| 00:00.35 | Katty | well i just found one |
| 00:00.37 | Kailtor | the only reason i'm using free pbx is cause it came with asterisknow |
| 00:00.59 | Kailtor | so i'm probably just gonna wipe the whole thing and do it all myself |
| 00:01.13 | Katty | auto tune the newsssssssssssssssss (autotunethenews) |
| 00:01.18 | nickaugust | whats the question Kailtor? |
| 00:01.20 | leifmadsen | awesome :) |
| 00:02.01 | Katty | grooves |
| 00:02.44 | *** join/#asterisk bmoraca_work (~bmoraca@66-242-174-254.ceres.bvn.net) |
| 00:03.08 | leifmadsen | rocks |
| 00:03.17 | *** join/#asterisk jksM (jks@193.189.93.254) |
| 00:03.35 | Katty | what are you listening to? |
| 00:04.47 | leifmadsen | The Computer Fans |
| 00:04.52 | leifmadsen | errr..... the computer fans :) |
| 00:05.00 | leifmadsen | I have no music other than what is going on in my head |
| 00:05.01 | Katty | never heard of it |
| 00:07.56 | Kailtor | nevermind guys, i'm just gonna reformat CentOS and install everything manually |
| 00:08.19 | Kailtor | and forget freepbx, it's easier to just write config files |
| 00:09.59 | *** join/#asterisk dkirker-openmobl (~dkirker@openmobl/ceo/dkirker) |
| 00:11.34 | p3nguin_ | I wonder if anyone ever removes the freepbx stuff from asterisknow after they find out freepbx is a bother to config files, then continues using the OS/asterisk the old fashioned way. |
| 00:12.52 | *** join/#asterisk voipmonk (~voipmonk@CPE002191f85581-CM001692568382.cpe.net.cable.rogers.com) |
| 00:12.56 | bmoraca_work | p3nguin_: my first asterisk box was asterisknow, back when it defaulted to asterisk-gui, and it was so anemic that I didn't even bother using the GUI |
| 00:13.35 | voipmonk | Anemic |
| 00:14.27 | SeriousMatters | does freepbx has call hunting? |
| 00:14.35 | SeriousMatters | and follow-me? |
| 00:15.10 | voipmonk | Yes |
| 00:15.18 | voipmonk | SeriousMatters: Yes it does |
| 00:18.17 | *** join/#asterisk pawz (~pawz@ppp118-208-171-146.lns20.bne4.internode.on.net) |
| 00:18.35 | *** join/#asterisk voipmonk (~voipmonk@CPE002191f85581-CM001692568382.cpe.net.cable.rogers.com) |
| 00:19.07 | *** join/#asterisk nix8n82 (~AndChat@63.162.27.14) |
| 00:20.05 | *** join/#asterisk voipmonk (~voipmonk@CPE002191f85581-CM001692568382.cpe.net.cable.rogers.com) |
| 00:31.14 | *** join/#asterisk ELBunce (~erik@kde/developer/bunce) |
| 00:33.33 | *** part/#asterisk ruben23 (~AGENT@122.55.48.243) |
| 00:35.00 | hipitihop | when setting up my sip provider in sip.conf, the book uses ip address e.g. host=10.251.55.100 can I also use a domain name ? e.g. host=sip.gotalk.com ? |
| 00:35.14 | [TK]D-Fender | hipitihop: yes |
| 00:37.21 | *** join/#asterisk phix (~threat@123-243-44-131.tpgi.com.au) |
| 00:38.08 | phix | hi gang |
| 00:39.31 | *** join/#asterisk hluesea (~hulusikah@88.247.127.66) |
| 00:43.02 | *** join/#asterisk nickaugust (~nickaugus@74-95-115-29-Colorado.hfc.comcastbusiness.net) |
| 00:43.55 | *** join/#asterisk jakent (~john@soleil.johnkent.mooo.com) |
| 00:44.17 | nickaugust | haha! polycom online! |
| 00:44.40 | phix | hi |
| 00:44.49 | nickaugust | spoofed the phones mac address from my macbook then logged onto the server and switched the cable to the phone |
| 00:44.51 | nickaugust | hah! |
| 00:44.55 | nickaugust | no one can stop me! |
| 00:45.03 | nickaugust | world domination here I come! |
| 00:45.05 | nickaugust | :P |
| 00:45.06 | nickaugust | |
| 00:45.11 | phix | :D |
| 00:45.13 | phix | Can I help? |
| 00:46.30 | hipitihop | [TK]D-Fender, thanks, done that and reloaded sip and diaplan, now running "sip show registry" It shows a registration, can I take that that it has actually contacted provider and registered ? |
| 00:47.50 | fink | nickaugust: nice |
| 00:48.15 | hipitihop | it shows a registration time which seems to change every two minutes so I'm guessing that part is working |
| 00:48.23 | *** join/#asterisk CoderForLife (~Miranda@cpe-174-101-155-51.cinci.res.rr.com) |
| 00:49.15 | *** join/#asterisk goldenear (~goldenear@2a01:e35:2e58:8df0:21c:b3ff:fec6:943a) |
| 00:49.17 | *** join/#asterisk voipmonk (~shido6@CPE002191f85581-CM001692568382.cpe.net.cable.rogers.com) |
| 00:51.18 | goldenear | hi. I've got a problem with asterisk 1.6.2 compiled from svn. It consumes 100% of the CPU all the time. |
| 00:51.25 | goldenear | is this normal ? |
| 00:51.38 | techie | take a guess |
| 00:52.11 | goldenear | I guess it a thread issue or something |
| 00:52.19 | goldenear | it's* |
| 00:52.29 | phix | who knows much about tdm400p cards? |
| 00:52.45 | voipmonk | phix - whats your question? |
| 00:52.48 | phix | Mine seems to think a phone has been picked up |
| 00:53.02 | phix | I have FXO and FXS ports |
| 00:53.05 | voipmonk | what part of the world are you in? |
| 00:53.22 | phix | when a call comes from FXO I get it to dial FXS port, which answers the call straight away for a few seconds then hangs up |
| 00:53.29 | phix | voipmonk: AU |
| 00:54.29 | phix | Near Lat 33.950001 south, long 151.183334 east |
| 00:54.40 | goldenear | techie: what could be the problem ? is there something special to care of during the compilation ? |
| 00:55.08 | techie | goldenear: Does it happen with any other version? |
| 00:56.53 | hipitihop | can someone give me a hand to get my Linksys PAP2t side configured to talk to asterisk instead of my sip provider |
| 00:57.14 | goldenear | techie: I don't know. I was using 1.4.21 before and I never had any problem. |
| 00:57.56 | *** join/#asterisk xpot-mobile (~xpot@70-91-210-233-BusName-Utah.hfc.comcastbusiness.net) |
| 00:59.14 | voipmonk | hipitihop: sure |
| 00:59.51 | goldenear | techie: I found the problem, this wasn't the fault of asterisk but the init.d/asterisk script |
| 01:00.16 | *** part/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 01:00.40 | goldenear | I found the bug is already known : https://issues.asterisk.org/view.php?id=16784 |
| 01:00.59 | hipitihop | voipmonk, thanks... I'm just setting up a dialplan as per Chapter 4 now that I seem to have my sip provider registering..2 secs |
| 01:01.21 | goldenear | I juste remove the -c option from the command line and no more 100% cpu use :) |
| 01:02.33 | phix | http://pastebin.ca/1802194 |
| 01:02.40 | phix | ^---- voipmonk |
| 01:02.40 | Katty | Qwell: BANG |
| 01:02.44 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 01:03.07 | jblack | eek! |
| 01:03.36 | techie | goldenear: Funny you didn't have the same problem with 1.4.x |
| 01:03.59 | techie | goldenear: That same bug was reported for 1.4.x |
| 01:04.34 | Katty | Qwell: next next next |
| 01:04.43 | goldenear | bug the 1.4.x had been installed with apt-get, not from source |
| 01:04.54 | techie | goldenear: yuck |
| 01:04.55 | Katty | Qwell: i put a unique thing int he background :P |
| 01:07.39 | hipitihop | voipmonk, ok ... I have this in sip.conf [LinksysPAP] ...type=friend ...context=phones ..host=192.168.0.102 |
| 01:09.17 | hipitihop | voipmonk, are you familiar with the Linksys PAP2t ? |
| 01:10.20 | voipmonk | i am very familiar with it |
| 01:10.25 | voipmonk | my sister in law has one |
| 01:10.30 | voipmonk | my mother in law has one |
| 01:10.32 | voipmonk | i have one |
| 01:10.35 | voipmonk | my father in law has one |
| 01:10.53 | voipmonk | I've been configuring these devices for a few years |
| 01:10.59 | phix | wow this linksys SPA Is quiet |
| 01:11.22 | voipmonk | what do you mean by quiet? :) |
| 01:11.32 | voipmonk | the device itself should make no nosie |
| 01:11.34 | voipmonk | noise |
| 01:11.45 | adnc | is there somewhere an example how to define dialplans where one can select the provider by dialing *xx# in front of the number? |
| 01:12.36 | voipmonk | adnc exten => *21X.,1,Dial(SIP/myprovider/${EXTEN:3}) |
| 01:12.40 | voipmonk | or.. |
| 01:12.58 | voipmonk | exten => *21X.,1,Dial(SIP/${EXTEN:3}@myprovider) |
| 01:13.03 | voipmonk | or..... |
| 01:13.08 | voipmonk | u get the idea |
| 01:13.09 | phix | what is FXS Port Impedance:? |
| 01:13.10 | adnc | wow, hold on, i did check it |
| 01:13.17 | adnc | so simply remove the first three chars |
| 01:13.19 | phix | will it make the voice louder if I fiddle with it? |
| 01:13.22 | phix | voipmonk: ? |
| 01:13.27 | adnc | thank you voipmonk |
| 01:14.06 | adnc | voipmonk, do i need to define seprate context for it? |
| 01:16.47 | phix | fxs port input gain |
| 01:16.50 | phix | -5 |
| 01:16.52 | tkrn | voipmonk is my HERO! |
| 01:16.53 | phix | hmmm, isthat in DB? |
| 01:16.56 | tkrn | :-D :-D :-D |
| 01:17.20 | phix | port impendence what is that? |
| 01:22.45 | Katty | struggles |
| 01:22.49 | Katty | tries to resist |
| 01:22.53 | Katty | twitches |
| 01:22.57 | Katty | phix: your mom |
| 01:24.52 | adnc | Katty, ;) |
| 01:25.13 | adnc | Katty, how are you doing? |
| 01:29.43 | phix | Katty: :D |
| 01:29.47 | phix | Katty: <3 |
| 01:29.56 | phix | You always know the right thing to say in any situation |
| 01:30.44 | fink | Katty: hi |
| 01:30.48 | fink | Katty: hi hi hi |
| 01:31.38 | adnc | voipmonk, unfortunately this rule doesnt match |
| 01:31.39 | adnc | [kurzwahl_kd_out] |
| 01:31.39 | adnc | exten => *22#X.,n,Dial(SIP/${EXTEN:4}@terrasip) |
| 01:33.36 | phix | Katty: hiihihihihihihihihihihihihi |
| 01:33.55 | phix | fink: repeating hi x number of times doesn't seem to work |
| 01:34.13 | voipmonk | adnc: what do you mean this rule doenst match? |
| 01:34.17 | fink | phix: you need pyirc |
| 01:34.50 | *** join/#asterisk lordmortis (~lordmorti@203-206-117-94.dyn.iinet.net.au) |
| 01:34.51 | phix | fink: nonononono irssi ftw |
| 01:34.52 | voipmonk | exten => *22#X.,n,Dial(SIP/${EXTEN:4%7D@terrasip) <---- wrong u need exten => *22#X.,n,Dial(SIP/${EXTEN:4%7D@terrasip) |
| 01:39.26 | *** join/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 01:39.58 | *** join/#asterisk voipmonk_ (~voipmonk@CPE002191f85581-CM001692568382.cpe.net.cable.rogers.com) |
| 01:40.02 | *** join/#asterisk Tech_Travis (~Administr@cpe-76-168-191-127.socal.res.rr.com) |
| 01:40.25 | voipmonk_ | Back |
| 01:42.21 | adnc | voipmonk, thank you very much |
| 01:46.05 | jaskew | is having Aastra 480i weirdness - or maybe I didn't read the manual... |
| 01:46.24 | *** join/#asterisk coppice (~chatzilla@94.201.17.210.dyn.pacific.net.hk) |
| 01:47.27 | jaskew | So I'm on L1 and a call comes in on L2. I hit the "Goodbye" button to hang up L1, but the phone instead drops the ringing call on L2. Is that normal? |
| 01:48.51 | jaskew | It's like DEC designed the darn thing. |
| 01:57.09 | *** join/#asterisk n0cturnal (~n0c@203.161.121.158.static.amnet.net.au) |
| 02:07.40 | p3nguin_ | Like a conference? |
| 02:13.14 | jaskew | no - was trying to hang up the call I was on to pick up the other line. I was surprised when it terminated the incoming call and left me on the line with the first call |
| 02:15.34 | phix | hihihihihi |
| 02:15.35 | adnc | does someone know flowroute? |
| 02:16.08 | phix | I am sure someone does |
| 02:16.10 | jaskew | is afk for dinner |
| 02:16.12 | phix | not me though |
| 02:16.18 | phix | what you having for dinner jaskew ? |
| 02:16.45 | jaskew | the usual - whatever is on the table |
| 02:16.46 | phix | A big fat steak? |
| 02:16.51 | *** join/#asterisk voipmonk_ (~shido6@CPE002191f85581-CM001692568382.cpe.net.cable.rogers.com) |
| 02:17.24 | phix | I recommend Kangaroo, cheaper than steak, less fat and lots of protein, it is win |
| 02:17.36 | phix | (steak == beef in this context_ |
| 02:17.37 | phix | ) |
| 02:17.50 | jaskew | You get that at Jack in th Box, right? |
| 02:18.42 | jaskew | gotta run - bbl |
| 02:18.53 | phix | nfi |
| 02:18.55 | phix | never been there |
| 02:19.00 | phix | kk |
| 02:28.48 | p3nguin_ | adnc: Did you have a question pertaining to flowroute, or just taking a poll to see if anyone has heard about it? |
| 02:29.18 | adnc | p3nguin_, well, i'm looking for cheap routes to turkey |
| 02:29.25 | adnc | with good quality |
| 02:29.40 | adnc | and i was wondering if someone has done good experiences with them |
| 02:29.44 | p3nguin_ | US to Turkey? |
| 02:29.49 | adnc | germany to turkey |
| 02:30.31 | p3nguin_ | I don't know if a US-based ITSP is the best choice for calls originating in Germany. |
| 02:31.47 | phix | I concur |
| 02:49.07 | *** join/#asterisk Jasonwert (~jasonwert@97-83-97-13.dhcp.trcy.mi.charter.com) |
| 02:50.46 | *** join/#asterisk D-Boy (~Geek@unaffiliated/cain) |
| 02:54.23 | D-Boy | Hi, when installing AsteriskONE on a dedicated machine i have those errors on freepbx web interface : http://pastebin.com/m15e4d52b |
| 02:56.02 | Kobaz | #freepbx |
| 02:58.31 | drmessano | AsteriskONE? |
| 02:59.20 | D-Boy | AsteriskNOW |
| 03:01.32 | *** join/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 03:02.22 | *** join/#asterisk ManxPower-work (~EWieling@89.sub-75-235-48.myvzw.com) |
| 03:03.50 | ManxPower-work | ~answers |
| 03:03.51 | infobot | i heard answers is Asterisk Book: http://astbook.asteriskdocs.org & SIP w/ NAT: http://www.aocomputing.net/?p=3 & Wiki: http://www.voip-info.org & docs: /path/to/src/asterisk/doc & 1.4 .vs. 1.6.0 .vs. 1.6.1: : /path/to/src/asterisk/UPGRADE*.txt |
| 03:08.12 | Katty | oh |
| 03:08.13 | Katty | phix: hi |
| 03:08.23 | Katty | adnc: also hi |
| 03:08.40 | phix | Katty: :D |
| 03:08.43 | *** join/#asterisk pyite (~dschreibe@unaffiliated/pyite) |
| 03:08.47 | phix | Katty: you are lagged :) |
| 03:08.48 | Katty | i was raiding |
| 03:08.54 | phix | haha WoW |
| 03:09.01 | Katty | yesh |
| 03:09.06 | phix | you play STO as well? |
| 03:09.11 | Katty | yesh |
| 03:09.14 | phix | or pure WOW? |
| 03:09.16 | phix | awesome |
| 03:09.17 | Katty | no |
| 03:09.24 | phix | lets STO!!!! |
| 03:09.26 | Katty | i also play roulette |
| 03:09.34 | phix | texas holden poker as well? |
| 03:09.37 | Katty | no. |
| 03:09.40 | Katty | reddit or chat roulette |
| 03:09.42 | phix | aawww |
| 03:09.47 | phix | 1 out of 4 :) |
| 03:09.53 | phix | lets STO |
| 03:10.00 | Katty | i dont' wanna |
| 03:10.04 | Katty | gamed out |
| 03:10.18 | Katty | raiding is exhausting |
| 03:10.39 | Katty | s/raiding/healing/ |
| 03:11.26 | Katty | Qwell: remind me to tell you about webcammax |
| 03:14.39 | phix | Katty: aawww, what's your handle in sto? I will catch up later :) |
| 03:15.31 | Katty | @gwennora |
| 03:16.19 | pyite | where is jsmith |
| 03:18.06 | phix | :D |
| 03:19.40 | phix | Katty: what rank are you? |
| 03:23.42 | Katty | no idea |
| 03:25.07 | phix | heh |
| 03:25.16 | phix | how long have you played as one character then? :) |
| 03:25.37 | *** join/#asterisk lordmortis (~lordmorti@203-206-117-94.dyn.iinet.net.au) |
| 03:26.34 | Katty | well i got to 16 in beta |
| 03:26.42 | Katty | but since it started for real i'm like lvl 3 |
| 03:26.54 | phix | heh |
| 03:26.57 | phix | awww |
| 03:27.01 | phix | I will tank you :) |
| 03:27.12 | Katty | <- science ship |
| 03:27.50 | Katty | roulettes |
| 03:28.08 | phix | heh |
| 03:28.12 | phix | snake eyes! |
| 03:28.49 | phix | I am playing some STO now! |
| 03:28.49 | Katty | no reddit rouletting! |
| 03:28.54 | phix | ooohhh |
| 03:28.56 | phix | what is that? |
| 03:29.31 | Katty | http://redditroulette.appspot.com/ |
| 03:29.41 | *** part/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 03:29.53 | Katty | come find me |
| 03:32.07 | phix | 251Mb STO patch :/ |
| 03:32.33 | Katty | no fun |
| 03:36.09 | Katty | Qwell: ping |
| 03:45.13 | *** join/#asterisk cnr0616 (~cj@74.195.93.109) |
| 03:45.28 | phix | pong |
| 03:45.36 | cnr0616 | Is the switchboard available for Asterisk Business Edition? |
| 03:45.38 | adnc | SIP/2.0 488 Not acceptable here |
| 03:45.46 | adnc | does someone know how to interprete this |
| 03:46.06 | florz | probably a codec problem ;-) |
| 03:46.24 | adnc | could it be a matter of NAT? |
| 03:47.25 | voipmonk | um |
| 03:47.30 | voipmonk | adnc: whats up? |
| 03:47.38 | adnc | hi voipmonk |
| 03:48.13 | adnc | i' can not receive calls. so i looked into the sip debug of asterisk and found that error |
| 03:48.19 | adnc | SIP/2.0 488 Not acceptable here |
| 03:48.24 | cnr0616 | Is the switchvox switchboard available for Asterisk Business edition? |
| 03:48.35 | ChannelZ | maybe you should show us what that is in response to. |
| 03:49.40 | voipmonk | thats good adnc |
| 03:49.46 | voipmonk | that means u need to set a codec |
| 03:49.57 | voipmonk | what codec do you have set adnc? |
| 03:50.11 | adnc | disallow=all |
| 03:50.11 | adnc | allow=alaw |
| 03:50.16 | adnc | thats in my sip.conf |
| 03:51.04 | voipmonk | good |
| 03:51.12 | voipmonk | whats in your sip device? |
| 03:52.00 | voipmonk | if your sip device uses anything but alaw ur going to get a 488 |
| 03:53.22 | adnc | i see. i've a-law und u-law |
| 03:53.36 | phix | voipmonk is my pal ) |
| 03:53.37 | phix | :) |
| 03:55.03 | voipmonk | remove ulaw |
| 03:55.11 | adnc | ok |
| 03:55.11 | voipmonk | or |
| 03:55.21 | voipmonk | add allow=ulaw in sip.conf |
| 03:55.28 | voipmonk | its up to you |
| 03:55.31 | voipmonk | but you see how that works? |
| 03:55.32 | voipmonk | :) |
| 03:55.42 | adnc | i understand |
| 03:55.55 | voipmonk | :) |
| 03:56.03 | adnc | ;) |
| 03:56.07 | adnc | i did allow it |
| 03:56.13 | phix | I use alaw |
| 03:56.13 | adnc | allowing is better than forbidding |
| 03:56.32 | adnc | u-law is more used in the usa |
| 03:56.37 | adnc | ;) |
| 03:56.50 | adnc | like giving is better than taking |
| 03:57.38 | phix | I use alaw here in the AU :) |
| 03:58.35 | adnc | strange, i did allow ulaw and alaw but still the same |
| 03:58.43 | phix | I HAS TRIBBLES!!! |
| 03:59.12 | Katty | :>>>>>>>>> |
| 03:59.15 | adnc | hold on someone was writing that max. package size needed to be changed |
| 04:00.58 | ManxPower-work | Generally you don't want to allow both ulaw and alaw |
| 04:01.09 | p3nguin_ | Not much reason to, really. |
| 04:02.12 | ManxPower-work | sip debug will tell you what codecs the device is trying to use |
| 04:02.34 | *** join/#asterisk tkrn (~tkrn@216.196.213.246) |
| 04:02.56 | adnc | ok |
| 04:07.49 | adnc | http://asterisk.pastebin.ca/1802332 |
| 04:07.57 | adnc | this is what i get as debug |
| 04:08.16 | adnc | but looks ok |
| 04:17.35 | Katty | :< |
| 04:17.47 | Katty | ryan showed me a youtube video of a person with a cat in a harness, trying to walk it |
| 04:17.57 | Katty | but the cat obviously didn't want to go for a walk...so they dragged the cat across the floor :/ |
| 04:18.02 | Katty | and they were just sittin there lauging |
| 04:18.05 | Katty | i hate people. |
| 04:18.50 | phix | Katty: was it long cat? |
| 04:19.02 | Katty | your statement does not parse |
| 04:19.04 | Katty | please try again |
| 04:19.06 | phix | long cat is long |
| 04:19.23 | Katty | syntax error |
| 04:20.09 | phix | for i in xrange(20): |
| 04:22.01 | *** join/#asterisk stallione (~stallione@c-67-173-76-139.hsd1.il.comcast.net) |
| 04:22.18 | stallione | Hi |
| 04:23.22 | phix | oh noes! Hull integrity below 50% |
| 04:23.42 | Katty | OH NEOS |
| 04:23.43 | phix | Damn Romulans |
| 04:23.46 | Katty | EVASIVE MANUVERS STAT |
| 04:23.56 | phix | Exit sector block stat :) |
| 04:23.56 | stallione | tyring to figure out why is it that I get all circuits are busy if I add a prefix of '1' (calls go out if I don't add the prefix) |
| 04:24.11 | Katty | emergency power to engines! |
| 04:24.28 | phix | warp ftw |
| 04:24.50 | phix | I hate it when enemy signal contacts get ya, it auto loads you in there |
| 04:25.09 | phix | Then you have to close all of the skill windows so you can GTFO of there |
| 04:25.17 | phix | any wya, back to asterisk :P |
| 04:25.21 | phix | stallione: hmmmm |
| 04:25.39 | phix | pastebin.ca config and version information? |
| 04:25.39 | stallione | sorry to interrupt your game :) |
| 04:26.30 | phix | Where is starbase 39? hmmmm |
| 04:26.34 | phix | stallione: hehe :) |
| 04:26.42 | phix | np |
| 04:27.05 | phix | I probably shouldnt be playing it, but I have to wait for this copy to finish before I can do any more work |
| 04:27.13 | stallione | embarrased to say I don't even know which files to pastebin.. (the install is a mix of freepbx, lmce and asterisk).. When it broke, I started to configure everything manually |
| 04:27.39 | stallione | what are you guys playing.. (I am jealous...) |
| 04:27.57 | Katty | i'm playing chatroulette |
| 04:27.59 | Katty | well i was. |
| 04:28.02 | Katty | but now i'm going to bed. |
| 04:28.25 | Katty | gnighters |
| 04:28.33 | stallione | nite |
| 04:28.59 | stallione | feel its' got something to do with my dialplan |
| 04:34.19 | p3nguin_ | stallione: If changing the dialed number changes the results, then it's a dialplan issue. |
| 04:37.49 | idespinner | stallione, i recommend watching the asterisk console while dialing the number |
| 04:38.11 | idespinner | then do 'dialplan show xxxx' for the pertinant dialplan portions |
| 04:39.15 | stallione | p3nguin, idespinner, thx, I'll try that and see what happens when I dial |
| 04:39.47 | *** join/#asterisk xmitter (~xmitter@c-24-21-213-242.hsd1.or.comcast.net) |
| 04:40.28 | phix | stallione: StarTrek Online |
| 04:40.40 | phix | I has tribbles!!! :) |
| 04:40.46 | adnc | i've such a simple entry in my dialplan http://asterisk.pastebin.ca/1802351 where i redirect premium-rate calls to a privacy announcement. if i also want to match number of the form _0049900. do i need one more context or can i somehow put it into the same context |
| 04:40.51 | adnc | and if so how |
| 04:42.46 | idespinner | adnc, you can do include statements in your dialplan i beleive |
| 04:42.59 | idespinner | include a context within another |
| 04:55.36 | *** join/#asterisk amol112 (~amol111@203.129.255.188) |
| 04:57.01 | amol112 | hi all, i want to setup call center application using softphone.which softphone is better and at http://www.asterisk.org/downloads there is no zaptel source available for latest asterisk |
| 04:58.21 | *** join/#asterisk Kevin` (~kevin@rrcs-67-52-47-69.west.biz.rr.com) |
| 05:00.33 | amol112 | i am new to asterisk |
| 05:01.13 | amol112 | hi all, i want to setup call center application using softphone.which softphone is better and at http://www.asterisk.org/downloads there is no zaptel source available for latest asterisk |
| 05:02.19 | ChannelZ | zaptel is now DAHDI |
| 05:02.59 | *** join/#asterisk dkirker-openmobl (~dkirker@openmobl/ceo/dkirker) |
| 05:03.03 | *** join/#asterisk Jak_o_Shadows (~Fake@121.209.35.170) |
| 05:04.36 | amol112 | ChannelZ, thanks |
| 05:04.57 | amol112 | ChannelZ, i think i am referring old documents |
| 05:05.20 | amol112 | ChannelZ, which is best free softphone for linux? |
| 05:05.50 | ChannelZ | dunno sorry |
| 05:06.50 | amol112 | ChannelZ, AsteriskNOW is created on which linux distro? |
| 05:07.00 | ChannelZ | centos |
| 05:07.46 | *** join/#asterisk Faustov (user@gentoo/user/faustov) |
| 05:08.16 | amol112 | ChannelZ,can you give me latest documentation for configuration of latest asterisk server. |
| 05:13.41 | p3nguin_ | ~book |
| 05:13.42 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 05:17.03 | *** join/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 05:19.08 | p3nguin_ | ~answers |
| 05:19.09 | infobot | methinks answers is Asterisk Book: http://astbook.asteriskdocs.org & SIP w/ NAT: http://www.aocomputing.net/?p=3 & Wiki: http://www.voip-info.org & docs: /path/to/src/asterisk/doc & 1.4 .vs. 1.6.0 .vs. 1.6.1: : /path/to/src/asterisk/UPGRADE*.txt |
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| 05:41.18 | sun28 | moin /o\ |
| 05:43.28 | amol112 | ChannelZ, after installing dahdi also i not got zaptel driver .. |
| 05:52.26 | p3nguin_ | amol112: That should have been obvious. |
| 05:52.49 | p3nguin_ | What zaptel driver are you wanting? |
| 05:53.33 | amol112 | p3nguin_, just for testing softphone,call center application |
| 05:53.38 | *** join/#asterisk mintos (~mvaliyav@nat/redhat/x-fauzehgjnbnzottv) |
| 05:53.48 | p3nguin_ | Does not parse. |
| 05:54.07 | p3nguin_ | Try answering the question I asked. You'll get better information that way. |
| 06:03.07 | *** part/#asterisk benngard (~benngard@90-230-92-67-no148.tbcn.telia.com) |
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| 06:11.04 | phix | hihihihihihi |
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| 06:23.13 | *** join/#asterisk Tulga (~chatzilla@203.91.113.10) |
| 06:23.48 | Tulga | hi guys. I installed asterisknow. in my hardware documentation, it says install dahdi and after that install hardware driver |
| 06:24.03 | Tulga | but no guide to install dahdi. how to install it? |
| 06:24.28 | ChannelZ | I think it's already there |
| 06:24.43 | Tulga | I checked 'rpm -q dahdi' in shell |
| 06:24.54 | Tulga | 'package dahdi is not installed' |
| 06:25.30 | Tulga | I installed asterisknow from ISO. but it uses centos and I'm not familiar redhat like systems |
| 06:25.49 | ChannelZ | Why do you assume it's a package? Look at your startup |
| 06:26.25 | ChannelZ | /etc/rc.d/init.d/dahdi - should have loaded when *now booted |
| 06:26.48 | Tulga | ohh yes. there is dahdi |
| 06:27.07 | ChannelZ | see /usr/sbin/dahdi* |
| 06:27.22 | Tulga | how to check dahdi version? |
| 06:28.17 | ChannelZ | most of the tools should know - dahdi_cfg -help |
| 06:28.43 | Tulga | ok great |
| 06:29.06 | *** join/#asterisk JAMMAN2110 (~james@unaffiliated/jamman2110) |
| 06:29.09 | Tulga | I'm planning to make small call center. asterisknow is best solution? |
| 06:31.46 | p3nguin_ | tulga: rpm -qa|grep -i dahdi |
| 06:32.42 | ChannelZ | maybe, maybe not |
| 06:32.53 | Tulga | p3nguin_: thanks |
| 06:33.02 | Tulga | ChannelZ: what you suggest? |
| 06:33.05 | ChannelZ | I haven't used asterisknow much, I installed it in a virtual machine just to look at it.. the gui is actually quite terrible and limiting |
| 06:33.07 | p3nguin_ | If it is a package, that should reveal it. rpm -q dahdi would not. |
| 06:33.16 | Tulga | yes. gui is not good |
| 06:33.23 | ChannelZ | I prefer to just configure asterisk myself |
| 06:33.40 | Tulga | asterisknow is asterisk + gui? |
| 06:33.41 | p3nguin_ | Because the package name is not dahdi -- it contains a version, such as dahdi-1.2.3. |
| 06:33.56 | Tulga | p3nguin_: ohh yes. that is problem I think |
| 06:34.00 | ChannelZ | Yes it's just a linux distro + asterisk + dahdi + gui |
| 06:34.18 | ChannelZ | It's nothing special besides preconfigured |
| 06:34.40 | p3nguin_ | I think AsteriskNOW is kind of nice. |
| 06:34.41 | Tulga | ok. so I try asterisknow first time. then after that I switch own build |
| 06:35.02 | Tulga | because there is lot of tools that I don't understand |
| 06:35.04 | p3nguin_ | I don't personally use it, but I did play with it once. |
| 06:35.30 | Tulga | I remember 4 years ago our boss using it on small office |
| 06:35.39 | Tulga | play IVR, some extension etc |
| 06:35.43 | p3nguin_ | I bet it is completely different now. |
| 06:35.56 | Tulga | I think so |
| 06:36.57 | ChannelZ | It used to be a different GUI I think; FreePBX I'm not so much impressed with |
| 06:37.30 | ChannelZ | I mean it seems OK if you just want to configure a bunch of devices and have voicemail and not go to a lot of trouble |
| 06:38.10 | Tulga | I prefer CLI. but I'm very newbie asterisk |
| 06:38.39 | Tulga | did you guys use voice mail? |
| 06:38.47 | ChannelZ | do..yeah |
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| 07:24.12 | benngard | any that can help me with a t.38 question? |
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| 07:27.24 | tuxx- | morningggg |
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| 07:28.12 | benngard | did get 4 inches of snow last night :( |
| 07:28.38 | benngard | wants SUN |
| 07:29.00 | creativx | ive got sun |
| 07:29.06 | *** join/#asterisk ReDNeQ (~ReDNeQ@70.114.229.58) |
| 07:29.06 | creativx | and -8,5 centigrade |
| 07:29.40 | benngard | around 0 centigrade in gothenburg and very cloudy |
| 07:29.49 | benngard | probably gonna get more snow today |
| 07:30.02 | creativx | those bastards are saying we are gonna get more snow this weekend too |
| 07:30.07 | creativx | I refuse to believe it |
| 07:30.13 | benngard | where do u live? |
| 07:30.17 | creativx | bergen, no |
| 07:30.30 | creativx | so right across the border, and then some |
| 07:30.32 | benngard | not far from me :) |
| 07:30.38 | benngard | exactly |
| 07:31.26 | benngard | i have like 200 km to "svinesund" the bridge, i guess u know where it is |
| 07:31.35 | creativx | yep been over there a couple of times |
| 07:31.45 | creativx | albeit mostly via karlstad/karlskoga and then mantorp race park |
| 07:31.55 | benngard | :) |
| 07:32.10 | benngard | about 300 km to karlstad |
| 07:32.33 | creativx | mjölby |
| 07:32.48 | creativx | been there 3 or 4 times for bugrun |
| 07:32.55 | benngard | east side of the laken amed vättern |
| 07:33.04 | benngard | lake named* |
| 07:33.18 | creativx | yea |
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| 07:38.03 | creativx | and theres some bike race around those lakes |
| 07:38.38 | *** part/#asterisk Tech_Travis (~Administr@cpe-76-168-191-127.socal.res.rr.com) |
| 07:39.59 | benngard | yupp |
| 07:40.27 | *** join/#asterisk oej (~olle@ns.webway.se) |
| 07:42.50 | benngard | but why cant my stupid avaya and my asterisk negotiate t.38 correctly? :( |
| 07:43.55 | *** join/#asterisk mpe (~mpe@gate.ipvision.dk) |
| 07:44.57 | benngard | fax - ata - asterisk - ata - fax, works, switches to t.38 and so... fax - ata - asterisk - avaya - fax, works if i disable t.38 in avaya but breaks if i enable t.38 :( |
| 07:45.40 | *** join/#asterisk oej (~olle@ns.webway.se) |
| 07:45.51 | benngard | can (maybe) live with that but hate when there is stuff that i dont understand :( |
| 07:59.43 | *** join/#asterisk basty (~basty@212.218.65.219) |
| 07:59.45 | basty | morning |
| 08:00.09 | basty | is somebody using asterisk 1.4.29 with snom phones ? I have a little Problem with the call forwarding |
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| 08:54.12 | dangerdan | Anyone alive? |
| 08:56.02 | *** join/#asterisk nickaugust (~anonymous@74-95-115-29-Colorado.hfc.comcastbusiness.net) |
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| 09:03.41 | dangerdan | Looking for a way to use asterisk across many nodes which are geograpihcally dispersed |
| 09:03.48 | dangerdan | Is this something dundi is for? |
| 09:05.40 | dangerdan | Is dundi supported? |
| 09:05.51 | *** join/#asterisk s2krish (~chatzilla@113.199.146.147) |
| 09:06.02 | s2krish | hay guys |
| 09:06.15 | *** join/#asterisk oej (~olle@static-213-115-251-100.sme.bredbandsbolaget.se) |
| 09:06.30 | s2krish | this tag helps to dial soft phone <a href="sip:205">dial 205</a> |
| 09:06.38 | s2krish | is there any way I can hang up shoftphone? |
| 09:10.32 | florz | sip:205 is not a URI, so that's not HTML |
| 09:11.08 | s2krish | if you render that code in browser, it will lunch softphone and dial 205 number |
| 09:11.34 | florz | no |
| 09:11.39 | s2krish | i tested this. |
| 09:11.45 | s2krish | works perfectly |
| 09:11.47 | *** join/#asterisk oej (~olle@static-213-115-251-100.sme.bredbandsbolaget.se) |
| 09:11.47 | florz | you can't "test" that |
| 09:11.51 | s2krish | why not.. |
| 09:12.02 | s2krish | i have 203, 204 and 205 extension |
| 09:12.12 | s2krish | and i did it from browser |
| 09:12.17 | s2krish | and doing now as well :-) |
| 09:12.29 | s2krish | now, my want some sort of link to hangup softphone call |
| 09:12.44 | *** join/#asterisk oej (~olle@static-213-115-251-100.sme.bredbandsbolaget.se) |
| 09:12.59 | florz | because the browser is free to do anything it likes if you give it non-HTML declared as HTML |
| 09:13.33 | norrec | hey florz do you know anything about adding in codecs? |
| 09:13.36 | florz | that is did what you wanted it to do is merely an accident |
| 09:13.46 | s2krish | i know that, "sip" lunch ur softphone |
| 09:13.49 | s2krish | and ask to dial |
| 09:14.07 | s2krish | like <a href="mailto:abc@mail.com">mail me</a> lunch ur mail client |
| 09:14.23 | s2krish | same way <a href="sip:205">dial 205</a> lunch softphone and dial to 205 |
| 09:14.34 | norrec | not all softphones tie into the browser |
| 09:14.38 | florz | what you think to "know" there is just your very own invention |
| 09:14.48 | s2krish | u try |
| 09:14.58 | s2krish | firefox, ie and chrome |
| 09:15.00 | florz | you still can't "try" that |
| 09:15.13 | s2krish | this is what I am looking here |
| 09:15.22 | florz | what happens is simply a matter of chance |
| 09:15.28 | norrec | what he's saying is that, gj it works for you, it may not work for everyone because its not considered part of standard html |
| 09:15.30 | s2krish | i am looking something to which send hang up |
| 09:15.42 | *** join/#asterisk scardinal (~supreme@0905ds1-rdo.0.fullrate.dk) |
| 09:15.44 | florz | anything could happen and it would be perfectly acceptable by the standards |
| 09:17.17 | norrec | do you know if you can add in iLBC support w.o recompileing asterisk? |
| 09:19.53 | florz | (and this is not about "not part of _standard_ html", but rather "this is not HTML and this won't ever be HTML") |
| 09:20.00 | *** join/#asterisk oej (~olle@static-213-115-251-100.sme.bredbandsbolaget.se) |
| 09:20.26 | florz | (for, this is not a SIP URI, and never will be) |
| 09:22.11 | s2krish | watever, it still serve my purpose with. that's good yet. as you can do <a href="mailto:mail@mail.com">mail me</a> |
| 09:22.30 | s2krish | if u can tell me how to hang up sotfphone, that makes sense here |
| 09:22.32 | *** join/#asterisk scardinal (~supreme@0905ds1-rdo.0.fullrate.dk) |
| 09:24.34 | s2krish | at least it works with X-lite, this is what wanted. |
| 09:25.24 | dangerdan | Has dundi support died? |
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| 09:28.06 | s2krish | do you mean, i can do that with help of dundi? |
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| 10:01.38 | amol112 | can anyone suggest me open source calll center application which has ACD ,PABX |
| 10:01.42 | kaldemar | dangerdan: dundi is one way to implement that, but it is not necessary |
| 10:02.29 | kaldemar | dangerdan: you can use dundi to lookup numbers from the other nodes |
| 10:03.29 | *** join/#asterisk d-k-t (~D@112.202.232.46) |
| 10:07.16 | basty | is somebody using asterisk 1.4.29 with snom phones ? I have a little Problem with the call forwarding |
| 10:08.31 | kaldemar | ~ask |
| 10:08.32 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
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| 10:44.55 | ruyo | Hello, does anyone know if/how I can catch DTMF for an AGI application on an ongoing call? |
| 10:45.35 | ruyo | On an ongoing call as a call between two parties talking to eachother, not stream/ivr/etc. |
| 10:49.39 | *** join/#asterisk faiz_grw (~faiz_grw@209.17.184.117) |
| 10:57.17 | Amorsen | Heh, Asterisk dial plan is a funny language |
| 10:57.20 | Amorsen | exten => _X.,n,Verbose(${FILTER(0-9,1)2)}) |
| 10:57.30 | Amorsen | That one actually works and FILTER returns 12 |
| 10:57.59 | Amorsen | I wonder how it knows that it can eat the first bracket but not the second |
| 10:58.44 | faiz_grw | hi guy's, just wondering if i can get some help on fake ring with asterisk |
| 10:59.53 | faiz_grw | i need fake ring but i don't want to user option "r" with dial plan coz i won't transfer any audio from the callee |
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| 11:27.45 | asteriskATmarmuD | hello everyone |
| 11:28.15 | asteriskATmarmuD | can someone give me infos on how to get connection status from asterisk to other connected serevrs |
| 11:29.03 | asteriskATmarmuD | for example we need to trigger some stuff, if the connection is established and need to delete unavailable numbers |
| 11:29.26 | asteriskATmarmuD | and of course if the line is "busy" |
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| 12:17.10 | *** join/#asterisk Victor_Yure (~victor@unaffiliated/victoryure/x-837844) |
| 12:24.46 | *** join/#asterisk Orbixx (Orbixx@office.exoware.net) |
| 12:27.05 | plundra | In my dialplan, how do I acquire the status of a certain queue-member, if it's paused or not, is what I want to know. |
| 12:27.38 | Orbixx | Why would registration be failing due to 'Not a local domain'? |
| 12:27.44 | plundra | My first thought of a hack, was to try a Unpause/Pause, but they won't return something like "was already paused". Just NOTFOUND/<it-was-successful> |
| 12:30.43 | *** join/#asterisk krion (~seb@unaffiliated/krion) |
| 12:32.07 | FSB_1 | ~book |
| 12:32.07 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 12:32.30 | FSB_1 | Thank you infobot |
| 12:32.30 | infobot | FSB_1: bitte |
| 12:38.20 | *** join/#asterisk anonymouz666 (~anonymouz@189.24.62.66) |
| 12:39.33 | benngard | can any halp me with a nat/t38 problem? fax - ata - no nat - asterisk - no nat -ata fax works with t38, fax - ata - nat - asterisk - no nat - ata -fax fails completly |
| 12:40.09 | benngard | i did try nat=yes, directmedia=no, any more stuff that i can try? |
| 12:41.46 | *** join/#asterisk OrNix (~ornix@l151-249-47.static.cn.ru) |
| 12:45.36 | kaldemar | benngard: rest of the parameters that affect nat, see: |
| 12:45.39 | kaldemar | ~sipnat |
| 12:45.40 | infobot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 12:49.45 | garymc | Hey Oneliner that stuff you gave me last night works a charm. Thanks alot. It helped me to get a good sleep then do it today Thanks your a GEM! |
| 12:55.48 | amol112 | hi all , i have compiled asterisk and installed on debian lenny now searching for open source call center application.can anyone suggest me |
| 13:03.59 | *** join/#asterisk hatpanda (~pete-joh@triton.dsv.su.se) |
| 13:04.47 | amol112 | which application should i use for creating call center using asterisk |
| 13:05.04 | amol112 | vicidial or any other? |
| 13:05.36 | *** part/#asterisk benngard (~benngard@213.88.138.230) |
| 13:09.22 | hatpanda | Anyone tired using asterisk as 3g gateway, SIP Terminal<=GW SIP/ISDN PRI=>3G Cell phone? |
| 13:10.23 | *** part/#asterisk GabrielPiassetta (~gabriel@189.114.202.96.dynamic.adsl.gvt.net.br) |
| 13:12.02 | kaldemar | hatpanda: plenty of people |
| 13:13.22 | *** join/#asterisk tkrn (~tkrn@216.196.213.246) |
| 13:13.23 | amol112 | kaldemar, i am new to asterisk and call center application.so got confused with choosing call center application |
| 13:13.57 | hatpanda | kaldemar: Do you need some kind of partnership with a 3g operator for videocalls, or will it work if the protocol negotiation is correct? |
| 13:14.01 | amol112 | it will be better if any one suggest best call center application |
| 13:16.24 | *** join/#asterisk albasheers (~basheer@188.116.235.226) |
| 13:17.12 | *** join/#asterisk mefiX (~mef@meff03.synserver.de) |
| 13:17.23 | kaldemar | hatpanda: i'm not 100% sure but i'd say that there is no support for video calls over PRI. only VoIP. |
| 13:17.58 | amol112 | i think i am asking wrong question? |
| 13:18.36 | mefiX | hey guys. how can i force asterisk to prepend a "0" to the number that is dialed when using one of the local capi-controllers? asterisk is running behind isdn |
| 13:18.40 | albasheers | Best way to learn asterisk , any video tutorials |
| 13:19.02 | kaldemar | amol112: you better elaborate your needs a bit. "call center application" is quite vague. |
| 13:20.10 | kaldemar | mefiX: show your dialplan, i.e. how you handle the number when the call comes in. |
| 13:20.14 | kaldemar | ~book |
| 13:20.15 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 13:20.18 | kaldemar | albasheers: ^ |
| 13:20.41 | albasheers | thanks kaldemar |
| 13:20.57 | mefiX | kaldemar: http://pastebin.com/m2a72cb50 |
| 13:21.12 | mefiX | kaldemar: this works, but i have to prepend a "0" manually |
| 13:21.38 | kaldemar | mefiX: replace CAPI/ISDN1/${EXTEN} with CAPI/ISDN1/0${EXTEN}. that's it. |
| 13:21.46 | amol112 | kaldemar, i am looking for handling calls which are coming from mobile or landline |
| 13:21.48 | mefiX | that would be simple |
| 13:21.53 | *** join/#asterisk oej (~olle@static-213-115-251-100.sme.bredbandsbolaget.se) |
| 13:22.03 | amol112 | and having database of question answer etc |
| 13:22.24 | amol112 | call forwarding,conference |
| 13:26.48 | hatpanda | kaldemar: according to voip-info its possible to multiplex video and audio over one isdn channel. So it possible, but i cant try it because i dont have a pri connection :( |
| 13:26.49 | *** join/#asterisk niekie (~niek@CAcert/Assurer/niekie) |
| 13:27.13 | hatpanda | kaldemar: also, im not sure if the receiving network operator will accept video from my pri channel |
| 13:27.25 | *** join/#asterisk slidesinger (~slidesing@c-68-44-99-50.hsd1.nj.comcast.net) |
| 13:27.43 | hatpanda | kaldemar: or if i need to talk them into doing it, by either being diplomatic or by using maffia methods |
| 13:28.45 | *** join/#asterisk [TK]D-Fender (~chatzilla@216.191.106.163) |
| 13:28.51 | amol112 | kaldemar, asterisk require at client side or only at server side? |
| 13:29.30 | hatpanda | amol112: There is a good book for you. Please read The future of Telephony |
| 13:29.57 | *** join/#asterisk Akiraa (~Akira@92.81.218.220) |
| 13:30.16 | *** join/#asterisk e4 (~e4@rrcs-76-79-59-194.west.biz.rr.com) |
| 13:32.55 | kaldemar | hatpanda: and you'd need video support for POTS, the telco would need to be able to convert it into 3G format.. i have my fingers crossed for you to get it working some day. :) |
| 13:33.14 | kaldemar | amol112: yes, asterisk is a server or a client. listen to hatpanda. |
| 13:34.19 | *** join/#asterisk SeriousMatters (~Sirius@87.115.170.210) |
| 13:34.38 | *** join/#asterisk donatas (~donatas@office.kis.lt) |
| 13:36.19 | hatpanda | kaldemar: No side is using pots, one side is using SIP and the other side is 3G. |
| 13:36.52 | kaldemar | it's going through PRI is it? |
| 13:37.38 | hatpanda | kaldemar: Yep. the gateway communicates with ISDN to 3g network operator |
| 13:37.52 | hatpanda | ISDN PRI |
| 13:40.14 | *** join/#asterisk jaytee (~jforde@unaffiliated/jaytee) |
| 13:46.55 | *** join/#asterisk c0rnoTa (~c0rnoTa@94.25.175.235) |
| 13:55.12 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
| 13:57.11 | *** join/#asterisk rossand (~aross@CPE000c413a19a3-CM00159a025ad4.cpe.net.cable.rogers.com) |
| 14:04.05 | *** join/#asterisk spenguin[work] (~penguin@59.162.86.164) |
| 14:04.12 | *** join/#asterisk coppice (~chatzilla@94.201.17.210.dyn.pacific.net.hk) |
| 14:05.04 | *** join/#asterisk Ze^3k (~anonymous@pdpc/supporter/active/zeeek) |
| 14:06.51 | *** join/#asterisk atis_work (~atis_work@193.238.212.171) |
| 14:07.23 | *** join/#asterisk nickaugust (~anonymous@74-95-115-29-Colorado.hfc.comcastbusiness.net) |
| 14:08.01 | Katty | hi |
| 14:09.38 | *** join/#asterisk andres833 (~andres833@190.144.75.22) |
| 14:10.06 | asteriskATmarmuD | anyone got a good AMI solution to get the status of all channels transferred to another server. we want to take action on the status of the channels |
| 14:10.33 | asteriskATmarmuD | if busy do... if answering machine... and so on |
| 14:11.40 | *** join/#asterisk jmacz (~jmacz@190.144.75.22) |
| 14:13.28 | asteriskATmarmuD | and if the called person picks up, connect to inbound analog phone |
| 14:14.30 | Katty | stretches, yawns |
| 14:14.40 | Katty | blinks |
| 14:15.20 | Ze^3k | {{{{{Katty}}}}} |
| 14:15.47 | Katty | gosh |
| 14:15.50 | Katty | what happened to your /nick |
| 14:17.20 | Ze^3k | Katty: someone truncated it, fining it too long |
| 14:17.36 | Ze^3k | wait it's the same number of chars |
| 14:18.06 | NotKatty | Much better |
| 14:18.43 | NotKatty | what's new in the asterisk realm? |
| 14:19.40 | kaldemar | topping |
| 14:19.46 | nickaugust | Ok, I think I' going crazy... but are call in the queue handled differently than parked calls? I can transfer parked calls by passing thier channel name to the manager along with the desired extension... but when I try to do the same thing with queued calls I get channel does not exist... even though it does! and it hangs up on the caller... any ideas? |
| 14:20.25 | *** join/#asterisk H4U (~AdamH@79-121-154-141.eurotel.managedbroadband.co.uk) |
| 14:21.47 | nickaugust | also, can I transfer a call back into a queue like I can the parking lot? |
| 14:21.55 | *** join/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 14:22.44 | Katty | hugs Ze^3k |
| 14:23.26 | Ze^3k | feels much better now |
| 14:24.22 | *** join/#asterisk moy (~moy@74.12.129.100) |
| 14:27.16 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 14:27.16 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 14:27.47 | Katty | i watched a group of green wing macaws play ball this morning |
| 14:28.53 | Ze^3k | Is there a gree wing macaws webcam? |
| 14:29.01 | Ze^3k | leifmadsen: |
| 14:29.14 | leifmadsen | Ze^3k: sir yes sir?! |
| 14:29.24 | Ze^3k | Tennnnnnnnnnnch Hut! |
| 14:29.34 | leifmadsen | stands at attention |
| 14:29.52 | Ze^3k | leifmadsen: do you think much has changed in VoIP or Asterisk since March 2007 (when the VUC began)? |
| 14:29.59 | leifmadsen | yes. |
| 14:30.07 | Ze^3k | I mean major, not 1,000,000,000 bug fixes |
| 14:30.13 | *** join/#asterisk Buklov (~buklov@213.138.71.254) |
| 14:30.18 | leifmadsen | ummmm, depends :) I say yes. |
| 14:30.26 | leifmadsen | there are some really cool new features! |
| 14:30.30 | Ze^3k | so can you start making a list for our 3rd anniversary show? |
| 14:30.34 | H4U | Any Skype and Asterisk (switchvox) solutions? I found something yesterday for about £60 but carnt seem to find it, already have skype in number. not fond on ditch'ing it |
| 14:30.39 | leifmadsen | oh, homework... I see! |
| 14:30.41 | Ze^3k | We need to do this on March 26th, 2010 |
| 14:31.02 | leifmadsen | Ze^3k: yes, can you send me an email so I can do some research and schedule time in to do it? |
| 14:31.09 | leifmadsen | today is already packed (not that you need it today) |
| 14:31.10 | Ze^3k | If buzz wroked better we could do it there. If Wave worked better, we could use that. |
| 14:31.25 | leifmadsen | ya... no kidding, but email seems to still be the thing to use |
| 14:31.32 | Ze^3k | I suppose we might be able to drag a few people back to wave |
| 14:31.38 | leifmadsen | meh |
| 14:31.42 | leifmadsen | depends on what you're going to use it for |
| 14:31.46 | leifmadsen | an actual document, sure |
| 14:31.51 | Ze^3k | thought on the three year thing |
| 14:31.59 | Ze^3k | Actually that'd be good |
| 14:32.00 | leifmadsen | a social networking tool? that's not what wave is for |
| 14:32.04 | Ze^3k | because you could do an outline |
| 14:32.07 | leifmadsen | yep |
| 14:32.10 | Ze^3k | naw, the doc |
| 14:32.13 | leifmadsen | right |
| 14:32.14 | Katty | Ze^3k: no, but it was on youtube. |
| 14:32.17 | Ze^3k | thern everyone could add shit to it |
| 14:32.18 | Ze^3k | $ |
| 14:32.21 | Katty | Ze^3k: just as adorable tho |
| 14:32.26 | *** join/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 14:32.40 | leifmadsen | gah, already wasted 30 mins of my allocated 60 mins of blogging time |
| 14:32.42 | leifmadsen | back in 30 |
| 14:32.46 | Ze^3k | heh |
| 14:32.50 | Ze^3k | poor leif |
| 14:33.02 | Ze^3k | all these younguns are in such a hurry these days |
| 14:33.03 | leifmadsen | I even had to schedule lunch in because my day is so packed |
| 14:33.12 | leifmadsen | well, I'm leaving at 3:30pm today, so... :) |
| 14:33.24 | Katty | oh yeah |
| 14:33.26 | Ze^3k | I not only had lunch but took my 7,000 step walk before going on the Friday beer run |
| 14:33.26 | Katty | it's friday isn't it |
| 14:33.38 | Ze^3k | VUC.ME IT'S FRIDAY! |
| 14:33.44 | Ze^3k | ^^^^^yes it is ^^^^^^ |
| 14:33.53 | Ze^3k | http://vuc.me |
| 14:34.22 | *** join/#asterisk Geminizer (~me@cpe-76-180-27-4.buffalo.res.rr.com) |
| 14:34.35 | *** join/#asterisk mnt_real (~sinan@bas1-montreal43-1177754737.dsl.bell.ca) |
| 14:34.53 | Katty | heh. i think it's funny how vista just asked me 6 times if i really wanted to replace these two files |
| 14:35.15 | Geminizer | why are people still using vista? |
| 14:35.38 | jaytee | money? |
| 14:35.39 | Katty | that's a silly question |
| 14:35.42 | Katty | you should know the answer |
| 14:36.28 | Geminizer | win 7 demands the same hardware requirements as vista and is far better in terms of usability |
| 14:36.43 | Ze^3k | I use XP |
| 14:36.59 | jaytee | Win 7 is much better. the asynchronous driver loading really speeds up the boot process |
| 14:37.00 | Katty | why would a company go out and buy windows 7 for all their workstations when xp and vista is still working fine for their users |
| 14:37.04 | Ze^3k | Skipped Vista. New laptop uses W7. Not much better than XP |
| 14:37.11 | tuxx- | why would a company ever use windows? |
| 14:37.14 | tuxx- | that the big question |
| 14:37.14 | tuxx- | :D |
| 14:37.17 | tuxx- | *thats |
| 14:37.19 | Ze^3k | no can Vista, use XP or 7 |
| 14:37.26 | jaytee | because it's so ubiquitous |
| 14:37.27 | Katty | tuxx-: because that's what their people know |
| 14:37.37 | Katty | tuxx-: why invest tons of money to train all your people if they already understand a platform |
| 14:37.41 | tuxx- | Katty: i call that PEBKAC |
| 14:37.51 | Katty | call it whatever you want |
| 14:37.54 | tuxx- | :) |
| 14:38.00 | Katty | but that's how companies operate |
| 14:38.02 | Katty | money money money |
| 14:38.03 | jaytee | and most of the dolts in society that barely understand what a computer is let alone how it functions use Windows so big business buys what people know |
| 14:38.04 | Geminizer | good point, Angela |
| 14:38.06 | tuxx- | sadly, it it :P |
| 14:39.30 | jaytee | until an OS comes along that instantly captivates over 85% of the population and leaves them going "Oooooh! Sparkly!!! Shiny!!! Me Want!!!!" then Windows will remain dominant. |
| 14:39.58 | Ze^3k | dare I say it? PENETRATION |
| 14:40.06 | jaytee | sure you can |
| 14:40.07 | Ze^3k | INERTIA |
| 14:40.16 | jaytee | SEDIMENTATION |
| 14:40.32 | Ze^3k | and just as I did, "Updates are ready for your computer" popped up on the one next to me |
| 14:40.43 | Ze^3k | ATROPHY |
| 14:41.18 | leifmadsen | people who say "Why would anyone use Windows in a company?" obviously has never managed dozens or hundreds of workstations for a company |
| 14:41.24 | Ze^3k | I boot XP once a week because I need tow computers to do the VoIP Users Conference |
| 14:41.25 | leifmadsen | elitist bastards I say :) |
| 14:41.28 | jaytee | well, Linux will never dominate the desktop because for the average user it's still too confusing and there are too many distro choices |
| 14:41.38 | *** join/#asterisk ccesario (~ccesario@189-19-6-236.dsl.telesp.net.br) |
| 14:41.57 | Ze^3k | jaytee: that's why all other distros should die except for ${YOURS} |
| 14:42.43 | jaytee | it's like one of those crew races at an Ivy League college. You have the Windows devs in one boat all rowing in basically the same direction in rhythm and the Linux devs all rowing at different points of the compass and out of synch with each other. |
| 14:42.59 | Ze^3k | interesting analogy |
| 14:43.45 | jaytee | the development efforts in the linux community are admirable but there is so much waste on duplicated effort with out a cohesive standard. |
| 14:43.59 | *** join/#asterisk jabka (~bazbar@109.64.125.33) |
| 14:44.02 | *** join/#asterisk Faithful (~Faithful@ns.linuxterminal.com) |
| 14:44.13 | ManxPower-work | hugs his linux desktop |
| 14:44.25 | jaytee | this system is running Ubuntu |
| 14:44.29 | rue_house | before I read the manual again, I can "text" a polycom phone? |
| 14:44.51 | rue_house | oh it prolly has a mwi light |
| 14:44.53 | rue_house | hmm |
| 14:44.59 | jabka | Just to be sure is it a common task to assign IVR command to run spesific task (start the washer , close the lightes , port scan some laptop) |
| 14:45.05 | rue_house | cant remember if the aastra has one |
| 14:45.06 | drmessano | I can give you dozens of examples of simple things that needed complicated workarounds on even Ubuntu 9.10 |
| 14:45.24 | jabka | I mean easy task |
| 14:45.28 | rue_house | jabka, a guy I know made it flush his toilet |
| 14:45.32 | jaytee | I also have a Win 7 machine on a KVM to share the monitor and keyboard. At home I have 2 linux boxes, 1 Ubuntu and the other CentOS and then one Win 7 pc |
| 14:46.30 | jabka | lol , nice |
| 14:46.32 | jabka | rue_house , so it is common right ? I wish to email the gays at sce saing that an IVR then is connected to other facilities isn't their invention |
| 14:46.34 | visz | thinclients connected to a linux server with kde works ok |
| 14:46.37 | jaytee | do you know the main difference in the average linux developer and the average Windows developer? the former drives a Hyundai and the latter drives a BMW. |
| 14:47.00 | visz | allthough the lusers do bitch about openoffice time to time |
| 14:47.11 | jabka | and you can use one IVR to dosens of diffrent hardware |
| 14:47.24 | jaytee | they'd bitch about MS Office if they were running that, user bitch no matter what |
| 14:47.25 | drmessano | To run Skype on Ubuntu and get video with my LOGITECH cam, bash -c 'LD_PRELOAD=/usr/lib32/libv4l/v4l2convert.so skype' So easy, my Dad could do it |
| 14:47.31 | *** join/#asterisk muiro (~muiro@unaffiliated/muiro) |
| 14:47.34 | jaytee | s/user/users |
| 14:47.40 | rue_house | hmm I need to hook mine to the furnace control |
| 14:47.50 | rue_house | I havn't done it |
| 14:47.50 | jaytee | drmessano, LOL |
| 14:48.10 | jabka | furnace ? |
| 14:48.27 | Chainsaw | leifmadsen: Could 16470 block 1.6.2.3 please? |
| 14:48.33 | ManxPower-work | jabka: household heat |
| 14:48.38 | jabka | oo |
| 14:48.40 | leifmadsen | M16470 |
| 14:48.42 | MuffinMan | [ready for testing] [Asterisk] General 0016470: [patch] 'core stop when convenient' causes segfault reported by kjotte https://issues.asterisk.org/view.php?id=16470 |
| 14:48.50 | jabka | I thoght that I missunderstood |
| 14:48.54 | Chainsaw | leifmadsen: (It seems to also solve an intermittent memory corruption for me) |
| 14:49.02 | jabka | press one to freeze the house |
| 14:49.08 | drmessano | jaytee lets not get into nvidia drivers being HORRIBLY broken and not detecting DVI monitors properly, thus requiring 3 worksarounds I had to learn seperately to get my 22" LCD working |
| 14:49.11 | leifmadsen | Chainsaw: ok, I'll make the request, but I can't guarantee anything |
| 14:49.11 | jaytee | back when I was running Dapper it took me about an hour of messing with my xorg.conf and trying different versions of my nvidia driver to get my widescreen display for 1440x900 working. in XP and Vista it took less than a minute |
| 14:49.19 | Chainsaw | leifmadsen: *nod* Thank you. |
| 14:49.20 | *** join/#asterisk c0rnoTa (~c0rnoTa@94.25.175.235) |
| 14:49.22 | krion | quick question |
| 14:49.35 | jaytee | drmessano, haha we've both felt the pain then :-) |
| 14:49.37 | leifmadsen | Chainsaw: please add a note as to why you think this should be a blocker, and I'll follow up on it |
| 14:49.39 | drmessano | LOL |
| 14:49.42 | drmessano | Jaytee... YES |
| 14:49.45 | krion | is it possible with asterisk 1.2 acting as a centrex do to only sip signaling |
| 14:49.46 | Ze^3k | Katty: I am uploading a cute animal video I shot at lunhc. (not the animal, the video) |
| 14:49.47 | jaytee | drmessano, it's gotten much better though |
| 14:49.54 | drmessano | lol |
| 14:49.58 | drmessano | Not as of last week |
| 14:50.03 | krion | and not manage rtp flux if you know what i mean... |
| 14:50.15 | jaytee | well, for me anyways. as long as there's not a major kernel update |
| 14:50.38 | leifmadsen | the rtp flux capacitor? |
| 14:50.49 | ManxPower-work | krion: no. Asterisk is not a SIP proxy. |
| 14:50.58 | jaytee | mmmm, mango chipotle seasoning. I'm gonna have to buy some of that! |
| 14:52.01 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 14:52.01 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:52.05 | Chainsaw | leifmadsen: "Could this block 1.6.2.3 as applying the attached patch solves an intermittent memory corruption in the SIP register address for me (corruption manifests as an OR with 0x40 on the 11th, 12th or 13th character)." <- Note added. |
| 14:52.05 | drmessano | jaytee: Hooked up the monitor with the VGA cable, saved the EDID, set xorg.conf to ignore the CRC, switched back to DVI, and a reboot later, bam.. So easy my Mom could do it |
| 14:52.28 | Chainsaw | drmessano: Yuck, that sounds like a ViewSonic. |
| 14:52.33 | drmessano | Acer |
| 14:52.45 | drmessano | The Viewsonic second monitor was easy |
| 14:53.18 | drmessano | But the Viewsonic is not DVi |
| 14:55.23 | Chainsaw | drmessano: We had entire batches of those self-destruct on DVI. |
| 14:55.50 | Chainsaw | drmessano: *image blank* *image comes back* *image blank* loops for a long time, getting progressively worse. And then in the end the DVI port is dead. |
| 14:56.16 | *** join/#asterisk nickaugust (~anonymous@74-95-115-29-Colorado.hfc.comcastbusiness.net) |
| 14:56.43 | *** part/#asterisk eject_ck (~eject_ck@haitek.cosmonova.net.ua) |
| 14:56.53 | *** part/#asterisk mefiX (~mef@meff03.synserver.de) |
| 14:57.02 | drmessano | http://www.pastebin.ca/1802701 <-- Plug N Play |
| 14:58.30 | asteriskATmarmuD | anyone got a good AMI solution to get the status of all channels transferred to another server. we want to take action on the status of the channels (by channels I mean all outbound lines the asterisk server is connected to or trying to connect to) |
| 15:00.02 | rue_house | jabka, http://www.voip-info.org/wiki/view/Asterisk+cmd+System |
| 15:00.12 | *** join/#asterisk mlarsen (~mlarsen@212.37.141.188) |
| 15:00.29 | jabka | rue_house , wow |
| 15:01.13 | drmessano | I will say that thankfully Google has indexed the *nix community well, because once I realize WHY something isnt working and know what to ask the great oracle specifically, its not hard to find from there |
| 15:01.29 | drmessano | But "Why doesnt my %$%#$ video work" is really overly vague |
| 15:01.46 | ManxPower-work | Video on linux is one of the last major issues. |
| 15:01.49 | Ze^3k | Cute animal video from today: http://www.youtube.com/watch?v=6o1xuvL78BI |
| 15:01.56 | Ze^3k | Katty: ^^^^ |
| 15:02.00 | ManxPower-work | The desktop issue is just crappy menu design. |
| 15:03.24 | rue_house | hu? |
| 15:03.34 | rue_house | ManxPower-work, what distro were you looking at? |
| 15:05.10 | Ze^3k | linux is one of the last major issues, period |
| 15:05.55 | Katty | Ze^3k: orly? |
| 15:05.58 | Katty | watches |
| 15:06.07 | *** join/#asterisk SeriousMatters (~Sirius@87.115.170.210) |
| 15:06.12 | ManxPower-work | rue_house: Fedors XFCE Spin |
| 15:06.15 | mlarsen | Would it be valid to use for example: GotoIfTime(*,sat-tue,*,*?context,s,1) or would I need to split it to two lines? |
| 15:06.17 | ManxPower-work | and Fedora too. |
| 15:06.45 | rue_house | well those are all junky redhat distros |
| 15:06.47 | drmessano | My three big issues were getting my monitors working on the desktop, getting my laptop screen brightness adjust and power management working so the fan didnt run at 100% constantly, and then Adobe Flash and Air just suck horribly and theres just not enough wrappers to wrap up that smelly crap |
| 15:06.54 | rue_house | try unbuntu |
| 15:08.01 | rue_house | I use true debian, but I dont mind battling it over a few things every so often |
| 15:08.04 | drmessano | libwrapperforlibflashwrappertowraptheoriginalwrapper.so |
| 15:08.41 | jaytee | ubuntu with two 20" widescreens with the desktop spanning both monitors, Compiz Fusion and AWN make for a real nice desktop experience. |
| 15:08.57 | jaytee | but it was a major PITA getting it tweaked right |
| 15:09.23 | rue_house | I think that adobe stuff is so terribly inefficient that not even free range public developers can make an efficeint replacement |
| 15:09.25 | drmessano | In order to get some things working in Linux, you must find a way to set the compatibility layer to that of a programmable coffee maker |
| 15:09.35 | coppice | 2 of Dell's new 27" 2560x1440 monitors is what I call a nice desktop |
| 15:10.17 | rue_house | your too windows, you dont need compatability layers if your doing things right |
| 15:10.27 | drmessano | Not correct |
| 15:10.30 | rue_house | I'm going to 2x17" and a 15" soon |
| 15:10.47 | Katty | work is so boring today |
| 15:10.52 | rue_house | time for me to get going to work |
| 15:10.53 | Katty | i think everyone took off work |
| 15:11.36 | drmessano | Actually need to use 32-bit flash with a wrapper on 64-bit Ubuntu because the 64-bit .so lags behind (it's based on 10.0 not 10.1), and the aforementioned Skype workaround |
| 15:11.55 | Ze^3k | no work |
| 15:12.00 | drmessano | Mom had a hard time with that one, but Dad helped her out |
| 15:12.03 | Katty | you're right. it's friday |
| 15:12.06 | Katty | there's no work on fridays |
| 15:12.08 | Katty | what was i thinking |
| 15:12.21 | Ze^3k | Friday is a reserved_day |
| 15:13.27 | visz | work week almost over |
| 15:13.40 | visz | still need to make an announcement for a customer |
| 15:13.46 | Ze^3k | work wee k over here |
| 15:13.47 | visz | pain |
| 15:14.48 | visz | jaytee, xmonad is the display manager of choice for me |
| 15:14.52 | *** join/#asterisk chazzm (~chazz@173-24-238-25.client.mchsi.com) |
| 15:14.53 | drmessano | Oh and my Acer 22" worked great once I set it up as analog, then told xorg.conf "No, really.. that's not a teapot" |
| 15:15.00 | visz | i don't like using mouse |
| 15:17.06 | *** join/#asterisk jmacz (~jmacz@190.144.75.22) |
| 15:18.50 | *** join/#asterisk Naikrovek (~jjohnson@unaffiliated/naikrovek) |
| 15:19.21 | Katty | sighs |
| 15:19.36 | Katty | Harry Rediger, one of the people running for Mayor just tried to friend me on facebook |
| 15:20.03 | jaytee | don't accept! |
| 15:20.05 | Naikrovek | heh |
| 15:20.25 | Katty | i don't think people should be promoting their own political agenda on facebook |
| 15:20.52 | jaytee | true, but when did shameless and politician ever become mutually exclusive? |
| 15:21.17 | jaytee | "Lies, lies, lies and more damn lies!" |
| 15:22.01 | Kobaz | or is it, lies, damned lies, and statistics |
| 15:23.37 | Katty | jaytee: point |
| 15:24.35 | jaytee | Katty: thataway ----> |
| 15:24.46 | jaytee | brb |
| 15:30.05 | Katty | Qwell: ping |
| 15:30.31 | Kobaz | do de do |
| 15:30.37 | Kobaz | making progress with my custom call queues |
| 15:32.12 | *** join/#asterisk voipmonk (~shido6@CPE002191f85581-CM001692568382.cpe.net.cable.rogers.com) |
| 15:32.19 | *** join/#asterisk Naikrovek (~jjohnson@unaffiliated/naikrovek) |
| 15:36.36 | *** join/#asterisk Deeewayne (~dwayne@75.76.254.162) |
| 15:36.36 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 15:38.38 | *** join/#asterisk steliosk (~Stelios@ipa107.2.tellas.gr) |
| 15:38.45 | Ze^3k | In about 1 hour, c'mon by to #vuc and join us for the VUC, the VoIP users Conference. Call in at sip:200901@login.zipdx.com in G722 or g711 |
| 15:39.32 | Kobaz | what's a vuc |
| 15:39.44 | visz | vuc me |
| 15:40.04 | Kobaz | vuc you |
| 15:40.08 | Katty | do we complimentary naps? |
| 15:40.08 | visz | sideways |
| 15:40.14 | *** join/#asterisk [8none1] (~8none1]@cerberus.franklinamerican.com) |
| 15:41.48 | Katty | takes a snooze on the couch |
| 15:43.09 | Katty | i need to use up some ground turkey and celery. |
| 15:43.11 | Katty | anyone got any ideas |
| 15:43.53 | Deeewayne | Katty, TurCelDucken ? |
| 15:43.54 | Ze^3k | The words after VUC, VoIP Users Conference |
| 15:44.05 | Katty | Deeewayne: what's that? |
| 15:44.15 | Deeewayne | like a turducken, but with celery too |
| 15:44.19 | *** join/#asterisk tnsampaio (~tiago@201-42-159-10.dsl.telesp.net.br) |
| 15:44.32 | Kobaz | duckturken |
| 15:44.42 | Katty | i don't get it |
| 15:44.49 | *** join/#asterisk benngard (~benngard@90-230-92-67-no148.tbcn.telia.com) |
| 15:44.50 | Kobaz | it's a southern meal |
| 15:44.53 | Katty | is this an actual dish |
| 15:44.56 | Katty | or are you making a funny |
| 15:45.00 | Kobaz | it's turkey stuffed with duck, stuffed with chicken |
| 15:45.04 | Deeewayne | katty, http://en.wikipedia.org/wiki/Turducken |
| 15:45.08 | Katty | oh |
| 15:45.09 | Katty | eww |
| 15:45.12 | Kobaz | hah |
| 15:45.14 | Kobaz | what eww? |
| 15:45.20 | Kobaz | it's food |
| 15:45.29 | Deeewayne | I've never actually seen a turducken |
| 15:45.33 | Katty | doesn't sound very yummy |
| 15:45.39 | Katty | besides this is ground turkey |
| 15:45.40 | Deeewayne | I but they could use celery |
| 15:45.41 | Katty | not whole turkey |
| 15:55.05 | Katty | maybe chili |
| 15:55.17 | Geminizer | is there a way to force drop a channel in asterisk? |
| 15:55.31 | Katty | hangup |
| 15:55.34 | *** join/#asterisk Tim_Toady (~moi@77.49.178.169.dsl.dyn.forthnet.gr) |
| 15:56.11 | Geminizer | I have set a timeout in my dialplan (e.g. 30 seconds). The call drops from the caller's perspective (no more sound), but the channel persists on the pbx. Strange, huh? |
| 15:56.20 | Geminizer | Thanks, Katty... I will try that |
| 15:56.32 | ManxPower-work | Geminizer: only if your dialplan is broken |
| 15:56.41 | Kobaz | Geminizer: in dialplan, Hangup() to hang up your channel, SoftHangup() to hang up other channels |
| 15:57.12 | ManxPower-work | Geminizer: which of the dozen or so timeouts are you setting? |
| 15:57.46 | *** join/#asterisk jpeeler (~jpeeler@asterisk/digium-software-dev/jpeeler) |
| 15:57.57 | Kobaz | Geminizer: call logs are always helpful |
| 15:58.00 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 15:58.25 | Geminizer | I have the following in order: exten => 1,n,Set(TIMEOUT(absolute)=30) |
| 15:58.25 | Geminizer | exten => 1,n,Dial(SIP/....,300,gr) |
| 15:59.06 | Geminizer | though now that you mention it, the timeout would apply to the "parent" channel (the one that called Dial(...)), right? |
| 15:59.07 | *** join/#asterisk elwerene (~lalala@vpn-s-8d3a328e.campus.uni-stuttgart.de) |
| 16:00.10 | ManxPower-work | Geminizer: you need to start with priority 1 |
| 16:00.12 | [TK]D-Fender | gemyes |
| 16:00.22 | ManxPower-work | and don't retype, paste\ |
| 16:01.10 | Geminizer | those two lines are in the middle of a context (I have stuff happening before these lines).. |
| 16:01.32 | ManxPower-work | as the priority after dial run the Hangup |
| 16:01.37 | elwerene | i'm having sound issues with my asterisk installation. Outgoing calls(zap -> * -> voip) are working, incoming calls(voip -> * -> zap) are ringing, but when i answer the phone there's no sound. |
| 16:01.52 | Ze^3k | said In about 1 hour, c'mon by to #vuc and join us for the VUC, the VoIP users Conference. Call in at sip:200901@login.zipdx.com in G722 or g711 |
| 16:01.53 | ManxPower-work | elwerene: set canreinvite=no in sip.,conf [general] |
| 16:01.59 | benngard | ipnat |
| 16:02.05 | benngard | ~sipnat |
| 16:02.06 | infobot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 16:02.09 | Geminizer | is there a way to set a timeout in the Dial(...) call? |
| 16:02.14 | benngard | ~natsip |
| 16:02.15 | ManxPower-work | elwerene: Are you behind NAT? |
| 16:02.15 | Ze^3k | NOYTE: That was a while ago, now the conference room will open in 45 minutes |
| 16:02.19 | elwerene | ManxPower-work: it's not nat problem |
| 16:02.23 | ManxPower-work | Geminizer: you already set that to 300 |
| 16:02.25 | elwerene | i am behind a nat |
| 16:02.31 | elwerene | but hello world works |
| 16:02.34 | *** join/#asterisk adnc (~numer@unaffiliated/adnc) |
| 16:02.35 | Geminizer | ahh, I see |
| 16:02.37 | ManxPower-work | elwerene: did you set localnet and externip? |
| 16:02.43 | *** join/#asterisk bmoraca_work (~bmoraca@66-242-174-254.ceres.bvn.net) |
| 16:02.49 | elwerene | only externip |
| 16:02.57 | ManxPower-work | elwerene: then it won't work |
| 16:03.41 | ManxPower-work | did you also forward the RTP and SIP ports on your NAT box? |
| 16:03.56 | elwerene | ManxPower-work: i can't do it |
| 16:04.02 | ManxPower-work | elwerene: Then it sucks to be you. |
| 16:04.10 | bmoraca_work | lol |
| 16:04.13 | elwerene | that's my problem, don't have the rights to do so |
| 16:04.14 | elwerene | :D |
| 16:04.24 | ManxPower-work | elwerene: then you won't be able to do what you want to do. |
| 16:05.22 | elwerene | the asterisk is only there as sip gateway to sipgate, i don't want to use it with a sip phone from external |
| 16:06.00 | Geminizer | so, if I have set a timeout of 30 seconds prior to calling Dial(SIP/..., 300, gr), the "parent" channel will drop out, leaving the secondary one open until Dial() completes (either due to hangup on either end or the timeout of 300 is reached)? |
| 16:06.04 | ManxPower-work | I don't care if you want to use Asterisk as a space ship. You are not going to get audio two way audio on incoming calls unless you set it up right. |
| 16:06.04 | elwerene | i had a hello world test in it and could here it when i called from outside |
| 16:06.15 | ManxPower-work | Geminizer: no. |
| 16:06.24 | ManxPower-work | elwerene: that is ONE WAY AUDIO |
| 16:06.30 | elwerene | damn |
| 16:06.50 | *** join/#asterisk rossand (~aross@CPE000c413a19a3-CM00159a025ad4.cpe.net.cable.rogers.com) |
| 16:06.55 | *** join/#asterisk javierg (~javier@200.49.23.142) |
| 16:07.01 | elwerene | so no chance, dump that asterisk box and stick to the expensive isdn line? |
| 16:07.06 | ManxPower-work | But feel free to spend days working on the issue. |
| 16:07.22 | bmoraca_work | elwerene: if all you're doing is registering asterisk to an outside ITSP, then you don't need port forwards. |
| 16:07.34 | ManxPower-work | elwerene: or fix your NAT problems or use something like SER/OpenSER, which unlike Asterisk, is an actual SIP proxy |
| 16:07.35 | bmoraca_work | elwerene: you WILL, however, need both externip and localnet specified. |
| 16:07.37 | javierg | How can i connect 2 asterisk servers one behind other and course all calls normaly? |
| 16:08.11 | Geminizer | ~book |
| 16:08.12 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 16:08.57 | bmoraca_work | elwerene: that does, of course, depend on your provider. for instance, the wholesale SIP provider I use for toll-free origination works by IP address only. in that case, I do need ports forwarded. |
| 16:09.20 | adnc | does someone know astlinux? there are versions that include asterisk 1.4 and 1.6 what would you experienced user say, gonig for 1.6 or 1.4? |
| 16:09.48 | Katty | OH NOES |
| 16:09.49 | Katty | OUT OF TEA |
| 16:09.50 | ManxPower-work | adnc: We don't go GUIs here. |
| 16:09.50 | elwerene | hmm but as the hello world test works, why can't i here my self when i am speaking from an isdn phone on zap? |
| 16:09.56 | Katty | NATIONAL CRISIS |
| 16:10.07 | Geminizer | Katty, are you from England ? |
| 16:10.14 | elwerene | it's the same direction.. |
| 16:10.17 | *** join/#asterisk Godfather_ (~Godfather@205.Red-88-0-3.dynamicIP.rima-tde.net) |
| 16:10.17 | ManxPower-work | elwerene: I ewish you the BEST of luck. |
| 16:10.33 | adnc | ManxPower-work, ohh, right it has a webgui. what about the asterisk version? |
| 16:10.35 | elwerene | ManxPower-work: thanks :) |
| 16:10.46 | Chainsaw | Geminizer: You don't have to be British to appreciate tea :) |
| 16:10.50 | ManxPower-work | adnc: since we don't use GUIs here, how would we know what version? |
| 16:11.02 | adnc | no, not the gui |
| 16:11.06 | adnc | asterisk version |
| 16:11.16 | Geminizer | :) Well, for it to be a national crisis, perhap |
| 16:11.17 | Geminizer | s |
| 16:11.29 | ManxPower-work | adnc: The current Asterisk versions can be found on downloads.digium.com. We don't know what version the AST linux people used. |
| 16:11.33 | ManxPower-work | Why don't you ASK THEM? |
| 16:11.35 | adnc | i suppose because i mentioned astlinux the answer is now dead |
| 16:11.46 | adnc | ManxPower-work, it is ok. thank you |
| 16:11.57 | Chainsaw | Geminizer: A few hundred years back, there was a big row about tea in a certain harbour. |
| 16:12.04 | Chainsaw | Geminizer: Remember that? :) |
| 16:12.28 | Geminizer | Nope... not a history buff, sorry |
| 16:12.29 | voipmonk | http://www.youtube.com/watch?v=mnH3EAES8_8 - Katty |
| 16:12.33 | ManxPower-work | Chainsaw: he's have to be pretty old to remember it. |
| 16:12.44 | voipmonk | one flight too long with a skymall magazine |
| 16:12.48 | javierg | It is possible connect 2 asterisk servers one (sip client) behind other (sip sever)? |
| 16:12.48 | Chainsaw | ManxPower-work: This is true. I was hoping it was discussed in school at one point. |
| 16:12.53 | Katty | crisis averted! |
| 16:12.56 | Katty | drink aquired :> |
| 16:12.57 | Chainsaw | Katty: Pfew :) |
| 16:13.03 | Katty | i knows, right :> |
| 16:13.53 | Katty | i thought i was going to have to make a tea run |
| 16:14.16 | eppigy | hello |
| 16:14.18 | eppigy | I am dave |
| 16:14.22 | Katty | oh em gee |
| 16:14.25 | Katty | it's dave :> |
| 16:14.27 | Katty | hugs eppigy |
| 16:14.28 | eppigy | HI |
| 16:14.32 | eppigy | huggles Katty |
| 16:14.39 | jaytee | DAVE! |
| 16:15.10 | [TK]D-Fender | javierg: Yes |
| 16:15.40 | eppigy | HI jaytee |
| 16:15.46 | [TK]D-Fender | javierg: * is jsut a SIP UA. Nothing more. Add peers. Register where needed. Set dialplan. The End. |
| 16:15.47 | Ze^3k | [TK–D-Fender will you consider joining us for our 3rd anniversary conference on March 26th? You were one of the early guests! |
| 16:15.57 | [TK]D-Fender | Ze^3k: 1st IIRC ;) |
| 16:16.02 | javierg | [TK]D-Fender: Do you have some guide or tutorial to do this ? |
| 16:16.14 | Ze^3k | [TK–D-Fender before Mark Spencer? |
| 16:16.24 | [TK]D-Fender | javierg: same as any ITSP or phone you've ever installed with * |
| 16:16.33 | [TK]D-Fender | Ze^3k: think so... |
| 16:16.36 | javierg | [TK]D-Fender: I did exactly that, but it doesnt work |
| 16:16.47 | Ze^3k | [TK]–D-Fender I can't remmeber how you were "billed"? |
| 16:16.53 | [TK]D-Fender | javierg: Do I see you showing me your code and failed call attempts with SIP DEBUG? |
| 16:16.57 | ManxPower-work | javierg: You understand that Asterisk isn't really a PBX, right? Asterisk is a TOOLKIT that helps you build a PBX, much like a compiler and libraries help you build an application. |
| 16:17.01 | Ze^3k | nor am I able to spell your nick :) |
| 16:17.06 | [TK]D-Fender | Ze^3k: on Polycom, Presence, NAT, etc |
| 16:17.07 | *** join/#asterisk atis_work (~atis_work@193.238.212.171) |
| 16:17.30 | [TK]D-Fender | javierg: PASTEBIN is your friend... |
| 16:17.32 | [TK]D-Fender | ~pb |
| 16:17.33 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , or apt-get install pastebinit |
| 16:17.50 | *** join/#asterisk garymc (~garymc@81.138.225.161) |
| 16:18.04 | ManxPower-work | Hi, I just started using GIMP. No, I've not read the documentation. How can I make a movie? |
| 16:18.20 | *** join/#asterisk Faithful (~Faithful@ns.linuxterminal.com) |
| 16:18.24 | Ze^3k | [TK]D-Fender Looks like April to me |
| 16:18.40 | Ze^3k | Every session was about Polycom in those days :) |
| 16:18.41 | garymc | Hey Guys Ive been offered a Cisco PoE 48 port switch model SGE2010P anyon egot one of these. Should I get it. Its pretty cheap. To buy one new would cost £1500 |
| 16:19.07 | ManxPower-work | garymc: would that be Cisco PoE, or the IEEE PoE? |
| 16:19.26 | garymc | I think its IEEE POE whats the differnce? |
| 16:19.31 | Naikrovek | important difference |
| 16:19.40 | Naikrovek | cisco PoE doesn't work on anything modern, except cisco |
| 16:19.44 | ManxPower-work | garymc: Cisco PoE only works with Cisco stuff, |
| 16:19.57 | Chainsaw | IEEE PoE involves a negotiation sequence. Cisco PoE appears to be *bam* *POWER!* |
| 16:19.57 | garymc | Ok well it says it is IEEE PoE 802.3af |
| 16:19.58 | Naikrovek | IEEE 802.3af is what all modern (including modern cisco stuff) uses |
| 16:20.01 | Ze^3k | If only there were something better than GIMP |
| 16:20.14 | ManxPower-work | Naikrovek: Is that switch a "modern" one? |
| 16:20.18 | Ze^3k | which is actually called The GIMP IIRC |
| 16:20.20 | Naikrovek | no idea |
| 16:20.24 | Naikrovek | but it has modern PoE |
| 16:20.30 | Naikrovek | and is worth a closer look |
| 16:20.44 | Naikrovek | he gave the price for a new one so maybe it is modern |
| 16:20.44 | ManxPower-work | garymc: find the specs on Cisco's web site and confirm. |
| 16:20.46 | Chainsaw | My Foundry switches will take a "legacy" option to do Cisco-style PoE. |
| 16:20.48 | garymc | So Ive got Polycom IP330's and Ip550 and Ip650 this should be fine then |
| 16:20.51 | garymc | I would assume |
| 16:21.00 | Naikrovek | garymc, double check on cisco's site, but it sounds like it |
| 16:21.05 | garymc | Yeah thats where i got my info |
| 16:21.09 | garymc | cisco website |
| 16:21.11 | Naikrovek | ah okay |
| 16:21.12 | Chainsaw | So IEEE PoE may not mean immediate doom. But yes, check first and be sure before money changes hands. |
| 16:21.13 | ManxPower-work | garymc: do you want to take a chance on blowing a phone? |
| 16:21.35 | garymc | Would this blow a phone? |
| 16:21.41 | ManxPower-work | and Cisco PoE does not blindly put power on the line. |
| 16:21.59 | garymc | Do they require setting up ? |
| 16:21.59 | ManxPower-work | garymc: I don't know, but I'm not going to try and gind out. |
| 16:22.16 | garymc | Well people must use PoE mustn they |
| 16:22.22 | ManxPower-work | chances are it just won't work. |
| 16:22.29 | Naikrovek | i think it'll work |
| 16:22.35 | Naikrovek | cisco site says 802.3af |
| 16:22.35 | bmoraca_work | garymc: SGE2010P isn't a real Cisco switch. it's a Linksys switch rebranded. it'll be 802.3af |
| 16:22.39 | ManxPower-work | I think we can't know |
| 16:22.55 | Naikrovek | it'll be fine |
| 16:22.56 | garymc | yes its a CISCO linksys |
| 16:22.56 | ManxPower-work | bmoraca_work: I thought the model number looked wonky. |
| 16:23.02 | Naikrovek | as did i |
| 16:23.07 | Naikrovek | it'll be fine |
| 16:23.09 | ManxPower-work | garymc: Nothing like leaving out that information until the end. |
| 16:23.13 | garymc | Cool Im gonna buy it then |
| 16:23.13 | Naikrovek | and it'll have poe turned on by default |
| 16:23.21 | garymc | yeah i do that alot :P |
| 16:23.24 | Naikrovek | heh |
| 16:23.27 | Geminizer | If TIMEOUT(absolute) is assigned a value in a context prior to calling Dial(...), does the "absolute" timeout apply to the call made through that Dial() (should it, that is) ? |
| 16:23.35 | bmoraca_work | ManxPower-work: yeah...they should prominantly put SMALL BUSINESS on all of them |
| 16:23.42 | bmoraca_work | or SOHO |
| 16:23.58 | bmoraca_work | additionally, real Cisco POE switches DO work fine with Polycom phones... |
| 16:24.01 | [TK]D-Fender | Geminizer: Yes |
| 16:24.15 | ManxPower-work | garymc: don't worry about it. Some day you'll forget to mention an alergy and die when someone gives you a peanutbutter sandwitch |
| 16:24.21 | [TK]D-Fender | [11:23]<bmoraca_work>additionally, real Cisco POE switches DO work fine with Polycom phones... <- POST cisco-Standard PoE models that is |
| 16:24.33 | ManxPower-work | Fitting when you think about it. |
| 16:24.37 | [TK]D-Fender | garymc: Be EXtREMELY sure that your switch supports 892.3af |
| 16:24.38 | Geminizer | k, thanks [TK]D-Fender |
| 16:24.38 | javierg | [TK]D-Fender: http://www.pastebin.org/94082 |
| 16:24.39 | bmoraca_work | the only caveat is that polycom phones don't respond to CDP in the bootloader, only in the application, so you have to tell it to wait 5 minutes |
| 16:24.41 | Naikrovek | i need to talk my management into a few 2960 PoE switches or, if i'm feeling lucky, some 3750s |
| 16:25.00 | bmoraca_work | Naikrovek: just get some used 3550s, they work great |
| 16:25.07 | Naikrovek | are they poe? |
| 16:25.12 | bmoraca_work | they can be |
| 16:25.17 | Naikrovek | need poe models |
| 16:25.24 | bmoraca_work | so get PoE models |
| 16:25.27 | javierg | [TK]D-Fender: http://www.pastebin.org/94082 -> thats my configuration on both servers. |
| 16:25.29 | Naikrovek | fine! i will! |
| 16:26.12 | Ze^3k | In 5 minutes it will be officially BEER O'CLOCK |
| 16:26.31 | Ze^3k | please adjust your watches accordingly |
| 16:26.40 | [TK]D-Fender | javierg: Where is the FAILED CALL with SIP DEBUG like I asked for? |
| 16:27.51 | garymc | [TK]D-Fender : Well according to the Cisco website that model number supports IEEE 802.3af |
| 16:28.39 | javierg | [TK]D-Fender: this is my CLI asterisk out (in client asterisk) http://www.pastebin.org/94083 |
| 16:29.18 | ManxPower-work | garymc: any Linksys PoE switch would be IEEE PoE |
| 16:29.27 | [TK]D-Fender | javierg: That shows NOTHING. Where is the DIAL attempt? Where is any sign that you actually tried to do something/ |
| 16:29.52 | ManxPower-work | [TK]D-Fender: I bet you'll still be working with him at 5pm |
| 16:30.20 | *** join/#asterisk hipitihop (~denis@203.132.229.187) |
| 16:30.27 | javierg | [TK]D-Fender: how can i view that?.... i do #asterisk -vvvr on debian with asterisk 1.4 |
| 16:30.30 | [TK]D-Fender | ManxPower-work: No, I'll drop it like a rock LONG before that comes to pass :) |
| 16:30.51 | [TK]D-Fender | javierg: "sip set debug on" |
| 16:32.53 | garymc | ok people, have funs im off. Have a good weekend |
| 16:32.56 | *** part/#asterisk garymc (~garymc@81.138.225.161) |
| 16:33.00 | ManxPower-work | javierg: remove the # at the start of the command |
| 16:35.03 | *** part/#asterisk adnc (~numer@unaffiliated/adnc) |
| 16:37.01 | javierg | Ok [TK]D-Fender this is my out, hope works for u: http://www.pastebin.org/94091 |
| 16:39.59 | [TK]D-Fender | javierg: That call enters the dialplan, calls that AGI which does nothiNG and just quits. It is not even trying to dial. |
| 16:40.03 | Ze^3k | leaving for the conference now #vuc or call sip:200901@login.zipdx.com in about 10 Minutes or see http://vuc.me |
| 16:40.05 | *** part/#asterisk Ze^3k (~anonymous@pdpc/supporter/active/zeeek) |
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| 16:41.33 | javierg | [TK]D-Fender: so the problem is in the a2billing configuration? |
| 16:42.00 | [TK]D-Fender | javierg: Yes |
| 16:42.17 | *** join/#asterisk albasheers (~basheer@188.116.235.226) |
| 16:42.32 | javierg | [TK]D-Fender: Ok... But the rest of config files like sip.conf on both servers are good? |
| 16:42.57 | [TK]D-Fender | javierg: How can we know?... you haven't tested them yet. |
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| 16:43.45 | javierg | [TK]D-Fender: Ok... Thank you so much [TK]D-Fender. I go for launch and later i gonna see whats the problem... Bye |
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| 16:54.18 | nickaugust | [TK]D-Fender: is it possible to have all incomming calls sent to the parking lot? |
| 16:54.42 | [TK]D-Fender | nickaugust: Sure |
| 16:55.11 | nickaugust | [TK]D-Fender: do I have to do it manually in the config file? theres no option in the gui |
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| 16:55.33 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
| 16:55.40 | [TK]D-Fender | nickaugust: Yes its possible. Can you do this conveniently in FreePBX? Porbably not |
| 16:55.47 | *** join/#asterisk adnc (~numer@unaffiliated/adnc) |
| 16:55.48 | Katty | so what's for lunch |
| 16:56.51 | adnc | ManxPower-work, astlinux is not really a gui for asterisk |
| 16:57.52 | krion | ManxPower-work: ok thanks, guess i should have a look at opensips |
| 16:59.04 | nickaugust | what conf file are the inboud routes configured in? |
| 16:59.47 | ManxPower-work | nickaugust: Asterisk does not have "inbound routes" |
| 17:00.02 | [TK]D-Fender | nickaugust: Doesn't work that way.... it gets generated by MagicClickyGUI. |
| 17:00.31 | [TK]D-Fender | nickaugust: You'll need to change the destination of all of your inbound routes, including the default to a custom context to do this. |
| 17:00.46 | ManxPower-work | ~freepbx |
| 17:00.47 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there, or fpbx |
| 17:02.21 | *** join/#asterisk binbash_ (~peter@ip4da53781.direct-adsl.nl) |
| 17:02.30 | nickaugust | [TK]D-Fender: well thats not a problem I only have one inbound route! |
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| 17:12.47 | iratik | Silly slightly offtopic question. I work in a customer contact center and currently for the supervisors who walk from each team center on the floor they use cordless handsets hooked up to ATA's. If we want to add the ability to monitor another simultaneous call (increase number of channels that can be monitored simultaneously) .. we must add an ATA and another cordless phone. Is there some sort of wireless base station out there that can ha |
| 17:12.47 | iratik | ve an arbitrary number of headsets and an arbitrary number of sip registrations so I don't keep having to purchase ATA's and cordless phones? |
| 17:13.18 | iratik | some sort of multi-registration capable wireless ip handset system? |
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| 17:16.53 | krion | Interarrival jitter: 1073 |
| 17:16.58 | krion | is that bad ? :-D |
| 17:17.18 | ManxPower-work | krion: that is so much worse than bad, you can't even imagine. |
| 17:17.53 | ManxPower-work | krion: can you even make phone calls? |
| 17:18.01 | krion | that's why i've huge blank when i call echo from a visiophone (g729+h264) |
| 17:18.28 | ManxPower-work | Fix your network. |
| 17:18.41 | iratik | I've never head of interarrival jitter.. is that a quality metric .. how do i obtain that metric for my own network? |
| 17:18.51 | p3nguin_ | iratik: Many of the cordless IP phones will use up to six handsets and support registrations for each. That doesn't really fit your need of an arbitrary number, though, since it is defined as six only. |
| 17:18.56 | krion | iratik: it's in the RTCP packets |
| 17:19.15 | iratik | p3nguin: recommendation for any cordless ip phones? |
| 17:19.26 | ManxPower-work | all cordless ip phones suck |
| 17:20.26 | Chainsaw | But in subtly different ways. |
| 17:21.25 | krion | ManxPower-work: i don't have that problem when i don't enable h.264 |
| 17:21.56 | *** part/#asterisk asteriskATmarmuD (~mundt@193.158.65.23) |
| 17:22.16 | krion | zsh: segmentation fault wireshark test.pcap |
| 17:22.19 | krion | Oops. :) |
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| 17:24.10 | p3nguin_ | iratik: Maybe Siemens Gigaset A580 IP |
| 17:24.20 | BCS-Satori | I am trying to use the built in http to asterisk to service phoneprov to polycom phones; however when I go to the URL I get "not implemented. attempt to use unimplemented/unsupported method -- Asterisk Server". I am using 1.6.2.2 |
| 17:24.25 | iratik | Thanks |
| 17:24.36 | ManxPower-work | BCS-Satori: I highly doubt that will work |
| 17:24.48 | paulc | iratik: how about IP enabled cell phones.. Nokia N95 worked great for me. |
| 17:25.00 | *** join/#asterisk nickaugust (~anonymous@74-95-115-29-Colorado.hfc.comcastbusiness.net) |
| 17:26.13 | p3nguin_ | iratik: $70 USD for the Gigaset A580 IP base with one handset. Extra A58H handsets are around $40 USD each. |
| 17:26.36 | *** join/#asterisk rubberneck (~chatzilla@ext-52.sagetelecom.net) |
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| 17:28.15 | BCS-Satori | ManxPower-work: Highly doubt what will work? |
| 17:28.41 | p3nguin_ | iratik: For $270 USD, you can have the base and six handsets. |
| 17:29.00 | ManxPower-work | BCS-Satori: getting your Polycoms to provision off the very limited build in "web server" Asterisk 1.6.x has. |
| 17:29.23 | ManxPower-work | not to mention you'll have quite a few issues with lost settings since the phones can't upload their local config changes. |
| 17:31.00 | BCS-Satori | ManxPower-work: I am not sure I want the users to be able to save local config settings; which is why i choose the http rather then ftp. However I am having great issues figuring out the phoneprov; can't seem to find a lot of documenation on it. I only always done it by hand in 1.4 |
| 17:31.41 | ManxPower-work | BCS-Satori: so you want the phones to lose their directory/contacts every time a firmware upgrade happens? |
| 17:32.08 | ManxPower-work | as well as their call forwarding settings, DND, custom ring tones, etc. |
| 17:33.51 | [TK]D-Fender | BCS-Satori: There is extremely little. It was a mistake to create.... |
| 17:34.46 | BCS-Satori | ManxPower-work: If the bug is fixed (which I am sure it is after 1 year) then yes I'll let them save it; however 1 year ago when someone adjusted something (ie ringtone) it would wipe my whole configuration out and replace it with just 1 line about the ringtone. But thats minor; i'm still trying to figure out the phoneprov |
| 17:35.02 | BCS-Satori | [TK]D-Fender: the http server or phoneprov? |
| 17:35.11 | [TK]D-Fender | Ok, I have an idea....bcsBOTH |
| 17:35.14 | [TK]D-Fender | BCS-Satori: BOTH |
| 17:35.16 | ManxPower-work | I thought phoneprov was only for IAX phones. |
| 17:36.53 | paulc | BCS-Satori: In recent playings with Polycoms, I've had no problems with local config changes (ring tones etc) wiping out global config settings (using FTP) |
| 17:37.31 | paulc | FTP = simple, it works, correct permissions prevent screwing up of static config data, NICE. |
| 17:38.36 | bmoraca_work | polycom phones write a completely different file for user-specified configs such as ringtone or wallpaper |
| 17:39.01 | bmoraca_work | if it's wiping out important things, such as your registration information, then you're putting that in the wrong place |
| 17:39.13 | Naikrovek | there's a note about that in the polycom documentation |
| 17:39.44 | Naikrovek | don't name your config files [MAC]-phone.cfg. name them [MAC]-phone1.cfg (note the 1) the former will get overwritten by local phone directory |
| 17:39.49 | Naikrovek | if you configure them a certain way |
| 17:40.13 | BCS-Satori | That may have been my mistake. |
| 17:40.16 | Naikrovek | my directory and overrides go into separate folders |
| 17:40.38 | *** join/#asterisk jdnWEST (~jdn@mx1.westparkcom.net) |
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| 17:42.50 | BCS-Satori | Thanks again all; valuable insight. |
| 17:43.22 | *** join/#asterisk xpot-mobile (~xpot@173-14-232-121-Utah.hfc.comcastbusiness.net) |
| 17:43.38 | ManxPower-work | READ the Polycom provisioing guide and whitepaper |
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| 17:45.54 | *** part/#asterisk moldy (~rene@unaffiliated/moldy) |
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| 17:53.49 | *** join/#asterisk bakermd (~bakermd@38.104.0.102) |
| 17:55.07 | bakermd | Question: I am trying to get app_swift to work. When Asterisk starts, it says that it cannot locate libceplang_es.so.5 The file exists in /opt/swift/lib -- where does Asterisk look for libs referenced by .so files? I imagine I can just make a symlink |
| 17:58.27 | ManxPower-work | bakermd: add that directory to /etc/ld.so.conf, then run ldconfig -v and try it again |
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| 18:00.22 | *** part/#asterisk asteriskATmarmuD (~mundt@193.158.65.23) |
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| 18:08.36 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 18:09.50 | bakermd | ManxPower-work: Thanks - did that from the docs, but they did not say -v .. that seems to have fixed it. Thanks!! |
| 18:11.47 | p3nguin_ | The only difference between ldconfig and ldconfig -v is that with -v you get to see what directories were scanned. |
| 18:11.59 | Katty | peeks in |
| 18:12.07 | Kobaz | pokes in |
| 18:13.17 | p3nguin_ | ssh sigma |
| 18:13.32 | p3nguin_ | Err... DAMN YOU, Alt+Tab! You have failed me. |
| 18:14.16 | Katty | dns failure. |
| 18:15.45 | Katty | who wants to play roulette |
| 18:16.03 | Kobaz | spins |
| 18:16.14 | Katty | redditroulette.appspot.com |
| 18:16.20 | Katty | see if you can find me! |
| 18:18.09 | b14ck | Katty, you read reddit? |
| 18:18.10 | p3nguin_ | It seems to be broken. I click Allow, but it does nothing. |
| 18:18.22 | Katty | cause you don't have a webcam? |
| 18:18.32 | Katty | just hit play |
| 18:18.33 | p3nguin_ | I do, actually. |
| 18:18.40 | Katty | select it from your drop down menu |
| 18:18.41 | Katty | for video |
| 18:18.41 | p3nguin_ | There's no play. |
| 18:18.59 | Katty | ^_- |
| 18:19.28 | Katty | does chattroulette work for you? |
| 18:19.42 | Katty | err chatroulette |
| 18:20.23 | Katty | it'll take forever to find me there tho |
| 18:20.31 | p3nguin_ | http://imagebin.org/85471 |
| 18:20.48 | Katty | looks |
| 18:20.57 | p3nguin_ | Everything is shaded, waiting for me to click Allow. |
| 18:21.18 | Katty | can you hit deny? |
| 18:21.24 | Katty | or tab |
| 18:21.26 | Katty | and then space |
| 18:21.28 | p3nguin_ | It won't accept a click to either. |
| 18:21.44 | Katty | what browser is it? |
| 18:22.37 | *** join/#asterisk joako (~joako@opensuse/member/joak0) |
| 18:22.41 | p3nguin_ | Tabbing cycles through the highlighted buttons on the page, but I can't space/enter on them. It's firefox 3.5. |
| 18:22.50 | Katty | that's crazy |
| 18:23.01 | joako | I need to quickly setup a machine with Sangoma T1 card and asterisk just as a SIP gateway. Should I use 1.4 or 1.6? |
| 18:23.03 | Katty | mine's 3.5.8 |
| 18:23.04 | *** join/#asterisk nix8n82 (~AndChat@63.162.27.14) |
| 18:23.14 | p3nguin_ | I just upgraded flashplugin last night, too. |
| 18:23.19 | Katty | hmm |
| 18:23.22 | Katty | try dumping your cache |
| 18:23.25 | p3nguin_ | What's the URI for the other one? |
| 18:23.34 | Katty | chatroulette.com |
| 18:23.58 | Katty | does that one work for you? |
| 18:25.44 | p3nguin_ | How friggin' annoying. Everything looks good at first, then I click Preview my cam, and the same flash permission box pops up... which I cannot click the buttons on. |
| 18:25.57 | Katty | )= |
| 18:26.02 | Katty | major bummer |
| 18:26.11 | p3nguin_ | Maybe a different browser will work better. |
| 18:26.19 | Katty | k |
| 18:26.42 | *** join/#asterisk lanning (~lanning@208.87.235.224) |
| 18:27.28 | p3nguin_ | It's a flashplugin problem. |
| 18:27.41 | p3nguin_ | Same results on Chromium. |
| 18:28.32 | p3nguin_ | [2010-02-18 18:34] upgraded flashplugin (10.0.22.87-1 -> 10.0.45.2-1) |
| 18:28.53 | Katty | :< |
| 18:29.13 | p3nguin_ | [2010-02-19 12:29] upgraded flashplugin (10.0.45.2-1 -> 10.0.22.87-1) |
| 18:29.30 | bmoraca_work | how many times does a telco need to reject a port-out for spurious reasons before I should file a complaint with the PUC and the FCC? |
| 18:29.49 | Katty | no idea, i'm not a telco |
| 18:29.51 | Katty | however |
| 18:29.58 | Katty | do you have a orange shrimp recipe? |
| 18:30.08 | bmoraca_work | no, i forgot to bring it in :( |
| 18:30.18 | bmoraca_work | er, peach shrimp one, anyway |
| 18:30.18 | Katty | :<<<<< |
| 18:30.21 | Qwell | pokes Katty |
| 18:30.26 | Katty | hi Qwell |
| 18:30.38 | Katty | you can roulette with me since p3nguin_'s flash is borken |
| 18:30.58 | Qwell | if this build finishes before I leave |
| 18:31.08 | Katty | (= |
| 18:34.31 | *** join/#asterisk adnc (~numer@unaffiliated/adnc) |
| 18:34.37 | Katty | hi |
| 18:35.44 | ecrane | Filter yer wildcards http://seclists.org/fulldisclosure/2010/Feb/349 |
| 18:36.05 | ecrane | (for those who do not read the mailinglist...) |
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| 18:51.17 | *** join/#asterisk puzzled (~foobar@puzzled.xs4all.nl) |
| 18:51.42 | benngard | is a pretty happy son of a bi*, all avaya/fax/audiocodes have been moved to asterisk/fax/spa2102 users, and TH E SHIT IS WORKING! |
| 18:52.11 | *** join/#asterisk nsgn (~nsgn@rrcs-24-153-206-251.sw.biz.rr.com) |
| 18:52.23 | p3nguin_ | benngard is a son of a bi! |
| 18:52.43 | nsgn | network with 12 polycom phones and a * server. two of the 12 phones don't beep/light up when they have a new voice message. i can't find out WHY |
| 18:53.08 | Naikrovek | nsgn: make sure they have the same configs |
| 18:53.18 | ManxPower-work | nsgn: the mailbox= setting for those peers in sip.conf |
| 18:53.25 | nsgn | they do. configs are being sent centrally via HTTP. phones download them every reboot |
| 18:53.28 | benngard | p3nguin_: i dont care what u think i am, i like the result ;) |
| 18:53.37 | ManxPower-work | this is an asterisk thing, not a phone thing. |
| 18:53.38 | nsgn | ManxPower-work, i'll verify that in the individual configs |
| 18:53.54 | p3nguin_ | benngard: I quoted what you said. I don't have an opinion of you, personally. |
| 18:53.59 | nsgn | ManxPower-work, oh on the * sip file, not the phone's? |
| 18:54.15 | ManxPower-work | It's hard enough to make polycom phones not beep, I doubt you messed up the phone config. |
| 18:54.17 | bmoraca_work | ManxPower-work: could be a phone thing if it's not configured to subscribe...though it's likely an asterisk peer configuration |
| 18:54.23 | ManxPower-work | nsgn: yes, sip.conf for Asterisk |
| 18:54.31 | benngard | p3nguin_: .) |
| 18:54.34 | ManxPower-work | bmoraca_work: the phone does not have to subscribe to get MWI |
| 18:54.41 | benngard | p3nguin_: :) |
| 18:55.56 | bmoraca_work | really? if you zero out the msg.mwi.1.subscribe config param, the phone still gets MWI? i've never noticed that...though i've also never had a mismatch between my phone and asterisk |
| 18:56.39 | ManxPower-work | bmoraca_work: If you want the messages button to work, you need the various msg.mwi settings correct, |
| 18:57.38 | bmoraca_work | well, i knew you needed msg.mwi.1.callBack to tell it what to dial when you hit the "messages" button...but i thought that the MWI was tied to the msg.mwi.1.subscribe parameter |
| 18:57.46 | bmoraca_work | learn something new every day |
| 18:58.37 | nsgn | ManxPower-work, found some of my devices have just the extension number, non working ones seem to have extension number followed by @ symbol |
| 18:58.47 | nsgn | and other working ones have extension number, @ symbol, then the word device |
| 18:59.30 | bmoraca_work | nsgn: the stuff after the @ is the voicemail context. if you only have one context, it's pretty much ignored |
| 18:59.40 | vader-- | Do you guys know of any Open source asset management tools for keeping track of hardware? |
| 18:59.48 | nsgn | but the ones with an @ but no context don't function |
| 18:59.56 | nsgn | just threw device on the end of them and it appears it instantly started working |
| 18:59.56 | nsgn | wild |
| 19:00.28 | bmoraca_work | nsgn: why would you put a @ and nothing after it? |
| 19:00.42 | nsgn | i don't know. it was like this. another one of my techs screwing up? |
| 19:00.51 | bmoraca_work | or ignorance |
| 19:01.11 | nsgn | it's why i didn't check that off the bat; i just didn't figure it was that way |
| 19:01.21 | nsgn | i'd always just put in extension number and nothing else. lines i have set up that way function |
| 19:01.31 | nsgn | though for some reason the majority here have @device after the extension |
| 19:01.38 | nsgn | which apparently functions too |
| 19:01.38 | nsgn | *shrug* |
| 19:01.42 | *** join/#asterisk adnc (~numer@unaffiliated/adnc) |
| 19:02.16 | *** join/#asterisk jmacz (~jmacz@190.144.75.22) |
| 19:07.28 | leifmadsen | doh! Chainsaw just left |
| 19:07.37 | Katty | bmoraca_work: so where did that orange or peach shrimp recipe come from? |
| 19:07.43 | Katty | bmoraca_work: any chance i could find it somewhere on google? |
| 19:10.46 | *** join/#asterisk tnsampaio (~tiago@189.111.7.210) |
| 19:14.04 | p3nguin_ | leifmadsen: Yes, but there are other people here that you can socialize with. |
| 19:14.28 | p3nguin_ | Anyone here familiar with SME Server? |
| 19:16.04 | *** join/#asterisk atis_work (~atis_work@193.238.212.171) |
| 19:16.43 | p3nguin_ | ~modem |
| 19:16.44 | infobot | modem is probably (Modulator/Demodulator) A device to turn digital signals to analog ones and back again, so they can be transmitted and translated back to digital at another modem without loss. Used for communication through means of audio, telephone, CB, etc. Random disconnects? S10=255 sure to do the trick! |
| 19:17.41 | p3nguin_ | What's the reason Asterisk can't use a PCI phone modem as an FXO port for connecting to a POTS wall jack? |
| 19:17.46 | Katty | i'm not |
| 19:18.30 | ManxPower-work | p3nguin_: 1) the voice versions are too high latency for real time voice, but they could work fine, in theory for an IVR. |
| 19:19.02 | Katty | is that part of the reason behind why faxing is not very good |
| 19:19.05 | Katty | the latency issue |
| 19:19.08 | Katty | or is that something else |
| 19:19.25 | ManxPower-work | 2) nobody with the required skills has managed to get the secret driver info from vendors that has also been interested in writing a driver for a hobby device like a "voice" modem. |
| 19:20.06 | ManxPower-work | Mark spencer wrote a driver for a specific intel and motorola chipsets. Those chipsets were discountinued 5 years ago |
| 19:20.25 | ManxPower-work | That is what we call an X100P/X101P |
| 19:20.25 | Katty | :< |
| 19:20.43 | *** join/#asterisk tzafrir_laptop (~tzafrir@bzq-218-155-146.cablep.bezeqint.net) |
| 19:20.47 | Katty | hi tzafrir_laptop |
| 19:32.01 | *** part/#asterisk fainsys (~fainsys@c-98-242-73-30.hsd1.ga.comcast.net) |
| 19:33.43 | *** join/#asterisk Alagar (~Administr@122.164.33.35) |
| 19:34.33 | *** join/#asterisk Xetrov` (~xetrov@unaffiliated/xetrov/x-827361) |
| 19:36.25 | *** join/#asterisk voipmonk (~shido6@CPE002191f85581-CM001692568382.cpe.net.cable.rogers.com) |
| 19:37.52 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 19:43.28 | bmoraca_work | Katty: the peach shrimp recipe i have is from a slow cooker recipe book...i don't recall the name, but I don't think it'd be available online...i'll do a quick search |
| 19:48.27 | bmoraca_work | has anyone else here used VoipInnovations or M2Tel or GlobalpopsVoip? i'm wondering when they're going to start charging me for the service they've been giving me for a week... |
| 19:49.50 | *** join/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 19:50.34 | Katty | yawns a bit |
| 19:59.30 | *** join/#asterisk TheDavidFactor (~chatzilla@96.57.203.22) |
| 20:00.50 | TheDavidFactor | is there any documentation on the options for the configure script that you run before make when you install asterisk from source? |
| 20:03.01 | *** join/#asterisk tgunr (~tgunr@2001:0:53aa:64c:0:fbff:bd06:59f4) |
| 20:03.13 | *** join/#asterisk Whtsup (~sssi@119.63.136.50) |
| 20:03.26 | Whtsup | hello |
| 20:03.27 | Whtsup | ? |
| 20:03.38 | Katty | http://boldt.us/8524-2/deer_cat_window.jpg |
| 20:03.53 | *** join/#asterisk Cain` (~Geek@unaffiliated/cain) |
| 20:04.04 | Whtsup | is there any way how can i check voice quality between 2 asterisk server |
| 20:05.52 | idespinner | Whtsup, an wireshark capture of the SIP and RTP packets would go a long way... |
| 20:05.59 | Whtsup | hmmm |
| 20:06.50 | *** join/#asterisk Ad-Hoc (~nimbus@62.1.108.182.dsl.dyn.forthnet.gr) |
| 20:08.14 | tgunr | I am confused about localnet and externip in my sip.conf , I have a LAN 10.0.0.0/22 and on my WAN side I have 2 devices I also want to use 66.249.166.8/29. No matter what settings I use for localnet I continue to get "not a local domain" from my softphones i am testing with. Could use a clue or two :) |
| 20:09.19 | tgunr | if my asterisk server is sitting on 66.249.166.11 is it legal to have a localnet = 66.249.166.8/29? |
| 20:09.48 | [TK]D-Fender | tgunr: pastebin your failed call with SIP debug and your sip.conf |
| 20:09.49 | Corydon76-dig | tgunr: the purpose of those is for when your asterisk server sits BEHIND the firewall |
| 20:10.22 | tgunr | hmm, ok, the server is not behind firewall, will paste |
| 20:10.29 | Corydon76-dig | If your Asterisk server has links on both networks, then there's no need to use localnet/externip |
| 20:10.43 | tgunr | so just remove the settings then? |
| 20:11.02 | Corydon76-dig | Correct. Asterisk uses the routing table for that |
| 20:11.12 | tgunr | cc, trying |
| 20:11.27 | Corydon76-dig | Make sure you restart, to ensure the channel driver doesn't cache the information |
| 20:11.41 | idespinner | Corydon76-dig, how does asterisk set an affinity for which interface to use(internal vs public)? |
| 20:11.49 | Katty | http://dobermanpuppiesblog.com/files/dobe_litter.jpg |
| 20:11.56 | idespinner | for its outgoing sip packets? |
| 20:12.11 | Corydon76-dig | idespinner: it depends upon the registration of the phones (i.e. the destination IP) |
| 20:12.12 | rubberneck | tgunr: And close some of those ports man, you got that thing hangin out there. |
| 20:12.39 | Corydon76-dig | idespinner: destination IP determines source IP, according to the routing table |
| 20:12.40 | tgunr | yeh, i know, just testing stuff right now |
| 20:13.08 | idespinner | yes but asterisk still must put the source ip/contact in the sip header... doesnt it? |
| 20:13.09 | rubberneck | zeroshell netservices. |
| 20:13.32 | Corydon76-dig | idespinner: I don't think you understand. Asterisk is directly querying the routing table |
| 20:13.38 | idespinner | i ask only because i dont understand... |
| 20:14.00 | idespinner | and based on the routing table, the sip packet is formed? |
| 20:14.22 | ChannelZ | Katty: those are some big feet! |
| 20:14.22 | bmoraca_work | idespinner: it's almost always simpler just to let asterisk not reside on both networks |
| 20:14.25 | Corydon76-dig | It's not just relying on the kernel to set its IP layer correctly, but actually querying the kernel for routing information, so it knows what IP to put inside the protocol on an outgoing packet |
| 20:14.48 | idespinner | Corydon76-dig, understood, thanks. |
| 20:15.11 | idespinner | bmoraca_work, I kind of agree'd, I always set up a single NIC asterisk when I install... |
| 20:15.19 | Katty | ChannelZ: aren't they :> |
| 20:15.33 | Corydon76-dig | It was one of my first contributions to Asterisk, way back in the 0.20 days |
| 20:15.39 | *** join/#asterisk seanbright (~sean@asterisk/contributor-and-bug-marshal/seanbright) |
| 20:16.19 | *** join/#asterisk fainsys (~fainsys@c-98-242-73-30.hsd1.ga.comcast.net) |
| 20:16.45 | idespinner | are you digium? |
| 20:16.56 | tgunr | still getting the error? :( |
| 20:16.57 | tgunr | http://paste.lisp.org/display/95274#1 |
| 20:17.46 | Corydon76-dig | idespinner: at the time, I wasn't, but I am now |
| 20:18.16 | Katty | hi |
| 20:18.23 | Katty | seanbright: |
| 20:18.41 | p3nguin_ | Poop, there's no iperf package in CentOS 4. |
| 20:19.07 | Katty | show me the way to go home home home |
| 20:19.10 | Katty | i'm tired and i wanna go to bed |
| 20:19.24 | tgunr | had a little one |
| 20:19.31 | idespinner | thats fun, my company has been a reseller for about 2.5 years... |
| 20:20.07 | visz | oh show me the way to the next little girl |
| 20:20.14 | *** join/#asterisk cesar_CR (~cesar@201.192.86.30) |
| 20:20.15 | visz | don't ask my why |
| 20:20.18 | visz | oops |
| 20:20.22 | visz | it's friday |
| 20:20.24 | Katty | mhmm |
| 20:20.40 | tgunr | duh, the copy for sip.conf didn't take, on first pass, it is now in the paste http://paste.lisp.org/display/95274#2 |
| 20:20.55 | visz | oh crap |
| 20:20.57 | Katty | i need to kill an hour |
| 20:21.02 | Katty | anyone have some suggestions |
| 20:21.16 | visz | almost forgot i had to do some perl to test a shdsl line |
| 20:21.24 | tgunr | yeh, you can help me! :) |
| 20:21.41 | Katty | i said kill an hour |
| 20:21.45 | Katty | not make it drag for eternity |
| 20:21.47 | tgunr | LOl |
| 20:24.13 | *** join/#asterisk evil_gordita (~evilgordi@ip70-188-50-186.rn.hr.cox.net) |
| 20:25.28 | *** join/#asterisk ruben23 (~AGENT@122.55.48.243) |
| 20:26.36 | ManxPower-work | It's Jim Morrison |
| 20:26.50 | *** join/#asterisk evil_gordita (~evilgordi@ip70-188-50-186.rn.hr.cox.net) |
| 20:27.05 | bmoraca_work | come on baby light my fire? |
| 20:27.33 | Katty | set the world on fiiiiiireeeeeee |
| 20:29.10 | *** join/#asterisk tgunr (~tgunr@2001:0:53aa:64c:0:fbff:bd06:59f4) |
| 20:30.38 | Corydon76-dig | Katty: just weighed Johnny. He's 15.5 lbs. |
| 20:30.56 | Katty | :>>>> |
| 20:31.05 | Katty | that's one ginormous bunbun |
| 20:31.23 | Corydon76-dig | He's just under a year old now |
| 20:33.24 | *** join/#asterisk atis_work (~atis_work@193.238.212.171) |
| 20:34.21 | Katty | Corydon76-dig: he's gonna be all grown up soon :> |
| 20:36.57 | *** join/#asterisk hluesea (~hulusikah@88.247.127.66) |
| 20:37.40 | *** join/#asterisk imcdona (imcdona@173.160.189.69) |
| 20:38.24 | *** join/#asterisk sleazye (~ernesto@64-71-24-110.static.wiline.com) |
| 20:39.25 | Corydon76-dig | Katty: if he gets any bigger, I might need a bigger house |
| 20:39.51 | sleazye | Bug reporting question: A bug was reported, closed with a patch but the patch doesn't work - how do I re-open the bug? |
| 20:39.51 | sleazye | https://issues.asterisk.org/view.php?id=16470 |
| 20:40.32 | Corydon76-dig | sleazye: yes, the patch works |
| 20:40.41 | Corydon76-dig | sleazye: are you talking about 1.4? |
| 20:40.48 | sleazye | yeah, 1.4 |
| 20:40.58 | sleazye | restart when convenient does not restart, it stops |
| 20:41.00 | Corydon76-dig | sleazye: Use SVN to update to the latest branch |
| 20:41.35 | sleazye | i just pulled the loader.c file into 1.4.29.1 - that work? |
| 20:41.42 | Corydon76-dig | I know the patch doesn't apply to 1.4 |
| 20:41.54 | Corydon76-dig | Which one, the one from SVN? |
| 20:42.03 | Corydon76-dig | SVN 1.4, rather? |
| 20:42.52 | sleazye | Corydon76-dig: http://svn.digium.com/view/asterisk?view=rev&revision=248012 |
| 20:43.16 | Corydon76-dig | Yes, that one |
| 20:43.21 | Corydon76-dig | should work fine on 1.4 |
| 20:43.41 | *** join/#asterisk D-Boy (~Geek@unaffiliated/cain) |
| 20:43.42 | sleazye | Corydon76-dig: Right..it *should* but I'm reporting that it's not... |
| 20:43.48 | tgunr | If my asterisk server is not behind a firewall, and has 2 interfaces LAN and WAN, shouldn't sip show domains show both interfaces? |
| 20:44.38 | Corydon76-dig | sleazye: I'll take a look at it |
| 20:44.55 | sleazye | tgunr: what do you have for bindaddr in sip.conf? |
| 20:45.04 | tgunr | 0.0.0.0 |
| 20:45.10 | sleazye | Corydon76-dig: thx |
| 20:47.48 | Corydon76-dig | sleazye: I just tried it, and it works fine |
| 20:48.17 | Katty | blah 45mins to go |
| 20:48.27 | sleazye | Corydon76-dig: what command did you try? |
| 20:48.41 | Corydon76-dig | "restart when convenient" |
| 20:49.27 | Corydon76-dig | sleazye: you running any modules not distributed with Asterisk? |
| 20:49.34 | sleazye | Corydon76-dig: psssh..i just tried it again and it worked fine - i haven't re-compiled or anything |
| 20:50.36 | sleazye | Corydon76-dig: No, pretty standard stuff. The "restart when conveninet" used to do a half restart...asterisk would continue to run but some modules were unloaded (like iax) |
| 20:51.16 | bmoraca_work | you don't NEED iax :P |
| 20:51.47 | *** join/#asterisk brezular (~brezular@adsl-dyn-24.95-102-194.t-com.sk) |
| 20:51.51 | sleazye | bmoraca_work: true. problem solved. :) |
| 20:52.10 | p3nguin_ | restart when convenient probably unloads anything not being used and then restarts when there are no more calls. |
| 20:52.36 | sleazye | p3nguin_: it was when no calls were going on |
| 20:52.45 | bmoraca_work | i just tell the people "your calls are going to drop!" and then pull the plug |
| 20:52.51 | p3nguin_ | I like restart gracefully, since it won't let anymore calls go through, then restarts when all active calls finally end. |
| 20:53.24 | p3nguin_ | when convenient could be hours of even days, depending on usages. |
| 20:53.34 | *** join/#asterisk Fulg0re (~jago@216.17.67.126) |
| 20:53.34 | p3nguin_ | hours or even days, I mean. |
| 20:54.39 | Fulg0re | Where's a good place to purchase DID's? |
| 20:55.20 | p3nguin_ | Flowroute or VoIP.ms or CallCentric |
| 20:55.24 | bmoraca_work | your local telco.. |
| 20:55.28 | p3nguin_ | Or that! |
| 20:55.50 | bmoraca_work | whoever provides you with termination right now will likely give you the best deal |
| 20:55.51 | Fulg0re | local telco...would they have good call rates? |
| 20:56.09 | bmoraca_work | i could hardly begin to speculate. might consider asking. |
| 20:56.55 | bmoraca_work | the problem i've noticed with most ITSPs is that they charge you for origination (inbound calls). you generally won't get charged that going with a traditional CLEC or ILEC |
| 20:57.07 | Fulg0re | termination....that would be the people who provide the sip trunk? |
| 20:57.12 | sleazye | Corydon76-dig: let a few calls go by and restarted when convenient successfully - what should i capture when i'm doing this and next time it breaks? |
| 20:57.39 | Corydon76-dig | sleazye: exactly console output when it quits instead of restarts |
| 20:57.39 | bmoraca_work | it's a secret, but telcos pay precisely $0 to receive phone calls from other telcos. telcos pay only when they need to place a call to another telco |
| 20:58.02 | bmoraca_work | Fulg0re: could be. i don't know how your system is set up. |
| 20:58.18 | sleazye | Corydon76-dig: k - i'll keep testing but gotta run for a bit..u going to be on here for a bit? |
| 20:58.18 | Fulg0re | its not...setting one up for the first time so im trying to find out what i need :) |
| 20:59.18 | bmoraca_work | Fulg0re: well, if you intend on placing calls out of your system or receiving calls into your system, you'll need some sort of PSTN connectivity. depending on your call patters and capacity requirements, there are lots of options, and one is bound to be right for you. |
| 21:00.13 | Fulg0re | yes I would be placing both inbound and outbound calls...so what type of pstn connectivity should i look for? |
| 21:00.33 | Fulg0re | i'd like to have it all routed over the internet and hopefully not need additional hardware |
| 21:00.42 | Fulg0re | except the phone and asterisk server |
| 21:00.54 | bmoraca_work | Fulg0re: then you'll want an ITSP probably. |
| 21:00.57 | [TK]D-Fender | Fulg0re: what phone? |
| 21:00.59 | bmoraca_work | ~itsp |
| 21:01.00 | infobot | [~itsp] An ITSP is an Internet telephony Service Provider or "VoIP Telephone company". The allow you to either PLACE calls to the PSTN (called Termination), RECEIVE calls from the PSTN (called Origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs |
| 21:02.15 | Corydon76-dig | sleazye: No, I've gotta take a rabbit to the vet |
| 21:02.48 | bmoraca_work | uhg...i have a customer who wants to try and set up a multitenant phone system using Fonality and he wants to try and do it all himself...i feel huge fail coming on here pretty quickly... |
| 21:02.56 | Fulg0re | purcahsed a used plycomm ip soundpint 500 to test out the setup |
| 21:03.27 | Fulg0re | ~itsplist-us |
| 21:03.28 | infobot | somebody said itsplist-us was Here are some popular ITSPs (USA) starting with the more respected ones : http://www.teliax.com , http://www.voicepulse.com/connect/ , http://www.broadvoice.com, http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net, or http://flowroute.com , http://voip.ms |
| 21:05.49 | Fulg0re | if i get phone number from someone like teliax than i wouldnt need anything else correct? |
| 21:09.50 | [TK]D-Fender | Fulg0re: Nope |
| 21:10.51 | Fulg0re | if i get one phone number, and have 4 phones total. the other 3 would have extensions? do i set that up or the DID provider? |
| 21:11.06 | *** join/#asterisk Sargun (~Sargun@atarack/Staff/Sargun) |
| 21:11.41 | hardwire | From: <sip:+1907xxxxxxx@xx.xxx.xx.xx:5060;isup-oli=70>;tag=gK077440b4 |
| 21:11.43 | p3nguin_ | fulg0re: If you have one DID, you would let Asterisk route calls to phones based on extensions dialed. |
| 21:11.51 | hardwire | any idea how to check for isup-oli=70 in the dialplan? |
| 21:11.55 | *** join/#asterisk hluesea (~hulusikah@88.247.127.66) |
| 21:12.03 | hardwire | I guess I'd pull up the sip FROM header and do a string check. |
| 21:12.27 | hardwire | Does From information get turned into kwargs at any point? |
| 21:12.35 | Fulg0re | so if someone called in i could set up a recording asking them to dial the other extensions? |
| 21:13.01 | p3nguin_ | fulg0re: Yes. That would be known as a simple IVR. |
| 21:14.14 | Fulg0re | but to have more than one phone dial outbound would i need more DID's or what? |
| 21:15.06 | p3nguin_ | Nope. DID is Direct Inward Dialing ... aka a phone number. |
| 21:15.30 | p3nguin_ | You don't even have to any DID at all to make outbound calls. |
| 21:15.45 | Fulg0re | than to get more than one inbound call i would have to get multiple DIDs right? |
| 21:15.46 | p3nguin_ | You just need to have some type of call termination service. |
| 21:16.05 | p3nguin_ | No, you can have multiple calls on a single inbound phone number. |
| 21:16.26 | p3nguin_ | The channels available on the DID determine how many inbound calls you can have at a time. |
| 21:16.58 | Kobaz | Fulg0re: a DID is like an ip address |
| 21:17.08 | p3nguin_ | You can get free DIDs from IPkall, IPcomms, and sipgate. I think those all give you two channels, which means you can have two calls at the same time on the phone number. |
| 21:17.28 | Kobaz | Fulg0re: with one ip address you can have a hundred computers or more, all accessing different websites for example |
| 21:17.37 | Fulg0re | ahhh...so i have to make sure theres more than one channel to handle multiple inbound calls? |
| 21:17.47 | Fulg0re | ahhh...yea the ip analogy makes ense |
| 21:17.48 | Fulg0re | sense |
| 21:17.48 | p3nguin_ | correct |
| 21:18.17 | Kobaz | Fulg0re: a did is used only for one thing, which is incomming call handling |
| 21:18.41 | Kobaz | and you route your calls accordingly... and make sure you have enough lines/channels/resources available to handle the number of calls you expect |
| 21:18.44 | p3nguin_ | IPkall gives me two inbound channels, so you can call me and kobaz can call me at the same time. If hardwire tries to call me, he'll get a circuits busy message until one of you two disconnect and free up a channel. |
| 21:19.27 | Fulg0re | do you or kobaz talk a lot? :) |
| 21:19.31 | Fulg0re | kidding |
| 21:19.35 | Kobaz | nope |
| 21:19.39 | Kobaz | i just sit here quietly |
| 21:19.42 | Fulg0re | hehehe |
| 21:19.59 | Fulg0re | if i get one or 2 channels...than what does the outbound caller id show up as? |
| 21:20.03 | p3nguin_ | On the other hand, my toll-free DID from VoIP.ms gives me unlimited channels. So I can have as many incoming calls as my Asterisk system and internet conneciton can handle. |
| 21:20.06 | Fulg0re | just the one did? |
| 21:20.09 | Kobaz | Fulg0re: it depends |
| 21:20.22 | Kobaz | Fulg0re: if your provider allows it... you can set the callerid to whatever you want |
| 21:20.31 | p3nguin_ | You can often set your CID number to whatever you want. |
| 21:20.37 | Kobaz | Fulg0re: if you provider has a fixed callerid, it'll be whatever they set it to, per each account |
| 21:20.41 | p3nguin_ | Normal people will set it to their DID number. |
| 21:21.03 | p3nguin_ | That way if you call me, I see your DID phone number where I can call you back. |
| 21:21.13 | p3nguin_ | It wouldn't make sense to set it to a phone number you don't have. |
| 21:21.13 | Kobaz | it's funny when you forget to set the callerid to a did, and you get a call on your cell phone from '5000' |
| 21:21.45 | Fulg0re | voip.ms looks like it has better rates then the other providers |
| 21:21.58 | Kobaz | Fulg0re: vitelity is even better, if you sign up for wholesale |
| 21:22.10 | Kobaz | it's cheaper than voip.ms... and voip.ms uses vitelity as one of their providers |
| 21:22.16 | tgunr | If I have a LAN setup as 10.0.0.0/22 and my WAN as 66.249.166.8/29 and sip.conf domain=10.0.0.1 what should a 'sip show domains' look like? I get this: http://pastie.textmate.org/833382 |
| 21:22.18 | *** join/#asterisk Naikrovek (~jjohnson@unaffiliated/naikrovek) |
| 21:22.40 | Kobaz | Naikrovek: my new piston came |
| 21:22.47 | Naikrovek | ooh |
| 21:22.52 | Naikrovek | fit it in there yet? |
| 21:22.55 | Kobaz | it's much less gouged than the old one |
| 21:23.02 | Kobaz | not yet.. still need to write more code |
| 21:23.03 | Naikrovek | hehe |
| 21:23.10 | p3nguin_ | fulg0re: Flowroute is slightly lower, if you care about a few pennies. |
| 21:24.13 | p3nguin_ | fulg0re: US to US calling with VoIP.ms is 1.05 cents per minute, where Flowroute is 0.98 cents per minute. |
| 21:24.28 | Fulg0re | vitelity or flowroute....will they offer multiple channels? |
| 21:24.34 | Kobaz | sure |
| 21:24.42 | Fulg0re | quality is good for both of those providers/ |
| 21:24.44 | Fulg0re | ? |
| 21:25.02 | Kobaz | quality really comes down to your own internet connection, and the path it takes to get to your provider |
| 21:25.07 | hardwire | p3nguin_: but I hate phones! |
| 21:25.17 | p3nguin_ | I know, I do, too. |
| 21:25.22 | hardwire | heh |
| 21:25.24 | Kobaz | computers suck |
| 21:25.27 | hardwire | happy friday peoples. |
| 21:25.36 | hardwire | apparently ${ANI2} is my friend |
| 21:27.46 | Fulg0re | vitelity shows $20 for 2-5 channels...is that $20 per channel than? so 4 channles would be $80 plus the costs of the minutes used? |
| 21:27.53 | Fulg0re | and the cost of the did |
| 21:28.59 | Kobaz | that's if you have unlimited incoming |
| 21:29.10 | Kobaz | if you just pay per minute, it's unlimited channels |
| 21:29.14 | Kobaz | and you just pay for the minutes |
| 21:29.47 | Kobaz | and the 'unlimited' incoming, i think is like 1500 minutes or something |
| 21:29.52 | Fulg0re | kobaz, what does it mean hwn you say if you pay per minute its unlimited channels? |
| 21:30.20 | p3nguin_ | They have pay-per-minute services as well as "unlimited" minute plans. |
| 21:30.44 | bmoraca_work | i think it's criminal that these people charge for inbound calls. |
| 21:31.07 | p3nguin_ | If you don't get wholesales rates, you're going to spend a lot more signing up with Vitelity directly. |
| 21:31.10 | idespinner | does vitelity allow you to set your outbound callerid to anything? i know alot of sip trunk providers dont |
| 21:31.25 | Kobaz | with per-minute, you can get as many calls as you want, and you pay per minute |
| 21:31.37 | Fulg0re | but for wholesale they require $299 monthly minimum |
| 21:31.50 | p3nguin_ | see flowroute.com or voip.ms |
| 21:31.54 | Kobaz | Fulg0re: i signed up for wholesale and got a dedicated server |
| 21:32.06 | Kobaz | Fulg0re: so the $199 a month was all that was needed |
| 21:32.41 | p3nguin_ | If you want unlimited outbound calling rather than pay-per-minute, CallCentric is the only one I know of that offers it. |
| 21:35.32 | Fulg0re | kobaz you got the server at vitelity/ |
| 21:35.33 | Fulg0re | ? |
| 21:36.30 | Kobaz | yeah |
| 21:36.47 | Kobaz | 25 feet from their pstn lines |
| 21:38.15 | Fulg0re | how'd you get around the $299 monthly minimum? |
| 21:38.22 | Kobaz | i dunno |
| 21:38.31 | Kobaz | maybe they didn't have that when i signed up |
| 21:38.46 | Kobaz | call them up and ask for a quote... tell em what you're doing, and what sort of volume you expect |
| 21:39.08 | *** join/#asterisk hluesea (~hulusikah@88.247.127.66) |
| 21:39.50 | *** join/#asterisk mpe (~mpe@0x4dd624b2.adsl.cybercity.dk) |
| 21:45.38 | Fulg0re | cool, im getting some ideas :) |
| 21:45.51 | Fulg0re | where is a good place to get hardware phones? |
| 21:45.56 | Fulg0re | i need 4 of em |
| 21:46.00 | Fulg0re | for now :) |
| 21:46.10 | p3nguin_ | Depends on what kind of phones you want and how much you want to spend. |
| 21:46.26 | p3nguin_ | Depends on what features you need in the phone, as well. |
| 21:46.49 | Fulg0re | for a small office, nothing to fancy, put users on hold, transfer calls, answer more than one call at a time |
| 21:47.12 | Fulg0re | forwarding buttong, dnd button maybe |
| 21:47.24 | p3nguin_ | Do you want to be able to put one call on hold and then dial another number? |
| 21:47.55 | Fulg0re | yep |
| 21:48.11 | *** join/#asterisk Defraz (~Defraz@c72co-edge-router.fuzecore.com) |
| 21:48.13 | tgunr | why is setting up sip.conf so freaking hard! |
| 21:48.22 | p3nguin_ | It isn't. |
| 21:48.49 | tgunr | been at this for hours now, can't get a simple softphone to register |
| 21:50.00 | *** join/#asterisk skrite (~skrite@69.55.17.185) |
| 21:50.06 | Kobaz | Fulg0re: if you have 4 phones, you don't need dedicated server hosting and wholesale accounts |
| 21:50.09 | Pan3D | all together now... |
| 21:50.10 | skrite | hello all |
| 21:50.12 | Pan3D | You're doing it wrong. |
| 21:50.32 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 21:50.32 | p3nguin_ | tgunr: http://pastebin.com/d3ad5026b |
| 21:50.45 | Kobaz | tgunr: just make sure ann the usernames, secrets, and ips match, and you're good to go |
| 21:50.46 | Pan3D | tgunr: if it's taking you hours, there's something wrong with either your configuration or your understanding. What is it that you're trying to do specifically? |
| 21:51.27 | Fulg0re | kobaz, yea i think that will be over kill for now |
| 21:51.37 | tgunr | regardless of what i set in sip.conf, I get "Not a local domain" |
| 21:52.20 | tgunr | http://paste.lisp.org/display/95274#2 |
| 21:52.27 | p3nguin_ | Oops, forgot type=peer |
| 21:52.53 | p3nguin_ | Probably ought to throw in a host=dynamic too. |
| 21:53.06 | tgunr | my host is static |
| 21:53.30 | p3nguin_ | http://pastebin.com/d5c8baaa6 |
| 21:57.15 | *** join/#asterisk imcdona (imcdona@173.160.189.69) |
| 21:58.32 | tgunr | p3nguin_: same thing, not a local domain |
| 21:58.51 | p3nguin_ | Then the problem is not in the peer definition. |
| 21:59.04 | p3nguin_ | It's either in the general section if sip.conf or on the phone. |
| 21:59.12 | tgunr | no, i think it is in sip.conf |
| 21:59.21 | tgunr | somewhere |
| 21:59.49 | p3nguin_ | domain = 66.249.166.11 is probably causing an issue. |
| 22:00.09 | tgunr | I have tried many variations, all same result |
| 22:00.44 | p3nguin_ | Maybe my config could help you. Give me 5 minutes to get it. |
| 22:02.04 | skrite | hey all, from the CLI in asterisk, i am trying to issue 'zap show status' and it tells me the command is not available. |
| 22:02.10 | skrite | no such command |
| 22:03.17 | Pan3D | dahdi show status |
| 22:04.38 | *** join/#asterisk QubeZ (~qube@64.128.254.34) |
| 22:04.58 | skrite | i dont have that one either. now, it brings me to a point, i was trying to use zaptel instead of dahdi |
| 22:05.05 | *** join/#asterisk Defraz (~Defraz@c72co-edge-router.fuzecore.com) |
| 22:05.19 | QubeZ | hello all. I am running Asterisk 1.4 and I have a group of people complaining that their entire group (10-15 ppl) hear silence all at once while on a call then their customer is able to hear them again. Any ideas on how to start attacking this issue? |
| 22:05.52 | ChannelZ | Voip? |
| 22:05.56 | ChannelZ | Maybe your connection is stalling |
| 22:06.03 | skrite | i think i may have screwed up my installation. I have to use asterisk 1.4 because i am using adhearsion ( a ruby asterisk system) but it is incompatible with version 1.6 |
| 22:10.24 | QubeZ | no one else downstairs is affected |
| 22:10.25 | Pan3D | skrite: check to be sure the modules with those commands are loaded |
| 22:10.32 | skrite | Pan3D, i am mixed between zap and dahdi |
| 22:10.32 | skrite | trying to fix now. |
| 22:10.32 | *** join/#asterisk moy (~moy@74.12.129.100) |
| 22:10.32 | Pan3D | I am mixed between Mexican and Italian. |
| 22:10.32 | Pan3D | errr... never mind |
| 22:10.33 | Pan3D | ;) |
| 22:10.33 | Pan3D | good luck |
| 22:11.16 | atis_work | umm, does asterisk have some kind of filesystem read cache for sounds played? |
| 22:11.42 | p3nguin_ | tgunr: http://pastebin.com/d7e493007 |
| 22:11.49 | tgunr | looking |
| 22:11.51 | skrite | i am getting no such command 'show status' |
| 22:11.52 | atis_work | i'm thinking of what's the best way how to distribute files through several machines and glusterfs |
| 22:11.53 | p3nguin_ | I meant 11 minutes. :) |
| 22:12.16 | tgunr | could this have something to do with the problem? |
| 22:12.17 | tgunr | [Feb 19 14:08:21] NOTICE[17826]: chan_sip.c:17859 reload_config: Can't add wildcard IP address to domain list, please add IP address to domain manually. |
| 22:13.42 | p3nguin_ | tgunr: Just take out the domain line completely. |
| 22:14.09 | tgunr | i have done that before, same error |
| 22:14.52 | p3nguin_ | I don't know what else to tell you. I took that off my personal system that I use every day, and 123 is used by a soft phone. |
| 22:15.00 | *** join/#asterisk puzzled (~foobar@puzzled.xs4all.nl) |
| 22:15.09 | minotaur01 | quick question: is there a way to shop the MOH from repeating when you put some one on park? By repeating i mean the music starts over after the user hangs up after the parked location is played... |
| 22:15.45 | p3nguin_ | Use a stream instead of a file. |
| 22:16.21 | minotaur01 | well we have specially recorded MOH |
| 22:16.41 | voipmonk | so stream it |
| 22:17.14 | minotaur01 | k ill tray that |
| 22:17.22 | minotaur01 | try* |
| 22:19.34 | skrite | ok, weirder. asterisk starts with asterisk -vvvv and i can go in a different terminal and asterisk -r to get to the CLI, but there i am not able to see the status of any channels |
| 22:20.18 | skrite | dahdi show channels and zap show channels both reply with that there is no such command |
| 22:20.32 | p3nguin_ | skrite: Maybe there is no such command. |
| 22:20.43 | tgunr | p3nguin_: thanks for your effort, will keep plugging away at it |
| 22:20.51 | skrite | p3nguin_, he he |
| 22:21.14 | skrite | i am looking at the commands from the voip-info.org wiki |
| 22:21.47 | *** join/#asterisk lordsporkton (~Lawrence@ip70-181-108-246.oc.oc.cox.net) |
| 22:22.15 | skrite | ok, just show channels says i have no channels |
| 22:22.52 | lordsporkton | im looking for info on how to get a clec license in ca, anyone have any pointers? |
| 22:23.08 | p3nguin_ | skrite: Many things on that site are completely outdates and sometimes just plain wrong. |
| 22:23.21 | p3nguin_ | outdated |
| 22:23.42 | skrite | ok |
| 22:24.19 | minotaur01 | voipmonk: do you have recommanded program i can use to stream music on the asterisk box? |
| 22:24.45 | voipmonk | it would help you to google that question |
| 22:24.52 | voipmonk | stream file asterisk <-------- google that |
| 22:24.54 | minotaur01 | i did |
| 22:25.01 | voipmonk | its ok - it doesnt hurt too much |
| 22:25.27 | bmoraca_work | lordsporkton: are you just providing ITSP services or are you providing hardline services? |
| 22:25.30 | voipmonk | you did - and what did you get from your 2 second research |
| 22:26.06 | minotaur01 | i found all the info to modify the musiconhold.conf |
| 22:26.20 | minotaur01 | but they dont recommand a specific program |
| 22:26.28 | minotaur01 | they use shoutcast |
| 22:26.42 | p3nguin_ | mpg123, ogg123, et cetera |
| 22:27.44 | *** join/#asterisk Jak_o_Shadows (~Fake@121.209.35.170) |
| 22:28.42 | minotaur01 | p3nguin_: thank you |
| 22:29.47 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 22:30.08 | *** join/#asterisk cguerrero (~cuauhtemo@200.79.231.94) |
| 22:30.57 | lordsporkton | bmoraca_work: whats ITSP? im providering hardlines i think,data t1s and voip over it |
| 22:31.20 | p3nguin_ | ~itsp |
| 22:31.20 | infobot | [~itsp] An ITSP is an Internet telephony Service Provider or "VoIP Telephone company". The allow you to either PLACE calls to the PSTN (called Termination), RECEIVE calls from the PSTN (called Origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs |
| 22:32.06 | bmoraca_work | lordsporkton: are you billing the customer for the data T1 and allowing them to pass their own data over it or are you just billing them for the VoIP while the data T1 is in your name? |
| 22:32.46 | bmoraca_work | lordsporkton: the reason i ask is because the California PUC does not recognize VoIP providers as utilities, and thus you do not need to become a CLEC or pay any of the regulatory taxes and fees. |
| 22:34.25 | lordsporkton | bmoraca_work: the former. i did read that part about the voip not being recognized but i was unsure as you usually end up back ending to a traditional TDM voice circuit unless your upstream voice provider is something voip like what level3 offeres wholesale, which i may consider, however I will be billing customers for the data t1 and they will be using it for genereal internet access etc |
| 22:35.21 | bmoraca_work | lordsporkton: ahh...then you'll need to make sure you bill all appropriate fees, etc. billing nightmare :P |
| 22:35.50 | bmoraca_work | lordsporkton: might want to check this out: https://clec.att.com/clec/shell.cfm?section=2163 |
| 22:37.29 | lordsporkton | bmoraca_work: im reading it now |
| 22:38.43 | skrite | ok, when i run the ztcfg, my three channels are found, but nothing is there when i run my cli show channels |
| 22:39.39 | skrite | just cannot figgure out what is wrong |
| 22:40.26 | ChannelZ | are the channels actually configued to be used in chan_dahdi.conf ? |
| 22:42.22 | [TK]D-Fender | skrite: What command exactly? |
| 22:43.17 | skrite | well, i am wanting to use zaptel, not dahdi |
| 22:46.50 | skrite | show channels is just showing me 0 active channels and 0 active calls |
| 22:46.50 | *** join/#asterisk neurosys (~neurosys@c-71-196-20-208.hsd1.fl.comcast.net) |
| 22:46.51 | skrite | if what is on voip-info is old, where do i find better docs? |
| 22:47.02 | carrar | src |
| 22:47.30 | p3nguin_ | Your source tarball should include documentation. |
| 22:47.49 | skrite | ok |
| 22:48.53 | skrite | thanks guys |
| 22:49.57 | *** part/#asterisk Tech_Travis (~Travis@mail.techglia.com) |
| 22:50.02 | carrar | Do itashi mashite |
| 22:51.20 | *** join/#asterisk jayrod422 (~jayrod422@pool-71-99-94-9.tampfl.dsl-w.verizon.net) |
| 22:52.35 | jayrod422 | any know why with x-lite (behind nats) and "newer" verisons of asterisk I am getting a retrans_pkt: Maximum retries exceeded on transmission which drops outbound calls (initiated from x-lite user) after about 30 secs... but works fine for inbound (asterisk initiated) |
| 22:52.39 | skrite | ok, how do i know from asterisk if there are any kind of zap channels available? |
| 22:52.47 | *** join/#asterisk ManxPower-work (~EWieling@163.sub-75-234-126.myvzw.com) |
| 22:53.13 | cguerrero | skrite: zap show channels |
| 22:53.42 | p3nguin_ | If zap is not a valid command, then you probably don't have the zaptel channel driver loaded. |
| 22:54.00 | skrite | i am getting a 'no such command' |
| 22:54.07 | p3nguin_ | see above |
| 22:55.42 | skrite | well i have zaptel and ztdummy as well as the wctdm24xxp for the card are there when i do an lsmod, is that what you are referring to p3nguin_ ? |
| 22:55.58 | cguerrero | try zt_cfg -vvv |
| 22:56.12 | cguerrero | check if you have any channel configured |
| 22:56.14 | *** join/#asterisk ManxPower-work (~EWieling@163.sub-75-234-126.myvzw.com) |
| 22:57.12 | idespinner | skrite, try 'dahdi show channels' |
| 22:57.17 | skrite | ztcfg -vvv show i have channel 01, and channel 02, and channel 03 |
| 22:57.22 | idespinner | you may just be using dahdi instead... |
| 22:57.27 | skrite | Kewlstart (all) |
| 22:57.33 | skrite | ok |
| 22:57.36 | idespinner | nm |
| 22:57.45 | idespinner | if ztcfg works, you have zaptel |
| 22:58.03 | skrite | ok |
| 22:58.04 | idespinner | sounds like your zaptel channel driver isnt in asterisk... |
| 22:58.14 | idespinner | how did you build/install asterisk? |
| 22:58.33 | skrite | source tarball, ./configure make sudo make install |
| 22:59.26 | idespinner | skrite, did you install libpri and zaptel first? |
| 22:59.26 | Katty | hi |
| 22:59.35 | idespinner | err scratch libpri... |
| 22:59.44 | skrite | zaptel, i believe so, |
| 23:00.56 | skrite | however, when i am in the /etc/asterisk folder, there was no zapata.conf like i would have expected, i copied it over from the old server ( this is all an upgrade to better hardware) |
| 23:01.12 | skrite | i had tried dahdi, but removed it. |
| 23:01.48 | idespinner | do you have /etc/asterisk/chan_dahdi ? |
| 23:01.53 | skrite | i am still kinda new at this |
| 23:02.04 | skrite | yes i do |
| 23:02.14 | skrite | /etc/asterisk/chan_dahdi |
| 23:02.17 | skrite | exisits |
| 23:02.19 | idespinner | so... err sounds like you have dahdi installed.... |
| 23:02.23 | idespinner | asterisk -vvvvvvvvr |
| 23:02.31 | idespinner | then do dahdi show channels |
| 23:03.00 | skrite | gives me no shuch command 'dahdi show channels' |
| 23:03.15 | p3nguin_ | module show like zap |
| 23:03.15 | skrite | maybe i need to start clean? |
| 23:03.18 | idespinner | well... yea |
| 23:03.25 | idespinner | i dont know at this point |
| 23:03.34 | skrite | no dahdi shows up in lsmod. |
| 23:04.01 | idespinner | maybe asterisk was compiled for dahdi, then you removed and installed zaptel? |
| 23:04.03 | *** join/#asterisk JAMMAN2110 (~james@unaffiliated/jamman2110) |
| 23:04.12 | skrite | ok, if i needed to totally purge asterisk, dahdi, and zaptel from the system to start over.... .... |
| 23:04.20 | skrite | idespinner, that is possible. |
| 23:04.27 | skrite | been working on this a couple of days now |
| 23:04.53 | ChannelZ | why do you want zaptel over dahdi? |
| 23:04.56 | idespinner | If I were more of an expert i could tell you how, but yea |
| 23:05.08 | idespinner | yea, usually people use dahdi now.. |
| 23:05.12 | p3nguin_ | Which version of asterisk are you using? |
| 23:05.14 | skrite | well, zaptel is what i had configured before |
| 23:05.15 | idespinner | zaptel is gone... |
| 23:05.21 | skrite | i have to use 1.4 of asterisk |
| 23:05.28 | thehar | 1.4.22 to be exact |
| 23:05.46 | skrite | yes, the latest stable tarball |
| 23:05.53 | p3nguin_ | That's not 1.4.22. |
| 23:05.58 | idespinner | yea.. |
| 23:05.58 | skrite | soorry |
| 23:06.03 | skrite | 1.4.29 |
| 23:06.17 | p3nguin_ | You pretty much HAVE to use dahdi on 1.4.29. |
| 23:06.20 | thehar | 1.4.22 is the last stable release that will work with zaptel afaik |
| 23:06.23 | thehar | p3nguin_: yes |
| 23:06.39 | ManxPower-work | you do not have to use dahdi with 1.4.anything. |
| 23:06.41 | p3nguin_ | You have enough trouble, you certainly won't get zaptel to work on this version of asterisk. |
| 23:06.54 | skrite | ah |
| 23:07.00 | skrite | ok |
| 23:07.03 | idespinner | well, i dont think you need to format |
| 23:07.14 | p3nguin_ | You do if the file system is damaged. |
| 23:07.16 | idespinner | just remove zaptel as much as possible |
| 23:07.18 | skrite | no, have a webserver, etc... |
| 23:07.27 | idespinner | then install dahdi |
| 23:07.33 | idespinner | recompile asterisk |
| 23:07.37 | skrite | ok, remove zaptel and asterisk? |
| 23:07.43 | ManxPower-work | Only the actual channel driver is renamed to chan_dahdi.so. It still expects a zapata.conf is zapel is loaded and chan_dhadi.conf is DAHDI is loaded. It will use zap CLI commands if zap is loaded, etc. |
| 23:07.44 | idespinner | yea i guess |
| 23:07.46 | skrite | ok, because it was compiled with zaptel? |
| 23:07.50 | p3nguin_ | Remove zaptel. Install dahdi and dahdi-tools. Rebuild asterisk completely. |
| 23:08.12 | skrite | ok |
| 23:08.15 | idespinner | skrite, yea, what p3nguin_ said |
| 23:08.15 | skrite | will do. |
| 23:08.23 | skrite | ok, will do. |
| 23:08.24 | ManxPower-work | why don't you just try removing the dahdi config from /etc/asterisk? |
| 23:08.41 | skrite | since it was all compiled, i should be able to remove them all with make uninstall, right? |
| 23:09.20 | p3nguin_ | No clue. I never condone installation of any software without it being in packaged form. |
| 23:09.31 | thehar | hahaha |
| 23:09.34 | p3nguin_ | Which distro are you using? |
| 23:09.36 | thehar | laughs |
| 23:09.41 | skrite | ubuntu 9.10 |
| 23:09.51 | *** join/#asterisk hfb (~hfb@pool-98-112-226-53.lsanca.dsl-w.verizon.net) |
| 23:09.56 | skrite | but apt wants to install version 1.6 of zaptel |
| 23:10.02 | p3nguin_ | You can use checkinstall INSTEAD of "make install" to make your build into a package. |
| 23:10.32 | p3nguin_ | Once you make your build into a package, install it with dpkg. Now you can remove it more easily. |
| 23:10.42 | idespinner | p3nguin_, could you explain that more? does that do RPM aswell? |
| 23:10.45 | skrite | cool, thanks ? |
| 23:10.50 | p3nguin_ | Do this every time you compile anything. |
| 23:10.50 | skrite | ! |
| 23:10.54 | p3nguin_ | idespinner: yes. |
| 23:10.56 | skrite | right, will do |
| 23:11.03 | idespinner | ive never heard of that... |
| 23:11.22 | idespinner | (and it wasnt in any of the readme's) |
| 23:12.44 | *** join/#asterisk Torrieri (~Torrieri@nelug/crew/torrieri) |
| 23:13.52 | idespinner | p3nguin_, maybe a silly question but does checkinstall work with all apps you compile in linux? |
| 23:15.00 | p3nguin_ | idespinner: Assuming you are using something RH-based, install the rpmforge rpm, then you can install checkinstall via yum. It's not anything related to Asterisk. |
| 23:15.11 | p3nguin_ | idespinner: Yes, you run it instead of "make install". |
| 23:15.35 | idespinner | thx. |
| 23:15.41 | p3nguin_ | It's a sysadmin's friend. |
| 23:15.53 | idespinner | whats the debian equivalent to rpmforge? |
| 23:16.06 | p3nguin_ | Only masochists don't use packaging systems. |
| 23:16.16 | Katty | ^- his mom |
| 23:16.30 | idespinner | thats what I always thought but i never knew of any alternative... :/ |
| 23:16.41 | p3nguin_ | checkinstall could be installed on Debian by default. apt-cache search checkinstall |
| 23:17.07 | thehar | Katty: HI |
| 23:17.14 | Katty | ohai |
| 23:17.20 | thehar | how goes it |
| 23:17.24 | Katty | bored |
| 23:17.38 | thehar | his mom will keep you company |
| 23:17.40 | jaytee | hi Katty |
| 23:18.06 | *** join/#asterisk _mwoodj_ (~mwoodj@pdpc/sponsor/digium/hyper-eye) |
| 23:19.59 | p3nguin_ | idespinner: http://www.debian-administration.org/articles/147 |
| 23:20.03 | Katty | hi jaytee |
| 23:20.08 | Katty | Qwell: I'm going to farm Jeeves |
| 23:36.49 | tgunr | p3nguin_: found it! |
| 23:36.50 | tgunr | ; autodomain = yes |
| 23:39.09 | ruben23 | hi i have unusual logs on my asterisk which display like this ---> [Feb 20 07:37:39] NOTICE[4758]: chan_sip.c:12669 handle_response_peerpoke: Peer 'delridgeis' is now Lagged. (1238ms / 1000ms) |
| 23:40.05 | ruben23 | but when i duplicate the console my asterisk and have ping the same carrier ip of delridges, on my ping i only get 256ms its not actually getiing 1238ms at all...anyone have idea..? |
| 23:41.34 | jaskew | ruben23: Latency varies. It could be that it was 1238ms at the time the check was done, and then returned to normal |
| 23:42.18 | jaskew | ruben23: is the peer a remote peer, or on your LAN? |
| 23:44.15 | ruben23 | jaskew: as the warning display 1238ms along as i continues ping it on the other console same server i only gettting 256ms continues then jumps to 280ms.. |
| 23:44.22 | ruben23 | like that. |
| 23:44.39 | ruben23 | the peer si on remote yes, its a voip carrier.. |
| 23:44.48 | *** join/#asterisk ecrist (ecrist@pdpc/supporter/professional/ecrist) |
| 23:45.04 | ecrist | anyone in here run digium on freebsd? |
| 23:45.18 | jaskew | ruben23: I assume you have "qualify=yes" on that peer? |
| 23:45.32 | *** part/#asterisk sleazye (~ernesto@64-71-24-110.static.wiline.com) |
| 23:45.38 | ruben23 | qualify=1000 |
| 23:46.19 | ruben23 | when that waring appears callersa re not able to hear clients and getting choppy lines. |
| 23:46.22 | *** join/#asterisk tzafrir (~tzafrir@212.179.75.202) |
| 23:47.33 | *** part/#asterisk ecrist (ecrist@pdpc/supporter/professional/ecrist) |
| 23:47.36 | jaskew | ruben23: Really sounds like latency - VoIP competing with other traffic. |
| 23:48.03 | jaskew | ruben23: How are you connected to the Interenet? Is there a firewall and/or router? Is it DSL, T1 or something else? |
| 23:48.44 | ruben23 | im on E1 2Mbps line uisng voip trunks, with g729 codec, using softphones.. |
| 23:49.23 | jaskew | What kind of firewall/router? Is it VoIP aware? |
| 23:49.52 | ruben23 | im not using firewall at all, my asterisk is using public IP, open to WAN.. |
| 23:50.35 | jaskew | So the asterisk box is the only thing connected to the E1, or is something else bridged in with another public IP? |
| 23:51.28 | ruben23 | E1 is used also by another box my firwall where data traffic pass thorugh, and my asterisk server where vocie passes through.. |
| 23:51.56 | ruben23 | so E1 at center point is accepting data and voice traffic with the 2 box, |
| 23:52.07 | ruben23 | firewall/box and asterisk box. |
| 23:52.28 | jaskew | OK - so the asterisk box and the firewall are "sharing" the E1 |
| 23:52.39 | ruben23 | the callers are using softphones, while accessing some sites on the internet.. |
| 23:52.50 | ruben23 | yes your right |
| 23:53.36 | jaskew | So if someone downloads/uploads a large file, the firewall can use all of the E1 bandwidth and cause asterisk to loose packets. |
| 23:53.54 | ruben23 | but i see on my mrtg monitoring im not over utilizing ther E1 connection..why is it..? |
| 23:54.02 | jaskew | hm - hang on - phone call |
| 23:54.09 | ruben23 | ok |
| 23:54.25 | p3nguin_ | tgunr: I'm pretty sure I don't use that setting, so there's probably no reason you need it. |
| 23:54.47 | *** join/#asterisk seanjohn (~john@ns1.sheltoncomputers.com) |
| 23:55.14 | seanjohn | in the call file, what lines would I put to make the call retry every 15 seconds for 30 minutes? |
| 23:55.17 | tgunr | cc, now I have a softphone registering but getting message too big when i try a dial |
| 23:55.28 | tgunr | :( |
| 23:56.09 | seanjohn | I have made an awesome busyredial dialplan I will gladly share if I can get this straightened out; this is the ONLY problem. |
| 23:57.09 | p3nguin_ | seanjohn: probably RetryTime: 15 and MaxRetries: 7200 |
| 23:57.42 | ruben23 | jaskew: what you can suggest..? if ever like in time i will expand then my connection is 4Mbps but burstbale to like 20Mbps...but still the data and vocie are shared...is it still a problem...? |
| 23:58.48 | p3nguin_ | seanjohn: Well, that might not be so good. Perhaps WaitTime: 30 RetryTime: 15 and MaxRetries: 5400 would be better. |
| 23:58.54 | ManxPower-work | ruben23: You can only do a limited amount of QoS if you only control one end of your internet connection. However, with plenty of bandwidth that limited amount of QoS may work well for you. |
| 23:59.24 | ManxPower-work | It will still be a pain in the ass to get QoS working. |
