| 00:00.07 | Qwell[] | ^ |
| 00:00.24 | *** join/#asterisk BZBW (i=BZBW@ip67-153-142-109.z142-153-67.customer.algx.net) |
| 00:00.55 | JASON99 | I'm wondering why they should press 9 too? heh |
| 00:01.13 | GarethTheGreat | they could press any digit from 0-9, doesn't matter |
| 00:01.23 | GarethTheGreat | though i'm thinking of 0 to repeat the menu |
| 00:01.26 | BZBW | emm, anyone knows what kinda variable I can use to refer to the extension number that initiate a call? |
| 00:01.44 | GarethTheGreat | naain: where do i start configuring this? |
| 00:02.01 | GarethTheGreat | the tutorials i'm reading don't give much clues |
| 00:02.04 | BZBW | i.e, exten => ${fromEXTEN}000 |
| 00:02.17 | JASON99 | BZBW: ${CALLERIDNUM} ? |
| 00:02.58 | BZBW | JASON99: you mean this is the extension number that initiate the call? |
| 00:04.02 | JASON99 | BZBW: that will show you the number that is set in the callerid line |
| 00:05.08 | BZBW | exten => _*2, 1, ParkAndAnnounce(pbx-transfer:PARKED|120|SIP/${EXTEN:2}|my_context,${EXTEN:2},1) |
| 00:05.31 | *** join/#asterisk Qwell (n=north@unaffiliated/qwell) |
| 00:05.57 | BZBW | Above is what I'm trying to do, when user press *2, the call will be announced to the caller and park it into a parking lot. |
| 00:06.09 | asterboy | NOTICE[19229]: callerid.c:322 callerid_feed: Caller*ID failed checksum |
| 00:06.37 | asterboy | tried increasing the gains. |
| 00:06.39 | *** join/#asterisk Ariel_ (n=Ariel@dsl-20-177.cofs.net) |
| 00:06.52 | BZBW | and I want to pass the caller extension number to SIP/${EXTEN} |
| 00:06.56 | asterboy | still get this on 1 out of 5 calls approx. |
| 00:07.46 | JASON99 | BZBW: I'm not familiar enough with that to help you.. I'm sorry.. I'm pretty new myself |
| 00:07.50 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
| 00:12.36 | *** join/#asterisk SuperLag (n=aaron@gentoo/developer/SuperLag) |
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| 00:39.17 | GarethTheGreat | getting this when i dial in: |
| 00:39.19 | GarethTheGreat | Asterisk Ready. |
| 00:39.19 | GarethTheGreat | Found route to 213.166.5.130, output from our address 81.174.255.77. |
| 00:39.19 | GarethTheGreat | Check for res for |
| 00:39.20 | GarethTheGreat | <PROTECTED> |
| 00:39.21 | GarethTheGreat | Stopping retransmission on '2707C1F2-F1D011DA-8C97C62B-D7E64309@213.166.5.133' of Response 101: Found |
| 00:48.40 | Jaxxan | GarethTheGreat: you should use pastebin or #flood |
| 00:49.43 | GarethTheGreat | sorry |
| 00:50.24 | brettnem | ~pastebin |
| 00:50.29 | jbot | [pastebin] a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com, or http://bzflag.pastebin.ca/ |
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| 01:03.37 | *** part/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
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| 01:17.36 | Sponge_bob | anyone here use cisco as their voice gateway? |
| 01:19.50 | *** join/#asterisk wunderkin (i=kev@ip68-226-113-228.ph.ph.cox.net) |
| 01:22.34 | *** join/#asterisk naain (n=nain@202.59.90.178) |
| 01:23.25 | *** join/#asterisk mog_home (n=mogorman@68.62.237.103) |
| 01:25.58 | *** join/#asterisk TripleFFFF (n=Miranda@147-102.mc.cite.net) |
| 01:26.14 | TripleFFFF | if my upstream is sending me 1XXXXXXXXXX how can i trim if there and not trim if not ? |
| 01:26.30 | TripleFFFF | i mean .. change the src number directly |
| 01:29.08 | JASON99 | I would do something like this.. Not sure if there is a better way.. |
| 01:29.09 | JASON99 | [context] |
| 01:29.09 | JASON99 | exten => _1XXXXXXXXXX,1,Goto(context,${EXTEN:1},1) |
| 01:29.09 | JASON99 | exten => _XXXXXXXXXX,1,DoWhatever |
| 01:30.27 | TripleFFFF | trying |
| 01:30.48 | JASON99 | If it matchs a 1 it will resend it to the context without the 1 |
| 01:31.08 | JASON99 | matches rather |
| 01:32.02 | TripleFFFF | not too good yet |
| 01:32.23 | TripleFFFF | had _ instead of - |
| 01:32.25 | TripleFFFF | lol works |
| 01:32.45 | TripleFFFF | i hear weird clicks when i call it lol |
| 01:32.47 | TripleFFFF | weird |
| 01:32.58 | TripleFFFF | like .. clack .. clack.then all ok |
| 01:33.03 | *** join/#asterisk iq|mobile (n=iq@71-215-55-11.omah.qwest.net) |
| 01:33.20 | TripleFFFF | weird..but hey got quebec numbers nowlol |
| 01:34.43 | GarethTheGreat | i just setup my first IVR |
| 01:34.45 | GarethTheGreat | woohoo |
| 01:36.39 | JASON99 | TrippleFFFF do you want other canadian numbers |
| 01:36.52 | Jaxxan | hrm |
| 01:36.55 | JASON99 | hehe |
| 01:37.09 | Jaxxan | one-touch pause/unpause for agents in SVN TRUNK |
| 01:37.14 | Jaxxan | well that's kewl |
| 01:37.57 | TripleFFFF | lol |
| 01:38.00 | TripleFFFF | how much |
| 01:38.01 | TripleFFFF | ;) |
| 01:40.14 | *** join/#asterisk bjohnson (n=bjohnson@i216-58-92-11.cybersurf.com) |
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| 01:49.47 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 01:50.29 | *** join/#asterisk Dr-Linux (n=Linux@202.59.73.131) |
| 01:52.36 | *** join/#asterisk Assid (n=assid@203.115.83.214) |
| 01:55.07 | Dr-Linux | yo |
| 01:55.14 | naain | Hi Can any one explain me why asterisk is picking up dynamic SIP port other then 5060????? |
| 01:55.15 | Dr-Linux | howdy |
| 01:55.17 | Dr-Linux | salaam |
| 01:55.44 | Dr-Linux | naain: whats in your sip.conf? |
| 01:57.01 | Jaxxan | is there a variable that will return your agent ID ? |
| 01:57.07 | Jaxxan | ie: Agent/1000 |
| 01:57.10 | naain | Dr-Linux>: [general] |
| 01:57.10 | naain | context=default |
| 01:57.10 | naain | bindaddr=0.0.0.0 |
| 01:57.10 | naain | bindport=5060 |
| 01:57.38 | Dr-Linux | naain: what you are using bindport? |
| 01:57.52 | naain | Dr-Linux:>i have explicitly define 5060 port in softphone at client end but asterisk is picking up dynamic port. |
| 01:58.02 | naain | Dr-Linux>:yes bindport |
| 01:58.44 | naain | Dr-Linux: is there any thing wrong with bindport? |
| 01:59.32 | Dr-Linux | naain: how you know that asterisk is picking up dynamic ports other than 5060? |
| 02:00.09 | naain | Dr-Linux>: sip show peers |
| 02:00.28 | Dr-Linux | naain: awww |
| 02:00.29 | naain | Dr-Linux>: even in Diagnostic log of Sip phone |
| 02:00.37 | Dr-Linux | that's not sip ports man |
| 02:00.41 | Dr-Linux | that's source ports |
| 02:01.03 | Dr-Linux | naain: what port you can see when you do "sip show status" ? |
| 02:04.17 | Dr-Linux | naain: what sip client you are using? |
| 02:04.52 | naain | Dr-Linux>:Eyebeam, Bosoft etc... |
| 02:05.28 | Dr-Linux | naain: what sip port you see when you do "sip show status"? |
| 02:05.38 | naain | Dr-Linux>: I didn't see any command in help "sip show status"? |
| 02:05.53 | naain | No such command 'sip show status' (type 'help' for help) |
| 02:07.03 | Dr-Linux | naain: hhm.. sorry my box is not infront of me, so i don't remember the exact command |
| 02:07.17 | Dr-Linux | but you are wrong |
| 02:07.21 | Dr-Linux | these are not sip ports |
| 02:07.30 | Dr-Linux | those are source ports from client |
| 02:09.02 | naain | Dr-Linux>: but if these are the source port then why some time it binds to the particular port or if even if 5060 port is block by ISP then why it client wont register to asterisk on 5060 port |
| 02:12.14 | Dr-Linux | naain: where from you? |
| 02:12.27 | naain | Dr-Linux>: When i did "sip show settings" it shows me "SIP Port: 5060" |
| 02:12.40 | Dr-Linux | naain: bcoz server sip port 5060 |
| 02:12.43 | Dr-Linux | correct |
| 02:12.51 | Dr-Linux | so your sip port is 5060 |
| 02:13.12 | Dr-Linux | naain: whre from you, if you ISP is blocking SIP port? |
| 02:13.17 | naain | Dr-Linux>: Yes sip port is 5060 but sip show peers or sip log shows dynamic port connecting |
| 02:13.29 | naain | Dr-Linux>: Pakistan |
| 02:13.55 | markus99 | I have an issue with a sip termination account not passing audio (neither party can hear the other) with no asterisk errors when a call is placed from a sip device, any ideas? |
| 02:14.16 | Dr-Linux | naain: where in Pakistan? |
| 02:15.01 | Dr-Linux | naain: most of Paki ISP do not allow sip port |
| 02:15.09 | Dr-Linux | naain: but you can use any other port for SIP |
| 02:15.26 | Dr-Linux | naain: or better use 3 ports for SIP ;) |
| 02:15.39 | Dr-Linux | naain: kia samjhey ;) |
| 02:16.11 | naain | Dr-Linux:> My 5060 port is open, i have tried it by binding other port as well and it works, but the thing is that the port on which i bind to sip client some time it works and some time it pick up the dynamic port |
| 02:16.45 | naain | Dr-Linux>: exactly wahi jo aap samjha, |
| 02:16.48 | naain | :) |
| 02:16.58 | Dr-Linux | hhmm.. |
| 02:17.05 | naain | Dr-Linux>: 3 ports sa kia muraad ? |
| 02:17.21 | Dr-Linux | naain: i mean 2 ports for sip |
| 02:18.18 | Dr-Linux | naain: well, on client side use 8080 for sip and on the server redirect 8080 port to 5060 and fuck PTCL ;) |
| 02:18.43 | Dr-Linux | naain: you purchase eyeBeam or cracked version? :S |
| 02:18.48 | Jaxxan | http://bugs.digium.com/view.php?id=5531 should be in SVN Trunk right ? |
| 02:19.02 | Jaxxan | like, i should have it available to me right? |
| 02:19.26 | naain | Dr-Linux>: I still didn't got your point, I can bind port other then 5060, and i have done it and it's working but this is not my problem, the only thing is that why asterisk is not showing port that cilent is connecting to 8060, some time it connects to 8060 but some time dyanmic port like 63831 etc... |
| 02:20.00 | *** join/#asterisk joat (n=joat@ip70-160-147-169.hr.hr.cox.net) |
| 02:20.09 | naain | Dr-Linux>: i have even tried X-Lite with same issue.... |
| 02:20.33 | Dr-Linux | naain: that's not an issue dude |
| 02:20.37 | naain | and even free ware sip dialer ... |
| 02:21.29 | naain | Dr-linux>: then what u suggest |
| 02:21.39 | Dr-Linux | naain: your soft clients are not registering with the server? |
| 02:21.57 | Dr-Linux | naain: city? |
| 02:22.01 | naain | Dr-Linux>: Successfully Registering with server and i can make call and recieve as well |
| 02:22.17 | Dr-Linux | naain: so what? |
| 02:22.17 | naain | Dr-Linux>: lhr |
| 02:22.23 | Dr-Linux | what's your problem? |
| 02:22.29 | Dr-Linux | naain: where in lhr? |
| 02:22.47 | naain | Dr-Linux>: Actually i am facing voice breakage issue while i have enough bandwidth........ |
| 02:23.12 | Dr-Linux | naain: so that's not ports issue |
| 02:23.16 | naain | I noticed that client is binding to different port other then i specified althought it can't be integrated with voice break but just for info |
| 02:23.28 | Dr-Linux | naain: you are using any cards? |
| 02:23.36 | naain | Dr-Linux> No dear |
| 02:23.56 | naain | Dr-Linux>: From where do you belong in PK? |
| 02:24.10 | Dr-Linux | tribal |
| 02:24.41 | Dr-Linux | naain: check your client side setting. |
| 02:25.15 | naain | Dr-Linux>: for example what to check, I have override Sip Listen port, Outbound Port, and even proxy with port binding |
| 02:25.48 | Dr-Linux | naain: nope, just mic/volume setting etc |
| 02:26.00 | Dr-Linux | forget about ports |
| 02:26.08 | Dr-Linux | naain: now adays i'm in lhr |
| 02:26.14 | Dr-Linux | don't ask where :P |
| 02:26.16 | naain | Dr-Linux>: for satisfication i have retune the client for voice but same results |
| 02:26.35 | naain | Dr-Linux>: Good to know |
| 02:27.07 | naain | Dr-Linux>: What are you doing here any special task.... |
| 02:27.54 | Dr-Linux | naain: no way, just walking around ;) |
| 02:28.05 | techman97_andy | evenin' all - anyone worked with the * Manager API using C# or VB.NET? |
| 02:30.01 | *** join/#asterisk Eight (n=blake@12-227-169-99.client.mchsi.com) |
| 02:37.55 | Jaxxan | hrm |
| 02:37.59 | Jaxxan | dlynes_home: |
| 02:38.07 | Jaxxan | dlynes_office: you there ? |
| 02:42.44 | Jaxxan | how do i add a custom function ? |
| 02:44.35 | *** part/#asterisk cfassoni (n=cfassoni@c911444e.rjo.virtua.com.br) |
| 02:49.46 | *** join/#asterisk kaz0358 (n=kurt@asterisk.telecom.ksu.edu) |
| 02:54.31 | *** join/#asterisk e-Hernick (n=vx@modemcable222.104-130-66.mc.videotron.ca) |
| 02:54.59 | *** part/#asterisk e-Hernick (n=vx@modemcable222.104-130-66.mc.videotron.ca) |
| 02:55.09 | kaz0358 | has anyone ever gotten back a 404 bad request on an invite? The message says "Malformed/Missing Contact field". i have enabled sip debugging and i have noticed something stange. when my asterisk box is doing an invite it is putting From: "callerid name" <sip:extension@asterisk-dns@asterisk-ip> ... that sip URI does not look avalid to me..is it? |
| 02:55.26 | dlynes_home | Jaxxan: ? |
| 02:55.27 | kaz0358 | i have googled around, but there is almost nothing on that weird 404 message |
| 02:56.11 | Jaxxan | hey man |
| 02:56.16 | dlynes_home | Jaxxan: no idea what a custom function is :) |
| 02:56.22 | Jaxxan | http://bugs.digium.com/view.php?id=5531 |
| 02:56.28 | Jaxxan | i want to install that patch |
| 02:56.33 | Jaxxan | which adds a function |
| 02:56.33 | dlynes_home | Jaxxan: or you mean like your own dialplan application? |
| 02:56.43 | Jaxxan | i'm just not sure how to do it |
| 02:57.19 | dlynes_home | yeah..i'm bringing it up |
| 02:57.37 | dlynes_home | it might not be a patch against the latest stable...usually patches are applied against trunk |
| 02:57.52 | Jaxxan | hrm |
| 02:59.29 | Jaxxan | ok |
| 02:59.29 | Jaxxan | so i use agents in my queue |
| 02:59.43 | Jaxxan | and let's say for example, i do an agentcallbacklogin and my agent ID is 1000 |
| 03:00.09 | Jaxxan | within my dialplan, i want to use pausequeuemember/unpausequeuemember, but i have to know what the agent ID is. |
| 03:00.39 | dlynes_home | Jaxxan: it's in the current trunk |
| 03:00.41 | Jaxxan | how can i retrieve the Agent ID based on callerid to insert into pause/unpausequeuemember applications |
| 03:00.55 | dlynes_home | Jaxxan: i haven't checked to see if it's in 1.2.8 or not |
| 03:01.05 | kaz0358 | anyone have problems making a call from asterisk to a Cisco SIPGateway? |
| 03:01.07 | dlynes_home | Jaxxan: but it should be in 1.4 for sure |
| 03:01.11 | Jaxxan | i'm using 1.2.6 |
| 03:01.19 | dlynes_home | Jaxxan: it's not in 1.2.7.1 |
| 03:01.35 | Jaxxan | this is a production box |
| 03:01.49 | GarethTheGreat | http://pastebin.com/754872 |
| 03:02.06 | GarethTheGreat | in mainmenu here, i have a few SIP calls |
| 03:02.30 | GarethTheGreat | i want them to actually dial the same SIP address but somehow make the person picking up aware of what the nature of the call is |
| 03:02.37 | dlynes_home | Jaxxan: i'm downloading 1.2.8 to see if it's there or not, too |
| 03:02.52 | dlynes_home | Jaxxan: but normally, you would go into your src directory |
| 03:03.12 | GarethTheGreat | what's the best way to implement this? |
| 03:03.12 | dlynes_home | Jaxxan: and then type patch < filename.patch |
| 03:03.12 | dlynes_home | Jaxxan: then remake and reinstall |
| 03:03.24 | Jaxxan | ok |
| 03:03.47 | dlynes_home | GarethTheGreat: modify callerid(num) to reflect the nature of the call |
| 03:04.02 | Jaxxan | so you're saying that agent function is in SVN ? |
| 03:04.02 | Jaxxan | erm... Trunk |
| 03:04.10 | dlynes_home | Jaxxan: correct |
| 03:04.17 | Jaxxan | hrm |
| 03:04.20 | GarethTheGreat | dlynes_home: how does one do that? |
| 03:04.25 | dlynes_home | Jaxxan: but it's also in that patch file that you see on the bugs.digium.com page |
| 03:04.46 | Jaxxan | so i can just copy that patch file and it should work against my 1.2.6 ? |
| 03:05.01 | dlynes_home | Jaxxan: did I say that? I did not. |
| 03:05.11 | Jaxxan | i didn't think you did (= |
| 03:05.19 | dlynes_home | Jaxxan: I said _____try_____ applying that patch against 1.2.6 |
| 03:05.25 | dlynes_home | Jaxxan: i did not say it would work |
| 03:05.39 | Jaxxan | worst case scenario, what happens if it doesn't work? |
| 03:05.40 | dlynes_home | Jaxxan: it's written against trunk, not any particular stable release |
| 03:05.54 | dlynes_home | Jaxxan: just untar your source files again, and your source code directory is back to normal |
| 03:06.02 | Jaxxan | alrighty |
| 03:06.22 | Jaxxan | can you lemme know if it's in 1.2.8? |
| 03:06.49 | Jaxxan | if it is i'll just upgrade to that |
| 03:06.55 | dlynes_home | Jaxxan: it's not |
| 03:07.02 | dlynes_home | Jaxxan: guess you're going to have to wait for 1.4 |
| 03:07.04 | Jaxxan | k |
| 03:07.31 | GarethTheGreat | can SetCallerID take a string? |
| 03:07.32 | dlynes_home | Jaxxan: btw...1.2.7.1 is considerably more stable than 1.2.6 |
| 03:07.40 | dlynes_home | GarethTheGreat: that's asterisk 1.0 |
| 03:07.50 | Jaxxan | ok |
| 03:07.56 | dlynes_home | GarethTheGreat: Use Set(CALLERID(number)=xxxxxx) |
| 03:08.05 | dlynes_home | GarethTheGreat: or Set(CALLERID(name)=xxxxxx) |
| 03:08.35 | dlynes_home | GarethTheGreat: To get the caller id, you can do Set(mystring=CALLERID(number)) |
| 03:08.56 | dlynes_home | GarethTheGreat: num will also work if you're too lazy to type out number |
| 03:08.58 | Jaxxan | what do you think of http://bugs.digium.com/view.php?id=6650&nbn=10 |
| 03:09.08 | *** join/#asterisk anonymouz666 (n=anonymou@200.218.193.6) |
| 03:09.14 | anonymouz666 | hello all |
| 03:09.17 | anonymouz666 | I am drunked |
| 03:09.22 | anonymouz666 | I love Johnnie Walker |
| 03:09.45 | GarethTheGreat | exten => 2,1,Set(CALLERID(name)=parent) |
| 03:09.47 | GarethTheGreat | is that valid? |
| 03:09.54 | dlynes_home | GarethTheGreat: yes |
| 03:09.58 | GarethTheGreat | woohoo |
| 03:10.01 | anonymouz666 | kram is got rich |
| 03:10.13 | anonymouz666 | he doesnt not access anymore? |
| 03:12.02 | GarethTheGreat | next part - how do i make it dial whoever answers from a pool of SIP or PSTN users? |
| 03:12.41 | *** join/#asterisk coppice (n=chatzill@84.157.17.210.dyn.pacific.net.hk) |
| 03:13.11 | Jaxxan | so trunk just has all the kewl stuff that doesn't quite make it to stable right ? |
| 03:13.36 | GarethTheGreat | dlynes_home? |
| 03:14.20 | dlynes_home | GarethTheGreat: how many would be in the pool? |
| 03:14.24 | dlynes_home | GarethTheGreat: and how big is the office? |
| 03:14.27 | Jaxxan | grabbing trunk |
| 03:14.31 | dlynes_home | GarethTheGreat: and how many pools? |
| 03:14.32 | Jaxxan | i think i'm gonna try it out this weekend |
| 03:14.46 | dlynes_home | Jaxxan: that issue was closed, but I can't figure out why |
| 03:15.08 | dlynes_home | Jaxxan: so, I wouldn't recommend using it, for fear that it might not be stable |
| 03:15.15 | Jaxxan | one touch pause/unpause seems kewl though (= |
| 03:15.22 | dlynes_home | Jaxxan: doesn't seem to be any review of the code in question |
| 03:15.23 | GarethTheGreat | dlynes_home: not a physical office, but there'd be 3 pools |
| 03:15.38 | justinu | are those pools heated? |
| 03:15.51 | GarethTheGreat | well, actually would be more efficient to have 1 large pool and get volunteers to handle all 3 types of call |
| 03:15.53 | dlynes_home | GarethTheGreat: are these three pools for three small offices, or something? |
| 03:15.58 | GarethTheGreat | no |
| 03:16.06 | GarethTheGreat | let's just call it one large pool |
| 03:16.07 | dlynes_home | GarethTheGreat: so it's a call center then? |
| 03:16.11 | GarethTheGreat | yes |
| 03:16.17 | dlynes_home | GarethTheGreat: then you want to look at agents and queues |
| 03:16.56 | GarethTheGreat | when going through to an agent can the agent still see the CallerID name? |
| 03:17.15 | dlynes_home | Jaxxan: yeah...I wouldn't know if it was cool or not...I know nothing about the needs of a call center :) |
| 03:17.21 | dlynes_home | GarethTheGreat: yes, afaik |
| 03:17.31 | dlynes_home | GarethTheGreat: jaxxan might be able to answer that question better than me though |
| 03:20.04 | dlynes_home | Dr-Linux: you there? |
| 03:20.30 | GarethTheGreat | feel like a n00b asking this but can multiple calls actually come in from the voip-user.org number? |
| 03:20.45 | dlynes_home | voip-user.org? |
| 03:20.56 | GarethTheGreat | gives free PSTN numbers with SIP |
| 03:21.15 | dlynes_home | ah...that would depend entirely on whether they allow it or not |
| 03:21.24 | GarethTheGreat | http://www.voipuser.org/mynumbers.html |
| 03:21.34 | dlynes_home | but yes, voip trunks are capable of sending many simultaneous calls |
| 03:21.55 | GarethTheGreat | so it is technically possible but depends on whether they allow it? |
| 03:21.57 | Jaxxan | gareth |
| 03:22.08 | GarethTheGreat | Jaxxan? |
| 03:22.46 | Jaxxan | lemme catch up real quick. you're setting up a call center, and want calls to rollover to overflow groups ? |
| 03:22.59 | dlynes_home | GarethTheGreat: correct |
| 03:23.21 | GarethTheGreat | Jaxxan: i'm setting up a virtual call center of sorts |
| 03:23.29 | Jaxxan | are you using queues ? |
| 03:23.34 | GarethTheGreat | not yet |
| 03:23.55 | GarethTheGreat | so far i've got a basic menu that prompts for the purpose of the call and sets the caller id name to reflect it |
| 03:23.55 | Jaxxan | how many people you have in this virtual call center ? |
| 03:24.06 | Jaxxan | ok |
| 03:24.12 | GarethTheGreat | well, it's still under development so none as of yet |
| 03:24.18 | Jaxxan | lemme give you a quick rundown of how i handle a call center |
| 03:24.21 | GarethTheGreat | will probably have around 10 people max |
| 03:24.45 | Jaxxan | i have a 24/7 call center with about 4 agents logged in at a time that handle about 1000+ calls a day |
| 03:25.19 | GarethTheGreat | doubt i'll get that amount of traffic |
| 03:25.27 | Jaxxan | i have one queue. and multiple types of calls (ie: customer care, directory, operator calls, etc...) |
| 03:26.03 | Jaxxan | each call that i dump into a queue, i tag their calleridname to reflect the type of call (ie: customercare, directory, operator, etc...) |
| 03:26.11 | GarethTheGreat | similar to my idea |
| 03:26.27 | Jaxxan | i also set that calls accountcode for easy CDR parsing |
| 03:26.39 | GarethTheGreat | err, CDR? |
| 03:26.46 | GarethTheGreat | accountcode? |
| 03:26.49 | Jaxxan | CDR= Call Detail Record |
| 03:27.15 | Jaxxan | so i can go back and bill, or just see how many calls were handled in a timeframe etc... |
| 03:27.42 | GarethTheGreat | not required for basic functionality though, right? |
| 03:27.43 | Jaxxan | i use agentcallbacklogin application for my agents. |
| 03:27.47 | Jaxxan | not required no |
| 03:27.48 | TheCops | Someone know an easy way to make a load test with asterisk ? |
| 03:27.57 | dlynes_home | TheCops: sipx |
| 03:28.03 | dlynes_home | TheCops: erm sipp i mean |
| 03:28.12 | Jaxxan | it works out pretty decent for my callcenter |
| 03:28.30 | TheCops | dlynes_home, I read on that, seem to be powerful, do you have an xml example ?! |
| 03:28.36 | naain | Jaxxan: Hi |
| 03:28.38 | Jaxxan | i also use queuemetrics to analyze the call center so the manager knows what's going on |
| 03:28.47 | dlynes_home | TheCops: nope...haven't had time to shit, much less try sipp :) |
| 03:28.50 | Jaxxan | but that's probably outside of what you're trying to do with a basic call center |
| 03:28.57 | GarethTheGreat | i just need something extremely basic |
| 03:29.15 | GarethTheGreat | going to all be run by volunteers |
| 03:30.04 | Jaxxan | GarethTheGreat: what you want is http://www.voip-info.org/wiki/view/Asterisk+call+queues |
| 03:30.17 | GarethTheGreat | so, do all agents just get shoved into a context where AgentLogin() is done? |
| 03:30.26 | GarethTheGreat | i.e a different context for each agent and then: |
| 03:30.42 | GarethTheGreat | exten => s,1,AgentLogin(whatever) |
| 03:30.46 | GarethTheGreat | that correct? |
| 03:30.46 | Jaxxan | all the agents get added to my single queue |
| 03:30.56 | Jaxxan | agentcallbacklogin is what i use |
| 03:31.07 | Jaxxan | go to that web page i just linked and read man |
| 03:31.17 | Jaxxan | everything you want to know is there |
| 03:31.17 | GarethTheGreat | reading |
| 03:31.21 | GarethTheGreat | i'll get playing with this |
| 03:31.25 | GarethTheGreat | thanks for your help |
| 03:31.25 | Jaxxan | naain: sup ? |
| 03:31.26 | naain | Jaxxan: For inbound QueueMatric and agents application work fine. How can we utilize it for Outbound Call Center |
| 03:32.07 | Jaxxan | naain: that's a good question, my call center manager just asked me if they could monitor outbound calls for our collections department. |
| 03:32.33 | Jaxxan | to be honest, i dont know at this time and haven't tackled it yet. |
| 03:32.55 | Jaxxan | also, i dont really plan to work on it either (= |
| 03:33.05 | Jaxxan | but... |
| 03:33.16 | Jaxxan | i know queuemetrics has that oubound feature |
| 03:33.26 | Jaxxan | so it must be doable |
| 03:34.06 | *** join/#asterisk P-NuT (n=P-Nut@CPE-60-225-220-3.nsw.bigpond.net.au) |
| 03:34.09 | P-NuT | Hi all, |
| 03:34.22 | GarethTheGreat | any way for agents to put callers on hold? |
| 03:34.45 | Jaxxan | GarethTheGreat: yeah, they press the hold button on their phone |
| 03:35.19 | P-NuT | I've installed asterisk and zaptel from source and it can't seem to find my x100p card. I heard something about udev stuffing it up? Apparently when I run ztcfg it says line 0: Unable to open master device '/dev/zap/ctl' |
| 03:35.21 | GarethTheGreat | but will that play hold music? |
| 03:35.29 | coppice | come on. the very first thing you develop in any telephone system is caller on hold, with really bad music :-) |
| 03:35.32 | GarethTheGreat | it's client-side isn't it? |
| 03:35.33 | Jaxxan | GarethTheGreat: yes |
| 03:35.45 | Jaxxan | GarethTheGreat: no |
| 03:35.56 | GarethTheGreat | that's cool |
| 03:37.45 | GarethTheGreat | ok, so to transfer my inbound calls to a queue do i just do Queue(whatever) ? |
| 03:39.03 | Jaxxan | yes |
| 03:39.33 | GarethTheGreat | http://pastebin.com/754903 |
| 03:39.38 | GarethTheGreat | so, this will work then? |
| 03:40.37 | Jaxxan | pretty much |
| 03:40.37 | Jaxxan | but, might i suggest a macro |
| 03:40.40 | Jaxxan | let me show you an example |
| 03:41.54 | dlynes_home | P-NuT: are you sharing interrupts? |
| 03:42.36 | P-NuT | umm.. |
| 03:42.39 | P-NuT | no? |
| 03:42.49 | dlynes_home | P-NuT: so you don't know? |
| 03:42.56 | P-NuT | yeah. |
| 03:42.57 | GarethTheGreat | [amy] |
| 03:42.58 | GarethTheGreat | exten => s,1,AgentLogin(1001) |
| 03:42.58 | GarethTheGreat | exten => s,2,Hangup |
| 03:43.03 | GarethTheGreat | is that correct? |
| 03:43.27 | dlynes_home | P-NuT: lspci -v | grep IRQ |
| 03:43.37 | Jaxxan | yeah, dont use agentlogin though |
| 03:43.42 | Jaxxan | use agentcallbacklogin |
| 03:43.42 | dlynes_home | P-NuT: do you see any IRQ's that have the same number there? |
| 03:43.55 | Jaxxan | unless you want to listen to holdmusic for hours on end while waiting for calls |
| 03:44.18 | *** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239) |
| 03:44.33 | GarethTheGreat | bah, i don't mind |
| 03:46.35 | Jaxxan | GarethTheGreat: This is kind of what i do http://pastebin.com/754913 |
| 03:46.43 | P-NuT | I have this.. |
| 03:46.49 | P-NuT | oh hang on.. |
| 03:46.53 | P-NuT | pastebin |
| 03:47.36 | *** part/#asterisk anonymouz666 (n=anonymou@200.218.193.6) |
| 03:47.38 | P-NuT | http://pastebin.ca/61256 |
| 03:47.43 | Jaxxan | i call other macro's within my version for special emergencies and stuff, but in a nutshell that's an easier way to handle your call center |
| 03:48.12 | P-NuT | does this make any sense to you>? |
| 03:48.27 | kaz0358 | btw, i figured out my problem with the invalid SIP URI from field being invalid on an INVITE. the short answer is do not put an "at" symbol in your callerid. i was following the ISN cookbook. hopefully they will correct it soon after i email them. |
| 03:48.27 | P-NuT | oooh |
| 03:48.32 | P-NuT | I think I have to go. |
| 03:48.38 | P-NuT | I'll come back and ask |
| 03:48.39 | dlynes_home | P-NuT: as you can see you've got two shared interrupts |
| 03:48.42 | P-NuT | oh ok. |
| 03:48.47 | P-NuT | I'll work from there then |
| 03:48.48 | dlynes_home | P-NuT: i'm guessing that's your problem right there |
| 03:48.49 | P-NuT | thanks |
| 03:50.08 | Jaxxan | GarethTheGreat: if for whatever reason, there aren't any agents logged into my queue, and a call comes in destined for that queue, the call will continue to 8 and dial every single phone in my office |
| 03:50.14 | *** join/#asterisk postel_ (n=jp@unaffiliated/postel) |
| 03:50.26 | Jaxxan | so i can ensure that someone answers the call |
| 03:51.00 | Jaxxan | also, the SetCIDName(blah) is deprecated |
| 03:51.06 | Jaxxan | dont use that |
| 03:51.16 | *** join/#asterisk bmg505 (n=leon@c1-28-1.rndf.isadsl.co.za) |
| 03:51.28 | *** join/#asterisk asteriskwannabe (n=fholmes@cpe-72-177-253-50.houston.res.rr.com) |
| 03:51.36 | Jaxxan | although it still works (= |
| 03:53.51 | *** join/#asterisk hayburn (i=hayburn@concorde.hayburn.net) |
| 03:55.32 | GarethTheGreat | having trouble getting an agent login |
| 03:58.03 | GarethTheGreat | login and get no hold music |
| 03:58.55 | *** join/#asterisk digdug (n=dam@emperorzurg.infowest.com) |
| 03:59.10 | Jaxxan | hold music is defined in your zapata.conf |
| 03:59.20 | Jaxxan | and your musiconhold.conf |
| 03:59.27 | digdug | are there any software sip clients to which you can attach a file (image or pdf, or something) to fax? |
| 04:00.51 | Jaxxan | hrm |
| 04:01.51 | Jaxxan | hey dlynes_home, do you think it's possible to have an application modify the light on my cisco 7960 ? |
| 04:02.09 | Jaxxan | like, it lights up when i have voicemail and when my phone is ringing |
| 04:02.35 | dlynes_home | Jaxxan: you mean make it green instead of purple? |
| 04:02.43 | Jaxxan | i need some sort of visual indicator for my agents as to whether they're logged into the queue or not |
| 04:02.51 | GarethTheGreat | after logging it hangs up within a few seconds |
| 04:02.55 | Jaxxan | and none of their phones have voicemail attached to it |
| 04:03.58 | dlynes_home | Jaxxan: do those phones have buddy lights? |
| 04:04.13 | Jaxxan | it has a red light on the handset when it's in the cradle |
| 04:04.26 | *** part/#asterisk digdug (n=dam@emperorzurg.infowest.com) |
| 04:04.28 | Jaxxan | i dunno what you mean by a buddy light |
| 04:05.48 | dlynes_home | buddy light is a light that lights up when a particular user is on the phone |
| 04:05.59 | dlynes_home | so that you can see the status of the other extensions in the office |
| 04:06.00 | Jaxxan | oh no |
| 04:06.34 | Jaxxan | my old legacy pbx handsets had that functionality though |
| 04:06.54 | Jaxxan | no one in my call center cares about that, they'll go right on talking anyways (= |
| 04:07.35 | dlynes_home | Jaxxan: they can't tell if they're logged into the queue or not? |
| 04:07.48 | dlynes_home | Jaxxan: i would think if they were, their phone would ring |
| 04:07.49 | dlynes_home | no? |
| 04:07.53 | *** join/#asterisk neillt (n=neillt@cpe-24-165-2-22.san.res.rr.com) |
| 04:08.16 | Jaxxan | that's not what i mean, every once in a while, i get a user that *thinks* they logged into the queue, but didn't. |
| 04:08.35 | dlynes_home | Jaxxan: ah |
| 04:08.43 | dlynes_home | Jaxxan: anyways...one way of doing it |
| 04:08.58 | dlynes_home | Jaxxan: is assign that user a mailbox |
| 04:09.19 | dlynes_home | Jaxxan: and whenever they're logged into the queue, dump a voicemail with a corresponding txt file into their inbox directory |
| 04:09.33 | Jaxxan | hey that's doable |
| 04:09.36 | dlynes_home | Jaxxan: when they log out, move the voicemail and the txt file out of their inbox |
| 04:09.53 | dlynes_home | Jaxxan: you could probably achieve something like that with an agi script |
| 04:10.36 | Jaxxan | i'll look into that next week sometime |
| 04:10.48 | Jaxxan | for now, i'm going to go home |
| 04:10.54 | Jaxxan | have a great weekend everyone |
| 04:10.59 | [TK]D-Fender | That idea is way to flakey |
| 04:11.14 | Jaxxan | [TK]D-Fender: it would totally work though |
| 04:11.24 | [TK]D-Fender | they leave their desk for 2 mins, someone calls them and leaves a message direct and wham, they think they're logged in... |
| 04:11.38 | dlynes_home | [TK]D-Fender: not if you don't forward to voicemail |
| 04:11.43 | dlynes_home | [TK]D-Fender: he said he doesn't use voicemail |
| 04:11.55 | Jaxxan | as long as i dont add voicemail to their dialplan it wont be an issue |
| 04:12.01 | [TK]D-Fender | yes it can work, but it unreliable... what are they going to do? Leave the VM in there? Constanlt look in a secondary box not knowing if the missed one? |
| 04:12.18 | dlynes_home | [TK]D-Fender: listen |
| 04:12.19 | [TK]D-Fender | You're talking Cisco.... for crying out loud use the XML browser! |
| 04:12.22 | Jaxxan | my call center people dont use voicemail |
| 04:12.22 | dlynes_home | [TK]D-Fender: he's not using voicemail |
| 04:12.32 | dlynes_home | [TK]D-Fender: well, i know nothing about cisco, either :) |
| 04:12.45 | Jaxxan | i have 6 phones dedicated to the call center that aren't attached to voicemail |
| 04:12.59 | Jaxxan | you have something else in mind [TK]D-Fender ? |
| 04:13.32 | Jaxxan | oi! |
| 04:14.04 | Jaxxan | XML yeah, but i'm not sure if i can turn the red light on and off with that |
| 04:15.03 | Jaxxan | anyways, i'm taking off, ttyl |
| 04:16.34 | GarethTheGreat | hmm, found why agent login wasn't working |
| 04:16.38 | GarethTheGreat | DTMF not recogonised |
| 04:22.52 | GarethTheGreat | Inband DTMF is not supported on codec gsm. Use RFC2833 |
| 04:22.56 | GarethTheGreat | get flooded with that |
| 04:23.04 | GarethTheGreat | when i try rfc2833 it still doesn't respond |
| 04:23.08 | GarethTheGreat | responds fine with dialins |
| 04:29.35 | *** join/#asterisk viler (i=1000@200.114.70.228) |
| 04:29.37 | *** join/#asterisk eipi (n=eipi@139-213-126-200.fibertel.com.ar) |
| 04:45.14 | jarrod | any reason why reinvite with asterisk on sip would cause voice quality to sound like crap when the rtp stream is redirected to the pstn gateway |
| 05:01.23 | *** join/#asterisk feld_ (n=feld@12.148.212.157) |
| 05:06.36 | *** join/#asterisk kernel20 (n=kernel20@203.160.223.26) |
| 05:06.39 | kernel20 | hi |
| 05:06.55 | kernel20 | how can i connect my analog phones to asterisk? |
| 05:08.42 | [TK]D-Fender | kernel20 : Buy ATA's. Sipura/Linksys are very affordable and decent |
| 05:10.07 | *** join/#asterisk x86 (n=x86@p3m/member/x86) |
| 05:10.07 | neillt | kernel20: check out http://www.voip-info.org/wiki/view/Analog+Telephone+Adapters |
| 05:10.13 | kernel20 | [TK]D-Fender: our company has and old PBX programmed through KSU, now all telephone modules where tapped at some points on our outlets |
| 05:10.33 | kernel20 | what i want is to retain all the modules |
| 05:10.38 | kernel20 | all the phones |
| 05:10.55 | kernel20 | just by a device that could connect all the phones( around 50+) |
| 05:11.05 | kernel20 | [TK]D-Fender:? |
| 05:11.46 | *** join/#asterisk P-NuT (n=P-Nut@CPE-60-225-220-3.nsw.bigpond.net.au) |
| 05:11.51 | P-NuT | Hi all. |
| 05:12.06 | kernel20 | the card that has fxs, only provides 1 fxs port, so in my understanding only one analog phone can be connected from there |
| 05:12.29 | P-NuT | I have an irq conflict w/ 2 pci devices. how do i resolve it? |
| 05:13.00 | kernel20 | is my understandingh correct? |
| 05:13.02 | neillt | kernel20: you want to retain the KSU modules? |
| 05:13.31 | kernel20 | no the lines |
| 05:13.35 | kernel20 | connected to it |
| 05:13.39 | neillt | ahhhh |
| 05:13.39 | kernel20 | i want to transfer ity |
| 05:13.45 | kernel20 | to asterisk box |
| 05:13.58 | neillt | and the phones are analog? |
| 05:14.01 | kernel20 | yeap |
| 05:14.07 | kernel20 | about 50+ |
| 05:14.10 | neillt | ok... you want what is called a channel bank |
| 05:14.18 | neillt | you will need several for those kinds of numbers |
| 05:14.31 | kernel20 | channel bank? |
| 05:14.33 | neillt | they take your phone lines in, and spit out a digital T1 like |
| 05:14.38 | [TK]D-Fender | kernel20 : What kind of system are you replacing? |
| 05:15.15 | *** join/#asterisk mog_home (n=mogorman@68.62.237.103) |
| 05:15.22 | kernel20 | is there a wiki for that? |
| 05:15.36 | kernel20 | the KSU [TK]D-Fender |
| 05:16.14 | [TK]D-Fender | kernel20 : KSU is like saying its a "car motor". I need to know exactly what KIND it is. Whats the model/manufacturer? |
| 05:16.32 | kernel20 | PANASONIC |
| 05:16.57 | [TK]D-Fender | kernel20 : Those old phones are worthless then. |
| 05:16.57 | neillt | kernel20: http://www.voip-info.org/wiki/view/Asterisk+Channel+Bank |
| 05:17.08 | neillt | [TK]D-Fender: I was afraid of that |
| 05:17.13 | kernel20 | yeah sort of but we want to retain it |
| 05:17.15 | [TK]D-Fender | kernel20 : they cannot be used |
| 05:17.28 | kernel20 | ? |
| 05:17.39 | neillt | kernel20: if they are the panasonic phones with the display then they won't work with anything other than the panasonic system |
| 05:17.56 | [TK]D-Fender | kernel20 : those are digital set for which there is no way to adapt them for use with * |
| 05:18.09 | neillt | actually, they don't even need a display to be the hybrid digital/analog type of phones |
| 05:18.15 | P-NuT | Can anybody help me with my x100p card. |
| 05:18.16 | P-NuT | ? |
| 05:18.30 | [TK]D-Fender | its a Panasonic KSU. 100% useless |
| 05:18.33 | kernel20 | hmm, got confused with u guys |
| 05:19.40 | kernel20 | what is your recommendation [TK]D-Fender? |
| 05:21.03 | *** join/#asterisk feld_ (n=feld@12.148.212.157) |
| 05:21.23 | [TK]D-Fender | kernel20 : Nothing to recommend. Your old equipment is worthless. You're going to have to start from scratch. |
| 05:21.56 | kernel20 | HMMM |
| 05:22.00 | kernel20 | neillt? |
| 05:22.36 | neillt | kernel20: if you have the panasonic hybrid digital/analog telephones, then [TK]D-Fender is 100% percent correct |
| 05:23.06 | neillt | kernel20: if you have "old skool" analog phones (no feature buttons, displays, etc) then those will work |
| 05:23.21 | neillt | but to be honest, you will spend just as much on a good channel bank as new phones |
| 05:28.20 | kernel20 | hmmm |
| 05:29.14 | neillt | kernel20: yeah, it's a bummer, I know |
| 05:31.32 | P-NuT | Guys, when I build the zaptel drivers from source, I use the following commands. make, make install, ....... what else should I do? make devices? |
| 05:32.55 | neillt | P-NuT: Normally you only need make && make install. You may have to modprobe zaptel to get it to load at first. |
| 05:33.14 | [TK]D-Fender | P-NuT : "modprobe zaptel" and whatever module is required for your card, check "cat /proc/interrupts" to ensure your card got its own IRQ. |
| 05:33.47 | [TK]D-Fender | P-NuT : Set upt your zaptel.conf and zapata.conf and the run "ztcfg -vvvv" to ensure that everything looks ok |
| 05:34.03 | P-NuT | ok |
| 05:34.06 | P-NuT | here's the go... |
| 05:34.18 | P-NuT | modprobe zaptel gives me --> FATAL: Module zaptel not found. |
| 05:34.33 | P-NuT | /proc/interupts doesnt list th ecard at all |
| 05:34.50 | [TK]D-Fender | P-NuT : Did you have your kernel sources & headers ready when you compiled? |
| 05:35.01 | P-NuT | um.. |
| 05:35.06 | P-NuT | yes. |
| 05:35.10 | P-NuT | I had, |
| 05:35.22 | P-NuT | kernal-sources |
| 05:35.37 | P-NuT | and the kernel-image (exact server version) |
| 05:35.45 | P-NuT | so yeah |
| 05:35.54 | [TK]D-Fender | P-NuT : What distro? |
| 05:35.56 | P-NuT | I've had this prob on ubuntu b4, |
| 05:36.04 | P-NuT | ubuntu dapper 6.06 |
| 05:36.23 | [TK]D-Fender | P-NuT : and you've installed the tons of app devel stuff you need for *? |
| 05:36.35 | *** join/#asterisk chandi (n=burni13@modemcable237.178-37-24.mc.videotron.ca) |
| 05:36.35 | P-NuT | ummm..... |
| 05:36.39 | P-NuT | like what? |
| 05:37.53 | P-NuT | like dev packages like gcc? |
| 05:37.57 | [TK]D-Fender | P-NuT : GCC, make, bison, and so on.. I forget the list since I only work with general-purpose distro's |
| 05:38.18 | [TK]D-Fender | P-NuT : It's mostly listed on asterisk.org |
| 05:38.34 | P-NuT | yeah I got those.. |
| 05:38.47 | P-NuT | it all compiles ok, |
| 05:39.09 | P-NuT | but zaptel doesnt come up when I do a modprobe |
| 05:39.21 | chandi | hi folks, I've got little questions about the dial command in extensions.conf. I'm trying to make asterisk send dtmf after the called party answers. But Dial(SIP/15143867626@account,130,A(transfer)rD(DTMFTOBESENT)) doesn't work. It never sends the dtmf and never bridges the calls even though it says it does on the console |
| 05:40.09 | P-NuT | [TK]D-Fender: What do you think? |
| 05:40.19 | chandi | it works only when I don't have options A(x.gsm) and r |
| 05:41.58 | P-NuT | Basically, I run ztcfg and it spits out "line 0: Unable to open master device '/dev/zap/ctl'" |
| 05:42.29 | P-NuT | I don't know if it can't find the card bacuse it's sharing an IRQ or what? |
| 05:42.53 | *** join/#asterisk Mother (n=mother@93.Red-80-32-127.staticIP.rima-tde.net) |
| 05:42.59 | [TK]D-Fender | P-NuT : OH... being Ubuntu there is the issue of how you become root to compile and install all this stuff... thats another can of worms I don't know the details around. |
| 05:43.05 | P-NuT | I don't see the card listed when I tail /var/log/messages |
| 05:43.35 | P-NuT | right...... |
| 05:43.38 | P-NuT | so...... |
| 05:43.45 | *** part/#asterisk Mother (n=mother@93.Red-80-32-127.staticIP.rima-tde.net) |
| 05:43.50 | P-NuT | ubuntu is a big no no then.... |
| 05:44.42 | P-NuT | Ok, |
| 05:44.47 | P-NuT | here's another scenario |
| 05:45.00 | P-NuT | what if I did this whole thing on Debian? |
| 05:45.06 | P-NuT | would that be easier? |
| 05:45.48 | [TK]D-Fender | I would definately think so. |
| 05:46.23 | [TK]D-Fender | Just pick something standard. RHEL, CentOS, FC, Debian, Slackware.... |
| 05:46.36 | [TK]D-Fender | SUSE tends to cut a number of things out lately. |
| 05:46.46 | P-NuT | Ubuntu isnt standard? |
| 05:47.00 | P-NuT | I know it's a total shit to compile things on./ |
| 05:47.17 | P-NuT | ok, I'll try Debian Sarge 3.1r2 |
| 05:47.17 | [TK]D-Fender | Not to say you can't get it running on just about anything its a question of inconveniences along the way. |
| 05:47.25 | P-NuT | yeah |
| 05:47.28 | P-NuT | I hear you there. |
| 05:47.38 | [TK]D-Fender | Ubuntu was built to be a desktop distro and run off binaries.... |
| 05:47.39 | P-NuT | I just want it to be clear cut and up and running. |
| 05:47.45 | P-NuT | yeah true |
| 05:47.53 | P-NuT | alright... |
| 05:48.10 | P-NuT | well I'm comfortable with Debian. |
| 05:48.13 | P-NuT | Sound good? |
| 05:48.37 | [TK]D-Fender | For me Slackware has always meant instant success. Debian is really solid. FC can have issues, centOS has 1 or 2 well documented ones so its very suggestable. |
| 05:49.25 | P-NuT | Yeah, I kinda want to intergrate VHCS or ISPConfig with it, so I think Debain would be better for that. |
| 05:50.02 | P-NuT | On the asterisk with x100p front though, do we all concur that it can be done with debian 'sarge'? |
| 05:50.56 | [TK]D-Fender | P-NuT : You could do it on Ubuntu I'm sure... just not HOW :) |
| 05:51.19 | [TK]D-Fender | Debian has always been a solid choice. pick a full install mode and you should be good to go. |
| 05:51.24 | P-NuT | Yeah, I need a clear cut, do this then this, then this. |
| 05:51.36 | P-NuT | not working stuff out for a month completely on my own. |
| 05:51.55 | P-NuT | So if you have heard that it can be done on debian, then I will give that a go. |
| 05:51.57 | h3x0r | the freebsd port rocks |
| 05:52.00 | h3x0r | as long as you dont need t1s |
| 05:52.01 | h3x0r | heh |
| 05:52.55 | h3x0r | wait no |
| 05:53.00 | h3x0r | theres a wct4xxp.ko |
| 05:53.16 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 05:53.22 | [TK]D-Fender | Avoid BSD unless you know what you're doing... |
| 05:53.27 | P-NuT | full install? what do you mean by that? |
| 05:53.43 | [TK]D-Fender | P-NuT : Well dont skimp on "minimal packages". |
| 05:53.55 | h3x0r | its f'n easy |
| 05:53.56 | h3x0r | heh |
| 05:54.06 | P-NuT | I have a netinstall CD |
| 05:54.22 | h3x0r | freebsd is easier than debian or fedora |
| 05:54.32 | P-NuT | no it's not. |
| 05:54.42 | P-NuT | anyway, |
| 05:54.51 | h3x0r | "make install" |
| 05:54.51 | h3x0r | heh |
| 05:54.55 | P-NuT | I'll give it a go with debian |
| 05:55.03 | h3x0r | its like gentoos instructions with asterisk |
| 05:56.25 | P-NuT | apart from these packages, what else do I need? --> ssh subversion linux-source-2.6.15 linux-headers-2.6.15-23-server libncurses5 libncurses5-dev openssl libssl0.9.8 libssl-dev bison make gcc |
| 05:56.43 | *** join/#asterisk jsaunders (i=jsaunder@S01060060971c5817.va.shawcable.net) |
| 05:56.57 | [TK]D-Fender | P-NuT : that looks good. no need for subversion really, but why not... |
| 05:57.10 | [TK]D-Fender | I always suggest download from FTP for the release versions. |
| 05:57.22 | P-NuT | well, the sources you cant get via cvs anymore |
| 05:58.08 | [TK]D-Fender | FTP <- |
| 05:58.15 | [TK]D-Fender | direct linked off asterisk.org |
| 05:58.31 | justinu | that's a good suggestion |
| 05:58.49 | P-NuT | well, I'll be back |
| 05:58.55 | P-NuT | thanks for your help |
| 05:59.03 | [TK]D-Fender | np |
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| 06:13.56 | stephane_ | jour |
| 06:18.57 | [TK]D-Fender | soir :) |
| 06:20.52 | [TK]D-Fender | Ok, I'm finished... later all..... |
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| 07:10.33 | dlynes_home | h3x0r: the wct4xxp is the only thing on freebsd drivers that's stable :) |
| 07:14.27 | x86 | sweet, just got my MySQL-backed AGI script working to allow 7 digit dialing :) |
| 07:19.07 | dlynes_home | what's great about 7-digit dialing? |
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| 07:42.46 | Eric-xx | channel.c: Unable to find a codec translation path from g729 to slin |
| 07:42.52 | Eric-xx | does anyone know what is slin |
| 07:43.29 | Qwell | signed linear |
| 07:44.20 | Eric-xx | whats that |
| 07:44.35 | kay2 | Qwell: 8khz 16bit ? |
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| 07:51.36 | *** join/#asterisk TonyM (n=TonyM@softins.claranet.co.uk) |
| 08:01.24 | dlynes_home | Eric-xx: 16-bit raw signed audio data |
| 08:01.45 | Eric-xx | hmm stange don't think i enable such codec in any conf |
| 08:02.24 | dlynes_home | Eric-xx: did you enable ulaw or alaw? one of those two is slin...I can't remember which |
| 08:02.46 | dlynes_home | Eric-xx: i think it's the alaw |
| 08:03.31 | Eric-xx | i see |
| 08:03.44 | Eric-xx | but in my trunk i did not enable ulaw/alaw |
| 08:03.55 | Eric-xx | i have set my trunk to only have gsm|ilbc now .. The xlite i used is using libc , does asterisk always try to use alaw for the first contact? |
| 08:04.10 | dlynes_home | Eric-xx: do you have disallow=all? |
| 08:04.16 | Eric-xx | no |
| 08:04.20 | Eric-xx | i need that? |
| 08:04.31 | dlynes_home | yeah, otherwise alaw and ulaw are allowed by default |
| 08:04.50 | Eric-xx | i see |
| 08:05.11 | Eric-xx | ok another question |
| 08:05.21 | Eric-xx | what codec does asterisk supports? |
| 08:05.52 | Eric-xx | cause when i try just now , it looks like it don't support any other besides gsm/ilbc |
| 08:06.06 | dlynes_home | Eric-xx: type show codecs in the cli |
| 08:06.30 | Eric-xx | hmm |
| 08:07.09 | dlynes_home | Eric-xx: now all the ones you show in that list, asterisk doesn't support transcoding on all of them |
| 08:07.25 | dlynes_home | Eric-xx: it might only support passthrough on some, such as g.729 and g.723 |
| 08:07.36 | dlynes_home | Eric-xx: g.729 you can purchase licenses for it, for transcoding |
| 08:08.17 | Eric-xx | ok example i don't use g.729 |
| 08:08.42 | Eric-xx | allow=g.723,g726,ilbc,gsm, |
| 08:08.49 | dlynes_home | oh yeah...and g726, asterisk only supports g726-32 |
| 08:09.11 | dlynes_home | asterisk doesn't support g726-16, g726-24, or g726-40 |
| 08:10.10 | Eric-xx | allow=g726,ilbc,gsm |
| 08:10.19 | Eric-xx | this will be it right |
| 08:11.25 | dlynes_home | Eric-xx: g726,ilbc,gsm,ulaw,alaw |
| 08:11.44 | Eric-xx | oh i took ulaw and alaw out.. bandwidth consuming too high |
| 08:11.48 | dlynes_home | Eric-xx: speex as well, if you download it...it doesn't come with asterisk because someone owns the patent |
| 08:12.06 | dlynes_home | Eric-xx: but they allow people to freely implement the engine |
| 08:12.48 | Eric-xx | so does g726,ilbc and gsm do auto translation |
| 08:13.12 | dlynes_home | Eric-xx: yes, asterisk can handle transcoding between them |
| 08:13.43 | *** join/#asterisk Teeli (n=Tili@cm109.gamma248.maxonline.com.sg) |
| 08:16.45 | Eric-xx | great |
| 08:16.57 | Eric-xx | so if i wish to install other codecs , will that be hard? |
| 08:17.13 | Eric-xx | what are the codecs that i can choice anyway |
| 08:17.28 | dlynes_home | g.729 and speex |
| 08:17.47 | dlynes_home | g.729 you need to pay for...licenses are $10USD/ea from Digium |
| 08:17.59 | dlynes_home | speex is free, but it's patented |
| 08:18.12 | dlynes_home | you need to download it from another site besides digium's |
| 08:18.28 | dlynes_home | try looking on the wiki for speex to find out where to download it from |
| 08:18.34 | dlynes_home | there's also lpc10 |
| 08:18.40 | dlynes_home | but apparently that codec's really really bad |
| 08:19.07 | Eric-xx | i see |
| 08:19.12 | Eric-xx | wow .. thank you very much |
| 08:19.17 | Eric-xx | you really helped me a lot |
| 08:20.33 | Eric-xx | channel.c: Didn't get a frame from channel: SIP/AGLOW-70bf |
| 08:20.46 | Eric-xx | hmm do i need to put the frame size anywhere? |
| 08:21.13 | dlynes_home | nope |
| 08:21.23 | dlynes_home | what's the full error? |
| 08:21.45 | Eric-xx | Jun 3 16:19:47 VERBOSE[8377] logger.c: -- Attempting native bridge of SIP/1000001-64cc and SIP/AGLOW-70bf |
| 08:21.45 | Eric-xx | Jun 3 16:19:49 DEBUG[31167] chan_sip.c: Auto destroying call '6a4b08a32dd2c9bc10e9746c6c6fc5cd@127.0.0.1' |
| 08:21.45 | Eric-xx | Jun 3 16:19:53 DEBUG[8377] channel.c: Didn't get a frame from channel: SIP/AGLOW-70bf |
| 08:21.45 | Eric-xx | Jun 3 16:19:53 DEBUG[8377] channel.c: Bridge stops bridging channels SIP/1000001-64cc and SIP/AGLOW-70bf |
| 08:21.46 | Eric-xx | Jun 3 16:19:53 DEBUG[8377] chan_sip.c: update_call_counter(91886260) - decrement call limit counter |
| 08:21.46 | Eric-xx | Jun 3 16:19:53 DEBUG[8377] app_dial.c: Exiting with DIALSTATUS=ANSWER. |
| 08:21.49 | dlynes_home | dood |
| 08:21.54 | dlynes_home | use pastebin for shit like that :) |
| 08:21.58 | Eric-xx | opps |
| 08:22.01 | Eric-xx | forgotten |
| 08:22.02 | Eric-xx | sorry |
| 08:22.22 | dlynes_home | It looks fine to me |
| 08:22.40 | dlynes_home | What makes you think there was a problem? |
| 08:22.51 | Eric-xx | err.. everything is smooth |
| 08:23.01 | dlynes_home | and that's bad? |
| 08:23.05 | Eric-xx | just saw this log about not getting a frame channel |
| 08:23.15 | Eric-xx | so wonder if i need to do anything to improve it |
| 08:23.19 | dlynes_home | nah |
| 08:23.22 | dlynes_home | that's normal |
| 08:23.40 | dlynes_home | you'll get dropped frames every once in a while...it's one of the hazards of using udp |
| 08:23.45 | dlynes_home | however |
| 08:23.51 | dlynes_home | if you get excessive dropped frames |
| 08:23.54 | dlynes_home | then that's a problem |
| 08:24.12 | Eric-xx | i see |
| 08:24.22 | dlynes_home | when you do get a dropped frame, however |
| 08:24.32 | dlynes_home | you'll notice degradation in call quality |
| 08:24.34 | Eric-xx | after reading some stuffs about g.729 , i understand that pass-thru is okay |
| 08:24.42 | dlynes_home | Eric-xx: exactly |
| 08:25.16 | Eric-xx | so if i am just running asterisk, one of the extension uses g.729 and i pass it to my SIP provider (which supports g.729) , than asterisk will just pass over right |
| 08:25.57 | dlynes_home | sending end needs to be g.729 and receiving end also needs to be g.729, and you cannot attempt autonegotiation, unless canreinvite=yes |
| 08:28.01 | Eric-xx | client g.729 ---> Me ---> SIP provider which supports g.729 |
| 08:28.25 | Eric-xx | so i have to make sure client is g.729 and sip provider is g.729 right |
| 08:29.13 | dlynes_home | Eric-xx: is either end behind a firewall, and passing through that firewall to get to asterisk? |
| 08:29.53 | *** join/#asterisk gr0mit (n=tim@extrt.txrx.org.uk) |
| 08:30.01 | Eric-xx | only the sip provider have a firewall |
| 08:33.01 | x86 | anyone know where i can get local weather information in text form? |
| 08:33.08 | dlynes_home | weather.com |
| 08:33.16 | x86 | I would use NOAA.gov, but I don't want to parse HTML that could change next week |
| 08:33.25 | x86 | dlynes_home: they have text output? |
| 08:33.31 | dlynes_home | Eric-xx: yeah, so you're going to have to set g.729 as preferred codec on both ends then |
| 08:33.53 | dlynes_home | x86: afaik, yeah....ummm...what's that stupid xml schema again.... |
| 08:34.41 | x86 | dlynes_home: XML and HTML are markup... not text ;) |
| 08:34.58 | dlynes_home | x86: oh..you wanted pure text |
| 08:35.12 | dlynes_home | x86: say text/plain then, not text :) |
| 08:35.26 | dlynes_home | x86: thought you wanted something other than mp3s |
| 08:36.11 | x86 | lol |
| 08:36.37 | x86 | dlynes_home: well XML may work... |
| 08:36.49 | x86 | dlynes_home: you know where to get XML wheather info per a zip code? |
| 08:36.57 | dlynes_home | x86: rdf format...that's what it was |
| 08:37.37 | x86 | that's better than HTML :P |
| 08:38.01 | dlynes_home | x86: thought cause it's not plain text, it's not useful? :) |
| 08:38.37 | x86 | well i still have to write a parser for it, but XML would be more predictable than crap HTML :P |
| 08:39.09 | dlynes_home | x86: xml you don't have to write a parser for...just use xerces |
| 08:40.06 | x86 | dlynes_home: for my application i would... anyway, what's the URL you're getting this from? |
| 08:40.15 | dlynes_home | x86: one sec |
| 08:40.40 | Eric-xx | ok |
| 08:42.17 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 08:42.56 | dlynes_home | http://www.weather.com/weather/rss/subscription?from=servicesindex |
| 08:43.03 | dlynes_home | rss, not rdf |
| 08:43.08 | dlynes_home | that's what it was |
| 08:46.22 | *** join/#asterisk RoyK (n=roy@10.80-203-106.nextgentel.com) |
| 08:48.35 | Eric-xx | what if |
| 08:48.53 | Eric-xx | client g.729 ---> Me (suppports g.729)---> SIP provider which does NOT support g.729 |
| 08:49.07 | Eric-xx | i will be able to do the conversation before sending to sip provider right |
| 08:49.39 | dlynes_home | Eric-xx: nope, not when SIP provider is behind a firewall, because asterisk will be part of the media path |
| 08:50.00 | dlynes_home | Eric-xx: and asterisk does not autonegotiate properly when it is part of the media path |
| 08:50.20 | Eric-xx | so if my provider asterisk does not have a firewall will it works? |
| 08:50.52 | dlynes_home | Eric-xx: correct |
| 08:51.03 | Eric-xx | i see |
| 08:52.47 | Eric-xx | i believe the $10 from digium is a one time fee right |
| 08:52.51 | Eric-xx | not monthly |
| 08:52.57 | dlynes_home | correct |
| 08:53.03 | dlynes_home | $10/channel |
| 08:53.18 | Eric-xx | do i need a zap card for this |
| 08:53.23 | dlynes_home | no |
| 08:53.29 | Eric-xx | ok |
| 08:53.37 | *** join/#asterisk _4d4m_ (n=adam@62.69.102.99) |
| 08:53.40 | Eric-xx | hmm but the cpu consuming is real high |
| 08:53.43 | *** join/#asterisk feld_ (n=feld@12.148.212.157) |
| 08:54.02 | Eric-xx | i think i have to put another server seperately that supports g.729 |
| 08:54.25 | dlynes_home | Eric-xx: possibly...all depends on how many calls you plan to handle, and the speed of your current cpu |
| 08:55.05 | Eric-xx | if i separate them its better right? cause i have a P42.66gig and i wish to use this as the basic server (gsm,libc and such) |
| 08:55.21 | Eric-xx | than i will purchase another server to accept (g.729 alone) |
| 08:55.39 | dlynes_home | Eric-xx: yeah...go for it |
| 08:56.00 | dlynes_home | Eric-xx: if you're planning to handle a lot of calls, figure on about 30 licenses, and a dual cpu, or a dual-core |
| 08:56.11 | dlynes_home | maybe as many as 40 licenses |
| 08:56.30 | Eric-xx | hmm is there away to create a asterisk load balancing system |
| 08:57.33 | dlynes_home | Eric-xx: yeah...see the section on large asterisk systems on the front page of the asterisk wiki |
| 08:57.45 | Eric-xx | ok |
| 08:58.12 | Eric-xx | g.723 looks expensive |
| 08:59.50 | dlynes_home | Eric-xx: you can't currently get it for asterisk |
| 09:00.10 | dlynes_home | Eric-xx: however, digium does have plans to bring out a card with dsp codec support for g.723 and g.729 |
| 09:00.19 | Eric-xx | i see |
| 09:00.25 | dlynes_home | Eric-xx: they haven't set a release date, or even an approximate release date |
| 09:00.34 | Eric-xx | so for now only g.729 is something i need to setup |
| 09:00.52 | dlynes_home | Eric-xx: correct |
| 09:01.04 | dlynes_home | Eric-xx: and if you encounter a provider that needs g.723, you're pretty much hooped |
| 09:01.51 | Eric-xx | hmm i think it's okay .. since is so expensive, don't think a lot of provider will be using it |
| 09:02.00 | Eric-xx | does asterisk support h.232 ? |
| 09:02.09 | *** join/#asterisk abatista (n=Ariel@dsl-20-177.cofs.net) |
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| 09:40.16 | Qwell | heh, g723 is ungodly expensive |
| 09:41.38 | x86 | moreso than g729? |
| 09:42.15 | Qwell | x86: ungodly |
| 09:43.39 | x86 | hmm |
| 09:43.48 | x86 | why dont people just use GSM or g728? |
| 09:43.57 | MGSsancho | like $20? |
| 09:48.20 | Qwell | try tens of thousands, before they'll even talk to you |
| 09:49.28 | RoyK | <PROTECTED> |
| 09:49.37 | RoyK | x86: g.728? |
| 09:49.59 | x86 | RoyK: was that a question? |
| 09:50.56 | RoyK | didn't think there was any g.728 support in asterisk |
| 09:51.35 | RoyK | g.723.1 is insanely priced |
| 09:52.01 | Qwell | it's far beyond "unreasonable" |
| 09:52.29 | MGSsancho | is there any major advantages? i dont meanlike 2-5%. |
| 09:53.15 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.220) |
| 09:54.56 | *** join/#asterisk danalien (n=danalien@unaffiliated/danalien) |
| 09:55.20 | RoyK | MstlyHrmls: advantages of what? |
| 09:55.51 | RoyK | hm. wtf is ast_frame declared? |
| 09:56.24 | MGSsancho | g.723 over g.728 |
| 09:57.33 | RoyK | why G.728? There's no support for that in * |
| 09:57.59 | MGSsancho | oh hmmm |
| 09:58.12 | MGSsancho | whats recommended? |
| 09:58.17 | RoyK | for what? |
| 09:58.21 | RoyK | G.711a |
| 09:58.30 | MGSsancho | k |
| 09:58.33 | MGSsancho | thanks |
| 09:58.40 | Qwell | RoyK: silly non-US codec :P |
| 09:58.47 | RoyK | Qwell: only sane one :) |
| 09:59.00 | Qwell | REAL men use G.711u |
| 09:59.02 | MGSsancho | T_T |
| 09:59.09 | RoyK | Qwell: bloody USAnians |
| 09:59.19 | Qwell | RoyK: our codec is better than yours! ;) |
| 09:59.28 | MGSsancho | why not use .wav |
| 09:59.31 | RoyK | Qwell: :) |
| 09:59.42 | dpryo | MGSsancho: wav is a container format ;P |
| 09:59.49 | MGSsancho | oh |
| 09:59.55 | MGSsancho | hahahhaha |
| 10:02.18 | MGSsancho | so g.711a for non US, and g.711u for US? |
| 10:02.49 | RoyK | g.711a is used in the sane world on PSTN |
| 10:02.55 | RoyK | g.711u is used elsewhere :) |
| 10:03.12 | MGSsancho | ahh ok |
| 10:04.09 | MGSsancho | gsm for ) and gsm cell phones. any simularties? or just the name? |
| 10:04.09 | MGSsancho | * |
| 10:04.30 | *** join/#asterisk af_ (n=af@ip-164-240.sn2.eutelia.it) |
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| 10:29.55 | Eric-xx | using g.729, how much in and out bandwidth if i talk for one hour ? |
| 10:30.23 | MGSsancho | 3.14tb <__< |
| 10:30.34 | Eric-xx | .... |
| 10:30.41 | MGSsancho | PI tb |
| 10:31.21 | MGSsancho | mmmmmmmmm pi. *goes offto get some cake while win98SE installs on a clients laptop* |
| 10:32.40 | Eric-xx | seriously |
| 10:32.49 | Eric-xx | any people able to tell me how to calculate ? |
| 10:35.15 | Eric-xx | wiki explained as a voip provider view |
| 10:35.35 | MGSsancho | oh |
| 10:35.53 | Eric-xx | but if i am a consumer, and i wish to know by using g.729 and talk for one hour, how much up and down bandwidth i have used |
| 10:36.19 | MGSsancho | 8 Kbps |
| 10:36.40 | Eric-xx | for an hour? |
| 10:36.56 | MGSsancho | thats 28,880Kb |
| 10:37.10 | MGSsancho | 3,600KB |
| 10:37.20 | Eric-xx | what's the formul |
| 10:37.23 | MGSsancho | 3.515625 MB |
| 10:37.57 | Eric-xx | hmm ... for one hour is already 3MB? damn... |
| 10:38.03 | GarethTheGreat | anyone here familiar with kphone? |
| 10:38.13 | MGSsancho | 8kb x 3600 secs in hour = 28,880kb |
| 10:38.46 | MGSsancho | that / 8 = 3,600Kb. then 3,600Kb / 1024 = 3.515625MB |
| 10:38.53 | Eric-xx | ghoss 8kb of transfer per second ??! |
| 10:39.21 | MGSsancho | killo bits |
| 10:39.25 | MGSsancho | not bytes |
| 10:40.05 | Eric-xx | ya but 1 hour takes 3 mb? that's a lot :( |
| 10:40.27 | MGSsancho | a 14.4K modem can handle that lol |
| 10:41.13 | Eric-xx | hmm but 8kb is based on upload + download |
| 10:41.23 | MGSsancho | yeah |
| 10:41.32 | Eric-xx | okay |
| 10:42.12 | Eric-xx | thanks |
| 10:42.59 | MGSsancho | np |
| 11:12.17 | X-Gen | BOO |
| 11:12.44 | *** join/#asterisk postel (n=jp@unaffiliated/postel) |
| 11:14.25 | *** join/#asterisk starlein (i=star@fo0bar.de) |
| 11:25.10 | [Airwolf] | Can someone tell me if the mp3's for Asterisk moh need to be in a special frequency ? |
| 11:29.10 | [Airwolf] | Because I keep getting this: |
| 11:29.11 | [Airwolf] | Jun 3 13:28:33 WARNING[10444]: interface.c:215 decodeMP3: Junk at the beginning of frame 49443303 |
| 11:36.11 | MGSsancho | 4:35 am = bed time ga'night and good luck |
| 11:48.58 | stephane_ | re |
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| 12:00.48 | *** join/#asterisk psyco-obiwan (n=cschnee@2001:4060:4419:b1:0:0:0:2) |
| 12:01.58 | psyco-obiwan | can i have sipsak let a sip phone ring ? i want to alert over voip from an internet server |
| 12:05.28 | psyco-obiwan | basically i need some scriptable ua for a shell env. |
| 12:06.54 | tzafrir_laptop | psyco-obiwan, sipp may be a better choice |
| 12:07.05 | *** join/#asterisk mosty (i=mostynm@60-241-198-194.static.tpgi.com.au) |
| 12:07.10 | tzafrir_laptop | or another asterisk |
| 12:07.23 | tzafrir_laptop | what do you need it for? |
| 12:08.46 | *** join/#asterisk Weezey (n=ohno@206.210.111.31) |
| 12:08.52 | tzafrir_laptop | just establish one SIP connection? |
| 12:08.57 | Weezey | What's up with -addons ? why won't it compile |
| 12:09.21 | tzafrir_laptop | Also, the iaxclient distribution includes some simple IAX comamnd-line clients |
| 12:09.23 | psyco-obiwan | i have a wireless webcam, the cam images get grabbed from a shell account on an internet site and saved on motion detection....what i want is to let a phone ring upon motion detection beside storing the image, so that people can switch on their monitor and see the fox live ;-) |
| 12:10.11 | psyco-obiwan | a whole asterisk seems to complex for the job of just letting a phone ring, but the iaxclients sound interesting.. |
| 12:11.03 | psyco-obiwan | tzafrir_laptop: apt-cache search sipp yields: sipp - create and render 3-d scenes |
| 12:11.21 | tzafrir_laptop | on Debian it's called 'sip-tester' |
| 12:11.33 | tzafrir_laptop | but maybe sipsak will be better |
| 12:11.39 | tzafrir_laptop | or linphonec |
| 12:13.30 | psyco-obiwan | thx for the hints, ill check them |
| 12:19.41 | Weezey | dammit. Why would they leave -addons in the fucked position? |
| 12:40.27 | *** join/#asterisk RoyK (n=roy@static-213-115-144-122.sme.bredbandsbolaget.se) |
| 12:43.32 | RoyK | <PROTECTED> |
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| 12:43.44 | RoyK | ~nickometer [a]freebsd_fan |
| 12:44.29 | [a]freebsd_fan | your daddy is 98% lame :P |
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| 13:11.26 | kay2 | tzafrir: what's sip-tester |
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| 14:31.40 | *** join/#asterisk brockj49464_home (n=chatzill@63.87.56.153) |
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| 14:33.30 | *** join/#asterisk ManxPower (n=ewieling@dpc67142183150.direcpc.com) |
| 14:36.37 | brockj49464_home | Why would my outgoing calls end after 120 minutes? How can I remove this? |
| 14:36.57 | ManxPower | brockj49464_home, We don't know. |
| 14:37.07 | dlynes_home | brockj49464_home: Are you using any parameters to the Dial command? |
| 14:37.10 | ManxPower | Are you using something like Asterisk@Home? |
| 14:37.31 | brockj49464_home | AAH upgraded manually a few times |
| 14:37.44 | dlynes_home | brockj49464_home: please read the topic |
| 14:38.04 | *** join/#asterisk Ariel_ (n=Ariel@70.46.87.158) |
| 14:38.25 | brockj49464_home | So if I look in the dial command in the log what would it look like if a limit was set? |
| 14:38.32 | dlynes_home | brockj49464_home: unfortunately, amp/freepbx's configuration files are such a mess, we're not familiar enough with them in this topic |
| 14:38.41 | dlynes_home | s/topic/channel/ |
| 14:39.14 | dlynes_home | brockj49464_home: it would have some bracketted parameters after it, in the third position of the dial command |
| 14:39.49 | ManxPower | dlynes_home, There could also be an AbsoluteTimeout set before thge Dial |
| 14:39.58 | dlynes_home | ManxPower: that too |
| 14:40.24 | ManxPower | Calls also may NOT be ending at exactly 120 mins. That could be caused by busydetect or callprogress options to Zap. |
| 14:40.34 | dlynes_home | ManxPower: btw...what's the point to the absolutetimeout variable? |
| 14:40.51 | ManxPower | dlynes_home, To limit the time a call can run |
| 14:41.05 | dlynes_home | ManxPower: yeah, but those parameters to the dial command already manage that |
| 14:41.19 | dlynes_home | ManxPower: oh...nvm...they don't control that for incoming calls though |
| 14:41.43 | [Airwolf] | Is it possible in the dailplan to use a goto and return to the same context after the goto is finished ? |
| 14:41.44 | ManxPower | dlynes_home, those Dial options are reasonably recent additions. Also, as you said Dial isn't run for IVR, etc. |
| 14:42.03 | ManxPower | [Airwolf], Use a Gosub or a Macro |
| 14:42.14 | dlynes_home | [Airwolf]: gosub |
| 14:42.21 | brockj49464_home | Jun 2 22:43:47 VERBOSE[19176] logger.c: -- Executing Dial("SIP/164-9406", "SIP/Z8921d0b95Out/16417744539|120|r") in new stack |
| 14:42.24 | [Airwolf] | Thankyou, |
| 14:42.29 | ManxPower | [Airwolf], asterisk does not do any of that sort of checking. you can goto anywhere. |
| 14:42.40 | [Airwolf] | I found that out :P |
| 14:42.47 | dlynes_home | brockj49464_home: it's not your dial command then...there's probably an absolutetimeout set |
| 14:43.32 | *** join/#asterisk Assid (i=assid@203.115.83.214) |
| 14:43.35 | ManxPower | brockj49464_home, "r" will hide any error messages. |
| 14:43.52 | ManxPower | That 120 in your dial says the call must be answered by the far end within 120 SECONDS |
| 14:43.57 | dlynes_home | ManxPower: ah...didn't know that...thanks for the info |
| 14:44.15 | dlynes_home | ManxPower: the ',r', that is |
| 14:44.54 | ManxPower | dlynes_home, Imagine this: You dial an extension and it sends the call to a cell phone with something like Dial(Zap/1/5551212). If the cell phone is turned off or out of area or the number is disconnected you will hear that message from the carrier. |
| 14:45.18 | ManxPower | If you add ,,r then you will hear ringing instead of the "the number you have dialed is disconnected" or other telco message. |
| 14:45.29 | dlynes_home | ManxPower: yeah, makes sense |
| 14:45.47 | ManxPower | It is also not REQUIRED ever. |
| 14:46.09 | ManxPower | I use it so the callers do NOT hear the telco message because after the timeout they will be sent to the local voicemail. |
| 14:46.21 | Weezey | Manx: I have some IAX connections where if I don't put in the ring, the user hears dead air until the other party answers |
| 14:46.25 | ManxPower | only really useful for hiding the telco messages |
| 14:46.28 | dlynes_home | ManxPower: not according to someone on here earlier...although, they never bothered to mention in which extenuating circumstances it's required |
| 14:46.31 | brockj49464_home | grep "absolutetimeout" /etc/asterisk/* did not find anytime... |
| 14:46.42 | ManxPower | Weezey, then there is some OTHER problem, like lack of /etc/asterisk/indications.conf on one end. |
| 14:46.42 | dlynes_home | ManxPower: and why would you want to hide them? |
| 14:46.44 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 14:46.54 | Weezey | ManxPower: cool, good to know. |
| 14:47.01 | ManxPower | dlynes_home, so the caller does not hangup before the call goes to voicemail. |
| 14:47.56 | ManxPower | dlynes_home, also it makes the users stop yelling about it. |
| 14:48.07 | dlynes_home | ManxPower: ah...but how are you going to know if the carrier is going to tell you all circuits are busy? |
| 14:48.19 | dlynes_home | ManxPower: erm actually |
| 14:48.22 | ManxPower | dlynes_home, in THIS specific case, I don't care. |
| 14:48.31 | dlynes_home | ManxPower: in that case you wouldn't be forwarding to voicemail, anyways |
| 14:48.53 | ManxPower | Since the caller dialed a 4-digit extension they don't even know what number was actually dialed. |
| 14:48.58 | dlynes_home | ManxPower: the only reason I can think of for that, is if you're forwarding an extension to the user's cell phone or something |
| 14:49.12 | ManxPower | dlynes_home, That is what I was describing. |
| 14:49.15 | dlynes_home | ManxPower: ah |
| 14:49.30 | ManxPower | ManxPower dlynes_home, Imagine this: You dial an extension and it sends the call to a cell phone with something like Dial(Zap/1/5551212). If the cell phone is turned off or out of area or the number is disconnected you will hear that message from the carrier. |
| 14:49.48 | ManxPower | NORMALLY you want to hear the carrier's messages. |
| 14:50.02 | dlynes_home | ManxPower: yeah, and in that case you don't want the user to hear it because as you said, they'll hang up |
| 14:50.12 | brockj49464_home | I do find this, could this be the reason the call drops after 120min? "Didn't get a frame from channel:" |
| 14:50.28 | dlynes_home | ManxPower: the only other time i've enabled it is when I've got users bitching about how long it takes before they start hearing a ringing |
| 14:51.03 | dlynes_home | ManxPower: most of them are able to deal with it after I tell them sometimes it takes a while before it starts ringing on voip |
| 14:51.09 | dlynes_home | ManxPower: but others can't deal with that concept |
| 14:52.05 | [Airwolf] | dlynes_home, if I would execute a Macro, would Asterisk always return to that context, no matter what I do in the macro ? |
| 14:52.10 | dlynes_home | brockj49464_home: that shouldn't be an issue, unless you're getting a lot of those |
| 14:52.38 | *** join/#asterisk TheCops (i=nobody@got.securebinary.com) |
| 14:52.51 | dlynes_home | [Airwolf]: i wouldn't know...manxpower would probably be more qualified to answer that...i've never used macros |
| 14:52.57 | brockj49464_home | 22 in 24 hours |
| 14:53.09 | dlynes_home | brockj49464_home: on the same call? |
| 14:53.20 | brockj49464_home | in the full log file |
| 14:53.30 | dlynes_home | brockj49464_home: how many in a single call? |
| 14:53.38 | dlynes_home | brockj49464_home: more than one? |
| 14:54.09 | dlynes_home | brockj49464_home: also, what kind of internet connection are they on? |
| 14:54.22 | [Airwolf] | Like if I do something like this: s,1,macro(blaat,100) s,2,system(do something) [macro-blaat] s,1,goto(call,{arg1},1) |
| 14:54.26 | brockj49464_home | 1 per call |
| 14:54.39 | [Airwolf] | ManxPower, do you know this ? :) |
| 14:54.58 | dlynes_home | brockj49464_home: is it a telco dsl line, or a cable carrier internet service? |
| 14:55.18 | brockj49464_home | Motorola Canopy. |
| 14:55.26 | dlynes_home | brockj49464_home: wtf is taht? |
| 14:55.37 | brockj49464_home | wireless broadband |
| 14:55.48 | dlynes_home | as in 802.11? |
| 14:56.07 | dlynes_home | or something similar? |
| 14:56.09 | brockj49464_home | no in 5.8ghz |
| 14:56.32 | dlynes_home | is it cellphone band or soemthing then? |
| 14:56.43 | brockj49464_home | uses a dish aimed at an atenna, not cell phone |
| 14:56.53 | TheCops | brockj49464_home, canopy own :) |
| 14:57.10 | dlynes_home | TheCops: is it something available in montreal? |
| 14:57.21 | TheCops | For services ? or to buy ? |
| 14:57.26 | brockj49464_home | can't get dsl or cable |
| 14:57.28 | dlynes_home | for services, yeah |
| 14:57.50 | TheCops | for services, Xittel is offering services and Hypertelecom (I've started this compagny 2 yeras ago) |
| 14:57.51 | ManxPower | [Airwolf], a goto breaks all macro or gosub returns |
| 14:57.54 | TheCops | but this is not at montreal |
| 14:57.59 | TheCops | this is in little city |
| 14:58.07 | dlynes_home | TheCops: ah...i think [TK]D-Fender told me you were in Montreal |
| 14:58.11 | TheCops | nop |
| 14:58.14 | TheCops | Valleyfield |
| 14:58.23 | TheCops | 1 hours from Montreal :) |
| 14:58.25 | [Airwolf] | ManxPower, ok, then I have to find out how to do what I want then. :) |
| 14:58.28 | dlynes_home | well, close enough |
| 14:58.30 | dlynes_home | sheesh |
| 14:58.34 | TheCops | heh |
| 14:58.38 | dlynes_home | That's a suburb to me |
| 14:58.45 | ManxPower | [Airwolf], you can call a macro inside of a macro |
| 14:59.02 | TheCops | dlynes_home, where you are living if you can't get DSL or cable ?! |
| 14:59.15 | brockj49464_home | if my provider terminated the call what kind of message would I see in the log? |
| 14:59.16 | dlynes_home | TheCops: it's brockj49464_home that can't get dsl or cable, not me |
| 14:59.20 | TheCops | ok |
| 14:59.26 | [Airwolf] | ManxPower, well I want to have a function that records a call and e-mails the recording after the call. |
| 14:59.32 | dlynes_home | brockj49464_home: offhand, i wouldn't know for sip |
| 14:59.33 | TheCops | brockj49464_home where do you live |
| 14:59.37 | dlynes_home | brockj49464_home: never really looked at it |
| 14:59.51 | dlynes_home | TheCops: btw...i'm in vancouver |
| 15:00.04 | [Airwolf] | But I have some lcr, so for call I need to go to that context. |
| 15:00.16 | TheCops | dlynes_home, yeah I just remember:) |
| 15:00.22 | ManxPower | brockj49464_home, Dial sets the variable HANGUPCAUSE check the status of that Noop(HANGUPCAUSE=${HANGUPCAUSE}) after the Dial |
| 15:00.29 | ManxPower | you may need the "g" option to dial |
| 15:00.38 | Dr-Linux | hi |
| 15:00.40 | dlynes_home | TheCops: ah...didn't think i'd mentioned it to you before...never really talked to you before that I remember |
| 15:01.09 | TheCops | hehe true, but someday with Fender around we was talking hehe |
| 15:01.16 | dlynes_home | ah |
| 15:01.26 | dlynes_home | yeah...CunningPike's just across the river from me |
| 15:01.29 | dlynes_home | He's in Richmond |
| 15:01.34 | TheCops | :) |
| 15:01.35 | dlynes_home | and works in North Van |
| 15:02.06 | dlynes_home | I work all over the place...I'm a subcontractor for an interconnect |
| 15:02.06 | TheCops | interconnect? |
| 15:02.06 | TheCops | Bell ? |
| 15:02.12 | dlynes_home | interconnect, not ilec |
| 15:02.30 | TheCops | dlynes_home, this is a road job ? |
| 15:03.06 | dlynes_home | interconnect is a company that does cabling from demarc into the suite, runs all the data and phone runs, terminates them into jacks and mod ends, and patch panels, ..., and sells phone systems and does all the programming for the phone systems |
| 15:03.20 | TheCops | Ha :) |
| 15:03.35 | TheCops | dlynes_home, and sell annual contract I guess hehe |
| 15:03.45 | dlynes_home | We also sometimes have to run new cable up through the risers too |
| 15:04.02 | *** join/#asterisk Dr-Linux (n=Linux@202.59.73.131) |
| 15:04.03 | dlynes_home | TheCops: atm, we don't do that |
| 15:04.10 | TheCops | dlynes_home, ok |
| 15:04.12 | dlynes_home | TheCops: but I've been trying to push the boss to start doing that |
| 15:04.23 | TheCops | dlynes_home, this is why Bell exist today. |
| 15:04.32 | TheCops | years contract $$ |
| 15:04.33 | TheCops | ;) |
| 15:04.53 | dlynes_home | then it would give us some guaranteed income we could start growing the business on and it would make the customers have the warm fuzzies going with us instead of someone else |
| 15:04.56 | TheCops | dlynes_home, Do you have an URL ? |
| 15:05.08 | dlynes_home | http://www.247communications.com/ |
| 15:05.16 | dlynes_home | It needs some major work on the website :) |
| 15:05.23 | dlynes_home | some idiot wrote the whole thing in flash |
| 15:05.42 | TheCops | LOL |
| 15:06.15 | dlynes_home | It'll probably be getting redone next month |
| 15:06.34 | dlynes_home | The owner's getting married in another week, so he's not thinking about anything else atm |
| 15:07.07 | [Airwolf] | Is it better to have all the functions in my dailplan as macro's ? |
| 15:07.25 | TheCops | dlynes_home, I'm consultant for new technologies, VoIP, Barcode/RFID, automate, basic,advanced,isp networking, security and stuff like that. |
| 15:07.25 | [Airwolf] | So I can call them more easy. |
| 15:07.45 | TheCops | dlynes_home, I dont see Cisco stuff on your website, if you need Cisco, just call me hehe |
| 15:07.52 | TheCops | I'm reseller stuff and services |
| 15:07.57 | *** join/#asterisk queuetue (n=scott@toronto-HSE-ppp4122670.sympatico.ca) |
| 15:08.15 | queuetue | Hello. How would I append a "1" to outgoing calls that don't have it? |
| 15:08.44 | dlynes_home | TheCops: heh...i'll keep it in mind |
| 15:08.57 | mosty | queuetue, use 1${EXTEN} in your Dial command ? |
| 15:09.01 | [Airwolf] | queuetue, Dial(SIP/1{EXTEN}) something like that |
| 15:09.04 | dlynes_home | TheCops: at the current time though, we don't need cisco...cisco's priced themselves out of the market |
| 15:09.12 | [Airwolf] | mosty, don't be so fast :P |
| 15:09.22 | TheCops | dlynes_home, 24/7 services is very the best services to make money easy hehe |
| 15:09.45 | dlynes_home | Yeah...we truly are 24/7 too |
| 15:09.56 | dlynes_home | We just got a2zcommunications.com, too :) |
| 15:10.01 | dlynes_home | I'm surprised it wasn't taken :) |
| 15:10.12 | queuetue | mosty, [Airwolf] Won't that always add a 1? I only need to add one when the user did not... |
| 15:11.02 | [Airwolf] | queuetue, then you should use the if statement. If the exten doesn't contain a 1 then add it. |
| 15:11.20 | dlynes_home | queuetue: use a GotoIf |
| 15:11.32 | dlynes_home | [Airwolf]: there's no such thing as an if statement |
| 15:11.34 | queuetue | I'm pretty sure you can just do it with the numeric notation... |
| 15:11.39 | [Airwolf] | And use substring on the exten if you want it to be on a special position. |
| 15:11.42 | dlynes_home | [Airwolf]: unless of course there's on in AEL |
| 15:11.47 | dlynes_home | s/on/one/ |
| 15:12.13 | [Airwolf] | dlynes_home, I'm unsure how to call it in Asterisk. |
| 15:12.22 | ManxPower | In general anything you can do in AEL you can do in regular dialplan since AEL is translated into standard dialplan stuff. |
| 15:12.27 | mosty | queuetue, send that user to a specific context, and do it in that context |
| 15:12.29 | [Airwolf] | I just had java at college and they called it and if statement. |
| 15:12.31 | ManxPower | AEL is never RUN, it's translated |
| 15:13.05 | [Airwolf] | s/and/an |
| 15:13.06 | dlynes_home | [Airwolf]: java has an if statement, asterisk extensions.conf has a gotoif statement; the two are not related; java is a programming language, extensions.conf is not |
| 15:13.18 | [Airwolf] | hmm |
| 15:13.27 | [TK]D-Fender | dlynes_home : More like 30mins ;) |
| 15:13.32 | dlynes_home | [Airwolf]: you need the final forward slash |
| 15:13.34 | [TK]D-Fender | TheCops : PM |
| 15:13.35 | mosty | methinks asterisk is turing complete, most likely |
| 15:13.42 | [Airwolf] | dlynes_home, tnx. |
| 15:14.02 | [Airwolf] | But according to the wiki extensions does have an if thing. :) |
| 15:14.12 | dlynes_home | GotoIf |
| 15:14.21 | dlynes_home | [Airwolf]: but like i said...AEL might have an if |
| 15:14.35 | dlynes_home | [Airwolf]: i've never used ael, so i don't know for sure |
| 15:14.38 | [Airwolf] | dlynes_home, http://www.voip-info.org/wiki/index.php?page=Asterisk+func+if |
| 15:15.29 | [Airwolf] | dlynes_home, but it doesn't matter. :P |
| 15:15.33 | dlynes_home | ah...never used functions |
| 15:15.39 | [Airwolf] | exactly |
| 15:15.40 | dlynes_home | only used dialplan applications |
| 15:15.57 | dlynes_home | always assumed functions were for manager api, agi, or ael or something |
| 15:16.35 | [Airwolf] | since 1.2 not more apperently. |
| 15:16.40 | dlynes_home | well, i've seen people use cut, but i thought that was a dialplan application |
| 15:17.00 | dlynes_home | [Airwolf]: yeah, and iwth 1.4 AEL2 will be available |
| 15:17.59 | [Airwolf] | dlynes_home, but I have another question. Right now I have a dial plan and I have made diffrent sections in my dial plan (like external, local, services) and in those sections I have context for every function. |
| 15:18.10 | dlynes_home | heh...that's weird...seeing a country singer with an earring in both ears |
| 15:18.20 | [Airwolf] | But I really don't know if that is the best way to setup a dail plan. |
| 15:18.41 | [Airwolf] | And I can find any best pratices, it's like trail and error. |
| 15:18.53 | dlynes_home | [Airwolf]: whichever way is the easiest to work with and readable for you |
| 15:19.20 | dlynes_home | for me, I have an entire tree structure set up with common files in each directory |
| 15:19.52 | dlynes_home | and so my main extensions.conf file just consists of a whole bunch of exten => did,1,Goto(context,s,1) |
| 15:20.22 | dlynes_home | and obviously a whole bunch of #includes too |
| 15:20.29 | [Airwolf] | I get it. Well, my extentions.conf is still one long file. But I'm running into the fact that it's no longer handable |
| 15:20.56 | dlynes_home | [Airwolf]: are you doing any ivr stuff in there? |
| 15:21.14 | [Airwolf] | yes, also |
| 15:21.20 | [TK]D-Fender | [Airwolf] : Pastebin the whole thing and let me hav a look. |
| 15:21.24 | [Airwolf] | It's about 315 lines now |
| 15:21.30 | dlynes_home | yeah...you might want to move your ivr off into a separate file |
| 15:21.42 | dlynes_home | and maybe put your outbound routing into another file |
| 15:21.57 | [Airwolf] | I think I will do that. |
| 15:21.58 | dlynes_home | and your extensions into another file |
| 15:22.14 | [Airwolf] | [TK]D-Fender, well I'm going to upload it somewhere |
| 15:22.24 | dlynes_home | then the only thing left in your main file will be your macros and your incoming dialplan |
| 15:23.07 | [TK]D-Fender | ~pb |
| 15:23.08 | jbot | i heard pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
| 15:23.25 | [TK]D-Fender | 315? hah! |
| 15:24.02 | dlynes_home | I think mine if it was all in one file would probably be about 2000 or 3000 lines |
| 15:24.07 | [Airwolf] | [TK]D-Fender, www.ymav.nl/extentions.conf |
| 15:24.22 | [TK]D-Fender | One of my AstDB driven STDEXTEN macro's and their supporting setup scripts is 326 all by itself! |
| 15:24.25 | [Airwolf] | dlynes_home, well it's a small organisation with alot of voip wishes. :P |
| 15:25.03 | [Airwolf] | And for me, I just am a student who like to play alot with Asterisk |
| 15:25.07 | dlynes_home | [Airwolf]: yeah, same here |
| 15:25.14 | [Airwolf] | And make some money on the side. |
| 15:25.21 | dlynes_home | and i'm the one that gets saddled with implementing it |
| 15:25.22 | dlynes_home | yay |
| 15:27.23 | dlynes_home | [Airwolf]: try putting all those *-out contexts into a separate file called extensions/outbound.conf |
| 15:27.24 | [TK]D-Fender | [Airwolf] : Your setup isn't too bad. A little cleanup, and conversion required for proper 1.2 compliance, but I wouldn't mess with that... |
| 15:28.07 | dlynes_home | [Airwolf]: and maybe move your menus out into menus/menuname.conf |
| 15:28.28 | [Airwolf] | dlynes_home, I will do that. Thank you for the tip |
| 15:28.35 | [Airwolf] | [TK]D-Fender too |
| 15:28.44 | dlynes_home | [Airwolf]: and extensions/internal.conf for the [internal] context |
| 15:28.51 | *** join/#asterisk ms345 (n=mike_sim@64.74.198.10) |
| 15:29.01 | dlynes_home | [Airwolf]: and extensions/macros.conf for the macros |
| 15:29.02 | [Airwolf] | But first I'm going to finish the last functions they want. |
| 15:29.23 | [Airwolf] | dlynes_home, so like all the sections I made just in seperate files. |
| 15:29.24 | dlynes_home | [Airwolf]: and extensions/queues.conf, extensions/services.conf for your queues and services |
| 15:29.31 | [Airwolf] | :) |
| 15:29.45 | dlynes_home | [Airwolf]: then you can find everything easily |
| 15:29.57 | dlynes_home | [Airwolf]: and it'll be easier to follow your logic |
| 15:30.13 | asterboy | dlynes_home, looks like turning up the txgain has reduced the number of callerID checksum errors. |
| 15:30.25 | dlynes_home | asterboy: that's what i told you :) |
| 15:30.29 | dlynes_home | asterboy: to adjust your gains :) |
| 15:30.30 | [Airwolf] | dlynes_home, I get it. |
| 15:30.39 | asterboy | but you don' |
| 15:30.47 | asterboy | don't work for digium do you? |
| 15:30.50 | dlynes_home | nope |
| 15:30.52 | [Airwolf] | dlynes_home, do ever do anything with the queue application ? |
| 15:31.06 | asterboy | interesting that when I asked that, none of the digium staff on here repsonded. |
| 15:31.09 | dlynes_home | asterboy: actually, if you want, I have a c program to allow you to loop through your gains until you get it just right |
| 15:31.24 | asterboy | ya, that sounds perfect |
| 15:31.33 | dlynes_home | [Airwolf]: nah...never used queues or agents |
| 15:31.38 | dlynes_home | [Airwolf]: never had to set up a call center |
| 15:31.55 | [Airwolf] | Ah ok |
| 15:32.30 | [Airwolf] | I needed a way for checking if there where any agents logged in for a specfied queue. |
| 15:32.44 | [Airwolf] | But it seems that feature isn't yet avalible in Asterisk. |
| 15:33.35 | *** join/#asterisk Dr-Linux (n=Linux@202.59.73.131) |
| 15:41.44 | dlynes_home | asterboy: try http://www.ancient-legacy.org/letsdoit.c |
| 15:42.05 | dlynes_home | asterboy: just do gcc -Wall -pedantic -o letsdoit letsdoit.c to compile it |
| 15:42.17 | dlynes_home | asterboy: then type ./letsdoit to get the usage instructions |
| 15:42.51 | *** join/#asterisk burizaa (n=freeee@cm107.omega96.maxonline.com.sg) |
| 15:43.13 | dlynes_home | asterboy: it's a pretty crude tool, but it does the job, without manually having to make a whole bunch of changes |
| 15:43.39 | *** join/#asterisk ManxPower (n=ewieling@dpc67142183150.direcpc.com) |
| 15:43.59 | *** join/#asterisk juanjoc (n=juanjoc@248-32-235-201.fibertel.com.ar) |
| 15:51.21 | *** join/#asterisk wunderkin (i=kev@ip68-226-113-228.ph.ph.cox.net) |
| 15:51.24 | asterboy | I ran dlynes_home and now this message comes up: "Sucker, you've benn rooted!" |
| 15:51.28 | asterboy | what does that mean? |
| 15:51.36 | burizaa | where i can get the newest asterisk-addons |
| 15:51.56 | asterboy | asterisk.org? |
| 15:52.07 | dlynes_home | asterboy: lol...liar |
| 15:52.54 | asterboy | :P |
| 15:52.57 | burizaa | found it ! |
| 15:53.23 | dlynes_home | asterboy: and it's ftp.digium.com |
| 15:54.03 | *** join/#asterisk Zer0HiT (n=Z@cpe-72-231-175-199.nycap.res.rr.com) |
| 15:55.45 | asterboy | letsdoit -6.3 6.3 .1 |
| 15:56.41 | asterboy | Seems to be stuck with this message repeating: This run used rxgain=-7, txgain=-7 |
| 15:56.41 | dlynes_home | -7, -7? |
| 15:56.41 | *** join/#asterisk cfassoni (n=cfassoni@c911444e.rjo.virtua.com.br) |
| 15:56.41 | asterboy | is the min allowed to be negative? |
| 15:56.55 | asterboy | also, I have three rx/txgain lines in my zapata.conf |
| 15:56.58 | dlynes_home | asterboy: yes, but I also only use integers |
| 15:57.03 | dlynes_home | asterboy: I don't use floating point |
| 15:57.37 | asterboy | ok, that seems to do it. |
| 15:57.42 | asterboy | letsdoit -6 6 1 |
| 15:58.02 | burizaa | guys, anyone tried to hook asterisk with gnugk ? |
| 15:58.03 | dlynes_home | might want to try -6 12 1 |
| 15:58.50 | dlynes_home | wtf is this stupid flavor of love show supposed to be about? |
| 15:59.05 | dlynes_home | Flavor Flave's the host of the show |
| 15:59.26 | asterboy | it set 6 for both |
| 15:59.39 | asterboy | I had a -6.3 for txgain and 2.0 for rxgain |
| 15:59.44 | dlynes_home | asterboy: ? |
| 15:59.57 | dlynes_home | asterboy: Yeah..i increase both |
| 16:00.12 | dlynes_home | asterboy: Like I said...it's quite crude |
| 16:00.16 | asterboy | be nice to have it test each individually |
| 16:00.24 | dlynes_home | asterboy: so modify it then :) |
| 16:00.29 | dlynes_home | asterboy: you've got the c code :) |
| 16:00.59 | asterboy | /usr/src/letsdoit -5 12 1 |
| 16:01.13 | asterboy | trying that in case it just took my arguments |
| 16:01.17 | asterboy | as what to set |
| 16:01.35 | asterboy | c code...been a while |
| 16:02.21 | dlynes_home | asterboy: anyways...it restarts asterisk every time if I remember correctly, so you can test each increment |
| 16:02.28 | dlynes_home | asterboy: it's been a while since I wrote it :) |
| 16:02.36 | asterboy | ya that's what it did...took my arguments to set the file. |
| 16:02.58 | dlynes_home | yeah...sleeps 2M milliseconds each iteration |
| 16:03.18 | dlynes_home | so 2000 seconds |
| 16:03.34 | dlynes_home | erm microseconds |
| 16:03.40 | dlynes_home | so sleeps for 2 seconds |
| 16:04.16 | dlynes_home | between iterations |
| 16:04.28 | dlynes_home | so you have to hit ctrl-c on asterisk each time after you've finished testing |
| 16:04.50 | dlynes_home | then it'll increment to the next gain setting |
| 16:08.11 | *** join/#asterisk adorah (n=Asterjet@87.69.72.228) |
| 16:13.52 | burizaa | for h323, which one is better? h323, oh323, ooh323c, woomera ? |
| 16:17.37 | *** join/#asterisk eimajenthat (n=jamie@cpe-70-123-133-94.austin.res.rr.com) |
| 16:18.38 | iq | hi |
| 16:19.35 | dlynes_home | burizaa: none of the above :) |
| 16:20.13 | dlynes_home | burizaa: if you can possibly avoid h323, avoid it like hte plague |
| 16:20.39 | *** join/#asterisk saaib (n=nabudoco@75.7.229.85) |
| 16:20.46 | *** join/#asterisk PMantis (n=pmantis@cpe-66-66-115-197.rochester.res.rr.com) |
| 16:20.48 | *** join/#asterisk iq|mobile (n=iq@71-215-55-11.omah.qwest.net) |
| 16:21.22 | PMantis | Hi... can someone clarify: |
| 16:21.23 | PMantis | MeetMe([confno][,[options][,pin]]) |
| 16:21.58 | PMantis | Does the [pin] parameter *enter* the pin for the person, or prompt the use for that pin, overriding the pin n meetme.conf? |
| 16:23.03 | *** join/#asterisk markit (n=konversa@host119-245.pool8172.interbusiness.it) |
| 16:23.49 | markit | please, I need help with the translation of a sound (I can't understand it, I need some explaination): vm-saveoper.gsm press 1 to accept this recording, otherwise, please continue to hold |
| 16:24.07 | markit | "continue to hold"... what happens? what is the meaning of "hold"? |
| 16:25.24 | eimajenthat | If I have an asterisk server running at my house, how does it connect to people with regular phones or on other VOIP services? If you can't tell, I'm a complete ignoramu about this asterisk stuff, but ti sounds interesting. Would love to read anything like an "Asterisk for Dummies" page, if you've got one. |
| 16:26.03 | *** join/#asterisk chapeaurouge (n=chapeaur@user-85-201-82-146.tvcablenet.be) |
| 16:26.17 | eimajenthat | I checked the page listed in the subject. |
| 16:26.21 | PMantis | eimajenthat, There are many options... You can buy a card that allows your * server to connect to your home phone line. |
| 16:26.56 | PMantis | eimajenthat, ...or you can sign up with a VoIP service that assigns you a pone number, then sends/receives calls to/from yur * server over the internet. |
| 16:26.58 | russellb | eimajenthat: you should check out the asterisk o'reilly book |
| 16:27.01 | mosty | eimajenthat, you have to pay someone who has a link to the regular phone system, either via a regular phone line at your house, or over the internet to someone who has a connection to the regular phone system |
| 16:27.03 | russellb | ~thebook |
| 16:27.24 | [TK]D-Fender | ~book |
| 16:27.24 | jbot | extra, extra, read all about it, book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 16:28.21 | PMantis | Also, read my local article here: http://www.techny.com/articles.cfm?getarticle=801&go=0.677435032263 |
| 16:28.28 | *** join/#asterisk ToyMan (n=stuq@74-32-70-93.dsl1.mdl.ny.frontiernet.net) |
| 16:28.29 | PMantis | (well, local to my city) :-) |
| 16:29.10 | eimajenthat | interesting |
| 16:29.37 | eimajenthat | so, Vonage and companies like that have to pay to connect to the regular phone network, right? |
| 16:29.46 | *** join/#asterisk mmmmmToop (n=mmmmToop@firewall.datapro.co.za) |
| 16:29.47 | PMantis | eimajenthat, Exactly. |
| 16:30.16 | eimajenthat | but if I paid Vonage, I could hook my asterisk box into the network via Vonage |
| 16:30.17 | PMantis | eimajenthat, They get a PRI, the run a VoIP PBX system that takes the PRI phone calls, and sends them over the internet to your LinkSys PAP2. |
| 16:30.36 | PMantis | eimajenthat, no, Vonage only allows you to use *their* device. |
| 16:30.53 | eimajenthat | oh, hrm |
| 16:31.02 | kukhuvud | can anyone recomend some * billing software? i dont need it to process CC's or anything, but it needs to at least link extensions to a group/company and tell me how many minutes they've used, how much i'm charging per minute for them etc |
| 16:31.03 | burizaa | dlynes_home: i tried to avoid h323, but my friend in singapore has quintum which just support h323 only :( so sad.. and i can't connect to his VoIP SS (MERA) ... |
| 16:31.07 | PMantis | eimajenthat, Check out viatalk. www.viatalk.com or broadvoice www.broadvoice.com, voipjet, connect.voicepulse.com, etc |
| 16:31.21 | eimajenthat | will do |
| 16:31.31 | mmmmmToop | any ideas on retrieving which an agent is getting a call from Queue() cmd? |
| 16:31.44 | eimajenthat | so all these will let you use your own asterisk box? |
| 16:31.46 | PMantis | ViaTalk and BroadVoice offer inlimited inbound/outbound plans. |
| 16:32.09 | PMantis | eimajenthat, The companies I mentioned allow for Asterisk, on a BYOD (bring your own device) otion. |
| 16:32.12 | eimajenthat | what about SIP. What's that all about? |
| 16:32.22 | PMantis | That's the protocol. |
| 16:32.34 | PMantis | There's H.323, SIP, IAX2, and some others. |
| 16:32.56 | PMantis | They're different ways of sending voice over and IP network. |
| 16:33.02 | PMantis | s/and/an |
| 16:33.12 | eimajenthat | Skype uses SIP, no? |
| 16:33.30 | PMantis | Heh, actually Skype has their own version. |
| 16:33.45 | eimajenthat | oh |
| 16:34.05 | PMantis | There's a bounty out there enticing anyone to decipher the Skype protocol, and make a channel module for Asterisk... nothing yet. |
| 16:34.41 | PMantis | So, anyon have a clue about MeetMe? |
| 16:34.53 | PMantis | Does the [pin] parameter *enter* the pin for the person, or prompt the use for that pin, overriding the pin n meetme.conf? |
| 16:36.22 | eimajenthat | anyone here using ViaTalk? $15 a month sounds like an awesome deal. |
| 16:36.30 | PMantis | Yup, me |
| 16:36.52 | PMantis | I signed up when they had the $200/year special with 1 year free |
| 16:37.07 | PMantis | So... $100/year for unlimited calling?? I'll take that. |
| 16:37.27 | PMantis | My biggest complaint is that they're very rigid in their plans, and wont' budge... |
| 16:37.42 | *** join/#asterisk tzafrir_laptop (n=tzafrir@local.xorcom.com) |
| 16:37.47 | PMantis | They allow for simultaneous 2 inbound and 2 outbound calls. |
| 16:38.13 | PMantis | eimajenthat, What kind of internet connection you have? |
| 16:38.37 | eimajenthat | Cable, TimeWarner RoadRunner |
| 16:38.46 | PMantis | ok, me too. |
| 16:38.58 | *** join/#asterisk zotz (n=zotz@24.244.133.115) |
| 16:39.03 | PMantis | You'll want a "Broadband Booster" it *really* cleared up my calls. |
| 16:39.16 | eimajenthat | what is it? |
| 16:39.25 | eimajenthat | a hardware device, or a service? |
| 16:39.38 | PMantis | A little $50 box that you plug your cable modem into, then on the other side, your router or computer. |
| 16:40.00 | eimajenthat | what's it do? |
| 16:40.12 | eimajenthat | or is that one of those great mysteries of life? |
| 16:40.20 | PMantis | It simply prioritizes the voice communications, so that a download doesn't cause a oice cal to break up and sound choppy, robotic, etc. |
| 16:40.30 | PMantis | ugh |
| 16:40.36 | PMantis | s/oice/voice |
| 16:40.58 | eimajenthat | so it's specifically for people running VOIP stuff |
| 16:41.01 | Zer0HiT | but wouldn't a QoS setting in your router do the same thing as that? |
| 16:41.13 | *** join/#asterisk anthm (n=anthm@000-450-480.area4.spcsdns.net) |
| 16:41.13 | *** mode/#asterisk [+o anthm] by ChanServ |
| 16:41.24 | PMantis | Zer0HiT, Essentially, yes... but I tried that and it didn't help me. |
| 16:41.27 | PMantis | This device did. |
| 16:41.28 | PMantis | http://www.hawkingtech.com/products/productlist.php?CatID=36&FamID=80&ProdID=216 |
| 16:41.46 | PMantis | I think buy.com has it on sale with free shipping, too. |
| 16:41.53 | Zer0HiT | sweet currently between VoIP companies as i'm tired of vonage BS and haven't tried the QoS setting personally. |
| 16:42.21 | Zer0HiT | does viatalk only do SIP? |
| 16:42.42 | eimajenthat | I'm doing alright for now with just my cell phone. But my parents have dial-up and Bell phone service. I'm not sure how much their phone bill is, but I'm thinking if I got them to go with RoadRunner and VOIP, it might be pretty close to the same price. |
| 16:42.46 | PMantis | Zer0HiT, if you use their router, it should include QoS... |
| 16:43.10 | *** part/#asterisk saaib (n=nabudoco@75.7.229.85) |
| 16:43.31 | PMantis | eimajenthat, viatalk is great for that... they offer LinkSys PAP2's and have no problem giving you the admin password so you can unlock it ater, fi you want. |
| 16:43.45 | PMantis | Vonage will *never* help you unlock the device they give you. |
| 16:44.09 | PMantis | Zer0HiT, viatalk does SIP or IAX2, but they discourage IAX2. |
| 16:44.18 | PMantis | Strongly, too. |
| 16:44.55 | Zer0HiT | really i thought IAX2 was preferred over SIP? |
| 16:45.14 | Dr-Linux | Zer0HiT: NO |
| 16:45.21 | wunderkin | if it worked right it would be |
| 16:45.26 | Zer0HiT | haha |
| 16:45.27 | Dr-Linux | SIP rocks! |
| 16:45.30 | PMantis | SIP and IAX2 both have their place... but what makes SIP annoying for NAT makes is preferable for some situations. |
| 16:45.43 | markit | please, I need help with the translation of a sound (I can't understand it, I need some explaination): vm-saveoper.gsm press 1 to accept this recording, otherwise, please continue to hold |
| 16:45.44 | Zer0HiT | how so? |
| 16:45.47 | markit | "continue to hold"... what happens? what is the meaning of "hold"? |
| 16:46.21 | PMantis | Zer0HiT, If a provider resold SIP servive, you'd have this: |
| 16:46.39 | PMantis | Zer0HiT, Provider1--->Provider2--->Asterisk |
| 16:47.26 | [Airwolf] | Can someone tell me if it's possible to execute something when a channel is disconnected ? |
| 16:47.27 | PMantis | With SIP and reinvites, the SIP control channel takes that path, but the RDP packets (voice) go straight from Provider1-->Asterisk... avoiding the extra lag of the middle provider. |
| 16:47.44 | PMantis | [Airwolf], exten => h,1,DoSomething() |
| 16:47.53 | Zer0HiT | but w/ IAX2 it'd still go through the middle provider(provider2)? |
| 16:48.03 | [Airwolf] | PMantis, what does the h stand for ? |
| 16:48.23 | PMantis | Zer0HiT, Yes, AFAIK IAX2 doesn't support reinvites, because it's all in one TCP connection. (better for NAT, however) |
| 16:48.28 | PMantis | [Airwolf], h=hangup |
| 16:48.41 | [Airwolf] | Ah, thank you |
| 16:48.42 | *** join/#asterisk salviadud (n=ralfalfa@201.133.207.93) |
| 16:48.43 | PMantis | [Airwolf], t=timeout, i=invalid... |
| 16:49.12 | Zer0HiT | PMantis: thank you, i'll have to read up on IAX2 more, a friend suggested i should look for companies that provide IAX2 service over SIP |
| 16:49.23 | PMantis | Zer0HiT, and in my case it's even worse sometimes... :) |
| 16:49.28 | *** join/#asterisk Eggplant (i=No@dsl-72-19-46-175.cascadeaccess.com) |
| 16:49.42 | PMantis | Zer0HiT, Provider1-->Provider2-->Asterisk-->my remote device/laptop |
| 16:50.23 | salviadud | that's kind of odd |
| 16:50.41 | Zer0HiT | ah |
| 16:50.43 | salviadud | PMantis, why would it go through 2 providers first? |
| 16:51.24 | PMantis | salviadud, A SIP provider may buy their telephone service from another SIP provider (Reseller), *or* they may have their own PRI's. |
| 16:51.51 | PMantis | salviadud, Just using it as an example, to show where SIP has an advantage. |
| 17:03.57 | *** join/#asterisk feld_ (n=feld@12.148.212.157) |
| 17:04.08 | *** join/#asterisk Qwell (n=north@unaffiliated/qwell) |
| 17:06.54 | *** join/#asterisk burizaa (n=freeee@cm107.omega96.maxonline.com.sg) |
| 17:07.10 | burizaa | how to set SIP codec ? |
| 17:07.45 | Dr-Linux | burizaa: at the server? |
| 17:08.21 | Dr-Linux | burizaa: you can do that in sip.conf in general section |
| 17:09.07 | burizaa | hmmm |
| 17:09.18 | burizaa | i want to do it for once trunk only... |
| 17:09.30 | burizaa | they ask me to use g723 :( no idea how to set it out |
| 17:09.48 | Dr-Linux | burizaa: you can do that |
| 17:10.00 | burizaa | i put disallow=all allow=g723 i think it's correct right ? |
| 17:10.12 | Dr-Linux | burizaa: just put there "allow=g723" |
| 17:10.14 | file | g723.1 |
| 17:10.24 | burizaa | file: need to put .1 ? |
| 17:10.33 | Dr-Linux | g723.1? : |
| 17:10.40 | burizaa | Dr-Linux: should i put "disallow=all" ? |
| 17:10.50 | file | meh, they're the same |
| 17:11.15 | Dr-Linux | burizaa: yes, but we should follow file's guide. |
| 17:11.38 | *** join/#asterisk lunaphyte (n=lunaphyt@pool-71-115-145-155.gdrpmi.dsl-w.verizon.net) |
| 17:11.48 | Dr-Linux | file: what's difference between allow=g723 and g723.1 ? |
| 17:12.01 | burizaa | file: i put allow=g723.1 and remove "disallow=all" right ? |
| 17:12.10 | file | internally they're the same because everyone put g723 and forgot the .1 |
| 17:13.52 | Dr-Linux | i see |
| 17:14.31 | *** join/#asterisk Ariel_ (n=Ariel@dsl-20-177.cofs.net) |
| 17:14.44 | [Airwolf] | hmm |
| 17:14.59 | Dr-Linux | burizaa: in the gerneral section use "allow=g723" and then disallow=all |
| 17:15.01 | [Airwolf] | I use the system command to send a mail after a call is hangup |
| 17:15.13 | [Airwolf] | I see on the console that it gets executed |
| 17:15.25 | *** join/#asterisk Tili (n=Tili@cm109.gamma248.maxonline.com.sg) |
| 17:15.26 | Dr-Linux | burizaa: but you must have g723 codec's module, it's not free |
| 17:15.41 | [Airwolf] | But the mail doesn't even get send. |
| 17:16.20 | Dr-Linux | [Airwolf]: check mail logs /var/log/maillog |
| 17:16.35 | *** part/#asterisk markit (n=konversa@host119-245.pool8172.interbusiness.it) |
| 17:16.36 | dlynes_home | burizaa: why can't you connect to his mera softswitch? |
| 17:17.22 | dlynes_home | Dr-Linux: there is no g723 codec module; only a g723 format module |
| 17:17.36 | bon | -win µ+ |
| 17:17.48 | dlynes_home | bon? |
| 17:17.50 | Dr-Linux | dlynes_home: same i said, |
| 17:18.04 | dlynes_home | Dr-Linux: no, it's not |
| 17:18.09 | dlynes_home | Dr-Linux: the g723 format module is free |
| 17:18.17 | dlynes_home | Dr-Linux: the g723 codec module is simply non-existent |
| 17:18.41 | *** join/#asterisk burizaaa (n=freeee@cm107.omega96.maxonline.com.sg) |
| 17:18.46 | burizaaa | sorry just got disconnected |
| 17:18.52 | Dr-Linux | dlynes_home: hmm.. it's ok thanks :) |
| 17:18.59 | dlynes_home | Dr-Linux: the codec format modules allow pass through |
| 17:19.07 | dlynes_home | Dr-Linux: the codec modules allow translation |
| 17:19.11 | Dr-Linux | burizaa: your net connection is still better than mine :) |
| 17:19.16 | burizaaa | lolz |
| 17:19.29 | burizaaa | dlynes_home: do you know my problem? :p |
| 17:19.35 | dlynes_home | burizaa: lar |
| 17:19.37 | burizaaa | i was dc before |
| 17:19.38 | [Airwolf] | Dr-Linux, I did but no new log. |
| 17:19.51 | [Airwolf] | It seems that Asterisk doesn't execute it. |
| 17:19.58 | [Airwolf] | But hey says he does. :P |
| 17:20.03 | dlynes_home | burizaa: your problem with what lah? |
| 17:20.12 | Dr-Linux | dlynes_home: i have separate modules for g729 and for g723 , i put them in modules dir and i load them from CLI |
| 17:20.18 | Dr-Linux | and i can see in them in traslation |
| 17:20.27 | dlynes_home | dlynes_home: yeah...format_g729 and format_g723 |
| 17:20.45 | dlynes_home | erm |
| 17:20.51 | dlynes_home | Dr-Linux: unless of course you mean the illegal versions |
| 17:20.53 | Dr-Linux | heh |
| 17:20.53 | Dr-Linux | :S |
| 17:21.07 | Dr-Linux | dlynes_home: mine is not like that |
| 17:21.24 | dlynes_home | Dr-Linux: so you must be using the illegal versions then |
| 17:21.37 | burizaaa | oke... i got SIP provider they using voipswitch and only accept g723. i try to call asterisk using express talk (sip) and i saw from the console that the call is answered but it's got disconnected after that |
| 17:21.42 | Dr-Linux | dlynes_home: nope i use legal stuff |
| 17:21.43 | dlynes_home | Dr-Linux: i.e. the ones that are legal in countries that don't respect intellectual property rights |
| 17:22.26 | florz | dlynes_home: what's otherwise illegal about them? |
| 17:22.41 | dlynes_home | florz: nothing...just a patent licensing issue |
| 17:22.57 | Dr-Linux | dlynes_home: i'm paid for them, so it's legal for me |
| 17:23.11 | florz | dlynes_home: Well, that's what I was thinking, which is why I wondered :-) |
| 17:23.13 | dlynes_home | florz: and i doubt very much Dr-Linux has paid $100K for the rights, and $50/seat or whatever ridiculous amount it is they charge for the licensing |
| 17:23.17 | Dr-Linux | dlynes_home: check PM |
| 17:24.14 | florz | dlynes_home: But if you are in a country where there is not patent on it? Or if you are just using it privately? |
| 17:24.24 | dlynes_home | florz: i'm not using it, period |
| 17:24.33 | dlynes_home | florz: i'm in a country where the patent is quite enforcable |
| 17:24.55 | Dr-Linux | dlynes_home: but not in my country |
| 17:25.02 | florz | dlynes_home: Even when you are using it just for your own private pleasure? |
| 17:25.23 | dlynes_home | florz: i don't know what the specifications are regarding that, but it's irrelevant to me |
| 17:25.30 | dlynes_home | florz: I work for an interconnect |
| 17:25.36 | dlynes_home | florz: everything has to be completely legal |
| 17:25.41 | Dr-Linux | dlynes_home: i'm damn sure, many peoples are using in the same way .. but do no tell!!! |
| 17:26.08 | florz | dlynes_home: sure ;-) |
| 17:29.38 | dlynes_home | dood |
| 17:29.41 | Dr-Linux | anybdoy know what softphone supports g723 codec? |
| 17:29.46 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 17:29.49 | dlynes_home | what's with all these peeps using ipv6 addresses lately? |
| 17:30.35 | salviadud | it's the future |
| 17:30.46 | salviadud | soon, my microwave oven will have an ipv6 address |
| 17:30.56 | florz | salviadud: Only one? |
| 17:31.13 | salviadud | yeah with 128 bits... i think it's enough |
| 17:31.29 | salviadud | i'll get an ip for everthing |
| 17:31.40 | salviadud | YES GENTLEMEN |
| 17:31.49 | salviadud | finally, we will be able to ping that missing tubesock |
| 17:32.20 | dlynes_home | florz: yeah...i noticed you were using one |
| 17:32.22 | Dr-Linux | when ipv6 is coming out? |
| 17:32.31 | dlynes_home | florz: pastebin.ca's dns servers are using ipv6, too |
| 17:32.51 | florz | dlynes_home: I guessed so, yeah :-) |
| 17:32.55 | dlynes_home | florz: but where do you get the ipv6 address allocated from? |
| 17:33.07 | florz | dlynes_home: From my provider? :-) |
| 17:33.12 | dlynes_home | florz: i.e. who decides what ipv6 addresses you're allowed to have? |
| 17:33.27 | dlynes_home | florz: ok, and where did they get it allocated from? |
| 17:33.39 | dlynes_home | and what are the reserved subnets for private networks? |
| 17:33.41 | florz | dlynes_home: RIPE I guess |
| 17:34.04 | justinu | cool thing about ipv6 is you get a /64 by default |
| 17:34.14 | Dr-Linux | DIANA decides |
| 17:34.15 | justinu | if you ask, you can get a/48 |
| 17:34.18 | burizaaa | how to i get th g723 codec? dl from http://kvin.lv/pub/Linux/Asterisk right? then put the file under /usr/lib/asterisk/modules, am i correct ? |
| 17:34.18 | florz | dlynes_home: AFAIK those are deprecataed (as in "there are none") |
| 17:34.45 | burizaaa | *how do i, typo wrongly |
| 17:34.45 | dlynes_home | florz: so you're stuck with ipv4 for private subnets then? |
| 17:35.08 | florz | dlynes_home: No, just use some of your IPv6 addresses!? |
| 17:35.21 | dlynes_home | burizaaa: well, I don't help people get around patent issues |
| 17:35.33 | justinu | there's no point for private address space ipv6 |
| 17:35.53 | burizaaa | dlynes_home: okay sorry :) |
| 17:36.03 | *** join/#asterisk feld_ (n=feld@12.148.212.157) |
| 17:36.05 | dlynes_home | florz: yeah, and then if someone else on the internet has the same ipv6 address as you, your network isn't going to know where to route that ip address |
| 17:36.34 | florz | dlynes_home: I mean yours as in yours, not as in someone else's =:-) |
| 17:37.09 | salviadud | well, i know an IRC server that uses ipv6, and you can get connected via port 8080 |
| 17:37.11 | dlynes_home | justinu: i remembered about the cd this time, but i think i might have left it in the van, instead of taking it home in the car :p |
| 17:37.17 | salviadud | it's an efnet server |
| 17:37.25 | justinu | :) |
| 17:37.37 | salviadud | pretty useful if you're at work and you can't get irc |
| 17:37.46 | salviadud | doesn't apply to me though |
| 17:37.59 | justinu | with IPv6, you get assigned a network prefix, then your computers basically auto append their MAC to that prefix |
| 17:38.13 | justinu | and your "subnet" is 64bit address space |
| 17:38.21 | florz | dlynes_home: I mean, amongst those 18446744073709551616 there will be some to spare for that purpose, no?! :-) |
| 17:38.27 | justinu | so the idea that other people will have the same adress as you is kinda silly |
| 17:39.32 | dlynes_home | justinu: well, i'm sure mac addresses get reused, too :) |
| 17:39.34 | burizaaa | dlynes_home: how do i get the legal codec? |
| 17:39.49 | justinu | MACs only have to be linklevel unique |
| 17:39.52 | justinu | so it's not a problem. |
| 17:40.05 | dlynes_home | burizaaa: pay the patent owner $100K for the right to use their codec, and then an additional $50 or something per channel |
| 17:40.14 | *** join/#asterisk chandi (n=burni13@modemcable237.178-37-24.mc.videotron.ca) |
| 17:40.24 | dlynes_home | burizaaa: or wait for digium to come out with a codec for it |
| 17:40.29 | dlynes_home | burizaaa: or sangoma |
| 17:40.32 | dlynes_home | burizaaa: or whoever |
| 17:41.04 | burizaaa | omg ! |
| 17:41.19 | burizaaa | 100k.... cool ! |
| 17:41.20 | florz | erm, and BTW, I didn't even ask for a /48 and still got one, so in case 18446744073709551616 are not enough, that really should do ;-) |
| 17:41.26 | dlynes_home | yeah...exactly why I don't use g723 |
| 17:41.51 | dlynes_home | The patent on g723 from what i understand will be expiring soon, anyways |
| 17:42.05 | dlynes_home | So when the patent expires, I'm sure a free implementation will emerge |
| 17:42.22 | burizaaa | lol |
| 17:42.42 | dlynes_home | afaik though, g729 has a while before it expires |
| 17:42.47 | justinu | g729 sounds just as good, and is cheaper |
| 17:43.13 | dlynes_home | justinu: yeah, but unfortunately a lot of the carriers are using g.723 |
| 17:43.43 | justinu | who? |
| 17:43.50 | justinu | i've only run across g729 and g711 |
| 17:44.03 | dlynes_home | i've only run across g729 and g723 |
| 17:44.22 | justinu | they don't offer 711? that sucks |
| 17:44.34 | dlynes_home | about 80% of my wholesalers carriers within north america do g723 |
| 17:44.34 | chandi | hey guys, I need help from someone with a good imagination ;) I want to use Dial dialplan's command to call a number. I want it to hang up if the callee has NOT sent a DTMF within 15secondes. How can I do that ? I might be willing to hack the code if it's impossible |
| 17:44.39 | dlynes_home | the remainder do g729 |
| 17:44.49 | florz | BTW, TelDaFax in .de seems to be nice: They seem to support any codecs * knows of and the prices are quite affordable, especially for low volume ... |
| 17:44.49 | dlynes_home | most of the overseas traffic is g729 |
| 17:45.29 | dlynes_home | florz: the transatlantic lag would kill us :) |
| 17:46.42 | dlynes_home | I need to find a wholesale carrier that'll do iax in north america, though |
| 17:46.49 | justinu | why? |
| 17:46.50 | dlynes_home | Preferably Canadian |
| 17:46.53 | florz | dlynes_home: Well, I don't know whether they're affordable for international calls, but rates to landline within .de are pretty good - and in that case you couldn't avoid the lag anyway =:-) |
| 17:46.55 | *** join/#asterisk lorinc (n=ang@caracas-1198.adsl.interware.hu) |
| 17:47.01 | dlynes_home | justinu: to cut down on our bandwidth costs |
| 17:47.13 | dlynes_home | justinu: sip's bandwidth consumption is killing us |
| 17:47.31 | justinu | sip, or rtp? |
| 17:47.49 | dlynes_home | obviously the rtp portion :) |
| 17:49.21 | dlynes_home | I just want to use iax trunking to cut down on the channel overhead |
| 17:49.42 | justinu | does that even work right? |
| 17:49.53 | dlynes_home | justinu: what do you mean? |
| 17:50.02 | justinu | i remember a lot of people having trouble with trunking |
| 17:50.08 | dlynes_home | justinu: hasn't it been being used for a couple of years now? |
| 17:50.23 | dlynes_home | justinu: i've never had a problem with it, doing trunking between my pbxes and my main softswitch |
| 17:50.52 | jsaunders | Anyone run accross this? "frame.c:179 __ast_smoother_feed: Dropping extra frame of G.729 since we already have a VAD frame at the end" |
| 17:51.01 | dlynes_home | the only issue i've ever had with it, is that it doesn't seem to work with asterisk boxes that are on a dynamic ip address |
| 17:51.12 | dlynes_home | jsaunders: yeah...all the time |
| 17:51.17 | dlynes_home | jsaunders: just ignore it |
| 17:51.17 | jsaunders | Heheh |
| 17:51.53 | dlynes_home | jsaunders: for whatever reason, they didn't do a if( option_verbose>5 ) { printf( "frame.c:179 __ast_smoother_feed: Dropping extra frame of G.729 since we already have a VAD frame at the end" ) ; } on it |
| 17:52.02 | burizaaa | any idea which softphone using g723 ? |
| 17:52.10 | dlynes_home | burizaaa: probably none |
| 17:52.14 | justinu | eyebeam might |
| 17:52.18 | burizaaa | WAW ! |
| 17:52.30 | burizaaa | justinu: free one |
| 17:52.34 | justinu | no free one |
| 17:52.35 | jsaunders | dylnes_home: Audio quality is horrible though, I believe it's dropping packets. |
| 17:52.41 | jsaunders | * 1.2.8 |
| 17:53.14 | justinu | how could someone pay 100k + $50 per seat and then give the software away? |
| 17:53.19 | jsaunders | It's not including annexb=no in SDP which I believe was the fix as per RFC. Once again, not sure. Just goin' off what I'm reading thus far. |
| 17:53.55 | burizaaa | hrrrrr |
| 17:53.58 | burizaaa | goes crazy :D |
| 17:54.26 | jsaunders | Found this article. http://bugs.digium.com/view.php?id=5539&nbn=19 |
| 17:54.48 | jsaunders | Apparenlty it was dealt w/ in 2005 but it's creating a problem for me w/ a certain provider. |
| 17:54.59 | justinu | jsaunders: you said you're not getting annexb=no in your SDP? |
| 17:55.01 | justinu | that's a problem |
| 17:55.18 | jsaunders | k, thanks, I'll do some more sniffing in that direction justinu. :) |
| 17:55.38 | justinu | otherwise your provider probably thinks they can send you annexb 729, which has VAD enabled. |
| 17:55.45 | jsaunders | Gotcha |
| 17:55.45 | justinu | which we all know asterisk can't handle |
| 17:55.52 | jsaunders | As do I, now. |
| 17:55.53 | jsaunders | Heh |
| 17:56.45 | *** join/#asterisk opc0de (n=adam@CPE006008148866-CM000f9fa8c50a.cpe.net.cable.rogers.com) |
| 17:57.03 | opc0de | hey can anyone tell me if it's possible to do a three way call through an asterisk system from a remote line? |
| 17:57.22 | justinu | "remote line"? |
| 17:57.30 | justinu | be specific here, we can take it |
| 17:57.51 | opc0de | I'mm at home with a POTS phone line, want to make a conference call through my asterisk box at work |
| 17:57.54 | dlynes_home | justinu: no we can't...we'll just run away screaming mad |
| 17:57.59 | justinu | wuss :P |
| 17:58.03 | opc0de | we've got 4 POTS lines going into the asterisk box |
| 17:58.18 | justinu | opc0de: you want to take a look at app_meetme |
| 17:58.36 | opc0de | I thought meetme required people dialing into the system |
| 17:58.43 | opc0de | I want to be able to initiate both outgoing calls |
| 17:58.58 | justinu | ah. not sure if zap channels offer 3 way calling features |
| 17:59.36 | opc0de | yeah I can do it easily enough from a SIP phone through the asterisk box, but never done it through a POTS phone remotely |
| 17:59.37 | dlynes_home | jsaunders: yeah...it creates a problem for me with certain sip devices |
| 18:00.04 | justinu | you'd need some way to trigger the 3 way calling, and flash hook ain't gonna work thru PSTN |
| 18:00.36 | justinu | however, i'm sure you can come up with some thing creative to make it work |
| 18:00.49 | jsaunders | dlynes_home: There's a hack @ http://bugs.digium.com/view.php?id=5539&nbn=19 that apparently removes the dropping of packets, may prove fruitful. Want to do some more debugging 1st. |
| 18:01.18 | justinu | certain g729 implementations maybe ignoring the annexb=no |
| 18:01.26 | justinu | in that case, i think you're hosed. |
| 18:01.30 | jsaunders | Heheh |
| 18:02.09 | justinu | i don't understand why digium doesn't put time in implementing things like VAD |
| 18:02.25 | dlynes_home | heh |
| 18:02.28 | jsaunders | heheh |
| 18:02.50 | dlynes_home | Well, they're not making any money from asterisk directly |
| 18:02.56 | jsaunders | In which case you respond w/ an invoice for them. :D |
| 18:03.05 | justinu | hah |
| 18:03.07 | dlynes_home | They're the same as every other consultant...they make money by supporting asterisk |
| 18:03.15 | jsaunders | Prolly wouldn't go over too well. Heh. |
| 18:03.20 | dlynes_home | But they also make money from the g729 licensing and from selling their cards |
| 18:03.40 | justinu | yeah... they sell a half ass implementation of g729 |
| 18:03.52 | dlynes_home | better than none |
| 18:03.54 | justinu | i mean don't get me wrong |
| 18:03.55 | justinu | it works |
| 18:03.57 | justinu | and I use it |
| 18:04.13 | dlynes_home | yeah, i know what you mean |
| 18:04.23 | dlynes_home | it doesn't fully implement g729, and it's less than optimal |
| 18:04.25 | justinu | i'm just an impatient and whiny mofo |
| 18:04.30 | russellb | what is half-ass about it? |
| 18:04.35 | justinu | lack of VAD support? |
| 18:04.52 | russellb | that's something that isn't supported in asterisk |
| 18:05.13 | russellb | that is absolutely not specific to the g729 module |
| 18:05.21 | justinu | i don't see how that changes anything |
| 18:05.48 | russellb | why would the g729 module support g729b, when asterisk can't do anything with it |
| 18:06.05 | justinu | so the real question is why doesn't asterisk support it? |
| 18:06.30 | dlynes_home | because justinu hasn't implemented that functionality yet |
| 18:06.34 | justinu | heh |
| 18:06.34 | russellb | because it's a non-trivial thing to do, and we have a million non-trivial things on our list to do |
| 18:06.51 | dlynes_home | russellb: and what i said :) |
| 18:06.58 | russellb | yes, that too |
| 18:07.51 | justinu | yay! |
| 18:07.55 | *** join/#asterisk topping (n=topping@207.47.6.245.static.nextweb.net) |
| 18:08.01 | russellb | ;) |
| 18:08.10 | *** join/#asterisk ToTo (n=ToTo@host88-86.pool8256.interbusiness.it) |
| 18:08.17 | *** join/#asterisk Qwell (n=north@unaffiliated/qwell) |
| 18:08.37 | justinu | now /that's/ customer service |
| 18:08.56 | dlynes_home | russellb: btw...for that app_voicemail2.c do I just submit the entire thing after I'm done, or should I submit it in stages? |
| 18:09.22 | dlynes_home | russellb: the entire thing after I'm done, I'm guessing right? |
| 18:10.03 | *** join/#asterisk JakBeatZ (n=JakBeatZ@trek.tor1.ebit.ca) |
| 18:10.48 | JakBeatZ | Any current user experiences with asterisk + FreeBSD? Need to build a simple system with MoH, Meetme and a TDM400P with an FXO. |
| 18:11.58 | salviadud | freebsd is da' devil! |
| 18:12.08 | Nugget | if you need meetme I strongly encourge you to stick with linux for asterisk. |
| 18:12.10 | blitzrage | I don't think Zaptel works in FreeBSD |
| 18:12.17 | justinu | people make it work |
| 18:12.20 | Nugget | zaptel works in freebsd, but just barely. |
| 18:12.23 | blitzrage | I guess... |
| 18:12.33 | salviadud | better off using slackware |
| 18:12.34 | *** join/#asterisk mitcheloc (n=mitchelo@70-32-188-167.lmdaca.adelphia.net) |
| 18:12.36 | blitzrage | seems like a "make work" project |
| 18:12.37 | dlynes_home | blitzrage: it does, but it's only considered mostly stable for the wct4xxp driver |
| 18:12.39 | salviadud | if you like bsd that much |
| 18:12.40 | blitzrage | just use Linux and be done with it |
| 18:13.04 | blitzrage | no point in overcomplicating matters |
| 18:13.15 | Nugget | Yeah, it sucks but it really is the best way to go. |
| 18:13.23 | justinu | blitzrage: you can't insult people's religion like that man |
| 18:13.25 | justinu | :) |
| 18:13.25 | dlynes_home | well, i think freebsd sucks |
| 18:13.30 | dlynes_home | but that's my personal opinion |
| 18:13.58 | Nugget | if you don't need zaptel, then asterisk will happily run on just about any unix you can think of |
| 18:14.03 | dlynes_home | that crap about freebsd being more secure I think is just freebsd religion |
| 18:14.12 | [Airwolf] | dlynes_home, I'm using the system command to execute a script. |
| 18:14.22 | dlynes_home | [Airwolf]: now that truly is not secure :) |
| 18:14.25 | [Airwolf] | But the problem is, Asterisk says it is executing the system command. |
| 18:14.27 | Nugget | dlynes_home: freebsd has much better tools for securing the system against local attacks. |
| 18:14.35 | Nugget | for remote attacks I agree, the differences are minimal |
| 18:14.36 | [Airwolf] | But it doesn't execute the scru[t |
| 18:14.39 | [Airwolf] | script |
| 18:14.46 | [Airwolf] | And I don't know why |
| 18:14.47 | justinu | i think there's a key difference... linux is a kernel |
| 18:14.48 | blitzrage | justinu: I insult religion all the time :) |
| 18:14.50 | justinu | freebsd is an OS |
| 18:14.56 | dlynes_home | Nugget: and why are you letting someone try to access your system locally that's not authorized? |
| 18:15.01 | JakBeatZ | What are people's opinions of the different linux OS'? Gentoo vs. Slackware? |
| 18:15.08 | dlynes_home | Nugget: a padlock on the cage works just as well |
| 18:15.09 | JakBeatZ | (for asterisk, that is) |
| 18:15.18 | dlynes_home | JakBeatZ: slackware ownz all |
| 18:15.19 | justinu | JakBeatZ: people say use what you like |
| 18:15.19 | JakBeatZ | don't want to start a shitstorm :) |
| 18:15.23 | Nugget | dlynes_home: well, it could be that I sell shell service account, or it could be that I don't trust phpbb or other shitty code I run to keep people out. |
| 18:15.28 | Nugget | there are plenty of reasons |
| 18:15.36 | blitzrage | I use CentOS happily |
| 18:15.37 | dlynes_home | Nugget: i don't use php anywhere |
| 18:15.43 | blitzrage | php r0x |
| 18:15.45 | Nugget | good for you. tens of millions of people do. |
| 18:15.53 | Nugget | the point is still valid even if you don't use php |
| 18:15.58 | Nugget | that's just a single example |
| 18:15.59 | *** join/#asterisk bruno_asterisk (n=root@200.218.180.35) |
| 18:16.06 | dlynes_home | Nugget: yeah, but you can throw them all into UML |
| 18:16.29 | Nugget | yes, that's an approach that does add another layer of security |
| 18:16.45 | dlynes_home | Nugget: or you could chroot them into their home directories |
| 18:16.47 | JakBeatZ | well, I was using gentoo, but I really don't like it's system utilities.. compared to FreeBSD that is.. what's why I wanted to stick with FreeBSD, but if Linux is the cats ass for Asterisk, then I'll have to find a suitable distribution. |
| 18:16.48 | Nugget | but, the point still stands. freebsd provides better tools for protecting against local exploits. |
| 18:17.05 | JakBeatZ | *system management utilities, that is |
| 18:17.06 | dlynes_home | Nugget: yeah, but if that's the case you can just port those tools to linux, too |
| 18:17.23 | Nugget | good luck with that. if anyone ever does then the situation will be different |
| 18:17.44 | Nugget | I think it would be more of a challenge than you let on/think, though |
| 18:17.55 | salviadud | local exploits? come one, not everybody is an BOFH... |
| 18:17.56 | dlynes_home | I mainly just don't like freebsd ports tree, or the lack of all the commands i'm used to being able to use in linux |
| 18:18.10 | Nugget | what commands don't exist in freebsd? |
| 18:18.22 | asterboy | lfs rulez! |
| 18:18.29 | dlynes_home | Nugget: lemme see....lspci -v, cat /proc/..., ... |
| 18:18.36 | justinu | ~asterboy |
| 18:18.36 | jbot | methinks asterboy is a weed smoker |
| 18:18.40 | justinu | what's up dude? |
| 18:18.43 | dlynes_home | lol |
| 18:18.43 | Nugget | "proc" really isn't a "command" |
| 18:18.53 | dlynes_home | no, it's a filesystem |
| 18:18.59 | Nugget | and yeah, if you're used to the bastardized linux proc I can see how you'd miss it in other unixes. |
| 18:18.59 | Dr-Linux | http://durak.org/sean/pubs/kfc/ :P |
| 18:19.00 | JakBeatZ | dlynes_home: That's interesting.. I think exactly the opposite.. I can't find a linux equivilent to pkg_version to tell me new versions of packages compared to what's already installed.. I can't find that for gentoo.. does one exist for slackware? |
| 18:19.15 | dlynes_home | but it still gives me a lot of info that's easier to find than cat /proc/... |
| 18:19.28 | dlynes_home | erm sysctl or whatever the equivalent freebsd command is |
| 18:19.29 | JakBeatZ | Also, I really love periodic(8) but it doesn't seem to exist (or an equivalent) in Linux |
| 18:19.31 | justinu | JakBeatZ: centos has yum, debian based distros have apt-get |
| 18:19.49 | Nugget | and lspci is linux specific, but it's not like there isn't a totally equivalent command in freebsd. |
| 18:20.00 | salviadud | JakBeatZ, slackware has pkgtool |
| 18:20.07 | JakBeatZ | There is an lspci in freebsd |
| 18:20.10 | dlynes_home | salviadud: it's not the same thing |
| 18:20.10 | JakBeatZ | it's in ports |
| 18:20.21 | dlynes_home | JakBeatZ: ls /var/log/packages in asterisk |
| 18:20.28 | dlynes_home | erm slackware i mean |
| 18:20.30 | salviadud | i compile everything from source dammit, why would you want binaries? |
| 18:20.55 | dlynes_home | salviadud: i only compile from source once, and deploy binary packages everywhere else |
| 18:20.57 | Nugget | I think proc is a hideous, ugly, maldesigned mess. but now we're just talking preferences, not anything actually subjective or compelling. |
| 18:21.13 | bruno_asterisk | hi all! how can i Know if TE405P is dead? lspci is not finding it |
| 18:21.15 | salviadud | dlynes_home, are you using gentoo? |
| 18:21.15 | dlynes_home | Nugget: you can also use devfs |
| 18:21.17 | JakBeatZ | dlynes_home: that will just give a list of installed packages, right? That's not what I'm looking for |
| 18:21.21 | dlynes_home | salviadud: slackware |
| 18:21.30 | dlynes_home | JakBeatZ: correct |
| 18:21.30 | [Airwolf] | hmm |
| 18:21.33 | salviadud | i'm using slackware too |
| 18:21.35 | [Airwolf] | I hate this |
| 18:21.36 | JakBeatZ | salviadud: I compile everything from source too.. I don't use binaries. |
| 18:21.43 | justinu | hey nugget, i bought a macbook |
| 18:21.48 | justinu | for wife |
| 18:21.48 | Nugget | freebsd has a dynamic dev too. that's nothing unique |
| 18:21.49 | salviadud | how the hell do you compile from source on installation? |
| 18:21.52 | [Airwolf] | justinu, me too |
| 18:21.53 | dlynes_home | JakBeatZ: you're wanting to see if there's newer versions on the net of what you've got currently installed? |
| 18:21.58 | [Airwolf] | I'm working on it right now |
| 18:21.59 | [Airwolf] | :) |
| 18:22.04 | JakBeatZ | dlynes_home: Right, I need something that's going to compare installed packages to current versions and tell me what needs to be upgraded. FBSD has that built-in |
| 18:22.12 | *** join/#asterisk burizaa (n=freeee@cm107.omega96.maxonline.com.sg) |
| 18:22.14 | justinu | [Airwolf]: i took mine apart and redid the thermal grease on the heatpipe |
| 18:22.17 | dlynes_home | JakBeatZ: i.e. something like portsnap ; ports manager or whatever the command is in freebsd? |
| 18:22.31 | JakBeatZ | pkg_version is what I use in FBSD, yes. |
| 18:22.31 | [Airwolf] | justinu, do you have the pro or the normal one ? |
| 18:22.42 | justinu | white macbook 2.0ghz |
| 18:22.44 | [Airwolf] | Because I don't have any heat problems what so ever |
| 18:22.53 | dlynes_home | JakBeatZ: yeah...for something like taht in linux, you'd have to write a shell script i believe |
| 18:22.57 | justinu | mine still runs kinda hot when the CPUs are busy |
| 18:23.02 | salviadud | dlynes_home, is it possible to install slackware like a source distro? |
| 18:23.04 | justinu | but it's much cooler at idle now |
| 18:23.11 | JakBeatZ | Darn :( |
| 18:23.12 | Nugget | yay macbooks. |
| 18:23.19 | dlynes_home | salviadud: i don't follow you? |
| 18:23.20 | JakBeatZ | Ya, that's why I was looking for FreeBSD stability :( |
| 18:23.24 | Nugget | I've got a trip next week and it's really tempting to pick one up beforehand. |
| 18:23.24 | justinu | the fans don't kick in until the CPU gets above 70C or something |
| 18:23.31 | justinu | which seems ludicris |
| 18:23.42 | [Airwolf] | justinu, why ? |
| 18:23.42 | dlynes_home | ludicris is that rap star |
| 18:23.43 | JakBeatZ | For me, the freebsd system management utils are a little better than linux.. |
| 18:23.43 | salviadud | yeah, slackware has a bunch of packages at the beginning of the install |
| 18:23.48 | dlynes_home | i think you mean ludicrous :) |
| 18:23.53 | [Airwolf] | These core duo's can take 120c |
| 18:23.56 | salviadud | could you use the source cds to package everything yourself |
| 18:24.01 | salviadud | from source? |
| 18:24.03 | justinu | yeah, but my legs can't take 120C :) |
| 18:24.07 | dlynes_home | salviadud: yes |
| 18:24.12 | [Airwolf] | justinu, hehe |
| 18:24.16 | dlynes_home | salviadud: you mean to make your own binary distributions, right? |
| 18:24.22 | salviadud | dlynes_home, exactly |
| 18:24.22 | justinu | nugget: over all, an impressive machine. |
| 18:24.30 | dlynes_home | salviadud: yeah...entirely possible |
| 18:24.32 | JakBeatZ | Well, I have some time so I think I'm going to mess around with FreeBSD + Asterisk and see how it works |
| 18:24.33 | justinu | i really like the display too |
| 18:24.38 | salviadud | dlynes_home, i've never tried it, i don't know if its worth it though... |
| 18:24.39 | justinu | nice res, nice size, nice color |
| 18:24.47 | bruno_asterisk | Anyone has just installed a TE405P? I'm with a big problem or my motherboard does not work with the Digium card or My digium card is broken. LSPCI not show up the TE405P |
| 18:24.50 | dlynes_home | salviadud: i've made my own slackware distribution |
| 18:25.06 | salviadud | dlynes_home, what makes me curios is the fact that my boss installed gentoo, and it was fast |
| 18:25.10 | dlynes_home | salviadud: had to, so that the admin over in our China office could install everything easily |
| 18:25.11 | justinu | i'm not a fan of hugebooks |
| 18:25.23 | justinu | my laptop is a thinkpad x60s, and the macbook feels huge! |
| 18:25.27 | JakBeatZ | I need the screen realestate and the FW800 port |
| 18:25.28 | dlynes_home | salviadud: the admin in China was a Windows admin |
| 18:25.33 | JakBeatZ | I love my 17" powerbook |
| 18:26.04 | dlynes_home | salviadud: so I had to create a Slackware distro that would install Oracle 9i, KDE, Java, Tomcat, JBoss, and a bunch of other crap |
| 18:26.17 | Nugget | I've got a 15" powerbook that really ought to have been replaced a year ago. |
| 18:26.19 | justinu | i'm used to 1024x768, so the 1280x800 on the macbook seems like opulence |
| 18:26.42 | salviadud | dlynes_home, what are the necesary packages so i can build everything from scratch? |
| 18:26.51 | Nugget | I'm used to 2560x1600, so any laptop feels like I'm wearing shoes that are a size too small. |
| 18:27.02 | salviadud | dlynes_home, i've read its GCC and some other compiling packages, i don't remember |
| 18:27.10 | dlynes_home | salviadud: i think slackware only includes the source code for the build tools |
| 18:27.21 | dlynes_home | salviadud: erm for the packaging tools i mean |
| 18:27.26 | justinu | nugget: i know what you mean... i run dual 1600x1250 screens on the desktop |
| 18:27.45 | JakBeatZ | I came from Dell Inspiron lane where I had 1600x1200 resolution so I was always looking for screen realestate so to go down to 1400 on the powerbook was a sacrifice for me, but the payoffs of using OS X far outweighed the screen realestate. |
| 18:27.49 | JakBeatZ | *land |
| 18:27.49 | dlynes_home | salviadud: if you want the source code for all the packages, you'll have to download the source code, and use the packaging scripts |
| 18:27.54 | justinu | but yeah... OSX is running really nicely on the core duo |
| 18:28.17 | justinu | i run windows inside parallels for a few things, and windows runs fast |
| 18:28.21 | dlynes_home | salviadud: anyways..gotta run...gotta get a couple pbxes set up and a backup completed |
| 18:28.30 | justinu | work! |
| 18:28.31 | Nugget | just shut up about the macbook, justinu. You're going to make me go to Fry's and buy one. |
| 18:28.36 | salviadud | dlynes_home, sure thing, thanx for the info |
| 18:28.36 | justinu | the curse of the drinking man! |
| 18:28.44 | Nugget | I'm dreading taking this TiBook to London |
| 18:28.51 | justinu | nugget: i'm responsible for a lot of people spending a lot of money :P |
| 18:28.59 | justinu | you should get one |
| 18:29.09 | justinu | i just sold my 15" Ti 667 |
| 18:29.50 | *** join/#asterisk lorinc (n=ang@caracas-3785.adsl.interware.hu) |
| 18:30.18 | justinu | oh, the superdrive seems to be picky about ejecting certain discs |
| 18:30.21 | justinu | dunno what that's about |
| 18:32.26 | *** join/#asterisk jeffpc (n=jeffpc@ool-18ba4c2d.dyn.optonline.net) |
| 18:33.24 | *** join/#asterisk mjh001 (n=Michael@c-68-37-78-102.hsd1.nj.comcast.net) |
| 18:34.18 | jeffpc | hrm |
| 18:34.29 | bruno_asterisk | Has anyone done Oracle Database Integration with asterisk? |
| 18:34.55 | jeffpc | for whatever reason, I'm getting "No authority found" with my psql/realtime setup |
| 18:35.07 | bruno_asterisk | or tryed |
| 18:35.22 | chandi | hey guys, I need help from someone with a good imagination ;) I want to use "Dial" command to call a number. I want it to hang up if the callee has NOT sent a DTMF within 15secondes. How can I do that ? I might be willing to hack the code if it's impossible |
| 18:36.20 | salviadud | mmmmm |
| 18:36.27 | salviadud | blank background |
| 18:36.32 | salviadud | with 15 second timeout |
| 18:36.42 | chandi | salviadud yup ;) |
| 18:37.15 | chandi | I just want to detect if it's me that answers or if it's my cell phone provider's voice message that says I'm unavailable |
| 18:37.28 | justinu | it's possible, but you will need to modify the code |
| 18:37.30 | chandi | to transfer it to *'s voicemail |
| 18:38.02 | justinu | there's an app included in openpbx called app_icd |
| 18:38.06 | justinu | intelligent call distributer |
| 18:38.12 | justinu | maybe more suited to what you want to do |
| 18:38.22 | *** join/#asterisk lilo (i=levin@freenode/staff/pdpc.levin) |
| 18:38.23 | chandi | justinu I'm actually reading app_dial.c and I wanted to insert a timeout from the t option that transfers a call if * is pressed |
| 18:38.37 | justinu | good luck |
| 18:38.41 | chandi | hahaha thanks |
| 18:38.45 | justinu | app_dial is a nightmare |
| 18:39.18 | chandi | justinu but the code that "receives" and deals with the * dtmf is not in app_dial. I'm trying to find it. But I don't want to end up working in 10 different source files |
| 18:39.24 | *** join/#asterisk archvile (n=fdsf@70.88.61.54) |
| 18:39.34 | justinu | yeah, all the dsp stuff is in dsp.c |
| 18:39.45 | justinu | but there's a pseudo api that app_dial uses to talk to it |
| 18:39.51 | justinu | so just try and work thru that API |
| 18:40.09 | archvile | is there anyway to specifiy the timeout range for the trunk? for some reason i keep getting timeouts |
| 18:40.20 | chandi | justinu ok... but I'm trying to find where that dsp.c would tell app_dial that it has received that DTMF |
| 18:40.32 | justinu | chandi: that happens thru control frames, iirc |
| 18:41.06 | chandi | justinu ok. Is app_dial running for all the duration of the call or is it only initiating it ? |
| 18:41.26 | justinu | it runs for the duration in most situations |
| 18:42.17 | chandi | justinu okk.. and what do these control frames look like in the code ? |
| 18:42.57 | justinu | afaik, what happens is this: app_dial realizes that you want to attach a DSP DTMF decoder to the channel (assuming inband DTMF), so it calls a function that patches the dsp code into the incoming RTP stream |
| 18:43.33 | justinu | it's actually technology agnostic, so there's some internal representation of audio |
| 18:43.33 | chandi | justinu ok..mine is inband |
| 18:44.08 | chandi | justinu I don't actutally want to work with that representation, only the signals that are sent to app_dial |
| 18:44.13 | justinu | whenver that DSP code detects a digit, it sends DTMF control frames to the file descriptor the channel driver is reading from |
| 18:44.57 | justinu | case AST_CONTROL_DTMF: |
| 18:45.02 | justinu | try looking for that |
| 18:45.11 | justinu | err AST_FRAME_DTMF |
| 18:45.23 | chandi | justin ok.. 1 sec |
| 18:45.37 | justinu | you might ask about this kinda stuff on the #asterisk-dev channel |
| 18:45.53 | justinu | maybe you'll have better luck getting an explanation that I did |
| 18:46.23 | jeffpc | any why I would get "No authority found" with my psql/realtime |
| 18:46.28 | jeffpc | ? |
| 18:46.37 | jeffpc | it happens when I want to place a call |
| 18:46.47 | jeffpc | as far as I can tell, registration works well |
| 18:46.50 | chandi | justinu well.. you've been really helpfull! thanks! I'm going to read the code now |
| 18:47.11 | justinu | no prob, good luck |
| 18:48.39 | chandi | justinu the AST_FRAME_DTMF seems to be the dtmf coming from the caller not from the callee. Have you got any idea ? |
| 18:48.56 | archvile | does anyone where to specifiy the timeout range for the trunk? |
| 18:49.01 | justinu | it just depends on which channel you're reading from. |
| 18:49.05 | justinu | the inbound caller, or outbound caller channel |
| 18:49.11 | burizaa | how do i detect which codec in use ? |
| 18:49.38 | chandi | justinu in app_dial it seems to be used only for the caller's channel.. I'll keep on reading ;) |
| 18:49.38 | justinu | chandi: i've spent a fair bit of time trying to figure out what's going on in app_dial, but it's still not very clear to me |
| 18:50.35 | chandi | justinu ahaha ;) |
| 18:50.55 | justinu | one of the problem is bad variable names |
| 18:51.32 | justinu | alot of this stuff needs a refactoring that no one has the time, skill, or motivation to do i guess |
| 18:52.59 | justinu | chandi: search for the line "f = ast_read(winner);" |
| 18:53.28 | justinu | i believe that is where app_dial is reading from the outbound call, forwarding control indications and voice frames to the inbound call |
| 18:53.39 | chandi | justinu ohh thanks |
| 18:54.59 | *** join/#asterisk angom_h (n=angom@200.76.230.166) |
| 18:58.48 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.220) |
| 19:03.19 | *** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com) |
| 19:04.19 | jeffpc | what's "restricting registation for peer 'jeff' to 60 seconds (requested 300)" ? |
| 19:04.36 | justinu | do you have maxexpiry=60 set in sip.conf? |
| 19:04.42 | Tili | is it possible in SIP to re-route RTP after some specific time to another gateway. even though first time RTP stream is opened to first gateway |
| 19:04.54 | chandi | justinu the only place I see something about the flags is line 1556 of svn 1.2 : if (ast_test_flag(peerflags, OPT_CALLEE_TRANSFER)) |
| 19:04.58 | chandi | <PROTECTED> |
| 19:05.00 | *** part/#asterisk mitcheloc (n=mitchelo@70-32-188-167.lmdaca.adelphia.net) |
| 19:05.01 | chandi | that's it |
| 19:05.01 | jeffpc | justinu: I don't use sip |
| 19:05.04 | justinu | tili: yes, that's called a reinvite. but you can't make it happen after a specific time. |
| 19:05.12 | jeffpc | justinu: only iax |
| 19:05.19 | justinu | jeffpc: then check for the same thing in iax.conf |
| 19:05.24 | chandi | justinu that's weird. |
| 19:05.27 | Tili | jutinu: so if RTP is opened it is opened. later cant change path |
| 19:05.42 | justinu | chandi: the problem with those AST_FEATURE things is that they only apply after the call is bridged |
| 19:05.49 | chandi | justinu it only detects '*' from the caller in app_dial. I don't get it |
| 19:06.01 | justinu | tili: it can... like if you transfer the call, etc. |
| 19:06.09 | chandi | justinu so it's not app_dial that deals with it anymore, is that what you're saying ? |
| 19:06.16 | justinu | after the call is bridged no |
| 19:06.18 | Tili | jutinu: yeah but still be in control of call to monitor it |
| 19:06.22 | justinu | that happens is res_features.c, i believe. |
| 19:06.29 | justinu | tili: then you can't do reinvites |
| 19:06.34 | chandi | justinu okkk |
| 19:06.43 | jeffpc | justinu: ah, that did it |
| 19:06.44 | jeffpc | thanks |
| 19:06.45 | justinu | chandi: however, as I understand it, you want to do this before the call is bridged, no? |
| 19:06.51 | Tili | yeah so even if i transfer call media will go through Gateway |
| 19:06.59 | justinu | tili: if you're recording, yeah. |
| 19:07.02 | Tili | the one where it first came |
| 19:07.09 | Tili | no we dont want to record |
| 19:07.21 | Tili | just need to monitor call hangup |
| 19:07.26 | justinu | oh, that's different. |
| 19:07.29 | Tili | here is situatioon |
| 19:07.34 | justinu | in SIP, signalling and media plane are two different things. |
| 19:07.46 | justinu | so media can not pass thru asterisk, yet you can still monitor for hangup. |
| 19:07.58 | Tili | jutinu: I know that |
| 19:08.00 | Tili | listen to this |
| 19:08.21 | *** join/#asterisk TripleFFFF (n=Miranda@147-102.mc.cite.net) |
| 19:08.31 | chandi | justinu nope, after it is. I want it to happen within the first 15 seconds after the call is bridged |
| 19:09.08 | justinu | ok... then start checking into res_features, i believe. |
| 19:09.27 | chandi | justinu thanks! |
| 19:09.47 | TripleFFFF | an y regex expert ?i need a way to match any number...as in 555-1212 up yo 1-(555)-555-1212 |
| 19:09.52 | Tili | Call comes in to gateway A adn RTP is opened to play prompt. after prompt I want RTP to go to Gateway B. |
| 19:10.03 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-88-154-17-113.red.bezeqint.net) |
| 19:10.15 | justinu | tili then set reinvite=yes for the peer entries |
| 19:10.24 | justinu | er canreinvite=yes |
| 19:10.51 | Tili | jutinu: but then will it be able change RTP path dynamically from what it was initially and what it is later |
| 19:11.32 | justinu | yeah, after * plays the prompt, it'll reinvite itself out of the RTP path. |
| 19:11.52 | justinu | as far as reinviting itself INTO the media path, i've never experimented with that. |
| 19:12.03 | justinu | s/INTO/back INTO/ |
| 19:12.08 | Tili | no |
| 19:12.13 | Tili | back into may not be possible |
| 19:12.28 | Tili | but in such case it will receive the SIP headers |
| 19:12.31 | justinu | yes |
| 19:13.32 | Tili | now the point is how I tell asterisk to send re-invite after it has accepted the call |
| 19:14.27 | justinu | happens automatically |
| 19:15.13 | jeffpc | bleh, I removed all the realtime/psql bits, and it still gives me "no authority found" when I try to dial locally |
| 19:16.01 | justinu | trying to dial an IAX peer? |
| 19:16.29 | jeffpc | just iaxcomm -> asterisk extension with a simple playback |
| 19:16.53 | justinu | ok, your entry in iax.conf for iaxcomm probably doesn't have the right context set. |
| 19:17.48 | *** join/#asterisk Blackvel (n=blackvel@dslb-084-057-068-063.pools.arcor-ip.net) |
| 19:18.11 | Blackvel | hi all |
| 19:18.37 | Tili | jutinu: I want to play a prompt from Gateway A and then after prompt want SIP client to start sending RTP to B instead of A anymore |
| 19:19.13 | justinu | then call B w/ app_dial after playing the prompt |
| 19:19.33 | jeffpc | context looks fine.. |
| 19:19.53 | *** join/#asterisk lorinc (n=ang@caracas-0983.adsl.interware.hu) |
| 19:20.10 | justinu | jeffpc: i'm not very good with iax... maybe someone else here knows more about the basics of getting it running |
| 19:20.29 | jeffpc | :) |
| 19:20.38 | jeffpc | I'll try to poke around the iax config file |
| 19:20.40 | *** join/#asterisk Assid (i=assid@203.115.83.214) |
| 19:20.49 | *** join/#asterisk sfbosch (n=sfbosch@dsl-cap-66-18-218-36-cgy.nucleus.com) |
| 19:20.58 | sfbosch | hello |
| 19:21.13 | justinu | i remember seeing that same problem (no authority found), and it had something to do with sending the call into an invalid context, or context that iax user didn't have access too |
| 19:21.22 | justinu | but it's been a while, and my memory sucks |
| 19:21.24 | sfbosch | I am getting TDM PCI Master Abort messages and my system locks up hard |
| 19:21.39 | jeffpc | justinu: ok, thanks |
| 19:22.01 | *** join/#asterisk Fanatic (n=fanatic@c-68-82-43-225.hsd1.de.comcast.net) |
| 19:22.16 | justinu | sfbosch: check for IRQ conflicts, try a different slot, etc. |
| 19:22.32 | sfbosch | justinu: have you seen this before? Do those measures help? |
| 19:22.48 | justinu | never that specific problem |
| 19:22.58 | justinu | but yeah, IRQ conflicts with digium wildcards is common issue |
| 19:23.22 | sfbosch | okay, I will try moving the card; I have already tried manually assigning an IRQ |
| 19:23.30 | justinu | cat /proc/interrupts |
| 19:23.47 | justinu | make sure the kernel module for your TDM card isn't sharing an IRQ with something else. |
| 19:23.52 | sfbosch | okay |
| 19:24.48 | justinu | IME, running the SMP kernel (even on a single CPU machine) can help with IRQ conflicts |
| 19:25.03 | justinu | since it activates the APIC support in linux, i guess |
| 19:25.32 | Blackvel | is anyone running bristuff RC8r? |
| 19:26.08 | Blackvel | looks like there are some configuration changes. |
| 19:26.09 | Blackvel | J |
| 19:26.15 | Tili | jutinu: then all stream goes via A and then B |
| 19:26.37 | Blackvel | I get the error when starting asterisk: WARNING[10222]: chan_zap.c:955 zt_open: Unable to specify channel 1: No such device or address |
| 19:27.24 | jeffpc | sanity check...context=default in iax.conf will make all the calls made by that user start in [default] in extensions.conf, correct? |
| 19:27.47 | justinu | yeah, at least thats how it works in the SIP world |
| 19:28.35 | jeffpc | it's the same for both, IIRC |
| 19:29.46 | Tili | jutinu: yeah that is what I dont want. |
| 19:30.23 | justinu | i dunno then |
| 19:30.25 | Tili | ok is there anyway to to play sound while ringing. I know in territorial networks some telcos do that. they play message like THE NUMBER IS BUSY when the numebr is busy instead of tudu tudu tudu ring |
| 19:30.45 | justinu | yes, that is called SIP Early Media |
| 19:30.48 | Tili | in such case they dont actually pickup the call |
| 19:31.07 | justinu | media that is played before the 200 OK |
| 19:31.14 | Tili | justinu: yes exactly. how do we control that in asterisk. |
| 19:31.14 | justinu | triggered by a 183 Session Progress |
| 19:31.22 | Tili | yeah during 183 session in progress or 100 Trying |
| 19:31.31 | Tili | ok we are now coming on smae level |
| 19:31.33 | Tili | ummm |
| 19:31.52 | justinu | try using Playback(prompt|noanswer) |
| 19:31.54 | Tili | Now I wonder if SIP client can send DTMF during this also |
| 19:32.00 | justinu | as the first command in your dialplan entry |
| 19:32.26 | justinu | however, it will /not/ work if your provider isn't sending RTP to you during early media phase |
| 19:32.34 | justinu | asterisk cannot generate one way RTP without patching. |
| 19:33.02 | Blackvel | oh, I fixed my problem with bristuff 8r |
| 19:33.14 | Blackvel | I forgot to call ztcfg before starting asterisk :) |
| 19:33.17 | justinu | :P |
| 19:33.43 | Tili | jutinu: yeah. but that I can fix may be in * |
| 19:33.50 | Tili | or is there patch available for that? |
| 19:33.56 | justinu | yes |
| 19:33.58 | justinu | there is a patch. |
| 19:33.59 | Blackvel | really NEVER do computer stuff after fitness training :) |
| 19:34.09 | justinu | search for "Async RTP" on the bugs.digium.com site |
| 19:34.14 | justinu | i use the patch sucessfully |
| 19:35.30 | *** join/#asterisk AltnTab (n=ecs@nrjsoft13.networx-bg.com) |
| 19:36.33 | Tili | justinu: thanks a lot man. |
| 19:36.36 | justinu | no prob |
| 19:39.16 | Blackvel | how lives in san diego or LA? |
| 19:39.26 | justinu | i'm in LA |
| 19:39.42 | justinu | it sucks, 100F outside now |
| 19:41.16 | *** join/#asterisk ToTo (n=ToTo@host20-145.pool870.interbusiness.it) |
| 19:41.40 | Qwell | justinu: feels like shit outside |
| 19:42.02 | justinu | heh |
| 19:42.38 | *** join/#asterisk ambriento (n=ambrient@www.cobranet.com.br) |
| 19:43.32 | jeffpc | argh |
| 19:43.44 | jeffpc | justinu: it was peer vs. friend :) |
| 19:44.04 | justinu | jeffpc: ahh... nice and intuitive, isn't it :) |
| 19:44.24 | justinu | sounds good |
| 19:44.26 | jeffpc | justinu: I thought I used to use peer for everything.. |
| 19:44.34 | jeffpc | Qwell: :) |
| 19:44.49 | justinu | i just started using friend for everything |
| 19:45.05 | blitzrage | I never use friend -- I just use peer |
| 19:46.20 | MikeJ[Laptop] | blitzrage, I thought I was your friend? |
| 19:46.38 | blitzrage | MikeJ[Laptop]: pfft |
| 19:46.41 | sfbosch | justinu: okay, I switched to PIC in the BIOS, then forced the TDM-400 PCI slot to IRQ11 and restarted |
| 19:46.44 | MikeJ[Laptop] | :( |
| 19:46.46 | blitzrage | MikeJ[Laptop]: if you bring rye -- then yes :) |
| 19:46.47 | sfbosch | same behaviour |
| 19:46.50 | Qwell | MikeJ[Laptop]: denied |
| 19:46.57 | MikeJ[Laptop] | so you do use friends... |
| 19:47.04 | blitzrage | IRQ11? isn't that used by like... everything? |
| 19:47.04 | MikeJ[Laptop] | for their rye |
| 19:47.07 | jeffpc | :) |
| 19:47.12 | sfbosch | When I try to load the FreePBX home page, the card locks |
| 19:47.21 | sfbosch | IRQ11 is not in use, no |
| 19:47.22 | Qwell | sfbosch: see topic |
| 19:47.33 | jeffpc | Qwell: btw, thanks for reminding me to watch some more stargate sg-1 :) |
| 19:47.33 | blitzrage | freePBX? please go to #freepbx |
| 19:47.40 | blitzrage | sg-1? ewww |
| 19:47.40 | Qwell | jeffgus: eh? |
| 19:47.46 | sfbosch | guys, this has nothing to do with FreePBX |
| 19:47.49 | sfbosch | the card is locking up |
| 19:47.50 | MikeJ[Laptop] | freepbx haters here? |
| 19:47.58 | jeffpc | blitzrage: what? |
| 19:48.04 | sfbosch | I get a "TDM PCI Master Abort" message |
| 19:48.06 | MikeJ[Laptop] | sfbosch, what does locking up mean? |
| 19:48.06 | blitzrage | ? |
| 19:48.18 | sfbosch | I get the above noted message on the console |
| 19:48.32 | sfbosch | I cannot break out of it, I lose network connectivity, everything |
| 19:48.34 | MikeJ[Laptop] | when you start asterisk? |
| 19:48.37 | sfbosch | I have to hard reset the machine |
| 19:48.45 | sfbosch | no, asterisk starts okay |
| 19:48.50 | blitzrage | wierd... when hidd shuts down... it segfaults |
| 19:48.55 | sfbosch | it happens whenever I try to change configurations |
| 19:48.57 | blitzrage | I don't even know what hidd is :) |
| 19:48.59 | MikeJ[Laptop] | trying to figure out what that has to do with starting a webpage |
| 19:49.03 | Qwell | blitzrage: hid or hidd? |
| 19:49.08 | blitzrage | hidd |
| 19:49.09 | sfbosch | It varies |
| 19:49.14 | Qwell | never heard of it |
| 19:49.17 | blitzrage | ditto |
| 19:49.24 | sfbosch | sometimes I get as far as the Setup option in FreePBX, but when I try to apply changes, bang |
| 19:49.29 | MikeJ[Laptop] | sfbosch, what specifically are you doing in asterisk when it does that? |
| 19:49.37 | *** join/#asterisk lilo (i=levin@freenode/staff/pdpc.levin) |
| 19:49.39 | Qwell | sfbosch: #freepbd :p |
| 19:49.43 | Qwell | freepbx too |
| 19:49.56 | sfbosch | Qwell: thanks for the tip |
| 19:49.59 | sfbosch | Now, as I was saying |
| 19:50.13 | Qwell | ~amp |
| 19:50.14 | jbot | from memory, amp is NOT supported here! People using it should join #freepbx (FreePBX is the new name of AMP) |
| 19:50.17 | *** join/#asterisk somegeek (i=levin@tor/regular/somegeek) |
| 19:50.27 | sfbosch | I am trying to add an inbound route |
| 19:50.29 | justinu | HIDD? sounds like something for usb devices |
| 19:50.35 | Qwell | justinu: That'd be hid |
| 19:50.41 | Qwell | human interface device |
| 19:50.47 | justinu | so it couldn't be the HID daemon? |
| 19:50.49 | MikeJ[Laptop] | sfbosch, does it happen if you just do a reload in asterisk console too? |
| 19:51.00 | sfbosch | yes, it has happened at least once there |
| 19:51.08 | sfbosch | I can try doing that again |
| 19:51.18 | sfbosch | let me restart the machine (it is locked up again) |
| 19:51.24 | MikeJ[Laptop] | after any config changes? |
| 19:51.34 | MikeJ[Laptop] | or just randomly when you do a reload it does that? |
| 19:52.20 | MikeJ[Laptop] | wassup w/ the freepbx hatred in here? |
| 19:52.30 | blitzrage | its not asterisk :) |
| 19:52.36 | blitzrage | you have to be hardcore :) |
| 19:52.45 | MikeJ[Laptop] | blah |
| 19:52.48 | MikeJ[Laptop] | that's lame. |
| 19:52.53 | blitzrage | down with pants! up with skirts! |
| 19:52.59 | MikeJ[Laptop] | heh |
| 19:53.02 | *** join/#asterisk Dr-Linux (n=Linux@202.59.73.131) |
| 19:53.09 | justinu | no one wants to support the freepbx generated dialplans |
| 19:53.10 | Qwell | even Dr-Linux doesn't use freepbx |
| 19:53.13 | tzafrir_laptop | MikeJ[Laptop], my rule is: we answer Asterisk questions here. |
| 19:53.13 | Qwell | right Dr-Linux? |
| 19:53.13 | justinu | i think that's the only bias |
| 19:53.25 | MikeJ[Laptop] | how bout zaptel ? |
| 19:53.25 | Qwell | justinu: dialplans, and configs |
| 19:53.30 | Qwell | MikeJ[Laptop]: ^ |
| 19:53.38 | sfbosch | MikeJ: Okay, the machine has been restarted |
| 19:53.41 | tzafrir_laptop | So if you have a freebpx system, feel free to ask Asterisk questions but not freepbx questions |
| 19:53.43 | MikeJ[Laptop] | I understand not dealing with configuring freepbx |
| 19:53.51 | MikeJ[Laptop] | I don't know how to either |
| 19:53.58 | sfbosch | we're not talking about FreePBX |
| 19:54.05 | sfbosch | We are talking about a hardware problem |
| 19:54.06 | MikeJ[Laptop] | but the guy is having an issue with zaptel choaking. |
| 19:54.10 | sfbosch | yes, exactly |
| 19:54.13 | Dr-Linux | freepbx just gives bunch of shit macros , and it's setting takeover you configs |
| 19:54.14 | sfbosch | so sod off |
| 19:54.22 | Qwell | MikeJ[Laptop]: ever seen the zaptel configs generated by freepbx? :) |
| 19:54.36 | MikeJ[Laptop] | yeah.. |
| 19:54.39 | sfbosch | This is a Trixbox install |
| 19:54.45 | sfbosch | FreePBX is just one part of it |
| 19:54.46 | MikeJ[Laptop] | the dialplans are not eiven that bad |
| 19:54.49 | Qwell | sfbosch: /msg trixter help |
| 19:54.57 | tzafrir_laptop | Qwell, actually they use an old version of my script |
| 19:54.58 | Qwell | MikeJ[Laptop]: They are if you have to follow the ratsnest of macros |
| 19:55.17 | MikeJ[Laptop] | to find a zaptel hardware problem... |
| 19:55.21 | MikeJ[Laptop] | I think not. |
| 19:55.22 | Qwell | heh |
| 19:55.36 | sfbosch | MikeJ: So, I can log into the box and get an asterisk console. |
| 19:55.40 | MikeJ[Laptop] | so.. sfbosch, what kind of cards ya got? |
| 19:55.50 | sfbosch | It's a TDM-400 -- the dev-kit one |
| 19:55.55 | MikeJ[Laptop] | k |
| 19:56.03 | sfbosch | there's also an intel ethernet card |
| 19:56.05 | Dr-Linux | who likes freepbx? :S |
| 19:56.09 | MikeJ[Laptop] | are you able to pass calls at all? |
| 19:56.14 | sfbosch | No |
| 19:56.19 | MikeJ[Laptop] | ok |
| 19:56.21 | MikeJ[Laptop] | lspci |
| 19:56.22 | sfbosch | I can't get to the point where I can configure the card |
| 19:56.31 | tzafrir_laptop | in the CLI, what do you get for: 'zap show channels' ? |
| 19:56.33 | MikeJ[Laptop] | what do you see? |
| 19:56.48 | tzafrir_laptop | ~pb |
| 19:56.49 | jbot | hmm... pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
| 19:57.00 | sfbosch | okay |
| 19:57.02 | sfbosch | hang tight |
| 19:57.03 | MikeJ[Laptop] | both questions :P |
| 19:58.25 | Dr-Linux | ~dict anticipation |
| 19:58.26 | sfbosch | MikeJ: http://pastebin.ca/61404 |
| 19:58.26 | tzafrir_laptop | BTW: if they have not fixed it, the version of genzaptelonf that they use has a broken TDM400p detection. I only fixed it recently |
| 19:59.01 | MikeJ[Laptop] | you can just do zaptel.conf manually, can't you? |
| 19:59.27 | *** join/#asterisk ToTo (n=ToTo@host20-145.pool870.interbusiness.it) |
| 19:59.34 | MikeJ[Laptop] | sfbosch, oky.. so the card is there... |
| 19:59.41 | MikeJ[Laptop] | what about your conf files. |
| 19:59.44 | MikeJ[Laptop] | zaptel.conf |
| 19:59.46 | tzafrir_laptop | sfbosch, next: modprobe wctdm |
| 20:00.02 | tzafrir_laptop | after that, what do you see in /proc/zaptel ? |
| 20:00.02 | MikeJ[Laptop] | stuff all setup in the conf files right? |
| 20:00.17 | tzafrir_laptop | (ignore errors. They may be due to incorrect zaptel.conf) |
| 20:00.19 | sfbosch | http://pastebin.ca/61407 |
| 20:00.30 | sfbosch | i added some things |
| 20:00.37 | sfbosch | checking conf files |
| 20:01.13 | tzafrir_laptop | asterisk sees the channels. Good. no need to mess with zaptel.conf etc. |
| 20:01.28 | tzafrir_laptop | Asterisk would have failed to load if those channels were invalid |
| 20:01.57 | sfbosch | tzafrir: So, don't bother with /proc/zaptel? |
| 20:02.10 | tzafrir_laptop | no |
| 20:02.21 | MikeJ[Laptop] | so when does it die on you? |
| 20:02.34 | sfbosch | I can log into trixbox |
| 20:02.42 | sfbosch | When I click on FreePBX, I get the FreePBX home page |
| 20:02.54 | sfbosch | the moment I click "Setup", it will die |
| 20:02.54 | MikeJ[Laptop] | shhh |
| 20:03.00 | sfbosch | lol |
| 20:03.02 | MikeJ[Laptop] | don't say the bad words around here |
| 20:03.05 | sfbosch | right |
| 20:03.08 | tzafrir_laptop | That's a freepbxquestion. |
| 20:03.09 | MikeJ[Laptop] | heh |
| 20:03.19 | Qwell | ask it as an asterisk question, and we might be able to help |
| 20:03.23 | tzafrir_laptop | not an asterisk q. |
| 20:03.25 | Qwell | remove freepbx from the equation |
| 20:03.43 | sfbosch | okay, I will now try to restart asterisk from the CLI |
| 20:03.44 | MikeJ[Laptop] | I am more thinking more about what's up w/ your PCI errors your getting |
| 20:03.48 | sfbosch | it locked up at least once doing that |
| 20:04.17 | sfbosch | let me give that a whirl |
| 20:05.57 | sfbosch | alright |
| 20:06.18 | sfbosch | how am I going to get this to work so that I can make outbound calls from my Polycom 501 through the PSTN interface on the TDM-400? |
| 20:06.34 | sfbosch | I'm prepared to do it from the console |
| 20:06.44 | MikeJ[Laptop] | add it to the dialplan |
| 20:06.54 | MikeJ[Laptop] | ~docs |
| 20:06.54 | jbot | i guess docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 20:07.07 | MikeJ[Laptop] | lots o info on basic config on the wiki |
| 20:07.18 | sfbosch | I know, I've read lots of it |
| 20:07.20 | tzafrir_laptop | (Actually freepbx makes this more difficult than it should be) |
| 20:07.22 | *** join/#asterisk mog_home (n=mogorman@68.62.237.103) |
| 20:07.23 | sfbosch | it's rather overwhelming |
| 20:07.29 | MikeJ[Laptop] | there are -additional files in freepbx that you can put your own stuff in that won't get run over |
| 20:07.30 | sfbosch | don't say that word |
| 20:07.37 | Dr-Linux | sfbosch: why don't you use simply asterisk new version 1.2.8? |
| 20:07.47 | sfbosch | I'll consider it |
| 20:07.53 | sfbosch | but i don't want to spend days messing around |
| 20:08.28 | tzafrir_laptop | sfbosch, I suggest you simply dump trixbox and start with a default installation of Asterisk. That is: if you want to have any chance of actually understanding what's happening |
| 20:08.28 | MikeJ[Laptop] | from #freepbx : <bduncan1975> okay...gotta linux question |
| 20:08.28 | MikeJ[Laptop] | <PROTECTED> |
| 20:08.28 | sfbosch | I like the Trixbox because I can start it from a livecd and have it up and running fast |
| 20:08.28 | MikeJ[Laptop] | :P |
| 20:08.28 | tzafrir_laptop | The dialplan of Freepbx is a complete mess |
| 20:08.34 | MikeJ[Laptop] | sfbosch, not this time :P |
| 20:08.42 | sfbosch | well, yeah |
| 20:08.53 | sfbosch | I'm a gentoo user and gentoo installs take time |
| 20:09.00 | sfbosch | but you're right, I'm wasting a lot of time here |
| 20:09.18 | Dr-Linux | MikeJ[Laptop]: thre is no #linux channel though :) |
| 20:09.23 | MikeJ[Laptop] | centos seems to get a lot of attention |
| 20:09.33 | Qwell | Dr-Linux: there is. It forwards to ##linux |
| 20:09.34 | MikeJ[Laptop] | Dr-Linux, that's funnier then |
| 20:09.46 | *** join/#asterisk mitcheloc (n=mitchelo@70-32-188-167.lmdaca.adelphia.net) |
| 20:09.52 | sfbosch | MikeJ: yes, but it's a redhat based distro |
| 20:09.53 | Dr-Linux | Qwell: yes it's then ##linux |
| 20:09.59 | sfbosch | i prefer source builds |
| 20:10.23 | MikeJ[Laptop] | and I prefer things to just work... |
| 20:10.35 | MikeJ[Laptop] | but we can't always get what we want now can we :P |
| 20:10.35 | sfbosch | so do i -- generally, source builds do |
| 20:10.40 | sfbosch | but they take longer |
| 20:10.56 | MikeJ[Laptop] | you can do your own asterisk source build onto a live cd :P |
| 20:11.10 | sfbosch | I think I'll try that, then |
| 20:11.28 | sfbosch | So, I guess you guys all detest Trixbox/AMP/FreePBX/Asterisk@Home? |
| 20:11.56 | Dr-Linux | Qwell: yes night i was facing a weird problem with international dialing. |
| 20:12.18 | MikeJ[Laptop] | no. |
| 20:12.20 | MikeJ[Laptop] | just linux |
| 20:12.22 | MikeJ[Laptop] | :P |
| 20:12.35 | sfbosch | hah. |
| 20:12.45 | *** join/#asterisk adorah (n=Asterjet@87.69.72.228) |
| 20:13.18 | chandi | Hi, i've got questions about Macros |
| 20:13.37 | chandi | I'm writing one that is ran from the dial application |
| 20:13.46 | chandi | somebody knows about macros ? ;) |
| 20:14.12 | Dr-Linux | chandi: yes , Qwell is master of Macros :) |
| 20:14.17 | chandi | greeat ;) |
| 20:14.18 | Qwell | lies |
| 20:14.39 | chandi | haha but qwell knows about what I'm actually working on |
| 20:14.44 | chandi | this is my small macro : |
| 20:14.51 | Qwell | ~pb |
| 20:14.51 | jbot | somebody said pb was a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
| 20:15.02 | chandi | ok |
| 20:15.04 | Dr-Linux | Qwell knows everything except dance ;) |
| 20:15.10 | Qwell | Dr-Linux: I can dance |
| 20:15.42 | Dr-Linux | Qwell: coool :) |
| 20:16.30 | Dr-Linux | chandi: what does your nick meaning? |
| 20:16.33 | Qwell | blitzrage: yes, that |
| 20:16.48 | *** join/#asterisk tsurk0 (n=tsurko@digsys226-159.pip.digsys.bg) |
| 20:16.48 | chandi | my macro : http://pastebin.com/756285 |
| 20:17.10 | Dr-Linux | some guys can dance good with girls, but can't do that alone |
| 20:17.22 | blitzrage | numbered priorities? eww :) |
| 20:17.42 | chandi | Dr-Linux : I've got 2 answers to give you ;) It actually means "Silver moon" and it's my first name :) |
| 20:18.04 | blitzrage | chandi: you don't need to set that globally -- the MACRO_RESULT will be availabel to the channel even outside the macro |
| 20:18.20 | chandi | Dr-Linux : it's in Sanskrit, the sacred language of india. I've been conceived in India |
| 20:18.31 | chandi | blitzrage ok, but the thing is that it never goes into timeout |
| 20:18.46 | blitzrage | you don't have a 't' extension |
| 20:18.48 | chandi | blitzrage : it always end up bridging the 2 channels |
| 20:19.06 | chandi | blitzrage line 13 and 14 ? |
| 20:19.11 | blitzrage | and you have a space after the comma on line 17 |
| 20:19.18 | blitzrage | oops -- missed it :) |
| 20:19.35 | Dr-Linux | chandi: is it something like: >> chandi , sona , mooti , hera ? :P |
| 20:19.36 | Qwell | and no space after exten on lines 2 through 9 |
| 20:20.03 | chandi | blitzrage should the 't' extension be in the context or in the macro ? |
| 20:20.14 | Qwell | chandi: in the macro, I'd think |
| 20:20.17 | chandi | Okkk spaces do change things |
| 20:20.42 | blitzrage | yah -- in the macro |
| 20:21.26 | Dr-Linux | chandi: i'm also Desi , that's why i asked ;) |
| 20:21.42 | chandi | Dr-Linux ahhhhhhh. great. Where are you from ? |
| 20:22.13 | chandi | Dr-Linux I've actually spent 6 months in India last year. I did take some Hindi classes but I forgot a lot of it since I'm not practicing a lot here ;) |
| 20:22.13 | Dr-Linux | but i don't think chandi could be a male name :S |
| 20:22.18 | Dr-Linux | chandi: Pak |
| 20:22.35 | justinu | Dr-Linux: how goes? |
| 20:22.38 | chandi | Dr-Linux hahaha It's been the name of a male poet (can poets be male?) |
| 20:22.39 | Assid | that would be chandni |
| 20:22.44 | chandi | Dr-Linux great! |
| 20:22.46 | chandi | ohh shit |
| 20:22.51 | Dr-Linux | justinu: hey my friend, |
| 20:22.52 | chandi | I've got a girl's name |
| 20:22.53 | chandi | :I |
| 20:23.04 | Dr-Linux | justinu: my fuckin yahoo doesn't work :( |
| 20:23.07 | justinu | Dr-Linux: how is your wife? |
| 20:23.38 | Dr-Linux | chandi: dude, chandi name is for a girl :) |
| 20:23.41 | *** join/#asterisk Mattwj2006 (n=Matt@user-12l3n74.cable.mindspring.com) |
| 20:23.53 | chandi | If I come back to my Macro... the lines that you've pointed me aren't the ones I've got trouble with :I |
| 20:24.01 | Mattwj2006 | hey guys how good is bluetooth support with linux? |
| 20:24.04 | chandi | Dr-Linux grrrrr shiiit... |
| 20:24.07 | Dr-Linux | justinu: i was just talking to her on phone, she is nice .. for bad thing is that, we are far away from each other |
| 20:24.14 | justinu | :( |
| 20:24.15 | Dr-Linux | justinu: how about jen? |
| 20:24.18 | chandi | Dr-Linux I've noticed pics of "Chandi Mason" on the web, what a girl |
| 20:24.20 | Mattwj2006 | I was thinking of getting bluetooth headset |
| 20:24.21 | justinu | jen is happy |
| 20:24.32 | *** join/#asterisk r0d3nt (n=RatMan@foster.stonedcoder.org) |
| 20:24.49 | Dr-Linux | justinu: great , and you? :) |
| 20:24.53 | blitzrage | chandi: agreed |
| 20:25.05 | Dr-Linux | chandi: believe me chandi is a girl name in Pak/Ind |
| 20:25.19 | justinu | Dr-Linux: here is a photo: http://justinu.smugmug.com/photos/67502740-L.jpg |
| 20:25.19 | chandi | Dr-Linux I believe you. I'm just a little ashamed :I |
| 20:25.51 | Dr-Linux | justinu: lemme see |
| 20:25.58 | Dr-Linux | chandi: where're you right now? |
| 20:26.16 | justinu | Dr-Linux: click the link |
| 20:26.24 | chandi | Dr-Linux Montreal, Canada. U ? |
| 20:26.26 | Mattwj2006 | I guess I could always use it in Windows ;) |
| 20:27.49 | Dr-Linux | justinu: wowwwwwww very nice, really i'm happy to see you both :) |
| 20:28.00 | justinu | :) |
| 20:28.01 | Dr-Linux | chandi: Pakistan |
| 20:28.23 | Dr-Linux | justinu: did you guys feel any change in the life? :) |
| 20:28.27 | justinu | not much |
| 20:28.30 | justinu | a little bit |
| 20:28.46 | chandi | Dr-Linux whereabout in Pakistan are you ? |
| 20:28.53 | Dr-Linux | justinu: hhm.. you will ... when you have a small cute Justin ;) |
| 20:29.03 | chandi | Dr-Linux I wanted to go last year but I got sick and had to go back home :( |
| 20:29.05 | Dr-Linux | chandi: Lahore |
| 20:29.06 | justinu | Dr-Linux: unknown at this time :) |
| 20:29.22 | Dr-Linux | chandi: to pakistan? |
| 20:29.23 | chandi | Dr-Linux a few friends of mine loved pakistan a lot |
| 20:29.26 | chandi | Dr-Linux yup!! |
| 20:29.36 | chandi | Dr-Linux but I got sick in India |
| 20:29.42 | Dr-Linux | justinu: hehe .. you won't, but she will be crazy to have one :) |
| 20:30.00 | Dr-Linux | chandi: pakistan is cool |
| 20:30.12 | Dr-Linux | chandi: but tribles rocks |
| 20:30.16 | justinu | Dr-Linux: she's not so crazy yet... we will see |
| 20:30.29 | Dr-Linux | justinu: i'll be waiting for you guys in Pakistan :) |
| 20:30.35 | justinu | i hope to visit someday |
| 20:30.44 | justinu | i have met many friends in lahore from the internet |
| 20:30.53 | justinu | lahore and islamabad |
| 20:31.17 | Dr-Linux | justinu: great, you should visit then :) |
| 20:31.29 | Dr-Linux | lemme save your pic, i'll show it to my fiance |
| 20:31.42 | justinu | cool |
| 20:32.48 | Dr-Linux | justinu: our all US employee is coming to pakistan this month, to clelebrate AGM |
| 20:32.56 | justinu | AGM? |
| 20:33.25 | Dr-Linux | Anual Grand Meeting |
| 20:33.50 | Dr-Linux | most of them are coming to pk for the first time |
| 20:34.53 | chandi | Dr-Linux I've been to the pakistaneese border not too far from Amritsar(india) to see the closing of the border ceremony. It was amazing fun |
| 20:35.45 | Dr-Linux | chandi: heh do you know India and pakistn are ..... you know :P |
| 20:35.48 | chandi | so.. does somebody understands why my macro never goes into timeout ? |
| 20:35.50 | justinu | Dr-Linux: that's cool... |
| 20:35.53 | chandi | Dr-Linux I know I know!!! |
| 20:36.48 | chandi | Dr-Linux I've read as most as I could about the creation of Pakistan |
| 20:36.56 | chandi | Dr-Linux and about kashmir |
| 20:36.56 | Dr-Linux | chandi: i live in Lhr , but i'm from Kohat |
| 20:37.26 | *** join/#asterisk chino (n=daquino@c-68-84-57-212.hsd1.nj.comcast.net) |
| 20:37.40 | Dr-Linux | chandi: yes, Kashmir is the main issue between pakistan and india |
| 20:38.25 | chandi | Dr-Linux what's the situation like now ? is there a ceasefire ? |
| 20:38.43 | chino | i have a linksys pap2t-na phone adapter but i cant figure out what ip address it has to configure it via its web app |
| 20:39.50 | *** join/#asterisk timscott (n=a@d198-53-23-18.abhsia.telus.net) |
| 20:39.53 | Dr-Linux | chandi: the situation is as always, but govt: do not let it open now. |
| 20:40.09 | *** join/#asterisk gcarrillog (n=gcarrill@201.155.92.48) |
| 20:40.11 | gcarrillog | hi |
| 20:40.13 | chandi | Dr-Linux ok.. |
| 20:40.20 | timscott | Hello there. :) |
| 20:40.23 | gcarrillog | alguien habla español? |
| 20:40.27 | gcarrillog | :) |
| 20:40.42 | Dr-Linux | gcarrillog: wtf did you just said? :S |
| 20:40.44 | timscott | No, sorry. :S |
| 20:41.06 | gcarrillog | xoks |
| 20:41.21 | timscott | socks? |
| 20:41.25 | gcarrillog | oks |
| 20:41.28 | timscott | :p |
| 20:41.30 | chandi | gcarrillog I speak french if you do |
| 20:41.30 | Dr-Linux | sucks? |
| 20:41.37 | gcarrillog | LOL |
| 20:41.44 | chandi | ;) |
| 20:41.50 | gcarrillog | thanks but i dont speak french |
| 20:41.59 | chandi | gcarrillog let's try mandarin then |
| 20:42.04 | gcarrillog | lol |
| 20:42.10 | *** join/#asterisk stf4449 (n=stf@HSE-Montreal-ppp133176.qc.sympatico.ca) |
| 20:42.14 | Dr-Linux | i prefer Punjabi and urdu , Pashtu will be much better ;) |
| 20:42.29 | gcarrillog | im trying configure asterisk |
| 20:42.36 | gcarrillog | i have a X100p on FreeBSD |
| 20:42.40 | chandi | gcarrillog we are all trying to do that :I ;) |
| 20:42.48 | gcarrillog | but xDD |
| 20:42.49 | chandi | it's a work in progress |
| 20:42.54 | Dr-Linux | lol |
| 20:42.56 | gcarrillog | ya |
| 20:43.20 | stf4449 | Anybody knows why what causes this after receiving a SIP INFO request "X-Asterisk-HangupCause: Normal Clearing" |
| 20:43.29 | gcarrillog | i have wekks trying with little advance |
| 20:44.15 | gcarrillog | i need documentacion more soft than asterisk handbook |
| 20:45.03 | Dr-Linux | gcarrillog: why don't you like a bit hard? |
| 20:45.29 | gcarrillog | i found a ebook of asterisk from o´reylli |
| 20:46.22 | gcarrillog | Dr-Linux i dont understand many tecnics clues |
| 20:46.35 | gcarrillog | i need some more basic |
| 20:46.47 | Dr-Linux | gcarrillog: it takes some time to understand |
| 20:47.38 | Dr-Linux | gcarrillog: TFOT or something is a best book for asterisk |
| 20:47.44 | Dr-Linux | ~book |
| 20:47.45 | jbot | [book] a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 20:47.50 | gcarrillog | http://www.nufone.net/downloads/asteriskdocs/AsteriskTFOT.zip |
| 20:47.52 | gcarrillog | :D |
| 20:48.05 | gcarrillog | so i will print that book |
| 20:48.19 | Dr-Linux | gcarrillog: this is one is nice, ever i found |
| 20:48.28 | gcarrillog | ok thanks |
| 20:50.39 | timscott | is there an html or .ods version of that anywhere? |
| 20:51.22 | timscott | of the Asterisk TFOT book, I mean. |
| 20:51.25 | MikeJ[Laptop] | go buy the book |
| 20:51.49 | MikeJ[Laptop] | or print the book.. and paypal the authors :P |
| 20:52.11 | *** part/#asterisk Mattwj2006 (n=Matt@user-12l3n74.cable.mindspring.com) |
| 20:52.37 | timscott | Can't afford it ATM. |
| 20:53.03 | gcarrillog | MikeJ[Laptop] thats good idea |
| 20:53.06 | Dr-Linux | timscott: why you need in html? |
| 20:53.11 | timscott | Don't need, want. |
| 20:53.16 | gcarrillog | because the send is too expensive |
| 20:53.21 | gcarrillog | more than book |
| 20:53.21 | gcarrillog | :S |
| 20:53.23 | timscott | I prefer to read in HTML format, as opposed to PDF... |
| 20:53.36 | gcarrillog | i dont have pda |
| 20:53.41 | timscott | then I can open it in my web browser, and have a whole ton of different pages open at the same time, etc. |
| 20:53.46 | Dr-Linux | PDF is very nice over the HTML text |
| 20:53.49 | timscott | but whatevs |
| 20:53.59 | timscott | Well, maybe PDF for printing, but I prefer HTML texts for reading online |
| 20:54.07 | gcarrillog | but you can read html in console |
| 20:54.08 | gcarrillog | :P |
| 20:54.11 | timscott | "online", ie on my computer screen |
| 20:54.29 | timscott | pdf2html anyone? |
| 20:54.31 | timscott | :) |
| 20:54.55 | gcarrillog | xD |
| 20:56.51 | Dr-Linux | timscott: download Acrobat Reader |
| 20:57.03 | timscott | I have it. |
| 20:57.07 | timscott | What about it? |
| 20:57.59 | Dr-Linux | timscott: but you said you dn't have PDF file reader? |
| 20:58.14 | timscott | No, I didn't say that |
| 20:59.08 | timscott | :p |
| 20:59.37 | Dr-Linux | what's pokes mean?:S |
| 20:59.45 | timscott | poke...like... |
| 20:59.54 | timscott | uhh |
| 21:00.05 | timscott | What's your first language? |
| 21:00.18 | stf4449 | anybody knows something about this -> "X-Asterisk-HangupCause: Normal Clearing" |
| 21:00.59 | Dr-Linux | timscott: Pashtu (Tribals Language) 2nd lang... is Urdu (Paki) |
| 21:01.09 | timscott | Dr-Linux, poke is like, "pushing someone with the end of my index finger" |
| 21:01.16 | timscott | does that make sense? |
| 21:01.23 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.220) |
| 21:01.26 | *** join/#asterisk r0d3nt|m (n=RatMan@ip68-108-184-243.lv.lv.cox.net) |
| 21:01.31 | dlynes_office | timscott: oh...i thought you meant in the sexual sense |
| 21:01.34 | timscott | oh |
| 21:01.35 | timscott | haha |
| 21:01.50 | Dr-Linux | :S |
| 21:01.53 | timscott | dlynes: working on a saturday? :S |
| 21:02.14 | dlynes_office | yeah...i work 7 days |
| 21:02.14 | timscott | ooch |
| 21:02.14 | Dr-Linux | timscott: my understanding is the same like dlynes_home |
| 21:02.15 | gcarrillog | :O |
| 21:02.29 | dlynes_office | setting up two new pbxes, and a backup softswitch |
| 21:02.37 | dlynes_office | and backing up and restoring another machine |
| 21:02.45 | timscott | I've got a question about symmetric vs. asymmetric RTP streams |
| 21:02.55 | timscott | what is the advantage of symmetric RTP streams? |
| 21:03.00 | timscott | what is the difference? |
| 21:03.07 | dlynes_office | timscott: are you trying to be funny? |
| 21:03.09 | timscott | No. |
| 21:03.14 | dlynes_office | symmetric == the same |
| 21:03.19 | dlynes_office | asymmetric == different |
| 21:03.20 | timscott | I know what the *words* mean. |
| 21:03.28 | dlynes_office | lol |
| 21:03.32 | timscott | I'm asking what is the difference in reference to SIP RTP streams. |
| 21:03.59 | timscott | anyone? |
| 21:04.00 | *** join/#asterisk feld (n=feld@ruc-mwt-gw-1.ruc.mwt.net) |
| 21:04.13 | dlynes_office | feld knows |
| 21:04.22 | timscott | dlynes: do you know? |
| 21:04.23 | feld | of course |
| 21:04.26 | timscott | ah, whatever. i'll ask google |
| 21:04.27 | feld | lol |
| 21:04.27 | dlynes_office | nope |
| 21:04.51 | dlynes_office | i'm the opinion that if it works, don't ask why it works..just be happy :) |
| 21:06.14 | dlynes_office | man, i abhor windows |
| 21:06.19 | timscott | Ah, I guess all it means is that it uses the same socket/port for sending and reciveing RTP traffic |
| 21:06.32 | timscott | when running symmetric, of course. |
| 21:06.37 | dlynes_office | ah |
| 21:06.53 | dlynes_office | and asymmetric uses different ports for sending than it does for receiving |
| 21:06.57 | timscott | mmhmm |
| 21:07.06 | dlynes_office | Dr-Linux: don't poke me there again, ghey boy |
| 21:07.23 | znoG | just wondering.. if I have 10 G729 licenses and all are in use, is there a fallback codec? |
| 21:07.34 | timscott | I usually just use gsm as my fallback codec |
| 21:07.52 | timscott | If you're in a bandwidth bind, and need g729, then you'll probably want to use gsm as your fallback |
| 21:09.16 | znoG | yeah, so basically disallow=all, allow=g729, allow=gsm ? |
| 21:09.29 | znoG | that's how you specify the fallback? |
| 21:09.35 | timscott | i'd say so, that's just how I did it |
| 21:11.01 | znoG | so basically if I convert all prompts from gsm to g729, and all my ATAs use G729, then in theory I wouldn't need that many licenses |
| 21:11.20 | timscott | I dunno how you're set up |
| 21:11.25 | timscott | that could make it better, or make it worse |
| 21:11.35 | znoG | except when I receive calls from FXO which then get transcoded to G729 and sent to the ATA |
| 21:11.50 | timscott | how is your system set up? |
| 21:12.06 | znoG | i have an asterisk box at each of the 2 branches |
| 21:12.26 | znoG | and a big asterisk box at the central branch which is connected to a Lucent Definity PBX through a TDM2400 card (many FXS/FXO ports) |
| 21:12.33 | *** join/#asterisk adker (n=adker@70-100-239-157.br1.glv.ny.frontiernet.net) |
| 21:13.00 | znoG | so when the branches want to call each other, they do it over the Internet (asterisk<->asterisk) |
| 21:13.13 | timscott | Is that where you're using g729? |
| 21:13.20 | znoG | if they want to call an extension that is connected to the Lucent PBX, the central branch Asterisk box calls out through the FXS modules |
| 21:13.21 | *** join/#asterisk [hC] (n=hardcore@S01060004e21ea953.vc.shawcable.net) |
| 21:13.29 | timscott | ah |
| 21:13.41 | znoG | well i want to use G729 mostly when the branches want to call out of their systems |
| 21:13.48 | timscott | so for the remote branches to call the Lucent, they have to go through the central one, eh? |
| 21:13.48 | znoG | ie. interbranch connections |
| 21:13.52 | znoG | yep |
| 21:14.11 | znoG | the central one is more or less a VoIP gateway for in/out calls to/from the lucent |
| 21:14.26 | timscott | well, you only need to use a g729 license for converting g729 into something else |
| 21:14.31 | znoG | thats right |
| 21:14.46 | znoG | so for the 2 branches (NOT the central one) there is no transcoding done |
| 21:14.51 | znoG | as its all G729 |
| 21:15.04 | znoG | but when they call the central branch, it would be transcoding there |
| 21:15.09 | timscott | so then you're probably better off recording all your prompts/voicemail/etc in g729 |
| 21:15.11 | znoG | so probably about 10 channels needed there |
| 21:15.14 | timscott | oh |
| 21:15.17 | znoG | yeah I'll be doing that |
| 21:15.37 | znoG | i'm currently using GSM but I thought G729 would give me better quality and less bandwidth usage |
| 21:15.54 | timscott | well, slightly better quality, and slightly less bandwidth use ;) |
| 21:15.55 | znoG | when both the outside branches call the central office during peak hours, i think the gain from using g729 over GSM will be noticeable |
| 21:15.57 | timscott | nothing to write home about. :) |
| 21:16.11 | timscott | if someone is calling through the lucent, they'd need codecs to talk to the branch offices |
| 21:16.17 | znoG | yep |
| 21:16.27 | timscott | is voicemail all run off the central server? |
| 21:16.31 | timscott | voicemail/prompts/etc |
| 21:16.38 | znoG | as it comes in as analog over FXO and needs to transcode to G729 |
| 21:16.51 | znoG | nope, each asterisk box at each branch has the voicemail |
| 21:16.51 | timscott | MOH is on the central server? |
| 21:16.56 | timscott | oh, okay, that's cool then |
| 21:16.57 | znoG | nope, on each server |
| 21:17.13 | timscott | i was gonna say, if it was all centralized, then you'd have to transcode to reach your MOH or voicemail ;) |
| 21:17.17 | znoG | if they had a dedicated pipe between the branches then I would have more stuff centralized |
| 21:17.19 | timscott | which would probably be a pain in the ass |
| 21:17.33 | znoG | but since they don't, I do it this way |
| 21:17.49 | timscott | do you expect to run out of codecs? |
| 21:17.54 | feld | Can someone give me an idea of difficulty here on a scale from 1-10? Asterisk setup, 1 server, voicemail, extensions for phones with a few that will be dynamic addresses. Real phone numbers needs to be called out, too. |
| 21:18.12 | timscott | Because if your prompts are recorded in g729, and you're out of licenses, you won't be able to decode them, iirc. |
| 21:18.15 | znoG | i put it together mainly using a complicated dial plan and a AGI script that I coded in Perl.. but I'll always wonder if there are better ways of doing what I'm doing |
| 21:18.40 | *** join/#asterisk topping (n=topping@209-204-141-95.dsl.static.sonic.net) |
| 21:18.50 | znoG | timscott: yeah, bit of a worry.. not sure if I'm gonna go ahead with this since I'm not sure how much better quality I'll get out of this |
| 21:18.56 | timscott | hee, there probably are, i'm sure someone else could find a better way, but not likely me ;) |
| 21:19.11 | timscott | well, even having a fallback codec won't really matter if your prompts are g729, since no one will be able to reach them |
| 21:19.23 | timscott | is anyone going to need to hear an IVR calling between branch offices? |
| 21:19.43 | timscott | or alternatively, the central office, are they going to need to hear an IVR on the branch machines? |
| 21:20.01 | timscott | or are they just calling phones like, directly, without going through an IVR at the branches? |
| 21:20.35 | *** part/#asterisk chino (n=daquino@c-68-84-57-212.hsd1.nj.comcast.net) |
| 21:20.59 | znoG | they don't usually go through the IVR |
| 21:21.01 | znoG | but sometimes they do |
| 21:21.03 | TripleFFFF | whatn the best way to start asterisk on boot ? in a screen |
| 21:21.08 | znoG | kinda hard to determine how many codecs they'll need |
| 21:21.32 | timscott | hmm |
| 21:22.09 | feld | TripleFFFF: it can be backgrounded and u can connect and disconnect to the asterisk console at will |
| 21:22.14 | timscott | well, you could maybe build an IVR in gsm, and force calls coming from the central server to the branches onto the gsm IVR. |
| 21:22.33 | timscott | that way, it would save you from having to use g729 until you actually connected with the party on the branches |
| 21:23.35 | timscott | just an idea. |
| 21:24.02 | TripleFFFF | ?? |
| 21:24.09 | TripleFFFF | i mean from freebsd |
| 21:24.11 | TripleFFFF | as in |
| 21:25.28 | TripleFFFF | as in /etc/rc.conf -> screen -d -m asterisk -vgc |
| 21:27.09 | TripleFFFF | meant /etc/rc.local |
| 21:31.11 | chandi | Hi guys, somebody good with macros ? |
| 21:33.02 | chandi | I've got this macro here that never goes into 't' timeout extension |
| 21:33.02 | chandi | http://pastebin.com/756441 |
| 21:35.19 | *** join/#asterisk Qwell (n=north@unaffiliated/qwell) |
| 21:38.50 | tzafrir_laptop | Asterisk is a daemon. Why run it in screen? |
| 21:39.05 | tzafrir_laptop | for a remote console, use asterisk -r |
| 21:39.30 | justinu | i run my remote consoles in screen |
| 21:39.30 | tzafrir_laptop | to tail logs, use tail -f /var/log/asterisk/full |
| 21:40.18 | tzafrir_laptop | screen is lousy at scrolling |
| 21:40.47 | tzafrir_laptop | you can't use the terminal's native scrolling. You must use screen's |
| 21:40.54 | timscott | chandi: sorry, i'm waiting to hear what the answer to your question is, but I can't answer it myself :S |
| 21:41.36 | justinu | oh, yeah... i don't mind screens history tho, i just crank it up to something like 50,000 lines |
| 21:41.44 | justinu | you can search back in it, cut/paste, etc. |
| 21:41.46 | justinu | kinda nice |
| 21:42.27 | tzafrir_laptop | but you can simply scroll up and down in an xterm |
| 21:42.40 | justinu | yeah, i have to hit ^A-ESC |
| 21:42.42 | justinu | then page up |
| 21:42.48 | tzafrir_laptop | You have to use the wierd ctrl-a-somethings |
| 21:42.49 | justinu | big deal :) |
| 21:44.01 | tzafrir_laptop | justinu, if you like that so much, run an asterisk -r in a screen session |
| 21:44.13 | justinu | i do |
| 21:44.15 | *** join/#asterisk Dr-Linux (n=Linux@202.59.73.131) |
| 21:44.20 | blitzrage | me too |
| 21:44.23 | justinu | one window on asterisk -r |
| 21:44.28 | Dr-Linux | me too |
| 21:44.32 | justinu | one on tail -f /var/asterisk/messages |
| 21:44.43 | justinu | etc |
| 21:45.27 | Dr-Linux | its not >> /var/log/asterisk/messages ? |
| 21:45.33 | justinu | yeah |
| 21:47.09 | gcarrillog | i ve already installed my x100p but when i call the number, they not respond |
| 21:47.36 | chandi | timscott I've just found out. you want to know how I made it work ? |
| 21:53.43 | timscott | chandi: yes please. :) |
| 21:54.15 | timscott | By all rights, I'm still learning about asterisk, so I'm always interested in seeing other people's problems and solutions. ;) |
| 21:54.58 | chandi | timscott 1 sec, I'll put it on pastebin |
| 21:55.23 | jhiver | hi all |
| 21:55.30 | jhiver | anybody gave a try to sipxpbx? |
| 21:55.33 | jhiver | any comments? |
| 21:57.11 | chandi | timscott : http://pastebin.com/756492 the highlited line is the line that made it work |
| 21:58.10 | timscott | oh yeah, that makes sense >_< |
| 21:58.11 | timscott | :D |
| 22:01.26 | tzafrir_laptop | jhiver, what issippbx? |
| 22:01.43 | dlynes_home | gcarrillog: ok, adn what does them not answering have to do with your x100p card? |
| 22:02.30 | Dr-Linux | can i use pattern in the GotoIf application? |
| 22:02.34 | jhiver | sipxpbx is a sip based pbx which looks nice |
| 22:02.42 | jhiver | it's open source too |
| 22:02.51 | jhiver | I'm downloading the VMWare image to give it a try |
| 22:02.59 | Qwell | jhiver: There is this other new open source pbx, that's pretty cool |
| 22:03.07 | Qwell | I think it's called asterisk, or something |
| 22:03.12 | jhiver | lol |
| 22:03.27 | dlynes_home | there's like ten different open source pbxes now |
| 22:03.38 | dlynes_home | and they all have their problems |
| 22:03.50 | dlynes_home | why doesn't everyone just focus their time and energy on one to make it better? |
| 22:03.53 | mitcheloc | and 8 of them are based on asterisk |
| 22:04.01 | jhiver | I was just wondering if somebody gave that one a try because if it's crap I don't want to be wasting my time with it :) |
| 22:04.06 | dlynes_home | mitcheloc: yeah, no kidding :) |
| 22:04.39 | dlynes_home | jhiver: but asterisk is good, it's relatively stable |
| 22:04.45 | jhiver | Asterisk is cool, sure :) |
| 22:04.55 | jhiver | although I've had my share of problems with it |
| 22:04.57 | dlynes_home | jhiver: it just needs some loving tender care, and some rewrites of some poorly written code |
| 22:05.02 | jhiver | maybe I should get the business edition :) |
| 22:05.05 | mitcheloc | uh you guys better say that or i'll have denon/ressellb kick you haha |
| 22:05.23 | jhiver | BTW do you know which version of * is used in the business edition? |
| 22:05.28 | dlynes_home | mitcheloc: ummm...russellb and denon would probably agree with me |
| 22:05.38 | jhiver | the problem also is that asterisk isn't terribly suited for what I want to do |
| 22:05.52 | jhiver | what I need is more of a proper softswitch than a pbx |
| 22:06.07 | dlynes_home | jhiver: then why don't you go with freeswitch, or yate? |
| 22:06.12 | Dr-Linux | Qwell: can i use pattern in the GotoIf() application, if not then what should i do |
| 22:06.12 | jhiver | but commercial softswitches are so expensive :) |
| 22:06.21 | Qwell | Dr-Linux: yep, $[] |
| 22:06.25 | jhiver | freeswitch doesn't look ready for production |
| 22:06.30 | dlynes_home | jhiver: neither one of them is really designed to be a pbx |
| 22:06.31 | Dr-Linux | exten => _91NXXXXXXXXX,1,GotoIf($["${CALLERID(num)}" = "_4XXX"]?5:2) |
| 22:06.35 | jhiver | I haven't heard about yate |
| 22:06.42 | Qwell | Dr-Linux: yep, exactly |
| 22:06.49 | dlynes_home | jhiver: yate's been around almost as long as asterisk...maybe longer |
| 22:06.53 | Qwell | or, wait, no |
| 22:06.58 | Qwell | not patterns like _4XXX |
| 22:06.59 | jhiver | Is it any good? |
| 22:07.03 | dlynes_home | jhiver: or if you want to roll your own, bayonne has a good sip stack |
| 22:07.13 | Dr-Linux | Qwell: then how can i use? :S |
| 22:07.14 | dlynes_home | jhiver: well, there's a lot of telcos using it |
| 22:07.19 | dlynes_home | jhiver: same with bayonne |
| 22:07.24 | dlynes_home | jhiver: that should tell you something |
| 22:07.32 | jhiver | cool |
| 22:07.37 | jhiver | I'm gonna look at it then :) |
| 22:07.39 | jhiver | thanks :) |
| 22:07.44 | dlynes_home | jhiver: try #bayonne and #yate |
| 22:09.00 | Dr-Linux | Qwell: i want to monitor only 4XXX extensions while they dialout. so i'm not understand what should i do :S |
| 22:09.03 | *** join/#asterisk WiredX (n=matthew@rnas.arach.net.au) |
| 22:09.35 | Dr-Linux | exten => _91NXXXXXXXXX,1,GotoIf($["${CALLERID(num)}" = "4040"]?5:2) << a single extension works for me |
| 22:09.47 | Dr-Linux | but now sure how can i put there a pattren |
| 22:11.14 | Dr-Linux | s/now/not |
| 22:12.49 | blitzrage | Dr-Linux: what do you mean? |
| 22:12.55 | blitzrage | ${EXTEN:1} ? |
| 22:13.16 | Hmmhesays | this game is impossible to fine |
| 22:13.17 | Hmmhesays | *find |
| 22:13.39 | Dr-Linux | blitzrage: |
| 22:13.50 | Dr-Linux | exten => _91NXXXXXXXXX,1,GotoIf($["${CALLERID(num)}" = "4040"]?5:2) << a single extension works for me |
| 22:14.14 | Dr-Linux | blitzrage: but i want patterns instead of 4040 extension |
| 22:14.33 | blitzrage | REGEX() ? |
| 22:14.37 | Dr-Linux | exten => _91NXXXXXXXXX,1,GotoIf($["${CALLERID(num)}" = "_4XXX"]?5:2) << like this |
| 22:14.48 | blitzrage | yah -- regular expressions |
| 22:15.07 | Dr-Linux | blitzrage: does it work, as i pasted in last? |
| 22:15.12 | blitzrage | no |
| 22:15.26 | Dr-Linux | blitzrage: so what should i do? :S |
| 22:15.36 | blitzrage | regular expressions |
| 22:15.49 | Dr-Linux | blitzrage: i don't understand, how can i do that? |
| 22:15.50 | blitzrage | using REGEX() |
| 22:16.21 | Dr-Linux | blitzrage: what it does? can you give me an example to put this in my above line? |
| 22:16.21 | blitzrage | www.regluar-expressions.info I think |
| 22:16.30 | blitzrage | no -- I suck at regular expressions |
| 22:16.49 | Dr-Linux | :S |
| 22:17.00 | Dr-Linux | ? |
| 22:17.15 | blitzrage | ?? |
| 22:17.36 | Dr-Linux | Server not found |
| 22:17.37 | Dr-Linux | Firefox can't find the server at www.regluar-expressions.info. |
| 22:17.52 | Dr-Linux | oopsss lemme remove . |
| 22:18.10 | Dr-Linux | same happend |
| 22:19.33 | timscott | regluar? |
| 22:19.42 | timscott | *regular... |
| 22:19.45 | timscott | that might be your problem, mate |
| 22:20.17 | timscott | www.regular-expressions.info |
| 22:20.32 | Dr-Linux | timscott: but blitzrage didn't explain anything, or i didn't understand :S |
| 22:20.50 | blitzrage | 'show function REGEX |
| 22:20.52 | timscott | what is the question? |
| 22:21.08 | timscott | oh |
| 22:21.09 | timscott | I see |
| 22:21.14 | timscott | you want patterns instead of 4040. |
| 22:21.21 | timscott | Sorry, I'm not familier with the REGEX() function |
| 22:22.01 | Dr-Linux | timscott: yess |
| 22:23.10 | Dr-Linux | WIKI also don't know :S |
| 22:23.16 | *** join/#asterisk kFuQ (n=somedude@c-67-185-114-199.hsd1.wa.comcast.net) |
| 22:24.42 | dlynes_home | ~regex |
| 22:24.43 | jbot | somebody said regex was ^[$%]?s/.*?:(?:\\\\\\\\\\\\\)+/.*[^\\\\\\\\\\\\\\\]/[i]?$ |
| 22:24.46 | dlynes_home | jbot knows |
| 22:25.04 | chandi | hi guys, in the dialplan, how can I read a variable from a text file ? |
| 22:25.22 | *** join/#asterisk jorgito (n=jorge@snat2.arachne.czfree.net) |
| 22:26.02 | dlynes_home | chandi: why would you want to? |
| 22:26.07 | Dr-Linux | wtf is this, http://www.regular-expressions.info/quickstart.html |
| 22:26.09 | Dr-Linux | how to understand |
| 22:26.13 | jorgito | hi have a problem with asterisk , i had in paste time three registers in sip.conf , two i have commented out (;) but they are still shown when i do sip show peers |
| 22:26.21 | jorgito | and also are working |
| 22:26.31 | jorgito | but the one i need that has to work is not ... |
| 22:26.46 | dlynes_home | jorgito: did you do a sip reload? |
| 22:26.53 | jorgito | maybe this is in some cache how clear cache, |
| 22:27.02 | Dr-Linux | dlynes_home: any clue on my question? |
| 22:27.05 | jorgito | dlynes_home, i did reload of asterisk, nothing changed |
| 22:27.11 | chandi | dlynes_home : I've got a script that writes the phone number to reach me in a text file |
| 22:27.15 | dlynes_home | jorgito: did you do a restart of asterisk? |
| 22:27.23 | jorgito | dlynes_home, i did also reload of whole server did not help |
| 22:27.28 | *** join/#asterisk chaoscon (n=ph33r@smartserv/ceo/chaoscon) |
| 22:27.29 | jorgito | dlynes_home, looks like bug |
| 22:27.44 | dlynes_home | chandi: if you've got a script to do that, get the script to write you a whole context |
| 22:27.48 | chandi | dlynes_home : I actually use it for other functions for asterisk |
| 22:28.10 | dlynes_home | chandi: and then just do an extensions reload after you've rewritten it |
| 22:28.22 | dlynes_home | chandi: make sure you use #include to include that file |
| 22:28.30 | chandi | dlynes_home : how can I make the script do an extension reload ? |
| 22:28.44 | dlynes_home | chandi: asterisk -rx "extensions reload" |
| 22:29.33 | dlynes_home | jorgito: sounds like you've got some stale entries in your astdb |
| 22:29.33 | Dr-Linux | hhm.. |
| 22:29.50 | chandi | dlynes_home thanks |
| 22:29.51 | Dr-Linux | maybe file will help me, once he gets in |
| 22:29.53 | mitcheloc | asdf |
| 22:29.53 | dlynes_home | Dr-Linux: i've got a clue on your question, but it seems to me you're not trying hard enough to learn |
| 22:29.58 | jorgito | dlynes_home, how to remove this entries ? |
| 22:30.21 | dlynes_home | Dr-Linux: you need to read the documentation for regex carefully to understand it |
| 22:30.27 | dlynes_home | Dr-Linux: open your mind, don't close it |
| 22:30.31 | Dr-Linux | dlynes_home: i'm trying, but what to try? |
| 22:30.38 | dlynes_home | Dr-Linux: regex is difficult to understand |
| 22:30.49 | dlynes_home | Dr-Linux: but once you learn it, you'll wonder how you ever managed without it |
| 22:31.09 | dlynes_home | jorgito: i don't know...i'm not terribly familiar with astdb |
| 22:31.22 | jorgito | dlynes_home, ok |
| 22:31.27 | dlynes_home | jorgito: but i would imagine there's a cli command like database refresh or something like that |
| 22:31.45 | dlynes_home | jorgito: or database clear all |
| 22:31.47 | Dr-Linux | dlynes_home: regex is asterisk application or what? |
| 22:31.54 | dlynes_home | jorgito: or anyways...you get the idea |
| 22:32.22 | jorgito | ok |
| 22:32.58 | dlynes_home | Dr-Linux: show function regex at the cli |
| 22:34.02 | blitzrage | type REGEX though because its case sensitive on the CLI |
| 22:34.05 | jorgito | does anybody know how to delete a register from registry in astdb ? |
| 22:34.11 | dlynes_home | blitzrage: no it isn't |
| 22:34.17 | dlynes_home | blitzrage: i just typed it all in lowercase |
| 22:34.17 | blitzrage | since when? |
| 22:34.21 | blitzrage | thats new then |
| 22:34.25 | dlynes_home | blitzrage: oops....my mistake |
| 22:34.28 | dlynes_home | my memory's bad :p |
| 22:34.30 | blitzrage | :) |
| 22:34.43 | dlynes_home | i typed in a lower case r, and hit tab ;) |
| 22:34.50 | blitzrage | ah |
| 22:34.54 | dlynes_home | it modified it to be a capital R |
| 22:35.12 | dlynes_home | i'm too lazy to type shit in |
| 22:35.16 | dlynes_home | i always use tab completion |
| 22:35.47 | blitzrage | REGEX is a dialplan function that will return a 0 or 1 if the regular expression matches or not -- if you place that in the $[ ] part of GotoIf, then the 0 or 1 will be interpreted as false or true |
| 22:35.52 | blitzrage | dlynes_home: I hear that |
| 22:36.18 | Qwell | blitzrage: No need to put it into an expression |
| 22:36.21 | dlynes_home | blitzrage: yeah...all programmers hear that :) |
| 22:36.24 | Qwell | all $[] does is return 0 or 1 |
| 22:36.36 | blitzrage | Qwell: but if you're using GotoIf()... |
| 22:36.39 | Qwell | no need :) |
| 22:36.43 | dlynes_home | blitzrage: not to mention sys admins :) |
| 22:36.45 | blitzrage | still seems like good form :) |
| 22:36.57 | Qwell | GotoIf(1,s:1) |
| 22:37.05 | Qwell | blitzrage: maybe so |
| 22:37.09 | blitzrage | or else you get burned on things like not using $[ ] While() :) |
| 22:37.21 | blitzrage | $[ ] in While()* |
| 22:37.24 | Dr-Linux | :S |
| 22:37.27 | blitzrage | Qwell: s/,/? |
| 22:37.35 | Dr-Linux | blitzrage: any example for my case? |
| 22:37.45 | blitzrage | Dr-Linux: no -- I suck at regular expressions (still) |
| 22:37.47 | Qwell | blitzrage: yeah, whatever :p |
| 22:37.51 | blitzrage | :D |
| 22:37.54 | Dr-Linux | exten => _91NXXXXXXXXX,1,GotoIf($["${CALLERID(num)}" = "_4XXX"]?5:2) |
| 22:37.57 | Qwell | 1?s,1: |
| 22:37.58 | De_Mon | in my dialplan I've got a Read(), but if no numbers are entered before the timeout it jumps to the next priority instead of the t extension.. what gives? |
| 22:38.33 | dlynes_home | gotoif(regex("011[0-9]*","01191923473984")?0:1) |
| 22:38.35 | Qwell | De_Mon: Things That Are Supposed To Happen - Chapter 6, Page 2 |
| 22:39.28 | blitzrage | dlynes_home: GotoIf(${REGEX("011[0-9]*","01191923473984")}?0:1) I think is what you want there... |
| 22:39.28 | dlynes_home | De_Mon: maybe read() doesn't honor the timeout extension? |
| 22:39.45 | dlynes_home | blitzrage: why the ${...}? |
| 22:39.56 | Qwell | dlynes_home: It's a function |
| 22:39.57 | blitzrage | because its a function and you need to use ${ } to return a value |
| 22:40.14 | dlynes_home | oh...thought qwell was complaining about how that wasn't necessary? |
| 22:40.21 | blitzrage | that $[ ] |
| 22:40.22 | Qwell | $[] != ${} |
| 22:40.24 | dlynes_home | ah |
| 22:40.28 | dlynes_home | square brackets |
| 22:40.31 | dlynes_home | heh |
| 22:40.36 | blitzrage | the way I'd do it would be: |
| 22:40.41 | Qwell | GotoIf(${REGEX("011[0-9]*","01191923473984")}?0:1) vs GotoIf($[${REGEX("011[0-9]*","01191923473984")}]?0:1) |
| 22:40.50 | dlynes_home | yeah...agreed |
| 22:40.56 | dlynes_home | blitzrage's solution looks damned ugly |
| 22:40.57 | Qwell | either way will work |
| 22:40.57 | blitzrage | GotoIf($[${REGEX("011[0-9]*","01191923473984")}]?0:1) I think is what you want there... |
| 22:41.24 | De_Mon | hmm, okay... so check the length of the variable after reading it |
| 22:41.26 | dlynes_home | braces and no square brackets looks much more eloquent |
| 22:41.34 | blitzrage | standard form for GotoIf() is: GotoIf($[<expression>]?true:false) |
| 22:41.49 | jorgito | how to delete register from registery field in astdb ? |
| 22:42.07 | blitzrage | database deltree registry (if thats the family name) |
| 22:42.16 | blitzrage | notice that will delete ALL registrations |
| 22:42.33 | dlynes_home | what does database deltree asterisk do? |
| 22:42.35 | blitzrage | database del <family> <key> |
| 22:42.48 | blitzrage | dlynes_home: it deletes all keys within a family |
| 22:42.58 | dlynes_home | and database deltree *.*? |
| 22:43.04 | blitzrage | nada work |
| 22:43.11 | *** join/#asterisk assorted_mike (n=assorted@S01060012171a89fc.wp.shawcable.net) |
| 22:43.23 | assorted_mike | hey all can anyone help me out with a few problems? |
| 22:43.23 | jorgito | nothing works |
| 22:43.30 | blitzrage | assorted_mike: just ask a question |
| 22:43.43 | jorgito | blitzrage, nowthing what you mentioned works |
| 22:43.53 | dlynes_home | wp? where hte hell is that? |
| 22:43.57 | De_Mon | assorted_mike don't ask to ask just ask |
| 22:43.59 | blitzrage | jorgito: database show <enter> for the form |
| 22:44.15 | blitzrage | errr |
| 22:44.20 | blitzrage | database del <enter> |
| 22:44.32 | assorted_mike | i setup an asterisk box using the trixbox install, it works inside my netowrk fine however when itry and connect over the internet x-lite connects but when i dial a number it doesnt ring on the other end and it says number is unavalible. any ideas? |
| 22:44.36 | jorgito | blitzrage, yes i did database del *.* but if i do database show it is still there |
| 22:44.40 | dlynes_home | assorted_mike: where's wp.shawcable.net? |
| 22:44.44 | blitzrage | jorgito: ummm... I just said that won't work |
| 22:44.48 | assorted_mike | manitoba |
| 22:44.50 | *** join/#asterisk DarKnesS_WolF (n=wolf@196.218.71.147) |
| 22:44.54 | dlynes_home | ah |
| 22:45.05 | dlynes_home | winterpeg? |
| 22:45.05 | blitzrage | assorted_mike: check the topic |
| 22:45.36 | assorted_mike | it says its asterisk 1.2.8 |
| 22:45.39 | assorted_mike | thats what i am using |
| 22:45.45 | assorted_mike | oh sorry |
| 22:45.48 | dlynes_home | assorted_mike: nah...he means about the trixbox stuff |
| 22:45.51 | blitzrage | assorted_mike: trixbox* |
| 22:45.52 | assorted_mike | freepbx is where i want to go |
| 22:45.56 | blitzrage | aye :) |
| 22:46.03 | assorted_mike | thanks guys |
| 22:46.06 | dlynes_home | what the heck is trixbox, anyways? |
| 22:46.12 | dlynes_home | a bastardized freepbx? |
| 22:46.15 | blitzrage | its a@h |
| 22:46.17 | blitzrage | renamed |
| 22:46.22 | blitzrage | afaik |
| 22:46.24 | dlynes_home | ah...the new name of it? |
| 22:46.28 | blitzrage | aye |
| 22:46.32 | dlynes_home | ghey |
| 22:46.39 | blitzrage | agreed |
| 22:46.49 | *** part/#asterisk cfassoni (n=cfassoni@c911444e.rjo.virtua.com.br) |
| 22:47.11 | dlynes_home | it's all part of the conspiracy to get #asterisk to support freepbx :p |
| 22:47.23 | blitzrage | :D |
| 22:47.26 | blitzrage | brb |
| 22:47.30 | jorgito | does anybody know how to delete a register from registry in astdb ? |
| 22:47.38 | Qwell | jorgito: dude, he told you |
| 22:50.29 | Dr-Linux | gotoif(regex("011[0-9]*","01191923473984")?0:1) << still thinking but not sure how can i modify it to need :S |
| 22:52.04 | dlynes_home | Dr-Linux: GotoIf(${regex("011[0-9]*","01191923473984")}?0:1) |
| 22:53.00 | timscott | well, i'll be damned. I didn't even know asterisk had regex support :S |
| 22:53.07 | *** part/#asterisk assorted_mike (n=assorted@S01060012171a89fc.wp.shawcable.net) |
| 22:53.16 | dlynes_home | timscott: you can use it for extension names, too |
| 22:53.46 | dlynes_home | timscott: but in taht case, you don't use the regex() function |
| 22:54.16 | dlynes_home | Dr-Linux: type man regex |
| 22:57.19 | De_Mon | regex is best learned by example, not all that jibber jabber in the manpage |
| 22:57.38 | blitzrage | timscott: yah -- but its a fairly simple implementation unfortunately |
| 22:57.59 | blitzrage | although its supposed to support extended regex |
| 22:58.49 | blitzrage | GotoIf(${REGEX("4[0-9][0-9][0-9]","${CALLERID(number)}")?true:false) |
| 22:59.12 | blitzrage | me thinks |
| 22:59.21 | blitzrage | Dr-Linux: try that |
| 23:00.05 | Dr-Linux | hhm.. |
| 23:00.17 | Dr-Linux | blitzrage: someone said this works >> GotoIf($["${CALLERID(num)}" : "4..."]?5:2) |
| 23:00.18 | blitzrage | actually -- replace the , with a space |
| 23:00.32 | blitzrage | Dr-Linux: thats another way of doing regex - yes |
| 23:00.51 | blitzrage | I had forgotten about it though |
| 23:00.58 | Dr-Linux | blitzrage: so that was is very easy i think |
| 23:01.57 | Dr-Linux | s/was/way |
| 23:06.06 | *** join/#asterisk ToTo (n=ToTo@host20-145.pool870.interbusiness.it) |
| 23:09.27 | Dr-Linux | blitzrage: GotoIf($["${CALLERID(num)}" : "45.."]?5:2) << is also fine? |
| 23:09.29 | Dr-Linux | right? |
| 23:09.59 | *** join/#asterisk _4d4m_ (n=adam@62.69.102.99) |
| 23:13.10 | *** join/#asterisk zagaya972 (n=d2s-comp@APointe-a-Pitre-102-1-3-9.w81-248.abo.wanadoo.fr) |
| 23:13.13 | blitzrage | Dr-Linux: try it and let me know how it went |
| 23:13.26 | *** part/#asterisk jeffpc (n=jeffpc@ool-18ba4c2d.dyn.optonline.net) |
| 23:14.12 | blitzrage | Dr-Linux: if you're using 1.2 or later, you should really not be using priority numbering |
| 23:15.39 | *** join/#asterisk Bullseye_Network (n=info@72.1.186.66) |
| 23:16.02 | TripleFFFF | darn |
| 23:16.09 | TripleFFFF | anyone good with ajax ? |
| 23:16.21 | Dr-Linux | blitzrage: i don't know priority numbering, but i just use it for GotoIF , bcoz i need that |
| 23:16.32 | Bullseye_Network | why would I constantly get: SIP/2.0 404 Not Found - from SIP phones not in use? |
| 23:16.44 | Dr-Linux | blitzrage: i'm at home and i was searching for this solution since 1 month |
| 23:17.18 | Dr-Linux | blitzrage: and why i sholdn't use priroity numbering? |
| 23:17.42 | blitzrage | TripleFFFF: I wish.. :( |
| 23:18.49 | blitzrage | Dr-Linux: priority number is bad because when you want to insert things with bigger dialplans, you have to renumber everything. Plus, when you do something like GotoIf($[ ]?5:2) -- what the heck is happening at 5 and at 2? sooo.... |
| 23:19.15 | blitzrage | exten => 123,n,GotoIf() |
| 23:19.56 | blitzrage | so all the numbers are figured out automatically by Asterisk -- now, you might ask... "Well how do I know where to go?". You use priority labels to do that |
| 23:20.06 | blitzrage | exten => 123,n(call),Dial() |
| 23:20.14 | Dr-Linux | blitzrage: that i know, i asked maybe there is something else |
| 23:20.36 | blitzrage | what something else? you aren't using priority labels -- you're using numbers -- don't do that. |
| 23:20.38 | Dr-Linux | blitzrage: i'm not talking about >>> ]?5:2) |
| 23:20.39 | *** join/#asterisk |ryan| (n=foo@c-24-7-159-130.hsd1.ca.comcast.net) |
| 23:20.58 | blitzrage | I know you're talking about the RegEx part of it -- and I told you to actually try it. |
| 23:21.08 | Dr-Linux | blitzrage: i only use it for gotoif() application |
| 23:21.26 | blitzrage | Dr-Linux: ONLY? thats one of the worst times to use priority numbers |
| 23:21.31 | blitzrage | if not THE worst |
| 23:21.44 | Dr-Linux | blitzrage: if i'm using all "n" then what should i use at the >>> ]?5:2) ? |
| 23:21.47 | blitzrage | use priority labels |
| 23:21.54 | |ryan| | hi, can someone reccomend a good howto for a first time asterisk user with an x100p card? |
| 23:21.57 | blitzrage | I was trying to tell you that -- but you said you already knew it |
| 23:22.11 | blitzrage | so where I left off... |
| 23:22.26 | blitzrage | exten => 123,n(call),Dial() <-- this is how you define a priority label |
| 23:22.43 | Dr-Linux | blitzrage: sorry, i said, i already knew that pirority numbering is hard to manage when insert new. |
| 23:22.53 | blitzrage | exten => 123,n,GotoIf($[<expression>]?call:hangup) <-- this is how you use them |
| 23:23.10 | blitzrage | much nicer to read eh? |
| 23:23.15 | Bullseye_Network | thats cool. I didnt know I could do that. |
| 23:23.20 | Dr-Linux | lol |
| 23:23.23 | blitzrage | yah -- thats what you really should be using |
| 23:23.37 | blitzrage | priority number is so 90's |
| 23:23.41 | Bullseye_Network | That makes it alot eaiser to... awesome |
| 23:23.57 | blitzrage | Bullseye_Network: a lot <-- two words :) |
| 23:24.15 | Bullseye_Network | :P |
| 23:24.22 | Bullseye_Network | lol |
| 23:24.27 | Dr-Linux | blitzrage: thanks, but if i have all "n" what should i'll use at "true:faluse" area? |
| 23:25.26 | blitzrage | you use the priority label |
| 23:25.29 | |ryan| | hello? |
| 23:25.41 | blitzrage | exten => 123,n(call),Dial() <-- this is how you define a priority label |
| 23:25.45 | blitzrage | exten => 123,n,GotoIf($[<expression>]?call:hangup) <-- this is how you use them |
| 23:25.50 | Dr-Linux | blitzrage: what's >> ?5:2) , so if i'm using "n" what i'll use instead? |
| 23:26.03 | blitzrage | Dr-Linux: are you seeing? |
| 23:26.13 | Dr-Linux | yes |
| 23:26.25 | *** join/#asterisk WiredX (n=matthew@gateway.ozpacific.net.au) |
| 23:26.53 | SplasPood | anyone happen to have or know where I can find a list of voicepulse connect's domestic us rates now that they have this FlexRate stuff.. |
| 23:27.52 | Dr-Linux | awww |
| 23:28.07 | Dr-Linux | blitzrage: i'm sorry, i just understood now. i seee |
| 23:28.40 | Dr-Linux | exten => 123,n(call), << this is totally new to me |
| 23:30.13 | blitzrage | excellent -- glad you learned something |
| 23:30.25 | Dr-Linux | blitzrage: thanks :) |
| 23:31.34 | Dr-Linux | blitzrage: asterisk will be the hotest product in my country by next year :) |
| 23:32.55 | blitzrage | Dr-Linux: what country? |
| 23:33.04 | blitzrage | India? |
| 23:33.17 | Dr-Linux | blitzrage: Pakistan |
| 23:34.07 | SplasPood | Dr-Linux: When I learned about using 'n' and priority labels it was a near religious experience :P |
| 23:34.55 | Dr-Linux | SplasPood: what's religous in it? :) |
| 23:35.08 | blitzrage | everything is religious |
| 23:36.07 | Dr-Linux | i didn't get though |
| 23:36.31 | mitcheloc | asterisk isn't a product |
| 23:38.15 | Nugget | flying into KAUS yesterday some moron asked tower for the "localizer 34 practice approach". controller said she didn't even understand what e wanted and that they were too busy, so try someplace else. |
| 23:38.37 | Nugget | the guy persisted and she finally explained that KAUS doesn't have a runway 34 or a localizer approach. that seemed to work. |
| 23:38.50 | Nugget | it's scary who else is in the sky sometimes |
| 23:39.25 | *** join/#asterisk jorgito (n=jorge@snat2.arachne.czfree.net) |
| 23:39.26 | jorgito | hi |
| 23:39.40 | jorgito | how do i clear some channels which are hanging on * ? |
| 23:39.48 | Qwell | jorgito: soft hangup |
| 23:39.56 | Nugget | jorgito: shutdown -r now :) |
| 23:40.06 | Dr-Linux | soft hangup <channel> |
| 23:40.31 | Dr-Linux | jorgito: noooooooooo |
| 23:41.48 | Dr-Linux | Qwell: in what case channel gets hanged? |
| 23:42.11 | Dr-Linux | and how can we verify if the channel is in use or hanged? |
| 23:42.38 | blitzrage | Qwell: !! |
| 23:42.55 | blitzrage | Dr-Linux: show channels |
| 23:43.31 | Dr-Linux | blitzrage: that only shows bridged channel. but this is not question |
| 23:43.34 | Qwell | blitzrage: y0 |
| 23:43.48 | blitzrage | sip show channels? |
| 23:45.09 | Dr-Linux | that will show connected channel, hanged channel is also connected. |
| 23:45.36 | Dr-Linux | hhm... but maybe that helps :S |
| 23:46.51 | blitzrage | no idea... never thought about showing channels that weren't bridged |
| 23:47.07 | mitcheloc | has anyone seen ZK on source forge? |
| 23:47.43 | *** join/#asterisk topping (n=topping@adsl-68-124-19-44.dsl.lsan03.pacbell.net) |
| 23:47.56 | topping | hoi |
| 23:48.06 | Dr-Linux | hoi? |
| 23:48.10 | topping | hi |
| 23:48.22 | Dr-Linux | ohh hi |
| 23:48.25 | topping | :) |
| 23:49.17 | topping | anyone using a mac to set up asterisk ivr prompts? would like to be able to edit voice, save as gsm |
| 23:49.19 | Qwell | Nugget: So, who was the idiot? ATC, or the pilot? |
| 23:50.38 | Dr-Linux | topping: i suggest you should use WavePad program, thats very easy and good |
| 23:50.47 | topping | nice, thanks |
| 23:51.03 | Nugget | the pilot |
| 23:51.18 | Qwell | Nugget: That went WAY over my head. |
| 23:51.22 | Qwell | Why was it a stupid thing to ask? |
| 23:51.22 | Nugget | heh |
| 23:51.39 | topping | Dr-Linux: that's windows... need OS-X |
| 23:51.44 | topping | looks nice tho |
| 23:51.58 | Nugget | he was asking (at a busy airport at a busy time of day) to perform a practice procedure on a runway that didn't exist, using ground facilities not present at that airport. |
| 23:52.05 | Qwell | heh |
| 23:52.27 | topping | are you guys pilots? |
| 23:52.35 | Qwell | I, obviously, am not |
| 23:52.40 | Nugget | only recreationally |
| 23:52.45 | topping | nice, me too |
| 23:53.04 | topping | used to have an old mooney |
| 23:53.08 | Nugget | nice |
| 23:53.18 | Nugget | I just rent. I don't fly enough to justify buying anything. |
| 23:53.38 | topping | haha, yah i learned that i barely broke even buying versus renting |
| 23:54.07 | Nugget | with the new g1000 glass cockpit stuff now, I'm glad I didn't buy anything with the old instrumentation, too. |
| 23:54.22 | Nugget | we just got a new 182 at my place with the g1000 and it's really slick. |
| 23:55.03 | topping | i got rid of mine after a really hairy trip home once. didn't have turbo to go over it, de-ice to go through it, or a stormscope to go around it |
| 23:55.15 | Nugget | oof |
| 23:56.06 | *** part/#asterisk sfbosch (n=sfbosch@dsl-cap-66-18-218-36-cgy.nucleus.com) |
| 23:56.06 | topping | thought i would get a nicer platform that i could put some of that stuff on and get the resale out of it, but then saw the literature on the eclipse (and gas prices) and decided things might change pretty quickly in the industry |
| 23:56.50 | Nugget | yeah, I agree with that completely |
| 23:57.22 | topping | buying a share of an eclipse would rock |
| 23:57.46 | Nugget | yeah, no doubt. even if that doesn't become totally approachable, it's got to kill prices on turboprops. |
