42.03 | durex | having problem to specify a realm to a registry in sip.conf |
| 17:42.18 | *** join/#asterisk Thus0 (~Thus0@dyn-83-157-176-138.ppp.tiscali.fr) |
| 17:44.14 | Romik | anybody can help me with regext in asterisk? "Zap/23" : "/Zap/huyap/" this expression will replace Zap to huyap in the asterisk? |
| 17:45.49 | *** join/#asterisk MajestiK (~MajestiK@S01060800208687ec.ed.shawcable.net) |
| 17:45.58 | *** join/#asterisk heison (~heison@ns.somanetworks.com) |
| 17:46.22 | tandrews | Dumb question - what is "huyap" ? |
| 17:46.53 | *** join/#asterisk Gh0sty (~Ghosty@ip-81-11-209-19.dsl.scarlet.be) |
| 17:48.24 | Romik | tandrews: just a sample |
| 17:50.15 | *** join/#asterisk L|NUX (~linux@202.5.145.54) |
| 17:50.33 | *** join/#asterisk comintel (~tom@spc1-hava1-4-0-cust62.cosh.broadband.ntl.com) |
| 17:51.00 | comintel | whats the command to show the dial statement |
| 17:51.10 | tandrews | Romik: I use agi to munge stuff like that - that way I can use sqlite in the equation |
| 17:52.41 | *** join/#asterisk vpp (~noone@host-83-146-50-131.bulldogdsl.com) |
| 17:52.43 | vpp | hi guys |
| 17:52.52 | *** join/#asterisk rcam (~rcammobil@adsl-218-151-77.jax.bellsouth.net) |
| 17:53.02 | vpp | my asterisk is stuck on 'Loading zaptel hardware modules:' |
| 17:53.09 | vpp | any idea's? |
| 17:54.45 | jontow | would "PRI got event: HDLC Bad Fcs (8) on Primary D-channel of span 1" be caused by using regular ethernet-spec'd cat5 instead of a t1 cable? |
| 17:54.50 | jontow | or negligible overall? |
| 17:55.38 | Romik | tandrews: no way to make simple regular expression replacement? |
| 17:56.31 | *** join/#asterisk marky (emes@65.114.80.8) |
| 18:02.53 | ManxPower | Does Europe use the same battery sizes as the USA? AA, AAA, C, D, etc? |
| 18:03.05 | ManxPower | Romik: see README.variables |
| 18:03.08 | ManxPower | or use an AGI |
| 18:03.26 | RaYmAn-Bx | ManxPower: those definitely exist at least..whether it's the same size I have no idea.. |
| 18:03.33 | RaYmAn-Bx | They also generally have alternative names on them |
| 18:03.44 | marky | anybody care to point me in a direction |
| 18:03.54 | ariel_ | ManxPower, yes they have the same type of battery |
| 18:03.57 | marky | i've been looking at the tutorial for asterisk and broadvoice |
| 18:04.09 | marky | i've got a voip account with my ISP and i'm not getting it to work |
| 18:04.19 | ariel_ | marky, bv seems to be down allot right now. |
| 18:04.32 | marky | i don't have them....just referencing that tut |
| 18:04.36 | wisdom | they've had a hard time keeping up with customer demand |
| 18:04.41 | ManxPower | ariel_: I thought so, but it would be VERY annoying to arrive and then find out they don't |
| 18:05.20 | harryvv | hi Manx, To your knowledge is there a asterisk context that will echo back a different dtmf tone if a caller calls a extension? May be a usefull selling feature to anyone who needs access to a dtmf solinoid door access. |
| 18:05.20 | ariel_ | ManxPower, the biggest difference are the electrical plugs. |
| 18:05.28 | marky | i'm just looking at getting a calling in/calling out system that I can mess with |
| 18:05.38 | marky | i don't deal with creating situations in my head |
| 18:05.52 | ManxPower | ariel_: got that dealt with |
| 18:06.14 | marky | works best if i just put it together and get it working |
| 18:06.14 | Romik | manxpower: AA and AAA is for sure |
| 18:07.33 | harryvv | actually what may work is a recording of the dtmf and use that as playback |
| 18:09.36 | DrWho17 | marky: I've been using voipjet for 1 month, no troubles, their web interface sucks, their CDR's aren't quite right, but the voice service has been fine |
| 18:10.49 | Romik | manxpower: still do not undestand how to make regexp to replace Zap to Sip for ex: Zap/23 to Sip/23 ? |
| 18:11.43 | marky | can anybody point me a tutorial that might explain the trunks and whatnot |
| 18:11.43 | vaewyn | JerJer: got a customer asking me if their nufone account will let them call Canadian 800 numbers... I am assuming no... correct? |
| 18:11.46 | Romik | manxpower: do not forget in europe everywhere is 220-230 volts...most euipments in states is 110v |
| 18:11.52 | *** join/#asterisk Moc_ (~mochouina@h66-201-214-109.gtconnect.net) |
| 18:11.56 | marky | i just printed off the asterisk handbook v2......so that may or may not help |
| 18:11.59 | marky | i've got service |
| 18:12.13 | RaYmAn-Bx | ManxPower: when you say dealt with, I assume you know that most European countries have different electricity sockets? (but most at least has the same voltage at between 220 and 240) |
| 18:12.21 | harryvv | ManxPower: where are you going |
| 18:12.38 | harryvv | I was going to say that |
| 18:12.39 | harryvv | :) |
| 18:12.52 | harryvv | also the frequency is different |
| 18:13.01 | *** join/#asterisk r0d3nt|m (~RatMan@wsip-24-234-241-84.lv.lv.cox.net) |
| 18:13.02 | harryvv | 50 hz vs 60 hz here |
| 18:13.25 | *** join/#asterisk Trionnis (buffy@12-203-113-15.client.insightBB.com) |
| 18:13.32 | vpp | if i have a T1 card setup in span 1 |
| 18:13.35 | harryvv | ManxPower: call radio shack thay may still sell frequency/voltage conversion adapters. |
| 18:13.37 | RaYmAn-Bx | I think it's 50hz here..but most things you buy here can take both 50 and 60 (and a lot can take 110V as well) |
| 18:13.38 | vpp | what do i put in extensions to dial out of it? |
| 18:13.46 | vpp | zap/1 |
| 18:13.47 | vpp | ? |
| 18:14.30 | RaYmAn-Bx | I have a canadian iPAQ where the powersupply works just fine here with a simple converter that just makes the plug match up |
| 18:15.00 | marky | apple computer equipment is simalar as well |
| 18:15.05 | marky | $80 accessory kit... |
| 18:17.42 | vaewyn | vpp: zap/1 for channel 1 or zap/g1 to use next in span |
| 18:17.58 | vaewyn | assuming your span is setup as a group |
| 18:18.02 | vaewyn | which it should be |
| 18:18.32 | *** join/#asterisk zwhitley (~zwhitley@69-162-31-243.stcgpa.adelphia.net) |
| 18:19.32 | zwhitley | Any Digium employees here? |
| 18:19.45 | vpp | oh ok.. how do i set it up as a group? |
| 18:21.08 | vaewyn | vpp: group=1 in your zapata.conf |
| 18:21.25 | vaewyn | or group=2 if you want it to be group 2... etc... |
| 18:22.38 | *** join/#asterisk jjhall (~chatzilla@24-119-114-94.cpe.cableone.net) |
| 18:22.40 | Romik | regex make me crazy!!!! :( i can't find way to do it |
| 18:22.52 | vaewyn | zwhitley: There are... but they don't interact much... if you have a setup/config issue though feel free to ask and someone can try to help you |
| 18:23.19 | vpp | ok cool |
| 18:23.43 | vaewyn | vpp: if you used the zapata.conf.sample then it will already be there |
| 18:24.46 | zwhitley | I don't have any config questions. (at this moment) just wanted to mention something to a Digium employee that was directly relateed to Digium. |
| 18:25.04 | vaewyn | zwhitley: ahh :} |
| 18:25.05 | hardwire | how dare you |
| 18:25.55 | vaewyn | hardwire: because I use swisscheese for underwear?!? |
| 18:26.00 | vaewyn | :} |
| 18:26.10 | DrWho17 | Romik: download the file to an editor, and do a search and replace then |
| 18:26.11 | zwhitley | if there are any out there msg me, if not I'll just write an email. thanks. |
| 18:26.42 | vpp | Dial(zap/g1,50) |
| 18:26.45 | vpp | whats the 50 for? |
| 18:26.50 | vpp | is that right? |
| 18:26.57 | vaewyn | number of seconds to wait for answer |
| 18:27.00 | vaewyn | Yep |
| 18:27.01 | vpp | ahh ok |
| 18:27.02 | ManxPower | the 50 time to wait before giving up |
| 18:27.14 | Juggie | ManxPower, have any expirence with agi processes going defunt |
| 18:27.15 | vpp | ok cool |
| 18:27.15 | DrWho17 | vpp: digium has a manual on their site for basic questions |
| 18:27.16 | Trionnis | anyone have suggestions for a provider to replace broadvoice? |
| 18:27.18 | ManxPower | vpp: I assume these are FXS ports? |
| 18:27.22 | Juggie | but none the less still terminating and asterisk moving on |
| 18:27.29 | vaewyn | ManxPower: T span |
| 18:27.32 | Juggie | asterisk knows they are over, but they still hang around in the process list. |
| 18:27.33 | Trionnis | I kinda need the unlimited EU calling, and of course, asterisk support |
| 18:27.33 | Romik | <PROTECTED> |
| 18:27.34 | ManxPower | Juggie: only when I forget to have a callback handler for when people hang up. |
| 18:28.01 | Romik | manxpower: could you advice with regex replacement? |
| 18:28.02 | DrWho17 | Romik: ok, I thought you just wanted to batch change zap's to sip's |
| 18:28.07 | *** join/#asterisk joe (~jsauer@ip66-107-33-196.z33-107-66.customer.algx.net) |
| 18:28.12 | Juggie | ManxPower, none of my agi code does dialplan functionality just database, setting vars and stuff, how would i implement a call back handler? |
| 18:28.13 | ManxPower | Romik: only what is already in README.variables |
| 18:28.19 | vpp | ManxPower.. its a T1 card |
| 18:28.24 | vpp | PRI |
| 18:28.37 | Romik | manxpower: there no sample for : operator |
| 18:28.40 | ManxPower | vpp: So you don't need to dial a phone number or anything like that? |
| 18:28.51 | dalabera | Has anyone experience echo problems transferring calls from a legacy PBX into Asterisk? |
| 18:28.54 | ManxPower | Romik: I have never done REGEX unless it's in an AGI |
| 18:29.04 | vaewyn | Reminds me... anyone that is planning on hooking to a Norhell... use 5ess pri_net on your end or it won't let you dial anything but 'local' extensions |
| 18:29.04 | vpp | well it comes in SIP, goes out on the T1 |
| 18:29.10 | ManxPower | Juggie: The examples for asterisk-perl should give you the needed info. |
| 18:29.19 | ManxPower | See also the pastebin I'm creating in a moment |
| 18:29.20 | Romik | manxpower: any other way to replace substring inte string? |
| 18:29.36 | Juggie | yah i'm looking now, but my agi's dont do anything that rely on input from the user |
| 18:29.37 | vaewyn | s/Zap\//SIP\//g or the other way around |
| 18:29.40 | Juggie | so i dont see how they could be blocking |
| 18:29.42 | vaewyn | :} |
| 18:29.47 | Juggie | i'm not getting channels tied up |
| 18:30.03 | Juggie | i've seen that happen, where a zap channel doesnt terminate because the agi is still running |
| 18:30.10 | Juggie | but thats not happening in this case |
| 18:30.21 | ManxPower | Juggie: http://pastebin.ca/11603 |
| 18:30.23 | vaewyn | Juggie: anything on 'show channels' ? |
| 18:30.45 | Juggie | well i just restarted to make them go away, but i dont think so |
| 18:30.54 | vaewyn | hm... odd then |
| 18:30.58 | Juggie | i'll check again when some more processes build up |
| 18:31.04 | *** join/#asterisk Lee__ (~Lee__@cpe-69-203-211-144.nyc.res.rr.com) |
| 18:31.07 | Juggie | there were like 200 defuncts |
| 18:31.16 | Juggie | if they were trying up lines |
| 18:31.16 | DrWho17 | ouch |
| 18:31.20 | ManxPower | Juggie: using asterisk-perl? |
| 18:31.22 | Juggie | all my lines would have been held up |
| 18:31.24 | Juggie | no, phpagi |
| 18:31.30 | ManxPower | Juggie: I can't help you then |
| 18:31.37 | ManxPower | My pastebin is for asterisk-perl |
| 18:31.49 | Juggie | yah i know... its the same idea tho i'll check |
| 18:32.18 | *** join/#asterisk cmaj (~chris@65-37-6-42.nrp2.roc.ny.frontiernet.net) |
| 18:33.45 | cmaj | hello |
| 18:34.25 | *** join/#asterisk jets (~brian@guardian.pmt.org) |
| 18:36.34 | ManxPower | Juggie: And if you laugh at my code I will be forced to hurt you. 8-) |
| 18:37.47 | *** part/#asterisk PCadach (~paul@www.east.telecom.kz) |
| 18:39.21 | Juggie | ManxPower, i dont do perl :) |
| 18:40.33 | CyberKnet | Every man has had a go on Perl. How did you miss out? =) |
| 18:40.37 | vaewyn | ManxPower: actually not bad... if you ever turned use strict on though it would slap you down :} |
| 18:40.55 | sudhir492 | ManxPower: Its time you too moved from Perl to Python. |
| 18:41.15 | vaewyn | sudhir492: it's time python got over itself ;P |
| 18:41.23 | CyberKnet | hah |
| 18:41.53 | Lee__ | Perl is great. Python is great. Java isn't. there. it's settled :) |
| 18:41.58 | vaewyn | python was an answer to problems that perl use to have... |
| 18:42.06 | vaewyn | notice the past tense |
| 18:42.15 | vaewyn | Lee__: hehehe |
| 18:42.20 | sean | I love how I have 2 java apps running on my desktop, and also two full JVMs. how nice. |
| 18:42.24 | vaewyn | your new here arn't you? ;P |
| 18:42.45 | Lee__ | fairly. one month. |
| 18:42.53 | anthm | do what you want, just don't be dissing perl |
| 18:42.54 | Trionnis | anyone have suggestions for a provider to replace broadvoice? |
| 18:42.57 | Trionnis | I kinda need the unlimited EU calling, and of course, asterisk support |
| 18:43.00 | vaewyn | Lee__: I was trying to be facetious... but anywyas :P |
| 18:43.32 | *** join/#asterisk |Vulture| (~V@95.236.204.68.cfl.res.rr.com) |
| 18:43.36 | vaewyn | Lee__: depending on the mood wars can go on for days in here :P |
| 18:43.52 | Lee__ | I'm new but I'm a few steps away from getting Asterisk on an embedded platform. |
| 18:43.55 | Corydon-w | It's time to move from Python to assembly language. |
| 18:44.12 | Corydon-w | At least in assembly, you can do thing in multiple ways |
| 18:44.14 | vaewyn | Lee__: Cool... congrats :} |
| 18:44.39 | vaewyn | Lee__: You should hook up with kpfleming... he got it running on the Linksys WRT54G boxes :P |
| 18:44.42 | Lee__ | thanks. it's my first hack at embedded Linux. |
| 18:44.56 | Lee__ | woah, a $50 * appliance :) |
| 18:45.02 | vaewyn | yep :P |
| 18:45.03 | Lee__ | probably can serve one channel though. |
| 18:45.26 | anthm | how did vi take the loss did it need a hankey? |
| 18:45.35 | vaewyn | As long as you don't transcode it handles quite a few calls |
| 18:45.54 | Lee__ | cool. I really need to get one of those things |
| 18:46.09 | vaewyn | emacs lost... it just didn't notice cause the results were in plain english :P |
| 18:46.47 | *** join/#asterisk Nuxi (~nuxi@cust-sdsl-204-250-82-202.bzn-co-i1000-01.bridgeband.net) |
| 18:46.49 | vaewyn | emacs is like S&M.... it loves control... |
| 18:47.10 | vaewyn | ctrl-x ctrl-w |
| 18:47.12 | vaewyn | hehehe |
| 18:47.42 | ManxPower | EMACS OS: Your nightmare has just begun! |
| 18:48.37 | CyberKnet | Ever wonder why in web forms suffix drop down box they have "Jr" or "Sr" or "III"... but not even once have I ever seen one that said "Master Of The Universe". I mean, what am I supposed to do? |
| 18:48.37 | anthm | every feature I ever added to asterisk that you shamelessly use was coded in emacs so have that in mind as you think of ways to knock it. |
| 18:49.01 | *** join/#asterisk Inv_Arp (junya@adsl-3-244-124.mia.bellsouth.net) |
| 18:49.10 | ManxPower | CyberKnet: Leave it blank. |
| 18:49.14 | sean | CyberKnet: I intend to sign up as Senator next time I buy an airline ticket online. |
| 18:49.26 | ManxPower | Future Emporer of the Planet is never there either. |
| 18:49.31 | *** join/#asterisk Dishwasha (~chatzilla@208.251.32.70) |
| 18:49.44 | ManxPower | sean: that will keep you off the plane ya know. |
| 18:49.46 | Inv_Arp | hmm anyone having incoming BV issues? |
| 18:50.02 | CyberKnet | sean: definitely. |
| 18:50.06 | ManxPower | Is there a day when someone does NOT ask "Anyone here having problems with BroadVoice?" |
| 18:50.11 | *** join/#asterisk bannerman (~bannerman@209.216.176.42) |
| 18:50.17 | Dishwasha | Hey, I figured I might as well mention that zaptel-1.0.7.tar.gz and libpri-1.0.7.tar.gz have corrupt tar files from ftp.digium.com, so if anybody here works at digium you might mention that |
| 18:50.21 | sean | ManxPower: bah. it's an innocent "mistake" |
| 18:50.29 | vaewyn | anthm: but you coulda done the same thing in any editor... so I still knock it :P |
| 18:50.51 | vaewyn | ManxPower: before they opened for business ;P |
| 18:51.11 | sean | plus, I'm in Canada. We're much less uptight about air travel (even though we ARE still uptight about it). |
| 18:51.12 | vaewyn | anthm: Hey..... I'm on your side on the perl thing.... |
| 18:51.20 | Dishwasha | Oh wait, nevermind, seems to be an issue with microsoft ftp, weird how asterisk, asterisk-sounds, and asterisk-addons worked fine tho |
| 18:51.52 | vaewyn | anthm: my main complaint with emacs is that it isn't small enough to work off a boot disk... and hence I would have to keep 2 editors in my mind |
| 18:51.58 | Dishwasha | Well, this is cool, I have finally deployed my first asterisk server complete with dialplan |
| 18:52.07 | jsharp | "issue with microsoft" is usally the answer to most of life's problems. |
| 18:52.18 | Dishwasha | jsharp: hah |
| 18:52.19 | blitzrage | OT: anyone know how to cut a number of characters from a string in PHP |
| 18:52.22 | vaewyn | anthm: heck... I don't even use vim... I use nvi just so it doesn't screw with my mind :P |
| 18:52.25 | anthm | ahh but when would you do software develompent on a boot disk! |
| 18:52.25 | Lee__ | Dishwasha: congrats. I hope Windows didn't get in the way too much |
| 18:52.26 | blitzrage | just need a pointer to the right function |
| 18:52.30 | jsharp | use substr() |
| 18:52.34 | blitzrage | jsharp: thanks! |
| 18:52.48 | Dishwasha | Lee__: Oh no, fortunately I'm using Linux for everything but downloading |
| 18:53.02 | vaewyn | anthm: I don't... but then again when I do software development I use EPIC under eclipse now... that is taking over as my only method of perl coding |
| 18:53.24 | Dishwasha | So, if anybody needs help configuring asterisk to integrate with MCI's VOIP service, let me know. I might even write up a little instruction for the Wiki |
| 18:53.37 | vaewyn | anthm: You should try it sometime... it really does speed up development... especially in CVS/shared environments |
| 18:55.12 | anthm | they only way left for me to speed up development is clone myself |
| 18:55.32 | Dishwasha | anthm: You should play the game "Evil Genius" |
| 18:56.57 | Lee__ | argh! configuring a serial console boot Linux has so many insane steps. I feel so old skool. |
| 18:57.36 | Dishwasha | Does your getty look like spaghetti? |
| 18:57.40 | Lee__ | yeah |
| 18:57.53 | harryvv | Anyone seen a case while in in vm option 3 does not respond? |
| 18:57.58 | Lee__ | I got past grub and Linux, only to find that getty's not configured. |
| 18:58.14 | Lee__ | harryvv: nope |
| 18:58.14 | Dishwasha | ew, most distros do weird, which one? |
| 18:58.19 | Lee__ | Debian |
| 18:58.39 | Dishwasha | that probably explains it, I haven't used debian (directly) as of yet |
| 18:58.47 | Dishwasha | I'm an ol' slackware fuddy duddy |
| 18:58.51 | Lee__ | I was able to install it via netboot, which was kind of rad. |
| 18:58.53 | *** join/#asterisk juice (~juice@mo-69-68-105-244.dyn.sprint-hsd.net) |
| 19:00.49 | *** part/#asterisk Grooby (~Grooby@66.160.105.186) |
| 19:01.30 | jets | How do you change the IRQ on a te410 |
| 19:01.39 | jsharp | Move it to a different PCI slot. |
| 19:01.42 | vaewyn | move it to another slot |
| 19:01.56 | *** join/#asterisk drmac (~drmac@216.54.143.2) |
| 19:01.59 | Dishwasha | watch out for motherboards that do PCI slot sharing |
| 19:02.00 | jets | That's the only PCI slot in this server! |
| 19:02.03 | vaewyn | or if you have a good bios... reassign that slot |
| 19:02.14 | drmac | anyone here help me with ztcfg errors? |
| 19:03.23 | harryvv | something is really bugy with this voicemail. Always studders and some times does not respond to prompts like selecting advanced options #3. |
| 19:03.27 | *** join/#asterisk nytefall (~n2nightfa@69.24.142.129) |
| 19:03.48 | jontow | i wonder how to get an easy cdr_mysql.so module going in asterisk built from freebsd ports |
| 19:04.04 | nytefall | got a question for asterisk T-1 guru |
| 19:04.42 | nytefall | what DT type does an E&M signalling need to pick up dialed digits? |
| 19:05.12 | nytefall | is it MF or DTMF? |
| 19:05.17 | *** join/#asterisk tzafrir_laptop (~tzafrir@bzq-179-40-134.cust.bezeqint.net) |
| 19:05.46 | *** join/#asterisk NewSole (dave@i216-58-44-245.avalonworks.net) |
| 19:05.56 | *** join/#asterisk leandro_pt (~leandro@bl6-126-215.dsl.telepac.pt) |
| 19:06.07 | *** join/#asterisk tld (~tld@80.203.70.227) |
| 19:06.31 | drmac | I get this error when running ztcfg: "ZT_CHANCONFIG failed on channel 1: Invalid argument (22)" |
| 19:06.46 | jsharp | nytefall: You can use DTMF or MF, depening on which flavor of E&M you select. |
| 19:07.02 | jsharp | drmac: It means you've got your signalling set wrong for the particular card |
| 19:07.57 | *** part/#asterisk leandro_pt (~leandro@bl6-126-215.dsl.telepac.pt) |
| 19:09.44 | drmac | hmm..everything is pri, b8zs, esf.. |
| 19:10.35 | nytefall | jsharp --> can you direct me to a source where I can get more details? |
| 19:10.54 | vaewyn | drmac: pri_net or pri_cpe? not just 'pri' correct? |
| 19:11.06 | jsharp | nytefall: Look at zaptel.conf under signalling type. |
| 19:11.40 | vaewyn | zaptel doesn't have signalling... that is zapata |
| 19:11.50 | jsharp | oops, yes. |
| 19:11.52 | vaewyn | zaptel is framing/timing |
| 19:11.52 | jsharp | zapata.conf |
| 19:11.53 | vaewyn | :P |
| 19:12.11 | jets | The ident dial on these cards don't change anything |
| 19:12.29 | jontow | what the heck.. since when did the UNIQUEID field become something like: asterisk-47909-1115925106.0 |
| 19:12.30 | jsharp | Uh. Ident dial? |
| 19:13.26 | jets | this te410p has an ident card 0-9 and a-f |
| 19:14.55 | rcam | Anyone here setup AMP before? |
| 19:15.00 | jsharp | Can you stick your zaptel.conf on pastebin.ca? |
| 19:16.12 | drmac | pri_cpe |
| 19:16.45 | vaewyn | drmac you have bchan=1-23 and dchan=24 or something akin to that? |
| 19:16.49 | drmac | yep |
| 19:17.05 | jets | Does the te410 run at 133mhz |
| 19:17.07 | drmac | hang on..i think its something else.. |
| 19:17.10 | drmac | brb |
| 19:22.24 | *** join/#asterisk techie (gus@asterisk.horizonte.us) |
| 19:23.17 | ChulJin | Anyone here having problems with BroadVoice? |
| 19:24.29 | opus_ | yeah, broadvoice does not work |
| 19:24.32 | rcam | ChulJin I tried to call them earlier to sign up... And even there phone system was messing up. |
| 19:24.56 | rcam | I figured if they could not handle their own phones, they can't handle mine. |
| 19:25.01 | ChulJin | I don't touch BV...I just wanted to get ManxPower's goat. :P |
| 19:25.21 | *** join/#asterisk Defraz (~t0tal@65.103.222.4) |
| 19:26.03 | *** join/#asterisk Juxt (~Juxt@64.135.20.202) |
| 19:26.03 | rcam | ChulJin NuFone works well. |
| 19:26.31 | *** join/#asterisk dsfr (~dsfr@207.111.174.1) |
| 19:26.33 | vaewyn | NuFone works very good |
| 19:26.43 | Lee__ | so does my getty! yea! |
| 19:27.02 | Lee__ | is NuFone accepting any new customers yet? |
| 19:27.13 | Juxt | hello |
| 19:27.14 | vaewyn | yep |
| 19:27.14 | marlowe | Lee__: www.nufone.net |
| 19:27.43 | drmac | ok..got chan_zap up but now im getting this: |
| 19:27.44 | drmac | <PROTECTED> |
| 19:27.44 | drmac | May 12 14:32:51 NOTICE[16360]: app_dial.c:968 dial_exec_full: Unable to create channel of type 'ZAP' (cause 0) |
| 19:27.44 | drmac | <PROTECTED> |
| 19:27.44 | drmac | <PROTECTED> |
| 19:28.03 | drmac | cause 0 isnt defined anywhere |
| 19:28.18 | Juxt | i have 6 fxo cards in my box |
| 19:28.31 | Juxt | what's the simplest way to cycle thru them to find an avaiable one for an outgoing call? |
| 19:28.35 | vaewyn | drmac: what does a pri show span 1 say? (pastebin.ca it) |
| 19:28.46 | *** join/#asterisk netofsickcoder (~netofsick@200.121.129.178) |
| 19:28.47 | *** join/#asterisk Singod (~ud@12.129.197.229) |
| 19:28.57 | vaewyn | Juxt: make them one group and call zap/g1 instead of zap/1 |
| 19:29.03 | bjohnson | Juxt: groups in zapata |
| 19:29.10 | drmac | http://pastebin.ca/11607 |
| 19:29.30 | Juxt | so create goup=1 |
| 19:29.34 | Juxt | and add channel=1 |
| 19:29.36 | bjohnson | Juxt: and/or use superdial macro that could then let you call out via voip provider if the zap group is busy |
| 19:29.38 | Juxt | channel=2, etc |
| 19:29.56 | ManxPower | Juxt: SIX FXO cards? You sick bastard! How did you get them on their own IRQs |
| 19:30.11 | bjohnson | Juxt: btw, 6 pci cards is likely gonna be a problem |
| 19:30.14 | vaewyn | drmac: umm... what is the /R1/ for? that should be zap/g1/18005551212.... |
| 19:30.24 | drmac | R is for reverse round robin |
| 19:30.32 | vaewyn | ahh |
| 19:30.33 | vaewyn | hehehe |
| 19:30.38 | vaewyn | forgot about that one |
| 19:30.45 | Juxt | yeah i am expecting problems :-) |
| 19:30.57 | Juxt | not all of them have their own irqs |
| 19:31.00 | Juxt | 2 of them share 1 |
| 19:31.02 | Juxt | not good i know |
| 19:31.17 | jontow | why not get a pair of the 4port card? |
| 19:31.18 | jontow | (s) |
| 19:31.25 | drmac | changed it to g1 just to be sure and got same behavior |
| 19:31.26 | Juxt | this is a temporary solution |
| 19:31.27 | vaewyn | drmac: You arn't by any chance connecting to a Norhell box are you? |
| 19:31.31 | drmac | nope |
| 19:31.36 | Juxt | i have these phone lines for 3 more months |
| 19:31.38 | vaewyn | ok... just checking :P |
| 19:31.43 | Juxt | not worth buying 4 port cards |
| 19:31.51 | Juxt | so in zapata |
| 19:32.05 | drmac | you can get digium X100P's off ebay for about $10 each |
| 19:32.06 | Juxt | do i just add channel=n one per in the group? |
| 19:32.43 | vaewyn | group=1 channel => 1-23 |
| 19:32.44 | vaewyn | or such |
| 19:32.48 | Juxt | oh! |
| 19:32.52 | vaewyn | drmac: you got something like that also? |
| 19:32.55 | drmac | vae: i just enabled pri debug span 1 |
| 19:33.03 | drmac | and when i made the call again.. |
| 19:33.06 | drmac | same thing |
| 19:33.09 | drmac | no debug stuff |
| 19:33.12 | Juxt | what does pickgroup do? |
| 19:33.15 | drmac | which means the call never made it to the pri |
| 19:33.26 | drmac | pickupgroup is for *8 |
| 19:33.28 | vaewyn | drmac: do you have the group=1 and channel => 1-23 lines in zapata.conf? |
| 19:34.15 | drmac | http://pastebin.ca/11609 |
| 19:34.16 | jsharp | zap show channels shows you all 23 channels? |
| 19:34.27 | drmac | shows all 96 channels *wink* |
| 19:34.37 | jsharp | Oh. |
| 19:35.08 | jsharp | And your D channel is up? |
| 19:35.16 | drmac | yep..all 4 |
| 19:35.57 | Juxt | ok i got the channels working, thank you guys |
| 19:36.07 | drmac | app_dial is the one giving the error.. |
| 19:36.30 | drmac | pri debug doesn't show anything because app_dial never sends the call to the pri |
| 19:37.28 | drmac | dial_exec_full: Unable to create channel of type 'Zap' (cause 0) |
| 19:37.33 | ManxPower | drmac: pri debug will show stuff regardless |
| 19:37.44 | ManxPower | drmac: PRIs are chatty beasts |
| 19:37.46 | drmac | no..pri intense debug will |
| 19:37.57 | ManxPower | drmac: just wait. |
| 19:38.17 | drmac | enabled "pri debug" on all 4 |
| 19:38.39 | vaewyn | umm... drmac... ztcfg has run right? and you have restarted asterisk since you last changed the zaptel.conf or zapata.conf correct? |
| 19:38.39 | jontow | what would you guys like to see in pbx_gtkconsole.so ? |
| 19:39.04 | vaewyn | drmac: full restart... not just a reload |
| 19:39.44 | drmac | yes.. |
| 19:39.50 | *** join/#asterisk firestrm (firestrm@S010600047577bccd.gv.shawcable.net) |
| 19:39.57 | vaewyn | hmm... I am puzzled then... your configs look good |
| 19:40.37 | Juxt | ok this superdial macro thinger |
| 19:40.45 | Juxt | i am looking at the attibutes, they confuse me :-) |
| 19:40.45 | drmac | debug isnt helpful either... |
| 19:40.46 | drmac | app_dial.c: Exiting with DIALSTATUS=CHANUNAVAIL. |
| 19:42.39 | jsharp | Positive your D channels are up? |
| 19:42.54 | drmac | yep |
| 19:43.13 | drmac | <PROTECTED> |
| 19:43.13 | drmac | <PROTECTED> |
| 19:43.13 | drmac | <PROTECTED> |
| 19:43.13 | drmac | <PROTECTED> |
| 19:43.28 | jsharp | hsm |
| 19:43.44 | Singod | bye |
| 19:43.47 | Singod | exit |
| 19:44.07 | drmac | guess i'll call the card manuf's again |
| 19:46.02 | Juxt | can someone show me a working samle of superdial? |
| 19:47.10 | ManxPower | drmac: you are using BRI? |
| 19:47.18 | ManxPower | or have 4 PRIs? |
| 19:47.19 | vaewyn | ManxPower: PRI |
| 19:47.36 | drmac | 4 pri |
| 19:48.09 | drmac | btw, i have yet to see any pri debug messages |
| 19:48.33 | jsharp | Can you try just Dial(Zap/1/18005551212) |
| 19:48.38 | jsharp | Without using the group? |
| 19:48.52 | vaewyn | not a bad idea to try |
| 19:49.57 | drmac | k,..hang on |
| 19:50.28 | drmac | WOAH! that worked! |
| 19:50.32 | drmac | what the hek |
| 19:50.44 | vaewyn | groups are messed up |
| 19:50.48 | drmac | somebody dont like my groups |
| 19:50.54 | Defraz | Umm, what things can I check to see why I have echo. I have plenty of bandwidth, in fact I am only using it on my network(asterisk) and I hit the local pstn with a pri |
| 19:51.05 | Defraz | Can seem to find anything that might fix it. |
| 19:51.05 | drmac | they be here: http://pastebin.ca/11609 |
| 19:51.36 | vaewyn | drmac: just out of curiosity... try zap/g5/.... |
| 19:51.45 | vaewyn | I think it is overwriting g1 |
| 19:51.50 | *** join/#asterisk Carp1 (carp_xigon@ip-204-97-151-131.modem.logical.net) |
| 19:51.53 | vaewyn | since is the same channels |
| 19:52.21 | Carp1 | Right now I'm writing my AGI scripts in PHP.. Do you think it would be better to learn Perl to write them or learn C and just create an application? |
| 19:52.22 | drmac | hang on.. |
| 19:53.09 | *** join/#asterisk casterman (~casterman@83.214.24.202) |
| 19:57.42 | drmac | nope g5 gave same error as before |
| 19:58.13 | vaewyn | Hmm... odd... I would try only using g1-g4 with no overlap... and build up from that and see where it breaks |
| 19:58.38 | vaewynAFK | have fun all |
| 19:59.05 | drmac | yes..gonna do that |
| 20:03.01 | Nuxi | Carp1, what are your goals? |
| 20:05.37 | *** join/#asterisk zotz (~zotz@24.231.32.109) |
| 20:05.45 | drmac | Carp1, what is wrong with PHP? |
| 20:05.51 | drmac | all my AGI's are in php |
| 20:05.55 | drmac | work wonderfully |
| 20:06.27 | *** join/#asterisk darby_t (~tom@dnp96.neoplus.adsl.tpnet.pl) |
| 20:06.41 | *** join/#asterisk HD (~HD@82-136-197-93-mx.xdsl.tiscali.nl) |
| 20:09.51 | Juxt | can you write Fast AGI in php? |
| 20:10.20 | jontow | can you write Fast AGI in Fortran? (WATFOR only) |
| 20:11.47 | Carp1 | drmac: From what I hear, its a resource hog |
| 20:12.38 | Nuxi | That's why nobody uses it for servers. What kind of idiot would run php on a server? |
| 20:14.02 | Carp1 | What do you mean no one uses it? |
| 20:14.06 | Carp1 | I know alot of people that use it. |
| 20:14.25 | jontow | i like writing AGI with /bin/sh |
| 20:14.31 | ChulJin | mod_vb.net |
| 20:14.33 | ChulJin | *duck* |
| 20:14.40 | *** join/#asterisk shmaltz (~chatzilla@ool-43551098.dyn.optonline.net) |
| 20:14.54 | jontow | firestrm; talkabout FastAGI :) |
| 20:15.31 | shmaltz | anybody knows if *8 (call pickup) works with queues? |
| 20:15.37 | Nuxi | FastAGI is absolutely necessary to write agis for java. |
| 20:15.44 | firestrm | jontow, that why i only use assember for embedded systems.. nothing faster.. maintainance is a bitch.. but its fast. |
| 20:16.28 | *** join/#asterisk santiago (~santiago@63.245.86.229) |
| 20:18.10 | Nuxi | Carp1, if php is too slow, resource intensive, etc, then you probably want to write an app, as that would be the most efficient. |
| 20:24.20 | *** join/#asterisk ManxPower (~eric@dsl-209-16-67-160.datasync.com) |
| 20:25.44 | shmaltz | anybody knows if *8 (call pickup) works with queues? |
| 20:26.23 | *** join/#asterisk makkia (~pippo@host16-50.pool8250.interbusiness.it) |
| 20:27.53 | *** join/#asterisk {sean} (~sean@adsl-69-214-130-169.dsl.lgtpmi.ameritech.net) |
| 20:28.24 | *** join/#asterisk jsolares (~jsolares@200.6.213.204) |
| 20:29.05 | Juxt | if i use firefly, how can i transfer calls? |
| 20:30.10 | bjohnson | true geeks run mod_logo |
| 20:30.21 | Beirdo | pen up |
| 20:30.34 | Beirdo | forward 100 |
| 20:30.36 | Beirdo | pen down |
| 20:30.43 | bjohnson | right 90 |
| 20:31.18 | jsolares | hehe |
| 20:31.19 | Beirdo | heh |
| 20:31.20 | Juxt | can i program say *<extension> to transfer a call to that extension? |
| 20:31.29 | bjohnson | Juxt: yes |
| 20:31.31 | Beirdo | that brings back memories |
| 20:31.40 | Juxt | bjohnson: how is that done |
| 20:31.48 | opus_ | * ChulJin will start writing mod_apl as soon as his special 300-key apl keyboard is delivered. |
| 20:31.57 | opus_ | whoah that would rock!!!! |
| 20:31.57 | bjohnson | Juxt: during a call or at dial tone? |
| 20:32.04 | Juxt | during a call |
| 20:32.22 | bjohnson | Juxt: you would have to reprogram the code to look for * instead of # |
| 20:32.34 | bjohnson | and then use the tT dial options |
| 20:32.35 | Juxt | hmm well... |
| 20:32.43 | |Vulture| | anyone know why I might be getting "Out of g.729 Decoder Licenses" errors, I have 2 licenses and 1 7960 is connected happens only when I try to bridge the call to Zap or IAX |
| 20:32.49 | Juxt | i don't hav ea problem with # either |
| 20:32.51 | bjohnson | Juxt: you're aware that # will work right? |
| 20:32.57 | Juxt | it doesn't seem to |
| 20:33.05 | bjohnson | you used tT? |
| 20:33.14 | *** join/#asterisk Nix (~Nix@81.213.125.220) |
| 20:33.29 | Juxt | err no |
| 20:33.41 | ManxPower | Juxt: Is your device too stupid to support it's own transfers? |
| 20:33.46 | bjohnson | find and read about the options to the dial() command |
| 20:34.06 | ManxPower | For Zap you use FLASH/RECALL/LINK, for SIP you use the Transfer key on your phone. |
| 20:34.12 | ManxPower | Quite simple, really. |
| 20:34.17 | Juxt | yeah i am using firefly |
| 20:34.21 | Juxt | doesn't have a transfer key |
| 20:34.34 | ManxPower | All SoftPhones suck. |
| 20:34.41 | Juxt | that might be true |
| 20:34.49 | bjohnson | likely been discussed on the mailing list .. probably needs the tT |
| 20:35.09 | Juxt | yeah the mailing list archive isn't really searchable |
| 20:35.13 | Juxt | unless i am missing something |
| 20:36.05 | ManxPower | ~mailinglist |
| 20:36.06 | jbot | mailinglist is probably Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search. Browse the mailing list archive at http://lists.digium.com/ |
| 20:36.34 | ManxPower | ~docs |
| 20:36.35 | 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 |
| 20:36.48 | ManxPower | Maybe I don't post those links often enough |
| 20:36.54 | Juxt | rofl |
| 20:36.57 | jsolares | maybe |
| 20:37.53 | *** join/#asterisk MrSam2 (~smachin@home.bs8.org.uk) |
| 20:38.32 | MrSam2 | anyone using a SpeedTouch 716 with asterisk |
| 20:39.35 | ManxPower | ~google site:lists.digium.com speedtouch |
| 20:39.38 | harryvv | Are there any other voip suppliers near the pacific nw ? atacomm and voip suppy are just to far away. |
| 20:42.12 | MrSam2 | already tried that, no help |
| 20:42.44 | MrSam2 | asterisk is returning a 407 when I try to dial from the fxs ports |
| 20:44.12 | Beirdo | harryvv: I think there's one in Vancouver, BC, close enough? |
| 20:44.16 | *** join/#asterisk docelm0 (~docelm0@67.106.194.90.ptr.us.xo.net) |
| 20:44.48 | docelm0 | does anyone know how to pull the dialstatus information from an oh323 channel? I am having quite some trouble with this. Works with IAX and SIP but not the oh323 |
| 20:46.22 | *** join/#asterisk kietlak (~kietlak@11-mo3-6.acn.waw.pl) |
| 20:48.55 | harryvv | Beirdo, if its the same company that sold me the sipura ata 1000 then thay really jack up the price. Thay charged like $130 for my ata. Phones are corospondingly expensive also. |
| 20:49.09 | Beirdo | ah, that could be |
| 20:49.19 | opus_ | damn |
| 20:49.28 | opus_ | anyone here run x86_64? |
| 20:49.33 | Beirdo | but that's still cheaper than the bend-over-and-take-it brokerage fees in cross-border shipping |
| 20:49.40 | opus_ | does gaim and firefox take half a gig of memory like my system does? |
| 20:50.39 | harryvv | Actually yes or mabey no. I have imported PC hardware and still payed the same gst/pst as it was purchaced locally. |
| 20:50.57 | MrSam2 | can I disable the 407 authentication requests |
| 20:51.02 | harryvv | Under what cases are brokerage fees are charged? |
| 20:51.22 | docelm0 | opus yes |
| 20:51.31 | docelm0 | dual opteron 248 |
| 20:52.17 | harryvv | doc, I have the 244 but only one cpu at this moment. |
| 20:52.30 | docelm0 | I have 15 of these servers |
| 20:54.53 | harryvv | doc, what are thay being used for. |
| 20:55.07 | *** join/#asterisk vooduhal (~cmcbee@64-18-104-139.adsl.catt.com) |
| 20:55.33 | vooduhal | Can anyone here help with a swig/perl question? |
| 20:56.12 | vooduhal | I've got ast_readstring working now but I'm not sure how to handle the char buffer length issue. |
| 20:57.08 | vooduhal | The prototype of ast_readstring is: int ast_readstring(struct ast_channel *c, char *s, int len, int timeout, int ftimeout, char *enders) |
| 20:57.27 | vooduhal | s is the location to put the user input, and len is the length. |
| 20:57.30 | *** join/#asterisk _DAW (~bob@cable-68-114-110-210.sli.la.charter.com) |
| 20:57.40 | docelm0 | harry, asterisk call processing |
| 20:57.48 | vooduhal | But how do I handle the length of a string in perl so that I can properly give it a length? |
| 20:57.59 | docelm0 | I have 5 setup loadbalanced running calls |
| 20:59.36 | harryvv | good |
| 20:59.45 | *** part/#asterisk Juxt (~Juxt@64.135.20.202) |
| 21:02.28 | opus_ | doccelm0 - do applications take like 10 times the amount of memory on your system? I know that with 64 bit addressing, everything is big, but like this is ridicolus |
| 21:02.42 | opus_ | firefox takes 400mb started with no page loaded |
| 21:03.00 | opus_ | gaim taikes 189mb |
| 21:03.15 | opus_ | X is using 2 gigs of virtual memory |
| 21:03.27 | docelm0 | No.. I dont use much of anything on my x64 boxes |
| 21:03.41 | `Sauron | look at the RSZ/RSS size |
| 21:03.48 | *** join/#asterisk meppl (mephisto@p54AAF3EB.dip.t-dialin.net) |
| 21:03.56 | `Sauron | because a large chunk is shared libs accounted for more than once |
| 21:04.11 | opus_ | what utility can show me this? |
| 21:04.11 | meppl | guten abend |
| 21:05.04 | opus_ | fuq, i'm going to have to upgrade to like 4 gigs of ram just to browse the web |
| 21:05.35 | meppl | nett . |
| 21:06.36 | docelm0 | 4812 1 root 313m 4400 8364 S 0.4 175m 0.0 305m 27:04.26 asterisk |
| 21:06.50 | docelm0 | that is my top report of my asterisk daemon |
| 21:07.17 | |Vulture| | anyone know why I might be getting "Out of g.729 Decoder Licenses" errors, I have 2 licenses and 1 7960 is connected happens only when I try to bridge the call to Zap or IAX |
| 21:07.17 | ChkDigit | Es ist tag hier, meppl... =) |
| 21:07.43 | opus_ | 14107 root 18 0 167m 6708 3844 S 0.3 0.7 0:00.10 asterisk |
| 21:08.02 | meppl | good evening chkdigit |
| 21:08.56 | *** join/#asterisk in-side (~Lowgitek@es-217-129-31-172.netvisao.pt) |
| 21:09.21 | *** join/#asterisk kb1_kanobe (~krisbouti@h24-207-80-55.cst.dccnet.com) |
| 21:09.38 | in-side | Hi |
| 21:10.14 | in-side | does anybody knows a good billing frontend ? |
| 21:10.27 | in-side | or a nice and adaptative billing system ? |
| 21:10.28 | docelm0 | custom coded.. :) |
| 21:10.40 | in-side | ya i have one custom coded already |
| 21:11.20 | in-side | it is not opensource and i haven't time to continue to developping it |
| 21:11.21 | in-side | :S |
| 21:14.32 | niZon | anyone use a sipura-841? |
| 21:14.38 | ChkDigit | Yup. |
| 21:14.45 | niZon | how is it? |
| 21:15.08 | ChkDigit | It is ok. Reboots sometimes. But all-in all good for the money. |
| 21:15.14 | ChkDigit | Reboot only takes 3 seconds. |
| 21:16.48 | niZon | reboots during calls? |
| 21:16.55 | pussfeller | whats the downside to using an ata vs using a pci unit? |
| 21:17.20 | ChkDigit | niZon: It has never done that. Just on occasion while sitting around. |
| 21:17.30 | in-side | niZon: I have some spa 3000 units |
| 21:17.32 | pussfeller | the upside seems to be no irq worries |
| 21:17.33 | in-side | wazz up? |
| 21:18.21 | in-side | pussfeller: I see no downside in using a ata to be honest |
| 21:18.41 | in-side | ChkDigit: wich firmware are you using ? |
| 21:18.48 | pussfeller | im running out of pci slots :) |
| 21:18.56 | in-side | did you checked the cables ? |
| 21:19.10 | in-side | pussfeller: in fact I preder ata always |
| 21:19.25 | niZon | in-side: you like the 3000s? |
| 21:19.26 | in-side | the only downside could be prices |
| 21:19.32 | in-side | niZon: very nice |
| 21:19.33 | niZon | ChkDigit: how's the speaker phone? |
| 21:19.44 | in-side | documentation lacks in advanced information |
| 21:19.55 | niZon | i should get an SPA-841 and SPA-3000 for some testing |
| 21:20.01 | in-side | no much futher information is gave |
| 21:20.07 | in-side | I would prefer spa 3000 |
| 21:20.14 | in-side | as it is much more extwnsible |
| 21:20.22 | in-side | and you could keep your old phones |
| 21:20.37 | niZon | which is why i said AND :P |
| 21:20.38 | in-side | till now I just had a problem with spa 3000 |
| 21:20.45 | in-side | with sip compact messages |
| 21:20.53 | niZon | how did you solve it? |
| 21:21.00 | in-side | firmware upgrade |
| 21:21.04 | in-side | not a big issue |
| 21:21.07 | Romik | somebody can help me with regex - Zap/23/sadfsdfsdf/sdfsdfsdf/ i need to strip all / and put into $1 = ([^/.]*) not work right :( ? |
| 21:21.48 | niZon | ah |
| 21:21.59 | in-side | sound are very nice |
| 21:22.01 | in-side | very real |
| 21:22.08 | jalsot | mc |
| 21:22.13 | jalsot | sorry |
| 21:22.17 | jalsot | wrong windo ;) |
| 21:22.20 | in-side | much control over sip parameters |
| 21:22.21 | in-side | and so on.. |
| 21:22.22 | niZon | i wonder how good the GXP-2000s are |
| 21:22.34 | pussfeller | im stuck with kphone right now which sucks cause I can't play music for fear of getting a call and the sound card won't release |
| 21:22.54 | pussfeller | which is a whole rant in and of itselfr |
| 21:23.00 | in-side | niZon: well it is nice... normal I would say |
| 21:23.08 | ManxPower | pussfeller: welcome to the world of softphones |
| 21:23.13 | niZon | hm |
| 21:23.23 | niZon | i'm bored of playing with softphones |
| 21:23.28 | pussfeller | thats more welcome to the world of lousy linux software mixing |
| 21:23.33 | in-side | I had test some ut ya.. it is enough |
| 21:23.39 | in-side | I have some cheap brand units |
| 21:23.44 | in-side | and they work nicelly |
| 21:23.54 | in-side | have 2 lines visor |
| 21:23.57 | pussfeller | supposedly it can be done on an i810 |
| 21:24.11 | in-side | softphones real sux |
| 21:24.19 | niZon | there aren't enough canadian companies that sell voip equipment |
| 21:24.21 | in-side | but is nice to debug |
| 21:24.22 | pussfeller | i can hear the mic and music at the same time, why not sound from an application |
| 21:24.38 | ChkDigit | in-side: I can't recall the firmware, and the phone is elsewhere right now... |
| 21:25.06 | ChkDigit | niZon: The speakerphone is a problem. The mic appears to be on the bottom, and does not pickup sound well. |
| 21:25.06 | in-side | ChkDigit: eheh ok |
| 21:25.16 | in-side | I would check if there ar eno firmware upgrade first |
| 21:25.22 | in-side | then next check the power cable |
| 21:25.29 | ChkDigit | You may be right. |
| 21:25.39 | in-side | some of my phones |
| 21:25.46 | in-side | have the cables behind to tight |
| 21:25.53 | ChkDigit | A friend is using it, and did the install... |
| 21:25.59 | in-side | I don't know if sipura ones suffers from the same |
| 21:26.15 | niZon | ChkDigit: you're talking about the GXP-2000? |
| 21:28.16 | ChkDigit | niZon: No the Sipura SPA-841. |
| 21:28.52 | HeadachesAbound | anybody here got polycom IP500s? |
| 21:29.04 | ChkDigit | I have one of those too.. |
| 21:29.18 | ChkDigit | That one, I love. |
| 21:29.19 | HeadachesAbound | what kind of headset jack do those have on them? |
| 21:29.37 | ChkDigit | RJ-11 I believe... |
| 21:29.49 | HeadachesAbound | is there somepleace that i can detailed specs / pics of the ip500? |
| 21:29.57 | ChkDigit | I think they're amped. |
| 21:30.09 | ChkDigit | www.polycom.com? |
| 21:30.42 | *** join/#asterisk rcam (~rcam@adsl-218-151-77.jax.bellsouth.net) |
| 21:31.34 | HeadachesAbound | will be back in a bit, must go home now! |
| 21:32.15 | ChulJin | reminds me... |
| 21:32.49 | ChulJin | do you suppose anyone makes adapters to allow 2.5mm headsets to be plugged into RJ[narrow] headset jacks? |
| 21:33.44 | niZon | ChulJin: make your own? :P |
| 21:33.48 | ChkDigit | I've never seen any. There should be plenty of headsets for RJ-14 or 2.5 though. |
| 21:34.01 | niZon | the RJ11 should have 4 pins, 2 for mic and 2 for speaker |
| 21:34.58 | *** join/#asterisk jbot (ibot@apt.bot.TimRiker.active.supporter.pdpc) |
| 21:34.58 | *** topic/#asterisk is Asterisk: The Open Source PBX || Astricon Europe -- Madrid, Spain -- June 15-17 || ClueCon Dev Conf Aug 3rd - 5th - http://www.cluecon.com (Registration Open Now) || Dev meeting today 1pm CST IAX2/guest@switch-3.asterlink.com/996 |
| 21:35.39 | ChkDigit | ~jbot How ya doing... =) |
| 21:35.51 | ChkDigit | Hum, |
| 21:36.00 | ChulJin | oh yeah |
| 21:36.20 | niZon | ~squee |
| 21:36.22 | ChkDigit | ~sipura |
| 21:36.23 | jbot | well, sipura is selling out to Cisco... fags. |
| 21:36.40 | niZon | lol |
| 21:36.52 | ChkDigit | Nice. |
| 21:37.32 | jontow | :-/ does that mean sipura is going to be high priced and suck now? ;) |
| 21:38.34 | ChkDigit | It could have been worse, the Canadian Gov't could have given Nortel enough money to do it, and make it exhorbitantly expensive. Then croak and die. |
| 21:38.56 | joe | ChkDigit: believe it or not I think radioshack has some |
| 21:39.21 | ChkDigit | 2.5-RJ14 adators? |
| 21:39.32 | ChkDigit | Or Cisco/Sipura/LinkSys phones? |
| 21:39.42 | joe | 2.5-RJ14 adators |
| 21:40.18 | sean | radioshack is such a ripoff for that stuff, if there's a _real_ electronic shop around. |
| 21:40.54 | sean | went to buy some phono plugs earlier this week.. $5.99 at RS. The local electronics shop (where they have aisles of capacitors and resistors): $0.89. |
| 21:41.16 | sean | yeah, true.. |
| 21:42.10 | *** join/#asterisk bsgr (~doc@p548B05D2.dip0.t-ipconnect.de) |
| 21:42.32 | bsgr | hi |
| 21:43.02 | *** join/#asterisk ariel_ (~Ariel@dsl-20-177.cofs.net) |
| 21:47.44 | opus_ | i had the manager at my local radio shack try to outsmart me last sunday |
| 21:47.56 | ChkDigit | Do tell... >=) |
| 21:48.26 | opus_ | tried to chew me out, i bought a power supply, used it on something, took it back. said that I damaged it. volt meter said 15.5 volts for a 12v adapter |
| 21:48.42 | ChkDigit | Sure, no load. |
| 21:48.59 | opus_ | yeah, he woulnd't give me a refund |
| 21:49.11 | opus_ | so I bought another one, showed him, and he got pissed. |
| 21:49.17 | ChkDigit | So, did you hit him with it? |
| 21:49.20 | opus_ | had to do refund for two :) |
| 21:49.44 | *** join/#asterisk dr123 (~temp@12-202-51-38.client.insightBB.com) |
| 21:49.55 | opus_ | later |
| 21:51.03 | dr123 | hey can anyone help me witha registration problem... I have 2 asterisk servers on the same network behind the same nat and I have them registred to one another and IAX2 show registry does show they are registered but when I place a call that should port over from one server to the other the call is immediatly terminated |
| 21:55.07 | |Vulture| | anyone know why I might be getting "Out of g.729 Decoder Licenses" errors, I have 2 licenses and 1 7960 is connected happens only when I try to bridge the call to Zap or IAX |
| 21:55.22 | *** join/#asterisk Milligan (~support@wkstn6.gnwd-noc.valuelinx.net) |
| 21:56.06 | |Vulture| | but the call is in g729... |
| 21:56.30 | |Vulture| | and I have every SIP channel set to disallow=all;allow=ulaw except for the one i want to have 729 |
| 21:56.36 | *** join/#asterisk Grooby (~Grooby@ip24-250-126-171.dc.dc.cox.net) |
| 22:02.15 | *** join/#asterisk sneak (~sneak@64.220.234.21.ptr.us.xo.net) |
| 22:04.23 | Nuxi | anybody got a cdr to store calls in the windows registry? |
| 22:04.38 | *** part/#asterisk grolloj (~chatzilla@slim-eth0.horizonlive.net) |
| 22:04.59 | *** part/#asterisk santiago (~santiago@63.245.86.229) |
| 22:06.00 | ChulJin | nuxi: place a bounty, I'll do it. |
| 22:06.09 | *** join/#asterisk HeadachesAbound (~asterisk-@adsl-70-244-228-14.dsl.tulsok.swbell.net) |
| 22:06.21 | ChulJin | WB Head'ound |
| 22:06.45 | HA | ty CJ. |
| 22:07.00 | HA | I think I prefer the shortened form. |
| 22:07.15 | HA | I think, therefore I am HA! |
| 22:07.52 | shido6 | -s |
| 22:08.34 | comintel | whats the command to show the dial statement, in the CLI? |
| 22:08.46 | ChulJin | show application dial |
| 22:09.00 | comintel | ahar |
| 22:09.03 | comintel | cool, thanks |
| 22:09.31 | joe | anyone here runing on centos4? |
| 22:09.46 | joe | running, even |
| 22:09.51 | ChrisHodgetts | I was wondering if anyone could help with this, its a NAT problem I assume, but I dont get why :) |
| 22:11.06 | Nuxi | nat -> problem :: cardinality(problem) >= cardinality(nat) |
| 22:14.03 | *** join/#asterisk MajestiK (~MajestiK@S01060800208687ec.ed.shawcable.net) |
| 22:14.12 | ChrisHodgetts | I run linphone on my laptop |
| 22:14.21 | ChrisHodgetts | asterisk on an internal box within my network |
| 22:14.37 | ChrisHodgetts | and I have portforwarded rtp ports to the internal box |
| 22:15.57 | ChrisHodgetts | I have registered with a sip proxy provider |
| 22:16.15 | ChrisHodgetts | you call the number they have allocated, and it reachses an extenetion within my pabx, and audio is heard on the remote |
| 22:16.16 | ChrisHodgetts | end |
| 22:17.05 | ChrisHodgetts | then when the laptop makes a call out, rtp packets are seen by the laptop, but no audio is heard, and from a tcpdump /ethereal says that icmp destination port is unreachable -- on the pabx machine |
| 22:17.31 | ChrisHodgetts | but when you call an internal extention I dont see these errors, and rtp packets flow both directions |
| 22:18.18 | *** join/#asterisk r0d3nt (~RatMan@wsip-24-234-241-84.lv.lv.cox.net) |
| 22:18.22 | ChrisHodgetts | wondering what I could/am doing wrong |
| 22:19.55 | *** part/#asterisk bsgr (~doc@p548B05D2.dip0.t-ipconnect.de) |
| 22:21.08 | ChulJin | having * behind NAT |
| 22:22.51 | HA | anybody know where to get headsets for a polycom ip500 for cheap? |
| 22:25.40 | JerJer | ebay.com |
| 22:26.17 | firestrm | 4 days still no nufone.. |
| 22:26.53 | rcam | firestrm Hello... What's wrong with Nufone? |
| 22:27.26 | firestrm | rcam, sent payment ~4 days ago, still no credit.. |
| 22:27.38 | rcam | firestrm It takes a good while. |
| 22:27.39 | tzanger | firestrm: have you emailed support@ |
| 22:27.40 | tzanger | ? |
| 22:27.44 | tzanger | have you bugged shido6? |
| 22:27.48 | r0d3nt | HA, many standard headsets are compatible. |
| 22:27.50 | rcam | firestrm Call and get on them. |
| 22:28.09 | rcam | firestrm Have you ever used AMP... |
| 22:28.10 | firestrm | rcam, i dont want to be annoying though. |
| 22:28.21 | firestrm | rcam, no whats AMP? |
| 22:28.21 | rcam | firestrm Yeah I know how it is. |
| 22:28.26 | pussfeller | do those generic x100p 's handle caller id |
| 22:28.56 | rcam | firestrm Asterisk Management Portal |
| 22:29.00 | firestrm | tzanger, i might resort to that.. |
| 22:29.07 | ChrisHodgetts | yeah sorry ChulJin I do have * behnd nat |
| 22:29.24 | firestrm | rcam, ive seen it.. but no.. never used it.. im a Vi kinda guy.. |
| 22:29.29 | JerJer | firestrm: or instead of just bitching about it send the transaction reference id number to fucking billing@nufone.net |
| 22:29.39 | HA | how about an rj11 to 2.5mm adapter or cable? |
| 22:29.40 | firestrm | JerJer, i did.. |
| 22:29.42 | JerJer | i hate whiny bitches |
| 22:29.58 | pussfeller | JerJer, do you work for nufone or something |
| 22:30.02 | rcam | firestrm JerJer = Nufone. |
| 22:30.13 | firestrm | :) |
| 22:30.19 | rcam | Howdy JerJer, how have you been? |
| 22:30.20 | pussfeller | what kinda image do you think that gives nufone |
| 22:30.40 | firestrm | JerJer, should i send it again? i sent it 3 days ago.. |
| 22:30.44 | HA | well, off to play parent for a while. |
| 22:30.52 | JerJer | pussfeller: then don't buy our service |
| 22:30.57 | JerJer | i could really care less |
| 22:31.02 | pussfeller | i guess so |
| 22:31.32 | firestrm | JerJer, it sould like you'r havign a bad day :( |
| 22:31.55 | JerJer | nope - i just hate whiny bitches |
| 22:32.04 | JerJer | either do something about the problem or shut the fuck up |
| 22:32.18 | JerJer | i see 3 email to billing@nufone.net that have not been dealt with |
| 22:33.20 | *** join/#asterisk Nethab (~chatzilla@adsl-67-113-141-170.dsl.sntc01.pacbell.net) |
| 22:33.58 | firestrm | JerJer, well.. Im trying to do somthing about it. but unfortunatly your page says nothing about billing@nufone.net, although it does list sales@nufone.net, i know you might be busy, or even just dont care, but you really cant blame people when you dont give them the correct information on how to conduct a transaction. |
| 22:33.59 | *** join/#asterisk Alvaro123 (Alvis@200.105.128.59) |
| 22:34.56 | *** join/#asterisk AgiNamu (~AgiNamu@216.230.151.230) |
| 22:35.23 | *** join/#asterisk darth-timeus (darth@200.105.128.61) |
| 22:35.28 | darth-timeus | hello |
| 22:35.36 | darth-timeus | i'm configuring a new asterisk server |
| 22:35.40 | AgiNamu | Hi Darth |
| 22:35.47 | darth-timeus | with fedora 3 |
| 22:35.51 | AgiNamu | Do you need help configuring your Asterisk server? |
| 22:35.57 | darth-timeus | yes, please |
| 22:36.03 | darth-timeus | my problem is this |
| 22:36.06 | AgiNamu | Sure, what difficulties are you experiencing? |
| 22:36.08 | MikeJ[Laptop] | ~docs |
| 22:36.09 | 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 |
| 22:36.14 | MikeJ[Laptop] | there you go :D |
| 22:36.20 | Alvaro123 | Hello |
| 22:36.23 | firestrm | darth-timeus, have you read asteriskdocs.org yet? |
| 22:36.38 | darth-timeus | yes |
| 22:36.43 | firestrm | darth-timeus, that the best place to begin. |
| 22:36.45 | AgiNamu | OK, why don't you ask your question. |
| 22:36.58 | AgiNamu | Then, depending on the content, we will either help you, mock you, or offer you consulting. |
| 22:37.19 | darth-timeus | when i place a call i can't hear what the other end conversation |
| 22:37.24 | darth-timeus | but they can hear me |
| 22:37.26 | AgiNamu | Are you using SIP? |
| 22:37.33 | darth-timeus | no, h323 |
| 22:37.41 | AgiNamu | Oh, sorry, I don't use H323. |
| 22:37.42 | MikeJ[Laptop] | darth-timeus, which driver |
| 22:38.32 | Nethab | darth-timeus: that's a common NAT issue |
| 22:38.37 | Nethab | one way audio |
| 22:38.41 | darth-timeus | openh323 |
| 22:38.48 | darth-timeus | yes |
| 22:38.59 | darth-timeus | but my network, is not using nat |
| 22:39.03 | darth-timeus | or firewall |
| 22:39.20 | MikeJ[Laptop] | darth-timeus, which channel driver |
| 22:40.00 | darth-timeus | chan_h323 |
| 22:40.04 | Nethab | anyone read the open letter from broadvoice |
| 22:40.21 | Nethab | isn't chan_h323 the broken one that comes with asterisk? |
| 22:40.21 | MikeJ[Laptop] | how old? |
| 22:40.53 | MikeJ[Laptop] | JerJer, Nethab has a question for you ^^ :) |
| 22:41.05 | MikeJ[Laptop] | darth-timeus, how old is your copy? |
| 22:41.08 | daork | MikeJ[Laptop]: hahaha |
| 22:41.14 | Nethab | i don't need new service silly head |
| 22:41.15 | AgiNamu | Nethab, which open letter? |
| 22:41.35 | Nethab | the letter from their president about their service problems |
| 22:41.42 | darth-timeus | is the 0.1.0 |
| 22:42.25 | JerJer | firestrm: Pay attention |
| 22:42.26 | JerJer | Send money via PayPal to billing@nufone.net to continue to fund your account, if you can. |
| 22:42.26 | JerJer | Wire transfer information is available upon request. |
| 22:42.37 | *** join/#asterisk NewSole (dave@i216-58-44-245.avalonworks.net) |
| 22:43.02 | firestrm | jerjer, done 4 days ago.. |
| 22:43.11 | *** join/#asterisk Legend (~legend@24.244.142.133) |
| 22:43.32 | firestrm | JerJer, and i just resent the transaction to billing |
| 22:45.20 | *** join/#asterisk outtolunc (~me@ppp-69-237-32-168.dsl.pltn13.pacbell.net) |
| 22:45.37 | darth-timeus | i use the openh323-1.15.1 |
| 22:45.45 | darth-timeus | to compile it |
| 22:45.53 | darth-timeus | i guess is a little old |
| 22:46.25 | firestrm | oh,, my mistake.. 3 days ago.. |
| 22:46.30 | MikeJ[Laptop] | darth-timeus. read the readme carefully, it requires specific versions |
| 22:48.58 | darth-timeus | i use the version in the readme, for both the openh323 and the pwlib |
| 22:49.11 | MikeJ[Laptop] | ok. |
| 22:49.12 | darth-timeus | do you think, the problem is a compilation issue? |
| 22:49.23 | MikeJ[Laptop] | asterisk cvs head?? |
| 22:50.41 | darth-timeus | Asterisk CVS-HEAD-04/28/05-15:20:51, Copyright (C) 1999 - 2005 Digium. |
| 22:51.03 | MikeJ[Laptop] | update to current head |
| 22:51.11 | MikeJ[Laptop] | I beleive that issue was fixed |
| 22:51.23 | firestrm | JerJer, thanks.. |
| 22:51.28 | darth-timeus | ok, i'll give it a try |
| 22:51.30 | darth-timeus | thanks |
| 22:51.32 | Nethab | yeah cause old head isn't very good |
| 22:51.42 | AgiNamu | lol |
| 22:51.43 | Nethab | er, wait... |
| 22:51.46 | vpp | darth-timeus: turn debug on |
| 22:51.47 | MikeJ[Laptop] | uhhhhh |
| 22:51.48 | Nethab | that came out wrong |
| 22:51.49 | MikeJ[Laptop] | yeah |
| 22:51.51 | vpp | and check the RTP address |
| 22:51.54 | vpp | i've had the same issue |
| 22:51.59 | vpp | gave up on it and moved to oh323 |
| 22:52.22 | MikeJ[Laptop] | jerjer commited a bunch of fixes the last few days, get up to date then troubleshoot |
| 22:52.51 | darth-timeus | ok, i'll let you know |
| 22:52.54 | darth-timeus | thanks |
| 22:53.01 | darth-timeus | good bye |
| 22:53.06 | AgiNamu | Thanks for playing. |
| 22:59.15 | xkev | http://www.voip-info.org/wiki-Asterisk+echo+cancellation |
| 22:59.33 | xkev | that page is no help. anyone understand the alternate echo cancelers? |
| 23:00.17 | kb1_kanobe | no. the reference implementation behind mec2 is good and valid. As an implementation though mec2 is somehow flawed. :=-) |
| 23:01.36 | kb1_kanobe | M2820 |
| 23:01.47 | harryvv | any of you have cbc news turn on you tv if you do biz with voip in canada. Canadian goverment is thinking of regulating voip service. |
| 23:02.17 | kb1_kanobe | xkev: Hmmm.... take a look at bug 2820 in Mantis |
| 23:03.12 | Nethab | their considering regulating voip in as much as preventing the ILEC's from undercharging and forcing everyone out of business |
| 23:06.27 | jontow | it'll have to be regulated someday, or it'll be shitty underpriced service with no guarantee you'll ever get ahold of an emergency dispatch agent in time of trouble, and things will generally be chaotic :) |
| 23:06.58 | *** join/#asterisk roamer323 (~sing@toronto-HSE-ppp4090567.sympatico.ca) |
| 23:08.11 | *** join/#asterisk Ahewes (~rsb@adsl-69-107-53-145.dsl.pltn13.pacbell.net) |
| 23:11.39 | xkev | kb1_kanobe thx |
| 23:11.40 | *** join/#asterisk ilium007 (~brantwint@220-253-92-177.QLD.netspace.net.au) |
| 23:15.04 | *** part/#asterisk makhtar (~ageller@mail.bulletinnews.com) |
| 23:15.11 | xkev | anything to worry about wrt echotraining=800 |
| 23:15.13 | xkev | ..etc? |
| 23:16.05 | xkev | and use -DAGGRESSIVE_SUPPRESSION on that? |
| 23:20.40 | Nethab | what's the deal with the new ${DB()} stuff |
| 23:20.52 | *** join/#asterisk Sedorox (brandon@sedorox.staff.smartserv) |
| 23:20.55 | vpp | hmm |
| 23:20.58 | vpp | Dial(Zap/g1,30) |
| 23:21.01 | vpp | is that corect? |
| 23:21.18 | bkw_ | regulating voip is like trying to heard minnows |
| 23:21.28 | bkw_ | you can try |
| 23:21.30 | bkw_ | but you will fail |
| 23:21.35 | Sedorox | ahahahah I like that |
| 23:21.54 | *** topic/#asterisk by bkw_ -> Asterisk: The Open Source PBX || Astricon Europe -- Madrid, Spain -- June 15-17 || ClueCon Dev Conf Aug 3rd - 5th - http://www.cluecon.com (Registration Open Now) |
| 23:22.32 | vpp | how do i set the callerid before i dial? |
| 23:22.42 | dr123 | anyone using asterisk on a WRT Linksis Wireless router.. I have it installed and working but I dont have any idea how many extensions it can handle at once |
| 23:24.20 | tzanger | dr123: so try it |
| 23:24.27 | tzanger | dr123: then tell us so we have a defininitive answer |
| 23:24.34 | tzanger | it's open source, contribute some data |
| 23:25.16 | dr123 | I am contrubting I am trying to get a useful doc up on the wiki on voip-info for how to like to asterisk servers together using IAX |
| 23:25.24 | dr123 | as thiers sucks |
| 23:25.28 | tzanger | good :-) |
| 23:26.21 | dr123 | but before i can publish i need help with the dial string |
| 23:26.23 | Grooby | Dr you using wrt54g? |
| 23:26.27 | dr123 | yeah |
| 23:26.48 | Grooby | what do you mean by # of extensions? |
| 23:27.02 | Grooby | like..load-wise on the internal network? |
| 23:27.03 | dr123 | how many simultanious dials can be used |
| 23:27.07 | dr123 | yeah load |
| 23:27.24 | Grooby | i don't think it uses a lot...i would say the limit would be on your bandwith isn't it? |
| 23:27.34 | Dishwasha | Quick question, does anybody know how much a new Mortel Meridian 1 Option 11C mini costs roughly? |
| 23:27.40 | Dishwasha | er, Nortel |
| 23:27.54 | dr123 | the bandwidth is 100 Mb |
| 23:28.01 | Grooby | ahhh |
| 23:28.03 | Grooby | lol |
| 23:28.03 | *** join/#asterisk bjohnson (~bjohnson@66.11.188.6) |
| 23:28.05 | dr123 | i dont think that is the limiting factor |
| 23:28.33 | Grooby | you got a 100mb out? |
| 23:28.36 | dr123 | has anyone linked 2 asterisks servers via iax on the same network |
| 23:28.39 | Grooby | or we talking the 4 ports internal lanes? |
| 23:28.43 | vpp | how do i view the current codec for a channel |
| 23:28.46 | dr123 | 4 lan lanes |
| 23:29.12 | Grooby | that's a good question then...i really dunno.... |
| 23:29.23 | Grooby | #wrt54g? |
| 23:29.37 | dr123 | what does that mean |
| 23:29.42 | Grooby | channel |
| 23:29.42 | dr123 | # ... |
| 23:29.46 | Grooby | you might be able to ask in there |
| 23:29.46 | dr123 | oh |
| 23:29.47 | dr123 | thanks |
| 23:29.55 | dr123 | i didnt know that is what we are talking about |
| 23:30.00 | dr123 | ill look there |
| 23:30.16 | Grooby | =) |
| 23:30.34 | *** join/#asterisk roamer323 (~sing@toronto-HSE-ppp4090567.sympatico.ca) |
| 23:30.48 | dr123 | Grooby have you ever linked 2 asterisks servers on the same network behind the same NAT |
| 23:30.51 | *** join/#asterisk Astrak (~astrak@122.Red-217-126-181.pooles.rima-tde.net) |
| 23:31.07 | Astrak | hi ppl |
| 23:31.29 | Dishwasha | Does anybody know how much a new Nortel Meridian 1 Option 11C mini costs roughly or something equivalent? |
| 23:34.08 | Grooby | dr, no..why? |
| 23:34.18 | Grooby | you having trouble linking them? |
| 23:38.16 | bjohnson | linking * servers via iax is same whether on the same network or not |
| 23:45.42 | *** join/#asterisk bkw__ (~brian@ppp-70-243-95-193.dsl.tulsok.swbell.net) |
| 23:46.25 | *** join/#asterisk fefede (~fna@r200-125-52-67-dialup.adsl.anteldata.net.uy) |
| 23:46.29 | fefede | i am try to get working some ipphone's... I have some problems in the codec negotiation... How can I do, to make asterisk don't interfere in the negotiation?? |
| 23:46.30 | fefede | I like that the end point resolve de codec to use....! |
| 23:47.41 | fefede | some way to doit???? |
| 23:48.55 | *** join/#asterisk bkw_ (~brian@bkw.developer.and.friend.of.asterisk) |
| 23:48.55 | *** mode/#asterisk [+o bkw_] by ChanServ |
| 23:52.41 | *** join/#asterisk HA (~asterisk-@adsl-70-244-228-14.dsl.tulsok.swbell.net) |
| 23:52.59 | *** join/#asterisk fefede (~fna@r200-125-52-67-dialup.adsl.anteldata.net.uy) |
| 23:53.57 | *** join/#asterisk dslx (~jay@network-operations-center.dslx.net) |
| 23:54.03 | fefede | hi... I made some question some minutes ago... but i can't read answer because I lost my conection.. |
| 23:54.18 | *** join/#asterisk BoRiS (boris@S01060040ca1e5b54.wp.shawcable.net) |
| 23:54.50 | HA | where can i get cheap headsets for polycom ip500s |
| 23:54.52 | dslx | Any ISP's here using * as a hosted PBX platform? |
| 23:56.12 | fefede | some ideas? I need to leave my IPphone's negotiate the codec with out asterisk interaction... |
| 23:56.14 | shido6 | fefede you have something misconfigured in your sip.conf |
| 23:56.18 | shido6 | disallow=all |
| 23:56.20 | shido6 | allow=ulaw |
| 23:56.23 | shido6 | not allow=all |
| 23:56.27 | shido6 | should fix your problem |
| 23:58.18 | fefede | no but i like that the end point make the choice |
| 23:59.09 | jontow | has anyone actually tried a VoIP call with a specific codec over a 24.6kbit dialup connection? :) |