00:00.05 | _Vile | harlequin, look at generating "call files", in /var/spool/asterisk/outgoing |
00:00.15 | harlequin516 | dialout commandline control? |
00:00.21 | cripito | :)) |
00:00.43 | harlequin516 | Ah right!!!Thanks _Vile, I remember reading this... |
00:01.04 | _Vile | Harle - http://www.voip-info.org/wiki-Asterisk+tips+Wake-Up+Call+PHP |
00:01.05 | _Vile | may help |
00:01.28 | tzanger | harlequin516: you did not read my message |
00:04.56 | Dr-Linux | tzanger: i faced a problem today, when i call reciever hears me very low, but i can hear him fine, what things need to be chacked? |
00:05.00 | *** join/#asterisk BhaalWK (i=bhaal@freenode/staff/bhaal) |
00:05.35 | *** join/#asterisk newl (n=newlook@203-59-61-62.dyn.iinet.net.au) |
00:05.38 | tzanger | sounds like your txgain is set too low |
00:06.29 | *** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin) |
00:06.33 | _Vile | tzang, ltns -- ever played with routing on diversion headers? |
00:07.25 | _Vile | is bkw still around? |
00:07.54 | tzanger | _Vile: nope |
00:08.28 | *** join/#asterisk Vco (n=Vco@S01060050da6df072.sc.shawcable.net) |
00:08.32 | harryvv | seen Ariel |
00:08.37 | _Vile | seen bkw- |
00:09.58 | Corydon-w | bkw decided he'd be better off by himself. His choice. |
00:10.14 | mog_home | heh |
00:10.56 | _Vile | his time of the month already? |
00:11.52 | Corydon-w | More like, time of the year. It's been several months |
00:11.58 | mog_home | ow |
00:12.09 | mog_home | no he is in cough opbx if you want him |
00:12.10 | _Vile | hm, i've been gone for 4-5.. maybe 6 |
00:12.29 | _Vile | interesting |
00:12.31 | mog_home | that sounds about right then |
00:12.53 | Corydon-w | He basically renamed a whole bunch of APIs... and left it at that |
00:12.55 | _Vile | branch? |
00:13.06 | harlequin516 | tzanger: Did not get a response from you, looked like you tried to respond but had a mixup |
00:13.14 | mog_home | hey they have autoconf.... |
00:13.30 | _Vile | that sucks, he's a good developer |
00:13.35 | Corydon-w | Wow, they can autoconf... |
00:13.45 | mog_home | heh whatever we have make menuconfig ^_^ |
00:13.51 | Corydon-w | Boy, that's a good reason to step back 4 months in development time |
00:14.01 | _Vile | did he branch? |
00:14.13 | Corydon-w | Yep |
00:14.20 | Corydon-w | Fork, actually |
00:14.30 | Corydon-w | but it might as well be a branch |
00:15.00 | mog_home | i would have disagreed but when they played catch up when we went 1.2 made me think again |
00:15.02 | _Vile | ouch |
00:15.49 | _Vile | well shit, who's going to answer my questions now? ;) |
00:15.55 | mog_home | ask |
00:15.58 | mog_home | and we might answer |
00:16.04 | _Vile | trying to route on diversion sip headers |
00:16.11 | _Vile | sip diversion headers |
00:16.16 | _Vile | not RDNIS |
00:16.19 | rob0 | I'll answer but it will be wrong :\ |
00:16.27 | rob0 | 42? |
00:16.31 | file[laptop] | 42! |
00:16.31 | _Vile | well, RDNIS, but the "reason" part |
00:16.45 | _Vile | reason=no-answer |
00:16.51 | _Vile | answer to life and everything?!?! |
00:17.05 | Corydon-w | Don't forget the universe |
00:17.08 | _Vile | or reason=user-busy |
00:17.10 | _Vile | oh yeah |
00:17.21 | _Vile | "Diversion: <sip:1231230001@123.12.0.123>;reason=no-answer" -- example header |
00:17.39 | _Vile | I can get the 1231230001 |
00:19.28 | Dr-Linux | <tzanger> sounds like your txgain is set too low |
00:19.40 | Dr-Linux | tzanger: but the same everything was working fine |
00:19.51 | Dr-Linux | txgain = 0.0 |
00:19.59 | Dr-Linux | and thats in zapata.conf |
00:20.09 | Corydon-w | _Vile: have you filed a bug report? |
00:20.13 | Dr-Linux | i m not saying problem with zap channels |
00:23.51 | Dr-Linux | txgain is related only with zap channels/ or its matter if we are using only softphones? |
00:26.54 | *** join/#asterisk Vco (n=Vco@S01060050da6df072.sc.shawcable.net) |
00:27.32 | _Vile | Cory, of course not |
00:27.44 | _Vile | I don't know if it's a bug or not |
00:27.50 | jimmy_deanPB | Hmmhesays, getting this error when trying to compile asterisk-1.2.1 on debian stable: "configure: error: termcap support not found" |
00:28.04 | _Vile | seems like a feature request, if it isn't available? trying to figure out if that reason code is routable before I file anything |
00:28.04 | jimmy_deanPB | the only package with "termcap" in it I installed |
00:28.36 | jimmy_deanPB | s/Hmmhesays/hmm |
00:28.54 | _Vile | Scenario is: Sipura -> MetaSwitch -> Sipura (no answer) -> MetaSwitch (routes on NA) -> Asterisk |
00:28.57 | _Vile | Asterisk picks up |
00:29.15 | _Vile | but I need to figure out how to determine busy or no answer |
00:29.42 | _Vile | if it's not there, I could probably add the code and submit a patch |
00:31.26 | Corydon-w | _Vile: if you file a feature request on the bugtracker in the next 5 minutes, I'll upload a patch. |
00:31.39 | _Vile | OK |
00:40.40 | Corydon-w | 10 minutes later, still no feature request... |
00:40.42 | litage | if userA is connected to *boxA and makes a call to userB who is connected to *boxB, and everyone's using h323, which * box would need an h323 gatekeeper? |
00:41.09 | Corydon-w | Neither |
00:41.54 | Corydon-w | A gatekeeper is never really needed... it's a convenience when you don't know which gateway you need to connect to |
00:42.05 | *** join/#asterisk Sedorox (i=brandon@smartserv/cna/Sedorox) |
00:42.21 | Corydon-w | but you can go gateway to gateway just fine and never encounter a gatekeepr |
00:42.48 | _Vile | Cory, I hadn't logged into Mantis in 6 months -- took me a couple of minutes to grab my pw |
00:42.49 | _Vile | sorry |
00:42.50 | _Vile | http://bugs.digium.com/view.php?id=5995 |
00:42.54 | litage | Corydon-w: ah i see. but you can only go gw-gw IFF you know which gw you want to connect to? |
00:43.28 | Corydon-w | Correct |
00:43.49 | litage | Corydon-w: awesome, that *really* clears up that question =) |
00:43.50 | Corydon-w | A gatekeeper is essentially a directory service |
00:44.35 | litage | Corydon-w: so a gk needs to know about all available gateways. do gateways register with 1+ gatekeepers? |
00:45.03 | Corydon-w | They can |
00:45.51 | Corydon-w | Typical installs, though, you have a single gatekeeper, and you hardcode that gatekeeper to send all calls starting with e.g. 1615 to a certain gateway, 1423 to another gateway, etc. |
00:45.51 | robl^ | and the gatekeep has to know about the dungeon master and have the keys to the bridges. :)\ |
00:47.07 | _Vile | u rock |
00:47.14 | _Vile | thx |
00:47.16 | litage | Corydon-w: is gnugk the most "popular"/widely used/"best" gk? |
00:48.03 | Corydon-w | You'd have to ask someone who actually gasp enjoys using h323 |
00:48.20 | litage | Corydon-w: hahah. why does everyone i talk with hate h323? |
00:48.32 | Corydon-w | because it's an overly complex protocol |
00:48.36 | }btorch{ | how can I convert a time like 10:15:00 to UTC format ? I guess if I enter 10:15:00 to the SAY TIME command that won't work |
00:48.39 | }btorch{ | right ? |
00:49.16 | Corydon-w | It implements every single condition that could possibly occur on a PRI, even though most of those conditions will never occur on an IP connection... |
00:49.30 | litage | }btorch{: how about an agi that runs the 'date' command? |
00:49.49 | litage | hrmm |
00:50.15 | }btorch{ | no you are talking about shell dat cmd right |
00:50.17 | Corydon-w | Unfortunately, the solution for H.323 was also designed by committee, and suffers from some of the same maladies as H.323: SIP |
00:50.31 | litage | i'm stuck with h323 for my protocol, and g729 for my codec, for 98% of my comms |
00:50.39 | Corydon-w | However, SIP is still a whole lot better than H.323 |
00:50.40 | }btorch{ | now you are talking about shell date cmd, right? |
00:50.43 | *** join/#asterisk wunderkin (i=kev@ip68-226-113-228.ph.ph.cox.net) |
00:50.53 | litage | }btorch{: yes |
00:51.37 | }btorch{ | litage: You see I get a pre- determined time from a database field |
00:52.04 | }btorch{ | can the date command conver the time value I get into UTC ? |
00:52.15 | }btorch{ | checking man page |
00:52.17 | litage | }btorch{: i can't remember off-hand. ``man date'' |
00:52.34 | rob0 | date -u |
00:52.34 | litage | }btorch{: you can convert times and timezones and whatnot in perl, too |
00:53.44 | Corydon-w | date is essentially just a command line frontend to strftime, anyway |
00:55.12 | }btorch{ | tried -u with date looks the same as date |
00:55.24 | }btorch{ | tried date -u -d 10:15 |
00:56.35 | }btorch{ | isn't it wiered though that SAY time only takes the time in seconds ? |
00:56.38 | Corydon-w | Try + |
00:57.40 | litage | }btorch{: as Corydon-w said, don't forget the mandatory "+" prefix on the date string |
00:58.39 | rob0 | Wed Dec 14 10:15:00 UTC 2005 |
00:59.18 | }btorch{ | Ithat's what I got |
00:59.34 | rob0 | what did you want/expect? |
00:59.42 | *** join/#asterisk adker (n=adker@67-51-239-152.dsl1.glv.ny.frontiernet.net) |
01:00.26 | rob0 | try date --help to see the ways you can format it |
01:00.27 | }btorch{ | well here ... according to "SAY TIME time escaped_digits " time is the number of seconds elapsed since 00:00:00 on january 1,1970 |
01:00.30 | Dr-Linux | txgain is related only with zap channels/ or its matter if we are using only softphones? |
01:00.44 | rob0 | date +%s (in GNU date only) |
01:01.28 | }btorch{ | just saw that on the man page |
01:02.07 | }btorch{ | cool thank you all |
01:02.15 | }btorch{ | got test it now |
01:04.25 | *** join/#asterisk techie (n=gus@adsl-068-209-242-072.sip.mia.bellsouth.net) |
01:05.48 | *** part/#asterisk techie (n=gus@adsl-068-209-242-072.sip.mia.bellsouth.net) |
01:05.51 | jimmy_deanPB | Dr-Linux, only zap channels |
01:06.54 | *** join/#asterisk ncconquer (n=info@203.113.172.79) |
01:06.59 | Dr-Linux | jimmy_deanPB: then what could be the problem, when i call i can hear fine, but reciever hears me very low? |
01:07.03 | ncconquer | hello |
01:07.12 | ncconquer | i've problem w asterisk |
01:07.32 | ncconquer | when i try: genzaptelconf -s -d |
01:07.40 | ncconquer | <PROTECTED> |
01:07.54 | ncconquer | Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters. |
01:08.15 | litage | voip-info.org (voip-info.org/wiki/index.php?page=Asterisk+sip+type) says that a peer sends calls, and a user receives calls. does that mean that a peer can't receive calls, and that a user can't send calls? |
01:08.32 | *** join/#asterisk g0mb0 (n=test@external.micom.mng.net) |
01:08.58 | *** join/#asterisk riddlebox (n=james@24-217-15-91.dhcp.stls.mo.charter.com) |
01:09.21 | ncconquer | plz help |
01:09.36 | litage | ncconquer: please don't forget what i told you yesterday |
01:09.52 | ncconquer | ? |
01:09.56 | ncconquer | how litage |
01:10.40 | litage | [12:25] litage ncconquer: continually asking for help will only annoy people |
01:11.10 | robl^ | can someone help me annoy people? |
01:11.16 | litage | ncconquer: after asking a proper question (which you didn't do; you only said you have a "problem"), you wait for a response. you don't say "please help" |
01:11.41 | ncconquer | ok |
01:12.10 | litage | great :) |
01:12.32 | ncconquer | do u know how about my error? |
01:13.08 | litage | ncconquer: i don't. however, all you gave us was an error. we need more information than that to help you. what did you do? what did you expect to happen? what happened? etc.. |
01:13.45 | *** join/#asterisk asterboy (n=Snake@S01060204ee2b6007.ed.shawcable.net) |
01:14.04 | ncconquer | can u login my system? |
01:14.12 | ncconquer | u can know how about |
01:14.50 | *** join/#asterisk MikeJ[Laptop] (n=ircatjer@d14-69-8-30.try.wideopenwest.com) |
01:14.57 | jimmy_deanPB | Dr-Linux, you are using all SIP based things, no analog phone lines? |
01:16.03 | Dr-Linux | jimmy_deanPB: yes |
01:16.38 | Dr-Linux | jimmy_deanPB: it was workinf fine before :S |
01:16.55 | jimmy_deanPB | Dr-Linux, it could be your SIP phone then...bad quality or something...I'm not sure how to adjust the volume though for SIP connections |
01:16.56 | Dr-Linux | i'm not sure if my net speed were slow this time or what:S |
01:17.15 | Dr-Linux | jimmy_deanPB: its X-Lite |
01:17.22 | asterboy | What is the cheapest *unlimited* local and long distance number going? |
01:17.32 | *** join/#asterisk bweschke (n=bweschke@24-75-229-26.atlaga.adelphia.net) |
01:17.44 | asterboy | Vonage has $40/month...anyone cheaper? |
01:17.58 | De_Mon | broadvoice is only $10 or something |
01:18.20 | asterboy | $10 is nice, is that all the features and long distance? |
01:18.32 | De_Mon | yeah |
01:18.55 | De_Mon | support isn't that great, but if it works who cares! |
01:18.56 | asterboy | no, that $10 is in state only. |
01:19.08 | asterboy | $20 for world |
01:19.14 | De_Mon | nation wide = longdistance |
01:19.34 | De_Mon | overseas != long distance |
01:20.18 | asterboy | broadvoice is only $10 for in state, not nation wide |
01:20.26 | asterboy | still dam cheap |
01:20.51 | De_Mon | I'm trying to setup some x-lite sip softphones, followed the wikis and howtoos, yet I'm getting Registration from '<sip blalblabla>' failed for 'myip' |
01:20.59 | asterboy | why would you get DID? |
01:21.31 | asterboy | guess if you have low volume on appearances it would work out ok. |
01:21.49 | De_Mon | running 1.2 |
01:21.52 | De_Mon | .0 |
01:22.14 | asterboy | otherwise you might as well pay only $10 and talk all you want and have inbound an outbound capabilities. |
01:22.29 | asterboy | This landscape is changing fast. |
01:23.51 | asterboy | no doubt we have already reached 'singularity' |
01:24.17 | *** join/#asterisk apardo (n=apardo@29.Red-81-39-85.dynamicIP.rima-tde.net) |
01:24.52 | asterboy | http://en.wikipedia.org/wiki/Technological_singularity |
01:24.55 | De_Mon | theres a Bring Your Own Device rate of $6 |
01:26.46 | asterboy | nice. |
01:27.12 | asterboy | for $6 DID is almost dead. |
01:27.37 | *** join/#asterisk rezEdit (n=rezEdit@zapdos.omnigroup.com) |
01:30.45 | riddlebox | hello, I am trying to get a sip phone working with asterisk but I get the registration fails due to a username/authname mismatch? |
01:32.03 | rezEdit | riddlebox: what kind of phone? |
01:32.13 | riddlebox | xten xlite |
01:32.25 | rezEdit | I am using that with Asterisk as well |
01:32.48 | riddlebox | how did you set it up in sip.conf? |
01:33.43 | rezEdit | [700] |
01:33.43 | rezEdit | type=friend |
01:33.43 | rezEdit | nat=yes |
01:33.43 | rezEdit | incominglimit=3 |
01:33.43 | rezEdit | username=700 |
01:33.43 | rezEdit | secret=1111 |
01:33.45 | rezEdit | host=dynamic |
01:34.12 | rezEdit | Then Authorization User and Username are both 700 in X-Lite |
01:35.37 | rezEdit | X-Lite seems to be pretty unstable for me, unfortunately. Crashes a lot. |
01:35.51 | harryvv | never has for me. always stable. |
01:36.07 | riddlebox | I just want to get something running lol |
01:36.59 | harryvv | go buy a cheap ip phone |
01:37.16 | rezEdit | X-Lite will sometimes crash on launch, several times over, then be fine. Other times, after several crashes it automatically re-creates it's preferences |
01:37.31 | rezEdit | I just use it for testing, have Polycom 501's for real use |
01:38.06 | *** join/#asterisk los415 (i=los415@los.race.com) |
01:38.32 | rezEdit | I am having an issue with ChanIsAvail(), which doesn't seem to be working when I use it to test a SIP extension for availability. Anyone else having issues? For me it always returns like the channel is free when in fact, it is not. |
01:39.02 | rezEdit | I am using it like: exten => 100,n,ChanIsAvail(SIP/224|sj) |
01:40.14 | rezEdit | and it always goes to n+1, never to n+101, even when the phone is on another call |
01:41.21 | harryvv | rez, only complaint i dont like about the polycoms is no lit display. |
01:41.54 | rezEdit | harryvv: yeah for sure. I work on the system late at night sometimes, with very little light around and it sucks |
01:41.58 | harryvv | I dont know if thats a optional feature of was never considered when thay were designed. |
01:42.14 | MstlyHrmls | I just turn a light on... |
01:42.17 | rezEdit | I don't think it's available at all. |
01:42.26 | rezEdit | the 4000 conf phone has a backlight |
01:42.27 | MstlyHrmls | you could get an IP 4000s :-) |
01:42.34 | harryvv | I have a bright halogen light on behind me and the display is still on the dark side. |
01:42.55 | MstlyHrmls | I've just got a desk lamp, and that works fine |
01:43.10 | harryvv | ip4000? |
01:43.44 | MstlyHrmls | SoundStation IP 4000 |
01:43.55 | MstlyHrmls | conference phone version of the SoundPoints |
01:44.28 | rezEdit | yeah the 4000 has the backlight, but the screen is also very tiny |
01:44.37 | los415 | anyone ever setup gr303 in asterisk ? |
01:45.03 | MstlyHrmls | rezEdit: yeah, and you have to lean over it to see anything |
01:45.16 | *** join/#asterisk mog_work (n=mogorman@gateway.digium.com) |
01:45.40 | rezEdit | MstlyHrmls: yup. thankfully we don't use it all that often |
01:48.38 | litage | voip-info.org/wiki/index.php?page=Asterisk+sip+type says that a peer sends calls, and a user receives calls. does that mean that a peer can't receive calls, and that a user can't send calls? |
01:49.06 | *** join/#asterisk klictel (n=klictel@modemcable185.108-200-24.mc.videotron.ca) |
01:49.35 | *** join/#asterisk bch1 (n=bch@CPE-24-26-173-250.mn.res.rr.com) |
01:50.34 | bch1 | anyone have trouble entering extensions on a cell phone and having Asterisk process it right? |
01:51.02 | bch1 | say I enter extension 1234, sometimes it goes to extension 1233 |
01:51.13 | bch1 | or 1123 |
01:51.15 | *** join/#asterisk milkyflava (n=milkyfla@240-7-237-24.gci.net) |
01:51.19 | milkyflava | hello |
01:51.46 | bch1 | no matter how long or short I hold down each key |
01:52.25 | bch1 | is there some kind of workaround for that? |
01:52.49 | milkyflava | should echocancel equal yes or some value i.e. 16,32,64,128? |
01:53.53 | milkyflava | same thing with echo training, should it just be "yes" or 400 or 800 |
01:55.46 | litage | milkyflava: i've no idea, but why not try each value and see what happens? |
01:56.18 | riddlebox | rezEdit: this is the error I get: chan_sip.c:10817 handle_request_register: Registration from '523 <sip:523@192.168.0.101>' failed for '192.168.0.3' - Username/auth name mismatch |
01:56.48 | milkyflava | litage: I was asking because I have heard people say use yes and others say use 800, in the case of echotraining |
01:57.21 | *** join/#asterisk xachen (i=justin@magnum.thisgeek.com) |
01:57.25 | xachen | Hey all :) |
02:00.36 | _Vile | riddle, check the username/password on the box |
02:00.52 | _Vile | then, make sure "use auth id" is off |
02:01.19 | riddlebox | _Vile:in sip.conf? or what I am trying to use xten-xlite softphone |
02:01.23 | _Vile | or auth name.. not sure what box you have |
02:01.25 | _Vile | hm |
02:01.28 | jimmy_deanPB | milkyflava, having any of them = yes just uses a default number...that's the only difference that I'm aware of |
02:01.28 | _Vile | on the xten |
02:01.42 | _Vile | make sure "use auth name" is off |
02:02.04 | milkyflava | jimmy_deanPB: Thanks, thats what I was thinking it did but wasn't sure |
02:02.19 | jimmy_deanPB | yeah, I was just dealing with those config options today a lot :) |
02:02.22 | jimmy_deanPB | so they're fresh in my mind |
02:02.25 | _Vile | make sure you have an entry for username |
02:02.29 | milkyflava | I will be tonight |
02:02.30 | _Vile | make sure your password is right |
02:02.31 | _Vile | will work |
02:02.43 | riddlebox | _vile:I do not see where to turn use auth name? |
02:02.49 | _Vile | blah sec |
02:02.55 | milkyflava | I read on the * list that Sangoma has a little utility to show how many taps are needed |
02:03.24 | jimmy_deanPB | didn't know that, did google turn up anything about that for you? |
02:03.27 | milkyflava | So I want to fire that up and make a call and fine tune my echocancel and echotraining |
02:03.44 | jimmy_deanPB | milkyflava, try out fxotune too |
02:03.57 | milkyflava | No, it was posted on the list today under the subject of X100p |
02:04.00 | milkyflava | I am going to |
02:04.04 | jimmy_deanPB | ok |
02:04.08 | _Vile | i'm wrong |
02:04.12 | _Vile | on the xlite |
02:04.17 | jimmy_deanPB | milkyflava, what version of asterisk by the way? |
02:04.21 | _Vile | you need "Authorization user" and username to be the same |
02:04.27 | _Vile | as what is in sip.conf |
02:04.29 | milkyflava | now if I use fxotune I still need to setthe echocancel, echotraining but not the tx/rx settings, correct? |
02:04.32 | milkyflava | 1.2 |
02:04.41 | _Vile | and password to be the same as the secret= you use in sip.conf |
02:04.45 | jimmy_deanPB | ok, have you tried things out yet to see if there is echo? |
02:04.54 | milkyflava | there is echo |
02:04.54 | _Vile | Enabled must be switched to YES - otherwise nothing will work even if your user registration is correct. Display Name may be whatever you want to be displayed. Username and Authorization User must be as in sip.conf and in our example it is ivan. Password has to be same as secret which is set in sip.conf for the user. |
02:04.54 | _Vile | Domains and Proxy have to be the IP addresses of your asterisk server. Note that here my asterisk IP is 10.3.3.25 but in your case it might be different. |
02:04.59 | _Vile | oops... |
02:05.00 | jimmy_deanPB | I was told that just using * v1.2 makes a huge difference by default |
02:05.19 | jimmy_deanPB | milkyflava, with or without echocancel, echotraining, etc? |
02:05.19 | milkyflava | probably because it uses a different echo can than 1.1 |
02:05.21 | _Vile | http://www.asteriskguru.com/tutorials/xlite_softphone.html |
02:05.33 | milkyflava | I have them both set to yes right now |
02:05.38 | jimmy_deanPB | ok, hmm |
02:05.59 | jimmy_deanPB | that makes me a little nervous, I'm hoping an upgrade to version 1.2.1 for me takes care of my echo problems |
02:06.25 | milkyflava | You might try using the MG2 echo can or maybe it is MG1 I never can remember |
02:06.35 | jimmy_deanPB | I am using MG2 now |
02:06.38 | milkyflava | how bad is your echo? |
02:06.41 | jimmy_deanPB | I already changed to that |
02:06.49 | milkyflava | I used MG2 but reverted back to KB1 |
02:06.58 | jimmy_deanPB | milkyflava, users of the system I setup described the phone system as "unusable" because of the echo |
02:07.17 | milkyflava | have you adjusted the tx/rx gains? |
02:07.24 | jimmy_deanPB | yep, everything |
02:07.29 | jimmy_deanPB | spent all day today working on that |
02:07.39 | milkyflava | did you use fxotune and make sure it starts before asterisk on a reboot |
02:07.43 | jimmy_deanPB | yep |
02:08.02 | milkyflava | did you reset your rx/tx gains back to 0.0 after using fxotune |
02:08.07 | *** join/#asterisk ldnblk (n=Just@optbom1.uk.access.vodafone.net) |
02:08.25 | jimmy_deanPB | milkyflava, what do you mean by that exactly? |
02:08.45 | milkyflava | When you use fxotune you shouldn't adjust your rx/tx gains |
02:08.50 | milkyflava | use either one or the other |
02:09.21 | milkyflava | fxotune takes the place of adjusting the rx/tx gains |
02:09.22 | jimmy_deanPB | oh right, yeah I did that |
02:09.24 | jimmy_deanPB | right |
02:09.25 | milkyflava | ok |
02:09.35 | milkyflava | so you had echo on both |
02:09.51 | milkyflava | after fxotune and when you tried to adjust rx/tx gains |
02:10.03 | milkyflava | and how bad is the echo you are hearing |
02:10.13 | jimmy_deanPB | yeah, best I could get it to was no echo to begin with, then 3-4 seconds later the echo was back to full strength |
02:10.29 | jimmy_deanPB | milkyflava, the echo is as loud as the voice going into the handset microphone |
02:10.30 | milkyflava | wow |
02:10.47 | milkyflava | what are you using for a phone |
02:10.50 | jimmy_deanPB | you can't hear yourself think when using it |
02:10.54 | milkyflava | i bet |
02:11.16 | *** join/#asterisk TheCops (i=nobody@got.securebinary.com) |
02:11.16 | jimmy_deanPB | milkyflava, cordless phone, low-end SIP phone, and high-end PolyCom SIP 501 all have the same echo |
02:11.51 | jimmy_deanPB | so the cordless phone is a direct bridge |
02:12.03 | jimmy_deanPB | on the TDM400P card |
02:12.12 | jimmy_deanPB | and yet even it echos just the same as the SIP phones |
02:12.27 | *** join/#asterisk apardo (n=apardo@29.Red-81-39-85.dynamicIP.rima-tde.net) |
02:12.41 | milkyflava | how far down did you adjust your tx andrx when you were trying to adjust them |
02:12.45 | litage | using the sample configs, when i connect to extension 500 and disconnect, i get a busy tone for a second. why might that be? |
02:13.08 | jimmy_deanPB | milkyflava, I got far enough down to where you couldn't hear the incoming at all nor hear the outgoing |
02:13.19 | milkyflava | lol and still echo? |
02:13.23 | jimmy_deanPB | no echo then |
02:13.26 | jimmy_deanPB | nothing then |
02:13.31 | jimmy_deanPB | no echo, but no voice either :) |
02:13.35 | TheCops | I guess that PRI dont have any echo on that type of technology ? |
02:14.09 | milkyflava | thats because there are two hybrids removed out of the equation with pri lines |
02:14.21 | milkyflava | it is 4 wire to 4wire from you to the CO |
02:14.30 | jimmy_deanPB | nice |
02:14.36 | jimmy_deanPB | but costs a lot more :) |
02:14.39 | milkyflava | I have been thinking of doing the same thing |
02:14.43 | milkyflava | yep |
02:14.54 | jimmy_deanPB | doing what? |
02:15.03 | milkyflava | what are your settings in zapata.conf |
02:15.11 | milkyflava | buying a pri line instead of analog |
02:15.25 | _Vile | most recurring problem I've heard w/ * is echo.... |
02:15.52 | milkyflava | yep, plan and test before you implement _Vile |
02:15.58 | jimmy_deanPB | milkyflava, how much are they if you know? |
02:16.11 | _Vile | tested, planned, implemented.. |
02:16.14 | milkyflava | I am not sure but I have heard 400-2000 per month |
02:16.15 | jimmy_deanPB | echo is a hard thing with phone systems in general |
02:16.19 | _Vile | echo only happens ocassionally on our PBX |
02:16.20 | jimmy_deanPB | dang! |
02:16.27 | jimmy_deanPB | not cheap |
02:16.30 | drray | I've hardly got any echo |
02:16.38 | drray | I've been blessed, I guess |
02:16.44 | milkyflava | echo happens at the beginning of the call and then once in awhile during the conversation |
02:16.46 | _Vile | speakerphone is an issue on 7940s |
02:16.52 | _Vile | w/ echo |
02:17.06 | jimmy_deanPB | the funny thing is, this is our second * PBX...the last one was a way underpowered one running an old Asterisk@Home version with the same TDM400P card |
02:17.22 | jimmy_deanPB | no echo, but lots of chirps and clicks due to a weak processor |
02:17.44 | milkyflava | thats caused by shared IRQs with the TDM card and some other device |
02:17.58 | litage | can users/peers/friends specified in h323.conf register with the asterisk server? |
02:18.17 | jimmy_deanPB | perhaps, but really I know it's partially due to it only being a Celeron 350 Mhz with 192 MB of RAM |
02:18.26 | milkyflava | lol |
02:18.26 | jimmy_deanPB | way under the minimal requirements |
02:18.34 | jimmy_deanPB | yet it was useable |
02:18.42 | jimmy_deanPB | most of the time |
02:19.01 | jimmy_deanPB | it was running, IAX, SIP and zaptel |
02:19.10 | jimmy_deanPB | I'm amazed it did as well as it did |
02:19.17 | milkyflava | how many users |
02:19.39 | jimmy_deanPB | 1-5 |
02:19.52 | rob0 | I had to give up my PII 400 because of the lack of PCI 2.1 or whatever. Didn't work with the TDM card. |
02:20.04 | jimmy_deanPB | mog_work, how many users for you? |
02:20.15 | mog_work | its just my house |
02:20.21 | mog_work | i think i have had 4 calls bridged on it |
02:20.22 | jimmy_deanPB | ok, yeah :) |
02:20.32 | mog_work | i also use for apache |
02:20.34 | mog_work | jabber |
02:20.40 | mog_work | and samba file server |
02:20.44 | mog_work | its pretty busy box |
02:20.47 | jimmy_deanPB | wow, no kidding |
02:20.48 | mog_work | avg load is 1.4 or so |
02:20.51 | jimmy_deanPB | hehe |
02:20.58 | jimmy_deanPB | linux? |
02:21.01 | mog_work | yeah |
02:21.05 | jimmy_deanPB | which distro? |
02:21.06 | mog_work | its a cobalt raq 2 |
02:21.09 | mog_work | got it for 50 bucks |
02:21.11 | milkyflava | ah |
02:21.12 | mog_work | debian |
02:21.13 | jimmy_deanPB | nice |
02:21.21 | mog_work | its also firewall and ssh front end |
02:21.24 | jimmy_deanPB | debian, the savior of the world :) |
02:21.26 | mog_work | damn spiffy little box |
02:21.31 | mog_work | no dma on hd though |
02:21.47 | mog_work | so its not much fun to work on |
02:22.42 | mog_work | i got 128mb more ram for it today though so all is good |
02:22.49 | jimmy_deanPB | good :) |
02:22.56 | jimmy_deanPB | it'll thank you |
02:23.11 | milkyflava | it also does the laundry and the dishes and can change out the main bearings in a 78 Ford? |
02:23.39 | mog_work | hmm /me thinks about hooking it up to dish washer and laundry over its serial port |
02:23.47 | milkyflava | lol |
02:23.50 | milkyflava | post the pics |
02:23.55 | mog_work | will do |
02:23.56 | jimmy_deanPB | harryvv, I do that sorta thing for a living :) |
02:24.01 | mog_work | its a pretty little box for what it is |
02:24.05 | jimmy_deanPB | oops |
02:24.08 | jimmy_deanPB | dang tab completion |
02:24.16 | jimmy_deanPB | ha is what I meant |
02:24.55 | mog_work | yeah |
02:25.41 | milkyflava | jimmy_deanPB: have you made sure you tdm is on its own IRQ? |
02:25.46 | *** join/#asterisk Nukemizer (n=Nuke@67.137.28.165) |
02:25.58 | jimmy_deanPB | milkyflava, I haven't no, how do I make sure of that? |
02:25.59 | litage | the * CLI has commands such as ``iax show registry'' and ``iax show peers''. why is there no ``h323 show registry'' or ``h323 show peers''? |
02:26.02 | mog_work | you do linux home integration jimmy_deanPB? |
02:26.11 | milkyflava | cat /proc/interrupts should tell you |
02:26.27 | jimmy_deanPB | mog_work, no, device that communicate over RS232 and other busses |
02:26.41 | jimmy_deanPB | milkyflava, ok, I'll try that |
02:26.55 | jimmy_deanPB | there's a chance it could be shared...it's in a new dell dimension 3000 desktop box |
02:26.58 | mog_work | ahh is there a washer i can get that has serial? |
02:27.08 | milkyflava | jimmy_deanPB: Like my little IR reciever for my remote controller for MythTV? |
02:27.23 | jimmy_deanPB | you could easily put your own serial transceiver on it |
02:27.40 | mog_work | really? |
02:27.41 | jimmy_deanPB | you just need a max232 chip and the proper power |
02:27.51 | milkyflava | the transceiver will both recieve and send, correct? |
02:27.52 | jimmy_deanPB | and a few caps and resistors |
02:27.55 | jimmy_deanPB | yep |
02:28.07 | mog_work | want to make me one? |
02:28.09 | jimmy_deanPB | runs at +12volts usually |
02:28.10 | milkyflava | lol |
02:28.24 | jimmy_deanPB | mog_work, google for "max232 projects" or something like that |
02:28.31 | jimmy_deanPB | should come up with stuff |
02:28.34 | mog_work | okies |
02:28.42 | mog_work | i really need a washer /drier on my lan |
02:28.47 | milkyflava | lol |
02:28.50 | jimmy_deanPB | I haven't actually done that from scratch yet, but all the products we make at work use that chip |
02:28.58 | jimmy_deanPB | haha |
02:29.02 | milkyflava | try Pluto home automation |
02:29.12 | jimmy_deanPB | but you can't get the clothes into it automatically |
02:29.14 | jimmy_deanPB | so what's the point? |
02:29.23 | mog_work | heh |
02:29.30 | mog_work | nah but i would want it to run at 3 in morning |
02:29.36 | jimmy_deanPB | that you could do |
02:29.41 | mog_work | so i dont have to worry about no hot water in shower |
02:29.46 | jimmy_deanPB | monitor water temp, level, ph level, etc |
02:30.22 | jimmy_deanPB | and if you press '3' on your * server, it'll start a new load :) |
02:30.40 | jimmy_deanPB | from your phone I mean |
02:30.59 | milkyflava | get bluetooth proximity |
02:31.10 | jimmy_deanPB | now you're talking! |
02:31.22 | milkyflava | Thats in Asterisk |
02:31.23 | mog_work | man that would be cool |
02:31.34 | jimmy_deanPB | whoa, what does that do in *? |
02:31.51 | milkyflava | If it detects your bluetooth phone it rings the office |
02:32.07 | milkyflava | if it doesnt then it rings your cell phone |
02:32.23 | mog_work | i dont have any bluetooth devices |
02:32.42 | milkyflava | your cell doesnt have it or do you have one |
02:32.54 | mog_work | i dont have a cellphone |
02:32.57 | mog_work | i hate the damn things |
02:33.02 | milkyflava | word |
02:33.14 | mog_work | my favorite parts of the day are when i am completely out of reach |
02:33.27 | milkyflava | mine are when the batteries in the cell die |
02:33.32 | mog_work | heh |
02:34.00 | Nukemizer | is it possible to configure a PRI for 12 B channels ? If so, would I need to leave my D channel as 24 ? thank you for any help |
02:34.09 | *** join/#asterisk ahattar (n=kjsd@ool-435292d6.dyn.optonline.net) |
02:34.12 | jimmy_deanPB | milkyflava, that's cool! |
02:34.21 | mog_work | yes and yes |
02:34.26 | mog_work | well not ness. 24 |
02:34.31 | mog_work | but most telcos run d on 24 |
02:34.36 | mog_work | even if you dont have full span |
02:34.41 | jimmy_deanPB | so what, you hook up a bluetooth proximity device to your local desktop or laptop computer and that communicates with asterisk? |
02:35.03 | Nukemizer | mog_work, thank you . That helps |
02:35.18 | milkyflava | I'm not sure, let me see if I can find the article for you. |
02:35.24 | jimmy_deanPB | cool |
02:35.25 | jimmy_deanPB | thanks |
02:35.54 | Chuji | Is walmart.com working for anyone? |
02:36.03 | milkyflava | http://mundy.org/blog/index.php?p=78 |
02:36.08 | milkyflava | POW! |
02:36.13 | jimmy_deanPB | hehe |
02:37.13 | milkyflava | Thats a good site |
02:37.18 | *** join/#asterisk Sedorox (i=brandon@smartserv/cna/Sedorox) |
02:37.25 | milkyflava | I'm going to eat and then battle the dragon of echo |
02:37.30 | milkyflava | you all have a good night |
02:38.27 | *** join/#asterisk froguz (n=froguz@244-142-222-201.adsl.terra.cl) |
02:40.18 | *** join/#asterisk `lyme (i=lyme@dsl081-121-070.dfw1.dsl.speakeasy.net) |
02:40.26 | froguz | why doesn't zaptel work if i install it by a bash script called by /etc/rc.local? (missing /dev/zap! on startup) |
02:40.36 | litage | do commands such as ``h323 show registry'' or ``h323 show peers'' only appear in the asterisk CLI if you're using an h323 gate*keeper*? |
02:40.49 | froguz | do i need tu add sudo to the install script commands? |
02:41.10 | litage | froguz: by default, the init script will be run as root |
02:41.13 | jimmy_deanPB | froguz, no, don't add sudo...you don't want to type your password everytime you run that |
02:41.53 | froguz | why doesn't works then? |
02:43.01 | froguz | but it works if i install it by hand |
02:43.26 | litage | froguz: "doesn't work" doesn't tell us anything about your problem |
02:44.06 | froguz | i think the problem could be the user, just root is permitted in my udev.permissions files |
02:45.16 | froguz | litage, on linux startup, when zaptel is loading (waiting zaptel to come online) it says missing /dev/zap! |
02:45.50 | litage | froguz: are you using udev? |
02:46.28 | froguz | but i install it typing the same commands as root i have no problem, and it load lots of modules like ztdummy on startup |
02:46.30 | froguz | yep |
02:46.32 | froguz | udev |
02:46.48 | litage | froguz: maybe udev hasn't created /dev/zap yet |
02:46.49 | froguz | Centos |
02:47.10 | litage | froguz: i know nothing about zaptel though |
02:48.02 | jimmy_deanPB | froguz, are you sure you did "modprobe zaptel" "modprobe wcfxs" modprobe "wctdm" if you are using a TDM card? |
02:49.13 | *** join/#asterisk jbot (i=ibot@pdpc/supporter/active/TimRiker/bot/apt) |
02:49.13 | *** topic/#asterisk is Asterisk 1.2.1 has been released! -//- http://www.asterisk.org |
02:49.37 | froguz | it supose zaptel do that for me, because i did a make config after make install |
02:51.47 | froguz | any clue? |
02:52.10 | jimmy_deanPB | I recommend doing a manual modprobe for them |
02:52.48 | froguz | i can't, because it complains about the missing /dev/zap device |
02:53.20 | Qwell | what commands do you run that makes it work? |
02:54.04 | froguz | doing exactly the same zaptel install process, but writing directly |
02:54.33 | froguz | the problem is just running the script |
02:54.37 | Qwell | what happens if after you reboot, you run /etc/init.d/zaptel start? |
02:54.41 | froguz | wich is called by rc.local |
02:55.18 | froguz | i haven't tried that |
02:56.05 | froguz | i would have to do that tomorrow |
02:56.10 | marcus2 | why doesnt atacomm send any email confirmations of orders? grr |
02:56.40 | froguz | so, you think is not about the script user |
02:56.47 | litage | how does chan_h323 compare to ooh323c? |
02:58.34 | Nukemizer | I could use some zaptel assitance as well.. zaptel start will not work for me, but if I manually load wcte11xp I can get the module to load. when running asterisk -vvvvcg 12 of my PRI B channels will register and then un register .. then "broken pipe" I do not know where to log for further trouble shooting |
02:59.07 | Qwell | Nukemizer: Did you edit the config file for the zaptel init script, to tell it you need wctc11xp? |
02:59.48 | Qwell | or, that must be just mandrake |
02:59.49 | Nukemizer | Qwell, box had been working for a month until i had to reboot tonight |
02:59.58 | Nukemizer | ahh yes Mandrake it is |
03:00.28 | Nukemizer | I take it Zaptell and Mandrake are a bad mix ? |
03:00.30 | Qwell | yeah, then either edit /etc/sysconfig/zaptel or /etc/init.d/zaptel and change the modules |
03:00.37 | Qwell | MODULES= |
03:01.12 | Qwell | so, in your case |
03:01.24 | Qwell | MODULES="zaptel wcte11xp" |
03:01.51 | Nugget | zaptel and mandrake are fine. zaptel and failing to read the documentation is apparently a real problem, though. |
03:02.06 | Qwell | s/documentation/init script/ |
03:04.34 | Nukemizer | Well, one could say that. But I have had Digium support login to box prevoiusly and unsuccesfully to get modules to load properly.. I was just wishing that something might have changed for me :( Thank you both I will read more |
03:05.27 | Nukemizer | Qwell, I am trying your suggestion. thank you for the help |
03:07.22 | *** join/#asterisk toddf (n=toddf@ns0.fries.net) |
03:08.28 | Flauto | qwell, i got the problem fix |
03:08.33 | Flauto | ed |
03:09.20 | Flauto | i got a script to solve the modprobe problem |
03:09.53 | Qwell | Flauto: what was the problem? |
03:10.16 | Flauto | i had to modprobe everytime i start my computer |
03:10.22 | Qwell | I know what the problem was |
03:10.24 | Qwell | what was the cause? |
03:10.45 | Flauto | there was not starting script |
03:10.46 | *** join/#asterisk techie (n=gus@adsl-068-209-242-072.sip.mia.bellsouth.net) |
03:10.53 | Flauto | i friend of mine send me one |
03:11.02 | Flauto | then, that is it |
03:13.11 | *** join/#asterisk franx (n=garra@215-78-246-201.adsl.terra.cl) |
03:13.37 | trixter | the pens that digium gives out are nice, uniball |
03:13.49 | file[laptop] | yeah |
03:13.53 | file[laptop] | I used my pen all up :( |
03:13.58 | file[laptop] | I'll grab more next time |
03:14.00 | trixter | I got 30 of em today |
03:14.23 | Qwell | 30? |
03:14.26 | trixter | for the sacramento asterisk users group along with mouse pads, some marketing materials and screwdrivers |
03:14.40 | trixter | give aways for the meet |
03:14.44 | Qwell | not a day goes by where I don't use my digium screwdriver, heh |
03:14.49 | Qwell | or...at least... |
03:14.56 | Qwell | not a day goes by where I don't think about stabbing somebody with it |
03:15.05 | trixter | well it is reversable |
03:15.12 | file[laptop] | I need another mousepad |
03:15.24 | file[laptop] | I stacked up on them last time but they got packed :( |
03:15.25 | Nukemizer | Flauto, I would be interested.. just got back from the server room and no luck thus far.. would you be willing to share script ? |
03:15.26 | trixter | I just prefer uniball pens so I was suprised they werent hte standard cheap bic ones |
03:15.39 | file[laptop] | but yeah, most excellent pens |
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03:23.22 | *** join/#asterisk monachoi (n=bonvoyag@cpe-24-174-162-34.satx.res.rr.com) |
03:24.45 | rezEdit | I am having an issue with ChanIsAvail(), which doesn't seem to be working when I use it to test a SIP extension for availability. Anyone else having issues? For me it always returns like the channel is free when in fact, it is not. |
03:24.54 | rezEdit | I am using it like: exten => 100,n,ChanIsAvail(SIP/224|sj) |
03:25.01 | rezEdit | and it always goes to n+1, never to n+101, even when the phone is on another call |
03:25.14 | rezEdit | anyone else using that app? |
03:25.21 | Qwell | rezEdit: svn trunk? |
03:25.27 | rezEdit | 1.2.0 |
03:25.32 | Qwell | The functionality of chanisavail and many others changed... |
03:25.42 | Qwell | they don't jump to n+101 anymore by default |
03:25.47 | Qwell | show application chanisavail |
03:25.54 | rezEdit | yeah that's why the j is there |
03:25.59 | Qwell | right |
03:26.40 | Qwell | perhaps it doesn't work :P Have you checked the returned variables? |
03:27.02 | rezEdit | yeah |
03:27.05 | Qwell | and? |
03:27.11 | rezEdit | one sec |
03:27.22 | Qwell | I'd NoOp all three |
03:27.55 | rezEdit | Executing ChanIsAvail("SIP/227-26ef", "SIP/224|sj") in new stack |
03:27.55 | rezEdit | <PROTECTED> |
03:28.17 | Qwell | So what happens when you dial out with SIP/227? |
03:28.21 | Qwell | erm, 224 |
03:28.42 | litage | how does chan_h323 compare to ooh323c? |
03:28.46 | rezEdit | to test this, I am calling 224 from 700 |
03:29.00 | rezEdit | the calling an extension from 227 that tests 224 (which IS on a call to 700) |
03:29.51 | rezEdit | I checked bugs.digium.com as well, doesn't seem to be anything there on it |
03:30.40 | rezEdit | that NoOp line is NoOp(${AVAILCHAN},${AVAILORIGCHAN},${AVAILSTATUS}) |
03:31.16 | rezEdit | ahhh: http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+ChanIsAvail |
03:31.27 | rezEdit | in the comments there is a list of states |
03:32.06 | rezEdit | <PROTECTED> |
03:34.00 | distortion | litage: http://www.voip-info.org/wiki-Asterisk+H323+channels |
03:34.12 | distortion | there's a comparison section on that page |
03:34.37 | Qwell | rezEdit: yeah, it checks for <= 1 |
03:34.55 | Qwell | So, if the device is in an unknown state... |
03:35.13 | rezEdit | yeah. unfortunately, that doesn't help me understand why it's unknown |
03:35.42 | rezEdit | * has all the info it needs to know that it's in use. |
03:36.03 | Qwell | Is the device registered? |
03:36.08 | rezEdit | yeah |
03:36.22 | litage | distortion: yeah i've read that, but it's not much of a comparison |
03:36.29 | Qwell | rezEdit: Is call limit set? |
03:36.41 | rezEdit | Qwell: Yeah, to 3 |
03:36.57 | Qwell | peer? |
03:37.07 | rezEdit | Qwell: I needed to fdo that to get 'sip show inuse all' to work |
03:37.16 | rezEdit | Qwell: friends |
03:37.21 | Qwell | so, yes |
03:38.29 | asterboy | Anyone using the Cisco 7920 wireless IP Phone |
03:38.31 | asterboy | ??? |
03:39.18 | *** join/#asterisk mtupper (n=mtupper@pc-14-172-104-200.cm.vtr.net) |
03:39.25 | distortion | last i checked it was not upgradable to sip, and the skinny support in * is a little light |
03:41.17 | Qwell | distortion: it's a little skinnt |
03:41.19 | Qwell | skinny* |
03:41.26 | Qwell | and no, actually, sccp rocks my world |
03:42.38 | denon | SCCP on Asterisk rocks your world? |
03:42.43 | Qwell | it does |
03:42.49 | denon | is it really that cool yet? |
03:42.50 | mtupper | please help on completing a call from * to PSTN through a SPA-3000!!! I have googled it all over the place, tried dozens of configurations and I just can't get the PSTN line to dial-out. I HAVE been able to complete calls from the Phone jack in the SPA-3000 to an x-ten softphone via SIP... but the PSTN line is not having any of it!!! |
03:42.52 | file[laptop] | Qwell: that was bad |
03:42.53 | Qwell | denon: for sure |
03:42.55 | *** join/#asterisk znoG (n=gs@33-138-114-200.fibertel.com.ar) |
03:42.58 | denon | better than SIP? |
03:42.59 | file[laptop] | Qwell: therefore, I sentence you to... DEATH BY MUFFINS! |
03:43.02 | Qwell | denon: yes |
03:43.03 | Qwell | :P |
03:43.07 | denon | details, details |
03:43.08 | Qwell | I don't like SIP though... |
03:43.11 | denon | oh well .. |
03:43.15 | denon | you're one of those guys... |
03:43.18 | Qwell | I converted my 7960 to sccp at home, and I've got a few at home |
03:43.20 | Qwell | at work... |
03:43.31 | denon | so you run OSX too, right? just because you hate microsoft. . |
03:43.33 | Qwell | nah, it works really well though. |
03:43.50 | Qwell | denon: I wrote some realtime support for it...it's hot |
03:44.01 | denon | im not much of a realtime guy |
03:44.03 | Qwell | I use the chan_sccp from berlios.de |
03:44.04 | denon | I like config files |
03:44.15 | denon | so what fun stuff does it do, that I cant do with sip? |
03:44.18 | denon | (mostly 7960s here) |
03:44.34 | Qwell | device messaging is cool |
03:44.48 | Qwell | can they use hints with SIP? |
03:44.53 | Qwell | You can with sccp... |
03:45.59 | Qwell | and I think you can only do XML pushing with sccp...that's real nice |
03:46.39 | rezEdit | hmmm. 'sip show registry' doesn't show anything.... |
03:46.41 | Qwell | file[laptop]: ha! tcp only. You're stuck on udp. ;] |
03:46.46 | rezEdit | 'sip show peers' does though |
03:46.49 | file[laptop] | bah |
03:46.53 | file[laptop] | UDP over TCP |
03:46.54 | file[laptop] | HA! |
03:46.56 | file[laptop] | :P |
03:46.57 | Qwell | eh? |
03:47.01 | Vco | stuck on "teh udp" |
03:47.58 | *** join/#asterisk mtgh (n=chatzill@dsl093-001-038.det1.dsl.speakeasy.net) |
03:48.17 | Qwell | oh, and...duh...the obvious |
03:48.22 | Qwell | you can't use the addons with sip |
03:49.00 | *** part/#asterisk mtupper (n=mtupper@pc-14-172-104-200.cm.vtr.net) |
03:49.03 | *** join/#asterisk _daver_ (n=daver@ns1.tmok.com) |
03:49.07 | Qwell | and immediate dialplan checking...that |
03:49.12 | Qwell | 's cool. Just like zap |
03:49.20 | Qwell | or the iaxy |
03:49.26 | rezEdit | Qwell: regarding your earlier question about whether my devices were registered... shouldn't they show up in the list with the command 'sip show registry'? |
03:49.34 | litage | what's the difference between the 'host' and 'ip' settings for a friend/peer/user in sip.conf or iax.conf? |
03:49.39 | Qwell | rezEdit: dunno, I don't use sip :p |
03:49.48 | rezEdit | Qwell: heh ok. |
03:51.11 | *** join/#asterisk bmg505 (n=leon@dsl-146-19-135.telkomadsl.co.za) |
03:53.39 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
03:53.52 | distortion | If i wanted to make sure that ulimit -c unlimited and ulimit -n 65535 |
03:53.58 | distortion | err |
03:54.31 | distortion | If i wanted to make sure that ulimit -c unlimited and ulimit -n 65535 was added to the initscript for redhat, would i just add it to the /etc/init.d/asterisk file? |
03:55.20 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
03:59.09 | *** join/#asterisk L|NUX (n=linux@202.5.145.58) |
03:59.50 | SkramX | Hiya. |
04:01.48 | litage | distortion: what's ulimit? |
04:07.53 | *** join/#asterisk insomni (n=insomni@x1-6-00-e0-18-6f-34-ad.k455.webspeed.dk) |
04:08.14 | L|NUX | litage : ulimit is the command :) |
04:08.15 | *** join/#asterisk stkn_ (i=nobody@gentoo/developer/pdpc.active.stkn) |
04:08.23 | L|NUX | litage : man ulimit |
04:08.30 | Vco | the opposite of melimit |
04:08.56 | L|NUX | heh |
04:09.10 | L|NUX | ulimit is the bash builtin command |
04:10.24 | *** join/#asterisk twisted (n=twisted@asterisk/friend-and-developer/pdpc.professional.twisted) |
04:10.24 | *** mode/#asterisk [+o twisted] by ChanServ |
04:10.34 | franx | hi, what softphone for linux do you recommend? |
04:10.55 | L|NUX | franx : sjphone |
04:11.12 | L|NUX | franx : gnomemeeting or xten xlite :) |
04:11.58 | _Vile | go xten, less questions for this chan |
04:12.18 | litage | L|NUX: ah. don't have ulimit on my machine |
04:13.09 | *** join/#asterisk BhaalWK (i=bhaal@freenode/staff/bhaal) [NETSPLIT VICTIM] |
04:13.09 | *** join/#asterisk newl (n=newlook@203-59-61-62.dyn.iinet.net.au) [NETSPLIT VICTIM] |
04:14.07 | franx | L|NUX: thanks a lot |
04:14.10 | *** join/#asterisk Tili (i=Tili@202-133-67-94-dialup.sat.net.pk) |
04:14.50 | L|NUX | franx : np |
04:15.03 | L|NUX | litage : humm |
04:15.13 | L|NUX | litage : which distro you are using ? |
04:15.28 | *** join/#asterisk CoaxD (i=coax@shell1.cornernet.com) |
04:19.46 | franx | is it possible to test asterisk without having to buy extra hardware? |
04:20.24 | Qwell | franx: sure, got speakers and a mic? |
04:24.16 | rob0 | litage: ulimit is a bash builtin |
04:24.57 | rob0 | IOW, if you have bash, you have ulimit |
04:25.23 | trixter | franx: yes if you have all the hardware you need |
04:25.46 | trixter | astlinux.org even has a vmware image for the free vmware player which lets you test without reformatting, they have a liveCD if you would rather that, and there are a few other lvie CDs out there |
04:26.29 | trixter | if you only want to play with call mnagers, do voip, etc you dont need hardware other than a system, perhaps the one you are using to get on irc perhaps another, if you want to interface with the pstn directly you may need hardware to do that if you dont already have it |
04:27.23 | trixter | so depending on what you mean by 'test asterisk out' the answer could go either way ... |
04:31.38 | tessier | Looks like quite a few of these things trade hands on ebay |
04:31.47 | tessier | I always see a bunch on there. |
04:31.57 | tessier | It seems to have driven the price down a bit too. |
04:32.31 | rob0 | tessier: how much are you wanting? |
04:35.19 | tessier | rob0: It's on ebay. No reserve. So...whatever I can get! :) |
04:36.09 | rob0 | what do you think they will go for? Seems like most of this stuff costs more than I want to pay ... /dev/cheap0 :) |
04:36.33 | tessier | $200 or so probably |
04:39.03 | rob0 | 5841378897 ? |
04:39.15 | rob0 | haha the item number not the bid :) |
04:39.27 | tessier | 5842276412 and 5842276731 |
04:39.34 | tessier | crap...I screwed up the title on the listing for my ata |
04:42.50 | trixter | can you skype:5842276731@ebay.com ? or do they not yet have that integration? :P |
04:43.16 | *** join/#asterisk inv_arp (n=junya@c-66-176-211-109.hsd1.fl.comcast.net) |
04:47.11 | shido6 | $10 |
04:47.12 | shido6 | ? |
04:49.33 | *** join/#asterisk ComputerWarm (n=workingg@66.244.235.210) |
04:49.57 | ComputerWarm | Hello all question music on hold anyway i can get it to play music from a online radio station/ |
04:50.26 | asterboy | yes |
04:50.47 | asterboy | but you will go to jail if you do. |
04:50.49 | ComputerWarm | asterboy sorry i should have finished that off. while they are in Queue? |
04:51.01 | asterboy | :P |
04:51.12 | ComputerWarm | not with this radio station |
04:51.15 | asterboy | lol |
04:51.28 | asterboy | I just say that cause its funny in here. |
04:51.45 | SkramX | ComputerWarm: somepeople have weird senses of humor.. |
04:51.45 | ComputerWarm | oh ok... anyway is there anyway to get it to play in queue? |
04:51.54 | ComputerWarm | SkramX i see that lol |
04:52.17 | Sedorox | tessier: power supply with the ata? |
04:52.42 | asterboy | someone sell a X100m! |
04:53.00 | tessier | Sedorox: Yep |
04:53.03 | loud | my cat peed my 7960. |
04:53.09 | SkramX | loud: lol |
04:53.13 | Sedorox | loud: hopefully it wasn't a male cat... |
04:53.19 | loud | no, female |
04:53.23 | loud | fucking hate her now. |
04:53.27 | Sedorox | ah.. so it shouldn't smell for long :p |
04:53.30 | Sedorox | if you cleaned it |
04:53.47 | Sedorox | lol |
04:53.49 | tessier | My male cat peed on my nice leather jacket |
04:53.53 | tessier | Had to have it dry cleaned |
04:53.55 | Sedorox | :/ |
04:54.04 | Sedorox | my cat took a dump on my bed one time... |
04:54.06 | tessier | Then after it aired out for a while it was good as new. |
04:54.14 | Sedorox | I deserved it tho.. forgot to clean her box for a while.... |
04:54.21 | asterboy | easy fix...with a 22 cal. gum |
04:54.26 | Sedorox | lol |
04:54.37 | asterboy | sticks to the cat.= :P |
04:54.58 | Sedorox | personally I still prefer cats over dogs... |
04:55.34 | *** join/#asterisk Ayano (n=Ayano_@adsl-70-245-190-90.dsl.spfdmo.swbell.net) |
04:55.37 | *** join/#asterisk SwK (n=SwK@12-219-151-128.client.mchsi.com) |
04:56.16 | Ayano | What is the web interface port for a polycomm 301? |
04:59.12 | *** join/#asterisk toddf (n=toddf@ns0.fries.net) |
05:00.26 | *** join/#asterisk EriSan (n=erisan@81-174-42-85.f5.ngi.it) |
05:01.49 | SwK | ayano it should be port 80... but use the XML from a ftp or a tftp server instead you'll be a lot happier |
05:03.40 | twisted | SwK, fsck off :P |
05:05.22 | litage | i just installed ooh323c, and it said it installed several libraries into /usr/lib . how do you get asterisk to use ooh323c though? |
05:05.30 | litage | ^--- http://rafb.net/paste/results/uZuIoU20.html |
05:11.37 | *** join/#asterisk amir (n=amir@gentoo/developer/amir) |
05:11.41 | *** join/#asterisk mog_home (n=mogorman@user-24-236-84-48.knology.net) |
05:11.43 | SwK | twisted stfu |
05:11.44 | SwK | :P |
05:11.55 | twisted | well, that only took 8 minutes |
05:12.01 | SwK | heh |
05:12.12 | twisted | SwK, i have most of my voice back |
05:12.12 | *** join/#asterisk trixter (n=trixter@65.172.209.246) |
05:12.14 | SwK | had to go find some beer |
05:12.29 | twisted | i'm guessing not talking hardly at all today helped |
05:12.32 | SwK | yeah |
05:12.43 | SwK | talking yesterday probably didnt help much |
05:12.47 | twisted | no, probably not |
05:13.09 | twisted | but i can talk now, as long as I don't try to get loud or high pitched, then it cuts out |
05:13.32 | SwK | twisted add me to you msg list |
05:13.36 | twisted | lol |
05:14.02 | SwK | I don't accepting private messages or dialogs at this time. Please talk with me in the public channel. { Automessage } |
05:14.04 | SwK | heh |
05:14.18 | twisted | k |
05:14.38 | twisted | took me a minute to remember the command |
05:14.56 | Nugget | so clearly you should keep messaging so you're sure to notice when they do start accepting them. :) |
05:15.16 | twisted | lol |
05:17.08 | bsdfreak | herh |
05:19.41 | *** join/#asterisk inv_arp (n=junya@c-66-176-211-109.hsd1.fl.comcast.net) |
05:21.12 | *** join/#asterisk Gamercjm (n=gamercjm@pool-70-104-145-143.lsanca.fios.verizon.net) |
05:21.35 | Gamercjm | Is there a way to record calls? |
05:21.46 | Nugget | yes. |
05:22.16 | *** join/#asterisk alephcom (n=alephcom@host75.net14.mcsnet.ca) |
05:22.23 | alephcom | Greetings Everyone. |
05:22.37 | sbingner | greetings earthlings |
05:22.47 | Gamercjm | i tried to use the *1 but would not work, just says not found.. |
05:22.49 | inv_arp | Gamercjm: monitor |
05:22.53 | Nugget | http://justfuckinggoogleit.com/?q=asterisk+record+calls |
05:23.37 | sbingner | heh |
05:24.17 | inv_arp | wow domain actually exists |
05:24.38 | tessier | Wish I had thought of that. |
05:25.03 | Nugget | me too |
05:25.20 | sbingner | this is retarded, I can burn a DVD at 16x in 6 mins, but then it only can verify it at about 7x which takes like 12 mins |
05:25.37 | SwK | hah |
05:25.43 | SwK | thats lame |
05:25.51 | bsdfreak | yeh |
05:26.21 | sbingner | although it's been pretty reliable on this media I may just give up verifications heh |
05:32.29 | *** join/#asterisk Assid (n=assid@203.115.64.62) |
05:32.31 | Assid | heya |
05:32.52 | Assid | anyone found any issues updating to 1.2.1 from an old CVS head? |
05:33.07 | Qwell | Assid: modules still existing that shouldn't |
05:33.22 | Qwell | rm /usr/lib/asterisk/modules/* |
05:33.27 | Qwell | then make install |
05:33.40 | Assid | ahh.. so i gotta shut it down for a bit then |
05:34.08 | Assid | i thought i could do an update by just reloading |
05:34.25 | *** join/#asterisk santiago (n=santiago@208.195.215.160) |
05:34.34 | Assid | do i unload the zaptel ? |
05:38.47 | *** join/#asterisk trixter (n=trixter@65.172.209.246) |
05:39.46 | *** join/#asterisk enmaca (n=enmaca@dsl-201-129-118-166.prod-infinitum.com.mx) |
05:41.33 | enmaca | hi all |
05:44.59 | alephcom | hi |
05:45.34 | Rowter | anyone has a Dual AMD® Opteron™ CPU with asterisk? I wonder if works ok as in Intel processors. |
05:46.01 | mog_home | intels seem to work a little better |
05:46.03 | mog_home | but not much |
05:46.18 | MikeJ[Laptop] | you got home in 2 min? |
05:46.27 | mog_home | lol |
05:46.28 | mog_home | no |
05:46.33 | MikeJ[Laptop] | heh |
05:46.35 | Rowter | mog_home, you test it out? |
05:46.41 | mog_home | my work machines switch doesnt always work |
05:46.43 | litage | after compiling ooh323c, how do you generate chan_ooh323.so? |
05:46.52 | mog_home | we have a few of everything in the lab rowter |
05:47.01 | MikeJ[Laptop] | hehe |
05:47.08 | Rowter | mog_home, excelent.. so its a safe choice? |
05:47.15 | mog_home | yeah its fine |
05:47.25 | mog_home | asterisk is not very cpu specific |
05:47.30 | mog_home | i run it on my mips box at home |
05:47.42 | enmaca | hi have a sunfire v20z working for two weeks ago |
05:47.44 | MikeJ[Laptop] | yes... but it is anti windows! |
05:47.47 | enmaca | its working fine |
05:48.07 | mog_home | nice |
05:48.28 | Rowter | mog_home, thanks a lot |
05:48.45 | mog_home | no problemo |
05:51.37 | enmaca | anyone have worked whit a sonus psx, gsx gateway? |
05:51.47 | enmaca | and *? |
05:54.58 | Assid | umm.. can someone paste me a copy of their musiconhold |
05:55.10 | Assid | it first said my copy is incompatible.. |
05:55.22 | Flauto | <PROTECTED> |
05:55.28 | mog_home | you have your /usr/src/asterisk/configs/musiconhold.conf.sample |
05:55.29 | Flauto | what is this message about |
05:55.36 | Assid | so i deleted my copy.. and ran a make install on asterisk again.. now it says Dec 14 00:53:58 WARNING[12318] res_musiconhold.c: No music on hold classes configured, disabling music on hold. |
05:56.02 | Flauto | would anyone tell me? |
05:57.26 | *** join/#asterisk kusznir (n=kusznir@pool-70-110-34-187.sea.dsl-w.verizon.net) |
05:57.36 | mog_home | it means flauto dnandnana peer is gone |
05:57.51 | mog_home | aka it has lost contact with it for a set amount of time |
05:58.11 | shido6 | http://cgi.ebay.com/Cisco-7960-IP-Phone-with-AC-Adapter_W0QQitemZ5841191851QQcategoryZ51204QQrdZ1QQcmdZViewItem |
05:58.23 | Flauto | do you mind if i send you pravite messages? mog |
05:58.24 | shido6 | sonus! |
05:58.26 | shido6 | LORD |
05:58.31 | shido6 | reminds me of Global Crossing |
05:58.34 | mog_home | nope |
05:58.46 | Flauto | thanks |
05:58.59 | Qwell | shido6: eww, sip |
05:59.15 | *** join/#asterisk Umaro (n=umaro@68.142.142.105) |
06:00.01 | Umaro | hey guys.. I have a tormenta 2 PCI card, and i'm told it should have a pci id of d00d.. but mine has a pci id of 1000. it's a 4 port T1/E1 card.. does anyone have experience with this? |
06:02.13 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
06:06.35 | *** join/#asterisk s0lid (n=usahnem@58.69.102.22) |
06:08.40 | sbingner | d00d. |
06:09.00 | _Vile | wha |
06:09.12 | sbingner | heh |
06:09.29 | Assid | hrmm.. what do i do if the moh sound is too less |
06:10.40 | _Vile | turn your volume up a little? |
06:10.58 | Assid | err.. its max.. |
06:11.11 | _Vile | there's a setting in *, sec |
06:12.54 | _Vile | how are users connecting to your moh? |
06:13.16 | _Vile | rather, what kind of cards do you have in you * box? |
06:13.20 | Assid | whenever they are put on hold.. OR.. i have a extension where i play it from |
06:13.24 | Assid | no cards.. |
06:13.29 | Assid | all over ip |
06:13.30 | _Vile | all SIP? |
06:13.38 | _Vile | ok |
06:13.41 | _Vile | sec |
06:13.58 | Assid | sip for phones.. iax for providers |
06:14.14 | _Vile | what version of * are you running? |
06:14.44 | Assid | 1.2.1 |
06:15.11 | Assid | shouldnt mpg123 do something? |
06:15.36 | MikeJ[Laptop] | who the hell uses mpg123 for moh |
06:15.42 | MikeJ[Laptop] | use native |
06:15.51 | Assid | native huh? |
06:16.49 | _Vile | you don't need mpg123 |
06:17.50 | Assid | how do i make it change? just override the settings inside the default context? |
06:20.28 | Assid | Dec 14 01:19:58 WARNING[12580]: file.c:508 ast_openstream_full: File /var/lib/asterisk/personelmoh/londonl_auto does not exist in any format |
06:20.38 | Assid | its a .mp3 file.. |
06:21.07 | mog_home | man compiling all of asterisk on my box |
06:21.10 | mog_home | takes foreva |
06:21.36 | Qwell | mog_home: distcc |
06:21.57 | mog_home | yeah maybe |
06:21.59 | mog_home | next time |
06:22.07 | Qwell | You'll never do it :p |
06:22.25 | mog_home | heh yeah |
06:22.32 | mog_home | but im not gonna compile it again on this box |
06:22.36 | Qwell | heh |
06:22.38 | mog_home | i just need to get gsm working for it |
06:24.08 | Assid | wow.. this is weird |
06:24.23 | Assid | my show translation shows almost 1 ms more for most of my codecs |
06:24.30 | Assid | what happened |
06:24.30 | Qwell | Assid: than? |
06:24.39 | Assid | the last time.. using CVS head |
06:24.40 | MikeJ[Laptop] | Assid, use format_mp3, or even better, convert the file to raw or gsm or somthing better than mp3 |
06:24.51 | *** join/#asterisk edwar64896 (n=edwar648@220-244-62-6.static.tpgi.com.au) |
06:25.36 | Assid | MikeJ[Laptop]: just curious.. how does it help? |
06:26.28 | drumkilla | Assid: he's saying Asterisk doesn't know how to read an mp3 file by default |
06:26.35 | drumkilla | unless you install format_mp3 from asterisk-addons |
06:26.46 | drumkilla | and the other option is to just convert your mp3's into another format that asterisk can read |
06:27.10 | Assid | just curious.. whats so bad abt mpg123? |
06:27.34 | drumkilla | well, that's another topic, but ... |
06:27.36 | _Vile | hm, slow |
06:27.39 | drumkilla | it's just an external dependency |
06:27.44 | _Vile | and we can debate that for an hour |
06:27.48 | drumkilla | we have to execute an external process |
06:27.55 | drumkilla | and we depend on a specific version of it ... |
06:28.24 | drumkilla | it's much more efficient to use files of a native format |
06:28.28 | drumkilla | and let asterisk read them directly |
06:28.52 | _Vile | . /|\ |
06:29.10 | konfuzed | ive probably asked before but is gsm a non commercial license encumbered codec as in open source |
06:29.16 | Assid | okay .. so whats suggested? raw? |
06:29.28 | _Vile | gsm |
06:29.53 | Assid | gsm? hrmm.. is the transcoding overhead worth it? |
06:30.01 | _Vile | heh |
06:30.02 | konfuzed | gsm is lightest vis a vis high compression and cross telecom network compatible |
06:30.05 | konfuzed | is that right |
06:30.28 | konfuzed | s/vis a vis/via/ |
06:30.35 | *** part/#asterisk s0lid (n=usahnem@58.69.102.22) |
06:30.41 | _Vile | http://www.marko.net/asterisk/archives/0212/0350.html |
06:30.43 | drumkilla | well actually there isn't a best |
06:30.50 | _Vile | go there |
06:30.54 | _Vile | follow directions. |
06:31.00 | drumkilla | the *best* is the same format as the other leg of the call |
06:31.16 | drumkilla | the *best* is to have it available in every format, and let asterisk pick the best one for you :) |
06:31.25 | _Vile | stop confusing him |
06:31.28 | _Vile | ;) |
06:31.33 | konfuzed | ok that works for mwe |
06:31.37 | *** part/#asterisk Ayano (n=Ayano_@adsl-70-245-190-90.dsl.spfdmo.swbell.net) |
06:31.42 | konfuzed | ;^) |
06:31.45 | drumkilla | confusing? |
06:31.51 | konfuzed | not for me ;^) |
06:32.39 | _Vile | s/confusing/konfusing |
06:32.41 | Assid | in that case.. ulaw would be the best for me.. coz thats what my incoming and outgoing stream is in.. EXCEPT for my personal SIP account, sinec i am remote |
06:33.10 | _Vile | Assid, follow sox directions. |
06:33.21 | konfuzed | id like to do all IAX/gsm |
06:33.33 | *** join/#asterisk tekati (n=captain@cpe-66-75-215-63.bak.res.rr.com) |
06:33.34 | konfuzed | but hey thats just me |
06:34.39 | Assid | oh crap.. upgrading broke my voicemail config |
06:34.40 | Assid | Dec 14 01:34:21 WARNING[19236]: app_voicemail.c:2384 leave_voicemail: No entry in voicemail config file for '201' |
06:34.56 | Assid | im guessing i gotta use u201@context ? |
06:34.57 | _Vile | add an entry for 201 |
06:35.06 | Assid | there already is.. in the ila context |
06:35.09 | _Vile | in the proper context. |
06:35.26 | _Vile | perhaps you must do an @context. |
06:35.28 | _Vile | yes. |
06:36.06 | _Vile | beer break bbiam |
06:38.20 | *** join/#asterisk edwar64896 (n=edwar648@220-244-62-6.static.tpgi.com.au) |
06:38.26 | *** join/#asterisk tartar (n=tartar@CPE0004e27b716e-CM014370001917.cpe.net.cable.rogers.com) |
06:38.38 | *** part/#asterisk edwar64896 (n=edwar648@220-244-62-6.static.tpgi.com.au) |
06:38.48 | Assid | oay now to make it into files |
06:39.09 | *** join/#asterisk edwar64896 (n=edwar648@220-244-62-6.static.tpgi.com.au) |
06:41.10 | _Vile | read sox, google: sox site:voip-info.org |
06:41.17 | _Vile | or |
06:41.18 | _Vile | sec |
06:41.37 | *** join/#asterisk krisguy (n=krisguy@h216-170-039-057.adsl.navix.net) |
06:44.12 | Assid | yeah i already have it |
06:44.18 | Assid | sox blah.mp3 blah.ul |
06:44.47 | *** join/#asterisk Ayano (n=Ayano_@adsl-70-245-190-90.dsl.spfdmo.swbell.net) |
06:44.52 | Qwell | they need to be mono 8hz |
06:44.55 | Qwell | 8khz... |
06:45.03 | _Vile | ;exten => metaswitch,1,Answer |
06:45.03 | _Vile | ;exten => metaswitch,2,Playback(/tmp/asterisk-recording) |
06:45.03 | _Vile | ;exten => metaswitch,3,Wait(2) |
06:45.04 | _Vile | ;exten => metaswitch,4,Record(/tmp/asterisk-recording:gsm|2|20) |
06:45.04 | _Vile | ;exten => metaswitch,5,Wait(2) |
06:45.04 | _Vile | ;exten => metaswitch,6,Playback(/tmp/asterisk-recording) |
06:45.05 | _Vile | ;exten => metaswitch,7,Wait(2) |
06:45.07 | _Vile | ;exten => metaswitch,8,Goto(metaswitch,4) |
06:45.30 | _Vile | good |
06:45.52 | _Vile | dont include the extension on a Playback |
06:46.43 | _Vile | dont include the ;'s either |
06:46.52 | _Vile | comments |
06:47.18 | Ayano | I just got a used polycom 300, How do I configure the phone itself. I have asterisk configured for it, now what? |
06:47.26 | _Vile | enjoy |
06:47.38 | Assid | hrmm.. still gotta use sox |
06:48.30 | _Vile | Assid, no, only for conversion |
06:48.31 | Assid | yeah |
06:48.36 | Assid | -v 1.5 -c 2 |
06:48.39 | Assid | should do the trick |
06:48.50 | _Vile | you got it then |
06:48.57 | _Vile | Ayano |
06:49.21 | _Vile | What's the problem, 300<->*? |
06:50.10 | Ayano | I don't know where to put the asterisk ip, and user info. I have used the cisco 7940 a lot, but this doesn't have a place to put it |
06:50.12 | *** join/#asterisk edwar64896 (n=edwar648@220-244-62-6.static.tpgi.com.au) |
06:50.19 | *** join/#asterisk L|NUX (n=linux@202.5.145.58) |
06:50.34 | *** part/#asterisk edwar64896 (n=edwar648@220-244-62-6.static.tpgi.com.au) |
06:50.36 | _Vile | sip.conf |
06:50.42 | _Vile | in /etc/asterisk |
06:51.05 | _Vile | sec |
06:51.14 | Ayano | I have sip.confi configured. How do i configure the phone itself? |
06:51.48 | _Vile | can't help ya there |
06:52.02 | _Vile | tell the phone to look at * |
06:52.27 | Ayano | yea, that's what I can't find is where to put that |
06:52.48 | _Vile | explain what 'options' you have |
06:53.04 | _Vile | should allow you to put in an auth name |
06:53.14 | _Vile | and display name |
06:53.14 | Ayano | none that have anything to do with a auth name or pass |
06:53.15 | _Vile | and a password |
06:53.25 | _Vile | then your phone is probably locked |
06:53.35 | _Vile | try logging in as admin |
06:53.46 | Ayano | lol tried that |
06:53.52 | _Vile | locked. |
06:54.01 | _Vile | where did you get your phone from? |
06:54.28 | _Vile | if you say eBay I will kick you in the shin |
06:54.41 | mog_home | ebay ^_^ |
06:55.07 | _Vile | hide all you want |
06:55.08 | _Vile | ;) |
06:55.15 | Ayano | no, it was from some dealer I think. I don't know, its a friend of mine. |
06:55.21 | _Vile | ok |
06:55.34 | _Vile | can you SEE the display name? |
06:55.34 | Ayano | have you used a polycom 300? |
06:55.43 | Ayano | no, it doesn't get that far |
06:55.44 | *** join/#asterisk edwar64896 (n=edwar648@220-244-62-6.static.tpgi.com.au) |
06:55.46 | _Vile | no, never. |
06:55.54 | _Vile | I know the functions. |
06:56.07 | Ayano | I didn't think so, your describing cisco stuff. lol |
06:56.07 | _Vile | I use 7940's |
06:56.10 | Assid | man.. it sounds bad |
06:56.28 | _Vile | and 7960's |
06:56.59 | _Vile | but, I suspect that you're locked out of any good config options -- mog please confirm? |
06:57.37 | *** part/#asterisk edwar64896 (n=edwar648@220-244-62-6.static.tpgi.com.au) |
06:58.25 | _Vile | Assid, what sounds bad? |
06:59.21 | *** join/#asterisk edwar64896 (n=edwar648@220-244-62-6.static.tpgi.com.au) |
06:59.24 | Assid | the moh |
06:59.51 | _Vile | explain... |
07:00.01 | Assid | hrmm.. cracks |
07:00.09 | Assid | i think i will leave it on mp3 for now |
07:01.03 | _Vile | cracks? |
07:01.39 | *** part/#asterisk EriSan (n=erisan@81-174-42-85.f5.ngi.it) |
07:01.43 | _Vile | explain better |
07:02.38 | _Vile | is it consistent? |
07:02.56 | _Vile | or random |
07:03.21 | _Vile | bbiam, smoke break |
07:03.27 | Assid | consistent |
07:03.31 | _Vile | ok |
07:03.39 | Assid | my bass just goes boom boom.. every few seconds |
07:03.41 | _Vile | SIP? |
07:03.55 | Assid | im gonna try this locally on my asterisk box here.... to see what the issue is |
07:03.59 | Assid | yeah.. over sip |
07:04.02 | _Vile | ok |
07:04.04 | _Vile | try local |
07:04.08 | _Vile | i'll bbiam |
07:04.12 | Qwell | using the right kind of mp3? |
07:04.21 | Qwell | 8khz mono |
07:04.34 | _Vile | I tell everyone to use the right format |
07:04.39 | _Vile | noone listens to me, I dunno |
07:04.55 | Qwell | _Vile: because you aren't saying it in 8khz mono |
07:04.59 | Qwell | duh |
07:05.02 | _Vile | ok |
07:05.05 | _Vile | Assid |
07:05.13 | _Vile | try 8khz Mono |
07:05.24 | Qwell | with no id3 |
07:05.27 | _Vile | see what happens lemme know in 5 min |
07:05.30 | _Vile | yeah |
07:05.31 | Qwell | seems to help for whatever reason |
07:05.33 | _Vile | and with no id3 |
07:06.24 | mog_home | yay i have gsm compiled |
07:06.24 | mog_home | unoptimized |
07:06.24 | mog_home | but compiled all the same |
07:06.24 | _Vile | optimize that |
07:06.29 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
07:06.32 | mog_home | meh i dont need it |
07:06.37 | mog_home | only a channel of audio at any point |
07:06.57 | _Vile | im going to smoke |
07:07.12 | _Vile | I need one, assid is busy |
07:07.52 | _Vile | interesting, I allow 1024 appearances to my * server... |
07:08.01 | Assid | yeah let me finish this |
07:08.04 | Assid | got a client one sec |
07:08.07 | _Vile | and I think 70 simultaneous |
07:08.24 | _Vile | I should probably reconfigure the appearance list |
07:08.26 | _Vile | hm |
07:08.27 | _Vile | blah |
07:08.28 | _Vile | bbiam |
07:09.54 | litage | do h323 users/friends register with a gateway or gatekeeper? |
07:09.59 | Assid | back |
07:10.24 | Assid | _Vile: i sox'd it from mp3 to ulaw |
07:10.33 | Assid | sox -c 1 blah.mp3 blah.ul |
07:10.41 | Assid | i also removed the mp3 files |
07:11.00 | Assid | so it only plays the ulaw ones.. but all i get is a FART outta the bass |
07:11.19 | *** join/#asterisk bartpbx (n=bartpbx@p54B0411C.dip0.t-ipconnect.de) |
07:11.30 | Assid | hrmm.. i think i better go head for a shower |
07:12.12 | litage | Assid: after being farted at, a shower's a good idea |
07:12.40 | Assid | yep |
07:12.52 | Ayano | how do you unlock a polycom ip 300? |
07:13.00 | Assid | unlock? |
07:13.15 | Ayano | to get to the configuration |
07:13.28 | Assid | use the username and password Polycom/456 |
07:13.32 | Qwell | Assid: You need to make it 8khz too :p |
07:13.41 | Assid | hrmm.. gotta find out how |
07:15.05 | Qwell | -r I think |
07:15.11 | Qwell | -r8000 |
07:16.09 | bartpbx | realtime voicemail does not seam to work with 1.2.1 anyone has the same problem? |
07:16.40 | Assid | Qwell: will reload res_musiconhold.so work aftr that? |
07:18.04 | *** join/#asterisk lorinc (n=ang@caracas-2904.adsl.interware.hu) |
07:19.09 | *** join/#asterisk koperniqs (n=koperniq@129.187.15.143) |
07:19.11 | koperniqs | hi |
07:22.26 | *** join/#asterisk |cleric| (n=dacleric@p5482A960.dip0.t-ipconnect.de) |
07:22.27 | Assid | bah.. i will do it locally.. remote is tough |
07:24.54 | *** join/#asterisk trixter (n=trixter@65.172.209.246) |
07:24.54 | *** part/#asterisk Ayano (n=Ayano_@adsl-70-245-190-90.dsl.spfdmo.swbell.net) |
07:26.00 | Qwell | bartpbx: yeah, hold on |
07:26.45 | Umaro | does anyone here have a tormenta 2 PCI card? quad-T1? |
07:26.54 | _Vile | Qwell, stop what you are doing. |
07:27.10 | bartpbx | It seams to we working in current branch 1.2 but not in the released 1.2.1 |
07:27.17 | Qwell | bartpbx: http://bugs.digium.com/view.php?id=5960 |
07:27.25 | _Vile | Qwell. |
07:27.26 | Qwell | bartpbx: yeah, it was fixed in the branch |
07:27.54 | _Vile | u done? |
07:27.54 | bartpbx | Qwell, thanks. so I'll use the branch |
07:28.04 | Qwell | _Vile: sure |
07:28.09 | _Vile | beer break man. you work too hard. |
07:28.15 | Qwell | break? please |
07:28.20 | _Vile | shup |
07:28.22 | _Vile | drink |
07:28.57 | Qwell | who says I'm not? |
07:29.08 | _Vile | you type too fast |
07:29.27 | _Vile | bbiam |
07:29.33 | _Vile | smoke |
07:29.37 | Umaro | some people practice typing fast with one hand while busy with the other, you know :P |
07:29.55 | _Vile | haha, I can type with three hands |
07:29.57 | edwar64896 | sox input.wav -r 8000 -c 1 output.ul |
07:30.07 | _Vile | bbiam |
07:30.47 | kuku5 | anyone have echo problems on a pri ? |
07:30.56 | bartpbx | is there an easy way to find changes from 1.2.1 to the current branch? |
07:31.19 | shido6 | im not touching your keyboard _Vile |
07:31.51 | Qwell | bartpbx: diff |
07:32.20 | bartpbx | ok |
07:32.35 | Umaro | so, guys? no one here has a tormenta 2 PCI card? it looks like this card should have a pci id of d00d, but mine has an id of 1000 |
07:32.55 | mog_home | nope umaro |
07:32.57 | Qwell | d00d, that's a 1337 card |
07:33.06 | mog_home | but just change the subvender id in the driver |
07:33.10 | mog_home | blamo it will probably work |
07:33.14 | Umaro | and tor2.ko won't load it |
07:33.19 | mog_home | yeah it wouldnt |
07:33.26 | Umaro | mog_home: tried that, couldn't find the id in the code yet |
07:33.50 | mog_home | look for PCI_ANY_ID |
07:33.51 | Qwell | ztd-eth.c |
07:33.56 | mog_home | in tor2.c |
07:33.57 | Qwell | 0xd00d? |
07:34.07 | Umaro | ztd-eth??? |
07:34.16 | Qwell | dunno |
07:34.19 | mog_home | why qwell would it be there |
07:34.27 | Qwell | got me. That's what grep said |
07:34.37 | Qwell | ztd-eth.c:#define ETH_P_ZTDETH 0xd00d |
07:34.47 | mog_home | yikes i dont have zaptel |
07:34.55 | mog_home | oh well i cant find it for you at the moment umaro |
07:34.59 | Qwell | That's the only place d00d is defined...will try -i |
07:35.30 | Qwell | tor2.mod.c? tor2-hw.h? |
07:35.40 | Qwell | tor2-hw.h: { PCI_VENDOR_ID_PLX, 0xD00D, PCI_ANY_ID, PCI_ANY_ID, 0, 0, (unsigned long)"Tormenta 2 Quad T1/PRI or E1/PRA" }, /* Tormenta 2 */ |
07:35.40 | Qwell | tor2.mod.c:MODULE_ALIAS("pci:v000010B5d0000D00Dsv*sd*bc*sc*i*"); |
07:35.40 | Umaro | oic, tor2-hw.h |
07:36.04 | Qwell | tor2ee.c:#define EEPROM_MAGIC 0xD00dF00d |
07:36.13 | Qwell | Who wants to bet they were incredibly hungry at the time? |
07:36.48 | Umaro | woot woot |
07:37.14 | Umaro | now let's hope the te210p works in this other box |
07:37.23 | Umaro | it should, supposedly it's just an alias to wct4xxp |
07:37.58 | mog_home | they use a "unified" driver |
07:40.36 | Umaro | yay, te210P works too |
07:40.38 | Umaro | today is a good day |
07:40.45 | mog_home | always |
07:40.52 | litage | do h323 users/friends register with a gateway or gatekeeper? |
07:42.00 | _Vile | qwell, why do you get the easy questions |
07:42.26 | Qwell | umm |
07:42.35 | Qwell | volume? |
07:42.38 | _Vile | lit, gw |
07:42.52 | _Vile | h.323 hmm |
07:42.56 | litage | _Vile: are you sure? |
07:43.05 | _Vile | lit, no |
07:43.16 | Qwell | Is there a difference between the two? |
07:43.29 | litage | _Vile: gnomemeeting has username and password options in its gatekeeper tab, but not its gateway tab |
07:43.35 | litage | Qwell: yessum |
07:44.02 | _Vile | lit, use the same, connect ton the same ip |
07:44.17 | litage | _Vile: ? |
07:44.24 | *** part/#asterisk edwar64896 (n=edwar648@220-244-62-6.static.tpgi.com.au) |
07:44.34 | _Vile | ignore gnomeet's wording |
07:44.35 | *** join/#asterisk edwar64896 (n=edwar648@220-244-62-6.static.tpgi.com.au) |
07:44.41 | litage | k |
07:45.12 | *** part/#asterisk edwar64896 (n=edwar648@220-244-62-6.static.tpgi.com.au) |
07:45.18 | *** join/#asterisk edwar64896 (n=edwar648@220-244-62-6.static.tpgi.com.au) |
07:45.40 | litage | _Vile: with chan_h323 loaded, * doesn't have commands like 'h323 show peers' |
07:45.43 | _Vile | now, back to important stuff.. |
07:46.01 | _Vile | use sip |
07:46.03 | Umaro | h323.. need to play with that soon.. dreading that |
07:46.29 | litage | _Vile: i'm forced to use h323 |
07:46.32 | _Vile | dont use h323 unless you feel like getting a beating |
07:46.44 | _Vile | ok |
07:46.56 | _Vile | I defer to others then |
07:47.00 | litage | doh |
07:47.05 | _Vile | im a sip man |
07:47.40 | _Vile | any possibility for sip? |
07:47.44 | litage | none |
07:47.45 | _Vile | actually |
07:47.53 | _Vile | explain your hardware to me |
07:48.35 | litage | _Vile: not much at the moment. just a couple of asterisk boxes |
07:48.53 | litage | _Vile: but my carrier only supports h323 |
07:49.12 | _Vile | lit, explain a sample route to me |
07:49.16 | _Vile | via h323 |
07:49.20 | Umaro | if I don't actually have a T1 hooked up to my digium card, but I want it to be used as a timer, do I need to put anything in zaptel.conf and zapata.conf, or do I just need to have the module loaded? |
07:50.03 | _Vile | I'd the differ this question already |
07:50.16 | litage | _Vile: ? |
07:50.24 | _Vile | but noone's awake? |
07:50.57 | litage | bob's softphone connects to *box1. joe's softphone connects to *box2. bob calls joe, so the call goes from bob to *box1 to *box2 to joe |
07:51.00 | mog_home | set with a 0 |
07:51.03 | litage | _Vile: is that what you meant? |
07:51.07 | _Vile | If noones awake, gimme errors, I can work erroes |
07:51.11 | _Vile | errors |
07:51.13 | mog_home | int zapata.conf |
07:51.17 | mog_home | er zaptel.conf |
07:51.24 | Umaro | mog_home: how do you mean? |
07:51.28 | _Vile | yes that's what I meant |
07:51.31 | _Vile | sec |
07:51.35 | _Vile | mog, you got this? |
07:51.37 | Umaro | mog_home: set a span? |
07:51.50 | mog_home | X,X,X,esf,b8zs first number is span, second number is timing source, third is build out |
07:52.06 | steff | hi all |
07:52.06 | mog_home | a 0 there meens card provides timing for the board |
07:52.19 | mog_home | 1 means take timing from far end and replicate it on board |
07:52.26 | mog_home | all that hardware can only have one source |
07:52.30 | mog_home | er each card |
07:52.38 | mog_home | so you can only have one 1 |
07:52.41 | mog_home | but many 0s |
07:52.44 | Umaro | right |
07:52.45 | Umaro | cool |
07:52.48 | mog_home | and on that note |
07:52.50 | mog_home | i sleep |
07:52.56 | kuku5 | would any of those cause echo ? |
07:52.58 | mog_home | peace we outa here |
07:52.59 | _Vile | mog |
07:53.00 | *** part/#asterisk mog_home (n=mogorman@user-24-236-84-48.knology.net) |
07:53.05 | _Vile | shiet |
07:53.47 | _Vile | litage |
07:54.56 | litage | _Vile |
07:54.59 | _Vile | that is what I meant |
07:55.06 | _Vile | though... |
07:55.24 | _Vile | explain the problem from boxX to me. |
07:55.50 | _Vile | pretend I'm bob. |
07:56.33 | litage | _Vile: the problem is that i can't get bob to register with *box1 using h323 |
08:02.00 | _Vile | I have to defer the question to someone who knows h323 |
08:02.27 | _Vile | <PROTECTED> |
08:02.37 | litage | thanks for trying to help :) |
08:03.22 | _Vile | gimme logs and I can maybe do more |
08:03.45 | _Vile | yw |
08:06.13 | Qwell | bed |
08:07.33 | _Vile | Qwell >>>> I can't answer litage's question |
08:08.14 | _Vile | Qwell <<<< going to bed, too. |
08:08.40 | _Vile | litage, go to bugs.digium.com. |
08:08.47 | *** join/#asterisk konfuzed (n=KonfuzeD@H135.C72.B0.tor.eicat.ca) |
08:09.02 | *** join/#asterisk BadaBing (n=BadBug@c-24-61-4-191.hsd1.ma.comcast.net) |
08:09.20 | BadaBing | hello everyone |
08:09.32 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
08:09.32 | _Vile | If it is not found there, then submit a 'feature request'. |
08:09.58 | BadaBing | <PROTECTED> |
08:10.08 | _Vile | Digium will take care of you from there. |
08:10.14 | BadaBing | section 3.7 and now i get this error Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?) |
08:10.36 | BadaBing | mp is working fine but it show that asterisk is stoped. when i call it works fine but can't connect to CLI |
08:10.59 | BadaBing | can anyone help ? |
08:11.53 | _Vile | how are you trying to connect |
08:11.57 | _Vile | SIP? |
08:12.01 | BadaBing | asterisk -r |
08:12.02 | _Vile | or IAX? |
08:12.17 | _Vile | protocol wise from one box to the other..... |
08:12.53 | BadaBing | Everything was working fine b4 I follow the wiki |
08:13.03 | BadaBing | I am using IAX and SIP |
08:13.07 | _Vile | ignore then wiki |
08:13.11 | _Vile | s/then/the |
08:13.26 | BadaBing | yeah but how can I fix this problem now |
08:13.42 | _Vile | explain to me what you were doing before the new changes |
08:13.47 | _Vile | first |
08:13.47 | BadaBing | I follow section 3.7 |
08:13.54 | _Vile | no, explain |
08:14.00 | BadaBing | okay |
08:14.28 | BadaBing | first I changed the the defult FOP Password |
08:15.05 | BadaBing | after that i issued a amportal restart and was still fine |
08:15.23 | BadaBing | everything was working fine... |
08:15.56 | BadaBing | then "nano /usr/sbin/safe_asterisk " changed CONSOLE=yes to no. |
08:16.03 | *** join/#asterisk JohnJacob (n=m00p@pcp0011543623pcs.mainf01.in.comcast.net) |
08:16.27 | BadaBing | I also uncomented the line "NOTIFY=your@emailadress.com" and placed my emailaddress |
08:16.36 | _Vile | what file |
08:16.54 | BadaBing | /usr/sbin/safe_asterisk |
08:17.09 | _Vile | OK anything else? |
08:17.32 | _Vile | look at what you changed, first. |
08:17.39 | BadaBing | then "MACHINE='yourmachinename' " then changed it to MACHINE='asterisk'' |
08:17.42 | _Vile | reverse what you changed. |
08:18.04 | _Vile | I'll be here for a couple more minutes |
08:18.13 | _Vile | the MACHINE= doesn't matter. |
08:18.21 | BadaBing | even the amp password? |
08:18.25 | _Vile | nor does the NOTIFY |
08:18.37 | _Vile | change back everything. |
08:19.50 | BadaBing | okay I changing them back |
08:20.16 | BadaBing | one problem though can't remeber MACHINE=`asterisk`what was it |
08:20.32 | _Vile | use localhost. |
08:20.39 | BadaBing | okay |
08:23.23 | BadaBing | _Vile, okay everything is back the way it was |
08:23.35 | _Vile | ok, now tell me your problems. |
08:24.25 | _Vile | one at a time. |
08:24.26 | BadaBing | when i issue asterisk -r |
08:24.40 | BadaBing | i get Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?) |
08:25.08 | testmachine | hmmm |
08:25.17 | _Vile | asterisk -vvvvr, if you can't get that -- do a ps -auxww | grep ast |
08:25.23 | testmachine | i am bussy with some zyxel prestige 2002 |
08:25.31 | testmachine | in asterisk |
08:25.37 | _Vile | let me know if you get a response from the ps |
08:26.04 | _Vile | actually, just do a ps -auxww| grep ast and msg me what you get |
08:26.05 | BadaBing | okay got responce from ps |
08:26.22 | _Vile | "ps -auxww|grep ast" |
08:26.33 | _Vile | msg me |
08:26.34 | BadaBing | okay |
08:26.46 | _Vile | test, explain. |
08:27.21 | _Vile | testmachine, explain. |
08:29.07 | _Vile | or not |
08:29.12 | _Vile | I'll be in bed in 5m |
08:30.56 | Assid | hrmm |
08:31.05 | Assid | note to self.. dont use CFLAGS when compiling asterisk |
08:32.04 | *** join/#asterisk chapeaurouge (n=chap@85.201.81.201) |
08:33.35 | testmachine | _Vile: |
08:33.35 | testmachine | ehm |
08:33.58 | testmachine | with these things you can call over a vpn from server to server, and one has a modem poty |
08:34.02 | testmachine | port |
08:34.06 | testmachine | wich can call too the outside |
08:41.15 | *** join/#asterisk trixter (n=trixter@65.172.209.246) |
08:45.31 | litage | is okay to run a gatekeeper (eg: gnugk) and asterisk on the same machine? |
08:47.16 | *** join/#asterisk habakuk (n=chatzill@c-24-6-173-113.hsd1.ca.comcast.net) |
08:49.27 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
08:49.35 | *** join/#asterisk af_ (n=af@ip-142-84.sn1.eutelia.it) |
08:49.36 | habakuk | hi can anyone enlighten me on Manager events? How do you "subscribe" to them? |
08:56.40 | edwar64896 | events are set by manager interface by default unless you explicitly don't want them |
08:57.17 | edwar64896 | you do this by setting "Events: Off" in the authentication clause. |
08:57.48 | edwar64896 | http://www.voip-info.org/wiki-Asterisk+manager+API is a good place to look for this sort of stuff |
09:00.00 | habakuk | edwar64896: thanks. I just saw that note there. Do you know if it's possible to subscribe only to certain types of events? |
09:00.48 | edwar64896 | event docco on voip-info is a bit light on in this regard... |
09:01.45 | *** join/#asterisk Gunnar (n=gunnar@bkkb-gw.bitcon.no) |
09:02.06 | *** join/#asterisk Gladius (n=hapchu@82.208.156.94) |
09:02.15 | Gladius | hello guys |
09:02.40 | Gladius | I have a problem with the queue agents. I use AgentCallbackLogin and it's just not working |
09:03.05 | edwar64896 | habakuk: there are 7 types of events (include/asterisk/manager.h) |
09:03.28 | *** join/#asterisk oej_ (n=Olle@apollo.webway.se) |
09:03.30 | *** part/#asterisk telmich (i=telmich@gpm/telmich) |
09:03.47 | edwar64896 | I think you subscribe using the read/write config lines in /etc/asterisk/manager.conf |
09:04.12 | habakuk | edwar64896: hmm.. interesting. I'll have to take a further look at that |
09:04.27 | edwar64896 | don't think that the subscription is via the API though. |
09:04.58 | testmachine | hm |
09:05.17 | testmachine | these zyxel things are kinda hwphones |
09:06.15 | habakuk | well if a user with low privelages will only get a low level of events |
09:06.57 | habakuk | there is an eventmask you can set |
09:08.23 | *** join/#asterisk darkskiez (n=darkskie@194.247.78.146) |
09:08.38 | Gladius | is anybody here using agents ? (agents.conf) |
09:13.37 | *** join/#asterisk tengulre (n=tengulre@222.90.66.4) |
09:14.46 | tengulre | hi,all! |
09:14.51 | tengulre | anybody active? |
09:17.18 | mover | sure :-) |
09:17.49 | mover | Gladius yes |
09:19.23 | Gladius | mover, my problem is very weird.... I can see the agent as loged in, but on show queue, it's Unavailable |
09:19.57 | Gladius | thing is, I'm using alphanumeric names for agents, not just digits |
09:21.02 | *** join/#asterisk ptiggerdine (n=ptiggerd@c220-239-93-118.rochd1.qld.optusnet.com.au) |
09:21.21 | ptiggerdine | mISDN still has a few bugs :) |
09:21.36 | mover | Gladius you have added the agents as members in queue.conf? |
09:22.18 | *** join/#asterisk densin (n=x@203-144-187-18.static.asianet.co.th) |
09:22.47 | Gladius | of course :) |
09:22.59 | densin | if I connect FXO to line which alread have phone, can it use for record ? |
09:23.14 | densin | hello all |
09:23.14 | Gladius | test-agent (test agent) available at '52' (musiconhold is 'default') |
09:23.26 | Gladius | Agent/test-agent (Unavailable) has taken no calls yet |
09:26.53 | viperdude | hi all |
09:27.04 | Gladius | hi viperdude |
09:27.10 | densin | if I connect FXO to line which alread have phone, can it use for record ? (I mean hardware capability , need to programming that ok ) |
09:27.49 | viperdude | just upgraded from 1.0.7 to 1.2.1 and found when a caller enters a queue it goes straight to MOH with out a welcome ot queue anoouncment, what do i need to enable it? |
09:33.45 | *** join/#asterisk digime (n=digime@60.49.10.234) |
09:40.01 | *** join/#asterisk rza (i=rza@todellisuus.net) |
09:40.11 | rza | Good morning! |
09:43.43 | *** join/#asterisk t0ke (n=t0ke@51.Red-83-46-136.dynamicIP.rima-tde.net) |
09:43.43 | rza | im sorry for disturbing possibly stupid question, but can i use asterisk to control ericsson md110 ? |
09:43.52 | *** join/#asterisk nagl (n=nagl@rih.zid-nw.wu-wien.ac.at) |
09:44.20 | rza | I got this client who wants a "Call this number" button to their intranet and I am lost with how should I initiate a call |
09:44.47 | rza | i dont mind paying for detailed help |
09:45.01 | t0ke | anyone with fedora core 3 and TE410P? |
09:45.21 | t0ke | when I do modprobe wct4xxp I receive this: ZT_CHANCONFIG failed on channel 97: No such device or address (6) |
09:45.56 | *** join/#asterisk zagaya971 (n=almeli@APointe-a-Pitre-102-1-3-9.w81-248.abo.wanadoo.fr) |
09:50.44 | *** join/#asterisk gnosys (n=ksford@ip68-9-201-250.ri.ri.cox.net) |
09:50.56 | *** join/#asterisk tzafrir_laptop (n=tzafrir@local.xorcom.com) |
09:53.31 | t0ke | anyone with fc3 and TE410P? |
09:54.37 | rza | t0ke: i presume thats more of a fc issue than asterisk issue |
09:55.23 | rza | have you tried asking #fedora or #linux? |
09:55.59 | rza | http://www.google.com/search?q=ZT_CHANCONFIG+failed+on+channel+97%3A+No+such+device+or+address+%286%29&sourceid=mozilla-search&start=0&start=0&ie=utf-8&oe=utf-8&client=firefox&rls=org.mozilla:en-US:unofficial |
09:56.04 | rza | gives quite many results |
09:57.04 | *** join/#asterisk TonyM (n=TonyM@adsl-solo-80-168-225-85.claranet.co.uk) |
09:57.19 | t0ke | rza..I looke a possible reason to that fail but I wanted to talk with anyone for to be sure ok? |
09:58.49 | t0ke | problem would to be jumpers setting about T1/E1 but I am not sure |
10:09.35 | *** join/#asterisk fulgas (n=fulgas@213.58.130.46) |
10:09.52 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
10:10.47 | *** join/#asterisk cjk (n=cjk@80.92.64.103) |
10:10.59 | cjk | hi, is there any yada user inside? how stable is it? |
10:15.26 | psirac | hi, i need help about asterfax, is there any asterfax user here ? |
10:19.39 | *** join/#asterisk bangawanga (n=andhecke@ppp-82-135-84-226.mnet-online.de) |
10:20.29 | bangawanga | hello, can anybody tell me how to listen the same call on more than one phone? |
10:21.18 | *** part/#asterisk bangawanga (n=andhecke@ppp-82-135-84-226.mnet-online.de) |
10:21.28 | *** join/#asterisk bangawanga (n=andhecke@ppp-82-135-84-226.mnet-online.de) |
10:21.42 | bangawanga | is anybody on line? |
10:22.23 | astcryz | always |
10:23.14 | bangawanga | we want to do sales training on our staff and i wonder how to listen to the same call on two phones. is it phone specific? |
10:23.29 | bangawanga | something to configure in asterisk? |
10:27.13 | bangawanga | no way to get an answer? silly question? |
10:31.40 | bangawanga | i am new to irc and asterisk, some hints would be useful. i do not know where to start serching. i dont want to set up smoething like conferencing |
10:32.07 | bangawanga | the called party should not know that more than one person is listening |
10:35.25 | *** join/#asterisk TonyM (n=TonyM@adsl-solo-80-168-225-85.claranet.co.uk) |
10:38.39 | *** join/#asterisk areski (n=areski@polar.es6.egwn.net) |
10:43.31 | edwar64896 | bangawanga? you still on? |
10:43.38 | bangawanga | yes |
10:43.45 | edwar64896 | Just saw your post. |
10:43.58 | edwar64896 | <PROTECTED> |
10:44.09 | rza | big butt |
10:44.28 | edwar64896 | i like big..... |
10:44.35 | edwar64896 | can you finish the song? |
10:44.59 | edwar64896 | the bigg butt is that if you do, you _will_ segfault your box. |
10:45.09 | edwar64896 | (assuming 1.2 svn trunk) |
10:45.11 | rza | butts and i can not lie |
10:45.20 | edwar64896 | ;-) |
10:45.35 | rza | edwar64896: did you see my post? |
10:45.45 | edwar64896 | check 0005941 on bugs.digium.com |
10:46.01 | edwar64896 | which has been the subject of much debate for the past week or soo. |
10:46.20 | edwar64896 | soz - which one? |
10:46.25 | edwar64896 | just got back on... |
10:46.33 | rza | edwar64896: about md110 integration |
10:46.51 | rza | i can explain briefly |
10:46.58 | edwar64896 | ta |
10:47.04 | rza | i need to make application which initiates call from number y to number x |
10:47.16 | rza | with ericsson md110 and application link |
10:47.23 | rza | just wondering if asterisk can help me here |
10:49.08 | edwar64896 | so, if I get this straight, you want to tell the md110 to "originate" a call from an extension (y) to another number (x) |
10:49.17 | edwar64896 | a bit like the "asterisk" origination command through AMI? |
10:49.22 | rza | yes |
10:49.28 | rza | using csta standard |
10:49.43 | rza | and im willing to pay for detailed help |
10:51.45 | edwar64896 | does md110 fully support ECMA CSTA std? |
10:51.51 | rza | yes |
10:51.59 | rza | you mean all phases ? |
10:52.06 | rza | or just the standard in general? |
10:52.36 | edwar64896 | where r u Timezone wise? |
10:55.18 | edwar64896 | I am looking through the ECMA documentation ... |
10:55.54 | edwar64896 | I think I am probably not the right person for this - no experience with CSTA and even less with the md110 |
10:56.19 | edwar64896 | If the md110 has a published API, it might be possible to interface to this from a client process and originate a call though... |
10:56.48 | bangawanga | sorry, i was absent because telephone |
10:56.54 | bangawanga | thank you for your time |
10:56.55 | rza | hmm |
10:57.12 | rza | edwar64896: if it has published api |
10:57.25 | rza | but it just says it conforms to ecma csta phase 1 |
10:57.58 | rza | im in gmt+2 |
10:58.36 | bangawanga | segfaul my box? what do you mean? |
10:59.16 | edwar64896 | banga: er... you box will fall over if you try chanspy one channel from two phones. |
10:59.32 | edwar64896 | read 005941 for the gory details... |
11:00.10 | edwar64896 | OK CSTA Phase one refers to Standard ECMA-180 |
11:00.36 | rza | edwar64896: i know, but i didnt find those standards very helpfull in implementation wise |
11:01.19 | edwar64896 | It's all written in ASN.1 format, |
11:01.29 | *** join/#asterisk mmmToop (n=chatzill@196.31.11.194) |
11:01.40 | rza | i know that also |
11:01.46 | rza | ASN.1 is scary |
11:01.55 | rza | all im looking for is where to start |
11:02.00 | rza | what apis to use in linux |
11:02.04 | rza | example applications |
11:02.06 | rza | so on |
11:03.02 | edwar64896 | you would need to start looking somewhere else - the MD110 docco |
11:03.13 | edwar64896 | (someone step in If I am getting it wrong...) |
11:03.32 | edwar64896 | The standard is simply a standard, and if ericsson appear to offer support for this standard |
11:03.43 | edwar64896 | then there must be some sort of implementation and/or interface to the implementation. |
11:04.02 | edwar64896 | Your best start is to look for a developers kit for the md110 and identify what tools are provided. |
11:04.19 | edwar64896 | You will then need to build an application and link in any development libraries provided by ericsson. |
11:05.03 | edwar64896 | It might be possible that they supply a sockets-type interface, but that would assume your md110 can get itself an IP address. |
11:08.41 | rza | i got application link |
11:08.49 | rza | which listens to tcp/ip port |
11:08.59 | rza | and i should be able to communicate that way |
11:10.35 | edwar64896 | so you can telnet to your md110 |
11:12.09 | rza | yes basically |
11:12.14 | rza | but i shouldnt have to do that |
11:12.27 | rza | sec |
11:12.38 | edwar64896 | http://cc.borland.com/Item.aspx?id=22836 |
11:12.43 | edwar64896 | interesting link... |
11:12.51 | edwar64896 | refers to an intel API which probably does what you want. |
11:13.02 | rza | http://www.ericsson.com/enterprise/library/brochures_datasheets/MD110/MD110_1022528.pdf |
11:15.04 | edwar64896 | looks like you need that media kit |
11:15.14 | rza | yeah |
11:15.21 | rza | that explains the applicationlink |
11:15.25 | rza | which i already got |
11:17.07 | mutilator | anyone have experience on a USR TC1000 v92 modem bank? |
11:24.54 | *** join/#asterisk Delvar (n=irc@host-83-146-53-34.bulldogdsl.com) |
11:30.33 | edwar64896 | Anyone on who has PRI expererince on E1 span? |
11:32.53 | *** part/#asterisk tartar (n=tartar@CPE0004e27b716e-CM014370001917.cpe.net.cable.rogers.com) |
11:35.02 | *** join/#asterisk taec (n=phil@eventhorizon.hosting365.ie) |
11:35.13 | *** join/#asterisk lehel (n=ddd@82.79.20.17) |
11:35.23 | taec | is there any way to display the original caller id, or at least flag the call in a certain way on a blind transfer? |
11:36.00 | rza | edwar64896: theres no sdk for applicationlink |
11:36.12 | rza | theres zero resources in developers section |
11:36.18 | rza | at ericsson.com |
11:37.11 | *** join/#asterisk zotz (n=zotz@24.231.47.168) |
11:38.57 | edwar64896 | looks like p4 of that ericsson flyer has the key |
11:39.21 | edwar64896 | "communication with the md110...... via tcp/ip over ethernet using ECMA-CSTA protocol" |
11:39.37 | edwar64896 | time to brush off your ASN.1 ;-) |
11:45.51 | rza | heh |
11:45.57 | rza | i was hoping to not go there |
11:48.38 | docelmo | whadup whadup? |
11:49.31 | cjk | is there a way to set the lastdata field in cdr? |
11:50.46 | *** join/#asterisk dushyanthh (n=chatzill@203.199.114.33) |
11:56.54 | docelmo | no not really... |
11:57.16 | docelmo | But whatever you want to be there is the last command you need to call before hanging up the call |
12:03.39 | cjk | docelm0: do you have an idea how i can easily add fields to the cdr? |
12:12.46 | *** join/#asterisk dummie (n=dummie@202.155.89.122) |
12:13.15 | dummie | can anyone help me how to set up phpagi? i'm very confused.. |
12:19.18 | docelmo | yes.. |
12:19.26 | *** join/#asterisk Whisk (n=whisk@whisk.gotadsl.co.uk) |
12:19.49 | docelmo | cjk, you have to add them to mysql then go into the cdr_mysql logger and put the information in for the fields your looking for |
12:20.14 | docelmo | «dummie» What do you wanna know? I am one of the developers for PHPAGI |
12:22.56 | lehel | `s gone.. |
12:27.36 | *** join/#asterisk gnosys (n=ksford@ip68-9-201-250.ri.ri.cox.net) |
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12:42.20 | *** join/#asterisk akrot (n=akrot@p54A75901.dip.t-dialin.net) |
12:42.31 | akrot | when i start asterisk with "asterisk -vvvgc" everything works fine. but if i start /etc/init.de/asterist start every call is "403 Forbidden" in xlite :/ |
12:52.46 | *** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin) |
12:52.48 | PakiPenguin | evening |
12:54.34 | *** join/#asterisk jimmy_deanPB (n=jhodapp@72.244.232.226) |
13:03.38 | dummie | hi docelmo |
13:03.57 | dummie | docelmo «dummie» What do you wanna know? I am one of the developers for PHPAGI |
13:03.58 | docelmo | took you long enough.. |
13:04.01 | RoyK | Filesystem Size Used Avail Use% Mounted on |
13:04.01 | RoyK | /data/var 2.3T 111M 2.2T 1% /var |
13:04.04 | RoyK | :D |
13:04.15 | docelmo | roy thats sick dude |
13:04.21 | dummie | just come back.. |
13:04.24 | docelmo | How mnay HD's do you have in that raid? |
13:04.30 | RoyK | docelm0: just 6 |
13:04.41 | RoyK | 6x500gig in raid5 |
13:04.50 | dummie | may i ask you how to set up phpagi? i'm a little confused with that.. |
13:05.27 | dummie | i just need to require('phpagi.php'); |
13:05.30 | dummie | that's all? |
13:12.45 | docelmo | I dont DCC chat.. |
13:12.52 | docelmo | nice dude |
13:13.02 | docelmo | 500? How much did that set you back? |
13:13.23 | docelmo | one line needs to be: require('phpagi.php'); |
13:13.38 | docelmo | next shoud be: $agi = NEW agi; |
13:13.39 | rza | <?php require('phpagi.php'); ?> |
13:13.47 | docelmo | thats it. You didnt download the examples? |
13:14.23 | rza | whats phpagi? |
13:14.27 | taec | is there any way to display the original caller id, or at least flag the call in a certain way on a blind transfer? |
13:14.52 | dummie | yes, i download it |
13:15.02 | dummie | but it seems to stuck.. don't work |
13:15.21 | dummie | does it require to insert it to extensions? |
13:16.00 | docelmo | exten => s,1,AGI(script.php) |
13:16.16 | *** join/#asterisk coppice (n=chatzill@82.194.17.210.dyn.pacific.net.hk) |
13:16.40 | dummie | my apache document root is in /var/www/html |
13:17.32 | dummie | should i place the script.php in /var/lib/sterisk/agi-bin ? |
13:18.06 | docelmo | yes |
13:18.20 | docelmo | PHPAGI has nothing to do with Apache just so you kno |
13:18.22 | docelmo | know |
13:18.56 | dummie | oh.. ic.. then it seems i've misunderstood about it.. :D |
13:19.29 | dummie | then can i run that script.php from web? |
13:19.46 | docelmo | No.. |
13:19.51 | docelmo | PHPAGI is ONLY for asterisk. |
13:20.12 | docelmo | the only output it produces is for asterisk to interperet. Not the web |
13:20.36 | *** join/#asterisk santiago (n=santiago@208.195.215.160) |
13:21.34 | dummie | oh ic.. btw.. if i have to view the sip show peers from web.. can i do it? |
13:22.21 | docelmo | yes.. You would use PHP or something and do a $output = `/usr/sbin/asterisk -rx "sip show peers"`; |
13:22.47 | docelmo | or use the manager api of phpagi and do it that way |
13:23.41 | dummie | <<docelmo>> or use the manager api of phpagi and do it that way |
13:24.14 | docelmo | You didnt read 1 of the docs did you? |
13:24.14 | dummie | can i have a tutorial about it? :) |
13:24.23 | docelmo | um, no.. |
13:24.31 | docelmo | http://phpagi.sf.net |
13:24.37 | docelmo | click on 2.0 documentation |
13:25.18 | sivana | is there a way to tranfer a call using a hook flash? |
13:25.30 | docelmo | depends on your ATA |
13:25.33 | *** join/#asterisk Dead-Bum (n=Satan@tor/session/x-d0a5b60ea055d5fd) |
13:30.38 | dummie | should i use the phpagi-asmanager.php? |
13:31.53 | *** join/#asterisk [TK]D-Fender (n=joe@toronto-HSE-ppp4122655.sympatico.ca) |
13:33.30 | docelmo | yes.. |
13:33.53 | docelmo | I am not going to go into great detail.. But yes.. Look at the documentation on the site. |
13:33.59 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
13:34.36 | dummie | umm.. yes.. i will go go read about it |
13:34.40 | dummie | thanx. :D |
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13:39.22 | znoG | crazy, i go into voicemail, press 0 then 4 to change my password and it sends me to change my temporary greeting |
13:40.16 | *** join/#asterisk FaithX (n=FaithX@202-6-145-116.ip.adam.com.au) |
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13:43.15 | mutilator | W00T! the insurance company came through, not even a deductable to fix the car |
13:43.30 | iCEBrkr | How'd you pull that off? |
13:44.42 | mutilator | i got sideswiped in a parking lot |
13:44.45 | mutilator | and i wasn't around |
13:44.58 | iCEBrkr | and you didn't have to pay your deductable? |
13:45.09 | mutilator | fix the while passenger side of the car from bumper to bumper |
13:45.10 | iCEBrkr | Merry Christmas |
13:45.19 | mutilator | yea because of the michigan no fault law or something |
13:45.23 | iCEBrkr | haha |
13:45.27 | mutilator | it pays the deductible for me |
13:45.40 | iCEBrkr | Florida is a 'no fault' state, but WTF does that mean when both people get cited for the accident?!? |
13:45.48 | jimmy_deanPB | hey [TK]D-Fender, I tried doing 1.2.1 here at work and now I can't get it to load anymore because of chan_zap.so |
13:45.58 | mutilator | i dunno |
13:46.01 | mutilator | i'm just super happy |
13:46.13 | iCEBrkr | mutilator: I would be |
13:51.40 | znoG | anyone use voipbuster? |
13:51.58 | iDunno | OK - seriously - I need new eyes. |
13:52.06 | iDunno | I just read voipbuster as vibrator :/ |
13:52.22 | znoG | ok then, anyone use a vibrator? |
13:52.25 | znoG | interesting question all the same |
13:52.27 | znoG | :) |
13:53.00 | *** join/#asterisk ctooley (n=ctooley@rrcs-24-227-212-162.sw.biz.rr.com) |
13:53.14 | ctooley | good morning |
13:54.18 | *** join/#asterisk Sedorox (i=brandon@smartserv/cna/Sedorox) |
13:56.07 | [TK]D-Fender | jimmy_deanPB : You'll need to download Zaptel as well |
13:56.13 | [TK]D-Fender | Anddo a rebuild of it |
13:56.42 | jimmy_deanPB | I did |
13:56.46 | jimmy_deanPB | blew everything up |
13:56.55 | jimmy_deanPB | now I'm in panic mode trying to go back to 1.0.7 |
13:58.44 | *** join/#asterisk paryl (n=paryl@216-201-177-82.res.logixcom.net) |
14:00.42 | paryl | g'morning all |
14:00.58 | Dandan | gm |
14:01.59 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
14:04.00 | *** join/#asterisk Theuni (n=ctheune@alphastar.gocept.com) |
14:04.08 | Theuni | howdi |
14:04.26 | docelmo | iCEBrkr more or less in flordia everyone is gonna pay no matter the fault |
14:04.36 | docelmo | btw.. WHADUP #ASTERISK! |
14:05.18 | Theuni | Weird. When I connect calling from outside (isdn trunk) within asterisk he gets the music on hold. except when i dial the number i want to connect him to and then hangup. he gets the dialtone _but_ is connected after the third party picks up the phone. any idea? |
14:06.09 | iCEBrkr | docelmo: yea, really. |
14:06.25 | docelmo | kinda sucks.. This is why I avoid them.. |
14:06.39 | docelmo | or try at least.. But people on the veterans.. geesh |
14:06.43 | iCEBrkr | docelmo: When I got down here, I was debating getting insuranc for my motorcycle and the lady on the phone was saying how 20% of the motorists down here don't have insurance. That's when I said "CUZ THEY CAN'T F'N AFFORD IT!" and I hung up |
14:07.04 | docelmo | haha |
14:07.28 | iCEBrkr | Dude, the insurance on my motorcycle is what I pay on my Blazer back home. |
14:07.37 | iCEBrkr | Ok,so it's a crotch-rocket, but still. |
14:08.40 | paryl | when i first set up my te205p, someone helped me get the config straight and had me set pridialplan and prilocaldialplan to 'local' in both cases. what do those parameters do? |
14:09.23 | iCEBrkr | docelmo: Good news. I just realized during yet another moment of insomnia last night that I'll be able to test my system as soon as I get that Sangoma card. |
14:09.23 | paryl | the reason i ask is, my telco has been debugging a call that won't go through from our asterisk system, and they said this: "We see private networking messaging from the PBX going to the network which may be killing the call" |
14:09.45 | docelmo | haha |
14:10.01 | docelmo | I fell asleep at like 9pm on the couch |
14:10.12 | iCEBrkr | Now if I can easily configure that Sangoma card to use CAS is another story |
14:10.13 | *** join/#asterisk Dandre (n=Dandre@was59-3-82-236-48-30.fbx.proxad.net) |
14:11.02 | iCEBrkr | Man, I got insomnia bad. I can be dead tired and head to bed, then all of a sudden the brain starts thinking about what I gotta do tomorrow, what I wanna do tomorrow, what I gotta get done, which bills I have to pay, etc, etc, the list goes on and on |
14:11.22 | docelmo | hehe.. fun stuff |
14:11.30 | docelmo | Um, I will stick with passing out.. |
14:11.39 | iCEBrkr | I'd fix it with 2 beers, but there's something wrong with pounding two beers before bed at 2am |
14:12.17 | *** join/#asterisk gpreche (n=gprenche@200-127-6-8.cab.prima.net.ar) |
14:12.17 | docelmo | um, ya you belong in Ybor |
14:12.28 | iCEBrkr | uugh |
14:12.39 | iCEBrkr | I haven't been to a single club in Ybor since I've moved her. |
14:12.41 | docelmo | incase people are wondering we live about 20mi from each other |
14:12.42 | iCEBrkr | +e |
14:12.59 | iCEBrkr | I've been to Banana Joes at Channelside tho. |
14:13.28 | docelmo | its not bad.. I recommend going to ybor on friday.. around 12am |
14:13.32 | iCEBrkr | The only times I've been to Ybor was on the motorcycle and that's when we all just loiter in the bank parking-lot across from Green Iquana |
14:13.35 | docelmo | thats when it gets fun.. |
14:13.42 | docelmo | hehe |
14:14.11 | docelmo | Joe's is cool.. Im thinking about doing channel side for new years.. Dont know whats happening there tho.. Maybe disney.. they are supposed to have a huge new years bash |
14:14.28 | iCEBrkr | Oh shit, I never thought about Disney for NYE |
14:14.56 | *** join/#asterisk UlbabraB (n=caplaz@host241-43.pool8172.interbusiness.it) |
14:14.59 | iCEBrkr | My roomie and I were thinking about head'n down to the Keys for NYE, but it'd be $200 a night on top of all the alochol + gas + food. |
14:15.16 | gpreche | Does anybody know if a Dialogic D41/D 4 channels/analog ISA board can be used with Asterisk ? |
14:15.26 | iCEBrkr | gpreche: Nope. |
14:16.11 | gnosys | Hey folks: polycom 501 phones... is the default firmware SIP software or something else (as with Cisco)? Also, good source of documentation on configuring them? Do they do they same thing as Cisco phones, searching tftp server for SIPDefault.cnf and SIPmacaddress.cnf? |
14:16.20 | gpreche | Thanks: Nope meaning "no it wont work" or "not tried" ? |
14:16.27 | paryl | my telco has never heard of a PRI dial plan, or overlap dialing |
14:16.30 | paryl | :\ |
14:16.46 | *** join/#asterisk gambolputty (n=gambolpu@64.74.225.131) |
14:16.49 | iCEBrkr | gpreche: Haven't tried and I'm pretty sure they don't work. |
14:17.12 | iCEBrkr | paryl: they probably never heard of LCR either. |
14:17.31 | gpreche | Me too ... Just found a ton of these and had some hope .... |
14:17.39 | iCEBrkr | Found a ton? |
14:17.52 | iCEBrkr | at $3k a piece... Hrrm, hit-up eBay |
14:18.07 | gpreche | Asterisk does not yet support E1 CAS (MFC R2), right ? |
14:18.29 | paryl | gnosys... that's a loaded question, but i've gone through it all with the 501 |
14:18.33 | iCEBrkr | gpreche: I sure hope it supports CAS |
14:18.59 | gnosys | which question is loaded? |
14:19.11 | docelmo | «paryl» its in asterisk.. whats the exact issue? |
14:19.52 | brad_mssw | gnosys: it depends on where you got the polycom from .... voipsupply ships sip firmware, others don't necessarily |
14:20.11 | docelmo | «gpreche» its based on card.. |
14:20.15 | docelmo | but yes I believe it does |
14:20.20 | gpreche | <iCEBrkr>: These old cards ??!! That expensive ??!! We have just replaced several equipments and have a bunch of them |
14:20.34 | paryl | docelmo: when we try to dial one of our vendors, 3 toll-free numbers, 1 long distance, it goes onto the PSTN, waits about 10 seconds and comes back with hangupcause 34 |
14:20.48 | paryl | docelmo: this is the ONLY number we've had issues with |
14:20.56 | gnosys | i got these from atacomm |
14:21.07 | docelmo | have you tried to diagnose the q.931 error? |
14:21.12 | iCEBrkr | gpreche: Seriously, go check out eBay. I guess it depends on the model, but they're a pretty penny |
14:21.15 | paryl | docelmo: but our telco (a CLEC) sees it failing on the local telco's network |
14:21.21 | gpreche | docelmo, iCeBrkr: Thank you ! |
14:21.29 | docelmo | for what? |
14:21.46 | paryl | gnosys... look here: http://paryl.blogspot.com/2005/10/making-polycom-501-work-we_113034274646163867.html |
14:21.59 | gpreche | docelmo: For your answer ... that it supports MFC R2 .... |
14:22.00 | paryl | i wrote that after screwing with them a lot |
14:22.05 | gnosys | thanks, paryl |
14:22.10 | paryl | np |
14:22.22 | iCEBrkr | gpreche: www.voip-info.org |
14:22.23 | tzafrir_laptop | if I want an incoming call from an analog line not to be Answer()-ed immedietly, is there any way to make Dial answer it if the call was actually made? |
14:23.47 | gpreche | Any idea if Asterisk would support an Aculab Primary rate E1 cas ISA card ? ((I know it supports the Prosody)) Today I have received old components and hope to be able to use it for anything ... |
14:24.26 | iCEBrkr | gpreche: www.voip-info.org |
14:24.30 | *** join/#asterisk fugitivo (n=ajf@209.13.242.149) |
14:25.11 | Gladius | I have a problem with agents in queues |
14:25.12 | tzafrir_laptop | in short: any way to avoid the "auto-answer" on analog lines? |
14:25.17 | Gladius | I'm using AgentCallbackLogin |
14:25.31 | [TK]D-Fender | gnosys : Poly's don't ship with the firmware, you typically get it from your vendor (who gest it off Poly's private support site). |
14:25.36 | Gladius | the login is successfull, but the agent still apears Unavailable in queue |
14:25.40 | gpreche | iCEBrkr: thanks |
14:25.46 | [TK]D-Fender | gnosys : What version of BR & SIP are you running now? |
14:26.05 | docelmo | «Gladius» Did you add them to the Queue? |
14:26.19 | docelmo | Member => Agent/??? |
14:26.26 | Gladius | of course |
14:26.43 | docelmo | Well they will show unavailable untill they take a call |
14:26.48 | Gladius | iuliana (Iuliana) available at '34' (musiconhold is 'default') |
14:26.57 | Gladius | Agent/iuliana with penalty 5 (Unavailable) has taken no calls yet |
14:27.03 | docelmo | ok then whats the issue? |
14:27.06 | docelmo | ohh |
14:27.09 | gnosys | thanks, [TK]D-Fender. This one reported a version number 1.6.2.0041 (I think) in booting up, but not sure exactly what that's referring to. From the boot messages, it seemed to have SIP software/firmware loaded. I got it from atacomm. |
14:27.16 | docelmo | Well send her a call and try it.. |
14:27.21 | gnosys | how do i find out BR & SIP versions? |
14:27.26 | docelmo | Which Im guessing you didnt being that your here |
14:27.40 | gnosys | paryl, at the end of that web page you sent me to, I see this: "The phone will automatically download the latest software and configurations and boot to working state. " |
14:27.49 | [TK]D-Fender | gnosys : They always come with SIP installed. Now you just need the XML sample pack and you're ready to go. |
14:27.56 | Gladius | docelmo, I actually did call the hotline |
14:28.03 | gnosys | Does that mean that the phone downloads the firmware from my ftp server or from a polycom server or what? |
14:28.04 | Gladius | her extension works |
14:28.04 | docelmo | And the outcome? |
14:28.15 | Gladius | can't enter queue |
14:28.25 | docelmo | then check your settings for the queue |
14:28.29 | paryl | so what i posted above... "We see private networking messaging from the PBX going to the network which may be killing the call", that's what the telco says. they say their software identifies something coming from my asterisk switch as being 'private networking messages' |
14:28.36 | docelmo | I setup 6 yesterday and they worked off the bat.. |
14:28.42 | docelmo | what version * you using? |
14:28.47 | iCEBrkr | BLARRRGH |
14:28.47 | [TK]D-Fender | gnosys : Its means that the phone will auto-upgrade if you put a newer firmware on your boot server. The boot server is your responsibility to set up, be it HTTP, FTP, or TFTP. |
14:28.54 | Gladius | I have 3 queues, but I'm only using sip/user |
14:28.58 | Gladius | and they work |
14:29.00 | [TK]D-Fender | I use FTP personally and life is good. |
14:29.02 | iCEBrkr | my Cablemodem at my apartment is lagged to shit and back.. 2000ms ping times |
14:29.03 | Seldon1975 | all your base, your base, base base |
14:29.14 | Gladius | but the problem is that an agent is still called while he's in a call |
14:29.18 | [TK]D-Fender | Seldon1975 : you have no chance to survive. Make your time. |
14:29.23 | Seldon1975 | ;) |
14:29.33 | *** join/#asterisk klictel (n=klictel@207.107.208.137) |
14:29.37 | Gladius | 1.2.0 bristuff |
14:30.29 | paryl | gnosys: yes, that's what i outlined in the blog post... put the files on your ftp server, and the phones dl it at boot |
14:30.29 | [TK]D-Fender | gnosys : Click here to get the support pack - http://www.freedomphones.net/polycom/files/SoundPoint_IP_SIP_1_6_2.zip |
14:30.43 | gnosys | [TK]D-Fender: how do I discover BR and SIP versions? Boot messages? |
14:30.43 | [TK]D-Fender | gnosys : what paryl said... |
14:30.57 | docelmo | iCEBrkr GET FIOS! |
14:30.59 | gnosys | thanks, guys. |
14:31.21 | [TK]D-Fender | You just look through the phone's menu (menu/status/platform, menu/status/phone |
14:31.28 | paryl | gnosys: i created a database and a python script to parse all of the params and create the config files on the ftp server... so now all i have to do is enter a new line in the table with the MAC, and boot the phone ;) |
14:31.28 | docelmo | ~pastebin' |
14:31.30 | docelmo | ~pastebin |
14:31.32 | jbot | i guess pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.com/ |
14:31.38 | docelmo | send your queues.conf and agents.conf there |
14:31.41 | docelmo | let me take a look at em |
14:31.50 | gnosys | Very nice, paryl! |
14:32.04 | gnosys | is it a mysql database or what? |
14:32.08 | paryl | i think once you get a handle on it, you'll love how easy they are to set up |
14:32.09 | [TK]D-Fender | I ust copy and rename mine and change 3 lines!!! Whhe for MEE! |
14:32.16 | iCEBrkr | FIOS? |
14:32.28 | docelmo | Verizon Fios.. |
14:32.34 | docelmo | 30/5 internet connection |
14:32.36 | iCEBrkr | Fibre? |
14:32.41 | docelmo | what I have at home.. :P |
14:32.50 | iCEBrkr | You think I'm made of money? |
14:33.01 | docelmo | static-71-251-95-4.tampfl.fios.verizon.net |
14:33.05 | docelmo | ya arent most? |
14:33.07 | paryl | D-Fender: well, for all of my users, i've got different SIP params and whatnot. also i create new directory.xml files for each phone based on the data in the table |
14:33.09 | iCEBrkr | Palm tress grow down here, not money trees :P |
14:33.40 | iCEBrkr | docelmo: That's the sad thing, I make good money, and I'm still broke as a joke |
14:34.01 | paryl | lol... 'broke as a joke' |
14:34.03 | docelmo | define good money to the 10's of thousands.. |
14:34.27 | docelmo | Cause your idea and mine are probably way different.. |
14:34.29 | Gladius | docelmo: http://pastebin.ca/33650 for queues.conf |
14:34.42 | docelmo | for instance I make around 70's.. and I like you am broke |
14:34.48 | iCEBrkr | docelmo: I said good money, not great money |
14:35.10 | [TK]D-Fender | paryl : Guess I'm lucky. I have only 2 profile types for my users. |
14:35.14 | iCEBrkr | I should be making $70k/yr |
14:35.21 | gnosys | So what's the general consensus on the preferred server to use with polycom phones? ftp, http, tftp? |
14:35.26 | Gladius | docelmo: http://pastebin.ca/33652 for agents.conf |
14:35.30 | iCEBrkr | docelmo: Doesn't matter, I'd still be broke living in a dumpy apartment |
14:35.40 | docelmo | You couldnt put both in the same? geesh |
14:35.50 | [TK]D-Fender | gnosys : FTP is the easiest and more secure than TFTP so it gets my vote... |
14:36.01 | Gladius | :) |
14:36.30 | paryl | gnosys: yes, ftp |
14:36.33 | docelmo | Well I own a house on water front w/ 1 ACRE, 2 cars, a lot of technology goodies.. |
14:36.52 | [TK]D-Fender | Damn, they completely depreciated the ASTDB stuff my dialplan used and now I have some serious re-writing to do :( |
14:37.44 | paryl | iCEBrkr: do you just like it there? otherwise... why stay? |
14:37.48 | file[laptop] | [TK]D-Fender: yup, because we're evil |
14:37.49 | docelmo | what is this: member => sip/horatiu,5 <---- ,5 what does that do? |
14:38.10 | Gladius | docelmo: it's the priority in the queue |
14:38.14 | paryl | docelmo: penalty |
14:38.22 | docelmo | ahh |
14:38.22 | iCEBrkr | paryl: I can't say I totally enjoy it. But I make a difference and I do see some sort of payoff. I'm in a good position here to make decent money |
14:38.26 | docelmo | but be new |
14:38.56 | iCEBrkr | paryl: Like for instance, I think my boss is getting pushed up to administration, my goal is to fill his shoes. |
14:38.56 | docelmo | Gladius, ok dude.. take out all of your member => sip/.... and change them to member => agent/... |
14:39.01 | docelmo | that should fix your issue |
14:39.16 | Gladius | docelmo, check the #2 queues :) |
14:39.39 | Gladius | the ones with sip/ are used now, and they work.... with the quirk I told you about |
14:39.43 | docelmo | My goal is to have my bosses job.. VP Operations.. |
14:39.45 | Seldon1975 | can someone elaborate on these messages I got: |
14:39.46 | Seldon1975 | == Spawn extension (default, t, 1) exited non-zero on 'SIP/226-e710<ZOMBIE>' |
14:39.46 | Seldon1975 | <PROTECTED> |
14:40.01 | docelmo | You got a internal server error |
14:40.10 | docelmo | :) |
14:40.17 | Seldon1975 | theres noi device on my network with that IP |
14:40.28 | Gladius | I just upgraded today from 1.0.8 to 1.2.0 |
14:40.30 | Seldon1975 | does Asterisk think there is? |
14:40.39 | Seldon1975 | maybe in sip.conf... |
14:40.44 | Gladius | and it behaves the same way |
14:41.03 | paryl | iCEBrkr: i'm in a privately owned company, lucked out really, and i'm as high as i can get in position, but whenever i feel like i'm not making enough, i normally get a raise because they're scared i'll leave ;) |
14:41.21 | *** join/#asterisk Katty (n=angela@68.112.15.110) |
14:41.24 | docelmo | hehe paryl same here.. |
14:41.32 | Gladius | me too :)) |
14:41.34 | syle | what do you say to get that? |
14:41.39 | docelmo | speaking of which I have an evaluation tomorrow.. But I have the day off!!!! |
14:41.40 | Katty | hi lads. |
14:41.40 | iCEBrkr | paryl: ew |
14:41.43 | paryl | docelmo: yay for being indispensible! |
14:42.02 | iCEBrkr | paryl: Lets just put it this way, I was only here 4 months, and I Got a $350 christmas bonus last year. |
14:42.15 | docelmo | I am IT Admin(only guy in building who knows all of the quirks of our network), Sr. Data Engineer and Telecom Engineer |
14:42.18 | iCEBrkr | paryl: Ya know, that's the only sigificant bonus I've EVER gotten? |
14:42.48 | iCEBrkr | I've NEVER gotten a christmas bounus. Not to mention, I wasn't even here a full year and they still included me |
14:42.49 | docelmo | I got a 500 and a raise |
14:42.50 | docelmo | :) |
14:43.02 | iCEBrkr | I got run of the mill her, I can't complain |
14:43.05 | iCEBrkr | +e |
14:43.06 | docelmo | I was there for 4 months also.. |
14:43.51 | Gladius | docelmo: do you know a way to stop * from calling an agent while he's already on a call ? |
14:44.01 | docelmo | How long have you been with your comapny ice? I have been with mine since September before last |
14:44.19 | docelmo | is the agent directly connect to that asterisk box? |
14:44.31 | iCEBrkr | docelmo: Just a little over a year.. Maybe closer to a year and a half. |
14:44.43 | Gladius | docelmo: yes, using SJPhone (and it doesn't have a setting to take only one call) |
14:44.46 | iCEBrkr | docelmo: I just moved here last Aug. |
14:44.46 | docelmo | Cause my situation is different than yours.. I have agents in Aussie and tampa.. So my asterisk routes some calls via pstn or voip.. |
14:45.06 | iCEBrkr | I used Asterisk to get this job :P |
14:45.10 | docelmo | hehe |
14:45.22 | iCEBrkr | ie. Picked up a Tampa phone number, used my buddy's Tampa address. :P |
14:45.29 | iCEBrkr | Meanwhile, the phone rang in Cleveland |
14:45.32 | docelmo | I fell into asterisk as the guy leaving knew it and I was hired cause I heard of it. |
14:46.10 | paryl | iCEBrkr: before i got into my sitch, i was woo'd by a company in SF, and i went there to tour, put in my notice, etc. so i took a month to train my replacement (out of the kindness of my heart) and at the end of the month he explained that there was no way he'd be able to do it. at that point things started getting better because the president realize how important i am |
14:46.22 | paryl | so i've gotten much better raises, etc |
14:46.40 | docelmo | Gladius, I Dunno dude.. Sounds like you have a unique situation if they all register to the same box |
14:46.54 | iCEBrkr | paryl: Kinda sucks you gotta do things that way |
14:46.55 | paryl | iCEBrkr: maybe that's what you need to do :) |
14:47.15 | paryl | very true, but sometimes the suits don't know who hold the real power in their company |
14:47.26 | paryl | not talking revolution or anything... they just don't get it |
14:47.28 | iCEBrkr | paryl: I've already sold my soul to this place. We're changing technology and programming language. My boss asked me to promise that I'd stick around and make things happen. |
14:47.46 | Dandre | Hello all, |
14:47.53 | paryl | iCEBrkr: back when i put in my notice i used the following line: |
14:48.33 | iCEBrkr | paryl: and like I said, I'm in a good position here-- which also means I'm aware of the financials. |
14:48.33 | Dandre | I have problem setting moh on my 1.2 installation: |
14:48.33 | paryl | iCEBrkr: "it sucks to have to choose between eating and paying bills" |
14:48.34 | iCEBrkr | paryl: ie. they can't afford me. |
14:48.34 | paryl | the owner understood what he was doing at that point |
14:48.34 | docelmo | anywho.. off to do some home repairs.. Im out.. |
14:48.34 | iCEBrkr | docelmo: l8r |
14:48.36 | paryl | iCEBrkr: ah |
14:48.36 | iCEBrkr | docelmo: oh! |
14:48.38 | iCEBrkr | docelmo: wait |
14:48.42 | iCEBrkr | docelmo: Mon the 19th |
14:48.47 | iCEBrkr | SLUG meeting |
14:48.57 | gnosys | hi Katty, wha's up? |
14:49.16 | docelmo | ok will get with you on how the hell to get there |
14:49.19 | iCEBrkr | ok |
14:49.42 | paryl | iCEBrkr: automate business processes so that the company saves money. softcosts are IT paydirt :) |
14:49.49 | iCEBrkr | paryl: That's the idea. |
14:49.59 | iCEBrkr | paryl: Apparently, there's only a handful of us that understand this |
14:50.06 | iCEBrkr | Out with the old, in with the new |
14:50.11 | paryl | indeed |
14:50.14 | Dandre | http://pastebin.ca/33654 |
14:50.56 | paryl | Dandre: that happens if the box is underpowered |
14:51.04 | paryl | (not the ONLY cause, mindyou) |
14:51.32 | Dandre | The moh doesn't play at all |
14:52.17 | Dandre | I don't think it is underpowerd because my other setup with 1.0.7 is same architecture |
14:52.20 | paryl | Dandre: yeah... it'll either play very choppy or not play at all |
14:52.40 | paryl | Dandre: you could try doing a rawplayer type setup |
14:52.52 | paryl | of course, you'll have to convert the mp3s to raw |
14:53.06 | paryl | but i've seen it improve performance quite a bit |
14:53.23 | KriS83 | Anyone here using Asterisk 1.2.0 with Junghanns cards? |
14:53.36 | Dandre | I don't understand why moh stops just after starting |
14:53.37 | Gladius | KriS83: me |
14:53.55 | KriS83 | Gladius, may I query you for 2 mins? |
14:54.08 | *** join/#asterisk DaPrivateer (i=Privatee@CRIMSON.OFF-HOURS.COM) |
14:54.10 | *** join/#asterisk trixter_ (n=trixter@65.172.209.246) |
14:54.15 | iCEBrkr | dial bitch |
14:54.17 | iCEBrkr | WTF? |
14:54.28 | Dandre | I don't have any zap channel in my setup is that the reason? |
14:54.31 | Gladius | KriS83: better talk here |
14:54.36 | KriS83 | k |
14:55.17 | paryl | do changes to the PRI settings in zapata.conf require a total restart of asterisk, or can i just reload a portion? |
14:55.36 | KriS83 | Gladius, may I ask where you are from? The problem I have is that even when setting Set(CALLERID(name)="") the number is passed to the party being called |
14:56.04 | Gladius | KriS83: it can be a "feature" of your telco |
14:56.09 | Gladius | I'm from Romania |
14:56.31 | file[laptop] | KriS83: you're setting the name, not the number to nothing |
14:56.31 | [TK]D-Fender | paryl : at the minimum you'll need to reload zaptel |
14:56.32 | iCEBrkr | ahh there we go |
14:56.53 | iCEBrkr | Oh sure, now he doesn't answer |
14:56.54 | KriS83 | Well with our normal Agfeo PBX the number is set to anonymous correct... |
14:57.11 | KriS83 | file, I thought for the name it would be CALLERID(name) |
14:57.19 | file[laptop] | it is... that's what you said |
14:57.20 | KriS83 | ops |
14:57.26 | Gladius | exten => s,1,SetVar(CALLERIDNUM=212312) |
14:57.29 | KriS83 | sorry.. above I mean number |
14:57.32 | KriS83 | not name :) |
14:57.35 | file[laptop] | Gladius: you can't do that |
14:57.46 | Gladius | file[laptop]: wanna bet ? :D |
14:57.50 | file[laptop] | and setvar is deprecated |
14:57.54 | file[laptop] | it's set now |
14:57.55 | paryl | Dandre: have you set up an extension to go straight to MOH? |
14:58.13 | file[laptop] | Gladius: yes I'll bet, because the callerid stuff isn't a normal variable |
14:58.51 | Gladius | file[laptop]: it's been working for me for the last months |
14:59.04 | file[laptop] | are you sure? |
14:59.10 | KriS83 | But as said above with the normall ISDN PBX I can do calls without sending the number.. |
14:59.16 | file[laptop] | because there's a reason there was functions to change the callerid |
14:59.22 | KriS83 | with * it always sends a number :( |
14:59.22 | file[laptop] | er applications |
14:59.26 | file[laptop] | which have now become functions |
14:59.39 | Gladius | file[laptop]: yes, but on 1.0.8 |
14:59.51 | KriS83 | Gladius, I was asking for 1.2.0 though :) |
14:59.52 | file[laptop] | using set to change the variable does it as a variable on the channel, using the application or function actually changes it on the channel |
14:59.52 | Gladius | I've just upgraded today to 1.2.0 and didn't test it yet |
15:00.18 | Dandre | No I use the hold from another extension |
15:00.22 | Gladius | so, what's the "proper" way to do it in 1.2.0 ? |
15:00.45 | file[laptop] | Gladius: you use dialplan functions with set |
15:00.47 | KriS83 | exten,s,3,Set(CALLERID(number)=12345) |
15:01.07 | paryl | which echo canceller do you guys prefer for sound quality? i've been using MARK2 and the person on the PSTn always gets broken up |
15:02.07 | KriS83 | Is there some dial flag to override the set callerid? |
15:03.04 | Katty | it's a beautiful day in the neighborhood, a beautiful day for a neighbor |
15:03.45 | Dandre | paryl: I have added a MOH Test extension and same result |
15:04.32 | [TK]D-Fender | paryl : I find the A104d echo canceller to be the next best thing since sliced bread :) |
15:04.45 | Katty | iDunno: you, sir, have insaned. |
15:05.02 | iDunno | Katty: I have not! I was always this way! |
15:05.02 | paryl | [TK]D-Fender: >:-| |
15:05.11 | Katty | iDunno: oh ah, i see. |
15:09.08 | iDunno | sometimes I'm slightly less insane, but that doesn't generally last very long :) |
15:10.21 | *** join/#asterisk wunderkin (i=kev@ip68-226-113-228.ph.ph.cox.net) |
15:10.59 | iDunno | Katty: I think I can win on the bored stakes, I'm reinjecting mail in to an exim mail spool |
15:11.20 | iDunno | (or rather, at the moment I'm waiting for the queue to disolve a bit so that I can reinject some more) |
15:11.49 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
15:11.49 | *** mode/#asterisk [+o anthm] by ChanServ |
15:12.44 | Katty | iDunno: i see. |
15:13.16 | anthm | hello |
15:13.36 | *** join/#asterisk tRSS (n=tRSS@202.174.142.2) |
15:17.11 | Katty | Hmmhesays: come on man, it's not even 10 yet |
15:17.25 | Hmmhesays | red bulls were 3 for 5 bucks today |
15:17.36 | Katty | eww, red bull has taurine |
15:17.42 | Katty | do you know what you're drinking? |
15:17.46 | synthetiq | whats wrong with taurine |
15:17.48 | Hmmhesays | bull piss? |
15:17.51 | Katty | ye |
15:17.59 | Hmmhesays | crushed baby rabbits? |
15:17.59 | Katty | that's what it is. |
15:18.07 | Katty | made from the bile of oxen! |
15:18.13 | Hmmhesays | mmm good stuff |
15:18.20 | Katty | you, sir, have insaned. |
15:18.33 | Katty | look at what that stuff is doing to you! |
15:18.43 | Hmmhesays | can't be any worse than that hundreds of cigarettes and gallons of beer I drink every month |
15:18.47 | Katty | next you'll be riding that bike in the ice |
15:18.57 | Katty | like some sort of rebel. |
15:19.09 | synthetiq | yea i makes u give attention to katty which is not allowed in this channel |
15:19.10 | Katty | Hmmhesays: you should stop smoking |
15:19.43 | Hmmhesays | the bassist brought over his ibanez rg guitar ot practice a couple days ago, I felt like an 80's hair band rockstar |
15:20.01 | *** join/#asterisk coppice (n=chatzill@82.194.17.210.dyn.pacific.net.hk) |
15:20.23 | Katty | Hmmhesays: you practically are. |
15:20.30 | Katty | Hmmhesays: except you're too young, and don't have enough hair yet |
15:20.42 | Hmmhesays | i'm going to keep it at this level of spikedness |
15:21.18 | *** join/#asterisk trym (n=trym@062016209171.customer.alfanett.no) |
15:21.24 | Hmmhesays | i remember this summer after ribfest like 10 people came and asked me if I was joan jett's guitarist, he had the same hair |
15:21.31 | trym | is the newest version in the svn rep 1.2.1 ? |
15:21.44 | Katty | Hmmhesays: haha. |
15:21.57 | Katty | Hmmhesays: maybe that's why there were following you around at the bar last time ;) |
15:22.03 | *** join/#asterisk Spla4t1 (n=splat1@rrcs-24-172-35-197.midsouth.biz.rr.com) |
15:22.06 | Katty | Hmmhesays: s/there/they/ |
15:22.16 | Hmmhesays | lol |
15:22.42 | Katty | Hmmhesays: of course i was the only SANE one that decided to remain sober |
15:22.59 | Hmmhesays | LOL |
15:23.14 | *** join/#asterisk kpettit (n=keith@69.15.174.114) |
15:23.18 | Hmmhesays | we need karaoke at the next cluecon |
15:23.24 | Katty | oh god. |
15:23.32 | Hmmhesays | or open mic night |
15:23.39 | synthetiq | coed naked karokee |
15:23.50 | Hmmhesays | i'd be the nights entertainment |
15:23.56 | Katty | Hmmhesays: you would. |
15:24.01 | Katty | Hmmhesays: i'd laugh at you allllll night long. |
15:24.10 | Hmmhesays | lol |
15:25.01 | Hmmhesays | "this is my artistic interpretation of 80's glam rock as played on a mandolin, thank you" |
15:25.12 | *** join/#asterisk aldem (n=ww@80.237.176.16) |
15:26.06 | *** join/#asterisk ian_k (n=ian@gateway.digium.com) |
15:26.40 | *** join/#asterisk zpn (n=xpn@gateway.digium.com) |
15:26.56 | [TK]D-Fender | Hmmhesays : My latest toy : http://www.bosscorp.co.jp/products/en/GT-8/index.html |
15:27.05 | *** part/#asterisk zpn (n=xpn@gateway.digium.com) |
15:27.27 | Hmmhesays | damn you [TK]D-Fender |
15:27.29 | *** join/#asterisk diego_br (n=Diego@200.208.241.178) |
15:27.39 | Hmmhesays | i'm torn between that and a podxt live |
15:28.02 | Katty | speaking of toys. |
15:28.06 | Katty | i need a new moocow. |
15:28.11 | [TK]D-Fender | Pod is a great amp/speaker emulator, but they don't have an all-in-one that reeally shines. |
15:28.24 | Hmmhesays | the xt live does |
15:28.33 | *** join/#asterisk Abbas (i=Abbas@203.81.194.195) |
15:28.41 | Hmmhesays | 30+ amp models and 80+ effects |
15:28.42 | [TK]D-Fender | Behringer's VAMP makes Pod's smaller stuff obsolete at a fraction on the price. |
15:28.53 | Katty | Hmmhesays: find me one. |
15:28.56 | [TK]D-Fender | The GT-8 replaced my defective Boss VF-1 half-rack |
15:29.01 | Katty | Hmmhesays: a cute one :> |
15:29.12 | Hmmhesays | we run behringer power amps for our tops |
15:29.23 | [TK]D-Fender | German audio = good :) |
15:29.30 | Hmmhesays | god priced audio |
15:29.39 | [TK]D-Fender | I need to replace my dead Samson wireless rig and I'm set... |
15:29.52 | Hmmhesays | i need to go wireless |
15:29.58 | Hmmhesays | i hate being tethered |
15:30.05 | Katty | Hmmhesays: but you're so cute tethered. |
15:30.06 | [TK]D-Fender | (andget the electical bits on my Dean fixed up) |
15:30.15 | [TK]D-Fender | rawr ;) |
15:30.36 | Hmmhesays | but if I go wireless instrument I gotta go wireless monitors also |
15:31.09 | [TK]D-Fender | For monitors? Power requirement sucks.. not worth the extra lag |
15:31.42 | Hmmhesays | yeah but its hard to play when you can't hear it |
15:31.54 | Hmmhesays | especially for me who likes to spend a fair amount of time off stage |
15:32.01 | [TK]D-Fender | No I mean monitors are good, just frget the wireless... that set stage gear |
15:32.11 | [TK]D-Fender | I miss the thrill of the spolight.... |
15:32.50 | Hmmhesays | i love the thrill of mingling with the audience playing some kickass riff |
15:32.59 | Hmmhesays | "you give love a bad name" comes to mind |
15:33.10 | Katty | eww, mingling. |
15:33.16 | *** join/#asterisk Samoied (n=Samoied@popeye.opens.com.br) |
15:33.24 | Katty | Hmmhesays: that's dirty :< |
15:33.29 | paryl | how do you set the rxgain/txgain from the CLI? |
15:33.34 | paryl | is it possible? |
15:33.49 | [TK]D-Fender | I played with a Van Halen tribute band (KB / guitar), and did tons of Bon Jovi, Warrant, Whitesnake, Poison, and so on... |
15:33.52 | Katty | paryl: i've never done it......cause i think you have to stop asterisk and start it again for changes to take effect. |
15:34.30 | paryl | :\ |
15:34.59 | [TK]D-Fender | paryl : you need to reload ZAptel in CLI if you don't want a total restart |
15:35.06 | Hmmhesays | the 80's stuff is a lot of fun once you've leared it. |
15:35.17 | Hmmhesays | killer to learn though especially if you play lead |
15:35.19 | Dandre | paryl: Do you know how should I convert from mp3 to raw files? |
15:35.36 | [TK]D-Fender | I love my 80's....... |
15:35.51 | Hmmhesays | my rig right now consists of a mesa boogie, and my les paul |
15:36.13 | Hmmhesays | solid body, dual humbuckers and a tube amp |
15:36.27 | *** join/#asterisk shanky (i=jramirez@217.11.114.145) |
15:36.31 | paryl | Dandre: yes! |
15:36.32 | [TK]D-Fender | My problem is I am hooked on 24 fret w/ floating trems. Makes my choices more limited... |
15:36.34 | shanky | hi, good afternoon |
15:36.35 | paryl | :) |
15:36.42 | *** join/#asterisk negatendo (n=negatend@c-24-9-136-152.hsd1.co.comcast.net) |
15:36.43 | Dandre | :-) |
15:36.50 | Dandre | would you tell me how? |
15:37.28 | paryl | Dandre: you need sox and lame |
15:37.31 | paryl | http://pastebin.ca/33661 |
15:37.51 | [TK]D-Fender | Hmmhesays : Here's some ancient junk of mine http://www.geocities.com/tk_dfender/ |
15:37.52 | paryl | i wrote that to help automate the task... i put the mp3 file in the same directory as that script |
15:38.10 | paryl | and use something like ./covert music |
15:38.23 | paryl | it converts and puts the file in the moh directory |
15:39.10 | Dandre | I don't have lame :-( |
15:39.22 | Katty | sometimes i don't have a clue what you guys are talking about. |
15:39.27 | *** join/#asterisk negatendo (n=negatend@c-24-9-136-152.hsd1.co.comcast.net) |
15:40.06 | paryl | Dandre: what distro? |
15:40.16 | Dandre | debian sarge |
15:40.26 | Hmmhesays | cool |
15:40.52 | Katty | yay for debian! |
15:41.03 | iDunno | debian++ |
15:41.22 | Katty | yay for people helping you set up asterisk too ;) |
15:41.23 | paryl | just download the src and compile it |
15:41.27 | paryl | it take 2 minutes |
15:41.35 | paryl | takes* |
15:42.04 | paryl | Katty: boo-yah |
15:42.42 | Katty | huh? |
15:42.48 | Katty | that does not parse. |
15:43.19 | paryl | erm... i agree |
15:43.20 | *** join/#asterisk digime (n=digime@60.49.10.234) |
15:43.54 | paryl | http://www.urbandictionary.com/define.php?term=booyah |
15:44.29 | Katty | clinching fist and thrusting their elbow downward vertically ?! |
15:44.35 | Katty | that sounds painful. |
15:44.45 | paryl | only the first time |
15:44.49 | Gladius | is there a way to reload queues without restart ? |
15:44.53 | testmachine | debian is the future :P |
15:44.54 | paryl | after the nerve damage it really isn't so bad |
15:45.02 | paryl | viva la debian! |
15:45.11 | testmachine | :) |
15:45.11 | Katty | why can't people just yay? |
15:45.23 | testmachine | becouse i dont like the word yay |
15:45.24 | Gladius | yay |
15:45.24 | paryl | because then you would /get it/ |
15:45.24 | testmachine | :P |
15:45.51 | *** join/#asterisk darkskiez (n=darkskie@194.247.78.146) |
15:46.39 | wunderkin | Gladius, reload app_queue.so |
15:46.56 | Gladius | wunderkin, thanks |
15:47.36 | Gladius | guys, do you know what is the Internet for ? |
15:47.37 | iDunno | how about hoorah! |
15:47.49 | iDunno | Gladius: depends on who you believe... |
15:48.03 | iDunno | Gladius: if you believe the academics, then it's for sharing information |
15:48.22 | Gladius | well, these guys know for sure (and I kinda agree with them) that it's for p0rn :) |
15:48.25 | Gladius | http://www.youtube.com/watch_fullscreen?video_id=lr_HR-iIlYg&l=196&fs=1&title=WOW%20meets%20Porn |
15:48.28 | iDunno | Gladius: if you believe the marketting peeps it's to "get your company known, and provide another place to do business" |
15:48.41 | iDunno | Gladius: and if you believe the porn industry, it's all about the porn. |
15:49.04 | *** join/#asterisk GXTi (n=scuba@freenode/developer/GXTi) |
15:49.20 | *** join/#asterisk MikeJ[Laptop] (n=ircatjer@mi.origenfinancial.com) |
15:49.24 | Gladius | It's all about the Pentiums, baby :) |
15:49.29 | Gladius | do you know this song ? |
15:49.34 | paryl | seriously... what would the internet be without porn? think about it. |
15:49.55 | shanky | how could I know how many calls (sip or Zap ones) is happening in a asterisk? is there any command for the CLI? |
15:50.00 | Gladius | watch the movie :) |
15:50.11 | Gladius | shanky: show channels |
15:50.38 | iDunno | paryl: be useful again? |
15:51.02 | Katty | paryl: uhh |
15:51.04 | Gladius | what would the internet be without spam ? |
15:51.06 | Katty | paryl: that's cause i AM a girl |
15:51.14 | shanky | Gladius: thanks |
15:51.19 | paryl | oh noes |
15:51.20 | iDunno | Gladius: less damned annoying and processor intensive? |
15:51.27 | Dandre | paryl: I haven't found the sources for lame do you have a pointer? |
15:51.29 | paryl | there's a girl in the house |
15:51.36 | paryl | Dandre: one sec |
15:51.36 | Katty | oh grow up |
15:51.40 | Gladius | :)) |
15:51.47 | _Sam-- | f'n teliax |
15:51.52 | Gladius | paryl, she didn't say how old she is |
15:52.02 | Gladius | I knew once an ircop who was 65 :) |
15:52.12 | Gladius | horny as hell, I might add :) |
15:52.23 | paryl | haha |
15:52.34 | paryl | is sourceforge down? |
15:52.46 | Gladius | iDunno, she hit on me A LOT ! |
15:52.57 | paryl | there it goes |
15:53.06 | iDunno | Gladius: ahh - horny *and* insane, then? :) |
15:53.10 | _Sam-- | sourceforge may be up....but teliax is down |
15:53.18 | _Sam-- | and so if the phone service for my business |
15:53.21 | paryl | Gladius: i'm notoriously slow, so i wouldn't know when one was hitting on me. |
15:53.23 | _Sam-- | if = is |
15:53.36 | synthetiq | me thinks katty is mid 20s. has a butch hair cut and wears thick glasses |
15:53.44 | paryl | Dandre: http://prdownloads.sourceforge.net/lame/lame-3.97b2.tar.gz?download |
15:53.45 | *** join/#asterisk wrzf (n=chrisf0r@adsl-70-227-44-29.dsl.dytnoh.ameritech.net) |
15:53.47 | synthetiq | pale white |
15:53.54 | synthetiq | chunky |
15:53.57 | Gladius | paryl: same here, and I figured it out... so it was obvious as hell |
15:54.06 | wrzf | Good morning all |
15:54.11 | synthetiq | willign to bet 3$ paypel |
15:54.12 | paryl | Gladius: that's the best kinda flirting, imho |
15:54.15 | synthetiq | paypal |
15:54.15 | [TK]D-Fender | synthetiq : Please leave "thinking" to qualified professionals :) |
15:54.16 | Dandre | thanks paryl |
15:54.18 | paryl | w00t! slutty girls! |
15:54.22 | paryl | Dandre: np |
15:54.33 | synthetiq | 3$ paypal bet who wants to take me on |
15:54.43 | wrzf | Got a ? if somebody is up to it.. (-: |
15:54.57 | iDunno | wrzf: woah - next you'll tell us it's asterisk related? |
15:55.08 | Gladius | wrzf, why do you make me turn my head the opposite direction ? |
15:55.16 | wrzf | hahahah |
15:55.20 | iDunno | wrzf: and not to do with porn, or (potentially) offending Katty |
15:55.23 | wrzf | Good morning to you too 9-; |
15:55.31 | wrzf | hahaha |
15:55.41 | wrzf | probably easy for you guys |
15:55.52 | paryl | yes, probably |
15:55.52 | Gladius | iDunno, did you see the movie ? |
15:55.57 | paryl | IF YOU WOULD ASK IT |
15:56.00 | paryl | ;) |
15:56.07 | Katty | iDunno: possibly. |
15:56.29 | iDunno | Gladius: nah - had the word WOW in it, which I was guessing was World Of Warcraft, so I didn't bother :) |
15:56.30 | Katty | synthetiq: fyi, i am pale. |
15:56.36 | Katty | synthetiq: but hardly chunky ;) |
15:56.38 | wrzf | I have an inbound did pointing at my asterisk IP address. Do I need to run another program to get that did to ring in. I get Line congestion erros when I call it. |
15:56.41 | Gladius | trust me, it's funny :) |
15:56.50 | paryl | pale = best ever |
15:56.52 | Katty | synthetiq: but feel free to ask anyone who's met me in person (= |
15:56.54 | synthetiq | damn, why didnt soemone bet me, i would have 3$ more paypal |
15:57.06 | synthetiq | anythign bigger than size 6 pants is chunky |
15:57.17 | synthetiq | :discuss: |
15:57.42 | iDunno | synthetiq: just which countries sizes are you going by? and you could politely STFU ;) |
15:57.43 | paryl | wrzf: how can you have a did pointing at an ip address? |
15:57.50 | paryl | a did is PSTN |
15:57.52 | Katty | synthetiq: perhaps you belong in south korea then (= |
15:57.59 | wrzf | its as UK Number just pointing at my IP |
15:58.02 | synthetiq | this country? |
15:58.10 | brettnem | "this" |
15:58.12 | iDunno | synthetiq: which is? |
15:58.22 | synthetiq | united states |
15:58.38 | wrzf | when I call it I get line Congestion errors I put in my inbound routing table but i still get these errors |
15:58.41 | *** join/#asterisk alexhopper (n=a27386@CPE000103d29ae2-CM001225dfdfe0.cpe.net.cable.rogers.com) |
15:58.56 | Katty | synthetiq: girls aren't barbies. |
15:58.58 | iDunno | UK size 6 is insanely small, what the hell would you get hold of?! |
15:59.03 | Katty | synthetiq: if you don't like that, go find a male. |
15:59.04 | paryl | wrzf: what does the debug show? |
15:59.06 | synthetiq | my ex was a size 4 |
15:59.08 | Hmmhesays | www.realdoll.com nsfw |
15:59.09 | _Sam-- | if anyone knows the guy from teliax tell him his stuff is broke |
15:59.19 | jimmy_deanPB | ok, just wanted to contribute this: if people complain about echo problems with their zaptel channels, have them upgrade from * 1.0.x series to 1.2.x...completely cured my problems. |
15:59.26 | synthetiq | me bets size 10 |
15:59.32 | paryl | wrzf: when the call is coming in... do you see it at all? |
16:00.09 | sivana | heh |
16:00.18 | Katty | iDunno: twisted would do it, too |
16:00.19 | wrzf | yes |
16:00.26 | sivana | HRISTMAS SPECIAL! |
16:00.26 | sivana | Place your doll order before Dec. 31st with payment in full |
16:00.26 | sivana | by check, Money Order or Wiretransfer |
16:00.29 | *** join/#asterisk VincentMX (n=vincent@c3eea0597.cable.wanadoo.nl) |
16:00.39 | sivana | 250 off! |
16:00.40 | VincentMX | hi |
16:01.09 | VincentMX | can you use asterisk to call to a normal phone from a computer? |
16:01.15 | sivana | yes |
16:01.18 | VincentMX | without modem? |
16:01.24 | alexhopper | yes |
16:01.25 | sivana | you can use your computer to call a normal phone with * |
16:01.31 | VincentMX | ok |
16:01.44 | VincentMX | is asterisk hard to use? |
16:01.50 | sivana | depends |
16:01.58 | file | if it keeps you from using it, and asking questions, yes! |
16:01.59 | alexhopper | yeah - it jsut sends the phone signal out over the internet...and then conects youinto the land lines |
16:02.02 | sivana | :) |
16:02.07 | [TK]D-Fender | VincentMX : You need some sort of interface to the PSTN. A card in your * server, a VoIP gateway (SPA-3000), a VoIP PSTND provider (take your pick of the dozens out there) |
16:02.40 | [TK]D-Fender | HAPPY MORNING TO YOU FILE!!!!!! |
16:02.43 | Seldon1975 | anyone know if it's possible to rollback the sip.ld on a polycom 301 |
16:02.52 | file | [TK]D-Fender: ah go to hell :P |
16:03.14 | VincentMX | what's a PSTN? |
16:03.44 | Seldon1975 | i updated the sip.ld and now my phone doesnt talk to the * server |
16:03.46 | [TK]D-Fender | VincentMX : PSTN is just another term for the normal phone network. |
16:03.51 | Seldon1975 | i need to roll back |
16:03.55 | Katty | file: suffocater. |
16:03.55 | [TK]D-Fender | VincentMX : Typically just an analog phone line. |
16:04.01 | VincentMX | ok |
16:04.07 | Katty | alexhopper: paws off, plskthx. |
16:04.08 | VincentMX | like a modem? |
16:04.23 | file | modem != analog phone line |
16:04.28 | file | modem = modulator/demodulator |
16:04.40 | VincentMX | ok |
16:04.49 | Katty | that was on the a+ test. |
16:04.52 | alexhopper | an analog phone line is just the traditional phone system most everyone has |
16:05.03 | Katty | most everyone that's not in here, you mean. |
16:05.06 | VincentMX | so i do need a modem for calling to normal phones? |
16:05.06 | file | analog is silly |
16:05.15 | alexhopper | Naturally |
16:05.24 | file | well, if you use a modem to get on the internet and plan to use a provider - SURE! |
16:05.43 | VincentMX | i use cable to go to Internet |
16:05.54 | iDunno | Katty: when putting people in to boxes, don't forget to cut air holes. |
16:05.57 | alexhopper | then thats how you calls will go out |
16:06.05 | file | if you choose to use a VoIP provider. |
16:06.06 | Katty | iDunno: file is special and can transcend walls. |
16:06.15 | file | and transcend reality! |
16:06.21 | iDunno | coo. |
16:06.30 | iDunno | that's quite an impressive feat. |
16:06.32 | _Sam-- | just dont use teliax |
16:07.05 | VincentMX | alexhopper, so i just install asterisk and then i can call to normal phones? |
16:07.11 | iDunno | hmm - I need to do some asterisk with normal analogue lines at sometime in the not too distant :/ |
16:07.14 | Dandre | Ok now moh works with raw files but not with mp3 files. May be I have fogotten sth |
16:07.18 | _Sam-- | or if you can live with your business and employees not having phone service on a regular basis, then you should choose teliax. |
16:07.30 | file | VincentMX: you have to pay either a VoIP provider, or get hardware that allows you to connect your phone line into asterisk |
16:07.38 | VincentMX | ok |
16:07.45 | *** join/#asterisk slak- (i=slak@shudup.before.you.get.rewted.biz) |
16:07.46 | *** join/#asterisk Smidge (n=steve@steve.enta.net) |
16:08.14 | slak- | hi, i just set up email notification for vm and .wav attachments, i'd like to run it for all old messsages for one mailbox |
16:08.15 | alexhopper | It's just like any phone service, you have to pay for it. |
16:08.17 | slak- | can that be done? |
16:08.20 | *** join/#asterisk t0ke (n=t0ke@51.Red-83-46-136.dynamicIP.rima-tde.net) |
16:08.24 | t0ke | hi |
16:08.27 | file | slak-: not really no |
16:08.46 | _Sam-- | just forward them from the voicemail options to a new box |
16:08.47 | slak- | put it on your list file |
16:08.50 | _Sam-- | and then have the new box email |
16:08.55 | file | my list is already long enough |
16:08.58 | file | like other stuff of mine! |
16:09.19 | t0ke | I have connected one E1 to one of my TE410P ports, anyone know if there is any command which show what ports are connected? |
16:09.34 | slak- | type "help |
16:09.35 | slak- | " |
16:09.37 | slak- | in the cli |
16:09.56 | t0ke | how I launch cli mode in asterisk? with -g? |
16:10.14 | slak- | -r |
16:10.15 | file | asterisk -vvvvgc |
16:10.18 | Spla4t1 | asterisk -r |
16:10.20 | slak- | if its running |
16:10.24 | file | will start asterisk running in console |
16:10.32 | file | asterisk -r to attach to an exisiting running asterisk |
16:10.43 | file | and kill -9 to make users scream |
16:10.55 | t0ke | tnks |
16:11.22 | Dandre | I haven't installed mpg123 :-( |
16:11.32 | alexhopper | Outside the snow is falling and friens are calling to you, come on it's lovely weathr for a sleighride together with you! |
16:11.43 | *** part/#asterisk VincentMX (n=vincent@c3eea0597.cable.wanadoo.nl) |
16:11.50 | Hmmhesays | anyone know of any uk anac numbers? |
16:12.10 | paryl | Dandre: you don't need to |
16:12.16 | viperdude | anyone know how to get asterisk to stop presenting a call from a queue to a agent who is already on a call? |
16:12.54 | *** join/#asterisk funxion (n=nunya@mtnuser.icgws.com) |
16:12.58 | paryl | viperdude: the best method would be for their ACD extension to be only one line |
16:13.06 | Hmmhesays | i'm still trying to sort out the BT clid issue |
16:13.11 | Dandre | I have installed it and it works now Why shouldn't I install mpg123? |
16:13.17 | viperdude | is that configured on asterisk or on the phone? |
16:13.30 | paryl | Dandre: because it sucks |
16:13.57 | Seldon1975 | Dandre there is a security flaw |
16:14.06 | Seldon1975 | but it does work |
16:14.06 | Katty | paryl: lolzitSUCKSkthx |
16:14.10 | Katty | paryl: how about a /real/ answer |
16:14.15 | Seldon1975 | i just installed it yesterday |
16:14.29 | paryl | also license issues... you don't want to use it for a busines |
16:14.30 | paryl | s |
16:14.41 | paryl | Katty: yes, ma'am |
16:14.49 | Dandre | So how should I play mp3 moh ? Is there another tool for that? |
16:14.51 | taec | is there any way to display the original caller id, or at least flag the call in a certain way on a blind transfer through asterisk? |
16:14.52 | Katty | paryl: good boy. |
16:15.04 | paryl | Dandre: i just gave you the stuff you need for raw |
16:15.40 | Dandre | shure but the sound quality isn't the same. Maybe a configuration issue? |
16:15.45 | klictel | use rawplayer |
16:15.52 | paryl | Dandre: try this: http://www.orderlyq.com/asteriskqueues.html |
16:15.59 | paryl | it explains using rawplayer |
16:16.11 | alexhopper | sont sing |
16:16.14 | alexhopper | *dont |
16:16.51 | LostFrog | How can you afford the long distance charges? <grin> |
16:16.57 | alexhopper | lmao |
16:17.07 | file | that was bad |
16:17.11 | paryl | who's the guy that everyone says is a queues god? i need help |
16:17.12 | file | but ironically he's just across town |
16:17.15 | file | :P |
16:17.23 | file | which makes it harder to avoid him |
16:17.26 | alexhopper | yeah - 20 minute walk |
16:17.28 | alexhopper | MWa ha ha |
16:17.33 | alexhopper | I'm coming over! |
16:17.39 | file | probably longer today, considering the storm :P |
16:19.05 | alexhopper | hehe = yeah |
16:19.05 | iDunno | heh |
16:19.12 | LostFrog | Noone ever calls me.. |
16:19.17 | LostFrog | I like it that way. :) |
16:19.21 | Dandre | I have seen that 1.2 has its own mp3 player. Why doesn't it ply mp3 moh on my box? |
16:19.25 | file | LostFrog: you probably have an extension that's hard to get to |
16:19.27 | file | :P |
16:19.33 | Katty | file: ! |
16:19.38 | file | Katty: ! |
16:19.41 | iDunno | not getting called is good. |
16:19.46 | alexhopper | alexhoppper: ! |
16:19.47 | iDunno | it means that nothing has broken :) |
16:19.51 | LostFrog | I have people who give messages to other people to forward to me, because they are afraid to call me. :) |
16:19.56 | file | alexhopper: wow, people sure like you |
16:20.01 | alexhopper | They do |
16:20.17 | paryl | hey, i found a wierd memory leak yesterday, but i don't know why it happens |
16:21.04 | file | paryl: do you know how it happens, ie: how to reproduce it? |
16:21.13 | Katty | file: are you not answering your messageseses? |
16:21.14 | file | if so, file a bug report with details and maybe someone will help track it down |
16:21.17 | file | mmmkthxbye |
16:21.28 | file | Katty: I got none! |
16:21.43 | paryl | file: if you have a timeout in a Dial with the next rule Goto a queue, and that extension is dialed from a queue... it kills * within about 5 minutes |
16:21.56 | file | tjat |
16:21.59 | file | that's odd |
16:22.04 | file | Katty: lemme exit and reopen iChat, 1 sec |
16:22.11 | Katty | file: k'then |
16:23.09 | file | that's very very odd |
16:23.20 | paryl | file: depending on the number of incoming calls, natch, but it does go down. and before it goes down, every call that comes into the queue goes to dead air |
16:23.22 | iDunno | # why do you get all the love in the world |
16:23.33 | Katty | file: and prime. |
16:23.44 | paryl | file: but SIP-SIP calls work fine up until * dies |
16:24.15 | LostFrog | Beyond 3, are there any prime triangular numbers? |
16:24.25 | *** join/#asterisk ahmuck_jr (n=chatzill@p115n4.ruraltel.net) |
16:24.47 | paryl | LostFrog: short answer 'yes' with an 'if', long answer 'no' with a 'but' |
16:24.54 | paryl | :) |
16:24.57 | *** join/#asterisk fulgas (n=fulgas@213.58.130.46) |
16:25.10 | ahmuck_jr | can one take a standard pc and put three modems in it and use it ? |
16:25.20 | file | modems? |
16:25.23 | file | like regular ol' modems? |
16:25.27 | file | and use them for voice? |
16:25.34 | ahmuck_jr | yes |
16:25.37 | LostFrog | ahmuck_jr: with IRQ sharing or full PCI modems, yes. |
16:25.41 | *** join/#asterisk lorinc (n=ang@caracas-1065.adsl.interware.hu) |
16:25.54 | file | no, you can't use modems like that... |
16:25.59 | file | not for asterisk |
16:26.04 | ahmuck_jr | so if i have three lines in my house, take a standard pc and put three us robotics or 3com modems in them |
16:26.10 | Katty | file: what about muffins. |
16:26.13 | LostFrog | ohh.. not for asterisk. |
16:26.16 | Katty | file: can you use muffins like that in asterisk? |
16:26.24 | file | Katty: yes, thanks to chan_muffin! |
16:26.44 | Delvar | you lot realy are insane |
16:26.59 | Katty | file is practically my twin. |
16:27.03 | LostFrog | I prefer chan_bakedgoods |
16:27.09 | Katty | but impractically not so. |
16:27.18 | Katty | he's smarter. |
16:27.29 | Katty | .........in voip. |
16:27.39 | Katty | but i'll clearly slingshot around the earth before he does. |
16:27.40 | LostFrog | lol.. I like the qualification. |
16:28.32 | kpettit | I'm trying to use astfax to send faxes through asterisk |
16:29.09 | LostFrog | Unfortunately, Katty, you will accidentely reach escape velocity. |
16:29.11 | paryl | Katty: you better plan your trajectory well |
16:29.16 | kpettit | There is only breif install instructions and they are specific to qmail and not very informitive. Anybody know of any more general instructions |
16:29.26 | kpettit | or how to get astfax to work with a different MTA? |
16:29.30 | paryl | LostFrog: we had the same thought :) |
16:29.41 | `Sauron | kpettit: welcome to open source |
16:30.04 | LostFrog | peryl: I am sure she will plan ok.. It's that extra .5 seconds of burn time that will get her. |
16:30.11 | Katty | i'm always well away of escape veloticity requirements, etc. |
16:30.16 | Katty | s/away/aware/ |
16:30.20 | kpettit | `Sauron, yeah it's fun. But this app dosen't even have a help or a readme which makes it more difficult to experiment |
16:30.40 | kpettit | if you rrun the app it just sits there doing nothing so I'm not sure what to do with it to test |
16:30.44 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
16:30.45 | Katty | 7mps |
16:31.10 | `Sauron | Katty: Mew |
16:31.31 | Katty | v_esc=sqrt(2*G*M/R) (= |
16:31.36 | Katty | `Sauron: mew. |
16:31.36 | paryl | Katty: is that your major? (given our previous conversation on photons being split) |
16:31.46 | Katty | twisted[asteria]: ! |
16:32.34 | iDunno | Katty: what's R in this case? |
16:32.42 | slak- | RATE |
16:32.51 | LostFrog | distance from the center of the earth, iDunno. |
16:32.52 | *** join/#asterisk SpaceBass (n=SpaceBas@static-71-251-230-2.rcmdva.fios.verizon.net) |
16:32.54 | Katty | radius |
16:32.55 | LostFrog | Radius |
16:33.04 | slak- | roach |
16:33.07 | twisted[asteria] | Katty, :P |
16:33.08 | iDunno | right - cool - that's what I thought :) |
16:33.12 | alexhopper | Katty: You dont wanna be twins w/ fily |
16:33.13 | twisted[asteria] | hard math in the morning makes me combust |
16:33.21 | Katty | twisted[asteria]: oops, sowwie. |
16:33.23 | file | bah |
16:33.30 | LostFrog | Beans do that to me, twisted[asteria]. |
16:33.32 | twisted[asteria] | Katty, heh, s'ok |
16:33.45 | iDunno | only an hour till home time though :) |
16:33.54 | iDunno | (or, in this case, pub o'clock) |
16:34.27 | Katty | twisted[asteria]: we really need to go raid nasa :> |
16:34.28 | paryl | iDunno: can we swap timezones? just for today? sigh. |
16:34.36 | Katty | twisted[asteria]: did you ever figure out what to do with the semi? |
16:34.38 | *** join/#asterisk Porks (n=nao@200.231.120.138) |
16:34.58 | slak- | mark spencer also works on gaim right? |
16:35.00 | iDunno | paryl: fraid not, I like this timezone |
16:35.00 | alexhopper | My physics teacher used to work for nasa! |
16:35.06 | twisted[asteria] | Katty, actually, yea, there's a new road being built, and an overpass that could easily conceal it |
16:35.08 | slak- | this homeboy here claims he went to school with mark spencer |
16:35.08 | twisted[asteria] | :P |
16:35.11 | alexhopper | Maybee he can get us codes! |
16:35.13 | paryl | stupid central time... |
16:35.17 | slak- | in alabama |
16:35.25 | iDunno | though, if someone could give me some sunlight to counteract this flickering flourescant tube, that'd be great. |
16:35.32 | twisted[asteria] | slak-, it's not unlikely. |
16:36.12 | *** join/#asterisk mog_work (n=mogorman@gateway.digium.com) |
16:36.27 | *** join/#asterisk Lurr (i=user@adsl-067-034-122-207.sip.mia.bellsouth.net) |
16:36.34 | iDunno | ta :) |
16:37.13 | paryl | can upping txgain actually reduce echo? that seems counter-intuitive to me |
16:37.15 | LostFrog | Umm.. that is a metric ruler, Paryl.. It's 50mm. |
16:37.27 | *** part/#asterisk Lurr (i=user@adsl-067-034-122-207.sip.mia.bellsouth.net) |
16:37.42 | Katty | twisted[asteria]: excellent |
16:37.44 | Katty | twisted[asteria]: when do we go? |
16:37.50 | paryl | again with the stupid place i live! why can't we freaking use real measurements? |
16:37.52 | Hmmhesays | yes I smoked another pc |
16:37.55 | Hmmhesays | i've got the touch |
16:38.31 | slak- | dudes can i do this: PHONE=${MACRO_EXT} exten => ${PHONE},1,Macro(macroname,${PHONE} |
16:38.34 | LostFrog | Hmmhesays: that doesn't take much. |
16:38.35 | slak- | will that work |
16:38.41 | slak- | for all my extensions |
16:38.57 | LostFrog | I didn't know you could match on a variable. |
16:39.25 | paryl | slak: what are you trying to accomplish? |
16:39.34 | slak- | minimize my extensions.conf |
16:39.37 | slak- | right now i have: |
16:39.38 | twisted[asteria] | Katty, soon. when the eagle lands on fertile soil, and the mouse rolls up in pygme oil |
16:39.41 | slak- | PHONE15=119 |
16:39.41 | slak- | PHONE16=120 |
16:39.45 | slak- | exten => ${PHONE00},1,Macro(oneline,${PHONE00}) |
16:39.45 | slak- | exten => ${PHONE01},1,Macro(oneline,${PHONE01}) |
16:39.49 | slak- | i want to just have two lines |
16:39.54 | slak- | for all 20 extensions |
16:40.01 | paryl | dude, it sooo much easier than that |
16:40.14 | slak- | and use the extensions dialed for the number |
16:40.20 | *** join/#asterisk generalhan (i=general_@ip67-90-64-2.z64-90-67.customer.algx.net) |
16:40.31 | twisted[asteria] | Hmmhesays, smoking PC's is bad for your health |
16:40.38 | iDunno | slak-: erm - well, surely you know the range of numbers for the phones, so use... |
16:40.41 | generalhan | whats up guys ! |
16:40.47 | *** join/#asterisk mut (n=animenod@65.111.201.79) |
16:40.56 | file | twisted[asteria]: don't you have stupid people to help become the next Vonage? |
16:41.09 | iDunno | exten => _1XX,1,Macro(oneline,${EXTEN}) |
16:41.17 | iDunno | or similar. |
16:41.21 | slak- | hmm |
16:41.33 | paryl | slak: http://pastebin.ca/33669 |
16:41.36 | slak- | ty |
16:41.51 | paryl | slak: and that even makes it so when they call their own extension it dials VM :) |
16:42.07 | paryl | i've got a block of 100 DID's, and that works for all of them |
16:42.22 | generalhan | I have a quick question ... I have my calls being recorded by ${CALLERIDNUM} but the recordings are being saved as "user" for $CALLERIDNUM anyone have any idea why its not showing the phone number of the caller ? |
16:42.46 | slak- | paryl: your extensions are 40XX? |
16:43.14 | twisted[asteria] | file, pardon? |
16:43.26 | file | twisted[asteria]: nevermind then! |
16:43.39 | alexhopper | Dont mind him, he's just an idiot |
16:44.03 | alexhopper | file: just kiding |
16:44.09 | file | :p |
16:44.29 | Smace | file: Vonage? |
16:44.33 | iCEBrkr | DIE DIE DIE DIE DIE DIE |
16:44.39 | file | I refuse to die |
16:44.49 | iCEBrkr | kill -9 file |
16:45.00 | twisted[asteria] | nooo |
16:45.02 | twisted[asteria] | rm -rf file |
16:45.04 | alexhopper | Smace: A little known company that uses something calle VoIP |
16:45.07 | Smace | killall file |
16:45.15 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
16:45.16 | twisted[asteria] | you cannot kill a file, but you can delete it |
16:45.16 | iCEBrkr | file: fine.. init 0 |
16:45.23 | file | lies |
16:45.27 | file | you can't delete me |
16:45.30 | file | I control your filesystem! |
16:45.30 | iCEBrkr | haha |
16:45.31 | file | muahahahaha |
16:45.32 | twisted[asteria] | wanna bet? |
16:45.33 | shmaltz | I like this: |
16:45.34 | shmaltz | http://www.engadget.com/entry/1234000497072377/ |
16:45.37 | *** part/#asterisk file (n=jcolp@mctnnbsa24w-142167049176.nb.aliant.net) |
16:45.42 | iCEBrkr | LOLOLOL |
16:45.42 | paryl | slak: sorry, yes |
16:45.45 | *** join/#asterisk file (n=jcolp@mctnnbsa24w-142167049176.nb.aliant.net) |
16:45.46 | *** part/#asterisk file (n=jcolp@mctnnbsa24w-142167049176.nb.aliant.net) |
16:45.51 | generalhan | HAHAHAHA |
16:45.51 | *** join/#asterisk file (n=jcolp@mctnnbsa24w-142167049176.nb.aliant.net) |
16:45.55 | file | that's cheating |
16:45.55 | iCEBrkr | haha |
16:45.56 | twisted[asteria] | file, told ya :) |
16:46.05 | slak- | paryl; i dont understand your macro |
16:46.08 | slak- | can you break it down |
16:46.13 | slak- | in msgs? |
16:46.15 | twisted[asteria] | that's a fun little trick. |
16:46.16 | LostFrog | file: I can just remove the inode entry. |
16:46.17 | iCEBrkr | twisted[asteria]: he's like a virus.. You delete him and he keeps coming back |
16:46.40 | alexhopper | And throwing muffins at you... |
16:47.42 | twisted[asteria] | file, lol |
16:47.49 | twisted[asteria] | file, i don't answer calls that don't have ANI |
16:48.50 | file | :P |
16:49.17 | file | that was my cellphone |
16:50.51 | iDunno | ANI? |
16:52.06 | Ikarus | Automatic Number Identification |
16:52.09 | Ikarus | Caller ID |
16:52.45 | iDunno | ahh |
16:52.47 | file | well |
16:52.47 | drumkilla | twisted[asteria]: i know, he tries that crap with me too |
16:52.53 | file | callerid and ANI are two different things |
16:52.58 | iDunno | right. |
16:53.03 | generalhan | can anyone help me out with my monitor issue ?? i dont understand this at all. |
16:53.13 | iDunno | so what's ANI? it maps it to an entry in a phone book? |
16:53.30 | generalhan | I have my calls being recorded, Monitor(), by ${CALLERIDNUM} but the recordings are being saved as "user" for $CALLERIDNUM anyone have any idea why its not showing the phone number of the caller ? |
16:53.36 | Katty | twisted[asteria]: but what if i call? |
16:53.42 | LostFrog | We heard you the first time, generalhan. |
16:53.47 | Katty | twisted[asteria]: i don't think that has ANI on it |
16:54.08 | Katty | twisted[asteria]: though i suppose it does........you do know /who's/ calling |
16:54.12 | Ikarus | file: ANI and CID are used to mean the same thing here |
16:55.22 | file | ANI and callerid can be two completely different things |
16:55.29 | iDunno | right - OK |
16:55.42 | file | for example |
16:55.43 | iDunno | so what *exactly* is ANI as apposed to CID? |
16:56.01 | file | a US cellphone customer roaming in Canada can have a US callerid, but a Canadian ANI |
16:56.36 | file | now if you bill that off callerid when someone calls a toll-free number in the US, you would be missbilling it |
16:56.45 | file | because they were actually calling from Canada - which usually has a higher rate |
16:56.59 | Ikarus | file: ah, that is implemented differently here |
16:57.49 | iDunno | file: *ahhh* - I see what you mean. right. cool. |
16:57.57 | Ikarus | file: here the company you are roaming with calculates the extra charge and forwards it to your own telco, the rest of the cost stays the same |
16:58.18 | Ikarus | So there is no need for the local telco to billingwise know where the phone is |
17:00.18 | shmaltz | anybody seen the treo 650 with Linux? |
17:00.40 | fugitivo | url? |
17:01.34 | *** join/#asterisk asteriskmonkey (n=phil@69.156.197.242) |
17:02.18 | asteriskmonkey | Can anyone tell me if asterisk supports presence yet? |
17:02.36 | file | well |
17:02.45 | file | it can tell you if somebody's phone is inuse, whatever |
17:02.55 | asteriskmonkey | yes i know using hints and limits |
17:03.18 | asteriskmonkey | but no official presence support atm? |
17:03.29 | file | correct, it doesn't accept it from the phone and that |
17:04.08 | slak- | hey how do i know what 10x priority needs to be my if busy goto vm line |
17:04.15 | slak- | exten => s,4,Dial(SIP/${ARG1},20,t) |
17:04.18 | slak- | exten => s,104,Voicemail(b${ARG1}) |
17:04.18 | slak- | exten => s,105,Hangup |
17:04.22 | slak- | is it 104 |
17:04.22 | slak- | ? |
17:04.47 | paryl | slak: i'm looking for that |
17:04.52 | slak- | thanks |
17:05.44 | paryl | i'm pretty sure it's +101 |
17:05.46 | KriS83 | Isn't it normally prio + 101 |
17:05.48 | paryl | not +100 |
17:05.53 | slak- | ok so 105 |
17:05.58 | paryl | jawohl |
17:06.04 | Qwell | slak-: What version of asterisk? |
17:06.08 | slak- | 1.2.1 |
17:06.19 | Qwell | Then you need to do +1, and check ${DIALSTATUS} |
17:06.28 | slak- | eh |
17:06.31 | slak- | thats that new config |
17:06.33 | slak- | i dont like it |
17:06.55 | Qwell | Then use option j, but it's cheesy and deprecated |
17:07.10 | slak- | exten => s,104,Voicemail(b${ARG1}) |
17:07.10 | slak- | exten => s,105,Hangup |
17:07.12 | slak- | er |
17:07.17 | slak- | this you mean |
17:07.17 | slak- | exten => s,1,Dial(${ARG2},20) ; Ring the interface, 20 seconds maximum |
17:07.18 | slak- | exten => s,2,Goto(s-${DIALSTATUS},1) ; Jump based on status (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER) |
17:07.18 | fugitivo | dial+101 |
17:07.39 | Qwell | slak-: not exactly, but that would work |
17:13.06 | *** join/#asterisk truescot (n=truescot@j242050.upc-j.chello.nl) |
17:14.34 | slak- | hi |
17:14.47 | slak- | can someone take a look at http://pastebin.ca/33675 and tell me if that will work |
17:15.01 | slak- | its my macro and [extensions] lines...about 10 lines |
17:15.06 | *** join/#asterisk toddf (n=toddf@net-66-210-104-104.theshop.net) |
17:15.09 | *** part/#asterisk Dandre (n=Dandre@was59-3-82-236-48-30.fbx.proxad.net) |
17:15.42 | truescot | please can anyone help me, every call on my asterisk box gets dropped within 5 minutes and i get the error PRI got event: HDLC Bad FCS (8) on |
17:15.42 | truescot | Primary D-channel of span 1 |
17:16.04 | Delvar | <slak->: that should work |
17:16.19 | *** join/#asterisk spunz_ (n=spunz@h081217096096.dyn.cm.kabsi.at) |
17:16.21 | truescot | i am using an e1 connection on a te110p card |
17:17.15 | syle | asteriskmonkey: even better case statements, check extensions.ael |
17:17.29 | iDunno | slak-: looks good to me |
17:17.40 | slak- | okay i just have one question |
17:17.44 | *** join/#asterisk adker (n=adker@67-51-239-152.dsl1.glv.ny.frontiernet.net) |
17:17.45 | slak- | extensions 106 is a special case |
17:17.49 | slak- | that cant follow that macro |
17:17.54 | asteriskmonkey | syle: ive not gone to 1.2 yet still on 1.0.9 :) looking forward to it though :D |
17:18.00 | slak- | but the line exten => _1XX,1,Macro(oneline,${EXTEN}) will send it to the macro |
17:18.00 | viperdude | on the cli, if i do a add queue member xxxx to xxxx can I specify the context for the member ie Local/100@mycontext |
17:18.06 | slak- | how can i get around that |
17:18.35 | slak- | if i put 106 above exten => _1XX,1,Macro(oneline,${EXTEN}) will that fix it? |
17:19.42 | *** join/#asterisk tzafrir_laptop (n=tzafrir@local.xorcom.com) |
17:20.45 | *** join/#asterisk EriSan (n=erisan@81-174-42-85.f5.ngi.it) |
17:21.16 | slak- | [extensions] |
17:21.16 | slak- | exten => 106,1,Macro(remote,${NICKOFFICE}) |
17:21.17 | slak- | exten => _1XX,1,Macro(oneline,${EXTEN}) |
17:21.19 | *** join/#asterisk Suckysucky (i=Borgon@70-100-55-164.dsl1.tbr.ga.frontiernet.net) |
17:21.21 | Suckysucky | hello |
17:21.23 | slak- | will 106 in this case never match _1XX |
17:21.28 | slak- | i dont want it to |
17:21.33 | slak- | 106 follows a different Macro |
17:22.02 | Suckysucky | If i got a international number to call and am using a softphone, how would i insert the number to call direct for example +639157435091 .. i tried 011639157435091 but nothing |
17:23.36 | *** join/#asterisk Ariel_ (n=Ariel@adsl-068-157-125-248.sip.mia.bellsouth.net) |
17:24.15 | slak- | everyone took off for lunch of what |
17:26.11 | Delvar | sorry :) |
17:26.24 | Delvar | slak-: asterisk will match 106 before _10XX |
17:26.25 | *** join/#asterisk Dr-Linux (n=Nothing@202.125.141.6) |
17:26.40 | Delvar | asterisk as i understand it sorts them alphabeticly |
17:26.48 | Delvar | and _ comes after numbers :) |
17:27.02 | *** join/#asterisk jtodd (n=jtodd@dsl027-191-178.sfo1.dsl.speakeasy.net) |
17:28.38 | *** join/#asterisk oej_ (n=oej@apollo.webway.se) |
17:29.14 | LostFrog | There is a comment on the wiki that it isn't determinate which will match first.. |
17:30.00 | Ariel_ | Just wondering looks like the user list is down. I have not gotten any emails in over an hour. |
17:30.01 | LostFrog | I don't know if that changed for 1.2 |
17:32.25 | nextime | can i set multiple sip friend with same ip? |
17:32.37 | oej_ | No, one login per peer |
17:32.55 | Delvar | LostFrog: i have lots of setups and i can assure you it will amtch 106 before _1XX |
17:32.59 | Delvar | 100% sure :) |
17:33.06 | loud | they will flap. |
17:33.20 | nextime | oej_ are you sure? this is a bad notice for me... |
17:33.21 | loud | although both will be able to call though. |
17:33.54 | LostFrog | Delvar: works for me.. but whoever wrote the wiki entry probably knows what he/she was talking about. |
17:34.02 | oej_ | nextime: If you're talking about how many phones you can connect to one account in Asterisk, as a sip peer, I am very sure. |
17:34.25 | Delvar | maybe they have a dodgy version of asterisk... :) |
17:35.04 | oej_ | The wiki is no longer, I am sad to say, to be fully trusted at all times. |
17:35.07 | nextime | oej_ : they aren't phones, but is the same thing, i'm proxyng h323 to sip with yate, and i need separated login to recognize different h323 users |
17:35.24 | nextime | but as you say this is pretty impossible |
17:36.03 | oej_ | I don't know how yate translates caller IDs and such, maybe you can separate on something else in the dial plan |
17:37.10 | nextime | oej_: yes i can recognize from clid or maybe from an addictional extension prepended to the called one, but isn't good for accounting and billing... |
17:38.04 | oej_ | Then use asterisk all the way without involving yate |
17:38.18 | *** join/#asterisk hugov6 (n=foo@p54AD54EB.dip.t-dialin.net) |
17:38.24 | shmaltz | linux on treo 650: |
17:38.25 | shmaltz | http://www.engadget.com/entry/1234000497072377/ |
17:38.29 | hugov6 | hiho |
17:38.38 | *** join/#asterisk zgor (n=zgor@61.Red-80-36-3.staticIP.rima-tde.net) |
17:38.41 | zgor | hi :) |
17:39.10 | *** join/#asterisk heison (n=heison@ns.somanetworks.com) |
17:39.26 | nextime | oej_ h323 support in * is bad, no gk at all and many problems with some h323 remote gateways |
17:39.33 | hugov6 | is there another option to set the callerid for sip-phones than callerid and fromuser in sip.conf? |
17:40.03 | Delvar | setCIDName |
17:40.08 | Delvar | setCIDNum |
17:40.10 | Delvar | in dailplan |
17:41.16 | hugov6 | Delvar: thx. 'll have a look at it. |
17:41.36 | Smace | hello |
17:41.51 | [TK]D-Fender | Delvar : the old CID connamds are now depreciated. |
17:41.58 | Delvar | yeah |
17:42.13 | [TK]D-Fender | I just finished adapting my setup for the DB stuff... |
17:42.14 | Delvar | SetVar(callerid(name)=bla) now isnt it? |
17:42.24 | shmaltz | hugov6, yes, RPID, or Set(CALLERID(datatype)) |
17:42.35 | drumkilla | yeah, don't use SetVar :) |
17:42.40 | Delvar | hehe :) |
17:42.45 | drumkilla | use Set ... SetVar has been completely removed from the trunk |
17:42.52 | Delvar | been a while since i used any of that |
17:43.12 | Smace | I'd like to ask for suggestion for one project. We've one large network here, over 50km of wireless networks and 10km of wired networks. We would like to use VOIP here, the point is i'm not sure about which hardware would fit our needs. |
17:44.19 | [TK]D-Fender | Smace : Depends on how many end-points, what kind of hardware (wifi-phones needed?), PSTN termination is any, etc.... |
17:44.22 | shmaltz | Smace, how many phones? |
17:44.38 | Smace | I was thinking about 50 for start... |
17:44.38 | [TK]D-Fender | Who are the users, what kind of phones, and any PSTN involved? |
17:45.11 | Smace | first I need to test it using 3 phones. |
17:45.52 | Smace | This network covers 3 cities. I want to use this network we have to make calls to IP-IP. |
17:46.11 | *** join/#asterisk wshs (i=screwy@24-52-105-100.bflony.adelphia.net) |
17:46.33 | shmaltz | Smace, what is the maximum of users you ever want this system to support? |
17:47.12 | Smace | at least 3 for start. but the right number would be between 50 and 100 users for 2006. |
17:47.31 | jimmy_deanPB | hey asteriskmonkey, upgrading to asterisk 1.2.1 pretty much got rid of all echo. we've had one call with echo in it but otherwise no echo. |
17:48.12 | *** join/#asterisk zgor (n=zgor@61.Red-80-36-3.staticIP.rima-tde.net) |
17:48.15 | zgor | hi people :) |
17:48.25 | LostFrog | 1.2.1 is out? |
17:48.29 | LostFrog | Duh.. |
17:48.32 | shmaltz | Smace, then go with a Dual Xeon, and ~2GB RAM, or you could use Dual Core 64bit AMD |
17:48.50 | Smace | shmaltz, but it's not at the same time. |
17:48.54 | shmaltz | Smace, that is if you will not use *any* recording or heavy AGI |
17:49.06 | shmaltz | Smace, what are you trying to say? |
17:49.09 | *** join/#asterisk junbug (n=junya@c-66-176-211-109.hsd1.fl.comcast.net) |
17:49.33 | zgor | is there any way to make SIP extension gain audio ? Im using Sipura 841 with updated firmware and set gain level on web interface, but still being very low level. Any idea ? |
17:49.40 | Smace | shmaltz, I've 50 points in our network I'd like to have ip-phone there. |
17:49.42 | shmaltz | Smace, what PSTN connection you going to have? |
17:50.02 | shmaltz | ~pstn |
17:50.03 | jbot | rumour has it, pstn is Pubic Switched Telephone Network, or "please stop the nonsense" |
17:50.16 | shmaltz | Smace, how you connecting to the telephone company? |
17:50.20 | Smace | shmaltz, I'm getting started do you suggest me any site? |
17:50.32 | Smace | shmaltz, we have some lines here. We can use that. |
17:50.35 | shmaltz | ~docs |
17:50.37 | jbot | it has been said that docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com |
17:50.56 | Suckysucky | If i got a international number to call and am using a softphone, how would i insert the number to call direct for example +639157435091 .. i tried 011639157435091 but nothing |
17:51.15 | shmaltz | Smace, see you in around 3 days, I'm sure you will be more confused. :P |
17:51.22 | Smace | lol |
17:51.36 | shmaltz | Suckysucky, ask your provider what they expect |
17:52.09 | Smace | shmaltz, is there any comercial ip-phone that make calls from ip to ip ? something like one access point. |
17:52.26 | Dandan | how do you force first time user to set up his vm? |
17:52.36 | Dandan | setting password to his vm number doesn't do it |
17:53.00 | shmaltz | Smace, Cisco, Polycom, Grandstream, Sipura, Snom, Zytux, and RTFM on the wiki you should be able to find more |
17:53.16 | Suckysucky | shmaltz: they say do 011 with area code |
17:53.24 | LostFrog | RTFW? |
17:53.25 | Smace | shmaltz, is it configured like one access point? |
17:53.40 | rob0 | hey, off topic but someone here would know ... I have telephone headsets with the itty bitty plug. Is there an adapter/splitter which would allow that to be used for a computer sound card speaker/mic jacks? |
17:53.50 | Smace | shmaltz, I can set up it to use one external voip provider and also make ip2ip calls? |
17:53.59 | shmaltz | Danda, you will have to write your own AGI for it, as asterisk DP cannot do that, since asterisk doesnt have an app that will drop you just to the change password in app_voicemail |
17:54.07 | rob0 | (and what would I ask for?) |
17:54.13 | shmaltz | Smace, where you located? |
17:54.57 | shmaltz | rob0, visit your local radio shack |
17:55.00 | Smace | rio de janeiro brazil |
17:55.44 | rob0 | :) |
17:55.45 | Dandan | for new years :) |
17:56.17 | shmaltz | Smace, you paying fare? |
17:57.26 | *** join/#asterisk mutilator (n=animenod@65.111.201.79) |
17:59.22 | LostFrog | Just for the record, the only prime triangular numbers are 1 and 3. |
17:59.50 | Dandan | how do you force first time user to set up his vm? |
17:59.56 | Dandan | pls? |
18:00.55 | Smace | shmaltz, we make to much calls between these 3 cities. as we have one wide wireless network that covers it, i thought i could avoid paying for these calls. |
18:01.03 | Smace | using our own private network |
18:01.36 | shmaltz | Dandan, you will have to write your own AGI for it, as asterisk DP cannot do that, since asterisk doesnt have an app that will drop you just to the change password in app_voicemail |
18:01.55 | Dandan | hm, i thought i had that a long time ago... |
18:02.00 | Dandan | but thx shmaltz |
18:02.18 | shmaltz | Smace, you could, but the most CPU power will be needed when converting between PSTN and VoIP, thats why I asked what I asked |
18:02.20 | *** join/#asterisk hugov6 (n=foo@p54AD6B94.dip.t-dialin.net) |
18:02.25 | hugov6 | hmz. |
18:03.47 | Smace | shmaltz, i believe CPU is not gonna be the problem, the problem is gonna me to do it :) |
18:04.03 | shmaltz | Smace, I told you, just pay the fare :P |
18:04.21 | Smace | how much is the fare? |
18:04.23 | Smace | hehe |
18:05.16 | Smace | shmaltz, where are you from? |
18:05.20 | shmaltz | NJ |
18:05.52 | Smace | New Jersey? |
18:06.12 | hugov6 | hmmz. anyone an idea on how to set outgoing callerid for sip-phone and mISDN? |
18:06.50 | shmaltz | ~nj |
18:06.52 | jbot | from memory, nj is home to the sopranos. Fogedaboudit! |
18:07.13 | shmaltz | Smace :) |
18:10.24 | Dandan | shmaltz: forcename=yes and forcegreetings=yes does that |
18:10.32 | Dandan | ~ct |
18:10.39 | shmaltz | Dandan, what? |
18:10.49 | shmaltz | where? in voicemail.conf? |
18:10.54 | Dandan | forces user to record name/greetings |
18:10.55 | Dandan | yes |
18:12.30 | *** join/#asterisk ahmuck_jr (n=chatzill@p115n4.ruraltel.net) |
18:12.42 | shmaltz | Dandan, but nothing for the password |
18:12.55 | asteriskmonkey | is there a protocol you can tie in to some voip phones that will present it with an address book? |
18:12.57 | ahmuck_jr | i don't quite understand what the hardware is for ? |
18:13.02 | Dandan | yes, if your password = your ext. it will ask you to change pass... |
18:13.22 | Dandan | shmaltz: i used in in 1.0.6 afair :) |
18:13.25 | shmaltz | Dandan, is that new as well? |
18:13.27 | Dandan | in my testbed :) |
18:13.38 | Dandan | shmaltz: no, but it is not documented |
18:13.41 | shmaltz | Dandan, then why did you ask? |
18:13.53 | *** join/#asterisk genmud (n=genmud@ip68-98-82-206.ph.ph.cox.net) |
18:13.59 | Dandan | cause I didn't know where it was |
18:14.02 | Dandan | :)))) |
18:14.11 | Dandan | and I have no conf files to grep |
18:14.21 | Dandan | sorry to bother |
18:15.16 | ahmuck_jr | i don't quite understand what the hardware is for ? |
18:16.58 | shmaltz | asteriskmonkey, thridlane makes an MS Outlook plug in that allows you to click on any phone# in Outlook to dial it thru * |
18:17.25 | asteriskmonkey | smaltz: is it free |
18:17.28 | Dandan | or is it *THIRDLANE? |
18:17.57 | Dandan | shmaltz: how about a pop-up caller id for your pc? |
18:18.00 | shmaltz | asteriskmonkey, also, some phones (Polycom, and Cisco), will allow you to create an xml file that will show up on the phone, no it's not free, and yes its thirdlane |
18:18.08 | shmaltz | Dandan, use FOP |
18:18.14 | Dandan | FOP? |
18:18.17 | tainted- | anyone have experience doing voip in mexico? |
18:18.18 | asteriskmonkey | i know xten makes a softclient that links into outlook and does the same.. i was looking for something that moreso actuaclly populates a phonebook on a sipphone (hardphone) |
18:19.07 | nextime | is the open source implementation of g729 supporting g729r8? |
18:19.13 | shmaltz | asteriskmonkey, thats where the xml files for the Polycom Cisco comes in |
18:19.20 | Dandan | ~fop |
18:19.21 | jbot | An XSL formatter written in Java that outputs PDF. URL: http://www.jtauber.com/fop/, or the Flash Operator Panel |
18:19.22 | LostFrog | I think it just supports 16. |
18:19.25 | nextime | or only g729a like the others? |
18:19.41 | Dandan | a |
18:19.45 | Dandan | that thing :) |
18:20.28 | nextime | Dandan : "a" was for me? |
18:21.06 | asteriskmonkey | shmaltz: tell me more :) |
18:21.30 | asteriskmonkey | im also looking for some hacks info trick for a wip-5000 if anyone has one of those aka hitachi ip 5000 |
18:21.35 | shmaltz | asteriskmonkey, more about what? |
18:22.10 | asteriskmonkey | the xml extension stuff is there documents on the polycom web? |
18:22.27 | shmaltz | asteriskmonkey, yes the docs are in the admin manual |
18:22.32 | shmaltz | same goes for the cisco |
18:23.18 | shmaltz | I think the cisco uses http for that, which makes it very easy to be dynamic, while the polycom only supports static from ftp/tftp, or with the newer bootroms http/https |
18:24.40 | asteriskmonkey | so is there a common format for those phones? |
18:24.51 | shmaltz | asteriskmonkey, now RTFM |
18:25.15 | Suckysucky | Can anyone direct me on how tom ake an international call? am using voicepuse and my provider said type 011 + country code.. the number am dialing is +639157435091 |
18:25.41 | asteriskmonkey | shmaltz: thanks ... but is the syntax same for polycom and cisco ? or is it a completely different xml structure? |
18:26.04 | *** join/#asterisk sysdebug (n=sysdebug@200.163.193.247) |
18:26.43 | shmaltz | Suckysucky, so you trying to dial the philipines, first ask voicepulse if they have service to the philipines |
18:26.47 | bsdfreak | heh |
18:26.55 | Suckysucky | shmaltz: they do |
18:26.58 | shmaltz | asteriskmonkey, I think it's completely different |
18:27.11 | shmaltz | Suckysucky, how do you know? |
18:29.22 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
18:34.20 | ahmuck_jr | how do the cards work ? |
18:39.04 | [Lamer] | I'm running asterisk with some fxo cards to do the auto-dialout |
18:39.35 | [Lamer] | how do I know when the other side picks up the phone? |
18:40.42 | shmaltz | http://www.local6.com/news/5527928/detail.html |
18:41.06 | shmaltz | [Lamer], if you use a PRI, you would know because of the answer state |
18:41.10 | *** join/#asterisk SplasPood (i=jwb@ludicrous.paravolve.net) |
18:42.44 | *** join/#asterisk marc32422 (n=marc3234@206-248-134-12.dsl.teksavvy.com) |
18:43.32 | *** join/#asterisk jahani (n=BaT@adsl196-219-27-206-196.adsl196-1.iam.net.ma) |
18:43.49 | *** join/#asterisk robl^ (n=robl@dsl093-025-118.hou1.dsl.speakeasy.net) |
18:43.50 | [Lamer] | shmaltz: so, there's no way to recognize it using POTS? |
18:44.09 | shmaltz | http://www.firstcoastnews.com/news/topstories/news-article.aspx?storyid=48816 |
18:44.59 | shmaltz | [Lamer], no, not with POTS, since its answered when it goes off hook, remember POTS uses inband signalling |
18:46.05 | *** join/#asterisk kuonSama (n=kuon@alragore.goyman.com) |
18:46.06 | shmaltz | [Lamer], most SIP porviders, will also give you the answer state, but not all of them |
18:46.06 | kuonSama | hello |
18:47.11 | bsdfreak | heh |
18:47.14 | [Lamer] | shmaltz: I see. I tried with the D41E from dialogic and it seems to recognize those though |
18:48.18 | [Lamer] | shmaltz: but only when I called to non-mobile phone users |
18:49.16 | LostFrog | Damn.. you can't even buy your friend a hooker in class any more. |
18:50.59 | robl^ | LostFrog: buy him a gift certificate |
18:52.08 | LostFrog | robl^: Did you read the site that shmaltz posted? |
18:52.38 | robl^ | no.. I am at work... |
18:52.38 | *** join/#asterisk sysdebug (n=sysdebug@200.163.193.247) |
18:52.41 | LostFrog | It's not dirty. |
18:53.07 | LostFrog | A 18 year old kid paid a 16 year old girl to go down on another minor in class. |
18:53.13 | robl^ | they have LOTS of "filters" preventing web browsing outisde a few select sites here. |
18:53.20 | rob0 | Some content filters would block the phrase in the title. |
18:53.30 | rob0 | namely, Dansguardian. |
18:53.55 | robl^ | but then again.. since I have remote access to my home computer.. they don't block outbound ssh... :) |
18:54.10 | LostFrog | secure-tunnel.com |
18:54.13 | LostFrog | :) |
18:54.21 | rob0 | "Not indicative of what is happening in schools" .. LOL |
18:54.54 | LostFrog | Yeah.. we at least used to go into the bathrooms or supply rooms. :) |
18:55.09 | *** join/#asterisk r0d3nt (i=nobody@wsip-24-234-241-145.lv.lv.cox.net) |
18:55.25 | *** join/#asterisk SpaceBass (n=sp@static-71-251-230-2.rcmdva.fios.verizon.net) |
18:55.30 | SpaceBass | hey folks |
18:55.55 | LostFrog | Hey, SpaceBass. |
18:56.07 | SpaceBass | dire straits.... trying to order a SSL cert and it requires an automated system to call me... I have to press # which asterisk thinks is me iniating a transfer |
18:56.11 | SpaceBass | stupid asterisk@home |
18:56.20 | LostFrog | Anyone know of a *good* windoze hylafax client? |
18:56.27 | SpaceBass | with out editing my entire dial plan, is there a quick way around that |
18:56.33 | SpaceBass | ie disable # transfers |
18:57.11 | sivana | SpaceBass: do you know the macro that rings your phone? |
18:57.28 | sivana | SpaceBass: if so, you can remove the t or T from the dial() |
18:57.40 | *** join/#asterisk VxJasonxV (n=jason@unaffiliated/VxJasonxV) |
18:57.42 | SpaceBass | sivana thats what I'm trying to avoid editing |
18:57.56 | SpaceBass | damn... times like this that A@H is a real pain |
18:58.09 | sivana | I don't use A@H.. so I dunno |
18:58.17 | sivana | I pretty to be able to do things :) |
18:58.19 | sivana | I prefer |
18:58.44 | sivana | you can search for dial( string |
18:59.04 | robl^ | A@H is useful for a quick base install.. then forget AMP and do it all by hand with custom config files .. :) |
18:59.22 | Dandan | I used A@H to learn and study config files :) |
18:59.22 | LostFrog | robl^: That's what I did. |
18:59.23 | SpaceBass | robl^ thats pretty much my philosophy |
18:59.40 | SpaceBass | but some of the scripts are so embedded that its hard to manually configure |
18:59.50 | [TK]D-Fender | All * GUI = shit-on-a-stick. Thats my story... AND I'M STICKING TO IT> |
18:59.52 | *** join/#asterisk tris (i=tristan@camel.ethereal.net) |
18:59.52 | SpaceBass | found it |
19:00.01 | *** join/#asterisk rezEdit (n=rezEdit@zapdos.omnigroup.com) |
19:00.21 | SpaceBass | nice... /etc/asterisk/extensions_additional.conf |
19:00.30 | SpaceBass | had a dial string var option |
19:00.38 | SpaceBass | wooo pannic over |
19:00.41 | SpaceBass | ssl cert on its way |
19:00.43 | robl^ | SpaceBass: I delete ALL the config files... then install samples from asterisk source and edit away |
19:00.59 | file | [TK]D-Fender: that's a great story |
19:01.09 | sivana | heh |
19:03.07 | [TK]D-Fender | I use ScopServ at work, and have helped people get away from A@H & AMP....... |
19:03.28 | SkramX | Hi all. |
19:05.45 | SpaceBass | I hear a lot of knocks agains A@H but frankly I am at home and it works great |
19:08.49 | [TK]D-Fender | It does work, but ... ICK. the point of * is control, and A@H turns * into another cookie-cutter PBX ableit free. |
19:09.09 | [TK]D-Fender | And makes nasty to maintain and volatile code. |
19:09.35 | file | [TK]D-Fender: !!! |
19:09.35 | *** join/#asterisk asterisk99 (n=dunoowhy@modemcable169.194-130-66.mc.videotron.ca) |
19:09.41 | asteriskmonkey | can anyone tell me if there is a difference between the 406 and 411 quad t1 card from digium other than the voltagwe? |
19:10.09 | robl^ | SpaceBass: A@H is really good for experimentation or a VERY VERY basic setup. Its a pain to upgrade. It limits flexibility. |
19:10.26 | asterisk99 | anyone here good at zaptel boot-up (script) procedures (gentoo specifically) |
19:10.39 | [TK]D-Fender | file: !!! |
19:10.40 | asterisk99 | (that was a question) |
19:10.45 | file | [TK]D-Fender: ...hi |
19:14.40 | jarrod | where can i get latest polycom ip500 sip firmware |
19:15.12 | asteriskmonkey | the polycom website :P |
19:15.17 | Seldon1975 | http://www.freedomphones.net/polycom/files/ |
19:15.22 | Seldon1975 | get them from there |
19:15.35 | Seldon1975 | I believe that is ok for ip500 |
19:15.44 | Seldon1975 | better check |
19:16.36 | Dandan | can someone call me over fwd? |
19:16.58 | rezEdit | asterisk99: I wouldn't say 'good' but what's the issue...? |
19:16.59 | *** join/#asterisk hamish (n=hamish@196-28-87-71.wdsl.co.za) |
19:18.26 | asterisk99 | rezEdit: after boot, asterisk fails to load... if I manually run ztcfg -vv and the asterisk, it starts up fine How do I figure out if the zaptel startup script is executing OK or getting an error? |
19:19.08 | rezEdit | well ztcfg does not load zaptel |
19:19.29 | hamish | Anyone using a BRI hfc card on ubuntu breezy? |
19:19.30 | asteriskmonkey | is there a prituner like fxotune? |
19:20.04 | rezEdit | it only parses the config. As long as ztcfg funs fine you should try running the zaptel script |
19:20.06 | asterisk99 | rezEdit: I know... but there's a startup script(s) somewhere that loads ztcfg and asterisk |
19:20.25 | rezEdit | yes... /etc/rc.d/init.d/zaptel |
19:20.26 | truescot | asterisk99 it tells you whether it loaded or failed in the boot up when its loading all the driver |
19:20.53 | *** join/#asterisk nagl (n=nagl@213.235.241.6) |
19:20.54 | asterisk99 | rezEdit: does ztcfg have to execute every time after boot? |
19:21.11 | rezEdit | the zaptel script runs it |
19:21.49 | rezEdit | but it doesn't have to.... I worked aorund some bugs in an older version I was using by altering the script to ingore the failure it was getting from ztcfg |
19:22.06 | *** join/#asterisk Sedorox (i=brandon@smartserv/cna/Sedorox) |
19:22.27 | rezEdit | you should also be able to run the zaptel script using /sbin/service zaptel start |
19:23.26 | asteriskmonkey | any digiums people around? |
19:23.40 | mog_work | nope |
19:23.44 | mog_work | we are hiding |
19:23.46 | mog_work | errr working |
19:24.22 | asteriskmonkey | mog_work: do you know if there is anything to autotune a pri like fxodoes on the pots cards |
19:24.23 | Seldon1975 | does /etc/rc.d/rc.local get run after boot? |
19:24.38 | mog_work | you cant do that |
19:24.41 | mog_work | as its impossible |
19:24.44 | mog_work | sorries |
19:25.00 | rezEdit | asterisk99: so does the zaptel script run OK for you? |
19:25.13 | asteriskmonkey | mog_work: :( is there any tools available for aiding in setting right tx rx gain levels? |
19:25.46 | mog_work | ztmonitor |
19:25.56 | mog_work | is levels wrong |
19:25.58 | mog_work | or whats going on |
19:26.13 | asterisk99 | rezEdit: (I'm using gentoo btw) You way ztcfg does not need to run before asterisk? |
19:26.18 | asteriskmonkey | mog_work: when setting up the te110p cards most people have mad echo until they get there levels spot on |
19:27.07 | mog_work | bah |
19:27.09 | truescot | i gotta say i goy a te110p card and have no problems with echos and tx and rxgain are set to default |
19:27.18 | mog_work | using tx and rx are the worst way to attack echo |
19:27.21 | mog_work | they are last resort |
19:27.27 | rezEdit | asterisk99: the zaptel script needs to run before asterisk, and the zaptel script runs ztcfg |
19:27.33 | mog_work | i dont know why everyone hits em first |
19:27.52 | asteriskmonkey | mog_work: running asterisk 1.0.9 with 1.2 zaptel.. and kb1 echo can still have minor echo issuse occasionally |
19:28.19 | rezEdit | asterisk99: what I was saying is that I modified the zaptel script to NOT run ztcfg because there was a bug that was causing errors, and even though everything should have loaded fine, because of the bug it did not. |
19:28.20 | mog_work | hrmm you bump up the echocanceling to 256? |
19:28.29 | mog_work | or try mg2? |
19:29.09 | asterisk99 | rezEdit: I found /etc/init.d/zaptel and in it is a command [ -f /sbin/ztcfg ] !! exit 0 |
19:29.18 | rezEdit | asterisk99: The szaptel script basically just runs modprobe zaptel and then modprobe again for each additional driver that needs to load. |
19:29.46 | asterisk99 | rezEdit: Thos are not !!, they are really the long-stroke (my keyboard has been remapped on me) |
19:29.56 | rezEdit | asterisk99: It sounds like ztcfg runs fine for you, so don't worry about that |
19:29.58 | asteriskmonkey | mog_work : sec ill pastebin you my zaptel its often far end that hears echo after sometime |
19:30.28 | asteriskmonkey | echos alot on iaxys over 80ms away |
19:30.33 | mog_work | okay |
19:30.34 | asterisk99 | rezEdit: it runs --- but only manually It looks like it's not running on boot |
19:30.50 | rezEdit | asterisk99: if anything is failing for you, it seems it's the zaptel script itself. |
19:31.02 | asterisk99 | rezEdit: how can I tell? Is there a log somewhere? |
19:31.07 | brad_mssw | think iaxy's have a tendency to echo ... the single iaxy we have is the only device that echos ... |
19:31.20 | rezEdit | asterisk99: so do you see a message at startup that zaptel is loading? |
19:31.37 | asteriskmonkey | mog_work: http://pastebin.ca/33692 |
19:31.38 | rezEdit | asterisk99: Maybe in boot.log? or messages.log? |
19:31.41 | brad_mssw | all our sip phones, and zap (TDM400P FXS connected) phones are fine :/ |
19:31.49 | asterisk99 | machine is 600 miles away... I use Putty for the console |
19:31.56 | asteriskmonkey | kb1 is an improvment of mg2 is it not? |
19:32.07 | mog_work | bah iaxys dont echo... |
19:32.15 | mog_work | mg2 is improvement of kb1 |
19:32.17 | truescot | mog_work can you tell me if the correct line for my zaptel.conf should be, 1,0,0,ccs,hdb3,crc4 or 1,1,0,ccd,hdb3,crc4 i have a te110p with an e1 line from the telco plugged in |
19:32.32 | mog_work | you want the 1,1 as you are taking timing from telco |
19:32.42 | truescot | ok cool |
19:33.16 | mog_work | i would get rid of gain asteriskmonkey and set echocancel=256 |
19:33.36 | truescot | i have been bangin my head off the walls at work trying to find out the problem and only just realised that it should prob be 1,1 but now not in front of the machine to check so thanks for putting my mind at rest :) |
19:33.45 | mog_work | no prob |
19:34.44 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
19:34.53 | asteriskmonkey | mog_work: had that before and it slapped echo no matter what really bad i think :P its a production box so not much time to play this early in day :) |
19:35.06 | asteriskmonkey | should i just upgrade the whole thing to 1.2.1? |
19:36.04 | asterisk99 | rezEdit: hmmmmmm I am looking at /var/log/messages there are messages ... ''rcscripts: /sbin/rc-update /etc/init.d/ztcfg not found; aborting'' That must be it!!!! |
19:36.05 | truescot | btw i changed echocancel in zapata.conf to echocancel=no and my line sounds much better for it |
19:36.33 | asteriskmonkey | truescot: some echo cans must be in place from your provider then.. |
19:36.34 | mog_work | probably but it wont change to much |
19:36.47 | asterisk99 | rezEdit: typo above ... that was rc-scripts |
19:36.58 | asteriskmonkey | most of my clients use the iaxys |
19:37.09 | asteriskmonkey | i see there is some new updates for those things in the 1.2.1 |
19:37.24 | rezEdit | asterisk99: yes, probably. you need to either specify the fill path in the script or add the path to your $PATH env var |
19:38.01 | asterisk99 | rezEdit: that would be in /etc/init.d/zaptel ?? |
19:38.11 | Seldon1975 | does anyone know a Windows program that plays Asterisk gsm sound files. I've been looking for this myself but come up dry |
19:38.23 | asteriskmonkey | just get the codec |
19:38.44 | mog_work | quick time will |
19:38.52 | *** join/#asterisk _-Jon-_ (i=jon@CPE00112f6dfbee-CM00111a232a80.cpe.net.cable.rogers.com) |
19:38.55 | mog_work | and you can do wav47 or whatever it is |
19:39.00 | mog_work | that is gsm for windows like codec |
19:39.14 | _-Jon-_ | hey can someone help me with a nat problem? |
19:39.16 | asterisk99 | rezEdit: the line has a full path... /sbin/ztcfg (that fiule exists) |
19:39.19 | rezEdit | asterisk99: uh wait... I am not sure what rc-scripts is doing there. |
19:39.32 | _-Jon-_ | Our asterisk server just seems the client (which is behidn NAT) as coming from it's internal IP |
19:39.36 | asterisk99 | rezEdit: that makes two of us :) |
19:40.01 | rezEdit | asterisk99: yeah... one sec |
19:40.11 | truescot | does anyone know of a good dect interface for asterisk that allows roaming? |
19:40.13 | asterisk99 | rezEdit: 'spose I edit it to /usr/src/zaptel/ztcfg ? |
19:40.33 | brad_mssw | _-Jon-_: did you set nat=yes in sip.conf ? |
19:40.34 | rezEdit | asterisk99: no, that won't do anything. |
19:40.54 | _-Jon-_ | brad_massw, I sure did |
19:40.55 | brad_mssw | _-Jon-_: also, if your SIP device supports STUN, it may help as well |
19:41.01 | Seldon1975 | asteriskmonkey: 'just get the codec' can you tell me where from? |
19:41.44 | _-Jon-_ | brad_mssw, one step ahead of you :) Tried testing out the client to my personal Asterisk server at home and it worked fine, so it's got to be a problem with the server |
19:42.03 | _-Jon-_ | The only difference that I can tell is the one that isn't working is the stable build, and the one I'm running is the latest head |
19:42.58 | *** join/#asterisk numbshot (n=numbshot@dsl-201-133-74-110.prod-infinitum.com.mx) |
19:43.22 | rezEdit | asterisk99: what happens when you run /etc/init.d/zaptel as root? |
19:43.25 | *** join/#asterisk paryl (n=paryl@216-201-177-82.res.logixcom.net) |
19:43.35 | *** join/#asterisk sysdebug (n=sysdebug@200.163.193.247) |
19:43.49 | asterisk99 | rezEdit: Permission denied |
19:44.20 | rezEdit | asterisk99: uh!?! running it as root? is the file executable? |
19:44.31 | fugitivo | of course not |
19:44.37 | asterisk99 | rezEdit: it's -rw-r--r-- |
19:44.40 | fugitivo | ;) |
19:44.48 | asterisk99 | rezEdit: chmod 777 it? |
19:44.50 | fugitivo | asterisk99: chmod +x it |
19:45.13 | rezEdit | asterisk99: yeah what fugitivo said! |
19:46.15 | asterisk99 | rezEdit: got Line 23 /etc/init.d/functions: No such file or directory |
19:46.36 | *** join/#asterisk sysdebug (n=sysdebug@200.163.193.247) |
19:47.44 | rob0 | 777: the executable privileges of the Beast |
19:48.21 | asterisk99 | rezEdit: the script is plugged ''system=redhat'' and line 22-24 read is [ $system = 'redhat'] ; then ./initdir functions fi |
19:48.23 | rob0 | asterisk99: what OS? |
19:48.39 | asterisk99 | rob0: gentoo |
19:48.57 | _-Jon-_ | So no other ideas? I set localhost=192.168.2.0/24, set externip, set nat=yes and canreinvite=no, but still no go.. reports something about max retries and then @192.168.0.99 |
19:50.02 | rob0 | did you "emerge asterisk"? |
19:50.17 | paryl | i have a channel bank, and a fax machine with two lines suddenly stopped making outgoing calls. every other machine is fine, and this machine will take incoming calls. outgoing gets congestion tone immediately |
19:50.24 | fugitivo | asterisk99: what's the problem? |
19:50.34 | *** join/#asterisk m160858 (n=jsaenz@200.89.12.46) |
19:50.55 | asterisk99 | fugitivo: looks like ztcfg fails to execute on boot |
19:51.01 | brad_mssw | _-Jon-_: hope you mean localnet not localhost |
19:51.08 | rob0 | fugitivo: asterisk99 is trying to run a RedHat-style init script in Gentoo |
19:51.10 | brad_mssw | _-Jon-_: plus, did you set externip ?? |
19:51.19 | fugitivo | asterisk99: did you emerge asterisk or manually install? |
19:51.30 | rob0 | and there is no /etc/init.d/functions file to source |
19:51.39 | rob0 | that is what I asked too |
19:51.42 | Ikarus | rob0: try selling him a gentoo training or funrolloops |
19:51.56 | rezEdit | asterisk99: yeah, I am using a Red Hat derivitave here (Yellow Dog) so I will not be much help. |
19:51.58 | m160858 | hi everyone |
19:52.09 | fugitivo | asterisk99: if you emerged asterisk, gentoo installs it's own init scripts |
19:52.10 | _-Jon-_ | brad_mssw, err yes, localnet, my mistake. and yes, externip is set. I just dont' get it.. when I do 'sip show peers' it shows the external IP of the client, but when the errno max retries error comes up, it shows it's internal |
19:52.30 | asterisk99 | fugitivo: I tried emerge at first ... then later I did the CVS install |
19:52.49 | m160858 | I need that you recommend a adaptor to me |
19:52.50 | fugitivo | asterisk99: ok, then you should use the right init scripts, not redhat init scripts |
19:52.59 | brad_mssw | _-Jon-_: and you have host=dynamic and nat=yes in sip.conf for your device .... |
19:53.13 | m160858 | which adapter is better than the SPA ? |
19:53.24 | asterisk99 | fugitivo: ok... sounds like a game-plan... where can I download such a beast? |
19:53.28 | brad_mssw | _-Jon-_: that should really be all there is to it ... though I'm running asterisk 1.0.10 ... hopefully migrate to 1.2 this weekend |
19:53.30 | fugitivo | asterisk99: check your contrib/init.d/rc.gentoo.asterisk script in asterisk sources |
19:53.31 | JonR800 | _-Jon-_: set nat to yes on the client. |
19:54.21 | _-Jon-_ | JonR800, that strange thing is, when I connect to a different Asterisk server the same client works. So I'm think it's an issue on the server side, BUT why would X-Lite report it's internal IP int he first place |
19:54.39 | asterisk99 | fugitivo: can you give me the full path to that please? |
19:54.44 | JonR800 | because nat is set to no, so it thinks it's registering locally. |
19:54.53 | fugitivo | asterisk99: no, because i don't know where are your asterisk sources |
19:55.04 | slak- | hey |
19:55.05 | Seldon1975 | hey guys |
19:55.07 | fugitivo | asterisk99: for example /usr/src/asterisk/contrib/init.d/rc.gentoo.asterisk |
19:55.15 | Seldon1975 | i think someone is hacking our switch |
19:55.17 | JonR800 | the other asterisk servers may have ser or some other proxy?.. something to do the footwork themselves and fix up broken clients. |
19:55.20 | Seldon1975 | can someone have a look at this: http://pastebin.com/464161 |
19:55.20 | slak- | in exten => 106,1,Macro(remote,${NICKOFFICE}) |
19:55.21 | slak- | exten => _1XX,1,Macro(oneline,${EXTEN}) |
19:55.25 | slak- | which will match first |
19:55.26 | slak- | ? |
19:55.29 | slak- | the 106? |
19:55.34 | Seldon1975 | does it look like an incoming call bridged to the outside world? |
19:55.36 | Seldon1975 | http://pastebin.com/464161 |
19:55.41 | *** join/#asterisk genmud (n=genmud@ip68-98-82-206.ph.ph.cox.net) |
19:55.54 | JonR800 | _-Jon-_: all else fails do a sip debug and get an idea of what's going on. |
19:56.53 | m160858 | hello? |
19:57.32 | *** join/#asterisk sshadow (n=sshadow@213-84-101-107.adsl.xs4all.nl) |
19:57.50 | m160858 | which adapter is better than the SPA ? |
19:57.53 | asterisk99 | fugitivo: I found that... it's pretty tiny... 3 lines for start() { ebegin "Starting Asterisk" /usr/sbin/asterisk eend "Failed to start Asterisk" |
19:58.11 | fugitivo | asterisk99: what else do you need? |
19:58.28 | asterisk99 | fugitivo: there's no line for ztcfg |
19:59.08 | fugitivo | you don't need that |
19:59.32 | slak- | heyyy |
19:59.32 | fugitivo | tell modprobe to load the module and then /sbin/ztcfg |
19:59.37 | slak- | come on people |
19:59.38 | asterisk99 | fugitivo: does ztcfg not need to run always before asterisk? |
19:59.38 | fugitivo | for example |
19:59.43 | slak- | which one gets matched first |
19:59.43 | fugitivo | install wctdm /sbin/modprobe --ignore-install wctdm && /sbin/ztcfg |
19:59.50 | slak- | exten => 106,1,Macro(remote,${NICKOFFICE}) |
19:59.50 | slak- | exten => _1XX,1,Macro(oneline,${EXTEN}) |
19:59.58 | slak- | i need 106 to match first |
20:00.03 | fugitivo | asterisk99: that's a single line of modprobe.conf |
20:00.04 | slak- | its using a diff macro |
20:00.37 | fugitivo | slak-: 106 first |
20:00.42 | slak- | no matter what? |
20:00.44 | slak- | always? |
20:00.54 | fugitivo | slak-: if you dial 106, it'll match with 106 |
20:01.03 | slak- | but it also matches _1XX |
20:01.17 | fugitivo | well, test it yourself |
20:01.59 | *** part/#asterisk sshadow (n=sshadow@213-84-101-107.adsl.xs4all.nl) |
20:05.59 | rob0 | asterisk99: I bet it's the path to asterisk. Default $PREFIX is /usr/local, so change that to /usr/local/sbin/asterisk |
20:06.52 | asterisk99 | rob0: how can I see the path? |
20:07.14 | fugitivo | asterisk99: do what i told you, it should work |
20:07.45 | rob0 | asterisk99: you REALLY should go through the Gentoo user guides! |
20:08.06 | asterisk99 | fugitivo: I tried that command install: unrecognized option '--ignore' |
20:08.13 | rob0 | as root do "which asterisk" |
20:08.13 | fugitivo | asterisk99: lol |
20:08.14 | asterisk99 | rob0: yes, I should |
20:08.26 | fugitivo | asterisk99: you should put that lines in your modprobe.conf |
20:08.40 | fugitivo | asterisk99: so at boot time, it'll load the right modules, and then do a ztcfg |
20:08.56 | *** part/#asterisk Samoied (n=Samoied@popeye.opens.com.br) |
20:09.27 | *** join/#asterisk hhoffman (n=hhoffman@tor/session/x-b198c2110185c500) |
20:09.44 | fugitivo | asterisk99: if you want, i have the init script for zaptel used by gentoo |
20:10.03 | hhoffman | hi, just bought a pap2 from staples and trying to flash it... seems to require a admin password... any idea as to what the password is? |
20:10.17 | fugitivo | hhoffman: blank? |
20:10.18 | rob0 | asterisk99: so many things are going against you. I am an experienced Linux admin, know my way around /bin/bash very well, and asterisk is a challenge for me. |
20:10.22 | sivana | hhoffman: chances are it's locked to Vonage |
20:10.39 | sivana | hhoffman: the default I think is admin/admin? |
20:10.40 | rob0 | but it's fun :) |
20:10.50 | fugitivo | default is admin/blank |
20:10.55 | asterisk99 | fugitivo: 1. modprobe.conf says not to edit it (says it's generated by modules-update) |
20:11.10 | asterisk99 | fugitivo: 2. can I see your script? |
20:11.29 | fugitivo | asterisk99: http://pastebin.com/464187 |
20:11.47 | hhoffman | sivana: yeah, it's locked... I'm trying to unlock it... admin/admin doesn't work :-( |
20:11.55 | hhoffman | neither does admin/blank |
20:12.07 | fugitivo | hhoffman: you should buy the pap2-na |
20:12.12 | fugitivo | hhoffman: that's unlocked |
20:13.34 | hhoffman | they're a bit tough to get a hold of |
20:13.43 | asterisk99 | fugitivo: that file is saved as /etc/init.d/zaptel ?? |
20:13.53 | sivana | not if you get authorized with a supplier |
20:14.10 | fugitivo | asterisk99: yes, then do rc-update add zaptel default |
20:17.02 | paryl | why would i be able to dial into a Zap channel but not out of it? |
20:17.24 | sivana | what's your Zap channel.. PRI or just a Bell line? |
20:17.42 | *** join/#asterisk bangawanga (n=andhecke@ppp-82-135-14-116.mnet-online.de) |
20:18.08 | bangawanga | did anybody succeed configuring asterisk with ser and livecomm?? |
20:18.14 | Seldon1975 | this may seem like a basic question, but how do I check my voicemail messages on the * server from my SIP andset? |
20:18.28 | sivana | Seldon1975: create an extension to VoicemailMain() |
20:18.29 | Seldon1975 | handset |
20:18.37 | Seldon1975 | aha |
20:18.40 | Seldon1975 | ok |
20:18.48 | Seldon1975 | thx |
20:19.48 | asterisk99 | fugitivo: K ... did as you suggested... rc-update add zaptel complains zpatel already installed in runlevel default; skipping |
20:20.09 | asterisk99 | fugitivo: K ... rc-update del zaptel ?? |
20:20.10 | fugitivo | asterisk99: ok, then reboot and test it |
20:20.19 | fugitivo | no, leave it |
20:20.34 | asterisk99 | fugitivo: ok - here goes nuttin' |
20:21.23 | paryl | sivana: it's a te205p connected to a rhino channelbank |
20:21.29 | *** join/#asterisk razu (n=razu@217-159-240-134-dsl.est.estpak.ee) |
20:22.08 | bangawanga | nobody who has the challenge to get those programs working together? |
20:24.18 | *** join/#asterisk bangawanga (n=ahecker@ppp-82-135-14-116.mnet-online.de) |
20:24.29 | asterisk99 | fugitivo: did not work - had to manually do ztcfg and asterisk after boot |
20:24.54 | fugitivo | asterisk99: what does it say? |
20:28.11 | asterisk99 | fugitivo: message .... /etc/init/d/zaptel syntax errors |
20:28.16 | bangawanga | is there anybody who tested asterisk <> ser <> livecomm 2005 ?? ser can translate between tcp<->udp, right? |
20:28.48 | asterisk99 | fugitivo: i think I see it |
20:28.49 | Seldon1975 | can I get something straight; if you bought an echo cancelling card and you enable echo cancelling; should you still expect to hear echo? |
20:29.07 | Seldon1975 | is it generally accepted that you cant be completely free of iut? |
20:29.09 | asterisk99 | fugitivo: vestige of HTML |
20:29.20 | *** join/#asterisk mtupper (n=mtupper@pc-14-172-104-200.cm.vtr.net) |
20:30.10 | mtupper | anyone here successfully call SIP > PSTN using * and a spa-3000? |
20:30.37 | asterisk99 | fugitivo: I fiexed it, and if I manually do a ./zaptel start, I still get syntax error |
20:31.01 | asterisk99 | fugitivo: or..... is that a script I can't run on my own? |
20:31.11 | *** join/#asterisk CleanerX (n=nix@nat-ph3-wh.rz.uni-karlsruhe.de) |
20:31.32 | *** join/#asterisk Igbothom_III (n=HiltonT@203-206-114-129.dyn.iinet.net.au) |
20:32.39 | Seldon1975 | mtupper yes |
20:32.43 | Seldon1975 | mtupper: i have |
20:33.23 | m160858 | which adapter is better than the SPA ? |
20:33.26 | mtupper | Seldon1975, i beg you for a quick tutorial! I am at my wits end... |
20:33.37 | Seldon1975 | i had trouble with it too, |
20:33.50 | *** join/#asterisk |cleric| (n=dacleric@p54829A91.dip0.t-ipconnect.de) |
20:33.55 | Seldon1975 | until i found out that you HAVE to turn use authentication with the * server |
20:34.07 | fugitivo | asterisk99: if i don't know the error, i can't help you |
20:34.14 | *** join/#asterisk nagl (n=nagl@213.235.241.6) |
20:34.19 | Seldon1975 | ie: set a username and password on the SPA3000 that matches username and secret in your SIP.conf for that device |
20:34.33 | paryl | will reload chan_zap.so reset the active zap channels? |
20:35.11 | *** join/#asterisk nagl (n=nagl@213.235.241.6) |
20:35.26 | hugo-v6 | ok. found out how to get cid screened on outgoing calls with misdn. |
20:35.36 | mtupper | Seldon1975, forgive me for being such a rookie, but could you specify more there... where do I modify this? in which .conf? |
20:36.06 | Seldon1975 | ok first you need to set your configuration on the SPA3K by using a browser to visit the IP address of the SPA device |
20:36.11 | mtupper | Seldon1975, by the way, I am using astbill on top of my asterisk... |
20:36.27 | Seldon1975 | there you can go to Advanced settings and sell it where your SIP server (*) is |
20:36.36 | Seldon1975 | i have no idea about Astbill |
20:37.08 | Seldon1975 | after that you just need to edit Asterisk's sip.conf and add an entry there corresponding to the SPA3K's IP address |
20:37.24 | asterisk99 | fugitivo: syntax error |
20:37.27 | mtupper | Seldon1975, yes, I have configured the spa-3000 about 50 times in the last 24 hours... I can successfully call SIP > SIP from the phone plugged into the SPA to my x-lite softphone... but I cant for the life of me get my SPA to use the PSTN line... |
20:37.55 | Seldon1975 | hmm |
20:37.57 | Seldon1975 | how odd |
20:38.05 | hugo-v6 | i guess thats better than every existing tool |
20:38.07 | Seldon1975 | I have no issue calling PSTN |
20:38.08 | asterisk99 | fugitivo: ''/etc/init.d/zaptel' has syntax errors in it; not executing...'' |
20:38.23 | Seldon1975 | mtupper I didnt need to do any thing 'special' to achieve this |
20:38.35 | syle | rrdtool is kewl, used that like 5 years ago, i thought there would be something better by now :) |
20:38.49 | Seldon1975 | mtupper do you have other SIP devices calling PSTN lines on your network? |
20:38.50 | hugo-v6 | Seldon1975: its more reliable |
20:38.53 | Nivex | syle: sometimes you just get something good an it endures |
20:38.54 | asterisk99 | fugitivo: 'i found it!!!!!!! |
20:39.02 | mtupper | Seldon1975, not yet |
20:39.05 | hugo-v6 | syle: nothing better than rrdtool :) |
20:39.15 | Seldon1975 | ok so your issue may be nothing specific to the SPA3000 |
20:39.20 | asterisk99 | fugitivo: one & became & ... I didn;t see it |
20:39.31 | syle | good to know |
20:39.34 | sivana | I need an AGI script that query's a databas and returns 1 or 0... who wants to do it |
20:39.56 | Seldon1975 | mtupper: but you CAN make SIP-SIP calls on your network? |
20:40.45 | hugo-v6 | sivana: write it yourself. not that hard |
20:41.00 | mtupper | Seldon1975, yes, I have completed many calls from SIP to SIP via combinations of softphones and SPAs (1001, 2100 and 3000)... |
20:41.00 | syle | sivana: perldoc DBD::mysql |
20:41.01 | sivana | hugo-v6: if I wanted to do that, do you think I would have posted it here |
20:41.07 | syle | big 5 lines of code to do that |
20:41.08 | sivana | name your price |
20:41.36 | hugo-v6 | sivana: well whats to do exactly? i get 75euros/h + taxes |
20:42.13 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
20:42.16 | hugo-v6 | syle: shhhhhh! |
20:42.18 | sivana | I need to query a table with the CID area code and return 1 if its canada or 0 if not |
20:42.30 | hugo-v6 | syle: i would bill him 1h fpr each line ;) |
20:42.35 | syle | lol |
20:42.42 | shmaltz | anybody here know what is normal to pay for an independed agent for mr charges? |
20:43.01 | *** join/#asterisk asteriskmonkey (n=phil@69.156.197.242) |
20:43.28 | hugo-v6 | sivana: well ull see we're not that cheap here ;) better do it yourself. and its really easy |
20:44.00 | sivana | alright.. clearly you're not interested your too busy |
20:44.10 | syle | depends |
20:44.12 | syle | how much? |
20:44.27 | sivana | I need an AGI script that query's a databas and returns 1 or 0... who wants to make a couple of bucks (besides hugo-v6) |
20:44.27 | hugo-v6 | syle: dont do it cheaper or i d0s you ;)) |
20:44.49 | syle | #DEFINE coupleofbucks |
20:44.59 | sivana | if it's such an easy script.. should be easy money for you.. $25-30 |
20:45.09 | asteriskmonkey | mog_work: does the iaxys only update there bin when its a new asterisk or when a newer .bin file is found... what is the process also.. when an iaxy connects does it do a request for a bin update? |
20:45.17 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
20:45.20 | hugo-v6 | i dont understand why u dont write it yourself? |
20:45.22 | syle | lol, i wouldn;t bother doing anything for 30 bucks |
20:45.34 | sivana | syle: is it really worth my time to spend the 2-3 hours learning Perl? |
20:45.42 | sivana | no.. my time is better spent making money |
20:45.43 | hugo-v6 | lol |
20:45.50 | mog_work | yes |
20:45.56 | hugo-v6 | sivana: then pay the price |
20:45.59 | asterisk99 | fugitivo: BUDDA BING BUDDA BOOM!!!!! That did the trick!!!!!! :) Thanks fugitvo and rob0 ... I owe you BEERS!!!!! :) |
20:46.01 | puzzled | evening all |
20:46.06 | Seldon1975 | mtupper have you configured your fxs modules in zapata.conf? |
20:46.21 | mtupper | Seldon1975, NO!!! |
20:46.23 | asteriskmonkey | does anyone here have a te406p? |
20:46.23 | sivana | hugo-v6: you've yet to give me a serious price... |
20:46.31 | Seldon1975 | mtupper you gotta do that |
20:46.37 | *** join/#asterisk Gunnar (n=gunnar@34.80-203-209.nextgentel.com) |
20:46.39 | Seldon1975 | mtupper to make PSTN calls |
20:46.41 | mtupper | let me look at that, can you add more details... |
20:46.52 | syle | then i;d suggest you don;t insult people with 30 dollars :) |
20:47.00 | syle | i think people in here are worth more than that |
20:47.05 | asteriskmonkey | is kram around? |
20:47.24 | mog_work | in france |
20:47.30 | sivana | syle: are you kidding me... it's a simple script |
20:47.36 | mog_work | or actually he is probably somewhere over the ocean |
20:47.38 | hugo-v6 | sivana: well. i got enough todo in my workinghours which brings 75euros/h + taxes (own comp.) so ill better do my own job ;) |
20:47.43 | Seldon1975 | I have configured all my fxs modules (18-24) to use fxs_ks signalling, you might as well try that |
20:47.47 | sivana | hugo-v6: ya, you should |
20:47.49 | *** join/#asterisk trym (n=trym@062016209171.customer.alfanett.no) |
20:47.52 | asteriskmonkey | mog_work: did you read my msg above re the the iaxys? |
20:47.53 | mog_work | what do you want sivina? |
20:47.56 | mog_work | yes |
20:47.59 | mog_work | they will autoupdate |
20:48.12 | asteriskmonkey | based on what the bin or asterisk version |
20:48.17 | mog_work | bin |
20:48.19 | sivana | mog_work: I need a simple script to query a table with the first 3 digits of CID and return 1 if it's canada value or 0 f not |
20:48.25 | sivana | agi script |
20:48.36 | Seldon1975 | mtupper then you need a rule in your dialplan (extensions.conf) to route outgoing calls throuugh those Zap iterfaces which represent your PSTN lines |
20:48.46 | mog_work | why does it need to be agi |
20:48.50 | mog_work | and what kind of tablee |
20:48.53 | asteriskmonkey | sivana: just make a php agi script :) |
20:48.56 | sivana | it doesn't |
20:48.56 | mog_work | you can do it in dialplan most likely |
20:48.57 | sivana | lol |
20:49.04 | sivana | postgresql table |
20:49.33 | mog_work | is there app_postgres in addons? |
20:49.37 | mog_work | like app_mysql? |
20:49.40 | file | mog_work: !!! |
20:49.53 | mog_work | file: !!!!!!!!!!!!!!!!!!!!! |
20:50.00 | sivana | I don't really want to load another module just for that one call |
20:50.25 | mog_work | agi is much more work than a module |
20:50.30 | mog_work | its simple logic |
20:50.42 | mog_work | i dont generally do agi, as C agi just seems dumb |
20:50.55 | file | C agi |
20:50.56 | mog_work | and I only write in c or dial plan for free these days |
20:50.59 | mog_work | it exists |
20:50.59 | syle | you smoking dope? |
20:51.03 | mog_work | i had to do it for someone |
20:51.03 | sivana | I'm not asking for free |
20:51.06 | sivana | never did |
20:51.14 | sivana | hehe |
20:51.16 | mog_work | its a conflict of intrest |
20:51.21 | mog_work | you can pay digium |
20:51.22 | mog_work | to pay me |
20:51.27 | mog_work | but that can be expensive |
20:51.30 | mog_work | as im not cheap |
20:51.31 | sivana | :) |
20:51.47 | file | mog_work: you're not a cheap programmer whore? :( |
20:51.52 | syle | higher than a 2 cent hooker? |
20:52.01 | mog_work | im more of a programming slut |
20:52.04 | mog_work | ill give it away |
20:52.12 | mog_work | but offering me cash insentives is pointless |
20:52.39 | mog_work | mmm muffinlicious |
20:53.03 | file | mog_work: did you get one of the cookies I sent Mark? |
20:53.12 | mog_work | you sent mark cookies |
20:53.14 | *** join/#asterisk nagl (n=nagl@213.235.241.6) |
20:53.19 | mog_work | i cant believe he didnt share..... |
20:53.20 | asteriskmonkey | im cheap ill work for food/sex |
20:53.26 | mog_work | lol |
20:53.39 | syle | i think he should jsut get billing software anyways, from what he described hes just selling tollfree numbers based on a CDN or US rate |
20:53.46 | syle | not much more |
20:54.31 | hugo-v6 | asteriskmonkey: well. for sex is a point... but what if its your worst dreams type? |
20:54.41 | mog_work | eh its not hard |
20:54.45 | mog_work | but shouldnt be agi |
20:54.56 | syle | i have it done in c |
20:54.59 | syle | as a module |
20:55.10 | mog_work | well you dont even need a specific |
20:55.11 | mog_work | one |
20:55.18 | mog_work | isnt there app_postgresql |
20:55.31 | mog_work | or did i imagine it |
20:55.37 | hugo-v6 | mog: afaik ther is at least a app_mysql |
20:55.40 | syle | yeah its in the source tree |
20:55.44 | mog_work | i know of app_mysql |
20:55.49 | syle | under cdr directory i beleive |
20:55.58 | *** join/#asterisk harryvv (n=none@S010600a0c93f6f7e.vs.shawcable.net) |
20:56.00 | asteriskmonkey | hugo-v6: if she be hidious its sex pluse tdmxxx or texxx cards |
20:56.04 | mog_work | i just wasnt sure if there was app_odbc and app_psgql |
20:56.18 | harryvv | is there any DID providers for the 604 area code? |
20:56.29 | *** join/#asterisk lesouvage (n=lesouvag@82.74.114.137) |
20:56.35 | sivana | syle: it's much more than that |
20:56.39 | asterisk99 | My next big question is... how can I use something like cronolog to switch Asterisk's logs daily at midnight? Or if not cronolog, what? :) :) :) |
20:56.46 | hugo-v6 | asteriskmonkey: hrhr |
20:57.04 | syle | sivana is there? tollfree is only thing i know of that uses callerid of incomming call |
20:57.21 | syle | or maybe some attempt at a calling card solution |
20:57.32 | sivana | syle: any PSTN interface sends CID |
20:57.33 | mog_work | there is cron rotate or something |
20:57.47 | lesouvage | I have this output in the cli: "Dec 14 15:54:52 WARNING[3247]: chan_oh323.c:4147 oh323_gk_check: Gatekeeper discovery failed." What should I do to get rid of this h323 isn't used. |
20:57.50 | hugo-v6 | sivana: not in .de |
20:57.52 | mog_work | umm no sivana |
20:57.55 | syle | yes but normally you route based on what they called not where they're calling from |
20:57.59 | mog_work | most CAN send |
20:58.01 | mog_work | not all do |
20:58.04 | sivana | true |
20:58.07 | *** join/#asterisk nagl (n=nagl@213.235.241.6) |
20:58.12 | asterisk99 | syle: Inbound CallerID is useful... I use it to route calls to different extensions |
20:58.19 | sivana | syle: you have no idea the purpose of the script :P |
20:58.23 | *** join/#asterisk fulgas (n=fulgas@a81-84-117-84.cpe.netcabo.pt) |
20:58.45 | mog_work | logrotate |
20:58.51 | slak- | hey what does Answer do |
20:58.52 | mog_work | thats what its called i thought |
20:58.56 | slak- | is it harmless? |
20:58.58 | mog_work | lol |
20:59.00 | mog_work | depends |
20:59.01 | paryl | what does "progress with cause code 127" mean? |
20:59.04 | mog_work | but mostly yes |
20:59.11 | syle | well talking about asterisk not SER, yes you;d use SER to route calls wherever on what areacode dialed |
20:59.20 | slak- | mog_work: i have Answer as the s,1 in my macro for local extensions |
20:59.28 | slak- | s,1,Answer |
20:59.32 | slak- | s,2,Wait |
20:59.33 | mog_work | thats harmless |
20:59.36 | *** join/#asterisk limo (n=limo@limonet.opf.slu.cz) |
20:59.38 | *** join/#asterisk file[desk] (n=jcolp@mctnnbsa24w-142167049176.nb.aliant.net) |
20:59.39 | slak- | s,3,Dial |
20:59.49 | mog_work | harmless |
20:59.49 | slak- | why would one want an Answer there |
20:59.54 | harryvv | traditional pbx passes cid? |
21:00.07 | mog_work | its a just in case |
21:00.07 | *** join/#asterisk insomni (n=insomni@x1-6-00-e0-18-6f-34-ad.k455.webspeed.dk) |
21:00.19 | slak- | ok |
21:00.23 | syle | btw what are you guys using for SER stateless redirects or stateful? |
21:00.46 | Seldon1975 | how cna i stop a specific Zap channel |
21:00.51 | asteriskmonkey | do alot of people use ser in conjunction with asterisk? |
21:00.59 | Seldon1975 | i cant do 'zap stop 19' |
21:01.00 | file | syle: depends how evil you get... |
21:01.23 | limo | please can somebody help me with message "Bridge stops bridging" ? |
21:02.10 | syle | asteriskmoney, anyone big does |
21:02.10 | Katty | FILE |
21:02.33 | paryl | "progress with cause code 127" brings up 4 results on google, all from the same thread :\ |
21:02.42 | *** join/#asterisk santiago (n=santiago@208.195.215.154) |
21:02.48 | paryl | and none of them really explain anything |
21:03.00 | syle | its fairly new to me, but i;m quite impressed at their c code |
21:03.37 | file | KATTY |
21:03.45 | Katty | file: hi. |
21:03.52 | file | Katty: hi. |
21:04.02 | *** join/#asterisk nagl (n=nagl@213.235.241.6) |
21:04.41 | *** join/#asterisk nagl (n=nagl@213.235.241.6) |
21:04.45 | *** join/#asterisk stoffell (n=stoffell@d51A4D148.access.telenet.be) |
21:05.37 | Katty | wow, i've not gotten any email in 5 hours. |
21:05.40 | Seldon1975 | ive been looking for ages and still cant find a GSM codec for windows to play the asterisk sound files |
21:06.04 | *** join/#asterisk loick (n=loick@APuteaux-151-1-74-135.w83-204.abo.wanadoo.fr) |
21:06.10 | file | yup |
21:06.12 | file | it's all my fault |
21:06.16 | Katty | i /knew/ it! |
21:06.20 | syle | echo "here you go baby" | mail katty@pornstar.com |
21:06.38 | Katty | ... |
21:07.30 | *** join/#asterisk Seldon1975 (n=someone@CPE0013105d0913-CM0014e8b6162c.cpe.net.cable.rogers.com) |
21:09.28 | syle | people still use windows for mail :( |
21:09.35 | Seldon1975 | can someone tell me where the vulnerability is in this dialplan? it seems that someone is dialling in, and then dialling out onto the PSTN again somehow: http://pastebin.com/464260 |
21:09.41 | gnosys | could someone point me (again) to that config & xml zip file for polycom phones? |
21:10.29 | Seldon1975 | gnosys: http://www.freedomphones.net/polycom/files/ |
21:10.51 | gnosys | thx! |
21:10.57 | *** join/#asterisk EriSan (n=erisan@81-174-42-85.f5.ngi.it) |
21:11.11 | Seldon1975 | np |
21:11.24 | Seldon1975 | now maybe someone will look at my dialplan issue |
21:11.31 | Seldon1975 | http://pastebin.com/464260 |
21:11.39 | Seldon1975 | D-Fender? |
21:11.43 | Seldon1975 | asteriskmonkey? |
21:13.07 | gnosys | wish i could help with your dialplan, but i'm sure it's more sophisticated than I could manage at this point in my learning. |
21:13.30 | Seldon1975 | ;) |
21:14.16 | hamish | anyone using hfc based bri cards? |
21:14.38 | *** join/#asterisk bartpbx (n=bartpbx@p54B0411C.dip0.t-ipconnect.de) |
21:14.54 | bartpbx | hello |
21:15.07 | fugitivo | Seldon1975: well, your dial out logic is inside your default context where your incoming calls go |
21:15.24 | Corydon-w | Seldon1975: Yeah, anybody calling in can dial an external extension |
21:15.42 | Corydon-w | Seldon1975: you should instead be using a different context for incoming than for internal calls |
21:15.42 | bartpbx | how can i convert an existing wav file to get it accepted by asterisk? I always recive " Unexpected header size 16" |
21:15.51 | Seldon1975 | ok |
21:15.53 | Seldon1975 | thanks guys |
21:16.03 | Seldon1975 | I set up these contexts in Zapata.conf? |
21:16.04 | syle | file: what can i use to compare times, ie if current hour is between 9pm and 6am etc |
21:16.12 | fugitivo | Seldon1975: you should create another context for dialing out, and another context for local extensions, include the dialout context in your extensions context |
21:16.13 | file | gotoiftime? |
21:16.13 | *** join/#asterisk malcolmd (n=malcolmd@pdpc/sponsor/digium/malcolmd) |
21:16.16 | Seldon1975 | ie: the Zap channel/Context correspondence |
21:16.22 | syle | using time.h |
21:16.32 | sivana | mog_work: I'll try the app_sql_postgres.so |
21:16.34 | Seldon1975 | ok thanks |
21:16.35 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
21:16.47 | Seldon1975 | do you by any chance have an example of this type of dialplan? |
21:16.52 | Corydon-w | No, you set up contexts in extensions.conf |
21:16.58 | bartpbx | oh.. I'm to silly.. I've just fixed it my self.. sorry |
21:16.59 | mog_work | awesome |
21:17.03 | fugitivo | Seldon1975: check asterisk config samples |
21:17.23 | syle | suppose i could use that builtin function, but i noticed a problem with gotoiftime to that it doesn;t work after 11:59 pm |
21:17.33 | Seldon1975 | Corydon-W: but dont I map Trunk lines (specific Zap interfaces) to the extensions.conf contexts in zapata.conf? |
21:17.42 | Corydon-w | Seldon1975: you already have a separate context for faxes |
21:17.54 | Seldon1975 | yes; I set that up in zapata.conf too |
21:18.03 | syle | so if i say gotoiftime sun-thurs 9pm-8am its not accurate |
21:18.04 | Seldon1975 | so that Zap21 uses that context |
21:18.05 | Corydon-w | You map channels, whether or not they're in a trunk or not |
21:18.13 | Seldon1975 | right |
21:18.15 | *** join/#asterisk zotz (n=zotz@24.231.47.168) |
21:18.29 | Seldon1975 | so in zapata.conf I mapped Zap/21 to 'faxes' |
21:18.40 | syle | would have to do fri 00-00 to 8am :( |
21:18.41 | *** join/#asterisk cyberjew (n=jhochber@adsl-065-007-156-054.sip.asm.bellsouth.net) |
21:18.42 | tainted- | Seldon1975 is this your dayjob or a client? |
21:19.06 | cyberjew | does anyone have the cisco 6.3 firmware? |
21:19.42 | tainted- | cause u might not want to attempt any more production stuff until u get a better understanding of asterisk |
21:20.09 | asteriskmonkey | seldon1975: you rang? |
21:20.32 | Seldon1975 | asteriskmonkey yes its ok thank others have stepped up |
21:21.23 | SkramX | Hi all |
21:21.27 | mrinvader | hi all! I have a goofy question - how/where would i go to set the dtmf tone length/spacing? i have a ksu that i believe cant handle digits as fast as asterisk sends them. myasterisk ver is 1.0.9 . |
21:22.09 | asteriskmonkey | seldon1975: ok cool :D |
21:22.37 | cyberjew | howdy y'all...i've got a cisco 7960g here, but i need the 6.3 SIP firmware |
21:23.05 | cyberjew | cant find it anywhere |
21:23.13 | syle | so login to cisco site and get it |
21:23.22 | cyberjew | my support contract is not setup yet |
21:23.37 | mtupper | bit-torrent? |
21:23.46 | cyberjew | ordered from cdw weeks ago...they are slacking |
21:23.56 | cyberjew | and i'm getting frusterated |
21:23.59 | mrinvader | dern slackers :) |
21:24.01 | SkramX | heh, hi cyberjew |
21:24.04 | *** part/#asterisk m160858 (n=jsaenz@200.89.12.46) |
21:24.37 | cyberjew | any of you be able to help me out? |
21:24.52 | cyberjew | <<would |
21:25.00 | paryl | so i tried the MARK2 echo cancellation and it chopped up the PSTN side, MARK3 is unreliable and sometimes screams at us, MG2 acts as though it's not even there... can anyone suggest another to try? |
21:25.09 | kuku5 | How can I make a dial command AFTER 5:30 pm and a different one after 8:30a m |
21:25.14 | mog_work | we do not promote piracy cyberjew |
21:25.15 | mog_work | sorry |
21:25.17 | stoffell | paryl, doesn't MG2 help you? |
21:25.27 | mog_work | gotoiftime kuku |
21:25.32 | Katty | right. some people need to ping me. |
21:25.32 | kuku5 | ok |
21:25.33 | paryl | stoffell, not really... same ol' echo |
21:25.33 | mog_work | which is now just a function in trunk |
21:25.38 | *** part/#asterisk cyberjew (n=jhochber@adsl-065-007-156-054.sip.asm.bellsouth.net) |
21:25.40 | kuku5 | mog_work: thanks |
21:25.41 | stoffell | what phones you using paryl? |
21:25.45 | paryl | polcom 501 |
21:25.46 | LostFrog | I don't understand how companies can charge for firmware updates. |
21:25.49 | paryl | polycom* |
21:25.50 | mog_work | you can do show functions to see that |
21:25.57 | mog_work | yeah it doesnt make sense |
21:26.03 | syle | paryle KB2? |
21:26.13 | mog_work | mg2 not kb2 |
21:26.21 | syle | he renamed it? |
21:26.39 | asteriskmonkey | what there is a kb2 now? |
21:26.45 | stoffell | paryl, i just read this today, haven't tried it yet : http://sangoma.editme.com/wanpipe-linux-asterisk-debugging |
21:27.00 | mrinvader | wouldnt a firmware update sometimes count the same way as say going from win95 to win98 tho? |
21:27.05 | mog_work | no mg2 |
21:27.10 | *** join/#asterisk shido6 (n=shido6@d221-68-216.commercial.cgocable.net) |
21:27.15 | kuku5 | mog_work: what if i dont want to GOTO - I want to just DIAL |
21:27.27 | *** join/#asterisk Katty (n=angela@68-112-15-110.dhcp.cpgr.mo.charter.com) |
21:27.34 | mog_work | then you want the fuctionstuff |
21:27.41 | kuku5 | which is what stuff ? |
21:27.48 | Katty | mog_work: would you tell me what my ping reply is please. |
21:27.48 | mog_work | other wise you are just gonna have to prefix that dial with the gotoitftimes |
21:27.48 | *** part/#asterisk bartpbx (n=bartpbx@p54B0411C.dip0.t-ipconnect.de) |
21:27.49 | asteriskmonkey | mog_work: damn had my hopes up |
21:28.18 | mtupper | Seldon1975: i could use your assistance again... |
21:28.21 | mog_work | 2 seconds katty |
21:28.25 | mog_work | sorries |
21:28.29 | Katty | thanks. |
21:28.34 | *** join/#asterisk backblue (n=moo@87-196-38-64.net.novis.pt) |
21:28.38 | mog_work | and kuku do show functions |
21:28.41 | mog_work | in asterisk |
21:29.11 | mtupper | Seldon1975: so I have uncommented signalling=fxo_ls in my zapata.conf is this what you were referring to? |
21:29.25 | Seldon1975 | mtupper not quite |
21:29.39 | mtupper | ummm... |
21:29.42 | Seldon1975 | mtupper: your fxs devices are the ones to the PSTN |
21:30.09 | Seldon1975 | you need an entry in there to mirror what you have in zaptel.conf |
21:30.18 | Seldon1975 | can you pastebin your zaptel.conf? |
21:30.33 | Seldon1975 | http://pastebin.com/ |
21:30.55 | mtupper | k, one sec... but FXO to PSTN FXS to phone, no??? |
21:31.19 | mrinvader | anyone here play with dtmf timings?? |
21:31.20 | stoffell | paryl, you're readin it now, aren't ya ? :) |
21:31.32 | Seldon1975 | it depends if you're talking about the hardware modules themselves, of the signalling |
21:31.34 | harryvv | no need to |
21:31.34 | Katty | Hmmhesays: FOP is being evil :< |
21:31.36 | paryl | stoffell: yeah :) interesting |
21:31.38 | Seldon1975 | which are exactly opposite |
21:31.42 | LostFrog | mtupper: on a FXO device, you use FXS signalling, and vice versa. |
21:31.43 | Seldon1975 | i know - its confusing |
21:31.45 | Katty | Hmmhesays: you should stabbity it. |
21:31.50 | stoffell | yes paryl, I'm gonna try it out asap also :) |
21:31.59 | mtupper | ahhhh! thx |
21:32.20 | *** join/#asterisk nagl (n=nagl@213.235.241.6) |
21:32.25 | Seldon1975 | so the setup in zapata.conf is the OPPOSITE of what you have in zaptel.conf |
21:32.32 | *** join/#asterisk heath__ (n=root@12-215-32-62.client.mchsi.com) |
21:32.33 | Beirdo | nothing like breaking stuff to release some tension |
21:32.34 | Seldon1975 | ...which makes for fun for newbies |
21:32.51 | mtupper | LostFrog, Seldon1975: but I ummmm, dont have a zaptel.conf, just a zapata.conf .... same thing? |
21:32.53 | mrinvader | i want to stabbity asterisk for trying to talk to my phone system too fast :P LOL |
21:32.54 | kuku5 | anyone have a iftime() example? |
21:33.20 | paryl | stoffell: thing is, i don't know if my echo issues are what they're describing |
21:33.25 | Katty | Beirdo: :> |
21:33.25 | Seldon1975 | mtupper: you do have it - just not in the same place |
21:33.37 | Seldon1975 | mtupper: /etc/zaptel.conf |
21:33.39 | stoffell | paryl, indeed, first thing one should do is, find the 'source' of the echo |
21:33.46 | stoffell | wich is not always an easy task |
21:33.47 | Seldon1975 | mtupper: you need to set up both |
21:33.51 | mtupper | k, damn, I feel like the rookie I am... |
21:34.24 | stoffell | paryl, this page has good resources: http://www.voip-info.org/wiki-Asterisk+echo+cancellation |
21:34.30 | stoffell | at the bottom some nice links |
21:34.31 | paryl | they mention muting the other side. if the other side mutes, the echo goes totally away. |
21:34.39 | paryl | yeah, i've been over that page over and over |
21:34.44 | stoffell | aha |
21:34.55 | paryl | problem is the echo is only present on about 30% of our outside calls |
21:35.04 | stoffell | hm, same problem here :( |
21:35.10 | kuku5 | same here |
21:35.13 | heath__ | true or false: there's no way to do command substitution in the dialplan without doing something hack assed like System(OUTPUT=`echo "whatever"` ; /usr/sbin/asterisk -rx dbput family key ${OUTPUT}) |
21:35.21 | paryl | and of course, those agents with headsets are CAUSING echo issues |
21:35.29 | paryl | PRI |
21:35.35 | kuku5 | PRI |
21:35.55 | *** join/#asterisk genmud (n=genmud@ip68-98-82-206.ph.ph.cox.net) |
21:36.37 | heath__ | i'd rather give my server Adderall |
21:36.48 | paryl | i'd rather go to the bar |
21:37.01 | paryl | a nice glass of Wild Turkey on the rocks... mmm |
21:37.03 | paryl | rocks |
21:37.06 | Beirdo | eek |
21:37.14 | asteriskmonkey | would be sweet to map audio to key presses so people can be like 1... one etc :D |
21:37.21 | mtupper | Seldon1975: not there either!!! i'm running Asterisk 1.2 on Debian Sarge with MySQL 5, and Astbill (and if you didn't notice I am a total rookie, only 6 weeks using linux)!!! |
21:37.22 | mrinvader | shotgun it with acapulco gold... |
21:37.27 | stoffell | paryl, you won't have any echo then :D |
21:37.28 | Beirdo | I should make some more bourbon sauce |
21:37.36 | Beirdo | that Jim Beam needs to get finished |
21:37.40 | paryl | stoffell: lol |
21:37.55 | mrinvader | mmmmm bourbon sauce... |
21:37.57 | paryl | you guys see the Good Eats about bourbon? i never knew that stuff |
21:38.07 | paryl | Alton Brown is the bee's knees |
21:38.08 | Beirdo | not yet, when was it on? |
21:38.19 | paryl | a couple weeks ago |
21:38.21 | paryl | i tivo'd it |
21:38.22 | Seldon1975 | mtupper: if you dont have etc/zaptel.conf then I have no idea how you are set up.... sorry! |
21:38.31 | Beirdo | ah, it will take a while to be shown in Canada |
21:38.37 | paryl | actually.. i guess it was egg nog, but a large part was about bourbon |
21:38.42 | Beirdo | ah |
21:38.45 | Beirdo | the egg nog one? |
21:38.48 | Beirdo | let me look |
21:39.05 | Beirdo | bourbon and egg nog do NOT go in the same sentence for me |
21:39.23 | LostFrog | RUM.. only RUM. |
21:39.24 | paryl | yeah... the color/flavor of bourbon comes from the wood of the burned out barrels they age it in |
21:39.38 | Beirdo | yeah, I knew that |
21:39.39 | Beirdo | :) |
21:39.43 | Beirdo | same goes for rum |
21:39.59 | Beirdo | the color's from the barrels... that were previously used for bourbon |
21:40.02 | mrinvader | rum is brown sugar based tho isnt it? |
21:40.02 | Beirdo | in some cases |
21:40.07 | paryl | i never knew they burned them! that's so odd |
21:40.13 | Beirdo | molasses-based, yes |
21:40.22 | mrinvader | ya |
21:40.51 | mtupper | Seldon1975: thanks anyway |
21:40.53 | mrinvader | so the darker rums have more cane solids in them. |
21:41.00 | Beirdo | umm, no |
21:41.06 | Beirdo | the darker rums are from aging |
21:41.14 | mrinvader | ahhh. k. |
21:41.23 | paryl | Beirdo: 1/4 coconut run, 1/4 creme de banana, 1/4 peach schnapps, 1/4 mountina dew code red |
21:41.26 | Seldon1975 | Im about to try tweaking my SIP Phones Audio Processing settings to reduce echo. My JitterBuffer min/shrink/max are 40/500/160. can someone suggest other values to try? |
21:41.26 | paryl | rum* |
21:41.32 | *** join/#asterisk jsolares (n=jsolares@200.6.233.132) |
21:41.51 | Beirdo | I like my rum in a glass. :) |
21:41.54 | Beirdo | jummy |
21:41.54 | paryl | i call it hawaiian punch |
21:42.13 | jsolares | hmmm how is ringing indication supposed to work with E1/T1's? |
21:42.30 | mrinvader | rum in a vanilla/eggnog milkshake. w00t |
21:42.43 | Beirdo | mmmm, pint of Havana Club dark. |
21:42.49 | jsolares | if i answer the incomming zap channel before dialing a sip phone i get ringing, if i dont answer i dont get ringing, am i doing something wrong? :S |
21:42.52 | mrinvader | XD |
21:43.17 | Beirdo | and my (now) fiancee witnessed that on the webcam |
21:43.18 | Beirdo | hehe |
21:43.31 | Beirdo | boozy goodness |
21:43.45 | mrinvader | rofl |
21:43.53 | paryl | jsolares: with either a ring or a polarity reversal |
21:44.43 | mrinvader | or power fluctuations in the trans-warp oscillators |
21:44.52 | jsolares | hehe |
21:45.04 | mrinvader | %P |
21:45.34 | jsolares | paryl: but this is an E1 not a fxo/fxs channel :S |
21:46.19 | jsolares | maybe i should make asterisk believe that it has answered the channel when in fact it hasnt so it sends the ringing audio... weee |
21:46.51 | jsolares | or get the sip phones to send ringing audio? hmmm this is confusing |
21:46.57 | mrinvader | everything i have is up and working.. except that when i dial from the server or from a sip phone it randomly will not make it out to the pstn ... im figgerin its gotta be asterisk sending digits too fast. |
21:47.41 | paryl | jsolares: where are you? |
21:47.50 | jsolares | in central america |
21:48.00 | syle | no shit |
21:48.03 | syle | where abouts? |
21:48.18 | jsolares | in guatemala city |
21:48.20 | mrinvader | [sip-phone] to [asterisk server 1.0.9] to [vodavi triad 2 ksu] to pstn |
21:48.21 | paryl | jsolares: ah... i don't know what the network does there. i've never seen that problem |
21:48.30 | *** join/#asterisk Gamercjm (n=gamercjm@71.254.175.163) |
21:48.31 | jsolares | crap |
21:48.36 | *** join/#asterisk harryvv (n=none@S010600a0c93f6f7e.vs.shawcable.net) |
21:49.10 | syle | hehe kewl |
21:49.13 | jsolares | well if i make the line digital so asterisk wont send that there's audio inband to the E1 it works if i call in from a big telco, seems they get to generate the ringing tone for me |
21:49.20 | syle | aren;t they holding a survivor series there? |
21:49.32 | jsolares | they were, it ended yesterday or so |
21:50.00 | jsolares | but that was way up north in the jungle, the migthy jungle where the lions sleep at night.... or something |
21:50.08 | jsolares | :) |
21:50.11 | syle | lol |
21:50.12 | Gamercjm | I tried setting up DISA and when i test i get the dialtone and call other extensions, but wont allow outbound calls, get busy sound. Is DISA able to make outbound calls? |
21:50.15 | *** join/#asterisk Dio_ (n=dima@82.207.0.44) |
21:50.48 | Beirdo | DISA goes into the context you tell it to |
21:50.56 | syle | not sure what the problem is but have you tried the r option to dial to get the audio to the sip phones? |
21:51.09 | Beirdo | if the context you put it in can't do outbound calls, then DISA won't do outbound calls |
21:51.27 | jsolares | the problem is not getting the audio to the sip phones, but getting it to the incomming calls via the zap pri |
21:51.30 | brettnem | THE 'r' OPTION!!! |
21:51.42 | syle | lol |
21:51.42 | mrinvader | the only time ill ever watch realitard tv is when they have thr unning man in real life and ppl actually die from eating rancid whale anus or fighting a rabid randy water buffalo barehanded. |
21:51.49 | *** join/#asterisk shido (n=shido@d221-68-216.commercial.cgocable.net) |
21:51.50 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
21:51.54 | Gamercjm | oh I just set to defualt, ill check into context then |
21:52.00 | syle | the evil 5979 right :) |
21:52.07 | jsolares | tried with and with no r, no luck |
21:52.09 | Beirdo | mrinvader: or when they do "Survivor: Alaska" |
21:52.11 | LostFrog | Damn, steal my idea for a show, why don't you, mrinvader. |
21:52.25 | mog_work | do you get calls on this sip phone |
21:52.32 | LostFrog | mrinvader: "Survivor: North Pole" |
21:52.34 | mrinvader | LO friggidy L |
21:52.43 | jsolares | everything works, it's just ringing indication that wont |
21:52.56 | mog_work | going to or from this sip device |
21:53.07 | jsolares | from the sip to zap |
21:53.11 | Beirdo | a lot harder to survive in the north than in the frigging rain forest/paradise island |
21:53.12 | jsolares | E1 digium card |
21:53.14 | *** join/#asterisk toddf (n=toddf@net-66-210-104-104.theshop.net) |
21:53.30 | mog_work | well if you have r it should give you ring nomatter what |
21:53.35 | mog_work | what if you do m |
21:53.39 | mog_work | do you get musiconhold |
21:53.58 | *** join/#asterisk Igbothom_III (n=HiltonT@203-206-114-129.dyn.iinet.net.au) [NETSPLIT VICTIM] |
21:55.14 | *** join/#asterisk stkn (i=nobody@gentoo/developer/pdpc.active.stkn) |
21:56.06 | *** join/#asterisk jsolares (n=jsolares@200.6.224.146) |
21:56.15 | jsolares | meh this internet is killing me |
21:56.47 | jsolares | i'd rather be in a frozen tundra with a reliable internet than this paradise jungle with crappy connection... |
21:57.29 | Beirdo | Survivor: IRC? |
21:58.17 | jsolares | since those would all be geeks, they'd die within minutes of the show XD |
21:58.27 | LostFrog | lol.. all the lamers would be killed the first day. |
21:59.21 | Beirdo | all that would be left would be the ops |
21:59.23 | LostFrog | I know.. take 12 average windows users, install Linux to their machine.. |
21:59.28 | LostFrog | Survivor: Linux |
21:59.39 | Beirdo | to make it more fun... |
21:59.47 | mrinvader | nah the geeks would fashion a nuclear furnace out of radium watch dials and old at power supply.. |
21:59.49 | Beirdo | give them a Gentoo CD, no web access. |
22:00.00 | Beirdo | and watch em squirm |
22:00.02 | LostFrog | owww. |
22:00.06 | LostFrog | You are mean, Beirdo. |
22:00.08 | Beirdo | :) |
22:00.19 | Beirdo | with the docs, Gentoo's easy enough |
22:00.23 | LostFrog | By web, I assume you mean internet.. |
22:00.26 | Beirdo | without... have fun |
22:00.36 | Beirdo | no, I mean specifically web |
22:00.38 | mrinvader | nah.. microsoft evangelists.. not just winders users.. |
22:00.43 | LostFrog | Ohh.. ok then. |
22:00.49 | LostFrog | I can do it without the web. |
22:00.52 | Beirdo | they can emerge -sync all they want |
22:00.55 | Nugget | Microsoft Flight Simulator is a great product! :) |
22:01.05 | Beirdo | yeah, but you ain't a Microsoft-only gumby |
22:01.11 | mrinvader | yeah im glad they bought it arent you? |
22:01.16 | LostFrog | Without internet, it would suck. |
22:01.54 | Beirdo | hmmm |
22:01.58 | mrinvader | now are we talking the cheapbytes complete gentoo set or one from jerkit city.. |
22:02.05 | Beirdo | well, I've had enough of work for today |
22:02.07 | Beirdo | stage 1 |
22:02.08 | Beirdo | duh |
22:02.18 | Beirdo | no shortcuts allowed |
22:02.56 | Beirdo | or if you are especially cruel. |
22:03.08 | Beirdo | give em a box of 50 floppies with OLD Slackware on it |
22:03.11 | LostFrog | LFS with no host OS. |
22:03.11 | Beirdo | circa 1992 |
22:03.12 | twisted[asteria] | wake me up |
22:03.15 | twisted[asteria] | can't wake up |
22:03.17 | twisted[asteria] | save me |
22:03.48 | LostFrog | Been there, Beirdo.. done that. |
22:03.52 | mrinvader | YAH!!! watch em fall to the ground and seize/soil themselves.. |
22:03.58 | Beirdo | LostFrog: me too :) |
22:04.14 | LostFrog | I want to buy a Yggdrasil CD. |
22:04.17 | Beirdo | "what's LaTeX"? |
22:04.26 | nextime | anyone with a call termination with DialMex? |
22:04.33 | LostFrog | Umm.. that's the stuff that rubbers are made of. |
22:04.42 | Beirdo | no, that's "latex" :) |
22:04.55 | Beirdo | not even pronounced the same |
22:04.55 | LostFrog | Ohh.. then it's a document markup language. |
22:05.01 | mrinvader | i was gonnasay Phylactixx Pro v.69 |
22:05.03 | LostFrog | Well.. TeX is. |
22:05.11 | Beirdo | exactly, that's what the nooobs would be wondering though |
22:05.50 | Beirdo | make em compile gcc and then ghostscript |
22:05.51 | Beirdo | heh |
22:05.57 | Beirdo | oh, and glibc |
22:06.03 | LostFrog | ok.. gcc can be a pain. |
22:06.08 | Beirdo | by hand, no emerging, that's for wusses |
22:06.13 | mrinvader | anyone have any ideas about tone timings? no.. XFree 4.1 |
22:06.15 | LostFrog | It seems not to compile 50% of the time. |
22:06.18 | Beirdo | and qt |
22:06.20 | fugitivo | linuxfromscratch.org |
22:06.32 | LostFrog | fugitivo: been there, got the t-shirt. |
22:06.40 | fugitivo | i use lfs for my servers |
22:06.45 | fugitivo | no crap inside |
22:06.49 | LostFrog | I use minimal debian. |
22:06.59 | Beirdo | sure there's crap, but you put it all there :) |
22:07.10 | Beirdo | so you know WHAT crap is there :) |
22:07.10 | mrinvader | me II. minimal debian runs my firewall. |
22:07.37 | mrinvader | debian potato. |
22:07.49 | mrinvader | on a p133/32mb machine |
22:07.57 | Beirdo | OpenBSD 3.6 |
22:07.58 | LostFrog | I run sarge everywhere now. |
22:08.00 | fugitivo | hmm, that's nice for openbsd |
22:08.00 | Seldon1975 | Im about to try tweaking my SIP Phones Audio Processing settings to reduce echo. My JitterBuffer min/shrink/max are 40/500/160. can someone suggest other values to try? |
22:08.01 | Beirdo | anyways. |
22:08.15 | Beirdo | have fun, people |
22:08.24 | mrinvader | sarge is cool. did a fw in indiana with sarge. |
22:08.32 | mrinvader | c ya beirdo |
22:08.34 | fugitivo | openbsd and pf |
22:08.45 | Nugget | pf+altq is tasty. |
22:08.51 | fugitivo | i love pf |
22:08.56 | fugitivo | i want a pf's tshirt |
22:09.14 | mrinvader | que es pf? |
22:09.21 | mrinvader | just curious |
22:09.31 | Nugget | a really great packet filter/firewall stack |
22:09.39 | mrinvader | ahhhh |
22:09.39 | fugitivo | much better than iptables |
22:09.54 | fugitivo | better syntax |
22:10.06 | mrinvader | bsd is very reliable for stuff like that too. i use it for mailservers. |
22:10.11 | LostFrog | If I could get over my SYSV racism, I would install BSD> |
22:10.17 | LostFrog | . |
22:10.18 | fugitivo | and you can do bridging |
22:10.29 | fugitivo | transparent |
22:10.30 | LostFrog | You can do bridging in Linux. |
22:10.36 | iDunno | FreeBSD isn't so bad :) |
22:10.38 | LostFrog | Yes.. transparent. |
22:10.52 | mog_work | bsd and linux can all do same things now |
22:10.52 | fugitivo | LostFrog: but, pf is cool :) |
22:10.54 | mog_work | woopity do |
22:10.58 | iDunno | and linux bridging is fucking easy these days. |
22:11.00 | mog_work | not the place for this fight |
22:11.13 | fugitivo | what fight? |
22:11.14 | mrinvader | gotta love those holy wars.. i love it all. |
22:11.18 | fugitivo | linux and bsd are friends |
22:11.19 | Nugget | openbsd (and freebsd to a lesser degree) is still more capable of protecting itself from local users. |
22:11.23 | mrinvader | yaa |
22:11.27 | iDunno | and iptables is nicer (for me) than pf/ipf/ipfw/fuckingputtheevilbsdtoolofdoomhere |
22:11.38 | astcryz | Anyone had issues with asterisk realtime and md5password? |
22:11.39 | fugitivo | pf is 37337 |
22:11.44 | Nugget | linux is better at using crap like $9 tape drives that you bought at hamvention |
22:11.46 | mog_work | i dont know i think linux and bsd can be pretty pissy with each other |
22:11.47 | mrinvader | lololiololol |
22:11.53 | Nugget | (which is great for users who have $9 worth of data) |
22:12.28 | LostFrog | I think BSD is good.. for those who like it. |
22:12.44 | fugitivo | i like openbsd soundtracks |
22:12.45 | mog_work | i think bsd/linux is not much more than prefrence these days |
22:12.50 | fugitivo | linux doesn't have soundtracks |
22:12.50 | mog_work | obsd songs rule |
22:12.54 | mog_work | lol |
22:12.57 | mrinvader | i like native bsd process accounting for some things and linux's native baremetal speed for others |
22:13.00 | mog_work | ill make one for you fugitivo |
22:13.07 | Nugget | linux bridging cheats inside, which makes it unworkable in some situations. (like needing to do bidirectional filtering in bridged mode) |
22:13.15 | fugitivo | mog_work: thank you |
22:13.29 | LostFrog | Nugget: that works now, as well. As well as accounting. |
22:13.31 | Nugget | to be fair, so does freebsd's |
22:13.33 | tdonahue | SIP/2.0 407 Proxy Authentication Required |
22:13.38 | Nugget | when did they fix it? |
22:13.39 | mog_work | like i said |
22:13.42 | tdonahue | what setting would cause asterisk to start sending a 407 response? |
22:13.54 | mog_work | little advantages here and there |
22:13.59 | mrinvader | it ran out of 406'es? |
22:14.03 | LostFrog | Some time midway through 2.4. |
22:14.04 | mog_work | i use linux because zaptel hardware works better in linux |
22:14.06 | Flauto | mog_work, how are you doing |
22:14.10 | mog_work | Grand |
22:14.10 | *** join/#asterisk heison (n=heison@ns.somanetworks.com) |
22:14.17 | fugitivo | well, but that's like mysql vs pgsql, now mysql has a lot of new functions, but i moved to pgsl years ago, i'll not try again mysql |
22:14.17 | LostFrog | lol@mrinvader. |
22:14.36 | synthetiq | i use macos to run asterisk because i want to be leet |
22:14.43 | mog_work | yeah |
22:14.51 | mog_work | i agree gutiov |
22:14.58 | mog_work | err fugitivo |
22:15.00 | synthetiq | v.9 !!!!!!!!!!!11one |
22:15.01 | fugitivo | lol |
22:15.01 | LostFrog | Send me OSX86, and I'll run * on it. |
22:15.17 | fugitivo | give me solaris for x86! |
22:15.17 | iDunno | LostFrog: why on earth?! |
22:15.26 | iDunno | OSX is just hideously broken ;) |
22:15.28 | LostFrog | iDunno: just to say I could. |
22:15.33 | synthetiq | * on pico bsd =x |
22:15.36 | Nugget | OSX is doubleplusgood. |
22:15.48 | iDunno | and OpenSlowarsis has a semi broken licence |
22:15.51 | fugitivo | osx is a os for sissy girls |
22:15.57 | synthetiq | osx double playchild toy good |
22:16.15 | iDunno | OSX is fine for 3 year olds that only want bouncy icons and no productivity... |
22:16.16 | fugitivo | is the "barbie" theme out for osx? |
22:16.19 | mog_work | ugh... |
22:16.25 | mog_work | os fight |
22:16.26 | mog_work | over |
22:16.28 | mrinvader | osx can get ppl off winders.. an enemy of my enemy is my friend :LOL |
22:16.32 | iDunno | kinda like XP is fine for all telly tubbies ;) |
22:16.47 | mrinvader | idunno : XD XD XD |
22:17.01 | Nugget | OSX is unix on the desktop that doesn't totally suck. |
22:17.06 | Nugget | it's the *only* one, imho. |
22:17.07 | mrinvader | i think all the *nices need to ally |
22:17.20 | iDunno | mrinvader: the figures actually show that OSX takes people away from the real operating systems to OSX, they were generally linux or FreeBSD users before hand ;) |
22:17.22 | fugitivo | Nugget: kde doesn't suck |
22:17.26 | tdonahue | so no one knows why my account settings that worked 2 days ago no longer work? |
22:17.30 | brookshire | kde is great! |
22:17.32 | iDunno | Nivex: no it isn't. |
22:17.41 | iDunno | erm - Nugget: no it isn't. |
22:17.46 | fugitivo | kde is functional for me, and i do a lot of work with it |
22:17.47 | Nugget | I find KDE to be pretty miserable, but a lot of the blame lies with x.org/xfree86. |
22:17.56 | Nugget | X11R6 is just too crude to do anything modern with |
22:17.56 | iDunno | Nugget: OSX is a bastardisation of a Unixoid. |
22:18.17 | iDunno | it's FreeBSD with all the useful bits ripped out and replaced with shit. |
22:18.26 | fugitivo | Nugget: why? i have 3d, dual head |
22:18.28 | Nugget | clipboard sucks, it can't even do font kerning properly. |
22:18.31 | iDunno | also, it has a broken python *by default* |
22:18.34 | mog_work | ugh |
22:18.36 | mog_work | lets move on |
22:18.47 | iDunno | mog_work: agreed. |
22:18.49 | fugitivo | Nugget: clipboard? i have 2 clipboards with my kde |
22:19.07 | Nugget | so you can copy an image out of mozilla and paste it into openoffice? |
22:19.16 | mog_work | so there is this guy |
22:19.20 | mog_work | he walks into a bar |
22:19.27 | mog_work | and orders a small drink and a cactus |
22:19.37 | LostFrog | lol.. a cactus. |
22:19.54 | mog_work | now think about that for a minute |
22:20.02 | mrinvader | it _is_ all just preference outside of doze. i unite with all the unixes/alikes. we have the entirety of the non-billy world now. |
22:20.36 | fugitivo | Nugget: no, but it copies the link to that image :) |
22:20.47 | Nugget | heh |
22:20.53 | Nugget | how almost useful. |
22:21.02 | fugitivo | anyways, i never had to do that |
22:21.04 | mog_work | back to the cactus |
22:21.08 | Nugget | probably because you can't. :) |
22:21.30 | LostFrog | I've never done it in windoze.. Always saved the image and imported it. |
22:21.34 | fugitivo | i didn't know i can't :) |
22:21.39 | fugitivo | me too |
22:21.50 | brad_mssw | can anyone in here verify that teliax is down right now ? |
22:21.58 | fugitivo | i don't like that drag'n drop stupid thing, hehe |
22:22.07 | *** join/#asterisk Seyr (n=Seyr@rrcs-24-227-133-226.sw.biz.rr.com) |
22:22.09 | file[laptop] | brad_mssw: call and ask if they're down? lol |
22:22.27 | brad_mssw | file[laptop]: haha ... not like they answer their phones even when they're up |
22:22.31 | fugitivo | brad_mssw: simple, does it work? |
22:22.40 | LostFrog | There was a time when you couldn't call the broadvoice support line from a broadvoice number. :) |
22:22.47 | fugitivo | if it doesn't then the service is down |
22:22.55 | brad_mssw | fugitivo: want to make sure my ISP didn't just start blocking this outgoing port ... |
22:23.09 | fugitivo | are you in china? |
22:23.12 | Seyr | im using Realtime and set mailbox=100@mycontext , but when i call Voicemail(u100@mycontext) the SQL query that Realtime does, has context="" .. so it is not getting the context. anyone seen this before? |
22:23.13 | brad_mssw | fugitivo: they've been known to do stupid things like that ... |
22:23.37 | brad_mssw | fugitivo: no, not in china ... in US, but my ISP has primarily windows boxes, so they close down about any port they see a lot of traffic on |
22:23.39 | LostFrog | Do a nmap from another machine. |
22:23.44 | brad_mssw | fugitivo: don't ask :/ |
22:23.58 | fugitivo | brad_mssw: ok :) |
22:24.07 | *** part/#asterisk mtupper (n=mtupper@pc-14-172-104-200.cm.vtr.net) |
22:24.17 | brad_mssw | just looking for anyone else that uses teliax that can confirm |
22:24.26 | mrinvader | brad_mssw: what are you trying to connect out on? |
22:24.46 | brad_mssw | mrinvader: voip-co3.teliax.com .. iax2 registration is failing ... port 4569 |
22:25.35 | tainted- | brad_mssw teliax was having issues |
22:25.45 | mrinvader | yeah i doubt that it would be blocked.. thats an unprivileged port. they usually only kill ports like 25 and such. |
22:26.10 | tainted- | mrinvader what do u mean 'unprivileged' |
22:26.14 | mrinvader | my isp unblocked me because i host a real mailserver |
22:26.23 | LostFrog | tainted-: less than 1024. |
22:26.32 | mrinvader | root/admin access .. thanks lost! |
22:26.48 | synthetiq | what you need is a good iax2 termination service brad_mssw =] |
22:26.50 | mrinvader | i cnt type fast enuf *nix geek or not. |
22:26.57 | tainted- | yea but how does that mean 'unprivileged' |
22:27.10 | mrinvader | or one not down right now at least LOL |
22:27.13 | tainted- | brad_mssw does teliax go down a lot? |
22:27.36 | LostFrog | tainted-: only root-owned process can open any port less than 1024 in *nixes. |
22:27.42 | mrinvader | 1025+ do not req. admin privs on a server to open up. |
22:27.50 | mrinvader | zacly |
22:28.03 | brad_mssw | tainted-: this is the first time it's gone down on me |
22:28.04 | tainted- | i did not know what |
22:28.05 | synthetiq | yiu need izx2 termination on a loadbalanced set up =] |
22:28.09 | iDunno | LostFrog: not true, there are kernel patches that give privileges to certain users for other ports. |
22:28.20 | tainted- | s/what/that |
22:28.30 | mrinvader | well right but under normal circumstances... |
22:28.43 | tainted- | brad_mssw i was going to port a # to them when they had this downtime |
22:28.45 | LostFrog | right.. and the kernel runs at ring 0. |
22:28.49 | tainted- | brad_mssw their portal went down too |
22:28.51 | iDunno | under a usual distro, in usual circumstances, granted ;) |
22:28.52 | LostFrog | It can do anything it wants. |
22:29.08 | mrinvader | like httpd runs as a user and so does ssh with privilege separation. |
22:29.16 | brad_mssw | tainted-: hmm, their portal is up ... this is for business unfortunately, so it hurts more ... ported our 800 and local numbers over |
22:29.25 | iDunno | but the selinux patches, IIRC, allow you to actually provide a more open policy on ports |
22:29.33 | LostFrog | mrinvader: httpd starts as root and chroots as a normal user. |
22:29.39 | iDunno | and thus you don't have to even start daemons as root ;) |
22:29.58 | *** join/#asterisk JakBeatZ (n=JakBeatZ@gw2tor1nat161.beanfield.net) |
22:30.07 | *** join/#asterisk nagl (n=nagl@213.235.241.6) |
22:30.15 | mrinvader | and mahcrusawf jes allows anyone to do that stuff as a user... including frobbing packets :) LOL |
22:30.33 | tainted- | brad_mssw do u terminate through them or do u have others? |
22:30.38 | mrinvader | lostfrogtrue that |
22:31.52 | *** join/#asterisk clarity (n=chatzill@84.255.196.166) |
22:31.55 | brad_mssw | tainted-: we have a single PTSN line coming in for 911 routing but everything else is through teliax ... we have a direct 100Mbps fiber connection to our ISP (only 2 doors down), so we thought reliability wouldn't be a problem, at least from our end |
22:31.57 | JakBeatZ | Is it possible to use a variable as an extension in a context? For example assume a [global] EXT => 100 can you have exten => $EXT,1,Dial(SIP/100) or is that an invalid use of an extension? |
22:32.06 | JakBeatZ | er.. invalid use of a variable, rather |
22:32.59 | LostFrog | I don't believe you can. |
22:33.34 | mrinvader | brad_mssw: it might not be on your end.. | wow... MMMmmm.. Fiber... i need to get more fiber in my network diet. |
22:33.34 | clarity | good evening everyone... i have a bit of a strange question: I'm trying to force plain authenthication to a SIP phone, does anyone know if this is possible? i tried auth=plain but asterisk that I use doesn't send anything different with that option, and phone still does md5 auth |
22:35.17 | tainted- | brad_mssw yea origination u can't do anything about.. but termination u should probably look into some redundancy |
22:35.26 | *** join/#asterisk brookshire (n=nubb@gateway.digium.com) |
22:35.29 | Seyr | Anyone know why Realtime is giving me: [SELECT * FROM vmail WHERE mailbox = '555' AND context = '' ] even though I am calling it with "555@mycontext" ??? |
22:35.36 | LostFrog | How can you have redundancy for incoming termination? |
22:35.42 | Seyr | its cutting out the context |
22:35.56 | tainted- | termination meaning outgoing calls |
22:36.03 | _Sam-- | f'n teliax, down for the 2nd time today....been down a total of over an hour today...got 15 employees sitting around twiddlin thumbs |
22:36.17 | mrinvader | idunno: why would one want to run a demon on a 1024- port as a user.. isnt that kinda dangerous? |
22:36.36 | tainted- | _Sam-- i think there are several people peeved with teliax |
22:37.23 | iDunno | mrinvader: think httpd |
22:37.44 | lesouvage | How can I prevent hisax and isdn modules to load during startup. I can remove them with rmmod but I don't want them to load at all. |
22:37.50 | iDunno | mrinvader: that's a user. not one that I'd *ever* give a password or login access, but a user all the same. |
22:38.28 | LostFrog | Why would you want to run a daemon as root, if you didn't want to, mrinvader? |
22:38.36 | LostFrog | didn't need to, even. |
22:39.37 | Seldon1975 | do the 'context's in zapata.conf correlate to the 'context's in extensions.conf? |
22:39.45 | brad_mssw | _Sam--: yeah, i'm pissed at teliax right now ... |
22:39.52 | brad_mssw | _Sam--: guess they're not a good business solution for us |
22:40.08 | Faithful | Hey guys, I have looked on google... I have a problem when I use a particular provider I get chan_iax2.c:5067 socket_read: meta trunk cmd 1 received, I only understand 0 (perhaps the remote side is sending trunk timestamps?) |
22:40.47 | _Sam-- | brad_mssw: i feel your pain and then some. |
22:40.51 | Faithful | and I don't get incoming sound... outgoing is fine |
22:40.55 | _Sam-- | who will you switch to? |
22:42.10 | *** part/#asterisk Seyr (n=Seyr@rrcs-24-227-133-226.sw.biz.rr.com) |
22:43.47 | *** join/#asterisk testmachine (n=assink@ip237-239-58-62.adsl.versatel.nl) |
22:43.48 | clarity | perhaps someone can tell me what does auth=plain actually do? just allow plain auth? doesn't force it? |
22:43.57 | clarity | any help is much appreciated |
22:45.34 | mrinvader | well all its been a lot of fun.. ill be back tamarra at the butt-crack o' 10am (EST). gnight all. |
22:46.42 | brad_mssw | _Sam--: looks like teliax just came back online |
22:47.37 | _Sam-- | indeed |
22:48.01 | *** join/#asterisk |Vulture| (n=V@216.84.158.117) |
22:48.48 | Seldon1975 | hey guys, if i have contexts [incoming] and [outgoing] in extensions.conf, how do I configure the context in zapata.conf for my Zap interfaces to the outside world |
22:48.48 | Seldon1975 | which are used for both incoming and outgoing |
22:48.48 | |Vulture| | Anyone seen an instance where a polycom phone keeps trying to get an IP from a DHCP server? The server issues the DHCPDISCOVER then DHCPOFFER until it fails? |
22:49.44 | _Sam-- | nice of teliax...call our toll free number it says it has been disconnected |
22:50.23 | Sedorox | ahaha |
22:50.59 | Sedorox | dunno how far you are from canada..... but I know a good canadian place that does toll free... never had a problem with them |
22:51.46 | _Sam-- | they fixed it now |
22:52.30 | _Sam-- | if you can recommend a good IAX provider id love to know |
22:52.40 | _Sam-- | or maybe i should go sip? |
22:53.15 | Sedorox | http://www.thinktel.ca |
22:53.18 | Sedorox | they do both sip and iax.. |
22:53.30 | Sedorox | think they prefer to do sip tho.. but not sure :p |
22:54.05 | _Sam-- | what would be the downside to switching server communications from IAX to SIP? |
22:54.37 | Sedorox | I was told there are still issues with sip trunking and stuff on asterisk... |
22:54.41 | Sedorox | but I haven't done any tests myself... |
22:57.08 | *** join/#asterisk ldnblk (n=Just@212.183.128.185) |
22:57.47 | gnosys | hey folks, Polycom phones. paryl sent me to his blogspot site: http://paryl.blogspot.com/2005/10/making-polycom-501-work-we_113034274646163867.html but I have a question about this... what are folks doing for the ftp server? vsftpd? proftpd? other? setting it up with local users in /etc/passwd? a "virtual user" database? ldap repository? other? |
22:58.20 | *** join/#asterisk JakBeatZ (n=JakBeatZ@gw2tor1nat161.beanfield.net) |
22:58.40 | Flauto | is iaxtel still working |
22:59.05 | mog_work | sometimes |
22:59.37 | Flauto | mog_work, i can never get it registered |
23:01.53 | paryl | gnosys: i had success with pure-ftpd |
23:01.54 | Flauto | itis always not registered |
23:02.17 | gnosys | how did you set up the user list? in /etc/passwd? other? |
23:02.18 | paryl | gnosys: http://paryl.blogspot.com/2005/10/setting-up-pure-ftpd.html :) |
23:02.33 | gnosys | oh! thanks, paryl. |
23:02.38 | paryl | i knew logging my experiences would come in handy :) |
23:02.46 | JakBeatZ | If I have a callback context configured in voicemail.conf and the callback context requires a dialing prefix, how would that work? The number called back would be based on the callerid so the dialing prefix wouldn't be added to the callerid. Conversely, if I had an outbound context configured in my voicemail.conf and I wanted to dial an outbound number that required a different dialing prefix, will that "just work"(tm) or would I need |
23:02.47 | JakBeatZ | <PROTECTED> |
23:03.37 | Flauto | or, the context you can use by default is fromvm |
23:03.50 | gnosys | it definitely did for me, paryl! thanks. |
23:03.52 | Flauto | you can create that context in your exensions.conf |
23:04.05 | paryl | gnosys: awesome! glad to know i could help |
23:04.17 | Flauto | to define what dialout channel you want to use |
23:05.13 | Flauto | mog, thanks for your answers last night. i have some more questions |
23:05.17 | Flauto | if you dont' mind me asking |
23:05.47 | Flauto | whatis the difference in between enumlookup and dundi? |
23:06.06 | Flauto | i know that i can set up some channels for dundi |
23:07.00 | mog_work | dundi is kinda like distributed enum |
23:08.23 | _Sam-- | if i have 12 incoming trunks via IAX, would there be any problems switching them to SIP? |
23:08.30 | mog_work | no |
23:08.45 | _Sam-- | ty |
23:08.53 | *** join/#asterisk kshumard_home (n=ksh@pcp01931374pcs.huntsv01.al.comcast.net) |
23:08.56 | Flauto | mog, how exactly should i use dundi |
23:09.05 | Flauto | that is the part i am not sure |
23:09.18 | Flauto | there is examples in the extensions.conf |
23:09.21 | sivana | mog_work: http://pastebin.ca/33709 |
23:09.45 | sivana | thank you :) |
23:10.49 | mog_work | yay! |
23:10.59 | mog_work | now isnt that better than an ugly agi? |
23:11.04 | sivana | I hate the module, but it works :) |
23:11.23 | mog_work | sorries |
23:11.27 | mog_work | im glad it works though |
23:11.49 | mog_work | dundi is not a real trivial thing to do honestly |
23:11.50 | sivana | ya.. the parameter thing is weak, maybe some day I'll make it a function instead |
23:12.02 | mog_work | took me a bit of time to set up first time |
23:13.23 | mog_work | hehe im a genius, the bug is gone in my library now i just need to code it |
23:15.33 | Flauto | mog, i know people can monitor calls to see where the call is from |
23:15.37 | *** join/#asterisk xachen (i=justin@magnum.thisgeek.com) |
23:15.41 | Flauto | what should i do to set it up |
23:15.48 | Flauto | like when a call is coming in |
23:15.55 | Flauto | you see it on your cli |
23:16.01 | Flauto | but you don't see where the call is from |
23:16.56 | mog_work | you can do a noop(${CALLERIDNUM}) |
23:17.14 | Flauto | where should i set it though |
23:17.23 | mog_work | zap show chanel |
23:18.15 | Flauto | i am not using zap channel much |
23:18.19 | Flauto | i use broadvoice |
23:18.22 | *** part/#asterisk Jestre (n=ack@dargo.trilug.org) |
23:18.27 | Flauto | let me try |
23:19.06 | mog_work | oh iax2 show channels |
23:19.11 | mog_work | or your channel |
23:20.57 | Flauto | broadvoice is sip. so it should be sip show channels? |
23:21.04 | *** join/#asterisk jsaunders (i=js@S01060060971c5817.va.shawcable.net) |
23:21.06 | mog_work | broadvoice i thought was both |
23:21.08 | mog_work | but yes |
23:21.43 | jsaunders | Anyone know of any articles discussing Sangoma Shark cards? |
23:21.48 | shido | . |
23:22.01 | Flauto | sip only |
23:22.14 | mog_work | havent seen any jsaunders |
23:22.15 | mog_work | just picks |
23:22.23 | mog_work | vapor drivers i think still |
23:22.35 | Flauto | done |
23:22.38 | jsaunders | Likewise |
23:22.52 | mog_work | looks spiffy though |
23:23.00 | jsaunders | Indeed |
23:23.37 | Flauto | 147.135.12.128 3129620228 53732565368 02691/00000 unkn No |
23:23.41 | Flauto | this is what i see |
23:24.10 | mog_work | okies |
23:24.13 | mog_work | what do you want to know |
23:24.24 | drray | I want a 24xxp |
23:24.59 | jsaunders | What the heck is a 24xxp? Heheh |
23:25.08 | mog_work | tdm 2400p |
23:25.21 | jsaunders | Make/model? |
23:25.29 | mog_work | http://www.digium.com/index.php?menu=product_detail&category=hardware&product=TDM2400P |
23:25.31 | drray | digiums new FXS/FXO 23 port card |
23:25.34 | drray | er, 24 |
23:25.34 | jsaunders | Tnx |
23:26.08 | mog_work | i want for my apt |
23:26.18 | mog_work | and just have a phone in every knook and cranny |
23:26.18 | jsaunders | Quite the beast. |
23:26.22 | mog_work | yes |
23:26.25 | jsaunders | :D |
23:26.26 | mog_work | full height full length |
23:26.40 | drray | it's expandable, which is what I like, I can grow to 24 |
23:26.48 | mog_work | yup |
23:26.51 | drray | as opposed to just buying 24 and lumping it |
23:27.07 | mog_work | or change modules out |
23:27.10 | jsaunders | That backplane bus idea from Sangoma is pretty slick. |
23:27.17 | mog_work | its intresting |
23:27.25 | mog_work | but i dont have a machine with 4 pci slots |
23:27.28 | jsaunders | Heheh |
23:27.38 | mog_work | and wouldnt want to use em all for a card |
23:27.46 | mog_work | rather have 96 ports of tdm ^_^ |
23:27.48 | jsaunders | Makes sense. |
23:28.00 | jsaunders | Biased is ok. |
23:28.03 | jsaunders | Don't beat yerself up. |
23:28.07 | mog_work | no if you do a whois on my ip |
23:28.20 | gnosys | paryl, you still here? |
23:28.46 | jsaunders | mog_work: Heheh. Gotcha. ;) |
23:29.19 | mog_work | just pluggin away |
23:29.33 | gnosys | anybody here know polycom phones? |
23:29.42 | mog_work | yeah |
23:30.24 | gnosys | paryl pointed me to his blogspot on configuring them, but he mentions files that I don't see in the xml zip file: 000000000000-phone.cfg |
23:30.37 | drray | mac address? |
23:31.06 | Flauto | mog, noop works |
23:31.08 | mog_work | thats the default one it will spray to all |
23:31.11 | mog_work | yay flauto |
23:31.22 | Flauto | i see callerid when someone is calling in |
23:31.24 | Flauto | thank |
23:31.25 | Flauto | s |
23:31.28 | gnosys | well, he says to fill in the mac address into the zeros here, but there's no file by that name. There is a phone1.cfg, is that it? |
23:31.34 | mog_work | no prob |
23:31.44 | mog_work | thats probably sameone |
23:32.04 | gnosys | so i should rename it following the mac address naming convention i guess? |
23:32.20 | mog_work | i thik all 000s flushes any phone |
23:32.39 | shido | no |
23:32.39 | shido | man |
23:32.42 | shido | no 0' |
23:32.48 | shido | th phone will use the 0's file |
23:32.50 | shido | for default |
23:32.55 | shido | if no mac address cfg file is found |
23:32.59 | mog_work | rigth |
23:33.01 | shido | u can read that in the -log file in the same dir |
23:33.03 | shido | if u use ftp |
23:33.06 | shido | insetead of tftp |
23:33.07 | mog_work | otherwise you can use teh mac of the phone |
23:33.13 | shido | I suggest using ftp because when u make a change to a file |
23:33.14 | mog_work | and use ftp |
23:33.15 | *** join/#asterisk CyberPony (n=CyberPon@cpe-069-132-017-022.carolina.res.rr.com) |
23:33.16 | shido | the phone reboots |
23:33.18 | mog_work | as tftp is unreliable |
23:33.21 | shido | and updates |
23:33.39 | Seldon1975 | if I have issued 'restart when convenient' in # console; can I cancel it? |
23:33.50 | mog_work | hmm i dont know |
23:33.51 | gnosys | thanks folks... i'll try it. |
23:33.54 | mog_work | why would it matter? |
23:34.00 | shido | its not convenient for asterisk to cancel a command like that |
23:34.11 | Seldon1975 | i dont want it to restart |
23:34.18 | Seldon1975 | when the lines all stop |
23:34.18 | kshumard_home | maybe `abort halt` |
23:34.20 | mog_work | hmm maybe no or done |
23:34.24 | shido | brb |
23:34.41 | jsaunders | Golf anyone? http://img.photobucket.com/albums/v493/odog/golfcarrier.jpg |
23:34.50 | kshumard_home | `help abort halt` makes it seem like that's what you want, Seldon1975 |
23:35.21 | drray | restart when convenient, stops incoming calls.. waiting for zap to clear. so it's not a good option for us |
23:35.25 | mog_work | kshumard_home: has mad asterisk cli skills |
23:35.42 | kshumard_home | mog_work, giggity giggity |
23:35.45 | Seldon1975 | yes |
23:35.47 | drray | oh yeah |
23:35.51 | Seldon1975 | i think abort halt might work |
23:36.15 | ldnblk | jsaunders do the Sangoma cards do the polling on the card as apposed to doing it by software ? |
23:36.28 | mog_work | ldnblk: ? |
23:36.55 | *** join/#asterisk apardo (n=apardo@29.Red-81-39-85.dynamicIP.rima-tde.net) |
23:37.29 | ldnblk | well what do the Sangoma do ? |
23:37.37 | mog_work | ??? |
23:37.42 | jsaunders | Heheh |
23:37.43 | Corydon-w | Ooops... help <tab> no longer works |
23:37.50 | twisted[asteria] | lol |
23:38.14 | jsaunders | Honestly, couldn't tell ya ldnblk. I've only seen pictures, no details thus far. |
23:38.46 | Corydon-w | How exactly would you do 'polling on the card'? |
23:39.10 | twisted[asteria] | by smoking a large blunt. |
23:39.24 | Corydon-w | You're reading from the card... |
23:39.45 | Corydon-w | mog_work: I don't think he does, either. |
23:39.51 | paryl | gnosys: i'm back, sorry |
23:39.56 | paryl | did you figure it out? |
23:40.06 | gnosys | np... i may have solved it... |
23:40.12 | ldnblk | jsaunders: I was at a linux expo, and the card that u mentioned was billed as an equivelent to the digum cards |
23:40.15 | paryl | ah, great |
23:41.20 | mog_work | yeah its equiv to tdm400p |
23:41.26 | mog_work | when they get drivers done |
23:41.34 | jsaunders | Well, Digium does have more presence. |
23:41.37 | Corydon-w | So you can buy a card from Sangoma... or you could buy an equivalent card from Digium and finance further Asterisk development.... |
23:41.46 | Corydon-w | Hmmmm, what to choose, what to choose... |
23:41.50 | mog_work | keep mog off the streets |
23:41.52 | mog_work | and in college |
23:41.53 | jsaunders | Heheh |
23:41.58 | mog_work | sounds like a good plan to me ^_^ |
23:42.19 | jsaunders | Chi eyes there mog? Heh. |
23:42.38 | mog_work | im big fan of that smiley face as apposed to :) |
23:43.13 | *** join/#asterisk bjohnson (n=bjohnson@i216-58-58-83.cybersurf.com) |
23:44.45 | Corydon-w | "Keep Other Matt in college. Buy Digium." |
23:44.59 | jsaunders | mog: Did you get to do the photoshoot for that chick on digium.com? :D |
23:45.23 | jsaunders | Not bad, not bad indeed. |
23:45.24 | mog_work | lol |
23:45.25 | mog_work | i wish |
23:45.29 | mog_work | but i got a woman |
23:45.48 | Corydon-w | mog_work: damn. There goes my chances. |
23:46.03 | mog_work | lol |
23:46.05 | *** join/#asterisk riddlebox (n=james@24-217-15-91.dhcp.stls.mo.charter.com) |
23:46.19 | mog_work | keep the 6 matts off the streets is a good reason to support digium ^_^ |
23:46.34 | Corydon-w | There are now 6 Matts? |
23:46.37 | mog_work | yes |
23:46.48 | mog_work | 4 in support |
23:46.51 | mog_work | 1 in web dev |
23:46.52 | Corydon-w | Crap. 4 of you are "Other Matt" ? |
23:46.54 | mog_work | 1 in shipping |
23:47.03 | xachen | creepy |
23:47.10 | jsaunders | What about the 1 in front of the door? Heheh. |
23:47.17 | xachen | If I change my name to Matt can I wor there? :D |
23:47.20 | Corydon-w | I thought Matt F was Asterisk development |
23:47.21 | jsaunders | Oh, that was good. |
23:47.25 | DaPrivateer | gah |
23:47.29 | mog_work | yeah its a gen rule |
23:47.30 | DaPrivateer | my nick alert wont stop going off! |
23:47.31 | mog_work | well he is |
23:47.32 | DaPrivateer | lol |
23:47.37 | mog_work | i guess really |
23:47.40 | mog_work | 3 matts are in dev |
23:47.42 | *** part/#asterisk DaPrivateer (i=Privatee@CRIMSON.OFF-HOURS.COM) |
23:47.44 | Corydon-w | and Matt B is the web dev |
23:47.44 | mog_work | 1 in support |
23:47.46 | mog_work | 1 in shipping |
23:47.49 | mog_work | 1 in web dev |
23:48.03 | jsaunders | 1 on ground in front of door |
23:48.27 | mog_work | heh |
23:48.34 | mog_work | nah all matts are always on the move |
23:48.39 | mog_work | we are raid 0 |
23:48.42 | mog_work | no redundancy |
23:48.55 | Corydon-w | Do you have more than one Matt with the same initial last name? |
23:49.08 | mog_work | nope |
23:49.22 | mog_work | we have b,r,n,o,f,s |
23:49.26 | mog_work | fnorbs |
23:49.56 | drray | Yate is interesting |
23:50.05 | jsaunders | Shall I run and hide? Heheh. |
23:50.11 | nextime | anyone using chan_h323 on SVN-trunk-r7230 ? |
23:50.14 | mog_work | meh we dont care |
23:50.16 | drray | Yate has windows drivers |
23:50.19 | mog_work | be happy |
23:50.22 | paryl | yate? huh? |
23:50.23 | jsaunders | Exactly. |
23:50.24 | mog_work | yay! |
23:50.40 | jsaunders | Too bad he went and killed himself though. Sheesh. |
23:51.00 | mog_work | ? |
23:51.05 | *** join/#asterisk Druken (n=druken@CPE00121716da99-CM000e5cde4ca2.cpe.net.cable.rogers.com) |
23:51.07 | jsaunders | What was his name? Bobby McFaren? |
23:51.09 | jsaunders | Somethin' like that. |
23:51.11 | Corydon-w | and there's what, 2 people developing yate? |
23:51.29 | jsaunders | If that. Heh. |
23:51.38 | mog_work | hey incourage our oss breathren |
23:51.40 | *** join/#asterisk GaVa (n=GaVa@201.14.147.30) |
23:51.41 | Corydon-w | No, it's Diana and her man |
23:51.44 | mog_work | dont knock em |
23:51.46 | drray | If that is how they want to spend their time.. |
23:51.47 | jsaunders | He's doin' a pretty good job I must say. |
23:51.52 | mog_work | i mean people should do what makes em happy |
23:51.56 | drray | amen mog |
23:51.56 | jsaunders | I like Paul, nice guy. |
23:52.13 | jsaunders | Diana means well, but high strung at times though. |
23:52.18 | Druken | drray: haven't gotten that board yet... should i have? |
23:52.29 | drray | they said 5 days |
23:52.34 | drray | business days |
23:52.46 | Corydon-w | jsaunders: She comes off as manic |
23:52.51 | jsaunders | Indeed |
23:53.02 | Druken | drray: when was it sent again? |
23:53.14 | drray | I want to say last wednesday maybe? |
23:53.22 | drray | I don't recall for sure |
23:53.26 | drray | maybe longer |
23:53.41 | Druken | aight.. no biggie |
23:53.42 | litage | is okay to run a gatekeeper (eg: gnugk) and asterisk on the same machine? |
23:53.53 | Druken | i'm sure it'll get here when it does :) |
23:54.04 | drray | well, I declared it at zero value |
23:54.11 | drray | maybe they opened it? |
23:54.33 | Druken | that's possible... OH NO IT'S A BOMB!!!!!! |
23:54.43 | drray | well, drugs |
23:55.05 | Druken | did you pack it in cofee grounds? :) |
23:55.14 | drray | no, the stinky ass newspaper |
23:55.48 | Druken | ahh, so i'll know the news from where ever your from |
23:55.59 | drray | well seattle |
23:56.09 | drray | and you'll know what date I boxed it and mailed it |
23:56.11 | drray | :) |
23:56.17 | Druken | pfft, nothing special there... :) |
23:56.32 | Druken | good point.. |
23:56.38 | Druken | unless it was an old paper |
23:56.58 | Druken | something tells me ya didn't buy the paper to use it for packing |
23:57.02 | drray | did I email you telling it was sent? |
23:57.15 | drray | we get the paper everyday here |
23:57.16 | Druken | no, i think you msg'd me |
23:57.18 | drray | it was snowing |
23:57.22 | drray | the day I mailed it |
23:57.23 | drray | I think |
23:57.48 | jarrod | hey what is the format for a subscribecontext as specified in sip.conf? |
23:57.55 | jarrod | is it an extensions context or a sip context? |
23:58.36 | drray | Yate, is interesting to me because it uses windows versions of the tor2 drivers |
23:58.45 | drray | and it's just interesting |
23:58.53 | mog_work | ? |