| 00:01.13 | *** join/#asterisk ProActive (n=ProActiv@bl7-131-26.dsl.telepac.pt) |
| 00:01.55 | ProActive | Hi. Anyone uses voipcheap with asterisk? |
| 00:07.35 | *** join/#asterisk JayTee52 (n=jtforde0@c-69-137-243-25.hsd1.in.comcast.net) |
| 00:08.33 | CunningPike | jimbo-: Any Day Now(tm) |
| 00:09.02 | CunningPike | bluregard: You can't edit the softkey labels or functions |
| 00:09.26 | CunningPike | bluregard: All you can do is enable and disable features |
| 00:09.32 | *** join/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.236.140.Dial1.SanJose1.Level3.net) |
| 00:11.36 | jimbo- | is early media support on SIP ported to the 1.2 branch? |
| 00:11.57 | jimbo- | i've only seen it mentioned on 1.4.x |
| 00:12.51 | SwK | jimbo yes |
| 00:13.29 | file | SwK: ! |
| 00:14.24 | SwK | file! |
| 00:14.48 | file | how goes? |
| 00:15.30 | [TK]D-Fender | <PROTECTED> |
| 00:15.44 | CunningPike | [TK]D-Fender: Do tell...... |
| 00:16.08 | CunningPike | [TK]D-Fender: Make my 'Transfer' softkey dial voicemail |
| 00:16.15 | [TK]D-Fender | CunningPike : Think creative and examine exactly what comes in the firmware provisioning zip :) |
| 00:16.50 | CunningPike | [TK]D-Fender: Hmm - well, if the OP likes messing around with the SIP image, I guess........ |
| 00:17.16 | [TK]D-Fender | CunningPike : nope.... getting colder... |
| 00:17.33 | [TK]D-Fender | CunningPike : You need to think a little more "outside the box" |
| 00:17.44 | CunningPike | [TK]D-Fender: I don't - the OP does - tell him |
| 00:18.58 | [TK]D-Fender | CunningPike : So disappointed...... it's its own xml file.... LOOK..... |
| 00:19.26 | CunningPike | [TK]D-Fender: I don't care - tell the original poster. |
| 00:21.16 | jjshoe | is anyone aware of a way to figure out if a phone is in DND or forwarded without sending an invite? If you send an invite, the obvious response from the phone is a ring, and I want to check without ringing the phone |
| 00:21.27 | [TK]D-Fender | CunningPike : In the SoundPointIPLocalization you have a folder per language with #'d prashes and what should display on-screen. So yes you can rename any soft-key you want.... |
| 00:21.36 | [TK]D-Fender | phrases* |
| 00:21.54 | [TK]D-Fender | CunningPike : Not to any productive end, but yes, you CAN rename them. |
| 00:27.05 | *** join/#asterisk sloth (n=josh@cpe-69-203-212-182.nyc.res.rr.com) |
| 00:29.24 | Math` | jjshoe: my opinion would be that you'd have to implement that on a per-brand basis |
| 00:30.27 | SwK | file: it goes |
| 00:30.36 | JayTee52 | Is it possible to route outbound calls to an FXO port on a Digium 100 card if that channel is also being used for inbound calls assuming that the line would not be busy as it is only used by one SIP phone? |
| 00:30.57 | SwK | file: so when is Quebec going to be its own country? |
| 00:31.11 | JT | digium 100? |
| 00:31.17 | JayTee52 | hi JT |
| 00:31.17 | file | SwK: it already is, we just don't tell outsiders |
| 00:31.18 | jimbo- | SwK: thanks fro the reply |
| 00:31.23 | JT | hi |
| 00:31.31 | file | JayTee52: X100P? |
| 00:31.38 | JayTee52 | JT, yes |
| 00:31.39 | *** part/#asterisk jimbo- (n=dsgsdfga@modemcable252.110-83-70.mc.videotron.ca) |
| 00:31.51 | JayTee52 | it's an OEM x100p |
| 00:32.13 | SwK | hah |
| 00:32.52 | JayTee52 | I got it working for inbound calls to a specific extension after you helped me the other night. I've been reading through all the dialing plan info on setting up outbound calls. |
| 00:35.26 | *** join/#asterisk sahafeez (n=sahafeez@67.109.14.227.ptr.us.xo.net) |
| 00:40.18 | JayTee52 | JT, I guess what I'm asking is do the FXO channels on the Digium X100P or their newer TDM cards allow both inbound and outbound dialing? I've setup T1's with FXO on the PBX end and FXS on the station end to have 4 digit dialing to retail stores for one company I worked for but that was copper. |
| 00:41.56 | file | you can't dial both in and out at the same time |
| 00:42.02 | file | but you can receive calls, and place calls over it |
| 00:44.09 | *** join/#asterisk robin_sz (n=robin@rapid2.gotadsl.co.uk) |
| 00:44.27 | robin_sz | hmmm ... my zap_hfc seems to be sharing the same interrupt as my scsi controller. is this a bad thing? |
| 00:44.27 | JayTee52 | file, I know and each line would be a standard analog line from my legacy PBX, a Meridian Option 11C and would be dedicated to one SIP phone per FXO port/channel. If this works then we can get budget approval for buying a 4 port T1 line card next year and route everything through the asterisk server and then any legacy extensions would route to PRI circuits to the old pbx. |
| 00:46.03 | JayTee52 | but since we're a non-profit and my boss likes to do everything on the cheap for now I'm trying to set this up for analog FXO to SIP with one port for each SIP phone. |
| 00:46.04 | sahafeez | anyone know how to reboot a polycom 501 from the web interface. seems you can do anything but |
| 00:46.43 | file | JayTee52: okay, I'll be wild and guess it's the dialplan logic for outbound dialing that is killing you |
| 00:46.52 | CunningPike | sahafeez: You can reboot them from the CLI a whole lot easier if you have sip-notify set up for them' |
| 00:47.15 | sahafeez | do not think i have that setup. |
| 00:47.22 | JayTee52 | it's a whole new language for me. My head is still stuffed with a half million 3 letter and 4 letter acronyms from Nortel M1 programming. |
| 00:47.28 | Shaun2222 | whats the recommended method for playing hold music... right now i'm using moph3 or whatever but it sucks because it always plays the same shit because it starts the music as soon as the customer is placed on hold.. |
| 00:47.55 | sahafeez | CunnignPike - link/docs?? |
| 00:48.08 | sahafeez | hum. CunningPike that is - link/docs |
| 00:48.19 | CunningPike | sahafeez: Search the wiki for 'polycom reboot sip-notify' |
| 00:48.35 | sahafeez | thanks will do |
| 00:48.42 | CunningPike | Shaun2222: We use native MOH with the random option set |
| 00:48.45 | CunningPike | ~moh |
| 00:48.54 | jbot | it has been said that moh is Music On Hold. Good information about how to set it up in the various possible ways can be found at http://www.voip-info.org/wiki/index.php?page=Asterisk+config+musiconhold.conf |
| 00:49.03 | CunningPike | jbot ate too much turkey |
| 00:49.14 | robin_sz | <PROTECTED> |
| 00:49.21 | JayTee52 | i ate too much turkey and too many other things besides. |
| 00:49.25 | robin_sz | from /proc/interrupts |
| 00:49.33 | robin_sz | ^^ thats not good is it? |
| 00:50.20 | robin_sz | and zttest figures around the 80% mark? |
| 00:51.14 | *** join/#asterisk derek (n=derek@ip68-3-24-234.ph.ph.cox.net) |
| 00:57.24 | CunningPike | robin_sz: No, that's not good |
| 00:57.52 | robin_sz | some stuff, like ring tones as its ringing an extension sound absolutely crystal clear |
| 00:58.05 | CunningPike | robin_sz: The two worst things you can share interrupts with (remembering that you shouldn't be sharing them at all) are disk and network |
| 00:58.10 | robin_sz | when it drops to voicemail it distorts and is covered in intermodulation |
| 00:58.47 | CunningPike | robin_sz: That's because progress tones aren't audio - they're signalling |
| 00:58.53 | jjshoe | is anyone aware of a way to figure out if a phone is in DND or forwarded without sending an invite? If you send an invite, the obvious response from the phone is a ring, and I want to check without ringing the phone |
| 00:59.11 | robin_sz | CunningPike, that woudl explain it then |
| 00:59.38 | CunningPike | robin_sz: The ringing an outside caller hears is generated by their CO as a result of signaling sent to it |
| 00:59.39 | Shaun2222 | anybody have the 9 prefix working on a 601 phone so that when 9 is pressed a dialtone is given after that... |
| 00:59.48 | robin_sz | CunningPike, so, what is likely to be a solution ?? just swapping the card around PCI slots until it repents? |
| 00:59.49 | Shaun2222 | i set 9,1[2-9] but it's not working |
| 01:00.02 | CunningPike | Shaun2222: In your digitmap? |
| 01:00.13 | robin_sz | CunningPike, are you SURE about that? |
| 01:00.13 | Shaun2222 | in my sip.cfg |
| 01:00.32 | CunningPike | robin_sz: Yes - you need to get that card on it's own IRQ |
| 01:00.34 | JT | robin_sz: zttest 80% avg, are you kidding?? |
| 01:01.12 | CunningPike | Shaun2222: Aye |
| 01:01.13 | Shaun2222 | CunningPike? |
| 01:01.13 | CunningPike | Shaun2222: What is your digitmap string? |
| 01:01.13 | robin_sz | CunningPike, no i meantr |
| 01:01.18 | Shaun2222 | <digitmap dialplan.digitmap="9,1[2-9]|[1-8]xx" dialplan.digitmap.timeOut="3"/> |
| 01:01.20 | robin_sz | sure about where the tones are generated |
| 01:01.26 | JT | robin_sz: anything below 99.97..% is bad |
| 01:01.34 | CunningPike | robin_sz: No, I made it up |
| 01:01.41 | JT | are you sure you mean 80%? |
| 01:01.53 | JT | i have no idea how the D channel would even stay up |
| 01:02.32 | robin_sz | CunningPike, I was trying to reconcile that with why when calling a foreign number (eg back end of botswana) you get to hear their crazy ringtones rather than just your local ones, as handed on by signalling? |
| 01:02.48 | robin_sz | JT, well not always 80% .. sometimes less |
| 01:02.56 | JT | 80.xxx%? |
| 01:02.58 | rob0 | I had my TDM400P card crash the other day, "TDM PCI Master abort" thousands per second, overwhelmed syslogd. But unload and reload wctdm, zttest > 99.98%, and all is well since. |
| 01:03.04 | JT | that's beyond awful |
| 01:03.12 | robin_sz | sorry my bad .. 89% |
| 01:03.21 | rob0 | (Still getting over 1K interrupts on cpu#1 per second.) |
| 01:03.29 | JT | yeah, i have issues below 99.97% |
| 01:03.36 | CunningPike | robin_sz: Not sure how international stuff works |
| 01:03.42 | robin_sz | Opened pseudo zap interface, measuring accuracy... |
| 01:03.42 | robin_sz | 89.819336% 89.611816% 89.587402% 89.831543% 89.648438% 89.514160% 89.855957% |
| 01:03.48 | JT | :O |
| 01:03.58 | JT | err yeah |
| 01:04.08 | JT | anyway, they do that with early B3 audio i believe |
| 01:04.15 | robin_sz | so thats sub-optimal then? |
| 01:04.19 | JT | there's also signalling to tell you it's ringing then |
| 01:04.31 | *** join/#asterisk vexorg (n=vexorg@CPE0003478eef7c-CM0016b531e87c.cpe.net.cable.rogers.com) |
| 01:04.33 | JT | yes, get it to at least 99.97% |
| 01:04.34 | *** join/#asterisk sloth (n=josh@cpe-69-203-212-182.nyc.res.rr.com) |
| 01:04.39 | JT | what's the specs/config? |
| 01:05.38 | robin_sz | right. |
| 01:05.38 | CunningPike | Shaun2222: What's the first part of your digitmap supposed to be doing? Allowed 2-digit numbers? |
| 01:05.38 | CunningPike | s/Allowed/Allowing/ |
| 01:05.39 | *** join/#asterisk lters (n=tech@mrtcdsl-433.mis.net) |
| 01:05.39 | robin_sz | dual 800mhz Xeon I think ... numernous crappy PCI cards (wcfxo, framgrabber, "cologne" ISDN) |
| 01:05.41 | Shaun2222 | CunningPike: i just pasted this one in to see if it works... [2-9]11|0T|011xxx.T|9011xxx.T|91[2-9]xxxxxxxxx|9[2-9]xxxxxx|*xx|[8]xxx|[1-7]xx |
| 01:05.48 | Shaun2222 | but added ,'s after the 9's |
| 01:05.53 | JT | why does it need a framegrabber in the same box |
| 01:06.09 | CunningPike | Shaun2222: And did it? |
| 01:06.16 | Shaun2222 | well i'm trying to just basically make it so that when they press 9 they still hear a dialtone |
| 01:06.24 | robin_sz | JT, for the motion detection video cameras around the building .. |
| 01:06.29 | CunningPike | Shaun2222: Well, the comma is supposed to do that, alright |
| 01:06.29 | Shaun2222 | it's still loading, these phones are so slow to load.. |
| 01:06.31 | robin_sz | JT, is that a bad thing> |
| 01:06.34 | robin_sz | ?? |
| 01:06.48 | CunningPike | Shaun2222: Yup - "this may take a few seconds" - yah - like 600 |
| 01:06.53 | Shaun2222 | lol |
| 01:07.09 | robin_sz | I have a spare 800mhz Dell poweredge, loaded with Xorcomm BRI-stuffed |
| 01:07.12 | Shaun2222 | expecially when updating firmware... shit was like 15-20 minutes |
| 01:07.20 | robin_sz | its beginning to sound like a better option |
| 01:07.32 | Shaun2222 | CunningPike: you dont happen to have access to the 2.0.3 firmware do you ;) |
| 01:07.52 | JT | robin_sz: well if you're having these issues, yes |
| 01:08.00 | JT | framegrabbing is pretty interrupt intensive |
| 01:08.01 | CunningPike | Shaun2222: Nope - the latest I've heard of is 2.0.2 |
| 01:08.11 | Shaun2222 | i'm hoping it will make phones on a NAT connecting to public asterisk box work better. |
| 01:08.25 | CunningPike | Shaun2222: We run 1.6.6/1.6.7 at the moment |
| 01:08.26 | Shaun2222 | i found 2.0.2 on that freephone site, polycom's site has a 2.0.3 on it |
| 01:08.39 | robin_sz | JT, does sounding like its being processed by passing it through a small 2 stroke engine count as an "issue"? |
| 01:08.48 | JT | yes |
| 01:08.50 | Shaun2222 | CunningPike: i was running that... then hoped nat would be improvved in 2.0.2 |
| 01:08.56 | Shaun2222 | they did add a nat keep alive.. |
| 01:08.57 | JT | let's put it this way |
| 01:09.00 | JT | for a digital channel |
| 01:09.13 | JT | i am VERY surprised it is working at all with such low zttest results |
| 01:09.14 | Shaun2222 | nope.. still doesnt give me a dialtone |
| 01:09.25 | CunningPike | Shaun2222: Weird |
| 01:09.46 | Shaun2222 | anyway to pull the dialmap set in the phone to see if it's actually listening to what i told it |
| 01:09.51 | Shaun2222 | i can see that its' pulling the config |
| 01:10.10 | Shaun2222 | in the cisco phones i could do like a sh dialplan or somthing |
| 01:10.20 | robin_sz | JT, fair enough ... it does always seem to manage to pick up the call and answer it thoug |
| 01:10.42 | robin_sz | JT i dont think it explains why I cant dial out on that channel does it? |
| 01:10.44 | JT | aren't you getting HDLC errors to your server's console? |
| 01:10.53 | JT | or pri errors in ast console |
| 01:10.59 | robin_sz | nope |
| 01:11.06 | Shaun2222 | hmm |
| 01:11.12 | robin_sz | just crap sounding audio |
| 01:11.13 | Shaun2222 | wait i have two dialplan sections... |
| 01:11.13 | JT | this is bri right? |
| 01:11.16 | robin_sz | yeah |
| 01:11.20 | Shaun2222 | ones in the phone.cfg and one's in sip.cfg |
| 01:11.23 | JT | what drivers are you using? |
| 01:11.29 | robin_sz | zap-hfc |
| 01:11.43 | JT | bristuff? |
| 01:11.46 | robin_sz | yip |
| 01:11.54 | JT | hrm |
| 01:12.08 | JT | and yes, scsi and isdn card on same interrupt very bad |
| 01:12.37 | robin_sz | sigh |
| 01:12.54 | robin_sz | so ... this box has sat here waiting for the line to convert to ISDN |
| 01:13.00 | JT | try playing the card swapping game |
| 01:13.04 | robin_sz | bt have fucked up the order time and time agai ... |
| 01:13.17 | robin_sz | and finally, they swap the lien on the moring we have a job advert out |
| 01:13.30 | robin_sz | and .. the phones are down .. great |
| 01:13.38 | JT | mm |
| 01:13.55 | robin_sz | "get it diverted to another line" you say ... |
| 01:14.19 | robin_sz | well we tried that, after 6 hours of calling BT I gave up |
| 01:15.35 | CunningPike | Shaun2222: ah - they load in the reverse order as they are listed in the phone's <mac-address>.cfg file, so it's likely that the one in your phone.cfg file is loading last and taking precedence |
| 01:15.49 | robin_sz | I think I might just try putting the card into the dell/xorcomm box |
| 01:16.00 | *** join/#asterisk Mportnoy (n=test@201.199.68.150) |
| 01:16.02 | JT | asterisk boxes with hardware should be pretty much dedicated |
| 01:16.06 | robin_sz | then i can bug tzafrir about it ;) |
| 01:16.26 | Mportnoy | hello, anyone knows if AMD LX-800 will compile Asterisk lastest edition and all other modules? |
| 01:16.53 | robin_sz | I run some other boxen with eicon divas in them.. they seem immune to issues of interrupts and workload |
| 01:16.56 | Shaun2222 | CunningPike: the one in phone.cfg is blank |
| 01:17.13 | JT | what do you use the xorcom for? |
| 01:17.18 | CunningPike | Shaun2222: Get rid of it altogether, I would try |
| 01:17.51 | CunningPike | Shaun2222: The dialplan section, not the file :) |
| 01:18.32 | Shaun2222 | lol |
| 01:18.34 | Shaun2222 | ya, thanks |
| 01:18.40 | Mportnoy | hello, anyone knows if AMD LX-800 will compile Asterisk lastest edition and all other modules? |
| 01:18.41 | Shaun2222 | i removed dialplan and digitmap |
| 01:18.47 | Shaun2222 | and some other shit like emergency and stuff |
| 01:19.01 | Mportnoy | hello, anyone knows if AMD Geode LX 800 will compile Asterisk lastest edition and all other modules? |
| 01:20.02 | Shaun2222 | CunningPike: my phone downloads phone.cfg first and then sip.cfg |
| 01:20.23 | CunningPike | Shaun2222: Remember it's in reverse order....... |
| 01:20.27 | Shaun2222 | hmm, then it trys to download MAC-phone.cfg |
| 01:20.33 | Shaun2222 | thats weird, i didnt tell it to do tha |
| 01:20.33 | lters | any known longdistance outages with nufone or others? |
| 01:20.55 | CunningPike | Shaun2222: That's where is stores user preferences chosen via the phone menus |
| 01:21.00 | CunningPike | s/is/it/ |
| 01:21.49 | CunningPike | Shaun2222: I have to run now, but if you're still having trouble, pastebin your files and someone (probably [TK]D-Fender :) ) will take a look at them |
| 01:21.56 | CunningPike | ~pb |
| 01:22.05 | jbot | [pb] a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 01:22.08 | Shaun2222 | ya still not working.. |
| 01:22.10 | Shaun2222 | thanks though. |
| 01:22.29 | *** part/#asterisk ProActive (n=ProActiv@bl7-131-26.dsl.telepac.pt) |
| 01:24.51 | CunningPike | Shaun2222: What is se.appLocalEnabled set to? |
| 01:26.41 | rob0 | Mportnoy: LX800 is x86-compatible, no? |
| 01:26.51 | Mportnoy | mmmm |
| 01:26.53 | Mportnoy | dont know |
| 01:26.54 | CunningPike | Shaun2222: If it isn't 1, set it to 1 and see if that helps |
| 01:27.02 | Mportnoy | thats why im asking |
| 01:27.04 | CunningPike | Gotta run |
| 01:28.04 | rob0 | well, is it running a Linux? "uname -a"? |
| 01:29.33 | Mportnoy | no |
| 01:29.36 | Mportnoy | thats why im asing |
| 01:29.38 | Mportnoy | to see if |
| 01:29.40 | Mportnoy | I buy it |
| 01:32.04 | rob0 | What hardware would be in it? I think you might have trouble with some Zaptel hardware. But asterisk itself should be no problem. |
| 01:32.33 | rob0 | (IOW, fine with ATA's.) |
| 01:39.03 | Shaun2222 | ya it's set to 1 |
| 01:42.02 | *** join/#asterisk ivanfm (n=ivanfm@c93481ec.virtua.com.br) |
| 01:53.22 | *** join/#asterisk sloth (n=josh@cpe-69-203-212-182.nyc.res.rr.com) |
| 02:05.53 | *** join/#asterisk sloth (n=josh@cpe-69-203-212-182.nyc.res.rr.com) |
| 02:07.35 | *** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox) |
| 02:09.22 | *** join/#asterisk drcode (n=chatzill@87.69.59.186.cable.012.net.il) |
| 02:09.23 | *** join/#asterisk stkn__ (i=nobody@gentoo/developer/pdpc.active.stkn) |
| 02:09.30 | drcode | hi all |
| 02:09.57 | drcode | I have linux with asterisk box |
| 02:10.16 | drcode | In this linux box I don't have gui |
| 02:10.34 | drcode | can I run IAX client or sip client under console? |
| 02:13.29 | *** join/#asterisk upod (n=chibondk@cpe-66-108-211-222.nyc.res.rr.com) |
| 02:19.18 | rob0 | drcode: the asterisk console can be a softphone, if you have sound. |
| 02:19.39 | drcode | I have sound card |
| 02:19.50 | drcode | I need to load module for that? |
| 02:21.05 | rob0 | well ... you would probably want sound for *any* IAX/SIP client, no? :) |
| 02:21.17 | rob0 | unless you're faxing :) |
| 02:21.26 | *** join/#asterisk kristalino (n=kristali@cl-157.dub-01.ie.sixxs.net) |
| 02:21.33 | drcode | k |
| 02:29.33 | *** join/#asterisk _Vile (n=vile@bc182112.bendcable.com) |
| 02:29.43 | *** join/#asterisk tr2x (n=alvar@80-218-150-90.dclient.hispeed.ch) |
| 03:23.30 | *** join/#asterisk brif8 (n=brif8@rrcs-67-78-24-179.se.biz.rr.com) |
| 03:23.41 | Mportnoy | Qwell: hello ? |
| 03:24.03 | *** part/#asterisk brif8 (n=brif8@rrcs-67-78-24-179.se.biz.rr.com) |
| 03:28.15 | *** join/#asterisk Crumbles (i=Crumbles@71-81-29-162.dhcp.gwnt.ga.charter.com) |
| 03:28.41 | Crumbles | How can I make Asterisk know that I'm done dialing, after I've dialed 911 ? |
| 03:28.52 | Crumbles | instead of waiting for it to timeout |
| 03:28.56 | Crumbles | on an analog phone |
| 03:30.19 | *** join/#asterisk mishehu (i=mishehu@cshells.shavedgoats.net) |
| 03:36.29 | Crumbles | anyone? |
| 03:37.14 | *** join/#asterisk jeebusroxors (n=jeebusro@cpe-75-80-228-131.dc.res.rr.com) |
| 03:37.38 | jeebusroxors | *sigh* anyone run * on a sparc? or 64 bit system? |
| 03:37.40 | *** join/#asterisk zmef420 (n=zmef420@metarb3-pool1-11.mtco.com) |
| 03:41.15 | *** part/#asterisk justinu|laptop (n=Justin@cpe-76-170-91-201.socal.res.rr.com) |
| 03:48.31 | Math` | Crumbles: that depends on your ATA |
| 03:50.19 | Crumbles | can you suggest a good ATA? |
| 03:50.20 | *** join/#asterisk bmg505 (n=leon@c1-73-16.rndf.isadsl.co.za) |
| 03:52.06 | emphyrio | jeebusroxors I have had no problems with it on a pure 64bit gentoo install |
| 03:52.23 | Crumbles | can anyone suggest a good ATA to buy? |
| 03:52.51 | emphyrio | iaxy ^_^ |
| 03:52.51 | emphyrio | spa2k or the linksys equivalent also good |
| 03:53.26 | emphyrio | oooh speaking of which can anyone suggest an ar1688 based ata or phone that can currently be bought in the US? |
| 03:54.15 | Crumbles | so, the linksys spa-2002 linksys |
| 03:54.22 | Crumbles | is good is good? |
| 03:55.37 | emphyrio | I have only used sipura which is quite good, the linksys version is called pap2 and is identical. not sure which one can be had cheaper |
| 03:56.00 | *** join/#asterisk coppice (n=chatzill@213.143.17.210.dyn.pacific.net.hk) |
| 03:56.44 | jeebusroxors | emphyrio: did you have to use anything? my pbx_functions.so is missing |
| 03:58.02 | emphyrio | jeebusroxors hmmm all have been doing is installing from the stock ebuild, currently 1.2.13 |
| 03:58.02 | emphyrio | $ slocate pbx_fun |
| 03:58.02 | emphyrio | /usr/lib64/asterisk/modules/pbx_functions.so |
| 03:58.22 | *** join/#asterisk JT (n=jon@unaffiliated/jt) |
| 03:59.15 | jeebusroxors | emphyrio: was that a pc? |
| 04:00.01 | emphyrio | sorry yes it is. I thought you meant a sparc OR a 64bit...? |
| 04:00.15 | jeebusroxors | emphyrio: both heh, sparc64 |
| 04:00.22 | emphyrio | oic ^_^ |
| 04:01.35 | emphyrio | does solaris separate lib dirs for 64 and 32? it could just be a library path problem... |
| 04:01.50 | jeebusroxors | emphyrio: im running debian on it |
| 04:02.43 | emphyrio | jeebusroxors oh ok. are you building from source or installing a package? |
| 04:02.49 | jeebusroxors | emphyrio: both lol |
| 04:02.59 | jeebusroxors | this one was source |
| 04:03.33 | emphyrio | does the library get built at all? |
| 04:04.44 | jeebusroxors | how would i check that? |
| 04:05.13 | emphyrio | hm are you building from svn or tarball, or using dpkg or something? |
| 04:05.34 | jeebusroxors | tarball |
| 04:06.11 | emphyrio | ok in the funcs dir is where pbx_functions.so gets built so it should be there assuming there was not any build error |
| 04:06.22 | jeebusroxors | emphyrio: no it wasnt |
| 04:06.36 | jeebusroxors | there were no errors apparent, it went through config make and make install... |
| 04:06.43 | jeebusroxors | the rest of the libs are there... |
| 04:07.05 | emphyrio | odd since the main Makefile just makes subdirs and does not have dependencies on arch I think |
| 04:07.15 | *** join/#asterisk SwK (n=Silik0nJ@12-214-191-109.client.mchsi.com) |
| 04:07.56 | emphyrio | jeebusroxors did any objects get built in funcs/ at all, like pbx_functions.o? |
| 04:08.19 | jeebusroxors | emphyrio: where is that? |
| 04:08.56 | emphyrio | jeebusroxors also in funcs, are there any .o files at all? |
| 04:09.04 | jeebusroxors | /var/lib/ast? |
| 04:09.18 | emphyrio | oh sorry no I mean where you unzipped the tarball |
| 04:09.25 | jeebusroxors | oh ok |
| 04:09.39 | jeebusroxors | oh... .o's heh ima dumbass |
| 04:10.46 | jeebusroxors | http://pastebin.isawsome.net/pastebin.php?show=83 |
| 04:11.17 | jeebusroxors | i dont see it |
| 04:11.43 | emphyrio | why yes I am an idiot... sorry I am still back in 1.2 days, guess things were rearranged for 1.4 |
| 04:12.07 | jeebusroxors | hm |
| 04:12.26 | emphyrio | try from the top dir something like |
| 04:12.26 | emphyrio | find . -name "pbx_functions.*" -ls |
| 04:12.33 | jeebusroxors | yea its goin |
| 04:12.44 | jeebusroxors | nothin... |
| 04:12.48 | emphyrio | alternately maybe it doesnt exist at all in 1.4? |
| 04:13.13 | jeebusroxors | you have 1.4 installed? |
| 04:13.46 | emphyrio | no like I said 1.2.13 |
| 04:14.09 | jeebusroxors | well..i took it out and i boots... |
| 04:14.28 | emphyrio | wait you had pbx_functions.so in modules.conf? |
| 04:14.50 | jeebusroxors | yea |
| 04:14.53 | emphyrio | 'cos there is various stuff that is different now in the config files I know |
| 04:15.03 | jeebusroxors | heh, i guess i have a 1.2 install here... |
| 04:15.09 | jeebusroxors | didnt even notice the 1.4 :\ |
| 04:15.18 | emphyrio | 1.2 style config files yeah sounds like |
| 04:15.57 | emphyrio | ugh that is one thing I have been meaning to do, figure out exactly what modules I need so I can get rid of auto loading |
| 04:16.08 | jeebusroxors | autoloading is bad mmkay ;) |
| 04:16.39 | jeebusroxors | yea |
| 04:16.48 | jeebusroxors | man i feel like a dumbass... |
| 04:17.03 | emphyrio | hey you got it working that is all that counts |
| 04:17.50 | jeebusroxors | lol i always ask for help and someone tells me something and im like OOOOH YEA!! *slap forehead* |
| 04:17.58 | *** join/#asterisk techie (n=techie@c-67-182-22-174.hsd1.ca.comcast.net) |
| 04:18.30 | emphyrio | eh |
| 04:18.42 | emphyrio | this is why the intarnets is a good thing |
| 04:18.53 | jeebusroxors | mmhmm |
| 04:19.00 | *** join/#asterisk Joel1978 (n=Joel1978@71.39.87.5) |
| 04:20.01 | emphyrio | I am just hoping to find some nice soul's example providers.conf so I can get asteriskNOW to actually do something useful.... |
| 04:20.16 | jeebusroxors | heh |
| 04:20.21 | jeebusroxors | wow 1.4 is so different |
| 04:20.25 | jeebusroxors | why would you change it so much |
| 04:20.41 | emphyrio | subtle admission of kludgy problems in earlier codebase |
| 04:20.48 | jeebusroxors | meh |
| 04:20.54 | jeebusroxors | now i have to relearn everything |
| 04:21.10 | jeebusroxors | and the user friendlyness is paramount! |
| 04:21.20 | emphyrio | yeah I am not looking forward to having to rewrite my extensions.conf for 1.4 |
| 04:21.55 | Joel1978 | is there an outline or article about the differences between 1.2 and 1.4? |
| 04:22.13 | emphyrio | there's console debug errors :) |
| 04:23.13 | jeebusroxors | lol how the hell do i restart!? |
| 04:24.41 | Joel1978 | stop now then reboot? |
| 04:24.42 | Joel1978 | lol |
| 04:25.02 | emphyrio | reload will catch most changes |
| 04:25.14 | Joel1978 | except for zapata.conf |
| 04:25.17 | Joel1978 | but ya |
| 04:26.47 | *** part/#asterisk Joel1978 (n=Joel1978@71.39.87.5) |
| 04:27.05 | *** join/#asterisk Tako-san (n=jmkiffia@142.179.12.58) |
| 04:27.49 | jeebusroxors | there is no restart or stop... |
| 04:29.19 | jeebusroxors | nm...messed up modules conf |
| 04:38.07 | *** part/#asterisk techie (n=techie@c-67-182-22-174.hsd1.ca.comcast.net) |
| 04:42.52 | codefreeze | jeebusroxors, Joel1978: In the 1.4 release dir, there's UPGRADE, and CHANGES; get out the hot chocolate, put up your feet, plump up the cushions. They are both a long read. |
| 04:43.16 | jeebusroxors | yea, i got the manual ill read through |
| 04:44.01 | *** join/#asterisk [F] (n=f@pool-71-163-158-24.washdc.fios.verizon.net) |
| 04:44.15 | [F] | does all an Asterisk PBX require is a box with an ethernet card |
| 04:44.39 | [F] | to have conferencing, voicemail, caller-id, etc. ? |
| 04:45.06 | jeebusroxors | yes |
| 04:45.14 | [F] | jeebusroxors: wow. i am a moron. |
| 04:45.18 | [F] | i was waiting to get enough cash for this: |
| 04:45.20 | jeebusroxors | lol |
| 04:45.27 | [F] | http://www.voipsupply.com/product_info.php?products_id=1646 |
| 04:45.34 | [F] | i was thinking |
| 04:45.37 | [F] | i could do nothing without that. |
| 04:46.18 | [F] | what does that SPA do theN? |
| 04:46.22 | [F] | if its not necessary? |
| 04:46.41 | codefreeze | jeebusroxors, Joel1978: As one of the [many] perpetrators of the many changes, all I can say in my/our defense, is that there's a lot of cool things added. Hopefully, cool enough to justify the time to upgrade! |
| 04:47.44 | jeebusroxors | yea...its just annoying....every command i have typed in is depreciated |
| 04:48.12 | emphyrio | l8r sk8rs |
| 04:49.02 | codefreeze | jeebusroxors: Ha, yes, they felt the CLI needed some "organizing". And in 1.6, it will see some "reorganizing", but not as major. |
| 04:49.33 | jeebusroxors | codefreeze: its not too bad...after my initial frustration i am calming down and i see the patern.... |
| 04:49.48 | jeebusroxors | would anyone happen to know the FWD extension for echo tests? |
| 04:50.19 | [F] | question: does asterisk work on netbsd? |
| 04:50.32 | codefreeze | [F]: last I heard, yes. |
| 04:51.15 | codefreeze | [F]: that is, FreeBSD... I'm woefully ignorant of my BSD-isms. |
| 04:51.25 | [F] | codefreeze: alright. |
| 04:54.41 | Luke-Jr | [F]: IIRC, not all of it |
| 04:54.52 | [F] | Luke-Jr: not all of what? |
| 04:56.24 | Luke-Jr | for example, I don't think zaptel works |
| 04:56.55 | [F] | question: do I need a digium card for conferencing with asterisk? |
| 05:02.45 | codefreeze | [F]: as I understand it, no, not really. You can set up ztdummy, and use that for timing. That's the real issue in using conferencing without a zaptel [digium] card-- timing. 8Khz interrupts. |
| 05:04.02 | codefreeze | [F]: wait, 1Khz; 8 bits per byte. |
| 05:04.03 | *** join/#asterisk axisys (n=axisys@c-69-143-190-152.hsd1.va.comcast.net) |
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| 05:55.00 | Joel1978 | anyone able to get the asterisknow.org site to load? |
| 05:55.19 | Joel1978 | blah...nevermind |
| 05:55.22 | Joel1978 | they changed it to .com |
| 06:00.28 | Joel1978 | wow that site is really messed up! |
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| 06:57.36 | *** join/#asterisk brannfenix (n=brannfen@ip68-230-133-70.ri.ri.cox.net) |
| 06:59.29 | brannfenix | hi :) im looking to get 2 lower end display phones for my business asterisk system... I these will not be really hardcore used phones so my main goal is to get something on the cheap that isnt junk |
| 06:59.40 | brannfenix | can someone please recommend some models ? |
| 07:00.49 | [TK]D-Fender | brannfenix : Polycom IP 301, Aastra 9112i |
| 07:01.21 | brannfenix | also im looking for an adapter to hookup my fax machine to the box... so any leads on a good something for that would be awesome too |
| 07:01.31 | brannfenix | let me have a look at those |
| 07:05.19 | *** join/#asterisk aao_pwner (n=aao_pwne@c-24-21-91-140.hsd1.mn.comcast.net) |
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| 07:14.46 | argos73 | i like the polycoms |
| 07:16.06 | *** join/#asterisk mosty (i=mostyn@60-241-198-194.static.tpgi.com.au) |
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| 08:17.51 | *** join/#asterisk Terinjokes (n=spader@adsl-144-240-152.mia.bellsouth.net) |
| 08:18.17 | Terinjokes | hey, can i use asterisk without all the fancy telephone hookups and stuff? |
| 08:18.53 | mosty | what do you mean by telephone hookups? |
| 08:19.20 | mosty | you can use it just voip gear, if that's what you mean |
| 08:19.30 | mosty | with just voip gear, i mean |
| 08:20.16 | Terinjokes | i mean, i don't actually need to have/configure telephones, if i just want to setup asterisk to be a voicemail for some SIP lines? |
| 08:20.24 | *** part/#asterisk VoipMasta (n=fabio@dial-148-240-36-84.zone-1.dial.net.mx) |
| 08:20.56 | mosty | all you need is asterisk, plus a network card then |
| 08:21.26 | Terinjokes | i got the asterisk sources (check), i'm online (check) |
| 08:23.52 | Terinjokes | how long does it take for asterisk to compile? |
| 08:24.49 | mosty | depends how fast your machine is, probably about 5 minutes |
| 08:26.09 | mosty | what kind of cpu? |
| 08:26.22 | Terinjokes | PPC |
| 08:26.42 | SwK | what kinda PPC |
| 08:26.46 | mosty | ahh, well i've only used asterisk on p4 2.8G and up |
| 08:26.52 | SwK | its doesnt take that long on my G4 1Ghz |
| 08:28.29 | *** join/#asterisk Joel1978 (n=Joel1978@71.39.87.5) |
| 08:28.31 | Terinjokes | I'm testing asterisk on my iBook G4 1.33Ghz, after i understand it, move it over to the server |
| 08:28.59 | SwK | should work fine |
| 08:29.01 | Terinjokes | nice ASCII on ./configure BTW :) |
| 08:29.15 | SwK | I do 1/2 my dev on my g4 power book |
| 08:30.02 | Joel1978 | does anyone know the initial username and password for the GUI login on AsteriskNOW? |
| 08:30.19 | *** join/#asterisk jacekowski (n=jacek@aaof81.neoplus.adsl.tpnet.pl) |
| 08:30.26 | mosty | joel: never heard of "asteriskNOW", what do their docs say? |
| 08:31.05 | SwK | mostly thats digiums version of trixbox (for lack of a better analogy) that was released this week |
| 08:31.06 | Joel1978 | AsteriskNOW is Digiums answer to TrixBox.....there are no docs except a quick start guide. |
| 08:31.08 | SwK | its brand new |
| 08:31.10 | Joel1978 | ya |
| 08:31.20 | Joel1978 | it would be nice if they gave you the information to login and setup |
| 08:31.20 | Joel1978 | lol |
| 08:31.36 | SwK | theres a conf file in there somewhere I am sure |
| 08:31.45 | Terinjokes | mosty, it's kinda in the channel's topic |
| 08:32.11 | *** join/#asterisk dlynes_laptop (n=dlynes@S0106001346f7843f.vc.shawcable.net) |
| 08:32.31 | mosty | i don't see "asteriskNOW" in the topic |
| 08:32.53 | Terinjokes | mosty: "Join #asterisk-gui to learn about the new Asterisk GUI framework" |
| 08:33.22 | Joel1978 | they acutally have asterisk-now for support on it but no one responded in the room...i waited for probably 15 minutes |
| 08:33.38 | Joel1978 | so i thought i'd ask |
| 08:33.45 | Terinjokes | true it doesn't say 'asteriskNow', but i said 'kinda' |
| 08:33.53 | mosty | oh ok |
| 08:33.56 | Joel1978 | someone in here earlier had been discussing it |
| 08:34.33 | dlynes_laptop | asterisk-gui is usually busy during daylight hours, north american time |
| 08:34.38 | dlynes_laptop | other than that, it's usually dead |
| 08:34.54 | dlynes_laptop | as it gains momentum, i'm guessing it'll be busy round the clock |
| 08:34.56 | Joel1978 | i just popped in there...thanks for the suggestion |
| 08:35.05 | Joel1978 | i hope it gains momentum |
| 08:39.16 | dlynes_laptop | btw |
| 08:39.37 | dlynes_laptop | does anyone know of a supplier for machines that will power back on automatically after a power outage? |
| 08:39.57 | Terinjokes | hrm, most modern computers will... |
| 08:40.05 | dlynes_laptop | nah...most modern computers are atx |
| 08:40.14 | dlynes_laptop | not at |
| 08:40.30 | mosty | it's a bios setting on most modern computers |
| 08:40.37 | dlynes_laptop | if you short out the power switch on an atx machine, it'll go into a reboot cycle |
| 08:40.52 | Joel1978 | sometimes i've seen people enable the Wake On Lan if the BIOS supports it and the LAN activity will turn it on |
| 08:40.59 | Joel1978 | but that's only on certain BIOS sets |
| 08:41.15 | dlynes_laptop | wake on lan...isn't that for coming back out of suspend mode? not power-down? |
| 08:41.35 | Joel1978 | well i once had a few computers that would turn on immediately after i powered them down |
| 08:41.36 | mosty | dlynes_laptop, the dell bios lets you select "stay off" "turn on" or "last state" upon the return of power |
| 08:41.43 | Terinjokes | on my computer it turns back on after a power outage (i know, as I normally shut it down by killing the power) |
| 08:42.02 | dlynes_laptop | ah, cool |
| 08:42.13 | dlynes_laptop | I've only dealt with the cheap generic computers |
| 08:42.18 | dlynes_laptop | Never dealt with dell |
| 08:42.29 | dlynes_laptop | Do the dell poweredge machines also have that option? |
| 08:43.05 | mosty | the sc430 does, but i think it's fairly common to most if not all of the other dell machines |
| 08:46.19 | mosty | all my other computers have similar settings, they are all name-brand motherboards |
| 08:46.45 | dlynes_laptop | ic |
| 08:48.31 | Terinjokes | ok, time for real life, will mess around with asterisk tomorrow (i guess i need to get asterisk to pickup my SIP number, and then play out an audio file) |
| 08:48.48 | Terinjokes | this seems about right? |
| 08:50.04 | dlynes_laptop | asterisk -rx "stop when convenient" |
| 08:50.36 | *** join/#asterisk lorinc (n=ang@caracas-5075.adsl.interware.hu) |
| 08:51.03 | Terinjokes | hrm, asterisk is running in foreground mode (i guess i need to set it up as a deamon as well) |
| 08:51.21 | dlynes_laptop | ah, in foreground, just type 'stop when convenient' at the CLI |
| 08:51.48 | Terinjokes | sorry, no such command |
| 08:52.14 | Terinjokes | nvm, typo |
| 08:52.34 | dlynes_laptop | heh...was gonna say |
| 08:54.18 | Terinjokes | though README tells me i can type 'dial', but that fails... |
| 08:54.27 | Terinjokes | 'night |
| 08:55.08 | *** part/#asterisk Terinjokes (n=spader@adsl-144-240-152.mia.bellsouth.net) |
| 09:09.33 | SheriF_SpacE | i read about asterisknow |
| 09:09.51 | SheriF_SpacE | but i don't want to get a full ISO made by rpath can i get the GUI sperated ? |
| 09:10.54 | Joel1978 | ha |
| 09:11.29 | Joel1978 | there's so much unknown about it....i got it installed from the ISO but i couldn't find documentation on which username / password to use |
| 09:11.56 | Joel1978 | but i just came across this: http://astrecipes.net/?n=217 |
| 09:12.17 | Joel1978 | instructions on how to use SVN to checkout the source, compile and setup w/o rPath |
| 09:13.30 | SheriF_SpacE | Joel1978: thx ;-) |
| 09:14.08 | Joel1978 | i hope that helps...i haven't tried it out yet |
| 09:17.57 | *** join/#asterisk Xen^ (n=linux@unaffiliated/lnux/x-10290) |
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| 09:34.42 | *** join/#asterisk knarfly (n=bwoodwar@c-65-34-177-3.hsd1.fl.comcast.net) |
| 09:35.43 | knarfly | newbie here...with not much experience patching code...what is the syntax for applying the patches to the latest * tar.gz files ? 8-) |
| 09:37.02 | mosty | uncompress the .tar.gz files first |
| 09:37.29 | dlynes_laptop | knarfly: normally patch < filename.patch |
| 09:38.10 | *** join/#asterisk dasenjo (n=dasenjo@208.195.215.94) |
| 09:38.52 | *** join/#asterisk Slimey (n=simon@virtual.bogons.net) |
| 09:39.52 | Slimey | anyone awake can help with a Set() / 1.4 question? |
| 09:40.08 | dlynes_laptop | Slimey: Set's changed again? |
| 09:40.09 | dlynes_laptop | damn |
| 09:40.21 | Slimey | well |
| 09:40.27 | Slimey | I just upgraded from 1.2.9.1 to 1.4-beta3 |
| 09:40.47 | Joel1978 | somewhere in voip-info.org i found a half-way decent primer on the new way |
| 09:40.50 | dlynes_laptop | Yeah...Set went through a bit of a change from 1.0 to 1.2, too |
| 09:40.51 | Slimey | I used to have the line: |
| 09:40.54 | Slimey | exten => 155,2,Set(MEETME_RECORDINGFILE_1=/var/spool/asterisk/meetme/meetme-conf-rec-${CONFNO}-${TIMESTAMP}) |
| 09:41.12 | Slimey | now, it's expanding the ${CONFNO} and ${TIMESTAMP} before it gets to MeetMe() |
| 09:41.35 | Joel1978 | there needs to be a few more brackets i believe? |
| 09:41.45 | Slimey | I've tried backslashing the $ and that doesn't work |
| 09:41.52 | dlynes_laptop | Slimey: Isn't that the way you expect it to work? |
| 09:42.08 | Joel1978 | heh :) |
| 09:42.10 | Slimey | no, because it won't know CONFNO until the MeetMe() app has asked for it |
| 09:42.16 | dlynes_laptop | Slimey: i.e. to expand it in-place? |
| 09:42.24 | Slimey | as a programmer, yes, I'd expect that |
| 09:42.28 | Slimey | but in 1.2 it didn't |
| 09:42.53 | dlynes_laptop | oh...that is weird...I would never expect it to expand it after the fact :) |
| 09:43.09 | dlynes_laptop | i would expect it to expand it when it executes that line :0 |
| 09:43.49 | *** join/#asterisk SheriF_SpacE (n=sherif@chatty.shellshark.net) |
| 09:43.51 | Slimey | yeah |
| 09:43.58 | Slimey | which is why I'm wondering how it worked at all in 1.2 ;) |
| 09:45.16 | knarfly | dlynes_laptop: do I patch the tar.gz file or do I unpack it first? I'm not understanding which state the file is to be in before I apply the patch? |
| 09:45.28 | Slimey | untar first |
| 09:45.31 | Slimey | then apply patch |
| 09:45.35 | dlynes_laptop | tar zxvf filename.tar.gz |
| 09:45.44 | dlynes_laptop | then find the .patch file, and do patch < filename.patch |
| 09:47.36 | knarfly | okay...I'll give it a whirl |
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| 10:12.53 | knarfly | dlynes_laptop: I'm sooooo confused...can you offer some more advice on the patch? |
| 10:13.11 | *** join/#asterisk Theuni (n=ctheune@uter.gocept.com) |
| 10:13.18 | Theuni | good morning (or whatever) |
| 10:13.48 | dlynes_laptop | knarfly: how could i? I don't even know what patch you're applying |
| 10:13.49 | Joel1978 | here's a patch how-to: http://www.linuxhq.com/patch-howto.html |
| 10:14.03 | knarfly | the patch for *-1.4.0-beta3 came as a gz file not tar.gz. So I gunzip'ed it and now I have a text file called asterisk-1.4.0-beta3 |
| 10:14.17 | dlynes_laptop | knarfly: ah |
| 10:14.23 | Theuni | I'm trying to get attended transfer working for my setup and have some problems. I want to activate the ATXFER feature with a single key-stroke, as some of my phones have very small key and people can't hit them fast enough. |
| 10:14.28 | knarfly | Joel1978: thanks...I'll check it out |
| 10:14.28 | dlynes_laptop | knarfly: you probably need to download beta 1 first |
| 10:14.35 | *** join/#asterisk sergee (n=opera@195.94.224.197) |
| 10:14.37 | dlynes_laptop | knarfly: then download the beta2 patch |
| 10:14.43 | dlynes_laptop | knarfly: apply the beta2 patch |
| 10:14.44 | Theuni | I'm stumbling over the problem that I have normally called people and a queue. |
| 10:14.48 | dlynes_laptop | knarfly: then apply the beta3 patch |
| 10:15.26 | Theuni | I seem not to be able to map "#" to ATXFER, it seems to always endup as an unattended transfer. And using "*" doesn't work for the queue. |
| 10:15.29 | knarfly | dlynes_laptop: no I got it working with the *-beta3 , which really surprised me..I was just going to apply the patch and rebuild |
| 10:15.30 | Theuni | Any tips? |
| 10:15.54 | dlynes_laptop | Theuni: is this for a sip channel? |
| 10:16.04 | Theuni | both. SIP and EuroISDN mixed. |
| 10:16.10 | *** join/#asterisk adorah (n=admin@84.94.112.67.cable.012.net.il) |
| 10:16.12 | sergee | patch for tarnsfer available now http://bugs.digium.com/view.php?id=8413 - any volounteers ro test it? :) |
| 10:16.25 | sergee | to* |
| 10:16.27 | Theuni | both channels work in general, i just can't get a single key-combo to work. |
| 10:16.32 | Theuni | sergee: talking to me? :) |
| 10:16.57 | sergee | Theuni: you are welcome :) |
| 10:17.16 | knarfly | if I go to the /usr/src/asterisk-1.4.0-beta3 folder, where I originally untarred the source, and then do patch < asterisk-1.4.0-beta3-patch, how will it know which file to patch...I'm lost here....but then again I'm not that up on programming and patching. |
| 10:17.54 | Theuni | sergee: hmm. does |
| 10:18.02 | Theuni | sergee: hmm. does'nt look like what I need ... |
| 10:18.18 | dlynes_laptop | knarfly: the patch files are usually a conglomeration of patches to multiple files |
| 10:18.34 | dlynes_laptop | knarfly: typically you go to the root directory of the source, and run the patch |
| 10:18.46 | knarfly | okay...i think i'm getting it....back in a flash |
| 10:18.51 | dlynes_laptop | knarfly: i can't say for sure that's how you do it with that specific patch, but that's how you normally do it |
| 10:18.51 | sergee | Theuni: and what do you need? |
| 10:19.38 | Theuni | sergee: I need to be able to map ATXFER to either "#" or "*" reliably (again: using "#" fails for me, it always becomes unattended, using "*" doesn't work with queues) |
| 10:20.16 | sergee | The_Ritz: yeah, that's the different storry :) |
| 10:20.28 | sergee | Theuni: yeah, that's the different storry :) |
| 10:21.11 | Theuni | you don't have an answer for that story, I guess ;) |
| 10:21.35 | sergee | Theuni: nope :) |
| 10:21.37 | knarfly | okay that worked but I got alot of Reverse or previously applied patch detected messages....assume -R (y). I guess this means the patch for beta3 was to patch to be applied to beta2. beta3 does not need the patch applied...? |
| 10:24.07 | dlynes_laptop | probably not, no |
| 10:24.20 | knarfly | dlynes_laptop: thanks dude...your advice is always appreciated. |
| 10:24.26 | dlynes_laptop | I don't see why they would call the patch for beta3 to become something else as the fielname you gave me |
| 10:24.27 | adorah | Hi does anyone here using ASTLINUX or any other embedded build of *? |
| 10:24.30 | dlynes_laptop | it made no sense |
| 10:24.53 | Joel1978 | IMHO: stay away from embeded and learn it from scratch |
| 10:24.58 | dlynes_laptop | adorah: there's also another embedded version for running on Linksys routers and network appliances |
| 10:25.27 | knarfly | i am assuming that asterisk-1.4.0-beta3 already has the asterisk-1.4.0-beta3-patch applied |
| 10:25.34 | adorah | I need the tiny build for a very specifi implementation already learned it from scrath thx:) |
| 10:26.08 | adorah | <dlynes_laptop>which one? |
| 10:26.11 | knarfly | the beta3 patch is for *-1.4.0-beta2 and earlier....but don't quote me...I'm still fumbling my way through this |
| 10:26.42 | dlynes_laptop | adorah: one sec |
| 10:27.07 | knarfly | Joel1978: IMHO - the low cost of hard drives has made embedded systems take on a whole new meaning.... |
| 10:27.16 | dlynes_laptop | adorah: nslu2 |
| 10:28.00 | adorah | <dlynes_laptop>does it have some documentation? coz I find it hardto find the .conf files in AstLinux.. |
| 10:28.06 | dlynes_laptop | adorah: nslu2-asterisk is the group name on yahoogroups |
| 10:28.15 | knarfly | i found a 120GB drive yesterday for $39 |
| 10:28.20 | adorah | <dlynes_laptop>Thx |
| 10:28.21 | dlynes_laptop | adorah: don't know about docs, but it does have a mailing list |
| 10:28.48 | Joel1978 | wow |
| 10:28.49 | dlynes_laptop | adorah: and the fellow moderating the list is quite knowledgable about it...he's the one that actually developped it |
| 10:29.09 | Joel1978 | knarfly: was it 7200rpm? |
| 10:29.10 | Joel1978 | lol |
| 10:29.19 | adorah | <dlynes_laptop>Many thx.. |
| 10:29.27 | dlynes_laptop | adorah: His name is Alistair...he's on these chat channels from time to time; he's local to Vancouver, Canada |
| 10:29.46 | dlynes_laptop | adorah: He's actually part of Vancouver Linux Users' Group, too |
| 10:30.28 | adorah | Well I'm a bit far from Vancouver but I know the natives there r friendly:) |
| 10:30.40 | Joel1978 | haha sorta! |
| 10:30.41 | Joel1978 | jk |
| 10:30.42 | knarfly | yep |
| 10:31.38 | knarfly | ah....Vancouver....they got some great smoke in that town...on my way to Amsterdam in a few weeks as well to try some of the .....coffee! |
| 10:31.58 | dlynes_laptop | ahahaha |
| 10:32.13 | dlynes_laptop | Yeah...Blunt Brothers Cafe and the Amsterdam Cafe |
| 10:32.19 | Joel1978 | lol |
| 10:32.20 | adorah | OK another one did any one here tried the asterisk now GUI? is it a commercial or open? |
| 10:32.24 | dlynes_laptop | I think Amsterdam's closed down now, but Blunt Brothers is still there |
| 10:32.44 | Joel1978 | it's open and commercial |
| 10:32.47 | dlynes_laptop | Mark Emery got his ass busted, and now he's fighting extradition to the US |
| 10:32.56 | adorah | Bluntness is always good for business..hehe |
| 10:33.25 | knarfly | adorah: I'm trying to get it running today....I use FreeBSD so it's not so easy...all the docs are linux based and the chatrooms for #freebsd are full of "holier-than-thou" snobs with bad attitudes towards newbies. |
| 10:33.53 | dlynes_laptop | knarfly: asterisk installs pretty straightforward on freebsd |
| 10:34.04 | dlynes_laptop | knarfly: assuming you use the one in the ports tree |
| 10:34.37 | knarfly | dlynes_laptop: tell me more, please...where do I get it...no it's not in the ports tree. and the current port will only install *1.2.13 which does not run the gui...or am I wrong |
| 10:34.42 | adorah | <knarfly>Well FreeBSD users r hard nosed snubs..keeping themselves high above the Hoi-Poloy |
| 10:34.56 | Theuni | sergee: i solved it. i had to remap "#0" from blind transfer to "*something" to allow "#" to be picked up for attended transfer. |
| 10:35.26 | knarfly | adorah: some of us are just down to earth users...so don't lump me in with those a**holes I meet yesterday |
| 10:35.59 | knarfly | dlynes_laptop: I just updated my ports...is the gui in there....? |
| 10:36.06 | adorah | I've downloaded the asterisk NOW linux based but I couldn't configure anything there I don't like systems that doesn't let me access to the console |
| 10:36.08 | dlynes_laptop | knarfly: correct...you're hooped for 1.4 until the asterisk ports tree maintainer fixes up 1.4 to build on freebsd |
| 10:36.33 | dlynes_laptop | knarfly: he usually lags behind 1 or 2 minor versions, and doesn't usually do betas |
| 10:37.13 | knarfly | dlynes_laptop: yep, betas are taboo in FreeBSD ports...so is asking for help with them it would seem. |
| 10:37.35 | knarfly | dlynes_laptop: so where can I download the gui from? |
| 10:37.54 | dlynes_laptop | knarfly: same place as everyone else downloads it from |
| 10:38.02 | dlynes_laptop | knarfly: www.asterisk.org |
| 10:38.10 | dlynes_laptop | knarfly: and follow the links for asterisk-gui |
| 10:38.36 | dlynes_laptop | knarfly: you don't want asterisk-now...that's a complete distribution of linux and asterisk and asterisk-gui |
| 10:39.23 | *** join/#asterisk DaPrivateer (i=Privatee@crimson.66fruit.com) |
| 10:40.13 | knarfly | dlynes_laptop: I guess I'm missing the link to the gui....still looking. |
| 10:40.45 | Joel1978 | i'd have to agree with dlynes....it's not really a good option unless you don't really want to do anything creative |
| 10:41.17 | Joel1978 | http://www.astrecipes.net/index.php?from=1&q=astrecipes/installing+the+asterisk+gui |
| 10:41.18 | knarfly | No...I'm staying with FreeBSD...it's been my OS of choice now for 9 years |
| 10:41.37 | Joel1978 | that link has instructions on checking out the source for the gui via SVN |
| 10:41.45 | Joel1978 | or u can download via ftp |
| 10:42.17 | dlynes_laptop | knarfly: from the /topic for #asterisk-gui: Topic for #asterisk-gui is: Welcome to Asterisk GUI -- Asterisk finally has a (very alpha) GUI! || http://svn.digium.com/view/asterisk-gui/trunk/ || Screenshots: http://www.asterisk.org/node/111 |
| 10:42.21 | knarfly | Joel1978: I'm there dude...thanks |
| 10:42.41 | dlynes_laptop | knarfly: NOTE: "(very alpha)" |
| 10:42.59 | dlynes_laptop | would seem to kinda defeat the purpose of running freebsd |
| 10:43.51 | adorah | Well I wish for a GUI that won't take over the .conf files and will overwrite my manul configuration everytime I restart the system like FreePbx |
| 10:44.02 | Joel1978 | goodluck finding that |
| 10:44.35 | knarfly | dlynes_laptop: that's what I made this test system for yesterday...my real FreeBSD server is still a virgin...never had an alpha or beta and never will. |
| 10:45.02 | knarfly | adorah: backups dude...backups |
| 10:45.19 | *** part/#asterisk Theuni (n=ctheune@uter.gocept.com) |
| 10:45.32 | adorah | <knarfly>I don't get u.. |
| 10:45.54 | dlynes_laptop | adorah: what's not to get? |
| 10:45.54 | adorah | <knarfly>do u think I should restore every time I reboot? |
| 10:46.33 | dlynes_laptop | adorah: nah...he's just saying if freepbx overwrites yoru config files and borks them, you can always restore a previous backup |
| 10:47.04 | knarfly | no...I know it's a pain but sometimes you just gotta do what you gotta do when using Open Source (Open Sores) software... |
| 10:47.23 | Joel1978 | well freebpx has an extra config file with -custom.conf that you can put configs in that won't get overwritten |
| 10:47.35 | Joel1978 | u just have to be more creative in how you setup your contexts |
| 10:48.06 | adorah | Well I do reboots about every week for production systems..I don't think restoring once a week is a sensible commercial attitude.. |
| 10:48.37 | knarfly | adorah: script files dude...script files....cron baby...cron |
| 10:48.44 | Joel1978 | lol |
| 10:48.51 | adorah | yeah sure |
| 10:48.58 | Joel1978 | now don't start chatting in klingon next |
| 10:49.12 | knarfly | like I said...open sores software is a pain...but you can't beat the price |
| 10:49.31 | Joel1978 | ha...open sores |
| 10:49.46 | adorah | Just that backups is one of the thing I really don't like croned.. |
| 10:50.15 | lters | backups get easy with bacula |
| 10:51.21 | knarfly | hey Joel1978. that website says |
| 10:51.22 | knarfly | The first thing you'll have to do is to download the GUI, currently through Subversion only: |
| 10:51.36 | knarfly | can you tell me where the ftp is? |
| 10:52.50 | Joel1978 | ahhh ok...my bad i just assumed they had it on ftp but it makes sense since it's alpha that it's not up on ftp yet |
| 10:52.51 | Joel1978 | blah |
| 10:52.54 | Joel1978 | my bad |
| 10:53.00 | Joel1978 | im a wishful thinker i guess |
| 10:53.17 | dlynes_laptop | svn, dood |
| 10:53.19 | dlynes_laptop | svn |
| 10:53.37 | dlynes_laptop | lters: bacula? |
| 10:53.54 | lters | yes. great for the mix group |
| 10:54.09 | *** join/#asterisk ToTo (n=ToTo@host97-157-dynamic.2-87-r.retail.telecomitalia.it) |
| 10:54.11 | lters | xp/debian or whatever. |
| 10:54.21 | *** join/#asterisk daysmen3 (n=primus@host86-139-50-199.range86-139.btcentralplus.com) |
| 10:54.22 | Joel1978 | lol |
| 10:54.25 | knarfly | I'm still learning cvsup...guess I'll have to throw that one on the scrap heap now and move do subversion |
| 10:54.33 | lters | we use a 16 tape(400.400g) tape box |
| 10:54.42 | Joel1978 | cvs is crap |
| 10:54.49 | lters | s/400/400/ |
| 10:55.09 | Joel1978 | svn is easy to learn i mean it's a one line command to checkout the source |
| 10:55.15 | lters | 400 gig or 800 compressed |
| 10:55.26 | Joel1978 | but you need to have svn installed |
| 10:55.34 | lters | cvs/svn are nearly the same to use. |
| 10:55.36 | knarfly | Joel1978: it used to be the neatest thing since blunts....did I tell you guys, I invented blunts.... 8-) |
| 10:55.44 | Joel1978 | LMAO |
| 10:56.36 | knarfly | well, Al Gore helped a little bit |
| 10:57.02 | adorah | <knarfly>R u *tht* ancient? hehe |
| 10:57.18 | knarfly | ancient is as ancient does.... |
| 10:57.36 | adorah | weel almost like me..daa |
| 10:57.51 | knarfly | closing in on the big 50...if you know what I mean... |
| 10:58.01 | Joel1978 | actually he slipped up when he said he invented the internet....he was supposed to say something along the lines of kicking off the II2 |
| 10:58.12 | adorah | I do know more than u think..hehe |
| 10:58.54 | knarfly | yes but he invented global warming all on his own...maybe with a little help from Rev. Al Sharpton |
| 10:59.26 | adorah | Well the bloke is just to nice to be a president..that's all |
| 10:59.33 | adorah | =too.. |
| 11:00.29 | *** join/#asterisk Aximas (n=Aximas@gw-100.selfnet.de) |
| 11:01.25 | knarfly | Al Gore.....too nice.....Nice like Hillary is nice to her travel agents |
| 11:04.33 | Joel1978 | thankfully he won't be running again!!!! |
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| 12:06.58 | EmleyMoor | If I've seemingly hidden ekiga, how do I unhide it? (I appreciate it's not really an * query) |
| 12:08.02 | *** join/#asterisk backblue (n=moo@87-196-100-233.net.novis.pt) |
| 12:13.03 | xheliox | It should be in your tray, as long as you have a tray applet enabled. |
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| 12:22.31 | qdk | can i combine templates with realtime sip-account? |
| 12:53.12 | *** join/#asterisk apardo (n=apardo@87.217.147.43) |
| 13:02.48 | *** join/#asterisk tRSS (n=tRSS@124.29.203.59) |
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| 13:07.49 | tRSS | i have a quick question: i have a conexant modem installed in my asterisk machine and i am suspicious that it is getting detected as an fxo card. here is http://pastebin.ca/257521 what i have in my messages log. can someone explain line 3 and line 7 on pastebin? |
| 13:08.18 | *** join/#asterisk henrique (n=henrique@201-1-135-75.dsl.telesp.net.br) |
| 13:09.00 | tRSS | ooh, and the modem is a pci win(soft)modem |
| 13:09.38 | isladelobos | Ein? Conexat modem 56k? |
| 13:09.47 | tRSS | yes sir |
| 13:10.19 | isladelobos | Work fine? |
| 13:10.33 | tRSS | never used it as a modem :) |
| 13:14.04 | tRSS | i think, it is using ztdummy, and hence it is loading the USB wildcard driver (wcusb).... i guess, i get it now :) but if i am wrong, maybe someone can correct me |
| 13:14.50 | shellshark | ztdummy should not load wcusb |
| 13:15.14 | *** join/#asterisk SheriF_SpacE (n=sherif@chatty.shellshark.net) |
| 13:15.37 | SheriF_SpacE | how can i force asterisk to forget about UNREACHABLE SIP node ? |
| 13:16.04 | tRSS | then may be you can help me understand my var/log/messages, pasted here: http://pastebin.ca/257521 |
| 13:17.27 | rob0 | Maybe hotplug loaded something else? What drivers ARE loaded? tor2? |
| 13:18.46 | tRSS | it says it registers a Tormenta2 PCI. i am not sure, what that is? I have nothing plugged in my usb ports. I have a lan card (built in my mobo) and a modem (conexant chipset) in the pci slot of the mobo |
| 13:20.03 | rob0 | lsmod, find the Zaptel drivers. |
| 13:20.33 | tRSS | but when I add a fxsks=1 in zaptel.conf and do a ztcfg -vvv, i get this: ZT_CHANCONFIG failed on channel 1: Invalid argument (22) |
| 13:20.45 | rob0 | FWIW I believe there's little difference between a softmodem and a FXO card. |
| 13:21.36 | tRSS | actually, i have read on voip-info that x100p was basically a modem and certain modems can be detected as a single port fxo (x100p clone) by asterisk. I could be wrong though |
| 13:22.49 | tRSS | results from lsmod are pasted here: http://pastebin.ca/257526 |
| 13:22.50 | rob0 | "Being detected as FXO" and "being made to work as FXO" might be a different story. |
| 13:23.40 | rob0 | Yikes, that's a bunch of zaptel drivers! |
| 13:24.22 | tRSS | true.... hence, the reason, I want to make sure. I don't need an fxo port really, but it would be nice, if this one works :) |
| 13:24.38 | shellshark | it wont work |
| 13:24.55 | tRSS | shellshark: i am cool with that. |
| 13:26.46 | *** part/#asterisk jeffik (n=Jeff@pool-71-101-218-122.tampfl.dsl-w.verizon.net) |
| 13:27.20 | tRSS | zttool should be able to tell me if a card was successfully detected as an fxo, correct? because zttool is only showing me ztdummy |
| 13:29.12 | *** join/#asterisk Koshatul (n=evangeli@ppp188-33.static.internode.on.net) |
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| 13:40.13 | SheriF_SpacE | how can i force asterisk to forget about UNREACHABLE SIP node ? |
| 13:44.00 | *** join/#asterisk daysmen3 (n=primus@host86-139-50-199.range86-139.btcentralplus.com) |
| 13:44.51 | dlynes_laptop | SheriF_SpacE: just fail gracefully and handle it appropriately |
| 13:45.14 | dlynes_laptop | SheriF_SpacE: based on your ${DIALSTATUS} variable when you try to call that unreachable sip node |
| 13:46.37 | *** join/#asterisk moon06 (n=michael@cim06-1-82-228-240-97.fbx.proxad.net) |
| 13:47.10 | SheriF_SpacE | dlynes_laptop: no now i have UNREACHABLE device okay i want asterisk to mark it as unknwen so the device can register again it's Nokia E60 . |
| 13:47.35 | dlynes_laptop | SheriF_SpacE: umm...asterisk isn't preventing it from registering |
| 13:47.43 | dlynes_laptop | SheriF_SpacE: the device just isn't re-registering |
| 13:47.59 | SheriF_SpacE | yes |
| 13:48.08 | *** join/#asterisk daysmen3 (n=primus@host86-139-50-199.range86-139.btcentralplus.com) |
| 13:48.14 | dlynes_laptop | SheriF_SpacE: i.e. it timed out on a re-register, and asterisk hasn't heard back from it since |
| 13:51.13 | *** join/#asterisk voopianer (n=chatzill@99.82-134-107.bkkb.no) |
| 13:52.16 | *** join/#asterisk daysmen3 (n=primus@host86-139-50-199.range86-139.btcentralplus.com) |
| 13:52.20 | voopianer | Greetings! Anybody with good knowledge of the Queue app? |
| 13:53.45 | dlynes_laptop | voopianer: lots of peeps |
| 13:53.48 | dlynes_laptop | just not me :) |
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| 14:14.36 | xheliox | voopianer: Whatcha need? |
| 14:16.04 | *** join/#asterisk daysmen3 (n=primus@host86-139-50-199.range86-139.btcentralplus.com) |
| 14:16.22 | saftsack | is the new chan_misdn which is delivered by asterisk the chan_misdn from beronet? |
| 14:19.11 | *** join/#asterisk daysmen3 (n=primus@host86-139-50-199.range86-139.btcentralplus.com) |
| 14:21.20 | *** join/#asterisk AsteriskAlbania (n=info@217.24.244.130) |
| 14:22.08 | AsteriskAlbania | What about FOIP, does ASTERISK support FAX from SIP to PSTN ? if anyone knows smth please reply, I read everywhere that is not yet supported |
| 14:22.18 | *** join/#asterisk daysmen3 (n=primus@host86-139-50-199.range86-139.btcentralplus.com) |
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| 14:26.08 | rob0 | daysmen3: This is annoying, please stop the quits/joins. |
| 14:26.31 | blitzrage | rob0: most clients let you turn off the join/part msg's |
| 14:27.10 | rob0 | I know, but his are every 3 minutes. :) I like to see comings and goings, but this is a bit much. |
| 14:27.52 | blitzrage | gotta take the bad with the good. If it's happening a lot, more than likely he isn't even there and it's his network connection |
| 14:28.22 | AsteriskAlbania | What about FOIP, does ASTERISK support FAX from SIP to PSTN ? if anyone knows smth please reply, I read everywhere that is not yet supported |
| 14:28.32 | *** join/#asterisk daysmen3 (n=primus@host86-139-50-199.range86-139.btcentralplus.com) |
| 14:29.09 | blitzrage | AsteriskAlbania: fax by nature does not allow SIP to PSTN and vice-versa unless you're converting it to T.38 first. Asterisk 1.4 branch has T.38 pass-through support |
| 14:30.06 | AsteriskAlbania | blitzrage: Is it possible to send fax from a ANALOG FAX connected to ATA HANDY TONE 386 registered to ASTERISK ? |
| 14:30.34 | blitzrage | AsteriskAlbania: that should work fine as long as its not going over a busy network. I'd say like 95% effective. |
| 14:30.59 | blitzrage | AsteriskAlbania: unless you try sending it over the Internet -- in which case it probably will not work. |
| 14:31.05 | blitzrage | Fax does not like packetized networks |
| 14:31.20 | AsteriskAlbania | I have tried it and all I hear is the FAX voice on the other side |
| 14:31.20 | blitzrage | If you keep it on a LAN, it might work |
| 14:31.25 | AsteriskAlbania | no handshaking done |
| 14:31.39 | *** join/#asterisk daysmen3 (n=primus@host86-139-50-199.range86-139.btcentralplus.com) |
| 14:31.39 | blitzrage | AsteriskAlbania: handshaking with what? Asterisk will not receive the fax. |
| 14:32.00 | blitzrage | AsteriskAlbania: sounds like you want Hylafax |
| 14:32.04 | AsteriskAlbania | handshaking between 2 faxes one connectet to ATA 386 and the other on PSTN |
| 14:32.38 | blitzrage | if you set up asterisk right, then one fax machine should call the other fax machine |
| 14:34.46 | *** join/#asterisk daysmen3 (n=primus@host86-139-50-199.range86-139.btcentralplus.com) |
| 14:34.49 | AsteriskAlbania | I call the fax on PSTN from teh fax connected to ATA 386 registered to asterisk , the FAZ on the PSTN side respond with the fax noise as you were calling it from a regular phone |
| 14:35.25 | voopianer | xheliox: Still there? |
| 14:36.11 | *** join/#asterisk DarKnesS_WolF (n=wolf@217.54.209.1) |
| 14:37.15 | xheliox | Kinda sorta. |
| 14:37.39 | voopianer | Does that mean you're busy? :) |
| 14:37.57 | *** join/#asterisk daysmen3 (n=primus@host86-139-50-199.range86-139.btcentralplus.com) |
| 14:38.01 | xheliox | Kinda, not really. |
| 14:38.05 | xheliox | What's the problem you're having? |
| 14:38.27 | voopianer | I'm looking for best practice in the following scenario: |
| 14:39.54 | voopianer | When incoming call is placed in a queue, I have two devices (sip devices) ringing. They are just ordinary members of the queue in queue.conf. After x seconds timeout, I'd like to include a third device. |
| 14:40.38 | voopianer | I have messed about with different approaches, but they all seem to mess up either timeouts, MOH or the messages played. |
| 14:40.48 | blitzrage | voopianer: you'll probably have to use the Local channel as a static member |
| 14:41.03 | voopianer | It seems to me it would be a common real worls scenario? |
| 14:41.23 | voopianer | I tried that too: |
| 14:41.30 | blitzrage | yep -- I do staggered dialing all the time, but you have to do some Local/ trickery |
| 14:42.07 | voopianer | In the local context, I had wait(x seconds), then dial(SIP/device3), but that kills the MOH |
| 14:42.22 | voopianer | ...and gives me ringing |
| 14:42.22 | blitzrage | use the 'm' flag in the Dial() ? |
| 14:42.36 | blitzrage | that'll give you MoH again |
| 14:42.54 | voopianer | That I have not tried... |
| 14:43.26 | voopianer | I guess then, that using local is best practice for such a need? |
| 14:43.57 | voopianer | In your oppinion? |
| 14:44.06 | blitzrage | well, it's the only way to branch multiple Dial()'s at a staggered interval |
| 14:45.00 | blitzrage | Once you get into a Dial(), you're stuck, so the only way to run multiple dialplan logics that call other Dial()'s at staggered intervals is by calling the Local/ channel to control your Dial()'s to the end points separately |
| 14:46.22 | *** join/#asterisk daysmen3 (n=primus@host86-139-50-199.range86-139.btcentralplus.com) |
| 14:46.33 | blitzrage | file: you are a tomato! |
| 14:46.42 | voopianer | Yes. As I said, I have tried that approach too. It makes sence to use the 'm' option though. Didn't think of that. What do you think would happen if I also had periodic announce? Will it stop? Of course I will try it anyway :) |
| 14:48.02 | blitzrage | it might or might not.... once you Dial(Local/....), the Queue() might not be able to send audio down the embedded Dial()'s, but not sure. But using the Local channel is pretty much the only way to stagger your Dial()'s to endpoints |
| 14:50.04 | voopianer | Ok. I'll try it and see. Thanks for your input (both of you :)) |
| 14:50.21 | *** join/#asterisk robin_z (n=robin@rszemeti.gotadsl.co.uk) |
| 14:51.33 | *** join/#asterisk bigjb (n=nnbigjb@195.60.10.114) |
| 14:52.48 | *** join/#asterisk Ebola (n=Ebola@host86-136-86-71.range86-136.btcentralplus.com) |
| 14:53.48 | saftsack | florz, hi are you available? |
| 14:53.59 | florz | saftsack: kindof |
| 14:55.00 | saftsack | florz, i want to undestand junghanns driver. do you have tips for me howto to this? i saw that the initial doing of the driver is are the fifos, is this right? and where does the driver communicate with the chip? |
| 14:55.43 | florz | saftsack: Which of the drivers exactly? |
| 14:56.35 | *** join/#asterisk daysmen3 (n=primus@host86-139-50-199.range86-139.btcentralplus.com) |
| 14:56.40 | saftsack | the easiest one would be the best (zaphfc?) |
| 14:57.14 | florz | saftsack: Well, that sepends on your aims :-) |
| 14:57.27 | florz | depends even |
| 14:58.29 | saftsack | i want to understand the driver and how it is working and furthermore i want to understand, what to do to implement reroute (the "anlagenanschluss" zapCD) |
| 15:00.06 | florz | saftsack: That would have to be done in asterisk's chan_zap, not in any hardware driver |
| 15:00.56 | saftsack | ok. i thought that is be done in a low level ;) but then i will look there. is there a documentation project for the driver? |
| 15:01.35 | saftsack | florz, can you give me a short description what the driver does and what not? i thought that the driver is also responsable for doing things like the full signalling. |
| 15:02.31 | robin_z | ok, so I reinstalled asterisk bri-stuffed on a new box (from the xorcomm CD) and have run genzaptelcofg and ztcfg and zttest is now at 100% .. this is good, right? |
| 15:02.57 | florz | saftsack: Well, there is the datasheet for the chip. When you've read that (and know a bit about computer architecture and linux driver programming), the driver is pretty much self-documenting |
| 15:04.12 | saftsack | my problem is that i dont know anything about linux driver programming and i wanted the driver from junghanns to be such a document ;) but i will try it with the datasheet :> thanks |
| 15:04.17 | robin_z | but .. heres the thing, the sync light is on on the ISDN box on the wall, so its seeing my zpahfc card .. but when I dial in, nothing happens .. it waits 10 seconds and then the line drops .. I see neothing at all on the asterisk CLI no matter how high I turn the debugging up ... |
| 15:04.49 | florz | saftsack: What it does is that it copies the data of both b and the d channels from the card to zaptel and the other way around - that's it. |
| 15:06.16 | saftsack | florz, ok thank you :) this seems to be a simple task ^^ so the hard task would be the one in zaptel, right? like implementing the features of the bri channels ... |
| 15:07.37 | saftsack | florz, the datasheet shows awesome functionalities for brdging many hfc-s cards :) |
| 15:08.08 | florz | saftsack: In case you wanna learn something about linux driver programming, this might be useful: http://www.xml.com/ldd/chapter/book/ |
| 15:08.21 | robin_z | sigh ... |
| 15:08.51 | saftsack | florz, thank <ou :) |
| 15:09.02 | florz | saftsack: And no, most of those features are actually implemented inside asterisk |
| 15:09.15 | florz | in userspace, that is |
| 15:09.54 | robin_z | so .. how can I dubug this .. the ISDN line is connected to the * box, but nothing seems to happen in * when a call comes in |
| 15:10.21 | florz | robin_z: have you tried dialing out? |
| 15:11.42 | robin_z | not yet ... |
| 15:12.01 | *** join/#asterisk Iamnacho (i=Iamnacho@ip68-103-140-124.ks.ok.cox.net) |
| 15:12.02 | robin_z | the CLI seems to be lacking a dial command |
| 15:14.03 | florz | robin_z: You are sure that you are using asterisk? |
| 15:15.08 | florz | robin_z: Or just not loaded chan_oss or chan_alsa? |
| 15:16.15 | robin_z | florz: probably no sound supprt on the server, anyway, i have a real phoen connected now |
| 15:17.13 | robin_z | doesnt dial out, just drops the call with Channel 1/1, span 1 got hangup |
| 15:17.13 | robin_z | <PROTECTED> |
| 15:18.21 | blitzrage | no dial command on the CLI unless you have chan_oss or chan_alsa loaded |
| 15:18.33 | SheriF_SpacE | very strange the noka E60 was registering very fine and i did a call and now it's getting UNAUTHORIZED and nothing changed |
| 15:21.04 | robin_z | well, anyway, the cli dial command is no big deal ... not worried |
| 15:21.16 | robin_z | so . debugging the zaptel stuff ... |
| 15:22.42 | florz | robin_z: could you put your zaptel.conf and your zapata.conf online somwhere? |
| 15:22.49 | *** join/#asterisk benjk (n=benjamin@f8a01-0357.din.or.jp) |
| 15:23.43 | robin_z | sure |
| 15:23.49 | robin_z | oh woow |
| 15:24.16 | blitzrage | ~pb |
| 15:24.24 | jbot | [pb] a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 15:24.29 | robin_z | <PROTECTED> |
| 15:24.53 | robin_z | Zaptel cards initial configuration: cat: /sys/module/xpp/parameters/zap_autoreg: No such file or directory |
| 15:24.53 | robin_z | Updating XPP sync source failed (used XPP_SYNC='') |
| 15:24.53 | robin_z | zaptel. |
| 15:25.12 | robin_z | whats that all about? |
| 15:25.29 | *** join/#asterisk ltd (n=z@202-161-1-61.dyn.iinet.net.au) |
| 15:27.06 | *** join/#asterisk riddlebox (n=james@75-132-205-166.dhcp.stls.mo.charter.com) |
| 15:27.13 | blitzrage | what version of zaptel did you checkout? |
| 15:28.03 | saftsack | florz, i cant find the dss1 protocoll description. there are just things like what it is in google. do you have an idea? |
| 15:28.52 | *** join/#asterisk derek (n=derek@ip68-3-24-234.ph.ph.cox.net) |
| 15:29.05 | robin_z | <PROTECTED> |
| 15:29.22 | robin_z | blitzrage: I used the bristuffed thing on the xorcomm cd |
| 15:29.56 | blitzrage | ahhhh... that's why the XPP |
| 15:30.21 | robin_z | xpp? |
| 15:30.55 | blitzrage | xpp is the stuff the xorcom guys use for their hardware |
| 15:31.46 | *** join/#asterisk lintechnokrats (n=chikki@61.17.68.129) |
| 15:31.50 | robin_z | so I would be better off with basic bri-stuffed build? |
| 15:32.03 | blitzrage | no idea to be honest. I've never compiled stuff for BRI |
| 15:32.11 | blitzrage | I'm in north america |
| 15:32.39 | robin_z | so ... anyway ... to debug this ... where should I start? |
| 15:34.24 | *** join/#asterisk perlmonk (n=pierre@201.38.238.66) |
| 15:36.43 | *** part/#asterisk Slimey (n=simon@virtual.bogons.net) |
| 15:37.43 | *** part/#asterisk perlmonk (n=pierre@201.38.238.66) |
| 15:50.27 | backblue | robin_z: why are you using bristuff? dont you have any chance of using misdn? |
| 15:53.24 | *** join/#asterisk tr2x (n=alvar@80-218-150-90.dclient.hispeed.ch) |
| 15:54.15 | florz | saftsack: Well, you can buy it from the ITU. Sometimes you find such things on the web with some luck. But apart from buying it, it's usually difficult to come by ... |
| 15:56.45 | coppice | DSS1 isn't an ITU protocol. its an ETSI protocol. you can download specs for free from ETSI |
| 15:57.16 | florz | Oh, IC. But I think ITU does have some number for it, too, no? |
| 15:58.12 | coppice | nope. the ITU defines the generic Q.921 and Q.931 protocols. others define specific variants. DSS1 is by far the most widely used variant, and its defined in ETSI specs |
| 16:00.10 | florz | makes sense, actually, given that it was a european development =:-) |
| 16:01.30 | florz | robin_z: The config looks fine, actually |
| 16:04.06 | florz | robin_z: you do have a PtP line? |
| 16:10.58 | robin_z | florz sorry, was a way fro kb |
| 16:11.11 | robin_z | florz: its point to multipoint |
| 16:11.16 | *** join/#asterisk Grind1 (n=Grind1@adsl-074-237-085-125.sip.mia.bellsouth.net) |
| 16:11.46 | *** join/#asterisk SomethingISODD (n=dan@209NTL226-89-30.nt.net) |
| 16:11.49 | robin_z | backblue: I have ISDN .. its on a little box on the wall |
| 16:11.58 | SomethingISODD | Hello, does anyone know of any applications that will enable me to broadcast iptv? |
| 16:12.12 | backblue | robin_z: that should answer my question? |
| 16:12.34 | robin_z | whats "misdn" ? |
| 16:13.09 | backblue | it's what you should be using insteed of bristuff |
| 16:13.14 | backblue | :) |
| 16:13.17 | robin_z | URL? |
| 16:13.23 | backblue | www.google.com |
| 16:13.51 | *** join/#asterisk k84 (n=pyro@86.84-48-44.nextgentel.com) |
| 16:14.28 | backblue | robin_z: it's from beronet (www.beronet.com), but the module for asterisk, comes allready with it. you just have to install mISDN drivers and libs, and chan_misdn for asterisk will be compiled also. |
| 16:14.50 | backblue | you have pretty good docs on their website, and it works much better than bristuff. |
| 16:19.38 | robin_z | hmmm |
| 16:19.44 | robin_z | looks very new |
| 16:20.54 | *** join/#asterisk Grind1 (n=Grind1@adsl-074-237-085-125.sip.mia.bellsouth.net) |
| 16:21.36 | robin_z | many people using it yet? |
| 16:23.33 | robin_z | im finding it hard enough to get help with bri ... I suspect misdn might be almost impossible to find anyone to ask .. |
| 16:23.42 | florz | robin_z: For PtMP, you should be using signalling bri_cpe_ptmpt |
| 16:23.48 | florz | erm |
| 16:23.52 | florz | bri_cpe_ptmp |
| 16:23.59 | robin_z | oh, OK |
| 16:24.39 | robin_z | span=1,1,1,ccs,ami in zaptel.conf is OK though? |
| 16:25.16 | *** join/#asterisk zmef420 (n=zmef420@metarb3-pool1-11.mtco.com) |
| 16:26.27 | robin_z | ahah ... now I get "Rejecting callwaiting SETUP on channel 0/1 span 1 1" when I try to dial in |
| 16:28.34 | florz | robin_z: Yep, should do. |
| 16:29.23 | robin_z | the "on channel 0/1" .. what does that mean? |
| 16:35.44 | backblue | robin_z: digium. :P |
| 16:36.03 | backblue | robin_z: i can answer anything to you about misdn. |
| 16:36.15 | backblue | and you have the misdn ML |
| 16:36.18 | backblue | that you can use. |
| 16:36.28 | robin_z | well ... hmmm |
| 16:36.30 | riddlebox | does anyone use the budget tone GXP 2000 phones? |
| 16:36.39 | robin_z | riddlebox: sure ... |
| 16:36.43 | backblue | riddlebox: i have tested them. |
| 16:36.45 | robin_z | riddlebox: I have several |
| 16:36.46 | backblue | but i dont use them |
| 16:37.10 | robin_z | backblue: they work perfectly for us ... we have found them ideal for severla uses |
| 16:37.14 | riddlebox | do they work good? I may get one for christmas |
| 16:37.21 | robin_z | backblue: holding doors open |
| 16:37.30 | riddlebox | its either that one or the aastra phone |
| 16:37.46 | backblue | they worked for us too, but we have better |
| 16:37.59 | robin_z | backblue: if you put one on top of the other, (one upside down) you can slide then to make a sort of adjustable height wedge ... |
| 16:38.35 | robin_z | riddlebox: what did you want to use them as? |
| 16:38.56 | riddlebox | robin_sz, just as a home phone thats all |
| 16:39.03 | robin_z | oh as a phone? |
| 16:39.12 | robin_z | no ... I wouldnt recommend them fo that use |
| 16:39.35 | riddlebox | not use a phone as a phone? |
| 16:39.49 | Supaplex | use it as a backscratcher? |
| 16:40.15 | robin_z | riddlebox: its not really a phone ... it may LOOK like a phone, but its not |
| 16:40.49 | riddlebox | what do you mean? the budget tone or the aastra |
| 16:41.02 | *** join/#asterisk shido6 (i=shido6@d221-68-200.commercial.cgocable.net) |
| 16:41.12 | robin_z | infact .. I can think of several household objects that are better SIP phones than the grandstream |
| 16:41.32 | robin_z | I have a block of soap that has a more rleiable GUI than that phone |
| 16:41.46 | *** join/#asterisk Cybersoft (i=cyversof@unaffiliated/cybersoft) |
| 16:41.49 | Cybersoft | is sitting in ur channel, logging ur conversations |
| 16:42.20 | monsted | err |
| 16:42.24 | monsted | good for you? |
| 16:42.33 | Cybersoft | err I meant to put a /me before this |
| 16:42.55 | monsted | still doesn't make much sense :) |
| 16:43.27 | Cybersoft | it does when the topic says "No public logging" |
| 16:44.46 | Supaplex | big woopie dog do do? |
| 16:44.48 | riddlebox | robin_z, what about the aastra one then I believe the 480i phone, I want to have buttons, that I can program for other extensions and stuff |
| 16:45.09 | robin_z | backblue: and misdn will work with my "cologne" chipset BRI card? |
| 16:46.48 | backblue | y |
| 16:47.01 | backblue | whatever works with bristuff works with beronet. |
| 16:48.32 | *** part/#asterisk Cybersoft (i=cyversof@unaffiliated/cybersoft) |
| 16:49.04 | *** join/#asterisk adorah (n=admin@84.94.112.67.cable.012.net.il) |
| 16:49.42 | robin_z | so I still need the zaptel zap_hfc driver? |
| 16:51.20 | backblue | robin_z: i just use zaptel for ztdummy, yes you should use zaptel only for ztdummy., |
| 16:51.36 | backblue | read the docs, you have on beronet.com |
| 16:51.53 | robin_z | so misdn will load a kernel module for the PCI card I have ... |
| 16:52.19 | backblue | y |
| 16:52.26 | robin_z | 'k |
| 16:52.32 | robin_z | installing .. :) |
| 16:53.14 | robin_z | just installing " a few" dependencies .. |
| 16:53.17 | robin_z | like gcc ... |
| 16:53.43 | backblue | yes, keep the coments for you, i'm tyred of installing that all the week. |
| 16:56.59 | *** join/#asterisk jaike (n=jaike@210.213.168.120) |
| 16:59.51 | robin_z | backblue: is the make-install script your work? |
| 17:00.15 | robin_z | backblue: I found a dependency the very comprehensive list didnt check and install |
| 17:00.43 | robin_z | udev |
| 17:01.12 | backblue | make-install? no |
| 17:01.31 | robin_z | overall though, its very impressive as an install script |
| 17:01.47 | backblue | robin_z: what are you using? |
| 17:02.19 | robin_z | the "install-asterisk" makefile from misdn.org |
| 17:02.20 | backblue | distro |
| 17:02.41 | robin_z | oh debian .. started of as the xorcomm cd, so Sarge with bits missing |
| 17:04.04 | backblue | robin_z: http://www.beronet.org/downloads/install-asterisk.tar.gz -> this will do all the work for you. |
| 17:04.18 | backblue | in debian |
| 17:04.19 | robin_z | thats what I have I think |
| 17:04.28 | backblue | i have made the same for centos |
| 17:05.07 | backblue | and i changed the install-misdn-mqueue so it does not install chan_misdn from beronet svn |
| 17:05.20 | backblue | and leave it for asterisk chan_misdn |
| 17:09.53 | jaike | has anyone compared scalability of 1.4beta3 with 1.2? |
| 17:11.03 | backblue | both disaster? |
| 17:18.43 | FuriousGeorge | is anyone on the asterisk video task force... |
| 17:19.02 | FuriousGeorge | i was looking at video conferencing hardware. its priced like PBXs. they sell it by the seat |
| 17:19.31 | FuriousGeorge | the difference between a 4 person conferencing setup and one for 12 people is like 8,000USD |
| 17:19.55 | FuriousGeorge | and thats from Polycomm. Basically you get a better camera and an arbitrary pricing jump |
| 17:22.42 | robin_z | FuriousGeorge: its called "profiteering" |
| 17:24.03 | FuriousGeorge | more like racketeering :) |
| 17:24.40 | FuriousGeorge | anyway, perhaps some oh323 video support in * 1.6 will take some of the sass out of them |
| 17:25.06 | robin_z | perhaps |
| 17:25.29 | robin_z | I bet my Polycom lives in a bigger and better house than all of us, so I suspect hes not totally stupid |
| 17:25.32 | robin_z | mr |
| 17:25.37 | *** part/#asterisk jaike (n=jaike@210.213.168.120) |
| 17:26.42 | FuriousGeorge | robin_z: i know what you mean, but i dont think we should make it easy on mr. polycomm |
| 17:27.24 | FuriousGeorge | the 2nd half of their solution is always basically a pc running collaboration software. there is so much of that stuff in existence today that its really just a matter of how you do with the video |
| 17:28.42 | *** join/#asterisk perlmonk (n=pierre@201.38.238.66) |
| 17:29.26 | *** part/#asterisk perlmonk (n=pierre@201.38.238.66) |
| 17:30.05 | FuriousGeorge | it bothers me how the voip and videoconferencing worlds are like unrelated in practice. you'd think they'd fall under the same umbrella |
| 17:36.26 | *** join/#asterisk prh (n=paul@X80.mjr.org) |
| 17:37.44 | Supaplex | in sip.conf I have working [XXXXX...] sectioins for clients to login to. How do I dial those direct? |
| 17:38.07 | Supaplex | s/sectioins/sections/ |
| 17:39.26 | SheriF_SpacE | Supaplex: u need to add exten => XXXX,1,Dial(SIP/XXXX) for each client in ur dialplan |
| 17:39.56 | robin_z | backblue: w00t! ... just got my fiorst incoming call ... crystal clear :) |
| 17:40.05 | Supaplex | I thought so. I'll review again. thanks |
| 17:40.29 | robin_z | backblue: so ... for outgoing where I used to have zap/g1 ... I should now have? |
| 17:41.52 | *** join/#asterisk adorah (n=admin@87.68.171.192.cable.012.net.il) |
| 17:42.00 | robin_z | mISDN/1-2? |
| 17:45.04 | backblue | zoa: dial(mISDN/1/${EXTEN}) |
| 17:45.06 | backblue | or |
| 17:45.19 | backblue | zoa: dial(mISDN/g:group/${EXTEN}) |
| 17:45.24 | backblue | ups it was for robin_z |
| 17:46.01 | robin_z | so ... port 1 covers both B channels? |
| 17:46.16 | backblue | yes |
| 17:46.28 | robin_z | so dial mISDN/1/${EXTEN} will use whichever channel is free? |
| 17:46.34 | backblue | port1 means 1 port. |
| 17:46.43 | backblue | robin_z: y |
| 17:46.49 | robin_z | kewl dude! |
| 17:47.40 | *** join/#asterisk jeebusroxors (n=jeebusro@cpe-75-80-228-131.dc.res.rr.com) |
| 17:48.58 | robin_z | hmmm ... nealry works .. |
| 17:49.14 | robin_z | Executing Dial("SIP/robin-081a89f8", "mISDN/1/01562862001}") in new stack |
| 17:49.14 | robin_z | <PROTECTED> |
| 17:49.41 | backblue | you have "}" wrong there |
| 17:49.51 | robin_z | but then I just get dialtone and finally "please hang up and try again" .. but atle ast its coming from the external network, which is more than I ever got out of bri :) |
| 17:50.13 | backblue | you can do misdn set debug 3 |
| 17:50.26 | backblue | in asterisk CLI, and get all the debug at isdn level |
| 17:51.00 | robin_z | argh .. it works :) |
| 17:51.05 | robin_z | heres a thing ... |
| 17:51.17 | robin_z | ive doen that trailign } before |
| 17:51.27 | backblue | you can dinamycally enable/disable echo cancelation and auto-gain |
| 17:51.39 | robin_z | except last time I had to fly to bloody Geneva to fix it! |
| 17:51.43 | backblue | robin_z: speak in english at least. |
| 17:52.11 | robin_z | apologies, I have mild dyslexia, it happens |
| 17:52.38 | backblue | apoligies accepted |
| 17:52.57 | robin_z | thnaks for the help .. I now have a working phoen system again! |
| 17:53.43 | robin_z | friday was HELL ... the telco replaced the pots line with a ISDN ... we plugged in the astersik box and ... nothing. no phones. |
| 17:54.08 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.216.2) |
| 17:54.17 | robin_z | worse, due to a fsck up, they were unable to divert the old number to our remaining working Pots line |
| 17:54.35 | robin_z | so ... getting this working is a very VERY good thing! |
| 17:56.01 | *** join/#asterisk VibroMax (n=phisto@83.229.70.154) |
| 18:01.45 | Supaplex | there it is |
| 18:05.52 | rob0 | They should have a crank on the side of the Web site, just like those old-fashioned pre-dial telephones used for calling the Operator. |
| 18:13.19 | Supaplex | I was thinking OLPC :P |
| 18:13.44 | Supaplex | it's one of the many reasons I'd like a sql dump of the wiki to keep a local mirror. |
| 18:21.55 | *** join/#asterisk fnordus (n=dnall@24.85.128.203) |
| 18:22.51 | [TK]D-Fender | Supaplex : "Real men don't backup. They upload their stuff to the WWW and let the world mirror them" - Linus Torvalds |
| 18:23.16 | backblue | hoo dear... |
| 18:24.24 | Supaplex | too bad I can't mirror them then huh? :) |
| 18:25.53 | *** join/#asterisk upod (n=chibondk@cpe-66-108-211-222.nyc.res.rr.com) |
| 18:28.42 | [TK]D-Fender | Supaplex : "Life sucks, but rarely swallows" - Me. |
| 18:33.04 | *** join/#asterisk Grind1 (n=Grind1@adsl-074-237-085-125.sip.mia.bellsouth.net) |
| 18:36.59 | *** join/#asterisk Xen^ (n=linux@unaffiliated/lnux/x-10290) |
| 18:43.49 | robin_z | mISDN works nicely for me :) |
| 18:45.05 | *** join/#asterisk justinu|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 18:57.02 | *** join/#asterisk Skarmeth (n=Skarmeth@201009101247.user.veloxzone.com.br) |
| 18:57.21 | *** join/#asterisk sahafeez (n=sahafeez@ip68-6-223-92.sd.sd.cox.net) |
| 18:58.12 | *** join/#asterisk jacekowski (n=jacek@aaof81.neoplus.adsl.tpnet.pl) |
| 18:59.48 | *** join/#asterisk olinux (i=olinux@ip68-107-4-202.sd.sd.cox.net) |
| 19:00.41 | sahafeez | i have a strange issue with a polycom 501. it is on a remote vpn. the vpn has 100% connection back to the main network. when the phone boots i can ping it from the remote end after it dhcp's so i know the connection is good. the dhcp server is local and the option tftp-server-name "voice-gateway" is set. (i have tried both IP and name. names can resolve locally). the phone keeps saying it cannot find the boot server. it is ignoring the d |
| 19:01.15 | sahafeez | tcpdump of the dhcp packets shows the tftp server is getting passed |
| 19:04.53 | *** join/#asterisk Chr1ztian (n=Christia@2001:5c0:8fff:ffff:0:0:0:cd) |
| 19:06.55 | robin_sz | did I mention how happy I was with * and mISDN? |
| 19:07.00 | backblue | sahafeez: insteed of voice-gateway put ip addr |
| 19:07.27 | sahafeez | yes, i tried that. |
| 19:07.32 | sahafeez | same issue |
| 19:07.50 | sahafeez | the phone seems to be ignoring the tftp option. |
| 19:08.11 | sahafeez | which is strange because i have 30 other phones in 3 location that work fine. |
| 19:09.20 | backblue | sahafeez: have you selected tftp option in the setup of the phone? |
| 19:10.03 | sahafeez | the phone has been erased. formated file system. it is the default at that point. |
| 19:10.05 | backblue | and your dhcpd have to be with option66 |
| 19:10.24 | backblue | i dont know what it's the default in 501, but for 601 it's ftp. |
| 19:10.39 | sahafeez | which is option tftp-server-name "voice-gateway" under dhcpd in linux. |
| 19:10.40 | backblue | you have to manually specify tftp |
| 19:10.48 | sahafeez | no you do not |
| 19:10.56 | sahafeez | i have never had to with 29 other phones |
| 19:10.57 | backblue | yes, if you sare using dhcpd from ISC |
| 19:11.11 | backblue | sahafeez: dont know what to say, in 601 you have too. |
| 19:11.16 | backblue | never used 501 |
| 19:11.31 | backblue | s/sare/are/g |
| 19:12.02 | [TK]D-Fender | sahafeez : pastebin your dhcpd.conf |
| 19:12.03 | sahafeez | once you format that file system there is nothing there. the phone will dhcp and try to get its files. the error on the phone is "cannot get file macadd.cfg from boot server" |
| 19:12.44 | backblue | sahafeez: after errased |
| 19:12.53 | backblue | it boots for tftp or ftp or http or https? |
| 19:12.54 | sahafeez | yes. |
| 19:13.01 | sahafeez | yes |
| 19:13.02 | *** join/#asterisk jacekowski (n=jacek@aaof81.neoplus.adsl.tpnet.pl) |
| 19:13.10 | backblue | yes what? it boot with which one? |
| 19:13.19 | backblue | what it's the default? |
| 19:13.22 | sahafeez | ftp is default. i have done this on 29 other phones. that is why i am stumped. |
| 19:14.00 | backblue | if the ftp it's the default, are you getting packets on port 21 of the tftp server? |
| 19:14.16 | backblue | have you seen, if he is trying to reach port 21? |
| 19:14.31 | *** join/#asterisk aao_pwner (n=aao_pwne@c-24-21-91-140.hsd1.mn.comcast.net) |
| 19:14.34 | [TK]D-Fender | sahafeez : did you set the boot server to 0.0.0.0 on the phone? |
| 19:14.48 | sahafeez | no. the phone was erased. |
| 19:14.49 | [TK]D-Fender | sahafeez : And double checked the rest of the BootROM options |
| 19:15.07 | backblue | yes sahafeez be shore of the boot server type you are using and he is with 0.0.0.0 |
| 19:15.18 | sahafeez | http://rafb.net/paste/results/BOB2UZ98.html |
| 19:15.53 | sahafeez | hum. i will have to go and get the phone. not anywhere near me. ;) |
| 19:16.21 | backblue | sahafeez: no one near it, you can give a call? |
| 19:16.52 | sahafeez | not right now. i will have to grab the phone today and have a look at it. i will bring it home and try from my vpn connection ;) |
| 19:17.34 | backblue | go head, your done, next... |
| 19:17.53 | sahafeez | like i said i am a bit perturbed and i have done this and have it working at 3 other locations. i have checked everything 100 times...and even dumped the packets. i think the phone has lost it. |
| 19:17.57 | sahafeez | thanks |
| 19:18.23 | backblue | sahafeez: it can be, happends to everyone. |
| 19:18.24 | [TK]D-Fender | sahafeez : You are spanning subnets with DHCP. You're not supposed to do that.... |
| 19:18.40 | sahafeez | the dhcp server is local. |
| 19:18.53 | [TK]D-Fender | sahafeez : It is? looks whacked |
| 19:19.17 | backblue | why [TK]D-Fender ? |
| 19:19.26 | sahafeez | it looks fine. the local net is 121. the stuff handed out does not have to be local except for the ip address of the client. |
| 19:19.46 | [TK]D-Fender | actually yeah... looking again closely, hrm... |
| 19:20.27 | [TK]D-Fender | sahafeez : And 121.1 routes to 119.X? |
| 19:20.35 | sahafeez | yes, no issues |
| 19:21.01 | sahafeez | i can ping the phone after it dhcpd from the 192.168.119.4 box |
| 19:21.16 | sahafeez | at this point i think it is the phone. |
| 19:22.20 | sahafeez | i love the polycom's but they can be a bit strange sometimes. |
| 19:22.45 | sahafeez | the phone was pulled working from the local 119 net before being moved. that why i erased it to make it forget it old cfg. |
| 19:29.19 | [TK]D-Fender | sahafeez : Is your boot server using the default user/pass? |
| 19:29.25 | sahafeez | yes |
| 19:29.31 | sahafeez | and it is booting 29 other phones ;) |
| 19:29.38 | sahafeez | and have been working for over a year |
| 19:31.28 | robin_sz | hmmm .. the directory where voicemail messages go is? |
| 19:31.57 | [TK]D-Fender | robin_sz : Typically "/var/spool/asterisk/voicemail/[context]/[box] |
| 19:33.03 | robin_sz | ta |
| 19:33.19 | robin_sz | weirdly .. I expected to find that set in voicemail.conf ... |
| 19:33.25 | robin_sz | shrug .. found it now |
| 19:36.56 | [TK]D-Fender | robin_sz : asterisk.conf.... |
| 19:37.13 | robin_sz | ahh |
| 19:38.14 | *** join/#asterisk zotz (n=zotz@208.196.247.175) |
| 19:38.41 | *** join/#asterisk ToTo (n=ToTo@host97-157-dynamic.2-87-r.retail.telecomitalia.it) |
| 19:39.46 | *** join/#asterisk derek (n=derek@ip68-3-24-234.ph.ph.cox.net) |
| 19:43.17 | sahafeez | going to get the phone...will let you know...thanks!@ |
| 19:45.10 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 19:45.32 | Supaplex | if balance_due_54 = $5.00 and custnum0 = 54, how else would I access ${balance_due_54} ? ${balance_due_${custnum0}} doesn't interpolate like I expected. |
| 19:49.06 | Supaplex | eval it is. |
| 19:50.06 | SwK | are you trying to do that with global vars or just channel vars? |
| 19:50.38 | *** join/#asterisk dasenjo (n=dasenjo@208.195.215.94) |
| 19:50.40 | Supaplex | channel |
| 19:50.56 | SwK | ${balance_due_${custnum0}} should resolve to ${balance_due_54} which would in turn resolve to 5.00 |
| 19:51.15 | SwK | if you are using a channel var why not just set bal_due |
| 19:51.29 | SwK | instead of including the custno |
| 19:51.33 | SwK | just simplify it down |
| 19:51.38 | Supaplex | some have multiple accounts. |
| 19:51.53 | *** join/#asterisk SGM (n=stoyan@home.marinov.us) |
| 19:51.58 | *** part/#asterisk Joel1978 (n=Joel1978@71.39.87.5) |
| 19:52.12 | Supaplex | I guess I'll worry about those 2%ers later. not a bad idea. |
| 19:52.40 | SwK | are you using AGI to do this? |
| 19:52.44 | SGM | hi guys |
| 19:52.50 | Supaplex | yes and no |
| 19:52.58 | Supaplex | agi to populate values |
| 19:53.03 | SGM | i found some strange 1.4 behaviour |
| 19:53.04 | SwK | do everything in the AGI |
| 19:53.16 | SwK | everything possibly except for calling Dial() |
| 19:53.23 | SGM | its not logging the disposition correctly |
| 19:53.24 | SGM | in the cdr |
| 19:54.06 | Supaplex | SwK: it seems to break0r when I call another agi from one agi. it's ignored/nothing happens. |
| 19:54.11 | SwK | dont start an agi then stop it then start another one wash rinse repeat... that is very costly... do everything possible in the AGI |
| 19:54.27 | SwK | do it in 1 agi |
| 19:54.57 | Supaplex | I see.. |
| 19:56.06 | *** join/#asterisk jacekowski (n=jacek@aaoo215.neoplus.adsl.tpnet.pl) |
| 20:02.04 | *** join/#asterisk kink0 (n=k@161.pool62-37-205.static.orange.es) |
| 20:02.11 | kink0 | good night |
| 20:02.47 | *** join/#asterisk jacekowski (n=jacek@aaoo215.neoplus.adsl.tpnet.pl) |
| 20:02.48 | *** join/#asterisk h0 (n=fakhir@unaffiliated/fakhir) |
| 20:02.57 | *** join/#asterisk BSDTech (n=RNeese@adsl-69-230-170-5.dsl.irvnca.pacbell.net) |
| 20:02.59 | kink0 | off-topic any idea how to store a class 0 SMS ? or any idea where to search ? |
| 20:14.52 | *** join/#asterisk [Airwolf] (n=airwolf@86.92.174.235) |
| 20:26.09 | *** join/#asterisk riddlebox (n=james@75-132-205-166.dhcp.stls.mo.charter.com) |
| 20:33.19 | robin_sz | hey riddlebox |
| 20:33.35 | riddlebox | hey had to take the dog for a walk |
| 20:33.50 | robin_sz | wow .. that brings another idea to mind |
| 20:34.01 | robin_sz | you know you where asking about GXP2000's? |
| 20:34.07 | robin_sz | well ... |
| 20:34.28 | robin_sz | split them in half, remove the circuitry and they make great poop scoops. |
| 20:35.17 | *** join/#asterisk tsurk0 (n=tsurko@145-226.go.evo.bg) |
| 20:36.51 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 20:40.02 | *** join/#asterisk axisys (n=axisys@c-69-143-190-152.hsd1.va.comcast.net) |
| 20:40.56 | *** join/#asterisk SkalTura (n=none@a88-113-70-135.elisa-laajakaista.fi) |
| 20:53.05 | riddlebox | robin_sz, lol |
| 20:53.12 | riddlebox | you really dont like those phones |
| 20:53.22 | robin_sz | those what? |
| 20:53.27 | riddlebox | lol |
| 20:53.32 | riddlebox | those things |
| 20:53.36 | robin_sz | oh, they are sipposed to be a phone ... |
| 20:53.46 | robin_sz | no. I don't |
| 20:54.13 | robin_sz | they shipped them early with non-working firmware |
| 20:54.25 | riddlebox | really |
| 20:54.28 | robin_sz | they shipped an irreversible "upgrade" |
| 20:54.59 | robin_sz | the "upgrade" doesnt work with all the phones .. certain phones are not compatibel |
| 20:55.08 | robin_sz | and they decline to give you your money back |
| 20:55.31 | Supaplex | that suxorz :-/ |
| 20:55.41 | robin_sz | I wish they had simply shipped me a wooden phone. it would have been better. |
| 20:55.52 | robin_sz | 1) I wouldnt have wasted time trying to get it to work |
| 20:55.58 | *** join/#asterisk dlynes_laptop (n=dlynes@S0106001346f7843f.vc.shawcable.net) |
| 20:56.00 | robin_sz | 2) I could have burnt it for heating. |
| 20:56.35 | *** join/#asterisk jsandnes (i=jsandnes@sip.meet24.com) |
| 20:56.57 | robin_sz | I understand some phones do work with the later firmware, mine doesnt and theres no way back to the earlier stuff |
| 20:58.09 | jsandnes | Hi, I am using meetme and background agi, and when someone pushes 0-9 the agi should playback a sound, the problem is that the agi plays it only to the person who initiates the playback, any idea on how I can make the agi play it to all the users in the conference? |
| 20:58.14 | SkalTura | robin_sz: if you've got the older firmware, you can ask some electronics shop to make you a new chip to it, they charge something like chip + 5euros |
| 20:58.53 | robin_sz | SkalTura, you have doen this? |
| 20:58.56 | *** join/#asterisk xnon (i=xnon@200.8.87.44) |
| 20:59.06 | SkalTura | robin_sz: my dad has |
| 20:59.15 | robin_sz | on a grandstream? |
| 20:59.24 | SkalTura | robin_sz: mobo |
| 20:59.55 | SkalTura | robin_sz: it's flash chip where the firmware is, and if it has a connector, and not soldered on the PCB it's really easy job |
| 21:00.04 | robin_sz | this I know ... |
| 21:00.11 | robin_sz | and it releis an you having a rom image |
| 21:00.15 | SkalTura | robin_sz: but before calling electronics (hobbyist) shop you gotta check what kind of flash chip |
| 21:00.26 | SkalTura | releis? |
| 21:00.30 | robin_sz | relies |
| 21:00.42 | robin_sz | and it relies on you having a rom image |
| 21:00.53 | SkalTura | if you can get the older firmware, as full, like they usually are, this ain't a problem |
| 21:01.02 | robin_sz | and most consumer electronics ... no, id say nearly all, do not use socketed flash |
| 21:01.17 | SkalTura | well, you can solder, yes? |
| 21:01.44 | Supaplex | surface mount? yuck |
| 21:02.47 | SkalTura | If it's surface mount, it ain't that easy, but still manageable |
| 21:02.47 | robin_sz | I can ... its 0.05 pitch smd though ... |
| 21:02.47 | robin_sz | well, thanks for trying .. |
| 21:02.47 | robin_sz | so ... all I would have to do is: |
| 21:02.47 | robin_sz | make a PCB |
| 21:02.47 | robin_sz | solder the new flash chip to it |
| 21:02.51 | robin_sz | get hold of the bootloader |
| 21:03.03 | robin_sz | put it in the right place (whereever that may be) on the image |
| 21:03.17 | robin_sz | desolder the chip |
| 21:03.29 | robin_sz | desolder the GXP200 flash chip |
| 21:03.35 | robin_sz | put the new chip in ... |
| 21:03.41 | SkalTura | nope you don't need to make a PCB |
| 21:03.52 | riddlebox | wow, that defeats everything I thought sip phones worked on |
| 21:04.02 | riddlebox | I thought you could get the new firmware and upgrade thenm |
| 21:04.12 | robin_sz | OK, so .. how many EXACTLY programming headers have you seen for smd flash chips? |
| 21:04.21 | SkalTura | open the device, find the flash chip, desolder it, take to a store and get it reflashed to older firmware(better perhaps replace with a new flash chip tho), resolder, assemble, use |
| 21:05.07 | robin_sz | riddlebox, for normal phones yes you can ... grandstream released upgrades that made it impossible to go back again |
| 21:05.18 | robin_sz | I repeat .., so .. how many EXACTLY programming headers have you seen for smd flash chips? |
| 21:05.22 | SkalTura | robin_sz: dude, i'm just trying to help you, enuff with the BS |
| 21:05.34 | robin_sz | thnaks, but trust me, its not an option |
| 21:06.01 | SkalTura | programming headers? Dude, they are usually in format of RAW / Image |
| 21:06.07 | robin_sz | sigh ... |
| 21:06.07 | riddlebox | I have heard from others that the Grand Stream phones are not the best |
| 21:06.41 | SkalTura | robin_sz: i think you are thinking too complicated now |
| 21:06.48 | robin_sz | SkalTura, when you mount a flash chip in a chip programmer, you need to put the chip in a header... it adapts the standard pinout of the programmer to the chip in question ... |
| 21:06.55 | SkalTura | gotta remember to: Keep It Simple Stupid |
| 21:07.45 | Supaplex | does it support jtag or serial console etc? |
| 21:08.02 | robin_sz | I say this, having owned at least 5 different device programmers over the years in my employment as an electronics engineer with a major national broadcaster ... |
| 21:08.04 | SkalTura | robin_sz: i'm sure they have that kind of adapters, lol. If i got your meaning right --> adapter to make it fit (PIN order) ... yeap, english ain't my native language |
| 21:08.18 | robin_sz | I have never seen an adapter for SMD device ... |
| 21:08.44 | kink0 | SkalTura, try www.digikey.com for adapters |
| 21:08.44 | Supaplex | same concept as a socket T processor |
| 21:08.56 | kink0 | but is rare smd adapters |
| 21:09.20 | SkalTura | i'm not in the search of something like that, just trying to help robin_sz |
| 21:09.34 | kink0 | see u later !! time for lunch, if anyone knows how to read ussing AT commands a flash (class 0) incoming SMS, let me know :) |
| 21:09.41 | *** join/#asterisk Assid (i=assid@59.183.15.26) |
| 21:09.47 | robin_sz | trust me, its not an option ... a plug in flash chip on a mobo is way different from an SMD device |
| 21:10.17 | robin_sz | if it aint soldered down, it aint getting programmed. period |
| 21:10.52 | robin_sz | the only real hope would be to access the serial pins on the phone that they must use to give it "first birthday" during manufacture |
| 21:11.10 | Assid | wassssup |
| 21:11.14 | SkalTura | well, then you could ask vendor to send in a new chip, but i doubt they will |
| 21:11.22 | robin_sz | but, even then .. youd still have a crap phone .. as the original firmware was also crap |
| 21:11.46 | SkalTura | well, without field, what you gonna do with shit? |
| 21:12.14 | robin_sz | well, the phoens make decent plant pots ... |
| 21:12.20 | robin_sz | or poop scoops. |
| 21:12.43 | SkalTura | poop scoops... i don't even want to know what that is |
| 21:12.52 | robin_sz | when you take the dog for a walk |
| 21:12.59 | robin_sz | and it craps on the sidewalk ... |
| 21:13.07 | robin_sz | you scoop it up and bin it ... |
| 21:13.13 | SkalTura | from another point of view: if you can reliable receive and make calls, you've got a good phone >;D |
| 21:13.25 | robin_sz | well, yes. I agree |
| 21:13.29 | robin_sz | but, it cant. |
| 21:13.34 | robin_sz | crashes randomly |
| 21:13.42 | robin_sz | display goes blank |
| 21:13.55 | SkalTura | oh in that case, it should have been trashed without eevn trying firmware upgrade, that's by default garbage |
| 21:14.03 | robin_sz | poor audio |
| 21:14.16 | SkalTura | display doesn't matter imo, but audio quality does |
| 21:14.18 | SkalTura | so garbage |
| 21:14.21 | robin_sz | how about: the on-hook switch doesnt get pressed by the receiver |
| 21:14.33 | robin_sz | you put the phoen down .. it doesn hang up |
| 21:14.38 | SkalTura | hell, i would probably try to get my money back on the basis of item not delivering what promised |
| 21:14.46 | robin_sz | tried. |
| 21:14.48 | robin_sz | failed. |
| 21:14.50 | SkalTura | what it exactly were ... grand something? |
| 21:15.08 | robin_sz | grandstream GXP2000 .. like a phone, but crap. |
| 21:15.26 | SkalTura | ok, reminder for self: never get any grandstream crap |
| 21:15.35 | robin_sz | indeed |
| 21:15.37 | SkalTura | robin_sz: try again, try atleast 3 times ;) |
| 21:15.42 | *** join/#asterisk Juggie (n=Juggie@wiley-459-29776.roadrunner.nf.net) |
| 21:15.47 | robin_sz | they were cheapest on market |
| 21:15.51 | robin_sz | but now there are others |
| 21:16.06 | SkalTura | price doesn't matter, product / service has to deliver what promised |
| 21:16.15 | SkalTura | atleast, in business that's been my philosophy |
| 21:16.17 | robin_sz | the Elmeg 290 is almost same price |
| 21:16.24 | robin_sz | re-badged Snom190 |
| 21:16.36 | robin_sz | and .. theres some open source code for it .. |
| 21:16.37 | SkalTura | really, i don't jack about SIP phones |
| 21:16.57 | robin_sz | it runs Linux :) |
| 21:17.11 | SkalTura | only reason i'm here is because i almost got lucky and almost received a huge PBX deal |
| 21:17.20 | robin_sz | almost? |
| 21:17.29 | SkalTura | long story |
| 21:17.37 | robin_sz | still possible? |
| 21:17.58 | SkalTura | shamefully not, i'm not going to ever call that jackass again |
| 21:18.06 | robin_sz | heh |
| 21:18.27 | riddlebox | SkalTura, http://www.voipsupply.com/product_info.php?products_id=331 |
| 21:18.37 | SkalTura | He made the impression that the job is mine already, i had been redying a 3 PBX demo system, and he was supposed to deliver some gear for the demo day |
| 21:18.48 | robin_sz | and? |
| 21:19.24 | SkalTura | never heard of again of him, and i were managing his servers too. I worked hard for the bizarre systems he insisted on having, just to hear he scrapped them and got just about stock & basic systems for 10-20x the price |
| 21:20.14 | robin_sz | http://www.voip-info.org/wiki/view/GXP-2000 |
| 21:20.41 | robin_sz | heh |
| 21:20.46 | robin_sz | people suck |
| 21:21.13 | SkalTura | indeed |
| 21:21.26 | monsted | yep |
| 21:21.37 | monsted | i've got 13000 colleagues and 12800 of them suck |
| 21:21.42 | SkalTura | and i made all that for him under the regular price, because i wanted to learn from him (I'm a small time do'er, his a big time do'er) |
| 21:22.04 | robin_sz | well, big timers get there by steppign on small timers |
| 21:22.31 | SkalTura | most funniest thing was, he's an IT security expert... He asked a secure system, i built a secure system, he got angry cos it was too hard to use by his opinion |
| 21:22.52 | monsted | security is fine, as long as people don't have to do anything for it |
| 21:22.59 | robin_sz | security != difficult |
| 21:23.34 | SkalTura | robin_sz: true, but he also lost a lot of possible revenue by this |
| 21:24.17 | SkalTura | robin_sz: yup, he got annoyed by the fact that he couldn't login as root directly |
| 21:24.31 | robin_sz | im waiting for them to introduce fingerprint and foreskin scanners ... fingerprints, well yeah ... but easy to copy off someone .. you need SECURE ... foreskin logins! its the only sane way!! |
| 21:24.53 | SkalTura | lol! x) |
| 21:25.00 | robin_sz | root logins are nto good |
| 21:25.38 | SkalTura | yeap, i gave him "admin" shell access which to use to login to shell, and then first root pwd to login as root, and later i put in a sudo config |
| 21:25.38 | robin_sz | ssh, require public keys, no password logins, no root login |
| 21:26.15 | SkalTura | i make always nowadays no root login, v2 only, specific users only, specific IPs only |
| 21:26.26 | robin_sz | and NO TELNET |
| 21:26.31 | SkalTura | lol! :D |
| 21:26.45 | robin_sz | anyway ... |
| 21:28.32 | SkalTura | robin_sz: yes, i know couple ports are open on the system YOU see ;) |
| 21:29.28 | jsandnes | anyone? |
| 21:29.31 | jsandnes | Hi, I am using meetme and background agi, and when someone pushes 0-9 the agi should playback a sound, the problem is that the agi plays it only to the person who initiates the playback, any idea on how I can make the agi play it to all the users in the conference? |
| 21:30.25 | monsted | "Aaaargh, your firewall is broken! I can't log in on my nortel box!", "Oh? Works fine from here.", "Nooo, i need rlogin!" |
| 21:30.39 | monsted | uhm, yeah, about that... go play in traffic? |
| 21:33.56 | Juggie | jsandnes, yes. |
| 21:34.32 | jsandnes | Juggie: do you mind helping me out? := |
| 21:34.54 | Juggie | jsandnes, instead of the background agi doing a playback, you are going to have to have it create a fake call on the local channel using AMI: Action Originate, then in your context for the local channel, make it Answer, play what you want to play, and then hangup. |
| 21:35.05 | riddlebox | hrmm maybe I should put some time into redesigning my little projects webpage lol http://vallista.idyll.org/~james/ |
| 21:35.25 | Juggie | essentially, you will have to cause a local channel to start, join the meetme conference, play the file, and hangup. |
| 21:35.31 | jsandnes | Juggie: i think that would take too much time |
| 21:35.48 | Juggie | jsandnes, its the only way to broadcast to the entire conference |
| 21:35.58 | jsandnes | Hm. |
| 21:36.05 | jsandnes | Is there any way to do it in a 2way call? |
| 21:36.18 | Juggie | not that i know of, its not that hard. |
| 21:36.24 | SkalTura | riddlebox: shameless advertising, but do check out my hosting services @ http://www.artichost.net :) |
| 21:36.39 | riddlebox | ok |
| 21:36.40 | jsandnes | hm |
| 21:36.46 | jsandnes | Juggie: okay thanks alot :-) |
| 21:37.01 | Juggie | jsandnes, msg me if you get really stuck, i have the code for it somewhere. |
| 21:37.17 | Juggie | but, it involves joining a local channel into the meetme conference. |
| 21:37.35 | Juggie | and making it do what you want. |
| 21:37.35 | SkalTura | i think i'll head to bed now |
| 21:37.37 | SkalTura | l8er |
| 21:37.39 | Juggie | look up local channels on the wiki for more info. |
| 21:37.47 | *** join/#asterisk zotz (n=zotz@208.196.247.175) |
| 21:39.20 | BSDTech | anyone get asterisknow working on bsd ? |
| 21:39.33 | BSDTech | or just linux |
| 21:39.33 | Qwell | ...what? |
| 21:39.40 | Qwell | asterisknow is a Linux distro |
| 21:39.52 | BSDTech | ok I mean the asteriskgui |
| 21:40.07 | BSDTech | that digiuim released |
| 21:40.19 | Qwell | BSDTech: it should work anywhere |
| 21:40.45 | BSDTech | hmm |
| 21:41.11 | BSDTech | even thou on linux apache is in /var/www and on bsd its in .usr/local/www |
| 21:41.14 | BSDTech | ok |
| 21:41.19 | BSDTech | I will test it |
| 21:41.27 | BSDTech | I just was looking at it |
| 21:41.29 | Qwell | it doesn't use apache |
| 21:41.39 | BSDTech | what soe si t use |
| 21:41.46 | Qwell | asterisk |
| 21:41.55 | BSDTech | yes but its a php interface |
| 21:42.01 | Qwell | Says who? |
| 21:42.16 | BSDTech | I have looked at the src for it |
| 21:42.23 | BSDTech | its also html pages |
| 21:42.24 | Qwell | It's...not php |
| 21:43.35 | BSDTech | ok what webserver does it use to display the config pages |
| 21:43.43 | Qwell | asterisk |
| 21:43.46 | BSDTech | did they embedded one |
| 21:43.49 | BSDTech | ok wow |
| 21:44.37 | BSDTech | and no its does not work /install on bsd |
| 21:44.44 | BSDTech | it cant find the zaptel |
| 21:44.52 | BSDTech | I just tried installing |
| 21:46.10 | *** join/#asterisk osiris (n=osiris@c-71-205-27-131.hsd1.mi.comcast.net) |
| 21:48.54 | robin_sz | hmmm .... mISDN and callerid ... |
| 21:49.03 | robin_sz | doesnt seem to be happening for me |
| 21:50.49 | riddlebox | hrmm I should be able to do something like this right exten => *23,1,SayDigits(${CALLERIDNUM})? |
| 21:52.35 | *** part/#asterisk [Airwolf] (n=airwolf@86.92.174.235) |
| 21:54.04 | Corydon76-home | riddlebox: assuming the channel is already answered, yes |
| 21:54.14 | BSDTech | thisis crap it seems the gui is hardcoded for linux asterisk installs |
| 21:55.59 | riddlebox | Corydon76-home, what I want to do is have an Anac code, so I can call in dial a code and be told what number I am dialing from |
| 21:56.20 | Corydon76-home | riddlebox: are you using Zap or SIP channels? |
| 21:56.27 | riddlebox | sip |
| 21:56.36 | Corydon76-home | riddlebox: then you need to Answer first |
| 21:56.42 | riddlebox | ahh ok |
| 21:57.05 | Corydon76-home | riddlebox: and may need a Wait(1) after the answer to give time for the media to be established |
| 21:57.29 | riddlebox | ok I will try it thanks |
| 22:02.29 | riddlebox | Corydon76-home, if I am on an inside extension, will that work as well? |
| 22:02.40 | Corydon76-home | Yes |
| 22:02.44 | riddlebox | hrmm its not |
| 22:02.52 | Corydon76-home | but your callerid may be different |
| 22:03.05 | Corydon76-home | It might not even be set |
| 22:03.06 | riddlebox | it should read like 522 |
| 22:03.40 | Corydon76-home | Did you set the callerid in sip.conf? |
| 22:04.15 | riddlebox | yeah |
| 22:04.25 | *** join/#asterisk BSDTech (n=RNeese@adsl-69-230-170-5.dsl.irvnca.pacbell.net) |
| 22:04.26 | Corydon76-home | Remember, channel name is NOT the same thing as callerid |
| 22:05.07 | robin_sz | hmmm .... for $someReason the "press 1 for sales, press 2 for accounts" menu thing that works fine from SIP phones, does not seem to work from the incoming ISDN |
| 22:05.39 | Corydon76-home | robin_sz: do you have other extensions that start with 1 or 2 in the same context? |
| 22:05.47 | robin_sz | nope |
| 22:06.01 | Corydon76-home | Are you even starting in the same context? |
| 22:06.11 | Corydon76-home | Is the line answered? |
| 22:06.23 | riddlebox | you talking to me? |
| 22:06.30 | robin_sz | yeah, it answers, plays the "press 1 for blah" and then .. nothing happens |
| 22:07.06 | Corydon76-home | robin_sz: On ISDN, you can hear prompts even without answering the line. Are you sure the line is answered? |
| 22:07.26 | robin_sz | well, I can hear the prompts ... |
| 22:07.42 | Corydon76-home | Did you explicitly run Answer ? |
| 22:07.48 | robin_sz | yep |
| 22:07.59 | robin_sz | and I have "disposition; answered" in CDR |
| 22:08.21 | Corydon76-home | Did you run the prompt with Background ? |
| 22:09.12 | robin_sz | yep |
| 22:09.37 | Corydon76-home | Is autofallthrough set to 'yes' or 'no' in extensions.conf? |
| 22:09.46 | robin_sz | hmmm |
| 22:09.48 | robin_sz | not knowing |
| 22:09.49 | *** join/#asterisk KeNroM (n=sdfgas@72.22.149.238) |
| 22:09.55 | riddlebox | weird I can do exten => SayDigits(522) and it works, but in that line it wont |
| 22:10.02 | KeNroM | Hi name is kenley |
| 22:10.12 | Corydon76-home | riddlebox: then callerid is not set |
| 22:10.16 | *** join/#asterisk perlmonk (n=pierre@201.38.238.66) |
| 22:10.25 | KeNroM | i`m looking for a vicdail\asterisk consultant |
| 22:10.44 | KeNroM | any in here? |
| 22:10.55 | robin_sz | Corydon76-home, what shoudl it be set to? |
| 22:11.04 | riddlebox | Corydon76-home, it is set in my sip.conf for extension 522 |
| 22:11.06 | Corydon76-home | KeNroM: try the asterisk-biz list |
| 22:11.28 | Corydon76-home | riddlebox: callerid=Foo <522> |
| 22:11.29 | KeNroM | thats a irc room on this server? |
| 22:11.46 | Corydon76-home | KeNroM: no, that's a mailing list on lists.digium.com |
| 22:12.08 | Corydon76-home | robin_sz: there isn't a right answer |
| 22:12.13 | KeNroM | ty |
| 22:12.38 | robin_sz | Corydon76-home, wait I PasteBin the bits ... |
| 22:12.47 | robin_sz | ~pb |
| 22:12.53 | jbot | i heard pb is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 22:12.53 | Corydon76-home | riddlebox: the angle brackets are not optional |
| 22:13.51 | riddlebox | Corydon76-home, I have callerid=Cordless<522> |
| 22:14.11 | Corydon76-home | riddlebox: space? |
| 22:14.25 | [TK]D-Fender | riddlebox : Should look like - callerid = "Cordless" <522> |
| 22:14.38 | Corydon76-home | [TK]D-Fender: actually, quotes are bad |
| 22:14.53 | [TK]D-Fender | riddlebox : Please pastebin the exact entry and the dialplan bits you're talking about (perhaps more than bits would be helpful) |
| 22:15.14 | [TK]D-Fender | Corydon-w : Not in my experience.... where have you encountered problems with them? |
| 22:15.28 | Corydon76-home | [TK]D-Fender: the quotes show up as part of the callerid |
| 22:15.37 | *** part/#asterisk perlmonk (n=pierre@201.38.238.66) |
| 22:15.37 | KeNroM | i have a asterisk dailer....and when i have agents dailing it is tellling me dailer timed out please contact administrator |
| 22:15.43 | KeNroM | how do i fix this |
| 22:15.49 | [TK]D-Fender | Corydon-w : Not anywhere that I've ever seen.... |
| 22:16.07 | Corydon76-home | [TK]D-Fender: they're unnecessary, in any case |
| 22:17.24 | [TK]D-Fender | Corydon76-home : If you say so. I personally find it looks cleaner and feels not natural in a "programming" sense to indicate a string. |
| 22:17.44 | riddlebox | Corydon76-home, http://pastebin.ca/257894 |
| 22:18.22 | Corydon76-home | riddlebox: did you run a 'sip reload' since the last time you edited that? |
| 22:19.16 | riddlebox | Corydon76-home, yes |
| 22:19.20 | [TK]D-Fender | riddlebox : please pastebin the defective dialplan bits... |
| 22:20.21 | robin_sz | http://channels.debian.net/paste/4594 |
| 22:20.33 | robin_sz | Corydon76-home, there you go .. the public menu |
| 22:20.37 | riddlebox | [TK]D-Fender, http://pastebin.ca/257898 |
| 22:20.39 | robin_sz | embryonic |
| 22:21.21 | robin_sz | it doesnt even seem to be decoding the pressed DTMF |
| 22:21.22 | *** join/#asterisk bkw_ (n=brian@24.249.5.154) |
| 22:21.40 | [TK]D-Fender | riddlebox : You should be using the newer CID functions..... exten => *23,3,SayDigits(${CALLERID(num)}) |
| 22:21.41 | robin_sz | bkw_, NEXT! |
| 22:21.44 | Corydon76-home | robin_sz: is this a cell phone, perchance? |
| 22:21.58 | *** join/#asterisk Ashura (n=Ashura@adsl-ull-205-143.49-151.net24.it) |
| 22:22.01 | Ashura | hello |
| 22:22.19 | riddlebox | ohh it changed in 1.4 then,I guess I need to read up on what changed |
| 22:22.24 | robin_sz | Corydon76-home, nah, plain old telephone .. coming in over isdn |
| 22:22.36 | Corydon76-home | robin_sz: which card? |
| 22:22.37 | robin_sz | mISDN on the * box |
| 22:22.40 | [TK]D-Fender | riddlebox : I believe the stuff that got deprecated in 1.2 got removed entirely in 1.4 |
| 22:22.51 | robin_sz | with some random "cologne" chipset card |
| 22:22.59 | Corydon76-home | robin_sz: not familiar with that hardware, sorry |
| 22:23.13 | *** part/#asterisk sergee (n=opera@195.94.224.197) |
| 22:23.44 | riddlebox | [TK]D-Fender, that did it, it is working now |
| 22:24.02 | robin_sz | theDTMF tones are decoded by the hardware normally then? |
| 22:24.22 | [TK]D-Fender | riddlebox : Quite welcome |
| 22:24.24 | Corydon76-home | robin_sz: AFAICT, you're doing everything correct on the Asterisk side, so you'll probably need to talk with the chan_misdn folks |
| 22:25.05 | Corydon76-home | robin_sz: I'm not familiar enough with the hardware to say one way or the other |
| 22:25.30 | robin_sz | right |
| 22:25.43 | Corydon76-home | robin_sz: it might even be something as simple as a config option in the .conf file |
| 22:25.49 | robin_sz | could be ... |
| 22:26.53 | [TK]D-Fender | ok, GTG, BBL |
| 22:28.49 | *** join/#asterisk nick125 (n=ndevito@atarack/staff/nick125) |
| 22:29.00 | nick125 | afternoon everyone |
| 22:29.12 | *** join/#asterisk XSMerlin (n=me@host-87-74-5-98.bulldogdsl.com) |
| 22:29.59 | riddlebox | now if I could get asterisk connected to broadvoice |
| 22:30.53 | XSMerlin | lo, anyone here had any luck with installing the digium b410p 4port bri card? |
| 22:32.26 | *** join/#asterisk knarfly (n=bwoodwar@c-65-34-177-3.hsd1.fl.comcast.net) |
| 22:33.33 | *** join/#asterisk sahafeez (n=sahafeez@ip68-6-223-92.sd.sd.cox.net) |
| 22:35.20 | robin_sz | Corydon76-home, they are a passive ISDN card, I would expect the detection of dtmf to be done somehwre in * I guess |
| 22:35.43 | riddlebox | when I said that I meant, keep asterisk connected to broadvoice, it always looses connection |
| 22:35.58 | robin_sz | turning up the misdn debug doesn show anything happening as I press the '1' key etc |
| 22:42.04 | Supaplex | if I'm recording a voice prompt, should I record at 8khz first, or rescale after it's converted to gsm? |
| 22:45.07 | *** join/#asterisk SirThomas_Home (n=tomc@209-169-198-159.us.warpdriveonline.com) |
| 22:46.38 | *** join/#asterisk hfb (n=hfb@cpe-66-27-96-242.san.res.rr.com) |
| 22:47.08 | SirThomas_Home | are the lights on the digium tdm cards supposed to light up? |
| 22:47.16 | Qwell | SirThomas_Home: Did you call Digium? |
| 22:47.24 | Qwell | we give free installation support... |
| 22:47.38 | SirThomas_Home | figured digium would be closed on sat |
| 22:47.46 | Qwell | this is probably true |
| 22:47.51 | SirThomas_Home | cool.. .didn't think of that. |
| 22:47.55 | Qwell | (you'd think I'd know..) |
| 22:48.10 | XSMerlin | i tried to call a min ago, seems theres noone there |
| 22:48.13 | SirThomas_Home | the lights on the back by the plugs, do they light up when powered? in use? |
| 22:48.18 | XSMerlin | which is a real shame |
| 22:48.24 | XSMerlin | im desperate for some support |
| 22:48.26 | Qwell | SirThomas_Home: when the modules are loaded, iirc |
| 22:48.38 | riddlebox | Qwell, I didnt know you work for digium |
| 22:48.43 | SirThomas_Home | Qwell: ok. thanks. That's what I figured. |
| 22:49.01 | Qwell | I know almost nothing about configuring our cards, heh |
| 22:49.15 | SirThomas_Home | Qwell: I suspect I got a dead card. :-( |
| 22:49.23 | XSMerlin | do u know when theres supposed to be someone back on teh suport desk? |
| 22:49.31 | Qwell | XSMerlin: Monday at the latest |
| 22:49.36 | XSMerlin | kk |
| 22:49.48 | XSMerlin | im a dead man by then, lol |
| 22:50.02 | XSMerlin | SirThomas_Home: which card? |
| 22:50.22 | SirThomas_Home | tdm21b. two fxs, one fxo |
| 22:50.50 | XSMerlin | k |
| 22:50.55 | Qwell | What happens? |
| 22:51.10 | XSMerlin | how far thru install are you? |
| 22:51.33 | XSMerlin | heh, are u running trixbox? |
| 22:51.42 | robin_sz | has any smoke come out of it yet? |
| 22:51.48 | XSMerlin | lol |
| 22:53.45 | Supaplex | keep the magic smoke so the factory can put it back in. |
| 22:54.27 | *** join/#asterisk hwt (n=hwt@curb.thorkildssen.com) |
| 22:54.38 | hwt | hi, how do i catch those Unable to create channel of type 'SIP' (cause 3 - No route to destination), and playback a message instead? |
| 22:54.45 | hwt | i use a stdexten macro for inbound calls. |
| 22:55.29 | SirThomas_Home | sorry -- checking digium docs. |
| 22:55.45 | SirThomas_Home | I (re-)installed trixbox and it basically doesn't detect my card |
| 22:55.58 | SirThomas_Home | no lights light up on the card itself. |
| 22:56.02 | *** join/#asterisk shido6 (i=shido6@d221-68-200.commercial.cgocable.net) |
| 22:56.23 | XSMerlin | have u deited zaptel and zapata .conf's ? |
| 22:56.28 | SirThomas_Home | no (detectable) smoke has escaped |
| 22:56.31 | XSMerlin | *edited |
| 22:56.53 | Qwell | SirThomas_Home: We use invisible smoke |
| 22:56.59 | SirThomas_Home | lol |
| 22:57.00 | Qwell | (patent pending) |
| 22:57.14 | XSMerlin | my smoke has a bluish tint ;) |
| 22:57.25 | Qwell | XSMerlin: You must be Euro |
| 22:57.29 | XSMerlin | :) |
| 22:57.34 | hwt | anyone? |
| 22:57.38 | XSMerlin | i am indeed |
| 22:57.45 | SirThomas_Home | XSMerlin: I edited zaptel. Told it I had one FXO, two FXS and it complained about them not existing |
| 22:58.10 | XSMerlin | are u sure uv got it edited for the right ports? |
| 22:58.14 | *** join/#asterisk henrique (n=henrique@201-1-135-75.dsl.telesp.net.br) |
| 22:58.53 | SirThomas_Home | XSMerlin: yup, I tried that. I lied to it. It still complained about the device not existing instead of the "you screwed up" message. |
| 22:58.56 | XSMerlin | and what do u see when u run zttool -vvv |
| 22:59.19 | SirThomas_Home | zttool gives me "ZTDUMMY" as my only card. |
| 22:59.30 | XSMerlin | ok |
| 22:59.32 | XSMerlin | lspci |
| 22:59.40 | XSMerlin | run that, can u see the car dinthat list? |
| 23:00.03 | SirThomas_Home | Is this it? --> 02:08.0 Communication controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface |
| 23:00.46 | *** join/#asterisk darkskiez (n=mbryars@bb-87-81-62-203.ukonline.co.uk) |
| 23:00.58 | XSMerlin | possibly, im not sure its supposed to show it as a isdn car tho, let me log itno one of my other boxes and check... |
| 23:01.38 | SirThomas_Home | 02:09 is my network card, which is right next to it... and they are my only two PCI cards in the machine. |
| 23:03.25 | XSMerlin | yep yep |
| 23:03.25 | XSMerlin | thats right |
| 23:03.29 | XSMerlin | ok |
| 23:03.36 | XSMerlin | do lsmod |
| 23:03.46 | XSMerlin | paste teh results into pastebin |
| 23:04.48 | SirThomas_Home | ok... |
| 23:05.10 | XSMerlin | gonna have to be quick tho, i really gotta work on my own problems, lol |
| 23:05.59 | SirThomas_Home | sorry... pastebin being slow. :-( |
| 23:06.05 | XSMerlin | np |
| 23:06.25 | SirThomas_Home | http://pastebin.ca/257928 |
| 23:07.46 | XSMerlin | ok, all ur modules are loading fine, so im guessing its ur zaptel.conf |
| 23:08.38 | SirThomas_Home | http://pastebin.ca/257930 <-- /etc/zaptel.conf |
| 23:09.10 | XSMerlin | hard for me to tell if tis right, cos u need to match it up to the ports that the modules are plugged into the card in |
| 23:09.22 | SirThomas_Home | ZT_CHANCONFIG failed on channel 3: No such device or address (6) |
| 23:09.39 | SirThomas_Home | that's what I get from "ztcfg -vv" |
| 23:10.11 | SirThomas_Home | XSMerlin: Thanks for your help & time. I will most likely be calling them on Monday for help. |
| 23:10.41 | XSMerlin | dont worry bout it, i like agood mystery, im no expert tho |
| 23:10.46 | XSMerlin | anyway |
| 23:10.52 | XSMerlin | few thoughts |
| 23:11.06 | XSMerlin | u need to have the molex plugged itnot he card for the fxs ports |
| 23:11.30 | XSMerlin | you need to define channels=> in zaptel |
| 23:11.37 | XSMerlin | and |
| 23:12.01 | XSMerlin | u need to look phsically on the card and see which ports which module is plugged itno and set each one up wit the right signalling |
| 23:12.12 | SirThomas_Home | ok |
| 23:12.14 | XSMerlin | remember for the fxo port u gotta provide fxs signalling |
| 23:12.19 | SirThomas_Home | yup |
| 23:12.23 | XSMerlin | and vice versa |
| 23:12.43 | XSMerlin | but im fairly certian that that zaptel.conf isnt right |
| 23:13.09 | SirThomas_Home | that sounds promising. |
| 23:13.51 | XSMerlin | <PROTECTED> |
| 23:13.53 | robin_sz | you could do worse than trying genzaptelconf ... |
| 23:13.59 | robin_sz | the auto config generator? |
| 23:13.59 | XSMerlin | no u couldnt |
| 23:14.17 | SirThomas_Home | robin_sz: yup, tried that a couple of times. gives me nothing. :-( |
| 23:14.21 | XSMerlin | that app has only ever gone wrong for me, but im sure thats not the case for everyone |
| 23:14.22 | robin_sz | oh, ok, its always worked for me .. |
| 23:14.26 | robin_sz | fair enuff |
| 23:15.10 | robin_sz | you usually need to touch zapat.conf if there isnt one ... just to tickle it |
| 23:15.14 | SirThomas_Home | yup, just re-tried it and I now have an empty zaptel.conf. :-) |
| 23:15.22 | XSMerlin | lol |
| 23:15.37 | robin_sz | I shall assumeyou copied it first |
| 23:15.57 | XSMerlin | get the card showing under zttool first, then mess with zapata |
| 23:16.01 | SirThomas_Home | another empty zaptel.conf |
| 23:16.03 | robin_sz | true |
| 23:16.14 | robin_sz | zttool is your friend |
| 23:16.21 | SirThomas_Home | zttool not showing it made me thing it was a dead card. |
| 23:16.28 | SirThomas_Home | but it does show under lspci... |
| 23:16.37 | robin_sz | even if your card is unconfiured zttool should see it |
| 23:16.38 | XSMerlin | nah, pretty sure its just not configged wright |
| 23:16.49 | XSMerlin | thats true |
| 23:17.13 | XSMerlin | have u checked theres no irq conflict? |
| 23:17.16 | robin_sz | as long as the appropriate module is modprobed, zttool will see it |
| 23:17.41 | SirThomas_Home | and that module is "wctdm" -- correct? |
| 23:17.50 | SirThomas_Home | irq conflict... um.. .no. |
| 23:17.55 | SirThomas_Home | haven't checked for that. |
| 23:18.06 | robin_sz | cat /proc/interrupts |
| 23:18.07 | XSMerlin | do lspci again and pastebin |
| 23:18.23 | robin_sz | theres an iraq conflict .. is that similar? |
| 23:18.30 | XSMerlin | lol |
| 23:19.10 | robin_sz | apparently there is more violence in iraq now than there was during Saddams time |
| 23:19.15 | SirThomas_Home | http://pastebin.ca/257937 <-- /proc/interrupts and lspci |
| 23:19.24 | robin_sz | so its not been a waste of effrot then ;) |
| 23:19.30 | SirThomas_Home | lol |
| 23:19.54 | *** join/#asterisk Winkie (n=urmom@cpc2-stre3-0-0-cust344.bagu.cable.ntl.com) |
| 23:20.24 | XSMerlin | i think that looks ok |
| 23:20.31 | XSMerlin | lol |
| 23:20.59 | XSMerlin | anyone else? |
| 23:23.33 | *** part/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 23:24.16 | *** join/#asterisk Winkie (n=urmom@cpc2-stre3-0-0-cust344.bagu.cable.ntl.com) |
| 23:26.03 | SirThomas_Home | I'm going to try it without the ztdummy module loaded. |
| 23:28.20 | SirThomas_Home | now zttool shows nothing. :-( |
| 23:28.26 | XSMerlin | lol |
| 23:28.34 | XSMerlin | u were expecting somethign different? |
| 23:28.56 | SirThomas_Home | I was hoping without the dummy module it would...um...look harder. or something. |
| 23:29.07 | XSMerlin | no such luck |
| 23:30.01 | XSMerlin | have u checked the positioning of the modules on the card? |
| 23:30.31 | SirThomas_Home | should I try it with them shuffled? |
| 23:31.14 | XSMerlin | if u look on the main board there will be a number by each module, i need to match that number up with the zaptel.conf |
| 23:31.15 | SirThomas_Home | my card isn't numbered 1.2.3.4. It was numbered strange, like 1.2.4.3 |
| 23:31.37 | EmleyMoor | Anyone prepared to give me a call on FWD 794933? |
| 23:31.42 | SirThomas_Home | but based on the numbering I had a fxs in 1 and 4 and a fxo in 2. |
| 23:32.25 | Ashura | I'd like to read more about asterisk and its capabilities... could you please address me to a valid documentation ? |
| 23:33.04 | EmleyMoor | www.asterisk.org |
| 23:33.07 | *** join/#asterisk Winkie (n=urmom@cpc2-stre3-0-0-cust344.bagu.cable.ntl.com) |
| 23:33.08 | SirThomas_Home | Channel 02: FXS Kewlstart (Default) (Slaves: 02) --> gives me --> ZT_CHANCONFIG failed on channel 2: No such device or address (6) |
| 23:33.23 | qdk | Does Asterisk CDR silently ignore if CDR USERFIELD is greater than the 255 limit? |
| 23:34.09 | EmleyMoor | Anyone used a Trimphone with *? |
| 23:34.12 | SirThomas_Home | If I lie, and change it to "fxoks=2" I still get --> ZT_CHANCONFIG failed on channel 2: No such device or address (6) |
| 23:34.23 | XSMerlin | hang on |
| 23:34.42 | Ashura | I mean... some "Asterisk for dummies" or "foolproof" docs :) |
| 23:34.52 | Ashura | i need a getting started guide |
| 23:35.19 | XSMerlin | asterisk: the future of technology |
| 23:35.25 | XSMerlin | blinding book |
| 23:35.28 | EmleyMoor | http://www.oreilly.com/catalog/asterisk/ |
| 23:35.32 | XSMerlin | *telephony |
| 23:35.35 | EmleyMoor | It is available for online download |
| 23:35.36 | XSMerlin | heh |
| 23:35.43 | Ashura | Tnx |
| 23:35.46 | XSMerlin | oh oh oh |
| 23:36.04 | *** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox) |
| 23:36.10 | XSMerlin | SirThomas_Home: theres a great bit in that book about zaptel .conf, go read it |
| 23:36.19 | qdk | Ashura: http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 23:36.31 | SirThomas_Home | yup, I read a bit of it... is there good troubleshooting info? |
| 23:37.05 | XSMerlin | read the section on zaptel.conf |
| 23:37.35 | XSMerlin | i reeeely gotta crack on with this isdn card |
| 23:37.46 | XSMerlin | its 11.40pm here |
| 23:37.55 | XSMerlin | and the missis ig startign to get pissed offf |
| 23:37.57 | SirThomas_Home | XSMerlin: again, thanks for your help. |
| 23:37.59 | EmleyMoor | XSMerlin: UK? |
| 23:38.12 | XSMerlin | SirThomas_Home: sry cldnt b more help |
| 23:38.18 | XSMerlin | EmleyMoor: yep |
| 23:38.43 | XSMerlin | cool |
| 23:39.46 | XSMerlin | ok |
| 23:40.08 | EmleyMoor | Had to recompile wctdm to get the dial to work - but it works now |
| 23:40.46 | XSMerlin | why oh why wold u want a rotary dial phone nowadays? |
| 23:40.49 | XSMerlin | lol |
| 23:40.51 | XSMerlin | nutter |
| 23:40.59 | EmleyMoor | XSMerlin: I like its bell |
| 23:41.22 | XSMerlin | lol |
| 23:42.07 | EmleyMoor | Besides, it has an amplifier, which means my partner could have a conversation on it (though he can't dial on it) |
| 23:42.21 | *** join/#asterisk nassy (n=nassy@207-38-197-201.c3-0.wsd-ubr1.qens-wsd.ny.cable.rcn.com) |
| 23:42.32 | XSMerlin | ur on a computer |
| 23:42.40 | EmleyMoor | Yes |
| 23:42.45 | EmleyMoor | I use softphones too |
| 23:42.55 | XSMerlin | im pretty sure u could replicate teh bells n amplification on a soft [hone |
| 23:43.04 | XSMerlin | heh, beat me to it :P |
| 23:43.36 | EmleyMoor | I could, but my laptop resides in my drawer at work between working days, only being brought home at weekends |
| 23:44.00 | KeNroM | any vicidal experts in here? |
| 23:44.15 | EmleyMoor | Besides, David likes a phone to be a phone (as long as it's push-button dialing, LD or MF) |
| 23:44.27 | *** join/#asterisk ariel_ (n=Ariel@dsl-20-177.cofs.net) |
| 23:44.56 | XSMerlin | fair enuff |
| 23:45.28 | Qwell | LD? |
| 23:45.49 | EmleyMoor | Loop Disconnect |
| 23:45.52 | *** join/#asterisk robin_sz (n=robin@rapid2.gotadsl.co.uk) |
| 23:45.54 | Qwell | oh |
| 23:46.00 | EmleyMoor | (equivalent to rotary) |
| 23:46.41 | *** join/#asterisk jm|home (n=jamiem@zen.jamiem.com) |
| 23:47.28 | XSMerlin | lol, its prety cool purely in its kitshness |
| 23:49.03 | EmleyMoor | If I could get MF dial and an amplifier on same kind of phone easily, I'd sell both of my existing Yeomen |
| 23:49.40 | EmleyMoor | (there is a technical issue with having both, that needs battery power for the amplifier) |
| 23:49.59 | EmleyMoor | I have a rotary table model and a wall MF. |