| 00:02.17 | SwK_ | anyone have any 1 or 2 port t1 cards they wanna part with cheap? |
| 00:04.15 | *** join/#asterisk idefine (n=panaii@lawn-128-61-19-228.lawn.gatech.edu) |
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| 00:09.44 | BBHoss | SwK_, whats cheap? |
| 00:09.59 | BBHoss | i have a 2 port T1 card (Digium) |
| 00:11.16 | BBHoss | brand new, how does $800 sound? |
| 00:11.58 | *** join/#asterisk magic_hat (n=geoffdou@h-74-2-87-16.chcgilgm.covad.net) |
| 00:12.17 | *** join/#asterisk asdx (n=diego@adsl-145-110.click.com.py) |
| 00:12.45 | magic_hat | I'm looking at buying a new 'puter to serve as our * box. Anyone have recs on chip, RAM, HD size and Linux flavor? |
| 00:13.06 | magic_hat | we've got an office with 10-15 softphones, up to 5 people talking at the same time. |
| 00:13.36 | SwK_ | magic_hat, dood get a dell |
| 00:14.01 | fujin | magic_hat: dell 2950 |
| 00:14.04 | fujin | Ubu |
| 00:14.05 | fujin | or db |
| 00:14.06 | SwK_ | like a Dell SC440 will cost you ~400 shipped 80G HDD, 1G Ram, and have fun... |
| 00:14.06 | fujin | deb |
| 00:14.15 | SwK_ | 2950 is over kill for what he's doign |
| 00:14.16 | fujin | get n+1 servers |
| 00:14.26 | fujin | 5 people at the same time |
| 00:14.27 | fujin | hrmg |
| 00:14.40 | SwK_ | a 2950 can do like 150 concurrent calls |
| 00:14.55 | SwK_ | you get it with the dual quad cores and it can do way more then that |
| 00:15.07 | fujin | lol aye |
| 00:15.13 | fujin | I've got two HP DL360's |
| 00:15.18 | fujin | they were purchased before I arrived |
| 00:15.23 | fujin | I would have preferred Dellkit |
| 00:15.28 | Qwell | only 150? You'll get far more than that |
| 00:16.09 | Qwell | that isn't even 8 T1s |
| 00:16.13 | fujin | ;P |
| 00:16.35 | magic_hat | hrm.... so SC440 sounds good. |
| 00:16.38 | magic_hat | cheep! |
| 00:16.39 | magic_hat | lol |
| 00:17.10 | magic_hat | and 'nix flavor? is the panel united on Debian? |
| 00:17.24 | fujin | ubu or deb, imho |
| 00:17.31 | fujin | although I build * from source wherever I go |
| 00:17.33 | fujin | generally :} |
| 00:17.44 | fujin | whatever you're comfortable in |
| 00:18.29 | SwK_ | use whatever linux you like it does matter |
| 00:18.37 | SwK_ | Centos, Deb, Gentoo whatever |
| 00:20.13 | magic_hat | sweet. sc440 comes with Red Hat. |
| 00:20.19 | fujin | yuck |
| 00:21.12 | SwK_ | magic_hat, d00d order w/ no OS |
| 00:21.27 | SwK_ | you'll save some cash and then just drop a Centos ISO in it |
| 00:22.05 | BBHoss | magic_hat, debian is a safe bet |
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| 00:24.25 | magic_hat | Yeah, I'm just realizing the RHEL is $200 extra. No way I pay for that. |
| 00:24.33 | SwK_ | heh for real |
| 00:25.02 | SwK_ | they have a T105 w/ a dual-core opteron right now 1G ram and a 80G hdd for like 375 |
| 00:25.32 | *** join/#asterisk d00gster (n=doughant@bas1-toronto12-1128684591.dsl.bell.ca) |
| 00:26.06 | blitzrage | CentOS w00t |
| 00:27.36 | SwK_ | leif loves the redhat |
| 00:27.47 | blitzrage | :) |
| 00:27.55 | SwK_ | whats up mang |
| 00:28.05 | SwK_ | blitzrage, was talking about you on saturday |
| 00:28.07 | blitzrage | nada much bud.. just got back from the gymnasium |
| 00:28.17 | blitzrage | oh ya? |
| 00:28.36 | SwK_ | yeah ran into russel and kevin at the pool hall |
| 00:29.51 | craigk | does anybody know if i can use the management interface to redirect a call that is on hold ? So I use a sip phone to place a call on hold, then <insert magic here> redirect the held call ? |
| 00:30.08 | craigk | i just want to know if it is possible, not how to do it :) |
| 00:30.23 | SwK_ | craigk, that should be possible |
| 00:30.41 | craigk | SwK_: thanks |
| 00:30.45 | SwK_ | there might actually be something for that already in the AMI code |
| 00:31.08 | craigk | there is a redirect command there ... i just was not sure it would work on a held call |
| 00:31.18 | SwK_ | hell try it :P |
| 00:31.23 | SwK_ | all it can do is fail |
| 00:31.25 | craigk | my concern is that the call will be left on hold at the new extension |
| 00:31.58 | craigk | true ... maybe that will be a weekend project :) |
| 00:32.10 | craigk | man i know how to party ;) |
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| 00:38.06 | generalhan | can anyone point me to some docs on the best way to get 2 remote phones behind the same router to connect to my local * machine ? |
| 00:48.38 | idefine | is anyone familiar with the floor control mchanism for IMS Pish to Talk Over Cellular (PoC) |
| 00:48.54 | idefine | mechanism*, Push* |
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| 01:07.02 | webman | anyone noticed that current asterisk 1.4 SVN keeps core dumping within 30 minutes of starting up on our extremely lightly loaded system ? |
| 01:07.16 | fujin | no, I can't say I've noticed that on your system. |
| 01:08.32 | webman | fujin: errr, have you noticed it on your system ? |
| 01:08.42 | BBHoss | heh |
| 01:09.30 | webman | maybe I should have started by asking if anyone is using an 1.4svn version from the last 24 hours ? |
| 01:09.33 | BBHoss | u using trunk or a branch |
| 01:10.13 | blitzrage | lol |
| 01:10.13 | JT | BBHoss: 1.4svn |
| 01:10.23 | JT | he's said it twice |
| 01:10.26 | blitzrage | webman: sorry, I've been using trunk, not the 1.4 branch |
| 01:11.06 | BBHoss | damn, brainfart |
| 01:11.53 | blitzrage | its ok -- just be thankful you're not the girl I just talked to on cable... now there's a permanent brainfart |
| 01:12.27 | blitzrage | took forever to just add a channel to my cable... the problem is she was SOOOOOOOOOOOO nice and trying to be so helpful, but she was talking to me like I was her 80 y/o grandmother |
| 01:12.52 | BBHoss | heh |
| 01:12.54 | blitzrage | I was laughing by the time I hung up, so it definitely had its entertainment value |
| 01:13.08 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 01:13.17 | webman | ~pastebin |
| 01:13.18 | jbot | [~pastebin] 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 |
| 01:14.02 | webman | http://pastebin.com/d34feb860 |
| 01:14.43 | webman | does that look like an asterisk problem, or is it saying it is a problem in the libc library (which probably means my hardware error) |
| 01:15.56 | tzafrir | Astrisk passed a bad pointer to libc |
| 01:16.27 | webman | tzafrir: so it is more likely a asterisk problem ? |
| 01:16.35 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 01:16.35 | *** mode/#asterisk [+o anthm] by ChanServ |
| 01:16.36 | tzafrir | yes |
| 01:18.02 | jer | any way i can use GotoIfTime() to specify multiple times? that is, if i want to skip to a priority between 17:00 and 17:30, and also between 07:30 and 08:00 to the same priority without having to have two GotoIfTime() statements? |
| 01:19.25 | *** join/#asterisk mosty (n=mostyn@ppp191-34.static.internode.on.net) |
| 01:19.30 | blitzrage | jer: you need two statements for that |
| 01:19.41 | nitrus | what is a sip response 500? |
| 01:19.46 | nitrus | it that a generic error like http |
| 01:19.48 | blitzrage | hrmm... actually |
| 01:20.22 | jer | blitzrage, damn, that complicates things greatly |
| 01:20.54 | blitzrage | GotoIfTime(0730-0800&1700-1370,,,,,) |
| 01:21.00 | blitzrage | I'm curious if that works |
| 01:21.04 | jer | hrmm i'll give it a try |
| 01:21.11 | *** join/#asterisk icel (n=icel@63.78.162.72) |
| 01:22.20 | jer | blitzrage, it does! great |
| 01:22.25 | blitzrage | hawtness |
| 01:23.31 | mosty | is there a dialplan function that is the inverse of strftime? |
| 01:23.53 | icel | question: I recently switched hardware for my asterisk server and I am using a digium 205p instead of a 405p. When * starts it works find locally, but T1 doesn't come up. After running ztcfg -vvv and restarting asterisk a couple of times the T1 starts working. Anyone know what is up with that? |
| 01:25.55 | webman | icel: you need to modprobe the module for your card, and run ztcfg before you start asterisk |
| 01:26.37 | mosty | i want to do the equivalent of "date -d 09:00 +%s" in the dialplan, is there a way to do it without agi or similar? |
| 01:26.57 | webman | I'm trying to log a bug for my core dump, but am not sure how to describe it, or even how to attempt to reproduce it.... any suggestions ? |
| 01:27.11 | webman | mosty: system application ? |
| 01:28.58 | mosty | webman, how do i get stdout from the command? |
| 01:29.32 | webman | mosty: can you set a variable perhaps? I've never used system to do anything, just know it exists... maybe look for some example in the wiki ? |
| 01:30.07 | nitrus | anyone here know what a sip response 500 "nice try" means? |
| 01:30.24 | jer | blitzrage, correction, it doesn't quite work |
| 01:30.34 | mosty | webman, if it was a function i'd be sorted, but since it's a dialplan application i'd have to write to a file or somethign |
| 01:30.44 | blitzrage | jer: no? I was guessing on syntax that might work... what is wrong with it? |
| 01:32.13 | jer | blitzrage, it plays always. i had tested with using two times, one 00:00-00:01&20:20-20:30 (it was 20:21 when i tested); played the message i wanted it to then. when i changedi t to the real times (07:30-07:59&17:00-17:29) tested again, it still played that message |
| 01:32.18 | jer | not sure why |
| 01:32.20 | jer | but it did |
| 01:32.48 | blitzrage | maybe you don't need the : |
| 01:33.06 | jer | hrmm, ok i'll try it without |
| 01:33.08 | mosty | webman, ahh, i wish STRPTIME was in asterisk 1.2 |
| 01:33.48 | jer | blitzrage, nope, still played it. i can post the relevant section of the dialplan if you'd like |
| 01:34.04 | blitzrage | jer: sure... its probably just invalid syntax |
| 01:34.10 | jer | ok |
| 01:34.13 | blitzrage | or I'm not sure what the syntax is |
| 01:34.38 | jer | blitzrage, http://pastebin.com/m556566a1 |
| 01:34.47 | jer | it always plays my "on the road" message |
| 01:34.49 | blitzrage | maybe Corydon76-dig knows |
| 01:35.52 | *** join/#asterisk techie (n=techie@adsl-76-214-17-132.dsl.lsan03.sbcglobal.net) |
| 01:37.09 | jer | oops, error there too on the second gotoiftime... but it's not being caught atm |
| 01:37.25 | blitzrage | might need to use two lines |
| 01:37.35 | jer | yeah, i probably will; was just hoping there was a way around that =] |
| 01:37.43 | icel | webman: thanks. I put the wct4xxp module to load at boot time |
| 01:37.49 | jer | my ivr config is horrible |
| 01:38.01 | icel | webman: but how do i make ztcfg run at the same time? |
| 01:38.04 | blitzrage | jer: first of all -- stop using priority numbers |
| 01:38.14 | jer | just use 'n' ? |
| 01:38.17 | blitzrage | yes |
| 01:38.20 | blitzrage | and labels |
| 01:38.28 | jer | ok |
| 01:38.36 | blitzrage | 100,1,NoOp(always need prio 1) |
| 01:38.47 | jer | right |
| 01:38.50 | blitzrage | 100,n(loop),Verbose(1,Well hello there!) |
| 01:38.54 | blitzrage | 100,n,Goto(loop) |
| 01:38.59 | *** join/#asterisk JayTee52 (n=jforde05@c-69-137-243-25.hsd1.in.comcast.net) |
| 01:39.08 | blitzrage | that is BAD |
| 01:39.20 | blitzrage | (infinite loop)_, but its only 3 lines to show you how to use a priority label |
| 01:39.27 | icel | webman: nevermind, i see the init script. Thanks! |
| 01:39.40 | *** join/#asterisk quah (i=kilgore_@cpe-075-181-049-064.carolina.res.rr.com) |
| 01:39.57 | jer | blitzrage, right, i appreciate it. what about jumps? i.e., n + 101 ? |
| 01:40.00 | blitzrage | icel: using a Fedora Core or some other RH based distro? Do 'make config' in the asterisk source directory |
| 01:40.08 | jer | just, 100,n+101,... ? |
| 01:40.38 | icel | blitzrage: using gentoo, fedora and cent os kept locking up on this box for some reason...I am trying to edit the zaptel init.d script now so it will run |
| 01:40.41 | blitzrage | jer: you don't use that logic anymore. Most applications returns a "STATUS" variable. i.e. Dial() uses ${DIALSTATUS}, then you can use that to direct calls based on status |
| 01:40.53 | jer | ah |
| 01:40.56 | blitzrage | jer: yes, that logic works too |
| 01:41.04 | blitzrage | if you are converting a dialplan |
| 01:41.19 | blitzrage | or label+101 works too |
| 01:41.28 | jer | blitzrage, i wrote this fresh a few days ago, and it's just for my home office, so i'm not too worried about changing it =] |
| 01:41.45 | blitzrage | so when you are not on the very first priority, you can offset the +101 |
| 01:41.56 | jer | you'd recommend using the dial status? |
| 01:42.03 | blitzrage | but +101 logic is easy to get dialplan that jumps to places you don't expect |
| 01:42.13 | blitzrage | always use the STATUS variable when you can |
| 01:42.19 | jer | in this case, i can |
| 01:42.28 | blitzrage | infact, +101 is deprecated, in 1.4, and removed in 1.6 |
| 01:42.41 | jer | ah |
| 01:43.20 | quah | hi - i've a new asterisk running on Centos 5, asterisk 1.4.10 - with a Rhino R8FXX card for incoming pots lines..in the past 2 weeks the box stopped seeing incoming calls 2 times - needing a reboot to start working again. Can anyone give me some adive on how to start troubleshooting this? |
| 01:43.24 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 01:43.24 | *** mode/#asterisk [+o russellb] by ChanServ |
| 01:43.32 | blitzrage | anywhere a +101 was required, the status can be found in a variable. See the doc/variables.txt file for info |
| 01:43.44 | jer | will do |
| 01:43.53 | blitzrage | quah: why is a new asterisk running 1.4.10 when 1.4.15 is out? |
| 01:44.12 | quah | because I didn't want to change what "seemed" to be working <G> |
| 01:44.47 | blitzrage | keep the code base, install 1.4.15 to see if it fixes the problem. You can always just reinstall the copy you have now with 'make install' |
| 01:45.01 | blitzrage | just make sure you keep the directory with the source currently installed as a backup |
| 01:45.28 | blitzrage | jbot: tell jer about book |
| 01:45.43 | *** join/#asterisk waverly360 (n=waverly@ns2.dalcon.com) |
| 01:45.44 | jer | blitzrage, yup, i'm flipping through it right now =] |
| 01:45.58 | jer | it's quite a bit, been slowly going through it |
| 01:46.13 | quah | ok..along those same line - is there any way I can "notice" that it is not working - for example, this system is in a pizza shop - customers might be calling and only getting ringing - people working the shop dont even know anything is wrong. |
| 01:46.37 | waverly360 | Hey guys, this may be beyond the scope of this room, but is there a way to create a tc filter that filters on multiple ports instead of a single one? |
| 01:47.23 | idefine | does anyone know of any company providing Push and Talk Over Cellular (PoC)...and what the rates are? |
| 01:48.32 | *** join/#asterisk |omni| (n=rob@c-67-185-144-206.hsd1.wa.comcast.net) |
| 01:56.36 | webman | well, it looks like I can't use 1.4svn because it crashes on every call shortly after it comes in.... |
| 01:56.47 | fujin | why run svn? |
| 01:57.00 | fujin | have you filed a bugreport with your coredump? |
| 01:57.09 | webman | idefine: if push and talk is the same as push to talk then www.optus.com.au does it :) |
| 01:57.28 | fujin | idefine: telecom NZ does it, so does vodafone NZ |
| 01:57.45 | fujin | I believe you pay a flatrate, which gives you a minute quota |
| 01:57.48 | webman | fujin: because 1.4svn is meant to be the version for production use, it is meant to have all the bug fixes with no new bugs introduced.... |
| 01:58.01 | fujin | webman: no, trunk is meant to be the production use one |
| 01:58.11 | fujin | svn is where the development on 1.4 is done |
| 01:58.15 | webman | idefine: optus used to pay a monthly fee, and then no extra |
| 01:58.17 | file | what? no |
| 01:58.21 | fujin | other way around? |
| 01:58.27 | fujin | trunk/branhc |
| 01:58.29 | fujin | fucked if I know |
| 01:58.35 | file | 1.4 branch in SVN receives bug fixes only, but we are human |
| 01:58.38 | webman | trunk is development for 1.6 |
| 01:58.46 | fujin | oic |
| 01:58.51 | fujin | webman: do you experience the issue with 1.4.15? |
| 01:58.56 | JayTee52 | which is the stable version of 1.4? |
| 01:59.03 | webman | file: yeah, of course, I understand bugs can happen :( |
| 01:59.09 | idefine | webman: nice, thanks. Do you know of any companies in the USA offering this service? |
| 01:59.25 | webman | fujin: haven't tried that yet .... that is next on my list... |
| 01:59.41 | webman | idefine: dunno... never looked ... |
| 01:59.53 | fujin | are you running the latest svn? when was your last svn up? |
| 02:00.12 | webman | latest svn update was about 2 or 3 hours ago now.... |
| 02:00.28 | fujin | & did you file a bugreport? |
| 02:00.49 | webman | http://bugs.digium.com/view.php?id=11486 |
| 02:01.58 | russellb | webman: i'll take a quick look ... |
| 02:02.39 | fujin | there ya go |
| 02:02.40 | webman | current svn (10 secs ago) just has a patch for rtp.c .... so I don't think that will fix it.... |
| 02:02.50 | fujin | many heart <3 russellb |
| 02:03.14 | webman | russellb: thank you |
| 02:03.21 | russellb | if it happens every time, it shouldn't be too hard to spot |
| 02:03.22 | russellb | ... in theory |
| 02:03.30 | russellb | give me a few mintues to dig ... |
| 02:03.52 | fujin | webman: you might want to re-build Asterisk, wit the optimizations disabled |
| 02:03.57 | fujin | so that the coredump is a little better |
| 02:04.40 | russellb | webman: can you execute a couple gdb commands for me? |
| 02:04.43 | webman | that is the "DONT_OPTIMIZE" flag right ? any others I should do ? |
| 02:04.49 | webman | russellb: yeah sure |
| 02:05.00 | fujin | webman: forget what I said, let russell guide your hand |
| 02:05.04 | russellb | webman: (gdb) frame 5 ...... (gdb) p di |
| 02:06.07 | webman | russellb: added to the bug/ticket |
| 02:06.38 | webman | that was using the core file from the last report I added if that makes a difference |
| 02:06.41 | fujin | definitely looks like you'll want to recompile without optimisations |
| 02:06.48 | fujin | to provide a little more useful information there |
| 02:07.13 | russellb | yeah, what fujin said |
| 02:07.19 | russellb | if you don't mind ... |
| 02:07.25 | webman | ok, compile is in progress.... |
| 02:07.29 | russellb | yay |
| 02:07.48 | russellb | you're lucky, i have some time to kill in between flights :-p |
| 02:08.34 | webman | my star sign said "I could bend reality today" ... I just wish it didn't mean I could break asterisk :) |
| 02:09.09 | russellb | ha |
| 02:09.13 | russellb | you didn't break it ... we did |
| 02:12.09 | webman | russellb: ok, got a core with dont_optimize, do you want all the same commands from gdb ? |
| 02:12.54 | russellb | yes please |
| 02:13.44 | JayTee52 | I just got a new server that came with RHEL 5. I want to install Asterisk 1.4.? on it. Is there any reason why I should avoid using RPM's? |
| 02:13.55 | webman | ok, added to bug report |
| 02:14.14 | russellb | perfect! |
| 02:14.15 | [TK]D-Fender | JayTee52: Do * from source. RPMS are dated and zaptel needs to match your kernel anyways |
| 02:14.19 | russellb | will fix in a second |
| 02:14.31 | JayTee52 | [TK]D-Fender, thanks man! |
| 02:16.31 | JayTee52 | [TK]D-Fender, what's the latest stable 1.4.* release? |
| 02:16.41 | russellb | JayTee52: see the topic :) |
| 02:19.03 | russellb | webman: svn update to revision ... *waits for the commit to finish* ... |
| 02:19.09 | russellb | 91675 |
| 02:22.40 | russellb | webman: do you use Local channels as members in your queue? |
| 02:22.51 | fujin | russellb, webman: good work |
| 02:23.00 | russellb | thanks |
| 02:23.32 | webman | russellb: yes |
| 02:23.56 | webman | I've downloaded the new version, am just waiting for the compile to finish... the only update was app_queue ? |
| 02:24.01 | russellb | yes |
| 02:24.09 | russellb | however, i'm curious what happens after this update |
| 02:24.24 | russellb | i'm reading the part of the code i just fixed and it looks like it breaks Local channel queue members, too ... |
| 02:24.30 | russellb | makes it so they won't ever get called |
| 02:24.36 | *** join/#asterisk tengulre (n=tengulre@125.71.208.16) |
| 02:24.37 | russellb | but if you don't mind, just try this and let me know what happens |
| 02:24.40 | mosty | can i do goto from an agi script followed by exit to jump to another context? |
| 02:25.00 | mosty | or do i need to set a variable and do a Goto in the dialplan? |
| 02:25.49 | webman | russellb: does that mean I should have left the DONT_OPTIMIZE on ? |
| 02:25.57 | russellb | nope |
| 02:25.59 | russellb | it won't crash anymore |
| 02:26.54 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 02:27.28 | webman | ok, still compiling... will let you know shortly |
| 02:32.12 | webman | russellb: ok, the caller is in the queue, but it isn't attempting to call any of the queue members (all local channels) |
| 02:33.02 | webman | did you want any particular information ? |
| 02:33.24 | russellb | nope, i know what the problem is |
| 02:33.26 | russellb | i will fix it in a minute |
| 02:33.34 | russellb | talking on the phone to the guy that wrote that part |
| 02:33.52 | *** join/#asterisk bkw__ (n=brian@adsl-64-149-47-251.dsl.tul2ok.sbcglobal.net) |
| 02:33.53 | webman | ok, thanx |
| 02:37.35 | *** join/#asterisk outtolunc (n=me@c-67-174-216-60.hsd1.ca.comcast.net) |
| 02:37.49 | russellb | webman: 91677 |
| 02:37.59 | *** part/#asterisk bkw__ (n=brian@adsl-64-149-47-251.dsl.tul2ok.sbcglobal.net) |
| 02:38.34 | webman | russellb: downloading now... will let you know shortly. ... thank you again! |
| 02:38.43 | *** join/#asterisk Kumbang (n=dsp@rusnas.paume.ITB.ac.id) |
| 02:38.43 | russellb | you're welcome |
| 02:38.48 | russellb | thanks for the helpful debugging and testing |
| 02:39.28 | russellb | all is good in the world again, i think |
| 02:46.11 | *** join/#asterisk shtoom (n=shtoom@59.93.121.201) |
| 02:46.50 | webman | russellb: now I can't dial in at all, the zap lines are 'busy' and "zap show channels" returns no output at all ..... |
| 02:47.14 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
| 02:47.43 | russellb | argh! |
| 02:47.57 | russellb | :( |
| 02:48.08 | russellb | do any cli commands return output? |
| 02:48.52 | webman | I killed it, (kill -9) and restarted, and now it seems to work .... |
| 02:49.04 | russellb | weird. |
| 02:49.11 | russellb | stupid open source software ... |
| 02:49.47 | webman | I think I might have called in too quickly - before it was fully started or something.... |
| 02:50.35 | webman | I have now called, the call was added to the queue, and rang my sip extension, I answered, and talked to myself... seems like "all is good in the world again" :) |
| 02:50.44 | russellb | yay! |
| 02:50.55 | russellb | that will be $1000. |
| 02:50.55 | russellb | kthx! |
| 02:51.23 | webman | thanks again for your help - $1000 .... hmmm :) |
| 02:51.28 | russellb | j/k :-p |
| 02:51.32 | russellb | you're welcome |
| 02:51.35 | russellb | 'tis my job |
| 02:51.57 | webman | yeah, but must be kinda outside office hours wherever you are .... |
| 02:53.07 | russellb | yeah ... |
| 02:53.15 | russellb | but i'm addicted to asterisk :-/ |
| 02:53.46 | webman | now I gotta get back to ... my work :) |
| 02:54.01 | russellb | alrighty, glad we got your system fixed up. |
| 02:54.23 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 02:56.48 | blitzrage | wow... asterisk just crashed my vmware fusion :) |
| 02:57.10 | russellb | nice. |
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| 02:57.23 | russellb | blitzrage: i fixed iax2 on mac btw |
| 02:57.32 | blitzrage | russellb: it was broked? |
| 02:57.42 | russellb | yeah, i remember you mentioning it on irc one day |
| 02:57.46 | russellb | ... i think |
| 02:58.00 | blitzrage | I think I mremeber that now... |
| 02:58.13 | blitzrage | this is why I don't use iax2... every time I try, I find a bug :) |
| 02:58.16 | russellb | ha |
| 02:58.31 | blitzrage | I find some obscure ones sometimes... |
| 02:58.35 | russellb | you're good at that. |
| 02:58.43 | tzanger | you're obscure |
| 02:58.45 | blitzrage | I'm doing a CLI audit right now |
| 02:58.48 | russellb | yay |
| 02:59.09 | blitzrage | hopefully we'll end up with a solid cli syntax out of this |
| 02:59.15 | russellb | i sure as hell hope so |
| 02:59.19 | russellb | we've gone this far ... |
| 02:59.22 | blitzrage | this is my goal... |
| 02:59.30 | russellb | while we still have everyone pissed off, might as well finish the job :-p |
| 02:59.37 | blitzrage | and the code dealing with the cli commands can't be that difficult can it? |
| 02:59.41 | blitzrage | probably just tedious... |
| 02:59.45 | russellb | tedious, yes |
| 02:59.47 | russellb | but not difficult |
| 02:59.54 | russellb | and i would be happy to answer any questions you have |
| 02:59.56 | blitzrage | I'm good at tedious |
| 02:59.58 | russellb | as would a number of other people .. |
| 03:00.11 | russellb | heh |
| 03:00.23 | russellb | welcome to the asterisk janitor team! |
| 03:00.27 | blitzrage | cool.. well, first step is the audit |
| 03:00.29 | blitzrage | :) |
| 03:01.05 | tzanger | hmm |
| 03:01.07 | tzanger | you are tedious too |
| 03:01.18 | tzanger | how many other adjectives can you attribute to asterisk that you personally reflect? |
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| 04:04.32 | tzanger | heh |
| 04:05.39 | blitzrage | goooooooo freenode |
| 04:05.58 | blitzrage | up to ]q]! |
| 04:06.03 | blitzrage | errr.. 'q' |
| 04:08.04 | [TK]D-Fender | tzanger: Nope, can't say that I have... |
| 04:09.39 | tzanger | [TK]D-Fender: http://www.theglobeandmail.com/servlet/story/RTGAM.20071203.wsords1203/BNStory/National/?page=rss&id=RTGAM.20071203.wsords1203 |
| 04:15.04 | tzanger | go up to q? |
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| 04:34.40 | ZaVoid | hey all |
| 04:34.56 | ZaVoid | so this doesn't make much sense but maybe i'm looking at it wrong |
| 04:35.13 | ZaVoid | if my allow line is g723/g729/ulaw (formatted correctly obviously) |
| 04:35.46 | ZaVoid | but i send a call to the asterisk via a device that only has g723.... and the invite only comes with g723 in the SDP... why would asterisk advertise g723/ulaw and g729 to the device i send the call to? |
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| 04:37.19 | ZaVoid | hmm n etsplit |
| 04:37.39 | ZaVoid | : if my allow line is g723/g729/ulaw (formatted correctly obviously) but i send a call to the asterisk via a device that only has g723.... and the invite only comes with g723 in the SDP... why would asterisk advertise g723/ulaw and g729 to the device i send the call to? |
| 04:42.06 | mosty | because asterisk advertises what it supports |
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| 04:54.12 | [TK]D-Fender | Oh yeah, that lasted f'n long.... |
| 04:54.22 | blitzrage | jeebuz... wtf |
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| 05:05.49 | [TK]D-Fender | ~whee |
| 05:05.50 | jbot | [~whee] Weeeeeeeee! http://www.albinoblacksheep.com/flash/weeee |
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| 05:13.23 | tzanger | *sigh* |
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| 05:24.54 | Morrocco | Hi, I need some help configuring a Polycom 330 phone with asterisknow, what do I need to do to asign the phone an extension so I can use it with my asterisk system? |
| 05:25.26 | fujin_ | hrmph, no idea with asterisknow, although I'd expect it to be relatively painless being a clickybutton gui. |
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| 05:26.28 | Morrocco | yes, its painless, but I want to know the basic information, like to use the phone do I use SIP ? |
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| 05:27.06 | fujin_ | Morrocco: yes, you use SIP |
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| 05:28.24 | Morrocco | ok, Cool, so I just need to configure the phones to use sip via the web interfece of the phone? |
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| 05:31.48 | *** mode/#asterisk [+o russellb] by ChanServ |
| 05:32.08 | Morrocco | do you guys know of a simple softphone that I can use with asterisk Im using windows vista |
| 05:32.32 | russellb | try zoiper |
| 05:32.37 | [TK]D-Fender | ~zoiper |
| 05:32.38 | jbot | [~zoiper] Zoiper (Formerly known as Idefisk) is a free SIP and IAX soft-phone for Windows, MacOSX, and Linux that can be found at http://www.zoiper.com |
| 05:32.40 | [TK]D-Fender | ~xlite |
| 05:32.40 | jbot | [~xlite] X-Lite is a free SIP soft-phone for Windows, MacOSX, and Linux that can be downloaded from http://www.counterpath.com/ |
| 05:32.59 | russellb | [TK]D-Fender: greetings |
| 05:33.05 | [TK]D-Fender | russellb: howdy... |
| 05:33.11 | [TK]D-Fender | Almost out of the office.... |
| 05:33.18 | Morrocco | thank you :) |
| 05:33.43 | russellb | [TK]D-Fender: cool ... what timezone? it's late :) |
| 05:33.55 | [TK]D-Fender | russellb: 12:33am :/ |
| 05:34.07 | russellb | ooh, eastern ... that's where i am now |
| 05:34.15 | russellb | go home, crazy man. |
| 05:35.33 | russellb | [TK]D-Fender: do you do asterisk consulting full time? |
| 05:36.09 | [TK]D-Fender | russellb: Nope, I'm IT for a plumbing distribution company. |
| 05:36.22 | [TK]D-Fender | russellb: and right now in working on budget consolidation. |
| 05:36.29 | russellb | ahhh, cool deal. |
| 05:36.38 | [TK]D-Fender | russellb: the other bane of my existance here. Excel sheets from hell. |
| 05:36.41 | russellb | when are you coming to work for Digium? :) |
| 05:36.52 | [TK]D-Fender | russellb: * consulting is a few extra $ for my hobbies |
| 05:37.05 | russellb | gotcha |
| 05:37.07 | [TK]D-Fender | russellb: Make me an offer ;) |
| 05:37.19 | russellb | willing to relocate? |
| 05:37.30 | [TK]D-Fender | russellb: ummm...... thats rough. |
| 05:37.37 | russellb | understood |
| 05:37.41 | [TK]D-Fender | russellb: dunno, guess it'd have to be a pretty good offer :) |
| 05:38.04 | blitzrage | too bad HSV didn't == California or something :) |
| 05:38.04 | russellb | hehe ... getting people into the US is a pain, too ... |
| 05:38.18 | russellb | blitzrage: yeah, no kidding |
| 05:38.21 | blitzrage | first round of the CLI audit done! |
| 05:38.25 | russellb | blitzrage: awesome |
| 05:38.34 | russellb | [TK]D-Fender: you never know, there are remote possibilities ... |
| 05:38.36 | blitzrage | and now my brain hurts... that took nearly 4 hours |
| 05:38.37 | [TK]D-Fender | russellb: Now Extraordinary Rendition.... THAT'D be easy ;) |
| 05:38.51 | [TK]D-Fender | russellb: Nice double entendre ;) |
| 05:39.00 | russellb | :-D |
| 05:39.05 | [TK]D-Fender | russellb: likely unintentional as it is. |
| 05:39.12 | russellb | yeah, it was not on purpose ... |
| 05:40.44 | russellb | blitzrage: how much are we going to have to change? :-/ |
| 05:40.54 | [TK]D-Fender | ok, thats it, I'm fried. Heading home. Later all. |
| 05:41.15 | blitzrage | russellb: ummm... I haven't done a thorough audit, but it actually didn't look like too many |
| 05:41.17 | Morrocco | ok, I have zoiper, under domain do I put the ip address of my asterksnow box? |
| 05:42.10 | blitzrage | there are a few things that should have been deprecated, but got missed in 1.4... so either we fix and deprecate now in 1.4, or... we fix the CLI and just make it right and docment it very well and make all the changes in trunk and get rid of all the defunct commands |
| 05:42.23 | russellb | Morrocco: yeah, probably .. |
| 05:42.24 | blitzrage | trying to support both methods at the same time is making the CLI look very messy |
| 05:42.33 | blitzrage | Morrocco: yes you do |
| 05:42.57 | russellb | blitzrage: yeah, but we have already done so much of it ... |
| 05:43.04 | russellb | blitzrage: maybe we should aim to have all of the new stuff in 1.4 |
| 05:43.13 | russellb | so that we can remove all deprecated stuff from trunk/1.6 |
| 05:43.48 | blitzrage | russellb: ya, I'd be in favour of that... I just don't want to have yet another round of deprecation for commands that should have been fixed in 1.4. I guess that makes it considered a bug fix eh? |
| 05:43.52 | russellb | ooh, and we could even have an option to turn off the deprecated commands for people that find them annoying ... |
| 05:44.00 | jql | core voip sip show set debug status # exaggerated griping from the peanut gallery |
| 05:44.18 | blitzrage | core set cli [new|old] |
| 05:44.21 | Morrocco | ok, Im trying to call the voicemail service at extension 850 but its not working, any settings that Im missing? |
| 05:44.22 | russellb | blitzrage: yeah, i don't mind doing it in 1.4 ... |
| 05:44.22 | blitzrage | that'd be a nice command |
| 05:44.29 | russellb | to be honest, the changes have irritated the crap out of me, too |
| 05:44.34 | blitzrage | totally |
| 05:44.38 | russellb | i want to get it right and have it done with |
| 05:44.58 | blitzrage | totally agreed. I've got the CLI tree built now... so I can look at it and figure out what is missing and whatnot |
| 05:45.05 | blitzrage | make sure we get it all done right this time |
| 05:45.10 | jql | core add command alias 'set debug' 'core set debug' |
| 05:45.10 | russellb | sounds good. |
| 05:45.21 | jql | core save command alias file |
| 05:45.23 | blitzrage | it certainly was a huge job, and whoever did the original crack at it did a good job |
| 05:45.40 | russellb | yeah, agreed |
| 05:46.01 | blitzrage | jql: I don't like the idea of people being able to make non-standard commands because then if they go to work on another system, they'll be lost |
| 05:46.03 | russellb | but if there is anything left that isn't consistent, let's go ahead and fix 'em up in 1.4 |
| 05:46.21 | russellb | one of these days, i'll make the cisco-style command completion :) |
| 05:46.26 | blitzrage | russellb: yep, I made a few notes in my bug tonight |
| 05:46.30 | jql | which is exactly what makes asterisk frustrating right now. heh |
| 05:46.30 | russellb | and it will make me famous in asterisk land |
| 05:46.42 | blitzrage | you're already famour |
| 05:46.45 | blitzrage | famous* |
| 05:46.47 | jql | now, everyone is lost |
| 05:46.57 | Morrocco | wow |
| 05:46.59 | Morrocco | its working |
| 05:47.05 | russellb | Morrocco: yay :) |
| 05:47.10 | russellb | it's *magic* |
| 05:47.41 | russellb | blitzrage / jql ... there are probably multiple lessons to be learned from this CLI changeup |
| 05:47.59 | *** join/#asterisk shtoom (n=shtoom@59.93.123.185) |
| 05:48.00 | blitzrage | asterisk is just a babe.... we're all learning |
| 05:48.14 | russellb | heh, still a babe, huh? |
| 05:48.19 | blitzrage | I think so :) |
| 05:48.24 | blitzrage | maybe a young adult |
| 05:48.26 | russellb | i've been working on it long enough that i feel like we should be done |
| 05:48.32 | blitzrage | haha, totally |
| 05:48.33 | russellb | but there is so much to do ... |
| 05:48.48 | file | indeed |
| 05:48.48 | blitzrage | that's the problem with a piece of software that does everything... |
| 05:49.09 | blitzrage | wow... I didn't realize just HOW low on food I was... :) |
| 05:49.09 | blitzrage | I'm eating a block of cheese.... |
| 05:49.21 | file | blitzrage: I told you! |
| 05:49.50 | blitzrage | you certainly did... |
| 05:49.59 | blitzrage | although I don't like chinese, and it's so expensive |
| 05:50.14 | blitzrage | always seems like you have to order $40 worth of food |
| 05:50.15 | russellb | we should remove the functionality that does everything |
| 05:50.17 | file | your face is expensive |
| 05:50.17 | russellb | and make it do nothing |
| 05:50.50 | blitzrage | that'd make my job easier |
| 05:50.54 | Morrocco | I have to go guys, cya tomorrow |
| 05:51.15 | Morrocco | thank you for all your help |
| 05:51.17 | Morrocco | :D |
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| 06:01.25 | bmcghee`home | im having a mysql issue |
| 06:01.28 | bmcghee`home | i get this error |
| 06:01.30 | bmcghee`home | http://pastebin.com/d30f98e19 |
| 06:02.07 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 06:07.46 | bmcghee`home | anyone? |
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| 06:39.50 | BeeBuu | hello,all |
| 06:40.24 | BeeBuu | how can i get the callerid when i make call with a iax client? |
| 06:41.12 | *** join/#asterisk shinao1 (n=shinao1@196.207.1.30) |
| 06:47.09 | fujin_ | ${CALLERID(all)} |
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| 06:52.07 | BeeBuu | thanks. |
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| 07:13.23 | BeeBuu | how can i logout a agent? |
| 07:14.41 | BeeBuu | after agentcallbacklogin |
| 07:16.13 | Juggie | i think execute agentcallbacklogin agains |
| 07:16.15 | Juggie | *again |
| 07:16.19 | Juggie | w/o any parms maybe |
| 07:16.38 | BeeBuu | again? |
| 07:16.51 | BeeBuu | type password one more? |
| 07:17.09 | Juggie | try it w/ AgentCallBackLogin(agentid) |
| 07:17.54 | Juggie | check here, http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20AgentCallbackLogin |
| 07:18.57 | BeeBuu | w/ ? |
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| 07:23.12 | *** join/#asterisk RedStalker_Mike (n=mike@unixway.tversu.ru) |
| 07:23.36 | RedStalker_Mike | hi all |
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| 07:46.39 | zeeesh | in linux how check a directory or file propertie? |
| 07:46.54 | JT | for real? |
| 07:50.27 | endre | stat |
| 07:50.53 | JT | ls -la |
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| 07:57.04 | salzh | hi, what's the default passwd for root account on centos where trixbox is installed |
| 07:57.22 | JT | ~trixbox |
| 07:57.23 | jbot | [~trixbox] Trixbox is a full linux distro that includes FreePBX and other 3rd party add-ons. It is all this extra stuff which makes trixbox VERY difficult to support, and is not supported in #asterisk. Try asking in #trixbox or on their forums & wiki at http://www.trixbox.org |
| 07:59.32 | salzh | ~trixbox is failed |
| 07:59.33 | jbot | ...but trixbox is already something else... |
| 08:00.01 | JT | salzh: eh? |
| 08:00.08 | JT | salzh: READ the text |
| 08:00.17 | JT | salzh: you are in the wrong channel |
| 08:00.43 | salzh | *grin* |
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| 08:26.09 | fukz | Hello all |
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| 08:27.22 | fukz | I have a problem while calling from a Siemens HiCom to Asterisk over E1. I use Q.SIG and it works, |
| 08:27.54 | fukz | but when HiCom sends a call in an channel > 15, I have no audio signal. |
| 08:28.15 | fukz | Channels <= 15 works. |
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| 08:51.21 | codejunky | Hi, any ideas what could be causing this error: sk |
| 08:51.22 | codejunky | drwxr-xr-x 3 root root 17 Sep 19 10:05 cache |
| 08:51.25 | codejunky | argh |
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| 08:51.44 | CrummyGummy | Hi all, what do I use to connect a G703 pipe to a 64K card? |
| 08:51.58 | codejunky | This error I mean: http://rafb.net/p/7JVBtB89.html, sorry |
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| 08:53.07 | *** join/#asterisk joelsolanki (i=joelsola@202.160.161.94) |
| 08:54.12 | joelsolanki | Hi all |
| 08:54.20 | joelsolanki | good morning |
| 08:54.46 | joelsolanki | i configured this in sip.conf |
| 08:54.49 | joelsolanki | #include /etc/asterisk/sip-phones/* |
| 08:54.53 | joelsolanki | and it worked gr8. |
| 08:55.04 | joelsolanki | same way i want to include extensions |
| 08:55.13 | joelsolanki | i did configured this in extensions.conf |
| 08:55.29 | joelsolanki | #include "/etc/asterisk/sip-extensions/*" |
| 08:55.41 | joelsolanki | tried to remove "" and check but it is not getting included |
| 08:56.05 | joelsolanki | did dialplan reload |
| 08:56.06 | joelsolanki | [Dec 7 14:26:58] WARNING[32632]: config.c:864 config_text_file_load: '/etc/asterisk/sip-extensions/internal-extensions' is not a regular file, ignoring |
| 08:56.22 | joelsolanki | any hints plz |
| 08:58.52 | loompek | ls -l /etc/asterisk/sip-extensions/internal-extensions |
| 08:59.30 | loompek | file /etc/asterisk/sip-extensions/internal-extensions |
| 08:59.34 | joelsolanki | -rw-r--r-- 1 root root 44 Dec 7 14:19 4024645971.conf |
| 08:59.52 | joelsolanki | -rw-r--r-- 1 root root 43 Dec 7 14:20 4024648707.conf |
| 08:59.59 | loompek | it seems that internal-extensions is a directory |
| 09:00.07 | joelsolanki | yes it is directory. |
| 09:00.11 | loompek | well? |
| 09:00.16 | loompek | it's not a regular file :D |
| 09:00.22 | joelsolanki | i want to include all files in internal-extensions directory |
| 09:00.38 | joelsolanki | it worked in sip.conf |
| 09:03.43 | joelsolanki | i hope anybody had worked on this already |
| 09:03.44 | joelsolanki | ? |
| 09:04.49 | loompek | nope |
| 09:04.50 | loompek | not me |
| 09:05.12 | loompek | what line are you using for including |
| 09:05.45 | loompek | http://www.voip-info.org/wiki/index.php?page=Asterisk+config+extensions.conf |
| 09:06.20 | joelsolanki | let me check |
| 09:06.23 | joelsolanki | 1 sec |
| 09:06.51 | loompek | #include <exts/...> |
| 09:11.07 | joelsolanki | include => /etc/asterisk/sip-extensions/* |
| 09:11.12 | joelsolanki | i did this now. |
| 09:11.33 | joelsolanki | -- Including context '/etc/asterisk/sip-extensions/*' in context 'default' |
| 09:11.33 | joelsolanki | <PROTECTED> |
| 09:11.34 | joelsolanki | [Dec 7 14:42:30] WARNING[32679]: pbx.c:6300 ast_context_verify_includes: Context 'default' tries includes nonexistent context '/etc/asterisk/sip-extensions/*' |
| 09:11.55 | joelsolanki | says warning |
| 09:18.47 | *** part/#asterisk BeeBuu (n=beebuu@59.38.99.237) |
| 09:19.23 | loompek | what if you put ... instead of * |
| 09:19.27 | loompek | just for fun |
| 09:20.20 | *** join/#asterisk vale-ICS (n=vale@icsnet.demon.co.uk) |
| 09:21.39 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
| 09:25.29 | joelsolanki | let me check |
| 09:25.51 | *** join/#asterisk KermitTheFragger (n=siepkes@118-197.bbned.dsl.internl.net) |
| 09:26.20 | joelsolanki | [Dec 7 14:57:15] WARNING[32728]: pbx.c:6300 ast_context_verify_includes: Context 'default' tries includes nonexistent context '/etc/asterisk/sip-extensions/' |
| 09:26.21 | joelsolanki | :) |
| 09:26.35 | R1ck | I'm trying to dial out, but get the following message: "Dec 7 10:21:51 NOTICE[3723] app_dial.c: Unable to create channel of type 'ZAP' (cause 34 - Circuit/channel congestion) |
| 09:26.38 | R1ck | " |
| 09:26.42 | R1ck | whats causing that? |
| 09:27.57 | kaldemar | joelsolanki: you have "include => ..." which includes contexts in other contexts, "#include <file>" includes other files. |
| 09:29.36 | joelsolanki | kaldemar: i want to includes all files under /etc/asterisk/sip-extensions/ |
| 09:29.46 | joelsolanki | i dont want to include individual file |
| 09:31.44 | *** join/#asterisk sergey (n=sergey@91.189.233.71) |
| 09:32.19 | joelsolanki | kaldemar: u want me to use #include /etc/asterisk/sip-extensions/* ??? |
| 09:32.21 | kaldemar | joelsolanki: well, i just tried that and "#include /path/to/files/*" works fine. |
| 09:32.37 | joelsolanki | let me do that. |
| 09:33.21 | joelsolanki | kaldemar: that worked. |
| 09:33.42 | joelsolanki | actually i was trying to included one folder too that was creating problem |
| 09:34.04 | joelsolanki | so now i did is #include /etc/asterisk/sip-extensions/internal-extensions/* |
| 09:34.08 | joelsolanki | this worked |
| 09:39.38 | *** part/#asterisk CrummyGummy (n=dude@dsl-244-240-16.telkomadsl.co.za) |
| 09:40.59 | *** part/#asterisk sergey (n=sergey@91.189.233.71) |
| 09:49.11 | R1ck | I'm trying to dial out, but get the following message: "Dec 7 10:21:51 NOTICE[3723] app_dial.c: Unable to create channel of type 'ZAP' (cause 34 - Circuit/channel congestion) |
| 09:49.15 | R1ck | whats causing that? |
| 09:49.15 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 09:50.06 | kaldemar | R1ck: without more information, no one can tell you what causes that. |
| 09:52.08 | kaldemar | if you pastebin your zaptel.conf, zapata.conf, and the full trace for the call with maximum verbosity, you might get somewhere. |
| 09:53.32 | R1ck | http://www.pastebin.ca/808149 |
| 09:55.23 | kaldemar | is that your whole zapata.conf? i only see group 1 and your dialing group 0. |
| 09:55.57 | R1ck | eh, yeah, thats it.. |
| 09:56.15 | kaldemar | replace g0 with g1 in the dial line if it's the first span you're trying to use. |
| 09:56.23 | *** join/#asterisk SuurMyy (n=SuurMyy_@195.238.211.98) |
| 09:58.25 | *** join/#asterisk oej (n=olle@193.126.30.214) |
| 10:10.45 | R1ck | kaldemar: allright, getting a little further now.. |
| 10:11.41 | R1ck | I now get this: http://www.pastebin.ca/808156 |
| 10:15.23 | R1ck | also i keep getting these: |
| 10:15.27 | R1ck | Dec 7 11:14:48 DEBUG[4260] chan_sip.c: Stopping retransmission on '11d1396a4be2bea91f4c6f30693b0355@192.168.20.2' of Request 102: Match Found |
| 10:15.27 | R1ck | Dec 7 11:15:02 DEBUG[4260] chan_sip.c: Auto destroying call 'OWE4Y2UyMDM1ZWQ5ZWNkZGQxMWMyMzU1YmEwMTAxMDc.' |
| 10:15.31 | R1ck | whats that |
| 10:19.22 | *** join/#asterisk jm|home (n=jm|home@zen.jamiem.com) |
| 10:23.43 | *** join/#asterisk klyrelion (n=Kevin@icsnet.demon.co.uk) |
| 10:40.49 | *** part/#asterisk klyrelion (n=Kevin@icsnet.demon.co.uk) |
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| 10:51.15 | *** join/#asterisk fukz (n=johns@p4FC2E513.dip.t-dialin.net) |
| 11:06.22 | R1ck | Dec 7 12:06:15 WARNING[3889] chan_zap.c: 1 !! Got S-frame while link down |
| 11:06.28 | R1ck | why does it keep saying that |
| 11:08.37 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
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| 11:29.20 | *** part/#asterisk dec (n=tom@unaffiliated/dec) |
| 11:37.09 | DJ_InstincT | any1 here know how I can test UK CLI is being sent by bt [without calling them?] |
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| 11:43.17 | EitanS | Hi, I would like to know where I can get info in solving a problem I am having with Asterisk, our sangoma wanpipe card says its disconnected even though the cable is in fact connected. |
| 11:55.35 | *** join/#asterisk bantu (n=Miranda@p54A32A1D.dip0.t-ipconnect.de) |
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| 12:23.39 | EitanS | Hi, I would like to ask a question concerning the sangoma card and problems being connected. We get a disconnected status on all our lines. its an NT setup. Can anyone help? |
| 12:28.55 | *** join/#asterisk sECuRE_ (n=sECuRE@ist.eine.netzhure.de) |
| 12:29.43 | sECuRE_ | hi.. i'm having problems with mISDN (several oops until the system completely hangs). are there any chances to get it fixed or is mISDN currently really broken? (version is 1.1.7 on kernel 2.6.24-rc3 with asterisk 1.2) |
| 12:31.07 | sECuRE_ | the kernel oops aren't even in the misdn-code but in sysfs and kcryptd.. |
| 12:32.20 | R1ck | I got this on an incoming call: Extension '0263844911' in context '"from-zaptel"' from '0263611107' does not exist. Rejecting call on channel 0/1, span 4 |
| 12:33.27 | *** join/#asterisk coppice (n=chatzill@235.202.17.210.dyn.pacific.net.hk) |
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| 12:45.13 | *** mode/#asterisk [+o Corydon76-vcch] by ChanServ |
| 12:47.25 | tzafrir_home | R1ck, just what the message says |
| 12:47.56 | tzafrir_home | to see that context: dialplan show from-zaptel |
| 12:49.59 | tzafrir_home | EitanS, In what way does the sangoma card say it is disconnected? What exactly do you see? |
| 12:50.02 | *** join/#asterisk zerohalo (n=zeroHalo@pool-71-162-97-18.bstnma.east.verizon.net) |
| 12:50.04 | *** join/#asterisk _ys (i=yuri@91.151.196.254) |
| 12:50.08 | tzafrir_home | Did it work before? |
| 12:51.20 | EitanS | No it hasnt worked before, its a new installation. Using woomera, where wanrouter status shows all lines as disconnected. |
| 12:53.29 | EitanS | We are using an Sangoma A500 card, we struggling to debug |
| 12:54.14 | tzafrir_home | Sorry. /me only knows bristuff and not woomera (and netbricks) |
| 12:55.04 | EitanS | A500 is BRI if that helps any ? |
| 12:56.12 | R1ck | tzafrir_home: No such command 'dialplan' (type 'help' for help) |
| 12:56.38 | tzafrir_home | ah, you use 1.2. show dialplan from-zaptel |
| 12:56.59 | tzafrir_home | R1ck, in short: you need to handle that in your dialplan |
| 12:57.17 | coppice | I wonder how asterfax gets hosted at sourceforge, when it isn't open source? |
| 12:57.41 | tzafrir_home | coppice, what isn't free there? |
| 12:58.05 | coppice | the source does not appear to be available |
| 12:58.33 | *** join/#asterisk lirakis (n=lirakis@65.200.191.253) |
| 12:59.45 | tzafrir_home | any idea when they have last updated http://asterfax.sourceforge.net/Licensing%20AsterFax.html ? |
| 13:01.28 | coppice | dunno, but the sf.net download pages seems to only have binaries |
| 13:02.22 | coppice | and although there is a free binary, it has restricted functionality |
| 13:05.29 | tzafrir_home | http://web.archive.org/web/20060619202621/http://asterfax.sourceforge.net/Licensing+AsterFax.html |
| 13:06.11 | loompek | exten => _100431XXX.,1,Dial(SIP/${EXTEN:7},20) |
| 13:06.23 | coppice | is that one different? it looks very similar |
| 13:06.29 | tzafrir_home | They fully admit it. And have not acted for probably over a year and a half |
| 13:06.39 | loompek | in case i have a trunk... then this should dial that number... right? |
| 13:06.39 | tzafrir_home | right |
| 13:07.25 | loompek | like.. if i get a call for 100431230 it should dial 230.. right? |
| 13:08.20 | tzafrir_home | The page above is the first one recorded in the wayback machine |
| 13:08.21 | kaldemar | loompek: :7 will chop 7 digits off EXTEN. so it would dial your SIP user 30. |
| 13:08.23 | *** join/#asterisk phillipk (n=pkey@fw.datafax.net) |
| 13:08.36 | loompek | bummer |
| 13:09.04 | loompek | exten => _100431XXX,1,Dial(SIP/${EXTEN:6},20) |
| 13:09.08 | loompek | okay.. this is better? |
| 13:09.17 | loompek | dialing 100431XXX should dial XXX |
| 13:09.55 | *** join/#asterisk webman (n=adamg@52.87.233.220.exetel.com.au) |
| 13:15.21 | *** join/#asterisk saftsack (n=saftsack@p4FC745DC.dip.t-dialin.net) |
| 13:25.54 | EitanS | tzafrir, where would i go to find a application to do a line test ? we think the problem may be starting there? |
| 13:27.38 | *** join/#asterisk Nivex (n=kjotte@user-0c8hvoj.cable.mindspring.com) |
| 13:29.34 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 13:29.35 | R1ck | what voip phones are recommended for use with asterisk? |
| 13:29.35 | [TK]D-Fender | ~phones |
| 13:29.41 | jbot | i guess phones is http://bani.anime.net/phones/. While personal preference will dictate which phone works best for you, general consensus on a rough order of quality and suggestibility is as follows: Polycom (any), Aastra 480i, Aastra 5i Series, Cisco 7940+, Linksys SPA-9XX, Snom, and finally everything else. Do not consider Grandstream phones. Ever. places ... |
| 13:30.34 | R1ck | thanks [TK]D-Fender |
| 13:30.57 | [TK]D-Fender | R1ck: Actually I'd place Linksys above Cisco for the mostpart |
| 13:31.15 | R1ck | but Polycom is best? |
| 13:32.05 | [TK]D-Fender | R1ck: pretty much |
| 13:32.13 | R1ck | cool tnx |
| 13:32.43 | [TK]D-Fender | R1ck: Aastra & snom have a few more features, but Polycom wins on quality and basic call handling. |
| 13:34.15 | coppice | [TK]D-Fender: have they offered you a sales position, yet? |
| 13:34.30 | [TK]D-Fender | coppice: Not yet.... |
| 13:34.51 | Qwell | he probably sells more than people on their sales team :p |
| 13:35.42 | coppice | I think Polycom sucks (trying to achieve fair balance here) |
| 13:36.00 | EitanS | Can anyone help, when running wanpipemon -i xxx -u 9000 -c xm for modem status I get the following message: "Protocol: unkown support not compiled in!" |
| 13:36.31 | R1ck | allright, gonna order a Polycom 301 (for testing) |
| 13:37.24 | coppice | if you don't like it, the handset is fairly robust, so you can beat [TK]D-Fender senseless with it |
| 13:37.38 | cpm | nice point |
| 13:38.16 | [TK]D-Fender | Might make right... and I am very VERY right ;) |
| 13:38.25 | [TK]D-Fender | R1ck: I'd advise against the 301 |
| 13:38.42 | cpm | what's wrong with the 301? |
| 13:38.47 | [TK]D-Fender | R1ck: the IP320/330 completely de-validate it as a choice |
| 13:39.14 | R1ck | the 320 doesnt have a hub thingie |
| 13:39.22 | [TK]D-Fender | cpm: 301 costs more than them, and the new ones have native PoE, Speakerphone, pixel display and microbrowser capabilities. |
| 13:39.30 | [TK]D-Fender | R1ck: IP 330 does |
| 13:39.39 | R1ck | hmm yeah |
| 13:39.41 | coppice | dot ehy make good coffee? |
| 13:39.42 | cpm | ah |
| 13:39.48 | cpm | yeah, nice point |
| 13:39.58 | R1ck | 330 seems ok |
| 13:39.59 | [TK]D-Fender | coppice: Mine does thanks to X-10 :) |
| 13:40.00 | cpm | I'll bet they don't, and don't bother disclosing that little detail |
| 13:40.03 | R1ck | its cheaper :P |
| 13:40.17 | cpm | typical |
| 13:40.28 | [TK]D-Fender | R1ck: like I said |
| 13:41.14 | cpm | most IP ethernet phones don't make decent coffee, and you won't find out about that feature missing until it's too late |
| 13:43.16 | coppice | I blame it on these fine geometry ARM and MIPS core. they don't cook anything very well |
| 13:44.48 | R1ck | hmm, the 320 and 330 are un-orderable |
| 13:44.48 | *** join/#asterisk reber (n=reber@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr) |
| 13:45.07 | cpm | indeed. POE *should* be able to handle a decent coffee maker |
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| 13:46.03 | *** mode/#asterisk [+oo Corydon76-vcch Qwell] by irc.freenode.net |
| 13:46.46 | [TK]D-Fender | coppice: Well the IP 650 has a USB expansion port. I'm not sure its powered, but if it is, there are USB mug warmers out there, so this might just be viable! |
| 13:47.14 | coppice | what's the USB port for? |
| 13:47.25 | coppice | other than coffee making |
| 13:48.01 | *** join/#asterisk docelmo (n=vircuser@c-76-99-153-242.hsd1.de.comcast.net) |
| 13:48.27 | cpm | yeah, but the POE isn't up to it. |
| 13:49.26 | *** join/#asterisk GiantPickle (n=GiantPic@S01060016b600537f.gv.shawcable.net) |
| 13:49.52 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.125.196) |
| 13:50.19 | [TK]D-Fender | cpm: IIRC PoE pushes far more than USB. |
| 13:51.00 | cpm | [TK]D-Fender, yeah, but this tiny, efficient cpus used in these phones don't waste enough heat to properly brew coffee. |
| 13:51.23 | coppice | USB is only 2.5W. PoE does a lot more than that |
| 13:51.32 | [TK]D-Fender | coppice: thats a little grey right now. guesses for things in the works. : USB Wi-Fi, local memory expansion for custom software / directories, authentication, etc |
| 13:51.49 | cpm | coppice, you have a coffee pot running off PoE? |
| 13:52.00 | [TK]D-Fender | cpm: I said via an extension USB mug warmer! not the CPU direct! |
| 13:52.17 | *** join/#asterisk joelsolanki (i=joelsola@202.160.161.94) |
| 13:52.23 | joelsolanki | Hi all. Good morning |
| 13:52.30 | cpm | [TK]D-Fender, mug warmer! weak. |
| 13:52.42 | joelsolanki | I have a specific requirement. |
| 13:52.42 | cpm | chrome 20cup percolator |
| 13:53.31 | joelsolanki | i have 2 plans Unlimited US/Canada and A-Z |
| 13:53.52 | Qwell | A-Z? All of my providers allow 0-9 |
| 13:54.22 | *** join/#asterisk jbondc2 (n=jbondc@modemcable158.97-203-24.mc.videotron.ca) |
| 13:54.24 | coppice | their operating costs are lower if they don't offer 0-9 |
| 13:54.32 | cpm | yup. |
| 13:54.44 | Qwell | I wonder if I can remove the 0-9 service from my accounts, and save a few bucks. |
| 13:54.47 | cpm | or allow phantom powered coffee brewers |
| 13:55.02 | joelsolanki | this is my config in extensions.conf |
| 13:55.05 | joelsolanki | [digitalphone-unlimited] |
| 13:55.06 | joelsolanki | #include /etc/asterisk/sip-extensions/internal-extensions/* |
| 13:55.06 | joelsolanki | #include /etc/asterisk/sip-extensions/external-providers/digitalphone-unlimited.conf |
| 13:55.06 | joelsolanki | [digitalphone-az] |
| 13:55.06 | joelsolanki | #include /etc/asterisk/sip-extensions/internal-extensions/* |
| 13:55.06 | joelsolanki | #include /etc/asterisk/sip-extensions/external-providers/digitalphone-az.conf |
| 13:55.07 | jbondc2 | hi everyone, have anyone ever played with a sipura 2102? |
| 13:55.30 | [TK]D-Fender | jbondc2: What about it? |
| 13:55.34 | jbondc2 | i'm troubleshooting it and it looks like there's a: SIP/2.0 489 Bad event |
| 13:55.42 | [TK]D-Fender | joelsolanki: Do not spam like that again, PASTEBIN it |
| 13:55.47 | joelsolanki | I m sorry. |
| 13:55.50 | jbondc2 | when it tries to send a NOTIFY |
| 13:56.12 | *** join/#asterisk rbd (n=rbd@adsl-074-229-183-112.sip.rmo.bellsouth.net) |
| 13:56.14 | [TK]D-Fender | jbondc2: WHO is sending the notify, and what is it for? |
| 13:56.42 | De_Mon | http://pastebin.ca/808319 |
| 13:56.57 | De_Mon | Any ideas how to get asterisk to do what it says its going to do on timeout? |
| 13:57.15 | jbondc2 | The sipura 2102 is behind NAT, when I enable keep-alive on it, it sends: NOTIFY sip:pbx.gdesolutions.com:5060 SIP/2.0 |
| 13:57.30 | jbondc2 | asterisk replies: SIP/2.0 489 Bad event |
| 13:57.43 | rbd | hi guys, I'm trying to use a simple callflow that answers, plays back a prompt, then hangs up. I dial in from my sip phone, the call is answered and looks like the prompt is played, however I don't hear anything and the call never hangs up afterwards. no errors in asterisk using -cvvvvvvvvv level verbosity, etc. Using wireshark, I see that the sip messaging looks good, and there is RTP (g711 ulaw), but the RTP payload is all blank (F |
| 13:58.00 | rbd | any ideas? |
| 13:58.11 | rbd | tried with two different sip softphones, same results with either |
| 13:58.47 | rbd | and machines are on the same network (private IPs)...no firewall between then, etc. |
| 13:59.20 | De_Mon | rbd what eversion of asterisk |
| 13:59.30 | twisted | jbondc2: so? it's just a keep-alive. just because asterisk says "bad event" doesn't mean the keepalive isnt' working. |
| 13:59.34 | rbd | 1.4.10 |
| 13:59.45 | De_Mon | edit logger.conf and enable debugging on the console and pastebin the call with verbose and debug turned on |
| 14:00.22 | *** part/#asterisk xachen (n=justin@pdpc/supporter/student/xachen) |
| 14:00.23 | De_Mon | (logger reload then core set debug 10) |
| 14:00.39 | twisted | jbondc2: all a keep-alive does is keep the nat ports open so that two way communication isn't interrupted. it's just like asterisk sending a qualify. if you don't want to see the bad event, then enable qualify on your sip device in the asteirsk config, and disable the keep-alive within the sipura. |
| 14:01.50 | [TK]D-Fender | ^^^ |
| 14:02.53 | rbd | De_Mon: hmm, ok I did that (both issued that command on the console, and edited logger.conf)...I only get the 3 normal lines when I make the call (answering, playback, playing prompt)...I started asterisk with "asterisk -vvvvvvvvvvc" ... |
| 14:02.57 | joelsolanki | HI all. |
| 14:03.01 | joelsolanki | plz see this pastebin |
| 14:03.09 | joelsolanki | with complete details |
| 14:03.10 | joelsolanki | http://www.pastebin.ca/808328 |
| 14:04.29 | jbondc2 | well the phone seems to be working, i guess your right, the packets going out keep the port open =] Sadly qualify isn't an option... the firewall is picky |
| 14:05.40 | jbondc2 | is there an advantage of having NAT keep-alive AND qualify as well? |
| 14:05.53 | twisted | joelsolanki: SPECIFY the file you're #including for internal extensions |
| 14:06.01 | twisted | just like you're doing for the digitalphone files |
| 14:06.03 | *** join/#asterisk _ShrikE (n=ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 14:06.08 | De_Mon | rbd oh yea do a 'core set debug chan_sip.c' or 'sip set debug' to get the sip messages added |
| 14:06.12 | joelsolanki | yes 1 sec |
| 14:06.38 | [TK]D-Fender | joelsolanki: reverse the order of your includes. |
| 14:06.55 | De_Mon | looks like the core set requires a verbosity |
| 14:07.13 | twisted | [TK]D-Fender: it shouldn't matter, exact matches take precedence over pattern matches |
| 14:07.15 | R1ck | yay. 330 has been ordered. |
| 14:07.16 | joelsolanki | 4024645971.conf 4024648707.conf are the file internal-extensions directory |
| 14:07.17 | *** join/#asterisk Pagautas (n=bigman@83.171.14.250) |
| 14:07.33 | joelsolanki | reverse means ? |
| 14:07.43 | [TK]D-Fender | twisted: What happened to the old precedence? |
| 14:07.59 | twisted | [TK]D-Fender: AFAIK, exact matches always take precedence, and always have |
| 14:08.42 | [TK]D-Fender | twisted: Maybe what I'm referring to has to do with only patterns.... |
| 14:09.12 | joelsolanki | <[TK]D-Fender>: waiting for your input |
| 14:09.22 | twisted | [TK]D-Fender: sure, but you were responding to joelsolanki, which, if you look at the pastebins, has two exact matches |
| 14:09.29 | [TK]D-Fender | joelsolanki: Get a translator, its a basic word. |
| 14:09.39 | twisted | anywho, off to work |
| 14:09.42 | rbd | De_Mon: http://www.pastebin.org/10801 ....looks like maybe something with the SIP_ALREADYGONE .... |
| 14:10.24 | rbd | De_Mon: actually I think that's just when I hung up... |
| 14:10.39 | joelsolanki | <[TK]D-Fender>: plz explain little. i m confused. :( |
| 14:11.12 | [TK]D-Fender | joelsolanki: change the order of your includes. |
| 14:11.21 | joelsolanki | u want me put digitalphone-unlimited.conf include above internal-extension/* ? |
| 14:11.26 | *** join/#asterisk dioedu (n=dioedu@201.7.117.114) |
| 14:13.08 | *** join/#asterisk zydrunas (n=Miranda@216.49.228-ip-184.ckt.net) |
| 14:13.20 | [TK]D-Fender | joelsolanki: Yes. |
| 14:14.26 | joelsolanki | ok doing... |
| 14:15.05 | joelsolanki | oh gr8. that worked. |
| 14:15.12 | joelsolanki | :) |
| 14:15.25 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 14:15.40 | *** join/#asterisk JayTee52 (n=jforde@207-67-84-185.static.twtelecom.net) |
| 14:15.40 | De_Mon | hrm |
| 14:16.25 | [TK]D-Fender | \o/ |
| 14:16.28 | De_Mon | I was thinking it has something to do with [Dec 7 09:07:52] DEBUG[11303]: chan_sip.c:2139 __sip_ack: Stopping retransmission on 'NmMwNGI5MTVhOGNlNGUyM2RkMDA5ZTI3MGQyNjA5YzM.' of Response 1: Match Not Found |
| 14:18.15 | JayTee52 | I'm installing * version 1.4 the tutorial says to build the libpri modules first, then zaptel and then asterisk. I will be using 2 TDM04B cards at first and then swapping out 1 for a TE210P card at a later point in time. Do I need to load the libpri modules when I first install Asterisk then or can I wait till I'm ready to use the TE210P card? |
| 14:18.40 | [TK]D-Fender | JayTee52: Do it all now. |
| 14:19.09 | JayTee52 | [TK]D-Fender, damn skippy! I'm gonna have to put you on the payroll! :-) |
| 14:19.35 | [TK]D-Fender | JayTee52: My rates are very accessable :) |
| 14:19.39 | JayTee52 | so if the card isn't in the server it Asterisk won't pitch a fit? |
| 14:20.49 | [TK]D-Fender | JayTee52: noe. |
| 14:20.53 | [TK]D-Fender | nope* |
| 14:20.53 | De_Mon | tbd just checked a debug of a valid playback and everything looks normal |
| 14:21.07 | JayTee52 | [TK]D-Fender, if you do consulting gigs and have experience with routing T1 PRI ISDN circuits in Asterisk we might want to use your services. |
| 14:21.31 | De_Mon | rdb even |
| 14:21.47 | De_Mon | rbd damnit sorry I can't spell your 3 letter nick :) |
| 14:22.09 | De_Mon | rbd try playing back a file that we know contains audio like welcome |
| 14:22.57 | [TK]D-Fender | rbd: And put a wait(2) in front. |
| 14:23.14 | [TK]D-Fender | JayTee52: PM me details |
| 14:23.50 | *** join/#asterisk Toerkeium (i=oo@201.216.206.221) |
| 14:24.48 | *** join/#asterisk MrTelephone (n=test@S010600195b946fa3.ls.shawcable.net) |
| 14:24.55 | *** join/#asterisk funxion (n=x@63.214.236.169) |
| 14:25.06 | MrTelephone | hey shouldn't there be a Contact: header on the register request from asterisk with www-authenticate? |
| 14:25.58 | *** join/#asterisk pigpen (n=mark@fw.seamans.cc) |
| 14:26.26 | *** join/#asterisk ming_zym (n=ming_zym@124.14.236.56) |
| 14:26.30 | pigpen | hi all, I am running 1.4.11 pretty successfully, any reason not to go to 1.4.15? (yes, I use realtime pgsql) |
| 14:27.11 | MrTelephone | read the changelog |
| 14:27.12 | [TK]D-Fender | pigpen: If it ain't broke... |
| 14:27.47 | pigpen | [TK]D-Fender, well, I know there are some new enhancements to iax in 1.4.11+, and I run quite a bit of it. |
| 14:28.21 | rbd | De_Mon: ok did what you and [TK]D-Fender said...still no audio (didn't have welcome, so I used hello-world.gsm)... in my wireshark output I see that the rtp payload for all the packets is a mix of F, E, D and 7 only ...like FEFED7FF7E7E7F7 ... |
| 14:28.36 | rbd | it's like that for every prompt I try |
| 14:28.51 | *** join/#asterisk billybongo (n=rich@85-189-96-153.rcg-global.managedbroadband.co.uk) |
| 14:29.25 | *** join/#asterisk putnopvut (i=putnopvu@nat/digium/x-b9820a83e0eac07d) |
| 14:31.18 | [TK]D-Fender | pigpen: not really. |
| 14:33.04 | funxion | anyone knwo of a reason the cdr_sqlite.so would fail when starting *? |
| 14:34.28 | funxion | tk? |
| 14:34.38 | *** join/#asterisk twisted (n=twisted@pdpc/supporter/active/twisted) |
| 14:34.38 | *** mode/#asterisk [+o twisted] by ChanServ |
| 14:34.50 | funxion | Dec 7 14:24:12 ERROR[4705]: cdr_sqlite.c:178 load_module: cdr_sqlite: Unable to create table 'cdr': table cdr already exists |
| 14:34.50 | funxion | Dec 7 14:24:12 WARNING[4705]: loader.c:345 ast_load_resource: cdr_sqlite.so: load_module failed, returning -1 |
| 14:34.50 | funxion | <PROTECTED> |
| 14:34.50 | funxion | Dec 7 14:24:12 WARNING[4705]: loader.c:440 load_modules: Loading module cdr_sqlite.so failed! |
| 14:35.56 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 14:35.58 | [TK]D-Fender | funxion: no idea |
| 14:36.55 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 14:36.55 | *** mode/#asterisk [+o anthm] by ChanServ |
| 14:40.49 | *** join/#asterisk rpyne (n=rpyne@69.77.169.14) |
| 14:41.03 | rpyne | Hello World |
| 14:41.10 | De_Mon | rbd duno, everything else looks right |
| 14:41.34 | rpyne | Good Day, I'm having a problem with voicemail |
| 14:42.35 | rpyne | I have remote users accessing the voicemail system and only receiving 20 seconds of their messages. |
| 14:43.31 | rpyne | Any thoughts as to why this is happening. Other remote users have tried accessing their voicemail under exactly the same conditions and were successful, |
| 14:43.48 | rpyne | I'm just having the problem with these 2 users |
| 14:44.03 | *** join/#asterisk tobias (n=tobias@user-0ce2hr3.cable.mindspring.com) |
| 14:46.13 | [TK]D-Fender | rpyne: what phones, describe their connection in detail and pastebint he CLI output of a failed attempt at verbose 10, and channel debug enabled |
| 14:46.42 | *** join/#asterisk PepOSX (n=pepOSX@201.248.215.16) |
| 14:50.27 | *** join/#asterisk datachomper (n=russ@75.146.194.61) |
| 14:50.49 | datachomper | Is there anyway to get ser to register with a registrar, similar to the way you can register => sip@host with asterisk? |
| 14:53.45 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 14:53.45 | *** mode/#asterisk [+o russellb] by ChanServ |
| 14:55.25 | rbd | De_Mon: ok thanks. I am trying to reinstall asterisk and try a few other things. I'll let you know if something changes |
| 14:56.10 | *** join/#asterisk tsearle (n=torrey@97.110-246-81.adsl-static.isp.belgacom.be) |
| 14:57.30 | errr | do Hints work with realtime? |
| 15:02.41 | *** join/#asterisk frigidzephyr (i=frigidze@nat/digium/x-0c7d8c57639bca6a) |
| 15:03.24 | rpyne | [TK]D-Fender: We are using Aastra 480i; I will paste the debug now |
| 15:03.55 | rpyne | -- Executing Answer("SIP/450-ac7fb0d8", "") in new stack |
| 15:03.55 | rpyne | <PROTECTED> |
| 15:03.55 | rpyne | <PROTECTED> |
| 15:03.55 | rpyne | <PROTECTED> |
| 15:03.55 | rpyne | <PROTECTED> |
| 15:03.56 | rpyne | <PROTECTED> |
| 15:03.58 | rpyne | <PROTECTED> |
| 15:04.00 | rpyne | <PROTECTED> |
| 15:04.02 | rpyne | <PROTECTED> |
| 15:04.06 | rpyne | <PROTECTED> |
| 15:04.08 | rpyne | <PROTECTED> |
| 15:04.10 | rpyne | <PROTECTED> |
| 15:04.12 | rpyne | <PROTECTED> |
| 15:04.14 | rpyne | <PROTECTED> |
| 15:04.14 | *** kick/#asterisk [rpyne!i=north@pdpc/sponsor/digium/Qwell] by Qwell (pastebin) |
| 15:04.28 | errr | amazing |
| 15:04.51 | [TK]D-Fender | twit |
| 15:05.05 | [TK]D-Fender | s(qwell)tched! |
| 15:05.11 | errr | heh |
| 15:05.11 | *** join/#asterisk rpyne (n=rpyne@69.77.169.14) |
| 15:05.25 | [TK]D-Fender | RypPn: Ok, WAKE UP TIME. Pastebin it, do NOT spam this channel |
| 15:05.27 | [TK]D-Fender | ~pb |
| 15:05.28 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 15:05.29 | [TK]D-Fender | ^^^^^^^^^^^ |
| 15:06.32 | [TK]D-Fender | rpyne: and we do not see the complete call from beginning to end, nor do we see any SIP DEBUG. |
| 15:08.13 | rpyne | User dials *97 for their voicemail and an execute Answer() initiates the process as shown |
| 15:08.45 | rpyne | I will use pastebin to show everything |
| 15:10.44 | *** join/#asterisk enalert (n=trelane@2001:4830:150c:0:20d:61ff:fe31:a58) |
| 15:10.45 | *** join/#asterisk grandpapadot (n=no@adsl-074-185-089-046.sip.bhm.bellsouth.net) |
| 15:11.33 | enalert | I got a batch of polycom phones with a background image on them. Is there something I can put in the autoprovisioning scripts to clear it? (is there a good setup guide for the various items that the polycom phones download to provision?) |
| 15:11.39 | grandpapadot | Hi all. Anyone seen this error? I just started getting it out of nowhere. I have reinvite off for all the peers that display the error. Doesn't really seem to affect anything (that I can tell) but just out of nowhere, no updates, nothing: handle_response_invite: Forbidden - wrong password on authentication for INVITE to <channel info> |
| 15:12.34 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 15:12.36 | [TK]D-Fender | enalert: the only way they got that is through provisioning in the first place. |
| 15:12.45 | [TK]D-Fender | enalert: So naturally yes, thats the way to clear it. |
| 15:13.15 | [TK]D-Fender | enalert: Grab the firmware from your vendor, and the admin guides off of Polycom. check the Wiki for a guide to setting them up |
| 15:13.39 | Ryushin | Every time I restart asterisk, I'm getting this error: chan_zap.c: Unknown signalling method 'pri_cpe' |
| 15:14.00 | Ryushin | I just upgraded from zaptel 1.4.5 to zaptel 1.4.7 and Asterisk 1.4.11 to 1.4.15. |
| 15:14.16 | Qwell | Ryushin: update libpri too |
| 15:14.25 | Ryushin | I have libpri installed. I've compiled from source several times. This is confusing the heck out of me. |
| 15:14.27 | Qwell | then in asterisk, re-run ./configure and make install |
| 15:14.29 | enalert | [TK]D-Fender, would you happen to have one of those manual thingies to point me to that might elaborate? |
| 15:14.39 | Ryushin | I've been using libpri that comes with debian. |
| 15:14.42 | *** join/#asterisk etfonhomey (n=chatzill@66.148.161.90) |
| 15:14.43 | Ryushin | Does it need a newer version? |
| 15:14.52 | Qwell | dunno, but probably |
| 15:14.53 | [TK]D-Fender | enalert: I just told you where to go. |
| 15:15.01 | Ryushin | Okay, I'll try that. |
| 15:15.10 | Qwell | Ryushin: do you also have the -dev package of it? |
| 15:15.23 | Qwell | that might be the problem, but IMO, if you install one from source, install all from source |
| 15:16.25 | etfonhomey | [TK]D-Fender, other than the number of channels is there any limitation on the features (Caller ID, DNIS, etc.) on a BRI? |
| 15:16.30 | enalert | [TK]D-Fender, sorry didn't see it, thanks :) (I blame my IRC Client) |
| 15:17.45 | [TK]D-Fender | etfonhomey: I have never personally worked with one, but I believe the overall functionality is about the same... |
| 15:17.57 | *** join/#asterisk fskrotzki (n=fskrotzk@host198.textwise.com) |
| 15:18.03 | grandpapadot | Any recommendations for a SIP provider on the table these days? |
| 15:18.34 | etfonhomey | That may be the most asked question on here. I don't think any are very reliable unless they are also your ISP. |
| 15:18.35 | mocker | on the table? |
| 15:18.59 | coppice | desk phones |
| 15:19.00 | etfonhomey | 2nd most asked question is "What's the best IP phone?" |
| 15:19.03 | grandpapadot | mocker: i.e., since the last time the issue was debated, have any 'surfaced' to the top as above average? |
| 15:19.04 | Ryushin | Yea, I had the deb package too. |
| 15:19.13 | Ryushin | I'm downloading the libpri source now as well. |
| 15:19.15 | [TK]D-Fender | grandpapadot: For what area? |
| 15:19.34 | grandpapadot | [TK]D-Fender: US/South East |
| 15:19.37 | [TK]D-Fender | ~itsplist-us |
| 15:19.38 | jbot | [~itsplist-us] Here are some popular ITSPs (USA) starting with the more respected ones : http://www.teliax.com , http://connect.voicepulse.com , http://www.nufone.net , http://www.broadvoice.com |
| 15:19.39 | [TK]D-Fender | ^^^ |
| 15:19.39 | grandpapadot | [TK]D-Fender: I'm in bham. |
| 15:19.46 | grandpapadot | Thanks. |
| 15:19.54 | Qwell | [TK]D-Fender: broadvoice is respected? |
| 15:20.17 | grandpapadot | broadvoice sucks |
| 15:20.24 | [TK]D-Fender | Qwell: Sure they have issues, but they are at least "solid". Figure they should be removed? |
| 15:20.27 | grandpapadot | big green <edited> |
| 15:20.32 | Qwell | [TK]D-Fender: nah |
| 15:20.35 | coppice | Al Capone was respected. Its all a matter of viewpoint |
| 15:20.42 | grandpapadot | So was hitler |
| 15:21.08 | Qwell | [TK]D-Fender: I'm just trolling - they used to be A LOT worse a year or two ago |
| 15:21.16 | mocker | Isn't there some rule about invoking hitler? |
| 15:21.17 | cpm | So was [TK]D-Fender |
| 15:21.26 | Qwell | every few days somebody would come in asking if broadvoice was down - yet again |
| 15:21.39 | *** join/#asterisk novinder (n=Novinder@CPE000f664f0f37-CM0014045a95ea.cpe.net.cable.rogers.com) |
| 15:21.44 | Qwell | mocker: isn't there some rule about bringing up the rule? |
| 15:22.07 | mocker | ~hitlerrule |
| 15:22.07 | [TK]D-Fender | The first rule about Fight Club ..... |
| 15:22.10 | cpm | if someone invokes hitler, and no one case, is godwin invoked? |
| 15:22.24 | cpm | if someone invokes hitler, and no one cares, is godwin invoked? |
| 15:22.26 | cpm | rather |
| 15:22.33 | coppice | you cared |
| 15:22.40 | mocker | As far as ITSPs go, I've been using Vitelity for some time now w/ no real issue. |
| 15:22.49 | Qwell | there has to be a corollary drawn. |
| 15:22.50 | cpm | i only cared about the question |
| 15:23.02 | coppice | its enough |
| 15:23.04 | Qwell | read godwin's original post |
| 15:24.29 | coppice | there seemed to be a comparison of hitler and a telco. sounds like a fine corollary to me |
| 15:24.38 | cpm | ah, but there was nothing inflammatory about how 'h' was used in this context, so godwin's isn't invoked. |
| 15:24.52 | grandpapadot | mocker: We use and are please with Vitelity. Our customer base has grown so much we need to "spread it out" a bit and not have all of our eggs in one basket in the (unlikely) event that Vitelity goes away. |
| 15:25.09 | mocker | grandpapadot: Gotcha. |
| 15:25.21 | mocker | grandpapadot: You resell their service? |
| 15:25.25 | etfonhomey | I know of at least 2 ISP's who offer "Dynamic" T1's. What they're really giving you is a data T1 with a managed router on your premises. Then they QoS VoIP to the managed router and "convert" the channels to a PRI or they can even give you analog lines. |
| 15:25.41 | grandpapadot | mocker: My only issue with vitelity has been DTMF, we solved that by doing inband to vitelity, rfc2833 from. |
| 15:26.01 | etfonhomey | So, that when you have no voice calls, you get a full T1. Each active voice call takes a fraction of the T1's bandwidth. |
| 15:26.56 | grandpapadot | mocker: We have an "on-net" pop. |
| 15:27.03 | etfonhomey | The ISP in my area that is doing this is getting ready to offer SIP termination. |
| 15:27.11 | mocker | grandpapadot: I have no idea what that means. :P |
| 15:27.37 | grandpapadot | mocker: We have servers on or very close to vitelity's network (5 hops, < 2ms). |
| 15:28.06 | etfonhomey | I've been playing with Vitelity's service off and on for a few months. |
| 15:28.22 | etfonhomey | Any one else here have experience with BRI's? |
| 15:28.38 | grandpapadot | mocker: We've also written extensive PHP classes as wrappers for their simple web api, which is by far one of there better features. |
| 15:29.16 | mocker | grandpapadot: So you resell their service? :) |
| 15:29.54 | *** join/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
| 15:30.08 | grandpapadot | mocker: not really ... We sell a hosted PBX product with flat-rate fees ... We actually terminate through a number of ITSP's |
| 15:30.31 | grandpapadot | mocker: We even just recently started doing flat-rate international for $29.95/trunk (virtual trunk) |
| 15:30.36 | mocker | Ahh. |
| 15:30.39 | mocker | I've always been scared of that. |
| 15:30.47 | grandpapadot | mocker: Which part? |
| 15:30.51 | mocker | grandpapadot: Shoot me your URL. |
| 15:31.08 | mocker | grandpapadot: Relying on ITSPs. |
| 15:31.52 | etfonhomey | grandpapadot, where are you located? |
| 15:32.06 | grandpapadot | mocker: We've had really great results. There's occassional issues, but the low price of our service generally offsets the need to worry about it (from a customer's perspective). They save a fortune and get all the cool enterprise pbx features without a capital investment including the natural redundancy of our service. |
| 15:34.10 | *** part/#asterisk frigidzephyr (i=frigidze@nat/digium/x-0c7d8c57639bca6a) |
| 15:34.26 | *** join/#asterisk MrTelephone (n=MrTeleph@h697179-171.picriverisp.net) |
| 15:34.29 | MrTelephone | RUSSEL you there bud? |
| 15:34.33 | MrTelephone | :P |
| 15:34.37 | MrTelephone | I need some advice |
| 15:35.00 | grandpapadot | mocker: We have 4 pops, each with two OpenSER servers, two custom Asterisk servers, one MySQL server. We do push db replication with asterisk realtime and rsync evertyhing else between each pop. |
| 15:35.49 | mocker | grandpapadot: nice. |
| 15:35.49 | grandpapadot | etfonhomey: Our corporate office is in Birmingham, AL, USA. |
| 15:36.15 | grandpapadot | etfonhomey: In the shadow of our master, Digium. |
| 15:36.28 | grandpapadot | Nugget: Cool, what part of town? |
| 15:36.37 | Nugget | center point and alabaster |
| 15:36.53 | grandpapadot | I grew up in Center Point, then spent 8 years in the Navy, live in greystone now. |
| 15:36.57 | Nugget | cool |
| 15:37.02 | etfonhomey | I'm in Lexington, KY. We have an ISP here that's regional called Nuvox. Not sure they have a presence down there, but they are the ones rolling out SIP termination and the dynamic T1 product. |
| 15:37.02 | Nugget | I moved away in 2000. |
| 15:37.17 | grandpapadot | Center Point is a DMZ these days |
| 15:37.25 | Nugget | yeah, I believe it |
| 15:38.04 | grandpapadot | And we actually have a "Little Mexico" in hoover, it's hilarious, they just kind of made there own town. |
| 15:38.10 | Nugget | heh |
| 15:38.35 | Nugget | I lived there long enough ago to remember when Hoover and Riverchase were the happenin' places |
| 15:38.36 | grandpapadot | The local immagrants have changed my POV on immagration. |
| 15:38.46 | phillipk | I used to work at a restaraunt in Hoover. We'd drive over to little Mexico to hire dishwashers. |
| 15:38.53 | Nugget | way before greystone :) |
| 15:39.12 | coppice | don't Hoover make dishwashers? |
| 15:39.18 | grandpapadot | They work there ass off. I say we have a one-to-one swap program at the border: 1 welfare puppy for 1 hard working mexican. |
| 15:39.24 | Qwell | coppice: I hear they suck |
| 15:39.32 | Ryushin | Yea, it needed the latest version of libpri. I guess I won't be using debian's built pri on etch anymore. |
| 15:40.22 | *** join/#asterisk nirz (n=nnscript@bzq-79-178-22-251.red.bezeqint.net) |
| 15:40.34 | MrTelephone | qwell, p->fullcontact? |
| 15:40.46 | Qwell | ? |
| 15:41.14 | MrTelephone | I have some clients that don't switch to dns srv #2 very well.. I think its because the WWW-Authenticate response doesn't have a contact header |
| 15:41.28 | MrTelephone | p stands for peer right? |
| 15:41.29 | MrTelephone | :P |
| 15:41.44 | MrTelephone | Am I completely out to lunch here or what |
| 15:42.50 | datachomper | Is there anyway to get ser to register with a registrar, similar to the way you can register => sip@host with asterisk? |
| 15:42.51 | MrTelephone | I can see the client sends a register to srv #2, srv#2 responds with the new www-authenticate but the client responds to that on srv#1 again until it times out.. then it sends to srv#2. the delay causes asterisk to reject authentication |
| 15:43.06 | MrTelephone | stupid clients :( |
| 15:43.24 | MrTelephone | and i read through the feature bitmap and I can turn off multiple authorization headers |
| 15:43.26 | MrTelephone | hahaha |
| 15:43.44 | MrTelephone | so why didn't tech support tell me that? instead they just tell me its not their problem |
| 15:46.02 | billybongo | anyone know what's involved in doing 999 calls in the uk over sip? |
| 15:48.04 | MrTelephone | no |
| 15:48.10 | MrTelephone | what is a 999 call |
| 15:48.16 | billybongo | emergency call |
| 15:48.21 | *** join/#asterisk blitzrage (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 15:48.21 | *** mode/#asterisk [+o blitzrage] by ChanServ |
| 15:48.25 | MrTelephone | do it like any other call? |
| 15:48.45 | billybongo | I'm sure that's what happens in the end |
| 15:48.47 | blitzrage | do it like its 1984 |
| 15:48.48 | *** join/#asterisk Cyon (n=cyon@216.179.31.170) |
| 15:49.01 | billybongo | but it seems that they want you to register phone numbers and locations |
| 15:49.09 | billybongo | if you dial from a landline they know where you are |
| 15:49.23 | billybongo | if you dial from a mobile they triangulate your location |
| 15:49.29 | billybongo | if you dial from voip you could be anywhere |
| 15:49.34 | billybongo | apparently this causes them a problem |
| 15:49.52 | *** join/#asterisk mindCrime (n=chatzill@66.83.208.218.nw.nuvox.net) |
| 15:50.07 | billybongo | just wondering if anyone hear has been through the various hoops with this |
| 15:50.11 | billybongo | s/hear/here |
| 15:52.33 | MrTelephone | yeah its a problem everywhere |
| 15:52.37 | *** join/#asterisk IOscanner (n=IOscanne@76.187.195.124) |
| 15:53.03 | MrTelephone | in canada there is a law that says you have to warn your customers to tell the operator their location |
| 15:53.24 | MrTelephone | also there are companies that you forward calls to that transmit the location on a per DID or LDN basis |
| 15:53.31 | IOscanner | Is there a way to find out what DTMF tones are sent over a SIP channel. |
| 15:54.09 | MrTelephone | if your using rfc2853 dtmf the number is part of the rtp payload header? |
| 15:54.18 | blitzrage | MrTelephone: yep |
| 15:54.18 | *** join/#asterisk beasty (n=beasty@about/apple/macbook/beasty) |
| 15:54.21 | beasty | hi all |
| 15:54.33 | beasty | anyone knows how i configure a SIP trunk into my asterisk ? |
| 15:54.37 | blitzrage | I think wireshark will show you the DTMF in the RTP stream |
| 15:54.39 | MrTelephone | ~siptrunk |
| 15:54.40 | jbot | i heard siptrunk is Asterisk does not support SIP Trunks. Set trunk=no in sip.conf and then set up the device normally in sip.conf. |
| 15:54.45 | *** join/#asterisk ManxPower (n=manxpowe@46.sub-70-221-44.myvzw.com) |
| 15:54.54 | blitzrage | jbot: tell beasty about book |
| 15:54.58 | IOscanner | IT is inband DTMF carrier can't seem to get rfc to work. |
| 15:55.08 | blitzrage | beasty: I wrote a how-to in the book for how to do that |
| 15:55.14 | MrTelephone | too much packet jitter or loss for inband then |
| 15:55.23 | blitzrage | and that siptrunk thing is just wrong -- there is no trunk=foo option in sip.conf |
| 15:55.28 | blitzrage | that's an iax.conf thing |
| 15:55.33 | Qwell | blitzrage: exactly |
| 15:55.34 | beasty | jbot: tell beasty about book |
| 15:55.37 | MrTelephone | inband requires a clean network with no packet reorder |
| 15:55.38 | ManxPower | blitzrage: I know. I created that for a reason |
| 15:55.40 | beasty | what book blitzrage ? |
| 15:55.41 | Qwell | blitzrage: subtle humor |
| 15:55.57 | blitzrage | ManxPower: its slightly confusing... :) |
| 15:56.03 | ManxPower | People seem to INSIST there is such a thing as a "sip trunk". I got tired of arguing about it, so I created ~siptrunk |
| 15:56.05 | IOscanner | yep I am aware of that. Is there away to find out what DTMF tones are sent? |
| 15:56.14 | ManxPower | ~trunk |
| 15:56.15 | jbot | from memory, trunk is is a word with varying definitions. In Asterisk, a trunk is a "stream of UDP packets containing IAX2 frames from more than 1 call"; in telecom, a trunk is a "single voice channel between two pieces of switching equipment."; in Ethernet a trunk carries more than one 802.1q VLAN. There is no such thing as a "SIP Trunk" -- Don't use the term. |
| 15:56.18 | ManxPower | that is the correct one. |
| 15:56.32 | blitzrage | IOscanner: logger.conf --> console => warning,notice,dtmf |
| 15:56.37 | MrTelephone | mode +b manxpower |
| 15:56.37 | blitzrage | *CLI> logger reload |
| 15:56.38 | MrTelephone | :P |
| 15:56.44 | MrTelephone | mode +trunk manxpower |
| 15:56.55 | ManxPower | Leave my "trunk" alone! |
| 15:57.06 | MrTelephone | is it a coincidence that b is the first letter in bad AND ban? |
| 15:57.06 | IOscanner | that will show me inband DTMF. Great thanks. |
| 15:57.25 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 15:57.28 | MrTelephone | good knowledge blitzrage, i didn't know you could do that |
| 15:58.11 | MrTelephone | apparantly i don't know anything because I keep running into snags with my project |
| 15:58.36 | blitzrage | asterisk is a big peice of software -- always test before deploying |
| 15:58.45 | blitzrage | and when you've think you've tested enough -- test it 2 more times at least |
| 15:58.58 | IOscanner | you got that right |
| 15:59.10 | MrTelephone | i have everything working good.. im working on redundancy now.. again polycom and cisco dns srv works good.. but my other clients won't authorize with the second asterisk server |
| 15:59.15 | IOscanner | and keep the stupid admins out of it. |
| 15:59.22 | MrTelephone | both servers have the same credentials for the user and yet it won't authenticate |
| 15:59.56 | MrTelephone | i think the client waits too long to send the Authorization |
| 16:00.05 | iCEBrkr | blitzrage: Dude, dedicate a full day... I gave up thinking I can get things upgraded in just a few hours :P |
| 16:00.06 | MrTelephone | is there a time limit on that before asterisk says.. this client isn't responding? |
| 16:00.55 | blitzrage | iCEBrkr: a day? I spent 2 months at 12+ hours a day before I really "got" how things all worked together (this was before documentation existed) |
| 16:01.13 | rpyne | exit |
| 16:01.44 | iCEBrkr | blitzrage: oh oh oh, I thought you meant upgrading and stuff.. Yea, I didn't quite 'get' asterisk for a few months. |
| 16:01.47 | ManxPower | 2 months of 8 hour days learning Asterisk sounds about right. |
| 16:01.52 | beasty | blitzrage: i would love to read the book ... only thing is that my computer crashes when i open adobe reader |
| 16:02.15 | blitzrage | beasty: sounds like you have other problems |
| 16:02.20 | beasty | yeah |
| 16:02.21 | iCEBrkr | haha |
| 16:02.25 | beasty | i just hate plain pdf file |
| 16:02.27 | MrTelephone | my main sip client is designed for carrier grade sip application servers.. |
| 16:02.28 | iCEBrkr | RTFM *BOOM* |
| 16:02.34 | beasty | <3 text clients :p |
| 16:02.34 | MrTelephone | and breaks asteirsk continually |
| 16:02.35 | blitzrage | beasty: tfot.leifmadsen.com |
| 16:02.52 | MrTelephone | how much does a carrier grade sip server cost ? |
| 16:03.01 | Qwell | MrTelephone: 80 million dollars |
| 16:03.15 | blitzrage | iCEBrkr: oh for an upgrade from 1.2 to 1.4... expect to spend anywhere from 2 days to 2 months, depending on the complexity of your setup |
| 16:03.16 | twisted | i'll sell you one for 70 million, and i'll throw in the brookly bridge, too |
| 16:03.22 | twisted | *brooklyn |
| 16:03.27 | ManxPower | MrTelephone: "Carrier Grade"? That would be tens of thousands of dollars. |
| 16:03.40 | ManxPower | Perhaps you are looking for "corporate grade". |
| 16:03.41 | MrTelephone | ciscos is around 200k i know |
| 16:03.50 | blitzrage | too many people upgrade their production systems without having ever installed it on a test box |
| 16:03.58 | blitzrage | I have no pity for those people |
| 16:04.11 | twisted | blitzrage: But it's just supposed to work perfectly! |
| 16:04.22 | iCEBrkr | blitzrage: I dunno, moving over to the new extensions deal was a full nights work just for my home system :) |
| 16:04.25 | MrTelephone | dns srv is not handled properly by a lot of systems |
| 16:04.29 | ManxPower | blitzrage: I have pity for the people that do it the first time, I have no pity for people that do it more than once. |
| 16:04.48 | blitzrage | iCEBrkr: exactly |
| 16:04.52 | twisted | my favorite customer quote: |
| 16:04.56 | MrTelephone | polycom handles it AWESOME, if i iptables drop a phone from asterisk-1, it connects to asterisk-2 in seconds |
| 16:04.58 | ManxPower | You should EXPECT bugs and problems when you upgrade. These issues can frequently be "show stoppers". It is sad, but true. |
| 16:05.10 | twisted | "Why should there be an acceptance or testing period? This should just work if it's done properly" |
| 16:05.22 | blitzrage | took me 2 days even to get my home system setup just right (I was also trying some new things I hadn't done before, while at the same time trying to stablize my network... so I had a lot of other things going on :)) |
| 16:05.26 | ManxPower | I'm just glad my other software usually doesn't have that issue. |
| 16:05.44 | MrTelephone | can you guys tell me if there is a way to get more detail on the authentication rejection? |
| 16:05.48 | blitzrage | twisted: haha... ya... the problem is that people don't understand "properly" == "testing period" |
| 16:05.54 | blitzrage | sip debug |
| 16:05.59 | twisted | blitzrage ;) |
| 16:06.01 | *** join/#asterisk neoalex (n=chatzill@user-0ccengj.cable.mindspring.com) |
| 16:06.06 | MrTelephone | I tested my system in production and I told my customers that.. but they were still happy to use it |
| 16:06.20 | blitzrage | sometimes you have to save the customer from themselves |
| 16:06.34 | blitzrage | if they won't let you do that, then get new customers. The stress will kill you. |
| 16:06.34 | MrTelephone | I am a guinea pig for asterisk in a broadband production environment |
| 16:07.06 | MrTelephone | I had to disable threewaycalling cause too many people are hook flashing and then the ATA ringsback.. so many complaints about that |
| 16:07.14 | ManxPower | It would really suck if Firefox, Thunderbird, Postfix, Courier, or Apache upgrades screw up. |
| 16:07.26 | _x86_ | is it possible to use iaxmodem to provide a PPP session? |
| 16:07.32 | Qwell | ManxPower: they doscrew up |
| 16:07.46 | _x86_ | ManxPower: thunderbird updates fuck me over all the time |
| 16:07.50 | Qwell | heh |
| 16:07.56 | iCEBrkr | 11:06 <@blitzrage> if they won't let you do that, then get new customers. The stress will kill you. |
| 16:07.59 | iCEBrkr | AMEN |
| 16:08.07 | twisted | if only it were that simple. |
| 16:08.13 | neoalex | hi guys... I have a problem with a Wildcard X101P |
| 16:08.23 | ManxPower | Qwell: not EVERY release. 8-) |
| 16:08.30 | neoalex | specifically zttool shows a RED Alarm on it |
| 16:08.44 | neoalex | though there is a POTS line plugged in |
| 16:08.45 | iCEBrkr | neoalex: Did it ever work? |
| 16:08.45 | ManxPower | neoalex: that means "no line voltage detected" |
| 16:09.01 | ManxPower | neoalex: if you plug a phone into the 2nd port on the card, do you get dialtonr? |
| 16:09.03 | *** join/#asterisk juanjoc (n=juanjoc@190.2.0.145) |
| 16:09.07 | ManxPower | and dialtone too. |
| 16:09.34 | iCEBrkr | dialtonr... a reverse dialtone |
| 16:09.50 | _x86_ | heh |
| 16:09.52 | neoalex | I didn't try a phone... but I believe it worked with PBXnSIP or something I wasn't the one who set that up at the time |
| 16:10.07 | ManxPower | neoalex: you need to test it with a standard analog phone. |
| 16:10.14 | neoalex | what's a reverse dialtone |
| 16:10.15 | *** part/#asterisk harpal (n=Harpal@124.125.255.24) |
| 16:10.24 | ManxPower | neoalex: it was a typo. |
| 16:10.32 | neoalex | oh... :D |
| 16:10.33 | coppice | enotlaid |
| 16:10.40 | Qwell | coppice: telling |
| 16:10.42 | Qwell | so very very telling |
| 16:10.53 | iCEBrkr | LOL |
| 16:10.58 | neoalex | what lights should be on on the card? |
| 16:11.07 | neoalex | I have the first 3 ports and the last one is off |
| 16:11.20 | twisted | uhm |
| 16:11.26 | twisted | neoalex: a x101p is a single fxo card |
| 16:11.28 | *** join/#asterisk zydrunas (n=Miranda@216.49.228-ip-184.ckt.net) |
| 16:11.50 | neoalex | I have a 4 port card |
| 16:11.51 | twisted | you should have TWO ports, one for phone line in, the other for lifeline out |
| 16:11.54 | neoalex | 1 fxs 3 fxo |
| 16:11.55 | twisted | then you do not have an x101p |
| 16:12.08 | neoalex | which is what I said |
| 16:12.09 | ManxPower | neoalex: then you do not have an X101P or X100P |
| 16:12.17 | Qwell | <neoalex> hi guys... I have a problem with a Wildcard X101P |
| 16:12.20 | twisted | <neoalex> hi guys... I have a problem with a Wildcard X101P |
| 16:12.26 | neoalex | yes yes... sorry |
| 16:12.28 | MrTelephone | yeah its pretty stressful |
| 16:12.28 | neoalex | ok |
| 16:12.41 | MrTelephone | i have to do home installs as well as maintain the servers :( |
| 16:12.44 | neoalex | zttool shows a X101P and a TDM400P |
| 16:12.45 | ManxPower | neoalex: You just wasted 15 mins of our life, which we will never get back. |
| 16:12.54 | MrTelephone | integrating the lines with all their current household wiring |
| 16:12.55 | MrTelephone | :( |
| 16:13.08 | ManxPower | I'm not going to helpyou anymore if you are going to give wrong information |
| 16:13.17 | iCEBrkr | MrTelephone: Why the frown? Where's your sense of adventure? |
| 16:13.47 | neoalex | zttool says it's a Z101P but the card I know has 3 FXO and 1 FXS |
| 16:13.56 | neoalex | X101P rather |
| 16:14.06 | iCEBrkr | MrTelephone: I had my home system setup with a POTS gimmick and a VOIP provider for LD. |
| 16:14.06 | neoalex | it also shows a TDM400P |
| 16:14.40 | MrTelephone | icebrkr, im waiting for enough revenue to get a 20kw gensat |
| 16:14.45 | MrTelephone | if the power goes out im screwed |
| 16:15.11 | ManxPower | neoalex: The X101P and the TDM400P are totally different cards. |
| 16:15.13 | iCEBrkr | MrTelephone: Well, prior to DSL I had ISDN so, I was screwed if power went out regardless. |
| 16:15.18 | ManxPower | neoalex: Now go find out what cards you have and then come back |
| 16:16.03 | neoalex | I would but I'm 10 miles away, and it seems I'm also getting erroneous info from people there |
| 16:16.25 | ManxPower | neoalex: we can't help you if you don't have the correct information |
| 16:19.39 | *** join/#asterisk kand (n=kanderso@139.148.188.72.cfl.res.rr.com) |
| 16:19.59 | neoalex | ok... it seems I have two cards in that system the TDM400P shows up as ok... how do I configure that and just ignore the X101P |
| 16:20.15 | MrTelephone | find out what span is what |
| 16:20.18 | ManxPower | neoalex: you configure the card and ignore the other one.l |
| 16:20.21 | MrTelephone | cat /proc/zaptel/1 or 2 |
| 16:20.34 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
| 16:20.37 | ManxPower | the best bet is, of course, to remove the card, the 2nd best is stop loading the kernel module for that card on boot. |
| 16:20.44 | MrTelephone | yeah that too |
| 16:20.54 | kand | hey all, I have had a one way audio issue for callers coming off hold and Level 3 has said it is because my RTP timestamps are slipping. Anybody know what could cause this? |
| 16:21.23 | MrTelephone | maybe asterisk 1.2.25 is bunk and not authorizing properly |
| 16:21.32 | _x86_ | is there a way to use a zap channel and iaxmodem to provide "dial-in mgetty" ? |
| 16:21.46 | neoalex | http://pastebin.com/d4dc86016 |
| 16:22.42 | MrTelephone | just got a call, "can't get into voicemail with *98" solution: press talk first |
| 16:22.43 | MrTelephone | hhaha |
| 16:23.43 | iCEBrkr | all your voicemail are belong to us! |
| 16:23.50 | neoalex | ok which is the kernel module for X101P |
| 16:24.26 | ManxPower | neoalex: wcfxo, which you would know if you read the README in the Zaptel source directory. |
| 16:24.32 | MrTelephone | my voicemail :P |
| 16:24.40 | MrTelephone | like the cookie monster likes cookie |
| 16:24.40 | MrTelephone | s |
| 16:25.10 | neoalex | ManxPower: you're right I should come back after some more RTFM-ing |
| 16:25.37 | *** join/#asterisk Deeewayne (i=dwayne@nat/digium/x-e3de0baa6758bebb) |
| 16:25.37 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 16:26.34 | MrTelephone | why would asterisk completely ignore a register attempt from a client |
| 16:26.48 | mocker | MrTelephone: Firewall. |
| 16:26.54 | iCEBrkr | Talk to the hand, cuz Asterisk doesn't want to hear it |
| 16:27.18 | MrTelephone | maybe its a netmask mistake or something on the interface? |
| 16:27.18 | MrTelephone | hmm |
| 16:27.29 | MrTelephone | it registers with my accounts with usernames with 3 characters |
| 16:27.40 | MrTelephone | is there a delinter script for asterisk configs? |
| 16:27.53 | ManxPower | MrTelephone: no |
| 16:28.15 | MrTelephone | im using all 1.2.24 configs with 1.2.25.. so im going to compile 1.2.24 and see if it keeps happening |
| 16:28.20 | *** join/#asterisk dijungal (n=kdaniel@63.175.159.171) |
| 16:28.27 | MrTelephone | according to changelog though there was only a couple changes |
| 16:28.27 | dijungal | hello |
| 16:28.40 | iCEBrkr | MrTelephone: Famous last words! |
| 16:28.50 | MrTelephone | hahaha |
| 16:28.56 | MrTelephone | yeah I hear ya partner |
| 16:29.22 | dijungal | is it best practice to store recordings on the same server as asterisk or a separate server? |
| 16:29.48 | ManxPower | dijungal: Yes. |
| 16:29.52 | iCEBrkr | lol |
| 16:30.10 | iCEBrkr | dijungal: Depends if you want to put all your eggs in one basket. |
| 16:30.12 | dijungal | ManxPower: why? |
| 16:30.43 | ManxPower | dijungal: because it depends on many things, including codecs, number of voicemail boxes, system speed, etc. |
| 16:31.10 | dijungal | what i'm trying to decide is should i tell asterisk to write the recordings from the queue on a separate server or on the same server then move them later |
| 16:31.45 | ManxPower | dijungal: there isn't a whole lot of difference from an Asterisk perspective |
| 16:31.45 | blitzrage | store them in the database :) |
| 16:31.47 | iCEBrkr | dijungal: What would eat up less CPU? |
| 16:31.57 | iCEBrkr | blitzrage: um, ew. |
| 16:31.59 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 16:32.01 | coppice | tip them down the drain |
| 16:32.13 | blitzrage | I'd write them to the same drive, then batch them off to a remote server |
| 16:32.30 | iCEBrkr | "for safe keeping" |
| 16:33.12 | iCEBrkr | I think I'd soxmix'm at night over to another server. |
| 16:33.23 | dijungal | i would think putting on the same server then moving after shift when the server is idle would be beter |
| 16:33.39 | ManxPower | dijungal: is your server ever idle? |
| 16:33.41 | dijungal | i just do a find and move |
| 16:33.43 | dijungal | yes |
| 16:33.52 | iCEBrkr | So your Asterisk box doesn't get loaded up with archived calls which will save the head of adding more diskspace later, or having to shuffle those archives off |
| 16:33.57 | dijungal | it only works for about 9 to 7 |
| 16:34.27 | iCEBrkr | s/head/heahache |
| 16:34.53 | ManxPower | dijungal: I think you need to see a therapist about this obsession with being vague. |
| 16:35.31 | dijungal | here's my overall issue i was having REALLY bad call quality, so i restarted the server, linux forced a disk check and made some fixes, now the audio on the calls are great. So i'm wondering if my current process of storing the recordings on the server and then moving it overnight is causing a disk issue... |
| 16:35.35 | ManxPower | If it's 9am to 7am then you have only 2 hours of idle time, if that is 9am to 7pm, then you have many hours of idleness. |
| 16:35.52 | coppice | through most of history being vague was excellent protection against getting your head chopped off |
| 16:35.54 | ManxPower | dijungal: No it did not. |
| 16:36.04 | iCEBrkr | dijungal: Don't process the calls during peak hours. |
| 16:36.10 | iCEBrkr | dijungal: That chews up CPU |
| 16:36.14 | *** join/#asterisk techie (n=techie@adsl-76-214-26-98.dsl.lsan03.sbcglobal.net) |
| 16:36.28 | *** join/#asterisk Zeeek (n=IceChat7@pdpc/supporter/active/Zeeek) |
| 16:36.37 | dijungal | ICEBrkr: indeed |
| 16:36.37 | iCEBrkr | dijungal: Plus, that channel isn't released until after the encoding has completed. |
| 16:36.45 | dijungal | k |
| 16:36.46 | *** join/#asterisk hfb (n=hfb@pool-71-106-219-180.lsanca.dsl-w.verizon.net) |
| 16:36.57 | dijungal | so the best thing to do is what i am currently doing |
| 16:37.12 | *** join/#asterisk kand (n=kanderso@139.148.188.72.cfl.res.rr.com) |
| 16:37.13 | dijungal | store them locally then move them off on low periods? |
| 16:37.19 | ManxPower | dijungal: did you ever tell us your VERSION of Asterisk? |
| 16:37.19 | blitzrage | sure |
| 16:38.11 | iCEBrkr | god, the asterisk page is ugly. |
| 16:38.20 | iCEBrkr | who's idea was this? |
| 16:38.36 | iCEBrkr | I suppose I should say 'still ugly' |
| 16:38.54 | ManxPower | #asterisk-newbie: What is causing the headache I have. [30 mins of troubleshooting pass] #asterisk-newbie: I drank a quart of whiskey last night. Do you think that might have anything to do with it. |
| 16:39.07 | Zeeek | get on up |
| 16:39.28 | dijungal | ManxPower: Sorry Sir ASterisk 1.4.15 |
| 16:39.45 | kand | Can anybody help me with issue in my RTP stream? In one example, the timestamps for callers placed on hold increments by 41,120 while the wireshark capture time only increments by .123238. What would cause something like this, is it an * bug? |
| 16:39.46 | dijungal | all the latest downloaded from the ugly asterisk website |
| 16:40.00 | *** join/#asterisk c0rnflake (n=c0rnflak@38.112.4.210) |
| 16:40.15 | ManxPower | kand: Asterisk does not support silence supression, that might have something to do with it. |
| 16:42.06 | iCEBrkr | Dude.. Is there a changelog online somewhere? So I don't have to download asterisk? |
| 16:42.38 | iCEBrkr | http://downloads.digium.com/pub/asterisk/ChangeLog-1.4.15 |
| 16:42.40 | iCEBrkr | ah |
| 16:42.44 | kand | ManxPower: It is disabled |
| 16:43.17 | ManxPower | kand: Level3 has silence supression disabled? |
| 16:43.50 | kand | ManxPower: That is what they tell me. |
| 16:44.24 | *** part/#asterisk ManxPower (n=manxpowe@46.sub-70-221-44.myvzw.com) |
| 16:44.34 | Zeeek | VOIP Users Unite! http://VoipUsersConference.org IRC #voip-users-conference right now |
| 16:44.46 | *** join/#asterisk myiagy (n=myiagy@200.215.59.133) |
| 16:46.25 | Zeeek | or not. |
| 16:47.41 | *** join/#asterisk mindCrime (n=chatzill@66.83.208.218.nw.nuvox.net) |
| 16:50.03 | beasty | ok |
| 16:50.25 | beasty | anyone knows how to a handle a single sip account with multiple numbers on it ? |
| 16:52.17 | _x86_ | eh? |
| 16:52.34 | _x86_ | Dial(SIP/number@peer|100|t) ? |
| 16:52.42 | _x86_ | usually how I do it |
| 16:52.50 | _x86_ | well, I usually use IAX though ;) |
| 16:53.06 | beasty | that's why my voip provider also sugested |
| 16:53.14 | beasty | but now i'm waiting on his reply |
| 16:53.52 | *** join/#asterisk Defraz (n=t0tal@fw.fuzecore.com) |
| 16:58.02 | blitzrage | beasty: Dial(SIP/peer_setup_in_sip_conf/${EXTEN},30) |
| 16:58.47 | MrTelephone | if you were to choose a type for registrar and you had the options ipv4 and dns.. what would dns be? |
| 17:00.27 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 17:00.33 | dioedu | hi, can someone explain me why asterisk send a link and a unlink manager message in every DTMF received in sip channels ? |
| 17:00.41 | iCEBrkr | wow there's a lot of fixes since 1.4.6 |
| 17:00.51 | russellb | iCEBrkr: ha, yeah ... |
| 17:00.54 | russellb | hundreds |
| 17:01.02 | russellb | in fact, i have a script that tells me how many, let me check :) |
| 17:01.06 | iCEBrkr | I'm paging through the changelog. |
| 17:01.09 | iCEBrkr | hahaha |
| 17:01.22 | russellb | $ ./changes_since asterisk 1.4.6 |
| 17:01.22 | russellb | 763 |
| 17:01.31 | iCEBrkr | haha nice |
| 17:01.42 | MrTelephone | russellb, how come there is no contact in www-authenticate 401 not authorized reponses? |
| 17:01.43 | iCEBrkr | russellb: I see your name on quite a few :P |
| 17:02.08 | MrTelephone | your going to see my name on a tombstone soon |
| 17:02.20 | iCEBrkr | MrTelephone: Sweet, what's it's extension? |
| 17:02.36 | russellb | MrTelephone: i know nothing |
| 17:02.39 | iCEBrkr | hrrm, there appears to be a bunch of memory leak fixes. |
| 17:03.03 | iCEBrkr | I wonder if this is what's causing my system to act up after awhile. |
| 17:03.03 | russellb | iCEBrkr: heh, i don't doubt it ... i'm one of the lucky ones that gets paid to work on asterisk full-time |
| 17:03.11 | iCEBrkr | haha |
| 17:03.32 | MrTelephone | 1800tombstone |
| 17:03.40 | MrTelephone | are they accepting applicants? |
| 17:03.46 | MrTelephone | im joining |
| 17:04.03 | iCEBrkr | russellb: It's a bit early to say, but if things go well, I'll be working for Kristian. Everything is still fairly liquid, so who knows. |
| 17:04.20 | russellb | awesome, that would be fun |
| 17:04.22 | endre | who is kristian? |
| 17:04.29 | iCEBrkr | endre: The astLinux guy |
| 17:04.30 | endre | or what :) |
| 17:04.39 | endre | oh i see |
| 17:04.56 | endre | good for you |
| 17:05.01 | endre | that would be awesome |
| 17:05.14 | iCEBrkr | The job description sounds like a job made in heaven. |
| 17:05.21 | iCEBrkr | :P |
| 17:05.28 | endre | i demand a link |
| 17:05.33 | iCEBrkr | But I think that's because I'm trapped in this ASP shop at the moment. |
| 17:05.56 | iCEBrkr | I'd really like to get back into a linux based environment. |
| 17:06.37 | iCEBrkr | Actually, I think I just want a job that I'm happy at :P |
| 17:07.13 | iCEBrkr | I think working for a linux based shop is a plus---- working with Asterisk is just a extra headache^H^H^H^H^Herrr bonus |
| 17:08.42 | MrTelephone | russelb, does asterisk discard the www-authenticate response if no peer authorization is received.. after a period of time? |
| 17:08.54 | russellb | MrTelephone: i don't know |
| 17:09.00 | MrTelephone | I got a client that takes 5 seconds to respond to the digest |
| 17:09.08 | MrTelephone | by that time asterisk forgets and issues a new nonce |
| 17:09.13 | russellb | it probably does, yeah |
| 17:09.23 | *** join/#asterisk thansen|laptop (n=thansen@19.243.sfcn.org) |
| 17:09.30 | *** join/#asterisk Navion (n=billp@75-105-41-123.cust.wildblue.net) |
| 17:09.31 | MrTelephone | is it a global timeout ? :( |
| 17:09.31 | iCEBrkr | Hrrm, I think I'll 'upgrade' to 1.4.15 |
| 17:10.07 | iCEBrkr | See if that clears up my issues |
| 17:10.43 | MrTelephone | why would a client send the digest back to the server it just failed to get a response from.. I'm mad at the japanese right about now |
| 17:11.28 | *** join/#asterisk Strom_M (n=strom@208.127.172.112) |
| 17:11.52 | MrTelephone | client resq -> dns srv 1 -> timeout client resq -> dns srv 2 -> response from asterisk client resq (authorization) -> dns srv1 timeout, then it send to dns srv2 but then its too late |
| 17:12.27 | _x86_ | using IAXmodem + zap channel, is it possible to get agetty to listen on the "modem"'s device entry and allow for remote dial-in access? |
| 17:16.15 | coppice | ask yourself "what is iaxmodem?" |
| 17:16.35 | *** join/#asterisk objective (n=objectiv@ool-43536c3b.dyn.optonline.net) |
| 17:17.17 | MrTelephone | reminds me of internet dialup |
| 17:20.37 | myiagy | i'm having trouble with pickup application, it appears to run correctly, -- Executing Pickup("SIP/111-b0411ce0", "107@ramais") in new stack |
| 17:20.50 | myiagy | the context [ramais] has the exten => 107,1,dial(SIP/107) |
| 17:21.32 | myiagy | all extensions have the same callgroup/pickupgroup.. but once the Pickup runs, it immediatly exits non-zero.. it doesn't answer the call.. i tried searching the mailing lists, but got nothing.. |
| 17:21.40 | myiagy | any ideas where i might have wronged? |
| 17:22.14 | *** join/#asterisk Simon-- (n=sim@staff-nat.netnation.com) |
| 17:22.31 | *** join/#asterisk fluidicslave (n=fluidics@adsl-75-36-221-71.dsl.pltn13.sbcglobal.net) |
| 17:22.39 | Simon-- | is there a way to turn on zaptel link status logging? eg: it would be really nice if it just logged "link down" like nic drivers do :) |
| 17:23.40 | Simon-- | hrm. maybe it was just signalling and not link, because link gives "alarm"s... |
| 17:25.45 | *** join/#asterisk NightMonkey (n=NightMon@pdpc/supporter/sustaining/NightMonkey) |
| 17:30.50 | *** join/#asterisk blepsoaf (n=pbaker@167.206.216.189) |
| 17:31.32 | fluidicslave | are there any known problems with zapbarge that might explain a signfigant number of warnings and failures on zap chans |
| 17:32.14 | MrTelephone | does asterisk store the last nonce? |
| 17:34.11 | fluidicslave | I had some one perform a zapbarge yesterday and affter an error setting conference it seems like all hell borke lose on ever channel that was zapbarged affter that |
| 17:38.56 | *** join/#asterisk jarg (n=jarg@200.56.225.61) |
| 17:39.15 | *** join/#asterisk ManxPower (n=manxpowe@129.sub-75-200-103.myvzw.com) |
| 17:40.56 | *** join/#asterisk atisss (n=atisss@193.238.212.171) |
| 17:41.33 | MrTelephone | does asterisk generate a new nonce every so often or something? |
| 17:41.38 | *** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net) |
| 17:41.50 | *** join/#asterisk RoyK (n=roy@ip-56-15-149-91.dialup.ice.no) |
| 17:42.51 | *** join/#asterisk ptblank (n=MURDER1@cpe-75-84-47-107.socal.res.rr.com) |
| 17:43.33 | MrTelephone | SIP AUTO DESTORY |
| 17:43.34 | MrTelephone | hmm |
| 17:44.17 | _x86_ | SIP AND DESTROY |
| 17:44.25 | _x86_ | that's how that metallica song should go ;) |
| 17:44.29 | MrTelephone | my clients timeout at 20 seconds so I changed it to 25 |
| 17:45.53 | MrTelephone | hahaha |
| 17:46.21 | MrTelephone | I love that ngrep tool |
| 17:46.28 | MrTelephone | ngrep -W byline -t -port 5060 |
| 17:46.54 | MrTelephone | it kicks some serious rear end |
| 17:47.43 | MrTelephone | why the hell do you need a 20 second timeout on a dns srv response |
| 17:47.44 | MrTelephone | beats me |
| 17:49.01 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 17:53.02 | *** join/#asterisk spatulamaan (n=ggilmore@207.188.8.252) |
| 17:54.52 | florz | can someone in here explain why the chan_cip source as well as the default sip.conf claim that SIP wasn't capable of hairpin calls? |
| 17:56.06 | MrTelephone | what is a hairpin call |
| 17:56.35 | *** join/#asterisk cjk (n=cjk@d90-129-18-139.cust.tele2.lu) |
| 17:56.54 | *** join/#asterisk bantu (n=Miranda@p54A32A1D.dip0.t-ipconnect.de) |
| 17:56.57 | florz | I'm not sure in this particular case, either, but in general it's a call that somehow is "routed back to the part of the network it came from" |
| 17:57.06 | cjk | hi, can my ata box with a fax connceted dial out a zap channel and send faxes over t38? |
| 17:57.50 | MrTelephone | right |
| 17:57.58 | MrTelephone | which indicates a loop condition |
| 17:58.06 | MrTelephone | i think asterisk checks for tag= |
| 17:58.10 | florz | MrTelephone: hu? |
| 17:58.13 | MrTelephone | im not 100% sure |
| 17:58.19 | *** join/#asterisk dfas (n=none@10.201.216.81.static.s-o.siw.siwnet.net) |
| 17:58.48 | MrTelephone | florz, its a common problem when using openser with asterisk |
| 17:59.27 | florz | MrTelephone: I mean, routing loops certainly could be considered a subset of hairpin calls, but that wouldn't make every hairpin call a loop!? |
| 18:00.08 | MrTelephone | for some reason asterisk rejects a loop even if it doesn't hit max forwards? I read that on the web in some opernser documentation |
| 18:00.19 | florz | MrTelephone: Yeah, I'm kindof having that problem with OpenSER + Asterisk |
| 18:01.10 | florz | MrTelephone: Yeah, Asterisk by default considers just an equal Call ID on incoming request and outgoing transaction a "loop" |
| 18:01.11 | coppice | cjk: no |
| 18:01.49 | florz | MrTelephone: Which is what blows up when rewriting the R-URI with OpenSER and then routing the request back to the asterisk it came from ... |
| 18:01.52 | cjk | coppice, hmmm any other solution? |
| 18:02.32 | *** part/#asterisk dfas (n=none@10.201.216.81.static.s-o.siw.siwnet.net) |
| 18:02.51 | MrTelephone | florz, im sure there must be a resolution on the web.. i didn't get that far to look at it |
| 18:03.30 | MrTelephone | take out the comparison of callid routine in chan_sip.c |
| 18:04.05 | *** part/#asterisk Simon-- (n=sim@staff-nat.netnation.com) |
| 18:04.20 | florz | MrTelephone: Well, as usual ... ;-) - I did find a bug report in the BTS with a patch attached, but that one somehow manages to break registration, not sure yet exactly why ... |
| 18:04.35 | MrTelephone | im getting really pissed off because asterisk times out the sip auth response before my client responds |
| 18:04.50 | *** join/#asterisk hohum (n=dcorbe@h-74-1-66-114.lsanca54.covad.net) |
| 18:05.17 | *** join/#asterisk atomicd (n=Atomicd@74-206-0-80.static-ip.m.telepacific.net) |
| 18:05.24 | MrTelephone | florz, that sucks.. whats the error message? |
| 18:05.36 | MrTelephone | compare your sip messages before and after |
| 18:05.39 | rbd | De_Mon: The problem I was running into with Playback not working earlier that you helped with was due to ztdummy/ACPI conflict: http://forums.digium.com/viewtopic.php?p=46221&sid=410ecce657b99d06ed012a8062a01aa4 |
| 18:05.54 | beasty | anyone ever see this error ? |
| 18:05.55 | beasty | [Dec 7 19:02:46] WARNING[14782]: chan_sip.c:2841 sip_call: No audio format found to offer. Cancelling call to gnelisse |
| 18:07.19 | De_Mon | I didn't think ztdummy was involved in simple Playback() |
| 18:07.53 | De_Mon | interesting.. thanks for the update rbd |
| 18:08.37 | florz | MrTelephone: It's just 401, but I got already a bit further in the source as to how this happens - I just don't know yet what the logic error in the patch is |
| 18:08.43 | rbd | De_Mon: no problem. thanks for your help earlier! |
| 18:09.46 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 18:10.07 | MrTelephone | what does the patch look like |
| 18:10.08 | MrTelephone | pastbin it |
| 18:10.17 | *** join/#asterisk Victor_Yure (n=aaa@postfix.tradein.com.br) |
| 18:11.41 | florz | MrTelephone: somehow it manages not to correlate new REGISTER transactions with previous ones with the same call-id, which is why the authentication code doesn't have access to the nonce it sent in the transaction before, so it keeps issuing new nonces ... |
| 18:11.55 | florz | MrTelephone: http://bugs.digium.com/view.php?id=7403 |
| 18:12.12 | florz | MrTelephone: the sip_spiral.patch one |
| 18:13.18 | florz | MrTelephone: But actually I was just asking that question because I was curious whether there actually is a valid reason to claim that SIP "can't to hairpin calls". |
| 18:14.18 | MrTelephone | people think "why should a call goto a proxy where it came from"? |
| 18:14.23 | coppice | its like Yugos can't do hairpin bends |
| 18:14.44 | MrTelephone | because your application servers such as voicemail shouldn't be on the proxy server |
| 18:15.11 | MrTelephone | its like you want to travel west in your car but you start off traveling east first and goto the coast and then start driving west |
| 18:15.34 | MrTelephone | proxy servers aren't supposed to act like sine waves |
| 18:15.52 | florz | hu? =:-) |
| 18:16.11 | *** join/#asterisk D|eHeLL (i=D_eHeLL@170.57.49.60.klj02-home.tm.net.my) |
| 18:16.12 | MrTelephone | a call shouldn't hit the same proxy server twice |
| 18:16.21 | florz | why not? |
| 18:16.25 | MrTelephone | but it will if your pstn access or voicemail is on the same box as your proxy |
| 18:16.31 | beasty | [Dec 7 19:02:46] WARNING[14782]: chan_sip.c:2841 sip_call: No audio format found to offer. Cancelling call to gnelisse |
| 18:16.33 | MrTelephone | then you have no choice |
| 18:16.35 | *** join/#asterisk Buhntz (i=Boones@port-212-202-41-252.dynamic.qsc.de) |
| 18:16.35 | beasty | anyone knows this error ? |
| 18:17.03 | MrTelephone | why would an invite goto a remote proxy then back to your originating proxy? |
| 18:17.13 | MrTelephone | it doesn't make sense but it does with asterisk |
| 18:17.27 | MrTelephone | because you want to hit voicemail after trying to reach your sip client remotely |
| 18:17.41 | MrTelephone | anyone can correct me if im wrong |
| 18:17.56 | florz | MrTelephone: I think it makes perfect sense. If the remote user whishes to redirect calls somewhere else, why shouldn't he? |
| 18:18.35 | MrTelephone | i think in big scenerios it doesn't happen |
| 18:18.49 | MrTelephone | if your big enough to need openser you should have seperate pstn access and voicemail boxes |
| 18:19.09 | MrTelephone | then you won't have a looping problem |
| 18:19.11 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 18:19.29 | MrTelephone | florz, like i said I will run into the same situation is you soon |
| 18:19.38 | MrTelephone | I'm tired of fighting with asterisk as a registrar |
| 18:20.29 | florz | MrTelephone: Well, it doesn't happen because what is currently being done isn't quire "VoIP" but rather IP-mediated PSTN access, I guess. But look at email: It's perfectly valid to redirect mails coming from domain a to domain b back to (a different account at) domain a, no? |
| 18:20.45 | MrTelephone | I can't even get asterisk to hold a registration attempt in memory for 25 seconds |
| 18:21.40 | MrTelephone | florz, yeah.. I just think asterisk was programmed to be over protected |
| 18:21.48 | florz | MrTelephone: And actually, the scenario I am trying to solve is this: Asterisk is the PSTN gate, call comes in, is routed to OpenSER, OpenSER has no registration, but a fallback PSTN number, so it should route the call back to the PSTN gate, billed to the account whose number has been called. |
| 18:22.18 | MrTelephone | it wont' do that without error? |
| 18:22.39 | florz | MrTelephone: No, asterisk rejects that with a 482 (Loop detected) |
| 18:22.41 | MrTelephone | dial(openserbox) and if that fails dial(zap/1) |
| 18:22.48 | florz | no |
| 18:22.57 | MrTelephone | don't tell openser to do the forwarding of the failure |
| 18:23.06 | *** join/#asterisk bkruse (i=bkruse@nat/digium/x-0ee9b92726190abb) |
| 18:23.06 | *** mode/#asterisk [+o bkruse] by ChanServ |
| 18:23.09 | florz | not a 302, just rewriting the URI and forwarding the request back to Asterisk |
| 18:23.27 | MrTelephone | why don't you let asterisk handle the failure? |
| 18:24.37 | MrTelephone | the idea of manipulating a sip message by rewriting the uri may be a bad solution.. if you do that you have to erase the callid or change it |
| 18:24.52 | MrTelephone | or you might have to branch the call instead? |
| 18:25.02 | MrTelephone | but I don't know much about branching |
| 18:25.04 | *** join/#asterisk slima (i=slima@unaffiliated/slima) |
| 18:26.06 | florz | MrTelephone: Hu? No, rewriting the URI is perfectly OK. That exactly what a SIP registrar does - or at least one option it has, the other would be redirection. |
| 18:26.12 | florz | +'s |
| 18:26.47 | MrTelephone | in asterisk extensions you should do exten => 1,Dial(openserbox) | exten => 2,Dial(localpstn) |
| 18:27.11 | florz | MrTelephone: After all: If there were a registration, it would do exactly that: replace the URI with the one from the registered contact. |
| 18:27.14 | MrTelephone | what happens is openser responds with a Not Available and asterisk says, ok, fuck you then, :P and dials out the pstn |
| 18:27.34 | MrTelephone | florz, yeah you got me on that one |
| 18:27.58 | florz | And how do I tell Asterisk who is gonna pay for it? |
| 18:28.07 | MrTelephone | its just that its the same "CALL" and you redirected it back to asterisk |
| 18:28.13 | *** part/#asterisk Zeeek (n=IceChat7@pdpc/supporter/active/Zeeek) |
| 18:28.43 | *** join/#asterisk codazoda (n=chatzill@70-96-185-203.directbb.com) |
| 18:29.04 | MrTelephone | not sure |
| 18:30.10 | florz | Well, that's why the current strategy in principle would be very nice - it's just an outgoing call, not much different from one coming from one of the "real" OpenSER clients |
| 18:30.49 | MrTelephone | before it dials pstn set a billing flag or change one of your cdr variables |
| 18:31.13 | *** join/#asterisk ice_croft (n=nolan@213.132.86.246) |
| 18:31.13 | MrTelephone | someone calls from the pstn to a sip client? the sip client isn't there so you forward the call out the pstn to who? |
| 18:31.16 | ice_croft | hi ppl |
| 18:31.28 | MrTelephone | I don't 100% understand the configuration your looking for |
| 18:31.36 | ice_croft | where can i get rc.d script for freebsd? |
| 18:32.05 | ice_croft | * installed from cources |
| 18:32.07 | ice_croft | * installed from sources |
| 18:32.34 | florz | MrTelephone: Well, there is a DB with all the account information, amongst which there is a field for a E.164 number that calls are supposed to be redirected to in case OpenSER doesn't have any registrations for that account. |
| 18:32.55 | florz | MrTelephone: OpenSER uses that DB for all its routing and authentication. |
| 18:33.25 | *** part/#asterisk atomicd (n=Atomicd@74-206-0-80.static-ip.m.telepacific.net) |
| 18:33.56 | florz | MrTelephone: Where OpenSER is basically to interface to the "Internet", that is, customers. |
| 18:34.02 | florz | s/to/the/ |
| 18:34.51 | ice_croft | gentlemen, where can i get rc.d script for freebsd? |
| 18:34.53 | ice_croft | please |
| 18:36.20 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 18:37.32 | ice_croft | thanx anyway |
| 18:41.57 | *** join/#asterisk mcab (n=mb@mostly-harmless.ca) |
| 18:42.55 | hmmhesays | good lord itsp's suck some days |
| 18:43.54 | MrTelephone | florz, oh I see now.. you'll have to figure out the looping issue i guess |
| 18:46.19 | *** join/#asterisk NightMonkey (n=NightMon@pdpc/supporter/sustaining/NightMonkey) |
| 18:54.13 | *** join/#asterisk vale-ICS (n=vale@icsnet.demon.co.uk) |
| 19:02.31 | blitzrage | hmmhesays: just some? :) |
| 19:09.10 | *** join/#asterisk roe_ (n=roe___@216-164-160-36.c3-0.eas-ubr10.atw-eas.pa.static.cable.rcn.com) |
| 19:09.40 | roe_ | does anyone know of a sip presence monitor (kind of like a buddy list)? |
| 19:12.00 | kand | roe_: the only one I know of is the Flash operator panel at http://www.asternic.org/ |
| 19:12.23 | roe_ | nothing for ekiga or xlite or any other softphone? |
| 19:12.49 | kand | roe_: sorry, none that I know of |
| 19:13.27 | roe_ | rats, and a flash implemtation also makes me sad, oh well thanks anyway |
| 19:14.53 | *** join/#asterisk angom (n=angom@200.38.31.239.dsl.dyn.telnor.net) |
| 19:14.55 | kand | roe_: Just on a note, the FOP has ajax based display. I have used that to some effect as a narrow browser window on a few of my clients. |
| 19:16.08 | hmmhesays | most end users are morons also |
| 19:16.31 | mocker | roe_: For awhile there was a jabber server that would show when someone was on a call. |
| 19:16.37 | mocker | Let me find the link. |
| 19:16.45 | roe_ | thanx |
| 19:17.07 | mocker | http://www.igniterealtime.org/projects/openfire/index.jsp |
| 19:17.20 | ice_croft | ppl, a question. what's zapata channel module depends of? |
| 19:17.56 | ice_croft | it's disabled in menuselect, i'm dispaired |
| 19:18.32 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 19:19.53 | ice_croft | any advices? |
| 19:21.34 | kand | ice_croft: Probaly wont help but on a fedora base install all I need are compilers and kernel-devel (and usbutils for the newer zaptel) |
| 19:26.04 | *** part/#asterisk Victor_Yure (n=aaa@postfix.tradein.com.br) |
| 19:28.52 | ice_croft | kand> i'm on freebsd, and zaptel's intalled |
| 19:31.01 | codazoda | I want to use mixmonitor to record all calls to an IVR system. I need it to pick a random filename for each call recording. What's the best way to do this? |
| 19:31.18 | kand | ice_croft: I dont really know freebsd but mabey ./configure --with-zaptel=PATH ? |
| 19:31.29 | *** join/#asterisk gleydson_barbosa (n=gleydson@201.20.71.12) |
| 19:32.18 | ice_croft | kand> i did. i red it needs libpri. where can i get it? |
| 19:32.47 | *** join/#asterisk pjezek (n=pj@193.85.164.154) |
| 19:32.57 | kand | ice_croft: http://downloads.digium.com/pub/libpri/ |
| 19:32.57 | gleydson_barbosa | hi, i need help about modem over voip! |
| 19:33.51 | florz | gleydson_barbosa: That's easy: forget it. |
| 19:33.52 | codazoda | Oh, actually, I think I'll use RAND. Then I can keep my calls to a limited number as well (overwriting some with newer ones), which is fine and will keep the disk usage under control. |
| 19:36.39 | *** join/#asterisk calvinhp (n=calvinhp@rrcs-24-172-172-88.central.biz.rr.com) |
| 19:37.04 | calvinhp | what is the recommended SIP firmware version for a Cisco 7940 when using it with Asterisk? |
| 19:37.12 | calvinhp | should I stick with 7.4 or go to 8.8? |
| 19:39.40 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 19:39.59 | cjk | how do you guys replace existing pbxes and solve the fax issue? how do your customers do outgoing fax using their existing devices? |
| 19:44.58 | twisted | faxing is so 20 years ago |
| 19:45.01 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 19:45.16 | blitzrage | calvinhp: I've used 8.8 with some success on my 7960 |
| 19:45.30 | blitzrage | just at home -- no strong testing |
| 19:48.14 | calvinhp | blitzrage: what issues does "some" success come with? |
| 19:48.27 | blitzrage | it means it registered and let me place and receive calls |
| 19:48.29 | Qwell | no matter how much you change the firmware, it'll still be a cisco |
| 19:48.34 | blitzrage | :) |
| 19:48.47 | calvinhp | sounds like working to me :-) |
| 19:49.35 | *** part/#asterisk myiagy (n=myiagy@200.215.59.133) |
| 19:49.40 | calvinhp | We've been using our 7940's in our office for the last 3 years and love them |
| 19:49.45 | calvinhp | sound quality is great |
| 19:49.54 | blitzrage | I like the speakerphone on my 7960 more than the IP501 |
| 19:50.16 | ice_croft | what is zaptel_vltdmf? |
| 19:54.34 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 19:54.43 | hmmhesays | I can't remember my freaking godaddy login |
| 19:55.42 | cpm | dang |
| 19:56.24 | gleydson_barbosa | i have any POS Terminals with pap2t linksys and asterisk 1.4, but i cant make work! |
| 19:56.28 | gleydson_barbosa | please any help! |
| 19:58.03 | blitzrage | gleydson_barbosa: ask an actual question please |
| 19:58.20 | blitzrage | hmmhesays: I hate when that happens :) |
| 19:58.23 | muiro | ok, having a strange problem. I'd love to do a pastebin but I don't see anything in the cli output that is informative. Recently moved asterisk from out test server onto the server we want to use for production. This is a 64 bit machine. I have a sip trunk that I've been using to test the asterisk box. On the old server, everything was working fine. However, on this machine, calls through the sip trunk are behaving rather oddly. The first |
| 19:59.29 | *** join/#asterisk shinao1 (n=shinao1@196.207.1.30) |
| 20:00.55 | ice_croft | kand> damn! |
| 20:01.01 | muiro | after the first call, asterisk doesn't pick up at all. I know the connection to the sip trunk is fine because normally it would send the call to the sip provider's voicemail. It seems as if asterisk is getting the call but not work with the dialplan at all. |
| 20:01.16 | ice_croft | cant figure how to build chan_zap on freebsd |
| 20:01.18 | muiro | I can pastebin, but even with debug and 10 verbosity it shows nothing during these calls |
| 20:02.27 | blitzrage | you added 'console => debug' and then did a logger reload ? |
| 20:02.37 | muiro | ah, wait. I'm seeing somehting now |
| 20:02.40 | blitzrage | and what about sip debug |
| 20:02.43 | muiro | Really destroying SIP dialog '50536e75639243e20a1d68bb152217ce@127.0.0.1' Method: REGISTER |
| 20:02.56 | blitzrage | your remote end seems like its sending the wrong IP.... |
| 20:03.17 | blitzrage | my gut tells me an invalid context name |
| 20:03.52 | muiro | context names haven't changed. I copied the sip.conf and the dialplan over exactly. Sip has it go to one context, that context answers the phone |
| 20:04.20 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 20:04.35 | muiro | I'll look them over though |
| 20:05.15 | muiro | well, now it stopped |
| 20:05.37 | muiro | for some reason I had externrefresh=60 in sip.conf |
| 20:05.47 | muiro | it looks fine now |
| 20:06.38 | muiro | yeah, all's good again |
| 20:07.36 | *** join/#asterisk implicit (n=implicit@ip68-105-92-210.sd.sd.cox.net) |
| 20:07.37 | ice_croft | cant figure how to compile chan_zap on freebsd. its marked XXX |
| 20:07.44 | ice_croft | what i do wrong? |
| 20:08.34 | Mavvie | you should install it from the ports collection. |
| 20:09.29 | ice_croft | Mavvie> u mean asterisk? |
| 20:09.36 | Mavvie | yes |
| 20:10.16 | ice_croft | Mavvie> i have 6.2 with old ports. is it really neccesary to usr port instead of sources? |
| 20:10.16 | muiro | blitzrage: ok, listen to this. When I have sip set debug turned on, it works. When I turn debugging off... it stops working? |
| 20:10.40 | Mavvie | ice_croft: run portsnap fetch and portsnap update (or extract if it is the first time) |
| 20:10.47 | *** part/#asterisk gleydson_barbosa (n=gleydson@201.20.71.12) |
| 20:11.01 | ice_croft | Mavvie> oh man. that's not cool. |
| 20:17.55 | *** join/#asterisk yoanis (n=yoanis@murphy.uh.cu) |
| 20:18.22 | yoanis | hello there |
| 20:18.52 | yoanis | i wonder if there's an IAX client with proxy support (HTTP,socks,etc)? |
| 20:25.52 | grandpapadot | <PROTECTED> |
| 20:26.12 | yoanis | but is there any? |
| 20:26.26 | grandpapadot | Probably not, because call quality would really suck |
| 20:26.36 | yoanis | i see |
| 20:29.33 | *** join/#asterisk syzygyBSD (n=chatzill@66.226.228.204.cpe.speedyquick.net) |
| 20:36.43 | muiro | does anyone know if lumenvox will run at all on 64 bit systems? |
| 20:37.15 | grandpapadot | muiro: Distro? |
| 20:37.35 | muiro | RHEL5 |
| 20:41.55 | *** part/#asterisk yoanis (n=yoanis@murphy.uh.cu) |
| 20:44.31 | *** join/#asterisk remmo (n=junk@203.32.47.250) |
| 20:46.25 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 20:52.16 | blepsoaf | Could someone look at http://pastebin.com/m13a6c7e4 - im trying to understand why asterisk is always setting the externip when I have matchexterniplocally=yes this is causing my openser proxy to attempt to send information to the public ip which isnt possible in the internal network due to elbow routing. |
| 20:54.30 | [TK]D-Fender | blepsoaf: matchexterniplocally=yes <- never heard or, and you should have nat= YES under [general] |
| 20:56.05 | blepsoaf | [TK]D-Fender: http://bugs.digium.com/view.php?id=8821 for the matchexterniplocally |
| 20:58.12 | [TK]D-Fender | blepsoaf: pastebin actual SIP debug. |
| 20:58.49 | blepsoaf | [TK]D-Fender: sure, just two secs |
| 21:01.35 | *** join/#asterisk dbtid (i=j4ynr9je@cpe-71-72-252-171.columbus.res.rr.com) |
| 21:02.42 | blepsoaf | [TK]D-Fender: http://pastebin.com/d721590b |
| 21:03.42 | *** join/#asterisk [hC] (n=hardcore@66.119.167.162) |
| 21:03.44 | *** join/#asterisk katsuodo (n=musashi@ool-44c7e914.dyn.optonline.net) |
| 21:04.20 | katsuodo | hello |
| 21:04.43 | [TK]D-Fender | blepsoaf: Oh, and set "Nnat=no" for [openser] ..... |
| 21:05.12 | katsuodo | [TK]D-Fender Halo |
| 21:05.16 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 21:08.34 | katsuodo | here at site a broadband with a gateway phone, linksys (openwrt) gateway, port 5060 enabled, want to call analog phone from sip phone, sip conf setup, extensions.conf setup, able to dialout from analog phone, call sip by extension, phone ring, hear no voice |
| 21:08.48 | blepsoaf | [TK]D-Fender: same result - I set nat=no for openser and still having issues |
| 21:08.56 | blepsoaf | also set nat=yes under [general] |
| 21:09.42 | [TK]D-Fender | katsuodo: Go read : |
| 21:09.43 | [TK]D-Fender | ~sipnat |
| 21:09.44 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 21:10.02 | [TK]D-Fender | blepsoaf: Did you comment out that funky parm? |
| 21:10.23 | dbtid | can anyone tell me if anyone's running asterisk on embedded ppc platforms?? |
| 21:10.56 | *** join/#asterisk NightMonkey (n=NightMon@pdpc/supporter/sustaining/NightMonkey) |
| 21:11.38 | blepsoaf | [TK]D-Fender: yes I've read that, and I've tested with that commented out just now and still doesnt work |
| 21:11.45 | [TK]D-Fender | blepsoaf: :/ |
| 21:12.00 | blepsoaf | [TK]D-Fender: openser is unable to respond due to the asterisk publishing the incorrect route information |
| 21:12.11 | *** join/#asterisk techie (n=techie@adsl-76-214-26-98.dsl.lsan03.sbcglobal.net) |
| 21:17.09 | *** part/#asterisk reber (n=reber@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr) |
| 21:19.08 | blepsoaf | [TK]D-Fender: well i guess its time to look at the chan_sip.c :S |
| 21:19.59 | MrTelephone | i feel like choking a software developer at arris |
| 21:20.10 | MrTelephone | give me a name :-/ |
| 21:20.18 | dbtid | i'm new here; what's arris? |
| 21:20.30 | MrTelephone | a cable modem manufacturer |
| 21:21.33 | dbtid | ah |
| 21:22.15 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 21:22.51 | *** part/#asterisk sid (i=unstable@tor/regular/sid) |
| 21:24.33 | *** part/#asterisk MrTelephone (n=MrTeleph@h697179-171.picriverisp.net) |
| 21:29.19 | *** join/#asterisk JunK-Y (n=junky@modemcable183.17-83-70.mc.videotron.ca) |
| 21:30.00 | *** join/#asterisk pfn (n=pfnguyen@hanhuy.com) |
| 21:31.41 | *** join/#asterisk adorah (n=Michael@87.69.57.246.cable.012.net.il) |
| 21:31.45 | *** join/#asterisk Alowishus (n=jpenix@fwsdo.projectdesign.com) |
| 21:33.24 | *** join/#asterisk jetlagmk2 (i=jetlag@pool-70-17-58-187.pskn.east.verizon.net) [NETSPLIT VICTIM] |
| 21:34.58 | *** join/#asterisk denon (n=denon@tooth.decay.org) [NETSPLIT VICTIM] |
| 21:34.58 | *** mode/#asterisk [+o denon] by ChanServ |
| 21:35.21 | *** join/#asterisk implicit (n=implicit@ip68-105-92-210.sd.sd.cox.net) [NETSPLIT VICTIM] |
| 21:35.36 | *** join/#asterisk blitzrage (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 21:35.36 | *** mode/#asterisk [+o blitzrage] by ChanServ |
| 21:36.49 | *** join/#asterisk bkw_ (n=brian@adsl-64-149-47-251.dsl.tul2ok.sbcglobal.net) |
| 21:43.35 | *** join/#asterisk groogs (n=gregmac@d38-54-164.commercial1.cgocable.net) [NETSPLIT VICTIM] |
| 21:43.42 | *** join/#asterisk angler (i=angler@pdpc/sponsor/digium/angler) |
| 21:43.42 | *** mode/#asterisk [+o angler] by ChanServ |
| 21:43.42 | *** join/#asterisk orcimrepus (n=orcimrep@74-130-123-207.dhcp.insightbb.com) [NETSPLIT VICTIM] |
| 21:43.42 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) [NETSPLIT VICTIM] |
| 21:43.46 | *** join/#asterisk Corydon76-home (i=eight@pdpc/supporter/bronze/Corydon76-home) [NETSPLIT VICTIM] |
| 21:43.46 | *** mode/#asterisk [+o Corydon76-home] by ChanServ |
| 21:44.07 | *** join/#asterisk ectospasm (n=ectospas@c-68-62-219-11.hsd1.al.comcast.net) [NETSPLIT VICTIM] |
| 21:44.23 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) [NETSPLIT VICTIM] |
| 21:44.43 | *** join/#asterisk atisss (n=atisss@193.238.212.171) |
| 21:46.40 | *** join/#asterisk wyoming (n=steve_mu@216.166.159.235) |
| 21:47.22 | *** join/#asterisk teknoprep (n=chris@unaffiliated/teknoprep) |
| 21:47.23 | teknoprep | hey all |
| 21:47.30 | teknoprep | does anyone know of an application like hudlite |
| 21:47.37 | teknoprep | but isn't hudlite lol |
| 21:47.49 | *** join/#asterisk fnordus (n=dnall@24.84.160.227) |
| 21:47.54 | teknoprep | where i can record / transfer calls from a PC for that users extension |
| 21:48.19 | De_Mon | teknoprep that was about as clear as mud |
| 21:49.01 | teknoprep | well if you have ever used hudlite |
| 21:49.19 | teknoprep | you would know what i am looking for |
| 21:49.43 | teknoprep | hudlite has a ridiculous install process for non-trixbox servers |
| 21:50.44 | lirakis | later everyone |
| 21:50.47 | *** part/#asterisk lirakis (n=lirakis@65.200.191.253) |
| 21:52.41 | De_Mon | oh |
| 21:55.54 | De_Mon | why does hudlite want an irc server? |
| 21:56.23 | ice_croft | Mavvie> u still here? |
| 21:56.53 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 21:57.55 | *** join/#asterisk fskrotzki (n=fskrotzk@cpe-74-74-245-250.rochester.res.rr.com) |
| 22:02.32 | *** join/#asterisk Rapani (n=rapani@u211.ip1.netikka.fi) |
| 22:02.35 | Rapani | hello |
| 22:02.55 | Rapani | I have a problem with registration |
| 22:03.37 | Rapani | registration server is sip.mydomain.com but "realm" should be mydomain.com when registering.. how can I do this? |
| 22:03.48 | *** join/#asterisk ThatKidKel (n=Kelvin@66.236.241.67.ptr.us.xo.net) |
| 22:04.07 | Rapani | with myaccount:mypasswd@sip.mydomain.com/myaccount doesn't work |
| 22:04.21 | Rapani | with myaccount:mypasswd@mydomain.com/myaccount tries to wrong server |
| 22:04.31 | De_Mon | ah hah Yet, HUDlite is written for the latest (quite different and often quite unstable) version of Asterisk. |
| 22:04.46 | ThatKidKel | anyone have any experience putting OpenSER in front of Asterisk. I'd like it to handle my phone registrations. How will Asterisk know which context a call should be put into when it comes from OpenSER? |
| 22:04.51 | De_Mon | lil jab there eh fonality? |
| 22:10.30 | ice_croft | damn ports |
| 22:11.13 | *** join/#asterisk UCFmethod (n=UCFmetho@office.eyestreet.com) |
| 22:11.13 | Rapani | so how can I change "realm" in registering |
| 22:11.48 | Rapani | realm needs to be different than registration server address - is it possible with asterisk? |
| 22:17.04 | *** part/#asterisk ThatKidKel (n=Kelvin@66.236.241.67.ptr.us.xo.net) |
| 22:17.18 | *** part/#asterisk bkw_ (n=brian@adsl-64-149-47-251.dsl.tul2ok.sbcglobal.net) |
| 22:18.01 | *** join/#asterisk RoyK (n=roy@ip-76-25-149-91.dialup.ice.no) |
| 22:19.27 | *** join/#asterisk UCFmethod (n=UCFmetho@office.eyestreet.com) |
| 22:20.46 | *** join/#asterisk dexpdx (n=jason@66-162-134-242.static.twtelecom.net) |
| 22:21.04 | *** join/#asterisk UCFmethod (n=UCFmetho@office.eyestreet.com) |
| 22:21.06 | dexpdx | does anyone make a T1/PRI bank that would work for asterisk? |
| 22:21.42 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 22:22.11 | [TK]D-Fender | dexpdx, And what exactly is a "T1/PRI bank"? |
| 22:22.32 | dexpdx | a chassis that allows me to add pri cards that terminate to asterisk |
| 22:22.35 | dexpdx | you know like a real switch |
| 22:23.00 | Rapani | ok, solved realm problem but now asterisk sends wrong call-id |
| 22:23.19 | [TK]D-Fender | dexpdx, well you can get PCI cards for T1, or use SIP/PRI gateways, etc |
| 22:23.38 | dexpdx | [TK]D-Fender: yeah I know I've already got 4 quad sangoma cards |
| 22:23.42 | dexpdx | which work great |
| 22:24.03 | dexpdx | exempt if I want to bring up a new pri on a new card without dumping active calls |
| 22:24.35 | [TK]D-Fender | dexpdx, Then the only live way is with a VoIP gateway |
| 22:24.45 | dexpdx | like an AS5400 |
| 22:24.52 | dexpdx | ? |
| 22:24.55 | dexpdx | that's sad |
| 22:25.24 | dexpdx | I guess that only that's even close are those foneBridges |
| 22:26.10 | [TK]D-Fender | dexpdx, not really.. you neeed to restart * for those, and its only 1 per NIC IIRC |
| 22:29.37 | dexpdx | [TK]D-Fender: lame |
| 22:30.30 | *** join/#asterisk jdunck (n=jdunck@adsl-70-247-106-166.dsl.rcsntx.swbell.net) |
| 22:31.38 | [TK]D-Fender | dexpdx, indeed they are. one of several reasons nobody gives a rats ass about TDMoE |
| 22:32.16 | *** join/#asterisk citats (n=james@mrplow.gnuinternet.com) |
| 22:32.54 | Rapani | ok, now it works like charm :) |
| 22:40.44 | ice_croft | Mavvie> chan_zap still unavaliable |
| 22:41.34 | ice_croft | Mavvie> installed * from ports |
| 22:43.47 | [TK]D-Fender | ice_croft, and did you grab Zaptel from there as well? |
| 22:45.41 | *** join/#asterisk RoyK (n=roy@ip-76-25-149-91.dialup.ice.no) |
| 22:46.46 | *** join/#asterisk ManxPower (n=manxpowe@129.sub-75-200-103.myvzw.com) |
| 22:46.52 | kand | can someone help with commpartners registration? |
| 22:47.10 | jdunck | hey all. is there a way i can start testing SIP support prior to my SIP trunk going in? |
| 22:47.30 | ManxPower | ~siptrunk |
| 22:47.31 | jbot | [siptrunk] Asterisk does not support SIP Trunks. Set trunk=no in sip.conf and then set up the device normally in sip.conf. |
| 22:49.01 | ice_croft | [TK]D-Fender> no i didn't. there is an especial zapata driver for freebsd for my device |
| 22:49.34 | [TK]D-Fender | ice_croft, if you didn't install zaptel you're certainaly not going to get chan_zap |
| 22:50.07 | ice_croft | [TK]D-Fender> u mean i should install it from ports too? |
| 22:50.29 | [TK]D-Fender | ice_croft, obviously |
| 22:51.12 | ice_croft | [TK]D-Fender> hmm.. |
| 22:51.18 | ice_croft | shall do |
| 22:52.22 | ice_croft | [TK]D-Fender> anyway, i need to install cronyx zaptel over the standard port |
| 22:52.41 | blepsoaf | [TK]D-Fender: I just made a bug report ( http://bugs.digium.com/view.php?id=11493 ) |
| 22:55.02 | jdunck | ManxPower: to be clear on sip trunk support -- i'm trying to use asterisk on a sipconnect IAD from cbeyond (cisco 2430)... i'm totally new to telecom, so i'm not sure if this is a "sip trunk" or not |
| 22:55.13 | ManxPower | jdunck: it isn't. |
| 22:55.34 | ManxPower | In fact, "SIP Trunk" is TOTALLY a marketing term. |
| 22:55.58 | ManxPower | It's about as technically accurate as "new and improved". How can you improve something that did not exist before? |
| 22:56.03 | jdunck | yes, i've gotten *lots* of marketing terms so far, and very little clarity. |
| 22:56.27 | ManxPower | jdunck: Have you read The Book? |
| 22:56.37 | *** join/#asterisk Givemelove (n=foo@216.70.173.176) |
| 22:56.37 | jdunck | i've started, but am only on ch2 |
| 22:56.48 | jdunck | i guess that says it's still mostly-accurate for 1.4.* ? |
| 22:56.52 | *** part/#asterisk Givemelove (n=foo@216.70.173.176) |
| 22:57.00 | jdunck | is there a backwards-incompat list from book to 1.4.*? |
| 22:57.03 | ManxPower | jdunck: Correct. |
| 22:57.32 | ManxPower | For the most part, just read UPGRADE.txt that comes with 1.4 and you should see what you need. |
| 22:57.54 | *** join/#asterisk Greek-Boy (n=email@41.221.58.5) |
| 22:58.23 | jdunck | ok. thanks. :) i'll read the book; but back to original question-- if i get asterisk and a SIP phone, is there a way i can test in/out calls before the pipe from cbeyond is in? |
| 22:58.35 | jdunck | i'm just trying to minimize downtime while i figure out what i'm doing :) |
| 23:01.01 | *** mode/#asterisk [+o codefreeze] by ChanServ |
| 23:03.29 | jdunck | ManxPower: gotta run, but thanks for pointers |
| 23:05.43 | ice_croft | men, still need help with zapata |
| 23:06.04 | ice_croft | what should i do to make it avaliable for compile? |
| 23:07.22 | ice_croft | <PROTECTED> |
| 23:07.29 | ice_croft | that's what i have here |
| 23:07.32 | ice_croft | :) |
| 23:07.34 | ice_croft | :(( |
| 23:08.17 | *** join/#asterisk BigCanOfTuna (n=chatzill@dsl-mac-66-18-226-119-cgy.nucleus.com) |
| 23:09.09 | *** join/#asterisk bmd (n=bmd@72.54.252.34) |
| 23:09.09 | *** mode/#asterisk [+o bmd] by ChanServ |
| 23:10.45 | BigCanOfTuna | Excuse my (linux) ignorance, but when I dump a call file in the outgoing spool as username:asterisk when asterisk is started through its init.d script (startup; root), it will not pick up and process the call file. However, if I start asterisk as username, it picks up the file just fine. What is the best way to get this working assuming I'd like the init.d script to be used, but I don't want... |
| 23:10.45 | BigCanOfTuna | ...to be root to drop the call file into the outgoing spool? |
| 23:16.19 | denon | chown it after you put it in? |
| 23:16.30 | *** join/#asterisk JayTee52 (n=jforde05@c-69-137-243-25.hsd1.in.comcast.net) |
| 23:24.25 | *** join/#asterisk DaPrivateer (i=Privatee@crimson.66fruit.com) |
| 23:24.44 | JayTee52 | has anyone here run Asterisk on a x86-64 kernel? |
| 23:33.10 | *** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox) |
| 23:38.45 | *** join/#asterisk Weezey (n=ohno@206.210.101.254) |
| 23:39.04 | De_Mon | here we go, my first dialplan using bridge. please don't blow up or catch on fire... |
| 23:39.10 | Weezey | Anyone know how to get Xlite/eyebeam to "alert" a bluetooth module so that you can just press the button to answer a call? |
| 23:40.44 | *** join/#asterisk watchy (n=watchy@h200.176.255.206.cable.cmdn.cablelynx.com) |
| 23:40.50 | watchy | hi |
| 23:40.58 | watchy | whats a cheap place to get sangoma |
| 23:41.21 | De_Mon | wait what? execIf only handles true? |
| 23:41.31 | De_Mon | oookay |
| 23:42.02 | ice_croft | men, still need help with zapata |
| 23:42.10 | ice_croft | what should i do to make it avaliable for compile? |
| 23:42.18 | ice_croft | i have XXX 16. chan_zap |
| 23:42.24 | ice_croft | :( |
| 23:44.54 | De_Mon | when you select that menu item doesnt it tell you what it wants? |
| 23:48.47 | *** part/#asterisk Navion (n=billp@75-105-41-123.cust.wildblue.net) |
| 23:49.13 | ice_croft | De_Mon> it tells |
| 23:49.28 | ice_croft | but i cant figure out how to fix it |
| 23:49.43 | ice_croft | i mean i have zaptel and libpri |
| 23:49.48 | *** join/#asterisk mindCrime (n=chatzill@cpe-065-190-188-124.nc.res.rr.com) |
| 23:50.03 | ice_croft | Depends on: res_smdi(M), zaptel_vldtmf(E), zaptel(E), tonezone |
| 23:50.29 | ice_croft | what are the else things? |
| 23:50.49 | De_Mon | you have to have zaptel compiled and loaded as a kernel module |
| 23:50.54 | ice_croft | yes |
| 23:50.56 | De_Mon | and libpri if you're using it |
| 23:51.23 | ice_croft | oh. libpri needs kldload? |
| 23:51.34 | De_Mon | I think it gets compiled into zaptel |
| 23:51.45 | De_Mon | I dont use pri's so I cant say for sure |
| 23:52.36 | ice_croft | well, anyway i have libpri installed |
| 23:52.55 | De_Mon | what does lszaptel say? |
| 23:53.22 | ice_croft | min |
| 23:53.28 | ice_croft | lszaptel? |
| 23:53.33 | ice_croft | what's that? |
| 23:53.45 | ice_croft | ls zaptel? |
| 23:53.54 | De_Mon | its an app that tells you about zaptel hardware installed |
| 23:54.08 | De_Mon | pretty sure its included in zaptel package |
| 23:54.24 | ice_croft | a can get ztcfg. does it fit? |
| 23:54.39 | De_Mon | how about a zttest? |
| 23:55.24 | De_Mon | after you compile and install zaptel you have an init script that loads and configures the module |
| 23:55.49 | ice_croft | wait a minute, and i'll get it |
| 23:56.37 | De_Mon | the other thing, is to make sure those modules it depends on are selected. they will probably give reasons why they arnt available if theres a problem |
| 23:56.58 | ice_croft | it counts percents for now |
| 23:57.08 | ice_croft | ast# zttest |
| 23:57.08 | ice_croft | Opened pseudo zap interface, measuring accuracy... |
| 23:57.14 | ice_croft | 100.000000% 99.987793% 100.000000% 100.000000% 100.000000% 100.000000% 100.000000% |
| 23:57.14 | ice_croft | 100.000000% 100.000000% 100.000000% 100.000000% 100.000000% 100.000000% 100.000000% 100.000000% |
| 23:57.14 | De_Mon | okay sounds like it knows something is there |
| 23:57.56 | ice_croft | so, what's next? |
| 23:57.56 | De_Mon | nice numbers I get 99.9[6-9]% usually |
| 23:58.04 | ice_croft | yes, 100-99.9 |
| 23:58.26 | De_Mon | look for those modules it depends on and make sure they are selected |
| 23:58.35 | De_Mon | err -- menu items |
| 23:58.45 | ice_croft | res_smdi(M), zaptel_vldtmf(E) |
| 23:58.55 | ice_croft | i don't know what is it :( |
| 23:59.48 | ice_croft | what is it? |