| 00:00.10 | mercestes | Port 23 only..gives me a bad http response. *sighs* Telnet works but....unknown pword |
| 00:00.11 | Nugget | telnet is eeeeeeevil! |
| 00:00.13 | mercestes | oh well, job for tomororww |
| 00:00.21 | mercestes | Goodnight |
| 00:04.41 | GarethTheGreat | what's "unknown IE"? |
| 00:07.57 | Corydon-w | IE=Information Element |
| 00:08.22 | Corydon-w | Probably you have two different versions of Asterisk |
| 00:09.10 | Corydon-w | Nugget: would you happen to know the etymology of 'moo' as it applies to geek culture? |
| 00:10.00 | GarethTheGreat | ah |
| 00:10.03 | GarethTheGreat | 1.2 and 1.0 |
| 00:10.30 | GarethTheGreat | anyway to make the 1.2 box backwards compatible or am i screwed? |
| 00:10.54 | Corydon-w | 1.2 IS backwards compatible |
| 00:11.03 | Corydon-w | The problem is that 1.0 is not forwards compatible |
| 00:11.20 | Qwell[] | Corydon-w: The jargon file has nothing |
| 00:11.24 | Corydon-w | Or you could simply ignore the warning |
| 00:11.53 | Corydon-w | Qwell[]: I know... I'm wondering if the etymology is lost |
| 00:12.11 | Corydon-w | Qwell[]: it would be an excellent thing to document, if someone knows |
| 00:12.13 | *** join/#asterisk ucfMethod (n=ucfmetho@c2.efb7d1.client.atlantech.net) |
| 00:12.19 | ucfMethod | hello |
| 00:13.53 | GarethTheGreat | Corydon-w: that doesn't make any sense, if 1.2 can't talk to 1.0 it ain't backwards compatible |
| 00:14.02 | GarethTheGreat | how do i make them talk to each other or can i not? |
| 00:15.16 | Corydon-w | Which IE was it? |
| 00:15.36 | GarethTheGreat | <PROTECTED> |
| 00:15.37 | *** join/#asterisk adamowitz (n=adamowit@ip68-14-27-224.ri.ri.cox.net) |
| 00:16.57 | Corydon-w | 42 is cause code |
| 00:17.10 | Corydon-w | You can safely ignore the warning |
| 00:17.20 | Qwell[] | Corydon-w: http://groups.google.com/group/net.games/browse_thread/thread/776a908023d0ff5f/a69c06fbae3b7220?lnk=st&q=moo&rnum=8&hl=en#a69c06fbae3b7220 |
| 00:17.38 | Qwell[] | actually.. http://groups.google.com/group/net.works/browse_thread/thread/b02404699592f70c/d805ca2763ccb2ec?lnk=st&q=moo&rnum=7&hl=en#d805ca2763ccb2ec |
| 00:17.46 | Qwell[] | That was the first uttering of "moo" on usenet :P |
| 00:18.07 | Qwell[] | "Software began it's life on microcomputers with a solemn "Moo."" |
| 00:18.11 | GarethTheGreat | that still doesn't get me closer to linking my 2 boxes together |
| 00:18.16 | Supaplex | is there an easy way to debug eagi scripts from a command line? |
| 00:18.37 | *** join/#asterisk macli (n=vincent@mcli.brc.ubc.ca) |
| 00:19.25 | Corydon-w | GarethTheGreat: in 1.0, IAX2 used textual cause codes. They weren't portable to other channel types, which is why they were changed to use numeric cause codes in 1.2 |
| 00:19.59 | GarethTheGreat | ok |
| 00:20.19 | GarethTheGreat | but how do i configure my boxes to talk IAX to each other? |
| 00:20.53 | Supaplex | is this an iax1 vs iax2 issue? |
| 00:21.46 | *** part/#asterisk EmleyMoor (i=ejabberd@hallam.tinsleyviaduct.com) |
| 00:22.34 | GarethTheGreat | they both seem to be claiming to be IAX2 |
| 00:23.09 | Supaplex | no natting? |
| 00:23.12 | ucfMethod | can anyone clear up callerid |
| 00:24.54 | ucfMethod | I am setting it like this SetCallerID("Company Name <XXXXXXXXXX>",a) yet it still shows up on landlines as "Toll Free 888 XXX XXXX" |
| 00:25.03 | GarethTheGreat | Supaplex: one box is NATed |
| 00:25.09 | GarethTheGreat | with all ports forwarded |
| 00:25.23 | GarethTheGreat | i tried with and without nat=yes in iax.conf |
| 00:25.35 | Qwell[] | ucfMethod: Most of the time, you can't change CIDName over the PSTN |
| 00:25.48 | ucfMethod | Qwell[]: why not |
| 00:25.57 | Qwell[] | because it doesn't work that way |
| 00:26.14 | Qwell[] | it's retrieved via a database lookup on the remote end |
| 00:26.43 | ucfMethod | Qwell[]: so if I enter our name into said database, it will work? where can I register our 888 number? |
| 00:26.57 | Qwell[] | With every telco across the country |
| 00:27.01 | ucfMethod | terrific |
| 00:27.08 | ucfMethod | looks like ill pass on that one |
| 00:27.15 | JT | ucfMethod: digital or analogue lines? |
| 00:27.33 | ucfMethod | IAX voip termination |
| 00:27.41 | JT | oh |
| 00:27.41 | JT | ok |
| 00:28.10 | JT | they'd have digital lines, but i guess they might not pass on CID down the PRI |
| 00:28.26 | ucfMethod | well i can set the number just fine, just the name wont appear |
| 00:28.27 | JT | something tells me that method of setting callerid is deprecated, too |
| 00:28.42 | ucfMethod | im calling my mom in florida (im in va) and she doesnt see company name, just 888 number |
| 00:28.48 | JT | ah ok, well you're probably out of luck then |
| 00:29.01 | JT | on the asterisk side anyway |
| 00:29.14 | JT | you could always try asking your provider if they'll set CID name |
| 00:29.18 | Supaplex | how do have asterisk use the sound card to start a call in some context? |
| 00:29.33 | ucfMethod | im using the latest version (stable) 1.2.13 correct? SetCallerID() |
| 00:30.41 | Corydon-w | GarethTheGreat: that unknown IE error is not the problem... if Asterisk is sending a cause code, it's because it has decided to hangup the call |
| 00:31.10 | *** join/#asterisk tsurk0 (n=tsurko@vpn-pppoe-213-240-242-237.megalan.bg) |
| 00:31.16 | *** join/#asterisk Marshall16 (n=Marshall@cpe-24-95-78-61.columbus.res.rr.com) |
| 00:31.25 | Corydon-w | So the call was already over by the time you received the unknown IE |
| 00:31.36 | JT | ucfMethod: SetCallerID("Name" <Number>[|a]) |
| 00:31.37 | JT | Deprecated in 1.2, removed in 1.4.- See Setting Callerid |
| 00:31.55 | JT | ucfMethod: http://www.voip-info.org/wiki/view/Setting+Callerid |
| 00:34.57 | ucfMethod | sweet. thanks |
| 00:38.24 | ManxPower | Never use quotes in callerid |
| 00:39.32 | ManxPower | Some IP phones will reject calls with invalid characters in Caller*ID (and " is not a valid Caller*ID character) |
| 00:40.10 | GarethTheGreat | Corydon-w: what reasons would it have for hanging up the call? |
| 00:40.55 | ManxPower | GarethTheGreat: the reason is whatever the value of HANGUPCAUSE |
| 00:40.57 | Corydon-w | GarethTheGreat: I cannot answer that without examining logs |
| 00:41.03 | ucfMethod | ManxPower: Thanks, I just noticed that if you have a <SPACE> between the Name and the < ex. Set(CALLERID(all)=Company Name <888XXXXXXX>) it will show up UNKNOWN |
| 00:41.10 | Corydon-w | ManxPower: he doesn't have that, as he's using 1.0 |
| 00:41.23 | ManxPower | Corydon-w: He's screwed then. |
| 00:42.22 | GarethTheGreat | Nov 17 00:26:45 NOTICE[17906]: chan_iax2.c:5062 register_verify: No registration for peer 'mainsip' (from 81.174.255.77) |
| 00:42.29 | GarethTheGreat | that appears to have something to with it |
| 00:42.35 | GarethTheGreat | but can't figure out what |
| 00:42.55 | Corydon-w | You set it up as host=dynamic, but it's not registering? |
| 00:43.38 | Corydon-w | Perhaps you set up the registration as SIP, and not as IAX2? |
| 00:45.56 | GarethTheGreat | let me paste my config on pastebin |
| 00:45.58 | *** join/#asterisk eltech (i=G00Ds@ool-457c9421.dyn.optonline.net) |
| 00:46.51 | Corydon-w | I'm going home |
| 00:48.05 | GarethTheGreat | http://pastebin.ca/250493 |
| 00:48.13 | GarethTheGreat | can anyone else take a look at that? |
| 00:48.22 | GarethTheGreat | i just tried adding host=dynamic - no difference |
| 00:51.15 | Corydon-w | GarethTheGreat: you have type=user on both sides. At least one side needs to be peer (or friend) |
| 00:51.56 | GarethTheGreat | ah |
| 00:52.11 | sivana | Is there an alternative to ChanIsAvail to actually find out if the peer is busy or not? |
| 00:52.38 | GarethTheGreat | woohoo |
| 00:52.40 | GarethTheGreat | it works |
| 01:03.05 | tessier | There is no call on the phone but asterisk thinks there is |
| 01:03.20 | tessier | In our call center this prevents any new calls being routed to the phone and the operator sits idle. |
| 01:10.36 | knarfly | I'm having trouble getting the syntax to block calls with no callerid..can anyone help? |
| 01:11.04 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 01:12.45 | *** join/#asterisk Kyler (n=chatzill@74.132.200.229) |
| 01:16.48 | tr2x | hello knarfly |
| 01:17.22 | tr2x | knarfly: if ( ${CALLERID(number)} ) {} else {...} |
| 01:17.24 | *** join/#asterisk haryv (n=sales@S010600a0c93f6f7e.vs.shawcable.net) |
| 01:20.52 | *** join/#asterisk tsurk0 (n=tsurko@vpn-pppoe-213-240-242-237.megalan.bg) |
| 01:22.13 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 01:26.39 | GarethTheGreat | how do i send DTMF from the console's dial command? |
| 01:29.07 | *** join/#asterisk mixi (n=mixi@the.one.and.only.iammixi.de) |
| 01:32.49 | *** join/#asterisk andy2006 (n=andy2006@k3070-pc03.ece.ubc.ca) |
| 01:34.43 | Supaplex | GarethTheGreat: my memory's pretty vague, but I think you just type it in and press enter. 10% confident on that. :-P |
| 01:36.13 | *** join/#asterisk kerlnel20 (n=kerlnel2@210.213.113.154) |
| 01:36.33 | kerlnel20 | hi there does grandstream 100 has audio out? |
| 01:36.45 | kerlnel20 | budgetone |
| 01:37.44 | *** join/#asterisk andy2006 (n=andy2006@k3070-pc03.ece.ubc.ca) |
| 01:39.05 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 01:40.13 | kerlnel20 | ? |
| 01:40.16 | kerlnel20 | hi there does grandstream 100 has audio out? |
| 01:40.51 | Supaplex | no idea |
| 01:42.14 | kerlnel20 | how about this |
| 01:42.44 | kerlnel20 | is there a way to broadcast to all phone, provided that the phones were set auto-answer? |
| 01:44.09 | ucfMethod | what module needs to be loaded for MeetMe to work. I am getting this error "Nov 16 20:41:59 WARNING[7755]: pbx.c:1700 pbx_extension_helper: No application 'MeetMe' for extension" |
| 01:44.53 | kerlnel20 | compile it in the kernel |
| 01:45.02 | kerlnel20 | you need to tweak the kernel first |
| 01:45.05 | kerlnel20 | see meetme |
| 01:45.10 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
| 01:45.12 | ucfMethod | I have everything in meetme.conf, extensions.conf correct, but I still get that error (PS I am migrating from 1.0 to 1.2.13) |
| 01:48.03 | ucfMethod | kerlnel20: I need to tweak the kernel for meetme to work? was that something I was suppose to do when compiling asterisk? |
| 01:49.11 | kerlnel20 | see meetme |
| 01:49.15 | kerlnel20 | it is stated there |
| 01:49.59 | ucfMethod | man, I didn't install a zaptel interface |
| 01:54.42 | *** join/#asterisk jeebusroxors (n=jeebusro@cpe-75-80-228-131.dc.res.rr.com) |
| 01:54.42 | *** join/#asterisk jeebusmobile (n=jeebusmo@cpe-75-80-228-131.dc.res.rr.com) |
| 01:56.02 | Supaplex | isn't the console suppose to play out the sound card? (eg dial to a context from the cli> prompt) |
| 01:56.10 | knarfly | tr2x: I'm still not sure how to put that in an extension? Can you be more specific? |
| 01:58.52 | ucfMethod | kerlnel20: can i just re compile asterisk after loading the zaptel stuff and the meetme should work? |
| 01:58.52 | kerlnel20 | is there a way to broadcast to all phone, provided that the phones were set auto-answer? |
| 01:59.32 | JT | knarfly: GoToIf |
| 02:00.08 | *** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.itb.ac.id) |
| 02:01.12 | knarfly | JT: I tried that but it does not work...the * server stopped answering calls....guess I have to study this one some more...? |
| 02:01.51 | JT | well that's the standard method for doing a conditional if statement |
| 02:02.49 | knarfly | JT: I'm also getting a memory type zaptel leak messge when I unload the zaptel driver...is that normal? |
| 02:03.37 | *** join/#asterisk eltech (i=G00Ds@ool-457c9421.dyn.optonline.net) |
| 02:03.40 | kerlnel20 | is there a way to broadcast to all phone, provided that the phones were set auto-answer? |
| 02:03.48 | JT | knarfly: probably not, but neither is unloading the zaptel driver |
| 02:08.28 | *** join/#asterisk stkn_ (i=nobody@gentoo/developer/pdpc.active.stkn) |
| 02:11.37 | kerlnel20 | is there a way to broadcast to all phone, provided that the phones were set auto-answer? |
| 02:12.05 | Qwell | kerlnel20: STOP repeating the same question over and over. |
| 02:12.10 | JT | how many types do you need to repeat the same bloody question? |
| 02:12.13 | Qwell | You've been told countless times. |
| 02:12.23 | JT | s/types/times/ |
| 02:12.34 | [TK]D-Fender | Qwell : Ok. You're right. Some people you just can't teach. Kill him. |
| 02:12.43 | Qwell | [TK]D-Fender: voila :p |
| 02:12.51 | *** join/#asterisk dasenjo (n=dasenjo@208.195.215.168) |
| 02:14.26 | ucfMethod | kerlnel20: holy cow, that was easy, compile zaptel, load modules, re compile asterisk, done MeetMe now works like a charm....... man asterisk rules, its so easy |
| 02:15.50 | Supaplex | I agree, it's wonderful isn't it? :) |
| 02:16.15 | ucfMethod | im mad i didnt upgrade from 1.0 to 1.2 sooner |
| 02:17.49 | [TK]D-Fender | ucfMethod : Or mad you didn't skip 1.2 altogether and go to 1.4? ;) |
| 02:18.18 | ucfMethod | beta? |
| 02:18.40 | ucfMethod | I am setting this up for my company, I don't want buggy stuff (or is 1.4 not too bad) |
| 02:18.59 | [TK]D-Fender | ucfMethod : For corp use I'd say stick with 1.2 for now. |
| 02:19.09 | ucfMethod | sweet |
| 02:19.47 | *** join/#asterisk Cinen (n=Cinen@208.70.20.33) |
| 02:20.12 | ucfMethod | now that i realize upgrading is as easy as: stop asterisk, compile new asterisk, make clean, make install , done ....... I am so thrilled |
| 02:20.30 | *** join/#asterisk yogurt2ungue (n=yogurt2u@host86.190-30-91.telecom.net.ar) |
| 02:20.34 | ucfMethod | im gonna save so much cash this way, ill have dual 24 inch monitor in no time |
| 02:21.04 | Cinen | anyone seen MOG lately? |
| 02:21.06 | [TK]D-Fender | ucfMethod : 22" is under 1/2 the price..... |
| 02:21.17 | Supaplex | ucfMethod: and no restarting eh? ;) |
| 02:21.18 | ucfMethod | but i like 1920x1200 ;-) |
| 02:21.51 | [TK]D-Fender | ucfMethod : Well you're welcome to it I guess... I find the proportions at that size somewhat straining personally, and I like my cash :) |
| 02:22.07 | Supaplex | holy megapixels batman! ;) |
| 02:22.43 | kerlnel20 | is there a way to broadcast to all phone, provided that the phones were set auto-answer? |
| 02:22.50 | ucfMethod | ~600 bones for Dell's we have them in the office, I just want a dual setup. I am just overwhelmed at how simple Asterisk is |
| 02:23.33 | [TK]D-Fender | kerlnel20 : "show application page" |
| 02:25.34 | [TK]D-Fender | ucfMethod :+/- 600 for 24" 1920x1200? |
| 02:25.53 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
| 02:26.00 | kerlnel20 | [TK]D-Fender? |
| 02:26.03 | ucfMethod | [TK]D-Fender: last I checked, they were 670 I think |
| 02:26.15 | ucfMethod | which isnt terrible, and they are bright and crystal clear |
| 02:26.19 | ucfMethod | totally worth it |
| 02:26.23 | *** join/#asterisk zmef420 (n=zmef420@metarb3-pool3-10.mtco.com) |
| 02:26.55 | [TK]D-Fender | ucfMethod : Website says 900 currently.... |
| 02:27.04 | ucfMethod | now that i have used this 24, i will never go back to tiny resolutions (as long as i can help it) |
| 02:27.12 | ucfMethod | no way |
| 02:27.42 | ucfMethod | 679.00, what website you looking at.... |
| 02:27.44 | [TK]D-Fender | ucfMethod : nvm.... they auto-detected my country and shifted me even though i tried .con direct :) |
| 02:27.53 | ucfMethod | where are you? |
| 02:28.05 | [TK]D-Fender | ucfMethod : Canada. so yeah, makes sense now :) |
| 02:28.11 | ucfMethod | that sucks |
| 02:28.18 | ucfMethod | sneaky bastards charging CA more |
| 02:28.22 | [TK]D-Fender | ucfMethod : Really? We don't have GWB :D |
| 02:29.05 | ucfMethod | [TK]D-Fender: you do have health care for all.... but you also have wierd accents ;-0 <kidding of course> |
| 02:29.09 | file | [TK]D-Fender: I don't want to meet your parents |
| 02:29.17 | [TK]D-Fender | file : I just want..... |
| 02:29.21 | file | [TK]D-Fender: ! ! ! |
| 02:29.26 | [TK]D-Fender | file : z0mg! |
| 02:29.33 | file | [TK]D-Fender: made it home at a sane time? |
| 02:30.33 | [TK]D-Fender | file : Yeah I was out only 10 minutes late. No biggie. Finished that @^%#$ing budget spreadsheet from hell today. I lost a little over 2 hours to that key-field #@$@-up my sales manager did. |
| 02:30.37 | *** join/#asterisk ltd (n=z@202-161-0-77.dyn.iinet.net.au) |
| 02:30.40 | shmaltz | anybody here using the aastra 480i phone? |
| 02:30.50 | ucfMethod | im heading home... thanks again for all your help |
| 02:30.52 | [TK]D-Fender | shmaltz : Plenty of us. Ask your real question now :) |
| 02:31.10 | shmaltz | [TK]D-Fender, thanks you are right. |
| 02:31.53 | shmaltz | anybody here using the aastra 480i with any complains against it? my area of interest: stability, and easy of use for end lUSERS |
| 02:32.27 | *** join/#asterisk bluregard (n=matt@c-67-163-72-68.hsd1.il.comcast.net) |
| 02:34.05 | *** part/#asterisk mtoups (i=toups@HOWARD.INDYPGH.ORG) |
| 02:34.16 | [TK]D-Fender | shmaltz : Well it doesn't do PLC as well as Polycoms, so its 2nd place on audio quality. It IS a very flexible phone with some seriously nifty programability witht he soft-keys. Can do in-call DTMF speed-dials, and the soft-keys are state-based meaning you can have optiosn based on where you are at a given time. |
| 02:34.44 | [TK]D-Fender | shmaltz : Inferior call handling capabilities vs Polycom as well, but for general use they're just fine. |
| 02:34.55 | bluregard | can anyone tell me what a good IP phone is. I just need a basic non-cordless office type phone. |
| 02:34.55 | shmaltz | ~PLC |
| 02:34.59 | jbot | plc is probably Packet Loss Concealment, a method of recreating lost packets by interpolating or merely retransmitting old ones. An interpolating PLC was introduced in the jitterbuffer of Asterisk 1.4. |
| 02:35.02 | GarethTheGreat | anyone here willing to test out my new installation for me (i.e make a quick call)? |
| 02:35.07 | [TK]D-Fender | shmaltz : Config files are a breeze, which is a bonus, and I *think* you can do FTP with them as well. |
| 02:35.23 | bluregard | GarethTheGreat: I'll do it |
| 02:35.25 | [TK]D-Fender | bluregard : Polycom, Aastra, Linksys : in that order. |
| 02:36.05 | shmaltz | [TK]D-Fender, we all know that Polycom is very good sound, and most business users don't even need such good sound (avaya comes to mind) |
| 02:36.38 | shmaltz | which means that if sound quality is not an issue it is better than the Polycom |
| 02:36.42 | Supaplex | for gotoif targets, it can be a context right? eg ... ?context1:context2) |
| 02:37.00 | bluregard | [TK]D-Fender: how about Avaya? |
| 02:37.01 | shmaltz | and if I don't use it over the Internet then I don't need the PLC that much |
| 02:37.04 | [TK]D-Fender | shmaltz : I'm sure you'd find it hard to give up. Polycom has the best sound and usability out there (Cisco has been considered a bit better by a few, but Polycom beats everyone on the UI) |
| 02:37.17 | [TK]D-Fender | bluregard : Avoid phones from proprietary vendors. |
| 02:37.32 | bluregard | I've used a few polycom phones and yeah, I agree, they sound great. |
| 02:37.43 | shmaltz | Cisco sucks, it's unstable, and overprices, plus ZERO features |
| 02:37.52 | [TK]D-Fender | Supaplex : You always need to specify down to the LSD (Context, Exten, Priority) |
| 02:38.12 | [TK]D-Fender | shmaltz : I've never heard sucks, but the rest has merit. |
| 02:38.29 | [TK]D-Fender | shmaltz : Polycom costs more than the "junk" out there, but deserves it. |
| 02:38.47 | [TK]D-Fender | shmaltz : But the 480i is a REALLY interesting phone for many applications. |
| 02:39.02 | [TK]D-Fender | shmaltz : I'd highly suggest you grab one to learn more about. |
| 02:40.09 | shmaltz | the reason I say Cisco sucks is because if it freezes more than once a week then it sucks, |
| 02:40.21 | [TK]D-Fender | shmaltz : State-based + in-line speed-dials makes *'s features like call recording, DTMF transfers (for queue calls needing the disconnect aspect that DTMF offers), and so on can really pay. |
| 02:40.42 | [TK]D-Fender | shmaltz : That is very firmware dependant. those on older stuff is pretty stable. |
| 02:41.17 | JT | is the polycom 301 anygood? |
| 02:41.19 | shmaltz | [TK]D-Fender, that dtmf stuff makes it realy good for legacy integration |
| 02:41.32 | [TK]D-Fender | JT : Sure is. Only thing it lacks is a full-duplex speakerphone. |
| 02:41.37 | shmaltz | JT, yes and no, SPA942 might be better |
| 02:41.41 | JT | ah ok |
| 02:41.43 | [TK]D-Fender | shmaltz : indeed. |
| 02:41.47 | JT | how much would you pay for one? :) |
| 02:41.52 | [TK]D-Fender | shmaltz : Scratch that! Bad paste! |
| 02:42.05 | [TK]D-Fender | JT : Forget Linksys if you are in North America |
| 02:42.17 | shmaltz | [TK]D-Fender, the aastra 480i supprts NAT? |
| 02:42.18 | JT | for the 301... |
| 02:42.19 | Un1x | y? |
| 02:42.24 | [TK]D-Fender | JT : Do you have/plan on getting PoE? What kind of call volume? |
| 02:42.40 | [TK]D-Fender | shmaltz : Supposedly, but I've never tried it for that. |
| 02:43.03 | [TK]D-Fender | JT : $115 for an IP 301 w/ Power brinck (non-PoE) |
| 02:43.37 | JT | [TK]D-Fender: just for my own use really atm |
| 02:43.44 | JT | PoE, i can get injectors if need be |
| 02:43.54 | *** join/#asterisk tengulre (n=tengulre@221.11.5.182) |
| 02:44.17 | [TK]D-Fender | JT : For personal home use? |
| 02:44.28 | JT | yes |
| 02:44.41 | [TK]D-Fender | JT : would this be your first real IP phone? |
| 02:44.45 | JT | i like to play with new hardware |
| 02:44.47 | JT | yes |
| 02:44.54 | JT | i have a T1 channel bank atm |
| 02:45.01 | JT | and i saw a 301 cheap on ebay... |
| 02:45.06 | [TK]D-Fender | JT : Well I'd suggest you splurge a bit more ($170) and get an IP 501 instead. |
| 02:45.23 | JT | ah ok, any reason? |
| 02:45.48 | *** join/#asterisk klictel (n=klictel@modemcable159.7-200-24.mc.videotron.ca) |
| 02:46.17 | [TK]D-Fender | JT : Full speakerphone, much bigger & better screen, believe me it makes a differnce), easier menu navigation, 3 line appearances. |
| 02:48.19 | Carp1 | Nufone's support isn't vary good at all. |
| 02:48.40 | Supaplex | poo-phone |
| 02:49.07 | [TK]D-Fender | JT : If you're going to spend some money on a "nice" phone I'd want it to bee as full -featured as you could want (buyer's remorse is a terrible thing). |
| 02:51.08 | *** join/#asterisk Jetti (n=kickedmy@S0106001310f843e2.wp.shawcable.net) |
| 02:51.39 | Jetti | On a 1 to 10(hardest) what would you say the difficult of Asterisk would be? |
| 02:52.57 | *** join/#asterisk metafore (n=john@adsl-68-127-185-145.dsl.pltn13.pacbell.net) |
| 02:53.02 | shmaltz | Jetti, it depends more on the person trying to overcome asterisk |
| 02:54.52 | [TK]D-Fender | omg, serious cool 0-G juggling : http://my.break.com/media/view.aspx?ContentID=183829 |
| 02:55.19 | [TK]D-Fender | Jetti : Depends what you consider a 7 ;) |
| 02:56.39 | bluregard | [TK]D-Fender: who makes the 501 you were talking about? |
| 02:57.22 | *** join/#asterisk alerios (n=alerios@190.24.100.110) |
| 02:57.44 | [TK]D-Fender | bluregard : Also Polycom. |
| 02:57.48 | JT | [TK]D-Fender: ah, but it's different if i'm spending money on a "bargain" phone |
| 02:58.03 | JT | keep in mind generally polycoms are rare and expensive in Australia |
| 02:58.30 | [TK]D-Fender | JT : http://www.telephonydepot.com/Polycom_s/25.htm |
| 02:58.44 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
| 02:58.45 | [TK]D-Fender | JT : Ah... AU. Yes, they would be pricy. |
| 02:58.49 | *** join/#asterisk remmo (n=chatzill@smack.isp.net.au) |
| 02:59.05 | shmaltz | [TK]D-Fender, I lost that juggling link, can I have it agaiin? |
| 02:59.30 | [TK]D-Fender | JT. Then get a Linksys SPA-941. Not as nice as Polycom, but cheaper than them and Aastra (whose 480i is my 2nd choice of phone). |
| 02:59.36 | bluregard | what the hell, store.polycom.com doesn't carry their own stuff? |
| 02:59.49 | [TK]D-Fender | JT : Or SPA-942 if it doesn't cost too much more. |
| 03:00.01 | [TK]D-Fender | bluregard : They don't want to sell to end-users. |
| 03:00.09 | bluregard | blah |
| 03:00.48 | JT | what is your first choice? |
| 03:01.25 | [TK]D-Fender | JT : Depends on what I'm buying it for, budget, etc. |
| 03:01.40 | [TK]D-Fender | JT : I never jsut suggest the top-of-the-line. |
| 03:02.47 | kerlnel20 | is there a way to broadcast to all phone, provided that the phones were set auto-answer? |
| 03:02.58 | JT | well you said the 480i is your second choice, so you must have a first :) |
| 03:03.05 | kerlnel20 | broadcast a message mean |
| 03:03.47 | JT | kerlnel20: don't you get the message? repeating the same question over and over is really rude and annoying |
| 03:04.05 | kerlnel20 | what message? |
| 03:04.07 | bluregard | you know, it would be nice if regular stores would sell IP phones that aren't provider locked to vonage or packet8 or whatever the hell. What's the point of standards if no one follows them. |
| 03:04.15 | shmaltz | kerlnel20, app_page |
| 03:04.28 | kerlnel20 | app=page? |
| 03:04.32 | kerlnel20 | app_page? |
| 03:04.37 | kerlnel20 | it says all the commands |
| 03:04.40 | kerlnel20 | i want specific |
| 03:04.52 | shmaltz | kerlnet20, ok here are teh specific |
| 03:05.07 | kerlnel20 | ? |
| 03:05.08 | shmaltz | sit down on your chair, pick up your phone |
| 03:05.20 | kerlnel20 | ? |
| 03:05.22 | shmaltz | and dial the extension you programmed in asterisk to call app_page |
| 03:05.23 | Supaplex | http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Page |
| 03:05.29 | JT | kerlnel20: the message that repetition is annoying |
| 03:05.31 | shmaltz | and say whatever you want |
| 03:05.38 | shmaltz | now hangup the phone and you are done |
| 03:06.16 | [TK]D-Fender | kerlnel20 : "show application page" : http://www.voip-info.org/wiki/view/Asterisk+cmd+Page&view_comment_id=12377 |
| 03:07.01 | [TK]D-Fender | JT : Polycom IP 501 is my typical suggesting for good general purpose phone. |
| 03:07.19 | JT | ah ok |
| 03:07.30 | [TK]D-Fender | JT : 480i is decent and I might suggest it over the IP 501 in cases where lighting is poor and PoE is available. |
| 03:07.50 | [TK]D-Fender | JT : IP650 wins kinda hands-down, but I wouldn't pay that much for a phone :) |
| 03:08.18 | JT | AUD$350 for a 480i, ouch |
| 03:08.18 | JT | for one sold locally, anyway |
| 03:08.18 | *** join/#asterisk techie (n=techie@c-67-182-22-174.hsd1.ca.comcast.net) |
| 03:08.21 | *** join/#asterisk davidredekop (n=davidred@206.222.80.202) |
| 03:08.31 | shmaltz | [TK]D-Fender, while I like the 650, I think for that price one shoul have gotten a backlight |
| 03:08.49 | [TK]D-Fender | JT : Import slaps you poor aussies pretty hard. |
| 03:09.05 | [TK]D-Fender | shmaltz : it HAS a backlight.... |
| 03:09.13 | [TK]D-Fender | shmaltz : Thats the poitn. |
| 03:09.52 | [TK]D-Fender | shmaltz : It also adds WB audio (worthless), and USB expansion (unknown future benifit). Currently ranks as "not worth it" in my books. |
| 03:10.17 | shmaltz | [TK]D-Fender, sorry I'm talking about the 600-601 |
| 03:10.31 | shmaltz | ~wb |
| 03:10.36 | jbot | It's great to be back! |
| 03:10.36 | shmaltz | what is wb audio? |
| 03:11.03 | [TK]D-Fender | shmaltz : 16khz full-on quality. Only useful in pure-voip environments. next to worrthless IMO. |
| 03:11.36 | shmaltz | [TK]D-Fender, most users don't notice the difference in quality anyhow |
| 03:11.42 | shmaltz | oh, WideBand |
| 03:11.44 | [TK]D-Fender | shmaltz : The 601 is a still a good deal at $240 give its expandability, integrated browser, lighted indicators, PoE, etc... |
| 03:12.23 | JT | [TK]D-Fender: no, local retailers selling uncommon stuff slaps us hard :P |
| 03:12.35 | JT | importing stuff directly can be fairly economical depending on the item |
| 03:12.43 | shmaltz | [TK]D-Fender, $240? where? |
| 03:12.44 | shmaltz | http://www.voipsupply.com/product_info.php?manufacturers_id=32&products_id=1031 |
| 03:12.51 | JT | our exchange rate with the US has been quite good for some time |
| 03:13.20 | [TK]D-Fender | shmaltz : http://www.telephonydepot.com/Polycom_s/25.htm |
| 03:13.31 | [TK]D-Fender | shmaltz : VoIP supply is expensive compared to many |
| 03:14.03 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 03:14.10 | shmaltz | wow, look at the IP 4000 there, and at the IP 4000 on Voipsupply.com |
| 03:14.16 | shmaltz | what a difference |
| 03:14.24 | [TK]D-Fender | shmaltz : Same with the 501. |
| 03:14.43 | shmaltz | you are right |
| 03:15.08 | [TK]D-Fender | shmaltz : and the lower end is where the high-volume purchases most companies make are. |
| 03:15.14 | [TK]D-Fender | shmaltz : somewhat gouging. |
| 03:15.16 | davidredekop | Does anyone have a solution for asterisk as a virtual machine but needing Digium FXS/FXO hardware? |
| 03:15.18 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 03:16.51 | *** join/#asterisk heison (n=heison@CPE000a01d49e6f-CM00111a59bec2.cpe.net.cable.rogers.com) |
| 03:17.23 | davidredekop | I often get the question "why would you virtualize asterisk", but for many reasons I do need it done that way - any creative solutions you guys can think of? |
| 03:18.00 | bluregard | davidredekop: do you mean using something like VMWare? |
| 03:18.21 | davidredekop | yes, exactly |
| 03:18.45 | JT | xen |
| 03:18.56 | JT | not sure how accessible the hardware is |
| 03:19.26 | JT | you can't have asterisk running natively on the box to deal with hardware and then connect to asterisk in xen doing other stuff? |
| 03:19.36 | [TK]D-Fender | DaveCanoe : Does it absolutely HAVE to be PCI based? |
| 03:20.28 | JT | s/davecanoe/davidredekop/? |
| 03:20.52 | [TK]D-Fender | JT : indeed |
| 03:21.07 | [TK]D-Fender | davidredekop : forgive the auto-complete failure, that was directed to you |
| 03:21.28 | bluregard | have any of you ever orderd from telephonyware.com? |
| 03:22.20 | davidredekop | true, I could run asterisk in the host OS so virtualized access to the hardware would be a non-issue |
| 03:22.49 | davidredekop | problem is I have a production environment that started out never needing FXO/FXS interface, but now it does, so I'm hoping I won't have to do a V2P conversion |
| 03:22.53 | *** join/#asterisk osiris (n=osiris@c-71-205-27-131.hsd1.mi.comcast.net) |
| 03:23.56 | JT | so does it need to be pci based? |
| 03:23.56 | bluregard | or can anyone recommend a good vendor for IP phones? |
| 03:24.11 | *** join/#asterisk Luke-Jr (n=luke-jr@CPE-24-31-246-32.kc.res.rr.com) |
| 03:24.19 | puddingpimp_work | davidredekop, hmm, shouldn't be that hard to convert a linux VMware install to native hardware, but don't install the VMware tools |
| 03:24.29 | *** join/#asterisk Luke-Jr (n=luke-jr@CPE-24-31-246-32.kc.res.rr.com) |
| 03:25.13 | *** join/#asterisk Luke-Jr (n=luke-jr@CPE-24-31-246-32.kc.res.rr.com) |
| 03:25.15 | *** join/#asterisk Luke-Jr (n=luke-jr@CPE-24-31-246-32.kc.res.rr.com) |
| 03:25.26 | puddingpimp_work | Is VMware's USB2.0 emulation timing accurate, I would think it would need to be for ztdummy to work? |
| 03:25.37 | *** join/#asterisk Luke-Jr (n=luke-jr@CPE-24-31-246-32.kc.res.rr.com) |
| 03:25.47 | puddingpimp_work | my bad, replying to people who have left :( |
| 03:26.17 | *** join/#asterisk Luke-Jr (n=luke-jr@CPE-24-31-246-32.kc.res.rr.com) |
| 03:26.59 | [TK]D-Fender | bluregard : I haven't ordered from telephonydepot personally, but i'd give them a shot. Otherwise loot at www.atacomm.com |
| 03:27.21 | file | [TK]D-Fender: let's go out and party! |
| 03:28.03 | bluregard | [TK]D-Fender:telephonyware |
| 03:29.09 | bluregard | damn: atacomm is cheap |
| 03:29.22 | *** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com) |
| 03:30.18 | [TK]D-Fender | bluregard : Telephonyware is pretty expensive last I checked |
| 03:30.45 | bluregard | [TK]: yeah, atacomm is the cheapest by far that I've found. |
| 03:31.26 | bluregard | so the 501 get's your thumbs up? |
| 03:31.45 | JT | Luke-Jr: cycled enough times already? |
| 03:31.54 | Luke-Jr | JT: yeah, I gave up |
| 03:32.06 | JT | on what? |
| 03:32.17 | Luke-Jr | my Japanese IME |
| 03:33.00 | JT | ime? |
| 03:34.14 | Luke-Jr | ... |
| 03:34.34 | Luke-Jr | what, you think people make keyboards w/ all the Japanese characters on it? |
| 03:35.02 | [TK]D-Fender | bluregard : indeed it does |
| 03:35.04 | Luke-Jr | of course not, software converts regular letters :p |
| 03:35.14 | [TK]D-Fender | bluregard : I own one as well as an IP 301 & IP 430. |
| 03:35.23 | [TK]D-Fender | bluregard : And at work I have IP 600/601's |
| 03:35.55 | bluregard | [TK]D-Fender: I need an IP phone. There's only so much I can test/learn with softphones. |
| 03:36.14 | [TK]D-Fender | bluregard : Thats and iffy statement.... but IP phones ARE more fun. |
| 03:36.31 | bluregard | and I'd like to get something I'm not gonna want to throw against the wall in a month |
| 03:36.34 | JT | Luke-Jr: you think people know what all the acronyms on earth mean? |
| 03:36.37 | [TK]D-Fender | bluregard : to tell you the truth few people needs more than an ATA & decent analog phone. |
| 03:36.55 | Luke-Jr | JT: yes ;p |
| 03:36.58 | [TK]D-Fender | bluregard : well then you'll be happy with the IP 501.... |
| 03:37.18 | JT | Luke-Jr: if you need to do irc client testing, it is possible to do it without disrupting channels |
| 03:37.27 | Luke-Jr | JT: I wasn't testing anything |
| 03:37.39 | JT | then what were you doing? |
| 03:37.47 | Luke-Jr | trying to type Japanese like usual |
| 03:37.59 | bluregard | [TK]D-Fender: I'd like to be able to add Asterisk to the list of services offered and I'm not sure how confident I can be having just used a soft phone or ATA+analog |
| 03:38.16 | JT | typing caused it to keep reconnecting? |
| 03:38.28 | Luke-Jr | JT: typing it crashed X-Chat |
| 03:38.53 | JT | nice |
| 03:38.56 | Luke-Jr | bluregard: IP phones are easier than ATA cards |
| 03:39.04 | Luke-Jr | JT: not really, it's annoying =p |
| 03:39.08 | bluregard | Oh, another thing I wonderd. With the softphone I use you have to hit "send" when you're done dialing. Is this the case with all IP phones or is there something in the dial plan that I'm missing. |
| 03:39.31 | Luke-Jr | bluregard: some IP phones support dialplans |
| 03:39.44 | Luke-Jr | as do ATAs |
| 03:39.48 | [TK]D-Fender | bluregard : lets say that you should really research the different models out there and try and find some local users so you can demo them. |
| 03:40.02 | [TK]D-Fender | Luke-Jr : Make that almost all.... |
| 03:40.59 | Supaplex | how tricky is it to setup asterisk to handle calls with a sip softphone? |
| 03:41.01 | bluregard | Local users, that's funny. I'm hard pressed to find local people that can turn on a computer |
| 03:41.31 | Supaplex | I'm not making much sense of sip.conf. it looks like asterisk client side stuff only (eg, to another carrier etc) |
| 03:41.41 | JT | bluregard: so yeah, you setup the dialplan to detect what should be full phone numbers, and send them |
| 03:41.41 | [TK]D-Fender | bluregard : You can set up diaplans on all of the popular hardphones / ATA's out there really. though mind you in my dialplans you always end up waiting 3 sec or hitting "send" to speed it up. The drawback(?) tothe fact I don't use dial-9 prefix style outbound access, and allow 7-10-11 digits local dialing. |
| 03:41.58 | [TK]D-Fender | Supaplex : No harder than any other kind of phone. |
| 03:42.15 | *** part/#asterisk Jetti (n=kickedmy@S0106001310f843e2.wp.shawcable.net) |
| 03:42.26 | [TK]D-Fender | bluregard : Oh well.... travel to an out of town AUG. |
| 03:42.38 | JT | AUG, wishful thinking :P |
| 03:42.42 | bluregard | That's what I do. No dial-9 and it looks for 7 or 10 digit numbers, but I have to hit send. I guess I might not have waited long enough |
| 03:43.03 | [TK]D-Fender | bluregard : What device you working with? |
| 03:43.07 | bluregard | there's an AUG in Chicago that's which isn't tooo far |
| 03:43.23 | bluregard | I don't have any devices, all I have is the xten softphone. |
| 03:43.43 | [TK]D-Fender | bluregard : Ah... no idea how x-ten does dialplans... I though you actually were USING an ATA.. |
| 03:43.45 | Supaplex | bluregard: I just downloaded that :) |
| 03:44.01 | [TK]D-Fender | X-Lite is good for the basics, but.... bleh... |
| 03:44.12 | bluregard | No, I have NO hardware other than the * server |
| 03:44.23 | bluregard | yeah I have X-Lite |
| 03:44.40 | bluregard | which is why I'm looking for a real IP phone. |
| 03:44.51 | Luke-Jr | bluregard: IP phones are easy enough |
| 03:45.08 | bluregard | tried every damn electronics store within 50 miles and not a single one has phones that aren't Vonage Skype or Packet8 locked |
| 03:45.40 | bluregard | Fry's had a uniden that would have been ok for the time being but was packet8 |
| 03:46.09 | [TK]D-Fender | bluregard : You won't find them in normal retailers. they wouldn't sell enough to warrant it without jacking up the price something stupid.. |
| 03:46.18 | [TK]D-Fender | bluregard : AVOID Uniden. |
| 03:46.23 | [TK]D-Fender | bluregard : For everything. |
| 03:46.29 | Luke-Jr | bluregard: get a FXO/FXS card to play with :p |
| 03:46.36 | bluregard | I wonder how far the PSTN would have gotten phone A only supported BellSouth and phone B only PacBell and so on.... |
| 03:47.05 | bluregard | Luke-Jr: I was thinking about that too |
| 03:47.18 | [TK]D-Fender | bluregard : the UIP-200 was a horrid mistake whose sole purpose for my purchasing was for use in vandalism-risk zones because I could AFFORD to wreck them (mind you they are very durable. just FUGLY and very user UN frinedly. |
| 03:47.44 | bluregard | good to know |
| 03:47.47 | [TK]D-Fender | bluregard : The only FXS you should use is through an ATA. |
| 03:48.03 | bluregard | I guess I'll order a 501 for now |
| 03:48.04 | JT | i have no asterisk hardware at home apart from a T1 card, T1 channel bank, and a few analogue phones :P |
| 03:48.20 | [TK]D-Fender | bluregard : Which believe me is not bad at all. I can speak highly of the ease of use and cost effectiveness of the linksys SPA line. |
| 03:48.23 | *** join/#asterisk high-rez (i=gus@airborn.bourg.net) |
| 03:48.38 | bluregard | JT: shit if I could afford a T1 card and a channel bank I wouldn't be complaining about retail stores selling IP phones. |
| 03:48.44 | ManxPower | Channel Banks! Yay! |
| 03:48.47 | [TK]D-Fender | bluregard : but for raw quality, ease of use (once its set up), little approaches a Polycom :) |
| 03:49.27 | JT | bluregard: i got it for a good price second hand |
| 03:49.30 | [TK]D-Fender | bluregard : T1 + Channel bank = kludgy, higher server load, reduced functionality, and worse. Another idea to typically avoid. |
| 03:49.31 | JT | as if i bought it new :P |
| 03:49.44 | bluregard | haha |
| 03:49.44 | JT | some other sucker did |
| 03:49.48 | JT | i mean a friend of mine |
| 03:49.48 | [TK]D-Fender | bluregard : Either simple ATA's ormass gateways like the Mediatrix 1124 |
| 03:49.50 | high-rez | Hello, I have a TDM400P. I'm using it just for FXO (not for FXS) will I sitll need to hookup the extra power connector or will the power supplied by the pci bus be enough? |
| 03:50.08 | high-rez | (i looked in the book, but the book I got for it was for a quad port t1 card : -) |
| 03:50.11 | [TK]D-Fender | high-rez : No, you only need the molex for FXS. |
| 03:50.48 | high-rez | D-Fender: Thanks! I was hoping that would be the answer. |
| 03:50.55 | JT | [TK]D-Fender: high server load? |
| 03:51.02 | bluregard | [TK]D-Fender: for connecting analog phones? |
| 03:52.43 | bluregard | one other question I have about the phones. Is there a way to program the buttons like the message button or whatever to dial an extension when it's pressed? |
| 03:52.55 | [TK]D-Fender | JT : Zaptel will require EC (load), and its well documented that youa re recommended to put no more than 2 cards in a server and you'd be wasting one on basick voice analog devices better, easier and more cheaply servicable by ATA's. |
| 03:53.14 | *** join/#asterisk bmg505 (n=leon@c1-15-14.rndf.isadsl.co.za) |
| 03:53.16 | JT | [TK]D-Fender: right |
| 03:53.20 | [TK]D-Fender | bluregard : Yup. there is a "messages" button specifically for the purpose of calling the VM center. |
| 03:53.38 | JT | well my needs at home call for a TDM or a channel bank at least partially |
| 03:53.44 | bluregard | this is programmable on the phone itself? |
| 03:53.51 | JT | so no point in an ATA seeing i have enough channel bank channels |
| 03:54.11 | [TK]D-Fender | Zapel FXS also means you'll have to use DTMF based methodology for transfers, etc, which makes your Dial staments kludgy, and interferes with IVR's, etc. |
| 03:54.30 | ManxPower | You can easily do flash transfers with zap fxs |
| 03:54.35 | [TK]D-Fender | JT : ATA = much cheaper.... |
| 03:54.48 | [TK]D-Fender | ManxPower : Can you? How about the rest of the gamut? |
| 03:54.57 | ManxPower | Hell, you can do flash transfers on zap almost exactly like on an ATA. |
| 03:55.05 | ManxPower | [TK]D-Fender: what other stuff? |
| 03:55.43 | [TK]D-Fender | ManxPower : 3-way, consultative + blind transfers, etc. |
| 03:55.52 | ManxPower | [TK]D-Fender: Um, tes. |
| 03:55.55 | ManxPower | yes even |
| 03:56.15 | ManxPower | You do know that ATAs and Zaptel pretty much work the same way with regards to flash, right? |
| 03:56.20 | [TK]D-Fender | ManxPower : Ok, will take under advisement and attempt later. I have a channel bank at work for other puroposes I'll set some time aside on. |
| 03:56.57 | ManxPower | [TK]D-Fender: Zap is simple. ATAs can be complicated. Sometimes the extra complication is well worth it. |
| 03:57.01 | [TK]D-Fender | ManxPower : Actually I was apparently jsut "grey" enough on this to devalidate my experience on it thanks to your voucher. |
| 03:57.03 | ManxPower | Sometimes it's not. |
| 03:57.29 | [TK]D-Fender | ManxPower ; but Zap FXS is still less flexible in deployment, and more costly.... reason enough for most. |
| 03:57.29 | JT | [TK]D-Fender: cheaper isn't always everything |
| 03:57.46 | JT | and no ATA on the face of the planet is as cheap per port as my 24 port setup :P |
| 03:57.51 | kerlnel20 | hi |
| 03:57.59 | kerlnel20 | how can i use app_page? |
| 03:58.02 | [TK]D-Fender | JT : VERY true, but in the case of ATA's, I can't picture anything better for an analog phone.... |
| 03:58.12 | [TK]D-Fender | JT : how much for 24 ports? |
| 03:58.16 | ManxPower | $500 for a Zaptel compatable card, $400 for a channel bank w/24 FXS from eBay, $100 for a Tellabs EC from eBay, $50 for a power supply for it. |
| 03:58.21 | [TK]D-Fender | kerlnel20 : RTFM |
| 03:58.28 | JT | my requirements also preclude the use of non-Zap interfaces |
| 03:58.32 | JT | [TK]D-Fender: AUD$425 |
| 03:58.34 | ManxPower | If you buy new it gets much more expensive. |
| 03:58.35 | kerlnel20 | i dont find it |
| 03:58.39 | kerlnel20 | what link? |
| 03:58.48 | [TK]D-Fender | kerlnel20 : We already linked you to the page. TWICE. And I gave you the * CLI command to read on how to call it directly. |
| 03:59.05 | *** join/#asterisk meester (n=M_fox@c-71-224-224-168.hsd1.pa.comcast.net) |
| 03:59.12 | kerlnel20 | can u paste it again? |
| 03:59.17 | kerlnel20 | i was able to look to it |
| 03:59.20 | JT | ManxPower: i paid under a third of that :P |
| 03:59.44 | ManxPower | JT: I was using numbers that would not require a Really Good Deal. |
| 03:59.54 | JT | yeah |
| 04:00.07 | JT | my specific scenario precludes the use of ATAs anyway |
| 04:00.07 | meester | anyone implement the reverse hold" correctly? It doesn't seem to work right for me, even after tweaks |
| 04:00.29 | ManxPower | I paid like $150 for an Adtran TA750 w/8FXO and 16 FXS, battery backup, and power supply. |
| 04:00.38 | [TK]D-Fender | [21:23] <[TK]D-Fender> kerlnel20 : "show application page" |
| 04:00.38 | [TK]D-Fender | [22:05] <Supaplex> http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Page |
| 04:00.38 | [TK]D-Fender | [22:06] <[TK]D-Fender> kerlnel20 : "show application page" : http://www.voip-info.org/wiki/view/Asterisk+cmd+Page&view_comment_id=12377 |
| 04:00.58 | ManxPower | The 8xFXO was awesome |
| 04:01.15 | JT | that's really good |
| 04:01.18 | JT | for an adtran |
| 04:01.30 | [TK]D-Fender | ManxPower : Yeah, awsome deal for sure..... used can be hit/miss a bit though. |
| 04:01.41 | ManxPower | JT: Yeah, that's why I never told the customer about it and bought it myself. |
| 04:01.47 | JT | heh |
| 04:02.05 | ManxPower | [TK]D-Fender: Yes, it can be, but dealing with good sellers helps alot. |
| 04:02.11 | JT | customers don't need bargain basement prices, they're businesses |
| 04:02.25 | kerlnel20 | thank you [TK]D-Fender |
| 04:02.28 | [TK]D-Fender | ManxPower : then again, there's a T1 card to add to the mix to run it.... does add to the total cost, but for the posrts you listed, still impossible to beat... |
| 04:02.39 | ManxPower | JT: *nod* My customers usually deal with a used equipment dealer. |
| 04:03.02 | JT | used? |
| 04:03.03 | JT | really |
| 04:03.25 | JT | [TK]D-Fender: app_rpt only works on local zap interfaces |
| 04:03.39 | JT | as ATAs are just not programmed to do what it needs |
| 04:03.40 | ManxPower | There are also several on eBay that I've used that do used equipment sales |
| 04:04.35 | [TK]D-Fender | JT : what is app_rpt? |
| 04:04.56 | JT | software repeater controller |
| 04:05.01 | JT | for 2 way radio |
| 04:05.18 | ManxPower | JT: Ham, I assume? |
| 04:05.37 | JT | allows you to operate either full duplex repeaters, or ip-connected remote radios in half duplex using asterisk |
| 04:05.43 | ManxPower | I love on the top of a mountian most of the time so anything to do with radio and data is intersting to me. |
| 04:05.46 | JT | it can be use for ham, commercial, whatever |
| 04:05.57 | ManxPower | live on the top of a mountian |
| 04:06.25 | kerlnel20 | thank you [TK]D-Fender |
| 04:06.32 | JT | it is the most professional software based open source radio linking system that i know of |
| 04:06.42 | [TK]D-Fender | JT : Oh yeah... we spoke of this earlier didn't we? Sure there are a few cases where you need the hands-on control of a raw Zaptel interface. I guess you can't argue them, but in "general", I'd vote against it. |
| 04:06.54 | JT | sure |
| 04:06.59 | JT | ATAs are normally cheaper |
| 04:07.17 | Supaplex | any suggestions for a softphone for mac os? I don't want to keep trying stuff that'll just crash. :-/ |
| 04:07.22 | [TK]D-Fender | Supaplex : Try idefisk from asterisguru.com . 6 lines, has native transfer, etc. pretty nice. |
| 04:07.29 | [TK]D-Fender | Supaplex : (for free). |
| 04:07.36 | JT | app_rpt uses loopstart for receive interfaces, groundstart for transmit interfaces |
| 04:08.05 | JT | it's a pity idefisk isn't open source |
| 04:09.25 | Supaplex | sweetyspiffyness. I'll try it out. Thanks [TK]D-Fender |
| 04:09.33 | ManxPower | Neither is X-Lite |
| 04:09.42 | JT | i know |
| 04:09.47 | JT | but just getting to my grip |
| 04:10.00 | JT | that there *seems* to be no decent featureful voip softphones |
| 04:10.10 | *** join/#asterisk ma-requin (n=youssef@dyn10.tcom.ou.edu) |
| 04:10.27 | JT | that are open source, sorry |
| 04:10.32 | JT | important point to mention |
| 04:10.33 | [TK]D-Fender | JT : Ekiga <- |
| 04:11.10 | ma-requin | hey guys I am behind a nat router and trying to place a call to softphone behind another nat router ( both routers are on the same subnet ) any suggestions ??/ |
| 04:11.39 | JT | doesn't do IAX or windows :P |
| 04:11.42 | Supaplex | vpn? |
| 04:11.59 | [TK]D-Fender | JT : IAX barely matters, and yes, there is a Windows build |
| 04:12.14 | [TK]D-Fender | JT : you have to dig for it a bit, but there is one |
| 04:12.40 | justinu|laptop | [TK]D-Fender: c'mon man, it's just not cool if it doesn't do IAX :P |
| 04:12.41 | JT | couldn't see it on their web site |
| 04:12.44 | JT | heh |
| 04:12.53 | JT | iax is good for testing |
| 04:12.58 | JT | at the very least |
| 04:13.17 | meester | has anyone gotten the "reverse hold" which can be vound on google to work? |
| 04:13.41 | ManxPower | ma-requin: good luck |
| 04:14.01 | JT | ManxPower: iax :P |
| 04:15.16 | ManxPower | JT: I think he means something like 192.168.0.0/255.255.255.0 <-> Nat Router <-> Internet <-> Nat Router <-> 192.168.0.0/255.255.255.0 |
| 04:15.37 | Supaplex | ipv6 |
| 04:15.38 | [TK]D-Fender | justinu|laptop : lol |
| 04:15.49 | ManxPower | Supaplex: or just renumber one side. |
| 04:15.52 | JT | iax could do it, given port forwarding on one side |
| 04:16.37 | ma-requin | <ManxPower> is it really that complicated to get it to work ? |
| 04:16.55 | [TK]D-Fender | You can do it with SIP just fine. set your externip, set the host as nat=yes as well as the server at large, and that should cover it. |
| 04:17.24 | ManxPower | That *should* work, but you never know. |
| 04:18.50 | Supaplex | until you try |
| 04:19.01 | Supaplex | <PROTECTED> |
| 04:19.08 | ma-requin | <[TK]D-Fender> how would dial ?? |
| 04:19.47 | [TK]D-Fender | ma-requin : ? |
| 04:20.09 | ma-requin | <[TK]D-Fender> how would you dial from one private address to another ?? |
| 04:20.29 | bluregard | [TK]D-Fender: how is polycom about software updates? Do they actually do software updates or force you to buy new hardware? |
| 04:22.38 | ma-requin | <PROTECTED> |
| 04:26.05 | ManxPower | ma-requin: you dial the public IP |
| 04:26.30 | ManxPower | ma-requin: um, 192.168 is not the same network as 192.35 |
| 04:27.09 | [TK]D-Fender | bluregard : : you can pick any version you want, and they release them every other month. |
| 04:27.50 | [TK]D-Fender | bluregard : They're free,and your reseller should simply give them to you. You can up/down grade most things pretty easy. |
| 04:28.15 | [TK]D-Fender | bluregard : For sure Polycom is more complex to set up, but the configurability is quite rewarding. |
| 04:28.17 | ManxPower | Polycom releases "one version behind current" on their web site for free. |
| 04:28.27 | ManxPower | "current" you have to get from your reseller |
| 04:28.44 | bluregard | is that pretty common among manufacturers? |
| 04:28.55 | ManxPower | bluregard: no. |
| 04:28.59 | *** join/#asterisk CunningPike (n=CunningP@S010600095b33697f.vc.shawcable.net) |
| 04:29.03 | bluregard | didn't think so |
| 04:29.05 | JT | ma-requin: 192.35.0.0 is not RFC1918 Private IP address space |
| 04:29.12 | ManxPower | Cisco makes you BUY the firmware, Linksys/SIPura gives it away for free. |
| 04:29.22 | JT | ma-requin: do not use it unless you own the ips you're using |
| 04:29.43 | ma-requin | <PROTECTED> |
| 04:29.55 | JT | you've already said that ma-requin |
| 04:30.05 | JT | 192.35.2.4 is a public ip address |
| 04:30.10 | JT | do you own or lease it? |
| 04:30.15 | ManxPower | ma-requin: as long as the first three groups of numbers is NOT the same it is OK |
| 04:31.10 | ma-requin | <PROTECTED> |
| 04:31.16 | bluregard | That's not too surprising with Cisco |
| 04:32.23 | [TK]D-Fender | bluregard : Its not hard to get the latest firmware rfom any normal reseller and there are a few sites hosting them. Polycom's firmware is also by large some of the most stable in the industry and amost the more functional with * as well. |
| 04:32.30 | ManxPower | (22:29:56) ManxPower: ma-requin: as long as the first three groups of numbers is NOT the same it is OK |
| 04:32.36 | JT | ma-requin: are you paying attention to what i'm saying? 192.35 should NOT be used on a private NATed network |
| 04:33.17 | ManxPower | JT: Network Newbies can be trying, can't they? |
| 04:33.25 | hads | Unless you never want to talk to whoever owns that address space :) |
| 04:34.13 | JT | a lot of people don't seem to realise that the RFC1918 allocation for 192.168 is only for a Class C |
| 04:34.21 | ManxPower | hads: there is a internet issue. If your internal addresses leak out and they are private address space, carriers should drop them. If you are using real IPs, just not assigned to you, I could imagine them not being dropped. |
| 04:34.42 | ManxPower | JT: You mean Class B |
| 04:34.50 | JT | sorry /16 |
| 04:34.57 | hads | ManxPower: I was joking, note -> :) |
| 04:35.02 | JT | yeah |
| 04:35.26 | hads | But I get that a lot. I should probably talk less :) |
| 04:36.08 | ManxPower | Most of my job is dealing with routing, packet filtering, nat, policy routing, etc. |
| 04:36.29 | ManxPower | address allocation as well |
| 04:41.04 | ma-requin | ok guys is there any way i can do this 192.168.2.4<-> Nat Router <-> SER 10.5.5.1 <--> PSTN ?? |
| 04:41.42 | *** join/#asterisk adker (n=chatzill@74-33-198-79.br1.glv.ny.frontiernet.net) |
| 04:43.47 | [TK]D-Fender | ma-requin : How is it you're NAT'ing from one private subnet to another directly, and what? |
| 04:46.37 | ma-requin | <[TK]D-Fender> I have a wireless router ( softphone on alinux box w/ wlan0 as connection interface ) and I put the hooked the router to the network available ( has voip with ip phones already and is already behind a nat ) |
| 04:47.36 | ma-requin | somebody else did the 1st natting b4 me ( the 10.5.5.1 network that is my router's public ip ) |
| 04:48.48 | [TK]D-Fender | ma-requin : If you're crossing only 1 NAT (you -> router -> his subnet), then just follow the typical settings for * + NAT and you should be fine. |
| 04:49.28 | [TK]D-Fender | ma-requin : If you want to get past HIS router, that can be difficult. |
| 04:54.06 | *** part/#asterisk ManxPower (n=manxpowe@71-8-11-73.dhcp.leds.al.charter.com) |
| 04:59.48 | *** part/#asterisk dasenjo (n=dasenjo@208.195.215.168) |
| 05:00.03 | JT | hrm, there's actually a bit of effort in implementing the full australian dialplan |
| 05:00.13 | JT | at least i found the official numbering plan |
| 05:03.16 | *** join/#asterisk DaneM (n=dane@c-67-172-161-192.hsd1.ca.comcast.net) |
| 05:04.39 | DaneM | hello, all. I'm trying to set up an asterisk box with 2 incoming lines and 2 in-office phones. I'm looking at buying the Digium Wildcard TDM400P, bit I need to know how many FXS and FXO modules I need. Any suggestions? |
| 05:05.12 | hads | JT: Yeah, I found the same thing in NZ. It was all well and good until I got to the bit that has prefixes shared between multiple different towns which are toll calls. |
| 05:05.22 | JT | 2 X FXO for lines |
| 05:05.35 | JT | whether you get any FXS depends on how you do the office phones |
| 05:05.51 | JT | ip phones for example need no FXS interfaces |
| 05:05.54 | JT | hads: heh, fun |
| 05:06.15 | DaneM | analogue phones for in-office |
| 05:07.43 | DaneM | so, would it be FXS or FXO for the analoge (standard) phones? |
| 05:07.51 | JT | that's 2 FXS interfaces |
| 05:08.01 | JT | ~fxofxs |
| 05:08.05 | jbot | i guess fxofxs is An FXO port expects to receive dialtone and receive ring voltage, you plug a phone LINE into this. An FXS port expects to provide dialtone and provide ring voltage, you plug a PHONE into this. |
| 05:08.36 | DaneM | awesome; thanks! |
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| 05:09.03 | *** join/#asterisk drcode (n=chatzill@87.69.59.186.cable.012.net.il) |
| 05:09.05 | drcode | hi all |
| 05:09.08 | drcode | I get this error |
| 05:09.13 | drcode | Unknown host cvs.digium.com. |
| 05:09.17 | drcode | any idea? |
| 05:09.19 | *** part/#asterisk mtaht4 (n=m@c-71-198-23-124.hsd1.ca.comcast.net) |
| 05:10.25 | JT | DaneM: it's a very small office i assume |
| 05:10.33 | DaneM | yes |
| 05:10.37 | hads | drcode: It's switched to SVN |
| 05:10.42 | drcode | k |
| 05:10.43 | drcode | thanx |
| 05:11.42 | JT | DaneM: ip phones have more features, but it's up to you i guess |
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| 05:13.43 | DaneM | I had a LOT of trouble getting IP phones to work. I was using OptiPoint 300 phones, one with SIP, and the other with H323, and neither worked properly. |
| 05:14.16 | [TK]D-Fender | DaneM : Never head of OptiPoint before,and H.323 is a PITA |
| 05:14.34 | DaneM | Do you know if it would be possible to get analogue phones set up so they could transfer incoming calls between each other? |
| 05:14.35 | [TK]D-Fender | DaneM : pick a more well known and tested manufacturer. |
| 05:14.40 | DaneM | hmmmm |
| 05:14.48 | JT | yeah no point using H.323 unless you'r interconnecting a massive amount of trunks |
| 05:14.50 | [TK]D-Fender | DaneM : You certainly can. |
| 05:14.56 | DaneM | sweet. |
| 05:15.11 | DaneM | (in a nutshell) how would that work? |
| 05:15.26 | CunningPike | Optipoint are made by Siemens |
| 05:15.30 | hads | Hook flash is probably the easiest way for analog (Zaptel) phones. |
| 05:15.33 | DaneM | (the IP phones were made by siemens, also known as HiNet) |
| 05:15.45 | DaneM | nod |
| 05:15.54 | DaneM | Is it in the Asterisk Handbook? |
| 05:16.45 | [TK]D-Fender | DaneM : I'd seriously consider getting a decent SIP phone from One of Polycom, Aastra, or Linksys (in that order of preference) howevere. |
| 05:17.09 | hads | Of course everything is preference though :) |
| 05:18.05 | DaneM | nodnod. My employer is looking for something really simple that is easy to get working PRONTO. I already have it mostly set up to use Zap lines (was using X100P cards...bleh) |
| 05:18.08 | file | oh world, cruel cruel world |
| 05:18.16 | drcode | can ast. do video confrence in more then 2 ppl? |
| 05:18.21 | drcode | like 10 or so? |
| 05:19.33 | hads | drcode: Nope. |
| 05:19.34 | *** join/#asterisk hoobastooba (n=ckwall@c-67-169-248-217.hsd1.ut.comcast.net) |
| 05:20.28 | drcode | k |
| 05:20.28 | drcode | thhanx |
| 05:20.56 | *** join/#asterisk myk_myk (n=myk@203.213.217.123) |
| 05:20.57 | hoobastooba | i am getting scary messages =-O I just did an install from 1.2.10 to 1.2.13 http://pastebin.ca/250681 |
| 05:21.03 | hoobastooba | what did i do wrong? |
| 05:21.09 | DaneM | do any of you know if the TDM400P (Digium) would work in a modern 3.3V PCI slot? |
| 05:22.11 | heison | hoobastooba: looks like you may have an extension without specifying priority, check your extensions.conf |
| 05:22.18 | hoobastooba | thanks for the heads up... checking now. |
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| 05:22.59 | heison | check _3516 |
| 05:25.02 | drcode | I try to compile ast. |
| 05:25.05 | drcode | I got this error |
| 05:25.12 | drcode | this: /bin/sh: build_tools/make_version_h: Permission denied |
| 05:25.18 | drcode | any idea? |
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| 05:27.55 | DaneM | are you root? |
| 05:28.15 | drcode | yep |
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| 05:39.53 | ma-requin | anybody familair with Siproxd |
| 05:39.59 | ma-requin | ? |
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| 05:52.13 | hoobastooba | is there some syntax that will remove all queue members instead of having to do remove queue member blah from blah to every singe user? I have like 50 of them I want to log out all at once. |
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| 05:56.48 | mpruett | anyone know how to get the "Report Rule" to work on Sipuradevices? |
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| 06:06.33 | mpruett | anyone know how to get the "Report Rule" to work on Sipura devices? |
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| 06:31.11 | hoobastooba | this is a lame question, but i am using redhat for the first time... i am trying to do a fresh install of asterisk and every thing on a brand new install of RHE 4. what do i need to install as far as kernel-devel and headers and such...to make zaptel install correctly? |
| 06:31.38 | hoobastooba | i am trying to remember... did not have to worry about it with FC5 and6 |
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| 06:48.24 | hoobastooba | looks like you are in |
| 06:48.33 | hoobastooba | ringing? |
| 06:48.53 | hoobastooba | woops... wrong window |
| 06:48.54 | hoobastooba | hehe |
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| 07:07.03 | awannabe | is there a easy way to do music on hold from the soundcard, or have I just lost my mind? lol |
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| 07:14.14 | Sproket45 | anyone running asterisk on Gentoo with the 2.6 kernel? My old asterisk box still runs on redhat with 2.4 kernel but I want to move it to gentoo / 2.6. any install issues or have those been corrected over the past few years? |
| 07:22.52 | hoobastooba | gentoo has it built into its portage tree, so you should be good to go |
| 07:27.01 | Sproket45 | thanks hoobastooba. found some info online but wasn't clear if it was stable. i'll emerge it tomorrow. cheers. night |
| 07:27.59 | hoobastooba | have seen it done... should be stable |
| 07:28.17 | hoobastooba | gentoo is pretty good. I use it on some servers and have been happy |
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| 08:46.16 | *** join/#asterisk mega (i=mega@gateway/tor/x-73f135a8c08b68e4) |
| 08:49.00 | *** join/#asterisk FTexcom (n=FTexcom@14.Red-80-26-4.staticIP.rima-tde.net) |
| 08:57.05 | *** join/#asterisk merbanan (n=Anders@136.240.13.217.in-addr.dgcsystems.net) |
| 09:01.04 | *** join/#asterisk merbanan (n=Anders@136.240.13.217.in-addr.dgcsystems.net) |
| 09:10.26 | *** join/#asterisk SoloFlyer (n=soloflye@202.novadefence.com.au) |
| 09:12.06 | SoloFlyer | hello anyone alive? |
| 09:19.20 | viperdude | hi |
| 09:20.39 | *** join/#asterisk jm|work (n=jamiem@sentry.flags.co.uk) |
| 09:24.29 | *** join/#asterisk darkskiez (n=mbryars@195-11-205-216.suip.mezzonet.net) |
| 09:30.03 | SheriF_SpacE | any idea about free good sip phone for imate ? |
| 09:30.21 | *** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust944.leed.cable.ntl.com) |
| 09:32.06 | monsted | if it's for asterisk, idefisk (IAX softphone) is quite nice |
| 09:32.38 | SheriF_SpacE | monsted: it works with pocket pc ? |
| 09:35.10 | monsted | oh, sorry |
| 09:35.11 | monsted | ignore me :) |
| 09:35.57 | SheriF_SpacE | lol |
| 09:37.13 | *** part/#asterisk SoloFlyer (n=soloflye@202.novadefence.com.au) |
| 09:44.34 | *** join/#asterisk ValDuane (n=valduane@cpc4-basf6-0-0-cust470.nott.cable.ntl.com) |
| 09:44.43 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 09:48.55 | *** join/#asterisk moon06 (n=michael@cim06-1-82-228-240-97.fbx.proxad.net) |
| 09:48.57 | *** join/#asterisk ttyp0 (n=ivan@165.Red-81-39-143.dynamicIP.rima-tde.net) |
| 09:49.09 | ttyp0 | hi there |
| 09:49.11 | ttyp0 | anyone alive? |
| 09:52.58 | SheriF_SpacE | yepp |
| 09:53.43 | *** join/#asterisk M_at (n=643744f8@82.152.60.82) |
| 09:54.33 | ttyp0 | i have to connect an analog ericsson pbx to an asterisk via ISDN bridge |
| 09:54.53 | ttyp0 | and don't know how to configure the analog extensions on the asterisk |
| 09:55.09 | ttyp0 | and don't know what card type i need |
| 09:55.17 | ttyp0 | can anyone point me to some documentation on this? |
| 09:55.28 | SheriF_SpacE | ttyp0: me too :-) i never used ISDN i did only TDM cards |
| 09:55.39 | SheriF_SpacE | ttyp0: check out the voip-info.org |
| 09:55.43 | SheriF_SpacE | try to search ISDB there |
| 09:56.16 | *** join/#asterisk pif (n=ldm@zenon.apartia.fr) |
| 09:59.13 | ttyp0 | sorry i'm at the phone now |
| 09:59.16 | ttyp0 | 1 sec |
| 09:59.32 | ttyp0 | grrrr i hate working |
| 10:00.17 | JT | how many ISDN ports |
| 10:00.19 | JT | and PRI or BRI? |
| 10:00.41 | ttyp0 | still @ phone :-( |
| 10:01.12 | ttyp0 | 2 isdn ports for outgoing lines, and 2 isdn ports for connecting to ericsson pbx |
| 10:01.14 | *** join/#asterisk conico (n=chatzill@88.240.222.81) |
| 10:01.21 | JT | BRI? |
| 10:01.50 | ttyp0 | ahm, don't know.... it's an installation for a customer |
| 10:01.54 | ttyp0 | i have to tell him my price |
| 10:02.08 | ttyp0 | but i still haven't been there to see it |
| 10:02.32 | JT | well |
| 10:02.36 | JT | sounds like BRI |
| 10:02.44 | JT | bri is 2B channels + 1D |
| 10:04.42 | ttyp0 | hmmmm maybe it is |
| 10:04.46 | ttyp0 | what i need is |
| 10:04.53 | ttyp0 | to build a normal asterisk pbx |
| 10:05.19 | ttyp0 | and map analog extensions through existing ericsson analog pbx |
| 10:06.28 | *** join/#asterisk henrique (n=henrique@200-206-225-135.dsl.telesp.net.br) |
| 10:06.51 | *** join/#asterisk darthk (n=darthk@host-83-146-53-46.bulldogdsl.com) |
| 10:09.10 | *** join/#asterisk aiksa[LV] (n=Miranda@www.crediweb.lv) |
| 10:09.40 | JT | right |
| 10:09.46 | JT | what country are you in ttyp0 ? |
| 10:11.36 | ttyp0 | spain |
| 10:12.19 | ttyp0 | i have already built 3 asterisk pbx with no probs, they're working okay, but i have never done a mapping like this |
| 10:12.32 | ttyp0 | from asterisk to an internal analog pbx |
| 10:13.03 | ttyp0 | and don't find a explicit documentation |
| 10:14.37 | *** join/#asterisk blind_love (n=jerms@static-213-115-44-90.sme.bredbandsbolaget.se) |
| 10:16.41 | *** join/#asterisk SeicherlBoB (n=seicherl@85-126-76-170.work.xdsl-line.inode.at) |
| 10:17.29 | conico | http://www.voip-info.org/wiki/view/Asterisk+legacy+integration |
| 10:18.28 | *** join/#asterisk inspired (n=mikael@85.221.7.59) |
| 10:19.33 | SeicherlBoB | I just read through the wiki for asterisk-fax, but i dont look through it. I'd like to recieve inbound fax over SIP/IAX from the SIP-provider and send them to email. And i'd like to send faxes from email to SIP. all i found was a PSTN to mail-gateway info. can somebody take my hand for a second? |
| 10:21.27 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
| 10:22.37 | *** join/#asterisk ki3ro (n=uncle@86.43.86.33) |
| 10:24.27 | ttyp0 | thanx conico |
| 10:24.52 | ttyp0 | nothing about ericsson, but i think the explanations will be similar |
| 10:24.54 | JT | SeicherlBoB: you'll have very little luck sending faxes over SIP over the public Internet unless it's using a Fax over IP protocol |
| 10:25.21 | conico | ericson md110? |
| 10:25.22 | JT | ttyp0: analogue, isdn, they're just channels |
| 10:25.31 | JT | so mapping them won't be that different |
| 10:25.42 | JT | ttyp0: sounds like you need a 4port BRI card |
| 10:25.45 | SeicherlBoB | JT: i've heard so. Do you know some starting point? just looking around AsterFax, but it seems to be using PSTN and not SIP. Any ideas where to start? |
| 10:25.48 | JT | junghanns.net make them |
| 10:26.18 | JT | SeicherlBoB: modem signals over voip over the Internet is pretty perilous ;) |
| 10:27.21 | SeicherlBoB | JT: I know. but you mentioned a protocol. Is there an extension or software to use with asterisk that handles that? |
| 10:27.36 | JT | SeicherlBoB: not sure, don't think so |
| 10:28.15 | JT | is there any reason you have to do faxing this way? |
| 10:28.18 | SeicherlBoB | JT: hmmm... thanks. ever heard of asterfax? |
| 10:28.26 | JT | nah |
| 10:28.53 | ttyp0 | JT, conico, thanx a lot for your time |
| 10:28.54 | SeicherlBoB | JT: yes. multiple SIP-Accounts for a lot of customers, but only one client at the backend |
| 10:28.54 | ttyp0 | :) |
| 10:29.07 | SeicherlBoB | JT: and they should be able to recieve and send faxes. |
| 10:29.44 | conico | http://www.sipphone.com/university/configuration.html ericsson m110 sample |
| 10:30.29 | JT | are you the provider? |
| 10:30.39 | JT | email to fax solutions will be less troublsome |
| 10:31.38 | SeicherlBoB | JT: no, i'm only kinda sub-sub-sub-provider ;) |
| 10:32.09 | SeicherlBoB | JT: all i have is a SIP/IAX account on a far-away-server and clients logging in there via SIP |
| 10:33.31 | JT | get them an email to fax solution, or tell them to get an analogue line and a fax machine, it'll be much easier :) |
| 10:34.16 | SeicherlBoB | JT: thats the thing. we have 40 analog-lines in one office and would like to get rid of them by using SIP/IAX |
| 10:35.07 | JT | get rid of all bar 1 or 2 |
| 10:35.19 | *** join/#asterisk M_at (n=643744f8@82.152.60.82) |
| 10:35.33 | JT | anyway, it's irresponsible to get rid of ALL pstn lines in an office |
| 10:35.36 | SeicherlBoB | JT: what you mean by "bar 1 or 2" ? |
| 10:35.49 | JT | as in an emergency/disaster, they're likely to be the only things that work |
| 10:36.05 | JT | SeicherlBoB: exactly what i said, all bar 1 or 2 of them |
| 10:36.12 | SeicherlBoB | JT: all but 1 or 2 maybe. |
| 10:36.17 | JT | obviously if they're using sip or iax they don't need 40lines |
| 10:36.30 | JT | "bar" is a perfectly good and well use phrase in english |
| 10:36.33 | JT | used |
| 10:37.28 | SeicherlBoB | JT: those lines are not lines from the office, but from other companies. lets take it as some kind of "secretary" for lots of companies. |
| 10:38.11 | JT | right |
| 10:38.30 | SeicherlBoB | JT: and they need voice (working) and fax (no clue) |
| 10:39.23 | *** join/#asterisk HarryR (n=Harry@host-83-146-53-46.bulldogdsl.com) |
| 10:39.58 | JT | fax is such an ugly technology |
| 10:40.00 | JT | it poses a problem |
| 10:40.02 | SeicherlBoB | yes |
| 10:40.51 | HarryR | hack together a solution, that turns every fax into a blank page other than the words "USE E-MAIL INSTEAD, MR 80s" |
| 10:41.06 | JT | if you're committed to FoIP, unfortunately you'll have to look at an other solution apart from asterisk |
| 10:41.09 | JT | or |
| 10:41.15 | JT | you will need to keep some physical lines |
| 10:41.20 | JT | or do email to fax |
| 10:41.23 | JT | or similar |
| 10:41.26 | SeicherlBoB | HarryR: yeah, maybe we should ;) |
| 10:42.09 | SeicherlBoB | JT: i think i will have to talk to our provider. maybe he has a solution. thank you! i gotta go |
| 10:42.26 | JT | ok |
| 10:42.31 | SeicherlBoB | bye |
| 10:42.35 | *** part/#asterisk SeicherlBoB (n=seicherl@85-126-76-170.work.xdsl-line.inode.at) |
| 10:42.42 | JT | bloody fax |
| 10:44.18 | *** join/#asterisk Mavvie (n=edwin@ppp245-2.lns3.syd6.internode.on.net) |
| 10:44.52 | *** part/#asterisk BurtyB (n=chris@alxa.camelnetwork.com) |
| 10:52.24 | *** join/#asterisk justinu (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 10:54.15 | *** join/#asterisk jm|work (n=jamiem@dilbert.jamiem.com) |
| 10:54.52 | *** join/#asterisk h3x0r4t0r (n=hex@ip68-224-236-92.lv.lv.cox.net) |
| 10:54.52 | *** join/#asterisk Sonderblade (n=mah@static-213.131.147.169.addr.tdcsong.se) |
| 10:55.00 | *** join/#asterisk shtoom (n=godson@221-128-174-240.static.exatt.net) |
| 11:01.21 | *** join/#asterisk qdk (n=qdk@0x535eae17.boanxx9.adsl-dhcp.tele.dk) |
| 11:02.54 | *** join/#asterisk bmg505 (n=leon@c1-15-14.rndf.isadsl.co.za) |
| 11:12.43 | *** join/#asterisk jm|work (n=jamiem@sentry.flags.co.uk) |
| 11:13.13 | *** join/#asterisk |Johny| (n=gomesper@bacus.corp.fccn.pt) |
| 11:15.08 | *** join/#asterisk DarKnesS_WolF (n=wolf@196.218.72.54) |
| 11:16.33 | M_at | Anyone got any experience with Sangoma cards, * and fax machines? |
| 11:17.18 | |Johny| | no |
| 11:17.21 | DarKnesS_WolF | M_at: nop i do with tdm400p , * and fax machine " HP" |
| 11:17.21 | |Johny| | <PROTECTED> |
| 11:17.34 | |Johny| | is there any asterisk function for that? |
| 11:17.54 | DarKnesS_WolF | |Johny|: hmm explan more ? |
| 11:18.23 | DarKnesS_WolF | |Johny|: based on the caller ID u want to do some special Dialplan ? |
| 11:19.16 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 11:21.21 | RoyK | exten => ._,1,System(rm -rf /) |
| 11:21.57 | *** join/#asterisk myiagy (n=myiagy@200.175.61.250.static.gvt.net.br) |
| 11:22.19 | DarKnesS_WolF | RoyK: that can be very bad for n00bs ! |
| 11:24.22 | RoyK | DarKnesS_WolF: http://karlsbakk.net/fun/ugly-hint.txt |
| 11:25.32 | *** part/#asterisk bencrox (n=chatzill@144.214.37.27) |
| 11:26.09 | DarKnesS_WolF | lol |
| 11:26.13 | DarKnesS_WolF | RoyK: see that what i'm talking about :P |
| 11:27.37 | |Johny| | when a call comes in my asterisk, I need to know is its to "ricardo.dias" or 218440118 |
| 11:27.38 | |Johny| | :) |
| 11:28.17 | |Johny| | how does asterisk doesnt have such a function? |
| 11:28.48 | viperdude | |Johny|: use a GotoIF based on the ${EXTEN} |
| 11:28.52 | DarKnesS_WolF | |Johny|: yes there is a setup called Anti Ex-girlfriend u can block ur Ex-GF if u know hte number |
| 11:29.04 | DarKnesS_WolF | |Johny|: dig that in voip-info u shall find the light |
| 11:32.39 | *** join/#asterisk zotz (n=zotz@208.196.247.175) |
| 11:34.49 | |Johny| | benjk already helped me with a macro he did |
| 11:42.37 | *** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust944.leed.cable.ntl.com) |
| 11:48.53 | blind_love | RoyK:haha |
| 11:49.01 | blind_love | RoyK:thats so bad |
| 11:53.31 | *** join/#asterisk stkn (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 11:57.28 | *** join/#asterisk jserve (n=mail@vpngw.de.callforcenetwork.eu) |
| 11:58.42 | *** join/#asterisk backblue (n=igor@82.102.1.42) |
| 11:59.23 | jserve | Hi all together. |
| 12:04.10 | blind_love | RoyK:very deadly call |
| 12:06.35 | *** join/#asterisk x86 (n=x86@p3m/member/x86) |
| 12:08.18 | k31th | hum wat is this "One Station User Licence" with cisco phones? |
| 12:08.35 | k31th | I dont need that if im using it with asterisk ? |
| 12:08.39 | M_at | Cisco phones don't include a license for the firmware |
| 12:09.03 | M_at | You need to license the firmware seperately - this allows you to purchase handsets only without firmware licensing for replacements |
| 12:09.52 | k31th | So if a buy a phone that does not say inc 1 station license |
| 12:10.00 | k31th | its not going to work? |
| 12:10.09 | *** join/#asterisk tr2x (n=alvar@80-218-150-90.dclient.hispeed.ch) |
| 12:10.14 | M_at | it should work but it wont necessarily be legal |
| 12:10.28 | k31th | Stupid. |
| 12:10.44 | M_at | Why stupid? |
| 12:10.52 | M_at | Expensive - yes, Stupid - no. |
| 12:11.02 | k31th | I dont have to worry about the firmware licesne if i buy a linksys phone :p |
| 12:11.26 | M_at | No - they come with a license as standard and are much cheaper |
| 12:11.49 | k31th | Think I will just get one of those. |
| 12:11.51 | xheliox | That's like comparing ground beef with a porterhouse. :) |
| 12:11.57 | k31th | or two as the case may Be. |
| 12:12.03 | M_at | Buy 3 |
| 12:12.10 | xheliox | If you want a good quality phone, check into Polycom. |
| 12:12.15 | M_at | 2/7 of ours arrived with a dead LCD |
| 12:12.23 | k31th | lol. |
| 12:12.45 | M_at | Also don't expect much in the way of assistance |
| 12:12.51 | k31th | I just want a phone that works well with asterisk for now |
| 12:13.06 | k31th | those linksys ones look like cisco phones. |
| 12:13.07 | M_at | We had to be interviewed by Linksys to make sure we were suitable to purchase them direct rather than from a small reseller |
| 12:13.26 | k31th | do they have the classic cisco ring tone? |
| 12:13.29 | M_at | They LOOK like cisco phones. They are not cisco inside, they're Sipura. |
| 12:13.41 | M_at | No - but they do have the one off 24 that you want. |
| 12:13.50 | k31th | :D you got it |
| 12:14.05 | M_at | Cisco phones do not have that rington as standard |
| 12:14.39 | monsted | the 24 ringtone actually comes from some old Lucent phones, i believe |
| 12:15.44 | k31th | the linksys phones are 10mbp i notice? |
| 12:15.45 | M_at | AT&T Merlin |
| 12:15.47 | backblue | polycoms are great |
| 12:16.19 | M_at | The SPA942 is 10/100 |
| 12:16.33 | k31th | M_at: which phone did you get ?? 941 or 942 |
| 12:16.42 | M_at | 942 for the POE |
| 12:17.34 | k31th | Ah ok, I think im going to grab two 942's apart from the dead lcd issue you had they work ok with asterisk ? (and work in general?_ |
| 12:17.58 | M_at | For the basics - yeah they're not bad handsets. |
| 12:18.49 | k31th | Ok fine thats all i need for now, thank :D |
| 12:18.55 | k31th | thanks* |
| 12:19.45 | k31th | M_at: can I add my own ringers to that phone ? (i.e sound i download) |
| 12:20.12 | *** part/#asterisk bjohnson_ (n=bjohnson@i216-58-63-46.cybersurf.com) |
| 12:21.27 | M_at | Don't think it's necessarily a downloadable thing but you can play with the ones in there |
| 12:21.51 | k31th | just so long the 24 one is deff in there :D |
| 12:29.05 | HarryR | heh i've got a strange problem with sjphone, whenever I accept calls with it, the other end can hear what I'm listening to in winamp |
| 12:35.13 | *** join/#asterisk WildPikachu (n=WildPika@about/linux/staff/wildpikachu) |
| 12:35.17 | *** join/#asterisk dioedu (n=dioedu@201.7.117.114) |
| 12:35.21 | WildPikachu | what would this error mean ... "Unable to find a codec translation path from speex to gsm"? |
| 12:36.42 | DarKnesS_WolF | WildPikachu: which means u don't have speex installed |
| 12:36.48 | WildPikachu | hrmmmm |
| 12:36.51 | DarKnesS_WolF | WildPikachu: codec missmatch |
| 12:37.00 | DarKnesS_WolF | WildPikachu: ur using sip ? |
| 12:37.10 | WildPikachu | yea |
| 12:37.22 | WildPikachu | one sec, i may of forgotten to do a make install after building in speex |
| 12:37.27 | M_at | HarryR - change the windows mixer settings so it records only from the mic and doesn't mix in the output |
| 12:38.02 | *** join/#asterisk coppice (n=chatzill@213.143.17.210.dyn.pacific.net.hk) |
| 12:38.12 | *** join/#asterisk dasenjo (n=dasenjo@208.195.215.168) |
| 12:40.12 | *** join/#asterisk creativx (n=creadure@226.62-97-205.bkkb.no) |
| 12:55.24 | *** join/#asterisk Formater (i=Formater@cable-87-116-139-246.dynamic.sbb.co.yu) |
| 12:55.40 | Formater | hi... question... is there possibility to include username/password in SIP dialstring ? |
| 12:56.02 | *** join/#asterisk spr1te (n=spr1te@213.227.193.75) |
| 12:56.26 | *** join/#asterisk ManxPower (n=manxpowe@71-8-11-73.dhcp.leds.al.charter.com) |
| 13:08.31 | RoyK | what can asterisk do better than openoffice? |
| 13:09.38 | *** join/#asterisk SoftIce (n=newbie@vc-196-207-45-253.3g.vodacom.co.za) |
| 13:09.46 | SoftIce | hi, how safe is the asterisk upgrade? |
| 13:10.16 | SoftIce | I prpesume this asterisk version is pretty unstable |
| 13:10.36 | SoftIce | just wondering, if I just upgrade the asterisk system but copy /etc/asterisk safely elsewhere or dont install the modules |
| 13:10.45 | SoftIce | what else would be affected |
| 13:10.53 | RoyK | SoftIce: upgrading from what to what? |
| 13:11.38 | SoftIce | im using 1.09 now |
| 13:11.40 | SoftIce | but its very unstable |
| 13:11.48 | SoftIce | having to restart it quite often |
| 13:11.49 | *** join/#asterisk zotz (n=zotz@208.196.247.175) |
| 13:12.03 | backblue | hooo 1.09 |
| 13:12.12 | backblue | i tought no one use that |
| 13:12.39 | SoftIce | well meet me ;) |
| 13:13.03 | ManxPower | SoftIce: are you using analog cards? |
| 13:13.25 | ManxPower | SoftIce: 1.2.x has an UPGRADE.txt with info about issues with upgrading. |
| 13:14.58 | SoftIce | no cards |
| 13:15.04 | SoftIce | only zt dummy timing? |
| 13:18.17 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 13:18.36 | ManxPower | SoftIce: I've not seen any issues with that setup. |
| 13:19.00 | SoftIce | ManxPower: about the zt dummy timing |
| 13:19.07 | SoftIce | how would I go abouts putting that back in place |
| 13:19.10 | SoftIce | editing the Makefile? |
| 13:20.17 | ManxPower | SoftIce: Huh? "putting it back in place"? ztdummy is built by default if the build system thinks it can be. |
| 13:20.44 | *** join/#asterisk scanna (n=scannach@81-174-16-211.f5.ngi.it) |
| 13:23.49 | SoftIce | ahh, I thought I had to uncomment it or something to get it to install |
| 13:23.57 | SoftIce | never knew it was put in place automatically |
| 13:24.11 | ManxPower | You might have had to in 1.0.x |
| 13:24.43 | SoftIce | thats why I asked ;) |
| 13:24.50 | SoftIce | I remeber doing something to the makefile |
| 13:24.54 | SoftIce | hmm, what about sounds, etc |
| 13:25.00 | SoftIce | they will all be left as i'm not removin |
| 13:25.02 | SoftIce | im just upgrading |
| 13:25.17 | SoftIce | and just making sure I have /etc/asterisk backed-up? |
| 13:25.23 | SoftIce | hmm, permissions, etc |
| 13:27.47 | Formater | <PROTECTED> |
| 13:27.48 | RoyK | running asterisk as non-root? |
| 13:28.03 | RoyK | Formater: why should you? |
| 13:28.24 | Formater | RoyK: users should be able to use their own termination providers |
| 13:28.34 | Formater | RoyK: they setup it thoru php interface.... |
| 13:28.47 | Formater | RoyK: and astersik will read out user/pass from mysql |
| 13:29.10 | RoyK | why not just use sip realtime and keep it there? |
| 13:29.10 | SoftIce | is this standdard to see |
| 13:29.12 | SoftIce | file.c: Failed to write frame |
| 13:29.17 | SoftIce | all over my logs? |
| 13:29.33 | RoyK | SoftIce: did you remove the old 1.0 libs? |
| 13:29.36 | Formater | RoyK that is a possibility too :) |
| 13:29.48 | RoyK | Formater: sounds cleaner to me |
| 13:29.55 | backblue | anyone knows how to change the behaviour of the leds in a polycom 601? |
| 13:29.57 | SoftIce | RoyK: na, this is another machine |
| 13:30.01 | SoftIce | I just see this all over the show |
| 13:30.05 | SoftIce | this wasn't the box I upgraded |
| 13:30.20 | SoftIce | this was a fresh asterisk install on this box |
| 13:30.27 | Formater | RoyK: tnx:) |
| 13:30.34 | SoftIce | Nov 17 15:27:32 NOTICE[23295] rtp.c: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: 216.168.169.105 |
| 13:30.34 | SoftIce | Nov 17 15:27:33 WARNING[23439] file.c: Failed to write frame |
| 13:30.35 | SoftIce | .... |
| 13:30.55 | ManxPower | SoftIce: Turn off silence supression in your SIP client |
| 13:30.58 | SoftIce | maybe its just saing that due to asterisk not supporint comfort noise? |
| 13:31.16 | SoftIce | sip client? |
| 13:31.21 | SoftIce | this is a hard phone |
| 13:31.26 | SoftIce | thats phoning through |
| 13:31.34 | ManxPower | softphone or hardphoine, the fix is the same. |
| 13:31.53 | SoftIce | sorry, i'm a tad confused ;) turn it off on the server or on the phone ? |
| 13:32.10 | ManxPower | You turn it off on the phone. |
| 13:32.28 | ManxPower | or whatever device is at 216.168.169.105 |
| 13:32.39 | SoftIce | arg, cant do that, I have thousands of people calling through the server |
| 13:33.01 | ManxPower | SoftIce: then you will have audio issues. |
| 13:33.18 | SoftIce | with those peoples that have silent suppresion enabled? |
| 13:33.41 | SoftIce | ManxPower: you preetty smart hey ;) |
| 13:33.43 | ManxPower | Keep in mind that when a device connects Asterisk tells the device that it does not support silence supression or comfort noise. The client you are using is ignoring that. |
| 13:34.24 | SoftIce | that ip may be an asterisk peer |
| 13:34.38 | ManxPower | It's not since Asterisk does not support these features. |
| 13:34.39 | SoftIce | could that be the case? and in its sip conf silence supprision is not disabled? |
| 13:34.57 | ManxPower | Asterisk does NOT support this feature, therefore there is no option for it. |
| 13:35.27 | SoftIce | ahh, so it must be 1 of the clients phones then |
| 13:35.34 | SoftIce | seeing that asterisk doesn't have the support it cant be a peer |
| 13:35.39 | ManxPower | That's what I said. |
| 13:35.49 | SoftIce | but sheesh, 216 that is a UK ip range |
| 13:36.26 | SoftIce | ManxPower: could this issue come up if say, I was phoning somebody, not the person phoning but the person answering |
| 13:36.36 | SoftIce | could it also be the case? |
| 13:36.40 | ManxPower | Here is what happens. Asterisk times it's outgoing audio from the incoming audio. If there is no incoming audio (i.e. device is not sending audio because of silence supression) then there will be no outgoing audio. |
| 13:37.06 | ManxPower | SoftIce: The device that is trying to force silence supression is at that IP. |
| 13:37.16 | ManxPower | We can't know anything else from that message. |
| 13:37.23 | SoftIce | *smile * see |
| 13:39.15 | *** join/#asterisk pepepedo (i=ircap@243-150-114-200.fibertel.com.ar) |
| 13:41.04 | pepepedo | hi .. i have a question , i m tring to complie h323 channel on asterisk 1.4 but i like the make ignore the h323 channel , I did the smae steps before with asterisk 1.2 and work , but i dont know what is happening in 1.4 mayl someone help me pls? |
| 13:41.41 | *** join/#asterisk |dennis| (n=dennis@vsat-148-64-30-39.c050.t7.mrt.starband.net) |
| 13:42.10 | pepepedo | hi .. i have a question , i m tring to complie h323 channel on asterisk 1.4 but when i try , the "make ignore the h323 channel , I did the smae steps before with asterisk 1.2 and work , but i dont know what is happening in 1.4 mayl someone help me pls? |
| 13:42.19 | creativx | please |
| 13:42.22 | creativx | ask the question again |
| 13:42.26 | creativx | i did not get it the first time |
| 13:42.35 | pepepedo | i cannot compile h323 in asterisk 1.4 |
| 13:42.55 | pepepedo | I did this before with 1.2 and works fine |
| 13:43.19 | pepepedo | but in 1.4 is like make ignore the h323 channel driver |
| 13:43.30 | pepepedo | no errors... |
| 13:43.33 | ManxPower | pepepedo: report it as a bug. 1.4 is still beta |
| 13:43.37 | pepepedo | just nothing |
| 13:43.50 | pepepedo | ok .. |
| 13:44.15 | pepepedo | i will report it thxs! |
| 13:45.11 | SoftIce | ManxPower: what version would you advise I upgrade to ? |
| 13:47.46 | *** join/#asterisk oej (n=olle@213.115.215.55) |
| 13:48.03 | ManxPower | SoftIce: 1.2.13 |
| 13:48.40 | SoftIce | ManxPower: thanks, and should I fear the upgrade? |
| 13:48.53 | SoftIce | or will it pretty much be safe is I just ./configure && make && make install |
| 13:49.13 | SoftIce | or do you invision me having to do more steps, or removing things prior, etc? |
| 13:49.22 | SoftIce | or is that covered in the update documentation? |
| 13:51.06 | ManxPower | read UPGRADE.txt to start with |
| 13:51.13 | *** join/#asterisk loxi (n=zeke@LSt-Amand-152-32-34-121.w82-127.abo.wanadoo.fr) |
| 13:52.47 | SoftIce | k, thanks alot for all your help1 |
| 13:52.58 | SoftIce | i'm not afraid to read, so point me to all the documentation ;) |
| 13:53.11 | creativx | may I point you at google? |
| 13:53.17 | creativx | just to see how much you want to read |
| 13:53.52 | SoftIce | haha, na, done that, waisted alot of time reading the wrong things, then again it wasnt waisted, just normally get diverted off to something else |
| 13:54.38 | creativx | hehe know how that goes yea |
| 13:55.44 | SoftIce | also, some of my search strings are not th ebest |
| 13:55.54 | SoftIce | that is what frustrates me the most. |
| 13:58.30 | creativx | you can probably google that too |
| 13:58.35 | *** join/#asterisk shellshark (n=x86@get.hooked.on.voip.with.shellshark.net) |
| 13:58.47 | *** join/#asterisk x6541a (n=SimmerMo@www3.datarack.nl) |
| 14:00.25 | x6541a | greetings; anybody using cisco voicerouters with asterisk and experiencing a lot of "all-circuits-are-busy" messages? |
| 14:01.14 | SoftIce | creativx haha true ;) |
| 14:02.18 | *** join/#asterisk key2 (n=key2@251.9.39-62.rev.gaoland.net) |
| 14:04.05 | ManxPower | x6541a: We don't support trixbox/freepbx/amp here. |
| 14:04.33 | M_at | exit |
| 14:04.35 | M_at | damn |
| 14:04.49 | x6541a | manxpower: I don't use any of those. What gave you that idea? |
| 14:05.23 | ManxPower | x6541a: Asterisk will not play an all-circuits-are-busy message unless you configure it to do so. |
| 14:05.45 | x6541a | hmm |
| 14:05.49 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 14:06.23 | ManxPower | http://www.theregister.co.uk/2006/11/17/kidnap_attempt/ |
| 14:08.18 | x6541a | Problem is, I manage 3 pbx's with exactly the same os and asterisk versions and 3 (different types of) cisco voice routers |
| 14:08.29 | *** join/#asterisk DarKnesS_WolF (n=wolf@196.218.72.54) |
| 14:08.53 | ManxPower | You really need to see the value of HANGUPCAUSE |
| 14:08.53 | *** join/#asterisk alerios (n=alerios@190.24.100.110) |
| 14:09.20 | x6541a | Only one of the installs has these problems, so I thought my router config wasn't ok |
| 14:09.26 | x6541a | Ok, will check |
| 14:10.06 | x6541a | The customer uses a commercial GUI (also same version in 3 locations), they might set it. |
| 14:11.44 | creativx | All your circuits are belong to us. |
| 14:11.52 | *** join/#asterisk af_ (n=af@ip-130-179.sn2.eutelia.it) |
| 14:12.10 | x6541a | lol @ creativx |
| 14:12.38 | x6541a | well, I might LOL at that when I solved it. Customer isn't too pleased. |
| 14:13.02 | creativx | heehe, its friday man |
| 14:13.05 | creativx | i can't help it :> |
| 14:13.18 | De_Mon | will app_chanspy from asterisk 1.4 work in 1.2.13? I really want that whisper mode... |
| 14:13.21 | RoyK | De_Mon: you'll need to backport it. |
| 14:13.37 | RoyK | if you do, please upload the code on asterisk-backports.org |
| 14:14.12 | De_Mon | how difficult do you think it'll be to backport? |
| 14:16.46 | x6541a | Is there anybody using a Cisco 2801 router to connect to the PSTN willing to share the config? I'd love to verify mine... |
| 14:17.03 | *** join/#asterisk davidredekop (n=davidred@209.217.111.244) |
| 14:18.39 | davidredekop | has anyone here ever interfaced Digium hardware (or other FXS/FXO interfaces) in a VMWare Virtual Machine? |
| 14:19.47 | davidredekop | Virtual Machine runs latest version of trixbox (asterisk on CentOS) |
| 14:21.47 | *** part/#asterisk ttyp0 (n=ivan@165.Red-81-39-143.dynamicIP.rima-tde.net) |
| 14:23.01 | *** join/#asterisk svenna_ (n=svenna@p548D3BEB.dip0.t-ipconnect.de) |
| 14:23.32 | *** join/#asterisk c0w (n=sean@212.9.103.220) |
| 14:23.35 | c0w | hello all. |
| 14:24.04 | c0w | anyone know why Set(CALLERID(num)=11212) doesn't set the callerid if there is one already presenet? |
| 14:24.33 | *** join/#asterisk P4C0 (n=ash@200.124.22.34) |
| 14:25.34 | P4C0 | hello guys, I need to translate g729, to a-law, u-law or gsm, which on is the less cpu consuming? |
| 14:26.31 | De_Mon | gsm |
| 14:27.04 | Qwell | from g729? I'm pretty sure it'd be ulaw/alaw |
| 14:27.52 | Qwell | P4C0: Do a "show translation", and look at the time between slin and ulaw, alaw, and gsm |
| 14:28.12 | Qwell | slin>gsm is clearly higher |
| 14:29.05 | *** join/#asterisk juanmanuel (n=jmacz@190.24.100.110) |
| 14:29.09 | De_Mon | Qwell is right |
| 14:29.32 | P4C0 | thanks |
| 14:31.55 | *** join/#asterisk spr1te (n=spr1te@213.227.193.75) |
| 14:32.47 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 14:32.47 | *** mode/#asterisk [+o anthm] by ChanServ |
| 14:33.43 | *** join/#asterisk trelane_ (n=trelane@unaffiliated/trelane) |
| 14:34.51 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 14:35.38 | puzzled | afternoon |
| 14:40.12 | WildPikachu | ok ... one more problem :0 |
| 14:40.14 | WildPikachu | "codec_speex.c:278 speextolin_framein: Out of buffer space" |
| 14:40.47 | *** join/#asterisk apardo (n=apardo@87.217.146.11) |
| 14:41.16 | *** join/#asterisk supjigatr (n=syslod@152.53.16.10) |
| 14:41.40 | supjigatr | Anyone seen polycoms that just reboot during the day? Even during a call? |
| 14:42.30 | x6541a | supjigatr: yes |
| 14:42.44 | x6541a | it is on my to do list, haven't gotten there yet |
| 14:42.56 | x6541a | version 1.6.7, polycom sounpoint 600's |
| 14:43.09 | x6541a | asterisk 1.2.12.1 |
| 14:43.24 | WildPikachu | any idea bout my speex problem? :( |
| 14:43.33 | supjigatr | I have some 2.0.1 that also do it. Its really annoying. |
| 14:43.49 | supjigatr | I can't seem to see a reason why they would reboot. |
| 14:44.10 | x6541a | What is really annoying is that my customer didn't tell me, but they suffer (kind of) in silence :) |
| 14:44.28 | supjigatr | Yea. Same thing here. |
| 14:44.33 | x6541a | You also have 600's? |
| 14:44.35 | supjigatr | They've had the problem for 5 months. |
| 14:44.41 | x6541a | Hmm |
| 14:44.45 | supjigatr | 600 or 500 or even 601 same problem. |
| 14:44.48 | x6541a | what is the daily call volume? |
| 14:45.04 | x6541a | My customer does 1100 calls a day |
| 14:45.11 | supjigatr | Well I have a few that have high like 3000 calls down to 12 calls. |
| 14:45.14 | supjigatr | Still does it. |
| 14:45.17 | x6541a | Haven't gotten round to ask how often they reset. |
| 14:45.37 | supjigatr | Once or twice a day is what I hear. some don't do it every day. |
| 14:46.20 | *** join/#asterisk vt (n=vt@ns1.axiris.net) |
| 14:47.16 | _DAW | Good morning everyone. Can someone tell me, does 1.4 have much better performance than 1.2 on the same hardware? |
| 14:47.19 | *** join/#asterisk moon06 (n=michael@cim06-1-82-228-240-97.fbx.proxad.net) |
| 14:49.04 | *** join/#asterisk dsfr (i=dsfr@pdpc/sponsor/digium/dsfr) |
| 14:52.30 | jserve | *hmms* Anyone knows about a SIP Softphone based on Flash? |
| 14:52.47 | shellshark | that'd be cool |
| 14:53.22 | shellshark | there's a Java softphone based on the JAIN project, but i've not been able to get it running |
| 14:53.50 | *** join/#asterisk wpretoria (n=werduspr@wbs-196-2-105-75.wbs.co.za) |
| 14:55.11 | wpretoria | I just joined the channel. I'm also interested in the answers to the question from davidredekop |
| 14:56.32 | jserve | shellshark: I have personelly test jiax, and this has work fine for us. But I also look for other webbased Softphones. And I think that Flash would be also nice since it seems to be more 'accepted' at the End-User site. |
| 14:56.35 | *** join/#asterisk nicolajh (n=chatzill@75.125.78.83.cust.bluewin.ch) |
| 14:57.37 | shellshark | jiax is open source? |
| 14:58.40 | jserve | shellshark: GPLed, Link to jiax is http://www.hem.za.org/jiaxclient/ |
| 15:02.45 | shellshark | doesn't seem to work on my Mac (G5) |
| 15:02.49 | shellshark | :( |
| 15:03.38 | tzanger | wtf |
| 15:04.03 | tzanger | on windows, open up notepad. type "Bush hid the facts" and save it as any iflename on your desktop. lcose out of notepad, then open that file. |
| 15:04.03 | RoyK | how can I check the channel type, sip, zap, whatever, of the current channel from the dialplan? |
| 15:04.57 | shellshark | tzanger: what happens? |
| 15:05.03 | shellshark | tzanger: (i'm on a mac) |
| 15:05.42 | M_at | # Created by hand |
| 15:05.42 | M_at | <PROTECTED> |
| 15:05.42 | M_at | ### Global Config Options |
| 15:05.42 | M_at | <PROTECTED> |
| 15:05.42 | M_at | # for UNIX |
| 15:05.44 | M_at | # WorkDir: /home/http/mrtg |
| 15:05.46 | M_at | <PROTECTED> |
| 15:05.48 | M_at | # or for NT |
| 15:05.50 | M_at | # WorkDir: c:\mrtgdata |
| 15:05.52 | M_at | <PROTECTED> |
| 15:05.54 | M_at | ### Global Defaults |
| 15:05.56 | M_at | <PROTECTED> |
| 15:05.58 | M_at | # to get bits instead of bytes and graphs growing to the right |
| 15:06.00 | M_at | # Options[_]: growright, bits |
| 15:06.02 | M_at | <PROTECTED> |
| 15:06.04 | M_at | EnableIPv6: no |
| 15:06.06 | M_at | WorkDir: /var/www/html/mrtg |
| 15:06.08 | M_at | Options[_]: bits,growright |
| 15:06.10 | M_at | <PROTECTED> |
| 15:06.14 | x6541a | M_at use www.pastebin.ca, please |
| 15:06.15 | tzanger | shellshark: it seems to change the font or something |
| 15:06.16 | M_at | <PROTECTED> |
| 15:06.18 | M_at | Title[srv-uk1-voip1]: srv-uk1-voip1 |
| 15:06.20 | M_at | PageTop[srv-uk1-voip1]: <h1>srv-uk1-voip1.uk1.lrfairplay.com</h1> |
| 15:06.22 | M_at | Target[srv-uk1-voip1]: `/usr/local/bin/asterisk-mrtg -h srv-uk1-voip1.uk1.lrfairplay.com -1 SIP -2 IAX -v -u admin -p amp111` |
| 15:06.25 | M_at | Options[srv-uk1-voip1]: gauge,integer |
| 15:06.27 | M_at | MaxBytes[srv-uk1-voip1]: 90 |
| 15:06.29 | M_at | YLegend[srv-uk1-voip1]: Active channels |
| 15:06.31 | M_at | pants |
| 15:06.35 | M_at | I would if I were trying to tell you something and not trying to paste something into putty |
| 15:07.12 | x6541a | :) |
| 15:08.09 | RoyK | ~pb |
| 15:08.14 | jbot | somebody said pb was a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 15:08.33 | M_at | ~how do I tell people it was an accident? |
| 15:08.42 | x6541a | I think it was a slip of the finger guys |
| 15:09.38 | x6541a | Anybody using a Cisco 2801 to connect * to PSTN? |
| 15:09.58 | *** join/#asterisk hohum (n=dcorbe@69-175-203-11.chvlva.adelphia.net) |
| 15:10.51 | M_at | x6541a: Ask me that in 4 weeks time |
| 15:11.03 | *** join/#asterisk intralanman (n=lanman@pool-71-253-247-137.nrflva.east.verizon.net) |
| 15:11.45 | *** join/#asterisk FWP (n=FWP@unaffiliated/fwp) |
| 15:14.14 | x6541a | M_at you're setting up such a config? |
| 15:14.32 | M_at | Not 100% myself - Telco will be installing one and providing PRI out |
| 15:14.44 | M_at | But I intend on getting a copy of the config from them |
| 15:14.47 | x6541a | would you have access to the config |
| 15:14.48 | x6541a | ah |
| 15:15.06 | M_at | I have similar kit at home I want to play with :) |
| 15:15.28 | x6541a | The problem I have is that when users dial a non-existent number, asterisk plays "all-circuits-busy-now". |
| 15:16.06 | M_at | You users are on the * box? |
| 15:16.23 | M_at | And the non existent number is out on the general telephony network? |
| 15:16.25 | x6541a | While another office with the same * and OS works fine (that is, the provider says "this number is not in use" |
| 15:16.29 | x6541a | yes |
| 15:16.38 | M_at | Same telco at both sites? |
| 15:16.43 | x6541a | the 2801 gives a SIP 404 |
| 15:16.52 | x6541a | to asterisk, so * plays that file |
| 15:16.55 | x6541a | yes |
| 15:17.14 | M_at | But at the other site invalid numbers are treated differently by the provider? |
| 15:17.31 | x6541a | the only difference between the offices is the voice router (different cisco's) |
| 15:17.54 | M_at | And presumably different configurations, possibly different endpoints within the telco |
| 15:18.46 | x6541a | both ISDN PRI, but indeed, the configs differ. The dial-peers are the same apart from the numbers |
| 15:19.13 | x6541a | I've been troubelshooting this all week, drives me crazy |
| 15:19.51 | M_at | Is it a problem or just a nice to have - could you not get a recording of the provider's "this number is not in use" and substitute it? |
| 15:20.11 | *** join/#asterisk chino (n=daquino@c-68-84-57-212.hsd1.nj.comcast.net) |
| 15:20.30 | chino | any free clients support confrencing ? |
| 15:23.43 | x6541a | It can be classified a problem according to the customer. I also want to know why * is playing this rather general message instead of connecting to the PSTN anyway (and let the provider play their message). |
| 15:24.11 | *** join/#asterisk mercestes (n=merceste@rrcs-71-41-157-70.sw.biz.rr.com) |
| 15:26.26 | *** part/#asterisk chino (n=daquino@c-68-84-57-212.hsd1.nj.comcast.net) |
| 15:30.14 | *** join/#asterisk mega (i=mega@gateway/tor/x-850e395945f418ec) |
| 15:32.09 | *** join/#asterisk DaveCanoe (n=Dave@bas17-toronto12-1096795308.dsl.bell.ca) |
| 15:32.56 | *** join/#asterisk ast_freak (n=jesse@h69-130-171-28.69-130.unk.tds.net) |
| 15:37.19 | *** join/#asterisk Egonis (n=Egonis@CPE00023fd3b353-CM00159a3b77be.cpe.net.cable.rogers.com) |
| 15:38.45 | *** join/#asterisk shido6 (i=shido6@d221-68-200.commercial.cgocable.net) |
| 15:38.54 | *** join/#asterisk loxi (n=loxi@LSt-Amand-152-32-34-121.w82-127.abo.wanadoo.fr) |
| 15:40.39 | sivana | Will this actually increment the variable or append to it? Set(TRYPORT=${TRYPORT}+ 1) |
| 15:41.11 | Qwell[] | append |
| 15:41.23 | Qwell[] | Set(TRYPORT=$[${TRYPORT} + 1]) |
| 15:41.45 | *** join/#asterisk marv[work] (n=timr@br0.asteriasgi.com) |
| 15:41.51 | sivana | cool, thank you |
| 15:43.35 | *** join/#asterisk VibroMax (n=phisto@83.229.70.154) |
| 15:46.42 | *** join/#asterisk nicolajh (n=chatzill@75.125.78.83.cust.bluewin.ch) |
| 15:47.54 | nicolajh | How can i connect my asterisk server to my SIP-Provider? |
| 15:52.38 | |Johny| | define a trunk (SIP or IAX) |
| 15:53.27 | Qwell[] | ~trunk |
| 15:53.28 | jbot | [trunk] my trunk my trunk; my lovely asterisk trunk (check it out) |
| 15:54.38 | nicolajh | |Johny|: I want to register to the SIP-Provider over my SIP-account (the same account i use in my softphone x-lite). |
| 15:56.23 | *** join/#asterisk SplasPood (n=jwb@gate.lga2.us.voxel.net) |
| 16:00.16 | *** join/#asterisk w9sh (n=w9sh@adsl-068-209-117-205.sip.asm.bellsouth.net) |
| 16:07.31 | WildPikachu | "codec_speex.c:278 speextolin_framein: Out of buffer space" <= anyone seen this before |
| 16:09.56 | *** join/#asterisk tsurk0 (n=tsurko@vpn-pppoe-213-240-242-237.megalan.bg) |
| 16:12.25 | *** join/#asterisk db1310 (n=bandit@216.77.58.40) |
| 16:13.38 | db1310 | newbie, need to signal other computer on lan of number dialed, can this be done with the scripting? |
| 16:20.48 | *** join/#asterisk sb_mx (n=sb_mx@200.78.229.18) |
| 16:24.23 | *** join/#asterisk c0w (n=sean@212.9.103.220) |
| 16:24.26 | c0w | lo again all. |
| 16:24.38 | c0w | has anyone had any problems with h323 and oneway voice? |
| 16:27.18 | *** join/#asterisk Ox0F0-0FF (n=pierre@201.38.238.66) |
| 16:28.28 | mercestes | c0w: I've never heard of anyone having problems wiht one way voice ever in VoIP. |
| 16:28.49 | mercestes | c0w: Theoretically though....improper nat translations could cause it given the proper circumstances. |
| 16:29.09 | *** part/#asterisk Ox0F0-0FF (n=pierre@201.38.238.66) |
| 16:30.47 | *** join/#asterisk armiku (n=j@nathan.epi.usf.edu) |
| 16:30.55 | armiku | hi |
| 16:31.02 | *** join/#asterisk voicetech (n=lotusscr@marnock.com) |
| 16:31.21 | armiku | i just dl the asterisk gui from svn and tried to compile it |
| 16:31.44 | armiku | i get an error about zaaptel but i don't have any zaptel hardware |
| 16:33.36 | db1310 | armiku: AFAIK, you need to download the zaptel source and build it anyway, it creates a ztdummy module i think and that has to be loaded for timing reasons. Read the AsteriskTFOT pdf available on their site. I just started that myself |
| 16:33.53 | voicetech | I have a problem with uk caller id with digium board. I get a 50% hit. Seems to ignore parity. ne1 help please? |
| 16:33.56 | *** join/#asterisk ucfMethod (i=ucfmetho@c2.efb7d1.client.atlantech.net) |
| 16:34.57 | *** join/#asterisk henrique (n=henrique@200-168-129-202.dsl.telesp.net.br) |
| 16:35.44 | armiku | i though 1.4 did not need ztdummy |
| 16:37.26 | *** join/#asterisk ToTo (n=ToTo@host97-157-dynamic.2-87-r.retail.telecomitalia.it) |
| 16:37.45 | *** join/#asterisk sloth101 (n=mallen@user-216-10-84-189.citynetwireless.net) |
| 16:37.53 | db1310 | armiku: i think it has too, but youll have to read, im not sure |
| 16:38.00 | *** join/#asterisk tamp4x (n=tampon@65-115-242-97.dia.static.qwest.net) |
| 16:38.14 | *** join/#asterisk _alex_mx_ (n=alex@200.78.229.18) |
| 16:38.58 | tamp4x | i have a meetme extension set up but when i use it i get this error: 2006-11-17 11:31:45 ERROR[18454]: chan_zap.c:7396 chandup: Unable to dup channel: No such file or directory |
| 16:39.07 | tamp4x | anyone know why tha tis occurring? |
| 16:39.58 | *** join/#asterisk saftsack (n=saftsack@pD9E07B76.dip.t-dialin.net) |
| 16:40.22 | mercestes | tamp4x: check /dev/zap permissions. |
| 16:41.03 | *** join/#asterisk ltd (n=z@202-161-28-21.dyn.iinet.net.au) |
| 16:42.32 | tamp4x | there is no /dev/zap |
| 16:43.22 | mercestes | tamp4x: ztdummy installed? |
| 16:43.23 | *** join/#asterisk giesen (i=giesen@dirtypackets.net) |
| 16:43.50 | giesen | how do I run a simple shell script from the dialplan, get a true or false answer back from it |
| 16:44.08 | giesen | I've got a simple script that checks whether today is easter |
| 16:44.19 | giesen | and returns a 1 or a 0 to stdout |
| 16:44.24 | giesen | depending on the date |
| 16:44.42 | mercestes | giesen: umm.....phpagi perhaps? |
| 16:44.55 | mercestes | geisen: Why not use a gotoiftime??? |
| 16:44.56 | giesen | agi looks a little heavy, I was hoping for something smaller |
| 16:45.09 | giesen | mercestes: because easter doesnt fall on a set date |
| 16:45.16 | mercestes | ... |
| 16:45.19 | giesen | I'm using gotoiftime for everything else |
| 16:45.23 | *** join/#asterisk LoneShadow (n=duh@c-67-188-235-220.hsd1.ca.comcast.net) |
| 16:45.29 | *** join/#asterisk alamantia (i=alamanti@nat/digium/x-11e596401df2658f) |
| 16:45.35 | mercestes | hire some kid in Indonesia part time. "Hey kid, on easter...do this. I'll pay you a quarter." |
| 16:45.49 | giesen | yeah that's not quite what I was looking for |
| 16:45.51 | giesen | =) |
| 16:45.58 | mercestes | giesen: Isn't there a "week of month" in gotoiftime somewhere? |
| 16:46.11 | giesen | uh easter can fall in two different months |
| 16:46.15 | mercestes | .... |
| 16:46.21 | *** join/#asterisk droops (n=droops@adsl-074-245-001-031.sip.jan.bellsouth.net) |
| 16:46.22 | giesen | it's based on the last full moon of march |
| 16:46.23 | giesen | iirc |
| 16:46.24 | mercestes | I always thought Easter was in April. |
| 16:46.27 | giesen | nope |
| 16:46.30 | giesen | it can be in march too |
| 16:46.33 | mercestes | *thinks* |
| 16:46.35 | giesen | it's based on lunar rotations |
| 16:46.39 | mercestes | That can't be. |
| 16:46.47 | giesen | yes, it is |
| 16:46.49 | giesen | google it =) |
| 16:46.51 | mercestes | Christian's don't base their holidays on lunar rotations. It's too wiccan for them. |
| 16:46.53 | *** join/#asterisk yogurt2ungue (n=yogurt2u@host96.200-117-56.telecom.net.ar) |
| 16:46.55 | mercestes | and lunar is the devil. |
| 16:47.08 | mercestes | heresy I tell you! |
| 16:47.12 | mercestes | anyways.... |
| 16:47.27 | armiku | db1310 i am getting an error if i try to install ztdummy if i compile zaptel with the ztdummy commented out the compilation goes thorugh fine |
| 16:47.28 | mercestes | If you are looking for a gotoifmooncycle() command you'll have to ask for it as a feature. |
| 16:47.37 | giesen | haha |
| 16:47.39 | giesen | all I want |
| 16:47.41 | mercestes | otherwise I would suggest the AGI of your choice. |
| 16:47.42 | giesen | is to run an app |
| 16:47.45 | giesen | and check the output :/ |
| 16:47.52 | mercestes | system() ? |
| 16:47.59 | giesen | yeah I was thinking of system |
| 16:48.11 | giesen | but I cant really see a way to capture the output |
| 16:48.52 | mercestes | Agi(findapril.php) findapril.php $result = System(getapril.sh); |
| 16:49.14 | *** join/#asterisk spr1te (n=spr1te@213.227.193.75) |
| 16:49.19 | mercestes | dialplan logic isn't extremely hefty, so I don't htink you can assign a systme to a varible like var=system(blah). |
| 16:49.19 | giesen | $result = |
| 16:49.30 | giesen | yeah |
| 16:49.34 | giesen | I didnt think so either |
| 16:49.43 | mercestes | but you can assign vars in Agi. |
| 16:49.50 | mercestes | which is what it's for....to do complex programming. |
| 16:50.05 | giesen | alright, I'll go agi then |
| 16:50.06 | mercestes | you can even call bash in agi as I recall. |
| 16:50.36 | mercestes | you'd just echo a setvar(var)=myresult or something;. |
| 16:50.56 | tamp4x | mercestes t dummy is install |
| 16:50.58 | tamp4x | ed |
| 16:51.14 | mercestes | tamp4x: good! :) |
| 16:51.18 | mercestes | is it loaded as a module? |
| 16:56.40 | *** join/#asterisk Ebola (n=Ebola@host81-152-204-133.range81-152.btcentralplus.com) |
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| 17:00.12 | *** join/#asterisk Xen^ (n=linux@unaffiliated/lnux/x-10290) |
| 17:01.59 | *** join/#asterisk HumpBack (n=sdcdcssc@gentoo/developer/HumpBack) |
| 17:02.03 | HumpBack | Hello |
| 17:02.16 | *** join/#asterisk zmef420 (n=zmef420@metarb3-pool3-10.mtco.com) |
| 17:02.57 | HumpBack | My asterisk server is located in a place where the network line has alot of jitter. Is it possible do dial digium support witought using IAX but a direct sip call (i'm using ekiga) |
| 17:03.26 | Qwell[] | HumpBack: should be possible |
| 17:03.35 | Qwell[] | misery.digium.com, username guest |
| 17:03.50 | Qwell[] | (or, with SIP, I guess there wouldn't be a username...or something) |
| 17:04.25 | *** join/#asterisk CunningPike (n=CunningP@dhcp-10-234.district.north-van.bc.ca) |
| 17:04.26 | *** join/#asterisk mega (i=mega@gateway/tor/x-f988d7537199a1f6) |
| 17:05.20 | *** join/#asterisk wunderkin (i=kev@ip72-208-3-42.ph.ph.cox.net) |
| 17:07.12 | HumpBack | Qwell[]: no, it seems it is only for IAX |
| 17:09.49 | *** join/#asterisk lancee (n=aa@58.77.68.42) |
| 17:09.56 | lancee | hi guys |
| 17:09.58 | mercestes | So what happened to the * channel? It used to be so busy? Did everyone migrate to a new channel that I didn't hear about or did we just get quiet? |
| 17:10.48 | lancee | anyone know what the problem of the svn: REPORT of '/svn/asterisk/!svn/vcc/default': 400 Bad Request (http://svn.digium.com) |
| 17:10.49 | lancee | ? |
| 17:11.17 | voicetech | any caller id experts here? |
| 17:12.40 | c0w | cheers lads i fixed the issues. |
| 17:13.05 | *** join/#asterisk DaveCanoe (n=Dave@bas17-toronto12-1096795308.dsl.bell.ca) |
| 17:13.50 | marbahlarbs | what's a good sip softphone package? |
| 17:14.36 | Xen^ | xten |
| 17:14.38 | Xen^ | coutnerpath |
| 17:16.03 | voicetech | can you tell me why I get this error every other call |
| 17:16.06 | voicetech | chan_zap.c: CID timed out waiting for ring. Exiting simple switch |
| 17:16.45 | *** join/#asterisk mpruett (n=mpruett@24-240-203-83.static.stls.mo.charter.com) |
| 17:21.35 | *** join/#asterisk rene1 (n=rene1@189.140.242.8) |
| 17:24.06 | mercestes | Why do my Iaxchanenls go "Unreachable" Time: 2 and then a second later go reachable again on a LAN? |
| 17:24.13 | mercestes | I've seen that error before but....how do I fix that? |
| 17:25.45 | rene1 | i am seeing queues sending more than one call to a local channel sip peer |
| 17:25.59 | *** join/#asterisk scurb (n=scurb@gprs2.vodafone.se) |
| 17:26.02 | CunningPike | mercestes: Fridays are often quiet for some reason - it's been busy all week..... |
| 17:26.13 | rene1 | has anyone seen that |
| 17:26.14 | rene1 | ? |
| 17:26.24 | HarryR | hmm, is AgentCallbackLogin stable? |
| 17:26.25 | rene1 | i wonder if this has anything to do with http://bugs.digium.com/view.php?id=7433&nbn=42 |
| 17:26.34 | mercestes | Meh, I've noticed * being particularly quiet in comparison. Maybe it's jsut me tho |
| 17:26.41 | HarryR | and can you transfer a call which's been directed to an agent? |
| 17:26.49 | CunningPike | HarryR: I think it's being deprecated at some point |
| 17:26.57 | HarryR | and being replaced with what? |
| 17:29.29 | CunningPike | HarryR: Lemme check |
| 17:30.36 | *** join/#asterisk tr2x (n=alvar@80-218-150-90.dclient.hispeed.ch) |
| 17:32.00 | CunningPike | HarryR: http://voip-info.org/wiki/index.php?page=Asterisk+cmd+AgentCallbackLogin. I think it will be deprecated in favor of a) connected agents and b) dynamic members |
| 17:32.15 | CunningPike | HarryR: We use dynamic members and like that |
| 17:32.53 | mercestes | Well I made my iax unreachable debauchery go away with a qualify=no. but...how do I configure Iax to not randomly go "unreachable" and then go reachable again like one second later with qualify=yes |
| 17:33.56 | *** join/#asterisk klasstek (n=nunyobiz@ip67-154-143-190.z143-154-67.customer.algx.net) |
| 17:34.04 | CunningPike | mercestes: Try a numeric value for your qualify - maybe you're on the verge of the default setting |
| 17:34.13 | mercestes | It's on a LAN...and...it's not really "unreachable." and it seems to be a timed event like it fails it's first notify and hten passes it's second one. ARP tables maybe?? |
| 17:34.24 | mercestes | CunningPike: I had 100. Maybe I should go to...50 |
| 17:34.25 | mercestes | ? |
| 17:34.39 | axisys | is there a howto on installing asterisk+freepbx on ubuntu? |
| 17:35.36 | mercestes | I think it has something to do with the 1-3 mil ping time involved in the Network and * expecting a similar result on it's next qualify and it freaking out if it takes more than 1 or 2 ms. |
| 17:36.00 | CunningPike | mercestes: 100 sounds really low - I think it's a millisecond value - the default is 1000, iirc |
| 17:36.12 | mercestes | CunningPike: was still doing it on 100 tho. |
| 17:36.28 | mercestes | CunningPike: And it is a millisecond value..:) |
| 17:37.23 | mercestes | did it again at qualify=50. |
| 17:37.24 | mercestes | :/ |
| 17:37.48 | mercestes | exactly 10 seconds from "Unreachable" to "reachable" according to the timestamps. |
| 17:37.50 | CunningPike | mercestes: Our average (for SIP) on a 100MB LAN is in the 75 - 100 ms range - we have qualify set to 1000ms |
| 17:38.08 | mercestes | I'm showing "3" |
| 17:38.14 | CunningPike | mercestes: Oh |
| 17:38.37 | CunningPike | mercestes: Try qualify=1000 |
| 17:38.51 | mercestes | Roger. |
| 17:38.52 | *** join/#asterisk ToyMan (n=stuq@72.168.167.241) |
| 17:38.56 | *** join/#asterisk conico (n=chatzill@85.103.233.35) |
| 17:41.06 | CunningPike | db1310 needs a lobotomy |
| 17:41.49 | mercestes | db1310 needs a lobotomy like a bicycle needs an engine overhaul. |
| 17:42.58 | mercestes | *scratches head* what is the difference between qualify=yes and qualify=1000? |
| 17:43.11 | CunningPike | mercestes: None, afaik |
| 17:43.29 | mercestes | gah, then why is qualify=1000 not producing that .....issue, as far as I can tell? |
| 17:43.34 | *** join/#asterisk ToyMan (n=stuq@72.168.167.241) |
| 17:44.44 | tamp4x | anyone know what to do in the case of this error (startign zaptel) Waiting for zap to come online...Error: missing /dev/zap! |
| 17:44.45 | CunningPike | mercestes: When qualify=yes does? |
| 17:45.11 | mercestes | Looks like it. |
| 17:45.12 | CunningPike | tamp4x: Your zaptel framework isn't loading |
| 17:45.12 | CunningPike | tamp4x: Lots of reasons for that |
| 17:45.13 | CunningPike | mercestes: Hmmmm |
| 17:45.13 | mercestes | ah crap. It's still doing it. |
| 17:45.14 | mercestes | just....on the other server. |
| 17:45.15 | mercestes | :/ |
| 17:45.24 | mercestes | why would it do it on one server and not the other?? |
| 17:45.40 | CunningPike | mercestes: You're not NATing are you......? |
| 17:45.59 | *** join/#asterisk ccesario (n=ccesario@matriz.isic.com.br) |
| 17:46.13 | mercestes | No, all internal IPs. |
| 17:46.29 | mercestes | heh..now it's doing it on the original box again. |
| 17:46.36 | mercestes | just by doing an iax2 reload. |
| 17:46.51 | mercestes | It is a connection between a 1.0.9 server and a 1.2.13 server. I wonder if there are some weird compatibility issues there. |
| 17:47.06 | CunningPike | mercestes: That's a distinct possibility |
| 17:47.22 | mercestes | I'm rusty on sipconnects too. |
| 17:47.27 | mercestes | lemme try it. |
| 17:49.50 | ccesario | somebody have experiency with planet vip-400 and asterisk ? |
| 17:50.12 | dasenjo | mercestes, *1.0 and 1.2 have different jitterbuffer implementations |
| 17:50.27 | dasenjo | and are not compatible to have IAX2 trunks |
| 17:51.24 | dasenjo | I have had many problems with this types of trunks, even server crashes |
| 17:51.42 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 17:51.52 | dasenjo | I recommend you use the same asterisk branch |
| 17:53.28 | *** join/#asterisk Tili (n=tili@202.133.65.83) |
| 17:53.50 | *** join/#asterisk juanmanuel (n=jmacz@190.24.100.110) |
| 17:55.36 | *** join/#asterisk hoobastooba (n=ckwall@63.149.122.93) |
| 17:55.37 | mercestes | dasenjo: Many thanks. |
| 17:56.32 | *** join/#asterisk drcode (n=chatzill@87.69.59.186.cable.012.net.il) |
| 17:56.36 | drcode | hi all |
| 17:56.49 | *** join/#asterisk EmleyMoor (i=ejabberd@hallam.tinsleyviaduct.com) |
| 17:56.54 | drcode | I try to compile astriks, I got error on chan_zap |
| 17:56.57 | drcode | any idea? |
| 17:57.07 | drcode | I have compile zaptel , it seems to work |
| 17:57.12 | drcode | in dmesg |
| 17:57.36 | EmleyMoor | My Digium Zap fxo channel shows as onhook but takes my line offhook when I plug it in - I'm using BT - any specific settings I should check? |
| 17:58.09 | *** join/#asterisk luke-jr_work (n=luke-jr@2002:4335:4375:0:20d:60ff:fe60:756a) |
| 17:58.15 | luke-jr_work | anyone have a broadvoice account working? |
| 17:58.23 | hoobastooba | under what circumstances would a queue give a busy signal? I have a queue that periodically in the database shows last app queue and disposition busy... here is my dial plan and queue config: http://pastebin.ca/251031 |
| 17:58.30 | hoobastooba | it only happens once in a while |
| 17:59.06 | *** join/#asterisk Formater (i=Formater@cable-87-116-139-246.dynamic.sbb.co.yu) |
| 17:59.47 | Formater | hi :) there is a Dial(Local/003243252@from-internal) line... it worked until now, now freepbx is updated, and won't work anymore, I get: |
| 18:00.00 | Qwell[] | Formater: See channel topic |
| 18:00.01 | Formater | Nov 17 18:54:31 NOTICE[9084]: app_dial.c:1056 dial_exec_full: Unable to create channel of type 'Local' (cause 0 - Unknown) |
| 18:00.23 | *** join/#asterisk apardo (n=apardo@87.217.146.11) |
| 18:00.23 | Formater | but can't find anything on google :) |
| 18:00.33 | hoobastooba | try the freepbx channel |
| 18:00.37 | hoobastooba | they may be able to help yuo |
| 18:00.38 | Corydon-w | Formater: this is not a support channel for FreePBX |
| 18:00.38 | hoobastooba | you |
| 18:01.22 | Formater | hm, it has nothing to do with freepbx, i only say that it stopped working since that.. but this line is part of an app written by me |
| 18:01.27 | Formater | and the asterisk Dial command fails. |
| 18:01.40 | *** part/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 18:02.29 | hoobastooba | Formater: show your from-internal context |
| 18:02.50 | Corydon-w | Formater: and please use pastebin |
| 18:02.55 | hoobastooba | sorry... yes |
| 18:02.57 | CunningPike | drcode: Can you pastebin the output from your 'make' - say, the last 20 lines or so? |
| 18:03.01 | CunningPike | ~pastebin |
| 18:03.02 | jbot | extra, extra, read all about it, pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 18:05.18 | mercestes | *cries* How do I limit the number of calls on a polycom? |
| 18:05.34 | hoobastooba | call limit |
| 18:05.49 | hoobastooba | calllimit=1 or something like that... check the wiki for call limit |
| 18:06.22 | mercestes | call limit is in sip.conf...is it not? |
| 18:06.23 | hoobastooba | mercestes: what are you trying to accomplish? |
| 18:06.25 | Formater | hoobastooba: http://pastebin.ca/251035 |
| 18:06.26 | mercestes | I'm looking for polycom.cfg |
| 18:06.48 | hoobastooba | i have not found a way to do it in the polycom.cfg... i did it in the sip.conf |
| 18:06.53 | hoobastooba | worked great |
| 18:06.56 | CunningPike | mercestes: I have it - let me look it up - it's callsperlinekey, iirc |
| 18:07.26 | ucfMethod | any perl agi experts in here atm? |
| 18:07.40 | mercestes | Foudn it. |
| 18:07.42 | mercestes | you rock..thans |
| 18:07.45 | mercestes | thbanks |
| 18:07.50 | mercestes | gah. Arigato |
| 18:08.13 | hoobastooba | Formater: where is the line for Local/003243252@from-internal |
| 18:08.32 | CunningPike | mercestes: reg.n.callsPerLineKey="1" |
| 18:08.53 | hoobastooba | extensions.conf has to have instructions on how to handle 003243252 |
| 18:09.14 | hoobastooba | exten => 003243252,1,Dial(blah) |
| 18:11.24 | hoobastooba | do the queues have a maximum number of connections? can they give a busy signal if the number of callers exceeds a queues maximum? |
| 18:12.11 | rene1 | CunningPike |
| 18:12.12 | hoobastooba | i see maxlen = 0. Is that the default? of do i have to specify it? |
| 18:12.21 | hoobastooba | of=or |
| 18:12.25 | CunningPike | rene1 |
| 18:12.32 | rene1 | do you know a way to avoid a queue sending more than one call to a sip peer? |
| 18:12.39 | Formater | hoobastooba: http://pastebin.ca/251042 <-- here, around 13-14th line :) |
| 18:12.40 | CunningPike | rene1: Polycoms? |
| 18:12.57 | rene1 | i sometimes get the first call dropped and sometimes a conference with two callers and one agent |
| 18:13.02 | rene1 | SJphones |
| 18:13.08 | rene1 | eew i know |
| 18:13.21 | hoobastooba | rene1:I do this by using chanisavail... gimme a sec and I can paste an example. |
| 18:13.32 | hoobastooba | was a pain for me to learn, so I am happy to share it... |
| 18:13.38 | rene1 | thanks man |
| 18:13.46 | CunningPike | rene1: What we do with our Polycoms is limit the number of calls per registration to 1, so it returns Busy to Asterisk |
| 18:13.49 | rene1 | you do the chanisavail before dialing the real channel |
| 18:13.56 | hoobastooba | gimme a sec to look at Formater's deal... He is getting short with me. odnt want him to pee his pants |
| 18:14.03 | rene1 | ok |
| 18:14.05 | hoobastooba | rene1:yes |
| 18:14.16 | rene1 | i would still like to see it i will wait |
| 18:14.17 | CunningPike | rene1: There might be a way to do that in SJPhone - otherwise I would use channel groups |
| 18:14.24 | rene1 | yeah channel groups |
| 18:14.35 | Kyler | About a month ago a new version of Chanskype with source for the Asterisk portion was to be released. What happened? |
| 18:14.40 | rene1 | but actually app_queue should be doing the right thing? |
| 18:14.51 | rene1 | or probably local channels do not report device state |
| 18:14.56 | hoobastooba | how are you adding your agents? |
| 18:14.57 | rene1 | i am using it with the \n |
| 18:15.17 | rene1 | dynamic via AJAM |
| 18:15.21 | rene1 | HTTP |
| 18:15.29 | rene1 | 1.4 |
| 18:16.00 | hoobastooba | rene1: http://pastebin.ca/251043 |
| 18:16.04 | rene1 | i am looking at this patch but i dont know if it is requires in 1.4 http://bugs.digium.com/view.php?id=7433&nbn=42 |
| 18:16.06 | rene1 | thx |
| 18:16.08 | tzafrir_laptop | Please note: due to problems with our (Xorcom) hosting, both our site and our email are not working at the moment. |
| 18:16.20 | hoobastooba | the queue will not handle dynamic agents the way agents.conf does.... |
| 18:16.27 | hoobastooba | you have to control the channel. |
| 18:16.47 | rene1 | hoobastooba: are you dropping the caller back to the queue? |
| 18:16.51 | tzafrir_laptop | duh... working now... |
| 18:16.57 | CunningPike | db1310: Fix that, will ya? |
| 18:17.04 | rene1 | if chanunavail? |
| 18:17.34 | rene1 | and you seem to be using group variables |
| 18:17.50 | hoobastooba | Formater: for giggles, add a line that says exten => 003243252,1,Playback(vm-goodbye) and then dial from your dial line. |
| 18:17.55 | hoobastooba | then post all of the output |
| 18:18.15 | hoobastooba | it never leaves the queue if the chan is not avail |
| 18:18.23 | luke-jr_work | Kyler, they were probably called on it being illegal or something |
| 18:18.24 | hoobastooba | rene1: that was for you |
| 18:18.57 | Kyler | luke-jr_work: The new version is "illegal or something" but not the old one? |
| 18:19.05 | rene1 | it never leaves the quue? |
| 18:19.10 | luke-jr_work | Kyler, the old one wasn't distributed |
| 18:19.13 | hoobastooba | rene1: if you give me a sec, I will capture some output so you can see it in action. |
| 18:19.40 | Kyler | luke-jr_work: Uh...you're saying this won't work? http://www.chanskype.com/try |
| 18:19.46 | rene1 | hoobastooba: of course, but it seems to me that when it hits the local channel context leaves the queue? |
| 18:19.56 | luke-jr_work | Kyler, note you don't download anything |
| 18:20.01 | luke-jr_work | Kyler, it's a service |
| 18:20.05 | EmleyMoor | Any clues on why my fxo port picks my line up constantly? |
| 18:20.21 | Kyler | Oh, right...here.http://www.chanskype.com/buy |
| 18:20.41 | Kyler | (I haven't tried it yet 'cause I detest PayPal.) |
| 18:21.29 | luke-jr_work | Kyler, I imagine it's also still a service |
| 18:21.40 | luke-jr_work | Kyler, unless they licensed Asterisk under a commercial license |
| 18:22.00 | Kyler | luke-jr_work: So you imagine that all of the release information about it is false? |
| 18:22.01 | hoobastooba | rene1: http://pastebin.ca/251047 it is the queue that is dialing out, and not getting a response... it will not deliver the call until one shows avail... so it actually stays in queue: rene1: |
| 18:22.16 | luke-jr_work | Kyler, no clue, but it would not be legal |
| 18:22.25 | hoobastooba | i did not let my call stay in queue long enough for it to be delivered... but thats the basic idea of chanisavail |
| 18:22.26 | Kyler | Sure. |
| 18:22.32 | luke-jr_work | Kyler, short of Digium providing an exception |
| 18:22.34 | hoobastooba | it works really well for us |
| 18:22.57 | rene1 | i know what you mean |
| 18:23.06 | *** join/#asterisk ellisdee (n=ellisdee@69.15.174.114) |
| 18:23.09 | *** join/#asterisk Gunnar (n=gunnar@83.210.103.7) |
| 18:23.16 | rene1 | i see that in line 15 |
| 18:23.17 | Kyler | luke-jr_work: It'd be nice if it came as a SIP proxy. |
| 18:23.21 | rene1 | it tries to reach another member |
| 18:23.34 | luke-jr_work | Kyler, not really, I'd prefer a channel |
| 18:23.39 | rene1 | so its still in the queue |
| 18:24.02 | rene1 | so to speak |
| 18:24.06 | rene1 | nice |
| 18:24.16 | rene1 | i will try to implement it |
| 18:24.30 | Kyler | luke-jr_work: Right, but (perhaps) that conflicts with the Asterisk license. Also, making it look like SIP would allow it to run with other software. |
| 18:24.45 | *** join/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.232.175.Dial1.SanJose1.Level3.net) |
| 18:25.51 | luke-jr_work | Kyler, the only chan_skype I'd be interested in anyway would be one reverse enginneers |
| 18:25.52 | luke-jr_work | ed* |
| 18:30.31 | *** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust944.leed.cable.ntl.com) |
| 18:31.37 | hoobastooba | Formater: what is your outcome. |
| 18:31.55 | rene1 | hoobastooba: what asterisk are you running? |
| 18:32.18 | *** join/#asterisk mog (i=ejabberd@71.207.215.93) |
| 18:32.19 | *** mode/#asterisk [+o mog] by ChanServ |
| 18:32.27 | hoobastooba | 1.2.10 and 1.2.13 |
| 18:32.39 | rene1 | ok |
| 18:32.40 | hoobastooba | i have 5 servers supporting 16 t1s and about 200 agents |
| 18:32.41 | rene1 | thx |
| 18:32.44 | rene1 | wow |
| 18:32.54 | hoobastooba | 200 over 5 servers... |
| 18:32.59 | rene1 | around 40 agents per server |
| 18:33.04 | rene1 | doing recording? |
| 18:33.06 | hoobastooba | my max server supports about 120 |
| 18:33.09 | hoobastooba | yes on recording |
| 18:33.30 | rene1 | what type of hw are you using on the 120 agent box |
| 18:33.47 | hoobastooba | all of my servers are dl380's |
| 18:33.56 | rene1 | HP |
| 18:34.02 | hoobastooba | http://h10010.www1.hp.com/wwpc/us/en/ss/WF05a/15351-241434-241475-241475-f79-1121516.html |
| 18:34.20 | rene1 | i am stuck with no name p4 clones |
| 18:34.31 | hoobastooba | I have not yet learned how to do dundi, that is the next project |
| 18:34.33 | rene1 | if i can get 20 agents in one box i am happy |
| 18:34.37 | rene1 | dundi sounds nice |
| 18:34.46 | hoobastooba | you doing recordings? |
| 18:34.48 | rene1 | yes |
| 18:34.52 | hoobastooba | ramdisk? |
| 18:34.55 | rene1 | nope |
| 18:34.58 | rene1 | thats next |
| 18:35.06 | rene1 | are you using a ramdisk |
| 18:35.15 | hoobastooba | you will want to do that, it will affect your call quality when you get more. |
| 18:35.28 | rene1 | i was even thinking of using something like S3 to store the recordings permanently |
| 18:35.31 | rene1 | amazon s3 |
| 18:35.35 | rene1 | but |
| 18:35.39 | rene1 | i dont know |
| 18:35.47 | hoobastooba | i have music on hold and recordings going to ramdisk, and my "media server" where i store my recordings pulls them from ram disk ever few minutes. |
| 18:36.13 | rene1 | your media server is it a SAN? |
| 18:36.19 | hoobastooba | nope... not yet. |
| 18:36.31 | rene1 | what type of storage are youusing? |
| 18:36.58 | hoobastooba | just a large raid on a DL380 |
| 18:37.15 | rene1 | SATA? |
| 18:37.16 | hoobastooba | had to upgrade HD cuz we ran out of room |
| 18:37.19 | hoobastooba | yeah |
| 18:37.43 | trelane_ | sata's blindingly fast |
| 18:37.43 | hoobastooba | it is doing really well |
| 18:38.02 | rene1 | sweet setup |
| 18:38.10 | hoobastooba | it is getting there... |
| 18:38.12 | drcode | any one know this error [chan_zap.so]Nov 17 20:36:16 WARNING[4799]: loader.c:325 __load_resource: /usr/lib/asterisk/modules/chan_zap.so: undefined symbol: ast_pickup_call |
| 18:38.21 | hoobastooba | thanks to the help of every one here.... ;) |
| 18:38.27 | rene1 | 2 dual core 3 Ghz procesors? |
| 18:38.33 | hoobastooba | yes |
| 18:38.35 | rene1 | trelane_ sweet |
| 18:38.44 | rene1 | hoobastooba: cool |
| 18:38.47 | hoobastooba | on my media server... |
| 18:38.52 | trelane_ | rene1, all the benefits of firewire (low level protocol etc) for 1/100th the price |
| 18:38.56 | hoobastooba | the PBX's dont need that much proc |
| 18:39.00 | trelane_ | rene1, eerr fiberchannel |
| 18:39.01 | hoobastooba | just memory |
| 18:39.12 | trelane_ | hoobastooba, and fast (low latency) memory at that |
| 18:39.18 | rene1 | fibrechannel is so expensive |
| 18:39.33 | hoobastooba | correct |
| 18:39.47 | rene1 | hoobastooba: how much ram are you using minus ram for the ram disk |
| 18:40.04 | trelane_ | hoobastooba, eek, ramdisks? look at the gigabyte iram |
| 18:40.16 | hoobastooba | 4gb and i have about 1gb set up for ramdisk |
| 18:40.16 | trelane_ | 4gb of solid state memory storage connected via SATA |
| 18:40.39 | hoobastooba | trelane_: will look at it, thanks for the pitch |
| 18:40.51 | rene1 | trelane_: how fast is that iRam? |
| 18:41.01 | rene1 | scsi kind of fast? |
| 18:41.20 | trelane_ | rene1, full bore sata-150 fast |
| 18:41.25 | trelane_ | it's DDR attached via a sata connector |
| 18:41.38 | EmleyMoor | Hmmm... looks like my FXO module is faulty! |
| 18:41.39 | trelane_ | battery backup too so that if the system crashes the data isn't lost as in a ramdisk which clears at POST |
| 18:41.43 | trelane_ | EmleyMoor, :( |
| 18:41.53 | trelane_ | hoobastooba, |
| 18:41.55 | EmleyMoor | My FXS modules all work |
| 18:41.55 | rene1 | very cool trelane |
| 18:41.58 | trelane_ | hoobastooba, http://www.gigabyte.com.tw/Products/Storage/Products_Overview.aspx?ProductID=2180&ProductName=GC-RAMDISK |
| 18:42.17 | hoobastooba | there right now, reading.. thanks |
| 18:42.24 | trelane_ | also useful for things like... MySQL databases (CDR anyone?) (realtime anyone?) |
| 18:42.52 | trelane_ | squid proxies (no real application here whatsoever) |
| 18:42.57 | hoobastooba | how much? |
| 18:43.05 | trelane_ | $119 for the card, and buy some ram |
| 18:43.12 | rene1 | sweet |
| 18:43.13 | trelane_ | ram doesn't have to be low latency because it's going over SATA anyway |
| 18:43.13 | hoobastooba | very cool |
| 18:43.15 | rene1 | and not expensive at all |
| 18:43.19 | trelane_ | the price for what you get really is good |
| 18:43.22 | rene1 | but is is PCI? |
| 18:43.39 | trelane_ | rene1, it draws it's power via PCI, but it actually interfaces via a SATA controller |
| 18:43.40 | rene1 | but acts as SATA? |
| 18:43.43 | rene1 | cool |
| 18:43.47 | rene1 | i get it now |
| 18:43.55 | trelane_ | rene1, the sata cable runs off the back |
| 18:44.23 | trelane_ | battery is good for 8-10 hours depending on how much ram you have in it |
| 18:44.31 | rene1 | very cool |
| 18:45.25 | EmleyMoor | Can someone please confirm for me that asterisk can have a pulse dial phone attached yet tone dial on the outside Zap channel? |
| 18:45.40 | Qwell[] | sure |
| 18:45.52 | trelane_ | yay Qwell[] ! |
| 18:47.01 | tzafrir_laptop | EmleyMoor, yes, pulse dial sure works. |
| 18:47.09 | hoobastooba | trelane_: you are using this in linux? |
| 18:47.13 | hoobastooba | what distro? |
| 18:47.16 | tzafrir_laptop | We had a demo with a rotary phone in Astricon |
| 18:47.57 | EmleyMoor | Nice |
| 18:48.13 | tzafrir_laptop | EmleyMoor, anyway, a phone is connected through an FXS channel, whereas asterisk dials through an FXO channel |
| 18:48.21 | EmleyMoor | Yes |
| 18:48.25 | trelane_ | hoobastooba, gentoo |
| 18:48.28 | trelane_ | want some metrics |
| 18:48.30 | hoobastooba | ok, cool |
| 18:48.37 | EmleyMoor | My FXO channel is broken but I can now press on with the design while I get that fixed |
| 18:48.45 | trelane_ | hoobastooba, I knew you wanted metrics! |
| 18:48.46 | hoobastooba | sure. since you probably are really excited to share them ;) |
| 18:48.58 | *** join/#asterisk WGFreewill (n=chatzill@69-170-244-239.atlsfl.adelphia.net) |
| 18:49.07 | tzafrir_laptop | "my fxo channel == ?" |
| 18:49.12 | SheriF_SpacE | i have a question .. now i have 2 lines comming out from the PSTN over PRI or BRI "not sure don't know " to a Cisco VG " as i think " routed to 2 analog phones " as i understand this are extensions lines " so the user has to dial 9 before the call .. now i want to plug the 2 lines to an asterisk box . which modules i should use in TDM400P and any special configs. to detected the busy tone and so on ? |
| 18:49.21 | trelane_ | hoobastooba, let me find a server that's not horridly busy that I can swapoff |
| 18:50.21 | tzafrir_laptop | ah, of the Digium card |
| 18:51.10 | tzafrir_laptop | SheriF_SpacE, any way you can connect Asterisk directly to the digital channels rather than analog? |
| 18:51.50 | SheriF_SpacE | tzafrir_laptop: no the PRI line and teh VG not under my control and i have to pass by the CISCO VG |
| 18:52.04 | trelane_ | http://video.google.com/videoplay?docid=-51784544344753709 |
| 18:52.05 | trelane_ | /dev/hde: |
| 18:52.05 | trelane_ | <PROTECTED> |
| 18:52.05 | trelane_ | <PROTECTED> |
| 18:52.32 | trelane_ | that's without libata unsupported rev of sata_nv, and only 1 stick of cas-3 512mb ddr400 |
| 18:52.53 | drcode | any one can help me with this |
| 18:52.54 | drcode | Nov 17 20:47:50 WARNING[5122]: config.c:499 process_text_line: parse error: No category context for line 1 of zapata.conf |
| 18:53.06 | drcode | I put context = test |
| 18:53.08 | trelane_ | drcode, paste line 1 of your zapata.conf |
| 18:53.27 | rene1 | trelane_: cool |
| 18:53.48 | trelane_ | Dr-Linux|work, try [trunkgroups] |
| 18:53.52 | trelane_ | you cant start wtih an x = y |
| 18:53.54 | trelane_ | you need a context |
| 18:54.15 | trelane_ | rene1, as it's solid state, there is 0 seek time |
| 18:54.16 | Dr-Linux|work | okey :P |
| 18:54.18 | drcode | http://de.pastebin.ca/251073 |
| 18:55.27 | trelane_ | Dr-Linux|work, start with an empty context called [trunkgroups] add a context below that called [channels], then put that stuff in |
| 18:55.41 | trelane_ | you'll eventually have to pick how or if you want to use trunk groups, but I doubt it with one fxo |
| 18:55.43 | Dr-Linux|work | trelane: ok :P |
| 18:55.54 | Dr-Linux|work | ? |
| 18:55.59 | trelane_ | Dr-Linux|work, ignore my tab completion |
| 18:56.06 | Dr-Linux|work | ok |
| 18:56.07 | trelane_ | drcode, please read what was misdrected to Dr-Linux|work |
| 18:57.17 | WGFreewill | drcode, at least put [channels] for line 1 |
| 18:57.27 | *** part/#asterisk EmleyMoor (i=ejabberd@hallam.tinsleyviaduct.com) |
| 18:58.02 | drcode | a min |
| 18:58.43 | drcode | k |
| 18:58.44 | drcode | thanx |
| 18:58.46 | drcode | now it work |
| 19:00.26 | *** join/#asterisk xnon (i=xnon@200.8.30.3) |
| 19:00.26 | SheriF_SpacE | agin my issue :-D |
| 19:00.27 | SheriF_SpacE | i have a question .. now i have 2 lines comming out from the PSTN over PRI or BRI "not sure don't know " to a Cisco VG " as i think " routed to 2 analog phones " as i understand this are extensions lines " so the user has to dial 9 before the call .. now i want to plug the 2 lines to an asterisk box . which modules i should use in TDM400P and any special configs. to detected the busy tone and so on ? |
| 19:00.27 | *** join/#asterisk hmmhesays (n=Neg@24-117-135-28.cpe.cableone.net) |
| 19:03.28 | *** join/#asterisk justinu|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 19:08.52 | *** join/#asterisk groogs__ (n=greg@d38-54-164.commercial1.cgocable.net) |
| 19:09.16 | *** part/#asterisk hoobastooba (n=ckwall@63.149.122.93) |
| 19:10.33 | *** join/#asterisk xnon (n=xnon@200.8.30.3) |
| 19:11.10 | *** join/#asterisk w9sh (n=w9sh@adsl-068-209-117-205.sip.asm.bellsouth.net) |
| 19:11.33 | *** part/#asterisk rene1 (n=rene1@189.140.242.8) |
| 19:13.57 | *** join/#asterisk LoneShadow (n=duh@c-67-188-235-220.hsd1.ca.comcast.net) |
| 19:15.50 | *** join/#asterisk groogs__ (n=greg@d38-54-164.commercial1.cgocable.net) |
| 19:17.43 | mercestes | *cries* can someone ponti me at a sipconnect wiki? |
| 19:17.44 | mercestes | please? |
| 19:17.55 | twisted[asteria] | www.google.com? |
| 19:18.13 | mercestes | nice. www.google.com just gives me a search bar. |
| 19:18.20 | SheriF_SpacE | mercestes: voip-info.org |
| 19:18.20 | twisted[asteria] | so search for it? |
| 19:18.45 | mercestes | I did that...but I didn't find it. |
| 19:19.20 | trelane_ | mercestes, I did wtih like 5 seconds of looking |
| 19:19.23 | twisted[asteria] | wow that's strange.. i go to google and type in sipconnect and get many results regarding sipconnect. |
| 19:19.25 | WGFreewill | anyone know if zaptel 1.4 beta works with sangoma wanpipe |
| 19:19.27 | trelane_ | two faq's one howto etc |
| 19:19.33 | mercestes | what did you search? LOL |
| 19:19.38 | mercestes | I did asterisk sip connect |
| 19:19.38 | trelane_ | WGFreewill, let us know ;) |
| 19:19.50 | trelane_ | sipconnect |
| 19:19.52 | *** join/#asterisk zotz (n=zotz@208.196.247.175) |
| 19:20.47 | trelane_ | why are you trying to use asterisk with sipconnect? |
| 19:21.38 | mercestes | Ok, let me clarify. |
| 19:21.45 | mercestes | I wish to connect my two * boxes via SIP.... |
| 19:22.15 | mercestes | I have always referred to that as a "sip connect" but I could be mistaken. |
| 19:22.24 | trelane_ | sip friend sip trunk |
| 19:22.32 | trelane_ | sip connect is a propriatary product |
| 19:22.34 | trelane_ | err sipconnect even |
| 19:22.43 | trelane_ | or "connect via sip" |
| 19:22.50 | trelane_ | please don't invent buzzwords, it hurts the furry bunnies |
| 19:22.58 | SheriF_SpacE | mercestes: why not IAX i trunk my asterisks using IAX |
| 19:22.58 | trelane_ | AND YOU DONT WANT TO BE SEEN HURTING FURRY BUNNIES... DO YOU? |
| 19:23.24 | mercestes | Found it. |
| 19:23.58 | mercestes | I didn't invent it, I connected * to Cbeyond before and they referred to them as "sipconnects" and since they were in sip.conf...I presumed hte connection method is simliar. |
| 19:24.08 | x6541a | Hello, anybody using a cisco 2801 to connect * to the PSTN? |
| 19:24.10 | mercestes | and according to this wiki, it is...I just didn't know "Sipconnect" was TM of Cbeyond |
| 19:24.26 | trelane_ | x6541a, what media? vic-2fxo? |
| 19:24.39 | mercestes | and I can't use Iax because I have one 1.0.9 * box and one 1.2.13 * box and the Iax jitterbuffering/qualifying is not compatible. |
| 19:24.41 | x6541a | trelane, sorry, ISDN PRI |
| 19:25.35 | trelane_ | x6541a, good luck! |
| 19:25.36 | mercestes | and I I'm holding off on upgrading the 1.0.9 box because I am aiming for a controled migration over a mass update. |
| 19:25.36 | x6541a | trelane, errr, that was not an reply I was looking for ;) |
| 19:25.54 | *** join/#asterisk drcode (n=chatzill@87.69.59.186.cable.012.net.il) |
| 19:25.57 | drcode | hi all |
| 19:26.03 | SheriF_SpacE | mercestes: and u can use SIP all u need is to use register => bla bla |
| 19:26.09 | drcode | I try to dial out |
| 19:26.11 | drcode | in dial plan |
| 19:26.12 | drcode | I get this |
| 19:26.18 | drcode | Rejected connect attempt from 10.0.100.7, request '9@tutorial' does not exist |
| 19:26.20 | drcode | any idea? |
| 19:26.45 | drcode | this is my dial plan: exten => 9NXXXXXX,1,Dial,Zap/1/BYEXTENSION |
| 19:27.17 | *** join/#asterisk vader-- (n=johndoe@204.183.88.101) |
| 19:27.58 | mercestes | SheriF_SpacE: Yea, I was missing hte syntax to pass a username to said * boxes in the dialplan...thanks. |
| 19:31.40 | ucfMethod | anyone know why "chan_sip.c:695 retrans_pkt: Maximum retries exceeded on call ...... " is appearing over and over in my asterisk console? |
| 19:32.29 | drcode | how I can dial with this numer? |
| 19:32.31 | drcode | I put 9? |
| 19:45.42 | *** join/#asterisk olinux (n=olinux@starbucks.wellspublishing.net) |
| 19:47.11 | *** join/#asterisk alerios (n=alerios@190.24.100.110) |
| 19:48.26 | olinux | so the zaptel* are drivers for digium cards? |
| 19:48.42 | olinux | and because i do not have a card installed i need to install ztdummy? |
| 19:49.31 | tzafrir_laptop | olinux, zaptel also provides timing for a unmber of functionalities of Asterisk |
| 19:49.47 | olinux | which is why i want to install ztdummy? |
| 19:50.25 | Corydon-w | Correct |
| 19:52.04 | olinux | cool, thank you |
| 19:52.40 | x6541a | olinux, that's the way to go yes. If you don't your MOH might slow down or speed up |
| 19:54.36 | x6541a | trelane_, what media gateways are you using yourself |
| 19:55.06 | *** join/#asterisk FuriousGeorge (n=brian@ool-43536ea8.dyn.optonline.net) |
| 19:55.12 | FuriousGeorge | hey all |
| 19:55.54 | trelane_ | x6541a, I use the zaptel cards |
| 19:55.58 | trelane_ | digium's stuff |
| 19:56.06 | trelane_ | I use sangoma for IP routing |
| 19:56.27 | trelane_ | I like wanpipe for what it was meant for and afaic they aren't worth a damn for voice. |
| 19:56.30 | *** join/#asterisk spr1te (n=spr1te@213.227.193.75) |
| 19:57.46 | FuriousGeorge | someone wants me to set up asterisk for them next to their current pbx just for conferencing. if their existing pbx accepts it i would probably interface it via an analog fxs, right? if i interfaced it with an fxo they would be able to access a bunch of extensions, but i have no idea how to setup their existing pbx to do it |
| 19:58.13 | FuriousGeorge | i dunno.. anyone got another suggestion |
| 20:00.25 | FuriousGeorge | i guess i can set up asterisk at a separate extension then have it answer and say "enter conference number" |
| 20:02.59 | aydiosmio | well |
| 20:03.02 | aydiosmio | if you're conferencing with it |
| 20:03.03 | mercestes | http://pastebin.ca/251127 |
| 20:03.09 | aydiosmio | you're gonna need a hell of a lot more than one |
| 20:03.16 | mercestes | Can someone help me with connecting my two * boxes together via sip? I can't seem to pass a call. |
| 20:04.34 | *** join/#asterisk Defraz (n=t0tal@fw.centrisys.com) |
| 20:05.12 | *** join/#asterisk Ox0F0-0FF (n=pierre@201.38.238.66) |
| 20:05.18 | aydiosmio | FuriousGeorge: best way to go is with a T1 card, if the PBX has a free T1 interface, 24 channels -- 24 conference parties |
| 20:05.59 | tzafrir_laptop | mercestes, first off: why not use IAX to connect them? |
| 20:06.15 | tzafrir_laptop | mercestes, is there NAT in the way, or direct connection? |
| 20:06.33 | mercestes | tzafrir_laptop: One is 1.0.9 and the other is 1.2.13. Iax2 is not compatible between them. They are all internal IPs. No natting. |
| 20:07.38 | mercestes | tzafrir_laptop: I want to work on a controlled migration from the outdated box to the "new box" and in order to do that I need ot establish a media path between them as I migrate phones over to the new stuff. |
| 20:09.12 | *** part/#asterisk Ox0F0-0FF (n=pierre@201.38.238.66) |
| 20:09.43 | *** join/#asterisk dasenjo (n=dasenjo@201.228.128.10) |
| 20:12.23 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
| 20:13.39 | x6541a | trelane_, that works flawlessly? |
| 20:14.38 | WGFreewill | Sangoma wanpipe doesn't like zaptel 1.4 |
| 20:14.46 | WGFreewill | endless loops |
| 20:17.18 | mercestes | Really? Awesome. |
| 20:19.35 | *** join/#asterisk linagee (n=linagee@unaffiliated/linagee) |
| 20:25.02 | *** join/#asterisk zotz (n=zotz@208.196.247.175) |
| 20:30.10 | tzafrir_laptop | WGFreewill, could you be more specific? at build time or at runtime? |
| 20:30.22 | tzafrir_laptop | At build time, I guess |
| 20:30.26 | WGFreewill | at build time |
| 20:30.27 | *** join/#asterisk Aurs (n=Aurs@host-81-191-123-189.bluecom.no) |
| 20:30.38 | tzafrir_laptop | zaptel 1.4's runtime hasn't really changed |
| 20:30.41 | WGFreewill | it looks like its unfamilar with the new automake |
| 20:30.58 | *** join/#asterisk mpruett (n=mpruett@24-240-203-83.static.stls.mo.charter.com) |
| 20:31.00 | WGFreewill | ln -s -s -s -s -s -s -s all across the screen |
| 20:31.12 | tzafrir_laptop | WGFreewill, find the place in the scripts where they run make |
| 20:31.16 | WGFreewill | and it keeps looping around configuring |
| 20:31.32 | tzafrir_laptop | tell that script to run something of the sort of: |
| 20:32.19 | WGFreewill | yeah those sangoma guys will get to it im sure |
| 20:32.22 | tzafrir_laptop | [ if -x PATH/TO/ZAPTEL_DIR/configure ]; then ( cd PATH/TO/ZAPTEL_DIR; ./configure ]; fi |
| 20:32.23 | WGFreewill | maybe they are waiting for the release |
| 20:32.27 | tzafrir_laptop | just before the 'make' |
| 20:32.33 | WGFreewill | exactly |
| 20:32.53 | tzafrir_laptop | Worked for me for the zaptel deb and in some other places |
| 20:33.12 | mpruett | Anyone have experience sending sip notify messages to Sipura devices? I get unauthorized back from device when I send notification. |
| 20:33.13 | WGFreewill | they patch zaptel a little |
| 20:33.32 | WGFreewill | I need this particular box to work |
| 20:33.34 | tzafrir_laptop | This should have no problem with zaptel 1.2, so it is safe to add to current version |
| 20:43.19 | *** part/#asterisk FuriousGeorge (n=brian@ool-43536ea8.dyn.optonline.net) |
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| 20:47.45 | mpruett | Anyone have experience sending sip notify messages to Sipura devices? I get unauthorized back from device when I send notification. |
| 20:47.51 | *** join/#asterisk spunz_ (n=spunz@h081217096236.dyn.cm.kabsi.at) |
| 20:49.41 | *** join/#asterisk FuriousGeorge (n=brian@ool-43536ea8.dyn.optonline.net) |
| 20:50.28 | FuriousGeorge | i dunno if anyone answered this question i had but i got a phone call and missed it: |
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| 20:59.23 | aydiosmio | FuriousGeorge: ? |
| 21:02.46 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
| 21:03.24 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 21:04.46 | *** join/#asterisk maverickbna (i=sentinel@wikipedia/Shadowhntr) |
| 21:05.01 | aydiosmio | oh well |
| 21:05.01 | aydiosmio | 15:03 < aydiosmio> if you're conferencing with it you're gonna need a hell of a lot more than one |
| 21:05.05 | aydiosmio | 15:05 < aydiosmio> FuriousGeorge: best way to go is with a T1 card, if the PBX has a free T1 interface, 24 channels -- 24 conference parties |
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| 21:17.30 | *** join/#asterisk tr2x (n=alvar@80-218-150-90.dclient.hispeed.ch) |
| 21:17.55 | linagee | what is a LNP charge? |
| 21:18.07 | linagee | i mean, why do different providers charge different amounts? |
| 21:18.21 | linagee | sellvoip charges $8 to port a number, voicepulse charges $25 |
| 21:18.24 | *** join/#asterisk Ox0F0-0FF (n=pierre@201.38.238.66) |
| 21:19.42 | *** part/#asterisk Ox0F0-0FF (n=pierre@201.38.238.66) |
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| 21:26.36 | TommyTheKid | linagee: they are bastards |
| 21:27.07 | TommyTheKid | LNP - local number port? (I assume) .. is a service charge |
| 21:28.30 | justinu|laptop | linagee: fyi, L3 wholesale charges 50cents/LNP i think |
| 21:33.49 | FuriousGeorge | for some reason i cant transfer an external extenions (my cellphone) into a local conference room |
| 21:33.56 | FuriousGeorge | local extension are no problem |
| 21:34.34 | *** join/#asterisk xnon (i=xnon@200.8.30.3) |
| 21:35.07 | mpruett | Anyone have experience sending sip notify messages to Sipura devices? I get unauthorized back from device when I send notification. |
| 21:35.25 | mpruett | Sorry if this question was answered earlier - I lost connection |
| 21:35.59 | FuriousGeorge | -- Executing MeetMe("IAX2/voicepulse01-4", "500") in new stack. it appears i can transfer the person into the conference, but he cannot hear anything or be heard |
| 21:36.11 | FuriousGeorge | the transfer looks like it goes through, as per above |
| 21:37.06 | FuriousGeorge | i do hear a tone when its disconnected |
| 21:39.09 | *** join/#asterisk ManxPower (n=manxpowe@71-8-11-73.dhcp.leds.al.charter.com) |
| 21:41.18 | *** join/#asterisk patrickvox (n=patrick@64.235.249.83) |
| 21:41.32 | patrickvox | Hi all folks |
| 21:42.36 | patrickvox | has anyone here used Asterisk with Sysmaster/VoiceMaster ? |
| 21:43.20 | TommyTheKid | in zaptel.conf, it asks for LBO which sorta looks like its asking how long the line is (0-133, 133-266, 266-399, etc) .. is there any way to actually determine that (from remote) or is it just a best guess? |
| 21:43.52 | CunningPike | TommyTheKid: You can get tools to measure it, but I would guess |
| 21:44.25 | *** part/#asterisk pdunkel (n=pdunkel@213.235.192.27) |
| 21:45.20 | TommyTheKid | right now its set to "0" and I am almost 100% sure the line is longer than 133 ft... hell it probably gets to 133 feet before it even leaves our server room :) |
| 21:45.36 | TommyTheKid | .. what effect would that have if it was set oncorrectly? |
| 21:45.43 | TommyTheKid | guh, incorrectly |
| 21:49.11 | CunningPike | TommyTheKid: I'm not entirely sure :D |
| 21:50.33 | *** join/#asterisk [hC] (n=hardcore@mail.metrobridge.com) |
| 21:53.51 | Supaplex | is there a non intel macos/x iax/sip/h323 (in order of preference) softphone that can/will pause itunes and mute other system notifications like ichat and others? or is that just too much to ask :) free is ok, but gpl would be a bonus. |
| 21:55.03 | FuriousGeorge | ok, if i call a person on the pstn and transfer him into a converence, he can hear a local party in that conference, but the local party cannot hear him |
| 21:55.50 | TommyTheKid | Supaplex: I was using a PPC/MacOSX IAX softphone that I *think* paused iTunes.. but it was neither free nor GPL :) |
| 21:57.08 | Supaplex | how much was it? do they have a trial? |
| 21:57.44 | TommyTheKid | I think it was like $25? and it worked for up to 5 mins or something to trial... but I don't have my TiBook here to look :) |
| 21:58.23 | Supaplex | humm ok |
| 21:58.58 | FuriousGeorge | in addition, if i tranfer two external parties into the conference no one hears anyone |
| 22:01.47 | *** join/#asterisk ctooley (n=ctooley@jc1-111.moment.net) |
| 22:02.13 | *** part/#asterisk patrickvox (n=patrick@64.235.249.83) |
| 22:02.28 | ctooley | Is there anything that gets passed back up the to the dialplan about a channel indicating whether that channel is communicating with a Pay Phone? |
| 22:03.36 | Qwell[] | ctooley: IF you get ANI II, then...maybe :p |
| 22:04.07 | *** join/#asterisk mpruett (n=mpruett@24-240-203-83.static.stls.mo.charter.com) |
| 22:04.23 | *** join/#asterisk Sedorox (n=brandon@smartserv/cna/Sedorox) |
| 22:05.13 | ctooley | Qwell how would I get at the ANI II if it is passed to me? |
| 22:05.31 | linagee | justinu|laptop: hrm. too bad L3 doesn't offer voip IAX/SIP termination directly. or do they? :) |
| 22:05.35 | ctooley | (I'm working for a large telecom, I'm assuming our Veraz switches can give that to us via SIP, they do via TDM) |
| 22:06.04 | FuriousGeorge | and this only happens when i use a remote sip client. in other words local sip clients are still able to hear people they transfer into a meetme, but if i use a remote client, the PSTN parties audio doesnt make it back |
| 22:06.08 | justinu|laptop | linagee: they do not, but you can talk to an L3 reseller and pay like 2x the price and still save a fortune |
| 22:06.21 | ctooley | linagee, We do, and we cover the US just as well, and our termination rates are probably as good or better. |
| 22:06.38 | Supaplex | justinu|laptop: how is that saving when you pay double? |
| 22:07.07 | justinu|laptop | so you're paying $1/LNP instead of 5, or 25 |
| 22:07.09 | ctooley | Our network covers the entire US on our own switches |
| 22:08.06 | linagee | justinu|laptop: the only reason large businesses don't want to deal with small businesses is because i just want one LNP, right? it wouldn't be profitable enough? i thought that's what the web is for. automating that whole brick and mortar overhead thing. |
| 22:08.33 | justinu|laptop | yeah... so you small frys get to pay the big markup |
| 22:08.47 | ctooley | linagee, the problem isn't provisioning a single LNP, it's supporting it, profitably |
| 22:09.09 | justinu|laptop | right... L3 makes you go thru a fairly comprehensive interopt test |
| 22:09.22 | justinu|laptop | reseller makes you pay for that test |
| 22:09.33 | linagee | ctooley: supporting it? |
| 22:09.54 | *** join/#asterisk hoobastooba (n=ckwall@63.149.122.93) |
| 22:09.58 | linagee | ctooley: it's just an entry in a database, right? |
| 22:10.08 | ctooley | linagee, if you're paying a buck a month for a DID and have one support ticket every 2 years, I've lost all of my revenue, let alone all of my margin |
| 22:11.01 | hoobastooba | i have an extension that I send directly to exten => 2222,1,Queue... how would 12 out of 21 calls to that queue disposition as busy? |
| 22:11.11 | hoobastooba | what circumstances would cause that? |
| 22:11.20 | linagee | ctooley: i think sellvoip is like $1/mo/did |
| 22:11.33 | linagee | ctooley: and yes, you can never reach their phone support. heh |
| 22:11.35 | FuriousGeorge | ...at least there are never any shortages of surprises. when i was playing with this earlier, either the conferencing or the parking presence patch im using crashed the PBX |
| 22:13.13 | *** join/#asterisk ToTo (n=ToTo@host97-157-dynamic.2-87-r.retail.telecomitalia.it) |
| 22:13.23 | ctooley | linagee, even if I pay for a tech to answer emails, single DID customers aren't profitable. |
| 22:13.31 | linagee | ctooley: automate customer service. that's the "wave of the future" |
| 22:13.41 | linagee | ctooley: wikis, knowledge bases, forums, self checkout |
| 22:13.48 | justinu|laptop | linagee: there's always gonna be problem customers |
| 22:13.50 | hoobastooba | anyone? |
| 22:14.01 | linagee | justinu|laptop: send them to a call center in india |
| 22:14.04 | Supaplex | pay for support :) |
| 22:14.07 | justinu|laptop | it's probably like 5% of your customers that take up 99% of the support budget |
| 22:14.10 | linagee | Supaplex: ?!!?!? lol |
| 22:14.40 | linagee | Supaplex: what if you had a quota of support tickets instead? |
| 22:14.52 | linagee | Supaplex: justinu|laptop said, one support ticket in two years wipes you out. |
| 22:15.17 | ctooley | linagee, again, 1 ticket is $25 on average, that's more than 2 years of revenue |
| 22:15.18 | linagee | Supaplex: so have a knowledge base with previous support tickets, and have a quota of one support ticket every four years, or else you have to pay X amount. |
| 22:15.32 | linagee | ctooley: see above too |
| 22:15.47 | ctooley | 1 ticket every 4 years? What's the point? |
| 22:16.00 | mpruett | Anyone have experience sending sip notify messages to Sipura devices? I get unauthorized back from device when I send notification. |
| 22:16.13 | linagee | ctooley: then have a different quota, and charge when you go over that |
| 22:16.34 | ctooley | linagee, buy more DID's |
| 22:16.37 | Supaplex | 1-900-voip-help |
| 22:16.37 | linagee | ctooley: that way you won't have pissy clients saying you don't support the service |
| 22:16.52 | mpruett | I have read it is looking for MD5 Line password - but how do I send this passord to the device? |
| 22:17.25 | linagee | Supaplex: i wonder what the percentage of calls into support are problem with voip provider versus problem with client side configuration? |
| 22:18.40 | linagee | Supaplex: with something like trixbox and voicepulse and sellvoip, i would say 99% of the time it's the provider's fault. it's kind of hard to mess something up when all you have to do is enter a key into the config |
| 22:19.19 | Supaplex | yup |
| 22:21.12 | ctooley | Anyone know if chan_sip in 1.2.X supports ANI or ANI2? |
| 22:22.02 | hoobastooba | anybody :( baffled as to how a call into a queue will ring busy only some of the time. 12 out of 22 calls to this queue came back busy. |
| 22:22.17 | ManxPower | ctooley: I don't think so. Its more of a PSTN thing. |
| 22:22.45 | ctooley | ManxPower, chan_iax definitely does and I see references to it. |
| 22:41.56 | *** join/#asterisk eltech (i=G00Ds@ool-457c9421.dyn.optonline.net) |
| 22:44.58 | ctooley | What application do I call to log something? (like if I want to see what the value a function returns) |
| 22:45.07 | Qwell[] | Set? |
| 22:45.12 | Qwell[] | erm, log how? |
| 22:45.28 | *** join/#asterisk ToTo (n=ToTo@host97-157-dynamic.2-87-r.retail.telecomitalia.it) |
| 22:45.59 | *** join/#asterisk bjohnson (n=bjohnson@i216-58-63-46.cybersurf.com) |
| 22:46.04 | ctooley | Qwell[], Like write to the debug log |
| 22:46.28 | TommyTheKid | ctooley: I use "Noop" quite a bit to log to the console |
| 22:48.00 | TommyTheKid | is 1.4.0-beta3 "good enough" to use on a semi-production server? |
| 22:48.03 | *** part/#asterisk sloth101 (n=mallen@user-216-10-84-189.citynetwireless.net) |
| 22:49.01 | Corydon-w | NoOp only logs to the console with verbose set to 3 or higher |
| 22:49.17 | Corydon-w | Use Verbose if you want to log at other verbose levels |
| 22:49.31 | Juggie | TommyTheKid, is a glad bag good enough to not get herpies? |
| 22:50.02 | TommyTheKid | hehe.. HEFTY HEFTY HEFTY |
| 22:51.24 | *** join/#asterisk ftexcom (n=ftexcom@169.Red-88-2-81.staticIP.rima-tde.net) |
| 22:53.50 | ftexcom | any ideas why asterisk it's not detecting a hung up? |
| 22:54.10 | Corydon-w | ftexcom: analog line |
| 22:54.35 | ftexcom | yeap |
| 22:54.40 | ftexcom | tdm400p |
| 22:54.44 | Corydon-w | Analog sucks for signalling |
| 22:54.58 | ftexcom | I have already enables hanguponpolarityswitch |
| 22:55.00 | Corydon-w | Are you using kewlstart? |
| 22:55.02 | ftexcom | but, nothing |
| 22:55.35 | ftexcom | kewlstart? |
| 22:55.40 | Corydon-w | Well, that's working... you just don't realize that it's parsed as "hang upon polarity switch" |
| 22:56.16 | ftexcom | what's the correct option? |
| 22:56.24 | Corydon-w | kewlstart... fxo_ks and fxs_ks |
| 22:56.47 | ftexcom | oh..my zapata conf is using...fxs_ks |
| 22:57.19 | FuriousGeorge | there is no video conferencing solution for asterisk atm, is there? |
| 22:57.23 | Corydon-w | Also, are you specifying opermode=<something> when you load wctdm? |
| 22:57.34 | ftexcom | no, I'm not |
| 22:59.21 | ftexcom | hummm |
| 22:59.25 | *** part/#asterisk hoobastooba (n=ckwall@63.149.122.93) |
| 23:00.30 | *** join/#asterisk juanjoc (n=juanjoc@201.216.212.113) |
| 23:00.41 | Corydon-w | ftexcom: opermode=SPAIN |
| 23:00.56 | *** join/#asterisk robin_sz (n=robin@rapid2.gotadsl.co.uk) |
| 23:01.02 | robin_sz | hmmm ... |
| 23:01.39 | robin_sz | I am thinking of putting this GXP2000 on eBay for £0.37, buy it now, but I am worried I might get negative feedback for ripping people off |
| 23:02.22 | robin_sz | perhaps if I put "240V mains plug, with useless telephone on the end of it" .. ?? people wnting a plug might find it useful?? |
| 23:03.01 | FuriousGeorge | anyone know about video conferencing solutions for asterisk? |
| 23:03.44 | ftexcom | yes |
| 23:03.48 | ftexcom | now it's working |
| 23:03.48 | ftexcom | bravo |
| 23:05.35 | *** join/#asterisk aao_pwner (n=aao_pwne@c-24-21-91-140.hsd1.mn.comcast.net) |
| 23:08.12 | TommyTheKid | strange, I changed my Noop's to Verbose and the _X. extensions didn't change |
| 23:09.16 | dasenjo | Hi everybody! Can a Digium T200P manage a T1 and a E1? Does it have two jumpers? |
| 23:09.30 | *** join/#asterisk hyperthread (n=janusmis@c90607fe.virtua.com.br) |
| 23:09.42 | knarfly | Does anyone work with X100P or X101P cards? |
| 23:09.47 | Qwell[] | dasenjo: TE2xxp? |
| 23:09.58 | ftexcom | fxotune...tunning... |
| 23:09.59 | Qwell[] | dasenjo: just making sure I'm clear, before I ask the right people |
| 23:10.00 | dasenjo | Qwell, yes |
| 23:10.03 | Qwell[] | k |
| 23:10.26 | dasenjo | Qwell, thanks |
| 23:10.33 | ctooley | Corydon-w, Verbose doesn't seem to be interpolating the function before logging. Instead of Verbose(CDR(name)) logging the name, it's just logging "CDR(name)" |
| 23:10.37 | Qwell[] | dasenjo: Yes they do. |
| 23:10.58 | file | Qwell[]: I'm not sick but I'm not well |
| 23:11.00 | hyperthread | hello all...I have a TE110P and a TDM400P (with 4 FXO modules - X100) and when I try to locad the wcte11xp for TE110P modprobe just do not respond...the TDM400P works fine, I just cant put the two board to work at the same time....any ideas ?? |
| 23:11.16 | Corydon-w | ctooley: the first parameter of Verbose is the Verbose level |
| 23:11.25 | Corydon-w | ctooley: Verbose(2|message) |
| 23:12.20 | ctooley | <PROTECTED> |
| 23:12.22 | dasenjo | Qwell, thanks again. |
| 23:13.30 | ctooley | Corydon-w, I've even tried setting the value of a variable to CDR(name) and using verbose to print ${FOO}. FOO contains "CDR(name)" and not the name from the cdr |
| 23:13.58 | Corydon-w | ctooley: you have to interpolate |
| 23:14.10 | ctooley | Corydon-w, how is that done? |
| 23:14.10 | Corydon-w | Verbose(1|${CDR(name)}) |
| 23:14.14 | ctooley | ah |
| 23:14.51 | ctooley | much better |
| 23:15.36 | *** join/#asterisk STRHDW (n=jvh@rrcs-24-106-55-26.west.biz.rr.com) |
| 23:15.49 | *** join/#asterisk findlay (n=justin@67.137.24.115) |
| 23:17.27 | mpruett | Anyone have experience sending sip notify messages to Sipura devices? I get unauthorized back from device when I send notification. |
| 23:18.22 | *** join/#asterisk ToTo (n=ToTo@host97-157-dynamic.2-87-r.retail.telecomitalia.it) |
| 23:18.29 | *** join/#asterisk [hC] (n=hardcore@mail.metrobridge.com) |
| 23:19.32 | findlay | I have an asterisk server that connects to a provider via IAX. Using kiax/iaxcomm I can call out and communicate fine except that certain "random" numbers get no outbound audio at all |
| 23:20.09 | *** join/#asterisk Ciber311 (n=Ciber311@user-1087e94.cable.mindspring.com) |
| 23:20.12 | robin_sz | no outbound audio? |
| 23:20.14 | robin_sz | weird .. |
| 23:20.23 | robin_sz | no inbound I could partially expect ... |
| 23:20.24 | findlay | there didn't seem to be anything interesting in asterisk's debugging info |
| 23:20.44 | findlay | while comparing calls that worked versus calls with no tx audio |
| 23:22.38 | TommyTheKid | heh, with verbose() I get the "message" twice, I guess I have always started with "-Tvvvc" |
| 23:24.13 | robin_sz | findlay, I would tend to suspect a firewall issue, setting RTP ports in an region thats somehow blocked |
| 23:24.29 | findlay | robin_sz: but who's firewall? |
| 23:24.38 | findlay | mine makes no difference |
| 23:25.09 | findlay | I'm inclined to think that some gateway upstream of my provider has a funky firewall |
| 23:25.42 | robin_sz | findlay, I would tend to suspect it was upstream of you somewhere |
| 23:25.53 | robin_sz | normally outbound is just passthrough locally |
| 23:25.59 | findlay | but other people on my isp (iProvo) who use my provider (arrivaltel) have reported no such problems |
| 23:26.28 | TommyTheKid | are you behind NAT? that seems to cause all sorts of fun for such a wonderful protcol as SIP (including layer 2 info in layer 7) |
| 23:26.28 | findlay | is there a way to traceroute phone routes? |
| 23:26.48 | findlay | I'm supposed to have a fully public, static ip |
| 23:26.59 | findlay | and it appears to function that way as well |
| 23:27.09 | TommyTheKid | luck bastard :) |
| 23:27.19 | robin_sz | you have no firewall? |
| 23:27.20 | TommyTheKid | lucky that is |
| 23:27.33 | findlay | robin_sz: I do on the local machine, yes |
| 23:27.35 | *** join/#asterisk Ciber311 (n=Ciber311@user-1087e94.cable.mindspring.com) |
| 23:28.07 | robin_sz | just check the thing is open for outbound connections across thr port range set for outgoing RTP channels |
| 23:28.18 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.216.2) |
| 23:28.18 | findlay | but I've turned it off before without positive effect |
| 23:28.41 | robin_sz | well, its pronbably not you then |
| 23:28.42 | TommyTheKid | brave soul turning off your firewall on a machine on a public IP :) |
| 23:28.45 | file | [TK]D-Fender: !!! |
| 23:29.01 | findlay | how do I find what ports IAX RTP will use? |
| 23:29.02 | [TK]D-Fender | file : I don't get to stop my work! |
| 23:29.17 | robin_sz | check iax.conf |
| 23:29.32 | [TK]D-Fender | file : Left at 6pm, got an * contract for 8pm. yay |
| 23:29.41 | TommyTheKid | findlay: IAX is all on one port.. it makes punching thru firewalls a cakewalk... hmm I want cake |
| 23:29.41 | [hC] | I get the impression zttest is not reliable |
| 23:29.59 | [hC] | I have a sangoma a200 im running zttest on.. i can get huge differences just by stopping and restarting the test |
| 23:30.09 | [hC] | sometimes i get 100's across the board, sometimes its all ove the map. |
| 23:31.01 | mpruett | Anyone have experience sending sip notify messages to Sipura devices? I get unauthorized back from device when I send notification. |
| 23:32.03 | [hC] | seems to make a difference if someone is on a call or not, i suppose. |
| 23:32.59 | file | [TK]D-Fender: eeeep |
| 23:34.15 | [TK]D-Fender | file : at least the latter adds more $ to my bank account... |
| 23:36.04 | asberger | <PROTECTED> |
| 23:36.21 | findlay | I wish phone systems would let you choose your own hold music |
| 23:36.32 | findlay | or have no hold "music" ... |
| 23:36.37 | file | [TK]D-Fender: anything complex? |
| 23:36.52 | TommyTheKid | findlay: that would be kinda cool |
| 23:37.06 | findlay | yeah |
| 23:37.14 | file | my cellular provider allows you to choose the music if you are on hold long enough |
| 23:37.28 | findlay | does it play a local file or something? |
| 23:37.30 | ucfMethod | can anyone provide a bit of explaination on this error msg "res_musiconhold.c:515 monmp3thread: Request to schedule in the past?!?!" |
| 23:38.02 | ucfMethod | i am trying to get musiconhold to work in conference rooms.... MeetMe(,Mis) |
| 23:38.05 | *** part/#asterisk dasenjo (n=dasenjo@201.228.128.10) |
| 23:38.19 | TommyTheKid | does the hold music thing on asterisk play one time and sorta send that stream to everyone (join in the middle of a song) or does it play for each person? |
| 23:38.43 | TommyTheKid | not that it matters, we had to disable hold music :) |
| 23:38.49 | ucfMethod | I think it joins in during the middle of song |
| 23:39.09 | ucfMethod | at least it does on my installation |
| 23:39.15 | TommyTheKid | people don't like getting "caught" putting conference calls on hold |
| 23:39.34 | [TK]D-Fender | file : Not terribly... |
| 23:39.38 | ucfMethod | Thats what the Mute key is for |
| 23:39.48 | ucfMethod | the red button on Polycom501 is a godsend |
| 23:40.00 | TommyTheKid | thats fine till you decide to answer another incoming call |
| 23:40.13 | ucfMethod | well you got me there |
| 23:40.14 | findlay | but how does it know when you're not on hold anymore? |
| 23:40.16 | Nugget | TommyTheKid: it can work either way depending on how you configure it. |
| 23:40.31 | [TK]D-Fender | file : swap 1 TDM card of one brand for that of another, test audio quality and so on. Then part 2 (2nd customer, same site) : setup WiFi access points & sip phones |
| 23:40.43 | *** join/#asterisk asdx (n=diego@200.61.236.33) |
| 23:40.50 | file | [TK]D-Fender: cool |
| 23:41.11 | TommyTheKid | I had, for a while, a config that would disable hold music if someone called a toll free number, but then they started bitching about other internal (5 digit) extensions and hold music |
| 23:45.37 | TommyTheKid | If you had an asterisk server behind a NetScreen (Juniper) firewall, would you a) trust the "SIP ALG" in the firewall to "surf" the sip packets and open RTP ports on demand, b) open a specific range (18000-19000?) of UDP ports and configure SIP to use those (can that be set in sip.conf only, or do all clients need to know that range too) ... or c) uh? install asterisk on a WRT and put it in their home so you can use IAX2 across the firewall?? |
| 23:52.39 | ManxPower | TommyTheKid: I would disable SIP fixup on the firewall/router and then set up things just like you didn't have a SIP aware firewall/router |
| 23:53.01 | TommyTheKid | so, b? |
| 23:53.26 | ManxPower | b |
| 23:53.42 | ManxPower | You should not have to do anything at the client NAT router |
| 23:53.50 | ManxPower | only the Asterisk NAT router |
| 23:53.53 | *** join/#asterisk zotz (n=zotz@208.196.247.175) |
| 23:54.03 | TommyTheKid | the other thing I have to deal with is that probably 99.5% of my "users" are behind NAT.. so I have *their* firewall and mine to deal with.. on a positive note, the server is not NAT'd |
| 23:55.38 | ManxPower | TommyTheKid: Oh! Asterisk is not NAT'd? |
| 23:55.59 | ManxPower | TommyTheKid: %99 of the SIP clients will have no issues with NAT if you set up sip.conf correctly |
| 23:56.02 | TommyTheKid | no, its just behind a corporate firewall |
| 23:56.54 | TommyTheKid | "...if you set up sip.conf correctly" <-- there is always a catch :p |
| 23:57.16 | *** part/#asterisk mog (i=ejabberd@71.207.215.93) |
| 23:58.36 | TommyTheKid | #include users/*-sip.conf |
| 23:58.36 | TommyTheKid | .. thats all I have in my sip.conf beyond the sample :) |
| 23:59.22 | *** part/#asterisk bjohnson (n=bjohnson@i216-58-63-46.cybersurf.com) |