| 00:04.49 | *** join/#asterisk lunaphyte (n=lunaphyt@pool-71-120-136-36.gdrpmi.dsl-w.verizon.net) |
| 00:06.00 | *** join/#asterisk hatamen (i=hatamen@222.183.20.9) |
| 00:10.08 | lunaphyte | has anyone taken a look at this tycho voice mail manager? http://sip-syndication.com/ |
| 00:12.04 | Lyfe | anyone have any suggestions as to why i might hear clicks on calls originating from IP phones and travelling out a T1 (or vice versa) every 30 seconds? If I restart asterisk, the problem seems to go away for a period of time (say, between 24 & 48 hours). |
| 00:12.10 | *** join/#asterisk kratzers (n=kratzers@kratzers.static.pa.net) |
| 00:12.11 | *** part/#asterisk TripleFFFF (n=TripleFF@145-27.mc.cite.net) |
| 00:14.08 | *** join/#asterisk saftsack (n=saftsack@p54A7D54A.dip.t-dialin.net) |
| 00:17.09 | Lyfe | nevermind, it might not be asterisk-related, i just realized there was more that got reset recently. I thought for a moment that I might have narrowed it down to asterisk, but I'm mistaken. |
| 00:17.38 | *** join/#asterisk rvhi (n=rv@66.175.65.89) |
| 00:26.10 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
| 00:27.39 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
| 00:28.48 | *** join/#asterisk justinu|laptop (n=Justin@12.44.122.130) |
| 00:32.38 | *** join/#asterisk r0d3nt (n=RatMan@foster.stonedcoder.org) |
| 00:38.23 | *** part/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 00:38.45 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 00:38.45 | *** mode/#asterisk [+o russellb] by ChanServ |
| 00:45.23 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.216.2) |
| 00:48.49 | *** join/#asterisk marv (n=ilovekim@c-71-228-189-127.hsd1.al.comcast.net) |
| 01:01.42 | *** join/#asterisk hads (n=hads@mail.nice.net.nz) |
| 01:02.07 | pyrom | how do i disable http digest auth. |
| 01:02.58 | *** join/#asterisk saftsack (n=saftsack@p54A7D54A.dip.t-dialin.net) |
| 01:13.47 | *** part/#asterisk pyrom (n=pyro@86.84-48-44.nextgentel.com) |
| 01:14.10 | *** join/#asterisk lowlevel (n=Stuart@CPE0050ba71c82f-CM000f9f7d6742.cpe.net.cable.rogers.com) |
| 01:14.47 | *** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.ITB.ac.id) |
| 01:15.42 | *** join/#asterisk quid246 (n=quid24@CPE00131078ba5d-CM000f9f7eff1e.cpe.net.cable.rogers.com) |
| 01:16.41 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 01:18.55 | *** topic/#asterisk by russellb -> Asterisk: The Open Source PBX -=- http://www.asterisk.org -=- Asterisk 1.2.11, Asterisk-addons 1.2.4, and Zaptel 1.2.8 released! (August 22, 2006) -=- FreePBX/AMP/Asterisk@Home/Trixbox support in #freepbx |
| 01:19.09 | Qwell | yay russellb! |
| 01:19.16 | russellb | :) |
| 01:19.28 | quid246 | wow, what good timing I have |
| 01:20.07 | Nivex | ooooh new toys! |
| 01:20.33 | russellb | no, Asterisk 1.4 will be the new toy |
| 01:20.40 | russellb | this is just for bug fixes :) |
| 01:21.05 | quid246 | hehe... I wonder why Digium abandoned IAX as a trademark a few years back? |
| 01:21.55 | Nivex | wow, talk about hot off the presses. the link for zaptel on asterisk.org doesn't even work yet :) |
| 01:22.08 | russellb | Nivex: heh |
| 01:22.18 | russellb | Nivex: if it doesn't within about 10 more minutes, please let me know :D |
| 01:22.24 | russellb | mirrors have to update |
| 01:23.08 | Nivex | Is Pound Key gonna get these updates? |
| 01:23.26 | russellb | i suppose |
| 01:23.34 | Nivex | I've considered tossing my Debian install and going to that, but I noticed that it's not updated as frequently. |
| 01:24.00 | quid246 | zaptel link working now |
| 01:24.24 | [TK]D-Fender | sfasdfjaskl;dfjskl;djf OMGZ! |
| 01:24.41 | Nivex | [TK]D-Fender: omgwtfbbq? |
| 01:24.47 | [hC] | rofflemayo |
| 01:25.02 | [TK]D-Fender | Nivex : kjhfdg! |
| 01:25.08 | Nivex | [TK]D-Fender: gesundheit |
| 01:25.16 | *** join/#asterisk justinu (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 01:25.21 | [TK]D-Fender | Nivex : Danke |
| 01:25.23 | jbroome | Aww come on, i just got 1.2.10/1.2.7 working right! :P |
| 01:25.26 | Nivex | [TK]D-Fender: Bitte. |
| 01:25.41 | Nivex | jbroome: tell your boss to stop making you switch hardware every three days! |
| 01:25.56 | jbroome | Nivex: this is a client machine, not ours |
| 01:26.06 | jbroome | thank jeebus |
| 01:26.08 | Nivex | which reminds me, we need to have a chat about some MeetMe configs |
| 01:26.26 | file | reminders are not allowed! |
| 01:26.51 | jbroome | I'm currently working on an * project that's due this week (14 poly 501s!) and trying to fix a freepbx install someone else did |
| 01:26.57 | jbroome | so maybe next week. :) |
| 01:27.10 | Nivex | so you say now :) |
| 01:27.26 | [TK]D-Fender | jbalcomb : Polycom's aren't a problem, Trixbox... well... Trix are for kids! |
| 01:27.35 | jbroome | yeah, i may not want to look at an extensions.conf for a while after this |
| 01:27.59 | [TK]D-Fender | jbroome : rather |
| 01:28.04 | jbroome | :) |
| 01:28.11 | jbroome | jb tab-complete claims another victim |
| 01:30.34 | *** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.ITB.ac.id) |
| 01:33.00 | *** join/#asterisk joburg (n=voipmagi@vc-196-207-36-133.3g.vodacom.co.za) |
| 01:33.22 | joburg | hi |
| 01:35.06 | *** join/#asterisk Trakkasure (n=Sgemtum@24-50-26-239.atlsfl.adelphia.net) |
| 01:35.20 | joburg | hi |
| 01:36.09 | *** join/#asterisk znoG_ (n=gs@162-148-235-201.fibertel.com.ar) |
| 01:40.33 | Nivex | are the rumours about 1.4 being able to use a posix timer for meetme true? |
| 01:41.02 | *** join/#asterisk doolph (n=doolph@200.46.148.58) |
| 01:42.46 | [shodan] | eh there's an error in the asterisk book on page 100 "remember that when you reference a variable, you can call it by its name, but when you refer to a variable's value, you have to use the dollar sign and brakets around the variable name" , it's not brackets , it's curly braces, brackets are for expressions right ? |
| 01:44.20 | joburg | right |
| 01:45.11 | [shodan] | k, so page 100/chapter six in the footer |
| 01:46.11 | *** join/#asterisk Frogdude (n=chris@c-24-16-72-159.hsd1.wa.comcast.net) |
| 01:46.35 | [shodan] | also it says that 8885551212 will match "exten => _555XXXX,1,Playback(digits/1)" , it won't right ? (page 94 chapter 5) |
| 01:47.12 | joburg | no absolutely not |
| 01:48.47 | *** join/#asterisk Flauto (n=zhao@adsl-75-3-139-218.dsl.chcgil.sbcglobal.net) |
| 01:49.05 | Flauto | Aug 22 20:39:13 WARNING[22405]: chan_sip.c:2561 sip_write: Asked to transmit frame type 64, while native formats is 4 (read/write = 64/64) |
| 01:49.10 | Flauto | what is that for |
| 01:49.29 | Flauto | that is what i got when i answered a wakeup call |
| 01:49.37 | Flauto | is there anything i can do? |
| 01:50.24 | joburg | wakeup call on your sip phone? |
| 01:50.40 | Flauto | yes |
| 01:50.46 | Flauto | i use a sipura spa 3000 |
| 01:50.57 | joburg | codec? |
| 01:51.02 | Flauto | the strange thing is that when i dial musiconhold, i hear music |
| 01:51.25 | Flauto | but when the wakeup call agi put on musicohhold, i get that message |
| 01:52.19 | Flauto | ulaw is the first choice |
| 01:52.44 | joburg | and music on hold mp3? |
| 01:52.48 | Flauto | yes |
| 01:52.51 | Flauto | mp3 |
| 01:52.58 | Flauto | and i use native |
| 01:53.21 | joburg | there is transcoding issues |
| 01:53.31 | Flauto | i agree |
| 01:53.37 | joburg | you can convert your mp3's into ulaw.... |
| 01:53.49 | Flauto | but i dont know what i could do |
| 01:54.14 | [shodan] | ~thebook |
| 01:54.16 | jbot | thebook is, like, a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 01:54.17 | Flauto | well, the thing is that i can hear music when i dial my musiconhold |
| 01:54.36 | Flauto | only i got that error message when i pick up a wakeup call |
| 01:54.49 | joburg | are you sure your callfile is correct? |
| 01:55.06 | Flauto | what you mean by call file |
| 01:56.15 | joburg | there should be a callfile which generates the wakeupcall |
| 01:56.27 | Flauto | it should be |
| 01:56.40 | Flauto | i did not have this problem earlier |
| 01:56.47 | joburg | this callfile send the call to a [context] in your dialplan when you answer |
| 01:57.45 | Flauto | it happened when i upgraded to asterisk 1.2.10 |
| 01:58.16 | joburg | well that explains alot! |
| 01:58.55 | Flauto | so, i should go back to the earliers version? |
| 01:58.59 | joburg | i don't use agi for my wakeupcall , i did it manually using crontab and a callfile |
| 01:59.08 | joburg | no don't go back ! |
| 01:59.28 | Flauto | whta should i do |
| 02:00.39 | joburg | you know howto use crontab right? |
| 02:01.46 | Flauto | not really |
| 02:01.58 | Flauto | if youdont' mind teach me |
| 02:02.09 | joburg | what linux distro? |
| 02:02.10 | *** join/#asterisk Deeewayne (n=dwayne@ool-44c0d56e.dyn.optonline.net) |
| 02:02.24 | Flauto | mandriva 2006 |
| 02:03.01 | joburg | know howto use vi ? |
| 02:03.09 | Flauto | yes sir |
| 02:04.01 | joburg | if you do a crontab -l , vi will open if you don't have anything in quit , let me know if you got anything |
| 02:04.21 | joburg | sorry thats crontab -e |
| 02:04.33 | joburg | crontab -l will list your crontab |
| 02:05.03 | *** join/#asterisk lordbaron (n=redbaron@host55-226.rancor.birch.net) |
| 02:05.48 | Flauto | i tried -l it tells me that i don't have any crontab on root |
| 02:06.00 | Flauto | when when did crontab -e |
| 02:06.04 | Flauto | it opened a new file |
| 02:06.15 | joburg | ok your crontab is a scheduling tool in linux |
| 02:06.18 | Flauto | now, i have a new file |
| 02:06.29 | joburg | you can use it to schedule pretty much anything |
| 02:07.00 | Flauto | the only thing i have done on mandriva is that i added a schedule to the sip.conf to reeload |
| 02:07.12 | Flauto | i edited the crontab file under /etc |
| 02:07.22 | lordbaron | I have 3 pots lines connected to a tdm400p. SBC. I am trying to use Kewlstart. |
| 02:07.23 | lordbaron | Incoming calls work, but outgoing calls hangup the zap by the first ring |
| 02:07.40 | lordbaron | is there any way to adjust the timing or settings? |
| 02:09.00 | Flauto | joburg, would you show me how to set up wake up call by using your way? |
| 02:09.05 | [TK]D-Fender | lordbaron : I suggest turning off call progress, etc.... |
| 02:09.35 | joburg | i'll try |
| 02:09.41 | Flauto | thanks |
| 02:09.49 | lordbaron | callprogress=no; busydetect=yes; --> should the busydetect be turned off? |
| 02:09.54 | Flauto | i have a new blank file open here now |
| 02:10.07 | joburg | first you have to understand the crontab |
| 02:10.16 | Flauto | okay |
| 02:11.12 | joburg | the usage is : 1) caracter is the minutes 2nd) the hours 3rd) the day 4ht) the day of the week and 5th) the month then the command |
| 02:11.19 | Flauto | is there any chance in 1.2.10 for musiconhold? |
| 02:12.25 | joburg | so if you go crontab -e add the following to the 1st line 30 6 * * * cp /home/wakeupcall /var/spool/asterisk/outgoing |
| 02:12.47 | joburg | this will activate your wakeupcall every morning at 6h30 |
| 02:13.02 | [TK]D-Fender | lordbaron : I'd do taht if I were you. |
| 02:13.21 | joburg | the you'll have to create your wakeupcall in your /home directory |
| 02:13.34 | lordbaron | [TK]D-Fender: Changed busydetect=no, and removed echotraining lines. Still no dice |
| 02:14.02 | *** join/#asterisk lunaphyte (n=lunaphyt@pool-71-120-136-36.gdrpmi.dsl-w.verizon.net) |
| 02:14.06 | joburg | flauto : yes moh will work |
| 02:14.36 | [TK]D-Fender | lordbaron : What does DialStatus say afterwards? |
| 02:15.05 | Flauto | is there a way to setup time by calling in the system though |
| 02:15.10 | *** join/#asterisk juanjoc (n=juanjoc@248-32-235-201.fibertel.com.ar) |
| 02:16.22 | doolph | how can I make asterisk dial faster, example if the number start with 2 and the user pressed 7 keys make asterisk dial inmediatly, not to wait those 2 or 3 secs |
| 02:16.50 | lordbaron | [TK]D-Fender: not sure how to check dialstatus from console |
| 02:17.33 | joburg | no i guess thats what agi is for.... |
| 02:17.49 | *** join/#asterisk techie (n=gus@adsl-068-209-242-072.sip.mia.bellsouth.net) |
| 02:18.19 | joburg | doolph : hit the # key |
| 02:18.41 | doolph | joburg i don't want to hit the #key |
| 02:18.46 | doolph | i want asterisk hit it for me |
| 02:19.04 | doolph | or its just impossible |
| 02:19.58 | joburg | hmmm |
| 02:21.58 | joburg | if your pattern matches * should dial quicker |
| 02:23.35 | doolph | uh |
| 02:23.39 | doolph | how pattern matches |
| 02:24.02 | joburg | in your dialplan.... |
| 02:24.16 | *** join/#asterisk r0d3nt (n=RatMan@foster.stonedcoder.org) |
| 02:24.29 | r0d3nt | 1.2.11 |
| 02:24.33 | [shodan] | I just made a call . sip phone => fxo => some pstn phone number but when I talked I could hear myself in echo about 200ms later , where do I need to enable echo cancellation ? in the phone , in the fxo configuration , in asterisk ? |
| 02:24.37 | r0d3nt | so i guess 1.0.7 is pretty old..... |
| 02:24.42 | r0d3nt | i found a old pbx i installed a while ago |
| 02:24.46 | r0d3nt | 381 days uptime.... |
| 02:24.53 | r0d3nt | OS and Asterisk uptime..... |
| 02:25.14 | doolph | rofl |
| 02:25.27 | r0d3nt | a couple hundred defunct mpg123's, but other then that.. it was running perfectly... just like the day i left it.... |
| 02:25.28 | r0d3nt | lol |
| 02:25.49 | r0d3nt | Dell SC420, sata 80gig hd, gentoo, 2.6 kernel... |
| 02:26.03 | r0d3nt | 512mb of ram... 0k swap used... |
| 02:26.36 | r0d3nt | i even forgot the root pw, and had a local account without sudo and not in wheel, so i had to local kernel exploit it to get root back without rebooting it... =) |
| 02:26.48 | jbroome | hahah |
| 02:26.52 | r0d3nt | ya... |
| 02:26.54 | r0d3nt | good times.... |
| 02:27.14 | r0d3nt | i had 3 techs who swore up and down they knew what the root pw was.... but no go.... |
| 02:27.28 | r0d3nt | normally we all have our own accounts, and we sudo, but i guess someone didn't get to that... |
| 02:27.33 | r0d3nt | ( me ) |
| 02:27.51 | r0d3nt | anyways </ramble> |
| 02:27.57 | r0d3nt | just thought i would share that with #asterisk |
| 02:27.58 | r0d3nt | <3 |
| 02:28.04 | joburg | shodan : what fxo card what sip phone? |
| 02:28.38 | r0d3nt | oh, and it's running 2 quad FXS/FXO pci cards.. 8 lines PSTN... .. |
| 02:28.40 | lordbaron | [TK]D-Fender: DIALSTATUS=answer |
| 02:30.02 | joburg | where from r0d3nt? |
| 02:30.36 | *** join/#asterisk Bazinn (n=pbaker@ool-457805b1.dyn.optonline.net) |
| 02:31.20 | [TK]D-Fender | lordbaron : Not good.. like the other side hung up on you.... does it always happen? |
| 02:31.27 | Bazinn | hi all, I was hoping someone can give advice on how to convert telephone numbers to letters for searching, IE sort of what the directory application does.. |
| 02:31.27 | lordbaron | yes |
| 02:31.41 | lordbaron | if I change to Loopstart, no problems, but then I lose hangup detection |
| 02:31.48 | Bazinn | IE for searching for names like 'jsmith' par example |
| 02:32.00 | r0d3nt | joburg, where am I from ??? California / Nevada USA... |
| 02:32.06 | r0d3nt | <paste> |
| 02:32.07 | r0d3nt | tkcvoip*CLI> show uptime |
| 02:32.07 | r0d3nt | System uptime: 1 year, 2 weeks, 2 days, 9 hours, 40 minutes, 8 seconds |
| 02:32.07 | r0d3nt | Last reload: 47 weeks, 4 hours, 37 minutes, 10 seconds |
| 02:32.07 | r0d3nt | 18:10:54 up 381 days, 3:08, 5 users, load average: 2.05, 1.85, 1.20 |
| 02:32.13 | r0d3nt | </paste> |
| 02:32.15 | [TK]D-Fender | Bazinn : Simple search & replace by char. A=2, B=2, c=2,d=3, etc, and thens tring comp |
| 02:32.45 | joburg | impressive! |
| 02:33.01 | r0d3nt | thanks.... |
| 02:33.24 | r0d3nt | i only installed it... i owe it all to dell/gentoo/digium/asterisk |
| 02:33.42 | r0d3nt | but ya thanks.. i guess i did something right.... |
| 02:33.58 | Bazinn | 1850? |
| 02:34.31 | joburg | TKD : do you search and replace in the dialplan? |
| 02:34.40 | Bazinn | be AGI |
| 02:39.54 | lordbaron | Any Debug I could enable to determine why kewlstart is detecting a hangup? |
| 02:43.25 | lordbaron | this page seems to indicate there is a issue with this in australia..I am Texas USA: http://www.voip-info.org/wiki/index.php?page=Asterisk+Disconnect+Supervision |
| 02:43.41 | *** join/#asterisk _deg_ (n=deg@201-40-223-25.ctame704.dsl.brasiltelecom.net.br) |
| 02:50.35 | joburg | i agree the calprogress and busydetect did it for me |
| 02:51.24 | lordbaron | I just loaded the wctdm with debug=1. I get a polarity reversed 0 -> -1 message |
| 02:51.40 | lordbaron | does this mean kewlstart will not work? |
| 02:53.12 | *** join/#asterisk hads|home (n=hads@mail.nice.net.nz) |
| 02:53.47 | *** join/#asterisk Axklor (n=ollo@ultrasparc.l33t.net.au) |
| 02:55.32 | adelas | hey somebody have a reject sound file? |
| 02:55.39 | adelas | like your call has been rejected b/c ur a loser |
| 02:55.44 | adelas | or something like that lol? |
| 02:56.31 | adelas | or something like, this number has been disconncted because your a loser |
| 02:57.25 | joburg | simply create it yourself |
| 02:58.32 | Faithful | When my adsl disconnects/reconnects * cannot place calls with my IAX2 terminations, they dial but never connect... ideas? |
| 02:58.33 | adelas | i was looking for something like funny with effects heh |
| 02:58.52 | Faithful | I have to reboot my * box |
| 03:00.52 | adelas | oo |
| 03:00.54 | adelas | sweet found one |
| 03:00.55 | adelas | haha |
| 03:01.00 | adelas | i'll just call in record, and take for my use :P |
| 03:02.11 | [TK]D-Fender | joburg : i'D DO IT IN agi |
| 03:04.15 | joburg | where did you find it? |
| 03:05.04 | [shodan] | $(5) <= this would return the contents of the variable name 5, not "5" , right ? |
| 03:05.54 | joburg | ${5} you mean |
| 03:06.08 | [shodan] | oops. yes |
| 03:07.25 | [shodan] | in the book there is this example exten => 123,1,Set(TEST=example) |
| 03:07.26 | [shodan] | exten => 123,2,SayNumber(${LEN(${TEST})}) |
| 03:08.50 | [shodan] | it says this example would execute SayNumber(7) but is should be the same as SayNumber(${7}) right ? and since there isn't a variable named 7 , it would be the same as SayNumber() ? |
| 03:09.46 | joburg | nope to get it to say the number 7 , SayNumber(7) will do it |
| 03:12.41 | [shodan] | oops that is was a it |
| 03:13.07 | [shodan] | I mean exten => 123,2,SayNumber(${LEN(${TEST})}) is the same as exten => 123,2,SayNumber(${7}) and not exten => 123,2,SayNumber(7) |
| 03:14.09 | joburg | no ${LEN${TEST} = 7 therefore it's SayNumber(7) |
| 03:15.01 | *** join/#asterisk teknoprep (n=teknopre@unaffiliated/teknoprep) |
| 03:15.25 | [shodan] | shouldn't the example just say exten => 123,2,SayNumber(LEN(${TEST})) , why the extra ${} ? |
| 03:15.26 | teknoprep | what module needs to be loaded for call waiting? |
| 03:16.42 | joburg | because ${LEN} calculated the LENGHT of the value of ${TEST} which results in 7 |
| 03:18.34 | Qwell | LEN() is a function. functions are called with ${} |
| 03:19.40 | [shodan] | oh ok, you need the extra ${} to encapsulate another function it's not the same just ${name_of_a_variable} |
| 03:20.18 | joburg | yip |
| 03:20.38 | [shodan] | I thought the ${} was only to return the value of a variable |
| 03:21.08 | [shodan] | kinda confusing that the same operator has different meanings |
| 03:21.29 | hads|home | You are returning the value of the function. |
| 03:21.41 | *** join/#asterisk Coeus (n=Coeus@ip24-255-125-43.dc.dc.cox.net) |
| 03:22.43 | teknoprep | hmm |
| 03:22.47 | [shodan] | yeah, make sense |
| 03:25.49 | [shodan] | the book says "show functions" should return a list of available functions , but I get "No such command 'show functions'" is there another command the get the list, has it been moved ? |
| 03:25.53 | [shodan] | (using 1.0.11 btw) |
| 03:26.31 | joburg | time to upgrade.... |
| 03:27.07 | file | dialplan functions don't exist in 1.0.11 |
| 03:27.31 | [shodan] | so much for gentoo being "cutting edge" pfft ! :\ |
| 03:29.26 | [shodan] | any config file consideration in upgrading from 1.0. to 1.2. / will my config files mostly work with 1.2 ? (just found a masked 1.2.9 ebuild in portage) |
| 03:29.55 | joburg | yes configs needs no change |
| 03:30.29 | [shodan] | k, I'll upgrade right away then |
| 03:30.32 | [TK]D-Fender | joburg : Watch the double negatives. |
| 03:30.48 | Corydon76-home | They will work, but you should change some syntax, in preparation for the release of 1.4 |
| 03:30.51 | [TK]D-Fender | [shodan] They might need nothing at att, they may require a noticable rewrite... depends |
| 03:30.51 | joburg | double negatives? |
| 03:31.13 | Corydon76-home | Certain syntaxes are deprecated in 1.2 and will be removed in 1.4 |
| 03:31.16 | CunningPike | [shodan]: You'll get some 'deprecated' warnings in the CLI, but otherwise you should be OK. We went from 1.0.x to 1.2 with no problems |
| 03:31.19 | [TK]D-Fender | joburg : Actually... that wasn'ta double negative... |
| 03:31.28 | [TK]D-Fender | joburg : Just a wierd way of say what you wanted. |
| 03:31.40 | CunningPike | Greets, [TK]D-Fender |
| 03:31.48 | file | [TK]D-Fender: ! ! ! |
| 03:31.48 | [TK]D-Fender | CunningPike : y0 y0 y0 'sup! |
| 03:32.01 | [TK]D-Fender | file : I don't want relationship! |
| 03:32.13 | CunningPike | [TK]D-Fender: Not much - on vacation this week |
| 03:32.58 | joburg | ofcourse like SetVar will become Set etc.... |
| 03:33.47 | hads|home | There should be UPGRADE(.txt) or something in the 1.2 source to tell you what's changed etc. |
| 03:33.48 | joburg | nice change with the n priority from 1.0 to 1.2 ! |
| 03:34.50 | CunningPike | [shodan]: The real gotcha might be +101 jumping - make sure you have that enabled, if you used it in 1.0.x |
| 03:34.51 | [shodan] | k, anyway I have only the extensions.conf.example as my dialplan I'm in the middle of reading thebook to make my dialplan so I hope the rest didn't change too much |
| 03:35.36 | [TK]D-Fender | [shodan] : No big deal tos tart from scratch anyways.... |
| 03:35.42 | *** join/#asterisk Floodbar (n=Flood@ip72-192-124-29.ok.ok.cox.net) |
| 03:36.08 | [TK]D-Fender | CunningPike : Iwasted 1 week of mine, and plan on the Bahamas after christmas |
| 03:36.09 | Floodbar | is russel on |
| 03:36.28 | CunningPike | [TK]D-Fender: Bahamas! Nice |
| 03:36.44 | CunningPike | [TK]D-Fender: We have family visiting this week, from Ireland |
| 03:37.23 | [TK]D-Fender | CunningPike : Nice place when its peaceful... They like Canucks up there too :) |
| 03:37.30 | CunningPike | :D |
| 03:37.32 | joburg | was in Miami in November for the DCAP , yes the bahamas is chill |
| 03:37.56 | file | Bahamas... what a good idea |
| 03:38.08 | CunningPike | The next Astricon should be there :D |
| 03:38.18 | [TK]D-Fender | Even better since I'll be travelling off-season and staying with family that lives there :) |
| 03:38.28 | Floodbar | file maybe you could help me |
| 03:38.37 | file | Floodbar: if you ask a question, someone may answer |
| 03:38.48 | joburg | DCAP what a waste of money... |
| 03:38.54 | [TK]D-Fender | Ascii stupid question get a stupid ansi! ;) |
| 03:38.57 | file | dCAPitation! |
| 03:39.21 | CunningPike | [TK]D-Fender: Cool |
| 03:39.38 | Floodbar | I have put in a bug a few weeks ago on the queue and it still is asking for feedback but I have given them the information I was just wondering if it had been closed or have they found anything yet |
| 03:39.53 | [TK]D-Fender | Floodbar : Try looking at Mantis |
| 03:40.01 | Floodbar | okay |
| 03:40.10 | Floodbar | I don't want to be a pest |
| 03:40.13 | Floodbar | just checking |
| 03:40.19 | file | what bug? |
| 03:40.32 | Floodbar | queue not playing music on hold |
| 03:41.12 | [TK]D-Fender | Floodbar : MoH works everywhere else? |
| 03:41.45 | file | I have a vague recollection... first caller in queue not getting MOH? |
| 03:41.46 | Floodbar | yeah it just doesn't work for the first person in the queue if no agents are logged into the queue |
| 03:41.51 | Floodbar | right |
| 03:42.02 | file | what number? |
| 03:42.22 | file | ah found it |
| 03:42.59 | file | okay, I put it in my notes for tomorrow |
| 03:43.05 | Floodbar | thank you |
| 03:46.50 | adelas | oo this is pwnage now |
| 03:46.53 | adelas | rejection! |
| 03:46.53 | adelas | haha |
| 03:47.02 | adelas | i reject your call, u go to my reject hotline |
| 03:47.04 | adelas | how sweet is that |
| 03:47.11 | Floodbar | hah |
| 03:47.14 | file | s/sweet/dangerous/ |
| 03:47.49 | file | send them to a call center... like the student loan one that thinks I'm a girl and that calls my cellphone |
| 03:49.09 | file | (I'm not bitter at all after having called 3 times to have my number removed) |
| 03:50.41 | benjk | file, set PRI_CAUSE to 1 and hangup before picking up their call ;) |
| 03:50.55 | file | it's my cellphone ... |
| 03:51.31 | benjk | then you need to migrate the number first :) |
| 03:51.48 | file | this is Canada, no wireless number portability yet |
| 03:51.56 | benjk | ah, too bad |
| 03:51.59 | file | (totally lame) |
| 03:52.17 | CunningPike | file: 2007...... :) |
| 03:52.47 | benjk | we get number portability for mobile phones in 2 months |
| 03:52.55 | [shodan] | oh no , i just upgraded to asterisk 1.2.9_p1 zaptel 1.2.6 and speex 1.1.12 but now asterisk is crashing ! |
| 03:53.09 | file | [shodan]: you DID wipe out the modules directory, right? |
| 03:53.09 | [shodan] | <PROTECTED> |
| 03:53.09 | [shodan] | Aug 22 23:51:27 WARNING[26540]: loader.c:499 load_modules: Loading module chan_modem.so failed! |
| 03:53.14 | [shodan] | nope |
| 03:53.38 | [shodan] | I just naively run "emerge asterisk zaptel speex" ;) |
| 03:53.40 | file | you need to modify your /etc/asterisk/modules.conf as well and remove the lines that load the chan_modem stuff, it's gone |
| 03:53.48 | [shodan] | k |
| 03:54.26 | CunningPike | [shodan]: Why not just download the latest tarballs? |
| 03:54.58 | joburg | whats speex used for? |
| 03:55.12 | hads|home | the speex codec |
| 03:55.15 | [shodan] | I really prefer to use the package management system , maybe I could just modify the ebuild instead to get the very latest |
| 03:55.27 | CunningPike | joburg: Making your calls sound crappy ;) |
| 03:55.48 | *** join/#asterisk Winkie (n=urmom@cpc3-stre1-0-0-cust656.bagu.cable.ntl.com) |
| 03:56.02 | joburg | hehe |
| 03:56.31 | *** join/#asterisk lowlevel (n=Stuart@CPE0050ba71c82f-CM000f9f7d6742.cpe.net.cable.rogers.com) |
| 03:56.34 | AvoidingDeadlock | *PUNT* |
| 03:56.37 | joburg | so stay away ? |
| 03:56.53 | benjk | stay away from what? |
| 03:57.02 | file | radioactive materials |
| 03:57.07 | joburg | stay away from speex? |
| 03:57.20 | CunningPike | joburg: I'm not qualified to comment, to be honest - I've never used it, but I believe it's not that great |
| 03:57.26 | [shodan] | what was that chan modem stuff ? for using voicemodems as fxos ? |
| 03:57.31 | CunningPike | s/believe/have read/ |
| 03:57.55 | benjk | speex is cool |
| 03:57.56 | joburg | hehe |
| 03:58.04 | hads|home | It requires a lot of CPU from memory? |
| 03:58.07 | benjk | use it whenever you can in place of g729 |
| 03:58.16 | joburg | benjk : elaborate |
| 03:58.23 | CunningPike | benjk: It's good, then? |
| 03:58.46 | benjk | it uses about the same amount of resources g729 does and its free and open, no patents |
| 03:59.15 | benjk | and it sounds at least as good as g729, probably better |
| 04:00.11 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 04:00.23 | joburg | hwtas the badiwdth usage like? |
| 04:00.26 | benjk | codecs shouldn't be patentable anyway |
| 04:00.37 | benjk | they are nothing other than mathematical formulas |
| 04:00.52 | joburg | speex: what's the bandwidth usage like? |
| 04:00.55 | benjk | and mathematical formulas are explicitly excluded from patentability |
| 04:01.08 | AvoidingDeadlock | speex is a CPU WHORE |
| 04:01.16 | benjk | not anymore than g729 is |
| 04:01.22 | AvoidingDeadlock | actually WRONG |
| 04:01.32 | AvoidingDeadlock | Speex takes 10% of a dual core 2ghz opteron per call |
| 04:01.34 | AvoidingDeadlock | g729 doesn't |
| 04:01.44 | benjk | the cost on my system is exactly the same |
| 04:01.47 | joburg | outch! |
| 04:01.52 | benjk | show translation |
| 04:02.06 | AvoidingDeadlock | I'm not talking in the context of Asterisk |
| 04:02.17 | AvoidingDeadlock | and I'm also talking WideBand :P |
| 04:02.19 | joburg | benjk : can u send us a show translation ? |
| 04:02.38 | AvoidingDeadlock | show translation is about as accurate as the coding style of the whole project |
| 04:03.02 | *** join/#asterisk cmurphy (n=cmurphy@24-155-147-68.ip.grandenetworks.net) |
| 04:03.04 | joburg | the console don't lie |
| 04:03.14 | AvoidingDeadlock | riiight |
| 04:03.29 | benjk | on this system I am on here I don't have g729, but I was doing comparisons on three different boxes with a bunch of codecs incl g729 earlier this year |
| 04:04.35 | benjk | I didn't test Speex with wideband |
| 04:04.52 | joburg | AvoidingDeadlock : what do u use for showing cpu usage ? |
| 04:05.08 | benjk | also there is a bunch of optimisations you can do |
| 04:05.11 | lordbaron | having given up on KewlStart, I am trying to get LoopStart to detect hangup. I have Busydetect=yes, BusyCount=6, and BusyPattern=100,100. Doesn't work |
| 04:05.25 | lordbaron | I measured the busytones, and they are .1 sec of sound, with .1 sec of silence |
| 04:05.34 | AvoidingDeadlock | top ;) |
| 04:06.08 | lordbaron | callprogress=no |
| 04:06.27 | benjk | lordbaron, you can waste man months on that and never get it right |
| 04:06.37 | joburg | AvoidingDeadlock : how do you isolate the usage of speex in top ? |
| 04:06.43 | benjk | often it is dependent on the exchange you are connected to |
| 04:06.43 | AvoidingDeadlock | its only 1 CALL up |
| 04:07.02 | lordbaron | benjk: :> I can account for 48 hours so far |
| 04:07.15 | benjk | joburg, he already said he didn't talk about speex in the context of asterisk |
| 04:07.28 | benjk | and he was using it with wideband audio as well |
| 04:07.36 | lordbaron | benjk: so what is the best real-world answer for this? (besides get a pri) |
| 04:07.44 | joburg | i see... |
| 04:08.07 | joburg | whats wideband audio? |
| 04:08.16 | AvoidingDeadlock | something asterisk only wishes it could do |
| 04:08.33 | benjk | I don't know if there are best answers, but I personally concluded that I should either shut down my business or go all digital and make customers not ever want to use analog |
| 04:09.05 | joburg | benjk : the bottom line remains - stay away from analogue ... |
| 04:09.27 | benjk | analog has turned out to be one of those things that when you're done you reflect upon the sense of it all and find that you might have earned more money if you had taken an job at McDonalds |
| 04:09.43 | joburg | i only use it to interface with existing analogue pbx's - cause you control both sides ... |
| 04:09.57 | joburg | lol |
| 04:10.24 | lordbaron | joburg: I can confirm..that part of this connection works on this pbx..the Telco side is the issue |
| 04:10.43 | *** join/#asterisk JunK-Y (n=junky@modemcable205.175-81-70.mc.videotron.ca) |
| 04:10.48 | lordbaron | we are going to nortel for 3 lines, and those are fine, with KewlStart |
| 04:10.51 | benjk | but hey, YMMV, maybe you get lucky and get a reasonable result before wasting that much time |
| 04:11.11 | benjk | analog is like gambling |
| 04:11.22 | benjk | sometimes you win, sometimes you lose |
| 04:11.36 | *** join/#asterisk barspi (n=barspi@r200-125-54-2-dialup.adsl.anteldata.net.uy) |
| 04:11.37 | lordbaron | any experience with swbell/at&t not providing disconnect supervision? |
| 04:11.38 | benjk | but you can't beat the casino |
| 04:11.45 | hads|home | I wouldn't call it gambling. I have several systems which have no troubles at all. |
| 04:11.56 | benjk | lucky streak |
| 04:12.14 | hads|home | Not at all. |
| 04:12.22 | barspi | hi.. anyone know what's fw2h.c in the new zaptel? |
| 04:12.37 | *** join/#asterisk Strom_C (n=strom@fl-65-41-146-225.sta.embarqhsd.net) |
| 04:12.48 | hads|home | As long as you have disconnect supervision then you shouldn |
| 04:12.54 | hads|home | 't have trouble |
| 04:13.00 | benjk | I can take you to an area in Tokyo which is served by an exchange where you never ever get analog working no matter what you do |
| 04:13.24 | hads|home | I'm not in Tokyo. |
| 04:13.27 | Strom_C | benjk, is it a crossbar switch? :) |
| 04:13.34 | benjk | you move one street out of that area where a different exchange server the line and you can get reasonable results |
| 04:13.55 | *** join/#asterisk Nukemizer (n=Nuke@160.7.239.13) |
| 04:14.09 | CunningPike | lordbaron: Where are you located? |
| 04:14.16 | benjk | probably NEC, most of NTT's stuff is NEC |
| 04:14.27 | joburg | try south africa : we have a monopoly and guess what else - asterisk is illegal in south africa ! |
| 04:14.30 | lordbaron | CunningPike: Fort Worth, TX |
| 04:14.44 | *** part/#asterisk cmurphy (n=cmurphy@24-155-147-68.ip.grandenetworks.net) |
| 04:14.58 | CunningPike | lordbaron: See if a local telco with do partial PRI |
| 04:15.00 | Strom_C | joburg, Telkom doesn't play nice? there's a surprise :) |
| 04:15.11 | benjk | the trouble is that that area is the financial services district |
| 04:15.35 | benjk | and I have come across things like that in many other countries |
| 04:15.38 | joburg | in some of our neibouring countries like botswana , voip is illegal ! |
| 04:15.47 | benjk | asterisk != voip |
| 04:15.49 | joburg | oh you know about Telkom ? |
| 04:15.53 | lordbaron | CunningPike: Ok, good idea. Thanks |
| 04:16.08 | Strom_C | joburg, I've got family in south africa |
| 04:16.14 | CunningPike | lordbaron: We had one with 3 channels from Allstream |
| 04:16.24 | joburg | Strom_C : how so ? |
| 04:16.28 | benjk | lordbaron, they may not recognise partial PRI, try "fractional PRI" |
| 04:16.45 | Strom_C | joburg, mainly around cape town |
| 04:17.04 | lordbaron | CunningPike: is the cost comparable to pots * 3 for the 3 channels? |
| 04:17.07 | joburg | Strom_C : cape town is magnificent ! |
| 04:17.24 | L|NUX | hello every one |
| 04:17.27 | benjk | depends on the telco and location, in some places ISDN is actually cheaper |
| 04:17.28 | L|NUX | i have little question |
| 04:17.38 | benjk | in some its about the same, in some its more expensive |
| 04:17.47 | lordbaron | figures |
| 04:17.48 | CunningPike | lordbaron: Not quite sure - a full PRI is about US$800 |
| 04:17.54 | benjk | you will only know if you ask |
| 04:18.06 | CunningPike | lordbaron: For us - ymmv |
| 04:18.28 | lordbaron | anyone ever use Lightyear PRI? |
| 04:18.37 | joburg | what's "ymmv" |
| 04:18.39 | lordbaron | the resell MCI and others |
| 04:18.45 | benjk | and there may be a competitve telco whtat might be much more affordable than the one you use now |
| 04:18.45 | lordbaron | joburg: your mileage may vary |
| 04:18.52 | L|NUX | i am using svn trunk and its working good but when i forward my call from another * server which is using stable version 1.2.x then it will some time give me message on svn version Host 192.168.129.11 failed to authenticate as super |
| 04:19.03 | L|NUX | what will be the possible cause :) |
| 04:19.08 | joburg | thanks |
| 04:19.19 | L|NUX | and after two three tries it will work |
| 04:19.35 | CunningPike | ymmv is 'your mileage may vary' where I am - ymmv elsewhere though....... |
| 04:19.46 | Strom_C | ymmv on ymmv? |
| 04:19.48 | CunningPike | ~ymmv |
| 04:19.52 | jbot | rumour has it, ymmv is Your Mileage May Vary |
| 04:20.33 | benjk | young masochist male veterinary |
| 04:20.41 | *** join/#asterisk barspi (n=barspi@r200-125-54-2-dialup.adsl.anteldata.net.uy) |
| 04:20.50 | CunningPike | L|NUX: Authentication problem :) |
| 04:21.09 | CunningPike | L|NUX: And the pipe in your nick is a PITA ;) |
| 04:21.46 | L|NUX | CunningPike : man its not authentication problem because it work some time |
| 04:22.02 | L|NUX | and i have checked my username and password |
| 04:22.49 | *** join/#asterisk _mwoodj_ (n=mwoodj@pdpc/sponsor/digium/hyper-eye) |
| 04:22.51 | CunningPike | L|NUX: Did you try a 'sip debug'? |
| 04:23.29 | benjk | how do you know he's using SIP? |
| 04:23.39 | benjk | sixth sense? |
| 04:23.50 | AvoidingDeadlock | http://video.google.com/videoplay?docid=-4613750174577358330&q=cluecon |
| 04:23.53 | benjk | his question didn't give any details |
| 04:23.53 | AvoidingDeadlock | woops wrong window |
| 04:24.17 | L|NUX | CunningPike : let me try |
| 04:24.25 | CunningPike | benjk: Good point. I think everyone in the world is exactly like me :D |
| 04:24.28 | joburg | IAX is best fro connecting * servers |
| 04:25.50 | L|NUX | benjk : i am connecting server using IAX |
| 04:26.04 | L|NUX | let me explain |
| 04:27.03 | L|NUX | Server A have extension 11 and if some one call on it will dial on Server B which have 11 and it will ring to my phone but when i dial it i will get Host 192.168.129.11 failed to authenticate as super message on Server B |
| 04:27.10 | benjk | so much for your mind reading abilities CunningPike :D |
| 04:27.43 | CunningPike | Guess that explains why I haven't won the Lottery yet..... |
| 04:27.49 | benjk | heh |
| 04:27.57 | benjk | do you play the lottery |
| 04:28.06 | CunningPike | I run a syndicate at work |
| 04:28.51 | benjk | more trouble? |
| 04:28.55 | benjk | :D |
| 04:29.00 | joburg | L|NUX : are u using md5 / rsa ? |
| 04:29.43 | CunningPike | ~lart benjk |
| 04:29.47 | CunningPike | See? |
| 04:29.49 | CunningPike | ;) |
| 04:30.01 | benjk | I personally think I left all the trouble behind by abandoning everything to do with analog |
| 04:30.08 | *** join/#asterisk xachen (i=justin@pdpc/supporter/student/xachen) |
| 04:30.14 | CunningPike | benjk: Good call |
| 04:30.22 | L|NUX | joburg : nope |
| 04:30.29 | L|NUX | joburg : just plain text |
| 04:32.45 | joburg | try using md5 , at least it's more secure , and might give you something else in debug , giving you more to work with |
| 04:33.10 | L|NUX | okies |
| 04:33.30 | joburg | remember to add on both sides ! |
| 04:35.40 | L|NUX | ok |
| 04:41.25 | *** join/#asterisk I-MOD (i=opticron@c-71-207-209-230.hsd1.al.comcast.net) |
| 04:41.49 | *** join/#asterisk mmurdock (n=vircuser@c-24-10-190-87.hsd1.ut.comcast.net) |
| 04:41.57 | mmurdock | howdy all. |
| 04:42.46 | *** join/#asterisk Strom_C (n=strom@fl-65-41-146-225.sta.embarqhsd.net) |
| 04:43.07 | Strom_C | this hotel's wifi really sucks |
| 04:43.31 | file | Strom_C: I'm blocking the signal. |
| 04:43.43 | Strom_C | oh, so it's all your fault then |
| 04:43.50 | file | yup |
| 04:44.18 | denon | better? |
| 04:44.19 | mmurdock | I've got a quick parking question. When I park a call I am not hearing which extension the call is being parked on. Any suggestions why? The call is being parked. |
| 04:44.42 | Strom_C | mmurdock, you've got to do an attended transfer |
| 04:44.47 | denon | mmurdock: are you doing a blind transfer? you shouldnt bre |
| 04:44.48 | denon | be |
| 04:45.12 | mmurdock | denon: I dont' belive so, I'm just hitting the transfer button and then 70. |
| 04:45.18 | denon | what kinda phone? |
| 04:45.24 | mmurdock | Snom 320 |
| 04:45.30 | denon | ah, dunno on snom |
| 04:45.38 | denon | do you have an option for an attended transfer? |
| 04:45.55 | mmurdock | When watching the the asterisk console I see it try to say the number seve. |
| 04:45.58 | mmurdock | seven. |
| 04:46.21 | mmurdock | now that's interesting. |
| 04:46.40 | mmurdock | If I hit # it says transfer and then I dail 70 it then tells me the extension. |
| 04:46.49 | denon | well .. |
| 04:46.56 | denon | that's a ... not a native sip way of doing it |
| 04:47.03 | denon | it works, but its kinda crappy |
| 04:47.07 | mmurdock | mmm. |
| 04:47.21 | denon | your phone should have an attended transfer option |
| 04:47.26 | denon | it sounds like it's doing a blind xfer by default |
| 04:47.42 | mmurdock | denon: yea. |
| 04:47.55 | mmurdock | Would the consol tell me what type of transfer it was doing? |
| 04:48.07 | denon | nope |
| 04:48.08 | denon | but .. |
| 04:48.14 | denon | when you transfer a call to another person .. |
| 04:48.19 | denon | do you talk to the other person first? |
| 04:48.23 | denon | before you send the caller to them |
| 04:48.27 | denon | or does the call just disappear |
| 04:49.26 | mmurdock | I hit transfer and then the extension and then it disappears. |
| 04:49.36 | denon | ok, so its blind |
| 04:49.39 | denon | (nothing to do with call parking) |
| 04:49.53 | mmurdock | mm. off to the config file. |
| 04:49.53 | denon | google your phone model for attended transfer, hit the manual, or use # :) |
| 04:50.20 | joburg | see features.conf for transfer options ... |
| 04:51.02 | joburg | benjk : still around ? |
| 04:51.55 | benjk | sure |
| 04:52.42 | *** join/#asterisk celophane (n=e@ip68-104-251-230.ph.ph.cox.net) |
| 04:52.52 | *** join/#asterisk jets (n=jets@root.ownsu.com) |
| 04:52.55 | joburg | benjk : just remember something : check indications.conf for country options to work with busydetect etc ... |
| 04:53.30 | benjk | nah, trust me, I have been there - done that, its not that simple |
| 04:53.34 | Un1x | benjk you know that email you gave me asteriagi.com i emailed them they didn't respond been like 4 days... |
| 04:54.00 | benjk | its asteriasgi.com |
| 04:54.08 | *** join/#asterisk bhrobinson (n=brobinso@mail1.nt-it.com) |
| 04:54.16 | *** join/#asterisk pbx1 (i=pbx1@netblock-66-245-193-236.dslextreme.com) |
| 04:54.40 | Un1x | ya benjk i got the email i copy/pasted it from whaty you gave me i even checked the site i got the right email |
| 04:54.42 | joburg | benjh : we are using the [za] for south africa and we still have to minupulate settings for different regions of south africa - cause the telco use diff exchanges |
| 04:54.44 | bhrobinson | anyone here a T-1 expert? |
| 04:54.45 | Un1x | im just saying they have not replyed. |
| 04:54.50 | celophane | Hello! We just purchased Asterisk Business Edition from Digium... Our original phone vendor flaked out. One of our techs who is very familiar with Linux and VoIP is taking over for him. This might be a question he's better suited to ask, but I want to try to find out as much as I can... |
| 04:55.14 | Strom_C | bhrobinson, what do you need to know? |
| 04:55.54 | bhrobinson | I have a TE210P working fine on the Asterisk. In port 1, I have the zaptel using channels 1-5 for voice and 24 for data |
| 04:56.00 | joburg | have no idea why anyone would purchase software that's for free ? |
| 04:56.25 | denon | joburg: its not identical, obviously |
| 04:56.31 | SwK | un1x what up? |
| 04:56.37 | bhrobinson | for port 2, I want to hook it to my existing phone system. Is it better to connect all 23 channels, or only the 5 I need for outbound? |
| 04:56.42 | SwK | msg me about your asteria issue? |
| 04:56.56 | Strom_C | bhrobinson, are you doing CAS T1 or PRI? |
| 04:56.57 | celophane | Our previous vendor was going to have us set up the server at a data center that has fiber running into it. We're putting everything in our own rack. We have two offices that are going to be connecting to the phone server from here in town (Phoenix area). He was supposed to have ordered a PRI line. I'm not sure what roll he was planning on that PRI line taking. Something about it being for local lines? |
| 04:57.01 | joburg | denon : u r correct the free one is more up-to-date |
| 04:57.11 | bhrobinson | StromC, PRI |
| 04:57.23 | benjk | Un1x, talk to SwK |
| 04:57.26 | denon | joburg: "up to date" doesnt mean it's been more stress-tested |
| 04:57.29 | Un1x | heh yea tlaking now :) |
| 04:57.30 | Strom_C | bhrobinson, do you have a good reason for /not/ connecting all 23 B-channels? |
| 04:57.35 | benjk | ok, good |
| 04:57.49 | denon | joburg: it also comes with a year of support |
| 04:57.59 | SwK | thanks ben |
| 04:58.03 | benjk | welcome |
| 04:58.04 | bhrobinson | StromC, on the Adit to Asterisk, or the Asterisk to Samsung phone switch? |
| 04:58.07 | joburg | yippee ! |
| 04:58.54 | denon | joburg: it also has the option for native cepstral, and speech recognition modules |
| 04:59.09 | *** join/#asterisk Strom_C (n=strom@fl-65-41-146-225.sta.embarqhsd.net) |
| 04:59.11 | denon | you've gotta do a fair bit of hacking to get those to work on regular asterisk |
| 04:59.13 | wunderkin | celophane, yes.. normally.. i have one for long distance though.. im local, if you need help |
| 04:59.19 | bhrobinson | StromC, on the Adit to Asterisk, or the Asterisk to Samsung phone switch? |
| 04:59.33 | Strom_C | bhrobinson, whichever one is on span 2, like you said |
| 04:59.40 | benjk | you gotta do a fair bit of hacking to get regular asterisk to work properly anyway :) |
| 04:59.43 | denon | joburg: besides, to many businesses, a supported product, and well-tested/etc is more important than "look, its free!" |
| 04:59.43 | Strom_C | the one that goes to your existing system |
| 04:59.53 | bhrobinson | strom_c,no reason |
| 05:00.00 | benjk | which you cant if you use ABE cause as I understand it doesn't come with sources |
| 05:00.11 | benjk | so you're more likely to get stuck |
| 05:00.12 | Strom_C | bhrobinson, so just provision all 23B + D and avoid headaches |
| 05:00.34 | bhrobinson | ok |
| 05:00.43 | celophane | He also mentioned that we needed a TDM card...I looked those up, they appear to be analog cards. Isn't a PRI line digital lines? Why would we need an analog card for digital lines? The PRI line is not in. He keeps telling me it should be installed any time. He also mentioned that we could use Vonage until the PRI line is installed... This seem like horsesh*t. Vonage sucks... |
| 05:00.47 | benjk | well testes, LOL |
| 05:00.52 | bhrobinson | strom_c, done |
| 05:00.55 | Strom_C | remember....simpler solution is better |
| 05:00.57 | benjk | s/testes/tested |
| 05:01.05 | bhrobinson | now how do I need to put it in the zapata file |
| 05:01.23 | Un1x | SwK: is alseep i think lol |
| 05:01.31 | benjk | don't use Vonage |
| 05:01.48 | denon | vonage is silly in a business environment |
| 05:01.59 | denon | there are so many ip carriers that are so much more flexible |
| 05:02.04 | benjk | and try to avoid any analog stuff like the plague if you can |
| 05:02.13 | denon | and if you plan to seriously abuse vonage, they'll close your account anyway |
| 05:02.27 | denon | "unlimited" isnt really what you think |
| 05:02.28 | CunningPike | celophane: Digium's TDM cards aren't TDM cards :D I have no idea why they called them that. For a PRI, you need a TE card like the TE110P |
| 05:02.52 | benjk | for most ITSPs unlimited calls means 1000 or 1500 minutes per month |
| 05:02.53 | file | ugh that annoys me too |
| 05:03.16 | denon | file: wussat? "unlimited"? |
| 05:03.26 | file | denon: unlimited and the naming of the analog boards |
| 05:03.35 | bhrobinson | cunningpike, you might also want to remind then that those cards are 3.3v too... I was a little upset when I noticed that :) |
| 05:03.35 | denon | limited unlimited is nothin new, you guys remember the days of dialup |
| 05:03.43 | benjk | unlimited is a misnomer, it should be "unmetered" |
| 05:03.53 | denon | benjk: or "high cap" |
| 05:04.03 | benjk | and then qualified with "up to X mins per Y" |
| 05:04.04 | bhrobinson | strom_c, what do I need to do with the zapata.conf? |
| 05:04.14 | CunningPike | bhrobinson: afaik, there are 2 versions of each card - one does only 3.3 and the other does both - or something like that |
| 05:04.19 | Strom_C | bhrobinson, provision the full 23 b-channels |
| 05:04.21 | joburg | Sangoma T1 cards are the absolute best ! |
| 05:04.31 | denon | heh, if you dont mind the wanpipe overhead |
| 05:04.47 | Strom_C | yeah, or kludging zaptel |
| 05:04.55 | bhrobinson | cunningpike, I did not know that. I had to go buy a new machine |
| 05:04.57 | joburg | don't mind it at all |
| 05:04.57 | benjk | since you are from South Africa, I knew you would say that ;) |
| 05:05.23 | Strom_C | I like Mrs. H. S. Ball's chutney |
| 05:05.32 | benjk | craptel is not nice |
| 05:05.35 | Strom_C | that comes from south africa :) |
| 05:06.07 | joburg | Yes Digium and E1 is a real mess ... |
| 05:06.12 | bhrobinson | my 2 cents on the digium, the built in echo cancellation routines are intesive to the procs of the box, therefore limiting you to no more than a few hundred simulatious connections |
| 05:06.41 | CunningPike | bhrobinson: 410P is 3.3V - 405P is 5V |
| 05:07.25 | bhrobinson | cunningpike, that is more what I thought... I was pretty sure the 210 was only 3.3... and what a kick in the teeth that was <grin> |
| 05:07.27 | CunningPike | ~seen dlynes_laptop |
| 05:07.30 | jbot | dlynes_laptop <n=dlynes@S01060016b6c052ee.vc.shawcable.net> was last seen on IRC in channel #asterisk, 8d 7h 17m 37s ago, saying: 'Good afternoon, cp'. |
| 05:07.39 | benjk | hopefully, Sangoma will get rid of craptel soon |
| 05:07.55 | CunningPike | bhrobinson: :) |
| 05:08.01 | bhrobinson | cunningpike, you were the guy that helped me all day saturday, right? |
| 05:08.03 | benjk | does anybody know the status on that? bkw? |
| 05:08.22 | denon | benjk: sangoma's always going to be a step behind.. cant really be bleeding edge when you have nothing to do with dev |
| 05:08.38 | CunningPike | bhrobinson: Um, not sure :) That was a long time ago - and I'm on vacation. What was your problem? |
| 05:08.58 | benjk | once they get rid of craptel as a layer in their software, theirs will really be nice |
| 05:09.32 | bhrobinson | cunningpike, just wanted to thank you if it was. I could not get the 210 to sync to the T-1 |
| 05:09.50 | *** join/#asterisk rvhi (n=rv@66.175.65.89) |
| 05:09.55 | CunningPike | bhrobinson: Doesn't ring a bell......... but you're welcome anyway :D |
| 05:10.35 | hads|home | benjk: Stop trolling |
| 05:10.43 | benjk | no trolling |
| 05:10.53 | bhrobinson | cunningpike, are you in new zealand? |
| 05:11.02 | hads|home | "craptel" - I'd call that trolling |
| 05:12.02 | benjk | craptel is so bad that Linus Thorvalds was adamant to make sure it will not get into the linux kernel |
| 05:12.34 | benjk | apparently he said something like he didn't want any such crap in his kernel |
| 05:12.36 | denon | benjk: if that was true, he should probably spend his time elsewhere -- there's a lot worse crap in the linux kernel |
| 05:13.02 | Strom_C | "thorvalds"? |
| 05:13.19 | benjk | did I misspell it? |
| 05:13.25 | Strom_C | you lose |
| 05:13.29 | denon | Torvalds. |
| 05:13.34 | benjk | oh well |
| 05:13.45 | CunningPike | That's Mr. Torvalds |
| 05:14.12 | benjk | doesn't change anything about the messiness of craptel |
| 05:15.08 | joburg | craptel = captel ? |
| 05:15.31 | denon | benjk: http://www.ussg.iu.edu/hypermail/linux/kernel/0207.2/1372.html |
| 05:15.36 | denon | is that the thread you're referring to? |
| 05:15.49 | benjk | dunno, I got it second hand |
| 05:16.05 | hads|home | So you're trolling with second hand info eh? |
| 05:16.17 | denon | probably better not quote it unless you know for sure |
| 05:16.56 | joburg | seeya |
| 05:16.57 | benjk | I am not trolling |
| 05:17.00 | *** part/#asterisk joburg (n=voipmagi@vc-196-207-36-133.3g.vodacom.co.za) |
| 05:17.04 | Qwell | denon: heh, that isn't even the same thread |
| 05:17.10 | *** join/#asterisk VAXpirate (i=arab@12-240-51-111.client.mchsi.com) |
| 05:17.13 | VAXpirate | hi. *waves* |
| 05:17.18 | benjk | I probably have looked into craptel more intensively that you |
| 05:17.35 | benjk | we have been trying to get a contractor to port the damn stuff |
| 05:17.52 | benjk | nobody wanted to touch it unless paid shameless amounts of money |
| 05:18.16 | benjk | whilst quotations for porting alternative solutions would come in at about 10-15% |
| 05:18.46 | benjk | and everybody was complaining about craptel, how it was structured (or lack of structure) and so forth |
| 05:18.47 | denon | Qwell: whoops, wrong link - but you see the one I meant |
| 05:19.12 | hads|home | benjk: Fantastic! |
| 05:19.19 | denon | benjk: people complain about everything, but I don't think you should say Linus personally called zaptel "crap" unless you can quote it |
| 05:19.42 | benjk | developers in various countries who make a living of porting drivers |
| 05:19.48 | benjk | not just "people" |
| 05:19.55 | hads|home | benjk: Lovely. |
| 05:20.02 | denon | benjk: so fix it |
| 05:20.07 | benjk | nope |
| 05:20.11 | benjk | replace it |
| 05:20.16 | denon | that's the beauty of open source, nobody's allowed to whine |
| 05:20.41 | denon | go ahead then, write something better, submit a patch - I bet digium would throw zaptel out if they had something better completed |
| 05:21.19 | benjk | Unicall is a damn fine professionally designed architecture and eventually the lowest layer won't be zap, that'll replace the damn thing |
| 05:21.20 | *** join/#asterisk [hC] (n=hardcore@S01060004e21ea953.vc.shawcable.net) |
| 05:21.51 | *** join/#asterisk operat0r (n=h0msar@adsl-19-78-76.asm.bellsouth.net) |
| 05:23.01 | operat0r | ok so with http://www.yellowpages.com/sp/yellowpages/ypout.jsp?linkType=2&outURL=http%3A//smartpages.yp.ingenio.com/Listings/Action.aspx%3FCustomer%3D466249038%26Listing%3D318316994%26Appearance%3D318316994002654001%26Phone%3D6784504044%26Directory%3D314310%26Heading%3D8010077%26Tier%3D30%26SC%3DfxfiEvFYCmjoN7v46CEGGw%253D%253D%26Adver%3Dhttp%253A%252F%252Fwww.isconsulting-inc.com&impressionId=87&listingId=56207574 |
| 05:23.23 | Strom_C | holy catse |
| 05:23.32 | operat0r | if I forward that call to a lan line the call is free both ways right ? |
| 05:23.33 | Strom_C | have you not heard of tinyurl? |
| 05:24.12 | operat0r | Strom_C yes sorry did not know the url was that long |
| 05:24.26 | operat0r | the idea is to use that service to make a free call |
| 05:25.05 | *** join/#asterisk danalien (n=danalien@unaffiliated/danalien) |
| 05:25.29 | operat0r | you can seup the account for free on yellowpages and it does not appear to charge me with I forward the call to a LAN line |
| 05:25.47 | Strom_C | what is a "LAN line"? |
| 05:26.03 | Qwell | Strom_C: stop cutting phone lines |
| 05:26.12 | Qwell | silly vandal |
| 05:26.14 | benjk | denon, Mark has been trying to get his hands on Unicall for a long time, but Steve won't move off the GPL so Mark is stuck with the mess of the zap chain |
| 05:26.41 | denon | benjk: well then it's no better than Nortel having a better solution, and not wanting to share, is it? |
| 05:26.51 | benjk | are you kidding me? |
| 05:26.54 | denon | you're not forced to use asteisk - use something else |
| 05:27.00 | benjk | its GPL |
| 05:27.01 | denon | if you want it in asterisk, talk to steve |
| 05:27.07 | benjk | not sure if you heard of the GPL |
| 05:27.16 | denon | asterisk's licensing requirements have been around for a very long time |
| 05:27.21 | benjk | I don't even have to talk to Steve |
| 05:27.26 | benjk | I can just download it |
| 05:27.33 | Qwell | So then just download it |
| 05:27.35 | benjk | and run make ; make install |
| 05:27.36 | denon | I'm sorry, I've forgotten .. why are you whining again? |
| 05:27.46 | *** part/#asterisk Axklor (n=ollo@ultrasparc.l33t.net.au) |
| 05:27.49 | benjk | I am not whining |
| 05:27.55 | benjk | nor am I trolling |
| 05:27.56 | hads|home | Yes, yes you are. |
| 05:27.59 | Qwell | Good, then we'll move on |
| 05:28.03 | Juggie | * is also gpl |
| 05:28.08 | operat0r | http://0pencircuit.net/t0c/index.php?topic=171 So far so good I check my cell bill and fwd is free |
| 05:28.21 | Juggie | the problem isnt gpl, its just that i assume steve wont give him ownership permission to distro it and sell it. |
| 05:28.39 | Qwell | Juggie: No, he can still sell it and such if it's disclaimed |
| 05:28.42 | Qwell | That isn't the issue. :) |
| 05:28.46 | Juggie | no he cant. |
| 05:28.57 | benjk | I was just pointing at something that shows how things can be done right and how the zap chain isn't |
| 05:28.59 | Juggie | in abe, * isnt under the gpl. |
| 05:29.05 | Juggie | its under a commercial license |
| 05:29.11 | *** join/#asterisk Strom_C (n=strom@fl-65-41-146-225.sta.embarqhsd.net) |
| 05:29.12 | benjk | since you accused me of not being able to back up what I said |
| 05:29.18 | Juggie | so theres no way to change unicall from gpl -> commercial license |
| 05:29.23 | benjk | Unicall is proof that you can do thing proper |
| 05:29.30 | Juggie | without ownership of the code |
| 05:29.37 | denon | benjk: I didnt accuse you of anything, I just said you should ensure you have proper backing on what you do say |
| 05:29.40 | benjk | that's all there is to it, delivering proof |
| 05:29.49 | benjk | no whining, no trolling, just proof |
| 05:29.58 | Strom_C | Trolling...delivered! |
| 05:30.00 | CunningPike | Is this helping anyone run asterisk? |
| 05:30.07 | Juggie | thats why when you submit patches you disclaim everything to digium, since they own the code, they can license it however they want, what they cant do however is take someone elses GPL code and change the license, which is the problem w/ unicall. |
| 05:30.18 | benjk | of course, many folks run Unicall with Asterisk |
| 05:30.35 | Qwell | enough |
| 05:30.38 | benjk | its not a patch either |
| 05:30.45 | Qwell | This is completely non productive |
| 05:30.53 | denon | agreed |
| 05:30.55 | denon | next |
| 05:31.05 | Juggie | none the less, digium could distro it if they wanted to, but not as part of BE. |
| 05:31.19 | Juggie | because they cannot change the license, so thats the problem, case closed. |
| 05:31.38 | benjk | Asterisk is a toolkit, it says so at the top of every source file |
| 05:32.00 | benjk | so you can use some of the tools in that toolbox and mix it with tools from another toolbox |
| 05:32.04 | *** part/#asterisk hads|home (n=hads@mail.nice.net.nz) |
| 05:32.48 | [hC] | The biggest problem with licensing is so many people think they know how it works, and operate accordingly, yet they were misinformed. Especially with the GPL |
| 05:32.48 | [hC] | There was something productive that came out of that whole spew up there... |
| 05:32.52 | [hC] | I'd never heard of unicall before. |
| 05:32.52 | benjk | why people get excited if you choose a different screwdriver than the one that came with some kit you got, escapes me |
| 05:33.11 | Juggie | i've never used unicall, but i've heard of it |
| 05:33.19 | [hC] | I wonder what makes unicall any better, and if my sangoma cards work w/ it? |
| 05:33.25 | CunningPike | #unicall, #gpl, #enoughalready |
| 05:33.35 | benjk | there is no #unicall |
| 05:33.43 | [hC] | Personally, I could give a crap on what's licensed as what, I just want to build the best system. |
| 05:33.46 | CunningPike | #benjk |
| 05:33.52 | [hC] | benjk: it was a joke. |
| 05:33.53 | benjk | its a library for softswitches like asterisk |
| 05:34.10 | [hC] | Yeah, I gathered that. |
| 05:34.23 | benjk | you don't tell anyone to have sex and travel if they are interested in H323 or OH323, so you? |
| 05:34.57 | denon | the heck? |
| 05:35.03 | denon | ok, really, that's enough. |
| 05:35.04 | benjk | and mind you, there are some libraries that people use with asterisk without controversy even though those libs are also GPL only |
| 05:35.09 | *** join/#asterisk achandra (n=achandra@static-71-103-255-118.lsanca.dsl-w.verizon.net) |
| 05:35.44 | benjk | so the controversy here is simply that Unicall challenges the zap chain because it shows how you can do a much better architecture |
| 05:36.00 | Juggie | does unicall support all zaptel hardware? |
| 05:36.17 | achandra | hello. all... Hey guys ive been playing with the ideas of Virtual Machines...and using XEN or VMWARE...any luck getting it working...assuming...I have VT enabled in bios and also seperate dedicated NICS? |
| 05:36.20 | denon | benjk: I think it's past your bedtime |
| 05:36.21 | benjk | not the amateur radio repeater stuff I think |
| 05:36.29 | [hC] | What is the benefit, aside from it being tidier under the hood? Does it add any functionality or stability, or anything at all? |
| 05:36.49 | Strom_C | [hC], it adds pixies |
| 05:36.49 | VAXpirate | achandra: oh, c'mon. VMware is so obsolete. Head for VirtualPC. |
| 05:36.50 | denon | achandra: vmware/etc is going to add a bit of latency no matter how you do it .. |
| 05:36.51 | benjk | its a unified call model, like Dialogic's globalcall |
| 05:37.10 | denon | achandra: Ive heard of people having really good luck running * under vmware, but it seems like it could be troublesome |
| 05:37.17 | benjk | makes every tech look exactly the same from the top |
| 05:37.23 | Qwell | denon: I've heard of people have very bad luck |
| 05:37.41 | denon | Qwell: I guess it's safe to say people have a variety of luck with it then |
| 05:37.57 | denon | but I have talked to at least 2 people who swore up and down they had it running *well* under vmware |
| 05:38.04 | CunningPike | achandra: Are you planning to use Digium cards? |
| 05:38.06 | [hC] | OOoOO Pixies! |
| 05:38.08 | denon | what they tweaked, I Dont know |
| 05:38.21 | achandra | the thought here...is that with somethinglike a dell 850 which handles approx 120 simultaneous calls with rtp to spread it out with vms, and use opensers dispatcher.so module to load balance... |
| 05:38.24 | achandra | ;) |
| 05:38.32 | [hC] | benjk: you didnt really answer my question. |
| 05:38.43 | benjk | http://www.soft-switch.org/unicall/unicall/index.html |
| 05:39.05 | achandra | CunningPike: nope |
| 05:39.19 | denon | achandra: I think you'd be much better off trying to LB over real hardware, and saturating the iron |
| 05:39.36 | CunningPike | achandra: Well, you might have some success then. Many of the troubled tales I have read were around getting the hardware to work |
| 05:39.53 | denon | achandra: you might also want to wait until hardware virtualization is a bit more mature |
| 05:40.09 | achandra | hmmmm.all valid points..... |
| 05:40.11 | denon | (more mature on x86, anyway - it's plenty mature on, say, as/400s) |
| 05:41.25 | CunningPike | denon: Or Tandems :) |
| 05:41.34 | [hC] | benjk: yeah, I saw the diagrams, it still doesnt tell me how as an asterisk user, it benefits me at all. |
| 05:41.38 | achandra | virtualization is pretty embraced by some some big hitter now...ie some huge mortage companies using j2ee apps....etc.. ebay uses it...not mature enough?? |
| 05:41.40 | *** join/#asterisk kmilitzer (n=km@office-gw.westend.com) |
| 05:41.44 | denon | er, sorry, i5 |
| 05:42.15 | denon | IBM LPAR is just so darn cool |
| 05:42.19 | *** join/#asterisk Strom_C (n=strom@fl-65-41-146-225.sta.embarqhsd.net) |
| 05:42.30 | achandra | LPAR is cool...will give ya that :) |
| 05:42.38 | Strom_C | this hotel wifi blows |
| 05:42.56 | [hC] | that reminds me i need to book my astricon tickets |
| 05:42.57 | benjk | for starters, it makes just about everything simpler |
| 05:42.58 | CunningPike | Strom_C: So you said :) |
| 05:44.18 | benjk | once Steve has completed the lower layer interface how a driver interfaces with the unicall core, then you can easily plug in a variety of different vendors' cards with little porting effort and without replicating work |
| 05:44.20 | Aurs | when a caller has 2 caller-ids and call in on our PRI, the last number will always be presented. is there a way to choose which CID to use? |
| 05:44.40 | benjk | you wont have such things as chan_misdn, chan_capi, chan_foo, chan_bar, chan_baz |
| 05:44.43 | [hC] | Hey, any of you guys seen any information about using asterisk to control door locks? |
| 05:44.46 | Strom_C | Aurs, what do you mean "2 caller IDs" |
| 05:44.55 | Qwell | [hC]: mitcheloc |
| 05:45.11 | Qwell | I think it was him anyhow |
| 05:45.36 | Aurs | Strom_C: here in norway there are some cell phones that have 2 numbers on the same phone |
| 05:45.59 | benjk | in fact Unicall is much closer to Jim Dixon's original vision for his zaptel project |
| 05:46.02 | Aurs | when these guys call, our * server will always use CID #2 |
| 05:46.16 | Strom_C | Aurs, surely that's a telco issue |
| 05:46.16 | CunningPike | [hC]: Ed Guy did a good presenation at Astricon last year about integrating asterisk and Mister House |
| 05:46.36 | Qwell | CunningPike: Was that Astricon? |
| 05:46.43 | Qwell | I guess it was |
| 05:46.51 | CunningPike | Qwell: Yup - last year |
| 05:47.02 | [hC] | CunningPike: I was in that presentation |
| 05:47.04 | [hC] | er |
| 05:47.06 | [hC] | present for |
| 05:47.13 | [hC] | but I dont recall any specifics about door locks |
| 05:47.53 | CunningPike | [hC]: No - nothing specific, but just about the whole home automation thing. |
| 05:48.48 | [hC] | Yeah, I mean it should be as easy as integrating some sort of X10 module and a shell script together, but Im hoping for something a bit less... 'duct taped' |
| 05:48.51 | [hC] | :) |
| 05:48.59 | CunningPike | :D |
| 05:57.38 | *** join/#asterisk adelas (n=booger@rrcs-24-199-21-138.west.biz.rr.com) |
| 05:57.43 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 05:59.33 | Aurs | Strom_C: I can show you some more details here... just a sec |
| 06:10.06 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.221) |
| 06:10.43 | [shodan] | any way I can sms to my phone from * for free ? (canadian cellphone carrier) |
| 06:12.13 | [shodan] | or paid .. |
| 06:13.18 | CunningPike | [shodan]: I'm struggling with the concept of using a voice system to send sms messages..... can you explain what you are trying to do? |
| 06:13.46 | *** join/#asterisk daysmen3 (n=primus@host217-44-109-83.range217-44.btcentralplus.com) |
| 06:14.10 | [hC] | I'm about to integrate w/ SMS modem for things like SMS callback |
| 06:14.27 | [hC] | i text * a number, it dials that number and connects it back to me (I get unlimited cell incoming) |
| 06:14.30 | [shodan] | this is just an example but .. user mike received a voice mail or a fax , I want to sent a sms to his cell to that effect |
| 06:14.56 | CunningPike | [shodan]: Can't you just email it to 123456789@msg.telus.net or wahtever? |
| 06:15.04 | *** part/#asterisk celophane (n=e@ip68-104-251-230.ph.ph.cox.net) |
| 06:15.20 | [hC] | CunningPike: telus charges you an arm and a leg to receive email sms messages, as well |
| 06:15.28 | [hC] | [shodan]: you'll want to look into integrating w/ Kannel |
| 06:15.32 | [hC] | its a linux SMSC |
| 06:16.01 | [shodan] | CunningPike, maybe but if I can send an sms I don't have to worry about the user's cellphone cie |
| 06:16.46 | CunningPike | [shodan]: True |
| 06:17.06 | benjk | Kannel won't help you unless you have a service contract with an operator to let you exchange messages with their SMSC directly |
| 06:17.19 | CunningPike | [hC]: Ya - I have no friends, so I don't get that many :) |
| 06:17.41 | [hC] | benjk: Its as simple as connecting a GSM modem and buying a SIM card that has a text messaging plan |
| 06:18.30 | benjk | that's not SMSC mode though |
| 06:18.46 | [shodan] | aren't there voip providers that offer sms ? |
| 06:19.08 | benjk | you could always get one of those GSM PCI cards from Junghanns |
| 06:19.22 | benjk | its fully integrated with Asterisk |
| 06:19.24 | [hC] | benjk: Kannel is still an SMSC, wether or not you connect directly to another SMSC, or wether you broadcast via GSM modem. |
| 06:19.29 | benjk | for voice and sms |
| 06:19.47 | [hC] | I didnt realize GSM PCI cards existed... |
| 06:20.22 | benjk | Junghanns.net |
| 06:20.37 | benjk | 1 to 4 transceivers |
| 06:21.09 | benjk | and its all integrated with asterisk, not fiddling with external delivery systems |
| 06:21.43 | [hC] | no prices on their site i guess huh? |
| 06:23.08 | benjk | that's because usually they sell through resellers |
| 06:23.30 | [shodan] | no prices anywhere, check a bunch of their resellers , they're all sellings "solutions" |
| 06:23.37 | [hC] | holy SHIT |
| 06:23.39 | [shodan] | expect mucho expensive |
| 06:23.42 | [hC] | 1185 euros?? |
| 06:23.53 | [hC] | I'll take a gsm modem and kannel, thanks! |
| 06:24.07 | benjk | is that for the quad card? |
| 06:24.09 | [shodan] | that's like 4 times the cost of a ethernet gsm gateway |
| 06:24.36 | [hC] | pardon me 885 euros for the single channel |
| 06:24.45 | *** join/#asterisk breakdisk (n=breakdis@62.149.122.2) |
| 06:24.53 | benjk | you mean that crap from Siemens that goes through analog? |
| 06:25.57 | [shodan] | that kinda thing I guess http://cgi.ebay.ca/Tri-Band-GSM-Cellular-Terminal-Gateway-IP-PBX-VOIP-GPRS_W0QQitemZ110022493391QQihZ001QQcategoryZ61839QQssPageNameZWDVWQQrdZ1QQcmdZViewItem |
| 06:25.59 | benjk | I wouldn't mind paying 900 EUR if they had a UMTS version of it |
| 06:26.23 | benjk | but we don't have GSM over here :( |
| 06:26.31 | [shodan] | what tech do they use in Canada ? is it gsm or cdma or something else mostly ? |
| 06:26.38 | benjk | both |
| 06:26.54 | benjk | in CA they have both GSM in the US bands and CDMA |
| 06:27.07 | [shodan] | is bell gsm and telus cdma ? |
| 06:27.27 | benjk | you can always look it up at the GSMA |
| 06:27.37 | benjk | under "roaming" or "international" |
| 06:27.56 | [hC] | bell and telus are both cdma |
| 06:28.02 | [hC] | rogers/fido is gsm |
| 06:28.21 | *** join/#asterisk af_ (n=af@ip-173-144.sn1.eutelia.it) |
| 06:28.49 | benjk | http://www.gsmworld.com/index.shtml |
| 06:29.36 | CunningPike | Don't forget Mike, our iDEN network |
| 06:29.51 | *** join/#asterisk Strom_C (n=strom@fl-65-41-146-225.sta.embarqhsd.net) |
| 06:30.03 | [shodan] | oh I thought bell was gsm because a friend of mine has a bell and he said he had a sim card , or does cdma use sim cards too or something similar ? |
| 06:30.12 | *** join/#asterisk |ryan| (n=foo@c-24-23-17-75.hsd1.ca.comcast.net) |
| 06:30.27 | diablopico | hello ... i am having trouble with calls that start with great quality for about 3 minutes , and then it gets so bad that i cant understand what the called party is saying , any ideas what to look for ? |
| 06:30.53 | Strom_C | diablopico, what kind of calls? |
| 06:31.00 | benjk | is it a SIM or a USIM? |
| 06:31.27 | diablopico | incoming from h323 on oh323 for asterisk , and out on unicall for r2 signalling |
| 06:32.04 | benjk | diablopico, sounds like your h323 packets are piling up and the machine cant handle them fast enough |
| 06:32.06 | [shodan] | dunno , he called it a "sim" but he probably just refered to any sim-like object that plugged in his phone |
| 06:32.12 | |ryan| | I know this isn't the channel for it, but has anyone here gotten a linksys pap2 to talk to a sip server? |
| 06:32.29 | Strom_C | |ryan|, yes, i have one talking to my asterisk box |
| 06:32.35 | benjk | UMTS (aka 3G) systems also use SIMs, referred to as USIMs |
| 06:32.45 | benjk | and those are technically CDMA |
| 06:32.49 | *** join/#asterisk UlbabraB (n=UlbabraB@host241-43-static.72-81-b.business.telecomitalia.it) |
| 06:33.05 | |ryan| | Strom_C: mine is giving me a busy signal when i try to dial |
| 06:33.08 | diablopico | benjk: that makes sense , but it is only one call on the system ,, does that sound possible . |
| 06:33.11 | diablopico | ? |
| 06:33.17 | Strom_C | |ryan|, a busy signal, or a reorder tone? |
| 06:33.19 | benjk | but considered either CDMA descendant (CDMA-2000) or GSM descendant (3GSM) |
| 06:33.29 | |ryan| | Strom_C: what's the diffrence? |
| 06:33.40 | Strom_C | busy signal is 60ipm, reorder is twice the speed |
| 06:33.45 | benjk | diaplopico, sometimes this can be network related too |
| 06:33.59 | |ryan| | ipm? |
| 06:34.04 | Strom_C | impulses per minute |
| 06:34.17 | |ryan| | i guess it's probably a reorder then |
| 06:34.30 | Strom_C | alright....reorder signals an error condition |
| 06:34.34 | |ryan| | i'm trying to connect to a remote sip server |
| 06:34.41 | Strom_C | is the pap2 registering with the server? |
| 06:34.41 | [shodan] | hmm |
| 06:34.54 | |ryan| | yeah, it's registered. |
| 06:35.06 | Strom_C | are you controlling the server as well? |
| 06:35.07 | benjk | you should be able to send sms from a 3G system to GSM though |
| 06:35.34 | |ryan| | No, not my server. |
| 06:35.59 | |ryan| | I am behind NAT, but I've got ports forwarded |
| 06:36.08 | Strom_C | well, that makes debugging much more difficult |
| 06:37.34 | |ryan| | one odd thing i noticed |
| 06:37.43 | |ryan| | right before it gives me the tone |
| 06:37.54 | |ryan| | it sends out some broken DNS requests. |
| 06:39.05 | *** join/#asterisk hads|home (n=hads@mail.nice.net.nz) |
| 06:39.13 | |ryan| | 23:16:47.834011 IP (tos 0x0, ttl 250, id 2274, offset 0, flags [none], length: 58) 192.168.x.121.14773 > 192.168.x.1.53: [udp sum ok] 1+ Type1891 (Class 28525)? . (30) |
| 06:39.13 | |ryan| | 23:16:47.839464 IP (tos 0x0, ttl 64, id 14, offset 0, flags [DF], length: 45) 192.168.x.1.53 > 192.168.x.121.14773: [udp sum ok] 1 NXDomain q: Type1891 (Class 28525)? . 0/0/0 (17) |
| 06:39.23 | |ryan| | wtf is that? |
| 06:40.04 | |ryan| | i mean, i can see it talkign to the SIP and STUN server... |
| 06:40.21 | Strom_C | beats me; I am not a DNS expert |
| 06:40.36 | Strom_C | why not try getting it working with your asterisk box for testing purposes |
| 06:40.48 | |ryan| | yeah, I suppose I could do that. |
| 06:41.22 | Strom_C | oh? |
| 06:41.39 | |ryan| | it doesn't work in my second pci slot |
| 06:42.05 | Strom_C | odd - which card is it? |
| 06:42.38 | |ryan| | X100P |
| 06:42.57 | Strom_C | a real x100p, or a clone? |
| 06:43.11 | |ryan| | from x100p.com, claims to be a real one. |
| 06:43.26 | hads|home | claims to :) |
| 06:43.27 | Strom_C | claims :) |
| 06:43.34 | *** join/#asterisk Frogdude (n=chris@c-24-16-72-159.hsd1.wa.comcast.net) |
| 06:43.38 | |ryan| | It is detected as a real one. |
| 06:43.45 | Strom_C | yeah, there's some hack to get it to do that |
| 06:43.53 | |ryan| | it does work. |
| 06:43.58 | Strom_C | digium hasn't manufactured those for....god, ive forgotten how long |
| 06:44.00 | benjk | there is no such thing as a real one and not-real one |
| 06:44.08 | Strom_C | oh christ, here we go again |
| 06:44.12 | |ryan| | just gets pissy if i put it in the lower PCI slot. |
| 06:44.13 | benjk | they are all the same, or they wouldn't work |
| 06:44.17 | CunningPike | "They're all crap" lol |
| 06:44.20 | Strom_C | yes |
| 06:44.30 | benjk | its called an Ambient MD3200 PCI modem |
| 06:44.47 | hads|home | That's nice. |
| 06:45.02 | |ryan| | i also have a voicetronix openswitch card, but I can't get the bloody driver for it to build. |
| 06:45.17 | *** join/#asterisk jeebusmobile (n=jeebusmo@12.180.154.130) |
| 06:45.30 | benjk | voicetronix stuff usually only works with a particular version of asterisk |
| 06:45.53 | benjk | you need to call them up (or send email) and ask which is the latest matching release |
| 06:46.40 | *** join/#asterisk UlbabraB (n=UlbabraB@host241-43-static.72-81-b.business.telecomitalia.it) |
| 06:46.54 | benjk | but you are well advised to invest your time into getting the vt driver to work and forget about the Ambient modem |
| 06:47.20 | Snake-Eyes | is there any way to increase the number of columns the 'show' comands use in Asterisk CLI (eg "show channels" ) |
| 06:47.32 | benjk | because the vt card is much better hardware |
| 06:47.39 | |ryan| | benjk: I'm only being lent the vt card. |
| 06:47.40 | *** join/#asterisk adorah (n=Administ@84.94.133.70.cable.012.net.il) |
| 06:47.45 | benjk | oh, ok |
| 06:48.25 | |ryan| | this is just for my home system so i can learn a bit of asterisk |
| 06:48.26 | benjk | I have had better results with Sipura 3000s than the ambient modems |
| 06:48.50 | benjk | and I still have ambient modems which were purchased directly from Digium years ago |
| 06:49.01 | benjk | when the chipsets were still being manufactured |
| 06:50.09 | benjk | so if you can afford the 65 bucks a Sipura 3000 costs, you can safe yourself a lot of trouble |
| 06:50.28 | mmurdock | Sorry to interupt, I've got another question. |
| 06:50.34 | Strom_C | just ask |
| 06:51.23 | mmurdock | I've got a tdm24 with a fxs in slot one and two fxo's in slot 2 and 3 |
| 06:52.06 | mmurdock | I set my zaptel.conf with fxsks=1-4 and fxoks=5-12 This is right isn't it? |
| 06:52.14 | Strom_C | no |
| 06:52.25 | Strom_C | fxs modules use fxo signaling |
| 06:52.28 | Strom_C | and vice versa |
| 06:52.35 | CunningPike | ~fxofxs |
| 06:52.36 | jbot | methinks fxofxs is An FXO port expects to receive dialtone and receive ring voltage, you plug a phone LINE into this. An FXS port expects to provide dialtone and provide ring voltage, you plug a PHONE into this. |
| 06:52.36 | mmurdock | So I need to reverse it? |
| 06:52.55 | Strom_C | mmurdock, exactly |
| 06:52.59 | mmurdock | Got it. |
| 06:53.18 | Snake-Eyes | sorry I was wonder if I can increase column length of Asterisk CLI when commands like show channels is run ? |
| 06:53.32 | |ryan| | asterisk server rebooting |
| 06:53.36 | mmurdock | cool, no errors when running ztcfg -vvvv |
| 06:53.58 | benjk | yes, you can, but last time I looked it required a change in the source code |
| 06:54.59 | *** join/#asterisk Assid (i=assid@203.115.83.215) |
| 06:55.00 | |ryan| | hmm |
| 06:55.05 | |ryan| | wcfxo: Out of space to write register 06 with e0 |
| 06:55.05 | |ryan| | wcfxo: Out of space to write register 0f with 00 |
| 06:55.06 | *** join/#asterisk DarKnesS_WolF (n=wolf@212.103.170.135) |
| 06:55.08 | Assid | 1.2.11 ? |
| 06:55.20 | |ryan| | anyone know what that mean/how to fix it? |
| 06:56.30 | Snake-Eyes | benjk, were you reffering to my question ? |
| 06:56.32 | mmurdock | so since my zaptel.conf seems to be correct, I just need to add the channels to asterisk, correct? |
| 06:56.39 | |ryan| | hmm |
| 06:56.55 | Strom_C | mmurdock, zapata.conf |
| 06:56.57 | benjk | grep "Out of space to write register" wcfxo.c |
| 06:57.02 | benjk | in zaptel |
| 06:57.20 | |ryan| | i think it's my kernel |
| 06:57.25 | benjk | that will point you at the code and probably tell you under which circumstances this is generated |
| 06:57.38 | benjk | could come from the kernel yes |
| 06:57.42 | |ryan| | i have two kernels installed, and i don't remember which one worked |
| 06:59.10 | |ryan| | damnit |
| 06:59.13 | |ryan| | doesn't work |
| 06:59.16 | |ryan| | :( |
| 07:00.10 | |ryan| | no, not my kernel. |
| 07:00.10 | |ryan| | hmm |
| 07:00.18 | |ryan| | this _DID_ work before. |
| 07:02.15 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 07:03.55 | |ryan| | wtf. |
| 07:05.35 | *** part/#asterisk operat0r (n=h0msar@adsl-19-78-76.asm.bellsouth.net) |
| 07:18.20 | *** join/#asterisk lowlevel (n=Stuart@CPE0050ba71c82f-CM000f9f7d6742.cpe.net.cable.rogers.com) |
| 07:24.44 | CunningPike | Is it normal for 'modprobe wct2xxp' to result in wct4xxp showing up in lsmod? |
| 07:25.44 | Qwell | CunningPike: it's an alias iirc |
| 07:25.54 | CunningPike | OK - thanks, Qwell |
| 07:28.57 | *** join/#asterisk fnordus (n=dnall@s142-179-111-243.bc.hsia.telus.net) |
| 07:33.24 | bionoid | Good morning everyone. I woke up to some error messages that indicates that a user is trying to exceed his call quota (of 1 call), apparantly the users softphone crashed, and asterisk did not notice the fact that the call was terminated. Is there a way to check for dead SIP clients to prevent this kind of thing? (it did eventually free up, though, but I don't know when). Extract from log: |
| 07:33.30 | bionoid | Aug 22 21:26:00 ERROR[1630] chan_sip.c: Call from user 'olav' rejected due to usage limit of 1 |
| 07:35.22 | *** join/#asterisk chapeaurouge (n=chapeaur@80.92.83.35) |
| 07:36.04 | CunningPike | bionoid: Would qualify help? |
| 07:36.27 | |ryan| | erg |
| 07:36.40 | |ryan| | the damn thing doesn't like sharing an IRQ |
| 07:36.50 | |ryan| | is there any way to force it to use a specific one? |
| 07:37.11 | bionoid | CunningPike: ? |
| 07:37.42 | CunningPike | bionoid: In sip.conf, qualify=yes |
| 07:37.44 | bionoid | |ryan|: "force" is relative - depends on your hardware. Some motherboards allow you to control the resources from the BIOS (down to every last detail), others don't |
| 07:38.19 | bionoid | |ryan|: You might want to move the card to a different PCI slot, it may (or may not, unfortunally) have the desired effect ;) |
| 07:38.21 | |ryan| | bios doesn't let me |
| 07:38.27 | bionoid | CunningPike: Will look that up, thanks |
| 07:38.30 | |ryan| | changing PCI slots does not help. |
| 07:38.44 | CunningPike | |ryan|: What type of PC? |
| 07:38.49 | |ryan| | um |
| 07:39.14 | |ryan| | some dell optiplex someone gave me. P3 450 |
| 07:39.31 | bionoid | |ryan|: What is it conflicting with? |
| 07:39.32 | |ryan| | it does have acpi support |
| 07:39.39 | |ryan| | AGP. |
| 07:39.46 | |ryan| | on-board AGP. |
| 07:40.03 | |ryan| | this WAS working previously. |
| 07:40.08 | |ryan| | which perplexes me. |
| 07:40.38 | |ryan| | there's nothing on IRQ 7 |
| 07:43.10 | bionoid | Hm that sucks |
| 07:43.24 | *** join/#asterisk alphaque (n=alphaque@219.94.80.162) |
| 07:43.34 | bionoid | One possibility, obviously, is to install a pci display adapter and disable the onboard one |
| 07:43.48 | bionoid | but that shouldn't be necessary if it _was_ working fine.. |
| 07:44.04 | CunningPike | |ryan|: Disable USB if you don't need it |
| 07:44.38 | *** join/#asterisk Strom_C (n=strom@fl-65-41-146-225.sta.embarqhsd.net) |
| 07:49.26 | hads|home | bionoid: 'show channels' from the CLI will show you any calls to the softphone channel which you can kill with 'soft hangup SIP/foo' |
| 07:50.10 | bionoid | Yes, that I know, but I don't wanna get up at night to do it.. :P |
| 07:50.18 | |ryan| | CunningPike: can't disable USB |
| 07:50.29 | |ryan| | I turned off everything i can |
| 07:50.34 | hads|home | bionoid: OK, so long as you konw :) |
| 07:50.39 | bionoid | ;) cheers |
| 07:50.39 | |ryan| | the first pci slot always uses IRQ10 |
| 07:50.45 | CunningPike | |ryan|: And what is it sharing with? |
| 07:50.57 | |ryan| | whach is used by the video card. which can't be disabled. |
| 07:51.12 | |ryan| | irq11, which it gets in the second slot is free |
| 07:51.23 | |ryan| | but the card has a diffent problem in that slot. |
| 07:51.39 | CunningPike | |ryan|: What's the problem in the second slot |
| 07:51.39 | bionoid | what problem is that? |
| 07:51.45 | *** join/#asterisk inspired (n=mikael@85.221.7.59) |
| 07:52.39 | |ryan| | wcfxo: Out of space to write register 06 with e0 |
| 07:52.39 | |ryan| | wcfxo: Out of space to write register 0f with 00 |
| 07:52.46 | |ryan| | Failed to initailize DAA, giving up... |
| 08:01.53 | *** join/#asterisk Dibbler_ (n=Dibbler@dsl-217-155-254-174.zen.co.uk) |
| 08:03.33 | *** join/#asterisk ptblank (n=MURDER1@68.233.142.186) |
| 08:06.10 | *** join/#asterisk BladeRunner05 (n=feelme@81-174-56-54.f5.ngi.it) |
| 08:09.45 | *** join/#asterisk Tebi_ (n=rantis@gw.aller.fi) |
| 08:09.53 | *** join/#asterisk bhrobinson (n=brobinso@mail1.nt-it.com) |
| 08:11.22 | bionoid | |ryan|: That sounds very much like hardware error |
| 08:11.27 | bionoid | (sorry afk, boss walked in;p) |
| 08:12.03 | *** join/#asterisk freebsd_fan (n=ebola@i-83-67-73-117.freedom2surf.net) |
| 08:12.46 | bionoid | |ryan|: "I'm feeling lucky" result http://scottstuff.net/blog/articles/2005/02/04/upgrading-asterisk-can-be-a-whole-lot-of-fun |
| 08:13.04 | *** join/#asterisk Dibbler_ (n=Dibbler@dsl-217-155-254-174.zen.co.uk) |
| 08:13.09 | CunningPike | lol - I was just reading that very thread |
| 08:15.04 | bionoid | Q.What does this type of error mean? "wcfxo: Out of space to write register 05 with 0a" |
| 08:15.07 | bionoid | A.The means that the fxo device is not receiving interrupts. The hardware needs to be on it's own irq. |
| 08:17.53 | *** join/#asterisk Dibbler_ (n=Dibbler@dsl-217-155-254-174.zen.co.uk) |
| 08:18.44 | *** join/#asterisk axscode (n=axscode@203.213.217.123) |
| 08:18.54 | axscode | hi guyz.. anyone tried F1000G on asterisk!? |
| 08:19.13 | CunningPike | axscode: The wifi phone? |
| 08:19.26 | axscode | yups.. the Wifi Phone F1000G on asterisk? |
| 08:19.38 | CunningPike | axscode: Yes, we have one |
| 08:20.01 | axscode | coz i tried it here.. it can register... but it cant call.. do you happen to know whats on sip.conf and the config on the fone please? |
| 08:20.03 | *** join/#asterisk luchshiy (n=anonymou@d212-53-104-193.cust.tele2.ch) |
| 08:20.40 | CunningPike | axscode: It's very similar to most other UAs..... |
| 08:21.19 | *** join/#asterisk af_ (n=af@ip-173-144.sn1.eutelia.it) |
| 08:21.24 | *** join/#asterisk luchshiy (n=anonymou@d212-53-104-193.cust.tele2.ch) |
| 08:21.24 | axscode | well... i dont know whats the problem yet... |
| 08:21.35 | CunningPike | axscode: If it's registered properly, it should call - what do you get in the CLI? |
| 08:21.38 | axscode | when i dial. i get nothing from the CLI |
| 08:21.49 | CunningPike | axscode: Can you call it? |
| 08:21.58 | axscode | i cant |
| 08:22.03 | axscode | nothing happens on my CLI |
| 08:22.38 | CunningPike | axscode: Then you have bigger problems, my friend - what verbose level are you running at? |
| 08:22.46 | axscode | verbose 100 |
| 08:23.20 | CunningPike | Well, if nothing shows on the CLI when you try to call it, it's not a problem with the wifi phone. |
| 08:23.40 | CunningPike | Enter an invalid extension - do you see anything in the CLI? |
| 08:23.55 | axscode | hmm ill see wait |
| 08:24.55 | *** join/#asterisk Nebukadneza (n=daddel9@i3ED6E720.versanet.de) |
| 08:25.39 | *** join/#asterisk x86 (n=x86@p3m/member/x86) |
| 08:25.45 | axscode | i got error |
| 08:25.46 | axscode | hehe.. |
| 08:25.55 | axscode | cannot find extension context. |
| 08:25.57 | axscode | :) |
| 08:28.48 | |ryan| | bionoid: yeah, i saw that page |
| 08:29.18 | |ryan| | bionoid: it wasn't sharing an intrupt |
| 08:29.20 | CunningPike | axscode: OK - now dial your wifi phone from the same phone |
| 08:29.34 | *** join/#asterisk darkskiez (n=mbryars@thirtythree.103.wightcablenorth.net) |
| 08:29.43 | axscode | using? the existing context? |
| 08:30.12 | |ryan| | this bios if fucking retarded |
| 08:30.44 | *** join/#asterisk znoG (n=gs@162-148-235-201.fibertel.com.ar) |
| 08:31.18 | _Vile | back |
| 08:32.10 | CunningPike | axscode: As you would expect to be able to |
| 08:32.55 | _Vile | cannot find extension is an easy error |
| 08:33.18 | *** join/#asterisk Nebukadneza (n=daddel9@i3ED6E720.versanet.de) |
| 08:33.59 | _Vile | load the exten => xxxx in the right context |
| 08:34.17 | _Vile | or difer to the right context on inbound |
| 08:40.19 | *** join/#asterisk Jeffjohnson (n=Jeffjohn@unaffiliated/jeffjohnson) |
| 08:40.45 | _Vile | axs, 3 min |
| 08:42.15 | _Vile | one more smoke |
| 08:42.27 | _Vile | 2 min |
| 08:45.54 | Jeffjohnson | hello |
| 08:46.06 | Jeffjohnson | i alway get the error message "Aug 23 10:47:28 NOTICE[23252]: chan_sip.c:9683 handle_response_invite: Failed to authenticate on INVITE to '""" <sip:2815025e0@sipgate.de>;tag=as486bf92e' |
| 08:46.06 | Jeffjohnson | <PROTECTED> |
| 08:46.34 | _Vile | failure to auth |
| 08:47.11 | _Vile | means ur trying to initate a call |
| 08:47.40 | _Vile | to a sip provider |
| 08:48.01 | Jeffjohnson | failure to auth, can't be the problem... cause calls from the sip provider to asterisk work |
| 08:48.03 | _Vile | that doesn't see you as being registered |
| 08:48.09 | Jeffjohnson | iam registered |
| 08:48.18 | _Vile | no you are not. |
| 08:48.31 | Jeffjohnson | asterisk*CLI> sip show registry |
| 08:48.31 | Jeffjohnson | Host Username Refresh State |
| 08:48.31 | Jeffjohnson | dus.net:5060 000387224998 105 Registered |
| 08:48.31 | Jeffjohnson | sipgate.de:5060 2815025e0 105 Registered |
| 08:48.36 | Jeffjohnson | you see :) |
| 08:48.46 | Jeffjohnson | _Vile: state: registered |
| 08:50.20 | Jeffjohnson | _Vile: or not? |
| 08:50.55 | _Vile | registered maybe |
| 08:51.03 | Jeffjohnson | but? |
| 08:51.03 | _Vile | check the callerid num |
| 08:51.06 | axscode | CunningPike: just want to ask. what codec u use for your F1000G? |
| 08:52.10 | _Vile | caller id number not being right when you initiate a call can on some switchesngive a 401 unauthorized |
| 08:53.51 | _Vile | do you know what kind of switch you are talking to, curiosity |
| 08:54.32 | *** join/#asterisk ptblank (n=MURDER1@68-169-166-65.lmdaca.adelphia.net) |
| 08:55.15 | Jeffjohnson | _Vile: k thx, now it works. It looks like that i need the callerid |
| 08:55.22 | _Vile | ;) |
| 08:55.41 | *** join/#asterisk Vec (n=Vector@dsl-165-182-202.telkomadsl.co.za) |
| 08:56.07 | _Vile | do you know what kind of switch you are talking to, curiosity |
| 08:56.13 | _Vile | ? |
| 08:56.43 | Assid | whats refresh? |
| 08:58.36 | _Vile | no reach-around, gotta love it |
| 08:58.46 | _Vile | Assid, meaning? |
| 08:58.53 | Assid | the refresh column |
| 08:58.58 | _Vile | on |
| 08:59.07 | _Vile | ahhh |
| 08:59.13 | _Vile | sip show registry? |
| 08:59.56 | Assid | yes |
| 09:00.22 | _Vile | registration refresh #, # of seconds until the next "ping" is made, ping is I think in sip "options" |
| 09:02.20 | _Vile | i can double check in code, but I believe that's what it means |
| 09:02.41 | Assid | hmm |
| 09:02.47 | Assid | how do i set it? |
| 09:03.05 | *** part/#asterisk bhrobinson (n=brobinso@mail1.nt-it.com) |
| 09:03.25 | _Vile | interesting question |
| 09:03.33 | _Vile | never had to -- why? |
| 09:06.22 | *** join/#asterisk Sonderblade (n=mah@static-213.131.147.169.addr.tdcsong.se) |
| 09:07.44 | _Vile | axs is plugged into a meta or a taqua |
| 09:08.27 | *** join/#asterisk apardo (n=apardo@87.217.146.232) |
| 09:08.33 | _Vile | maybe a cisco cm |
| 09:08.43 | _Vile | depending on config on the cm |
| 09:10.44 | _Vile | s/axs/Jeffjohnson |
| 09:10.54 | *** join/#asterisk BugKham (n=bugkham@ppp-58.8.11.241.revip2.asianet.co.th) |
| 09:11.16 | Jeffjohnson | _Vile: /w/t/f? |
| 09:11.46 | *** join/#asterisk ghenry (n=ghenry@82-69-192-46.dsl.in-addr.zen.co.uk) |
| 09:11.51 | BugKham | anyone knows how to skip "STREAM FILE" in agi using any dfmf? |
| 09:12.14 | BugKham | s/dfmf/dtmf |
| 09:12.15 | _Vile | Bug, explain |
| 09:12.28 | _Vile | dtmf detect on playback? |
| 09:12.33 | _Vile | Jeff, |
| 09:12.53 | _Vile | Jeff, curious do you know your upstream voip carriers equipment? |
| 09:13.06 | BugKham | _Vile, STREAM FILE currently allows escape digits or a blank value |
| 09:13.36 | _Vile | explain escape |
| 09:13.43 | BugKham | but I wanna skip it using any single dtmf digit |
| 09:13.46 | _Vile | like *? |
| 09:14.35 | _Vile | what application are you looking at modifying, voicemail? |
| 09:14.53 | BugKham | ok, STREAM FILE in agi currently requires three parameters right? |
| 09:15.06 | _Vile | gimme a sec to read up |
| 09:15.10 | BugKham | filename, escape digits, and sample offset |
| 09:15.17 | _Vile | sec |
| 09:15.25 | _Vile | lemme catchup |
| 09:16.03 | BugKham | wiki/view/stream+file |
| 09:18.17 | _Vile | different approach, what are you trying to do |
| 09:18.42 | BugKham | why's that? |
| 09:18.54 | _Vile | because i asked |
| 09:19.21 | *** join/#asterisk delmar (n=delmar@ip-58-28-158-154.ubs-dsl.xnet.co.nz) |
| 09:20.03 | _Vile | what are you trying to do? |
| 09:20.06 | delmar | hey everyone. i have a wierd little problem... it has to do with the use of the "Answer" command and it's effects on voicemail when calls are being redirected. |
| 09:20.12 | BugKham | what I'm trying to do is to let a caller to skip the voice prompt if they do not want to listen by using any key on their telephone |
| 09:20.29 | _Vile | ahh |
| 09:20.30 | BugKham | not a particular one |
| 09:21.14 | _Vile | that'll be in the dialplan |
| 09:21.48 | BugKham | yeah, there'll be no problem on the dialplan |
| 09:21.51 | delmar | BugKham, pick a number and add it to the ivr .. then set like .. exten => X,1,Goto(blah,x,x) |
| 09:22.03 | _Vile | and you need to jump to the end context |
| 09:22.04 | delmar | where X is the number |
| 09:22.13 | _Vile | when they dial any digit |
| 09:22.24 | _Vile | so |
| 09:22.27 | *** join/#asterisk nicox (n=nicox@83-64-42-210.prater.xdsl-line.inode.at) |
| 09:22.31 | nicox | Hello |
| 09:22.32 | _Vile | 0-9 and * and # |
| 09:22.35 | delmar | that would be the " i " |
| 09:22.45 | BugKham | delmar, hmm, I wanna do it in my agi script |
| 09:22.50 | BugKham | is there a way? |
| 09:22.55 | _Vile | in the agi |
| 09:22.58 | _Vile | detect a digit |
| 09:23.04 | delmar | exten => i,1,Goto(blah,x,x) |
| 09:23.07 | _Vile | 0-9 or * |
| 09:23.10 | nicox | did anybody kknow, when the res_snmp is available in svn-trunk? |
| 09:23.12 | delmar | i = invalid number |
| 09:23.13 | _Vile | or # |
| 09:23.18 | _Vile | and send it ot the context |
| 09:23.26 | delmar | so if they press a number that is not in your menu... it will action " i " |
| 09:23.41 | _Vile | del, he's doing it within AGI tho |
| 09:23.53 | delmar | Why? |
| 09:23.54 | BugKham | _Vile, I think I better put all in the "escape digits" |
| 09:24.03 | BugKham | _Vile, let me try |
| 09:24.04 | delmar | thats just making something complicated when it doesn't need to be :P |
| 09:24.30 | _Vile | he probably has some perl code |
| 09:24.36 | _Vile | who knows |
| 09:24.39 | delmar | yeah. guess so. |
| 09:24.50 | delmar | anyway.... here is my problem.... |
| 09:24.59 | *** join/#asterisk Newbie___ (n=me@211.24.146.11) |
| 09:25.07 | delmar | lets say there are two extensions... 801 and 802.... |
| 09:25.15 | delmar | both are Polycom phones... |
| 09:25.15 | Newbie___ | hi, anyone uses a Zhone 24FXS ? |
| 09:25.16 | BugKham | yeah, that works =) |
| 09:25.26 | _Vile | enjoy bug |
| 09:25.29 | nicox | did anybody kknow, when the res_snmp is available in svn-trunk? |
| 09:26.06 | *** join/#asterisk nassy (n=nassy@207-38-197-201.c3-0.wsd-ubr1.qens-wsd.ny.cable.rcn.com) |
| 09:26.22 | BugKham | _Vile, delmar: didn't realise that it's "escape digits", the plural one |
| 09:26.46 | BugKham | probably, it's my english problem |
| 09:26.57 | bXi | okay i finally know that my isdn card is loaded in asterisk |
| 09:27.00 | bXi | how does one test it? |
| 09:27.25 | _Vile | delmar, 801, 802 ... registered phones, go ahead |
| 09:27.43 | delmar | So, two Polycom phones .. 801 and 802... .. 801 sets call forward (within the phone) to 802... ... 801 gets a call, and tells asterisk... we see Got SIP response 302 "Moved Temporarily" on the console.. everything works great... until... voicemail comes along... we get 802's voicemail... when we should get 801's voicemail... |
| 09:27.57 | delmar | the solution to this is to remove the "Answer" part of the dialplan for 802.... |
| 09:28.02 | delmar | and I understand why this is.. |
| 09:28.04 | delmar | but when we do this... |
| 09:28.09 | delmar | we loose our ring tone. |
| 09:28.14 | _Vile | no |
| 09:28.17 | delmar | but thats also no big deal.... |
| 09:28.28 | _Vile | give me one prob at a time |
| 09:28.30 | _Vile | first of all |
| 09:28.31 | _Vile | a 302 |
| 09:28.39 | _Vile | will push a phone to the next |
| 09:28.44 | delmar | yeah that works fine.. |
| 09:28.47 | delmar | there is no problem there... |
| 09:28.51 | _Vile | so by default voicemail will hit on 803 |
| 09:28.54 | _Vile | er 802 |
| 09:28.59 | delmar | no.. not the case |
| 09:29.01 | _Vile | ok |
| 09:29.08 | delmar | let me show ya.. |
| 09:29.10 | _Vile | now next probsec let me read |
| 09:29.11 | delmar | juse a sec |
| 09:29.16 | _Vile | oh |
| 09:29.22 | _Vile | thatdoesn't work? |
| 09:29.26 | _Vile | that must work |
| 09:29.35 | _Vile | if a 302 is handled |
| 09:29.46 | _Vile | then it forwards to the next phone |
| 09:29.46 | delmar | ok here is exactly what happens... |
| 09:29.51 | _Vile | i would look at config |
| 09:29.56 | _Vile | for 802 |
| 09:29.59 | delmar | 801 and 802 have lines in the dialplan like this.. exten => s,1,Dial(SIP/801,30,rt) |
| 09:30.02 | diablopico | can anyone tell me what a normal g729 packet size would be ? |
| 09:30.16 | _Vile | 801 is 302 forwarded.... |
| 09:30.17 | nicox | anyone tested res_snmp? |
| 09:30.18 | [shodan] | anyone got videophones to suggest ? |
| 09:30.34 | delmar | if we add yeah 801 is 302 forwarded care of the Polycom |
| 09:30.48 | delmar | ok.. sorry.. explain further i guess... |
| 09:30.54 | delmar | ill start again.. lol |
| 09:31.00 | _Vile | hah ok |
| 09:31.21 | delmar | exten => 801,1,Goto(801-menu,s,1) |
| 09:31.37 | delmar | THEN... under [801-menu]... |
| 09:31.45 | delmar | we can have two ways... |
| 09:32.26 | delmar | first way... just have s,1,Dial(SIP/801,30,rt) .. then the following lines to handle voicemail.. and the "o" to handle operator (calls cell phone if 0 is pressed from voicemail) etc etc |
| 09:32.28 | delmar | OR.. |
| 09:32.49 | delmar | we add exten => s,1,Answer first... |
| 09:33.09 | delmar | IF we use the nasty "Answer" at the top... |
| 09:33.30 | delmar | we get nice .. correct.. ring tones .. care of the "r" on the Dial command |
| 09:33.40 | _Vile | rt is ring and transfer ability... |
| 09:33.48 | _Vile | not the prob |
| 09:34.00 | delmar | no. "r" generates ring tones. t is the transfer |
| 09:34.09 | _Vile | rt together |
| 09:34.11 | delmar | hence rt |
| 09:34.14 | _Vile | is ring and transfer |
| 09:34.24 | _Vile | not the prob. |
| 09:34.39 | _Vile | by ring |
| 09:34.45 | _Vile | i mean gen ring |
| 09:34.48 | delmar | "r" instructs * to generate ring tones... t means allow the called party to initiate a transfer.. |
| 09:35.02 | delmar | yes. r = generate ring tones to the calling party.. |
| 09:35.03 | Jeffjohnson | r also generate ring tone, if the called phone don't ring .o or if it is a mobile and the mobile is powered off |
| 09:35.05 | _Vile | by transfer it allows the called party |
| 09:35.06 | delmar | lets not go into that too much... |
| 09:35.13 | _Vile | to transfer the cal |
| 09:35.15 | _Vile | l |
| 09:35.18 | _Vile | but yeah |
| 09:35.20 | delmar | anyway !! lol |
| 09:35.26 | _Vile | not worth looking at |
| 09:35.30 | _Vile | yes |
| 09:35.34 | _Vile | anyway |
| 09:35.41 | delmar | lets get down to the real problem... this "Answer" command.. |
| 09:35.52 | delmar | so.. 801 is 302'd to 802 .... |
| 09:36.00 | _Vile | right |
| 09:36.49 | _Vile | 802 doesn't answer, should go to voicemail..? |
| 09:37.13 | delmar | if we do NOT have the "Answer" .. we loose the ring tone generation... fine... when we have this removed.. and ring 801... it diverts to 802... and eventually.. gets 801's voicemail.. YAY.. good.. because.. it was the person at 801 that the calling party wanted... |
| 09:37.44 | _Vile | Answer on 802 ahh |
| 09:37.55 | delmar | when we have "Answered" invovled in the little dialplan for the extension.... we are screwed.... |
| 09:38.04 | delmar | it will 302 to 802, then hit 802's voicemail |
| 09:38.09 | _Vile | remove answer from 801 and 802 |
| 09:38.18 | delmar | right... that fixes the problem... |
| 09:38.19 | delmar | now.. |
| 09:38.20 | _Vile | have an answer on the channel |
| 09:38.25 | _Vile | incoming channel |
| 09:38.57 | _Vile | no answer on the extensions |
| 09:39.16 | delmar | yep.. so.. all fixed... 800 calls 801, gets 302 Redirected to 802, and ... the caller on 800 gets 801's voicemail... |
| 09:39.32 | delmar | this is perfect... sadly we can't have the ring tones... but it works.... |
| 09:39.38 | delmar | but there is still a huge problem... |
| 09:39.51 | _Vile | make sure you do a dial w/ a rt |
| 09:40.06 | _Vile | when dialing 801 and 802 |
| 09:40.09 | delmar | same scenario... but now a call comes in on the PSTN or on the DID .. (whatever)... whats the first thing we gotta do before it hits the IVR? "ANSWER" !!!! |
| 09:40.21 | _Vile | yes |
| 09:40.22 | _Vile | but |
| 09:40.27 | _Vile | you answer the channel |
| 09:40.31 | _Vile | and THEN |
| 09:40.33 | delmar | the moment it does that... we are screwed again... person on the IVR dials 801... gets 802... then gets 802's voicemail |
| 09:40.37 | _Vile | redirect ot the exten |
| 09:40.58 | _Vile | ivr |
| 09:41.05 | _Vile | first of all |
| 09:41.10 | _Vile | when someone calls you |
| 09:41.12 | _Vile | you answer. |
| 09:41.21 | delmar | yup. cant be avoided |
| 09:41.33 | _Vile | if the person dials 801 |
| 09:41.33 | delmar | line needs to be answered for the IVR..... |
| 09:41.35 | _Vile | or 802 |
| 09:41.40 | _Vile | or 803 or whatever |
| 09:41.45 | delmar | yep... |
| 09:41.50 | _Vile | that line is already answered |
| 09:41.53 | delmar | yup |
| 09:42.00 | _Vile | in your 801 or 802 extens |
| 09:42.05 | _Vile | get rid of answer. |
| 09:42.10 | _Vile | no need |
| 09:42.12 | delmar | that is the case already... |
| 09:42.18 | _Vile | ok |
| 09:42.31 | _Vile | then it should work if the timeout is set correctly |
| 09:42.32 | delmar | once the call was answered before the IVR.. we were screwed at that point |
| 09:42.42 | _Vile | and you are timing out correctly |
| 09:42.47 | _Vile | ahh |
| 09:43.00 | delmar | I can still see why it does this :P |
| 09:43.02 | _Vile | then the problem is before the IVR |
| 09:43.12 | delmar | the problem is the use of "Answer" |
| 09:43.33 | *** join/#asterisk bhrobinson (n=brobinso@mail1.nt-it.com) |
| 09:43.41 | _Vile | doubt it, www.pastebin.ca your extensions.conf |
| 09:43.52 | _Vile | ill tell you how to fix |
| 09:43.56 | delmar | not easy. too much sensitive stuff in it |
| 09:44.01 | _Vile | heh |
| 09:44.04 | _Vile | ok |
| 09:44.14 | bhrobinson | is there anyone here that can help me sync my t1 card to my phone system? |
| 09:44.20 | _Vile | show me your entry point on pastebin.ca |
| 09:44.28 | _Vile | the inbound call |
| 09:44.36 | _Vile | show me the ivr menu |
| 09:44.40 | delmar | ive tested this quite extensivly... as i say.. I do understand why it works like this... i just want to work around it |
| 09:44.42 | _Vile | and show me how it gets to 801 |
| 09:44.53 | _Vile | and private msg me |
| 09:45.08 | delmar | just a sec. ill whip something up. |
| 09:45.27 | _Vile | bh no, but what's the problem? |
| 09:46.38 | bhrobinson | _Vile, I spent 5 hours with someone else on this.. the D channel was having troubles connecting. I finally put the card in another slot and it came up no problem |
| 09:46.56 | bhrobinson | issue now is I have a Samsung phone system that has a PRI card. |
| 09:47.21 | bhrobinson | I want to hook the TE210P from the asterisk to the samsung on port 2 |
| 09:47.37 | bhrobinson | and from the asterisk to the IAD(Adit 600) on port 1 |
| 09:49.15 | _Vile | putting the card in another slot was not a software problem |
| 09:49.30 | _Vile | samsung |
| 09:49.35 | bhrobinson | agreed, but I have never hooked up what I am talking about |
| 09:49.37 | _Vile | does it do PRI? |
| 09:49.44 | bhrobinson | yes |
| 09:49.48 | _Vile | ok |
| 09:49.56 | _Vile | do you know what class of service? |
| 09:50.02 | _Vile | 5ess, dms, etc? |
| 09:50.27 | _Vile | as in |
| 09:50.28 | bhrobinson | yeah. It is the same setup that I have the asterisk setup as. |
| 09:50.37 | _Vile | what type of signalling does it want |
| 09:50.44 | bhrobinson | b8zs |
| 09:50.50 | _Vile | esf b8zs |
| 09:50.51 | _Vile | yes |
| 09:50.53 | _Vile | but |
| 09:51.04 | _Vile | you need to know what switchtype it wants |
| 09:51.14 | bhrobinson | national |
| 09:51.23 | _Vile | good |
| 09:51.54 | _Vile | ok |
| 09:52.06 | _Vile | checked timing in /etc/zaptel.conf? |
| 09:52.40 | bhrobinson | yeah. If I pull the t1 cable from port1 and plug it straight into the samsung, it works fine |
| 09:55.54 | *** join/#asterisk zedkatuf (n=zedkatuf@82-32-57-69.cable.ubr08.azte.blueyonder.co.uk) |
| 09:57.44 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.221) |
| 09:57.54 | _Vile | no |
| 09:57.58 | _Vile | not what i mean |
| 09:58.03 | _Vile | check /etc/zaptel.conf |
| 09:58.18 | *** join/#asterisk ptblank (n=MURDER1@68-169-166-65.lmdaca.adelphia.net) |
| 09:58.19 | _Vile | to see if you are "providing timing" or "pulling timing" from the span |
| 09:58.38 | _Vile | I think it is called master/slave on timing, not sure on ur samsung |
| 09:58.48 | _Vile | bbiafm |
| 09:58.51 | *** join/#asterisk backblue (n=igor@82.102.1.42) |
| 09:59.02 | bhrobinson | ok |
| 09:59.05 | backblue | how do i send one channel down? on a running call? |
| 10:00.03 | meppl | guten morgen |
| 10:03.19 | bhrobinson | ok |
| 10:03.29 | bhrobinson | got the status up, but not reading anything from it |
| 10:04.57 | *** join/#asterisk evol-emil (n=emile@landi.oddi.is) |
| 10:07.03 | *** join/#asterisk _Vile (n=vile@90.b160.bendtel.net) |
| 10:09.27 | bhrobinson | _vile, it has connected up.... but nothing coming across |
| 10:10.01 | Newbie___ | hi, anyone using a Zhone 24FXS ? |
| 10:10.08 | _Vile | explain |
| 10:10.36 | _Vile | like |
| 10:10.41 | _Vile | calls not coming across? |
| 10:10.51 | bhrobinson | on the phone system, if I dial 708 (trunk 1) and then dial a number, I get a CO disconnect on the phone, and nothing reported on the asterisk |
| 10:11.10 | Newbie___ | _Vile: i am getting a red alarm on zhone and TE110 is not detecting it |
| 10:12.00 | _Vile | bh, is the light green on the back of the card? |
| 10:12.04 | bhrobinson | yes |
| 10:12.06 | *** join/#asterisk moon06 (n=michael@cim06-1-82-228-240-97.fbx.proxad.net) |
| 10:12.12 | _Vile | Newbie sec |
| 10:12.15 | Newbie___ | _Vile: no, is red all the way |
| 10:12.31 | bhrobinson | _vile, asterisk1*CLI> pri show spans |
| 10:12.31 | bhrobinson | PRI span 1/0: Provisioned, Up, Active |
| 10:12.31 | bhrobinson | PRI span 2/0: Provisioned, Up, Active |
| 10:12.33 | _Vile | bh, then your problem is now a config issue |
| 10:13.09 | nicox | do anybody know, when the res_snmp is coming into the svn-trunk? |
| 10:13.10 | Newbie___ | _Vile: i plug a straight cable onto WAN1 on zhone. is that right? |
| 10:13.34 | _Vile | Try a T-1 Cross-over Newbie |
| 10:13.50 | Newbie___ | did that, and same |
| 10:14.02 | _Vile | ok |
| 10:14.20 | _Vile | Seems like a config issue |
| 10:14.31 | Newbie___ | _Vile: config on zhone or * ? |
| 10:14.41 | _Vile | config |
| 10:15.03 | _Vile | need to modify /etc/zaptel.conf |
| 10:15.10 | _Vile | and re-run ztcfg |
| 10:15.11 | [shodan] | what is parking for ? if you receive a call and forward it to the parking so that another user can answer the call , then why not transfert the call directly to the other person ? |
| 10:15.32 | _Vile | probably need to make the span |
| 10:15.35 | _Vile | loopsatart |
| 10:15.43 | _Vile | loopstart |
| 10:15.55 | _Vile | 1-24 as the channels |
| 10:16.08 | _Vile | span 1 I assume |
| 10:16.25 | _Vile | esf |
| 10:16.28 | _Vile | b8zs |
| 10:16.36 | tzafrir | nicox, why not see for yourself? |
| 10:16.38 | *** join/#asterisk moonaddict (i=b864dd6a@213.129.253.62) |
| 10:16.43 | _Vile | PROVIDE timing |
| 10:17.00 | nicox | where? |
| 10:17.03 | _Vile | and make sure the channel bank is setup to pull timing |
| 10:17.18 | moonaddict | hey all. I have a new asterisk (trixbox) setup with 1 pstn line to the outsinde world connected via zaphfc interface |
| 10:17.24 | tzafrir | nicox, http://svn.digium.com/svn/asterisk/trunk/res/ |
| 10:17.35 | tzafrir | res_snmp.c is clearly there |
| 10:18.05 | moonaddict | now I am seeing "received TEI check request for TEI = 127" or "= 0" all the time in * and * answers all outbound dialling with "Everyone is busy/congested at this time (1:0/1/0)" |
| 10:18.11 | Newbie___ | _Vile: bought it off ebay and was told is pre configure. just hope is true |
| 10:18.19 | hads|home | There is even a doc/snmp.txt and configs/res_snmp.conf.sample |
| 10:18.23 | moonaddict | I cannot quite follow what's going on here. |
| 10:18.58 | hads|home | moonaddict: If you are using trixbox you may get more help in #freepbx |
| 10:19.55 | *** join/#asterisk Bambr (n=Bambr@213-35-236-25-dsl.end.estpak.ee) |
| 10:20.06 | _Vile | Newbie |
| 10:20.20 | _Vile | the zhone is probably pulling clock from the span |
| 10:20.35 | _Vile | which means zaptel needs to provide the clock |
| 10:21.03 | Newbie___ | _Vile: any ideas ? |
| 10:21.07 | _Vile | and it's probably b8zs |
| 10:21.10 | _Vile | esf |
| 10:21.23 | _Vile | and loopstart |
| 10:21.36 | _Vile | which means read /etc/zaptel.conf |
| 10:21.56 | _Vile | it's a one liner |
| 10:22.29 | _Vile | then once that's done |
| 10:22.41 | _Vile | run ztcfg again |
| 10:22.56 | Jeffjohnson | what hardware i need for 10 telephone conversions concurrently? alaw/ulaw codec need less cpu load than compressed codecs, right? |
| 10:23.07 | Newbie___ | _Vile: as in ztcfg -vv |
| 10:23.13 | _Vile | sure |
| 10:23.16 | *** join/#asterisk _omer (i=_omer@202.166.161.23) |
| 10:23.21 | _Vile | that gives verbose |
| 10:23.22 | Newbie___ | _Vile: tks. trying out |
| 10:23.43 | _Vile | now you gotta modify zaptel.conf |
| 10:23.46 | _Vile | for this to work |
| 10:24.00 | _Vile | but it gives you a lot of abilities |
| 10:24.07 | _Vile | and is very well documented |
| 10:24.46 | _Vile | you need to be master for timing on *THAT* span, which i think is 1, 0 on your other span |
| 10:25.05 | Newbie___ | _Vile: i have span=1,0,0,esf,b8zs |
| 10:25.10 | _Vile | esf, b8zs, that should be default |
| 10:25.11 | _Vile | ok |
| 10:25.28 | _Vile | try span=1,1,0,esf,b8zs |
| 10:25.34 | _Vile | on your first span |
| 10:25.57 | _Vile | on your second use span=1,0,0,esf,b8zs |
| 10:26.12 | _Vile | sorry |
| 10:26.17 | _Vile | span=2,0,0,esf,b8zs |
| 10:26.28 | _Vile | and use your second as your upstream |
| 10:26.32 | _Vile | or vice versa |
| 10:27.14 | Aurs | when a cell phone caller has 2 caller ids... how do you decide which one to use? asterisk seems to use the last one, but "everybody else" is using the first one |
| 10:27.20 | _Vile | and you have a lit T-1 at work? |
| 10:27.31 | Aurs | possibly a norwegian phenomenon, but.. any ideas? |
| 10:27.34 | _Vile | if so, use a straight-through |
| 10:27.48 | _Vile | do not use a T-1 crossover |
| 10:28.25 | _Vile | an inbound call will always have only one caller id. |
| 10:28.29 | Aurs | nope |
| 10:28.29 | Newbie___ | _Vile: i only have 1 span TE110P |
| 10:28.39 | Aurs | _Vile: i can show you the pri debug :) |
| 10:28.56 | _Vile | Aurs, they may have an ani and an ani2... |
| 10:29.07 | _Vile | that is not caller id. |
| 10:29.16 | _Vile | please do. |
| 10:29.20 | Aurs | ok |
| 10:29.25 | Aurs | http://pastebin.com/773789 |
| 10:29.27 | Aurs | line 26 |
| 10:29.33 | Aurs | or 27.. hehe |
| 10:30.11 | _Vile | Newbie... one span... ok.. provide timing use my second example, span=1,1,0,esf,b8zs. |
| 10:30.34 | Aurs | the thing is.. that when this guy calls "regular" phone systems, the 1st number is shown (99xxxx), but when he calls to voip, the second number is shown |
| 10:30.40 | Aurs | and i have no clue on why |
| 10:31.03 | Newbie___ | _Vile: ok |
| 10:31.18 | _Vile | Aurs, unknown number type, that url doesn't show me much |
| 10:31.24 | _Vile | ine 27 |
| 10:31.45 | _Vile | ?? |
| 10:32.06 | _Vile | bad outbound calling patterns is looks to me on the q.931 crap |
| 10:32.15 | Aurs | lines 29 and 35 |
| 10:32.24 | _Vile | u sure that's the right paste? |
| 10:32.46 | _Vile | ahhh |
| 10:32.48 | _Vile | hmm |
| 10:32.50 | _Vile | sec |
| 10:32.55 | Newbie___ | _Vile: still RED |
| 10:33.08 | _Vile | newbie |
| 10:33.38 | *** join/#asterisk Irulka (n=irina@213-35-236-25-dsl.end.estpak.ee) |
| 10:33.41 | *** join/#asterisk Nebukadneza (n=daddel9@i3ED6F487.versanet.de) |
| 10:34.15 | _Vile | Newbie right after the span line |
| 10:34.16 | _Vile | fxols=1-20 |
| 10:34.25 | _Vile | fxols=1-24 actually |
| 10:34.32 | _Vile | got that there? |
| 10:34.35 | Aurs | _Vile: what do you mean, "right paste" ? |
| 10:34.37 | Newbie___ | yup |
| 10:34.40 | _Vile | ok |
| 10:34.46 | _Vile | double check your connections |
| 10:35.04 | _Vile | you should be green at this point |
| 10:35.10 | _Vile | do a loop back at the far-end |
| 10:35.36 | *** join/#asterisk RoyK (n=roy@213.160.242.91) |
| 10:36.08 | _Vile | Aurs, I was curious, sec let me read the paste, I only glanced |
| 10:36.12 | Newbie___ | Changing signalling on channel 1 from FXO Kewlstart to FXO Loopstart |
| 10:36.23 | _Vile | Newbie you need FXS |
| 10:36.36 | _Vile | actually |
| 10:36.39 | Newbie___ | yes, i need fxs |
| 10:36.44 | Newbie___ | 24 fxs |
| 10:36.44 | _Vile | ok |
| 10:37.24 | Newbie___ | hrm. still getting red |
| 10:37.25 | *** join/#asterisk daaku (n=daaku@202.88.167.108) |
| 10:37.32 | Aurs | _Vile: sure. just say "Aurs" if you find anything that makes sense. bottom line: when he calls my users, the second number (46....) is shown. when he calls other cell phones, or the national PSTN, the first one is presented (99....) |
| 10:37.39 | _Vile | well if you are red on the t-1 |
| 10:37.50 | _Vile | then it's a cable? issue |
| 10:38.08 | Newbie___ | in zapata.conf i have not chage the signallig |
| 10:38.29 | _Vile | Newb sec let me finish |
| 10:38.34 | Newbie___ | ok |
| 10:38.39 | _Vile | Aurs |
| 10:38.56 | _Vile | What ANI are you presenting? |
| 10:39.07 | _Vile | 46 or 99? |
| 10:39.09 | _Vile | And |
| 10:39.22 | _Vile | What does the telco GIVE you as your BTN? |
| 10:39.52 | _Vile | It feels like |
| 10:40.02 | _Vile | your Telco is presenting an ANI for the 99 number |
| 10:40.28 | _Vile | you're probably (maybe) passing an ANI for a 46, Telco is ignoring, passing the 99 as the ANI |
| 10:40.54 | _Vile | And that's why it appears correctly to your users, not to the telco |
| 10:41.12 | _Vile | s/telco/PSTN |
| 10:41.15 | Irulka | does anybody know what changes are made in call transfer in the new version of * 1.2.11? on the 1.2.9.1 transfers were made successfully, but now i get: "WARNING[3476]: chan_sip.c:2561 sip_write: Asked to transmit frame type 4, while native formats is 2 (read/write = 4/4)" when trying to do that... |
| 10:41.30 | *** join/#asterisk inspired (n=mikael@85.221.0.46) |
| 10:41.34 | _Vile | Aurs. |
| 10:41.36 | daaku | hi all |
| 10:43.10 | _Vile | OK Newb |
| 10:43.21 | _Vile | Do you have a T-1 Loopback PLug? |
| 10:43.31 | Newbie___ | no |
| 10:43.33 | *** join/#asterisk muppetmaster (n=jasongoe@169.red-81-184-73.user.auna.net) |
| 10:43.33 | _Vile | Or cable.. doesn't matter. |
| 10:43.50 | _Vile | ok, a) get one or b) create one |
| 10:43.58 | Newbie___ | b) |
| 10:44.07 | _Vile | on a t-12 |
| 10:44.09 | _Vile | t-1 |
| 10:44.23 | _Vile | you use pairs 1,2 rx,tx 3,5 rx,tx |
| 10:44.38 | _Vile | 1,2 need to connect to 3,5 on both ends |
| 10:44.48 | _Vile | easy cross-connect |
| 10:44.59 | Newbie___ | ok |
| 10:45.12 | Newbie___ | 1-3 , 2-5 |
| 10:45.28 | _Vile | wait |
| 10:45.46 | _Vile | that's the cross-connect spec for a t-1 cross-over |
| 10:45.52 | Aurs | yes _Vile? |
| 10:45.55 | _Vile | im too tired. |
| 10:46.01 | _Vile | Aurs /|\ |
| 10:46.16 | _Vile | 3:38 msgs |
| 10:46.30 | Aurs | hehe |
| 10:46.31 | Newbie___ | _Vile: that will be on 1 RJ45 |
| 10:46.45 | _Vile | Newb ok you got it then :) |
| 10:47.00 | Aurs | _Vile: it is a special kind of cell phone thing. translated to something like "twin sim" or something |
| 10:47.18 | axscode | anyone has a copy of default extensions.conf ? |
| 10:47.40 | _Vile | Aurs, you know the diff betw caller id, ani, and ani2 right? |
| 10:48.16 | Aurs | not really |
| 10:48.35 | _Vile | it's all one number |
| 10:48.45 | _Vile | caller id shows up on your phone |
| 10:48.57 | _Vile | ani is the "BILLABLE" number |
| 10:49.03 | _Vile | the number that called you |
| 10:49.13 | _Vile | which may or may not show up on your phone |
| 10:49.37 | Aurs | it shows up allright |
| 10:49.50 | bhrobinson | Vile, |
| 10:49.55 | Aurs | but if this guy call me on my cell phone, it shows the 99 number |
| 10:50.01 | Newbie___ | _Vile: i am ready, u want me to plug in to TE110P |
| 10:50.11 | Aurs | and when he calls on my voip number (asterisk), it shows the 46 number |
| 10:50.21 | daaku | i'm looking to setup something where i can call a SIP phone and use that as a bridge to make calls through a normal (non-SIP local) line [kinda new to this, not sure if i'm using the right terms] |
| 10:50.28 | Aurs | but I guess I've already pointed that out |
| 10:50.29 | bhrobinson | I want to send anything coming on port 1 of the TE210 to port 2 of the TE210... how hard is that to do? |
| 10:50.37 | _Vile | ani2 would be like an rdnis or something of that nature |
| 10:50.40 | *** join/#asterisk lorinc (n=ang@caracas-1810.adsl.interware.hu) |
| 10:50.40 | _Vile | newb yeah |
| 10:50.45 | Newbie___ | ok |
| 10:51.12 | daaku | basically i wanna call a SIP phone in another country (which is really cheap) - and call non-VoIP phones through that which are local to that country |
| 10:51.16 | _Vile | do you have a female<->female rj45 connector for the other end as well to check that end of the cable? |
| 10:51.33 | Newbie___ | _Vile: i'll be damn .is green now |
| 10:51.42 | _Vile | your cable |
| 10:51.44 | _Vile | ;) |
| 10:51.56 | Newbie___ | grrrr |
| 10:52.03 | _Vile | 1,2,3,5 |
| 10:52.09 | _Vile | straight through |
| 10:52.15 | Newbie___ | as i did earlier ? |
| 10:52.18 | Newbie___ | 1-3 , 2-5 |
| 10:52.18 | _Vile | or the t-1 from the channel banks doesn't work |
| 10:52.23 | _Vile | yeah |
| 10:52.32 | _Vile | s/banks/bank |
| 10:52.46 | _Vile | sec daak |
| 10:52.48 | Newbie___ | was told a cross over |
| 10:52.50 | _Vile | let me read |
| 10:52.52 | _Vile | no |
| 10:52.58 | _Vile | well |
| 10:53.07 | _Vile | no should be straight |
| 10:53.11 | _Vile | from a cb |
| 10:53.18 | Newbie___ | roger that |
| 10:54.48 | *** join/#asterisk Dr-Linux|work (n=Linux@202.125.139.198) |
| 10:54.56 | bhrobinson | _Vile, can you tell me easily how to route all calls from port1 of my TE210 to Port2 of my TE210 and vice versa? |
| 10:55.16 | *** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust944.leed.cable.ntl.com) |
| 10:56.19 | Newbie___ | run out of RJ45 |
| 10:56.19 | _Vile | you need to answer in * and direct to the corrresponding Zap ports on 2 |
| 10:56.54 | _Vile | im going to bed |
| 10:56.59 | _Vile | daaku |
| 10:57.01 | muppetmaster | Goodnight |
| 10:57.02 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 10:57.04 | bhrobinson | thanks |
| 10:57.11 | Newbie___ | _Vile: thanks a bunch |
| 10:57.17 | _Vile | explain |
| 10:57.46 | _Vile | daaku |
| 10:57.54 | _Vile | I gotta defer off |
| 10:58.05 | _Vile | to someone else |
| 10:58.13 | _Vile | it is 4am here |
| 10:58.23 | bhrobinson | you talking to me? |
| 10:58.36 | _Vile | night yall |
| 10:58.40 | bhrobinson | that was sincere... I appreciate it |
| 11:00.09 | puzzled | morning all |
| 11:00.21 | Aurs | g'nite _Vile |
| 11:01.43 | *** join/#asterisk Newbie___ (n=me@211.24.146.11) |
| 11:03.30 | Dr-Linux|work | _Vile, who is daaku? |
| 11:04.46 | _Vile | he's in the chan |
| 11:05.12 | _Vile | having problems I suspect w/ CID & ANI & ANI2 sep |
| 11:06.00 | bXi | are there any known bugs with grandstream handytone 286 devices? |
| 11:06.12 | bXi | whenever i call another phone it hangs up immediatly |
| 11:06.21 | bXi | an asterisk doesnt give any info whatsoever |
| 11:07.16 | _Vile | later |
| 11:07.20 | daaku | _Vile: still here |
| 11:08.08 | daaku | _Vile: sorry, wasnt looking |
| 11:08.23 | daaku | _Vile: but i can come back tomorrow |
| 11:09.05 | _Vile | daaku - Dr-Linux will help you as much as he can, if not will refer you to someone else |
| 11:09.18 | daaku | _Vile: cool - thanks |
| 11:09.27 | _Vile | im hittin the sack |
| 11:09.42 | daaku | _Vile: later |
| 11:09.44 | _Vile | 4:10am here |
| 11:09.48 | _Vile | later |
| 11:10.06 | daaku | _Vile: 4:40pm here - :D |
| 11:10.13 | viperdude | hi i am getting random audio dropouts on calls... any ideas what i need to check? |
| 11:11.00 | Jeffjohnson | _Vile: Dr linux, do you have an idea why I don't have an dial tone when calling from an ISDN phone with dusnet as provider, but I have an dialtone when i use sipgate.de as provider. With an connected VoIP phone i have a dialtone with both providers |
| 11:12.03 | puzzled | Jeffjohnson: it's not surprising that you have a dialtone with a VoIP phone since the phone generates the tone. With an ISDN phone the tone is generated by the switch |
| 11:12.58 | Jeffjohnson | puzzled: ok, so what to do to have correct dialtone with an isdn phone? |
| 11:13.20 | Jeffjohnson | puzzled: ringing application in dialplan? :o |
| 11:13.31 | *** join/#asterisk parag_ast (n=root@dxb-b16451.alshamil.net.ae) |
| 11:13.34 | puzzled | Jeffjohnson: afaik you can not do anything. the switch does it and in a normal situation you can't change the switch |
| 11:14.38 | Jeffjohnson | puzzled: but the support from the voip provider sayed me that it should work :) they test it with asterisk+isdn phone |
| 11:14.38 | *** join/#asterisk scubasteve (n=steve@ns1.misel.com) |
| 11:14.42 | scubasteve | Good morning! |
| 11:14.51 | puzzled | Jeffjohnson: to what is your ISDN phone connected? asterisk box with ISDN cards? |
| 11:15.02 | puzzled | scubasteve: morning |
| 11:15.10 | parag_ast | My some zap channels don't hangup and when i manually hangup i get the error 481 "Call leg/trasanctions dosn't exists"... It generally happens in incomming call... |
| 11:15.20 | Jeffjohnson | puzzled: to an telephone system |
| 11:15.29 | scubasteve | Does anyone know how to do SIP-CGI with OpenSER? I've read all about it, completely understand it, but can't find any docs on configuring SER to use it. |
| 11:16.09 | puzzled | Jeffjohnson: so your ISDN phone is connected to a PBX which is connected to an Asterisk box which connects with your VoIP providers? |
| 11:16.18 | Jeffjohnson | puzzled: yes |
| 11:17.27 | puzzled | Jeffjohnson: ok, then I guess you need to compare the Dial statements in your Asterisk box to both your VoIP providers and see if you can spot the difference |
| 11:18.33 | Jeffjohnson | puzzled: k I will try it, thx :) the providers support also writes me that I must set "progressinband=yes", but it doesn't help |
| 11:18.46 | puzzled | scubasteve: what's SIP-CGI? |
| 11:19.40 | scubasteve | puzzled: It's like AGI/CGI for SER. |
| 11:20.17 | *** join/#asterisk gitano3344 (n=yo@62.36.227.220) |
| 11:20.18 | puzzled | Jeffjohnson: while you are searching for the solution you could use the "r" option in the Dial statement to dusnet |
| 11:20.35 | puzzled | so Asterisk provides the ringing |
| 11:20.48 | Jeffjohnson | puzzled: yes, Im using allready the r paramter :) buts problematic with mobile phones that powered off |
| 11:20.56 | puzzled | scubasteve: ah right, the CGI part sounded familiar :) |
| 11:22.02 | *** join/#asterisk Mandrak3 (n=io@81.27.211.30) |
| 11:22.07 | axscode | hi guyz.. whats wrong if F1000G UTStarcom wifi fone to wifi fone aint workin?? |
| 11:22.27 | puzzled | Jeffjohnson: then use a GotoIf. If the recipient is a mobile phone then use a dial statement without "r", else use "r" |
| 11:22.56 | puzzled | axscode: probably because the F1000G is a piece of crap :) |
| 11:23.13 | puzzled | axscode: the least you should do is upgrade it to the latest firmware version |
| 11:23.32 | Jeffjohnson | puzzled: no, calling without r is bad :) cause you don't really now what the dial state is atm, and than suddenly you hear a voice :o |
| 11:23.54 | puzzled | Jeffjohnson: can imagine that spooks people a bit :) |
| 11:26.09 | RoyK | viperdude: install a jitterbuffer |
| 11:26.40 | Mandrak3 | Hi everybody.... Anyone knows how to check if an outgoing trunk is busy or unavalaible? I'm trying to set up my LCR engine |
| 11:27.15 | puzzled | show application chanisavail or something like that |
| 11:28.02 | Mandrak3 | I tried chanisavail..... but it returns always "0" value |
| 11:28.21 | Mandrak3 | in AVAILSTATUS |
| 11:29.17 | puzzled | is works for channels in a trunk. dunno about an entire trunk |
| 11:29.23 | Mandrak3 | in the manual is written that chaisavail is not useful for this purpose |
| 11:29.44 | puzzled | just do failover. search the list or check voip-info.org |
| 11:29.52 | *** join/#asterisk saftsack (n=saftsack@p54A7DB5B.dip.t-dialin.net) |
| 11:31.06 | Mandrak3 | uhm... I'll search better! |
| 11:31.06 | axscode | exten => _777XXXXX,1,DIAL(SIP/${EXTEN},20,rt) <-- this will call a any number that starts why 777 right? or somethings missing? |
| 11:31.29 | *** join/#asterisk benjk (n=benjamin@f8a01-0357.din.or.jp) |
| 11:32.00 | puzzled | it will call a number starting with 777 and that consists of a total of 8 digits |
| 11:33.28 | axscode | yups... my problem is it says... no destination? |
| 11:35.13 | Jeffjohnson | puzzled: the problem is dusnet only sends an 183 and no 183 ringing. But progressinband=yes don't fix this... but it should? :o |
| 11:36.51 | puzzled | jeffgus: then I guess you should file a bug |
| 11:37.00 | Jeffjohnson | puzzled: as per "http://bugs.digium.com/print_bug_page.php?bug_id=4105" its providers fault? |
| 11:37.03 | puzzled | oops stupid nick completion |
| 11:39.53 | puzzled | Jeffjohnson: I agree with what kpfleming said at the end. If dusnet sends inband easly audio with preceding it with a 183 ringing than they should fix it |
| 11:40.08 | puzzled | s/with/without |
| 11:40.17 | Jeffjohnson | puzzled: they send an 183 ringing |
| 11:40.34 | Jeffjohnson | :) |
| 11:40.48 | puzzled | ah :) then file a bug with the dialplan and a trace |
| 11:40.52 | puzzled | and the config used |
| 11:41.15 | Jeffjohnson | don't be sure it is a bug, or my fault .o |
| 11:43.56 | *** join/#asterisk rcsw (n=richard@mail.shout-telecoms.com) |
| 11:44.01 | puzzled | Jeffjohnson: so if you hook up a SIP phone to the asterisk box and call dusnet than you hear ringing? |
| 11:44.39 | ESCulapio__ | hola quien habla espanol |
| 11:45.48 | Mandrak3 | How can I see asterisk mailing list? |
| 11:46.09 | puzzled | ESCulapio__: no habla espanol. but there is the #asteriskbrasil.org channel. maybe they can help you |
| 11:46.50 | ESCulapio__ | puzzled, ok thanks |
| 11:46.57 | puzzled | Mandrak3: http://lists.digium.com/mailman/listinfo/ |
| 11:50.29 | *** join/#asterisk brif8 (n=Administ@ns1.ttienterprises.org) |
| 11:51.00 | ESCulapio__ | help my please with Intel Dialogic DI/0408-LS analog |
| 11:52.00 | puzzled | ESCulapio__: did you buy the Dialogic drivers from Digium? |
| 11:52.15 | Jeffjohnson | puzzled: yes with a sip phone it works :) |
| 11:52.44 | puzzled | Jeffjohnson: then I guess the problem must be between the asterisk box and the pbx |
| 11:53.16 | Jeffjohnson | puzzled: but... |
| 11:53.35 | Jeffjohnson | puzzled: the problem appears only with dusnet as provider... with sipgate it works with isdn phones too |
| 11:54.03 | ESCulapio__ | just the business solution suports DI/0408-LS card |
| 11:55.03 | ESCulapio__ | puzzled, just the business solution suports DI/0408-LS card |
| 11:55.11 | puzzled | Jeffjohnson: ah yes that's right. Don't have a clue here. Did you try sending all your config/console output/trace to the mailinglist? |
| 11:55.31 | puzzled | ESCulapio__: then call Digium because the Business Edition comes with support |
| 11:55.35 | Jeffjohnson | puzzled: No, I will try it... thx for your help |
| 11:57.41 | ESCulapio__ | puzzled, I do not have the business version, informed to me that single the business version supports the card that I have |
| 11:58.50 | puzzled | ESCulapio__: maybe you can buy the driver from Digium without the Business Edition |
| 11:59.34 | gitano3344 | Buenas, donde podría encontrar información para un principante en Asterisk? |
| 12:00.01 | ESCulapio__ | gitano3344, voip-info.org |
| 12:00.49 | ESCulapio__ | gitano3344, este canal es solo en ingles puede q te ban si continuamos hablando en espanol |
| 12:01.26 | ESCulapio__ | puzzled, and I am looking for the form to install it in the version that I have the 1.2.9 |
| 12:02.27 | puzzled | ESCulapio__: I don't think the driver is available. I think it is a commercial driver that you need to buy |
| 12:03.12 | ESCulapio__ | no gitano3344 ya yo tengo instalada esa version |
| 12:03.23 | *** join/#asterisk af_ (n=af@ip-173-144.sn1.eutelia.it) |
| 12:03.23 | puzzled | ESCulapio__: are you in Cabarete? |
| 12:03.40 | ESCulapio__ | gitano3344, mi problema esta con una tarjeta intel Dialogic analoga |
| 12:04.21 | ESCulapio__ | gitano3344, pero si puedes ayudarme con gusto |
| 12:05.08 | ESCulapio__ | puzzled, no Santo Domingo Republica Dominicana |
| 12:05.58 | puzzled | ESCulapio__: ah ok. I only have been to Cabarete |
| 12:06.12 | meppl | what do you mean are the best drivers for hfc-cards? |
| 12:06.23 | *** join/#asterisk [TK]D-Fender (n=joe@toronto-HSE-ppp4122655.sympatico.ca) |
| 12:06.36 | meppl | http://66.102.9.104/search?q=cache:TUHnw3SzZ7oJ:www.asteriskguru.com/tutorials/bri.html+asterisk+hfc+tutorial&hl=de&gl=de&ct=clnk&cd=6 |
| 12:07.00 | gitano3344 | anyone can say me why fail my make install? |
| 12:07.05 | *** join/#asterisk burus (n=burus@87.248.161.141) |
| 12:07.06 | gitano3344 | i received this error: /usr/bin/ld: no se puede encontrar -lh323_linux_x86_r |
| 12:07.43 | puzzled | gitano3344: there were h323 bugfixes in the latest asterisk 1.2.11. maybe try that one |
| 12:08.34 | puzzled | meppl: mISDN/chan_misdn does not work on CentOS and RHEL. vISDN does work I'm told |
| 12:08.42 | burus | I have problem with using ExternalIVR app |
| 12:09.33 | ESCulapio__ | gitano3344, puzzled dice q intenten con el asterisk 1.2.11 |
| 12:10.41 | meppl | puzzled, thx |
| 12:11.04 | burus | I'm execute ExternalIVR app with params: ExternalIVR(example.py) |
| 12:12.14 | ESCulapio__ | puzzled, that cabarete is a very pretty place well that this in sosua, in the North part of the island |
| 12:12.15 | *** join/#asterisk murf (n=steve_mu@216.166.159.235) |
| 12:12.26 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 12:12.35 | burus | in the python scrip are string: sys.stdout.write("S,has-been-cleared\n") |
| 12:13.03 | burus | has-been-cleared.gsm - standart sound addon |
| 12:13.28 | ESCulapio__ | somebody can help with a card intel me Dialogic DI/0408-LS |
| 12:13.33 | burus | but ExternalIVR generator doesn't work :( |
| 12:13.55 | burus | plz .. help .. may be it's some bug |
| 12:14.33 | jbalcomb | [TK]D-Fender: you around? |
| 12:14.46 | benjk | Dialogic hardware needs a commercial driver from Digium |
| 12:14.52 | [TK]D-Fender | ~8ball Am I actually here? |
| 12:14.54 | jbot | Negative. |
| 12:15.03 | [TK]D-Fender | jbalcomb: Appears not, sorry! |
| 12:15.07 | benjk | if you have purchased a license for that, you should call Digium for support |
| 12:15.36 | jbalcomb | [TK]D-Fender: aw damn, that sucks. well, i'll PM you and you get get it later when you get here |
| 12:15.37 | [TK]D-Fender | *yawn* |
| 12:15.41 | brif8 | From a latency and network congestion point of view. Is the G.729 better and gives more quality than G.711 or not ? I realize that the packet is smaller 8kbps vs. 64kbps. But won't more smaller packets just mean more congestion or load on the network ? |
| 12:15.57 | benjk | G729 increases latency |
| 12:16.04 | benjk | reduces bandwidth use |
| 12:16.53 | brif8 | benjk: bandwidth is not much of a problem right now I;m using a 7 Mbps fibre connection. but latency is and esp. call quality |
| 12:16.53 | jbalcomb | i thought G.729 was the same information just in smaller packets? |
| 12:17.23 | benjk | G729 takes time to calculate, this adds to latency |
| 12:17.41 | benjk | G711 can send packets out faster |
| 12:17.47 | [TK]D-Fender | brif8: G729 is a much more compressed codec taking up less bandwitdh and noticably LOWERING the quality. |
| 12:17.57 | jbalcomb | What is taking the time for G.729 to calculate? |
| 12:18.14 | [TK]D-Fender | jbalcomb: High data compression. |
| 12:18.17 | benjk | its a very CPU intensive algorithm |
| 12:18.41 | jbalcomb | So we are talking about increasing CPU latency as opposed to network latency? |
| 12:18.49 | brif8 | ok so if you have bandwidth (which in my case I do) the g711 is better than g729 , thanks guys |
| 12:19.22 | benjk | not only that, g729 is patent encumbered and you have to pay for licenses on a per channel basis |
| 12:19.29 | benjk | g711 is free |
| 12:19.50 | jbalcomb | benjk: people like paying for stuff so it's not so much a bother |
| 12:20.11 | benjk | silly if you pay for bogus patents |
| 12:20.25 | benjk | a codec is nothing other than a mathematical formula |
| 12:20.45 | benjk | mathematical formulas are always *explicitly* excluded from patentability |
| 12:21.11 | benjk | so you're paying for somebody's dirty bribes |
| 12:21.14 | *** join/#asterisk bjohnson_ (n=bjohnson@jecinc.tor.istop.com) |
| 12:21.35 | Mandrak3 | Everyone knows application app_trunkisavail.c - load balancing between mulitple trunks ? |
| 12:21.57 | Mandrak3 | I found in bugs.digium.com |
| 12:26.14 | puzzled | Mandrak3: yes I know it. Haven't used it though |
| 12:28.40 | *** join/#asterisk murf (n=steve_mu@216.166.159.235) |
| 12:29.20 | *** join/#asterisk ivanfm (n=ivanfm@201.52.129.236) |
| 12:29.37 | *** join/#asterisk postel_ (n=jp@unaffiliated/postel) |
| 12:30.40 | Mandrak3 | could be useful for LCR ? |
| 12:31.01 | Mandrak3 | and is there a stable version? |
| 12:31.22 | *** join/#asterisk vaddineni (n=vaddinen@cpe-72-181-71-206.houston.res.rr.com) |
| 12:34.07 | brif8 | Where is there a good source to read on Asterisk clustering, or sharing multiple physical servers to each do their own part eg Server 1: SIP Registration , Server 2: CDR records , Server 3: Extensions 1 - 40 and Server 4: Extensions 41 - 80 etc.. ? |
| 12:35.48 | puzzled | brif8: there is no good source for that |
| 12:35.59 | *** join/#asterisk coppice (n=chatzill@229.166.17.210.dyn.pacific.net.hk) |
| 12:36.08 | brif8 | puzzled: any source at all ? |
| 12:36.18 | [TK]D-Fender | brif8: You're looking at SER there, not *. |
| 12:36.36 | puzzled | brif8: not that I know. best kept secret in the asterisk community |
| 12:37.06 | brif8 | [TK]D-Fender: SER just handles SIP stuff right ? it could the SIP registrations but not all the other or am I wrong ? |
| 12:37.42 | burus | some body used application ExternalIVR ? |
| 12:38.02 | [TK]D-Fender | brif8: I do believe it does CDR, and load balanced termination. *'s role would be that of application server (VM / MeetMe), and PSTN termination. |
| 12:38.27 | brif8 | ok let me check more on SER, thanks |
| 12:50.48 | bXi | is it possible to include *.conf in a config file? |
| 12:51.37 | *** join/#asterisk uwe (n=uwe@dogbert.palnet.com) |
| 12:54.49 | *** join/#asterisk myiagy (n=myiagy@200.175.61.250.static.gvt.net.br) |
| 12:56.20 | [TK]D-Fender | bXi: ...huh? |
| 12:58.01 | Aurs | [TK]D-Fender: think he wants to know if you can do #include <*.conf> |
| 12:58.11 | Aurs | instead of 10 lines with the individual file names |
| 12:58.37 | [TK]D-Fender | Aurs: I was hoping not... didn't want to even suggest it myself :) |
| 12:58.50 | Aurs | hehe |
| 12:58.52 | [TK]D-Fender | Aurs: Bad karma and all |
| 12:58.58 | *** join/#asterisk juanjoc (n=juanjoc@200.73.189.82) |
| 12:59.09 | [TK]D-Fender | bXi: But if you're referring to what I never EVER said... well.. NO. |
| 12:59.53 | Aurs | /etc/asterisk/conf.d/ :P |
| 13:01.22 | *** join/#asterisk aep (n=naep@hosting-technology.com) |
| 13:02.30 | aep | hi, im fed up with the avarage mobile phone(never found a working interface) and i had the idea to use a pda with an gsm pcmcia card. anyone got any tips for me? is it posible/stupid idea ? |
| 13:03.37 | aep | i think this is the right channel to ask , as if it is poisble i hope to find someone who e.g. conected such a card to his asterisk |
| 13:09.12 | coppice | I assume you mean a GSM compact flash card. they usually have a big limitation - no digital audio interface through the compact flash bug. you need to use a headset plugged into the card for voice calls |
| 13:10.12 | *** join/#asterisk syn (i=syn@kenobi.sceen.net) |
| 13:10.25 | syn | hi |
| 13:10.39 | bionoid | aep: That sounds like a whole shitload of work for little or no benefit? I'd just go to benefon.com and get some hardware ;-) |
| 13:10.42 | syn | where is the best place to submit an addon ? |
| 13:11.08 | *** join/#asterisk ssokol (n=ssokol@dsl017-122-217.mci1.dsl.speakeasy.net) |
| 13:11.22 | aep | bionoid: i'll have a look thanks |
| 13:11.32 | docelmo | STEVE! |
| 13:11.44 | bXi | [TK]D-Fender: it seemingly works :) |
| 13:11.55 | ssokol | Yep. It's me. |
| 13:12.04 | bXi | but i'm still having a curious issue with a handytone 286 |
| 13:12.06 | syn | is posting on the bug tracker ok ? |
| 13:12.11 | aep | bionoid: whats so special about that device? |
| 13:12.38 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:14.45 | aep | bionoid: at all the site dosnt work in konqueror |
| 13:15.35 | aep | coppice: ah sorry didnt see your post. what do you mean? dosnt that depend on the card? |
| 13:16.06 | coppice | i've never seen a compact flash or pcmcia card that supports digital voice |
| 13:16.24 | aep | oh |
| 13:16.53 | aep | i never seen a card with a headphone plug outside |
| 13:17.14 | coppice | there are some GSM modules that do. I think the lack of standard AT commands for the audio has inhibited people adding it to their modems |
| 13:17.52 | aep | oh realy no i see what you mean |
| 13:18.06 | *** join/#asterisk hatamen (n=hatamen@222.183.30.146) |
| 13:18.06 | coppice | i've never seen a PCMCIA of CF card without a headset jack. look carefully. its a small hole, usually in an out of the way corner |
| 13:19.59 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 13:20.04 | aep | ok, no digital audio means, no bluetooth headset, sucks |
| 13:21.42 | *** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust944.leed.cable.ntl.com) |
| 13:23.34 | *** join/#asterisk roving_prole (n=Harper@72-254-127-253.client.stsn.net) |
| 13:26.31 | aep | coppice: you know any device that works with linu and does digital audio? |
| 13:26.53 | coppice | a bluetooth adaptor :-) |
| 13:26.57 | aep | eh? |
| 13:27.12 | aep | pda ~~~bluetooth ~~ mobile ? |
| 13:27.36 | *** join/#asterisk inspired (n=mikael@85.221.0.46) |
| 13:28.24 | coppice | a bluetooth capable PDA to GSM handset works OK |
| 13:28.55 | aep | what is a "gsm handset" ? |
| 13:30.08 | *** join/#asterisk jmesquita (n=jmesquit@201.7.117.114) |
| 13:30.24 | coppice | an ordinary phone |
| 13:30.30 | aep | ah |
| 13:30.46 | aep | are you sure? i think this will not route phone calls to the pda |
| 13:31.14 | aep | the problem is i dont want to see any mobile phone anymore, i'm fed up with them |
| 13:31.25 | *** join/#asterisk baskew (n=brad@24.214.206.158) |
| 13:31.39 | coppice | if the PDA supports the bluetooth audio profile it will act like a headset to the phone |
| 13:31.48 | aep | randomly crashing overloaded coulourfull animatet crap |
| 13:32.10 | coppice | crashing? you must have a WinCE phone :-) |
| 13:32.10 | aep | coppice: heh that dosnt give me the posibility to control calls |
| 13:32.37 | *** join/#asterisk SkoZombie (n=awhalan@203-217-86-249.dyn.iinet.net.au) |
| 13:32.37 | aep | i tryed hundrets ( eh not that much) of phones and none of them works like i want |
| 13:32.45 | aep | no matter what brand/os |
| 13:32.52 | aep | even the linux phones suck |
| 13:33.07 | aep | bad gui concept is os independent |
| 13:36.26 | *** join/#asterisk breakdisk (n=breakdis@62.149.122.2) |
| 13:36.54 | breakdisk | hi huys,anyone experience updating zaptel driver of asterisk@home2.5? |
| 13:37.19 | breakdisk | because im going to update it and its in production environment. |
| 13:39.28 | *** join/#asterisk TrixVox (n=Spy007@c-69-248-121-104.hsd1.nj.comcast.net) |
| 13:41.45 | *** join/#asterisk darkskiez (n=mbryars@bb-87-81-62-203.ukonline.co.uk) |
| 13:41.51 | SkoZombie | I've got a wildcard tdm400P with X100M FXO modules in sockets 1 & 2, genzaptelconf says that each channel is inactive |
| 13:41.55 | SkoZombie | any suggestions? |
| 13:44.28 | *** join/#asterisk inv_arp[work] (i=junya@c-71-206-88-100.hsd1.fl.comcast.net) |
| 13:45.27 | *** join/#asterisk SaTLaN32 (n=satlan32@212.150.142.211) |
| 13:45.40 | SaTLaN32 | hello |
| 13:45.56 | SaTLaN32 | need some help with zaptel not hanging after call is finished |
| 13:46.39 | SaTLaN32 | i call one line connected to the card, then from the dial plan i call through the second line on the card, and even when both sides hang up asterisk still keep the call |
| 13:48.01 | *** part/#asterisk hatamen (n=hatamen@222.183.30.146) |
| 13:53.58 | jbroome | aww yeah, MWI works on polycoms without me doing anything. :) |
| 13:54.37 | jbroome | stutter tone and blinky light. </technical> :) |
| 13:54.37 | aep | bad gui concept is os independent/ |
| 13:57.28 | *** join/#asterisk marv[work] (n=timr@64.89.118.139) |
| 13:58.39 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 13:58.39 | *** mode/#asterisk [+o anthm] by ChanServ |
| 14:02.06 | *** join/#asterisk littleball (n=littleba@cm82.epsilon172.maxonline.com.sg) |
| 14:02.46 | littleball | hello, i am looking for voip billing artical about how to build a voip billing system |
| 14:02.54 | littleball | who can recommend? |
| 14:03.07 | littleball | any standard? |
| 14:07.13 | *** join/#asterisk Dibbler_ (n=Dibbler@dsl-217-155-254-174.zen.co.uk) |
| 14:10.46 | burus | I have problem with using ExternalIVR app |
| 14:10.51 | burus | I'm execute ExternalIVR app with params: ExternalIVR(example.py) |
| 14:10.55 | burus | in the python scrip are string: sys.stdout.write("S,has-been-cleared\n") |
| 14:11.04 | burus | has-been-cleared.gsm - standart sound addon |
| 14:11.12 | burus | but ExternalIVR generator doesn't work :( |
| 14:12.35 | *** join/#asterisk bweschke (n=bweschke@66.152.207.97) |
| 14:14.18 | *** part/#asterisk kmilitzer (n=km@office-gw.westend.com) |
| 14:21.47 | phearless | where can I find a good doc about g1, g2 etc ? |
| 14:21.54 | phearless | it is used for zaptel |
| 14:22.00 | phearless | I am confused |
| 14:22.17 | parag_ast | what is the confusion |
| 14:22.20 | phearless | <PROTECTED> |
| 14:22.25 | phearless | I dial this |
| 14:22.28 | parag_ast | so |
| 14:22.35 | phearless | 9 is used by my phone line, it is normal |
| 14:22.36 | parag_ast | its going from fist line |
| 14:23.08 | burus | plz help me with ExternalIVR command |
| 14:24.57 | phearless | so... |
| 14:25.10 | *** join/#asterisk sxpert (n=sxpert@raph.imag.fr) |
| 14:25.15 | phearless | when I call, with asterisk, I got nothing |
| 14:25.15 | *** part/#asterisk sxpert (n=sxpert@raph.imag.fr) |
| 14:25.31 | res_segfault | AvoidingDeadlock: hey, stop messing around, or else I'll have to restart you! |
| 14:25.32 | phearless | i got one ring and then I got just static sounds |
| 14:25.35 | phearless | like crrr crrrr |
| 14:26.02 | AvoidingDeadlock | *PUNT* |
| 14:26.13 | phearless | and my FXO card is on the first slot |
| 14:26.24 | phearless | of the TDM400 PCI card |
| 14:26.51 | phearless | and when I call with a real normal phone plugged in this phone line, it works |
| 14:27.24 | res_segfault | who wants to use me on your asterisk machines? |
| 14:28.32 | *** join/#asterisk monkey13 (n=monkee13@69.7.217.140) |
| 14:29.13 | fafnir | sooo |
| 14:29.32 | fafnir | anyone used voip over tor yet? |
| 14:30.06 | res_segfault | tor as in the onion router? |
| 14:30.16 | fafnir | yup |
| 14:30.29 | fafnir | i was thinking about setting up a hidden asterisk server on the tor network |
| 14:30.42 | res_segfault | nah, but wouldn't that add random latency? |
| 14:31.07 | fafnir | probably |
| 14:31.21 | burus | plz help me with ExternalIVR command |
| 14:31.31 | phearless | hello ! |
| 14:31.40 | burus | hello |
| 14:31.45 | phearless | so nobody has read my question I think |
| 14:31.53 | phearless | my problem is : |
| 14:32.03 | phearless | <PROTECTED> |
| 14:33.01 | phearless | <PROTECTED> |
| 14:33.27 | phearless | does it looks ok ? My FXO module is on the first TDM400 slot |
| 14:33.40 | caio1982 | does someone here knows what's the font name used in the asterisk.org logo? |
| 14:34.17 | burus | funny channel .. very helplees |
| 14:34.37 | burus | sorry for my english :d |
| 14:34.44 | *** join/#asterisk _deg_ (n=deg@200.163.193.247) |
| 14:34.50 | *** join/#asterisk JimVanM (n=jimvanm@HSE-Toronto-ppp3490740.sympatico.ca) |
| 14:35.05 | caio1982 | burus: it's a general purpose channel (afaik), please hold until someone answer your call |
| 14:36.38 | phearless | hello ! |
| 14:36.55 | phearless | anybody ever used asterisk ? |
| 14:37.05 | phearless | I am on the VoIP solution chan, no ? |
| 14:37.20 | docelmo | No NO ONE HERE HAS USED ASTERISK ass.. |
| 14:37.28 | phearless | damn ! |
| 14:37.33 | phearless | i am so unlucky |
| 14:37.56 | phearless | what does look like your log files when you make an outbound call, docelmo ? |
| 14:38.17 | lunk | what's asterisk? |
| 14:38.28 | docelmo | Go learn proper english then come back and ask questions |
| 14:38.57 | AvoidingDeadlock | phearless, spanish is your native lang? |
| 14:39.17 | phearless | french |
| 14:39.47 | phearless | I am surprised by the attitude of docelmo |
| 14:39.53 | lunk | haha |
| 14:39.58 | phearless | docelmo: you should take an anger management course |
| 14:40.10 | lunk | install french module |
| 14:40.10 | AvoidingDeadlock | phearless, to be honest my attitude is much worse than his.. i'm just being nice today! ;) |
| 14:40.11 | phearless | it would help you |
| 14:40.22 | lunk | AvoidingDeadlock: hahah |
| 14:40.43 | phearless | <lunk> install french module |
| 14:40.45 | phearless | wtf is this |
| 14:40.58 | phearless | I ask what does look like the logs file during an outbound call |
| 14:40.58 | lunk | i was growing my vagina |
| 14:41.01 | lunk | sorry |
| 14:41.18 | caio1982 | lol |
| 14:41.20 | Nivex | yay, let's all continue to precipitate the image of ugly Americans at light speed on the Internet. |
| 14:41.21 | phearless | please talk about your vagina to docelmo , not me |
| 14:41.40 | docelmo | piss off frenchy |
| 14:41.44 | *** join/#asterisk kb3ien (n=kb3ien@ool-182f7b34.dyn.optonline.net) |
| 14:42.04 | zoa | omg |
| 14:42.08 | phearless | your case, docelmo , is worst than what i thought ! |
| 14:42.10 | Nivex | docelmo: Go to hell. |
| 14:42.12 | *** join/#asterisk bmg505 (n=leon@dsl-165-130-108.telkomadsl.co.za) |
| 14:42.30 | AvoidingDeadlock | phearless, the best way to know what you want is make a call and see the logs |
| 14:42.51 | zoa | tss behave missies |
| 14:42.52 | phearless | I just pasted 2 times what happen when I make a call |
| 14:43.02 | Nivex | docelmo: I'm having a bad day, but you don't see me deriding random people who come by for help. |
| 14:43.05 | zoa | docelmo, are we having a bad day ? :) |
| 14:43.10 | kb3ien | hello. I seek help compiling asterisk on NetBSD. Anyone have any experience with that? |
| 14:43.23 | docelmo | Been there.. They kicked me out.. |
| 14:43.38 | zoa | having a bad goatie hair day ? :) |
| 14:43.43 | AvoidingDeadlock | OMG ITS ZOA |
| 14:43.50 | zoa | omg now that you say |
| 14:43.54 | zoa | it is me |
| 14:44.00 | *** join/#asterisk sb_mx (n=sb_mx@200.78.229.18) |
| 14:44.10 | zoa | omg its brian! |
| 14:44.13 | docelmo | No.. My day is good thus far.. |
| 14:44.15 | AvoidingDeadlock | ya ya |
| 14:44.30 | zoa | brian's privates |
| 14:45.08 | phearless | it is hard to imagine docelmo when he got a bad day |
| 14:45.27 | docelmo | bad day's == things being thrown around my house.. |
| 14:45.28 | phearless | maybe he use to shot random people in the streets |
| 14:45.38 | phearless | or kill puppies |
| 14:45.40 | *** join/#asterisk blebleble (i=godie@caesar.godie.net) |
| 14:45.52 | Nivex | phearless: I'm guessing the latter |
| 14:45.54 | Nivex | :) |
| 14:47.11 | kb3ien | Im a little concerned about spurious error messages that i dont understand from gmake. |
| 14:47.14 | kb3ien | [[: not found |
| 14:48.48 | kb3ien | I don't think that the Makefile for 1.2.10 is very well debugged. |
| 14:48.48 | phearless | is is a smiley I think |
| 14:48.55 | phearless | [[: |
| 14:49.05 | kb3ien | hrm, its making me frown. |
| 14:49.06 | phearless | it is* |
| 14:49.23 | kb3ien | the other problem i keep running into is lncurses not found. |
| 14:49.55 | phearless | so |
| 14:50.03 | phearless | nobody ever made an outbound call in this chan |
| 14:50.07 | *** join/#asterisk EnoCix (n=jsloan@gateway.digium.com) |
| 14:50.08 | phearless | great ! |
| 14:50.20 | *** part/#asterisk EnoCix (n=jsloan@gateway.digium.com) |
| 14:50.21 | jbroome | nope, i don't like talking to people |
| 14:50.45 | phearless | ah jbroome is waking up to participate to the conversation |
| 14:50.55 | phearless | thank you jbroome |
| 14:50.58 | Nivex | must have just finished his coffee |
| 14:51.07 | kb3ien | I am told that '-L/usr/pkg/lib' must prefix -lncurses for NetBSD, but i have installed the /devel/ncurses package, and it still hasnt created anything in /usr/pkg/lib |
| 14:51.28 | kb3ien | Is it looking for an object file, or a src file? |
| 14:51.40 | kb3ien | object file surely? |
| 14:51.48 | jbroome | i don't finish coffee, i just take breaks btw cups. :) |
| 14:51.59 | burus | plz help me with ExternalIVR command |
| 14:55.58 | *** join/#asterisk ghenry (n=ghenry@suretecsystems.plus.com) |
| 14:56.11 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 14:56.11 | *** mode/#asterisk [+o mog] by ChanServ |
| 14:59.29 | *** join/#asterisk klasstek (n=nunyobiz@ip67-154-143-190.z143-154-67.customer.algx.net) |
| 15:00.28 | *** join/#asterisk negativecreep (n=xaeem@210.2.151.110) |
| 15:00.34 | negativecreep | hi folks |
| 15:00.57 | negativecreep | is it possible to make asterisk listen sip on multiple ports ? |
| 15:01.18 | docelmo | with asterisk no |
| 15:01.20 | docelmo | with SER yes |
| 15:01.32 | negativecreep | so i have asterisk running on this server |
| 15:01.36 | negativecreep | and i dont have another server. |
| 15:01.45 | negativecreep | SER and asterisk can run on the same serveR? |
| 15:01.58 | docelmo | yes.. either different bound IP's or Ports.. |
| 15:02.33 | docelmo | you can run multiple instances of asterisk on the same server.. I dont reccommend it as the deadlocking would probably crash it.. but who knows.. if it was beefy enough |
| 15:03.11 | negativecreep | nopes..not multiple asterisk.. |
| 15:03.17 | negativecreep | one ser instance and one asterisk |
| 15:03.30 | docelmo | uhh yes you can run multiple.. |
| 15:03.35 | docelmo | dont tell me I have done it. |
| 15:03.51 | *** join/#asterisk smackus (n=ckwall@63.149.122.93) |
| 15:04.04 | *** join/#asterisk ToyMan (n=stuq@74-32-51-182.dsl1.mdl.ny.frontiernet.net) |
| 15:05.28 | negativecreep | how was the experience? |
| 15:05.37 | negativecreep | any wiki or howto for this particular scenario? |
| 15:05.42 | *** join/#asterisk Deeewayne (n=dwayne@ool-44c0d56e.dyn.optonline.net) |
| 15:06.43 | *** join/#asterisk wunderkin (n=kev@ip68-226-113-228.ph.ph.cox.net) |
| 15:09.43 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 15:09.43 | *** mode/#asterisk [+o anthm] by ChanServ |
| 15:12.50 | *** join/#asterisk crlshn (i=kvirc@operaciones3.globalnet.hn) |
| 15:13.35 | *** part/#asterisk tecnico (n=tecnico@24.96.146.69) |
| 15:14.07 | *** join/#asterisk tecnico (n=tecnico@24.96.146.69) |
| 15:23.20 | *** join/#asterisk Avalone (n=Avalone_@dial-285.vl-cen-as1.avtlg.ru) |
| 15:24.46 | *** join/#asterisk aydiosmio (i=aydiosmi@judecca.aculei.net) |
| 15:25.14 | smackus | I am reading on the wiki how to set up dundi... also, I am looking on google for definitions on what the heck it all means. Is there a more detailed, beginners guide to dundi somewhere? |
| 15:25.19 | aydiosmio | [custom-out] |
| 15:25.20 | aydiosmio | exten => s,1,Dial(SIP/12127773456@voipswitch) |
| 15:25.40 | aydiosmio | how do I set this so it dials the number in the SIP TO: header instead of 12127773456? |
| 15:26.23 | *** join/#asterisk [TK]D-Fender (n=joe@toronto-HSE-ppp4122655.sympatico.ca) |
| 15:26.26 | aydiosmio | all the asterisk variables don't seem to work |
| 15:28.17 | aydiosmio | [TK]D-Fender: hi |
| 15:28.38 | phearless | asteriiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiisk !!!!!! |
| 15:28.54 | phearless | it makes me crazy, sorry |
| 15:28.58 | mog | hi |
| 15:29.11 | phearless | hi |
| 15:29.20 | aydiosmio | no one has an idea? |
| 15:32.11 | aydiosmio | ${SIP_HEADER(TO)} is just junk |
| 15:32.28 | file | define "junk" |
| 15:32.57 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 15:34.07 | aydiosmio | hm, just a minute |
| 15:34.37 | *** join/#asterisk SplasPood (n=jwb@206.252.198.101) |
| 15:41.53 | aydiosmio | ah |
| 15:42.21 | aydiosmio | Aug 23 11:41:16 DEBUG[20135] pbx.c: Function result is '<sip:12127773456@216.32.221.000>' |
| 15:42.28 | *** join/#asterisk matlj (n=mlejeune@213.56.232.90) |
| 15:43.05 | matlj | hi |
| 15:43.13 | file | that's a SIP URI, not junk - pretty standard for a To header |
| 15:43.13 | matlj | I need help |
| 15:43.14 | aydiosmio | how do I get the phone number out of that? |
| 15:43.31 | aydiosmio | file: yeah I was using something else before and I got the SIP call ID |
| 15:43.44 | matlj | I can't get any incoming sip call (outgoing ok) |
| 15:45.08 | matlj | can I paste revelant items from my config files here ? |
| 15:45.19 | *** join/#asterisk alexhopper (n=a27386@CPE000103d29ae2-CM001225dfdfe0.cpe.net.cable.rogers.com) |
| 15:45.45 | jbroome | ~paste |
| 15:45.47 | jbot | well, paste is see http://paste.husk.org, or http://paste-it.net |
| 15:45.52 | malverian | We have some issues with DTMF from mobile phones here. Do you think using relaxdtmf=yes would help the situation? |
| 15:46.09 | *** join/#asterisk scastromx (n=scastro@200.38.91.142) |
| 15:46.17 | matlj | jbot: thanks |
| 15:46.17 | jbot | matlj: gern geschehen |
| 15:47.27 | scastromx | hello everyone, can anybody help with a TE110P connected to a 3Com NBX, it worked great for a couple of weeks and now the dchannel is down on the nbx side |
| 15:49.57 | kb3ien | ahh! i have compiled asterisk1.2.9.1 binary,as ELF and it works. |
| 15:50.07 | kb3ien | are there any other binaries that asterisk needs? |
| 15:50.54 | Qwell | kb3ien: Did you not compile it as ELF previously? |
| 15:51.29 | [TK]D-Fender | aydiosmio: So whats the issue? That is a perfectly normal header value. |
| 15:52.23 | matlj | here it is : |
| 15:52.25 | matlj | http://pastebin.com/774074 |
| 15:52.33 | matlj | thanks in advance.. |
| 15:53.10 | jbroome | is voicepulse connect down for anyone else for incoming? |
| 15:53.32 | aydiosmio | exten => s,1,Dial(SIP/${SIP_HEADER(TO)}@voipswitch) |
| 15:53.38 | aydiosmio | I want to dial the TO number |
| 15:53.43 | matlj | with this config, when I call my sip number, the sip provider tells me that I'm offline (not registered) |
| 15:53.48 | aydiosmio | need to know how to get those digits out of there |
| 15:53.50 | Qwell | aydiosmio: You're gonna need to do some parsing there, bud |
| 15:54.04 | aydiosmio | yeah I'm moving it reluctantly into a an AGI |
| 15:54.08 | aydiosmio | an |
| 15:54.15 | aydiosmio | I wish DNIS worked. |
| 15:54.50 | [TK]D-Fender | aydiosmio: How are you getting to that exten? Pastbin your code for everything related to the call path. |
| 15:54.52 | [TK]D-Fender | !pb |
| 15:55.17 | [TK]D-Fender | ~pb |
| 15:55.19 | jbot | pb is, like, a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca/, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com/, or http://bzflag.pastebin.ca/, or for images use http://imageshack.us/, or http://www.paste-it.net/ |
| 15:56.10 | *** join/#asterisk funxion (n=nunya@63.214.236.169) |
| 15:56.24 | funxion | does anyone know of a mini pci dual t1 card? |
| 15:56.44 | Qwell | funxion: I've heard of converter dealies |
| 15:57.01 | [TK]D-Fender | funxion: mini-pci? |
| 15:57.06 | funxion | I saw something from sangoma but would rather have digium |
| 15:57.08 | funxion | yes |
| 15:57.14 | funxion | mini PCI |
| 15:57.15 | matlj | is someone looking at my config ? I'm really stuck.. |
| 15:57.17 | Qwell | supposedly, there are converters that go from minipci, which accept a pci card |
| 15:57.21 | [TK]D-Fender | funxion: Just a 2U bracket? |
| 15:57.28 | coppice | funxion: there are dual T1/E1 PCMCIA cards, but I haven't heard of mini-PCI ones |
| 15:57.29 | funxion | something like that |
| 15:57.33 | Qwell | maybe it was the other way around...but |
| 15:57.38 | funxion | trying to use a aopen mini pc |
| 15:57.40 | Qwell | probably worth looking into |
| 15:58.12 | funxion | I think I make the te110p mini pci wiht a little mod |
| 15:58.17 | funxion | but not the dual |
| 15:58.19 | coppice | funxion: you mean a 2U high PCI card? that's isn't mini-PCI |
| 15:58.24 | funxion | I think the circuitry is too big |
| 15:58.46 | Qwell | yeah, are you talking minipci or low profile pci? |
| 15:58.48 | funxion | no I mean mini PCI |
| 15:58.56 | funxion | its the same thing |
| 15:58.59 | Qwell | no... |
| 15:59.19 | coppice | mini-PCI is a tiny card that does in notebooks |
| 15:59.22 | jbroome | no it's not |
| 15:59.25 | coppice | s/does/goes |
| 15:59.34 | funxion | thats PCMCIA |
| 15:59.46 | Qwell | and minipci |
| 15:59.51 | *** join/#asterisk ToTo (n=ToTo@host212-207.pool872.interbusiness.it) |
| 15:59.56 | Qwell | pcmcia is external to the notebook |
| 16:00.00 | coppice | PCMCIA is a module to plug into notebooks. mini-PCI goes inside |
| 16:00.03 | jbroome | holy shit, you have no idea what you're talking about |
| 16:00.06 | Qwell | minipci is *inside* of it, for like wireless cards and such |
| 16:00.13 | funxion | ok |
| 16:00.19 | Qwell | low profile PCI is what you put into servers and such |
| 16:00.21 | funxion | http://minipc.aopen.com/Global/spec.htm |
| 16:00.26 | funxion | Im trying to use one of those |
| 16:00.30 | funxion | check the specs |
| 16:00.33 | funxion | it says mini pci |
| 16:00.38 | funxion | thats why I ask |
| 16:01.06 | funxion | need to put it in a place where space is at a premium |
| 16:01.17 | funxion | needs to be ultrasmall |
| 16:01.32 | funxion | need dual t1 |
| 16:01.53 | coppice | I think that's because it is a notebook in a bigger box. they really mean mini-PCI, and its a tiny card with no external bracket |
| 16:02.38 | funxion | looking at the pics of the rear of the box it has a bracket |
| 16:02.47 | funxion | external |
| 16:02.56 | jbroome | then it's a pci card with a small bracket |
| 16:03.02 | *** part/#asterisk monkey13 (n=monkee13@69.7.217.140) |
| 16:03.11 | funxion | exactly |
| 16:03.26 | funxion | but does anyone know of a dual t1 card that I could fit in there |
| 16:03.34 | funxion | hmm |
| 16:03.45 | funxion | maybe Ill measure out one of the dual's that I have |
| 16:03.46 | *** join/#asterisk Gregabyte (n=greg@gateway.digium.com) |
| 16:03.56 | coppice | i can't see any pictures of the back |
| 16:04.22 | funxion | http://linuxdevices.com/news/NS8464432110.html |
| 16:04.29 | funxion | theres one on that page |
| 16:05.42 | coppice | I can't see any card slot in that picture, and the spec says it doesn't have any |
| 16:05.43 | funxion | they sell them at tiger direct |
| 16:05.49 | funxion | Im going to go by the outlet store and check it out |
| 16:05.57 | jbroome | that's firewire, usb, lan, DVI, svideo and sound |
| 16:06.09 | jbroome | and a vent |
| 16:06.57 | funxion | hmmm |
| 16:06.59 | coppice | nice that they have DVI instead of a 15 pin D. too many things still lack a DVI connector |
| 16:07.02 | funxion | well that would suk |
| 16:07.36 | *** join/#asterisk eKo1 (n=eKo1@190.4.7.90) |
| 16:09.41 | aydiosmio | exten => s,n,Dial('SIP/${DNID}@voipswitch') |
| 16:09.47 | aydiosmio | seems to work just fine |
| 16:11.12 | *** join/#asterisk matlj (n=mlejeune@mat.zapto.org) |
| 16:11.19 | matlj | hi again |
| 16:11.37 | matlj | did someone look at my problem, please ? |
| 16:11.52 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 16:12.01 | *** join/#asterisk Trakkasure (n=Sgemtum@adsl-2-32-56.mia.bellsouth.net) |
| 16:14.23 | *** join/#asterisk Givemelove (n=non@208.57.229.162) |
| 16:17.44 | *** join/#asterisk Seba_soy (n=s@64.76.126.27) |
| 16:17.47 | Seba_soy | hi all! |
| 16:18.48 | *** join/#asterisk dasenjo (n=dasenjo@208.195.215.127) |
| 16:18.56 | Seba_soy | I have a FXO-CLONE card with a pstn-line connected, each time a make a call I got "Zap 1-1 answered SIP...." instead RINGING when called phone starts to ring... |
| 16:19.06 | Seba_soy | there is some config to resolv it? |
| 16:25.52 | *** join/#asterisk okdo (n=goldenol@65.171.196.18) |
| 16:26.07 | okdo | anyone have any tips for getting the timing good enough on a sangoma or digium card to use rxfax with a PRI? |
| 16:28.50 | *** join/#asterisk NDT (n=nunya@cpe-24-195-66-214.nycap.res.rr.com) |
| 16:29.27 | NDT | anyone running cepstral with multiple voices and using app_cepstral.so? |
| 16:31.27 | [TK]D-Fender | okdo: Works pretty much stock for me... |
| 16:34.41 | okdo | [TK]D-Fender: I am getting broken up faxes via spandsp (http://soft-switch.org/spandsp_faq/ar01s08.html#id2621606) which says my timing with the telco is most likely causing frame slips? |
| 16:34.58 | okdo | [TK]D-Fender: unfortunately I have no clue where to start to get the timing setup properly |
| 16:35.03 | [TK]D-Fender | What card? |
| 16:35.34 | coppice | okdo: show me your zaptel.conf file |
| 16:35.38 | *** join/#asterisk Budairc (n=chatzill@proxy01.mhnet.com.br) |
| 16:37.47 | *** join/#asterisk jtodd (n=jtodd@adsl-75-24-91-221.dsl.pltn13.sbcglobal.net) |
| 16:41.43 | *** join/#asterisk florz (n=florz@2001:1a50:503c:0:0:0:0:1) |
| 16:42.23 | *** join/#asterisk anthonyl (n=anthony@gateway.digium.com) |
| 16:43.47 | coppice | okdo: having looked at your zaptel.conf it looks like you are clocking the card OK. You might have PCI problems. People get data dropped there on some machines |
| 16:44.24 | *** join/#asterisk profounded (n=profound@ool-44c4eae2.dyn.optonline.net) |
| 16:44.41 | coppice | Digium hide the error information from the T1, so its hard to look for subtle problems there, unless you use the old Tormenta 2 cards. |
| 16:44.54 | okdo | this is actually a sangoma card |
| 16:45.06 | mog | lol |
| 16:45.19 | okdo | AFT-A101c T1/E1 |
| 16:46.00 | *** join/#asterisk darkskiez (n=mbryars@bb-87-81-62-203.ukonline.co.uk) |
| 16:46.06 | *** join/#asterisk justinu|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 16:46.11 | mmealling | woot.... got the SPA-2002 in and made my first call with it in less than 5 minutes after unpacking it. |
| 16:46.18 | *** join/#asterisk krisguy (n=krisguy@h216-170-039-057.adsl.navix.net) |
| 16:46.21 | coppice | mog: what's funny? hiding those errors is a huge PITA |
| 16:46.53 | mog | no you trolling on us |
| 16:46.57 | mog | and it ended up being sangoma |
| 16:46.59 | mog | thats all |
| 16:47.10 | [TK]D-Fender | okdo: Call the telco and have them give you a timed error count. When I first started out on a digital circuit, my first 2 cards refused to accept telco clock, and were throwing off frame slips & filps like nuts. |
| 16:47.19 | coppice | I'm not trolling. its a disgrace that nothing gets done about that |
| 16:48.11 | okdo | [TK]D-Fender: once I get a timed error count what do I do with it? |
| 16:48.19 | coppice | mog: I don't know if sangoma report them or not. I put the infrastructure in years ago, and made sure the tor2 driver reported everything. Its far easier to help people when they use tormenta 2 cards |
| 16:48.20 | *** join/#asterisk RoyK (n=roy@ti211310a080-4327.bb.online.no) |
| 16:48.25 | [TK]D-Fender | mmealling: Yup, dead easy to set up, though you'll want to tweak it a bit I'm sure to match dialplans, etc. |
| 16:48.46 | [TK]D-Fender | okdo: Well think about that once the telco confirms the state of your link. |
| 16:49.04 | mmealling | won't have to that much.... I just send everything to telasip. Anything from telasip rings on all extensions.... |
| 16:49.11 | [TK]D-Fender | okdo: And while Sangoma's share IRQ's rather well you SHOULD still try to assure that it has its own if possible. |
| 16:49.41 | [TK]D-Fender | mmealling: Yeah, the SPA's really get you ready to rock'n'roll fast... |
| 16:50.05 | mmealling | that and just a simple setup on Asterisk on my home server. Since it _is_ the firewall it makes NAT traversal easier. |
| 16:51.09 | mmealling | Although I can tell I'm going to have to setup a dedicated firewall/asterisk box.... call quality drops whenever I'm compiling something. |
| 16:51.42 | okdo | [TK]D-Fender and coppice: thank you |
| 16:51.58 | mmealling | even setting a specific nice level to the asterisk process doesn't solve that problem.... |
| 16:52.33 | *** join/#asterisk Bullseye_Network (n=info@72-166-37-114.dia.static.qwest.net) |
| 16:53.23 | RoyK | are there any open, GPLed or similar speech recognition engines? |
| 16:54.43 | mog | sphinx |
| 16:56.22 | *** join/#asterisk spatulamaan (n=ggilmore@207.188.8.252) |
| 17:00.31 | *** join/#asterisk c4t3l (n=c4t3l@69.15.174.114) |
| 17:00.53 | *** join/#asterisk miller7 (n=999@adsl19-195dynamic.athens.acn.gr) |
| 17:02.52 | coppice | i think sphinx is the only complete one |
| 17:03.09 | Qwell | sphinx isn't great though |
| 17:03.17 | Qwell | I mean, it's good, don't get me wrong |
| 17:03.28 | coppice | its pretty much as good as anything else |
| 17:03.40 | Qwell | well, it's probably the best "free" one |
| 17:04.02 | coppice | do you know something commercial that is much better? |
| 17:04.36 | Qwell | file showed me the lumenvox one - it seemed very good |
| 17:05.49 | *** join/#asterisk profounded (n=profound@ool-44c4eae2.dyn.optonline.net) |
| 17:06.28 | coppice | any of them *can* seem good. its doing it consistently in a reasonably open ended context that always defeats them. people play with sphinx in a rather open ended way, and think its weak. |
| 17:07.00 | miller7 | Anyone knows of the best way to receive a call from PRI, then send it out to BRI on an asterisk box? I want this for a data call (call comes from PRI and the data box is another one with a BRI card on it) |
| 17:07.44 | miller7 | I have junghans BRI card and digium PRI one so I can put them on the same box. |
| 17:09.13 | *** join/#asterisk obscurant (n=obscuran@12-32-45-95.static.blackfoot.net) |
| 17:09.34 | RoyK | miller7: exten => 123,1,Dial(Zap/g2/234) |
| 17:09.38 | RoyK | ~docs |
| 17:09.39 | jbot | hmm... docs is Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 17:09.39 | RoyK | ~book |
| 17:09.42 | jbot | i heard book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 17:09.42 | RoyK | ~rtfm |
| 17:09.45 | jbot | it has been said that rtfm is Read The F*cking Manual (TM). It is a suggestion to do your homework before posting a question. Sometimes used as RTFM $SPECIFIC_MANUAL to refer to a specific source of information. See also http://uncyclopedia.org/wiki/RTFM. |
| 17:09.54 | miller7 | RoyK: I'm talking about data call, not voice |
| 17:10.07 | RoyK | is there a difference? |
| 17:10.13 | miller7 | I don't know, I'm asking :) |
| 17:10.22 | RoyK | i don't think there is |
| 17:12.42 | [TK]D-Fender | coppice: Sphinx-ter ;) |
| 17:13.27 | RoyK | ~say wtf |
| 17:13.28 | jbot | wtf |
| 17:14.30 | RoyK | ~say ~lart jbot |
| 17:14.31 | jbot | ~lart jbot |
| 17:14.42 | Qwell | heh |
| 17:15.25 | RoyK | ~say ~say ~say wtf |
| 17:15.27 | jbot | ~say ~say wtf |
| 17:15.35 | RoyK | not recursive. boring :P |
| 17:19.04 | *** join/#asterisk mtaht4 (n=m@c-71-198-23-124.hsd1.ca.comcast.net) |
| 17:19.20 | coppice | miller7: are your PRI and BRI both connected to the PSTN? |
| 17:19.34 | miller7 | PRI is, BRI is on the asterisk box |
| 17:19.57 | miller7 | call should come in from PRI, and out the BRI to the 2nd box |
| 17:20.00 | miller7 | (ideally) |
| 17:20.30 | coppice | If they are clocked from the same source data will probably work OK. If they are not, you'll get data slips |
| 17:20.46 | miller7 | PRI is clocked from the telco |
| 17:21.04 | miller7 | BRI, I don't know. From the PC I guess? |
| 17:21.16 | miller7 | or there are other options? |
| 17:22.23 | *** join/#asterisk Zodiacal (i=hehehe@bdsl.66.14.242.199.gte.net) |
| 17:22.35 | Zodiacal | anyone know a site to download ringtones? |
| 17:22.55 | Qwell | Zodiacal: what kind of phone? |
| 17:22.59 | Qwell | and what service? |
| 17:23.07 | Qwell | If you use Sprint, I know a great one |
| 17:23.17 | Zodiacal | qwell asterisk polycom 601 |
| 17:23.20 | Qwell | oh, heh |
| 17:23.21 | Zodiacal | but just any .wav file will do |
| 17:26.05 | RoyK | Zodiacal: http://karlsbakk.net/fun/modem.wav |
| 17:26.37 | Zodiacal | umm |
| 17:27.09 | jbroome | humm, single number extensions don't work so well |
| 17:28.22 | Nivex | only an an IVR context |
| 17:28.45 | *** join/#asterisk arkonadev (n=arkonaj@65.203.186.131) |
| 17:29.06 | arkonadev | whats the easiest way to make an extension dial out to a external number though the manager api? |
| 17:29.21 | jbroome | i have 1-15. I dial 15 and get something else. |
| 17:29.27 | jbroome | i'm just going to change them to 100 - 115 |
| 17:30.11 | designdream | anyone have an opinion on spa-841 vs 941? |
| 17:30.50 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
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| 17:37.56 | _MDC_ | I've got a trouble connecting a grandstream budgetone behind a netgear router. What are the settings I should touch? |
| 17:38.13 | aydiosmio | probably should increase the quality |
| 17:38.15 | smackus | quick question. I have a large number of people doing call recordings on my asterisk systems. I am starting to see the effects of that. So I am considering recording to a ramdisk partition. Has anyone been in my same situation to tell me if this helps? |
| 17:38.22 | aydiosmio | and lower the penny-pinching |
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| 17:39.31 | *** join/#asterisk CrossRoad (n=SilentVa@209.172.67.146) |
| 17:39.46 | _MDC_ | I can here MoH but no other sound, RTP problem? |
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| 17:41.04 | CrossRoad | Is it true that IAX handles DTMF better to SIP? I'm having issues like callers reach the wrong EXT.. so our provider says that I should switch to IAX instead of SIP.. is this true or? any help is appreciated |
| 17:42.54 | CrossRoad | VoicePulse: is this someone from Voicepulse or just a random name? the reason I'm asking is our provider is VoicePulse! |
| 17:43.42 | jbroome | omg |
| 17:44.19 | *** join/#asterisk lowlevel (n=Stuart@CPE0050ba71c82f-CM000f9f7d6742.cpe.net.cable.rogers.com) |
| 17:45.02 | VoicePulse | CrossRoad: Yes, I am someone from VoicePulse. However, if you are already in touch with someone from our Support department, they will be best equipped to resolve your problem. I suggest you try IAX instead of SIP and see if it improves your situation at all -- either way, that information will be helpful to our engineer in resolving the issue. |
| 17:46.48 | arkonadev | ok on an originate action i can set the extension to dial to but how do i set the extension it is originating from? |
| 17:47.27 | CrossRoad | VoicePulse: Ok I'll go with that.. but is it enough if I just switch the communicate between VP and us to IAX or should I change the complete extention.conf (how I call the ext as well)? thanks for your response |
| 17:50.31 | CrossRoad | VoicePulse: meaning.. can I just change the communication between VP and our PBX switch to IAX and leave the communication between our PBX switch and the extentions (phones) to SIP, will that work or do I need to change to whole thing to IAX |
| 17:51.00 | VoicePulse | CrossRoad: You can change just PBX<-->VoicePulse to IAX2 and leave the rest as SIP. |
| 17:51.14 | IOscanner | Anyone know a source to buy DID's for Dallas, Texas? Everyone seem to be out. I need about 20. |
| 17:51.17 | CrossRoad | VoicePulse: Thanks alot appreicate it |
| 17:51.50 | *** join/#asterisk The_LightSide (n=lightsid@wbs-196-2-109-10.wbs.co.za) |
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| 17:53.40 | bweschke | IOscanner: what's the NPA-NXX ? |
| 17:53.55 | The_LightSide | hi all, does anyone know if the early media issue has been dealt with in the latest release of zaptel/libpri? |
| 17:54.07 | backblue | designdream: the one in the midle it's better! :D |
| 17:55.17 | TrixVox | Speaking of VoicePulse, they had 214 dallas last time I checked |
| 17:56.10 | IOscanner | 972 214 469 |
| 17:56.39 | IOscanner | Yes, but I dont' need to pay for service. I have termination. I only need to buy DID's. |
| 17:56.42 | VoicePulse | CrossRoad: No problem, if you have any problems, send in your *.conf files and our support guys will look at them for you. |
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| 17:59.03 | bweschke | IOscanner: nope - sorry.. my wholesaler doesn't have them either atm |
| 17:59.07 | [TK]D-Fender | designdream: 841 is a cheap piece of junk, the 9XX series is pretty decent, but not worth it in North America and any country with access to good Polycom pricing. |
| 17:59.14 | TrixVox | you want DIDs and incoming calls for free? |
| 17:59.20 | wunderkin | i am finding that out also, polycoms are much better and cheaper.. |
| 17:59.31 | IOscanner | I have 3 and all of them are out. If I buy them and put them on our network then yes inbound is free. |
| 18:06.15 | IOscanner | I have 2000 channels with a carrier. Just need DID for some people in dallas. |
| 18:06.42 | IOscanner | Guess I get to play the watch and wait game. |
| 18:06.50 | *** join/#asterisk syzygyBSD (n=chatzill@66.226.228.204.cpe.speedyquick.net) |
| 18:07.15 | [TK]D-Fender | wunderkin: Better yes, cheaper... well... maybe cheaper than they once were depending where you are. |
| 18:07.15 | TrixVox | So, you want to purchase DIDs from somewhere and port them to your existing carrier? |
| 18:07.15 | wunderkin | do i have this right, total calls per phone: 301=16, 430=16, 501=24, 601=144? |
| 18:07.16 | [TK]D-Fender | wunderkin: Sounds about right... kinda crazy to think ANYONE would fill even a 301 |
| 18:07.16 | wunderkin | cool |
| 18:07.16 | wunderkin | the 301 is pretty much out unless you dont want poe |
| 18:07.16 | [TK]D-Fender | wunderkin: And of course thats using the scroll keys to naviage multiple calls/key across multiple keys... challenging to say the least. |
| 18:07.22 | Cresl1n | The_LightSide: what early media issue? |
| 18:07.40 | wunderkin | yes, thats what makes it fun |
| 18:07.57 | *** join/#asterisk smackus (n=ckwall@63.149.122.93) |
| 18:07.58 | [TK]D-Fender | wunderkin: Pretty much. tThe 301 with only the PoE cable and no brick is about $135 USD, and the IP 430 w/ BOTH is $150. Makes the 301 a real bottom dollor fixed purpose phone. |
| 18:07.58 | IOscanner | Trixvox: yes |
| 18:08.24 | [TK]D-Fender | Hey : Off-topic general question someone here likely has the easy answer to : in RH based systems, how do I restrict a user from loggin in through SSH? |
| 18:08.31 | [TK]D-Fender | (specific users clearly) |
| 18:08.35 | wunderkin | non existant shell? |
| 18:08.53 | designdream | [TK]D-Fender: polycom pricing? |
| 18:08.57 | Qwell | yeah, either /bin/false or /sbin/nologin or something |
| 18:09.02 | [TK]D-Fender | wunderkin: Might work, but I might want it to work on the box direct, just not SSH |
| 18:09.06 | [TK]D-Fender | designdream: Please rephrase your question.... |
| 18:09.10 | wunderkin | shitty, hmm |
| 18:09.15 | designdream | [TK]D-Fender: i was just going to order 10 spa-942's.. and then i read your comment on north americans having access to good polycom pricing.. |
| 18:09.20 | [TK]D-Fender | Thoguht there must be a ssh_disallow list or something pbvious I just overlooked. |
| 18:09.20 | smackus | I have seen that you can adjust the gain on polycom phones... does this affect the quality of the calls? I would like to get a little more volume out of these suckers. Is this possible? If so am I going in the right direction? |
| 18:09.24 | designdream | [TK]D-Fender: what phone do you suggest over spa-942s'? |
| 18:09.37 | [TK]D-Fender | designdream: www.telephonydepot.com |
| 18:22.43 | [TK]D-Fender | designdream: Virtually any Polycom > any SPA. (IP 301 IS a little limited) |
| 18:22.46 | *** join/#asterisk oej (n=oej@63.117.53.60) |
| 18:22.57 | *** join/#asterisk pigpen (n=mark@fw.seamans.cc) |
| 18:23.00 | Qwell | [TK]D-Fender: http://marc.theaimsgroup.com/?l=secure-shell&m=92799861811646&w=2 |
| 18:23.03 | [TK]D-Fender | designdream: Lets say on general I'd suggest IP430'a for general users, 501's anywhere you don't need PoE (instead of the IP 430), and 601's + attendant modules for receptionists and ego-trip bosses. |
| 18:23.15 | pigpen | hi all, quick question. I have Flash Operator Panel setup on an * box at a remote site. It works fine on the local lan, but remotely, it shows no status. Any way to make it ... well... work? |
| 18:23.15 | designdream | [TK]D-Fender: uhm. dual port ethernet switch in phone? |
| 18:23.18 | Qwell | [TK]D-Fender: also http://www.karkomaonline.com/article.php/20030829212356235/print |
| 18:23.20 | Qwell | specifically AllowGroups |
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| 18:23.35 | *** part/#asterisk helleub (i=helleub@APuteaux-154-1-28-90.w83-199.abo.wanadoo.fr) |
| 18:23.38 | wunderkin | designdream, i believe they all do |
| 18:23.40 | [TK]D-Fender | Qwell: 2nd link is exactly the kind of info I was looking for. Many thanks. |
| 18:23.44 | Qwell | ~thanks |
| 18:23.44 | jbot | Qwell: my pleasure |
| 18:23.44 | Qwell | Just send money.... |
| 18:23.49 | wunderkin | designdream, take a look at the data sheets |
| 18:23.53 | Qwell | :p |
| 18:23.56 | designdream | i am.. they do |
| 18:24.01 | [TK]D-Fender | designdream: Indeed they all do. |
| 18:24.04 | syzygyBSD | there was a version released yesterday, is it stable? |
| 18:24.08 | syzygyBSD | I am updating a client from cvs head and want to make sure it is stable |
| 18:24.13 | [TK]D-Fender | Qwell: When the time comes for me to learn Cisco, then I have a reason to flip you a few $ The quick stuff we all give out free.. kinda like pushers ;) |
| 18:24.16 | Qwell | :p |
| 18:24.16 | syzygyBSD | can't wait to clear off my tasklist so I can dive into ss7 |
| 18:24.22 | designdream | [TK]D-Fender: i dont need PoE on any of them...but there is quite a price diff between 430 and 501.. |
| 18:24.25 | The_LightSide | i beleive its called early media |
| 18:24.27 | wunderkin | designdream, $15? |
| 18:24.29 | designdream | wunderkin: i am seeing $50 |
| 18:24.37 | wunderkin | try froogle.google.com |
| 18:24.39 | designdream | yay! 169 |
| 18:24.40 | [TK]D-Fender | Qwell: Actually.. not working yet... restarted sshd & xinetd, but am continuing my research |
| 18:24.51 | The_LightSide | Cresl1n? |
| 18:24.54 | [TK]D-Fender | designdream: Just use the link I gave you. Lowest prices I've seen in one place. |
| 18:25.08 | *** part/#asterisk smackus (n=ckwall@63.149.122.93) |
| 18:25.10 | [TK]D-Fender | designdream: IP 430 has only 2 line-keys (regs possible), lighted indicators, native PoE, and a smaller footprint. IP 501 has a bigger nicer screen, 3 line keys (regs possible), but no lights or native PoE. |
| 18:25.12 | *** join/#asterisk mcreedjr (n=mcreedjr@adsl-75-13-62-230.dsl.toldoh.sbcglobal.net) |
| 18:25.24 | designdream | yay! =D |
| 18:25.32 | [TK]D-Fender | So if you're not considering PoE then I'd say 501's around..... |
| 18:25.34 | mcreedjr | Hi all, I'm having trouble receiving CID from the USA PSTN on a TDM400 series card. Here is a pastebin with relevant information: http://pastebin.com/774213. Any ideas? |
| 18:25.41 | pigpen | I thought the new sip 2.0x software was not going to run on the 50x series? |
| 18:25.41 | designdream | [TK]D-Fender: thanks for all the help... ordering 10 right now |
| 18:25.48 | pigpen | Only the 430 & the 601... |
| 18:25.53 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
| 18:26.01 | [TK]D-Fender | pigpen: I'm running the 2.0 beta on my IP 501 |
| 18:26.04 | [TK]D-Fender | pigpen: Release still pending of course |
| 18:26.06 | pigpen | ah...ok..so it must not run on the 300/500/600 then.... |
| 18:26.24 | pigpen | personally, I am not going to run it for some time.... |
| 18:26.28 | [TK]D-Fender | pigpen: 300/500 is believable, the 600 HAD the extra memory to begin with and the reason for creating the 601 was for the attendant module support. |
| 18:26.35 | MstlyHrmls | pigpen: no, my understanding is that it will run on the x00 series as well |
| 18:42.32 | MstlyHrmls | they just won't support all the new, fancy features |
| 18:42.35 | [TK]D-Fender | MstlyHrmls: Depends on final image size as well... no idea how it scales out.... (or will in the near future) |
| 18:42.46 | pigpen | ah...cool..I have a few 500's... |
| 18:42.52 | [TK]D-Fender | MstlyHrmls: I DO hope that they use the new IP 430 colour scheme for the rest of the series though.... |
| 18:42.55 | pigpen | but I have deployed about 500 601's... |
| 18:42.55 | MstlyHrmls | [TK]D-Fender: well, that'd be why they wouldn't support some fo the features on the 2 Meg platforms. The features wouldn't be built into the 2 Meg image. |
| 18:42.59 | Qwell | ... |
| 18:43.03 | [TK]D-Fender | pigpen: I run about 25 over here |
| 18:43.06 | pigpen | personally, we only have about 10 in our company.... |
| 18:43.09 | [TK]D-Fender | Qwell: Don't worry about it.... he's even slower to APOLOGIZE! :D |
| 18:43.13 | MstlyHrmls | [TK]D-Fender: new 430 colour scheme? |
| 18:43.17 | pigpen | but I am "going live" with a customer on Wednesday of about 200 |
| 18:43.20 | [TK]D-Fender | MstlyHrmls: If you want to call grey-scale like that, yeah :) |
| 18:43.23 | pigpen | a mix of 430's & 601's |
| 18:43.27 | [TK]D-Fender | MstlyHrmls: IP 430 uses an inverse video theme for the soft keys, etc, and less "framing" in the menus. |
| 18:43.30 | *** part/#asterisk aep (n=naep@hosting-technology.com) |
| 18:43.30 | *** join/#asterisk rbordeaux (i=hidden-u@80.169.196.234) |
| 18:43.40 | MstlyHrmls | [TK]D-Fender: ahhh, for the graphic U/I? |
| 18:43.56 | *** join/#asterisk matlj (n=mlejeune@mat.zapto.org) |
| 18:44.03 | [TK]D-Fender | MstlyHrmls: Yes, the phone-LCD |
| 18:44.13 | matlj | hi |
| 18:44.16 | [TK]D-Fender | MstlyHrmls: Forget the Web interface..... |
| 18:44.20 | MstlyHrmls | [TK]D-Fender: heh |
| 18:44.25 | MstlyHrmls | [TK]D-Fender: I try to, very very hard |
| 18:44.25 | [TK]D-Fender | MstlyHrmls: I would GLADLY see it removed to make room for FEATURES ;) |
| 18:44.32 | pigpen | I second that! |
| 18:44.35 | MstlyHrmls | [TK]D-Fender: haha |
| 18:44.38 | pigpen | like the "MyStat" to actually do something.... |
| 18:44.39 | pigpen | but yes...that is also an * integration thing... |
| 18:44.52 | matlj | could someone look at http://pastebin.com/774074 and tell me why the incoming calls don't work, please ? |
| 18:45.00 | [TK]D-Fender | pigpen: I just want bweschke's ACD patch to see complete merge into the main-line * tree. |
| 18:45.08 | [TK]D-Fender | matlj: You only have a peer, no user entry, and no details about your [general] section to show even an un-auth'd call can come in. |
| 18:45.15 | pigpen | ok...ACD patch??? |
| 18:45.19 | [TK]D-Fender | pigpen: Well its more like an entire branch of CVS for it. not sure exactly how much needed to be changed to support it... |
| 18:45.24 | [TK]D-Fender | pigpen: For use with the Polycom ACD login/out functionailty. |
| 18:45.33 | pigpen | ah..yes....that would be nice. |
| 18:46.32 | matlj | [TK]D-Fender: thanks. Which general section are you talking about ? In extensions or sip ? |
| 18:46.36 | [TK]D-Fender | matlj: sip.conf |
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| 19:03.48 | mcreedjr | Hi all, I'm having trouble getting CID to work with my TDM400P series card. Here are my configs: http://pastebin.com/774233. Any suggestions? |
| 19:03.48 | *** join/#asterisk Kylun (i=StarHawk@adsl-068-157-090-228.sip.bct.bellsouth.net) |
| 19:03.48 | *** join/#asterisk lowlevel (n=Stuart@CPE0050ba71c82f-CM000f9f7d6742.cpe.net.cable.rogers.com) |
| 19:03.59 | wunderkin | it looks like that the spa-942 does not support multiple appearances per line key? |
| 19:04.06 | mcreedjr | wunderkin: not like a Cisco phone, no, there are no line overlays |
| 19:04.09 | wunderkin | ok, thanks |
| 19:04.20 | mcreedjr | wunderkin: you can configure one extension on the phone and then each line key can be a call instance for the configured extension |
| 19:04.31 | mcreedjr | c'mon one of you smart asterisk dudes help me with my CID problem :) |
| 19:04.34 | [TK]D-Fender | wunderkin: Nope, nor multiple CALLS.... SPA = bleh |
| 19:04.36 | wunderkin | heh, yes, i am just trying to compare it to the polys, since the spa-942 was their origional decision |
| 19:04.41 | [TK]D-Fender | wunderkin: Pay for more than 2 regs? Ew. 1 call per line key FIXED? ew. Puny LCD? Ew. I evven like the 301 better than the SPA's |
| 19:04.44 | *** join/#asterisk mercestes (n=merceste@216.54.143.2) |
| 19:04.51 | [TK]D-Fender | wunderkin: Also Polycom's superior audio quality. |
| 19:05.08 | mcreedjr | [TK]D-Fender: not to mention the auto-answer page functionality on the SPA-942, if you're paged while on an active call, it puts the call on hold to answer the page. Yuck! |
| 19:05.24 | [TK]D-Fender | mcreedjr: I won't comment on that as it'll be a plus for 1 guy, and a minus the next. I like CHOICE personally. Polycom offers me more choices than pretty much anyone else. |
| 19:05.29 | [TK]D-Fender | Quality products, decent price. I'm willing to pay for the good stuff... |
| 19:05.33 | mcreedjr | [TK]D-Fender: well thats part of my argument, that functionality on the SPA is certainly not ideal for me, and I have no way to change it |
| 19:05.39 | [TK]D-Fender | mcreedjr: Change your dial-plan around it obviously! |
| 19:05.42 | wunderkin | wow the digium installation guides are crappy now |
| 19:05.42 | mcreedjr | [TK]D-Fender: done so, but i mean no options on the phone itself :) |
| 19:05.45 | wunderkin | are gains set in zaptel.conf? mcreedjr, maybe your gains are too low/high? i dont have zaptel on my current box.. |
| 19:05.49 | [TK]D-Fender | mcreedjr: Yeah.... oh well.... SPA's are a budget choice for cheaper companies outside of reasonable Polycom pricing. |
| 19:05.52 | mcreedjr | wunderkin: i guess thats the last thing i haven't played with much |
| 19:06.04 | mcreedjr | wunderkin: any other ideas? |
| 19:06.20 | wunderkin | not that i know of.. |
| 19:06.28 | *** join/#asterisk Assid (i=assid@203.115.83.215) |
| 19:06.29 | Assid | heya |
| 19:06.29 | mcreedjr | wunderkin: thanks... |
| 19:06.29 | Assid | anyone here using voicepulse |
| 19:21.49 | *** join/#asterisk r0d3nt (n=RatMan@foster.stonedcoder.org) |
| 19:21.49 | aydiosmio | amg |
| 19:21.49 | aydiosmio | it's r0d3nt |
| 19:21.49 | *** join/#asterisk obiwanmikenolte (n=obiwanmi@mail.efc-intl.com) |
| 19:21.50 | syzygyBSD | so when is someone going to hack linux onto a polycom? |
| 19:21.50 | mcreedjr | [TK]D-Fender: Does BLF on the Polycoms work with Asterisk? |
| 19:21.55 | [TK]D-Fender | mcreedjr: Yup. I've got a 601 w/ 2 Modules fully loaded and lit up like a Christmas tree :) |
| 19:21.55 | [TK]D-Fender | syzygyBSD: Why bother... I jsut want the phone to do its thing.... hom much more should we ask for... its not a strong processor as it is. |
| 19:22.03 | mcreedjr | [TK]D-Fender: sweet... care to share any tricks, or isn't is that difficult? |
| 19:22.07 | [TK]D-Fender | mcreedjr: Not much to say that isn't on the WIKI and in the admin guide. |
| 19:22.07 | syzygyBSD | [TK]D-Fender: how bout the ability for it not to restart and take 10minutes everytime I want to update any part of the configuration |
| 19:22.12 | Assid | BLF = MWI ? |
| 19:22.16 | mcreedjr | [TK]D-Fender: cool, thanks |
| 19:22.20 | mcreedjr | Assid: Busy lamp field indicator |
| 19:22.29 | mcreedjr | Assid: line key blinks when extension is in use. |
| 19:22.33 | syzygyBSD | or a nicer web gui for configuring |
| 19:22.37 | [TK]D-Fender | syzygyBSD: Already done... its called PROVISIONING ;) |
| 19:22.37 | [TK]D-Fender | vi / emacs / whatever! |
| 19:22.47 | obiwanmikenolte | syzgyBSD: or you'll have to use snoms or Grandstreams and deal with worse call quality |
| 19:22.55 | syzygyBSD | yet to hear a web gui... |
| 19:23.03 | syzygyBSD | and what if the phone isn't on the same network as a server you can provision from? |
| 19:23.06 | [TK]D-Fender | syzygyBSD: Who needs it? Not I.... |
| 19:23.11 | *** join/#asterisk batphone (n=will@69.15.174.114) |
| 19:23.16 | obiwanmikenolte | Use FTP |
| 19:23.30 | *** join/#asterisk CrummyGummy (n=wayne@dsl-145-99-158.telkomadsl.co.za) |
| 19:23.33 | batphone | anyone know what would cause polycoms to hang up on a caller when trying to retrieve a call put on hold? |
| 19:23.41 | syzygyBSD | obiwanmikenolte: the call quality on my spa is fine |
| 19:23.45 | aydiosmio | syzygyBSD: modifying the existing firmware woud be easier. |
| 19:23.48 | syzygyBSD | FTP only works if it is setup on the phone.. |
| 19:23.52 | syzygyBSD | aydiosmio: yes, but I don't have a copy of that |
| 19:24.05 | [TK]D-Fender | syzygyBSD: And that takes 5 seconds flat coming out of the box..... |
| 19:24.09 | syzygyBSD | dun know, I have always configured it through the web |
| 19:24.13 | Assid | okay anyone here using voicepule |
| 19:24.19 | syzygyBSD | but I use it for more testing then my main phone |
| 19:24.23 | aydiosmio | solution looking for a problem |
| 19:24.27 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 19:24.31 | syzygyBSD | so I change servers/ports frequently, provisioning only works when it is setup/restarted doesn't it? |
| 19:24.35 | [TK]D-Fender | SPA audio quality is better than most. a clear 3rd. |
| 19:24.43 | [TK]D-Fender | syzygyBSD: Same with any change.... |
| 19:24.43 | Bullseye_Network | why would I get "No Match Their CallID" many times in debug? |
| 19:24.48 | [TK]D-Fender | syzygyBSD: the phone looks to pickup its config every time it restarts and just uses the last one it got if it can't reach the provisioning server. |
| 19:24.53 | syzygyBSD | yes, but if the phone isn't on my desk.. say at a clients, then I cna't restart it with provisioning |
| 19:24.56 | mcreedjr | syzygyBSD: Some SIP endpoints support rebooting via SIP NOTIFY IIRC |
| 19:25.01 | *** part/#asterisk CrummyGummy (n=wayne@dsl-145-99-158.telkomadsl.co.za) |
| 19:25.01 | aydiosmio | that sounds like fun. |
| 19:25.05 | syzygyBSD | hmm, may have to look into that |
| 19:25.09 | aydiosmio | what are you changing? |
| 19:25.09 | aydiosmio | serversa and ports? |
| 19:25.14 | hmmhesays | [TK]D-Fender: so I got my eye on a hamer studio |
| 19:44.00 | *** join/#asterisk charles___ (n=charles@fw.invosat.com) |
| 19:44.17 | [TK]D-Fender | hmmhesays: link me... |
| 19:44.21 | syzygyBSD | and other things, depends on what I am trying to do that day |
| 19:44.25 | hmmhesays | http://www.musiciansfriend.com/product/Hamer-USA-Studio-Custom-Electric-Guitar?sku=516246 |
| 19:44.28 | *** part/#asterisk matlj (n=mlejeune@mat.zapto.org) |
| 19:44.28 | [TK]D-Fender | hmmhesays: OUCH |
| 19:44.31 | hmmhesays | beautiful guitar though, my uncle owns one |
| 19:44.34 | hmmhesays | i love it |
| 19:44.34 | aydiosmio | that's not bad |
| 19:44.37 | [TK]D-Fender | hmmhesays: Seriously... OUCH |
| 19:44.40 | aydiosmio | the Gibson Customs are lik 6-7000 |
| 19:44.40 | hmmhesays | yeah spendy bugger |
| 19:45.15 | charles___ | hey guys |
| 19:45.15 | [TK]D-Fender | My last one had cosmetic damage and I got ita half for $300. I walked in with an empty case ready to pay full price when I found out what happened. |
| 19:45.15 | charles___ | how do you get the callforwarding variable from console ? |
| 19:45.15 | hmmhesays | last what? |
| 19:45.15 | [TK]D-Fender | charles___: What call forwarding variable? |
| 19:45.16 | charles___ | [TK]D-Fender: it's probably using DBPUT here |
| 19:45.16 | [TK]D-Fender | charles___: ... "it"? |
| 19:45.16 | charles___ | [TK]D-Fender: probably dbput(CFIM |
| 19:45.16 | charles___ | [TK]D-Fender: is there a way to get DBGET or something similar on console ? |
| 19:45.16 | charles___ | echo {DB} |
| 19:45.16 | charles___ | ? |
| 19:45.16 | [TK]D-Fender | charles___: "database show [family] [key]" |
| 19:45.16 | Cresl1n | syzygyBSD: what do you do with ss7? |
| 19:45.17 | charles___ | [TK]D-Fender: great thanks |
| 19:45.17 | syzygyBSD | well, nothing yet |
| 19:45.17 | *** join/#asterisk ellisgl (i=keefejoh@seraph.techwareit.com) |
| 19:45.17 | charles___ | yeah looks like the extension got jammed |
| 19:45.17 | ellisgl | Question: If my signalling on my t1 blinks out for a couple ms - how to do I get asterisk from dropping the connection? |
| 19:45.17 | charles___ | it says in use, I already restarted the cisco 7940 |
| 19:45.17 | charles___ | but still show inuse |
| 19:45.37 | [TK]D-Fender | charles___: what shows "inuse"? I think you'd better be a little more forthcoming with what your talking about here..... |
| 19:45.53 | *** join/#asterisk Waverly360 (n=mirc@209.12.249.243) |
| 19:45.55 | hmmhesays | hmm is there any way to make ${CDR(dst)} writeable? |
| 19:45.55 | *** join/#asterisk QbY (n=Kelvin@cm-64-221-172-66.dhcp.southerncoastalcable.net) |
| 19:45.55 | syzygyBSD | any ideas why rxfax would be segfaulting on me? |
| 19:45.55 | QbY | does anynone know what would cause asterisk to not join the two monitor recordings when in queues.conf it states "monitor-join = yes" -- My queue calls are being recorded, however i end up with two -in & -out |
| 19:45.55 | syzygyBSD | QbY: do you have soxmix? |
| 19:45.55 | QbY | let me check |
| 19:45.55 | syzygyBSD | somix |
| 19:45.55 | QbY | well hell no.. that'd be to simple.. |
| 19:45.59 | Zodiacal | is it legal for an employer to record an employees phone calls with out telling the employee? |
| 19:46.02 | *** join/#asterisk }btorch{ (n=kvirc@adelphi.geofocus.com) |
| 19:46.10 | *** join/#asterisk Mandrak3 (n=io@81.27.211.30) |
| 19:46.10 | }btorch{ | is there a channel for zaptel related issues only ? |
| 19:46.10 | }btorch{ | my echos have gotten so worse after my asterisk/zaptel/libpri latest update |
| 19:46.10 | }btorch{ | it sucks |
| 19:46.10 | Cresl1n | }btorch{: which version did you update to? |
| 19:46.11 | ellisgl | Hm... #zaptel would be nice - I need that too.. |
| 19:46.11 | Mandrak3 | Hi everybody! |
| 19:46.11 | }btorch{ | the latest at the time which was 1.2.10 and 1.2.7 I think |
| 19:46.11 | Mandrak3 | I have a problem with Congestion.... I'm trying to place calls with an outgoing trunk |
| 19:46.11 | Mandrak3 | after 2 calls this trunk go in Congestion |
| 19:46.11 | }btorch{ | ellisgl: that would be awosome |
| 19:46.17 | ellisgl | I need to figure out how to make the t1 not reset when the link blinks on and off really quick |
| 19:46.21 | }btorch{ | ellisgl: what you mean ? |
| 19:46.24 | *** join/#asterisk arkonadev (n=arkonaj@65.203.186.131) |
| 19:46.31 | }btorch{ | ellisgl: the spans are reloading ? |
| 19:46.38 | *** join/#asterisk n00dle (n=ccraft@hillel.springsips.com) |
| 19:46.52 | Mandrak3 | How can i manage this ? I need it to setting up LCR |
| 19:46.57 | arkonadev | what channel do i want to use when using the OriginateAction to make a call to an external phonenumber |
| 19:47.01 | *** join/#asterisk [hC] (n=hardcore@mail.metrobridge.com) |
| 19:47.05 | ellisgl | my setup - asterisk box to a m13 mux - to some ds3 equipment to simple 2 node sonet ring |
| 19:47.14 | n00dle | Good day, everyone. :) Anyone know if an Ariel RS4200 can be convinced to talk to *? |
| 19:47.14 | ellisgl | and if I do a protection switch over on my ds3 or sonet stuff - the t1 is dropped on the asterisk side |
| 19:47.18 | vosque | is there anything I can do to make a .call file avoid going to voicemail other than keeping the WaitTime less than the voicemail timeout? |
| 19:47.22 | ellisgl | and takes 2 seconds to come backup and the establish call is gone |
| 19:47.22 | hmmhesays | can anyone tell me where ${CDR(dst)} is set read only? |
| 19:47.26 | ellisgl | oh forgot that I also have a channel bank - and test that side - no problem with it.. |
| 19:47.31 | *** join/#asterisk mivck (i=1000@200.114.70.228) |
| 19:47.35 | charles___ | [TK]D-Fender: show hints was showing the extension as InUse |
| 19:47.38 | }btorch{ | ellisgl: I'm not sure if this is the same problem that you are having but I did had a reset of span issue a while back |
| 20:02.53 | charles___ | [TK]D-Fender: but phone has restart several times, so I have had to restart asterisk to clear |
| 20:03.04 | ellisgl | }btorch{: how did you fix it? |
| 20:03.11 | }btorch{ | ellisgl: the calls would drop and asterisk would reload the spans .. I had to increase my resetinterval to 30000000 |
| 20:03.16 | ellisgl | which file is that in? |
| 20:03.16 | }btorch{ | zapata.con |
| 20:03.19 | }btorch{ | f |
| 20:03.22 | }btorch{ | I think that was the only change I did to fix that but I can't really remember since it was some time ago |
| 20:03.29 | ellisgl | going to try it now. |
| 20:03.29 | charles___ | [TK]D-Fender: is there a way to someway fix that particular extension ? |
| 20:03.33 | r0d3nt | aydiosmio, i'm always here. |
| 20:03.36 | r0d3nt | <SecNews> Title: [Full-disclosure] [MU-200608-01] Multiple Vulnerabilities in Asterisk 1.2.10 |
| 20:03.39 | r0d3nt | <SecNews> Link: http://lists.rootsecure.net/?p=view&l=full_disclosure&m=38021 |
| 20:03.39 | r0d3nt | <SecNews> |
| 20:03.42 | Mandrak3 | Anyone can help me with LCR problems? |
| 20:03.45 | r0d3nt | that is rare.... |
| 20:03.54 | }btorch{ | how do you guys troubleshoot the any echo problems ... I haven't been able to fix that and I have tried several ways |
| 20:03.54 | arkonadev | in order to dial out to an external nubmer asterisk uses the zap/1-1 channel correct? |
| 20:03.59 | Cresl1n | }btorch{: what is your setup |
| 20:04.02 | }btorch{ | I uses zmonitor and played around with my rxgain and txgain settings, changed my zconfig.h file used several softphones, used bluetooth and regular headsets.... etc |
| 20:04.05 | ellisgl | }btorch{: that didn't work. =/ |
| 20:04.05 | bkw_ | Aug 23 15:44:18 WARNING[25744]: channel.c:787 channel_find_locked: Avoided initial deadlock for '0x817d498', 10 retries! |
| 20:04.09 | bkw_ | *PUNT* |
| 20:04.12 | hmmhesays | bkw_: help me |
| 20:04.15 | bkw_ | SCORE |
| 20:04.15 | hmmhesays | pleeeaaase |
| 20:04.18 | *** join/#asterisk techie (n=gus@adsl-068-209-242-072.sip.mia.bellsouth.net) |
| 20:04.22 | bkw_ | hmmhesays, yes? |
| 20:04.24 | hmmhesays | well i kind of have a hangover, and I'm trying to figure out how I make CDR(dst) writeable via Set |
| 20:04.31 | }btorch{ | Cresl1n: my setup is something like PSTN -> asterisk -> SIEMENS PBX and voip |
| 20:04.33 | Cresl1n | ok |
| 20:04.36 | bkw_ | you can't its read only if I recall |
| 20:04.36 | Cresl1n | what echo canceler do you have selected in zconfig.h? |
| 20:04.39 | hmmhesays | bkw_ I know this |
| 20:04.41 | }btorch{ | ellisgl: your problem is probably something else |
| 20:04.41 | hmmhesays | I want to make it rw |
| 20:04.43 | bkw_ | check cdr.h |
| 20:04.49 | hmmhesays | like accountcode |
| 20:04.49 | bkw_ | and cdr.c |
| 20:05.07 | }btorch{ | Cresl1n: KB1 the default one |
| 20:05.11 | hmmhesays | CDR(accountcode) looks the same as the ro variables |
| 20:05.15 | }btorch{ | Cresl1n: I tried all the other ones but they just made it worse |
| 20:05.31 | }btorch{ | specially MG2 |
| 20:05.34 | Cresl1n | }btorch{: what kind of lines are they, analog or T1? |
| 20:05.38 | }btorch{ | T1 |
| 20:05.38 | bkw_ | I don't think its been setup to allow you to set DST |
| 20:05.42 | Cresl1n | what signalling? |
| 20:05.51 | bkw_ | in the function thingy |
| 20:05.51 | hmmhesays | in the Set function? |
| 20:06.07 | }btorch{ | pri_cpe |
| 20:06.07 | Cresl1n | ok |
| 20:06.10 | bkw_ | in the CDR() Function |
| 20:06.17 | Cresl1n | can you pastebin your zapata.conf? |
| 20:06.20 | }btorch{ | sure |
| 20:06.20 | Mandrak3 | Anyone can help me with LCR problems? |
| 20:06.33 | hmmhesays | bah I'm looking |
| 20:06.36 | }btorch{ | http://pastebin.ca/146659 |
| 20:07.09 | Cresl1n | first of all, set echotraining=no |
| 20:07.09 | Cresl1n | that usually shouldn't be used with PRI |
| 20:07.10 | Cresl1n | I can't remember if it actually does anything with PRI signalling, but whatever it does is probably wrong |
| 20:07.10 | }btorch{ | Cresl1n: today I even tested from a voip(using a bluetooth headset) to call an internal phone behind the PBX and also my cell phone. both had horrible echo |
| 20:07.10 | }btorch{ | Cresl1n: you sure |
| 20:07.10 | Cresl1n | }btorch{: yes |
| 20:07.10 | Cresl1n | ok, let me ask a question |
| 20:07.15 | aydiosmio | that'll be a dollar. |
| 20:07.15 | }btorch{ | shoot |
| 20:07.24 | Cresl1n | who hears the echo, the person calling from the PSTN, or the person behind asterisk on the PBX? |
| 20:07.28 | }btorch{ | whomever is either behind the PSTN or a regular siemens phone which is behind the PBX |
| 20:07.28 | charles___ | why would show hints , show the extension INUSE while it's not ? |
| 20:07.28 | charles___ | is there a way to change the status of the extension ? |
| 20:07.32 | *** part/#asterisk QbY (n=Kelvin@cm-64-221-172-66.dhcp.southerncoastalcable.net) |
| 20:45.58 | }btorch{ | when I call from voip to voip I don't remember hearing a lot of echo but I have not tested with this bluetooth headset |
| 20:46.11 | [TK]D-Fender | charles___: Restart * |
| 20:46.11 | }btorch{ | I do remember hearing some kind of white noise |
| 20:46.11 | *** join/#asterisk gbodemantv (n=gbodeman@216.142.38.154) |
| 20:46.11 | Cresl1n | }btorch{: you answer wasn't clear enough |
| 20:46.11 | }btorch{ | sorry |
| 20:46.11 | Cresl1n | }btorch{: who hears the echo, the PSTN side or the side on the inside of asterisk |
| 20:46.11 | Assid | okay.. VP's servers are going up and down like a yoyo |
| 20:46.11 | }btorch{ | ok let me try again |
| 20:46.11 | gbodemantv | hi all |
| 20:46.12 | gbodemantv | so if I wanted to write a she ll script to capture the last 500 lines in the full.log, does anyone know how I do that |
| 20:46.12 | ellisgl | where the t1 information logged too? |
| 20:46.14 | gbodemantv | clueless here |
| 20:46.23 | Juggie | gbodemantv, tail -n 500 full.log |
| 20:46.37 | charles___ | [TK]D-Fender: that will kill everyone talking |
| 20:46.41 | }btorch{ | Assume I'm using my voip account and I use idefisk and I call a cell phone . The person on the cell phone hears the echo |
| 20:46.58 | }btorch{ | Cresl1n: so the person behind the PSTN |
| 20:47.12 | charles___ | [TK]D-Fender: any way to kill the inuse but not kill everyone else :P |
| 20:47.12 | [TK]D-Fender | charles___: well if hints think its in ue then you need to find the dead channel |
| 20:47.24 | gbodemantv | any way to make it a script and write it to a text file? |
| 20:47.28 | Cresl1n | that means that the echo is being generated by your phones or something |
| 20:47.32 | gbodemantv | like 500.txt |
| 20:47.35 | charles___ | [TK]D-Fender: is there a way to kill one channel only ? |
| 20:47.41 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
| 20:47.44 | [TK]D-Fender | charles___: "soft hangup [channel] |
| 20:47.48 | Cresl1n | so the extensions on the siemens PBX don't hear the echo, but the person they talk to do? |
| 20:47.48 | aydiosmio | gbodemantv: use tail |
| 20:47.53 | aydiosmio | man tail |
| 20:48.10 | Juggie | gbodemantv, tail -n 500 full.log > last500.log |
| 20:48.10 | charles___ | [TK]D-Fender: great thanks |
| 20:48.26 | }btorch{ | Cresl1n: I don't know about that ... well hold on the extension behind the siemens PBX also have the echo issue if I call them from whithin my voip extension |
| 20:48.38 | Cresl1n | ok, ok |
| 20:48.45 | Cresl1n | well, here's what I'll do |
| 20:49.06 | gbodemantv | Juggie: Thanks |
| 20:49.10 | aydiosmio | gbodemantv: you're welcome |
| 20:49.16 | Cresl1n | I'll give you a list of steps you can do to attempt to make this better if I understand you correctly |
| 20:49.25 | }btorch{ | Cresl1n: let me ask you this though |
| 20:49.38 | gbodemantv | aydiosmio: Thanks |
| 20:49.43 | Cresl1n | oh, and what kind of T1 card are you using? |
| 20:50.00 | }btorch{ | digium TE2XX and a TE1XX |
| 20:50.22 | Cresl1n | w/o hardware echocan? |
| 20:50.38 | Juggie | those dont have echocan so i would assume yes. |
| 21:11.43 | Cresl1n | }btorch{: what was your question? |
| 21:11.43 | kb3ien | back |
| 21:11.44 | *** join/#asterisk bethaud (n=eamonn@host-84-9-27-232.bulldogdsl.com) |
| 21:11.44 | ellisgl | what log is the t1 info stored in - ie connects and disconnects of the phyiscal? |
| 21:11.45 | }btorch{ | Cresl1n: sorry I'm back |
| 21:11.46 | kb3ien | trying to conquer this problem : in the compile `gcc -g3 -o asterisk -Wl,-E io.o sched.o logger.o frame.o loader.o config...stdtime/libtime.a -lncurses -lm -lpthread -lcrypto -lm -L/lib -R/lib -lncurses -lssl` |
| 21:12.28 | *** join/#asterisk QbY (n=Kelvin@cm-64-221-172-66.dhcp.southerncoastalcable.net) |
| 21:12.39 | *** join/#asterisk razu (n=rasmus@tln-kontor.norby.ee) |
| 21:12.55 | kb3ien | i get horrible error: ld: cannot find -lncurses |
| 21:13.03 | QbY | holy hell.. my music on hold is making people's ears bleed.. how do i lower the volume? |
| 21:13.06 | Qwell | kb3ien: Do you have ncurses installed? |
| 21:13.12 | Qwell | QbY: quietmp3? |
| 21:13.23 | bethaud | would you expect to see ~50% on the RX channel in ztmonitor while idle? I'm guessing dial-tone, but ... |
| 21:13.31 | QbY | Qwell.. That's what its set at.. |
| 21:13.31 | Qwell | QbY: nice. you could custom tune the mpg123 params |
| 21:13.35 | QbY | mode=quietmp3 |
| 21:13.40 | }btorch{ | Cresl1n: well my Siemens PBX is connected to span2 of the digium box on my asterisk and span1 is connected to the PSTN.. when I call from a siemens extension the quality is great no issues |
| 21:13.54 | }btorch{ | Cresl1n: is that just a voip issues ? latency ? |
| 21:14.06 | }btorch{ | Cresl1n: what you mean by hardware cancelling? I hope those cards have that they not cheap |
| 21:14.21 | *** join/#asterisk jhiver (n=jhiver@LReunion-151-2-164.w193-253.abo.wanadoo.fr) |
| 21:14.28 | jhiver | hi all |
| 21:14.42 | }btorch{ | what I really hate is the fact that everytime I have to update my asterisk I have to completly remove everything, zapel, libpri, asterisk to be able to install a new one .. why is that ? why can't it just be overwriten ? |
| 21:14.52 | jhiver | i have a pretty stange bug when using SER in conjunction with Asterisk |
| 21:15.05 | Cresl1n | }btorch{: latency |
| 21:15.14 | jhiver | I have: Asterisk A ------> SER -------> Asterisk B |
| 21:15.20 | Cresl1n | }btorch{: you shouldn't have to remove everything for it to upgrade |
| 21:15.24 | jhiver | I run the echo() app on Asterisk B |
| 21:15.41 | Cresl1n | }btorch{: ok, so this is what you try next |
| 21:15.45 | Cresl1n | turn off echotraining |
| 21:15.45 | jhiver | I try to call asterisk B from asterisk A through SER, and when I do that I have no audio |
| 21:15.52 | }btorch{ | that's what I think but others have told me different |
| 21:16.01 | }btorch{ | ok |
| 21:16.07 | Cresl1n | try turning off echocancelwhenbridged |
| 21:16.43 | Cresl1n | that's usually not what you want |
| 21:16.51 | jhiver | however if I call directly without using SER, I have audio |
| 21:16.56 | jhiver | any ideas what's going on? |
| 21:16.56 | Cresl1n | see how it sounds |
| 21:17.06 | }btorch{ | ok |
| 21:17.18 | Cresl1n | if it still sounds bad, this is the next step |
| 21:17.30 | jhiver | it seems that Asterisk B can't send audio to Asterisk A for some reason |
| 21:17.41 | Cresl1n | download zaptel from trunk |
| 21:17.41 | Cresl1n | use the mg2 echo canceler from there |
| 21:17.45 | Cresl1n | see if it sounds better |
| 21:17.56 | Cresl1n | if not, try setting echocancel=256 |
| 21:32.03 | Cresl1n | see if it sounds better |
| 21:32.06 | }btorch{ | I do remember that in the past the rxgain and txgain settings did help but I set them back to 0.0 for some reason , don't remeber wele |
| 21:32.11 | }btorch{ | ok |
| 21:32.11 | Cresl1n | don't mess with gains yet |
| 21:32.15 | Cresl1n | the best echo canceler so far is MG2 in trunk |
| 21:32.18 | Cresl1n | that's what you should be using to test all of this on |
| 21:32.55 | Cresl1n | if echocancel=256 sounds worse, try turning back on echocancelwhenbridged |
| 21:33.05 | Cresl1n | that may help |
| 21:33.05 | }btorch{ | for some reason I have had worse echo with that canceler .. first I will try changing the zapata.conf settings |
| 21:33.17 | }btorch{ | recompiling the zaptel I can only do that at night time |
| 21:33.25 | Cresl1n | MG2 from 1.2 and MG2 from trunk are different |
| 21:33.38 | }btorch{ | oh hold |
| 21:33.53 | Cresl1n | and if that still doesn't work and you're REALLY fed up, turn on the AGGRESSIVE_SUPPRESSOR option in zconfig .h |
| 21:33.53 | }btorch{ | http://svn.digium.com/svn/zaptel/trunk ? |
| 21:34.03 | Cresl1n | that'll fix it, but I always hate telling people to use that |
| 21:34.06 | Cresl1n | yep |
| 21:34.13 | }btorch{ | ok |
| 21:34.16 | Cresl1n | it should work with asterisk-1.2 |
| 21:34.20 | Cresl1n | (IIRC) |
| 21:34.23 | bethaud | Cresl1n: would you recommend using zaptel from trunk with * 1.2.10 ? |
| 21:34.30 | Cresl1n | bethaud: if 1.2 version of zaptel works for you, don't mess with it |
| 21:34.34 | }btorch{ | hey if I recompile my zaptel I shouldn't have to recompile asterisk right ? it uses shared modules |
| 21:34.47 | Cresl1n | }btorch{: mmmm..... you shouldn't have to IIRC |
| 21:34.50 | pigpen | My echo issues are only in the first 5 seconds of a call...using a pri and polycom phones...(the user probably has the volume too high on the handset as thy can hear themselves...) |
| 21:34.57 | pigpen | ... and only the receptionist is complaining... |
| 21:35.05 | Cresl1n | pigpen: who hears it, the SIP phone or the person at the other end of the PRI? |
| 21:35.22 | pigpen | Receptionist on a Polycom 601... |
| 21:35.32 | pigpen | Remote end never hears it. |
| 21:35.32 | *** join/#asterisk bethaud (n=eamonn@host-84-9-27-232.bulldogdsl.com) |
| 21:35.40 | Cresl1n | yeah, it's possible that their volume is too high on the handset |
| 21:35.48 | [TK]D-Fender | pigpen: EC is all the resposibility of the PSTN termination, not SIP phones. |
| 21:36.03 | pigpen | yeah..she has a "penetrating" voice... |
| 21:36.29 | Cresl1n | is it for every person on every call, or just this one person on a particular number? |
| 21:36.37 | [TK]D-Fender | pigpen: You can selectively lower the gain on her mic, but Polcoms already have AEC as it is. If she's actually responsible for it well... something is really wrong |
| 21:36.42 | bethaud | Cresl1n: apologies, my client crashed. I'm having some difficulties with volume and hangup detection in the UK with an X101P & zaptel 1.2.8 |
| 21:36.51 | pigpen | Only the receptionist is complaining...but we went live 1 week ago, then school started...so everyone is busy...even with complaints...(yes, it is a school) |
| 21:36.54 | *** join/#asterisk meshuga (i=meshuga@c-71-231-141-145.hsd1.or.comcast.net) |
| 21:37.02 | pigpen | I will probably have her try out a plantonics headset... |
| 21:57.01 | [TK]D-Fender | pigpen: Well some people won't even complain... verify the scope of the problem first, not just its primary whiner ;) |
| 21:57.25 | pigpen | yeah...good idea... |
| 21:57.36 | [TK]D-Fender | pigpen: I can tell you from more than a few different Polycom installs I can't believe its the phone.... |
| 21:57.40 | Cresl1n | pigpen: well, you can try following the same steps I gave }btorch{ |
| 21:57.45 | [TK]D-Fender | pigpen: maybe HER, but not the phone.... |
| 21:57.48 | Cresl1n | if you really want to |
| 21:57.54 | Cresl1n | or you could try adjusting her phone for her |
| 21:57.54 | Cresl1n | that's what I would do if it's the exception |
| 21:57.54 | Cresl1n | a lot easier |
| 21:57.54 | pigpen | yeah...the only issue I have had was due to someone having their handset too loud... |
| 21:57.54 | Cresl1n | pigpen: you could try setting your txgain to -1 or something like that |
| 21:57.54 | Cresl1n | that might help a bit |
| 21:58.00 | pigpen | hmm..can you force the volume in the conf file? |
| 21:58.04 | Cresl1n | but I'd try to fix it through technology first, rather then tweaking gains |
| 21:58.04 | pigpen | (for the polycom that is) |
| 21:58.08 | [hC] | mitcheloc: you awake? |
| 21:58.16 | pigpen | but yes...I would hate to knock down the gain if it works fine after the first few seconds... |
| 21:58.32 | [TK]D-Fender | pigpen: Then you'll want a good HWEC card... |
| 21:58.32 | Cresl1n | pigpen: nah, maybe we can fix it |
| 21:58.35 | pigpen | I heard the new one from digium helps with the first few seconds of echo.... |
| 21:58.47 | Cresl1n | pigpen: have you tried MG2 as well? |
| 21:58.51 | pigpen | no..I have not modifed the source... |
| 21:58.57 | Cresl1n | pigpen: the new one is amazing. It will kill that echo dead. |
| 21:58.57 | Cresl1n | pigpen: make sure you try MG2 |
| 21:59.00 | pigpen | what is default? |
| 21:59.06 | Cresl1n | pigpen: it's the latest and greatest in echo can technology |
| 21:59.11 | pigpen | MG(1) |
| 21:59.18 | Cresl1n | pigpen: in 1.2, KB1 is |
| 21:59.18 | Cresl1n | it was good |
| 21:59.21 | Cresl1n | MG2 from trunk is the best though |
| 21:59.23 | [hC] | Anyone used asterisk to control door locks yet? |
| 21:59.27 | Cresl1n | pigpen: do you have echotraining enabled? |
| 21:59.27 | pigpen | Cresl1n, yes. |
| 21:59.47 | Cresl1n | pigpen: this is on PRIs right? |
| 21:59.58 | pigpen | yes...a lovley PRI from SWB |
| 21:59.59 | Cresl1n | turn that off |
| 22:00.33 | Cresl1n | echotraining generally speaking shouldn't be enabled w/PRIs |
| 22:00.33 | pigpen | really? |
| 22:00.50 | Cresl1n | yeah, it was made for analog lines |
| 22:00.54 | pigpen | Well... I guess in theory, PRI's are digital, so is * and the Polycom's are...so I guess No echo should be possible... |
| 22:00.58 | Cresl1n | psssh |
| 22:01.06 | Juggie | Cresl1n, suggest echotraining=yes/no/allways |
| 22:01.18 | Juggie | yes=analog, no=no allways=analog/t1 |
| 22:01.34 | Cresl1n | echotraining=no on PRIs |
| 22:01.38 | Cresl1n | at least when you're debugging echo problems |
| 22:01.58 | *** join/#asterisk dserban (n=dserban@caliban.lodgingcompany.com) |
| 22:02.18 | Juggie | i know i'm just saying for ppl who want to force it. |
| 22:02.18 | [hC] | what about echocancelwhenbridged ? :) |
| 22:02.18 | Cresl1n | when you're debugging, keep it off |
| 22:02.19 | [hC] | what about when you're not debugging |
| 22:02.19 | kb3ien | Im trying to compile asterisk, but cannot resolve this error: ld: cannot find -lncurses |
| 22:02.19 | Cresl1n | [hC]: only if it improves it |
| 22:02.19 | Juggie | kb3ien, what distro. |
| 22:02.27 | [hC] | Ive had echotraining and cancelwhenbridged on on my pbx for ever. Never noticed a problem. |
| 22:02.29 | dserban | Hi! love to barge in and ask questions ;). I have a tdm2400p and it answers the phone!, it plays my greeting!, it can't hear anything via the ztmonitor tool for the channel that's being dialed. nothing incoming, where do I begin (I've been hunting for answers for a few hours now) |
| 22:02.29 | Cresl1n | [hC]: it won't always cause problems. If you haven't heard one, then they're obviously not causing one for you |
| 22:02.29 | pigpen | I haven't tried it with it off....hehe |
| 22:26.05 | [TK]D-Fender | pigpen: Not true. Sure the T1 is digital, but what aabout the OTHER side of the call? Latency can cause it as well.. so many different little things. And HELL YEAH you want HWEC on your card.... |
| 22:26.05 | Cresl1n | but other people have different setups |
| 22:26.05 | hmmhesays | please tell me whyyyyy |
| 22:26.05 | hmmhesays | my car is in the front yard and I am sleeping with my clothes on |
| 22:26.06 | Cresl1n | hmmhesays: you drank too much |
| 22:26.06 | [TK]D-Fender | hmmhesays: Lit? |
| 22:26.06 | hmmhesays | [TK]D-Fender: ja |
| 22:26.06 | hmmhesays | Cresl1n: last night I did |
| 22:26.06 | [hC] | hmmhesays: I came in thru the window last night! |
| 22:26.06 | hmmhesays | we have practice tonight though bah |
| 22:26.06 | dserban | pressing numbers to go to menu options does nothing... |
| 22:26.06 | [TK]D-Fender | hmmhesays: Another great piece in the key of EMaj |
| 22:26.07 | hmmhesays | and your...gone.... GONE |
| 22:26.07 | *** join/#asterisk Nitrus^ (n=Nitrus_@72-34-76-86.skyriver.net) |
| 22:26.07 | hmmhesays | it transitions into 99 red balloons perfectly |
| 22:26.07 | [TK]D-Fender | hmmhesays: I've medley'd up that, plus "Movies" from AAF, Basket Case before... |
| 22:26.10 | hmmhesays | eeeeeeeeee f######### aaaaaaaaa bbbbbbbb |
| 22:26.22 | hmmhesays | punk can be fun and drunk people love it |
| 22:26.31 | *** part/#asterisk bethaud (n=eamonn@host-84-9-27-232.bulldogdsl.com) |
| 22:26.31 | [TK]D-Fender | hmmhesays: Yup.... |
| 22:26.31 | hmmhesays | and most importantly I can play it drunk |
| 22:26.39 | [TK]D-Fender | lol... ok, I'm outta here.... |
| 22:26.39 | [TK]D-Fender | bbiab |
| 22:26.45 | hmmhesays | later |
| 22:27.00 | *** join/#asterisk morex (i=morex@host86-133-31-162.range86-133.btcentralplus.com) |
| 22:27.07 | morex | Hello Asterisk Community :-) |
| 22:27.26 | pigpen | The Community says Hi back. |
| 22:27.26 | morex | Does anyone out there have experience connectin an Avaya Definity to Asterisk or Yate over H.323? |
| 22:27.27 | morex | Hi Pigpen |
| 22:27.27 | mercestes | Does * block anonymous calling?? |
| 22:27.27 | Nitrus^ | im having a huge delay problem with asterisk or possibly my phone provider. when someone calls in and says hello, my end doesnt hear the hello, but after about 2 seconds the conversation goes as normal. it seems like a delay in connecting the channels. does anyone know what might be causing this? |
| 22:27.27 | [hC] | Nitrus^: how are you connecting to your phone provider? Sounds like it could be a channel bridging delay |
| 22:27.43 | Nitrus^ | im using vonage on adtran 750 FXO ports |
| 22:27.49 | *** join/#asterisk razu (n=rasmus@tln-kontor.norby.ee) |
| 22:27.51 | Nitrus^ | and the internal phones are analog on the FXS |
| 22:28.11 | jhiver | why, why oh why! |
| 22:28.11 | pigpen | jhiver, I have some extra room on this rope.... |
| 22:28.17 | jhiver | I try to call asterisk B from asterisk A through SER, and when I do that I have no audio, but calling directly without using SER, it works fine |
| 22:28.22 | Nitrus^ | so how do i stop this channel bridging delay? |
| 22:28.23 | jhiver | pigpen, no I'm not into bondage, thx :) |
| 22:28.35 | pigpen | sheesh! |
| 22:28.41 | pigpen | Suicidal...yes... |
| 22:28.47 | arkonadev | hey anyone got some time to help out a newbie? |
| 22:28.50 | kb3ien | progress "Shared object "libncurses.so.5" not found |
| 22:28.52 | kb3ien | at least i have a binary again! |
| 22:28.55 | ellisgl | arkonadev: depends on what you need |
| 22:43.51 | arkonadev | im trying to do just a simple originate action to an external phone number |
| 22:43.51 | arkonadev | and nohting seems to work |
| 22:43.51 | Juggie | then your doing it wrong :) |
| 22:44.03 | arkonadev | i have to use the zap/1-1 channel to call external number right? |
| 22:44.03 | Juggie | no, dont do that. |
| 22:44.07 | *** join/#asterisk giengus (n=giengus@71-37-118-187.slkc.qwest.net) |
| 22:44.07 | Juggie | use the local channel |
| 22:44.07 | arkonadev | a sip channel? |
| 22:44.13 | Juggie | no, the local channel driver, so you can throw the call into the dialplan to parse the number. |
| 22:44.24 | arkonadev | can you give me an idea where i can find that info? |
| 22:44.26 | Juggie | http://www.voip-info.org/wiki/index.php?page=Asterisk+Manager+API+Action+Originate |
| 22:44.34 | Juggie | but instead of Channel: Zap/g1/number |
| 22:44.39 | Juggie | do |
| 22:44.42 | Juggie | Local/number/n |
| 22:44.42 | Juggie | where number=5551234 or whatever |
| 22:44.49 | Juggie | oh also you'll need the context |
| 22:44.55 | Juggie | so Local/number@somecontext/n |
| 22:44.55 | arkonadev | so literraly the word "local"? |
| 22:44.59 | Juggie | then, within your extensions.conf youl'll have a [somecontext] |
| 22:45.12 | Juggie | yes. |
| 22:45.23 | Nitrus^ | if i want all my zap channels to ring at once on an incoming call i just do Zap/1&Zap/2&...Zap/n |
| 22:45.23 | Nitrus^ | correct? |
| 22:45.27 | arkonadev | k ill try that out |
| 22:45.30 | arkonadev | thanks |
| 22:45.30 | Juggie | and within there you will pattern match the number dialed |
| 22:45.37 | Juggie | and then do a Dial(Zap/.....) on that number, just like if a sip phone dialed it |
| 22:45.40 | Mandrak3 | I'm having problem with Congestion...... who can help me? |
| 22:45.48 | Juggie | except its the local channel. |
| 22:45.48 | justinu|laptop | juggiewhat's the /n mean? |
| 22:45.51 | Juggie | http://www.voip-info.org/wiki/index.php?page=Asterisk+local+channels |
| 22:45.58 | arkonadev | so how would i get it so extension 700 will dial 555555555 |
| 22:46.03 | Juggie | justin, theres an exlpanation there. |
| 22:46.08 | Juggie | *explanation |
| 22:46.12 | Nitrus^ | can echo cancelling cause bridging delays? |
| 22:46.45 | Juggie | if you do an action originate directally on your zap devices (which is bad) you can end up in a situation where you have multiple routes |
| 22:46.48 | Juggie | eg, an internal pri and a external pstn pri |
| 22:46.51 | Cresl1n | Nitrus^: if you're doing software echo can and you have a card that can natively bridge on the card, it does |
| 22:46.54 | Dovid | i have a sangoma card. will installing festival ruin my configs ? |
| 22:46.54 | Juggie | and yo have to make that decision on which route to use in your action originate script rather then let * do it |
| 22:46.57 | arkonadev | so what would be the easiest way to make a certain extension call a certain external number |
| 22:47.01 | Juggie | which is obviously bad |
| 22:47.07 | Juggie | hence the reason to use the local channel. |
| 22:47.14 | Juggie | exten=> 400,1,Dial(Zap/g1/5551234) |
| 22:47.18 | arkonadev | but thats bad because that would be doing an originate on the zap device? |
| 22:47.18 | *** join/#asterisk pdavid (n=chatzill@adsl-072-151-167-100.sip.mob.bellsouth.net) |
| 22:47.27 | Juggie | your confusing two things. |
| 22:47.31 | Nitrus^ | cresl1n: all my channels are zap channels on a channel bank, i have echocancel, echocancelwhenbridged, and echotraining turned on |
| 22:47.35 | arkonadev | lol |
| 22:47.40 | arkonadev | sorry |
| 22:47.40 | Juggie | what are you attempting to do. |
| 22:47.46 | arkonadev | i trying to use the managare API to make a certain extension call a certain external number |
| 22:47.51 | *** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com) |
| 22:47.55 | Juggie | so your trying to do click to talk essentially. |
| 22:48.00 | *** join/#asterisk X-Gen (n=X-Gen@dsl-145-209-108.telkomadsl.co.za) |
| 22:48.09 | arkonadev | exactly |
| 22:48.12 | Juggie | hold on. |
| 22:48.15 | arkonadev | i can do it with internal phones fine but when i start dialing out it goes all bad |
| 22:49.03 | Juggie | arkonadev,i'm going to show you a really ugly example |
| 22:49.05 | Juggie | but none the less, you'll get the point |
| 22:58.02 | arkonadev | k |
| 22:58.02 | arkonadev | sounds good |
| 22:58.03 | Juggie | heres the horrible php code... |
| 22:58.04 | Juggie | http://pastebin.ca/146735 |
| 22:58.04 | denon | oh man that's awful! |
| 22:58.07 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
| 22:58.07 | Juggie | which as you can see you just do file.php?num1=5551234&num2=5551111 |
| 22:58.08 | Juggie | and it connects the two. |
| 22:58.10 | denon | Juggie: isnt that .. um .. really insecure? |
| 22:58.14 | aydiosmio | lol haxd |
| 22:58.14 | Juggie | yes. |
| 22:58.16 | Juggie | its just a demo i did for someone |
| 22:58.16 | aydiosmio | file.php?num1=5551234&num2=5551111 |
| 22:58.17 | denon | Juggie: why not make it an extension? |
| 22:58.18 | Juggie | which i wrote in like 5minutes |
| 22:58.21 | denon | Juggie: is this live somewhere? |
| 22:58.22 | Juggie | hell no. |
| 22:58.22 | denon | if so, what url? I'd like to uh .. test it |
| 22:58.23 | denon | :) |
| 22:58.24 | Juggie | its just a demo i did to prove a concept |
| 22:58.25 | Juggie | anyways |
| 22:58.25 | Juggie | i'll get you the contexts |
| 22:58.26 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.216.2) |
| 22:58.27 | denon | ah, bummer .. I wanted to call a friend of mine at 1-900-4-pbx-fun |
| 22:58.29 | arkonadev | so whats up the cttnumber variable its used the dialplan? |
| 22:58.29 | arkonadev | *in |
| 22:58.32 | Juggie | one sec. |
| 22:58.33 | Juggie | http://pastebin.ca/146744 |
| 22:58.34 | Juggie | there. |
| 22:58.35 | aydiosmio | cctnumber? |
| 22:58.36 | arkonadev | cttnum |
| 22:58.36 | arkonadev | whoops |
| 22:58.37 | arkonadev | it what i meant |
| 22:58.37 | arkonadev | is |
| 22:58.38 | arkonadev | jeeze having a hard time typing today |
| 22:58.38 | aydiosmio | oh |
| 22:58.38 | Juggie | thats it... my [internals] is confusing because i have two local exchanges |
| 22:58.43 | Juggie | so i do a db lookup to see if a number is local or not. |
| 22:58.44 | *** part/#asterisk mtaht4 (n=m@c-71-198-23-124.hsd1.ca.comcast.net) |
| 22:58.44 | Juggie | the php is nasty, but the 2 astrisk context's are fine. |
| 22:58.45 | arkonadev | where did you put the contexts? |
| 22:58.49 | arkonadev | oh nvm |
| 22:58.51 | Juggie | i have them in their own files |
| 22:58.56 | Juggie | internals.conf & clicktotalk.conf |
| 22:58.57 | arkonadev | i didnt see your second link |
| 22:58.58 | Juggie | and then include those into extensions.conf |
| 22:59.00 | Juggie | but thats up to you |
| 22:59.02 | pdavid | what is usually considered the best choice codec for bandwidth concerns? (free) |
| 22:59.02 | denon | gsm or speex |
| 22:59.05 | Juggie | gsm |
| 22:59.15 | denon | gsm's the most compatible |
| 22:59.17 | denon | speex does well on weird wireless and such |
| 22:59.34 | pdavid | yeah, im using gsm now |
| 22:59.36 | denon | speex is also more tweakable |
| 22:59.39 | pdavid | how is ilbc? |
| 22:59.42 | arkonadev | thanks |
| 22:59.46 | Assid | okay anyone suggest a good provider for incoming lines.. whom i can port a number to ? |
| 22:59.46 | Cresl1n | pdavid: file |
| 22:59.47 | Juggie | *AVOID AVOID* |
| 22:59.50 | pdavid | weird wireless? |
| 22:59.56 | denon | well that depends, do you need to hear any detail? |
| 22:59.56 | denon | if you dont mind just hearing the noise of another person's voice |
| 22:59.58 | denon | ilbc is fine |
| 23:00.04 | pdavid | gotcha |
| 23:00.19 | pdavid | gsm would be a better choice, then |
| 23:00.21 | file | wha? |
| 23:00.21 | Juggie | yes |
| 23:00.24 | arkonadev | so juggie there isnt really anyway to do it without modifying the dialplan huh |
| 23:00.26 | pdavid | my voip doesnt support speex |
| 23:00.26 | *** join/#asterisk svemuri1 (n=svemuri1@nat.ftc.bz) |
| 23:00.33 | Assid | hey file, which providers you use? |
| 23:00.39 | denon | pdavid: but in many cases, its really worth the dough for g729 |
| 23:00.46 | Juggie | arkonadev, you can do it without touching the dialplan yes. |
| 23:00.49 | Juggie | but the only way to do it right is to modify the dialplan |
| 23:00.49 | file | none, I use misery! |
| 23:00.52 | [TK]D-Fender | Juggie : Ottawa area? |
| 23:01.11 | Juggie | yes. |
| 23:14.29 | pdavid | i wish i could test it first, but i don't think voicepulse supports g729... |
| 23:14.30 | file | :D |
| 23:14.34 | [TK]D-Fender | Juggie : just mixed your host-name in with the area-codes listing in your dialplan :) |
| 23:14.36 | Juggie | ottawa & gatineau :) |
| 23:14.44 | Juggie | but hostname? |
| 23:14.53 | [TK]D-Fender | Juggie : Figured you were bordering :) |
| 23:15.01 | arkonadev | when you say $num1@internals it sets num1 to s correct? |
| 23:15.04 | [TK]D-Fender | Juggie : Rogers tells me which side of the border you're on... |
| 23:15.05 | svemuri1 | Any one tried compiling 1.2.11 released today? I am getting compile errors |
| 23:15.13 | Juggie | tk, no it doesnt |
| 23:15.21 | *** join/#asterisk r_evolution (n=no@208.251.203.208) |
| 23:15.28 | Juggie | i'm using a proxy server in newfoundland |
| 23:15.32 | r_evolution | you know |
| 23:15.35 | Juggie | so you will be getting an ip from the rock :) |
| 23:15.44 | [TK]D-Fender | Juggie : Never seen them on my side... Videotron all the way for cable.... |
| 23:15.44 | r_evolution | someone was REALLY SUPER unoriginal |
| 23:15.44 | r_evolution | when they named newfoundland |
| 23:15.44 | puzzled | svemuri1: I had no issues |
| 23:15.44 | [TK]D-Fender | Juggie : Ok... so it COULD be misleading! |
| 23:15.44 | r_evolution | it's like... hmmm whats the MOST unoriginal name we can came up with for this New Found Land |
| 23:15.44 | Juggie | arkonadev, no |
| 23:15.45 | svemuri1 | chan_zap.c:9025: error: structure has no member named `call' |
| 23:15.45 | Juggie | $num1 could be 5551234 and $num2 coudl be 5554321 |
| 23:15.45 | [TK]D-Fender | Juggie : At work I flag as TO anyways.... |
| 23:16.04 | [TK]D-Fender | r_evolution : Springfield was already taken ;) |
| 23:16.27 | Juggie | i believe you might want to change Extension: $num1 to Extension: s depending on if you have autofallback on or off |
| 23:16.32 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 23:16.33 | Juggie | i forget which |
| 23:16.33 | Juggie | but yah thats kinda wrong |
| 23:16.33 | Juggie | or you could just change the [clicktotalk] from exten=> s to exten => _.X |
| 23:16.33 | Juggie | your pick. |
| 23:16.33 | Qwell | _X. |
| 23:23.15 | Juggie | nice Qwell, you broke irc. |
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| 23:32.58 | Nivex | sweet mother of netsplits! |
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| 23:33.33 | ellisgl | oh boy net splits. |
| 23:33.34 | Dovid | hi all |
| 23:33.34 | shmaltz | hi Dovid |
| 23:33.34 | *** part/#asterisk QbY (n=Kelvin@cm-64-221-172-66.dhcp.southerncoastalcable.net) |
| 23:33.34 | Dovid | i have a zangoma a200. i just downloaded the new verison of asterisk and ran wanpipe again and for some reason now the cFXO's dont work on the card |
| 23:33.34 | Dovid | shmaltz: can i PM ? |
| 23:33.34 | shmaltz | sure |
| 23:33.46 | *** join/#asterisk CoffeeIV_ (n=CoffeeIV@www.airlinksystems.com) |
| 23:33.46 | [TK]D-Fender | b00m |
| 23:33.46 | CoffeeIV_ | can you do nested GoSubs in asterisk dialplan ? |
| 23:33.47 | Assid | anyone know any good places for hosting incoming DID's? |
| 23:33.47 | shmaltz | CoffeIV_, why not? |
| 23:33.47 | shmaltz | Assid, you mean buy? |
| 23:33.55 | Assid | well.. i need to port |
| 23:34.00 | shmaltz | Assid, location? |
| 23:34.02 | ESCulapio__ | somebody knows if this card is supported by asterisk Intel Dialogic DI/0408-LS |
| 23:34.02 | CoffeeIV_ | because I'm trying it and it seems to be returning to the first call, not the inner ones |
| 23:34.03 | Assid | 212 - ny |
| 23:34.06 | shmaltz | ESCulapio__, might be in ABE |
| 23:34.06 | shmaltz | Assid, anybody will do it |
| 23:34.07 | shmaltz | Voange, myphonecompany.com |
| 23:34.10 | shido6 | any max tnt & sip users? |
| 23:34.13 | Assid | shmaltz: need iax/sip based |
| 23:34.16 | Assid | vonage. is a POS |
| 23:34.16 | shmaltz | Assid, read above, but they don't give IAX |
| 23:34.17 | shmaltz | Assid, whats POS? |
| 23:34.21 | Assid | piece of poo |
| 23:34.22 | designdream | anyone in here familiar with broadvoice? |
| 23:34.22 | obiwanmikenolte | Haha. Acronyms |
| 23:34.27 | shmaltz | Assid, why? they are still the best quality ones |
| 23:34.28 | *** join/#asterisk RoyK (n=roy@ti211310a080-4327.bb.online.no) |
| 23:34.32 | [TK]D-Fender | Will everyone please stop trying to use God-damned (and he has indeed damned them!) Dialogic cards with * and use something COMPATIBLE!!!!! |
| 23:34.39 | Nivex | s/he/He/ |
| 23:34.43 | CoffeeIV_ | any idea how I could cofirm/deny that Gosub() is not nestable, other than looking at the C code ? voip-info and show application don't seem to say one way or the other |
| 23:35.01 | }btorch{ | is there a way to make the voicemail directory 644 mode ? |
| 23:35.20 | *** join/#asterisk ruskie (n=ruskie@sourcemage/mage/ruskie) |
| 23:35.28 | }btorch{ | I mean the mailbox folders that are created by asterisk |
| 23:35.30 | [TK]D-Fender | Nivex : My God doesn't believe in capitalization, punctuation, or YOU! ;P |
| 23:35.39 | ellisgl | hmm - asterisk is not showing when I'm dialing or anything.. |
| 23:36.31 | *** join/#asterisk cekc (n=cekc@rrcs-24-199-36-210.west.biz.rr.com) |
| 23:36.32 | [TK]D-Fender | ellisgl : Maybe you can validate that with some dialplan pastebins and maybe so debug info where appropriate... |
| 23:36.33 | cekc | so... looks like I'm getting a T1 line. My company will save almost $400 a month to switch from analog to digital |
| 23:36.33 | shmaltz | CoffeeIV__, what are you trying to do? |
| 23:36.33 | [TK]D-Fender | cekc : scary that you'd save so much.... |
| 23:36.33 | shmaltz | CoffeeIV__, it should work whatever it is, the return should go to the last gotosub |
| 23:36.33 | ellisgl | [TK]D-Fender: Well I can dial where ever - but doing asterisk -vvvvr |
| 23:36.33 | cekc | well, AT&T is raping us in long distance costs at the moment |
| 23:36.39 | shmaltz | CoffeeIV__, otherwise you can use Goto(lable) |
| 23:36.41 | Waverly360 | I believe I have a presence related issue with my PBX. It involves this error message: Incoming call: Got SIP response 500 "Internal Server Error" back from 192.168.200.146 |
| 23:36.43 | Waverly360 | Anyone know what causes that? |
| 23:40.40 | cekc | but this way we'll have internet and phone combined into one service |
| 23:40.40 | [TK]D-Fender | Waverly360 : Just a Polycom formatting error... don't worry about it. |
| 23:40.41 | ellisgl | nothing pops up |
| 23:40.41 | ellisgl | [TK]D-Fender: I will pastebin my stuff thou |
| 23:40.41 | Waverly360 | But it's irritating the hell out of me...there's no way to get rid of it? |
| 23:40.41 | [TK]D-Fender | ellisgl : What kind of phone? |
| 23:40.45 | [TK]D-Fender | Waverly360 : Not that I've heard of yet. Not sure the full deatails on why, but from experience its harmless... just annoying |
| 23:40.46 | *** join/#asterisk Tak (n=Tak@66.230.25.38) |
| 23:40.46 | Waverly360 | [TK]D-Fender: Annoying it is... Just makes it difficult to watch the CLI for errors and such when that keeps cluttering everything up. |
| 23:40.46 | Waverly360 | [TK]D-Fender: I suppose it's my OCDish twitch that I want my CLI to be as clean as possible. |
| 23:40.46 | mountainm2k | Any ABE "experts" ? I have version A, and since version B just came out, I'd like to, you know, upgrade... |
| 23:40.47 | ellisgl | [TK]D-Fender: |
| 23:40.47 | [TK]D-Fender | Waverly360 : Know the feeling.... just let go... |
| 23:40.47 | ESCulapio__ | shmaltz, that it is ABE? |
| 23:40.47 | *** part/#asterisk Tak (n=Tak@66.230.25.38) |
| 23:40.47 | [TK]D-Fender | mountainm2k : Considered asking Digium Directly? |
| 23:40.47 | CoffeeIV_ | shmaltz: are you saying that you have actually done nested GoSubs, or that you just can't believe they would really write one that wasn't nestable ? |
| 23:40.50 | shmaltz | ~ABE |
| 23:40.53 | [TK]D-Fender | mountainm2k : Most ABE users don't come to places like here since they paid for "real?" support. |
| 23:40.53 | Waverly360 | [TK]D-Fender: Thanks |
| 23:40.53 | shmaltz | CoffeeIV_, go and test it, there is no point in reading the C code, when testing will take all of 2 minutes |
| 23:40.53 | file | [TK]D-Fender: some do, strangely enough |
| 23:40.53 | mountainm2k | <PROTECTED> |
| 23:40.53 | [TK]D-Fender | file : True, and in the same category of freaks of nature we find the duck-billed platypus.... |
| 23:40.54 | *** join/#asterisk ruskie (n=ruskie@sourcemage/mage/ruskie) |
| 23:40.54 | file | if you do see someone who is an ABE, do send them to support |
| 23:40.54 | file | er ABE user |
| 23:40.54 | eKo1 | [TK]D-Fender: Leave the echinoderms alone. |
| 23:40.54 | ellisgl | http://pastebin.ca/146783 |
| 23:40.54 | mountainm2k | [TK]D-Fender: They're saying, basically, that I need to wipe/re-install from scratch... |
| 23:40.54 | mountainm2k | But they did suggest I try their distro, "Poundkey Linux" |
| 23:40.54 | ellisgl | [TK]D-Fender: SIP and T1 to a channel bank |
| 23:40.54 | mountainm2k | (great thanks) |
| 23:40.54 | [TK]D-Fender | ellisgl : LOL AMP! |
| 23:40.54 | [TK]D-Fender | ellisgl : You are in the wrong place my friend...... |
| 23:40.54 | eKo1 | err, i mean monotremes |
| 23:40.55 | [TK]D-Fender | ellisgl : read the channel topic... |
| 23:41.08 | *** join/#asterisk pigpen (n=mark@fw.seamans.cc) |
| 23:41.11 | *** join/#asterisk r0d3nt (n=RatMan@foster.stonedcoder.org) |
| 23:41.13 | ellisgl | let me join there |
| 23:41.14 | ellisgl | but I had all that information before.. |
| 23:41.23 | [TK]D-Fender | mountainm2k : Well * runs on any sane distro I've heard of. Some with a bit more work than others of course. |
| 23:41.27 | [TK]D-Fender | mountainm2k : Shouldn't need to redo your whole DISTRO for it thoguh... |
| 23:41.27 | mountainm2k | I prefer CentOS... I had to trick ABE into accepting CentOS as being equal to RHEL |
| 23:41.30 | *** join/#asterisk DarKnesS_WolF (n=wolf@196.218.41.132) |
| 23:41.31 | mountainm2k | perhapps I can trick it into accepting CentOS 3 instead of RHEL 4... |
| 23:41.33 | [TK]D-Fender | ellisgl : AMP's dialplan and its supporting (LOL!!!) macro's are a hideous mess that only the brave, paid, or stupid would waste time trying to debug. Keeping in mind it probably works and its either your phone or your AMP setup that is to blame |
| 23:41.43 | [TK]D-Fender | mountainm2k : i've taken a liking to it as well, and am learning the RH way of doing things so as to grow out of Slackware a little |
| 23:41.51 | ESCulapio__ | that it is ABE? |
| 23:41.56 | mountainm2k | <PROTECTED> |
| 23:41.59 | mountainm2k | <PROTECTED> |
| 23:42.01 | shmaltz | CoffeeIV_, here: |
| 23:42.03 | shmaltz | http://pastebin.ca/146786 |
| 23:42.13 | pigpen | Ok..next issue... voicemail update to a database: |
| 23:42.13 | shmaltz | pigpen, mind to explain? |
| 23:42.14 | pigpen | I have "externpass=/usr/sbin/1.sh" in my voicemail.conf |
| 23:42.15 | pigpen | 1.sh contains "echo -e "$1 $2 $3 $4" > /tmp/biteme.txt" |
| 23:42.18 | pigpen | I can su to asterisk and run the script just fine...but when I change a pass in *, it seems as if * never attempts to execute it...and of course, the txt file is not created. Ideas? |
| 23:42.18 | shmaltz | pigpen, what user is asterisk running as? |
| 23:42.19 | pigpen | I am pretty lost on this one...after it can do a simple script...then I will stick in a script that will actually update the database... |
| 23:42.20 | pigpen | User: asterisk |
| 23:56.14 | [TK]D-Fender | pigpen : You running * as root? checked your permissions? |
| 23:56.19 | dserban | hmm anyone else have problems with their 2400p cards not hearing anything incoming? |
| 23:56.19 | [TK]D-Fender | pigpen : su to your * user and try |
| 23:56.34 | pigpen | no..* is running as asterisk.... |
| 23:56.39 | pigpen | yes...(I just confirmed) that if I su to asterisk..it creates the file fine.... |
| 23:56.44 | pigpen | I was hoping for a file permission honestly...it is easier... |
| 23:56.47 | pigpen | when I jump into the asterisk cli...and change a pass...I do not see it trying to execute it... |
| 23:57.00 | CoffeeIV_ | thanks schmaltz -- obviously I have some other problem in my dialplan, looking for it now |
| 23:57.00 | pigpen | Asterisk PID: asterisk 4975 0.0 1.2 25560 12300 ? Ssl Aug17 8:10 /usr/sbin/asterisk -U asterisk |
| 23:57.02 | shmaltz | CoffeeIV_, you couldn't test that for yourself? |
| 23:57.12 | pigpen | The way I understand it, that once the option is set, permissions are fine...it should just "work".... |
| 23:57.29 | shmaltz | pigpen, so you telling me that everything works even as user asterisk, but not when you change the password? is that correct? |
| 23:57.31 | pigpen | Correct. |
| 23:57.34 | pigpen | I have tried this on a test box with * ver. 1.2.7.1 |
| 23:57.37 | dserban | :( Am I the only asterisk user to ever have this prob? google isn't helping me at all. |
| 23:57.45 | pigpen | dserban, sorry..I have only fxs's on mine.... |
| 23:57.50 | shmaltz | dserban, whats the problem? |
| 23:57.53 | shmaltz | pigpen, did you reload asterisk? |
| 23:57.54 | syzygyBSD | dserban: it has nothing to do with what kind of card you have |
| 23:57.54 | dserban | pigpen: that's cool :) but I'm frustrated as all heck. |
| 23:57.57 | diablopico | Hello ... is there a version of asterisk-oh323 that is compatable with asterisk v 1.2.9.1 ? |
| 23:58.00 | dserban | shmaltz: I get no incoming tones or voice on the tdm2400p |
| 23:58.02 | dserban | syzygyBSD: que? What does it have to do with? my channel setup? |
| 23:58.03 | *** join/#asterisk Givemelove2k (n=bob@208.57.229.162) |
| 23:58.06 | shmaltz | dserban, asterisk is started and shows that the tdm2400P is configured right but no sound? |
| 23:58.07 | Givemelove2k | Hi there |
| 23:58.07 | syzygyBSD | dserban: you see a call come in right |
| 23:58.13 | Givemelove2k | I do have an issue with Asterisk |
| 23:58.17 | dserban | yes, I can dial into it and hear my menus... but it can't hear anything (using ztmonitor) |
| 23:58.23 | syzygyBSD | did you do an answer() |
| 23:58.23 | Givemelove2k | Let's say I pick up the phone onto line #1 |
| 23:58.26 | Givemelove2k | but I don't pass any call |
| 23:58.30 | dserban | syzygyBSD: yes I can see it come in |
| 23:58.37 | Givemelove2k | when I try to call #1 from #2 nothing happens |
| 23:58.48 | dserban | I set it to call an internal extension via sip |
| 23:58.56 | syzygyBSD | dserban: that isn't what i asked. in your dialplan do you have exten => _X.,1,Answer() |
| 23:59.04 | dserban | the caller (from outside can hear me) and I can't hear anything |
| 23:59.18 | syzygyBSD | or something with Answer() |
| 23:59.23 | dserban | yes |
| 23:59.28 | dserban | first thing |
| 23:59.28 | pigpen | shmaltz, yes...I have reloaded and restarted asterisk |
| 23:59.31 | *** part/#asterisk Samoied (n=Samoied@ip47092.static.poa.virtua.com.br) |
| 23:59.31 | dserban | exten => s,1,Answer() |
| 23:59.36 | syzygyBSD | dserban: put a wait(1) before the answer |
| 23:59.39 | syzygyBSD | lol.. is everything in extension s? |
| 23:59.46 | syzygyBSD | pastebin your dialplan |