| 00:00.13 | x86 | [TK]D-Fender: well, which field is it? |
| 00:00.42 | x86 | 100 => 100,Some User,user@domain.tld,attach=no|tz=central |
| 00:00.47 | x86 | that's my line from voicemail.conf |
| 00:01.22 | x86 | user's mailbox is 100, i tried also using 100 for the password but that didnt work |
| 00:02.25 | x86 | hmm |
| 00:02.39 | x86 | i think maybe it's not the voicemail setup, but it's handling DTMF wron |
| 00:02.41 | x86 | wrong* |
| 00:10.23 | *** join/#asterisk microcape (n=microcap@las-cust-208.57.57.94.mpowercom.net) |
| 00:11.13 | *** join/#asterisk kpettit (n=keith@69.15.174.114) |
| 00:11.13 | ManxPower | x86, you are missing several commas |
| 00:12.09 | ManxPower | 2320 => 1234,John DeMajo,,5045551212@mycingular.com,|tz=central |
| 00:12.13 | ManxPower | notice the extra cmmas |
| 00:13.14 | *** part/#asterisk microcape (n=microcap@las-cust-208.57.57.94.mpowercom.net) |
| 00:15.50 | *** join/#asterisk SibRphrek (i=SibrPhre@user-12lccke.cable.mindspring.com) |
| 00:18.47 | rezzen | i'm receiving "403 Forbidden" on incoming call to my asterisk server. Do I have to explicitly allow invites for phones/extensions already defined in sip.conf and extensions.conf as per docs at voip-info.org? |
| 00:20.52 | ManxPower | rezzen, it's good to do so. |
| 00:21.06 | ManxPower | but reinvites would not cause a 403 error |
| 00:21.23 | ManxPower | a 403 error would be a 1) username, 2) password, or ) destination problem |
| 00:21.33 | ManxPower | but 3) should really cause a 404 |
| 00:21.40 | *** join/#asterisk nagl (n=nagl@vie-086-059-104-148.dsl.sil.at) |
| 00:21.48 | x86 | ManxPower: i'm going by the howto, and it works ;) |
| 00:21.58 | x86 | ManxPower: my problem was incorrect DMTF handling |
| 00:22.09 | ManxPower | Ah. That would do it too. |
| 00:22.15 | x86 | ManxPower: it was unset (so defaulted to RFC), i set it to auto and it works fine |
| 00:22.17 | nettie | hi guys, anyone know what this message means please? - chan_sip.c:3588 process_sdp: No compatible codecs! |
| 00:22.46 | ManxPower | however you do have the email address in the email field (long message) rather than the pager field (short message) Is that what oyu want. |
| 00:22.56 | rezzen | ManxPower: i don't understand your response. good to return 403 on incoming? |
| 00:23.02 | x86 | ManxPower: only thing is, when someone calls an extension and it goes to voicemail, it does not play their greeting or anything, it just beeps to have them start recording... |
| 00:23.15 | x86 | ManxPower: how do i make it play their recorded unavailable message? |
| 00:23.16 | ManxPower | nettie, That means the list of supported codecs in Asterisk and the list of supported codecs for the device do not overlap |
| 00:23.30 | nettie | ahhh |
| 00:23.31 | nettie | ok |
| 00:23.34 | ManxPower | rezzen, It's good if you don't want random people using your PBX |
| 00:23.36 | nettie | the call works anyway |
| 00:23.38 | rezzen | if i remove the ext/phone from the register string, the incoming comes to default "s" extension. that works (asterisk accepts call, i hear auto-attendant) |
| 00:24.11 | nettie | is it more a codec priority error? |
| 00:24.34 | ManxPower | nettie, Why not just allow the one specific codec you want? |
| 00:24.43 | rezzen | MaxPower: so then there is a directive somewhere to (at least for the purpose of learning) to allow that call? its interesting none of the getting started docs mention it, if this is the case. |
| 00:24.44 | ManxPower | nettie, it might happen if you did an allow=all |
| 00:24.50 | *** join/#asterisk rajiv (n=irc@gentoo/developer/rajiv) |
| 00:24.50 | nettie | yeah in the sip.conf related to that phone |
| 00:24.55 | nettie | yeah |
| 00:24.58 | nettie | that's the reason |
| 00:25.01 | nettie | I suppose |
| 00:25.04 | nettie | checkign now.. |
| 00:25.10 | x86 | ManxPower: any ideas on my issue? :) |
| 00:25.12 | ManxPower | rezzen, if you remove the entry and it works then you have a problem in the entry |
| 00:25.20 | ManxPower | x86, What issue? |
| 00:25.33 | ManxPower | Ahrimanes, that issue. |
| 00:25.43 | rezzen | ManxPower, the only thing i remove is the trailing "/ext" from the register directive. |
| 00:25.48 | ManxPower | x86, "make sounds" in asterisk source directory or maybe "make datafiles" |
| 00:26.06 | x86 | ManxPower: i installed from apt-get ;) |
| 00:26.08 | rezzen | and then a call is accepted. |
| 00:26.09 | ManxPower | rezzen, you keep adding important information |
| 00:26.30 | x86 | ManxPower: when the user records their own greeting, how do i make VoicemailMain play it? |
| 00:26.40 | ManxPower | x86, it will do so by default |
| 00:26.47 | *** part/#asterisk akrall (i=user@201.152.155.171) |
| 00:27.03 | ManxPower | unless you do something stupid like use the "s" option to Voicemail |
| 00:27.05 | x86 | it does not, is what i'm saying ;) |
| 00:27.07 | x86 | AH! |
| 00:27.08 | nettie | ManxPower: specified ulaw on both user and peers -- works great now thanx |
| 00:27.11 | x86 | dont use s? |
| 00:27.18 | ManxPower | do you know what "s" means? |
| 00:27.35 | justinu | don knotts is dead. RIP |
| 00:28.14 | ManxPower | x86, simplify simoplify simplify |
| 00:28.41 | x86 | hmm changed it to 'u' and works better now ;) |
| 00:29.31 | ManxPower | *grumble* My boss and his wife are having a fight via phone and text messages with their kid. |
| 00:29.47 | justinu | what does that have to do with you? |
| 00:29.47 | ManxPower | I have to stay with them because there have not been available hotel rooms in the area since Katrina |
| 00:29.52 | justinu | ah |
| 00:30.22 | mishehu | ManxPower: hahaah that sounds funny |
| 00:30.25 | ManxPower | Their kid lied and said she was spending the night with a female friend, she was actually going to spend the night with her BF |
| 00:30.39 | justinu | how could she!? |
| 00:30.43 | mishehu | and having some snu-snu with him too I bet. |
| 00:30.50 | justinu | death by snu snu! |
| 00:31.02 | ManxPower | Oh, and it's mardi gras weekend |
| 00:31.14 | mishehu | *face of horror* *face of extreme joy* *face of horror* *face of extreme joy* |
| 00:31.20 | *** join/#asterisk asterisk99 (n=astguy@modemcable169.194-130-66.mc.videotron.ca) |
| 00:31.22 | *** join/#asterisk skwashd (n=skwashd@phpgroupware/skwashd) |
| 00:31.25 | skwashd | hi all |
| 00:31.32 | mishehu | I did snu-snu |
| 00:31.38 | justinu | heh |
| 00:31.51 | skwashd | i am playing with *@home ... and i am having some issues with the tarball install |
| 00:32.08 | ManxPower | So they are driving down to New Orleans (about 45 min drive away) to bring the kid home. |
| 00:32.10 | skwashd | i am using centos 4.2 ... but it doesn't seem to create the asterisk user duiring setup |
| 00:32.16 | ManxPower | ~amp |
| 00:32.17 | jbot | i guess amp is NOT supported here! people using it should join #amportal |
| 00:32.36 | skwashd | does the user and group ids matter? |
| 00:33.01 | asterisk99 | Anyone know why one would get tons of "Remote UNIX connection disconnected" messages?? This started after I reconfiged zapata (tho I can;t see why that had anything do do with it) |
| 00:33.31 | tzafrir_laptop | asterisk99, is that amp? |
| 00:33.46 | asterisk99 | amp?? |
| 00:33.52 | ManxPower | asterisk99, that would be someone doing "asterisk -r" |
| 00:33.55 | tzafrir_laptop | amportal |
| 00:34.22 | asterisk99 | ManxPower: No ... I'm all alone on this machine |
| 00:34.50 | ManxPower | asterisk99, You are not. |
| 00:34.51 | tzafrir_laptop | "Remote UNIX connection disconnected" is also when connecting to the manager interface though localhost, IIRC |
| 00:34.53 | asterisk99 | ManxPower: Unless asterisk failed on startup - which it did |
| 00:35.03 | nettie | manx, I actually have multiple outbound providers what's the best way to use all of them as outbound channel randomically? just create an outboard context and put there there? |
| 00:35.10 | ManxPower | tzafrir, Ah. I thought that was a slightly different message |
| 00:35.38 | tzafrir_laptop | not sure (this is too late at night now...) |
| 00:35.52 | ManxPower | nettie, that is a farily advanced thing. you have to Dial, then check the value of DIALSTATUS then determine to try another provider or stop because the line was busy or answered |
| 00:36.22 | *** part/#asterisk bkw__ (n=bkw_@adsl-70-234-37-160.dsl.tul2ok.sbcglobal.net) |
| 00:36.47 | ManxPower | asterisk99, have asterisk in one session (asterisk -rvvv) then in another session do another "asterisk -rvvv" and then do a quit. look at the first session |
| 00:37.04 | nettie | ManxPower ah.. OK, thanx for the hint, I'll find out some dcs.. |
| 00:37.13 | ManxPower | "show application dial" |
| 00:37.45 | asterisk99 | ManxPower: I agree. That would happen id I did multiple -r's, but I didn't |
| 00:39.07 | skwashd | nm ... was $PATH issues from su ... reinstalling fine now |
| 00:39.10 | *** part/#asterisk skwashd (n=skwashd@phpgroupware/skwashd) |
| 00:45.52 | rezzen | i've receiving 403 Forbidden on incoming calls from ITSP. The itsp context is setup such that insecure=very. what else do i need to do to allow calls? |
| 00:47.19 | *** join/#asterisk asterisk99 (n=astguy@modemcable169.194-130-66.mc.videotron.ca) |
| 00:47.54 | justinu | verboten |
| 00:48.12 | asterisk99 | Asterisk is not starting on reboot - dunno why - I can;t see any error msgs in /var/log/syslog ... is this the right place?? |
| 00:48.56 | rezzen | asterisk99 is it failing to start on reboot, or is it not configured to start on reboot? |
| 00:49.37 | asterisk99 | rezzen: I set that up already... I moved 2 FXO modules on my Digium card and all hell broke loose |
| 00:50.38 | asterisk99 | rezzen: I assume it's trying to start ... and failing (yes, I modified /etc/zaptel.conf |
| 00:51.59 | rezzen | asterisk99: if you're assuming its trying to start, i'll start by assuming its not trying to start. does it start manually? ( # asterisk -vvvc ) |
| 00:53.47 | asterisk99 | rezzen: It does start if I do an asterisk -c |
| 00:54.18 | rezzen | asterisk99, what distro are you using? |
| 00:54.38 | asterisk99 | rezzen: Then, I get a bazillion Remote UNIX Connection disconnected messages (It just started doing that) |
| 00:55.26 | rezzen | stop it, then try again with asterisk -vc. |
| 00:55.40 | asterisk99 | rezzen: SVN-branch-1.2-r10409M |
| 00:55.46 | rezzen | is there any sign that asterisk is already running? (i'm kinda clutching at straws now) |
| 00:56.07 | asterisk99 | rezzen: After boot, it's not running .... asterisk -r tells me so |
| 00:56.56 | rezzen | and asterisk has been running following previous reboots? |
| 00:56.59 | asterisk99 | rezzen: *(&(&(*&*(&&^*&%^&$^ I had this working nicely until I moved the 2 FXO modules !!!! Zaptel is too easily broken!!! |
| 00:57.25 | asterisk99 | rezzen: (soory, I'm not (*&(*& at you) |
| 00:57.32 | rezzen | "had this working" doesn't answer my question. |
| 00:58.04 | asterisk99 | rezzen: Had this working => *was* working nicely for 2 days up until 30 minutes ago |
| 00:59.02 | rezzen | and you rebooted in that time, and asterisk was running after the reboot? |
| 00:59.08 | asterisk99 | rezzen: I wonder if I need to recompile zaptel |
| 00:59.23 | asterisk99 | rezzen: Yuppers ... rebboted nicely |
| 00:59.25 | rezzen | does everything work after you manually start asterisk? |
| 00:59.35 | *** join/#asterisk brockj49464 (n=brockj49@63.87.56.235) |
| 01:00.06 | asterisk99 | rezzen: No ... I get all these Remote UXIX Connection messages (never had 'em b4) |
| 01:00.22 | asterisk99 | rezzen: It's the TWILIGHT ZONE |
| 01:01.35 | *** join/#asterisk Gir19 (i=Gir@67.189.110.174) |
| 01:06.35 | asterisk99 | rezzen: This is weird... I found Unable to open /dev/zap/channel: Permission denied messages in /var/log/asterisk/messages |
| 01:07.50 | Gir19 | any of you recommend any type of training available for asterisk, but not the basic stuff, I am looking to send a technician of mine to learn more about asterisk, but he already knows a most of the basics. |
| 01:09.14 | *** join/#asterisk r0d3nt (n=RatMan@foster.stonedcoder.org) |
| 01:09.15 | airwaves | Good question... does Digium or a reseller do actual training? |
| 01:09.52 | Gir19 | airwave, I have found a few places that do training, but it is mainly for the basics. |
| 01:09.57 | justinu | there's dCAP |
| 01:10.01 | justinu | asterisk bootcamp |
| 01:10.02 | mogorman | astricon group does training airwaves |
| 01:10.45 | file | why in my day we didn't have that new fangled dCAP |
| 01:11.02 | mogorman | marko got dcapped a while ago |
| 01:11.11 | file | really? poor him... |
| 01:11.13 | file | ;) |
| 01:11.28 | mogorman | i didnt ask him what his score was |
| 01:11.41 | mogorman | id find it funny if he missed any |
| 01:11.49 | airwaves | Yeah, I'd be looking for *very* in-depth |
| 01:12.02 | Gir19 | I won't pay to have anyone get the dCAP, cause it just teaches you have to go through the bootcamp toeven qualify for the dCAP, it's just not worth it for the basic training. |
| 01:12.34 | file | it teaches you everything... but not how it all works... which is what I enjoy |
| 01:12.36 | mogorman | i think hard part of the training issue is very few people are really know asterisk |
| 01:12.57 | justinu | airwaves: read the code |
| 01:13.03 | file | I mean, who can honestly say they understand how masquerading works? |
| 01:13.28 | file | mogorman: reallllllly? |
| 01:13.32 | justinu | heh |
| 01:13.33 | mogorman | but i have to be drunk |
| 01:13.34 | mogorman | ^_^ |
| 01:13.37 | mogorman | yeah i do |
| 01:13.38 | Gir19 | lol |
| 01:13.38 | file | yeah ... it's warped |
| 01:13.42 | mogorman | its hard |
| 01:13.49 | mogorman | bitch to finda bug |
| 01:15.52 | airwaves | justinu: I am... but for * to move to more scalable environments... it really needs to get to the point where you don't need to read the source code to understand the architechture |
| 01:16.24 | mogorman | you dont need to read source to understand arch. |
| 01:16.54 | justinu | you do when there's no docs |
| 01:16.58 | airwaves | Not for the basics, mogorman.... but to get in-depth for certain things i've found you have to. |
| 01:17.08 | file | we do have doxygen docs ^_^ |
| 01:17.12 | mogorman | well thats true of anything.... |
| 01:18.05 | rt | "doxygen" and "docs" are mutually exclusive. |
| 01:18.17 | file | picky people |
| 01:18.40 | rt | doxygen is only good at documenting the things that aren't worth documenting. |
| 01:18.41 | mogorman | honestly |
| 01:18.43 | mogorman | its oss |
| 01:18.47 | airwaves | grrr... nobody is calling her.e |
| 01:18.53 | file | if you don't like it, patches are welcome |
| 01:18.57 | mogorman | if you want it handed on a plater |
| 01:19.06 | mogorman | you are gonna have to pay some one to train you |
| 01:19.25 | mogorman | sorry |
| 01:19.42 | justinu | doxygen still requires some basic C skillz to interpret |
| 01:19.43 | airwaves | mogorman: I know... and while it's just OSS I have zero expectations...because you get what you pay for. Honestly, it's not bad for OSS. |
| 01:19.45 | *** join/#asterisk ReD-MaN (i=redman@dhcp-0-2-b3-9a-4a-5b.cpe.quickclic.net) |
| 01:20.00 | file | we do have a doc directory for things too |
| 01:20.21 | airwaves | However... that being said.. just like Red Hat made it possible for enterprises to move towards linux as a viable data center platform... something will have to happen to * at some point for that to take place as well |
| 01:20.34 | mogorman | you can do that with be |
| 01:20.38 | mogorman | and digium services |
| 01:20.44 | mogorman | as well as thousands of asterisk pros |
| 01:20.46 | mogorman | to do it for you |
| 01:21.35 | airwaves | Just as a background... while I'm *very* new to *... I learn quickly, and I have well over 7 years of design-engineering-operational experience on major wide-scale VoIP deployments |
| 01:21.42 | justinu | other than the 10+ page functions and the terrible variable names, the code isn't that bad |
| 01:21.54 | justinu | airwaves has mad selius skillz tho |
| 01:21.55 | file | oh, there's bad code |
| 01:22.26 | file | there's... ugyy code... very very ugly |
| 01:25.56 | file | so mogorman, how goes? |
| 01:26.16 | mogorman | word airwaves old school |
| 01:26.21 | mogorman | im packing to move |
| 01:26.24 | mogorman | i hate packing |
| 01:26.28 | mogorman | more than anything |
| 01:26.34 | airwaves | I'm sitting in a radio station. |
| 01:26.36 | file | did you leave it till the last minute? |
| 01:26.36 | mogorman | if i could have it my way id just torch my old place |
| 01:26.47 | mogorman | you on the radio airwaves ? |
| 01:26.52 | mogorman | well i have 2 months |
| 01:26.58 | mogorman | but i am moving sunday/monday |
| 01:27.06 | airwaves | I'm floating in and out of the studio... mostly I'm on IRC and doing stuff in the back room. |
| 01:27.06 | file | ...right |
| 01:27.08 | mogorman | i own my apt for 2 months |
| 01:27.13 | mogorman | and i am moving to new one |
| 01:27.29 | file | to kp's? |
| 01:27.36 | mogorman | yes |
| 01:27.46 | file | cool cool |
| 01:27.52 | mogorman | where are you on the dial, city? airwaves |
| 01:28.02 | airwaves | in Washington DC |
| 01:28.47 | mogorman | what station? |
| 01:28.50 | mogorman | my sister is out ther |
| 01:28.51 | mogorman | e |
| 01:30.04 | airwaves | You don't want to even know the POS system we're using as the main office phone system |
| 01:30.09 | airwaves | I really need to revamp this place |
| 01:30.11 | *** join/#asterisk r0d3nt|m (n=RatMan@foster.stonedcoder.org) |
| 01:30.59 | mogorman | lol i do |
| 01:31.06 | mogorman | i have seen the worst |
| 01:31.58 | justinu | airwaves: what kinda circuit do you use to transmit the music to the transmitter? |
| 01:32.39 | mogorman | so airwaves you in radio? |
| 01:32.45 | airwaves | mogorman: You could say that |
| 01:32.46 | mogorman | or just at the station... |
| 01:32.51 | airwaves | Justinu: It varies based on the station..... |
| 01:32.52 | mogorman | whats the deal with hd radio? |
| 01:32.59 | airwaves | For many stations, we use a microwave STL.... |
| 01:33.00 | mogorman | is it gonna be the big thing? |
| 01:33.01 | justinu | airwaves: always been curious how that's done |
| 01:33.11 | mogorman | or is it just hype |
| 01:33.15 | justinu | HD radio isn't even CD quality |
| 01:33.19 | mogorman | i know |
| 01:33.23 | justinu | but it's better than stereo FM |
| 01:33.28 | mogorman | but i keep hearing about it hear in huntsville |
| 01:33.34 | airwaves | For a microwave STL it's usually a 900 MHZ point to point dish... usually one way... at about 25 watts or so... |
| 01:33.37 | mogorman | and i love am radio |
| 01:33.45 | mogorman | so if i could get a better receiver for that |
| 01:33.46 | mogorman | i might |
| 01:33.54 | airwaves | in the beginning it was just analog with mad frequency response... as in 15-20000 HZ |
| 01:33.55 | blkremedy | anyone here know how spa3k compare to tdm400p in terms of quality of sound. |
| 01:34.01 | airwaves | brb |
| 01:34.12 | mogorman | its all 8000khz mono blkremedy |
| 01:35.14 | airwaves | brb - on air |
| 01:35.20 | justinu | airwaves: there's a few bearer cap types in ISDN for transmitting audio over the PSTN |
| 01:35.22 | justinu | like 7khz |
| 01:35.29 | justinu | and some other bonded channel type things |
| 01:38.17 | *** join/#asterisk ibob63 (n=hp@bb-87-82-15-9.ukonline.co.uk) |
| 01:38.58 | ibob63 | is there any alternative to using mpg123 - I am using ubuntu and can't work out how to install mpg123. |
| 01:39.46 | justinu | yeah - you can get format_mp3 |
| 01:39.54 | justinu | or convert your MoH to ulaw (probably best idea) |
| 01:39.57 | airwaves | will go over all of that momentarily .... |
| 01:40.08 | airwaves | we're running a contest right now - i'm manmning the phones |
| 01:40.12 | justinu | heh |
| 01:40.22 | russellb | airwaves: doing a radio show, huh? |
| 01:40.26 | russellb | airwaves: talking about Asterisk? :) |
| 01:41.05 | ibob63 | justinu - what do you mean convert MoH to ulaw? |
| 01:41.49 | justinu | convert your music on hold to ulaw format |
| 01:41.54 | justinu | so you don't need mpg123 or format_mp3 |
| 01:42.24 | ibob63 | hum... |
| 01:42.35 | airwaves | converting to mu-law in advance is highly desireable |
| 01:42.44 | lunaphyte | is it acceptable to have a voicemailbox number be the same as an extension? |
| 01:43.08 | justinu | it's common |
| 01:43.14 | justinu | common practice |
| 01:43.46 | ibob63 | how to i go about doing the conversion? |
| 01:43.55 | lunaphyte | it kind of seemed logical - i sort of expected to see mention of if everywhere i looked, so when i didn't, i got curious. |
| 01:44.36 | justinu | ibob63: a program called sox. and no, no one here will help you figure it out. |
| 01:44.37 | ibob63 | it is just a config? or do I need to do some file conversions? |
| 01:45.00 | ibob63 | okay, I will do some more research. Thanks for your help :) |
| 01:45.04 | justinu | np :) |
| 01:47.26 | ManxPower | ~docs |
| 01:47.27 | jbot | methinks docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 01:47.31 | airwaves | back in a few minuts - i'm on air right now |
| 01:47.43 | justinu | ~sox |
| 01:47.44 | jbot | hmm... sox is Sound Processing Tool. URL: http://sox.sourceforge.net/ |
| 01:54.07 | lunaphyte | could i solicit some examples of voicemail contexts? |
| 01:54.23 | justinu | i only use one |
| 01:54.33 | justinu | haven't needed to get fancy yet |
| 01:54.50 | lunaphyte | i'm wondering what sort of applications might call for multiple. |
| 01:55.09 | justinu | when you need duplicate mailbox numbers for different people/organizations |
| 01:55.21 | lunaphyte | ahh. good example. |
| 01:57.26 | lunaphyte | what about besides the need to duplicate numbers? |
| 01:57.35 | justinu | i can't think of anything else |
| 01:57.42 | justinu | i'm hardly a voicemail expert tho |
| 01:57.54 | lunaphyte | more so than i, i'm sure :) |
| 01:58.21 | lunaphyte | [vociemail] it is then :) |
| 01:58.29 | lunaphyte | er - [voicemail] |
| 01:58.34 | *** join/#asterisk bjohnson_ (n=bjohnson@i216-58-62-67.cybersurf.com) |
| 01:59.37 | ibob63 | stupid question - what do you call the commands you type into the phone to pickup answer machine messages? |
| 01:59.48 | justinu | DTMF |
| 01:59.56 | lunaphyte | numbers? |
| 02:00.00 | justinu | ~dtmf |
| 02:00.01 | jbot | DTMF: Dual Tone Multi-Frequency. The technical term describing Touch Tone dialing. Basically the combining of two tones, one low frequency and one high frequency. |
| 02:00.15 | ibob63 | no - nothing that complicated. |
| 02:00.38 | lunaphyte | passcode? |
| 02:00.38 | ibob63 | basically, I am trying to work out how to transfer calls etc. |
| 02:00.45 | *** join/#asterisk backblue (n=moo@87-196-36-43.net.novis.pt) |
| 02:01.04 | justinu | oh, VSI |
| 02:01.16 | justinu | vertical service activation codes |
| 02:01.17 | ibob63 | <PROTECTED> |
| 02:01.23 | justinu | like *69 |
| 02:01.37 | ibob63 | yeah that sounds right |
| 02:01.46 | justinu | http://www.voip-info.org/wiki/view/Asterisk+vertical+service+activation+codes |
| 02:02.21 | ibob63 | thats the one - thanks |
| 02:02.40 | nettie | guys, anyone know if the linksys PAP2 supports t.38 please? it's not clear.. I read that settint it to ReInvite will enable t.38 is that true? thanx in adv. |
| 02:05.24 | *** join/#asterisk Ad-Hoc (n=Nimbus@ppp37-adsl-149.ath.forthnet.gr) |
| 02:08.54 | lunaphyte | so why would someone be willing to hand out "free" dids? |
| 02:10.00 | ManxPower | to collect money for usage |
| 02:10.12 | lunaphyte | i.e. ipkall... |
| 02:10.40 | lunaphyte | no strings attached, except for sometimes sketchy audio |
| 02:21.52 | *** join/#asterisk angom_h (n=angom@red-corp-201.130.135.223.telnor.net) |
| 02:22.30 | *** join/#asterisk frans-th (n=frans@202.155.120.247) |
| 02:22.39 | frans-th | hi anyone success install asterix in ubuntu? |
| 02:23.40 | lunaphyte | how can i default VoiceMailMain to use the calling party's extension? |
| 02:25.03 | *** part/#asterisk ibob63 (n=hp@bb-87-82-15-9.ukonline.co.uk) |
| 02:25.43 | airwaves | ok i'm back for a few minuts.... |
| 02:26.01 | *** join/#asterisk Ad-Hoc (n=Nimbus@ppp30-adsl-138.ath.forthnet.gr) |
| 02:26.06 | airwaves | So... to answer the earlier question... microwave STL's... analog...have huge frequency response... better than the 25-15Khz FM has.... |
| 02:26.18 | airwaves | typically 10-25 watts @ 900 mhz |
| 02:26.31 | airwaves | most STL's now are digital though, using AES/EBU as a standard.... |
| 02:26.56 | airwaves | there are a good number of stations that use telco circuits as a studio-to-transmitter link.... |
| 02:27.28 | robin_sz | ummm |
| 02:27.36 | airwaves | in almost all cases a clear channel T-1 is used.... |
| 02:27.52 | *** join/#asterisk RoyK (n=roy@10.80-203-106.nextgentel.com) |
| 02:28.33 | airwaves | that allows for the full 44.1 khz to be sent digitally without compression... for years..even back in 80's/early 90's a QEI Catlink would be used just like a regular telco mux...except for broadcast audio.... |
| 02:29.01 | robin_sz | 44.1? |
| 02:29.18 | robin_sz | audio response ?? |
| 02:30.11 | robin_sz | you are confusing the bit rates with the frequency response ... |
| 02:31.07 | robin_sz | fm stations roll off the response at around 12 to 15 khz |
| 02:31.55 | robin_sz | because the stereo pilot signal is on 19khz |
| 02:33.14 | robin_sz | and .. the US uses 44.1khz as a sampling frequency, because it ties in well with NTSC base rates, the rest of the planet uses 48khz, to tie in with PAL |
| 02:33.51 | *** join/#asterisk a1fa (n=a1fa@207.210.210.202) |
| 02:34.42 | airwaves | the one that someone mentioned in here about ISDN BRI.... BRI's are still actually used for live remote broadcasts, and for slinging audio around ... TelosZephyr...among others... |
| 02:34.50 | airwaves | they're dedicated hardware MP3 codecs... |
| 02:35.20 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
| 02:35.21 | airwaves | robin: Nope.. I'm not confusing it. Usually they're PCM, 44.1K, Stereo.... these are the studio-to-transmitter links |
| 02:36.01 | airwaves | and yes, there is a roll-off at around 15 KHZ. The reason for that is because the stereo pilot.... (not the actual stereo information, just the boolean signal to say 'hey..i'm in stereo') that pilot is transmitted at 19 KHZ |
| 02:36.16 | a1fa | anyone? any voip apps for sidekick? |
| 02:36.18 | airwaves | The actual stereo data is transmitted beginning at 38 KHZ..... |
| 02:36.26 | airwaves | a1fa: No. Not that I've seen, |
| 02:36.37 | a1fa | sick |
| 02:36.38 | a1fa | ') |
| 02:37.04 | airwaves | From 0 - 15 KHZ is L+R, 38K is L-R... and the 38Khz is transmitted using compressed single sideband. FM Stereo radios do simple algebra to extract the left and right channels. |
| 02:37.22 | airwaves | HD Radio uses IBOC (in band on channel) and is a longer conversation |
| 02:38.26 | airwaves | oops... just saw robin mentioned the 19khz pilot as well.... |
| 02:38.54 | *** part/#asterisk zaf (n=tfournet@ip68-226-162-240.lf.br.cox.net) |
| 02:39.58 | *** join/#asterisk airwaves (n=support@c-68-50-99-18.hsd1.va.comcast.net) |
| 02:40.15 | airwaves | alrighty |
| 02:40.46 | a1fa | it would be sweet to have VOIP on T-MObile SideKick |
| 02:40.52 | a1fa | $29.99/month |
| 02:40.58 | a1fa | unlimited transfer |
| 02:41.30 | airwaves | a1fa: The way the backend at most cell carriers doesn't make that simple right now |
| 02:41.55 | file | the sidekick doesn't directly expose you to the internet either... uses a proxy system |
| 02:42.06 | a1fa | file: you can ssh out |
| 02:42.07 | airwaves | Yeah... don't even get me on Danger Inc right now |
| 02:42.16 | airwaves | I use the SSH on the sidekick all the time |
| 02:42.21 | a1fa | so i dont see why you cant use SIP |
| 02:42.40 | airwaves | a1fa: The processor on the sidekick 2 is not very powerful. |
| 02:42.46 | airwaves | Most of the work is done on the back end |
| 02:42.56 | a1fa | that sucks |
| 02:43.02 | a1fa | i dont have any use to ssh out |
| 02:43.18 | a1fa | i need more ;P |
| 02:43.40 | a1fa | i want VIPN |
| 02:44.02 | a1fa | it only costs $800 |
| 02:44.42 | a1fa | http://www.vipn.ch/uk/produitsuk.htm |
| 02:44.42 | airwaves | the sidekick 2 merely a thin client |
| 02:44.53 | airwaves | a1fa: Get a PPC phone |
| 02:45.00 | airwaves | those have enough power... |
| 02:45.06 | [av]bani | ... |
| 02:45.07 | airwaves | and there are VOIP clients out there |
| 02:45.13 | a1fa | i want this AXIA |
| 02:45.29 | a1fa | AXIA 308 |
| 02:45.50 | *** join/#asterisk ptblank (n=MURDER1@68-169-161-61.lmdaca.adelphia.net) |
| 02:46.04 | a1fa | 802.11g is a must |
| 02:50.35 | *** join/#asterisk MstlyHrmls (n=mh@melbourne.mostly-harmless.ca) |
| 02:51.55 | airwaves | brb - on air |
| 02:54.42 | IOscanner | anyone know what change in 1.2? Meetme conference rooms don't work in flash operator panel and also meetme2 UI. It seems maybe the commands changed or options change to get this information |
| 03:02.22 | *** join/#asterisk firestrm (n=no@S01060013105c1c69.gv.shawcable.net) |
| 03:03.15 | firestrm | anyone here payed with callerid incoming over zap channels? Its been so long, i cant remember how to set it up.. |
| 03:10.42 | *** join/#asterisk Ad-Hoc (n=Nimbus@ppp55-adsl-211.ath.forthnet.gr) |
| 03:10.47 | Katty | mew? |
| 03:12.00 | firestrm | woof |
| 03:14.25 | firestrm | i hate it when i learn new stuff.. i forget old stuff.. |
| 03:16.28 | *** join/#asterisk Splas (n=jwb@brooklyn.paravolve.net) |
| 03:20.07 | [av]bani | yay |
| 03:33.19 | *** join/#asterisk file (n=joshnet@mctnnbsa24w-142167032200.pppoe-dynamic.nb.aliant.net) |
| 03:45.22 | Katty | Netgeeks: be careful of jawline. it's still sore. |
| 03:46.03 | rob0 | ouch! But you're still alive. :) |
| 03:47.21 | Katty | yes, and on many a painkiller. |
| 03:49.47 | *** join/#asterisk bmg505 (n=leon@dsl-165-138-131.telkomadsl.co.za) |
| 03:53.56 | *** join/#asterisk bkw__ (n=bkw_@adsl-70-234-37-160.dsl.tul2ok.sbcglobal.net) |
| 03:55.50 | Qwell | bkw__: !! |
| 03:55.55 | Qwell | (one ! per _) |
| 03:57.21 | bkw__ | what!!!!!!!!!! |
| 03:57.29 | Qwell | umm |
| 03:57.31 | Qwell | hi |
| 03:57.35 | bkw__ | HO |
| 03:57.54 | Qwell | I so am not |
| 03:59.02 | Qwell | silly google, broke google video |
| 04:04.14 | *** join/#asterisk angom_w (n=angom@red-corp-201.130.135.223.telnor.net) |
| 04:08.49 | *** join/#asterisk angom_w (n=angom@red-corp-200.79.145.219.telnor.net) |
| 04:17.15 | *** join/#asterisk sevard (i=sev@24-179-181-160.dhcp.dlth.mn.charter.com) |
| 04:17.38 | *** join/#asterisk Flauto (n=zhao@71.194.194.48) |
| 04:17.45 | Flauto | hi there |
| 04:17.47 | Flauto | one question |
| 04:18.00 | Flauto | i don't want people to call cell numbers in taiwan |
| 04:18.06 | Flauto | so i tried to block it |
| 04:18.10 | Flauto | by using |
| 04:18.29 | Flauto | 0118869XXXXXXXX,1,Congestion |
| 04:18.35 | Flauto | it really does not work |
| 04:18.38 | Flauto | why? |
| 04:19.18 | Flauto | while i have _011886.,1,Dial(blahblahblah.30) |
| 04:19.44 | Abydos313 | mosely is beating the crap out of vargas :)) |
| 04:22.01 | Flauto | okay |
| 04:22.05 | Flauto | got it work now |
| 04:26.00 | *** join/#asterisk coppice (n=chatzill@78.193.17.210.dyn.pacific.net.hk) |
| 04:29.42 | *** join/#asterisk Ad-Hoc (n=user@ppp55-adsl-211.ath.forthnet.gr) |
| 04:35.35 | x86 | hmm... |
| 04:35.58 | x86 | if i put a call on hold, it tells me "Starting music on hold" and immediately says "Stopping music on hold" |
| 04:36.02 | x86 | why would it do that? |
| 04:36.30 | x86 | then like 30 seconds later (after the call is no longer on hold), it gives a message about music on hold being scheduled in the past or something |
| 04:43.40 | *** join/#asterisk iq (n=iq@71-214-4-12.omah.qwest.net) |
| 04:44.48 | *** join/#asterisk arctic_import (n=arctic@209-112-170-172-cdsl-rb1.nwc.acsalaska.net) |
| 04:47.32 | x86 | <PROTECTED> |
| 04:47.40 | x86 | <PROTECTED> |
| 04:48.41 | x86 | Feb 25 22:46:57 NOTICE[4577]: res_musiconhold.c:507 monmp3thread: Request to schedule in the past?!?! |
| 04:50.54 | sevard | http://www.google.com/search?hs=cds&hl=en&lr=&safe=off&client=firefox-a&rls=org.mozilla%3Aen-US%3Aofficial&q=res_musiconhold.c%3A507+monmp3thread%3A+Request+to+schedule+in+the+past&btnG=Search |
| 04:54.16 | *** join/#asterisk coppice (n=chatzill@212.197.17.210.dyn.pacific.net.hk) |
| 04:55.29 | *** join/#asterisk florz (n=florz@2001:1a50:503c:0:0:0:0:1) |
| 04:56.41 | *** join/#asterisk zaf (n=tfournet@ip68-226-162-240.lf.br.cox.net) |
| 05:04.23 | *** join/#asterisk hellop (n=hellop@cpe-72-130-252-68.hawaii.res.rr.com) |
| 05:09.15 | hypa7ia | hey |
| 05:09.26 | hypa7ia | oops, wrong windows :) |
| 05:11.04 | *** join/#asterisk asterisk99 (n=astguy@modemcable169.194-130-66.mc.videotron.ca) |
| 05:11.46 | asterisk99 | anyone here running Asterisk under userid other than root???? |
| 05:14.30 | x86 | asterisk99: i am |
| 05:14.56 | x86 | asterisk 4569 0.0 1.2 17368 7820 pts/2 Sl 19:02 0:08 /usr/sbin/asterisk -p -U asterisk -vvvg -c |
| 05:16.34 | asterisk99 | x86: assuming u have a zaptel card, how do you set the permissions for /dev/zap? |
| 05:16.36 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 05:17.35 | asterisk99 | x86: Asterisk won't start up until it gets permission to access files in /dev/zap, which I did with chown, but next reboot, it's reset back to root:root |
| 05:25.23 | x86 | err |
| 05:25.32 | x86 | modify your /etc/devfs.conf or /etc/udev.conf |
| 05:25.53 | x86 | make it auto-create the device as owned by your asterisk user |
| 05:27.46 | *** join/#asterisk Guest^DJ (i=me@211.24.146.12) |
| 05:30.21 | asterisk99 | ...and by 'device' that's the files in /dev/zap right? |
| 05:31.00 | asterisk99 | (sorry 4 delay ... multitasking badly) |
| 05:32.53 | arctic_import | anyone here using gentoo with asterisk? I think I'm having some udev permission problems. If I start asterisk with the user/group asterisk I get errors |
| 05:33.12 | arctic_import | asterisk99: I think we are having the same problem. |
| 05:34.00 | asterisk99 | sat night ... what can I say??? |
| 05:34.13 | arctic_import | if I chown -R asterisk:asterisk /dev/zap it seems to work correctly. |
| 05:34.30 | arctic_import | however if I unload wtcdm and reload it its goofed again. |
| 05:34.54 | arctic_import | if I run as root all is good. |
| 05:34.59 | [av]bani | _Sam-- around? |
| 05:36.02 | arctic_import | my udev rules are setting the mode to 0660 and group to dialout. adding user asterisk to the dialout group doesn't fix the problem. perhaps the mode needs to be 0770? |
| 05:36.35 | arctic_import | asterisk99: are you using udev as well? What distor? |
| 05:36.40 | arctic_import | distro rather. |
| 05:40.55 | arctic_import | I'm getting errors about unable to load channel chan_zap.so |
| 05:41.36 | arctic_import | seems to be a much bigger issue because even root is having problems now. |
| 05:42.34 | arctic_import | guess I'll try a power cycle of the box. Seems like the card has freaked out. |
| 05:43.52 | *** join/#asterisk airwaves (n=support@c-68-50-99-18.hsd1.va.comcast.net) |
| 05:44.54 | arctic_import | Looks like my chanel 1 is gone now. |
| 05:47.18 | *** join/#asterisk airwaves (n=support@c-68-50-99-18.hsd1.va.comcast.net) |
| 05:51.30 | asterisk99 | ubuntu 5.10 ... (I think that's udev) |
| 05:55.02 | arctic_import | changing my mode to 0770 fixed my problem |
| 06:22.23 | *** join/#asterisk shido6 (n=shido6@d221-68-216.commercial.cgocable.net) |
| 06:23.53 | Abydos313 | quick question, would two x100p pci cards function the same as a spa3k? |
| 06:25.34 | shido6 | err |
| 06:25.39 | shido6 | u want an FXS in there dont ya |
| 06:25.48 | shido6 | two x100p's would act like 2 FXO's |
| 06:26.03 | Abydos313 | yeah |
| 06:26.39 | Abydos313 | i want to use * as my home answering machine with long distant extensions |
| 06:26.58 | shido6 | kewl |
| 06:27.15 | shido6 | how many phone lines do u have coming into the place that you can use for your * ? |
| 06:27.20 | Abydos313 | so when i'm overseas i can get my phone calls and all calls i make back here are local since i'll be dialing from my local pstn |
| 06:27.30 | Abydos313 | just one |
| 06:27.40 | shido6 | picked out a voip provider yet? |
| 06:27.55 | Abydos313 | nope, i actually won't need one |
| 06:28.13 | shido6 | err K. |
| 06:28.26 | shido6 | so when you are overseas you're using a softphone or ata? |
| 06:28.28 | Abydos313 | i don't want to dial thru provider. i just want to be able to have an extension to my house |
| 06:28.47 | Abydos313 | i was going to get ata adapter for both sides |
| 06:29.25 | Abydos313 | i thought spa3k for * location and cheaper one for remote location. but i really wanted to run iax and those are supported |
| 06:29.53 | shido6 | kewl. |
| 06:30.49 | Abydos313 | we already pay 20 bucks for unlimited calls on the house phone. so this extension will be dialing from that number so all calls will be free from overseas to family and friends here |
| 06:31.22 | Abydos313 | only testing so far with softphones. trying to get advise on equipment to solve what i want to do |
| 06:49.49 | *** join/#asterisk znoG (n=gs@109-130-89-200.fibertel.com.ar) |
| 06:51.15 | arctic_import | Abydos313: I'm doing this. |
| 06:51.47 | arctic_import | Abydos313: Works really well now when I'm out of town I use a softphone to have local dialtone its great. |
| 06:52.00 | [av]bani | Abydos313: i think you should ask the asgard. i think they have a lot of experience with softphones. |
| 06:53.17 | arctic_import | I need to find a voip provider that works with asterisk and is cheap though. My LD isn't a very good package, any recommendations on a byod provider? |
| 06:54.30 | [av]bani | ~itsp |
| 06:54.32 | jbot | itsp is, like, Internet Telephony Service Provider. An ITSP is a "VoIP Phone Company" |
| 06:54.59 | [av]bani | http://www.voip-info.org/wiki-VOIP+Service+Providers |
| 06:56.44 | Abydos313 | [av]bani hehe |
| 06:57.09 | *** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin) |
| 06:57.34 | Abydos313 | [av]bani would you get iax capable ata adapters because both locations will be behind nat |
| 06:58.39 | *** join/#asterisk shido6 (n=shido6@d221-68-216.commercial.cgocable.net) |
| 07:11.12 | [av]bani | :< |
| 07:17.03 | coppice | I assume the T in ITSP stands for terrible |
| 07:17.50 | *** join/#asterisk Gand_DJ (n=gandalf@stnbmb01dc1-238-197.dynamic.mts.net) |
| 07:23.48 | *** join/#asterisk r0d3nt (i=nobody@wsip-24-234-241-145.lv.lv.cox.net) |
| 07:27.24 | *** join/#asterisk laichzeit (n=ubermens@c4-163-1.pta.dial-up.net) |
| 07:28.26 | laichzeit | hi all, I have a Digium TDM400P, when I make an outgoing call I get a terribly loud distorting noise, does anyone know what is causing this? |
| 07:32.59 | *** join/#asterisk chr|s_ (n=chris@217.171.51.175) |
| 07:34.17 | coppice | maybe you shouldn't call Steve Ballmer :-) |
| 07:35.37 | rob0 | ba-da BOOM ching! |
| 07:39.17 | *** join/#asterisk AxelBp (n=alexlsf@200.12.40.89) |
| 07:41.26 | AxelBp | HI guys...i'm new on this, any tutorial for the asteristic for windows? |
| 07:41.51 | chr|s_ | Axel, I think it involves running VMWare |
| 07:42.03 | Gand_DJ | pfft.... use virtual PC :) |
| 07:42.08 | Gand_DJ | or virtual server 2005 |
| 07:42.28 | chr|s_ | I think you can download a free vm of asterisk@home though |
| 07:42.31 | chr|s_ | soi I would use vmware |
| 07:42.32 | chr|s_ | not those |
| 07:42.34 | WasPhantom | heh - VPC on a mac is painful |
| 07:42.47 | AxelBp | i have an xp |
| 07:42.49 | laichzeit | seriously, no one has had shit with asterisk 2.3 and zaptel 1.2.3 with the Digium TDM400P cards? |
| 07:42.51 | chr|s_ | AxelBp, google VMWare 'asterisk at home' |
| 07:43.17 | AxelBp | why cant be install in the same os? |
| 07:43.22 | chr|s_ | linux only |
| 07:43.24 | chr|s_ | that is why |
| 07:43.44 | AxelBp | ohhhhh ok, i don't know any linux |
| 07:43.48 | chr|s_ | but I read a good article about running it in a VM, you download one isngle file and viola it works |
| 07:43.54 | chr|s_ | AxelBp, good time ot learn |
| 07:44.11 | AxelBp | i have vmware but can't do any linux |
| 07:44.40 | chr|s_ | AxelBp, google what I said, and download that file, you only need ot use the web based front end for it |
| 07:44.47 | chr|s_ | it is simplicity itself |
| 07:45.01 | *** part/#asterisk Guest^DJ (i=me@211.24.146.12) |
| 07:45.08 | Gand_DJ | http://www.vmwarez.com/2006/01/asteriskhome-voip-opensource-pbx.html |
| 07:45.18 | *** join/#asterisk kizmet (n=kizmet@freematrix/sponsor/kizmet) |
| 07:45.30 | kizmet | *yawn* |
| 07:46.00 | chr|s_ | Yeah that one! :-) jolly fine link |
| 07:46.10 | AxelBp | great |
| 07:46.13 | AxelBp | let me check |
| 07:46.18 | kizmet | http://www.woot.yawn |
| 07:46.44 | Gand_DJ | lol. I run *@Home in virtual server 2005, which runs on win2k3 ent server. |
| 07:47.00 | chr|s_ | Gand_DJ, which you run on WINE |
| 07:47.01 | kizmet | Gand_DJ, uh ? rofl |
| 07:47.06 | chr|s_ | on an old 32mb 386 |
| 07:47.10 | chr|s_ | with a creaky hard disk |
| 07:47.23 | Gand_DJ | haven't had it up in like 6 months though, since I moved in with fiance... she uses that for WinXP (dual boot system) |
| 07:47.31 | Gand_DJ | It's athlon 750mhz w/ 512mb ram |
| 07:47.37 | chr|s_ | Gand_DJ, you shoudl see a doctor about that |
| 07:47.52 | chr|s_ | "haven't had it up in like 6 months though, since I moved in with fiance..." << Too easy man |
| 07:47.52 | Gand_DJ | :P |
| 07:48.13 | chr|s_ | sorry must be british humour I think |
| 07:48.13 | Gand_DJ | I knew what you were refering to lol |
| 07:48.18 | Abydos313 | i have asterisk@home 2.5 running in vmware at work. runs fine in xp |
| 07:48.44 | AxelBp | can i do a voip pbx like a mini vonage? |
| 07:48.49 | Gand_DJ | why that guy recommends Idefisk for a softphone is funny.. just use eyebeam or x-lite |
| 07:48.57 | chr|s_ | hrm, I like the idea of being able to quickly get the system up again if problems |
| 07:49.00 | AxelBp | i mean just voip no pstn? |
| 07:49.11 | kizmet | i have debian w/ asterisk running on vmware esx server.... |
| 07:49.18 | chr|s_ | revert to yesterdays snapshot, but how many 'vm' work arounds do you have to do? does it affect performance a lot? |
| 07:49.33 | Abydos313 | kizmet pulls the trump card in this game.heh |
| 07:49.53 | Abydos313 | kizmet 2.51 or newer? |
| 07:50.00 | Gand_DJ | the * box I have setup now (in virtual PC) runs just fine. |
| 07:50.06 | kizmet | Abydos313, nfi actually |
| 07:50.06 | kizmet | lol |
| 07:50.14 | Gand_DJ | host is this pc.... Athlon 2600+ w/ 1GB ram |
| 07:50.23 | Abydos313 | nice |
| 07:50.36 | laichzeit | damn I should upgrade for that Oblivion game |
| 07:50.37 | chr|s_ | same here (2800+ 1GB) |
| 07:50.41 | kizmet | Abydos313, im assuming the latest... |
| 07:50.46 | kizmet | brb |
| 07:50.50 | chr|s_ | wondering if that is what I should do all along... |
| 07:51.16 | Gand_DJ | I'm thinking of putting my old P200 to use, and install asterisk@home onto it for a dedicated pbx system |
| 07:51.16 | chr|s_ | I mean, even running the asterisk@home in a vm ON linux |
| 07:51.21 | chr|s_ | because I like the idea of redundancy |
| 07:51.24 | chr|s_ | and snapshotting |
| 07:51.29 | chr|s_ | as long as DB is on another macine |
| 07:51.29 | Abydos313 | vm's are great for testing |
| 07:51.32 | *** join/#asterisk Guest^DJ (i=me@211.24.146.12) |
| 07:51.48 | chr|s_ | does asterisk die in the middle of things? |
| 07:51.53 | Guest^DJ | hi guys, how do i verify that i have successfully installed h323 ? |
| 07:52.03 | Gand_DJ | Test videophone? |
| 07:52.05 | chr|s_ | say you have 15 phone calls happening at once, can it just stop? lag? kill? |
| 07:52.11 | Gand_DJ | unless I'm thinking of wrong codec |
| 07:53.04 | chr|s_ | can you install asterisk at home in a more normal way? I am worried aboutkilling my linux with it, I want it to install ONTO my linux, not decimate it |
| 07:53.17 | chr|s_ | like download a complete a@h stack |
| 07:53.37 | laichzeit | chr|s_, I've recently done an asterisk@home install on gentoo |
| 07:54.14 | laichzeit | chr|s_, everything is working except outgoing calls :/ |
| 07:55.07 | laichzeit | the ast@home installation scripts are pretty much a big hack |
| 07:55.18 | Gand_DJ | hrm, I have a copy of VMware around here somewhere... lol.. not sure on which HD, or what version of software |
| 07:57.28 | Gand_DJ | looking at vmware website.... seems esx is just a virtual server competitor. I'll stick to VS2k5 :) |
| 07:58.29 | Abydos313 | i haven't tried that yet |
| 07:59.31 | Gand_DJ | I like it. I had 1 domain controller, 1 backup domain controller, and an exchange server all running at the same time (3 VM's)... and *@home |
| 07:59.52 | Abydos313 | one cpu machine? |
| 07:59.55 | Gand_DJ | yep |
| 07:59.59 | Gand_DJ | small test network |
| 08:00.15 | Gand_DJ | my 750mhz athlon |
| 08:00.25 | Gand_DJ | worked fine for the most part |
| 08:01.06 | Gand_DJ | I had Live Communication Server setup also (2003). |
| 08:01.13 | Gand_DJ | tried to get 2005 working, but that was a bit of a pain |
| 08:01.30 | Abydos313 | how was the communications server? |
| 08:02.02 | Gand_DJ | Was nice running a private IM server. Had to use Windows Messenger for it though. |
| 08:02.14 | Gand_DJ | Also 2003 had to work within the network. Didn't allow outside linking. |
| 08:02.23 | Abydos313 | so the same as runnig IM off exchange2k? |
| 08:02.26 | Gand_DJ | 2005 allows outside connects through proxies |
| 08:02.33 | Gand_DJ | yeah. |
| 08:02.42 | Abydos313 | what protocol? |
| 08:03.01 | Abydos313 | could it accept sip or iax connects? |
| 08:03.08 | laichzeit | does anyone know what's wrong if the zaptel channels don't show up in the Trunk section of the Flash Operator Panel in Asterisk@home? |
| 08:03.19 | *** join/#asterisk spunz_ (n=spunz@h081217096096.dyn.cm.kabsi.at) |
| 08:03.24 | Gand_DJ | SIP |
| 08:03.53 | Abydos313 | nice, can it dialout either thru a moden or ata converter? |
| 08:04.28 | Gand_DJ | Don't know. Just tested linking windows messenger to the Live Comm Server. |
| 08:04.41 | Gand_DJ | It might work through VPN, but I didn't have a vpn partner to test with |
| 08:04.44 | Abydos313 | ok, well it was worth asking |
| 08:05.11 | Gand_DJ | Same thing as using messenger to sign into your hotmail / .net acct |
| 08:05.19 | Gand_DJ | but this is a local server |
| 08:05.50 | laichzeit | where do you get genzaptelconf ? |
| 08:06.35 | Abydos313 | right now i have no ata adapters. i have my asterisk box running with ztdummy loaded and i have .25c credit for outgoing calls..hehe so it's really in the works. my real asterisk server sits behind nat so i want to use iax. any suggestions |
| 08:07.01 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
| 08:07.38 | Abydos313 | i hate to buy spa3k sipura and find out i have problems with remote clients and sip |
| 08:08.19 | Gand_DJ | link to FWD w/ iax.... or another provider via SIP? |
| 08:08.25 | Gand_DJ | iax would be better for nat |
| 08:08.30 | Gand_DJ | just have to open 1 port |
| 08:08.41 | Abydos313 | i like that alot better |
| 08:09.00 | Gand_DJ | then just have your softphone / ata link to * |
| 08:09.06 | Abydos313 | can i link you to a device i was looking at? give me an oppinion |
| 08:09.28 | *** join/#asterisk websae (n=icechat5@adsl-64-149-206-121.dsl.milwwi.sbcglobal.net) |
| 08:09.36 | websae | how's everyone doing tonight? |
| 08:09.39 | Gand_DJ | for ata? I have been thinking of getting an spa2002 |
| 08:09.58 | Abydos313 | that device only supports sip |
| 08:09.58 | Gand_DJ | also to link my home phone to *@Home, maybe the spa3k |
| 08:10.17 | Gand_DJ | well if you run *, sip is fine from ata <-> * |
| 08:10.26 | Gand_DJ | then * <-> ISP is iax |
| 08:10.37 | websae | whose your termination provider Grand? |
| 08:10.52 | Abydos313 | i wanted to run iax all the way around |
| 08:10.55 | Gand_DJ | I am signed up with a few free accts... |
| 08:11.09 | Gand_DJ | fwd, sipphone, iaxtel, |
| 08:11.11 | Gand_DJ | hrm |
| 08:11.19 | Gand_DJ | yak, firefly, |
| 08:11.22 | Abydos313 | http://www.x100p.com/products_2.htm |
| 08:11.42 | Abydos313 | this is the one i've been looking at. all the features of spa3k plus iax support |
| 08:12.00 | Abydos313 | anyone use this device or hear about it? |
| 08:12.12 | websae | looks great |
| 08:12.18 | Gand_DJ | What's the purpose of going all iax? iax is really only beneficial over nat from internet |
| 08:13.08 | Abydos313 | Gand_DJ one of my remote clients will be my dad, he wants to get calls from his house in the US to his house overseas and be able to call anyone around here free |
| 08:13.13 | websae | IAX is great for those who don't like dealing with NAT |
| 08:13.38 | websae | where is his house overseas? |
| 08:13.40 | websae | what country? |
| 08:13.44 | Abydos313 | israel |
| 08:13.50 | websae | ahh |
| 08:13.52 | websae | understandable |
| 08:13.59 | websae | are you implementing this system for him? |
| 08:14.04 | Abydos313 | yeah |
| 08:14.17 | Gand_DJ | websae, that's what I said...lol.. nat from over the internet. |
| 08:14.25 | *** join/#asterisk fm (n=7457@217.17.237.7) |
| 08:14.30 | Gand_DJ | linking ata to internal * has no nat issues |
| 08:14.35 | *** join/#asterisk Tene (n=tene@poipu/supporter/slacker/tene) |
| 08:14.52 | Gand_DJ | since * deals with nat to internet. |
| 08:14.56 | Gand_DJ | but in this case.... yeah |
| 08:15.07 | Gand_DJ | going from usa to * in overseas |
| 08:15.18 | Gand_DJ | there's that IAXy box |
| 08:15.21 | Abydos313 | Gand_DJ are you sure. with softphone i was having sound issues with sip and no issues with iax. i have 5060-5061 udp and 10000-10100 udp forwarded to server |
| 08:15.56 | fm | is it possible to create a fax server which deals with email-to-fax and fax-to-email service over internet using asterisk. Can anyone point me to a good doc/artcle regarding this. |
| 08:16.25 | websae | yes it is fm |
| 08:16.28 | Gand_DJ | Abydos313, depends on link..... if * AND ata is on local side of router... then there is no nat issues since you just give the 192 IP of * to ata |
| 08:16.33 | websae | google asterisk fax |
| 08:16.50 | Gand_DJ | if ata is at someone's house.... and you run * at your house behind router.... then IAX is good for linking those 2 |
| 08:16.55 | Gand_DJ | since you ONLY need to open 1 port |
| 08:17.37 | websae | no rtp packets going on |
| 08:17.38 | websae | that's nice |
| 08:19.01 | Gand_DJ | heh.... the spa3k is better then this box |
| 08:19.09 | Gand_DJ | from what I read |
| 08:19.35 | Gand_DJ | From how I read it.... the FXO is only a pass-through for when voip isn't working. (life line) |
| 08:19.40 | Abydos313 | both sides will be behind cable or dsl |
| 08:19.41 | websae | sipura makes a great ATA :) |
| 08:19.55 | Gand_DJ | I don't think you can program the FXO to forward the PSTN call to * or anything |
| 08:19.55 | websae | all of my clients use sipura ata adapters for VoIP in their houses |
| 08:20.19 | Gand_DJ | websae, how's the SPA-2002 compared to the old SPA-2000 (or was it 2001) |
| 08:20.20 | Abydos313 | and is your * behind nat? |
| 08:20.53 | *** join/#asterisk mrtwister|mobile (n=andrius@cable-10-68.cgates.lt) |
| 08:21.07 | websae | put the SPA 2000 in front of router/firewall |
| 08:21.09 | websae | and you'll be fine |
| 08:21.35 | Gand_DJ | For an IAXy box... if he just wants to make calls, he can get one of those IAXy boxes... or if he wants to link his PSTN to it as well (doesn't have 2 phones).. then get this other iax box |
| 08:21.53 | Gand_DJ | if this box is cheaper then iaxy, I might just get this box |
| 08:22.05 | websae | i would look for reviews on that box |
| 08:22.10 | Gand_DJ | yeah |
| 08:22.16 | Abydos313 | i don't want junk |
| 08:22.18 | websae | i have yet to to try it |
| 08:22.20 | websae | exactly |
| 08:22.56 | Abydos313 | i've been patient looking around for weeks now and hoping for someone who knows what's up to give me advise on hardware |
| 08:23.17 | Gand_DJ | websae, yet to try it.... directed in regards to spa2002 compared to spa2000? |
| 08:24.05 | Abydos313 | * will sit here on dsl and connected to regular landline. remote client will be on cable modem in israel and also behind nat. i know he will be able to call here and out, but will his extension ring with sip |
| 08:24.17 | websae | yes |
| 08:24.21 | websae | easily can set that up |
| 08:24.30 | websae | have nat=yes |
| 08:24.56 | Abydos313 | so what devices would you suggest for all this? spa3k and 2002 or something |
| 08:25.11 | websae | how many phone lines does he need ? |
| 08:25.22 | Abydos313 | he just needs one |
| 08:25.31 | Gand_DJ | SPA-1002? |
| 08:25.40 | Gand_DJ | you don't need SPA-3000 for him to call out |
| 08:25.46 | Gand_DJ | alot of money for nothing :) |
| 08:25.50 | Abydos313 | he just wants to be able to use his phone there and have the house dialout here |
| 08:26.09 | Abydos313 | Gand_DJ i won't have a sip provider |
| 08:26.44 | websae | any sipura ata will do |
| 08:26.59 | Gand_DJ | well you'll need the spa3k for your place then.. for him to call your *, and then route out through your spa3k to pstn call |
| 08:27.23 | Abydos313 | that is what i thought |
| 08:28.01 | Gand_DJ | isn't there a way to link 2 atas together directly? |
| 08:28.04 | Gand_DJ | without needing * |
| 08:28.05 | Abydos313 | later on i plan on adding a sip provider and basically having a second line in the house |
| 08:28.24 | Abydos313 | Gand_DJ is there? |
| 08:28.35 | Gand_DJ | That's what I'm asking in channel. |
| 08:28.42 | Gand_DJ | I think there is a way to directly link 2 ata together |
| 08:29.16 | Gand_DJ | since sipura ata has that SSL cert you can use for direct ATA <-> ATA encrypted talking |
| 08:29.17 | Abydos313 | i want more extensions and i want to learn asterisk. so i do want the server :) |
| 08:29.30 | [av]bani | ... |
| 08:29.33 | Gand_DJ | yeah. I have the server here running in virtual environment. |
| 08:29.39 | Gand_DJ | I am looking into getting some atas |
| 08:30.06 | Abydos313 | [av]bani do you agree with spa2k and any other sipura for what i want to do? |
| 08:30.10 | Abydos313 | spa3k |
| 08:30.11 | Gand_DJ | probably spa3000 for linking our house to *... and then the FXS on the spa3k & also an spa2k2 setup in basement for call routing |
| 08:30.55 | Abydos313 | what do you mean by call routing? |
| 08:30.57 | Gand_DJ | sure I could just get a cordless phone w/ 3-4 handsets lol |
| 08:31.14 | coppice | i think only the 2100 does T.38 if that is important to you. also, most of the sipuras have weedy DSP, and can only do one channel of G.729 |
| 08:31.46 | Abydos313 | coppice so what hardware would you suggest? |
| 08:31.50 | Gand_DJ | call routing.. as in press 1 for me, press 2 for fiance, etc.... or maybe I will just setup all phones to ring at same time |
| 08:31.56 | websae | but you can do faxing on any ulaw connection |
| 08:32.36 | *** join/#asterisk krisguy (n=krisguy@h216-170-039-057.adsl.navix.net) |
| 08:32.37 | coppice | Abydos313: why do you expect suggestions? only you know what you need |
| 08:32.43 | coppice | websae: wrong |
| 08:32.50 | websae | nope |
| 08:32.59 | websae | we do it just fine, all the time |
| 08:33.04 | websae | using ulaw |
| 08:33.15 | Abydos313 | i know the features i want, but i was hoping to get help on what works well and don't want to buy junk |
| 08:33.46 | coppice | websae: which means exactly what? that with one particular piece of kit, in one particular installation, it hangs together by the skin of its teeth and does something useful? |
| 08:33.50 | Gand_DJ | coppice... he's wanting to setup an ATA in isreal to link to * in usa... and then route through * to PSTN for calling |
| 08:34.05 | coppice | its recommendations like that which cause so many support problems with VoIP |
| 08:34.28 | websae | nope, on any ATA implemented that allows ULAW---it is supported |
| 08:34.45 | websae | we have tested on many different ATA adapters |
| 08:34.58 | coppice | websae: you've just been lucky |
| 08:35.14 | websae | no, we just use a non compressed codec |
| 08:35.17 | Gand_DJ | doesn't the spa2002 have fax firmware upgrade on it? |
| 08:35.30 | websae | that's the key to getting the fax :) |
| 08:35.40 | websae | g711 |
| 08:36.18 | [av]bani | Abydos313: ? |
| 08:36.45 | Abydos313 | hi |
| 08:37.17 | coppice | well, for a start G.711 is compressed, but we'll ignore that. Try http://www.soft-switch.org/foip-with-real-atas.html and http://www.soft-switch.org/foip.html |
| 08:39.52 | coppice | Abydos313: if you want one call at a time, or you don't care about bit rate, you'll find the sipuras fine. if you expect to use two channels of G.729 you won't like them. as I said, only you know what you need |
| 08:40.23 | Abydos313 | i would like to have two channels in the future |
| 08:40.41 | Abydos313 | and i don't mind buying the codecs |
| 08:40.57 | [av]bani | Abydos313: what you want to do? |
| 08:41.52 | *** join/#asterisk krasavin (n=chatzill@ip-217-24-113-226.parma.ru) |
| 08:41.54 | Abydos313 | have asterisk hooked up to my home phone for in/out calls and have softphone extenion at work and ata extenstion in israel so dad can be reached on an extension and he call out to US for free |
| 08:42.10 | krasavin | hi all! |
| 08:42.17 | Abydos313 | hi |
| 08:42.21 | coppice | The snag is that pretty much every piece of VoIP kit currently available has a lot of limitations. there isn't something you can really recommend that just does everything people want to their satisfaction |
| 08:42.40 | [av]bani | Abydos313: dad won't like the lag |
| 08:42.54 | krasavin | do anybody next: * + d-link dvg-1104TH? |
| 08:42.58 | trixter | even if it did everything on a technical level that would make it cost prohibitive which means that there is a group that still wouldnt be happy with it |
| 08:43.15 | Abydos313 | damn, so is there a diff setup you'd suggest |
| 08:44.33 | [av]bani | Abydos313: ata in israel will always suffer from lag no matter what vendor |
| 08:45.09 | trixter | yeah the mossad needs a little bit of time to process the audio :P |
| 08:45.12 | Abydos313 | so maybe just a sip accout and adapter is next best thing |
| 08:45.17 | Abydos313 | haha |
| 08:45.42 | coppice | [av]bani: is israeli internet really that bad |
| 08:46.14 | [av]bani | coppice: i am assuming israel pings to us are not very low, since nowhere in europe to us is low |
| 08:46.21 | krasavin | i have PBX LG LDK-100 and i want to get to call from ip-phone to the PBX |
| 08:47.30 | coppice | [av]bani: depends what you call bad. tried .ph :-) |
| 08:48.00 | [av]bani | coppice: i've tried .jp and .in, both are quite bad |
| 08:48.03 | Gand_DJ | Abydos313, you could always setup your dad with an FWD acct, and you setup FWD acct also for *. he calls your * number |
| 08:48.13 | [av]bani | coppice: its like walkie-talkie |
| 08:48.38 | Abydos313 | Gand_DJ so how exactly would that increase the quality |
| 08:48.54 | [av]bani | Abydos313: it wouldnt |
| 08:48.59 | Gand_DJ | he's going through fwd server (might have a server in europe?) |
| 08:49.05 | Gand_DJ | or another free provider? |
| 08:49.08 | Gand_DJ | in his area |
| 08:49.22 | Gand_DJ | fwd peers with alot of voip companies |
| 08:49.31 | Gand_DJ | or sip broker maybe / voxalot |
| 08:49.46 | [av]bani | coppice: UK is bad enough on pstn to oftentimes get uncontrollable echo even the telcos cant fix... |
| 08:50.01 | Abydos313 | how's this. if he just has a vonage type service here and takes the atapter to israel will the call quality be good? |
| 08:50.14 | [av]bani | Abydos313: quality good... lag no |
| 08:50.17 | krasavin | can anybody help me? |
| 08:50.32 | [av]bani | Abydos313: you will always face the lag issue simply because of geographic location of israel |
| 08:50.38 | [av]bani | Abydos313: there's no way to get around that |
| 08:50.58 | Abydos313 | ok so pretty close to what a regular phone line sounds like |
| 08:51.00 | Gand_DJ | have him ping your IP address and see what his response is. |
| 08:51.12 | Gand_DJ | if you are behind router, setup your pc for dmz or hook direct to modem |
| 08:51.16 | [av]bani | Abydos313: better than regular phone line for audio quality, just the lag which you have to ge tused to |
| 08:51.27 | Abydos313 | ok |
| 08:51.38 | [av]bani | Abydos313: basically, wait for the other end to stop talking completely, then speak a complete sentence, then stop |
| 08:51.45 | coppice | [av]bani: geographic location? Its not exactly a long way at speed_of_light/refractive_index_of_fibre :-) |
| 08:51.56 | Abydos313 | he is not over there yet. he wont' leave for a few weeks, kinda wanted to have a device picked out and config'd for him to take with him |
| 08:52.35 | [av]bani | coppice: add 30% for copper, then 5-10ms per router |
| 08:53.05 | [av]bani | Abydos313: dont forget israel ISPs are widely blocking voip |
| 08:53.16 | Gand_DJ | might not block iax port |
| 08:53.20 | Abydos313 | your kidding right |
| 08:53.22 | [av]bani | maybe not |
| 08:53.30 | Gand_DJ | if so, change voip port in * |
| 08:53.36 | [av]bani | Abydos313: nope. most of middle east is blocking voip |
| 08:53.49 | [av]bani | Gand_DJ: blocking sip bodies... youll need tls to get round that |
| 08:53.51 | Gand_DJ | might have him use a softphone to see how it sounds first |
| 08:53.51 | Abydos313 | i had no idea |
| 08:54.08 | coppice | most of the planet is blocking VoIP, unless its the telco's VoIP |
| 08:54.24 | [av]bani | i think pakistan is just about the only country that isnt |
| 08:54.35 | Gand_DJ | canada doesn't block voip |
| 08:54.39 | Gand_DJ | we don't block anything :) |
| 08:54.40 | [av]bani | in the middle east |
| 08:54.55 | Gand_DJ | coppice said most of the planet lol |
| 08:55.01 | FLeiXiuS | Canada doesn't even have a military, pshh :-) |
| 08:55.13 | Gand_DJ | I don't think USA blocks either |
| 08:55.14 | [av]bani | theyre protecting old govt monopolies |
| 08:55.23 | coppice | most of the middle east. much of asia. some of south america |
| 08:55.24 | Abydos313 | wow, i had no idea. this blows my whole project down the tubes |
| 08:55.43 | Gand_DJ | Abydos313, he would have to try using iax softphone maybe |
| 08:55.48 | *** join/#asterisk lorinc (n=ang@caracas-0192.adsl.interware.hu) |
| 08:55.51 | [av]bani | most of them havent broken up/deregulated/etc the telcos.. theyre like 20-30 years behind |
| 08:55.59 | [av]bani | iax might get around it |
| 08:56.11 | Gand_DJ | firefly uses iax |
| 08:56.16 | coppice | its not just a monopoly thing |
| 08:56.19 | Gand_DJ | too bad xlit or eyebeam doesn't |
| 08:56.43 | Abydos313 | how did you get a copy of eyebeam? i didn't see downloads for that one |
| 08:56.47 | [av]bani | well, some of it is police state govts, like syria |
| 08:56.59 | coppice | where things have been deregulated the government has often franchised a few people, and they get pissed if the governent lets that paid for franchise fall apart |
| 08:57.34 | [av]bani | with israel its probably just cronysim :)) |
| 08:57.42 | [av]bani | cronyism |
| 08:57.53 | Abydos313 | i'd believe it |
| 08:58.05 | [av]bani | it may depend on the particular isp as well |
| 08:58.13 | coppice | you mean packets are delivered by a cron job? :-\ |
| 08:58.16 | Gand_DJ | http://www.freshtel.net/firefly/download/ |
| 08:58.23 | [av]bani | but i have seen complaints about voip being widely blocked in israel by ISPs |
| 08:58.59 | Gand_DJ | eyebeam is hard to find a copy of ;) |
| 08:59.06 | [av]bani | im actually somewhat suprised pakistan doesnt block it |
| 08:59.10 | Gand_DJ | I have an older copy |
| 09:00.27 | [av]bani | Abydos313: http://blog.tmcnet.com/blog/tom-keating/voip/israel-blocks-voip.asp |
| 09:00.55 | *** join/#asterisk bartpbx (n=bartpbx@p54B03C0F.dip0.t-ipconnect.de) |
| 09:01.01 | bartpbx | hellp |
| 09:02.02 | Abydos313 | wow, now this really sucks, i really don't want to buy equipment for nothing |
| 09:02.22 | Gand_DJ | Abydos313... have him test softphone first :) |
| 09:03.37 | Abydos313 | good idea |
| 09:04.03 | PakiPenguin | [av]bani, pakistan doesnt block what? |
| 09:04.08 | PakiPenguin | voip is blocked for starters :) |
| 09:04.43 | Abydos313 | how do you get around it |
| 09:05.07 | [av]bani | PakiPenguin: i see lots of people using spa3k and voip in pakistan.. how you guys get around it? |
| 09:05.24 | [av]bani | PakiPenguin: and i see pakistan being one of the most active countries for voip, next to india |
| 09:08.48 | bartpbx | I have a question about variables and agi |
| 09:09.00 | bartpbx | anyone workin with fastagi / agi here? |
| 09:10.07 | bartpbx | I have Problems reading DIALEDTIME via AGI if the call was canced |
| 09:10.47 | bartpbx | but the way I understand the variable DIALEDTIME it schuld be availible even when a call was canced or notaswered |
| 09:11.46 | *** join/#asterisk PakiPenguin_ (n=uppal@linuxpakistan/admin/pakipenguin) |
| 09:15.18 | [av]bani | o_O |
| 09:15.30 | [av]bani | PakiPenguin: i see lots of people using spa3k and voip in pakistan.. how you guys get around it? |
| 09:15.35 | [av]bani | PakiPenguin: and i see pakistan being one of the most active countries for voip, next to india |
| 09:24.48 | *** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin) |
| 09:29.47 | *** join/#asterisk PakiPenguin_ (n=uppal@linuxpakistan/admin/pakipenguin) |
| 09:31.31 | *** part/#asterisk krasavin (n=chatzill@ip-217-24-113-226.parma.ru) |
| 09:37.01 | *** join/#asterisk akrall (i=user@201.152.155.171) |
| 09:37.18 | akrall | Anybody using unicall and iaxmodem/hylafax? |
| 09:45.37 | *** join/#asterisk DrData (n=michael@p54B259B7.dip.t-dialin.net) |
| 09:47.43 | *** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin) |
| 09:52.30 | *** join/#asterisk r0d3nt (i=nobody@wsip-24-234-241-145.lv.lv.cox.net) |
| 09:54.29 | *** part/#asterisk akrall (i=user@201.152.155.171) |
| 09:54.35 | *** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin) |
| 09:57.53 | *** join/#asterisk TallAndy (n=dfa@81-178-205-180.dsl.pipex.com) |
| 09:59.38 | [av]bani | ... |
| 10:00.26 | a1fa | god |
| 10:00.34 | a1fa | i am trying to create a freakin dns record |
| 10:00.37 | a1fa | and its not working |
| 10:00.41 | a1fa | i am trying to take over * |
| 10:01.28 | Gand_DJ | dns record in linux / *? |
| 10:01.48 | a1fa | bind |
| 10:02.01 | a1fa | i am trying with zone "*" |
| 10:02.06 | a1fa | but that not working |
| 10:03.02 | Gand_DJ | I've only messed with DNS stuff in windows server. know nothing on linux :) |
| 10:07.34 | a1fa | gdd |
| 10:07.37 | a1fa | its 4 am |
| 10:07.43 | a1fa | and i am still fucking with it |
| 10:07.43 | FLeiXiuS | 5am here. |
| 10:07.49 | FLeiXiuS | Whats the problem? |
| 10:07.58 | a1fa | trying to take over the root zone |
| 10:08.07 | a1fa | i am making a capturing portal |
| 10:08.25 | FLeiXiuS | And the problem is? |
| 10:08.28 | a1fa | bind |
| 10:08.35 | a1fa | zone "." |
| 10:08.53 | FLeiXiuS | http://www.catb.org/~esr/faqs/smart-questions.html |
| 10:09.17 | a1fa | lol |
| 10:09.19 | a1fa | thanks |
| 10:09.32 | a1fa | it wont take over the zone file |
| 10:10.59 | a1fa | and no error |
| 10:18.33 | a1fa | since when a NS record can not be an address? |
| 10:19.46 | *** join/#asterisk sulex (n=sulex@pdpc/supporter/active/sulex) |
| 10:20.47 | *** join/#asterisk mexuar-tim (n=mexuar-t@host-212-158-206-61.bulldogdsl.com) |
| 10:23.03 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
| 10:24.36 | *** join/#asterisk DrData (n=michael@p54B259B7.dip.t-dialin.net) |
| 10:31.33 | Vyeperman | http://the-edge.blogspot.com/2005/10/worldss-smallest-ip-pbx-at-astricon.html |
| 10:47.27 | *** join/#asterisk stoffell (n=stoffell@d51A4D52E.access.telenet.be) |
| 10:49.30 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 10:49.54 | *** join/#asterisk niZon (n=ilt@S010600080db4ab60.wp.shawcable.net) |
| 10:53.12 | *** join/#asterisk mtgh (n=chatzill@dsl093-001-038.det1.dsl.speakeasy.net) |
| 10:56.03 | x86 | hmm |
| 10:56.25 | x86 | i want extensions defined in one context to be able to dial extensions defined in another context, but not vice-versa |
| 10:56.31 | x86 | is that possible? |
| 11:00.27 | *** join/#asterisk nitram (i=foo@superblob.com) |
| 11:10.17 | *** join/#asterisk af_ (n=af@ip-165-17.sn2.eutelia.it) |
| 11:12.00 | tzafrir | x86, Goto can come in handy for that sometimes |
| 11:26.50 | *** join/#asterisk lithi (n=interp3@67.71.44.152) |
| 11:30.08 | lithi | Can someone help me compile app_rxfax/app_txfax on 1.2.4.. after I do patch <apps_Makefile.patch I get a 1 out of 2 hunks FAILED -- saving rejects to file Makefile.rej and I dont know what to do from there.. |
| 11:30.38 | stoffell | lithi, read the patchfiles, you should be able to figure out how to do it yourselves.. |
| 11:30.58 | lithi | yea I dont undestand what the patchfiles are saying |
| 11:31.02 | lithi | thats the problem |
| 11:32.19 | *** join/#asterisk Shoragan (n=shoragan@d072.apm.etc.tu-bs.de) |
| 11:32.21 | lithi | stoffell: Could I pastbin the Makefile.rej and maybe you could tell me? |
| 11:32.41 | stoffell | lithi, you can, but i think it's better to start from scratch and do the makefile actions manually |
| 11:33.07 | lithi | could you explain what the actions I need to make are? |
| 11:33.16 | lithi | cause I just dont understand how to read the makefile at all |
| 11:33.18 | stoffell | sure, pastebin the Makefile, we'll scroll along |
| 11:34.44 | lithi | http://pastebin.ca/43213 thats the apps_Makefile.patch |
| 11:35.43 | stoffell | okay, first open the file Makefile |
| 11:36.01 | lithi | ok |
| 11:36.13 | stoffell | then find a line looking similar like line 6 in the pastebin (APPS+-app_osp..) |
| 11:36.40 | lithi | ok |
| 11:37.35 | lithi | replace that section with lines 9-11? |
| 11:37.35 | stoffell | good, now if u look in the pastebin, lines 9-12 have a + in front of them.. |
| 11:37.52 | stoffell | this means, you should ADD these lines right after the section that is mentioned before.. |
| 11:37.56 | lithi | oh ok |
| 11:38.03 | stoffell | so: add 9-12 "after" the endif lines.. |
| 11:38.09 | lithi | without the ++ i assume |
| 11:38.16 | lithi | the + symbols |
| 11:38.21 | stoffell | indeed, it should read: ifneq ($(wildcard $(CR... |
| 11:38.38 | stoffell | then you will see it will start to look like the rest of the makefile |
| 11:38.48 | *** join/#asterisk my007ms (n=my@213.158.171.162) |
| 11:38.51 | my007ms | hello all |
| 11:38.58 | stoffell | after adding these lines, just yell :) |
| 11:39.00 | stoffell | hello |
| 11:39.12 | lithi | ok added and ready for next step |
| 11:39.43 | stoffell | good, now just look further in pastebin, same thing, but different lines.. find lines 21 and 22 in your makefile |
| 11:39.57 | stoffell | then, after this section, add the "new" lines (app_rxfax..) etc.. |
| 11:40.21 | stoffell | so, lines 24-29 should go in-between the app_curl and app_sql_postgres sections |
| 11:40.33 | my007ms | i have problem with config TDM card |
| 11:41.19 | my007ms | when i do lspci from any distore i can't see pci card but from livecd i see TDM card |
| 11:41.33 | my007ms | any one have idea |
| 11:41.35 | my007ms | ?? |
| 11:41.49 | stoffell | what livecd did you use? |
| 11:42.45 | Gand_DJ | http://www.freshtel.net/products/3010.php |
| 11:42.56 | my007ms | LFS |
| 11:43.53 | blkremedy | how is the spa3000 compared to the tdm400p? |
| 11:44.43 | my007ms | the probem mybe i think from my intel Expres chipst |
| 11:44.58 | my007ms | any one have idea |
| 11:44.58 | my007ms | ? |
| 11:45.38 | stoffell | my007ms, try it in other pc if possible? |
| 11:45.56 | my007ms | but i busy this server for this card |
| 11:46.04 | my007ms | it's TDM2424E |
| 11:46.13 | my007ms | as u see i need strong PC |
| 11:46.29 | stoffell | yes, but just to test.. |
| 11:48.28 | lithi | stoffell: I got a curl problem then I installed the curl devel package which fixed that now I get a Makefile:110: *** missing separator. Stop. |
| 11:48.49 | my007ms | i see that new kernel have support this kind of chipst |
| 11:49.04 | my007ms | but is this from kernel or udev |
| 11:49.05 | my007ms | ?? |
| 11:49.07 | stoffell | ouch, something went wrong while editing the makefile, can u pastebin it? |
| 11:50.20 | lithi | stoffell: I think I fixed it, it seemed to want a [tab] then $(CC) $(SOLINK) -o $@ ${CYGSOLINK} $< ${CYGSOLIB} $(CURLLIBS) |
| 11:50.27 | lithi | insted of 7 or 8 spaces |
| 11:50.41 | stoffell | great, correct :) |
| 11:50.51 | lithi | stoffell: Thanks so much btw |
| 11:51.10 | stoffell | no problem:) |
| 11:52.21 | x86 | anyone here from UK? |
| 11:53.25 | *** join/#asterisk RoyK (n=roy@10.80-203-106.nextgentel.com) |
| 11:53.33 | stoffell | x86, belgium, so we're neighbours :p |
| 11:55.04 | *** part/#asterisk RoyK (n=roy@10.80-203-106.nextgentel.com) |
| 11:57.17 | NotFreak | and the other neighbours as well .nl :P |
| 11:57.39 | lithi | stoffell: I get libspandsp.so.0: cannot open shared object file: No such file or directory when I try to start asterisk.. any idea where it wants that file cause its in /usr/local/lib |
| 11:57.53 | stoffell | hehe NotFreak, looks like a family gatherin :) |
| 11:57.57 | NotFreak | yup |
| 11:58.23 | NotFreak | bleh i need to get outside i think been inside for 4 days now :S |
| 11:58.34 | NotFreak | maybe i should go walk the dog or something |
| 11:59.09 | stoffell | don't stay out too long NotFreak, it's freezin' |
| 11:59.20 | NotFreak | yup it does |
| 11:59.35 | stoffell | lithi, hmm.. but the file exists? (you have installed spandsp) |
| 11:59.40 | NotFreak | that's why i stayed inside and because i was trying to hack some hardware |
| 12:00.13 | lithi | stoffell: Yea |
| 12:00.29 | lithi | stoffell: Yea, /usr/local/lib/libspandsp.so.0 |
| 12:00.31 | my007ms | stoffell this is from kernel or udev |
| 12:00.32 | my007ms | ?? |
| 12:00.41 | NotFreak | i've been crazy enough to implement a polarity reversal detection on my X100P card :S but the friggin thing is still unable to detect the DTMF callerid that is used in .nl |
| 12:00.47 | stoffell | sorry my007ms, i don't have any idea |
| 12:01.25 | my007ms | :) never mind |
| 12:01.49 | my007ms | did u know software to make sure that i have PCI 2.2 |
| 12:01.50 | my007ms | ? |
| 12:01.59 | my007ms | the manual say nothing |
| 12:02.15 | stoffell | NotFreak, nice :) |
| 12:02.45 | stoffell | lithi, weird, you used latest versions of spandsp and app_rx/txfax? |
| 12:02.55 | NotFreak | stoffell yup nice now it says polarity reversal detected but after 2 seconds i get DTMFCID timed out waiting for ring. :S |
| 12:03.10 | stoffell | you're half way :) |
| 12:03.20 | [av]bani | ... |
| 12:03.32 | NotFreak | seems it doesn't monitor the channel in onhook mode i guess |
| 12:03.39 | [av]bani | NotFreak: probably easier to buy a spa-3000 and be done with it |
| 12:03.50 | NotFreak | yeah sure true that |
| 12:04.02 | NotFreak | but it's a challenge to figure it out on x100p :P |
| 12:04.04 | [av]bani | x100p is the fxo from hell |
| 12:04.12 | NotFreak | and since i'm a student and on a really low budget heh |
| 12:05.41 | [av]bani | iirc someone was working on pre-ring cid to support countries which use it |
| 12:05.52 | [av]bani | you can find it on bugs.digium.com |
| 12:06.08 | lithi | stoffell: ah I needed to add /usr/local/include to my ld.so.conf (duh) |
| 12:06.10 | NotFreak | yeah i know about that |
| 12:06.19 | [av]bani | i dont think you need polarity reversal, you just need to always monitor for dtmf |
| 12:06.21 | NotFreak | but seems to only work for v23 |
| 12:06.28 | [av]bani | and buffer them, then use teh ring |
| 12:06.34 | NotFreak | true |
| 12:06.42 | lithi | stoffell: anyways thanks again for all the help, now I have a working asterisk fax machine |
| 12:06.52 | [av]bani | just collect digits at all times, then when a ring arrives, send the last buffer collected |
| 12:07.25 | NotFreak | http://lusyn.com/resources/asterisk/usehist.htm that's what this hack does |
| 12:07.51 | stoffell | lithi, good luck ;) |
| 12:07.57 | [av]bani | yeah, use that |
| 12:09.12 | [av]bani | or you can just move to another country which has different CID signalling :) |
| 12:09.27 | NotFreak | hahah |
| 12:09.35 | *** join/#asterisk __AK__ (n=ak@blm93-1-82-231-201-7.fbx.proxad.net) |
| 12:09.42 | stoffell | try belgium, it's close :p |
| 12:09.42 | __AK__ | hi |
| 12:09.43 | [av]bani | or ring up the telco and ask them if they can change your signalling to post-ring |
| 12:10.19 | [av]bani | stoffell: belgium uses chocolate & waffles CID signalling |
| 12:10.25 | robin_sz | re hi |
| 12:10.31 | stoffell | lol |
| 12:10.36 | NotFreak | stoffell yeah belguim is close indeed i can see the border from here haha |
| 12:10.42 | [av]bani | also: belgium does not exist |
| 12:10.45 | __AK__ | i'm trying to setup enum, i already can do enumlook for number@domain but I would like to do lookup for name@domaine |
| 12:10.53 | robin_sz | belgians have a lot of inner problems |
| 12:10.55 | *** join/#asterisk mexuar-tim (n=mexuar-t@host-212-158-206-61.bulldogdsl.com) |
| 12:11.00 | stoffell | lol |
| 12:11.02 | [av]bani | http://www.zapatopi.net/belgium/ |
| 12:11.03 | stoffell | yeah |
| 12:11.13 | robin_sz | they face very difficult choices that the rest of us do not have to make. |
| 12:11.35 | robin_sz | such as ... |
| 12:11.44 | robin_sz | to be a Flem or a Walloon ;) |
| 12:11.47 | robin_sz | some choice |
| 12:11.52 | stoffell | that's an easy choice ! :) |
| 12:12.33 | [av]bani | robin_sz: they are the root of all evil.. EU HQ is in brussels... |
| 12:12.34 | robin_sz | Flems speak French |
| 12:12.47 | robin_sz | unless there any foreigners about |
| 12:12.51 | [av]bani | robin_sz: also nato HQ |
| 12:12.55 | robin_sz | in which case they pretend to only speak flemish |
| 12:12.56 | stoffell | robin_sz, flemish is dutch :) |
| 12:13.12 | robin_sz | not quite, but close enough |
| 12:13.13 | __AK__ | anyone has succeed puting name@domaine in a dns for an enumlookup? |
| 12:13.17 | stoffell | so you're assuming we are dutch? okay , can live with that :) |
| 12:13.28 | NotFreak | hehehe |
| 12:13.32 | [av]bani | instead of speaking 1 silly language, you speak 3 |
| 12:13.39 | x86 | hmm |
| 12:13.54 | stoffell | that's the only positive thing 'bout it [av]bani ,we learn 3 languages |
| 12:14.01 | x86 | i have a free inbound DID that is supposed to forward to my FWD SIP account, but it does not seem to work |
| 12:14.14 | x86 | i actually have 2 of them, one in the US and another in the UK |
| 12:14.35 | x86 | +1-360-227-6548 (US) and +44-871-3094407 (UK) |
| 12:15.05 | x86 | when the numbers are called, the caller gets some message basically saying the user is unavailable, yet i see nothing at all coming into asterisk |
| 12:15.21 | x86 | i have set verbose 9999, iax2 debug, sip debug |
| 12:15.23 | robin_sz | did you register in sip.conf? |
| 12:15.34 | x86 | i registered the FWD account, yeah |
| 12:15.54 | x86 | do i have to do something different to register also for the DID's? |
| 12:16.14 | robin_sz | I have no idea what a FWD account is |
| 12:16.21 | robin_sz | but uou have to have a line |
| 12:16.27 | x86 | FreeWorldDialup |
| 12:16.38 | x86 | i can get incoming calls from FWD just fine |
| 12:16.50 | robin_sz | you need a register line for each incoming sip |
| 12:16.58 | x86 | would it be different? |
| 12:17.06 | robin_sz | differetn from what? |
| 12:17.19 | x86 | from just the FWD register? |
| 12:17.29 | x86 | i need addition registers for each DID? |
| 12:17.36 | robin_sz | well of course. if it was the same it would just register with FWD twice |
| 12:17.46 | x86 | hmm |
| 12:17.51 | x86 | how would i represent that? |
| 12:17.57 | x86 | my username and password is still on FWD :P |
| 12:18.06 | robin_sz | eh? |
| 12:18.21 | robin_sz | what is that supposed to mean? |
| 12:19.27 | robin_sz | anyway .. its all clearly documented in sip.conf |
| 12:19.59 | x86 | well this is how it's setup right now (my asterisk) <--(SIP)--> (FWD) <--> [ (DID provider A) (DID provider B) ] |
| 12:20.12 | x86 | i register once with FWD, that works fine |
| 12:20.20 | x86 | both DID providers are forwarding to FWD |
| 12:20.23 | robin_sz | right thats one incoming sip account ... |
| 12:20.30 | x86 | *nod* |
| 12:20.34 | robin_sz | no idea what that means |
| 12:20.43 | x86 | ugh |
| 12:20.47 | x86 | anyone else here have a clue? |
| 12:20.57 | robin_sz | both DID providers are forwarding to FWD? .. .sorry, no clue what that means |
| 12:21.04 | x86 | right |
| 12:21.13 | x86 | they are SIP forwarding to FWD (FreeWorldDialup) |
| 12:21.25 | robin_sz | normally, your DID provider just pgives you a SIP account to register |
| 12:21.26 | x86 | actually, I'm doing IAX2 to FWD |
| 12:21.33 | x86 | not these |
| 12:21.37 | x86 | they are forwarders only |
| 12:21.37 | robin_sz | how odd |
| 12:21.42 | x86 | they're free ;) |
| 12:21.48 | x86 | ipkall.com |
| 12:21.50 | x86 | ipstar.us |
| 12:21.50 | robin_sz | never come across that concept |
| 12:21.59 | robin_sz | sounds useless |
| 12:22.42 | x86 | they could send it directly to my asterisk, but i didnt set it up like that |
| 12:23.02 | robin_sz | "send it" ? |
| 12:23.20 | x86 | trunk it, whatever you want to call it... they say forward |
| 12:23.29 | *** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin) |
| 12:23.55 | robin_sz | if your * made a registration with a SIP peer, I can see how that would work . |
| 12:24.08 | x86 | right |
| 12:24.14 | x86 | the SIP peer in this case is FWD |
| 12:24.24 | robin_sz | I presume you have made some arrangement with FWD to have it register with the ipkall peer then |
| 12:24.41 | x86 | hmm |
| 12:24.45 | x86 | it didnt tell me i needed to |
| 12:24.53 | robin_sz | well, does it work? |
| 12:25.06 | x86 | no |
| 12:25.07 | x86 | heh |
| 12:25.11 | robin_sz | well, ... |
| 12:25.14 | x86 | would insecure=very help? |
| 12:26.29 | robin_sz | I just use sipgatre.co.uk, and have my * box establish the registration |
| 12:26.31 | robin_sz | also free |
| 12:26.38 | robin_sz | sipgate.co.uk |
| 12:26.55 | robin_sz | at least that way, it will still work when FWD is down |
| 12:30.43 | *** join/#asterisk chr|s_ (n=chris@217.171.51.175) |
| 12:30.53 | chr|s_ | Winkie, hey man how is it going? |
| 12:33.13 | robin_sz | sigh .. poxy GXP2000 ... thsi si REALLY beginning to bug me |
| 12:33.57 | Falle | robin_sz: why? the gxp is great :) |
| 12:41.32 | *** join/#asterisk tuxinator_linux (n=tuxinato@70-32-106-248.ontrca.adelphia.net) |
| 12:41.49 | x86 | robin_sz: can you try dialing that UK number i gave earlier? |
| 12:42.12 | x86 | +44-871-3094407 |
| 12:43.17 | *** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239) |
| 12:44.28 | my007ms | did any one know abut PCI Express |
| 12:49.39 | Gand_DJ | lol. AGP / PCI replacement |
| 12:51.07 | *** join/#asterisk zotz (n=zotz@24.244.133.10) |
| 12:52.54 | *** join/#asterisk Curus (n=Curus@x1-6-00-12-17-df-1b-be.k182.webspeed.dk) |
| 12:53.08 | chr|s_ | my007ms, been out for ages right? |
| 12:53.32 | *** join/#asterisk Gunnar (n=gunnar@34.80-203-209.nextgentel.com) |
| 12:54.49 | robin_sz | x86: dialling now ... |
| 12:55.36 | [av]bani | robin_sz: at least you didnt pay $200 for the gxp2000 ... i have expensive phones equally as bug ridden :/ |
| 12:55.57 | [av]bani | in fact,more buggy :| |
| 12:55.58 | robin_sz | x86: "we are sorry, you have reached a number that has been disconnected or is no longer in service" |
| 12:56.17 | robin_sz | I have 50 Snoms .. all seem fine |
| 12:56.23 | [av]bani | for $80 phone the bugs are annoying. for $200 the bugs are compltely unacceptable |
| 12:56.40 | robin_sz | 25 320s, 25 360s ... no major problems |
| 12:56.48 | [av]bani | except crashes? |
| 12:56.53 | robin_sz | the GXP is blank after less than 5 seconds now :( |
| 12:57.02 | robin_sz | no, no cashes AFAIK |
| 12:57.11 | [av]bani | the damn snoms lockup on transfers still |
| 12:57.22 | robin_sz | not for me |
| 12:57.35 | [av]bani | and i guess g729 is buggy tooo.. fortunately i can use g711 |
| 12:57.36 | robin_sz | but htey are expensive for what you get |
| 12:58.07 | [av]bani | i just wish snom would stop being so difficult about bugs |
| 12:58.33 | [av]bani | its obvious they have never used their phones in a production setting outside their own offices |
| 12:58.35 | robin_sz | what are Aastra like? |
| 12:58.55 | [av]bani | aastra had terrible firmware for a long time, i guess they are just now to the point where its usable |
| 12:59.07 | robin_sz | oh |
| 12:59.07 | [av]bani | whereas snom still have major issues |
| 12:59.25 | robin_sz | snom are too expensive for my own business |
| 12:59.27 | [av]bani | lockups, wrong indications, terrible ui |
| 12:59.30 | robin_sz | fine for clients though :) |
| 12:59.52 | robin_sz | the Aastra looks OK |
| 12:59.54 | mexuar-tim | [av]bani:, I've got 4 snoms - they are fine - which model do you have/ firmware version |
| 13:00.03 | [av]bani | and they keep introducing new "features" in firmwares, which break provisioning and introduce new bugs |
| 13:00.07 | [av]bani | mexuar-tim: snom 360 |
| 13:00.26 | [av]bani | mexuar-tim: in 5.3 they introduced a "new feature" which completely broke provisioning |
| 13:00.28 | robin_sz | also |
| 13:00.44 | robin_sz | provisioning is most fscking odd on snoms |
| 13:01.00 | [av]bani | eh, its ok, i just wish snom would stop breaking shit |
| 13:01.12 | [av]bani | or if they introduce a new "feature", make it fecking default to OFF |
| 13:01.17 | robin_sz | I copied the settings dump into a file ... renamed it to macaddy.txt |
| 13:01.29 | mexuar-tim | Ah, I've got the 'old' 190's - can't recall the firmware version. I've only got 4 so provisioning is not an issue. |
| 13:01.33 | [av]bani | ahahaa.. youre not supposed to do that |
| 13:01.49 | robin_sz | no? |
| 13:01.53 | [av]bani | not really |
| 13:02.02 | robin_sz | why not? |
| 13:02.08 | [av]bani | the settings dump is informational only, not supposed to be used for provisioning |
| 13:02.13 | robin_sz | how else do you get a dump of current settings? |
| 13:02.29 | [av]bani | its informational, not used 1:1 as a provisioning file |
| 13:02.48 | robin_sz | then they are fsckwits |
| 13:03.00 | [av]bani | afaict its really just for debug |
| 13:03.06 | robin_sz | sigh |
| 13:03.13 | robin_sz | anyway ... |
| 13:03.29 | robin_sz | I copied it over, since we use DHCP I delted the lines related ot ip addy etc |
| 13:03.39 | robin_sz | rebooted and it killed the phone |
| 13:03.50 | [av]bani | yep |
| 13:04.32 | [av]bani | well, theres stuff in there like keys and things, which yo arent supposed to set via provisioning afaict |
| 13:04.32 | robin_sz | i have to say if the settings dump is in a format other than what the provisioning file needs, they are compleat idiots |
| 13:04.39 | [av]bani | so no wonder you bricked your phone |
| 13:05.03 | robin_sz | no, any half-assed softaware would just ignore stuff it didnt need |
| 13:05.06 | [av]bani | though thats the other thing, they dont have an end user recovery system |
| 13:05.13 | [av]bani | even grandstream has that |
| 13:06.02 | robin_sz | "end user recovery"? |
| 13:06.13 | [av]bani | yes, eg recover a busted phone without having to RMA it |
| 13:06.15 | robin_sz | if my end users get lost, I rejoice, not recover |
| 13:06.29 | robin_sz | ahh that |
| 13:06.48 | robin_sz | you mean some form of factory reset? |
| 13:06.55 | [av]bani | no, eg |
| 13:07.00 | [av]bani | power outage during flash upgrade |
| 13:07.04 | [av]bani | gxp = recoverable |
| 13:07.06 | [av]bani | snom = rma |
| 13:07.37 | robin_sz | im not usre the gxp is recoverbale on the new firmware |
| 13:07.40 | [av]bani | it is |
| 13:07.43 | [av]bani | i've _done_ it |
| 13:07.48 | robin_sz | reverts to what? |
| 13:07.52 | *** join/#asterisk pengyong (n=lala@218.93.154.119) |
| 13:07.54 | [av]bani | what? |
| 13:08.02 | [av]bani | it reverts to whatevr you recover it to |
| 13:08.04 | robin_sz | recovers to what? |
| 13:08.10 | [av]bani | anything you give it |
| 13:08.13 | robin_sz | v1 firmware? |
| 13:08.15 | robin_sz | yeah? |
| 13:08.38 | robin_sz | so I could deliberatly pull the power during an upgrade as a means of getting back to the "old" firmware? |
| 13:08.42 | AndyCap | robin_sz: it's not hard to make a bootloader that can receive new firmware somehow if the old won't vboot. |
| 13:08.52 | [av]bani | heh, dunno if you an downgrade |
| 13:09.03 | [av]bani | but you can recover from complete busted upgrade |
| 13:09.16 | robin_sz | the new firmware is totally broken on my phone, dead. |
| 13:09.20 | [av]bani | because i had to :| |
| 13:09.27 | robin_sz | I now have two phones at home |
| 13:09.34 | robin_sz | GXP2000, no display |
| 13:09.38 | [av]bani | web? |
| 13:09.52 | robin_sz | Zyxel wifi phoen ... desinged to emulate a turd |
| 13:10.14 | [av]bani | hahaha.. the wip300? |
| 13:10.22 | robin_sz | prestige 2000 |
| 13:10.26 | [av]bani | :o |
| 13:10.27 | robin_sz | bag of crap |
| 13:10.31 | [av]bani | o: |
| 13:10.38 | robin_sz | battery life measured in milliseconds |
| 13:10.50 | robin_sz | cradel that charges it one time in 10 |
| 13:11.04 | robin_sz | protocols are busted every which way |
| 13:11.16 | [av]bani | welcome to zyxel. enjoy your visit |
| 13:11.25 | robin_sz | its even too thick to make a decent door wedge |
| 13:11.41 | [av]bani | producing shoddy product since 1985 |
| 13:11.50 | robin_sz | yeah |
| 13:11.58 | [av]bani | and you havent learned, yet? |
| 13:11.59 | [av]bani | :)) |
| 13:12.06 | robin_sz | I didnt buy it |
| 13:12.20 | robin_sz | client bought a bunch "here install these" |
| 13:12.33 | [av]bani | snom is disappointing because its a german company, and snom are supposed to be uber |
| 13:12.39 | robin_sz | "I tried installing them, but they wont flush ...." |
| 13:13.08 | robin_sz | german stuff USED to be uber, now? nah |
| 13:13.18 | [av]bani | :| |
| 13:13.26 | robin_sz | .de quality has dived in the last 10 years |
| 13:13.30 | [av]bani | well its more, nice hardware damn shame about the software |
| 13:13.36 | robin_sz | exposure to market forces, re-unification etc |
| 13:13.39 | [av]bani | they should either fix the fecking bugs or opensource the code |
| 13:14.07 | [TK]D-Fender | [av]bani : Betrayed by EVERY IP phone manufacturer now are we? You should change your nick to something like "jilted" ;) |
| 13:14.19 | [av]bani | [TK]D-Fender: i dont hate polycom half as much as i used to |
| 13:14.27 | robin_sz | in .de, there is atill a tendendcy to use .de products. use anything else and people look funny at you .. but in private, they admit that some foreign products are better now |
| 13:14.36 | [TK]D-Fender | A glowing review! You heard it here first! |
| 13:14.46 | [av]bani | [TK]D-Fender: though i still never forgive them for not implementing this newfangled technology called "backlight" |
| 13:14.59 | [av]bani | [TK]D-Fender: i have a cisco 7970 on order :| |
| 13:15.15 | robin_sz | all you bank account belong to us |
| 13:15.22 | [TK]D-Fender | [av]bani : Yeah, and tell that to the other 90% that don't including Cisco who is even more expensive and restrictive. |
| 13:15.23 | [av]bani | i got it on special |
| 13:15.38 | [TK]D-Fender | 7970 is SCCP only right now, no? |
| 13:15.42 | [av]bani | yep |
| 13:15.47 | robin_sz | ick |
| 13:15.57 | [av]bani | this is for home, i dun care what it speak as long as * can talk to it |
| 13:16.03 | [av]bani | it could speak klingon for all i care |
| 13:16.10 | robin_sz | never got * and sccp to work |
| 13:16.27 | robin_sz | seemed either: |
| 13:16.29 | robin_sz | busted |
| 13:16.31 | robin_sz | or |
| 13:16.36 | robin_sz | undebuggable |
| 13:16.40 | [av]bani | sounds like |
| 13:16.41 | [av]bani | snom |
| 13:16.42 | [av]bani | or |
| 13:16.42 | [TK]D-Fender | I wouldn't touch it unless I KNEW is was going to have SIP and be able to profit from the extras under it. SCCP is somewhat implemented in *, but I don't want to be beholden to * either.... |
| 13:16.44 | [av]bani | grandstream |
| 13:16.59 | [av]bani | [TK]D-Fender: CCM! yay |
| 13:17.03 | *** join/#asterisk insomni (n=insomni@x1-6-00-e0-18-6f-34-ad.k455.webspeed.dk) |
| 13:17.07 | [av]bani | problem solved |
| 13:17.18 | acidchild | hello all, i just ordered a vonage pack, is that easy and stuff to set up on asterisk? |
| 13:17.18 | [av]bani | :)) |
| 13:17.34 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 13:17.36 | robin_sz | we had a dect gateway on sccp .. * declined to speak to it |
| 13:17.49 | [av]bani | robin_sz: did you use chan_sccp2 ? |
| 13:18.05 | robin_sz | no |
| 13:18.10 | [TK]D-Fender | [av]bani : Or problems created... |
| 13:18.13 | robin_sz | dont think so |
| 13:18.26 | [av]bani | robin_sz: http://chan-sccp.berlios.de/ |
| 13:18.29 | robin_sz | we used a "fed ex box" to solve that problem |
| 13:18.53 | robin_sz | put gateway in "fed ex box" ... problem solved |
| 13:19.06 | *** join/#asterisk ibob63 (n=hp@bb-87-82-15-9.ukonline.co.uk) |
| 13:19.41 | ibob63 | I've installed the asterisk-gtk-console on ubuntu - can anyone tell me how to launch it? I just can't figure it out. |
| 13:20.05 | robin_sz | [av]bani: no mention of sccp2 on that page |
| 13:20.09 | [av]bani | [TK]D-Fender: course, all you have is polycoms so you have no frame of reference |
| 13:20.37 | [av]bani | robin_sz: its known as chan-sccp2, so it doesnt confuse with other chan-sccp projects... of which there are three |
| 13:21.53 | robin_sz | well, then I might have used it |
| 13:22.01 | robin_sz | or I might have used one of the others |
| 13:22.07 | [TK]D-Fender | [av]bani : No, I have Uniden's, Sipura's, and have seen others. |
| 13:22.10 | robin_sz | I used chan_sccp |
| 13:22.20 | robin_sz | whther that was a 2 or not, I dont know |
| 13:22.30 | *** join/#asterisk nagl (n=nagl@vie-086-059-104-148.dsl.sil.at) |
| 13:22.31 | [TK]D-Fender | Have tried a BT-101.... *ick* |
| 13:22.38 | [av]bani | [TK]D-Fender: spa-841 ... |
| 13:22.52 | [av]bani | for $160 no less... |
| 13:22.52 | *** join/#asterisk Riedel (i=riedel@darktower.versahell.net) |
| 13:23.04 | robin_sz | the logic of calling a project chan_sscp2 and naming the files chan_sccp in order "to avoid confusion" is ... perhaps, unique. |
| 13:23.16 | [av]bani | robin_sz: :) |
| 13:23.46 | [av]bani | robin_sz: this is asterisk, where peer+user = friend |
| 13:23.53 | [av]bani | it doesnt have to make sense |
| 13:24.12 | Riedel | I have a strange issue with Asterisk 1.2.4. Sometimes, like 1 out of 10 times, Asterisk answers but passes no audio to the called party. Note that the connection is not NATed and it appears it happens randomly. Anyone got an idea what the issue might be ? |
| 13:24.27 | Riedel | calling party, even. Not called party. |
| 13:24.46 | robin_sz | ok, so what command runs asterisk-gtk-console once its installed then? |
| 13:25.21 | my007ms | hi all |
| 13:25.25 | x86 | robin_sz: xterm + asterisk -r? |
| 13:25.35 | my007ms | i wish i have answer to my Q this time |
| 13:25.40 | my007ms | i have TDM card |
| 13:26.03 | my007ms | when i do lspci from any distore i can not but i can from livecd |
| 13:26.34 | my007ms | what can i do any idea?? |
| 13:34.17 | ibob63 | <PROTECTED> |
| 13:34.31 | [av]bani | [TK]D-Fender: snom's jitter buffers are first rate though, better than polycom's |
| 13:34.41 | [av]bani | [TK]D-Fender: probably the only positive i've found lately? :| |
| 13:38.55 | *** join/#asterisk Kernel_core (i=Kernel_C@217.218.80.227) |
| 13:38.57 | Kernel_core | hi all |
| 13:39.06 | Kernel_core | anybody here familiar with Astbill ?! |
| 13:40.58 | Riedel | Hm. This patch seems to have worked for my issue: |
| 13:40.59 | Riedel | http://bugs.digium.com/file_download.php?file_id=9420&type=bug |
| 13:41.16 | bartpbx | hello |
| 13:41.25 | Riedel | hello |
| 13:41.34 | bartpbx | i need some help with agi and variables |
| 13:41.51 | bartpbx | anyone working with agi / fast Agi here? |
| 13:41.55 | Riedel | I'm kindof an asterisk noob,but shoot |
| 13:41.57 | Riedel | I use AGI's |
| 13:42.05 | bartpbx | ok |
| 13:42.39 | bartpbx | i have the problem that if a call is canced a get Variable DIALEDTIME ist 0 |
| 13:43.01 | bartpbx | if a call is answerd all varibales are filled. But not in case of cancel or noanswer |
| 13:43.08 | [TK]D-Fender | [av]bani : Well I haven't ever heard of anyone experiencing any kind of audio/network quality issue with them anywhere.... |
| 13:43.25 | bartpbx | but as far as i udnderstand the vaibale DIALEDTIME it shuld get set in case of NOANSWER |
| 13:43.32 | [av]bani | [TK]D-Fender: i get some dropouts/clicks with polycom on loaded lines... snom is perfect |
| 13:43.49 | [av]bani | [TK]D-Fender: and the weird polycom sound truncating when you connect... |
| 13:44.21 | Riedel | bartpbx: I havn't had the need to use that variable yet so I can't really be of help there |
| 13:44.45 | bartpbx | :-( |
| 13:47.20 | tzafrir | ibob63, what do you mean "did not work for me"? |
| 13:47.37 | tzafrir | in general, skip that gtkconsole. Not worth the trouble |
| 13:48.02 | tzafrir | And you shouldn't be running a local X server with Asterisk, so I hope the X display is remote |
| 13:48.18 | [TK]D-Fender | [av]bani : That truncation appears to be unique to you so far... seen it documented anywhere else on the mailing lists? |
| 13:51.02 | my007ms | pleas all |
| 13:51.31 | my007ms | any one all idea how to knwo if my pci is express pci or 2.2 noraml pci |
| 13:51.33 | my007ms | ? |
| 13:51.56 | my007ms | software tools coz the manual don't see much |
| 13:52.04 | *** join/#asterisk led-zep (n=led-zep@lns-bzn-49f-81-56-191-95.adsl.proxad.net) |
| 14:02.18 | Gand_DJ | Might want to read your motherboard manual |
| 14:02.28 | Gand_DJ | If new motherboard... you might have both |
| 14:04.04 | my007ms | no it's don't have lot info |
| 14:04.29 | my007ms | and when i do lspci from livcd i see the card |
| 14:04.38 | *** join/#asterisk virterm (n=virterm@204.225.113.73) |
| 14:05.16 | stoffell | my007ms, check the manufacturers' website |
| 14:05.51 | my007ms | it's gigabyte |
| 14:05.57 | my007ms | have the same manual |
| 14:06.02 | my007ms | that i ahve |
| 14:06.11 | my007ms | wait i will send u the URL |
| 14:06.14 | stoffell | my007ms and what do the tech specs say? |
| 14:06.52 | my007ms | just PCI i don't see if it was pc 2.2 or not |
| 14:07.18 | stoffell | my007ms what distro do u want to use? |
| 14:08.20 | my007ms | Centos |
| 14:08.24 | my007ms | any othere ok |
| 14:08.34 | my007ms | i try fc3 but not wotk |
| 14:08.51 | stoffell | my007ms and lspci doesn't show card? |
| 14:09.12 | my007ms | it show unknow divice |
| 14:15.57 | my007ms | http://www.gigabyte.com.tw/Support/Motherboard/Manual_Model.aspx?ProductID=1906 |
| 14:16.08 | my007ms | this is my mothereboard |
| 14:25.49 | my007ms | http://www.gigabyte.com.tw/Products/motherboard/Products_Spec.aspx?ClassValue=motherboard&ProductID=1909&ProductName=GA-8I945G |
| 14:39.29 | my007ms | hello all |
| 14:47.15 | *** join/#asterisk asterisk99 (n=astguy@modemcable169.194-130-66.mc.videotron.ca) |
| 14:50.26 | *** join/#asterisk CoolAcid (n=jason@216.99.98.39) |
| 14:51.34 | *** join/#asterisk clive- (n=pirch@dsl-145-33-235.telkomadsl.co.za) |
| 14:53.56 | asterisk99 | I'm having trouble with udev on ubuntu... I want to run Asterisk as non-root... It won't start since /dev/zap/* are all root:root:660 even tho I put in the rules /etc/udev/udev.rules and permissions.rules --- anyone figure this out? |
| 14:57.18 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 14:57.18 | *** mode/#asterisk [+o anthm] by ChanServ |
| 14:58.18 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 14:58.52 | ManxPower | asterisk99, the Wiki page was not helpful? |
| 15:06.51 | *** join/#asterisk asterisk99 (n=astguy@modemcable169.194-130-66.mc.videotron.ca) |
| 15:07.42 | ManxPower | ~docs |
| 15:07.47 | jbot | methinks docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 15:09.52 | *** join/#asterisk Kernel_core (i=Kernel_C@217.218.80.227) |
| 15:09.55 | Kernel_core | hi all |
| 15:10.01 | Kernel_core | anybody familiar with ASTBILL ?! |
| 15:13.09 | ManxPower | Not I, I don't bill for calls. |
| 15:15.23 | asterisk99 | Need help: Trying to get asterisk to run non-root... all is ok EXCEPT /dev/zap files are all root:root despite my adding rules to udev control files |
| 15:17.59 | ManxPower | asterisk99, the Wiki page was not helpful? |
| 15:18.32 | asterisk99 | ManxPower: It was... I added all the rules it said to add |
| 15:19.03 | asterisk99 | ManxPower: But, alas, despite my modifying udev's files, it's a no-go |
| 15:20.07 | Kernel_core | ManxPower: do you have any solution to record calls ( I mean in GSM Format ) in SQL ? |
| 15:20.13 | asterisk99 | ManxPower: The only thing I haven't tried is running /usr/bin/safe_asterisk (cuz it says somewhere it only runs under a shell) |
| 15:20.23 | ManxPower | Kernel_core, I also don't record calls |
| 15:20.48 | ManxPower | I don't run asterisk as non-root. I just know there's a Wiki page about it. |
| 15:22.24 | asterisk99 | ManxPower: OK. I'm close; really close!!! If I go and chmod u=+r the /dev/zap files , asterisk immediately starts |
| 15:23.02 | asterisk99 | ManxPower: I could try modifying the startup script, if I knew where to find it :) |
| 15:27.16 | Gand_DJ | for allowing the g.729 codec in *, just use allow=g729 ? |
| 15:27.32 | Gand_DJ | same with allow=iLBC ? |
| 15:27.52 | Kernel_core | Gand_DJ: yes |
| 15:27.59 | Gand_DJ | k :) |
| 15:37.13 | *** join/#asterisk MarkAngels (n=Publiken@h114n1fls32o925.telia.com) |
| 15:49.03 | *** join/#asterisk riddlebox (n=james@24-171-11-166.dhcp.stls.mo.charter.com) |
| 15:50.08 | *** join/#asterisk danalien (n=danalien@unaffiliated/danalien) |
| 15:50.24 | lunaphyte | how can i make VoiceMailMain use the calling party's extension? |
| 15:58.52 | riddlebox | what would cause a source build not to start when I type asterisk? |
| 15:59.39 | tzafrir | "make" is a better excuse for it to start |
| 16:00.44 | tzafrir | lunaphyte, use the callerid. e.g: VoiceMailMain(${CALLERIDNUM}) |
| 16:00.59 | ibob63 | how does asterisk know where the music files are stored? is there a config somewhere? |
| 16:01.00 | Corydon76-home | lunaphyte: by using the database to associate the calling party's channel with the extension |
| 16:01.14 | tzafrir | ibob63, in /etc/asterisk |
| 16:01.19 | Corydon76-home | /var/spool/asterisk/mohmp3 |
| 16:01.30 | tzafrir | Although IIRC sounds/ is a bit hard-wired |
| 16:02.12 | Corydon76-home | Actually, the music files are configurable in the musiconhold.conf file |
| 16:03.47 | riddlebox | I have compiled from source asterisk with make, make install and checked /var/run/asterisk.ctl does exist, but I cannot get asterisk to start? |
| 16:04.01 | Gand_DJ | hrm.. curious. when I dial out through *, the other line will ring right aways like normal. If I call into my * box (using another voip acct) the call says established, but * doesn't pass it to my ext to "ring" for like 3-5 sec |
| 16:07.43 | ibob63 | Corydon76: thanks that was what I was looking for. |
| 16:09.07 | ibob63 | what does the /var/lib/asterisk/sounds folder store? |
| 16:09.57 | ibob63 | it seem my asterisk is conf to use /mohmp3 as the music on hold directory |
| 16:12.14 | riddlebox | nm I just apt-get'd it and it works |
| 16:12.29 | *** join/#asterisk Dr-Linux (n=Dreamer@host202-147-168-130.lhr.dancom.net.pk) |
| 16:12.48 | *** join/#asterisk rogier (n=rogier@16-65-dsl.ipact.nl) |
| 16:16.39 | *** join/#asterisk A-Tuin (n=a-tuin@steves.ip.v4.me.uk) |
| 16:19.04 | *** join/#asterisk _DAW (n=bob@adsl-6-66-81.msy.bellsouth.net) |
| 16:21.40 | *** join/#asterisk tfrevor (n=lpzovyz@c-24-1-238-49.hsd1.tx.comcast.net) |
| 16:22.13 | tfrevor | Hullo, all. Hoping I might be able to get a little help with an asterisk-related question. |
| 16:23.51 | *** join/#asterisk puzzled (n=yeahrigh@puzzled.xs4all.nl) |
| 16:24.40 | *** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin) |
| 16:24.46 | puzzled | hi |
| 16:25.18 | tfrevor | Howdy, puzzled. For so many people registered, I think we're the only two talkin'... |
| 16:25.19 | *** join/#asterisk Leob (n=chatzill@pool-71-126-240-250.bstnma.fios.verizon.net) |
| 16:26.42 | puzzled | tfrevor: seems so although that's not unusual |
| 16:27.09 | tfrevor | Maybe, but it sure makes getting assistance rather difficult... :) |
| 16:27.11 | Leob | QUESTION about configuring Asterisk as a server: Hello there, how can I configure Asterisk to run as server with the -C option (so that I can read the config files from a specific directory)? |
| 16:27.45 | puzzled | tfrevor: guess so but you can always use the mailinglist too |
| 16:27.52 | Leob | Sorry, I meant service: how can I configure Asterisk to run as service with the -C option (so that I can read the config files from a specific directory)? |
| 16:27.54 | tfrevor | Leob: Gratned, I'm a newbie to asterisk myself, but I would think that you could just change the paths in the asterisk.conf file, couldn't you? |
| 16:28.27 | puzzled | Leob: why don't you just put the config files in /etc/asterisk? |
| 16:30.08 | tfrevor | puzzled: I can see why I would put the other config files elsewhere (shared between multiple servers, backup directory, etc). |
| 16:30.10 | Corydon76-home | Or alter /etc/init.d/asterisk to specify the -C option |
| 16:30.40 | Leob | hmm, so I'd have to change init.d by hand ... |
| 16:30.56 | Corydon76-home | tfrevor: yes, but you could put a singleton /etc/asterisk/asterisk.conf file to change the paths |
| 16:31.08 | puzzled | tfrevor: well people asking such questions don't usually have reached the level of multiple servers and backup scenarios :) |
| 16:31.49 | tfrevor | puzzled: Better safe than sorry... :D |
| 16:31.54 | Leob | I'm trying to leave all my config files in a separate disk ... is that a bad idea? |
| 16:32.29 | Leob | this way, if I have to reinstall everything I don't have to worry about configs |
| 16:32.38 | tfrevor | Leob: It's not a bad idea (IMO), but not a great one either. I like to leave my config files in /etc ismply to have everything together and easy to find. |
| 16:33.33 | Leob | that's what I'm doing, too ... but moving everything from /etc to my /data/etc folder |
| 16:34.13 | tfrevor | Leob: Why not just link the /etc directory then? Move your stuff to /data/etc, but still have the link to /etc ? |
| 16:34.41 | Leob | would asterisk accept the link? |
| 16:34.52 | tfrevor | I don't see why not. |
| 16:35.01 | Leob | let me try |
| 16:37.03 | tfrevor | Guess I'll just throw my question out, then... I've recently converted the D-Link DVG-1120M to an 1120S for testing purposes with Asterisk (and, yes... I know about the quality). Works great on the two FXS. however, it also has a "line-in" and I read where it is supposed to be an FXO. Telnetting into the box does say it has FXO options. However, is that just a passthru to the included FXS or can it be used as a "true" FXO, passin |
| 16:37.50 | puzzled | no idea. didn't the manual say something about it? |
| 16:38.17 | tfrevor | No manual. It was a branded 1120M (AT&T CallVantage), which I head to convert to use as a SIP phone. |
| 16:38.31 | tfrevor | (Well, it had a manual, but only so far as using it with CallVantage) |
| 16:39.12 | Leob | alright, the link works just fine! thanks for the help! |
| 16:39.21 | tfrevor | Glad to help Leob! |
| 16:40.11 | *** join/#asterisk af_ (n=af@ip-165-17.sn2.eutelia.it) |
| 16:41.08 | tfrevor | For what it's worth, I can't even find the definition for the different fxo signal settings... It shows "fxo signal <Normal | PR | CPC | NTT>", but I'm not finding anything on those either... |
| 16:44.01 | *** join/#asterisk hertell (n=rene@jumbo52.adsl.netsonic.fi) |
| 16:45.14 | tzafrir | Leob, a good init.d script should require no hand configuration and instead source /etc/sysconfig/<servicename> for config |
| 16:45.26 | hertell | Good evening everyone! |
| 16:45.41 | tzafrir | (On Debian: default instead of sysconfig) |
| 16:45.56 | tfrevor | Hullo, Hertell. |
| 16:46.15 | hertell | can someone point me to a somekind of troubleshoot-checklist? |
| 16:46.24 | hertell | hi tfrevor |
| 16:46.47 | Leob | tzafrir, thanks for the tip, but I don't think Asterisk's init.d script allows for a -C option... |
| 16:46.49 | hertell | i'm really loosing my hair.. ;-) |
| 16:46.59 | tfrevor | Sorry, Hertell... Don't know of anything like that. |
| 16:47.41 | hertell | ok.. i'm just having trouble in getting voice transmitted between myself and anyone who is calling.. |
| 16:47.53 | tfrevor | Hertell: Behind a firewall? |
| 16:48.10 | riddlebox | is there a way to asterisk to look to a different dir for all sounds? |
| 16:48.48 | riddlebox | or better yet, is the default dir, /etc/asterisk/sounds? |
| 16:49.36 | tzafrir | Leob, if they don't (in that way) then I consider it a bug. But this may be a matter of personal taste. The Debian package's init.d script does. |
| 16:50.33 | *** join/#asterisk shakuhashi (n=teste@200.103.120.88) |
| 16:50.48 | *** join/#asterisk tfrevor (n=vrcaduc@c-24-1-238-49.hsd1.tx.comcast.net) |
| 16:50.58 | tfrevor | sorry 'bout that... internet hiccup. |
| 16:52.50 | *** part/#asterisk shakuhashi (n=teste@200.103.120.88) |
| 16:53.37 | *** join/#asterisk dlublink (n=dlublink@modemcable114.38-201-24.mc.videotron.ca) |
| 16:54.02 | asterisk99 | Need help: Trying to get asterisk to run non-root... all is ok EXCEPT /dev/zap files are all root:root despite my adding rules to udev control files |
| 16:55.37 | lunaphyte | ~docs |
| 16:55.38 | jbot | it has been said that docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 16:57.17 | hertell | tfrevor: yes, I'm behind a firewall |
| 16:58.19 | tfrevor | Okay... I remember reading something about that. |
| 16:58.57 | tfrevor | Your phone on one side and the asterisk box on another. What are you using for the dialing? |
| 16:59.40 | hertell | my phone and asterisk are in the same net, but calling out is like gambling. either you're lucky, or then you'r not.. :-( |
| 16:59.46 | hertell | i found this: http://www.voip-info.org/tiki-index.php?page=Asterisk+firewall+rules |
| 16:59.54 | *** join/#asterisk Eitch (n=hugo@unaffiliated/eitch) |
| 17:00.08 | hertell | i'll check that right away, so that I can test that example |
| 17:00.51 | tfrevor | Well, you said that it's just your media stream that's not working? The two systems do connect up? |
| 17:01.28 | tfrevor | Reason I ask is that some FXS devices don't like to work through firewalls. However, if they connect but no media stream, it could be that you don't have the 10000-20000 ports open, which it uses for the actual media stream. |
| 17:01.38 | asterisk99 | Need help: Trying to get asterisk to run non-root... the only thing stopping me is UDEV and permissions (keep resetting back to root) |
| 17:02.34 | hertell | the 10000 and 20000 should be open.. |
| 17:03.06 | *** join/#asterisk dpryo (i=hn@donatello.nesland.net) |
| 17:03.32 | tfrevor | Is your local net being NAT-ted? |
| 17:03.47 | *** join/#asterisk mrtwister|mobile (n=andrius@cable-10-68.cgates.lt) |
| 17:04.31 | *** join/#asterisk peanuter (n=peanuter@216.176.177.138) |
| 17:04.43 | *** join/#asterisk razu_ (n=razu@adsl26141.estpak.ee) |
| 17:05.31 | *** join/#asterisk daguerro (n=test@lss-67-33.ee.itb.ac.id) |
| 17:06.26 | tfrevor | hertell: Don't know if you've seen it, but http://www.voip-forum.com/?p=131&more=1 might answer some of the questions. |
| 17:07.45 | tfrevor | Hope it helps. I need to head out. |
| 17:15.27 | *** join/#asterisk frenzy (n=frenzy@196.45.144.41) |
| 17:15.48 | frenzy | hello all |
| 17:16.04 | frenzy | is there an open source IAX webphone available? |
| 17:17.41 | dlublink | best option is to use sip instead |
| 17:17.44 | dlublink | if possible |
| 17:17.58 | frenzy | is there one available? |
| 17:18.38 | dlublink | Haven't seen one, IAX is interAsterisk Exchange |
| 17:18.47 | dlublink | it is meant for asterisk servers to communicate, I doubt you will find one |
| 17:18.51 | frenzy | what about for SIP |
| 17:19.06 | dlublink | I have spent ages trying to find softphones, and the only ones I ever found were SIP |
| 17:19.28 | dlublink | xlite is free |
| 17:19.30 | dlublink | and works well |
| 17:19.44 | dlublink | just if you use XLite, right click on the grey part of the phone and choose diagnostic log |
| 17:19.49 | dlublink | it will make your life easier |
| 17:20.20 | dpryo | I want a xlite-workalike with GUI... The spaceship-thingie-looking xlite is horrible ;P |
| 17:20.49 | dpryo | Scares the hell out of my users |
| 17:21.13 | hertell | try linphone |
| 17:21.57 | Gand_DJ | Firefly does iax |
| 17:22.01 | dlublink | It is horrible, but if you use the diagnostic log, it makes it 10 times easier |
| 17:22.02 | Gand_DJ | also ide-something |
| 17:22.04 | SkramX | Hi All. |
| 17:22.07 | Gand_DJ | idefisk? |
| 17:22.21 | dpryo | I'm testing idefisk now... It looks clean. |
| 17:23.40 | dlublink | Hey, can anyone assist me with some basic extensions.conf file questions? |
| 17:26.50 | *** join/#asterisk xmark (n=brg@c-68-43-129-244.hsd1.mi.comcast.net) |
| 17:29.09 | xmark | Good morning -- I was hoping I might ask a few asterisk config questions -- |
| 17:30.20 | *** topic/#asterisk by russellb -> Asterisk: The Open Source PBX -=- http://www.asterisk.org |
| 17:31.24 | xmark | Does anyone know how to make an extension dial out on a particular device. Sorry this seems like a simple quetion, but I'm having a hard time trackin down the answer |
| 17:32.42 | frenzy | does SER have a room on freenode? |
| 17:33.19 | xmark | <PROTECTED> |
| 17:34.36 | frenzy | now that would kill my PC |
| 17:34.38 | frenzy | :) |
| 17:35.30 | lunaphyte | how can i change the folders available through voicemail? |
| 17:37.45 | asterisk99 | Can anyone tell me if /etc/udev/rules.d/udev.rules is the same as /etc/udev/rules.d/50-udev.rules ??? (Me use ubuntu) |
| 17:38.54 | asterisk99 | xmark: U mean dial out on a particular Zap line or PRI chanel? |
| 17:40.27 | xmark | Yes on a zap port. I have a TDM22B card 2 FXO Ports I'd like to have a particular extension always dial out on the "first port" |
| 17:41.14 | tzafrir | frenzy, there can't be a real webphone , I suppose. There is mozphone which is a mozilla extension. There is also one ActiveX phone |
| 17:41.51 | asterisk99 | xmark: Dial(Zap/1... or Dial(Zap/2... |
| 17:41.51 | *** join/#asterisk poemius (n=poemius@adsl-70-48-192-81.adsl.iam.net.ma) |
| 17:41.52 | tzafrir | xmark, use g0 |
| 17:42.19 | tzafrir | Dial(Zap/gNUM/number) , where NUM is the group number |
| 17:42.19 | poemius | hello :) |
| 17:42.51 | xmark | Doesn't Dial mean for dial the extension? |
| 17:43.00 | asterisk99 | xmark: I think u want to specify the exact port --- I don't think you want a group number |
| 17:43.29 | xmark | a99: I think your right |
| 17:43.31 | asterisk99 | xmark: Yes, you are going to "Dial" the outside line |
| 17:43.34 | poemius | in a sense it dials anything... an extension or even a remote extension |
| 17:43.46 | xmark | Here is how I have my extension defined |
| 17:43.49 | xmark | exten => 2000,1,Dial,Zap/2|20 |
| 17:43.49 | xmark | exten => 2000,2,Voicemail,u2000 |
| 17:43.49 | xmark | exten => 2000,3,Hangup |
| 17:44.39 | *** join/#asterisk iaxy (n=iaxy@modemcable236.55-131-66.mc.videotron.ca) |
| 17:44.48 | *** join/#asterisk peppertrader (n=peppertr@216.134.5.86) |
| 17:45.21 | asterisk99 | xmark: (Assuming that's from your SIP context (section)) that takes care of answering extensions ... are you not asking about dialing PSTNs? |
| 17:45.27 | poemius | wow, lots of people in asterisk-unregistered :)... I didn't think there would be that many people |
| 17:46.15 | dpryo | :P |
| 17:46.17 | xmark | a99: Yes I need them to dial out on the PSTN. They are actually analog devices attached to an FXS port but I think the concept is the same |
| 17:46.48 | *** join/#asterisk jpablo (n=jpablo@201.139.55.46) |
| 17:47.08 | poemius | :) bridging with a legacy PABX? |
| 17:47.22 | dpryo | What is the best way of programaticly make my sip-phone call somebody? Say I want a webpage where users can log in and make calls.. Should I use originate through asterisk manager? |
| 17:47.23 | tzafrir | asterisk99, what's wrong with a group dial here? |
| 17:47.36 | tzafrir | It will use the first availble phone of the group, right? |
| 17:47.54 | jpablo | dpryo: I do it creating files in /var/spool/asterisk/outgoing |
| 17:48.28 | asterisk99 | tzafrir: I thought xmark wanted to call out on "a particular device"? |
| 17:48.43 | dpryo | jpablo: So then the users phone will ring, and when he picks up, the phone continues to ring the other party? |
| 17:48.46 | *** part/#asterisk rob0 (i=1002@cardinal.lizella.net) |
| 17:49.03 | xmark | a99, tz maybe my terminology is not correct -- |
| 17:49.08 | jpablo | dpryo: that's rigth |
| 17:49.11 | tzafrir | jpablo, moving files to there, I hope |
| 17:49.16 | xmark | by device I mean FXO port... |
| 17:49.17 | jpablo | tzafrir: yeah |
| 17:49.39 | jpablo | creating them in /tmp then renaming them to outgoing |
| 17:49.44 | dpryo | jpablo: Thanks.. Btw, you know if there is a way to also transfer calls? |
| 17:49.53 | asterisk99 | xmark: Then I assumed correctly (I think) |
| 17:49.54 | xmark | I need exten 2000 to always dial out on one particular port |
| 17:50.05 | jpablo | dpryo: humm, that probably will involve the manager api |
| 17:50.23 | jpablo | dpryo: but i have never done that |
| 17:50.28 | dpryo | Ok :) |
| 17:51.16 | jpablo | i created a simple webbased company directory, then allowed people to call each other with a click on the browser. |
| 17:51.41 | poemius | :) sounds pretty neat :) |
| 17:51.53 | dpryo | Yeah, I've also created such thing, via the asterisk manager api.. Just wondered if there was another or better way of doing it. |
| 17:52.02 | dpryo | It really impresses bosses :) |
| 17:52.07 | asterisk99 | xmark: you don;t quite understand the exten => 2000, setup in extensions.conf ... That's what happens WHEN you dial 2000 (in whatever context you happen to be ... example: you might define all your SIP phones to be in one particular context) |
| 17:52.27 | poemius | definitely looks neat :) |
| 17:52.34 | asterisk99 | xmark: what you want is a plan for dialing PSTNS ... that's different |
| 17:52.48 | xmark | a99 -- Ok I follow |
| 17:53.06 | xmark | Where do I create a dialplane for PSTN? |
| 17:53.11 | hertell | guys.. does anyone know any tool to debug *? |
| 17:53.22 | poemius | asterisk -rvvvv |
| 17:53.26 | asterisk99 | xmark: so what u want is something like exten => _XXXXXXXXXX,1,Answer() |
| 17:53.39 | jpablo | dpryo: i think the manager api is a clearner way than creating files |
| 17:53.54 | hertell | mainly what I want is to understand why eg. I can't hear a squat when calling for example to the free world dialup echo-test.. |
| 17:54.25 | poemius | hertell: may be firewall rules |
| 17:54.37 | hertell | thsy should be correct.. |
| 17:54.43 | asterisk99 | xmark: the word "context" was confusing to me when I first saw it ... think "section" ... or "subroutine" |
| 17:54.55 | poemius | hertell: using X10? |
| 17:55.17 | hertell | i'm forwarding port 5060, 4569, 5036 and 10000:20000 |
| 17:55.22 | hertell | noup |
| 17:55.32 | hertell | i'm using a spa3k |
| 17:55.44 | *** join/#asterisk stoffell (n=stoffell@d51A583D9.access.telenet.be) |
| 17:55.47 | hertell | both * and my spa is behind a nat-firewall |
| 17:55.58 | hertell | i can call * |
| 17:56.27 | xmark | a99 exten => _XXXXXXXXXX,1,Answer() -- doesn't this mean to just anwer the call? |
| 17:56.45 | hertell | check voicemail etc, but calling to fwd:s 613 echo test gives me just an quiet line.. |
| 17:57.31 | poemius | not sure which ones are necessary, but I know I opened 5060:5063 5036:5040 8000:8003, 4569, 4520 (tcp and udp) |
| 17:57.37 | asterisk99 | xmark: as for extension 2000, ${CHANNEL} will tell you WHICH phone is the caller ... you'll have to GotoIf() that [good luck! GotoIf() is the kinkiest syntax I've ever seen ... except for a horrible Canadian-invented computer language called APL] |
| 17:57.38 | poemius | I did the config a long time ago |
| 17:58.24 | hertell | poemius: are you sure that also tcp? |
| 17:58.39 | hertell | poemius: i have just udp.. |
| 17:58.41 | poemius | some are udp, others tcp |
| 17:59.03 | poemius | but one port you don't seem to have is 8000, this one is udp |
| 17:59.10 | poemius | just to be sure, I opened both :) |
| 17:59.15 | hertell | 8000? |
| 17:59.52 | tronix | xmark: what hardware card do you have connected to the PSTN? TDM400P? X100P? |
| 17:59.57 | hertell | hmm. darn.. according to http://www.voip-info.org/tiki-index.php?page=Asterisk+firewall+rules there is no 8000 |
| 18:00.50 | tronix | poemius: also if you run SIP, need to open up RTP ports |
| 18:00.55 | tronix | which will be an UDP port range |
| 18:01.00 | poemius | I think it has to do with nat traversal |
| 18:01.03 | poemius | reception |
| 18:01.23 | poemius | :) you don't lose anything, try |
| 18:01.24 | asterisk99 | NAT traversal is a b**ch |
| 18:01.41 | xmark | tronix - I have a TDM400P (2FXO and 2FXS) I can dial out on the the PSTN via FXO or I can dial out via iax peer but I wonder how to tell an extension how to connect the call (IAx or which FXO port) |
| 18:01.50 | poemius | oh it works fine:) |
| 18:02.16 | poemius | they made progress... I use regularely my ipaq to dial my asterisk pbx through wifi |
| 18:02.24 | tronix | xmark: ahh, I see. here is an example: |
| 18:02.56 | poemius | http://corp.deltathree.com/technology/nattraversalinsip.pdf they mention port 8000 |
| 18:03.11 | tronix | [outbound-analog] |
| 18:03.25 | tronix | exten => _2.,1,Dial(Zap/g1/${EXTEN:1},${TIMEOUT}) |
| 18:03.37 | tronix | that means if you use 2 then number |
| 18:03.40 | tronix | it will put through PSTN |
| 18:03.42 | tronix | and |
| 18:03.48 | tronix | [outbound-voip] |
| 18:04.09 | tronix | exten => _91NXXNXXXXXX,1,Dial(IAX2/user@provider/${EXTEN:1},${TIMEOUT}) |
| 18:04.27 | tronix | means if you use 9 then 1 then area code then number (for north america), it will send through VOIP |
| 18:04.35 | *** join/#asterisk nagl (n=nagl@vie-086-059-104-148.dsl.sil.at) |
| 18:04.43 | tronix | that's a basic example of how to configure a dial plan using patterns |
| 18:04.46 | tronix | (and prefixes) |
| 18:04.51 | lunaphyte | how can i change the folders available through voicemail? |
| 18:05.07 | xmark | tronix -- thanks -- I'll give it a whirl |
| 18:05.14 | tronix | also, EXTEN:1 means strip first digit |
| 18:05.21 | tronix | so for 2<number>, EXTEN:1 will strip '2' |
| 18:05.24 | tronix | before dialing |
| 18:05.41 | tronix | if you had a prefix of 213 <number> then use EXTEN:3 to strip first 3 digits. etc |
| 18:08.37 | *** join/#asterisk nurfe (n=rgff@h24-207-10-162.dlt.dccnet.com) |
| 18:08.54 | SkramX | whats wrong with SetCIDNum(${CIDNUM})? It doesnt work, but doing SetCIDNum(NPANXXXXXXX) does work *shrug* |
| 18:09.20 | tronix | could it be that your voip provider is picky about checking? |
| 18:09.36 | SkramX | but i use the exact same number in [global |
| 18:09.39 | tronix | also, not sure if dashes are allowed in CIDNUM. I know it works fine in that format. |
| 18:09.42 | tronix | (NPA...) |
| 18:09.43 | SkramX | ] as i did... |
| 18:10.23 | SkramX | its not the carrier (asterlink) its asterisk.. i have CIDNUM=XXXXXXXXXX in [globals] that means I can use ${CIDNUM} where-ever I want, right? |
| 18:10.34 | *** join/#asterisk psyk0 (n=doc@023.adsl123.bie05.lan.ch) |
| 18:10.43 | psyk0 | Hello |
| 18:10.50 | *** join/#asterisk RoyK (n=roy@host-81-191-147-248.bluecom.no) |
| 18:10.53 | psyk0 | I have a problem with register => user:pw@sipprovider.org/extension |
| 18:10.59 | psyk0 | <PROTECTED> |
| 18:11.05 | psyk0 | but when I comment [sipprovider-out] everything works well except I can't make outgoing call |
| 18:11.12 | psyk0 | anybody have a solution? |
| 18:11.32 | tronix | xmark: don't forget that, also, in the context where you defined your softphones... also have two lines: 'include => outbound-analog' and 'include => outbound-voip' (or whatever you call these contexts) so they will be able to use these contexts to place outbound calls. and reload *, of course. |
| 18:14.27 | psyk0 | no solution? |
| 18:20.21 | *** join/#asterisk Eitch (n=hugo@unaffiliated/eitch) |
| 18:22.01 | jpablo | psyk0: your [sipprobla is pointing to the right context? |
| 18:28.06 | *** join/#asterisk troyb1 (n=central@CPE00907f17e478-CM014300013422.cpe.net.cable.rogers.com) |
| 18:28.10 | *** join/#asterisk Ad-Hoc (n=Nimbus@ppp55-adsl-211.ath.forthnet.gr) |
| 18:30.09 | psyk0 | jpablo: i have found the problem |
| 18:30.35 | psyk0 | I must use the same extension as my SIP username account |
| 18:30.53 | psyk0 | thanks |
| 18:33.18 | *** join/#asterisk ast_freak|Laptop (n=jesse@12.104.247.2) |
| 18:39.11 | *** join/#asterisk mexuar-tim (n=mexuar-t@host-212-158-206-61.bulldogdsl.com) |
| 18:49.39 | *** join/#asterisk brockj49464 (n=brockj49@63.87.56.235) |
| 18:52.48 | iaxy | Is there a way to start asterisk and have it dump to a log file when it doesn't start? I can't start asterisk and it is not logginganything |
| 18:53.07 | iaxy | actually there is no /var/log/messages |
| 18:53.15 | iaxy | actually there is no /var/log/asterisk/messages |
| 18:54.41 | *** join/#asterisk pixolex (n=chatzill@87-196-156-160.net.novis.pt) |
| 18:54.58 | RoyK | iaxy: logger.conf defines what files to log to |
| 18:56.46 | iaxy | actually that should be asterisk.conf, no? |
| 18:58.44 | stoffell | iaxy: no |
| 18:59.08 | TallAndy | Is there a way when using 'Originate' to get the line status back into a variable in phpagi. Eg line: engaged, dead, no answer, dialed sucessfully? |
| 18:59.26 | *** join/#asterisk NewSole (n=dave@d226-105-29.home.cgocable.net) |
| 19:00.19 | iaxy | hmmm.... asterisk.conf => path to log file |
| 19:00.42 | stoffell | iaxy: and logger.conf defines 'what' to log |
| 19:01.12 | iaxy | so where do I find out what file it logs to? |
| 19:01.23 | stoffell | iaxy: and make sure your verbosity is high enough |
| 19:02.00 | iaxy | My * isn't starting.... so I am trying -gdc to find out why, but its not logging anywhere. |
| 19:02.17 | stoffell | iaxy; tail /var/log/asterisk/full ? |
| 19:02.37 | stoffell | iaxy, start it like: asterisk -vvvvvvvvvvvvvvvc |
| 19:02.55 | iaxy | damn...there it is |
| 19:03.51 | iaxy | unable to open zap channel.. chan_zap.so failed.... I didn't change zap*.conf files..... |
| 19:04.13 | stoffell | iaxy, edit modules.conf and disable zap for now.. |
| 19:04.36 | stoffell | put noload => chan_zap.so for the moment |
| 19:05.50 | tronix | iaxy: run 'ztcfg -vvv' first; it may reconfigure zaptel |
| 19:05.57 | tronix | (to match what you actually ahve) |
| 19:09.18 | iaxy | Notice: Configuration file is /etc/zaptel.conf |
| 19:09.18 | iaxy | line 0: Unable to open master device '/dev/zap/ctl' |
| 19:10.08 | iaxy | WTF , I didn't change it. |
| 19:10.26 | Qwell | iaxy: got the modules loaded? |
| 19:10.27 | Abydos313 | make clean, make && make install from /usr/src/zaptel :)) |
| 19:10.51 | stoffell | maybe let iaxy try without zaptel for the moment... ? :) |
| 19:11.22 | Qwell | iaxy's don't need zap anyhow :p |
| 19:11.33 | stoffell | yeah, it did sound silly, didn't it? :p |
| 19:13.23 | iaxy | it goes without chan_xap |
| 19:22.39 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 19:23.18 | *** join/#asterisk Dr-Linux (n=Dreamer@host202-147-168-130.lhr.dancom.net.pk) |
| 19:23.23 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
| 19:23.25 | iaxy | something is putting channel=>zap/1 in my zap conf, and thats breaking it |
| 19:23.36 | hertell | does anyone know of any online-nat-debugging tool so that I could test if i have screwed up something with my firewall? |
| 19:23.56 | hertell | mainly for testing the SIP-traffic |
| 19:24.48 | stoffell | iaxy, try putting noload => chan_zap.so in modules.conf |
| 19:25.23 | stoffell | hertell, tail -f your firewall logs? ;) |
| 19:25.36 | iaxy | stoffell, I did that and it started. I started looking for the trouble and found that line in zap conf |
| 19:26.11 | stoffell | iaxy, okay. zaptel gives an error probably because your /etc/zaptel.conf or /etc/asterisk/zapata.conf are wrong |
| 19:26.21 | stoffell | iaxy, do you have zaptel-hardware and the needed drivers? |
| 19:26.23 | hertell | my firewall has no logging-features.. |
| 19:26.39 | iaxy | hehehe.... channel=>zap/1 is wrong |
| 19:26.53 | iaxy | should read channel => 1 |
| 19:27.34 | stoffell | iaxy, indeed, your zapata.conf is wrong |
| 19:31.39 | *** join/#asterisk MikeJ[Laptop] (n=vircuser@d14-69-8-30.try.wideopenwest.com) |
| 19:33.39 | *** join/#asterisk Leorg (n=Britson@adsl-223-42-191.aep.bellsouth.net) |
| 19:37.43 | *** join/#asterisk modulus` (n=modulus@shell.blacksun.net) |
| 19:37.45 | modulus` | yo |
| 19:41.25 | *** join/#asterisk cosa (n=tmalkut@fw.orasoft.net.pl) |
| 19:41.44 | *** part/#asterisk cosa (n=tmalkut@fw.orasoft.net.pl) |
| 19:46.58 | *** join/#asterisk bitmap (n=mdl@c-24-118-12-11.hsd1.mn.comcast.net) |
| 19:47.15 | *** join/#asterisk Mw3_ (n=mw3@national.t-error.hu) |
| 19:51.24 | *** join/#asterisk Grizzy (i=Generic@adsl-68-127-29-121.dsl.pltn13.pacbell.net) |
| 19:52.00 | riddlebox | does the debian package of asterisk change where sounds are stored from the default source compile? |
| 19:52.17 | Qwell | probably |
| 19:52.19 | Grizzy | Is there a tutorial for x-lite -> firewall -> asterisk ? |
| 19:52.24 | modulus` | hi qwell |
| 19:52.27 | Qwell | Grizzy: at asteriskguru |
| 19:52.34 | troyb1 | greetings Qwell |
| 19:52.37 | Qwell | I think it mentions firewall |
| 19:52.48 | modulus` | qwell can you help me with something? |
| 19:52.51 | Qwell | I can try |
| 19:52.58 | Katty | yawn. |
| 19:53.14 | modulus` | qwell where can i find a good wall mount pc or pc case to run my ass tricks box on? |
| 19:53.18 | Qwell | Katty: just thought you'd like to know...I went to subway the other night :P |
| 19:53.26 | Grizzy | does asterisk implement STUN, or do I need a separate package? |
| 19:53.28 | Qwell | modulus`: get a rackmount case |
| 19:53.33 | modulus` | ick |
| 19:53.36 | modulus` | i hate racks |
| 19:53.39 | modulus` | i work in a colo |
| 19:53.40 | troyb1 | why? |
| 19:53.42 | Qwell | :p |
| 19:53.42 | Katty | Qwell: yay :> |
| 19:53.49 | Qwell | well, they probably make wallmount mini-itx cases |
| 19:53.52 | troyb1 | if you work in a colo rack servers should make you jump for joy. |
| 19:54.05 | modulus` | i need 4-5 pci slots for fxo interfaces though |
| 19:54.08 | Qwell | umm |
| 19:54.11 | Qwell | why so many? |
| 19:54.18 | riddlebox | Qwell, where are the sounds by default in a source compile? |
| 19:54.22 | Qwell | just get a single tdm400p instead of 4 cheap ones |
| 19:54.26 | SibRphrek | hi |
| 19:54.30 | Qwell | riddlebox: /var/lib/asterisk/sounds/, I believe |
| 19:54.33 | modulus` | i already have fxo interfaces lying around |
| 19:54.35 | modulus` | single port |
| 19:54.38 | modulus` | nothing to do with them |
| 19:54.42 | Qwell | modulus`: 4 ports is bad... |
| 19:54.47 | SibRphrek | ? for y'all |
| 19:54.50 | Qwell | rather, using 4 pci slots is |
| 19:54.53 | SibRphrek | I am setting up a phone tree |
| 19:54.59 | modulus` | i never had irq conflicts |
| 19:55.05 | SibRphrek | but when the user enters ext 1,2 or 3, it never goes anywhere |
| 19:55.08 | Katty | Qwell: what did you eat for me? |
| 19:55.23 | Qwell | Katty: meatball - extra meat, extra cheese :( |
| 19:55.40 | Katty | :< |
| 19:56.08 | SibRphrek | does this look right to y'all |
| 19:56.08 | SibRphrek | http://pastebin.com/573714 |
| 19:56.23 | modulus` | it's horrid |
| 19:56.45 | riddlebox | Qwell, thats so wierd, I did a backup of my old asterisk stuff, then had to reinstall debian, so I just used the package in apt, and my custom sounds that I recorded dont work, unless I put the full path in? |
| 19:56.48 | Qwell | SibRphrek: looks fine |
| 19:57.00 | Qwell | riddlebox: yeah, debian is silly |
| 19:57.52 | SibRphrek | Qwell: the DTMF tones from ym cell phone aren't registering properly with that code |
| 19:58.11 | Qwell | SibRphrek: make sure your dtmfmode matches the provider |
| 20:03.35 | *** join/#asterisk vgster (n=vg@spc1-ledn1-3-0-cust136.seac.broadband.ntl.com) |
| 20:05.32 | modulus` | anyone here ever install asterisk on a wall mount pc case? |
| 20:06.55 | wisdom | yes |
| 20:07.10 | modulus` | do you have a link or a url for the wall mount case? |
| 20:07.21 | wisdom | there are a lot of them out there |
| 20:07.31 | modulus` | i can't seem to find any |
| 20:09.01 | poemius | lol, just read the april 1st asterisk 2.0 release :)... for a second I felt like... wow, I missed a lot of things :) |
| 20:09.44 | russellb | poemius: lol |
| 20:09.52 | russellb | poemius: you'd be surprised how many people took that seriously |
| 20:10.46 | poemius | lol it did sound believable :)... until I hit the date at the end :) |
| 20:11.10 | poemius | especially, as I have not followed things much lately :) |
| 20:11.47 | poemius | lol :) eliminate April 1sts , Friday the thirteenn |
| 20:11.50 | poemius | all mondays :) |
| 20:11.58 | poemius | put twice as many fridays :) |
| 20:12.07 | xmark | exit |
| 20:12.11 | xmark | quit |
| 20:12.13 | xmark | exit |
| 20:12.22 | xmark | oops sorry |
| 20:13.17 | airwaves | http://lists.digium.com/pipermail/asterisk-users/2005-April/098601.html |
| 20:13.20 | airwaves | hehe that's funny as hell |
| 20:14.21 | poemius | moved to c# and all :) |
| 20:14.24 | poemius | funky stuff :) |
| 20:14.50 | airwaves | i skimmed through it only briefly... then i got to the part where it said they removed SIP... I said 'wait a sec'.. looked at the date, and did the great DUHHHHHHHH |
| 20:15.20 | poemius | " And I'm so |
| 20:15.20 | poemius | <PROTECTED> |
| 20:15.20 | poemius | <PROTECTED> |
| 20:16.29 | airwaves | hehehe... i didn't see that part until i went through the whole thing in detail |
| 20:16.56 | airwaves | one funny part was where they were talking about an oracle DB for the whole thing... there are systems out there where the backends are entirely SQL |
| 20:17.56 | airwaves | but even those systems still befriend... some...arctic bird..or something |
| 20:18.39 | russellb | i'm going to write some april 1st code |
| 20:18.44 | SibRphrek | where do i go to fix |
| 20:18.44 | SibRphrek | WARNING[26809]: app_voicemail.c:2384 leave_voicemail: No entry in voicemail config file for '' |
| 20:19.24 | russellb | SibRphrek: you didn't specify a mailbox number |
| 20:19.32 | russellb | probably :) |
| 20:19.38 | SibRphrek | where? |
| 20:19.38 | Qwell | 1.2.1...I bet |
| 20:19.43 | russellb | Qwell: d'oh |
| 20:19.52 | russellb | SibRphrek: what version are you running |
| 20:20.16 | SibRphrek | 1.2.4 |
| 20:20.26 | Qwell | ooo |
| 20:21.32 | russellb | Qwell: 0wn3d |
| 20:28.59 | *** join/#asterisk backblue (n=moo@87-196-11-111.net.novis.pt) |
| 20:32.26 | *** join/#asterisk r0d3nt (i=nobody@wsip-24-234-241-145.lv.lv.cox.net) |
| 20:35.14 | *** join/#asterisk juanjoc (n=jcomella@222-32-235-201.fibertel.com.ar) |
| 20:39.26 | SibRphrek | dtmf does not work from cells phoens |
| 20:40.37 | modulus` | what's a dtmf? |
| 20:40.50 | *** join/#asterisk cosa (n=tmalkut@fw.orasoft.net.pl) |
| 20:40.56 | SibRphrek | when you press 1 it registers you pressed it |
| 20:41.01 | SibRphrek | dial tone something |
| 20:41.41 | modulus` | liar |
| 20:42.25 | SibRphrek | liar? |
| 20:42.33 | SibRphrek | i'm only talking about my dtmf |
| 20:42.37 | SibRphrek | in reference ot what i wrote earlier |
| 20:46.34 | [Airwolf] | Can anyone advise me in a VoIP phone with SIP/IAX support and some standard functions ? |
| 20:46.42 | [Airwolf] | What is a populair/good hardphone |
| 20:47.07 | modulus` | polycom ip601 |
| 20:47.10 | modulus` | rawk. |
| 20:49.59 | [Airwolf] | modulus`, the website doesn't say it supports iax |
| 20:51.15 | modulus` | airwolf, i never heard of a voip phone that supports iax |
| 20:52.49 | *** join/#asterisk eth00 (i=r00t@user-12lmut3.cable.mindspring.com) |
| 20:54.10 | [Airwolf] | modulus`, there are phones who do support it |
| 20:55.10 | modulus` | show me a link or url with such a phone |
| 20:55.21 | *** join/#asterisk Dr-Linux (n=Dreamer@host202-147-168-130.lhr.dancom.net.pk) |
| 20:56.25 | meshuga | i've used a couple of horribly crappy ones |
| 20:56.35 | meshuga | the iax ATA is ok |
| 20:56.51 | modulus` | that's a phone? |
| 20:57.20 | meshuga | no |
| 20:57.28 | [Airwolf] | modulus`, I have seen one at a friend of mine. But I have no idea what type it is. |
| 20:57.29 | meshuga | theres no good iax phones that i have seen as well |
| 20:57.48 | __AK__ | hi all, anyone know if it possible to enumlookup a name, ex instead of enumlookup +33170707070, i'd like to enumlookup name@domain.com |
| 20:57.59 | [Airwolf] | But I thought today more hard phones would support iax |
| 20:58.14 | [Airwolf] | But I'm wrong then. |
| 20:58.24 | meshuga | they should |
| 20:58.31 | meshuga | but they dont |
| 20:58.44 | [Airwolf] | I really hate Sip for the nat issues |
| 20:58.47 | SibRphrek | nice i got my crazy extentions.conf to work - still no dtmf off my cell tho |
| 20:59.00 | meshuga | SibRphrek: dtmfmode=inband |
| 20:59.06 | SibRphrek | meshuga: i tried that |
| 20:59.08 | *** join/#asterisk Olobola (n=casper_s@216.218.221.166) |
| 20:59.09 | meshuga | [Airwolf] : stun fixes that |
| 20:59.15 | *** join/#asterisk Ethon (i=arne@Oldman.steinkamm.com) |
| 20:59.24 | SibRphrek | i dunno if it has to be set to inband on my asterisk server or the one it's friending to |
| 20:59.34 | [Airwolf] | meshuga, well I don't have a stun server. |
| 20:59.34 | [Airwolf] | :P |
| 20:59.44 | meshuga | [Airwolf] : then start one. or use xten's |
| 21:00.03 | meshuga | SibRphrek: if you're going from * to *, rfc2833 should be fine. |
| 21:00.13 | [Airwolf] | meshuga, I have to look it up how that works and all. |
| 21:00.21 | meshuga | i'd debug iax and see if that even works |
| 21:00.22 | meshuga | [Airwolf] : its just a proxy. |
| 21:00.30 | meshuga | er, debug iax and see if you even see the dtmf tones |
| 21:00.35 | meshuga | on local |
| 21:00.38 | meshuga | then debug remote and see the same thing. |
| 21:00.47 | [Airwolf] | meshuga, is it hard to setup ? |
| 21:00.54 | Ethon | Hi |
| 21:01.08 | meshuga | [Airwolf] : not really |
| 21:01.26 | Ethon | Does someone has any information why Caller Presentation is broken on bri channels in asterisk 1.2.1 ? |
| 21:02.25 | [Airwolf] | meshuga, ok |
| 21:02.38 | SibRphrek | meshuga: rfc2833 nor inband work |
| 21:02.47 | meshuga | SibRphrek: i told you, debug the iax on either side |
| 21:02.53 | meshuga | and see if the dtmf tones are even being passed across. |
| 21:03.07 | meshuga | first locally, then remotely. |
| 21:03.16 | SibRphrek | they work from my softphone |
| 21:03.19 | SibRphrek | just not my cell phone |
| 21:05.07 | Olobola | will an unplugged (no phone line) wildcard cause asterisk to fail when starting? |
| 21:05.09 | *** part/#asterisk frenzy (n=frenzy@196.45.144.41) |
| 21:07.09 | Qwell | Olobola: no |
| 21:08.00 | russellb | Qwell: you have an svn server, don't you? |
| 21:08.08 | Qwell | russellb: at work, yeah |
| 21:08.14 | russellb | Qwell: how long does it take to set up |
| 21:08.25 | Qwell | not long at all, if you don't need mod_dav_svn |
| 21:08.39 | Qwell | create the repos, and thats about it, heh |
| 21:09.09 | russellb | ok, well i was about to do it on a LUG's server, but didn't want to start if I wasn't going to finish :) |
| 21:09.55 | Olobola | Qwell: thanks |
| 21:10.32 | Qwell | russellb: there is the easy way, and the way kpfleming did it :p |
| 21:11.44 | russellb | Qwell: lol |
| 21:12.19 | Corydon76-home | Actually, the way kpfleming did it isn't much more difficult |
| 21:12.43 | Qwell | with keys and all? It's a bit more involved |
| 21:12.47 | Corydon76-home | Kevin just added private keys to the mix |
| 21:13.23 | Corydon76-home | but there's no difference in the configuration of mod_dav_svn between the way Kevin did it and the way most people do it |
| 21:13.45 | *** join/#asterisk RoyK (n=roy@host-81-191-147-248.bluecom.no) |
| 21:14.15 | Corydon76-home | The difference is in the SSL layer |
| 21:14.26 | Olobola | Here is my problem: Loading module chan_modem_bestdata.so failed! |
| 21:14.46 | Qwell | Olobola: delete that file, and the others |
| 21:14.51 | Corydon76-home | Olobola: rm -f /usr/lib/asterisk/chan_modem* |
| 21:14.53 | *** join/#asterisk chapeaurouge (n=chap@user-85-201-82-146.tvcablenet.be) |
| 21:15.08 | russellb | /usr/lib/asterisk/moduiles/chan_modem* |
| 21:15.09 | russellb | :) |
| 21:15.14 | Corydon76-home | jinx |
| 21:15.19 | Qwell | modules |
| 21:15.24 | Qwell | russellb wins :p |
| 21:15.25 | russellb | s/moduiles/modules/ |
| 21:15.25 | Corydon76-home | Oh, modules, yeah |
| 21:15.35 | poemius | rm -R / anyone? :) |
| 21:15.48 | russellb | poemius: oh noes! |
| 21:15.49 | poemius | :) do not try this at home |
| 21:16.04 | Corydon76-home | Why oh why did we ever create that extra directory? |
| 21:16.22 | poemius | you mean the / directory? |
| 21:16.30 | Corydon76-home | No, modules |
| 21:16.39 | Corydon76-home | /usr/lib/asterisk/modules |
| 21:16.44 | russellb | no idea |
| 21:16.54 | Corydon76-home | /usr/lib/asterisk doesn't have anything else in it |
| 21:17.08 | poemius | maybe because modules starts with the same letters as moose? |
| 21:17.08 | Olobola | russellb: Asterisk Ready. Thank you. |
| 21:17.11 | russellb | i guess, just in case we wanted to put something else in there? |
| 21:17.23 | Corydon76-home | poemius: penis? |
| 21:17.34 | *** join/#asterisk chapeaurouge (n=chap@user-85-201-82-146.tvcablenet.be) |
| 21:17.46 | russellb | poemius: that doesn't make any sense :-p |
| 21:17.47 | Olobola | Qwell: thanks |
| 21:17.48 | poemius | mmm, I guess it's a hard one to figure out :) |
| 21:18.24 | poemius | brb phone |
| 21:18.47 | Qwell | because there are subdirs in /var/lib/asterisk/ |
| 21:18.51 | Qwell | consistancy... |
| 21:22.31 | modulus` | touch it |
| 21:22.37 | modulus` | oops! wrong window!! |
| 21:23.12 | *** join/#asterisk shido6 (n=shido6@d221-68-216.commercial.cgocable.net) |
| 21:26.32 | poemius | oh I feel touched :) |
| 21:29.32 | Corydon76-home | Where do you feel touched? :-) |
| 21:33.19 | justinu | touched by the hand of god |
| 21:34.03 | poemius | well I would not mind if it was by the hands of angelina jolie, cindy crawford, or... :) |
| 21:34.15 | poemius | I'm not picky :) |
| 21:34.35 | Corydon76-home | What about Brad Pitt? |
| 21:35.08 | poemius | mmmmmmmmm :) nah :), if he asks, I'll give him your number if you want |
| 21:36.56 | Corydon76-home | Heh, I wouldn't mind that at all |
| 21:37.53 | poemius | sounds good, if I ever meet him, I'll transmit the message :) |
| 21:38.53 | *** join/#asterisk fugitivo (n=ajf@201.216.246.181) |
| 21:45.15 | *** join/#asterisk frenzy (n=frenzy@196.45.144.41) |
| 21:45.30 | frenzy | can someone please tell me more about the FXO module |
| 21:45.39 | frenzy | does it come with ports? |
| 21:47.09 | russellb | um ... I don't understand your question |
| 21:47.21 | russellb | are you asking about the TDM400P? |
| 21:47.24 | frenzy | asin I take the TDM400P |
| 21:47.31 | frenzy | it has four ports |
| 21:47.40 | russellb | the FXO module enables one of the ports to be an FXO port |
| 21:47.40 | frenzy | and I take another FXO module |
| 21:47.44 | frenzy | so I can have 8 ports |
| 21:47.59 | frenzy | ohh |
| 21:48.08 | frenzy | if I want more than 4 ? |
| 21:48.13 | frenzy | Analog |
| 21:48.15 | russellb | then you can get the TDM2400P |
| 21:48.17 | *** join/#asterisk mexuar-tim (n=mexuar-t@host-212-158-206-61.bulldogdsl.com) |
| 21:48.19 | russellb | which allows 24 ports on one card |
| 21:48.30 | frenzy | the Developer kit |
| 21:48.35 | frenzy | how many ports does it have? |
| 21:48.45 | russellb | probably 1 FXS and 1 FXO |
| 21:48.59 | russellb | if you want to be able to extend beyond 4, you should consider the 2400P |
| 21:49.26 | frenzy | 2400P from the screen shot I see a connector |
| 21:49.32 | frenzy | are the ports external? |
| 21:49.41 | russellb | yes, it's an amphenol connector |
| 21:50.22 | frenzy | where can I buy the external port device? |
| 21:50.42 | russellb | anyone selling the TDM2400P probably sells one ... |
| 21:51.08 | frenzy | I dont see it in the digium store |
| 21:51.18 | russellb | http://www.netxusa.com/products/digium/2400pbundle.php |
| 21:57.33 | *** join/#asterisk shawarma (n=sh@sirius.linux2go.dk) |
| 21:58.01 | shawarma | Hi! Can any of you explain the difference between type=peer and type=friend in sip.conf? |
| 21:58.57 | *** join/#asterisk veepster (i=veepster@c-69-143-163-86.hsd1.va.comcast.net) |
| 21:59.20 | *** join/#asterisk RoyK (n=roy@host-81-191-147-248.bluecom.no) |
| 22:00.25 | veepster | hey, I just installed asterisk and zaptel software on an ubuntu box. where do I go from here? I have some basic questions : why do I need zaptel. what kind of hardware do I need to get this to work. Do I need two nics. whats a good way to test etc |
| 22:00.28 | *** join/#asterisk _DAW (n=_DAW@adsl-6-66-81.msy.bellsouth.net) |
| 22:01.26 | _DAW | Hello all |
| 22:02.07 | Corydon76-home | You only need zaptel if you have TDM hardware |
| 22:03.31 | Corydon76-home | and you only need two nics if your network layout demands it |
| 22:11.36 | veepster | thanks. should I be worried if I dont know what TDM means? |
| 22:11.43 | dpryo | No :) |
| 22:12.03 | veepster | ok:) |
| 22:12.24 | fugitivo | ~seen coppice |
| 22:12.31 | jbot | coppice <n=chatzill@212.197.17.210.dyn.pacific.net.hk> was last seen on IRC in channel #asterisk, 13h 14m 18s ago, saying: 'you mean packets are delivered by a cron job? :-\'. |
| 22:14.33 | *** part/#asterisk bitmap (n=mdl@c-24-118-12-11.hsd1.mn.comcast.net) |
| 22:16.13 | veepster | anyway, whats the basic hardware I need in addition to a linux PC with asterisk installed to test voip functionality? As mentioned earlier, I have also installed zaptel but dont know why |
| 22:24.41 | poemius | you need ztdummy for timing purposes |
| 22:24.52 | poemius | as well as a soft client like x lite or sjphone |
| 22:25.11 | poemius | :) as provided in the compiling instructions |
| 22:28.23 | veepster | apt-get in ubuntu bypasses compilation unfortunately |
| 22:30.13 | shawarma | I'm having trouble understanding the difference between a friend and a peer in sip.conf. The info on voip-info.org seems to indicate, that a peer is someone you only place calls TO, but a sample asterisk conf from a sip provider I found says to use type=peer... How does that make sense? |
| 22:31.14 | poemius | welcme back :) |
| 22:31.29 | kram | thank ye |
| 22:32.54 | Corydon76-home | shawarma: because a peer has evolved over time |
| 22:33.10 | shawarma | Corydon76-home: So... which one is true? |
| 22:33.15 | Corydon76-home | Both |
| 22:33.39 | shawarma | Corydon76-home: figures... |
| 22:33.54 | *** join/#asterisk wrench (n=signal@68-118-224-178.dhcp.oxfr.ma.charter.com) |
| 22:34.15 | Corydon76-home | Due to the SIP spec, we were unable to confine a peer in the SIP stack into only receiving calls |
| 22:34.44 | Corydon76-home | It had to be able to send new calls back to us, as well |
| 22:34.58 | sivana | so for SIP, does that mean user/peer no longer and friend/peer are the same? |
| 22:35.06 | shawarma | So what is the difference? |
| 22:35.30 | Corydon76-home | For the time being, it's ambiguous |
| 22:35.43 | Corydon76-home | A user is restricted to only sending calls to the PBX |
| 22:35.53 | Corydon76-home | However, a peer is not so restricted |
| 22:36.50 | [av]bani | ... |
| 22:36.54 | Corydon76-home | If you want to resolve the ambiguity in a calm, sophisticated way, look up the authors of the SIP spec, find them on the street, and give them a bloody nose. |
| 22:37.35 | zamba | i'm trying to set up a cisco 7940GP on asterisk using sccp |
| 22:37.39 | [av]bani | yay |
| 22:37.50 | zamba | i can get the phone seemingly connected.. i see the line number i gave it on the display |
| 22:37.57 | zamba | but after that, it just freezes totally |
| 22:38.14 | shawarma | Corydon76-home: Will do.. Until then, which one should I choose? |
| 22:38.17 | zamba | using tcpdump i see that the phone tries to get a tlv-file |
| 22:38.32 | Corydon76-home | shawarma: whichever one works |
| 22:38.50 | shawarma | Corydon76-home: :-D |
| 22:39.22 | shawarma | Corydon76-home: I see.. Does the difference by any chance have anything to do with the peer having to authenticate against me on incoming calls? |
| 22:39.53 | Corydon76-home | It depends greatly on how you configured it. |
| 22:41.27 | Corydon76-home | If we had built in only 3 switches to configure everything, Asterisk would be a lot simpler, but it wouldn't interoperate with as much equipment. Hence, there's a lot of choices when it comes to configuration. |
| 22:41.46 | Corydon76-home | Hence, I can't give you a more clear answer |
| 22:43.52 | shawarma | Corydon76-home: Right, ok. Well, the actual problem is that my voip provider is apparantly chaning some software, so they want me to change the hostname of the server, I'm connecting to, but as soon as I change it, any calls I place go dead after about three seconds. sip debug tells me that the server sends me a 503 of some sort.. It |
| 22:43.57 | shawarma | whoops. |
| 22:44.20 | shawarma | it's really quite odd and their support is not very helpful. |
| 22:44.25 | *** join/#asterisk kizmet (n=kizmet@freematrix/sponsor/kizmet) |
| 22:45.30 | Mavvie | digiums phone number is broken :-) |
| 22:45.45 | kizmet | Mavvie, Maybe they upgraded Asterisk :P |
| 22:48.04 | Abydos313 | kizmet heh, good one |
| 22:48.48 | kizmet | Abydos313, Some things in 1.2.X arnt the same as in 1.0.X :) |
| 22:49.08 | *** join/#asterisk samueltc (n=samuel@levinux.UQAR.UQUEBEC.CA) |
| 22:49.10 | Abydos313 | it's not just an upgrade? |
| 22:49.36 | Abydos313 | still funny when you think of it. they upgrade their own shit and it stops working. that would be funny |
| 22:49.44 | samueltc | hi, any recommendation for a terminaison provider with inbound dtmf support? |
| 22:50.20 | sivana | are there that many providers with no dtmf support? |
| 22:50.29 | kizmet | sivana, yes lol |
| 22:50.45 | sivana | wow.. didnt' think dtmf was such a big deal |
| 22:51.35 | kizmet | sivana, 3/5 SIP providers in Australia dont provide a stable DTMF service. |
| 22:51.46 | *** join/#asterisk franck (n=franck@tikiwiki/franck) |
| 22:52.06 | Abydos313 | kizmet doesn't that make it difficult to integrate older phone systems to voip |
| 22:52.32 | kizmet | Abydos313, yup |
| 22:52.50 | Abydos313 | sucks not to be able to dail |
| 22:52.52 | Abydos313 | dial |
| 22:53.00 | kizmet | Abydos313, hehe. |
| 22:53.16 | kizmet | Abydos313, it usually affects IVR more than anything. |
| 22:53.44 | Abydos313 | i bet that would be a pain very quickly |
| 22:54.44 | Abydos313 | just got a dell xps 933mhz box for free :)) it's pretty damn fast for a p3 |
| 22:57.22 | *** join/#asterisk forao (n=fasdfasd@pool-138-89-168-16.mad.east.verizon.net) |
| 23:00.14 | samueltc | so nobody wants my money? |
| 23:01.01 | kizmet | samueltc, your in the usa i assume ? |
| 23:01.07 | kizmet | or canada |
| 23:02.05 | kizmet | Abydos313, I got a Quad Opteron Dual Core box last night delivered to my door :) |
| 23:02.07 | samueltc | canada, but I need terminaison in UK, canada and US |
| 23:02.24 | kizmet | samueltc, Uhm Gimme a sec i'll do some research for you :) |
| 23:02.34 | samueltc | kizmet: I appreciate |
| 23:02.46 | poemius | oh if you want to give me money just like that :) no problem :) |
| 23:03.03 | samueltc | hehe |
| 23:03.11 | kizmet | samueltc, http://www.band-x.com |
| 23:03.17 | kizmet | samueltc, for UK termination |
| 23:03.33 | kizmet | samueltc, I use them as one of my upstreams :) |
| 23:04.21 | samueltc | I need short term (for a demo tomorrow) |
| 23:04.43 | kizmet | samueltc, http://www.voip-user.org *whistle* |
| 23:04.56 | kizmet | it might not have a - |
| 23:05.11 | xachen | it does |
| 23:05.13 | xachen | erm |
| 23:05.14 | xachen | sec |
| 23:05.21 | xachen | I was thinking voip-info there :) |
| 23:05.27 | SkramX | http://pastebin.com/574019 <== Cisco Phone (SKINNY) errors, please help :) |
| 23:05.55 | SkramX | Feb 26 17:05:37 ERROR[16097]: chan_skinny.c:2363 handle_message: Rejecting Device SEP00D0BA8474DA: Device not found |
| 23:06.08 | kizmet | The register has an article blaming the uprise in P2P traffic as a result of file sharing and systems like BtTorrent for a downturn in the quality of Skype calls. |
| 23:06.15 | lunaphyte | SkramX: i have trouble with a skinny phone too - someone suggested chan_sccp, which has worked great so far. |
| 23:06.18 | kizmet | rofl Skype sucks anyways *sigh* |
| 23:06.25 | lunaphyte | s/have/had/ |
| 23:06.28 | SkramX | i think i may have fixed it. |
| 23:06.29 | SkramX | wooo |
| 23:06.34 | SkramX | lunaphyte: what do you mean? |
| 23:06.38 | Mavvie | man... digium phone support is hell++ |
| 23:07.07 | SkramX | lunaphyte: link? |
| 23:07.15 | SkramX | it keeps "requesting load id" |
| 23:07.38 | lunaphyte | chan-sccp.berlios.de |
| 23:07.59 | x86 | MY INBOUND FROM PSTN.... IT IS TEH WORKIE! |
| 23:08.05 | x86 | finally :) :) |
| 23:08.23 | franck | hi |
| 23:08.31 | franck | I get some cross talk on my zap lines |
| 23:08.37 | franck | How comes? |
| 23:10.45 | SkramX | lunaphyte: was it a 12SP+? |
| 23:10.51 | lunaphyte | yes |
| 23:11.10 | Mavvie | franck: sound from other channels or misconnected channels? |
| 23:12.07 | franck | Mavvie: it seems sound from other channels |
| 23:12.14 | franck | zap channels |
| 23:12.24 | Mavvie | franck: aha, don't have that problem yet. |
| 23:12.41 | kizmet | franck, Analog or PRI ? |
| 23:12.51 | franck | I run my card in debug mode... so I will remove that and see... but wonder if it is a well known issue |
| 23:13.02 | x86 | anyone here in the UK? |
| 23:13.10 | SkramX | <PROTECTED> |
| 23:13.10 | SkramX | Device SEP00D0BA8474DA is attempting to register |
| 23:13.10 | SkramX | Feb 26 17:12:22 ERROR[32341]: chan_skinny.c:2363 handle_message: Rejecting Device SEP00D0BA8474DA: Device not found |
| 23:13.13 | kizmet | x86, I have a number in the Uk heh |
| 23:13.19 | SkramX | but its in my skinny.conf! |
| 23:13.29 | x86 | kizmet: me too, i'm trying to debug it ;) |
| 23:13.42 | kizmet | x86 who with ? |
| 23:13.53 | kizmet | x86 i can call u for cheap if you want.... |
| 23:13.54 | x86 | ipstar.us |
| 23:14.08 | x86 | ok call me if you would |
| 23:14.11 | x86 | +44-871-3094409 |
| 23:14.12 | franck | kizmet: analog |
| 23:14.17 | franck | the new tdm24000 cards |
| 23:14.29 | kizmet | franck, hmmm i have had that problem before are you using a X101P clone ? |
| 23:14.33 | franck | wctdm24xxp |
| 23:14.35 | SkramX | lunaphyte: how do you find out your version? |
| 23:15.04 | franck | What is a X101P clone |
| 23:15.06 | lunaphyte | what version? |
| 23:15.31 | kizmet | x86, All i get are beeps in 3 second lengths with half a sec time between |
| 23:15.54 | x86 | you sure you're dialing international correctly? |
| 23:15.55 | SkramX | SENTIEN*CLI> skinny show devices |
| 23:15.58 | SkramX | florian SEP(00D0BA8474D 70.116.9.213 0 N 1 |
| 23:16.03 | SkramX | Device SEP00D0BA8474DA is attempting to register |
| 23:16.03 | SkramX | Feb 26 17:15:41 ERROR[7075]: chan_skinny.c:2363 handle_message: Rejecting Device SEP00D0BA8474DA: Device not found |
| 23:16.06 | SkramX | that makes no sense! |
| 23:16.28 | kizmet | x86, other than being connected through http://www.band-x.com for my UK termination yes :) |
| 23:17.07 | *** join/#asterisk NewSole (n=dave@d226-105-29.home.cgocable.net) |
| 23:17.17 | NewSole | file file where is file |
| 23:17.35 | file | what |
| 23:17.54 | NewSole | need to talk... shido6 sent me |
| 23:18.25 | file | ... |
| 23:19.50 | SkramX | lunaphyte: so how does sccp work.. it is instead of using skinny? |
| 23:19.50 | poemius | :) wow 3 . :) |
| 23:20.04 | lunaphyte | SkramX: yes |
| 23:20.44 | SkramX | and you just followed the instructions and it just worked? |
| 23:23.18 | x86 | kizmet: i would verify your band-x.com setup... maybe it doesnt let you dial national numbers or something, but I had another native UK'er test it and it worked fine :) |
| 23:24.00 | NewSole | ~pb |
| 23:24.01 | jbot | [pb] a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
| 23:24.31 | lunaphyte | SkramX: yes |
| 23:27.00 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 23:27.00 | *** mode/#asterisk [+o anthm] by ChanServ |
| 23:27.52 | acidchild | vonage + asterisk |
| 23:27.54 | acidchild | can it be done? |
| 23:28.07 | poemius | acidchild : broadvoice or voicepulse |
| 23:28.17 | Mavvie | aaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaargh++ |
| 23:28.18 | Mavvie | 1300 88 686 doesn't work via our PRI provider, it doesn't work via my mobile, but it does work via my home phone. |
| 23:28.23 | Mavvie | how to explain that to customers? |
| 23:28.54 | acidchild | poemius: i wondered what i could do with asterisk |
| 23:29.07 | kizmet | Mavvie, I can test from my voip if you want. |
| 23:29.09 | acidchild | just for the fun of it :) |
| 23:29.16 | kizmet | Mavvie, You seem to be an australian :) |
| 23:29.22 | poemius | Mavvie : vonage locks their people with mac address with their unlimited plan |
| 23:29.27 | poemius | maybe with softphone |
| 23:29.37 | poemius | oops directed at acidchild |
| 23:29.37 | acidchild | poemius: is there away i can let my mate use my phone(voip) line? via like asterisk client? |
| 23:29.52 | poemius | with broadvoice yes |
| 23:30.00 | acidchild | ah cool |
| 23:30.08 | Mavvie | kizmet: by residence, not by nature. |
| 23:30.12 | kizmet | Mavvie, Whats the full number your missing a digit above :) |
| 23:30.19 | Mavvie | kizmet: none. |
| 23:30.28 | kizmet | Mavvie, siff ;) |
| 23:30.41 | kizmet | 1300 88 686 |
| 23:30.46 | kizmet | your missing a digit |
| 23:31.05 | Mavvie | kizmet: the number gets called, and connected, when I dial it from my home phone. |
| 23:31.11 | Mavvie | I even get a person on the line! |
| 23:31.16 | kizmet | heh |
| 23:31.45 | SkramX | can i change the ringer for my 12SP+? |
| 23:31.56 | SkramX | lunaphyte: *hug*, chan_sccp worked |
| 23:32.12 | lunaphyte | good |
| 23:32.14 | *** join/#asterisk FuriousGeorge (n=Brian@ool-44c5a9b8.dyn.optonline.net) |
| 23:32.34 | *** join/#asterisk zimba022 (n=zimba022@200.77.174.41) |
| 23:32.55 | Mavvie | kizmet: but that could indeed be the reason it didn't get through on the other networks. |
| 23:33.01 | x86 | hmm |
| 23:33.02 | SkramX | lunaphyte: were you able to set it to say something on the screen? |
| 23:33.03 | kizmet | Mavvie, :) |
| 23:33.19 | FuriousGeorge | ll |
| 23:33.22 | FuriousGeorge | hi all |
| 23:33.24 | kizmet | Mavvie, Telstra & Optus lines are smart (sorta) they guess numbers :) |
| 23:33.32 | x86 | when i get international calls, it shows me the CallerID correctly, but it does not give me the orginating country code... any way to append that? |
| 23:33.52 | kizmet | x86, unless your incoming passes it no. |
| 23:33.57 | x86 | well |
| 23:34.12 | x86 | i know everything that comes through a given SIP extension will be in the UK |
| 23:34.25 | x86 | so i could append it manually with SetCallerID or something? |
| 23:34.28 | kizmet | unless you were to 'guess' the country code on the pattern of the callers number :) |
| 23:34.57 | lunaphyte | SkramX: time and date. |
| 23:35.00 | kizmet | x86, you could Set(CALLERIDNAME="Uk Caller"); I think thats right :/ |
| 23:35.08 | kizmet | x86, checking |
| 23:35.43 | kizmet | x86, Set(CALLERID(name)="1300 663 721 (Support)"); |
| 23:35.49 | zimba022 | set(callerid(name)="uk caller") |
| 23:35.53 | kizmet | as an example will set the callerid name of the call :) |
| 23:36.11 | kizmet | Mavvie, yes that is my 1300 number :) |
| 23:36.30 | [av]bani | http://www.killsometime.com/Video/video.asp?ID=421 |
| 23:36.34 | Mavvie | aeria networks. |
| 23:36.40 | kizmet | Mavvie, :) |
| 23:37.26 | Mavvie | google is great for the blackpages :-) |
| 23:37.38 | kizmet | :) |
| 23:38.45 | x86 | kizmet: there is also SetCallerID("UK Caller (${CALLERIDNAME})" <+44-${CALLERIDNUM}>) |
| 23:38.52 | SkramX | lunaphyte: what is the line in extensions.com to call the phone? |
| 23:39.05 | kizmet | x86, yes :) |
| 23:40.06 | kizmet | Mavvie, there used to be greypages.com.au but they got knocked off the net |
| 23:41.38 | *** join/#asterisk _DAW (n=_DAW@adsl-6-66-81.msy.bellsouth.net) |
| 23:41.46 | kram | any bay area folks here? |
| 23:41.59 | file | kram: sadly no :( |
| 23:42.03 | kram | hrm |
| 23:43.31 | russellb | kram: !!!!!!!! |
| 23:43.38 | kram | russellb! |
| 23:43.53 | file | russellb: I miss drumkilla >.< |
| 23:44.01 | russellb | I'm sorry :( |
| 23:44.19 | SkramX | lunaphyte: what is the line in extensions.com to call the phone? |
| 23:44.25 | SkramX | thats all I need to know |
| 23:44.27 | poemius | bay area... anywhere near casablanca? |
| 23:44.37 | poemius | we have a bay too, you know :) |
| 23:44.58 | poemius | plenty of camels too |
| 23:45.14 | russellb | ok, everyone makes calls through iaxtel |
| 23:45.16 | russellb | all at once |
| 23:45.53 | poemius | synchronize watches |
| 23:46.02 | poemius | this message will self destruc |
| 23:46.31 | poemius | . t |
| 23:52.07 | SkramX | Feb 26 17:52:01 WARNING[21545]: rtp.c:1017 ast_rtp_settos: Unable to set TOS to 184 |
| 23:52.10 | SkramX | ? |
| 23:52.29 | *** join/#asterisk Psykick (n=anon@203.167.226.250) |
| 23:52.32 | Psykick | hi guys |
| 23:52.53 | Psykick | how do I specify the default codec for asterisk to use on all calls? |
| 23:54.06 | kizmet | Psykick, add a 'disallow =all' line and add a 'allow=<codec>' line for eatch codec in priority of how you want them to be used. This is in your sip.conf or iax.conf |
| 23:54.34 | kizmet | Please not that unless you have purchased a G729 licence from Digium you cannot use it for IVR purposes. |
| 23:55.05 | Psykick | ok |
| 23:55.12 | Psykick | I'm trying out the g726 |
| 23:55.28 | *** join/#asterisk dja_ (n=alden@cpe-24-95-62-160.columbus.res.rr.com) |
| 23:57.56 | Psykick | does that also work for inbound? |
| 23:58.07 | Psykick | setting the priority in iax.conf? |
| 23:58.12 | kizmet | Psykick yes. |
| 23:58.56 | Psykick | ok |
| 23:59.18 | Psykick | keep seeing gsm even though I've get ulaw and alaw above it for normal inbound calls |
| 23:59.24 | Psykick | oops I mean got |
| 23:59.38 | kizmet | does the other host support ulaw and alaw ? |
| 23:59.40 | *** join/#asterisk RoyK (n=roy@10.80-203-106.nextgentel.com) |