00:00.22 | justinu | why oh why can I not get the PBX to send me any RTP when I make sip calls? |
00:00.22 | hardwire | http://pastebin.ca/24075 |
00:00.23 | Pegger | FuriousGeorge ok i will owrk on it |
00:00.24 | hardwire | there |
00:00.32 | justinu | ever since upgrade to 1.2, it's broken |
00:00.42 | hardwire | good quick and dirty way of doing IVR for real quick autoattendants and different DID's |
00:00.44 | justinu | iax calls work |
00:01.00 | FuriousGeorge | justinu: what ports are you forewarding how much nat is going on |
00:01.23 | justinu | a fair bit of nat, but it doesn't seem to stop it from working if I route the inbound calls into an IVR |
00:01.43 | justinu | but if I try to route the inbound call into a Dial(SIP/exten), no rtp at all |
00:01.52 | FuriousGeorge | IVRs are outside of my scope, what ports are you forwarding |
00:02.07 | FuriousGeorge | are both the client and server behind nat |
00:02.20 | justinu | yes |
00:02.35 | justinu | i'm not explicitly forwarding any udp ports for media |
00:03.49 | FuriousGeorge | justinu: you need to set externip and localnet in sip.conf |
00:03.55 | justinu | done already |
00:04.12 | FuriousGeorge | you also need to foreward (iirc) 10000-20000 to server and client |
00:04.16 | FuriousGeorge | udp |
00:04.23 | FuriousGeorge | and 5060 tcp and udp |
00:04.51 | justinu | i'll try it... wondering why it worked on the older pbx version without any changes tho |
00:06.07 | FuriousGeorge | justinu: also 8000 udp and maybe 5061 udp and tcp not sure about that last one. basically there is no elegant way to do it for many clients w/o 3rd part software or taking * out from behind nat |
00:06.43 | *** join/#asterisk Connor (n=billy@198-144-165-65.knx.tn.nxs.net) |
00:07.04 | JerJer | FuriousGeorge: do you even know what you are saying? |
00:07.44 | hardwire | weird |
00:07.47 | FuriousGeorge | JerJer: yeah, i actually got itworking, the prot numbers are straight from voip-info, and someone here told me about externip and localnet, and it didnt work till i did it |
00:07.48 | shido6 | insert filter |
00:07.50 | hardwire | my snom phones are recieving hints |
00:07.54 | hardwire | but only once they boot |
00:08.04 | hardwire | so I see all the extensions in use on my network |
00:08.07 | hardwire | only on the bootup |
00:08.11 | hardwire | the LED's never change |
00:08.15 | JerJer | FuriousGeorge: and you followed this info? |
00:08.31 | JerJer | first off asterisk only ever deals with UDP on sip, so port forwarding tcp is useless |
00:08.42 | JerJer | second, whatever ports are specified in rtp.conf would need to be forwarded |
00:09.01 | JerJer | and sip has nothing to do with udp port 8000, unless you very specifically set that port in sip.conf |
00:09.24 | justinu | well, riddle me this... i have no ports forwarded for media, but I can see the media packets going in and out when I point an inbound call at a Playback statement. |
00:09.29 | FuriousGeorge | first i forewarded the ports to client and server as per voip-info then when that didnt work i set localnet and externip and voila |
00:09.40 | justinu | i'm looking at tcpdump on the * host. |
00:10.02 | justinu | they're on exactly the right ports that the SDP says they should be on. |
00:10.18 | JerJer | so then turn off the firewall |
00:10.30 | justinu | BUT, if I change the exten => command, and point the inbound DID at a Dial(SIP/exten) command, no more RTP comes at all. |
00:10.31 | FuriousGeorge | JerJer: i said i wasnt sure about 5061, and i prefaced the port numbers with an "iirc" and you gotta admit i nailed the default udp ports in rtp.conf ;) |
00:10.59 | justinu | why do I need to turn off the firewall if it's obvious RTP is making it thru? |
00:11.15 | JerJer | is it? |
00:11.19 | justinu | yes |
00:11.22 | JerJer | then why don't you hear audio? |
00:11.24 | justinu | i do |
00:11.28 | FuriousGeorge | u said it wasnt a second ago |
00:11.39 | justinu | I said that if I point the inbound DID to a playback statement, I get RTP, and hear audio. |
00:12.18 | justinu | but if I point the inbound DID to a Dial(SIP/whatever), there is no RTP at all (to either the phone, or the sip server outside my network) |
00:12.19 | justinu | and no audio |
00:12.45 | justinu | signalling works fine, the phone rings, answers, hangs up, all that is fine. |
00:13.00 | JerJer | then something is blocking it |
00:13.06 | JerJer | or it is not getting routed correctly |
00:13.29 | justinu | what would be blocking outbound RTP from the * server? |
00:13.45 | JerJer | firewall |
00:13.47 | justinu | i'm watching tcpdump running on the same box. |
00:13.50 | JerJer | or incorrect SDP |
00:13.53 | justinu | the firewall has nothing to do with it. |
00:14.38 | justinu | is * doing any media processing at this point, or is it just redirecting rtp? |
00:15.06 | JerJer | you tell us |
00:15.22 | FuriousGeorge | im off |
00:16.19 | justinu | you're obviously the expert |
00:16.24 | justinu | why don't you tell me? |
00:19.25 | *** join/#asterisk Snake-Eyes (n=blog@203.201.97.178) |
00:24.41 | Corydon-w | justinu: is your firewall wide open? |
00:25.32 | justinu | no, and it's also natting |
00:25.37 | Corydon-w | If not, you better try that first, after JerJer's excellent suggestion |
00:25.43 | justinu | i'm just befuddled, because this was working fine with 1.0.7 |
00:25.52 | justinu | upgrade to 1.2 and broken. |
00:26.17 | Corydon-w | 1.2 isn't out yet |
00:26.20 | JerJer | did you read UPGRADE.txt ? |
00:26.35 | justinu | no, i'll take a look at that right now. |
00:28.22 | phpboy | I seriously need some help with ISDN support in BSD :< |
00:28.36 | justinu | i know PRI well, but not bsd |
00:29.22 | phpboy | shit :< |
00:31.39 | phpboy | looks like I'm gonna have to install fedora :/ |
00:31.50 | hardwire | file: wheres your pickup app? |
00:31.59 | hardwire | I have a shitload oof whiney employees now |
00:33.24 | justinu | if * places an outbound call to a SIP peer, and gets a 302 Moved Temporarily back, is there any way to get it to automatically follow that new SIP URI and send the call to it? |
00:34.37 | Ariel_ | any one knows how to forward a channel to another one? on freenode here? |
00:35.00 | *** join/#asterisk l-fy (n=diana@yate/developer/l-fy) |
00:35.02 | l-fy | hello |
00:35.10 | Ariel_ | phpboy, use CentOS instead of fedora |
00:35.10 | l-fy | did someone ever used a sangoma a104 ? |
00:35.20 | Ariel_ | l-fy, hello |
00:35.29 | l-fy | hi Ariel_ |
00:38.05 | *** join/#asterisk Snake-Eyes (n=blog@203.201.97.178) |
00:39.34 | xheliox | Really stupid question, if I want to bring in a T1 to support inbound and outbound calls, do I need to put a CSU/DSU in front of it before I terminate it to a 410P? |
00:39.57 | justinu | what's a 410p? |
00:40.13 | xheliox | TE410P? The quad T1 card from Digium |
00:40.15 | phpboy | Ariel_: perhaps I should just install asterisk@home str8? |
00:40.18 | JerJer | TE410P |
00:40.31 | justinu | usually you only need a CSU if you're running data |
00:40.43 | JerJer | xheliox: no the TE410P has an intergrated CSU/DSU |
00:40.56 | xheliox | JerJer: Is that right? Very nice. |
00:41.13 | JerJer | just come right off your smart jack or mux (whatever you gots) |
00:41.28 | xheliox | Hrm. That seems just too easy. :) |
00:42.49 | Ariel_ | phpboy, that is your choice |
00:42.54 | *** join/#asterisk Moc (n=mochouin@modemcable173.101-70-69.mc.videotron.ca) |
00:43.39 | phpboy | I think it'll prolly be best |
00:43.49 | Moc | hi all |
00:43.55 | JerJer | phpboy: then don't expect people to support you in here |
00:44.20 | phpboy | ok |
00:44.25 | phpboy | CentOS it is then |
00:44.28 | phpboy | I'll get that |
00:44.38 | Ariel_ | phpboy, you can use #amportal |
00:44.46 | Ariel_ | and there are some that will help you here |
00:44.47 | Moc | centos is working good for me |
00:45.05 | phpboy | I've got one of the quad Junghanns cards... need it installed and working by the weekend... |
00:45.09 | Ariel_ | I have switch all my servers over to centOS and I have not had any problems |
00:45.11 | phpboy | CentOS will be fine with this? |
00:45.23 | JerJer | phpboy: then asterisk@home is the worst thing you can do |
00:45.25 | phpboy | where's the best place to get CentOS? |
00:45.30 | phpboy | ah, ok |
00:45.33 | Moc | phpboy: www.centos.org |
00:45.44 | Moc | there is multiple mirror... find the fastest for you |
00:45.51 | phpboy | cool, I should obviously get the newest one? |
00:46.08 | JerJer | phpboy: last i knew his cards required patches and what not |
00:46.11 | phpboy | am I gonna have to do kernel reconfigs... etc? to get the card going? |
00:46.16 | JerJer | most likely |
00:46.21 | JerJer | i don't know |
00:46.23 | phpboy | fuck :/ |
00:47.17 | JerJer | hire a consultant (not me) |
00:47.25 | Ariel_ | phpboy, there are two different ones for them. One based on 2.4 kernel and the otherone as 2.6 kernel |
00:47.38 | phpboy | I see... |
00:47.46 | phpboy | where's the best place to get docs for that? |
00:52.25 | AgiNamu | anyone here know a bit about 5350? I'd like to hire you for a bit |
00:53.15 | AgiNamu | what kinda switch type is most probably for a GC PRI? 5ess right? |
00:55.43 | Ariel_ | 5ess |
00:57.37 | *** join/#asterisk shido6 (n=shido@d221-68-210.commercial.cgocable.net) |
01:00.38 | hardwire | why the f*ck does m,ake clean on CVS HEAD result in a large CPU intense loop? |
01:02.29 | X-Rob | I'm fucking GOOD |
01:02.31 | X-Rob | FUCKING GOOD. |
01:02.35 | hardwire | you fuck good |
01:02.41 | hardwire | bah |
01:02.51 | hardwire | app_paging.c ? |
01:02.53 | X-Rob | Works with GXP2000's, Snoms, Cisco and Polycoms |
01:03.02 | X-Rob | nah. it's an agi. |
01:03.09 | hardwire | ah |
01:03.12 | hardwire | lets write an app |
01:03.18 | X-Rob | drags 'em into meetme. |
01:03.30 | hardwire | I wrote an agi that does it |
01:03.39 | hardwire | worked really damn well |
01:03.42 | X-Rob | hardwire - wouldn't take much. Take app_conference, plug it into Dial, and add the Call-Info header. |
01:03.43 | hardwire | but its such a damn hack |
01:04.14 | X-Rob | hardwire - how is it a hack? |
01:04.26 | hardwire | no offense to your fine work |
01:04.38 | hardwire | lets talk about this in 48 horus again if you are still upset that I called it a hack :) |
01:04.41 | *** join/#asterisk Johnsie (n=john@acs-24-154-53-217.zoominternet.net) |
01:05.10 | hardwire | http://bugs.digium.com/bug_view_page.php?bug_id=0003644 |
01:05.16 | hardwire | I really wanted that to just work |
01:05.17 | hardwire | but no |
01:05.23 | hardwire | it seems like most of it is in CVS HEAD anyways |
01:05.29 | hardwire | so I am off to use CVS HEAD from now on |
01:05.32 | X-Rob | Yeah |
01:05.34 | *** part/#asterisk l-fy (n=diana@yate/developer/l-fy) |
01:05.35 | X-Rob | this only works with HEAD |
01:05.39 | X-Rob | you need SipAddHeader |
01:05.44 | hardwire | I use SipAddHeader |
01:06.18 | X-Rob | If you've written an agi, and have found problems, tell me, coz I want to avoid them 8) |
01:09.11 | *** join/#asterisk anthm (n=anthm@h4608ac83.area4.spcsdns.net) |
01:09.11 | *** mode/#asterisk [+o anthm] by ChanServ |
01:10.10 | *** join/#asterisk jskcr (n=jskcr@unaffiliated/jskcr) |
01:10.59 | jskcr | hy all |
01:12.05 | hardwire | wow |
01:12.07 | hardwire | CVS HEAD is just |
01:12.10 | hardwire | looping |
01:12.14 | hardwire | over and over and over and over |
01:12.27 | hardwire | any ideas? |
01:12.29 | doughecka | its olympic! |
01:12.35 | hardwire | quite! |
01:12.40 | hardwire | man I want a beer |
01:12.45 | *** join/#asterisk Sk3tCh (n=wrickout@86.127.41.114) |
01:12.46 | Sk3tCh | hi |
01:13.03 | jskcr | looping? |
01:13.15 | jskcr | please define looping a little better. |
01:13.52 | hardwire | http://pastebin.ca/24080 |
01:13.55 | hardwire | good enough? |
01:14.51 | Sk3tCh | how can i make a call forward? |
01:15.00 | Sk3tCh | my iptelephon has FWD button |
01:15.05 | hardwire | neat |
01:15.11 | hardwire | maybe its for Free World Dialup :) |
01:15.14 | hardwire | now Forward :) |
01:15.19 | hardwire | or maybe its for transferring :) |
01:15.27 | jskcr | wow thats a new one |
01:15.31 | X-Rob | it's a broken build |
01:15.38 | X-Rob | rm -rf /usr/src/asterisk |
01:15.41 | Sk3tCh | yes but i must set something in asterisk? |
01:15.41 | hardwire | yeh |
01:15.43 | X-Rob | check it out again |
01:15.43 | hardwire | resnagging |
01:15.52 | hardwire | takes so long to snag |
01:15.53 | hardwire | soo |
01:15.56 | X-Rob | subversion DOESN'T HAVE THESE PROBLEMS |
01:16.03 | hardwire | anybody here use the 686 optimizations |
01:16.05 | X-Rob | hardwire - use '-z9' on your cvs co |
01:16.06 | drumkilla | it's not a cvs issue |
01:16.08 | drumkilla | do make clean |
01:16.11 | hardwire | or the MMX optimizations to GSM? |
01:16.15 | *** join/#asterisk JunK-Y (n=junky@Toronto-HSE-ppp3742731.sympatico.ca) |
01:16.20 | hardwire | whats -z9 btw? |
01:16.21 | *** join/#asterisk anthm (n=anthm@h4608ac83.area4.spcsdns.net) |
01:16.21 | *** mode/#asterisk [+o anthm] by ChanServ |
01:16.26 | X-Rob | drumkilla - I've had that and make clean didn't fix it. |
01:16.34 | jskcr | I think I see the problem |
01:16.35 | X-Rob | hardwire - compresses the data with gzip -z9 |
01:16.41 | hardwire | ohyeh |
01:16.59 | hardwire | some 1999 files in here |
01:17.10 | jskcr | remove then echo; else \ |
01:17.25 | jskcr | then the \ at the end of the next line |
01:17.26 | hardwire | X-Rob: haha.. make clean was doing that too |
01:17.35 | hardwire | it keeps moving a vresion file where it wants it |
01:17.37 | hardwire | then removing it |
01:17.40 | hardwire | and moving in another |
01:17.42 | hardwire | then removing it |
01:18.10 | jskcr | try that hardwire |
01:18.36 | hardwire | hold |
01:18.43 | hardwire | lets see if it even wants to compile |
01:18.49 | hardwire | I just rechecked it all out |
01:19.27 | hardwire | it wants to compile now. |
01:19.30 | Sk3tCh | i use actos. there is a program better like this? |
01:19.33 | *** join/#asterisk Snake-Eyes (n=blog@203.201.97.178) |
01:19.42 | hardwire | Sk3tCh: probably |
01:19.54 | Sk3tCh | what is the name? |
01:19.56 | Sk3tCh | :D |
01:19.59 | hardwire | make clean worked |
01:19.59 | hardwire | yay |
01:20.18 | hardwire | Sk3tCh: other. |
01:20.22 | Sk3tCh | pff |
01:20.55 | Sk3tCh | what do u use for config? (simple edit conf?) |
01:21.07 | jskcr | vi |
01:21.09 | Sk3tCh | vi |
01:21.13 | Sk3tCh | i thought |
01:21.25 | hardwire | Sk3tCh: I use jed :) |
01:21.45 | Sk3tCh | i think jed is like vi |
01:21.50 | hardwire | X-Rob: know much difference inbetween CVS and 1.2.0-beta ? |
01:21.52 | Sk3tCh | i am newbie |
01:22.03 | hardwire | Sk3tCh: no friggin way |
01:22.18 | X-Rob | hardwire - bug fixes mainly. I don't think theres any new functionality. |
01:22.25 | hardwire | or config changes I hope |
01:23.44 | FG-away | is there some way to control the volume of an mp3 in musiconhold.conf? |
01:23.55 | hardwire | quietmp3 |
01:23.57 | hardwire | vs mp3 |
01:24.04 | hardwire | or.. resample the mp3 :) |
01:24.08 | hardwire | with quieter volume |
01:24.15 | FG-away | hardwire: thats what im using. its beyond loud. im using the default file |
01:24.24 | hardwire | use an mp3 that says "to lower the volume.. move handset away from head" |
01:24.31 | *** join/#asterisk santiago (n=santiago@63.245.86.245) |
01:24.31 | FG-away | ill just reencode my own tracks |
01:24.43 | hardwire | or change the code for app_musiconhold |
01:24.44 | hardwire | :) |
01:24.45 | hardwire | or |
01:24.58 | FG-away | *hopes |
01:24.58 | hardwire | you could probably use custom |
01:25.05 | hardwire | and just doethe mpg123 line yourself |
01:25.23 | hardwire | then all is happy in FuriousGeorge land |
01:25.27 | hardwire | I will be here all week |
01:25.30 | hardwire | I will |
01:25.34 | hardwire | and all night |
01:25.37 | hardwire | all day |
01:25.39 | hardwire | all night |
01:25.56 | FuriousGeorge | i hope you at least take a break to copulate with a humanoid |
01:26.08 | hardwire | ######### More GSM codec optimization |
01:26.09 | hardwire | ######### Uncomment to enable MMXTM optimizations for x86 architecture CPU's |
01:26.11 | hardwire | ######### which support MMX instructions. This should be newer pentiums, |
01:26.11 | hardwire | ######### ppro's, etc, as well as the AMD K6 and K7. |
01:26.11 | hardwire | K6OPT = -DK6OPT |
01:26.11 | hardwire | ] |
01:26.11 | hardwire | kinda neat |
01:26.19 | hardwire | I copulated earlier |
01:26.21 | hardwire | thanky ou very much |
01:26.28 | FuriousGeorge | fornicator |
01:26.38 | hardwire | I guess |
01:27.01 | hardwire | I hope to high heaven the notify devstate crap works somewhat in CVS |
01:27.08 | hardwire | cause I am gonna freak out if it doesnt |
01:27.54 | FuriousGeorge | its just wierd. some installs its at a nice level, and on some its quite loud. its the default MoH so the only variable is the actual hardware in the box |
01:28.22 | FuriousGeorge | hey is it ilegal to just rip my own mp3s for cds and use them for MoH |
01:28.31 | FuriousGeorge | from cds* |
01:28.51 | wolfson | yes |
01:28.56 | FuriousGeorge | nuts |
01:29.05 | wolfson | its called rebroadcasting |
01:29.32 | FuriousGeorge | but its legal to play those same cds in my waiting room (if i had one) |
01:29.35 | hardwire | FuriousGeorge: resample.. normalized and 50% volume |
01:29.40 | hardwire | and use like.. rawplayer |
01:29.44 | hardwire | since you are going to do all that crap |
01:29.52 | wolfson | furiousgeorge: actually that can get you in trouble as well |
01:30.24 | wolfson | they come around and bug restaurants all the time for royalties |
01:30.59 | FuriousGeorge | hardwire: apparently im gonna be using public domain classical pieces |
01:31.19 | hardwire | as in |
01:31.22 | hardwire | you are being forced to? |
01:31.36 | hardwire | I would still normalize and make native or use rawplayer |
01:32.05 | FuriousGeorge | hardwire: well i dont wanna do anything ileagal, and apparently its ilegal to rebroadcast my cds as mp3 for MoH |
01:32.18 | FuriousGeorge | i got a link thats supposed to be encoded for * use |
01:32.19 | jskcr | FuriousGeorge: umm can you find anything better after a few minutes on hold customers may commit suicide |
01:32.34 | FuriousGeorge | jskcr: whats wrong with mozart |
01:32.39 | X-Rob | FuriousGeorge - get the MOH MP3s from AussieVoip. They're PD and you have permission to rebroadcast. |
01:33.01 | FuriousGeorge | X-Rob: facinating, ill look into that |
01:33.04 | *** join/#asterisk spackle (n=spackle@209.234.83.19) |
01:33.33 | hardwire | see |
01:33.37 | hardwire | where do you go to get permissions |
01:33.43 | hardwire | there needs to be a moh agency |
01:33.50 | Hogie | riaa |
01:33.51 | Hogie | :P |
01:33.55 | hardwire | no |
01:33.57 | hardwire | somebody else |
01:33.58 | hardwire | :) |
01:34.05 | spackle | AASCAP |
01:34.06 | hardwire | sweet.. upgradded to CVS |
01:34.11 | hardwire | its like 500 times more chatty |
01:34.15 | spackle | or BMI |
01:34.16 | hardwire | on single verbose |
01:34.21 | hardwire | yeh |
01:34.32 | hardwire | but why would I.. the little guy.. wanna contact them |
01:34.32 | FuriousGeorge | havent the grateful dead "public domained" any of their concerts or something :) |
01:34.36 | hardwire | why not get a broker to do it |
01:35.04 | spackle | hardwire, because if they catch, you they can fine you. |
01:35.36 | Hogie | just make your own music |
01:35.42 | Hogie | I sing on my moh |
01:35.48 | FuriousGeorge | spackle: he's trying to do it legally, and i think the reason is simply b/c the royalties to "Miss You" by the "Rolling Stones" is gonna be more $ than its worth |
01:37.14 | X-Rob | Man that was a funky video clip |
01:37.18 | X-Rob | I'm sure I've got it somewhere, too |
01:37.27 | spackle | hardwire, it really isn't an unreasonable license fee, as far as unreasonable license fees go. |
01:37.50 | jskcr | FuriousGeorge: hmmm thats true any of the the greatful dead bootlegs are public domain |
01:37.54 | *** join/#asterisk NDT (n=me@cpe-24-195-219-245.nycap.res.rr.com) |
01:37.57 | hardwire | well |
01:37.58 | X-Rob | ah. I've got it on DVD. How inconvenient. |
01:38.00 | hardwire | I hate all things |
01:38.06 | hardwire | hows that .. RIAA! |
01:38.15 | FuriousGeorge | spackle: but if you wanted say ten songs from each the beatles, the rolling stones, etc. till you had a hundred tracks for MoH how much would that be |
01:38.28 | Hmm-home | she's an 8 she's a 9 she's a 10 i know, shes got really big tits blonde hair blue eyes and i'm about to kiss my heart goodbye |
01:38.43 | spackle | it's based on how many times it gets heard daily, IIRC |
01:38.46 | hardwire | Sep 27 17:38:27 NOTICE[25279]: chan_sip.c:10484 handle_request_subscribe: Failed to authenticate user <sip:1135@voip001.corp.anc.tdxnet.com>;tag=5zqf610ps0 for SUBSCRIBE |
01:38.48 | hardwire | Sep 27 17:38:27 NOTICE[25279]: chan_sip.c:10484 handle_request_subscribe: Failed to authenticate user <sip:1135@voip001.corp.anc.tdxnet.com>;tag=0od1tlhxhm for SUBSCRIBE |
01:38.50 | hardwire | Sep 27 17:38:27 NOTICE[25279]: chan_sip.c:10484 handle_request_subscribe: Failed to authenticate user <sip:1135@voip001.corp.anc.tdxnet.com>;tag=wxqf871jyi for SUBSCRIBE |
01:38.50 | hardwire | god |
01:38.52 | FuriousGeorge | jskcr: i was tempted for a second but i think dead bootlegs arent quite professional enough for MoH |
01:38.53 | hardwire | I am so tired of this notice |
01:38.57 | spackle | so you can fudge your numbers a little. |
01:38.59 | hardwire | I really think this should be at a higher verbose setting |
01:39.06 | hardwire | cause it kills my console at v 1 |
01:39.06 | Supaplex | well quit fscking pasting it multiple times :P |
01:39.14 | hardwire | its different each time |
01:39.15 | hardwire | 20 times |
01:39.16 | hardwire | per phone |
01:39.19 | hardwire | times 25 phones |
01:39.27 | hardwire | every few minutes |
01:39.51 | hardwire | don't you dare. |
01:39.54 | hardwire | I love my enter key. |
01:40.03 | spackle | looks like you can find out everything you want here: http://www.ascap.com/weblicense/webintro.html |
01:40.03 | jskcr | FuriousGeorge: there are some really good greatful dead bootlegs and remember the quality of a phone line is alot less |
01:40.23 | hardwire | ass cap? |
01:40.25 | jskcr | http://www.iq451.com/music/sites/grateful-dead-web.htm |
01:40.28 | X-Rob | FuriousGeorge - You've got me listening to Talking Heads now. |
01:41.00 | FuriousGeorge | jskcr: but arent they bootleg of concerts with their 20 min incoherent to the non-stoned jams, and cowd noise |
01:41.05 | spackle | This is not my beasutiful asterisk this is not my beautiful sip phone |
01:41.08 | *** join/#asterisk probonic (i=probonic@host-84-9-223-181.bulldogdsl.com) |
01:41.14 | spackle | letting the calls go by... |
01:41.17 | hardwire | bah.. it may or may not be worth something to just become a company that deals with liscencing music for Music on Hold |
01:41.18 | hardwire | heh |
01:41.25 | hardwire | but how do DJ's at say.. a party do it? |
01:41.36 | spackle | illegally in many instances |
01:41.41 | FuriousGeorge | X-Rob: people sleep, sleep in the daytime, uf they want to, if they want to |
01:42.41 | *** join/#asterisk mog_home (n=mogorman@user-24-236-84-48.knology.net) |
01:42.54 | X-Rob | w00t fear. |
01:43.16 | hardwire | file: ! |
01:43.30 | file[laptop] | oh god what |
01:43.36 | X-Rob | lookout. |
01:43.42 | hardwire | wheres your pickup app? |
01:43.53 | file[laptop] | in CVS. |
01:43.56 | hardwire | mo to the fo |
01:44.01 | X-Rob | app_pickup? |
01:44.07 | file[laptop] | app_directed_pickup |
01:44.12 | spackle | file: were you watching all the fireworks today? |
01:44.13 | hardwire | show application picjup |
01:44.14 | hardwire | damn |
01:44.15 | hardwire | it worked |
01:44.17 | hardwire | pickup |
01:44.33 | jskcr | I think phish also lets you make booty's at there concerts too |
01:44.43 | X-Rob | file - yeah. That was me that said 'It's fucking broken, cocksucker' |
01:44.55 | X-Rob | ..but slightly nicer than that. |
01:45.06 | hardwire | can it pickup a queue? |
01:45.09 | file[laptop] | it was broken on commit |
01:45.12 | X-Rob | yeah I know |
01:45.20 | file[laptop] | I don't use it :P |
01:45.23 | X-Rob | heh |
01:45.30 | X-Rob | Obviously no-one had used it until I tried. |
01:45.36 | file[laptop] | yeah |
01:45.37 | file[laptop] | fixed now! |
01:45.39 | FuriousGeorge | jskcr: yeah but phish has the same problem, can you imagine the average joe sitting through a half hour divided sky jam |
01:45.53 | FuriousGeorge | or worse, 5 minutes of it while on hold |
01:46.06 | FuriousGeorge | the radio on the other hand is legal |
01:46.14 | jskcr | FuriousGeorge: check out http://www.archive.org/audio/etreelisting-browse.php?collection=etree&cat=Grateful%20Dead |
01:46.15 | X-Rob | FuriousGeorge - Not in .au it's not. |
01:46.56 | FuriousGeorge | X-Rob: but in the us i think its ok, or atleast everyone does it, does * support interfacing with an fm tuner card |
01:47.14 | jskcr | FuriousGeorge: theres open source audio on that link too |
01:47.36 | hardwire | file[laptop]: yay. |
01:47.37 | hardwire | it worked |
01:47.40 | hardwire | only ringing right? |
01:47.44 | file[laptop] | correct |
01:47.47 | hardwire | good |
01:47.57 | hardwire | and if I get dropped calls.. I have your current addy right? |
01:48.00 | hardwire | for the mail bomb |
01:48.02 | file[laptop] | unfortunately :P |
01:48.14 | FuriousGeorge | jskcr: i bet if i look through that i can find an hour of acceptable music |
01:48.22 | hardwire | it worked so fine |
01:48.29 | hardwire | now get devstate workign |
01:48.39 | FuriousGeorge | chop chop |
01:48.49 | jskcr | yup theres some really good GD tracks on there too |
01:49.09 | spackle | FuriousGeorge, what do you mean the radio is legal? |
01:49.47 | FuriousGeorge | spackle: either its legal to play radio for MoH or everyone just does it <-- the essence of what i said |
01:50.23 | spackle | I think everyone just does it. Like putting a radio behind the counter at a store. |
01:50.38 | spackle | You can still get spanked if they catch you. |
01:51.30 | FuriousGeorge | all classical music is public domain though right? |
01:51.35 | hardwire | great |
01:51.36 | doughecka | no |
01:51.38 | jskcr | no |
01:51.41 | hardwire | X-Rob: chanisavail I use for paging |
01:51.42 | FuriousGeorge | d'oh |
01:51.44 | hardwire | and its funky in CVS HEAD |
01:51.44 | spackle | no, the music is but the performances are not |
01:51.53 | doughecka | exactly |
01:52.01 | doughecka | spackle: ever get your problem resolved? |
01:52.27 | spackle | doughecka, yeah they have me pennacillan. Oh, you mean THAT problem |
01:52.35 | doughecka | yea |
01:52.48 | spackle | I made the changes but haven't checke on it yet. |
01:53.03 | doughecka | lol, cool :) |
01:53.35 | spackle | the project you were interested in is on sourceforge acrophobia.sourceforge.net |
01:53.38 | FuriousGeorge | spackle: what about covers? can a cover band put out a covers album and have it be public domain |
01:54.14 | spackle | FurousGeorge, I'm not a lawyer, don't even play one on TV, but I work for some. So this is speculation. |
01:54.26 | spackle | But I think they would have to get a license to do it legally. |
01:54.54 | JamesDotCom | "all classical music is public domain though right?" |
01:54.55 | JamesDotCom | hahaha |
01:54.56 | JamesDotCom | wtf? |
01:55.42 | spackle | "all classical music" -> I assumed he meant mozart, etc, not recent material. |
01:55.52 | FuriousGeorge | u assumed right |
01:55.58 | JamesDotCom | ahh fair enough |
01:56.30 | spackle | FurouseGeorge - Just like software licensing, music is a freakin' minefield to do legally. |
01:56.40 | doughecka | thx |
01:56.50 | spackle | mustpeople just go to ascap and ask "what to I pay" |
01:57.13 | spackle | or muzak probably has special rates. |
01:57.16 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
01:57.27 | *** join/#asterisk znoG (n=gs@200.115.218.81) |
02:01.09 | *** join/#asterisk TheCops (n=dump@206.248.136.146) |
02:01.11 | TheCops | Hi |
02:01.14 | hardwire | wow |
02:01.16 | hardwire | sendtext worked |
02:01.19 | hardwire | wow |
02:01.39 | spackle | hardwire - on a hard phone? |
02:01.45 | hardwire | my snom 360 |
02:01.47 | *** join/#asterisk digime (n=drooth@ip68-111-235-172.sd.sd.cox.net) |
02:01.52 | hardwire | exten => 664,1,Answer() |
02:01.52 | hardwire | exten => 664,n,SendText("Monkey") |
02:01.52 | hardwire | exten => 664,n,MusicOnHold(default) |
02:01.55 | hardwire | I got off the line |
02:01.59 | hardwire | and I had an SMS icon |
02:01.59 | TheCops | Someone know if with SPA-2002 adapter you can configure 2 different SIP users for each FXS port |
02:02.04 | digime | What have people done with Asterisk? I'm particularly interested in DIY projects and things that can be done on a small/home office (or even hobbiest's) budget. If you have clever hacks or creative functionality you've implemented, I'd love to hear what a few people have come up with |
02:02.22 | hardwire | the MWI blinked even while I dialed 664 |
02:02.24 | hardwire | and I could press it |
02:03.06 | TheCops | hardwire, ho, nice! I guess it is working with snom 320 |
02:03.18 | hardwire | weird he? |
02:03.23 | hardwire | I heard it did |
02:03.26 | hardwire | just hadn't cared |
02:03.33 | hardwire | let me do something really nasty |
02:03.54 | hardwire | exten => 665,1,SendText("MonkeyLove") |
02:03.55 | hardwire | mohaha |
02:03.59 | TheCops | :P |
02:04.06 | hardwire | now to place a call file |
02:04.19 | hardwire | you must answer first |
02:05.26 | hardwire | well |
02:05.30 | hardwire | thats officially annoyingh |
02:05.40 | hardwire | sipsak is much easier |
02:05.53 | X-Rob | Didn't work for me. |
02:05.59 | hardwire | well |
02:06.04 | hardwire | I am using the 4.3 firmware |
02:06.06 | X-Rob | <PROTECTED> |
02:06.06 | X-Rob | <PROTECTED> |
02:06.06 | X-Rob | <PROTECTED> |
02:06.08 | hardwire | and I have all sorts of things turned off |
02:06.10 | X-Rob | Ah, I'm using 4.1 |
02:06.13 | hardwire | most of the things needed for intercom |
02:06.13 | JamesDotCom | 4.3 :| |
02:06.15 | JamesDotCom | new firmware every week |
02:06.16 | JamesDotCom | haha |
02:06.21 | hardwire | 4.3 has the worst changelog |
02:06.25 | hardwire | fixed this |
02:06.26 | hardwire | fixed that |
02:06.29 | hardwire | nyeh |
02:06.29 | JamesDotCom | worst as in best? |
02:06.32 | JamesDotCom | cool |
02:06.34 | hardwire | no |
02:06.39 | hardwire | its a very lame changelog |
02:06.53 | hardwire | I can't wait to see "Fixed shanes speakerphone woes" |
02:07.06 | hardwire | "Fixed for option to use multi line CID" |
02:07.07 | TheCops | hardwire, intercom, do you mean a direct paging system ? |
02:07.11 | hardwire | removed need for animated bell |
02:07.24 | hardwire | TheCops: I have an intercom extension |
02:07.26 | hardwire | *nxxx |
02:07.36 | hardwire | and a paging extension to page certain areas |
02:07.38 | TheCops | and you speak on all phone directly ? |
02:07.43 | hardwire | yes |
02:07.43 | TheCops | ho god |
02:07.48 | spackle | anyone have sendtext work with Polycom or is it only supported on certain phones? |
02:07.50 | TheCops | this is fucking nice |
02:07.54 | hardwire | what |
02:07.57 | TheCops | I was searching how to do that |
02:07.58 | hardwire | hearing your voice down the hall? |
02:08.00 | TheCops | lol |
02:08.04 | hardwire | yes |
02:08.07 | Hogie | hardwire: we have an outside speaker also for our pageall... |
02:08.14 | hardwire | I made every phone.. including the one down the street with a page all |
02:08.19 | hardwire | I want to get a paging application made |
02:08.20 | TheCops | I'm using asterisk for my business, and this is very useful :) |
02:08.21 | hardwire | for zone paging |
02:08.27 | hardwire | as well as intercomming |
02:08.30 | hardwire | yeh |
02:08.31 | hardwire | well |
02:08.34 | hardwire | they won't shut up about it here |
02:08.39 | hardwire | so I got their damn paging working |
02:08.44 | hardwire | and now they annoy everybody |
02:09.02 | hardwire | Supaplex: hows my enter key now bitch! |
02:09.05 | Hogie | today, I silently transfered someone to the outside pager... talk about funny, since they were at their desk singing with the radio |
02:09.05 | hardwire | :) |
02:09.15 | TheCops | lol |
02:09.16 | hardwire | Hogie: thats mean yo |
02:09.27 | spackle | good clean fun |
02:09.38 | hardwire | TheCops: talk to X-Rob about his handy dandy AGI :) |
02:09.41 | Supaplex | meh |
02:09.52 | Hogie | I built in my own silent intercom numbers, the normal ones give a ding when they autoanswer, my other ones dont:) |
02:09.52 | hardwire | Supaplex: sorry I scroll your terminal :) |
02:09.55 | TheCops | For intercom ? |
02:09.57 | hardwire | its all the hacking I can do. |
02:10.14 | hardwire | I am gonna DOS your scrollback buffer! |
02:11.23 | Supaplex | DOS is my friend |
02:11.47 | spackle | that's just sad |
02:11.52 | TheCops | hardwire, do you have some docs about intercom and paging ? |
02:12.00 | hardwire | no |
02:12.09 | hardwire | and you cannot get them no matter how much you ask :) |
02:12.16 | hardwire | what phones are you using anyways? |
02:12.41 | TheCops | snom 320 |
02:13.12 | hardwire | yeh |
02:13.16 | hardwire | what versino of * ? |
02:13.19 | *** join/#asterisk cp5 (n=samy@adsl-69-232-108-255.dsl.irvnca.pacbell.net) |
02:13.22 | jskcr | anyone use the zip4x5? |
02:13.25 | TheCops | 1.0.9 |
02:13.31 | hardwire | TheCops: can't help yah :) |
02:13.38 | hardwire | well |
02:13.40 | hardwire | I sorta can |
02:13.41 | cp5 | anyone ever have the agent module (with persistence) deadlock on them? |
02:13.41 | TheCops | you are using 1.2.0 ? |
02:13.44 | hardwire | let me get you something |
02:13.57 | cp5 | or know of issues with chan_agent.so deadlocking in previous CVS versions? |
02:13.59 | TheCops | I'm reading about chaing the SIP header, you can make the phone to answer |
02:14.01 | hardwire | exten => s,1,SetVar(VXML_URL=intercom=true) |
02:14.14 | Hogie | TheCops: what type of phone are you using? |
02:14.15 | hardwire | exten => s,2,ChanIsAvail(Sip/${ARG1}|s) |
02:14.18 | *** join/#asterisk dev2008 (n=dev2003@222.33.36.205) |
02:14.19 | *** join/#asterisk Heng (n=hengff@61.6.65.226) |
02:14.21 | hardwire | TheCops: use that setvar |
02:14.25 | hardwire | and use chanisavail |
02:14.30 | hardwire | and then you should be fine |
02:14.32 | TheCops | Hogie, snom 320 |
02:14.33 | hardwire | the setvar sets the intercom |
02:14.41 | hardwire | and chanisavail checks to see if its in use |
02:14.43 | TheCops | hardwire, and after, I need to dial the phone ? |
02:14.44 | hardwire | otherwise you hose the 320 |
02:14.51 | hardwire | just dial the phone after that |
02:14.57 | TheCops | nice |
02:14.58 | hardwire | exten => s,3,Dial(Sip/${ARG1},12,TtA(beep)) |
02:14.59 | Hogie | damn, that's cool |
02:15.03 | hardwire | if you were using 1.2.0 |
02:15.09 | Hogie | I wish our c7960's would do that |
02:15.16 | hardwire | exten => _*[123]xxx,1,SipAddHeader(Call-Info: 209.112.194.40\; answer-after=0) |
02:15.16 | hardwire | exten => _*[123]xxx,n,ChanIsAvail(Sip/${EXTEN:1}|s) |
02:15.18 | hardwire | exten => _*[123]xxx,n,NoOp(${AVAILSTATUS}) |
02:15.19 | hardwire | exten => _*[123]xxx,n,Cut(ICHANNEL=AVAILCHAN,,1) |
02:15.19 | hardwire | exten => _*[123]xxx,n,Dial(${ICHANNEL},12,Tt) |
02:15.19 | hardwire | exten => _*[123]xxx,n,Hangup() |
02:15.29 | hardwire | that does it the snom compliant way |
02:15.32 | hardwire | where the snom beeps |
02:15.36 | hardwire | not your * box |
02:15.44 | hardwire | don't dos my IP |
02:15.48 | TheCops | :) |
02:15.58 | hardwire | Call-Info: can be 0.0.0.0 for the IP |
02:16.04 | dev2008 | E400P port2 and port 4 can not work |
02:16.04 | hardwire | but thats freaking verbatim |
02:16.12 | dev2008 | http://lists.digium.com/pipermail/asterisk-dev/2005-September/015681.html |
02:16.15 | hardwire | Hogie: you can do that really easily with c7960's |
02:17.45 | hardwire | AFAIK it works with SipAddHeader(Call-Info: \; answer-after=0 |
02:18.30 | hardwire | but if not just add a new line to it |
02:18.30 | hardwire | 1 + the first line extension |
02:18.30 | hardwire | and make new sip peers for it |
02:18.30 | hardwire | and make that second line auto answer |
02:18.30 | Hogie | that's how I have it now |
02:18.30 | Hogie | and its on 3rd line |
02:18.31 | Hogie | intercom_XXX where XXX is the extension |
02:18.34 | Hogie | is the peer |
02:19.11 | hardwire | yeh |
02:19.11 | TheCops | Incoming call: Got SIP response 489 "Bad Event" back from 192.168.50.107 <--- did you already seen this error hardwire with snom ? |
02:19.17 | hardwire | try SipAddHeader(Call-Info: \; answer-after=0) |
02:19.21 | hardwire | on line 1 |
02:19.23 | hardwire | see if that works |
02:19.27 | hardwire | TheCops: yeh |
02:19.33 | hardwire | you sorta also have to do a lot of crap tot he snom |
02:19.37 | *** join/#asterisk thiboxnk (n=thiboxnk@66-188-89-49.dhcp.mdsn.wi.charter.com) |
02:19.42 | TheCops | After a reload |
02:19.46 | TheCops | I see this |
02:19.52 | hardwire | Set "Filter Packets from Registrar" to OFF |
02:19.52 | Hogie | hardwire: is SipAddHeader something newly added? |
02:19.57 | hardwire | 1.2.0 and CVS |
02:20.23 | Hogie | ah, not upthat far |
02:20.29 | hardwire | as well as under line 1.. sip.. support broken registrar on and long-sip-contact off |
02:20.33 | hardwire | got that TheCops |
02:20.34 | Hogie | and... if I upgrade, its gonna break half my dial plan |
02:20.47 | TheCops | thanks |
02:20.47 | hardwire | Hogie: yeh of course |
02:20.49 | thiboxnk | need help.. having dropped calls during call waiting ne1 have problems with call waiting? |
02:21.26 | Hogie | I dont have any problems right now, and I dont really have time to upgrade our 2 * boxes |
02:21.43 | TheCops | hardwire, the setvar command dont work, the phone still ring |
02:21.44 | TheCops | hehe |
02:21.46 | bkw_ | OH thats rich... callerid on em_w is broken now |
02:23.27 | *** join/#asterisk konfuzed (n=konfuzed@H129.C72.B0.tor.eicat.ca) |
02:23.30 | hardwire | TheCops: its a pain |
02:24.00 | Hogie | thanks for the info hardwire |
02:24.30 | hardwire | I just wish devstate was working correctly in CVS |
02:24.35 | hardwire | so my snom leds would be happy |
02:24.53 | TheCops | in 1.2.0 ? |
02:25.02 | hardwire | CVS > 1.2.0 |
02:25.20 | hardwire | and ChanIsAvail is broken in CVS :( |
02:25.25 | TheCops | hardwire, duh |
02:25.35 | TheCops | devstate come by default in CVS? |
02:25.40 | hardwire | not at all |
02:25.45 | hardwire | hence it no workie |
02:26.41 | bkw_ | chan_zap.c:4561 zt_read: DTMF digit: * on Zap/73-1 |
02:26.42 | *** join/#asterisk uma1 (n=sudhir@pool-71-114-84-37.washdc.dsl-w.verizon.net) |
02:26.43 | bkw_ | haha |
02:26.44 | bkw_ | thats rich |
02:27.33 | hardwire | bkw_: get out more I guess |
02:27.47 | hardwire | heh |
02:28.00 | hardwire | there should be a "CVS Reliability" extension via IAXtel |
02:28.06 | hardwire | where it accepts 1 through 9 |
02:28.24 | hardwire | then we just plot it on a chart :) |
02:30.15 | dev2008 | who is familiar with tormenta2.vhd? |
02:30.56 | thiboxnk | wow all this * knowledge and nobody has had a problem with call waiting.. maybe everyone in the world ids using ip phones not ata's or channel banks so they dont have these problems |
02:31.11 | TheCops | Executing SIPAddHeader("SIP/105-5f44", "Call-Info: 0.0.0.0; answer-after=0") in new stack |
02:31.12 | TheCops | Ho yeah |
02:31.14 | TheCops | hardwire |
02:31.17 | TheCops | all is working proprely |
02:31.18 | TheCops | nice! |
02:31.20 | Saaib | umhh.. why is that /var/tmp is hardcoded into channels/chan_iax2.c instead of using env(TMP) or env(TMPDIR) ? |
02:31.34 | hardwire | oh |
02:31.39 | hardwire | when did you get SipAddHeader? |
02:31.53 | TheCops | What you mean ? |
02:31.57 | hardwire | Saaib: to piss you off |
02:32.01 | hardwire | TheCops: I thought you were 1.0.9 |
02:32.05 | TheCops | yeah I am |
02:32.09 | hardwire | ... |
02:32.12 | hardwire | ok |
02:32.16 | TheCops | Ho god |
02:32.17 | hardwire | its heavily patched then IMHO |
02:32.17 | TheCops | no |
02:32.21 | TheCops | you are right |
02:32.26 | TheCops | god it is my friend |
02:32.31 | TheCops | Ha god |
02:32.33 | hardwire | ? |
02:32.37 | hardwire | are you on the drugs? |
02:32.46 | Saaib | hardwire: lol :P i have a very restrictive system and just found out that asterisk wont work with /var/tmp |
02:32.54 | TheCops | yeah, but it's not that, it's a friend of mine who have changed the version |
02:32.57 | Saaib | guess i'll replace that |
02:33.11 | TheCops | that's why my led dont work at all |
02:33.29 | TheCops | god he have changed a production version |
02:33.35 | TheCops | stupid guys |
02:34.18 | TheCops | hardwire, but, you told me before ChanIsAvail is not working on CVS |
02:34.28 | TheCops | and I'm using it in my config and it's working perfectly |
02:35.00 | thiboxnk | ok what has everyone been doing for MTU/MDU apartments ? |
02:36.07 | hardwire | TheCops: as of an hour ago? |
02:36.13 | hardwire | you are using the CVS? |
02:36.18 | TheCops | yeah |
02:36.33 | hardwire | its not working worth a damn here |
02:36.55 | TheCops | when I'm using chanisavail(ZAP/3&ZAP/3) |
02:37.02 | TheCops | it do what it is suposed to do |
02:37.35 | TheCops | ho...but I have problem with it fail (I mean, when these to channel are in use) it dont use the n+101 priority |
02:37.46 | *** join/#asterisk flenders (n=fserto@61.8.29.101) |
02:37.50 | TheCops | but the major feature is ok |
02:38.38 | hardwire | TheCops: I use the |s option |
02:38.39 | hardwire | at the end |
02:38.40 | hardwire | add that |
02:38.46 | hardwire | chanisavail(ZAP/3&ZAP/3|s) |
02:38.48 | hardwire | see |
02:38.51 | TheCops | What is the S options ? |
02:39.11 | hardwire | If the option 's' is specified (state), will consider channel unavailable |
02:39.11 | hardwire | when the channel is in use at all, even if it can take another call. |
02:39.20 | hardwire | see |
02:39.27 | hardwire | if you intercom a snom |
02:39.35 | hardwire | and the person is on the line.. eall waiting or not |
02:39.37 | hardwire | the intercom takes over |
02:39.39 | hardwire | and drops the call |
02:39.49 | hardwire | thats why its imperative to use chanisavail |
02:39.57 | TheCops | LOL |
02:39.57 | hardwire | so for now paging is off in my office |
02:40.03 | hardwire | otherwise.. if you do a mass page |
02:40.07 | TheCops | you're right |
02:40.09 | TheCops | just tried |
02:40.09 | hardwire | you do a mass call drop |
02:40.15 | hardwire | so add the s option |
02:40.16 | TheCops | the secretary was on phone! |
02:40.16 | hardwire | and try again |
02:40.17 | TheCops | lol |
02:40.57 | file[laptop] | anyone who is using my pickup app owes me a beer |
02:41.02 | file[laptop] | because I say so. |
02:41.07 | hardwire | file[laptop]: I can't voip you a damn beer |
02:41.10 | hardwire | otherwise I would |
02:41.12 | hardwire | and it works fine |
02:41.13 | bkw_ | better watch it |
02:41.14 | hardwire | its too simple to break |
02:41.22 | hardwire | bkw_: it cannot be done |
02:41.42 | bkw_ | I faxed 26 pages over IAX which kinda shocked me |
02:41.48 | hardwire | nice |
02:41.52 | FuriousGeorge | i know this is sooo 1/2 hour ago, but i used to work for a lawfirm that played a cd with "o fortuna" on it (the song that the movies play when the devil's about) anyone got a link to a public domain cover of that one :) |
02:41.53 | hardwire | none of them was a beer |
02:41.58 | thiboxnk | seems similar to what im experiencing .. im in a caal .. caller calls in on call waiting.. if i ignore it will disconnect all calls.. seems broken? |
02:42.14 | hardwire | FuriousGeorge: I have a midi om fortuna |
02:42.19 | hardwire | o, fortuna :) |
02:42.30 | Johnsie | Rather appropriate for a law firm. |
02:42.32 | hardwire | just use timidity to convert it :) |
02:42.46 | hardwire | what about rebroadcasting FM radio |
02:42.50 | hardwire | everybody and their dog does it |
02:42.56 | hardwire | why is nobodie or their dog getting busted? |
02:43.03 | hardwire | man my spelling could use some work |
02:43.09 | FuriousGeorge | hardwire: we collected debt :) |
02:43.19 | hardwire | FuriousGeorge: you got pinned? |
02:43.20 | *** join/#asterisk krisguy (n=krisguy@h216-170-039-057.adsl.navix.net) |
02:43.27 | spackle | hardwire, they do, they just get slapped with a fine, not carries away to jail like the RIAA would like |
02:43.33 | hardwire | you know another thing you can di is explicit permission of the artist |
02:43.40 | hardwire | we have that for the native music we use |
02:43.43 | FuriousGeorge | hardwire: i wonder the same thing, but then i start thinking: i dont see that as much as i used to |
02:43.47 | *** join/#asterisk AgiNamu (n=Michael@dsl081-096-215.den1.dsl.speakeasy.net) |
02:44.06 | FuriousGeorge | hardwire: i always wanted to write george clinton a letter |
02:44.07 | AgiNamu | wtf is ISDN FAS? |
02:44.09 | wolfson | rebroadcasting fm radio via MOH is just as bad |
02:44.14 | FuriousGeorge | he didnt od yet did he? |
02:44.31 | FuriousGeorge | o.d.* |
02:44.58 | hardwire | FuriousGeorge: what brings that up? |
02:45.08 | hardwire | also because its MoH for a company .. its illegal? |
02:45.21 | thiboxnk | AgiNamu isdn Nfas is usually when 2 pri's are controlled by the same d channels |
02:45.27 | hardwire | for instance my Rammstein MoH for my house isn't illegal because I don't plan to turn a profit with whoever is calling me? |
02:45.37 | FuriousGeorge | hardwire: i brought that up b/c u said i could get explicit permission from the artist |
02:45.42 | hardwire | thiboxnk: that sounds fun |
02:45.57 | hardwire | thiboxnk: how used up is the d-channel for most t1s? |
02:46.06 | hardwire | FuriousGeorge: oh |
02:46.07 | hardwire | George |
02:46.09 | hardwire | Funk |
02:46.16 | hardwire | jesus.. I thought Bill |
02:46.20 | hardwire | that made 0 sense |
02:46.27 | hardwire | unless you mean you have sax music |
02:46.30 | dev2008 | who use E400P? |
02:46.31 | hardwire | or sex music |
02:46.33 | hardwire | either way |
02:46.34 | thiboxnk | actually it is from the old dialin modem days for multiple PRI's doing dialin with a hunt |
02:46.42 | *** join/#asterisk ManxPower (i=eric@110.sub-70-197-211.myvzw.com) |
02:46.43 | AgiNamu | thiboxnk, but there's no N. my GC cutsheet says "Trunk type: ISDN FAS" |
02:46.48 | FuriousGeorge | Dear George, Can i get it on the good foot? Ha Hah! (feet don't fail me now) |
02:46.53 | hardwire | thiboxnk: to get that one extra channel? |
02:47.02 | hardwire | hmm |
02:47.04 | thiboxnk | so you dont waste a channel for the pri signalling on multiple PRI's |
02:47.13 | *** join/#asterisk hellagony (n=hellagon@c-24-130-45-125.hsd1.ca.comcast.net) |
02:47.20 | hardwire | you sorta got your artists mixed up :) |
02:47.32 | FuriousGeorge | no i didnt |
02:47.37 | hardwire | James brown != George Clinton |
02:47.38 | thiboxnk | is this a BRI or PRI |
02:47.46 | mmlj4 | oh, the irony... i'm working with phones now, yet i loathe talking on them |
02:47.49 | AgiNamu | PRI |
02:47.52 | FuriousGeorge | hardwire: that song is... im thinking |
02:48.00 | AgiNamu | thiboxnk, you have any exp. with a 5350? |
02:48.05 | thiboxnk | SINGLE OR MULTIPLE |
02:48.11 | spackle | mmlj4, I feel exactly the same way |
02:48.18 | hardwire | FuriousGeorge: thinking or googling |
02:48.28 | hardwire | http://letrasdecanciones.tomamusica.com/i/Ice-Cube-F-George-Clinton/BopGunOneNation_20339.htm |
02:48.30 | AgiNamu | im willing to hire someone that understands either PRI or 5350 :P |
02:48.34 | FuriousGeorge | i think its atomic dog |
02:48.34 | AgiNamu | its a single PRI |
02:48.43 | FuriousGeorge | im thinking, i know the lyrics i dont know the titile |
02:49.00 | thiboxnk | k what hardware are you using and what country/ switch type/ etc.. |
02:49.08 | *** join/#asterisk azzie (i=az@cpe-24-168-17-173.si.res.rr.com) |
02:49.21 | FuriousGeorge | now im googling |
02:49.29 | hardwire | FuriousGeorge: I already posted it |
02:49.31 | azzie | anybody can suggest a descent cisco sip phone 79xx ? |
02:49.40 | hardwire | azzie: decent.. |
02:49.41 | hardwire | hmm |
02:49.45 | hardwire | Granstream :) |
02:49.49 | hardwire | Grandstream |
02:49.50 | hardwire | Polycom |
02:49.58 | FuriousGeorge | not atomic dog |
02:50.05 | hardwire | Bop Gun. |
02:50.09 | bkw_ | can i have someone try to fax 866 536 7405 |
02:50.13 | FuriousGeorge | thats Ice Cube |
02:50.14 | mmlj4 | someone wanna call me and tell me how horrible my setup is? |
02:50.15 | azzie | hardwire: so, which cisco would that be? :) |
02:50.17 | *** join/#asterisk MikeJ[Laptop] (n=ircatjer@d14-69-8-30.try.wideopenwest.com) |
02:50.25 | hardwire | azzie: are you shitting with me? |
02:50.32 | azzie | hardwire? |
02:50.45 | hardwire | bkw_: uno momento |
02:50.57 | bkw_ | not sure I have all my clocking correct |
02:50.59 | FuriousGeorge | one nation under one groove |
02:51.48 | cp5 | anyone ever experience agent/queue deadlocks with cvs? |
02:52.04 | azzie | hardwire, can't be they are worth than GS :)) |
02:52.37 | mmlj4 | polycoms are cool, grandstreams are poop |
02:53.31 | hardwire | bkw_: you have been faxed |
02:53.46 | bkw_ | you wanna make a bet :P |
02:53.49 | bkw_ | I don't htink it went thru |
02:53.49 | bkw_ | haha |
02:53.53 | hardwire | mo fo |
02:53.58 | mog_home | faxing so old school |
02:53.59 | mog_home | .... |
02:54.00 | hardwire | I better go get the tx results from my fax |
02:54.03 | hardwire | its naughty |
02:54.05 | hardwire | don't wanna get fired |
02:54.36 | azzie | mmlj4, worth than GS is only windows messenger :-) |
02:54.43 | mmlj4 | heh |
02:55.06 | mmlj4 | so someone call me and gripe about my call quality: 504-272-2060 |
02:55.47 | azzie | csim start 15042722060 :) |
02:56.18 | mmlj4 | no workie? |
02:57.11 | mmlj4 | hey manx |
02:57.19 | mmlj4 | so call me, that's my teliax number |
02:57.46 | spackle | bkw_ you gettin too many faxes now? |
02:57.54 | mmlj4 | i can mostly call out, except this lousy grandstream refuses to think it's authorized to place calls |
02:58.05 | bkw_ | I don't think any of those worked |
02:58.09 | bkw_ | now that pisses me off |
02:58.24 | bkw_ | I think the clocking is off |
02:58.28 | bkw_ | let me try something else |
02:58.43 | ManxPower | Um, I left Texarkana today went thru Alexandria, Lafayette, Baton Rouge, to Hammond, met John in Hammond, then went back thru Baton Route to Lafayette where I'm staying. Getting up at 6am in the morning to go back to Covington |
02:58.57 | mmlj4 | ugh, what a road trip |
02:59.03 | ManxPower | mmlj4, Yeah. |
02:59.11 | mmlj4 | so, has UMC gotten a server ordered yet? |
02:59.11 | ManxPower | Left at 6am |
02:59.23 | ManxPower | mmlj4, no idea, they have not talked to me about it. |
02:59.28 | ManxPower | and they have my cell phone number. |
02:59.30 | bkw_ | ok changed the clock priority |
02:59.35 | hardwire | bkw_: busy |
02:59.40 | ManxPower | Last time I talked to them was about 1.5 weeks ago when I got tranks from them |
02:59.40 | hardwire | you gave me a bad number |
02:59.44 | mmlj4 | they keep thinking that one's been ordered |
02:59.45 | bkw_ | doubt ful |
02:59.47 | bkw_ | I just bounced the PRI |
02:59.58 | bkw_ | 8665367405 |
03:00.06 | ManxPower | mmlj4, The telco won't be able to even deal with converting the lines for at least a month |
03:00.15 | hardwire | bkw_: I am in alaska |
03:00.16 | mmlj4 | heh, nice |
03:00.19 | hardwire | thats why it isn't working |
03:00.24 | mmlj4 | and they think they're opening in a week |
03:00.25 | hardwire | duh |
03:00.33 | ManxPower | mmlj4, That's a optimistic estimate. |
03:00.41 | ManxPower | brb smoke break |
03:00.47 | mmlj4 | i smell a link to covington and a lot of transferred calls |
03:01.02 | bkw_ | ok this puzzles me to no end now |
03:01.36 | TheCops | hardwire, in 1.0.9 with setvar, intercom is not working |
03:01.36 | hardwire | puzzle muzzle guzzle fuzzle |
03:01.36 | bkw_ | MUHAHA |
03:01.37 | bkw_ | that was it |
03:01.43 | hardwire | TheCops: it happens |
03:01.44 | bkw_ | the clock priority in the TNT was wrong |
03:01.47 | hardwire | its a tricky pickle |
03:01.54 | hardwire | and you need to patch 1.0.9 |
03:01.56 | hardwire | I forgot :) |
03:01.59 | hardwire | and move a %s around |
03:02.06 | TheCops | ho |
03:02.06 | TheCops | lol |
03:02.10 | TheCops | I'm not a coder hehe |
03:02.18 | TheCops | I'll wait 1.2.0 release |
03:02.23 | hardwire | I haven't had papa johns in forever |
03:02.45 | mmlj4 | hardwire: not long enough, in my book |
03:02.46 | bkw_ | XXX Missing handling for mandatory IE 8 (cs0, Cause) XXX |
03:02.51 | hardwire | I curse on ChanIsAvail |
03:02.51 | X-Rob | TheCops - what sorta phone? |
03:02.52 | bkw_ | Love that one |
03:03.02 | TheCops | X-Rob snom 320 |
03:03.03 | hardwire | mmlj4: been goign to this place down the street for so long |
03:03.09 | X-Rob | http://www.voip-info.org/tiki-index.php?page=Asterisk+Paging+and+Intercom |
03:03.11 | hardwire | papa johns heavy carbs is probably going to kill me |
03:03.12 | X-Rob | See 'allpage.agi' |
03:03.15 | X-Rob | I've _just_ put it there. |
03:03.19 | hardwire | X-Rob: oh have you now |
03:03.25 | hardwire | I think I am going to one up you X-Rob |
03:03.30 | hardwire | if you can deal with the heat |
03:03.33 | X-Rob | hardwire - woot. |
03:03.36 | hardwire | woot |
03:03.37 | X-Rob | you go! |
03:03.41 | hardwire | I need to contribute to asterisk damnit |
03:03.50 | hardwire | so app_paging_system_of_doom.c will have to exist |
03:04.01 | X-Rob | well. |
03:04.06 | X-Rob | join #openpbx |
03:04.36 | Johnsie | hahaha |
03:07.16 | BrianR___ | Anyone know if palm PDA's can play wav49 encoded voicemail attachments? |
03:08.03 | *** join/#asterisk krisguy (n=krisguy@h216-170-039-057.adsl.navix.net) |
03:08.06 | FuriousGeorge | can anyone think of a good reason to use call parking when your client supports 5 conversations? |
03:08.27 | X-Rob | park call. page all systems 'Fred. Call Parked on 72. Fred. Call parked on 72' |
03:08.40 | FuriousGeorge | thanks x-rob |
03:08.48 | BrianR___ | FuriousGeorge: Moving a support call from your desk to the server room |
03:08.51 | ManxPower | mmlj4, on the bright side I *might* have broadband by saturday. |
03:08.57 | mmlj4 | cool |
03:09.05 | MikeJ[Laptop] | and monkeys |
03:09.20 | BrianR___ | FuriousGeorge: You can't just use transfer in that case because you won't be there yet when the phone starts ringing. |
03:09.31 | ManxPower | tranks = good |
03:09.39 | FuriousGeorge | BrianR___: yeah, i figured that was the advantage |
03:09.51 | mmlj4 | the only PC parts wholesaler in NOLA can't get DSL nor cable, they're hoping for wireless |
03:10.23 | mmlj4 | ManxPower: so call me, i need you to laugh at how horrible my call quality is: 504-272-2060 |
03:11.11 | FuriousGeorge | the only reserved numerical extension is 0, right? technically i could set up the parked calls to be on 1-20, for instance? the reason i ask is b/c these people are used to an old shcool switched phone system where they had a line 7 which corresponds to an actual telco did |
03:11.43 | MikeJ[Laptop] | key systems are great for that |
03:11.46 | MikeJ[Laptop] | :P |
03:11.50 | MikeJ[Laptop] | night all. |
03:11.53 | FuriousGeorge | night |
03:12.06 | spackle | sleep tight, don't let bed bugs bite |
03:12.18 | FuriousGeorge | though in a sense that is good b/c everyone gets a headsup status as to where there is a call. if its blinking its on hold, etc |
03:12.26 | MikeJ[Laptop] | the only bug in my bed is my wife ;) |
03:12.30 | spackle | if they do report them to mantis.digium.com |
03:12.42 | MikeJ[Laptop] | bugs.digium.com |
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03:13.25 | FuriousGeorge | is there any softphone which supports mapping status lights to * parked call slots? does asterisk support that? |
03:13.57 | websae | anyone here using MySql with Asterisk? |
03:14.00 | websae | having some issues... |
03:14.54 | *** part/#asterisk spackle (n=spackle@209.234.83.19) |
03:15.02 | hardwire | FuriousGeorge: yes and no |
03:15.16 | hardwire | I don't know any soft phones that do that |
03:15.23 | hardwire | other than the snom 360 softphone |
03:15.27 | hardwire | which you can download |
03:15.33 | hardwire | and set destination functino keys |
03:15.36 | hardwire | function keys |
03:15.40 | *** join/#asterisk argos73 (n=argos@65-85-207-125.client.dsl.net) |
03:15.43 | hardwire | however.. getting that to work with asterisk is a headache |
03:15.56 | blitzrage | evening all |
03:15.57 | hardwire | I can only get the snom to subscribe once and get a report on all extensions |
03:16.01 | hardwire | and then the leds' just stic |
03:16.02 | hardwire | k |
03:16.07 | hardwire | blitzrage: mornin |
03:16.27 | file | hey blitzrage |
03:16.37 | hardwire | well I was all set to fax bkw some porn |
03:16.44 | blitzrage | hardware, file. |
03:16.44 | hardwire | but his toll-free inbound hates AK |
03:16.48 | blitzrage | how goes? |
03:16.49 | *** join/#asterisk spackle (n=spackle@209.234.83.19) |
03:17.53 | blitzrage | good to hear - just playing with my new printer/scanner which I got last week (finally got a USB cable) |
03:18.15 | blitzrage | actually printing and scanning and signing all these NDAs I had but couldn't do anything with since I didn't have a printer :) |
03:18.17 | file | USB is overrated |
03:18.18 | ManxPower | most tollfrees hate AK |
03:18.22 | file | bluetooth is where it's at |
03:18.55 | blitzrage | I need a better scanning program... Windows doesn't handle multiple pages :) |
03:22.38 | FuriousGeorge | whats the snazziest iaxphone? |
03:22.47 | FuriousGeorge | im looking for an opinion :) |
03:22.49 | thiboxnk | still ne help with call waitng call drops will be appreciated.. |
03:23.07 | spackle | FuriousGeorge: hardphone? |
03:23.14 | FuriousGeorge | spackle: softphone |
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03:24.09 | pauldy | anyone have any suggestions on the cheapest way to send out bulk fax |
03:24.50 | FuriousGeorge | spackle: ok hardphone |
03:24.53 | blitzrage | anyone who asks to send out bulk faxes is sketchy in my books :) |
03:24.56 | spackle | pauldy, have a service do it. |
03:25.06 | pauldy | blitzrage, it isn't spam |
03:25.16 | blitzrage | pauldy: haha, cool :D |
03:25.26 | pauldy | its contracts multiple pages maybe 10-20 a day |
03:25.31 | blitzrage | pauldy: I'm just being a pain really... continue about your business :) |
03:25.44 | blitzrage | pauldy: yah, that'd be a lot of faxes |
03:26.14 | pauldy | jfax killed their account this month |
03:26.26 | ManxPower | I got an authentic looking e-mail from "paypal" tonight. Was the standard notification of a purchase made for an expensive watch, shipped to beverly hills. Even had a nice "dispute" link, which sent you to a site that was NOT paypal. |
03:26.29 | pauldy | I looked at what they were paying I know there has to be a cheaper solution |
03:26.34 | ManxPower | damn fscking scammers |
03:27.05 | ManxPower | sleep now. |
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03:37.25 | drbrown | any opinons on the TE110P used in conjunction with a rhino channel bank???? specifically echo???? |
03:38.46 | bjohnson | pauldy: local or long distance |
03:39.28 | websae | my company provides termination at $.017/minute |
03:40.11 | pauldy | bjohnson, mostly local but a bit of it might be inside the metro area but long distance |
03:40.32 | bjohnson | and they use jfax for local faxing? |
03:40.39 | pauldy | yup |
03:40.42 | bjohnson | can't they just use a fax machine? |
03:41.02 | pauldy | trying to find a service they can be happy with for reliability or just build a hylafax server |
03:41.31 | pauldy | bjohnson, they could |
03:41.32 | bjohnson | why not a regular fax machine on a pstn? |
03:41.45 | pauldy | because that doesn't cut down on the paper |
03:41.51 | BrianR___ | Is there any way to answer calls coming in over PRI with hylafax on a box also running asterisk? |
03:41.53 | bjohnson | for $30/mo for pstn and $150 for a fax machine with memory |
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03:42.02 | pauldy | never been in a big office waiting on someones 40 page contract have you |
03:42.04 | bjohnson | pauldy: use hylafax for incoming |
03:42.22 | bjohnson | pauldy: more lines? |
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03:42.42 | pauldy | listen now your just second guessing my advice |
03:42.43 | bjohnson | actually we use mgetty for incoming |
03:43.09 | pauldy | the idea is a service that requires little effort on my part to switch them over to |
03:43.22 | pauldy | next best thing is roll my own |
03:43.36 | pauldy | I was hiopping someone else has been down this road before in here |
03:44.04 | bjohnson | I don't see how a fax machine is such a problem |
03:44.52 | pauldy | well I understand why they don't like it but I guess that comes from experience |
03:45.05 | BrianR___ | having only one fax machine in a busy office sucks hardcore... |
03:45.17 | bjohnson | so get more than one |
03:45.24 | bjohnson | use it for outgoing only |
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03:45.54 | bjohnson | any other option requires original docs in electronic format .. not something that is acceptable in all offices |
03:45.59 | BrianR___ | fax machines are best reserved for outgoing only... |
03:46.02 | bjohnson | (or easy) |
03:46.49 | bjohnson | we use a fax machine for outgoing and a linux box for incoming |
03:46.54 | bjohnson | on a shared line |
03:47.14 | bjohnson | but maybe we don't do as much volume |
03:47.16 | BrianR___ | We're using multifunction copiers at my office... Take up two analog lines, but they're decent if someone sends a huge fax. Can also do outbound faxes via an email gateway - would like to get that set up.. |
03:47.42 | BrianR___ | I have a pri terminated on my asterisk box, would like to give every employee a direct dial fax->email for inbound... But app rxfax doesn't work so well. |
03:47.44 | bjohnson | a simple fact is that faxes take time, if one is too slow, the easiest solution is to use two |
03:48.34 | BrianR___ | bjohnson: Well.. If you're sending the faxes via an email gateway on a box with a PRI, you can run dozens of faxes at once... |
03:49.10 | BrianR___ | ditto for big office fax machines that can take multiple analog lines |
03:49.18 | bjohnson | what hardware sends faxes over PRI? |
03:49.18 | pauldy | not only that but the reciept comes to your e-mail and all sorts of nice things that just make it easier |
03:49.25 | websae | anyone know of a good DID provider? |
03:49.33 | pauldy | The copier thing might be a solution I had not thought of that |
03:49.45 | bjohnson | pauldy: if you have to scan in documents before faxing them, email gateways are not easier |
03:49.55 | pauldy | if they can get it on a lease and replace the copier they have it might work out to be worthwhile |
03:49.55 | BrianR___ | bjohnson: The combination of a multifunction copier with fax to email and a hylafax email to fax gateway... |
03:50.18 | bjohnson | and fax to PRI? |
03:50.24 | bjohnson | modems? |
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03:50.39 | BrianR___ | The toshiba copiers we're playing with can tack an email suffix onto any unqualified address the user enters, so it's nice and transparent for the end user. |
03:50.55 | BrianR___ | bjohnson: softmodem running on the asterisk box - either spandsp or t38modem |
03:51.04 | pauldy | BrianR___, do you happen to have the model info? |
03:51.08 | bjohnson | how stable are those now? |
03:51.21 | BrianR___ | pauldy: estudio 350, I think... |
03:51.27 | pauldy | ty |
03:51.30 | BrianR___ | bjohnson: My testing with spandsp so far shows it sucks |
03:51.41 | BrianR___ | About to do some testing with t38modem - word is it's much more mature. |
03:51.44 | bjohnson | so not really a solution then |
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03:52.00 | bjohnson | might be at some point .. but not now |
03:53.37 | FuriousGeorge | is cable generally a better voip solution than dsl? |
03:53.44 | pauldy | BrianR___, the copier you have used does it do email replies for the fax transmission results etc.. |
03:53.57 | FuriousGeorge | can we generalize about latencies and bandwidth in the US48 |
03:54.25 | blitzrage | OT: anyone know of some good software for linux that I can use to monitor traffic in a network? Ideally I'd like to break it down based on IP address (internal) |
03:54.27 | pauldy | FuriousGeorge, my guess is it would be move of a debate over pppoe vs dhcp |
03:54.37 | blitzrage | by monitor traffic I mean usage |
03:54.45 | blitzrage | pretty graphs would be nice too :) |
03:54.51 | spackle | blitzrage: ntop? |
03:54.57 | bjohnson | blitzrage: iptables with mrtg |
03:55.02 | BrianR___ | pauldy: haven't played with it enough - looks like the user has to enter their email address if they want a receipt. |
03:55.18 | FuriousGeorge | pauldy: is that what its about? i dont really care about authentication, im more worried about latencies and whatnot. let me do a little experiment |
03:55.19 | bjohnson | ntop doesn't do bandwidth accounting which I think he wants |
03:55.26 | blitzrage | bjohnson: ahhh yes, I used MRTG once... thanks |
03:55.36 | bjohnson | mrtg just graphs |
03:55.47 | bjohnson | you needs data for it to make the graphs from |
03:55.48 | blitzrage | yah.. bandwidth accounting is what I want - need to watch how much the room mates are downloading |
03:56.03 | bjohnson | most ipaccounting stuff on linux uses iptables as the back end |
03:56.13 | pauldy | pppoe inherently introduces more latency as packets must be filtered |
03:56.20 | blitzrage | bjohnson: hrmmm... that might be a bit of a problem since I"m already running gShield... |
03:56.34 | bjohnson | blitzrage: if you have an extra box, ipcop them into a separate subnet |
03:56.36 | pauldy | even if they use a method similar to the cutthrough method on switches it is still that extra milisecond |
03:57.03 | blitzrage | bjohnson: unfortunately no extra box - just a single exit point - needs to do all :( |
03:57.13 | FuriousGeorge | i got this one dsl connection. i got a download on an ftp going on @ 40kBps and the pings to google are around 1 second |
03:57.32 | bjohnson | blitzrage: if the exit point is a linux box, you're laughing |
03:57.43 | blitzrage | bjohnson: it is |
03:57.44 | bjohnson | otherwise you're out of luck |
03:58.18 | websae | anyone know of a good DID provider....? |
03:58.35 | bjohnson | make iptables rules so that data to/from their IPs go to custom tables, the tables will log the data transfered through them |
03:58.53 | blitzrage | websae: www.mixnetworks.com (yes, I work for them) |
03:59.13 | bjohnson | then you use mrtg to track the differences in the kB count over time increments |
03:59.25 | bjohnson | and display them |
03:59.37 | BrianR___ | it looks like t38modem is a softmodem which exposes a h323 interface on one side and a pty on the other end, so you can use hylafax even if asterisk is controlling your PRI. Very slick. |
04:00.19 | blitzrage | bjohnson: hrmmm... seems like more work than I'm able to spend right now :( |
04:01.03 | pauldy | blitzrage, see if ntop pushes your buttons |
04:01.22 | pauldy | it does do some abdnwidth accounting |
04:01.40 | pauldy | not per minute but cumulative |
04:01.43 | *** join/#asterisk cio (n=na@adsl-068-209-198-242.sip.bhm.bellsouth.net) |
04:02.07 | *** join/#asterisk pashah (n=pashah@ns.itconnection.ru) |
04:02.13 | pashah | hello |
04:02.24 | *** join/#asterisk ikey (i=ikey@220.226.28.95) |
04:02.34 | cio | Hi all. I'm hearing some chirping on the line in my * system, it just seemed to start a few days ago. Any ideas what may be causing it? I'm using Polycom IP phones. This is a localnet phone to asterisk to 1fb call out. |
04:02.44 | pashah | could anyone please tell me if te205p has a jumper to toggle t1/e1 |
04:02.56 | JunK-Y | unlock the PAP2-CA wohooo. |
04:03.09 | pauldy | cio, did you make sure you don't have any rats in your * box |
04:03.19 | spackle | or crickets |
04:03.26 | JamesDotCom | pashah: stop being so fucking lazy, http://www.digium.com/index.php?menu=product_detail&category=hardware&product=TE205P |
04:03.30 | cio | No, nothing like that ... |
04:03.33 | cio | heh |
04:03.40 | *** join/#asterisk argos73 (n=mike@65-85-207-101.client.dsl.net) |
04:03.40 | hypa7ia | blitzrage: TAUG! |
04:03.48 | BrianR___ | I suppose I could use hylafax normally if I put a two port ISDN card in the box and used a crossover cable... |
04:03.59 | blitzrage | pauldy: thats fine... what I'm looking for is just to track how much bandwidth each user on the network is using so I can blame someone for the 60GB of b/w used so far this month :) |
04:04.03 | cio | Any other suggestions? |
04:04.05 | pashah | jamesDotCom: it does not say if it has a jumper |
04:04.10 | blitzrage | hypa7ia: w00t! I found two people to carpool with tomorrow! |
04:04.17 | hypa7ia | blitzrage: nice! |
04:04.23 | pashah | JamesDotCom: have you seen the card? |
04:04.27 | hypa7ia | i hope we move it next month |
04:04.32 | hypa7ia | i'm missing TASK |
04:04.35 | JamesDotCom | "The TE205P supports both E1 and T1/J1 environments and is selectable on a per-card or per-port basis" |
04:04.48 | hypa7ia | and i forgot to call Mix today but i will tomorrow :-) |
04:05.00 | cio | On a completely different note, is there anyway to improve the quality of the standard asterisk voice prompts? If I record prompts on my own, the sound is crisp and clear as I record at 16k, but the stock recordings are only 8k I think. |
04:05.04 | *** join/#asterisk ronaldl79 (n=ronaldl7@c-24-8-54-203.hsd1.co.comcast.net) |
04:05.07 | blitzrage | hypa7ia: I doubt it'll get moved yet again... :( |
04:05.15 | ronaldl79 | G'day, room. |
04:05.21 | pashah | JamesDotCom: so you have not seen it? but it should have jumper similar to te110p i presume |
04:05.33 | blitzrage | hypa7ia: we actually just moved it this month :) |
04:05.37 | JamesDotCom | pashah: i see the jumpers i'm sure they are on the picture on that page |
04:05.46 | blitzrage | hypa7ia: bad idea to keep moving it each month... I would assume |
04:05.50 | ronaldl79 | Q: Will the sound drivers for Gnome or KDE override the audio output of chan_oss?? |
04:05.52 | JamesDotCom | so yes |
04:05.58 | pashah | JamesDotCom: thanks |
04:05.59 | argos73 | pashah: just installed one of those cards today.. |
04:06.03 | ronaldl79 | ...for the console |
04:06.04 | hypa7ia | blitzrage: i know... it's like my career debate but in user group for, security or telecom?!? arrgh! |
04:06.29 | JamesDotCom | pashah: np, sorry for the anger in the initial response |
04:06.29 | JamesDotCom | ahah |
04:06.43 | pashah | JamesDotCom: np |
04:07.58 | pauldy | hey blitzrage I would just find the user with the empty lotion bottle and blame it on that one, pr0n |
04:08.26 | blitzrage | pauldy: LOL |
04:08.42 | joelsolanki | Hi jamesdotcom |
04:09.47 | JamesDotCom | hi joelsolanki |
04:09.52 | *** join/#asterisk santiago (n=santiago@63.245.86.245) |
04:09.56 | JamesDotCom | how did you go yesterady? i had to leave sorry |
04:10.10 | Cresl1n | ping |
04:10.13 | joelsolanki | Yes i waited for u long but u didnt came :) |
04:10.16 | cio | argh... chirping seems to be related to something else running on that linux box.. I stopped everything but * and the chirping goes away.. What sucks is I have to go one at a time and test... |
04:10.54 | joelsolanki | i too had to move out of city so just came today only. no progress on it. here is the pastebin for the ser+asterisk on same box. http://pastebin.ca/24028 |
04:14.16 | drumkilla | hypa7ia: ! |
04:14.35 | hypa7ia | drumkilla! |
04:14.39 | blitzrage | drumkilla: ! |
04:14.45 | blitzrage | Cresl1n: ! |
04:14.45 | drumkilla | blitzrage: ! |
04:14.53 | blitzrage | lol |
04:14.58 | blitzrage | that never gets old for me |
04:15.00 | Cresl1n | blitzrage!!!! |
04:15.05 | joelsolanki | JamesDotCom: got any clue ? |
04:15.09 | drumkilla | Cresl1n: !!!1! |
04:15.22 | Cresl1n | drumkilla!!!! |
04:15.24 | blitzrage | wasn't me! |
04:15.55 | hardwire | file: reload chan_sip.so.. reloads the lib? |
04:16.06 | file | it calls the reload function |
04:16.07 | hardwire | like.. new apps and so forth tada? |
04:16.10 | argos73 | there something funky I need to do to get incoming caller id to work with a channelized T1? works when it's plugged into the channel bank, but doesn't when I send those channels to asterisk... |
04:16.11 | hardwire | oh |
04:16.14 | hardwire | damn the reload function |
04:16.51 | hypa7ia | eeek! |
04:17.47 | joelsolanki | JamesDotCom: U there? |
04:23.29 | *** part/#asterisk da_monumental_1 (n=da_monum@rrcs-24-172-102-198.midsouth.biz.rr.com) |
04:27.14 | *** join/#asterisk philm (n=a@r43h15.res.gatech.edu) |
04:28.33 | philm | I'm having trouble calling the console. I dial->get music on hold->and have to hangup and answer before I can talk. Why the music on hold? |
04:30.29 | *** join/#asterisk websae (i=websae@207-118-147-212.dyn.centurytel.net) |
04:30.43 | *** join/#asterisk zapa (n=zapa@200.52.208.107) |
04:30.59 | zapa | hi all |
04:31.25 | philm | hi |
04:31.45 | philm | Zapa ever use console/dsp? |
04:31.57 | *** join/#asterisk heath__ (n=root@12-215-32-56.client.mchsi.com) |
04:32.09 | zapa | no philm sorry |
04:33.44 | ronaldl79 | Can anyone recommend a BYOD provider comparable to BroadVoice? |
04:34.50 | zapa | i am having troubles with a PRI30 telrad, to Digium Card when i call from telrad pbx to the trunk i only hear dial tone but asterisk answer the ivr, but when i call from asterisk to telrad i don't have any problem, any idea? |
04:35.33 | websae | ronald179--i have a suggestion, i messaged you |
04:37.38 | *** join/#asterisk justinnnnn (n=justinnn@61.95.68.85) |
04:37.39 | justinnnnn | hey |
04:37.42 | justinnnnn | does anyone no if voipjet is down ? |
04:37.50 | justinnnnn | i havnt been able to call through em for like 24 hours now... |
04:37.51 | justinnnnn | ??? |
04:38.14 | file[laptop] | have you e-mailed their fastsupport? |
04:38.23 | justinnnnn | nah they usualy just ignore it |
04:39.35 | justinnnnn | do u use them ?? |
04:40.11 | ronaldl79 | Anyone have any ideas as to why a virtual number won't pass DTMF to Asterisk? |
04:41.53 | justinnnnn | are there any other good international places ? |
04:42.54 | FuriousGeorge | someone should start a bsd orchestra |
04:44.07 | FuriousGeorge | release public domain performances for music on hold |
04:44.36 | *** join/#asterisk krisguy (n=krisguy@h216-170-039-057.adsl.navix.net) |
04:45.28 | FuriousGeorge | ronaldl79: ur codec? passing it inband when u shouldnt? "lose" dtmf? |
04:45.36 | FuriousGeorge | ur client? |
04:45.52 | FuriousGeorge | a combination? |
04:47.22 | justinnnnn | international providers ??? |
04:47.23 | justinnnnn | anyone ? |
04:47.36 | ronaldl79 | furious: 711 |
04:48.06 | FuriousGeorge | huh? |
04:48.24 | ronaldl79 | furious: I had my mom and one other person test a virtual DID in Detroit .. which rings here in Denver...neither could navigate the IVR. However, all is fine on the primary DID. |
04:48.42 | FuriousGeorge | ronaldl79: im kinda new myself |
04:48.44 | ronaldl79 | furious: u/a law |
04:48.45 | FuriousGeorge | ~IVR |
04:48.46 | jbot | hmm... ivr is Interactive Voice Response |
04:48.54 | FuriousGeorge | they have that for asterisk? |
04:49.08 | FuriousGeorge | i use festival :) |
04:49.12 | ronaldl79 | Furious: Yes...and it's simple to build. Asterisk has everything. |
04:49.43 | ronaldl79 | So, the DTMF isn't being passed from the virtual DID to navigate the IVR menus. As for festival, I haven't really tried to get that working yet. |
04:49.47 | FuriousGeorge | snazzy.. i use ulaw and iax mostly with sip clients almost always behind nat with my * server. i never have problems |
04:50.21 | FuriousGeorge | eyebeam is my client but ive used xlite too |
04:50.58 | ronaldl79 | I have a Cisco ATA 186 I'm using. |
04:51.00 | *** join/#asterisk Knight_DKN (n=knight_d@61.95.68.85) |
04:51.14 | ronaldl79 | Just got it unlocked from Vonage a few days ago ... I'll only use a soft phone on a PDA or laptop now. |
04:51.21 | ronaldl79 | The call quality is superb. |
04:51.39 | FuriousGeorge | ronaldl79: i can tell u my clients dtmf settings |
04:52.11 | ronaldl79 | I just built another * PBX tonight for a non-profit ... installing it tomorrow to demo and test....they'll be quite pleased. I also just discovered the console/intercom feature ... * has so much sh*t ... it's hard to keep up. |
04:52.35 | FuriousGeorge | console intercom? |
04:52.43 | ronaldl79 | Yes .... |
04:52.46 | mog_home | use dial |
04:52.55 | FuriousGeorge | no sound card |
04:52.55 | mog_home | app_intercom is dead |
04:53.00 | ronaldl79 | The 'console' can be used a pager/intercom system. |
04:53.17 | FuriousGeorge | i was just thinking about that functionality |
04:53.17 | ronaldl79 | right, Dial(console/dsp) |
04:53.18 | Cresl1n | dingdong |
04:53.24 | Cresl1n | app_intercom is dead |
04:53.35 | FuriousGeorge | and all your channels ring? |
04:53.38 | ronaldl79 | I wasn't speaking of app_intercom to begin with...dingdong. |
04:54.07 | FuriousGeorge | ronaldl79: have you messed with the relaxed dtmf setting? |
04:54.13 | ronaldl79 | I haven't, furious. |
04:54.28 | ronaldl79 | * is just a big toy of toys ... lol .... I love it. |
04:54.32 | FuriousGeorge | niether have i :) |
04:55.23 | FuriousGeorge | what dtmf related settings does your sipphone have? |
04:56.12 | ronaldl79 | I can't recall from memory .. but it's not me.... it's those calling from the PSTN ... to the virtual DID ... which isn't passing DTMF.. |
04:56.34 | flenders | hi, I'm just setting up asterisk now, but I'm having some weird problems while testing: I can make and receive calls just fine, but though, voice is not going both ways. When ring from the PSTN line to my DID, I can hear voice coming from PSTN->VoIP, but not from VoIP->PSTN. and When calling from VoIP to PSTN (or mobile), I can't hear anything on any direction. any hints? |
04:56.38 | ronaldl79 | If they call the primary DID directly, DTMF passes just fine....so I'm wondering what the virtual DID is cuasing.... |
04:57.00 | spackle | ~NAT |
04:57.02 | jbot | methinks nat is Network Address Translation Usable in Asterisk sip.conf file with externip, localnet, and localmask setup properly. See docs. |
04:57.05 | ronaldl79 | Flenders: Nat |
04:57.29 | heath__ | flenders: which protocol? |
04:57.41 | FuriousGeorge | ronaldl79: i would assume it has to do with the codec and/or how your client is passing the dtmf. you can hear eachother over the virtual did right |
04:57.58 | ronaldl79 | Yes, furious. |
04:58.06 | flenders | heath__ on my sip.conf: disallow=all |
04:58.07 | flenders | allow=alaw |
04:58.08 | flenders | allow=ulaw |
04:58.17 | ronaldl79 | Remember, it's the PSTN passing DTMF ... it's not me... |
04:58.43 | FuriousGeorge | ronaldl79: i know what ur saying, but you cant change settings on the PSTN so... |
04:58.44 | kram | n |
04:58.49 | *** join/#asterisk mydssmojo (n=bib@S0106001217475911.cg.shawcable.net) |
04:59.07 | ronaldl79 | Furious -- Everything works well on the primary DID ... DTMF is passed, etc...it's the virtual DID where it's not happening. |
04:59.27 | *** join/#asterisk jmg (n=cartel@shinobi.thoughtcrime.org.nz) |
04:59.29 | jmg | hey all |
04:59.34 | ronaldl79 | So, I'm wondering if something's screwed up on BroadVoice's end... |
04:59.36 | justinnnnn | anyone using voipjet ? |
04:59.42 | ronaldl79 | I do, justinnn. |
04:59.42 | jmg | is there a howto for asterisk on sarge? |
04:59.43 | Knight_DKN | I am |
04:59.46 | FuriousGeorge | ronaldl79: whats the primary did? |
04:59.50 | FuriousGeorge | pots? |
05:00.00 | justinnnnn | ronald is it actualy working for u ? |
05:00.11 | mydssmojo | anyone using AMP? |
05:00.11 | ronaldl79 | No, it's VoIP via Global Crossing, Furious. |
05:00.14 | ronaldl79 | Yes, justinn. |
05:00.17 | Knight_DKN | Yep I'm using voipjet all day |
05:00.19 | ronaldl79 | I have, mydssmojo. |
05:00.30 | jmg | im trying to install now |
05:00.33 | FuriousGeorge | ronaldl79: my bad, i was misunderstanding. both use the same codec then? |
05:00.36 | jmg | whats the canonical install guide? |
05:00.41 | ronaldl79 | Knight -- I can't imagine how much money you spend if you're using it all day. |
05:00.47 | ronaldl79 | Yes, furious. |
05:01.30 | flenders | ronaldl79 , do I have to open port 5060 on my firewall if I'm using a SIP account? |
05:01.42 | FuriousGeorge | flenders: yes and many more |
05:01.43 | mydssmojo | Is there a current installation instructions for AMP on fedora? Thanks |
05:02.00 | flenders | someone in here told me me the other day I didn't need it |
05:02.03 | ronaldl79 | mydssmojo ... I don't believe so, but you can search on Google for some reference points. |
05:02.10 | ronaldl79 | Someone lied to you, flenders. |
05:02.17 | flenders | FuriousGeorge , do you think that may be the reason for my problems? |
05:02.32 | blitzrage | kram: evening |
05:02.33 | justinnnnn | i like to ride around on the leckie bus |
05:02.34 | mydssmojo | thanks ronald179! |
05:02.41 | Cresl1n | kram: hey :-) |
05:02.43 | FuriousGeorge | flenders: your just trying to log asterisk into a voip provider? |
05:02.50 | flenders | yes |
05:02.50 | ronaldl79 | yw, mydss |
05:02.54 | *** join/#asterisk santiago (n=santiago@63.245.86.245) |
05:02.57 | FuriousGeorge | flenders: udp 5060 |
05:03.00 | Knight_DKN | Yeah the leckie bus is a pretty good ride |
05:03.08 | FuriousGeorge | flenders: voip is sip right? |
05:03.18 | flenders | FuriousGeorge , yes |
05:03.24 | FuriousGeorge | flenders: and you have a nat or firewall |
05:03.37 | flenders | FuriousGeorge : yes |
05:03.57 | FuriousGeorge | and your sip client is on the same network as * |
05:04.07 | flenders | FuriousGeorge , yes |
05:04.23 | FuriousGeorge | just udp 5060 --> asterisk, i think |
05:04.30 | FuriousGeorge | can someone verify |
05:04.33 | flenders | * is behind the firewall on an 192.168.x.x network |
05:04.47 | FuriousGeorge | whats *'s ip on your network |
05:05.01 | flenders | 192.168.10.20 |
05:05.08 | flenders | client is 192.168.10.185 |
05:05.25 | FuriousGeorge | try it |
05:06.32 | FuriousGeorge | ronaldl79: the only thing that comes to mind is that when the call goes PSTN ---> BroadVoice's Server, it gets compressed to much or something |
05:07.27 | flenders | DNAT udp -- 0.0.0.0/0 203.x.x.x udp dpt:5060 to:192.168.10.20:5060 |
05:07.30 | flenders | right? |
05:07.47 | flenders | just added that rule on the firewall |
05:07.57 | FuriousGeorge | ronaldl79: i know you can play a recording of DTMFs into a telephone on the PSTN and it will call so i dont think theres much too it on that end |
05:08.15 | FuriousGeorge | flenders: dont ask me, i used ipcop's gui to fwrd ports |
05:08.32 | FuriousGeorge | it /looks/ ok... whats with the 203.X.X.X |
05:08.48 | flenders | that's my firewall/external ip |
05:09.24 | FuriousGeorge | oh i get it, youre censoring it :) |
05:09.34 | FuriousGeorge | right? :| |
05:09.47 | flenders | yup |
05:09.58 | flenders | just to avoid shit loads of probes on my firewall |
05:11.13 | *** join/#asterisk cio (n=na@adsl-068-209-198-242.sip.bhm.bellsouth.net) |
05:11.32 | FuriousGeorge | (dnt foget to restart asterisk) so? |
05:11.42 | flenders | just trying now |
05:11.44 | cio | Anyone ever see chirping with 1.0.7 on a debian system? I'm runnning samba, dhcp3, bind9, mysql, apach2, dovecot on that box w/asterisk. |
05:12.07 | FuriousGeorge | chirp=jitter? ive had that |
05:12.20 | cio | FuriousGeorge: What did you do to solve it? |
05:12.21 | FuriousGeorge | not on debian tho :) |
05:12.35 | flenders | SIPclient->PSTN not working |
05:12.49 | cio | About every four seconds theres a 'chirp'. |
05:13.00 | cio | Sounds like somebody taking a corless phone off the hook. |
05:13.05 | cio | I'm using polycom sip phones. |
05:13.06 | FuriousGeorge | used settings like echocancel and jitterbuffer in client ans server. used ulaw or gsm |
05:13.10 | cio | It started about three days ago. |
05:13.18 | cio | I'm runing g729, think that would do it? |
05:14.05 | FuriousGeorge | cio: how much bandwidth does it use? |
05:14.30 | cio | The phones and * are on a gig lan, barely utilized. |
05:14.43 | FuriousGeorge | but what about your isp |
05:15.01 | FuriousGeorge | your talking internally it chirps? |
05:15.06 | cio | No ISP in the loop, the phone is connected to a switch, switch conneted to *. |
05:15.16 | FuriousGeorge | ahh |
05:15.25 | flenders | FuriousGeorge : not working yet mate. |
05:15.42 | flenders | ronaldl79 : what did you say about NAT before? |
05:16.15 | FuriousGeorge | cio: you running X on debian box? |
05:16.46 | cio | Nope. It just started a few days ago. I did install SQUID in that time frame, but I've stopped it and no changes. |
05:17.31 | cio | I did put a single ip tables rule in there to get port 80 traffic redirected to 3213 or something like that. |
05:19.02 | FuriousGeorge | cio: i dunno, you tried changing codecs, clients, seeing if anything makes it go away |
05:19.12 | FuriousGeorge | ? |
05:19.20 | cio | Yea, pretty much everything. |
05:19.23 | joelsolanki | JerJer: Hello |
05:19.43 | FuriousGeorge | flenders: did you try setting externip or localnet in sip.conf |
05:19.43 | cio | Rebooting my phone, haven't done that ... |
05:20.04 | cio | Yea, same here. Thanks, though! |
05:20.54 | flenders | FuriousGeorge : yup |
05:21.08 | argos73 | cio: could be one of those other jobs doing frequent (probably large) disk accesses.. some motherboards get a little goofy... |
05:21.10 | flenders | FuriousGeorge : and I have nat=yes on sip.conf |
05:21.26 | cio | I'm going to kill everything again and see if it stops... |
05:21.36 | argos73 | cio: try killing mysql first |
05:21.45 | argos73 | then apache |
05:21.51 | cio | I will. You have problems there? |
05:22.08 | argos73 | doubt it's dhcp or bind - they're pretty lightweight |
05:22.11 | flenders | FuriousGeorge: I think I got it |
05:22.22 | flenders | FuriousGeorge: netmask on localnet |
05:22.29 | argos73 | oh, i've battled my share of these problems... |
05:22.55 | cio | Nadda.... hrm... |
05:23.20 | FuriousGeorge | flenders: lemme know how that goes |
05:24.01 | argos73 | hmm - dovecot's a mail handler? those can be pretty rough on a system. |
05:24.20 | cio | What's the least problematic codec? ulaw? |
05:24.49 | argos73 | if bandwidth isn't an issue, probably. |
05:24.55 | argos73 | less cpu-intensive |
05:25.23 | cio | Wasn't that. |
05:25.38 | argos73 | syslogd? |
05:25.48 | *** join/#asterisk r0d3nt (i=nobody@wsip-24-234-241-145.lv.lv.cox.net) |
05:26.05 | cio | Well, at this point I need to make a IP to IP call to make sure it's not my dual X100P's. |
05:26.20 | flenders | FuriousGeorge: all good now mate! thanks for your help |
05:26.22 | argos73 | you've checked for interrupt sharing, right? |
05:26.26 | cio | I have a couple phones that will be in use tomorrow. |
05:26.35 | FuriousGeorge | flenders: it was all you ;) |
05:26.52 | cio | argos73: This just started.. if it was interrupt, I would think it would have been around since the server was built, right? |
05:27.20 | argos73 | most likely, but stranger things have happened... |
05:27.32 | cio | Yea, I agree. |
05:27.40 | FuriousGeorge | cio: i had this box that had a slight echo despite all the usual settings, and it turned out i'd built * w/o mmx support |
05:28.01 | cio | Hrm... Well, I'm just using the standard 1.0.7 debian packaged... |
05:28.28 | cio | It's been great until now. |
05:29.05 | argos73 | FuriousGeorge: second that - made a world of difference when I found that commented-out mmx line |
05:29.31 | cio | I do have an echo with my X100P's, but I just contributed that to crappy wiring as my phone-to-phone calls, even remote, were great with no echoing. |
05:29.50 | FuriousGeorge | cio: how come u dont upgrade to 1.0.9 |
05:30.24 | argos73 | hmm - my puppy is insane... :) |
05:30.39 | cio | The debian package is just so easy to install and it's stable. Plus I've heard (read?) issues about 1.0.9. |
05:30.40 | FuriousGeorge | cio: i find that jitter is most often caused by a) bandwidth or b) the sip/client's computer being too busy |
05:31.16 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
05:31.17 | FuriousGeorge | im amazed that opening IE will audibly affect call quality on a barton 2800 with a gig of ram running eyebeam on xp |
05:31.35 | cio | The nature of x86 computing ... |
05:31.45 | argos73 | heh - just starting xp has the same effect... :) |
05:31.45 | *** join/#asterisk djin_ib (n=djin_ib@gridfox.xs4all.nl) |
05:32.17 | cio | I just got a dual-core p4 box, it's lightning fast, but XP kills it. I'm tempted to install windows 3.1 just to feel like I have a fast computer. |
05:32.37 | argos73 | i have a copy of windows 2 if you really want to feel special |
05:33.30 | cio | heh.. |
05:33.38 | cio | later all - |
05:33.39 | cio | tire |
05:33.40 | cio | d |
05:33.45 | argos73 | good luck - night |
05:33.54 | cio | :) |
05:36.28 | FuriousGeorge | here's another one. when i use eyebeam's xfer feature to send a call from the pstn back out to the pstn the sound quality is terrible |
05:38.45 | websae | anyone here know why the heck in MySQL when I try to start it...it starts........then turns off.... |
05:38.46 | websae | Starting mysqld daemon with databases from /var/lib/mysql |
05:38.46 | websae | STOPPING server from pid file /var/lib/mysql/localhost.localdomain.pid |
05:38.50 | websae | any ideas...anyone? |
05:38.59 | websae | trying to get MySQL to use with asterisk |
05:39.06 | websae | would appreciaate any insight/help |
05:40.03 | FuriousGeorge | websae: is that the output from an initscript |
05:40.23 | websae | from trying to run the daemon |
05:40.39 | *** join/#asterisk lehel (n=asd@82.79.20.17) |
05:40.41 | lehel | hello |
05:41.29 | lehel | why is that?: manager.c:468 authenticate: xxx.xxx.xxx.xxx failed to pass IP ACL as 'admin' |
05:44.13 | Vco | well....tdmoe is cool.... |
05:44.21 | Vco | it's an absolute bandwidth whoar |
05:44.26 | Vco | but it's cool |
05:44.38 | Vco | err.. |
05:44.44 | Vco | whore |
05:44.46 | Vco | hmm. |
05:46.15 | blitzrage | whore! |
05:47.52 | *** join/#asterisk jayk- (i=jayk@lasziv.reprehensible.net) |
05:48.00 | jayk- | i keep getting this messages on my asterisk console |
05:48.00 | jayk- | <PROTECTED> |
05:48.09 | jayk- | they happen periodically..is this normal? |
05:48.26 | wunderkin | omg its going to blow |
05:48.36 | Vco | stop playing with it |
05:49.16 | jayk- | haven't touched it |
05:49.25 | websae | anyone here good with MySQL? |
05:49.42 | websae | i am curious..trying to get it to work with asterisk...let alone get it to start (as in a service) |
05:50.53 | wunderkin | jayk-: yes.. |
05:51.07 | jayk- | how come the channels restart? will it cause hangups? |
05:51.43 | mydssmojo | Can you guys recommend GUIs? |
05:52.54 | *** join/#asterisk normal1 (i=IipsjLhN@ip70-181-165-140.sd.sd.cox.net) |
05:53.52 | blitzrage | jayk-: from my experience, no, it doesn't drop channels |
05:54.03 | Juggie | jayk-, it happens peridiocally yes |
05:54.06 | Juggie | no it doesnt drop calls |
05:54.16 | blitzrage | jayk-: not knowing for sure (no T1's here) I think it just happens periodically... my guess is to resync the channels? |
05:55.11 | zapa | hi alll i alredy resolve my trouble with e1 telrad the troueble was outgoing tone provisoning |
05:55.35 | zapa | thanks for all |
05:55.56 | zapa | in telrad pri30 card |
05:57.20 | jayk- | okie doke |
05:58.39 | *** join/#asterisk CleanerX (n=nix@nat-ph3-wh.rz.uni-karlsruhe.de) |
06:01.19 | *** join/#asterisk alkalineX (n=alkaline@CN-ESR1-69-61-204-172.fuse.net) |
06:03.08 | blitzrage | anyone know what the format for tcpIpApp.sntp.gmtOffset in the Polycom config file is? |
06:04.28 | Pegger | websae can you get mysql to start |
06:04.30 | *** join/#asterisk wasim_ (n=wasim@pdpc/supporter/active/wasim) |
06:04.31 | blitzrage | hrmmmm... going to just try -5, but I keep seeing examples of -21500 or something :) |
06:06.12 | *** join/#asterisk joelsolanki (i=joelsola@202.160.161.93) |
06:06.49 | joelsolanki | Hello JerJer |
06:09.01 | djin_ib | I have a question regarding zaptel modules. I'm trying to setup an as minimal as possible CentOS config and having problems modprobing zaptel. |
06:09.23 | djin_ib | /lib/modules/2.6.9-11.EL/misc/zaptel.ko is there but "modprobe zaptel" gives "FATAL: Module zaptel not found." |
06:09.43 | djin_ib | Am i overlooking something (hopefully simpel :) |
06:10.00 | *** join/#asterisk tzafrir_laptop (n=tzafrir@local.xorcom.com) |
06:10.03 | djin_ib | I mean simple (simpel = Dutch :) |
06:10.31 | pashah | djun_ib: depmod -a ? |
06:11.20 | djin_ib | That doesn't say or do anything? |
06:11.38 | *** join/#asterisk Johnsie (n=john@acs-24-154-53-217.zoominternet.net) |
06:11.55 | Dr_Ray | fxs modules from digium can take the rxgain/txgain argument in zapata.conf right? |
06:12.12 | blitzrage | pretty sure |
06:12.32 | blitzrage | man... I really gotta stop building GUIs for Asterisk and start using Asterisk again |
06:12.38 | djin_ib | Mmm, perhaps I have to edit modules.conf manually? |
06:12.43 | pashah | dkin_ib: does not say, but generates modules.dep |
06:13.14 | djin_ib | Mmm, here is doesn't. |
06:13.51 | djin_ib | oh, wait. |
06:14.31 | djin_ib | I'm sorry, it does :) |
06:18.18 | djin_ib | No zaptel, though :( |
06:19.39 | mydssmojo | anybody here from city of Calgary? |
06:20.43 | flenders | does anyone know a good VoIP provider in the UK? |
06:21.14 | pashah | flenders: gradwell |
06:22.08 | flenders | pashah: cheers :o) |
06:28.27 | pashah | later all |
06:32.13 | *** join/#asterisk clive- (n=pirch@ndn-165-131-217.telkomadsl.co.za) |
06:32.53 | *** join/#asterisk KaBewM (n=kabewm@66-215-7-137.dhcp.psdn.ca.charter.com) |
06:33.36 | *** join/#asterisk gordonjcp (n=gordonjc@cpc3-broo2-5-0-cust232.renf.cable.ntl.com) |
06:34.44 | *** join/#asterisk zedkatuf (n=audela@82-33-115-145.cable.ubr08.azte.blueyonder.co.uk) |
06:38.25 | *** join/#asterisk shido6 (n=curtis@d221-68-210.commercial.cgocable.net) |
06:40.00 | joelsolanki | Hi shido6 |
06:43.57 | shido6 | hello |
06:44.31 | *** join/#asterisk mmmToop (n=chatzill@196.31.11.194) |
06:49.18 | wasim_ | mmmToop always reminds me of that hansen song ... |
06:49.48 | mmmToop | thanks guys ; ) |
06:51.19 | joelsolanki | Shido6: i m still unable to setup asterisk+ser on same machine |
06:51.36 | clive- | howzit toop |
06:51.51 | joelsolanki | Shido6: can u look in to my config ? |
06:51.52 | mmmToop | all good & u? |
06:52.44 | shido6 | whats up? |
06:53.29 | clive- | lekker thanks |
06:53.52 | clive- | waiting for telkom to install some lines...for a change |
06:54.02 | mmmToop | us to... |
06:54.34 | mmmToop | where do you purchase your Digium cards etc.? |
06:54.57 | clive- | i bought one from digium and one from jerjer |
06:55.17 | *** join/#asterisk ikey (i=ikey@202.54.37.184) |
06:55.33 | mmmToop | oh...we have been buying stuff from miro & get the feeling that they are overpriced |
06:55.45 | clive- | you can get them from stee in cape town |
06:55.51 | clive- | steve |
06:55.59 | clive- | connection-telecom |
06:56.41 | mmmToop | ah....on their site now |
06:57.13 | mmmToop | u got any PRI cards lying around that we can use/ buy? |
06:57.22 | clive- | i do actually |
06:57.26 | clive- | e100p |
06:58.14 | mmmToop | has that guy got echo cancelation/ |
06:58.15 | mmmToop | ? |
06:58.40 | *** join/#asterisk lenne_dk (n=mirc@83.72.129.7.ip.tele2adsl.dk) |
06:58.51 | *** join/#asterisk _omer (i=p@203.215.180.250) |
06:59.07 | clive- | i decided to do bri's instead |
06:59.10 | _omer | how to allow the specific IP Addresses in the asterisk? |
06:59.28 | clive- | omer host= |
06:59.33 | *** join/#asterisk djin_ib (n=djin_ib@gridfox.xs4all.nl) |
06:59.52 | _omer | in the sip.conf [general] |
06:59.55 | _omer | ?? |
07:00.18 | Heng | omer: In each client. |
07:00.24 | Heng | context |
07:00.39 | _omer | but my client doesnt register to my asterisk...he sends call directly.... |
07:00.59 | Heng | omer: oh. I use firewall for that |
07:02.08 | _omer | I dont have firewall :( |
07:02.24 | Heng | pls look at Shorewall |
07:03.46 | _omer | okey |
07:03.49 | Dr_Ray | or even smoothwall |
07:04.22 | lenne_dk | I want to announce incoming calls over the speaker. I can make calls to the console/dsp, and it autoanswers. But when a call comes in, I don't want to connect it to the speaker, I just want to send a soundfile to the speaker, and the real phone extention should keep ringing |
07:05.05 | _omer | Dr_Ray: cant I do it in asterisk? allow IPs ..etc |
07:05.28 | shido6 | ok |
07:05.28 | shido6 | back |
07:05.30 | shido6 | what sup? |
07:05.34 | *** join/#asterisk heka (n=heka@80.80.174.140) |
07:06.04 | Dr_Ray | you could also use firestarter |
07:06.25 | Dr_Ray | firestarter would run on an asterisk box |
07:06.28 | shido6 | hrmm |
07:06.35 | shido6 | so use festival :) |
07:06.37 | shido6 | lenne_dk: |
07:07.02 | wasim_ | lenne_dk: chan_local |
07:07.37 | _omer | Dr_Ray: ok..thnx |
07:07.42 | lenne_dk | A litle more clue, wasim |
07:13.35 | lenne_dk | wasim, what do you mean? if I do a exten => 111,1,Dial(${CONSOLE}), the incoming call is connected to the console. |
07:13.42 | flenders | guys, if I have a bunch of DIDs, each of them beeing used to direct dial an extension, is there a way to configure it so when the extension dials out, the corresponding DID is shown on the caller ID on the callee's device? |
07:17.33 | flenders | also, just to let you know, I'm not forwarding port UDP/5060 to my * server (I'm using a SIP account) |
07:17.37 | *** join/#asterisk akrall (i=user@201.144.58.10) |
07:18.37 | akrall | Guys, whats the function of ztdummy and timing source for meetme, moh, etc? if you use x100p clones, do you still need ztdummy or by loading the wcfco driver are you covered? |
07:19.53 | *** join/#asterisk newl (n=newlook@203-59-91-217.dyn.iinet.net.au) |
07:20.07 | ronaldl79 | I figured out why my virtual DID wasn't passing DTMF to * -- The dtmf was set to 'inband,' instead of 'rfc2833'. Works like a charm now! |
07:20.12 | djin_ib | akrall, you covered by teh wcfo driver. |
07:24.34 | *** join/#asterisk shanky (n=shanky@85.137.127.142) |
07:24.37 | *** join/#asterisk \PsyKo\ (n=xxxxxxxx@golia.caltanet.it) |
07:25.07 | shanky | hi, good morning |
07:25.48 | lenne_dk | so how to use the ztdummy.ko under FreeBSD? |
07:26.56 | lenne_dk | -bash-2.05b# kldload ztdummy.ko |
07:26.56 | lenne_dk | kldload: can't load ztdummy.ko: No such file or directory |
07:26.56 | lenne_dk | - |
07:27.06 | zzzirk | anyone have any tips on getting a reasonable understanding of the various codecs available? |
07:27.17 | shanky | do you if is there any specific channel for asteriskt@home ? |
07:28.14 | lenne_dk | do what, zzzirk |
07:28.17 | lenne_dk | ? |
07:28.37 | lenne_dk | Sorry, not zzzirk, shanky |
07:28.38 | zzzirk | I just want to understand the differences, strenghts, weaknesses of the various codecs |
07:28.41 | zzzirk | ah, okay |
07:28.42 | zzzirk | heh |
07:28.42 | shanky | http://www.voip-info.org/tiki-index.php?page=Asterisk%20codecs |
07:29.25 | shanky | that's for zzzirk |
07:31.12 | wasim_ | lenne_dk: you can fork in * dialplan using chan_local ... Dial(Zap/1&LOCAL/s@AlsoDoThis&LOCAL/s@AndThisToo) |
07:31.53 | *** join/#asterisk konfuzed (n=konfuzed@H129.C72.B0.tor.eicat.ca) |
07:33.19 | shanky | well, I have install an asteriskt@home 1.5, I have add 2 extensions and I can call between them |
07:33.50 | wasim_ | shanky: welcome to hell |
07:34.02 | shanky | but now I'm trying to setup a trunk to my voip provider |
07:34.11 | ronaldl79 | Does anyone know the link to an asterisk alarm app??? |
07:34.15 | *** join/#asterisk Gunnar (n=gunnar@34.80-203-209.nextgentel.com) |
07:34.23 | zzzirk | thanks for the pointer shanky |
07:34.45 | shanky | I have already added the trunk, and reconfigure an outbound routing |
07:35.14 | lenne_dk | ronald: I have my app drop a .call file, it starts an agi, telling me the temp of the fridge |
07:35.44 | ronaldl79 | Pretty wild, lenne |
07:35.53 | ronaldl79 | * might even be able to turn on the oven for ya |
07:35.55 | ronaldl79 | lol |
07:35.59 | ronaldl79 | It's so damn extensible, it's wild. |
07:36.10 | ronaldl79 | I wonder if anyone's tied in any X10 stuff... |
07:37.56 | lenne_dk | I'm using the 1-wire bus for temperature sensors, rrdtools for storing the data, and big brother for monitoring and generating alarms. |
07:38.24 | lenne_dk | So if the fridge is left open for too long, I get a call. |
07:38.29 | shanky | lenne_dk: nice job |
07:40.53 | shanky | well, after I setup the trunk and the outbound routing, I try to do a external call an I get this: |
07:41.11 | shanky | 09:38 < lenne_dk> I'm using the 1-wire bus for temperature sensors, rrdtools for storing the data, and big |
07:41.13 | shanky | <PROTECTED> |
07:41.15 | shanky | ups |
07:41.18 | shanky | 09:38 < lenne_dk> So if the fridge is left open for too long, I get a call. |
07:41.28 | shanky | sorry |
07:41.32 | Dr_Ray | dallas semiconductor? |
07:41.42 | shanky | Capabilities: us - 0x8 (alaw), peer - audio=0x51d (g723|ulaw|alaw|g726|g729|ilbc)/video=0x0 (nothing), combined - 0x8 (alaw) |
07:41.46 | shanky | Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) |
07:41.49 | shanky | Looking for 952486557 in from-internal |
07:41.51 | shanky | Reliably Transmitting (NAT): |
07:41.54 | shanky | SIP/2.0 484 Address Incomplete |
07:42.06 | *** join/#asterisk KaBewM (n=kabewm@66-215-7-106.dhcp.psdn.ca.charter.com) |
07:42.08 | lenne_dk | Dr_Ray: Yes |
07:42.15 | Dr_Ray | kewl |
07:44.07 | lenne_dk | +4536949142 extension 79 or 36949142@musimi.dk extension 79 |
07:44.43 | shanky | any idea what I'm doing wrong? |
07:45.37 | shanky | what I don't understand is why asterisk is looking for 952486557 in the context "from-internal" |
07:46.57 | shido6 | yeah |
07:47.11 | shido6 | from-internal doesnt have that many digits or the amount of digits is off |
07:47.14 | shido6 | for the context, shanky |
07:48.55 | shanky | yeah, I have added "952486557" in the Dial Patterns and now it works |
07:48.56 | _omer | how to do "SIP Proxy Authentication" in * ? |
07:51.20 | lenne_dk | Is it possible to ignore that a module can not be loaded? My soundcard sometimes(?) need a powerdown to be detected after boot, a software reset is not enough. |
07:51.41 | lenne_dk | So chan_oss sometimes can't be loaded and asterisk will not start. |
07:58.06 | lenne_dk | one solution is to put * in a wrapper script. First start *, then have 'if [ -r /dev/dsp ] then asterisk -r -x "load chan_oss.so" fi' |
08:02.20 | akrall | djin_ib: thx! |
08:02.25 | *** join/#asterisk tobiasWolf (n=konversa@195.162.255.10) |
08:02.44 | akrall | anybody using a sipura spa841? |
08:06.24 | *** join/#asterisk Ilya (n=Ilya@mail.tex.kiev.ua) |
08:06.43 | Ilya | hi |
08:07.39 | *** part/#asterisk KaBewM (n=kabewm@66-215-7-106.dhcp.psdn.ca.charter.com) |
08:08.04 | *** join/#asterisk jeh_work (n=jeh@ext122.almare.com) |
08:08.12 | jeh_work | good morning |
08:08.26 | *** join/#asterisk darkskiez (n=darkskie@194.247.78.146) |
08:08.28 | Ilya | SipClient: Receiving message... |
08:08.32 | Ilya | SipClient: Received: 11:09:04.018 |
08:08.36 | Ilya | --------------------------------- |
08:08.41 | Ilya | SIP/2.0 401 Unauthorized |
08:08.44 | Ilya | Via: SIP/2.0/UDP 10.0.55.1:5062;branch=z9hG4bK214FDD9B |
08:08.48 | Ilya | From: "ilya" <sip:200@ilya> |
08:08.52 | Ilya | To: "ilya" <sip:200@ilya>;tag=as63decd7a |
08:08.57 | Ilya | Call-ID: 1700781529@10.0.55.1 |
08:09.00 | Ilya | CSeq: 6163 REGISTER |
08:09.05 | Ilya | User-Agent: Asterisk PBX |
08:09.08 | Ilya | Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER |
08:09.09 | Ilya | Contact: <sip:200@10.0.55.1> |
08:09.09 | Ilya | WWW-Authenticate: Digest realm="ilya", nonce="388437cf" |
08:09.09 | Ilya | Content-Length: 0 |
08:09.10 | Ilya | i'm trying to connect to newly installed asterisk with Kphone.. i get this |
08:09.27 | *** join/#asterisk rat1101 (n=vinay@ip68-100-31-133.dc.dc.cox.net) |
08:10.57 | jeh_work | can anyone point me to some docs as to what the difference would be when i try to redirect a call using AMI (Redirect action) and doing it on a real plastic phone using #xxxx? |
08:11.13 | jeh_work | the former never works while the latter works |
08:11.14 | Ilya | [general] |
08:11.18 | Ilya | context=default |
08:11.22 | Ilya | realm=ilya |
08:11.26 | Ilya | port=5060 |
08:11.30 | Ilya | bindaddr=10.0.55.1 |
08:11.34 | Ilya | [200] |
08:11.37 | *** join/#asterisk gordonjcp (n=gordonjc@lodge.glasgownet.com) |
08:11.38 | Ilya | type=friend |
08:11.42 | Ilya | username=200 |
08:11.46 | Ilya | secret=secret |
08:11.50 | Ilya | nat=no |
08:11.50 | Ilya | context=office |
08:11.50 | Ilya | callerid="ilya" <222> |
08:11.51 | Ilya | host=dynamic |
08:11.51 | Ilya | in sip.conf i have: |
08:13.36 | jeh_work | doing the redirect using the real phone gives a voice message asking for the new extension to transfer to. so apparently the "#" puts asterisk in some kind of state of mind that the Redirect action in code can't do |
08:13.58 | *** join/#asterisk jeffgus (n=jeffgus@2002:d856:c704:0:0:0:0:1) |
08:15.18 | *** join/#asterisk KeX_WorX (n=chris@83-65-129-46.paris-lodron.xdsl-line.inode.at) |
08:15.20 | KeX_WorX | hi |
08:15.33 | KeX_WorX | any1 users srtp in asterisk ? |
08:15.41 | KeX_WorX | or knows anything bout it ? |
08:18.27 | *** join/#asterisk musical_Duck (n=kvirc@wblv-146-224-229.telkomadsl.co.za) |
08:20.10 | musical_Duck | Can any1 help me with iax2? I can't dail my clients register initial but then I get lost of failed md5 auth messages |
08:22.04 | musical_Duck | My clients register initially but then I get lots of failed md5 auth messages, soz late night |
08:23.46 | *** part/#asterisk musical_Duck (n=kvirc@wblv-146-224-229.telkomadsl.co.za) |
08:25.37 | *** join/#asterisk musical_Duck (n=kvirc@wblv-146-224-229.telkomadsl.co.za) |
08:26.09 | musical_Duck | Any1 here? |
08:27.37 | *** join/#asterisk nesys (n=nesys@2001:1418:1a6:0:20d:93ff:fe28:3ef8) |
08:31.11 | *** join/#asterisk oej (n=Olle@apollo.webway.se) |
08:34.29 | *** join/#asterisk Thoran (n=Thoran@p54A5A6A9.dip0.t-ipconnect.de) |
08:35.21 | Thoran | Morning every1! |
08:36.18 | avizion | morning Thoran |
08:36.21 | avizion | ;) |
08:36.56 | musical_Duck | lo |
08:37.30 | musical_Duck | Say any of you guys use iax clients? |
08:37.58 | pauldy | damb getting ready for bed here sun already up on that side of the ppond huh |
08:38.16 | avizion | not much... but once in a while I use FireFly |
08:38.51 | musical_Duck | My clients register initialy but I get lots of failed md5 auth messages |
08:39.25 | musical_Duck | And I can't seem to dial them with Dial IAX2/clientbob wich used to word |
08:39.31 | musical_Duck | which rather |
08:39.40 | musical_Duck | and work |
08:39.41 | musical_Duck | :) |
08:43.53 | *** join/#asterisk ErMeS`wrk (n=ermsewrk@217.220.121.62) |
08:46.27 | *** join/#asterisk Gunnar (n=gunnar@bkkb-gw.bitcon.no) |
08:48.52 | *** part/#asterisk akrall (i=user@201.144.58.10) |
08:51.34 | *** join/#asterisk gordonjcp (n=gordonjc@cpc3-broo2-5-0-cust232.renf.cable.ntl.com) |
09:09.51 | *** part/#asterisk nesys (n=nesys@2001:1418:1a6:0:20d:93ff:fe28:3ef8) |
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09:22.47 | *** part/#asterisk Romik (n=romik_@1.fix.netvision.net.il) |
09:25.16 | jeh_work | i see asterisk logging this: "dialparties.agi: Dial return value was -1 and dialstring was SIP/1002|120|tr" |
09:25.24 | *** join/#asterisk ful|work (n=fulgas@213.58.130.46) |
09:25.32 | jeh_work | how can i figure out why the dialing failed? |
09:26.04 | JamesDotCom | by debugging the agi script |
09:26.07 | jeh_work | this is then done using AMI redirect. when done using a plastic ugly phone the return value is 0 and the redirect proceeds |
09:26.34 | jeh_work | ugh, it seems to be some kind of line noise |
09:28.16 | *** join/#asterisk RoyK (n=roy@80.239.107.80) |
09:28.40 | *** join/#asterisk nesys (n=nesys@2001:1418:1a6:0:20d:93ff:fe28:3ef8) |
09:29.26 | jeh_work | ah, i had agi debuggin disabled. i hope it prints me some more clues when it's enabled |
09:33.40 | X-Rob | jeh_work - you probably don't have Asterisk::AGI installed. |
09:34.00 | jeh_work | X-Rob: i should as it works for hardware phones |
09:34.01 | X-Rob | at your shell prompt, type /var/lib/asterisk/agi-bin/dialparties.agi |
09:34.05 | X-Rob | it'll tell you what's not working |
09:34.33 | jeh_work | it just freezes |
09:34.39 | X-Rob | push enter a few times. |
09:34.40 | jeh_work | waiting for args maybe? |
09:34.48 | X-Rob | but, ok, that means it's not a dialparties problem |
09:34.58 | X-Rob | ..also, I just read what you said about it working sometimes. |
09:35.22 | jeh_work | it always works when used with # from a real phone |
09:35.43 | jeh_work | but when try to do the same using AMI it never works (a Redirect action) |
09:36.13 | jeh_work | so i guess pressing # on the phone does something that Redirect can't mimic |
09:37.01 | jeh_work | just to answer a call Redirect works ok, but if i want to further Redirect that answered call it fails. but works when done with # |
09:37.27 | X-Rob | ok |
09:37.29 | X-Rob | the word is 'transfer' |
09:37.40 | X-Rob | or is it not? |
09:40.38 | jeh_work | yes, but in AMI it's down with Redirect? |
09:41.03 | X-Rob | jbot: ami? |
09:41.08 | X-Rob | ~ami |
09:41.30 | jeh_work | asterisk management interface |
09:41.49 | X-Rob | ok. So what does that have to do with transfers? |
09:42.21 | RoyK | jeh_work: management interface? as in the so-called Manager API? |
09:42.28 | jeh_work | yes |
09:43.01 | jeh_work | i use that interface to connect to asterisk and then in with my application control it |
09:43.02 | RoyK | has someone finally renamed it? |
09:43.20 | RoyK | jeh_work: what app was that? |
09:43.36 | jeh_work | RoyK: an inhouse app |
09:43.56 | jeh_work | RoyK: built in java using the asterisk-java bindings |
09:44.03 | RoyK | ok |
09:44.26 | jeh_work | i get access to all events i need, all channels, queues etc |
09:44.58 | jeh_work | i can answer calls, origiate calls, but a transfer/redirect of an answered call doesn't work |
09:45.09 | Ilya | do i need a sound card on a server to run asterisk? |
09:45.27 | Dr_Ray | no, unless you want sound on the console (probably not) |
09:46.28 | Ilya | is p-pro200(no MXX)/192mb RAM enough to run asterisk? |
09:46.57 | jeh_work | lunch time |
09:47.19 | Ilya | ) |
09:49.38 | X-Rob | Ilya - it -might- be. But I wouldn't put money on it. Jump onto ebay with $50 and buy a Piii500 or something. |
09:49.47 | *** join/#asterisk jaxkz (n=cyrieldo@tbnb-165-193-112.telkomadsl.co.za) |
09:49.54 | jaxkz | 'n morning |
09:50.50 | Ilya | X-Rob: i just have this machine running freebsd |
09:51.12 | jaxkz | Does anyone use the Eurorings Network for voip Termination? |
09:51.40 | X-Rob | Ilya - so? Get a Piii 500, run Centos 4.1 on it. |
09:51.55 | X-Rob | you won't have _any_ problems. |
09:53.53 | *** join/#asterisk Jzalae (n=sk@216-220-248-44.midmaine.com) |
09:55.09 | ennuyeux72 | i have a question about RTP timing in asterisk |
09:55.34 | ennuyeux72 | am i correct in saying that for every RTP packet that asterisk receives it will send one back in response |
09:56.13 | ennuyeux72 | i am getting a scenario where tethereal trace shows multiple RTP packets being received by asterisk from a phone |
09:56.29 | ennuyeux72 | without anything being sent back |
09:56.36 | ennuyeux72 | till a later time |
09:56.46 | ennuyeux72 | where multiple RTP packets are sent back |
09:57.07 | ennuyeux72 | i was under the impression that there was some sort of one to one deal going on with the RTP packets |
10:13.55 | *** join/#asterisk cjk (n=cjk@80.92.64.103) |
10:15.37 | X-Rob | ennuyeux72 - that is incorrect. |
10:15.39 | *** join/#asterisk bsd3 (n=bsd@203.134.194.176) |
10:15.55 | X-Rob | voicemail is a prime example of this. When asterisk is recording a voicemail message, it is _only_ receiving, not sending RTP at all. |
10:21.17 | *** join/#asterisk FABRIZIOxxx (n=FABRIZIO@81-208-26-86.ip.fastwebnet.it) |
10:21.31 | bsd3 | hi, friends! |
10:23.42 | bsd3 | is the refresh in iax2 and sip is like a keep-alive? |
10:24.06 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
10:24.36 | FABRIZIOxxx | hello all .. i set up an asterisk box with 2 tdm04b (8 fsx lines) .. the problem is that sometimes from the pstn network people occasionally hear a choppy sound.. how can i solve this? |
10:25.18 | lehel | X-Rob, i gotta a prob with recorded vm messages, it is recorded well, but when i want to play/download from the web-interface cannot becouse the owner of the *.wav is asterisk, not www-data |
10:27.18 | bsd3 | lehel: i have not tested it, but you need to use an agi script to do that and NOT a cgi |
10:29.02 | bsd3 | how do i change refresh times for an IAX and, or a SIP channel? |
10:29.03 | jaxkz | Do the wildcards support ATM? |
10:31.03 | *** join/#asterisk N4SH (n=Bernardo@c-67-180-105-69.hsd1.ca.comcast.net) |
10:31.14 | N4SH | i got a question |
10:31.33 | X-Rob | How can I help you N4SH 8) |
10:31.54 | lehel | bsd3: do you know an agi script for it? |
10:31.58 | N4SH | x-rob tnx. is asterisk capable of doing IP telephony/VOIP? |
10:32.04 | X-Rob | N4SH - heh. |
10:32.07 | X-Rob | Yes. Yes it is |
10:32.07 | lehel | ;p |
10:32.27 | bsd3 | lehel: sorry, no dear |
10:32.38 | N4SH | x-rob oh wow. i don tknow how to use asteisk is there a websiet for dummmies? |
10:32.58 | *** join/#asterisk frenzy (n=frenzy@193.220.82.108) |
10:32.59 | X-Rob | N4SH - try Asterisk@Home |
10:33.01 | X-Rob | google for it. |
10:33.03 | lehel | bsd3: deaR?!:P |
10:33.10 | N4SH | ok thanks.. |
10:33.56 | N4SH | where is it best to rn asterisk? |
10:34.02 | N4SH | run |
10:34.08 | X-Rob | ...on a linux machine? |
10:34.13 | N4SH | mepis? |
10:34.33 | X-Rob | CentOS 4.1/WhiteBox/RHEL is the 'best' distro, because that's what all the devels use. |
10:34.43 | X-Rob | but, really, there's not that much difference. Use a 2.6 kernel. |
10:35.16 | N4SH | oh cool... tnx. let me read the @home stuff |
10:35.28 | X-Rob | N4SH - |
10:35.29 | X-Rob | ~docs |
10:35.31 | jbot | methinks docs is Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk |
10:35.38 | X-Rob | ^^^ ses them. |
10:35.40 | X-Rob | ^^^ see them |
10:35.41 | X-Rob | even |
10:35.49 | N4SH | ok tnx |
10:36.09 | X-Rob | jbot: wiizard is an anal wart |
10:36.11 | jbot | X-Rob: okay |
10:36.18 | X-Rob | *snigger* |
10:37.15 | frenzy | hello |
10:37.24 | frenzy | How do I enable transcoding ? |
10:38.03 | jaxkz | that will be done automaticly i think |
10:38.12 | bsd3 | N4SH: if you look at the asterisk source, building tools like Makefile's etc does not show any such dependency |
10:39.17 | frenzy | I'm getting a very jittery connection |
10:39.44 | frenzy | When connecting dirclty to the provider using ilbc I get a clear line |
10:39.55 | bsd3 | N4SH: i'm running 1.2.0-beta1 on a Knoppix machine kernel version 2.6.12 |
10:40.20 | *** join/#asterisk A-Tuin|work (n=A-Tuin@nat.office.legend.net.uk) |
10:40.25 | frenzy | when setting * to use ulaw and my ata on ilbc the audio is jittery |
10:40.26 | N4SH | tnx for the advices... =) |
10:40.47 | *** join/#asterisk ^X-works (n=r0x0r@host34-3.pool871.interbusiness.it) |
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10:59.59 | *** join/#asterisk gein (n=gein@213.134.110.241) |
11:00.00 | gein | Im having problems with my queues, calls keeps getting placed in the queue even if there aren't any agents logged in, and I |
11:00.04 | gein | <PROTECTED> |
11:00.32 | Cresl1n | too early |
11:00.37 | gein | running v1.0.9 stable |
11:01.46 | gein | if the queue doesn't have any members, then I can't join.. seems like strict option doesn't really work in my case? |
11:04.40 | *** join/#asterisk iCEBrkr (i=icebrkr@rrcs-24-129-130-158.se.biz.rr.com) |
11:07.10 | *** part/#asterisk bsd3 (n=bsd@203.134.194.176) |
11:07.11 | *** join/#asterisk [jedi] (n=hhgds4@213.162.200.226) |
11:14.22 | [jedi] | I get 'chan_iax2.c:3052 iax2_trunk_queue: Maximum trunk data space exceeded to MY_IP:4659' when calling throught IAX between two asterisk servers |
11:14.27 | [jedi] | SIP works great |
11:15.56 | *** part/#asterisk nesys (n=nesys@2001:1418:1a6:0:20d:93ff:fe28:3ef8) |
11:16.03 | gein | no one who uses joinempty=strict in queues.conf and that it is actually working? |
11:18.42 | RoyK | what does that do? |
11:19.27 | [jedi] | what does that mean? |
11:19.50 | [jedi] | iax2 is saying continuously these messages when calling between IAX servers |
11:20.10 | [jedi] | and also voice only goes from calling to caller |
11:20.14 | [jedi] | but not from caller to calling |
11:20.27 | clive- | jedi I got that too, sop I swicthed off trunking |
11:20.48 | *** join/#asterisk jeh_work (n=jeh@ext122.almare.com) |
11:20.53 | [jedi] | trunking? |
11:21.01 | [jedi] | should be activated, shouldn't |
11:21.01 | [jedi] | ? |
11:21.16 | gein | RoyK: do you mean joinempty=strict? |
11:21.16 | clive- | i think its a bug |
11:21.29 | jeh_work | stupid asterisk can't log why it thinks my Dial() command must fail |
11:21.40 | RoyK | gein: yes |
11:21.56 | gein | it make calls not queued if there aren't any agents logged in which is a member to that queue |
11:22.09 | jeh_work | it's so nice to just be given a "-1" and nothing in the log and no error message anywhere |
11:22.31 | jeh_work | not even with verbosity and debugging at 100 and agi debug on |
11:23.21 | shanky | hey, I'm getting this error: -- Got SIP response 488 "Not Acceptable Here" back from (my voip provider) |
11:23.49 | shanky | it's suppose to be a codec problem, but I have try with allow=all and it doesn't work |
11:24.08 | gein | I just want the feature that if no member of a queue has logged in, then calls should not be placed in that queue, also, already queued calls should exit the queue |
11:25.52 | *** join/#asterisk mosty (i=mostynm@adsl-137-244.swiftdsl.com.au) |
11:26.03 | jeh_work | debugging perl in itself is a horrible fate, debugging perl without any meaningful logging is worse |
11:28.08 | jeh_work | what is the purpose of AGI and the external perl stuff anyway? |
11:28.17 | mosty | is it possible to allow particular codecs only when both src and dest of a call are SIP channels? |
11:28.43 | jeh_work | dialparties.agi even has to do manager connections back to asterisk to get some info. that's not too smart |
11:30.16 | [jedi] | jeh_work: perl isn't the only AGI choice you have |
11:30.18 | [jedi] | jeh_work |
11:30.43 | jeh_work | [jedi]: sure, but i'm not writing new stuff, i'm debugging why my transfers die |
11:31.23 | jeh_work | [jedi]: and that means diving into the code that does the dialing |
11:33.18 | *** join/#asterisk knobo (n=knobo@217.77.34.124) |
11:36.55 | frenzy | Sep 28 07:25:58 NOTICE[20598]: chan_sip.c:6704 handle_response: Peer '7777777' is now TOO LAGGED! |
11:36.56 | frenzy | Sep 28 07:26:09 NOTICE[20598]: chan_sip.c:6698 handle_response: Peer '7777777' is now REACHABLE! |
11:36.56 | frenzy | Sep 28 07:27:13 NOTICE[20598]: chan_sip.c:8085 sip_poke_noanswer: Peer '7777777' is now UNREACHABLE! |
11:36.56 | frenzy | Sep 28 07:27:52 NOTICE[20598]: chan_sip.c:6698 handle_response: Peer '7777777' is now REACHABLE! |
11:37.11 | frenzy | how do I go about tackling this ? |
11:37.48 | mosty | frenzy: reduce the load on the network between the client and the server? |
11:38.22 | frenzy | the server is remotly located (US) |
11:38.31 | *** join/#asterisk phpboy (n=shane@c1-114-12.tbnb.isadsl.co.za) |
11:38.44 | frenzy | while the ATA is behind a satellite connection |
11:38.51 | frenzy | in a different country |
11:38.56 | phpboy | to install a working CentOS for asetrisk... do I have to download all 4 CD's? |
11:39.06 | mosty | frenzy, 2-way satellite? |
11:39.11 | frenzy | yap |
11:39.54 | mosty | frenzy: perhaps satellite just has too high a latency. are you able to make calls, and if so how much delay is there? |
11:40.13 | frenzy | yes |
11:40.21 | frenzy | about 800 - 1000ms |
11:40.33 | mosty | oh yuck :) |
11:41.02 | frenzy | considering the ATA is in Africa |
11:41.09 | frenzy | I wouldnt say the quality is bad |
11:41.51 | phpboy | frenzy: where in africa? |
11:42.20 | frenzy | south |
11:42.32 | phpboy | cool, I'm from South Africa! |
11:42.33 | phpboy | askhd |
11:42.35 | Druken | frenzy: just take the qualify off |
11:42.51 | frenzy | my bad the 800 - 1000ms is when doing an echo |
11:42.59 | shanky | phpboy: you can try wit asterisk@home |
11:43.07 | frenzy | so most probably termination routes would be shorter |
11:43.42 | frenzy | removing qualify wont cause the ATA to get dissconnected ? |
11:43.56 | Druken | uhmm.... no... |
11:44.07 | frenzy | I do face that at times... it disconnects and then after couple seconds reconnects |
11:44.48 | phpboy | shanky: I need to get a Junghanns Quad ISDN card going on it... will it be possible? |
11:45.04 | frenzy | just for your info my qualify=2000 :) |
11:45.50 | Druken | qualify=0 |
11:46.09 | shanky | phpboy: take a look at asteriskathome.sourceforge.net |
11:48.11 | frenzy | also get Sep 28 07:48:06 WARNING[20598]: chan_sip.c:701 retrans_pkt: Maximum retries exceeded on call 66aa1986c5024e4e@192.168.1.101 for seqno 29700 (Non-critical Response) |
11:48.13 | frenzy | sometimes... |
11:48.14 | phpboy | ok, I'm prolly gonna need to build it in myself |
11:48.29 | phpboy | does anybody have the Junghanns Quad ISDN card? |
11:49.56 | Druken | ya know it's funny, all those commercials talk about the poorness in africa, yet we get people in here doing voip, and looking for massive routes and shit... |
11:49.58 | Druken | which is it? |
11:50.43 | *** join/#asterisk iDunno (n=brettp@mike.catnip.org.uk) |
11:51.02 | wasim | Druken: both, its the .0001% that make it here |
11:51.23 | Druken | oh.. |
11:51.51 | iDunno | can someone point me at some documentation that tells me about attended transfers? |
11:51.59 | wasim | iDunno: features.conf |
11:52.08 | Druken | ~voipinfo |
11:52.19 | Druken | ~voip-info |
11:52.21 | jbot | [voip-info] 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 |
11:52.22 | iDunno | been there, I appear to be missing something obvious. |
11:53.31 | iDunno | assume that I've got kphone as the client, and that the call has come in via sip, I want to do an attended transfer to another sip phone, and using DTMF appears to be failing me, and using the transfer button also appears to be failing me. |
11:56.10 | phpboy | guys, most of africa is a fuckup |
11:56.24 | phpboy | but South Africa is pretty well off |
11:58.39 | gein | can someone tell me how exactly joinempty in queues.conf works or where I can read about it? |
11:59.49 | Druken | i'm not sure what it means, i've always assumed it means people can join the queue even without any agents to answer their call |
12:01.18 | gein | and what is meant by "without any agents to answer their call"? I can't figure out if they mean that there are no agents assigned as members in queues.conf or if they mean agents who's not logged in |
12:02.41 | gein | when joinempty=strict that is |
12:02.55 | *** join/#asterisk Samoied (n=Samoied@popeye.opens.com.br) |
12:03.01 | dudes_ | gein - I'm not sure what he means, nor do I know what it means. But I've always found the asterisk source code to be helpful with certain questions |
12:03.58 | gein | I'm not that good at intepreting source code |
12:04.20 | gein | I'm experiencing the same problem as the author of this feature-request: http://bugs.digium.com/bug_view_advanced_page.php?bug_id=3754 |
12:04.28 | mosty | can you allow/disallow codecs from the dialplan? |
12:04.57 | dudes_ | mosty - I don't think so. |
12:05.15 | gein | And they closed that ticket, saying: 'not an issue'...huh? |
12:05.30 | Thoran | Any1 willing to help a newbie debug his asterisk conf. I can't make calls from my sip phones to the PSTN |
12:05.39 | mosty | dudes: is there a way i can allow certain codecs for sip->sip calls, but not for sip->zap calls? |
12:05.58 | mosty | thoran: what connection do you have to the pstn? |
12:05.59 | dudes_ | sip.conf and zapata.conf ? |
12:06.23 | RoyK | cansandstring.conf |
12:06.53 | mosty | dudes: i want to allow ulaw for sip->sip calls, but if i put allow=ulaw in sip.conf then won't that allow ulaw for sip->zap calls? |
12:07.03 | wasim | mosty: SetGroup |
12:07.26 | Thoran | mosty: ISDN |
12:07.27 | *** join/#asterisk shido6 (n=curtis@d221-68-210.commercial.cgocable.net) |
12:07.34 | dudes_ | gein - Bug Reported before option added. |
12:08.09 | gein | yeah, but then this should have been solved? And I can't even get it to work with CVS-HEAD |
12:08.46 | Druken | is it me, or is more shit broken in stable then in head? |
12:09.00 | dudes_ | Are you using head? |
12:09.15 | gein | I've tried both stable and head |
12:09.29 | gein | no luck at all... joinempty=strict seems to work as joinempty=no |
12:09.34 | dudes_ | Druken - A lot of docs are written about head, not stable. |
12:10.08 | mosty | wasim, i don't understand what setgroup does |
12:11.27 | mosty | Thoran, and you've setup your dialplan to dial out on a zap channel for pstn calls? |
12:11.45 | Druken | mostly: just don't allow ulaw on the zap channels |
12:12.02 | Druken | why you don't want ulaw to your zaps i don't know... but that should fix it |
12:12.04 | dudes_ | mosty - what are you trying to do |
12:12.38 | *** join/#asterisk lathos42 (n=lathos42@65-42-27-66.dowdingindustries.com) |
12:12.40 | mosty | druken: but won't that just make asterisk transcode ulaw to some other codec? i don't want to allow ulaw for sip calls unless it's sip->sip with asterisk just doing the signalling |
12:13.13 | mosty | dudes: i want to allow ulaw for sip->sip calls (where asterisk is just doing the signalling), but not allow ulaw for any other situation |
12:13.14 | dudes_ | mosty - zap uses slin which is similiar to ulaw in the fact that it's raw data |
12:13.14 | RoyK | anyone here that knows how I can use an analog link to send SMS? |
12:13.18 | Druken | mosty: probably... i highly doubt you can do what you want |
12:13.21 | RoyK | can i do that with asterisk? |
12:13.25 | *** join/#asterisk klictel (n=klictel@207.107.208.140) |
12:13.38 | klictel | morning all |
12:14.16 | Druken | w00t! costers! |
12:14.24 | Druken | er.. coasters |
12:14.29 | mosty | dudes: i don't want incoming network data to use anything but compressed codecs, say gsm or g729, but i don't mind if sip->sip calls use other codecs since asterisk isn't in the media path |
12:14.59 | Thoran | @mosty yes |
12:15.18 | Thoran | exten => _8.,1,Dial(Zap/g1/${EXTEN:1}/,60) |
12:15.18 | Thoran | exten => _8.,2,Congestion |
12:15.18 | Thoran | exten => _8.,3,Busy |
12:15.18 | Thoran | exten => _8.,4,Hangup |
12:15.29 | mosty | thoran: and what error do you get in the asterisk logs when you make such a call? |
12:15.46 | dudes_ | Then allow g729/gsm on all context that are outgoing |
12:15.49 | Thoran | sometimes I get cannot forward voice |
12:15.50 | dudes_ | or incoming |
12:16.06 | Thoran | like some codec problem |
12:16.15 | dudes_ | But that's only the first few bytes normally |
12:16.23 | Thoran | can I paste my logfile here ? |
12:16.30 | mosty | thoran: no, use a paste site |
12:16.37 | *** join/#asterisk gambolputty (n=gambolpu@72.240.241.108) |
12:17.54 | mosty | dudes: a context in the dialplan? |
12:18.43 | dudes_ | Set your codecs you want to use in the contexts |
12:19.10 | dudes_ | ie, sip.conf context for your incoming/outbound providers to disallow=all allow=g729 |
12:19.17 | Thoran | just a sec |
12:19.23 | dudes_ | then allow your sip phones to allow=all |
12:19.24 | Thoran | asterisk acting up atm |
12:20.50 | mosty | dudes: what is an incoming sip provider? is that the same as a phone? |
12:20.54 | *** join/#asterisk djin_ib (n=djin_ib@217-19-19-229.dsl.cambrium.nl) |
12:26.41 | RoyK | anyone here that knows how I can send SMS over an analog line with a modem or perhaps a digium board? |
12:27.24 | mosty | royk: i don't think it's possible over POTS lines |
12:27.36 | RoyK | mosty: i beleive it is... |
12:28.33 | mosty | royk: i'm pretty sure in australia you can't, i don't know about the rest of the world |
12:30.28 | dudes_ | iDunno - enable it and try it out |
12:31.26 | iDunno | dudes_: I, erm, did. if I hit *2 with the DTMF pad, it appears to just throw the DTMF through to the other end, if I use the transfer button on kphone, it appears to do a blind transfer :/ |
12:34.46 | dudes_ | so you see how it works |
12:37.14 | iDunno | ... erm, not at all?! ;) |
12:37.51 | *** join/#asterisk siggi (n=siggi@wlan-gw.denic.de) |
12:39.55 | dudes_ | it allows you to answer the call |
12:40.12 | *** join/#asterisk docE (n=docE@66.237.242.41.ptr.us.xo.net) |
12:40.22 | dudes_ | then if you hangup before the called party hangs up it's connected to the party ou were talking to |
12:40.49 | iDunno | erm, that's how I expect it to work, that's not what's happening... |
12:41.12 | docE | Anyone in here work on the h323 addons module for asterisk? I am having issues with it crashing and taking down asterisk |
12:41.14 | dudes_ | so you hit *2 ... hit the extensions |
12:41.29 | iDunno | dudes_: that appears to be the bit that doesn't work. |
12:41.36 | Nugget | I think I'll call him tonight at 03:30 netherlands time. |
12:41.55 | dudes_ | So what does it do |
12:42.07 | *** join/#asterisk arnis (n=arnizzz@82.148.188.56) |
12:42.43 | arnis | So have anyone had success with cmd Conference on Gentoo and asterisk 1.0.9? |
12:42.51 | iDunno | dudes_: the *2 just gets sent to the other phone, it doesn't get picked up by asterisk or anything else |
12:42.58 | iDunno | dudes_: as real tones. |
12:43.30 | dudes_ | Are you the called party or the calling party |
12:43.46 | iDunno | both, in this case. but for the *2 I'm the called. |
12:43.50 | arnis | My asterisk just hangs when a second caller tried to enter the conference. |
12:44.00 | arnis | tries* |
12:44.12 | iDunno | and the Dial has got a tr in it. |
12:44.20 | dudes_ | arnis - never had that. Get some debug info |
12:44.49 | dudes_ | Maybe your DTMF's are getting send right |
12:45.41 | iDunno | oh bother. |
12:46.05 | *** join/#asterisk djin (n=djin_ib@84-245-25-231.dsl.cambrium.nl) |
12:47.28 | iDunno | nope, that still wasn't quite right. |
12:49.41 | arnis | What do you mean by Debug information? |
12:50.00 | arnis | the information i get in asterisk -vvvvvgr? |
12:50.49 | arnis | I get loads of notices and a few warnings on unable to translate from unkown to unkown/alaw |
12:52.04 | dudes_ | iDunno - erm, I don't know then. What does your sip context look like? |
12:52.26 | dudes_ | arnis - probably an unknow codec that can't be translated |
12:52.47 | dudes_ | show translation on the CLI will show you what you can translate from to |
12:55.27 | arnis | k. thanks. My can't translate anything from g723 or g729. Guess i have to set my phones to g726 then? |
12:55.57 | dudes | arnis - that'd be about the case. Or buy the g729 license |
12:56.29 | dudes | Or if your provide supports g729 and your phone you can passthru |
12:57.32 | arnis | I can call and recieve just find with the current codecsettings so it's wierd if that should be the problem. Don't you think? |
12:59.08 | Thoran | one of the guys who wanted to help me still here ? I found out what a paste site is :) |
12:59.37 | arnis | ok. but thanks anyway. i have to log off to reach my phones. (stupid wireless think). |
12:59.55 | *** join/#asterisk cuco (n=elcuco@local.xorcom.com) |
13:00.06 | iDunno | dudes: http://share.runtime-collective.com/~brettp/sip-context.conf |
13:00.09 | cuco | does anyone has a FWD account? |
13:00.10 | mosty | thoran: like http://pastebin.ca/ |
13:00.11 | iDunno | it looks a little like that |
13:00.21 | tzafrir | hi, anybody here with a FWD account (preferebly IAX)? |
13:00.34 | cuco | tzafrir: me :) |
13:00.42 | mosty | thoran: or http://rafb.net/paste/ |
13:00.45 | tzafrir | oops, ignore (note IPs of cuco and tzafrir) |
13:02.44 | KeX_WorX | anyone knows if there is a deb package for asterisk-1.2-beta1 ? |
13:03.14 | cuco | KeX_WorX: not yet, that I know of |
13:04.54 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
13:05.18 | KeX_WorX | cuco, k. u know if ther is one in dev or planing ? |
13:06.33 | *** join/#asterisk nesys (n=nesys@2001:1418:1a6:0:20d:93ff:fe28:3ef8) |
13:07.51 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
13:07.57 | *** join/#asterisk MikeJ[Laptop] (n=ircatjer@mi.origenfinancial.com) |
13:07.59 | tzafrir | I hope that this is a matter of less than a week |
13:09.04 | RoyK | can someone help me understand how SMS is transported over isdn/analog line? using data/modem/spandsp, or how does this work? |
13:09.30 | *** join/#asterisk arnis (n=arnizzz@82.148.188.56) |
13:09.39 | docE | Anyone in here work on the asterisk-ooh323 addons package for asterisk? I am having some issues |
13:09.47 | arnis | nope. didn't change anything. |
13:10.20 | arnis | also tested with G711a 64k on my phone.. |
13:11.00 | *** join/#asterisk pussfeller (n=todd@12.150.129.171) |
13:13.14 | Thoran | mosty: http://pastebin.com/376763 |
13:13.35 | KeX_WorX | tzafrir, u mean the deb package, with less than a week ? |
13:14.30 | klictel | you have 8 Xs? |
13:16.31 | klictel | Thoran: the number showing in dial has 8 digits? |
13:17.23 | *** join/#asterisk coppice (n=chatzill@246.204.17.210.dyn.pacific.net.hk) |
13:17.28 | *** join/#asterisk psycodad (n=obiwan@2001:4060:4419:b1:0:0:0:2) |
13:18.04 | *** join/#asterisk ^X-works (n=r0x0r@81-208-62-98.ip.fastwebnet.it) |
13:18.23 | *** join/#asterisk Abbas (n=Abbas@203.81.196.92) |
13:18.54 | Thoran | i am not sure what you mean |
13:19.18 | Thoran | I dial 8 |
13:19.21 | Thoran | then the number |
13:19.39 | Thoran | The XXXXs are the number i dialed minus the 8 |
13:19.56 | Thoran | in this case the number has 8 digits yes |
13:22.01 | *** join/#asterisk riemensc (i=riemensc@83-169-155-130-dynip.superkabel.de) |
13:22.09 | riemensc | hello |
13:24.41 | arnis | voip*CLI> show conferences |
13:24.41 | arnis | Deprecated! Please use 'meetme' instead. |
13:24.48 | arnis | bah |
13:26.59 | Katty | yay, morningstarfarms has 'chicken' and 'steak' strips! |
13:27.10 | kajtzu | arnis: yes, you want "meetme list" |
13:27.52 | tzanger | how do you get a vegan *chicken* enchilada? |
13:28.25 | coppice | most chickens are vegans |
13:28.32 | iDunno | vegan!=vegetarian |
13:28.47 | kajtzu | "A vegan diet is one which excludes any animal products." |
13:28.49 | *** join/#asterisk Mike9 (n=sturdee@ireland.pathwaynet.com) |
13:28.56 | kajtzu | iDunno: try using google define:vegan |
13:29.00 | sivana | A vegan diet is one which excludes any animal products. |
13:29.10 | sivana | heh |
13:29.20 | *** join/#asterisk pussfeller (n=todd@12.150.129.171) |
13:29.20 | MikeJ[Laptop] | what about sproutists |
13:29.20 | iDunno | ah. |
13:29.23 | iDunno | hmm |
13:29.26 | sivana | as long as they pet the chicken first, before they kill it, than it's ok |
13:29.40 | gordonjcp | yeah |
13:29.50 | coppice | that means they can't east veggies, because those are made from animal droppings :-\ |
13:29.52 | kajtzu | ;-) |
13:29.54 | sivana | hehe |
13:29.54 | gordonjcp | I wondered about that, surely having chicken in precludes it being vegan |
13:30.00 | kajtzu | gordonjcp: me too ;) |
13:30.02 | Katty | you know, if you're going to tease people |
13:30.09 | Katty | find something worth teasing them about (= |
13:30.13 | sivana | :) |
13:30.15 | kajtzu | .. make sure they get it first? ;) |
13:30.23 | Katty | kajtzu: grow up ;) |
13:30.31 | gordonjcp | Katty: not teasing, just wondering how you can have a vegan chicken *anything* |
13:30.32 | Katty | tzanger: it's soy, silly rabbit. |
13:30.37 | kajtzu | Katty: nah |
13:30.45 | Katty | gordonjcp: it's not /real/ chicken. it's a soy based substitute. |
13:30.51 | kajtzu | Katty: growing up is for people iwth no imagination ;) |
13:30.52 | Katty | gordonjcp: duh :P |
13:30.52 | gordonjcp | kajtzu: surely that would be a tofu enchilada then? |
13:31.02 | Thoran | @mosty: Do you happen to have an idea, what I did wrong ? |
13:31.04 | tzanger | Katty: ok but if it's soy then what makes it chicken? Or do you mean chicken-flavoured? |
13:31.07 | *** join/#asterisk PoWeRKiLL (n=PoWeRKiL@corail-gw.clients.easynet.fr) |
13:31.17 | coppice | I think all those Buddist fak meat dishes are deeply against the spirit of being a vegitarian |
13:31.18 | gordonjcp | hang on, why would you make the tofu chicken-flavoured? |
13:31.19 | Katty | tzanger: it looks (and simulates) a non breaded chicken strip. |
13:31.24 | gordonjcp | coppice: indeed |
13:31.26 | PoWeRKiLL | hi coppice |
13:31.27 | tzanger | Katty: ah okay |
13:31.37 | PoWeRKiLL | I have a core file to give you |
13:31.44 | gordonjcp | Katty: why, though? surely if you don't want to eat meat, you wouldn't want to eat meat-flavoured things? |
13:31.53 | Katty | gordonjcp: there IS NO MEAT IN IT |
13:31.56 | Katty | gordonjcp: THE PRODUCT IS VEGAN |
13:31.58 | Katty | gordonjcp: kthx. |
13:32.04 | gordonjcp | Katty: yes, but it's meat-flavoured |
13:32.06 | coppice | PoWeRKiLL any core files are due to someone else's code |
13:32.09 | Katty | gordonjcp: it is not. |
13:32.17 | Katty | gordonjcp: it would not be vegan if there was meat flavoring. |
13:32.18 | gordonjcp | Katty: it's chicken flavoured |
13:32.25 | Katty | gordonjcp: what part of no meat don't you get? :P |
13:32.37 | gordonjcp | Katty: what part of my question don't you get? |
13:32.42 | Katty | gordonjcp: apparently all of it. |
13:32.43 | gordonjcp | it tastes like chicken, right? |
13:32.48 | Katty | no |
13:32.51 | Katty | it simply simulates it |
13:32.57 | PoWeRKiLL | coppice when a specific fax machine send me a fax it's crash asterisk |
13:33.19 | Katty | gordonjcp: vegans are againts the unethical treatment of animals. not the taste. |
13:33.24 | MikeJ[Laptop] | it tastes like chicken, not tastes of chicken |
13:33.25 | Thoran | ok I found it |
13:33.26 | Katty | what a moron ;) |
13:33.29 | sivana | now we have engineered chicken meat |
13:33.38 | Katty | sivana: they've had it for years. |
13:33.41 | sivana | hrm.. cancer not on the rise |
13:33.45 | Thoran | there was a / at a place where there shouldn't be one |
13:33.52 | coppice | PoWeRKiLL: can't you just look at it with GDB and tell me where it died? |
13:33.59 | Katty | anyway! |
13:34.00 | gordonjcp | Katty: mmm, that sounds a bit spurious to me |
13:34.02 | sivana | hehe |
13:34.06 | MikeJ[Laptop] | mmmm gotta love that soy protine :D |
13:34.23 | Katty | i'm plotting using these chicken soy strips to make enchiladas |
13:34.29 | gordonjcp | if I was a vegan, I wouldn't eat stuff that tasted like meat products |
13:34.36 | MikeJ[Laptop] | they are good... and I eat chicken |
13:34.53 | Katty | gordonjcp: that's your call then. |
13:35.00 | gordonjcp | Katty: yes, true |
13:35.04 | Katty | gordonjcp: i have no personal problem with it. they taste good. they're not harming animals in the process. |
13:35.08 | MikeJ[Laptop] | if your vegan, you need protine from somewhere.. simulated meat products are a very easy way to do it |
13:35.11 | gordonjcp | I am also against the unethical treatment of animals |
13:35.20 | coppice | the fake meat in the Buddist monestaries can be a pretty good fake |
13:35.22 | MikeJ[Laptop] | except chickens :D |
13:35.37 | MikeJ[Laptop] | cuz they are mean anyways |
13:35.45 | MikeJ[Laptop] | I'm not |
13:35.52 | sivana | me either :) |
13:35.52 | MikeJ[Laptop] | I am being fecious... |
13:35.56 | gordonjcp | Katty: I don't know about anyone else, but I'm not picking on you for being vegan |
13:36.01 | coppice | I'm against the unethical treatment of vegetables |
13:36.01 | tzanger | fecious? haha |
13:36.02 | Katty | excellent. |
13:36.04 | MikeJ[Laptop] | I have a veggy wife |
13:36.05 | Katty | then stop teasing me :P |
13:36.07 | tzanger | facetious maybe? |
13:36.13 | gordonjcp | I really just don't get why people who don't want to eat animals want to eat things made up to taste like animals |
13:36.14 | MikeJ[Laptop] | sure |
13:36.17 | PoWeRKiLL | coppice yes it's die in spandsp |
13:36.20 | MikeJ[Laptop] | I need a spell checker |
13:36.25 | PoWeRKiLL | gordonjcp where are you from ? |
13:36.29 | coppice | MikeJ: severe brain damage? :=) |
13:36.41 | Katty | gordonjcp: if you dont' want it. don't eat it. |
13:36.50 | gordonjcp | PoWeRKiLL: Scotland |
13:36.51 | Katty | gordonjcp: doesn't mean you have to go harping on people who do |
13:36.53 | Katty | gordonjcp: don't be a snob |
13:36.57 | coppice | PoWeRKiLL: can you be a little more specific? like the file and line number? |
13:37.09 | gordonjcp | Katty: I only eat animals that I know personally |
13:37.21 | Katty | i think we should just move on (= |
13:37.23 | Katty | NEXT |
13:37.31 | kajtzu | what I'm not getting is how a vegan anything can be chicken anything no matter what |
13:37.47 | gordonjcp | kajtzu: it tastes a bit like chicken, apparently |
13:37.49 | Katty | i give up. |
13:37.59 | kajtzu | gordonjcp: how can it, not being chicken ;) |
13:38.04 | coppice | i'm over 80% vegetarian. its only my mouth which eats meat |
13:38.07 | kajtzu | gordonjcp: at the very least it is deceptive advertising ;) |
13:38.09 | gordonjcp | coppice: lol |
13:38.09 | sivana | ok.. ok.. stop.. if vegans want to eat fake, man-engineered meat, then who cares |
13:38.13 | *** join/#asterisk Ariel_ (n=Ariel@adsl-068-157-125-248.sip.mia.bellsouth.net) |
13:38.19 | Katty | MikeJ[Laptop]: does your wife have any really good recipes? |
13:38.27 | kajtzu | anyway... vegatarian stuff is what food eats >:) |
13:38.30 | sivana | oops.. not politically correct |
13:38.35 | sivana | human-engineered |
13:38.41 | [jedi] | coppice: I'm trying to find out a bit more about the problem I'm having with spandsp + txfax... what would you need from my bug report so that it's useful? |
13:38.49 | *** join/#asterisk mutilator (n=animenod@65.111.201.79) |
13:38.59 | sivana | [jedi]: what problem you having? |
13:39.07 | coppice | jedi: a patch which fixes the problem :-) |
13:39.16 | [jedi] | sivana: txfax gets executed but does nothing |
13:39.17 | PoWeRKiLL | coppice #0 0x4227b54b in process_baud () from /usr/lib/libspandsp.so.0 |
13:39.19 | Katty | MikeJ[Laptop]: because if she does, you must commander them. |
13:39.30 | [jedi] | coppice: I wish I was able to do that :D |
13:39.45 | PoWeRKiLL | #9 0x0807be5e in pbx_extension_helper (c=0x8e5bbe0, |
13:39.45 | PoWeRKiLL | <PROTECTED> |
13:39.45 | PoWeRKiLL | <PROTECTED> |
13:39.45 | PoWeRKiLL | <PROTECTED> |
13:39.45 | PoWeRKiLL | #10 0x08075faa in ast_pbx_run (c=0x42253bb4) at pbx.c:1769 |
13:39.46 | PoWeRKiLL | #11 0x0807c521 in pbx_thread (data=0x0) at pbx.c:1992 |
13:39.48 | PoWeRKiLL | #12 0x4002aa21 in pthread_start_thread () from /lib/i686/libpthread.so.0 |
13:40.10 | coppice | PoWeRKiLL: which version of spandsp? |
13:40.19 | Katty | MikeJ[Laptop]: i mean commandeer. |
13:40.39 | coppice | jedi: I suppose you did use the "caller" parameter? |
13:40.52 | [jedi] | I'm using 0.0.2pre20, with a CVS asterisk from july 13+- |
13:41.04 | [jedi] | coppice: of course |
13:41.13 | *** join/#asterisk IPmonger (n=ipmonger@pcp0010577106pcs.coatsv01.pa.comcast.net) |
13:41.15 | [jedi] | coppice: in fact, when I dial my own phone, I can hear the "beep" |
13:41.37 | [jedi] | but when I dial a real fax machine it doesn't do anything, nor it shows anything in any log or in the screen, even with debug parameter |
13:42.08 | gordonjcp | Katty: I've got some semi-decent recipes, which I will chuck in your direction as soon as I get time to write them up |
13:42.14 | [jedi] | I've tried three different fax machines to discard a problem with a concrete fax |
13:42.27 | coppice | PoWeRKiLL: what happened to backtrace entries 1 to 8? |
13:42.36 | Katty | gordonjcp: i see. |
13:42.40 | Katty | gordonjcp: k |
13:42.49 | gordonjcp | Katty: you'd like the miso soup one |
13:42.54 | *** join/#asterisk Moc (n=mochouin@modemcable173.101-70-69.mc.videotron.ca) |
13:43.01 | coppice | everyone likes miso |
13:43.02 | Katty | gordonjcp: i'm not a big soup person, but sure. |
13:43.03 | Moc | morning |
13:43.05 | [jedi] | what's miso? |
13:43.09 | Katty | mister Moc (= |
13:43.27 | coppice | miso == key staple of the japanese diet |
13:43.33 | [jedi] | oh |
13:43.39 | Katty | rice? |
13:44.37 | gordonjcp | [jedi]: it's a kind of bean curd paste |
13:44.38 | *** join/#asterisk kahuna_ (n=booger@209-254-56-194.ip.mcleodusa.net) |
13:44.52 | kahuna_ | Hi |
13:44.55 | [jedi] | ok |
13:45.10 | *** join/#asterisk spackle (n=spackle@209.234.83.19) |
13:46.19 | [jedi] | coppice: have you ever heard of a problem like mine with txfax? or i'm the only one ? |
13:47.02 | coppice | txfax generally works, although people have been complaining lately of a high corrupt pages issue |
13:47.03 | Connor_ | When you guys setup you linux box for VoIP applications, how do you all setup the partitions.. |
13:47.23 | [jedi] | I wish I had page corruption, at least |
13:47.27 | [jedi] | :) |
13:47.44 | spackle | "Cheese Grommit." |
13:47.47 | *** join/#asterisk Corydon76-home (i=mauve@pdpc/supporter/sustaining/Corydon76-home) |
13:49.18 | Katty | gordonjcp: k'then |
13:49.46 | *** join/#asterisk jhiver (n=jhiver@AStDenis-105-1-4-4.w193-253.abo.wanadoo.fr) |
13:49.49 | jhiver | hi all |
13:50.13 | jhiver | I've *finally* finished 'frenchizing' and 'cleaning' up astcc, wohoo :) |
13:50.22 | jhiver | it was like, a big day's work :) |
13:50.23 | *** join/#asterisk doughecka_ (n=Miranda@unaffiliated/doughecka) |
13:50.31 | [jedi] | jhiver: I gave up on that months ago |
13:50.35 | zoid99 | what kind of "cleanup" did you do? |
13:50.37 | [jedi] | jhiver: developed my own calling card app |
13:50.43 | jhiver | use strict; :) |
13:51.08 | zoid99 | cool |
13:51.23 | jhiver | I've also done quite a few mods so that it tells the max. lenght of call |
13:51.32 | jhiver | and it speak like, french, now :) |
13:51.47 | djin | You do not appear to have the sources for the 2.6.9-11.ELsmp kernel installed (while compiling zaptel). I installed the kernel sources (2.6.9-11.EL) and made the links /usr/src/linux and /usr/src/linux-2.6 pointing to this source. What am I overlooking? |
13:51.55 | jhiver | audiocity came in handy but I have to do more work because my mike is shit and my voice ain't any better |
13:51.59 | zoid99 | one step forward.. 2 steps backwards :) |
13:52.05 | kajtzu | djin: missing kernel-devel-smp |
13:52.10 | kajtzu | (or something) |
13:52.32 | kajtzu | kernel-smp-devel-2.6.9-11.EL.x86_64 |
13:52.39 | kajtzu | (or whatever your arch is) |
13:53.09 | djin | I have kernel-devel, but no kernel-smp-devel. |
13:53.12 | jhiver | but it's cool! I don't know if it's faster starting from scratch, but considering it took one day to install Xorcom + AstCC + play around with it, and another day to customize it, i'd say 'no' |
13:53.25 | [jedi] | zaptel works well with x86_64 machines? |
13:53.27 | kajtzu | djin: you have a smp kernel, you need the kernel-smp-devel package |
13:53.28 | djin | kajtzu, but it's downloading ;) |
13:53.35 | kajtzu | [jedi]: yes |
13:53.37 | kajtzu | djin: check ;) |
13:53.38 | [jedi] | great |
13:53.47 | [jedi] | jhiver: astcc is quite limited |
13:53.59 | kajtzu | djin: you don't need kernel-devel thjougu |
13:54.00 | kajtzu | though |
13:54.08 | jhiver | yeah I realize that |
13:54.12 | jhiver | but hey, it's a start |
13:55.04 | *** join/#asterisk Laerte (n=io@195.47.232.200) |
13:55.05 | Laerte | hy |
13:55.07 | [jedi] | most calling-card providers do many tricks with the pricing of the cards, which can't be done with astcc |
13:55.22 | djin | kajtzu, very cool. thanks :) |
13:55.28 | kajtzu | djin: np |
13:55.28 | jhiver | yeah sure like disconnect charges and other horrible stuff ;) |
13:55.48 | [jedi] | hehe |
13:55.49 | [jedi] | yes |
13:55.58 | [jedi] | I had to develop a bunch of these tricks in my app |
13:56.17 | [jedi] | I really hated it but not my choice |
13:56.30 | jhiver | I don't quite get what 'brand' is all about with AstCC |
13:56.58 | jhiver | There's a few things I'm not familiar with |
13:57.02 | [jedi] | brand is a product |
13:57.11 | [jedi] | you may have different products, different cards |
13:57.12 | *** join/#asterisk dalfry (n=dalfry@gateway.ishisystems.com) |
13:57.38 | jhiver | mhhh |
13:57.50 | [jedi] | you may apply different tariffs to each product |
13:57.56 | jhiver | so 'brand name' I get that... |
13:58.04 | jhiver | then 'language' that seems fair enough... |
13:58.13 | jhiver | 'published number' and 'did' WTF? |
13:58.22 | jhiver | Inc = Increment I would guess |
13:58.30 | jhiver | Service Fee = ??? |
13:58.33 | [jedi] | I don't really know what uses these numbers astcc for |
13:58.38 | jhiver | Service Fee Days = ??? again |
13:58.42 | [jedi] | but all these are tricks |
13:58.49 | jhiver | and markup = ??? markup of what? |
13:58.53 | [jedi] | markup on pricing |
13:59.02 | [jedi] | you put a pricing for each route |
13:59.14 | [jedi] | then you put a markup on these pricing |
13:59.24 | [jedi] | astcc tariffication model is too simple |
13:59.30 | jhiver | OK so this is for some kind of reseller malarki |
13:59.40 | [jedi] | or at least it was last time I checked |
13:59.56 | [jedi] | most calling card distributors will want many features astcc hasn't |
14:00.03 | [jedi] | most of them want a clone of digitalk :D |
14:00.15 | jhiver | is this your product? |
14:00.22 | [jedi] | digitalk ? no hehe |
14:00.46 | [jedi] | digitalk is almost a 'standard' platform for calling cards |
14:00.51 | jhiver | ah ok |
14:00.58 | [jedi] | is a switching platform which has calling card facilities |
14:01.11 | mutilator | holy fscking nuts |
14:01.15 | coppice | where did that PoWeRKiLL fellow go with the other half of his bug report? :-\ |
14:01.30 | jhiver | ok and I guess that costs a mint as well doesn't it |
14:01.32 | mutilator | my brother just called my cellphone from his cell, i said hello, then all the sudden it got cut off and i was talking to someone else |
14:01.33 | clive- | I have digitalk, its a great product |
14:01.44 | clive- | I have used digitalk, i mean |
14:01.49 | mutilator | so i called my brother back, and he said the same, he got cut off and someone else was on the line |
14:01.52 | [jedi] | it's quite cryptic |
14:01.55 | [jedi] | but it's powerful, yes |
14:02.03 | mutilator | like the telco just crossed the lines or some weird thing |
14:02.07 | mutilator | was weird |
14:02.21 | jhiver | clive-, what's so cool about it? |
14:02.52 | clive- | jhiver it just has every feature in the callingcard book |
14:02.52 | mutilator | wonder how that happened |
14:03.06 | clive- | i probably costs a fortune as well |
14:03.11 | clive- | it* |
14:03.12 | [jedi] | jhiver: it can do anything. It's just quite difficult to manage |
14:03.17 | clive- | jeez my typing today |
14:03.21 | Thoran | Abbas ? |
14:03.29 | wasim | mutilator: its very common with mobilink in pk |
14:03.48 | jhiver | ok, so it's basically what Oracle is to RDBMS, some monster elephant that does it all but actually works |
14:04.03 | mutilator | he uses cingular and i have verizon wireless |
14:04.11 | mutilator | never even heard of that happenin before |
14:04.14 | [jedi] | jhiver: almost hehe |
14:04.24 | [jedi] | jhiver: oracle is not as pricey as digitalk |
14:04.37 | mutilator | he got connected to someone looking for parts at a store, and i got someone trying to reach some number down in detroit |
14:05.37 | jhiver | you guys have tried other CC apps? |
14:05.47 | jhiver | other than AstCC that is? |
14:06.41 | spackle | ~seen bkw_ |
14:06.45 | jbot | bkw_ is currently on #asterisk (19h 13m 32s). Has said a total of 85 messages. Is idling for 11h 3m 53s |
14:06.47 | clive- | jhiver astcc is just a perl script, you can easly just add in any feature you want |
14:07.02 | coppice | ~seen redder86 |
14:07.03 | jbot | redder86 <n=lee@gateway.howardsilvan.com> was last seen on IRC in channel #asterisk, 22h 48m 23s ago, saying: 'c'mon down!'. |
14:07.06 | jhiver | clive-, I know i've done that today :) |
14:07.06 | bkw_ | yes |
14:08.42 | jhiver | clive-, I've done a few mods to AstCC it's quite easy to work with although I would have preferred a nice OO perl code :) |
14:09.01 | *** join/#asterisk dudes (n=dudes@12-215-34-84.client.mchsi.com) |
14:09.04 | [jedi] | mine is done on java |
14:09.16 | jhiver | mhhh interesting |
14:09.31 | coppice | jedi: cheap indonesian labour? |
14:09.38 | jhiver | is there a page somewhere about it or it kept inhouse? |
14:09.42 | [jedi] | coppice: cheap my-hands labour |
14:10.01 | zoid99 | jedi: Did you use the asterisk-java suite? |
14:10.04 | [jedi] | zoid99: yes |
14:10.17 | skyflex | dudes: vad gör du då? |
14:10.19 | zoid99 | very nice framework |
14:10.21 | skyflex | who, wrong chan |
14:10.23 | [jedi] | yes, very powerful |
14:10.24 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
14:10.24 | *** mode/#asterisk [+o anthm] by ChanServ |
14:10.32 | *** join/#asterisk mhnoyes (n=mhnoyes@user-2ivfldr.dialup.mindspring.com) |
14:10.36 | *** part/#asterisk Moc (n=mochouin@modemcable173.101-70-69.mc.videotron.ca) |
14:10.51 | zoid99 | we've used Asterisk-Java on a few projects |
14:11.09 | [jedi] | I tried JAGI but its non-threaded approach was not as easy to manage as asterisk-java's |
14:11.13 | zoid99 | only on the manager side.. not agi |
14:11.34 | RoyK | yagi antenna |
14:11.35 | RoyK | :P |
14:11.45 | [jedi] | FastAGI + Hibernate + Spring == powerful robust well-architected Agi in a breeze |
14:11.45 | kahuna_ | I'm getting this: PRI Error: We think we're the CPE, but they think they're the CPE too. |
14:12.05 | *** join/#asterisk wsuff (n=wsuff@pcp04243496pcs.eatntn01.nj.comcast.net) |
14:12.08 | zoid99 | What is spring? |
14:12.09 | kahuna_ | When I switch signalling to pri_net I get the same except switch thinks theyre net too... |
14:12.17 | [jedi] | zoid99: spring framework... java stuff |
14:12.18 | kahuna_ | How do I fix that? |
14:12.40 | [jedi] | zoid99: www.springframework.org |
14:12.44 | zoid99 | Hybernate is DB persistence... is Spring for UI? |
14:12.48 | *** part/#asterisk wsuff (n=wsuff@pcp04243496pcs.eatntn01.nj.comcast.net) |
14:12.48 | [jedi] | no |
14:13.28 | [jedi] | spring is a framework which adds Aspect-oriented features, Inversion of Control, declarative transactions, all without messing with your own code |
14:14.03 | [jedi] | it does also transparent remoting throught many protocols and many other things, but I don't use these features right now |
14:14.44 | zoid99 | nice.. |
14:14.59 | *** join/#asterisk g__ (n=goakham@itd01fw-fibre.itdepartment.com) |
14:15.43 | kahuna_ | maybe my provider is running an interface loop? |
14:17.11 | *** join/#asterisk krisguy (n=krisguy@h216-170-039-057.adsl.navix.net) |
14:18.16 | L|NUX | evening all |
14:22.04 | *** join/#asterisk zobia (i=zobia@222.212.71.126) |
14:22.51 | zobia | hello everyone , i want to wait () how many sec equal to 4 rings ? i want to let it ring 4 times and transfer a inbound call |
14:23.12 | cpatry | zobia: take a watch and seee |
14:24.24 | zobia | cpatry. do u know what's difference between wait() and ring? |
14:25.07 | cpatry | no i dont. are they both animals? |
14:26.48 | RoyK | anyone using zaphfc with pci-s? |
14:27.36 | *** join/#asterisk file (n=jcolp@mctnnbsa31w-142166116178.nb.aliant.net) |
14:27.36 | *** join/#asterisk santiago (n=santiago@63.245.86.245) |
14:27.36 | *** join/#asterisk alerios (n=alerios@200.24.109.199) |
14:28.59 | Thoran | I am searching for SIP Phones which support VLAN Tags. I have found Grandstream GXP 2000, some cisco phones, and snom 320 and 360. Does any1 know other phones which support VLAN ? Preferably for a price under 200 Eu. |
14:29.06 | *** join/#asterisk lubomier (i=lubomier@217.118.109.179) |
14:29.15 | *** join/#asterisk azzie (i=az@cpe-24-168-17-173.si.res.rr.com) |
14:29.59 | lubomier | hi, it's here anybody who configuers avm fritz!pci 2.0 in NT mode? or point-to-point? |
14:30.08 | _m_ | Thoran: the snom 190 supports VLAN, too |
14:30.24 | *** join/#asterisk FABRIZIOxxx (n=FABRIZIO@81-208-26-86.ip.fastwebnet.it) |
14:30.39 | Thoran | ok thx |
14:30.39 | *** join/#asterisk dsfr (n=dsfr@pdpc/sponsor/digium/dsfr) |
14:31.11 | FABRIZIOxxx | hello .. i was just wondering which is the best echo cancellation method for the zaptel driver ..? mar, mark2, mark, steve or steve2 .. anyone have some experience?? |
14:31.54 | *** join/#asterisk Mc_Tr (n=Mc_Tr@bacterio.knet.es) |
14:32.04 | Mc_Tr | hi! |
14:32.05 | jake1932 | FABRIZIOxxx: kb1 is the current |
14:32.11 | Mc_Tr | has anybody running asterisk? |
14:32.19 | Mc_Tr | i need to do one question. |
14:32.22 | jake1932 | no - try the asterisk channel |
14:32.31 | FABRIZIOxxx | oh ok .. so do you mean i have to use the zaptel 1,2 driver? |
14:32.52 | jake1932 | FABRIZIOxxx: the newest echo can is kb1 |
14:32.58 | Mc_Tr | know how i do a Phone Book in asterisk? |
14:32.58 | jaxkz | Mc_Tr: just as the question |
14:33.34 | Mc_Tr | jaxkz: do you know i can have a centralized phone book? |
14:33.35 | jake1932 | http://lists.digium.com/pipermail/asterisk-users/2005-August/122944.html |
14:34.12 | jaxkz | Mc_Tr: i do not |
14:34.26 | Mc_Tr | ok, thanks jaxkz |
14:34.28 | jaxkz | I think you need an ldap implementation for that |
14:35.08 | Mc_Tr | i'm searching in voip-info.org and google, but nothing happens :( |
14:35.42 | *** part/#asterisk santiago (n=santiago@63.245.86.245) |
14:38.16 | *** join/#asterisk Blazint (n=blazin@cm225.epsilon203.maxonline.com.sg) |
14:38.54 | *** join/#asterisk e3g (i=p@203.215.180.250) |
14:38.56 | e3g | hi |
14:39.09 | *** join/#asterisk folsson (n=filip@h82n1fls32o985.telia.com) |
14:39.13 | e3g | I have a question about Asterisk Manager. |
14:39.53 | *** join/#asterisk PoWeRKiLL (n=PoWeRKiL@corail-gw.clients.easynet.fr) |
14:39.59 | e3g | I want my manager to receive only specific Events, currently he receives all the events. |
14:40.05 | PoWeRKiLL | coppice still here ? |
14:40.13 | PoWeRKiLL | coppice sorry i got a internet disconnection |
14:40.25 | PoWeRKiLL | coppice you want a bt full by mail ? |
14:41.06 | coppice | yep |
14:41.19 | *** join/#asterisk DagMoller (n=DagMolle@2001:5c0:8fff:ffff:0:0:0:41) |
14:42.23 | jake1932 | e3g: All connected terminals will receive all "Events" that happen on the Asterisk box |
14:42.29 | DagMoller | alguem fala portugues? |
14:42.42 | RoyK | jeg snakker ikke portugisik.... |
14:42.53 | ful|work | DagMoller: diz |
14:43.38 | *** join/#asterisk oej (n=Olle@apollo.webway.se) |
14:44.47 | RoyK | telco switch monkey says "you cannot send SMS on a PRI" |
14:44.55 | RoyK | oej: god afton |
14:45.06 | oej | God afton |
14:45.10 | mosty | royk: snakk du spansk? |
14:45.39 | RoyK | nei |
14:46.04 | mosty | i just think "spansk" sounds funny, heh |
14:46.04 | jaxkz | Anyone know a reliable voip network? |
14:46.11 | *** part/#asterisk lubomier (i=lubomier@217.118.109.179) |
14:48.09 | kahuna_ | how do I turn off musiconhold. I keep getting unable to spawn mp3player and Found no files in '/usr/share/asterisk/mohmp3' |
14:48.35 | *** part/#asterisk mog_home (n=mogorman@user-24-236-84-48.knology.net) |
14:48.45 | mosty | kahuna: just put some mp3 files in that directory |
14:50.40 | iDunno | mosty: that's not how to turn it off! that's how to stop the second message ;) |
14:50.49 | gein | no one into queues, who can help me out with my configuration? |
14:51.38 | mosty | if you want to turn music on hold off completely, i think you can turn off that module in the modules.conf file |
14:51.49 | *** join/#asterisk Moc (n=mochouin@modemcable111.229-203-24.mc.videotron.ca) |
14:53.08 | iDunno | noload => res_musiconhold.so |
14:53.12 | iDunno | (at a guess!) |
14:53.24 | iDunno | (that could be wildly inaccurate and eat your cat, or something) |
14:54.42 | tzanger | ha |
14:54.59 | kahuna_ | I already ate my cat! |
14:55.47 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
14:56.14 | jake1932 | was it tasty? |
14:57.19 | *** join/#asterisk fugitivo (n=ajf@201.255.106.137) |
14:57.25 | spackle | tastes like chicken? |
14:57.26 | fugitivo | hello |
14:57.35 | gein | hej hej |
14:57.39 | Katty | hmm. |
14:57.46 | kahuna_ | it's the other, other white meat! |
14:58.29 | Ariel_ | Hope your morning is going well |
14:58.37 | Katty | yes. new vegan food products :> |
14:58.43 | Ariel_ | nice |
14:59.09 | Katty | i'm plotting something like chicken enchiladas (= |
15:00.01 | gein | morning? bah! |
15:01.04 | azzie | who knows which cisco 79xx phones support SIP and work well ? |
15:03.09 | Nugget | 7940 and 7960 both use the exact same firmware. the only functional difference is the number of lines. |
15:03.55 | Nugget | the sip firmware for the cisco phones is not as nice as the call manager firmware, but it's functional. it's also a total pain in the ass to buy. |
15:04.25 | jake1932 | azzie: but if you get the phone with SIP firmware already loaded - life is much easier |
15:04.25 | spackle | and install it would seem |
15:04.29 | *** join/#asterisk redder86 (n=lee@gateway.howardsilvan.com) |
15:04.39 | bendy24 | would probably be better with a polycom phone |
15:04.43 | redder86 | coppice: hello |
15:05.07 | bendy24 | i threw my cisco 7905g in the trash |
15:05.11 | bendy24 | piece of crap |
15:05.21 | coppice | redder86: did you see my e-mail? |
15:05.26 | Nugget | I like my 7960s, but I wouldn't recommend them. |
15:05.38 | Nugget | They're expensive and a hassle to work with |
15:05.44 | RoyK | anyone using zaphfc with pci-s? please shout out loud |
15:05.52 | bendy24 | Nugget: yep |
15:05.53 | *** join/#asterisk FaithX (n=FaithX@202-6-145-116.ip.adam.com.au) |
15:06.00 | redder86 | coppice: yes, I did. What kind of modem do you have that does not echo "ATH0E0H0" when that same command is given to it and echo is on initially? |
15:06.07 | *** part/#asterisk mosty (i=mostynm@adsl-137-244.swiftdsl.com.au) |
15:06.32 | redder86 | coppice: I've tested USR Sportster, MultiTech MT5634, Conexant K56... all echo the full command |
15:07.09 | *** join/#asterisk Chotaire (i=chotaire@chotaire.net) |
15:07.21 | Chotaire | good morning vietnam. |
15:07.33 | RoyK | anyone here that can help me sort out how an incoming SMS looks on a PRI (with PRI DEBUG)? |
15:08.44 | *** join/#asterisk wunderkin (i=kev@12-219-162-233.client.mchsi.com) |
15:08.47 | file | stabby stabby stabby |
15:09.20 | MikeJ[Laptop] | e-mail again? |
15:09.27 | Moc | morning again |
15:09.29 | MikeJ[Laptop] | or Sip? |
15:09.32 | tzanger | I'd love to send the proper MWI message to a cell phone (i.e. my asterisk voicemail lights up a Telus cellphone MWI indicator with the right callback) |
15:09.52 | MikeJ[Laptop] | tzanger, oh would you now? |
15:09.53 | MikeJ[Laptop] | :P |
15:09.58 | jake1932 | bendy24: did you really throw away your 7905? i'd buy it off you |
15:10.14 | tzanger | there are companies who say they can do it but they want like $0.17/SMS sent |
15:10.28 | jake1932 | bendy24: unless it smells like garbage now |
15:10.38 | bendy24 | jake1932: naw, just kidding, i'm frustrated as hell with it |
15:10.48 | bendy24 | i *will* get it to work |
15:10.57 | redder86 | coppice: I don't think that the command interpretation is supposed to occur until the end-of-line, yet the fact that echo is on will require that the full command be echoed. As for case sensitivity, there's no problem toupper()ing everything internally. I'll send you an updated patch. As for invalidating At and aT... well, that seems rather silly no matter what the spec says. The modems I've tested don't care about case... not sure why we sho |
15:11.04 | kahuna_ | Nothing better than a 7905 covered in ranch dressing |
15:11.11 | Beirdo | bendy24: smashy smashy? |
15:11.11 | bendy24 | how true |
15:11.16 | *** join/#asterisk }btorch{ (n=kvirc@208.63.19.172) |
15:11.17 | bendy24 | Beirdo: please go ahead |
15:11.20 | Beirdo | hehe |
15:11.23 | bendy24 | :) |
15:11.28 | Beirdo | I'll bring over a baseball bat |
15:11.29 | azzie | Nugget, why are they hussle to work with? |
15:11.43 | }btorch{ | anyone here uses the call park (feature.conf) with X-Lite ? |
15:11.48 | kahuna_ | I like how you have to "license" the firmware :( |
15:11.51 | spackle | redder86 - Old modems used to be upper-case. |
15:12.07 | Nugget | loading the firmware is a pain, obtaining the firmware is an even larger pain. |
15:12.17 | }btorch{ | I configured my extension.conf file as the instructions mention on voip-info.org but doesn't seem to work |
15:12.18 | coppice | redder86: I just tried it with a PCtel and it seems to do what you said. I remember putting in complexity specifically to make it stop echoing at the E0 to match the modems I played with at that point |
15:12.21 | redder86 | spackle: some older modems that I've tested converted everything to upper case, yes |
15:12.30 | Nugget | and the documentation is poor |
15:12.30 | azzie | Nugget, and besides loading firmware? |
15:12.31 | kahuna_ | People should put the firmware up on bittorrent sites |
15:12.33 | }btorch{ | any ideas ? |
15:12.40 | brad_mssw | Hmmhesays: around? |
15:12.43 | Beirdo | kahuna_: hardly legal :) |
15:12.51 | Katty | brad_mssw: i can call him if you want |
15:12.59 | redder86 | coppice: what modems were those that acted differently? They'd not be doing immediate echo, I don't think. |
15:12.59 | Nugget | It took me two months to find someone who would sell me the support contract to obtain the sip firmware. |
15:13.00 | kahuna_ | Yea. |
15:13.01 | Beirdo | morning, Katty |
15:13.10 | kahuna_ | Firmware nazis are hardly ethical though. |
15:13.18 | Katty | Beirdo: hi (= |
15:13.20 | coppice | redder86: the reason I specifically look for AT and at is to be compliant. not a common concept in modems, i know :-) |
15:13.29 | Nugget | don't be such an OSShole, kahuna_. |
15:13.37 | }btorch{ | I have added the include =>parkedcalls right below my [defaults] and also right below mi [macro-stdexten] |
15:13.42 | kahuna_ | lol |
15:13.44 | Nugget | it's cisco's kit, they get to set the rules. |
15:13.56 | Nugget | if you don't like it, buy a polycom. |
15:13.58 | Beirdo | even when they suck greatly :) |
15:14.08 | Hmmhesays | yeah |
15:14.09 | Hmmhesays | what up? |
15:14.12 | kahuna_ | Butt nugget - I was just voicing my opinion. |
15:14.19 | Nugget | sure, but it's a boneheaded one. :) |
15:14.20 | kahuna_ | Err, s/butt/but/ |
15:14.23 | jake1932 | maybe one day someone will write Open79XX? |
15:14.24 | Beirdo | hehehe |
15:14.35 | brad_mssw | Hmmhesays: vonage says asterisk doesn't work with their business plus service |
15:14.36 | azzie | Nugget, what would you recommend to have as a office phone ? |
15:14.39 | Beirdo | jake1932: that's a lovely idea. get me specs :) |
15:14.40 | Beirdo | hehe |
15:14.47 | brad_mssw | Hmmhesays: they won't switch my company to it |
15:14.47 | Katty | brad_mssw: then maybe it doesn't? |
15:14.49 | jake1932 | i wish |
15:14.55 | Hmmhesays | brad_mssw, they lie |
15:15.00 | Nugget | People in here seem to like the polycoms and really hate the grandstreams, but I really couldn't say. |
15:15.04 | brad_mssw | Hmmhesays: I know ... its frustrating |
15:15.08 | zobia | I got the answer wait(5) equal to 1 ring |
15:15.11 | kahuna_ | I'd rather have something IAX based anyway. Pee on Vonage :) |
15:15.11 | Nugget | I haven't used enough of them to have solid opinions on the alternatives |
15:15.15 | brad_mssw | Hmmhesays: I have some lady on the phone that doesn't even know what SIP is |
15:15.18 | azzie | Nugget, I agree that GS sucks :) |
15:15.24 | Hmmhesays | brad_mssw, give me a call and we'll hook you up |
15:15.27 | brad_mssw | Katty: Hmmhesays uses the vonage business plus |
15:15.36 | kahuna_ | I have a few grandstreams for testing. I would never give them to the users though. |
15:15.42 | azzie | Nugget, thanks a lot |
15:15.45 | brad_mssw | Hmmhesays: do you know if I can switch phone numbers yet ?? we're currently with vonage |
15:15.49 | kahuna_ | I'd spend waaay too much time supporting them |
15:15.52 | [jedi] | uhmmm |
15:15.56 | Hmmhesays | yeah they are really dragging their ass on that |
15:15.57 | brad_mssw | Hmmhesays: and we'd need to keep the same numbers |
15:16.02 | [jedi] | TDMoE is able to carry a FAX signal ? |
15:16.06 | redder86 | coppice: I guess one could say that modems that accept At and aT are broken, but my take on it is that the spec is broken. :-) |
15:16.08 | *** join/#asterisk wolfson (n=hehe@65.174.122.198) |
15:16.08 | *** join/#asterisk TripleFFF2sdf (n=TripleFF@modemcable131.156-131-66.mc.videotron.ca) |
15:16.08 | Hmmhesays | the only thing we can do right now is call forward the old numbers |
15:16.08 | [jedi] | or just voice? |
15:16.20 | Hmmhesays | they are really pissing me off in that area |
15:16.21 | TripleFFF2sdf | LOL bkw.. just saw astricon performance.. |
15:16.24 | [jedi] | (IAX2 with A-Law can't carry a FAX signal, right?) |
15:16.26 | Katty | Hmmhesays: you sure are popular of late. |
15:16.29 | kahuna_ | tdm doesn't get compressed so why not? |
15:16.39 | Hmmhesays | me? why for? |
15:16.51 | brad_mssw | Hmmhesays: well there's only one number that needs to be forwarded since it's in a call hunt |
15:16.54 | Katty | Hmmhesays: because people are all talking to you, etc. |
15:17.16 | Katty | Hmmhesays: and drunk people following you around all night. |
15:17.35 | Hmmhesays | brad_mssw yeah you'd have to keep one plan and call forward the number until portablility is completely in place |
15:17.37 | tzanger | Katty: becaue they smell tofu that smells like chicken? <ducks> |
15:17.38 | kfuq | would anyone happen to know what the MTU is on ATM ? |
15:17.38 | Hmmhesays | Katty: haha |
15:17.40 | redder86 | coppice: I think that there's an "intent" or "spirit" behind the specification, and if the actual printing of that specification doesn't adequately describe the "intent" or "spirit", well, then the spec is broken. That said, one must be able to accept behavior per the specification, however, invalidating other behaviors seems... well... pedantic. |
15:17.55 | [jedi] | kfuq: same as on IP I think... |
15:17.57 | Hmmhesays | maybe people find it odd I don't talk much tech in here |
15:18.03 | Katty | tzanger: down boy. |
15:18.06 | brad_mssw | Hmmhesays: pvt msging you .. |
15:18.14 | kahuna_ | Tech is boring. |
15:18.24 | [jedi] | kfuq: or maybe not, provided ATM is circuit-based... don't know :) |
15:18.28 | Katty | despite that fact, it's very popular, kahuna_ |
15:18.38 | kahuna_ | lol |
15:18.52 | Nugget | Katty mostly hugs in here. |
15:18.56 | kahuna_ | I'd rather talk about whats for dinner. |
15:18.59 | kfuq | [jedi]: heh.. me either.. ima n00b with atm lol |
15:19.00 | Katty | and demands recipes, Nugget |
15:19.09 | Katty | Nugget: GIVE ME YOUR RECIPES |
15:19.15 | Beirdo | hehe |
15:19.29 | Nugget | we made some amazing cheese and pepper stuffed pork tenderloins the other night. |
15:19.38 | Katty | excellent, post it. |
15:19.44 | *** join/#asterisk royth1 (n=royth1@200.121.129.178) |
15:20.35 | Katty | speaking of such, i require a recipe that uses little strips of steak. sorta like fajita steak strips. |
15:21.24 | coppice | redder86: what about my other points? |
15:21.51 | bendy24 | Katty: i prefer bacon |
15:21.54 | Hmmhesays | Katty I sang karaoke last night |
15:22.08 | Hmmhesays | it was fun |
15:22.11 | Katty | Hmmhesays: did you sing a recipe? |
15:22.13 | Katty | Hmmhesays: using steak strips? |
15:22.19 | Hmmhesays | nope |
15:22.23 | Katty | sad. |
15:22.24 | }btorch{ | is voip-info.org down or is it just me ? |
15:22.29 | Hmmhesays | I sang "she thinks my tractors sexy" and "bad touch" |
15:22.34 | Katty | ..! |
15:22.37 | Katty | Hmmhesays: what is wrong with you?! |
15:22.48 | Hmmhesays | ? |
15:22.54 | Katty | country music >.< |
15:22.55 | *** join/#asterisk toddf (n=toddf@net-66-210-104-252.theshop.net) |
15:23.05 | Katty | don't make me smack you with this polycom handset! |
15:23.19 | Hmmhesays | Katty: i gotta sing like 6 country tunes in this band |
15:23.30 | Katty | Hmmhesays: eww. |
15:23.31 | bendy24 | chattahoochi! |
15:23.31 | Katty | Hmmhesays: that's dirty. |
15:23.35 | zobia | }btorch{ i also can not access voip-info now |
15:23.57 | Hmmhesays | you'd have fun |
15:24.01 | zobia | oh. now it's okay |
15:24.01 | }btorch{ | ist back |
15:24.08 | redder86 | coppice: the echo should be in the same case that it was supplied, I think. However internally the command can be converted to uppercase. No problem. I'll send a new patch. As for At and aT, I think that the modem needs to accept them as valid. As for the OK response to AT, I think we're in agreement there. Were there other points that I missed? |
15:24.11 | Katty | Hmmhesays: fun? |
15:24.13 | zobia | yes. it's back now |
15:24.22 | Katty | Hmmhesays: somehow, that doesn't seem to be my thing. |
15:24.26 | Hmmhesays | watching me play country |
15:24.33 | Katty | well, that might be amusing. |
15:24.36 | Hmmhesays | i like to stand on tables |
15:24.44 | Katty | you would. |
15:24.53 | Hmmhesays | yes, I would |
15:24.58 | coppice | redder86: I think that is it |
15:25.34 | Katty | Hmmhesays: you should record songs and post them like i do. |
15:25.36 | anthm | I think a pluggable logger would be nice too |
15:25.44 | Katty | anthm: morning (= |
15:25.50 | [jedi] | FAX can be carried with TDMoE? |
15:25.52 | anthm | so instead of stderr you can hook up a func pointer |
15:25.59 | anthm | hi |
15:26.01 | *** join/#asterisk mutilator (n=animenod@65.111.201.79) |
15:26.09 | Katty | anthm: how is doggy? |
15:26.18 | anthm | annoying as ever |
15:26.26 | Katty | annoying as wife? |
15:26.37 | gordonjcp | Hmmhesays: you know what happens if you play country music backwards, don't you? |
15:26.49 | Hmmhesays | you get your dog back your girl back and your house back |
15:26.56 | Katty | heh, that reminds me of the JWs and back tracking or whatever. |
15:27.07 | Katty | they said it was DEMONIC |
15:27.10 | gordonjcp | ... you get out of jail, you get your truck back... |
15:27.11 | SwK[Work] | Hmmhesays: you left out sobering up and gettingyour job back |
15:27.14 | gordonjcp | yeah, you heard it |
15:27.19 | Katty | there were hidden demonic messages in these songs. |
15:27.24 | Katty | and it was Very Bad to listen to such things |
15:27.25 | anthm | he eats EVERYTHING |
15:27.26 | Katty | bad bad bad! |
15:27.32 | Katty | anthm: gosh. |
15:27.32 | Hmmhesays | i dont' play any twangy country, ain't going down by garth, she thinks my tractors sexy, the race is on by sawyer brown |
15:27.34 | gordonjcp | Katty: I played some Iron Maiden records backwards |
15:27.42 | Hmmhesays | cadillac ranch |
15:27.51 | sivana | gordonjcp: did you get a secret msg? |
15:27.56 | SwK[Work] | SATAN! |
15:27.59 | anthm | oh besides jalipinos *snicker* |
15:28.02 | gordonjcp | they said "shlub yub veeerp yaab vip" |
15:28.10 | redder86 | coppice: something also needs to be done about Caller*ID and DID information. For example, the DTE needs to be able send a command to get the DCE to repeat that information. If that information is not stored internally in spandsp, then spandsp will need to interact with the spandsp-using application much more (for nearly every AT command) to allow the application to provide that kind of information. |
15:28.23 | gordonjcp | then the stylus broke and I had to go shopping for a new one |
15:28.35 | Nugget | Katty: http://www.marga.org/food/blog/archives/001738.html |
15:28.46 | Nugget | I suggest using two or three times the amount of cheese, though |
15:29.14 | coppice | redder86: I think I handled caller ID in my channel driver. take a look at that |
15:29.31 | gordonjcp | Nugget: that may not really be to Katty's taste |
15:30.32 | redder86 | coppice: yeah, I'm handling it in IAXmodem itself, too. But getting it to *repeat* that information by way of an AT-command is not possible without either putting that information into spandsp and without the application being able to participate in the AT command-response stuff. |
15:31.16 | FABRIZIOxxx | is it normal that asterisk does not answer pstn after a couple of hours that its working but only sip to sip works?? |
15:31.27 | redder86 | coppice: so my question is whether you would prefer to see this stuff internalized into the library or whether you would prefer to see interaction between spandsp and the application on every AT command? |
15:31.38 | Katty | gordonjcp: i told him to post it. |
15:32.30 | *** join/#asterisk convey (n=test@66.55.43.2) |
15:32.34 | gordonjcp | Katty: ah, nm then |
15:33.16 | coppice | redder86: I can't remember what I did. I'll take a look and get back to you. which AT commands do you expect to work for this? Most modems don't seem to support the standards |
15:35.04 | file | why won't you build statically |
15:35.28 | redder86 | coppice: okay, thanks. Yeah, every modem seems to have it's own set of additional commands... but the reason that is - is because the standard set of commands is inadequate. For example, I'd like a command such as AT+VRID=n where it would enable and disable the displaying of Caller*ID and DID information and would also (depending on n) cause the DCE to repeat the received Caller*ID and DID for that call. |
15:35.29 | bendy24 | file: i'd blame your gcc inculdes |
15:35.40 | file | kurg blah |
15:36.33 | *** join/#asterisk makhtar (n=ageller@mail.bulletinnews.com) |
15:38.40 | coppice | redder86: I think there is a caller ID command, but its somewhere obscure. you may have noticed I went through all the ITU and 3GPP specs, and put all the commands in the interpreter :-) |
15:39.04 | kahuna_ | how can I disconnect from the asterisk console once I connect via asterisk -r |
15:39.05 | redder86 | coppice: yeah, I saw a lot of commands there. I haven't gone through them all. |
15:39.33 | redder86 | coppice: I'll have another look and if I find something I'll work with that and send a patch. |
15:40.08 | *** join/#asterisk pashah (n=pashah@ns.itconnection.ru) |
15:40.16 | pashah | hi all |
15:40.20 | Katty | Nugget: margarita's stuff is all posh |
15:40.32 | *** join/#asterisk Firestorm-voip (n=Firestor@mail.mysoft.se) |
15:42.07 | pashah | has anyone an example of zaptel.conf to use te110p and tdm with 4 FXO in one * box, |
15:42.22 | hardwire | pashah: not that hard |
15:42.47 | hardwire | define the t1 span |
15:42.48 | coppice | redder86: there is some truly horrible buffering in t31.c which defintely needs improving before it can be called complete |
15:42.52 | hardwire | then define the fxo span |
15:43.31 | redder86 | coppice: oh, I've been meaning to talk to you about buffers. Can you elaborate? |
15:43.31 | wunderkin | and make sure the t1 is loaded first |
15:43.34 | }btorch{ | do I need to create a parkedbcalls extenstion on extention.conf ? |
15:43.43 | hardwire | yup |
15:43.45 | }btorch{ | to get thing to work ? |
15:43.50 | pashah | thanks |
15:44.08 | coppice | redder86: It buffers a whole page, which was a very temporary measure |
15:44.08 | hardwire | }btorch{: sorry.. you need to include => parkedcalls |
15:44.14 | hardwire | in whatever context the phone you are using is in |
15:44.20 | redder86 | coppice: ah, that may explain some things |
15:44.35 | }btorch{ | hardwire: noprobs ... I have but that still didn't work |
15:44.49 | hardwire | make sure its enabled in features.conf |
15:44.53 | redder86 | coppice: can the modems support getting little bits of data at a time or do they need the full page? |
15:45.15 | }btorch{ | where should I include it though ? right below [general] on extension.conf ? |
15:45.23 | *** join/#asterisk [ViRii] (n=virii@68-186-170-246.dhcp.smrt.tn.charter.com) |
15:45.28 | hardwire | Katty: did you get your steak information? |
15:45.28 | royth1 | hello people |
15:45.30 | redder86 | coppice: I implemented flow-control in IAXmodem (xon/xoff)... it seems to send immediately. |
15:45.30 | }btorch{ | hardwire: it's enabl;ed |
15:45.40 | *** join/#asterisk syllogism (n=syllogis@adsl-69-152-41-249.dsl.ltrkar.swbell.net) |
15:45.44 | hardwire | dial 700 from your phone |
15:45.48 | royth1 | someone uses asthome ? |
15:45.58 | redder86 | coppice: but I need a way to "look in" and see how much "buffer" is left. |
15:45.59 | *** join/#asterisk mithro (n=tim@c213-100-42-188.swipnet.se) |
15:46.08 | [ViRii] | hey the intercom function for asterisk, how do i do this? i imagine that an overhead speaker is registered as an extension? |
15:46.17 | Katty | hardwire: no |
15:46.28 | hardwire | Katty: you just have too much steak ? |
15:46.39 | redder86 | coppice: right now I do it by way of calculation ... using the DCE-DCE bitrate and the time elapsed to know when to feed more data to the DSP. |
15:46.52 | redder86 | coppice: but that can be grossly inaccurate. |
15:47.07 | *** part/#asterisk syllogism (n=syllogis@adsl-69-152-41-249.dsl.ltrkar.swbell.net) |
15:47.31 | pashah | laters |
15:47.36 | Katty | hardwire: i'm a vegan. (= |
15:47.45 | hardwire | Katty: you require odd things then. |
15:47.49 | }btorch{ | hardwire: it works when I dial 700 and it gives me 701 as the parked extension |
15:47.51 | Katty | hardwire: and, thusly, i'm adapting reicpes. |
15:47.59 | hardwire | Katty: ah.. use TVP strips |
15:48.04 | R3DB0x | what is better.....the polycom ip501 or the cisco 7970G |
15:48.09 | Katty | hardwire: morningstar farms has just come out with a 'steak' strip simulant. |
15:48.15 | hardwire | so TVP |
15:48.22 | hardwire | but it won't soak.. |
15:48.23 | hardwire | hmm |
15:48.25 | Katty | it has more flavoring than tvp |
15:48.27 | hardwire | and you are a vegan |
15:48.29 | hardwire | so no sour cream? |
15:48.31 | Katty | tvp tastes like crap |
15:48.34 | Katty | tofutti |
15:48.45 | Katty | tofutti makes vegan sour cream and cream cheese. |
15:48.48 | coppice | usually people just pump data into the modem, and let flow control take care of things. with non-blocking reads you have to keep accepting stuff. I should probably dynamic accept and ignore the device in the select calls, and control things that way |
15:48.51 | hardwire | eww |
15:48.51 | arp2 | tvp plain certainly does |
15:49.00 | royth1 | hello |
15:49.01 | }btorch{ | hardwire: but If I call another phone and pick the call on that other phone and try to park the call with #700 it doesn't work |
15:49.04 | hardwire | Katty: can I try to remind you that Soy is a muscle development inhibitor? |
15:49.05 | arp2 | tvp is great in veg chili |
15:49.07 | hardwire | but none the less |
15:49.12 | hardwire | soudns like you could go make some stroganoff |
15:49.18 | hardwire | with some nice wide noodles |
15:49.19 | arp2 | or veg tacos or whatnot |
15:49.27 | Katty | hardwire: yep, that's what i'm planning |
15:49.29 | coppice | eh? soy is packed with protein |
15:49.30 | }btorch{ | hmm I like stroganoff |
15:49.47 | royth1 | who knows the asthome |
15:49.51 | royth1 | ? |
15:49.54 | kahuna_ | I like doing that too |
15:50.00 | Katty | coppice: everyone likes to bitch aobut something. |
15:50.00 | hardwire | coppice: the soy protein blocks muscle development ;) |
15:50.06 | Katty | coppice: quite frankly, i'm tired of all the bitching :) |
15:50.08 | hardwire | I do like to bitch |
15:50.17 | FABRIZIOxxx | if i use the tdm04b 4 fxo modules do i need to plug the power supply into the cards? |
15:50.18 | arp2 | soy is also packed with chemicals that could potentiall interfere with one's hormones (for some people) |
15:50.20 | gordonjcp | hardwire: does it? |
15:50.24 | arp2 | (I'm one of those people, so I avoid soy now) |
15:50.28 | *** part/#asterisk TripleFFF2sdf (n=TripleFF@modemcable131.156-131-66.mc.videotron.ca) |
15:50.29 | hardwire | this all came about for me after a partner of mine did a study on the effects of a soy oil spill in the aleutians |
15:50.33 | *** join/#asterisk mydssmojo (n=smi23le@d209-89-197-205.abhsia.telus.net) |
15:50.33 | kahuna_ | I think the PS is for FXS |
15:50.36 | redder86 | coppice: I actually prefer the flow-control to come from the IAXmodem-side of things. But I do need to be able to peer-in to the buffer. |
15:50.51 | Katty | hardwire: and by the way, you better stop breathing too. |
15:50.54 | hardwire | to find out what effect it would have on the wildlife and kept running acrossed the results that we will basically just have smaller halibut and sea life in that area for a few years |
15:51.01 | Katty | hardwire: cause all that stuff is you breathe is positively disgusting. |
15:51.01 | kahuna_ | Oh, stroganoff is a food? |
15:51.13 | *** part/#asterisk mydssmojo (n=smi23le@d209-89-197-205.abhsia.telus.net) |
15:51.17 | arp2 | stroganoff is a recipe |
15:51.18 | redder86 | coppice: do you know right-off what state variable I'm looking for that tells me how big the buffer is at that moment? |
15:51.23 | spackle | chan_stroganoff |
15:51.26 | kahuna_ | lol |
15:51.30 | hardwire | Katty: just pointing out something I found out. |
15:51.32 | *** join/#asterisk jeffik (n=Jeff@CPE0020ed8494b8-CM0012c999ca4e.cpe.net.cable.rogers.com) |
15:51.35 | arp2 | usually a rich cream-based sauce |
15:51.35 | hardwire | not insulting how you do things |
15:51.38 | kahuna_ | I thought it was a reference to masturbation. |
15:51.51 | redder86 | coppice: i.e. are they modem state variables or T.31 state variable? |
15:52.13 | arp2 | <hardwire> Katty: can I try to remind you that Soy is a muscle development |
15:52.13 | arp2 | <PROTECTED> |
15:52.14 | coppice | the relevant bit is |
15:52.16 | coppice | <PROTECTED> |
15:52.18 | coppice | <PROTECTED> |
15:52.19 | arp2 | ... in certain people |
15:52.20 | coppice | <PROTECTED> |
15:52.20 | arp2 | not all |
15:52.28 | kahuna_ | that's why all vegetarians are wimps! |
15:52.37 | Katty | kahuna_: i'm no wimp. |
15:52.39 | redder86 | coppice: thank you |
15:52.44 | Katty | kahuna_: i'd beat you any day. |
15:52.51 | arp2 | I have a thyroid condition, so I have to stay away from soy |
15:52.54 | kahuna_ | I bet you would ;) |
15:53.14 | kahuna_ | But I was only joking, as I was a vegan for 3 years myself. |
15:53.16 | hardwire | arp2: the same with my g/f |
15:53.32 | redder86 | coppice: is it technically possible to transmit (say, V.17) and listen simultaneously throughout the entire transmission and pick up any signals that the other end may be transmitting? |
15:53.48 | sivana | lol |
15:53.49 | redder86 | coppice: or is the communication only half-duplex? |
15:53.50 | sivana | too funny |
15:53.56 | Katty | anyone have a stroganoff recipe? |
15:54.08 | arp2 | www.google.com I'm sure |
15:54.09 | tzanger | not without beef |
15:54.15 | kahuna_ | Soy wasn't my thing though. I got most of my protein via legumes |
15:54.17 | Katty | tzanger: just post the recipe |
15:54.18 | *** join/#asterisk mydssmojo (n=smi23le@d209-89-197-205.abhsia.telus.net) |
15:54.20 | gordonjcp | use something else insted of beef |
15:54.24 | gordonjcp | mushrooms maybe |
15:54.28 | Katty | blah |
15:54.35 | Katty | tzanger: i'll redo it how i want it. |
15:54.49 | coppice | V.17 is one half of V.32. V.32 is full duplex through echo cancellation. V.17 is not |
15:55.02 | kahuna_ | What about chicken stroganoff? Would that be doable? I'm not into beef all that much either. |
15:55.08 | Katty | kahuna_: sure, post that. |
15:55.09 | arp2 | yes |
15:55.20 | arp2 | its just meat in sauce |
15:55.33 | kahuna_ | I don't have the algo for that. I was wondering if it would tase good. |
15:55.36 | arp2 | I'm sure theres a chicken stroganoff recipe on the internets |
15:56.05 | mydssmojo | Can anyone recommand any good GUI for Asterisk, I know of AMP; but any others? |
15:56.50 | royth1 | mydssmojo, i install asthome |
15:56.59 | *** join/#asterisk ManxPower (n=eric@stirprop-s4-0-0-21.ndcr2.datasync.net) |
15:57.03 | [ViRii] | i have a fax machine, how can i send a fax through asterisk? |
15:57.36 | redder86 | ViRii: plug it in to a T1 channel bank |
15:57.42 | spackle | ViRii: using an ATA and a hope and a a prayer |
15:57.51 | [ViRii] | lol |
15:57.54 | [ViRii] | thanks |
15:58.02 | spackle | make sure G711 |
15:58.37 | mydssmojo | royth1 > I will be using it more on a commercial bases that's why I don't want to use ast@home |
15:58.38 | royth1 | but i can not raise my graphical environment |
15:58.46 | spackle | Redder and Coppice are discussing fax implementation right now. |
16:00.07 | ronaldl79 | G'Day, room. |
16:00.33 | royth1 | mydssmojo, of that it forms i raise the web |
16:00.54 | *** part/#asterisk spackle (n=spackle@209.234.83.19) |
16:01.29 | [ViRii] | ok fax to email requires email accnts to be setup on the asterisk server? |
16:01.45 | redder86 | spackle: yes, but we're talking about a softmodem... in ViRii's case the DSP is in the faxmodem, so the data is already modulated audio... in which case we're not really discussing a solution for that. The "ideal" solution would be for him to use a channel bank or a T.38 ATA (however, Asterisk does not support T.38 at the moment, so channel bank is the only choice that will be reliable). |
16:03.09 | coppice | Ah. T.38. I suppose you'll want a T.31 interface to that next :-) |
16:03.15 | olivier_ | i use FAX-->PAP2--SIP(g711)-->*---ISDN Provider work well in a local network |
16:03.31 | mydssmojo | anybody using Sangoma cards? |
16:03.50 | [ViRii] | ouch |
16:03.56 | coppice | olivier: it depends on luck and the particular ATA you use |
16:04.12 | redder86 | coppice: we already have a T.31 interface to T.38 in OpenH323's t38modem |
16:04.21 | *** join/#asterisk bsd3 (n=bsd@203.134.194.176) |
16:04.35 | coppice | but that doesn't integrate with * |
16:04.36 | royth1 | mydssmojo, help me to enter the environment AMP, please |
16:04.39 | redder86 | coppice: can't use it with Asterisk though because of the lack of T.38 support and because the H.323 support is poor |
16:05.06 | bsd3 | hi, friends! |
16:05.08 | }btorch{ | man now I have no voice going on when I make a call... I can record calls to voicemail |
16:05.17 | coppice | I ain't even trying to get T.38 with H.323 working. H.323 is obsolete :-) |
16:05.18 | bkw_ | who wants to receive faxes inside of atserisk? |
16:05.18 | redder86 | coppice: in my case I really wanted access to the actual DSPs. |
16:05.21 | bkw_ | thats just not right |
16:05.30 | }btorch{ | has anyone had this problem .... not been able to hear any sound |
16:05.31 | bkw_ | t38modem is a nice way to pipe that stuff into hylafax |
16:05.35 | mydssmojo | royth1 > sorry i don't understand, you want to know where you can download AMP? |
16:05.38 | royth1 | now i am in the way commands |
16:05.53 | coppice | t38modem looks really clunky |
16:06.01 | bkw_ | not really |
16:06.07 | sivana | bkw_: most places what to work with an actual fax machine harware |
16:06.12 | redder86 | bkw: yeah, but a T.38 gateway still requires that a large portion of the fax protocol stuff be on the gateway |
16:06.13 | sivana | s/what/want |
16:06.15 | *** join/#asterisk Juggie (i=agony@CPE00c049d9f271-CM00137186c8d8.cpe.net.cable.rogers.com) |
16:06.19 | bkw_ | sivana, then they need to get with the program |
16:06.35 | file[laptop] | fax fight! |
16:06.38 | sivana | bkw_: what's the program? :) |
16:06.42 | redder86 | so with T.38 you've got fax protocol happening on the sender, the gateway, and the receiver... not just the sender and receiver. |
16:06.46 | coppice | redder86: yep. that's what I am working on |
16:06.57 | bsd3 | i have question on refresh time for IAX and SIP |
16:07.05 | redder86 | coppice: yep, I know. :-) |
16:07.25 | sivana | bkw_: with hylafax, how do you take a piece of paper and fax it? |
16:07.38 | bkw_ | I have had better luck faxing over IP using IAX one on one than I have over Zap at the moment |
16:07.40 | royth1 | mydssmojo, a already have it installed in ,my laptop but not know like entering to the way AMP |
16:07.41 | redder86 | sivana: scanner? |
16:07.51 | sivana | redder86: not practical |
16:08.06 | bkw_ | yes it is |
16:08.17 | redder86 | sivana: you can also hook up a fax machine to send it's faxes through HylaFAX. |
16:08.19 | bkw_ | you just get one of those ones that will take a stack of paper and scan it |
16:08.30 | sivana | ya, try to explain that process to a pencil pusher |
16:08.37 | Katty | hmm. |
16:08.39 | bkw_ | its really a one button process |
16:08.42 | Katty | my office is going through a vegan recipe fad. |
16:08.42 | redder86 | sivana: many new fax machines support a verison (or a corruption) of T.37. You can get that into HylaFAX. |
16:09.02 | bkw_ | redder86, you're right now which one does that? |
16:09.02 | Hmmhesays | beverly hills, thats where I want to be |
16:09.03 | file[laptop] | yay, corruption |
16:09.14 | redder86 | sivana: but if you want to be really expensive you can plug the fax machine into a channel bank, send the fax to the HylaFAX server and have HylaFAX relay it. |
16:09.15 | *** join/#asterisk spackle (n=spackle@209.234.83.19) |
16:09.17 | coppice | redder86: how many fax machines support t.38? |
16:09.21 | redder86 | bkw_: the Panafax machines do that |
16:09.34 | redder86 | coppice: I don't know of any that support T.38 at the moment... just T.37. |
16:09.36 | sivana | interesting |
16:09.46 | bkw_ | so see their are ways to get around this :P |
16:09.50 | sivana | I have a channel bank, I have a spare server for hylaFax |
16:09.51 | [ViRii] | sms messaging = instant messaging? |
16:09.58 | bkw_ | [ViRii], in a sick world sure |
16:10.00 | coppice | redder86: really? that was the whole point of the TCP mode in T.38 |
16:10.18 | Hmmhesays | hey file |
16:10.26 | [ViRii] | how do i get ip to resolv to a name like Kenny |
16:10.27 | sivana | OR we can fix * to actually handle timing properly to support faxes |
16:10.30 | coppice | maybe the fax makers actually caught on to real time being dumb :-) |
16:10.33 | kahuna_ | wht do u mean bk? sms rulz! |
16:10.34 | [ViRii] | <--- entire newb. |
16:10.35 | blitzrage | does the 7970G have a SIP firmware image yet? |
16:10.39 | bkw_ | sivana, you want a lot |
16:10.41 | *** join/#asterisk Sedorox (n=Brandon@smartserv/cna/Sedorox) |
16:10.41 | redder86 | coppice: Yeah, most fax machines I've seen that have an ethernet jack in them only support a fax-to-email facility... sometimes T.37, sometimes a corruption of it. |
16:10.42 | sivana | hehe |
16:10.43 | blitzrage | morning all btw :) |
16:10.44 | sivana | I guess I do |
16:10.59 | mydssmojo | royth1 > sorry i don't understand your english, but i think your asking how to get into AMP? http://<your amp ip address>/admin |
16:11.16 | bkw_ | still fax to email is something you can warp into hylafax :P |
16:11.17 | file[laptop] | hi blitzrage |
16:11.32 | redder86 | coppice: I worry about reliablity in a sender -> gateway, gateway -> receiver scenario. That's two sets of fax protocol happening instead of just one. Double the trouble for doulble the fun. |
16:11.36 | sivana | bkw_: I'm going to play with hylafax.. if I can plug it into a channel bank |
16:11.40 | coppice | sivana: the only timing problem in * affecting fax is the TDM400P card problem |
16:11.57 | sivana | coppice: nope, I can't get past 8 pages on 405P |
16:11.58 | coppice | redder86: well that is what T.38 is all about |
16:11.59 | bkw_ | coppice, well thats not totally true either |
16:12.13 | coppice | sivana: then fix your installation |
16:12.14 | redder86 | coppice: end-to-end T.38 would be wonderful. |
16:12.52 | coppice | redder86: I still think real time is dumb. e-mail queues. it retries. it does all sorts of nice things |
16:12.53 | sivana | coppice: how much you charge per hour? :) |
16:13.11 | blitzrage | file[laptop]: ahoi hoi |
16:13.48 | redder86 | coppice: HylaFAX queues ;-) |
16:13.51 | coppice | sivana: I have seem literally hundreds of people complaining about timing when it is their fault. I have a mass of e-mailed audio files from spandsp with frame slips in them |
16:14.24 | coppice | redder86: yeah, kind of, but multiple concurrent deliveries beats that hands down |
16:14.38 | sivana | coppice: maybe so, but 8 pages in? |
16:14.59 | kahuna_ | that is the TDM400P problem (curious) |
16:15.13 | kahuna_ | err, s/that/what |
16:15.18 | coppice | sivana: whatever. some people want to fix their problems. others just want to moan |
16:15.21 | redder86 | coppice: with multiple lines out and multiple lines in on the other end you can do that with HylaFAX, too. The only difference is the communication medium... really. |
16:15.41 | sivana | coppice: just be easy money for you then |
16:15.59 | royth1 | mydssmojo, i understand, i was thinking that of the same pc, i can enter to environmente AMP, but it is done from another pc |
16:16.04 | coppice | "multiple lines" is a concept more for the 20th century than the 21st |
16:16.11 | *** join/#asterisk file[laptop] (n=jcolp@mctnnbsa31w-142166116178.nb.aliant.net) |
16:16.20 | redder86 | coppice: multiple channels? |
16:16.47 | redder86 | coppice: multiple ports? |
16:16.54 | *** join/#asterisk shmaltz (n=chatzill@69.28.255.210) |
16:16.58 | redder86 | choose the word that makes you feel most astute |
16:17.01 | coppice | e-mail is really good, you know. its *very* hard to beat as a reliable delivery mechanism |
16:17.28 | shmaltz | can I use an analog ADSI phone (like the Aastra PT-480e) with an Adit 600 and get all the nice features? |
16:17.34 | bkw_ | email isn't something you can rely on if you don't provide the services |
16:17.42 | kahuna_ | I agree too. Isn't email supposed to replace fax in most instances? |
16:17.46 | bkw_ | for example AOL blocks all our faxes we send to customers with AOL accounts |
16:17.47 | royth1 | mydssmojo, thanks |
16:17.56 | bkw_ | they end up in the spam folder for no good reason |
16:18.10 | kahuna_ | AOL sucks :( |
16:18.29 | bkw_ | <PROTECTED> |
16:18.29 | bkw_ | <PROTECTED> |
16:18.29 | bkw_ | <PROTECTED> |
16:18.32 | redder86 | hehe... well, we've had this debate. I don't have any problem with e-mail. The difference between e-mailing an image file and faxing an image file, though, is that the data format type is negotiated in the fax protocol while it is not negotiated in e-mail. Meaning I can send you a JBIG image to you in an e-mail attachment, and you won't be able to view it. But with fax if I try to send you a JBIG image file and your fax machine doesn't suppor |
16:18.34 | bkw_ | haha |
16:18.44 | kahuna_ | I get the same thing when I email my customers. AOL also tends to mangle the messages. |
16:19.05 | *** join/#asterisk \Grooby\ (n=Grooby@66.160.105.186) |
16:19.12 | spackle | stop using HTML email |
16:19.13 | kahuna_ | redder86: looks like your fax got cut off :) |
16:19.18 | spackle | <grin> |
16:19.29 | shmaltz | can I use an analog ADSI phone (like the Aastra PT-480e) with an Adit 600 and get all the nice features? |
16:19.35 | kahuna_ | spackle: I send in both HTML and plantext. AOL refuses to display them both at times. |
16:19.51 | spackle | you said it best - AOL sucks |
16:20.01 | [ViRii] | just a question, where did you all learn so much about asterisk =P |
16:20.10 | coppice | bkw_: lots of things start with "e-mail has flaws" and end with a huge failed attempt to do better :-) |
16:20.12 | blitzrage | [ViRii]: using it for 3 years will do it |
16:20.15 | kahuna_ | [ViRii]: My mom taught me. |
16:20.20 | [ViRii] | lol |
16:20.38 | \Grooby\ | hey guys.....I have 3 sip phones in Taiwan that register to my * here in DC and every night those 3 phones looses registration |
16:20.39 | blitzrage | [ViRii]: sitting in a dark room for 16 hours a day for 4 months talking to bkw_ taught me most of what I know :) |
16:20.47 | file[laptop] | blitzrage: LOL |
16:20.48 | kahuna_ | Really I just talked my boss into buying a quad span T1 card and some voip phones. |
16:20.52 | kahuna_ | In the name of R+D |
16:20.53 | \Grooby\ | as of now, i can see the phone register and then lose registration in about 3 minutes |
16:20.59 | spackle | My first job was programming binary load lifters, very similar to our moisture vaporators |
16:21.16 | [ViRii] | blitzrage: thats probably what ill have to do :P |
16:21.18 | \Grooby\ | where can I start debugging this? I don't think it's network problem cause my brother's there and he's online and I am chatting w/ him fine |
16:21.31 | [ViRii] | ive recently been employed to setup an office and i dont know jack about it |
16:21.32 | [ViRii] | :> |
16:21.39 | anthm | hey this r2 unit has a bad motivator! |
16:21.40 | *** join/#asterisk Moc (n=mochouin@modemcable111.229-203-24.mc.videotron.ca) |
16:21.52 | kahuna_ | \Grooby\: But it still may be a network problem. Get out ethereal or something. |
16:21.53 | spackle | schmaltz, it used to be there were Aastra phones sold by Digium to work with ADSI |
16:22.00 | shmaltz | ViRii, in the john |
16:22.28 | spackle | anthm: great moments in movie whining "But I was going in to Toshi station to pick up some power converters" |
16:22.31 | shmaltz | ViRii, neither did I a year ago |
16:22.37 | *** join/#asterisk argos73 (n=mike@adsl-70-228-109-5.dsl.akrnoh.ameritech.net) |
16:22.53 | redder86 | coppice: just sent you an updated patch |
16:22.54 | kahuna_ | I spent way to much time shooting womprats in beggars canyon |
16:23.22 | \Grooby\ | hmm ok |
16:23.22 | sigterm | lol |
16:23.23 | anthm | reviews say anakin was a whiner but if so his son was a chip off the ol' block => |
16:23.36 | [ViRii] | what kinda phones do you use shmaltz? |
16:23.37 | *** part/#asterisk Firestorm-voip (n=Firestor@mail.mysoft.se) |
16:23.40 | *** join/#asterisk thuper (n=thuper@gateway.digium.com) |
16:23.59 | ronaldl79 | pbx.c:1294 pbx_extension_helper: No application 'Playback ' for extension (default, *90, 4) -- Why is * prompting this when attempting to playback a recording? |
16:24.10 | shmaltz | ViRii, depends on the job |
16:24.24 | cpatry | cause u dont have app_playback.so loaded? |
16:24.28 | cpatry | show modules like playback |
16:24.35 | [ViRii] | polycoms? |
16:25.07 | file[laptop] | bah, bah hum bug! |
16:25.33 | ronaldl79 | Okay, I do not see app_playback.so in the list -- shouldn't this be a given? |
16:25.36 | kahuna_ | Anything but Grandstream |
16:25.51 | spackle | schmaltz, you should call voipsupply and ask them about ADSI, asterisk, and channel banks. |
16:25.59 | cpatry | ronaldl79: load app_playback.so |
16:26.23 | *** join/#asterisk Firestorm-voip (n=Firestor@ua-83-227-140-131.cust.bredbandsbolaget.se) |
16:26.37 | *** join/#asterisk Druken (n=druken@CPE00121716da99-CM000e5cde4ca2.cpe.net.cable.rogers.com) |
16:26.47 | ronaldl79 | Okay, cpatry, I just tried that and asterisk says it already exists. |
16:26.55 | Druken | anyone here have experince with wan networking? |
16:27.01 | cpatry | u just said it was not there. |
16:27.05 | file[laptop] | ronaldl79: you have a space... in the name of the application |
16:27.13 | file[laptop] | at the end of it |
16:27.35 | ronaldl79 | doh |
16:27.45 | [ViRii] | anyone using polycom phones with asterisk? |
16:27.48 | jake1932 | good catch file |
16:27.54 | jontow | virii; lots of people |
16:28.05 | [ViRii] | jontow: do you recomend that? |
16:28.07 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
16:28.07 | spackle | virii, they are sweet phones |
16:28.12 | *** join/#asterisk jtodd (n=jtodd@dsl027-191-178.sfo1.dsl.speakeasy.net) |
16:28.13 | kahuna_ | I can't think of a reason why ASDI would not work with a channel bank. |
16:28.34 | kahuna_ | It's all sent in-band |
16:28.51 | ronaldl79 | * can be picky, it seems! |
16:29.05 | *** part/#asterisk bsd3 (n=bsd@203.134.194.176) |
16:29.44 | ronaldl79 | Thanks for that catch, file ... that did the trick... |
16:29.58 | ronaldl79 | Would it be safe to say not to include ANY spaces in the dialplan? :) |
16:30.02 | ronaldl79 | I normally don't... |
16:30.37 | jake1932 | ronaldl79: there are places you need spaces |
16:31.03 | ronaldl79 | arrrggg |
16:31.45 | *** join/#asterisk TripleFFF2sdf (n=TripleFF@modemcable131.156-131-66.mc.videotron.ca) |
16:31.50 | TripleFFF2sdf | !seen damin |
16:31.58 | anthm | use parens and you wont have that issue |
16:32.06 | TripleFFF2sdf | http://www.astertest.com/forum/viewtopic.php?t=8 |
16:32.08 | anthm | Playback(woohoo) |
16:32.11 | TripleFFF2sdf | anyone have that file |
16:32.17 | TripleFFF2sdf | ttp://astricon.asterisk.pl/2004-09-recordings/astricon_developer_conference_breakout_failover_09-24-2004.wav |
16:33.11 | jake1932 | ronaldl79: read this (if you haven't already regarding spaces): http://www.voip-info.org/wiki-Asterisk+Expressions |
16:33.25 | drumkilla | that's a totally different issue |
16:33.31 | drumkilla | and, isn't relevant anymore :) |
16:33.46 | Abbas | any one used asterisk-oh323? |
16:33.53 | *** join/#asterisk svenna (n=svenna@p548D37C2.dip0.t-ipconnect.de) |
16:33.56 | ronaldl79 | I haven't, jake -- I'm discovering new things about * everyday. Thanks for the URL. |
16:34.13 | jake1932 | np |
16:34.35 | file | ugh I'm hungry |
16:34.49 | spackle | Hi hungry, I'm spackle |
16:35.03 | file | nice to meet you! |
16:35.16 | cpatry | nice to EAT you! |
16:35.27 | *** join/#asterisk myiagy (n=myiagy@200.138.215.78) |
16:35.27 | file | noooooo |
16:35.55 | cpatry | dont be scared boi! |
16:37.57 | cpatry | with chan_fax? |
16:38.46 | sivana | heh |
16:39.34 | *** join/#asterisk simbulu_ (n=chatzill@80.77.133.83) |
16:40.07 | tzanger | ugh |
16:40.20 | tzanger | F = (a*b)/(a+b) ... is there any way to figure out b? |
16:40.27 | tzanger | I can't get b alone on this |
16:41.16 | bendy24 | b=a |
16:41.53 | cpatry | =a(a*b)+b(a*b) |
16:43.56 | simbulu_ | does anyone know how to use the redhat/asterisk.spec with rpmbuild ? |
16:44.53 | drumkilla | tzanger: well, my calculator gave me b as a function of F |
16:44.57 | drumkilla | if you want that :) |
16:45.05 | tzanger | drumkilla: sure, but does it look pretty? |
16:45.57 | drumkilla | b = (-6 * F) / ( sqrt(3) * F - 6 ) |
16:46.04 | tzanger | what the fuck |
16:46.17 | tzanger | where did -6 and sqrt(3) get into there |
16:46.32 | Juggie | hah |
16:46.33 | Juggie | what a joke |
16:46.50 | Juggie | i call up digium to tell them i want to do a quote we got a while ago |
16:46.57 | Juggie | and they say "oh, our rates have doubled" |
16:47.06 | tzanger | how long is "a while ago" ? |
16:47.18 | ManxPower | Juggie, Did the quote specify an expiry date? |
16:47.23 | Juggie | yah, it is expired |
16:47.34 | Juggie | because in typical goverment fassion my boss dragged his ass for months |
16:47.45 | Juggie | none the less, i find it funny they decided to double their rates |
16:47.50 | jake1932 | i got b = F(a+b)/b |
16:47.59 | ManxPower | Another example of why I won't work for a govt agency. |
16:48.24 | Juggie | i think they are still going to get it done |
16:48.24 | jake1932 | can you take it more than that? |
16:48.43 | Juggie | but i dont understand why digium would double the rates |
16:48.53 | coppice | redder86: I implemented a solution for your 5 flags requirement that should be generic. I'll pass it over when I have tidied up some other stuff. Do you intend to implement "A/"? |
16:49.13 | spackle | to make more money? They can charge what the market will bear. |
16:49.33 | TripleFFF2sdf | [12:47] jake1932: i got b = F(a+b)/b |
16:49.43 | tzanger | jake1932: yeah that's what I ended up with but it doesn't help since b's on both sides :-) |
16:49.43 | TripleFFF2sdf | i got e=mc2 |
16:49.46 | *** join/#asterisk santiago (n=santiago@63.245.86.245) |
16:49.47 | TripleFFF2sdf | and e=ri |
16:50.40 | spackle | and F=M*V |
16:51.14 | spackle | D=R*T |
16:52.16 | redder86 | coppice: I hadn't gotten around to "A/" yet... but I was planning on getting to everything. |
16:52.48 | *** part/#asterisk redder86 (n=lee@gateway.howardsilvan.com) |
16:53.29 | *** join/#asterisk kahuna_ (n=booger@209-254-56-194.ip.mcleodusa.net) |
16:54.43 | sivana | tzanger: b = a / ( a/F - 1) |
16:56.19 | kahuna_ | I'm getting lots of messages like these for my channel bank channels: zt hook failed: Device or resource busy |
16:56.23 | kahuna_ | How can I fix it? |
16:56.34 | *** join/#asterisk BartAleph (i=ircap751@pc-64-208-120-200.cm.vtr.net) |
16:56.49 | BartAleph | hello |
16:57.53 | *** join/#asterisk stkn (n=stkn@gentoo/developer/pdpc.active.stkn) |
16:57.54 | coppice | redder86: I gave the hdlc_rx_init() call a new parameter to set the number of flags before "OK" is declared |
16:58.25 | kahuna_ | would flash and rxflash timings possible solve that problem? |
16:58.33 | MikeJ[Laptop] | I geb b=a/((a/F)-1)) |
16:58.37 | MikeJ[Laptop] | do I win a prize |
16:58.47 | MikeJ[Laptop] | damn... sivana |
16:58.48 | MikeJ[Laptop] | :( |
16:59.04 | MikeJ[Laptop] | at least we both got the same answer... |
16:59.05 | MikeJ[Laptop] | heh |
16:59.27 | BartAleph | i´m lookin for information about use and configure of asterisk like a call center...anybody could help me? (please, excuse my poor english) |
16:59.58 | kahuna_ | BartAleph: outbound dialling? |
17:01.24 | BartAleph | kahuna: just like a call center, you know, with options to the caller, voice-mail, forwarding calls, etc |
17:02.22 | BartAleph | kahuna: could you say me where i can find information? |
17:02.45 | kahuna_ | Well you can start with asteriskdocs.org and read the stuff there, |
17:02.56 | kahuna_ | That seems like a really basic asterisk config. |
17:02.57 | BartAleph | internet address, forums, anything |
17:03.20 | BartAleph | ok..i´m taking notes |
17:03.50 | *** join/#asterisk Veto (i=mdkuser@cpe-66-69-38-192.satx.res.rr.com) |
17:04.06 | *** join/#asterisk obiyoda (n=chatzill@24-119-167-174.cpe.cableone.net) |
17:04.24 | *** join/#asterisk loick (n=loick@per92-7-82-236-197-96.fbx.proxad.net) |
17:04.28 | kahuna_ | What does the term "debounce" mean when it comes to analog phones? |
17:04.54 | BartAleph | anyother useful address |
17:04.58 | [ViRii] | for overhead paging i need an fxo port? this port is on ,.... a soundcard? |
17:05.06 | kahuna_ | www.google.com |
17:05.30 | ronaldl79 | Anyone care to share why they prefer PRI over VoIP connectivity? |
17:05.38 | BartAleph | ok...i used it :)...but a need some guide to look...i'm new with asterisk |
17:05.45 | kahuna_ | [ViRii]: If I remember right you can just Dial() the console and the output will go out thru the soundcard |
17:07.26 | *** join/#asterisk veto (i=mdkuser@cpe-66-69-38-192.satx.res.rr.com) |
17:07.34 | *** join/#asterisk RoyK (n=roy@host-81-191-147-248.bluecom.no) |
17:07.52 | obiyoda | Hello. I am runnig A@H I have a queue but when all agents are on the phone it starts to beep at them when people are in the queue. I have turned off call waiting on the agents phone but still get the beep any ideas on what i can do? |
17:07.59 | *** join/#asterisk Cresl1n (n=matt@m415e36d0.tmodns.net) |
17:09.18 | *** join/#asterisk Beave (n=beave@vistech.org) |
17:09.19 | *** join/#asterisk syle (n=blag@unaffiliated/syle) |
17:11.57 | kahuna_ | what's the heck is kewlstart? I've never heard of this before I configured my * box. |
17:12.18 | blitzrage | kahuna_: its loopstart with far end disconnect supervision |
17:12.39 | kahuna_ | Ok |
17:13.17 | kahuna_ | So if I have lines going into a channel bank to be multiplexed into an * t1 port then ks is the right signalling? |
17:13.53 | blitzrage | kahuna_: kewlstart is usually the first thing to try - if it doesn't work, then its either groundstart or loopstart plain |
17:14.03 | kahuna_ | I see. |
17:14.20 | *** join/#asterisk ikey (i=ikey@220.226.37.186) |
17:14.45 | kahuna_ | I'm having trouble with users not pressing the hook button down long enough and instead of hanging up it actually initiates a transfer. |
17:15.12 | *** join/#asterisk Derkommissar (n=alberto@66.64.215.6.nw.nuvox.net) |
17:15.14 | Derkommissar | Halo |
17:15.25 | kahuna_ | There was no trouble at all when using loop start on the channel bank. |
17:15.37 | Derkommissar | question... i started to see a lot of this lately,,,,, Sep 28 13:09:11 WARNING[30442]: channel.c:1768 ast_indicate: Unable to handle indication 3 for 'SIP/5060-f539a738' |
17:15.37 | Derkommissar | <PROTECTED> |
17:16.33 | *** part/#asterisk \Grooby\ (n=Grooby@66.160.105.186) |
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17:18.24 | *** join/#asterisk Johnsie (n=john@acs-24-154-53-217.zoominternet.net) |
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17:19.44 | jarrod | can someone please give me an example mgcp.conf that works with Adit600 |
17:20.36 | }btorch{ | hmm .. whenever I enable the parkedcalls and add the Dial(...., rtT) ... if I call aanother phone there is no sound |
17:21.26 | *** join/#asterisk Gunnar (n=gunnar@34.80-203-209.nextgentel.com) |
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17:23.32 | *** join/#asterisk implicit (n=implicit@dhcp-248155.mobile.uci.edu) |
17:24.13 | kahuna_ | If I change something in my zapata.conf and reload will it kill any of my channels that are live? |
17:24.47 | *** join/#asterisk loick (n=loick@per92-7-82-236-197-96.fbx.proxad.net) |
17:25.14 | *** join/#asterisk TheCops (n=dump@206-248-136-187.dsl.teksavvy.com) |
17:25.16 | TheCops | Hi |
17:27.44 | Syncros | moo |
17:27.59 | Syncros | bon matin cops |
17:27.59 | ManxPower | }btorch{ don't use "r" and don't use "t" or "T" unless you HAVE TO |
17:28.20 | ManxPower | kahuna_, 1.0.x or CVS-HEAD/1.2beta? |
17:28.50 | *** join/#asterisk jeffgus (n=jeffgus@2002:d856:c704:0:0:0:0:1) |
17:29.08 | kahuna_ | 1.0.6-2 ubuntu .deb |
17:29.27 | *** join/#asterisk jief- (n=jief@modemcable163.182-80-70.mc.videotron.ca) |
17:29.40 | ManxPower | kahuna_, it will not kill any calls, but it will not update the settings either |
17:29.59 | kahuna_ | Ok. Best to wait until everybody is off the phone then :) |
17:30.03 | ManxPower | in 1.0.x you must unload chan_zap.so and then load chan_zap.so or stop and start asterisk for changes to take effect. |
17:30.09 | jief- | hello. i have to deploy a PBX with asterisk that will support up to 4000 phones. i was wondering if 1. anyone ever deployed such a large system 2. what kind of hardware should we use? |
17:30.26 | ManxPower | jief-, that is a question for the mailing lists |
17:30.35 | *** join/#asterisk danalien (n=danalien@unaffiliated/danalien) |
17:30.51 | jief- | ManxPower: just out of curiosity, why isn't this an irc question? |
17:31.37 | R3DB0x | does anyone have any suggestions bout the cisco 7920 phone....my customer wants a cordless setup and i heard that cordless VoIP stuff sucks and its better to use an analog phone with an IAXY converter |
17:31.43 | Druken | because 4000 phones is a major company, and major companies don't irc :) |
17:33.48 | JerJer | and that question leaves a lot open for discussion |
17:34.08 | JerJer | irc is only for verbally hating on those that annoy you |
17:34.26 | g__ | (All the more reason big companies should be on irc.) |
17:34.28 | jief- | JerJer: i would think its more because no one here knows the answer |
17:34.51 | JerJer | no - i have two entire school districts that i manage, using Asterisk |
17:35.00 | JerJer | they've got about 25,000 phones |
17:35.02 | [ViRii] | jerjer: what kinda phones? |
17:35.20 | JerJer | 7905s, 7960s and a few Sipura ATAs |
17:35.38 | [ViRii] | how do you get an overhead speaker to work with asterisk jerjer |
17:35.49 | JerJer | one of the sound card channels |
17:36.31 | JerJer | don't pm me |
17:36.35 | [ViRii] | sry |
17:36.38 | JerJer | bkw says i lie about people pm-ing me |
17:37.07 | [ViRii] | jerjer, a school has a pa system how does it work do you dial an extension and it activates? |
17:37.23 | JerJer | chan_oss or chan_alsa - depending on what sound system you use |
17:37.28 | JerJer | use/need - duno |
17:37.39 | jarrod | does anyone have an example mgcp.conf that works with the Adit 600? |
17:37.44 | [ViRii] | whats the difference between the two? |
17:37.45 | jief- | JerJer: since you've done such a large install, do you think one 2 way box can support up to 4000 phones? only SIP, no zap. |
17:37.57 | JerJer | depends on what's running on that box |
17:38.08 | [ViRii] | onboard sound. |
17:38.08 | JerJer | SER, certianly |
17:38.08 | JerJer | Asterisk, doubtful |
17:38.17 | jief- | JerJer: yeah would only be * |
17:38.23 | JerJer | plus you don't want just one media gateway / application server |
17:38.43 | JerJer | i would rather have 2 or more so i know not one outage will bring down the entire system |
17:38.51 | jief- | JerJer: well, * will makes calls through a Quintum Tenor CMS softswitch |
17:38.56 | JerJer | i'm sorry |
17:39.06 | jief- | JerJer: oh, its planned, there'll be a hotstandby |
17:39.16 | kahuna_ | [ViRii]: So you hook a line from your seaker out into the overhead pagind and dial the console in your dialplan. |
17:39.32 | kahuna_ | [ViRii]: Google it, there's examples all over. |
17:39.55 | jief- | JerJer: how would you split the load then? half the extensions on one box, and the other half on another? connected with iax2? |
17:39.59 | JerJer | no |
17:40.09 | [ViRii] | thanks |
17:40.11 | bjohnson | didn't someone have a way to use vonage economically? deal on at future shop and best buy this week .. $15 in your pocket after credits |
17:40.17 | JerJer | don't use vonage |
17:40.35 | kahuna_ | Vonage is * unfriendly |
17:40.46 | Juggie | whos goin to astricon? |
17:40.47 | JerJer | unless you are stupid enough to pay for the softphone account |
17:40.47 | [ViRii] | oh no |
17:40.51 | bjohnson | heh .. I got flack from some nitwits who I told shouldn't use skype |
17:40.52 | JerJer | Juggie: not me |
17:40.53 | hypa7ia | there are sooo many better canadian voip providers |
17:40.58 | [ViRii] | vonage =! asterisk? |
17:41.03 | file | jief-: if you're just doing SIP on a scale that large, just use SER routing calls to your Quintum and Asterisk for features |
17:41.13 | JerJer | file: word |
17:41.45 | Druken | one of these days i'll have to play with ser.... |
17:41.58 | fugitivo | where is documentation about ser_ |
17:42.08 | azzie | jief-, are you already doing SIP on Quintum CMS or just planning to? |
17:42.13 | JerJer | documentation? that's funny |
17:42.17 | *** join/#asterisk chiardon1 (n=chiardon@200.71.58.39) |
17:42.25 | }btorch{ | how come asterisks keeps giving me problems when I connect from the same IP as different users ? |
17:42.33 | JerJer | yeah i would avoid Qunitum like the plauge |
17:42.39 | bjohnson | the simplicity of vonage is very attractive to home users |
17:42.49 | jief- | azzie: its alraedy in use with 32 PRIs |
17:43.01 | }btorch{ | I change the SIP Proxy config on X-Lite and than I keep getting this Awaiting proxy login information |
17:43.02 | JerJer | bjohnson: if said home user wants to waste money, sure |
17:43.04 | azzie | jeif: 32 pri's and sip ? |
17:43.08 | *** join/#asterisk [hC] (n=hardcore@8.10.2.53) |
17:43.10 | jief- | azzie: yes |
17:43.10 | spackle | bjohnson: plug'n'play like AOL, brainless. But you pay for it. |
17:43.11 | [hC] | morning fellas. |
17:43.32 | azzie | jief-, impressive, so quintum managed to make SIP more-or-less stable? |
17:43.40 | JerJer | less |
17:43.57 | jief- | azzie: well, last i was there 2 days ago, and asterisk didnt seem to be able to talk with it. i was always getting a 503 error code |
17:44.03 | bjohnson | JerJer: I've run into numerous ones who insist that anything else is too hard |
17:44.05 | *** join/#asterisk klasstek (n=nunyobiz@sta-206-168-218-202.rockynet.com) |
17:44.05 | [hC] | I've upgraded to cvs head as of last night (from cvs head august 15th) and I'm still having odd IAX frames show up on an iax link of mine. Ive included output in a pastebin, incase anyone can take a look and offer a suggestion |
17:44.10 | [hC] | http://pastebin.ca/24128 |
17:44.10 | jief- | azzie: going back there tomorrow, hoping to get something done |
17:44.41 | azzie | jief: emm... so do you have standing sip calls on your CMS or you're just planning to have them in the future? |
17:44.51 | kahuna_ | You know who's the worst voip provider though? |
17:45.02 | kahuna_ | Nufone! |
17:45.03 | *** join/#asterisk steve___ (n=steve@store-fw.porchlight.ca) |
17:45.22 | jief- | azzie: the client used Ondo (from bekeke.com) for a while with the CMS. so SIP is fine |
17:45.38 | JerJer | yeah the cock suckers at NuFone will never return my calls |
17:45.44 | kahuna_ | lol |
17:45.53 | bjohnson | JerJer: I even met one bonehead who (talking about skype) said if it's easy now, who cares about ease of upgrading in the future. I had mentioned that they could get a standard SIP fxs and config that easily |
17:45.57 | jief- | azzie: just doesnt seem like it likes *. if i use a phone, and set the CMS' IP as the outbound gateway, i can make outgoing calls, but not if i route them thru * first |
17:46.11 | JerJer | bjohnson: yeah really |
17:46.39 | JerJer | jief-: Qunitum very much so does their own thing |
17:46.40 | hardwire | grr |
17:46.41 | azzie | jief-, i see... my company has a pile of quintums (all kinds) and I was just wondering if it's the time for SIP in this department :) |
17:46.44 | hardwire | I updated to CVS :) |
17:46.45 | kahuna_ | skype sucks unless you're stuck with dialup. period. |
17:46.46 | hardwire | and now the t1 doesn't work |
17:46.48 | hardwire | turns out we had an outage at the ILEC right when I changed |
17:46.50 | JerJer | we were forced to drop their bullshit from Interop testing |
17:46.59 | hardwire | kahuna_: skype also can use a wideband codec |
17:47.00 | ronaldl79 | Guys: Please refresh my memory ... but in the diaplan, how is the ampersand used to ring two SIP devices at once? |
17:47.05 | hardwire | more hz for the buck |
17:47.10 | hardwire | for skype to skype calls |
17:47.20 | JerJer | Dial,SIP/bob&SIP/dillan |
17:47.20 | kahuna_ | yes it can, but you can do standard voip then... |
17:47.30 | jief- | azzie: might have been my * config that was wrong though. in a macro, i had exten => s,1,Dial(SIP/1.2.3.4). and that's how i was getting that 503 error |
17:47.32 | hardwire | kahuna_: really? |
17:47.33 | azzie | jer: dropped Q-m's? |
17:47.35 | ronaldl79 | Yeah, that's what I did, JerJer...hmmm...thanks |
17:47.42 | hardwire | do you have some voip software that does wideband codecs? |
17:47.45 | }btorch{ | why does asterisk keeps showing me 71488/71488 192.168.120.212 D 255.255.255.255 5060 Unreachable ? |
17:47.50 | JerJer | Q-m's ? |
17:48.11 | [hC] | hardwire: haha. doh. I spent 20 minutes trying to debug why wcfxo wasnt inserting last night only to discover the card had been removed from the box i was upgrading earlier to debug. :) |
17:48.11 | azzie | jer: dropped quintums from interop testing? |
17:48.14 | JerJer | azzie: yes |
17:48.17 | azzie | jer: wow |
17:48.19 | JerJer | they suck |
17:48.20 | hardwire | [hC]: your a fucking moron. |
17:48.23 | hardwire | :) |
17:48.29 | hardwire | atleast I bet thats how you feel |
17:48.37 | *** join/#asterisk brookshire (n=matt@gateway.digium.com) |
17:48.40 | azzie | jer: true |
17:48.46 | azzie | jer: any particular reason? |
17:48.46 | [hC] | hardwire: yes. its worse though, cause the box is remote, and i couldnt actually check for myself. |
17:48.51 | hardwire | I am a fucking moron.. because I spent all night trying to add a new app to chan_sip.c |
17:48.58 | hardwire | just to find out there was a dialplan functino that does what I want |
17:49.04 | JerJer | not even the guy Qutium sent could configure the bastards right |
17:49.07 | jief- | ok, how do you send SIP calls, to another SIP device (like a softswitch)? Dial(SIP/1.2.3.4) ? |
17:49.07 | hardwire | same code.. same everything |
17:49.08 | [hC] | hardwire: haha. what were you trying to add? |
17:49.10 | kahuna_ | Well there are proprietary wideband codecs, so I retract my statement about "standard" voip/ |
17:49.18 | hardwire | I awnted to get sip peer information |
17:49.20 | hardwire | wanted |
17:49.30 | JerJer | jief-: use a type=peer, so you have more control |
17:49.33 | JerJer | ~docs |
17:49.34 | jbot | it has been said that docs is Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk |
17:49.34 | hardwire | like the UserAgent |
17:49.41 | [hC] | hardwire: ahh. |
17:49.48 | *** join/#asterisk chiardon (n=chiardon@200.71.58.39) |
17:49.48 | hardwire | so I added SipGetPeerData |
17:49.51 | JerJer | hardware sip show peer bob |
17:50.04 | hardwire | just to find out ${SIPPEER(1153:useragent)} exists |
17:50.15 | [hC] | hardwire: yeah. you are a fucking moron too, then. |
17:50.18 | JerJer | RTFM |
17:50.18 | hardwire | yup |
17:50.20 | [hC] | :) |
17:50.21 | kahuna_ | hehe |
17:50.32 | hardwire | JerJer: for use in another app |
17:50.36 | jief- | JerJer: yeah, i tried adding an entry in sip.conf for the softswitch, but with type=peer, but i always get a 503 error |
17:50.36 | hardwire | not from the CLI |
17:50.38 | azzie | JerJer, oh... their SIP is screwed in particular or you are talking in general? |
17:50.50 | [hC] | hey, has anyone bought that oreilly book on *? amazon has it for 25 bucks, i was thinking of picking it up for fun. i suspect it will be way too basic, though. |
17:50.56 | kahuna_ | you should go work for Microsoft then, trying to re-invent the wheel/ |
17:51.02 | JerJer | azzie: all of the above - they don't follow the reco |
17:51.09 | azzie | [hC], I did, but they still did not ship it :( |
17:51.12 | hardwire | kahuna_: ? |
17:51.13 | JerJer | they do their own thing quite a whole lot |
17:51.44 | hardwire | my span is freaking red alarm |
17:51.45 | hardwire | I hate this |
17:51.57 | ronaldl79 | JerJer: Sip/XXX&Sip/XXX -- is that the best way to ring two devices at once? Does the same apply to IAX, etc. too? |
17:51.58 | *** join/#asterisk AgiNamu (n=Michael@dsl081-096-215.den1.dsl.speakeasy.net) |
17:52.13 | JerJer | ronaldl79: you can even mix channel types |
17:52.19 | ronaldl79 | Good deal. |
17:52.26 | ronaldl79 | Gotta love *... |
17:52.35 | JerJer | other than using a queue setup, thats really the only way i can think of |
17:52.45 | azzie | JerJer, I see |
17:52.55 | azzie | JerJer, thank you for the info |
17:53.03 | JerJer | thank you, drive-thru |
17:53.08 | azzie | :) |
17:53.33 | JerJer | iax2 is male |
17:53.34 | kahuna_ | ouch |
17:53.45 | kahuna_ | A male with ovaries?! Good God! |
17:53.45 | spackle | just undecended? |
17:53.51 | JerJer | perhaps transgender, we don't really know |
17:53.53 | [hC] | JerJer: really... it seems like a bitch to me. :| |
17:53.59 | bendy24 | o0 |
17:54.02 | [hC] | pre-op or post-op? |
17:54.04 | chiardon | ovaries is a so deeply layer . .isn't it? |
17:54.05 | AgiNamu | Anyone here decent with configuring PRIs (non asterisk, but I'll pay for help) |
17:54.17 | hardwire | hmm |
17:54.21 | hardwire | is zaptel in CVS dead? |
17:54.22 | azzie | AgiNamu, what equipment |
17:54.27 | AgiNamu | azzie, cisco as5350 |
17:54.28 | }btorch{ | great .... asterisk is pissing me off |
17:54.30 | *** join/#asterisk `Sauron (i=sauron@h-69-3-12-50.hstqtx02.covad.net) |
17:54.36 | JerJer | AgiNamu: try #cisco |
17:54.42 | azzie | AgiNamu, what could be simpler then |
17:54.44 | AgiNamu | oh.. duh :P |
17:54.47 | [hC] | }btorch{: get in line, sunshine |
17:54.52 | }btorch{ | hehe |
17:55.09 | ronaldl79 | JerJer: Is it possible to ring more than two sip devices, e.g, Sip/XXX&Sip/XXX&Sip/XXX? |
17:55.24 | azzie | AgiNamu, set up t1/e1, isdn type and that's all you need... :) |
17:55.27 | *** part/#asterisk santiago (n=santiago@63.245.86.245) |
17:55.33 | *** join/#asterisk veto (i=mdkuser@cpe-66-69-38-192.satx.res.rr.com) |
17:55.34 | AgiNamu | azzie, yea, I did... that was easy |
17:55.39 | mutilator | ronaldl79 use a comma.. |
17:55.40 | AgiNamu | but it just stays at "TEI_ASSIGNED" :P |
17:56.12 | *** join/#asterisk veto (i=mdkuser@cpe-66-69-38-192.satx.res.rr.com) |
17:57.03 | azzie | AgiNamu, show contr t1 and make sure it's up and no errors; then debug isdn q921 and debug isdn q931 |
17:57.21 | ronaldl79 | How are you guys addressing instant termination in your dialplan? There's generally a delay unless you press "#". |
17:57.35 | AgiNamu | azzie, yep, T1 is up fine. I'm debugging q931 and 921 |
17:57.48 | file | ronaldl79: if it's a SIP device that's on the SIP device's dialplan, not asterisk |
17:57.56 | AgiNamu | I just see a bunch of "sending SABME", User TX.... and on 931 I see "Ux_DLRelInd: DL_REL_IND received from L2" |
17:57.59 | AgiNamu | that's it |
17:58.01 | ronaldl79 | What about IAX, file? |
17:58.08 | file | an IAX2 phone? |
17:58.15 | kahuna_ | I just configured my 1st PRI on * today. |
17:58.15 | ronaldl79 | Yes, file...hard or soft. |
17:58.26 | file | it's all the phone/ata... |
17:58.30 | azzie | AgiNamu, your side hopefully is user-side (not network) ? |
17:58.35 | ronaldl79 | Cool, file...thanks. |
17:58.36 | AgiNamu | right, userside |
17:58.39 | chiardon | kahuna_ clap . .clap . . .clap . . . |
17:58.44 | file | it doesn't communicate with Asterisk when you're dialing, it waits for you to finish before it goes |
17:59.30 | JerJer | ronaldl79: last i knew you could string together 254 devices to ring |
17:59.43 | JerJer | and its not a comma its an ampersand |
17:59.49 | kahuna_ | I'm sure you guys have done hundreds of them but you probably still remember the satisfaction of your first time. |
17:59.51 | ManxPower | What?!?!? |
17:59.57 | ronaldl79 | Whew, JerJer, that's amazing.... |
18:00.04 | ManxPower | Tech/device&tech/device |
18:00.34 | JerJer | ronaldl79: at some point a queue becomes more efficient |
18:01.07 | ManxPower | Queues are EVIL. However, there always has to be some evil in the world. |
18:01.28 | *** join/#asterisk fiber0pti (n=johndoe@207.114.199.98) |
18:01.53 | *** join/#asterisk extremis (i=extremis@equinox.alluvium.com) |
18:02.04 | fiber0pti | I'm trying to get some questions answered about Polycom 500/501 phones and asterisk. Is it possible to have these 3 line phones utilize more than 3 lines? |
18:02.13 | extremis | Is there a way on a 7960 to transfer an incoming call to voicemail imediately without having to pick up the phone? |
18:02.28 | JerJer | extremis: have a direct to voicemail extension |
18:02.55 | extremis | JerJer: as the user receiving a call, I want to bounce the incoming call to voicemail after it has been transfered to me |
18:02.55 | JerJer | like exten => 6XXX,1,Voicemail,u${EXTEN:1} |
18:02.55 | [ViRii] | jerjer: oss or alsa , got an onboard soundcard how to get it to recognize in linux/asterisk |
18:03.17 | JerJer | [ViRii]: duno - i have never had luck with either |
18:03.22 | [ViRii] | lol |
18:03.24 | [ViRii] | thanks :> |
18:03.43 | JerJer | extremis: where ${EXTEN} is their three digit mailbox, of course |
18:03.50 | extremis | for example, someone dials me from the main menu |
18:03.52 | ronaldl79 | Now, this is strange...I just built a Suse Linux box last night with Asterisk 1.0.9 ... I was just attempting to modify some things, and Pico is telling me I can't ... because it's a 'read only filesystem.' Damn, I hope this box isn't fucking up...I'm demoing some stuff today for a client!!!! |
18:03.54 | extremis | and I dont want to talk to them |
18:03.58 | extremis | and I don't want it to ring |
18:04.07 | JerJer | put your phone on DND |
18:04.09 | extremis | so I evaluate based on caller id and decide, send that person to voicemail now |
18:04.15 | JerJer | then do that |
18:04.28 | extremis | so I can put it in DnD in the middle of an incoming call, and it will get redirected? |
18:04.30 | JerJer | anti ex-girlfriend logic style |
18:04.34 | JerJer | no |
18:04.35 | [ViRii] | ronald: ack |
18:04.41 | JerJer | well doubtfull - test it |
18:04.43 | extremis | well, a client wants it, yes anti-exgirl logic |
18:04.50 | JerJer | then use it |
18:04.52 | extremis | they said their existing pbx does it |
18:05.02 | extremis | well, dnd is there, but does it work when the call is already coming? |
18:05.04 | JerJer | call routing based on Caller*id ? |
18:05.06 | lathos42 | extremis: The Polycom phones have a "Reject" feature.. You should see if the 7960 has a similar feature |
18:05.27 | extremis | again, the call is ringing at my phone, and at that exact moment, I want to bounce them to voicemail |
18:05.32 | extremis | the next call I may want to receive |
18:05.58 | bjohnson | transfer? |
18:05.58 | JerJer | thats' not anti ex-girlfriend logic then |
18:06.14 | bjohnson | but you would have to answer the call first |
18:06.16 | extremis | well, that is what they are used to |
18:06.29 | [hC] | hardwire: huh.. looking further, when i had thought that frame_control type 3 was video.... its actually a "remote end is ringing" indication. |
18:06.30 | bjohnson | you mean they want to see the callerid and then decide whether to answer or not? |
18:06.31 | extremis | they have a pbx that you can hit a button on and it dumps the current incoming call to voic email |
18:06.44 | bjohnson | why not just wait for the time out |
18:06.46 | extremis | bjohnson: yes, but if they don't want to answer, they want to send to voicemail imediately |
18:06.47 | JerJer | see if cisco's can do that |
18:07.04 | bjohnson | eg ring for 10 seconds and then send to vm |
18:07.07 | JerJer | see lathos42's comment |
18:07.07 | extremis | they don't want to wait for a timeout |
18:07.10 | *** join/#asterisk bzbw (i=bwz@ip67-153-142-109.z142-153-67.customer.algx.net) |
18:07.14 | bjohnson | if they don't want to answer, turn off the ringer |
18:07.22 | bjohnson | strange request |
18:07.27 | extremis | well, they are used to the feature |
18:07.29 | ManxPower | all that stuff is controlled BY THE DAMN PHONE |
18:07.29 | bzbw | <PROTECTED> |
18:07.30 | extremis | that is why they want it |
18:07.31 | bjohnson | that they would act that quickly |
18:07.38 | extremis | I can change their behavior, or I can find a way to do it |
18:07.44 | extremis | it doesn't look like I can find a way to do it |
18:07.44 | bjohnson | no I don't want to wait 10 seconds to not talk to this person |
18:07.44 | ManxPower | bzbw, yes, yes |
18:07.45 | JerJer | Asterisk v1.2 is not out yet |
18:07.55 | JerJer | extremis: just show them DnD feature |
18:08.04 | extremis | JerJer: any way to put it on the softmenu? |
18:08.10 | bzbw | I mean beta for 1.2 |
18:08.19 | JerJer | duno - read the TFTP configs for cisco |
18:08.21 | JerJer | perhaps |
18:08.24 | bjohnson | but dnd will not ring their phone correct? |
18:08.27 | bzbw | ManxPower, what is the driver name? |
18:08.31 | ManxPower | extremis, the 7905 supports rejecting the calls. |
18:08.41 | lathos42 | extremis: Cisco would have to support it in the phone.. Polycom has a feature in the Softmenu that does exactly what you want to do.. call is coming in, hit reject, it bounces immediately to voicemail |
18:08.44 | extremis | ManxPower: really? |
18:08.45 | ManxPower | bzbw, read the zaptel README file, it lists each driver and what board it is for |
18:08.47 | hardwire | [hC]: I freaking told you |
18:08.50 | ManxPower | extremis, yes. |
18:08.53 | hardwire | what did I freaking tell you |
18:08.57 | hardwire | that it was ringing |
18:08.59 | hardwire | I told you |
18:09.00 | bzbw | thx |
18:09.00 | bjohnson | easier to just turn off the ringer if it has a ringer switch |
18:09.02 | hardwire | I told you |
18:09.07 | ManxPower | extremis, But I don't think the other 79XX SIP phones support it |
18:09.12 | JerJer | lathos42: well if the call agent is setup to send them to voicemail :) |
18:09.26 | extremis | ManxPower: they have some 7912s... do those support it? |
18:09.33 | ManxPower | extremis, I would assume so. |
18:09.41 | lathos42 | JerJer: True :) I think i'm going to make poeple have to work to get to my voicemail box |
18:09.45 | JerJer | why wouldn't 7960 have it? its just a software function |
18:10.00 | ManxPower | When I had a 7905 for testing I had to update my dialplan to make the call go to the right place when you rejected the call |
18:10.11 | bjohnson | lathos42: guess a number between one to ten? |
18:10.16 | ManxPower | I think it's not called "REJECT" I think it's a softbutten labeled "toVM" or something like that |
18:10.25 | lathos42 | bjohnson: That's a good idea.. That might be step 5 :) |
18:10.28 | bjohnson | lathos42: all wrong answers they hear "that's not it. try again" |
18:10.30 | JerJer | lathos42: i can see it now..."Please dial Pi out to the 30th digit to leave VM for bob" |
18:10.58 | lathos42 | JerJer: Yep, and its their problem you can't dial . |
18:11.04 | gein | can someone help me configuring queues? (having problems with the joinempty option) |
18:11.19 | gein | configure* |
18:11.28 | JerJer | can we? |
18:12.05 | gein | may |
18:12.14 | JerJer | don't ask to ask, just ask |
18:12.23 | lathos42 | bjohnson: It would be even better if with every wrong answer, they have to start the menu over again from the top |
18:12.46 | gein | ok, sorry, seems like joinempty=strict has no effect at all... calls are placed in the queue even if there's no agents logged in |
18:12.58 | ManxPower | JerJer, The 7905/7912 use their own firmware, not the same at the 7940/7960, there are some differences between the two firmwares |
18:13.22 | JerJer | yea i believe the 7905/12 are based on the komoto ata shit |
18:13.32 | ronaldl79 | I know this is only *, but why the hell is this Suse box not loading anything -- everything is read-only now? If the timing couldn't be better!!!! Is something corrupted? |
18:14.01 | JerJer | press ctrl-alt-delete |
18:14.02 | }btorch{ | how do you do a database del in the CLI ? |
18:14.03 | ronaldl79 | I've never experienced this before, and here I am all excited about demoing this box ... and now I've got to troubleshoot before the 1:30 PM meeting! |
18:14.04 | JerJer | see if that fixes anything |
18:14.11 | ronaldl79 | I already did that, JerJer, it's still the same thing. |
18:14.21 | JerJer | dont' run suse |
18:14.23 | spackle | did you mount the volume read-only |
18:14.30 | ronaldl79 | No, spackle. |
18:14.33 | ManxPower | ronaldl79, NEVER EVER demo a box on short notice. |
18:14.34 | JerJer | drop a netinstall of debian on there - worst of both worlds |
18:14.38 | ronaldl79 | Everything was fine last night and this morning... |
18:14.50 | bjohnson | }btorch{: check the chan that supports your db .. ie #mysql for mysql |
18:14.55 | mutilator | short notice = ? |
18:14.55 | jarrod | hmm.. adit600 not registering with asterisk mgcp |
18:14.56 | ronaldl79 | I've always used Suse.... |
18:14.59 | mutilator | appropriate time = ? |
18:15.31 | bjohnson | }btorch{: there are typically commands to do that. If you got the db as part of a package, you could look at the files the package contains that go into some /bin dir |
18:15.32 | }btorch{ | bjohnson: I'm not using mysql though but when I do a database show it shows me to phones in the registry |
18:15.36 | JerJer | lathos42: hell yeah |
18:16.00 | [hC] | hardwire: stfu |
18:16.01 | [hC] | :) |
18:16.10 | }btorch{ | bjohnson: the strange thing is one phone is connect but the other one is not I closed the app and still shows as register |
18:16.12 | JerJer | BUFU |
18:16.14 | ManxPower | jarrod, I thought Asterisk's MGCP support was client only, not server. |
18:16.27 | jhiver | Hi All |
18:16.38 | jarrod | thats incorrect |
18:16.56 | file | it's server only |
18:17.17 | jhiver | Any body has some info about that GSM Asterisk card that has been advertised on the ML? |
18:17.51 | JerJer | its vapor until i see one |
18:17.53 | ManxPower | jarrod, must have changed in the past year then |
18:18.10 | JerJer | ManxPower: yeah i think someone contributed a serious amount to mcgp |
18:18.12 | JerJer | mgcp |
18:18.16 | ManxPower | I have all my projects on hold until I can get the Digium DS-3 card! |
18:18.17 | ManxPower | NOT! |
18:18.23 | jhiver | Sounds good though, I hope they manage to get some stuff working |
18:18.53 | jhiver | it would be good to have sub €500 / port GSM card |
18:18.56 | JerJer | Did atacom ever release their bullshit? |
18:20.17 | bjohnson | I don't know, but bullshit is generally freely available from multiple sources |
18:20.36 | bjohnson | I've got some here if you want it |
18:20.42 | [hC] | hardwire: i wonder then if this is even my FREAKING problem after all. |
18:20.56 | hardwire | probably not |
18:21.01 | hardwire | have yup upped to 1.2.0 or CVS yet? |
18:21.08 | hardwire | yup/you |
18:21.21 | [hC] | i was running cvs head 8/15, and cvsup'ed to last night's cvs... last night... |
18:21.25 | bjohnson | [hC]: if you have an underling, then it can quickly become somebody else's problem |
18:21.35 | [hC] | bjohnson: oh god, do i ever wish. |
18:21.40 | blitzrage | in stable, is it possible to strip a group of digits from the "centre" of the ${EXTEN} ? |
18:21.41 | [ViRii] | anyone got paging and intercom to work with polycom600's? |
18:21.49 | spackle | "underling" heh |
18:21.51 | hardwire | bjohnson: hahahaha |
18:21.56 | blitzrage | ie. 111222444 <- strip 222 |
18:22.03 | blitzrage | resulting string 111444 |
18:22.10 | hardwire | blah |
18:22.12 | [hC] | we call those ones 'boah' around here... as in, fetch me my coffee, boah. |
18:22.14 | hardwire | bbl |
18:22.16 | pauldy | ronaldl79, did you try a little mount -o remount,rw / |
18:22.21 | bjohnson | underling .. what do they call the buddies of the bad guys in the old movies? henchmen? |
18:22.37 | Beirdo | I like "peon" or "minion" |
18:22.38 | hardwire | board of animal health? |
18:23.06 | bjohnson | serf |
18:23.15 | spackle | grunt |
18:23.39 | Beirdo | if it's a guy (and only then): "bitch" |
18:23.40 | spackle | lathos42: when you get promoted you can be the IT mangler |
18:23.54 | JerJer | perhaps ${EXTEN:9:-3} |
18:23.55 | file | have a little faith in me! |
18:24.11 | JerJer | women are bitches though |
18:24.14 | Beirdo | some are |
18:24.22 | lathos42 | spackle: I was reading Slashdot yesterday and someone mentioned in a comment that he was able to make up his own job title once, so he chose "Master of the Devices" |
18:24.25 | Beirdo | but it's not polite to call them that in a work situation |
18:24.27 | pauldy | holy mother flooder |
18:24.34 | pauldy | this server must be lagged |
18:24.58 | spackle | pauldy - why did you send your message 20 minutes ago? |
18:25.00 | Beirdo | you can get slapped with sexual harassment SOOO fast |
18:25.24 | pauldy | spackle, see previous statement |
18:25.52 | *** join/#asterisk Mw3 (i=mw3@spool12-14.gatesgroup.hu) |
18:25.55 | Beirdo | woohoo, go linode |
18:26.07 | [hC] | <3 linode |
18:26.30 | Beirdo | it just bounced my IRC connections :) |
18:26.50 | Beirdo | at least I'm gonna blame that |
18:26.51 | Beirdo | heh |
18:27.25 | }btorch{ | is there a place to change some timeout value for when a user becomes unreachable |
18:27.35 | *** join/#asterisk MnxPower (n=eric@stirprop-s4-0-0-21.ndcr2.datasync.net) |
18:27.35 | *** join/#asterisk pauldy (n=pauldy@24-155-86-154.ip.grandenetworks.net) |
18:27.41 | pauldy | maybe that will help |
18:29.47 | bjohnson | }btorch{: you're gonna have to be way more specific |
18:29.47 | g__ | jerjer, followup question from yesterday: why are "agents" important to queues.. why doesn't it work properly otherwise? |
18:29.47 | JerJer | g__: i use queues without agents all of the time |
18:29.52 | JerJer | member = > SIP/bob |
18:30.07 | JerJer | but SIP/bob has to have call waiting turned off |
18:30.09 | }btorch{ | bjohnson: for some reason I had an sip phone registration that was still showing up if you did a database show on the asterisk CLI ... |
18:31.00 | }btorch{ | bjohnson: it took around 10min for the registration to go way ... it keept the port open and status as unreachable ... |
18:31.02 | *** join/#asterisk bockx (i=bock@128-5-112.adsl.terra.cl) |
18:31.41 | g__ | ..and if SIP/bob doesn't, he gets calls even when he's busy? Ok, then if SIP/bob doesn't pick up the phone, will it go to someone else? |
18:32.04 | }btorch{ | bjohnson: using the same PC with a new X-Lite SIP proxy configuration would register the new phone and bring the old one back online |
18:32.13 | JerJer | then SIP/bob need to unregister or set DND |
18:32.33 | JerJer | and yes, if SIP/bob does not pick up the call will get sent to another member of the queue |
18:33.05 | g__ | I've found if SIP/bob sets the phone to DND or forgets to unregister, calls get stuck on his phone. |
18:33.13 | JerJer | stuck? |
18:33.13 | malverian[work] | Hmm.. asterisk keeps getting killed by kernel.. I need to figure out why :-/ |
18:33.25 | JerJer | by kernel? |
18:33.28 | g__ | Ie, they don't pass on to another extension |
18:33.31 | dsfr | hah |
18:33.43 | JerJer | g__: then you are doing something wrong |
18:33.45 | malverian[work] | JerJer, Almost positive.. It's saying "Killed" and then closing.. might be an OOM condition. |
18:34.00 | malverian[work] | Though I'm not seeing anything in dmesg |
18:34.01 | JerJer | no someone is doing kill <pid of asterisk> |
18:34.54 | malverian[work] | JerJer, I'm the only person with access to the machine. |
18:34.59 | malverian[work] | Some_THING_ is doing it. |
18:35.03 | malverian[work] | And it's happening quite frequently. |
18:35.15 | g__ | Under some conditions, the kernel does pick arbitrary processes to kill.. but those conditions are usually "low memory" or something equally bad. |
18:35.25 | pauldy | while test true; do kill -9 $RANDOM;done |
18:35.30 | malverian[work] | g__, Yeah, that's what I was saying.. OOMkilller |
18:35.31 | pauldy | thats a fun one to run |
18:35.54 | JerJer | how does $RANDOM get set? |
18:35.56 | bjohnson | }btorch{: I believe SIP device registration timeouts are controlled on each device |
18:35.58 | g__ | oh right, sorry. |
18:36.02 | JerJer | or is that a special shell var? |
18:36.02 | malverian[work] | pauldy, Except 99% of the time, you won't get a valid pid ;) |
18:36.09 | *** part/#asterisk simbulu_ (n=chatzill@80.77.133.83) |
18:36.11 | malverian[work] | JerJer, Yeah |
18:36.13 | malverian[work] | JerJer, Bash var. |
18:36.15 | JerJer | ahh ok |
18:36.21 | pauldy | JerJer, shell var |
18:36.34 | pauldy | yea malverian[work] but its like playing rusian roulet |
18:36.39 | malverian[work] | pauldy, Heh :) |
18:36.45 | malverian[work] | Anyone know how to disable OOM killer? |
18:36.46 | JerJer | click click click click click |
18:36.49 | JerJer | boom |
18:36.51 | malverian[work] | Is it sysctl? |
18:36.52 | }btorch{ | bjohnson: are you talking about the timeout for trying to register or when you close the application ? |
18:36.54 | pauldy | and running it like that is bound to kill something |
18:36.56 | pauldy | maybe itself |
18:37.05 | g__ | I'll stick the Doom systems administration mode. |
18:37.14 | g__ | s/stick/stick with |
18:37.30 | *** join/#asterisk ManxPower (n=eric@stirprop-s4-0-0-21.ndcr2.datasync.net) |
18:37.32 | *** join/#asterisk psypete (n=psypete@pix-nat.makosurgical.com) |
18:38.29 | *** join/#asterisk zeedo (n=zeedo@www.reboot-robot.net) |
18:38.45 | }btorch{ | bjohnson: do you use x-lite ? |
18:38.50 | g__ | Ok, so the point of 'Agents' is to let people move around a bit more? |
18:39.04 | [hC] | this might sound stupid, but is it possible that a frame request for 'remote ringing' on an IAX channel, could cause other calls via that IAX link to cease audio rx/tx? |
18:39.14 | *** join/#asterisk Inkubot (n=pancho@200.75.4.7) |
18:39.37 | pauldy | g__, resource allocation and call management |
18:39.47 | pauldy | otherwise just use a ringgroup and be done with it |
18:40.13 | pauldy | plus I think they only way you cna have hold music in asterisk is in a queue, is that correct? |
18:40.17 | Inkubot | i can't dial trough a sip proxy to the pstn |
18:40.29 | Inkubot | i don't know what it's wrong |
18:40.30 | jake1932 | just saw this start - my screen repaints the CLI output every so often to the term window |
18:40.45 | jake1932 | was this added in HEAD recently? |
18:40.45 | convey | I need help.. lol |
18:40.47 | g__ | pauldy: no, you can put someone on hold as well |
18:40.54 | malverian[work] | g__, Hmm.. maybe it could be the program hitting a stack limitation? |
18:40.57 | g__ | Or just play them hold music for the fun of it. |
18:40.59 | ManxPower | pauldy, see the "m" option to "show application dial" |
18:40.59 | malverian[work] | g__, Eg. ulimit -s |
18:41.00 | pauldy | I mean while you ring the ext |
18:41.14 | pauldy | oh yea forgot all about that thanks MP |
18:41.25 | blitzrage | being on hold r0x |
18:41.56 | jake1932 | blitzrage: that all depends on the music |
18:42.11 | g__ | malverian: anything's possible.. for starters, what kernel are we talking about? Linux, FreeBSD, Darwin.. and what version? |
18:42.13 | blitzrage | jake1932: yah... I should have added an </sarcasm> tag :) |
18:42.19 | pauldy | I like calling in via 7777 dialing my own extension then putting myself on hold |
18:42.21 | pauldy | it soothes me |
18:42.29 | jake1932 | being on hold with metallica r0x |
18:42.30 | blitzrage | you're all sick, you know that right? :) |
18:42.41 | blitzrage | jake1932: I hate metallica (mostly for political reasons) |
18:42.44 | ManxPower | jake1932, only if you are using an uncompresed codec |
18:42.49 | jake1932 | lol |
18:43.17 | AgiNamu | metallica on ipc10 is awesoem |
18:43.28 | JerJer | LPC10 |
18:43.36 | AgiNamu | yea yea thats what I meant |
18:43.47 | AgiNamu | whats even cooler is that with LPC10 is that me just talking sounds like metallica |
18:44.12 | JerJer | strangely enough we do have a few ppl that regularly use LPC10 |
18:44.20 | AgiNamu | s/whatever I wrote/what I meant to write |
18:44.33 | *** join/#asterisk tdonahue (n=tdonahue@64.201.13.50) |
18:44.42 | malverian[work] | pauldy, # for a in `seq 1 10000`; do ps ax | cut -d' ' -f1 | grep $RANDOM | grep -v grep; done | sort | uniq -u | wc -l |
18:44.42 | malverian[work] | 4 |
18:44.45 | malverian[work] | pauldy, ;) |
18:44.46 | *** part/#asterisk TripleFFF2sdf (n=TripleFF@modemcable131.156-131-66.mc.videotron.ca) |
18:46.07 | *** join/#asterisk rking (n=rking@ip68-105-231-56.lu.dl.cox.net) |
18:46.11 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
18:46.11 | *** mode/#asterisk [+o denon] by ChanServ |
18:46.27 | pauldy | whats the point of that |
18:46.40 | tdonahue | hi all, i just got asked a question that I have no idea how to answer. how can I have a receptionist transfer directly to someone's voicemail box |
18:46.41 | *** join/#asterisk abcd100 (n=abcd@61.16.172.254) |
18:47.16 | brookshire | yeah |
18:47.18 | pauldy | *<ext> |
18:47.20 | arp2 | malverian, you dont do shell scripting for a living do you? :) |
18:47.37 | pauldy | tdonahue, for most close to default setups it will be just * plus the ext |
18:47.44 | Katty | what do you call someone who develops dialplans? |
18:47.46 | JerJer | tdonahue: have a direct to voicemail extension |
18:47.59 | arp2 | katty, communications specialist? |
18:48.04 | JerJer | exten => 6XXX,1,Voicemail,u${EXTEN:1} |
18:48.06 | Katty | hmmmmmmmmm. |
18:48.08 | CoffeeIV_ | tdonahue: you can (or may already -- it's in the sample configs) have the extension *123 be the voicemail for *123 |
18:48.08 | pauldy | kajtzu, a dial planner |
18:48.09 | [hC] | Katty: a sucker? |
18:48.09 | JerJer | is there an echo in here? |
18:48.21 | brookshire | ahaha.. a dial planner |
18:48.25 | tdonahue | thanks guys |
18:48.28 | brookshire | that's great |
18:48.31 | fugitivo | lol |
18:48.32 | CoffeeIV_ | tdonahue: so then she just transferes to *xxx |
18:48.39 | *** join/#asterisk dabigshiznizzle (n=dabigshi@gwsecure.sctelcom.com) |
18:48.54 | JerJer | where XXX is their three digit mailbox |
18:49.02 | JerJer | modify accordingly, of course |
18:49.32 | }btorch{ | bjohnson: I though asterisk is the one that takes care of changing SIP registration status from OK -> UNREACHABLE -> UNKOWN |
18:49.43 | kajtzu | pauldy: ? |
18:51.10 | pauldy | kajtzu, can I get an ahmen |
18:51.52 | netnameus | anyone have any opinions on the Grandsteam Budgetone 101 IP phones? Good or bad |
18:52.08 | JerJer | you mean BarbieTones |
18:52.16 | netnameus | lol |
18:52.20 | denon | netnameus: the good is that they're cheap |
18:52.21 | spackle | netnameus: they should be the fisher-price barbietone |
18:52.22 | pauldy | better than a hot poker in your eye but don[t get your hopes up |
18:52.23 | denon | the bad is that they're cheap |
18:52.44 | kajtzu | pauldy: anytime :) |
18:52.47 | netnameus | that's what i thought... just wanted to make sure |
18:52.51 | denon | personally I've had pretty decent luck with them, but I wouldnt say that's the norm |
18:53.01 | denon | and I wouldnt trust them to anything important |
18:53.03 | spackle | netnameus: they are link training wheels for Asterisk |
18:53.07 | pauldy | I spent a little more for the gxp-2000 still not happy with how it works |
18:53.09 | spackle | netnameus: they are like training wheels for Asterisk |
18:53.10 | denon | their speakerphone isnt even usable |
18:53.54 | netnameus | ha, well i guess i'll just use an spa-1001 like i had originaly planned |
18:53.55 | dabigshiznizzle | I have been using the Snom 190s with good luck...If you don't bother the handset falling off the phone from time to time |
18:54.30 | rking | what symptoms are there for using a software zaptel emulator for iax conferences (via the meetme app)? i don't have a card, and i'm curious what buying one might do. |
18:54.53 | pauldy | netnameus, for me it was more important I had something that looked imrpessive sitting on my desk in my office than the actual functionality |
18:54.55 | pauldy | YMMV |
18:55.06 | spackle | rking - ztdummy is the no-card substitute |
18:55.20 | cpatry | is there any cheap 2 FXO ports ? |
18:55.24 | cpatry | ATA |
18:55.48 | rking | spackle: right - and i'm just wondering what benefit i'll get from an actual card. |
18:55.55 | pauldy | cpatry, for a phone or phone line to plug into |
18:56.32 | spackle | cpatry, It all depends on what you want to do. |
18:56.59 | cpatry | for a line |
18:57.16 | cpatry | pauldy: if its a phone, that would be FXS. |
18:57.20 | pauldy | haven't seen anyhing under the 100 dollar mark yet |
18:57.36 | cpatry | pauldy: which 2 ports FXO is at 100$? |
18:57.37 | pauldy | cpatry, understood making sure you did also |
18:57.51 | bjohnson | cpatry: 2 spa 3000's are about as cheap as fxo get for small installs |
18:57.54 | pauldy | no it was like 130 |
18:58.07 | pauldy | check froogle on google thats how I found it |
18:58.20 | cpatry | i dont need any FXS, just FXO, and spa3000 is 1 o , 1 s |
18:58.21 | pauldy | voip fxo |
18:58.34 | bjohnson | cpatry: 2 port fxo digium card is competitive, but not ATA as you specified |
18:58.48 | cpatry | pauldy: which model did u get? |
18:58.52 | bjohnson | cpatry: yes. you would need to buy 2 spa 3000 |
18:59.04 | cpatry | bjohnson: i would prefer another solution. |
18:59.15 | bjohnson | cpatry: then you will pay more |
18:59.20 | cpatry | pauldy: i dont want asterisk there if possible. |
18:59.30 | cpatry | file: u know any solution? |
18:59.31 | *** join/#asterisk r0d3nt (i=nobody@wsip-24-234-241-145.lv.lv.cox.net) |
18:59.55 | file[laptop] | not really |
18:59.58 | file[laptop] | :( |
19:00.00 | cpatry | i saw 4 fxo, but its like 800$ |
19:00.04 | bjohnson | exactly |
19:00.06 | blitzrage | anyone know how I can debug DTMF? For some reason I'm getting duplicate digits when calling into VoicemailMain() using RFC2833 |
19:00.16 | cpatry | bjohnson: u know something cheaper? |
19:00.18 | spackle | ....and pulls out nothing. |
19:00.32 | spackle | 8-( |
19:00.32 | bjohnson | cpatry: yes. as said, cheapest is 2 spa 3000 |
19:00.36 | azzie | blitzrage, tcpdump is your friend |
19:00.49 | blitzrage | topology is: PRI --> Asterisk 1.0.x --> billing box --> Asterisk 1.0.x (#2) --> VoicemailMain() |
19:00.53 | cpatry | bjohnson: but 4 fxo would requires 4 spa3k. |
19:00.57 | pauldy | cpatry, I didn't purchase but I came across it |
19:01.07 | bjohnson | cpatry: and would still be cheaper than $800 |
19:01.12 | blitzrage | azzie: yah - I figured I should have seen it in a SIP debug, but no go - RFC2833 go on a different port? |
19:01.16 | bjohnson | cpatry: plus .. you said you only needed 2 |
19:01.24 | cpatry | bjohnson: true. |
19:01.27 | cpatry | ive different sites. |
19:01.34 | azzie | blitzrage, rfc2833 goes inside RTP packets. You need to watch for RTP |
19:01.43 | cpatry | that would be great seeing an ATA at 2 fxo. |
19:01.45 | blitzrage | azzie: ahhh yes... I'm so dumb today |
19:01.52 | pauldy | you would be cheaper just buying two atas |
19:01.58 | blitzrage | azzie: much obliged :) |
19:02.14 | azzie | blitzrage, and get ethereal - it shows RTP nicely, you'll be able to spot digits from codec packets |
19:02.21 | cpatry | pauldy: i never saw a 2 ports FXO ata |
19:02.23 | bjohnson | cpatry: you can get multiple fxo units |
19:02.33 | spackle | four ports is common |
19:02.33 | bjohnson | cpatry: no, they are not cheaper than 2 spa 3000 |
19:02.44 | blitzrage | azzie: yah, I have both of those - guess I need to dump tcpdump into a file, then download that from the server (console only) and analyze it in ethereal |
19:02.57 | blitzrage | probably my best bet |
19:02.58 | malverian[work] | arp2, Of course not :-P Why do you ask? |
19:03.05 | azzie | blitzrage, would work... there's text ethereeal too btw :) |
19:03.24 | blitzrage | azzie: yah... seems to be a bit harder to watch though :) |
19:03.25 | *** join/#asterisk advorak (n=advorak@12-211-14-43.client.insightbb.com) |
19:03.37 | blitzrage | azzie: the gui version is niiiiiiice |
19:04.10 | cpatry | bjohnson: if if see some ATA at 2 fxos, let me know please. |
19:04.26 | azzie | blitzrage, so, you'll see RTP type 101 many times for each digit, and RTP 101 with "(end)" marker (usuall 3 times). |
19:04.53 | blitzrage | azzie: right - thanks much |
19:05.13 | pauldy | cpatry, don't forget you can also get fxs to fxo converters |
19:06.01 | *** join/#asterisk advorak (n=advorak@12-211-14-43.client.insightbb.com) |
19:06.27 | malverian[work] | pauldy, Just that in 10,000 loops through it found 4 valid running processes using $RANDOM ;) |
19:06.34 | bjohnson | 2 fxo and 2 "bonus" fxs |
19:06.58 | spackle | but wait, there's more! Buy now and get double power supplies |
19:07.04 | bjohnson | pauldy: I thought they only came in fxo to fxs versions |
19:07.21 | *** join/#asterisk IPmonger (n=ipmonger@pcp0010577106pcs.coatsv01.pa.comcast.net) |
19:07.48 | *** part/#asterisk delink (n=delink@ziegchen.delink.net) |
19:07.51 | bjohnson | and 2 cat 5 cable |
19:07.58 | pauldy | bjohnson, http://worldcall.brinkster.net/pcphoneline/fxsfxo.htm |
19:08.13 | spackle | and a coupon for 10,000 free packets. |
19:08.24 | cpatry | pauldy: ya true. thx for hints. |
19:08.33 | pauldy | malverian[work], aah well in an infinite loop it will likely kil the box pretty quick |
19:08.50 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.cust.bezeqint.net) |
19:08.53 | spackle | use the xml lathos42 |
19:09.23 | g__ | Um, interesting. It appears that queues still accept callers, even if none of the agents are logged in. |
19:09.28 | lathos42 | spackle: I tried setting it once without success, but I didnt try very hard at that point |
19:09.50 | bjohnson | pauldy: not bad for $39 |
19:10.02 | bjohnson | pauldy: kind of like a dumb 2 port fxo device |
19:10.32 | cpatry | any suggestion on good fxs->fxs converter? |
19:10.38 | pauldy | can't say I would ever use it but there it is for someone who absolutly must have it |
19:10.53 | bjohnson | cpatry: phone line extension cord from dollar store |
19:10.55 | *** join/#asterisk mkrufky (n=mk@68.160.103.77) |
19:10.59 | bjohnson | cpatry: today only $5 |
19:11.05 | bjohnson | (plus shipping and handling) |
19:11.18 | cpatry | huh! |
19:11.42 | bjohnson | fxs to fxs converter? |
19:12.01 | cpatry | fxs->fxo converter, sorry boi :) |
19:12.02 | tzanger | bjohnson: called a piece of telephone wire :-) |
19:12.55 | bjohnson | tzanger: with a lump of plastic attached for "marketing" purposes |
19:13.26 | tzanger | :-) |
19:13.39 | bjohnson | patented plastic of course |
19:13.39 | pauldy | to many acronyms tla here tla there I got an fla for that that describes little or nothing of what the actual functionality is |
19:13.58 | bjohnson | don't need any competition to come between me and recovering my R&D costs |
19:14.51 | *** join/#asterisk bumblefsck (n=bumblefs@69-160-158-193.ontrca.adelphia.net) |
19:15.25 | *** join/#asterisk mrlovatt (n=fgd@62-30-33-29.cable.ubr01.pres.blueyonder.co.uk) |
19:15.27 | *** join/#asterisk hellagony (n=egutierr@200.202.206.247) |
19:15.29 | emp | what would cause overlap in a call recording between two parties? |
19:16.43 | cpatry | bjohnson: huh? dollar-store? |
19:17.20 | emp | cpatry, a store where everything they sell is $1 |
19:17.55 | Beirdo | or $2 |
19:18.02 | Beirdo | (these days) |
19:18.31 | spackle | Beirdo - It's getting to be you can't afford to change your mind anymore |
19:18.36 | Beirdo | yeah |
19:18.37 | Druken | dollar stores rule! |
19:18.47 | *** join/#asterisk ClubBarf (n=spamme@host-87-74-0-72.bulldogdsl.com) |
19:18.53 | Beirdo | dollar stores are great ways to buy CHHEEEEEEP crap |
19:18.53 | ClubBarf | Hey ppl! |
19:18.54 | Beirdo | :) |
19:19.06 | mutilator | yea.. crap |
19:19.19 | Druken | Beirdo: dollar stores are a great place to buy shit you don't care if it gets broken |
19:19.28 | Beirdo | bingo |
19:19.44 | mutilator | heh i care if my dollar store items are broken |
19:19.47 | mutilator | means i have to spend more money |
19:19.48 | ClubBarf | Anyone here know anything about connecting INDeX telephones to *? |
19:19.50 | Druken | i buy all my glasses and the like there |
19:20.00 | Druken | 3 cups for a buck.. how can ya go wrong? |
19:20.12 | mutilator | break all 3 cups |
19:21.34 | mutilator | omg |
19:21.35 | ClubBarf | I guess that's a no then... |
19:21.36 | ClubBarf | :p |
19:21.44 | mutilator | i just got a job offer for a 75k/yr job outta the blue today |
19:21.54 | mutilator | so i emailed the guy back askin where he got my info |
19:21.57 | mutilator | and i get the reply.. |
19:22.04 | mutilator | you had been in touch with us in '04 about a prospect with Pfizer...but this was sent in error...there is a candidate with a similar name in the NYC area...It was meant for him and his skill set...I apologize... |
19:22.04 | mutilator | <PROTECTED> |
19:22.05 | mrlovatt | any1 know a good cheap voip provider in the uk.. or how to use skype as a channel on a astersik pbx? |
19:22.34 | spackle | mutilator: look on the bright side: |
19:22.50 | *** join/#asterisk konfuzed (n=konfuzed@H129.C72.B0.tor.eicat.ca) |
19:22.51 | ClubBarf | mrlovatt - I use pipecall. They do nice, cheap 0845 numbers and have unlimited call accounts. |
19:22.52 | spackle | he didn't read about you in "ass-whores daily" |
19:23.06 | Druken | mutilator: ya stupid ass, i would have just said yes :) |
19:23.09 | mutilator | yea, i use my other email address for that ;) |
19:23.15 | spackle | lol |
19:23.21 | mutilator | Druken: yea. i also mentioned that in the email too |
19:23.32 | ClubBarf | And you can't use skype as a channel on asterisk, skype won't connect to any other VoIP systems. |
19:23.32 | mrlovatt | thanks i'll look into them |
19:23.34 | ClubBarf | Yet. |
19:23.45 | mutilator | sorry about the confusion..thanks for bringing it to my attention... |
19:23.45 | mutilator | Regards, |
19:23.45 | mutilator | Joe |
19:23.48 | Druken | mutilator: oh... well that sucks... i'd be pissed :) |
19:23.51 | }btorch{ | are there any other free Voip phones better than X-Lite ? |
19:23.57 | mrlovatt | no1's created a cheeky little hack yet :P? |
19:23.58 | ClubBarf | Oh, and I hear Dixons is going to do a supercheap VoIP service too, but I don't know any more about that. |
19:24.00 | mutilator | i think this "guy with a similar name" in nyc owes me big time |
19:24.08 | mutilator | i coulda just got him a nice job |
19:24.32 | Druken | i say ya go on a manhun for him... |
19:24.39 | ClubBarf | manhun? |
19:24.40 | mutilator | heh |
19:24.40 | Druken | lets say 20g a year? |
19:24.43 | ClubBarf | :p |
19:24.48 | Druken | manhunt |
19:24.56 | ClubBarf | I liked manhun better. |
19:24.58 | mutilator | i could do with an extra 5g a year |
19:25.05 | *** join/#asterisk hellagony (n=egutierr@224-123.81-161.gts.tkb.net.pl) |
19:25.13 | Druken | well, then he's still getting 70 :) |
19:25.20 | mutilator | gives ya more of a comfort level over my 15k i currently make |
19:25.23 | ClubBarf | Kinda like a german WW1 pilot's name. |
19:25.31 | *** part/#asterisk trig (n=jb@xob.neospire.net) |
19:25.42 | Druken | mutilator: i claimed... 11.5k last year |
19:26.02 | mutilator | yeh i had almost 13 |
19:26.17 | mutilator | yea buddy |
19:26.24 | bjohnson | mrlovatt: don't count on it. skype isn't interested in connecting to other systems and other systems aren't interested in connecting to them |
19:26.24 | ClubBarf | "Und zis is ze ace pilot Manhun, unt he vill be szchuutink down ze englander pigdogs" |
19:26.47 | mutilator | manthunt england pipedogs? |
19:27.17 | spackle | Manhun von rickitybritches |
19:27.25 | mutilator | .. um just by the off chance.. |
19:27.34 | mutilator | anyone in here ever built an accounting system to do financing? |
19:27.44 | ClubBarf | ja, das ist him, ja! |
19:27.45 | mutilator | and billing |
19:28.04 | Druken | doubtfull |
19:28.24 | ClubBarf | Anyone here connected INDeX systems to *? |
19:28.43 | mutilator | well i gotta integrate financing into my current system, and well it doesn't blend together very well |
19:28.49 | spackle | anyone here ever go outside during the day in summertime? |
19:28.51 | mutilator | wondering how others might have built theirs |
19:29.14 | Druken | spackle: you mean, enter the bright light? |
19:29.24 | bumblefsck | mutilator, define billing/financing. are you talking double entry? |
19:29.24 | spackle | it has chosen us. |
19:29.34 | spackle | the big blue room |
19:29.42 | cpatry | bjohnson: and whats the relation of the dollard-store with the fxs->fxo converter? |
19:30.03 | bjohnson | no idea |
19:30.03 | spackle | dullard store - I like that. |
19:30.10 | bjohnson | but they have lots of fxs-fxs converters |
19:30.31 | bjohnson | AND fxo-fxo converters |
19:30.31 | cpatry | fxs-fxs :P |
19:30.40 | ClubBarf | What's an ISDN FXS called? Is it still an FXS? |
19:30.45 | cpatry | i need fxo->fxs |
19:30.47 | mutilator | well we sell the customer their hardware at say $900, then they pay $20.50/month for 18 months or something until it's paid off |
19:30.50 | *** join/#asterisk eziman (i=superop@64.116.231.226) |
19:31.11 | eziman | asterisk:/lib/modules/2.4.31-1-686/misc# modprobe ztdummy |
19:31.11 | eziman | . /lib/modules/2.4.31-1-686/misc/zaptel.o: /lib/modules/2.4.31-1-686/misc/zaptel.o: unresolved symbol __stack_smash_handler |
19:31.23 | mutilator | problem i run into, that initial 900 is already billed into our AR and i can't add the 20.50 that shows on their bill to the AR each month, only the interest thats incorporated into the 20.50 can be added to AR |
19:31.24 | eziman | ideas for solving that ? |
19:31.40 | mutilator | so i need to invoice for the interest but still show they need to pay their 20.50 |
19:31.53 | bjohnson | ahhh .. the stack'em and smash'em handler |
19:32.22 | bjohnson | I have a son that does that |
19:32.44 | bjohnson | also a mailman that does that |
19:33.31 | eziman | so? any fix for that ? |
19:34.09 | *** join/#asterisk glm2k (n=GLM@rrcs-24-199-11-41.west.biz.rr.com) |
19:34.15 | bendy24 | duct tape or some rope may help |
19:34.22 | Katty | mew. |
19:34.27 | psypete | rawr |
19:34.32 | Katty | ... |
19:34.37 | psypete | so my company has an old shitty key system |
19:34.41 | psypete | with lotsa phones |
19:34.46 | Katty | twisted[asteria]: wake up! |
19:34.48 | bendy24 | bjohnson: although your son may squirm his way out of that |
19:34.50 | psypete | is there a place i can sell my phone system to? |
19:34.52 | Katty | twisted[asteria]: no napping! |
19:35.04 | bjohnson | psypete: ebay? |
19:35.07 | psypete | so i can then afford the PRI cards for Asterisk |
19:35.14 | psypete | bjohnson: well besides them |
19:35.30 | bjohnson | you might want to set up the new system before you remove and sell the old one |
19:35.36 | psypete | and i'd like to make a decent amount for it |
19:35.39 | rking | bjohnson++ # thinking like a pro |
19:35.42 | psypete | yeah i was kinda thinking the same thing bjohnson |
19:36.04 | psypete | my coworkers might have a word or two with me if the phones are gone for a month |
19:36.12 | Beirdo | otherwise.... print resume first |
19:36.13 | bjohnson | psypete: no decent amount to be had for used key systems |
19:36.16 | Beirdo | then sell phone system |
19:36.19 | psypete | "where did the phones go pete?" |
19:36.27 | psypete | bjohnson: frowny-face |
19:36.35 | rking | emotitext? |
19:36.36 | bjohnson | or in other-words .. you can only get what they're worth |
19:36.53 | psypete | well think 50 phones |
19:37.04 | spackle | or what someone thinks they are worth. |
19:37.07 | psypete | and i guess i could sell the patch panels too... |
19:37.13 | bjohnson | psypete: many people are interfacing * to their key systems |
19:37.16 | *** join/#asterisk Nix (n=Nix@81.213.125.220) |
19:37.33 | psypete | bjohnson: our key system is a bit limited in functionality |
19:37.34 | bjohnson | then use the same phones and existing wiring |
19:37.37 | spackle | psypete so what is the existing system? Brands types and models? |
19:37.37 | mutilator | bumblefsck ideas? |
19:37.46 | *** part/#asterisk pigpen (n=mark@fw.seamans.cc) |
19:37.54 | psypete | spackle: NEC EliteSomething 192 |
19:37.54 | bumblefsck | mutilator, honestly, I think you'd be better off writing code to interface with an existing accounting package. |
19:38.07 | bumblefsck | what do you use to do your accounting now? |
19:38.20 | mutilator | my code |
19:38.25 | mutilator | been using it 2 yrs or so |
19:38.34 | bjohnson | mutilator: accounting packages are complicated. make an interface to an existing system |
19:38.38 | bjohnson | look at sql-ledger |
19:38.56 | psypete | and BTW, a SIP phone, does it give you the more advanced options of other phones, like menus and stuff? |
19:39.11 | bjohnson | psypete: if you pay enough |
19:39.14 | mutilator | current system has been workin great |
19:39.17 | psypete | :| |
19:39.17 | ClubBarf | psypete - depends on the phone. |
19:39.19 | *** part/#asterisk IPmonger (n=ipmonger@pcp0010577106pcs.coatsv01.pa.comcast.net) |
19:39.20 | mutilator | just financing doesn't fit well into it |
19:39.25 | mutilator | i guess i can check out options |
19:39.31 | psypete | well like |
19:39.35 | ClubBarf | I have a basic SIP phone that is nothing more than a very basic phone. |
19:39.49 | psypete | i want a cordless phone i can take multiple calls on and transfer to different extensions |
19:39.55 | Katty | <coworker> zomgpolycomsarecrapcausethere'sechoonthesecalls |
19:39.56 | ClubBarf | I also have a Polycom IP300 on my desk that'll do everything bar the washing up. |
19:40.00 | psypete | and i want it to work with asterisk |
19:40.10 | bjohnson | Katty: shout in their ear and point out the "echo" |
19:40.12 | Katty | <coworker> zomgciscophonesatOTHER$companydon'tdothat. lolzkthxyousuck |
19:40.22 | spackle | and smell pretty in the springtime |
19:40.27 | ClubBarf | Psypete - a multi-line capable SIP phone that uses 802.11b? |
19:40.46 | psypete | Katty: i think you messed up your variable |
19:40.46 | bjohnson | psypete: multiple calls at the same time, or one at a time? |
19:41.03 | *** join/#asterisk devel (n=devel@wiggum.digitalcoven.com) |
19:41.17 | Katty | psypete: go away. |
19:41.31 | psypete | don't hate me because you suck at coding :) |
19:41.33 | spackle | I hear an echo. |
19:41.41 | Katty | psypete: don't bitch at me because you don't get the language i'm using. |
19:41.49 | *** part/#asterisk eziman (i=superop@64.116.231.226) |
19:41.51 | psypete | bjohnson: well like... for a receptionist |
19:41.54 | *** join/#asterisk mitcheloc (n=mitchelo@69-169-28-46.anhmca.adelphia.net) |
19:42.11 | bjohnson | psypete: just get an ATA and use a regular cordless phone |
19:42.12 | Katty | why does everyone presume i'm dumb? |
19:42.15 | psypete | why language differentiates between $company in a string "$companydon'tdothat" ? |
19:42.16 | Katty | that really pisses me off sometimes. |
19:42.23 | ClubBarf | Why would you want a cordless phone for a receptionist? |
19:42.33 | psypete | i don't presume you're dumb, you just did a dumb thing |
19:42.38 | bjohnson | psypete: or a SIP phone that can take a headset (BT or otherwise) |
19:42.43 | makhtar | you know what they say about presumptions |
19:42.50 | spackle | ClubBarf - she can fly so she's away from the phone a lot |
19:43.08 | ClubBarf | Ah, he's got supergirl for a receptionist? |
19:43.11 | psypete | bjohnson: but i need to like, be able to park calls and call up certain lines again, access voice mail via the buttons like on our existing phones, etc |
19:43.15 | bjohnson | me wonders wtf psypete is talking about |
19:43.26 | Katty | psypete: i'm trying VERY hard not to be rude to you |
19:43.30 | bjohnson | oops .. forgot the slash |
19:43.31 | psypete | theres a particular provider which basically provides this all-in-one managed service i've been looking at |
19:43.43 | psypete | it's expensive but its cool |
19:43.48 | Katty | psypete: i'm going to POLITELY suggest you drop the topic and move on. |
19:44.03 | ClubBarf | Katty - I think he has... |
19:44.11 | Katty | excellent. |
19:44.14 | Katty | NEXT |
19:44.23 | ClubBarf | psypete - Why do you want the phone to be cordless? |
19:44.24 | psypete | for instance you take your phone to your house, plug it into your router and you're connected to the office pbx |
19:44.39 | bjohnson | psypete: ANY phone can do all that .. phone model justs control whether there is a special button for it or not |
19:44.43 | psypete | i want it to be cordless because our dumb receptionist keeps leaving her desk and we never get any calls |
19:44.52 | bjohnson | psypete: and a cordless phone won't have ALL the buttons |
19:44.59 | mitcheloc | lol |
19:45.11 | psypete | bjohnson: right, i'd rather there be special buttons for certain features, and have those buttons supported with asterisk or whatever |
19:45.17 | bjohnson | psypete: get a BT headset |
19:45.37 | ClubBarf | I think you're looking at a Cisco 7920. |
19:45.39 | psypete | err... once i get her the headset, what's it gonna connect to? |
19:45.51 | bjohnson | psypete: a decent desk phone |
19:45.52 | spackle | hmmm, technology to fix a personnel issue. |
19:46.04 | psypete | a bluetooth headset? |
19:46.10 | bjohnson | spackle: it's been done since the beginning of history |
19:46.12 | psypete | that won't reach to her desk when she leaves it |
19:46.21 | spackle | psypete, where is all the moeny for this coming from again? |
19:46.24 | bjohnson | psypete: how far is she going? |
19:46.42 | bjohnson | psypete: maybe you need a cellular network |
19:46.42 | psypete | bjohnson: say 100-200 feet |
19:47.03 | psypete | BT typically only reaches like a couple meters |
19:47.16 | bjohnson | psypete: how many button are you picturing on this wifi cordless phone? |
19:47.17 | psypete | not to mention has no special buttons |
19:47.28 | mitcheloc | Can someone help me figure this out....I need to collaborate with an associate in Russia and his ISP blocks VOIP calls.....is there anyway to change up the ports or something? |
19:47.29 | psypete | bjohnson: 14? |
19:47.35 | ClubBarf | psypete Get an 802.11b/g access point and a Cisco 7920. |
19:47.40 | spackle | how about a leash? That's a kid of technology, isn't it? |
19:47.56 | bjohnson | spackle: I was thinking a club |
19:48.01 | ClubBarf | Secretary on a leash? Spackle, how S&M of you. |
19:48.14 | mutilator | any other systems you recommend i should look at other than sql-ledger? |
19:48.25 | CoffeeIV_ | get a cordless phone with a belt clip and headset, and teach her to park/transfer without specail buttons |
19:48.26 | spackle | I thought it sounded practical. Maybe hobbles instead? |
19:48.30 | bjohnson | mutilator: there are thousands of acounting systems out there |
19:48.37 | mutilator | exactly |
19:48.45 | mutilator | "you recommend |
19:48.46 | mutilator | " |
19:48.53 | ClubBarf | Spackle, maybe just spank her alot. |
19:48.57 | bjohnson | totally depends on your needs |
19:49.06 | bjohnson | spankle? |
19:49.20 | ClubBarf | Yup, that'll do. |
19:50.05 | [ViRii] | whats this feature for asterisk E911? |
19:50.22 | psypete | well if i'm going to use a standard cordless phone, is the calling party gonna hear the receptionist typing in the transfer code or whatever? i don't want it to be like a hack job or something if possible |
19:50.40 | ClubBarf | I still think 802.11x and a wifi voip phone would suit psypete's needs best. |
19:50.58 | bjohnson | psypete: compared to the existing system where the receptionist doesn't answer the phone? |
19:51.21 | psypete | i think my boss would rather a clean solution or none at all |
19:51.29 | bjohnson | psypete: replace her with an ivr |
19:51.32 | psypete | so far it's just me pissed that my calls aren't getting transferred in |
19:51.35 | psypete | eww |
19:51.43 | dabigshiznizzle | Has anyone had any luck getting the vmail.cgi to work? I can get my and listen to my messages, but if I try and delete or move them I get a Software error: Invalid Context....Any suggestions? |
19:51.54 | bjohnson | ClubBarf: depends on his budget |
19:51.58 | psypete | ClubBarf i've already responded to them |
19:52.03 | hypa7ia | bjohnson: "leave me alone or i will replace you with a very simple IVR" |
19:52.26 | [ViRii] | whats the difference between alsa and oss? |
19:52.31 | bjohnson | an engineering buddy once told a technician that she could be replaced with a spreadsheet |
19:52.33 | ClubBarf | No you havn't. |
19:52.36 | psypete | one sucks, the other blows, [ViRii] |
19:52.50 | [ViRii] | psypete: haha |
19:53.16 | psypete | one is new the other is old |
19:53.20 | Katty | why do incoming analog calls have echo? |
19:53.21 | psypete | one is complicated one is simple |
19:53.27 | Katty | and not outgoing analog calls? |
19:53.38 | Katty | like that 2 second echo bit |
19:53.41 | psypete | the incoming calls are making fun of you |
19:53.41 | Katty | before asterisk chops it off |
19:53.51 | Katty | Hmmhesays: explain this to me. |
19:53.58 | ClubBarf | Anyone here know anything about INDeX systems? |
19:54.04 | *** join/#asterisk SplasPood (n=jwb@dsl081-201-143.nyc2.dsl.speakeasy.net) |
19:54.22 | bjohnson | Katty: do you have echotraining on? |
19:54.29 | Katty | bjohnson: this is NOT an issue |
19:54.30 | mutilator | i'll have to use that sometime bjohnson |
19:54.31 | Katty | bjohnson: i want to know why |
19:54.40 | psypete | hmm... this cisco 7920 cordless phone says it supports XML applications to the display? is this something Asterisk supports? |
19:54.52 | ClubBarf | No idea. |
19:54.58 | ClubBarf | Do you think you'd be using that? |
19:54.58 | bjohnson | Katty: well IF you have echotraining on, it takes a few seconds to process it |
19:55.03 | [ViRii] | my polycom 600 supports xml |
19:55.07 | Katty | bjohnson: i don't care about processing it |
19:55.11 | Katty | bjohnson: i want to know /why/ there is echo |
19:55.13 | mitcheloc | No, it's something you program yourself ;). |
19:55.14 | mutilator | the support/sales team "watch out, i can replace you with a simple IVR" |
19:55.23 | mitcheloc | Just point it to a webserver and start serving up xml instead of html. |
19:55.27 | bjohnson | Katty: to much volume from the remote end |
19:55.32 | mitcheloc | (to psypete) |
19:55.33 | bjohnson | too |
19:55.41 | psypete | ClubBarf: the XML applications could do the more advanced feature crap i want though, right? |
19:55.49 | bjohnson | your signal is bouncing back |
19:56.04 | ClubBarf | ViRii - I've been sent a Polycom IP300. Dear god, have you ever come across a config file quite that complex? |
19:56.05 | spackle | Katty: there are many causes of echo. |
19:56.09 | Katty | spackle: elaborate. |
19:56.20 | bjohnson | typically playing with gain will help |
19:56.22 | ClubBarf | Psypete - I don't think XML is the solution you're looking for, no. |
19:56.25 | psypete | mitcheloc: oh it's HTTP based? |
19:56.33 | bjohnson | and turn echotraining off |
19:56.33 | Katty | spackle: from what i understand, there should be no echo |
19:56.38 | spackle | Katty: high latency can cause echo, conversion from analog to digital can cause echo. |
19:56.42 | Katty | spackle: if pots lines terminate at the 21x before they go to a client....... |
19:56.53 | Katty | spackle: then why is there 2 seconds of echo? |
19:57.02 | ClubBarf | psypete - I think you're looking for a phone that can handle more than 1 line, which isn't a terrably hard thing to do. |
19:57.19 | psypete | yeah but not multiple analog lines |
19:57.19 | bjohnson | psypete: and typically not needed |
19:57.37 | ClubBarf | I think you're looking for the Cisco 7920. |
19:57.44 | ClubBarf | I would email cisco about it. |
19:57.49 | spackle | katty, what end is the echo on? |
19:57.52 | psypete | well with our existing key system we have four buttons at the top of an LCD display which act as an interactive menu, which Cisco phones support iirc |
19:58.00 | Katty | spackle: when we get a call INTO the building |
19:58.08 | bjohnson | psypete: a one line phone can easily have asterisk pick the first line that is available = multi line phone |
19:58.09 | ClubBarf | zyxel also do a wifi sip phone, but I don't know if it supports multiple lines. |
19:58.11 | psypete | i'd like to emulate that sort of function so people don't have to listen to options |
19:58.25 | spackle | and the echo is on your end or the other end? |
19:58.34 | Katty | spackle: our end |
19:58.47 | ClubBarf | Ah, well Cisco phones let you remap buttons, but I don't know if you have enough spare buttons on a 7920... |
19:58.48 | Katty | spackle: echo cancelation kicks in in about 2 seconds and it's fine |
19:58.54 | Katty | spackle: i just want to know /why/ it takes 2 seconds. |
19:59.05 | spackle | katty, it has to train. |
19:59.06 | bjohnson | that's the "training" part |
19:59.08 | psypete | maybe i could build a wifi phone with Gumstix and some 802.11b chippy thingy... |
19:59.10 | ClubBarf | How many people here use a soundpoint phone? |
19:59.11 | Katty | spackle: train? |
19:59.13 | Hmmhesays | <PROTECTED> |
19:59.16 | Katty | Hmmhesays: me |
19:59.20 | bjohnson | learning |
19:59.23 | Hmmhesays | whats up? |
19:59.23 | spackle | it samples the echo and begins to correct it and tunes until it is canceled. |
19:59.32 | Katty | Hmmhesays: i'm attempting to understadn why i have echo on these lines |
19:59.35 | Katty | Hmmhesays: 2 seconds. |
19:59.39 | ClubBarf | Yo! Who's using a Polycom Soundpoint IPxxx phone? |
19:59.56 | Katty | Hmmhesays: and i don't want the answer, oh enable echo cancelation |
20:00.02 | Katty | Hmmhesays: there is a logical reason for echo |
20:00.03 | Hmmhesays | i was just reading the speech the edonkey guy gave in a senate testimony.... he blasted them |
20:00.09 | ClubBarf | lathos42 - do you know if there's any apps out there that can front-end the config files? |
20:00.09 | bjohnson | DISABLE |
20:00.29 | bjohnson | Katty: it is your signal bouncing off the far end |
20:00.34 | psypete | Katty: why not google for reasons that echo is introduced, as i'm sure many people have had this problem before and there must be technical explinations of how the stuff works |
20:00.35 | ClubBarf | I've been sent an IP300 for review, and the config (at first glance) is a fkn NIGHTMARE. |
20:00.40 | bjohnson | turn off echotraining and adjust the gain until the echo is gone |
20:01.02 | *** join/#asterisk darkskiez (n=darkskie@host86-132-168-185.range86-132.btcentralplus.com) |
20:01.04 | spackle | ClubBarf: it really makes a lot of sense once you get used to it. I used to feel the same way. |
20:01.06 | psypete | bjohnson: she doesn't want an answer, she wants a reason |
20:01.11 | *** join/#asterisk TheCops (n=dump@206-248-136-187.dsl.teksavvy.com) |
20:01.15 | bjohnson | she's had it |
20:01.16 | lathos42 | ClubBarf: I vaguely recall a script that will configure the phone, but I can't remember where its at |
20:01.20 | bjohnson | and the solution |
20:01.22 | TheCops | X-Rob, are you busy ? |
20:01.26 | psypete | well then she just can't be satisfied |
20:01.28 | lathos42 | spackle: Have you ever tried to remap one of the keys? |
20:01.29 | Katty | psypete: indeed. and bjohnson gave me plenty of reasons. |
20:01.33 | psypete | LIKE MOST WOMEN ROFL |
20:01.37 | *** join/#asterisk Raceman (n=bla@cust-02-E169.adsl.scarlet.nl) |
20:01.46 | Katty | psypete: wow, you're incredibly annoying (= |
20:01.49 | ClubBarf | spackle - No, I get why they did it that way - if I ever have to config 300 phones, they'll be sounpoints. But for a single phone, it's a damn nightmare. |
20:01.54 | psypete | Thanks! |
20:01.59 | Katty | my pleasure. |
20:02.04 | ClubBarf | soundpoints, even... |
20:02.14 | Hmmhesays | echo is usually caused by external impedance imbalances |
20:02.19 | Hmmhesays | now there is a lot of things that can cause that |
20:02.19 | *** join/#asterisk file[laptop] (n=jcolp@mctnnbsa31w-142166116178.nb.aliant.net) |
20:02.19 | spackle | lathos42: no, I messed with the directory some, haven't hit on the need yet. |
20:02.30 | Katty | Hmmhesays: how much echo is normal? |
20:02.30 | spackle | define external impedance imbalances |
20:02.33 | Katty | Hmmhesays: 2 seconds? |
20:02.39 | Katty | Hmmhesays: .1 seconds? |
20:02.41 | psypete | humidity? |
20:02.43 | Hmmhesays | i don't think there is a "normal" |
20:02.55 | ClubBarf | different is normal. |
20:02.56 | bjohnson | you're not talking about an echo 2 seconds later right .. just for te first 2 seconds of the call |
20:02.58 | lathos42 | spackle: Ahh.. I'm trying to change the useless "services" button into a call pickup button, but the phone isnt cooperating |
20:03.07 | Katty | bjohnson: indeed |
20:03.44 | *** part/#asterisk dabigshiznizzle (n=dabigshi@gwsecure.sctelcom.com) |
20:03.57 | Hmmhesays | check out the wikipedia for the definition of impedance and it will make sense |
20:03.58 | *** join/#asterisk dabigshiznizzle (n=dabigshi@gwsecure.sctelcom.com) |
20:04.00 | bjohnson | or 6 |
20:04.03 | ClubBarf | I just wish Polycom would use FTPS, SCP or something equally secure for EVERYTHING, not just config files on the 301, 501 etc. |
20:04.13 | Katty | i'm fairly happy with the answer i got. |
20:04.24 | Hmmhesays | k |
20:04.31 | spackle | katty: summarize please |
20:04.33 | Katty | it's just lolzenableechocancelation i don't want. |
20:04.39 | Katty | spackle: shan't. |
20:04.48 | bjohnson | DISABLE |
20:04.53 | bjohnson | notenable |
20:04.54 | psypete | omgkattythatsprettyannoyingafterthe10thtimemmkay |
20:04.57 | Hmmhesays | impedance is fun to read about, it makes my mind warp |
20:05.05 | Katty | psypete: omgwtfbbqlolzkthxbi |
20:05.13 | fugitivo | Hmmhesays: docs |
20:05.17 | psypete | gay |
20:05.19 | spackle | Hmmhesays: isn't there a pill for that? |
20:05.23 | Hmmhesays | fugitivo what? |
20:05.32 | Hmmhesays | spackle, LSD? |
20:05.35 | bjohnson | we've been learning the alphabet at my house using the abc song. Don't make me sing it here |
20:05.46 | spackle | Hmmhesays - no that's for BSD |
20:05.53 | Hmmhesays | X? |
20:05.59 | bjohnson | scared him away |
20:06.23 | psypete | happy belated birthday |
20:06.28 | ClubBarf | Or so some people seem to think... |
20:06.49 | psypete | now on to my next question |
20:06.51 | Hmmhesays | "i was born at night but not last night baby" - Kid rock |
20:06.53 | psypete | you guys will love this one |
20:07.05 | psypete | How do I spoof Caller ID? |
20:07.08 | lathos42 | ClubBarf: I've got some land in Antartica to sell you.. Beautiful Sunny beaches as far as the eye can see |
20:07.22 | bjohnson | Hmmhesays: I don't think he originated that |
20:07.23 | Hmmhesays | www.engrish.com <-- very funny today |
20:07.42 | Hmmhesays | bjohnson no, but he did make redneck music popular |
20:08.14 | bjohnson | psypete: use a PRI or a voip provider and set the CID info in your extensions.conf |
20:08.24 | psypete | Hmmhesays: kid rock popularized Country? |
20:08.31 | spackle | lathos42: there is probably some other silly switch or option you have to set to yes or no before it will all take effect. |
20:08.34 | bjohnson | no, just redneck |
20:08.42 | Hmmhesays | he created a whole new genre |
20:08.51 | psypete | wow... i didn't think i would get a serious answer from that, bjohnson |
20:08.58 | spackle | I though ashton Kutcher did that. |
20:08.59 | psypete | way to kill the joke :( |
20:09.09 | Hmmhesays | band practice tonight |
20:09.12 | Hmmhesays | woot |
20:09.31 | Hmmhesays | i get to bust out mas tequila |
20:09.44 | spackle | and inna gada da vita? |
20:10.01 | lathos42 | spackle: I'll have to read the guide again.. I'd contact Polycom, but I have this feeling they wouldnt be a whole lot of help |
20:10.46 | spackle | lathos42 - I dunno, they have an agreement with Digium now. Besides, its a phone feature not a pbx feature. |
20:11.20 | spackle | lathos42 - Probably have to get support from the reseller |
20:12.05 | Hmmhesays | i need to quit smoking |
20:12.10 | Hmmhesays | like today |
20:12.25 | lathos42 | spackle: I guess the worst they could tell me is No |
20:12.39 | bjohnson | Hmmhesays: tomorrow is likely soon enough |
20:12.51 | spackle | is it from tritechcoa or somewhere else? |
20:12.53 | Hmmhesays | bjohnson: bad |
20:13.26 | spackle | Hmmhesays: why quit, each one takes off the worst three minutes of your life. |
20:13.32 | bjohnson | ClubBarf: not directly |
20:13.39 | hypa7ia | ClubBarf: you need some third-party hardware to do that |
20:13.43 | Hmmhesays | spackle cause its making me sick and giving me lung problems |
20:13.45 | MikeJ[Laptop] | ClubBarf, yes, via bri cards |
20:14.00 | MikeJ[Laptop] | although I do not know specifically with those phones |
20:14.03 | lathos42 | spackle: This one is from tritechcoa, so i'm not sure what kind of support they have |
20:14.13 | ClubBarf | MikeJ - do you know of any BRI cards that can handle more than 1 phone per card? |
20:14.21 | MikeJ[Laptop] | ummm |
20:14.24 | MikeJ[Laptop] | there are several |
20:14.29 | ClubBarf | Otherwise it would be cheaper to replace all the phones with sip phones. |
20:14.35 | MikeJ[Laptop] | I think there are quad cards supported by midsn |
20:14.42 | MikeJ[Laptop] | and then there are the other ones.. hmmm |
20:14.45 | MikeJ[Laptop] | let me find them |
20:14.53 | bjohnson | ClubBarf: are they already connected to a pbx or key system? that's where most people try to connect * |
20:14.57 | ClubBarf | I was thinking a backplane server filled with quad ISDN cards, but I can't find any quad ISDN cards that are supported... |
20:15.15 | ClubBarf | bjohnson - yeah, they're live atm. |
20:15.18 | *** join/#asterisk VxJasonxV (n=jason@unaffiliated/VxJasonxV) |
20:15.33 | *** join/#asterisk fiber0pti (n=johndoe@pcp01876618pcs.sandia01.nm.comcast.net) |
20:15.37 | ClubBarf | They're connected to an Avaya INDeX PBX at the college I just started working for. |
20:15.51 | bjohnson | ClubBarf: consider planting * between the pbx and the telco |
20:16.02 | MikeJ[Laptop] | http://www.voip-info.org/tiki-index.php?page=Asterisk+Hardware |
20:16.05 | MikeJ[Laptop] | there is a list there |
20:16.05 | spackle | MikeJ, earilier sombody was asking if asterisk supported ADSI through an Adit channel bank for Aastra phones. Any idea? |
20:16.34 | MikeJ[Laptop] | ummm |
20:16.34 | bjohnson | I don't think so |
20:16.38 | ClubBarf | Well, I'm trying to find out if the hardware the phones plug into could be plugged into an * box directly. |
20:16.47 | MikeJ[Laptop] | spackle, adsi should work through a channel bank |
20:16.54 | ClubBarf | But the more I look into it, the more it seems that getting sip phones would be cheaper. |
20:16.57 | MikeJ[Laptop] | but I have never used one of those personally |
20:17.08 | bjohnson | really? for those analog AASTRAs with the big screens? |
20:17.19 | spackle | MikeJ, digium used to sell special Aastra phones, but I don't see that they do any more. |
20:17.19 | MikeJ[Laptop] | there are 8 port bri cards too |
20:17.24 | bjohnson | ClubBarf: depends on how much wiring is needed |
20:17.42 | MikeJ[Laptop] | spackle, not sure about digium selling them, but they are still out there |
20:18.00 | bjohnson | AASTRA also make real VOIP phones |
20:18.03 | hypa7ia | i've seen some channel-bank like things that suport old nortel and vaya isdn phones |
20:18.11 | hypa7ia | but i can't find them :-( |
20:18.31 | bjohnson | hypa7ia: yeah .. but they're like $100 per port. |
20:18.46 | TheCops | Someone is using allpage.agi script by X-Rob ? |
20:19.05 | *** join/#asterisk fri (n=fri@port84.ds1-sdb.adsl.cybercity.dk) |
20:19.06 | *** join/#asterisk raceman (n=bla@cust-02-E169.adsl.scarlet.nl) |
20:19.17 | hypa7ia | bjohnson: true |
20:19.43 | ClubBarf | bjohnson - that's the problem, I can't find the technical docs on the kit we have in place atm. |
20:19.46 | bjohnson | usually better to buy new phones |
20:19.53 | ClubBarf | I think that's probably true. |
20:20.12 | ClubBarf | Which is why Polycom have sent me an IP300. |
20:20.19 | hypa7ia | nice :-) |
20:20.22 | bjohnson | ClubBarf: yeah .. typical telco problem. Sell you the hardware then sell you partial docs .. nobody knows real specs |
20:20.39 | ClubBarf | And Aastra, Grandstream etc are sending me samples too. |
20:20.41 | hypa7ia | my boss was going to throw it out! |
20:21.04 | *** join/#asterisk ^X-works (n=r0x0r@host34-3.pool871.interbusiness.it) |
20:21.06 | ClubBarf | Well, I like the idea of going VoIP all the way, with POE etc. |
20:21.32 | *** join/#asterisk dalfry (n=dalfry@gateway.ishisystems.com) |
20:21.37 | hypa7ia | PoE is nice |
20:21.44 | darkskiez | ClubBarf: what did you say to get samples? |
20:21.46 | bjohnson | hypa7ia: I thought they weren't very standards compiant and hard to get working |
20:21.58 | hypa7ia | bjohnson: that would be exactly correct |
20:22.03 | bjohnson | "darkskiez: money is no object" |
20:22.11 | *** join/#asterisk IPmonger (n=ipmonger@pcp0010577106pcs.coatsv01.pa.comcast.net) |
20:22.16 | hypa7ia | ClubBarf: look at the Adtran PoE switches. they are snazzy. |
20:22.18 | ClubBarf | I just asked a reseller to send me samples, because we're likely to need about 300 phones soon. |
20:22.36 | darkskiez | Ah, I may need to do a deployment with 1500 |
20:22.37 | ClubBarf | They're not letting me keep them, but it's nice to compare phones side by side. |
20:23.05 | ClubBarf | darkskiez - tell a local reseller that. They'll bend over backwards for ya. |
20:23.14 | darkskiez | to begin with |
20:23.26 | ClubBarf | Just don't promise them anything other than to return the phones in good working order. |
20:23.53 | raceman | hi all. Anyone experience with the SendUrl() command in extensions.conf ? I tryed it with idefisk iax client, but did'nt received an url |
20:24.22 | bjohnson | none are voip though |
20:24.31 | *** part/#asterisk dabigshiznizzle (n=dabigshi@gwsecure.sctelcom.com) |
20:24.39 | bjohnson | since we're providing inventories |
20:24.45 | darkskiez | I've read the asterisk dimensioning page, but i'm not sure what you need to run a TE4xxp with 120 channels and 1500 clients. |
20:24.45 | ClubBarf | I have 2 voip phones (1 is mine, the other's a loan) and 3 softphone machines. |
20:25.03 | ClubBarf | It's cheaper to buy old PC's to use as softphones than to buy a hardware phone... |
20:25.21 | hypa7ia | ClubBarf: not when you look at the power bills :-/ |
20:25.27 | syle | is it |
20:25.28 | ClubBarf | Meh. |
20:25.30 | spackle | darkskiez>, that sounds more like multiple machines. |
20:25.37 | syle | checkout your electricity bill |
20:25.38 | ClubBarf | Those desks needed PC's anyway. |
20:25.44 | raceman | nobody? |
20:25.50 | ClubBarf | For excell, word etc. |
20:26.14 | ClubBarf | Nothing fancy, which is why a P2 350 with 256MB ram does the trick nicely. |
20:26.24 | darkskiez | spackle: but you can only put a TE4xxp in one machine :/ |
20:26.25 | jake1932 | hmm. is there a softphone out there for windows that consistantly works? |
20:26.28 | spackle | The Microspendaplenty office menagerie |
20:26.45 | darkskiez | spackle: and there will be no transcoding |
20:26.50 | ClubBarf | jake1932 - I'm beginning to think... No. |
20:27.31 | spackle | darkskiez>, maybe some machines to register to and then another machine to connect to the PSTN, or a two port cad instead of a four port card. |
20:27.59 | darkskiez | spackle: could SER handle registrations and run on same machine as asterisk? |
20:27.59 | ClubBarf | Anyhew, nice chatting with you fellas. I'm having a hot bath and going to bed. |
20:28.49 | spackle | darkskiez>: file could probably answer that better. But that's a lot of folk, better to not put all eggs in one basket unless you have a killer support contract on the machine and spare cards. |
20:28.54 | darkskiez | spackle: there is a chance i'll need 10 Pri lines with failover machines. |
20:29.31 | spackle | ClubBarf: TMI |
20:30.13 | darkskiez | was looking at http://www.junghanns.net/en/ISDNguard_produkt.html that, anyone used it? |
20:30.17 | file[laptop] | you can run SER on the same machine as Asterisk... just need to bind to different ports |
20:30.25 | hardwire | grr |
20:30.29 | spackle | darkskiez>: some kind of a call-center or just the whole company? |
20:30.34 | darkskiez | a hotel. |
20:30.36 | hardwire | the debian contrib init script exports to LD that we are 2.4.x |
20:30.43 | hardwire | and splits asterisk into a milion tiny little pids |
20:30.47 | hardwire | on my 2.6.x system |
20:31.04 | spackle | darkskiez>: in vegas? |
20:31.10 | darkskiez | in london |
20:31.23 | spackle | darkskiez>: sip phones in the rooms? |
20:31.25 | darkskiez | one there already has 16 PRI lines |
20:31.42 | darkskiez | spackle: u want to go and steal them? |
20:31.53 | spackle | darkskiez>: not me, but somebody |
20:31.55 | darkskiez | That concerns me too. |
20:32.00 | darkskiez | its not a done deal. |
20:32.13 | spackle | darkskiez>: we had this discussion a couple months ago here. |
20:33.40 | darkskiez | wonder if we can get someone to make a custom firmware that wont upgrade to generic and will be subtly incompatible. |
20:33.49 | darkskiez | Needs a custom sip header or something to work. |
20:34.05 | spackle | darkskiez>: but you don't find that out until the phone is gone. |
20:34.14 | bjohnson | darkskiez: SIP ATAs attached to wall with analog deskphones as a better option? |
20:34.29 | darkskiez | is there any SIP ATAs that take POE ? |
20:34.44 | bjohnson | darkskiez: that only helps if someone has already stolen them and word has gotten out that they are not usable |
20:34.56 | spackle | darkskiez>: Is the thought to deliver phone and IP to the room without extalines |
20:34.57 | bjohnson | darkskiez: make your own POE adapters |
20:35.26 | darkskiez | spackle: without extra plug sockets and lines etc. |
20:35.34 | darkskiez | ATA's are easy to nick too. |
20:35.35 | bjohnson | all it is doing is sending DC over 2 or more of the cat5 lines |
20:35.49 | spackle | darkskiez>: what kind of room phones have you spec'd? |
20:36.01 | darkskiez | bjohnson: there is a POE protocol to stop damaging non POE hardware, so its not that simple. |
20:36.02 | bjohnson | put in a cat5 wall box and wire out the power lines to a plug into the ATA |
20:36.34 | bjohnson | darkskiez: you would remove the power before it got to the ATA |
20:36.35 | spackle | yeah, like those 3com boxes? |
20:36.40 | darkskiez | bjohnson: oh. duh. true. |
20:36.55 | darkskiez | 'those 3com boxes' ? |
20:37.18 | spackle | they take POE, and are a switch or hub that fits in a standard wall-jack box. |
20:37.28 | spackle | offer 3 or 4 lines out |
20:37.38 | darkskiez | Oh, aye, them. |
20:37.51 | darkskiez | but with an ATA built in.. drool. |
20:38.11 | *** join/#asterisk shido6 (n=shido@d221-68-210.commercial.cgocable.net) |
20:38.13 | spackle | that would be neat. |
20:38.35 | hardwire | stop when convenient |
20:38.35 | hardwire | well |
20:38.37 | spackle | run, don't walk to the patent office. |
20:38.39 | hardwire | its convenient for the callers here |
20:38.41 | hardwire | but not for me |
20:39.28 | darkskiez | I've been making to make a patch to *. stop after apology |
20:39.32 | spackle | darkskiez> I bet some of the phone manufacturers would produce a special firmware in a volume like that. |
20:39.38 | darkskiez | plays a sample to all active channels and then stops. |
20:39.44 | bjohnson | darkskiez: http://www.nycwireless.net/poe/ |
20:39.50 | bjohnson | I was thinking something like that |
20:40.07 | bjohnson | then mount the ATA right besdie it on the wall |
20:40.31 | darkskiez | bjohnson: Except the POE switch wont deliver the power until after a handshake. |
20:40.41 | *** join/#asterisk Firestorm-voip (n=Firestor@ua-83-227-140-131.cust.bredbandsbolaget.se) |
20:40.48 | darkskiez | bjohnson: and I dont want to sit down with a dremel and soldering iron and make 1000 of them. |
20:41.03 | spackle | call taiwan |
20:41.32 | bjohnson | exactly |
20:42.15 | darkskiez | 00886 and dial random numbers till someone picks up? |
20:42.34 | bjohnson | or put the works, including the ATA into a larger box and just have the cat5 hookups showing |
20:43.19 | bjohnson | darkskiez: there's got to be something similar that is provided by POE switch providers |
20:43.36 | darkskiez | i think there is a cisco config command to force poe to be on etc. |
20:43.47 | bjohnson | I like this one http://www.nycwireless.net/poe/lap1000.jpg |
20:44.06 | bjohnson | except picture an ATA where the wifi AP is and both mounted to a wall |
20:44.07 | spackle | darkskiez: market is about ready for a "disposable" IP phone. oh wait, what about budgetones? |
20:44.27 | bjohnson | darkskiez: what phones currently in use? |
20:44.31 | raceman | Anyone experience with the SendUrl() command in extensions.conf ? I tryed it with idefisk iax client, but did'nt received an url |
20:44.42 | darkskiez | bjohnson: this particular spec is for a hotel that isnt built yet. |
20:45.12 | bjohnson | ip to rooms for data too? |
20:45.14 | darkskiez | so i have to spec hardware and a range of phones etc |
20:45.19 | darkskiez | bjohnson: and TV :) |
20:45.20 | spackle | darkskiez> - so you can spec a phone that hasn't been made yet? |
20:45.32 | bjohnson | so you would want a hub or switch in each room too |
20:45.41 | darkskiez | if the phone can be built in time |
20:45.53 | bjohnson | unlikely |
20:46.17 | bjohnson | or at least .. I wouldn't count on it |
20:46.19 | darkskiez | bjohnson: well, a phone with hub built in is whats being looked at. |
20:46.21 | Druken | bjohnson: not a hub or switch, i'd do two seperate networks, one voice, one data |
20:46.35 | bjohnson | why? |
20:46.42 | darkskiez | druken: the hub is only at the room level, not a high contention. |
20:46.48 | Druken | less chance of interference |
20:46.52 | bjohnson | do one cat5 to each room and split the voice/data at the switch |
20:47.18 | bjohnson | you can even limit the data bandwidth |
20:47.47 | spackle | you could do that with CoS, Qos and VLANs too couldn't you? |
20:48.05 | spackle | darkskiez>: is there an IP TV? |
20:48.07 | bjohnson | darkskiez: would a hotel desk analog phone be any cheaper than a voip phone anyway? |
20:48.32 | bjohnson | oh yeah .. ip TV might suck up the bandwidth |
20:48.34 | spackle | bjohnson: probbaly not at least by a lot. |
20:48.37 | jake1932 | some of the cable networks are using "digital" - might be ip based? |
20:48.56 | bjohnson | but not cat5 that I've seen |
20:49.05 | darkskiez | a phone timing out could raise a theft alarm |
20:49.14 | *** join/#asterisk ian_k (n=ian@gateway.digium.com) |
20:49.20 | AgiNamu | When someone says to loop the T1 at our end |
20:49.24 | AgiNamu | what do we physically do? |
20:49.25 | bjohnson | you'd need a box to convert the cat5 video stream to something the tv could use |
20:49.32 | Dr_Ray | ok, how do I write up the e-ink dev kit as something that bossman should buy me |
20:49.37 | spackle | mythtv only transmits the "channel" or show you are watching, it could easily share the pipe with a phone call and data given good QoS |
20:49.47 | *** join/#asterisk folsson (n=filip@lund-meje-sr0-vl101-249.perspektivbredband.net) |
20:49.54 | darkskiez | the TV takes up about 4Mbits |
20:50.13 | bjohnson | spackle: control for each TV would be killer with such a configuration |
20:50.22 | spackle | Dr_Ray call it critical developmental info fodder for future development |
20:50.24 | bjohnson | darkskiez: what hardware does that? |
20:50.40 | darkskiez | bjohnson: ours, i work for an iptv company |
20:51.02 | bjohnson | crap, roll the hub, POE, and ATA into THAT! |
20:51.07 | spackle | darkskiez, why don't you make the switchbox into your TVs or settop box then? |
20:51.13 | darkskiez | bjohnson: i'd like to but we dont make the hardware |
20:51.14 | bjohnson | yeah yeah |
20:51.15 | bjohnson | do it |
20:51.24 | spackle | somebody does |
20:51.29 | darkskiez | bjohnson: amino do |
20:51.56 | darkskiez | we vape their software stack and have our own |
20:52.10 | spackle | bummer |
20:53.06 | *** join/#asterisk jeffik (n=Jeff@CPE0020ed8494b8-CM0012c999ca4e.cpe.net.cable.rogers.com) |
20:53.50 | rayvd | AgiNamu: make a loopback cable and plug it into your CSU/DSU card on your router |
20:53.51 | darkskiez | they screw under tables and hide away nicely: http://www.aminocom.com/products/ipstb/aminet110h.html |
20:54.06 | AgiNamu | rayvd oh ok |
20:54.30 | rayvd | usually you cross a couple of wires and voila |
20:56.26 | syle | wtf is diff between xp pro and xp home |
20:56.50 | mitcheloc | running iis on it and connecting to a domain |
20:57.01 | mitcheloc | ...well the two features that are important to me |
20:57.25 | syle | you can still share shit right on home? and connect to samba shares right? |
20:57.55 | syle | apache > iis |
20:58.10 | hardwire | Sep 28 12:56:47 WARNING[10117]: chan_zap.c:8288 pri_dchannel: Ring requested on channel 0/1 already in use on span 1. Hanging up owner. |
20:58.10 | hardwire | Sep 28 12:57:05 WARNING[10117]: chan_zap.c:8288 pri_dchannel: Ring requested on channel 0/1 already in use on span 1. Hanging up owner. |
20:58.10 | hardwire | damnit |
20:58.31 | jake1932 | syle: http://www.microsoft.com/windowsxp/pro/howtobuy/choosing2.mspx |
20:58.52 | jake1932 | bunch of marketing material mixed in with facts |
20:59.34 | Dr_Ray | I need to get better at having a slushfund |
20:59.37 | Dr_Ray | at work |
20:59.39 | *** join/#asterisk websae (i=websae@207-118-145-168.dyn.centurytel.net) |
20:59.52 | darkskiez | hardwire: show channels? |
20:59.59 | hardwire | they all died |
21:00.10 | darkskiez | whats show channels say |
21:00.21 | mitcheloc | syle: you can't give up remote desktop either |
21:00.24 | hardwire | darkskiez: nothing |
21:00.25 | hardwire | they all died |
21:00.29 | hardwire | a few seconds later |
21:00.31 | hardwire | Sep 28 12:57:17 WARNING[10117]: channel.c:741 channel_find_locked: Avoided initial deadlock for '0x8730470', 10 retries! |
21:00.44 | darkskiez | cvs head? whend u build it |
21:00.45 | hardwire | same thing a little before |
21:00.52 | hardwire | upped from today |
21:00.55 | darkskiez | depressing |
21:00.56 | darkskiez | http://www.voip-info.org/tiki-index.php?page=Ring+requested+on+channel |
21:00.57 | hardwire | this morning about an hour ago |
21:01.04 | *** join/#asterisk DannyF (n=dannyf@c-794fe353.24-0099-74657210.cust.bredbandsbolaget.se) |
21:01.04 | darkskiez | Ive got a potential fix for it |
21:01.15 | darkskiez | hardwire: did u rebuild libpri too/ |
21:01.20 | hardwire | yeh |
21:01.26 | darkskiez | first and reiinstalled? |
21:01.36 | hardwire | build libpri first |
21:01.38 | hardwire | installed |
21:01.40 | hardwire | built zaptel |
21:01.41 | hardwire | installed |
21:01.42 | darkskiez | i described a potential fix there, but i dont know the code well enough |
21:01.45 | hardwire | useing wct1xxp |
21:01.52 | mitcheloc | how sad that you have to run that script... |
21:02.04 | hardwire | mitcheloc: it just started happening today |
21:02.05 | mitcheloc | i've never need to |
21:02.17 | darkskiez | i dont get it very often |
21:02.18 | hardwire | after the ILEC broke our T1 |
21:02.23 | darkskiez | dont run the script. |
21:02.25 | syle | damn |
21:02.29 | hardwire | they unhooked our pairs |
21:02.34 | darkskiez | but I added a comment |
21:02.34 | hardwire | then hooked them back up |
21:02.36 | syle | well wouldn;t mind getting my hands on media center |
21:02.43 | frenzy | hello |
21:02.44 | mitcheloc | why would the code freeze anyways? it shouldn't stop asterisk because of something like that |
21:02.50 | mitcheloc | so why need to restart it.. |
21:02.55 | hardwire | darkskiez: thanks for that |
21:03.13 | frenzy | whats the simplest way to create an extension and point to to a specific sip ? |
21:03.16 | darkskiez | hardwire: I dont run the script...but it might be good for you |
21:03.21 | hardwire | no |
21:03.25 | hardwire | I think the ILEC can fix it |
21:03.37 | hardwire | but it totally kills asterisk |
21:03.40 | hardwire | which I dislike |
21:03.41 | darkskiez | I think its a bug in asterisk too |
21:03.48 | darkskiez | i think it should be handled better |
21:03.52 | darkskiez | as i've described |
21:04.01 | frenzy | exten => 2000,1,Dial(SIP/2000) ? |
21:04.01 | hardwire | yeh |
21:04.16 | hardwire | frenzy: looks fine |
21:04.38 | hardwire | I need to get some graphiing going |
21:04.44 | hardwire | about how often our PRI is in use |
21:04.48 | hardwire | stuff like that |
21:05.13 | *** join/#asterisk hypa7ia (i=hypatia@silenceisdefeat.org) |
21:05.32 | syle | anyone know if windows media center is more like pro or home? |
21:05.40 | hypa7ia | pro |
21:05.41 | hypa7ia | ish |
21:05.53 | frenzy | how about if the user is SIP ? |
21:06.01 | frenzy | whoops |
21:06.05 | frenzy | meaning characters |
21:06.14 | darkskiez | asterisk doesnt handle chars so well |
21:06.57 | darkskiez | it can though |
21:07.02 | darkskiez | but there are caveats |
21:07.06 | frenzy | I see... |
21:07.08 | emp | can someone explain how exten => 1234,5,dial(${TRUNK}c/9871234321,20,r) works? especially the c/ part |
21:07.16 | frenzy | what about priorities ? |
21:07.53 | darkskiez | what about zebras? |
21:08.20 | spackle | "Cheese Grommit." |
21:08.24 | shido6 | ? |
21:08.24 | hardwire | ewwww |
21:08.25 | frenzy | :P |
21:08.26 | *** join/#asterisk PupenoL (n=pupeno@200.123.183.89) |
21:08.31 | PupenoL | hello. |
21:08.39 | hardwire | spackle: did you see the new movie? |
21:08.44 | frenzy | does it really matter setting it to 1 ? |
21:09.00 | spackle | hardwire, not yet. Any good? |
21:09.00 | darkskiez | frenzy: your talking disjointed jibberish |
21:09.04 | hardwire | dunno |
21:09.30 | *** part/#asterisk makhtar (n=ageller@mail.bulletinnews.com) |
21:09.34 | spackle | hardwire, It's lost it luster a little. |
21:09.40 | hardwire | :( |
21:09.42 | frenzy | heh |
21:10.01 | spackle | two day to Serenity |
21:10.07 | spackle | days that is |
21:10.31 | *** join/#asterisk sudhir492 (n=sudhir@pool-71-114-84-37.washdc.dsl-w.verizon.net) |
21:10.32 | hardwire | that looks interesting |
21:10.37 | *** join/#asterisk rat1101 (n=vinay@ip68-100-31-133.dc.dc.cox.net) |
21:10.38 | hardwire | the captain guy looks like he has a really broken nose |
21:10.48 | darkskiez | Hhahahaha love the slashdot story about a phone survey to find out if people were getting unwanted calls. Genious. |
21:11.13 | sudhir492 | On my Asterisk, I see this message quite frequently: stale nonce received from .... |
21:11.15 | spackle | hardwire: never saw the TV show? |
21:11.20 | hardwire | firefox? |
21:11.22 | hardwire | something like that |
21:11.26 | spackle | firefly |
21:11.29 | hardwire | yeh |
21:11.30 | hardwire | never saw it |
21:12.00 | darkskiez | there is a firefly torrent doing the rounds |
21:12.04 | spackle | I "found" the DVDs ripped onto my hard drive one time and watched them all. Grew on me really fast. |
21:12.04 | frenzy | I get stale nonce all the time too |
21:12.10 | sudhir492 | I am using Polycom phones, |
21:12.19 | darkskiez | ahh 90% woo |
21:12.23 | hardwire | frenzy: it happens |
21:12.24 | Dr_Ray | qwest - called me a few years ago asking if I wanted so subscribe to a service that stopped unwanted phone calls, "You mean like this one?". The guy on the phone was a good sport about it. Said yes, I told him I already had a mechanism for dealing with those kind of calls, thanked him and hung up. |
21:12.25 | *** join/#asterisk expressfone1 (n=expressf@62-15-97-163.inversas.jazztel.es) |
21:12.29 | expressfone1 | Hi |
21:12.29 | sudhir492 | what causes that, and what does that mean? |
21:12.39 | sivana | is there a PRI channel usage grapher? |
21:13.20 | *** part/#asterisk raceman (n=bla@cust-02-E169.adsl.scarlet.nl) |
21:13.30 | *** join/#asterisk Brixius (n=Brixius@162.96.15.48) |
21:13.47 | expressfone1 | any one can helpme to boot Astlinux on net4801 (CF), getting VFS: Cannot open root device "hda1" or unknown-block(0,0) |
21:16.07 | hardwire | its hde |
21:16.34 | frenzy | say, any good music on asteirks I can use to test playback ? |
21:17.01 | expressfone1 | thanks hardwire, let me test it |
21:17.54 | *** join/#asterisk KranZ (n=user@sme.bestline.net) |
21:19.08 | expressfone1 | hardwire> same error :-S |
21:19.15 | hardwire | sorry |
21:19.42 | expressfone1 | any one running astlinux on soekris net4801 ???? |
21:20.41 | spackle | expressfone1: I've had it working in the past. |
21:21.12 | spackle | expressfone1: are you running it off CD or a hard disk? |
21:21.18 | expressfone1 | CF |
21:21.29 | spackle | what brand of CF? |
21:21.35 | Connor_ | CVS-HEAD or 1.2 tar file ? |
21:21.42 | Connor_ | which one should I use now.. |
21:21.42 | expressfone1 | scandisk 128M |
21:22.28 | spackle | there are bios settings and I think a jumper for boot device settings, have you fiddled with either? |
21:23.43 | frenzy | Sep 28 17:19:21 WARNING[20598]: file.c:475 ast_openstream: File fpm-calm-river does not exist in any format |
21:23.44 | frenzy | Sep 28 17:19:21 WARNING[20598]: file.c:787 ast_streamfile: Unable to open fpm-calm-river (format ilbc): Resource temporarily unavailable |
21:23.44 | frenzy | Sep 28 17:19:21 WARNING[20598]: app_playback.c:83 playback_exec: ast_streamfile failed on SIP/7777777-dcde for fpm-calm-river |
21:23.44 | expressfone1 | i´m install monowall and run ok |
21:23.54 | expressfone1 | boot ok from cf |
21:23.56 | frenzy | is there a way I can play mp3 on ilbc ? |
21:24.03 | expressfone1 | my problem is boot astlinux |
21:24.39 | spackle | I guess so. How are you flashing it? using Windows or linux? |
21:25.02 | expressfone1 | xp |
21:25.44 | Uberbot | shido6? |
21:26.20 | spackle | I had trouble last time I tried to flash from XP too. That's why I don't have it working right now. Maybe check with the astlinux listserv? Kristian is really good at getting back to people. |
21:26.21 | CleanerX | anyone familiar with pri signalling |
21:26.45 | expressfone1 | ok spackle |
21:26.52 | CleanerX | i've figured out a problem with q932 protocoll |
21:26.55 | CleanerX | -l |
21:27.01 | jalsot | hi |
21:27.47 | jalsot | I have a problem with iax2 softphone and ringback tone [hearing local and remote RBT :( ] |
21:28.21 | jalsot | does anybody have an idea what can be wrong? even if I dial out with 'r' option, I hear 2 ringback tones |
21:32.15 | sudhir492 | from time to time I cannot make call from my Polycom phone. PAP2-NA does not have the same problem |
21:33.50 | *** join/#asterisk SplasPood (n=jwb@dsl081-201-143.nyc2.dsl.speakeasy.net) |
21:33.58 | KranZ | any way to force a progress indicator so asterisk generates the ringback when the calling party doesnt supply a progress indicator? |
21:34.08 | KranZ | happens with cell phones |
21:36.39 | *** join/#asterisk kshumard (n=root@gateway.digium.com) |
21:37.53 | *** join/#asterisk generalhan (i=general_@ip67-90-64-2.z64-90-67.customer.algx.net) |
21:38.06 | generalhan | whats going on everyone ? |
21:39.22 | websae | anyone here familiar with AstBILL? |
21:39.29 | websae | or has anyone tried it? |
21:41.17 | bkw_ | frenzy, if you go get format_mp3 from asterisk-addons |
21:41.46 | *** join/#asterisk afrosheen (n=test@txprotoa2.august.net) |
21:42.03 | *** join/#asterisk jtodd (n=jtodd@dsl027-191-178.sfo1.dsl.speakeasy.net) |
21:42.36 | *** join/#asterisk Gerriall (n=NonYa@209.42.198.18) |
21:42.36 | *** part/#asterisk Brixius (n=Brixius@162.96.15.48) |
21:43.53 | generalhan | Is there a feature in Asterisk that will allow me to dial a number to listen into someones extension? for like training purposes ?? |
21:45.04 | darkskiez | generalhan: yes |
21:45.19 | bkw_ | chan_spy |
21:45.21 | generalhan | darkskies: can you link me to some info |
21:45.26 | hardwire | I have yet to play with chan_spy |
21:45.37 | mitcheloc | i thought chan_spy was *discontinued*? |
21:45.46 | bkw_ | its in CVS |
21:45.52 | afrosheen | used to use zapbarge but I think it only works on zap trunks |
21:45.59 | generalhan | hmm that makes me a bit nervous when people that i see in here ALL THE TIME havent even tried it. makes a little concerned that i might not be able to use it at all ! lol |
21:46.03 | mitcheloc | oh ok, so its just *undocumented* ;) |
21:46.46 | malverian[work] | generalhan, I use app_chanspy |
21:46.52 | malverian[work] | generalhan, It works well and is in CVS HEAD now. |
21:47.02 | malverian[work] | generalhan, It works on all channels. |
21:47.13 | generalhan | how does that work in comparison to chan_spy ? |
21:47.14 | malverian[work] | generalhan, It is "chan_spy" .. app_chanspy.so |
21:47.24 | afrosheen | I think chan_spy is the new implementation, zapbarge is old school |
21:47.42 | malverian[work] | It's not really "chan_spy" because it is not a channel.. it is an application. |
21:47.46 | generalhan | anyone have a link that might explain to me how to set it up ? or is it in the wiki pages ? |
21:47.53 | malverian[work] | show application chanspy |
21:47.57 | malverian[work] | generalhan, ^ from cli |
21:48.01 | generalhan | ok |
21:48.02 | *** join/#asterisk goatmilk (n=goatmilk@130-127-45-11.chouse.resnet.clemson.edu) |
21:48.05 | *** part/#asterisk mkrufky-not-here (n=mk@68.160.103.77) |
21:48.50 | mitcheloc | damnit...i'm running asterisk 1.0.0, bah, i wanted to try chan_spy, it's probably newer then that time period anyways |
21:50.10 | *** join/#asterisk RoyK (n=roy@55.80-202-161.nextgentel.com) |
21:50.56 | generalhan | malverian: i get this when i try the show application cmd : PhoneBox*CLI> show application chanspy |
21:50.57 | generalhan | Your application(s) is (are) not registered |
21:51.36 | mitcheloc | what is your "show version" |
21:51.42 | generalhan | 1.0.9 |
21:51.58 | mitcheloc | they said it's only in cvs head, that doesn't sound like it's cvs head |
21:52.01 | sudhir492 | what WiFi SIP phone works well with Asterisk? |
21:52.09 | sudhir492 | Any inexpensive ones? |
21:52.29 | afrosheen | wifi sip? ohhhh... |
21:52.38 | generalhan | mitcheloc: who is "they" |
21:52.52 | afrosheen | are you talking something like a standard portable phone wifi or an actual cellphone-ish sip phone |
21:52.55 | *** join/#asterisk Moc (n=mochouin@modemcable111.229-203-24.mc.videotron.ca) |
21:52.56 | mitcheloc | mitcheloc: i thought chan_spy was *discontinued*? |
21:52.56 | mitcheloc | bkw_: its in CVS |
21:53.19 | wunderkin | im getting ready to use it now |
21:53.22 | mitcheloc | so basically it's not in the release versions, probably for some sort of liability reasons |
21:53.30 | generalhan | <malverian[work]> generalhan, It works well and is in CVS HEAD now. |
21:53.57 | generalhan | so when i read that i thought he meant that it was working AND is working in CVS HEAD |
21:54.27 | mitcheloc | well it wasn't very clear ;), anyhow you'll need cvs head |
21:55.01 | generalhan | well i really dont want to change anything, damnit |
21:55.13 | mitcheloc | same here, so i can't use it either lol |
21:55.42 | *** join/#asterisk AlexCTI (n=alex@weston-69.65.86.197.myacc.net) |
21:55.48 | AlexCTI | Hi.. |
21:56.21 | AlexCTI | Can some one help me to set up the music on hold? |
21:57.58 | generalhan | ~jbot Dict crack |
21:58.13 | generalhan | ~jbot Dict crack pipe |
21:58.16 | *** join/#asterisk srt (n=nobody@gw0-cgn.reucon.net) |
21:58.19 | generalhan | lol |
21:58.23 | mitcheloc | is jbot in the dictionary? |
21:58.31 | mitcheloc | ~jbot dict dict crack |
21:58.42 | generalhan | ~jbot dict jbot |
21:58.57 | RoyK | lol |
21:59.08 | mitcheloc | it should have errored when it ran into an unknown defition for itself... |
21:59.11 | mitcheloc | what crappy programming |
21:59.17 | RoyK | ~lart mitcheloc |
21:59.50 | generalhan | lol |
21:59.51 | generalhan | what other kind of commands can he do ? |
22:01.08 | RoyK | generalhan: people from the .us are evil and deserves nothingg but harrasment |
22:01.22 | FuriousGeorge | it seems the only way to "lower the volume" of my MoH is to resample the tracks and lower the gain(?). is there any way to tell what the gain is on these tracks now, or do i just shoot in the dark? |
22:01.36 | FuriousGeorge | it is gain, isnt it? |
22:03.05 | afrosheen | FuriousGeorge: yeah there's no volume tweaking for MOH, you have to normalize your tracks all by yourself |
22:03.16 | FuriousGeorge | i got that part |
22:03.27 | FuriousGeorge | im gonna use sox and everything |
22:03.31 | afrosheen | if you're handling mp3's, just decode them all to .wav, normalize it in your encoder |
22:03.33 | afrosheen | oh ok |
22:04.08 | generalhan | RoyK: whats wrong with people from the .us ? |
22:04.14 | FuriousGeorge | im just wondering what "value" i need to be lowering, whats it called. surely its not called "colume" |
22:04.18 | FuriousGeorge | *volume |
22:04.53 | fugitivo | FuriousGeorge: normalize |
22:04.54 | FuriousGeorge | generalhan: we're far enough from europe to not really care |
22:05.16 | FuriousGeorge | fugitivo: thanks |
22:05.26 | fugitivo | FuriousGeorge: that´s the name "normalize", some programs have preset values for speech, music, etc |
22:06.10 | fugitivo | FuriousGeorge: for speech i think it's -20db |
22:06.35 | FuriousGeorge | fugitivo: its mozart, the stuff you linked me to the other day actually |
22:06.40 | afrosheen | yeah, we'll invade in ford taurus' equipped with rocket launchers lol |
22:06.43 | *** join/#asterisk BasketCase (n=BasktCas@asylum.sanitarium.net) |
22:06.45 | RoyK | generalhan: quite a lot |
22:06.50 | RoyK | generalhan: sorry |
22:06.52 | Connor_ | OMFG someone fix tab-complete with the "like" operator for iax2/sip show peers |
22:06.56 | fugitivo | FuriousGeorge: what stuff? i don't remember :) |
22:07.10 | FuriousGeorge | fugitivo: exactly |
22:07.26 | gordonjcp | afrosheen: American tanks that can go about 50 miles on a tank of fuel, assuming they don't just go on fire |
22:07.48 | gordonjcp | and then if there is anything like a bend in the road, or a bump, they're stuffed |
22:07.51 | afrosheen | what is this, a ww2 history lesson of the Shermans? |
22:08.21 | FuriousGeorge | RoyK: generalhan: i view nationalism like a family, i can make fun of my family, and other people in my family can make fun of us, but outsiders cant |
22:08.31 | gordonjcp | afrosheen: it's just extrapolating from the American cars I deal with |
22:08.40 | FuriousGeorge | well, they can, but i reserve the right to not care |
22:08.52 | fugitivo | FuriousGeorge: ? |
22:08.57 | gordonjcp | I've never seen a car that could set itself on fire by leaking water |
22:09.03 | afrosheen | gordonjcp: most americans don't even drive american cars much if they can afford not to...and the tanks...well, they're world famous for blowing other things up |
22:09.13 | gordonjcp | until I started fixing yank tanks |
22:09.29 | FuriousGeorge | fugitivo: scroll up for the context |
22:09.29 | *** part/#asterisk BasketCase (n=BasktCas@asylum.sanitarium.net) |
22:09.38 | gordonjcp | afrosheen: I'm not surprised, they're just plain dangerous |
22:09.49 | R3DB0x | can anyone tell me anything good or bad about the cisco 7920 phones |
22:09.49 | generalhan | RoyK: "quite a lot"??? like what ? |
22:09.50 | afrosheen | ha, nearly got me |
22:10.01 | fugitivo | oh well, i only have 10 lines for buffer |
22:10.02 | gordonjcp | no brakes, no lights, no suspension |
22:10.17 | gordonjcp | slow too |
22:10.39 | FuriousGeorge | one true thing: they're all automatic |
22:11.08 | gordonjcp | yeah |
22:11.20 | *** part/#asterisk alerios (n=alerios@200.24.109.199) |
22:11.22 | gordonjcp | the worst I've ever driven was a Z28 |
22:11.30 | gordonjcp | quick, I'll give it that |
22:11.35 | gordonjcp | 0-60 in about 5 seconds |
22:11.37 | RoyK | generalhan: so far I've met ONE person from the US being reflective |
22:11.38 | fugitivo | you didn't drive a tracker eh? |
22:11.46 | gordonjcp | ... but that was it |
22:11.50 | gordonjcp | top speed 90mph |
22:11.51 | generalhan | interesting |
22:12.01 | generalhan | and where are you ? |
22:12.08 | RoyK | .no |
22:12.32 | gordonjcp | 8mpg and getting passed by everything on the road |
22:12.36 | generalhan | .no ? |
22:12.44 | KranZ | norway? |
22:12.52 | KranZ | .yes |
22:12.55 | generalhan | .lol |
22:12.59 | r0d3nt | So since when does Asterisk and Digium's hardware have a known problem with PRI echo ???????????????????????? |
22:13.05 | KranZ | .yesrway |
22:13.07 | afrosheen | since awhile |
22:13.16 | gordonjcp | it does? |
22:13.17 | r0d3nt | i loaded zaptel cvs head, no change. |
22:13.44 | r0d3nt | afrosheen, no one has reported this until today. |
22:13.58 | r0d3nt | the only pri asterisk installs i have trouble with echo are on telepacific. |
22:14.25 | Connor_ | msg bkw_ never mind |
22:16.10 | r0d3nt | so Chris from Digium said the magical fix for echo on PRI's was in the CVS HEAD of zaptel.... |
22:16.12 | r0d3nt | i loaded it... |
22:16.14 | r0d3nt | no change... |
22:16.16 | r0d3nt | NONE. |
22:16.25 | drumkilla | it has a new echo cancellation module |
22:16.26 | r0d3nt | and since he told telepacific that this is a known problem.... |
22:16.28 | drumkilla | er, not module |
22:16.35 | r0d3nt | guess what ?!?!?! THEY JUST TOLD ME TO FUCK OFF. |
22:16.37 | drumkilla | new software echo can, though |
22:17.14 | KranZ | drumkilla: do u have to edit zconfig.h to enable that? |
22:17.15 | r0d3nt | where is this new software ??? |
22:17.35 | r0d3nt | he said and I quote " do a cvs check out of zaptel and make install and you're done " |
22:17.51 | r0d3nt | and that is supposed to enable this new magical echo cancellation code... ??? |
22:17.59 | KranZ | did you reload the module? |
22:18.02 | r0d3nt | UH YES> |
22:18.05 | sivana | r0d3nt: the new code is on by default |
22:18.10 | r0d3nt | sivana, ok.... |
22:18.22 | KranZ | did you click your heels and count to 3? |
22:18.36 | drumkilla | r0d3nt: calm down, son :) |
22:18.39 | r0d3nt | KranZ, i've done all the voodoo magic i know. |
22:18.43 | drumkilla | it has fixed the problems for most people |
22:18.43 | KranZ | heh |
22:18.44 | *** join/#asterisk zotz (n=zotz@24.231.36.100) |
22:18.44 | afrosheen | did you utter a short anti-american rant just to be sure? |
22:18.46 | r0d3nt | drumkilla, ....... |
22:18.54 | sivana | r0d3nt: but.. did you say abra |
22:18.54 | r0d3nt | afrosheen, absolutely not. |
22:19.01 | drumkilla | if it's that bad, maybe you need to invest in a hardware echo can board |
22:19.02 | afrosheen | well there's your problem |
22:19.49 | r0d3nt | drumkilla, all our tests show that it's not within asterisk and it's not within the PRI and it's not within Telepacific, but that it happens when they hand off to Sprint or MPower or anothe CLEC.... |
22:19.50 | KranZ | or a separate pri echo can hardware to insert in the pri |
22:20.48 | KranZ | you might need to raise the echocancel taps |
22:20.50 | r0d3nt | drumkilla, so how would that effect the issue ?? onboard quad PRI echo canceller card would only help it if there is echo on the PRI no ??? |
22:21.07 | r0d3nt | KranZ, I've done all of the above... |
22:21.28 | KranZ | is it on inbound or outbound calls? |
22:21.31 | r0d3nt | 32, 64, 128, 256, ON OFF bridged, YES NO, jitterbuffer up and down... |
22:21.35 | r0d3nt | KranZ, BOTH. |
22:21.45 | r0d3nt | duplicatble with certain phone numbers 100% of the time |
22:21.49 | r0d3nt | mostly mpower customers |
22:21.53 | FuriousGeorge | sox looks like fun. ive got 4 * servers and only one has bareable moh |
22:22.06 | r0d3nt | telepacific says " we ran tests between TPAC and mpower and we show no echo " |
22:22.22 | FuriousGeorge | the other too are far too loud. the only variable i can find is that the good one has an installed sound card |
22:22.38 | KranZ | r0d3nt: you're not getting slips or having timing issues are you? |
22:22.43 | r0d3nt | none. |
22:22.55 | r0d3nt | no bit errors, no timing slips, no alarms. |
22:23.21 | *** part/#asterisk extremis (i=extremis@equinox.alluvium.com) |
22:23.58 | KranZ | are you calling from voip lines? |
22:24.25 | r0d3nt | cisco 7940g's |
22:24.50 | r0d3nt | cisco <-> sip <-> asterisk <-> PRI <-> telepacific --------------- |
22:24.59 | KranZ | ---- pooop |
22:25.03 | afrosheen | haha |
22:25.11 | r0d3nt | yes it is poop. |
22:25.13 | KranZ | there's the problem, too much poop |
22:25.16 | KranZ | hmm... |
22:25.23 | r0d3nt | KranZ, funny. |
22:25.30 | r0d3nt | i'm glad you all can joke and laugh @ me. |
22:25.32 | KranZ | heh |
22:25.39 | KranZ | im honestly trying to think up solutions |
22:25.46 | KranZ | but everynow and then.... |
22:25.59 | r0d3nt | well this is my job.. and i'm about ready to quit. |
22:26.09 | KranZ | you try a different box? |
22:26.13 | r0d3nt | uh |
22:26.15 | r0d3nt | i have 3. |
22:26.20 | KranZ | all echo? |
22:26.34 | afrosheen | is it local or far end echo |
22:26.36 | r0d3nt | dell sc420, dell 2850, no-name with dual piii 750's Asus momboard... |
22:26.47 | sivana | r0d3nt: what are you using for interface with PRI? |
22:26.49 | fugitivo | do you have echo on all your calls? |
22:26.49 | sivana | 405? |
22:26.54 | r0d3nt | afrosheen, far end echo.. only I/we can hear it.. the other person on the PSTN cannot hear it... |
22:27.02 | afrosheen | that's near end |
22:27.09 | r0d3nt | sivana, the single span PRi card, the T101P |
22:27.19 | fugitivo | r0d3nt: do you have echo on all your calls? |
22:27.21 | sivana | so get the one with onboard echo can |
22:27.23 | r0d3nt | afrosheen, telepacific says that is far end... |
22:27.29 | sivana | we have the same issue, software isn't going to solve it |
22:27.34 | KranZ | from their end |
22:27.46 | r0d3nt | fugitivo, only on certain calls.. reproducable to certain phone #'s 100% of the time... usually mpower customers.... |
22:28.00 | r0d3nt | sivana $2500 out of my pocket.... |
22:28.02 | afrosheen | r0d3nt: well from their point of view it is I think...not sure, but generally if you can hear echo in your ear but the person you're calling can't, that's considered 'near end' |
22:28.15 | sivana | r0d3nt: $2500 no echo, $0 echo |
22:28.16 | sivana | pick |
22:28.20 | KranZ | you got a diff voip device? |
22:28.28 | r0d3nt | sivana, i didn't even get paid $2500 for this job. |
22:28.29 | KranZ | to test |
22:28.34 | afrosheen | that was my next suggestion, something besides a cisco to try |
22:28.45 | r0d3nt | KranZ, yes.. same effect... |
22:28.56 | fugitivo | KranZ: echo is not generated on his side |
22:28.57 | r0d3nt | grandsteam, cisco, softphone. |
22:29.01 | afrosheen | so everything regardless of sip hardware is echoing |
22:29.13 | KranZ | fugitivo, but couldnt it get cancelled if he's hearing it |
22:29.16 | r0d3nt | only certain phone calls |
22:29.22 | r0d3nt | mostly to mpower customers |
22:29.34 | r0d3nt | which go from Telepacific, to Sprint, over the tandem to mpower |
22:29.43 | KranZ | fugitivo, cancelled at the voip device... |
22:29.46 | sivana | r0d3nt: we have the same issue, software isn't going to solve it |
22:30.54 | r0d3nt | sivana, i agree.. which is why I want to kill digium |
22:30.54 | sivana | why |
22:30.54 | KranZ | r0d3nt: have you tried messing with the tx gains? |
22:30.55 | afrosheen | will those ciscos train any of it out on their own? |
22:30.55 | r0d3nt | since they just told telepacific that it would. |
22:30.55 | KranZ | on the voip device |
22:30.55 | r0d3nt | KranZ, YES....... |
22:30.55 | KranZ | turning them all the way down |
22:30.55 | r0d3nt | KranZ, there is setting for the voip device. |
22:30.55 | fugitivo | r0d3nt: did you set gains to 0? |
22:30.55 | r0d3nt | fugitivo, yes |
22:30.56 | r0d3nt | default is 0 / 0 |
22:30.56 | *** join/#asterisk Umaro (n=umaro@209.140.74.64) |
22:30.56 | KranZ | is there a -xx? |
22:30.56 | r0d3nt | we are currently @ 0 and -6 on tx |
22:30.56 | KranZ | hm |
22:30.59 | *** join/#asterisk TrainConductor (n=jdenton@208.255.225.67) |
22:31.11 | r0d3nt | we tried -3 -3 per TPACs instructions. |
22:31.15 | afrosheen | -6 on tx is dangerous sometimes, we had issues with a low tx and everyone cried and cried about how they were too quiet to callers |
22:31.24 | Umaro | Hey guys.. trying to get asterisk going on my new athlon 64 box, but having some troubles getting mpg123 compiled.. anyone here that has done it before on x86_64 and can give me some advice? |
22:31.37 | fugitivo | Umaro: use gentoo |
22:31.38 | KranZ | emerge mpg123 |
22:31.38 | tzanger | well -6 is 1/4 normal volume |
22:31.42 | r0d3nt | afrosheen, ya.. well telepacific lines are so fucking loud.. it wasn't even noticable. |
22:31.47 | afrosheen | lol |
22:31.51 | Umaro | fugitivo: ew ;) |
22:32.07 | afrosheen | tzanger: in an office setting people talk at 1/4 normal volume anyway |
22:32.14 | tzanger | yup |
22:32.23 | r0d3nt | we ran everything @ -9 and -6 to balance the audio on the meters..... |
22:32.24 | tzanger | I would really like if people spoke quieter even |
22:32.25 | afrosheen | so that's 1/2 as loud as normal overall :) |
22:32.28 | KranZ | r0d3nt: what cisco's are u usin? |
22:32.36 | fugitivo | Umaro: honestly, i use madplay, not mpg123 |
22:32.37 | FuriousGeorge | Umaro: google the output of the compile where it fails |
22:32.42 | r0d3nt | KranZ, 7940g's and ATA 186's |
22:32.50 | r0d3nt | sip flash 7.5 |
22:32.53 | r0d3nt | on the 79xx's |
22:32.54 | FuriousGeorge | you gotta be more specific |
22:33.03 | FuriousGeorge | not that i know the problem :) |
22:33.23 | afrosheen | r0d3nt: how long have you been working on this |
22:33.36 | Umaro | FuriousGeorge: yeah, I tried that.. :/ |
22:34.06 | r0d3nt | afrosheen, since july. |
22:34.12 | *** join/#asterisk obsidian-studios (n=obsidian@c-66-177-188-197.hsd1.fl.comcast.net) |
22:34.20 | r0d3nt | afrosheen, i'm about to quit my job and put bullet in my head. |
22:34.24 | afrosheen | r0d3nt: 3 months for less than $2500? dude I'd leave right now if it was me |
22:34.57 | r0d3nt | ya well |
22:35.02 | r0d3nt | i just sold another system |
22:35.05 | r0d3nt | for over 50 handsets |
22:35.07 | r0d3nt | and guess what |
22:35.10 | r0d3nt | they have mpower. |
22:35.25 | r0d3nt | i'm going to tell them to go buy a panasonic hybrid system and to never call me again. |
22:35.26 | afrosheen | so you gotta learn this one way or another huh |
22:35.27 | obsidian-studios | greetings, I got some funkyness going on. I have a Cisco 7960 with 6 sip channels configured for it, 1001-1006, Inbound works great, if I ring SIP/1001 I get the first line on the phone, 1006, the sixth. However no matter what button is pressed for dialing out it uses SIP/1006? |
22:35.27 | fugitivo | experience has a cost |
22:36.09 | afrosheen | obsidian-studios: is * registering it as only 1006? |
22:36.16 | *** part/#asterisk TrainConductor (n=jdenton@208.255.225.67) |
22:36.31 | obsidian-studios | afrosheen: * sees all 6 registered as peers? I am thinking maybe it |
22:37.02 | afrosheen | so you get 1001-1006 on sip show peers? |
22:37.06 | r0d3nt | so is this card really all that ???? |
22:37.07 | *** join/#asterisk TheCops (n=dump@206.248.136.146) |
22:37.11 | r0d3nt | it'll handle echo from the entire work |
22:37.13 | r0d3nt | world* |
22:37.28 | obsidian-studios | I am thinking the phone might be keeping a memory of the # dialed? I know if a call comes in and you return the call, the phone will use the same line in came in on. I know once you dial a #, it will show that # if you have dialed it before, but not sure if it uses the same line as before or what |
22:37.41 | RoyK | <PROTECTED> |
22:37.41 | KranZ | r0d3nt: is this an in office box, so no latency from the handsets? |
22:37.54 | r0d3nt | 2-3ms latency to the handsets from the server. |
22:37.59 | obsidian-studios | afrosheen: yes, sip show peers shows all 1001-1006, all six. Also inbound works fine, if a call comes in, I can map it to any line, and it will ring that line |
22:38.05 | fugitivo | the problem is not on his side |
22:38.16 | afrosheen | he's saying it's mainly mpower customers so there's a trunking issue there, sounds like an echocan card would fix it but it's anybody's guess where the problem lies |
22:38.54 | afrosheen | the problem is probably somewhere between telepacific and mpower like he said earlier |
22:39.56 | afrosheen | obsidian-studios: so * sees it as 6 devices..but does it see itself as 6? there's probably some config voodoo for outbound you may have missed |
22:40.31 | *** join/#asterisk tessier (n=treed@wsip-68-15-4-13.sd.sd.cox.net) |
22:41.04 | afrosheen | I know the polycom multiline phones have 2 sections for each line, inbound and outbound registry |
22:42.51 | obsidian-studios | afrosheen: I can call on various lines that are directly mapped to various sip channels, and make 1001-1006 ring on the phone by dialing SIP/1001-1006 |
22:43.01 | *** join/#asterisk bjohnson (n=bjohnson@i216-58-62-113.cybersurf.com) |
22:43.35 | *** part/#asterisk Samoied (n=Samoied@popeye.opens.com.br) |
22:43.43 | afrosheen | right, but your problem is with outbound right, you say it only dials out via 1006 |
22:43.43 | obsidian-studios | afrosheen: however from the phone, no matter what button is pressed for the corresponding line, it all uses SIP/1006 acting like a 1 line phone, it's not allowing outbound stuff to use the other lines |
22:43.52 | obsidian-studios | afrosheen: yes |
22:43.53 | afrosheen | gotcha |
22:44.33 | obsidian-studios | afrosheen: I know if a call comes in on line 1004, and you miss it, or receive it, navigate the phones menus for missed or received calls, and place the call again, it goes out the line it came in |
22:45.33 | obsidian-studios | I also know that if I dial 555-1212 once, as I start to dial 555, the phone will show the rest so I do not have to dial a # in memory. I am wondering that if I dial a # out of 1001, and then pick up 1006, if it uses 1001 again, since the previous call was placed out that line? |
22:46.30 | convey | Can someone tell me how to create an extension that forwards to a sip address for example exten => s,3,Dial(sip/happy@10.0.0.1)? |
22:47.00 | r0d3nt | we've decided to try and talk with digium some more, being that it was their recommendation for the cvs head of zaptel, perhaps they have more ideas, before we go and buy a card. But I told everyone that, that is our last viable option is to use the quad echo cancelling card.... |
22:47.02 | obsidian-studios | another issue, in the SIP*.cnf file sent to the phone, I reversed the order or line1 and line2, to where line1=1002, and line2=1001, and they still showed up in order on the phone? |
22:47.09 | afrosheen | obsidian-studios: you're going over my head now, I'm not sure how the feature set works on that model |
22:47.26 | obsidian-studios | afrosheen: I am wondering if it's firmware version or something |
22:47.39 | *** join/#asterisk HaHaOok (n=norman@60-240-188-146.tpgi.com.au) |
22:47.43 | obsidian-studios | afrosheen: pretty sure it's he newest release, and it's very weird |
22:47.45 | r0d3nt | also, in talking with my other technician helping on the new pbx sale, it was decided a few days ago to only sell the quad echo cancel cards and to inform the client of the additional cost before they sign a contract and give deposit. |
22:48.30 | afrosheen | obsidian-studios: I'd consider rolling it back a version |
22:48.32 | r0d3nt | my execution has been delayed once again. |
22:48.48 | obsidian-studios | afrosheen: might have to, getting the firmware was a pain, got to get a smartnet contract so I can get them directly |
22:49.05 | wunderkin | r0d3nt: what card are you using now? |
22:49.12 | afrosheen | obsidian-studios: you'll run into fellow cisco users here that may email you images, I've seen cd's for sale on ebay also |
22:49.14 | obsidian-studios | afrosheen: pretty sure it's someone to do more so with the phone, than config, config is fairly straight forward |
22:49.17 | r0d3nt | single span T/E/1/PRI card |
22:49.30 | r0d3nt | x401p ? |
22:49.51 | wunderkin | r0d3nt: oh ok, i was going to say.. if you had the 405 or 410 you could just add the module.. oh well |
22:50.01 | r0d3nt | damn |
22:50.02 | r0d3nt | nope |
22:50.10 | r0d3nt | and i sold all my quad pri cards i had... |
22:50.15 | r0d3nt | of which i've purchased 3. |
22:50.24 | wunderkin | ok |
22:50.38 | obsidian-studios | afrosheen: it's also weird that the phone displays a different order than I tell it to in the cnf file, I am going to play with that a bit and see as well |
22:50.43 | r0d3nt | i didn't know it was a module like that.. it looked like a daughter board of sorts tho' |
22:50.54 | KranZ | wunderkin: add the module? |
22:51.05 | obsidian-studios | afrosheen: should show 1002, 1001, 1003-6 on the phone, but they are in order? |
22:51.19 | afrosheen | obsidian-studios: if what you're mapping and what it's displaying are different, that's definitely a problem |
22:51.35 | websae | any AstBILL users here at all? |
22:51.40 | r0d3nt | i would imagine that it would require firmware updates.... |
22:51.45 | afrosheen | then again maybe cisco thinks they're smarter than you and 'adjust' that automagically |
22:51.45 | wunderkin | KranZ: yes, you can just add the echo can module.. its just an addon card |
22:51.56 | KranZ | like a daughterboard? |
22:52.00 | wunderkin | yes |
22:52.03 | KranZ | link? |
22:52.52 | obsidian-studios | afrosheen: check this out http://pastebin.ca/24149 |
22:54.20 | obsidian-studios | afrosheen: makes no sense there to me, I assume the phone must want to put them in order, but at the same time it's mapping all outbound to 1006? Maybe a bug in the firmware? |
22:54.49 | *** join/#asterisk buddah (n=djbrianc@67.110.253.129) |
22:55.06 | wunderkin | KranZ: not sure, it doesnt really say on the website |
22:55.07 | buddah | anyone know if there is a way to use call forwarding with polycom ip 500/501s if i'm using g711? |
22:55.22 | buddah | i get an error message spamming in * and it screeches on the line of the caller when it does |
22:55.43 | wunderkin | KranZ: you would need to ask digium sales |
22:56.13 | KranZ | hmm |
22:56.31 | KranZ | anyone already know the price on the echo can module?? |
22:57.06 | drumkilla | it is an addon card, but it is not a user servicable part |
22:57.22 | drumkilla | it requires a change of firmware. however, i believe it can be added to an existing card if you ship it back |
22:57.42 | KranZ | ic |
22:58.35 | KranZ | they'd shit if a tool and firmware were leaked, potential lost revenue |
22:58.45 | *** part/#asterisk shido6 (n=shido@d221-68-210.commercial.cgocable.net) |
22:58.45 | websae | anyone here used ASTBILL? |
22:58.47 | KranZ | guess i wait till i get a spare card |
22:59.13 | KranZ | but its nice to know its not obsolete quite yet |
22:59.16 | ronaldl79 | Which TOS setting do you all prefer for SIP? |
22:59.22 | *** join/#asterisk hound (n=MrHound@tor/session/x-430f00bf6e96a805) |
22:59.23 | jayk- | how do i check out -head using cvs? |
22:59.30 | ronaldl79 | Currently, I am using 'lowdelay' -- which is best? |
22:59.37 | hound | So is the backdoor in asterisk underway for US law enforcement ? |
22:59.41 | wunderkin | once your card is upgraded to 2nd gen, you can upgrade the firmware yourself.. although ive never seen anywhere to download it.. or how to do that |
22:59.47 | ronaldl79 | Speaking of CVS, is 'CVS HEAD' a daily build of Asterisk? I never bothered to find out. |
22:59.56 | hound | or is it already there with monitor! good planning |
23:00.00 | jayk- | ronaldl79: i thought there were nightly builds somewhere, but they wern't being updated |
23:02.26 | ronaldl79 | BTW, guys, I demo'd an * box today for a non-profit -- The director was THRILLED! We covered voicemail, transfer and parking, calling, and IVR. |
23:02.41 | afrosheen | buddah: on our system with the polycoms we just do a *71 and enter a number, boom, call forwarding is done |
23:03.08 | AgiNamu | hound, or ChanSpy? |
23:03.41 | *** join/#asterisk kFuQ (n=somedude@c-67-185-114-199.hsd1.wa.comcast.net) |
23:03.45 | websae | anyone here use MySQL and ASTERISK together...the CDRs into a MySQL database...? |
23:04.16 | afrosheen | buddah: when the call hits the phone the phone tells asterisk to call the forwarded number I believe |
23:04.31 | glm2k | websae: i do. |
23:04.39 | generalhan | why doesnt Chan_Spy work with 1.0.9 ? this sux |
23:05.20 | *** join/#asterisk redder86 (n=lee@gateway.howardsilvan.com) |
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23:05.31 | KranZ | 6pm |
23:05.34 | file[laptop] | generalhan: backport it if you want it in stable |
23:05.59 | generalhan | file: what do you mean ?> |
23:06.01 | *** join/#asterisk Ariel_ (n=Ariel@dsl-20-177.cofs.net) |
23:06.23 | file[laptop] | well chanspy uses stuff that's in the core of asterisk... that has been put there for chanspy to use |
23:06.27 | *** join/#asterisk bzbw (n=wlwzhang@ip67-153-142-109.z142-153-67.customer.algx.net) |
23:06.37 | file[laptop] | you'd have to backport that stuff, plus the application... |
23:07.38 | generalhan | awww, i wouldnt know how to do that. My boss just came to me today and said that we wanted to be able to listen in to our new guys for training purposes and i told him that i couldnt do it. he seemed to just accept it ! |
23:07.53 | AgiNamu | hmm, if I have 2 calls in a Polycom, do I press "Join" or "Conference" to connect em both in a 3way? |
23:07.58 | *** join/#asterisk Rez (i=lorez@freenode/staff/lorez) |
23:09.23 | JerJer | try both |
23:09.33 | JerJer | see what works better for you |
23:09.36 | AgiNamu | join does it whew |
23:09.43 | AgiNamu | yea, I have 2 live calls :) |
23:09.57 | AgiNamu | where you at VON? |
23:10.05 | AgiNamu | s/where/were |
23:10.08 | JerJer | no - too damned expensive |
23:10.40 | AgiNamu | get digium to sponsor you? ;) |
23:11.11 | darkskiez | generalhan: if u are using zap, try zapscan |
23:11.36 | generalhan | hmmm |
23:15.26 | [hC] | Anyone have an idea why IAX indication requests and the indications themselves appear to be happening about 10 seconds apart, according to my logs? |
23:15.28 | [hC] | http://pastebin.ca/24147 |
23:15.33 | [hC] | <PROTECTED> |
23:15.51 | bzbw | anyone tried to set up * with line sharing? It is based on draft-anil-sipping-bla-02.txt for SIP client. |
23:16.21 | file[laptop] | [hC]: is there a problem? |
23:17.28 | *** join/#asterisk Johnsie (n=john@acs-24-154-53-217.zoominternet.net) |
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23:17.40 | bzbw | I talked to Grandstream, they said their next release will support line sharing. |
23:18.23 | afrosheen | the latest release of AMP has some cool stuff in it like separating users from devices |
23:18.52 | afrosheen | you sit down at any phone, dial *11 and your extension and you can take/make calls from your normal ext number |
23:19.34 | bzbw | what is AMP? |
23:19.49 | websae | anyone here use asterisk and MySQL?? for like CDRs, etc...??? looking for some help getting it configured right |
23:20.30 | *** join/#asterisk stkn (i=nobody@gentoo/developer/pdpc.active.stkn) |
23:20.56 | Johnsie | AMP = Asterisk Management Portal |
23:21.11 | redder86 | What's the difference between "congestion" and "busy"? |
23:21.11 | Johnsie | Basically a web suite to control Asterisk, automate configurations to an extent, etc. |
23:21.23 | Johnsie | Congestion = fast busy/reorder tone |
23:21.27 | [hC] | file[laptop]: well yeah but im trying to figure out what could be causing it. kind of a wild goose chase at this point. im getting one way audio drop outs on some calls. cant figure out the cause yet, so im trying to eliminate everything that shows up in my logs just incase. |
23:21.31 | Johnsie | Busy = slow busy/standard busy |
23:21.33 | fugitivo | websae: mysql is evil |
23:21.36 | [hC] | file[laptop]: usually lasts 5-10 seconds, one way |
23:22.51 | redder86 | Johnsie: so congestion will cause the dialplan to increase by 101 steps, right? And not busy? |
23:22.59 | bzbw | Johnsie: so you can configure extensions and dial plan without using editor to edit either "extensions.conf" and other files? |
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23:26.30 | ronaldl79 | What are the recommended TOS/QOS bit settings for *? I'm trying 0x18 now (low delay and throughput) |
23:27.37 | *** join/#asterisk Katty (n=katrina@68-112-15-110.dhcp.cpgr.mo.charter.com) |
23:28.06 | Katty | mew. |
23:28.36 | afrosheen | bzbw: basically, google amp asterisk and you'll see what it can do |
23:29.40 | mrfrenzy | are there any debian packages for amp? |
23:29.49 | ariel_ | mrfrenzy, you can ask on #amportal |
23:29.56 | ariel_ | I think they were working on one |
23:30.09 | Katty | ariel_: :> |
23:30.09 | afrosheen | lol yeah right, that's a KOA kampground in there |
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23:30.31 | ariel_ | Katty, hello |
23:31.26 | ariel_ | hope your day is going well. |
23:31.26 | mrfrenzy | thx ariel_ will do |
23:31.27 | Katty | ariel_: mostly (= |
23:31.27 | ariel_ | mine has been pretty shitty |
23:31.27 | Katty | :< |
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23:31.27 | ariel_ | too much to do too little time to do it in. |
23:31.48 | Dr_Ray | can, ick, have you tried the pouches? |
23:31.53 | Katty | ariel_: yeh, i know that feeling :< |
23:32.02 | afrosheen | ewww, I don't eat ANYTHING from pouches |
23:32.18 | Dr_Ray | the tuna in the pouch is nice |
23:32.33 | Katty | ariel_: :<<<<< |
23:32.37 | afrosheen | I was just joking..the only tuna I'll eat is yellow fin on sushi |
23:32.38 | Johnsie | redder86: I am not sure if congestion will do that or not, I don't believe so...it's just a means of error indication. |
23:33.01 | AgiNamu | if a T1 is delivered to a smartjack in the wall , do I use a T1 crossover or a straighthru rj45 |
23:33.06 | Johnsie | bzbw: I believe so, yes... if you do a Google search, you can see AMP in action on their SourceForge site. |
23:33.10 | ariel_ | Katty, family is not here. But thanks. (There at my Mom's) I am stuck working. |
23:33.16 | Johnsie | Sorry for the delay, I have about 20 windows going on here. |
23:35.18 | ariel_ | argh do people really belive it when they get an email saying. Be a travel agent in 1 week make lots of money just send us 1495 for cource. |
23:35.36 | redder86 | Johnsie: turns out that with busy the call goes to the 101th step after the current one. With congestion it will go to the next step in the dialplan. |
23:36.08 | Johnsie | redder86: Oh okay...my bad. Thank you for correcting me, though. |
23:36.39 | [hC] | wow, cool netsplit. |
23:36.41 | azzie | AgiNamu, whatever brings the T1 up |
23:36.47 | [hC] | file[laptop]: does it make sense that the timestamps would be skewed like that? |
23:38.24 | *** join/#asterisk Tiveron (n=someone@66.146.140.5) |
23:38.52 | Tiveron | has anybody got a couple minutes to answer a couple questions? |
23:39.12 | AgiNamu | azzie, ok :).... |
23:39.17 | file[laptop] | I don't see where you say the timestamps are being skewed... |
23:39.24 | ariel_ | Tiveron, hello and maybe |
23:39.44 | azzie | AgiNamu, the other cable would keep T1 in "loss of signal" state... |
23:39.51 | Tiveron | ariel: that's nice and definite :P |
23:39.53 | file[laptop] | [hC]: iax2 debug is a good thing |
23:40.54 | AgiNamu | azzie, yea, that's what I'm seeing with a straighthru. T1 crossover brings layer1 up |
23:40.54 | *** join/#asterisk roulduke (i=j5zw1bxx@p508D1BF1.dip0.t-ipconnect.de) |
23:40.57 | [hC] | oh boy, i cant iax debug for a particular peer! |
23:40.57 | [hC] | :) |
23:41.13 | afrosheen | sure you can, that's what grep is for |
23:41.15 | afrosheen | ;) |
23:41.23 | [hC] | haha :) |
23:41.35 | file[laptop] | I have an evil patch that may not apply that does iax2 debug peer and iax2 debug ip... |
23:41.47 | [hC] | im running cvs head last night |
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23:42.56 | [hC] | ...from. last night. |
23:42.56 | [hC] | gotta love freenode. |
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23:43.01 | Tiveron | can anyone tell me if there's any way to propegate a Message Waiting Indicator originating from a Zap FXO channel to the rest of my FXS/SIP clients? |
23:43.19 | [hC] | file: I could try incorporating the patch, if you wanna send it? |
23:43.54 | file[laptop] | gotta find it |
23:44.18 | file[laptop] | http://neutrino.file-radio.com/asterisk/iax2_specific_debug.diff |
23:44.36 | [hC] | heh. damned sure that lilo has nothing better to do than to sit at irc waiting for the next reason to wall. |
23:44.46 | [hC] | file[laptop]: thanks. i'll give it a go |
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23:48.57 | Tiveron | how do I enable call waiting on asterisk? |
23:50.15 | file[laptop] | on a zaptel channel? |
23:50.40 | file[laptop] | what technology/protocol... you have to be specific |
23:51.21 | Tiveron | on any channel. I've got a Zap FXO channel as my trunk, and a number of SIP and Zap FXS extensions |
23:51.42 | file[laptop] | it's not up to Asterisk to do call waiting on stuff like SIP |
23:51.48 | file[laptop] | it's up to the ATA or Phone |
23:52.29 | file[laptop] | on zaptel stuff it's configured in zapata.conf |
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23:53.39 | Tiveron | ok... I'll look into that... what about if I get a call waiting signal through my PSTN trunk? I can hear the tone on my SIP hardphone (GXP-2000), but can't flash to it. |
23:53.58 | hardwire | my mwi is blinking |
23:54.03 | hardwire | it doesn't need to be |
23:54.06 | hardwire | damnit |
23:54.06 | file[laptop] | you have to setup like an extension that flashes it |
23:54.25 | Tiveron | I don't follow... |
23:54.41 | file[laptop] | there's an application, Flash, that flashes zaptel channels |
23:54.46 | file[laptop] | that's how you have to do it |
23:55.01 | Tiveron | oh. an asterisk addon you mean? |
23:55.09 | file[laptop] | no... it's an included application... |
23:55.21 | file[laptop] | you can't flash from your phone, because you're not actually flashing the line |
23:55.45 | file[laptop] | you have to setup another extension that goes to the flash extension that will flash the zaptel line, then go back to your original call to the zaptel line |
23:56.13 | file[laptop] | and this is why analog through Asterisk to a SIP phone is bad, mmmk? |
23:56.25 | [hC] | file[laptop]: hate to sound ignorant here, but do you have any suggestions on what to look for in an iax2 debug, to notice errors? |
23:56.37 | Tiveron | heh. understood. |
23:56.42 | file[laptop] | [hC]: not really, I haven't done an iax2 debug in a very very long time |
23:56.46 | Tiveron | one more question... |
23:57.05 | [hC] | file[laptop]: heh.. yeah.. and because there's so much of it, going thru every line is a pita. :) |
23:57.17 | file[laptop] | I'm much more at home with SIP |
23:57.28 | [hC] | me too. |
23:57.35 | Tiveron | my PSTN trunk comes from my ILEC, and can generate a MWI signal. Is there anyway to recognize that with asterisk and generate an internal MWI signal? |
23:57.44 | [hC] | might just try switching from iax2 to sip, and see if the problem goes away first. |
23:58.03 | hardwire | heh |
23:58.08 | [hC] | ive spent almost 2 weeks on this now, and still havent nailed down the cause. |
23:58.10 | [hC] | hardwire: shut. up. |
23:58.13 | hardwire | all these subscriptions for SIP to my phone make my phone wiggle |
23:58.13 | [hC] | :) |
23:58.22 | file[laptop] | Tiveron: not really, no... |
23:58.26 | [hC] | wiggle?? |
23:58.30 | hardwire | jitter |
23:58.31 | hardwire | poop |
23:58.37 | [hC] | oh |
23:58.57 | [hC] | i envisioned a phone wiggling around your desk.. remided me of pager races, back in the day,.. |
23:58.59 | azzie | Tiveron, give up your fxo line and port it to some SIP provider... |
23:59.17 | Tiveron | Reccomendations for NA proviers? |