00:03.13 | goodbot | hopefully someone gets to handling those errors sooner or later :(. Thanks anyway folks. |
00:03.22 | goodbot | I'll log a bug now I know where to do it. |
00:03.25 | *** join/#asterisk stkn (i=nobody@gentoo/developer/pdpc.active.stkn) |
00:10.11 | *** join/#asterisk xnon (n=xnon@200.8.30.161) |
00:10.29 | xnon | amigos |
00:10.31 | xnon | friends |
00:10.41 | xnon | i have a warning about MOH |
00:10.56 | xnon | Oct 22 02:07:16 WARNING[4883]: res_musiconhold.c:838 moh_register: Unable to open pseudo channel for timing... Sound may be choppy. |
00:11.05 | xnon | what kind of warning is it? |
00:11.18 | xnon | i can to understand |
00:11.24 | macTijn | you have no zap device |
00:11.30 | xnon | emmm no |
00:11.31 | macTijn | so there's no real timing |
00:12.01 | xnon | but this warning it is important? |
00:12.09 | macTijn | so all sound stuff is happening based on what asterisk thinks is a timely fashion |
00:12.13 | macTijn | well |
00:12.23 | macTijn | does your moh sounds choppy ? |
00:12.39 | xnon | i dont know what is choppy! :p |
00:12.41 | xnon | jejejee |
00:12.44 | xnon | sorry |
00:12.47 | tzafrir_laptop | ztdummy should do |
00:12.55 | macTijn | tzafrir_laptop: true |
00:13.01 | macTijn | xnon: jittery |
00:13.03 | tzafrir_laptop | It should provide a pseudo channe |
00:13.22 | xnon | i have son songs in a directory /var/lib/asterisk/mohmp3 |
00:14.49 | macTijn | xnon: how does the moh sound ? |
00:15.03 | macTijn | does it sound it's playing "weird" ? |
00:15.09 | macTijn | or does it sound OK ? |
00:15.18 | xnon | sound ok i think so |
00:15.24 | macTijn | ok |
00:15.30 | xnon | but my asterisk console say this error |
00:15.32 | xnon | :( |
00:15.39 | macTijn | then my opinion is you can ignore the warning |
00:15.41 | macTijn | it's not an error |
00:15.44 | macTijn | it's a warning |
00:15.48 | xnon | and others errors that i cant understand |
00:15.59 | xnon | ok |
00:16.04 | macTijn | try googling |
00:16.07 | xnon | macTijn, look at this |
00:16.10 | xnon | Oct 22 02:07:17 WARNING[4883]: chan_iax2.c:9572 load_module: Unable to open IAX timing interface: No such file or directory |
00:16.18 | *** join/#asterisk Dovid (n=Dovid@barak.cellcom.co.il) |
00:16.21 | xnon | this warning it is important? |
00:16.30 | macTijn | same shit |
00:17.06 | xnon | :S |
00:17.08 | xnon | what? |
00:17.10 | xnon | jejejej |
00:17.29 | Dovid | warnings in asterisk are info - not soo much that ur gona crash |
00:18.03 | xnon | ok but why say this warning my console? |
00:18.17 | xnon | chan_iax2.c cant find it |
00:18.27 | xnon | where are this files? |
00:20.03 | hads | xnon: That warning is caused by the same thing as the other one. |
00:20.11 | *** join/#asterisk topping (n=topping@207.47.6.182.static.nextweb.net) |
00:21.03 | xnon | ummmm ok |
00:21.12 | xnon | so isnt important really? |
00:23.08 | *** join/#asterisk Defraz (n=t0tal@fw.centrisys.com) |
00:23.47 | Defraz | I know this is asterisk but while I was here does anyone know the name of that song that there is just a bunch of laughing in. It is a newer song. |
00:23.57 | Defraz | I hear it on the radio but I never hear the dang name of it. |
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00:31.33 | *** join/#asterisk saftsack (n=oliver@p54A7FB13.dip.t-dialin.net) |
00:36.16 | saftsack | hi |
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00:44.02 | teknoprep | i am at a loss... looking for some info on running high volume asterisk box in vmware |
00:44.24 | teknoprep | maby some pointers on keeping stutter and timing issues to a minimum |
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00:47.31 | saftsack | teknoprep, do you have * running on an openwrt? |
00:47.39 | teknoprep | no |
00:47.42 | teknoprep | inside of vmware |
00:47.48 | saftsack | ok |
00:47.59 | saftsack | why that? do you have more than one asterisk, or waht? |
00:48.05 | teknoprep | nope |
00:48.06 | saftsack | i mean more than one in one computer? |
00:48.48 | teknoprep | no... i run all my servers except of course the file servers inside of vmware |
00:48.55 | tzafrir_laptop | Asterisk is generally not the ideal application to run inside vmware |
00:48.57 | teknoprep | everything from fireall's to active directory... |
00:49.06 | teknoprep | well i do not run anything except VoIP |
00:49.21 | tzafrir_laptop | Asterisk, and real-time application in general, need timely reaction from the CPU |
00:49.35 | saftsack | tzafrir_laptop, do you know anything about * on openwrt? |
00:49.47 | teknoprep | but i am starting to see there are a few problems... like the stuttering... which is pretty much the only application i am worried about |
00:50.11 | tzafrir_laptop | teknoprep, get a dedicated box. It will be just as good |
00:50.29 | tzafrir_laptop | Putting Asterisk on the same box assomething else is not good |
00:51.37 | tzafrir_laptop | hmmm... firewall inside vmware? hmmm ... so you trust both the vmware host setup and the unbreakability of the vmware virtualization ... |
00:52.10 | tzafrir_laptop | saftsack, sorry. NO |
00:52.12 | saftsack | ok |
00:52.15 | tzafrir_laptop | no, that is |
00:52.43 | saftsack | is it possible that asterisk strictly divorces the media from the other signals in sip? |
00:53.22 | *** part/#asterisk Agrajag- (n=filip@c211-30-4-5.artrmn1.nsw.optusnet.com.au) |
00:55.31 | teknoprep | tzafrir_laptop, the NSA wrote up a report on it.. that it can be trusted.. so yes i do also |
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00:58.10 | tzafrir_laptop | teknoprep, but what does it give you? |
00:58.57 | teknoprep | tzafrir_laptop, what does what give me? vmware? well if you don't really know.. you should check itout... ease of backups, High Availability, Load Balancing, the list goes on |
00:59.34 | teknoprep | tzafrir_laptop, reverting to snapshots is easy too.. which is a very nice feature for planned upgrades that go bad |
00:59.39 | tzafrir_laptop | teknoprep, for special-purpose machines such as a firewall |
00:59.57 | teknoprep | tzafrir_laptop, but low CPU usage applications |
01:01.39 | teknoprep | tzafrir_laptop, lets agree to disagree... only thing i am agreeing with you on.. is that asterisk because of its audio and the human ear can detect the imprefections of audio.. i probably will need to put it on another box... BUT, vmware for most applications run them perfectly... |
01:03.02 | teknoprep | tzafrir_laptop, the only imperfections i can hear and/or anyone else with the setup i have now is transcoding IVR's / VoiceMail or any other pre-recorded applications... transcoding live audio sounds the same... its the only thing i am having a problem with and i don't know if it is becuase i store my OS on an iscsi target and i am adding a delay factor there or if it is the Virtualization proccess |
01:04.23 | teknoprep | hmmmm you know what.. at home i never hear the delay.. maby it is the storing of the OS on an iSCSI target... at home my IVR's or VM's never lag and i also run them inside of vmware.. but i run them on the localhost's array |
01:04.50 | tzafrir_laptop | teknoprep, the OS practically sits in the memory for the time of the operation. Sound files may be on a disk. So to isolate the problem, create a small ramdrivve and put the sound files in it (using loopback mounting) |
01:05.11 | tzafrir_laptop | I believe that this should eliminate most disk accesses |
01:05.45 | tzafrir_laptop | (a bit complex, but can help for testing) |
01:06.00 | teknoprep | naw i can get that done easy |
01:06.46 | teknoprep | why not just mount all VM and IVR recordings to a local RAM Disk ? |
01:06.46 | teknoprep | the directorys |
01:06.46 | teknoprep | that wouldn't be good on a failure tho |
01:09.09 | saftsack | does someone of you both have experience with patton products? |
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01:26.50 | Defraz | I know this is asterisk but while I was here does anyone know the name of that song that there is just a bunch of laughing in. It is a newer song. |
01:26.56 | Defraz | anyone heard of a song like this? |
01:27.00 | Defraz | Driving me nuts haha |
01:27.50 | *** join/#asterisk linagee (n=linagee@unaffiliated/linagee) |
01:28.35 | napkin | Defraz: don't worry, sounds like you're already nuts :) |
01:28.46 | Defraz | haha thanks |
01:29.04 | Defraz | Just heard it and I want to find it but I can't remember the names. |
01:29.08 | Defraz | It is so funny to me. |
01:30.29 | *** part/#asterisk gmaruzz (n=gmaruzz@217-133-80-112.b2b.tiscali.it) |
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01:38.37 | SuPrSluG | saftsack:i did * on wrt |
01:38.38 | saftsack | SuPrSluG, hi |
01:38.38 | saftsack | did it work? :) |
01:38.38 | saftsack | can you tell me something about it? |
01:39.50 | SuPrSluG | yes. only issue is with voicemail storage. if you want to do it right get an asus geluxe w/ usb ports for storage. I tried nfs but it was a bit choppy ymmv |
01:40.10 | saftsack | sounds quite well :) |
01:40.17 | saftsack | what was with the performance? |
01:40.26 | saftsack | how many clients did you manage on your openwrt? |
01:40.36 | SuPrSluG | well i wouldn't go over 4-5 users |
01:40.51 | SuPrSluG | couple |
01:40.59 | saftsack | asus deluxe or what? :> |
01:41.10 | saftsack | but isnt asterisk to big for openwrt? ^^ |
01:41.12 | SuPrSluG | they only have 200mz processors |
01:41.14 | saftsack | i mean for 6mb |
01:41.20 | SuPrSluG | no |
01:41.49 | SuPrSluG | you dump alot of non used modules |
01:42.00 | saftsack | sounds great :) but i dont know why i can just manage 5 users :> i mean asterisk doesnt do any voice processing does it do it? |
01:42.44 | SuPrSluG | if your using ulaw no. gsm or a compressed codec yes |
01:43.22 | saftsack | in the case that i use alaw i think its the same in processing as ulaw is it possible to handle about 15 people? |
01:43.44 | SuPrSluG | or i should say as long as * doesn't have to transcode |
01:44.01 | SuPrSluG | maybe |
01:44.21 | saftsack | sounds good so i will try this :) |
01:44.23 | SuPrSluG | i didn't stress test it |
01:44.25 | saftsack | do you know the fritz boxes? |
01:44.46 | SuPrSluG | no what ate they? |
01:45.07 | saftsack | this are boxes from a german manufactor which are newly supported |
01:45.18 | saftsack | i will try asterisk on it the next days i think |
01:45.46 | saftsack | this would be great if i have all my pbx on embedded devices which are working properly :) |
01:46.04 | SuPrSluG | you can always flash it back. but they do warn you that you can toast it |
01:46.24 | saftsack | hrhr |
01:46.49 | SuPrSluG | look at the asus deluxe router. it has 2 usb ports |
01:47.01 | *** join/#asterisk anthonyc (n=ac@adsl-75-31-162-88.dsl.frs2ca.sbcglobal.net) |
01:47.23 | napkin | can you guys assure me i'm not on crack, so i don't waste $300? i'm switching our office to voip, and we need 4 lines - that means we need 4 fxs ports right? :) |
01:47.44 | SuPrSluG | fxo |
01:47.48 | saftsack | ASUS WL-500g Deluxe |
01:47.51 | saftsack | do you mean this? |
01:48.18 | napkin | SuPrSluG: you're joking right? |
01:48.33 | SuPrSluG | that's the one they rave about. i did it on a linksys. nfs not great for voicemail |
01:48.50 | napkin | we're not getting any land-lines... everything will go over the internet |
01:48.53 | SuPrSluG | u have 4 incoming lines? |
01:49.11 | napkin | nah just 4 orgs who need different lines out |
01:49.24 | saftsack | sumasuma, rave? you mean this is the good one with usb ports? |
01:49.28 | *** join/#asterisk b11d (n=noway@234-200-29-134.hcc.mnscu.edu) |
01:49.36 | b11d | hello |
01:49.49 | b11d | should I be wasting my time with an extensions.conf, or should I go with this "ael" stuff? |
01:50.40 | SuPrSluG | fxo modules connects to lines fxs connects to phones |
01:51.06 | napkin | k i just had to ask someone, even after reading about it, to make sure i wasn't surprised. :) |
01:51.24 | saftsack | b11d, what is ael?= |
01:51.43 | SuPrSluG | ~ael |
01:51.45 | jbot | i guess ael is Asterisk Extension Language - a dialplan language with 'c like' syntax? |
01:52.07 | saftsack | are there any advantages why i should use ael? |
01:52.15 | SuPrSluG | so if your a programmer you might like it |
01:52.49 | b11d | so no one is really using it then? |
01:53.23 | napkin | b11d: well there are ~100 people in this channel, and 3 people answered. |
01:54.12 | b11d | wow, i totally didnt notice that. |
01:54.21 | b11d | thanks for "awakening" me |
01:54.25 | saftsack | here are exactly 244 people |
01:54.28 | SuPrSluG | i would imagine they are. took a hell of a lot of effort to write it, some somebody must use it |
01:54.29 | saftsack | including bots and so on |
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01:55.00 | SuPrSluG | they're all out having a smoke |
01:55.07 | anthonyc | im eating a hamburger |
01:55.19 | TheMafia | is there some service that I can register my sip phone with? |
01:55.35 | SuPrSluG | free or pay? |
01:55.51 | saftsack | umts telephony support video calls, or? |
01:55.57 | TheMafia | free |
01:56.06 | SuPrSluG | free world dialup |
01:56.24 | saftsack | what is if i have a sip phone with a camera? are there gateways for calling a umts user with a video call? |
01:56.44 | TheMafia | yep, that was the one I was trying to remember, thanks |
01:56.56 | SuPrSluG | never tried video |
01:57.17 | SuPrSluG | anytime |
01:57.34 | saftsack | i go to bed now |
01:57.40 | saftsack | it is 3:59 here ^^ |
01:57.48 | napkin | b11d: sorry, so many people join a channel, ask a question, and only wait a few minutes before jumping to the assumption that they won't be answered and leave... |
01:58.49 | b11d | yeah I know.. I hate that |
01:58.57 | b11d | its their loss though |
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02:46.09 | doolph | help |
02:46.15 | doolph | I am having this message after upgrade |
02:46.16 | doolph | Oct 21 21:44:56 VERBOSE[25104] logger.c: [cdr_pgsql.so]Oct 21 21:44:56 WARNING[25104] loader.c: libpq.so.4: cannot open shared object file: No such file or directory |
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02:48.13 | b11d | is anyone here familar with using MusicOnHold? |
02:48.38 | b11d | I can set the hold music I hear when someone else puts me on hold, but I'd like to define what music they hear when I put them on hold.. and I cant seem to do this with SetMusicOnHold |
02:54.59 | doolph | hi |
02:55.25 | b11d | hi |
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02:57.08 | Parvaresh | any recommendation on an embedded pc for asterisk? |
02:57.18 | Parvaresh | want to able to install full-length pci also on it |
02:57.27 | Parvaresh | as small and customize lookng as possible |
02:58.02 | napkin | hmmm soekris? should look it up. |
02:58.03 | SuPrSluG | doolph:do you have the postres module? if not go to /etc/asterisk/modules.conf and insert noload => cdr_pgsql.so |
02:58.35 | napkin | i guess soekris only does half-length |
02:58.43 | Parvaresh | yep |
02:58.49 | Parvaresh | and also not sure about its performance |
02:59.02 | napkin | well, full length pci and embedded might be a pipe dream? |
02:59.07 | Parvaresh | i am actually looking to make asterisk to a good looking hardware |
02:59.25 | napkin | does mini-itx have full length pci? |
03:00.18 | SuPrSluG | Parvaresh:http://www.x100p.com/products_5.htm |
03:00.20 | napkin | it's probably your best bet |
03:00.46 | Parvaresh | i hope you got my idea |
03:01.00 | Parvaresh | looking for something gives the customer more feeling of a PBX rather than a PC :) |
03:02.29 | napkin | there are probably some affordable mini-itx cases out there, too |
03:02.37 | napkin | that look less like big pcs |
03:05.41 | Parvaresh | expensive... |
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03:08.10 | Defraz | Anyone happen to know the feature in Sendmail.mc file to deny mail from domains that don't exist? |
03:09.45 | Defraz | why not go with an external spa3000 |
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04:52.23 | axscode | helow |
04:52.57 | axscode | bskg. |
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05:58.38 | stephane_ | jour |
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07:23.18 | vietasterisk | hi there |
07:24.25 | vietasterisk | hello , anyone worked with asterisk ACD hereeeeee |
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07:46.44 | bsdfreak | hi |
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09:42.45 | jgoo | hey guys, OT, but no network chan on here |
09:43.28 | jgoo | kinda on topic, as is about bandwidth, and asterisk. I have 4 adsl connections. I google for ADSL connection pooling, but found nothing. |
09:44.13 | jgoo | anyone have an idea how to pool these connections? or is it just some traffic shaping app to choose which connection to go through? (as I will have 4 ip's...) |
09:46.17 | jgoo | yeah... desperate measures |
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09:48.39 | qdk | jgoo: naah, it just always supprises me that novice ppl thinks that everyone on a IRC-channel are mind readers. |
09:49.08 | Qorky | can anyone help me with registering my g729 codecs please? |
09:50.11 | jgoo | qdk: mind readers how? |
09:51.15 | qdk | jgoo: dunno, havent meet anyone yet, but now that you are looking for one you could get back to me with an update. |
09:51.19 | Qorky | i can get the register program to run.. but just does nothing for a while then drops back to prompt. |
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09:53.20 | jgoo | qdk: I mean, why do you think that I think that everyone are mind readers :p |
09:53.44 | Qorky | can anyone help please? |
09:56.39 | qdk | jgoo: how many in here do you think knows about your hardware, OS and software? |
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10:26.18 | Qorky | can anyone get to www.digium.com ? I cant.. |
10:26.40 | Mavvie | telnet: connect to address 216.207.245.9: Connection refused |
10:26.40 | Nugget | telnet is eeeeeeevil! |
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10:50.39 | DarKnesS_WolF | what is the best billing system for asterisk ? |
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11:20.41 | hi365 | vm emails not getteing sent. how do i trobleshoot? |
11:21.19 | ramtha | look in you mail.log |
11:21.26 | eliXier | sendmail is installed? |
11:21.38 | eliXier | mailcmd = sendmail -t ? |
11:21.39 | ramtha | do you have "attache mail" option aktivated? |
11:26.45 | hi365 | ramtha: /var/log/maillog is blank |
11:27.53 | hi365 | ramtha: 201 => 8899,xxxxxx,xxxxxxxxx@gmail.com,,attach=yes|saycid=no|envelope=no|delete=yes |
11:28.14 | hi365 | eliXier: yes |
11:29.08 | hi365 | [root@asterisk1 ~]# service sendmail status |
11:29.08 | hi365 | sendmail (pid 3098 3090) is running... |
11:29.13 | hi365 | eliXier: |
11:29.20 | hi365 | [root@asterisk1 ~]# service sendmail status |
11:29.20 | hi365 | sendmail (pid 3098 3090) is running... |
11:30.57 | eliXier | try to test on ssh: |
11:31.07 | eliXier | sendmail -t your@mailaddy.com |
11:31.09 | eliXier | test |
11:31.11 | eliXier | . |
11:31.21 | TheMafia | Is the rtp the acutal audio in a sip to sip call? |
11:31.26 | eliXier | show if you receive the mail |
11:31.27 | TheMafia | 5060 is the control right? |
11:32.23 | *** join/#asterisk saftsack (n=saftsack@p54A7D890.dip.t-dialin.net) |
11:32.25 | *** join/#asterisk xlyz (n=xl@213-140-17-96.ip.fastwebnet.it) |
11:33.16 | hi365 | eliXier: sendmail -t nachmic@gmail.com test |
11:33.23 | hi365 | like that? |
11:33.28 | *** part/#asterisk xlyz (n=xl@213-140-17-96.ip.fastwebnet.it) |
11:33.46 | hi365 | it sort of hangs. how long should it take? |
11:34.11 | eliXier | nope |
11:34.13 | eliXier | start: |
11:34.22 | eliXier | sendmail -t xxxx@gmail.com |
11:34.23 | eliXier | test |
11:34.24 | eliXier | . |
11:34.27 | eliXier | end |
11:34.52 | qdk | TheMafia: yes, RTP carries the media and SIP is the signaling. |
11:35.29 | *** join/#asterisk Mattwj2005 (n=Matt@user-12l3nck.cable.mindspring.com) |
11:35.43 | Mattwj2005 | hey guys :) |
11:36.04 | TheMafia | qdk what is the standard port/range for rtp? |
11:36.24 | hi365 | eliXier: no email recived! |
11:36.26 | Mattwj2005 | I believe the range for sip is 10000-20000 |
11:36.51 | eliXier | now, it's a problem of sendmail, not asterisk |
11:37.12 | qdk | TheMafia: whatever you pick. |
11:37.17 | TheMafia | thanks |
11:37.31 | hi365 | eliXier: good point. any advice anyway? |
11:38.06 | eliXier | hi365: reinstall/reconfigure sendmail again... the standard-configuration of sendmail is ok |
11:38.19 | *** join/#asterisk DarKnesS_WolF (n=wolf@212.103.170.135) |
11:38.36 | hi365 | eliXier: thanks ill try |
11:38.36 | eliXier | anotherway.. gmail has a spamfilter? |
11:38.46 | eliXier | np |
11:39.03 | hi365 | it has worked in the past so i dont think thats the problem |
11:39.17 | eliXier | ok |
11:40.47 | coppice | anyone around who has used the codec simply called ADPCM in asterisk? |
11:41.03 | *** join/#asterisk Ox0F0-0FF (n=pierre@201.38.238.66) |
11:45.28 | hi365 | eliXier: /var/log/sendmail here. although its not asterisk, would please have a look? http://pastebin.ca/215099 |
11:46.05 | TheMafia | I have a vpn connection between two networks with sip configured on both sides, phones ring as should however there is no audio, should I have to do anything netowrk related since I am using a vpn? |
11:47.40 | *** join/#asterisk flot (n=flot@87.251.134.36) |
11:49.40 | flot | hi all. I use asterisk 1.4 (SVN) In last version NOT WORKING transfer. Say: [Oct 22 15:42:16] WARNING[4718]: res_features.c:771 builtin_atxfer: Did not read data. |
11:49.40 | flot | <PROTECTED> |
11:50.29 | ramtha | TheMafia: what vpn you use? ipsec? |
11:51.09 | ramtha | TheMafia: sounds that signaling goes over the vpn tunnel but the rtp goes directly to the endpoint ips? |
11:51.28 | TheMafia | yes ipsec |
11:51.45 | ramtha | use tcpdump to see where the paces go |
11:51.50 | ramtha | or etherreal |
11:52.10 | TheMafia | 192.168.15.11.6004 > 192.168.0.30.8000 is that a typical rtp port or is there no range at all? |
11:52.36 | ramtha | you can set rtp ports in rtp.conf of asterisk |
11:52.39 | ramtha | 8000 is typical |
11:52.53 | ramtha | and you see the error? |
11:53.23 | ramtha | can you ping from 192.168.15.11 to 192.168.0.30? |
11:54.19 | TheMafia | yes I can |
11:54.48 | ramtha | no look where 192.168.0.30 send its rtp stream |
11:56.57 | TheMafia | 192.168.0.30 is a softphone an I have no mic setup so I assumed there was no stream to send, I cannot hear any audio from 192.168.15.11, I am duiling a fax machine |
11:57.51 | *** join/#asterisk mrg82 (n=na@dsl82-163-126-23.as15444.net) |
12:00.49 | TheMafia | I can plug in my grandstream phone but it seems to do the same thing and I can see traffic in both directions then |
12:02.17 | *** join/#asterisk brif8 (n=brif8@67.78.24.178) |
12:03.57 | brif8 | Hi All, reading I see that a "SIP Proxy Server can request to stay in the communication path" how is this done within the * env. Can phone A be set for call1 to be in the loop and call2 out of the loop or what ?? |
12:05.23 | *** join/#asterisk AbsTradELic (n=vldmr@201.79.188.89) |
12:06.26 | AbsTradELic | good times for all ! |
12:06.39 | AbsTradELic | X:) |
12:06.55 | AbsTradELic | SuPrSluG: hi ! |
12:08.10 | TheMafia | when I dump the grandstream sip phone this is what I see over and over |
12:08.12 | TheMafia | 07:07:35.001548 IP 192.168.15.11.6004 > 192.168.0.185.5004: UDP, length: 172 |
12:08.32 | TheMafia | 07:07:34.972617 IP 192.168.0.185.5004 > 192.168.15.11.6004: UDP, length: 172 |
12:08.44 | TheMafia | Is that rtp to and from my grandstream? |
12:12.47 | festr__ | rtp |
12:12.56 | festr__ | try to use tethereal |
12:14.16 | *** join/#asterisk DataCompBoy (n=datacomp@217.151.230.182) |
12:14.19 | TheMafia | I can't from there, so if the rtp is getting to and from why would there be no audio? |
12:14.40 | DataCompBoy | Hi everybody! Is digium.com down? Or that only for me? |
12:15.26 | *** join/#asterisk kio (n=kio@ool-4577ae5e.dyn.optonline.net) |
12:17.05 | TheMafia | I am confident that the codecs will mate up |
12:18.22 | *** join/#asterisk Druken (n=jdumais@CPE000854ddcdb1-CM00137189cb0c.cpe.net.cable.rogers.com) |
12:18.26 | *** part/#asterisk brif8 (n=brif8@67.78.24.178) |
12:20.59 | *** join/#asterisk wl0 (n=ge@88-196-80-95-dsl.krw.estpak.ee) |
12:23.51 | *** join/#asterisk kristalino (n=kristali@cl-157.dub-01.ie.sixxs.net) |
12:25.20 | hi365 | DataCompBoy: same here |
12:33.14 | AbsTradELic | i have all packages installed and loaded now! |
12:33.23 | AbsTradELic | is allright ! |
12:33.36 | AbsTradELic | i'm a beginner |
12:33.50 | AbsTradELic | and now I'll adjust the /etc/asterisk files |
12:35.26 | AbsTradELic | I didn't have nothing more that my computer and softphone x-lite installed... no voip hardwares |
12:36.03 | DataCompBoy | edit /etc/sip.conf and /etc/extensions.ael or /etc/extensions.conf |
12:36.18 | DataCompBoy | i mean, {edit (/etc/sip.conf and (/etc/extensions.ael or /etc/extensions.conf))} |
12:36.37 | AbsTradELic | hum... ok |
12:36.50 | AbsTradELic | wait a few |
12:38.22 | *** join/#asterisk axscode (n=axscode@124.217.42.206) |
12:39.32 | AbsTradELic | DataCompBoy: ok... I view the files on /etc/asterisk |
12:40.35 | DataCompBoy | edit them :) |
12:41.01 | DataCompBoy | carefully read comments and asterisk pages @ http://www.voip-info.org/wiki-Asterisk |
12:45.13 | *** join/#asterisk Shedoks (n=nn@82.117.206.193) |
12:45.18 | Shedoks | hi |
12:45.47 | Shedoks | i need a little help for asterisk |
12:46.35 | DataCompBoy | well... try to ask |
12:49.57 | Shedoks | :) |
12:50.41 | Shedoks | ok i want to make asterix work just over the internet right now, i don't need voip phones or connecting with the fixed telephony |
12:50.42 | DataCompBoy | ... welll? |
12:50.49 | Shedoks | what i s confusing me is |
12:51.08 | Shedoks | do i have to make sip.conf for every user i made ? |
12:51.24 | Shedoks | or i can put every user into [username] brackets |
12:51.56 | DataCompBoy | you put all your users (if you have not much users) in sip.conf |
12:52.07 | DataCompBoy | every user info after [username] |
12:52.50 | DataCompBoy | define at least type, username, secret or md5secret, host and nat |
12:53.42 | DataCompBoy | (hm... nat can be global, if all your users possible behind nat) |
12:54.09 | Shedoks | well for testing purposes it will be |
12:54.21 | Shedoks | but how do i make works with nat |
12:54.34 | DataCompBoy | for testing just put several blocks to sip.conf and bee happy |
12:54.39 | Shedoks | becouse asterisk is behind nat and i would like to use it globaly later |
12:54.54 | DataCompBoy | asterisk behind nat? how -- did you port forward UDP to it? |
12:55.19 | Shedoks | well i plan to do that |
12:55.24 | Shedoks | which one 5060? |
12:55.27 | Shedoks | or 8000 |
12:55.29 | DataCompBoy | All :))) |
12:55.55 | DataCompBoy | the best is forward all UDP to asterisk, but will be success only 5060 and RTP ports defined in rtp.conf |
12:56.39 | Shedoks | ok thanks :) |
12:56.44 | Shedoks | just one more thing |
12:56.51 | Shedoks | how to add user trought CLI |
12:57.01 | DataCompBoy | no way |
12:57.16 | DataCompBoy | edit sip.conf and do in cli 'sip reload' |
12:57.41 | *** join/#asterisk DarKnesS_WolF (n=wolf@196.218.72.21) |
12:57.57 | Shedoks | thanks man :) |
12:58.34 | robin_sz | meep? |
12:59.14 | DataCompBoy | robin_sz: meep?! |
12:59.39 | robin_sz | so, this asterisk thing .. and the CLI. Did one of you steal my "Dial" command? |
13:00.34 | robin_sz | I dont think I did ./configure --dont-bother-with-a-dail-command-for-CLI |
13:01.22 | robin_sz | does it omit the CLI dial command from compile of it cant findany suitable local audio hardware? |
13:01.53 | DataCompBoy | robin_sz: nope, it compile dial command always, since it can dial on any channel |
13:01.56 | DataCompBoy | even on LOCAL/ |
13:02.20 | robin_sz | DataCompBoy, ok, so what do I have to do to enable it then? |
13:02.54 | TheMafia | If I have determined that rtp data is getting to and from both sides, and I still only get rings etc on the phone and no audio, what else shoudl I look at? |
13:03.20 | DataCompBoy | robin_sz: look at build log, and see what problem was when it tried to compile |
13:03.49 | robin_sz | I dont thin I had any problems with the compile at all ... |
13:05.00 | DataCompBoy | TheMafia: try to add nat=yes to phone configuration |
13:05.00 | DataCompBoy | TheMafia: i have saw that some times |
13:05.00 | DataCompBoy | robin_sz: look at build tree, is dial.o there? :) |
13:05.02 | DataCompBoy | robin_sz: what os? |
13:05.09 | robin_sz | Linux, obviously ;) |
13:05.15 | Rhizome | Anyone know why DTMF wouldnt work calling from asterisk to asterisk to PSTN? even with IAX it doesnt work. |
13:05.19 | DataCompBoy | robin_sz: distro? |
13:05.28 | DataCompBoy | robin_sz: what asterisk sources? |
13:05.31 | robin_sz | debian |
13:05.38 | robin_sz | latest bristuffed |
13:05.48 | DataCompBoy | robin_sz: how you build? dpkg-buildpackage ? |
13:06.03 | robin_sz | but its the same with other variants eg 1.2.9.1 |
13:06.08 | robin_sz | or 1.2.7 |
13:06.26 | robin_sz | obvioulsy the Dial() in extensions.conf works fine |
13:06.32 | robin_sz | its just from the CLI .. |
13:06.37 | DataCompBoy | robin_sz: Stop. |
13:06.45 | DataCompBoy | robin_sz: there NO cli command dial() ! |
13:07.12 | robin_sz | you certain? |
13:07.24 | DataCompBoy | there Dial() command from cli here only for chan_oss :) |
13:07.34 | DataCompBoy | when you use soundcard of asterisk server. |
13:07.42 | robin_sz | so ... |
13:08.02 | robin_sz | when I said: |
13:08.04 | robin_sz | does it omit the CLI dial command from compile of it cant findany suitable local audio hardware? |
13:08.07 | robin_sz | the answer was: |
13:08.26 | robin_sz | "yeah thats right, if theres no local audio hardware, you cant dial from the CLI" |
13:08.31 | robin_sz | ??? |
13:08.32 | DataCompBoy | robin_sz: there CLI dial() command is non-standard, and only suitable for chan_oss. |
13:09.17 | DataCompBoy | if you want to initiate call, read http://www.voip-info.org/wiki/view/Asterisk+auto-dial+out or http://www.voip-info.org/wiki/view/Asterisk+Manager+API+Action+Originate |
13:09.32 | robin_sz | OK |
13:10.09 | robin_sz | hmmm |
13:10.16 | robin_sz | neither of those seem useful |
13:10.49 | DataCompBoy | you are not right -- last one very useful and practical. |
13:11.25 | DataCompBoy | never tried first, since always use AMI :) |
13:11.42 | robin_sz | uses the manager API, |
13:12.01 | *** join/#asterisk ukh (i=ukh@visionary.svansen.se) |
13:12.04 | robin_sz | so, from the CLI I type what ? |
13:13.06 | wl0 | hi, advice me any cheap but stable and quality sip phones for work with asterisk |
13:13.09 | wl0 | please |
13:13.23 | DataCompBoy | robin_sz: what you want to do? how do you imagine your doints? |
13:13.54 | robin_sz | normally, in testing, I just type dial xxxx@<context> and watch the progress of the call, so I can see what happens from the CLI, watch timeouts. routings etc |
13:14.04 | robin_sz | I dont need any audio |
13:14.24 | robin_sz | I think the other boxes I have used must have had oss support by accident |
13:15.12 | robin_sz | the manager API is nto really an option, as the things you can connect to it all take a day or more to configure from what ive seen so far. |
13:15.26 | robin_sz | I just need a simple way of initiating a call and monitoring the progress |
13:16.20 | DataCompBoy | robin_sz: so, you want to dial xxxx@<context>, context/exten/prio right? |
13:16.28 | robin_sz | uh uhuh |
13:16.57 | DataCompBoy | robin_sz: you can just create test file as Asterisk+auto-dial+out. thats easy |
13:18.08 | TheMafia | is port 5060 that is used for control udp or tcp? |
13:18.35 | DataCompBoy | TheMafia: 5060 udp is control port, and 10000-20000 (by default) are RTP ports. |
13:18.46 | TheMafia | thanks |
13:20.17 | *** join/#asterisk eltech (n=eltech--@ool-457c9421.dyn.optonline.net) |
13:20.37 | robin_sz | hmmm ... |
13:21.16 | robin_sz | I guess I'll need some sort of script then to create call files for the various extensions and contexts |
13:21.50 | robin_sz | ah, stuff it, I'll just go and buy a sound card tomorow and stick it in |
13:22.22 | *** join/#asterisk eltech (n=eltech--@ool-457c9421.dyn.optonline.net) |
13:26.23 | Druken | robin_sz: what do you need an audio card for? |
13:27.06 | robin_sz | Druken, to test extensions.conf |
13:27.17 | DataCompBoy | robin_sz: why not just run oss for empty ? :) you not need hardware to run it. |
13:27.38 | robin_sz | explain that sentence please |
13:27.45 | robin_sz | why not just run oss for empty |
13:27.48 | robin_sz | ?? |
13:27.52 | DataCompBoy | robin_sz: also, it's pretty easy to write Dial() app that will do connect of origitated and local extension |
13:28.14 | DataCompBoy | robin_sz: you don't need hardware to run OSS mixer. |
13:28.17 | DataCompBoy | and chan_oss too |
13:28.23 | Druken | robin_sz: why not just use a phone? |
13:28.48 | DataCompBoy | robin_sz: also, Druken right, why not use simple softphone |
13:29.00 | robin_sz | Druken: I thought phones where restricted in context? |
13:29.36 | Druken | uhmm... asterisk is a telephone system... no?? |
13:29.45 | Druken | how do you think everyone else will access the system ? |
13:29.53 | robin_sz | so can I call xxxx@incoming and it will originate a call and place it in the incoming context, as it it had come from the PSTN? |
13:31.00 | robin_sz | Druken, this is called "testing" ... before it goes live, I like to test ... normally I do stuff like "dial xxxx@incoming" and watch how the call prgresses through the timeouts etc |
13:31.09 | Druken | make an extension, something like exten => 987,1,goto(xxx,local,1) |
13:31.16 | robin_sz | make sure the logic is correct |
13:31.38 | DataCompBoy | robin_sz: do, you already used Dial cli command? |
13:31.49 | robin_sz | DataCompBoy, on other systems yes, all the time |
13:31.50 | DataCompBoy | robin_sz: why then not load chan_oss and use it now ? |
13:32.03 | robin_sz | because I dont understand that statement? |
13:32.10 | Druken | testing is usually a good idea... me, i edit my system WHILE it's live.. hehehe thankfully i'm very good at logic |
13:33.17 | DataCompBoy | robin_sz: load chan_oss, and you will be happy |
13:33.48 | robin_sz | modules/chan_oss: cannot open shared object file: No such file or directory |
13:33.54 | robin_sz | next? |
13:34.14 | DataCompBoy | hm. so, you have not build it? |
13:34.28 | robin_sz | probably not ... |
13:34.34 | robin_sz | no sound support on this machine .. |
13:34.35 | DataCompBoy | then, build it! :) |
13:34.45 | DataCompBoy | install liboss, that enough |
13:34.50 | robin_sz | ahhh |
13:34.57 | robin_sz | now you begin to make sense :) |
13:37.47 | robin_sz | libossp-sa12 - Abstraction library for the Unix socket API |
13:37.51 | robin_sz | that one? |
13:38.15 | DataCompBoy | nope |
13:38.41 | robin_sz | libossp-uuid12 - OSSP uuid ISO-C - shared library ?? |
13:39.03 | DataCompBoy | install audiooss |
13:39.13 | DataCompBoy | (if you talk about debian) |
13:39.38 | robin_sz | i do ... |
13:39.43 | robin_sz | they hide things |
13:39.45 | robin_sz | :) |
13:40.46 | robin_sz | obviously, they dotn let that package show up for a search of liboss .. that wwould be too easy :) |
13:40.46 | *** join/#asterisk teknoprep (n=tekon@unaffiliated/teknoprep) |
13:44.36 | *** join/#asterisk eltech (n=eltech--@ool-457c9421.dyn.optonline.net) |
13:45.30 | *** join/#asterisk prttp (i=Ftv@159.Red-83-50-32.dynamicIP.rima-tde.net) |
13:48.47 | *** join/#asterisk markdrago (n=mdrago@ool-182d1b14.dyn.optonline.net) |
13:48.50 | TheMafia | i can see traffic traversing the nat on the two rtp ports defined on both ends, however there is still only rings and no audio. What shoudl I look at? if a codec cannot be negotiated the ring would or would not take place? |
13:49.25 | DarKnesS_WolF | TheMafia: in the sip.conf canreinvite=no |
13:49.56 | DarKnesS_WolF | TheMafia: for each extention |
13:50.10 | TheMafia | pcmu is the same as g.711 right? |
13:55.19 | DarKnesS_WolF | i'm not sure about that i know that ulaw =G.711U |
13:55.47 | Qwell | TheMafia: yes, I'm pretty sure it is |
13:58.53 | DarKnesS_WolF | Qwell: so pcmu is g.711 and it is ulaw and alaw? |
13:59.06 | Qwell | no |
13:59.10 | TheMafia | pcma is a law g.711 I think |
13:59.22 | Qwell | g711u is ulaw is pcmu, g711a is alaw is pcma |
13:59.36 | DarKnesS_WolF | ah goti t |
13:59.39 | DarKnesS_WolF | thx Qwell :-) |
14:01.12 | *** join/#asterisk wunderkin (i=kev@ip72-208-3-42.ph.ph.cox.net) |
14:06.39 | DarKnesS_WolF | what is the best asterisk billing system ? |
14:09.18 | *** join/#asterisk linagee (n=linagee@unaffiliated/linagee) |
14:09.30 | Qwell | ~best |
14:09.31 | jbot | best for what? please define what you mean by "best" Gloria Gaynor! Tina Turner! Aretha Franklin! Men without Hats! Women without Hats! Flock of Seagulls!, or fvwm! Women without clothes! |
14:09.45 | *** join/#asterisk Seggy (i=rbutler@tsss.org) |
14:09.50 | *** join/#asterisk Inez (i=faceoff@devel4.net) |
14:09.53 | Inez | Hi |
14:09.59 | hegemoOn | plop |
14:10.37 | Inez | Is it good idea to use app_sql_postgres to receive data from pg database in dialplan or maybe better way is to use AGI and perl/python script? |
14:14.00 | *** join/#asterisk SuPrSluG (n=SuPrSluG@pool-71-241-169-13.buff.east.verizon.net) |
14:14.04 | Qwell | Inez: use func_odbc |
14:14.20 | SuPrSluG | hell |
14:14.23 | SuPrSluG | hello |
14:14.47 | SuPrSluG | although i am in zaptel hell right now |
14:15.50 | DarKnesS_WolF | SuPrSluG: TDM400P ? |
14:15.57 | SuPrSluG | i keep getting a message when trying to call zt_handle_event: Ring/Off-hook in strange state 6 on channel 4 |
14:16.00 | SuPrSluG | yep |
14:16.34 | Inez | Qwell but its only possible to select one row and one field? |
14:16.46 | Qwell | Inez: one row, as many fields as you tell it |
14:18.09 | DarKnesS_WolF | SuPrSluG: try from command like ztcfg -vvvvvvv |
14:18.37 | Inez | Qwell how, I cant find it on http://www.voip-info.org/wiki/index.php?page=Asterisk+func+func_odbc Is it standard in asterisk 1.2? |
14:18.45 | Qwell | no |
14:19.00 | SuPrSluG | i look in dmesg and proc see this http://pastebin.ca/215241 |
14:19.06 | Qwell | there is a backport of it though on svncommunity.digium.com |
14:20.35 | DarKnesS_WolF | SuPrSluG: what is the output from ztcfg -vvvv ? |
14:21.12 | SuPrSluG | i have a single fxo and a tdm04b i removed the wcfxo module but it want to keep the 1st port |
14:22.03 | Inez | Qwell I should configure sql resource in res_config_odbc.conf or res_odbc.conf, right? |
14:22.14 | AbsTradELic | SuPrSluG: Hi! |
14:22.19 | SuPrSluG | hi |
14:22.26 | AbsTradELic | X:) |
14:23.07 | Druken | func_odbc is FUN :) |
14:23.15 | SuPrSluG | DarKnesS_WolF:http://pastebin.ca/215250 |
14:23.30 | Druken | i gave up on it, and use realtime.... |
14:23.54 | *** join/#asterisk Vahram (n=VX@client-231-114.xter.net) |
14:24.19 | SuPrSluG | as you can see from proc it still thinks the single fxo port is 1 and using only the 1st 3 on the tdv04b board |
14:25.42 | DarKnesS_WolF | SuPrSluG: u have 4 FXO ports ? |
14:25.53 | *** join/#asterisk ESCulapio__ (n=ESCulapi@188stb68.codetel.net.do) |
14:26.51 | SuPrSluG | yes 1 single and a 1 4 port. it won't let go of the single and zap channel 1 |
14:28.13 | SuPrSluG | so a total of 5 fxo. no line is plugged into the single. |
14:28.40 | DarKnesS_WolF | 5 FXO !? |
14:28.54 | SuPrSluG | yes |
14:28.58 | DarKnesS_WolF | how come ? |
14:29.15 | DarKnesS_WolF | i know that the card has 4 places for modules |
14:29.52 | Inez | i have problem with svncommunity.digium.com, is it work ok for you? for making co |
14:30.34 | SuPrSluG | 2 cards a single and a 4 port |
14:30.57 | DarKnesS_WolF | ah so u have 2 cards |
14:31.01 | DarKnesS_WolF | hm |
14:31.01 | SuPrSluG | yes |
14:32.07 | DarKnesS_WolF | SuPrSluG: ok check ur /etc/zaptel.conf |
14:32.08 | SuPrSluG | and the single is grabbing zap channel 1 and i don't want it to |
14:32.28 | SuPrSluG | want me to post on pastebin? |
14:34.15 | AbsTradELic | ok |
14:34.47 | AbsTradELic | SuPrSluG: I have dificulties to understand english! |
14:34.49 | ManxPower | SuPrSluG: what specific cards do you have? |
14:34.53 | AbsTradELic | X;) |
14:35.06 | AbsTradELic | ok |
14:35.08 | ManxPower | and what modules are on those cards? |
14:35.14 | SuPrSluG | generic clone and tdm04b |
14:35.33 | SuPrSluG | all fxo |
14:35.49 | ManxPower | SuPrSluG: so a Generic X100P and a TDM400P with 4xFXO (RED) modules? |
14:36.14 | AbsTradELic | can I put only one ip addres on sip.conf like this: 'localnet=192.168.1.1/255.255.255.0' ? |
14:36.40 | AbsTradELic | its correct ? |
14:36.41 | Qwell | AbsTradELic: That would be technically invalid |
14:37.04 | Qwell | 192.168.1.0/255.255.255.0 or for just one single IP, 192.168.1.1/255.255.255.255 |
14:37.36 | SuPrSluG | the generic is in red (has no line attached) and TDM04B has 4 lines attached. the generic won't give up zap channel 1 |
14:37.54 | ManxPower | SuPrSluG: define "won't give up" |
14:38.13 | AbsTradELic | Qwell: ok... the asterisk work like a server ? |
14:38.21 | Qwell | huh? |
14:38.27 | *** join/#asterisk delmar (n=delmar@ip-58-28-158-154.ubs-dsl.xnet.co.nz) |
14:38.37 | asdx | lol |
14:39.06 | DarKnesS_WolF | Qwell: i think AbsTradELic means it can run as an init.d |
14:39.12 | ManxPower | SuPrSluG: also what is the order of the drivers loading? wcfxo wctdm or wctdm wcfxo ? |
14:39.13 | DarKnesS_WolF | AbsTradELic: yes u can run it like mysql |
14:39.16 | SuPrSluG | so if i pull the wcfxo module and ztcfg -v it says 4 channels configured. i look in proc and http://pastebin.ca/215241 |
14:40.08 | AbsTradELic | Qwell: in one local network, I just need to put asterisk only one machine and be usefull to others computers clients ? |
14:40.13 | SuPrSluG | do i need to physically pull the single fxo to correct this? |
14:40.24 | ManxPower | SuPrSluG: you should not need to. |
14:40.50 | ManxPower | RED in the output means "no line connected" |
14:40.59 | AbsTradELic | ok |
14:41.36 | ManxPower | SuPrSluG: I still don't understand that the PROBLEM is. Can you not dial out, can you not receive calls, etc? |
14:41.37 | SuPrSluG | ztcfg will configure 5 channels. when i zap show channel 1 it is alway in alarm |
14:41.55 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.216.2) |
14:41.56 | ManxPower | SuPrSluG: correct. if a phone line is not plugged into the X100P it will show an alarm. |
14:42.08 | SuPrSluG | can't receive |
14:42.18 | ManxPower | you should not be alarmed by an alarm if you are expecting an alarm. |
14:42.45 | *** join/#asterisk eltech (n=eltech--@ool-457c9421.dyn.optonline.net) |
14:42.48 | ManxPower | SuPrSluG: so someone calls in on a line connected to channel 2 (port 1 on the TDM400P) and nothing shows up on the console? |
14:43.18 | *** join/#asterisk Iamnacho (i=Iamnacho@ip68-103-140-124.ks.ok.cox.net) |
14:43.39 | SuPrSluG | yes. but when i configure it to use only the tdm04b it shows zap channel 1 in red alarm and there is a line plugged in |
14:44.00 | ManxPower | SuPrSluG: then the line is not working. Can you plug a normal analog phone into that line and get a dialltone? |
14:44.43 | Inez | Qwell can you tell me how to select two vwo fields from database (in one row). |
14:45.07 | SuPrSluG | here's what shows Executing BackGround("Zap/4-1", "night-greet-eea") in new stack |
14:45.08 | Qwell | select foo, bar from table where baz='blah' |
14:45.09 | SuPrSluG | <PROTECTED> |
14:45.10 | SuPrSluG | <PROTECTED> |
14:45.37 | ManxPower | SuPrSluG: that is channel FOUR, you were talking about chanel ONE |
14:46.10 | ManxPower | (09:43:21) ManxPower: SuPrSluG: then the line is not working. Can you plug a normal analog phone into the line plugged into port ONE and get a dialltone? |
14:46.18 | SuPrSluG | you can't hear the message and when you hangup it continues |
14:46.28 | ManxPower | the ring/offhook message is normal and not a problem. |
14:46.36 | ManxPower | SuPrSluG: what country are you in>? |
14:47.03 | Inez | Qwell but how to using func_odbc |
14:47.13 | ManxPower | I'm sorry, but I cannot help you. You are trying to diagnose the issues, but not following my advice or questions. |
14:49.37 | SuPrSluG | sorry that's one of the other numbers. when i call channel one it just rings. unless i configure the single card and 5 channels the zap channel 2 gets the same message |
14:49.47 | SuPrSluG | us |
14:50.02 | SuPrSluG | sorry i had to ssh back in |
14:51.22 | SuPrSluG | Manxpower i'm not phyically there. Although i can call to get access to the building |
14:51.33 | *** join/#asterisk kilobit2001 (n=locid@206-248-134-27.dsl.teksavvy.com) |
14:52.15 | ManxPower | you have a wiring problem. You need to be there to diagnose it. bring an analog phone so you can test the lines outside of Asterisk, |
14:53.23 | SuPrSluG | we have a butt set and anolog phones are attached to a splitter for emergencies |
14:55.11 | SuPrSluG | why would it try to set rbs . in dmesg i get this -> zt_rbs: Tried to set RBS hook state 0 on channel WCFXO/0/0 while span WCFXO/0 lacks rbsbits or hooksig function |
14:55.25 | ManxPower | *nod* Anytime you see a RED Alarm you can assume that means "I don't see the voltage I would expect on a phone line" |
14:55.40 | SuPrSluG | robbed bit is for T's right? |
14:56.00 | ManxPower | SuPrSluG: I would assume the driver tries to set the RBS bits on all Digium cards, but only T-1/E-1 cards support that function. |
14:56.09 | *** join/#asterisk teknoprep (n=tekon@unaffiliated/teknoprep) |
14:56.15 | kilobit2001 | hi |
14:56.23 | ManxPower | put your /etc/zaptel.conf on pastebin and I'll glance at it. |
14:56.27 | kilobit2001 | does asterisk support dial by name? |
14:56.40 | SuPrSluG | so that'll affect all the other numbers too? |
14:56.45 | ManxPower | kilobit2001: voicemail does. |
14:56.50 | ManxPower | as does the Directory() app |
14:57.05 | ManxPower | SuPrSluG: I don't believe the RBS message is an issue. |
14:58.16 | *** join/#asterisk kio (n=kio@ool-4577ae5e.dyn.optonline.net) |
14:58.45 | kilobit2001 | is the following valid: _xxxxxxxxxx,1,goto(${exten},1,1) |
14:59.35 | *** join/#asterisk saftsack (n=oliver@p54A7F8DD.dip.t-dialin.net) |
15:01.34 | ManxPower | only if your context is ${EXTEN} |
15:01.55 | ManxPower | The format is Goto([context],[extenation],priority) |
15:02.09 | ManxPower | context and extenstion are optional, the priority is not. |
15:03.11 | *** join/#asterisk af_ (n=af@ip-173-17.sn1.eutelia.it) |
15:04.41 | ManxPower | kilobit2001: of course _xxxxxxxxxx,1,goto(${exten},1) would cause a loop and prolly make Asterisk crash |
15:05.19 | kilobit2001 | is there a limit on the amount of contexts allowed? |
15:05.34 | ManxPower | kilobit2001: I don't think so, |
15:05.55 | ManxPower | there really isn't any need to have zillions of contexts |
15:05.58 | kilobit2001 | in this setup, there'll be a jump to a different context, based on the number dialed. |
15:06.17 | ManxPower | kilobit2001: why? |
15:06.37 | kilobit2001 | manx-- how would you implement a multiple menu ivr system? |
15:07.10 | kilobit2001 | lets say 5 users, each with 4 nested menus. |
15:08.52 | *** join/#asterisk Rhizome (n=rhizome@rhizome.boldlygoingnowhere.org) [NETSPLIT VICTIM] |
15:08.52 | *** join/#asterisk smurf (n=smurf@debian/developer/smurf) [NETSPLIT VICTIM] |
15:08.52 | *** join/#asterisk kore (i=kore@mindwipe.org) [NETSPLIT VICTIM] |
15:08.57 | kilobit2001 | anyway to do this within one context? |
15:09.13 | ManxPower | 5 users? |
15:09.24 | ManxPower | I understand the 4 nested menus. |
15:09.57 | TheMafia | i can see traffic traversing the nat on the two rtp ports defined on both ends, however there is still only rings and no audio. What shoudl I look at? if a codec cannot be negotiated the ring would or would not take place? I am not using asterisk, but I thought that this would be a common problem |
15:10.36 | ManxPower | TheMafia: it's not a common problem |
15:10.44 | kilobit2001 | 5 users each with their own menus.. incoming calls are jumped to different context based on dialed numbers... |
15:10.47 | ManxPower | it happens with Asterisk, but for a variety of reasons |
15:11.07 | *** join/#asterisk hi365 (n=hi365@bzq-167-158.dsl.bezeqint.net) |
15:12.49 | DarKnesS_WolF | TheMafia: i told u |
15:12.54 | DarKnesS_WolF | canreinvite= no |
15:13.04 | DarKnesS_WolF | TheMafia: server and the 2 nodes in same lan ? |
15:13.12 | ManxPower | kilobit2001: so, you mean something like this pastebin. [incoming] is where calls from the PSTN come into. http://pastebin.ca/215297 |
15:13.25 | ManxPower | DarKnesS_WolF: he said he's not using Asterisk |
15:13.35 | DarKnesS_WolF | ManxPower: what he is using !? |
15:13.41 | ManxPower | DarKnesS_WolF: no idea |
15:13.44 | DarKnesS_WolF | ManxPower: sorry i didn't notice and why he is here then ! |
15:13.56 | DarKnesS_WolF | oh i have to go to break the fast |
15:14.01 | DarKnesS_WolF | it almost sunset |
15:14.04 | ManxPower | DarKnesS_WolF: We have gotten a repuation of being able to fix all things VoIP. |
15:14.16 | ManxPower | I've heard rumors that we use dark magic to accomplish this. |
15:14.22 | DarKnesS_WolF | ManxPower: haha good for us :D |
15:14.24 | DarKnesS_WolF | lol |
15:14.30 | DarKnesS_WolF | ManxPower: leave the goat alone :P |
15:14.39 | DarKnesS_WolF | ok gtg will be in 1 hour or so i'm starving |
15:15.12 | *** join/#asterisk Qwell (n=north@unaffiliated/qwell) |
15:15.12 | *** mode/#asterisk [+o Qwell] by ChanServ |
15:15.21 | wmandra | hey all, is anyone else having trouble getting their cisco phones to display the correct time today??? |
15:15.28 | ManxPower | kilobit2001: yes, your current design will work, but it is a bad design and confusing. Manually put in a Goto for each number. |
15:16.28 | ManxPower | wmandra: It would not surprise me. Apparently some daylight savings functions think that October will NEVER have 5 sundays. |
15:16.43 | wmandra | that what it's looking like |
15:16.55 | kilobit2001 | manx-- that example has one context per user. |
15:16.57 | ManxPower | wmandra: any chance on updating the cisco firmware? |
15:17.14 | ManxPower | kilobit2001: stop using the word "user". User means "person". |
15:17.17 | wmandra | maybe next weekend, but not today |
15:17.35 | wmandra | i really should though, i'm still using 6.3 |
15:18.40 | ManxPower | kilobit2001: what it looks like is that you are trying to set up IVRs for each company or department. |
15:18.51 | ManxPower | wmandra: by next weekend it should no longer be a problem |
15:18.56 | ManxPower | well, at least until 2011 |
15:19.49 | TheMafia | DarKnesS_WolF, I am not using asterisk so I can't set canreinvite=no, they are not in the same lan |
15:20.19 | kilobit2001 | manx-- yes. and to do that, the only way I have found is to create multiple contexts for called number. |
15:20.19 | wmandra | lol |
15:20.29 | ManxPower | kilobit2001: correct. |
15:27.58 | saftsack | SuPrSluG, are you here? |
15:29.40 | *** join/#asterisk lorinc (n=ang@caracas-2158.adsl.interware.hu) |
15:40.50 | robin_sz | is it possible to mute a warning in the CLI ?? |
15:41.08 | robin_sz | I get this: |
15:41.10 | robin_sz | Oct 22 16:40:23 WARNING[4871]: chan_zap.c:2506 pri_find_dchan: No D-channels available! Using Primary channel 4 as D-channel anyway! |
15:41.21 | robin_sz | every second. pian in the ass. |
15:42.03 | robin_sz | it will (hopefully) go away when my telco connects the ISDN |
15:42.40 | SuPrSluG | yes |
15:43.10 | ManxPower | robin_sz: unload chan_zap.so |
15:44.06 | SuPrSluG | ManxPower: got it to work by pulling wcfxo out of /etc/mod stuff. it finally let go of that 1st card and everything is working |
15:44.35 | SuPrSluG | saftsack:yes |
15:44.49 | robin_sz | ManxPower, not a bad plan ... |
15:45.34 | robin_sz | ManxPower, of course, thats killed the X100P as well, but hey .. its shut the damn thing up :) |
15:45.57 | *** part/#asterisk markdrago (n=mdrago@ool-182d1b14.dyn.optonline.net) |
15:46.35 | ManxPower | robin_sz: you can remove the pro channels from /etc/asterisk/zapata.conf |
15:47.31 | ManxPower | pro == pri |
15:50.31 | ManxPower | SuPrSluG: in that case, I'll be you are sharing IRQs |
15:50.35 | ManxPower | cat /proc/interrupts |
15:52.16 | SuPrSluG | yeah but the 2 cards aren't. 10: 808758 XT-PIC wctdm, VIA8233 |
15:52.18 | SuPrSluG | <PROTECTED> |
15:53.08 | SuPrSluG | ManxPower,what the best way to make sure they're on their own irq |
15:53.43 | SuPrSluG | eth0 and wcfxo are on 10 |
15:54.52 | saftsack | SuPrSluG, how many bogomips do you have with your box? or do you know how many bogomips are needed? |
15:55.02 | *** join/#asterisk creativx (n=creadure@226.62-97-205.bkkb.no) |
15:55.22 | ManxPower | SuPrSluG: that would do it. |
15:55.37 | SuPrSluG | what's a bogomip? |
15:55.49 | SuPrSluG | i'm looking it up now |
15:55.53 | saftsack | its an indicator for the cpu speed |
15:55.57 | saftsack | cat /proc/cpuinfo |
15:57.02 | SuPrSluG | well in that case i have bogomips : 2940.92 |
15:57.13 | saftsack | on your router? Oo |
15:58.12 | SuPrSluG | saftsack:are you up and running now |
15:58.48 | *** join/#asterisk xnon (n=xnon@200.8.30.161) |
15:58.51 | saftsack | yes im up and running but with the original firmware yet |
15:58.55 | *** join/#asterisk xnon_ (i=xnon@200.8.30.161) |
15:58.55 | saftsack | i will install openwrt now |
15:59.10 | saftsack | but do i need something more than the image of the buildchain? |
16:01.41 | SuPrSluG | the one they give you works. they even have a asterisk pkg pre-made. you can get rid of things you don't want to mak more room. |
16:02.01 | SuPrSluG | what router do you have? |
16:13.10 | saftsack | speedport w501v |
16:13.14 | saftsack | it is an ar7 device |
16:17.31 | *** join/#asterisk RaYmAn-Bx (i=rayman@kbhn-vbrg-sr0-vl212-213-185-15-16.perspektivbredband.net) |
16:20.39 | asdx | can you install asterisk in openwrt and put it in a router? |
16:21.40 | *** join/#asterisk wundaboy (n=asdf@c-24-21-100-201.hsd1.or.comcast.net) |
16:22.10 | saftsack | yes |
16:22.15 | saftsack | this is possible |
16:23.48 | *** join/#asterisk Nand0 (n=Nando@unaffiliated/nand0) |
16:25.34 | SuPrSluG | they have all supported hardware on their site |
16:25.53 | SuPrSluG | gotta walk the dog talk to ya soon |
16:28.26 | asdx | and how will you connect the phone to the router? |
16:36.41 | creativx | tcp ip |
16:37.20 | *** join/#asterisk DarKnesS_WolF (n=wolf@196.218.71.47) |
16:38.36 | *** join/#asterisk eltech (n=eltech--@ool-457c9421.dyn.optonline.net) |
16:41.02 | robin_sz | heh, so I thought a 2 channel ATA would be useless ... I just found a use for the second channel! the big external bell inthe workshop |
16:44.04 | *** join/#asterisk tsurk0 (n=tsurko@vpn-pppoe-213-240-242-81.megalan.bg) |
16:51.59 | *** join/#asterisk gmaruzz (n=gmaruzz@217-133-80-112.b2b.tiscali.it) |
16:57.30 | *** join/#asterisk Deeewayne (n=dwayne@ool-44c0d56e.dyn.optonline.net) |
16:57.54 | hi365 | is rtp tcp or udt? |
16:58.09 | *** join/#asterisk Malawar_ (n=Malawar@adsl-75-21-165-54.dsl.sgnwmi.sbcglobal.net) |
16:58.12 | Malawar_ | Oct 22 08:57:23 WARNING[25718]: channel.c:2752 ast_channel_make_compatible: No path to translate from SIP/line2-081a5d00(4) to IAX2/voxee-1(256) |
16:58.12 | Malawar_ | Oct 22 08:57:23 WARNING[25718]: app_dial.c:1602 dial_exec_full: Had to drop call because I couldn't make SIP/line2-081a5d00 compatible with IAX2/voxee-1 |
16:58.13 | Malawar_ | :( |
16:58.33 | Malawar_ | oh |
16:58.36 | *** join/#asterisk lukketto (n=lukketto@host129-133-dynamic.57-82-r.retail.telecomitalia.it) |
16:58.39 | Malawar_ | might have set up codecs wrong.. |
17:00.07 | *** part/#asterisk lukketto (n=lukketto@host129-133-dynamic.57-82-r.retail.telecomitalia.it) |
17:01.34 | *** join/#asterisk benjk (n=benjamin@f8a01-0357.din.or.jp) |
17:04.01 | ManxPower | Malawar_: "show codecs" |
17:04.02 | Malawar_ | yeah |
17:04.05 | Malawar_ | it was trying to use g729 |
17:04.09 | Malawar_ | but i don't have it installed/etc |
17:06.13 | Malawar_ | hmm |
17:06.20 | Malawar_ | i have an IAX connection to voxee |
17:06.24 | Malawar_ | i set disallow=all |
17:06.27 | Malawar_ | then allow=ulaw |
17:06.32 | Malawar_ | but it's still trying to use g729 |
17:07.09 | Malawar_ | voxee supports ulaaw, btw. |
17:07.38 | Malawar_ | oops |
17:07.38 | Malawar_ | nm |
17:07.49 | Malawar_ | missed thde disallow line :( |
17:15.28 | *** join/#asterisk aao_pwner (n=aao_pwne@c-24-21-91-140.hsd1.mn.comcast.net) |
17:28.04 | *** join/#asterisk tsurk0 (n=tsurko@vpn-pppoe-213-240-242-81.megalan.bg) |
17:31.55 | *** join/#asterisk ramtha (n=tk@p5088C21E.dip0.t-ipconnect.de) |
17:32.55 | Malawar_ | voxee is skipping :( |
17:38.23 | ManxPower | you need the device to not be allowed g729 |
17:38.36 | Malawar_ | i got it |
17:38.54 | Malawar_ | i was pasting the block of disallow/allows into all my devices but i missed the disallow line going out to voxee |
17:39.23 | *** join/#asterisk tsurk0 (n=tsurko@vpn-pppoe-213-240-242-81.megalan.bg) |
17:41.51 | *** join/#asterisk napkin (n=izaak@bas10-montrealak-1096753512.dsl.bell.ca) |
17:42.51 | napkin | does anyone use linksys pap2(t) devices as fxs devices for asterisk? for a while i was set on buying a digium card, then i found these things. I only need 4 fxs ports so 2 would suffice. |
17:43.46 | Strom_C | napkin: I use both |
17:44.08 | *** join/#asterisk Deeewayne (n=dwayne@ool-44c0d56e.dyn.optonline.net) |
17:44.18 | *** join/#asterisk findlay (n=justin@67.137.24.115) |
17:44.37 | findlay | are there any softphones that are scriptble? |
17:44.43 | findlay | s/scriptable/ |
17:44.44 | napkin | Strom_C: what is your recommendation? are they pretty flexible? |
17:45.19 | findlay | I want to be able to call a series of numbers without manually looking them up in a directory |
17:46.19 | *** join/#asterisk DataCompBoy (i=data@217.151.230.182) |
17:46.23 | DataCompBoy | Pfffr... Hi all! |
17:46.31 | DataCompBoy | Now, i'm need a some help with asterisk... |
17:46.48 | Strom_C | napkin: the tdm400 has slightly better audio quality and behaves more like a traditional telephone line. The only times I would recommend using the PAP2 over the TDM400 are either when you need an analog phone in a location where there is ethernet cabling but running new wiring for an analog pair is impractical, or in situations where the TDM400 is just too expensive. |
17:48.29 | napkin | thanks Strom_C. we are 4 orgs sharing a small office. we can probably handle the price difference. i am trying to get the highest quality service because a couple people here have been scared by voip and i want it to be good :) |
17:49.17 | napkin | but when i do workshops for local non profits i'll definitely recommend an asterisk+pap2 setup as it's so much cheaper... |
17:49.17 | DataCompBoy | asterisk won't send RTP packets. can't understand why... |
17:49.43 | *** join/#asterisk cbm11211 (n=Administ@66.28.182.170) |
17:49.49 | DataCompBoy | and looks like when i'm connect to inet with external IP -- nothing receive. when i'm connect thru NAT -- everything ok |
17:49.56 | DataCompBoy | in asterisk i'm have on peer set nat=yes |
17:49.57 | hi365 | Strom_C: hi |
17:50.07 | DataCompBoy | rtp debug doesn't show anything, absolutely! |
17:50.21 | hi365 | Strom_c: my pap2 wont stop beeping. any ideas? |
17:50.23 | *** join/#asterisk ToTo (n=ToTo@host35-167-dynamic.2-87-r.retail.telecomitalia.it) |
17:50.29 | *** join/#asterisk ESCulapio_ (n=ESCulapi@213stb68.codetel.net.do) |
17:50.34 | hi365 | i.e. what setting is it? |
17:50.36 | Strom_C | hi365: no. ive never encountered that problem. |
17:54.01 | Qwell | anybody know anything about buying hardware from the UK, and having it shipped to the US? :D |
17:54.17 | Strom_C | hi qwell :) |
17:54.21 | Qwell | Are there customs charges involved? |
17:54.23 | Qwell | Strom_C: hey |
17:55.33 | Strom_C | qwell: I don't know about shipping from the UK, but the one time I had something small shipped from hong kong, either the import duties were covered in the price of the postage, or they didn't exist at all |
17:55.44 | Qwell | ok |
17:55.51 | Qwell | ? == euro, or gbp? |
17:56.09 | Strom_C | hong kong? hong kong dollar |
17:56.21 | Qwell | no, ?.. as in ?619.93 |
17:56.42 | Strom_C | its just showing up as a question mark |
17:56.47 | Strom_C | stupid client |
17:57.16 | Qwell | .co.uk == euro or gbp? heh |
17:57.28 | Strom_C | gbp |
17:57.31 | Qwell | lame |
17:57.42 | Qwell | That means it costs more |
17:57.58 | Strom_C | so buy it from france or germany or something :) |
17:58.10 | Qwell | that's the thing... |
17:58.24 | Qwell | Acer went and made different configurations for different countries |
17:58.37 | Qwell | So, they only sell the configuration I want in the UK |
17:58.50 | Strom_C | what is it a configuration of? |
17:58.53 | Qwell | laptop |
17:58.58 | Strom_C | ah |
17:59.01 | Qwell | 5103wlmi |
17:59.16 | Qwell | in the US, we only have 5102... |
18:01.05 | Qwell | I'm gonna have to call them on Monday, and tell them to stop being nubs :p |
18:01.55 | Strom_C | im trying to figure out what the difference is |
18:02.40 | Malawar_ | does the caller id number format matter? |
18:02.50 | Malawar_ | i.e. can I do "Name" <555-5555> |
18:03.04 | Malawar_ | or does it have to be 12223334444 |
18:03.06 | *** join/#asterisk Darthclue (n=Darthclu@adsl-71-149-167-182.dsl.snantx.sbcglobal.net) |
18:03.17 | Strom_C | Malawar_: are you in north america? |
18:03.22 | Malawar_ | yeah. |
18:03.37 | Strom_C | you ideally want to put the number in format NXXNXXXXXX |
18:06.05 | Strom_C | i.e. just as NANPA states you should :) |
18:07.25 | Qwell | Strom_C: with what? |
18:07.57 | Strom_C | qwell: the difference between the UK-spec model and the US-spec model |
18:08.04 | Qwell | several things |
18:08.12 | *** join/#asterisk holmie (i=holm@blackedge.org) |
18:08.27 | holmie | so, the MWI NOTIFICATION stuff, Cisco or Digiums fault? |
18:08.28 | holmie | :-) |
18:08.30 | Qwell | first, we don't have the 5103wlmi here, which has the TL52 instead of the TL50 (both are Turion X2, but the 52 has double the cache) |
18:08.42 | holmie | (Cisco IP phones saying "399 Bad MWI NOTIFY") |
18:08.42 | DataCompBoy | Can anybody describe how to debug RTP path? |
18:09.00 | Qwell | second, their 5102wlmi model has a camera on the 512 and 1gb models, plus, I think theirs can do 4GB instead of 2GB |
18:09.08 | Strom_C | ah, i see |
18:09.28 | Qwell | our 5102wlmi with a camera has 2gb, and for some stupid reason, uses a 4200rpm drive instead of a 5400rpm drive like the lower models |
18:09.44 | *** join/#asterisk yogurt2ungue (n=yogurt2u@24-48-231-201.fibertel.com.ar) |
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18:09.54 | Qwell | it's just weird that it varies so much |
18:10.57 | Qwell | holmie: good luck with that... |
18:11.02 | holmie | ;-D |
18:11.14 | Qwell | holmie: MWI should work just fine on Cisco phones...it's just a SIP notify or something |
18:11.29 | holmie | I think it's a cisco bug |
18:11.40 | holmie | I found a page with lots of people complaining about it |
18:11.41 | Qwell | however, if it isn't working, then it's probably a bug (either in Asterisk, or more likely, Cisco not conforming to spec...which isn't unusual) |
18:11.53 | hi365 | my pap2 is beeping like a pager. how do i shut it up? |
18:12.02 | Juggie | throw it against the wall. |
18:12.03 | Qwell | hi365: unplug it? heh |
18:12.15 | Qwell | hi365: constantly, or every couple minutes? |
18:12.46 | Qwell | hi365: it's probably telling you that you have VM |
18:12.54 | holmie | Qwell: I think the c7970G don't handle the correct MWI notification from asterisk properly |
18:13.05 | Qwell | holmie: quite likely |
18:13.32 | holmie | Funky. |
18:13.32 | hi365 | Qwell: every 30 sec. or so |
18:13.45 | Qwell | hi365: It's telling you that you have voicemail. There are options to disable it |
18:15.23 | Strom_C | man, you have to tell me what that option is so I can use it to irritate my most annoying clients |
18:16.20 | Qwell | Strom_C: sip.conf, mailbox=, I think |
18:16.45 | Qwell | I think the problem, is that both the ata and asterisk try to do mwi stuff |
18:16.55 | Qwell | something funky like that, and it ends up b0rking itself |
18:16.58 | hi365 | Ring On No New VM = no |
18:17.01 | Strom_C | no no, the beeping option |
18:17.05 | Darthclue | How old is the 1.2 branch? |
18:17.14 | Strom_C | i know how to set mailbox= in sip.conf, qwell :) |
18:17.26 | Qwell | Darthclue: the last commit was probably a few days ago. That's a fairly loaded question |
18:17.59 | *** join/#asterisk TripleFFFF (n=TripleFF@145-27.mc.cite.net) |
18:18.19 | hi365 | Qwell; Ring On No New VM = no |
18:18.33 | Darthclue | I like asking loaded questions. Makes things more interesting. Can't get trunk to compile cleanly so i'm going with 1.2. |
18:18.35 | TripleFFFF | can i record in an agi ? |
18:18.42 | Qwell | Darthclue: define cleanly? |
18:18.42 | hi365 | VMWI Ring Splash Len: 0 |
18:18.51 | Qwell | TripleFFFF: should be able to |
18:18.57 | Qwell | TripleFFFF: I can't see why not |
18:19.01 | TripleFFFF | how lol |
18:19.10 | Qwell | dunno, I've never done AGI, heh |
18:19.12 | TripleFFFF | i wanna use manager to call an agi that records |
18:19.14 | Strom_C | Darthclue: well if it's any indication, I still regularly use 1.2 for all my professional installs |
18:19.17 | Qwell | I only code - I don't use the stuff |
18:19.33 | Qwell | (I actually do use it...) |
18:20.13 | TripleFFFF | wahts net sec lol |
18:20.31 | hi365 | and i dont even have new voicemail! |
18:20.52 | *** join/#asterisk ToTo (n=ToTo@host35-167-dynamic.2-87-r.retail.telecomitalia.it) |
18:20.57 | Qwell | TripleFFFF: it's a branch that let's you control firewall hardware from...umm...that company |
18:21.07 | *** join/#asterisk Dovid (n=Dovid@barak.cellcom.co.il) |
18:21.07 | TripleFFFF | lol |
18:21.10 | Qwell | I'm completely spacing on the name |
18:21.21 | Darthclue | make clean errors out, make install generates a couple of invalid modules (on zaptel). The last time I used Asterisk, it was always a cvs-head install. |
18:21.46 | Qwell | TripleFFFF: Ranch Networks |
18:21.51 | TripleFFFF | k heeh |
18:22.01 | Qwell | but, it uses an open protocol or something, so other vendors may support it too...dunno |
18:22.45 | Qwell | basically, it tells the firewall that it needs exactly X bandwidth, and it allocates it for you |
18:22.45 | *** join/#asterisk drcode (n=chatzill@87.69.59.186.cable.012.net.il) |
18:22.48 | drcode | hi all |
18:22.51 | Qwell | pretty cool stuff |
18:23.10 | TripleFFFF | hmm so i cant find monitor in the agi libs |
18:23.16 | TripleFFFF | i guess, its exec(monitor |
18:23.23 | Qwell | TripleFFFF: should be able to call it like any other application |
18:23.55 | TripleFFFF | yeah tought theres was an agi command for it |
18:25.54 | drcode | is there sip client in php / java that I can use from web site? |
18:26.05 | TripleFFFF | not java lol it sucks resources |
18:26.10 | drcode | k |
18:26.13 | TripleFFFF | drcode let me know im looking for one |
18:26.36 | drcode | I want to let users run client from web , I know there is some in activex |
18:27.08 | hi365 | Qwell: any final thoughts on pap2 chirps? |
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18:28.52 | DataCompBoy | Peeeeeeeopppleeee!! What I'm need to check to find out why no RTP sent from asterisk?! |
18:29.28 | hi365 | port? |
18:29.33 | hi365 | ?ports |
18:29.48 | DataCompBoy | ports open on both ends. |
18:30.00 | DataCompBoy | i'm even not receive "Sent rtp packet" |
18:30.05 | DataCompBoy | nothing! |
18:30.06 | DataCompBoy | :/ |
18:30.13 | hi365 | which ports? |
18:30.18 | DataCompBoy | 10000-20000 |
18:30.37 | DataCompBoy | open on firewall on server, i have disabled firewall |
18:30.54 | DataCompBoy | Yesterday everything worked :( |
18:31.36 | hi365 | :( |
18:31.48 | *** join/#asterisk clyrrad (n=ddd@CPE001195f553c7-CM0011aea484a4.cpe.net.cable.rogers.com) |
18:31.48 | hi365 | rtp.conf fine? |
18:31.54 | holmie | Qwell: can I keep asterisk from sending the MWI notification? |
18:31.59 | DataCompBoy | untouched, and yes - fine |
18:36.10 | *** join/#asterisk lrizzo (n=luigi@88-149-142-59.f5.ngi.it) |
18:36.35 | clyrrad | I have a question about DTMF - I have phones set to INBAND+INFO, and it works well with Asterisk as well as most IVR systems. Howerver some IVR systems will not accept the DTMF's - Just wondering if anyone has had this issue - or knows of a work around? |
18:37.44 | DataCompBoy | the best, i'm get is one line: "Sent RTP packet to 217.151.230.182:9000 (type 0, seq 8001, ts 160, len 160)" and nothing more... |
18:37.51 | *** join/#asterisk gmaruzz (n=gmaruzz@217-133-80-112.b2b.tiscali.it) |
18:37.55 | Qwell | holmie: unset the mailbox= option in sip.conf |
18:39.25 | DataCompBoy | oohmmm |
18:39.38 | DataCompBoy | looks like phone receive always on one port, send from other ?! |
18:39.58 | DataCompBoy | is that normal?! |
18:42.09 | Qwell | DataCompBoy: yes |
18:42.43 | DataCompBoy | Qwell: strange, but looks like Asterisk doesn't receive such packets. |
18:42.55 | Qwell | check your firewall |
18:43.21 | clyrrad | makesure your firewall has open the ports you defined in rtp.conf |
18:43.32 | DataCompBoy | firewall have open them |
18:43.42 | clyrrad | Qwell: do you know the answer to my DTMF query? |
18:43.50 | DataCompBoy | 8:43:15.314965 IP 217.151.230.182.9001 > orange.11469: UDP, length 84 --- packets arrives |
18:44.33 | clyrrad | you see that on your firewall? |
18:44.42 | DataCompBoy | yes |
18:44.48 | clyrrad | does anyting hit your CLI? |
18:44.59 | DataCompBoy | Sent RTP packet to 217.151.230.182:9000 (type 0, seq 26403, ts 160, len 160) |
18:45.11 | DataCompBoy | and i'm got that packet on firewall -- and it passed |
18:46.40 | DataCompBoy | and on server i'm receive reply RTP packets (server have sent one packet from 17892 to mine 9000, from my machine sent back from 9001 to 17893). |
18:46.58 | DataCompBoy | but no more RTP packets from asterisk, and no sound heared... |
18:48.57 | DataCompBoy | looks like after no rtp phone tries to send reply from 9000 back -- also without luck :/ |
18:49.52 | *** join/#asterisk Wall (n=mnose@host129.200-117-63.telecom.net.ar) |
18:49.59 | Wall | Hi |
18:50.12 | Wall | alguna que sepa español ? |
18:50.18 | Wall | alguno que sepa español ? |
18:52.14 | holmie | Qwell: figured, thanks. :-) |
18:52.23 | holmie | Qwell: I ask too fast! |
18:54.26 | *** join/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
18:54.45 | DataCompBoy | What more I can check?... |
19:03.27 | *** join/#asterisk tuck3r (n=tuck3r@unaffiliated/tuck3r) |
19:03.40 | Druken | anyone used asterfax? |
19:06.29 | *** join/#asterisk Strom_C (n=strom@netblock-66-159-243-60.dslextreme.com) |
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19:18.47 | TripleFFFF | anyway to see ip trying to connect as anonymous via dialplan like noop("IP") ? |
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19:32.37 | TripleFFFF | any reason my cisco 7960 when i talk it cuts the other one from talking .. i mean i cant hear him.. like VAD? |
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19:40.27 | TripleFFFF | ?? |
19:40.57 | *** join/#asterisk bmg505 (n=leon@c1-226-1.rndf.isadsl.co.za) |
19:41.47 | TripleFFFF | anyone ? |
19:48.03 | *** join/#asterisk dasenjo (n=dasenjo@208.195.215.163) |
19:51.56 | enots | .t |
19:52.00 | enots | oops |
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20:09.26 | *** join/#asterisk DEader (i=DEader@ool-44c64639.dyn.optonline.net) |
20:09.28 | DEader | hello |
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20:09.44 | DEader | does any one know how to tweek fromuser in asterisk |
20:10.21 | DEader | just like setting callerid in extension.conf |
20:10.31 | DEader | i want to be able to set fromuser there also |
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20:45.38 | *** join/#asterisk eido (n=eido@m015f36d0.tmodns.net) |
20:50.16 | eido | hey folks, i'm sort of exploring using wifi voip phones as convention / hotel-type radios. i know the concept of 'PTT' is still not really available to the mases (except it seems via Vocera's 'badges') - is there a recommendation of a type of VOIP wifi phone that is inexpensive but capable? I see the ZyXEL P-2000W's around on ebay for decent prices? |
20:52.20 | [TK]D-Fender | WiFi VoIP phones currently SUCK. ALL of them. |
20:52.37 | eido | heh. |
20:55.56 | robin_sz | The Zyxel WiFi phones are great ... |
20:56.16 | robin_sz | you can use them to hold most sorts of doors open |
20:56.51 | robin_sz | they do have one teeny weeny little drawback in terms of using them as a phone |
20:56.56 | eido | alrighty then :) |
20:57.18 | robin_sz | their complete inability to connect to other WiFi devices for short periods ... |
20:57.30 | [hC] | Ive tried the zyxel and linksys wip300 extensively |
20:57.38 | robin_sz | typically for the first hour or so after you turn them on ... |
20:57.41 | [hC] | im now having to eat a sale i did to a client of wip300's cause they suck |
20:57.43 | [hC] | and they dont want em |
20:57.53 | robin_sz | after that, the battery is flat and the problem goes away :) |
20:59.27 | robin_sz | best advice is to use DECT phones and ATAs |
20:59.28 | eido | ew. |
20:59.40 | robin_sz | theres a dutch SIP to DECT gateway ... |
20:59.41 | eido | er, the ew was on the various ocmments, not on the last one. |
21:00.43 | robin_sz | I think it takes up to 6 phones per base station |
21:00.46 | eido | well, here's what i'm tyring to do, see if i'm loopy. i do convention services - registration, badging, IT stuff, etc. i see so many events that use FRS radios as their communication net. they suck. "I can do better!" - i'm trying to figure out -how- :) |
21:01.00 | robin_sz | hmmm |
21:01.02 | eido | the best iv'e seen are the Vocera badges - that's pretty damned slick, but i bet it costs an arm and a leg. |
21:01.18 | [TK]D-Fender | WIP300 is nice for browsing networks, but the phone has vitually 0 functionality, and the interface is slow |
21:01.24 | robin_sz | wifi is unsuitable ... |
21:01.40 | robin_sz | due to jitter amongst other stuff |
21:01.56 | robin_sz | the tdm soon clogs up with a few active channels |
21:01.56 | eido | well, audio quality on the wifi network will be a -vast- improvement over FRS. really :) |
21:03.29 | robin_sz | does that matter? |
21:04.23 | eido | does which matter? |
21:04.29 | robin_sz | audio quality |
21:04.56 | robin_sz | is that high on the list of your clients priorities? |
21:05.04 | robin_sz | or is range and reliability more important? |
21:05.39 | robin_sz | woudl wired phones be too hard to do? |
21:06.04 | eido | range and coverage are the kicker. |
21:06.11 | eido | we need to be able to bridge between disparate locations. |
21:07.11 | robin_sz | out in the fields? |
21:07.11 | eido | so folks in the ballroom, vs folks down in registration vs folks on the 40th floor of the hotel. |
21:07.12 | robin_sz | in some woods? |
21:07.12 | eido | no, primarily in the same building. |
21:07.12 | robin_sz | in africa? |
21:07.12 | eido | we can carpet the place with our own wifi hardware. |
21:07.20 | robin_sz | or europe? |
21:07.29 | eido | ideally, a PTT solution for all the users - similar to FRS would be great. private communications (direct dial) is a nice, but general 'group chat' is really what we need. |
21:07.31 | eido | no, US. |
21:07.43 | robin_sz | in europe all the decent conference venues have networking spread all over them |
21:08.09 | eido | yah, most of my events are not in large commercial venues. |
21:08.23 | eido | they tend to be in at best, conference hotels (1500-room hotels) |
21:08.42 | eido | at worst, small cheezy hotels that have poor wireless. at very worst, its small cheezy hotels with commercial wireless services :) |
21:08.42 | robin_sz | and they aint networked? |
21:08.51 | eido | we'd put in our own wireless. |
21:09.06 | eido | i don't mind buying a dozen APs to set our own network and control it. |
21:09.14 | robin_sz | wireless trunks to router/gateways .. wired phoens from there |
21:09.14 | eido | it' the mobile units that are the problem. |
21:09.17 | robin_sz | or DECT |
21:09.34 | *** join/#asterisk shy_guy (i=shy_guy@c213-100-17-43.swipnet.se) |
21:09.54 | eido | i don't know DECT phones well... how do they handle hopping? can i move one from one conference room to another, or upstairs, and it'll stay on-network? |
21:10.19 | robin_sz | no idea |
21:10.32 | eido | yah, that's the problem. and we're still stuck with no PTT tech. |
21:10.43 | robin_sz | sounds like you need a UHF base station and a bunch of Motorola HTs |
21:10.50 | eido | thought of that too. |
21:10.58 | robin_sz | probably the way to go |
21:11.22 | robin_sz | VOIP is almost certainly NOT the way to go |
21:11.25 | shy_guy | eido:whats this hopping feature like to the end user? |
21:11.34 | eido | shy_guy: in my ideal world, seemless. |
21:11.41 | eido | see |
21:11.47 | eido | frs or gmrs or the like are fine for small venues. |
21:11.55 | eido | ala, everyone within, say, 1500 feet of each other. |
21:11.57 | shy_guy | eido:if u have an account on the conference server,cant u join from anywhere? whats missing |
21:12.03 | eido | but go beyond that, and the whole mechanism breaks. |
21:12.47 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
21:13.01 | shy_guy | i cant really get u. are you from the marketing? :D |
21:13.06 | eido | shy - that would mean a wireless SIP phone that's dialled in / 'off hook' permanently. PTT problems are already an issue - doing it that way would bring on battery problems. |
21:13.24 | eido | shy_guy: no. by the way, something is wrong with your 'yo' keys. |
21:14.06 | shy_guy | we do such wifi stuff |
21:14.06 | robin_sz | eido, as I thought. Motorola have it well covered. |
21:14.11 | robin_sz | check out iDEN |
21:14.12 | shy_guy | pretty seamlessly |
21:14.14 | robin_sz | http://idenphones.motorola.com/idenHome/common/what_is_iden.jsp |
21:14.28 | *** join/#asterisk saftsack (n=bla@pD9E04CF7.dip.t-dialin.net) |
21:14.33 | robin_sz | and some pretty cute phones too |
21:14.34 | robin_sz | http://idenphones.motorola.com/idenProducts/phonesHome.do?phones=1 |
21:15.12 | saftsack | hi |
21:15.24 | eido | shy_guy: your product is not what i'm trying to do. |
21:15.35 | eido | iden is what nextel uss, yes? |
21:16.38 | shy_guy | eido:whats this ptt issue? tell me |
21:16.39 | eido | yah, iden is neat, but is a lot higher level / more costly than what i'm looking for. i may be hosed here :( |
21:16.45 | eido | shy_guy: ever use a walkie talkie? |
21:16.54 | robin_sz | yeah, lots. |
21:16.55 | eido | like an frs or gmrs radio? |
21:17.10 | shy_guy | keep talkin eido |
21:17.24 | eido | er, that sentence ended with a ? - that implies "its your turn to talk" |
21:17.46 | robin_sz | i use Motorola 300's a LOT ... coverage is typically 1000s of metres |
21:17.56 | robin_sz | well, used |
21:17.57 | eido | robin_sz: they require a commercial license, yes? |
21:18.00 | robin_sz | not in that game anymore |
21:18.03 | robin_sz | yeah |
21:18.26 | eido | i have a bunch of motorola VHF radios tat i was going to adapt to GMRS frequencies - they can do it, but they're not "fcc approved" equipment. It's a little edgy. but the radios are cheap. |
21:18.47 | shy_guy | i worked on ultra wide band radio |
21:18.54 | robin_sz | 2watts on VHF goes a LONG way |
21:19.10 | eido | yeah, no kidding. |
21:19.10 | shy_guy | and now after four years of working when i left, its ericsson whose implementing it |
21:19.27 | shy_guy | god damn fcc and the club |
21:19.35 | shy_guy | politcs players |
21:19.47 | robin_sz | iDEN looks to be exactly what you need |
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21:20.09 | robin_sz | im pretty certain voip is not what you want |
21:20.16 | eido | yeah, but iden is a large scale network - it's a digital transport. i'd need radio coverage in the venue, plus the phones - plus the fcc licenseing to do it, plus a server to contorl / route the phones. |
21:20.24 | eido | nifty if i had a zillion dollars :) |
21:20.44 | robin_sz | wired phoens could work ... |
21:20.49 | robin_sz | but you dotn want that either |
21:20.58 | eido | yah, i know i could do voip + wired. |
21:21.02 | shy_guy | eido:lol, who the hell wud wish for iden |
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21:21.18 | robin_sz | radios would work .. |
21:21.19 | eido | that could link all the stations / departments / whatever together, without stringing pots lines (or using the hotel's PBX, which is traditoinally -garbage- |
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21:21.32 | eido | i'm unfortunately heading back to radios with some form of repeater. |
21:21.39 | robin_sz | rounds simplest |
21:21.44 | robin_sz | sounds |
21:21.44 | eido | robin_sz: did you say you used gp300's? |
21:21.51 | shy_guy | whats wrong with wifi and wifi phones doing all this |
21:21.58 | eido | shy_guy: listen to the convo please. |
21:22.04 | robin_sz | yeah, on vhf and uhf |
21:22.05 | eido | mostly because they can't. |
21:22.14 | eido | gp300's look inexpensive. hm. |
21:22.16 | shy_guy | by 2008, 80% mobile phones would have wifi support anyway |
21:22.20 | eido | i should look at an FCC commercial license. |
21:22.21 | Greek-Boy | if the wifi phones were reliable atleast you would have a covnerged solution but u gotta use the radios for now |
21:22.22 | robin_sz | shy_guy, there are no wifi phones |
21:22.44 | robin_sz | gp300s are solid and reliable and very configurable ... and multi use |
21:23.01 | eido | heh. i could... rig up one of the Gp300's to an audio gateway, subscribe it into an asterisk hosted conference. |
21:23.03 | robin_sz | you can use them to talk on, or to beat punters to death with |
21:23.10 | eido | folks could listen / chat in from VOIP clients on PC's. |
21:23.11 | eido | HAH |
21:23.13 | eido | that's getting interesting. |
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21:23.36 | eido | can the GP300's do freq offset stuff? ala ham radio repeater style? |
21:23.46 | robin_sz | oh yeah |
21:23.47 | robin_sz | easy |
21:23.52 | robin_sz | fully synthed |
21:23.54 | shy_guy | Greek-Boy:i dont want to market our products, but wifi works. we have a patent pending technology on that. |
21:24.08 | saftsack | SuPrSluG hi are you still here? |
21:24.17 | eido | i could... hmmmmmmmmmmmmmmm. interesting. FCC would love this. what if i had some form of simple audio gateway. hmm. tht won't work. i can't find out when it should transmit. or could it. Hm. |
21:24.30 | robin_sz | shy_guy, you'll be rich then, because as yet, all WiFI phoens suck |
21:24.49 | eido | i was just thinking of attaching a radio via a simple PTT trigger, coneting it via VOIP gateway to an asterisk hosted conference. VOIP clients on PC's would be able to hear the radio conversation. |
21:24.56 | shy_guy | robin_sz:why exactly do they suck? can u point to the core issues |
21:24.57 | eido | but how could they answer back? Hmm. |
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21:24.59 | *** mode/#asterisk [+o mog] by ChanServ |
21:25.12 | robin_sz | shy_guy, point 1, inabality to connect to gateways |
21:25.21 | robin_sz | shy_guy, poitn 2 battery live measured in hours |
21:25.25 | eido | hah. freq offset would be on permanent trnasmit? hm. |
21:25.36 | robin_sz | shy_guy, point 3 see poitn 1 |
21:26.01 | shy_guy | robin_sz:is that all? |
21:26.07 | robin_sz | ALL? |
21:26.18 | eido | uhh. i'd say that's "enough". it means i can't use them for this function. |
21:26.23 | robin_sz | I would have thought an inablity to connect to a gateway was failry serious no? |
21:26.43 | robin_sz | a MINIMUM pocket life of 8 hours is a MUST HAVE |
21:26.46 | eido | also, the voip phones don't AP-hop well, as i'm understanding. |
21:26.51 | eido | minimum, yes. |
21:27.00 | robin_sz | absolute minimum |
21:27.03 | eido | and for radio usage, they would have to be on permanet 'listen' mode. |
21:27.07 | robin_sz | the zyxel is ~2 hours |
21:27.10 | shy_guy | i have heard and answered more criticism than that robin_sz |
21:27.14 | eido | iden folks work because the y'wake up' when a connection / msg comes in. |
21:27.24 | eido | most voip phones are not that smart. |
21:27.26 | eido | er. |
21:27.34 | eido | i'd say all of them - since they' renot designed to work that way. |
21:27.46 | robin_sz | shy_guy, well, show me one that works then ... |
21:28.23 | shy_guy | robin_sz:you are livin in wifi stone age like prolly everyone else. but trust me there is a product that exceeds your expectations |
21:28.30 | eido | shy_guy: name it. |
21:28.32 | robin_sz | bollocks |
21:28.43 | shy_guy | it works on same wifi networks |
21:28.46 | eido | shy_guy: |
21:28.49 | eido | name what you're talking about. |
21:28.55 | eido | and stop fishing around like a marketing idiot. |
21:28.56 | shy_guy | runs on windows mobile and symbian platforms |
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21:29.12 | shy_guy | hehe |
21:29.33 | eido | anyway, shy_guy guys idiocies aside, i think i'm hosed on this. |
21:29.34 | shy_guy | eido: u... |
21:29.38 | eido | the -only-thing i saw was vocera. |
21:29.50 | eido | i'll drop them a aline and see if they can do something for something under the US military budget. |
21:30.14 | shy_guy | eido:you like to discuss things in which i have no interest. microwave,radios... um not even hearing. |
21:30.29 | robin_sz | shy_guy, so you going to name this mystery product then or what? |
21:30.33 | eido | shy_guy: and you keep spouting drivel with no information. |
21:30.42 | eido | so unless you ave actual facts, information, and details, just stfu. |
21:30.56 | saftsack | is it possible to build asterisk < 2mb? |
21:31.10 | Qwell | saftsack: sure |
21:31.22 | Qwell | saftsack: You'll need to remove quite a few modules, but...yeah |
21:31.32 | saftsack | and < 0.5mb in a sqaushfs system? |
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21:31.41 | Qwell | 500k? probably not |
21:31.59 | Qwell | at least, not with any usability |
21:32.06 | saftsack | hmm but this 500k are compressed (squashfs) |
21:32.39 | eido | hmmmmmmmmmmmm. |
21:33.02 | saftsack | Qwell do you know the compression factor of squashfs? |
21:33.13 | Qwell | no clue |
21:33.58 | saftsack | ok |
21:40.03 | shy_guy | robin_sz:guess you liked it, later mate! |
21:40.36 | eido | robin_sz: did shy_guy point you at his swedish whatever site? |
21:40.49 | robin_sz | interesting idea ... |
21:40.57 | eido | yes, and i told him it was not what i needed, but he wasn't listening. |
21:41.10 | eido | it's an auto-switch from voip to cellular net and back |
21:41.12 | robin_sz | shrug .. .its stil an interesting idea |
21:41.13 | eido | bfd? |
21:41.51 | robin_sz | it assumes you have some windows enabled, gsm and wifi capable PDA |
21:42.08 | robin_sz | do those things last 8 hours on in your pocket? |
21:42.17 | robin_sz | in WiFi mode> |
21:42.18 | robin_sz | ? |
21:42.20 | eido | not with the radio active. |
21:42.22 | eido | wifi is heavy. |
21:42.30 | robin_sz | oh |
21:42.38 | robin_sz | thats my minimum requirement |
21:51.14 | Greek-Boy | anyone here from Philippines? |
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22:01.38 | DEader | is it posible to set asterisk user setting example fromuser and context from the extension.conf file |
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