| 00:00.21 | protocols | :) |
| 00:01.18 | protocols | anybody an idea why I suddenly can not register anymore on my asterisk? I am behind a nat, but I forward all needed ports + configured sip.conf accordingly.. |
| 00:01.47 | [TK]D-Fender | protocols: And we should take that at face value? |
| 00:02.24 | protocols | hm? |
| 00:02.36 | [TK]D-Fender | protocols: Everyone thinks they did everything properly... |
| 00:03.02 | [TK]D-Fender | protocols: Often not the case. Show your configs, and show SIP debug of the conversation and maybe we'll have some advice for you |
| 00:03.36 | protocols | I did not say I did it properly, it would most probably work then ;) - I just wanted to say that I tried to consider my situation of being behind a nat |
| 00:04.05 | [TK]D-Fender | ~pb |
| 00:04.05 | 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 |
| 00:04.07 | [TK]D-Fender | ^^ your friend |
| 00:04.14 | protocols | thank you :) |
| 00:06.09 | protocols | this is what I get on client side: http://pastebin.ca/1240135 |
| 00:06.42 | protocols | i was able to use sipgate though (a sip provider), so I persume my client seems to be working |
| 00:06.58 | *** join/#asterisk gmfm (n=hithere@boise-office.itsatomic.net) |
| 00:07.12 | *** join/#asterisk shmaltz (n=chatzill@mail.dmaven.com) |
| 00:07.39 | [TK]D-Fender | protocols: please provider proper SIP debug from * CLI |
| 00:08.04 | shmaltz | anyone know a provider that will allow me to do compaign calls around 12000 calls in a 12 hour span, each call lasting for around 15 to 30 seconds? |
| 00:08.18 | protocols | is there any way to get except through # asterisk -r -vvvv -d ? |
| 00:08.44 | [TK]D-Fender | protocols: that is what I wasnt, albeit with sip debug enabled. Also your sip.conf masking only passwords |
| 00:09.27 | *** join/#asterisk farkus__ (n=chatzill@cpe-72-225-212-219.nyc.res.rr.com) |
| 00:09.53 | protocols | sip.conf http://pastebin.ca/1240139 |
| 00:11.25 | protocols | ah cool ok |
| 00:12.00 | *** join/#asterisk nikko (n=nikko@12-218-250-137.client.mchsi.com) |
| 00:12.42 | protocols | here it is: http://pastebin.ca/1240141 |
| 00:13.26 | [TK]D-Fender | protocols: SIP/2.0 401 Unauthorized <- bad auth |
| 00:13.30 | protocols | yes it looks weird, that I am trying to connect from the same ip, but I persume I get the same errors when connecting from outside.. internal connections work fine |
| 00:13.58 | protocols | hmm is there an auth difference between local and remote registration? |
| 00:14.55 | [TK]D-Fender | protocols: Contact: <sip:6000@10.11.12.11> <- also wondering why I'm seeing that IP there... |
| 00:15.09 | [TK]D-Fender | protocols: not sure here... check your peer as well |
| 00:15.57 | protocols | hm ok |
| 00:19.09 | *** join/#asterisk StooJ (n=stooj@johnston37.plus.com) |
| 00:20.06 | *** join/#asterisk edwin_quijada (n=macaruch@200.26.172.98) |
| 00:20.09 | edwin_quijada | Hi! |
| 00:20.16 | edwin_quijada | I have a weird errror |
| 00:20.33 | edwin_quijada | line 0: Unable to open master device '/dev/zap/ctl' |
| 00:20.52 | rob0 | ls -l /dev/zap/ctl |
| 00:21.05 | edwin_quijada | i get this when I did ztcfg -vv |
| 00:21.28 | edwin_quijada | there is no this file |
| 00:21.40 | rob0 | driver loaded? |
| 00:21.52 | edwin_quijada | zaptel start |
| 00:22.21 | edwin_quijada | Loading zaptel framework: FATAL: Module zaptel not found. |
| 00:24.02 | protocols | seems to be missing then ;) |
| 00:24.39 | edwin_quijada | but i compile and install? |
| 00:24.49 | protocols | no errors occured? |
| 00:24.54 | edwin_quijada | no |
| 00:25.03 | edwin_quijada | i will do again' |
| 00:25.07 | tzafrir_laptop | uname -r |
| 00:25.42 | tzafrir_laptop | any chance 'make install' failed? |
| 00:26.04 | edwin_quijada | i didnt get anything alert |
| 00:26.19 | tzafrir_laptop | what is the output of: uname -r |
| 00:26.21 | edwin_quijada | so I get the info that i should use make config |
| 00:26.38 | edwin_quijada | 2.6.9-78.0.5.ELsmp |
| 00:26.54 | tzafrir_laptop | find /lib/modules -name zaptel.ko |
| 00:28.37 | tzafrir_laptop | edwin_quijada, what's the output of that command? |
| 00:28.49 | edwin_quijada | g |
| 00:29.15 | edwin_quijada | $>/lib/modules/2.6.9-78.0.5.EL/extra/zaptel.ko |
| 00:31.23 | *** join/#asterisk Deeewayne (n=dwayne@76.29.245.9) |
| 00:31.23 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 00:33.31 | edwin_quijada | it is the output |
| 00:33.35 | tzafrir_laptop | there you have it |
| 00:33.45 | tzafrir_laptop | not installed |
| 00:33.52 | tzafrir_laptop | at least not for your kernel |
| 00:34.27 | tzafrir_laptop | don't try to install it manually . Something went wrong |
| 00:34.41 | tzafrir_laptop | I guess 'make install' actually failed |
| 00:35.36 | edwin_quijada | ok |
| 00:35.45 | edwin_quijada | i will try again so |
| 00:37.15 | edwin_quijada | i did it again and i get the same |
| 00:38.01 | tzafrir_laptop | can you pastebin the output of 'make install' ? |
| 00:38.43 | edwin_quijada | yes |
| 00:41.16 | edwin_quijada | http://pastebin.com/m4767abe |
| 00:41.29 | edwin_quijada | i have centos |
| 00:44.26 | tzafrir_laptop | what was the command you ran? |
| 00:44.49 | tzafrir_laptop | just 'make install' ? |
| 00:45.38 | edwin_quijada | yes |
| 00:46.22 | tzafrir_laptop | ls -l /lib/modules/2.6.9-78.0.5.ELsmp/build |
| 00:47.11 | edwin_quijada | There is no file |
| 00:48.05 | edwin_quijada | this file not exist |
| 00:48.08 | tzafrir_laptop | ls -l /usr/src/linux |
| 00:48.35 | *** join/#asterisk Carlos_PHX (n=Carlos@ip68-3-162-244.ph.ph.cox.net) |
| 00:48.41 | edwin_quijada | lrwxrwxrwx 1 root root 29 oct 27 19:49 /usr/src/linux -> kernels/2.6.9-78.0.5.EL-i686/ |
| 00:48.58 | tzafrir_laptop | rm /usr/src/linux |
| 00:49.07 | edwin_quijada | why? |
| 00:49.28 | tzafrir_laptop | because it points to the wrong kernel source tree |
| 00:49.36 | tzafrir_laptop | and it hid the error message |
| 00:49.43 | edwin_quijada | done1 |
| 00:49.50 | tzafrir_laptop | it will just delete the symbolic link |
| 00:50.04 | tzafrir_laptop | now: ls /usr/src/kernels |
| 00:50.47 | tzafrir_laptop | is there /usr/src/ kernels/2.6.9-78.0.5.ELsmp-i686 ? |
| 00:51.50 | tzafrir_laptop | anyway, I'm off |
| 00:54.02 | edwin_quijada | 2.6.9-78.0.5.EL-i686 |
| 00:54.16 | edwin_quijada | nop |
| 01:00.23 | *** join/#asterisk protocols (n=protocol@ip-88-153-205-180.unitymediagroup.de) |
| 01:01.15 | *** join/#asterisk BhaalWK (i=bhaal@freenode/staff-emeritus/bhaal) |
| 01:02.34 | kerx | exten => 1,2,Dial(SIP/grnvoip/1238040118183579709|M(macroname)) |
| 01:02.49 | kerx | Am I using the Dial macros incorrectly from the above statement? |
| 01:04.50 | *** part/#asterisk eightmotives (n=em@67.203.130.154) |
| 01:10.31 | *** join/#asterisk mog (n=mog@74.95.48.254) |
| 01:10.31 | *** mode/#asterisk [+o mog] by ChanServ |
| 01:12.06 | *** join/#asterisk jicksta (n=jicksta@c-24-6-87-187.hsd1.ca.comcast.net) |
| 01:12.36 | [TK]D-Fender | kerx: "core show application dial" <- read the instructions carefully |
| 01:12.45 | kerx | k |
| 01:12.49 | kerx | i foudn what i did wrong |
| 01:12.53 | kerx | i don't even want a macro |
| 01:13.46 | [TK]D-Fender | kerx: One of many then |
| 01:15.50 | StephenF[W] | whats the best file format to have voice talent record IVR prompts and MOH to? |
| 01:16.02 | StephenF[W] | the most versatile and easiest to transcode |
| 01:16.51 | Carlos_PHX | Uncompressed wave file. |
| 01:17.04 | StephenF[W] | then transcode that into GSM or ulaw or something? |
| 01:17.06 | Carlos_PHX | The use something like SOX to encode them. |
| 01:17.25 | Carlos_PHX | Right, depending on what formats you use. |
| 01:17.48 | StephenF[W] | awesome, thx |
| 01:17.57 | Carlos_PHX | No point in a GSM file if you don't take GSM calls. |
| 01:18.29 | [TK]D-Fender | Carlos_PHX: the best in one in the codec of the call so you don't HAVE to transcode at all |
| 01:18.30 | Carlos_PHX | And the only reason to do multiple formats is to avoid server transcoding load, so most of my files are just loaded as wave. |
| 01:18.50 | [TK]D-Fender | StephenF[W]: rather |
| 01:19.33 | Carlos_PHX | StephenF[W]: Keep in mind that Asterisk can only play a specific wave format. |
| 01:20.49 | *** join/#asterisk protocols (n=protocol@ip-88-153-205-180.unitymediagroup.de) |
| 01:20.51 | *** join/#asterisk Koshatul (n=evangeli@ppp188-33.static.internode.on.net) |
| 01:21.38 | StephenF[W] | what do you mean specific? |
| 01:21.45 | Carlos_PHX | 8k, signed linear, 1 channel |
| 01:22.06 | StephenF[W] | I wont have asterisk do the transcoding, I will transcode from the .wav file into my normal codec and put that onto the asterisk box |
| 01:22.10 | StephenF[W] | will that work? |
| 01:22.16 | Carlos_PHX | Sure |
| 01:22.31 | *** join/#asterisk nicoAMG (n=superunk@201.203.50.42) |
| 01:22.31 | StephenF[W] | do I have to worry about the 8k spec then? |
| 01:22.48 | Carlos_PHX | So I ask the voice talent for a high-quality uncompressed wave file and transcode. |
| 01:23.02 | Carlos_PHX | I've found it sounds better than starting with a compressed file from them, for reasons I can't understand. |
| 01:23.12 | Carlos_PHX | It shouldn't be so, but that's what I've seen. |
| 01:23.17 | StephenF[W] | yup, ok thats what I asked for. uncomressed wave |
| 01:23.24 | StephenF[W] | then I will transcode it myself |
| 01:23.47 | Carlos_PHX | And if you're using Allison, you get that by default. |
| 01:25.40 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-3e4e218d6d27036c) |
| 01:29.14 | [TK]D-Fender | Carlos_PHX: because transcoding a compressed file is like a 10th generation photocopy... crap |
| 01:29.54 | hardwire | are we talking about evil? |
| 01:30.12 | *** join/#asterisk Kumbang (n=dsp@rusnas.paume.itb.ac.id) |
| 01:31.03 | jaytee | made me think of the movie Mulitplicity, lol |
| 01:31.12 | Carlos_PHX | [TK]D-Fender: The weird thing is that when I receive a ready to run file that was compressed at recording time, it sounds poor. I suspect Windows may have crappy compressors. |
| 01:31.18 | jaytee | "we took the blade out so he wouldn't cut himself" |
| 01:32.17 | [TK]D-Fender | Carlos_PHX: recompressing something that is already compressed = much worse |
| 01:32.39 | Carlos_PHX | I got that part, but I mean a file that I get that was compressed right to start, so just loaded into *. |
| 01:32.56 | Carlos_PHX | I've tried it three times with three people. |
| 01:33.13 | Carlos_PHX | (End users, not pro voice talent, so they may have just been stupid) |
| 01:33.19 | jaytee | ever use Audacity? |
| 01:33.26 | [TK]D-Fender | Carlos_PHX: its the transcode that kills it. It sounds fine on a sound card because you aren't trying to downgrade it further |
| 01:35.35 | hardwire | Carlos_PHX: I record voice in as HQ as possible then downsample using sox |
| 01:37.07 | jeev | any Command and conquer fans ? |
| 01:37.07 | hardwire | resample -qs ftw |
| 01:37.12 | Carlos_PHX | hardwire: Same here, I recommend it. Was just commenting on why. |
| 01:37.36 | hardwire | my boss once asked why I can't compress mp3 files using zip.. then compress that with rar.. |
| 01:37.36 | *** join/#asterisk axisys (n=axisys@117.18.228.88) |
| 01:37.39 | hardwire | that was a long day. |
| 01:37.49 | Carlos_PHX | Heh |
| 01:38.04 | *** join/#asterisk devilsoulblack (n=aandaluz@srv.ec-gye.internet.geainternacional.com) |
| 01:38.12 | *** join/#asterisk jer (n=jer@unaffiliated/jer) |
| 01:38.13 | Carlos_PHX | "I also recommend using small fonts in all your documents to save space." |
| 01:38.23 | hardwire | the smaller the better.. |
| 01:38.27 | hardwire | wastes less paper too! |
| 01:41.24 | *** join/#asterisk farkus___ (n=chatzill@cpe-72-225-212-219.nyc.res.rr.com) |
| 01:41.35 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
| 01:44.44 | Carlos_PHX | You would think in this economy that people would like to sell stuff. |
| 01:45.10 | Carlos_PHX | But it's taken almost two weeks to get a quote from Verizon for 500 cell phones and new service, and it's still not what I asked for. |
| 01:45.30 | Carlos_PHX | AT&T...no call back at all. Alltel...no call back. WTF |
| 01:46.41 | jjshoe | onyl 500 phones? I'm not surprised you haven't gotten a call back |
| 01:47.42 | Carlos_PHX | 500 committed up front, 2000 expected in 18 months. I suppose it's small fish. |
| 01:48.11 | Carlos_PHX | "What is your timeframe to buy?" |
| 01:48.16 | Carlos_PHX | "Last week!" |
| 01:48.51 | jjshoe | I've worked at a company with over 10,000 pagers, and their own pager tower |
| 01:49.36 | Carlos_PHX | Checks Amex credit limit for enough to buy a cell network |
| 01:50.07 | file | as long as you pay it in full next month... |
| 01:50.28 | jjshoe | it was a cool setup |
| 01:50.30 | jjshoe | you get a page |
| 01:50.32 | jjshoe | run to a phone |
| 01:50.38 | jjshoe | enter your pager number |
| 01:50.43 | jjshoe | get connected to whomever was paging you |
| 01:51.15 | Carlos_PHX | Oh yeah, I used one of those services. "Meet me" they called it. I usually returned the call on a trunked mobile. |
| 01:51.26 | Carlos_PHX | Shows age again |
| 01:53.45 | [TK]D-Fender | pulls out his Acme Carbon-Dating Kit |
| 01:54.36 | wylie_coyote | ....Super genius... |
| 01:55.19 | Carlos_PHX | It tried dating carbon once, but it was kind of dry and boring. |
| 01:55.35 | Carlos_PHX | Er, I tried |
| 01:56.29 | Qwell | jjshoe: yeah, well, when you work for one of the biggest clinics... :p |
| 01:56.29 | jeev | thank god, world series about to end |
| 01:56.52 | jjshoe | Qwell :D |
| 01:56.55 | jjshoe | Qwell second best in the world! |
| 01:57.56 | jjshoe | Qwell one of the doctors invented the intercom i think |
| 01:58.01 | Qwell | O.o |
| 01:59.58 | jjshoe | story has it he told a sales guy what he wanted, and spent sig. time trying to convince him how it would work, and the sales guy told him to F off |
| 02:00.14 | jjshoe | so he asked to talk to an engineer, and after lots and lots of trying to convince the engineer how do it they finally did it for him |
| 02:01.02 | jaytee | the guy who invented voicemail lives here in Indianapolis. He's a major donor to our zoo. |
| 02:01.39 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 02:01.39 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 02:01.45 | jaytee | his name is Scott Jones |
| 02:02.06 | lmadsen | don't you love you when solve a whole bunch of problems that have been plaguing you for a couple of months in one day? |
| 02:02.20 | jaytee | now he's got a new startup called ChaCha that people text questions for just about anything and you get a text message answer back. |
| 02:02.28 | jjshoe | Qwell same doctor also invented pneumatic tubes |
| 02:02.29 | Carlos_PHX | lmadsen: Mass murder? |
| 02:02.57 | jaytee | I just love that "Eureka!" moment you get sometimes |
| 02:03.28 | jeev | dood, when i have this Polycom 330 connected to the switch at the office, the internet has packet loss |
| 02:03.29 | jeev | wtf |
| 02:03.50 | orkid | i'll try again |
| 02:03.53 | orkid | i have a general question related to telecom, i hope someone can help. i'm trying to do an LNP to les.net, and on their form it says "affected long distance" "carry over PIC: Yes / No" "Long Distance Provider (IXC):" ... currently my long distance goes through my local provider (Bell Canada) afaik... ie. the long distance appears on their bill and I have a long distance plan from them... so should i put "Bell Canada" under "Long Distance Provider (IXC): |
| 02:04.06 | jaytee | who is dood? I never see him logged in here. |
| 02:04.35 | jeev | ~dood |
| 02:04.36 | jbot | well, dood is a typo for d00d, or dutch for dead |
| 02:04.41 | jeev | i dunno i dunno what to do |
| 02:04.51 | jeev | we got legit internet now and it's doing the same shit |
| 02:04.54 | jeev | i need a wired router to test with. |
| 02:05.09 | lmadsen | ~dude |
| 02:05.10 | jbot | Be most excellent to each other! |
| 02:05.28 | lmadsen | jbot: dude is also Jim Dixon |
| 02:05.29 | jbot | okay, lmadsen |
| 02:05.29 | jaytee | should be something like "Mark it an 8, Donny" |
| 02:05.34 | lmadsen | ~dude |
| 02:05.35 | jbot | Be most excellent to each other!. Jim Dixon |
| 02:05.53 | lmadsen | that didn't work very well.... |
| 02:06.01 | orkid | anybody? bueller? bueller? |
| 02:06.13 | lmadsen | jbot: no, dude is Be most excellent to each other! Also the moniker of Jim Dixon. |
| 02:06.14 | jbot | lmadsen: okay |
| 02:06.26 | lmadsen | orkid: my guess is that you are correct... Bell Canada |
| 02:06.29 | *** join/#asterisk farkus___ (n=chatzill@cpe-72-225-212-219.nyc.res.rr.com) |
| 02:06.31 | lmadsen | ~dude |
| 02:06.32 | jbot | well, dude is Be most excellent to each other! Also the moniker of Jim Dixon. |
| 02:07.05 | jaytee | yay! |
| 02:07.17 | jjshoe | jaytee this isn't nam, we have rules here! |
| 02:07.45 | *** join/#asterisk nikko (n=nikko@12-218-250-137.client.mchsi.com) |
| 02:07.47 | jeev | 64 bytes from 71.x: icmp_seq=14 ttl=114 time=11.951 ms |
| 02:07.47 | jeev | 64 bytes from 71.x: icmp_seq=53 ttl=114 time=11.865 ms |
| 02:07.47 | jeev | wow |
| 02:07.59 | jaytee | jjshoe, lol |
| 02:08.36 | jaytee | jeev, that's not terrible timing, 11ms? |
| 02:08.46 | lmadsen | jaytee: look at the icmp_seq |
| 02:09.03 | jaytee | ah, yuck |
| 02:09.26 | jeev | yea |
| 02:09.27 | jeev | lol |
| 02:09.38 | jeev | first, i was stealing internet.. paying for tv.. and obviously hax0ring docsis |
| 02:09.39 | *** part/#asterisk farkus___ (n=chatzill@cpe-72-225-212-219.nyc.res.rr.com) |
| 02:09.43 | jeev | then i'm like, maybe we got a digital filter. |
| 02:09.48 | jeev | then, order it. and same shit. |
| 02:09.49 | jaytee | yeah, like 48 packets just gone |
| 02:10.03 | jeev | how do you figure 48 |
| 02:10.13 | jaytee | jeev, this isn't on your link that you're using GRE tunnelling is it? |
| 02:10.18 | jeev | 39 |
| 02:10.19 | jeev | no way |
| 02:10.27 | jeev | how'd you get 48 packets?!?!? |
| 02:10.30 | jeev | gre tunneling is working fine |
| 02:10.33 | jeev | not a single boo boo |
| 02:10.33 | jeev | since.! |
| 02:10.38 | jeev | http://x.jeev.net/diag.jpg FOR LIFE |
| 02:10.55 | jaytee | oops, my bad. yeah, more like 39 |
| 02:11.07 | jaytee | I need to go to bed early. brain is crispy |
| 02:11.11 | jeev | but.. 100 packets transmitted, 62 packets received, 38.0% packet loss |
| 02:11.13 | jeev | that's what it came out to |
| 02:11.20 | jeev | maybe it's doing some weird shit, counting a late receival |
| 02:11.20 | jeev | dunno |
| 02:11.24 | jaytee | jesus hernando christ! |
| 02:11.26 | lmadsen | ya, I'm outta here too... finally solved these issues, and am done! |
| 02:11.27 | jeev | or a +1 / -1 |
| 02:11.31 | Carlos_PHX | Brain looks like this? http://www.speedextreme.com/temp/oct/brain.gif |
| 02:11.35 | jaytee | nite leif |
| 02:11.46 | jeev | lol |
| 02:11.48 | jeev | carlos |
| 02:11.51 | jeev | are you still in your boat ? |
| 02:11.55 | jeev | my tomahawk failed. |
| 02:11.58 | jaytee | hahahaha |
| 02:12.00 | Carlos_PHX | No, home unfortunately. |
| 02:12.07 | Carlos_PHX | Not quite the same. |
| 02:12.11 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 02:12.17 | jaytee | I'm savin that pic |
| 02:12.42 | jeev | jaytee, you were never a Command and Conquer guy ? |
| 02:12.47 | jeev | Westwood Studios/ |
| 02:12.50 | jeev | then stupid electronic farts |
| 02:13.09 | jaytee | no, haven't played. sorry |
| 02:13.13 | jeev | damn |
| 02:13.17 | jeev | it's like a 10 year old thing |
| 02:13.18 | jeev | series |
| 02:13.21 | jeev | maybe even more |
| 02:13.34 | jaytee | I've heard of it, just never played it |
| 02:13.38 | jeev | wack |
| 02:13.44 | jeev | when is Obama's tv thing |
| 02:14.37 | jaytee | I used to play the Ultima games and a few others back in the day but I got burned out on them before computer games actually started getting good. Some of the CGI stuff now is amazing |
| 02:14.50 | jaytee | it was on at 8 here. it was excellent |
| 02:14.53 | jeev | oh |
| 02:14.54 | jeev | really |
| 02:14.56 | jeev | i hope i dont miss it |
| 02:15.00 | jeev | i gotta go to thes tore, put that shit on directly |
| 02:15.01 | jeev | and come back |
| 02:15.02 | jeev | bbiab |
| 02:15.04 | jaytee | don't! it's worth it |
| 02:15.08 | jeev | but i need to install command and conquer |
| 02:15.08 | jeev | GRR |
| 02:15.09 | jeev | bbiab |
| 02:15.16 | jaytee | later |
| 02:15.46 | Carlos_PHX | We're going to make a drinking game of it. One shot for "change," two for every weasel word, and three when he tells an outright lie. I'm hoping the liquor delivery truck is full. |
| 02:16.19 | *** join/#asterisk tengulre (n=tengulre@125.71.208.16) |
| 02:17.11 | Carlos_PHX | MMm, Chinese food delivery is here. |
| 02:20.24 | drmessano | yawn |
| 02:20.48 | jaytee | drmessano, are you voting for Saxby Chambliss? |
| 02:23.44 | jaytee | In the 1960s, during the Vietnam War, Chambliss was given five student deferments and he received a medical deferment (1-Y) for bad knees due to a football injury. |
| 02:24.51 | drmessano | Um, no way.. He's a republican |
| 02:24.53 | jaytee | just like Cheney except for the medical and yet he had the gall to question the patriotism of Max Cleland, his opponent in 2004 who served in Vietnam and lost 3 limbs there. What an incredible douchebag. |
| 02:25.14 | jaytee | it's a close race he's in. I hope he loses. |
| 02:26.42 | drmessano | Well, this is why I keep telling the wife we need to sell our condo and buy a house with a LARGE front lawn, even sacrificing back yard space to do so |
| 02:26.48 | drmessano | *yard signs* |
| 02:26.51 | jjshoe | why didn't someone tell that guy not to catch gernades? |
| 02:26.58 | jaytee | amazing how they play the patriot card. if you question your government or your president then you're not a patriot? what horseshit. It's the responsibility of every citizen to question their government and hold their leaders accountable. Seems the publicans only undertand sleaze. |
| 02:27.16 | jaytee | jjshoe, learn to spell before you make a lame attempt at a joke. |
| 02:27.31 | jjshoe | jaytee my my are we a wee bit touchy? |
| 02:28.06 | drmessano | jaytee: This country was founded on the premise of questioning the wrongs of a tyrannical government.. Seems that doesn't count anymore |
| 02:28.19 | jaytee | jjshoe, WHAT? TOUCHY? I'M FINE THANKS. BRB, GOTTA REFILL MY 27th CUP OF COFFEE. SHIT!!!! I'M OUT OF SUGAR! |
| 02:28.22 | *** join/#asterisk Hadi- (n=Hadi@CPE002129717ae3-CM001a668ee8b2.cpe.net.cable.rogers.com) |
| 02:28.26 | Hadi- | hi everyone |
| 02:28.40 | Hadi- | any hylafax experts here? ;) |
| 02:28.44 | jjshoe | everyone isn't here righ tnow |
| 02:28.52 | jaytee | Carlos? |
| 02:30.04 | Hadi- | well I need to find a way to prefix all outgoing fax going through hylafax with a special prefix so that they can be send to a correct trunk (one that is using g711) |
| 02:30.19 | Hadi- | looked at the etc/dialrules |
| 02:30.21 | [TK]D-Fender | jaytee: Touchy? Hah, thats nothing... people around here say I have a hair trigger temper, but by God the next time I hear someone say that I'm gonna cop their #&$%ING head off! |
| 02:30.26 | Hadi- | but cant seem to gwt it work |
| 02:30.44 | jaytee | [TK]D-Fender, hahaha. |
| 02:30.56 | jaytee | [TK]D-Fender, you and ManxPower |
| 02:31.54 | jaytee | funny how the people who know the most and try to help the most also catch the most flack. They told me that life wasn't fair when I was young but I stubbornly remained hopefully optimistic. |
| 02:32.12 | *** part/#asterisk km2 (n=x@32.178.16.54) |
| 02:32.20 | *** join/#asterisk km2 (n=x@32.178.16.54) |
| 02:33.49 | jaytee | only 11 more days and I get to make my Hajj to VOIP Mecca. I'll probably get to meet or have the Grand Ayatollah Jared as my instructor. Who knows? I may even get to meet the Prophet Mark Spencer, praise be unto him! |
| 02:35.51 | drmessano | jaytee: I am not allowed to attend any asterisk events |
| 02:35.59 | drmessano | <-- Middle name is "hussein" too |
| 02:36.04 | jaytee | rofl |
| 02:36.31 | jaytee | what'd you do? piss in the punch at a Digium company picnic or something? |
| 02:38.12 | jaytee | the class is expensive but you get some serious schweet swag out of the deal. A Polycom 330, a TE110 card and a TDM410 card, a backpack, mousepad and orange ice pen. |
| 02:38.26 | jaytee | hopefully I'll even learn something |
| 02:39.10 | drmessano | Well, for legal reasons, I cannot go into details.. but needless to say.. If you're gonna play Dungeons and Dragons with the leaders of the VoIP free world, there's lots of cool names to come up with that sound like you're invoking some sort of VoIP magical dominance over everyone else, but "callweaver" actually has TWO meanings... |
| 02:39.18 | drmessano | Talk about "being late to the party" |
| 02:40.56 | jaytee | hehe, you really need to start a "comedy voip blog". You could make some serious coin from ad placement, have fun and entertain the hell out of all of us. |
| 02:41.39 | jaytee | I bet you could even get Kerry to pony up some dough for Trixbox ads and links to his camera sites |
| 02:41.51 | drmessano | Throw up a bunch of adsense so I could pay for the lawyers? |
| 02:41.55 | drmessano | HA |
| 02:42.44 | drmessano | Yeah The "KerryCAM 9000", the only camera on the market that secretly takes pictures of YOU while you're snapping off pics of everyone else |
| 02:42.50 | jaytee | "A portion of the profits from this website will be donated to the Fund for Engineer Tim and Other Displaced And Screwed Over Fonality Employees" |
| 02:44.25 | *** join/#asterisk qdk_ (n=qdk@77.241.140.253.bredband.3.dk) |
| 02:45.08 | *** join/#asterisk mszathmar (n=justin@li36-15.members.linode.com) |
| 02:45.14 | drmessano | Thats actually a fantastic idea.. Just need to ask Obama about how to redistribute the wealth |
| 02:45.39 | mszathmar | anyone seen " error: invalid use of undefined type âstruct moduleâ" before when trying to build dahdi-linux? |
| 02:45.45 | LeddyHM | Looking for some ideas. We just upgraded to 1.4 form 1.2 and whenever a user calls in to the main number and then dials an extension the caller never hears a "ring ring". However when they dial the direct number you get it |
| 02:46.00 | LeddyHM | or when you dial from extension to extension (internally) you hear it |
| 02:46.05 | LeddyHM | any ideas? |
| 02:50.49 | lmadsen | lol... wow |
| 02:50.56 | lmadsen | now that's some serious scrollback |
| 02:51.03 | jaytee | what? |
| 02:52.10 | lmadsen | http://pastebin.ca/1240245 |
| 02:52.58 | jaytee | lmadsen, well....we aim to entertain :-) |
| 02:54.02 | justdave | LeddyHM: I think there were changes to "auto-fallthrough" in dialplans from 1.2 to 1.4 |
| 02:54.03 | drmessano | lol |
| 02:54.21 | justdave | you might have had something in the dialplan that depended on falling through at the end |
| 02:54.46 | jaytee | lmadsen, and I understand completely why you can't be there because you're very busy spreading the Word Of Spencer to the infidels of Toronto. |
| 02:54.57 | lmadsen | heh |
| 02:54.58 | lmadsen | hardly |
| 02:55.04 | justdave | LeddyHM: watch your console when someone tries the dial from the menu thing and see if any warnings show up on the console |
| 02:55.21 | lmadsen | I was actually solving a few issues in a clustered asterisk solution |
| 02:55.33 | jaytee | cool! |
| 02:55.43 | *** join/#asterisk mog (n=mog@c-68-62-219-86.hsd1.al.comcast.net) |
| 02:55.43 | *** mode/#asterisk [+o mog] by ChanServ |
| 02:56.15 | jaytee | clustered as in they all share the same dialplan? or like a "warm" spare failover scenario? |
| 02:56.32 | lmadsen | as in, the all share the same dialplan via func_odbc goodness |
| 02:56.41 | lmadsen | using realtime peers, and app_queue |
| 02:56.44 | LeddyHM | no errors show up |
| 02:56.48 | jaytee | gonna do a HowTo about it? |
| 02:56.50 | jeev | woo hoo |
| 02:56.50 | LeddyHM | I have logging at 14 |
| 02:56.53 | jeev | RED ALERT 3 BABY |
| 02:57.33 | lmadsen | jaytee: when this system is working the way I want it to, I may build a simple version of it, but for now, we'll see where we go from here. This has been nearly a year long project (January will be a year) |
| 02:57.44 | *** join/#asterisk DJ_HaMsTa (i=DJ_HaMsT@c-69-136-240-75.hsd1.nj.comcast.net) |
| 02:57.47 | LeddyHM | I believe I set auto fallthrough off |
| 02:57.58 | lmadsen | 2nd time I've built this type of system.. first time took me two years, where it was fronted with openser |
| 02:58.13 | DJ_HaMsTa | i just got a did did.voip.les.net, can someone please help me configure it with X-lite ? |
| 02:58.24 | LeddyHM | autofallthrough = no |
| 02:58.29 | lmadsen | but a good chunk of that was me building an E911 portal that spoke via SOAP updates |
| 02:58.32 | LeddyHM | help any? |
| 02:58.41 | jaytee | lmadsen, I'll bet there's alot of people who'd be interested in that kind of setup for scaling up their current system(s) |
| 02:58.45 | hardwire | lmadsen: round robin? |
| 02:58.46 | *** join/#asterisk newmember (n=chatzill@S010600036d1139bb.cg.shawcable.net) |
| 02:58.51 | hardwire | I'm thinking of doing that atm |
| 02:58.54 | lmadsen | jaytee: oh ya... I'm starting to get a lot of calls about it recently |
| 02:59.06 | hardwire | I have a cluster of * and rr inbound on the PRI as well. |
| 02:59.44 | jaytee | "then I get to move up to assistant manager! And that's when the big bucks come in." |
| 02:59.51 | LeddyHM | there is a noticable pause before vm picks up i.e. you would hear it ringing, whereas autofallthrough I'd imagine would go straight to vm |
| 02:59.52 | lmadsen | hardwire: this one I'm building is just multi-site call-centre. The other one was an ITSP, so slightly different implementations of the same tools. OpenSER was the registration point and distributed calls via the distribution module |
| 03:00.32 | jaytee | lmadsen, so which fork are you taking now? Kamailio or openSIPS? |
| 03:00.37 | hardwire | I wanna play with the distribution module |
| 03:00.54 | lmadsen | hardwire: ya... each physical site either has a couple PRIs or analog lines, depending on site size. Agents in larger call centres can login and answer calls for remote queues, otherwise, they failover to the main queue, then over to voicemail |
| 03:00.57 | drmessano | I was thinking about writing a SIP proxy called Godare |
| 03:01.05 | *** join/#asterisk joobie (n=joobie@201.023.dsl.mel.iprimus.net.au) |
| 03:01.08 | lmadsen | jaytee: when I built the ITSP it was still openser |
| 03:01.19 | lmadsen | I no longer maintain that system |
| 03:01.24 | drmessano | Like, "Hey you, dis call! Go dare!.. and you, over dare.. go dare" |
| 03:01.30 | jaytee | which is still available on sourceforge I believe |
| 03:01.41 | hardwire | lmadsen: it's a shame DNS SRV isn't as honored as it should be |
| 03:01.49 | hardwire | think of all the wasted cycles. |
| 03:01.58 | lmadsen | hardwire: indeed... but luckily I've never had to use DNS methods of load balancing |
| 03:02.08 | hardwire | had to? |
| 03:02.11 | hardwire | I'd prefer to in some simple cases. |
| 03:02.30 | lmadsen | let me rephrase; never pursued that method to solve my problem |
| 03:02.33 | drmessano | DNS SRV + DUNDI would be WIN |
| 03:02.39 | jaytee | lmadsen, have you read any of the particulars regarding the fork and the reasons each group had? any opinions as to which direction is the better one? |
| 03:03.10 | hardwire | drmessano: indeed |
| 03:03.19 | hardwire | I'd love my ITSP to suddenly rock me with some DUNDi |
| 03:03.52 | lmadsen | jaytee: nope, I didn't do the openser implementation in the ITSP, and I will probably at all costs ever avoid learning openser/fork_of_the_day. If I needed to front a bunch of asterisk boxes with a central registration server, I'd probably try it on freeswitch first. I've never looked at it, or tried it, but I can't imagine the syntax can be any more difficult than openser :) |
| 03:04.02 | hardwire | lmadsen: right now we are doing some wholesale with PRI to several boxes behind a single gateway, which uses nothing at the moment for inbound voip calls. |
| 03:04.16 | hardwire | so I'm on the wall with just using the first node as a redirect to the second or not. |
| 03:04.26 | hardwire | or using openser ala ubuntu packages |
| 03:04.38 | hardwire | I wish I could just use DNS SRV |
| 03:05.08 | lmadsen | I had an idea to use asterisk+dundi+group_count to act as a gateway to communicate with other asterisk boxes and redirect incoming calls to them |
| 03:05.33 | lmadsen | I've used dialplan functions in DUNDI mappings a bunch of times, so I'm pretty positive I could make it work in just a few mins |
| 03:05.56 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 03:06.26 | drmessano | hates touching media. (Fear of cooties) |
| 03:07.55 | jaytee | my boss installed one of those hand sanitizer lotion dispensers and insists we all use it. "I will be monitoring!". The man is a bigger germophobe than Howard Hughes. |
| 03:08.40 | [TK]D-Fender | jaytee: Those dispensers are great... at dispensing GERMS. |
| 03:08.57 | lmadsen | jaytee: I'd tell him, "That's how you create super bugs as you're leaving the strongest 0.1% of germs to multiply. I will not be an accessory to the death of the human civilization due to a modern day plague." |
| 03:09.03 | [TK]D-Fender | jaytee: Given that nobody remembers to use a sleeved elbow, paper towel as a buffer, etc |
| 03:09.19 | [TK]D-Fender | jaytee: So everyone touches IT and everyone gets in contact |
| 03:09.24 | drmessano | [TK]D-Fender: Touch the handle, I promise it wont bite |
| 03:09.37 | drmessano | [TK]D-Fender: ...much |
| 03:09.38 | hardwire | lmadsen: or.. you could just use DNS SRV via astrisks |
| 03:09.40 | [TK]D-Fender | doesn't take drugs..... germs take drugs to try to survive ME. |
| 03:09.51 | hardwire | since it supposedly honors round robin priorities |
| 03:09.56 | jaytee | I've learned that arguing with him is like taking a running sprint into a brick wall. |
| 03:10.10 | DJ_HaMsTa | http://www.dailymail.co.uk/health/article-1081359/Copper-door-handles-taps-kill-95-superbugs-hospitals.html |
| 03:10.13 | hardwire | but that's nowhere near as easy to just "plug in" |
| 03:10.14 | DJ_HaMsTa | give that to your boss |
| 03:10.18 | lmadsen | hardwire: if you want to randomly distribute calls. With my method you can load test a box to determine how many calls it can handle, then distribute calls to servers smartly. |
| 03:10.29 | jaytee | I read that about the copper doorknobs |
| 03:10.36 | hardwire | lmadsen: I'd thought about doing similar things with the management software I'm writing |
| 03:11.09 | hardwire | but it would be using distributed databases and an event bus to help each node understand it's neighbors |
| 03:12.05 | jaytee | [TK]D-Fender, I just finished reading Darwin's Radio by Greg Bear |
| 03:12.21 | hardwire | mysqldb + master/master-slave + triggers + python + blind faith |
| 03:12.47 | hardwire | or ndb + triggers |
| 03:13.20 | hardwire | lmadsen: but.. how are you measuring said quality? |
| 03:13.25 | hardwire | or capacity |
| 03:13.35 | hardwire | I read teh book but I didn't know the conditions of the tests |
| 03:13.59 | lmadsen | hardwire: eh>? |
| 03:14.19 | orkid | lmadsen: thanks. but what about PIC transfer? is it even possible? i mean, it's going to be a VoIP DID... ie. incoming only as far as i understand |
| 03:14.40 | lmadsen | orkid: I have no idea beyond what I thought may have been the right answer :) |
| 03:14.52 | lmadsen | is now tired... I have worked weeks worth of hours in 3 days |
| 03:14.59 | lmadsen | hardwire: we can continue this conversation tomorrow... |
| 03:15.02 | lmadsen | night all! |
| 03:15.34 | orkid | congrats! maybe u can rest for weeks now :) |
| 03:15.53 | hardwire | my eyes need to start focusing |
| 03:16.06 | hardwire | I think my body is telling me I need to focus on things further away |
| 03:16.47 | jaytee | I've been vertical for too long and need to get horizontal. Nite all |
| 03:16.48 | hardwire | thinks his beanbag needs to get squished |
| 03:19.45 | trelane | your quit message is gay |
| 03:19.47 | trelane | enjoy the raid though |
| 03:20.03 | trelane | has arranged a few of those... from time to time... for deserving souls |
| 03:20.21 | trelane | always nice to see deserving scumbags get their very own SWAT team moment |
| 03:21.06 | hardwire | man frodo is a wuss |
| 03:30.47 | *** join/#asterisk TenJack (n=chatzill@c-71-197-166-145.hsd1.or.comcast.net) |
| 03:30.57 | DJ_HaMsTa | whats a good site to get a free sip from ? |
| 03:31.20 | [TK]D-Fender | DJ_HaMsTa: 127.0.0.1 |
| 03:31.52 | TenJack | anyone know what this error means: Postgresql RealTime: Failed to connect database server asterisk on 127.0.0.1. Check debug for more info." ? |
| 03:31.58 | DJ_HaMsTa | whats a good provider to sighn up to that gives free sip numbers |
| 03:32.19 | TenJack | i just restarted ubuntu and now i cannot start astserisk |
| 03:38.43 | *** join/#asterisk BeeBuu (n=beebuu@125.95.101.97) |
| 03:39.25 | orkid | uhoh |
| 03:39.38 | BeeBuu | how can i convert "2008-10-30 12:34:56" to epoch? |
| 03:42.57 | [TK]D-Fender | TenJack: Well go verify that your DB serveris indeed running |
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| 04:27.46 | drmessano | ~book |
| 04:27.47 | jbot | methinks book is Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com or see ~buybook |
| 04:41.28 | jameswf | lm/me starts stalking people on linkedin |
| 04:41.35 | jameswf | starts stalking people on linkedin |
| 04:41.38 | jameswf | bahh |
| 04:41.48 | *** join/#asterisk ManxPower (n=manxpowe@230.sub-75-251-121.myvzw.com) |
| 04:54.25 | *** part/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194) |
| 04:58.41 | *** join/#asterisk IanBeyer (n=chatzill@cpe-76-92-203-240.kc.res.rr.com) |
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| 05:00.10 | IanBeyer | Carlos_PHX: how was your early-morning meeting? |
| 05:02.55 | drmessano | jameswf: I have people adding me on linkedin that I want NOTHING TO DO WITH |
| 05:03.34 | drmessano | "Network with me, please" "With you? I didn't like WORKING with you where our mutual presence was FORCED, why do I want to selectively NETWORK with you now?" |
| 05:14.42 | *** join/#asterisk jicksta (n=jicksta@c-24-6-87-187.hsd1.ca.comcast.net) [NETSPLIT VICTIM] |
| 05:14.45 | *** join/#asterisk trnzmeta (n=bleh@iinet.guard.com.au) |
| 05:21.40 | *** join/#asterisk TOrrIeri (n=torrieri@nelug/crew/torrieri) |
| 05:24.47 | trnzmeta | go go palin! |
| 05:25.46 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 05:27.57 | drmessano | GTFO |
| 05:28.20 | drmessano | I bet you are going to vote for them.. you PALINDRONE |
| 05:29.59 | trnzmeta | nah, I just want the comic relief if the replicans win |
| 05:30.06 | trnzmeta | it's like counter strike really |
| 05:30.51 | drmessano | She doesnt need to win for that to happen |
| 05:31.13 | drmessano | Shes the Hillary of the republican party.. better believe they'll be building her up for 2012 |
| 05:32.32 | trnzmeta | then we can have a proper scrag fight in 2012 |
| 05:32.46 | trnzmeta | HC vs SP |
| 05:34.56 | drmessano | Unless Obama really screws up, he'll be a two term president |
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| 05:57.58 | trnzmeta | maybe we can ufc the 2012 womens presidential race |
| 05:58.03 | trnzmeta | yeah... raunchy |
| 05:58.11 | trnzmeta | yeah... risque |
| 06:04.08 | *** join/#asterisk coolthreads (n=coolthre@203-97-238-71.cable.telstraclear.net) |
| 06:12.02 | trnzmeta | fuck I'm bored |
| 06:12.14 | trnzmeta | trying to connect up my asterisk box in sth africa |
| 06:12.22 | trnzmeta | man the connection there is obsolete |
| 06:12.36 | mDuff | so -- contexts created by pbx_lua are showing up as empty in 'show dialplan', with nothing but an 'Alt. Switch' entry. Is this expected behavior? |
| 06:13.30 | mDuff | (the bigger issue is that attempts to Goto a pbx_lua-created context result in none of the callbacks in lua_switch being invoked... but that's something I don't have any idea where to start with) |
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| 06:24.21 | jeev | 15% packet loss out of 3000 packets from 2 different networks both ~10-20 ms away |
| 06:24.24 | jeev | and my cable company doesn't care. |
| 06:25.00 | Linuturk | time to get a new company jeev |
| 06:25.16 | jeev | rthat's at my store |
| 06:25.17 | jeev | at home |
| 06:25.22 | jeev | it's 0% loss |
| 06:25.24 | jeev | at my office |
| 06:25.27 | jeev | it's 0% loss |
| 06:25.30 | jeev | and my office is a mile away. |
| 06:25.32 | jeev | pathetic |
| 06:25.46 | jeev | funny thing is that first, i was stealing the internet and it was always going down |
| 06:25.49 | jeev | and i'm like wtf. |
| 06:25.53 | jeev | ordered it and it still goes down. |
| 06:26.08 | jeev | making me think it's the polycom downing the network.. but it's the stupid copper in the building probably |
| 06:26.17 | jeev | i will defniitely cancel it and steal internet again, once they come fix it. screw them. |
| 06:27.05 | jeev | oh well, command and conquer time, brb |
| 06:29.11 | tAnkOSX | :) |
| 06:39.34 | *** join/#asterisk pcrane (n=pcrane@125-238-28-158.broadband-telecom.global-gateway.net.nz) |
| 06:39.51 | pcrane | evening all |
| 06:47.38 | *** join/#asterisk joobie (n=joobie@joobie.org) |
| 06:49.28 | joobie | guys anyone got references to help with deciding between 1.4 or 1.6? |
| 06:51.01 | drmessano | What decision is there? |
| 06:51.10 | drmessano | You either want newer or you dont.. |
| 06:55.14 | trnzmeta | err cutting edge isn't always the greatest |
| 06:55.22 | trnzmeta | ginuea pig... oink oink |
| 06:56.20 | joobie | interested to read about a feature comparison between the two, to see if there's any features in there that are worthwhile trying 1.6 for |
| 07:05.10 | drmessano | Yeah, then you end up with people running 1.2 forever and not sure why none of their dialplan works in 1.6 |
| 07:08.43 | joobie | hmm.. how long wil 1.4 be supported for btw? |
| 07:14.11 | *** join/#asterisk Chris-NB (n=chris@nfw.ecos.at) |
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| 07:28.00 | kaldemar | joobie: http://svn.digium.com/view/asterisk/tags/1.6.0/CHANGES?view=markup |
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| 07:32.39 | Daejeo | anyone from states or enum config guy? |
| 07:33.00 | Daejeo | who can call my toll free number |
| 07:33.17 | Daejeo | i want to test a voice quality |
| 07:33.50 | Daejeo | ping someone? |
| 07:34.01 | Daejeo | ping anyone? |
| 07:34.29 | Daejeo | ping *? |
| 07:34.54 | *** join/#asterisk kamanashisroy (n=kamanash@119.30.34.2) |
| 07:40.33 | joobie | thanks kaldemar |
| 07:40.46 | *** join/#asterisk thedonvaughn (i=jvaughn@unaffiliated/printk) |
| 07:48.37 | Daejeo | joobie: |
| 07:48.39 | Daejeo | can u |
| 07:48.47 | Daejeo | call |
| 07:49.36 | Daejeo | :( |
| 07:54.14 | aliraja | Hi background application in some of the IVR prompts(no silence is there in IVR prompts) take long time after pressing instruct digit to goto next menu ...any suggestion to make it fast |
| 07:59.15 | *** join/#asterisk mvanbaak (i=vanbaak@asterisk/contributor-and-bug-marshal/mvanbaak) |
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| 08:06.39 | kaldemar | Daejeo: call it yourself |
| 08:06.52 | *** join/#asterisk xacatecas (n=jkroon@dsl-242-193-146.telkomadsl.co.za) |
| 08:07.48 | Daejeo | kaldemar: i did, but i need an opinion of other persons |
| 08:10.51 | *** join/#asterisk denon (i=root@synapse.subneural.net) |
| 08:10.51 | *** mode/#asterisk [+o denon] by ChanServ |
| 08:13.43 | kaldemar | Daejeo: you should be able to decide for yourself if your own system has god enough quality. |
| 08:14.16 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
| 08:14.16 | *** mode/#asterisk [+o denon] by ChanServ |
| 08:15.19 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
| 08:15.23 | *** mode/#asterisk [+o denon] by ChanServ |
| 08:17.00 | xacatecas | if i issue a call comes in, and somebody answers it, then a channel is created to that phone right? now if that person transfers the call, does it destroy the channel to that phone before creating the new channel towards the transferred phone? |
| 08:17.30 | xacatecas | then also, i just made two outbound calls from a gxp2010 phone, then tried to transfer them to one another ... but was only able to eventually conference them, and as soon as i put down the phone it killed the entire conference. |
| 08:17.52 | xacatecas | any tips/pointers/stuff i can look at? |
| 08:42.04 | *** join/#asterisk steliosk (n=Stelios@ipa107.2.tellas.gr) |
| 08:48.19 | synthetiq | anyone know what this error means? ZT_SPANCONFIG failed on span 1: Invalid argument (22) |
| 08:50.27 | tzafrir_laptop | synthetiq, compare /proc/zaptel/1 to the span=1,... you have in /etc/zaptel.conf |
| 08:52.03 | synthetiq | i have it in zaptel.conf set up for an e1, but its showing channel config for a t1 |
| 08:52.07 | *** join/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 08:52.24 | tzafrir_laptop | synthetiq, what card do you have? |
| 08:52.55 | synthetiq | [ 9.259234] wcte12xp: Found a Wildcard TE122 |
| 08:53.17 | mDuff | finally talks the backported-from-trunk pbx_lua into working roughly as advertised, posts as much to -users, and goes to bed. |
| 08:54.01 | *** join/#asterisk gr0mit (n=tim@dhcp3.zuk40.mot-tools.co.uk) |
| 08:54.56 | synthetiq | any ideas what to do tzafrir_laptop ? |
| 08:55.40 | *** join/#asterisk stimpie (n=stimpie@213-73-211-48.cable.quicknet.nl) |
| 08:55.42 | tzafrir_laptop | I guess you'll have to play with t1e1override or with a jumper or so |
| 09:00.10 | kaldemar | t1e1override=0xff as a modprobe argument should make it E1. |
| 09:00.39 | tzafrir_laptop | is it just me or the editing wars in voip-info's front page are not funny anymore |
| 09:02.05 | synthetiq | modprobe t1e1override=0xff ? |
| 09:02.15 | synthetiq | or modprobe zaptel t1e1override=0xff |
| 09:02.58 | tzafrir_laptop | or add the line 'options wct4xxp t1e1override=0xff' to a file under /etc/modprobe.d |
| 09:02.58 | synthetiq | or should i just edit the source code |
| 09:03.29 | tzafrir_laptop | cat /sys/modules/wct4xxp/parameters/t1e1override |
| 09:03.45 | tzafrir_laptop | err... wrong module name |
| 09:03.56 | tzafrir_laptop | use wcte1xp instead |
| 09:04.26 | tzafrir_laptop | or better: wcte12xp :-) |
| 09:04.41 | synthetiq | wc->spantype = TYPE_E1; |
| 09:05.30 | tzafrir_laptop | synthetiq, if you edit the source, change the default vale of t1e1override |
| 09:05.46 | tzafrir_laptop | that's the safest way |
| 09:12.39 | *** join/#asterisk ana_micho (n=sniper_v@87.236.144.38) |
| 09:15.14 | *** join/#asterisk chendy (n=chatzill@58.60.221.66) |
| 09:15.25 | ana_micho | Hi all,I'm facing an issue with my asterisk server when an extension on an X-Lite softphone tries to register on it...A huge amount of packets is exchanged between endpoint and asterisk server while the X-Lite is online...Even when I sign out from X-Lite, the asterisk server continues sending packets to my machine...Can Someone help me in that? Please find the SIP packets between asterisk and X-Lite on http://pastebin.com/d85f913e |
| 09:20.00 | *** join/#asterisk xacatecas (n=jkroon@dsl-242-193-146.telkomadsl.co.za) |
| 09:24.27 | *** join/#asterisk mikkel (n=mikkel@84-238-113-66.u.parknet.dk) |
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| 09:27.19 | phpboy | ana_micho: this may be asterisk sending checks or keep alives |
| 09:35.46 | kaldemar | ana_micho: take a look at "qualify" options in sip.conf. |
| 09:42.39 | ana_micho | kaldemar, qualify is commented in sip.conf under general context |
| 09:49.13 | *** join/#asterisk LND (n=lee@92.233.208.244) |
| 09:50.52 | *** join/#asterisk qdk (n=qdk@85.235.253.141) |
| 09:57.32 | tzafrir_laptop | isn't it funny that people get the habbit of using a pastebin also when posting to a mailing list? |
| 09:57.34 | *** join/#asterisk netspex (n=netspex@151.59.78.212) |
| 10:00.56 | *** join/#asterisk protocols (n=protocol@p5791FD67.dip.t-dialin.net) |
| 10:01.04 | protocols | hiall |
| 10:01.26 | protocols | I get an error when login into asterisk.org |
| 10:01.46 | protocols | Query failed: Duplicate entry '60496' for key 1 INSERT INTO phpbb_users (`user_id`,`username`,`user_password`,`user_regdate`,`user_email`,`user_timezone`) VALUES(60496,'G�nter','xxx',1225360817,'guenter@grodotzki.ph',1.0) |
| 10:02.52 | *** join/#asterisk postel (n=jp@wikimedia/Postel) |
| 10:04.20 | *** join/#asterisk tiav (n=tiav@91.197.165.222) |
| 10:09.15 | *** join/#asterisk netspex (n=netspex@151.59.78.212) |
| 10:09.16 | yang | gee, digium support takes 5 days to answer my email |
| 10:13.30 | protocols | ah now it works, when I login via asterisknow |
| 10:20.26 | *** join/#asterisk danalien (n=danalien@unaffiliated/danalien) |
| 10:22.48 | *** join/#asterisk zydoon (n=zydoon@41.225.140.76) |
| 10:22.55 | *** part/#asterisk zydoon (n=zydoon@41.225.140.76) |
| 10:33.35 | *** join/#asterisk drew (i=drew@whitehat.org) |
| 10:35.58 | drew | a question: how is the translation to extensions coming in done, does the service provider send the NNNNN through, or is it a config related issue, I have two sites, each running different versions of asterisk, one gets sent NNNN the other NNNNNNN, each extensions.conf is configured for each of the incoming exntensions, but i'm trying to 'normalize' them so i'm curious as to where it originates from |
| 10:37.20 | *** join/#asterisk angryuser (n=gdobrovo@LPuteaux-151-42-35-99.w193-251.abo.wanadoo.fr) |
| 10:39.54 | kaldemar | drew: you might want to rephrase your issue. show a concrete problem. |
| 10:41.26 | drew | kaldemar: i'm trying to work out where asterisk gets the incoming extension |
| 10:42.23 | drew | is it sent from the service provider, or is the whole number parsed by asterisk first, then by some rule the checked against the extensions.conf |
| 10:42.33 | drew | so if i dial 1234567890 |
| 10:42.44 | drew | site a)'s extension it see's is 7890 |
| 10:42.52 | drew | site b)'s extension it see's is 4567890 |
| 10:43.12 | drew | parsing extension incoming -> sip/extension is fine |
| 10:43.48 | drew | site b) has exten => 4567890,etc etc |
| 10:43.50 | kaldemar | drew: the service provider sends a number and your dialplan does what you make it do to it. |
| 10:43.58 | drew | site a) has exten => 7890,etc etc |
| 10:44.24 | kaldemar | asterisk does what you tell it to do. |
| 10:44.31 | drew | agreed |
| 10:44.33 | kaldemar | it doesn't do anything by itself. |
| 10:44.56 | drew | i'm trying to find out where the raw number thats sent from the SP comes from / is |
| 10:45.24 | drew | ie: are they only sending me 4 numbers |
| 10:46.12 | kaldemar | put a exten => _X.,1,NoOp(${EXTEN}) in your incoming context and you'll see what you get from the provider. |
| 10:48.39 | drew | ok |
| 11:05.43 | *** join/#asterisk Chris-NB (n=chris@213162066194.public.t-mobile.at) |
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| 11:37.42 | *** join/#asterisk neuwald (n=neuwald@200.252.126.130) |
| 11:39.32 | neuwald | hi folks. I´m trying to use asterisk realtime (sip peers and users), and i´m getting a lot of messages on console screen, as you can see at http://www.pastebin.ca/1240512 - repetition messages of res_config_mysql debug |
| 11:39.50 | neuwald | and, I can´t make it work |
| 11:41.25 | *** join/#asterisk ast-thecode (n=asterisk@scalise.csata.tno.it) |
| 11:42.09 | neuwald | here u can see some other config files: http://www.pastebin.ca/1240516 |
| 11:42.18 | neuwald | and database schema and information |
| 11:55.25 | *** join/#asterisk steliosk (n=Stelios@ipa107.2.tellas.gr) |
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| 12:17.03 | *** join/#asterisk aksyn (n=aksyn@gw.na.nu) |
| 12:22.08 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 12:32.01 | *** join/#asterisk rwaite (n=richard@rrcs-74-218-125-86.central.biz.rr.com) |
| 12:35.07 | *** join/#asterisk Dr-Linux|home (n=Nothing@119.63.130.34) |
| 12:35.08 | *** join/#asterisk wolfelectronic (n=wolfelec@91.112.227.150) |
| 12:35.16 | Dr-Linux|home | anyobody is using Asterisk AGI in java? |
| 12:36.00 | rwaite | no, but it can't be much different than anything else |
| 12:36.10 | rwaite | ask your question |
| 12:37.18 | Dr-Linux|home | rwaite: not sure, why verbose is not working for me |
| 12:37.31 | Dr-Linux|home | this.verbose("Before getting DEST number",4); |
| 12:37.48 | Dr-Linux|home | i'm doing this, but it doesn't print anythig on CLI .. even agi works accordingly |
| 12:38.09 | [TK]D-Fender | Dr-Linux|home: That looks like somebody else's AGI wrapper... |
| 12:38.11 | rwaite | are you getting the response afterward? |
| 12:38.28 | rwaite | and what [TK]D-Fender said |
| 12:38.40 | [TK]D-Fender | Dr-Linux|home: And confirm your verbose level at CLI... 3 is default don't forget... |
| 12:38.43 | kaldemar | what? doesn't vanilla java have an asterisk class? |
| 12:38.52 | [TK]D-Fender | kaldemar: ABSURD! |
| 12:39.25 | Dr-Linux|home | rwaite: yes, it is crossing .. all working .. but not the verbose |
| 12:39.37 | Dr-Linux|home | [TK]D-Fender: verbose is set to 99 |
| 12:40.01 | Dr-Linux|home | [TK]D-Fender: i'm using Asterisk-java |
| 12:40.22 | [TK]D-Fender | Dr-Linux|home: Show us your complete attempt at verbose 10, AGI debug enabled. |
| 12:40.39 | [TK]D-Fender | Dr-Linux|home: And while you're at it, your code |
| 12:44.25 | kaldemar | [TK]D-Fender: just like giving a wrench with a burger meal, just in case. |
| 12:44.44 | [TK]D-Fender | kaldemar: "Now with sprinkles on top!" |
| 12:45.13 | *** join/#asterisk coppice (n=chatzill@33.155.17.210.dyn.pacific.net.hk) |
| 12:47.44 | *** join/#asterisk Nunners (n=james@mail.nadn.co.uk) |
| 12:47.57 | rwaite | Hmm, is there a "correct" way to issue a 'sip reload' from the dialplan? Sometimes it will return busy/congested and a 'sip reload' corrects it, so I'm thinking of coding some logic to try the reload once before playing a fastbusy |
| 12:48.16 | rwaite | I can hack it together easily, but just checking if there's some proper was to do it |
| 12:48.31 | [TK]D-Fender | rwaite: System(/usr/sbin/asterisk -rx "sip reload") |
| 12:48.51 | rwaite | sweet. do you see any caveats about my solution? |
| 12:49.09 | [TK]D-Fender | rwaite: nothing to care about |
| 12:49.20 | rwaite | Thanks |
| 12:49.21 | *** join/#asterisk jasonwoot (n=jasonrot@69.73.89.233) |
| 12:49.24 | jasonwoot | woot |
| 12:50.36 | Nunners | hi all - remember me?!!!! |
| 12:50.55 | rwaite | jason |
| 12:51.01 | [TK]D-Fender | Nunners: We can't afford the kind of psychotherapy it'd take to do otherwise... |
| 12:51.24 | Nunners | Oh thanks... was I that bad yesterday? |
| 12:51.52 | Nunners | You don't need to comment on that! |
| 12:56.00 | Nunners | For those of you who can remember me from yesterday, I have now successfully installed and configured (I believe) my TDm410p... |
| 12:57.06 | kaldemar | this sounds fun. by all means, continue. |
| 12:57.18 | Nunners | I can't now though get asterisk to recognise an incoming call - having following the book and loads of examples on * config... |
| 12:57.33 | *** join/#asterisk MoarDesu (i=xxx@togi.homeunix.org) |
| 12:58.16 | tzafrir_laptop | Nunners, counfiguring analog cards is simple |
| 12:58.18 | MoarDesu | Good morning... could anyone tell me how, if there's a way, to disable a single b channel on a Zaptel card? |
| 12:58.35 | tzafrir_laptop | MoarDesu, permanently? |
| 12:58.48 | MoarDesu | Well, I'd want to be able to re-enable it |
| 12:58.50 | tzafrir_laptop | zap destroy channel NNN |
| 12:58.57 | tzafrir_laptop | too bad :-( |
| 12:59.20 | MoarDesu | What does it take to reenable a channel after doing that? |
| 12:59.37 | [TK]D-Fender | Nunners: Feel free t pastebin your zapata.conf and your inbound dialplan context |
| 12:59.38 | tzafrir_laptop | restarting asterisk would do |
| 12:59.50 | MoarDesu | Don't need to reload the zap module or anything? Hmmm K. |
| 13:00.15 | tzafrir_laptop | Nunners, what's the output of lszaptel / lsdahdi . Are the channels (In use) ? |
| 13:00.35 | tzafrir_laptop | What the output of: asterisk -rx 'dahdi show channels' |
| 13:00.42 | MoarDesu | Ugh, this is why voip-info sucks: zap destroy channel: Destroy a channel |
| 13:01.03 | MoarDesu | How about defining "destroy" in the context of asterisk channels? |
| 13:01.03 | *** join/#asterisk unasi7 (n=unasi7@84-75-23-200.dclient.hispeed.ch) |
| 13:01.31 | unasi7 | cound anyone give a hand solving touble with incomming calls on asterisk 1.6? |
| 13:01.40 | Nunners | ok - I've just finally gone through it, and can't see what I've got wrong... zapata.conf & extensions.conf bits: http://pastebin.com/d5b860966 |
| 13:02.14 | *** join/#asterisk jer (n=jer@unaffiliated/jer) |
| 13:02.21 | Nunners | Asterisk sees the incoming call, but reports: "Starting simple switch on 'Zap/3-1'" so guess I have my referencing wrong? |
| 13:02.37 | rwaite | MoarDesu: it's a wiki. figure it out, and then add it to the wiki. jesus. |
| 13:02.51 | [TK]D-Fender | Nunners: Your signalling is whacked. |
| 13:02.54 | rwaite | people feel so entitled nowadays |
| 13:03.07 | Nunners | Ok - in what way? |
| 13:03.28 | [TK]D-Fender | Nunners: if 3 is your FXO module (red), then it should be fxs_ks, and #1 should be fxo_ls |
| 13:04.00 | [TK]D-Fender | Nunners: signalling=fxo_ks ; Use FXS signalling for an FXO channel <-- look at your own comment. Its commented one way and you do another... |
| 13:04.07 | Nunners | [TK} 1 is red... 3 green |
| 13:04.23 | Nunners | [TK] ignore the comments - that was fron an example... |
| 13:04.25 | [TK]D-Fender | Nunners: Next set your core debug to 10, and verbose 10 and pastebin a failed call after fixing & restarting * |
| 13:04.39 | [TK]D-Fender | Nunners: then you are getting even further mixed up |
| 13:04.58 | *** join/#asterisk Segnale007 (n=Segnale0@host243-255-dynamic.32-79-r.retail.telecomitalia.it) |
| 13:05.08 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 13:05.12 | [TK]D-Fender | Nunners: if 1 is rd then THAT is what your PSTN should be plugged into. |
| 13:05.21 | unasi7 | i just try to ask: sip -> sip is running,... sip -> outbound is running,.. outbound -> sip WONT run. Debug: http://pastebin.com/m2dddd6d1 any ideas? |
| 13:05.28 | [TK]D-Fender | Nunners: make sure before your fry your card |
| 13:05.42 | Nunners | ok.... will double/triple check thanks... |
| 13:06.04 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:06.23 | [TK]D-Fender | unasi7: SIP/2.0 401 Unauthorized <- bad auth. |
| 13:06.23 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
| 13:07.07 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 13:07.08 | unasi7 | [TK]D-Fender: yes. i see it. but i habe no idea, where i can change that. password is right (sip -> outbound is working). any hint? |
| 13:07.17 | kaldemar | unasi7: does the other end respond to your 401? |
| 13:07.43 | [TK]D-Fender | unasi7: Maybe they are sending you UN-AUTHED calls on inbound. |
| 13:07.51 | *** join/#asterisk kannan (n=kannan@123.201.136.118) |
| 13:08.01 | unasi7 | kaldemar: yes: http://pastebin.com/m2b9caf42 |
| 13:08.09 | [TK]D-Fender | unasi7: try "insecure=port,invite" for it |
| 13:08.40 | unasi7 | [TK]D-Fender is was working on another system with insecure=very .. but i give it a try |
| 13:08.41 | kaldemar | unasi7: a SIP secret is not symmetric, so working outbound authentication doesn't tell you anything about inbound authentication. |
| 13:09.21 | unasi7 | kaldemar .. okay. great hint. but how can i allow all incomming calls from a registry? |
| 13:09.21 | MoarDesu | rwaite: please, be more rude |
| 13:09.30 | MoarDesu | rwaite: i bet you are a popular guy |
| 13:09.52 | tzanger | heh |
| 13:09.53 | kaldemar | unasi7: try what [TK]D-Fender told you to try |
| 13:10.06 | rwaite | i'm the rude one? you're the one bitching about a fucking free wiki, provided to you |
| 13:10.07 | unasi7 | WOW: insecure=port,invite was it! |
| 13:10.12 | rwaite | get a grip |
| 13:10.28 | *** join/#asterisk stencil (n=stencil@pdpc/supporter/student/stencil) |
| 13:10.28 | MoarDesu | rwaite: i was pointing out a problem, you can view it constructively or not |
| 13:10.36 | [TK]D-Fender | rwaite: Cool it... |
| 13:10.43 | unasi7 | great ... [TK]D-Fender! ... thanks very much. will check the wiki / help for more infos about what i did. :) |
| 13:10.44 | rwaite | go back to 4chan, i'm done with you |
| 13:10.56 | MoarDesu | rwaite: thank heavens |
| 13:11.23 | MoarDesu | If it were limited to that one instance, fixing it would be the only necessary remedy. |
| 13:11.45 | [TK]D-Fender | MoarDesu: YYMV as far as voip-info goes. a lot of stuff isn't quite valid for 1.4 as it is. 1.6 is virtually non-existant there |
| 13:12.06 | [TK]D-Fender | MoarDesu: Guess people lost the will to maintain. |
| 13:12.23 | MoarDesu | [TK]D-Fender: we were discussing the poor quality of on-line * docs back at VON in boston last year |
| 13:12.32 | MoarDesu | kinda sad. :\ |
| 13:12.52 | [TK]D-Fender | MoarDesu: You can be part of the solution you know... |
| 13:13.31 | MoarDesu | I don't feel like I know enough |
| 13:13.59 | MoarDesu | That's why I want lots of documentation. :) But yeah, I do understand your point. |
| 13:14.42 | *** join/#asterisk festr_ (n=festr@ns.hiro.cz) |
| 13:14.55 | rwaite | I, I, I, I, I |
| 13:15.03 | rwaite | I? I.. I I! |
| 13:15.14 | MoarDesu | rwaite: Would you prefer me to use the royal we? |
| 13:15.17 | festr_ | is it possible to call from sip client something@somedomain and asterisk will handle this or i have to use some sip proxy? |
| 13:15.26 | MoarDesu | Does my grammar offend you? |
| 13:15.32 | MoarDesu | Sheesh. |
| 13:15.40 | rwaite | oh, sorry. i was just talking about myself |
| 13:16.06 | tzanger | wow fun times in #asterisk this morning |
| 13:16.25 | [TK]D-Fender | festr_: "allowguets=" under [general] and point them to a context |
| 13:16.46 | *** join/#asterisk Linuturk (n=Linuturk@fluxbuntu/developer/Linuturk) |
| 13:16.50 | [TK]D-Fender | rwaite: If your "I" offends you... |
| 13:16.52 | festr_ | [TK]D-Fender: yes but i want to forward request to somedomain |
| 13:16.55 | *** part/#asterisk Deeewayne (n=dwayne@76.29.245.9) |
| 13:16.57 | [TK]D-Fender | hands rwaite a spoon... |
| 13:17.07 | [TK]D-Fender | festr_: "core show application transfer" |
| 13:17.11 | festr_ | [TK]D-Fender: i can call blabla or blabla@anotherdomain |
| 13:17.29 | festr_ | [TK]D-Fender: so asterisk will Dial(SIP/blabla@anotherdomain) |
| 13:17.31 | rwaite | lol |
| 13:17.49 | [TK]D-Fender | festr_: No, Dial is not Transfer |
| 13:18.12 | festr_ | [TK]D-Fender: yes. i want to record the call |
| 13:18.15 | Dr-Linux|home | [TK]D-Fender: thanks vorbose start working |
| 13:18.17 | kaldemar | asterisk can even send blablabla audio to blabla@anotherdomain if you make it do it. |
| 13:18.24 | Dr-Linux|home | i'd also like to share my another problem |
| 13:18.30 | [TK]D-Fender | festr_: Then you'd better make uup your mind. You're changing stories on us now. |
| 13:18.41 | festr_ | my appologies |
| 13:18.52 | [TK]D-Fender | festr_: then just Dial it. |
| 13:19.13 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 13:19.14 | [TK]D-Fender | Dr-Linux|home: You didn't even share this last one. |
| 13:19.24 | festr_ | [TK]D-Fender: so it is possible to extract @domain from incoming invite sip uri |
| 13:19.26 | [TK]D-Fender | Dr-Linux|home: we asked for a pastebin and like usual got nothing. |
| 13:19.45 | [TK]D-Fender | festr_: *'s INVITE format is fixed. |
| 13:20.14 | Dr-Linux|home | [TK]D-Fender: Actually there is long long java code and many closes .. so not sure how to pasbin |
| 13:20.32 | [TK]D-Fender | Dr-Linux|home: Yeah, ok, fine, sure. |
| 13:20.51 | Dr-Linux|home | [TK]D-Fender: but figured out, agi verbose should not more then CLI console verbose |
| 13:21.00 | Dr-Linux|home | that's what i learnt |
| 13:21.00 | festr_ | ok i should be more exact. if i dial some number it goes to predefined context. but if i dial from sip client number@somefrienddomain.com I'd like to Dial(number@somefrienddomain.com) |
| 13:21.08 | *** join/#asterisk stephbul (n=stephbul@bulot.org) |
| 13:21.38 | Dr-Linux|home | my problem is: |
| 13:21.50 | [TK]D-Fender | festr_: * is not a proxy. it is a B2BUA. It will not match realms, etc. |
| 13:22.03 | festr_ | and the question is, how to extract somefrienddomain.com so i can Dial(SIP/${EXTEN}@${DOMAIN}) |
| 13:22.24 | angryuser | festr_: use proxy |
| 13:22.29 | [TK]D-Fender | festr_: And in that sample * IS that domain... so what is it going to do, call itself again? |
| 13:23.06 | festr_ | no the point is that * is not that domain |
| 13:23.27 | festr_ | when i call number@usa.cz i want from asterisk to call SIP/number@usa.cz |
| 13:23.29 | [TK]D-Fender | festr_: "sip client number@somefrienddomain.com " got him INTO your * server. Dial(number@somefrienddomain.com) will only point to * AGAIN. |
| 13:23.45 | [TK]D-Fender | festr_: Unless you're telling your SIP client that * is its proxy... which * ISN'T. |
| 13:24.10 | festr_ | yes it is not but i think it should be possible? |
| 13:24.19 | Dr-Linux|home | when i dial >>>> Dialplan ext that goes to agi >>> dial(2222) again dialplan exte which dials through SIP provider, when provider answers the call, it connect the first channel with provider channel but it drops the 2222 extensions bridged channel |
| 13:24.24 | [TK]D-Fender | festr_: Take a look at the SIP headers of your inbound call and see if there is something you can parse from it |
| 13:24.45 | Dr-Linux|home | sorry, not good explaination .. |
| 13:24.55 | [TK]D-Fender | Dr-Linux|home: I'm just going to sit here and wait for you to pastebin something useful... |
| 13:25.02 | festr_ | ok got it |
| 13:25.03 | festr_ | ${SIPDOMAIN} * SIP destination domain of an inbound call (if appropriate) |
| 13:25.24 | festr_ | so, Dial{SIP/${EXTEN}@${SIPDOMAIN}) |
| 13:25.28 | festr_ | i'll try |
| 13:25.37 | Dr-Linux|home | [TK]D-Fender: what should I pastebin? |
| 13:25.49 | festr_ | why I dont try to look at this first than asking? :) anyway thanks for conversation |
| 13:25.53 | *** join/#asterisk etech3 (n=chatzill@68-243-103-134.area7.spcsdns.net) |
| 13:26.18 | [TK]D-Fender | Dr-Linux|home: Ok, you can't describe your problem, you can't show us whats going on. I cannot help you. |
| 13:26.36 | [TK]D-Fender | goes off to do something productive. |
| 13:26.40 | Dr-Linux|home | ok no problem |
| 13:27.47 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
| 13:29.00 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
| 13:35.58 | *** join/#asterisk bkw_ (n=brian@freeswitch/developer/bkw) |
| 13:37.08 | *** join/#asterisk propellerhead (n=yogurt2u@host38.190-136-115.telecom.net.ar) |
| 13:37.17 | *** join/#asterisk sabotralala (n=sabo@193.34.64.75) |
| 13:37.36 | sabotralala | Hi, i have a problem with my isdn phone line, anyone could help me? thx |
| 13:37.45 | *** join/#asterisk Chris-NB (n=chris@nfw.ecos.at) |
| 13:39.26 | *** part/#asterisk stencil (n=stencil@pdpc/supporter/student/stencil) |
| 13:40.57 | tzafrir_laptop | sabotralala, what ISDN channel driver do you use? What card? |
| 13:42.39 | sabotralala | beronet bn4s0 |
| 13:42.40 | sabotralala | misdn |
| 13:42.42 | *** part/#asterisk drew (i=drew@whitehat.org) |
| 13:43.04 | sabotralala | the problem is that inbound calls are working perfectly but outbound calls are not |
| 13:43.44 | sabotralala | when i call someone through the isdn line, the line is hung up immediately when the remote party picks up the phone |
| 13:45.51 | Katty | dear lord. |
| 13:45.55 | Katty | if they hire ANY more females here |
| 13:45.58 | Katty | I'm going to shoot myself |
| 13:46.46 | Katty | Dear Powers That Be, please PLEASE send SANE MALES for me to work with. I can't stand the drama ANYMORE!!!! thx, Kat |
| 13:47.40 | *** join/#asterisk nikko (n=nikko@69.57.49.100) |
| 13:48.06 | *** join/#asterisk n3hxs (n=HAMming@63.68.135.4) |
| 13:48.18 | *** join/#asterisk TommyBJ (n=noosjent@193.160.28.100) |
| 13:49.31 | protocols | anybody here having a success story with * behind nat/router? version 1.4.19x ? |
| 13:49.34 | *** join/#asterisk stencil (n=stencil@pdpc/supporter/student/stencil) |
| 13:49.48 | Katty | protocols: 1.4.22 works properly |
| 13:49.56 | [TK]D-Fender | protocols: Works fine |
| 13:50.01 | protocols | .19 not? |
| 13:50.18 | Katty | 1.4.18 as well |
| 13:50.19 | TommyBJ | .17 works, so I presume .19 does |
| 13:50.35 | Katty | [TK]D-Fender: i need to borrow a sword. |
| 13:50.42 | Katty | [TK]D-Fender: something particularly sharp. |
| 13:50.53 | protocols | oh i just see that I am on .21 |
| 13:50.55 | protocols | hmm.. |
| 13:50.55 | [TK]D-Fender | Katty: that new one of mine is very sharp |
| 13:50.57 | Katty | [TK]D-Fender: and by sharp i mean cut a pineapple in one slice. |
| 13:51.07 | edwin_quijada | Hi! |
| 13:51.07 | [TK]D-Fender | is awaiting going for corrective surgery.... |
| 13:51.09 | protocols | Katty, use force |
| 13:51.09 | Katty | [TK]D-Fender: can it cut bone? |
| 13:51.15 | Katty | protocols: hmm? |
| 13:51.17 | Katty | edwin_quijada: ohai! |
| 13:51.22 | edwin_quijada | i AM getting error when I do ztcfg -vv |
| 13:51.31 | edwin_quijada | line 0: Unable to open master device '/dev/zap/ctl' |
| 13:51.37 | protocols | use your whole weight.. should do it |
| 13:51.39 | [TK]D-Fender | Katty: pretty sure.... I can guarantee it does human flesh quite well |
| 13:51.42 | protocols | even with a spoon |
| 13:51.50 | Katty | [TK]D-Fender: oh goody. |
| 13:51.51 | TommyBJ | edwin_quijada: Does the special file exist? |
| 13:51.58 | Katty | [TK]D-Fender: i'd like to setup an assassination appointment with you! *hee* |
| 13:52.20 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 13:52.20 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 13:52.26 | edwin_quijada | TommyBJ:no |
| 13:52.28 | Katty | lmadsen: oh |
| 13:52.30 | Katty | lmadsen: you again |
| 13:52.35 | protocols | edwin_quijada, please post somewhere the exact procedure |
| 13:52.38 | Katty | lmadsen: you're always in here. |
| 13:52.41 | lmadsen | totally me! |
| 13:52.45 | Katty | lmadsen: do we hug? |
| 13:52.45 | TommyBJ | edwin_quijada: check wether the zaptel module is loaded. What kind of card are you using? |
| 13:52.47 | lmadsen | totally tubular! |
| 13:52.55 | lmadsen | not always... I just wasn't a few mins ago! |
| 13:53.04 | edwin_quijada | TommyBJ: D110PG T1 openvox card |
| 13:53.07 | lmadsen | Katty: let me evaluate the situation, then I will update you |
| 13:53.07 | protocols | but i reckon you mentioning you were using centos |
| 13:53.14 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 13:53.14 | Katty | lmadsen: k, you have 10 seconds. |
| 13:53.14 | protocols | which I have no idea of |
| 13:53.17 | Katty | lmadsen: process quickly. |
| 13:53.27 | lmadsen | processes |
| 13:53.28 | Katty | [TK]D-Fender: also, Mew. |
| 13:53.30 | Katty | hugs [TK]D-Fender |
| 13:53.37 | [TK]D-Fender | Katty: Mew |
| 13:53.38 | lmadsen | determines a hug is allowed |
| 13:53.40 | lmadsen | hugs katty! |
| 13:53.41 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
| 13:53.44 | Katty | hugs lmadsen! |
| 13:53.55 | protocols | hmm is there any good tutorial for stupidos like me to get * behind nat working |
| 13:54.00 | TommyBJ | edwin_quijada: Are the drivers correctly loaded? .. |
| 13:54.01 | Katty | jbot: natting? |
| 13:54.03 | [TK]D-Fender | ~sipnat |
| 13:54.03 | 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 |
| 13:54.05 | *** join/#asterisk gsiener (n=gsiener@209.169.48.66) |
| 13:54.07 | [TK]D-Fender | ^^^^ |
| 13:54.08 | Katty | there we go. |
| 13:54.12 | edwin_quijada | TommyBJ: Opps! Sorry I didnt load the zaptel |
| 13:54.15 | [TK]D-Fender | protocols: You've been referred there before... |
| 13:54.16 | protocols | ah cool thank you |
| 13:54.17 | Katty | protocols: let me see if i blogged anything |
| 13:54.18 | edwin_quijada | Thks!!! :( |
| 13:54.24 | TommyBJ | edwin_quijada: :) |
| 13:54.30 | protocols | yeah? oops did not get it |
| 13:54.32 | Katty | i guess i need to blog something on natting |
| 13:54.37 | protocols | yes please |
| 13:54.45 | protocols | and ping back me when done :D :P |
| 13:55.05 | Katty | everything but natting, it seems. |
| 13:55.09 | *** join/#asterisk gsiener (n=gsiener@209.169.48.66) |
| 13:55.13 | Katty | i should blog something on natting. |
| 13:55.19 | [TK]D-Fender | Katty: Why? Its already blogged :) |
| 13:55.32 | Katty | [TK]D-Fender: personal reference. |
| 13:55.43 | Katty | [TK]D-Fender: the step by step dummy process helps me remember things |
| 13:55.46 | [TK]D-Fender | Katty: Ctrl-P ;) |
| 13:55.48 | rwaite | i read those nat documents and it looks like i have scenario 1 :( |
| 13:55.58 | protocols | hmm 1.4.22 forces dandhi already, or can I stick to zaptel? |
| 13:56.13 | rwaite | no, you can use zaptel |
| 13:56.15 | Katty | [TK]D-Fender: too much effort |
| 13:56.17 | [TK]D-Fender | protocols: DAHDI *is* zaptel, post-rename |
| 13:56.29 | Katty | i blogged dahdi! |
| 13:56.35 | protocols | i know, but dahdi is not yet supported by the gui |
| 13:56.36 | TommyBJ | protocols: You can set a parameter in asterisk.conf |
| 13:56.46 | gsiener | hi all. my cli has verbal diarrhea right now. just turned off debug for sip/iax, and core debug/verbose off, and I'm getting tons of stuff rolling through |
| 13:56.48 | protocols | hmm maybe I will try that |
| 13:56.48 | Katty | protocols: http://angela.sleekgeek.org/category/dahdi/ |
| 13:57.42 | rwaite | if i have a sip "trunk" with a provider, that would be considered a proxy, right? |
| 13:57.42 | tzafrir_laptop | protocols, 1.4.22 can work with either zaptel or dahdi . But it depends on how you build it |
| 13:57.42 | protocols | ok |
| 13:57.42 | [TK]D-Fender | protocols: Oh well... GUI's aren't supported here anyways... |
| 13:57.42 | Katty | Gooey is supported. |
| 13:57.42 | Katty | lmadsen is particularly fond of the gooey. |
| 13:57.42 | tzafrir_laptop | protocols, strings /usr/sbin/asterisk | grep /dev.*channel |
| 13:57.42 | protocols | yupp don't worry won't ask gui related questions here ;) I know the limitations |
| 13:57.42 | [TK]D-Fender | Katty: Only if it comes in chocolate chunks! |
| 13:57.50 | Katty | [TK]D-Fender: gooey is a universial term for YUM |
| 13:57.50 | protocols | thanks for the link katty |
| 13:58.00 | Katty | protocols: hope it helps you (= |
| 13:58.05 | lmadsen | Katty: I am fond of fondu |
| 13:58.23 | Katty | lmadsen: what sort of fondu? |
| 13:58.24 | protocols | I do hope, too |
| 13:58.31 | [TK]D-Fender | Katty: YUM YUM is a brand of potato chips... they'll do! |
| 13:58.33 | gsiener | What CLI commands can I use to stop logging? |
| 13:58.47 | Katty | logging is good for you :< |
| 13:58.53 | gsiener | agreed |
| 13:58.59 | Katty | turn the verbose down :> |
| 13:59.03 | [TK]D-Fender | lmadsen: I do fondue regularly, and its on special starting today too! 2$/ 350g pack! |
| 13:59.06 | Katty | what you don't see, won't hurt you! |
| 13:59.06 | gsiener | it's on 0 :( |
| 13:59.09 | TommyBJ | Hmm.. what is the general opinion on the sangoma cards? .. and their drivers? |
| 13:59.13 | Katty | oh :< |
| 13:59.24 | Katty | TommyBJ: Sangoma = <3 |
| 13:59.44 | TommyBJ | Katty: "better" than Digium? |
| 13:59.51 | Katty | TommyBJ: much. |
| 13:59.54 | Katty | TommyBJ: in my opinion |
| 14:00.03 | lmadsen | ~betterquestions |
| 14:00.03 | Katty | TommyBJ: be sure to get other opinions too. |
| 14:00.05 | gsiener | Katty: Getting tons of: Really destroying SIP dialog '2343b8e009109e506bc71ce429b0df7f@sip.islandschool.org' Method: NOTIFY |
| 14:00.20 | TommyBJ | lmadsen: :) |
| 14:00.20 | Katty | someone needs to rename that error |
| 14:00.21 | lmadsen | ~thebestquestions |
| 14:00.22 | jbot | hmm... thebestquestions is Whenever you ask a "what is the best..." or "who is the best..." type questions, you're asking for trouble, and possibly may be called a troll. These types of questions do not have answers. Your best bet is to rephrase the question as, "What kind of experience do people have with..." or "Who has experience with...". |
| 14:00.26 | Katty | destroying SIP dialog FOR REALZ |
| 14:00.30 | lmadsen | aha :) |
| 14:00.38 | TommyBJ | Katty: Sure... but thanks :) |
| 14:00.41 | [TK]D-Fender | gsiener: Ignore unless you have special reason to do otherwise |
| 14:01.08 | Katty | gsiener: don't have a lot of experience with the sip stuff. |
| 14:01.09 | gsiener | [TK]D-Fender: I'm trying to debug something and it's disconcerting to have them flying by continually |
| 14:01.15 | Katty | gsiener: unless it's a SIP phone |
| 14:01.24 | gsiener | Katty: yeah - sip phones |
| 14:01.26 | [TK]D-Fender | gsiener: Then don't be distracted by them |
| 14:01.30 | tzafrir_laptop | sabotralala, please pastebin a CLI trace from a call |
| 14:01.53 | Katty | lmadsen: have you heard the chicken techno song? |
| 14:01.59 | lmadsen | Katty: I believe so |
| 14:02.16 | Katty | lmadsen: :> |
| 14:03.01 | Katty | i feel the need to blog the chicken techno |
| 14:03.12 | lmadsen | dooooooooooooo it |
| 14:03.25 | gsiener | Katty: all my phones are continually registering - thoughts? |
| 14:03.37 | lmadsen | turn up the registration timeout |
| 14:03.47 | Katty | gsiener: what lmadsen said. |
| 14:03.50 | gsiener | ahh - right |
| 14:04.02 | gsiener | is there anyway to set the timeout for phones different than trunks? |
| 14:04.05 | *** join/#asterisk write_erase (n=Olivier@telindu015615-6.clients.easynet.fr) |
| 14:04.05 | Katty | can't say i've had that problem before. |
| 14:04.13 | lmadsen | gsiener: set it on the phone |
| 14:04.23 | Katty | gsiener: i believe the timeout is either at the phone or the sip.cfg controlling file |
| 14:04.30 | Katty | gsiener: if you're using polycoms, anyway |
| 14:04.30 | lmadsen | configure the phone to not re-register so quickly |
| 14:04.50 | gsiener | Katty: yep, polycoms |
| 14:04.53 | gsiener | thanks |
| 14:04.55 | Katty | looks at her 501 |
| 14:04.58 | [TK]D-Fender | gsiener: You never even actually described your problem.... |
| 14:05.04 | lmadsen | looks at his 501 |
| 14:05.14 | lmadsen | waits for Katty's 501 to ring his 501 |
| 14:05.17 | [TK]D-Fender | left his 501 at home... |
| 14:05.33 | lmadsen | [TK]D-Fender: then you are not permitted to enter the club today |
| 14:05.36 | Katty | i guess that would be under lines. |
| 14:05.47 | lmadsen | Katty: yes... am I not in the speed dial? |
| 14:05.54 | *** join/#asterisk whatever-thingy (n=whatever@79-77-67-129.dynamic.dsl.as9105.com) |
| 14:05.54 | rwaite | hmm. is sip_nat.conf deprecated, should i put it in sip.conf? |
| 14:05.55 | Katty | lmadsen: i don't think so |
| 14:05.57 | Katty | lmadsen: file is tho |
| 14:05.59 | lmadsen | I even have a SIP URI! :) |
| 14:06.05 | lmadsen | Katty: oh that is sooooooooooooo weak |
| 14:06.12 | Katty | lmadsen: /comfort |
| 14:06.18 | Katty | lmadsen: we don't talk anymore |
| 14:06.20 | [TK]D-Fender | rwaite: How can a file that has no implicit declaration in * be deprecated? |
| 14:06.26 | gsiener | [TK]D-Fender: yeah, but I figured out what was going on... |
| 14:06.34 | [TK]D-Fender | rwaite: that is not an * config file. |
| 14:06.35 | lmadsen | rwaite: sip_nat.conf has never been an asterisk configuration file... that sounds like a remnant of some sort of GUI based system |
| 14:06.44 | [TK]D-Fender | (FreePBX <-) |
| 14:06.46 | Katty | like trixbox |
| 14:06.49 | [TK]D-Fender | Yes... I know it well |
| 14:06.57 | Katty | i don't. |
| 14:06.59 | Katty | and i still don't like it. |
| 14:07.02 | lmadsen | me either |
| 14:07.06 | lmadsen | I don't use gui's :) |
| 14:07.08 | rwaite | oh, i'm just reading my itsp's docs |
| 14:07.10 | Katty | we must be elitest snobs. |
| 14:07.14 | lmadsen | totally |
| 14:07.29 | lmadsen | but I don't need asterisk to be an elitest snob |
| 14:07.32 | rwaite | sounded fishy to me so i asked |
| 14:07.44 | write_erase | I'm using chan_sccp, but I can see a lot of UDP packets between Asterisk and phones when a commnuication is established. Why UDP stream packets are going through ASTERISK ? There should only be signaling packets right ? |
| 14:07.45 | lmadsen | I live in downtown toronto y0! |
| 14:07.45 | Katty | always a good plan. |
| 14:08.06 | Katty | write_erase: asterisk works in UDP not TCP |
| 14:08.17 | lmadsen | Katty: that's what he said :) |
| 14:08.33 | *** join/#asterisk aksyn (n=aksyn@gw.na.nu) |
| 14:08.33 | Katty | lmadsen: you sure are fiesty this morning |
| 14:08.36 | Katty | someone ate his wheaties. |
| 14:08.46 | lmadsen | write__erase: I don't know chan_sccp... but it may be possible you have to enable a "reinvite" type option... ? |
| 14:08.46 | lmadsen | Qwell: ? |
| 14:09.19 | [TK]D-Fender | write_erase: SCCP isn't TCP that I recall, and the OTHER end of your call isn't SCCp so that channels voice is definitely UDP for RTP |
| 14:09.20 | write_erase | lmadsen, is reinvite in SIP responsible of RTP stream direction ? |
| 14:09.40 | lmadsen | Katty: nah... I've just worked a weeks worth of hours in 3 days, which have been spread over from 9am to 2am each day... so my sleep pattern is a little off |
| 14:09.41 | [TK]D-Fender | write_wWhy would SIP & SCCP reinvite to each other? |
| 14:09.48 | lmadsen | write__erase: only if you're using SIP |
| 14:09.50 | Katty | lmadsen: :< |
| 14:10.05 | Katty | lmadsen: go nap. |
| 14:10.33 | lmadsen | Katty: but it's ok because so far 3 major issues that have been plaguing me for 3 months got solved yesterday (upon initial conclusion) |
| 14:10.45 | write_erase | mmm... I just want the SCCP audio stream don't cross my Asterisk server, just want signaling goes through It |
| 14:11.00 | Katty | lmadsen: :> |
| 14:11.09 | Katty | lmadsen: this calls for an ice cream!! |
| 14:11.19 | Katty | lmadsen: do take yourself to dairy queen later today. |
| 14:11.45 | lmadsen | Katty: oh don't you worry... I'm taking off most of today probably in order to clean my condo, go grocery shopping, etc... |
| 14:11.58 | [TK]D-Fender | write_erase: Doesn't work that way. It isn't SCCP on the other side. * has to translate |
| 14:12.02 | Katty | hmm groceries. |
| 14:12.04 | Katty | i gotta get those too |
| 14:12.08 | Katty | and doggy biscuits. |
| 14:12.29 | lmadsen | ya, I'm pretty much out |
| 14:12.41 | Katty | of doggy biscuits? |
| 14:12.41 | lmadsen | speaking of which... I'm gonna finish off my milk and have some cereal |
| 14:12.55 | [TK]D-Fender | lmadsen: Imagine if you combined the two! |
| 14:13.16 | write_erase | [TK]D-Fender, I have 2 CISCO phone using SCCP , and 1 asterisk ... so no need to translate ? right ? |
| 14:13.52 | Katty | cisco seems complicated. |
| 14:13.53 | [TK]D-Fender | write_erase: Is * doing anything else with the call? |
| 14:14.03 | Katty | lmadsen: i had a whopper for breakfast. |
| 14:14.04 | lmadsen | guess I'm having half a bowl of cereal... less milk than I thought :) |
| 14:14.06 | [TK]D-Fender | write_erase: recording, listening for DTMF, etc? |
| 14:14.16 | lmadsen | Katty: I want a whopper and some skittles |
| 14:14.32 | [TK]D-Fender | write_erase: Do we see that SCCP even has a reinvite mechanism? Or that * has implemented it? |
| 14:14.38 | Katty | pesky skittles. they make me ill. |
| 14:14.47 | [TK]D-Fender | write_erase: Debug of an actual call might help... |
| 14:14.54 | lmadsen | write__erase: like do you have tT or wW flags in Dial()? |
| 14:15.10 | write_erase | yes tT |
| 14:15.12 | [TK]D-Fender | lmadsen: "Whopper" the burger, or "Whopper" the chocolate-coated malt ball? |
| 14:15.26 | [TK]D-Fender | write_erase: You shouldn't have Tt |
| 14:15.41 | [TK]D-Fender | write_erase: these phones have a transfer feature |
| 14:16.40 | TommyBJ | Are sangoma drivers free software licensed? |
| 14:17.03 | [TK]D-Fender | TommyBJ: GPL last I checked |
| 14:17.13 | write_erase | [TK]D-Fender, I'l try to remove these parametres . What's the problem with them ? |
| 14:17.20 | Katty | [TK]D-Fender: Burger King Whopper |
| 14:17.28 | [TK]D-Fender | write_erase: gives * a reason to sin in the middle |
| 14:17.31 | Katty | [TK]D-Fender: i can only handle so much egg. |
| 14:18.00 | *** join/#asterisk ManxPower (n=manxpowe@109.sub-70-220-163.myvzw.com) |
| 14:18.34 | bpgoldsb | Is there a way to tell between a call dropping and a call being properly hungup? |
| 14:18.39 | *** join/#asterisk theHub (n=theHub@69.177.93.21) |
| 14:18.41 | bpgoldsb | This is on a T1 |
| 14:18.52 | [TK]D-Fender | bpgoldsb: look at the PRI debug |
| 14:19.09 | Katty | morning Manx (= |
| 14:19.16 | ManxPower | bpgoldsb: Not really, as calls should never drop. You're not using callprogress=yes or busydetect=yes, are you? |
| 14:19.42 | ManxPower | waves to Katty |
| 14:19.45 | bpgoldsb | We've had upstream errors on our T1 (at the LEC's T3) the past few days. |
| 14:20.02 | bpgoldsb | It's manifested to us as both circuit outages (red alarm) and individual dropped calls. |
| 14:20.10 | bpgoldsb | I'm trying to figure out how to tell which is which |
| 14:20.11 | ManxPower | bpgoldsb: In that case, I don't know what the HANGUPCAUSE or DIALSTATUS would be. |
| 14:20.25 | ManxPower | a red alarm WILL drop calls |
| 14:20.30 | bpgoldsb | I know that :) |
| 14:20.41 | bpgoldsb | It's the ones without circuit drop that I can't determine |
| 14:20.46 | bpgoldsb | The only thing I have seen is... |
| 14:21.02 | ManxPower | bpgoldsb: Well callprogress=yes or busydetect=yes will randomly drop calls. |
| 14:22.30 | lmadsen | write__erase: tT tells asterisk to listen for DTMF transfers, which causes the stream to go through asterisk because.... it has to listen to dtmf |
| 14:22.57 | lmadsen | needs to update documentation to make it more clear that you don't need tT options to enable transfers for devices that can do it natively |
| 14:22.58 | bpgoldsb | ManxPower: http://pastebin.com/m3ce8aeb9 |
| 14:23.20 | bpgoldsb | Thats what I'm seeing in the logs when a call drops without a circuit outage |
| 14:25.00 | *** join/#asterisk Deeewayne (n=Deeewayn@nat/digium/x-fb3618f5c422c162) |
| 14:25.00 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 14:25.27 | n3hxs | Slippage such as Lose synch and re synch, will cause dropped calls (all in progress at the time of the slip) but does that cause an alarm? Possibly a yellow alarm. |
| 14:26.38 | jasonwoot | never gonna give, never gonna give |
| 14:26.44 | ManxPower | bpgoldsb: If you want me to help you then you need to answer my questions. |
| 14:26.45 | *** join/#asterisk ddunavant (n=David@75.145.240.14) |
| 14:26.53 | n3hxs | to really see what is happening most T1/PRI diagnostics require a T-Burd or similar equipment. |
| 14:27.21 | ManxPower | T-Berd |
| 14:28.52 | *** part/#asterisk wolfelectronic (n=wolfelec@91.112.227.150) |
| 14:28.58 | *** join/#asterisk KingOfDos (n=irssi@ht1-ix.kingofdos.net) |
| 14:29.15 | ManxPower | n3hxs: I was going to check the sync settings as soon as I eliminated possible callprogress settings. However bpgoldsb is not answering my questions so I'm stuck not being able to help him. |
| 14:29.35 | bpgoldsb | ManxPower: Sorry, I must have missed your question and I stepped away from my desk. |
| 14:29.43 | plundra | I've got two sip-trunks from my provider, which works great for incoming calls (Using two register => ... lines only), but when I add them as peers too, for outbound calls, things go wrong. Using the same name of the peer as the extension in the register-line, I can call in on _one_ of the numbers, on the other I get a username mismatch (the other accounts number/extension i used). Or if I use different names of the peers all together, I get a mismatch on both |
| 14:29.54 | plundra | Have I missed some crucial logical error or whatever? |
| 14:30.08 | bpgoldsb | /etc/asterisk/zapata.conf:busydetect=no /etc/asterisk/zapata.conf:callprogress=no |
| 14:30.13 | plundra | Oh, using insecure=invite makes it all better. Or just using a single account. |
| 14:30.35 | ManxPower | bpgoldsb: put a copy of your /etc/zaptel.conf on pastebin.ca |
| 14:30.37 | *** part/#asterisk stimpie (n=stimpie@213-73-211-48.cable.quicknet.nl) |
| 14:30.44 | plundra | A single account (register + peer) at a time, that is. |
| 14:31.06 | *** join/#asterisk seanmh (i=HydraIRC@216.31.101.31) |
| 14:31.08 | *** join/#asterisk MrNaz (n=mrnaz@ppp118-208-186-83.lns10.mel4.internode.on.net) |
| 14:31.40 | bpgoldsb | ManxPower: http://pastebin.com/m7e98163a |
| 14:32.34 | ManxPower | bpgoldsb: that is /etc/asterisk/zapata.conf I need to see /etc/zaptel.conf |
| 14:33.08 | bpgoldsb | ManxPower: My mistake http://pastebin.com/m3ca54f8c |
| 14:33.34 | ManxPower | bpgoldsb: your sync source looks good. |
| 14:33.56 | ManxPower | bpgoldsb: you're not getting HDLC Aborts on the CLI or the logs are you? |
| 14:34.32 | bpgoldsb | ManxPower: I am from time to time. Some correlate with the dropped calls, some with the red alarms. However, I still get dropped calls without HDLC aborts. |
| 14:35.15 | ManxPower | HDLC Abort indicates 1) Line problems or 2) corrupted data coming from the zaptel card. |
| 14:35.35 | bpgoldsb | ManxPower: Yes, we just replaced the TE110 with a TE120 in hopes of fixing that. |
| 14:35.49 | bpgoldsb | ManxPower: But in this case, I know the problem is with the LEC's T3/DS3 |
| 14:35.53 | ManxPower | You need to get the red alarm issues fixed before you can really so anything else. |
| 14:36.08 | *** join/#asterisk mog (n=mog@nat/digium/x-4534820a0f56f3c7) |
| 14:36.08 | *** mode/#asterisk [+o mog] by ChanServ |
| 14:36.12 | bpgoldsb | ManxPower: Our LEC confirmed our Red Alarms matched with errors on their T3 |
| 14:36.44 | rwaite | hey ManxPower, you said you do networks a lot? |
| 14:36.50 | n3hxs | Do you get more dropped calls when the circiut is loaded rather than when you have just a couple of calls? |
| 14:36.50 | bpgoldsb | But what I'm trying to do it determine when calls are dropping, when there isn't a Red Alarm. |
| 14:37.02 | ManxPower | rwaite: Depends on the network, but yes. |
| 14:37.11 | *** join/#asterisk unasi7 (n=unasi7@84-75-23-200.dclient.hispeed.ch) |
| 14:37.19 | ManxPower | bpgoldsb: you can get errors that will drop calls without getting a red alarm |
| 14:37.36 | rwaite | i am running 'netstat -an' to see the connections for the sip call, but i dont see it. i do see udp connections but only 0.0.0.0 for the addresses? |
| 14:37.43 | bpgoldsb | ManxPower: And I'm trying to determine how to catch those, so I can be alerted of them. |
| 14:37.54 | rwaite | do you know what the best way to 'see' the sip connections (and the rtp too) |
| 14:38.12 | n3hxs | We had a situation where the carrier had a switch that was mis-configured. Error correction would fix the problem until the call load got to the point where it couldn't keep up, then calls would drop. |
| 14:38.20 | unasi7 | tinyquestion: is there a way to have the current sip registry shown in a file (or sql)? |
| 14:39.14 | ManxPower | rwaite: o.o.o.o means "any interface/ip on the system" |
| 14:39.41 | bpgoldsb | n3hxs: It's definately a problem with the upstream equipment, they confirmed that. I'm not trying to diagnose the problem at this point, I'm asking how to determine when there is one. |
| 14:39.53 | rwaite | hmm. but the foreign address has that too. would that mean that the port is open for any ip that connects to it (since these are udp connections) |
| 14:39.56 | ManxPower | bpgoldsb: I have no more suggestions. |
| 14:40.00 | Kobaz | how do i tell asterisk to not send notify's without having first recieved a subscribe to a sip peer |
| 14:40.01 | bpgoldsb | Our Sales department doesn't always report when this happens, so we're trying to figure out when it happens via asterisk |
| 14:40.05 | Kobaz | [Oct 30 10:36:48] WARNING[3747]: chan_sip.c:12892 handle_response: Remote host can't match request NOTIFY to call '4067af922b552a1848fd7c0c22c3afcf@192.168.50.1'. Giving up. |
| 14:40.06 | bpgoldsb | ManxPower: Alright, thanks for trying. |
| 14:40.09 | Kobaz | i keep getting those |
| 14:40.16 | Kobaz | from my audiocodes gateway |
| 14:40.16 | ManxPower | rwaite: put the netstat -an on pastebin.ca |
| 14:40.31 | [TK]D-Fender | Kobaz: sounds like a VM warning |
| 14:40.39 | Kobaz | yeah |
| 14:40.48 | Kobaz | i don't have voicemail even enabled for the thing |
| 14:41.30 | [TK]D-Fender | Kobaz: Then the only reason for * to send it anything is you specifying a VM box in the peer |
| 14:42.09 | *** join/#asterisk sh0tt (n=sh0t@83.19.145.67) |
| 14:43.16 | Kobaz | here's my sip.conf |
| 14:43.27 | *** join/#asterisk Nunners (n=james@mail.nadn.co.uk) |
| 14:43.45 | Kobaz | http://pastebin.ca/1240615 |
| 14:44.04 | Nunners | Can someone suggest any reason why when I load dahdi module, i get red alarms on my two FXO cards? |
| 14:44.13 | rwaite | ManxPower http://pastebin.ca/1240616 |
| 14:44.26 | rwaite | that is with a single sip outbound |
| 14:44.36 | Katty | SWEET! |
| 14:44.41 | Madkiss | hi. |
| 14:44.47 | Katty | i don't have to pay a $154 medical bill |
| 14:44.53 | Madkiss | terminate called after throwing an instance of 'std::length_error' what(): vector::_M_fill_insert |
| 14:44.53 | Katty | happydances |
| 14:44.56 | Madkiss | Aborted |
| 14:45.01 | Madkiss | what kind of error-message is that when starting asterisk? |
| 14:45.06 | Kobaz | [TK]D-Fender: i dont have voicemail set up at all |
| 14:45.12 | Kobaz | [TK]D-Fender: lemme get a sip debug |
| 14:45.50 | [TK]D-Fender | Kobaz: * only sends NOTIFY for VM & presence |
| 14:46.20 | Kobaz | sip debug: http://pastebin.ca/1240617 |
| 14:46.39 | [TK]D-Fender | Katty: This was for that test last week for the headache/cough issue? |
| 14:47.02 | Katty | [TK]D-Fender: no, some lab test from the appendix surgery thing awhile back |
| 14:47.06 | ManxPower | rwaite: the Foreign Address 0.0.0.0:* means "don't care about the source IP or source port. |
| 14:47.17 | Katty | [TK]D-Fender: the doctor visit/medication came up to 50 bucks |
| 14:47.28 | ManxPower | Madkiss: check on #asterisk-dev too? |
| 14:48.04 | [TK]D-Fender | Kobaz: Yuo disabled SIP debug BEFORE the warning came in. SMRT |
| 14:48.34 | ManxPower | rwaite: I can't find a port 5060 in that list. SIP does not appear to be loaded |
| 14:48.53 | Kobaz | [TK]D-Fender: the warning is at the bottom |
| 14:49.00 | Kobaz | oh wait |
| 14:49.02 | Kobaz | whoops |
| 14:49.21 | Kobaz | second time's a charm |
| 14:49.31 | Katty | did i mention i don't have to pay 154 in medical bills? I"M RICH! |
| 14:49.37 | Katty | dances |
| 14:49.48 | ManxPower | Katty: Us people without health insurance hate you. |
| 14:49.55 | Katty | ManxPower: i'm sorry :< |
| 14:49.56 | Kobaz | Katty: i fought for 5 months to get my insurance company to pay $50 on this particular bill |
| 14:50.06 | Katty | Kobaz: :< |
| 14:50.08 | *** join/#asterisk bminish (n=bminish@89.19.93.63) |
| 14:50.11 | Katty | well i had to pay the bill. |
| 14:50.18 | Katty | i just paid it off las tmonth without knowing it |
| 14:50.20 | Kobaz | those pinks |
| 14:50.24 | Kobaz | heh |
| 14:50.26 | Nunners | try £35k tax bill... |
| 14:50.27 | ManxPower | Katty: until recently my medication was almost $250 EVERY MONTH |
| 14:50.29 | Katty | so my balance was 0 |
| 14:50.41 | Katty | ManxPower: gosh. what kind of medication? |
| 14:50.43 | ManxPower | thanks dog for patents expiring |
| 14:51.09 | [TK]D-Fender | rwaite: udp 0 0 0.0.0.0:5060 0.0.0.0:* <- looks like SIP to me... |
| 14:51.11 | ManxPower | Katty: Antidepression/anti-anxiety |
| 14:51.11 | rwaite | ManxPower: what would cause that to happen? |
| 14:51.11 | Katty | 8 dollars a pillow is crazy. |
| 14:51.13 | Kobaz | er |
| 14:51.18 | *** join/#asterisk darkskiez (n=mhb@cpc1-broo3-0-0-cust659.renf.cable.ntl.com) |
| 14:51.21 | Kobaz | demo1*CLI> sip reload |
| 14:51.21 | Kobaz | Previous SIP reload not yet done |
| 14:51.34 | ManxPower | [TK]D-Fender: maybe I need more coffee. |
| 14:51.34 | *** join/#asterisk kb3ien (n=root@isl177-max1.accesshighway.net) |
| 14:51.37 | Katty | ManxPower: there is no reasonf for anti-anxiety pills to cost 8 dollars :/ |
| 14:51.40 | rwaite | yeah, but i am also looking for the rtp - i just didnt understand why i couldnt find my provider's ip in there |
| 14:51.44 | ManxPower | Kobaz: you have DNS issues. |
| 14:51.47 | [TK]D-Fender | ManxPower: Eliminating $250 worth of monthly expenses on anti-depressants would make you feel a lot better. Win-win |
| 14:51.51 | Nunners | Sorry folks - but can someone explain red alarms on two fxo modules? |
| 14:51.52 | rwaite | but i think because its udp is why, stateless? |
| 14:51.53 | ManxPower | Katty: Sure there is. Patents. |
| 14:52.02 | Kobaz | there it goes |
| 14:52.07 | Katty | ManxPower: That's not a legit excuse for charging 8 bucks per pill |
| 14:52.08 | ManxPower | Since it went off patent the generic prices are $85/month |
| 14:52.19 | Katty | that's still stupid. |
| 14:52.23 | n3hxs | Nunners both T1s at the same time? |
| 14:52.27 | Katty | i hate medical stuff. |
| 14:52.28 | ManxPower | rwaite: Are you sure RTP is going thru your server? |
| 14:52.37 | rwaite | i have canreinvite=no set |
| 14:52.39 | Nunners | n3hxs@ Not sure what you mean |
| 14:52.46 | rwaite | so it should? |
| 14:52.56 | Kobaz | allrightey... show me the notify's |
| 14:52.59 | Katty | ManxPower: i feel your pain tho. my appendix surgery was 26 grand. |
| 14:53.08 | n3hxs | <PROTECTED> |
| 14:53.12 | rwaite | but if that 5060 line is right, then i think a lot of those could be rtp. i have the ports from 15000 to 20000 |
| 14:53.21 | Katty | ManxPower: not include the other little stuff. |
| 14:53.31 | rwaite | so i see about ... 4 lines within that range |
| 14:53.32 | Katty | ManxPower: CT scan, original doctor visit, follow visits, getting the staples out. |
| 14:53.46 | vader-- | ok i put that new card in and my asterisk box |
| 14:53.46 | Nunners | n3hxs: I've got a tdm410, two fxo two fxs, and have finally got the setup correct (or at least I thought so) and am now getting these alarms.... |
| 14:53.49 | *** join/#asterisk zydoon (n=zydoon@41.225.140.76) |
| 14:53.56 | *** part/#asterisk zydoon (n=zydoon@41.225.140.76) |
| 14:54.02 | ManxPower | Nunners: do you have lines plugged into the card? |
| 14:54.03 | rwaite | but why would there be four connections for one call. 2 i'd understand for in and out |
| 14:54.03 | Katty | ManxPower: that was just for 30 minutes of surgery |
| 14:54.10 | vader-- | and then i did a make, make install, make config on the zaptel |
| 14:54.11 | n3hxs | Nunners, sorry, jumped in where I don't have experience... |
| 14:54.15 | Nunners | ManxPower: into one of them... |
| 14:54.16 | vader-- | then same for asterisk |
| 14:54.23 | vader-- | now both zaptel cards show unconfigured |
| 14:54.37 | Nunners | Manxpower: the other is currently the other side of the wall and haven't got my drill out yet |
| 14:54.39 | ManxPower | vader--: unconfigured usually means "forgot to run ztcfg" |
| 14:54.52 | Kobaz | [TK]D-Fender: http://pastebin.ca/1240623 there it is |
| 14:55.04 | ManxPower | Nunners: Red alarm on analog means "no line connected" |
| 14:55.13 | vader-- | ZT_SPANCONFIG failed on span 1: Invalid argument (22) |
| 14:55.23 | ManxPower | vader--: well that's why |
| 14:55.29 | Nunners | Manxpower: ok - that would make sense on one, guess the cable might be knackered.... on the other |
| 14:55.45 | sh0tt | hmm.. does anybody use forkcdr() on 1.4.22 ? |
| 14:55.46 | vader-- | span=1,1,0,esf,b8zs |
| 14:55.53 | vader-- | bchan=1-23 |
| 14:56.06 | ManxPower | Nunners: if you have a phone line plugged into an FXS port the port will blow when the phone rings. Make sure you don't have that |
| 14:56.29 | ManxPower | vader--: so you have only one card installed right now? |
| 14:56.31 | Katty | nice. |
| 14:56.49 | *** join/#asterisk jameswf (n=james@dsl093-157-131.phx1.dsl.speakeasy.net) |
| 14:56.58 | Nunners | Manxpower: I don't think so - it's a red card - which is FXO - and that's what the line is plugged into? Am I correct? |
| 14:56.59 | [TK]D-Fender | Kobaz: +/- 460 = VM <---- |
| 14:57.10 | ManxPower | Nunners: Correct. |
| 14:57.19 | Kobaz | [TK]D-Fender: i dont have vm enabled on those peers though... |
| 14:57.24 | Kobaz | [TK]D-Fender: hmm |
| 14:57.36 | vader-- | i have the TDM2400P and a TE122P |
| 14:57.39 | ManxPower | Kobaz: do you have a mailbox= line for those peers? |
| 14:57.43 | tzafrir_laptop | vader--, cat /proc/zaptel/1 and compare |
| 14:57.46 | Kobaz | no, i don't |
| 14:57.50 | ManxPower | vader--: Was that the case yesterday? |
| 14:58.12 | ManxPower | vader--: I suspect the TDM kernel module is loaded before the T-1 kernel module. |
| 14:58.17 | *** join/#asterisk ddunavant (n=David@pool-96-231-70-169.washdc.east.verizon.net) |
| 14:58.19 | Nunners | Manxpower: alarm cleared... bl88dy cable |
| 14:58.29 | ManxPower | channels are set up based on the order the drivers are loaded |
| 14:58.34 | ManxPower | Nunners: Damn, I'm good. |
| 14:58.43 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 14:58.51 | ManxPower | vader--: what distro are you using? |
| 14:59.05 | Nunners | Hooooooray.... it worked...... |
| 14:59.41 | unasi7 | is there a chance to see the current sip-registration in a file (CSV) .. or how can i export / read the current sip-registrations? |
| 15:00.07 | ManxPower | unasi7: It sounds like you need to use Realtime |
| 15:00.27 | ManxPower | unasi7: What are you trying to ACCOMPLISH? |
| 15:00.34 | Kobaz | ManxPower: so umm... why is asterisk sending voicemail notifies... is there a setting i can use to explicitly disable voicemail for the peer |
| 15:00.45 | unasi7 | ManxPower small website with current "online" users ... |
| 15:00.54 | ManxPower | Kobaz: mailbox= causes asterisk to send notifies |
| 15:00.57 | Nunners | Manxpower: Quickly... and again I don't have anything plugged into it yet, but... what is this likely to mean? Unable to specify channel 3: Device or resource busy |
| 15:00.58 | Kobaz | yeah |
| 15:01.00 | Kobaz | but i don't have one |
| 15:01.04 | ManxPower | unasi7: Best of luck. |
| 15:01.14 | ManxPower | Kobaz: then it's not the problem I thought |
| 15:01.27 | [TK]D-Fender | unasi7: parse out "sip show peers" |
| 15:03.54 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 15:04.07 | unasi7 | [TK]D-Fender how can i do that in CMD without asterisk -r? |
| 15:04.20 | Nunners | Manxpower: Quickly... and again I don't have anything plugged into it yet, but... what is this likely to mean? Unable to specify channel 3: Device or resource busy |
| 15:05.02 | [TK]D-Fender | unasi7: AMI |
| 15:05.38 | unasi7 | ^^ .. hmm.. asterisk is a cool software. :) |
| 15:08.36 | *** join/#asterisk putnopvut (n=putnopvu@nat/digium/x-f14a376e434ac6e2) |
| 15:08.36 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 15:09.26 | *** join/#asterisk HeMan (n=jimmy@ssh.southpole.se) |
| 15:09.38 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 15:09.40 | *** join/#asterisk fskrotzki (n=fskrotzk@host198.textwise.com) |
| 15:10.33 | HeMan | Hi! When using queues, how can I make it ring on a phone directly when I hang up if I have calls in the queue? |
| 15:10.56 | creativx | you mean the cleanup time between calls? |
| 15:11.14 | HeMan | yes |
| 15:11.19 | HeMan | ah! found it! |
| 15:11.22 | creativx | ;) |
| 15:11.24 | creativx | gg |
| 15:11.31 | HeMan | wrapuptime or? |
| 15:11.56 | creativx | yes i think that was it |
| 15:13.57 | HeMan | hmm, wrapuptime=0 didn't work |
| 15:15.52 | [TK]D-Fender | HeMan: * will not reprioritize a queue member just because they become available. |
| 15:16.04 | [TK]D-Fender | HeMan: They will have to wait their turn as per your strategy |
| 15:16.24 | HeMan | i have ringall as strategy |
| 15:16.35 | Katty | hugs fskrotzki |
| 15:16.39 | Katty | fskrotzki: what's shakin, bacon? |
| 15:16.50 | lmadsen | HeMan: autofill=yes ? |
| 15:16.59 | Katty | lmadsen: landfill=no |
| 15:17.04 | lmadsen | Katty++ |
| 15:17.20 | *** join/#asterisk bkw_ (n=brian@freeswitch/developer/bkw) |
| 15:17.22 | HeMan | lmadsen: i hade autofill=yess (double s's) |
| 15:17.24 | Katty | lmadsen: did you know if the entire world consumed like the USA, we'd need 6 earths to keep up with the demand? |
| 15:17.28 | HeMan | lmadsen: i'll try again |
| 15:18.08 | lmadsen | Katty: I don't think that makes a whole lot of sense since everything we have we have gotten from this earth... :) |
| 15:18.27 | Katty | lmadsen: sure it makes sense. |
| 15:18.32 | Katty | lmadsen: the USA consumes WAY too much. |
| 15:18.42 | lmadsen | maybe if you need to spread out everything we dig up below the earth over a thin layer on the surface... sure :) |
| 15:18.45 | *** join/#asterisk jtodd (i=gbyjude4@ns2.loligo.com) |
| 15:18.56 | [TK]D-Fender | :(nomNOMnomNOMnomNOMnomNOMnomNOM) |
| 15:19.03 | Katty | lmadsen: USA is pretty small compared to the rest of the world |
| 15:19.13 | lmadsen | Katty: right... but everything it consumes comes from the same earth it is being consumed on..... I don't understand the logic that you need more earths to contain what is pulled from one |
| 15:20.02 | Katty | digs up reddit article |
| 15:20.07 | *** join/#asterisk ChkDigit (n=mike@static24-72-71-175.regina.accesscomm.ca) |
| 15:20.15 | lmadsen | I still think the logic is flawed :) |
| 15:20.30 | lmadsen | not that I dispute the argument that north americans consume too much |
| 15:21.02 | lmadsen | the tipping point will come, and either we'll learn and fix what we're doing, or we will just be yet another collapsed civilization |
| 15:21.14 | lmadsen | has been reading Collapse by Jared Diamond :) |
| 15:21.30 | HeMan | did not work even with autofill=yes |
| 15:21.45 | HeMan | or does it matter if it's in [general] or [myque]? |
| 15:21.48 | lmadsen | HeMan: I don't understand what your issue is.... |
| 15:21.58 | lmadsen | HeMan: does not matter... it is global or per-queue |
| 15:22.08 | lmadsen | asked that question 2 days ago |
| 15:22.50 | HeMan | lmadsen: if I have calls in the queue, I like my phone to start ringing when I hang up an "old" call |
| 15:22.51 | lmadsen | perhaps the device status is slow in updating... check with 'show queues' when the caller hangs up or is connected and check the device state of the queue members to make sure it is switching back quickly |
| 15:23.17 | *** join/#asterisk Firass-z0r (n=asadf@juicebox.vikcomm.wwu.edu) |
| 15:23.31 | *** join/#asterisk scrash08 (n=scrash08@unaffiliated/scrash08) |
| 15:23.33 | lmadsen | HeMan: regardless, because of the way app_queue is designed, there will most probably be some sort of delay. However, check the device state for your queue members with 'show queues' from the cli |
| 15:23.33 | [TK]D-Fender | HeMan: It will not do that. Queue is following your strategy. |
| 15:24.06 | lmadsen | [TK]D-Fender: huh? it should ring all available callers |
| 15:24.16 | lmadsen | s/callers/members/ |
| 15:24.19 | HeMan | [TK]D-Fender: strategy is ringall, should it ring all members? |
| 15:24.25 | lmadsen | yes |
| 15:24.31 | [TK]D-Fender | lmadsen: if the queue is in "wait" mode" is shouldn't preempt that jsut because a member becomes available, should it? |
| 15:24.41 | lmadsen | [TK]D-Fender: what do you mean "wait mode"? |
| 15:24.51 | [TK]D-Fender | lmadsen: In between callout cycles |
| 15:25.28 | lmadsen | [TK]D-Fender: if there are people queued, and someone hangs up then it should distribute the next caller at the next polling interval and ring all available members, even if it is just one |
| 15:25.30 | unasi7 | AJAM: is there a way to load /asterisk/mxml?action=status without logging in first (Cookie!)? |
| 15:25.56 | scrash08 | I've * v1.4.22. In & outbound calling works fine. At, seemingly, random times (usually overnight ...) * simply stops accepting calls. Restarting asterisk makes no difference; only rebooting the box seems to fix the problem -- until it happens again. I've learned how to debug calls; how might I start tracking *this* down? |
| 15:26.14 | [TK]D-Fender | lmadsen: tahts the thing though, he wasn't it IMMEDIATE, not at the next polling interval |
| 15:26.18 | [TK]D-Fender | wants* |
| 15:26.20 | lmadsen | scrash08: actually kinda sounds like a network issue |
| 15:26.39 | Kobaz | scrash08: ip calls or pstn calls |
| 15:26.43 | lmadsen | [TK]D-Fender: polling interval should be pretty quick, but no, it will not ring 5ms after he hangs up |
| 15:27.22 | Katty | lmadsen: "If the rest of the world lived and consumed like the United States there would need to be approximately five planet earths. worth of resources and energy." |
| 15:27.24 | scrash08 | lmadsen: Network on my box, on the 'net, or @ my VSP? Thoughts? |
| 15:27.25 | scrash08 | Kobaz SIP-only setup, SIP trunk to Callcentric ... |
| 15:27.29 | HeMan | i got i working but it's really slow |
| 15:27.33 | Katty | lmadsen: they took down the original article, but REF: http://www.semrau08.com/15.html |
| 15:27.41 | lmadsen | Katty: oh! well that's different :) I thought we were talking about landfill |
| 15:27.46 | Katty | lmadsen: no. |
| 15:27.51 | Kobaz | scrash08: have you tried doing a sip reload instead of rebooting the box |
| 15:28.06 | HeMan | over 15 seconds |
| 15:28.14 | scrash08 | Kobaz: No I have not, but wouldn't a restart of * accomplish the same thing? |
| 15:28.14 | Katty | lmadsen: i knew i was somehow wording it wrong |
| 15:28.19 | Kobaz | scrash08: yeah |
| 15:28.22 | Katty | lmadsen: there were a lot of Fun Facts on that reddit page |
| 15:28.26 | lmadsen | has now exceeded his #asterisk room patience |
| 15:28.31 | lmadsen | Katty: nice! |
| 15:28.35 | scrash08 | Kobaz: Ok, and that didn't help :-/ |
| 15:28.39 | Katty | lmadsen: prices of coca-cola |
| 15:28.41 | Kobaz | scrash08: i thought you said you were rebooting the server |
| 15:28.41 | HeMan | what is the timeout value for a queue? |
| 15:28.47 | Kobaz | scrash08: is the peer reachable? |
| 15:29.13 | lmadsen | HeMan: in queues.conf, timeout is the amount of time to ring a member for before assuming he is not available |
| 15:29.25 | scrash08 | Kobaz "...Restarting asterisk makes no difference; only rebooting the box seems to fix the problem ..." |
| 15:29.25 | scrash08 | Kobaz Yes. *OUT*bound continues to work. |
| 15:29.52 | HeMan | if I have ringall, does the timeout really do anything? |
| 15:30.23 | Kobaz | scrash08: have you done a sip debug |
| 15:30.44 | vader-- | uhggg |
| 15:30.55 | vader-- | the TDM2400P is loading before the TE122P now |
| 15:30.56 | jameswf | well ubuntu 8.10 is officialy out yet I still feel empty.... |
| 15:31.05 | vader-- | dude from digium kinda didn't have any solution |
| 15:31.19 | HeMan | hehe, lowering it to 1 gave me 1 second to answer! |
| 15:31.21 | vader-- | it's in the right order for modeuls |
| 15:31.39 | vader-- | modules he just said sometimes the te122p takes longer and the tdm2400 loads quicker |
| 15:31.39 | [TK]D-Fender | HeMan: "retry" <- |
| 15:31.41 | jameswf | vader blacklist and load with init |
| 15:31.45 | Qwell | vader--: The load order is non-deterministic. |
| 15:31.49 | vader-- | ? |
| 15:31.51 | Qwell | you need to blacklist - like he said |
| 15:32.00 | *** join/#asterisk grantm (n=grant@68.142.138.4) |
| 15:32.03 | vader-- | i am not familiar with that |
| 15:32.10 | jameswf | vader--: hire a consultant |
| 15:32.41 | jameswf | is not for hire :) |
| 15:32.54 | scrash08 | Kobaz: Yes. Nothing outputs on the INbound call. When this happens, callers hear either "Person you are calling is unavailable" OR "Number is out of service". |
| 15:33.10 | *** join/#asterisk asim- (n=sim@gateway1.beatthatquote.com) |
| 15:33.27 | jameswf | echo "blacklist module_name" >> /etc/modprobe.conf |
| 15:33.27 | Qwell | vader--: /etc/modprobe.d/blacklist |
| 15:33.36 | Qwell | eww, modprobe.conf? |
| 15:33.46 | Qwell | does anything use that monstrosity anymore? |
| 15:33.52 | jameswf | Qwell: may not be clean but everyone has one :) |
| 15:33.56 | Qwell | I don't. |
| 15:34.02 | jameswf | elitest |
| 15:34.06 | Qwell | Debian. |
| 15:34.19 | Qwell | ~msg |
| 15:34.20 | jbot | (1) Use private messages to the bots to reduce channel spam, but don't message people on #debian without asking permission first. Most questions should be asked on channel, so that others can benefit from the question and the answers received. (2) Always feel free to message freenode network staff. They're the people with hostnames ending in 'staff.freenode'. (3) Monosodium glutamate, a food additive (see http://truthinlabeling.org/). |
| 15:34.21 | Qwell | vader--: ^^^ |
| 15:34.28 | jameswf | my ubuntu is like a cool version of debian it does.... |
| 15:34.35 | jameswf | legacy stuff |
| 15:34.42 | vader-- | i need the tdm module though |
| 15:34.44 | vader-- | tdm2400 |
| 15:35.15 | jameswf | echo "blacklist module_name" > /etc/modprobe.d/mymodifications |
| 15:35.28 | Qwell | /etc/modprobe.d/blacklist :p |
| 15:35.35 | Qwell | Real distros use that |
| 15:35.58 | jameswf | vader--: blacklist from the kernel's autl load and load through /etc/sysconfig/zaptel and the init script |
| 15:36.06 | jameswf | heck load from rc.local |
| 15:36.10 | *** join/#asterisk marc7 (n=marc@S0106001c1024382d.gv.shawcable.net) |
| 15:37.06 | jameswf | *in dahdi /etc/dahdi/modules |
| 15:37.34 | wacky__ | hello.. |
| 15:38.00 | wacky__ | on two machines where I installed * 1.6, the GSM to uLaw (with the same files as in * 1.4) sounds horribly garbled |
| 15:38.14 | vader-- | there is a /etc/modprobe.d/zaptel |
| 15:38.18 | [TK]D-Fender | ~gsmbug |
| 15:38.19 | jbot | [~gsmbug] there is a bug compiling Asterisk with GCC 4.2 where optimization errors cause GSM transcoding to distort heavily. Compile with GCC 4.1 or lower, or follow this patch : http://bugs.digium.com/view.php?id=11243 |
| 15:38.19 | wacky__ | is this something you guys are aware of ? |
| 15:38.22 | vader-- | it has a bunch of lines |
| 15:38.31 | vader-- | it is missing wcte12xp |
| 15:38.47 | wacky__ | [TK]D-Fender: thanks ! fieew :) |
| 15:39.13 | *** join/#asterisk dmz (n=dmz@64.203.203.232.dyn-cm-pool-64.hargray.net) |
| 15:39.13 | wacky__ | is that patch in any release yet ? |
| 15:39.56 | [TK]D-Fender | wacky__: It should already work, but I'd suggest recompiling.... |
| 15:40.08 | [TK]D-Fender | wacky__: I've heard this a few times here |
| 15:40.26 | wacky__ | [TK]D-Fender: will the patch fix the problem in 1.6 ? or is it already merged ? |
| 15:40.47 | [TK]D-Fender | wacky__: Should be, but again I've heard it creep up even ther |
| 15:41.56 | wacky__ | ok I'll have a look into it |
| 15:42.36 | vader-- | if i create a blacklist file in modprobe.d will that automaticall take effect? |
| 15:42.43 | vader-- | because there isn't a blacklist file |
| 15:44.51 | riddlebox | if you use zap/g1 does g = use the first line first, or last line first? |
| 15:45.15 | Qwell | riddlebox: first. there is also G, which does last |
| 15:45.46 | riddlebox | Qwell, yeah I thought it was that way but wanted to make sure |
| 15:46.33 | ManxPower | vader--: the next thing I was going to tell you before you left was how to specify the load order of the modules |
| 15:46.41 | jameswf | you can also use r or R |
| 15:46.46 | ManxPower | and now I don't have time |
| 15:46.49 | *** join/#asterisk Telemac (n=telemac@213.223.113.74) |
| 15:46.51 | Telemac | Hello |
| 15:47.18 | vader-- | i put them in the right order in /etc/default/zaptel |
| 15:49.01 | vader-- | well the kernel is flipping out |
| 15:49.10 | vader-- | it doesn't like the blacklist |
| 15:49.21 | jameswf | ooohhh you will be able to buy a G1 at walmart,,,, |
| 15:50.35 | *** join/#asterisk _khan (n=shariq@202.133.77.4) |
| 15:51.36 | Qwell | jameswf: somebody mentioned it would be $30 less than t-mobile in-store price |
| 15:51.42 | vader-- | uhhh |
| 15:52.00 | Qwell | jameswf: my wife bought one last week... go get one. today. stop what you're doing and go right now. |
| 15:52.35 | Qwell | (and get me one too...they don't have any here yet) |
| 15:52.36 | Telemac | I'm trying to use dynamic features with asterisk 1.4.19 . I've added an test extension which just Set DYNAMIC_FEATURES and then do a Dial to a SIP peer. When I try *9 for testfeature, either from caller or callee, nothing happens. What am I missing ? |
| 15:52.40 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 15:52.44 | ManxPower | Qwell: My fave headset is the Plantronics M65 or one of those models. Buy it online, pay about $120, buy the EXACT same product, but with the verizon branding from a verizon store $60 |
| 15:52.53 | Qwell | ManxPower: nice |
| 15:53.26 | jameswf | I will probably buy the wife one, she says she will never get to use it, I was like you underestimate my ADD i will probably be bored with it in 10 minutes |
| 15:53.39 | *** join/#asterisk hfb (n=hfb@96.247.65.63) |
| 15:54.00 | Qwell | the only problem I saw with it, is the battery life... |
| 15:54.04 | Qwell | it's *terrible* |
| 15:54.10 | tzafrir_laptop | vader--, put them in the right order in /etc/modules as well, to make sure they are loaded at the correct order at boot time |
| 15:54.11 | jameswf | like a palm |
| 15:54.12 | *** join/#asterisk Firass-z0r (n=asadf@juicebox.vikcomm.wwu.edu) |
| 15:54.21 | Qwell | jameswf: my wife is getting about 24 hours out of it... |
| 15:54.45 | jameswf | my blackberry goes 3 days but i use it more as a laptop than a phone |
| 15:55.19 | wacky__ | hmm.. must I recompile the whole thing with gcc-4.1 ?? |
| 15:55.24 | jameswf | I talk 60min a month I probably push 5-10 gigs of data a month |
| 15:55.30 | *** join/#asterisk CunningPike (n=arodgers@204.239.8.157) |
| 15:55.58 | vader-- | tzafrir do you happen to have the lines that are in your /etc/modules |
| 15:56.31 | Daejeo | ManxPowe: do you have WomanxPower? |
| 15:56.46 | Daejeo | ManxPower: do you have WomanxPower? |
| 15:56.55 | Daejeo | it will ring the bell |
| 15:56.56 | Daejeo | :) |
| 15:57.12 | jameswf | this needs more cowbell |
| 15:57.31 | Daejeo | :) |
| 15:59.08 | jameswf | http://www.morecowbell.dj/ |
| 16:01.21 | *** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee) |
| 16:01.37 | wacky__ | [TK]D-Fender: do you know what must be compiled with gcc-4.1 .. I get some: cc1: error: unrecognized command line option "-fno-strict-overflow" when trying to compile the whole thing (1.6.0.1) with gcc-4.1 |
| 16:01.45 | HeMan | is there any documentation on all options on queues? |
| 16:01.57 | Daejeo | jameswf: it seems that ManxPower is selling remedies for sexual power |
| 16:02.00 | [TK]D-Fender | wacky__: Not sure... |
| 16:02.02 | HeMan | I've looked at voip-info.org but something seem to be missing there |
| 16:02.13 | wacky__ | [TK]D-Fender: but I guess that hit a lot of people, didn't it ?! |
| 16:02.17 | [TK]D-Fender | heLook in the sample configs |
| 16:02.20 | wacky__ | or maybe asterisk 1.6 isn't used that much yet ? |
| 16:02.27 | [TK]D-Fender | wacky__: that too |
| 16:03.54 | wacky__ | hmm.. recompilation with gcc-4.1 didn't solve the problem. |
| 16:03.58 | wacky__ | GSM still garbled |
| 16:04.19 | *** join/#asterisk freakazoid0223 (n=mattc@pool-71-242-218-29.phlapa.east.verizon.net) |
| 16:05.11 | jameswf | wacky__: you have to turn off optimization |
| 16:06.10 | vader-- | ok i think i got it |
| 16:06.19 | jameswf | dude they canceled the Meatloaf concert |
| 16:08.50 | vader-- | but i am getting this error chan_zap.c:2441 pri_find_dchan: No D-channels available! Using Primary channel 24 as D-channel anyway! |
| 16:09.04 | *** join/#asterisk Ritzerisk (n=Ritztech@nv-65-40-156-46.sta.embarqhsd.net) |
| 16:09.11 | ManxPower | vader--: that means "you don't have a d-channel" |
| 16:09.32 | vader-- | i have dchan=24 in zaptel.conf |
| 16:09.35 | ManxPower | If you get that message often, then you should recompile asterisk, zaptel, libpri again. |
| 16:09.46 | ManxPower | vader--: Yes, but there is no D-channel on THE LINE |
| 16:09.47 | wacky__ | jameswf, [TK]D-Fender: this page might be a solution! but hey, who would have known.. : http://forums.digium.com/viewtopic.php?p=116731&sid=3c65e49ec840380ed20f1c8426382b39 |
| 16:09.52 | wacky__ | I'll try it out |
| 16:09.58 | vader-- | weird |
| 16:10.00 | vader-- | the system is working |
| 16:10.05 | vader-- | it's just throwing that error out |
| 16:10.06 | ManxPower | vader--: not at all |
| 16:10.10 | *** join/#asterisk hi365 (n=hi365@bzq-219-141-66.static.bezeqint.net) |
| 16:10.14 | ManxPower | vader--: the system is not working correctly |
| 16:10.20 | Ritzerisk | is it possible to do like a Remote call forward to a set if not at the sip device .... my act was *72 followed by the ext |
| 16:10.20 | hi365 | at what point was devstate included in 1.4? |
| 16:10.27 | vader-- | i am able to make phone calls out and in, only when i run ztcfg i get that error |
| 16:10.33 | tzafrir_laptop | vader--, that error means: "I get no valid traffic on the D channel. Is it actually a D channel? But I'll use it anyway" |
| 16:10.52 | ManxPower | vader--: Yuy are NOT supposed to run ztcfg while asterisk is running. It will drop all calls |
| 16:10.57 | *** join/#asterisk ddunavant (n=David@75.145.240.14) |
| 16:12.17 | vader-- | gotcha |
| 16:12.20 | vader-- | alright seems to be ok |
| 16:12.32 | vader-- | going to let it run and see if the system flakes out |
| 16:12.52 | [TK]D-Fender | hi365: It wasn't. A backport can be separately installed |
| 16:13.23 | ManxPower | hi365: I guess you didn't know that no new features are added to 1.4 after it was released? |
| 16:13.53 | vader-- | thank you guys tremendously for your help |
| 16:14.01 | Qwell | vader--: That'll be $39.95 |
| 16:14.09 | jameswf | +tax |
| 16:14.09 | hi365 | hu? i didnt think the backport was included, but i seem to have it installed and i dont rememeber installing it... |
| 16:14.15 | Qwell | +fees |
| 16:14.20 | ManxPower | You've been around long enough you should know tht. |
| 16:14.43 | hi365 | scrathes his head... |
| 16:14.53 | ManxPower | Unfortunatly they will be adding new features to 1.6 after it is released. |
| 16:14.57 | [TK]D-Fender | hi365: Maybe if you're ona dev branch and not "release" |
| 16:14.58 | HeMan | can the queue only ring again on phones if it times out? |
| 16:15.00 | hi365 | no downloads/tar files here. are you guys sure it was included in 1.4.22? |
| 16:15.14 | Qwell | hi365: re-read what ManxPower said |
| 16:15.27 | HeMan | I get a missed call everytime it times out |
| 16:15.28 | [TK]D-Fender | ~devstate |
| 16:15.29 | jbot | [~devstate] Devstate is an Asterisk 1.4 module for custom BLF device state, see the following link -=- http://svncommunity.digium.com/svn/russell/asterisk-1.4/func_devstate-1.4/ |
| 16:17.38 | hi365 | yup. my func_devstate.so has a different build date |
| 16:17.42 | hi365 | must be getting old young |
| 16:20.05 | [TK]D-Fender | hi365: that rounds off as "prematurely middle-aged" |
| 16:20.20 | hi365 | what were we talking about? |
| 16:20.24 | hi365 | :) |
| 16:21.21 | *** join/#asterisk psy0nid3 (n=IT@69.73.89.233) |
| 16:21.45 | *** join/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 16:21.47 | Telemac | I'm trying to use dynamic features with asterisk 1.4.19 . I've added an test extension which just Set __DYNAMIC_FEATURES and then do a Dial to a SIP peer. When I try *9 for testfeature, either from caller or callee, nothing happens. What am I missing ? |
| 16:22.10 | ManxPower | Telemac: the problem could be caused by many things. |
| 16:24.03 | psy0nid3 | I was watching the command line and saw this: -- Executing [s@macro-delcallback:3] MYSQL("Zap/59-1", "Query r 13 DELETE FROM callers where uniqueid<=1225293459.120317 AND callback=0 AND queuename=12002") in new stack |
| 16:24.21 | Telemac | ManxPower: I know that dtfm pass, as Read get it properly, I've check application I'd like to trigger (Playback for test). I don't see what I can check next |
| 16:24.46 | psy0nid3 | I can not locate where this is being called from, I have looked in all of the .conf and nothing, any ideas? |
| 16:24.55 | psy0nid3 | it seems to happen at random times |
| 16:25.14 | ManxPower | psy0nid3: the line is in extension "s", priority 3 in the "macro-delcallback" macro |
| 16:25.33 | ManxPower | psy0nid3: you're running a GUI aren't you? |
| 16:25.45 | psy0nid3 | correct |
| 16:25.57 | ManxPower | Best of luck with that. |
| 16:26.00 | ManxPower | ~freepbx |
| 16:26.01 | jbot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 16:26.04 | Ritzerisk | which file would Forwarding take place |
| 16:26.08 | Telemac | ManxPower: In fact I just test with testfeature provided as sample in features.conf, between to local SIP peers |
| 16:26.15 | ManxPower | the above applies to most of the GUIs for Asterisk |
| 16:26.28 | ManxPower | Telemac: I have never used features.conf |
| 16:26.53 | Telemac | ManxPower: no way, thanx |
| 16:27.08 | ManxPower | psy0nid3: have you tried the correct place to get support for your GUI? |
| 16:27.14 | psy0nid3 | actually, we found that ont he command line, my apologies |
| 16:27.47 | putnopvut | Telemac: a common problem encountered with features.conf settings is that the featuredigittimeout setting is too short. |
| 16:28.22 | putnopvut | Telemac: by default it is set to 500 ms, meaning you only have half a second allowed between DTMF presses. |
| 16:28.46 | putnopvut | Telemac: Since your feature code requires two DTMF presses, this may be the problem for you too. |
| 16:28.47 | Telemac | putnopvut: oh, I will check that |
| 16:28.51 | lmadsen | putnopvut: ya, I think that should really be set to like 2000 by default |
| 16:28.56 | lmadsen | I've run into that issue a few times |
| 16:29.36 | putnopvut | lmadsen: I agree...perhaps I should just go do that in trunk right now while I'm thinking about it. |
| 16:29.41 | lmadsen | do it! :) |
| 16:29.48 | lmadsen | don't forget CHANGES! |
| 16:30.06 | putnopvut | lmadsen: all righty. |
| 16:30.23 | jeev | i need a shave, who wants to do it |
| 16:30.27 | *** join/#asterisk russellb (n=russellb@asterisk/digium-open-source-team-lead/russellb) |
| 16:30.27 | *** mode/#asterisk [+o russellb] by ChanServ |
| 16:30.51 | Telemac | putnopvut: You were right, great thanx :) |
| 16:30.58 | putnopvut | Telemac: my pleasure |
| 16:31.01 | Ritzerisk | or should i ask what Conf file handles forwarding i have to remotely take the feature off because im not physically at the phone |
| 16:32.35 | file | there is no conf file if it is done device side, like on a SIP device |
| 16:32.48 | file | if you do it server side in the dialplan then that can not be answered without knowing the logic |
| 16:33.53 | *** join/#asterisk bbryant (n=brett@68.208.65.50) |
| 16:35.52 | *** join/#asterisk psy0nid3 (n=IT@69.73.89.233) |
| 16:36.19 | psy0nid3 | ManxPower:sorry got dc'd, I was posing that question regarding delcallback for my boss, we use command line not gui, I just started here. Thank you for your help. |
| 16:36.32 | Ritzerisk | ahhh well im sure that its stored somewhere in a file but cant find it Darn or is there a code for like a remote call forward cancel |
| 16:36.44 | Ritzerisk | or remote call forward setup |
| 16:39.22 | *** part/#asterisk ddunavant (n=David@75.145.240.14) |
| 16:41.40 | *** join/#asterisk axisys (n=axisys@bbgw10.bdcom.net) |
| 16:43.11 | plundra | Can I send a MWI-flag or whatever it's called to a user from the CLI? (Or what is the easiest way of just testing it?) |
| 16:44.45 | ManxPower | Ritzerisk: um, there is no call forwarding code in Asterisk. You have to write it in your dialplan |
| 16:45.39 | ManxPower | Oh! actually, I think there is some silly CF app in 1.6, maybe 1.4. It would store it's data in a base (ast database, get it?) |
| 16:46.02 | [TK]D-Fender | CF in 1.6? totally inappropriate.. this is dialplan stuff |
| 16:46.10 | ManxPower | [TK]D-Fender: I agree. |
| 16:46.57 | ManxPower | [TK]D-Fender: I just checked 1.6, doesn't look like it's in there. |
| 16:47.42 | *** join/#asterisk cesar_CR (n=cesar@200.91.75.66) |
| 16:47.48 | ManxPower | Ritzerisk: I don't consider Asterisk a PBX. PBXs have things like call fowrarding, parking, etc. Asterisk is more of a PBX Toolkit that lets you build those features yourself. |
| 16:48.07 | *** join/#asterisk af_ (n=getsmart@88-149-241-251.dynamic.ngi.it) |
| 16:48.21 | ManxPower | [TK]D-Fender: I think I was thinking of the zap CF stuff, which is in the channel driver |
| 16:49.35 | [TK]D-Fender | ManxPower: Still BS in my books :) |
| 16:49.57 | ManxPower | [TK]D-Fender: I agree. In any case Ritzerisk is smelling a lot like a GUI |
| 16:50.41 | [TK]D-Fender | ManxPower: I see no reason for that conclusion yet... just a little more rope... |
| 16:51.48 | ManxPower | ... is there a code for like a remote call forward cancel...or remote call forward setup... |
| 16:51.55 | ManxPower | stinks like a GUI to me. |
| 16:53.36 | [TK]D-Fender | ManxPower: He's not ont he list currently |
| 16:53.58 | [TK]D-Fender | ManxPower: However his channel listing is plenty indicative |
| 16:54.40 | *** join/#asterisk lionel (n=lionel@ip-185.net-89-3-221.rev.numericable.fr) |
| 16:54.45 | ManxPower | Maybe he realized we toss GUI users into the swamp for gator food? |
| 16:55.31 | *** join/#asterisk talirk81 (i=434e2716@gateway/web/ajax/mibbit.com/x-e65f920eef547e8f) |
| 16:56.21 | talirk81 | Can Background() not play wav files, it docs just say that the file should bespecfied without an extension and asterisk will find the best match. But it cant seem to find wav files |
| 16:56.39 | talirk81 | and when i convert to gsm im getting nasty artifacts and scratchyness |
| 16:56.57 | ManxPower | ~centos52 |
| 16:57.05 | ManxPower | ~centos52bug |
| 16:57.06 | jbot | [~centos52bug] There is a bug compiling Zaptel up to 1.4.11/1.2.26 on CentOS/RHEL 5.2: error in xpp/xdefs.h. Workaround: uncheck "xpp" (Astribank) driver in menuselect. Fixed in SVN. Patches: http://bugs.digium.com/12889, or trouble compiling x86_64 packages? Report a bug to CentOS! ".i386 packages should not satisfy dependencies for .x86_64 packages." |
| 16:57.31 | ManxPower | background can play wav |
| 16:58.28 | *** part/#asterisk nikko (n=nikko@69.57.49.100) |
| 16:58.41 | talirk81 | In the console i see " -- Executing [s@Debt:3] BackGround("SIP/216.235.135.236-0826cbe0", "LevelCall/Debt/Welcome/1") in new stack" |
| 16:58.58 | talirk81 | and in /var/lib/asterisk/sounds/LevelCall/Debt/Welcome/ i have 1.wav |
| 16:59.32 | talirk81 | any ideas why im not hearing sound then |
| 16:59.37 | *** join/#asterisk Maliuta (n=foofbar@kiev.lusan.id.au) |
| 16:59.55 | [TK]D-Fender | talirk81: do you Answer first? |
| 16:59.58 | [TK]D-Fender | ~gsmbug |
| 16:59.59 | jbot | [~gsmbug] there is a bug compiling Asterisk with GCC 4.2 where optimization errors cause GSM transcoding to distort heavily. Compile with GCC 4.1 or lower, or follow this patch : http://bugs.digium.com/view.php?id=11243 |
| 17:00.01 | talirk81 | yes |
| 17:00.09 | talirk81 | if i use .gsm files in the same folder |
| 17:00.12 | talirk81 | it plays the sound |
| 17:00.17 | talirk81 | but if i place .wav it doesnt |
| 17:00.25 | tzafrir_laptop | Isn't that one already fixed with latest versions? |
| 17:01.10 | [TK]D-Fender | talirk81: * only plays wav in 8khz mono 16bit |
| 17:01.28 | talirk81 | ok let me check the hz maybe thats wrong |
| 17:01.57 | *** join/#asterisk harry_v (n=pcsuppor@S0106001d7e52cc78.vs.shawcable.net) |
| 17:02.03 | [TK]D-Fender | no gsmbug is <reply> [~gsmbug] there is a bug compiling Asterisk with GCC 4.2 where optimization errors cause GSM transcoding to distort heavily. Compile with GCC 4.1 or lower, or follow this patch : http://bugs.digium.com/view.php?id=11243 Also please read : http://forums.digium.com/viewtopic.php?p=116731&sid=3c65e49ec840380ed20f1c8426382b39 |
| 17:02.07 | *** join/#asterisk bminish (n=bminish@2001:770:180:0:0:0:0:10) |
| 17:03.49 | tzafrir_laptop | jbot, no gsmbug is <reply> [~gsmbug] there is a bug compiling Asterisk with GCC 4.2 where optimization errors cause GSM transcoding to distort heavily. Fixed in 1.4.20. Compile with GCC 4.1 or lower, or follow this patch : http://bugs.digium.com/view.php?id=11243 |
| 17:03.49 | jbot | okay, tzafrir_laptop |
| 17:05.02 | [TK]D-Fender | tzaSeems applicable in 1.6 as experinced by a few here... |
| 17:05.55 | *** join/#asterisk Firass-VC22 (n=firass@rza.vikcomm.wwu.edu) |
| 17:06.17 | *** join/#asterisk axisys (n=axisys@bbgw10.bdcom.net) |
| 17:07.49 | *** join/#asterisk etech3 (n=chatzill@68-243-103-134.area7.spcsdns.net) |
| 17:09.05 | *** join/#asterisk Nunners (n=james@mail.nadn.co.uk) |
| 17:09.26 | tzafrir_laptop | [TK]D-Fender, you mean that the bug is not fixed? If so, it should be reopened |
| 17:09.35 | Nunners | Does anyone know whether asteriskgui updates the config files, or does it use something else to store the config? |
| 17:09.46 | tzafrir_laptop | conifg files |
| 17:10.18 | Nunners | ok.... in which case it's done something weird with mine... drawing board again! |
| 17:10.36 | Katty | mmm |
| 17:10.37 | Katty | salad. |
| 17:10.50 | Nunners | Do you know if there's anyway to reset the files, and get it to recreate them from scratch? |
| 17:10.51 | Katty | salad and CRACKERS! sesame toasteds crackers, by keebler. |
| 17:10.57 | Katty | most nomable. |
| 17:11.02 | tzafrir_laptop | Nunners, it tends to do bad things to extensions.conf but store most "data" in users.conf |
| 17:11.25 | Nunners | It's doing the following for me: http://pastebin.com/d33238f50 |
| 17:11.33 | tzafrir_laptop | Besides those two files it mostly reads files |
| 17:11.42 | Katty | lmadsen: you're jealous. i know you are. |
| 17:11.49 | *** join/#asterisk dmz (n=dmz@64.203.203.232.dyn-cm-pool-64.hargray.net) |
| 17:11.58 | Katty | lmadsen: here you've been pulling crazy 18 hour shifts. |
| 17:12.03 | Katty | lmadsen: probably without any REAL food. |
| 17:12.05 | lmadsen | Katty: slightly! I am starving! I just showered and shaved... and now... I get to go grocery shopping! |
| 17:12.10 | [TK]D-Fender | Nunners: it is looking in [pstn] for "s" like it should and it simply isn't there... |
| 17:12.11 | *** join/#asterisk gr0mit (n=tim@81.187.32.146) |
| 17:12.16 | Katty | lmadsen: word of advice. |
| 17:12.23 | lmadsen | Katty: I've been living off of veggie wraps from subway for 2 days |
| 17:12.24 | [TK]D-Fender | Nunners: And I highly recommend you not try to start learning * via the GUI... |
| 17:12.24 | Katty | lmadsen: eat before grocery shopping |
| 17:12.32 | lmadsen | Katty: oh I know that rule well :) |
| 17:12.34 | Katty | lmadsen: else you spend $300 on groceries ;) |
| 17:12.41 | Katty | lmadsen: k, happy hunting |
| 17:13.09 | lmadsen | Katty: I'm actually pretty good. I have a list of things I typically buy, plus it all has to fit into a single backpack... so the chances of me spending more than $80 is negligable |
| 17:14.00 | tzafrir_laptop | [TK]D-Fender, actually the asterisk-gui can be useful if you only use it to observe and not to configure |
| 17:14.10 | tzafrir_laptop | Nunners, ==^ |
| 17:14.48 | tzafrir_laptop | useful in exposing the configuration of Asterisk, that is |
| 17:14.50 | Nunners | I think I'm starting to see that - I'm not completely new to * just new to working with hardware.... I had a complete IVR running using IAX & SIP before installing this bloody card! |
| 17:14.56 | Katty | lmadsen: lucky boy you |
| 17:15.04 | Katty | lmadsen: i usually fill up an entire cart. |
| 17:15.07 | Katty | lmadsen: but i shop for two (= |
| 17:15.13 | Katty | 2.75 |
| 17:15.17 | Katty | doggy and 4 ferrets too |
| 17:15.26 | Katty | maybe that counts as 3, i dunno |
| 17:15.27 | lmadsen | heh... well I don't have a car because I live in downtown Toronto :) |
| 17:15.33 | Katty | ah ;) |
| 17:15.38 | Katty | perfect sense |
| 17:15.44 | lmadsen | so everything I buy is schlepped by moi ) |
| 17:15.44 | lmadsen | :) |
| 17:15.45 | Katty | Now go mister shower and shaven! |
| 17:16.09 | lmadsen | does as told |
| 17:16.10 | Katty | the grocery store adventure awaits! |
| 17:16.46 | orkid | why is most of IRC canadian? |
| 17:17.02 | Carlos_PHX | Eh? |
| 17:17.04 | [TK]D-Fender | tzafrir_laptop: because users.conf dos funky stuff and dialplan gets generated live by it and so forth I can't see it as a teaching guide. Too much overassociation goes on... |
| 17:17.04 | putnopvut | orkid: ? |
| 17:17.17 | Katty | orkid: not true. |
| 17:17.20 | [TK]D-Fender | tzafrir_laptop: And the lack of quality dos on it and user.conf compounds it |
| 17:17.27 | Katty | orkid: we're from all over the place. |
| 17:17.33 | tzafrir_laptop | [TK]D-Fender, do you have any system with a asterisk-gui? |
| 17:17.43 | Katty | orkid: i'd wager probably 60% of the channel is usa |
| 17:17.47 | [TK]D-Fender | tzafrir_laptop: I does have potential with the right changes, but the GUI doesn't have a clear direction that I could see |
| 17:17.54 | [TK]D-Fender | docs* |
| 17:17.57 | Katty | orkid: and 10% gay |
| 17:18.06 | Katty | orkid: and in my opinion, more the merrier. |
| 17:18.16 | tzafrir_laptop | if so, edit sip.conf , generate some users (peers, friends, whatever) |
| 17:18.32 | tzafrir_laptop | use #include, #exec, use templates, whatever you want |
| 17:18.36 | Katty | shares sesame crackers with [TK]D-Fender |
| 17:18.50 | [TK]D-Fender | Katty: Thanks, but I'm cutting the carbs... |
| 17:19.03 | Katty | [TK]D-Fender: :< |
| 17:19.05 | Katty | [TK]D-Fender: how will you survive? |
| 17:19.13 | Katty | [TK]D-Fender: on protein and veggies alone?! |
| 17:19.18 | [TK]D-Fender | Katty: Eating smaller creatures :) |
| 17:19.26 | Katty | [TK]D-Fender: you meatasarious you. |
| 17:19.33 | [TK]D-Fender | "Big fish eat the little ones... big fish eat the little ones..." |
| 17:19.49 | [TK]D-Fender | Katty: Vagiterian ;) |
| 17:19.52 | Carlos_PHX | Is hungry for steak now. |
| 17:19.56 | Katty | mmm, steak |
| 17:19.58 | Katty | ny strip |
| 17:19.58 | [TK]D-Fender | is bad... SO bad... |
| 17:20.01 | orkid | maybe u just notice the ones who are from your area |
| 17:20.04 | Katty | [TK]D-Fender: yes. yes you are. |
| 17:20.07 | Katty | [TK]D-Fender: but we still love you. |
| 17:20.20 | Carlos_PHX | Wonders who is "we??" |
| 17:20.21 | orkid | usa is much bigger than canada... so more % or population might be in irc in canada than usa |
| 17:20.30 | orkid | but who knows |
| 17:20.37 | Katty | orkid: to quote the muppets |
| 17:20.41 | Katty | orkid: the question is, who cares? |
| 17:20.48 | Carlos_PHX | Besides, there's nothing else to do in Canada anyway, might as well be on IRC. |
| 17:20.50 | Katty | orkid: ;) |
| 17:20.56 | Katty | there's snow. |
| 17:20.59 | Katty | and [TK]D-Fender |
| 17:21.19 | Katty | can't be all THAT boring. |
| 17:21.29 | Katty | seanmh: i got your email. |
| 17:21.57 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 17:22.08 | Daejeo | Katty: transformed from CAT? |
| 17:22.27 | Katty | Daejeo: sorry, you do not parse. please try again. |
| 17:22.41 | Daejeo | Katty: :) |
| 17:23.10 | Katty | must have gone over my head. |
| 17:23.12 | Katty | jbot: cat? |
| 17:23.12 | jbot | i heard cat is officially used to concatenate files. cat is also used to display the contents of a file on screen. Syntax: cat (file1) (file2) ...(fileN) Where file1 through fileN are the files to display. Example: cat letters/from-mdw displays the file letters/from-mdw. or a clawed walking stomach that meows, or http://www.linux-on-line.net/downloads/fun/humorous-plus/beware_of_dogs.jpg |
| 17:23.32 | Daejeo | ah, my compilerhas some problem then |
| 17:23.45 | Katty | Daejeo: please cat your log file. |
| 17:23.59 | Daejeo | right right |
| 17:24.03 | Katty | Daejeo: a readme might be helpful, if you have one. i find interacting with people who have readmes easier to work with. |
| 17:24.13 | putnopvut | The link from ~cat gives me a 403 |
| 17:24.25 | Katty | boo :< |
| 17:24.31 | Katty | we need a new kitteh. |
| 17:24.39 | Katty | something from the itty bitteh kittey commiteh would be good. |
| 17:27.52 | Daejeo | Katty: what is the mystery behind the "katty" name |
| 17:27.56 | Daejeo | ? |
| 17:28.10 | *** join/#asterisk wiscados (n=mint@81.25.184.155) |
| 17:28.55 | [TK]D-Fender | Daejeo: if she told you... it wouldn't be a mystery now would it? |
| 17:29.06 | *** join/#asterisk TonyM (n=TonyM@softins.claranet.co.uk) |
| 17:29.34 | Daejeo | right right |
| 17:29.57 | Daejeo | I should not be asking this |
| 17:33.40 | Katty | grins |
| 17:33.54 | Katty | i guess i could tell now that he's gone. |
| 17:34.07 | Katty | nah. |
| 17:34.11 | Katty | goes back to paperwork |
| 17:37.20 | *** join/#asterisk propellerhead (n=yogurt2u@host38.190-136-115.telecom.net.ar) |
| 17:38.30 | *** join/#asterisk sah-work (n=Bawbatos@65-119-47-100.dia.static.qwest.net) |
| 17:40.39 | jaytee | hi all, I'm trying to use Background() with WaitExten to enter an extension in another context and jump to that context/extension but when I start dialing it only takes the first digit and then jumps which returns an invalid extension in that context. |
| 17:41.19 | jaytee | Does Background() only work with single digits? |
| 17:42.26 | Katty | hello there mister Jaytee! |
| 17:42.28 | *** join/#asterisk ghostrdr (n=hirhgoih@96.56.103.35) |
| 17:42.30 | [TK]D-Fender | jaytee: No. Pastebint he call & your dialplan. |
| 17:42.36 | jaytee | Hi Katty |
| 17:42.49 | Katty | jaytee: pastebin your dialplan. |
| 17:42.51 | Katty | jaytee: and some cli info |
| 17:43.11 | jaytee | [TK]D-Fender, ok. this is just a rough draft, I'm looking at streamlining it. Just a sec |
| 17:44.15 | kb3ien | ERROR[6566] codec_dahdi.c: Failed to open /dev/dahdi/transcode: No such file or directory what is this supposed to be, and how does it get made? |
| 17:45.12 | festr_ | anyone using asterisk 1.4 voicemail with realtime and mwi? |
| 17:45.14 | *** join/#asterisk CrazyTux (n=brandon@65-60-108-170.static-ip.telepacific.net) |
| 17:45.47 | festr_ | i can subscribe to voicemail, got valid response (NOTIFY) but after leaving new voicemail it does not send NOTIFY to subscriber |
| 17:46.08 | festr_ | i have to restart sip client to get refreshed status. |
| 17:46.11 | jaytee | [TK]D-Fender, here's the dialplan from extenions.conf and a failed call output from the CLI at the bottom. http://pastebin.ca/1240795 |
| 17:48.04 | [TK]D-Fender | jaytee: {recorder] only has ONE extension, and its the one running your IVR |
| 17:48.45 | [TK]D-Fender | jaytee: You seem to have forgotten an INCLUDE in there I'm pretty sure you intended to do. |
| 17:49.02 | [TK]D-Fender | jaytee: And never run IVR's off of numbered extens, always use "s" |
| 17:49.18 | [TK]D-Fender | jaytee: And while we're at it.... don't forget to set your timeouts :) |
| 17:56.26 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 18:03.52 | *** join/#asterisk gr0mit (n=tim@81.187.32.146) |
| 18:05.05 | *** join/#asterisk gr0mit (n=tim@81.187.32.146) |
| 18:06.15 | *** join/#asterisk keulin (n=cray@bne75-5-82-231-224-34.fbx.proxad.net) |
| 18:06.21 | *** join/#asterisk stencil (n=stencil@pdpc/supporter/student/stencil) |
| 18:10.29 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 18:22.59 | *** join/#asterisk tzafrir_laptop (n=tzafrir@local.xorcom.com) |
| 18:23.49 | *** join/#asterisk iratik (n=itariki@209.248.216.146.nw.nuvox.net) |
| 18:24.45 | *** join/#asterisk nicoAMG (i=asgalt@201.203.96.42) |
| 18:24.46 | iratik | Teliax is terrible (their quality is good though).. Can anyone recommend a sip trunking provider that allows you to manage termination and origination indepdently with unlimited channels on both paid on per-minute basis? |
| 18:24.52 | iratik | we've tried so many! |
| 18:26.36 | tzafrir_laptop | kb3ien, if you don't have a dahdi transcoder card you can ignore codec_dahdi |
| 18:27.38 | _khan | i am getting registration timeout error, my asterisk is behind NAT i m registering through remote locations what is the config for sip.conf in this case?? |
| 18:28.24 | [TK]D-Fender | _khan: ... |
| 18:28.26 | [TK]D-Fender | ~sipnat |
| 18:28.27 | 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 |
| 18:28.28 | [TK]D-Fender | ^^^^^ |
| 18:30.19 | *** part/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 18:30.26 | _khan | [TK]D-Fender: thank u, will get back to u if any problem... |
| 18:30.55 | *** join/#asterisk jicksta (n=jicksta@c-24-6-87-187.hsd1.ca.comcast.net) |
| 18:35.42 | *** join/#asterisk ddunavant (n=David@75.145.240.14) |
| 18:37.07 | *** join/#asterisk write_erase (n=Olivier@royale.aixmarseille.com) |
| 18:39.21 | jameswf | anyone doing automotive aplications 5,133 cars : http://arrays.googlecode.com/files/cars.sql |
| 18:42.39 | *** join/#asterisk stoffell (n=stoffell@d51A4D5A5.access.telenet.be) |
| 18:44.44 | Kobaz | hmmm |
| 18:45.04 | Kobaz | is there a thing to "really, i mean it this time... reset the local settings" for polycom phones |
| 18:45.16 | Kobaz | 99% of the time, reset local settings, never clears the local settings |
| 18:45.44 | Kobaz | the only thing that will really clear them, is a format |
| 18:46.01 | [TK]D-Fender | Kobaz: it'll refresh from provisioning server's backup of the manual settings.. make sure thats gone too. |
| 18:46.15 | [TK]D-Fender | Kobaz: Then format it is |
| 18:46.16 | Kobaz | yeah, they are gone |
| 18:46.33 | Kobaz | it doesn't even matter about those files though, it's not even connecting to the tftp |
| 18:46.48 | Kobaz | polycom really frustrates me sometimes |
| 18:47.27 | *** join/#asterisk stimpie (n=stimpie@213-73-211-48.cable.quicknet.nl) |
| 18:47.47 | Kobaz | aastra, despite being a lesser powerful phone, is 100 times easier to provision |
| 18:48.16 | Kobaz | i thought i had polycom provisioning down pat... but moving polycom phones from system to system is a royal pita |
| 18:48.50 | [TK]D-Fender | Kobaz: 100% good. Every time. |
| 18:49.43 | orkid | exxon record profits... again |
| 18:49.56 | Katty | carols. |
| 18:50.53 | Kobaz | [TK]D-Fender: do you ever need to do a reformat to move a polycom from one system to another? |
| 18:51.24 | Kobaz | the problem is, someone else has previously monkeied with the phone and set a bunch of manual settings |
| 18:51.32 | Kobaz | which reset local config is supposed to take care of |
| 18:51.37 | [TK]D-Fender | Kobaz: Almost never happens, but when I do I trash the configs on the orig server, and do a factory reset |
| 18:51.54 | Kobaz | yeah |
| 18:51.56 | *** join/#asterisk Zizou (n=zizou@190.75.194.51) |
| 18:52.40 | [TK]D-Fender | is int he process of backing up to his new office PC o/ |
| 18:53.05 | stoffell | [TK]D-Fender, that's almost always a fun job to do :) |
| 18:54.28 | [TK]D-Fender | stoffell: Warm & fuzzy feeling from doing new format on a substantially better machine. From AMD 3000+ 1Gig, to Intel C2D E8500 3.16 @ 4 gig. |
| 18:54.42 | [TK]D-Fender | stoffell: "comfy" upgrade |
| 18:54.59 | Zizou | hi, am new to asterisk, i have a litle problem related to SIP routing, see, mi asterisk server is behind a NAT, and i finaly could stablish a call with audio in woth ways, with a softphone outside (with a public ip), now im trying to conect to a softphone behind other NAT and i can get the audio in either way) |
| 18:55.02 | jaytee | [TK]D-Fender, I tried setting and include for the promptchoice context, changing every instance of extension 1 to s and setting a TIMEOUT(digit)=6 and Background was still jumping after only 1 digit. I'm gonna use Read instead and make the code tighter by setting filename variables. I tested Read and it works ok. |
| 18:55.04 | stoffell | I completely understand that warm & fuzzy feeling going through you ... :-) |
| 18:55.11 | Katty | doh. |
| 18:55.16 | Katty | i forgot the billy gilman warm and fuzzy track |
| 18:55.26 | Katty | stoffell: way to jog my memory |
| 18:55.57 | stoffell | lol |
| 18:56.14 | jaytee | [TK]D-Fender, wow, nice machine! |
| 18:56.15 | Katty | there. |
| 18:56.32 | Katty | my Drive Ryan Insane with Christmas Music CD is complete. |
| 18:56.41 | *** join/#asterisk simonr (n=simonr@209.183.22.220) |
| 18:56.43 | jaytee | lol |
| 18:56.53 | Zizou | is like this LocalSoftphone --> Asterisk -->> NAT <--- NAT <--> RemoteSoftphone |
| 18:57.23 | Kobaz | double nat? |
| 18:57.42 | rob0 | [TK]D-Fender, congrats, sounds nice |
| 18:57.51 | Katty | i see that as Softwarephone -> asterisk -> firewall -> internet -> firewall -> softwarephone |
| 18:57.56 | stoffell | hm, in 1.4 with zaptel there was a "make b410p" option to build mISDN, any idea where that is in dahdi and 1.4.22 ? |
| 18:57.57 | Zizou | Kobaz, the remote phone is behind a nat, and the asterisk server and de local soft are behin my nat |
| 18:58.03 | Katty | of course i could be wrong |
| 18:58.12 | Kobaz | Zizou: have fun with that |
| 18:58.34 | Katty | Zizou: was my above description correct? |
| 18:58.47 | Kobaz | Zizou: unless you do port forwarding, there really isn't any good way to have communication between devices that are both behind nat on opposite ends |
| 18:58.48 | Zizou | Katty, yeah you got it |
| 18:58.52 | Katty | Zizou: it's easy. |
| 18:59.14 | Katty | Zizou: on the asterisk firewall side you port forward udp 5060 and 10000 to 20000 udp rtp to the ip of the asterisk server |
| 18:59.25 | Katty | Zizou: on the other nat side, you allow incoming/outgoing traffice on the same ports |
| 18:59.35 | [TK]D-Fender | ~sipnat |
| 18:59.35 | 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 |
| 18:59.37 | [TK]D-Fender | ^^^^ |
| 18:59.51 | Katty | Zizou: there's aslo some settings on the asterisk server to set, see [TK]D-Fender's link above for those particulars |
| 18:59.54 | Katty | Zizou: it WILL work. |
| 18:59.55 | [TK]D-Fender | Why do people waste time retyping it by hand all the time when there is a botlet for it? |
| 18:59.56 | Katty | Zizou: i do it |
| 18:59.58 | Zizou | Katty, Kobaz this is working well right now: softphone --> asterisk --firewall --- sophtphone (public ip) |
| 19:00.12 | stoffell | answers his own question by reading dahdi README .. *blush* |
| 19:00.24 | Katty | [TK]D-Fender: i like to feel useful. |
| 19:00.37 | Kobaz | Zizou: yeap, that looks like it would be easy |
| 19:00.38 | Katty | [TK]D-Fender: and to freely give information :P |
| 19:00.41 | Katty | [TK]D-Fender: repeatedly :P |
| 19:00.57 | Zizou | Katty, ok, so i think the only things that im missing is the forwarding in the remote router no? |
| 19:00.58 | *** join/#asterisk tobias (n=tobias@user-0c8hj2l.cable.mindspring.com) |
| 19:01.12 | Katty | Zizou: no idea. but the remote router will need those ports open |
| 19:01.18 | [TK]D-Fender | echo-cancels Katty |
| 19:01.27 | Katty | turns dead silent |
| 19:01.30 | Katty | [TK]D-Fender: :< |
| 19:01.37 | [TK]D-Fender | Zizou: Remote phones behind their own NAT need NO forwarding |
| 19:01.38 | jeev | sometimes i sneeze and my heart hurts, wtf |
| 19:01.43 | *** join/#asterisk tobias (n=tobias@user-0c8hj2l.cable.mindspring.com) |
| 19:01.46 | Katty | jeev: oh, you too eh? |
| 19:01.50 | Zizou | Katty, should i forwad the same ports? 5060 and 10000 to 20000 to de ip where de remote softphone is? |
| 19:02.00 | jeev | it's rare but has happened twice this past week |
| 19:02.05 | Katty | Zizou: no |
| 19:02.07 | Katty | Zizou: just open the ports |
| 19:02.57 | Zizou | Katty, some times i can talk without problem, with te current conf..but i cant most of the time |
| 19:03.21 | Katty | jeev: have you been coughing a lot? |
| 19:03.29 | Katty | Zizou: i've had that problem before too |
| 19:03.31 | Zizou | Katty, with a public ip soofphone i always can |
| 19:03.34 | Katty | Zizou: check the log of your firewall |
| 19:03.38 | *** join/#asterisk voxter (n=voxter@S01060016b6b53c0c.vc.shawcable.net) |
| 19:03.40 | Katty | Zizou: you might see some policy violations |
| 19:04.14 | scooby2 | how would one check an agents status via AGI? |
| 19:04.25 | jeev | no Katty |
| 19:04.28 | [TK]D-Fender | Zizou: Your * side needs forwarding, your remote phone side does not. |
| 19:04.43 | [TK]D-Fender | scooby2: AGI has nothing to do with this. |
| 19:04.52 | *** join/#asterisk StephenF[W] (n=none@198.144.201.106) |
| 19:04.59 | Zizou | [TK]D-Fender, well thats my current conf |
| 19:05.11 | stoffell | mISDN segfaults when unloading modules ( http://pastebin.ca/1240860 ) on kernel 2.6.24-etchnhalf, any idea on how to workaround except rebooting ? (as I do now) |
| 19:05.16 | *** join/#asterisk watchy (n=watchy@adsl-69-152-41-251.dsl.ltrkar.swbell.net) |
| 19:05.22 | [TK]D-Fender | Zizou: Go follow the guide you were linked. If you still have problems, pastebin your sip.conf masking only passwords |
| 19:05.23 | [TK]D-Fender | ~pb |
| 19:05.24 | 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 |
| 19:05.25 | [TK]D-Fender | ^^^^^ |
| 19:05.26 | watchy | anyone know of a good atx case for a phone system? |
| 19:05.39 | Katty | jeev: i'd wager its irritating a nerve, or your bronchial tubes. |
| 19:05.47 | scooby2 | [TK]D-Fender: ok, then is there a way to check the status for agents in a queue? |
| 19:05.48 | Katty | jeev: the sneezing. |
| 19:05.54 | Katty | jeev: but go see a doctor. |
| 19:06.05 | [TK]D-Fender | scooby2: parse "show queues" or use AMI |
| 19:06.08 | Katty | jeev: that can also be a sign of a collapsing lung |
| 19:06.44 | Zizou | Katty, [TK]D-Fender other thing, is there any script or something that allow me refresh the externip parameter? apparently it cant resolv names |
| 19:06.52 | scooby2 | [TK]D-Fender: thx |
| 19:07.05 | [TK]D-Fender | Zizou: "extenhost" + "externrefresh" are for names. |
| 19:07.13 | [TK]D-Fender | Zizou: "externhost" + "externrefresh" are for names. |
| 19:07.28 | Katty | (= |
| 19:07.36 | Zizou | [TK]D-Fender, so should i use those insted externip? |
| 19:07.43 | jeev | how is my lung collapsing, other than lack of exercise lol |
| 19:07.47 | Zizou | instead* |
| 19:08.17 | [TK]D-Fender | Zizou: Yes |
| 19:08.19 | Katty | jeev: usually it's due to a puncture |
| 19:08.25 | Zizou | [TK]D-Fender, ok, thanks |
| 19:08.46 | watchy | anyone recommend a good wallmount atx case? |
| 19:09.11 | Katty | jeev: are you having a hard time catching your breath? |
| 19:10.08 | watchy | my friends had a collaped lung like 8 times |
| 19:10.10 | watchy | it just happens |
| 19:10.51 | Katty | jeev: i highly doubt it is that serious--unless you're having a hard time catching your breath from little things, like walking to your car. |
| 19:11.10 | Katty | jeev: odds are the force is hitting a nerve, your broncial tubes are inflamed, or you have 'air' pockets floating around. |
| 19:11.19 | Katty | jeev: air pockets are the reason it hurts to breathe sometimes. |
| 19:11.38 | Katty | jeev: what 'feels' like your heart, anyway |
| 19:11.48 | watchy | dr katty |
| 19:12.00 | Katty | i'm a hypochondriac |
| 19:12.02 | Katty | i know these things |
| 19:12.11 | Katty | it keeps me sane when i panic :< |
| 19:12.17 | lmadsen | has returned.... with food! |
| 19:12.23 | [TK]D-Fender | puts his jeev effigy back in a safe place... |
| 19:12.39 | scooby2 | necrophiliac? |
| 19:12.46 | watchy | hey tk what kinda case do you use for phone systems? |
| 19:12.58 | [TK]D-Fender | scooby2: The irresistable urge to crack open a cold one ;) |
| 19:13.06 | [TK]D-Fender | watchy: Rackmount |
| 19:13.16 | [TK]D-Fender | grabs a beer |
| 19:13.18 | watchy | you mount it to the wall? |
| 19:13.20 | watchy | or what |
| 19:13.21 | scooby2 | lol |
| 19:13.28 | lmadsen | prefers open air concept and places the MB on a static bag on the table |
| 19:13.29 | [TK]D-Fender | watchy: no, in a RACK |
| 19:13.33 | watchy | ah |
| 19:13.39 | watchy | we wanna mount the box to a wall |
| 19:14.01 | [TK]D-Fender | lmadsen: one of my best friends just mounted a CP minus a case to a bord on his wall. |
| 19:14.08 | [TK]D-Fender | lmadsen: I should take a pic for you.. |
| 19:14.13 | scooby2 | double sided tape can do wonders |
| 19:14.20 | lmadsen | [TK]D-Fender: sure :) |
| 19:14.23 | [TK]D-Fender | scooby2: No, its well screwed in. |
| 19:14.35 | [TK]D-Fender | scooby2: looks awesome. |
| 19:14.46 | jeev | no i dont have a hard time catching my breathe |
| 19:15.08 | watchy | goto the doc? |
| 19:15.23 | jeev | no dood, it's just the second time this week where i sneezed and got a little headache\ |
| 19:15.35 | Zizou | [TK]D-Fender, which kind of nat parameter should i use for my sip clients, right now i have it set nat=route |
| 19:16.06 | [TK]D-Fender | Zizou: for remote, almost always "nat=yes" |
| 19:16.13 | [TK]D-Fender | Zizou: Its in the guide. Read it |
| 19:16.23 | rob0 | Sneeze+headache sounds like maybe high blood pressure. Check your BP lately? |
| 19:16.29 | Carlos_PHX | Jeev, could be worse: http://www.drdaveanddee.com/headache.html |
| 19:16.50 | [TK]D-Fender | grabs his jeev effigy again and his MCI pin and gives it another good jab |
| 19:17.07 | Zizou | [TK]D-Fender, what guide you mean? im reading the oÅeallys book |
| 19:17.16 | [TK]D-Fender | ~sipnat |
| 19:17.16 | 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 |
| 19:17.20 | [TK]D-Fender | ^^^ |
| 19:17.24 | iratik | did you used to work at MCI fender? |
| 19:17.43 | Zizou | [TK]D-Fender, thanks |
| 19:17.45 | [TK]D-Fender | iratik: No. |
| 19:17.54 | iratik | j/c .... MCI pin |
| 19:18.01 | scooby2 | he worked at Fender silly |
| 19:18.17 | iratik | I wonder sometimes ... who this guy is |
| 19:18.20 | scooby2 | or likes Fender |
| 19:18.22 | iratik | dr. asterisk |
| 19:18.30 | [TK]D-Fender | Nope... has nothing to do with music... |
| 19:18.33 | scooby2 | I work at a place that sells Fender |
| 19:18.40 | iratik | I have a fender |
| 19:18.45 | [TK]D-Fender | though I have played guitar for almost 20 years. |
| 19:18.57 | [TK]D-Fender | and I play on Dean & Ibanez, never Fender |
| 19:19.04 | scooby2 | your counterstrike name? |
| 19:19.25 | [TK]D-Fender | scooby2: Action:Half-Life |
| 19:19.27 | *** join/#asterisk harry_v (n=pcsuppor@S0106001d7e52cc78.vs.shawcable.net) |
| 19:19.31 | [TK]D-Fender | scooby2: MANY years ago. |
| 19:19.44 | [TK]D-Fender | scooby2: Nostalgia isn't what it used to be! |
| 19:20.52 | scooby2 | Valve has billboards all over Chicago for their new game. Of course I cannot remember what its called. |
| 19:21.13 | scooby2 | action:half-life was awesome. Those were the days. |
| 19:21.29 | *** join/#asterisk ViKing78 (n=ViKing78@cerberus.franklinamerican.com) |
| 19:22.36 | LeddyHM | Looking for some ideas. We just upgraded to 1.4 form 1.2 and whenever a user calls in to the main number and then dials an extension the caller never hears a "ring ring". However when they dial the direct number you get it |
| 19:22.36 | [TK]D-Fender | scooby2: Not sure, but they should get Farrah Faucett to do ads for them. Sales would come pouring in! |
| 19:22.36 | [TK]D-Fender | turns off his pun-generator |
| 19:22.36 | Carlos_PHX | I have four fenders. |
| 19:22.36 | Carlos_PHX | On my truck. |
| 19:22.44 | LeddyHM | or when you dial from extension to extension (internally) you hear it |
| 19:24.35 | jeev | fender, my effigy of you hangs in my bathroom.. when i run out of TP, i use it ;) |
| 19:25.54 | kb3ien | anyone up on the polycom SoundPoint IP 650, i'm trying to wrap my head arround the boot options. |
| 19:27.43 | kb3ien | in an ideal world i would not "Cannot contact the boot server" |
| 19:28.09 | LeddyHM | we brought over sip, extensions, voicemail configs and used the default for the rest |
| 19:28.35 | jeev | is Autumn Reeser chick is pretty cute from Red Alert 3 |
| 19:28.37 | jeev | what a stupid name |
| 19:28.53 | *** join/#asterisk theHub (n=theHub@69.177.93.21) |
| 19:30.00 | *** join/#asterisk grEvenX (n=even@cB78D5AC1.dhcp.bluecom.no) |
| 19:36.17 | *** join/#asterisk jasonwoot (n=jasonrot@69.73.89.233) |
| 19:37.52 | jasonwoot | assclown of the day: Entuitive Voice. Here's what happens when you pay them for asterisk support and ask a question: " This is proprietary information that I am not going to elaborate on per company policy. |
| 19:37.52 | jasonwoot | Thanks, |
| 19:37.59 | *** join/#asterisk protocols (n=protocol@ip-88-153-205-180.unitymediagroup.de) |
| 19:38.01 | psy0nid3 | agreed |
| 19:38.48 | *** join/#asterisk Greek-Boy (n=email@41.222.89.114) |
| 19:39.27 | psy0nid3 | we were attempting to locate where this is being called from Executing [s@macro-delcallback:3] MYSQL("Zap/59-1", "Query r 13 DELETE FROM callers where uniqueid<=1225293459.120317 AND callback=0 AND queuename=12002") in new stackExecuting [s@macro-delcallback:3] MYSQL("Zap/59-1", "Query r 13 DELETE FROM callers where uniqueid<=1225293459.120317 AND callback=0 AND queuename=12002") in new stack |
| 19:39.29 | *** join/#asterisk hi365_m (n=hi365@213.151.36.38) |
| 19:39.41 | psy0nid3 | and that is the response we get from intuitive |
| 19:42.17 | Katty | joy. |
| 19:42.26 | Katty | i get to go talk a walk with the call center manager |
| 19:42.41 | psy0nid3 | that is always fun |
| 19:42.55 | psy0nid3 | ¬_¬; |
| 19:42.56 | Katty | i get to explain to her why i'm annoyed. |
| 19:43.02 | Katty | so she can be whiny and defensive |
| 19:43.07 | Katty | and tell the rest of the company what i say |
| 19:43.29 | psy0nid3 | our is a very "controlling" person |
| 19:43.40 | psy0nid3 | and they do not like change |
| 19:43.44 | kb3ien | BootSrv Opt: confuses me. where is this defined.? |
| 19:43.56 | Katty | psy0nid3: well this one is used to getting her way |
| 19:43.59 | Katty | psy0nid3: and she's whiny |
| 19:44.03 | Katty | psy0nid3: extremely whiny |
| 19:44.10 | Katty | psy0nid3: this place is like bloody high school |
| 19:44.23 | psy0nid3 | Katty: yes! our is too |
| 19:44.24 | psy0nid3 | !! |
| 19:44.50 | psy0nid3 | Katty: maybe it is all call centers |
| 19:45.05 | Katty | psy0nid3: well this is a small company of 30 |
| 19:45.06 | Carlos_PHX | If you own the Asterisk server, you can own her... |
| 19:45.09 | psy0nid3 | Katty: this is the 3rd one I have worked in and it is the same story different place |
| 19:45.10 | Carlos_PHX | Just sayin' |
| 19:45.13 | Katty | psy0nid3: there are only two people in that call center |
| 19:45.15 | Katty | psy0nid3: she manages them. |
| 19:45.23 | Katty | psy0nid3: she's also the acting sales rep for the phone systems |
| 19:45.27 | Katty | psy0nid3: so we have to get along |
| 19:45.43 | *** join/#asterisk sah-work (n=Bawbatos@140.221.239.249) |
| 19:45.46 | carrar | Anyone know how to get a PLUS symbole in the INVITE like "INVITE sip:+15553331212@1.1.1.1 SIP/2.0." |
| 19:45.49 | psy0nid3 | Katty: ah I understand, we are much larger |
| 19:46.00 | Katty | psy0nid3: lucky you |
| 19:47.19 | psy0nid3 | Katty: lol, sometime I guess. More whiny people, especially the sales reps |
| 19:49.04 | jameswf | I wonder if i should go to the ubuntu release party tonight.... open bar... |
| 19:49.13 | kaldemar | carrar: you have a nice button in your keyboard for it. just put a + in your dialplan. |
| 19:49.42 | carrar | That doesn't translate to the INVITE in the SIP headers |
| 19:49.52 | kaldemar | yes it does. |
| 19:49.58 | carrar | using what? |
| 19:50.07 | [TK]D-Fender | carrar: Show us your dial in CLI, and the SIP debug of the call |
| 19:50.12 | kaldemar | Dial(SIP/+12345@123.123.123.123) |
| 19:50.28 | carrar | I'll try that, I thought I tried that already |
| 19:50.30 | kaldemar | that definitely puts a + in there. |
| 19:51.37 | carrar | ok you're right, I'm on crack |
| 19:51.39 | carrar | nm |
| 19:51.49 | carrar | not sure wtf I was thinking |
| 19:54.49 | *** join/#asterisk lanning (n=lanning@66.151.128.195) |
| 19:54.50 | *** join/#asterisk Ariel_Calzada (n=aricalso@dsl-emcali-200.29.106.116.emcali.net.co) |
| 19:56.29 | *** join/#asterisk superpop02 (n=mozveren@se167-1-82-242-148-65.fbx.proxad.net) |
| 19:56.34 | superpop02 | hello all |
| 19:57.13 | superpop02 | question about manager api |
| 19:57.33 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 19:57.46 | superpop02 | I dont know how differenciate a incoming or a outcoming call for a device |
| 19:58.12 | superpop02 | for exemple :Event: Newstate |
| 19:58.13 | superpop02 | Privilege: call,all |
| 19:58.13 | superpop02 | Channel: SIP/sjphone100-09c198e8 |
| 19:58.13 | superpop02 | State: Up |
| 19:58.13 | superpop02 | CallerID: 100 |
| 19:58.13 | superpop02 | CallerIDName: <unknown> |
| 19:58.15 | superpop02 | Uniqueid: 1223475723.62 |
| 19:58.58 | superpop02 | how I can say its a incoming call for the device sjphone100 ? |
| 19:59.07 | superpop02 | or a outcoming ? |
| 19:59.35 | superpop02 | I am confusing about event reporting |
| 20:00.24 | *** part/#asterisk stencil (n=stencil@pdpc/supporter/student/stencil) |
| 20:00.27 | *** join/#asterisk aksyn (n=aksyn@gw.na.nu) |
| 20:01.00 | superpop02 | for example if I make a originate call to asterisk |
| 20:02.17 | *** join/#asterisk tobias (n=tobias@user-0ce2hvs.cable.mindspring.com) |
| 20:02.17 | superpop02 | how to know the caller and the callee for channel ? |
| 20:07.00 | *** join/#asterisk rwaite (n=richard@rrcs-74-218-125-86.central.biz.rr.com) |
| 20:07.08 | rwaite | ah!! this echo is killing me! now only local sip->sip calls have echo |
| 20:08.12 | *** join/#asterisk etech3 (n=chatzill@68-243-103-134.area7.spcsdns.net) |
| 20:09.23 | *** join/#asterisk newmember (n=chatzill@static-66-11-81-77.ptr.terago.net) |
| 20:09.28 | *** join/#asterisk exvito (n=exvito@195.245.132.93) |
| 20:10.10 | HeMan | I just listened to some of the sounds to asterisk, who decides what sounds to record? |
| 20:10.26 | *** join/#asterisk beek (n=klinebl@65.211.106.242) |
| 20:10.28 | HeMan | when is for example tt-monkeys used? |
| 20:10.40 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 20:12.30 | Carlos_PHX | You should play tt-monkeys before every auto-attendant, just to set the mood. |
| 20:12.32 | exvito | hi all... problem at hand: 2 systems, iax interconnected... A dials B, dialplan in B calls wait(2) + hangup(cc).. A always gets hangupcause=16, no matter what cc i use in B's dialplan... ideas ? |
| 20:12.57 | HeMan | Carlos_PHX: but of course! |
| 20:13.22 | Carlos_PHX | They're just fun, do with them as you will. We use tt-monkeys for a test, so now monkeys=the sound of success. |
| 20:13.24 | HeMan | and what-are-you-wearing just anytime? |
| 20:13.40 | Carlos_PHX | We also actually put "to hear monkeys scream dial 8" on our AAX and all our customers. |
| 20:13.44 | Carlos_PHX | Most love it. |
| 20:13.48 | HeMan | hehe |
| 20:13.48 | Carlos_PHX | About half have it removed. |
| 20:14.22 | Carlos_PHX | You don't know how many people have called and asked about the monkeys, it's a good ice breaker for new calls. |
| 20:14.41 | HeMan | my boss told me to remove we-dont-have-tech-support from our company asterisk... |
| 20:15.57 | *** join/#asterisk gr0mit (n=tim@router0.txrx.org.uk) |
| 20:18.08 | exvito | anyone ? |
| 20:18.09 | exvito | :) |
| 20:20.12 | *** part/#asterisk exvito (n=exvito@195.245.132.93) |
| 20:20.48 | Carlos_PHX | A customer just asked me for IVR functionality to do surveys. Anyone recommend a product that does this and lets an end user enter the questions/answers? Rather than me programming each one as a dialplan...ugh... |
| 20:21.20 | superpop02 | carlos, you can use switchvox ... |
| 20:21.59 | superpop02 | but you need tts plugin |
| 20:22.25 | jaytee | anyone else in here using LumenVox? |
| 20:22.50 | superpop02 | Is lumenvox a good product ? |
| 20:23.14 | *** join/#asterisk Telemac (n=cchantep@ANantes-157-1-143-9.w90-25.abo.wanadoo.fr) |
| 20:23.34 | Carlos_PHX | Do you know if Switchvox free works for this? Is it end-user-friendly? |
| 20:23.55 | Carlos_PHX | The questions would be recorded, so could do without TTS, if it allows that. |
| 20:24.00 | superpop02 | I think its not enough user friendly ... |
| 20:24.25 | superpop02 | I am working to create a mashup technology for asterisk |
| 20:24.28 | Carlos_PHX | They want to replace an analog IVR which the users programs. |
| 20:24.32 | Zizou | [TK]D-Fender, the link that you send me work perfect, thnks again |
| 20:24.42 | [TK]D-Fender | Zizou: You're welcome |
| 20:24.45 | jaytee | superpop02, depends I guess on your point of view. I'm having an issue right now with GotoIf where if I use DTMF and press any key 0-9 my speech score is 1000 but I've set my Threshold to 720 and if it's lower it's supposed to loop and higher it's supposed to proceed in the dialplan. Speech scores 720 and above work fine but DTMF at 1000 evaluates as false when it should evaluate true. |
| 20:25.32 | ViKing78 | Is anybody using jabber to set presence in call queues? I want to defer calls for agents either away or offline in their jabber client. |
| 20:26.02 | jaytee | [TK]D-Fender, what ya loading for an OS on your new puter? |
| 20:26.31 | ViKing78 | I saw on voip-info.org that the cmd jabberstatus might be able to do what I want but just looking for suggestions |
| 20:26.37 | ViKing78 | http://www.voip-info.org/wiki/view/Asterisk+Jabber |
| 20:27.37 | [TK]D-Fender | jaytee: Its an office PC so I partitioned 60 out of 80 gig for WinXP Pro |
| 20:27.53 | [TK]D-Fender | jaytee: I jsut got Ubuntu 8.10 and am thinking of virtualizing it. |
| 20:28.11 | [TK]D-Fender | jaytee: I also have VirtualBox under Windows for that level if I care. |
| 20:28.26 | [TK]D-Fender | jaytee: I ahve a lot to learn about this stuff |
| 20:28.43 | [TK]D-Fender | but... thats later... checkout time. Back in a bit. |
| 20:33.00 | kb3ien | how do i go about setting up dahdi (dummy) for the first time? |
| 20:33.23 | rwaite | virtualbox pwns |
| 20:33.35 | rwaite | kb3ien 'README' ;) |
| 20:35.58 | kb3ien | hm, have been READIN'EM for a while now. |
| 20:37.27 | kb3ien | i've built everything, but there is no /dev/dahdi and no file to make it. |
| 20:39.07 | Telemac | I'm trying asterisk features.conf . It seems that when DYNAMIC_FEATURES is set, featuremap is no used but only applicationmap, so for exemple blindxfer cannot be trigger with '#'. Am I wrong ? Is there any way to reactivate blindxfer (and so on) when DYNAMIC_FEATURES is set ? |
| 20:39.27 | Carlos_PHX | Huh, one company advertises an "interactive" IVR. I wonder how the non-interactive IVR systems work? |
| 20:40.23 | kaldemar | interactive interactive voice response et etc. |
| 20:41.38 | *** join/#asterisk Linuturk (n=Linuturk@fluxbuntu/developer/Linuturk) |
| 20:43.36 | *** join/#asterisk write_erase (n=Olivier@royale.aixmarseille.com) |
| 20:43.39 | *** join/#asterisk kfife (n=Miranda@home.chicagoventure.com) |
| 20:47.15 | jameswf | That would be a UVR |
| 20:47.43 | jameswf | like a network interface card card |
| 20:49.03 | jjshoe | Carlos_PHX what company? |
| 20:49.27 | jjshoe | Carlos_PHX I don't think it's a rediculous claim, how many people that arn't in the telephony industry knows what ivr stands for? |
| 20:49.43 | *** join/#asterisk gr0mit (n=tim@router0.txrx.org.uk) |
| 20:50.05 | Carlos_PHX | I closed the window, don't recall what company. |
| 20:50.23 | Carlos_PHX | That and other typos on the main page tells me I probably don't want to talk to them. |
| 20:50.52 | Carlos_PHX | Reminds self never to bring the whole bag of sugar-free cookies to my desk, resulting in absent-minded gorging. |
| 20:51.16 | kfife | Does anyone know of a way to set callerid(num) when using 'originate' from the CLI? In this case callerid is dyanmic, so I don't want to have to 'set' it in sip.conf, neither resort to .call files. Any ideas would be much appreciated!! I have a suspicion I could override the value for a given sip.conf context from the CLI, and THEN place the call? Am I on the right track? |
| 20:51.46 | jjshoe | kfife dunno, you lost me in your question. |
| 20:52.13 | jjshoe | kfife you want to set it dynamically without ever passing the caller-id in? |
| 20:52.28 | *** join/#asterisk jer (n=jer@unaffiliated/jer) |
| 20:52.50 | kfife | jjshoe: It's null unless specified. |
| 20:52.54 | kaldemar | kfife: originate the call through your dialplan |
| 20:53.47 | kaldemar | Local is a fine tech |
| 20:54.20 | superpop02 | klife: you can use the local channel |
| 20:54.22 | kfife | Using Dial()? The problem is that Dial wants me to connect it to another channel, and I really want to play some DTMF, a recording, then terminate |
| 20:54.56 | kfife | I've been trying to get my head around how to do that. Once I connect to a local channel, the call has already been originated, so ti's too late. |
| 20:55.06 | kfife | ...that is if I use 'originate' |
| 20:55.44 | kfife | and thanks for your help by the way. |
| 20:56.30 | kfife | in other words, i want to place a call, play some DTMF tones, play a recording, conditionally some more tones, then hang up. |
| 20:59.01 | *** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194) |
| 20:59.23 | kfife | Since Dial() is connecting channels togheher rather than connecting a call to a given spot in the dial plan, I believe I need to drop a .call file, or trigger the call through AMI, or do a system(asterisk-rx originate...). That works, but the callerID is null. |
| 21:00.00 | [TK]D-Fender | kfife: call files & AMI Originate both have the option to set the callerID |
| 21:00.26 | kfife | [TK]D-Fender: Great news. That's just what I'd hoped you'd say. |
| 21:00.30 | kb3ien | every module make by dahdi is dependant on something else, where is 'crc_ccitt_table' defined? |
| 21:00.42 | kfife | [TK]D-Fender: I was looking for that option, but couldn't find it |
| 21:01.51 | kfife | my syntax was: exten => s,n,system(asterisk -rx "originate SIP/${TMOVM}@telasip-gw extension s@tmo-up-deeplink") |
| 21:01.52 | superpop02 | crc_ccitt_table is a generic module of the kernel |
| 21:02.24 | [TK]D-Fender | kfife: And that way I doubt you can. |
| 21:02.32 | superpop02 | crc-ccitt.ko |
| 21:02.41 | superpop02 | modprobe crc-ccitt |
| 21:02.46 | kfife | [TK]D-Fender: :-) |
| 21:03.33 | kb3ien | makes sense, but where does one get crc-ccitt.ko |
| 21:04.07 | jjshoe | why you would do that from within asterisk like that... |
| 21:04.09 | jjshoe | that's so scary |
| 21:04.10 | kfife | [TK]D-Fender: could you give me an example? |
| 21:04.36 | [TK]D-Fender | kfife: both are well documented. Go read the samples |
| 21:06.14 | *** join/#asterisk Greek-Boy (n=email@41.222.89.114) |
| 21:07.08 | kfife | the doc I read didn't show any way to do it from the CLI. Are you saying there's a way to use originate from the CLI, setting the callerID? I definitely saw how to do it from AMI and .call files |
| 21:07.36 | [TK]D-Fender | kfife: No, I did not say there was a way from CLI. |
| 21:07.50 | kfife | [TK]D-Fender: I misunderstood. |
| 21:07.56 | [TK]D-Fender | kfife: and I referred you to the 2 ways that DO work. |
| 21:08.18 | [TK]D-Fender | kfife: And then reinforced that by referring to them again as "both", to exclude CLI |
| 21:09.07 | jjshoe | kfife the way you are trying to do this is retarded. |
| 21:09.13 | jjshoe | exten => s,n,system(asterisk -rx "originate SIP/${TMOVM}@telasip-gw extension s@tmo-up-deeplink") |
| 21:09.13 | jjshoe | bad |
| 21:09.37 | jeev | that's so bad, i wouldn't even do it |
| 21:10.34 | kb3ien | is ccitt something normally missing from the kernel? |
| 21:10.53 | superpop02 | the ccitt is not loaded by default ... |
| 21:11.31 | superpop02 | you have to configure your linux to load at init sequence |
| 21:11.38 | kb3ien | not even built by default afaict, wow. major dependency to leave out the docs. |
| 21:11.41 | superpop02 | or load manually with modprobe ... |
| 21:11.57 | kb3ien | maybe i can get it built tonight. |
| 21:12.20 | superpop02 | what is your distrib ? |
| 21:13.50 | kb3ien | ubuntu |
| 21:15.21 | superpop02 | just type sudo modprobe crc-ccitt |
| 21:16.12 | superpop02 | and enter sudo echo crc-ccitt >> /etc/modules |
| 21:16.41 | kb3ien | okay, ive been in the bsd world a while, but WHY DIDNT I SEE THAT MODULE? |
| 21:17.06 | jjshoe | we have no idea why you are typing in caps! :D |
| 21:17.28 | tzafrir_laptop | it will be loaded if you modprobe zaptel |
| 21:17.31 | tzafrir_laptop | or dahdi |
| 21:17.46 | tzafrir_laptop | no need to put it in /etc/modules |
| 21:18.25 | *** join/#asterisk StephenF[W] (n=none@198.144.201.106) |
| 21:18.26 | superpop02 | just a example to resolve the problem ... |
| 21:19.20 | jeev | hey guys, my failover script @ www.jeev.net/asterisk/failover worked today! i saw it in my logs, WOO HOO |
| 21:19.29 | jeev | not a single thing, nobody was on the phone but it switched gateways and everything continued.. |
| 21:19.34 | kb3ien | my caps are in indicate the hair im pulling out. |
| 21:20.44 | stoffell | is there any more info somewhere on using BRI with the newest libpri's ? |
| 21:20.52 | *** join/#asterisk gr0mit (n=tim@router0.txrx.org.uk) |
| 21:21.19 | kfife | [TK]D-Fender: Again, my misunderstanding about the CLI option. Thanks for your help. |
| 21:21.50 | kfife | jjshoe: Question: How would you do it? |
| 21:21.53 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 21:22.01 | *** join/#asterisk bbryant (n=brett@c-68-59-20-153.hsd1.sc.comcast.net) |
| 21:22.02 | kfife | I'd prefer a better way. |
| 21:22.57 | jjshoe | I have no freaking clue what you're trying to do with that mess, but I would just use dial. |
| 21:23.26 | [TK]D-Fender | kfife: Either way works just fine |
| 21:24.38 | *** join/#asterisk nikko (n=nikko@69.57.49.100) |
| 21:25.04 | kfife | jjshoe: Yes, the thought had occurred to me :-). How would you use dial to: place a call, play some DTMF tones, play a recording, some more DTMF tones and hang up. Keep in mind the call is not connected to another channel. It's an outbound call triggered by a dialplan event. |
| 21:26.25 | ViKing78 | kfife: you could use an AGI script |
| 21:29.39 | *** join/#asterisk joobie (n=joobie@201.023.dsl.mel.iprimus.net.au) |
| 21:31.29 | kaldemar | kfife: Dial -> SendDTMF -> Playback -> SendDTMF -> Hangup |
| 21:31.42 | kfife | ViKing78: Correct. A self-contained single line in the dial plan per system(...originate...) is simpler, lighter weight, and therefore would perhaps have been better, if there were a way to set callerID using Originate from teh CLI. |
| 21:32.43 | kfife | kaldemar: That would work great on an inbound call. |
| 21:33.10 | kfife | kaldemar: Dial is looking to bridge two channels. It's not designed to originate calls as far as I can see. |
| 21:33.12 | kaldemar | tell be a real concrete reason why it wouldn't work with an outbound call |
| 21:33.28 | kaldemar | be->me |
| 21:33.53 | kfife | kaldemar: because 'flow' will stop at the priority of Dial() until the call is terminated |
| 21:34.17 | kaldemar | you can do the rest in the other end when the callee answers |
| 21:34.46 | *** join/#asterisk LemensTS (n=matthew@adsl-70-238-149-98.dsl.stlsmo.sbcglobal.net) |
| 21:34.56 | kfife | kaldemar: That's the crux of the issue. There is no 'other end' |
| 21:34.59 | kaldemar | so, other suggestions? |
| 21:35.16 | kaldemar | do you want help or do you want to argue? |
| 21:35.21 | kfife | kaldemar: :-) |
| 21:35.28 | kfife | Sorry didn't mean to sound terse. |
| 21:35.32 | kaldemar | i've done that exact thing myself. |
| 21:35.52 | kfife | I truly appreciate your ideas. |
| 21:35.59 | LemensTS | I want to have asterisk call a number, and play a message to a user when they answer. How can asterisk know when they answer? (this will be calling to analog phones) |
| 21:36.07 | kaldemar | that's not an idea, it's a working solution. |
| 21:36.23 | kfife | Hmmm. |
| 21:36.34 | kfife | maybe I don't understand your solution. |
| 21:37.10 | kfife | As far as I understand, once you use dial, asterisk sits there bridging until the call ends. Id doesn't process any more priorities. |
| 21:37.32 | kfife | so sendDTMF, playback etc would never get proceessed. |
| 21:37.51 | *** join/#asterisk tobias (n=tobias@user-0ce2hvs.cable.mindspring.com) |
| 21:38.16 | kaldemar | when you trigger a call with a call file, it first "dials" the channel given in the call file. that can point to a local extension where you can set the caller id and make the dial. when the callee answers, it triggers the extension defined in the call file. the extension can take care of the dtmf and the playback. |
| 21:38.48 | *** join/#asterisk tobias (n=tobias@user-0ce2hvs.cable.mindspring.com) |
| 21:38.55 | kfife | Yes, this is trivial with a call file. I though you were talking about using Dial() |
| 21:39.09 | jjshoe | you can do the same with dial |
| 21:39.24 | jjshoe | you can dial a context which does exactly what he said |
| 21:39.32 | jjshoe | I said dial only oh, a half hour ago? bleh. |
| 21:39.53 | kaldemar | Dial has G(context^exten^pri), you can use it. |
| 21:39.54 | kfife | :-) sorry guys if I'm being dense. Maybe I've got a false paradigm stuck in my head. |
| 21:39.59 | *** join/#asterisk nicoAMG (i=asgalt@201.203.96.42) |
| 21:40.16 | kaldemar | you should read some documentation before being so sure about what asterisk can and cannot do. |
| 21:40.24 | jjshoe | agreed |
| 21:40.47 | jameswf | asterisk is horrible software and doesnt work run away now |
| 21:41.11 | superpop02 | if is asterisk the originator of a channel, the cid_num of the channel is always null ? |
| 21:41.47 | LemensTS | Im still lost on how it knows when to play the wav file when they answer? Sometimes they will answer after 1 ring, sometimes 3 rings.... |
| 21:41.55 | jjshoe | superpop02 if you don't set it |
| 21:41.56 | superpop02 | with other word, in the event newchannel, if the calleridnum is <undefined>, its always asterisk the caller ? |
| 21:42.19 | kaldemar | LemensTS: a call has a state |
| 21:42.21 | jameswf | LemensTS: the difference is caller ID |
| 21:42.32 | jameswf | 3 rings means no cid |
| 21:42.58 | jjshoe | LemensTS I'm not sure with asterisk 1.4 or 1.6 |
| 21:43.12 | jjshoe | but asterisk assumes an analog channel is answered when it dials it i think |
| 21:43.37 | *** join/#asterisk Ast001 (n=uros@81.18.55.102) |
| 21:43.42 | superpop02 | with the manager API, its impossible to know if the caller of channel is the pbx or a device ? |
| 21:45.09 | kaldemar | jjshoe: no, it assumes a channel is answered when it actually is answered. |
| 21:45.34 | kfife | jjshoe: You're right. I just set the callerID, dial the e.164 number, bridge it to a local channel that has the specified sequence of DTMF, recordings, DTMF etc. |
| 21:45.35 | LemensTS | jameswf: http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out is this what your talking about? |
| 21:46.01 | Ast001 | hi, is it possible to have multiple calls waiting in queue on 2xISDN BRI connected with asterisk with TDM 404B when all 4 lines are busy ? |
| 21:46.25 | kfife | jjshoe: Thanks a lot. I was getting hung up on the idea of dialing an outside nubmer, rather than Dial() ing a simple local channe. |
| 21:46.39 | *** join/#asterisk stencil (n=stencil@pdpc/supporter/student/stencil) |
| 21:46.42 | Ast001 | or what will get 5th caller busy signal or music on hold in queue ? |
| 21:47.07 | superpop02 | ast001, the 5th will a get a busy signal |
| 21:47.22 | superpop02 | but no due to asterisk, but due to the bri protocol |
| 21:47.25 | superpop02 | the trunk ... |
| 21:47.44 | Ast001 | can I somehow change that ? |
| 21:47.56 | superpop02 | no |
| 21:48.02 | superpop02 | i think no |
| 21:48.34 | superpop02 | the 5 th will get "all line of your peer are busy, please call later" |
| 21:48.52 | superpop02 | this message is sent by the operator |
| 21:49.01 | superpop02 | and not by asterisk |
| 21:50.04 | Ast001 | and what if I used some digital card for conenct to Asterisk |
| 21:51.08 | superpop02 | you need more B channels on your bri trunk .. |
| 21:51.08 | Ast001 | like B200P ? |
| 21:51.28 | Ast001 | i see |
| 21:51.57 | *** join/#asterisk edoceo (n=edoceo@c-71-197-244-147.hsd1.or.comcast.net) |
| 21:51.59 | Ast001 | if I go to telco and want 8 channels and fix 2 on one single port of bri ? |
| 21:52.49 | Ast001 | in bri manual i read every port (2 of them) can serve up to 3 numbers |
| 21:54.21 | kb3ien | NICE : Oct 30 18:03:51 white kernel: [3334907.477496] asterisk[8536]: segfault at 000000000000001e rip 000000000046c769 rsp 00000000407f6410 error 4 |
| 21:54.33 | edoceo | Anyone have an example of enable/disable multi-ring dynamically? |
| 21:55.05 | edoceo | I have a multi-extension dial() that I want to dial *72 or the like to enable/disable, macro so any extension can use? |
| 21:56.55 | [TK]D-Fender | edoceo: "core show application gotoif" , "core show function DB" |
| 21:58.08 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 21:58.08 | *** join/#asterisk stimpie (n=stimpie@213-73-211-48.cable.quicknet.nl) |
| 21:58.08 | [TK]D-Fender | edoceo: edoceo this is your job to code your macro and the other extens to set values you can check to see who should be included with your Dial |
| 21:58.34 | Ast001 | can I solve this problem with MSN numbers ( Mulitple subsriber numbers ? ) |
| 21:59.49 | Ast001 | ok thanks for help |
| 21:59.52 | *** part/#asterisk Ast001 (n=uros@81.18.55.102) |
| 21:59.59 | *** part/#asterisk nido (i=nido@5ED105AD.cable.ziggo.nl) |
| 22:01.50 | edoceo | the-asterisk-book.com - ftw! |
| 22:03.32 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 22:04.16 | kb3ien | hahahah #include causes a segfault if the file is awol! |
| 22:10.07 | *** join/#asterisk jameswf (n=james@dsl093-157-131.phx1.dsl.speakeasy.net) |
| 22:13.26 | edoceo | any good tutorial on string manipulation of vars in extension.conf? |
| 22:14.31 | seanbright | yeah, on voip-info |
| 22:15.00 | *** join/#asterisk hardwire (n=hardwire@srv001.gandi.brutetech.com) |
| 22:15.10 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 22:15.16 | seanbright | edoceo: http://www.voip-info.org/wiki/view/Asterisk+variables |
| 22:15.24 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 22:17.08 | StephenF[W] | what do you guys use to transcode a high quality .wav file to ulaw for use in asterisk? |
| 22:17.15 | implicit | sox |
| 22:18.12 | StephenF[W] | implicit awesome, thx |
| 22:18.18 | jaytee | sox on the server or Audacity on a Windows or linux box |
| 22:18.26 | StephenF[W] | Audacity? |
| 22:18.34 | StephenF[W] | I can figure out how to export to ulaw from Audactiy |
| 22:18.36 | jaytee | yeah, it's freeware |
| 22:18.58 | superpop02 | There is a simple way to know the channel direction with manager api ? |
| 22:19.00 | jaytee | set it to 8khz ulaw in the export |
| 22:19.04 | jaytee | mono |
| 22:19.08 | StephenF[W] | ohh |
| 22:19.41 | StephenF[W] | in preference |
| 22:19.45 | jaytee | and if you need to edit a .ulaw file you need to import it, not open it or it'll transcode it to the default |
| 22:20.13 | jaytee | and import as raw data |
| 22:20.30 | StephenF[W] | gotcha |
| 22:22.12 | StephenF[W] | jaytee is the ulaw setting in Preferences? |
| 22:22.31 | *** join/#asterisk d00gster (n=doughant@bas1-cooksville01-1176000374.dsl.bell.ca) |
| 22:22.36 | jaytee | when you go to export click the options button and select Format=Other Header=WAV(Microsoft) and Encoding= U-Law and then just make sure to name the file with a .ulaw extension. I think sox with resampling actually produces cleaner output though. |
| 22:23.03 | StephenF[W] | ok thats my problem, I have no option when I click Export as WAV |
| 22:23.11 | StephenF[W] | just a save file dialog, maybe because im on Vista |
| 22:23.49 | jaytee | must be, but I get the Options button on both XP and Ubuntu |
| 22:24.33 | jaytee | Anyone want to take a wild ass guess as to why this evaluates to false when the speech_score is 1000? exten => s,n,Gotoif($["${SPEECH_SCORE(0)}">"${THRESHOLD}"]?proceed:repeat) |
| 22:24.44 | StephenF[W] | weird... maybe im doing something wrong. You just open the .wav file and then goto Export as WAV right? |
| 22:24.51 | jaytee | and the threshold is set to 750 |
| 22:25.22 | jaytee | StephenF[W], I don't have an Export As Wav menu option, I have Export and Export Selection |
| 22:26.02 | StephenF[W] | hmm, ok |
| 22:26.11 | *** join/#asterisk freakazoid0223 (n=mattc@pool-68-162-71-132.phil.east.verizon.net) |
| 22:26.14 | StephenF[W] | i think I found it in edit > preferences |
| 22:26.33 | jaytee | You might have a newer version for Vista with extra "preferences" :-) |
| 22:26.39 | StephenF[W] | yippe |
| 22:27.28 | jaytee | I have this pc setup for dual boot with Vista but I haven't booted it into Vista in at least 2 months probably 3. |
| 22:27.32 | StephenF[W] | alright, I'll see how this sounds and then try SOX if I dont like it |
| 22:27.46 | jaytee | there's also file convert from the CLI |
| 22:28.14 | StephenF[W] | i switched my home machine over to ubuntu about 6mo ago and havent looked back. But im at the office now with Vista |
| 22:28.20 | LemensTS | Ok i setup a call file, but when i plays it dials my phone number and imediatly plays the background file before i answer...here is the config http://pastebin.com/m66262c65 |
| 22:28.37 | StephenF[W] | i do use a win xp VM at home for Quicken though... |
| 22:28.44 | kaldemar | jaytee: quotes make it false. don't quote plain numbers. |
| 22:29.12 | jaytee | kaldemar, thanks, I'll give that a shot. I got that from some example code from Lumenvox |
| 22:30.00 | jaytee | kaldemar, the odd thing is that this only affects DTMF, if I speak a menu option clearly it works, if I mumble the score is below threshold and repeats. |
| 22:30.30 | kaldemar | comparing numbers and strings is not the same thing. |
| 22:31.04 | jaytee | kaldemar, do you use Lumenvox at all? |
| 22:33.50 | kaldemar | jaytee: no |
| 22:34.33 | LemensTS | im not sure why |
| 22:35.39 | jaytee | it's not too bad and at what they charge for a 5 port 500 grammar license it's way cheaper to "roll your own" than buy a commercial Speech Recognition system. |
| 22:36.50 | jaytee | StephenF[W], I run XP in a VM at work for Outlook, IE7 to access our Sharepoint Portal and Visio and that's about it. |
| 22:37.30 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.static.bezeqint.net) |
| 22:37.48 | jaytee | and I have an old 1.7ghz clunker clone I built in early 2003 with XP on it here at home that I use just for VPN to my work. |
| 22:39.03 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 22:45.04 | *** join/#asterisk jov4n (n=jovan@87.18.99.198) |
| 22:45.08 | jov4n | Hi |
| 22:45.13 | hi365_m | ~devstate |
| 22:45.14 | jbot | [~devstate] Devstate is an Asterisk 1.4 module for custom BLF device state, see the following link -=- http://svncommunity.digium.com/svn/russell/asterisk-1.4/func_devstate-1.4/ |
| 22:48.09 | watchy | anyone here use openbsd? |
| 22:48.26 | *** part/#asterisk russellb (n=russellb@asterisk/digium-open-source-team-lead/russellb) |
| 22:48.30 | drmessano | I'm not really into pokemon |
| 22:48.41 | watchy | i like pokemon |
| 22:49.44 | *** join/#asterisk DarylVOIP (n=daryl@75.147.121.177) |
| 22:50.41 | stencil | hi watchy, yes I do |
| 22:51.13 | stencil | what is the problem? |
| 22:51.16 | *** join/#asterisk `Sean (i=Shaan@CPE0016d3fe20a0-CM001ade84fd0a.cpe.net.cable.rogers.com) |
| 22:51.30 | `Sean | hey can someone please tell me what are some cheap and good toll free did providers |
| 22:52.10 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 22:53.29 | lesouvage | . |
| 22:54.07 | watchy | stencil: how do you restart services? |
| 22:54.14 | watchy | is there a /etc/rc.d like in fbsd? |
| 22:54.57 | stencil | what service do you mean?? |
| 22:55.34 | *** join/#asterisk nikko (n=nikko@12-218-250-137.client.mchsi.com) |
| 22:55.57 | *** join/#asterisk anthm][ (n=anthm@99-205-234-209.area4.spcsdns.net) |
| 22:56.38 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 22:58.50 | *** join/#asterisk LiNeTuX (n=LiNeTuX@253.238.95.24.cfl.res.rr.com) |
| 22:59.43 | edoceo | I have softphone that can register and I can call it but when I dial out I get a 503? |
| 23:00.59 | iratik | Is there an AMI command to find out what number an extension is talking to? |
| 23:01.18 | iratik | ExtensionState doesn't show, and i can't figure out how to get the channel name |
| 23:01.23 | iratik | Status just keeps saying channel not found |
| 23:02.31 | kb3ien | okay, my polycoms ARNT hitting the asterisk box. how readdily will they syslog? do they take dhcp options for that? |
| 23:03.39 | *** join/#asterisk grantm (n=grant@68.142.138.4) |
| 23:06.16 | `Sean | hey can someone please tell me what are some cheap and good toll free did providers |
| 23:08.34 | ManxPower | `Sean: without you defining both cheap and good, the answers you get won't do much good. |
| 23:10.06 | rob0 | Cost less than $1/minute, over 10% uptime. |
| 23:10.49 | edoceo | Which var has the originating extension? |
| 23:12.51 | LemensTS | Ok i setup a call file, but when i plays it dials my phone number and imediatly plays the background file before i answer...here is the config http://pastebin.com/m66262c65 |
| 23:13.58 | `Sean | ManxPower not cheap well quality |
| 23:14.00 | StephenF[W] | hmm, so i downloaded Audacity 1.3 and now I have export options. But when I export the file to ulaw and try to play it in asterisk it sounds like an alien |
| 23:14.19 | ManxPower | Teliax and Vitelity would work. |
| 23:14.24 | ManxPower | ~itsp |
| 23:14.24 | jbot | [~itsp] An ITSP is an Internet telephony Service Provider or "VoIP Telephone company". The allow you to either PLACE calls to the PSTN (called Termination), RECEIVE calls from the PSTN (called Origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs |
| 23:14.48 | StephenF[W] | I resampled to 8kz, set tracks to mono, then exported as WAV (Microsoft) encoding: U-Law... |
| 23:14.54 | StephenF[W] | am I missing something? |
| 23:15.47 | ManxPower | StephenF[W]: load a file that works in asterisk and see what settings it has. Use the Record application if you don't have any working .WAV files |
| 23:16.12 | StephenF[W] | the files im trying to export are .WAV files |
| 23:16.16 | vader-- | 8 Hours and no PRI drop |
| 23:16.17 | jaytee | StephenF[W], not sure. Try converting the WAV to 8khz and saving and see how it sounds when you play it in Audacity. |
| 23:16.17 | vader-- | WHOOOOO |
| 23:16.31 | StephenF[W] | ok |
| 23:16.34 | jaytee | vader, 177 days and no PRI drop :-) |
| 23:16.40 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 23:16.49 | vader-- | well that was me befor |
| 23:17.20 | hardwire | I have iax2 peers set up between two servers |
| 23:17.55 | jaytee | hardwire, what a coincidence. So do I |
| 23:18.24 | *** join/#asterisk tobyrussell (n=tobias@farnstr.coreware.co.uk) |
| 23:19.49 | hardwire | but it seems like host={publicip} is being completely |
| 23:19.51 | hardwire | ignored |
| 23:19.57 | hardwire | jaytee: blah.. i was interrupted :) |
| 23:20.15 | ManxPower | host= is mainly for OUTBOUND connections. permit=/deny= is for access control |
| 23:20.21 | hardwire | ohyeh |
| 23:20.38 | `Sean | Hey, i had a weird question does anyone know how to set asterisk up to dail a list of numbers like froma big list dail like 5-10 numbers and then play a audio recording and if they press 1 then forward that call to a rep? |
| 23:20.48 | ManxPower | There seems to be some community amnesia about permit=/deny= |
| 23:21.11 | jaytee | I don't recall anyone discussing that :-) |
| 23:21.26 | hardwire | ManxPower: either way.. danke danke |
| 23:21.34 | ManxPower | hardwire: Rumor is there was some half-assed attempt to make host= do access control, but I'd never use it that way |
| 23:21.41 | jer | `Sean, can you put me on your DNCL ? |
| 23:21.43 | jer | =D |
| 23:21.44 | ManxPower | jaytee: exactly! |
| 23:21.58 | ManxPower | It's listed in the sip.conf.sample and nobody ever reads that |
| 23:22.03 | jaytee | I did |
| 23:22.23 | jaytee | I have my IAX2 peers between each server trunking just fine. |
| 23:22.46 | ManxPower | jaytee: and you are one of the people not asking questions that are answered by reading sip.conf.sample |
| 23:22.56 | StephenF[W] | sounds fine when exported as a .wav file... |
| 23:23.05 | *** join/#asterisk Ariel_Calzada (n=aricalso@dsl-emcali-200.29.106.116.emcali.net.co) |
| 23:23.59 | jeev | jaytee, my script worked today. i saw it ;) failover 100% |
| 23:24.08 | *** join/#asterisk km2 (n=x@mobile-166-217-049-099.mycingular.net) |
| 23:24.19 | StephenF[W] | so then its something with the U-Law encoding? |
| 23:24.32 | StephenF[W] | maybe im not using U-Law on my phones? |
| 23:24.39 | jaytee | now if I could just figure out why Background jumps to the defined context after only 1 digit I'd be happy. I had to modify some code I was writing to use Read instead. Even after [TK]D-Fender gave me some advice and I made the changes. I think it's something buggy in my system because I'm also using SpeechBackground with Lumenvox's Speech Rec engine. |
| 23:25.39 | jaytee | if I try similar code on my primary * server that doesn't use the Lumenvox connector Background works as expected. |
| 23:25.40 | StephenF[W] | i've got disallow=all allow=ulaw in my sip conf, so that should force my phones to use ulaw... |
| 23:26.02 | jaytee | StephenF[W], you need to set the preference for codecs on the phones too |
| 23:26.41 | jeev | jaytee, fender gives you advice? all he gives me is a bunch or two to the effigy |
| 23:26.43 | StephenF[W] | jaytee, think that could be causing this? |
| 23:27.14 | jaytee | StephenF[W], what could be causing what? |
| 23:27.29 | StephenF[W] | the audio file not playing right in asterisk |
| 23:27.48 | StephenF[W] | i exported it as 8khz ulaw encoding, and it uncomprehensible. sounds like an alien breathing |
| 23:27.57 | StephenF[W] | with gsm encoding i just get random noises.. |
| 23:27.58 | jaytee | StephenF[W], are you running the Lumenvox speech engine and the connector module in Asterisk? |
| 23:28.14 | StephenF[W] | um, i have no clue what that even is |
| 23:28.26 | jaytee | then no, that would be what's causing it :-) |
| 23:28.47 | StephenF[W] | im just trying to convert a .wav file I got from a voice talent to a usable format for asterisk |
| 23:28.49 | jaytee | try using sox instead of Audacity. there's tons of good examples for converting on the WIKI |
| 23:29.01 | StephenF[W] | jaytee ohh, no i was repying to yor "you need to set the preference for codecs on the phones too" |
| 23:29.24 | StephenF[W] | sox wont open my .wav file, file type 'auto' unknown |
| 23:29.32 | StephenF[W] | lemme try it again |
| 23:29.49 | jaytee | StephenF[W], if the phone is using a different codec and needs to transcode that could cause problems. There's a known bug with the GSM codec |
| 23:30.13 | ManxPower | I still think my suggestions was the best |
| 23:30.16 | jaytee | StephenF[W], before you waste any more time, go to the Wiki and read up |
| 23:30.23 | jaytee | ManxPower, which was? |
| 23:30.53 | jaytee | ManxPower, never mind I scrolled back. Yeah. |
| 23:31.01 | ManxPower | jaytee: you don't need to set the codec preferences on the phone. My suggestion that he get a .WAV what does work in Asterisk (or make one using Record) then find out what kind/hz/etc it is |
| 23:31.08 | jeev | checks his microphone |
| 23:31.39 | jaytee | ManxPower, if his phone is set to not even use ulaw as a preference then wouldn't * force a transcode? |
| 23:32.11 | ManxPower | jaytee: yes, but I've never ever seen a phone that did not default to allowing ulaw |
| 23:32.56 | ManxPower | jaytee: if the phone did not allow ulaw and asterisk had disallow=all / allow=ulaw, Asterisk would not even accept the call. |
| 23:33.02 | jaytee | ManxPower, that's what I thought but I've only used Grandsuck and Polycom. Couldn't be sure there might be some phones out there that had ulaw capability but not enabled as a choice at all. |
| 23:33.49 | jaytee | ManxPower, ah so in that scenario it wouldnt' force a transcode. I think I've seen that error it throws posted in a pastebin before. |
| 23:34.28 | StephenF[W] | im on Polycom anyway |
| 23:34.38 | ManxPower | "No compatible codecs found" |
| 23:36.38 | jeev | still searches for a codec to decode ManxPower |
| 23:37.30 | StephenF[W] | maybe because the files were 32 bit? |
| 23:38.17 | drmessano | wonders if lmadsen turned off the Jeev.sh Fail2Ban script running in #asterisk |
| 23:38.50 | *** join/#asterisk coppice (n=chatzill@33.155.17.210.dyn.pacific.net.hk) |
| 23:39.33 | StephenF[W] | bingo |
| 23:39.37 | StephenF[W] | needs to be 16bit |
| 23:39.39 | drmessano | You got a wav file from talent? |
| 23:39.44 | StephenF[W] | yes |
| 23:39.46 | StephenF[W] | is that weird? |
| 23:39.47 | drmessano | Odd |
| 23:39.49 | drmessano | Yes |
| 23:39.55 | StephenF[W] | what do you normally get? |
| 23:40.09 | drmessano | Most voice talent sends mp3s now |
| 23:41.00 | StephenF[W] | that we she normally would send, I asked for .wav because I thought it would end up sounding better |
| 23:41.05 | drmessano | If it's gonna be a WAV, 22khz 16bit mono is pretty standard |
| 23:41.38 | StephenF[W] | i figured mp3 is already compressed, then I would be transcoding it again |
| 23:41.45 | StephenF[W] | probly wouldnt have made a difference |
| 23:42.01 | ManxPower | mp3 sucks up MASSIVE amounts of CPU compared to the other codecs |
| 23:42.05 | drmessano | They're gonna downsample the wav to make it fit in an e-mail, so not really |
| 23:42.18 | StephenF[W] | no I asked for uncomressed |
| 23:42.24 | drmessano | I didnt say compressed |
| 23:42.43 | StephenF[W] | 44khz |
| 23:43.03 | ManxPower | 16-bit / 8Khz / mono |
| 23:43.03 | StephenF[W] | is that max? |
| 23:43.09 | ManxPower | that's what you want for Asterisk |
| 23:43.11 | jaytee | it sucks because I know I'll be dead and gone for a long time before they come out with a wideband VOIP solution that supports 32 bit 44,400hz stereo with 5.1 Dolby for MOH. :-) |
| 23:43.55 | jeev | damn |
| 23:44.00 | drmessano | You were probably better off with the 192k MP3 and converting it down |
| 23:44.03 | jeev | jaytee, we're all gonna die soon.. doesn't matter |
| 23:44.08 | jeev | doesn't the mayan calendar say 2012? |
| 23:44.18 | jaytee | I'm not Mayan :-) |
| 23:44.24 | ManxPower | drmessano: what did StephenF[W] ever do to you? |
| 23:44.31 | drmessano | Why? |
| 23:44.38 | edoceo | how to do string manipulation without ael? |
| 23:44.48 | edoceo | Hack some AEL into exteinosons.conf? |
| 23:44.55 | jaytee | but I am tired of listening to my stomach growl so I'm off to KFC for some original recipe strips and mashed taters. bbiab |
| 23:45.22 | ManxPower | edoceo: Your question makes no sense |
| 23:45.43 | StephenF[W] | lol anyways, its working now so I'm happy. Thx guys |
| 23:45.50 | ManxPower | There is nothing you can do in AEL that you cannot do in extensions.conf. IN FACT, AEL is converted to extensions.conf format when it loads |
| 23:45.57 | drmessano | Getting a 192k MP3 and downconverting it to 8k/16/mono is better than dealing with WAVs.. You're gonna end up with about the same |
| 23:46.09 | drmessano | WAV's being transported anyway |
| 23:46.30 | StephenF[W] | drmessano, well what if the wav was 44khz 32 bit ? |
| 23:46.42 | drmessano | Nothing like a nice uncompressed 7MB e-mail you're gonna smash into telephone headset quality |
| 23:46.43 | StephenF[W] | is that uncompressed and not down sampled? |
| 23:47.03 | StephenF[W] | hehe, well the file was like 16MB |
| 23:47.07 | *** part/#asterisk stencil (n=stencil@pdpc/supporter/student/stencil) |
| 23:47.08 | StephenF[W] | I guess that is overkill |
| 23:47.09 | drmessano | Again.. Why the hell have them send something you're gonna butcher the shit out of anyway? |
| 23:47.31 | StephenF[W] | hey, like I said wasnt sure. So I said "send the best" |
| 23:48.01 | StephenF[W] | next time ill just ask for mp3 |
| 23:48.14 | edoceo | ManxPower: Yes, my version of astersisk does not support AEL, but I want some string manipulation like what is possible with AEL |
| 23:48.17 | drmessano | "send the best since i'm gonna squeeze it down to 8khz, 16bit mono.. " |
| 23:48.36 | edoceo | Since I cannot upgrade, does anyone have examples of string manipulation with regular commands from extensions.conf |
| 23:48.38 | drmessano | Thats like getting premium $10 oranges to make xmas ornaments out of.. |
| 23:48.45 | drmessano | heh |
| 23:48.51 | drmessano | Ok, gotta go shopping.. bbiab |
| 23:48.53 | StephenF[W] | well now I know |
| 23:49.05 | StephenF[W] | ill get the $2 oranges next time |
| 23:49.10 | *** join/#asterisk tAnkOSX (n=tank@the.matrix.has-you.net) |
| 23:49.13 | ManxPower | edoceo: Set(FNORD=tommy) Set(MARY=${FNORD}morestuff_ |
| 23:49.22 | ManxPower | There! i've manipulated a string! |
| 23:49.30 | ManxPower | now what are you actually trying to do? |
| 23:49.51 | coppice | most prompt files sound far worse than the codec they use is capable of. even ones where people have spent real money on professional voices. people don't seen to take much care with these things |
| 23:50.10 | edoceo | From the string Sip/100&SIP/200&SIP/400 I want to remove SIP/200 and one of the & |
| 23:50.31 | ManxPower | edoceo: "core show application cut" or "core show function CUT" |
| 23:51.37 | ManxPower | edoceo: All the super secret don't tell anyone Asterisk documention is in "core show applications" "core show functions" as well as the individual core show application and core show function commands |
| 23:51.38 | `Sean | hey manxpower can i msg you? |
| 23:51.47 | ManxPower | `Sean: is it to send me money? |
| 23:52.16 | `Sean | no ManxPower i need help was wondering how can i get asterisk to call off a list and if the person press's one then to transfer to a agent |
| 23:52.19 | edoceo | ManxPower: Thanks! |
| 23:52.30 | ManxPower | `Sean: I do not do private consulting for free. |
| 23:52.49 | `Sean | ok can we discuss this in private then? |
| 23:53.06 | ManxPower | `Sean: No! |
| 23:53.42 | `Sean | ManxPower dude well will you be able to tell me wich application of asterisk can do that? |
| 23:53.54 | *** join/#asterisk telecos (n=sergio@153.166.219.87.dynamic.jazztel.es) |
| 23:59.41 | hardwire | ok.. I'm not used to unauthenticated sip w/o guest accounts |
