| 00:01.01 | mohsen | When I change my phone auth info and change back to the first auth info, I fall into some troubles with that phone. It can not place any call and CLI says: [Apr 2 16:36:25] WARNING[27901]: chan_sip.c:8377 check_auth: username mismatch, have <98018>, digest has <98017> |
| 00:01.03 | mohsen | [Apr 2 16:36:25] NOTICE[27901]: chan_sip.c:13815 handle_request_invite: Failed to authenticate user "98017" <sip:98017@sip.newgtel.com>;tag=SCeE4p7DObNjHv6I |
| 00:05.11 | *** join/#asterisk djs26 (n=djs@unaffiliated/djs26) |
| 00:05.26 | mohsen | 98017 the first username and 98018 is the second one. |
| 00:05.31 | mohsen | And I am using real time |
| 00:07.17 | *** join/#asterisk Katty (n=The@adsl-68-92-250-115.dsl.stlsmo.swbell.net) |
| 00:08.46 | Katty | so did anyone ever answer my question about needing ztdummy even if i have wanpipe? |
| 00:10.41 | *** join/#asterisk axisys (n=axisys@ip70-174-179-120.dc.dc.cox.net) |
| 00:10.57 | *** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.pa.comcast.net) |
| 00:13.27 | *** join/#asterisk riddlebox (n=james@ppp-70-225-32-179.dsl.covlil.ameritech.net) |
| 00:13.46 | *** join/#asterisk codazoda (n=codazoda@c-67-161-216-115.hsd1.ut.comcast.net) |
| 00:15.12 | codazoda | Asterisk won't start with init.d automatically. If I run "/etc/init.d/asterisk start", it works fine. "chkconfig --list asterisk" shows it "on". In the log file, even though it hasn't started really, I see "CDR simple logging enabled." then I see "147 modules will be loaded." then this repeats forever. |
| 00:15.41 | codazoda | This is in messages. Maybe I should look at another log file? |
| 00:16.07 | tzafrir | Katty, I believe you don't |
| 00:16.24 | jameswf-home | endles amusement makinf the tts say poop |
| 00:17.36 | Katty | k |
| 00:21.54 | codazoda | No clues as to where to look in my log files for some type of problem that is causing asterisk to start over and over again? I assume that's why loader.c keeps showing up in the messages file every 5 seconds. |
| 00:21.56 | *** join/#asterisk khronos (n=khronos@aquaman.perryinstitute.org) |
| 00:31.10 | *** join/#asterisk nighty^ (n=nighty@210.188.173.246) |
| 00:32.16 | drmessano | 1.4.19 |
| 00:32.38 | drmessano | HAVE A NICE TRIP RUSSELLB |
| 00:33.01 | *** join/#asterisk djs26 (n=djs@unaffiliated/djs26) |
| 00:33.26 | *** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.pa.comcast.net) |
| 00:33.46 | Katty | where's russell going? |
| 00:36.10 | codefreeze | he's going to get married. |
| 00:37.57 | *** join/#asterisk anthm (n=anthm@68-29-161-69.area4.spcsdns.net) |
| 00:37.57 | *** mode/#asterisk [+o anthm] by ChanServ |
| 00:38.15 | drmessano | I stalk him on Twitter |
| 00:38.19 | drmessano | I feel like i know him |
| 00:39.07 | drmessano | Maybe it's because I have blueprints of his house in PDF form |
| 00:39.11 | drmessano | shrugs |
| 00:41.34 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 00:41.51 | codefreeze | drmessano: scary, very scary... ;^) |
| 00:41.56 | drmessano | heh |
| 00:42.52 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 00:44.10 | *** join/#asterisk husimon (n=nhuisman@aeko.IfA.Hawaii.Edu) |
| 00:44.33 | husimon | Hi, I have a cisco 7940 and i'm using the soft buttons to setup a call forward but when I call the phone the cli messages say it's busy |
| 00:44.34 | husimon | any ideas? |
| 00:45.50 | husimon | http://pastebin.com/m5d18539a |
| 00:45.59 | husimon | it does say that the forwarding is prevented |
| 00:46.04 | husimon | but i'm not sure why, here is the cli output |
| 00:49.03 | husimon | um is anyone awake? |
| 00:49.07 | drmessano | Yes |
| 00:49.09 | husimon | hehe |
| 00:49.13 | drmessano | Why is the call going to local/blah |
| 00:49.19 | husimon | I dunno i was just looking at that |
| 00:49.25 | husimon | blah is correct |
| 00:49.28 | drmessano | Yes |
| 00:49.31 | husimon | that's what the softkey on the phone is doing |
| 00:49.31 | drmessano | Local/ is not |
| 00:49.35 | drmessano | So go figure that out |
| 00:50.47 | hmmhesays | the dnd softkey? |
| 00:51.00 | husimon | dnd? |
| 00:51.15 | hmmhesays | do no disturb |
| 00:51.22 | husimon | hmmhesays, well I have a menu "Redial", "Newcall", "Cfwdall" |
| 00:51.27 | husimon | i press the cfwdall and then enter a number |
| 00:51.32 | hmmhesays | cfwdall is the same as dnd |
| 00:51.41 | husimon | wtf? |
| 00:51.55 | hmmhesays | well, I believe they both issue a 302 redirect\ |
| 00:52.01 | husimon | i see |
| 00:52.03 | hmmhesays | I could be wrong or just really drunk |
| 00:52.21 | hmmhesays | anyway you need to look at the dialplan in the [phones] context |
| 00:52.22 | drmessano | That makes no sense |
| 00:52.24 | drmessano | Go sober up |
| 00:52.43 | husimon | well I don't understand why it would put Local in front of it |
| 00:52.46 | hmmhesays | I'm pretty sure a dnd issues a 302 on my linksys phone here |
| 00:53.01 | hmmhesays | because you got a redirect back from your phone, and that is how asterisk handles it |
| 00:53.17 | hmmhesays | -- Got SIP response 302 "Moved Temporarily" back from 128.171.77.50 |
| 00:53.17 | drmessano | But a call forward? |
| 00:53.51 | hmmhesays | - Now forwarding Zap/1-1 to 'Local/99389984@phones' (thanks to SIP/342-08686700) |
| 00:54.02 | hmmhesays | the answer to your question is in the [phones] context |
| 00:54.46 | husimon | the phones context has the outgoing context included so it should dial that number fine. |
| 00:55.20 | lmadsen | dnd should really issue 486 Busy Here |
| 00:55.29 | lmadsen | cfwd would reply with 302 Redirect |
| 00:55.45 | husimon | which it is |
| 00:55.49 | husimon | isn't that what that sip message means? |
| 00:56.21 | hmmhesays | yes |
| 00:56.24 | lmadsen | the Local channel is to redirect the call back into the dialplan. It makes sense. |
| 00:56.25 | hmmhesays | show us your dialplan |
| 00:56.30 | lmadsen | pastebin |
| 00:56.33 | husimon | kk |
| 00:56.54 | lmadsen | and the entire console output would be handy too |
| 00:57.22 | husimon | http://pastebin.com/m11cca9e6 -> extensions.conf |
| 00:57.36 | husimon | the previous pastebin is the entire cli |
| 00:57.42 | husimon | http://pastebin.com/m5d18539a |
| 00:57.53 | husimon | unless you want sip debug output |
| 00:58.03 | lmadsen | I had just missed it |
| 00:58.13 | husimon | feel free to tell me how that extensions.conf is lame and ways you would re-write it :) |
| 00:58.42 | husimon | oh ya I was suppose to remove all those redundant outgoing rules |
| 00:58.48 | hmmhesays | console dial 99389984@phones |
| 00:58.53 | hmmhesays | see if you can do that |
| 00:58.57 | husimon | k |
| 00:58.58 | lmadsen | ya, thats a good test |
| 00:59.17 | husimon | worked |
| 00:59.37 | lmadsen | hrmmm |
| 00:59.40 | husimon | http://pastebin.com/m60765a5d |
| 00:59.45 | lmadsen | nothing actually jumps out at me as wrong... |
| 00:59.48 | husimon | not sure what that dsp weirdness is |
| 01:00.31 | lmadsen | only thing I can think of is it isn't traversing into the includes.... |
| 01:00.41 | lmadsen | but I don't know why it wouldn't |
| 01:00.52 | husimon | ya cause I can call out on the phone fine |
| 01:00.55 | lmadsen | you could try adding the _9NXXXXXX part to your phones directly to see if it catches |
| 01:00.57 | husimon | which is in the phones context |
| 01:01.07 | husimon | spose that's a good test |
| 01:01.21 | husimon | but even if I forward it to a local extension like xxx |
| 01:01.24 | husimon | 395 or something |
| 01:01.25 | husimon | it doe sit |
| 01:01.28 | lmadsen | it does? |
| 01:01.32 | husimon | and 395 is in the phones context |
| 01:01.42 | lmadsen | directly in the phones context? |
| 01:01.51 | lmadsen | I see only includes... |
| 01:01.56 | husimon | oh wait, |
| 01:01.59 | lmadsen | sip debug might be handy at this point |
| 01:02.03 | husimon | ya you are right |
| 01:02.39 | *** join/#asterisk BeeBuu (n=beebuu@61.145.76.74) |
| 01:03.04 | husimon | http://pastebin.com/m3100dc39 |
| 01:03.29 | drmessano | My God, It's Full Of Includes! |
| 01:03.35 | husimon | hehe |
| 01:03.37 | drmessano | wonders who gets the reference |
| 01:03.41 | husimon | i hope everyone |
| 01:03.45 | lmadsen | not me |
| 01:03.51 | lmadsen | maybe I'm not old enough :) |
| 01:03.56 | husimon | space odyssey |
| 01:04.02 | drmessano | Yep |
| 01:04.14 | husimon | sip debug doesn't give much more info |
| 01:04.15 | drmessano | 2001 + 2010 = Great evening |
| 01:04.48 | drmessano | One day, Asterisk will pwn us all, like Hal did |
| 01:05.00 | hmmhesays | doubtful |
| 01:05.09 | hmmhesays | it will deadlock before it gets that intelligent |
| 01:05.19 | husimon | i think it needs to learn how to cluster first |
| 01:05.45 | drmessano | Maybe the deadlocks are well calculated intentional flaws meant to throw us off it's self awareness |
| 01:05.59 | drmessano | Bet you NEVER thought of THAT, huh? |
| 01:06.13 | husimon | Here is something odd too |
| 01:06.21 | husimon | exten => 342,1,Macro(stdexten,${EXTEN},SIP/${EXTEN}&LOCAL/8089389984@phones,ti), that works fine to forward |
| 01:06.26 | husimon | and that's the same thing the phone is doing |
| 01:06.43 | husimon | sending a local command to a number at the phones context |
| 01:06.52 | drmessano | hides his eyes and peeks through his fingers |
| 01:07.01 | husimon | sorry that's a macro |
| 01:07.12 | husimon | but it is getting the same destination |
| 01:08.40 | husimon | I added a 7 digit outbound pattern in the phones context and no change |
| 01:09.14 | *** join/#asterisk coppice (n=chatzill@99.166.17.210.dyn.pacific.net.hk) |
| 01:09.32 | *** join/#asterisk MmixX (i=mmixx@202.124.138.69) |
| 01:09.47 | husimon | ..... |
| 01:09.54 | husimon | ok wtf, I found out why |
| 01:10.28 | lmadsen | oh? |
| 01:10.41 | husimon | why the hell did I have an i on that extension |
| 01:11.13 | lmadsen | eh? |
| 01:11.20 | husimon | blah it must have been because I had it ringing two phones previously |
| 01:11.22 | husimon | i = don't forward |
| 01:11.27 | *** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.pa.comcast.net) |
| 01:11.38 | lmadsen | you had a variable set? |
| 01:11.46 | drmessano | KWHAT? |
| 01:12.21 | drmessano | That is "need to know".. should have directly followed "hey guys" |
| 01:12.24 | drmessano | Major fail |
| 01:12.30 | husimon | : Asterisk will ignore any forwarding requests it may receive on this dial attempt. (new in 1.4) Useful if you are ringing a group of people and one person has set their phone to forwarded direct to voicemail on their cell or something which normally prevents any of the other phones from ringing. |
| 01:13.01 | husimon | I changed the dialplan ages ago and then forgot what "i" meant and ignored it. |
| 01:13.05 | husimon | sigh |
| 01:13.32 | lmadsen | oh, 'i' in Dial() |
| 01:13.34 | husimon | ya |
| 01:13.43 | lmadsen | I thought you meant the 'i' extension |
| 01:13.45 | husimon | oh ;) |
| 01:13.48 | husimon | sorry i wasn't clear |
| 01:14.07 | lmadsen | aha, wish I had noticed that on the Dial() line sooner for ya |
| 01:14.12 | lmadsen | glad you found it though |
| 01:14.21 | husimon | np, at least I know asterisk was doing the right thing. |
| 01:14.22 | lmadsen | I'll try to remember that scenario :) |
| 01:14.25 | lmadsen | indeed |
| 01:14.30 | *** join/#asterisk BCS-Satori (n=BCS-Sato@204.52.131.22) |
| 01:14.33 | husimon | layer 0 error again |
| 01:14.38 | lmadsen | welp, off to put groceries away, have a glass of wine and clean up the living room |
| 01:14.44 | lmadsen | I like ot call it layer 8 |
| 01:14.49 | husimon | ah |
| 01:15.04 | BCS-Satori | has anyone been able to make zaptel on os x 10.5.2 ? |
| 01:15.06 | husimon | i guess that does make more sense |
| 01:15.52 | lmadsen | :) |
| 01:16.02 | lmadsen | my 4 years of schooling taught me that much |
| 01:16.12 | husimon | heh |
| 01:16.25 | lmadsen | bye! |
| 01:16.25 | husimon | it's funny because most problems are layer 1 or layer 8 |
| 01:16.31 | husimon | the middle layers, not so much. |
| 01:16.48 | *** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.pa.comcast.net) |
| 01:18.14 | *** part/#asterisk mauroC (n=incastri@host242-129-dynamic.12-79-r.retail.telecomitalia.it) |
| 01:18.25 | drmessano | Layer 9 is the real issue |
| 01:18.39 | drmessano | I call those "idiot friend problems" |
| 01:18.41 | husimon | haha |
| 01:18.42 | drmessano | As in |
| 01:18.51 | husimon | i was just thinking that must be what you mean |
| 01:18.56 | drmessano | "Hey, you told me to reboot and now my itunes doesnt work" |
| 01:19.02 | drmessano | "Umm, that had nothing to do with it" |
| 01:19.15 | drmessano | "Well, my friend used to work in IT, and he said it does" |
| 01:19.25 | drmessano | Your friend, is an effin idiot |
| 01:19.35 | husimon | no, it's more like "So do I need plug the monitor into the hard drive?" |
| 01:19.45 | husimon | or "I'll just bring the hard drive over to your house and you can try to fix it" |
| 01:19.50 | husimon | aka the CASE! |
| 01:19.53 | drmessano | There's always a damn friend |
| 01:19.58 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-50571451aeb31e5a) |
| 01:20.11 | drmessano | "My friend told me that zaptel runs on windows" |
| 01:20.13 | husimon | then again, i'm sure i would sound like a tard asking a mechanic about my car |
| 01:20.31 | husimon | the best is "I downloaded asterisk, now what?" |
| 01:21.27 | drmessano | I used to love fixing friends of friends computers for free.. and then they would screw it up, and get mad at the common friend for having me work on it.. "Well, he said another friend of his looked at it, and..." |
| 01:21.38 | drmessano | Well, should have called him in the first place.. saved me a weekend |
| 01:22.08 | real0ne | drmessano can u explain me how can make call with wlite |
| 01:22.14 | real0ne | xlite |
| 01:22.32 | drmessano | Do you have it connected to asterisk yet? |
| 01:23.23 | real0ne | that what i need to know:) |
| 01:23.33 | real0ne | how can i do it |
| 01:23.57 | drmessano | ~book |
| 01:23.58 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 01:24.02 | drmessano | Thats a good start |
| 01:24.08 | real0ne | i have read it |
| 01:24.13 | real0ne | but it to lange |
| 01:24.18 | *** join/#asterisk Kyoshi (i=whoa@pool-71-167-117-15.nycmny.fios.verizon.net) |
| 01:24.24 | husimon | http://www.asteriskguru.com/tutorials/xlite_softphone.html |
| 01:24.26 | drmessano | lange? |
| 01:24.39 | husimon | err, that page is crap |
| 01:25.15 | husimon | long |
| 01:25.42 | husimon | real0ne, You need to understand the dialplan (extensions.conf, sip.conf) before you can add that phone. |
| 01:26.05 | real0ne | husimon yes |
| 01:26.09 | husimon | do you? |
| 01:26.51 | real0ne | yes |
| 01:26.57 | real0ne | i understand |
| 01:27.13 | husimon | well the book does have a section that talks about setting up xlite |
| 01:28.11 | drmessano | Im sure you knew that.. having read it |
| 01:29.07 | real0ne | yes but to read all this book it two lange |
| 01:29.22 | drmessano | You need to read it |
| 01:29.26 | real0ne | and i have debian not CentOS like the book |
| 01:30.06 | drmessano | Well, good luck to you then.. it is all in the book |
| 01:30.22 | real0ne | ok |
| 01:30.24 | real0ne | :) |
| 01:34.01 | *** join/#asterisk lvl- (n=lvl@101.248.dhcp.lht.hhs.nl) |
| 01:34.45 | ectospasm | real0ne: your Linux distro is immaterial... it should work on any one of them... |
| 01:35.44 | lmadsen | the book is too large? eep... wait until we decide to write a definitive guide! :) |
| 01:36.06 | lmadsen | tomes I, II, III and IV |
| 01:36.33 | lmadsen | comes with a DVD of allison reading the entire book |
| 01:36.52 | husimon | lmadsen, blueray |
| 01:36.54 | lmadsen | video interviews with world renowned developers and visionaries |
| 01:37.04 | husimon | bluray ewps |
| 01:38.12 | drmessano | Surely that is a compliment |
| 01:39.36 | *** join/#asterisk Horvath (n=Horvath@CABLE-72-53-93-87.cia.com) |
| 01:41.21 | TJNII | heh. Makes me think of the manuals for my truck. 12" of info on thin paper. |
| 01:41.27 | Horvath | I'm trying to get T38 pass-through to work from T38 Provider -> Asterisk -> Callweaver. Its working fine for T38 Provider -> Asterisk -> Grandstream ATA but passing it to Callweaver appears to be a problem |
| 01:42.35 | Horvath | I have t38pt_udptl=yes in my SIP peer which again works for the grandstream ATA but somehow not for Callweaver |
| 01:46.25 | drmessano | Callweaver? AHEM |
| 01:48.21 | Horvath | Yes, the devil im sure but until asterisk can terminate t38 you do what you can |
| 01:48.44 | *** join/#asterisk putnopvut (n=putnopvu@76.29.239.233) |
| 01:48.44 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 01:49.44 | drmessano | Your issue appears to be WITH callweaver, as you stated.. so they would be the place to look for support |
| 01:51.10 | Horvath | Usually you can find everything on the voip-info wiki but apparently no one has written up how to connect an Asterisk box with Callweaver |
| 01:51.45 | drmessano | Your issue appears to be connecting callweaver to asterisk |
| 01:52.42 | CCFL_Man2 | those wall place with square hols in them that you snap in either blanks or rj11/45 jacks, where can one get those? |
| 01:53.10 | Horvath | CCFL_Man2, Home Depot |
| 01:53.14 | drmessano | Greybar, or pay markup at home depot or Lowes |
| 01:53.45 | CCFL_Man2 | Horvath: they have the ones with 6 holes in them? |
| 01:53.47 | jameswf-home | greybar..... hmmmm |
| 01:53.51 | drmessano | yes |
| 01:54.06 | CCFL_Man2 | i need rj11 and rj45 in one wall plate |
| 01:54.14 | drmessano | Ok |
| 01:54.25 | drmessano | Well, that's your answerr |
| 01:54.30 | Horvath | CCFL_Man2, Yes you buy a plate and snap in what you need |
| 01:54.35 | jameswf-home | yes most professional wallplates are modular |
| 01:54.42 | Horvath | CCFL_Man2, One sec ill find the brand name |
| 01:54.44 | drmessano | Home Depot, Lowes, or Graybar |
| 01:55.22 | CCFL_Man2 | Horvath: i see them in commercial places like here at work, but haven't seen them at many stores |
| 01:55.26 | jameswf-home | I would go graybr if you haveone local.... I couldnt afford to wire a place up at homedepot prices |
| 01:55.32 | Horvath | CCFL_Man2, http://www.leviton.com/OA_HTML/ibeCCtpSctDspRte.jsp?section=10730&minisite=10028 |
| 01:55.32 | jameswf-home | Frys even |
| 01:55.41 | CCFL_Man2 | ahh |
| 01:55.54 | drmessano | Home depot and lowes are high markup |
| 01:56.06 | drmessano | I would avoid at all costs if you have a graybar |
| 01:56.08 | *** join/#asterisk quigon (n=matias@190.3.121.15) |
| 01:56.08 | jameswf-home | but they have cheap labor outside |
| 01:56.16 | jameswf-home | or frys |
| 01:56.33 | Horvath | When you buy the 10 packs I dont find them expensive |
| 01:56.37 | drmessano | LOL |
| 01:56.49 | drmessano | Horvath: Have you ever bought in REAL bulk? |
| 01:57.01 | drmessano | Lowes and/or Home Depot is HUGE markup |
| 01:57.14 | drmessano | Even small quantity bulk elsewhere |
| 01:57.17 | drmessano | MUCH cheaper |
| 01:57.18 | jameswf-home | my wife says I have real bulk.... but I am on a diet |
| 01:57.33 | Horvath | Well sometimes you are willing to pay a huge markup for the convenience |
| 01:57.59 | *** join/#asterisk rickross (n=rickross@supporter/active/rickross) |
| 01:58.07 | drmessano | Well, if you do any real wiring of that stuff, it's not worth it |
| 01:58.11 | Horvath | but to answer your question no |
| 01:58.17 | jameswf-home | be all fancy and crap.... get a 3 hole plate put eth/voice/video |
| 01:58.58 | jameswf-home | which reminds me I need to go up in the attick before temps get over 90 |
| 01:59.04 | drmessano | 2 RJ45, 1 RJ11, 1 F, and 1 Fiber |
| 01:59.05 | drmessano | Done |
| 01:59.26 | drmessano | or |
| 01:59.37 | drmessano | 2 RJ45, 1 RJ11, 1 F, and a L and R RCA for audio |
| 01:59.40 | jameswf-home | has hes whole house on BNC with token ring topology... |
| 01:59.49 | CCFL_Man2 | ahh, leviton does have them |
| 02:00.09 | jameswf-home | thinks token is cool... |
| 02:00.33 | drmessano | thinks buying in bulk at home depot is cool... |
| 02:00.37 | Horvath | CCFL_Man2, Yep, their greap |
| 02:00.39 | Horvath | great* |
| 02:01.40 | CCFL_Man2 | yeah |
| 02:03.17 | *** join/#asterisk m4sk4r4 (n=m4sk4r4@189.13.183.165) |
| 02:03.54 | jameswf-home | heh Home Depot http://www.noob.us/humor/the-amazing-racist-home-depot/ |
| 02:05.23 | *** join/#asterisk PepOSX (n=angeldav@200.93.28.168) |
| 02:07.22 | drmessano | ROFL |
| 02:07.29 | CCFL_Man2 | * |
| 02:08.14 | jameswf-home | does not endorse that video and can neither confirm nor deny it is funny as he** |
| 02:09.14 | drmessano | I like to round up Callweaver installers and drive them over to digium HQ every now and then |
| 02:09.19 | drmessano | It's about as funny |
| 02:12.48 | rickross | hi all - anyone running a recent beta of asterisk 1.6.0 ? |
| 02:13.20 | CCFL_Man2 | i don't know if i should terminate this twi pair wire as a serial port or phone line |
| 02:14.32 | CCFL_Man2 | what do you think i should do? |
| 02:15.11 | rickross | I'm getting bizarre issues with the more recent betas, but earlier worked reasonably well |
| 02:15.13 | rickross | http://pastebin.ca/968510 |
| 02:15.47 | rickross | and I have no idea if this is because of FreePBX or if it is a core issue with asterisk? |
| 02:23.06 | drmessano | twisted pair for what? |
| 02:23.07 | *** join/#asterisk _mwoodj_ (n=MWoodJ@pdpc/sponsor/digium/hyper-eye) |
| 02:24.16 | andrewn | anyone here from indonesia? |
| 02:28.32 | CCFL_Man2 | drmessano: two pair |
| 02:28.41 | CCFL_Man2 | i mean |
| 02:32.57 | jameswf-home | yes andrewn you are |
| 02:35.08 | andrewn | I've never even been there |
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| 02:49.23 | *** join/#asterisk asdx (n=diego@adsl-151-219.click.com.py) |
| 02:49.27 | asdx | hi |
| 02:49.35 | CCFL_Man2 | what color snap in jacks can i use for a second phone line? |
| 02:49.36 | *** join/#asterisk greed (n=greed@mail.octapod.org) |
| 02:49.37 | asdx | how can i configure asterisk as a sip proxy |
| 02:50.24 | JT | you can't. |
| 02:50.34 | JT | use SER or OpenSER for that |
| 02:50.45 | drmessano | CCFL_Man2: i go with red RJ-45s and Grey RJ-11s |
| 02:51.13 | CCFL_Man2 | hmm.. |
| 02:51.16 | CCFL_Man2 | why red? |
| 02:51.20 | asdx | JT: why is not possible with * |
| 02:51.22 | drmessano | Sorry, orange |
| 02:51.40 | drmessano | Orange is just a common.. |
| 02:51.58 | CCFL_Man2 | oh |
| 02:52.07 | JT | asdx: because asterisk is a B2BUA |
| 02:52.11 | CCFL_Man2 | grey seems like a good color |
| 02:52.13 | JT | totally different |
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| 02:52.25 | drmessano | I've seen blue RJ-45s, but they don't interest me |
| 02:52.35 | CCFL_Man2 | i'll use grey for line 2, white and purple for line one |
| 02:52.45 | CCFL_Man2 | blue i use for ethernet |
| 02:52.47 | asdx | JT: ok |
| 02:53.24 | CCFL_Man2 | what color rj45 would you use for rs232? |
| 02:53.30 | drmessano | Orange is also pair 1 in most applications that use CAT5 |
| 02:53.32 | CCFL_Man2 | i'm thinking ivory or grey |
| 02:53.41 | CCFL_Man2 | yeah |
| 02:54.15 | drmessano | rs232 for what? |
| 02:55.09 | greed | Hey guys, Is there a way that I can do something on actual pickup of a call to the receiver. I have jabber messages but instead of them going to one person I want to send it to whatever extension answered the call. Is that possible ? |
| 02:57.05 | CCFL_Man2 | drmessano: a serial terminal |
| 02:57.32 | drmessano | Just gonna pipe that between two points? |
| 02:58.54 | rickross | is this an invalid invocation of ExecIf() ? |
| 02:58.57 | rickross | exten => s,n,ExecIf($["${REALCALLERIDNUM:1:2}" = ""],Set,REALCALLERIDNUM=${CALLERID(number)}) |
| 03:00.27 | lmadsen | rickross: looks ok.... |
| 03:01.05 | rickross | thx lmadsen - somehow it seems to be giving asterisk indigestion |
| 03:01.08 | rickross | [Apr 2 22:55:10] ERROR[8466] app_exec.c: Usage: ExecIf(<expr>?<appiftrue>(<args>)[:<appiffalse>(<args)]) |
| 03:01.51 | lmadsen | exten => s,n,Exec(${IF($["${REALCALLERIDNUM:1:2}" = ""]?Set(REALCALLERIDNUM=${CALLERID(number)}):NoOp())}) is how I do it.... |
| 03:02.03 | lmadsen | although your format looks ok..... |
| 03:02.21 | JT | drmessano: rs232 isn't dead yet ;) |
| 03:02.50 | rickross | lmadsen, yours looks more like what it seems to want |
| 03:03.02 | lmadsen | notice I'm not using ExecIf() |
| 03:03.53 | putnopvut | rickross you need the '?' to tell it where the end of your condition is and where the app you want to execute begins. |
| 03:04.06 | lmadsen | oh duh |
| 03:04.22 | lmadsen | that means it's bedtime |
| 03:04.23 | lmadsen | night! |
| 03:04.24 | rickross | thx, lemme try that |
| 03:04.42 | rickross | curiously, the previous form has been working in older versions for a long time |
| 03:05.06 | rickross | but with asterisk 1.6.0 betas it isnt so happy ;) |
| 03:05.50 | putnopvut | rickross, that's because it was changed in that branch to be more consistent with the other *If applications |
| 03:05.54 | putnopvut | Like GotoIf. |
| 03:07.34 | putnopvut | I agree that it's bedtime though :) |
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| 03:23.07 | JayTee52 | I'm getting alot of undefined references when trying to run make zttool for zaptel-1.4.9.2. I have ncurses, newt and newt-devel installed. I had no problem running this on RHEL5 64bit but now I'm trying to do it on CentOS 5.1 |
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| 03:49.03 | lesterc | I've got bitten by the infamous reboot loop from a polycom Soundstation IP 4000. Does anyone know how to recover from it? |
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| 04:10.34 | jameswf-home | just did yum install $(yum list lets see what happens) |
| 04:10.48 | jameswf-home | *just did yum install $(yum list) lets see what happens |
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| 04:14.19 | fujin | I've noticed that the output from 'queue show' is truncating my queue names |
| 04:14.23 | fujin | Is there a way to get the full name of a queue? |
| 04:14.27 | jameswf-home | yay node poop |
| 04:14.32 | fujin | (I'm looking for a concise option, like sip show peers concise or whatever it is.) |
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| 04:16.12 | fujin | pretyt much the same behaviour as this: http://lists.digium.com/pipermail/asterisk-users/2006-March/145320.html |
| 04:16.18 | fujin | I'd like to see the actual queue name, somehow. |
| 04:16.22 | fujin | Is manager my only option? |
| 04:17.03 | hmmhesays | why not use the manager simple and easy |
| 04:17.12 | hmmhesays | probably need 30 lines in perl to take care of it |
| 04:17.28 | fujin | That's more than the 6 lines of Perl I'm currently using to parse all active queues and remove all members from it. |
| 04:18.01 | fujin | thanks though, I figured it'd come down to using the AMI. |
| 04:18.13 | hmmhesays | nice, i'm guessing your're using the asterisk manager package from cpan then |
| 04:19.20 | fujin | no, I've not lookeda t it |
| 04:19.31 | fujin | the 6 lines is purely some magical regex and backticks. |
| 04:19.36 | fujin | and some loops |
| 04:20.21 | fujin | I didn't realise there was a CPAN module for AMI. |
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| 04:21.14 | fujin | hmmhesays: You're not talking about Asterisk::Manager are you? That looks entirely useless tbh |
| 04:21.23 | hmmhesays | yeah its a pain in the ass |
| 04:21.36 | hmmhesays | you must be using one of the socket packages if you're getting a connection out of 6 lines |
| 04:21.49 | fujin | I'm not using the AMI to achieve what I'm doing. |
| 04:22.07 | fujin | I felt it was a little excessive |
| 04:22.12 | hmmhesays | indeed |
| 04:22.21 | fujin | but the issue with 'queue show <blah>' truncating the name has forced me to re-evaluate |
| 04:22.23 | jameswf-home | if youremove whitspace you can do it in 1 line |
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| 04:23.56 | fujin | yes, well. |
| 04:24.06 | fujin | argh. I'm so lazy |
| 04:24.21 | fujin | I really don't want to rewrite this with perl/AMI just to get the full queue name |
| 04:26.14 | jameswf-home | patebin your perl |
| 04:26.43 | fujin | My currentl Perl? |
| 04:26.59 | fujin | The perl's not broken. |
| 04:27.00 | jameswf-home | yes maybe we can improve it to work |
| 04:27.08 | fujin | The output of 'queue show', is. |
| 04:27.31 | fujin | moment |
| 04:27.54 | fujin | http://rafb.net/p/sGJTOv40.html |
| 04:28.37 | fujin | (there's nothing wrong with the perl, the problem is with the output of queue show.) |
| 04:28.48 | fujin | is forced to rewrite with AMI |
| 04:30.36 | fujin | ugh |
| 04:30.41 | fujin | hmmhesays: that lib is poorly documented |
| 04:30.58 | hmmhesays | yeah I sure as hell wouldn't use it |
| 04:31.08 | hmmhesays | especially when you can just use socket connect, get your data and get out |
| 04:31.37 | jameswf-home | btw in bash asterisk -rx 'show queues' | head -1 | awk '{print $1}' |
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| 04:33.16 | Kernel_Core | hi all |
| 04:33.26 | Kernel_Core | anybody here worked with speex jitter buffer ? |
| 04:33.27 | fujin | jameswf-home: yes, so? |
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| 04:33.37 | fujin | 'show queues' truncates long queue names |
| 04:33.39 | fujin | *that* is the problem |
| 04:33.49 | fujin | not the parsing implementation |
| 04:35.21 | hmmhesays | change the source to not do that? |
| 04:35.25 | hmmhesays | should be pretty easy |
| 04:35.37 | fujin | That's a little messier than I'd like |
| 04:35.41 | fujin | will just use the AMI |
| 04:35.44 | hmmhesays | how is that messy? |
| 04:35.54 | hmmhesays | I bet its a couple line change maybe |
| 04:36.05 | fujin | because instead of writing an external interface script, I'm actually pissing about with Asterisk |
| 04:36.08 | fujin | and I don't want to be. |
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| 04:39.04 | fujin | hrm |
| 04:39.12 | fujin | might write it in Ruby, with RAMI for the fun of toying with ruby |
| 04:39.15 | fujin | thanks guys. |
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| 04:40.28 | jameswf-home | would it be easier to parse queues.conf for queue names.... |
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| 04:52.40 | rfernandez | hiya!!i have a box with 1.2.23 and we need to patch unicall but we cannot patch it cause the makefile reject our patch how can we fix it? or what package that we need? |
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| 06:15.55 | cmantito | if you're using SIP RealTime with rtcachefriends on, does a SIP reload cause it to re-read the DB so changes made will take effect? |
| 06:17.18 | cmantito | nevermind |
| 06:17.23 | RoyK | += 0xc0ffee |
| 06:17.45 | RoyK | cmantito: sip prune realtime peer xxx |
| 06:17.49 | RoyK | if it's a single peer |
| 06:18.02 | cmantito | hmm, that would probably work better. |
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| 06:18.20 | cmantito | thanks, RoyK |
| 06:21.44 | rfernandez | hiya!!i have a box with 1.2.23 and we need to patch unicall but we cannot patch it cause the makefile reject our patch how can we fix it? or what package that we need? |
| 06:23.34 | CCFL_Man2 | if i have a T1 card originating a CAS T1, do i need to specify CAS as the signaling? |
| 06:23.51 | CCFL_Man2 | the example configs don't |
| 06:24.07 | CCFL_Man2 | they just tell you to configure the channels you want |
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| 06:43.35 | tzafrir | CCFL_Man2, hmm, not really sure. I think that you just use something like signalling = fxo_ls |
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| 06:49.28 | JT | CCFL_Man2: you simply specify the channels |
| 06:49.28 | CCFL_Man2 | tzafrir: ahh, ok |
| 06:49.37 | JT | usually use kewlstart |
| 06:49.37 | CCFL_Man2 | JT: oh, thats easy |
| 06:49.51 | CCFL_Man2 | why wouldn't loopstart work? |
| 06:49.54 | JT | the framing must still be right in zaptel.conf |
| 06:50.01 | CCFL_Man2 | yeah |
| 06:50.02 | JT | it probably would |
| 06:50.12 | JT | but it's a handicapped version of kewlstart |
| 06:50.52 | CCFL_Man2 | will it work with a channel bank? |
| 06:51.04 | JT | sure |
| 06:51.29 | CCFL_Man2 | telcos use loopstart signaling though |
| 06:52.23 | CCFL_Man2 | i guess i'll use loopstart |
| 06:52.29 | CCFL_Man2 | err, kewlstart |
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| 06:59.03 | CCFL_Man2 | of course running asterisk on solaris has to complicate things |
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| 07:00.00 | JT | CCFL_Man2: umm, a lot do kewlstart, it really is just loopstart with extra features |
| 07:03.18 | CCFL_Man2 | oh, ok |
| 07:03.26 | CCFL_Man2 | then i'll definately use it |
| 07:13.34 | CCFL_Man2 | the zaptel installation on my sun box is wrong |
| 07:15.58 | cmantito | how well does * run on solaris anyway? |
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| 07:39.46 | mpwizard | Should I use GSM or Alaw for voicemail prompts? |
| 07:41.16 | JT | mpwizard: for what sort of channel? |
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| 07:43.33 | mpwizard | JT: SIP channels - aLaw |
| 07:43.46 | JT | then alaw |
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| 07:45.43 | mpwizard | JT: I've experienced choppy sound when using aLaw with Asterisk 1.2... |
| 07:46.09 | JT | yeah i've heard of that before |
| 07:46.19 | JT | tried .wav? |
| 07:47.47 | CCFL_Man2 | the "S" sound in human speech is high frequency |
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| 07:55.53 | *** join/#asterisk teapot (n=tandrews@mail.grok.org.za) |
| 07:55.58 | teapot | hi |
| 07:56.53 | teapot | tzafrir: are you here ? |
| 07:56.57 | tzafrir | yes |
| 07:57.03 | teapot | hi tzafrir |
| 07:57.14 | teapot | tzafrir: I have a question or two abour bristuff |
| 07:57.49 | teapot | I noticed that there are bristuff references in the patch for asterisk 1.4.18 |
| 07:58.05 | teapot | does that mean that the asterisk-bristuff deb is obsolete ? |
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| 08:01.59 | tzafrir | yes, debs as of ~1.4.10 are bristuffed |
| 08:02.27 | teapot | ok, so all of kapejod's 0.4.0-test patches in asterisk_1.4.18~dfsg-1.diff.gz |
| 08:02.49 | tzafrir | basically, yes |
| 08:02.52 | teapot | awesome |
| 08:02.56 | tzafrir | not all |
| 08:03.21 | tzafrir | actually in 1.4.18 we started cutting out some stuff that are not really needed and change interface |
| 08:03.39 | teapot | I'm not really using the bristuff mods - I'm just using qozap and the changes to libpri |
| 08:04.05 | tzafrir | all the better for you |
| 08:04.09 | teapot | I have a modified version of qozap, and I just want to move asterisk from 1.2 to 1.4 |
| 08:04.21 | tzafrir | modified? In what way? |
| 08:04.54 | teapot | Well I designed a bri based on the hfc chip, so I used his qozap driver because I like it |
| 08:05.35 | tzafrir | hmm... that driver needs some love and care. specifically, in the way it loads |
| 08:05.46 | tzafrir | did you see the recent post to the bristuff mailing list |
| 08:05.48 | tzafrir | ? |
| 08:05.52 | teapot | no |
| 08:06.02 | teapot | got a url? |
| 08:06.31 | tzafrir | first off: http://bristuff.org/i |
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| 08:07.18 | teapot | looks |
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| 08:21.12 | mpwizard | CCFL_Man2: All sounds are processed with a De-esser. |
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| 08:25.49 | dominic1 | anybody using snom hardware? |
| 08:26.07 | dominic1 | I am using snom 370 and not getting any pickup dialoginfo |
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| 08:33.18 | EvilRick | Hey guys.. has the multihome ip binding issue been sorted out? I am running 1.4.17 and am having problems using multiple interface ip aliases |
| 08:33.37 | *** join/#asterisk viperdude (n=viperdud@87-127-248-176.no-dns-yet.enta.net) |
| 08:36.35 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 08:45.44 | *** join/#asterisk qdk (n=qdk@85.235.253.139) |
| 08:46.58 | *** join/#asterisk wurstdog (n=jeschmi@dhcp-209.varengold.com) |
| 08:47.43 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 08:49.02 | wurstdog | how can i connect digital phones to asterisk? |
| 08:51.30 | wurstdog | which card i need to connect digital phones to asterisk pbx and alcatel office pbx? |
| 09:04.23 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 09:06.30 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 09:12.27 | tzafrir | wundaboy, what type of digital phones? |
| 09:19.54 | *** join/#asterisk shinao1 (n=shinao1@41.222.65.165) |
| 09:25.41 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.193) |
| 09:25.59 | hi365 | where can I "donate" IVR promts? |
| 09:26.18 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 09:26.48 | tzafrir | hi365, what do you mean? |
| 09:26.56 | tzafrir | publish? |
| 09:27.06 | tzafrir | ask to record? |
| 09:27.14 | hi365 | publish |
| 09:27.30 | patrick-- | Hey, im using asterisk 1.4.18 with 3 BN4S0 PCI cards and mISDN channeldriver. I have 4 Outgoing Lines to the PSTN on 2 Ports. How can i Program my dialplan to use the next Port if both Lines on Port 1 are occupied? |
| 09:27.46 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 09:28.12 | tzafrir | patrick--, does misdn have any equivalent to the chan_zap Zap/gNN ? |
| 09:28.15 | hi365 | dial(port1/x/x) |
| 09:28.24 | hi365 | dial(port2/x/x) |
| 09:29.10 | tzafrir | hi365, well, you can start by writing it in the voip-info page of international sound files |
| 09:29.23 | hi365 | it in english |
| 09:29.27 | patrick-- | tzafrir: what does gNN do? |
| 09:29.34 | tzafrir | if you lack the web space, I'm sure some folks will happiily provide you |
| 09:29.53 | tzafrir | dial to the first channel available in group NN |
| 09:30.02 | hi365 | there kinda basic, I though they should be included in the Asterisk distro |
| 09:31.11 | hi365 | Hello, good morning/afternoon/evening. <----------i'm suprised actualy that there not allready in there |
| 09:34.22 | *** part/#asterisk viperdude (n=viperdud@87-127-248-176.no-dns-yet.enta.net) |
| 09:35.24 | tzafrir | well, you can "ask" Alison to record it. I don't recall the exact procedure to get such a recorded sound into the freely-available sounds |
| 09:35.47 | tzafrir | To get into Asterisk it also has to be recorded in French and Spanish |
| 09:36.50 | hi365 | I have it recorded. |
| 09:36.58 | hi365 | http://bugs.digium.com/view.php?id=12360 |
| 09:37.17 | *** join/#asterisk viperdude (n=viperdud@87-127-248-176.no-dns-yet.enta.net) |
| 09:38.02 | *** join/#asterisk znoG (n=gs@host85.190-31-66.telecom.net.ar) |
| 09:44.30 | tzafrir | removed hello.wav |
| 09:45.34 | tzafrir | btw: next time please use zip, tar or whatever.... |
| 09:48.44 | tzafrir | hmm... unrar-free fails with it.... |
| 09:51.41 | wurstdog | tzafrir alcatel4039 |
| 09:57.48 | hi365 | tzafrir: added .zip |
| 09:58.31 | tzafrir | wundaboy, hmm.. I think those only work with alcatel PBXs, right? |
| 09:58.43 | wurstdog | yes |
| 09:59.46 | hi365 | tzafrir: BTW, are you doing any work on the hebrew patch? |
| 10:00.08 | tzafrir | I recall that there was some expensive <digital phones>->sip adapter. But I'm not really sure how much this is worth the bother |
| 10:00.38 | tzafrir | wundaboy, vs. replacing them with SIP phones |
| 10:01.01 | wurstdog | :) |
| 10:01.03 | tzafrir | hi365, have not had the time to look at it for a while. I really need to... |
| 10:03.49 | tzafrir | hi365, hello, good-morning and good-evening have a short "beep" in the end |
| 10:04.41 | hi365 | not at my end (al ta'am v'reiach....) |
| 10:04.44 | tzafrir | I figure it can easily be edited away |
| 10:05.17 | hi365 | if tis there.... |
| 10:05.23 | hi365 | :) |
| 10:10.37 | *** join/#asterisk Koshatul (n=evangeli@ppp188-33.static.internode.on.net) |
| 10:12.11 | *** join/#asterisk _ys (i=yuri@91.151.196.254) |
| 10:14.52 | *** join/#asterisk stimpie (n=stimpie@84-104-6-10.cable.quicknet.nl) |
| 10:15.52 | *** join/#asterisk mikkel (n=mikkel@84-238-113-66.u.parknet.dk) |
| 10:16.17 | hi365 | tzafrir: whats right abour 48kHz? |
| 10:16.25 | hi365 | i though asterisk can only do 8khz? |
| 10:16.44 | tzafrir | it can now also use 16 |
| 10:17.10 | hi365 | doesnt really understand much about all the audio stuff |
| 10:17.35 | tzafrir | 48kHz makes sure you'l lbe able to properly down-sample this to various rates |
| 10:17.45 | hi365 | ah. cool! |
| 10:18.11 | hi365 | btw, if you dont mind my asking, what do you do? |
| 10:18.50 | tzafrir | A programmer at Xorcom |
| 10:19.13 | hi365 | oh right |
| 10:19.16 | hi365 | knew that |
| 10:19.44 | hi365 | is cuco still there? havent seem him in these part in ages! |
| 10:20.28 | hi365 | *parts |
| 10:21.39 | *** part/#asterisk wurstdog (n=jeschmi@dhcp-209.varengold.com) |
| 10:22.11 | tzafrir | no, he's a full-time student |
| 10:22.21 | tzafrir | must be crazy of him |
| 10:22.52 | hi365 | what, to pop in? |
| 10:24.11 | tzafrir | nm |
| 10:33.25 | *** join/#asterisk lvl- (n=lvl@101.248.dhcp.lht.hhs.nl) |
| 10:40.37 | dominic1 | Hi, can anybody tell me why my asterisk is not executing the special extension s? |
| 10:47.22 | *** join/#asterisk lesterc (n=lesterc@193.91.233.220.exetel.com.au) |
| 10:52.16 | *** join/#asterisk teapot (n=tandrews@mail.grok.org.za) |
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| 11:00.29 | dominic1 | Does anybody use the s extension for the internal phones? |
| 11:00.44 | dominic1 | I only get the answer extension not found... |
| 11:02.36 | *** part/#asterisk teapot (n=tandrews@mail.grok.org.za) |
| 11:04.40 | *** join/#asterisk SteveTotaro (n=Administ@pool-151-196-236-69.balt.east.verizon.net) |
| 11:07.44 | Tili | i don't want asterisk to generate CDR while using DISA on asterisk 1.4. NoCDR works fine if call is answered but for other times CDR is still generated |
| 11:08.08 | *** join/#asterisk flujan (n=flujan@internet.nube.com.br) |
| 11:09.23 | dominic1 | Anybody using asterisk and outgoing overlapdial? |
| 11:10.16 | dominic1 | I have a sipphone which can do overlapdial and now I want that my asterisk checks if he can dial the number via isdn |
| 11:11.05 | *** join/#asterisk DagMoller (n=aguirre@unaffiliated/dagmoller) |
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| 11:52.13 | the_5th_wheel | is there a way i can override the tone a caller hears when he calls someone in * ? |
| 11:55.35 | *** join/#asterisk mitcheloc (n=mitchelo@216.207.245.1) |
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| 12:04.02 | *** join/#asterisk tvmkid (n=chatzill@tarpit.linuxense.com) |
| 12:06.17 | EvilRick | Hey guys.. has the multihome ip binding issue been sorted out? I am running 1.4.17 and am having problems using multiple interface ip aliases |
| 12:07.20 | *** join/#asterisk quigon (n=matias@200.61.187.185) |
| 12:08.00 | tvmkid | Does some one have a combined patch of http://lists.digium.com/pipermail/asterisk-dev/2008-January/031545.html and bugno 12127 for 1.4.x? |
| 12:10.32 | tvmkid | Agents in Local channels are not showing correct status, and above two patches seems to fix those bug, but I could not see any combined patch. |
| 12:10.41 | *** join/#asterisk Lawbringer (n=Lawbring@cpc4-rdng12-0-0-cust292.winn.cable.ntl.com) |
| 12:11.15 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 12:11.23 | *** join/#asterisk adjohn (n=adjohn@cpe-72-227-184-198.nyc.res.rr.com) |
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| 12:23.42 | dominic1 | anybody using asterisk and isdn overlapdial for outgoing |
| 12:23.43 | dominic1 | ??? |
| 12:27.42 | *** join/#asterisk drehlecom (i=ircbnc@2001:6f8:1153:2:208:c7ff:feac:d1fb) |
| 12:28.41 | *** join/#asterisk Mavvie (n=edwin@ppp121-44-22-136.lns10.syd7.internode.on.net) |
| 12:29.38 | tzafrir | dominic1, BRI? PRI? |
| 12:29.57 | tzafrir | for an ISDN phone? also: what channe? zap? misdn? |
| 12:30.00 | *** join/#asterisk anonymouz666 (n=anonymou@200.218.193.6) |
| 12:30.19 | tzafrir | (/me uses it with BRI and chan_zap) |
| 12:31.22 | dominic1 | PRI |
| 12:31.51 | anonymouz666 | I have a customer still using Asterisk < 1.0 from the time of linux-support.net complaining about issues. |
| 12:31.52 | anonymouz666 | argh |
| 12:32.16 | *** join/#asterisk coppice (n=chatzill@99.166.17.210.dyn.pacific.net.hk) |
| 12:32.25 | shinao1 | hi.. i have an interesting project i have been saddled with.. i need to be able to configure asterisk with an IVR that users will call to confirm a transaction, and this confirmation will result in an email being generated to ensure the process has accountability. Now, im not sure what components i need.. i expect the asterisk system will need to connect to the client's inhouse telephone system .. im not sure what they use but i expect its a legacy PAB |
| 12:32.26 | shinao1 | X system. I want to know if using a T1/E1 card is of any use here, as i expect a lot of calls to the system and i wouldnt want the asterisk box to be engaged all the time |
| 12:32.45 | dominic1 | I am using chan_misdn and sipphones |
| 12:32.49 | shinao1 | im not really sure what to use |
| 12:33.12 | dominic1 | and I want that asterisk checks everytime I press a new number if it's possible to dial the number |
| 12:33.23 | tzafrir | dominic1, misdn/PRI? |
| 12:33.29 | dominic1 | yes |
| 12:33.52 | tzafrir | The bero.net E1 card? |
| 12:34.15 | ber___ | yes you will use PRI to connect to their system or channelized T1 |
| 12:34.31 | *** part/#asterisk Lawbringer (n=Lawbring@cpc4-rdng12-0-0-cust292.winn.cable.ntl.com) |
| 12:34.37 | *** join/#asterisk waKKu (n=eurulo@unaffiliated/wakku) |
| 12:34.37 | ber___ | once the call gets to you, you will code an AGI script (or at least that is what i would do) |
| 12:34.57 | ber___ | that script will take the IVR digits, verify against whatever database, and fire off the appropriate external processes (email/etc) |
| 12:35.38 | ber___ | ~agi |
| 12:35.39 | jbot | i heard agi is the Asterisk Gateway Interface... similar to CGI for web applications AGI lets you script call control and access databases using your favorite language. AGI wrappers are available for Python (pyst), Perl (astperl?) and other languages |
| 12:35.57 | ber___ | how do you update jbot? |
| 12:36.03 | ber___ | i wanted to add the voip-info url |
| 12:36.58 | ber___ | anyways, here is the URL |
| 12:37.00 | ber___ | http://www.voip-info.org/wiki-Asterisk+AGI |
| 12:37.07 | ber___ | i have used php and perl in the past to code the logic |
| 12:43.01 | *** join/#asterisk b52laptop (n=b52lapto@41.249.250.195) |
| 12:43.03 | b52laptop | hi |
| 12:43.15 | b52laptop | ppl asterisk now is based on php ? |
| 12:43.21 | tzafrir | ~voip-info |
| 12:43.22 | jbot | i guess voip-info is the Voice Over IP wiki. It is a community resource which will answer all of your questions, from Asterisk to ZTDummy. You can find it over at http://www.voip-info.org - well worth bookmarking |
| 12:44.11 | tzafrir | b52laptop, no. the asterisk-gui is LAJ - Linux Asterisk Javascript |
| 12:44.35 | tzafrir | ~agi |
| 12:44.35 | jbot | [agi] the Asterisk Gateway Interface... similar to CGI for web applications AGI lets you script call control and access databases using your favorite language. AGI wrappers are available for Python (pyst), Perl (astperl?) and other languages |
| 12:44.45 | *** join/#asterisk adjohn (n=adjohn@cpe-72-227-184-198.nyc.res.rr.com) |
| 12:44.48 | b52laptop | ah cool pure javascript ingerface ! |
| 12:44.48 | EvilRick | Hey guys.. has the multihome ip binding issue been sorted out? I am running 1.4.17 and am having problems using multiple interface ip aliases |
| 12:44.49 | b52laptop | nice |
| 12:44.53 | b52laptop | tzafrir, thks |
| 12:44.55 | EvilRick | ~multihome |
| 12:45.31 | *** join/#asterisk saftsack (n=Customer@p54A75766.dip.t-dialin.net) |
| 12:45.35 | tzafrir | jbot, AGI is also <reply> See also http://www.voip-info.org/wiki-Asterisk+AGI |
| 12:45.36 | jbot | okay, tzafrir |
| 12:45.38 | tzafrir | ~agi |
| 12:45.39 | jbot | [agi] the Asterisk Gateway Interface... similar to CGI for web applications AGI lets you script call control and access databases using your favorite language. AGI wrappers are available for Python (pyst), Perl (astperl?) and other languages, or <reply> See also http://www.voip-info.org/wiki-Asterisk+AGI |
| 12:45.46 | *** join/#asterisk jm|laptop (n=jm|home@zen.jamiem.com) |
| 12:45.47 | tzafrir | duh |
| 12:45.58 | saftsack | hi, found nothing with the search function on voip-info.org. is it possible to force, that the rtp stream goes over asterisk? |
| 12:46.07 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 12:47.17 | viperdude | dont allow reinvites |
| 12:48.28 | shinao1 | hi.. i have an interesting project i have been saddled with.. i need to be able to configure asterisk with an IVR that users will call to confirm a transaction, and this confirmation will result in an email being generated to ensure the process has accountability. Now, im not sure what components i need.. i expect the asterisk system will need to connect to the client's inhouse telephone system .. im not sure what they use but i expect its a legacy PAB |
| 12:48.28 | shinao1 | X system. I want to know if using a T1/E1 card is of any use here, as i expect a lot of calls to the system and i wouldnt want the asterisk box to be engaged all the time |
| 12:48.30 | shinao1 | im not really sure what to use |
| 12:49.05 | saftsack | canreinvite=no .... this way? |
| 12:50.23 | dominic1 | yes I am using the beronetcard! |
| 12:50.43 | ^shark_ | hi guys i am running etch and i have just used this http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Record to set up festival.But it only prompts me to record my message and that is where is stops, am i missing something in my configs? |
| 12:51.35 | coppice | tzafrir: when is the cutoff for new stuff in Debian? |
| 12:51.44 | *** join/#asterisk ManxPower (n=manxpowe@118.sub-75-203-241.myvzw.com) |
| 12:53.49 | tzafrir | coppice, fluid. The final is in June |
| 12:53.50 | *** join/#asterisk putnopvut (n=putnopvu@216.207.245.1) |
| 12:53.50 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 12:54.07 | *** part/#asterisk b52laptop (n=b52lapto@41.249.250.195) |
| 12:54.15 | coppice | oooh, just like degree exams :-) |
| 12:54.47 | hmmhesays | is fluid a toy story character? |
| 12:55.05 | coppice | no. its what lovers exchange |
| 12:55.06 | tzafrir | I hope we won't start extending it after June like degree exams... |
| 12:55.22 | hmmhesays | no in demolition man |
| 12:55.26 | hmmhesays | s/no/not |
| 12:55.30 | coppice | my degree exams were never extended after June |
| 12:57.43 | *** join/#asterisk viraptor (n=viraptor@awh178.internetdsl.tpnet.pl) |
| 12:59.23 | *** part/#asterisk EvilRick (i=rob@41.207.226.226) |
| 12:59.33 | agx | is there any SIP phones that have a customizable LCD? I mean i can push my own information over it? AFAIK only Polycom (and maybe AAstra support it) |
| 13:00.09 | viperdude | Cisco |
| 13:00.16 | viperdude | XML support |
| 13:00.43 | [TK]D-Fender | Only Cisco IIRC, and only in SCCP |
| 13:01.19 | viperdude | nopeit works in Cisco SIP, we use it ti display call queue info on the screen |
| 13:01.35 | shasta | linksys spa962 has builtin RSS reader - you could use that to show some information |
| 13:01.45 | dominic1 | Does anybody know a example how to use overlapdial for dialing out? I am using 3 digits for all internal numbers anything beginning with 0 should dial out via isdn |
| 13:02.08 | agx | viperdude, ok ty |
| 13:02.12 | *** join/#asterisk jasonwoot (n=jasonrot@user-69-73-40-171.knology.net) |
| 13:03.38 | [TK]D-Fender | viperdude: In push mode? |
| 13:03.55 | [TK]D-Fender | viperdude: Request pushed from the server? |
| 13:03.55 | viperdude | no pull only i think |
| 13:04.16 | [TK]D-Fender | viperdude: I've reead you can truly push with SCCP. |
| 13:04.16 | viperdude | we get page to refresh ever 10 secs etc |
| 13:04.27 | viperdude | SCCP sucks with * |
| 13:04.35 | [TK]D-Fender | viperdude: Polycom is timed poll on idle + demand only. |
| 13:04.43 | agx | viperdude, AAstra support push |
| 13:05.09 | [TK]D-Fender | dominic1: What phones? |
| 13:05.18 | *** join/#asterisk bmg505 (n=leon@196-209-76-146-tbnb-esr-2.dynamic.isadsl.co.za) |
| 13:05.20 | dominic1 | snom370 |
| 13:05.25 | viperdude | I dont have any polycom phones so dont know, and I was not the one asking BTW |
| 13:05.37 | [TK]D-Fender | dominic1: then fix your phone's dialplan. |
| 13:05.54 | [TK]D-Fender | viperdude: Just adding info to the pile.... |
| 13:06.41 | dominic1 | I thought I can do this in asterisk |
| 13:06.46 | viperdude | Aastra phones are nice but crap NAT support no good for my hosted platform |
| 13:07.30 | [TK]D-Fender | dominic1: Your phone controls what it sends to *. |
| 13:07.40 | agx | viperdude, well i've no problem with NAT just looking for a phone that can display how many people are in technical support queue waiting :))) |
| 13:07.57 | [TK]D-Fender | dominic1: For which you typically use an MGCP RFP dialplan like just about every other SIP phone out there. |
| 13:08.07 | [TK]D-Fender | RFC* |
| 13:08.12 | viperdude | agx: wire the agents nads to the phone system and blast them when calls waiting |
| 13:08.43 | [TK]D-Fender | agx: viperdude does it on idle with Cisco, and I do it here for 2 queues, 4 agents on my Polycom's |
| 13:09.10 | viperdude | i have 5 queues 200 agents |
| 13:09.52 | viperdude | goes out for a cig |
| 13:11.47 | *** join/#asterisk jmesquita (n=jmesquit@201.7.117.114) |
| 13:12.02 | dominic1 | Yes, but following problem: I wan to dial a number 0456789. I enter the 0, then asterisk know to dial out via isdn,then asterisk gets 4,5 and 6 after the first four numbers asterisk trys the first time if the number exists, after the 9 he gets a dialtone and the user is bridged to the call |
| 13:12.37 | ManxPower | dominic1: it does not work that way when using SIP |
| 13:13.04 | *** join/#asterisk Katty (n=Katty@hera.copi-rite.com) |
| 13:13.40 | ManxPower | dominic1: Asterisk does not receive the digits one at a time, it receives it from the phone all in one block of digits. |
| 13:13.47 | ManxPower | This is controlled by the phone itself. |
| 13:14.07 | dominic1 | but I can read the digits as dtmf tones right? |
| 13:14.20 | ManxPower | dominic1: not really. |
| 13:14.43 | ManxPower | dominic1: Asterisk receives the dialed digits all at one time. |
| 13:16.01 | dominic1 | If I enable overlapdial on my phone asterisk receives every single digit, I tested that with my phone. The first context matching after the first few digits will be used. |
| 13:16.02 | [TK]D-Fender | dominic1: You're phone does ALL of the thinking before it decides to pass what it considers "complete" to * |
| 13:16.02 | x86 | interesting.... there is a Zap, MD, and a Zapata, Tx |
| 13:16.06 | x86 | had no idea |
| 13:16.24 | *** join/#asterisk linuxmonger (n=stevej@mail.joneslinux.com) |
| 13:16.27 | dominic1 | but after the match I will have to do a read and how can I initiate the isdn checks in background? |
| 13:16.34 | coppice | Dr Zap? :-\ |
| 13:17.12 | ManxPower | We've been having people dialing 9911 and hanging up, which is, of course a problem |
| 13:17.38 | ManxPower | [TK]D-Fender: unless early dial is enabled |
| 13:18.01 | ManxPower | dominic1: Not many phones support early dial, are you using a Grandstream? |
| 13:18.56 | dominic1 | I am using snom and if I am enabling this feature: http://wiki.snom.com/wiki/index.php/Settings/overlap_dialing every single digit is passed to asterisk |
| 13:19.12 | ManxPower | dominic1: great. So what is the problem? |
| 13:19.33 | ManxPower | dominic1: just remember early dial/overlap dialing has been buggy in phones in the past. |
| 13:20.08 | linuxmonger | I've just come upon an interesting bit of hardware, an Adtran TA-908 - it is a router with 2 T1 interfaces, 8 channels of VOIP, and an ethernet port. Does anyone have any scripts to get this to talk to an asterisk server by any chance? |
| 13:20.22 | dominic1 | How can I dial the isdn line in the background till I get the dialtone? The line should indicate if the dialed number is complete |
| 13:20.38 | [TK]D-Fender | dominic1: Just set your phones dialplan and be done with it. |
| 13:21.19 | ManxPower | dominic1: Have you researched how Asterisk does "overlap dial"? |
| 13:21.30 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 13:21.32 | [TK]D-Fender | dominic1: You can't have * constantly ask your ISDN line if the number you're in the middle fo dialing would be complete. * has to accept the call, or not. |
| 13:21.43 | x86 | ManxPower: dude I ripped localcallingguide.com yesterday (and into the night last night) with an automated script and I ended up with almost a million records in my database of unique npa/nxx combinations |
| 13:21.55 | x86 | ManxPower: complete with swtich and OCN info :P |
| 13:22.02 | ManxPower | x86: that does not surprise me. |
| 13:22.19 | dominic1 | I researched that, but I only found some things about incoming calls |
| 13:22.30 | ManxPower | x86: Thank Dog I don't bill for calls. |
| 13:26.14 | hmmhesays | what is "localcallinguide.com" |
| 13:26.38 | linuxmonger | From what I saw, it's troll bait. |
| 13:26.43 | hmmhesays | aye |
| 13:28.19 | *** part/#asterisk dominic1 (n=dob@213.221.82.242) |
| 13:29.09 | x86 | ManxPower: i don't either, just want to know where my users are calling :) |
| 13:30.05 | hmmhesays | Get an nap/nxx list from somewhere |
| 13:30.09 | hmmhesays | *npa even |
| 13:35.51 | x86 | hmmhesays: I had to make my own, since there is not a complete one that's publically available |
| 13:36.11 | *** join/#asterisk ZPertee (n=ZPertee@dhcp164-045.wireless.uakron.edu) |
| 13:36.12 | x86 | hmmhesays: using localcallingguide.com, you can make your own |
| 13:36.26 | davevg-btwtech | x86, you can purchase a list |
| 13:37.51 | x86 | davevg-btwtech: ...or make one for free |
| 13:38.09 | davevg-btwtech | probably violating the sites tos in the process.. |
| 13:38.21 | ZPertee | how do I go about getting information on what kind of telephone line I have connected to my * box? Do telephone customer service reps know things like kewl start, loop start, ground start, etc? |
| 13:38.26 | [TK]D-Fender | lol, I'm calling an ISP here looking for VDSL pricing, and they're running *. Loaded up with custom prompts from Allison.... |
| 13:38.42 | _ShrikE | x86: Better keep that script running, our list updates weekly. |
| 13:38.44 | [TK]D-Fender | ZPertee: "tech support". |
| 13:38.51 | *** join/#asterisk linuxmonger (n=stevej@mail.joneslinux.com) |
| 13:38.54 | *** join/#asterisk tobias (n=tobias@cpe-066-026-085-055.nc.res.rr.com) |
| 13:41.10 | ZPertee | [TK]D-Fender, ok makes sense. |
| 13:42.22 | x86 | _ShrikE: where do you source from? |
| 13:42.36 | _ShrikE | I work for a clec |
| 13:42.55 | x86 | right... where do you get your info from? |
| 13:43.05 | davevg-btwtech | we've purchased our lists from http://www.zipcodeworld.com/ |
| 13:44.15 | x86 | davevg-btwtech: so you're on some kind of subscription system? |
| 13:44.47 | _ShrikE | maponics also has some very detailed npa subcriptions |
| 13:45.02 | davevg-btwtech | yes, it is subscription based since the data changes |
| 13:47.28 | x86 | hmmm zipcodeworld.com is $600/year... that's not terrible |
| 13:48.02 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 13:48.02 | *** mode/#asterisk [+o lmadsen] by ChanServ |
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| 13:50.16 | *** mode/#asterisk [+o anthm] by ChanServ |
| 13:50.43 | _ShrikE | maponics.com is somewhere around that much as well. |
| 13:51.21 | x86 | where localcallingguide.com is free |
| 13:51.32 | x86 | albeit, probably less updated |
| 13:51.42 | x86 | but I don't need to get terribly accurate here |
| 13:55.31 | *** join/#asterisk rupa (i=rupa@gw.rupa.com) |
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| 14:14.38 | Yourname`` | Any _significant_ changes in .19? I use queues. |
| 14:15.22 | [TK]D-Fender | Yourname``: Go read the changelog |
| 14:16.41 | *** join/#asterisk JayTee52 (n=jforde@207-67-84-188.static.twtelecom.net) |
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| 14:22.26 | *** join/#asterisk Telemac (n=telemac@213.223.113.74) |
| 14:22.28 | Telemac | Hello |
| 14:24.32 | Telemac | I'm trying to setup an asterisk to working with isdn and digium b410 card |
| 14:26.02 | *** join/#asterisk macros73 (n=cs@dsl093-063-232.pit1.dsl.speakeasy.net) |
| 14:26.52 | Telemac | With debug I get "P[ 0] Got empty Msg.." message |
| 14:27.14 | Telemac | So I suppose card does not respond |
| 14:27.22 | *** join/#asterisk CunningPike (n=arodgers@204.239.12.183) |
| 14:27.37 | Telemac | ? |
| 14:30.22 | Telemac | How can I check b410 card is properly configured ? |
| 14:31.51 | *** join/#asterisk jayrod422 (n=jayrod42@node2.164.136.64.1dial.com) |
| 14:32.48 | jayrod422 | im trying to get the most calls per second out my asterisk box(s) |
| 14:32.57 | jayrod422 | <PROTECTED> |
| 14:34.23 | [TK]D-Fender | Telemac: Try using it |
| 14:34.27 | davevg-btwtech | jayrod422, load it up with sipp and see for yourself? |
| 14:34.32 | jayrod422 | i did |
| 14:34.42 | jayrod422 | it goes dead around 25-30 calls a sec |
| 14:34.47 | jayrod422 | asterisk just dies |
| 14:35.05 | [TK]D-Fender | jayrod422: And it might be an idea to describe the exact hardware being used. |
| 14:35.59 | jayrod422 | dual core amd opteron 2.6ghz w/ 4gb ram |
| 14:36.01 | Telemac | [TK]D-Fender: It doesn't work, but I'd like something to known what isn't correct :) |
| 14:36.52 | *** join/#asterisk mwalling (i=mwalling@you.dontlike.us) |
| 14:37.28 | [TK]D-Fender | Telemac: pastebin the following : "cat /etc/zaptel.conf" , "cat /etc/asterisk/zapata.conf", "cat /proc/interrupts", and the output of "ztcfg -vvvv" |
| 14:37.30 | [TK]D-Fender | ~pb |
| 14:37.30 | jbot | [~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://www.rafb.net/paste |
| 14:37.31 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^^^ |
| 14:37.58 | ManxPower | [TK]D-Fender: What settings in those files would impact performance? |
| 14:38.12 | jayrod422 | what im am trying to do is build a setup that can do 200 calls a sec with openser and asterisk |
| 14:38.20 | [TK]D-Fender | ManxPower: Pardon? |
| 14:38.32 | jayrod422 | 1.4 dies around 20-30 calls but 1.6 can go to 60 but doesnt recover well from spikes |
| 14:38.36 | [TK]D-Fender | ManxPower: I think you've jsut mixed issues up... |
| 14:38.46 | ManxPower | [TK]D-Fender: sorry, I thought that was for jayrod422, but it was for Telemac |
| 14:38.51 | jayrod422 | lol |
| 14:39.08 | ManxPower | jayrod422: you mean 20-30 calls PER SECOND. |
| 14:39.34 | ManxPower | jayrod422: Asterisk does not handle floods of calls very well. |
| 14:40.49 | jayrod422 | what about a bunch of asterisk boxes with openser doing some sort of load balancing |
| 14:41.37 | ManxPower | jayrod422: that really doesn't impact how Asterisk handles spikes/flood or call setup requests |
| 14:42.05 | jayrod422 | my idea was to have about 10 server with openser doing round robin to each * |
| 14:42.13 | jayrod422 | that way i would distribute the flood |
| 14:42.22 | ManxPower | jayrod422: I wonder why the Wiki pages and google pages about Asterisk scaling and clustering were not helpful to you. |
| 14:42.24 | viperdude | ManxPower: yes it can, depends how you want to handle the floods |
| 14:42.52 | ManxPower | viperdude: no, you are talking about PREVENTING floods, not making Asterisk handle them. |
| 14:42.54 | viperdude | jayrod422: thats what I do to, that being said I have well over 100 ci |
| 14:43.01 | viperdude | concurrent calls |
| 14:43.15 | viperdude | openser can detect the floods and handle accordingly |
| 14:43.30 | ManxPower | viperdude: the number of concurrent calls Asterisk can handle is not really related to how many call setup requests Asterisk can handle per second. |
| 14:43.34 | viperdude | 100 concurrent calls per asterisk box i should have said |
| 14:43.44 | viperdude | ManxPower: yep i know |
| 14:43.51 | jayrod422 | i can do more than 100 concurrent |
| 14:43.55 | jayrod422 | my issue is cps |
| 14:44.04 | jayrod422 | im only doing signaling with my * |
| 14:44.05 | viperdude | i possibly can do more but i have never tried |
| 14:44.44 | viperdude | jayrod422: are you relaying the media RTP? |
| 14:45.02 | jayrod422 | viperdude: no |
| 14:45.05 | viperdude | openser in front is way to go, |
| 14:45.05 | jayrod422 | viperdude: just signaling |
| 14:45.09 | *** join/#asterisk xtr-II (i=94752345@216.19.191.191.novuscom.net) |
| 14:45.12 | jayrod422 | viperdude: i have openser |
| 14:45.14 | viperdude | ok |
| 14:45.19 | viperdude | i relay all media on my boes |
| 14:45.20 | jayrod422 | viperdude: but have to drop to asterisk |
| 14:45.23 | viperdude | boxes |
| 14:45.58 | jayrod422 | viperdude: openser is not a back to back user agent |
| 14:46.05 | viperdude | i know |
| 14:46.12 | jayrod422 | viperdude: we want more control of the call than we can get with openser |
| 14:46.26 | viperdude | i use openser in front of * |
| 14:46.44 | jayrod422 | are you doing dispatching |
| 14:47.02 | *** join/#asterisk waKKu (n=eurulo@unaffiliated/wakku) |
| 14:47.14 | *** part/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
| 14:47.14 | viperdude | i wote a custom app to dispatch the calls based on the load on my media servers |
| 14:47.20 | *** join/#asterisk SteveTotaro (n=Administ@pool-71-166-127-227.bltmmd.east.verizon.net) |
| 14:47.34 | jayrod422 | with openser |
| 14:47.44 | viperdude | yes |
| 14:48.24 | jayrod422 | using the dispatch module or your own stuff? |
| 14:48.43 | *** join/#asterisk susinths (n=susinths@sos3-1x-dhcp065.studby.uio.no) |
| 14:48.46 | *** join/#asterisk shinao1 (n=shinao1@41.222.65.165) |
| 14:49.05 | viperdude | my own stuff |
| 14:50.15 | viperdude | use exec_dest to all the URI based on the output of the application i wrote |
| 14:50.24 | Telemac | [TK]D-Fender: http://pastebin.com/m20e7b2a8, http://pastebin.com/m425d9fcf, http://pastebin.com/m16840b57 |
| 14:51.37 | *** join/#asterisk shinao1 (n=shinao1@41.222.65.165) |
| 14:52.53 | [TK]D-Fender | Telemac: ok, from this point I can no longer help you but perhaps someone else here willbe able to |
| 14:53.47 | *** join/#asterisk shinao1 (n=shinao1@41.222.65.165) |
| 14:55.34 | *** join/#asterisk shinao1 (n=shinao1@41.222.65.165) |
| 14:56.03 | Telemac | ok thx |
| 14:56.39 | *** join/#asterisk nighty^ (n=nighty@p1022-adsau16honb13-acca.tokyo.ocn.ne.jp) |
| 14:57.26 | *** join/#asterisk shinao1 (n=shinao1@41.222.65.165) |
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| 15:00.57 | Yourname`` | [TK]D-Fender: I did. And obviously, I don't understand much. |
| 15:01.47 | [TK]D-Fender | Yourname``: Whats to understand? You want to know what changed. Go read the changelog. If there an item in there you're not cleaar about? |
| 15:01.54 | [TK]D-Fender | Is* |
| 15:01.58 | *** join/#asterisk axisys (n=axisys@155.70.141.45) |
| 15:01.58 | *** join/#asterisk shinao1 (n=shinao1@41.222.65.165) |
| 15:02.07 | Yourname`` | Yeah, I'll pb those to you.. |
| 15:02.09 | Yourname`` | one sec |
| 15:03.42 | *** join/#asterisk shinao1 (n=shinao1@41.222.65.165) |
| 15:05.16 | *** join/#asterisk stonedyak (n=stonedya@firebox.enta.net) |
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| 15:08.02 | *** join/#asterisk shinao1 (n=shinao1@41.222.65.165) |
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| 15:11.47 | *** join/#asterisk bkw___ (n=brian@ppp-70-128-111-181.dsl.tulsok.swbell.net) |
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| 15:14.30 | Katty | hmmhesays: ping? |
| 15:14.42 | hmmhesays | pong |
| 15:15.01 | Katty | hai |
| 15:15.07 | hmmhesays | hello |
| 15:18.55 | *** join/#asterisk doolph (n=doolph@201.218.103.170) |
| 15:22.37 | Telemac | Anyone has successfully configured a b410 to work as isdn card with asterisk ? |
| 15:24.28 | *** join/#asterisk jpeeler (n=jpeeler@216.207.245.1) |
| 15:28.57 | [TK]D-Fender | Telemac: I'd advise calling Digium for support on this personally. Very few people here running those cards |
| 15:34.52 | *** join/#asterisk batoloco80 (n=raul_tor@190.25.228.235) |
| 15:35.16 | batoloco80 | hello |
| 15:35.17 | asteriskmonkey | does the asterisk manager interface (the socket based one) have some form of flood protection that would close down a current connection? I am processing calls using an application that feeds originate commands in via the ami socket and when i put alot of calls into it it closes the connection |
| 15:35.21 | hmmhesays | wow it is a slow day in irc |
| 15:37.01 | *** join/#asterisk shinao1 (n=shinao1@41.222.65.165) |
| 15:38.02 | [TK]D-Fender | asteriskmonkey: Best to use astmanproxy in front. |
| 15:38.21 | [TK]D-Fender | asteriskmonkey: Doesn't handle high connection volumes well |
| 15:39.20 | *** join/#asterisk shinao1 (n=shinao1@41.222.65.165) |
| 15:41.01 | batoloco80 | i would like to register in a database the channel that answer a phone call after passing through a queue |
| 15:41.14 | batoloco80 | how can i make it |
| 15:42.33 | batoloco80 | the idea is to automatically open a browser window with the information of the caller |
| 15:42.59 | batoloco80 | any ideas? |
| 15:44.02 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 15:44.53 | [TK]D-Fender | batoloco80: Write a daemon that will monitor AMI for queue connect messages to see who to pop the screen for. |
| 15:47.32 | rupa | uggh |
| 15:48.21 | rupa | both my bank and my insurance company have this problem. I call them, go throught he IVR and then when they start transfering me to the agent there is a "beep bop" tone. On the second tone, my phone continues the tone continously and I have to hang up. weird |
| 15:49.33 | asteriskmonkey | [TK]D-Fender: thanks, its a single connection dumping originates in though... you saying astman proxy works better? |
| 15:50.06 | [TK]D-Fender | asteriskmonkey: Yeah, it sits in front and maintains a single persistant conenction to * to bypass these issues |
| 15:53.33 | *** join/#asterisk xenonex (n=xenonex@89.218.237.201) |
| 15:53.37 | *** join/#asterisk adjohn (n=adjohn@wifi.javitscenter.com) |
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| 15:57.59 | anonymouz666 | Sysmaster sucky. |
| 15:58.30 | anonymouz666 | and suck more when it uses asterisk version < 1.0 |
| 16:00.18 | CCFL_Man2 | solaris zaptel package installed zaptel in the wrong place |
| 16:00.35 | *** join/#asterisk trippss (n=tripps@c-98-194-65-246.hsd1.tx.comcast.net) |
| 16:01.19 | UnixDog | astrisk 1.0 |
| 16:01.27 | UnixDog | wow dude your like in the past |
| 16:01.41 | UnixDog | come to 2008 |
| 16:01.53 | UnixDog | 1.4.19-beta7.1 |
| 16:02.21 | keith4 | wow Dog you're like illiterate |
| 16:02.41 | UnixDog | today yes |
| 16:02.44 | UnixDog | lol |
| 16:02.51 | UnixDog | been up for 72 hours |
| 16:02.54 | lmadsen | I do all my installs on asterisk 0.9 |
| 16:03.01 | keith4 | hardcore |
| 16:03.06 | UnixDog | and about have zaptel 1.4.9.2 working on bsd |
| 16:04.08 | UnixDog | then I will submit it back to the zaptel-bsd project and get our port updated |
| 16:04.25 | UnixDog | would be nice to have zaptel uptodate |
| 16:04.34 | UnixDog | then I can work on asterisk 1.6 port |
| 16:05.49 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.141) |
| 16:08.09 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 16:08.58 | *** part/#asterisk agx (n=AGX@host63-216-static.34-88-b.business.telecomitalia.it) |
| 16:12.04 | x86 | lmadsen: hahaha |
| 16:12.30 | Jumpie | anyone had good success with the grandstream budget ip phones? :D |
| 16:12.37 | Qwell | ~gs |
| 16:12.37 | jbot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
| 16:12.49 | Jumpie | lol |
| 16:12.52 | Qwell | close enough |
| 16:13.05 | Jumpie | but how do i know thats not just cheerleaders of oher products saying so? :P |
| 16:13.13 | Jumpie | whats a "good" budget phone then? |
| 16:13.17 | Jumpie | i odnt wanna pay $400 a pop |
| 16:13.17 | Qwell | ~phones |
| 16:13.17 | jbot | phones is, like, http://bani.anime.net/phones/. While personal preference will dictate which phone works best for you, general consensus on a rough order of quality and suggestibility is as follows: Polycom (any), Aastra 480i, Aastra 5i Series, Cisco 7940+, Linksys SPA-9XX, Snom, and finally everything else. Do not consider Grandstream phones. Ever. places like such as |
| 16:13.27 | Qwell | only the really expensive Cisco's cost $400.. |
| 16:13.35 | Qwell | there are some damn fine Polycom's for around $130 |
| 16:13.36 | Jumpie | wow, grandstream really on the suck list, glad i caught that |
| 16:13.44 | Jumpie | cool |
| 16:14.03 | Jumpie | i wonder if i can get a deal where i buy 4 i get a deal? or is that not 'bulk' enough hehe |
| 16:14.11 | Qwell | not even close :) |
| 16:14.19 | Jumpie | what about aastra? |
| 16:14.25 | Jumpie | ah i see it |
| 16:14.31 | Qwell | never used an Aastra personally. |
| 16:14.43 | Qwell | I hear good things, but really, pretty much everybody here will recommend Polycom |
| 16:14.53 | Jumpie | cool |
| 16:15.02 | Jumpie | isnt it true you can only use cisco phones with cisco pbx solutions? |
| 16:15.26 | Qwell | the newer cisco phones have SIP firmware, or you can use chan_skinny (which isn't great, but it works) |
| 16:15.39 | codefreeze | Add Cortelco to the bad-as-grandstream list. |
| 16:15.45 | Qwell | codefreeze: really? |
| 16:15.47 | Jumpie | he |
| 16:16.00 | Qwell | looks at the analog coretelco phone sitting next to him |
| 16:16.16 | Qwell | looks at the "Ringer stopped working - 2/2007" note on it, written in black perm marker |
| 16:16.25 | codefreeze | I've had nothing but probs with my sip-cortelco |
| 16:16.25 | Qwell | fair enough |
| 16:17.03 | Jumpie | haha qwell |
| 16:17.14 | Jumpie | ya qwell lookin on voipsupply |
| 16:17.18 | Jumpie | polycom has $115 one |
| 16:17.44 | *** join/#asterisk [hC] (n=hardcore@vpn.voxter.com) |
| 16:18.21 | Qwell | Jumpie: see msg |
| 16:19.01 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 16:21.41 | [TK]D-Fender | Jumpie: http://www.telephonydepot.com/product_p/105-058-320.htm |
| 16:22.10 | Jumpie | appreciated |
| 16:22.22 | Jumpie | oh wow :D |
| 16:23.03 | *** join/#asterisk fedya (n=fedya@75.112.143.226) |
| 16:23.11 | Jumpie | interesting they offer power supply or poe |
| 16:23.20 | UnixDog | no you just dont know how to set them up |
| 16:23.34 | UnixDog | grandstrean gateways and gxp units work fine |
| 16:23.44 | UnixDog | I have setup loads of them with no issues |
| 16:24.19 | UnixDog | and they work with asterisk great |
| 16:24.21 | Jumpie | maybe you got lucky? |
| 16:24.22 | Jumpie | hehe |
| 16:25.02 | UnixDog | no I just know that if you update the firmware on them to the latest they all work |
| 16:25.52 | UnixDog | I have a gxp2000 and a 2010 and a gxw 4104 port fxo |
| 16:26.01 | UnixDog | 4 port |
| 16:26.07 | UnixDog | and it works fine |
| 16:26.40 | [TK]D-Fender | Grandstream has a history of flaky firmware and feel like cheap plastic crap. Their phones offer second rate user interface, audio & build quality |
| 16:27.09 | rupa | just paid $20 for an old seimens that can suposably be flashed with SIP firmware |
| 16:27.14 | UnixDog | I would agree on the older units |
| 16:27.16 | rupa | probably gonna be more trouble than its worth though |
| 16:27.26 | Jumpie | ill be honest, even though im new to asterisk, i am not new to telecom, and i have DEFINATELY heard of and use polycoms before |
| 16:27.35 | UnixDog | but not on the 1201/2010/2020 units |
| 16:27.38 | Jumpie | if i can get a poly on a budget price, and i see tk's link shows as such |
| 16:27.40 | Jumpie | ill probably get that |
| 16:27.40 | UnixDog | they did alot of changes |
| 16:28.07 | *** part/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 16:28.08 | Jumpie | UnixDog for myself it'd probably be ok |
| 16:28.13 | Jumpie | but what about mass distribution for clients? |
| 16:28.18 | Jumpie | thats my concern |
| 16:28.52 | UnixDog | thI have 15 clients all running 1201/2010/2020 units and they are very happy |
| 16:29.03 | UnixDog | I would not give a client a gxp 2000 |
| 16:29.08 | UnixDog | they had major issues |
| 16:29.16 | *** join/#asterisk tvmkid (n=chatzill@202.88.236.180) |
| 16:29.17 | Qwell | "work fine" is nowhere near "is amazing" |
| 16:29.44 | coppice | i get pretty amazed when any voip stuff works |
| 16:29.52 | UnixDog | ok the do all the needed codecs and work great I have had no complaints |
| 16:30.09 | UnixDog | the even support g722 |
| 16:30.11 | Jumpie | lol coppice |
| 16:31.04 | Qwell | g722 on a grandstream?! |
| 16:31.12 | coppice | yep |
| 16:31.12 | ManxPower | What good is G722 when the mic/earpiece/speakerphone is so terrible? |
| 16:31.17 | Qwell | oh god |
| 16:31.34 | ManxPower | That's like putting a porche engine in a ford pinto |
| 16:31.40 | Jumpie | ya i definately want want 'barely acceptable' lol |
| 16:31.46 | UnixDog | not on th 1201/2010/2020 |
| 16:31.52 | UnixDog | you need to go and test them |
| 16:31.53 | Jumpie | survey says: polycom |
| 16:32.02 | UnixDog | I would agree on the older units |
| 16:32.03 | Qwell | ManxPower: it's more like putting a porsche logo on the engine in a pinto |
| 16:32.08 | UnixDog | but not the newer |
| 16:32.28 | ManxPower | UnixDog: Grandstream has had horrible products, horrible firmware for AT LEAST 5 years. Why should I ever give them a chance? |
| 16:32.45 | UnixDog | and if you have issues why not relay them to bill at grandstream |
| 16:32.59 | ManxPower | UnixDog: because I'm not wasting any more money on them. |
| 16:33.07 | ManxPower | Polycom all the way |
| 16:33.12 | UnixDog | how are they to get better unless you point issues out to them |
| 16:33.22 | *** join/#asterisk ZPertee (n=ZPertee@rrcs-24-106-241-121.central.biz.rr.com) |
| 16:33.48 | ManxPower | UnixDog: Honestly? I don't care if Grandstream ever sells another product. Not my problem. |
| 16:35.12 | Jumpie | ManxPower you spoiled my party, i was just about to head out back and put a porsche engine in my pinto |
| 16:35.14 | Jumpie | ;o) |
| 16:35.31 | *** join/#asterisk Defraz (n=t0tal@24-116-152-177.cpe.cableone.net) |
| 16:36.08 | UnixDog | well unless those who get the devices and have issues report them back things will never get better. Its sad that noone helps the companies better their products |
| 16:36.11 | Defraz | I have this dialplan that works great but I am trying to figure out how to go to another server if the one I have it pointed to is offline. |
| 16:36.14 | UnixDog | Polycoms are nice |
| 16:36.36 | UnixDog | they work but some of the functionality still needs more work |
| 16:36.37 | Defraz | I love Polycom and actually Grandstream is stepping up to the plate too. |
| 16:36.45 | Defraz | Yes they need more speed dials. |
| 16:36.50 | UnixDog | I can say the same for snom and aastra |
| 16:36.58 | Defraz | yep aastra is as well. |
| 16:37.05 | plik | even the name "Grandstream" is a bit of a mis-nomer... "pissy-trickle" would be more appropriate |
| 16:37.05 | Defraz | Haven't used snom much. |
| 16:37.13 | UnixDog | speeddial = the directory you build yourself |
| 16:37.41 | [TK]D-Fender | Aastra's soft-keys destroy pretty much everything else out there. Unfortunately the 5i series has serious physical issues, unstable firmware. |
| 16:38.28 | *** join/#asterisk tcseke (n=chatzill@217.20.134.239) |
| 16:39.00 | Jumpie | lol |
| 16:39.08 | ManxPower | UnixDog: It is not the customer's job to beta test products, it's the company's job. |
| 16:39.15 | Defraz | exten=> _2082330110,1,Dial(SIP/2082330110@master-voip-1.fuzecore.com) is my dial plan |
| 16:39.33 | ManxPower | Defraz: that's pretty silly thing to do |
| 16:39.51 | Defraz | if that is off line I want it to go to VoIP-2 |
| 16:40.04 | Defraz | Whey is that. |
| 16:40.05 | ManxPower | Why would you want to dial by hostname and be limited to the [general] options of sip.conf |
| 16:40.17 | [TK]D-Fender | Defraz: exten=> _2082330110,2,Dial(SIP/2082330110@master-voip-2.fuzecore.com) |
| 16:40.41 | Defraz | hmm that didn't work but maybe I missed something. |
| 16:40.42 | ManxPower | Defraz: it's also silly to put a leading _ on a non-pattern extension |
| 16:40.43 | Defraz | I will try that. |
| 16:40.55 | Defraz | oh okay |
| 16:42.28 | Jumpie | what is the _ supposed to signify? |
| 16:42.45 | *** join/#asterisk gaetronik (n=gaetan@200.111.138.170) |
| 16:43.17 | *** join/#asterisk DirtyD (n=DirtyD@ool-18bddaa0.dyn.optonline.net) |
| 16:43.21 | DirtyD | Hi! |
| 16:43.23 | gaetronik | i think i found a bug |
| 16:43.23 | ManxPower | Jumpie: a wildcard pattern |
| 16:43.26 | gaetronik | Hi |
| 16:43.53 | gaetronik | when i use asterisk -r -x "queue show" it does not always give me the same queues |
| 16:43.53 | DirtyD | I'm very confused with single line appearance in asterisk. |
| 16:44.05 | Defraz | I have an Asterisk box off a PRI and I have other servers out there for backup. |
| 16:44.09 | ManxPower | <PROTECTED> |
| 16:44.17 | Jumpie | what is there to match if he has all diits? |
| 16:44.18 | Defraz | My clients have sip trunks to each one for outgoing and incoming |
| 16:44.22 | DirtyD | I have a PRI line that carries in 23 channels. |
| 16:44.24 | Jumpie | ah |
| 16:44.37 | Defraz | So if there is a 1 infront it will work too. |
| 16:44.55 | Defraz | If there is a one or if there isn't a 1 correct. |
| 16:45.00 | DirtyD | But how to I create the trunks that SLA needs from PRI channels? |
| 16:45.04 | Jumpie | so he's explictiely matchin on that number |
| 16:45.16 | Jumpie | i guess thats why you said _ is silly |
| 16:45.22 | ManxPower | DirtyD: PRIs do NOT have numbers assigned to the individual channels |
| 16:45.29 | *** join/#asterisk BZBW (n=wlwzhang@67.110.250.132.ptr.us.xo.net) |
| 16:45.57 | Defraz | Thanks for the help ManxPower |
| 16:46.27 | ManxPower | Defraz: feel free to send a donation via paypal to eric@fnords.org |
| 16:46.43 | Jumpie | lol |
| 16:46.59 | DirtyD | ManxPower: I know. That's why I'm having a problem. The sla stuff in 1.4 seems to want individual Channels ties to trunks. But I can't do it that way.. I need to some how define a DID from the PRI into a trunk. |
| 16:47.27 | ManxPower | Jumpie: I never understood why people thing that's funny (other than the irony that only 1 in 300 people that I help send anything) |
| 16:47.51 | ManxPower | DirtyD: You cannot do what you want to do. |
| 16:48.15 | ManxPower | each PRI trunk has every DID assigned to it. |
| 16:48.15 | Qwell | ManxPower: that many? |
| 16:48.46 | ManxPower | I assume you mean PRI trunk, since there is no such thing as a SIP trunk. |
| 16:49.02 | ManxPower | Qwell: most send less than $5 |
| 16:49.08 | Jumpie | ManxPower i guess because most people are joing when they say it |
| 16:49.12 | Jumpie | by all means you deserve it :D |
| 16:49.41 | ManxPower | If I count only ones that send any significant amount, I'd guess it would be closer than 1 in 1,000 or 1 in 2,000 |
| 16:49.48 | DirtyD | Manx: So then, how do I assign a button on all of my SIP phones that will represent a single DID coming in over the PRI? |
| 16:49.53 | *** join/#asterisk shinao1 (n=shinao1@smtp.gtbank.com) |
| 16:49.56 | ManxPower | I'm just glad I don't rely on Asterisk to make a living. |
| 16:50.36 | ManxPower | DirtyD: You can't (if I understand you correctly) |
| 16:50.38 | ZPertee | when I try to dial out I get an alarm in the middle of the callout process and then asterisk hangs up thus not allowing me to make outgoing calls. incoming works just fine though |
| 16:50.58 | ManxPower | DirtyD: I assume you want the button to show if the pri channel is in use. |
| 16:51.09 | [TK]D-Fender | DirtyD: What phone? And you would normally use a dedication registration for this. |
| 16:51.23 | ManxPower | ZPertee: the extensive discussions about this problem on the mailing lists was not helpful" |
| 16:51.46 | ManxPower | DirtyD: if you just want the correct callerid number to show up when dialing out, that's trivial |
| 16:52.17 | [TK]D-Fender | DirtyD: Indeed, what you are asking for makes NO sense. |
| 16:52.17 | ZPertee | ManxPower, how do I get to the mailing list, search in google and trust them to search them all for me or what? |
| 16:52.27 | [TK]D-Fender | DirtyD: SLA is for channels, not DID's |
| 16:52.31 | ManxPower | ZPertee: google does not have recent stuff |
| 16:52.46 | DirtyD | DirtyD: Not the channel, but the number DID. When I call is placed to a specific DID (feed to my box through a PRI) I want to be able to ring all my SIP phones. SNOM 370 |
| 16:52.50 | [TK]D-Fender | DirtyD: And it is completely impractical if all but the smallest analog deployments. |
| 16:52.57 | ManxPower | ZPertee: if you are not subscribed to the mailing list you are missing out on some of the best support resources for asterisk. |
| 16:53.17 | [TK]D-Fender | DirtyD: * does not do SLA. What we have is an ugly hack pretending to be SLA |
| 16:53.54 | ManxPower | DirtyD: exten => 5045551212,1,Dial(SIP/peer1&SIP/peer2&SIP/peer3) This is not rocket science and this should have been covered in The Book. Were you not able to find info about this in The Book? |
| 16:54.10 | ManxPower | where 5045551212 is one of your DIDs |
| 16:54.38 | Jumpie | is kinda starting to figure out this syntax |
| 16:54.40 | Jumpie | :P |
| 16:54.56 | ZPertee | ManxPower, yeah I can see your point but if I'm not planning on working on asterisk forever and so eventually it just becomes spam filling my mailbox, slowing me down from getting the important stuff. |
| 16:54.57 | [TK]D-Fender | ManxPower: Dno't dual-dial servers like that... ick |
| 16:55.14 | ManxPower | [TK]D-Fender: Huh? |
| 16:55.22 | ManxPower | I was multiple dialing SIP phones |
| 16:55.39 | [TK]D-Fender | ManxPower: My turn to mix up issues :) |
| 16:55.42 | DirtyD | Manx: Right, that will ring all the phones... BUT when someone answers, I'd like the a dedicated button to stay lit on all the phones to indicate to the rest of the team that there is an active call. I'd also like the other folks to be able to pick up this call if they need to. |
| 16:56.04 | [TK]D-Fender | DirtyD: Go install this : |
| 16:56.06 | [TK]D-Fender | ~devstate |
| 16:56.07 | jbot | [~devstate] Devstate is an Asterisk 1.4 module for custom BLF device state, see the following link -=- http://svncommunity.digium.com/svn/russell/asterisk-1.4/func_devstate-1.4/ |
| 16:56.16 | ManxPower | DirtyD: why don't you get it working without those advanced, hard to configure options first? |
| 16:56.20 | *** join/#asterisk qdk (n=qdk@0x573fe57e.bynqu2.broadband.tele.dk) |
| 16:56.20 | [TK]D-Fender | DirtyD: And get ready to do some fugly hacking |
| 16:56.38 | DirtyD | DirtyD: It is working. |
| 16:56.42 | DirtyD | err |
| 16:56.45 | DirtyD | manx. |
| 16:57.14 | ManxPower | DirtyD: You CANNOT make buttons light when a specific DID is active. You CAN make buttons light up when a specific phone is in use. |
| 16:57.28 | [TK]D-Fender | ManxPower: There are ways. |
| 16:57.48 | ManxPower | [TK]D-Fender: thanks for volunteering to help DirtyD. I'm off to run errands now. |
| 16:57.55 | ManxPower | is away |
| 16:57.56 | DirtyD | The SNOM 360/70 can do it, I just have to figure out how. |
| 16:58.03 | [TK]D-Fender | DirtyD: Go install devstate, use a GroupCount in your inbound processing and on count > 0 light it up, on 0, off... |
| 16:58.11 | Jumpie | haha |
| 16:58.30 | [TK]D-Fender | DirtyD: My way is all you've got. Period. |
| 16:58.55 | [TK]D-Fender | DirtyD: (devstate as a base) |
| 16:58.56 | Jumpie | so you are trying to tie specific lines on your phones to individual did DirtyD? |
| 16:59.10 | Jumpie | for incoming notification |
| 16:59.21 | tvmkid | hi room, i am trying to get queues to recognize correct agent states when using Local channles and AddQueueMember |
| 16:59.28 | *** join/#asterisk javar (n=javar@69.79.134.24) |
| 16:59.48 | tvmkid | I have 2 patches which address this issue |
| 16:59.54 | tvmkid | I got them from |
| 17:00.08 | tvmkid | http://lists.digium.com/pipermail/asterisk-dev/2008-January/031579.html |
| 17:00.10 | tvmkid | and |
| 17:00.17 | tvmkid | bug 12127 |
| 17:00.45 | tvmkid | is there any one who had combined and made them work for 1.4.x ? |
| 17:01.17 | tvmkid | just looking for a tested patch if possible ... |
| 17:01.39 | tvmkid | patch 12127 does not apply cleanely on 1.4.17 |
| 17:02.04 | *** join/#asterisk variable_office (n=variable@cerberus.iswan.net) |
| 17:02.19 | variable_office | can asterisk be the network side of a pri? |
| 17:02.54 | Qwell | variable_office: absolutely |
| 17:03.15 | Nugget | with enough muffins anything is possible. |
| 17:03.21 | UnixDog | I am on with bill at GS now |
| 17:03.34 | Jumpie | lol |
| 17:03.39 | Qwell | UnixDog: fix #1; make the phones not cheap plastic |
| 17:03.47 | Qwell | put a little weight in the handset |
| 17:04.02 | Jumpie | gotta save them benjis |
| 17:04.41 | variable_office | Qwell, would it be possible to bring calls into an asterisk box as a channelized T1 and bring them out as a PRI? |
| 17:04.48 | Qwell | it should be like a roll of quarters |
| 17:04.55 | Qwell | variable_office: umm, sure |
| 17:05.35 | coppice | If GS hired one of a hundred decent plastics designers working within in a km of their place in shenzhen, people would have a whole different impression of them |
| 17:05.59 | variable_office | Qwell, did what I said not make sense? |
| 17:06.08 | bkw_ | coppice: they seem to fix firmware bugs but not in any rapid pace |
| 17:06.41 | bkw_ | UnixDog: don't be mean to bill... he owes me a beer |
| 17:06.42 | Qwell | variable_office: no, I understood. just an odd request. it's possible |
| 17:06.53 | coppice | GS believe deeply in conservation the add and remove bugs at the same pace. |
| 17:07.04 | bkw_ | coppice: hehe |
| 17:07.33 | variable_office | Qwell, do the sangoma cards accept channelized t1? the other possibility over channelized t1 is gr303, can asterisk take gr303 links? |
| 17:07.44 | Jumpie | woot gr303 |
| 17:08.12 | Jumpie | gr303 is how most fios elements are deployed |
| 17:08.12 | ManxPower | Isn't GR303 very PRIish? |
| 17:08.15 | coppice | who would build a T1 that can't work channelized? :-\ |
| 17:08.21 | Jumpie | with some older tr08 that needs to be scrapped |
| 17:09.05 | Jumpie | as far as i know it, the class 5 was tagged as these trunks are gr303 capable |
| 17:09.10 | Jumpie | but it was up to the terminating end to provision it as such |
| 17:09.50 | variable_office | Jumpie, do you know if asterisk can take a set of gr303 links? |
| 17:10.01 | Jumpie | dude im here cause ima a noob |
| 17:10.08 | Jumpie | im more of a transport guy with verizon |
| 17:10.28 | *** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk) |
| 17:10.45 | Jumpie | but like i said, i didnt think a link was nessecarily tagged as a 'gr 303 link' |
| 17:10.48 | *** join/#asterisk john8675309tm (n=chatzill@216.51.145.1) |
| 17:10.49 | Jumpie | more that it was 'capable' |
| 17:10.53 | Jumpie | and your end point sets it up that way |
| 17:11.08 | Jumpie | but im also not a 5e/dms guru |
| 17:12.11 | Jumpie | we had elements that had tr08 and gr303 |
| 17:12.18 | Jumpie | on the same device |
| 17:13.02 | variable_office | humm, ya, time to go through some mailing list archives |
| 17:13.26 | Jumpie | so you are trying to make the asterisk box look like telco |
| 17:13.52 | variable_office | essentially. I need to somehow get from gr303 to pri |
| 17:13.57 | variable_office | I can do gr303 to ct1 |
| 17:14.08 | variable_office | but the to pri part has me confused |
| 17:15.15 | gaetronik | it's a bit off topic, but anyone has yet bought things on voipsupply site |
| 17:16.24 | Jumpie | heh |
| 17:16.27 | Jumpie | i defer |
| 17:16.41 | ManxPower | variable_office: there won't be a 1-to-1 relationship between the PRI channels and the CT1 channels |
| 17:17.00 | Juggie | ManxPower, you could make one. |
| 17:17.23 | *** join/#asterisk xenonex (n=xenonex@92.47.14.6) |
| 17:17.30 | Juggie | the wiki says gr303 is supported. |
| 17:18.50 | Jumpie | man i need some more shit i can do lab |
| 17:18.54 | Jumpie | but i am not gettin a pri to my house |
| 17:19.14 | UnixDog | ok just got of the phone with bill and he is going to try and get a rep in the channel to get feed back |
| 17:20.07 | ManxPower | Jumpie: you need to do more reading of config file samples when you upgrade. GR30# has been supported since 1.2 |
| 17:20.25 | Jumpie | ManxPower |
| 17:20.28 | Jumpie | i never said it wasnt supported |
| 17:20.39 | Jumpie | thats DirtyD's issue :D |
| 17:20.46 | gaetronik | no one? |
| 17:21.40 | *** join/#asterisk nny_2 (n=Scott_My@64.203.239.83) |
| 17:22.43 | nny_2 | so wav49 isn't supported in say, the iphone or the treo, and WAV files are too uncompressed for cellular email delivery, mp3 seems like the alternative, but mp3 support for the system is a slippery slope, any other options? |
| 17:23.18 | Juggie | wonders when mp3's got slippery |
| 17:23.29 | *** join/#asterisk mocker (n=kyle@mocker.org) |
| 17:23.47 | Qwell | nny_2: actually, wav49 should work now. |
| 17:23.50 | Qwell | there was a bug that was fixed like last week |
| 17:23.57 | nny_2 | hmm actually voip-info doesn't list mp3 as an option for voicemail.conf |
| 17:24.02 | nny_2 | Qwell: hmmm? |
| 17:24.11 | Qwell | nny_2: try it out |
| 17:24.13 | nny_2 | Qwell: which version of asterisk? |
| 17:24.17 | Qwell | 1.4+ |
| 17:25.11 | nny_2 | Qwell: do i need to recompile? running 1.4.17 atm on our in house system |
| 17:25.20 | Qwell | 1.4.19 should have the fix |
| 17:25.22 | ZPertee | [TK]D-Fender, I had asked you the other day about the signalling on my pots line. you had told me that it was probably kewl. I just got off of the phone will the phone company and they told me that it was loopstart. do I want to get this changed to kewlstart or will this work? |
| 17:25.26 | Qwell | erm, no |
| 17:25.26 | *** join/#asterisk hugorebelo (n=hugorebe@200-171-132-124.completo.com.br) |
| 17:25.29 | Qwell | wait, yes |
| 17:25.50 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 17:25.51 | nny_2 | Qwell: doing a fresh compile now with "latest" release.. will try it and report back, thanks |
| 17:25.53 | [TK]D-Fender | ZPertee: "Will" it work? More like "DOES" it work? |
| 17:26.17 | [TK]D-Fender | ZPertee: And ask them for Kewlstart,a nd to enable CDS as I suggested |
| 17:27.49 | *** join/#asterisk jmesquita (n=jmesquit@201.7.117.114) |
| 17:28.02 | ZPertee | [TK]D-Fender, If I am in midst of replacing an Avaya PBX I would probably want to wait until asterisk ready to go before hooking up? |
| 17:28.36 | [TK]D-Fender | ZPertee: Go deal with your liens. |
| 17:28.39 | [TK]D-Fender | lines* |
| 17:29.23 | ZPertee | [TK]D-Fender, you don't think it will mess up the other pbx if I do it before I am ready? The last thing I want to do Is take down the phone lines |
| 17:29.35 | ZPertee | before I am ready |
| 17:29.37 | [TK]D-Fender | ZPertee: Doubt it'll make any difference |
| 17:31.29 | ZPertee | [TK]D-Fender, ok sounds good. this page made me worry http://kb.digium.com/entry/44/30/ "Most real phone switches, and almost no PBX's (except Asterisk, of course) support this feature." |
| 17:32.38 | *** join/#asterisk hi365_w (n=hi365@mail.pcgeula.co.il) |
| 17:33.00 | *** join/#asterisk lucho1981 (n=el_luise@domain01.commlogik.com.co) |
| 17:33.13 | lucho1981 | Hi .. |
| 17:33.18 | hi365_w | anyone having problems with stanahone? |
| 17:33.32 | lucho1981 | I need to upgrade the firmware version of my AA50 appliance .. |
| 17:33.40 | lucho1981 | do you know how can I get the new version..? |
| 17:34.15 | [TK]D-Fender | lucho1981: www.digium.com |
| 17:34.16 | lucho1981 | I've been trying on" https://be.digium.com/downloads" |
| 17:34.52 | lucho1981 | thanks Fender, but I've been trying on " https://be.digium.com/downloads" and it just shows an error .. |
| 17:35.09 | lucho1981 | do you know any other link? may be a FTP address? |
| 17:36.02 | [TK]D-Fender | lucho1981: Call them |
| 17:36.17 | lucho1981 | ok .. ! thanks .. |
| 17:37.15 | *** join/#asterisk [hC] (n=hardcore@vpn.voxter.com) |
| 17:38.11 | *** join/#asterisk xenonex (n=xenonex@89.218.237.201) |
| 17:38.21 | Juggie | [hC], hiho |
| 17:38.27 | [hC] | So all of a sudden calls coming in from link2voip are being hung up on as soon as someone answers the phone, with a 'maximum retries on transmission <xxx> for seqno 102', then hanging up call, no reply to our critical packet. |
| 17:38.50 | [TK]D-Fender | [hC]: Sounds like a nat/reinvite error |
| 17:40.12 | [hC] | Yeah.. Strange thing is it was working yesterday. Maybe i'll try focing reinvites to off... odd |
| 17:40.19 | *** join/#asterisk backblue (n=igor@82.102.1.42) |
| 17:40.25 | backblue | hi, any snom guru? |
| 17:42.19 | [hC] | Juggie: hey mang... sup |
| 17:42.28 | [hC] | argh, forcing reinvites to no didnt help either. |
| 17:44.27 | Juggie | [hC], do a sip debug |
| 17:50.00 | [hC] | Ah, looks like link2voip is dicking with stuff |
| 17:50.03 | [hC] | now the numbers are all busy |
| 17:50.10 | [hC] | crappy ITSPs man.. |
| 17:50.38 | *** join/#asterisk Tourinho (n=tourinho@201.37.118.16) |
| 17:51.05 | variable_office | ManxPower, sorry I left for a minute, what do you mean it would not be 1:1 just because one is 23 channels and the other 24? |
| 17:51.11 | Tourinho | hello all.. is there a way for asterisk to get Audio IP (remote) ? |
| 17:51.35 | Jumpie | lol |
| 17:56.43 | keith4 | backblue: i have a snom, does that count? |
| 17:57.53 | lmadsen | tzanger: ping |
| 17:57.59 | lmadsen | tzanger: what is the name of 151 Front? |
| 17:58.05 | lmadsen | Juggie: tzanger would probably know |
| 17:58.07 | tzanger | uh |
| 17:58.11 | tzanger | it's just called "151 Front" |
| 17:58.13 | backblue | keith4: do you have ldap integration in the phone? |
| 17:58.29 | lmadsen | ya, that's what I thought :) |
| 17:58.36 | lmadsen | I didn't know it by any other name, heh |
| 17:59.03 | *** join/#asterisk r0d3nt (n=astrutt@backup.realprohosting.com) |
| 17:59.31 | keith4 | backblue: nope |
| 17:59.43 | backblue | keith4: forget then! :D |
| 17:59.55 | *** join/#asterisk adjohn (n=adjohn@12.177.160.114) |
| 18:09.06 | *** join/#asterisk Yourname` (n=chatzill@unaffiliated/yourname/x-837320) |
| 18:09.45 | asteriskmonkey | any plans to support video conferencing in asterisk? |
| 18:10.15 | x86 | it's not supported? |
| 18:10.28 | x86 | I was under the impression videosupport=yes in sip.conf meant something |
| 18:10.39 | Daviey | naa, it's a tease] |
| 18:10.44 | Qwell | conferencing |
| 18:10.44 | x86 | ah |
| 18:10.53 | Qwell | and...it's quite difficult |
| 18:10.59 | Qwell | especially when you don't support video transcoding |
| 18:11.16 | Qwell | there have been some ideas floating about though, that wouldn't require too much work (in theory) |
| 18:12.41 | x86 | Qwell: why don't you just allow same codecs? no transcoding involved |
| 18:13.30 | Qwell | how are we going to mix video if we can't read it? |
| 18:13.53 | Qwell | most video conferencing systems will show all (or a few) videos at once on the screen |
| 18:14.00 | Daviey | x86: you mean mix with rtp? |
| 18:14.07 | Daviey | ahh |
| 18:14.26 | Qwell | we would have to be able to shrink the video, make a new stream (in their codec) with all of the streams mixed...meh. it's a nightmare |
| 18:14.28 | x86 | Qwell: I was thinking more like a one-on-one conference :P |
| 18:14.39 | Qwell | that isn't a conference then, is it? |
| 18:14.50 | x86 | Qwell: guess not :p |
| 18:14.56 | Qwell | that already works |
| 18:15.03 | the_5th_wheel | can anyone help me with a singapore number? Im tryng to figure out wht i need to tell my employees to call to get to my asterisk server, while they are in singapore |
| 18:15.26 | x86 | the_5th_wheel: calling an american number? |
| 18:16.05 | Strom_C | usually when giving people international-format numbers, you just tell them + (country code) (number) |
| 18:16.20 | the_5th_wheel | no, i setup one of those IAX trunks from voicetrading, which gives me a singapore number |
| 18:16.27 | backblue | how do i compile ilbc support in asterisk 1.2 now? |
| 18:16.41 | the_5th_wheel | now, i have it in international format, but i have no idea what they must dial there. |
| 18:16.58 | x86 | and your users don't know how to call a singapore number _from_ singapore? |
| 18:17.05 | cmantito | anyone know of a SIP call originator I can port my US number to? AT&T are being tits about my semi-permenant divert. |
| 18:17.18 | x86 | backblue: 1.2 is no longer supported |
| 18:17.35 | x86 | cmantito: telasip |
| 18:17.46 | cmantito | danke sir; |
| 18:17.48 | backblue | x86: i still use 1.2, and i need ilbc support. |
| 18:18.06 | backblue | i think digium have removed ilbc support |
| 18:18.19 | coppice | nobody *needs* ilbc support. |
| 18:18.47 | backblue | and why not? |
| 18:19.25 | coppice | what does only ilbc and offers no reasonable alternative? |
| 18:19.45 | *** join/#asterisk angom (n=angom@201.170.65.143) |
| 18:19.49 | the_5th_wheel | x86: yeah. the first two cant phone the number. app when they put in the int dialing codes, they fail |
| 18:20.10 | the_5th_wheel | x86: its sales people there for work.... |
| 18:20.26 | Strom_C | the_5th_wheel: tell them to call the operator or look in a telephone directory or to just bloody well ASK someone there |
| 18:20.46 | Strom_C | you know, all the usual ways everyone figures out how to dial numbers in strange places |
| 18:20.50 | *** join/#asterisk lanning (n=lanning@ip67-152-85-190.z85-152-67.customer.algx.net) |
| 18:21.13 | the_5th_wheel | Strom_C: they are being lazy and running up our cell bills |
| 18:21.50 | *** join/#asterisk SteveTotaro (n=Administ@pool-151-196-236-69.balt.east.verizon.net) |
| 18:21.51 | Strom_C | "You will figure out how to use this number or we will fire your ass" |
| 18:22.22 | the_5th_wheel | alas, i can only do that with my minions. Sales im not allowed to touch.... |
| 18:22.34 | *** join/#asterisk ddunavant (n=David@75.145.240.14) |
| 18:23.06 | Strom_C | oh hey look |
| 18:23.26 | Strom_C | look what I found on the first page of Yahoo! search results for "singapore telephone dialing" |
| 18:23.28 | Strom_C | http://en.wikipedia.org/wiki/Singapore_telephone_numbering_plan |
| 18:23.37 | Strom_C | look look look look look |
| 18:23.58 | Strom_C | (pointing) |
| 18:26.28 | the_5th_wheel | lables the boss as someone who cant use google. |
| 18:26.36 | the_5th_wheel | She told me she googled for an hour.... |
| 18:27.10 | Strom_C | and you didn't try? |
| 18:28.24 | the_5th_wheel | I was tired. I Beg thy forgiveness |
| 18:30.02 | Strom_C | sigh. |
| 18:31.00 | Tourinho | guys.. is there a way to get remote audio IP? |
| 18:33.53 | ZPertee | If I have a pots line that I want to be able to dial out on but not receive any calls whatis the correct way to do this put it in a fake context or what? |
| 18:37.06 | *** join/#asterisk tiav (n=tiav@ram94-3-82-225-11-215.fbx.proxad.net) |
| 18:38.00 | *** join/#asterisk vladtser1 (n=Vladimir@adsl-065-015-017-142.sip.mia.bellsouth.net) |
| 18:39.04 | vladtser1 | i know this should be easy to resolve...I just setup a new phone at a remote location using my * server at home. when I make a call, I can hear them but they cannot hear me. what should I check first |
| 18:39.07 | *** join/#asterisk Rico29 (n=Rico@ARennes-358-1-85-250.w90-54.abo.wanadoo.fr) |
| 18:39.26 | vladtser1 | the same model phone works perfectly on my LAN, but not from a remote location |
| 18:41.49 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 18:42.06 | nny_2 | vladtser1: you need to have the asterisk server on the WAN or use STUN |
| 18:42.35 | *** join/#asterisk Tuxofred (n=Fred@ip-80-236-192-102.dsl.scarlet.be) |
| 18:42.38 | nny_2 | vladtser1: http://www.asteriskguru.com/tutorials/sip_nat_oneway_or_no_audio_asterisk.html |
| 18:43.05 | vladtser1 | nny_2: thanks...I'll check it out now |
| 18:44.02 | *** join/#asterisk adjohn (n=adjohn@wifi.javitscenter.com) |
| 18:44.31 | nny_2 | vladtser1: no problem, FWIW i use clarkconnect (Centos based) as my gateway device and asterisk box for this very reason |
| 18:44.56 | *** join/#asterisk nirz (n=nir@89-139-56-81.bb.netvision.net.il) |
| 18:45.04 | nny_2 | CC also has Intrusion prevention, nice web interface, etc |
| 18:45.52 | vladtser1 | nny_2: I'm still new to this.... I see the stun setting in the phone app...do I set the stun server as the ip address of my gateway box on my LAN? |
| 18:46.52 | Jumpie | nny_2 so any time i try to point a remote sip phone to my box, even with port forwarding, it'll have that problem without using stun? |
| 18:46.55 | nny_2 | vladtser1: you will need a STUN server out in the open, i have not used one personally, |
| 18:46.59 | nny_2 | yes |
| 18:47.03 | nny_2 | afaik that is |
| 18:47.03 | Jumpie | grr |
| 18:47.10 | vladtser1 | nny_2: I'm running FreeBSD servers, my LAN connects through one of these servers and it is my firewall |
| 18:47.18 | Jumpie | i dont understand if you provide the right port forwarding |
| 18:47.31 | Jumpie | i remmber someone else talking about this |
| 18:47.55 | *** join/#asterisk ceL_ (i=cel@CPE-75-81-232-46.we.res.rr.com) |
| 18:48.02 | nny_2 | Jumpie: I am not clear as to why myself, reading now |
| 18:48.14 | Jumpie | k..im gonna check out that site too |
| 18:48.41 | [TK]D-Fender | vladtser1: Read up : |
| 18:48.43 | [TK]D-Fender | ~sipnat |
| 18:48.43 | jbot | [~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 |
| 18:48.46 | vladtser1 | I have the right port forwarding...I can make & receive calls from inside my LAN perfectly...but this time I'm at my office and I want to use my GPX2000 phone from outside my LAN |
| 18:48.55 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 18:48.55 | *** mode/#asterisk [+o anthm] by ChanServ |
| 18:49.28 | Jumpie | vladtser1 im glad you brought this up, i wa about to try the same thing saturday |
| 18:49.30 | Jumpie | thanks tk |
| 18:50.26 | vladtser1 | please excuse my noobie status...is a stun server running a particular app, or is it just a name * uses for my gateway which is running NATD and forwarding packets |
| 18:53.29 | *** join/#asterisk hfb (n=hfb@pool-71-118-254-245.lsanca.dsl-w.verizon.net) |
| 18:55.23 | vladtser1 | [TK]D-Fender: my sip seems to have all the necessary setting described at http://www.aocomputing.net/?p=3 |
| 18:59.33 | ceL_ | if an isp was wanting to become a voip provider but basically just relay through another provider and buy phone numbers in bulk from that provider is this possible or is there any software that lets you do this? |
| 19:00.36 | Jumpie | vladtser1 http://sourceforge.net/projects/stun/ |
| 19:01.02 | vladtser1 | i think this DSL connection at my office is too slow |
| 19:01.04 | vladtser1 | WARNING[10056]: chan_sip.c:1949 retrans_pkt: Maximum retries exceeded on transmission 8fa178a75faacbb1@192.168.1.43 for seqno 56071 (Critical Response) |
| 19:01.19 | *** join/#asterisk The_X (i=chris@true.fiberpimp.net) |
| 19:01.29 | The_X | hi folks |
| 19:01.50 | The_X | anyone ever setup an adtran TA9xx to using the DSX port? |
| 19:07.02 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 19:07.16 | Jumpie | lol complicated |
| 19:07.24 | Jumpie | so a stun box sits on the outside he |
| 19:10.18 | [TK]D-Fender | vladtser1: pastebin your sip.conf masking only passwords |
| 19:10.20 | [TK]D-Fender | ~pb |
| 19:10.21 | jbot | [~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://www.rafb.net/paste |
| 19:11.04 | *** join/#asterisk SteveTotaro (n=Administ@pool-151-196-236-69.balt.east.verizon.net) |
| 19:11.33 | Jumpie | our problem seems relatd to http://www.asteriskguru.com/tutorials/sip_nat_oneway_or_no_audio_asterisk.html number 1.4.6 |
| 19:11.52 | Jumpie | it says you just need port forwarding and the sip.conf setup correctly with externip and localnet |
| 19:14.20 | [TK]D-Fender | vladtser1: Just show us what you're doing.... |
| 19:17.25 | vladtser1 | [TK]D-Fender: stand by.... |
| 19:17.39 | *** join/#asterisk RoyK (n=roy@cnbokcafe.uio.no) |
| 19:18.50 | *** join/#asterisk riddlebox (n=riddlebo@75-128-170-26.static.stls.mo.charter.com) |
| 19:19.50 | riddlebox | is there any tricks to getting a snom phone to work with asterisk, I have it registering, but when I make a call it says Call from '' ? |
| 19:20.33 | vladtser1 | [TK]D-Fender: http://pastebin.ca/969841 |
| 19:21.39 | vladtser1 | 2000 is the phone on the LAN and it works fine...so does the other phone with four extensions that's on the LAN. I left it out for expediency |
| 19:21.57 | vladtser1 | 3000 and 3001 is the phone at my office. |
| 19:23.01 | [TK]D-Fender | vladtser1: You are missing "nat=yes" under [general] |
| 19:23.19 | [TK]D-Fender | vladtser1: And should have "canreinvite=no" there too |
| 19:23.39 | vladtser1 | BTW - I did this once from another office using X-Lite and it worked |
| 19:24.10 | [TK]D-Fender | vladtser1: Because X-Lite does some smarter stuff of its own. |
| 19:24.12 | vladtser1 | do I need nat=yes under general even if I have it under the individual phone extensions |
| 19:24.20 | *** join/#asterisk avr4you (n=marcus@mail.koenig-datentechnik.de) |
| 19:24.29 | vladtser1 | I see said the blind man... |
| 19:24.38 | [TK]D-Fender | vladtser1: Yes, because under [general] you are telling * itself that it is behind NAT |
| 19:24.51 | vladtser1 | stand by...will test it now |
| 19:25.07 | [TK]D-Fender | vladtser1: under phone entries that tell* whether it shound trust the IP in the SIP header vs the packet header |
| 19:25.21 | [TK]D-Fender | vladtser1: Under which case you deifnitely need both |
| 19:25.49 | Jumpie | putting nat=yes under general will still not affect any lan based extensions whatsoever tho right tk? |
| 19:26.00 | [TK]D-Fender | Jumpie: Nope |
| 19:27.40 | Jumpie | hmm tryin to setup my port forwarding for rtp |
| 19:27.48 | Jumpie | is it 10000-20000 as source or dest |
| 19:28.13 | [TK]D-Fender | Jumpie: dest |
| 19:28.34 | *** join/#asterisk l2cache (n=l2cache@m4e5e36d0.tmodns.net) |
| 19:28.38 | riddlebox | nm on my snom phone issue it is a problem with going with ports not being forwarded |
| 19:29.11 | vladtser1 | [TK]D-Fender: no good...still can only hear on the IP Phone...the callee cannot hear me. |
| 19:29.33 | vladtser1 | oh excuse me,...I left off the can reinvite..let me try that too |
| 19:29.45 | [TK]D-Fender | vladtser1: What are you forwarding on the * side exactly? Also you should not be forwarding on the REMOTE side |
| 19:31.15 | Jumpie | can reinvite vladtser1 is that on the per extension portion? |
| 19:31.25 | vladtser1 | I have no control on the remote side...its a firewall appliance that only our IT guy has access too and if I knew I was setting up an IP phone he would shit-a-brick |
| 19:31.52 | Jumpie | hahah |
| 19:32.26 | vladtser1 | [TK]D-Fender: if I set canreinvite = yes in general does that affect the canreinvite in the extensions, which are all set to no |
| 19:33.01 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177585424.dsl.bell.ca) |
| 19:34.20 | *** join/#asterisk avr4you (n=marcus@mail.koenig-datentechnik.de) |
| 19:35.10 | vladtser1 | [TK]D-Fender: Maximum retries exceeded on transmission ans then the call hangs up...I think it's this remote side DSL that's either too slow or noisy |
| 19:35.22 | vladtser1 | can reinvite did not help |
| 19:35.35 | [TK]D-Fender | vladtser1: Just do it. Also note that you will have real problems with some D-Link routers, Cisco PIX, and a few others. You need to ensure your router is also not doing any kind of NAT transform for SIP. |
| 19:35.44 | vladtser1 | I'm the only one in the building right now so it shouldn't be too much traffic on our Internet connection |
| 19:36.11 | vladtser1 | it's in my general section now...should I omit it from the extension for the remote? |
| 19:36.41 | Jumpie | not nat transform, but definatley forwarding right |
| 19:37.29 | avr4you | WHO |
| 19:37.57 | Jumpie | lol vladtser1 im trying same thing with friend in norway |
| 19:39.17 | vladtser1 | i just hacked into our router...the dumb IT guy never changes the password...it has something called pinholes which I'm looking into now |
| 19:39.46 | vladtser1 | it's a Netopia appliance |
| 19:40.07 | Jumpie | vladtser1 i have same problem |
| 19:40.10 | Jumpie | call is connected |
| 19:40.24 | *** join/#asterisk jameswf-home (n=james@dsl093-157-131.phx1.dsl.speakeasy.net) |
| 19:40.41 | vladtser1 | which ports do I forward for sip 5060 thru 5080? |
| 19:40.52 | Juggie | no |
| 19:41.06 | Juggie | 5060 and whatever is configured in rtp.conf |
| 19:41.07 | Daviey | 5060 and rtp.confstated ports |
| 19:41.31 | jameswf-home | wee wee weee r t peeeeeeeeee |
| 19:41.32 | *** join/#asterisk lvl- (n=lvl@101.248.dhcp.lht.hhs.nl) |
| 19:42.26 | Jumpie | i did 5060 |
| 19:42.29 | Jumpie | and 10000-20000 |
| 19:42.35 | Jumpie | in xlite it sayhs 'not transmitting your voice' |
| 19:42.45 | Jumpie | nm |
| 19:43.04 | vladtser1 | [TK]D-Fender: it's asking for External Port Start and External Port End...that I know but it wants a single internal port...can you offer advice? |
| 19:43.16 | Jumpie | you should abve able to do a range |
| 19:43.54 | *** join/#asterisk s0lid (n=s0lid@210.213.198.56) |
| 19:44.55 | vladtser1 | it doesn't ask for a range for internal...only external...I'm leaving it at zero for now and will simple give it the internal IP address of the phone |
| 19:45.11 | Jumpie | i dont think thats wha tyou need to do |
| 19:45.21 | vladtser1 | is rtp 10000-20000 |
| 19:46.00 | Jumpie | yeah |
| 19:46.03 | vladtser1 | someone once told me to do 8000 to 20001 |
| 19:46.08 | *** join/#asterisk dssman (n=no@209.8.41.68) |
| 19:46.12 | Jumpie | strange i see the incoming call, but on remote end he says call failed sip error |
| 19:46.30 | dssman | where does asterisk store the log files... I randomly started dropping all my calls today :( |
| 19:46.57 | *** join/#asterisk tobias (n=tobias@cpe-066-026-085-055.nc.res.rr.com) |
| 19:48.28 | vladtser1 | dssman: /var/log/asterisk, also watch the CLI screen |
| 19:49.30 | *** join/#asterisk Katty (n=Katty@hera.copi-rite.com) |
| 19:49.48 | dssman | thanks... do you know how to jump to the obttom of a file in vi? |
| 19:50.21 | Katty | OHAI |
| 19:50.25 | Katty | i upgraded your pbx. |
| 19:51.06 | cpm | dssman, G |
| 19:52.14 | cpm | parks you at the beginning of the last line, which is where I think you want to be |
| 19:52.34 | Yourname` | vladtser1: I think that someone said that to you because Xlite/eyeBeam by default start their RTP from 8000. And Asterisk starts from 10000. |
| 19:52.42 | dssman | hmm |
| 19:52.49 | dssman | yea Ive got nothing that looks bad |
| 19:52.49 | *** join/#asterisk hohum (n=dcorbe@99-203-119-108.area1.spcsdns.net) |
| 19:53.01 | dssman | peers reachable |
| 19:53.05 | dssman | a regs timeout / retry |
| 19:53.43 | rupa | [TK]D-Fender, btw, doing the two levels of Dial() with the macro on the second level seems to be working much better. haven't had a chance to fully test, but the earlier problems I was having don't seem to be present... |
| 19:53.46 | Katty | file: ping |
| 19:53.54 | Qwell | Katty: proxypong |
| 19:54.00 | file | moo |
| 19:54.07 | Katty | file: hewwoes. |
| 19:54.07 | Katty | hugs file |
| 19:54.12 | file | tickle hugs Katty |
| 19:57.10 | Katty | so. |
| 19:57.24 | Katty | if i wanted to start asterisk on boot...pause about 5 seconds, and then startup two things that use manager.conf |
| 19:57.36 | Katty | i'd be writing that in bash, right? and linking it to rc.local |
| 19:57.50 | *** join/#asterisk shido6 (n=shido6@204.126.120.132) |
| 19:58.04 | Katty | :> |
| 19:58.06 | Katty | hugs shido6 |
| 20:02.38 | jayrod422 | anyone ever use sipp |
| 20:02.46 | *** join/#asterisk comprookie2000 (n=david@adsl-065-012-210-216.sip.bct.bellsouth.net) |
| 20:02.46 | dssman | yee |
| 20:03.20 | outtolunc | katty: your 'scripts' could be in anything you wanted, but yes, just use rc.local to load asterisk, sleep, load first script, sleep or not, load next script |
| 20:03.22 | jayrod422 | im trying to load test a sipp->openser->asterisk(s)-sipp setup and sipp keeps going to crap |
| 20:03.35 | jayrod422 | works fine on sipp->*->sipp |
| 20:04.00 | ZPertee | apparently there is some kind of problem with Zaptel 1.4.9.2 my tdm808b wouldn't work with it and I had to get a special release by getting a hold of digium technical support |
| 20:04.49 | Katty | outtolunc: if a put a lil somethin somethin together, would you give it a look see and tell me if i'm doing it right? i've never written a bash script before. |
| 20:05.03 | outtolunc | if i have time sure |
| 20:05.10 | Katty | kk |
| 20:05.22 | dssman | no idea :S |
| 20:07.32 | Yourname` | Does anyone know how to forcefully update eyeBeam? It used to ask for updating before, but clicked on "Dont notify me again" and clicked on Reject, lol.. and now change of mind and actually want to update. |
| 20:08.15 | outtolunc | wonders if you run out of gas often also? <G> |
| 20:08.44 | Yourname` | Wimmen get so much done in chatrooms so easy. |
| 20:09.15 | Yourname` | thinks of changing nick from this genderneutral one to something more gender specific. |
| 20:09.26 | outtolunc | there has to be a config/reg entry for the 'no update' |
| 20:09.35 | outtolunc | find it, and remove it or set to 0 |
| 20:11.24 | outtolunc | or you could attempt to uninstall an reinstall (see if they truly clean up after themselves) |
| 20:11.32 | asteriskmonkey | what file is mysql-addons looking for to qualify the mysql driver? Im running ubuntu and have everything install :/ |
| 20:11.47 | Yourname` | I was thinking of it, but that looks even more complicated to do all over the place. |
| 20:12.09 | gaetronik | is it possible to have a dialplan commad which execute in background |
| 20:12.22 | bitzero | I'm trying to find more information on Jabber integraton (in specific, I'm looking for what so many other people are looking for: shared presence notification. I want jabber users to appear as 'on the phone' when they are.) but I'm having a hell of a time finding any MEANINGFUL documentation on "asterisk as a jabber module". other than that its possible as of 1.4 and isn't that just wonderful. |
| 20:12.42 | gaetronik | i mean whic is executed, but without waiting for its completion to execute te commands with next priority |
| 20:12.54 | outtolunc | control + g = check for updates |
| 20:13.29 | Yourname` | outtolunc: That for me? |
| 20:13.46 | outtolunc | gee, anyone else asking about updates <G> |
| 20:14.03 | outtolunc | someone way in the back, raise your hand please |
| 20:14.15 | Yourname` | LOL |
| 20:14.30 | Yourname` | Well, I tried ctrl+G and that didn't work, so I thought you weren't talking to me. |
| 20:14.44 | Yourname` | Ctrl+G on eyeBeam/ |
| 20:15.30 | [TK]D-Fender | rupa: Excellent |
| 20:16.06 | Katty | outtolunc: if i wanted to run moo.pl /located/here |
| 20:16.18 | ZPertee | anyone able to get supervised disconnect on a verizon business line. they claim they have never heard of kewlstart or supervised disconnect. was wondering if anyone else was successful with verizon |
| 20:16.21 | Katty | outtolunc: do i do /located/here/moo.pl in rc.local? |
| 20:16.54 | rupa | [TK]D-Fender, actually, I take that back, I need to look at the log when I get a chance, but the call still supervices as soon as any extension picks up the call (rather than after call disposition is determined) |
| 20:17.12 | Katty | outtolunc: presuming of course i do ./moo.pl to run it manually. |
| 20:17.56 | bitzero | Jabber anyone? |
| 20:18.28 | Yourname` | #jabber |
| 20:19.02 | bitzero | Jabber/Asterisk presence integration anyone? |
| 20:19.15 | outtolunc | katty: yes, always path the app/script |
| 20:19.24 | outtolunc | and no |
| 20:19.31 | outtolunc | . is only for in dir |
| 20:19.47 | outtolunc | so just /path/to/script args(if any) |
| 20:20.01 | outtolunc | and if args, you might need to quote/bt |
| 20:20.40 | outtolunc | an example |
| 20:21.08 | outtolunc | would be to start asterisk with safe_asterisk from rc.local would be /usr/sbin/safe_asterisk |
| 20:21.50 | gaetronik | i use a NoOP for execting an ODBC query |
| 20:21.59 | gaetronik | is this the regular way to do this |
| 20:26.57 | *** join/#asterisk barros (n=barros@189-19-24-162.dsl.telesp.net.br) |
| 20:28.10 | barros | hello guys.. I've been out of asterisk progress for some time so I'm not aware of new features of 1.6 version.. so, is there any news about T.38 termination on this new major release? |
| 20:28.32 | *** join/#asterisk SteveTotaro (n=Administ@pool-151-196-236-69.balt.east.verizon.net) |
| 20:32.14 | *** join/#asterisk fskrotzki (n=fskrotzk@host.textwise.com) |
| 20:36.00 | rupa | ok, this should be a dirt simple question. Read() on timeout clears thvariable. I was going to do a simple: Set(ACCEPT = $[ ${ACCEPT} | 2 ]) |
| 20:36.00 | rupa | <PROTECTED> |
| 20:37.23 | rupa | paste: http://pastebin.ca/970069 |
| 20:38.24 | rupa | guess I can test for LEN |
| 20:40.33 | Jumpie | is it possible to create a 'catch all 'extension that somebody doesn't really "log into' but then therefore dials X real esxtensions? |
| 20:40.40 | Jumpie | i.e. if you dial 1000, 2000, 3000, and 4000 will ring |
| 20:40.45 | Jumpie | but nobody is really at 1000 |
| 20:41.05 | ZPertee | Jumpie, _XXXX |
| 20:41.36 | ZPertee | Jumpie, pattern matching is the ticket |
| 20:41.40 | *** join/#asterisk DirtyD (n=DirtyD@ool-18bddaa0.dyn.optonline.net) |
| 20:41.43 | ZPertee | ~thebook |
| 20:41.44 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com |
| 20:41.45 | DirtyD | Hi. |
| 20:41.53 | DirtyD | I |
| 20:42.05 | Jumpie | ZPertee but what's the pattern |
| 20:42.14 | Jumpie | 1, 2, 3, and 4 have nothin in common |
| 20:42.27 | Jumpie | ZPertee ya i have that book open in pdf now :D |
| 20:42.44 | DirtyD | I've been running 1.2 for a while now.. no problems.. I just upgraded to 1.4.19.. and my machine keeps locking up completely.. The only think I see is a do_IRQ: error on the console. |
| 20:42.45 | Jumpie | i have seen questions about pattenr matching before, but i guess im not getting how that is implemented |
| 20:43.06 | DirtyD | I've got two Zaptel devices.. |
| 20:43.28 | *** join/#asterisk draygon (n=draygon@208.76.99.5) |
| 20:43.39 | Yourname` | outtolunc: Uninstall/reinstall didn't work out.. : |
| 20:44.17 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 20:44.28 | bitzero | ZPertee: I think you're over complicating Jumpie's question. |
| 20:45.10 | DirtyD | The TDM2400P and a TE200P any one know why I'd be seeing getting do_IRQ: and locking up my machine? |
| 20:45.30 | Jumpie | ya maybe im asking it more complicated |
| 20:45.38 | Jumpie | i just want a 'group' extension i guess that isnt a real person |
| 20:45.50 | ZPertee | bitzero, simplify it then |
| 20:46.11 | bitzero | Doing so outside of the channel so as not to flood everyone with booring shit. |
| 20:46.17 | Jumpie | lolz |
| 20:46.34 | ZPertee | bitzero, much appreciated |
| 20:46.49 | DirtyD | do_IRQ: stack overflow |
| 20:46.50 | *** join/#asterisk fujin (n=aj@unaffiliated/fujin) |
| 20:47.06 | fujin | Anyone know if there is a way to programatically get queue names without the AMI? |
| 20:47.18 | fujin | (parsing out show queue <name> / show queues is truncating the queue name) |
| 20:53.54 | *** join/#asterisk VaNNi (n=VaNNi@lgb-static-216.70.165.200.mpowercom.net) |
| 20:54.25 | hohum | is asterisk capable of playing mp3 files as prompts? |
| 20:54.35 | hohum | if now how do I convert an mp3 file to like a .gsm file |
| 20:55.01 | keith4 | sox |
| 20:55.30 | mvanbaak | or use asterisk if you have the mp3 codec installed |
| 20:57.31 | ZPertee | hohum, the nice thing about using gsm is that it takes less cpu power to play. I guess it comes down to whether hard drive space (use mp3) or processor load (use gsm) is more important to you. |
| 20:58.25 | hohum | ZPertee: if I can convert the files to G711 it would take barely any CPU power at all to play because asterisk wouldn't need to transcode the files |
| 20:58.53 | ZPertee | hohum, correct |
| 20:59.02 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 20:59.22 | hohum | does sox understand g711? |
| 21:01.30 | fujin | yes |
| 21:01.32 | fujin | it does |
| 21:03.42 | asteriskmonkey | can you read/insert new realtime data using ami ? |
| 21:05.32 | fujin | realtime data? |
| 21:05.34 | fujin | realtime data is SQL. |
| 21:05.41 | fujin | that'd be dumb |
| 21:05.56 | *** join/#asterisk flujan (n=flujan@internet.nube.com.br) |
| 21:07.28 | flujan | hello all. I am having a strange problem with asterisk. |
| 21:08.08 | flujan | I just install asterisk on a box... created some peers... but the authentication is not working... i configure a exten on sip.conf but cannot log on the machine... very weird... any ideas? |
| 21:08.43 | flujan | sip debug show NOTHING on the machine... |
| 21:09.43 | gaetronik | flujan, any firewall |
| 21:09.46 | gaetronik | ? |
| 21:10.08 | flujan | no, no firewall. the machines are on the same network... 192.168.1.15 asterisk and 192.168.1.10 the peer. |
| 21:10.32 | flujan | gaetronik: any ideas how can I debug it? |
| 21:11.06 | gaetronik | have you done the reload after creating the peers |
| 21:11.37 | doolph | hello |
| 21:13.36 | flujan | gaetronik: yeap, even reboot the machine... |
| 21:13.42 | flujan | doolph: hello |
| 21:13.54 | gaetronik | what client do you use |
| 21:13.55 | flujan | i just install asterisk, without libpri and stuff |
| 21:13.55 | gaetronik | ? |
| 21:13.56 | doolph | hi flujan |
| 21:13.58 | flujan | eyebeam |
| 21:14.20 | flujan | it is not a network problem, since i can log in on another box(production) |
| 21:14.28 | *** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.pa.comcast.net) |
| 21:14.35 | gaetronik | ok |
| 21:14.51 | gaetronik | you've just changed the host in the conf of eyebeam? |
| 21:15.48 | doolph | what's your problem? |
| 21:17.38 | flujan | gaetronik: yeap |
| 21:17.57 | flujan | doolph: I installed asterisk on a box, cannot make it work... a simple peer cannot authenticate on the machine |
| 21:18.34 | doolph | any firewall? |
| 21:18.35 | flujan | it is not a network problem, i installed a webserver on the machine and it is working flawless... |
| 21:18.38 | flujan | doolph: no |
| 21:19.03 | doolph | did you check the configuration? |
| 21:19.20 | flujan | doolph: i enabled the sip debug and it is printing nothing... |
| 21:19.36 | flujan | it is like asterisk is not receiving the packets nor sending them |
| 21:19.48 | gaetronik | did you asterisk bind the port |
| 21:19.50 | flujan | ~book |
| 21:19.51 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 21:19.56 | gaetronik | netstat -l -n | grep 5060 |
| 21:20.53 | flujan | gaetronik: reinstalling asterisk |
| 21:21.10 | gaetronik | it's not a windows service |
| 21:21.42 | *** join/#asterisk codazoda (n=codazoda@c-67-161-216-115.hsd1.ut.comcast.net) |
| 21:21.45 | flujan | gaetronik: yeap, but shit... this error is driving me crazy |
| 21:22.37 | gaetronik | take care of compiling chan_sip |
| 21:23.08 | codazoda | My asterisk system doesn't register rings in the console until after the first ring, it doesn't answer the call until the second ring, even though the first thing I do is "answer()". Using Dialogic TDM404B cars. Is this normal behavior with these cards, or might I be able to get it working faster? |
| 21:23.29 | Qwell | codazoda: disable callerid |
| 21:23.47 | Qwell | it waits until it gets the callerid feed before ringing |
| 21:23.56 | codazoda | Ah. Makes sense. I'll do that. |
| 21:26.14 | flujan | gaetronik: udp 0 0 0.0.0.0:5060 0.0.0.0:* |
| 21:26.22 | flujan | so asterisk is working with it |
| 21:26.25 | gaetronik | yeap |
| 21:26.53 | flujan | wtf what is happening them, I have sure that the softphone/hardphone is working... also tried it on another machine... |
| 21:27.00 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 21:28.16 | codazoda | Qwell: Do you mean to do this on the lines themselves, or in a config option in Asterisk? A couple quick searches are not turning up the option. |
| 21:28.29 | flujan | gaetronik: any other idea, it is not working at all.. . |
| 21:28.33 | codazoda | *Turn off caller id. |
| 21:28.58 | flujan | gaetronik: I just removed and reinstalled asterisk twice... will try version 1.4.19 |
| 21:29.05 | Qwell | codazoda: zapata.conf |
| 21:29.30 | gaetronik | flujan, tcpdump -netttti eth0 -p udp |
| 21:29.38 | gaetronik | and look for what is going on |
| 21:29.42 | flujan | http://pastie.caboo.se/174881 |
| 21:31.36 | flujan | gaetronik: http://pastie.caboo.se/174881 |
| 21:32.56 | gaetronik | notihing interesting |
| 21:33.14 | *** join/#asterisk tobias (n=tobias@user-0ce2hpk.cable.mindspring.com) |
| 21:33.19 | *** join/#asterisk Tako-san (n=jmkiffia@24.108.192.144) |
| 21:34.07 | gaetronik | do a tcpdump -netttti eth0 udp and port 5060 |
| 21:34.16 | gaetronik | and try connecting |
| 21:34.33 | gaetronik | I've got to go to a meeting sorry |
| 21:37.08 | *** part/#asterisk nny_2 (n=Scott_My@64.203.239.83) |
| 21:38.08 | *** join/#asterisk susinths (n=susinths@sos3-1x-dhcp065.studby.uio.no) |
| 21:39.18 | *** join/#asterisk floppp (n=flopp@nat-staff.b3g-telecom.com) |
| 21:39.38 | DirtyD | ugg: do_IRQ: stack overflow: 448 |
| 21:39.44 | *** join/#asterisk RoyK (n=roy@ip-184-48-149-91.dialup.ice.no) |
| 21:40.33 | *** part/#asterisk RoyK (n=roy@ip-184-48-149-91.dialup.ice.no) |
| 21:40.52 | flujan | gaetronik: ok thanks |
| 21:42.48 | flujan | gaetronik: http://pastie.caboo.se/174881 |
| 21:48.34 | *** join/#asterisk nirz (n=nir@89-139-36-49.bb.netvision.net.il) |
| 21:48.38 | *** join/#asterisk generalhan (n=asd@ip67-90-64-2.z64-90-67.customer.algx.net) |
| 21:48.54 | generalhan | hey all ! |
| 21:50.06 | flujan | hello |
| 21:50.12 | generalhan | i need some info on how to enable DTMF on my TDM card ... i thought that was something that just worked by default. but im trying to get my fax server working behind asterisk, and their techs just keep telling me i need to "enable DTMF on my FXS ports" |
| 21:50.36 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177583046.dsl.bell.ca) |
| 22:07.43 | codazoda | Asterisk is not answering my calls until after the 2nd ring. I've set usecallerid=no in zapata.conf. "answer()" is the first thing in my context. Anything else I can do to make it faster? |
| 22:10.53 | Qwell | did you restart asterisk? |
| 22:10.54 | rupa | ok, I almost have the skeleton of my call screening working, but I have one big problem. If the callee picks up the extension and then hangs up then both legs of the call are hung up. Can I trap that hangup in the macro and set MACRO_RESULT=BUSY? |
| 22:11.28 | codazoda | Never mind. Typo. "usercallerid". |
| 22:11.40 | codazoda | That works now. Thanks for the help. |
| 22:12.48 | *** join/#asterisk macli (n=macli@nmc.brc.ubc.ca) |
| 22:13.09 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177581733.dsl.bell.ca) |
| 22:14.58 | *** join/#asterisk eric_hill (i=EricHill@204.94.175.11) |
| 22:16.07 | *** part/#asterisk flujan (n=flujan@internet.nube.com.br) |
| 22:17.15 | *** join/#asterisk lesterc (n=lesterc@vl10.gw.ok-labs.com) |
| 22:19.18 | *** join/#asterisk xenonex (n=xenonex@92.47.14.6) |
| 22:19.58 | *** join/#asterisk coil (i=scott@cpe-67-10-243-97.satx.res.rr.com) |
| 22:20.22 | generalhan | when you pass DTMF digits in a dial command, should the caller be able to hear the digits being sent ? |
| 22:20.33 | Qwell | generalhan: it depends |
| 22:21.03 | generalhan | im trying a Dial(ZAP/50,30,D(4333)) and when i dial that extension i just hear a mean echo and thats it |
| 22:22.28 | generalhan | the techs for my fax server are telling me that i need to send the DTMF digits in order for the server to recognize the number for the routing rules. all they told me i needed to do was "enable DTMF" on my TDM card. |
| 22:22.47 | generalhan | so i tried it by just passing the digits in the dial command, and its still not working |
| 22:23.57 | generalhan | Qwell: depends on what ?" |
| 22:27.43 | *** part/#asterisk coil (i=scott@cpe-67-10-243-97.satx.res.rr.com) |
| 22:31.47 | *** join/#asterisk bartoloco80 (n=raul_tor@190.25.228.235) |
| 22:31.52 | bartoloco80 | hello |
| 22:32.12 | bartoloco80 | please help me with a question |
| 22:32.58 | bartoloco80 | how can i know if a channel is being used after calling dial() |
| 22:33.56 | bartoloco80 | i need to know if a channel was answered in the exact moment when the user pick up the phone |
| 22:34.56 | bartoloco80 | i've used the channel status variable but it always set on unknown |
| 22:35.14 | *** join/#asterisk Toerkeium (i=oo@201.216.206.221) |
| 22:36.11 | *** part/#asterisk codazoda (n=codazoda@c-67-161-216-115.hsd1.ut.comcast.net) |
| 22:38.05 | bartoloco80 | i need some help |
| 22:40.12 | outtolunc | then rephrase the question as 'if something answered' it would be bridged to that call |
| 22:40.43 | outtolunc | (that leg anyways) |
| 22:41.24 | bartoloco80 | any ideas? |
| 22:42.02 | outtolunc | yes i have many, first one is to suggest you read what i said and REPHRASE the question |
| 22:42.24 | outtolunc | nevermind, not in the mood for this |
| 22:42.55 | bartoloco80 | ok |
| 22:43.00 | *** join/#asterisk rupa (i=rupa@gw.rupa.com) |
| 22:45.26 | *** join/#asterisk boster (n=boster@chi-pat.cashnetusa.com) |
| 22:46.27 | bartoloco80 | i need to store in a database the channel name after the dial appication only if the call is answered and before the call finish |
| 22:46.54 | bartoloco80 | can you tell me how to do it |
| 22:47.09 | rupa | is there a way to capture if the callee hangs up in a Dial() macro and do something *other* than having the macro "exit non-zero" and propogate all the way up and kill the Dial()? |
| 22:49.53 | *** join/#asterisk jblack (n=jblack@pool-72-79-189-156.sctnpa.east.verizon.net) |
| 22:50.56 | boster | If I use a sip phone, dial into an asterisk box, use agent login, and then pull the power plug on the phone, the agent is still logged in. even after the sip peer becomes unreachable, the phone powers back up, becomes reachable again, and tries to log back in, the "agent" is still logged in. THis presents two problems, 1. the human agent can't log in, as the machine thinks they are already in, and 2. the dead channel "agent" accepts calls from |
| 22:50.57 | boster | <PROTECTED> |
| 22:51.17 | boster | any ideas/workarounds/config changes I can make to fix this? |
| 22:51.27 | *** join/#asterisk Katty (n=The@adsl-68-92-251-223.dsl.stlsmo.swbell.net) |
| 22:51.59 | outtolunc | suggest removing all 'digits' (fingers) from the agents that are yanking out the power cords |
| 22:52.10 | boster | I know it sounds like a stupid, unlikely occurrence to happen, but with callcenter reps and softphones, they do it on a fairly frequent basis |
| 22:52.34 | Katty | wibbles |
| 22:52.39 | Katty | today was tooooo busy. |
| 22:52.41 | outtolunc | then you setup an exten they can call to 'clear' that hung login |
| 22:52.41 | Katty | far too busy. |
| 22:52.47 | *** join/#asterisk craigk (n=craigk@58.174.150.119) |
| 22:52.53 | Katty | it's good to be home (= |
| 22:53.15 | boster | outtolunc like an exten that does a system "agent logoff XXXX" command? |
| 22:53.48 | outtolunc | or focus on the channel |
| 22:56.05 | Katty | everyone having a good night? |
| 22:56.24 | boster | well that's another thing. the channel shouldn't exist, right? it's a 1 ended channel because the other side has rebooted, or is still powered off/unavailable |
| 22:58.11 | outtolunc | because 'something bad happened' it will need a watchgod/timer/timeout to cleanup after it |
| 22:58.19 | outtolunc | er dog |
| 22:58.22 | *** join/#asterisk Faithful (n=Faithful@vg106.vodafone.com.au) |
| 22:58.37 | boster | right, I'm just a bit surprised that suck a thing doesn't already exist |
| 22:58.40 | boster | eer such |
| 22:59.08 | outtolunc | it does, if the channel was 'attempted' to be used |
| 22:59.15 | outtolunc | which hasn't happened yet |
| 22:59.31 | outtolunc | 'something' has to trigger the process |
| 22:59.38 | boster | the other thing is that you can't hang up the agent channel if a call has dropped to it from the queue |
| 23:00.00 | boster | I'll see if I can hang up the sip channel if a queue call drops to it |
| 23:00.00 | outtolunc | why not |
| 23:00.09 | boster | it just doesn't |
| 23:00.17 | boster | it accepts the command but does nothing with it |
| 23:00.33 | outtolunc | you do realize there are 3 channels involved right? |
| 23:00.40 | bartoloco80 | i hava a problem checking the channel status |
| 23:00.50 | outtolunc | the agent interface, the agent channel and the caller channel |
| 23:01.00 | outtolunc | e callee |
| 23:01.01 | boster | yes |
| 23:01.21 | bartoloco80 | even if the channel is ringing it always seems the channel is down (0) status |
| 23:01.53 | bartoloco80 | CHANNEL STATUS [SIP/365] |
| 23:02.40 | bartoloco80 | but really the channel is ringing or has been answered |
| 23:02.40 | outtolunc | but it is not just that, it is SIP/365-or,something |
| 23:02.54 | outtolunc | ekk, mixing replies |
| 23:06.26 | outtolunc | heading, home bbiaf |
| 23:17.50 | *** join/#asterisk a1fa (n=fiddy@unaffiliated/a1fa) |
| 23:17.52 | a1fa | hello |
| 23:20.10 | a1fa | i dont know if this is a bug or not |
| 23:20.37 | a1fa | maybe my isp is changing my sip packets |
| 23:21.05 | a1fa | but my packets are comming to my sip server from a different IP |
| 23:21.33 | drmessano | What do you mean? |
| 23:21.38 | a1fa | my ip is |
| 23:21.42 | a1fa | <PROTECTED> |
| 23:21.59 | a1fa | my server is seeing packets from 66.158.179.51 |
| 23:22.00 | drmessano | starts crapflooding your router |
| 23:22.04 | drmessano | Ok |
| 23:22.10 | a1fa | i dont get it |
| 23:22.21 | a1fa | how can that be |
| 23:22.33 | a1fa | 66.158.178.159 |
| 23:22.39 | a1fa | thats what shows up on w |
| 23:22.43 | drmessano | Do you know whats on 66.158.179.51 ? |
| 23:23.16 | a1fa | no |
| 23:23.19 | a1fa | not my ip address |
| 23:23.30 | a1fa | it doesnt even respond |
| 23:23.41 | a1fa | well it does |
| 23:23.41 | a1fa | wtf |
| 23:24.19 | drmessano | So what you are telling me is.. |
| 23:24.31 | drmessano | You're seeing sip packets from 66.158.179.51 |
| 23:24.38 | a1fa | ok.. thats fucked up |
| 23:24.39 | a1fa | 66.158.179.51 127.0.0.1 UGHS 0 1020 33168 lo0 |
| 23:24.45 | a1fa | and my ISP is actually routing this... |
| 23:25.15 | drmessano | um ok |
| 23:25.50 | a1fa | how the hell is that possible>? |
| 23:26.25 | drmessano | Are you sure 66.158.178.159 is your IP? |
| 23:26.51 | a1fa | yup |
| 23:26.53 | a1fa | just checked it |
| 23:27.13 | fujin | talk to your ISP |
| 23:27.27 | fujin | if you do traffic to .51, does it reach .159? |
| 23:27.31 | *** join/#asterisk quigon (n=matias@190.3.121.15) |
| 23:27.33 | a1fa | nope |
| 23:27.35 | *** join/#asterisk _mwoodj_ (n=MWoodJ@pdpc/sponsor/digium/hyper-eye) |
| 23:27.36 | *** join/#asterisk cesar_CR (n=cesar@celord.ice.co.cr) |
| 23:27.45 | a1fa | i can get to .51 from the internet |
| 23:27.48 | a1fa | its someone elses machine |
| 23:28.07 | drmessano | Maybe you're being haxored |
| 23:28.15 | fujin | your own packets are coming to your sip server from a different ip? |
| 23:28.15 | a1fa | nope |
| 23:28.20 | a1fa | yes |
| 23:28.23 | fujin | where is your SIP device in relation to the server? |
| 23:28.30 | a1fa | internet |
| 23:28.35 | a1fa | ~20 hops |
| 23:28.46 | a1fa | now it worked fine |
| 23:28.53 | a1fa | i shut off my PAP2-NA |
| 23:28.55 | *** join/#asterisk r0d3nt (n=astrutt@backup.realprohosting.com) |
| 23:28.55 | fujin | NEEDMOREINFO |
| 23:29.02 | a1fa | and now packets are coming from the right IP |
| 23:29.07 | fujin | cool |
| 23:29.13 | drmessano | uh |
| 23:29.35 | fujin | (fail) |
| 23:29.36 | a1fa | it could be state table from pf |
| 23:29.37 | drmessano | My brain just exploded |
| 23:29.42 | fujin | +1 drmessano |
| 23:29.47 | a1fa | pf state table may be getting corrupted |
| 23:29.55 | fujin | Don't use pf? :> |
| 23:30.22 | drmessano | WTF just happened here? |
| 23:30.32 | fujin | does not compute |
| 23:31.21 | a1fa | that was really wierd |
| 23:31.26 | drmessano | a1fa: is this the same PAP2-NA you told me is failing? |
| 23:31.31 | a1fa | yeah |
| 23:31.36 | drmessano | The one you incessantly bitch about? |
| 23:31.38 | drmessano | Ok |
| 23:31.41 | a1fa | but i was doing tcpdumps on both ends |
| 23:31.45 | a1fa | one leaving my firewall |
| 23:31.49 | a1fa | one at the sip server |
| 23:32.01 | drmessano | Epic Fail |
| 23:32.13 | a1fa | and I was seeing a different source IP at the sip end |
| 23:32.20 | a1fa | PAP2 has RFC1918 |
| 23:32.29 | a1fa | so i know its not PAPs fault |
| 23:32.43 | a1fa | which points to: either my PF box, or ISP is doing something |
| 23:33.13 | drmessano | Remind me to rickroll you sometime |
| 23:33.18 | a1fa | plz dont |
| 23:33.19 | a1fa | i kill you |
| 23:33.34 | a1fa | rick rolling is so gay |
| 23:33.43 | a1fa | i want to kick the douchebag who invented it |
| 23:34.02 | drmessano | So is your PAP2 you've been complaining about for months |
| 23:34.05 | drmessano | Throw it away |
| 23:34.07 | drmessano | Move on |
| 23:34.28 | a1fa | no dude |
| 23:34.32 | a1fa | i dont think its the device now |
| 23:34.39 | drmessano | O.o |
| 23:35.30 | *** join/#asterisk xenonex (n=xenonex@89.218.237.201) |
| 23:35.33 | a1fa | here is aproof |
| 23:35.34 | a1fa | http://www.pastebin.ca/970317 |
| 23:35.49 | a1fa | so the whole layer3 transport is fucked up |
| 23:35.59 | drmessano | I can't take any more of your troubleshooting |
| 23:36.03 | drmessano | Make it stop |
| 23:36.08 | a1fa | nurse |
| 23:36.10 | a1fa | pass the hammer |
| 23:36.15 | a1fa | no dude, did you see that |
| 23:36.36 | a1fa | i need to check my state table |
| 23:36.39 | fujin | its your tubes man |
| 23:36.40 | fujin | get out |
| 23:37.18 | a1fa | my intra nets |
| 23:38.17 | drmessano | Thats.. pretty screwy |
| 23:38.37 | a1fa | i cant believe it dude |
| 23:38.42 | a1fa | thats the worst part |
| 23:38.43 | *** join/#asterisk mascool (n=george@65.29.106.235) |
| 23:39.08 | mascool | hello everyone, could anyone please clarify how codec selection on sip channels works ? |
| 23:39.58 | fujin | SDP negotiates between the devices, based on the allow/disallowed/preferred stuff. |
| 23:40.09 | mascool | our provider is sending SDP with rtmpmap for PCMU and G729, and it has allow=g729 and allow=ulaw , in that order |
| 23:40.48 | mascool | the 2 nd leg is to an aastra480 with allow=ulaw only |
| 23:41.08 | mascool | call fails with "No audio format found to offer" |
| 23:41.20 | fujin | odd |
| 23:41.29 | mascool | i know, right ? |
| 23:41.52 | mascool | provider: Codec Order : (g729:20,ulaw:20) |
| 23:42.08 | mascool | phone: Codec Order : (ulaw:20) |
| 23:42.38 | fujin | and the phone is configured to use ulaw (not in asterisk, but in the actual phone) |
| 23:42.41 | fujin | ? |
| 23:42.51 | mascool | the phone is set to use both |
| 23:42.55 | a1fa | lame |
| 23:42.58 | a1fa | /sbin/pfctl -k 192.168.1.35 |
| 23:43.12 | mascool | i have a tshark capture and * is not even sending an invite to the phone |
| 23:43.46 | mascool | it just rejects the invite from the provider with a 503 service unavailable |
| 23:44.08 | fujin | pastie your relevant sip.conf, secrets obfuscated |
| 23:45.33 | a1fa | d |
| 23:46.15 | mascool | provider: http://pastebin.com/m3ed5324c |
| 23:46.53 | mascool | they're pasted from mysql, sorry if they look crappy |
| 23:48.53 | fujin | mascool: if you reconfigure with allow all, no disallow lines |
| 23:48.53 | mascool | phone: http://pastebin.com/m9dd8eec |
| 23:49.00 | fujin | do you ge ta call up? |
| 23:49.27 | mascool | let me try it out |
| 23:51.48 | mascool | fujin, on both ends ? |
| 23:51.56 | fujin | Yeah, worth a shot. |
| 23:52.02 | mascool | doing it now |
| 23:52.13 | fujin | whenever I have codec negotiation issues I try and remove all limiting fators |
| 23:52.17 | fujin | and buid up from there |
| 23:52.50 | mascool | now both legs are ulaw, |
| 23:53.23 | a1fa | fuck |
| 23:53.27 | a1fa | i cant replicate the problem |
| 23:53.58 | mascool | i basically want it to default to g729 (most peers) but allow ulaw for peers that support it |
| 23:56.15 | *** join/#asterisk romanc (n=rchernob@owa.alicatscientific.com) |
| 23:56.22 | *** join/#asterisk rupa (i=rupa@gw.rupa.com) |
| 23:56.33 | romanc | how do i get the svn version zaptel on my asterisk now machine? |
| 23:56.38 | *** join/#asterisk outtolunc (n=me@c-67-174-216-60.hsd1.ca.comcast.net) |
| 23:57.11 | romanc | <PROTECTED> |
| 23:57.15 | fujin | install the kernel sources |
| 23:57.16 | Nugget | if you don't understand how to get the svn version then that's just the universe's way of protecting you from yourself. |
| 23:57.41 | romanc | yeah how do i install teh kernel sources do u know? |
| 23:57.44 | mascool | fujin, any ideas how I can accomplish this ? |
| 23:58.06 | romanc | @nugget: my asterisk now does not work with relese version zaptel so i am forced to upgrade |
| 23:58.07 | fujin | mascool: should be a matter of doing exactly what you've done |
| 23:58.25 | fujin | mascool: I assume you've licensed the g729 transcoder for asterisk? |
| 23:58.28 | fujin | if you've not, you'll need to |
| 23:58.41 | fujin | cause it will go provdier (g729) -> asterisk transcoder -> phone (ulaw) |
| 23:58.52 | mascool | yes but I've unloaded that because I don't want any transcoding |
| 23:59.02 | fujin | heh, well that's why |
| 23:59.07 | fujin | you *need* it |
| 23:59.13 | mascool | why would I need it |
| 23:59.18 | mascool | if the provider supports is |
| 23:59.23 | mascool | and the phone supports is ? |
| 23:59.27 | romanc | fujin: can you please tell me the command to install kernel sources |
| 23:59.30 | fujin | romanc: no |
| 23:59.37 | romanc | why not |
| 23:59.41 | mascool | ulaw is free |
| 23:59.58 | fujin | mascool: not sure how the g729 stuff works actually sorry |