00:00.08 | tessier | And I don't mean tech support for the OS, I mean commercial application support. |
00:00.16 | denon | yeah .. |
00:00.19 | denon | everything's redhat |
00:00.28 | denon | they gots the bucks |
00:00.35 | zoa | there is probably a bug in ast_sched_add and ast_shed del |
00:00.44 | zoa | maybe only on dual processor systems |
00:01.13 | denon | seems like I have some weird cvs issues once in a while |
00:01.13 | jsmith | zoa: I wouldn't be surprised... Mark thought there was one in there six months ago, only he never had time to track it down |
00:01.22 | ciy | could iptables filter on a zap interfaces? |
00:01.32 | jsmith | denon: I think it's just you, man. |
00:01.34 | Powerkill | I manage to register with a xten client behind a firewall to my asterisk box i see it on sip show peer but i can make calls |
00:01.35 | *** join/#asterisk killerbee (~Killer@ool-44c1013f.dyn.optonline.net) |
00:01.45 | joako | is JerJer around? |
00:01.45 | *** join/#asterisk Exomorph (Greg@134-9.bvcompuworks.com) |
00:01.49 | zoa | jsmith why did he think that ? |
00:01.50 | jsmith | Powerkill: Then learn how SIP works with firewalls. |
00:01.52 | Powerkill | when trying to dial i see this in sip debug SIP/2.0 407 Proxy Authentication Required |
00:02.07 | zoa | Powerkill: you are not registered |
00:02.14 | Powerkill | yes I am |
00:02.21 | Powerkill | <PROTECTED> |
00:02.36 | Powerkill | 700/700 212.195.110.242 (D) 255.255.255.255 63522 OK (239 ms) |
00:02.47 | Powerkill | I can't dial |
00:03.26 | outtolunc | 239ms ouch |
00:04.06 | Powerkill | yes depend sometime it's lower I'm just downloading some stuff |
00:04.39 | Powerkill | 700/700 212.195.110.242 (D) 255.255.255.255 63522 OK (55 ms) |
00:04.46 | Powerkill | so zoa any idea ? |
00:04.51 | *** join/#asterisk stonefly (~trillian@toby.stoneflytech.com) |
00:05.02 | *** join/#asterisk FryGuy (~fryguy@c-24-2-50-122.client.comcast.net) |
00:05.28 | stonefly | How do I change how many rings it takes for asterisk to answer? right now it takes two... |
00:06.22 | km- | it takes two rings to answer? |
00:06.24 | Powerkill | you can't set number of ring you are setting time |
00:06.29 | Powerkill | to answer |
00:06.36 | denon | Wait() before you answer |
00:06.52 | stonefly | With no Wait() it takes two rings... |
00:06.53 | Powerkill | in the dial command you put ,20,r |
00:07.12 | *** join/#asterisk pino (foobar@host115-28.pool21345.interbusiness.it) |
00:07.21 | km- | thats weird |
00:07.26 | km- | my x100p picks up instantaneously |
00:07.36 | denon | Powerkill: he's talking before Answer(), I think |
00:07.38 | jsmith | stonefly: Turn off caller-id, and it'll answer on the first |
00:07.43 | jsharp | put "usecallerid=no" in zapata.conf |
00:08.00 | jsmith | stonefly: Asterisk has to wait for the second ring to get caller-id info |
00:08.11 | stonefly | jsmith, jsharp, ok thank you.. |
00:08.21 | Powerkill | denon may be :) |
00:08.23 | km- | jsmith: howdy |
00:08.44 | *** join/#asterisk ambassador (~brad@h143.1.39.162.ip.alltel.net) |
00:08.46 | km- | jsmith: I think you have to disable chunked encoding in php, not in apache |
00:08.59 | jsmith | km-: Yeah, that's kind of what I remember... |
00:10.03 | enzo | bye |
00:10.06 | denon | someoen tell me how to do it once you guys get it ironed out <g> |
00:10.07 | *** part/#asterisk enzo (blueDJ@ALille-208-1-27-146.w81-51.abo.wanadoo.fr) |
00:10.21 | ambassador | anyone using asterisk and some provider as their "only link" to the PSTN? |
00:10.25 | km- | denon: oh dont worry, once I get it man... |
00:10.26 | ambassador | i.e. no land lines at all? |
00:10.31 | denon | hehe |
00:10.59 | jsharp | ambassador: I am for my business. |
00:11.04 | stonefly | I finaly figured out how to route calls based on what zapata channel they come in... contexts.. :) |
00:11.15 | ambassador | jsharp: are you satisified with the service and quality? |
00:11.25 | jsharp | Yes. |
00:11.30 | ambassador | what providor? |
00:11.33 | jsharp | nufone |
00:11.47 | jsmith | stonefly: Yeah... contexts are important :-) |
00:11.48 | *** part/#asterisk Alric (~nbowyer@masq.hyperusa.com) |
00:11.50 | ambassador | I'm seriously considering it for my place |
00:12.03 | ambassador | keep only a single analog line for fax |
00:12.11 | ambassador | and have all voice go over asterisk |
00:12.13 | espenz | Nov 21 01:10:25 guestbox kernel: isdn_tty: call from 91625817 -> 33324240 ignored |
00:12.18 | espenz | why doesnt asterisk answer? |
00:12.22 | stonefly | jsmith, I'm still getting used to asterisk config files... |
00:12.39 | jsmith | stonefly: I see... |
00:13.12 | jsharp | ambassador: Once the softfax gets more workable, the IAX providers will be able to offer fax services too if they want to. |
00:13.35 | Mike | bkw_: sipura is better than grandstream adapter? |
00:13.47 | pino | espenz: might be that you did not tell asterisk to bind to the right MSN |
00:14.04 | denon | jsharp: fax worth using yet? |
00:14.16 | ambassador | jsharp: ever have any trouble with people trying to reach you? |
00:14.16 | km- | that fax software is weird |
00:14.31 | denon | hehe |
00:14.32 | jsharp | Dunno. It no workee with my shitty quicknet hardware. |
00:14.42 | jsharp | ambassador: Only when my ISDN line falls over. |
00:14.45 | bevins | has anyone got * running on trustix? |
00:14.49 | jsmith | jsharp: So sell it on ebay and get real hardware! |
00:15.23 | pino | bevins: someone has. don't remember who, but someone has. :) |
00:15.51 | jsharp | jsmith: I'm holding out to afford a T100P...I've got a big beefy channel bank just dying to be used. |
00:16.14 | denon | got a few lines that dont hang up on their own very well |
00:16.16 | bevins | seems zaptel.c line 41... has trouble with usr/include with trustix |
00:16.34 | jrollyson | hmm. |
00:16.42 | jrollyson | just thought of something. |
00:16.46 | pino | if you give some more details, maybe someone may help you without running trustix... |
00:17.02 | jsmith | jsharp: I'm holding out for a bigger box for my TE410P... :-) |
00:17.06 | jrollyson | for hotel and telco installations, operator style barge in. |
00:17.29 | bevins | its kmod.h is where the problem is. |
00:17.41 | jrollyson | operator barges in silently, but hears scrambled audio. |
00:17.44 | denon | man, forgot to adjust wavgain before compile |
00:18.21 | bevins | the compile dies with zaptel.c line 41 which is #ifdef CONFIG_DEVFS_FS |
00:19.10 | pino | the line has nothing wrong in itself or nothing that could trigger an error; probably the compiler is giving you some more hints! |
00:19.36 | *** join/#asterisk nassy (~nassy@24-193-228-121.nyc.rr.com) |
00:19.43 | zoa | <espenz> Nov 21 01:10:25 guestbox kernel: isdn_tty: call from 91625817 -> 33324240 ignored --> you don't have an extension for that number in that context |
00:20.22 | bevins | pico: /usr/linux/modversions.h:5:1 unterminated #ifndef |
00:20.51 | pino | i suggest that you compile your own kernel and install its headers. |
00:22.43 | bevins | I was just trying trustix... I am looking for a good clean fast dist...any suggestions other than trustix? I will attempt to compile trustix kernel anyway. |
00:25.01 | pino | bevins: it's too often a matter of religion rather than technology. :) |
00:25.06 | *** join/#asterisk JerJer[ghost] (~NunYoBizN@pppoe-66-112-51-206-rb2.grp.centurytel.net) |
00:26.10 | zoa | debian Doh ! |
00:27.08 | ciy | debian rocks the house |
00:28.56 | *** join/#asterisk Poincare (~jeff@D577A9AE.kabel.telenet.be) |
00:29.15 | stonefly | I'm using the CLASS-like features from http://lists.digium.com/pipermail/asterisk-dev/2003-July/001070.html, but CALLERIDNUM for a sip client has a "-" and four random characters after it, and it breaks the features for sip clients. Is there anyway to fix that? |
00:29.29 | stonefly | Or is there a better call forwarding example? |
00:29.39 | *** join/#asterisk spr (~spr@cpe-66-191-202-120.spa.sc.charter.com) |
00:30.25 | zoa | stonefly maybe add a bug report |
00:30.57 | joako | stonefly setup something using astdb |
00:31.37 | stonefly | zoa, I don't think it is a bug, but something I'm doing, anyways the CLASS-like features isn't part of the main package... |
00:31.47 | stonefly | joako, that's what I was afraid of... |
00:31.48 | jsmith | Or just use a regular expression to strip of the -abcd |
00:32.09 | jsmith | s/of/off/ |
00:33.05 | stonefly | jsmith, I need to learn AGI anyways, so this is a good excuse... |
00:33.37 | jsmith | stonefly: Yes, it is... |
00:33.41 | danielq | anyone here using ztdummy? |
00:33.42 | stonefly | too bad you can't do regex in extensions.conf.. :( |
00:34.20 | pino | server* |
00:34.24 | *** join/#asterisk Exomorph (Greg@134-9.bvcompuworks.com) |
00:34.57 | tholo | Ooh! He's alive! |
00:35.09 | pino | now, less than 3MB of RAM for a working idle * ... |
00:35.13 | zoa | aha kram ! |
00:35.14 | jsmith | kram! |
00:35.33 | jsmith | kram, kram, he's our man! If he can't do it, uh.. well.. we're screwed! |
00:35.40 | jsharp | lol |
00:35.50 | zoa | hehe lol |
00:37.14 | denon | aw man, cant you edit the voicemail email subject in th config? |
00:37.14 | denon | bummer |
00:37.44 | mrgoby | has anyone used extendb ?? |
00:38.05 | stonefly | denon, are you using Voicemail, or Voicemail2? |
00:38.10 | mrgoby | (or written it ;-) |
00:38.55 | bevins | debian.org is down? hehe |
00:39.02 | zoa | debian.org is not down |
00:39.51 | stonefly | denon, I believe that with voicemail you can't edit the subject, but with voicemail2, you can. |
00:40.04 | denon | stonefly: there's no such thing as Voicemail1 anymore |
00:40.07 | denon | they're all vmail2 |
00:40.20 | denon | I see where you can do the body, dont see anything about the subject tho |
00:40.20 | stonefly | denon, oh, I missed that.. :) |
00:40.27 | espenz | anyone, with asterisk and ISDN experience? |
00:40.36 | bevins | I'm not able to get to it |
00:41.24 | bevins | sprintlink router..... |
00:41.42 | pino | bevins, same for me |
00:42.00 | pino | but if you're looking for it in order to download something, you might use one of the foreign mirrors |
00:42.10 | pino | e.g. ftp.uk.debian.org |
00:42.19 | bevins | I wanna read about it forst |
00:42.39 | *** part/#asterisk outtolunc (~me@h-66-166-19-148.SNVACAID.covad.net) |
00:42.45 | bkw_ | lalalal |
00:42.56 | Mike | bkw_: sipura is better than grandstream adapter? |
00:43.05 | danielq | bevins: www.uk.debian.org |
00:43.05 | bkw_ | um yes |
00:43.06 | pino | then http://www.uk.debian.org/ for example.. |
00:43.11 | bkw_ | strings and cans would work better |
00:43.20 | bevins | cool thanks |
00:43.27 | pino | espenz: you still stuck with the "ignored" message? |
00:43.52 | Mike | bkw_: i asked for 3 adapters from grandstream thou would be enought to put my wireless telefons |
00:44.26 | jets | Is there any way to end a Meetme conference, or kick everyone out... we have a script using meetme as a "paging system" of sorts... |
00:44.41 | *** join/#asterisk zeta_ (~zeta@207.88.150.254.ptr.us.xo.net) |
00:44.48 | zeta_ | is there a way to destroy a meetme bridge? |
00:44.50 | espenz | pino: yes |
00:44.55 | mrgoby | wondering if dynextendb has been used by anyone here? |
00:44.56 | espenz | and i changed the trunk to: |
00:45.01 | espenz | Modem/g1 |
00:45.07 | espenz | and driver: i4l |
00:45.41 | espenz | im testet that i works with the ivcall |
00:46.44 | pino | are your incomingmsn's right? |
00:46.58 | bkw_ | zeta_ what do you mean? |
00:47.09 | *** part/#asterisk ciy (~2mork@node-402405b2.sfo.onnet.us.uu.net) |
00:48.04 | espenz | it works |
00:48.05 | espenz | yeah! |
00:48.16 | jsmith-away | espenz: I'm glad to hear you got it working! |
00:49.05 | espenz | jsharp: :) |
00:49.08 | espenz | jsmith-away even |
00:49.14 | espenz | the fail was |
00:49.21 | espenz | that i forgot to take away the |
00:49.23 | espenz | ; |
00:49.27 | espenz | msn |
00:49.27 | espenz | :P |
00:50.34 | espenz | jsmith-away: did you call me? |
00:50.57 | zeta_ | bkw_, I mean, is there a way that I can programatically destroy a meet-me conference room? |
00:51.13 | zeta_ | bkw_, for example, say 3 people are in a room... can I issue a command that closes that channel? |
00:51.32 | bkw_ | soft hangup the channels in the meetme |
00:52.03 | zeta_ | i tried that but it didn't seem to work |
00:52.07 | zeta_ | all of the other people stayed on the line |
00:52.22 | km- | do any of you have problems with dropped calls on x100p? |
00:53.19 | zoa | nopez |
00:53.22 | Connor | zeta, why would you want to do that anyway ?? |
00:53.45 | zeta_ | Connor, trying to implement a paging system |
00:54.06 | km- | see my wife says calls always drop for her |
00:54.06 | Connor | ok, go on..... |
00:54.15 | zeta_ | i have it all working, except ZapBarge/Monitor will leave the meetme conference immediatly... and softhangup won't kill the users after the "pager" has left |
00:54.17 | km- | but nothing weird ever shows up on the asterisk console |
00:54.26 | km- | and then I try and sit down and make like ten calls and it doesnt cut out on me |
00:54.40 | km- | it really pisses me off |
00:54.59 | Connor | km-, you sure it isn't a ID10T error ? |
00:55.15 | Connor | on your wife's part? No offence or anything.. |
01:00.47 | *** join/#asterisk nocnoc (~cvenegas@atlas.ifxnw.cl) |
01:01.17 | *** join/#asterisk crontibs (~crontibs@ool-44c02950.dyn.optonline.net) |
01:01.20 | nocnoc | guys.. anybody know what this might be? NOTICE[57352]: File chan_mgcp.c, Line 1099 (find_subchannel): Gateway '10.0.0.20' (and thus its endpoint 'd001') does not exist |
01:09.18 | *** join/#asterisk tristan2 (~tristan@213.239.44.133) |
01:10.30 | *** join/#asterisk ricky (~ricky@hoochie.digium.com) |
01:11.39 | espenz | could i get asterisk to forward a call remote? |
01:11.56 | *** join/#asterisk ionix- (~ionix@MTL-HSE-ppp202388.qc.sympatico.ca) |
01:16.18 | *** join/#asterisk Muckl (johannes@pD9ED523A.dip.t-dialin.net) |
01:17.30 | Muckl | can someone post me his working [client] section in skinny.conf for a 7940/7960 skinny client? |
01:18.46 | *** join/#asterisk _spr (~spr@cpe-66-191-202-120.spa.sc.charter.com) |
01:24.21 | *** join/#asterisk cfo_ (~cfo@194.19.190.217) |
01:25.31 | *** join/#asterisk MagicMan (~alm971@APointe-a-Pitre-101-1-5-81.w81-53.abo.wanadoo.fr) |
01:29.06 | *** part/#asterisk _spr (~spr@cpe-66-191-202-120.spa.sc.charter.com) |
01:30.48 | *** join/#asterisk _spr (~spr@cpe-66-191-202-120.spa.sc.charter.com) |
01:31.01 | *** part/#asterisk _spr (~spr@cpe-66-191-202-120.spa.sc.charter.com) |
01:32.09 | km- | connor: yeah, I double checked it |
01:32.23 | km- | connor: her hands are nowhere near the endcall or the receiver hook |
01:33.05 | espenz | hm, how do i turn off the default sounds |
01:40.55 | *** join/#asterisk _spr (~spr@cpe-66-191-202-120.spa.sc.charter.com) |
01:40.56 | Mike | how much does digium take to deliver a X100p? |
01:41.13 | km- | YES!!!!!!!!!!!! |
01:41.13 | km- | DUDES |
01:41.23 | km- | denon: WAKE UP I FOUND IT :P |
01:42.00 | Mike | km-: found what |
01:42.04 | Mike | ? |
01:42.11 | km- | mike: the answer to why 7960's stall with web services on apache |
01:42.27 | tholo | Huh. Never stalled for me yet... |
01:42.31 | Mike | thats the wireless cisco right? |
01:42.38 | km- | tholo: the problem is chunked encoding (like jsmith said) |
01:42.40 | km- | adding this: |
01:42.44 | km- | BrowserMatch "Allegro-Software-WebClient/3\.10b1" nokeepalive |
01:42.44 | km- | downgrade-1.0 force-response-1.0 |
01:42.51 | km- | to apache.conf has fixed the problem |
01:43.04 | tholo | Oh. I guess maybe I haven't sent big enough responses yet... |
01:43.18 | km- | its only a problem if you're using php to generate your XML packets |
01:43.19 | Mike | km-: is that the wireless phone from cisco? |
01:43.21 | km- | if you're using raw xml it wont happen |
01:43.27 | km- | mike: no, it's the 6 line desk phone |
01:43.31 | tholo | Ah. I've been using perl so far. |
01:43.41 | Mike | anyone has the cisco wireless? |
01:48.07 | bkw_ | km- can you email that to me? brian@bkw.org |
01:48.16 | km- | I just fired it to the asterisk-dev list |
01:48.18 | km- | should be there shortly |
01:52.16 | km- | now that I've fixed that I'm going to have to hook up jsmith with some bugfixes for his directory |
01:53.32 | bkw_ | OMG |
01:53.39 | bkw_ | ok my blog is copyright Brian K. West |
01:54.02 | km- | did someone steal your blog? |
01:54.31 | bkw_ | some asshole said this about my Fuck Riaa comments: |
01:54.32 | bkw_ | Comments: |
01:54.32 | bkw_ | Oh right, and stealing is cool now I guess. |
01:54.32 | bkw_ | Ever thought that you're ripping off other people? |
01:54.32 | bkw_ | "Copyright ©2003 Brian K. West " made me laugh - what a fucken hypocrite. 'What you download is copyrighted you moron. |
01:55.08 | bkw_ | I don't feel thats being hypocrite.. |
01:55.09 | bkw_ | You are free: |
01:55.09 | bkw_ | to copy, distribute, display, and perform the work |
01:55.14 | bkw_ | anything on my website |
01:55.30 | Connor | what website is this bkw? |
01:55.42 | bkw_ | www.bkw.org |
01:56.05 | Connor | no, the site that stole your comment. |
01:56.40 | bkw_ | nobody did |
01:56.52 | km- | bkw: how much space have you exhausted with your ata-186 password? |
01:56.53 | bkw_ | some ass posted saying I was a hypocrite for having the copyright notice on my site |
01:56.58 | bkw_ | km- I gave up |
01:57.08 | km- | you gave up? |
01:57.09 | bkw_ | I might hook it back up later this week |
01:57.10 | km- | why? |
01:57.14 | bkw_ | I did about 10% of the space |
01:57.19 | bkw_ | I lost intrest in it |
01:57.23 | doughecka | Only during testing did they find that thermonuclear hand grenade's blast radius was further than anyone could throw it. |
01:57.29 | mrgoby | bkw_ stealing IS cool |
01:57.35 | mrgoby | :-d |
01:57.44 | bkw_ | copyright infringment isn't stealing |
01:57.46 | km- | dougchecka: hahahha |
01:57.51 | bkw_ | the law makes that clear |
01:58.04 | doughecka | bkw_: I could let it run on my system |
01:58.08 | doughecka | I have a x100p |
01:58.09 | doughecka | :) |
01:58.29 | bkw_ | na thats ok |
01:58.32 | *** join/#asterisk UnixDawg (~UnixDawg@ny-lasalle6c-203.buf.adelphia.net) |
01:58.35 | bkw_ | I will hook it back up next week |
01:58.41 | bkw_ | and let it be for a few weeks |
01:58.41 | doughecka | bleh :P |
01:58.45 | doughecka | heh |
01:58.53 | doughecka | how do you know if its the correct number? |
01:58.59 | doughecka | wont it keep dialing numbers? |
01:59.08 | bkw_ | yep but it will still reset and stop responding to numbers |
01:59.11 | doughecka | or will it do something that * can recognize? |
01:59.13 | doughecka | ah |
01:59.34 | doughecka | what is that internet phone jack on pulver? |
01:59.45 | bkw_ | phone patch |
01:59.51 | doughecka | hmm |
01:59.56 | bkw_ | it turns an FXO into an FXS or vice versa |
02:00.02 | doughecka | hrm |
02:00.08 | doughecka | sounds awfully complex |
02:00.25 | doughecka | so I hook my x100p to it, and I can hook a phone to it? |
02:03.07 | UnixDawg | anyone know what happen to benjk |
02:03.11 | learath | bkw_: how well do they work? |
02:03.17 | UnixDawg | he has not responded in 4 days |
02:03.43 | *** join/#asterisk glLoadIdentity (~asdfrt@abn139-91.interaktif.net.tr) |
02:04.05 | bkw_ | ok the ata is cracking again |
02:04.13 | bkw_ | started at 200000 and headed up |
02:04.38 | bkw_ | er 300000 |
02:04.46 | bkw_ | maybe i will get lucky |
02:06.46 | Mike | can i do a menu that says dial 1 to call x dial 2 for the analog phone to ring |
02:06.53 | Mike | is it posible? |
02:07.00 | Mike | that the analog phone rings? |
02:07.07 | Mike | after the x100p answers? |
02:07.54 | *** join/#asterisk Zebble (~Zebble@Sherbrooke-HSE-ppp3610369.sympatico.ca) |
02:09.07 | tholo | Almost] anything is possible. |
02:10.32 | jrollyson | hmm. |
02:17.11 | Muckl | oh wow, i just tried voicemail feature, its really great! |
02:17.47 | Muckl | km-: do you use your 7960 as SIP or skinny client? |
02:22.18 | km- | mucki: SIP |
02:23.24 | *** topic/#asterisk by kram -> From bug 519: "Yes this patch did fix the blocking issue, now it just crashes. Yes this patch did fix the blocking issue, now it just crashes." |
02:23.37 | km- | hahahaha |
02:23.44 | *** topic/#asterisk by kram -> From bug 519: "Yes this patch did fix the blocking issue, now it just crashes. Which from my point of view is much better!" |
02:24.19 | km- | kram! |
02:24.23 | kram | km-! |
02:24.24 | km- | when's the 4 port fxo coming |
02:24.24 | km- | :P |
02:24.34 | kram | i'm about to scream |
02:24.37 | km- | "RSN!" |
02:24.40 | kram | i'm working on it, *really* |
02:24.46 | km- | hehehe :P |
02:24.47 | kram | but i have this very weird problem |
02:24.59 | km- | oh really... |
02:25.34 | km- | been there |
02:28.11 | *** join/#asterisk cypromis (~michael@217.11.142.161) |
02:28.24 | km- | cypromis: howdy |
02:28.38 | cypromis | aloha :) |
02:28.57 | km- | cypromis: were you one of the guys who was interested in the answer to my php problem? |
02:29.03 | km- | cypromis: I posted to asterisk-dev about it |
02:29.07 | *** join/#asterisk ThunderZ (~lzhang@cm1-off.syd.au.oztell.com) |
02:29.24 | cypromis | nope |
02:29.30 | cypromis | we switched from php to java :) |
02:29.45 | km- | hehe |
02:29.55 | km- | what do you guys do with java and the 7960? |
02:30.46 | UnixDawg | ok who shot stealth |
02:30.49 | cypromis | 7960 ? |
02:30.54 | cypromis | shot ? |
02:31.02 | cypromis | I helped him setup his * connections lol |
02:31.06 | cypromis | but I did not shoot him |
02:31.08 | cypromis | hehe |
02:31.13 | cypromis | km-: we do telco stuff |
02:31.20 | cypromis | no 7960's yet |
02:31.22 | km- | hahaha mark's office is messy |
02:31.29 | cypromis | although we are playing with the tdm40b |
02:31.32 | UnixDawg | well I fixed it and cleaned it up and got his iax working but now he is mia |
02:31.40 | cypromis | and are 85% hapy with the results |
02:31.46 | cypromis | :)) |
02:31.51 | km- | kram: dude, get that invoice out of the camera |
02:31.52 | cypromis | what did you fix ? |
02:31.52 | km- | :P |
02:31.53 | cypromis | :)) |
02:32.05 | cypromis | the extension mess ? |
02:32.11 | UnixDawg | his extensions.conf and his sip.conf |
02:32.13 | cypromis | ok |
02:32.16 | km- | cypromis: makes it so php generated xml packets are shown nearly instantaneously |
02:32.19 | UnixDawg | they work now |
02:32.20 | *** join/#asterisk Carp (Carp@ip-204-97-151-185.modem.logical.net) |
02:32.24 | cypromis | we where playing before with g723.1 passthrough |
02:32.41 | cypromis | nice |
02:32.46 | Carp | Where is JerJer? |
02:32.48 | Carp | Anyone lknow |
02:32.51 | Carp | know*? lol |
02:32.58 | UnixDawg | well I was chatting him 15 min ago and he stopped chatting |
02:33.03 | cypromis | hmmm |
02:33.11 | cypromis | probably went to get some pizza ? |
02:33.27 | UnixDawg | and now when you call him on his ext 7000 it says he is on the phone but he is not |
02:33.32 | UnixDawg | ahh maybe |
02:33.43 | cypromis | hehe |
02:33.50 | cypromis | he probably played with it again |
02:34.02 | cypromis | it was not easy to explain to him that there is no g723.1 yet in asterisk |
02:34.03 | cypromis | :)) |
02:34.05 | UnixDawg | I needed to get back in to the server to see why its saying he is on the phone when he is not |
02:34.14 | UnixDawg | I will |
02:34.23 | UnixDawg | its not used yet |
02:34.36 | UnixDawg | I for get the new ip |
02:34.36 | Carp | I ordered an 800 number from JerJer 4 weeks ago and it still doesnt work. |
02:35.04 | km- | crap: that sucks. |
02:35.04 | UnixDawg | carp are you sure your authing right |
02:35.28 | Carp | UnixDawg: What you mean? |
02:35.40 | Carp | I tried to call the 800 number from my phone and ti says its not valid |
02:35.43 | UnixDawg | if its threw iax then you need to make sure your getting the connection iax2 show oeers |
02:35.51 | tholo | If you asked for a particular number, NuPhone is probably still waiting for it to be transferred to them. |
02:36.06 | km- | yeah |
02:36.19 | km- | vanity 800's take a while, as well as transfers |
02:36.22 | Carp | Unix: Its a number redirection |
02:36.45 | km- | hmm |
02:36.50 | km- | so I've got weather on the 7960 |
02:36.52 | Carp | If I knew it would take this long I would have had him just turn up a number he already had. |
02:36.55 | km- | and now I've got a directory |
02:37.12 | km- | what do I implement now for the 7960! |
02:37.29 | tholo | Video! |
02:37.38 | km- | hmm video |
02:37.42 | tholo | :) |
02:37.48 | km- | you can do that on 7960's? |
02:38.11 | km- | I thought that was only something that could be done on the skinny ones, not SIP |
02:38.22 | km- | i.e., showing a picture of the person who's calling |
02:38.40 | Carp | how can i figure out what phone system my school is using? I cant actually look at the system |
02:39.08 | tholo | Ask to see the documentation for it? ;-) |
02:39.09 | km- | what kinda phones?? |
02:40.00 | Carp | I dunno lol. They are the yellow analog ones, wall mountable, has just standers buttons and a button with a lightning bolt on it which I believe they oress toi check their mail. |
02:40.08 | Carp | press to* |
02:40.16 | km- | hmm |
02:40.24 | km- | prolly some older comdial system that used analog phones? |
02:40.35 | Carp | Its a new system as of last year |
02:40.40 | km- | really |
02:40.44 | Carp | last year was the first year the rooms got phones in then. |
02:40.46 | Carp | them* |
02:40.48 | km- | who knows what it is |
02:42.30 | Carp | how can i make my auto-attendant answer right as the call comes in? rather than 2 rings. |
02:42.31 | km- | trip over a desk and accidentally pick up the phone while simultaneously dialing random numbers! Maybe you'll get lucky! |
02:44.09 | Carp | OMG! lol |
02:44.19 | Carp | I can get into any teachers mailbox at school without a password lol |
02:46.05 | espenz | could anyone send me a "professional" working extensions.conf? ;p |
02:46.15 | learath | watch out they'll throw you in jail for that |
02:46.33 | jrollyson | espenz: check the wiki, theres some links |
02:48.25 | cypromis | espenz: there is no such thing |
02:48.38 | cypromis | it depends what kind of scenario you have |
02:48.45 | cypromis | there is no UNIVERSAL |
02:48.48 | cypromis | extensions.conf |
02:49.33 | espenz | i know.. |
02:50.11 | espenz | cypromis: is it possibly to make it do remote calls? |
02:50.21 | Carp | Everyones extensions.conf is different, custom to the setup. |
02:51.15 | espenz | of course. |
02:51.20 | cypromis | espenz: you an do a pletora of things with it |
02:51.37 | Carp | Does anyone know how to setup an intercom using the sound card? |
02:51.37 | km- | pretty much if you can think of it |
02:51.43 | carrar | plethora? |
02:51.44 | km- | and it involves voice transport over phone or internet |
02:51.52 | km- | you can do it with asterisk |
02:52.17 | carrar | pletora invloves using the tora drivers? :) |
02:52.49 | km- | pletora? |
02:53.00 | carrar | (scroll back) |
02:53.05 | km- | I know, I'm staring at it |
02:53.17 | km- | just wondering when the damned crickets will stop! |
02:53.17 | cypromis | greeklish |
02:53.19 | km- | :P |
02:53.46 | km- | anyone here have the cisco XML SDK |
02:53.58 | km- | I'd like to do "Good Things" with it but I cant remember my cisco login |
02:54.11 | *** join/#asterisk cybyc (~cybyc@Ottawa-HSE-ppp258893.sympatico.ca) |
03:02.22 | Mike | i can use asterisk as a soft phone? |
03:02.41 | cypromis | yes |
03:02.57 | tholo | Provided you have a soundcard with OSS drivers. |
03:03.20 | Mike | yes i have a soundcard with oss |
03:03.24 | *** part/#asterisk km- (pgrace@virgil.fierymoon.com) |
03:03.29 | Mike | so what do i need to make it a softphone? |
03:03.55 | *** join/#asterisk Carp (Carp@ip-204-97-151-127.modem.logical.net) |
03:04.07 | Carp | I tried to setup an intercom with the soundcard, what does this error mean" |
03:04.09 | Carp | : |
03:04.10 | Carp | << Call placed to 'dsp' on console >> |
03:04.11 | Carp | << Auto-answered >> |
03:04.11 | Carp | Called dsp |
03:04.11 | Carp | OSS/dsp answered Zap/1-1 |
03:04.11 | Carp | WARNING[114703]: File chan_oss.c, Line 402 (soundcard_setinput): Unable to re-open DSP device or resource busy |
03:04.13 | Carp | WARNING[114703]: File chan_oss.c, Line 561 (oss_write): Unable to set device to input mode |
03:04.15 | Carp | << Hangup on console >> |
03:04.15 | tholo | Make an extension for Console/dsp |
03:04.22 | Carp | I did. |
03:04.28 | tholo | Then you get commands "dial" and "hangup". |
03:04.40 | Carp | exten => 7777,1,Dial,console/dsp |
03:04.53 | Mike | seams to easy:P |
03:04.57 | tholo | Carp: Sounds like maybe something else was using the soundcard. |
03:05.23 | Carp | tholo, maybe Redhat doesnt reconize it? |
03:05.39 | Carp | I dont know if my OS thinks its installed, I've never used it with Linux |
03:05.41 | tholo | The actual device opened is /dev/dsp |
03:06.01 | Carp | how do I check if my soundcard is setup properly? |
03:06.04 | Carp | I dont know alot of Linux |
03:06.23 | tholo | I dunno -- I'm not really a Linux person. I run a couple of Linux systems just for Asterisk, since there is no hardware support on OpenBSD. :) |
03:07.22 | Carp | Does anyone else know? |
03:09.12 | cypromis | Carp: get the patch for chan_oss from kapejod at www.junghanns.net |
03:09.15 | cypromis | also |
03:09.20 | cypromis | check if your soundcard is full duplex |
03:09.26 | cypromis | and if you have the right drivers loaded |
03:09.29 | cypromis | best use alsa |
03:09.31 | cypromis | drivers |
03:11.31 | Carp | cypromis: I dont know much linux, so I dont know how to do any of that |
03:12.00 | *** join/#asterisk _Chotaire (chotaire@irc.chotaire.net) |
03:12.43 | *** part/#asterisk _spr (~spr@cpe-66-191-202-120.spa.sc.charter.com) |
03:15.22 | *** join/#asterisk decode (~decode@ip-wv-68-117-144-207.charterwv.net) |
03:17.25 | Mike | someone has gotten sipphone.com 1800 toll free numbers to work with asterisk and GS? |
03:17.30 | Mike | or FWD |
03:24.33 | Mike | exten => _1800XXXXXXX,1,SetVar(SIP_CODEC=g711) |
03:24.34 | Mike | exten => _1800XXXXXXX,2,SetCallerID(${FWDUSERID}) |
03:24.34 | Mike | exten => _1800XXXXXXX,3,SetCIDName(${FWDUSERID}) |
03:24.34 | Mike | exten => _1800XXXXXXX,4,Dial(SIP/*${EXTEN}@fwd.pulver.com) |
03:24.37 | Mike | trying that |
03:29.29 | decode | i bet my gf loved the away message i have set.. |
03:29.37 | decode | err will love |
03:30.18 | decode | heh, any good web admin shit for *? :) |
03:33.10 | *** join/#asterisk Tekati (~captain@cpe-66-75-211-60.bak.rr.com) |
03:33.57 | *** join/#asterisk JerJer[ghost] (~NunYoBizN@pppoe-66-112-51-206-rb2.grp.centurytel.net) |
03:34.15 | decode | "No, no! 'Rare' implies dangerously cooked. When I say rare I mean just let it look at the oven in terror, then bring it out to me." |
03:39.30 | doughecka | hahahahahha |
03:39.42 | doughecka | whats that out of? |
03:39.44 | Mike | iaxtel.com still works? |
03:40.45 | decode | doughecka it's from my ex's bitch-box err journal |
03:40.59 | doughecka | ah |
03:41.16 | decode | i have no idea, but if i ever speak to her again, i'll ask |
03:42.04 | decode | whoa! |
03:42.13 | decode | i'm on AIM using 2 clients with the same SN |
03:42.15 | *** join/#asterisk pbxtech (hash@65.204.194.35) |
03:42.21 | *** join/#asterisk Epitaph (~epitaph@pr37.nji.com) |
03:42.23 | doughecka | WOAH! |
03:42.27 | doughecka | thats AMAZING! |
03:42.33 | decode | its not supposed to allow that |
03:42.33 | doughecka | I _cant_ believe it! |
03:42.40 | decode | but apparently if gaim is the second client.. |
03:43.18 | doughecka | oh boo yea |
03:43.20 | doughecka | I love newegg |
03:43.26 | doughecka | I send a money order... |
03:43.37 | doughecka | and when it gets there, it ships the same bloody day |
03:44.48 | Mike | someone knows if iaxtel gives servers this days? |
03:44.52 | Mike | or is it dead? |
03:45.12 | doughecka | dead |
03:45.15 | doughecka | for me |
03:45.35 | Mike | what are you using for toll free? |
03:45.59 | doughecka | no |
03:46.13 | doughecka | I called manx's phone number |
03:48.00 | denon | anyone know if Matt Florell is ever on irc? |
03:48.19 | denon | or have any of you played with his gui * call manager yet? |
03:48.21 | doughecka | never heard of him |
03:48.41 | denon | http://sourceforge.net/projects/astguiclient/ |
03:48.47 | denon | he's on the * list |
03:48.54 | denon | I cant get the thing to run tho |
03:48.58 | denon | on win32, perl .. |
03:49.10 | denon | think Ive got everything it needs .. throwing lotsa errors though |
03:49.19 | doughecka | ah |
03:49.33 | denon | get it to work? |
03:49.40 | doughecka | never tried |
03:49.46 | denon | well try, man .. |
03:49.47 | denon | :) |
03:51.47 | espenz | does asterisk take mp3? |
03:51.53 | doughecka | no |
03:51.57 | doughecka | but it can play mp3s |
03:52.00 | espenz | how? |
03:52.00 | doughecka | using mpg123 |
03:52.05 | espenz | just mpg123? |
03:52.08 | espenz | not mplayer? |
03:52.17 | doughecka | nope |
03:52.23 | doughecka | has to be the REAL mpg123 |
03:52.44 | espenz | okey |
03:52.47 | espenz | how do i load it? |
03:52.50 | espenz | i just want to test |
03:52.56 | espenz | so if you got a line for me |
03:53.04 | doughecka | seems you need mysql.pm |
03:53.07 | doughecka | net-mysql |
03:53.09 | doughecka | find it |
03:53.09 | doughecka | :P |
03:53.13 | espenz | hm |
03:53.14 | espenz | mysql? |
03:53.14 | doughecka | unless you have it |
03:53.18 | espenz | why? |
03:53.20 | doughecka | net-mysql |
03:53.21 | espenz | to play mp3, uh? |
03:53.22 | espenz | :P |
03:53.26 | doughecka | doh |
03:53.28 | doughecka | wrong person |
03:53.39 | espenz | hehe |
03:53.58 | doughecka | Net::MySQL |
03:54.49 | Mike | bkw_: alive? |
03:55.20 | espenz | ;exten => 400,1,MP3Player,song8.mp3 |
03:55.31 | espenz | exten => 400,1,mpg123,song8.mp3 |
03:55.41 | espenz | where must the song be? in wich catalog? |
03:56.48 | doughecka | you can make it like this: |
03:57.23 | doughecka | exten => 2225,2,MP3Player(/root/ThroneEnd.mp3) |
03:57.38 | doughecka | and just tell it where it is |
03:57.45 | doughecka | and make sure you have the REAL mpg123 |
03:57.55 | doughecka | and one problem I have is mpg123 crashing * |
03:57.58 | doughecka | YMMV |
03:58.17 | decode | now to go shopping for a new gf, this one has pissed me off. |
03:58.29 | doughecka | LOL |
03:58.36 | denon | doughecka: I have net-mysql on it |
03:58.41 | denon | I read the readme .. :) |
03:58.42 | doughecka | oh |
03:58.48 | decode | i don't tolerate people telling me they'll be at X place at Y time then not being on time.. |
03:58.50 | doughecka | and NET::TELNET? |
03:58.55 | denon | yes... |
03:58.57 | decode | or even better just never being there |
03:59.00 | doughecka | you have DCC? :) |
03:59.11 | denon | the errors im getting seem to be more along the lines of coding errors |
03:59.14 | denon | unless im mistaken |
03:59.16 | doughecka | ah |
03:59.25 | doughecka | well I think it was in the alpha stage |
03:59.25 | doughecka | :) |
03:59.28 | decode | i bill at 60$/hr for wait time.. :) |
03:59.30 | denon | "my" variable $dbhA masks earlier declaration in same scope at C:\AstGui\AST_WIN |
03:59.30 | denon | phoneAPP_0.7.pl line 2187. |
03:59.30 | denon | Name "main::user_switching_enabled" used only once: possible typo at C:\AstGui\A |
03:59.30 | denon | ST_WINphoneAPP_0.7.pl line 141. |
03:59.33 | doughecka | haaa |
03:59.34 | decode | chicks just get dumped |
03:59.39 | denon | etc etc |
03:59.42 | denon | lotsa errors |
03:59.52 | doughecka | interesting |
03:59.57 | decode | if i have to wait 5 minutes for a client, that's an hour of wait time, at 60$/hr |
04:00.01 | doughecka | hah |
04:00.10 | espenz | WARNING[262161]: File pbx.c, Line 1160 (pbx_extension_helper): No application 'mpg123' for extension (demo, 400, 1) |
04:00.10 | espenz | <PROTECTED> |
04:00.13 | espenz | whats wrong? |
04:00.15 | decode | 1 hour 5 minutes == two hours, etc |
04:00.19 | denon | doughecka: got ideas for me? |
04:00.21 | doughecka | you need mpg123 |
04:00.25 | espenz | i have it? |
04:00.33 | espenz | root@guestbox:~# type mpg123 |
04:00.33 | espenz | mpg123 is /usr/bin/mpg123 |
04:00.33 | doughecka | no, did you put my line in there? |
04:00.39 | doughecka | exten => 2225,2,MP3Player(/root/ThroneEnd.mp3) |
04:00.46 | doughecka | its called MP3Player |
04:00.49 | doughecka | not mpg123 |
04:01.00 | espenz | root@guestbox:~# type MP3Player |
04:01.00 | espenz | sh: type: MP3Player: not found |
04:01.01 | espenz | ? |
04:01.08 | doughecka | in asterisk |
04:01.11 | espenz | ok |
04:01.12 | doughecka | its called mp3player |
04:01.28 | doughecka | the app that is needed for playing music is called mpg123 |
04:01.46 | decode | and you'd want to do type $(which mpg123) |
04:01.47 | decode | :) |
04:01.55 | doughecka | :P |
04:02.05 | decode | or `` instead of $() but that doesn't nest worth shit |
04:02.23 | doughecka | I WANT MY FILES IN 0.13 SECONDS! |
04:02.47 | decode | i just want a cute, geeky chick who understands i'm extremely impatient and will be on time or at least fucking call |
04:03.00 | doughecka | hahaha |
04:03.01 | decode | is that as bad as asking for google?:P~ |
04:03.10 | decode | I do not wait more than 5 minutes, for anyone |
04:03.22 | decode | unless they wanna pay me to wait |
04:03.35 | denon | think of it this way .. if you had google on your PC .. it'd take so much ram and storage, you couldnt even boot .. :) |
04:03.39 | doughecka | bah |
04:03.41 | denon | much less get 2 second queries :) |
04:03.43 | doughecka | they have google machines now |
04:03.50 | doughecka | a little 1u box |
04:03.52 | denon | yeah .. but its not the whole index |
04:03.53 | espenz | doughecka: it works, but it has a crazy sound |
04:03.55 | espenz | why? :P |
04:03.58 | doughecka | that searches your whole network |
04:04.02 | doughecka | hmm |
04:04.05 | denon | network yeah |
04:04.07 | denon | internet no |
04:04.07 | doughecka | what OS? |
04:04.11 | doughecka | denon: yea |
04:04.17 | denon | FreeBSD owns you |
04:04.22 | denon | and with that, im gonna go watch a West Wing |
04:04.24 | decode | how does it search? i have no ports open on 99% of my boxen |
04:04.31 | denon | its runs on your lan |
04:04.35 | doughecka | :P |
04:04.36 | doughecka | NFS |
04:04.38 | doughecka | whatever |
04:04.53 | decode | why not just set my root password to root, and enable remote root logins? :) |
04:04.55 | espenz | doughecka: do i have to choose bitrate? |
04:05.02 | espenz | og insert a encoder ? |
04:05.07 | doughecka | decode: haha |
04:05.11 | espenz | or |
04:05.12 | doughecka | espenz: no... |
04:05.20 | doughecka | what os is it? redhat? |
04:05.22 | espenz | why, is the sound so crazy.. :/ |
04:05.24 | espenz | slackware |
04:05.27 | doughecka | hmm |
04:05.28 | decode | that's my new phrase whenever a client asks for something insecure |
04:05.31 | Corydon76-home | denon: West Wing? You're lagged 26 hours, aren't you? |
04:05.48 | decode | "Sure.. and while i do that, i'll set root passwd to root and enable remote root logins" |
04:06.19 | doughecka | try getting another copy of mpg123 |
04:06.24 | doughecka | it might not be the "real" version |
04:06.26 | decode | like a client earlier today insisting we install wu-ftpd |
04:06.37 | doughecka | some distros put mpg321 and just link it |
04:06.40 | decode | eww |
04:06.43 | doughecka | HAHAHAHHAHAHAHHA |
04:06.43 | decode | mpg321 |
04:06.50 | Corydon76-home | wuftpd is perfectly safe, as long as it's the current version |
04:07.10 | Corydon76-home | It's the old versions that hurt. |
04:07.12 | decode | mpg321: "Now leaves a bitter taste in your mouth and a painful bleeding in your ears!" |
04:07.28 | decode | Corydon76-home *shrugs* based on it's past history, i still don't trust it |
04:07.48 | decode | same with pine and about anything else written by WU |
04:08.11 | decode | almost to the extent i'm literally afraid of sendmail :) |
04:09.00 | Corydon76-home | Actually, sendmail has a distinct advantage. |
04:09.05 | bevins | Hey guys, I have a bad module on a TDM400. I have no dial tone on it. I change it to another port and the problem moves. My question is how do I order just the port on the digium site? |
04:09.21 | bevins | Just the module. |
04:09.27 | Corydon76-home | If your machine is ever compromised by a spammer, they'll only be able to send a fraction of the spam of qmail. |
04:09.37 | decode | Corydon76-home i like postfix :) |
04:10.05 | Corydon76-home | But it's plenty fast for a small company |
04:10.13 | Mike | can i use a quicknet card to plug any telephone i want? and give it an extension? |
04:10.52 | Corydon76-home | bevins: why not place a sales call to Greg or Malcolm and get a swapout for that module? |
04:11.44 | decode | bc he accidently connected the 220v instead of 48v to some network gear and fried it resulting in the dead module? :P~ |
04:12.05 | Corydon76-home | Mike: I think so, but the TDM400P is ultimately cheaper per telephone. |
04:12.12 | decode | i had a moron client do something of that sort then try to say we did a faulty install |
04:13.03 | decode | how can you connect 220 to a PoE switch instead of the 48v bus? |
04:13.26 | bkw_ | blah |
04:13.47 | jsharp | blah indeed. |
04:15.11 | espenz | wtf |
04:15.13 | espenz | suddently |
04:15.15 | espenz | Frame# 1100 [ 0], Time: 0.03 [28.71], [ 459314] |
04:15.20 | espenz | came in the cli |
04:15.23 | espenz | uuh |
04:18.00 | decode | heh, anyone know where to get some decently cheap serial terms that can do ansi colour? :) |
04:18.25 | decode | i just acquired a 16 port cyclades card |
04:18.31 | decode | for 5$ |
04:20.57 | daork | heh, cool |
04:21.12 | jsharp | Get a bunch of DEC VT520s and you've got yourself a call center. |
04:21.18 | daork | now all you need is some 9k6 modems and you can make an isp! |
04:21.20 | jsharp | They're not ANSI color. |
04:21.23 | jsharp | though. |
04:21.36 | bevins | how do you kill a zap channel to knowck someone off? |
04:21.41 | decode | jsharp i'd really like colour |
04:23.00 | decode | jsharp and we just use p133's for that |
04:23.01 | decode | :) |
04:23.11 | decode | we can get about 150 p133's for 200$ |
04:23.34 | decode | but then monitors/etc cost too :) |
04:23.41 | jsharp | Yeah, but then you gotta add monitors, keyboards, hard drives or network bootable cards... |
04:23.52 | decode | jsharp they all have bootable NICs in em |
04:24.03 | decode | you can get working pulls from .gov installs cheap :) |
04:24.11 | jsharp | Yeah. |
04:24.38 | jsharp | I used to buy them almost by the truckload from a surplus dealer in Baltimore. |
04:24.43 | decode | i hate green/amber on black crt's.. and would never subject someone to such pain |
04:25.06 | jsharp | That's why you make sure to get the VT520s with the white phosphor. |
04:25.15 | decode | OOo |
04:25.20 | bevins | Is there someway to knock someone off a Zap channel? destroy? |
04:25.22 | decode | those would be eye friendly :) |
04:25.29 | decode | bevins softhangup |
04:25.42 | jsharp | Yah. I love mine to death. I'd never part with it. |
04:25.44 | decode | as for usage, i have no idea |
04:25.48 | decode | jsharp heh, est price? |
04:26.27 | *** join/#asterisk _gorman (~lehmann@pD950FE4B.dip.t-dialin.net) |
04:27.58 | decode | jsharp i'm pondering getting a 32 port cyclades card, some terminals, some decent ip phones, and setting up asterisk w/ a module to speak to some CRM sw w/ a curses interface heh |
04:28.22 | jsharp | I've been working on something similar. |
04:28.34 | jsharp | An answering service package. |
04:29.00 | decode | ahh, i'm debating doing outsourced cust. service for local small businesses :) |
04:29.06 | decode | and phone ordering/etc |
04:29.22 | jsharp | But I hadn't thought about doing it with dumb terminals...I was going to wrap the software and softphone all into one package. |
04:29.28 | decode | i have about 6 years of experience in marketing :) |
04:29.30 | jsharp | So you just need a PeeCee with a soundcard. |
04:29.38 | decode | well direct marketing |
04:29.46 | decode | heh.. aka telemarketing/fundraising/etcx |
04:29.48 | jsharp | Or hell, it doesn't even need to be a PeeCee. I was doing development on a Sparc 5. |
04:30.15 | atacomm | wow, AT&T rocks |
04:30.27 | *** join/#asterisk h3x (~hex@user57.net472.lv.sprint-hsd.net) |
04:30.28 | learath | Blasphemy! |
04:30.56 | atacomm | ... lets sue eBay and Paypal because we patented payments over a communications system in 1994, rofl |
04:30.59 | decode | jsharp i was thinking about actually having one of our dever's write a decent softphone that will fit into the curses interface for the CRM :) |
04:31.46 | jsharp | Yup. That's what I was doing. That part was easy. |
04:31.48 | decode | heh |
04:31.54 | atacomm | of course, maybe AT&T will team up with SCO since AT&T used to own it, buy them back, and continue the suing over linux/bsd |
04:31.55 | decode | i dunno what approach we'll take yet |
04:32.14 | decode | It's all going to depend on a few meetings :) |
04:37.37 | Corydon76-home | wb, kram |
04:37.42 | kram | thanks |
04:37.54 | Corydon76-home | Sure has been a busy week on the mailing list |
04:38.03 | Corydon76-home | and contentious |
04:39.39 | bkw_ | yep |
04:41.14 | *** join/#asterisk mrgoby (~mrgoby@pcp05304587pcs.wanarb01.mi.comcast.net) |
04:42.05 | mrgoby | howdy all !! |
04:43.08 | *** join/#asterisk h3x (~hex@user57.net472.lv.sprint-hsd.net) |
04:44.59 | *** join/#asterisk ant_wood (~ant_wood@hoochie.digium.com) |
04:51.42 | *** join/#asterisk phsdshft (foobar@h00080e299383.ne.client2.attbi.com) |
04:51.51 | phsdshft | bkw: you there? |
04:54.35 | phsdshft | kram: are you there? |
04:54.55 | kram | i'm having private chat time with bkw |
04:55.43 | mrgoby | what patch is the topic refering to? |
04:55.49 | mrgoby | or rather what is the bug? |
04:55.57 | mrgoby | wow, i must be lazy |
04:56.00 | mrgoby | the number is right there |
04:56.05 | mrgoby | :---) |
04:56.08 | bkw_ | phsdshft yes i'm here... |
04:56.09 | phsdshft | kram: oh hehe.. So your probably aware that I revised the original patch to not use strtok |
04:56.54 | phsdshft | bkw: I just was monitoring the bug db.. but I take it you are already talking to kram about it |
04:57.04 | bkw_ | phsdshft yes I informed him |
04:57.14 | bkw_ | I just msged you what kram recommended |
04:57.46 | phsdshft | cool |
04:57.47 | bkw_ | its more elegant anway.. because we can't assume that strsep will return non-null |
04:57.52 | phsdshft | ah |
04:58.21 | bkw_ | also not on my diff we need to move that first if ending } to the bottom of the dring stuff |
04:58.34 | bkw_ | because we are only going thru that mess if we have them set |
04:58.43 | phsdshft | ah.. I haven't even looked at your diff yet.. I have had like no time :( |
04:58.49 | bkw_ | :) |
04:58.49 | *** join/#asterisk Lafinion (~Lafinion@test.incracow.com) |
04:59.07 | bkw_ | I took the bug tracker today.. turned it upside down.. and beat the hell out of it! |
04:59.08 | bkw_ | haha |
04:59.16 | phsdshft | lol |
04:59.55 | *** join/#asterisk cman (~arun@202.51.76.140) |
05:00.01 | cman | hello |
05:00.15 | cman | i cannot dial iaxComm from zap phones |
05:00.21 | cman | i can di he reverse |
05:03.21 | *** part/#asterisk decode (~decode@ip-wv-68-117-144-207.charterwv.net) |
05:03.30 | *** join/#asterisk decode (~decode@ip-wv-68-117-144-207.charterwv.net) |
05:03.40 | decode | oh fscking peachy |
05:03.53 | decode | winamp just killed my sound drivers.. god i hate windows... |
05:03.58 | *** join/#asterisk heller (heller@voltar.wacked.org) |
05:04.36 | Adam | how is windows to blame for shitty sound drivers? |
05:04.41 | *** join/#asterisk cocoy (strange123@ipdial-171-178.tri-isys.com) |
05:04.50 | cocoy | hi people |
05:04.54 | cocoy | can any help me |
05:05.03 | Adam | ask and you shall see |
05:05.10 | cocoy | about setting SIP user on asterisk |
05:05.20 | *** join/#asterisk Zebble_ (~Zebble@66.207.107.50) |
05:05.35 | Adam | ask the question |
05:05.36 | cocoy | so that my SIP client software can loggin |
05:05.51 | cocoy | how do i create sip user on asterisk? |
05:06.34 | cman | [sipuser] |
05:06.34 | Adam | look at examples in /etc/asterisk/sip.conf |
05:06.43 | cman | type=friend |
05:06.48 | cman | username=sipuser |
05:06.55 | cman | secret=sipuser |
05:07.00 | cman | host=dynamic |
05:07.21 | cman | context=sip |
05:07.37 | cman | now in extensions.conf.. u have to diefine sip context |
05:07.39 | cman | [sip] |
05:07.57 | cman | exten=>100,1,Dial(SIP/phone1,20,r) |
05:08.00 | cman | tr |
05:08.05 | cman | r=tr |
05:08.07 | cman | etc.... |
05:09.16 | mrgoby | what are those extra parameters you put in dial cman ?? |
05:09.41 | cocoy | what if i have multiple SIP user who want to login to asterisk? |
05:09.59 | mrgoby | there is a 'catch all' user conf |
05:10.06 | mrgoby | at the top |
05:10.11 | mrgoby | of the sample config |
05:10.32 | cman | SIP can be replaced by Zap,IAX etc |
05:10.49 | cman | phone 1 is the username of the phone u dial.. such as sipuser |
05:10.56 | cman | or sipuser2 |
05:11.10 | cman | 20 is the no of sec o dial he called no |
05:11.11 | *** join/#asterisk loko_moko (loko-moko@c-67-165-107-230.client.comcast.net) |
05:11.13 | mrgoby | [general] |
05:11.13 | mrgoby | i think |
05:11.13 | mrgoby | in sip.conf |
05:11.18 | ant_wood | Anybody know chan_capi and/or Australian ISDN BRI? |
05:11.19 | cman | and tr helps u to transfer the call |
05:12.03 | mrgoby | cman wait, i was trying to figure this out before |
05:12.26 | mrgoby | do you know how to set up a call extension and have it automatically transfer the second that person picks up? |
05:12.35 | mrgoby | someone else advised i use meetme |
05:12.40 | mrgoby | and a .call file |
05:13.07 | mrgoby | is there a way i can do it in extensions.conf with a 'transfer' command? |
05:14.03 | mrgoby | i don't really want to set up a conference.... more of an automatic 2party bridge |
05:15.39 | bkw_ | cman username= is pretty much useless since the part in [] is used |
05:15.42 | cman | where is the hell password in xlite???????????????? |
05:15.51 | bkw_ | !info xlite |
05:15.52 | AiNFO | xlite - a popular SIP softphone available from http://www.xten.com for Windows and OSX. see also the getting started guide at http://www.zebraroaming.com/stuff/X-Lite-Getting-Started.pdf and example configuration for Asterisk at http://www.zebraroaming.com/stuff/X-Lite-and-Asterisk.pdf |
05:15.56 | cman | ok |
05:16.21 | jsharp | grrr. Why do telephony protocols have to be developed by committees. |
05:17.16 | bkw_ | NYQUIL IS GOOD! |
05:17.28 | mrgoby | bkw_ is that right??? the username in sip.conf is actually the bracketed name?? |
05:17.28 | bkw_ | jsharp dont think IAX was? |
05:17.28 | cman | mrgroby.. don't know |
05:17.36 | bkw_ | mrgoby yep |
05:17.49 | mrgoby | that is misleading IMHO |
05:17.49 | cman | yes mygroby.. it looks for the name in [] |
05:18.14 | cman | i had a headache when i named [] and username another... |
05:18.21 | mrgoby | well... it worked by accident in my case then :-) |
05:18.41 | cman | i couln't igure out what went wrong |
05:19.00 | ant_wood | !info chan_capi |
05:19.00 | AiNFO | chan_capi - http://www.junghanns.net/asterisk/ |
05:19.10 | ant_wood | !info fritz |
05:19.13 | mrgoby | yeah..... that can be troublesome... |
05:19.49 | cman | i need info in xlite.... |
05:19.59 | cman | hat is the codec o use... |
05:20.17 | cman | g711u.. ulaw? |
05:20.43 | bkw_ | yes |
05:20.53 | cman | i see g711u in xlite... if i need to define codec in sip.con i say allow=ulaw? |
05:21.53 | cman | i can' find out the password in xlite... damn, where has it gone? |
05:21.59 | cman | i want to reset pwd in there |
05:22.26 | cman | does anyone know it? |
05:23.26 | bkw_ | !info xlite |
05:23.27 | AiNFO | xlite - a popular SIP softphone available from http://www.xten.com for Windows and OSX. see also the getting started guide at http://www.zebraroaming.com/stuff/X-Lite-Getting-Started.pdf and example configuration for Asterisk at http://www.zebraroaming.com/stuff/X-Lite-and-Asterisk.pdf |
05:23.28 | bkw_ | read those |
05:23.35 | bkw_ | its pretty much a howto |
05:23.53 | cman | ok i found it |
05:23.53 | cman | thx |
05:25.32 | cman | ok i got it registered... but thesound quality is worst |
05:25.48 | cman | i see errors like.. Unknown RTP codec 72 reeived |
05:25.52 | cman | any idea? |
05:26.29 | *** join/#asterisk cfo (~cfo@194.19.190.217) |
05:27.18 | cocoy | i able to to logged my sip user, what extension number can I dial? |
05:27.43 | cman | how many users are there? |
05:29.14 | *** join/#asterisk decode (~decode@ip-wv-68-117-144-207.charterwv.net) |
05:29.15 | decode | weee |
05:29.32 | decode | blitzrage you alive? |
05:29.47 | cocoy | one only.. |
05:29.54 | cocoy | when i dial it shows " File chan_sip.c, Line 4489 (handle_request): Failed to authenticate user " |
05:30.18 | cocoy | sorry guys, am just starting to learn how this asterisk works |
05:30.43 | mrgoby | NOTICE[1217662256]: File sched.c, Line 209 (sched_settime): Request to schedule in the past?!?! |
05:30.47 | cocoy | i already added my user on "sip.conf" , am i right? |
05:30.50 | mrgoby | does anyone know what this means? |
05:30.54 | mrgoby | from the current build? |
05:30.56 | decode | brb, one more reboot |
05:30.57 | *** part/#asterisk decode (~decode@ip-wv-68-117-144-207.charterwv.net) |
05:31.52 | bkw_ | mrgoby it means you need to stop running in debug mode so you wont ask stupid questions |
05:32.04 | bkw_ | ;) |
05:32.15 | mrgoby | ouch |
05:32.26 | bkw_ | oh lord poor decode.. thinks he's running windows or something. |
05:32.28 | mrgoby | bkw_ that really hurts man |
05:32.31 | mrgoby | really |
05:32.33 | mrgoby | :-D |
05:32.40 | bkw_ | mrgoby haha sowwy.. i'm really blunt |
05:33.05 | cman | whats ur sip user??? |
05:33.07 | cman | xlite? |
05:33.11 | bkw_ | to be honest you shouldn't be in debug mode like that till you understand that most of that stuff is just informational and has really no meaning unless your hunting down a problem. |
05:33.20 | cocoy | i use xlite, ren@routpeer.com |
05:33.31 | mrgoby | i hear ya.... vvv is force of habit i guess |
05:33.38 | bkw_ | I use safe_asterisk |
05:33.40 | bkw_ | then asterisk -r |
05:33.46 | bkw_ | tail -f /var/log/asterisk/messages if I need debug |
05:33.55 | mrgoby | gotcha |
05:34.08 | bkw_ | that way you can exit without killing * |
05:34.11 | mrgoby | that's not a bad idea |
05:34.16 | bkw_ | or having to run it in a screen like some goobs do |
05:34.30 | cocoy | i got my user logged in to asterisk, but says can't authenticate user when i dial |
05:34.33 | mrgoby | screen? |
05:34.45 | bkw_ | mrgoby don't ask |
05:34.58 | cman | bkw |
05:35.00 | mrgoby | sorry, bad habit that |
05:35.15 | mrgoby | i'm an interrogative-type |
05:35.19 | *** join/#asterisk IronHelixz (IronHelix@ool-182c7020.dyn.optonline.net) |
05:35.30 | cman | what can be done??? i get unknopwn rtp codec 72 received.. in using xlite and the sound isn't that good |
05:35.55 | cman | if u have any idea? |
05:36.27 | bkw_ | cman its xlite. |
05:37.42 | cman | any guy from nufone? |
05:38.29 | bkw_ | just call for the NuFone gods.. they will decend apon you. |
05:38.57 | *** join/#asterisk GotX (~Om3gAnGeL@pcp01540411pcs.huntsv01.al.comcast.net) |
05:39.05 | cocoy | my sip user still Fails when dialing someone |
05:39.16 | bkw_ | cocoy what is the error message? |
05:39.44 | cman | did ur sip regiser ith *? |
05:39.57 | cocoy | <PROTECTED> |
05:40.03 | mrgoby | boy... sjphone sure is purty |
05:40.13 | cocoy | i use xlite |
05:40.16 | cman | that means u haven't registered |
05:40.18 | bkw_ | !info xlite |
05:40.18 | AiNFO | xlite - a popular SIP softphone available from http://www.xten.com for Windows and OSX. see also the getting started guide at http://www.zebraroaming.com/stuff/X-Lite-Getting-Started.pdf and example configuration for Asterisk at http://www.zebraroaming.com/stuff/X-Lite-and-Asterisk.pdf |
05:40.20 | bkw_ | did you read that? |
05:40.28 | cman | !info xlite |
05:40.28 | AiNFO | xlite - a popular SIP softphone available from http://www.xten.com for Windows and OSX. see also the getting started guide at http://www.zebraroaming.com/stuff/X-Lite-Getting-Started.pdf and example configuration for Asterisk at http://www.zebraroaming.com/stuff/X-Lite-and-Asterisk.pdf |
05:40.33 | bkw_ | cman you don't have to register to make calls.. you just have to have the info right. |
05:40.38 | cman | oh waaa.. ididn't kno tha |
05:40.39 | Mike | can i use a quicknet card to plug any telephone i want? and give it an extension? |
05:40.40 | mrgoby | !info sjphone |
05:40.42 | bkw_ | so when it goes to make the call it can auth with that info |
05:40.43 | mrgoby | doh! |
05:40.59 | bkw_ | cman stick around you will learn grass hopper |
05:41.01 | jsharp | Mike: Yes. |
05:41.11 | mrgoby | bkw_ how you do that? info? |
05:41.20 | GotX | !info |
05:41.21 | cocoy | my user already regsitered, why when i dial any number it says Fail to authenticate user <sipuser> |
05:41.24 | Mike | so i can put my quicknet card and plug the phone and how do i give it an extension |
05:41.24 | Mike | ? |
05:41.25 | cman | i knew that sip don't have to register |
05:41.26 | mrgoby | i'm new to irc as well..... NEEEWWWW-BEEEEE |
05:41.33 | bkw_ | show us your sip.conf entry |
05:41.39 | cman | i didn't know !info could bring that ino |
05:42.03 | GotX | whoa...my screen just turned pink |
05:42.06 | cman | from the error it seems that he is trying to register xlite to * |
05:42.08 | GotX | thats kinda cool |
05:42.11 | mrgoby | !info xlite |
05:42.13 | AiNFO | xlite - a popular SIP softphone available from http://www.xten.com for Windows and OSX. see also the getting started guide at http://www.zebraroaming.com/stuff/X-Lite-Getting-Started.pdf and example configuration for Asterisk at http://www.zebraroaming.com/stuff/X-Lite-and-Asterisk.pdf |
05:42.18 | mrgoby | yah! |
05:42.22 | mrgoby | sorry, guys |
05:42.23 | cocoy | [general] |
05:42.23 | cocoy | port = 5060 |
05:42.23 | cocoy | bindaddr = 203.144.224.167 |
05:42.29 | GotX | lol @ mrgoby |
05:42.33 | cocoy | [ren] |
05:42.34 | cocoy | type=friend |
05:42.34 | cocoy | host=dynamic |
05:42.34 | cocoy | secret=zxcasd |
05:42.34 | cocoy | context=sip |
05:42.34 | cocoy | permit=0.0.0.0/0.0.0.0 |
05:42.51 | bkw_ | oh shit |
05:42.53 | bkw_ | can I smack you know |
05:42.57 | cman | u don't have to define permi and bind addre.... |
05:43.14 | bkw_ | cman you do if the hostname won't resolve |
05:43.21 | cman | so make 2 or 3 sip users |
05:43.22 | bkw_ | because chan_sip won't load if the hostname isn't resolvable |
05:43.29 | bkw_ | that permit part can go |
05:43.42 | cman | ok |
05:43.56 | cman | is ur configuration in xlite- ok?? |
05:44.05 | cman | doe it say logging in.. and logged in?? |
05:44.12 | cman | when u start |
05:44.16 | cman | xlite |
05:44.41 | *** join/#asterisk Landrocker (siC591746@203-118-171-237.adsl.ihug.co.nz) |
05:45.02 | cocoy | yes says logged in |
05:45.16 | cocoy | but whne i dial says fail to authenticate |
05:45.19 | cman | so what did u define in extensions.conf |
05:45.36 | Landrocker | setting up x-lite with *? |
05:45.39 | bkw_ | you don't have to define it to dial |
05:45.46 | cman | ap |
05:45.48 | bkw_ | you just have to have a sane dialplan |
05:45.55 | bkw_ | or a dialplan at all for that matter |
05:46.00 | bkw_ | sip show peers |
05:46.02 | bkw_ | shows it registered? |
05:46.16 | cman | ye |
05:46.29 | cocoy | what do i need to put in realm? |
05:46.30 | bkw_ | show me |
05:46.34 | bkw_ | no realms |
05:46.39 | bkw_ | wel |
05:46.42 | bkw_ | thats the ip of your * server |
05:46.45 | bkw_ | if I recall |
05:46.47 | cman | in the CLI> type sip show peers |
05:46.56 | cman | u shpould get name/'username hosts etc |
05:46.56 | bkw_ | I don't use xlite because its crap |
05:47.10 | cman | what do u use bk_ |
05:47.14 | Landrocker | what do you use bkw_? |
05:47.43 | cman | i treid DIAX.. eco problem.. iaxcomm.. bugs.. no trying lite... sound problems.. |
05:47.43 | bkw_ | hardware phones.. ata-186's and 7960's |
05:48.06 | Adam | patience i say, await my soft iax phone |
05:48.33 | cocoy | am logged in now, what do i dial? |
05:48.43 | bkw_ | dial 1000 |
05:48.45 | Landrocker | bkw_: /me is too poor for voip phones :( |
05:48.45 | bkw_ | for the demo |
05:48.46 | Adam | actually i wanted to ask Mark some questions |
05:48.59 | bkw_ | Landrocker I am too.. but we all have to sacrifice something |
05:49.11 | cocoy | "404 Not found" |
05:49.20 | *** join/#asterisk clive- (~pirch@rndf-ip-nas-1-p160.telkom-ipnet.co.za) |
05:49.28 | bkw_ | what context you have in your sip.conf peer entry? |
05:49.28 | mrgoby | has anyone had problems with mic capture on linphone with alsa ??? |
05:49.45 | *** part/#asterisk GotX (~Om3gAnGeL@pcp01540411pcs.huntsv01.al.comcast.net) |
05:50.22 | Adam | Mark/Kram: With IAX2, user A isn't NAT'd user B is NAT'd, if * does a supervised transfer, will they realise this (A will receive a packet, B won't) and be able to talk directly? |
05:50.26 | cocoy | how do i call the voicemail? i dialed 1000 , it said "404 Not found" |
05:50.50 | bkw_ | 8500 |
05:51.01 | bkw_ | but WHAT IS YOUR CONTEXT= LINE IN SIP.CONF or do I need to yell? |
05:51.03 | bkw_ | :P |
05:51.12 | bkw_ | you have to specify a context for your device |
05:51.43 | cman | where do i define incoming callerid so that i can see it? |
05:52.21 | bkw_ | cman have you even googled yet? |
05:52.46 | mrgoby | well, cman , i do not recommend linphone |
05:52.54 | mrgoby | or at least not for me |
05:52.57 | bkw_ | hehe |
05:53.01 | bkw_ | use hardware phones they ROCK |
05:53.14 | mrgoby | i need to make my 'boss' buy one |
05:54.17 | mrgoby | i like sjphone so far, although the sip config is limited, or at least non-intuitive |
05:54.27 | mrgoby | but |
05:54.28 | cman | can anone tell me how to receive caller id if someone dials from outside |
05:54.33 | mrgoby | no leenuchs client |
05:54.43 | mrgoby | :-( |
05:54.43 | cman | i have usecallerid=yes in zapata |
05:54.53 | cman | i don't think that is the one |
05:55.08 | bkw_ | cman it just happens |
05:55.35 | cman | i am not geting caller id |
05:55.42 | bkw_ | is your telco sending it |
05:55.42 | mrgoby | anyone used dynExtenDB ??? |
05:56.31 | *** join/#asterisk monsieur (~monsieur@81-86-185-223.dsl.pipex.com) |
05:57.03 | cocoy | how can a sip user make a call that will pass thru * then to gatekeeper |
05:57.12 | cman | yes.. when i connect directly i gives me caller id.. but if i connect thru * it is not giving |
05:57.22 | monsieur | has someone got a script for running through all the kernel recompile steps in an automated way? |
05:57.33 | mrgoby | canreinvite=no forces the audio through * , no? |
05:57.42 | monsieur | mrgo: ye |
05:58.08 | mrgoby | cocoy i would start by adding canreinvite=no to sip.conf |
05:58.28 | mrgoby | then route through gatekeeper |
05:58.37 | cman | bk |
05:59.24 | cocoy | anymore things i need to change other than on sip.conf, to do sip->*->gk |
05:59.38 | mrgoby | dunno cocoy |
06:00.01 | cocoy | canreinvite=no, where? on [general] ? |
06:00.22 | *** join/#asterisk Exomorph (Greg@134-9.bvcompuworks.com) |
06:00.32 | *** part/#asterisk Exomorph (Greg@134-9.bvcompuworks.com) |
06:01.07 | mrgoby | i don't know [general] confuses me, you set globals there for the whole sip conf .... but which are valid there an which are valid under particular user configs? |
06:01.13 | Mike | anyone can help me with 1800 numbers? |
06:01.20 | Mike | with sipphone or fwd or aixtel |
06:01.22 | Mike | iaxtel |
06:01.25 | Mike | or any? |
06:03.34 | mrgoby | sssssooooooooooo, how about that dynamic extensions module, dynExtenDB, boy that's a cool one, eh?? anyone used that?? :-) |
06:03.48 | mrgoby | anyone? |
06:03.55 | mrgoby | ;-) |
06:05.09 | monsieur | mrgo: we used it. Apparently it doesn't scale very well but we haven't tried on big installations |
06:07.29 | mrgoby | that is what i read from Uriel's post |
06:07.33 | *** join/#asterisk ant_wood (~ant_wood@hoochie.digium.com) |
06:07.33 | mrgoby | is that you? |
06:07.36 | mrgoby | ;-D |
06:07.41 | Mike | <PROTECTED> |
06:07.41 | Mike | <PROTECTED> |
06:07.41 | Mike | <PROTECTED> |
06:07.41 | Mike | <PROTECTED> |
06:07.41 | Mike | <PROTECTED> |
06:07.43 | Mike | it dies |
06:08.22 | mrgoby | did you have to modify the source to get it to work monsieur ? |
06:09.01 | monsieur | no |
06:09.22 | mrgoby | hmmmm.... very interesting.... |
06:10.28 | cocoy | can anyone send me a working sip.conf sample? and extensions.conf too |
06:10.29 | *** join/#asterisk Exomorph_ (Greg@134-9.bvcompuworks.com) |
06:10.37 | *** part/#asterisk Exomorph_ (Greg@134-9.bvcompuworks.com) |
06:11.06 | mrgoby | monsieur , by not scaling, do you mean that on a single machine several sip extensions in the DB effects performance significantly?? |
06:11.57 | mrgoby | underscore rubbing you the wrong way gorman?? |
06:12.45 | *** join/#asterisk tristan2 (~tristan@213.239.44.133) |
06:12.52 | monsieur | i'm talking about 250+ extensions. But don't take my word for it. |
06:13.32 | mrgoby | cocoy: the sample config files work just fine for me as a basis for building new extensions..... what are you trying to do ?? |
06:14.25 | cman | i can't get caller id??????????? |
06:14.30 | cman | but why? |
06:15.39 | mrgoby | shoot ... andreasotto.net is down |
06:16.24 | *** join/#asterisk davidhindmarsh (~dhindmar@CPE-144-137-125-115.nsw.bigpond.net.au) |
06:17.21 | mrgoby | Mr. Otto .... are you there ?? |
06:17.59 | mrgoby | andio where are you?? |
06:18.38 | davidhindmarsh | Hi, anybody used the H.323 and Sip to provide a gateway betwwen the two protocols. SIP > H.323 and H.323 > SIP |
06:19.41 | mrgoby | davidc: yes |
06:19.52 | mrgoby | davidhindmarsh: yes |
06:20.10 | davidhindmarsh | I have a request to bring some H.323 users into asterisk, what components do I need to install and where |
06:20.45 | mrgoby | hmmmm ... honestly, i didn't set it up ... my partner did ... |
06:20.54 | mrgoby | have you tried it? |
06:21.04 | jsharp | read the README in asterisk/channels/h323 |
06:21.15 | mrgoby | i believe you have to recompile H.323 into your kernel |
06:21.19 | mrgoby | that is what we did |
06:21.27 | mrgoby | i believe |
06:21.49 | davidhindmarsh | I have installed the h.323 and that is apparently working, I see it loads and listens, then what? |
06:22.02 | mrgoby | did you try calling ? |
06:22.33 | davidhindmarsh | I could not get netmeeting to register with the asterisk server. |
06:23.17 | mrgoby | so, it could be a registration issue then ? |
06:23.30 | monsieur | does anyone know of a script which automatically recompiles kernel and all steps? |
06:23.31 | mrgoby | sorry, actually i'm not the guy you wanna talk to ... i'm a sip-er |
06:24.15 | davidhindmarsh | most likely, does the endpoint register with asterisk or does there need to be something else involved. |
06:24.39 | jsharp | asterisk is not a gatekeeper. You can' |
06:24.50 | jsharp | t register against asterisk with h323. |
06:25.13 | mrgoby | monsieur: do you have those dynextendb files handy??? otto's server is offline |
06:25.41 | davidhindmarsh | If you can't register how does it work |
06:26.20 | davidhindmarsh | Do i need to install a gatekeeper |
06:26.31 | mrgoby | ggk |
06:26.41 | mrgoby | gnu gate keeper is one i've heard is good |
06:26.54 | jsharp | yes. Gnugk |
06:27.34 | davidhindmarsh | Ok, so endpoint registers with GNUGK, how does it talk to asterisk endpoints who are on SIP |
06:27.53 | jsharp | It can talk directly to SIP clients. |
06:27.57 | jsharp | through chan_sip. |
06:28.33 | davidhindmarsh | how does gnugk now where sip users are. |
06:28.42 | *** join/#asterisk Lafinion (~Lafinion@test.incracow.com) |
06:28.58 | mrgoby | set extensions for them in * |
06:29.10 | mrgoby | is one way i would imagine |
06:29.36 | davidhindmarsh | what would such a config line look like |
06:29.56 | monsieur | mrgob: it was a while ago. I'll ask one of my staff |
06:29.58 | Mike | how long did iaxtel closed? |
06:30.12 | mrgoby | monsieur: thnx |
06:30.33 | mrgoby | i have it on a server, but it is offline as well !!! |
06:31.04 | danielq | monsieur: if you're using Debian, install the kernel-package deb |
06:31.11 | mrgoby | davidhindmarsh: exten 34 => Dial(SIP/user@host) |
06:31.28 | mrgoby | sorry taht is the wrong syntax |
06:31.30 | mrgoby | just a sec |
06:31.53 | monsieur | danielq: thx. |
06:32.01 | monsieur | on rh8 |
06:32.03 | monsieur | :-( |
06:32.27 | mrgoby | exten => 23,1,Dial(SIP/user@host) |
06:32.45 | mrgoby | that would dial directly to the sip user when you call 23@asteriskHost |
06:32.49 | davidhindmarsh | what about the h323 endpoints. |
06:33.19 | mrgoby | davidhindmarsh: what do you mean? |
06:33.42 | mrgoby | just call that extension on asterisk with the h323 UA |
06:33.57 | mrgoby | i haven't done this, but from my understanding that should work |
06:34.07 | jsharp | asterisk itself can register with the gatekeeper, then you have access to all of the aliases that are registered with the GK. |
06:34.08 | mrgoby | i'm kinda pulling this outta my ass though |
06:34.10 | mrgoby | :-D |
06:34.21 | mrgoby | thankyou jsharp |
06:34.26 | mrgoby | i'll shut up now |
06:34.39 | mrgoby | no advice > bad advice |
06:34.56 | mrgoby | listen to jsharp davidh |
06:35.04 | jsharp | So if you have a netmeeting client with a username of "crackbaby" registered to your GK and asterisk is registered to the GK as well, then you can set up an extension like "exten => 23,1,Dial(H323/crackbaby)" |
06:35.35 | jsharp | and extension 23 will dial the netmeeting client. |
06:36.27 | davidhindmarsh | How does the gk know about the asterisk extensions, so h323 can dial the sip ep |
06:38.18 | jsharp | your * server can register multiple aliases with the GK. |
06:38.33 | jsharp | You can register an alias for each extension. |
06:39.22 | davidhindmarsh | do you use sip@asterisk type aliases |
06:39.30 | mrgoby | jsharp does the j stand for jonas by any chance ??? |
06:39.51 | jsharp | nope. james. |
06:40.15 | mrgoby | :-) |
06:41.43 | *** join/#asterisk diana (~diana@home-25022.b.astral.ro) |
06:41.45 | diana | ehlo |
06:42.09 | jsharp | I think you'd need to register the extension numbers. |
06:42.11 | mrgoby | howdy diana |
06:42.29 | jsharp | I've not done a hybrid network before...so I'm just working on what I know. |
06:42.43 | *** join/#asterisk decode (~decode@ip-wv-68-117-144-207.charterwv.net) |
06:42.56 | l-fy_ | jsharp > what do you mean by hibrid? |
06:43.02 | l-fy_ | hybrid? |
06:43.02 | decode | anyone happen to know where i can perhaps obtain a database mapping NXX-NPA's to zip codes? :) |
06:43.05 | l-fy_ | damn english |
06:43.33 | davidhindmarsh | can i run the gnugk on the same machine |
06:43.50 | mrgoby | hybrid H323 <---> SIP |
06:43.53 | l-fy_ | davidhindmarsh > yes |
06:43.58 | mrgoby | !info hybrid |
06:44.02 | l-fy_ | mrgoby > use a proxy |
06:44.05 | decode | or hell, a complete list of NXX+NPA's preferably including carrier's for mobile NPA's? |
06:44.10 | decode | and service type designations? |
06:44.26 | decode | i just obtained a zip+4 database.. :) |
06:44.30 | mrgoby | !info proxy |
06:44.35 | mrgoby | doh! |
06:45.04 | mrgoby | i do know what that is ... just checking if AiNFO does |
06:45.21 | l-fy_ | who's AiNFO |
06:45.25 | l-fy_ | !info hybrid |
06:45.30 | l-fy_ | k |
06:45.42 | davidhindmarsh | can i run the gnugk on the same machine as asterisk |
06:45.47 | l-fy_ | i hate openh323 |
06:45.52 | jsharp | Tru http://www.nanpa.com/nanp1/AllCodes.zip |
06:45.55 | l-fy_ | takes way to much to do anything |
06:46.21 | jsharp | davidhindmarsh: No. Both the h323 channel of * and gnugk try to bind to the same tcp/udp ports. |
06:46.33 | decode | http://docs.nanpa.com/npa/allnpas.zip that's the NPA's.. hmm |
06:46.50 | decode | now what about exchanges? |
06:47.01 | l-fy_ | jsharp > you can bind on another port |
06:47.16 | l-fy_ | anyway gnugk want to bind himself on 1719 |
06:47.28 | l-fy_ | and asterisk on 1720 |
06:47.50 | jsharp | yes, you can change port bindings...then you have to change ports on everything that needs to talk to the application you changed ports on. |
06:48.10 | l-fy_ | jsharp > anuway gnugk is on 1719 |
06:48.14 | l-fy_ | les me take a look |
06:48.55 | l-fy_ | sorry |
06:48.57 | l-fy_ | on 1721 |
06:49.01 | l-fy_ | i was close |
06:49.08 | l-fy_ | CallSignalPort=1721 |
06:50.11 | cman | why ami not getting callerid? |
06:50.16 | cman | thru * box? |
06:51.03 | l-fy_ | cman > what driver |
06:51.03 | l-fy_ | ? |
06:51.05 | denon | so .. someoen tell me .. what's a good "poor man's" packet shaper? open source stuff? |
06:52.08 | phsdshft | denon: um.. your just wanting to do rate-shaping while using linux? |
06:52.10 | l-fy_ | hi denon |
06:52.17 | jsharp | denon: I'm using OpenBSD's pf. |
06:52.38 | denon | hey l-fy |
06:52.48 | cman | driver? meaning? |
06:52.55 | cman | FXO |
06:52.57 | jsharp | A combination of priority queing and cbq queing. |
06:53.01 | denon | think I'd stick to FreeBSD, now that it can do true bridging a bit better |
06:53.04 | jsharp | I wish I could spell. |
06:53.12 | denon | I like OpenBSD for that kinda stuff, but FreeBSD is just so damn smooth to use |
06:53.35 | l-fy_ | sorry to putting such a stupid question |
06:53.42 | l-fy_ | but why not linux? |
06:53.52 | l-fy_ | you have a bandwidth more the 2 Mb? |
06:53.52 | denon | I'm not real fond of linux .. |
06:54.04 | l-fy_ | what's fond? |
06:54.13 | denon | well .. yes, we do .. but I'd not trust our core network to a x86 router |
06:54.21 | denon | l-fy: I don't like it. |
06:54.24 | phsdshft | bkw: are you there? |
06:54.33 | l-fy_ | denon > ok, i can understand this |
06:54.34 | l-fy_ | btw |
06:54.36 | denon | we run juniper and packeteer for core applications |
06:54.43 | phsdshft | heh packeteer |
06:54.44 | clive- | denon I belive lartc.org is good |
06:54.48 | l-fy_ | i do have my core network on linux |
06:55.06 | denon | l-fy_: multiple OC12s and an OC192 sonet ring? :) |
06:55.13 | l-fy_ | denon > you need much more then i need |
06:55.22 | l-fy_ | denon > this why i have ask you |
06:55.24 | phsdshft | why use packeteer |
06:55.30 | phsdshft | why not use rate-shaping on the juniper? |
06:55.38 | denon | what I need is a poor man's .. somethin I can toss at remote offices to prioritize voice, etc |
06:55.43 | l-fy_ | i know the differences between bsd and linux, and how is junos |
06:56.25 | decode | blah |
06:56.35 | jsharp | OpenBSD/pf on a sparc machine. |
06:56.41 | decode | bbl, going to try and find more databases >:) |
06:57.19 | phsdshft | denon: Why not use rate-shaping on the juniper? |
06:57.27 | denon | phsdshft: these are at remote offices .. |
06:57.32 | denon | running crappy pipes like cable and dsl |
06:57.38 | denon | cheap, dinky little networks |
06:57.46 | denon | where we cant justify more than a cheapo router and stuff |
06:57.50 | phsdshft | You can specify source/remote ips to rate-shaping |
06:57.59 | phsdshft | on your core network (if you so desired) |
06:58.02 | denon | not when I dont control the .. yeah .. |
06:58.06 | phsdshft | or your gateway router |
06:58.06 | denon | see .. its not offices on our netowrk |
06:58.11 | phsdshft | ah well |
06:58.28 | denon | they're crappy places in timbuktu that buy local bw and vpn in |
06:58.49 | phsdshft | oh you use packeteer on their sites |
06:59.19 | denon | packeteer on medium sized stuff |
06:59.25 | denon | nothing on smaller stuff.. but I need to |
06:59.30 | Mike | anyone has 1800 numbers working with asterisk + GS with FWD sipphone.com or iaxtel??? i just need a little hand |
06:59.35 | denon | seeing as though dsl/cable/etc is pretty damn unrepdictable |
06:59.36 | phsdshft | lol coo |
06:59.39 | phsdshft | anyway |
06:59.49 | denon | this lartc.org looks kinda cool |
07:00.25 | decode | Mike exten => _1800,1,Dial(SIP/${EXTEN}@fwd) or such should work, no? |
07:00.37 | Mike | decode: its hanging up my call |
07:00.44 | Mike | decode: i have about 15 examples |
07:01.12 | decode | hmm any way to convert from an access db to a mysql db? :) |
07:01.24 | jsharp | exten => _1NXXNXXXXXX,Dial(SIP/${EXTEN}@fwd) |
07:01.33 | decode | besides warez'ing office, installing it, and exporting to csv? :) |
07:01.34 | *** join/#asterisk bwz (~w_w_zhang@66-215-24-240.mpk-eres.charterpipeline.net) |
07:01.44 | jsharp | decode: Yah, lemme find the software I just had to use. |
07:02.04 | decode | cool |
07:02.31 | decode | i'm slowly compiling a ton of different db's related to geographic info into one heh |
07:02.33 | jsharp | Took me forever to find it originally. |
07:02.33 | bwz | anyone knows what this "sip show peers" out mean with (D) in the middle? |
07:02.37 | bwz | 75561331111/755 210.22.24.65 (D) 255.255.255.255 54328 Unmonitored |
07:02.49 | decode | phone exchanges, street level maps, etc |
07:02.49 | decode | :) |
07:03.30 | Mike | http://mike.calle69.net/exten.txt |
07:03.33 | Mike | none of those work |
07:03.48 | jsharp | decode: I think this is the one I used http://www.fonlow.com/zijianhuang/dbconverter/ |
07:04.21 | Kumbang | guys, how can i make sip client connect to * without password |
07:05.05 | l-fy_ | Kumbang > you just setup the user without a secret |
07:05.14 | l-fy_ | ok guys |
07:05.30 | l-fy_ | is there anyone here who can provide me some DS2155 chips? |
07:06.07 | Kumbang | anyone of you guys using pingtel sipphone? kinda hard to connect it |
07:06.08 | jsharp | Building a T1 card? |
07:06.22 | l-fy_ | jsharp > not really |
07:06.28 | decode | jsharp that appears to a demo that only does 5 records |
07:06.36 | jsharp | Dammit. |
07:06.42 | decode | heh, don't worry about it |
07:06.49 | decode | i'll get my friend to export it to csv tomorrow |
07:07.01 | jrollyson | how hard would that be to build? |
07:07.02 | jsharp | My wife wiped the program I used off her laptop. |
07:07.35 | jsharp | Shouldn't be that hard to build. Supporting it on a PCI bus would be tricky. |
07:07.51 | Mike | <PROTECTED> |
07:07.51 | Mike | <PROTECTED> |
07:07.51 | Mike | <PROTECTED> |
07:07.51 | Mike | <PROTECTED> |
07:07.51 | Mike | <PROTECTED> |
07:07.54 | Mike | it gets hangup |
07:09.41 | decode | extra dry ginger ale + vodka is good :) |
07:10.11 | Mike | maybe i need some speacial code |
07:10.16 | Mike | for 1800 numbers |
07:10.17 | Mike | ??? |
07:10.21 | jsharp | Mike: Do you have canreinvite=no in your fwd entry in sip.conf? |
07:10.31 | Mike | jsharp: let me see |
07:10.47 | jsharp | And in the entry for your GS phone? |
07:11.02 | Mike | in the entry of my GS i have it on fwd part i dont |
07:11.08 | *** join/#asterisk miller7- (~none@adsl49-static-gw1.access.acn.gr) |
07:11.22 | jsharp | And where's the extra * in the dialed number coming from? |
07:11.27 | jsharp | "SIP/*18006927753@f |
07:11.29 | miller7- | g'day people |
07:11.31 | jsharp | specifically. |
07:11.41 | *** join/#asterisk cocoy (strange123@ipdial-171-178.tri-isys.com) |
07:11.50 | Mike | the * is needed by fwd to dial 1800 numbers |
07:11.55 | jsharp | Oh. |
07:11.57 | decode | it is? |
07:12.01 | Mike | yes it its |
07:12.17 | decode | i never used it w/ xlite |
07:12.36 | cocoy | need help on x-lite. "484 Address incomplete" |
07:12.47 | Mike | ok i added the canretrive no |
07:12.51 | Mike | and still its getting hangup |
07:12.54 | Mike | when it answer |
07:12.59 | decode | did you reload? |
07:13.04 | jsharp | not canretrive. canreinvite. |
07:13.06 | Mike | i even stop asterisk |
07:13.07 | Mike | :) |
07:13.11 | Mike | and start it back |
07:13.13 | decode | cocoy looks like you're trying to call an invalid extension |
07:13.20 | decode | i never got x-lite working well w/ fwd tho |
07:13.26 | Mike | username=77443 |
07:13.26 | Mike | host=fwd.pulver.com |
07:13.26 | Mike | canreinvite=no |
07:13.26 | Mike | reinvite=no |
07:13.34 | cocoy | but it's there on my extensions.conf |
07:13.47 | jsharp | Hrm. |
07:14.26 | cocoy | i noticed i have dynextendb.conf , do i have to change here? |
07:14.28 | Mike | jsharp: could it be a codec thing? |
07:14.46 | jsharp | dunno. What codecs are you using? |
07:15.19 | Mike | well |
07:15.27 | Mike | disallow=all |
07:15.28 | Mike | allow=ulaw |
07:15.28 | Mike | allow=alaw |
07:15.28 | Mike | allow=g729 |
07:15.28 | Mike | allow=gsm |
07:15.28 | Mike | allow=ilbc |
07:15.30 | Mike | allow=speex |
07:15.33 | Mike | allow=lpc10 |
07:15.41 | denon | better question, what codecs are you not using? |
07:15.55 | Mike | :P |
07:15.57 | mrgoby | cocoy: do you have the dynextendb package handy ??? otto's site is down |
07:16.05 | mrgoby | i was hoping to install it tonight |
07:16.35 | cocoy | someone already installed it on my linux box.. i don't how to install it and where to get it |
07:16.57 | mrgoby | k, thnx anyway |
07:18.16 | cocoy | mrgoby... what can i do to solve my problem? |
07:18.24 | cocoy | need help on x-lite. "484 Address incomplete" |
07:18.41 | cocoy | can u give me example extension |
07:19.02 | cman | x-liteecho problem |
07:19.53 | mrgoby | cocoy what is the problem? |
07:20.02 | cocoy | anyone, "484 address incomplete" |
07:20.28 | cocoy | mrgoby- error in xlite |
07:20.29 | mrgoby | what is generating that error? |
07:20.31 | mrgoby | oh |
07:20.34 | decode | bbl, going to have to sleep for a bit.. |
07:20.52 | mrgoby | i don't use xlite but sounds like you have bad syntax on your sip address |
07:21.09 | cocoy | what to u use? |
07:21.31 | mrgoby | try variations of sip:user@host , user@host , <sip:user@host> |
07:21.46 | mrgoby | also |
07:21.55 | decode | sip:user@domain works well enough |
07:22.03 | mrgoby | i noticed that sjphone doesn't like domain names.... |
07:22.05 | cocoy | what do i put on domain? |
07:22.21 | mrgoby | so.... maybe try to specify the IP number |
07:22.28 | espenz | <PROTECTED> |
07:22.32 | espenz | hm, its playing |
07:22.38 | espenz | but i get a realy wierd sound |
07:22.39 | espenz | why? |
07:22.39 | espenz | :P |
07:23.00 | cocoy | mrgoby- i tried also specifying the IP but not happened |
07:23.24 | mrgoby | sorry cocoy ... dunno what to tell you .... try sjphone |
07:23.38 | mrgoby | i've heard lots of people complain about xlite |
07:23.38 | cman | do we really need to have register=>line in the onf iles? |
07:23.40 | *** part/#asterisk decode (~decode@ip-wv-68-117-144-207.charterwv.net) |
07:23.53 | mrgoby | what OS cocoy ?? windows? |
07:23.57 | cocoy | ok, i have my sjphone.. wait |
07:24.41 | cman | which is he best sip sotphone? |
07:24.57 | mrgoby | i can give you my experience |
07:25.06 | mrgoby | but i think all are bad at certain things |
07:25.11 | mrgoby | that is |
07:25.41 | mrgoby | there are many permutations of interactions that don't work well because of the differences between each implementation |
07:26.00 | mrgoby | ya know? |
07:26.26 | mrgoby | i use linphone or sip-communicator for linux |
07:26.32 | cocoy | mrgoby- i have my sjphone now, which part of config i need to put my * ip on sjphone? |
07:26.55 | mrgoby | and sjphone, sip-communicator, or ubiquity UA for windows |
07:27.16 | mrgoby | rephrase that cocoy |
07:27.17 | cman | where to download sjphone? |
07:27.26 | mrgoby | sjlabs.com i think |
07:27.34 | mrgoby | google sjphone |
07:27.37 | cocoy | what do i need to confgiure on sjphone? |
07:27.59 | mrgoby | the configuration of sjphone is non-intuitive for me |
07:28.00 | mrgoby | but |
07:28.33 | mrgoby | you need to go into sip tab under preferences or options and set your name, NAT info |
07:28.52 | mrgoby | then put the sip address you are trying to dial on the text bar on the UA |
07:29.07 | mrgoby | but this is all in the help on the phone :-) |
07:32.16 | *** join/#asterisk renv (strange123@ipdial-171-178.tri-isys.com) |
07:32.27 | *** join/#asterisk Inv_Arp (junya@fiudial2-84.fiu.edu) |
07:32.27 | renv | what's a SIP URL? |
07:32.30 | *** join/#asterisk FryGuy- (~fryguy@c-24-2-50-122.client.comcast.net) |
07:32.38 | mrgoby | !info sip |
07:32.39 | AiNFO | sip - Session Initiation Protocol - SIP is the IETF protocol for VOIP. It can also support video and other services (see !info sip-and-nat) |
07:32.53 | cocoy | what's a SIP URL? |
07:33.04 | *** join/#asterisk usam (usam@2001:730:11:29:0:0:0:281) |
07:33.08 | mrgoby | !info sip-and-nat |
07:33.09 | AiNFO | sip-and-nat - there are various issues with SIP and NAT traversal. The following paper by DeltaThree provides a comprehensive discussion http://www.zebraroaming.com/stuff/SIP-and-NAT-Traversal.pdf |
07:34.09 | mrgoby | g'night gentlefolks |
07:34.11 | renv | !sjphone |
07:34.22 | renv | !info sjphone |
07:34.30 | mrgoby | duddn't work... tried it earlier |
07:34.33 | mrgoby | :-D |
07:34.43 | renv | !info sip.conf |
07:35.10 | renv | anyone who knows how to set sphone? |
07:35.17 | renv | sjphone |
07:36.27 | renv | anyone? |
07:36.33 | renv | anyone who knows how to set sjphone? |
07:37.36 | *** join/#asterisk oriontkn (~oriontkn@pool0098.cvx6-bradley.dialup.earthlink.net) |
07:37.54 | oriontkn | wowzers |
07:39.33 | oriontkn | does anybody know where I can find some updated info on asterisk and dialogic? |
07:41.06 | izo | oriontkn : mailing list but there isnt much support for it |
07:44.48 | espenz | i get a wierd sound when i play mp3s with asterisk, why could that be? |
07:44.50 | *** join/#asterisk edguy3 (~edguy@host-24-225-213-50.patmedia.net) |
07:44.51 | oriontkn | yeah, that was the impression that I got from looking at the list archives, but some of the posts were older so I thought maybe it had come a ways since then... I have a large dialogic investment |
07:45.27 | cman | sip is used to call sip phones only? |
07:45.29 | oriontkn | although the digium t1 cards look kind of attractive |
07:45.33 | cman | sjphone i mean |
07:45.50 | *** join/#asterisk Marlow (~marlow@3ffe:200:1:155:0:0:0:2) |
07:46.17 | *** join/#asterisk cocoy (strange123@ipdial-172-68.tri-isys.com) |
07:46.47 | cocoy | help on configuring sjphone? |
07:47.07 | cman | click the options tab |
07:47.33 | *** join/#asterisk Marlow (~marlow@ragnarok.marlow.dk) |
07:47.36 | cocoy | then |
07:48.17 | cocoy | cman |
07:48.37 | cman | sip |
07:48.53 | cocoy | then |
07:48.57 | cman | tick use local outbound |
07:49.05 | cocoy | ok |
07:49.23 | cocoy | then |
07:49.48 | cocoy | UDP or TCP? |
07:49.58 | cman | udp |
07:50.22 | cocoy | then after? |
07:50.27 | cman | i don't know but this phone seems to dial ip addresses only |
07:50.32 | cman | click ok |
07:50.57 | cman | so if u have another sip phone .. dial the ip of that phone.. it should ring |
07:51.12 | cman | did it register with *? |
07:51.28 | cocoy | error says "invalid caller ID" |
07:51.42 | cocoy | what do i need on callerid |
07:51.48 | cman | in caller id pu like tis |
07:51.48 | cman | sip:12345@192.168.0.5 |
07:51.57 | cman | sip:username@ip |
07:52.01 | cocoy | ok |
07:52.04 | cman | sip: user |
07:52.38 | cman | registered? |
07:52.42 | cocoy | yes |
07:52.46 | cocoy | what number can i dial? |
07:53.00 | cman | give ip... |
07:53.22 | cman | suh as 192.168.0.11 and Dial |
07:53.23 | cocoy | ip of what? |
07:53.30 | cocoy | ah ok |
07:53.32 | cman | sip of another pone |
07:54.04 | cocoy | how abt dialing an extension? like a voice mail |
07:54.33 | cman | don't kno... figure that out.. read the docs.. i tried but ouln't dial other no.. like loal nos.. |
07:54.36 | cman | localnos. |
07:54.39 | cman | 66 |
07:54.39 | cman | 77 |
07:54.44 | cman | etc |
07:55.06 | cman | it just seems to dial ips |
07:56.21 | cocoy | ok |
07:56.33 | espenz | could anyone try to call "217.118.59.34" with a voip client? |
07:56.47 | Marlow | sip or h.323 ? |
07:56.48 | cman | u can also dial as sip:66@192.168.0.11 |
07:57.00 | cman | tis is useless |
07:57.12 | espenz | Marlow: try both |
07:57.13 | cman | x-lite is better hat sjphone |
07:57.13 | Marlow | cman : seems so .. |
07:57.25 | Marlow | espenz : eheh .. did you set both up or what ? |
07:57.34 | espenz | no, i just dont know what it is |
07:57.35 | espenz | :P |
07:57.44 | Marlow | espenz : what application ? |
07:57.57 | cman | i am not able to get caller id.. |
07:57.58 | cocoy | ok, i have xlite wait |
07:58.05 | cman | wha the heck! |
07:58.21 | espenz | Marlow: asterisk |
07:58.21 | Marlow | cocoy : RTFM |
07:58.42 | Marlow | espenz : and you configured it where ? sip.conf ? |
07:59.03 | cocoy | what do i need to fill up on x-lite? |
07:59.10 | cocoy | ah ok! |
07:59.17 | cocoy | i got u.. |
07:59.34 | Marlow | espenz : definatly not H.323 ... |
07:59.38 | cman | got who? |
07:59.40 | Marlow | espenz : no phone running .. |
08:00.18 | espenz | Marlow: try now, sure? |
08:00.58 | espenz | do you have one up, that i can test? |
08:03.44 | discordia | morning gyus |
08:04.42 | Marlow | discordia : morning .. |
08:05.01 | espenz | Marlow? |
08:05.11 | discordia | out of the bed ... in front of coffee ... |
08:05.18 | Marlow | espenz : try dial 612@fwd.pulver.com |
08:05.26 | Marlow | espenz : that's a test-line .. |
08:05.31 | Marlow | espenz : sip, though .. |
08:05.43 | Marlow | discordia : hopefully real coffee .. |
08:06.03 | Marlow | discordia : the stuff they serve here can only be used to clean furniture with .. |
08:06.18 | discordia | :) |
08:06.23 | discordia | no chance |
08:06.41 | discordia | guatemalean it is ... really great! |
08:06.50 | Marlow | discordia : it's allready that bad, that i've thought of taking my own coffeemachine to work with me .. |
08:06.57 | discordia | :) |
08:07.09 | discordia | a man has to do what a man has to do ... |
08:07.09 | Marlow | discordia : would solve the problem .. |
08:07.18 | Marlow | discordia : yep .. |
08:07.26 | discordia | and would be cheaper with the time |
08:07.46 | Marlow | discordia : nah .. coffee is free here ... or .. no .. it's not coffee .. |
08:07.54 | Marlow | discordia : but what they call coffee .. |
08:07.59 | discordia | lol |
08:08.09 | *** join/#asterisk JerJer[ghost] (~NunYoBizN@pppoe-66-112-51-206-rb2.grp.centurytel.net) |
08:08.18 | Marlow | JerJer[ghost]: morn .. |
08:11.36 | JerJer | yep, its certianly morning |
08:12.22 | espenz | Marlow: it wont answer |
08:12.31 | cman | anyone from xlite? |
08:13.47 | *** join/#asterisk _mwoodj- (~a@user-24-214-189-40.knology.net) |
08:16.44 | voidptr | morning |
08:16.53 | espenz | Marlow |
08:16.59 | espenz | take phone? |
08:16.59 | miller7- | morn voidptr |
08:17.01 | espenz | ;p |
08:17.06 | espenz | lol |
08:17.28 | cman | i can't get caller id thru * for incoming calls.. any one> |
08:17.37 | espenz | they have been carried away with monkeys |
08:20.21 | espenz | any voip client for windows to recomend? |
08:21.29 | JerJer | espenz: nope |
08:21.43 | JerJer | cman: you set the caller*id on the source channel or via the exen line |
08:22.48 | discordia | espenz: if you have one please tell me, im searching for my own for good clients ... |
08:23.34 | discordia | i think on linux side it'll be * as client, that is possible right? with register in sip.conf im on the way or not ?? |
08:30.22 | blitzrage | espenz: x-lite |
08:33.10 | espenz | it wont work |
08:33.33 | *** join/#asterisk bwz (~w_w_zhang@66-215-24-240.mpk-eres.charterpipeline.net) |
08:34.25 | bwz | anyone know how to config extension.conf to make sip-->zaptel-->outbound to PSTN? |
08:35.10 | *** join/#asterisk Levch (~Levch@217.116.160.6) |
08:35.16 | blitzrage | espenz: what do you mean? |
08:35.25 | espenz | blitzrage: i wont ring? |
08:35.41 | blitzrage | ummmm... which way, what are you calling, etc..? |
08:36.04 | espenz | im calling the ip 192.246.69.223? |
08:36.34 | Levch | How does IAX2 trunk work?! |
08:40.53 | discordia | bkw_: sry, not familiar with zaptel devices ... i can tell you how to get sip-->outbound to PSTN |
08:49.22 | *** join/#asterisk Exomorph_ (Greg@134-9.bvcompuworks.com) |
08:54.25 | blitzrage | I'm trying to create a script which will log me into a remote CVS tree. If I do a cvs login, it asks for a password. How can I either: pass the password for the prompt, or: pass the password along with the cvs login command? |
08:56.12 | knight- | anyone use SmoothWall? |
08:57.33 | discordia | nope |
08:58.17 | *** join/#asterisk olivier (~olivier@APuteaux-105-2-1-45.w217-128.abo.wanadoo.fr) |
09:07.08 | cman | --> |
09:16.11 | discordia | aaarrrrrrgggggl ---> |
09:16.13 | discordia | NOTICE[10251]: File app_capiECT.c, Line 125 (capiECT_exec): call was answered |
09:16.13 | discordia | <PROTECTED> |
09:16.13 | discordia | Segmentation fault |
09:16.58 | Inv_Arp | blitzrage: expect |
09:18.02 | blitzrage | huh? |
09:19.00 | Inv_Arp | blitzrage: never used it but the "expect" command can be used to pass passowrds etc... |
09:19.12 | blitzrage | oh right.. thanks |
09:19.14 | RoyK | discordia: start with -g and run a backtrace on the core file |
09:19.20 | blitzrage | hehe.. I wasnt's ure what was going on there for a second :) |
09:20.54 | blitzrage | hrm... still can't seem to get it |
09:22.51 | *** join/#asterisk Bonbon (~bonbon@62.3.220.66) |
09:23.07 | *** join/#asterisk sobol__ (~sobol@router-1.szczecin.tpnet.pl) |
09:23.18 | discordia | RoyK: i will ... |
09:23.21 | Bonbon | someone told me about a kernel utility that would compile my kernel and everything required in one step. Does anyone know what I'm talking about? |
09:24.35 | blitzrage | Bonbon: if you create a bash script, maybe |
09:24.53 | voidptr | quick help: modversions.h ... when its missing in usr/include/linux... what is the problem? (missing kernel headers?) |
09:25.05 | Inv_Arp | Bonbon: its called ik |
09:25.16 | Inv_Arp | freshmeat.net |
09:25.33 | voidptr | i would guess zaptel needs module headers from the running kernel |
09:25.37 | voidptr | and not from libc |
09:25.43 | voidptr | but could be wrong |
09:25.52 | voidptr | on the other hand, its usespace |
09:28.14 | Inv_Arp | voidptr: what distor? |
09:28.19 | Inv_Arp | err distro |
09:29.13 | *** join/#asterisk tristan2 (~tristan@213.239.44.133) |
09:29.17 | voidptr | gentoo |
09:29.33 | Bonbon | thanks Inv |
09:29.41 | RoyK | grr. security.debian.org is down :( |
09:29.43 | Bonbon | who uses the jitterbuffer in iax.conf? |
09:30.34 | Inv_Arp | voidptr: seems like ya need full kernel src for yer distro |
09:30.36 | *** join/#asterisk fyman (~fyman@CPE-138-130-18-16.nsw.bigpond.net.au) |
09:31.02 | voidptr | oah, thats so crap |
09:31.18 | af_ | moin all |
09:31.18 | many | royk, you should read debian-user. |
09:32.22 | many | (and yes, that warning is overdrawn, however it contains true bits) |
09:34.18 | RoyK | oh well |
09:34.42 | blitzrage | <PROTECTED> |
09:34.48 | *** join/#asterisk Muckl (johannes@pD954CD6F.dip.t-dialin.net) |
09:35.23 | RoyK | http://www.bangedup.com/Current/MikeysOntherun35.jpg |
09:36.39 | *** join/#asterisk Marlow (~marlow@ragnarok.marlow.dk) |
09:37.12 | Marlow | arghh .. |
09:38.20 | *** join/#asterisk mbranca (~matteo@213.140.14.155) |
09:38.29 | mbranca | morning all |
09:38.51 | [Sim] | hmm I'm having an odd issue when I try to link an inbound IAX call via IAX to DIAX or iaxComm |
09:39.00 | blitzrage | mbranca: you do any bash scripting? |
09:39.00 | [Sim] | seem to be havind trouble agreeing on codecs |
09:39.24 | mbranca | mbranca, very little. nothing fancy |
09:39.34 | [Sim] | anyone else seen this ? |
09:39.41 | mbranca | ehm, I meant blitzrage, :) |
09:39.43 | blitzrage | hehe |
09:39.56 | blitzrage | damn... I'm trying to login to CVS with a password from within the script |
09:39.57 | voidptr | morn mbranca, sim |
09:40.30 | mbranca | morning voidptr |
09:42.16 | [Sim] | morning :) |
09:42.48 | af_ | ciao mbranca |
09:43.02 | mbranca | ciao af_ |
09:43.46 | mbranca | blitzrage, automagically? |
09:43.52 | blitzrage | yah |
09:44.04 | *** join/#asterisk Unmanaged (~unmanaged@lebanon-24-159-24-23.midtn.chartertn.net) |
09:44.31 | blitzrage | if I do a cvs login, then it asks for a password. I want to pass it automatically from within the script |
09:44.45 | mbranca | blitzrage, just use the .cvspass file in the home dir of the user you're running the script as |
09:45.01 | mbranca | [root@astro root]# cat .cvspass |
09:45.14 | mbranca | <PROTECTED> |
09:45.17 | mbranca | :) |
09:45.44 | blitzrage | Ay=0=h<Z << is this part the password? |
09:45.53 | mbranca | is the enc password |
09:46.00 | mbranca | for digium, is anoncvs |
09:46.01 | mbranca | :) |
09:46.22 | mbranca | ~seen wasim |
09:46.24 | | wasim is currently on #asterisk. Has said a total of 92 messages. Is idling for 17h 26m 58s |
09:47.14 | *** join/#asterisk blitzrage (~blitzrage@dsl-145.sarnia.xcelco.on.ca) |
09:48.37 | mbranca | blitzrage, look at that: |
09:48.50 | mbranca | [root@astro root]# cat update.sh |
09:48.50 | mbranca | #!/bin/sh |
09:48.50 | mbranca | TMPDIR="/tmp" |
09:48.50 | mbranca | CVS="/usr/bin/cvs" |
09:48.50 | mbranca | export CVS_RSH=ssh |
09:48.50 | mbranca | DATA=`/bin/date` |
09:48.54 | mbranca | DATA2=`/bin/date +"%a_%b_%d_%Y"` |
09:48.56 | mbranca | <PROTECTED> |
09:48.58 | mbranca | cd $TMPDIR |
09:49.00 | mbranca | mkdir cvs-asterisk |
09:49.02 | mbranca | cd cvs-asterisk |
09:49.04 | mbranca | $CVS -d:pserver:anoncvs:anoncvs@cvs.digium.com:/usr/cvsroot co -P zaptel zapata libpri asterisk gastman asterisk-addons |
09:49.07 | mbranca | $CVS -d:pserver:anoncvs:anoncvs@cvs.digium.com:/usr/cvsroot co -P nbs libiax gnophone |
09:49.09 | mbranca | $CVS -d:pserver:anoncvs:anoncvs@cvs.digium.com:/usr/cvsroot co -P g723.1 g723.1b$CVS -d:pserver:anoncvs:anoncvs@cvs.digium.com:/usr/cvsroot co -P astconfig libiax2 ztphoned libr2 |
09:49.15 | mbranca | this works with the .cvspass file |
09:49.19 | mbranca | all automaagically |
09:49.22 | mbranca | -a |
09:50.10 | blitzrage | wicked! I will look at that in the morning. |
09:50.15 | blitzrage | now I must get some food, and some sleep! |
09:50.16 | blitzrage | night all |
09:56.09 | *** join/#asterisk Nix (~Nix@195.174.60.197) |
10:06.59 | *** join/#asterisk Insy (~ask@e146244.upc-e.chello.nl) |
10:07.12 | Insy | ey guys! |
10:08.02 | Insy | question: Is it possible to use SIP with a SSH tunnel? |
10:08.30 | l-fy_ | Insy > why not? |
10:08.32 | cman | where can i find the country codes for zaptel... as us, uk,jp, etc |
10:08.39 | voidptr | no |
10:08.43 | levon | morning every1 |
10:08.50 | mbranca | ciao levon |
10:08.51 | Nix | gunaydin levon |
10:08.53 | discordia | morning levon |
10:09.11 | Insy | l-fy_: can ya give me a hint cause can't seem to get it to work! |
10:09.14 | RoyK | morning |
10:09.24 | levon | ciao mbranca |
10:09.25 | levon | hi discordia |
10:09.30 | levon | merhaba nix, canim, naber.... |
10:09.35 | levon | god morgen RoyK |
10:09.45 | levon | l-fy_, diana, my secret love. good morning to you, too ;) |
10:09.54 | discordia | oh this multiculture |
10:09.58 | discordia | i love it ... |
10:10.00 | levon | totally |
10:10.05 | Nix | iyilik kardes. sen? |
10:10.08 | l-fy_ | hello levon |
10:10.20 | levon | iyilik, canim. |
10:10.23 | l-fy_ | tunnel everything |
10:10.38 | Nix | optum, tatlim :-) |
10:10.39 | Insy | l-fy_: with SSH????? |
10:10.39 | levon | Nix, sen glLoadEntity`yi taniyormusun? |
10:10.53 | *** join/#asterisk tristan2 (~tristan@213.239.44.133) |
10:10.53 | Insy | l-fy_: I'm using putty |
10:11.14 | levon | sxpert_work, foood :( |
10:11.16 | Insy | l-fy_: can specify the ports to forward. |
10:11.21 | Nix | hayir levon. malesef, tanimiyorum :-( |
10:11.39 | RoyK | levon: god formiddag |
10:11.47 | cman | anyone from zaptel |
10:11.55 | sxpert_work | levon, I had to get the crappy VCD, because the XSVCD doesn't come... |
10:11.55 | RoyK | from zaptel??? |
10:12.05 | RoyK | cman: I'm from oslo... don't know where zaptel is |
10:12.07 | levon | "from" zaptel` ;) |
10:12.21 | levon | sxpert_work, sh****t.... :( |
10:12.22 | cman | zaptel.. digium...whatever |
10:12.33 | *** join/#asterisk cypromis (~michael@lisa.halo2.pl) |
10:12.37 | cypromis | aloha |
10:12.37 | cman | i need to find country codes such as us, jp, uk |
10:12.43 | levon | Nix, glLE ankara`dan geliyor, burada bahsan bisimle konisiyor. tanistiricam sizi ;) |
10:12.46 | cman | i am having trouble with caller id |
10:12.48 | cypromis | country codes ? |
10:12.55 | cman | ye |
10:12.56 | levon | good morning cypromis |
10:12.59 | cypromis | cman: www.numberplan.org |
10:13.30 | RoyK | Nix: how's istanbul today? quiet again? what did the politicians say? |
10:13.47 | cman | i need codes that are used in zaptel.conf files |
10:13.49 | Nix | levon glLE?? |
10:14.11 | Nix | RoyK: well I had trouble getting to my office as I had to divert around the closed road in front of HSBC |
10:14.50 | mbranca | a big bank |
10:14.55 | levon | Nix: glLE= glLoadEntity (short form) |
10:14.56 | levon | ;) |
10:15.22 | levon | RoyK, the bank that got bombed. |
10:16.13 | Nix | RoyK: HSBC is the second biggest bank in the world after Citibank |
10:16.36 | Nix | the branch that got bombed is the one that I was about to use 30min later |
10:17.22 | Nix | levon: I am not sure I understood your turkish.. glLE will come from ankara and you will introduce it to me?? |
10:19.06 | levon | Nix: I'm sorry, my turkish is *garbage*. I was trying to say that he is from ankara and is in here frequently too. I'd like to intodruce him to you |
10:19.15 | levon | We can build a turkish-*-force, in here ;) |
10:19.39 | Nix | ok. sounds good |
10:19.42 | *** join/#asterisk kapejod (~kapejod@pD9E837BF.dip.t-dialin.net) |
10:19.49 | levon | meep meep, mr. kapejod |
10:19.50 | mbranca | morning kapejod |
10:19.58 | discordia | morning kapejod |
10:20.03 | discordia | hehe |
10:20.04 | kapejod | morn |
10:20.10 | Nix | that makes me feel beter, I showed your sentence to my wife and she couldn't understand it properly either :-) |
10:20.11 | cypromis | moin kape |
10:20.14 | levon | kapejod, can you help me installing chan_capi? |
10:20.16 | cypromis | 13.30 ostbahnhof |
10:20.18 | levon | kapejod, can you help me installing chan_capi? |
10:20.27 | mbranca | 50⬠|
10:20.29 | levon | kapejod, can you help me installing chan_capi? |
10:20.40 | mbranca | 100⬠|
10:20.43 | levon | rotfl |
10:20.46 | kapejod | cypromis: passt gut |
10:20.50 | cypromis | ok |
10:20.55 | kapejod | levon: what is capi?? |
10:21.08 | cypromis | ich komm entweder aus warszawa oder aus poznan nur mit nem notebook und nem flugticket bewaffnet |
10:21.09 | cypromis | lol |
10:21.09 | kapejod | cypromis: let's met at friedrichstrasse then |
10:21.10 | levon | ~tell kapejod about capi |
10:21.15 | cypromis | ok |
10:21.18 | levon | pff |
10:21.46 | levon | bÀh, jetzt kann ich den nicht mal Àrgern... |
10:22.05 | kapejod | cypromis: so gegen 13:45 draussen vor der StäV, das ist auf der nordseite vom wasser |
10:22.17 | cypromis | ok |
10:22.22 | *** join/#asterisk detten2 (~john@213.219.141.57) |
10:22.22 | cypromis | ich werds scho finden |
10:22.25 | cypromis | sonst ruf ich dich an |
10:22.26 | kapejod | cypromis: StäV == Ständige Vertretung |
10:22.47 | kapejod | k |
10:24.26 | sobol__ | cypromis: czesc |
10:24.33 | levon | czesc sobol__ |
10:24.45 | levon | kruliczku ;) |
10:24.53 | kapejod | w00t w00t , my master server is up again (now with a C3) :) |
10:25.02 | levon | kapejod, no, you really did it? |
10:25.21 | cypromis | hmmm |
10:25.27 | cypromis | should I start in .pl, .ru ? |
10:25.40 | kapejod | levon: yes, that was fun ... because the itx board is so small the pci riser and the case wouldnt really fit |
10:25.44 | Marlow | RoyK : de har problemer, hvad ? |
10:25.50 | sxpert_work | levon, ok, it's up ;) |
10:26.08 | RoyK | Marlow: tyskere... |
10:26.17 | kapejod | levon: and it's software raid1 with 2.4.22-preempt :) |
10:26.27 | levon | kapejod, you're evil ;) |
10:26.35 | Marlow | RoyK: ehehe ... og det der ligner .. |
10:26.39 | levon | sxpert_work, yeaaah! ;) |
10:26.43 | mbranca | I have a diva server on a mini-itx board.... I had to buy lowprofile ram to make the 4.bri fit in the case |
10:27.43 | kapejod | vendor_id : CentaurHauls |
10:27.43 | kapejod | cpu family : 6 |
10:27.45 | kapejod | model : 9 |
10:27.45 | kapejod | model name : VIA Nehemiah |
10:27.45 | kapejod | stepping : 1 |
10:27.45 | kapejod | cpu MHz : 999.544 |
10:27.47 | kapejod | :) |
10:27.53 | levon | kapejod, show me da mips |
10:28.30 | kapejod | flags : fpu de tsc msr mtrr pge cmov mmx fxsr sse |
10:28.32 | kapejod | bogomips : 1992.2 |
10:29.13 | levon | wow |
10:29.23 | voidptr | well |
10:29.27 | voidptr | it are bogus mips |
10:29.27 | levon | exactly as much mips as the Athlon 1Gig... |
10:29.28 | voidptr | so |
10:29.28 | cypromis | bogus mips |
10:29.28 | cypromis | :) |
10:29.29 | voidptr | :) |
10:29.30 | levon | ;) |
10:29.47 | af_ | kapejod: which cpu has your mini-itx? |
10:30.15 | voidptr | woah |
10:30.19 | voidptr | totally gey |
10:30.26 | kapejod | af_: C3 1ghz |
10:30.40 | af_ | C3? what is that? |
10:30.49 | af_ | it's a VIA stuff? |
10:30.57 | kapejod | via c3 |
10:31.00 | af_ | I see |
10:31.12 | af_ | I bought a biostar S200, with a celeron |
10:31.25 | RoyK | it's a sloow el cheapo low power cpu |
10:35.56 | many | err |
10:36.08 | many | does asterisk support regexes like [1-9] in the dialplan? |
10:36.14 | mbranca | many, yep |
10:36.23 | voidptr | pom pom |
10:36.26 | *** join/#asterisk Levch (~Levch@217.116.160.6) |
10:36.33 | many | mh. |
10:36.52 | af_ | pum pum |
10:36.58 | mbranca | exten => _[0-2]XXX.1,Dial(blah....) |
10:37.00 | Levch | Can I use different codecs to the same server with different extensions? |
10:37.22 | mbranca | matches aany number starting with 0,1,2 , with at least 4 digits |
10:37.24 | many | yea, didnt know. |
10:37.30 | mbranca | but also 5,6,7.... |
10:38.32 | *** join/#asterisk nickkknight (~nick@217.206.219.198) |
10:39.07 | nickkknight | kapejod hello |
10:40.58 | *** join/#asterisk Tekati (~captain@cpe-66-75-211-60.bak.rr.com) |
10:41.00 | JerJer | Levch: use a different user or peer |
10:41.08 | *** join/#asterisk cypromis (~michael@lisa.halo2.pl) |
10:43.34 | kapejod | hi nickkknight |
10:44.06 | RoyK | kapejod: do you have any nice music for moh? |
10:44.18 | jrollyson | JerJer: I can't seem to find pricing on your site. |
10:44.37 | nickkknight | kapejod - problems with music on hold - any experience? |
10:47.19 | kapejod | nickkknight: you might need zaptel timing for MOH to work correctly |
10:47.33 | *** join/#asterisk Tili (~Tili@202.133.67.121) |
10:49.23 | *** join/#asterisk lele (~fsck@rivendell.windmill.it) |
10:49.42 | levon | ehila lele |
10:49.48 | mbranca | ciao lele |
10:50.35 | mbranca | -s |
10:50.37 | nickkknight | kapejod how do I find out about zaptel timing? |
10:51.02 | lele | morning levon, mbranca |
10:53.59 | lele | has anybody ever had the need of a queue with priorities? |
10:54.07 | kapejod | nickkknight: if you have usb-uhci try ztdummy (in the zaptel dir), else try zaprtc (on my page) |
10:57.30 | JerJer | jrollyson; because there is no such information on there |
10:58.07 | levon | ~tell nickkknight about zaptel-timing |
10:58.11 | levon | ~tell nickkknight about zaprtc |
10:59.00 | levon | updated ;) |
10:59.08 | voidptr | encryption, vpn, blablabla |
10:59.17 | JerJer | ~tell levon about the mystories of the universe |
10:59.24 | levon | lol |
10:59.38 | JerJer | 01[05:57] <jbot> i dunno what is 'the mystories of the universe'. |
11:00.23 | Ares | hello all |
11:01.01 | JerJer | moo |
11:01.04 | levon | meep meep |
11:01.29 | Ares | I have problem when I want to access to asterisk remote option /usr/sbin/asterisk -vvvvgr ERROR[1024]: File asterisk.c, Line 1346 (main): Unable to connect to remote asterisk |
11:01.45 | JerJer | then asterisk is not running |
11:01.45 | l-fy_ | bwahaha |
11:01.53 | l-fy_ | sipstack from vovida is a big mess |
11:01.56 | levon | lol |
11:02.04 | levon | Area: -r is for remote console |
11:02.18 | levon | Area: if you haven't a running instance, it's pretty much useless ;) |
11:02.19 | Ares | now if I do /usr/sbin/safe_asterisk |
11:02.19 | Ares | bash-2.05a$ Asterisk ended with exit status 1 |
11:02.19 | Ares | Asterisk died with code 1. Aborting. |
11:02.34 | levon | just start it up, without -r |
11:02.38 | levon | like -vvvvvgc |
11:02.39 | JerJer | something is broke |
11:02.42 | Ares | levon: I know |
11:02.42 | JerJer | -vvvgc |
11:02.43 | levon | yupp |
11:03.00 | levon | you know? then do it ;) |
11:03.19 | nickkknight | kapejod that kernel you built me - did you include rtc? |
11:04.32 | knight- | why do you need RTC? |
11:04.49 | levon | ~tell knight- about zaprtc |
11:05.00 | knight- | gotcha thanks |
11:05.07 | levon | ;) |
11:05.12 | knight- | :) |
11:05.19 | nickkknight | in the readme is says (of zaprtc) it says make sure you don't have rtc compiled in |
11:05.32 | Ares | levon : sorry I m bit mess... |
11:06.23 | Ares | When normally I should launch asterisk with safe-asterisk |
11:07.19 | Ares | and then when I want access to access to the cli console I have to run /usr/sbin/asterisk -vvvvgr |
11:07.25 | Ares | are for remote |
11:07.25 | *** join/#asterisk JerJer[ghost] (~NunYoBizN@pppoe-66-112-51-206-rb2.grp.centurytel.net) |
11:07.33 | Ares | are for the remote |
11:07.39 | Ares | isn't it ? |
11:07.43 | detten2 | try asterisk -vvvvvvvvvvvvgc |
11:07.56 | detten2 | and see why safe_asterisk is not working |
11:07.58 | Ares | instead of safe_asterisk |
11:08.13 | Ares | no safe asterisk seems to work |
11:08.26 | Ares | it's when I want to access to the console |
11:08.31 | zoa | asterisk -r |
11:08.33 | cypromis | ~seen miller7 |
11:08.34 | | miller7 <~none@adsl49-static-gw1.access.acn.gr> was last seen on IRC in channel #asterisk, 4d 3h 22m 17s ago, saying: 'good morning all'. |
11:08.47 | cypromis | hmopft |
11:09.05 | voidptr | hum |
11:09.07 | Ares | <PROTECTED> |
11:09.07 | Ares | ERROR[1024]: File asterisk.c, Line 1346 (main): Unable to connect to remote asterisk |
11:09.08 | voidptr | thats borked |
11:09.15 | voidptr | miller was here yesterday |
11:09.22 | cypromis | yeah I know |
11:11.20 | nickkknight | so in the zaptel directory - I can just run a make install? |
11:12.17 | Bonbon | has anyone got a spanish number that they can sell me? |
11:13.12 | voidptr | ok i'm going slightly nuts now |
11:13.57 | Ares | it seem that the pid files and the cli aren't created inside /var/run/asterisk/ |
11:15.24 | Bonbon | no-one has spanish number --> sip service? |
11:15.29 | voidptr | instead of module |
11:15.35 | voidptr | modules are just borked by design |
11:16.27 | Pj_ | yeah! |
11:16.31 | Pj_ | modules are for sissies |
11:16.52 | voidptr | aight! |
11:16.52 | voidptr | :P |
11:16.55 | Pj_ | real men patch their kernels in memory |
11:17.20 | voidptr | :)) |
11:18.02 | Ares | Bonbon: perhaps contact mediafusion if you want to buy |
11:18.38 | l-fy_ | who have ever install vovida stack? |
11:20.03 | kapejod | hey l-fy_ :) |
11:21.23 | *** join/#asterisk kd-93 (~chatzilla@pbd.urtc.ru) |
11:21.44 | kd-93 | hello people |
11:21.51 | *** join/#asterisk lichen (lichen@58.136.8.67.cfl.rr.com) |
11:21.58 | l-fy_ | hi kapejod |
11:22.35 | kd-93 | does * support outgoing h.323 channels, and what is the syntax? "h323/user@peer" or something else? |
11:23.41 | Bonbon | does anyone know how to turn on echo canceller training mode? I think that this was implemented recently. |
11:24.20 | zigman | echo "1" > /dev/hda1 |
11:24.22 | zigman | :P |
11:26.44 | JerJer | Bonbon: see zapata.conf.sample |
11:26.46 | zigman | for wha ? |
11:26.48 | zigman | ;) |
11:26.50 | Pj_ | dd if=/dev/echo of=/dev/phone1 count=much |
11:27.04 | zigman | no dev/null |
11:27.15 | zigman | he doesn't want the echo :P |
11:27.23 | Pj_ | For training :P |
11:27.27 | zigman | lol |
11:27.55 | many | (debian users: http://cert.uni-stuttgart.de/files/fw/debian-security-20031121.txt) |
11:28.07 | mbranca | Dial(/dev/null,inf) |
11:29.47 | *** part/#asterisk Nix (~Nix@195.174.60.197) |
11:35.26 | *** join/#asterisk kn0rki (~kn0rki@213.168.83.216) |
11:35.30 | discordia | kapejod: wanna have my newest update to app_capiECT? it should be in the chan_capi package cause of bugfixes + features :) |
11:36.53 | Bonbon | Area: what's the url? |
11:38.26 | kapejod | discordia: just mail it :) |
11:38.34 | discordia | k |
11:40.33 | Bonbon | Ares: what is the url for that? |
11:40.49 | *** join/#asterisk cfo (~cfo@194.19.190.217) |
11:41.07 | kapejod | wtf is an ISDM card? ;) |
11:42.05 | mbranca | integrated services digital moooo |
11:42.18 | levon | ;) |
11:42.30 | mbranca | a new way to speak to the cows |
11:42.31 | kapejod | integrates services digital mettwurst ;) |
11:42.50 | Pj_ | a typo. |
11:42.58 | Pj_ | :D |
11:44.11 | discordia | hmmmmmm mettwurst ;) |
11:44.53 | discordia | lol |
11:45.20 | *** join/#asterisk pino (foobar@host115-28.pool21345.interbusiness.it) |
11:45.20 | levon | ~time |
11:45.23 | | somebody said time was 5:21 |
11:47.57 | Bonbon | need a spanish local number to sip. |
11:49.56 | levon | m00, ]data[ |
11:52.46 | ]data[ | lo |
11:54.32 | JerJer | howdie |
11:55.26 | cypromis | unig *!*mspain@hiostu.aim.hsbr |
11:55.32 | RoyK | grr. when using nagios to check for HTTP on grandstream phones, HTTP dies after a couple of days............ |
11:55.40 | cypromis | bitchx suxx |
11:55.52 | voidptr | unig voidptr |
11:55.53 | voidptr | :) |
11:56.00 | voidptr | tog flood_p |
11:56.08 | cypromis | try that on a console |
11:56.34 | RoyK | cypromis: use telnet |
11:56.43 | cypromis | royk ???? |
11:57.21 | cypromis | ok found the unignore switch |
11:57.22 | cypromis | lol |
11:57.26 | RoyK | telnet irc.freenode.net 6667 |
11:57.53 | cypromis | RoyK: that suxx much more than bitchx |
11:57.56 | RoyK | :) |
11:58.04 | RoyK | then stop complaining :) |
11:59.18 | *** join/#asterisk stipe (~stipe@host107-64.pool80180.interbusiness.it) |
12:00.07 | JerJer | ok the sun is offically rising... us vampires must head to our coffins |
12:01.57 | *** join/#asterisk stipe81 (~stipe@host107-64.pool80180.interbusiness.it) |
12:03.12 | Pj_ | When I receive a call there's 1-2 seconds with a ring and then it's a different ring and the phones actually starts to ring.. is it normal ? how do I reduce / cancel it ? |
12:04.11 | Pj_ | (on a X100P) |
12:10.46 | mbranca | I think that's normal, since * must wait for cid on the x100p (the cid is sent between first and 2nd ring) |
12:10.54 | mbranca | same behaviour here |
12:11.57 | mbranca | ok. going home. see ya later |
12:12.00 | *** join/#asterisk dnc (~duncan@213.244.224.118) |
12:12.34 | cypromis | hi duncan :) |
12:12.43 | kd-93 | does anyone have any experisnce with * and h323 ? |
12:12.49 | cypromis | yep |
12:13.02 | cypromis | it's somewhere between ok and a major pain in the butt |
12:13.05 | cypromis | :) |
12:14.15 | kd-93 | cypromis: :) |
12:15.04 | Bonbon | cypromis: do you know where I can a spanish sip number? |
12:16.35 | cypromis | Bonbon: not yet |
12:19.22 | cman | hey what was the disa format?? |
12:22.44 | UnixDawg | ext,1,DISA,password|where to go |
12:23.07 | UnixDawg | cli > show application disa |
12:23.38 | UnixDawg | use it only when really nessesary |
12:25.24 | cman | guys help me out with this one.. i spent my whole day figuring tis out |
12:25.25 | cman | http://lists.digium.com/pipermail/asterisk-users/2003-November/028027.html |
12:26.57 | rajo | any te410p or pri experts around? |
12:27.33 | cypromis | rajo: what's your problem with them >? |
12:27.45 | Pj_ | cman: I guess the fact that you're in nepal is an explanation enough |
12:28.10 | Pj_ | Asterisk may not be able to handle your country's caller id |
12:28.35 | rajo | cypromis: te410p, 1 span to the telecom (Deutsche telekom), 1 span to an ascend box. calling out from * to the telecom is working. |
12:29.00 | rajo | cypromis: calling in from telecom to ascend (using bridging) via analog modem is working, too |
12:29.13 | rajo | cypromis: calling in via an isdn modem doesn't work (both data calls) |
12:29.18 | rajo | are there any issues? |
12:30.23 | cypromis | no idea |
12:32.52 | *** join/#asterisk sjoeperd (~sjoerd@213-84-218-42.adsl.xs4all.nl) |
12:38.45 | *** join/#asterisk zw1 (~chris@ewa-denver.com) |
12:42.32 | *** join/#asterisk gadams666 (~wileyuser@63.111.7.137) |
12:46.04 | *** join/#asterisk clive- (~pirch@rndf-ip-nas-2-p53.telkom-ipnet.co.za) |
12:49.31 | rajo | is there someone (not from germany) who could give me a short call via landline? (just ringing for some seconds) |
12:50.23 | *** join/#asterisk cfu (~kapejod@pD9E8193C.dip.t-dialin.net) |
12:53.02 | magg | sure |
12:53.10 | magg | rajo : what's your number? |
12:53.34 | rajo | Magg +49 681 379680 |
12:54.06 | magg | calling now |
12:54.12 | magg | hidden callerid.. |
12:54.15 | magg | ringing.. |
12:54.21 | magg | ok? |
12:54.23 | rajo | ok |
12:54.23 | rajo | thx |
12:54.28 | magg | no worries |
12:54.31 | rajo | magg: no, not hidden |
12:54.47 | magg | ahh, wrong phone :D |
12:55.05 | magg | *grin* |
12:55.31 | cman | is there anyway to change the pattern of * source file so that i can make it compatible with my countrys caller id????? |
12:55.36 | *** part/#asterisk sjoeperd (~sjoerd@213-84-218-42.adsl.xs4all.nl) |
12:57.38 | *** join/#asterisk torment0r|wk| (~someone@207.168.230.31) |
12:58.08 | torment0r|wk| | lolol@topic |
12:58.08 | UnixDawg | there is a bug * is not cleaning up mpg123 when you stop now |
12:58.20 | UnixDawg | it leaves mpg123 running in the background |
12:59.51 | *** join/#asterisk nitram (nitram@superblob.com) |
13:00.04 | RoyK | I've seen that... |
13:00.13 | RoyK | should've reported that... |
13:00.15 | RoyK | sori |
13:01.23 | torment0r|wk| | i got a cisco 12+ ip phone that uses h 323 for free. has anyone used one of these with asterisk before |
13:01.47 | rajo | does anyone know how to set the outgoing capability (info transfer capability) of a te410p? want to make data calls, but they're sent out as voice |
13:02.30 | *** join/#asterisk ToyMan (~stuq@smtp.dstoys.com) |
13:03.52 | torment0r|wk| | i agree with ya |
13:04.04 | torment0r|wk| | it looks like a pain to setup |
13:04.22 | h3x | rajo: |
13:04.22 | h3x | [Description]: |
13:04.23 | h3x | <PROTECTED> |
13:04.26 | h3x | <PROTECTED> |
13:04.34 | h3x | i think that will do it |
13:07.09 | torment0r|wk| | does anyone know of any examples of setting up sphinx with asterisk? |
13:07.26 | torment0r|wk| | besides the eagi script that comes with the source |
13:07.31 | rajo | h3x: Dial(Zap/g2/$(EXTEN)|160|c)? |
13:08.21 | rajo | h3x: this doesn't work :( |
13:09.48 | h3x | maybe its a documented unfeature |
13:10.04 | rajo | h3x: :) |
13:10.15 | *** join/#asterisk Prutser (Prutser@bitbucket.capcave.com) |
13:10.26 | h3x | or your telco rejects data calls |
13:10.51 | rajo | h3x: no, don't think so as this is the main purpose of this line |
13:10.54 | Prutser | Hi all... |
13:11.02 | Prutser | Anyone using * on DM3 hardware? |
13:11.18 | *** join/#asterisk _GiGi_ (gigi@disc.more.pl) |
13:14.59 | cman | is there a way to telnet private ip? like i have pulblic ip in router... and if i want to see my linux box with ip 192.168.0.4 |
13:15.17 | dutch_ | pat |
13:17.42 | *** join/#asterisk Kevorkian (~levo@ool-18bc8bc9.dyn.optonline.net) |
13:17.58 | *** join/#asterisk coppice_ (~Steve@210.17.194.2) |
13:18.49 | *** join/#asterisk lichen_ (~lichen@vanquish.cohpa.ucf.edu) |
13:20.45 | Kevorkian | Im trying to get my * server to talk to iaxtel .. anyone got a moment to helpo |
13:21.46 | *** join/#asterisk Vco (~Vco@h24-76-11-125.wp.shawcable.net) |
13:23.20 | *** part/#asterisk pino (foobar@host115-28.pool21345.interbusiness.it) |
13:38.48 | *** join/#asterisk Zebble_ (~Zebble@Sherbrooke-HSE-ppp3610369.sympatico.ca) |
13:39.47 | *** join/#asterisk DrJack (~levo@ool-18bc8bc9.dyn.optonline.net) |
13:40.41 | *** join/#asterisk cypromis (~michael@marge.halo2.pl) |
13:41.24 | *** join/#asterisk bjohnson (~bjohnson@jecinc.tor.istop.com) |
13:50.01 | *** join/#asterisk hi (~tauss@210.187.114.238) |
13:54.20 | *** join/#asterisk p0lar (~Miranda@155.101.253.115) |
13:57.52 | doughecka | http://tuxscreen.net/ |
13:58.52 | *** join/#asterisk phiberkut (~phiberkut@tele-free-hotspot.netlinkip.com) |
13:59.53 | *** join/#asterisk EagleRed (~eaglered@a213-22-50-141.netcabo.pt) |
14:01.52 | *** join/#asterisk marrandy (~marrandy@209.216.76.1) |
14:02.41 | marrandy | Hello. Trying to get a manual fax working on ch. 4 fxo (tdm400) but seeing problems. |
14:02.59 | marrandy | exten => 22,1,Dial(Zap/4-1,20,d) ; Ring for 20 seconds, request a low latency call |
14:02.59 | marrandy | exten => 22,2,Hangup |
14:03.35 | marrandy | ignore the -1, that's old and has been changed to Zap/4, |
14:03.55 | marrandy | log is as follows |
14:05.00 | marrandy | exten => 22,1,Dial(Zap/4,20,d) ; Ring for 20 seconds, request a low latency call |
14:05.00 | marrandy | exten => 22,2,Hangup |
14:05.17 | marrandy | anyone see anything wrong with that ? |
14:05.46 | marrandy | exten => fax,1,Goto(22,1) |
14:06.47 | marrandy | How many lines before I flood (for the log) ??? |
14:07.56 | marrandy | anyone here ? |
14:08.44 | marrandy | JerJer ? |
14:09.28 | voidptr | <PROTECTED> |
14:09.54 | marrandy | hello voidptr: |
14:09.59 | p0lar | hello |
14:10.22 | marrandy | ahh.another one. Hello p0lar |
14:10.27 | p0lar | hwy |
14:10.29 | p0lar | hey |
14:10.32 | marrandy | Pretty quiet here |
14:10.41 | p0lar | <- too early to type or think |
14:10.43 | _GiGi_ | ;-) |
14:11.02 | p0lar | I think irc is having probs |
14:11.09 | marrandy | Does that fax extions look O.K. ? |
14:11.13 | p0lar | I only have 20 people in my list |
14:11.21 | marrandy | extions = extension |
14:11.29 | p0lar | <- has never used faxes |
14:11.41 | p0lar | but it looks good. whats wroing? |
14:11.54 | tzanger | ~seen citats |
14:11.57 | | citats is currently on #asterisk. Has said a total of 10 messages. Is idling for 1d 19h 25m 50s |
14:11.57 | marrandy | How many lines before I flood ? |
14:12.13 | p0lar | flood? |
14:12.51 | marrandy | If you put too many lines of text in, the system can flood and either kick me off or kick everyone off - not good |
14:13.05 | tzanger | question for you telco folks -- I have a D50 -> BIX cable, and the telco guy who gave it to me says that the buildout is pair 1 and 2, then skip two pairs, then 3 and 4, then skip 2 again, .. and so forth... why? |
14:13.13 | tzanger | as in , why are they punched down in that manner? |
14:13.23 | marrandy | Let me break the log into 5 lines to be safe |
14:13.58 | marrandy | <PROTECTED> |
14:13.58 | marrandy | <PROTECTED> |
14:13.59 | marrandy | <PROTECTED> |
14:13.59 | marrandy | <PROTECTED> |
14:14.00 | marrandy | <PROTECTED> |
14:14.00 | marrandy | <PROTECTED> |
14:14.21 | marrandy | <PROTECTED> |
14:14.22 | marrandy | <PROTECTED> |
14:14.22 | marrandy | <PROTECTED> |
14:14.22 | marrandy | <PROTECTED> |
14:14.37 | marrandy | WARNING[327700]: File chan_zap.c, Line 1364 (zt_call): Unable to ring phone: Device or resource busy |
14:14.37 | marrandy | <PROTECTED> |
14:14.37 | marrandy | <PROTECTED> |
14:14.37 | marrandy | <PROTECTED> |
14:14.55 | p0lar | ahhh ok lemme look |
14:15.00 | marrandy | As I'm a newbie, that is basically 'Duh' to me |
14:15.32 | p0lar | is zap 1 your fxs port? |
14:15.46 | p0lar | is the fax ready? or could it really be busy? |
14:16.04 | DrJack | does iaxtel still allow 800 service outbound tru them ? |
14:16.22 | marrandy | zap 1 = fxo to the landline, zap 2-5 = fxs (tdm400p card) |
14:16.24 | tzanger | ok marrandy that was a flood |
14:16.43 | tzanger | and on this D50->Bix, is pair 25 the first skipped pair? |
14:17.15 | tzanger | i..e does the order go 1,2,13,14 2,3,15,16 4,5,17,18 and so on? |
14:17.24 | marrandy | p0lar: You just gave me a thought...let me check |
14:18.05 | p0lar | ok |
14:18.13 | p0lar | glad someone had a thought :) |
14:18.29 | *** join/#asterisk jp (~jp@hoochie.digium.com) |
14:19.32 | *** join/#asterisk Kevorkian (~levo@ool-18bc8bc9.dyn.optonline.net) |
14:22.14 | *** join/#asterisk jerkface (~me@67.70.231.218) |
14:22.36 | *** join/#asterisk raoul (~raoul@lmepool1.ugr.be) |
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14:25.00 | *** part/#asterisk p0lar (~Miranda@155.101.253.115) |
14:25.06 | *** join/#asterisk p0lar (~Miranda@155.101.253.115) |
14:25.07 | doughecka | gee, this sure looks like a nice phone: http://tuxscreen.net/ |
14:27.56 | *** join/#asterisk pino (foobar@host115-28.pool21345.interbusiness.it) |
14:28.09 | RoyK | grr |
14:28.25 | RoyK | I can park calls, but I can't pick them up |
14:28.35 | pino | does anyone have a minute? i'd like to test-call and be test-called over iax or anything else to test this phone... |
14:29.00 | raoul | hi all, what are the different possibilities in terms of digium hardware if I want to be able to have several conversations at the same time? |
14:29.48 | pino | raoul: several conversations over a single analog line, or something else? |
14:30.40 | raoul | pino: well, the only requirement is that we have only one phone number, ie several people can call us at the same time... |
14:31.18 | doughecka | dsl |
14:31.19 | doughecka | :) |
14:31.39 | Pj_ | raoul: ask your phone operator I think |
14:31.49 | Pj_ | you get several lines with the same number |
14:32.10 | Pj_ | then you connect them to your server either with a few X100P or (far more better), a T1 line |
14:32.45 | RoyK | can someone help me with parking probs? |
14:33.10 | pino | raoul: or, in the middle, a few ISDN lines in a hunt group (if it is called that way... I still don't know too much telco terminology) |
14:33.48 | raoul | well, my first plan was to have a few pstn lines (for example 3), and I asked to my operator to have the same number for all of them |
14:33.57 | raoul | they told me it was impossible |
14:34.30 | raoul | now, if I buy a T1 or E1 line, is it ISDN PRI? or is it something different? I'm lost with all those things ;) |
14:34.48 | h3x | raoul: where are you |
14:35.07 | pino | what they do here (Italy) is this: each single analog line has a different number, but the calls to the main line are "deflected" to the first non-busy of the other lines. |
14:35.09 | raoul | h3x: in Belgium |
14:35.14 | h3x | oh, heh |
14:35.47 | raoul | pino: mmh, so you have for example 3 analog lines with 3 different FXO cards and asterisk will manage that? |
14:35.48 | pino | E1 is ISDN PRI, but it will be extremely expensive... it is usually convenient when you have at least 20 channels ("lines", so to say.) |
14:35.48 | h3x | well anyway as long as you get trunks or PRI then every incoming call in the trunk group (usually the entire span) |
14:35.56 | h3x | will come in with the dialed number and calling number |
14:35.59 | tholo | If you specify you want a 30B+D PRI, that will be delivered as an E1. |
14:36.03 | h3x | so your system can route accordingly |
14:36.05 | *** join/#asterisk benjk (~benjamin@f8a01-0359.din.or.jp) |
14:36.13 | pino | raoul: yes, Asterisk will if your telco does that, but I'd tell you right away to use ISDN instead of analog lines. |
14:36.29 | h3x | You don't have to use ISDN |
14:36.33 | h3x | to get this functionality |
14:36.39 | h3x | its nice but not required |
14:36.45 | h3x | if you order E&M Wink start with the right features |
14:36.47 | tholo | Some places you can get a PRI with fewer channels actually turned up -- but still delivered on an E1 (in Europe). |
14:36.51 | h3x | (ANI/DNIS delivery) you get the same thing |
14:37.02 | pino | h3x: you don't, but i think it saves a lot of trouble... |
14:37.19 | h3x | well in some parts of the world, PRI is too expensive to use |
14:37.31 | pino | and at least here an ISDN 2B+D is cheaper than 2 POTS :) |
14:37.36 | h3x | for the only real features gained is being able to reject a call without answering it, and setting your caller id for outgoing calls |
14:37.43 | raoul | mmh and ISDN BRI is 2 voice channels only, to be used with the CAPI channel, right? |
14:37.53 | h3x | which can be done with MF tones and FGD |
14:38.10 | pino | raoul: that's the setting I'm using. |
14:38.16 | tholo | Or with the new, upcoming ZapBRI (should be out any day now, check with kapejod). |
14:38.19 | h3x | well im talking about using robbed bit signalling and tones on a E1 instead of PRI ISDN |
14:38.20 | pino | h3x: fgd? |
14:38.24 | h3x | Feature Group D |
14:38.34 | h3x | maybe a north american thing |
14:38.45 | raoul | pino: and I guess you have 2 phone numbers too, but you are able to receive 2 different calls on the same number. Right? |
14:38.54 | pino | h3x: maybe I read something about it, but I'm Not An Expert :) |
14:38.56 | h3x | i think you guys just use MFC R2 MF |
14:39.03 | pino | raoul: yes, it is very well possible. |
14:39.15 | h3x | im sure euroisdn has a better chance of working with asterisk though |
14:39.29 | tholo | h3x: Most of Europe uses Q.931, not MFC R2 for PRI signalling... |
14:39.35 | raoul | so if I want to have 2 phone calls at the same time, I can use 2 dialup lines with number redirection if busy, or 1 isdn BRI line with chan_capi |
14:39.43 | h3x | ok well there you have it |
14:39.43 | h3x | heh |
14:39.52 | raoul | if I want to have more than 2 phone calls at the same time, I have to use ISDN PRI = E1 |
14:40.01 | tholo | Yes. |
14:40.09 | pino | raoul: no, you can just use more than one ISDN BRI ... |
14:40.31 | raoul | pino: with the telco redirecting all lines to the same number on a first-available basis? |
14:40.33 | pino | you will not be able to set the same callerid on all outgoing calls, but you could receive all calls on a single MSN... |
14:40.33 | tholo | kapejod is about to deliver a product that can do 4 BRIs. |
14:40.50 | raoul | I need to check the prices |
14:40.50 | pino | raoul: exactly. |
14:41.12 | raoul | ok nice :) |
14:41.20 | torment0r|wk| | has anyone used a granstream budgetphone? do they work well? |
14:41.21 | raoul | and what is T1 compared to E1? |
14:41.32 | tholo | Might fit nicely in the niche between BRI and PRI. |
14:41.33 | torment0r|wk| | or should i just step up |
14:41.43 | pino | tholo: I think it could also be done with two "cheap" passive cards, but I suspect ZapBRI will soon be far better in terms of features and support :) |
14:41.50 | tholo | T1 is 1.5 Mbit, E1 is 2 Mbit. T1 is typically used in the US, E1 typically used in Europe. |
14:42.02 | tholo | pino: Yeah, I agree. |
14:42.25 | raoul | is T1 a telephonic norm or is there a relation with the T1/T3 internet lines? |
14:42.38 | tholo | There are also many more channel banks (for use with analog phones / phone lines) available with T1 interface than with E1 interface. |
14:43.08 | raoul | ok, but I guess E1 is very expensive |
14:43.17 | tholo | There is a relation between T1 and T3, yes... A T3 has multiple T1's worth of capacity -- the exact number eludes me at the moment. |
14:43.31 | tholo | That varies a lot from place to place. |
14:43.36 | raoul | ok nice |
14:43.50 | pino | and a T1 for telephone use and a T1 for internet lines are the same thing, but different signalling is used ... |
14:43.56 | tholo | Yes. |
14:44.01 | torment0r|wk| | i think it's 23 t1's worth is a t3 |
14:44.10 | raoul | a last question now (sorry), when they say on digium.com that E1 is 32 channels, how many voice channels (ie how many calls) is it possible to do? And what is the difference with the quad-E1? |
14:44.19 | tholo | And a T1 (or E1) can even be shared between the two, with some channels for each purpose. |
14:44.26 | *** join/#asterisk Cornfed (~nospam@24.236.221.138.gha.mi.chartermi.net) |
14:44.27 | raoul | pino: ok |
14:44.38 | tholo | The Quad E1 is simply a card that can connect 4 E1 lines. |
14:44.54 | raoul | ah ok |
14:45.07 | tholo | An E1 has 32 64-kbit channels -- when used with a PRI (ISDN), you get 30 voice channels and one signalling channel (30B + D) |
14:45.31 | tholo | A T1 has 24 channels, 23 voice and one signalling when used as a PRI. |
14:45.54 | *** join/#asterisk l-fy_ (~diana@home-25022.b.astral.ro) |
14:46.23 | raoul | tholo: ok, so if I ask for a E1 for phone here, it is an ISDN PRI with 30 voice channels |
14:46.50 | Mike | isnt a t1 a dedicated line for internet? |
14:46.50 | raoul | ah there is a last thing that worries me, the very last one, I promise :) |
14:47.00 | Mike | E1 the same but european standars? |
14:47.05 | Mike | 2200kbit? |
14:47.49 | kapejod | 2048 kbit |
14:47.50 | pino | Mike: t1 is a dedicated line for whatever you wish; e1 is very similar, but 2048Kbit/second. |
14:47.53 | tholo | raoul: Only if you ask for an ISDN PRI. There are other ways of using an E1. |
14:48.00 | kapejod | 32 channels x 64 kbit |
14:48.16 | tholo | And here is the guy with the (upcoming) OctoBRI. ;) |
14:48.22 | Mike | so you talk about t1s as for internet lines? |
14:48.27 | raoul | if I'm going for the setup of 3 FXO cards with 3 analog lines, in the asterisk config, I guess I will have to configure it to explicitely try each of the cards to find a non-busy line, but with an E1 or ISDN BRI, it would be automatic, right? |
14:48.30 | kapejod | who? |
14:48.36 | raoul | tholo: ok |
14:48.42 | tholo | T1s are usable both for phones and IP -- not limited to either, Mike. |
14:48.47 | *** join/#asterisk anthm (~anthmct@CPE-24-167-193-161.wi.rr.com) |
14:48.54 | tholo | If you get a PRI in the US, that comes on a T1, for instance. |
14:49.04 | *** join/#asterisk swirlnets (~Mike@dsl093-001-038.det1.dsl.speakeasy.net) |
14:49.05 | jensd | E1 32 channels - 1D = 31 but you only gets 30 voice channels - what happens with the last one? |
14:49.15 | pino | raoul: maybe the opposite... that is: |
14:49.15 | Mike | 23phonelines on a t1? |
14:49.18 | kapejod | some timing syncing stuff |
14:49.26 | tholo | Mike: Yes. |
14:49.37 | *** join/#asterisk point (1000@195.161.161.248) |
14:49.39 | Mike | tholo: how much a month for one of those? |
14:49.45 | pino | if you have 3 FXO cards, incoming calls will be directed by the telco; if you have a PRI, you configure * to do what you wish. |
14:49.52 | Kevorkian | jensd its 2 chans for the d .. |
14:49.59 | point | hi there all ... |
14:50.05 | RoyK | grr. can someone look through my config? I can't pick up parked calls anymore, after rewriting it... |
14:50.12 | raoul | pino: yes but I meant for outgoing calls |
14:50.19 | tholo | Rewrite it back to what it was? ;-) |
14:50.35 | Mike | tholo: how much for a t1 of those? |
14:50.39 | RoyK | tholo: neppe. it was a complete mess..... |
14:51.03 | tholo | Mike: The price for a T1 / PRI varies wildly with location and phone company. |
14:51.03 | raoul | pino: because in Dial I have to specify the interface to use |
14:51.25 | tholo | I thought it was 4pm there? |
14:51.30 | RoyK | it is |
14:51.33 | Mike | tholo: and intermedium price would be? |
14:51.58 | tholo | Mike: Anywhere from $150 to $1.5k |
14:52.05 | matt-1control | Is anyone else using NuFone right now and having trouble with it? |
14:52.10 | Mike | 150 is cheap |
14:52.15 | pino | raoul: I think that it is automatic then, but I've never tried to configure a PRI interface on *... |
14:52.16 | tholo | Yes. |
14:52.20 | point | kapejod has own way ... :) |
14:53.01 | raoul | pino: ok thank you :-) |
14:53.04 | Kevorkian | Mike .. 150 would be for a zero mile t .. to your equpment local at the co |
14:53.04 | matt-1control | Also, how does VoicePulse compare in reliability and customer service to NuFone? |
14:53.06 | tholo | And you can probably only get one for $150 if you are in a colo at an ILEC. |
14:53.21 | raoul | pino, tholo: thanks a lot for your help, everything is more clear for me now! |
14:53.39 | pino | raoul: don't mention it:) |
14:53.48 | raoul | :-) |
14:53.51 | tholo | I hope I didn't give out too much misinformation. ;-) |
14:54.22 | point | here is collocation for $25 per unit + incomming traffic |
14:54.23 | raoul | I think I will ask for 3 dialup lines and a redirection of the first of the 3 numbers to the first available one |
14:54.24 | Kevorkian | can anyone give me a good link that explains the systax of extention commands ? |
14:55.12 | pino | Kevorkian: the most up-to-date info is got by "show applications" and "show application <name>" |
14:56.34 | pino | otherwise you might look at jtodd's sample configs: http://www.loligo.com/asterisk/current/ |
14:58.06 | doughecka | ~seen wasim |
14:58.08 | | wasim <~wasim@202.179.137.13> was last seen on IRC in channel #asterisk, 22h 38m 42s ago, saying: 'eww'. |
14:59.18 | *** join/#asterisk bjohnson (~bjohnson@jecinc.tor.istop.com) |
15:10.49 | *** join/#asterisk lucifuge3 (trilluser@pcp05047461pcs.ivylnd01.pa.comcast.net) |
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15:12.38 | tzanger | wow |
15:12.52 | doughecka | wow |
15:13.04 | tzanger | I'm happier... the telco guy was full of shit. :-) I was worried I'd blown out an FXS port on this CB (wired it to a PSTN line) |
15:13.24 | tzanger | you'd ring it, it'd buzz like a bee on ludes, then hang up |
15:14.06 | JerJer | matt-1control: everything is good here |
15:15.02 | *** join/#asterisk Kalandor (~storm@inray.spnet.net) |
15:15.09 | JerJer | tzanger: consider yourself lucky |
15:15.32 | JerJer | if u would have done that with a PhoneHack it would have smoked |
15:15.50 | denon | man .. another unexplained * crash |
15:15.59 | denon | anyone having problems with current? |
15:16.01 | JerJer | gdb |
15:16.13 | denon | im just using safe_asterisk, no special parms |
15:16.17 | *** join/#asterisk jtodd (~jtodd@207.141.153.205) |
15:16.27 | JerJer | denon: run a bt see /tmp |
15:16.29 | Kalandor | Sorry for disturbing, but I am a developer just started to examine the asterisk and I have a question. Anyone willing to help me ? |
15:16.46 | JerJer | Kalandor: it helps if you ask a specific question |
15:17.02 | denon | ooh .. -rw------- 1 root root 41967616 Nov 21 09:03 core.9235 |
15:17.11 | Kalandor | okay I will proceed with it then |
15:17.19 | *** join/#asterisk gage_man (~svoto@host2.216.41.24.conversent.net) |
15:17.41 | Kalandor | Is there a chance of estabilishing a connection between to IPs in LAN without using the phone line, and how is this done |
15:18.23 | JerJer | Kalandor: go learn about VoIP |
15:18.42 | Kalandor | I am doing this now.. this is my first encounter with this subject |
15:19.01 | Kalandor | I mean using the asterisk for that kind of connection |
15:20.11 | denon | per JerJer's privmsg request: |
15:20.11 | denon | #0 ast_queue_frame (chan=0x8172840, fin=0x8121a0c, lock=0) at channel.c:368 |
15:20.11 | denon | #1 0x4069c3d2 in local_write (ast=0x813f328, f=0x8121a0c) at chan_local.c:111 |
15:20.11 | denon | #2 0x080586f7 in ast_write (chan=0x813f328, fr=0x8121a0c) at channel.c:1392 |
15:20.11 | denon | #3 0x4042fbe1 in wait_for_answer (in=0x815d8f0, outgoing=0x813f220, to=0xbd5feb2c, allowredir_in=0xbd5feb30, |
15:20.12 | denon | <PROTECTED> |
15:20.14 | denon | #4 0x40430cbb in dial_exec (chan=0x815d8f0, data=0xbd5ff304) at app_dial.c:635 |
15:20.16 | denon | #5 0x08060bb0 in pbx_exec (c=0x815d8f0, app=0x8107a48, data=0xbd5ff304, newstack=1) at pbx.c:396 |
15:20.33 | lucifuge3 | Kalandor: Yes, it can be done. There are many ways and your question is too vague to answer. Take a look here: http://asterisk.xvoip.com/ and get some basic information on *. It will probably answer all of your questions. |
15:20.36 | JerJer | Kalandor: Asterisk is a hybrid TDM / Packet voice (VoIP) PBX and IVR platform with ACD functionality |
15:20.56 | JerJer | lucifuge3: why not refer him to the offical documenation? |
15:21.05 | JerJer | http://www.digium.com |
15:21.16 | Kalandor | ok ... thanx... I will take a look and then if I have some more specific questions I'll ask |
15:21.17 | lucifuge3 | Because that's the firs bookmark in my list. Why don't YOU refer him to the official docs. |
15:21.31 | JerJer | i did |
15:21.37 | loko_moko | oh the hostility <G> |
15:21.37 | lucifuge3 | (Which are pitifully incomplete as of last check) |
15:21.50 | *** join/#asterisk Vco (~Vco@h24-76-11-125.wp.shawcable.net) |
15:21.52 | JerJer | lucifuge3: then fucking write better or shutup |
15:21.58 | lucifuge3 | Yeah...sorry. Didn't get my entire first cup of coffee down yet. ;) |
15:22.01 | lucifuge3 | lol...knew that was coming. |
15:22.19 | JerJer | Mark and Maritn code |
15:22.26 | JerJer | someone else needs to step up and document |
15:22.37 | *** join/#asterisk segal_home (~segal@cuscon4542.tstt.net.tt) |
15:22.41 | ]data[ | didnt you like er just goto bed JerJer ? |
15:22.59 | JerJer | yep, i'm sleeping now |
15:23.00 | *** join/#asterisk jules (~jules@63.105.150.196) |
15:23.02 | ]data[ | k |
15:23.19 | tholo | This is JerJer's at-sleep behaviour -- you didn't know? |
15:23.23 | *** part/#asterisk Kalandor (~storm@inray.spnet.net) |
15:24.58 | jules | Is there a bot I can bug the crap out of to find out about getting started with IVR? :) |
15:25.12 | Exomorph_ | Morning people |
15:25.24 | JerJer | jules: read the docs (i already posted the url) |
15:26.09 | jules | JerJer, I'm currently on Digium's site grepping the FAQ... which is as far as I got. When you say posted, where did you post it? |
15:26.51 | lucifuge3 | http://asterisk.xvoip.com/ |
15:26.56 | JerJer | i can tell |
15:27.04 | lucifuge3 | http://www.voip-info.org/tiki-index.php?page=Asterisk |
15:27.16 | lucifuge3 | http://www.asterisk.org |
15:27.18 | jules | lucifuge3, JerJer, thanks! Very much appreciated. |
15:27.44 | lucifuge3 | See....I gave him the official one that time too, JerJer. |
15:28.34 | JerJer | lucifuge3: there is a Unofficial Links section at digium.com that has all of those url's your spewing |
15:28.53 | JerJer | why not give out one URL and let them find those links? |
15:29.28 | tholo | That'd be the same page that has the handbook, as well as mailing list information.... |
15:30.26 | jules | Yeah, make the damn newbies work for it lucifuge3. |
15:30.43 | jules | Seriously, a few links like that are a great leg up. |
15:30.53 | lucifuge3 | Sorry. I'm not pissed off enough this morning. |
15:32.34 | doughecka | at least I am that generous |
15:33.07 | *** join/#asterisk cocoy (strange123@203.76.221.55) |
15:33.40 | cocoy | can anyone give me how to create an extension for OH323 ? |
15:34.14 | cocoy | help |
15:35.16 | *** join/#asterisk cypromis (~michael@marge.halo2.pl) |
15:35.34 | JerJer | cocoy: can't help you on that one |
15:35.58 | JerJer | damn your old |
15:36.16 | tholo | Just kids... |
15:40.50 | cypromis | *sigh* |
15:40.51 | cypromis | kids |
15:41.17 | *** join/#asterisk mawdawg (dav@205.124.232.173) |
15:41.29 | *** join/#asterisk Tekati (~captain@cpe-66-75-211-60.bak.rr.com) |
15:42.35 | *** part/#asterisk Cornfed (~nospam@24.236.221.138.gha.mi.chartermi.net) |
15:46.08 | JerJer | ok i think its time to unplug /dev/cerebral/cortex |
15:46.56 | doughecka | hahahaha |
15:47.20 | Tekati | Is AGI in the cvs repository or do you have to download it somewhere else? |
15:49.04 | *** join/#asterisk raulfragoso (~raulfrago@200-204-142-71.dsl.telesp.net.br) |
15:49.21 | *** join/#asterisk os2doc (~os2doc@hoochie.digium.com) |
15:50.58 | raulfragoso | Anyone here using Dialogic cards with Asterisk ? |
15:51.22 | JerJer | raulfragoso: no... use zaptel |
15:52.00 | JerJer | Tekati: read more docs |
15:53.00 | *** join/#asterisk Tddy (bartek@66.178.36.215) |
15:53.17 | JerJer | exten => XXX,1,AGI,foo.agi |
15:53.28 | p0lar69 | anyone seen wasim lately? |
15:53.30 | raulfragoso | I'm in trouble to install Dialogic on RedHat 9. |
15:53.30 | Tddy | Hello everone. I need BIG HELP! Just bought Cisco 7960 and I upgrade it to SIP |
15:53.45 | p0lar69 | ok gor for it |
15:53.47 | p0lar69 | Tddy |
15:53.52 | Tddy | But now I cannot change any configuration because configuration is locked |
15:54.04 | p0lar69 | in the Cisco phone? |
15:54.08 | Tddy | Yes |
15:54.14 | p0lar69 | *##* |
15:54.20 | Tddy | let me see |
15:54.22 | p0lar69 | to unlock |
15:54.23 | bkw_ | no the password is cisco |
15:54.30 | bkw_ | if its 5.0 and above |
15:54.30 | p0lar69 | or *## |
15:54.36 | bkw_ | or |
15:54.38 | p0lar69 | no dude |
15:54.42 | p0lar69 | SCCP code |
15:54.52 | p0lar69 | he does not have SIP on it yet |
15:54.54 | bkw_ | no *## dont work anymore |
15:55.04 | bkw_ | he said its upgraded to sip and its now locked |
15:55.05 | p0lar69 | it does not work on SIP code your right |
15:55.09 | h3x | JerJer: what carrier(s) do you use for inbound toll free? |
15:55.09 | Tddy | i tryed *## i do not work |
15:55.12 | p0lar69 | AHHH ^$%$% |
15:55.14 | Tddy | also *##* |
15:55.14 | p0lar69 | sorry |
15:55.18 | bkw_ | cisco is the password |
15:55.25 | p0lar69 | <- still asleep |
15:55.27 | Tddy | let me check |
15:55.29 | p0lar69 | yes |
15:55.41 | p0lar69 | thx bkw_ |
15:56.07 | Tddy | THANS |
15:56.11 | Tddy | IT IS UNLOCKED |
15:56.20 | Tddy | THANKS SO MUCH!!!!!!!!!!!!!!!!!!!!!!!!! |
15:56.26 | p0lar69 | no |
15:56.34 | tholo | Aw, bwk_, you make it too easy! ;-) |
15:56.35 | p0lar69 | thx to bkw that he was awake |
15:57.11 | *** join/#asterisk zono (~trillian@inray.spnet.net) |
15:57.13 | Tddy | Thank to all |
15:57.19 | denon | bkw_: you know, im surprised ... I reported that crash several minutes ago, and yet I see no patches from you yet .. :) |
15:57.25 | zono | hi all |
15:57.34 | bkw_ | denon I just got up.. and I have a cold |
15:57.41 | denon | ah |
15:57.44 | zono | I wont to ask about Asterisk |
15:57.48 | denon | well I'll let ya off the hook then .. |
15:57.53 | p0lar69 | ok zono |
15:58.36 | doughecka | ~eeks |
15:58.37 | | somebody said eeks was http://eeks.convergence.com.pk |
15:59.04 | Exomorph_ | wb jsmith :) |
15:59.23 | *** join/#asterisk xupinet (~xupinet@150.162.248.8) |
15:59.26 | doughecka | hey |
15:59.32 | doughecka | eeks.confergance.com is down |
15:59.35 | xupinet | hi all! one easy question |
15:59.42 | denon | +.pk |
15:59.43 | jsmith | xupinet: What's the question? |
15:59.44 | xupinet | in extensions.conf file |
15:59.55 | xupinet | what does include mean? |
16:00.01 | xupinet | what does it do? |
16:00.05 | doughecka | yea |
16:00.07 | xupinet | that's the question |
16:00.30 | xupinet | jsmith: can you answer me? |
16:00.31 | jsmith | xupinet: It includes the extensions from another context in the current context... |
16:00.32 | p0lar69 | it allows you to "include" other files with * configugratuon in them |
16:00.51 | *** join/#asterisk loko-moko (loko-moko@c-67-165-107-230.client.comcast.net) |
16:00.56 | Exomorph_ | So I'm gonna mod the code to display only sip messages from xxx ip. |
16:01.19 | p0lar69 | so you can keep all extensions from each context in seperate files |
16:01.22 | jsmith | Exomorph_: Just look at it with a packet sniffer, and use that to filter... |
16:01.28 | xupinet | but aren't they all include? why you must include them in context what just two lines above you have declared them? |
16:01.38 | jsharp | "include" in extensions.conf allows you to include contexts into other contexts...not include files. |
16:01.54 | jsmith | xupinet: Let me give you an example... |
16:02.06 | Exomorph_ | jsmith: That doesn't show all the logic/etc going on on the asterisk side tho. It only show's it they are talking to each other. |
16:02.16 | p0lar69 | #include is what I was talking about...... |
16:02.24 | p0lar69 | <- is going back to sleep |
16:02.27 | Exomorph_ | jsmith: and tcpdump doesn't show whats inside the packets themselves. |
16:02.39 | xupinet | jsmith: i am listening |
16:02.42 | jsmith | Exomorph_: Use ethereal.... |
16:02.50 | JerJer | Exomorph_: wrong |
16:02.53 | jsmith | xupinet: Give me just a second... I'm trying to build a good example... |
16:02.58 | *** join/#asterisk detten2 (~john@213.219.141.57) |
16:03.05 | xupinet | jsmith: yes of course! |
16:03.06 | Exomorph_ | JerJer: Explain? :) |
16:03.19 | JerJer | man tcpdump |
16:03.29 | Connor- | tcpdump can.. -s0 |
16:03.31 | zoa | <-- looking for a mirror with the debian 3.0r2 |
16:03.37 | Pj_ | Use ngrep |
16:03.48 | Exomorph_ | Ya, but it still doesn't show the nice logic that asterisk can. :) |
16:04.18 | JerJer | Connor: incomplete answer |
16:04.18 | *** join/#asterisk Shido6 (~shido@d57-81-103.home.cgocable.net) |
16:04.25 | JerJer | tcpdump -s0 -x |
16:04.32 | h3x | JerJer: what carrier(s) do you use for inbound toll free? |
16:04.34 | *** join/#asterisk anthm (~anthmct@CPE-24-167-193-161.wi.rr.com) |
16:04.34 | *** join/#asterisk ]data[ (~data@193.138.95.4) |
16:04.42 | h3x | im trying to figure out something for rozo |
16:04.42 | Connor- | tcpdump -s0 -xxxv -ieth# |
16:04.51 | rozo | hi |
16:05.00 | *** join/#asterisk llboy (~llboy@hoochie.digium.com) |
16:05.01 | JerJer | h3x: Global Crossings, Qwest and ATT |
16:05.10 | h3x | are you LCRing it ? |
16:05.10 | rozo | gx here as well |
16:05.15 | JerJer | no |
16:05.16 | Connor- | oh and -X also |
16:05.23 | h3x | heh |
16:05.24 | JerJer | -x |
16:05.25 | h3x | ok |
16:05.40 | Connor- | no -X prints ascii |
16:05.54 | JerJer | h3x: we have batches of numbers from each |
16:06.00 | Exomorph_ | Connor: That gives some nice hex output... but no text. |
16:06.08 | Exomorph_ | Connor: LOL read my mind. |
16:06.25 | Connor- | tcpdump -s0 -xxxvvvX -ieth# |
16:06.33 | p0lar69 | use ethereal |
16:06.34 | p0lar69 | :) |
16:06.35 | h3x | well i was thinking he could send his WA calls to you with sms/800 or toll free enhanced routing |
16:06.35 | Connor- | the capital X gives you ascii |
16:06.55 | h3x | but that might tilt the non-rboc in the wrong direction.... |
16:07.14 | *** part/#asterisk zono (~trillian@inray.spnet.net) |
16:07.15 | JerJer | ? |
16:07.21 | h3x | intrastate calls |
16:07.28 | JerJer | he can |
16:07.39 | JerJer | we have CLECs sending us traffic |
16:07.49 | *** join/#asterisk klicTel (~Claude@modemcable115.119-131-66.mc.videotron.ca) |
16:07.53 | h3x | thats cool |
16:08.28 | h3x | i think enhanced routing wont work because it would span two customer accounts |
16:08.34 | zigman | does anyone know a |
16:08.35 | Exomorph_ | Connor: Ok, that might give me text... But its not formated anywhere near what asterisk does... Nor does it show the logic behind what the Asterisk is doing... |
16:08.41 | zigman | NIC with ROL ? |
16:08.43 | JerJer | ? |
16:08.49 | h3x | hes got gx |
16:09.01 | h3x | it would prob have to be done in sms/800 |
16:09.27 | Connor- | tcpdump just gives you raw data.. you can tee it to a file then debug after the session |
16:09.32 | p0lar69 | Exomorph can you use ethereal? |
16:09.48 | JerJer | h3x: i guess i'm not following you |
16:09.59 | JerJer | prolly cuz i haven't slept in like 40 hours |
16:10.08 | h3x | me either :( |
16:10.18 | p0lar69 | Exomorph? |
16:10.25 | Exomorph_ | 01p0lar69: Could.... Have to install it first. |
16:10.35 | jsharp | JerJer: sleep gud. |
16:10.37 | p0lar69 | it decodes SIP nucly |
16:10.41 | p0lar69 | nicly |
16:10.55 | Exomorph_ | Hmmmm |
16:11.00 | JerJer | jsharp: wish i could... i keep hearing ringing in my sleep |
16:11.07 | JerJer | the fone that is |
16:11.10 | p0lar69 | i also found a HTML prog that took a dump file and made a web page that traced the SIP messages out |
16:11.24 | p0lar69 | you like something like that? |
16:11.30 | jsharp | I know the feeling. |
16:11.43 | jtodd | p0lar69: Yes, that would be pretty handy. What's the name of the parser? |
16:11.51 | p0lar69 | lemme look |
16:12.05 | JerJer | fun fun |
16:12.13 | jsharp | from Texas. |
16:12.15 | jtodd | jsharp: thxgiving holiday festivities starting early? |
16:12.19 | Exomorph_ | Ya... send me the url... If nothing else I'll include it in the asterisk doc some of us are making. |
16:12.19 | JerJer | oh shitty |
16:12.37 | p0lar69 | ok |
16:12.40 | JerJer | i thought driving to chi from michigan sucked |
16:12.42 | p0lar69 | its really cool |
16:12.49 | jsharp | yah. We're headed up to visit family. |
16:12.51 | p0lar69 | shows the whole call flow and SIP setup etc |
16:13.06 | jsmith | p0lar69: So give us the URL already :-) |
16:13.17 | p0lar69 | looking......... |
16:13.28 | p0lar69 | dude you do need some food |
16:13.30 | p0lar69 | :_) |
16:13.48 | p0lar69 | found it |
16:14.06 | jsharp | And hunt around chiland for employment as well. |
16:14.25 | jsmith | p0lar69: Sorry... low blood sugar this morning... |
16:14.59 | p0lar69 | http://www1.cs.columbia.edu/sip/download/sip_scenario/ |
16:15.11 | p0lar69 | use this |
16:15.11 | p0lar69 | tcpdump -s 0 -i eth0 'port 5060' -w /var/log/sip1.dump |
16:15.14 | jtodd | Another one of Henning's tools, no doubt. |
16:15.17 | p0lar69 | to make the dump file |
16:15.24 | p0lar69 | heh |
16:15.31 | p0lar69 | I smell EDU all over it |
16:15.43 | p0lar69 | http://www1.cs.columbia.edu/sip/download/sip_scenario/ |
16:15.46 | p0lar69 | base url |
16:15.51 | *** join/#asterisk ]data[ (~data@193.138.95.4) |
16:16.16 | p0lar69 | got it Exomorph? |
16:16.45 | Exomorph_ | Yup. Thanks. |
16:16.56 | Exomorph_ | Just a dir with files in it... :) |
16:17.08 | p0lar69 | yeah |
16:17.21 | p0lar69 | those are the tar files |
16:17.30 | jtodd | p0lar69: What I'd really love is a color-coded version that runs via an ANSI terminal session that hands out the same kind of data, instead of post-processing... |
16:17.34 | p0lar69 | use the tcpdump command and create a file |
16:17.34 | *** part/#asterisk xupinet (~xupinet@150.162.248.8) |
16:18.08 | *** join/#asterisk p0lar (~Miranda@155.101.253.117) |
16:18.12 | p0lar | ok yes |
16:18.15 | Exomorph_ | jtodd: So you'd like the sip debug fixed up to output a cleaner message format? |
16:18.17 | p0lar | that would be nice |
16:18.41 | p0lar | check this out it may help yer prob |
16:19.51 | *** join/#asterisk cocoy (strange123@ipdial-172-28.tri-isys.com) |
16:20.21 | cocoy | how do i make an extension for OH323 ? |
16:20.33 | JerJer | cocoy: can't help you on that one |
16:20.42 | cocoy | tnx jerjer |
16:20.51 | bkw_ | fuck |
16:20.54 | bkw_ | someone shoot me |
16:20.57 | jets | hey whats up p0lar |
16:21.05 | p0lar | hey |
16:21.05 | *** join/#asterisk ram (~ram@hoochie.digium.com) |
16:21.09 | Exomorph_ | :) |
16:21.09 | p0lar | trying to wake up |
16:21.21 | cocoy | anyone pls help me |
16:21.23 | *** join/#asterisk ManxPower (~eric@ip68-109-105-237.pn.at.cox.net) |
16:21.37 | p0lar | whats up bkw_? |
16:21.56 | p0lar | you cant be doing as bad as I have already today |
16:22.19 | bkw_ | good |
16:22.24 | bkw_ | denon what are you doing to crash * |
16:22.28 | JerJer | cocoy: i wrote the H.323 channel driver that's distrubuted with Asterisk |
16:22.30 | bkw_ | and are you running latest cvs? |
16:22.42 | Shido6 | hey |
16:23.08 | p0lar | Hey jets |
16:23.37 | p0lar | I talked to our Man over at Mountain States.... he said he just talked to you guys yeaterday too.... |
16:23.47 | cocoy | i think so.. my friend installed it, am just configuring * so it can connect to GK then make call |
16:23.56 | p0lar | he told me you names but all I know is your aliases |
16:24.11 | p0lar | Ryan Nelson |
16:25.27 | JerJer | cocoy: then he installed a useless driver, imho |
16:25.33 | *** join/#asterisk i8kl (~lar@indo1.indosoft.unb.ca) |
16:28.17 | denon | bkw_: see that bug .. cvs as of yesterday, didnt do anything, it just happened this morning |
16:28.33 | bkw_ | updated zaptel lately? |
16:28.39 | denon | updated everything yesterday |
16:28.41 | denon | I do all at once |
16:29.05 | jsmith | p0lar: Yeah, I pointed Jets over to Ryan at Mt. States... do I get a commission? |
16:29.11 | bkw_ | "bt full" gives more goodies when you do another backtrace |
16:31.01 | torment0r|wk| | JerJer: do you think it's possible to get a cisco +12 ip phone working with asterisk? |
16:32.41 | *** join/#asterisk skeeziks (tim@66-23-208-2.clients.speedfactory.net) |
16:33.43 | skeeziks | Is it possible to set the bitrate on audio recordings coming out of asterisk? On .wav files for example? |
16:36.49 | ManxPower | skeeziks, no, It's always 8000 since that's what telephony audio is. |
16:36.59 | skeeziks | 16-bit? |
16:37.06 | ManxPower | I don't recall. |
16:37.13 | ManxPower | prolly 16 bit |
16:38.21 | Corydon76-work | Yes, 16 bit |
16:38.37 | skeeziks | Cool, thanks guys |
16:38.45 | Corydon76-work | Although it's the frequency that matters, not the number of bits |
16:39.01 | ManxPower | you can always resample it to a higher bitrate if you app requires it. |
16:39.44 | i8kl | send text agi, what channels support it |
16:40.00 | Corydon76-work | Yes, but upsampling will give you worse results than downsampling |
16:40.06 | i8kl | I'm having a hard time finding info on it |
16:41.03 | skeeziks | So it's effectively a 128kb/sec stream... |
16:41.44 | skeeziks | Thanks guys |
16:42.53 | znoG | kapejod: around? |
16:43.07 | *** join/#asterisk mrgoby (~mrgoby@pcp05304587pcs.wanarb01.mi.comcast.net) |
16:43.29 | mrgoby | morning all |
16:44.06 | *** join/#asterisk l-fy_ (~diana@home-25022.b.astral.ro) |
16:45.49 | klicTel | can anyone give me hand with extensions.conf? My problem is that when an extension is on the phone, and another call comes in for that extension, it just keeps on ringing and ringing in the party's ear... is there a way to disable the second line? or to force a transfer to voicemail directly? |
16:46.07 | bkw_ | klicTel grandstream? |
16:48.07 | klicTel | i have a channel bank with analog phones |
16:48.33 | ManxPower | klicTel, uh, callwaiting=no and threewaycalling=no in zapata.conf |
16:48.34 | *** join/#asterisk stan (~stan@213.78.66.69) |
16:48.57 | ManxPower | Go to http://www.digium.com/index.php?menu=documentation and look at the "Unofficial Links" section. This section has links to a wide variety of 3rd party Asterisk related pages. My page is the "Asterisk Resource Pages". |
16:49.09 | klicTel | ManxPower: that completly deactivate the second line? |
16:49.23 | ManxPower | klicTel, It's not a second line, it's call waiting |
16:49.42 | klicTel | yes call waiting... sorry |
16:50.11 | *** join/#asterisk outtolunc (~me@h-66-166-19-148.SNVACAID.covad.net) |
16:50.14 | ManxPower | klicTel, I think my sample zapata.conf file on my asterisk page has callwaiting disabled, but you'd have to look to be sure. |
16:50.37 | klicTel | ManxPower: I'm gonna try it... thanks |
16:50.39 | h3x | three way calling is a good thing to have |
16:50.49 | h3x | call waiting is annoying as hell |
16:51.21 | h3x | i still wish there was a way to set up slave call appearances of zaptel devices |
16:51.24 | bkw_ | http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&item=3059326125&category=51256 |
16:51.35 | h3x | so you could have it roll over to line 2 key for instance on a phone |
16:51.50 | h3x | man this fuckin rules |
16:52.17 | gadams666 | hell, I'm just happy there is finally multiple presentations for a single extension on the cisco 6.0 SIP cdoe |
16:52.30 | h3x | i found a office space in the same building as williams, xo, xspedius, broadwing, and qwest for $0.63/ft^2 per month here in vegas |
16:52.43 | h3x | er .64 |
16:53.32 | znoG | isnt there some way to set the caller on hold for X secs/mins until the extension becomes free, otherwise voicemail? |
16:53.45 | h3x | znoG: call queues. |
16:53.48 | outtolunc | http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&item=3059325876 |
16:54.08 | gadams666 | h3x: where's that at (vegas native here) |
16:54.23 | znoG | h3x: isn't that a good solution to what klicTel wants? |
16:54.29 | h3x | sahara & lamb is where all the telcos are |
16:54.33 | h3x | i was gonna get colo space |
16:54.47 | h3x | but i can get like 2500 ft^2 for less than the price of a 8x8 cage |
16:54.47 | h3x | heh |
16:55.10 | gadams666 | environmental controls too? |
16:55.18 | h3x | and the money saved in like 3-4 months i can just buy a damned dc power plant, batteries, rectifiers and shit off ebay |
16:55.28 | gadams666 | only good thing I like about my co-lo space, A/C, power, and security all taken care of |
16:55.41 | gadams666 | h3x: ;) there ya go |
16:55.57 | h3x | hahahahaha. Rack Mount... ROTFL |
16:56.15 | h3x | I could just steal power from the neighbors :P |
16:56.30 | i8kl | is it possible to send custom text to display on snom200 sip phones |
16:56.35 | h3x | most of the spaces in this building have iron bars over all the windows and shit |
16:57.31 | gadams666 | well, considering the area, don't blame them. hehe. Sam's Town brings in a rough crowd. |
16:58.15 | h3x | yeah it sucks |
16:58.28 | h3x | but what the hell |
16:58.30 | znoG | which area? |
16:58.37 | h3x | even though i live in the west side, DI is a quick way to get there |
16:58.54 | h3x | since it turns into lamb anyway |
16:58.56 | gadams666 | yup - bars are good protection. get space to park your car inside too! |
16:59.02 | h3x | HAHAHAH |
16:59.09 | tclark | znoG:http://bugs.digium.com/bug_view_page.php?bug_id=0000509 |
16:59.26 | gadams666 | we used to do that up near Valley View and Industrial |
16:59.28 | *** join/#asterisk erik (~eanders@host-127-202-220-24.midco.net) |
16:59.44 | h3x | tommy rockers or something? |
17:00.02 | gadams666 | razor wire fencing didn't keep the punks from throwing rocks at his car. so he started parking in the bulidng |
17:00.14 | h3x | shit i live at flamingo & decatur and have a call center across the street from me |
17:00.18 | znoG | tclark: haha, thanks. so many cool things you can do with *! |
17:00.25 | gadams666 | Tommy's is up at Decatur, is it not? been a few years since I've been in town |
17:00.26 | h3x | and they are paying like $2.10 or some crap for their least |
17:00.29 | h3x | lease |
17:00.30 | h3x | no |
17:00.42 | h3x | actually its like flamingo & industrial |
17:01.40 | gadams666 | ah, prolly moved since he orniginally opened it. Tommy's ok, knew him back in UNLV daze |
17:01.40 | h3x | the reason all that shit is over there is Excel's fault |
17:01.52 | h3x | they put a big phone switch next door so all the telcos built up around there |
17:01.55 | h3x | theres 24 of them on that block |
17:02.27 | h3x | plus its a easy exit towards the phoenix/california/salt lake corner of the state where they have their fiber run |
17:02.28 | gadams666 | lol. better be good service to Green Valley and Henderson then. |
17:02.57 | h3x | shit... |
17:03.11 | h3x | i got some guys that are moving their call center onto greenspun property |
17:03.12 | gadams666 | yah. was never in the telecoms b4 moving out of Vegas |
17:03.17 | h3x | and you can't get anything but XO on that block |
17:03.41 | h3x | you cant even order a sprint POTS line anywhere near 215 and green valley pkwy |
17:04.01 | gadams666 | damn, that just aint right. |
17:04.07 | h3x | i wish we had global crossing here |
17:04.20 | h3x | when enron went BK, GX got wiped out too |
17:04.24 | gadams666 | GBLX - now that was an intersting company |
17:04.24 | h3x | they were coloed with em |
17:04.41 | h3x | so now we have these 230+ mile loops to LA |
17:04.56 | oriontkn | ouch |
17:05.57 | *** join/#asterisk GhostNr1 (~Ashmed@193.10.185.3) |
17:06.02 | gadams666 | STM-1 or above? |
17:06.06 | *** join/#asterisk marrandy (~marrandy@209.216.76.1) |
17:06.23 | h3x | is what stm-1 |
17:06.27 | marrandy | Hello. Fixed my earlier fax machine issue |
17:06.45 | marrandy | On to the next issue. |
17:07.06 | h3x | er well lets just say a t1 local loop is $750 |
17:07.08 | bkw_ | if I had 10 bucks for every time I wanted to shoot the telco.. I would be one rich mofo |
17:07.14 | gadams666 | stm-1 == oc-3 I think. then submarine fiber guys seem to talk STM only. sigh |
17:07.15 | h3x | actually now they sell em for 200 but |
17:07.27 | h3x | i know, but i didnt know what you were referring to |
17:07.36 | gadams666 | bkw_: $0.05 and you'd still be richer than Gates |
17:07.36 | marrandy | On a conventional fax/ans phone, you can monitor as people leave a message and pickup. |
17:07.38 | mrgoby | shoot, bkw_ if i had $0.01 |
17:07.43 | h3x | the problem is theres no way for me to get a 0 mile loop to GX |
17:08.16 | oriontkn | or if your cliped onto the vm ports :) |
17:08.17 | marrandy | How do you monitor/put through the pc speaker in * ? |
17:08.32 | mrgoby | kram: I suggest a new topic "Ma Bell beats her kids" |
17:08.34 | ManxPower | marrandy, Actually that's only the case with an answering machine. Not for PBX or Key Systems |
17:08.48 | ManxPower | Or voicemail systems. |
17:08.59 | oriontkn | panasonic, iwatsu and a few others have the ability to listen to messages as they are left |
17:09.08 | ManxPower | oriontkn, celever. |
17:09.10 | raoul | cu all :) |
17:09.11 | oriontkn | the kx-TD series in perticula |
17:09.12 | oriontkn | r |
17:09.12 | ManxPower | clever, even |
17:09.15 | marrandy | so you can't monitor as a message is being left in VM ? |
17:09.18 | gadams666 | GBLX still got US presence on their Frontier network? |
17:09.20 | bkw_ | People names Johnny Depp sexiest man |
17:09.25 | bkw_ | RIIIGHT |
17:09.32 | ManxPower | marrandy, Not without doing dialplan hacks |
17:09.33 | bkw_ | he's not sexy at all. eww eww eww |
17:09.51 | ManxPower | bkw_, I think he's kinda cute |
17:10.09 | bkw_ | ewww |
17:10.16 | gadams666 | god actor. esp in Tim Burton films. IMO |
17:10.19 | bkw_ | gag me with E.T.'s finger. |
17:10.25 | mrgoby | heheh |
17:10.45 | marrandy | hmm...that could be a problem. I am testing this at home |
17:11.07 | ManxPower | marrandy, Then buy an answering machine and don't use the voicemail system |
17:11.37 | oriontkn | theres no way to intercept the stream and copy it to another timeslot on another port? |
17:11.41 | marrandy | I have one. But in the testing phase, I want to replace it as I laern the system |
17:11.45 | h3x | Yeah, well they sell a local enhanced service product on the class 5 switches. |
17:11.45 | ManxPower | That feature can't be all THAT popular since almost everyone I know uses either the voicemail on their cellphone or the voicemail service from their LEC and neither of those support that feature. |
17:12.04 | ManxPower | marrandy, Yes. Be prepared to do some coding in C |
17:12.06 | oriontkn | or CLAES |
17:12.09 | marrandy | laern = learn |
17:12.34 | ManxPower | marrandy, You could prolly hack app_monitor to spit the audio at /dev/audio |
17:12.34 | mrgoby | marrandy you could write an AGI that plays your voicemail for you after they leave it ... although that might not be the desired effect |
17:13.03 | ManxPower | Heck you might even be able to give app_monitor /dev/audio as an output filename. |
17:13.05 | *** join/#asterisk HeeD (~db@s216-232-111-32.bc.hsia.telus.net) |
17:13.06 | ManxPower | I don't know. |
17:13.30 | marrandy | Well, you know how it is, you call screen and monitor the line and then decide whether to pickup. I was hoping to suplicate this or the misses is going to be unhappy |
17:13.31 | gadams666 | gotcha. GBLX, Southern Crossing, and the rest of the sub companies were HQ'ed down the street from us in Bermuda |
17:13.40 | marrandy | suplicate = duplicate |
17:13.42 | ManxPower | marrandy, You don't have callerid? |
17:13.42 | *** join/#asterisk ]data[_ (~data@193.138.95.4) |
17:14.13 | marrandy | It costs $6.95 plus taxes per month |
17:14.15 | gadams666 | ironically they were denied the abiltiy to land a fiber there. cust care and all data servers provided by clueless and worthless at the time. |
17:14.25 | ManxPower | marrandy, put an answering machine on the misses FXS port. |
17:14.47 | mrgoby | Does anyone know what is going on with DynExtenDB ?? The site is down, or at least not accepting my pings/http requests |
17:15.08 | marrandy | I want to play with it here, but obviously, have to keep the boss happy ;-) |
17:15.12 | *** join/#asterisk cocoy (strange123@203.76.221.68) |
17:15.28 | ManxPower | Yet another reason not to get married, but I digress. |
17:16.00 | marrandy | ManPower: you a * developer ? |
17:16.08 | cocoy | what do i need to put on extensions.conf so i can call thru a GK? |
17:16.32 | marrandy | ManxPower: you a * developer ? Damn...my typing is going to pot |
17:16.32 | tclark | marrandy ManxPower: invoke zapbarge on the fxo channel to listen to vms .. |
17:17.20 | marrandy | oohh...ahhh...can you also pickup ? |
17:17.23 | ManxPower | marrandy, No, but I play one on IRC. |
17:17.36 | mrgoby | cocoy i think last night someone said that all you have to do is register asterisk with the GK and then you can call through with Dial(h323/blah@blah) |
17:17.43 | ManxPower | marrandy, sapbarge is designed for bosses to evesdrop on their employee's phone calls. |
17:17.57 | ManxPower | zapbarge, even |
17:18.01 | h3x | sapbarge? is that a tree version? |
17:18.13 | ManxPower | tclark, how would you send the zapbarge audio out the sound card? |
17:18.19 | h3x | you hear the next 2 people under you at the same time? :) |
17:18.49 | tclark | ManxPower: missed that part, yea that would be a mod to zapbarge |
17:18.56 | marrandy | marrandy returns to learning mode as the discussion ensues |
17:18.57 | h3x | I have an idea, How about you send voicemail calls to a meetme |
17:19.10 | ManxPower | tclark, why not using app_monitor and give it a filename of /dev/audio? |
17:19.15 | h3x | and somehow set up that meetme to invite the voicemail and monitoring party. naaah nevermind |
17:19.23 | mrgoby | h3x that might work ... and make your console a silent listener |
17:19.41 | h3x | zapbarge would be better |
17:19.46 | HeeD | I got my g.729 license just now.... setup on * and cant wait to go home to try on ata-186 :) |
17:19.47 | mrgoby | maybe??? hehehe we are searching here :-) |
17:20.07 | tclark | ManxPower: wee with a small mod i have to invoke zapbarge I can talk as well as listen not sure about app_monitor .. |
17:20.35 | cocoy | mrgoby- can u give me an idea how can i make call from SIP to GK? |
17:20.42 | h3x | I don't get why monitor and barge are named what they are |
17:20.43 | h3x | monitor should be log or record |
17:20.44 | mrgoby | maybe |
17:20.48 | ManxPower | tclark, Can you also play a sound file? |
17:20.48 | h3x | and barge should be monitor |
17:21.11 | ManxPower | "Please disconnect now. The line is required for a 911 emergenct call" sort of thing. |
17:21.16 | mrgoby | cocoy this is just my understanding, which might be limited so don't take this to be from an expert, but |
17:21.27 | tclark | ManxPower: not right now but I guess you could make that an option & do ast_streamfile .. |
17:21.33 | h3x | who the hell would care for a 911 call. Just Hangup() the damned line |
17:21.35 | oriontkn | it needs to support autovon precedence |
17:22.04 | *** join/#asterisk Vco (~Vco@h24-76-11-125.wp.shawcable.net) |
17:22.05 | *** join/#asterisk Damin ([rtFCpCntc@nucleus.nacs.net) |
17:22.07 | cocoy | mrgoby - i'll gladly accept what u can give me |
17:22.16 | oriontkn | <beep> <beep> and dumps the caller |
17:22.23 | Damin | Alright.. So I got my first live demo of asterisk last night.. |
17:22.42 | mrgoby | you can register your * machine with the GK ... this allows * to call h323 clients... then, just make an extension that calls an h323 client who is also registered with the GK |
17:22.43 | tclark | ManxPower: if you look at zapbarge it is realy bridgeing the channels to a meetme conf, its the same code as meetme |
17:22.50 | mrgoby | then just call that extension on a SIP client |
17:23.01 | zoa | anyone with a sipbarge ? |
17:23.10 | Damin | I've known about it for a while, but I was just clued into how totally kick ass it is! |
17:23.10 | mrgoby | cocoy does that help? |
17:23.13 | ManxPower | h3x, I'm concerned about someone trying to dial right after they are disconnected |
17:23.20 | Damin | Anyone from Digium online? |
17:23.55 | mrgoby | is there a doctor in the house? |
17:24.02 | cocoy | mrgoby - that's what i need an extension syntax i used exten => _X.,1,Dial(OH323/h323:${EXTEN}@111.111.111.1,60,r) |
17:25.06 | ManxPower | For some reason I thought it's supposed to be OH323/ip:${EXTEN}@111.111.111.1,60,r |
17:25.13 | t4k | how do you increse taps for echocancelation? |
17:25.17 | ManxPower | But I don't run H323 so I don't know for sure. |
17:25.18 | mrgoby | cocoy , unfortunately my associates took the box that has our h323 extensions offline, and i boked too much smot in college to remember the syntax exactly for h323 |
17:25.29 | cocoy | mrgoby- i will try |
17:25.40 | *** join/#asterisk maharzan (~chandra@202.51.76.74) |
17:25.44 | mrgoby | sorry |
17:25.49 | mrgoby | that was poked too much smot |
17:25.53 | mrgoby | :-D |
17:26.04 | mrgoby | indeed |
17:26.19 | maharzan | wasim akram???? |
17:26.25 | maharzan | where r u? |
17:26.33 | mrgoby | kram |
17:26.37 | mrgoby | there he is |
17:26.47 | maharzan | hey any body outside US successful in setting up caller ID??? |
17:27.24 | HeeD | I am using voicemail... when dialing in on FXO card and as soon as woman's voice answers prompting to leave a msg after beep... if person calling hangs up it takes a few secs to recognize hangup and records a second or 2 of silence. Any way to add more time after the beep prompt? |
17:27.59 | ManxPower | HeeD, Why do you want more time after the beep prompt? |
17:28.17 | maharzan | beep, more seconds |
17:28.27 | maharzan | LOL |
17:28.28 | ManxPower | HeeD, Just fiddle with the silence stuff in voicemail.conf |
17:28.28 | HeeD | manx because if someone hears the voicemail prompt and decides to just hangup and not leave a msg, it still records a couple secs of silence |
17:28.37 | maharzan | 8-) |
17:28.49 | ManxPower | HeeD, You would rather the system not record the first few seconds of their message? |
17:29.04 | cocoy | mrgoby - do i need to change something else on extensions.conf ? any more ? how abt "default" |
17:29.15 | *** join/#asterisk rusty_ (~rusty@65-101-255-24.dnvr.qwest.net) |
17:29.26 | HeeD | no, I want the voicemail to wait a few seconds after the prompt to leave after beep before it actually starts recording |
17:29.40 | denon | HeeD: make a beep.gsm with a delay |
17:29.44 | denon | some silence |
17:29.57 | ManxPower | *beep* *wait* [missed the name and phone number] "so if you want to get togather tonight give me a call" |
17:29.59 | HeeD | denon that sounds like it would work |
17:30.19 | denon | HeeD: it'll work fine, but make sure you have a script copy your beep back after every CVS, cause it'll get overwritten |
17:30.20 | mrgoby | cocoy i'm not sure... i would put it in your default context if you don't understand how contexts work |
17:30.40 | ManxPower | HeeD, Do you know how every other user of Asterisk solves that problem? |
17:30.56 | HeeD | Manx no |
17:31.02 | *** join/#asterisk Mike (~mike@dsl-200-67-4-11.prodigy.net.mx) |
17:31.14 | cocoy | mrgobt - error "Channel 'H323:1820' sent into invalid extension 's' in context 'default', but no invalid handler" |
17:31.19 | ManxPower | ; How many seconds of silence before we end the recording |
17:31.19 | ManxPower | maxsilence=10 |
17:31.19 | ManxPower | ; Silence threshold (what we consider silence, the lower, the more sensitive) |
17:31.19 | ManxPower | silencethreshold=128 |
17:31.33 | Connor- | So, can anyone tell me why when the IVR answers, that 80% of the time, it custs of the first few seconds of speach ? |
17:31.43 | mrgoby | hmmmmmmmmmmmm what does your exention look like cocoy ?? |
17:31.52 | ManxPower | cocoy, Looks like the call is being sent to extension "s" in the [default] context. |
17:31.52 | mrgoby | give me the line in extensions.conf |
17:31.53 | Connor- | Mostly happens with SIP, but, sometimes with PSTN too |
17:31.56 | mrgoby | yes |
17:32.38 | maharzan | anybody not from US |
17:32.39 | cocoy | mrgoby - [sip] |
17:32.40 | cocoy | exten => _X.,1,Dial(OH323/ip:${EXTEN}@203.144.224.167,60,r) |
17:32.44 | maharzan | ye yeah |
17:32.49 | maharzan | say yeah |
17:33.29 | cocoy | mrgoby i have no [default] context under extensions.conf |
17:33.30 | Mike | where do i configure h323 in asterisk? |
17:33.32 | ManxPower | maharzan, CallerID works in a lot of countries. Not BT UK lines, however. |
17:33.47 | maharzan | india? |
17:33.54 | ManxPower | Mike, That question is a lot like "How do I drive a car." Read the README file. |
17:34.01 | ManxPower | maharzan, prolly not. 8-) |
17:34.03 | maharzan | i am in nepal.. they should be similar to india pk.. |
17:34.17 | bkw_ | ATTENTION H.323 SUCKS.. DONT USE IT IF YOU DONT HAVE TOO! |
17:34.20 | ManxPower | maharzan, How is CallerID delivered to you? |
17:34.21 | bkw_ | ok that said... |
17:34.26 | rozo | maharzan: we weren't able to get callerid to work in new delhi, india |
17:34.56 | ManxPower | maharzan, Is it FSK or DTMF. When is it delivered? Before the first ring or between the first and second ring? Does it support name as well as number |
17:35.25 | mrgoby | bkw_ didn't you field this last night ?? |
17:35.30 | maharzan | i 'll have to figure that out.... but i seee that when i connect directly to pstn.. the callerid appears after one ring.. as that os US.. don't know what they use.. FXS or other.. |
17:35.41 | i8kl | is it possible to send custom text to display on snom200 sip phones |
17:35.47 | mrgoby | an h323 question very similar |
17:35.56 | mrgoby | i can't remember if that was you |
17:35.56 | *** join/#asterisk anachron (~fradooj@adsl-64-169-89-27.dsl.snfc21.pacbell.net) |
17:36.00 | bkw_ | i8kl you have asked that like twice |
17:36.03 | bkw_ | post to the mailing list |
17:36.06 | maharzan | it gives me number only.. like 012222555 for local no |
17:36.21 | cocoy | mrgoby - that's not me, am just learning * now |
17:36.33 | ManxPower | maharzan, If it's sent as FSK after the first ring then either the X100P supports that callerid or it can easily be patched to do so. |
17:36.39 | cocoy | but am vey familiar with opengk |
17:37.10 | maharzan | whats DTMF?bbefore first ring? as in UK? |
17:37.12 | ManxPower | Actually if it's sent as FSK or DTMF after the first ring chances are that it can be made to work eventually |
17:37.13 | Mike | ManxPower: im trying to figure out if the quicknet card can do sip would make it easier |
17:37.19 | ManxPower | maharzan, Yes. |
17:37.21 | Mike | ManxPower: do you know any software? |
17:37.54 | Damin | q |
17:37.54 | ManxPower | Mike, No, the Quicknet card does not do SIP. Just like the ZapTel cards don't do SIP and the Dialogic cards don't do SIP. |
17:37.54 | ManxPower | Mike, I know lots of software. |
17:37.56 | cocoy | mrbogy - do i need only one line for OH323 to work? |
17:38.10 | Mike | ManxPower: so i need to get it working under h323? |
17:38.14 | *** join/#asterisk FlowerPower (~michal@otho.nask.waw.pl) |
17:38.30 | ManxPower | Mike, I have no idea. I know Netscape, MS Ofice, Linux, GNOME, but I don't know H323 |
17:38.37 | FlowerPower | hi |
17:38.44 | ManxPower | Mike, NO! These cards are PSTN cards not SIP cards. |
17:39.01 | mrgoby | well.... i don't know really .... exten => 420,1,Dial(SIP/600@111.111.111.111) |
17:39.09 | mrgoby | this is one that works for me with sip |
17:39.15 | *** join/#asterisk segal4 (~segal@cuscon4728.tstt.net.tt) |
17:39.21 | ManxPower | If you want to interface with SIP do it just like you would with a ZapTel card. Load the kernel drivers and set up your extensions.conf and the rest of the .conf files. |
17:39.23 | mrgoby | i would imagine changing the SIP to H323 would work |
17:39.26 | mrgoby | but i don't know |
17:39.35 | maharzan | callerid comes after the first ring.. b4 second... what can be done to fix it? |
17:39.41 | Mike | ManxPower: i have it on a windows computer |
17:39.45 | ManxPower | maharzan, Post a bug report and pray |
17:39.46 | FlowerPower | is here any1 who has succsessfully configured isdn with *? |
17:39.51 | ManxPower | Mike, I can't help you. |
17:39.59 | maharzan | :D |
17:40.13 | maharzan | some one told me that i can change callerid.c file |
17:40.15 | ManxPower | Mike, But I guess you would have to load SIP or H323 software on the Windows computer. |
17:40.22 | maharzan | i'll try to have a look around |
17:40.48 | FlowerPower | maybe you know some sites where the problem is described |
17:40.50 | FlowerPower | > |
17:40.51 | FlowerPower | ? |
17:41.03 | ManxPower | Go to http://www.digium.com/index.php?menu=documentation and look at the "Unofficial Links" section. This section has links to a wide variety of 3rd party Asterisk related pages. My page is the "Asterisk Resource Pages". |
17:41.11 | ManxPower | Look at the CAPI or kapejod pages |
17:41.14 | Mike | ManxPower: thats whay im looking sip software compatible with quicknet card |
17:41.32 | ManxPower | Mike, I doubt you are going to find anyone here that runs Quicknet on Windows. |
17:41.47 | ManxPower | Mike, Call up the makers of the card. |
17:42.10 | Mike | ManxPower: i will look for some box running linux with isa |
17:42.14 | bkw_ | kram kram kram |
17:42.16 | bkw_ | :) |
17:42.17 | Mike | i just found this windows box has isa |
17:42.21 | kram | lol |
17:42.26 | *** topic/#asterisk by kram -> "Ma Bell beats her kids" |
17:42.27 | Mike | thats my problems ISA is hard to find this days |
17:42.36 | ManxPower | Mike, Would it not just be cheaper to but an X100P? |
17:42.38 | Mike | and the only computer with an isa slot is a winXP computer |
17:42.47 | marrandy | Connor: Do you mean the first part of the announcemnet |
17:43.07 | Mike | ManxPower: a friend just told me to use it |
17:43.23 | *** join/#asterisk os2doc (~michael@65.115.136.98) |
17:43.26 | Mike | ManxPower: i dont want to use it as a zaptel card |
17:43.28 | *** join/#asterisk jdg (~chatzilla@202.3.246.151) |
17:43.34 | mrgoby | kram YOU RULE |
17:43.39 | Mike | ManxPower: i just want to plug a telefone to put an extension |
17:43.50 | ManxPower | You mean like a TDM10B? |
17:44.07 | Mike | sorry i dont know alot of the hardware yet |
17:44.09 | marrandy | TDM400 |
17:44.17 | ManxPower | Mike, I seriously doubt you will find any Asterisk user that has the configuration you are trying to do |
17:44.25 | marrandy | has expansion to 4 extensions |
17:44.30 | ManxPower | marrandy, No, that's a 4 ports card, not the 1 port card |
17:44.48 | *** join/#asterisk denon (denon@synapse.subneural.net) |
17:44.51 | ManxPower | Well technically they are both the same card. |
17:44.58 | Mike | ManxPower: this card can also be useful as a x100p? |
17:45.05 | ManxPower | Mike, No. |
17:45.13 | Mike | ok |
17:45.17 | Mike | then ill just return it |
17:45.18 | denon | haha .. that topic is great |
17:45.24 | ManxPower | The X100P is for using with a telephone line. The TDMx0B is for using with a phone |
17:45.31 | marrandy | Not at the moment (TDM400) they are working on FXO modules |
17:46.11 | marrandy | I bought the TDM400 with 1 module, then bought the extra 3 fxs modules |
17:46.35 | marrandy | I also have the X100P as I couldn't wait for the FXO modules to appear |
17:46.43 | *** join/#asterisk os2doc (~michael@65.115.136.98) |
17:46.44 | ManxPower | marrandy, The TDM400P is the card with no modules, the TDM10B is the card with 1 mondule. |
17:47.04 | *** join/#asterisk lucifuge3 (trilluser@pcp05047461pcs.ivylnd01.pa.comcast.net) |
17:47.22 | UnixDawg | does anyone know what stealths name is |
17:47.24 | ManxPower | marrandy, if you don't put P or B at the end of the name when you refer to the card people can get confuzzled. |
17:47.48 | silug | so is digium replacing the original tdm400p cards? |
17:47.53 | marrandy | http://www.digium.com/index.php?menu=wildcard_tdm400p |
17:48.06 | marrandy | Oh, I see. They have changed the terminology |
17:48.39 | bkw_ | what changed? |
17:48.58 | *** join/#asterisk rainer_home (~rainer@p508AF127.dip.t-dialin.net) |
17:49.40 | ManxPower | TDM400P is the card with no modules installed. |
17:50.59 | marrandy | Probably...I don't remember. I ordered mine in Mid August and it was delayed till September until the new TDM400P's came in |
17:51.00 | cocoy | mrgoby - can u give urls where can i see setting-up OH323 |
17:51.18 | bkw_ | cocoy don't do it if you don't need it |
17:51.24 | bkw_ | h323 is pure and evil.. |
17:51.29 | bkw_ | at the same time. |
17:51.35 | ManxPower | cocoy, There are no docs with the tarball you downloaded? |
17:52.00 | cocoy | bkw - i need to call from SIP to GK |
17:52.30 | *** join/#asterisk os2doc (~michael@65.115.136.98) |
17:52.47 | cocoy | manx - few docs on it |
17:53.00 | ManxPower | cocoy, Have you tried the H323 driver that comes with Asterisk |
17:53.40 | os2doc | Anyone want to answer a newbie question? Setting up a phone system |
17:54.00 | cocoy | manx - yes that's what am using |
17:54.02 | marrandy | If you want a connection to the CO and have extensions, then get http://www.digium.com/index.php?menu=developerskit_tdm |
17:54.17 | JerJer | THAT IS NOT OH323 THEN!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!! |
17:54.21 | ManxPower | cocoy, Uh, the chan_oh323 is not included with Asterisk. chan_h323 is included with Asterisk |
17:54.57 | ManxPower | cocoy, They are TOTALLY different software. |
17:55.27 | marrandy | I would get Two fxs modules so you can test between Two regular phones |
17:55.45 | cocoy | manx - how will i check if i have chan_oh323 , my friend said he already installed it |
17:55.58 | Corydon76-work | Well, they aren't totally different. But they are different enough that they aren't the same. |
17:56.20 | ManxPower | Corydon76-home: The codebase is totally different. The design is totally different. |
17:56.21 | JerJer | Corydon76-work: oh but they are tottally different |
17:56.35 | JerJer | the only simularity is they use the same H.323 stack |
17:56.36 | Corydon76-work | They implement the same protocol. |
17:56.59 | JerJer | Corydon76-home: LInux and windows use X86 processors also |
17:57.06 | ManxPower | Corydon76-home: Grandsream and Cisco 79xx both implement the same protocol too 8-) |
17:57.59 | Corydon76-work | JerJer: they aren't totally different, either. They're very different, but totally is not an appropriate qualifier |
17:58.21 | Exomorph_ | JerJer: you have to admit that having them both named so simular, can confuse alot of people. When I first looked at h323 channels for asterisk, I initially thought they were both the same project, just at some point they changed the name. (of course I now know different) |
17:58.33 | ManxPower | Corydon76-home: Think "totally different" is the correct term when talking to someone that's confusing the two. 8-) |
17:58.47 | bkw_ | ManxPower I agree |
17:58.56 | Corydon76-work | hyperbole |
17:58.57 | ManxPower | They should just call them chan_jerjer and chan_mbraca |
17:59.12 | Corydon76-work | mbranca didn't write chan_oh323 |
17:59.26 | ManxPower | Corydon76-home: or chan_whoeverwroteoh323 |
17:59.32 | *** join/#asterisk feanor (~feanor@63.245.86.109) |
17:59.52 | Corydon76-work | ManxPower: that's even more confusing |
18:00.11 | JerJer | technically its chan_h323 and asterisk-oh323 |
18:00.13 | ManxPower | Corydon76-home: Maybe so, but people would stop confusing the two. |
18:00.43 | Corydon76-work | ManxPower: considering what they do, not a chance |
18:01.39 | Exomorph_ | What needs to be done is a good Asterisk Doc made up (we're in the process, right jsmith?) that clearly states they are diffrent, and all the details with them. |
18:03.43 | Corydon76-work | how about chan_oh323_dynamic and chan_h323_static ? |
18:04.01 | JerJer | how about chan_broken_h323 and chan_h323 ? |
18:04.29 | *** join/#asterisk elbandy (~elbandy@pD9E3E5BB.dip.t-dialin.net) |
18:04.35 | bkw_ | FIGHT.. FIGHT.. FIGHT.. FIGHT.. |
18:04.38 | Corydon76-work | Unfortunately, oh323 is broken, so both channels are broken |
18:04.51 | *** part/#asterisk elbandy (~elbandy@pD9E3E5BB.dip.t-dialin.net) |
18:04.52 | os2doc | Would anyone be able to answer a few basic questions about setting up a basic phone system. I am having difficlty getting specific advice and information... |
18:04.52 | cypromis | chan no323 |
18:04.53 | jsmith | suggestion, not implementation... |
18:04.59 | JerJer | :) |
18:05.02 | jsmith | os2doc: Just ask... |
18:05.08 | bkw_ | chan_ho323 |
18:05.17 | JerJer | lol |
18:05.26 | Corydon76-work | chan_oink323 and chan_moo323 |
18:05.49 | bkw_ | its chan_orange and chan_apple |
18:06.01 | jsmith | chan_die_H323 |
18:06.14 | Exomorph_ | LOL |
18:06.14 | cypromis | you have 80 fowl oranges left .... beep ... select your destination party ... |
18:06.23 | mrgoby | chan_can_string.so |
18:06.40 | JerJer | we need an independant third party to setup and deploy each one, then whichever one he picks the other one goes totally away, forever |
18:06.44 | os2doc | Cool. ;) I am setting up a phone system for my practice, and want to use as much gnu software as possible. It is a medical practice. Anyway, 5 B1 lines coming in, and want to use asterisk for the phone system. The problem is trying to figure out what phones I can use with the system. Is it safe to assume that any PBX phone will work with Asterisk? Like the Nortel Meridian 2317 for example? |
18:07.05 | Corydon76-work | JerJer: could you really live with yours going away, though? |
18:07.17 | jsmith | os2doc: No... only analog phones, or VoIP phones |
18:07.28 | JerJer | Corydon76-work: sure...i've purged H.323 totally from my operation and will never go back to it |
18:07.29 | jets | os2doc why use old phones, when you can get nice voip phones |
18:07.31 | marrandy | I don't think any proprietry phone will work with asterisk |
18:07.44 | jsmith | os2doc: PBX phones are usually a proprietary digital format, which Asterisk has no way of talking to |
18:07.52 | cypromis | :) |
18:08.05 | cypromis | I am trying to rid my operations of h323 as well |
18:08.07 | marrandy | Unless you link asterisk to your PBX which sort of defeats the purpose |
18:08.09 | JerJer | Plus, its my code, if for some dumb reason there I find a need for it, i'll pull it out of my own cvs |
18:08.23 | JerJer | :P |
18:08.44 | Corydon76-work | That's not exactly your code going away, forever... :-P |
18:09.04 | JerJer | it wouldn't be in asterisk and i wouldn't support it |
18:09.09 | marrandy | JerJer: do you know if you can monitor and pickup someone leaving a Voicemail ? |
18:09.16 | JerJer | if it happened to loose the test |
18:09.16 | *** join/#asterisk anthm (~anthmct@CPE-24-167-193-161.wi.rr.com) |
18:09.20 | os2doc | jsmith: ok, so I use standard phones or voip phones. The problem is trying to figure out which phones are VOIP. I have been looking at phones on ebay to save money, but am getting really confused... I had assumed that the PBX phones would be voip. Bad assumption? |
18:09.44 | Corydon76-work | bad assumption. |
18:09.45 | JerJer | but anyone that has acutally used that other one knows the limitations |
18:09.56 | Corydon76-work | Most PBX phones are not VoIP phones |
18:10.06 | ManxPower | os2doc, PBX phones are NOT "standard phones" |
18:10.41 | JerJer | hence why i wrote mine (after the author wouldnt listen to me and told me i couldn't do any better) |
18:10.48 | ManxPower | PBX phones are "vendor phones made incompatable with any other system so the vendor can lock you into their phones and charge you incredibly high prices" |
18:11.02 | os2doc | That answers some questions. Now where to find phones. |
18:11.09 | Corydon76-work | Face it, you can either go for many different proprietary phones, that all implement a protocol in various different ways, or you can use expensive channel banks and cheap phones from Walmart... |
18:11.52 | ManxPower | os2doc, You'll find a lot of different suggestions. *I* think VoIP phones at a decent pricepoint at NOT mature and should NOT be used. I feel that people should use analog phones on a Zap port. |
18:11.56 | marrandy | How many incoming lines ? How many extenions ? |
18:11.59 | Corydon76-work | Pick your poison, as it were. |
18:12.03 | marrandy | T-1 ? |
18:12.04 | ManxPower | There are other people that think VoIP phones are just fine. |
18:12.07 | marrandy | pri ? |
18:12.10 | UnixDawg | where in the hell is stealth |
18:12.19 | UnixDawg | I am going to beat him with asterisk |
18:12.28 | os2doc | marrandy: 5 incoming lines to start, provably 15 extensions at max |
18:12.37 | Corydon76-work | Until VoIP phones are all $20 apiece, it's a still a problem of pick-your-poison |
18:13.12 | ManxPower | Corydon76-home: I'm waiting for phones similar to Cisco 79xx phones to be $100/each |
18:13.14 | h3x | which will never happen unless they make them with no dsps and no itu algos |
18:13.43 | JerJer | h3x: which would be fine |
18:13.50 | marrandy | sounds like a T-1 and a channel bank. Adtran 650 ? (newbie alert) that's what i've heard |
18:13.55 | h3x | yeah for asterisk users |
18:13.56 | JerJer | fuck the bastards running anything that's not asterisk |
18:14.01 | os2doc | Marrandy: no, pots analog b1 lines coming in. I know I willn eed a channel bank to multiplex to a data intefacce into Asterisk box |
18:14.25 | JerJer | they should be punshed for blantely wasted all that money |
18:14.26 | ManxPower | I have no problem leaving out phone systems that do not use codecs like Speex, iLBC, GSM, etc |
18:14.42 | JerJer | waisting |
18:15.13 | JerJer | I have faith in wasim and his crew |
18:15.43 | ManxPower | JerJer, Unfortunatly, I don't. |
18:15.48 | os2doc | I would like to be able to run just one set of cables for phones and network, wich I thought I could do with IP phones. But with analog phones, I get to run a second wire. Not a big deal, but cooler iwth no wires. |
18:16.07 | JerJer | ManxPower: i've already talked to him on his IAX hardfone |
18:16.11 | HeeD | what would I use to connect 6 analog overlines to asterisk? WOudl I have to get provisioned differently from telco...ie. T1 instead of 6 lines? |
18:16.34 | marrandy | price it out |
18:17.03 | JerJer | HeeD: you can start out with a T100P and a adtran or cac channel bank |
18:17.09 | ManxPower | JerJer, Oh, I'm sure he can build the EEKS phone, I just don't know if 1) he can get mfg costs down and 2) release it soon enough for me to care. |
18:17.32 | JerJer | heed: then when u get to about 11 lines upgrade to a PRI and flebay the channel bank |
18:17.37 | ManxPower | JerJer, wasim is a smart guy. |
18:17.49 | HeeD | jerjer ok thanks |
18:17.58 | JerJer | ManxPower: they are going after the barbietone market |
18:18.02 | marrandy | Listen to JerJer: he's a * guru |
18:18.09 | h3x | you can get fractional pri's |
18:18.16 | h3x | some clecs have minimum 4 lines |
18:18.28 | JerJer | h3x: yeah, sometimes they don't make much sense financially |
18:18.30 | os2doc | I priced a T1 with the local telco, 5 pots with hunt, 1 dedicated, 1 fax/DSL line, 280$, T1 890$ starting. No contest. the extra 600$ gets me a channel bank on ebay, and the asterisk server. |
18:18.35 | ManxPower | h3x, We are having a line installed today that's PRI with 6 channels along with 384K internet |
18:18.36 | Mike | ManxPower: in linux i just load the quicknet module and can i use it to conect a phone and give it an extension or just PSTN? |
18:18.37 | torment0r|wk| | i wanna use my bonephone.. lol.. the name is silly |
18:18.39 | torment0r|wk| | http://iptel.org/products/bonephone/ |
18:18.43 | h3x | JerJer: well xspedius here in vegas for 4 lines + d channel is like $200/mo |
18:18.47 | ManxPower | Mike, I don't know. |
18:18.48 | h3x | $100 if its robbed bit |
18:19.01 | h3x | many: thats an even better idea |
18:19.09 | h3x | er. manx |
18:19.12 | ManxPower | I think dog was smileing on me when UPS lost my quicknet order and the vendor no longer stocked them. |
18:19.15 | mrgoby | torment0r|wk|: did you get bonephone to work ??? |
18:19.23 | ManxPower | So I got an X100P and a TDM10B instead |
18:19.24 | JerJer | Mike: your first mistake was acutally spending money on quicknet hardware |
18:19.26 | mrgoby | i tried but never got it to run |
18:19.32 | torment0r|wk| | no.. i just seen it and started to laugh |
18:19.46 | mrgoby | it is awful ... i would stay away |
18:20.07 | ManxPower | Mike, Quicknet is NOT well supported with Asterisk. |
18:20.29 | JerJer | more correctly: 01Mike, Quicknet is NOT well supported |
18:20.36 | ManxPower | I don't know if that's the fault of the card/drivers (as most people seem to think) or the fault of Asterisk (which other people thing), but regardless it's not well supported. |
18:21.20 | JerJer | ManxPower: their LTAPI is a joke |
18:21.42 | JerJer | ManxPower: run valgrind and load chan_phone.so and u'll see for yourself |
18:21.52 | *** join/#asterisk l-fy_ (~diana@home-25022.b.astral.ro) |
18:22.19 | JerJer | or hell run any of the ixj sdk sample apps in valgrind |
18:24.10 | torment0r|wk| | i have 3 quicknet cards looking at me.. eak! |
18:24.19 | JerJer | flebay them, quickly |
18:24.54 | JerJer | i've got like 4-5 phonehacks and a linehack or three collecting dust around here somewhere myself |
18:24.58 | ]data[_ | valgrind, uck |
18:25.11 | torment0r|wk| | i've been searching today to find an cheap sip phone.. without and luck |
18:25.19 | JerJer | 7960 |
18:25.25 | torment0r|wk| | there like $300 |
18:25.29 | JerJer | yep |
18:25.31 | *** join/#asterisk stan_ (~stan@213.78.71.85) |
18:25.38 | JerJer | worth every penny |
18:25.52 | torment0r|wk| | i'd believe it.. but the issue is i'm going to college and i'm poor |
18:26.11 | torment0r|wk| | it's bad enough i live off tuna |
18:26.12 | JerJer | barbietone |
18:26.18 | ]data[_ | JerJer: innit :} |
18:26.18 | HeeD | my * box is takin a couple seconds to realize a call is hung-up on FXO card. Is this normal or is there some setting? Does have to do with speed of server (433Mhz Celeron) |
18:26.23 | JerJer | ahh that's the smell |
18:26.30 | torment0r|wk| | yep |
18:26.44 | jtodd | Anyone remember the three-finger salute to get a SNOM 200 to reboot without plug-yanking? |
18:26.48 | h3x | thats because the LEC is slow |
18:26.50 | ]data[_ | torment0r|wk|: if you want something CHEAP and CHEERFUL, get a grandstream |
18:27.04 | JerJer | barbietone |
18:27.25 | ManxPower | HeeD, That's normal on analog lines. |
18:27.25 | h3x | they have black phones now :PPPPP |
18:27.45 | torment0r|wk| | has anyone used one before though |
18:27.54 | JerJer | so Jamican barbietones |
18:28.02 | miller7- | HeeD: you're in luck cause my analog card does not understand hang up at all :P |
18:28.07 | h3x | haha |
18:28.16 | HeeD | manx ok, still tryin to fix my voicemail issue... I think I may attempt to make an IVR that answers and prompts user to press a key to leave voicemail |
18:28.17 | torment0r|wk| | i wanted a pink one |
18:28.25 | HeeD | miller haha |
18:28.40 | *** join/#asterisk glLoadIdentity (~asdfrt@abn139-91.interaktif.net.tr) |
18:28.44 | JerJer | miller7-: you need disconnect supervsion on ur line |
18:28.49 | HeeD | what is voicemail2, maybe that would help me ? |
18:28.53 | miller7- | JerJer: what do you mean? |
18:29.20 | torment0r|wk| | HeeD, you need to update if you still have voicemail2 |
18:29.23 | torment0r|wk| | that's so last week |
18:29.25 | JerJer | miller7-: find a old school fone...like one that lights up or something powered only by the PSTN voltage |
18:29.43 | JerJer | then make a call and let the far end hang up first, see if the lights go out for a fraction of a second |
18:29.45 | HeeD | I am using voicemail, not voicemail2 |
18:30.16 | HeeD | I d/l latest cvs last week, how do I check quickly if there is newer cvs ? |
18:30.21 | torment0r|wk| | you shouldn't have voicemail2 is what i'm getting at.. they replaced voicemail with voicemail2 |
18:30.35 | JerJer | HeeD: cvs status |
18:30.39 | JerJer | in the asterisk/ |
18:30.54 | miller7- | JerJer: well, the thing is that no matter how much I leave the card does not hang up. This happens in one line, others work fine (using busydetect=yes) |
18:31.01 | marrandy | [data[-: whats your config for the grandstream ? |
18:31.23 | JerJer | miller7-: what wakky country u in? |
18:31.53 | miller7- | JerJer: the specific telco has problems in the area I am in... |
18:31.56 | HeeD | and to update I type cvs update ? |
18:32.06 | JerJer | bitch, loudly |
18:32.15 | JerJer | HeeD: yeppers |
18:32.21 | miller7- | JerJer: mixed hardware from 2 vendors, old hardware etc... very old pop |
18:32.23 | ]data[_ | marrandy: asterisk config? |
18:32.32 | HeeD | JerJer, and then just make install again ? |
18:32.47 | JerJer | miller7-: show them asterisk |
18:32.53 | bkw_ | blah |
18:32.56 | JerJer | tell em u'll deploy it for a free PRI |
18:32.58 | JerJer | :) |
18:33.15 | torment0r|wk| | HeeD, do a make update |
18:33.16 | ]data[_ | mmm pri |
18:33.21 | torment0r|wk| | then make install |
18:33.21 | HeeD | ok |
18:33.30 | miller7- | JerJer: hell, yeah... they are slower than replay... would take them 50 years to browse through first screenshot |
18:33.33 | JerJer | torment0r|wk|: make update just updates the cvs date |
18:33.43 | JerJer | heed: make install will work |
18:34.01 | HeeD | I gonna backup my .conf files just in case :) |
18:34.05 | marrandy | anything special on the grandstream setup. I initially had it going to FWD but have registration errors. Anything special about settings on the grandstream itself ? |
18:34.06 | JerJer | miller7-: run your own copper then |
18:34.18 | miller7- | JerJer: :P |
18:34.24 | *** join/#asterisk Santo (~santosh@209.78.110.175) |
18:34.53 | marrandy | now I'm going to the asterisk box instead. |
18:35.00 | ]data[_ | marrandy: nope, its a doddle |
18:35.00 | torment0r|wk| | does anyone know why i can't get any woman? |
18:35.24 | jtodd | torment0r|wk|: Because you spend too much time on IRC. |
18:35.29 | ]data[_ | torment0r|wk|: tried 'cvs co woman' ? |
18:35.33 | marrandy | do you use inband signalling ? |
18:35.36 | torment0r|wk| | lolol |
18:36.00 | miller7- | JerJer: anyway, I won't use analog, this card was just for meetme tests. I am waiting for kapejod's ISDN card for small deployment before going to PRI |
18:37.11 | marrandy | jtodd: thought you were busy ;-) |
18:37.35 | torment0r|wk| | d00d.. people become unbusy when it's comes to chicks |
18:37.40 | jtodd | I am never to busy to slap someone around. |
18:37.43 | jtodd | :-) |
18:38.20 | tzanger | hahaha |
18:38.43 | marrandy | Well if you want to check my configs for security issues, or any other problems, the offer still stands |
18:38.43 | tzanger | ~seen citats |
18:38.45 | | citats is currently on #asterisk. Has said a total of 10 messages. Is idling for 1d 23h 52m 38s |
18:38.53 | tzanger | wow two days idle |
18:38.54 | ]data[_ | marrandy: dtmf is 'in-audio' |
18:39.21 | *** join/#asterisk sjoep125 (~sjoerd@213-84-218-42.adsl.xs4all.nl) |
18:39.46 | jtodd | Has anyone seen their SNOMs (2.02t) recently go into "endless loops" during authentication on any SIP messages? |
18:40.18 | Connor | change dtmf to info or rfc2833 |
18:40.33 | Santo | I am having some telnet issue |
18:40.38 | Santo | Redhat 7.2 --> telnet localhost 3128 ..... after few seconds I get connection reset by foreign host ???? |
18:41.02 | *** part/#asterisk raulfragoso (~raulfrago@200-204-142-71.dsl.telesp.net.br) |
18:41.11 | Santo | any telnet Gurus |
18:41.23 | Santo | bkw_ you there |
18:42.01 | Santo | I want my Voicegenie box to talk to * |
18:42.18 | ManxPower | Santo, Try #linux |
18:42.19 | Santo | anyone with VXML and * |
18:42.40 | ManxPower | Isn't 3128 the default Squid proxy port? |
18:42.45 | Santo | yes |
18:43.02 | ManxPower | Santo, What does this have to do with Asterisk? |
18:43.51 | Santo | no Once I set my VoiceGenie box I could use H.323 Gateway -->*-->Voicegenie box |
18:44.58 | Santo | VoiceGenie (VXML server) is SIP only Gateway is H.323 Only |
18:45.15 | *** join/#asterisk Keso (~kddand@63.80.216.3) |
18:45.27 | Santo | so * has to talk to VXML server |
18:46.08 | Santo | ManxPower any idea about telnet |
18:46.25 | ManxPower | Santo, No idea. |
18:46.40 | *** join/#asterisk qiu (~qiu@194.102.203.13) |
18:47.53 | Santo | hmm bkw_ you there |
18:48.02 | HeeD | hmm I updated cvs and now sip phone cant login |
18:48.11 | HeeD | I notice there is no sip show peers command in new console |
18:49.38 | HeeD | do I have to update libpri and zaptel also maybe ? |
18:49.41 | torment0r|wk| | no |
18:49.51 | torment0r|wk| | it should just work.. have you stoped and reloaded asterisk? |
18:49.54 | HeeD | yes |
18:50.00 | torment0r|wk| | bkw_!@#! |
18:50.03 | torment0r|wk| | <3 |
18:50.15 | UnixDawg | BKW is always here |
18:50.24 | UnixDawg | this is nothing new |
18:50.58 | HeeD | it is like sip is missing |
18:51.27 | *** join/#asterisk denon (denon@synapse.subneural.net) |
18:51.30 | torment0r|wk| | that's odd |
18:51.30 | HeeD | from /usr/src/asterisk I did.... make update and then make install |
18:51.35 | marrandy | Santo: telnet, what's up ? |
18:51.37 | ManxPower | HeeD, try sip show users |
18:51.39 | denon | heh .. guess someone likes dell. |
18:51.53 | HeeD | manx, it says no command |
18:51.54 | *** part/#asterisk marrandy (~marrandy@209.216.76.1) |
18:52.01 | HeeD | if I type help, there is no sip stuff at all |
18:52.04 | *** join/#asterisk marrandy (~marrandy@209.216.76.1) |
18:52.15 | HeeD | maybe I should try to re-download the full cvs ? |
18:52.29 | UnixDawg | sip show peers |
18:52.37 | UnixDawg | and sip show regisrty |
18:52.46 | bkw_ | http://bugs.digium.com/bug_view_page.php?bug_id=0000562 |
18:52.49 | bkw_ | JerJer you see that? |
18:52.49 | UnixDawg | sip show clevage |
18:53.10 | HeeD | no sip commands are working |
18:53.17 | torment0r|wk| | mmm.. clevage |
18:53.29 | ManxPower | HeeD, Looks like chan_sip didn't load then |
18:53.49 | bkw_ | Clearing connection ip$localhost/14668 reason=EndedByTransportFail |
18:53.52 | bkw_ | that dont look right |
18:54.16 | HeeD | hmm maybe cvs didnt update something... I noticed I was getting verify erros and said it was gonna retry |
18:54.46 | Santo | bkw_ howdy |
18:56.12 | HeeD | it doesnt even recognize command... stop now |
18:56.13 | HeeD | hmm |
18:56.20 | ManxPower | try show modules |
18:56.24 | ManxPower | look for chan_sip |
18:56.44 | HeeD | Module Description Use Count |
18:56.45 | HeeD | chan_sip.so Session Initiation Protocol (SIP) 0 |
18:56.45 | HeeD | chan_iax.so Inter Asterisk eXchange 0 |
18:56.45 | HeeD | res_monitor.so Call Monitoring Resource 1 |
18:56.45 | HeeD | res_indications.so Indications Configuration 0 |
18:56.45 | HeeD | res_crypto.so Cryptographic Digital Signatures 1 |
18:56.47 | HeeD | res_parking.so Call Parking Resource 1 |
18:56.49 | HeeD | res_adsi.so ADSI Resource 1 |
18:56.51 | HeeD | res_musiconhold.so Music On Hold Resource 1 |
18:56.53 | HeeD | chan_modem_aopen.so A/Open (Rockwell Chipset) ITU-2 VoiceMod 0 |
18:56.55 | HeeD | chan_modem.so Generic Voice Modem Driver 0 |
18:57.18 | many | blerg. |
18:58.07 | ManxPower | are you sure you are not typing show sip users and not sip show users |
18:58.17 | HeeD | yup |
18:59.06 | HeeD | if I type help, it does not list any commands for sip |
19:00.51 | *** join/#asterisk sp3cialk (~alex@d141-82-32.home.cgocable.net) |
19:01.55 | HeeD | maybe I have to restart the server |
19:03.35 | HeeD | Im rebootin my linux box now |
19:05.06 | bkw_ | HeeD its not windows |
19:05.39 | HeeD | yes I know, but wasnot workin so I thought I would try that |
19:07.02 | HeeD | now when I type asterisk -r it just hangs after displayin a couple lines grrr |
19:07.17 | ManxPower | HeeD, type asterisk -c |
19:09.05 | HeeD | manx it registers some countries.... then says SIP seeding and name of a sipphone at 192.168.0.41:5060 for 3500 |
19:09.18 | HeeD | and is stopped there |
19:09.24 | *** join/#asterisk sobol_ (~sobol@router-1.szczecin.tpnet.pl) |
19:10.20 | *** part/#asterisk lucifuge3 (trilluser@pcp05047461pcs.ivylnd01.pa.comcast.net) |
19:11.07 | ManxPower | looks like something is blocking |
19:11.51 | HeeD | yes strange |
19:12.55 | HeeD | even an iax connection wont connect to it |
19:12.59 | denon | any of you guys deal at all with those dinky little Dell PE 400SCs? |
19:13.02 | HeeD | ie. friend's * box |
19:13.42 | Bonbon | does anyone have a spanish number which they can sell / give me? Need it for sip termination. |
19:13.43 | ManxPower | If SIP is blocking then nothing will work |
19:13.53 | HeeD | maybe I should try to re-download entire cvs and make again |
19:14.01 | HeeD | rather than update |
19:14.17 | *** join/#asterisk adam_gafachi (~diddy@69-55-69-130.da.netsville.net) |
19:14.25 | bkw_ | HeeD that makes no diff |
19:14.26 | ManxPower | If your original Asterisk is more than 2 weeks old you need to do that as was announced on the mailing list. |
19:14.39 | HeeD | ahh |
19:14.55 | HeeD | maybe I should join the mailing list? :) |
19:14.59 | ManxPower | you have to delete the asterisk source code an and other of the modules. |
19:15.00 | bkw_ | um yes |
19:15.09 | bkw_ | ManxPower Accually you don't |
19:15.13 | ManxPower | HeeD, I don't help people that I know are not on the mailing list. |
19:15.16 | bkw_ | it you make update twice it will be fine |
19:15.20 | bkw_ | thats all I did |
19:15.25 | HeeD | manx, ok Ill join the list now |
19:15.33 | *** join/#asterisk phiberkut (~phiberkut@tele-free-hotspot.netlinkip.com) |
19:16.18 | *** part/#asterisk sjoep125 (~sjoerd@213-84-218-42.adsl.xs4all.nl) |
19:16.31 | t4k | is there any known issues w/ echo cancel over tdmoe? |
19:16.39 | t4k | for example... it dosen't work? |
19:17.21 | t4k | if I'm doing tdmoe from hostA -> hostB, what host should have echocancel turned on? both? |
19:17.50 | t4k | will echocancel work if it's just turned on, on hostB? |
19:18.18 | *** join/#asterisk defian (ircuser@client80-83-46-147.abo.net2000.ch) |
19:18.44 | defian | hi |
19:20.46 | Exomorph_ | Is there any way to run an agi script when the asterisk starts up and/or stops? |
19:20.48 | bkw_ | <PROTECTED> |
19:20.48 | bkw_ | <PROTECTED> |
19:20.48 | bkw_ | <PROTECTED> |
19:20.48 | bkw_ | <PROTECTED> |
19:20.48 | bkw_ | <PROTECTED> |
19:20.50 | bkw_ | muhahahahah |
19:20.54 | bkw_ | I love privacyManager |
19:21.07 | bkw_ | my phone dont ring much at all |
19:21.55 | Corydon76-work | It don't? |
19:22.03 | bkw_ | nope |
19:22.16 | *** join/#asterisk Vco (~Vco@h24-76-11-125.wp.shawcable.net) |
19:22.20 | Corydon76-work | --> doesn't |
19:22.30 | HeeD | ok, I subscribed to both lists... waitin for confirmation emails |
19:22.36 | bkw_ | ok english guru.. i'm in oklahoma.. we say don't |
19:22.43 | bkw_ | :P |
19:23.04 | jsmith | bkw_: You can add me... as long as you don't tell anyone :-) |
19:23.14 | bkw_ | haha online friends list is HUGE |
19:23.15 | HeeD | bkw, I did a make clean, make update and make install.... noticed make update didnt do anythin 2nd time I ran |
19:23.26 | bkw_ | HeeD it usually wont |
19:23.26 | HeeD | waitin for it to compile... fingers crossed |
19:23.32 | jsmith | HeeD: It's not supposed to... |
19:23.47 | espenz | what is the best method to use asterisk and hardware as a telephone server? |
19:23.53 | bkw_ | but I have seen CVS be crack headed once or twice and it has checked stuff out the second time around |
19:23.56 | espenz | i mean, which hardware should I have? |
19:24.04 | bkw_ | espenz depends on your goal |
19:24.19 | espenz | my goal is to have, lik this: |
19:24.22 | espenz | press one for billing |
19:24.27 | espenz | press two for sales |
19:24.46 | bkw_ | My last set of friends all got mad at me.. they showed up every single day... ate all my food.. drank all my soda/pop/coke(what ever you call it in your neck of the woods) and I finally got sick of that and told em off |
19:24.46 | espenz | and own people answering the telephone |
19:25.07 | bkw_ | espenz you apparently didn't understand my statement |
19:25.17 | defian | espenz: do you live in the US or Europe? |
19:25.25 | espenz | Europe |
19:25.34 | bkw_ | how many lines? |
19:25.35 | defian | espenz: in Europe the definitce choice is an ISDN PCI card |
19:25.35 | espenz | Norway that is |
19:25.47 | espenz | why? |
19:25.49 | Corydon76-work | ya damn oklahomo's... :-P |
19:25.57 | defian | espenz: for example with the AVM c4 you can have upto 4 ISDN lines, meaning 8 simultaneous connections |
19:25.57 | espenz | how could I then transfer calls? |
19:26.01 | bkw_ | Corydon76-work you got it! |
19:26.05 | *** join/#asterisk denon (denon@synapse.subneural.net) |
19:26.19 | defian | espenz: you then either transfer call using a P2P (point-to-point) ISDN bus on the c4, or via SIP to a cheap Grandstream phone |
19:26.32 | defian | espenz: I am doing just that on my server (AMD + c4 + SIP phones) |
19:26.38 | espenz | hm |
19:26.45 | espenz | P2P isdn bus? |
19:26.49 | espenz | mean, ordinary isdn card? |
19:26.51 | bkw_ | I wish ISDN was cheaper here |
19:27.02 | bkw_ | 189/mth for 2B+D |
19:27.04 | espenz | isdn is realy cheap in norway |
19:27.05 | espenz | :) |
19:27.05 | defian | yes, with the c4 you can in theory (never did that) connect a phone and the c4 does the NT |
19:27.15 | defian | here I don't do that I do SIP phones |
19:27.20 | Corydon76-work | Move to Tennessee... |
19:27.23 | defian | easier to install, cheaper, works quite well |
19:27.48 | defian | the c4 however is not cheap: if you only have one ISDN line (two simultaneous connectins) maybe a cheaper ISDN card might be better |
19:27.50 | many | huh? 189USD for dual channel isdn? per month? |
19:27.51 | espenz | a subscription costs about, 14 dollars pr month |
19:28.09 | espenz | and a isdn card, about 28 dollar |
19:28.10 | bkw_ | many yes from SBC (link extension charge is most of that) |
19:28.14 | HeeD | ok, that didnt work, so I should just del evertying in my /usr/src/asterisk and re-download the cvs? Should I delete the modules too? |
19:28.19 | Corydon76-work | $50/month for ISDN... same cost as two analog lines... |
19:28.20 | many | phew. |
19:28.23 | espenz | omg? |
19:28.24 | espenz | :P |
19:28.32 | defian | many: anyway in US ISDN is not really ISDN, it depends on the provider |
19:28.42 | defian | many: (compatibility is not assured) |
19:28.44 | defian | many: will all equipment |
19:28.45 | many | yes. 56k isdn sucks. :-P |
19:28.45 | espenz | is SBC a provider? |
19:28.55 | Corydon76-work | HellSouth here... |
19:28.56 | bkw_ | DOVBS |
19:29.06 | defian | many: in the US it's not that important since POTS lines have caller-ID |
19:29.10 | defian | many: and are quite cheap |
19:29.20 | defian | many: you don't really need ISDN |
19:29.28 | Corydon76-work | but the TN Public Utility Commission forced BS to give ISDN *anywhere* in TN for cheap |
19:29.38 | many | i like isdn alone for the fact that it transfer the type of the call. |
19:29.51 | many | callerids are on analog lines here, too. |
19:29.55 | many | if you want that. |
19:30.54 | espenz | defian: anyways, i want a simple solution, number to choose type of customerhelp |
19:31.14 | espenz | how can I do that? i have a 2 channels ISDN card, that works fine in asterisk using the i4l-driver |
19:31.16 | defian | espenz: so even with an analog voice modem it could (?) work. |
19:31.26 | defian | espenz: plus either softphones or SIP phones |
19:31.39 | espenz | okey, but where do i connect the SIP ? |
19:31.52 | defian | to Ethernet |
19:32.26 | espenz | hm |
19:32.32 | espenz | cool |
19:32.38 | espenz | how much does a SIP phone cost? |
19:33.16 | ManxPower | espenz, $60 - $1,000 |
19:33.33 | high-rez | Do you guys usually enable MMX on zaptel interfaces? |
19:33.34 | espenz | cheap, where could I order then? |
19:33.52 | ManxPower | grandstream.com I think. |
19:34.12 | *** join/#asterisk raulfragoso (~raulfrago@200-204-142-71.dsl.telesp.net.br) |
19:34.13 | espenz | USA is early with technology |
19:34.20 | *** part/#asterisk raulfragoso (~raulfrago@200-204-142-71.dsl.telesp.net.br) |
19:34.24 | espenz | and, sorry my bad english ;-P |
19:34.28 | ManxPower | Or www.sipphone.com or any of a number of sites that carry what people here call the BarbieTone (I don't recalls what the actual name is since I'd NEVER buy one of those gorrible phones) |
19:35.13 | defian | BudgeTone |
19:35.25 | defian | ManxPower: I have one on my desk and it does what it should, especially for the price. |
19:35.50 | ManxPower | defian, You don't have problems with stuck keys or the phone slideing across the desk? |
19:35.56 | bkw_ | 'SeaQuest DSV' actor Brandis dead at 27 |
19:36.09 | *** join/#asterisk ricky1 (~ricky@hoochie.digium.com) |
19:36.18 | defian | ManxPower: for the sliding I put some specially crafted anti slipping glue, and for the keys, not yet. |
19:36.28 | ricky1 | hello... |
19:36.47 | HeeD | manx, which mailing list wast that mentioned on, I am checking november archives and cant find... but there are 3 or 4 lists |
19:37.20 | blitzrage | off to work, have a good day all! |
19:37.27 | Corydon76-work | ManxPower: those phones do have a bracket on the bottom, so you can bolt them down, if that's a problem |
19:37.31 | blitzrage | I love the topic |
19:37.43 | espenz | defian, hm, i dit not find, orderstuff on the grandstream |
19:37.43 | espenz | did |
19:37.46 | blitzrage | if someone knows the key mappings of comedian mail, please email me at leif@hacklocalhost.com |
19:37.59 | defian | espenz: a customer of me bought it |
19:38.07 | defian | espenz: AFAIK it's referred from sipphone.com |
19:38.19 | *** join/#asterisk Beave (~beave@bundy.vistech.net) |
19:38.19 | bkw_ | hahahahaha |
19:38.22 | Corydon76-work | ManxPower: and I haven't noticed the keys sticking on the GS phones... (yes, though, on the SNOM 100) |
19:38.22 | bkw_ | love the topic |
19:38.24 | Beave | hey all. |
19:38.53 | Corydon76-work | Hey, Beave, where's Wally? |
19:39.22 | Beave | I have a TDM400P, with only one port enabled. It appears to be upgradable to 4 ports. I looked for pricing the upgrades but dont see it on there webpage. any ideas how much it cost for a upgrade module for the TDM400P? |
19:39.36 | Beave | Corydon76: yeah.. Thats new.. :) |
19:39.46 | Exomorph_ | I'm making an outgoing call via sip to another provider, and for some reason the asterisk sip stack freezes... :( |
19:39.47 | Corydon76-work | Call Digium and hit the button for sales |
19:39.54 | ManxPower | Beave, $99, I think, you have to call Digium direct to order those modules |
19:40.04 | Exomorph_ | I've just upgraded to the latest cvs and thats when I started getting this problem. |
19:40.15 | Beave | ok.. thanks manxpower. |
19:40.23 | ricky1 | anyone knows where can i get to read on iaxcomm? |
19:40.55 | ManxPower | Exomorph_, Did you read the announcement about the CVS changes? |
19:40.57 | *** join/#asterisk G0L3M (~golem@CPE0040f42c4568-CM013529902742.cpe.net.cable.rogers.com) |
19:41.15 | Exomorph_ | Nope. What changes? |
19:41.28 | ManxPower | Exomorph_, Are you on the mailing list? |
19:41.42 | Bonbon | what does this mean: -- SIP Seeding '3991' at 192.168.254.189:5060 for 3600 |
19:41.59 | Exomorph_ | ManxPower: Just the Users list. |
19:42.02 | Bonbon | asterisk just hangs on it |
19:42.05 | bkw_ | it seeds the sip registrations from astdb |
19:42.09 | bkw_ | at the cli |
19:42.14 | bkw_ | do a database show |
19:42.18 | bkw_ | you will see them |
19:42.30 | bkw_ | that way if * crashes you have your registrations still in the db |
19:42.40 | Bonbon | but I don't get "Asterisk Ready" |
19:42.50 | ManxPower | Message-Id: <200311131556.hADFuhmT005139@mail.sigmasoft.com> |
19:43.14 | ManxPower | That's the message that talks about the CVS changes |
19:43.23 | defian | could anyone just check that my IAX connection works? |
19:43.38 | marrandy | defian: On the Grandstream, have you managed to get any of the 'feature' buttons to work ? |
19:43.39 | defian | (by dialing my IAXtel/gnophone number 1-700-895-5211) |
19:43.56 | defian | marranty: TRANSFER worls; CALLED works; CALLER works |
19:44.00 | ManxPower | defian, You can test that by calling your own IAXTel number |
19:44.09 | defian | marrandy: I didn't try the rest |
19:44.24 | bkw_ | ACK |
19:44.29 | JerJer | REJ |
19:44.31 | marrandy | defian> really, none of mine work, not even the called and callers |
19:44.32 | bkw_ | new zaptel changes cause kernel panic |
19:44.39 | marrandy | or the redial |
19:44.40 | defian | marrandy: with Asterisk? |
19:44.46 | marrandy | or flash |
19:44.48 | defian | marrandy: SEND also works for redialing |
19:44.49 | Bonbon | yeah, and asterisk doesn't seem to start with latest cvs |
19:44.54 | defian | marrandy: strange. |
19:45.17 | ManxPower | Bonbon, as I understand it that only happens if you didn't read the message about the changes to the CVS server. |
19:45.37 | Bonbon | Manx: oh, obviously not. Which changes/ |
19:45.38 | Bonbon | ? |
19:45.50 | ManxPower | Bonbon, Read the message ID... |
19:45.52 | ManxPower | Message-Id: <200311131556.hADFuhmT005139@mail.sigmasoft.com> |
19:45.58 | marrandy | You would think that it would save the last number it had dialed, but no |
19:46.06 | ManxPower | Bonbon, it was posted to bother the users and developer lists. |
19:46.13 | bkw_ | GREAT |
19:46.29 | Bonbon | Manx: what was it then? |
19:46.41 | ManxPower | Bonbon, read the archives. I'm not your personal assistant |
19:46.43 | defian | marrandy: what firmware version do you have? |
19:46.53 | Bonbon | ok, ok. But would that cause * not to start? |
19:47.17 | ManxPower | Bonbon, I don't know. I did what the message told me so I don't have those problems |
19:47.51 | espenz | defian: now I have bought a SIP phone ;-) |
19:48.00 | jules | marrandy, is that you martin? |
19:48.00 | defian | espenz: can you buy just one? |
19:48.06 | espenz | UPS Worldwide Express |
19:48.08 | espenz | defian: yes? |
19:48.11 | marrandy | yes. Hi Jules |
19:48.12 | espenz | :) |
19:48.12 | *** join/#asterisk Marlow (~marlow@3ffe:200:1:155:0:0:0:2) |
19:48.19 | espenz | Sub-Total: $79.99 |
19:48.19 | espenz | Shipping & Handling: $85.46 |
19:48.19 | espenz | Tax: $0.00 |
19:48.19 | espenz | -------------------------------------------------------------------------------- |
19:48.19 | espenz | Total: $165.45 |
19:48.29 | denon | man, im gettin damn sick of waiting for the Pioneer A07 drive to come out |
19:48.29 | defian | expensive |
19:48.31 | defian | shipping |
19:48.33 | jules | marrandy, I didn't know you were a IVR type ;) |
19:48.35 | denon | maybe I'll just buy an A06 for the house |
19:48.48 | espenz | defian: its norway |
19:48.50 | defian | espenz: my customer had to buy two, but it was about the same price (?) |
19:48.59 | defian | espenz: an Expensive Country? but how nice! |
19:49.00 | espenz | huh |
19:49.02 | defian | espenz: :) |
19:49.05 | espenz | :P |
19:49.17 | espenz | wierd shopping, it was on lindows |
19:49.17 | jules | marrandy, drop me an email. I might have some work for you. |
19:49.25 | marrandy | defian>: SEND, there is no SEND button on my phone ??? |
19:49.42 | marrandy | Oohhh...work...yummy |
19:49.44 | defian | marrandy: you probably don't have a BugdeTone-100 then |
19:49.56 | marrandy | I have the 102 with the Two ports |
19:50.13 | marrandy | Goes and checks |
19:50.14 | defian | aha |
19:50.15 | marrandy | .... |
19:50.20 | defian | two ports but no send key |
19:50.24 | defian | can't have everything :-> |
19:50.26 | Marlow | espenz : what was that you ordered ? |
19:51.18 | espenz | http://www.sipphone.com/images/grandstream.gif |
19:51.38 | Marlow | espenz : ah .. are you sure, that there is no dealer in europe for that ? |
19:51.50 | Marlow | espenz : i wouldn't order it in the US . |
19:52.01 | espenz | hm, |
19:52.04 | t4k | does echo cancel work over tdmoe? |
19:52.05 | espenz | :/ |
19:52.13 | JerJer | kram: this time it wasn't me who broke CVS :P |
19:52.30 | ManxPower | t4k, Why are you using TDMoE and not IAX2? |
19:52.32 | kram | what's broken in cvs? |
19:52.34 | *** join/#asterisk Mike (~mike@dsl-200-67-4-11.prodigy.net.mx) |
19:52.41 | high-rez | Yeah he didn't break cvs, he just inserted broken code. |
19:52.41 | Marlow | espenz : jeg vil se, om jeg kan få fingrene i en Cisco i stedet .. |
19:52.43 | high-rez | ;) |
19:52.47 | t4k | ManxPower -> uh. |
19:53.00 | t4k | does echo cancel work over tdmoe? |
19:53.04 | espenz | Marlow: Hva mener du? |
19:53.09 | JerJer | kram: kernel panic on new zaptel changes (see bkw) |
19:53.19 | ManxPower | kram, People are claiming chan_sip is blocking on startup. I don't know if it's a real problem or the fact they are updating from a pre cvs layout changes to an after cvs layout changes |
19:53.24 | Marlow | espenz : den er noget dyrere .... men også en bedre telefon .. |
19:53.28 | kram | jerjer: no, he just didn't unload it right |
19:53.34 | JerJer | ahh ok |
19:53.41 | espenz | Marlow: Sikkert, men jeg skal bare teste det først. |
19:53.41 | marrandy | defian>: it's a 102. Do you really have a 100 ? or is it a 101 |
19:53.47 | ManxPower | t4k, I am not aware of anyone using TDMoE so I don't know the answer to that. |
19:53.47 | HeeD | I just downloaded an entire new cvs and it doesnt work... same sip problem |
19:53.48 | Marlow | espenz : klart .. |
19:53.54 | espenz | når ble du norsk? |
19:53.56 | espenz | :p |
19:54.00 | ManxPower | HeeD, you deleted your old source right? |
19:54.02 | Marlow | espenz : nej .. dansk :) |
19:54.13 | HeeD | yes manx, I moved /usr/src/asterisk to /usr/src/asteriskbak |
19:54.15 | Marlow | espenz : men ... i skift er der jo ikke den store forskel .. |
19:54.19 | HeeD | and cvs made entire new dir |
19:54.26 | Marlow | espenz : faktisk dansker i Sverige :o) |
19:54.28 | espenz | Marlow: vet du om en forhandler for SIP i Norge eller europa generelt? |
19:54.32 | JerJer | no comprenda ur jive, dig |
19:54.42 | Marlow | espenz : jeg søger .... |
19:54.51 | ManxPower | HeeD, Are you using zaptel or pri as well? |
19:54.52 | Marlow | espenz : skal nok finde nogen på et tidspunkt .. |
19:54.57 | high-rez | JerJer: Does you use MMX optimizations in zaptel? :) |
19:54.58 | espenz | okey, gi meg output hvis du finner noe da ;-) |
19:55.00 | HeeD | manx yes zaptel |
19:55.09 | ManxPower | HeeD, I assume you did the same for zaptel? |
19:55.10 | Marlow | espenz : selvfølgeligt .. |
19:55.22 | HeeD | no, someone here told me I didnt need to |
19:55.28 | JerJer | d'oh |
19:55.29 | HeeD | Ill try that |
19:55.35 | bkw_ | man ok... i'm happy now |
19:55.36 | ManxPower | HeeD, I don't know if you need to or not, but try it. |
19:55.38 | defian | je ne comprends pas non plus :) |
19:55.44 | ManxPower | If that doesn't fix it, file a bug report |
19:56.04 | JerJer | i'll be brave |
19:56.18 | Marlow | defian / JerJer : :o) |
19:57.08 | marrandy | defian>: it's a 102. Do you really have a 100 ? or is it a 101 |
19:57.18 | bkw_ | haha |
19:57.32 | espenz | Marlow: men det er uansett litt for sent nå, kontoen har blitt belastet |
19:57.41 | defian | Marlow: was that norvegian? |
19:57.42 | Marlow | espenz : jup .. |
19:58.03 | Marlow | defian : for espenz norwegian, for me danish .. but they are similar, especially in writing .. |
19:58.28 | defian | Marlow: ok |
19:58.40 | defian | marrandy: it's written BudgeTon-100 |
19:58.46 | defian | marrandy: it's written BudgeTone-100 (sorry) |
19:58.49 | bkw_ | we know |
19:58.51 | JerJer | BarbieTone |
19:58.53 | bkw_ | or BarbieTone |
19:58.57 | bkw_ | or CrapStream |
19:59.00 | JerJer | GMTA |
19:59.09 | bkw_ | JerJer yeppers |
20:00.14 | bkw_ | asterisk*CLI> stop now |
20:00.15 | bkw_ | No such command 'stop' (type 'help' for help) |
20:00.16 | bkw_ | haha |
20:00.32 | Marlow | hmmm ... |
20:00.34 | Marlow | bkw_ : panic mode ? |
20:00.37 | HeeD | I tried updating zaptel and reloaded modules, asterisk still has problem with sip |
20:01.00 | bkw_ | ACK |
20:01.06 | JerJer | REJ |
20:01.06 | JerJer | HeeD: you have any host names in sip.conf that aren't resolving |
20:01.08 | bkw_ | broken cvs |
20:01.24 | HeeD | bks I get same error.... that stop now wont work either |
20:01.53 | HeeD | jerjer not that I know of |
20:02.06 | JerJer | my slow ass home box is chewing on the compile now |
20:02.27 | Exomorph_ | For those that need the URL for the cvs compile problem(s)... Go to: |
20:02.27 | Exomorph_ | http://lists.digium.com/pipermail/asterisk-users/2003-November/027050.html |
20:02.40 | bkw_ | HeeD ack.. evil thing |
20:02.47 | defian | bkw_: works very well. (GS) |
20:03.12 | JerJer | oddness...mine hangs on chan_iax.so (like it does when gethostbyname() is blocking) |
20:03.19 | HeeD | hmm and I didn't backup my old cvs... have to get from friend later |
20:03.31 | JerJer | HeeD: cvs update -D '1 day ago' |
20:03.36 | HeeD | oh |
20:04.33 | marrandy | Got to go. Catch you all later. Thanks for the info. Regards and best wishes !!! |
20:05.09 | HeeD | jerjer ok I typed that, make install again... see what happens |
20:05.46 | JerJer | yeah thre is defainatly a problem |
20:06.14 | Exomorph_ | JerJer: With the current changes? |
20:06.21 | Exomorph_ | err cvs I mean. |
20:06.41 | JerJer | yes, cvs head |
20:07.12 | bkw_ | I compiled lastnight and it was fine |
20:07.23 | bkw_ | after we did all the bug fixes.. just to double check it |
20:07.36 | mrgoby | is dynExtenDB on the asterisk FTP site ???? |
20:08.11 | Marlow | mrgoby : it's here: http://andreasotto.net/asterisk/ |
20:08.18 | JerJer | mrgoby: don't bother...its not the right way to deal with dynamic extensions |
20:08.22 | Exomorph_ | JerJer I'm compiling everything from scratch right now and installing to see if that fixes the problem. |
20:08.36 | mrgoby | JerJer: what is ?? |
20:08.49 | mrgoby | Marlow: the site is down |
20:08.54 | JerJer | not an asterisk application |
20:09.01 | jsharp | Any recommendations on a SIP client for MacOS X? |
20:09.09 | Marlow | mrgoby : not here .. |
20:09.22 | mrgoby | Marlow: really....hhhhmmmmmmmmmmmm |
20:09.31 | mishehu | oh where oh where has my cisco 7960 gone... oh where oh where can it be... |
20:09.33 | Marlow | JerJer : better suggestion to solve that then ? |
20:09.43 | mrgoby | JerJer: yes please elaborate |
20:09.52 | mrgoby | i find this very intereseting |
20:10.04 | JerJer | Marlow: see retrieve_extensions_from_mysql.pl |
20:10.23 | Marlow | JerJer : i'll have a look at that .. |
20:10.40 | JerJer | that's a decent way.. we do something sorta simular |
20:11.06 | *** join/#asterisk os2doc (~michael@65.115.136.98) |
20:11.11 | jsmith | mishehu: I didn't steal it! |
20:11.48 | bkw_ | ok cvs is fixed |
20:12.05 | mishehu | jsmith: well, smoebody must have, becuase it's not arrived from shido yet... |
20:12.09 | Exomorph_ | bkw: What did you do to get it working? |
20:12.23 | jsmith | mishehu: I've only got 40 or so of them right now :-) |
20:12.26 | bkw_ | Exomorph_ kram just fixed it. cvs update again |
20:12.38 | Exomorph_ | Ok |
20:12.38 | HeeD | jerjer that 1 day ago worked :) |
20:12.47 | bkw_ | HeeD todays cvs will work now too! |
20:12.53 | bkw_ | asterisk*CLI> show version |
20:12.54 | bkw_ | Asterisk CVS-11/21/03-14:10:15 built by root@asterisk on a i686 running Linux |
20:12.57 | Exomorph_ | HeeD: You can use the latest cvs now... its fixed. |
20:13.01 | HeeD | oh haa |
20:13.07 | *** part/#asterisk stan (~stan@213.78.71.85) |
20:13.08 | bkw_ | Heed things move fast around here |
20:13.13 | HeeD | :) |
20:13.14 | mrgoby | JerJer: where can i find the ....mysql.pl |
20:13.18 | mrgoby | ? |
20:13.20 | os2doc | Is there any way to attach one of the big fancy multiline phones to the asterisk system, like one of the receptionist phones. It seems that all I can find are propriatary. |
20:13.23 | mishehu | jsmith: care to donate one? ;-) |
20:14.02 | jsmith | mishehu: Uh, no. They're all in production, and you'd have to pry them from my cold dead fingers... Everyone here *loves* them. |
20:14.12 | mishehu | os2doc: a voip phone doesn't need multiple physical lines |
20:14.20 | mrgoby | i just googled it and only found a message from Uriel asking you ( it said Jeremy) the same thing |
20:14.32 | mrgoby | :-D |
20:14.39 | Marlow | JerJer : but as far as i see, you write the config from the table to a static file here .. |
20:14.48 | Marlow | JerJer : and reload asterisk .. |
20:14.54 | JerJer | and the problem is? |
20:15.12 | Muckl | is the name in the [brackets] important for the registration of a SIP client or just for internal use in the extensions.conf? |
20:15.21 | Marlow | JerJer : tjaeh .. not a big one .. but it ain't nice .. |
20:15.37 | JerJer | ?! we send multi-megabit text config files at our class 4/5 switches |
20:15.44 | JerJer | and issue a reload command |
20:15.50 | os2doc | mishehu: yes, that seems the only option. It seems that my phone system if I use asterisk is going to be analog phones with my employees having to punch in the codes, with the receptionist with the large phone, or maybe a computer terminal phone. Thats a though. |
20:15.51 | Marlow | JerJer : what do you have against using the sql table directly ? |
20:16.01 | *** join/#asterisk pawpa (~bob@216.253.86.210) |
20:16.04 | bkw_ | Marlow stability |
20:16.08 | pawpa | hi |
20:16.13 | JerJer | what happens WHEN your db goes down? |
20:16.14 | bkw_ | why put more chances of instability into * |
20:16.20 | JerJer | not if |
20:16.23 | JerJer | WHEN |
20:16.33 | mrgoby | JerJer: nevermind i think i found it |
20:17.19 | Marlow | i know, that it can be an issue .. |
20:17.33 | Marlow | so the question is creating the possibilty of an fallback ... |
20:17.52 | Marlow | s/an/a/ |
20:18.11 | os2doc | mishehu: an IP phone doesn't need multiple lines running to it, but what about access to multiple lines, like all 6 incoming pots lines, intercom, etc, without having to type in codes. |
20:18.24 | JerJer | Marlow: it is all in proper design of the system |
20:18.28 | mrgoby | JerJer: is that the only reason for doing it the other way?? just because MySQL is not dependable to be running always? |
20:18.42 | JerJer | NuFone runs on a totally databased method of config and we have zero trouble |
20:18.51 | espenz | Marlow: hvordan fungerer det når jeg får telefonen? Setter den opp med IP, så den treffer burken min.. Men skal jeg bruke asterisk som SIP-gw på en måte? |
20:18.55 | mrgoby | i understand that point, don't get me wrong |
20:18.57 | mishehu | os2doc: not sure about a good phone yet... I'm waiting for a cisco 7960 to arrive |
20:18.58 | JerJer | mrgoby: 01[15:14] <bkw_> why put more chances of instability into * |
20:19.06 | espenz | Slik at andre SIP telefoner kan tryke f.eks "2" for å komme til en telefon |
20:19.13 | Marlow | espenz : jep .. det er helt rigtigt .. |
20:19.14 | torment0r|wk| | me no hablo espano |
20:19.17 | pawpa | Jerjer: i thought you said not to use a DB because you are at the mercy of it's availability |
20:19.18 | mrgoby | i see |
20:19.18 | espenz | (må seff) sette opp det også |
20:19.26 | Marlow | espenz : fungerer på samme måde, som med en softphone .. |
20:19.28 | espenz | Marlow: men er det instillinger for SIP gw? |
20:19.28 | JerJer | pawpa: that's not what i said |
20:19.31 | espenz | på telefonen |
20:19.36 | kapejod | oh, the vikings again ;) |
20:19.40 | espenz | burde vel være det |
20:19.41 | espenz | ;P |
20:19.41 | torment0r|wk| | hola? |
20:20.09 | *** join/#asterisk okrumm (~okrumm@dsl-200-95-104-110.prodigy.net.mx) |
20:20.15 | pawpa | jerjer: i thought you were saying that it isn't a good idea to use a DB (mysql) because you have to maintain an extra server |
20:20.28 | JerJer | um no...where'd u read that? |
20:20.32 | espenz | Marlow: har du satt opp noe opplegg hjemme som funker? |
20:20.48 | pawpa | hemm let's see |
20:21.00 | Marlow | espenz : jeg har ikke en hardphone, men jeg har * med softphone kørende .. |
20:21.01 | okrumm | Anyone has a Windows SIP phone working? |
20:21.07 | JerJer | wana bring a Cisco Call Manager to its knees: stop the MSSQL service |
20:21.14 | espenz | Marlow: kan jeg få teste? |
20:21.15 | Marlow | kapejod : wir können auch anders .. |
20:21.17 | jsmith | JerJer: Yup... :-) |
20:21.20 | JerJer | or simply DoS it |
20:21.21 | Marlow | espenz : klart .. |
20:21.24 | espenz | tenkte på innringer nr med routing til softphone? |
20:21.27 | mishehu | okrumm: a number of ppl have gotten xlite to work. |
20:21.35 | jsmith | JerJer: Lemme guess... it install MSSQL with a blank "sa" password |
20:21.41 | torment0r|wk| | is there anything that the cisco call manager does that asterisk can't do? |
20:21.46 | pawpa | jerjer: i have heard that cdr_mysql does not build correct CDR's...true? |
20:21.51 | kapejod | Marlow: lol |
20:21.58 | jsmith | torment0r|wk|: Cost tons of money? Crash often? Run on windows? |
20:22.05 | torment0r|wk| | well i know that |
20:22.10 | jsmith | :-) |
20:22.10 | JerJer | pawpa: absolutely...incomplete and implemented poorly |
20:22.11 | torment0r|wk| | but.. i've never seen one in action |
20:22.27 | JerJer | torment0r|wk|: you don't want to |
20:22.45 | torment0r|wk| | is it like compairing IIS to Apache kinda thing |
20:22.50 | pawpa | jerjer: are there modules available that generate correct CDRs or is the some thing i need to write? |
20:22.56 | bkw_ | I love this |
20:22.56 | bkw_ | asterisk*CLI> show uptime |
20:22.57 | bkw_ | System uptime: 3 weeks, 22 hours, 35 minutes, 3 seconds |
20:22.57 | bkw_ | Last reload: 2 weeks, 1 hour, 19 minutes, 29 seconds |
20:23.03 | bkw_ | not bad eh? |
20:23.09 | bkw_ | thats the production one |
20:23.27 | JerJer | pawpa: i have written my own entire real-time rating engine, not just a cdr inserter |
20:23.31 | mishehu | bkw_: uhm... aren't those supposed to match? ;-) |
20:23.39 | bkw_ | mishehu nope |
20:23.41 | JerJer | mishehu: nope |
20:24.01 | pawpa | jerjer: reasonable undertaking or daunting? |
20:24.02 | Marlow | du mener sip ? |
20:24.29 | mishehu | *CLI> show uptime |
20:24.30 | mishehu | System uptime: 2 weeks, 3 days, 11 hours, 40 minutes, 57 seconds |
20:24.54 | HeeD | yup works now.... CVS-11/21/03-12:08:31 |
20:24.54 | mishehu | not bad for mine, considering the fact that I keep changing shit in it. ;-) |
20:24.57 | bkw_ | mishehu do a reload and it will show las realod |
20:25.19 | JerJer | good lordy... two small charectors in sched.c blew up everythign :) |
20:25.43 | JerJer | - ast_mutex_lock(&con->lock); |
20:25.43 | JerJer | + ast_mutex_unlock(&con->lock); |
20:25.47 | *** join/#asterisk TK (~sudsboy@53.int43.dsl.garlic.net) |
20:26.50 | pawpa | jerjer: never heard of a AAA oriented object called a rating engine...where does the definition come from? |
20:26.51 | bkw_ | JerJer thats usually the case |
20:26.54 | mrgoby | JerJer: so do these implementations of the dynamic configurations and extensions that you referenced, do they put the configuration into a flat conf file and then just reload? |
20:27.01 | mishehu | garlic.net.... the dsl provider that smells bad |
20:27.18 | bkw_ | RADIUS... let the fight begine! |
20:27.20 | bkw_ | er begin! |
20:27.29 | bkw_ | Radius Kombat! |
20:27.39 | bkw_ | oh nice bug in radius today.. did you guys see it? |
20:27.51 | Marlow | bkw_ : radius aint bad .. |
20:27.54 | Exomorph_ | JerJer: THATS MY BUG! THATS MY BUG! Sweet... Now I won't have to reboot the linux box agian! :) |
20:27.55 | mrgoby | i find adding dynamicism to * very intriguing |
20:28.30 | JerJer | pawpa: don't sware at me |
20:28.46 | pawpa | huh? |
20:28.57 | JerJer | RADIUS will never be implemented in ANY VoIP operaiton I have control over |
20:29.43 | Connor- | I still don't see what you all have against radius |
20:29.44 | Marlow | no .. but not everybody shares that point .. |
20:30.07 | pawpa | jerjer: i can see plainly how authentication and authorization would be simple, but how do you send the time elapsed to the script after the hangup |
20:30.14 | Marlow | JerJer : nobody forces you .. |
20:30.33 | JerJer | why add another layer of potential failure? |
20:30.50 | JerJer | there are many different methods to talk to a databas without involving the complexitities of RADIUS |
20:31.14 | Marlow | JerJer: because you can add scalability ... and radius is implemented in many places allready .. |
20:31.21 | Marlow | JerJer: use of existing ressources .. |
20:31.45 | Connor- | That's all well and good.. Radius makes cross platform easier too. You ever try to talk to a MSSQL from linux? |
20:31.50 | JerJer | and as you scale your operation not only do you have to worry abuot keeping your database online (and available) you have to also worry about your RADIUS server(s). |
20:32.15 | Marlow | JerJer: who says, that i talk to a database server behind radius ? |
20:32.21 | HeeD | what I done movin my /usr/src/asterisk dir to a backup, will that affect my g.729 licenses in any way... asterisk says I still licensed |
20:32.29 | JerJer | Marlow: by default... because some lazy Cisco developer implemented it into their boxes |
20:32.29 | sizzzung | Connor-: have you ever tried to talk to mssql from linux? |
20:32.33 | sizzzung | Connor-: it seems not |
20:32.39 | JerJer | sizzzung: simple |
20:32.40 | Marlow | JerJer: most radius work from plain textfiles, but you would have a central repository for multiple boxes .. |
20:32.41 | sizzzung | Connor-: because you can do it. quite easily. |
20:32.44 | sizzzung | Connor-: FreeTDS |
20:32.54 | Marlow | JerJer: cisco is not the default .. |
20:32.54 | pawpa | freetds? |
20:32.56 | HeeD | ie. does the Registration do somethin ... cuz they say it has to be run from /usr/src/asterisk initially |
20:32.56 | JerJer | Marlow: like that will scale |
20:32.59 | Connor- | Last time I tried it was a pain in the arse |
20:33.17 | sizzzung | i was doing it 3 years ago easily. |
20:33.31 | Marlow | JerJer: nope ... it will not .. but you are not required to talk to a database .. there are other possibilities .. |
20:33.52 | sizzzung | This point release fixes some memory leaks. (3 August 2003) |
20:33.57 | kapejod | flat files!!! ;) |
20:33.58 | sizzzung | i guess they got it to the point where it's perfect! |
20:34.04 | pawpa | if not radius, how can you pass accounting infor (ie time elased) back your accounting script for the sql insert |
20:34.15 | JerJer | pawpa: there is no script |
20:34.29 | JerJer | Asterisk has built in facilities for dealing with accounting |
20:34.30 | Marlow | why shouldn't anybody have the possibility to choose for themselves ? .. |
20:34.34 | sizzzung | hrmm |
20:34.36 | sizzzung | damn it |
20:34.38 | pawpa | the acct flag? |
20:34.42 | sizzzung | when will my T1 be in? |
20:34.57 | JerJer | Marlow: by adding all the complexities of RADIUS? i think not |
20:35.02 | JerJer | cdr_csv.so |
20:35.05 | JerJer | there is your flat file |
20:35.12 | pawpa | ok |
20:35.14 | kapejod | yeeehaaa ;) |
20:35.16 | JerJer | even seperated out by account code |
20:35.17 | Connor- | JerJer too bad those facilites doen't talk to my accounting system, which DOES talk to radius. |
20:35.28 | Marlow | JerJer: nobody says, that you will have to use radius .. others allready base their business on it .. |
20:35.31 | pawpa | ok...politically correct CDRs? |
20:35.44 | *** join/#asterisk point (~litw@195.161.106.222) |
20:35.56 | JerJer | Marlow: time will tell how smart of a business decsion that will be |
20:36.12 | sizzzung | heh |
20:36.22 | Marlow | JerJer: might .. might not .. |
20:36.24 | JerJer | i've been there and seen how RADIUS fails with VoIP |
20:36.30 | Marlow | JerJer: it's still the freedom of choice .. |
20:36.33 | sizzzung | how many real telecom companies uses RADIUS for accounting? |
20:36.39 | sizzzung | Marlow: it's a freedom to make dumb decisions. |
20:36.46 | JerJer | sizzzung: right on |
20:36.56 | Connor- | How many ISP's use it for Dialup, DSL and other things? |
20:36.57 | mrgoby | OUCH |
20:36.58 | sizzzung | JerJer: high five, one owner to another. |
20:37.01 | Marlow | why should it be a dumb decision, when it has been used for account for years .. |
20:37.10 | sizzzung | Marlow: obviously you're not pushing enough traffic. |
20:37.16 | JerJer | Connor-: which is EXACTLY what radius was designed for |
20:37.20 | sizzzung | wait till you're pushing millions of CDRs a month. |
20:37.34 | sizzzung | you're going to use RADIUS for that? |
20:37.36 | JerJer | sizzzung: or even a few hundered a second |
20:37.39 | Marlow | for modem-dialup |
20:37.39 | Marlow | where is the difference ? |
20:37.45 | Connor- | Let see, I track dialup time via radius, hmm.. What's the damn difference? |
20:37.50 | JerJer | a whole hell of a lot |
20:37.50 | bkw_ | we do not use radius for dialu |
20:37.50 | Marlow | it simply depends on what setup you want to archieve .. |
20:37.50 | sizzzung | Marlow: there are a lot more records |
20:37.50 | bkw_ | p |
20:37.54 | sizzzung | you guys have no fucking experience |
20:37.56 | bkw_ | er DSL |
20:37.58 | sizzzung | jesus fucking christ |
20:38.01 | sizzzung | i swear to god |
20:38.01 | bkw_ | we use radius for dialup.. but not DSL |
20:38.09 | pawpa | whoa |
20:38.16 | pawpa | i believe you |
20:38.16 | pawpa | :) |
20:38.18 | sizzzung | most of you are fucking nerds sitting in your mom's basement trying to start a carrier with a linux box and PRI |
20:38.22 | JerJer | sizzzung: i feel your pain |
20:38.24 | pawpa | hahaha |
20:38.26 | pawpa | hah |
20:38.26 | pawpa | ha |
20:38.32 | Marlow | sizzzung : no . we have other demands .. |
20:38.34 | sizzzung | note, i said most, and not all. |
20:38.38 | pawpa | hah |
20:38.38 | pawpa | hya |
20:38.42 | pawpa | rotfl |
20:38.43 | sizzzung | Marlow: demands like pushing CDR's to a RADIUS box? |
20:39.08 | sizzzung | so much lack of clue. |
20:39.13 | kapejod | sizzzung: i dont even have a pri :( |
20:39.16 | Connor- | I'm not really wanting it for CDR's as much as replacing sip.conf using radius attributes |
20:39.18 | Marlow | sizzzung : that would be for integration in the current billing system, instead of developing a new one .. |
20:39.23 | JerJer | RADIUS is for sheep |
20:39.33 | sizzzung | Marlow: then your current billing system sucks. rewrite it, or buy mine. |
20:39.46 | JerJer | Connor-: you most certianly do not need radius to do that |
20:39.47 | pawpa | sizzung: i will HAPPILY peruse any documentation on an alternate setup for voip AAA...can you direct me? |
20:40.01 | pawpa | sizzzung: how much for your solution? |
20:40.07 | JerJer | pawpa: it doesn't exist in Asterisk |
20:40.08 | sizzzung | haven't decided on a price. |
20:40.15 | sizzzung | i'm thinking $125-$200k |
20:40.26 | Connor- | No, But sinze my freaking accounting package can handle that, it makes things damn easy. |
20:40.26 | sizzzung | mattering how much more mature it gets in the next two months |
20:40.35 | Marlow | i'm not saying radius is the solution .. but it is one way to go .. especially because many base their business on it allready .. |
20:40.42 | pawpa | jerjer: so i must roll the solution in house, eh? |
20:40.42 | sizzzung | okay. do this. |
20:40.51 | JerJer | pawpa: not necessarily |
20:40.56 | sizzzung | create a perl pre-parser that'll parse the csv cdrs |
20:40.58 | Connor- | granted, I probably could talk directly to the SQL server, reading those same damn attrbitues |
20:41.02 | sizzzung | or runs through the sql server |
20:41.02 | ManxPower | Does anyone know what message waiting indication Zaptel analog FXS cards use? FXS, 90v, 48v? |
20:41.07 | sizzzung | generate the summaries |
20:41.11 | sizzzung | and drop the summaries into RADIUS |
20:41.15 | sizzzung | there are plenty of radius perl modules. |
20:41.16 | point | JerJer, I have fixed my main problem with chan_h323 ... only after installation of new ethereal :) - the option noh245tunneling does not work for outgoing calls at last without gk .. |
20:41.20 | Marlow | sizzzung : a billing system is not easy replaced ... if it's integrated with other stuff .. |
20:41.52 | JerJer | Marlow: who is saying replace anything? he just gave you the way to hook asterisk into your existing crap |
20:41.58 | Connor- | btw, I WANT TO KNOW. WTF do you think RADIUS FAILS with VoIP ?? |
20:42.06 | sizzzung | Connor-: TOO MUCH TRAFFIC! |
20:42.16 | JerJer | Vendor Specific Attributes |
20:42.19 | Marlow | JerJer : i had a lag here .. got the last 15-20 messages in a bunch |
20:42.24 | JerJer | improperly impleemnted records |
20:42.32 | Marlow | JerJer: after i wrote .. |
20:42.36 | sizzzung | i'm just blaming traffic. |
20:42.45 | Connor- | Bullshit! |
20:43.06 | Connor- | If your damn radius can't handle the traffic, then it's not scalled correctly or done correctly. |
20:43.19 | kapejod | or spelled correctly |
20:43.35 | JerJer | Connor-: so your saying buy two high end servers instead of one? |
20:43.49 | JerJer | for each time you need to scale up |
20:43.59 | JerJer | where is the financial sense there? |
20:44.02 | Connor- | JerJer, You better have 2 if you want to have failover/reduntant server. |
20:44.12 | JerJer | lol no..doesn't work like that |
20:44.17 | ManxPower | I figure that is RADIUS support was so important someone would have written a RADIUS logger module for Asterisk already. |
20:44.17 | dant | but if 1 can't handle the load, then you need 3 |
20:44.19 | kapejod | but you dont want to have 4 |
20:44.31 | pawpa | sizzzung: using radius on the back end for it's accounting abilities? |
20:44.40 | pawpa | sizzung: jerjer: > what about prepaid? |
20:44.51 | JerJer | pawpa: we do pre-paid all day long |
20:44.54 | pawpa | do you use the cdr_csv the same way?> |
20:44.58 | JerJer | no |
20:45.01 | ]data[_ | cdr_flow 0wnz me |
20:45.02 | ]data[_ | :] |
20:45.34 | Connor- | look, the main reason, like I said, that I want radius, is for provisioning new VoIP customers. |
20:45.50 | JerJer | Connor-: and i'm asking why? |
20:45.59 | JerJer | its a blantent waste of resources |
20:46.02 | ]data[_ | the whole configfile structure of asterisk is a mess imo :-) |
20:46.02 | ManxPower | Does *does* RADIUS send a "disconnect" message when a dialup user has exceeded their monthly allowance? |
20:46.12 | Connor- | Because, My freaking billing/accounting solution makes it easy to do. |
20:46.21 | ManxPower | Seems to me that's close to pre-paid voiuce. |
20:46.23 | torment0r|wk| | there's really no point in using radius for VoIP.. |
20:46.41 | ]data[_ | there are entire radius billing platforms out there you can buy off the shelf |
20:46.41 | ManxPower | torment0r|wk|, There are lots of reasons to use RADIUS for VoIP billing. 8-) |
20:46.43 | ]data[_ | thats the point |
20:46.48 | JerJer | Connor: so your solution can set specific codecs on a per user/peer basis? |
20:46.48 | *** join/#asterisk tim27 (doug25@229-29.dr.cgocable.ca) |
20:46.56 | tim27 | hello everyone :) |
20:46.58 | torment0r|wk| | ManxPower, it's not accurate |
20:46.59 | pawpa | jerjer: sizzung's recommendations are very straightforward, but you mentioned that I wouldn't have to roll the solution myself...were you meaning that i wouldn't have to write it completely because there are some helper modules available? |
20:47.10 | rusty_ | jerjer - what are the readily available alternatives to radius, then? |
20:47.17 | Marlow | torment0r|wk| : wrong .. Connor- gave you one .. |
20:47.26 | torment0r|wk| | i work for an isp.. trust me.. my time and the time of Qwest never matches |
20:47.29 | ManxPower | torment0r|wk|, *shrug* That's not my problem, that the problem of someone that wants to use RADIUS. |
20:47.48 | torment0r|wk| | and it always leads to problems |
20:47.49 | JerJer | torment0r|wk|: sync to a stratum 2 time source |
20:47.51 | Connor- | JerJer: I can set any damn attribute I want via radius Attributes as long as I have them defined for Vender specific. |
20:48.02 | Exomorph_ | wb karm |
20:48.05 | JerJer | Connor: i think not |
20:48.21 | Connor- | Oh? |
20:48.23 | mrgoby | pawpaw: that is a good question Jer??? |
20:48.24 | ManxPower | kram: What message waiting indication does Zaptel analog FXS cards use? FXS, 90v, 48v? |
20:48.45 | rusty_ | JerJer: what are the readily available alternatives to RADIUS? |
20:48.51 | JerJer | rusty_: i've let bkw_ play with a first public version of my nufone_rating.so |
20:48.55 | tim27 | i read on the web that many people get echo problem when mixing FXO phone line and ip phone, any know solution to this prob... (some phone better than other ??? ) and there is a way to get a isdn for SOHO (BRI) ... |
20:49.11 | *** join/#asterisk Marlow_ (marlow@as2-6-3.tbg.s.bonet.se) |
20:49.16 | pawpa | jerjer: how much for your rating engine? |
20:49.17 | rusty_ | JerJer: last I heard that was being kept a secret |
20:49.38 | rusty_ | JerJer: and it sounds like it's going to be a commercial release (as opposed to OS)? |
20:50.05 | JerJer | rusty_: if your rating calls that means you are going to be billing customers |
20:50.11 | JerJer | and if your billing customers you are making money |
20:50.22 | rusty_ | JerJer: and? |
20:50.27 | pawpa | jerjer: i am interested in your solution, but see no info on nufone |
20:50.35 | bkw_ | nufone_rating.so rocks... very nice |
20:50.43 | Connor- | JerJer: Why do you say I can't added custom Radius attributes for codec's? |
20:50.48 | *** join/#asterisk ToyMan (~stuq@user-0cevdks.cable.mindspring.com) |
20:51.08 | JerJer | I have worked out a non-gpl license with Digum, which will help with the continued the development of Asterisk |
20:51.18 | mrgoby | yes... i'd be interested too jer |
20:51.22 | pawpa | so licensing is through digium |
20:51.23 | pawpa | ? |
20:51.29 | rusty_ | JerJer: so it's going to be a commercial license? |
20:51.48 | JerJer | Connor-: there is only so much u can do with a VSA |
20:52.20 | JerJer | why you think those billing software packages out there cost so fsckin much? |
20:52.20 | defian | cu ) |
20:52.26 | pbxtech | its everywhere you want to be |
20:52.32 | tim27 | any got a clue :) |
20:52.34 | rusty_ | JerJer: because they can |
20:52.40 | rusty_ | JerJer: it's a niche market |
20:52.48 | pbxtech | there are open source billing software |
20:52.52 | JerJer | lolololol |
20:52.57 | JerJer | pbxtech: ok |
20:53.00 | pawpa | what about freeside? |
20:53.09 | *** join/#asterisk stonefly (~trillian@toby.stoneflytech.com) |
20:53.10 | JerJer | or tarbas |
20:53.26 | pawpa | doesn't seem like trabas even has any SW there |
20:53.56 | pawpa | if you are dropping cdrs into radius via perl, couldn't you use freeside to maintain your accounts? |
20:53.59 | rusty_ | JerJer: so I'm still not clear on what kind of licensing you intend to have for your rating engine |
20:54.11 | point | who use trabas ? |
20:54.24 | torment0r|wk| | what exactly does your rating engine do.. just keep track of billing? |
20:54.38 | ]data[_ | it rates the calls :) |
20:54.39 | bkw_ | torment0r|wk| you know that bill you get from the phone comany? |
20:54.40 | JerJer | rusty_: non-gpl...you will recieve the shared object an example config and some support |
20:54.49 | tim27 | anyknow about echo ... ??? |
20:54.50 | bkw_ | with all the calls list and the prices.. you need a rating engine to do that |
20:54.51 | rusty_ | JerJer: cost? |
20:54.59 | Connor- | JerJer: I see nothing in sip.conf that couldn't be done with a VSA... |
20:55.03 | *** join/#asterisk FryGuy (~fryguy@c-24-2-50-122.client.comcast.net) |
20:55.07 | JerJer | Connor-: then do it |
20:55.13 | torment0r|wk| | bkw_, someone couldn't do that with the cdr_myql db |
20:55.18 | ]data[_ | i'm writing my own one for fun too btw :) |
20:55.19 | mrgoby | what about echo tim? |
20:55.44 | bkw_ | torment0r|wk| if you think you can rely on that.. go for it |
20:55.48 | torment0r|wk| | i'd be interested in it though |
20:55.49 | JerJer | Connor: and when you hit that brick wall you'll wish you would have listened to someone that has been there already |
20:55.55 | rusty_ | JerJer: doesn't nuphone use asterisk? |
20:55.57 | tim27 | i want to purchase equipment, is that true that i will get echo on SIP phone trought FXO phone line ??? |
20:55.58 | bkw_ | torment0r|wk| but you can't do prepaid.. and such with it. |
20:56.05 | JerJer | nufone is 100% asterisk |
20:56.06 | dant | JerJer, can I have the rating thing for free so I can bill the gf?? ;) |
20:56.18 | rusty_ | JerJer: gpl software |
20:56.22 | pawpa | sizzzung: are you using off the shelf solution for billing once the cdrs are in radius? |
20:56.26 | tim27 | on the webforum, many people complain about echo |
20:56.31 | JerJer | codec_g729 is not gpl |
20:56.39 | rusty_ | JerJer: right...so? |
20:57.04 | JerJer | so what? |
20:57.20 | *** join/#asterisk Padre (Padre@67.60.50.146) |
20:57.33 | mrgoby | you are saying that your system just works with open source, and does not contain open source, right Jer? |
20:57.55 | mrgoby | or GPL rather |
20:57.58 | bkw_ | You can do prepaid. postpaid.. any paid.. in any currency with nufone_rating.so |
20:58.14 | pawpa | how much is it? |
20:58.16 | JerJer | mrgoby: nufone and digium have worked out a non-gpl license... |
20:58.23 | pawpa | is digium licensing it at themoment? |
20:58.25 | rusty_ | mrgoby: yet his entire business is run on OS software (except for the codecs) |
20:58.44 | rusty_ | not just OS, GPL |
20:58.46 | pawpa | sounds like an efficient model to me |
20:59.02 | mrgoby | sure |
20:59.06 | pawpa | you can't feel bad about making money with OS |
20:59.08 | *** join/#asterisk sleepy_ (~sleepy@80.96.155.10) |
20:59.12 | mrgoby | no |
20:59.12 | pawpa | that's what itis there for |
20:59.23 | pawpa | i'm sure jerjer contributes in many ways |
20:59.40 | pawpa | answering 5 * 10^1200 questions a day is one way |
20:59.43 | bkw_ | um |
20:59.49 | mrgoby | Pirates, all a ya !!! pinko commies !!!! |
20:59.49 | bkw_ | chan_sip keeps losing my sip registrations |
20:59.51 | bkw_ | strange |
20:59.54 | JerJer | rusty_: show me the paragraph that says I can't make money using GPL |
21:00.10 | sleepy_ | hello all |
21:00.17 | Connor- | bkw_ you sure it isn't GS firmware? Had that problem already |
21:00.19 | mrgoby | howdy sleepster |
21:00.22 | dant | JerJer, I'm sure Microsoft have some papers on it ;) |
21:00.25 | rusty_ | JerJer: I realize that you can...show the paragraph that says I can't critize your method of making money |
21:00.27 | pawpa | jerjer: i'll probably be licensing your engine until i can get my own solution ready |
21:00.30 | tim27 | mrgoby, they said :Some changes were implemented around Oct 27th that improved the echo |
21:00.30 | tim27 | problem. To use that, put "echotraining=yes" in zapata.conf for each |
21:00.30 | tim27 | x100p card installed. As I understand this, it changed the way the |
21:00.30 | tim27 | echo canceller samples the analog line. Using the parameter is far |
21:00.30 | tim27 | better then previous, but there can still be some echo involved in |
21:00.31 | tim27 | some cases. |
21:00.46 | bkw_ | Connor- i'm talkin about my Cisco |
21:00.53 | kapejod | actually JerJer could have done it also without paying for a non-gpl license from digium |
21:01.15 | mrgoby | anyone wanna help tim27 ??? not I said the fly |
21:01.16 | bkw_ | kapejod how? |
21:01.20 | bkw_ | kapejod do tell! |
21:01.26 | Connor- | bkw_ not good |
21:01.26 | kapejod | bkw_: read the gpl |
21:01.32 | Exomorph_ | JerJer: Ok... New problem with the lastest cvs... When it does a restart on the B-Channel's, (pri's) asterisk hangs. |
21:01.38 | rusty_ | I'm sure that jerjer's engine will be very useful for some people |
21:01.44 | torment0r|wk| | JerJer, happen to have any documentation on exactly what it does.. i'd be interested to get something like that |
21:01.51 | sleepy_ | i tried to install asterisk on my computer with my gvc external modem and it can't be recognized |
21:01.54 | stonefly | tim27, does echotrainning=yes apply to T100p cards hooked up to a channel bank too? |
21:02.02 | Marlow | the GPL is not about free beer, it is about the freedom of Source .. |
21:02.02 | sleepy_ | can someone help me? |
21:02.12 | JerJer | torment0r|wk|: no, i'm still working on the code still... as i get feedback from the few ppl that are currently using it |
21:02.16 | Padre | Free BEER? HUH HUH! |
21:02.23 | Padre | Who's buying? |
21:02.28 | many | beer |
21:02.29 | ManxPower | sleepy_, That is correct. You will have hack the asterisk source code to make it work with a voicemodem. Even then it won't work very well |
21:02.30 | many | good one. |
21:02.38 | many | its 10pm, good time for a beer. |
21:02.46 | pawpa | 2 pm, here |
21:02.47 | Marlow | i've got mine allready .. |
21:02.51 | Marlow | before i came home .. |
21:02.52 | pawpa | still sounds like a good idea, though |
21:02.54 | rusty_ | But there is a constant schism between Qt vs GTK camps for the same reason. |
21:03.02 | rusty_ | GPL vs non-GPL |
21:03.06 | Padre | hey I am in the west coast only 1pm here time for a beer lunch brake |
21:03.09 | Marlow | rusty_ : nah .. that's another issue .. |
21:03.12 | Connor- | SO... If someone DID want to write a radius module for *, where would be good place to start? |
21:03.13 | torment0r|wk| | bbiab |
21:03.21 | Marlow | rusty_ : and has nothing to do with making money of source .. |
21:03.27 | rusty_ | Marlow - no, it's not. One's proprietary, and one isnt. |
21:03.30 | pawpa | someone is GPLing one at the end of the year |
21:03.31 | JerJer | Connor: good luck |
21:03.38 | Marlow | rusty_ : yes .. but it's not about money . |
21:03.38 | ManxPower | Connor: cdr_mysql or cdr_csv |
21:03.49 | Connor- | not talking for cdr's |
21:03.57 | pawpa | well, cdr_mysql is messed up |
21:03.59 | Connor- | for configing sip.conf |
21:04.02 | rusty_ | Marlow: I was never debating money - jerjer brought that up |
21:04.16 | ManxPower | Connor: oh. put it in ast_load then any of the config files could use it. |
21:04.33 | pawpa | ast_load? |
21:04.39 | rusty_ | Marlow: I was debating the principle. |
21:04.47 | ManxPower | ast_load is the function that loads and parses the asterisk config files. |
21:04.54 | Marlow | rusty_ : of course .. but that is his freedom to choose .. |
21:05.00 | ManxPower | at least I *think* it's called ast_load |
21:05.02 | pawpa | so put sip.conf in ast_load...don't follow |
21:05.04 | Marlow | rusty_ : as is yours to do it different .. |
21:05.12 | rusty_ | Marlow: yes, of course |
21:05.31 | *** join/#asterisk ssokol (~ssokol@64-151-38-185-dhcp-kc.everestkc.net) |
21:05.32 | Marlow | rusty_ : he has released the h323 chan driver, if i understood that right .. ... so he contributes .. |
21:05.36 | ManxPower | pawpa, Connor was asking where the best place to put RADIUS support was, I suggested he patch the ast_load function in the Asterisk source code. |
21:05.50 | pawpa | manx: sorry |
21:05.53 | Connor- | as there any sort of dynamic config file stuff * out yet? |
21:05.53 | Marlow | rusty_ : but nobody forces him to release everything, just because he uses OS |
21:06.02 | JerJer | can't forget about chan_skinny |
21:06.26 | ManxPower | Connor: No, but if you put RADIUS support in the asterisk config file parser.... |
21:06.54 | Connor- | Manx: That's the plan. |
21:07.01 | bkw_ | anyone else notice chan_sip just stops responding? |
21:07.13 | ManxPower | Of course then you would need to modify Asterisk to reload it's configs when some event happens |
21:07.17 | Exomorph_ | bkw: Me... I'm looking into it now. |
21:07.45 | pawpa | sizzung: you aroung? |
21:07.55 | Marlow | ManxPower : does dynextendb do that ? |
21:08.10 | Connor- | Manx: or setup a event schedule... is there a way to force a reload on individuale conf files? |
21:08.10 | Marlow | ManxPower : reload on any event ? |
21:08.10 | pawpa | i heard extendb is not the best way for dynamic extensions |
21:08.38 | ManxPower | Marlow, I have no idea. I am not aware of anyone using that external software |
21:08.53 | *** join/#asterisk jgaviria (~jgaviria@63.245.86.109) |
21:09.17 | pawpa | jerjer said that extendb was not the best way, but didn't mention a better way |
21:09.28 | JerJer | pawpa i certainaly did |
21:09.33 | pawpa | sorry |
21:09.37 | pawpa | what was that? |
21:09.41 | pawpa | missed that |
21:09.53 | Marlow | pawpa : yes .. but he didn't say way ... beyond that he doesn't think radius is the way either .. |
21:10.02 | ManxPower | I always thought "JerJer's way" was "modify the config files and issue reloads automatically" |
21:10.09 | JerJer | retrieve_extensions_from_mysql.pl |
21:10.17 | pawpa | tx!! |
21:10.19 | pawpa | you da man |
21:10.40 | pawpa | everybody: here this............retrieve_extensions_from_mysql.p is the way |
21:10.44 | pawpa | :^)) |
21:10.50 | pawpa | pawpa happy |
21:10.51 | rusty_ | I think radius will be fine for some installations - not everything has to scale to huge proportions. |
21:10.58 | JerJer | ManxPower: nufone runs on a database |
21:11.02 | JerJer | config |
21:11.25 | Marlow | JerJer: so when you add something .. when is it active ? |
21:11.34 | pawpa | immediately i'd guess |
21:11.42 | pawpa | it is dynamic |
21:11.47 | pawpa | ? |
21:11.57 | JerJer | dynamic enough |
21:12.06 | pawpa | heh |
21:12.07 | mrgoby | JerJer: what does that mean? |
21:12.12 | Marlow | does still not answer my question .. |
21:12.18 | ManxPower | JerJer, Yes, but the database writes the config files and then issues a reload doesn't it? |
21:12.27 | Connor- | So where IS retrieve_extensions_from_mysql.pl ? |
21:12.30 | mrgoby | is that how it works? |
21:12.37 | JerJer | ManxPower: not directly, no |
21:12.44 | pawpa | mrgoby: i believe itmeans you can modify and maintain your extensions via php or something and have dynamic control of them vs. a static file |
21:12.49 | mrgoby | yeah....asterisk.gnuinter.net/files/digium/asterisk-ng/retrieve_extensions_from_mysql.pl |
21:12.55 | dant | Connor, /usr/src/asterisk/retrieve_extensions_from_mysql.pl |
21:13.03 | ManxPower | ./asterisk/messages-expire.pl |
21:13.03 | JerJer | mrgoby: how about /path/to/asterisk/ |
21:13.03 | ManxPower | ./asterisk/retrieve_extensions_from_mysql.pl |
21:13.03 | ManxPower | ./asterisk/retrieve_sip_conf_from_mysql.pl |
21:13.03 | mrgoby | heheh |
21:13.27 | mrgoby | JerJer: sorry, google was closer |
21:13.29 | mrgoby | :-D |
21:13.35 | Marlow | JerJer : how often do you write the configfiles and reload ? |
21:13.48 | JerJer | Marlow: doesn't work like that |
21:13.57 | *** join/#asterisk bobman (~bobman@mube.psouth.net) |
21:13.59 | Marlow | JerJer: but ? |
21:14.10 | pawpa | jerjer: clarification needed....does retrieve_extensions load it once or lookit up like an AGI script...like extendb |
21:14.26 | JerJer | pawpa: its not a AGI script, no |
21:14.28 | ManxPower | pawpa, Try reading the file. |
21:14.36 | pawpa | ok |
21:14.37 | pawpa | good idea |
21:14.40 | pawpa | no, great idea |
21:14.42 | JerJer | yeah, do i smell a RTFM bomb again? |
21:14.49 | pawpa | damnit, guys |
21:14.51 | pawpa | don't do that |
21:14.52 | mrgoby | hehehe |
21:14.55 | pawpa | jeez |
21:14.58 | JerJer | last time i released one i brought down freenode |
21:14.59 | pawpa | i'm sorry |
21:15.06 | pawpa | i'm soorrryyyyyyyyyyy |
21:15.15 | mrgoby | grovel pawpaw |
21:15.23 | mrgoby | bow down like the newbie you are |
21:15.26 | pawpa | i just want to make money on GPL's software....i will RTFM again and again |
21:15.33 | *** join/#asterisk MamboKing (~mambo@66.207.107.50) |
21:15.45 | MamboKing | what's up with ma bell? |
21:15.56 | pawpa | halitosis |
21:15.58 | pawpa | sp? |
21:16.45 | MamboKing | anyways, anyone developed pageing via ADSI yet? |
21:16.52 | mrgoby | heheh kram changed the topic to that at my suggestion :-) i think it is funny |
21:17.05 | Connor- | What do those perl scripts do, make a new conf file everytime something is changed and issue a reload? |
21:17.07 | bkw_ | access-list 101 deny ip any 64.254.234.0 0.0.0.255 |
21:17.08 | bkw_ | access-list 101 deny ip any 65.49.50.0 0.0.0.255 |
21:17.10 | bkw_ | does tha tlook right? |
21:17.15 | pawpa | is there a repository of contrib files for asterisk |
21:17.27 | many | bkw: depends on whaddya want :-P |
21:17.30 | Connor- | bkw_ don't for get the permit ip any any at the end |
21:17.39 | bkw_ | Connor- um no |
21:17.44 | bkw_ | you never permit ip any any |
21:17.49 | bkw_ | you boob |
21:17.52 | pawpa | heh |
21:17.56 | bkw_ | you only permit things that are destin for your network |
21:18.15 | Connor- | bkw_ umm you do. Maybe not for what your doing.. But you do. |
21:18.27 | pawpa | u oh |
21:18.33 | bkw_ | no you don't you only allow your network in.. unless you like having traffic on your network that shouldn't be there. |
21:18.35 | dant | Connor, when doing logging? :) |
21:18.38 | pawpa | well what is it?...you do or you don't |
21:18.43 | bkw_ | anyway |
21:18.44 | bkw_ | next |
21:18.51 | pawpa | so you do? |
21:18.51 | bkw_ | I wanna block those IPs from my network |
21:18.57 | jensd | how would you reload asterisk as best way - through the manager interface or through shell? (in a perl script) |
21:19.08 | bkw_ | asterisk -r |
21:19.10 | bkw_ | type relaod at the cli |
21:19.16 | bkw_ | asterisk -rx reload |
21:19.16 | MamboKing | through the cli is better |
21:19.26 | Connor- | then you need to have access-list 101 permit ip any any at the end of it as well. |
21:19.40 | Connor- | the deny takes precedence, then the permit allows everything else. |
21:19.50 | jensd | why not through manager interface? (when there is this nice perl manager interface :-P) |
21:19.55 | Connor- | once it's denied, it breaks out of the access list. |
21:19.56 | bkw_ | Connor- yes I know this... |
21:20.16 | MamboKing | bkw_: do you know if kram has had a chance to look at that bug yet? (487) |
21:20.40 | bkw_ | not yet |
21:20.52 | MamboKing | awesome |
21:21.00 | MamboKing | kram: how's it going? |
21:21.25 | Connor- | Do I need to send in a disclaimer for that little patch I did? |
21:21.58 | kram | surviging at the moment |
21:22.15 | MamboKing | good to read |
21:22.24 | *** join/#asterisk Vco (~Vco@h24-76-11-125.wp.shawcable.net) |
21:22.25 | izo | anyone here has ETSI ISDN specification ? |
21:22.30 | rozo | hey kram, what kind of cam is that in your office? The one that's taking 180 degree pictures |
21:23.48 | MamboKing | kram: a colegue of mine was talking to you about enableing pageing from phone to phone via ADSI some time ago.. you had mentioned that it was possible but it would require development |
21:24.07 | MamboKing | what kind of cost do you think would be involved in such an undertaking? |
21:24.26 | jets | whats up gents |
21:25.22 | *** join/#asterisk voipguy (~voipguy@hoochie.digium.com) |
21:27.45 | JerJer | rozo: i think its homemade (if memory services me :) |
21:28.11 | kapejod | izo: www.etsi.org |
21:28.18 | MamboKing | anyone here done any poking around with ADSI? |
21:29.51 | izo | kapejod : i cant find it there :-( maybe you have downloaded ? |
21:30.20 | izo | kapejod: someone told me today that SUSPEND ISDN message is not available in PRI ETSI specification |
21:30.29 | kapejod | izo: click on "get a standard", sign up and download them for free |
21:30.47 | Marlow | kapejod : what is the hfcdummy for ? |
21:31.03 | rusty_ | JerJer: how would you quantify the difference between the way that RADIUS scales (or not), and the way that your rating engine would scale? |
21:31.09 | kapejod | izo: yes that makes sense |
21:31.15 | kapejod | Marlow: it generates zaptel timing out of a hfc-s pci a isdn card |
21:31.29 | JerJer | rusty_: my solution is intergrated into asterisk |
21:31.35 | izo | kapejod: well I've seen it working : |
21:31.56 | Marlow | kapejod : from the isdn network ? |
21:31.57 | rusty_ | JerJer: that doesn't tell me much. |
21:32.08 | JerJer | rusty_: and doesn't had a whole sepearate protocol worth of overhead just to rate calls and instert them into a DB |
21:32.08 | kapejod | izo: it depends on the switch i'd say |
21:32.25 | atacomm | hey jets, how goes it |
21:32.28 | pawpa | there is also a company that was listed on the voip-info list ( Buena Vista Communications ) and they have a nice cheap AAA system for asterisk, although they have told me that this release is not scalable...meant more for call center applications and whatnot...soho/smo |
21:32.29 | Marlow | kapejod : or does the chipset provide that ? |
21:32.32 | rusty_ | JerJer: how does that translate into actual numbers? |
21:32.36 | izo | kapejod: u r probably right thats why I want to check ETSI standards |
21:32.40 | kapejod | Marlow: no network needed, just the card |
21:33.05 | kapejod | izo: suspend is not a mandatory message |
21:33.27 | rusty_ | JerJer: I'm a PHB...help me understand why I'd choose your engine over a free RADIUS solution. |
21:34.03 | JerJer | rusty_: the radius solution isn't free |
21:34.03 | JerJer | thre is no radius hooks into asterisk |
21:34.12 | izo | JerJer there is |
21:34.15 | izo | but not published |
21:34.21 | izo | there was posts on mailingilst |
21:34.25 | pawpa | yes i saw that on google |
21:34.27 | JerJer | that was a troll |
21:34.30 | izo | naah |
21:34.33 | pawpa | troll? |
21:34.35 | JerJer | vaporware |
21:34.36 | izo | just contact those people |
21:34.41 | rusty_ | hell just grep the source |
21:34.41 | izo | and pay him a little |
21:34.43 | JerJer | no need |
21:35.01 | izo | JerJer did you try to contact them ? |
21:35.10 | rusty_ | JerJer: so you're saying that there's no way to use radius with asterisk? |
21:35.11 | JerJer | i have a little bird |
21:35.16 | JerJer | that told me |
21:35.18 | pawpa | btw, that same company with the AAA system has a radius solution rolling out soon |
21:35.22 | tzanger | so in an answering dialplan (s,1,Answer) what is the timeout it waits for a response... is that the DigitTimeout? |
21:35.23 | pawpa | but it isn't open |
21:35.45 | JerJer | looks like the AstGod has bestowed another fix into the cvs |
21:35.47 | Exomorph_ | bkw_: cvs update to fix your sip problem. |
21:35.57 | pawpa | but it is cheap |
21:36.11 | bkw_ | Exomorph_ just checking |
21:36.15 | bkw_ | Stealth_Man wake up |
21:36.30 | pawpa | they are the only company besides nufone that i've seen offereing these products |
21:37.00 | JerJer | um our solution isn't AAA garbage |
21:37.08 | mrgoby | pawpaw: who? |
21:37.19 | pawpa | buena vista communications |
21:37.26 | Marlow | what happened to the daily debian builds of the cvs ? |
21:37.27 | pawpa | well i've tried it out and it works well |
21:37.34 | pawpa | for the application that it targets |
21:37.40 | pawpa | VERY intuitive |
21:38.02 | pawpa | which is really nice for me, because the operators of our call centers don't have to know diddly |
21:38.17 | Connor- | what ths addi pawpa |
21:38.40 | pawpa | jerjer: like i mentioned earlier, though, they told me that it was not meant to scale...only for soho/smo call center type app...does prepaid, too |
21:38.48 | bevins | can I use irq 3 or 4 for pci slots with x110p and tdm400? I know these are for serial ports usually but....cat /proc/interrupts doesn't show these, so I guess they are not used. |
21:39.08 | bevins | asterisk -vvvvr |
21:39.16 | pawpa | and they offer hosted billing, too...which can have its advantages |
21:39.23 | JerJer | pawpa: why buy something that won't grow with you? |
21:39.49 | pawpa | jerjer: cost/performance (right tool for the job) i don't need scalability for my applications |
21:39.56 | JerJer | lol ok |
21:39.57 | rusty_ | JerJer: not everyone needs to grow in a way that requires the degree of scaling that you're talking about. |
21:40.04 | pawpa | jerjer: why do you say? |
21:40.13 | JerJer | famous last words |
21:40.14 | pawpa | jerjer: i'm only needing 24 phones at a time |
21:40.20 | JerJer | pawpa: today |
21:40.20 | bkw_ | Exomorph_ seems to fix it |
21:40.38 | pawpa | i'm not a carrier, jerjer, i'm a callcenter guy...that's what i do |
21:40.46 | JerJer | doesn't matter |
21:40.52 | bevins | can I use irq 3 or 4 for pci slots with x110p and tdm400? I know these are for serial ports usually but....cat /proc/interrupts doesn't show these, so I guess they are not used. |
21:40.58 | JerJer | what if u land a hell of a deal for 500 phones? |
21:41.01 | Marlow | JerJer : not everybody wants to grow endless .. |
21:41.04 | ManxPower | bevins, Try it and see. |
21:41.24 | ManxPower | bevins, Your motherboard will assign the IRQs to the cards. |
21:41.24 | Marlow | JerJer : if it doesn't benefit in the long run .. |
21:41.39 | ManxPower | bevins, IF your motherboard does not generate interrupts for devices that are disabled then in theory it should work |
21:41.50 | pawpa | jerjer: your point is very well taken, it is just that currently i don't have the capital to get into a more scalable solution and also, the BVC solution is working perfectly fo rme |
21:41.50 | sizzzung | yo jerjer. |
21:41.59 | Marlow | JerJer : callcenter business is hard .. and keeping people with work isn't easy .. |
21:42.05 | ManxPower | JerJer, Then he will pay for his shortsightedness 8-) |
21:42.07 | sizzzung | cellcenter business is a bitch. |
21:42.08 | bevins | ManxPower: thats what I am trying not to do because it assigns wcfxo and eth0 the same irq....irq sharing and it has locked up the channel a few times. |
21:42.17 | pawpa | jerjer: we are very happy with it...i don't even have to train anyone to use it becuase it is so intuitive |
21:42.18 | bkw_ | Specifically, the law contains: opt-out, authority for the FTC to set up a "Do-Not-SPAM" registry, criminal charges for fraudulent spam, including five years in prison, statutory damages of $2 million for violations, tripled to $6 million for intentional violations, unlimited damages for fraud and abuse. |
21:42.19 | sizzzung | you have to show up at their office with a gun to get them to pay their bill. |
21:42.20 | bkw_ | woot |
21:42.20 | JerJer | ManxPower: yep |
21:42.42 | ManxPower | bevins, Yup. Change the slot the card is in or figure out a way to make the motherboard assign a different IRQ for that slot. |
21:42.44 | pawpa | suzzzung: we've never had any problems with call center apps |
21:42.49 | JerJer | so much for sleeping today |
21:43.01 | sizzzung | pawpa: what kind of call center? |
21:43.07 | ManxPower | bkw_, It'll never pass. |
21:43.10 | pawpa | pre and post paid and internet cafe |
21:43.28 | sizzzung | ahh |
21:43.36 | pawpa | we were using quintum |
21:43.45 | bevins | ManxPower: I have another box that put wxfxs,wxfxo,eth0 on the same irq....That doesn't work too well |
21:44.06 | pawpa | but have switched to asterisk becuase of the cheaper entry costs and the BVC AAA system fits nicely with our needs |
21:44.12 | ManxPower | bevins, IRQ sharing is not supported with Digium cards |
21:44.19 | Marlow | i've been working for callcenters starting from 10 agents to over 300 now . |
21:44.37 | bevins | ManxPower: I know....thats what I have to tell my motherboard...:-) |
21:44.51 | pawpa | marlow: wow |
21:45.11 | ManxPower | bevins, on the motherboards I've used you tell the system to not auto assign IRQs and then it will let me spcify per slot IRQ assignments |
21:45.32 | pawpa | another reason i like the BVC is because they will do hosted billing and handle customer accounts, too |
21:45.33 | Marlow | and it's allways a bitch |
21:45.51 | bevins | ManxPower: thats why I wanted to know about irq 3 and 4..I will just try it. |
21:45.56 | pawpa | which is great for ease of setting up a new call-center |
21:46.10 | pawpa | they will ship a unit preconfigged |
21:46.16 | pawpa | and we just plug it in and go |
21:46.18 | ManxPower | bevins, I don't think 3 and 4 are "officially supported" but I suspect it will work. |
21:46.21 | *** join/#asterisk zwi (~chris@216.88.131.43) |
21:46.57 | Marlow | not only is it a bitchy job .. you also allways run into a M1 |
21:47.06 | bevins | as long as I don't run setserial on those tty's, even better disable onboard serial ports... |
21:47.13 | JerJer | pawpa: hosted billing is nothing special... hell we host asterisk based billing for a few of our highly motivated customers |
21:47.33 | *** join/#asterisk glLoadIdentity (~asdfrt@abn139-91.interaktif.net.tr) |
21:47.54 | ManxPower | I have a question! /etc/zaptel.conf what signaling should I use for a PRI? |
21:48.13 | pawpa | jerjer: i know it is nothing special i was just saying that I like BVC and that it is a service they offer |
21:48.16 | ManxPower | pri_cpe is only listed in /etc/asterisk/zapata.conf not in /etc/zaptel.conf |
21:48.25 | pawpa | jerjer: it is nice for my situation |
21:48.34 | pawpa | jerjer: and they are very reasonable |
21:48.35 | tzanger | I have 12 lines in a trunk group... how do I dial that group? |
21:48.44 | tzanger | (i.e. so it picks up any available in the group) |
21:49.34 | JerJer | pawpa: then run with it |
21:49.39 | bkw_ | lalllalalal |
21:50.38 | *** join/#asterisk anachron (~fradooj@adsl-64-169-89-27.dsl.snfc21.pacbell.net) |
21:52.23 | Marlow | bkw_ : what have you eaten ? |
21:52.39 | jets | 801 805 6034 |
21:52.45 | bkw_ | Marlow just being me |
21:53.28 | jsharp | Anyone available to help me with some PRI problems? |
21:53.30 | zeta_ | jets, eh? |
21:53.31 | Marlow | jets : and ? what happens if i dial that no. ? :o) |
21:53.54 | jets | you get tt-monkeys |
21:53.56 | jets | its awesome |
21:54.18 | tzanger | what is the difference between 'group' and 'callgroup' in zapata.conf? |
21:54.20 | zeta_ | heh |
21:54.46 | zeta_ | Does anyone know if it is possible to destroy a meetme bridge, or to kick users out of the meetme conference room? |
21:55.04 | Marlow | ehe ... i've got tt-allbusy, before you get to my VM |
21:55.16 | Marlow | <-- jets |
21:56.32 | jets | huh? |
21:57.17 | JerJer | zeta_: sorta... i've been playing with some manager hooks for app_meetme |
21:57.24 | Marlow | jets : i just found it too amusing, when i heard that the first time on 612@fwd |
21:58.04 | jsharp | Extension '8005095639' in context 'incomingpri' from '008307519917' does not exist. Rejecting call ecting call on channel 1, span 4 |
21:58.16 | jsharp | Except the extension does exist in the context. |
22:00.35 | bkw_ | does it now? |
22:01.02 | tzanger | bah |
22:01.04 | tzanger | you guys are right |
22:01.11 | tzanger | no far end disconnect on the channel bank I |
22:01.49 | *** join/#asterisk af_ (af@ip314-35-1.adsl.edisontel.com) |
22:01.52 | jsmith | tzanger: Yup... use an Adit 600 for FXO ports... |
22:02.08 | tzanger | jsmith: yup |
22:02.12 | tzanger | how do I kill the channels? |
22:02.18 | tzanger | can I tell * for forcibly hang them up? |
22:02.22 | Marlow | anyone an idea, what codec the fwdnet 0800 gateway requires now ? |
22:02.23 | tzanger | show applications |
22:02.23 | jsmith | soft hangup Zap/1 |
22:02.26 | tzanger | haha wrong one |
22:02.28 | tzanger | er window |
22:03.44 | tzanger | jsmith: thanks |
22:03.58 | tzanger | damn liars... Carrier Access says CB1 does Far-end discon |
22:04.00 | tzanger | oh well |
22:04.13 | ]data[_ | ooh er |
22:04.21 | ]data[_ | * Run-Time Check Failure #0 - The value of ESP was not properly saved across a function call. This is usually a result of calling a function declared with one calling convention with a function pointer declared with a different calling convention. |
22:04.47 | tzanger | later all tiem to pick up the kids |
22:06.51 | *** join/#asterisk woodsy (~woodsy@64.6.35.237) |
22:08.58 | zeta_ | JerJer, have you had any success at it? |
22:10.20 | JerJer | who what when where why? |
22:10.37 | zeta_ | JerJer, with kicking users out of a conference/channel? |
22:11.10 | outtolunc | seems to me, last time i viewed app_meetme.c the 'admin menu' was a bit lacking |
22:11.16 | zeta_ | heh yeah |
22:11.18 | *** part/#asterisk mrgoby (~mrgoby@pcp05304587pcs.wanarb01.mi.comcast.net) |
22:11.29 | JerJer | outtolunc: yeah thats a stubb for someone else to implement |
22:11.32 | JerJer | but its not needed |
22:11.37 | outtolunc | ah |
22:11.41 | zeta_ | I'm trying to find out if I have to code in support into the meetme source to actually accomplish this task |
22:11.45 | JerJer | _EAGLE_'s agi hook is ABSOTIVELY KILLER! |
22:12.06 | pawpa | what agi hook? |
22:12.16 | JerJer | show application meetme |
22:12.16 | outtolunc | yeah cough it up <G> |
22:13.07 | *** join/#asterisk espenz (espen@a217-118-51-166.bluecom.no) |
22:13.13 | espenz | yeh, yo |
22:13.16 | outtolunc | ah |
22:14.04 | pbxtech | where is his agi app at? |
22:14.25 | *** join/#asterisk FredtheBadger (~fred@128.113.36.123) |
22:14.38 | FredtheBadger | hi, I have a question about asterisk and DynExtenDB |
22:15.21 | Marlow | hmm .. skal lige se, om jensd også kender danske el. svenske .. han er jo dansk .. |
22:15.28 | JerJer | haven't we beat that dead horse enough ? |
22:15.34 | FredtheBadger | I need database support for extensions (unless there's an easy way to deal with a list of extensions that changes rediculously often in large quantities), but I've heard a lot of bad things after searching |
22:15.51 | FredtheBadger | I figured it was a dead horse, but I still need info on it :) |
22:16.05 | JerJer | its a hack |
22:16.23 | pbxtech | JerJer, is eagles agi hook public? |
22:16.30 | FredtheBadger | is there a better way to do it? |
22:16.36 | FredtheBadger | I'm assuming the answer's yes |
22:17.52 | JerJer | yes, vi extensions.conf |
22:18.07 | FredtheBadger | ? |
22:18.39 | JerJer | pbxtech: show application meetme |
22:18.54 | JerJer | for the lazy |
22:18.55 | JerJer | <PROTECTED> |
22:18.55 | JerJer | <PROTECTED> |
22:19.37 | bkw_ | Windows the Internet Whore.. catches more viruses and worms than a french whore in pheonix |
22:19.55 | JerJer | the zaptel only problem can be fixed, with proper motivation |
22:20.17 | bkw_ | well fix it |
22:20.18 | bkw_ | :) |
22:20.24 | bkw_ | hehe |
22:20.24 | JerJer | hence proper motivation |
22:20.46 | JerJer | it works absolutely flawlessly for me |
22:20.50 | JerJer | currently |
22:21.20 | mishehu | are there many french whores in phoenix? |
22:21.37 | sizzzung | jerjer. |
22:21.47 | sizzzung | JerJer: when are we going to party? |
22:21.53 | sizzzung | when are you coming down to south florida? |
22:22.31 | JerJer | when a ticket shows up here :) |
22:22.47 | sizzzung | plz |
22:24.12 | JerJer | now if u could provide some clean, geek friendly female units to party with, that might encurage me to jump in the C172 and head down |
22:24.54 | pbxtech | jerjer, did shido9 ever talk to you about being able to record a conference call? |
22:25.06 | bkw_ | its possible |
22:25.18 | bkw_ | and easy if you stop and think about it |
22:25.23 | JerJer | the recording it not hard at all |
22:25.38 | bkw_ | nope |
22:25.42 | bkw_ | I record all calls |
22:25.47 | pbxtech | just recording then mixing right |
22:25.53 | JerJer | its how you plan to manage those recordings after the fact is what gets tricky |
22:25.57 | JerJer | or can get |
22:26.12 | pbxtech | he never got back to me on a price to get something built |
22:26.36 | pbxtech | cant you have some monitor extension dial into it an record off 1 line? |
22:26.39 | pbxtech | or something |
22:27.23 | bkw_ | pbxtech you can.. or setup the conf admin as the monitor |
22:27.34 | bkw_ | once they enter.. record them |
22:27.50 | *** join/#asterisk indiam (~sipjic@81-86-244-189.dsl.pipex.com) |
22:28.01 | pbxtech | i just need something build and I was willing to pay to get it done |
22:28.36 | *** join/#asterisk Spaceboy (furgaw@205.124.232.209) |
22:28.39 | indiam | i have a specific question to use asterisk only as a User Agent to Register to SIP |
22:28.48 | indiam | can anybody help me ? |
22:29.55 | *** join/#asterisk RichA (~Rich@vegas.routers.com) |
22:30.22 | indiam | is nobody in this channel ?? |
22:30.39 | bkw_ | indiam no you just don't barge in and start asking questions.. thats rude |
22:30.43 | bkw_ | say hi |
22:30.44 | bkw_ | or something |
22:30.47 | bkw_ | jesus |
22:30.48 | indiam | hi |
22:30.50 | indiam | yes |
22:30.54 | indiam | sorry about that |
22:30.59 | bkw_ | you want to change the user agent on outbound sip registrations right? |
22:31.06 | bkw_ | if so then you need to edit the src code |
22:31.14 | *** join/#asterisk bevins (~bob@modemcable197.44-202-24.mc.videotron.ca) |
22:32.07 | indiam | I want to just use it as a SIP UA |
22:32.36 | drgalaxy | I keep getting "ast_load_resource): /usr/lib/asterisk/modules/chan_skinny.so: undefined symbol: ast_pickup_call" when I try to load module chan_skinny.so |
22:32.54 | drgalaxy | anyone have any ideas? |
22:34.17 | Corydon76-work | You forgot to load res_parking.so first |
22:34.23 | Spaceboy | I need some help with installing the h323 channel driver is there any one here that has done it before? |
22:34.55 | JerJer | Spaceboy: depends on which one your talking about |
22:35.03 | Corydon76-work | Spaceboy: follow the instructions in the README *to* *the* *letter* |
22:35.34 | JerJer | yeah what Corydon76-work said |
22:38.07 | JerJer | data[Zzz]: whimp |
22:38.33 | JerJer | ive been up almost two days straight now |
22:38.59 | mishehu | hmm... |
22:39.12 | mishehu | I wonder if The Precursors would sound good as music on hold |
22:39.28 | JerJer | i like russian-monkeys myself |
22:40.27 | bkw_ | JerJer |
22:41.40 | bkw_ | show me on the doll where the bad man thouched you. |
22:42.28 | mishehu | was the bad man who thouched you named michael jackson? |
22:42.37 | bkw_ | haha |
22:42.42 | bkw_ | mishehu I was thinking along those lines |
22:43.59 | mishehu | bkw_: criminal minds think alike |
22:44.15 | bkw_ | i'm not criminal |
22:44.20 | bkw_ | doesn't matter what my record says |
22:44.27 | bkw_ | I'M NOT NOT NOT NOT NOT |
22:46.15 | gage_man | Hey anyone Know max CALL-waiting callers |
22:46.19 | gage_man | I only have 2 |
22:47.18 | JerJer | 1-517-679-7772 <--- russian monkeys |
22:47.54 | sizzzung | i stink |
22:47.55 | mishehu | my record says nothing... I stopped buy lp's when I was... hm... probably 10... |
22:48.07 | *** join/#asterisk Apophis (~btatton@209.180.83.2) |
22:48.15 | sizzzung | JerJer: gvive me your phone number |
22:48.46 | JerJer | 1-517-679-7772 |
22:48.49 | JerJer | lol |
22:48.55 | sizzzung | damn it |
22:49.09 | sizzzung | seriously. |
22:49.11 | jets | awesome asterisk is awesome |
22:49.13 | sizzzung | let's talk. |
22:49.16 | sizzzung | i have questions. |
22:49.38 | tim27 | any know here if i will get echo using a SIP phone to make or receive call from PTSN |
22:49.47 | sizzzung | tim27: probably not. |
22:49.50 | sizzzung | but anything is possible. |
22:50.28 | mishehu | JerJer: I hope the russian monkeys weren't the products of soviet nuclear research |
22:50.29 | mishehu | heh |
22:50.51 | JerJer | lol |
22:51.04 | stonefly | POT lines hooked up to a channel bank are very frustrating!!! One call it is too quiet, another call it's too echo'y another call there is feedback... I wish frac T1's were cheaper here! |
22:51.17 | tim27 | sizzung : i read on the web many post complaining about echo ... on sip phone... why ??? |
22:51.36 | sizzzung | tim27: because people suck. |
22:51.37 | jsharp | bleh. I can't get this PRI problem straightened out. |
22:51.44 | tim27 | lol |
22:52.08 | tim27 | i have only 3 phone line... so i can only affort ... analog... |
22:52.33 | stonefly | tim27, same here... |
22:52.58 | tim27 | stonefly: you use fxo with sip phone ... or with analog phone... |
22:56.42 | *** join/#asterisk daork (~daork@202.89.35.252) |
23:01.48 | *** part/#asterisk Exomorph_ (Greg@134-9.bvcompuworks.com) |
23:02.22 | *** join/#asterisk cfo (~cfo@194.19.190.217) |
23:15.01 | *** join/#asterisk xeno42 (~xeno42@wtf.cx) |
23:16.15 | *** part/#asterisk stonefly (~trillian@toby.stoneflytech.com) |
23:16.24 | *** join/#asterisk stonefly (~trillian@toby.stoneflytech.com) |
23:17.22 | *** join/#asterisk Exomorph_ (Greg@134-9.bvcompuworks.com) |
23:19.23 | sizzzung | haha jerjer. |
23:19.27 | sizzzung | gizzzone. |
23:21.02 | *** join/#asterisk juhas (juhas@hot.juhas.net) |
23:22.42 | juhas | I'm having a problem with sync sources on a TE410P, could anybody help? Mainly, zttool tells that sync source is "internal" even when it should be the line itself. |
23:23.01 | JerJer | yeah gdb tends to cause a lil blocking |
23:23.31 | *** join/#asterisk okrumm (~okrumm@dsl-200-95-104-110.prodigy.net.mx) |
23:24.25 | jsmith | JerJer: Do you happen to have a CVS server we could use for this documentation stuff? |
23:25.04 | doughecka | http://www.st.cs.uni-sb.de/askigor/ |
23:25.06 | doughecka | interesting |
23:25.40 | izo | juhas when you change /etc/zaptel.conf do you do ztcfg -vvv |
23:25.49 | ManxPower | Hmm. Took me 5 mins to make my Asterisk server work after the PRI was installed. |
23:29.21 | jsharp | Come fix my PRI, then. |
23:31.24 | *** join/#asterisk espenz (espen@a217-118-51-166.bluecom.no) |
23:31.40 | *** join/#asterisk fyman (~fyman@CPE-138-130-18-16.nsw.bigpond.net.au) |
23:33.08 | *** join/#asterisk cfo (~cfo@194.19.190.217) |
23:33.17 | *** join/#asterisk pbx_noob (~pbx_noobi@hoochie.digium.com) |
23:33.29 | *** join/#asterisk anthm (~anthmct@CPE-24-167-193-161.wi.rr.com) |
23:35.12 | ManxPower | jsharp, They way you fix your PRI is to go call up a good CLEC 8-) |
23:35.20 | indiam | jsharp he said it is only for his asterisk server, not for yours dear |
23:35.23 | ManxPower | jsharp, what's the span= line in your zaptel.conf? |
23:37.43 | jtodd | Can accountcode be set from within the dialplan? I see that it's frustratingly only set in the channel configs... |
23:38.55 | ManxPower | jtodd, yes |
23:39.11 | jtodd | So, I can SetVar(ACCOUNTCODE)=blah ? |
23:39.22 | ManxPower | exten => 91411,1,SetAccount(${CALLERIDNUM}) |
23:39.38 | jtodd | Cool cool. I'll try... |
23:40.51 | jets | any way to import the "show channels" in to an agi script? |
23:41.40 | *** join/#asterisk Exomorph_ (Greg@134-9.bvcompuworks.com) |
23:42.00 | ManxPower | jsharp, what's the span= line in your zaptel.conf? |
23:44.39 | zoa | any asterisk internals for rent ? |
23:44.46 | pbx_noob | does anyone have a link to good doc for CLI? |
23:44.51 | jsmith | zoa: Rent? |
23:45.03 | jsmith | pbx_noob: Hit tab in the CLI... |
23:45.20 | pbx_noob | thanks |
23:45.25 | jsmith | pbx_noob: For example, type "iax [TAB]" |
23:46.13 | zoa | yes i need someone to take it up the ... |
23:47.24 | *** part/#asterisk tim27 (doug25@229-29.dr.cgocable.ca) |
23:47.54 | ManxPower | context = incoming |
23:47.54 | ManxPower | signalling = pri_cpe |
23:47.54 | ManxPower | group = 1-4 |
23:47.54 | ManxPower | channel => 1-4 |
23:47.59 | ManxPower | the group = is correct? |
23:49.19 | jtodd | ManxPower: Nope, no luck. ACCOUNTCODE is not settable. |
23:49.40 | ManxPower | jtodd, Maybe not on your asterisk, but it is on mine, using the line I pasted. |
23:50.23 | jtodd | Ah... sorry, I mis-read your line. I'll try again. (getting sloppy - I want to get the hell out of the office but I keep getting new jobs piled on me as I reach for my bags...) |
23:50.41 | jsmith | jtodd: I know that feeling! |
23:52.15 | *** join/#asterisk fmany (~fyman@CPE-138-130-18-16.nsw.bigpond.net.au) |
23:52.18 | *** join/#asterisk Conro (~rolko@pcp01843985pcs.lncstr01.pa.comcast.net) |
23:52.28 | Conro | Hey, I'm in need of some help :) |
23:52.40 | Conro | If anyone is available to help an oldbie turned newbie, PM me plz =) |
23:52.56 | Conro | *cries* |
23:52.59 | jsmith | Conro: Just ask your question... |
23:53.03 | Conro | Alright, thanks :) |
23:53.09 | Conro | I need to know how to register myself a channel on this server |
23:53.17 | Conro | I'm going to be using to to host a meeting for a website I work for :) |
23:53.20 | Conro | I forget how =p lol |
23:53.36 | Conro | #nexusatlas |
23:53.51 | jsmith | just type "/join nexusatlas" |
23:53.59 | jsmith | And you'll be in that channel |
23:54.09 | Conro | Doesn't exist, I'll need to create :) |
23:54.12 | Conro | How do I do that? lol =p |
23:54.14 | *** join/#asterisk bkw__ (~brian@hoochie.digium.com) |
23:54.21 | jsmith | just type "/join nexusatlas" and it'll create it |
23:54.26 | *** join/#asterisk dasenjo (~dasenjo@200.21.83.174) |
23:54.41 | Conro | - |
23:54.41 | Conro | nexusatlas That channel doesn't exist |
23:54.59 | bkw_ | what evah |
23:55.03 | Conro | err, forgot to put the # in front of it :) |
23:55.07 | Conro | It's created now :) |
23:57.40 | bkw_ | blah blahb blah |
23:57.52 | *** join/#asterisk bjohnson (~bjohnson@ip114-165.tor.istop.com) |
23:58.08 | *** join/#asterisk forkenb (~forkenb@dailup-219-90-63-226.appscorp.net) |
23:59.16 | sizzzung | you ain't noting but a hoochie momma |