| 00:00.33 | JimVanM | [TK]D-Fender: sorry, just figured out what PB means |
| 00:00.40 | JimVanM | [TK]D-Fender: http://pastebin.com/mfd069e2 |
| 00:00.54 | [TK]D-Fender | JimVanM: Just fixed it anyway... was only 1 line that really mattered |
| 00:01.39 | JimVanM | [TK]D-Fender: so what was I doing wrong? |
| 00:02.54 | [TK]D-Fender | JimVanM: The function returns nothing apparently when the conf doesn't exist therefor breaking your expression. You need something on each side of the operator |
| 00:03.34 | [TK]D-Fender | hrm |
| 00:03.41 | [TK]D-Fender | maybe I didn't read fully into that... |
| 00:04.15 | JimVanM | [TK]D-Fender: I'm thinking it'd be nice if the function returned *something* if the conference doesn't exist |
| 00:05.05 | JimVanM | [TK]D-Fender: of the top of my head, I can't see why '0' wouldn't work, certainly for the 'parties' keyword |
| 00:05.41 | [TK]D-Fender | JimVanM: PB the failed attempt |
| 00:07.21 | JimVanM | [TK]D-Fender: http://pastebin.com/m499bcba8 |
| 00:08.32 | JimVanM | [TK]D-Fender: my sense is that since the conference doesn't exist yet, the function can't find it, and thus complains |
| 00:09.09 | [TK]D-Fender | JimVanM: I think the function is returning a literal statement, which is the problem. Assign it to a var before calling your GotoIf to see |
| 00:10.42 | *** join/#asterisk blkry (n=blkry@64.147.222.130) |
| 00:12.09 | JimVanM | [TK]D-Fender: yep, that's the problem -- http://pastebin.com/m1df569e7 |
| 00:12.24 | JimVanM | [TK]D-Fender: a wee little bug, methinks? |
| 00:13.56 | *** join/#asterisk s14ck (n=s14ck@190.77.78.180) |
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| 00:17.14 | [TK]D-Fender | JimVanM: If not "tragic conceptual flaw" |
| 00:17.53 | JimVanM | [TK]D-Fender: well, I wouldn't call it 'tragic'. The dialplan keeps on going, so it's just a little verbiage tweak, really. |
| 00:18.15 | [TK]D-Fender | JimVanM: Where's your sense of drama?! |
| 00:19.04 | JimVanM | [TK]D-Fender: See July 24th -- http://www.baddogtheatre.com/modules/agendax/index.php?op=view&id=13 |
| 00:19.07 | WindowsUser | why is a Wait() needed before I Background() or Playback() something remotely? is there a DynamicWait that I can use that waits until the audio connection is ready? |
| 00:20.32 | JimVanM | WindowsUser:Do you have an Answer() in there? That can help |
| 00:22.33 | WindowsUser | yea I got an answer in there |
| 00:23.13 | [TK]D-Fender | WindowsUser: No. |
| 00:24.14 | *** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim) |
| 00:24.44 | [TK]D-Fender | JimVanM: With some more work you may even be able to charge admission ;) |
| 00:25.23 | JimVanM | [TK]D-Fender: I wouldn't pay to see me! |
| 00:26.17 | JimVanM | we need a func_dramatic_pause |
| 00:27.02 | [TK]D-Fender | JimVanM: It was deprecated in lieu of res_uncomfortable_silence |
| 00:28.08 | JimVanM | [TK]D-Fender: that's unfortunate. The caller is much more likely to hang up with an uncomfortable silence, but if we can give 'em a dramatic pause, they'll just be aching to know what happens next! |
| 00:29.02 | [TK]D-Fender | JimVanM: Ok, there is the third-party app_suspense.... |
| 00:31.17 | JimVanM | [TK]D-Fender: hmmm. that could work for some folks, but there's something creepy about it. Might scare some away. Mess up the stats. |
| 00:31.47 | JimVanM | how about: app_omg_this_is_gonna_be_awesome |
| 00:32.32 | [TK]D-Fender | JimVanM: Planned for * 1.vapor.ware |
| 00:33.10 | JimVanM | [TK]D-Fender: oh yeah! that's the one where *everything* is gonna be in there. it's gonna be great! |
| 00:53.46 | *** join/#asterisk sebbl (n=name@HSI-KBW-078-043-155-130.hsi4.kabel-badenwuerttemberg.de) |
| 00:54.03 | sebbl | i search a wakeup dialplan |
| 00:58.07 | sebbl | for asterisk 1.6.1.1 |
| 00:58.45 | *** join/#asterisk eppigy (n=Dave@216-139-245-58.aus.us.siteprotect.com) |
| 00:58.48 | eppigy | hello |
| 00:58.51 | eppigy | I am dave |
| 00:59.04 | [TK]D-Fender | eppigy: We saw that :) |
| 00:59.12 | eppigy | yesh |
| 00:59.28 | eppigy | how was your 4th of july [TK]D-Fender |
| 00:59.36 | eppigy | did you detonate obsolete hardware |
| 00:59.46 | [TK]D-Fender | <- Canuckian |
| 00:59.52 | eppigy | or possibly discharge small arms in to the air |
| 00:59.58 | eppigy | ala hood rat |
| 01:00.05 | eppigy | oh canuckistan native |
| 01:00.09 | eppigy | well that is ok |
| 01:00.31 | eppigy | know you are canadian I now think there may be some hope for the canuck geen pool |
| 01:00.31 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-91e0444fe8022912) |
| 01:00.37 | eppigy | please start sowing wild oats |
| 01:00.46 | eppigy | *knowing |
| 01:00.51 | eppigy | *gene |
| 01:01.15 | [TK]D-Fender | For instance I have that totally awesome head start on spelling ;) |
| 01:01.27 | jaytee | lol |
| 01:01.52 | eppigy | indeed |
| 01:05.06 | *** join/#asterisk carrar (i=tim@osburn.com) |
| 01:09.13 | sebbl | i have found this: http://das-asterisk-buch.de/1.0/call-file-weckruf.html but it dont work |
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| 01:49.18 | WindowsUser | sebbl: so uh what doesn't work? |
| 01:49.29 | AlmightyOatmeal | eyeballs WindowsUser |
| 01:49.49 | AlmightyOatmeal | at least your nick isn't AOLuser :P |
| 01:50.47 | WindowsUser | if i did that people would question the lack of an aol.com hostname |
| 01:50.57 | sebbl | the asterisk hang up |
| 01:50.58 | AlmightyOatmeal | good point |
| 01:50.59 | sebbl | [Jul 6 03:49:10] ERROR[4436]: sccp_actions.c:1700 sccp_handle_open_receive_channel_ack: SEP0018181587DF: No channel with this PassThruId! |
| 01:51.11 | AlmightyOatmeal | WindowsUser: i admire the size of your testicles for having a nick like that |
| 01:51.31 | AlmightyOatmeal | for *willfully* |
| 01:52.05 | WindowsUser | sebbl: so the call file doesn't work? or it fails during the commands to make it? |
| 01:53.21 | sebbl | i dont have a call file... only the dialplan |
| 01:53.32 | WindowsUser | sebbl: that thingy makes a call file |
| 01:53.39 | AlmightyOatmeal | WindowsUser: please don't tell me you're a grrl or i might feel a little embarassed for that last comment |
| 01:53.42 | WindowsUser | echo -e "stuffs " > callfile |
| 01:53.52 | WindowsUser | haha no im a duder |
| 01:53.58 | AlmightyOatmeal | hah k |
| 01:55.10 | eppigy | TRABAJO |
| 01:55.51 | sebbl | I call on the extension of the asterisk and submit immediately to |
| 01:55.53 | rob0 | I have a feeling that this channel has a lot of Windows users in it, so it's probably not as hostile as some others might be. But, it seems odd that someone would want to draw attention to themselves specifically as a Windows user. |
| 01:56.16 | eppigy | IM A PC |
| 01:56.24 | rob0 | OTOH, it's odd that someone would call himself rob0 ... |
| 01:56.27 | eppigy | AND I AM THIS MANY |
| 01:56.52 | rob0 | This little eppigy went to market. This little eppigy stayed home ... |
| 01:56.56 | WindowsUser | sebbl: you're kinda screwed if you cant realize that what you pasted makes a callfile :) |
| 01:57.20 | eppigy | :D |
| 01:58.50 | *** join/#asterisk dshap (n=dshap@216-165-39-50.DYNAPOOL.NYU.EDU) |
| 01:58.54 | sebbl | ok ... but the asterisk would have to wait times yes, at least till I tell him what time he should create a file |
| 01:59.14 | dshap | hey does anyone here know if i should be able to use the MYSQL() application to select multiple values from the DB in a single query? |
| 01:59.26 | dshap | or if i want to get 3 fields |
| 01:59.31 | dshap | do i have to do 3 seperate queries |
| 01:59.38 | WindowsUser | sebbl: it makes a callfile and then sets the modified time with that touch -t command |
| 01:59.44 | WindowsUser | sebbl: do you speak german? |
| 01:59.54 | sebbl | ja |
| 01:59.58 | eppigy | DA |
| 02:00.07 | eppigy | UND KEINER EIR |
| 02:00.26 | WindowsUser | thats good |
| 02:00.52 | eppigy | How do you say "Please butter the platypus" in german? |
| 02:01.04 | sebbl | I can now write German? :) |
| 02:01.13 | AlmightyOatmeal | rofl eppigy |
| 02:01.29 | rob0 | mmmm some nice Deutsche buttered platypus! |
| 02:02.14 | *** join/#asterisk mrichman (n=mrichman@adsl-067-035-107-190.sip.bct.bellsouth.net) |
| 02:02.31 | mrichman | I've got a Cisco IP Phone 7940, which I now have at home and have factory reset. Anything cool I can do with it using Asterisk? |
| 02:02.46 | AlmightyOatmeal | sebbl: pozhalujjsta namazh'te maslom platypus in russian :P |
| 02:02.52 | AlmightyOatmeal | mrichman: yeah, call people |
| 02:03.01 | mrichman | AlmightyOatmeal: like free VoIP? |
| 02:03.23 | AlmightyOatmeal | AlmightyOatmeal: same with any other phone that can do sip or iax.. depends on your provider and what your phone can do |
| 02:03.25 | sebbl | mrichman yes |
| 02:03.45 | mrichman | sebbl: sweet...I definitely have to d/l it now ;) |
| 02:04.13 | AlmightyOatmeal | mrichman: you can use that particular phone with sip without having to use asterisk as well, depends on how you configure the phone |
| 02:04.20 | dshap | anyone here have expierience with the MYSQL app? |
| 02:04.25 | WindowsUser | not i |
| 02:04.30 | rob0 | I got a wrong number on my ipkall, someone local in Seattle, and she started speaking Russian to me! "Ya ne goboriu po-russkie," I said. She was amazed that I knew even that much. :) |
| 02:04.38 | sebbl | mrichman you can user asterisk 1.6.1.1 and chan_sccp V3.1 |
| 02:04.39 | mrichman | AlmightyOatmeal: yes i've seen that, i think its SCCP right now...i would have to find the SIP firmware somewhere |
| 02:05.02 | sebbl | WindowsUserwhat name should the call file have? |
| 02:05.10 | sebbl | WindowsUser what name should the call file have? |
| 02:05.16 | florz | eppigy: "Bitte bestreichen Sie das Schnabeltier mit Butter." - anything else we can help you with? =:-) |
| 02:05.31 | AlmightyOatmeal | mrichman: not only will you need the sip firmware, but you will need a tftp server for it, the phone config files, and make a special dialplan for that phone to work with asterisk... i did the cisco 7911G ip phone and i realized, i dont like cisco IP phones |
| 02:05.39 | WindowsUser | sebbl: look in /tmp and /var/spool/asterisk/outgoing |
| 02:05.47 | mrichman | AlmightyOatmeal: what is a dialplan? |
| 02:05.52 | WindowsUser | <PROTECTED> |
| 02:05.53 | eppigy | florz: maybe a nice massage? |
| 02:06.15 | rob0 | And some oatmeal with the platypus! |
| 02:06.21 | eppigy | YES |
| 02:06.24 | AlmightyOatmeal | mrichman: the phone itself has a specific dialplan that you will need to pause a specific amount of time befoe pushing the dial to asterisk or you will dial nothing but extensions |
| 02:06.29 | WindowsUser | ~dialplan |
| 02:06.30 | infobot | dialplan is probably the thing configured in extensions.conf |
| 02:06.37 | AlmightyOatmeal | mrichman: the cisco ip phones themselves have a dialplan as well as asterisk |
| 02:07.15 | AlmightyOatmeal | speaking of that, i sold my 7911G on ebay and i need to email the buyer my config files |
| 02:07.19 | mrichman | I should probably start reading some docs |
| 02:07.33 | AlmightyOatmeal | mrichman: yes. |
| 02:07.34 | mrichman | What can a 7940 fetch on ebay? |
| 02:07.42 | florz | eppigy: I think that really would be a bit too off-topic in here. |
| 02:07.42 | AlmightyOatmeal | mrichman: why don't you check ebay.com and see? |
| 02:07.50 | florz | ! |
| 02:08.02 | eppigy | 8[] |
| 02:09.33 | AlmightyOatmeal | time to get baby ready for bed.. big day tomorrow, little surgery for her |
| 02:09.36 | AlmightyOatmeal | good luck mrichman |
| 02:09.40 | AlmightyOatmeal | you're going to need it |
| 02:09.43 | mrichman | AlmightyOatmeal: thanks lol |
| 02:09.49 | AlmightyOatmeal | is afk(babies,sleep,spandex); |
| 02:09.53 | rob0 | good luck to the poor baby! |
| 02:10.00 | AlmightyOatmeal | ty |
| 02:12.10 | sebbl | WindowsUser i dont understand what i to do. Ich have added the dialplan http://das-asterisk-buch.de/1.0/call-file-weckruf.html |
| 02:13.19 | sebbl | toll... jetzt versteh ich nix mehr |
| 02:15.23 | *** join/#asterisk propellerhead (n=yogurt2u@200.43.87.49) |
| 02:16.30 | sebbl | i need a wakeup call who can i aktivate from my phone |
| 02:22.10 | WindowsUser | that looks like a pain to use |
| 02:23.05 | *** join/#asterisk serph (n=serph@70.49.145.76) |
| 02:23.10 | WindowsUser | gotta punch in like *77*200907060700 for a 7am call tomorrow |
| 02:23.44 | [TK]D-Fender | WindowsUser: depends on your point of view. Coding wise its the simplest. No need for multiple prompts, validation, etc |
| 02:25.22 | [TK]D-Fender | And like usual we keep getting people who say "it doesn't work" without APSTEBIN-ing the attempt & failure for us to examine. |
| 02:28.33 | sebbl | tommorow? here is 04:28 at night :) |
| 02:29.40 | WindowsUser | anyways |
| 02:32.52 | sebbl | with touch -t *2000*200907060428 the server say wrong time format and with touch *2000*200907060428 the asterisk dont call the extension 2000 |
| 02:33.41 | WindowsUser | the touch is a timestamp |
| 02:33.48 | WindowsUser | the person to call is inside the file |
| 02:33.54 | WindowsUser | read the english docs on call files |
| 02:34.16 | WindowsUser | http://www.voip-info.org/wiki/view/Asterisk+Documentation+1.6.1+callfiles.txt |
| 02:36.19 | [TK]D-Fender | [22:32]<sebbl>with touch -t *2000*200907060428 the server say wrong time format and with touch *2000*200907060428 the asterisk dont call the extension 2000 <-- you don't put a "*" after the 2000 |
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| 03:19.06 | sebbl | ok |
| 03:19.44 | sebbl | it works... but who can i used this with the dialplan. |
| 03:23.57 | [TK]D-Fender | seb? |
| 03:24.04 | [TK]D-Fender | sebbl: ? |
| 03:24.47 | sebbl | i have this added to my extension.conf http://www.das-asterisk-buch.de/2.1/call-file.html#call-file-weckruf |
| 03:26.29 | sebbl | then i dial *77*200907060526 but the asterisk dont "wake me up" |
| 03:28.46 | [TK]D-Fender | sebbl: And I don't see you showing the failed attempt or the call file that got created, or your configs. |
| 03:31.56 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 03:32.16 | xpot | anyone have any ideas about implementing an outbound rule that looks up the area-code/country code of dialed number, checks the current time at the dialed location and plays an alert if you are calling that area too early or too late?? |
| 03:33.37 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-223fdf1c34bbd6d3) |
| 03:33.59 | rob0 | oh my, that sounds too smart |
| 03:34.32 | [TK]D-Fender | xpot :Looks up in what? |
| 03:34.54 | eppigy | xpot: god I have to do that here shortly |
| 03:35.20 | sebbl | [TK]D-Fender thank you :) i have found the error. the problem was that i dont use sccp |
| 03:35.22 | xpot | ?? maybe an internet engine somewhere? ;) I could probably find a timezone offset table somewhere and import it into a MySql table? |
| 03:36.25 | *** join/#asterisk jmacz (n=jmacz@166.210.186.203) |
| 03:36.43 | [TK]D-Fender | xpotWell thats the only bit of code to do. 1 little lookup and a comparison on the time |
| 03:37.56 | xpot | [TK]D-Fender: was thinking someone might have had a quicker fix already -=0) I will get working on it then. Thanks |
| 03:39.00 | [TK]D-Fender | xpot : Quick fix? You don't have a target resource yet. |
| 03:39.13 | [TK]D-Fender | xpot : Clues are in the bin to the left of the door.... |
| 03:40.03 | xpot | [TK]D-Fender: nope. ... I don't see the bin, I only see the water filtration device... |
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| 04:04.27 | _pepo_ | hi friends |
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| 05:51.25 | rob0 | I need to disable voicemail, copying a config over from a system which has voicemail to a temporary one, no more than 2 days. Is there a quick-n-easy solution, or do I just change the dialplan to bypass the voicemail()? |
| 05:52.19 | *** join/#asterisk Chris-NB (n=chris@85-126-61-10.work.xdsl-line.inode.at) |
| 06:04.56 | carrar | unload the voicemail mdule |
| 06:04.57 | carrar | module |
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| 06:13.37 | rob0 | thanks |
| 06:13.41 | rob0 | ~sipnat |
| 06:13.42 | infobot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
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| 07:01.02 | zeeesh | like asterisk speaks with gtalk.. is there any plugin or way to speak with skype or yahoo ? |
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| 07:33.36 | tzafrir_laptop | Doesn't yahoo use sip? |
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| 07:53.08 | zeeesh | there are some forums i red ,,, yahoo uses sip over TCP ... |
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| 08:16.31 | defswork | has a vodafone femtocell and would like to hack it |
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| 09:30.51 | viraptor | do I need to enable some specific option to make rtpkeepalive work? I set it to 2 seconds, but nothing is sent when the call is on hold :/ |
| 09:39.10 | viraptor | ok.... did anyone get rtpkeepalive to work? |
| 09:55.15 | *** join/#asterisk Ex_peter (n=peter@unaffiliated/expeter/x-019426) |
| 10:00.19 | *** join/#asterisk Slashman (n=Slash@ariane.fimasys.com) |
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| 10:29.26 | *** join/#asterisk arossouw (n=arossouw@dsl-146-28-60.telkomadsl.co.za) |
| 10:30.05 | arossouw | hi, what are the techniques for troubleshooting inbound calls to a asterisk server ? |
| 10:30.32 | arossouw | i set the asterisk to debug and verbose. The problem experienced is that ocassionally calls do not even get picked up by asterisk |
| 10:31.07 | *** join/#asterisk Faustov (i=user@gentoo/user/faustov) |
| 10:31.27 | arossouw | the ISDN hardware is Junghanns with 3 ISDN box'es connected to it |
| 10:32.28 | Faustov | hi, I'm seeing "ERROR[4987]: cdr_sqlite.c:160 sqlite_log: cdr_sqlite: attempt to write a readonly database" lines in the CLI each time a call is made - I checked /var/log/asterisk and the permissions are fine (asterisk is the owner of all files and dirs there) - what could be the reason? |
| 10:33.37 | arossouw | Faustov: is asterisk being run as the same user that owns the database file? |
| 10:34.44 | Faustov | yes |
| 10:35.10 | Faustov | that is, if the /var/log/asterisk/* contains the database file |
| 10:35.14 | Faustov | but i think it does |
| 10:35.45 | Faustov | -rw-r--r-- 1 asterisk asterisk 64512 Jul 6 10:08 cdr.db |
| 10:35.51 | Faustov | this is the file, right? |
| 10:36.14 | *** join/#asterisk XiXaQ (n=jes@135.137.34.95.customer.cdi.no) |
| 10:36.31 | arossouw | if you try chmod 664 /var/log/asterisk/cdr.db and restart asterisk, see what happens |
| 10:37.40 | arossouw | if that doesn't work start asterisk in debug mode,kill all asterisk instances first then execute asterisk -gvvvvvvvcd |
| 10:38.11 | Faustov | looks like it's gone |
| 10:38.16 | Faustov | but it's weird, since owner had read and write |
| 10:38.28 | Faustov | maybe it's a false warning? |
| 10:38.48 | arossouw | nah, i think you need -rw-rw-r-- on that file |
| 10:39.05 | arossouw | you had -rw-r--r-- |
| 10:39.08 | Faustov | so sometimes it uses user rights, sometimes group? doesn't make sense... |
| 10:39.11 | Faustov | well ok |
| 10:39.28 | Faustov | let me just observe it for a while if it doesn't reappear |
| 10:40.25 | arossouw | it should give you warnings if you start calling, if something is wrong |
| 10:51.24 | *** join/#asterisk Slashman (n=Slash@ariane.fimasys.com) |
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| 11:01.35 | Faustov | arossouw: ok thanks, all is ok |
| 11:08.11 | arossouw | great |
| 11:17.27 | *** join/#asterisk Borai (n=DYN@S0106001c109e98db.no.shawcable.net) |
| 11:17.27 | Borai | hello |
| 11:18.46 | Borai | I cant pass CFLAGS to make while compiling asterisk, the makefile will overwrite my CFLAG entry any idea on what to do ? |
| 11:18.53 | Borai | im trying to compile -m32 |
| 11:22.11 | *** join/#asterisk coppice (n=chatzill@204.196.17.210.dyn.pacific.net.hk) |
| 11:24.38 | tzafrir_laptop | OT: anybody elase having problems reaching afraid.org ? (as a name server) |
| 11:25.12 | coppice | I'd be afraid to even try |
| 11:25.27 | tzafrir_laptop | Borai, which makefile? main one? menuselect? |
| 11:27.03 | JT | tzafrir_laptop: i'm afraid i had no luck |
| 11:27.35 | tzafrir_laptop | it's the name server for tzafrir.org.il |
| 11:27.44 | coppice | tzafrir_laptop: unknown host ns1.afraid.org |
| 11:34.27 | Borai | the main one |
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| 11:49.44 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 11:50.23 | lenne_dk | Hi. I want to add an external number to a queue. In 1.2 I had a queue member SIP/12345@voisp |
| 11:50.48 | *** join/#asterisk ramindia (n=balajibh@202.63.96.10) |
| 11:50.49 | lenne_dk | In 1.4 the number is flagged as (Invalid) in show queue |
| 11:51.14 | lenne_dk | Is it not possible anymore, or is syntax changed? |
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| 11:52.23 | lenne_dk | I have local queue members as SIP/10 but they are registered. the external number is not. |
| 11:52.49 | lenne_dk | Anybody here? |
| 11:53.41 | lenne_dk | Just say no, if you are not here :-) |
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| 12:05.36 | lenne_dk | Any guru here? |
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| 13:22.18 | beek | morning jaytee |
| 13:22.35 | jaytee | morning beek |
| 13:25.03 | *** join/#asterisk telnettech (n=telnette@cpe-71-74-91-116.insight.res.rr.com) |
| 13:25.36 | telnettech | All....need some assistance what does the following error mean? chan_dahdi.c:8703 pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 |
| 13:26.27 | telnettech | I am unable to make outbound calls on the PRI and it was working before....trying to figure out if it is a Asterisk problem or Telco..... Im leaning Telco but the dynamic spans show ok |
| 13:26.51 | *** join/#asterisk vegbox (n=kevinle@24-205-54-7.dhcp.gldl.ca.charter.com) |
| 13:28.49 | [TK]D-Fender | telnettech: used to get those on the old revision PCI cards with IRQ misses, etc. "cat /proc/interrupts" |
| 13:28.49 | Nugget | telnet is eeeeeeevil! |
| 13:30.05 | telnettech | TK. Im not using a PCI card.....using redfone devic.....I know yuck!!!!! |
| 13:32.11 | Poincare | hou can I limit ONLY the incomming calls to a sip-phone without limiting out-going calls from that sip-phone? call-limit seems to work in both directions.. |
| 13:32.45 | telnettech | ok I did the interrupts and this is what i see http://pastebin.com/d1aac85ea |
| 13:33.14 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 13:35.31 | telnettech | i dont see anything ot of the ordinary from what i know |
| 13:35.39 | telnettech | ^^^out |
| 13:36.00 | *** join/#asterisk putnopvut (n=putnopvu@asterisk/master-of-queues/mmichelson) |
| 13:36.00 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 13:36.21 | *** join/#asterisk knipster (n=knipster@164.55.254.106) |
| 13:37.07 | *** join/#asterisk shazaum (n=98u89weu@unaffiliated/shazaum) |
| 13:37.49 | [TK]D-Fender | telnettech: Was only relevant to a Digium PCI card... no diea at this point. |
| 13:37.58 | telnettech | ok thanks |
| 13:37.59 | [TK]D-Fender | Poincare: What do you want to limit inbound to? |
| 13:38.37 | telnettech | I am thinking it is a Telco issue as the redfone device was working just 2 hours ago and we are not doing anything with the * server |
| 13:40.31 | Poincare | [TK]D-Fender: to one call |
| 13:43.05 | [TK]D-Fender | Poincare: You can use GROUP() in your dialplan to limit that. Do you want to not call in if they are on ANY call, or only 1 INBOUND? |
| 13:43.23 | Poincare | ANY call |
| 13:43.23 | [TK]D-Fender | telnettech: That is a very open-ended write-off.... |
| 13:43.32 | [TK]D-Fender | Poincare: then ChanIsAvail() <- |
| 13:43.59 | *** join/#asterisk nfi|ermes (n=ErMeS@217.220.121.62) |
| 13:44.35 | telnettech | TK: I know but that is the last thing i have to go on. I am checking the mailing list for anything else...i am reading about the interrupts but there also seems to be a thing about graphics so im looking into that as well |
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| 13:45.23 | Poincare | [TK]D-Fender: thanks |
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| 14:12.17 | JackTheNipple | [TK]D-Fender: |
| 14:12.51 | JackTheNipple | [TK]D-Fender: do you know how to statically switch of the callerid presentation in chan_dahdi.conf? |
| 14:13.26 | [TK]D-Fender | JackNot offhand if possible |
| 14:13.54 | JackTheNipple | [TK]D-Fender: srry |
| 14:14.03 | JackTheNipple | [TK]D-Fender: understand.... |
| 14:16.03 | *** join/#asterisk moy (n=moy@bas1-unionville55-1177733615.dsl.bell.ca) |
| 14:18.33 | rob0 | hmmm, why I am getting "Seattle WA" <2062040232> as my callerID, and not what I set90 ? |
| 14:18.37 | rob0 | set() |
| 14:19.10 | rob0 | when I call with cell, the caller ID is right. |
| 14:19.17 | JackTheNipple | rob0: you have the otherside of my problem ;-) |
| 14:21.56 | rob0 | no time today, oh well |
| 14:21.57 | rob0 | afk |
| 14:22.02 | JackTheNipple | rob0: me; i can suppress the number on DAHDI, |
| 14:22.18 | JackTheNipple | rob0: okay - else let me know if you're on DAHDI as well.... |
| 14:23.16 | rob0 | set() was working for my Sipura ATA. In fact it seems to work fine here, but not at the place where I am moving it to. |
| 14:24.17 | rob0 | when this * server gets there, I should be able to plug it in and have it work. |
| 14:24.37 | rob0 | anyway, gtg, bye. |
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| 14:58.43 | brah | Can I run Zaptel instead of DAHDI on 1.4? |
| 15:00.11 | *** join/#asterisk nkohh (n=justin@unaffiliated/kohh) |
| 15:01.00 | Joel | sure, just use a version of 1.4 that still supports zap |
| 15:02.48 | *** join/#asterisk andres833 (n=andres83@190.144.102.122) |
| 15:04.55 | viraptor | do I need to enable some specific option to make rtpkeepalive work? I set it to 2 seconds, but nothing is sent when the call is on hold :/ (only sometimes, it works for one phone, but not the other and I don't see any difference in traces) |
| 15:05.13 | brah | Alright, so asterisk 1.6 doesn't work on FreeBSD |
| 15:05.32 | eppigy | DONDE ESTA |
| 15:06.39 | *** join/#asterisk Deeewayne (n=dwayne@75.76.254.162) |
| 15:06.39 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
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| 15:08.24 | [TK]D-Fender | brah: http://www.freebsd.org/ports/net.html |
| 15:08.31 | [TK]D-Fender | brah: Begs to differ |
| 15:08.54 | brah | I should have been clearer. |
| 15:08.59 | brah | No DAHDI in FreeBSD |
| 15:11.53 | Joel | who knows, not a BlowSD user :D |
| 15:13.24 | brah | bee ass dee |
| 15:13.30 | viraptor | or maybe someone knows how to send rtcp responses while I'm on hold? |
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| 15:23.24 | ramindia | [TK]D-Fender: Hi |
| 15:25.33 | *** join/#asterisk pierrelux (n=pierrelu@IP-208-88-110-46.mtl.fibrenoire.ca) |
| 15:26.05 | *** join/#asterisk paramobile (n=pk@host217-114-156-190.pppoe.mark-itt.net) |
| 15:26.13 | *** join/#asterisk markbest (n=chatzill@66.236.6.194.ptr.us.xo.net) |
| 15:26.25 | paramobile | пÑивеÑÑÑвÑÑ. кодиÑовка ноÑм ? |
| 15:27.34 | russellb | english, please :-) |
| 15:27.47 | HeXiLeD | ноÑм ? |
| 15:27.50 | *** join/#asterisk Whitor (n=Whitor@rrcs-24-97-4-146.nys.biz.rr.com) |
| 15:27.54 | *** join/#asterisk mechbangirc (n=mechbang@203.99.48.49) |
| 15:28.00 | mechbangirc | hi |
| 15:28.03 | paramobile | any russian guy here ? |
| 15:28.07 | paramobile | ;) |
| 15:28.57 | mechbangirc | is there a way to specify subnet 200.xx.xx.0/24 in host field instead of single IP?? |
| 15:29.13 | [TK]D-Fender | mechbangirc: No. |
| 15:29.29 | [TK]D-Fender | mechbangirc: You can use the permit/deny to restrict connections though |
| 15:29.48 | mechbangirc | [TK]D-Fender: so if i have to allow some subnet what is the best approach |
| 15:30.00 | paramobile | who use skypiax for work with Skype via Asterisk ? |
| 15:30.25 | mechbangirc | [TK]D-Fender: ok and host field should be set to dynamic right? |
| 15:31.00 | [TK]D-Fender | mechbangirc: yes |
| 15:32.19 | paramobile | how to tell asterisk to hangup skypecall with "skypiax_hangup" command but not with "hangup" command when user hangs up? |
| 15:33.24 | mechbangirc | [TK]D-Fender: what if i have two subnets can I put two permit rules next to each other or do i have to write another sip context for next permit |
| 15:34.36 | [TK]D-Fender | mechbangirc: I believe you can issue multiple permit |
| 15:35.07 | mechbangirc | [TK]D-Fender: ok |
| 15:35.58 | mort_gib | Would a firewall not be better for restricting access?? |
| 15:36.44 | [TK]D-Fender | mort_gib: Not for just a given entry |
| 15:37.18 | mort_gib | Mm, But you WOULD have your FW configured regardless -Right?? |
| 15:37.24 | markbest | Does anyone know how to 'delay' the PAGE() or MEETME() commands? I need to conference in several phones, but the phones I'm using (Ascom i75 wifi phones) need two rings instead of just one - for autopickup. |
| 15:38.05 | [TK]D-Fender | mort_gib: Sorry, could you be a little MORE generic please? ;) |
| 15:38.18 | [TK]D-Fender | markbest: I already answered you |
| 15:38.24 | [TK]D-Fender | markbest: And it isn't changing.... |
| 15:38.39 | *** join/#asterisk PTorres (n=PTorres@200.68.87.146) |
| 15:38.42 | mort_gib | Sure How about your old favorite?? -Will a server be good enough for asterisk?? |
| 15:38.48 | markbest | I'm sorry i must have miss your answer... |
| 15:39.28 | [TK]D-Fender | markbest: Yeah, It was only the line immediately following your question and prefixed with your name ;) |
| 15:39.39 | wmfg | Hi everyone. Let me know if this is not the place to ask... |
| 15:39.44 | [TK]D-Fender | looks for a glowing neon sign.... |
| 15:39.50 | wmfg | We want to put an Asterisk server between our pre-PRI (CAS) Sprint PBX, and our Cbeyond T1 (which we will shortly switch to SIP trunking.) |
| 15:40.01 | wmfg | Currently testing hardware (Digium TE1100P) and trying to get a handle on CAS, E&M Wink, Direct Inward Dial. |
| 15:40.29 | *** join/#asterisk marv[work] (n=timr@router.asteriasgi.com) |
| 15:40.41 | wmfg | Temporarily connected Asterisk to the T1. Asterisk Dial out fine. Dial In...cannot figure out how to tell Asterisk to expect and wait for 4 Direct Inward Dial digits: If dial in to DID 3250 asterisk tries extension "25". Dial in to 3260 asterisk tries extention "260". Again this is temporary, trying to figure things out. |
| 15:40.52 | *** join/#asterisk jamesh1 (n=jhenders@xob.neospire.net) |
| 15:40.56 | wmfg | Connected Asterisk to the PBX. Dial PBX to Asterisk ok (except works and immediately accepts dial digits even though doesn't provide dial tone until about 4-5 seconds.) Dial Asterisk to PBX not work at all...I don't know how to provide the 4 Direct Inward Dial digits that the PBX is expecting. |
| 15:41.01 | markbest | I'm sorry but its not on my history list. Could you please re post as I didn't receive your response. (Although I assume the answer is no) |
| 15:41.20 | wmfg | ?comma pause?"w" pause?D() dial option? |
| 15:42.50 | Joel | wmfg so are you looking for a consultant, or are you going to ask a detailed question at some point? |
| 15:42.58 | [TK]D-Fender | [11:12]<markbest>Does anyone know how to 'delay' the PAGE() or MEETME() commands? I need to conference in several phones but the phones I'm using (Ascom i75) need two rings instead of just one - for autopickup. |
| 15:42.59 | [TK]D-Fender | [11:24]<[TK]D-Fender>markbest: Sit back and wait a few seconds. |
| 15:43.01 | wmfg | When I say "Asterisk to PBX not work at all" I mean it just rings and rings. |
| 15:43.11 | [TK]D-Fender | markbest: Only *6* lines happend since <- |
| 15:43.37 | wmfg | If I must hire a consultant, I'm willing, but I've gotten quite far on my own so far. |
| 15:43.55 | [TK]D-Fender | wmfg: PASTEBIN is your friend... show us yuor configs and failed call attempts |
| 15:44.24 | jamesh1 | anyone have trouble with certain cell phones and dtmf? |
| 15:44.33 | markbest | Roger. *Sits back and waiting* |
| 15:44.44 | *** join/#asterisk Iamnach0 (i=Iamnacho@ip174-70-137-120.ks.ks.cox.net) |
| 15:44.48 | jamesh1 | been knocking my head against the wall |
| 15:44.48 | *** part/#asterisk PTorres (n=PTorres@200.68.87.146) |
| 15:44.55 | wmfg | Specific question: How to tell Asterisk to push DID digits during DIAL. |
| 15:45.05 | wmfg | ...On a CAS T1 |
| 15:45.10 | Joel | wmfg Specific answer: what? |
| 15:45.32 | Joel | wmfg when asterisk calls out, it dials a number, that number is the did. |
| 15:46.01 | Joel | there is no way to "fake" the number you've dialed.... |
| 15:46.41 | *** join/#asterisk pmhaddad-lappy (n=pmhaddad@24-231-204-189.dhcp.bycy.mi.charter.com) |
| 15:46.49 | wmfg | That makes sense. |
| 15:47.09 | Joel | telcom 101 here :) |
| 15:47.10 | wmfg | But I thought I tried that with no result. |
| 15:47.18 | Joel | you thought you tried calling a number? :P |
| 15:47.51 | Joel | if you're doing sip->asterisk->legacy then good luck, you're probably not doing something your legacy expects |
| 15:48.00 | wmfg | I thought I had already tried a 4 digit DID number known to work...still rings and rings. Timing issue? |
| 15:48.03 | Joel | I would make sure your legacy is looking for 10 digits of the dialed number |
| 15:48.15 | Joel | and I would pass 10 digits |
| 15:48.30 | Joel | and if your legacy doesn't answer, then ask someone who's familiar with the legacy to take a look and tell you why |
| 15:48.57 | [TK]D-Fender | wmfg: Its all part of your dial |
| 15:49.38 | jamesh1 | Question: Anyone know of the problem/fix for certain cell phones not being recognized in asterisk1.4 in the ivr? (dtmf) |
| 15:49.41 | wmfg | Legacy is currently expecting 4, should be changable. Plan to hire with legacy telcom guy later this week, but was hoping to be well versed beforehand. |
| 15:50.07 | Joel | jamesh1 relaxdtmf=1 |
| 15:50.33 | jamesh1 | place that in chan_dahdi? |
| 15:51.11 | [TK]D-Fender | wmfg: Ask your telco the signalling order & # of digits expected |
| 15:52.55 | wmfg | d-fender: telco currently provides and PBX currently expects four DID digits. What would be the menu for signalling order? Is E&M Wink what you mean? |
| 15:52.58 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 15:53.16 | *** join/#asterisk ayeso (n=chatzill@ext-52.sagetelecom.net) |
| 15:53.48 | [TK]D-Fender | wmfg: do they expect a wink between DID and the target #? A pause? Continuous digits? |
| 15:54.01 | jamesh1 | joel: sorry wrong rebuttal. if I place that in the config is a restart or reload needed? |
| 15:55.24 | ayeso | How do I compare with an &? like this: GotoIf($["${CALLERID(ani)}" = "123456789" & "${CALLERID(DNIS)}" = "987654321"]?wherever,1,1) or like this: GotoIf($["${CALLERID(ani)}" = "123456789"] & ["${CALLERID(DNIS)}" = "987654321"]?wherever,1,1) |
| 15:57.24 | ayeso | basicly does the and go in 1 bracket or between 2 brackets? [var = val & var = val] or is it [var =val] & [var = val]? |
| 15:57.58 | [TK]D-Fender | ayeso: Go read the CHANNELVARIABLES doc again |
| 16:00.55 | ayeso | [TK]D-Fender: no luck, but I think i found my answer on the expressions page |
| 16:01.11 | [TK]D-Fender | ayeso: Its in the doc as well. |
| 16:01.13 | [TK]D-Fender | && <- |
| 16:01.19 | ayeso | its [val = var & val =var] |
| 16:04.51 | *** join/#asterisk ariel_ (n=chatzill@63.214.236.169) |
| 16:07.32 | *** join/#asterisk chendy (n=chatzill@116.30.195.158) |
| 16:08.42 | ariel_ | Hello folks |
| 16:09.20 | wmfg | joel, d-fender: Thanks for saying "dial IS did." I was being _that_ stupid. 4 digit Asterisk dial string works perfectly to PBX. |
| 16:10.46 | *** join/#asterisk kombi (n=kombi@cpe-68-175-101-211.nyc.res.rr.com) |
| 16:11.09 | *** join/#asterisk maximo (n=maximo@CPE001217b1920e-CM00111ade9528.cpe.net.cable.rogers.com) |
| 16:13.03 | kombi | trying to connect to asterisk with x-lite on mac but no answer on udp 5060, even when nmap'ed locally, what might be wrong? |
| 16:18.19 | angryuser | Kobaz, firewall, do a tcpdump on port 5060 locally |
| 16:19.04 | angryuser | Kobaz, >> kombi |
| 16:19.11 | moa_ | Anyone have some insight on solving this? It's causing choppy call quality every time this happens. http://pastebin.ca/1485815 |
| 16:21.35 | kombi | angryuser: firewall is completely off... I'll try tcpdump (if that exists on mac..) |
| 16:22.08 | angryuser | kombi, try to do a dum on the target machine * |
| 16:22.27 | angryuser | moa_, do you use asterisk 1.4.11 ? |
| 16:23.09 | *** join/#asterisk lucasb (n=bussey@s154-5-252-231.bc.hsia.telus.net) |
| 16:23.27 | moa_ | It's actually older, 1.4.4. I'm in the process of upgrading to the latest, I've never had a problem until now for some strange reason. |
| 16:23.49 | angryuser | moa_, https://issues.asterisk.org/view.php?id=9833 |
| 16:24.25 | angryuser | moa_, and next time google first |
| 16:24.33 | Qwell | no, next time *upgrade* first. |
| 16:25.22 | moa_ | I did google, and I am upgrading. Just thought I'd ask :) |
| 16:25.45 | Qwell | Changes since asterisk Version 1.4.4/ - svn revision 62252 |
| 16:25.45 | Qwell | 2701 |
| 16:26.22 | Qwell | so...2000 or so bugs fixed |
| 16:28.45 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 16:33.45 | *** join/#asterisk Olobola (i=Olobola@231.sub-75-209-143.myvzw.com) |
| 16:38.26 | jeff | Qwell: bad metric, since the revisions between would include work being done on trunk/HEAD and other branches, etc. |
| 16:38.46 | jeff | grins |
| 16:41.31 | Qwell | jeff: No it doesn't |
| 16:42.07 | jeff | Qwell: ah, i misunderstood, then. |
| 16:42.38 | jeff | ah, i see. yep, my mistake. sorry. :) |
| 16:46.03 | *** join/#asterisk icyValk77 (n=icyValk7@host81-153-93-26.range81-153.btcentralplus.com) |
| 16:46.15 | *** join/#asterisk thansen (n=thansen@74-36-210-251.dr01.hmdl.id.frontiernet.net) |
| 16:51.07 | Joel | pets Qwell |
| 16:52.45 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 16:54.43 | *** join/#asterisk JackTheNipple (n=JackTheN@static-87-79-237-194.netcologne.de) |
| 16:54.53 | *** join/#asterisk vegbox (n=kevinle@wireless-169-235-59-82.ucr.edu) |
| 16:56.09 | *** join/#asterisk k3rn3l (n=dam@infapen.com) |
| 16:56.16 | k3rn3l | hi |
| 16:56.19 | k3rn3l | good morning |
| 16:56.29 | k3rn3l | somebody speak spanish? |
| 16:57.29 | coppice | 西çå |
| 16:57.30 | voipheroes | n0t m3 |
| 16:57.35 | voipheroes | ;) |
| 16:57.36 | k3rn3l | :S |
| 16:57.38 | k3rn3l | i need some help |
| 16:57.44 | k3rn3l | i have a old panasonic pbx |
| 16:57.46 | [TK]D-Fender | k3rn3l: #drphil |
| 16:57.59 | k3rn3l | and i have a new pbx asterisk |
| 16:58.48 | k3rn3l | in my old pbx i need digit #*3225 to can cal to the cell phones |
| 16:58.51 | k3rn3l | **call |
| 16:59.11 | k3rn3l | and i want to send the #*3225 from asterisk to can have a dialtone |
| 16:59.35 | [TK]D-Fender | k3rn3l: So Dial that. |
| 17:01.19 | *** join/#asterisk bmoraca (n=chatzill@66.242.174.254) |
| 17:01.39 | k3rn3l | how? |
| 17:01.42 | k3rn3l | i have the next dialplan |
| 17:01.52 | *** join/#asterisk xpot-mobile (n=james@static-66-182-88-85.bbsc.net) |
| 17:02.02 | k3rn3l | exten => _9044XXXXXXXXXX,2,Dial(Zap/5/#*3525,30,D(${EXTEN:1}),,To) |
| 17:06.42 | *** join/#asterisk lanning (n=lanning@nat/yahoo/x-21aa73d36af8bce7) |
| 17:09.14 | [TK]D-Fender | k3rn3 : exten => _9044XXXXXXXXXX,2,Dial(Zap/5/#*3525wwww${EXTEN:1}),30,To) |
| 17:09.24 | [TK]D-Fender | k3rn3l: w = .5 second wait |
| 17:09.44 | k3rn3l | :o trying |
| 17:09.46 | [TK]D-Fender | k3rn3l: just to give the other PBX a chance to ACk the code & provide the second tone if required |
| 17:11.15 | *** join/#asterisk Whitor (n=Whitor@64.128.237.124) |
| 17:11.36 | k3rn3l | - Executing Dial("SIP/200-08264780", "Zap/5/#*3525wwww0449982126568|30|To") in new stack -- Called 5/#*3525wwww0449982126568 |
| 17:11.52 | k3rn3l | "The numbers is not in the dialplan" |
| 17:12.48 | k3rn3l | exten => _9044XXXXXXXXXX,2,Dial(Zap/5/#*3525wwww${EXTEN:1},30,To) |
| 17:16.19 | *** join/#asterisk MindTheGap (n=MindTheG@187.20.141.72) |
| 17:16.57 | *** join/#asterisk cusco (n=tralala@213.63.137.210) |
| 17:17.01 | cusco | hi |
| 17:17.03 | cusco | what sip client (softphone) would you recomend to test an asterisk server? I could use a command line one that I could run several instances |
| 17:17.06 | cusco | for linux |
| 17:17.38 | *** join/#asterisk ruben23 (n=RPL@124.107.3.178) |
| 17:18.01 | *** join/#asterisk Iamnacho (i=Iamnacho@ip174-70-137-120.ks.ks.cox.net) |
| 17:18.49 | ruben23 | hi, i have asterisk-getting low volume on clients voice in calls, all the volume adjustment of the headset is on high level. |
| 17:18.57 | [TK]D-Fender | k3rn3l: What interface are you using? |
| 17:19.03 | ruben23 | what could be causing it...? |
| 17:19.23 | k3rn3l | thks [TK]D-Fender |
| 17:19.32 | k3rn3l | [TK]D-Fender: zap |
| 17:19.40 | [TK]D-Fender | k3rn3l: What card & signalling? |
| 17:22.08 | eppigy | DONDE ESTA CPE |
| 17:22.45 | *** join/#asterisk ingenius (n=alektro@186.136.6.218) |
| 17:23.27 | ruben23 | hi,, |
| 17:23.33 | ruben23 | anyone..? |
| 17:24.01 | ruben23 | im using voip. with leased line 1 Mbps |
| 17:25.10 | [TK]D-Fender | ruben23: Internet connection doesn't make the fixed sound in your VoIP traffic low. |
| 17:25.13 | JackTheNipple | ruben23: on E1 there is a parameter called rxgain & txgain |
| 17:25.25 | [TK]D-Fender | ruben23: Its either the fault of the sender, or the receiver |
| 17:25.29 | *** join/#asterisk kombi (n=kombi@cpe-68-175-101-211.nyc.res.rr.com) |
| 17:25.43 | errr | if I have 2 aastra 55i phones, is it possible to direct sip dial from 1 phone to the other with out the call going though asterisk? |
| 17:25.54 | JackTheNipple | ruben23: maybe there is something on the client side like this? |
| 17:26.10 | *** join/#asterisk jmacz (n=jmacz@190.144.75.22) |
| 17:26.22 | k3rn3l | [TK]D-Fender: solved thnks:D |
| 17:26.27 | JackTheNipple | ruben23: as D-Fender says - VoIP does not have this kind of parameters... |
| 17:26.27 | k3rn3l | other question |
| 17:26.29 | k3rn3l | :D |
| 17:26.52 | kombi | trouble connecting with x-lite, keep getting 408. Works fine otherwise, just not with remote connection over port forward. What must I do? |
| 17:26.56 | *** join/#asterisk Winkie (n=urmom@ur.fa.gs) |
| 17:27.23 | k3rn3l | i need to make some app, that be like a bot, this bot have to call some numbers in different hours of the day and say "Hello you are {CustomerName} and you have to pay ${moneymount}" |
| 17:27.29 | [TK]D-Fender | Kobaz: You have a networking / firewall / NAT issue |
| 17:27.39 | k3rn3l | With what technology or commands i can do that? |
| 17:28.10 | [TK]D-Fender | k3rn3l: Read up on AGI in THE BOOK, and the mecahnism to call out will either be an AMI Originate, or "call-files" |
| 17:28.12 | [TK]D-Fender | ~book |
| 17:28.12 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 17:28.13 | JackTheNipple | k3rn3l: you want to look out for ".callfiles" |
| 17:28.19 | [TK]D-Fender | k3rn3l: All of this can be found in the book. |
| 17:29.04 | [TK]D-Fender | k3rn3l: Actually AGI probably isn't needed for this. |
| 17:29.14 | [TK]D-Fender | k3rn3l: Just a process to prepare the out-calls |
| 17:29.17 | k3rn3l | :o thanks soo much i have the first edition i download right now the second edition :D |
| 17:30.15 | kombi | better said: asterisk box is connected via port forwarding, x-lite can connect but not register. What might be wrong? |
| 17:30.45 | [TK]D-Fender | kombi: I don't see you showing us SIP DEBUG from *'s side showing the reg attempt and response... |
| 17:30.48 | [TK]D-Fender | ~pb |
| 17:30.48 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste , or , http://bin.cakephp.org/, or apt-get install pastebinit |
| 17:30.50 | [TK]D-Fender | ^^^^^^ |
| 17:31.14 | kombi | fender: ;) coming up! |
| 17:31.24 | ruben23 | [TK]D-Fender:what would be my steps to follow the root of the problem |
| 17:31.32 | ruben23 | :-( |
| 17:31.58 | [TK]D-Fender | ruben23: Test with another phone. If its better, then your headset rig is to blame. If its the same, then its your termination provider |
| 17:32.34 | [TK]D-Fender | cans ee a lot of people who wouldn't know the scientific process if it ran up and bit them in the face... |
| 17:43.14 | *** part/#asterisk icyValk77 (n=icyValk7@host81-153-93-26.range81-153.btcentralplus.com) |
| 17:43.40 | kombi | x-lite connection trouble: http://pastebin.se/198458 <- is it just wrong credentials? |
| 17:46.56 | [TK]D-Fender | kombi: <--- Transmitting (no NAT) to 192.168.1.101:3821 ---> |
| 17:47.04 | [TK]D-Fender | kombi: Bad NAT setup. go read the guide : |
| 17:47.06 | [TK]D-Fender | ~sipnat |
| 17:47.07 | infobot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 17:47.19 | *** join/#asterisk BuSyAnToS (n=31749@93-44-80-124.ip96.fastwebnet.it) |
| 17:47.36 | kombi | fender: bad NAT on server or client? |
| 17:47.50 | kombi | ..I'm reading.. |
| 17:47.56 | [TK]D-Fender | kombi: Server. |
| 17:48.45 | *** join/#asterisk errotan (n=errotan@5403E42F.catv.pool.telekom.hu) |
| 17:55.53 | markbest | [TK]D-Fender: Could you provide assistance with the PAGE() command? |
| 17:59.21 | *** join/#asterisk ttl- (n=patrick@d5153AE82.access.telenet.be) |
| 18:00.19 | eon` | [TK]D-Fender: What model polycom would you recommend to replace some Cisco 7960 IP Phones? |
| 18:01.13 | jamesh1 | Cisco stopped making IP phones, correct? |
| 18:01.36 | eon` | I have a whole load of Cisco 7960's causing me A LOT of grief. |
| 18:01.54 | jamesh1 | Those are all I use. |
| 18:02.10 | eon` | http://www.cdw.com/shop/products/default.aspx?EDC=1367000 these be ok? |
| 18:07.11 | *** join/#asterisk wonderworld (n=w@ip-62-143-16-28.unitymediagroup.de) |
| 18:08.03 | kombi | fender: ;) Sometimes things can be as easy as "nat = yes" in sip.conf.. thanks fender! |
| 18:09.11 | *** join/#asterisk [8none1] (n=[8none1]@cerberus.franklinamerican.com) |
| 18:09.34 | cusco | what sip client (softphone) would you recomend to test an asterisk server? I could use a command line one that I could run several instances for linux? |
| 18:09.50 | jamesh1 | xlite |
| 18:10.00 | cusco | can I lauch several? |
| 18:10.18 | jamesh1 | not in windows I don't believe so |
| 18:10.28 | cusco | in linux |
| 18:11.06 | jamesh1 | I have used ekiga and you can't open multiple with that. |
| 18:11.25 | ISO9001 | I don't think many will let you open multiple instances. |
| 18:12.44 | cusco | I need to test a big amount of calls with asterisk |
| 18:13.41 | KavanS | cusco, use ekiga |
| 18:13.51 | KavanS | and then use another box, to call with... |
| 18:14.10 | jamesh1 | ekiga won't let you open multiple clients |
| 18:14.22 | KavanS | sounds like an issue that is easily solved |
| 18:14.25 | KavanS | i.e. get another system |
| 18:14.45 | KavanS | if it was me, I'd not want to run two clients from the same system... |
| 18:14.56 | cusco | why...? |
| 18:15.09 | jamesh1 | same ip. |
| 18:15.10 | KavanS | uhm...isn't the idea to connect two remote end points to one another? |
| 18:15.22 | KavanS | ;) |
| 18:15.33 | jamesh1 | functionality tests though. |
| 18:15.36 | cusco | I can configure several ip's on this machine, if the client has the option to select the interface |
| 18:15.38 | KavanS | I'd want to test it as close to real world... |
| 18:15.46 | KavanS | cusco, you can get a sip adapter for 50$ shipped |
| 18:15.52 | KavanS | pretty cheap |
| 18:15.55 | cusco | sip adapter? |
| 18:15.59 | KavanS | yep |
| 18:17.27 | [TK]D-Fender | ~ATA |
| 18:17.28 | infobot | methinks ata is Analogue Terminal Adapter which provides an FXS and/or FXO and ethernet, see http://www.voip-info.org/wiki/view/ATA |
| 18:17.50 | [TK]D-Fender | And don't call it a "SIP adapter" |
| 18:18.14 | bmoraca | what's a good TDMoE device with 2 FXO ports? |
| 18:21.08 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 18:22.10 | ariel_ | TDMoE with 2 FXO I have not seen any, only ones with PRI/T1/E1's |
| 18:22.53 | bmoraca | yeah...it's not going to work anyway |
| 18:23.12 | eon` | ~[6~[6~[6~[6~[6~[6~[6~[6~[6~ |
| 18:23.25 | *** join/#asterisk GreyFoxx (i=greg@out.of.phaze.org) |
| 18:23.52 | eon` | wth |
| 18:23.59 | eon` | my terminal just flipped out |
| 18:25.43 | ariel_ | tzafrir_laptop: you around? I need some xorcom help if you can or able too |
| 18:26.26 | rene- | cusco: what about sipp |
| 18:26.48 | rene- | i believe sipp is exactly what u are looking fo |
| 18:26.49 | rene- | for |
| 18:27.16 | rene- | it is command line, it acts as a user agent, it can generate tons of calls to your system with audio |
| 18:27.33 | *** part/#asterisk GreyFoxx (i=greg@out.of.phaze.org) |
| 18:27.34 | *** join/#asterisk ttl_ (n=patrick@d5153AE82.access.telenet.be) |
| 18:27.56 | rene- | http://sipp.sourceforge.net/ |
| 18:28.22 | cusco | rene-: idt does not have voice |
| 18:28.56 | [TK]D-Fender | cusco: Why are you looking to push multiple voice calls off 1 box? |
| 18:29.22 | rene- | it has some audio samples you can use to test, and they explain how to create your own (it is complex) |
| 18:30.08 | *** join/#asterisk TrentCreek (n=kvirc@129.113.113.149) |
| 18:31.50 | cusco | because all the other boxes are being used |
| 18:31.55 | cusco | only have 2 or 3 available |
| 18:32.07 | cusco | and I would like to produce loads of calls |
| 18:32.20 | cusco | rene-: they do have audio samples? |
| 18:32.22 | cusco | thats great |
| 18:32.29 | cusco | only there is no man page for sip-test |
| 18:32.32 | cusco | only there is no man page for sip-tester |
| 18:32.33 | rene- | it does |
| 18:32.34 | cusco | lol |
| 18:33.06 | rene- | i havent figured out what in the world the voice in the audio file says, or what language is at least |
| 18:34.49 | cusco | hehe |
| 18:35.03 | cusco | I never used sip-tester I will try now |
| 18:35.29 | pmhaddad-lappy | anyone in here played with twilio yet? |
| 18:38.24 | rene- | nope, but adhearsion lets you do it in your own box |
| 18:38.30 | *** join/#asterisk korcan (n=korcan@99.23.50.73) |
| 18:39.19 | [TK]D-Fender | cusco: sipp or use another * box. |
| 18:39.22 | cusco | rene-: any tips in how to use sipp? |
| 18:39.29 | [TK]D-Fender | cusco: The multiple softphone idea is crazy |
| 18:39.48 | rene- | sipp can run in your same box, |
| 18:40.17 | [TK]D-Fender | rene-: Not a fair load tst since the tester and testee are on the SAME BOX. |
| 18:40.24 | [TK]D-Fender | rene-: Horrible "test" |
| 18:40.38 | rene- | cusco, i dont have my notes handy, read the manual |
| 18:40.52 | rene- | D-Fender: I agree |
| 18:41.10 | [TK]D-Fender | cusco: What is your planned load? |
| 18:41.32 | cusco | as much as I could lol |
| 18:41.44 | cusco | if I could get to 100 simultaneous calls, great |
| 18:42.07 | [TK]D-Fender | cusco: 100 calls doing what? What hardware? |
| 18:42.08 | cusco | rene-: it does not bring a man page lol! |
| 18:42.24 | cusco | but yea there is the docs that it sends to stoud |
| 18:42.45 | *** join/#asterisk jameswf (n=james@unaffiliated/jameswf-home) |
| 18:42.47 | rene- | go to the website http://sipp.sourceforge.net/doc/reference.html |
| 18:42.56 | [TK]D-Fender | cusco: You know Digium has been selling 4-port E1 cards for about almost a decade now.... thats 120 channels <- |
| 18:42.57 | cusco | I don't really know the specs but it is a rack server from intel that can have 2 primary connections.. lol |
| 18:43.05 | cusco | I can tell you the processor and the memory.. |
| 18:43.06 | *** join/#asterisk jeffgus (n=jeffgus@2002:ad33:b504:0:0:0:0:1) |
| 18:43.19 | [TK]D-Fender | cusco: So just spit out what you are expecting to do during those calls |
| 18:43.21 | cusco | each port has 60 channels |
| 18:43.25 | cusco | so we can handle 120 |
| 18:43.32 | *** join/#asterisk mltlnx (n=mltlnx@asams.mserve.com) |
| 18:43.53 | mltlnx | Hello? In 1.6 what is the best way to detect a fax? |
| 18:43.55 | cusco | asterisk has some odd behaviours from time to time and we have to test a massive load of calls on it |
| 18:44.02 | cusco | suddenly one queue stops ringing |
| 18:44.03 | [TK]D-Fender | cusco: So what will the box be DOING? And transcoding? Recording? All VoIP? Any TDM hardware? |
| 18:44.12 | rene- | download sipp, build it, then you would want to use the UAC with media scenario, learn how to tweak the amount of calls, and how to set your asterisk ip address and user using sipp command line switches |
| 18:44.14 | [TK]D-Fender | Any*? |
| 18:44.29 | rene- | D-Fender: is right again |
| 18:44.38 | [TK]D-Fender | cusco: 60 != massive. |
| 18:44.40 | cusco | only queueign and we may answer only 2 or 3 calls at a time |
| 18:45.11 | cusco | [TK]D-Fender: well, it suits our needs, our tops reach 60 calls queueing at the same time |
| 18:45.12 | [TK]D-Fender | cusco: And Queue-ing calls isn't anything special as load goes. |
| 18:45.20 | cusco | yes I know |
| 18:45.26 | cusco | its not about the load |
| 18:45.30 | [TK]D-Fender | cusco: So answer my other questions. |
| 18:45.38 | cusco | its about one of the queses that stops ringing |
| 18:45.48 | cusco | I don't have the answers, I don't know :( |
| 18:45.49 | cusco | hold |
| 18:46.02 | cusco | what is tdm hardware? |
| 18:46.23 | [TK]D-Fender | cusco: Telephony cards. E1, T1, POTS, etc |
| 18:46.52 | cusco | lspci returns: 02:01.0 Communication controller: Digium, Inc. Wildcard TE210P dual-span T1/E1/J1 card 3.3V (rev 02) |
| 18:47.02 | [TK]D-Fender | [14:45]<cusco>I don't have the answers, I don't know :( <-- You don't know how your calls will come in? You don't know if you'll be doing recording? Open ass, remove head :) |
| 18:47.12 | rene- | mltnx: you have spandsp rx_fax family, you can use hylafax with iax_modem, and you can use digium t.38 implementation, i have never used digium's, and spandsp rx_fax tends to be flaky, hylafax with iax_modem works ok for me |
| 18:47.16 | cusco | we are recording them, yes |
| 18:47.21 | [TK]D-Fender | cusco: And tahts what your calls will come in under? |
| 18:47.26 | cusco | Im new in the asterisk world, sorry |
| 18:47.34 | cusco | yes |
| 18:48.18 | cusco | well.. for now I only want to test a big amount of calls queuing |
| 18:48.45 | cusco | so I will be looking at sipp... if any of you could point me to some examples on usage.. thanks |
| 18:48.49 | [TK]D-Fender | cusco: And only 3 people actaully talking? A single P3 can do this |
| 18:49.32 | [TK]D-Fender | cusco: though not to look like a cheap-ass idiot I'd say a semi-decent C2D these days. |
| 18:49.38 | cusco | it could be more... depending on the machines available or the calls going in our production system |
| 18:49.54 | [TK]D-Fender | cusco: "Machines available"? As in...? |
| 18:50.02 | cusco | people answering calls or not |
| 18:50.08 | cusco | many calls at this time |
| 18:50.14 | cusco | or not |
| 18:50.15 | [TK]D-Fender | cusco: You only have about 4 people so far. |
| 18:50.19 | cusco | no |
| 18:50.25 | cusco | we are about 30 right now |
| 18:50.27 | [TK]D-Fender | cusco: this wasn't considered a load 5 years ago |
| 18:50.31 | cusco | but we are aswering calls |
| 18:50.46 | [TK]D-Fender | cusco: And in your "new plan"? |
| 18:50.52 | cusco | brb, 10 min. |
| 19:02.41 | eppigy | DONDE |
| 19:04.08 | *** join/#asterisk JackTheNipple (n=JackTheN@static-87-79-237-194.netcologne.de) |
| 19:05.39 | cusco | back |
| 19:07.36 | cusco | [TK]D-Fender: it is the same. tho when these problemas happen we just restart asterisk .. so now I would like to make it happen in our redundancy test server |
| 19:07.43 | cusco | anyway I will now look into sipp |
| 19:08.29 | [TK]D-Fender | cusco: What "problem"? |
| 19:09.55 | *** join/#asterisk fish-bulb (n=cstewart@nat/digium/x-361d551fea9893ed) |
| 19:10.12 | *** join/#asterisk jmacz (n=jmacz@190.144.75.22) |
| 19:16.12 | *** join/#asterisk raden (n=chatzill@24-240-62-13.dhcp.stpt.wi.charter.com) |
| 19:17.45 | raden | anyone recomend a good voip provider |
| 19:20.22 | [TK]D-Fender | ~itsplist-us |
| 19:20.23 | infobot | [itsplist-us] Here are some popular ITSPs (USA) starting with the more respected ones : http://www.teliax.com , http://www.voicepulse.com/connect/ , http://www.broadvoice.com, http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net |
| 19:20.43 | ariel_ | raden: what is good for me might not be for you, but I have been using voip pulse for over 4 years and have been happy with them. |
| 19:21.56 | raden | is there a program to measure a connection to test how many voip channels would be supported |
| 19:23.14 | *** join/#asterisk WhIteSidE (n=chatzill@209.236.250.9) |
| 19:24.51 | WhIteSidE | Just a quick question for the community: For installing a new asterisk server, 1.6 branch, what's the preferred version? 1.6.1.1? Or is there a more stable version? |
| 19:25.22 | [TK]D-Fender | raden: Its called a "caculator". I recommend Texas Instruments. |
| 19:25.47 | ariel_ | is still running on the 1.4.2X, have not moved up to 1.6 |
| 19:26.08 | [TK]D-Fender | WhIteSidE: 1.6.0 branch is more mature. 1.6.1 I'd wait on for another release or two just out of habit. |
| 19:26.09 | raden | [TK]D-Fender: calculator going to calculate my packet jitter or latency or how amny hops to specific call centeres >? |
| 19:26.54 | [TK]D-Fender | raden: hops = traceroute, jitter = you'd have to actually place calls, latency = ping |
| 19:27.06 | beek | WhIteSidE: 1.6.0.10 has been rock-solid for me. |
| 19:27.07 | WhIteSidE | Ok, thanks, I want to go with 1.6, as this will be a brand new dialplan, but I have been having a bit of trouble with 1.6.1.1, so I think I'll re-install. |
| 19:27.09 | WhIteSidE | Thanks |
| 19:27.17 | raden | there are programs out there that calculate jitter |
| 19:28.36 | [TK]D-Fender | raden: Yes, but as I mentioned, that should require you to actually place calls. That is a "I'm already dealing with provider XYZ" test |
| 19:31.05 | *** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk) |
| 19:32.26 | TrentCreek | Why would the dialparties.agi override CID of what is seup for each extension? |
| 19:32.39 | [TK]D-Fender | TrentCreek: Wrong channel... |
| 19:32.44 | jameswf | ~freepbx |
| 19:32.45 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 19:34.13 | TrentCreek | darn it...I dont want to go to that channel. |
| 19:34.43 | TrentCreek | too many back-stabbing jerks there |
| 19:34.51 | Qwell | TrentCreek: then...don't use freepbx? |
| 19:34.54 | [TK]D-Fender | stabs TrentCreek in the front... |
| 19:35.00 | TrentCreek | ouch! |
| 19:35.19 | bmoraca | TrentCreek: freepbx lets you set multiple CIDs for each extension and trunk...there's 50 places you could have inadvertantly told it to use a different callerid |
| 19:35.24 | TrentCreek | Qwell: I mean't the people on the channel, not the FPBX community |
| 19:35.36 | Qwell | they are one in the same |
| 19:35.47 | [TK]D-Fender | TrentCreek: they are the "community", just on a different media |
| 19:35.49 | Qwell | Are people in #asterisk not "the Asterisk community"? |
| 19:36.23 | TrentCreek | no....they talk about FPBX issues |
| 19:36.55 | TrentCreek | bmoraca: yes I am aware of that, and been using it for almost 2 years, but all of a sudden the dialparties.agi is sending out its own CID |
| 19:37.17 | *** join/#asterisk Zuchmir2 (n=Zuchmir-@ool-18bd3bfc.dyn.optonline.net) |
| 19:37.27 | TrentCreek | Guess I will head over to the PBX forum...thanks |
| 19:37.49 | Joel | TrentCreek back stabbing jerks? |
| 19:37.53 | TrentCreek | yes |
| 19:38.01 | Joel | TrentCreek you realize most of the #freepbx folks are also in here? lol |
| 19:38.13 | TrentCreek | not the ones I am referring to |
| 19:38.19 | Joel | and thanks for calling me a back stabbing jerk, way to motivate me to help, the newest freepbx developer :\ |
| 19:38.49 | [TK]D-Fender | jabs Joel with a hot poker |
| 19:38.57 | bmoraca | vomits like in the Microsoft commercial |
| 19:38.59 | [TK]D-Fender | Joel: I got 'yer "motivation" RIGHT HERE! |
| 19:39.11 | Joel | Signal To Noise ratio is so incredibly lousy in forums, and now it's getting to forum point on irc. yay. |
| 19:39.14 | [TK]D-Fender | jabs Joel with a hot poker again |
| 19:40.16 | TrentCreek | Someone over in the FPBX channel started some crap, and I had no idea what was going on...so I told this one person to f**k off and some other expletives, and have not been back since |
| 19:40.34 | bmoraca | good way to make friends |
| 19:41.03 | [TK]D-Fender | TrentCreek: "No-one can make you feel inferior without your consent" |
| 19:41.24 | outtolunc | but you can try, right fender <G> |
| 19:41.25 | [TK]D-Fender | TrentCreek: And you could have just "/ignored" them if it was that bad. |
| 19:41.26 | TrentCreek | well when you are accused of soemthing and you dont know what they are talking about, but yet they take sides, what else can one do? |
| 19:41.44 | TrentCreek | Well they went to one of the channel ops, to "complain" |
| 19:42.09 | [TK]D-Fender | TrentCreek: find a spine and stand up for yourself. |
| 19:42.32 | TrentCreek | hard to stand up when one wants to keep kicking you off |
| 19:43.01 | [TK]D-Fender | TrentCreek: As in kicking you out of the channel? |
| 19:43.03 | bmoraca | TrentCreek: if you can't provide logs of a call (freepbx and dialparties are fairly verbose in their logging), no one can help you...so why don't you start there. |
| 19:43.31 | [TK]D-Fender | TrentCreek: Ah yes... i see how |
| 19:43.33 | [TK]D-Fender | who* |
| 19:43.58 | [TK]D-Fender | TrentCreek: Yup, piss off an op and you in for trouble |
| 19:44.02 | TrentCreek | [TK]D-Fender: yes...I was trying to ask what the channel op what they were talking about, and he says "I believe someone I know for 3 years, over you" |
| 19:44.04 | [TK]D-Fender | your're* |
| 19:44.20 | [TK]D-Fender | TrentCreek: get another op to back you |
| 19:45.05 | TrentCreek | yeah I can do that, but I messaged him a bad message meant to be private, and I accidently sent it on the open channel, so that did not go over well |
| 19:45.33 | *** join/#asterisk iratik (n=itariki@209.248.216.146.nw.nuvox.net) |
| 19:45.59 | [TK]D-Fender | TrentCreek: Speak your mind to another op, be honest and see what happens. |
| 19:46.12 | *** join/#asterisk cervi (n=Iulia@89.39.205.57) |
| 19:46.17 | iratik | can you guys make any recommendations for a high-volume sip trunk provider (with no contracts involved, priced per channel/month) ? |
| 19:46.24 | *** join/#asterisk mykhyggz (n=mykhyggz@evolone.org) |
| 19:46.33 | [TK]D-Fender | TrentCreek: And passing off a bad message even in private is still a bad idea regardless. |
| 19:46.57 | [TK]D-Fender | iratik: shop through the usuals |
| 19:47.01 | [TK]D-Fender | ~itsplist-us |
| 19:47.02 | infobot | itsplist-us is probably Here are some popular ITSPs (USA) starting with the more respected ones : http://www.teliax.com , http://www.voicepulse.com/connect/ , http://www.broadvoice.com, http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net |
| 19:47.34 | TrentCreek | yeah I can do that. Someome said I was threatening them..so I showed the chat logs where he insults me and I reply with <PLONK> then I put him on ignore. Next think you know he is telling channel ops I "threatened" him. |
| 19:48.08 | TrentCreek | so one channel ops does not care because it's him "buddy" |
| 19:48.19 | [TK]D-Fender | TrentCreek: Ok, so it wasn't an OP you pissed off then... well go talk to them about it |
| 19:49.18 | *** join/#asterisk telnettech (n=telnette@cpe-71-74-91-116.insight.res.rr.com) |
| 19:49.28 | TrentCreek | Oh it was....It was a channel op that asked me about this "threat." Actually was not aking me but rather, "why are yoy threatening my friend" attitude |
| 19:50.08 | TrentCreek | anyway this was months ago |
| 19:50.15 | TrentCreek | well..back in May |
| 19:51.36 | TrentCreek | He was taking sides even though I could show my log, and he did not want to hear it so I told him off |
| 19:51.39 | Zuchmir2 | having trouble with SIP+DTMF |
| 19:52.08 | Zuchmir2 | when client presses keys (using X-Lite) not seeing it on server |
| 19:52.43 | *** join/#asterisk knipster (n=knipster@164.55.254.106) |
| 19:53.15 | Joel | TrentCreek this entire thing sounds extremely childish to me, sounds like you need to use ignore. |
| 19:53.15 | TrentCreek | bmoraca: dialparties.agi: Caller ID name is 'Name Unavailabl' number is '9999999999' |
| 19:53.26 | [TK]D-Fender | TrentCreek: Ah... from that chapter from "How to Not Win friends or Influence People (positively)" |
| 19:53.47 | bmoraca | TrentCreek: that doesn't tell me anything as it's out of context. |
| 19:53.48 | telnettech | have a question. I need to setup an operator option for people to be able to zero out of voicemail application. Where is that done in freepbx |
| 19:54.04 | [TK]D-Fender | telnettech: WRONG CHANNEL! |
| 19:54.06 | [TK]D-Fender | :p |
| 19:54.13 | [TK]D-Fender | telnettech: You already asked there, now sit on it! |
| 19:54.34 | TrentCreek | Joel: I did use ignore after I was insulted by this troll, and so days later he telling a channel op I was threatening him. |
| 19:54.43 | [TK]D-Fender | telnettech: And don't get all fidgety jsut because no-one answers in 5 minutes! |
| 19:54.56 | TrentCreek | bmoraca: you want to see the whole call progress? |
| 19:55.00 | telnettech | im sorry....i will show more patience |
| 19:55.06 | bmoraca | TrentCreek: that's what I would need, yes. |
| 19:57.50 | *** join/#asterisk ehsjoar (n=ehsjoar@c-24-9-91-203.hsd1.co.comcast.net) |
| 20:00.52 | *** join/#asterisk linu1 (n=winkelbr@g226176003.adsl.alicedsl.de) |
| 20:01.06 | bmoraca | TrentCreek: uhm, Dialparties isn't causing that. THat's the CID info it's getting from SIP/66.54.140.46 |
| 20:01.34 | bmoraca | or that's what CID is being forced from that inbound trunk |
| 20:01.48 | Zuchmir2 | any ideas why DTMF would not work w/SIP? (it works fine with PRI) |
| 20:01.48 | bmoraca | but that's happening before dialparties is executed |
| 20:02.01 | Joel | Zuchmir2 wrong dtmf mode |
| 20:02.18 | Joel | TrentCreek still sounds childish to me! |
| 20:02.22 | TrentCreek | bmoraca: This shows CID is correct, if I am not mistaken http://pastebin.com/m3f8c4899 |
| 20:03.00 | TrentCreek | Joel: well it is human nature to respond in kind when being attacked |
| 20:03.16 | bmoraca | TrentCreek: you're looking at two different calls. the one with the wrong CID is an inbound call, this call is an outbound call. |
| 20:03.32 | Zuchmir2 | joel: which dtmf should i be using? |
| 20:03.34 | Joel | TrentCreek your human nature, you mean. |
| 20:03.39 | Joel | Zuchmir2 which dtmf mode is your provider using? |
| 20:03.55 | TrentCreek | bmoraca: first case.. I called my cell phone...second case... I called a number pointing on my system, |
| 20:04.05 | Zuchmir2 | i'm trying to configure server against X-Lite |
| 20:04.19 | TrentCreek | Joel: so if someone come swinging a bat at you, you will just ignore them? |
| 20:04.26 | Joel | TrentCreek yup. |
| 20:04.32 | Joel | TrentCreek it's irc, not life, grow up. |
| 20:05.04 | *** join/#asterisk ingenius (n=alektro@186.136.6.218) |
| 20:05.17 | bmoraca | TrentCreek: in what circumstances, and on what phones, are you getting the incorrect callerid? |
| 20:05.19 | TrentCreek | well it is life, because I did talk to that channel op in real life on many occasions |
| 20:05.31 | Joel | like right now, I'm adding an ignore rule :D |
| 20:05.39 | Zuchmir2 | joel: in [general] i have dtmfmode=rfc2833 |
| 20:05.49 | TrentCreek | bmoraca: it seems all but, internal calls |
| 20:05.50 | [TK]D-Fender | TrentCreek: You like another shovel? |
| 20:05.53 | Joel | Zuchmir2 which dtmfmode does your provider use? |
| 20:05.55 | [TK]D-Fender | would* |
| 20:06.05 | TrentCreek | no need |
| 20:06.05 | Zuchmir2 | joel: i'm testing X-Lite |
| 20:06.12 | Joel | Zuchmir2 oh sorry, make sure it matches on both sides |
| 20:06.33 | [TK]D-Fender | TrentCreek: Also I don't see you debugging the actual call coming in... |
| 20:06.37 | bmoraca | TrentCreek: what does "all" mean? When you dial out, do you get that incorrect callerid on the destination phone? when someone dials in, do you get incorrect callerid? |
| 20:07.07 | TrentCreek | outgoing calls via trunk ... |
| 20:07.09 | Zuchmir2 | joel: i don't see that option in X-Lite |
| 20:07.23 | Joel | Zuchmir2 neither do I, you'll need to hit the x-lite website and research I guess |
| 20:07.34 | *** join/#asterisk nny_1 (n=Scott@64.203.244.146) |
| 20:07.54 | bmoraca | TrentCreek: That's the second call log you gave me...in which I do not see any wrong callerid information, meaning that if callerid info is wrong, it's because of SIP/5060 |
| 20:07.59 | [TK]D-Fender | Zuchmir2: it may not be an "option" in X-Lite. set * accordingly |
| 20:08.03 | *** part/#asterisk cervi (n=Iulia@89.39.205.57) |
| 20:08.20 | Zuchmir2 | joel: i have the exact same DTMF setting on another server, and X-Lite can do DTMF w/it |
| 20:08.35 | TrentCreek | bmoraca: Okay...I will check that part..thanks |
| 20:08.55 | nny_1 | noob question: Normally I can use the inbound number as an exten prefix like _XXXXXXXXXX,1,Goto(blah). On this one system everything dumps to s,1. Is this an issue with the way the telco is setup? (Using a series of FXO ports, I assume this isn't meant for such a setup) |
| 20:09.27 | [TK]D-Fender | nny_1: FXO = analog = "s" |
| 20:09.35 | nny_1 | [TK]D-Fender: roger |
| 20:09.39 | [TK]D-Fender | nny_1: there is no targeted number on analog |
| 20:09.39 | bmoraca | yep |
| 20:09.56 | nny_1 | heh too bad they share roll over lines with other internal companies.. sounds like an issue for the telco to change |
| 20:10.27 | bmoraca | nny_1: that's not going to work |
| 20:10.30 | [TK]D-Fender | nny_1: Sounds like the USER shouldn't be using analog any more |
| 20:14.11 | nny_1 | [TK]D-Fender: heh yeah. They either need dedicated rollover lines or a PRI. they only have like 6 lines, just called and found out the hunt group shares the extra lines :\ |
| 20:15.07 | TrentCreek | bmoraca: I just sent my provider a new ticket to verify their equipment. Thanks again, and [TK]D-Fender: |
| 20:15.13 | Zuchmir2 | tried dtmfmode=rfc2833, inband, info, and auto (with an asterisk -rx "sip reload" between each), none of them worked |
| 20:16.10 | Joel | Zuchmir2 some other problem maybe? |
| 20:16.28 | [TK]D-Fender | Zuchmir2: I don't see you showing us your configs or the problem. |
| 20:16.33 | [TK]D-Fender | ~pb |
| 20:16.34 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste , or , http://bin.cakephp.org/, or apt-get install pastebinit |
| 20:16.37 | [TK]D-Fender | ^^^^6 |
| 20:21.47 | Zuchmir2 | http://pastebin.com/d77afc78a |
| 20:21.54 | *** join/#asterisk Iamnach0 (i=Iamnacho@ip174-70-137-120.ks.ks.cox.net) |
| 20:24.47 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 20:24.55 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 20:25.52 | nny_1 | curses at Ubuntu |
| 20:26.08 | nny_1 | need to find a good solid distro that doesn't break everything trying to be helpful |
| 20:26.14 | asaleem | If I change the default sip port to 5061, how can I register x-lite using the thsi port? |
| 20:26.36 | asaleem | I need to change the port because 5060 is blocked by my ISP |
| 20:27.37 | Aiatek | asaleem change the default listen port of your xlite |
| 20:28.00 | asaleem | Aiatek, how? |
| 20:28.31 | asaleem | Aiatek, I did not find something related to that? |
| 20:28.55 | Aiatek | it has an option |
| 20:29.03 | Aiatek | Sip Listen port |
| 20:29.04 | asaleem | Aiatek, are the one under topolgy need to be changed? |
| 20:29.18 | nny_1 | asaleem: i think thats the rtp ports |
| 20:29.25 | nny_1 | asaleem: did you try domain:port? |
| 20:29.41 | asaleem | Aiatek, ohh, ok so I need to change the config file |
| 20:29.48 | asaleem | nny_1, I think so |
| 20:30.01 | asaleem | nny_1, yes I did |
| 20:30.11 | Aiatek | you asked how you connect your xlite if you change your sip default port |
| 20:30.12 | asaleem | nny_1, did not seem to work |
| 20:30.32 | Aiatek | i answered that |
| 20:30.41 | nny_1 | asaleem: there is suppose to be a network option, somewhere heh |
| 20:30.57 | nny_1 | http://74.125.47.132/search?q=cache:faXa1LKKsQ4J:www.pipecall.com/downloads/X-Lite%2520Softphone%2520Set-up%2520Guide.pdf+xlite+change+sip+port&cd=3&hl=en&ct=clnk&gl=us&client=firefox-a |
| 20:30.59 | asaleem | nny_1, there's none |
| 20:31.12 | nny_1 | asaleem: yeah updating mine to latest to see if I can find it |
| 20:31.17 | asaleem | nny_1, maybe, there is a config file, I will look for it |
| 20:31.29 | [TK]D-Fender | Zuchmir2: I don't see anything in there that requires DTMF, you aren't using rfc2833 as you were told it should be, you didn't set codecs for your peers. |
| 20:31.34 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.static.bezeqint.net) |
| 20:31.39 | nny_1 | asaleem: ahh |
| 20:31.45 | nny_1 | asaleem: click advanced at the bottom |
| 20:31.49 | nny_1 | asaleem: of the options menu |
| 20:31.51 | [TK]D-Fender | checkout time, BBIAB |
| 20:32.01 | asaleem | I will have to check it on my windows box |
| 20:32.12 | nny_1 | asaleem: yeah right click xlite -> options, advanced |
| 20:32.13 | *** join/#asterisk bbryant (n=brett@c-68-59-22-114.hsd1.sc.comcast.net) |
| 20:33.14 | asaleem | nny_1, let me check, I fired up my window vm |
| 20:33.28 | nny_1 | asaleem: yeah looking too, using older version |
| 20:33.58 | *** join/#asterisk blackest_mamba (n=blackest@c-71-239-160-143.hsd1.il.comcast.net) |
| 20:34.23 | nny_1 | asaleem: hmm not under network tab either :\ |
| 20:34.34 | asaleem | nny_1, oops |
| 20:36.00 | nny_1 | asaleem: dunno, each piece of info i find may or may not refer to an outdated version. Another says set the outbound proxy to domain:port, but that wouldn't do much for incoming |
| 20:36.22 | asaleem | nny_1, yep |
| 20:36.24 | nny_1 | although i guess that doesn't matter |
| 20:36.30 | nny_1 | er |
| 20:36.34 | nny_1 | for registering though it should |
| 20:36.38 | asaleem | did you find a config file for it? |
| 20:38.03 | nny_1 | asaleem: no |
| 20:38.26 | asaleem | nny_1, hmmm, no luck |
| 20:38.31 | nny_1 | asaleem: reading the fm right now |
| 20:38.40 | asaleem | nny_1, ok |
| 20:39.55 | *** join/#asterisk zeroHalo (n=zeroHalo@75.150.77.161) |
| 20:42.02 | nny_1 | asaleem: gah ha |
| 20:42.09 | nny_1 | asaleem: 3.0 manual is useless |
| 20:42.33 | nny_1 | asaleem: says you need the port* with an asterisk as if to say (read below, some kind of condition exists) but then nothing below :\ |
| 20:43.55 | nny_1 | asaleem: according to this support forum it's the port range on the topology page |
| 20:44.11 | nny_1 | If you have to specify the internal port (and you shouldn't) you can specify a range of 5060-5061 on the topology page of your SIP account. |
| 20:44.15 | nny_1 | from the forum |
| 20:44.18 | asaleem | nny_1, No idea how people do that with sip because firewalls are everywhere these day (fire is inexpensive,I guess to build walls from) |
| 20:44.33 | nny_1 | asaleem: feel free to try that, but just put 5061-5061 or w/e |
| 20:44.36 | nny_1 | asaleem: heh yeah |
| 20:44.52 | nny_1 | asaleem: i think asterisk tells the client which rtp ports to use after it registers, so that field makes sense |
| 20:45.03 | asaleem | nny_1, will give it a shot |
| 20:46.46 | bmoraca | I'd wager that if an ISP is going to go to the trouble of blocking SIP traffic on 5060, they'll probably block it on 5061, too |
| 20:46.46 | *** part/#asterisk nny_1 (n=Scott@64.203.244.146) |
| 20:47.13 | Zuchmir2 | <[TK]D-Fender>: i changed it to all ways in [general] section |
| 20:47.28 | Zuchmir2 | <[TK]D-Fender>: i left it by auto now |
| 20:49.48 | *** join/#asterisk vegbox (n=kevinle@wireless-169-235-59-82.ucr.edu) |
| 20:54.55 | *** join/#asterisk brian (n=brian@unaffiliated/brian) |
| 21:02.36 | *** join/#asterisk kn0x (n=pinochle@67.159.48.101) |
| 21:03.05 | kn0x | I keep getting a lot of HANGUPCAUSE 0's on SIP |
| 21:03.19 | kn0x | any idea how I can troubleshoot this? |
| 21:08.03 | RoyCrowder | anyone had a problem with the webvmail CGI having a file permissions problem? I've gone as far as 777 and chown'ing apache.apache still not working. |
| 21:08.09 | *** join/#asterisk Maximo (n=maximo@CPE001217b1920e-CM00111ade9528.cpe.net.cable.rogers.com) |
| 21:12.13 | Maximo | !hello |
| 21:20.05 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194) |
| 21:33.03 | Joel | RoyCrowder sounds like a non asterisk issue really... |
| 21:33.38 | Zuchmir2 | joel: any clues re: DTMF q. above ( http://pastebin.com/d77afc78a ) |
| 21:33.51 | *** join/#asterisk ingenius (n=alektro@host117.190-138-52.telecom.net.ar) |
| 21:37.01 | cusco | hi |
| 21:37.05 | cusco | no reply to our critical packet (see doc/sip-retransmit.txt) |
| 21:37.11 | cusco | Im trying to use sipp to test |
| 21:37.14 | cusco | what is missing? |
| 21:38.01 | [TK]D-Fender | Zuchmir2: I already told you 3 things that were wrong there, and I don't see you showing the PROBLEM |
| 21:38.29 | [TK]D-Fender | cusco: Networking issue (routing/firewall/NAT, etc) |
| 21:38.44 | kn0x | [TK]D-Fender: how can I troubleshoot HANGUPCAUSE 0 on SIP Dialing |
| 21:39.03 | [TK]D-Fender | kn0x: Look at the SIP debug and see whats really happening. |
| 21:39.36 | RoyCrowder | Joel: yea figured it out... was an SELinux issue.... |
| 21:43.28 | *** join/#asterisk xpot-mobile (n=james@144.35.20.74) |
| 21:45.07 | *** join/#asterisk ccesario (n=ccesario@200-160-106-130-ara.static.vivax.com.br) |
| 21:46.07 | Micc | Wouldn't it be possible to simulate multiple parking lots with ChannelRedirect? |
| 21:46.28 | Micc | I can't get multiple parking lots to work in 1.6.1.1 |
| 21:46.48 | Micc | So I figured I could do it myself with some dialplan magic. |
| 21:47.12 | [TK]D-Fender | Micc: that + dynamic conferences sure.... messy, but doable |
| 21:48.25 | Micc | TKD-Fender, why would I need dynamic conferences? Isn't there a way to just connect the two lines? |
| 21:48.47 | Micc | hmmm, I'm not sure how that work work I guess. |
| 21:49.16 | Micc | I could have it dial back the previous channel but thats not the same. |
| 21:49.19 | Zuchmir2 | <[TK]D-Fender>: http://pastebin.com/d6540d854 the problem is DTMF not working |
| 21:50.00 | [TK]D-Fender | Micc: the redirect only points to dialplan last I checked so there was no way to really target another device on pickup... |
| 21:50.20 | [TK]D-Fender | Zuchmir2: And I asked you to show a FAILED CALL |
| 21:50.50 | [TK]D-Fender | Micc: I suppose you could do a forced call-back method for it. |
| 21:50.52 | Micc | TKD-Fender, is there a way to bridge two channels? |
| 21:51.12 | [TK]D-Fender | Micc: Call exten that hangs up on you and then redirects to an exten that dials that same device back |
| 21:51.15 | Micc | TKD-Fender, but then the phone would ring. |
| 21:51.34 | Micc | hmm |
| 21:51.34 | [TK]D-Fender | Micc: 1.6's bridge might do it... never triend any of these methods personally. |
| 21:52.25 | Micc | TKD-Fender, I'll have to try that. Otherwise I think a dynamic conference would be ok. |
| 21:52.38 | Micc | Except last time I tried conferencing on my current server it crashed |
| 21:52.47 | Micc | But I think that was because of something else. |
| 21:53.03 | *** join/#asterisk ccesario_ (n=ccesario@189-92-7-41.3g.claro.net.br) |
| 21:53.23 | [TK]D-Fender | Micc: Doesn't 1.6.0 offer multiple lots? |
| 21:54.05 | kombi | on to the next issue: no sound.. Setup: xlite - dmz - internet - port forward(3478,4050,8000,8001) - asterisk. Must I forward 10000-20000 too? |
| 21:54.43 | [TK]D-Fender | kombi: Apparently you aren't forwarding SIP or IAX to your server either... |
| 21:54.48 | drmessano | <PROTECTED> |
| 21:54.54 | [TK]D-Fender | kombi: READ THE GUIDE |
| 21:54.56 | [TK]D-Fender | ~sipnat |
| 21:54.57 | infobot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 21:54.58 | drmessano | WTF are those ports? |
| 21:55.17 | [TK]D-Fender | drmessano: Twit(ter) ;) |
| 21:55.28 | drmessano | lol |
| 21:55.52 | kombi | will do..;) |
| 21:56.08 | Zuchmir2 | <[TK]D-Fender>: http://pastebin.com/d30bd37ce |
| 21:56.24 | Micc | TKD-Fender, yes it does offer multiple lots, but I can't figure out how to make it work. It seems broken to me. I can see the multiple lots, but using them only assigns from the main lot even though I have the correct parkinglot defined. |
| 21:56.35 | Micc | And the config settings don't seem to change the parkexten. |
| 21:56.37 | *** join/#asterisk CunningPike (n=CunningP@S01060014bf81366b.vc.shawcable.net) |
| 21:58.58 | [TK]D-Fender | Zuchmir2: I see SIP messages for DTMF, **no** RFC2833 which is carried over **RTP** |
| 21:59.18 | cusco | [TK]D-Fender: thanks that could be it as running from localhost resolved it |
| 21:59.30 | cusco | now its not getting to my extention - Unable to create channel of type 'SIP' (cause 20 - Unknown |
| 21:59.39 | cusco | what could it be ? |
| 21:59.51 | [TK]D-Fender | cusco: "sip show peer [thepeernamewithoutbraces]" |
| 22:00.25 | Zuchmir2 | <[TK]D-Fender>: so shouldn't dtmfmode=info work? |
| 22:00.52 | cusco | [TK]D-Fender: http://pastebin.com/m1ba3730 |
| 22:01.15 | [TK]D-Fender | Zuchmir2: You just showed me a config that specified rfc2833. I'm actually a little gray as to whether or not that is actually SIP INFO, or an "IM" |
| 22:01.47 | [TK]D-Fender | cusco: Addr->IP : (Unspecified) Port 5060 <--- your device has not registered and * has no idea how to contact it |
| 22:01.54 | cusco | let me check that |
| 22:02.46 | *** part/#asterisk zeroHalo (n=zeroHalo@75.150.77.161) |
| 22:03.55 | [TK]D-Fender | Zuchmir2: And I don't see at what point * should care what DTMF you enter. |
| 22:04.09 | kombi | fender: did everything as advertised, still no sound.. how do I best troubleshoot? |
| 22:04.32 | *** join/#asterisk SkramX (n=SkramX@cpe-70-112-25-138.austin.res.rr.com) |
| 22:05.26 | *** join/#asterisk gpled (n=gpled@firewall.fccfurn.com) |
| 22:06.02 | [TK]D-Fender | kombi: And the reason I should trust that is....? |
| 22:06.17 | kombi | from the manual: "Our * server will need the port range specified in rtp.conf forwarded to it (typically 10000 - 20000)" <- that being not the case might be the reason... |
| 22:06.43 | kombi | fender: pb coming up.. |
| 22:07.04 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 22:07.06 | *** part/#asterisk gpled (n=gpled@firewall.fccfurn.com) |
| 22:08.15 | *** join/#asterisk shazaum (n=98u89weu@unaffiliated/shazaum) |
| 22:08.43 | kombi | http://pastebin.se/pastebin.php |
| 22:08.54 | drmessano | ROFL |
| 22:08.55 | kombi | crap.. |
| 22:08.55 | [TK]D-Fender | ........ |
| 22:08.59 | kombi | wait.. |
| 22:09.08 | [TK]D-Fender | reaches for his ClueBat (tm) |
| 22:09.14 | drmessano | points and snickers |
| 22:11.00 | kombi | http://pastebin.se/198459 <- this time... |
| 22:11.49 | Zuchmir2 | <[TK]D-Fender>: http://pastebin.com/d757d2ad1 here's a phone call on a DID |
| 22:12.27 | drmessano | nat = no? |
| 22:13.01 | kombi | that's for the phones within the network |
| 22:13.02 | [TK]D-Fender | Zuchmir2: Found no matching peer or user for 'voip.providers.public.ip:5060' <--- your call isn't matching a peer so it uses the mode from [general] |
| 22:13.23 | [TK]D-Fender | kombi: Wheres the failed call? |
| 22:13.51 | kombi | fender: call goes through fine, just no sound, neither tx or rx |
| 22:13.56 | Zuchmir2 | <[TK]D-Fender>: the dialplan is a very complex one, what i pasted was the "entry", there's a whole gosub... |
| 22:13.58 | [TK]D-Fender | ..... |
| 22:14.09 | drmessano | If youre going to misuse a directive, at least misuse it properly and use "never", damnit |
| 22:14.30 | [TK]D-Fender | Zuchmir2: Nothing in what you PB'd before showed me that * cared about DTMF. Forget about receiving it... I don't see * CARING. |
| 22:14.36 | kombi | drmessano: was that for me? |
| 22:14.43 | [TK]D-Fender | BRB |
| 22:14.58 | drmessano | kombi: Youre a little slow, i take it? |
| 22:15.45 | Zuchmir2 | <[TK]D-Fender>: why would it not care? |
| 22:16.41 | kombi | drmessano: hmm.. |
| 22:17.14 | drmessano | kombi: Do you have SIP and RTP ports open or not? |
| 22:17.17 | *** join/#asterisk SkramX (n=SkramX@cpe-70-112-25-138.austin.res.rr.com) |
| 22:19.02 | kombi | drmessano: might have mentioned it, got udp 3478, 5040, 8000 and 8001 open, but not 10000 - 20000. Is that simply it? |
| 22:19.33 | [TK]D-Fender | Zuchmir2: I don't see any commands being issued that listen for DTMF and act upon it |
| 22:19.43 | drmessano | WTF are 3478, 5040, 8000, and 8001????????? |
| 22:19.55 | drmessano | SIP is 5060 and RTP is 10000-20000 by default |
| 22:20.02 | drmessano | Go forward those ports |
| 22:21.01 | kombi | drmessano: ok.. |
| 22:21.45 | drmessano | ~sipnat |
| 22:21.46 | infobot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 22:22.08 | drmessano | Our Asterisk server will also have to have ports 5060 (UDP), and the port range specified in ârtp.confâ (typically 10000-20000 UDP) forwarded to it. |
| 22:22.17 | Zuchmir2 | <[TK]D-Fender>: the line saying: Playing '/voice/cvt/menu/brooklyn/langmenu' is from a custom app which calls ast_control_streamfile() |
| 22:22.27 | drmessano | ^^^^^^^^^^^^^^ |
| 22:22.54 | kombi | ok,ok... i'll do it,) |
| 22:23.25 | [TK]D-Fender | Zuchmir2: Tell you what. Do a real test with normal apps who's actions I can trust. DON'T mask anything except passwords, and maybe we'll get somewhere |
| 22:24.49 | *** join/#asterisk DeVilSoulBlacK (n=aandaluz@srv.ec-gye.internet.geainternacional.com) |
| 22:25.21 | kombi | I think x-lite pointed me to those other weired ports... isp won't let me change a thing before tomorrow though, off to the grill then. Thanks people! |
| 22:25.24 | DeVilSoulBlacK | hi any one know where can get download this type of graphic for asterisk http://www.elastix.org/images/stories/screenshots/reports/channel_usage.png |
| 22:26.10 | Joel | DeVilSoulBlacK some sort of graphing package, personally I like using ofc2 for my charting needs |
| 22:26.15 | Joel | DeVilSoulBlacK look at the source code. |
| 22:27.08 | DeVilSoulBlacK | i dont use elastix , but i see interest that type of graphic |
| 22:28.08 | DeVilSoulBlacK | Joel: what ist ofc2 ?? |
| 22:28.27 | DeVilSoulBlacK | Open Flash Chart ? |
| 22:28.28 | [TK]D-Fender | kombi: ISP? What does your ISP have to do with this? |
| 22:30.45 | drmessano | [TK]D-Fender: Who cares? |
| 22:34.32 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 22:37.53 | Zuchmir2 | <[TK]D-Fender>: http://pastebin.com/d1222f969 |
| 22:40.39 | [TK]D-Fender | Zuchmir2 : New configs please... |
| 22:43.52 | Zuchmir2 | <[TK]D-Fender>: http://pastebin.com/d8d1d2ad |
| 22:52.03 | *** join/#asterisk double_cheesebur (n=chatzill@ip68-98-36-4.ph.ph.cox.net) |
| 22:52.56 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 22:53.25 | double_cheesebur | I just came across documentation that states the asterisk sms command only works in 6 countries...of which the US is not included. Can anyone recommend documentation that speaks to Asterisk SMS capabilities that have been developed for US markets? |
| 22:54.47 | *** join/#asterisk coppice (n=chatzill@204.196.17.210.dyn.pacific.net.hk) |
| 22:56.05 | [TK]D-Fender | Zuchmir2: Ok, I'm not sure whats going on at this point... what is the call coming in from? |
| 22:57.23 | Zuchmir2 | POTS |
| 22:58.07 | Zuchmir2 | i recentrly installed the g729 codec, and that's when the trouble began :-( |
| 23:00.53 | *** join/#asterisk shazaum (n=98u89weu@unaffiliated/shazaum) |
| 23:03.30 | [TK]D-Fender | Zuchmir2: ..no, what SIP service is sending you that call.... |
| 23:06.10 | SkramX | what app (if any) do you all use to map out db model structures and/or call flow? :) thanks |
| 23:06.34 | WindowsUser | visio? ^_^ |
| 23:06.43 | SkramX | im on a mac :\ |
| 23:06.49 | SkramX | but I have heard visio is pretty swell |
| 23:07.07 | WindowsUser | iWork should have something |
| 23:07.13 | [TK]D-Fender | SkramX: Dia |
| 23:07.14 | WindowsUser | iDiagramtool |
| 23:07.17 | coppice | visio used to be good. the MS bought it :-( |
| 23:07.51 | WindowsUser | microsoft needs to be more secretive about buying people |
| 23:08.03 | SkramX | ...no |
| 23:08.04 | WindowsUser | like buy them but dont let them know they've been bougt |
| 23:08.13 | WindowsUser | double secret probation |
| 23:08.15 | coppice | the need to stop wrecking what they bought |
| 23:09.20 | WindowsUser | brb, might be awhile if i killed this ubuntu install finally |
| 23:18.44 | *** join/#asterisk De_Mon (i=de_mon@fl-71-55-238-9.dhcp.embarqhsd.net) |
| 23:24.17 | *** join/#asterisk s14ck (n=s14ck@ccscliente156.ifxnetworks.net.ve) |
| 23:32.49 | Zuchmir2 | <[TK]D-Fender>: newtel |
| 23:34.55 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 23:53.33 | *** join/#asterisk hfb (n=hfb@pool-98-112-239-34.lsanca.dsl-w.verizon.net) |
