00:00.30 | DrukenHme | do ya see an rj45 jack ya can't figure out? :) |
00:00.48 | sivana | ya.. there's an RJ45 |
00:00.59 | DrukenHme | how many? |
00:00.59 | intrin | is everyone ignoring me? |
00:01.00 | intrin | :P |
00:01.03 | file[laptop] | D-ohhh! |
00:01.12 | sivana | not sure.. I'll have to check.. it's not here |
00:01.25 | DrukenHme | well, doesn't really matter anyways |
00:01.31 | DrukenHme | it'll take a t1 directly |
00:01.47 | DrukenHme | if it's 56k, it has to be digital... |
00:02.00 | eKo1 | isdn? |
00:02.00 | goatmilk | http://www.crn.com/sections/special/top25/top25_03.jhtml?ArticleID=52601053 <-- this is an awesome read |
00:02.17 | sivana | so theres no way to redirect traffic from *? |
00:02.24 | *** join/#asterisk cp5 (cp5@dsl093-032-201.snd1.dsl.speakeasy.net) |
00:02.27 | DrukenHme | of course there is |
00:02.37 | cp5 | is it possible to include a context on a sort of if-statement basis? |
00:02.59 | intrin | www.intritech.com/zap.html = my error, if anyone could be of serivce |
00:03.14 | sivana | based on the DNIS, I can tell which number the caller called and know if it's a dial up request |
00:03.25 | DrukenHme | exactly |
00:03.29 | sivana | just not sure how to send it out to the Portmaster for answering |
00:04.07 | DrukenHme | connect the portmaster to a t1 port, (like a pri) then dial using those ports |
00:04.17 | SipChat | Anyone know of any popular / busy conference / chat / party lines / numbers / rooms / etc. ? |
00:04.18 | DrukenHme | the port master should pick the calls up like normal |
00:04.25 | sivana | so I would need a second PRI then |
00:04.33 | sudoer | do i have any options if i have a linksys router, but i need to make surevoip packets get prioritized out |
00:04.35 | DrukenHme | well, a second port |
00:04.48 | sivana | I have a couple of pair gains |
00:04.56 | ariel_ | it's pizza night. |
00:05.29 | sivana | how would i create a T1 port between * and the portmaster? |
00:05.33 | *** join/#asterisk sudoer2 (~toy@pool-151-203-91-218.bos.east.verizon.net) |
00:05.42 | *** join/#asterisk jks (~jks@0x503e4c12.arcnxx4.adsl-dhcp.tele.dk) |
00:06.05 | SipChat | sudoer - how many users share the router? |
00:06.10 | jks | Hi Guys... anyone using SIPPhone.com with Asterisk and got Virtual Numbers working? |
00:06.34 | sudoer | only about 3 users, but thats enought to saturate the crappy dls connection |
00:06.39 | arrgh | funny, I'm just fiddling with that now. |
00:06.48 | jks | arrgh: sipphone? |
00:06.52 | SipChat | sudoer - you can run local software bandwidth management software |
00:06.52 | arrgh | yea |
00:06.56 | arrgh | or |
00:07.02 | arrgh | you can go in with a bat and solve your bandwidth problems. |
00:07.05 | jks | arrgh: it's a bit weird... I'm simple "stuck" |
00:07.06 | ariel_ | intrin, what system are you trying to install it on. |
00:07.10 | arrgh | *smack* *smack* *smack* |
00:07.12 | Nethab | anyone here connect to sipphone.com from * |
00:07.14 | intrin | debain |
00:07.15 | SipChat | sudoer - or run everything through a software router (arghh) ;) lol |
00:07.20 | SipChat | arrgh - ;) |
00:07.25 | jks | Nethab: me |
00:07.34 | Nethab | do you run CVS? |
00:07.34 | ariel_ | intrin, 2.4 or 2.6 |
00:07.38 | sudoer | SipChat, what kind of software? you mean sitting in the windows clients? |
00:07.41 | jks | Nethab: nope |
00:07.48 | Nethab | hmm |
00:07.48 | SipChat | sudoer - yep |
00:07.50 | intrin | hmm |
00:07.55 | intrin | i forget |
00:07.56 | intrin | how do i chk |
00:07.59 | sudoer | SipChat, do you know of any good software? |
00:08.12 | arrgh | I've got outbound through sipphone.com to work, it seems. |
00:08.23 | Nethab | jks: i keep getting an immediate hangup and a 488 Not Acceptable Here when connecting to sipphone.com now |
00:08.25 | ariel_ | intrin, cat /proc/version |
00:08.28 | arrgh | yea |
00:08.30 | arrgh | I get that too |
00:08.30 | jks | Nethab: it works fine for me |
00:08.41 | SipChat | sudoer - if you have to use a software solution best to use it on a gateway / router style dedicated pc |
00:08.46 | jks | I purchased a virtual number, but I can't figure out how to use it |
00:08.46 | intrin | Linux version 2.4.27-1-386 |
00:08.47 | SipChat | i.e. |
00:09.02 | jks | If I call it up I get "Unknown user"... I tried using sip register with the virtual number, but that doesn't work |
00:09.03 | Nethab | jks: use it how? |
00:09.06 | SipChat | that way if no one is using sip the spare bandwidth can be allocated to other uses |
00:09.11 | jks | Nethab: for dialing in to asterisk |
00:09.16 | ariel_ | intrin, did you get the files for cvs -r 1-0 or apt-get? |
00:09.28 | SipChat | suddoer - I wouldn't recommend it but .... good software if you do want to go this way is... |
00:09.29 | sudoer | SipChat, i used to run software NAT/router on bsd/linux, but its just not as stable as dedicated hardware |
00:09.35 | Nethab | even if your not registered sipphone.com should say "User is not online leave voicemail after tone" |
00:09.53 | jks | Nethab: my regular sipphone number works fine |
00:09.53 | Nethab | and then send the wav to you email address |
00:10.04 | jks | Nethab: it's just the virtual number that I can't get working |
00:10.04 | arrgh | hmm, thats where I'm at. |
00:10.12 | jks | Nethab: is there anything special to do with a virtual number? |
00:10.19 | SipChat | suddoer - netlimiter (net limiter) or netprioritizer (net prioritizer) |
00:10.20 | arrgh | calling through sipphone.com to 7474745000 gets me the "unacceptable" message |
00:10.41 | ariel_ | intrin, I have to go and pickup the pizza for dinner I will be back maybe in about 1 hour. |
00:10.41 | jks | arrgh: what does sip show registry say? |
00:10.42 | Nethab | calling any number including 12220000000 gives me unacceptable |
00:10.51 | arrgh | I can |
00:10.53 | arrgh | tho |
00:10.59 | arrgh | call my FWD number through sipphone.com |
00:11.09 | intrin | ariel_ cvs |
00:11.10 | SipChat | suddoer - :) software routers are not quite so stable ;) |
00:11.14 | intrin | i didnt use any flags tho |
00:11.37 | ariel_ | intrin, your using head then move down to stable |
00:11.50 | sudoer | well, is there a way in case the software router/nat goes down, the comps can use an alternative gateway |
00:12.00 | intrin | how do i do that? ;/ im new to all this |
00:12.05 | ariel_ | rm -rf directory before getting the stable. |
00:12.13 | sudoer | if that is possible, i would be comfortable to use the linux box as nat box also |
00:12.16 | intrin | reinstall? |
00:12.19 | intrin | everything? :/ |
00:12.20 | *** part/#asterisk evgenyt (~Miranda@81.24.128.90) |
00:13.15 | SipChat | sudoer - that should be possible but may be geting a bit complicated. |
00:13.40 | SipChat | sudoer - you could try looking at some windows based solutions they are quite flexible and stable |
00:13.54 | cursor | stable? Windows? Same sentence? |
00:13.59 | SipChat | sudoer - however - they rarely come close to hardware (although some software solutions can easily be remotely restarted) |
00:14.06 | SipChat | I know it amused me ! ;) |
00:15.00 | Nethab | are "Not Acceptable Here" messages only generated for codec mismatches? |
00:15.01 | tessier_ | hmmm |
00:15.20 | SipChat | The other option is either a remote script to run on the other stations to termporarily restrict their bandwidth when a SIP call is being made or... |
00:15.20 | tessier_ | It would seem that if you are listening to someone elses hold music and transfer them to another line you lose the hold music |
00:15.30 | arrgh | hmm |
00:15.37 | arrgh | Nethab: that's a good question |
00:15.43 | SipChat | Use a domain controller type arrangement :). |
00:16.10 | SipChat | Nethab - I think I have seen them for other problems. |
00:16.36 | jks | arrgh: do you have a virtual number? |
00:16.41 | Nethab | i get them for any sipphone number i dial |
00:16.50 | cursor | Windows router admin: "I have to change a route, so I'll throw everyone off while I reboot" |
00:17.03 | SipChat | Nethab - what proxy and dialing format are you using etc? |
00:17.11 | redder86 | hi |
00:17.11 | SipChat | cursor ; - ) |
00:18.09 | arrgh | jks: no, still fiddling with normal number for in/out functionality |
00:18.15 | Nethab | proxy01.sipphone.com and dtmf = rfc whatever |
00:18.16 | jks | okay :-( |
00:18.39 | SipChat | nethab - what have you tried to dial at proxy01.sipphone.com ? |
00:18.39 | cursor | "I have to change a route - who stole my mouse?" |
00:18.59 | SipChat | cursor - ;) lol (use that command prompt) |
00:19.02 | SipChat | lol |
00:19.09 | SipChat | you deserve it ;) |
00:19.52 | eKo1 | why oh why did these people decide to use windows for e-mail!? @#!%!$%!%$ |
00:19.58 | SipChat | lolo |
00:20.21 | jks | I wonder what the Number Assignment feature is for, if it's not for custom/virtual numbers? |
00:20.37 | Nethab | I've tried 12220000000, my own number, my friends number, and the number that guy said above |
00:22.19 | SipChat | Hmm. ok :) |
00:22.44 | MerTech_Wes | !! anyone know how VoIP providers like Vonage/Primus address these systems (i.e. sip:xxxx@primustel.com)?? |
00:22.45 | *** join/#asterisk Zaw (~zaw@cc.cirqular.com) |
00:23.52 | SipChat | Nethab - I recall GSM was working ok for you. |
00:23.54 | SipChat | ? |
00:24.03 | Nethab | yeah and you direct dialed my * box |
00:24.12 | Nethab | but i can't dial out to sipphone |
00:25.57 | *** join/#asterisk A-Tuin (~a-tuin@steves.ip.v4.me.uk) |
00:26.20 | Nethab | My CLI says -> invite 12220000000@proxy01.sipphone.com |
00:26.28 | Nethab | then <- 100 trying |
00:26.44 | bkw_ | sounds like someone needs to learn how to setup sip.conf |
00:26.53 | Nethab | then <- 200 ok contact conference+12220000000@198.65.166.130 |
00:27.05 | *** join/#asterisk ctooley (~ctooley@015-835-274.area5.spcsdns.net) |
00:27.22 | Nethab | it worked before no changed to sip.cond |
00:27.35 | ctooley | Anyone know what to put in the Polycom ftp files to be able to register an IP500 directly to Broadvoice? |
00:27.36 | bkw_ | here |
00:27.38 | bkw_ | [sipphone] |
00:27.38 | bkw_ | type=peer |
00:27.38 | bkw_ | username=NUMBER |
00:27.38 | bkw_ | fromuser=NUMBER |
00:27.38 | bkw_ | fromdomain=proxy01.sipphone.com |
00:27.39 | bkw_ | secret=xxxxxxx |
00:27.41 | bkw_ | host=proxy01.sipphone.com |
00:27.43 | bkw_ | qualify=yes |
00:27.46 | bkw_ | then |
00:27.52 | bkw_ | dial,SIP/number@sipphone) |
00:27.57 | bkw_ | dial(SIP/number@sipphone) |
00:28.15 | Nethab | [sipphone-out] |
00:28.19 | Nethab | type = peer |
00:28.23 | Nethab | host = proxy01.sipphone.com |
00:28.33 | Nethab | username = 1747667XXXX |
00:28.40 | Nethab | secret = XXX |
00:28.51 | Nethab | fromdomain = proxy01.sipphone.com |
00:28.58 | Nethab | fromuser 1747667XXXX |
00:29.13 | cursor | http://pastebin.ca/ |
00:29.15 | Nethab | exten => _91747NXXXXXX,1,Dial(SIP/sipphone-out/${EXTEN:1}) |
00:29.22 | bkw_ | you're dialing WRONG |
00:29.29 | Nethab | it worked before |
00:29.35 | Nethab | then upgraded CVS |
00:29.37 | bkw_ | that whole SIP/peer/exten stuff |
00:29.40 | bkw_ | I don't like it |
00:29.47 | bkw_ | I do SIP/exten@peer |
00:30.49 | arrgh | I discovered that the only way to show up as "online" appears to do a sip register for sipphone.com |
00:30.59 | *** join/#asterisk Darwin35 (~darwin35@24.3.204.71) |
00:31.06 | bkw_ | ya think? |
00:31.08 | *** join/#asterisk calvis (calvis@h-66-167-52-131.sttnwaho.covad.net) |
00:31.11 | Nethab | i register |
00:31.12 | bkw_ | :P |
00:31.14 | arrgh | ;) |
00:31.19 | Nethab | but that doesn't affect dialing out |
00:31.22 | arrgh | no |
00:31.47 | arrgh | my exten lines are similar to bkw's |
00:32.12 | Nethab | no change still Not Acceptable Here |
00:32.13 | *** join/#asterisk netvulture (1000@63.174.172.245) |
00:32.25 | netvulture | who |
00:32.32 | arrgh | I can 1393612 on my sipphone.com extension and get the fwd time server |
00:32.33 | netvulture | oops - wuz up guys |
00:32.40 | calvis | I am trying to cvs checkout zaptel on the CVS server, but I keep getting cvs server: Updating zaptel It has been doing that for the past 30 min. |
00:32.49 | cursor | (netvulture) ceiling |
00:32.54 | ZX81 | ~seen sokol |
00:32.56 | jbot | i haven't seen 'sokol', ZX81 |
00:33.01 | ZX81 | ~seen ssokol |
00:33.02 | jbot | ssokol <~ssokol@64-151-42-28-dhcp-kc.everestkc.net> was last seen on IRC in channel #asterisk, 42d 23h 50m 59s ago, saying: 'they were closed when I looked 6 months ago'. |
00:33.12 | cursor | ~seen WMD in Iraq |
00:33.13 | jbot | cursor: i haven't seen 'wmd in iraq' |
00:33.18 | Nethab | bkw_: I changed my extensions.conf and still get "Not Acceptable Here" |
00:33.26 | cursor | George Dubbya should have had a jbot |
00:33.27 | netvulture | Here's a dumb questions - what is the difference between Goto and Transfer? |
00:33.38 | arrgh | I think W is bluescreened |
00:33.46 | arrgh | ever seen that blank look he gets when they fire up the teleprompter? |
00:33.55 | eKo1 | netvulture: they are spelled differently\ |
00:34.02 | Atacomm | ~seen mark actually talk on IRC |
00:34.03 | jbot | Atacomm: i haven't seen 'mark actually talk on irc' |
00:34.03 | tessier_ | netvulture: Only difference I can see is that Goto can go to different contexts whereas transfer just jumps to another extension within the same context |
00:34.09 | Nethab | Goto is a dialplan command and transfer usually wants the other side to Answer? |
00:34.15 | ZX81 | ~seen kram |
00:34.16 | jbot | kram is currently on #asterisk |
00:34.21 | ZX81 | kram == mark |
00:34.27 | cursor | really? |
00:34.34 | Atacomm | zx81: i know, that doesnt mean he talks on here |
00:34.34 | eKo1 | oh no run for cover |
00:34.43 | ZX81 | yah he does |
00:34.50 | Atacomm | zx81: for 10 seconds yes |
00:34.53 | arrgh | nethab |
00:34.55 | ZX81 | ask him a question that only he can answer and he'll answer |
00:34.56 | ZX81 | :-) |
00:34.59 | Nethab | arrgh: yes? |
00:35.03 | arrgh | try and dial out to fwd, bet it works. |
00:35.16 | cursor | What is your shoe size |
00:35.20 | ZX81 | i.e. re sysmaster |
00:35.21 | ZX81 | :-0 |
00:35.21 | Nethab | arrgh: which number? |
00:35.22 | ZX81 | lol |
00:35.25 | bkw_ | haha |
00:35.26 | bkw_ | sysmaster |
00:35.31 | ZX81 | hmmmm |
00:35.34 | arrgh | 1393612 |
00:35.43 | Atacomm | i need to call sysmaster one of these days, they were asking us to sell their systems |
00:35.51 | Atacomm | among many reasons to call sysmaster |
00:35.55 | ZX81 | I wouldn't |
00:36.05 | ZX81 | I'd rather sell windows |
00:36.26 | Nethab | arrgh: through sipphone, or through fwd |
00:36.33 | arrgh | through sipphone |
00:36.34 | bkw_ | Atacomm sysmaster is asterisk |
00:36.36 | bkw_ | :P |
00:36.38 | arrgh | of course it'll work through fwd |
00:36.38 | Atacomm | lol, i'm not gpl religious |
00:36.53 | Atacomm | i know bkw, lol |
00:36.54 | ZX81 | Atacomm: your company name? |
00:36.55 | eKo1 | man, i wish there was a command to reset all the sip/zap/iax/h323 channels |
00:36.56 | arrgh | IAX YOUR DATA . NET PROTOCOL need flat roof |
00:37.00 | bkw_ | Atacomm my job is to make sure we get everything sysmaster can do in the GPL version |
00:37.03 | bkw_ | they done pissed me off |
00:37.05 | eKo1 | without having to restart asterisk |
00:37.05 | cursor | Church of GPL |
00:37.40 | tessier_ | bkw_: You get 'em! |
00:37.42 | Atacomm | bkw: lol, good, then they wont violate the gpl and mark wont be able to hold them hostage on the "use our gear or pay big fees" theme |
00:37.55 | bkw_ | Atacomm bingo |
00:38.09 | Atacomm | bkw: lol, uh, that plays more into our camp than it does digiums |
00:38.17 | bkw_ | yes |
00:38.24 | Nethab | arrgh: it did |
00:38.28 | bkw_ | anthm and I give that bug tracker hell EVERY DAY |
00:38.31 | Nethab | arrgh: got the date |
00:39.25 | Nethab | bkw_: do you think markster will accept the patch to fix it quickly |
00:39.30 | bkw_ | ya |
00:39.36 | bkw_ | I already talked to him on the phone |
00:39.41 | Nethab | nice |
00:39.46 | bkw_ | he was in his car |
00:39.47 | bkw_ | haha |
00:40.12 | Nethab | *car crash* there goes the * development schedule |
00:40.26 | bkw_ | can more people test this http://bugs.digium.com/bug_view_page.php?bug_id=0002927 |
00:40.39 | ctooley | bkw_, gone? really, I still have all kinds of issues with our tones being detected by other phone systems |
00:40.44 | Nethab | i can try what should i test for? |
00:40.56 | brc_ | ~seen tclark |
00:40.57 | jbot | tclark is currently on #asterisk |
00:40.58 | bkw_ | ctooley using inband eh? |
00:41.27 | *** join/#asterisk czero (~h@CPE0090f800c5b0-CM00e06f166c34.cpe.net.cable.rogers.com) |
00:42.03 | bkw_ | oh that was fixed too |
00:42.06 | bkw_ | like three days ago |
00:42.16 | ctooley | bkw_, yeah, Broadvoice |
00:42.21 | bkw_ | the seq numbers were rolling over |
00:42.28 | ZX81 | anyone heard of implex systems? |
00:42.34 | bkw_ | ctooley hehe |
00:42.38 | Atacomm | nope, i've heard of implex.net (they are our isp) |
00:42.52 | ZX81 | what do they do? |
00:42.57 | ZX81 | just isp? |
00:43.14 | ZX81 | maybe is not the same one |
00:43.17 | Atacomm | implex.net is an isp... we colocate there, they do fiber, wireless, t1, dialup and dsl in the minneapolis area |
00:43.25 | ZX81 | ok |
00:43.26 | ZX81 | maybe |
00:43.30 | ZX81 | ta |
00:43.45 | ctooley | bkw_, I need the flexibility of the IP based carriers... we've been moving the phone system frequently for the last 3 weeks. |
00:43.46 | _Vile | heh heh heh heh |
00:43.52 | _Vile | luff muh lnp full # check hack |
00:44.09 | czero | eveing all |
00:44.19 | Atacomm | bkw: hows your scalability stuff coming along |
00:44.21 | bkw_ | someone buy me a broadvoice account |
00:44.27 | bkw_ | i'll fix that too |
00:44.35 | ctooley | bkw_, besides both the TDM400P |
00:44.53 | Atacomm | what does P stand for anyways ? |
00:45.02 | *** join/#asterisk implicit (~implic1t@roam-15.roam.intelenet.net) |
00:45.03 | ZX81 | ~p |
00:45.05 | jbot | somebody said p was pee |
00:45.10 | ZX81 | :-) |
00:45.11 | bkw_ | power hungry? |
00:45.13 | Atacomm | lol |
00:45.19 | implicit | heh |
00:45.24 | implicit | hi everyone |
00:45.24 | ctooley | bkw_, and the single channel card had horrible quality |
00:45.26 | modulus_ | jbot pee? |
00:45.27 | jbot | Ok. |
00:45.32 | ZX81 | lol |
00:45.32 | Nethab | P = persnickity |
00:45.34 | Atacomm | hey implicit, how goes |
00:45.54 | implicit | Atacomm: not bad |
00:45.58 | Atacomm | good good |
00:47.46 | redder86 | PCI |
00:47.48 | *** part/#asterisk eKo1 (~abc@63.245.57.70) |
00:47.52 | Atacomm | implicit: the train is getting close to the end of the tunnel :) |
00:48.01 | redder86 | yeah, as if there were ever going to be an ISA version |
00:48.14 | Atacomm | could be a VESA Local or EISA |
00:48.27 | arrgh | nethab |
00:48.43 | arrgh | try canreinvite=no |
00:48.50 | implicit | implicit: oh ya? |
00:48.51 | arrgh | in you sip.conf entry for sipphone |
00:48.56 | implicit | :-p |
00:49.08 | Nethab | arrgh: yes the fwd number worked, and canreinvite is no and notransfer is yes already |
00:49.13 | redder86 | okay, so we need TDM400I, TDM400V, and TDM400E versions in addition to TDM400P |
00:49.25 | Atacomm | implicit: talking to yourself? |
00:49.37 | implicit | i am pretty out of it right now |
00:49.41 | Atacomm | rofl |
00:49.48 | implicit | lol |
00:49.50 | implicit | hahahaha |
00:49.54 | Atacomm | atacomm: i've never addressed myself before now |
00:49.55 | Nethab | Nethab: have you seen my pants, I know i left them somewhere around here |
00:49.58 | *** join/#asterisk bdeb5 (~bdeb4@alb-24-195-238-207.nycap.rr.com) |
00:50.09 | funraps-as | guys |
00:50.26 | Atacomm | implicit: yeah, converting between 4 to 6 layers right now, meeting with some important people next week in our supply chain |
00:50.29 | funraps-as | is there a cheap FXS card out there? just need 1 port |
00:50.35 | implicit | Atacomm: nice |
00:50.43 | redder86 | funraps-as: nope |
00:50.51 | bkw_ | someone paypal me some cash to get a broadvoice account :P |
00:50.59 | redder86 | funraps-as: the least-expensive thing would be an ATA |
00:51.11 | Atacomm | implicit: my vendor got mad that one of the suppliers was falling behind, so they are flying one of their reps in-state so we can bitch together at him |
00:51.18 | funraps-as | isn't ata fxs to usb or ethernet? |
00:51.50 | Nethab | yes ata's are usually inline from rj11 to rj45 |
00:51.53 | redder86 | funraps-as: okay, sure, it's still an FXS port, though |
00:52.13 | Nethab | or usb |
00:52.58 | funraps-as | gotcha, any cheapie you recommend to try out? |
00:53.14 | arrgh | nethab check your codecs? |
00:53.35 | Nethab | disallow = all |
00:53.35 | Nethab | allow = ulaw |
00:53.35 | Nethab | allow = alaw |
00:53.35 | Nethab | allow = gsm |
00:53.36 | arrgh | I'm getting the recording at 5000 now, so, sip.conf was malconfiged |
00:53.58 | *** join/#asterisk _Simon (irc@i216-58-40-193.avalonworks.net) |
00:54.14 | _Simon | hey guys |
00:54.40 | *** join/#asterisk SipChat (jirc@m71-mp1.cvx2-b.lng.dial.ntli.net) |
00:55.00 | bkw_ | the dns resolver in asterisk is fucking lame |
00:55.09 | HellHound | anyone know the url for the SixNet IAX itsp ? |
00:55.12 | bkw_ | OH lets block everything for X seconds just for shits and giggles |
00:55.39 | _Simon | was curious if anyone got iaxclient to work in .NET? I'm getting catestrophic errors trying to call methods from the object, it won't register with regsvr32 either |
00:57.08 | tessier_ | .NET? Catastrophic errors? |
00:57.23 | _Simon | I think its because the COM object won't register |
00:58.59 | *** join/#asterisk HitTop (~Miranda@Toronto-HSE-ppp3732702.sympatico.ca) |
00:59.38 | ta[i]nted | can someone send me their broadvoice sip.conf? |
01:01.01 | *** join/#asterisk juice (~juice@mo-205-240-40-86.dyn.sprint-hsd.net) |
01:01.10 | redder86 | can someone send me their broadvoice authentication information, their credit card information, and a signed form authorizing me to use them without limits |
01:01.23 | tessier_ | ok, I am confused. |
01:01.38 | ta[i]nted | redder86: i think any reasonable person would obscure their authentication information |
01:01.52 | cursor | (tessier) I'm cursor |
01:01.53 | redder86 | hopefully |
01:02.15 | ta[i]nted | glad to hear you're so caring of strangers |
01:02.15 | Nethab | arrgh: well it's connected now, but no sound so that's some kind of progress |
01:02.17 | ta[i]nted | now stfu |
01:02.26 | implicit | ta[i]nted: i don't like reasonable people for that reason |
01:02.49 | ta[i]nted | that's reasonable |
01:03.14 | cursor | reasomable people are always so unreasonable, for some reason |
01:03.41 | redder86 | ta[i]nted: I don't use Broadvoice anyway, but I stopped handing out my confs like candy because sometime, someday, I'm gonna forget to obsure my authentication information. |
01:03.45 | Nethab | it's unreasonable to reason that reasonable people would have a reason at all, no? |
01:03.53 | tessier_ | Ok, I have a PRI... |
01:03.58 | ta[i]nted | they wouldn't be reasonable people if they were unreasonable, even for a reasonable amount of time |
01:04.00 | tessier_ | And I have 100 DID's on this PRI |
01:04.14 | tessier_ | Is it possible to have several calls all coming in at the same time to the same DID? |
01:04.33 | redder86 | tessier_: yes, or of what value would your expensive PRI be? |
01:04.44 | cursor | What reason could any reasonable person have to suspect that anyone should be reasonable to an unreasonable person? |
01:04.48 | tessier_ | redder86: I thought I had to have a separate DID for each line I wanted coming in. |
01:05.05 | redder86 | tessier_: it's only one line, 24 channels |
01:05.11 | ta[i]nted | redder86: why don't you contribute to voip-info wiki then? and rid the hassle of routine sharing with one fell swoop! |
01:05.13 | *** join/#asterisk glm2k (~glm2k@rrcs-24-199-11-45.west.biz.rr.com) |
01:05.21 | redder86 | tessier_: 100 DIDs... all independent |
01:05.40 | redder86 | ta[i]nted: too busy, sorry |
01:06.10 | ta[i]nted | did u just say that in an IRC channel? |
01:06.21 | ta[i]nted | lol |
01:06.31 | ta[i]nted | the irony |
01:07.22 | bkw_ | http://codeen.cs.princeton.edu/codns/ |
01:07.28 | _Simon | anyone know how to get the COM object to work for IAXClient? |
01:07.47 | redder86 | wikis are an abomination to all that is holy about documentation: usefulness |
01:08.19 | *** join/#asterisk drbyte (~byte@byte.fedora) |
01:09.26 | bkw_ | redder86 if people would only verify what they write before putting it on the wiki |
01:09.44 | bkw_ | and by verify.. I mean read the code |
01:10.11 | Nethab | wiki's are ok, if you carry around your own grains of salt |
01:10.16 | redder86 | it's the root problem of a wiki, though, the design is to allow the lay-person the ability to write documentation for something for which they are not really "authorized" to do |
01:10.25 | file[laptop] | let's all go to the lobby |
01:11.02 | libpcp | hi all |
01:11.05 | czero | file[laptop] your not goign to start disco dancing agian are you |
01:11.17 | ta[i]nted | i propose an electoral wiki.. |
01:11.40 | libpcp | in latest cvs of zapata, does wcfxs replace by wctdm ? |
01:12.02 | libpcp | i couldnt find the wcfxs in the Makefile |
01:12.07 | Nethab | but then the red wiki states would have just as much say as the blue wiki states |
01:12.56 | Equinox | If I have *#1 registering as a peer to *#2.. *#2 needs to have a user entry, right?(IAX2) |
01:13.23 | libpcp | ah okay, the wcfxs was alias to wctdm |
01:13.44 | *** join/#asterisk drazvan (~drazvan@romsat011.fx.ro) |
01:13.59 | Nethab | equinox: yes otherwise #2 would show up as guest |
01:14.14 | Equinox | Nethab- Well, I wanted it unidirectional |
01:14.20 | drazvan | hi everyone. could anyone help me with some ANI + Asterisk questions? |
01:14.21 | Equinox | #1 -> #2 |
01:14.28 | Nethab | you should try trunking |
01:14.38 | Equinox | What advantage does trunking have? |
01:14.40 | redder86 | it would be nice if the people who wrote the code actually documented it outside of the code (in addition to inside it) |
01:15.04 | Nethab | less overhead per call if you make several calls between systems |
01:15.08 | Equinox | <PROTECTED> |
01:15.11 | Nethab | otherwise none |
01:15.15 | *** join/#asterisk Kumbang (~kumbang@167.205.22.54) |
01:15.23 | Equinox | Nethab- So without trunking the reduction in overhead isn't there? |
01:15.50 | *** join/#asterisk JerJer (~mine@d2-236.rt-bras.che.centurytel.net) |
01:16.17 | brc_ | ~seen JerJer |
01:16.19 | jbot | jerjer is currently on #asterisk (29s) |
01:16.19 | Nethab | if one * server is the main system, and has the connection to the PSTN or the main voicemail box, and calls from the second box need to go through it to get to the PSTN then trunking is very good |
01:16.20 | brc_ | er...doh |
01:16.35 | bkw_ | libpcp yes wctdm replaces wcfxs |
01:16.37 | bdeb5 | anyone here been able to unlock the pap2? |
01:16.44 | ta[i]nted | it would be nice if i was fellated regularly by random attractive women on a daily basis |
01:16.52 | _Simon | anyone here use IAXClient COM object? |
01:16.58 | redder86 | bdeb5: it's not locked |
01:17.01 | ta[i]nted | but we all have dreams, don't we |
01:17.03 | Nethab | i disbabled my COM service sorry |
01:17.21 | bdeb5 | redder86: it's locked into vonage, i can't get in to set my sip hostname, etc |
01:17.23 | redder86 | bdeb5: but don't count on anyone here telling you how to use it with Asterisk |
01:17.31 | _Simon | Nethab: hmm? |
01:17.42 | Nethab | i was being facetious |
01:17.50 | bdeb5 | redder86: how come? |
01:18.15 | _Simon | hehe ok, well I can't get the IAX ActiveX object to work |
01:18.28 | redder86 | bdeb5: "locked" would indicate that something in the firmware is forcing it to Vonage, and that's not true, it pulls a configuration file and gets the Vonage config from the config file. |
01:18.32 | ta[i]nted | bdeb5: that would be hax0ring |
01:18.37 | redder86 | bdeb5: those who know aren't telling |
01:18.43 | *** join/#asterisk PoWeRKiLL (~PoWeRKiLL@l02m-6-201.d4.club-internet.fr) |
01:18.55 | *** part/#asterisk drazvan (~drazvan@romsat011.fx.ro) |
01:18.57 | bdeb5 | og |
01:18.59 | ZX81 | what is the best interface to use with the Fritz cards? i.e. chan_capi, zaphfc etc |
01:18.59 | bdeb5 | oh |
01:19.07 | Nethab | everything i know i learned from the wiki |
01:19.12 | ZX81 | assuming European BRI |
01:19.18 | redder86 | bdeb5: the cardinal rule about hacking stuff and keeping your hack functional: don't tell anyone about it |
01:19.30 | bdeb5 | i've been able to roll back the firmware and get it to download the xml file, but it wont take the changes - it needs to be salted i think |
01:19.55 | bdeb5 | i see |
01:20.18 | ta[i]nted | salt your ATA, grasshoppa |
01:20.24 | cursor | http://www.cursor.biz/tmp/flame_form.txt |
01:20.33 | redder86 | salt on slugs is better than salt on ATAS |
01:22.12 | cursor | Not if you believe what they say in the adverts on British TV |
01:22.48 | redder86 | what, that slugs are good for the earth? |
01:23.13 | bkw_ | ok the dns resolver lib in asterisk needs to be replaced with something that doesn't block |
01:23.16 | redder86 | they eat decaying plant matter, help the food chain, save the whales? |
01:23.18 | ta[i]nted | cursor: i <3 flame_form |
01:23.32 | *** join/#asterisk david (~david@muffin.davidcoulson.net) |
01:23.33 | cursor | :-) |
01:23.51 | redder86 | bkw_: the IAXy just plain needs a resolver |
01:24.03 | ta[i]nted | someone needs Preparation H |
01:24.09 | redder86 | hehe |
01:24.14 | cursor | What's that? |
01:24.15 | ZX81 | bkw_: agree |
01:24.24 | cursor | http://www.rtfm.info/tmp/flame.txt |
01:25.01 | bkw_ | http://www.gnu.org/software/adns/ |
01:25.28 | redder86 | bkw_: they won't sign the disclaimer |
01:26.33 | nestAr | cursor: you might want to add "DIE IN A FIRE" to the FURTHERMORE: section.. :) |
01:26.51 | Equinox | I'm trying to get IAX2 working from my home machine, behind a Linksys NAT router, to another asterisk server on a DS-3.. This isn't working and I don't seem to see any debug information saying why. Any suggestions? |
01:28.00 | Nethab | if your behing NAT you should really forward the ports to the * box behind the linksys |
01:28.11 | redder86 | UDP port 4569 |
01:28.14 | bkw_ | you shouldn't have to |
01:28.24 | cursor | disclaimer? what's that then? |
01:28.29 | redder86 | for incoming calls you have to, unless you register, then you shouldn't have to. |
01:28.33 | cursor | don't answer that :-) |
01:28.35 | Equinox | I'm registering. |
01:28.41 | Nethab | if you originate calls from the non nat box to the nat box you should have the ports forwarded |
01:28.49 | Equinox | Even with registering? |
01:28.59 | Equinox | Actually I'm mostly concerned about outgoing calls from nat. |
01:29.03 | redder86 | Equinox: I forward the ports even though I register. |
01:29.27 | Equinox | Can't hurt.. 1 sec |
01:29.32 | Nethab | if you don't reinvite and leave the * box behind the nat in the media path you should be fine |
01:30.21 | ZX81 | does anyone have any questions they would like asked of Mark Spencer for an interview for the Asterisk Daily News? |
01:30.28 | brc_ | yes |
01:30.39 | ZX81 | mail me? |
01:30.39 | brc_ | what is the short term plan for 1.1 |
01:30.43 | ZX81 | ok |
01:30.44 | brc_ | and what is the long term vision for 2.0 |
01:31.00 | ZX81 | :-) |
01:31.15 | brc_ | there's tons of new stuff in head |
01:31.17 | Nethab | Equinox: i forwarded UDP/5060 for SIP as well |
01:31.33 | Nethab | brc_: like what new stuff? |
01:31.41 | brc_ | read the cvs list |
01:31.49 | Equinox | Where is the cvs list? |
01:31.50 | Nethab | i did didn't mean much to me |
01:32.00 | brc_ | Equinox, same place as all the mailing lists |
01:32.03 | Nethab | lists.digium.com |
01:32.03 | brc_ | asterisk.org |
01:32.05 | brc_ | look hard |
01:32.17 | brc_ | http://www.asterisk.org/index.php?menu=support |
01:32.17 | Equinox | Still not working. |
01:32.46 | Nethab | Equinox: are you in the * console (CLI)? |
01:32.50 | cursor | <brc_> read the cvs list |
01:32.51 | cursor | haha |
01:32.52 | cursor | "oops" |
01:33.10 | Mavvie | ZX81: I have a lot of questions, but I don't think any of them makes sense. |
01:33.17 | ZX81 | :-) |
01:33.20 | Equinox | Nethab- on both machines |
01:33.22 | ZX81 | mail to editor@sineapps.com |
01:33.31 | brc_ | cursor, wtf? |
01:33.32 | *** join/#asterisk michael12345 (mick1234@202.43.239.10) |
01:34.10 | cursor | Useless commit message of the week: |
01:34.11 | cursor | --- |
01:34.11 | cursor | Modified Files: |
01:34.11 | cursor | pbx_realtime.c |
01:34.11 | cursor | Log Message: |
01:34.11 | cursor | Realtime improvements |
01:34.12 | cursor | --- |
01:34.25 | michael12345 | anyone got the new oh323 module to compile |
01:34.26 | Nethab | and you don't get any messages with 'set verbose 4' or iax2 debug? |
01:34.40 | michael12345 | 0.70 |
01:34.40 | drumkilla | cursor: oh come on ... it is exactly what he said it was :p |
01:34.44 | cursor | :-) |
01:34.46 | brc_ | luke, read the code...ya know...the stuff right under that message |
01:34.59 | Equinox | I get some IAX2 messages, but they don't tell me anything. The set verbose thing same as -vvvv? |
01:35.05 | Nethab | yes |
01:35.07 | cursor | ok - how about this one, which was last week's useless commit message of the week: |
01:35.08 | cursor | ---Modified Files: |
01:35.08 | cursor | logger.c |
01:35.08 | cursor | Log Message: |
01:35.09 | cursor | Fix little logging issue |
01:35.10 | cursor | --- |
01:35.23 | drumkilla | well there you go |
01:35.28 | drumkilla | obviously it was pretty little |
01:35.32 | drumkilla | so you don't really need to worry. |
01:35.41 | michael12345 | please anyone |
01:35.41 | drumkilla | :) |
01:35.54 | brc_ | michael12345, contact the authors... |
01:35.56 | drumkilla | michael12345: i don't use it |
01:35.58 | brc_ | use cvs head |
01:36.05 | Nethab | because some of us are low level programmers and res = chan->pvt->setoption(chan,... means nothing to us |
01:36.08 | Equinox | Hmm |
01:36.11 | Equinox | Fixed it |
01:36.18 | Mavvie | cursor: and he wants us to read these to figure out what has changed in the configuration files.... |
01:36.19 | Equinox | Apparently Asterisk isn't happy being on a 2nd interface IP.. |
01:36.25 | michael12345 | brc_ are you using it |
01:36.29 | brc_ | no |
01:36.41 | Mavvie | Equinox: use bind addresses. |
01:36.57 | Equinox | Bind = 0.0.0.0 won't do it? |
01:37.03 | Equinox | Can I have multiple bind statements? |
01:37.12 | Mavvie | not if you want the eth0:1 |
01:37.13 | Nethab | bindaddr = 0.0.0.0 |
01:37.30 | Equinox | That's default isn't it? |
01:37.32 | cursor | 0.0.0.0 should still work |
01:38.43 | cursor | <Equinox> Can I have multiple bind statements? |
01:38.43 | cursor | no |
01:39.14 | Mavvie | lsof -n | grep asterisk | grep -E '(TC|UD)P' <- should all be bound on your eth0:1 IP address. |
01:39.36 | cursor | lsof: command not found |
01:39.37 | cursor | :-) |
01:39.39 | JerJer | 0.0.0.0 will work in IAX and SIP, maybe MGCP and most certianly NOT chan_h323 |
01:39.43 | drbyte | hello all. i have a Winbond 6692, and i'd like to configure asterisk to make use of that |
01:39.44 | Equinox | udp 0 0 0.0.0.0:4569 0.0.0.0:* |
01:39.48 | Equinox | Netstat -n -l |
01:40.08 | timecop | heh |
01:40.18 | timecop | asterisk behind nat = waste of time. |
01:40.20 | timecop | give up now. |
01:40.25 | Mavvie | Equinox: doesn't five you the process name. |
01:40.41 | Equinox | timecop- My main asterisk is on a static |
01:40.44 | Equinox | timecop- This is my home |
01:40.48 | Nethab | fuser 4569/udp? |
01:40.49 | Mavvie | I'm only interested in the asterisk side of the story. |
01:40.59 | timecop | equi: its still a waste of time |
01:41.03 | cursor | * sys-apps/lsof |
01:41.03 | cursor | <PROTECTED> |
01:41.03 | cursor | <PROTECTED> |
01:41.03 | cursor | <PROTECTED> |
01:41.10 | Equinox | Why? Outgoing calls to my other box? |
01:41.14 | timecop | use IAX. |
01:41.23 | drbyte | cursor: what rpm are you using? i.e. which repository is it from? i'm rebuilding from fedora.us to make it generally work |
01:41.26 | timecop | or better yet |
01:41.31 | timecop | give your "home machine" a external IP. |
01:41.37 | Nethab | my * box is behind a Linksys NAT and i can make calls and receieve them all day long |
01:41.46 | JerJer | LFS |
01:41.47 | Optic2 | hey |
01:41.49 | Equinox | timecop- I wanted to experiment with NAT. |
01:41.51 | bkw_ | yo yo yo |
01:41.57 | JerJer | tar zxf blah.tar.gz |
01:41.58 | Optic2 | what would cause like 2 seconds of lag on an iax conversation? :) |
01:41.58 | Equinox | Actually the problem isn't NAT at all.. |
01:42.02 | Equinox | It's asterisk not working on eth0:1! |
01:42.03 | Mavvie | cursor: only on linux machines :-) |
01:42.03 | Optic2 | audio is perfect, but very delayed |
01:42.03 | JerJer | ./configure && make install |
01:42.05 | Nethab | externip = <your ip> |
01:42.11 | Equinox | It works fine if I point the other config file to eth0... |
01:42.12 | Mavvie | Equinox: works fine here. |
01:42.12 | timecop | equin: ugh |
01:42.14 | Equinox | Is this a bug? |
01:42.18 | bkw_ | localnet=localnet/localmask |
01:42.22 | bkw_ | oh |
01:42.24 | bkw_ | FYI |
01:42.27 | Equinox | Mavvie- You have a default gw on eth0:1? |
01:42.35 | bkw_ | you can't do auth on a virtual interface using iax2 |
01:42.36 | Nethab | what's your route -n say |
01:42.43 | bkw_ | if say you have eth0 and eth0:1 |
01:42.52 | bkw_ | and you try to register to the ip on eth0:1 |
01:42.52 | Mavvie | Equinox: no, default gateway is pointing to eth0 IP address. |
01:42.54 | bkw_ | YOU CANT |
01:42.57 | Equinox | Oooh |
01:43.00 | _Simon | anyone here use the ActiveX IAX client? |
01:43.03 | *** join/#asterisk czero (~h@CPE0090f800c5b0-CM00e06f166c34.cpe.net.cable.rogers.com) |
01:43.04 | Equinox | Well that explains it. |
01:43.07 | bkw_ | its a bug |
01:43.11 | czero | kelw XP Pro bluescreens on MeTaBSD |
01:43.11 | bkw_ | I have been screaming about it for months |
01:43.19 | bkw_ | but nobody seems to think its really a bug |
01:43.19 | czero | first aht that has happended in a LONG time |
01:43.29 | bkw_ | Equinox are you getting hit by that? |
01:43.31 | znoG | shiiit, 37000 messages in my asterisk folder |
01:43.34 | Mavvie | bkw_: can you elaborate on that one? |
01:43.36 | znoG | time to delete 85% of them |
01:43.53 | bkw_ | eth0 = 192.168.1.1 and eth0:1 = 192.168.1.2 |
01:43.57 | cursor | folder - ugh |
01:44.02 | bkw_ | chan_iax2 is on 0.0.0.0 |
01:44.05 | bkw_ | ie all right? |
01:44.20 | bkw_ | if you try to register or do anything that requires auth on eth0:1 IT WILL FAIL |
01:44.40 | Mavvie | bkw_: is your subnet mask on eth0:1 /32 ? |
01:44.48 | bkw_ | doesn't matter |
01:44.49 | Equinox | I tried /32 and /30 |
01:44.53 | Equinox | Just for the record ;) |
01:44.57 | bkw_ | its a flaw in chan_iax2 |
01:45.18 | bkw_ | but I know it does give us hell |
01:45.19 | bkw_ | :P |
01:45.20 | Equinox | Glad it's a bug |
01:45.23 | Equinox | Was driving me insane. |
01:45.25 | tessier_ | redder86: I just talked to a PRI guru and he suspects they had each DID set up as an individual call route and not as a trunk and then changed it when I called up wanting a hunt group. |
01:45.35 | bkw_ | Equinox try a real ip on a real interface |
01:45.44 | Equinox | They are both real IPs. |
01:45.49 | bkw_ | ok try the other one |
01:45.50 | bkw_ | haha |
01:45.51 | Equinox | But I'm switching to the non aliased eth :) |
01:45.53 | bkw_ | the one thats not an alias |
01:46.17 | Mavvie | bkw_: it seems to work fine here, can you give some more information on how I can prove it is not working? |
01:46.18 | Equinox | bkw_ Yes.. You're right. I figured it out about 2 sec before you said it's a bug |
01:46.24 | Equinox | bkw_ Of course it's been torturing me for days . . . |
01:46.44 | bkw_ | Mavvie can you make calls and register to an aliased ip? |
01:47.09 | sudoer2 | is it possible to get a business address that isnt a po box and you dont live there? |
01:47.22 | bkw_ | sudoer mailboxes etc |
01:47.37 | cursor | Why not get a business address that points to your home/business? |
01:47.51 | Mavvie | bkw_: I think so. |
01:48.11 | bkw_ | try it to be sure |
01:48.33 | Mavvie | bkw_: I want to know where you think it is going wrong, since I can't see it here. |
01:49.15 | Optic2 | 3 second lag at the end of a 5 minute call |
01:49.24 | Optic2 | almsot no lag at the beginning |
01:49.26 | Mavvie | Optic2: same here. |
01:49.27 | Optic2 | jitterbuffer off |
01:49.40 | Optic2 | iaxComm <-> asterisk <-> sip |
01:49.48 | Equinox | If there is no auth=whatever statement are plaintext pws exhcnaged? |
01:50.16 | tessier_ | oh jeezus freakin' christ |
01:50.24 | tessier_ | The email said DO NOT discuss it on the mailing list. |
01:50.35 | Equinox | Mavvie- Um.. It isn't working. |
01:50.39 | Equinox | Mavvie- Trust me :) |
01:50.45 | Equinox | Mavvie- At least, in bkw and my case. |
01:50.49 | sudoer2 | cursor, i dont want people to have my home address |
01:50.50 | *** join/#asterisk Nethab (~Nethab@adsl-67-113-141-170.dsl.sntc01.pacbell.net) |
01:50.54 | Equinox | Mavvie- The one thing I didn't try was putting a route out eth0:1. |
01:51.03 | Mavvie | Equinox: yes, and then I ask for more information so I can emulate it here. |
01:51.13 | Equinox | Mavvie- Do you have a route out of your eth0? |
01:51.16 | Equinox | Mavvie- Do you have a route out of your eth0:1? |
01:51.29 | cursor | Don't you have a legal obligation to reveal your business address? |
01:51.33 | Mavvie | Equinox: eth0 |
01:51.36 | Nethab | The problem is when registering to the external * box the src IP will always be eth0 not 0:1 |
01:51.36 | cursor | so people can send you official letters etc. |
01:51.39 | Equinox | cursor- Your registered agent. |
01:51.50 | cursor | ok |
01:51.56 | cursor | It'd obviously different over here |
01:51.56 | Equinox | cursor- A few states have a minimum of a RA. |
01:52.02 | Equinox | cursor- Where ru? |
01:52.08 | cursor | England |
01:52.11 | Equinox | But in Florida my business address is public record. |
01:52.20 | Equinox | And most places in the US is the same. |
01:52.27 | cursor | ok |
01:52.47 | Nethab | the default route which is what is used to find the other * server says to go through eth0 not the IP alias on eth0:1 |
01:52.49 | cursor | In England, you have to reveal your "registered address" |
01:52.57 | cursor | even if that's different from your office address |
01:53.03 | Equinox | sudoer2- Just rent an office? |
01:53.10 | Equinox | sudoer2- It's 1. A business address, and 2. You don't live there. |
01:53.14 | Equinox | Your qualifications :) |
01:53.14 | yxa | anyone knows of a online site that sells digium products and ships internationally as well? |
01:53.15 | cursor | so one company might have several offices, but only one "registered address" |
01:53.21 | Equinox | cursor- Sure. |
01:53.25 | Mavvie | Nethab: yes, but if it bound to an IP address it will take that one anyway. |
01:53.28 | Optic2 | reading iax2 protocol doc |
01:53.32 | Equinox | cursor- The registered agent is there so you can sue them. |
01:53.42 | cursor | ok |
01:53.48 | sudoer2 | Equinox, its not a bad idea, unfortunately thats too exspensive for me |
01:53.51 | cursor | Again, slightly different over here :-) |
01:53.57 | Equinox | sudoer2- Then use your home addy.. I do ;) |
01:54.02 | Equinox | cursor- How so? |
01:54.22 | cursor | We don't tend to run around sueing one another |
01:54.27 | Equinox | I think it's a crime if your registerd agent isn't valid. |
01:54.28 | Mavvie | Equinox: can you show me the output of that lsof command I just pasted? |
01:54.31 | Equinox | Yes, I've heard that. |
01:54.41 | Equinox | mavvie- lemme find it again |
01:54.48 | Equinox | Repaste it? :) |
01:55.01 | Mavvie | lsof -n | grep asterisk | grep -E '(TC|UD)P' |
01:55.07 | timecop | so how do I get asterisk to setup for sccp |
01:55.11 | Nethab | lsof -n | grep UDP |
01:55.18 | cursor | Perhaps your accountant will allow you to use its address as your registered office |
01:55.47 | Equinox | Or your lawyer. |
01:55.58 | jks | Anyone here got a SIPPhone Virtual Number working? (with Asterisk) |
01:56.00 | cursor | The registered address (in England) is used for tax reasons - it's the address the tax man uses to send stuff |
01:56.18 | cursor | and the address anyone should be able to use to send stuff |
01:56.40 | cursor | so your accountant could be a good choice |
01:56.48 | Nethab | jks: nope, i can't even get regular calls to sipphone working right |
01:56.49 | cursor | (for a fee - probably) |
01:58.07 | *** join/#asterisk Equinox (~secret@star.l93.com) |
01:58.16 | cursor | Excess flood? |
01:58.17 | Equinox | If u want more just give me an email ;) |
01:58.21 | cursor | Are you in Florida? |
01:58.25 | Equinox | Yeah I pasted that thing mavvie wanted |
01:58.27 | Equinox | And owned myself |
01:58.32 | Mavvie | Equinox: port 2727 and port 4569 aren't bound to an IP address. |
01:58.33 | cursor | pastebin |
01:58.45 | Equinox | asterisk 22968 asterisk 15u IPv4 32679767 UDP *:2727 |
01:58.45 | Equinox | asterisk 22968 asterisk 16u IPv4 32679768 UDP *:4569 |
01:58.46 | Nethab | asterisk 18394 root 8u IPv4 242563 UDP *:4569 |
01:58.51 | Equinox | Mavvie- They are |
01:58.57 | Nethab | that means all ips |
01:58.58 | Equinox | Mavvie- I just died before you got those two lines |
01:59.00 | Nethab | not none |
01:59.02 | *** join/#asterisk syslod (~sysglod@65.114.0.198) |
01:59.07 | Mavvie | if they are they wouldn't be a * there |
01:59.11 | jks | noone using sipphone.com here? |
01:59.14 | Equinox | * means all? |
01:59.16 | Nethab | i am |
01:59.21 | Nethab | but not virtual bumbers |
01:59.21 | Mavvie | Equinox: yes |
01:59.22 | Gand_DJ | I think I have my asterisk server setup to link to fwd. |
01:59.29 | cursor | somewhere |
01:59.31 | Gand_DJ | outgoing calls work so far :) |
01:59.49 | Equinox | tcp 0 0 0.0.0.0:22 0.0.0.0:* LISTEN |
01:59.53 | Equinox | Ssh works just fine . . . |
01:59.54 | Equinox | (on all Ips) |
01:59.55 | jks | hmm, I think I wasted 69.99$ on my virtual number then :-( |
02:00.07 | Equinox | jks- What happened? |
02:00.07 | Mavvie | yes, ssh listens on every port. |
02:00.14 | jks | Equinox: well, it doesn't work |
02:00.15 | Gand_DJ | I use sipphone.com.... but don't pay for anything on it |
02:00.18 | Nethab | that's netstat not lsof silly |
02:00.24 | Equinox | jks- what's a virtual number? |
02:00.26 | jks | Equinox: my regular sipphone number works fine with asterisk |
02:00.28 | Equinox | Well yeah |
02:00.28 | Gand_DJ | I like fwd better |
02:00.36 | cursor | It's almost a number, but not |
02:00.36 | Equinox | But won't netstat show the listening ports? |
02:00.38 | jks | Equinox: virtual number = a real US number that can be dialed from PSTn |
02:00.50 | cursor | DDI |
02:01.03 | jks | cursor: sorry? |
02:01.06 | Nethab | the virtual number should simply forward to your sipphone number no difference, if the virtualnumber says User unknown then something else is wrong |
02:01.10 | Nethab | on sipphone.com's end |
02:01.11 | cursor | Direct Dialin number |
02:01.14 | Equinox | sshd 196 root 3u IPv4 350 TCP *:ssh (LISTEN) |
02:01.16 | Equinox | Fine |
02:01.17 | Equinox | There :) |
02:01.17 | jks | Nethab: how do you know that? |
02:01.19 | Equinox | Hehe |
02:01.20 | jks | cursor: thanks |
02:01.42 | Nethab | because either way, the connection should come through proxy01.sipphone.com |
02:01.50 | Nethab | or your local proxy |
02:01.59 | Mavvie | Equinox: for your iax.conf: |
02:02.03 | jks | Nethab: how do you know that? |
02:02.05 | Mavvie | [general] |
02:02.05 | Mavvie | bindport=4569 |
02:02.05 | Mavvie | bindaddr=218.1.2.3 |
02:02.12 | jks | Nethab: i.e. are you just guessing or do you know this/tried this? |
02:02.14 | cursor | are you having trouble with sipphone in or out? |
02:02.16 | cursor | or both? |
02:02.19 | Equinox | Mavvie- Actually I tried binding it directly to that IP, didn't work. |
02:02.25 | jks | cursor: regular sip phone works fine for me. |
02:02.26 | Equinox | Mavvie- I've tried lots of things ;) |
02:02.30 | Mavvie | Equinox: what didn't work about it? |
02:02.35 | Mavvie | the binding ? |
02:02.37 | Equinox | It was just as broken as it is now |
02:02.38 | jks | cursor: it's just this new number, I can't dial into it from PSTN.. it just says user unknown |
02:02.41 | Nethab | because the PSTN incoming termination is happeneing before it gets to sipphone that's how they know it's your number |
02:02.43 | Equinox | It wouldn't register |
02:02.44 | Equinox | Or pass traffic |
02:02.52 | jks | Nethab: okay, so you're just guessing |
02:03.06 | Nethab | well it's not terminating at your house |
02:03.07 | cursor | What does your register line look like in sip.conf? |
02:03.08 | Mavvie | can you, before you try again, please make sure they're *all* bound to an IP address? |
02:03.10 | jks | Nethab: i.e. I'm looking for a "missing link" here... something which must be done, which I didn't do |
02:03.23 | jks | Nethab: like for example I have to do number assignment from a web control panel, or I need to register with a new username or something |
02:03.30 | jks | cursor: like regular? |
02:03.36 | jks | cursor: do I need to change it? |
02:03.41 | cursor | Should be: register => phoneno:password@proxy01.sipphone.com/phoneno |
02:03.45 | Equinox | jks- Using sip or IAX? You using asterisk or a sip phone? |
02:04.04 | jks | Equinox: SIP, Asterisk |
02:04.29 | Nethab | even if you register using x-lite, the PSTN number should ring you through sipphone |
02:04.34 | cursor | You'll also need a type=user section |
02:04.39 | cursor | with username = phoneno |
02:04.53 | jks | cursor: yes, I have already have all that working |
02:04.54 | Equinox | Mavvie- The same setup works great if I go to the eth0 IP |
02:04.56 | Nethab | did you use the my.sipphone.com to get the virtual number? |
02:04.57 | jks | cursor: with my regular sip phone number |
02:05.01 | jks | cursor: which works fine |
02:05.08 | jks | cursor: now I bought a DDI number as an add-on |
02:05.13 | jks | cursor: do I need to change anything then? |
02:05.23 | timecop | hm |
02:05.26 | cursor | I don't know, but I suspect so |
02:05.28 | Mavvie | Equinox: then use the eth0 IP. |
02:05.31 | jks | cursor: I tried adding the DDI number the same way as the regular sip number, but Sipphone.com won't let me register then |
02:05.31 | timecop | cant get this cisco communucator thing to do anythign with asterisk |
02:05.34 | cursor | did you get a second account with the second number? |
02:05.38 | jks | Nethab: yes? |
02:05.40 | Equinox | Mavvie- I am.. I switched the DNS for the 2nd IP to the eth0 IP |
02:05.41 | cursor | The number I have is used as the username |
02:05.46 | jks | cursor: no, it's an addon to the same account somehow |
02:05.49 | Equinox | Mavvie- Just kind of.. Strange. Glad it's a known bug tho |
02:05.49 | cursor | so perhaps you'd need two accounts for two usernames |
02:05.51 | jks | cursor: same here |
02:06.03 | cursor | set up a second type=user block |
02:06.06 | jks | cursor: I tried using my new number as the account name, but that didn't work |
02:06.27 | Nethab | and under the "Premium features to your account" it shows your Virtual number |
02:06.29 | jks | cursor: I can't authenticate with the new number as account name, so it won't owrk. |
02:06.36 | cursor | ok |
02:06.43 | cursor | you'll need to call sipphone support then |
02:06.44 | jks | Nethab: There's no Premium features menu? |
02:06.45 | Equinox | I wonder if they just bind the 2nd # to your first account? |
02:06.48 | cursor | they'll probably know |
02:06.50 | Equinox | So you have 2 "lines" on your sip phone? |
02:06.51 | jks | cursor: I've written a ticket... they don't reply |
02:06.55 | Nethab | no use your sipphone user as the registry user name |
02:07.05 | cursor | 2:06am - give them time :-) |
02:07.08 | *** join/#asterisk tris (tristan@camel.ethereal.net) |
02:07.10 | jks | Nethab: sorry? |
02:07.18 | jks | cursor: well, I wrote in two days ago |
02:07.22 | jks | cursor: still no reply |
02:07.23 | timecop | this shit isnt working at all |
02:07.24 | timecop | ugh |
02:07.27 | Nethab | you only need to register to sipphone as 1747 whatever |
02:07.36 | cursor | phone them |
02:07.49 | jks | Nethab: done that |
02:07.53 | jks | Nethab: what more do I need then? |
02:08.06 | jks | cursor: I'll try that then... I just couldn't find their number the other day |
02:08.07 | cursor | What does your register line look like in sip.conf? |
02:08.19 | Nethab | the virtual number should get translated by their PSTN provider and sent through sipphone to your box |
02:08.32 | jks | Nethab: ofcourse |
02:08.45 | *** join/#asterisk Verliba (~Miranda@65-100-56-22.ptld.qwest.net) |
02:08.46 | jks | cursor: register => 1747667xxxx:9634@proxy01.sipphone.com/1000 |
02:08.54 | cursor | ok |
02:09.02 | *** join/#asterisk w0w0 (~apardo@80.26.167.46) |
02:09.11 | Nethab | the only thing i would try is don't register with asterisk, trying registering with a softpohone on your PC and see if that rings from the virtual number |
02:09.29 | Nethab | if not then it's on sipphones side |
02:10.07 | jks | Nethab: well, either that - or there's something I don't know |
02:10.22 | jks | I was wondering if there was any kind of "extra clue" you could send along in the sip registration to sign in for more than one number |
02:10.26 | cursor | This message probably won't change while you're staring at it. |
02:10.36 | Nethab | not really, X-lite and other softphones should receive the call |
02:10.37 | jks | or something I needed to do in their my.sipphone.com to assign the new virtual number to a specific sip phone number |
02:10.50 | cursor | This message is slightly different than the one that was here a minute ago. |
02:10.51 | Nethab | do you have more than one account? |
02:10.57 | Nethab | on sipphone? |
02:11.07 | jks | Nethab: yes, but that doesn't have anything to do with it |
02:11.12 | *** join/#asterisk Darwin35 (~darwin35@c-24-3-204-71.client.comcast.net) |
02:11.13 | Nethab | you can have multiple register commands |
02:11.19 | jks | Nethab: I know. |
02:11.54 | drbyte | i've setup asterisk like this: http://www.voip-info.org/wiki-Asterisk+quickstart, and am using kphone. but the registration repeatedly fails :( |
02:12.16 | Nethab | jks: if a softphone can't receive the call then it's not your fault |
02:12.31 | jks | Nethab: It could easily be for the reasons, I've just lined out. |
02:12.33 | cursor | I'm going to take a break for a while |
02:12.35 | Darwin35 | ok having some problems with trunking |
02:12.35 | cursor | back later |
02:12.37 | cursor | probably |
02:12.50 | Darwin35 | I add the needed info to iax.conf |
02:13.09 | Darwin35 | and the 2 boxes reject each other |
02:13.10 | *** join/#asterisk oncemore (~oncemore@222.33.36.198) |
02:13.30 | Nethab | jks: if the virtual number appears under the "Premium features on Your account" and it says status is on then you've done your part |
02:13.39 | timecop | so i'm still running 08/05/03 CVS of asterisk because anything past that (not sure of exact date) fails registering with my SIP providers. |
02:13.50 | timecop | and i'm now trying to use chan_skinny and its failing hard |
02:13.50 | jks | Nethab: as I said before, there's nothing in my.sipphone.com that says "Premium Features" |
02:14.01 | timecop | any suggestions on what to od? |
02:14.14 | jks | Nethab: okay, I found what you said now... it wasn't in the menu |
02:14.23 | jks | Nethab: it lists the number there fine... but as a Custom Number? |
02:14.33 | timecop | so much for opensores help |
02:14.37 | jks | Nethab: the status show "on line" |
02:14.40 | Nethab | jks: on my my.sipphone.com page i see "Basic Features on Your account"Basic Voicemail on |
02:14.44 | jks | Nethab: do you know the phone numebr for their setup ? |
02:14.46 | oncemore | hi |
02:15.07 | oncemore | how to make busydetect work? |
02:15.31 | Nethab | jks: if you dial ** into sipphone.com you should get your number back? |
02:15.38 | Nethab | is that what you mean |
02:15.46 | *** join/#asterisk funraps-as (~funraps@adsl-64-164-82-84.dsl.lsan03.pacbell.net) |
02:15.49 | oncemore | i set busydetect=yes in zapata.conf ,but it does not wok |
02:16.06 | jks | Nethab: I get my number back.. the sip phone number (1-747-etc) |
02:16.14 | jks | Nethab: sorry, not setp |
02:16.17 | Nethab | that means your connected |
02:16.19 | jks | Nethab: I meant, the number for their _support_ |
02:17.48 | Nethab | jks: not offhand sorry |
02:19.05 | oncemore | who can help me? |
02:19.19 | funraps-as | hi guys |
02:19.21 | oncemore | make busydetect work. |
02:19.22 | funraps-as | I got dissed |
02:19.39 | Nethab | i've never used busydetect sorry |
02:19.41 | *** join/#asterisk routerheads_atho (~root@user-12l2p2p.cable.mindspring.com) |
02:19.47 | funraps-as | is there a document online that shows ALL of the asterisk extensions.conf commands? |
02:20.04 | *** part/#asterisk routerheads_atho (~root@user-12l2p2p.cable.mindspring.com) |
02:20.12 | oncemore | O ~~ x100p can not detect pstn phone hungup |
02:20.21 | oncemore | o |
02:20.23 | *** join/#asterisk NormAst (NormAst@Toronto-HSE-ppp3677946.sympatico.ca) |
02:20.29 | oncemore | thanks |
02:20.30 | Mavvie | funraps-as: because they can be generated when you're using app_xxxx, no. |
02:20.43 | Optic2 | funraps: voip-info.org is the first place I go for asterisk doco! :) |
02:21.24 | *** join/#asterisk nassy (~mark@24-193-231-136.nyc.rr.com) |
02:21.46 | *** join/#asterisk n0where (~ken@o152244.ppp.asahi-net.or.jp) |
02:22.24 | *** join/#asterisk UdontKnow (udontknow@udontknow.staff.freenode) |
02:22.27 | UdontKnow | hello |
02:22.54 | oncemore | how understand dsp.c chan_zap.c channel.c ? |
02:22.57 | UdontKnow | uh, can someone recommend a console sip client besides cornfedsipua? cornfed doesnt support tcp... |
02:23.01 | *** join/#asterisk Johaan (~johaan@ip24-56-24-181.ph.ph.cox.net) |
02:23.14 | drbyte | i can't even get kphone to dial in properly :( |
02:23.15 | oncemore | I want to modify busydetect Alg. |
02:24.10 | Nethab | heh asterisk can dial out from console, does that count? |
02:24.35 | ariel_ | hello all |
02:24.43 | drumkilla | Nethab: probably overkill, but it will do the trick :) |
02:25.09 | sudoer2 | can i set macros per client in sip/iax conf, like CID, and then set the CID to whatever is set for the cid in their client setup in sip/iax conf? |
02:25.49 | drbyte | i followed exactly everything http://www.voip-info.org/wiki-Asterisk+phone+Kphone and i can't get it to work. i'm trying the "simple" config |
02:26.01 | *** join/#asterisk routerheads_atho (~root@user-12l2p2p.cable.mindspring.com) |
02:27.17 | Mavvie | drbyte: use sip-debugging on asterisk to figure out what is going wrong. |
02:28.51 | UdontKnow | Nethab: heh. do you have "full phone" on console with it? |
02:29.07 | UdontKnow | Nethab: like receiving, calling, waiting and so on |
02:29.32 | _Vile | hrmm |
02:29.36 | *** part/#asterisk Gand_DJ (~gandalf@ptr-207-54-104-24.ptr.terago.ca) |
02:29.40 | _Vile | wonder how I can make lynx post from the command line |
02:29.44 | Nethab | on linux is /dev/dsp is configured to your sound card asterisk will use the microphone and speaker to call with |
02:30.04 | Nethab | CLI> call 912220000000 |
02:30.28 | Nethab | will call and use the local mic and speakers |
02:30.43 | UdontKnow | Nethab: can I register an asterisk with fwd? or need to use e164 stuff? |
02:30.54 | drbyte | Mavvie: okay, i'm running that in the console. i just turned on sip debug. nothing. kphone is trying, but not working |
02:31.04 | Nethab | yes you can register asterisk to fwd and many other providers |
02:31.28 | Mavvie | drbyte: if it is trying, does that mean asterisk is receiving the packets? |
02:31.51 | drbyte | Mavvie: by the looks of it no. asterisk console has nothing responding to it. |
02:32.04 | UdontKnow | Nethab: nice |
02:32.26 | drbyte | using kphone 4.0.5, compiled off tarball |
02:32.27 | Nethab | but admittedly that's like carving a steak with a chainsaw |
02:32.53 | Mavvie | drbyte: does or doesn't asterisk receive the packets? do you see them being outputted on the asterisk console? |
02:33.09 | Chuji | that's good |
02:33.26 | Nethab | if your phone sees trying that came from the proxy or asterisk |
02:33.37 | Nethab | otherwise it would say connecting |
02:34.08 | *** join/#asterisk santiago (~santiago@63.245.86.75) |
02:36.05 | drbyte | Mavvie: it doesnt. nothing is being output on the console |
02:36.20 | Nethab | set verbose 4 |
02:36.32 | Nethab | whoops |
02:36.40 | Nethab | bad xwindows bad |
02:36.52 | drbyte | and i started it up with -vvvvgc |
02:38.14 | *** join/#asterisk soulz2 (~Soulz-@host-137-132-45-204.imcb.nus.edu.sg) |
02:38.20 | soulz2 | hello all |
02:39.24 | ZX81 | hello |
02:40.59 | UdontKnow | Nethab: some friend tells me that it chews 100% cpu when calling from asterisk... is that true? |
02:41.54 | Nugget | how would we know if your friend is lying? |
02:41.58 | Nethab | well not 100% but asterisk has to convert your raw audio to a voip codec instead of your pc so it's a little heavy yeah |
02:42.16 | zigman | but never ever 100% |
02:42.18 | *** join/#asterisk calvis (calvis@h-66-167-52-131.sttnwaho.covad.net) |
02:42.20 | Nethab | that and chan_oss sucks |
02:42.23 | zigman | unless its a 386 something |
02:42.57 | *** join/#asterisk Legend (~legend@24.244.142.134) |
02:43.26 | *** part/#asterisk sigtom (~sigtom@rrcs-67-78-17-122.se.biz.rr.com) |
02:43.31 | *** join/#asterisk sigtom (~sigtom@rrcs-67-78-17-122.se.biz.rr.com) |
02:43.51 | Nethab | ok those of you helping me get my sipphone outbound working again, I was able to get the 1222 number to work but only intermittently |
02:44.29 | Mavvie | drbyte: then asterisk isn't receiving anything. |
02:44.50 | *** join/#asterisk rustyb (~rustyb@68.235.250.116) |
02:45.45 | drbyte | Mavvie: any idea why that might be the case? is kphone an appropriate tool to use? |
02:46.15 | Mavvie | drbyte: I don't know why it happens. First thing I would do is to run tcpdump in the local and remote machine to see where the packets are actually going to. |
02:48.53 | drbyte | Mavvie: i think i solved it. what happened was i was running asterisk via ssh. now i run it locally on the console and it works! now to get the isdn card working (the winbond) |
02:49.23 | Mavvie | good luck :-) |
02:49.39 | drbyte | Mavvie: thanks. have you had experience in setting it up? |
02:51.54 | *** part/#asterisk KHague (~khague@72.39.cm.sunflower.com) |
02:52.51 | oncemore | why busydetect alg. can not detect busy tone? |
02:53.45 | implicit | lol |
02:53.47 | implicit | because it sucks |
02:57.05 | oncemore | ? |
02:57.08 | oncemore | why??? |
02:57.18 | oncemore | what means? |
02:57.23 | fraglet | hi all |
02:57.28 | fraglet | quick question if i may |
02:57.44 | Mavvie | drbyte: I did everything via ISDN4Linux |
02:57.46 | fraglet | we are looking to use pri card on asterisk, does it handle it well? |
02:58.04 | Mavvie | fraglet: no problems here using the zaptel driver |
02:58.22 | fraglet | mavvie: thats using one of the digium cards? |
02:58.34 | Mavvie | that's using the TE410P cards |
02:58.36 | sigtom | sucks=bad |
02:59.20 | fraglet | thats the one i was looking at. |
02:59.29 | oncemore | :( |
02:59.37 | oncemore | pool busydetect alg. |
02:59.39 | fraglet | you know if it scales well.. ie multiple te410`s in one box? |
02:59.52 | Mavvie | fraglet: I have two TE410Ps in one box. |
02:59.56 | oncemore | callprocess can not work outside US |
03:00.05 | oncemore | how to du? |
03:00.10 | oncemore | how to do? |
03:00.25 | fraglet | mavvie: thanks for you help, makes us feel more comfortable ordering up some hardware and pri to give it a go |
03:00.38 | Mavvie | no probs. |
03:01.04 | oncemore | who can help me to detect busytone outside US? |
03:01.19 | Nethab | you mean ringing tones hehe |
03:01.59 | oncemore | BUSY tone 450Mhz 0.35s<On Time(ms)> 0.35<Off Time(ms) |
03:02.08 | ta[i]nted | anyone here use broadvoice |
03:02.10 | jks | anyknow knows the exact meaning of this sipphone error message: "the user you're trying to reach is unknown" |
03:02.20 | Nethab | i was commenting on how european ring tones sound similar to US busy tones |
03:02.38 | oncemore | how to detect this tone? now busydetect alg. not work |
03:02.46 | oncemore | it is China tone plan |
03:04.37 | UdontKnow | ;[sip_proxy] ; For incoming calls only. Example: FWD (Free World Dialup) |
03:04.42 | UdontKnow | how does one use that? |
03:04.43 | Nethab | I get "The user you are trying to reach is unavailable" all the time |
03:04.52 | drbyte | Mavvie: well, thats what i'll be using. got to read up on isdn4linux now then. it seems it gets automatically detected on Fedora Core 3 (the card), but other than that, i guess i'm in for some fun... |
03:05.08 | oncemore | I also played with BUSYDETECT_MARTIN and/or BUSYDETECT_TONEONLY and it |
03:05.09 | oncemore | makes no difference. I also tried editing dsp.c and adjusting |
03:05.09 | oncemore | BUSY_MIN and BUSY_MAX, but nothing fixes these problems. |
03:06.13 | Mavvie | drbyte: make sure you have the patches from http://bugs.digium.com/bug_view_page.php?bug_id=0002704 in your code. |
03:06.30 | Mavvie | http://bugs.digium.com/bug_view_page.php?bug_id=0002667 that is. |
03:06.40 | *** join/#asterisk goatmilk (~goatmilk@cae168-249-184.sc.rr.com) |
03:07.25 | *** topic/#asterisk by bkw_ -> Asterisk: The Open Source PBX || DUNDi (http://www.dundi.com) #dundi || Find a bug... help squash it out! http://bugs.digium.com |
03:07.27 | drbyte | Mavvie: thanks, though i'm using stable? 1.0.2 iirc |
03:07.43 | Mavvie | drbyte: just check if that patch is in it. |
03:08.06 | drbyte | 1.0 is when its fixed. i will double check in a bit |
03:08.38 | goatmilk | anyone get the error about radfw.h file not found during compile of zaptel w/ make linux26 ? |
03:09.00 | NormAst | Anyone know why this won't work ??? Zap/G1/9056198844,,D(90565554000) |
03:10.12 | bkw_ | NormAst no clue |
03:10.16 | bkw_ | that should be fine |
03:10.22 | bkw_ | show your full dial line |
03:10.41 | bkw_ | goatmilk nope |
03:10.43 | file[laptop] | nowhere else I'd rathe rbe |
03:11.06 | *** join/#asterisk t3t (~t3t@207.67.0.20) |
03:11.15 | NormAst | bwk: it's getting called in C ret = pbx_exec(chan, app, data, 1); |
03:11.54 | goatmilk | bkw_: apparently you need to run make radfw.h before make linux26 .. |
03:12.05 | bkw_ | goatmilk no you don't |
03:12.07 | bkw_ | make clean |
03:12.09 | bkw_ | then do it again |
03:12.22 | goatmilk | well i did that, and it didn't work |
03:12.28 | goatmilk | so then i did my way, and it worked |
03:12.52 | bkw_ | just do make |
03:12.56 | bkw_ | it works the same |
03:13.00 | bdeb5 | how could i configure asterisk to call out to a prepaid calling card, dial my pin, and then have it dial the number that i want to dial? |
03:13.16 | bkw_ | the D option on app dial |
03:13.39 | NormAst | <PROTECTED> |
03:13.39 | NormAst | <PROTECTED> |
03:13.53 | puzzled | hey guys, what's the "List propostion" threat about? |
03:14.00 | bkw_ | try it other another way |
03:14.08 | bkw_ | NormAst try it witohut pbx_exec |
03:14.18 | NormAst | dbeb5: Create a macro.. |
03:14.25 | bkw_ | dont need a macro |
03:14.29 | bkw_ | the D flag can do it |
03:15.22 | file[laptop] | you don't need a reason, let the rain go on and on |
03:15.46 | NormAst | bkw: So how do I dial out without pbx_exec? |
03:15.56 | file[laptop] | bkw_: how are you oh Brian-like one? |
03:16.06 | bkw_ | exten => 555,1,Dial(Zap/g1/blah||D(1234)) |
03:16.17 | bkw_ | NormAst you know C and you had to ask that? |
03:16.27 | bkw_ | :P |
03:16.44 | NormAst | It's been 10 years since I played in C and linux :) |
03:16.49 | bkw_ | ;) |
03:17.23 | file[laptop] | Enya is nice. |
03:17.37 | bkw_ | yactopitc |
03:18.42 | bkw_ | file[laptop] yactopitc |
03:20.19 | NormAst | bkw: needs the || and not the ,, |
03:20.27 | bkw_ | both work |
03:20.54 | NormAst | I tried the ,, and pbx_exec and it did not work! |
03:21.18 | NormAst | bugs? |
03:21.23 | bkw_ | doubt it |
03:21.41 | NormAst | I bet it is.. |
03:21.51 | bkw_ | bet its not:P |
03:22.12 | bkw_ | file[laptop] do you wish me to smack you.. YACTOPITC |
03:22.12 | NormAst | bkw: PRI issue? |
03:22.24 | bkw_ | NormAst who knows let me try it |
03:22.30 | file[laptop] | whatttttttttt? I was reading code change reports |
03:23.13 | *** join/#asterisk mhnoyes (~mhnoyes@user-2ivfnjl.dialup.mindspring.com) |
03:23.23 | NormAst | bkw: Okay... What would you like to do? |
03:23.35 | file[laptop] | bkw_: you've lost me |
03:25.02 | bdeb5 | how could i configure asterisk to call out to a prepaid calling card, dial my pin, and then have it dial the number that i want to dial? |
03:25.39 | bkw_ | file |
03:25.42 | bkw_ | 666 |
03:25.45 | file[laptop] | one sec |
03:26.20 | brc_ | YACTOPITC! |
03:26.35 | brc_ | wtf |
03:26.36 | brc_ | bkw_, |
03:27.10 | NormAst | dbeb5: it was answered! |
03:27.34 | NormAst | exten => 555,1,Dial(Zap/g1/blah||D(1234)) |
03:28.10 | NormAst | Just change the 555 to the area code, and the blah to ${EXTEN} |
03:28.27 | NormAst | 555 to Area code and number ie: 7165551212 |
03:29.51 | UdontKnow | I have some doubt regarding fwd setup as I found on http://lists.digium.com/pipermail/asterisk-users/2004-January/034050.html |
03:30.05 | UdontKnow | the sample says |
03:30.06 | UdontKnow | exten => s,1,Hangup |
03:30.06 | UdontKnow | exten => 74928,1,Answer |
03:30.23 | UdontKnow | where can I find how to do it right for my setup? |
03:30.45 | UdontKnow | (the sample uses multiple fwd lines...) |
03:33.00 | oncemore | hi |
03:33.15 | brc_ | bye |
03:33.18 | oncemore | who understand dsp.c channel.c chan_zap.c |
03:33.30 | brc_ | nobody |
03:33.39 | brc_ | it's voodo magic |
03:33.48 | brc_ | it wrote it self really |
03:34.00 | czero | :) |
03:34.04 | czero | its selfaware |
03:34.18 | *** join/#asterisk alphaque (~Alphaque@218.208.238.245) |
03:34.18 | puzzled | unless "it" receives money off course. then it explains by itself |
03:34.34 | czero | if you try to modify it, it will appear in you office and steal your soul, only to grow more strong and ruthless |
03:34.43 | brc_ | genetic evolutionary code using highly advanced heuristic models |
03:35.01 | puzzled | some whisper it is more evil than bill himself |
03:36.00 | puzzled | which would explain the money part |