00:00.02 | nextime | for example, for many people freepbx is good and work very well |
00:00.12 | nextime | for others it suck |
00:01.00 | Ubeyguy | where can i download freepbx ? |
00:01.20 | nextime | freepbx.org? |
00:01.43 | brookshire | sf.net/ampportal |
00:01.45 | brookshire | lol |
00:01.48 | brookshire | i think |
00:04.53 | *** join/#asterisk CrippsFX (n=CrippsFX@Kitchener-HSE-ppp3568787.sympatico.ca) |
00:09.38 | CrashHD | how can I make calls authenticated (iax) based on ip? |
00:09.46 | CrashHD | I'm already using host=IP |
00:09.54 | CrashHD | but it doesn't seem to mark the calls as authenticated |
00:11.27 | *** join/#asterisk _Paulo_ (n=pirch@201-13-17-36.dsl.telesp.net.br) |
00:11.36 | *** join/#asterisk Vazir (i=anton@82.198.21.17) |
00:11.40 | _Paulo_ | ~seen coppice |
00:12.03 | jbot | coppice <n=chatzill@53.162.17.210.dyn.pacific.net.hk> was last seen on IRC in channel #asterisk, 18h 2m 51s ago, saying: 'its wrong that short DTMF is produced when you press keys on a cellphone. well controlled lengths are produced. for most makes of infrastructure the lengths are very long. they are never too short, though'. |
00:12.04 | Vazir | Does anyone used H323 with Asterisk? |
00:12.04 | Vazir | Hi Folks :) |
00:12.30 | _Paulo_ | Vazir, I tried but open323 makes my server freeze |
00:13.13 | Vazir | Yeah, I have some luck, but I need stable solution :) |
00:13.31 | *** join/#asterisk asterboy (n=kevin@S010600485480f4be.ed.shawcable.net) |
00:13.38 | Vazir | Maybe anyone knows stable combination of openh323/pwlib for example |
00:14.03 | asterboy | Anyone know a TFTP server for Linux, (non-rpm)? |
00:14.10 | asterboy | can' |
00:14.20 | Qwell | asterboy: there are several |
00:14.28 | asterboy | I've been looking to no avail. |
00:14.31 | Vazir | any linux have tftpd daemon |
00:14.41 | asterboy | not if your an LFS type like me. |
00:14.44 | _Paulo_ | asterboy, look at http://freshmeat.net |
00:14.58 | asterboy | sourforge.net |
00:15.01 | Vazir | than use debian |
00:15.01 | Qwell | selinux-tftpd, tftp-hpa, netkit-tftp, atftp, linksys-tftp |
00:15.11 | Qwell | at least 3 of those are tftpd's |
00:15.45 | Vazir | any chan_ss7 experience guys? |
00:16.01 | Qwell | asterboy: I use the hpa one |
00:16.29 | asterboy | looking... |
00:16.35 | nextime | Vazir : for the moment the only one stable solution for h323 with asterisk in my opinion is chan_oh323, isn't the most performant, but is the most stable, the second one is chan_woomera, the third is chan_ooh323 that i like but is really young, and the latest for stability but good for performance is chan_h323 |
00:17.07 | asterboy | thought sourceforge and freshmeat were the same. |
00:17.16 | nextime | anyway, i'm testing hard ooh323 but i'm using only oh323 in production at the moment |
00:17.33 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
00:17.35 | Vazir | :) funny - but for me chan_h323 was most stable - and passed 10000 calls before crash |
00:17.42 | *** join/#asterisk Skumling (n=skumling@fw.sg12.dk) |
00:17.44 | brookshire | aster: same people, different content |
00:17.45 | justinu | that's stable? |
00:17.46 | justinu | lol |
00:18.01 | nextime | Vazir : 10000 calls before crash isn't stable for me :) |
00:18.13 | Vazir | can you share any experience for your oh323 |
00:18.14 | nextime | stable is "don't crash at all" |
00:18.18 | *** part/#asterisk CrippsFX (n=CrippsFX@Kitchener-HSE-ppp3568787.sympatico.ca) |
00:18.22 | Vazir | Glad to find someone H323 ! |
00:18.43 | Vazir | combination of openh323/pwlib/asterisk versions? |
00:19.13 | nextime | Asterisk SVN-trunk-r7230 built by root @ voip01 on a i686 running Linux on 2005-12-18 12:45:23 UTC |
00:19.24 | nextime | asterisk-oh323-0.7.3 |
00:19.37 | nextime | openh323_Mimas_patch2 |
00:19.41 | nextime | pwlib_Mimas_patch2 |
00:19.55 | shmaltz | nextime, then you should try Avaya their Parner system will not crash by 10000 calls |
00:19.56 | Vazir | uhum... what distro do you use? |
00:20.09 | nextime | Vazir : debian |
00:20.22 | Vazir | The same as me |
00:20.35 | Vazir | I did tried both on stable and testing. What yours> |
00:20.36 | Vazir | ? |
00:20.38 | nextime | shmaltz : this combination of * and oh323 is really stable for me |
00:20.58 | shmaltz | nextime, good for you |
00:21.09 | Vazir | <PROTECTED> |
00:21.31 | nextime | i'm generating a about 200000 calls/day and is running good from 4 m without problem |
00:21.55 | shmaltz | nextime, for what type of business? |
00:21.59 | nextime | ( before i was using a oldest version of asterisk ) |
00:22.12 | Vazir | I do use SIP/H323 convertor for now and would try your combination today!... Hm strange if I did not try |
00:22.14 | Vazir | yet |
00:22.17 | nextime | shmaltz : traffic reselling |
00:22.35 | shmaltz | why did you add oh323 to the mix? |
00:22.49 | Vazir | I do use MVTS as a softswitch and want to have an Asterisk as a gateway to PSTN |
00:23.02 | shmaltz | nextime, you doing it in VoIP, not TDM, right? |
00:23.05 | asterboy | thx for the tftp-hpa |
00:23.21 | nextime | shmaltz : because some telco support *only* h323 |
00:23.34 | asterboy | looks good. Will save me from running my Polycom init files at a Windows box. |
00:23.35 | Vazir | I would say MOST telcos... |
00:23.38 | nextime | shmaltz : voip AND tdm |
00:23.44 | Vazir | H323 is in commercial world |
00:23.46 | shmaltz | so you have telco giving you traffic over h323? |
00:23.49 | asterboy | Strange that Sourceforge has no tftp servers. |
00:23.59 | shmaltz | Vazir, not here in the US |
00:24.02 | nextime | shmaltz : no, i give h323 traffic to telco :) |
00:24.10 | Ubeyguy | Anyone here use MetaSwitch? |
00:24.17 | Frogzoo | asterboy: sourceforge has atftpd dude iirc |
00:24.31 | Vazir | shmaltz, maybe for sure :) |
00:25.52 | alephco1 | I have a provider using it.... I wish I had one. |
00:25.52 | Vazir | But even with US partners I'm offered H323 first :) |
00:25.52 | shmaltz | Vazir, maybe for sure what? |
00:25.52 | nextime | shmaltz : anyway, i'm using sip and iax2 too |
00:25.52 | Ubeyguy | my company has a metaswitch 4/5 class |
00:25.52 | nextime | but in commercial world h323 is the most requested and most time is the only one solution in voip |
00:25.52 | asterboy | must have typed in my search wrong, couldn't find squat. |
00:25.52 | shmaltz | in any case if asterisk is configured correctly, you shoulnd't have a problem getting 10000 calls |
00:26.12 | CrashHD | how do I force an incoming iax2 call into a certain context? |
00:26.23 | shmaltz | CrashHD, iax.conf |
00:26.25 | nextime | shmaltz : so, my config is correct :) |
00:26.40 | shmaltz | nextime, I didn't see your config, I don't know |
00:26.55 | CrashHD | shmaltz: that I know, but it doesn't seem to be working |
00:27.10 | nextime | shmaltz : trust me, my config is working perfectly for my needs :) |
00:27.13 | shmaltz | CrashHD, you did a reload in asterisk CLI? |
00:27.17 | CrashHD | ya |
00:27.21 | shmaltz | nextime, I got no clue |
00:27.37 | CrashHD | OTICE[415]: chan_iax2.c:6799 socket_read: Rejected connect attempt from 71.16.179.149, request '6023571734@default' does not exist |
00:27.40 | shmaltz | CrashHD, and how do you know it's going to the wrong context? |
00:27.47 | CrashHD | those notices |
00:28.13 | shmaltz | does shoe dialplan default show you an extension for 6023571734? |
00:28.20 | shmaltz | shoe==show |
00:28.25 | CrashHD | there is no default in the dialplan |
00:28.34 | CrashHD | http://pastebin.com/611597 |
00:28.37 | CrashHD | is my iax context |
00:28.37 | shmaltz | CrashHD, then you answered your quesiotn |
00:28.47 | CrashHD | it shouldn't be going to the default |
00:28.58 | CrashHD | I have context set to inbound |
00:29.07 | shmaltz | CrashHD, but does your extension.conf have a inbound context? |
00:29.11 | CrashHD | yes |
00:29.23 | shmaltz | CrashHD, pb it |
00:29.50 | shmaltz | plus, pb the follwoing command from the CLI: |
00:29.52 | shmaltz | show dialplan inbound |
00:31.05 | CrashHD | http://pastebin.com/611601 |
00:31.08 | CrashHD | simple simon stuff |
00:31.15 | CrashHD | it's just not going to the right context |
00:31.43 | *** join/#asterisk Ridgeback (n=jircii@104.243.8.67.cfl.res.rr.com) |
00:32.13 | Ridgeback | anyone know if you can push SIP prescence status through an IAX trunk? |
00:33.07 | CrashHD | shmaltz any ideas? |
00:33.39 | Ridgeback | my buddy watch/status works fine locally. just would like it to span over the trunks |
00:33.40 | Frogzoo | asterboy: I lie: I got it from the ubuntu repos, but the copyright says: ftp://ftp.mamalinux.com/pub/atftp |
00:33.57 | shmaltz | CrashHD, looks like you are requesting to dial with the @default at the end |
00:34.11 | shmaltz | Redgeback, I doubt it |
00:34.24 | Ridgeback | shmaltz: ok thanks |
00:34.36 | CrashHD | so the inbound call is directing to the default context? |
00:35.11 | CrashHD | I'm pretty sure the dial is just DIAL(IAX2/${PEERNAME}/${EXTEN}) |
00:35.19 | CrashHD | any reason why it would want to veer toward default? |
00:36.16 | CrashHD | it is currently hitting the last context that is set in the iax.conf |
00:36.18 | CrashHD | why would that be? |
00:36.41 | asterboy | I'm trying to run it but it just exits without any message. |
00:36.46 | asterboy | compiled nice |
00:36.53 | asterboy | tried tftpd -l |
00:37.14 | asterboy | but I had to copy it from the compile directory, make install did not put it in /usr/bin |
00:37.26 | asterboy | what do you start it with? inetd? |
00:40.47 | Frogzoo | asterboy: either/or |
00:40.51 | shmaltz | CrashHD, try including the inbound context in the default context, see if the call goes thru |
00:41.06 | Frogzoo | asterboy: you remembered ./configure? |
00:41.14 | CrashHD | shmaltz: I think I figured out, the call is coming in as unauthenticated |
00:41.16 | asterboy | hmm, not sure why it won't start, strange |
00:41.35 | CrashHD | shmaltz: but why that is I'm unsure. I have the host=HOSTIP setup |
00:41.35 | asterboy | ohya, did the ./configure; make; make install |
00:41.44 | Frogzoo | asterboy: os depending, may need root perm to start |
00:41.46 | shmaltz | CrashHD, exactly thats why I told you to do the last test |
00:41.47 | asterboy | where does it put the log output? |
00:42.01 | asterboy | ya I should check the README again. |
00:42.23 | Frogzoo | asterboy: some OSs won't let you access ports < 1024 without root |
00:42.34 | asterboy | I'm ok there. |
00:42.44 | CrashHD | shmaltz: ok hold one sec |
00:44.34 | CrashHD | shmaltz: yes the calls go through at that point |
00:44.49 | CrashHD | shmaltz: so how can I get this call to auth? |
00:45.05 | shmaltz | I don't know, what is your setup? |
00:45.09 | shmaltz | try register first |
00:45.28 | CrashHD | shmaltz: the call is inbound from a voip provider (they don't register, just send the call) |
00:45.47 | shmaltz | CrashHD, then you have to register |
00:46.02 | *** join/#asterisk [ProB]CrazyMan (n=crazyman@p549F3734.dip0.t-ipconnect.de) |
00:46.16 | CrashHD | shmaltz: isn't there a way to do it only based on ip? |
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01:02.25 | *** join/#asterisk Ubeyguy (n=chatzill@vannuys-cuda2-70-35-20-230.vnnyca.adelphia.net) |
01:02.33 | Ubeyguy | hi is there a asterisk@home room? |
01:05.34 | Abydos313 | not really, what's your question |
01:06.22 | Ubeyguy | ok well i have a VoIP account and the only info i have is Username password and server |
01:06.46 | Ubeyguy | now when i try to add a trunk theres a out section a in section and i dont know what to put ( Using AMP) |
01:07.08 | Abydos313 | most providers have a howto on setting up with asterisk. |
01:07.15 | Abydos313 | is it an iax account or sip |
01:07.23 | Ubeyguy | SIP |
01:07.47 | Abydos313 | who is the provider. a quick search will probably show a howto. |
01:08.22 | Ubeyguy | well my company set up there own VoIP metaswitch and there is really no one i can talk to that manages it |
01:09.20 | *** join/#asterisk theorem_ (n=theorem@pool-71-127-251-111.nwrknj.fios.verizon.net) |
01:09.21 | Abydos313 | outgoing sounds like the place to put the info |
01:09.22 | theorem_ | wee |
01:09.57 | Ubeyguy | what about Incoming Settings? |
01:09.58 | theorem_ | ok, so if I just installed asterisk onto debian ... what should I do next ? |
01:10.32 | Ubeyguy | theorem_: format hd |
01:10.35 | Abydos313 | i've never done it thru asterisk@home . only put entries in sip.conf or iax.conf |
01:10.41 | theorem_ | dang |
01:10.54 | Ubeyguy | Abydos313: can you give me a sample of sip.conf you use |
01:10.58 | *** join/#asterisk exten123 (n=exten@60.49.6.190) |
01:11.47 | Abydos313 | they are all over voip-info.org |
01:13.40 | *** join/#asterisk Mw3_ (n=mw3@national.t-error.hu) |
01:17.44 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
01:17.45 | *** mode/#asterisk [+o russellb] by ChanServ |
01:20.22 | CrashHD | why is it that if no secret or user is given (or included in context) on an iax context and only host= is used the call still shows as unauthenticated? |
01:24.31 | russellb | if you provide no way to authenticate a user, it will always say that |
01:24.42 | orlock | Hmm.. |
01:24.42 | russellb | it's just telling you that the call was accepted without any authentication |
01:24.50 | orlock | I've got some grandstreams here, which can dial each other |
01:25.01 | orlock | but i'm having problems dialling out using a sip account |
01:29.49 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
01:30.18 | theorem_ | orlock - firewall problems ? |
01:32.57 | *** join/#asterisk |ryan| (n=foo@c-67-174-157-188.hsd1.ca.comcast.net) |
01:33.22 | |ryan| | does anyone know of a cheap (less then $100) 3.3V PCI FXO card? |
01:33.40 | mogorman | x100p |
01:34.00 | CrashHD | how come my inbound iax calls are showing unauthenticated |
01:34.41 | CrashHD | I have host=10.0.4.243 |
01:34.47 | CrashHD | and secret=secretkeyhere |
01:34.55 | CrashHD | still shows as unauthenticated |
01:35.15 | |ryan| | mogorman: no, the X100P is 5 volt |
01:35.36 | |ryan| | will not work in a 3.3 volt PCI socket. |
01:35.50 | |ryan| | something USB would also do |
01:35.58 | |ryan| | I have a soekris net4801 |
01:36.13 | *** join/#asterisk Damin (n=damin@nucleus.nacs.net) |
01:41.03 | CrashHD | $20 bucks via paypal to the first person to help me fix this damn iax problem |
01:41.14 | tsume | heh |
01:41.17 | CrashHD | heh |
01:41.21 | CrashHD | desperation |
01:41.35 | CrashHD | so it'll take you 5 minutes to solve |
01:41.40 | CrashHD | I'm just a big dummy |
01:41.58 | tsume | Chotaire: what is the problem? |
01:42.06 | tsume | oops |
01:42.08 | CrashHD | heh |
01:42.10 | tsume | CrashHD: what is the problem? |
01:42.24 | CrashHD | iax calls are not hitting the proper contexts' |
01:42.32 | CrashHD | and are showing unauthenticated |
01:42.48 | tsume | I have rules, I must see code ;) |
01:42.54 | tsume | I don't have ESP |
01:43.07 | CrashHD | sure, let me get it all up on pastebin |
01:43.07 | CrashHD | brb |
01:43.47 | CrashHD | just to give you some background |
01:43.56 | CrashHD | I have inbound traffic from a voip provider |
01:43.58 | CrashHD | via iax |
01:44.12 | *** join/#asterisk iq (n=iq@71-214-5-20.omah.qwest.net) |
01:44.14 | CrashHD | which goes to a * gateway machine (voip_gw_1) |
01:44.25 | CrashHD | some calls get passed to another * machine (voip_gw_2) |
01:44.30 | CrashHD | some get handled or passed else where |
01:44.48 | CrashHD | (voip_gw_2) also needs to dial out through (voip_gw_1) |
01:48.29 | CrashHD | check query |
01:48.54 | CrashHD | a call from gw_1 to gw_2 shows as authenticated |
01:49.21 | CrashHD | calls from [ipcomms] to gw_1 show unauth as well as from gw_2 to gw_1 (show unauth) |
01:49.49 | CrashHD | consequently inbound calls on iax fall into [inbound] because it is the last context= defined in the iax.conf |
01:50.34 | asterboy | finallyl got the tftp-hpa working. |
01:50.52 | asterboy | found an entry about user name in /var/log/sys.log |
01:51.13 | asterboy | looks like you need to invoke the command with a -u username. |
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01:51.23 | asterboy | also, added entries into /etc/services. |
01:51.31 | asterboy | would be nice if that was in the README. |
01:51.46 | asterboy | typical linux |
01:52.02 | Nugget | Linux is poo. |
01:54.17 | *** part/#asterisk tehdely (n=delysiid@home.teambarry.org) |
01:55.11 | asterboy | poo? as in corn and poo? |
01:55.48 | Frogzoo | Nugget: :X |
01:55.51 | asterboy | as in - dries on bum hair, makes itchy, make for smelly finger? |
01:56.13 | asterboy | as in - bacon strip |
01:56.25 | Frogzoo | atftpd works for me - just install the package & go |
01:56.50 | asterboy | lfs can be a litle picky |
01:57.08 | asterboy | usually missing some library the standard distros dump everywhere. |
01:57.26 | asterboy | I like my box tight. :-> |
01:57.52 | asterboy | as in - clamped around my finger |
01:58.47 | tsume | Windows is poo |
01:58.51 | tsume | linux is the savior |
01:59.02 | tsume | but if you want a better system than both.. use MacOSX :) |
01:59.21 | asterboy | They all make me money one way or the other. |
01:59.43 | asterboy | I like em all, but for Servers I love Linux. |
01:59.51 | adelas | has anyone heard of ACN here? |
01:59.56 | asterboy | ~acn |
02:00.13 | adelas | do you think the companies any good? |
02:00.16 | asterboy | not even jbot knows...what does it stand for? |
02:00.27 | adelas | ACN = phone company |
02:00.43 | adelas | a digital phone, analog phone, and dsl provider company |
02:00.53 | asterboy | Do they have technical support that directs you to India? |
02:02.09 | asterboy | Funny how the automated atendant never has an east indian accent. |
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02:26.09 | orlock | Hmm, i have inbound calls working, but outbound i;m getting a 404 |
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02:32.53 | Kumbang | guys, how can set flash hook time, my old PBX seems to reject it |
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02:40.34 | [hC] | I hate to ask this, but does anyone have, or us, the Zyxel P2000W wifi phone? |
02:40.41 | [hC] | us=use |
02:41.59 | Djeli | Not me, sorry |
02:44.56 | X-Rob | [hC], apparently they suck really really badly. |
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03:04.34 | Vazir | hi men! Anyone using OH323 in production? |
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03:19.57 | CrashHD | -- Attempting native bridge of IAX2/ipcomms149-12 and IAX2/voip_gw_2-13 |
03:19.57 | CrashHD | <PROTECTED> |
03:19.57 | CrashHD | <PROTECTED> |
03:20.00 | CrashHD | what does that mean? |
03:20.04 | CrashHD | does it everytime |
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03:22.40 | rpm | sounds like you are using incompatible codecs. |
03:22.47 | rpm | ls |
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03:30.26 | babyhuey | <PROTECTED> |
03:30.26 | babyhuey | <PROTECTED> |
03:30.26 | babyhuey | <PROTECTED> |
03:30.26 | babyhuey | <PROTECTED> |
03:30.39 | babyhuey | i get that every time i call in, then i cant do anything with the system, and i have to reboot |
03:30.44 | babyhuey | it scrolls that over and over |
03:30.52 | I-MOD | ~pb |
03:30.57 | jbot | pb is, like, a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
03:31.16 | babyhuey | sorry for that |
03:31.35 | |ryan| | does anyone know of a cheap (less then $100) 3.3V PCI FXO card? I've got a X100P, but it needs a 5v PCI socket :( |
03:35.18 | babyhuey | http://pastebin.com/611808 |
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03:43.46 | ambriento | babyhuey, what about some extensions.conf lines? |
03:44.28 | babyhuey | well, im using amp to set it up, dont know if that helps |
03:44.40 | babyhuey | still want me to post it? |
03:45.42 | *** join/#asterisk whatisthat (n=va2003ch@203.119.9.9) |
03:45.48 | whatisthat | Hi |
03:46.04 | whatisthat | can I use database to store sip user in Asterisk |
03:47.00 | ambriento | sure babyhuey, why not? :) |
03:47.25 | whatisthat | which database can be used to store sip user |
03:47.32 | ambriento | whatisthat, how is that? |
03:47.47 | I-MOD | whatisthat: yes, sql, voip-info.org |
03:47.50 | ambriento | :) |
03:48.08 | whatisthat | I want to use database to store user and password instead of using sip.conf file |
03:48.24 | babyhuey | ambriento: http://pastebin.com/611818 |
03:48.59 | whatisthat | is there any guide for this |
03:50.17 | whatisthat | I-MOD: can you tell me where to find document guide for this |
03:50.50 | I-MOD | http://voip-info.org |
03:51.04 | I-MOD | its on there somewher |
03:51.05 | I-MOD | e |
03:52.27 | *** join/#asterisk bmg505 (n=leon@dsl-146-32-34.telkomadsl.co.za) |
03:55.11 | ambriento | thats a lot of stuff babyhuey |
03:55.15 | babyhuey | yea, i know |
03:55.19 | babyhuey | its all generated by amp |
03:55.54 | I-MOD | actually, i think amp support issues are directed toward #amportal |
03:56.05 | babyhuey | i dont think its an amp problem |
03:56.15 | *** join/#asterisk BugKham (n=lamer@ppp-58.10.66.137.revip2.asianet.co.th) |
03:57.22 | ambriento | which verbose level are you using at CLI's? |
03:58.23 | babyhuey | i usually just hit it 4 times |
03:59.17 | X-Rob | whatisthat, you're looking for 'realtime' |
03:59.31 | X-Rob | that lets you store stuff in databases, rather than config files. See voip-info.org |
03:59.47 | X-Rob | I-MOD, #freepbx now |
04:03.22 | *** join/#asterisk tuxinator_linux (n=tuxinato@mobile-166-173-127-126.mycingular.net) |
04:10.06 | *** join/#asterisk tuxinator_linuxM (n=tuxinato@mobile-166-173-111-001.mycingular.net) |
04:17.41 | whatisthat | can you tell me some usb phone can be used with asterisk |
04:20.38 | exten123 | can we give FXS port a channel name like FXO port? |
04:20.59 | exten123 | I mean by digium hardware |
04:27.15 | Octothorpe[away] | CrashHD: Did you get your problem fixed? |
04:29.04 | *** join/#asterisk giggles (n=chatzill@ool-18bb0d86.dyn.optonline.net) |
04:29.09 | *** join/#asterisk FLeiXiuS (n=fleixius@c-68-50-206-161.hsd1.md.comcast.net) |
04:30.50 | *** join/#asterisk marv (n=ilovekim@12-219-145-181.client.mchsi.com) |
04:33.05 | *** join/#asterisk TheCops (i=nobody@got.securebinary.com) |
04:33.23 | TheCops | [TK]D-Fender, hey :) |
04:34.28 | *** join/#asterisk maxx4life (n=max4life@71-35-210-12.slkc.qwest.net) |
04:38.03 | CrashHD | how can you check t1/pri clocking? |
04:38.13 | CrashHD | Octothorpe[away] sort of |
04:38.33 | CrashHD | Octothorpe[away] I worked around the issue. I still have some questions about how * deals with iax authentication |
04:38.59 | *** join/#asterisk nayyares (n=Nayyar@mbl-65-151-218.dsl.net.pk) |
04:40.17 | nayyares | where can i find the a2billing installation howto? |
04:40.17 | *** join/#asterisk BugKham (n=lamer@202.8.86.163) |
04:41.13 | BugKham | Hi, I've got a quick newbie question. How do I know that the ISDN-PRI is connected properly? |
04:41.25 | I-MOD | zttool |
04:41.30 | BugKham | assuming that all configurations are right |
04:41.52 | BugKham | ztcfg? |
04:42.04 | I-MOD | you can also use pri show span 1 on the asterisk console |
04:42.28 | SwK_ | bugkham pri show span ${SPAN_NUMBER} |
04:42.39 | SwK_ | so like "pri show span 1" |
04:42.41 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
04:42.43 | BugKham | I-MOD: I normally 'cat /proc/zaptel/*' |
04:42.54 | BugKham | SwK_: k, thanks |
04:43.09 | SwK_ | zttool will only show you if the T1 has carrier pri show span will show you if the D is up |
04:43.19 | SwK_ | omg is blitzrage |
04:44.03 | blitzrage | omg! :) |
04:44.11 | brookshire | what a nub |
04:44.19 | blitzrage | brookshire: OH NO YOU DIDN'T |
04:44.29 | blitzrage | brookshire: I need you to fix something :) |
04:44.45 | blitzrage | nevermind -- you already fixed it |
04:44.58 | brookshire | blitz: lame lame lame |
04:45.31 | blitzrage | Llama's Ate My Eggo's |
04:45.31 | ambriento | babyhuey, I'm sorry but I got stuck in some other stuff over here |
04:45.57 | babyhuey | i got it working |
04:45.59 | ambriento | besides, its almost 2am and I have to got up really soon |
04:46.09 | babyhuey | thanks for the help |
04:46.09 | babyhuey | 2am? |
04:46.09 | babyhuey | where are you? |
04:46.09 | babyhuey | its only 11.45 here |
04:46.26 | ambriento | I'm in brazil |
04:46.31 | babyhuey | cool |
04:46.33 | ambriento | :) |
04:46.43 | babyhuey | alright, well, ill talk to you later then |
04:46.47 | ambriento | And no need to thanks me, I didn't do anything :) |
04:46.57 | ambriento | be my guest babyhuey. :) |
04:48.14 | *** join/#asterisk shido6 (n=shido6@d221-68-217.commercial.cgocable.net) |
04:58.22 | BugKham | my status says, Status: Provisioned, In Alarm, Down, Active |
04:58.44 | Math` | heh I got that too, but no D-Channel is config'd on the other side so I understand |
04:59.03 | BugKham | so, the link is down, am I correct? |
04:59.29 | Math` | it is |
04:59.54 | BugKham | Math`: hehe |
05:01.23 | BugKham | Math`: do u have to put crc parm in your zaptel.conf? |
05:02.00 | Math` | I just closed the pri's other end and its still displaying that status |
05:02.01 | Math` | I didnt |
05:02.30 | BugKham | Math`: this is the first time so I don't know if the crc parm is required |
05:07.03 | asterboy | Interesting that you can bypass password for the reset function of a polycom phone. |
05:08.02 | Math` | by unplugging the power and putting it back? |
05:08.30 | asterboy | lol |
05:08.48 | Math` | it sure bypasses the password |
05:08.54 | asterboy | why do they put that function in a password protected area? |
05:08.59 | Math` | I have no idea |
05:09.16 | Math` | well when I open a polycom which isnt provisioned yet, its not passwd-protected |
05:09.24 | *** join/#asterisk wundaboy (n=asdf@c-24-21-100-201.hsd1.or.comcast.net) |
05:09.27 | Math` | by in my config it is (config which I took from the Wiki(tm)) |
05:09.30 | asterboy | I just hold done the [Vol-] [Vol+] [Mic Mute] [Messages] keys for >3secs |
05:09.41 | Math` | ah ya thats documented too |
05:10.02 | asterboy | page 17 of user guide |
05:10.07 | Math` | yup |
05:10.19 | asterboy | when debugging it sure makes it faster to reboot |
05:10.43 | Math` | indeed |
05:13.30 | Math` | you can also send the phone a certain kind of sip message to make it reboot |
05:16.09 | *** join/#asterisk shido6 (n=shido6@d221-68-217.commercial.cgocable.net) |
05:18.52 | Math` | justinu: I finally tested the async rtp branch, I need to Background(silence/1|n) then do a Ringing(); for it to ring on the other end |
05:27.55 | *** join/#asterisk io_error (n=error@mdsnwikwbas08-pool7-a82.mdsnwikw.tds.net) |
05:40.30 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
05:42.22 | *** join/#asterisk julien[re] (n=julien[r@AStDenis-103-1-11-206.w80-8.abo.wanadoo.fr) |
05:42.28 | julien[re] | hi there |
05:43.08 | asterboy | When setting up a dial plan where I just want a line pickup to immediately goto another SIP extension, I have this: exten => 1,2,Dial(SIP/Upstairs)...must be missing something cause it doesn't work. Any suggestions? |
05:43.33 | orlock | Hmm |
05:43.40 | julien[re] | exten => s,1,Dial() |
05:43.44 | orlock | i can dial in to my asterisk system, but trying to call out i get a 404 |
05:43.52 | orlock | i'm setting it up using SAIL |
05:44.21 | asterboy | julien, how will it know where to Dial with just dial()? |
05:44.36 | julien[re] | i mean Dial(SIP/Upstairs) ;) |
05:44.57 | asterboy | oh ok, the s, |
05:45.00 | julien[re] | that's what I have on one of my DID and it works |
05:45.04 | mishehu | anybody here using iaxmodem to send faxes? I'm having a problem where iaxmodem doesn't seem to want to connect to the proper context (when dialing, it send 1234567@ ) |
05:45.08 | julien[re] | the first item is for the DID |
05:45.24 | asterboy | trying, thanx |
05:45.25 | julien[re] | (as far as i've understood) |
05:46.05 | julien[re] | btw, i've been wondering how to have such a SIP URI working: http://www.voip-info.org/tiki-index.php?page=Phone+Numbers |
05:46.10 | julien[re] | any1 could help? |
05:46.29 | julien[re] | i mean, i'd like to have this URI for my * |
05:47.47 | justinu | math: i used Playtones(ringing) |
05:48.41 | Math` | oh |
05:48.52 | Math` | now I Queue() so I can listen to moh when ringing :) |
05:50.21 | justinu | dial option m does that too, i think |
05:51.10 | asterboy | julien[re], I think that the SIP URI is a function of either a soft phone or a hardware phone where you can cut&paste the URI...some phones like my Polycom allow you to use the alpha keys to punch it in or you could do some fancy asterisk programming with IVR and do it with a regular phone via FXS adaption. |
05:51.43 | Math` | justinu: it does! is that a new option? |
05:52.02 | justinu | dunno |
05:52.58 | asterboy | Anyone here setup Polycom phones so you can dial another Polycom phone by hitting a line key. I want to just dial from one extension to another via two Polycom phones. |
05:53.03 | asterboy | ? |
05:53.18 | Qwell | asterboy: Do they have speeddial support? |
05:53.26 | asterboy | yes |
05:53.29 | asterboy | IP 600 |
05:53.29 | Qwell | then sure |
05:53.46 | asterboy | ah, ok..I was thinking I had to register the lines. |
05:53.48 | Qwell | or are the speeddials not on the line keys? |
05:53.56 | asterboy | Then setup a dial plan to call one or the other. |
05:54.17 | asterboy | I'm sure I can setup the speed dials on any of the line keys. |
05:54.38 | asterboy | So do the SIP URI thing? |
05:55.29 | Math` | you can override any keys in the configs |
05:55.44 | Math` | even arrows.... tho this is not recommanded (for obvious reasons) |
05:56.04 | justinu | i had a frustrating experience, my polycoms wouldn't use their custom ring tone |
05:56.05 | asterboy | lol, even the arrows...that is neat. |
05:56.15 | justinu | even tho it downloaded the ring tone, and I could set it manually |
05:56.23 | justinu | it would not pick the setting up from the config file |
05:56.42 | *** join/#asterisk alephcom (n=alephcom@host75.net14.mcsnet.ca) |
05:57.04 | justinu | yep, no matter what I set in the xml config, it would chose 2 |
05:57.41 | asterboy | justinu, I just had some of that frustration. |
05:57.54 | justinu | no kidding... |
05:57.55 | asterboy | realized it was a rogue "/>" |
05:57.57 | asterboy | end tag |
05:58.02 | justinu | yeah - i thought that too |
05:58.08 | asterboy | in the wrong spot due to my cutting and pasting. |
05:58.09 | justinu | but i'm pretty sure my xml is ok |
05:58.37 | asterboy | The line I setup would just not show up. |
05:58.50 | asterboy | totally ignored the changes. |
05:58.57 | justinu | i'll crank of the debug log level, see if the stupid thing complains about invalid xml |
05:59.10 | justinu | s/of/up/ |
05:59.33 | asterboy | cool, didn't jbot did that. |
06:00.10 | asterboy | s/didn't/didn't know/ |
06:00.39 | asterboy | even pickup up the "'" |
06:00.39 | *** join/#asterisk bjohnson_ (n=bjohnson@i216-58-43-154.cybersurf.com) |
06:00.39 | *** join/#asterisk stefan (i=stefan@home.stefan.id.au) |
06:00.39 | *** join/#asterisk Beirdo (n=gjhurlbu@unaffiliated/beirdo) |
06:01.16 | *** join/#asterisk Snake-Eyes (n=blog@202.168.41.172) |
06:02.33 | julien[re] | any1 could try a SIP URI for me? |
06:03.07 | asterboy | once I learn how to program my speed dials...sure |
06:03.12 | julien[re] | ;) |
06:06.11 | *** join/#asterisk mrtwister|mobile (n=andrius@cable-10-68.cgates.lt) |
06:07.20 | giggles | anybody know the format of the xml config file on a mediatrix 2102? |
06:07.34 | Qwell | giggles: I'm just guessing here, but... |
06:07.39 | Qwell | could it maybe be in...xml format? |
06:08.09 | asterboy | lol |
06:08.22 | giggles | har har har |
06:08.38 | asterboy | speaking of...just found Polycom's speed dial instructions on page 34 of Admin Guide. |
06:08.49 | mattodude | Qwell: no, it's in encoded binary XML format! |
06:08.55 | Qwell | eww |
06:09.18 | asterboy | <Ethernet address>-directory.xml |
06:09.28 | asterboy | hate having to reboot all the time. |
06:09.33 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
06:10.53 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
06:11.20 | giggles | can it be that nobody knows how to program this device? |
06:11.34 | giggles | or is it that no one wants to tell? |
06:11.48 | asterboy | not here anyway...did you try clusty.org? |
06:15.31 | giggles | ok something new, a lot of hits |
06:15.45 | giggles | not sure if there is anything useful, but thanks |
06:16.36 | *** join/#asterisk Foxamemnon (n=elrond@71.195.202.240) |
06:16.44 | x86 | anyone have sip extensions defined in a mysql database working correctly? |
06:17.16 | x86 | i'm using asterisk 1.2.4 on my Gentoo system (which takes the source and applies a few patches) |
06:18.19 | x86 | i have mysql in my USE flags, and I have CDR with mysql working (using cdr_addons_mysql) |
06:18.24 | x86 | CDR works great |
06:20.59 | Foxamemnon | Hello. I'm looking for some help getting Asterisk to record video with voicemail. |
06:21.24 | *** join/#asterisk littleball (n=littleba@26.203-123-30.leased.qala.com.sg) |
06:21.31 | asterboy | X86, I'd like to do this also...can't help much no since its new to me. |
06:21.43 | asterboy | s/no/now/ |
06:21.50 | Foxamemnon | When I make the connection, I get errors from Asterisk: "No translator path from unknown to unknown" and "Unable to translate to format h263, source format unknown" |
06:22.08 | asterboy | for my CDR, I just wrote a little shell script. |
06:22.41 | Foxamemnon | But I don't know what's wrong. My understanding is that Asterisk will just save the h261/h263 stream to disk. That's fine by me and Asterisk wouldn't really need to know the internals of the format. |
06:23.47 | asterboy | I had that error when trying to use a SIP connection without a license for G729 codec from Digium. |
06:24.09 | *** join/#asterisk subdolus (n=subby@dsl-202-72-158-93.wa.westnet.com.au) |
06:24.33 | Foxamemnon | Hmmm... in this case, the connection is GSM audio and h263 video. Shouldn't need any licenses for that, right? |
06:24.52 | asterboy | well I'm thinking that your missing a codec or something. |
06:25.03 | asterboy | liscense or no. |
06:25.22 | asterboy | can you test each codec individually? |
06:25.53 | asterboy | cause it looks like it want to convert from one to the other but it's missing the codec or the config to do it. |
06:26.06 | asterboy | s/want/wants/ |
06:26.41 | littleball | hello, who can give me a suggestion about setting up a minimum cost of asterisk system? basically, it collects to analog phone in one end and the other end it connects my main asterisk system through internet. |
06:27.01 | littleball | s/collect/connect/ |
06:27.16 | Foxamemnon | That's what I thought too. But I'm not sure how I can test the doces properly. I checked the source distribution (I'm using the Debian packages) and h263 is part of the core of Asterisk. |
06:27.58 | Foxamemnon | doces->codecs |
06:28.41 | *** join/#asterisk Assid (n=assid@59.183.13.48) |
06:28.47 | Assid | heya |
06:28.53 | Assid | Mar 20 11:55:44 mercury kernel: zaptel: disagrees about version of symbol struct_module |
06:29.04 | Assid | i cant manage to get the zaptel working |
06:29.22 | asterboy | littleball, $25 FXO adaptor from ebay, $10 PII 233MHz, 96Mb Ram machine, Your time ...priceless. |
06:29.38 | Assid | without struct i get those RTC errors |
06:30.40 | asterboy | yuk, I hate compile errors...either the hardware or the software...and software is tought to tell...I always have a spare box with a fresh install handy so I can eliminate the possibilities...otherwise goto clusty.org |
06:30.46 | Assid | anyone know how do i fix it? |
06:30.55 | asterboy | s/tought/tough/ |
06:32.54 | asterboy | Fox, I'm not up to par on those codecs, so can't help much, but I'd start by creating a test case...h263<->h263. |
06:33.32 | Assid | asterboy: i get that error when i boot up and zaptel doesnt wanna load |
06:33.42 | asterboy | ah |
06:33.57 | asterboy | Did you try different pci slots? |
06:34.05 | Assid | pci slots? |
06:34.24 | trixter | did you compile zaptel with the same kernel headers that were used to build your running kernel? |
06:34.27 | Assid | i dont have any zaptel device |
06:34.28 | asterboy | zaptel is pci hardware no? |
06:34.42 | Assid | nah.. using ztdummy |
06:34.44 | Assid | trixter: yes |
06:34.45 | asterboy | ah |
06:34.51 | Assid | lemme recompile again |
06:34.54 | trixter | if you did then that error shouldnt be present |
06:35.06 | trixter | that error indicates that there is a version mismatch :/ |
06:35.24 | Assid | without that.. i normally get some RTC error of 1024Hz |
06:37.17 | asterboy | If anyone wants a simple CDR script to save your eyes: |
06:37.18 | asterboy | cat /var/log/asterisk/cdr-csv/Master.csv |grep incoming |cut -d"," -f2,10,12 |sed -e "s/\"//g" |grep |
06:37.21 | asterboy | <PROTECTED> |
06:37.45 | io_error | asterboy: ah, that script hurt my eyes! |
06:37.49 | asterboy | been using that until I get SQL online |
06:37.52 | asterboy | lol |
06:38.02 | asterboy | its only temporary |
06:38.07 | io_error | :) |
06:38.47 | *** join/#asterisk MikeJ__ (n=vircuser@d14-69-8-30.try.wideopenwest.com) |
06:40.04 | Assid | as soon as i load ztdummy : i get rtc: lost some interrupts at 1024Hz. |
06:41.27 | Assid | trixter: any clue whats to be done? |
06:42.37 | asterboy | rtc = real time clock, all I can add. |
06:43.17 | Assid | i thin i had a similar issue with another box before |
06:43.24 | Assid | but it was related to USB |
06:43.30 | asterboy | why it's having an interrupt issue at some specific cycle...dunno. |
06:45.13 | Assid | trixter? |
06:45.37 | Foxamemnon | Well, I tried narrowing it down. The program I'm using supports only h261 for video. So I changed Asterisk's sip.conf to allow only gsm and h261. |
06:46.07 | Foxamemnon | I get the same error messages when trying to record a voicemail. It seems Asterisk keeps wanting the video stream to be h263 even though it's not. |
06:48.08 | asterboy | try a full reboot, (unecessary usually, but it will make sure your config sticks). |
06:48.43 | asterboy | I did a lot of changes that did stick often cause I was not changing the right file or reloading properly. |
06:49.10 | asterboy | s/t did/didn't/ |
06:49.23 | asterboy | lol |
06:49.26 | io_error | he |
06:49.30 | io_error | s/he/heh |
06:49.35 | io_error | s/he/heh/g |
06:49.41 | io_error | LOL |
06:49.46 | asterboy | :P |
06:50.21 | asterboy | it's getting late hear...me spelling is drifting. |
06:50.30 | asterboy | s/hear/here/ |
06:50.46 | io_error | io_error meant: the final / is generally optional, you stupid bot |
06:51.02 | Assid | i have a ohci usb |
06:51.10 | Assid | i even have the drivers in the kernel |
06:52.07 | Assid | i got USB Controller: Broadcom CSB6 OHCI USB Controller |
06:52.25 | asterboy | s/got/have/ |
06:52.58 | Foxamemnon | Okay, I restarted the machine. But it's still doing the same thing. |
06:53.39 | asterboy | Assid, try taking the card out for now...see if there is any change. |
06:53.47 | Assid | err.. onboard |
06:53.56 | asterboy | disable in bios |
06:54.26 | Assid | lemme quickly disable the ohci usb |
06:54.55 | asterboy | Foxamemnon, where did you make the change to tell * to use your codec. |
06:55.10 | Assid | from the kernel |
06:55.15 | Assid | making it into module |
06:55.28 | io_error | Assid: what do you have set in the kernel config for CONFIG_HZ ? |
06:55.29 | asterboy | Assid, may not help but it can eliminate the possibility |
06:55.38 | Foxamemnon | asterboy, In sip.conf. I set disallow=all, then allow=gsm and allow=h261 |
06:56.00 | asterboy | maybe do a pastebin |
06:56.17 | Foxamemnon | It does seem to have some effect. If I set allow=h263 instead of h261, then I get no video support, which is correct. |
06:56.32 | asterboy | ok that helps |
06:56.45 | Assid | # CONFIG_HZ_100 is not set |
06:56.46 | Assid | CONFIG_HZ_250=y |
06:56.46 | Assid | # CONFIG_HZ_1000 is not set |
06:56.46 | Assid | CONFIG_HZ=250 |
06:57.17 | asterboy | Assid, that may help to play with those settings. |
06:57.23 | io_error | looks pretty default to me. |
06:57.25 | asterboy | guessing though |
06:57.29 | io_error | Try it at 1000 |
06:58.44 | asterboy | Foxamemnon, what does "show codecs" at CLI give ya? |
06:58.55 | io_error | Assid: do you just see the error message once, or does it repeat? |
06:58.57 | Assid | err.. i had a similar issue on another box.. checking those |
06:59.01 | Assid | io_error: repeat |
06:59.11 | Assid | thats the same CONFIG_HZ there |
06:59.12 | io_error | Assid: what CPU and mobo? |
06:59.32 | Assid | p4 2.4 dellpowereddge 600sc |
06:59.41 | Assid | err.. lemme get the mobo |
06:59.59 | io_error | yeah, change the kernel to 1000 HZ and try again |
07:00.00 | Assid | no clue which mobo.. but its an intel from what i can see |
07:00.16 | asterboy | aha! see that's what you get for going to the darkside! Dell/Intel |
07:00.38 | Assid | err.. |
07:00.47 | Assid | its a dedicated server.. |
07:00.58 | io_error | oh... hmmm. |
07:01.08 | io_error | Assid: dedicated, or VPS? |
07:01.11 | *** join/#asterisk edwar64896 (n=edwar648@72.83.233.220.exetel.com.au) |
07:01.23 | Assid | dedicated.... |
07:01.38 | Foxamemnon | Okay, here's the "show codecs" output: http://pastebin.ca/46317 |
07:01.52 | io_error | Assid: what distro? |
07:02.03 | Assid | debian |
07:02.07 | Assid | err.. this is not good : Unable to find swap-space signature |
07:02.38 | edwar64896 | join #asterisk-bugs |
07:02.41 | edwar64896 | oops |
07:02.42 | edwar64896 | soz |
07:02.53 | io_error | Assid: I'm starting to suspect hardware problems |
07:03.31 | io_error | yow...building a kernel for debian is a complete pain :( |
07:03.51 | *** join/#asterisk chapeaurouge (n=chap@vilhost1.vision.lu) |
07:07.50 | x86 | ok, i think i got sip extensions from mysql working properly... |
07:08.02 | asterboy | Foxamemnon, that is exactly what I have. Looks in order. |
07:08.29 | x86 | is there a script i can use to convert all of my existing extensions in the sip.conf flat-file into the mysql database? |
07:09.41 | niZon | anyone know how to enable the http server on a cisco ip phone (sccp firmware)? |
07:10.24 | asterboy | x86, shouldn' be too hard to wip up a script, imagine someone has done it though and there may be something floating in interspace. |
07:10.57 | Qwell | niZon: I don't know if you can disable it |
07:11.24 | asterboy | Foxamemnon, wish I could get this going with ya, (be doing some of the same soon), anyway, gotta go to sleep ...have a good one. |
07:11.37 | Foxamemnon | asterboy, Thanks for your help tonight. Guess I'll work on it some more tomorrow. Goodnight. |
07:12.10 | asterboy | ya, I'll be on...like to do more on it. |
07:12.16 | asterboy | night all |
07:13.17 | niZon | Qwell: well my 7970 doesn't accept connections on port 80 |
07:16.55 | BugKham | what does this imply? -> "Detected alarm on channel 31: Yellow Alarm" |
07:18.29 | BugKham | chan_zap.c:1435 zt_disable_ec: Unable to disable echo cancellation on channel 19 |
07:20.15 | Assid | Mar 20 12:48:22 mercury kernel: rtc: lost some interrupts at 1024Hz. |
07:20.15 | Assid | Mar 20 12:48:22 mercury last message repeated 218 times |
07:20.20 | Assid | still cant get rid of it |
07:20.28 | Assid | had the usb disabled from bios too |
07:21.07 | io_error | Assid: hm, I think you should have the USB enabled, ztdummy will want it I think |
07:21.54 | Assid | err.. you just said to disable it |
07:22.09 | io_error | er, I didn't say anything about disabling USB |
07:22.15 | Assid | you didnt? |
07:22.22 | io_error | no, that was asterboy |
07:22.27 | io_error | who has left |
07:22.33 | Assid | dammit |
07:22.43 | Assid | the tech at the DC is gonna go nuts |
07:22.56 | Assid | ive made him go to my box 4-5 times in the past 2 hrs |
07:26.39 | *** join/#asterisk kmilitzer (n=km@office-gw.westend.com) |
07:27.01 | Assid | thig is .. last time this happened.. to another box |
07:27.04 | kmilitzer | Morning everyone ... |
07:27.07 | Assid | i dont know what i did to fix it |
07:28.02 | Assid | that time it was because of : USB Controller: Intel Corporation 82801FB/FBM/FR/FW/FRW (ICH6 Family) USB UHCI #1 |
07:28.13 | Assid | the 82801 usb uhci |
07:28.34 | io_error | Assid: does this box have IDE disks? |
07:29.24 | Assid | which one ? the one that doesnt work? yes.. IDE .. but it works through serverworks |
07:29.45 | io_error | Assid: try this: hdparm -u 0 /dev/hda |
07:30.53 | io_error | er, excuse me |
07:30.57 | io_error | Assid: try this: hdparm -u 1 /dev/hda |
07:31.20 | Assid | /dev/hda: |
07:31.21 | Assid | <PROTECTED> |
07:31.21 | Assid | <PROTECTED> |
07:31.26 | io_error | now try ztdummy :) |
07:31.33 | Assid | you kidding me? |
07:31.37 | io_error | no |
07:31.55 | Assid | nah |
07:31.57 | Assid | didnt work |
07:32.07 | io_error | hm, you're losing interrupts somewhere else then |
07:32.12 | Assid | but i made ehci and uhci into modules |
07:32.20 | io_error | A nice comment in the kernel source said that was probably the cause :) |
07:34.06 | io_error | Assid: do you have CONFIG_PREEMPT_* anything set in the kernel config? |
07:34.32 | Assid | CONFIG_PREEMPT_NONE=y |
07:34.32 | Assid | # CONFIG_PREEMPT_VOLUNTARY is not set |
07:34.54 | io_error | hm, my working system has CONFIG_PREEMPT_VOLUNTARY=y :) |
07:35.13 | io_error | and # CONFIG_PREEMPT_NONE is not set |
07:35.25 | *** join/#asterisk justnulling2 (n=justnull@ool-18bcc906.dyn.optonline.net) |
07:35.40 | Assid | this is what i have in my other workin: |
07:35.41 | Assid | # CONFIG_PREEMPT_NONE is not set |
07:35.41 | Assid | # CONFIG_PREEMPT_VOLUNTARY is not set |
07:35.41 | Assid | CONFIG_PREEMPT_BKL=y |
07:35.47 | io_error | hm, either way |
07:35.59 | io_error | but CONFIG_PREEMPT_NONE is probably a bad idea to run * |
07:36.06 | justnulling2 | i get "No application 'MeetMe' for extension" how can i fix it and i have ztdummy loaded? |
07:36.09 | Assid | whats it for? |
07:36.59 | io_error | realtime response |
07:37.17 | *** join/#asterisk g0mb0 (n=g0mb0@external.micom.mng.net) |
07:37.42 | Assid | wheres that in menuconfig? |
07:38.09 | io_error | CONFIG_PREEMPT - This option reduces the latency of the kernel when reacting to real-time or interactive events by allowing a low priority process to be preempted even if it is in kernel mode executing a system call. |
07:38.35 | io_error | Assid: Processor type and features |
07:39.50 | Assid | <PROTECTED> |
07:39.56 | Assid | ( ) Voluntary Kernel Preemption (Desktop) |
07:40.01 | Assid | ( ) Preemptible Kernel (Low-Latency Desktop) |
07:40.55 | io_error | I like this one ( ) Preemptible Kernel (Low-Latency Desktop) |
07:41.05 | io_error | But either of the two should be fine |
07:41.41 | Assid | should i still change the Timer frequency (250 HZ) ---> |
07:41.49 | io_error | yes |
07:41.55 | io_error | 1000 HZ |
07:42.24 | Assid | weird |
07:42.30 | Assid | none of the other boxes are running at that |
07:42.41 | io_error | Same hardware? |
07:42.59 | Assid | nah.. |
07:43.06 | Assid | actually |
07:43.07 | io_error | heh, well make the changes, if it doesn't work you can chagne it back :) |
07:43.18 | Assid | i got another box.. which used to run 2.6.11 and asterisk |
07:43.21 | Assid | same hardware |
07:43.39 | Assid | and the same settings i brought here |
07:44.37 | *** join/#asterisk af_ (n=af@ip-164-94.sn2.eutelia.it) |
07:45.01 | *** join/#asterisk subdolus (n=subby@dsl-202-72-158-93.wa.westnet.com.au) |
07:46.15 | Assid | hrmm |
07:46.18 | Assid | something went nuts |
07:46.28 | Assid | its redoing thr whole kernel compilation all over again |
07:47.09 | x86 | that's what it's supposed to do |
07:47.12 | x86 | you want that ;) |
07:48.28 | |ryan| | does anyone know of a cheap (less then $100) 3.3V PCI FXO card? I've got a X100P, but it needs a 5v PCI socket :( |
07:48.30 | Assid | man.. im supposed to be working . instead im sitting and figuring this crap out since 5 days |
07:48.37 | Assid | something or another always blowing up |
07:48.43 | io_error | Assid: hehehhe :) |
07:48.58 | Assid | first.. kernel doesnt wanna upgrade coz of devfs |
07:49.01 | Assid | lost 2-3 days there |
07:49.11 | Assid | then... grub blows up |
07:49.16 | x86 | 2-3 days for devfs? |
07:49.17 | Assid | not zaptel |
07:49.22 | x86 | you must be a linux newbie ;) |
07:49.36 | Assid | x86: its a redhat box converted to debian remotely |
07:49.38 | io_error | x86: he's running debian, of course he's a newbie :) |
07:50.12 | Assid | that too.. the old debian (woody).. [at the time of installation] |
07:50.32 | iDunno | ouch ;) |
07:50.47 | *** join/#asterisk linxroute (n=linxrout@58.187.123.87) |
07:50.49 | x86 | could be worse... |
07:50.54 | *** join/#asterisk medusaXX (n=medusaxx@p54A983D5.dip0.t-ipconnect.de) |
07:50.55 | x86 | could be potato ;) |
07:51.00 | Assid | then..... i finally just asked that tech to punch it into another debian box.. imaged the drive into this one.. and started it with sarge |
07:51.08 | *** join/#asterisk twisla (i=twisla@lutin.jard.in) |
07:51.25 | Assid | that just managed to get over.. and i started playing with zaptel.. when grub got messed up.. cause of the RTC.. |
07:51.41 | Assid | i was trying to grub-install .. when the system locked up cause of that rtc issue |
07:51.53 | io_error | Assid: yeah, it's sounding more and more like bad hardware |
07:52.06 | Assid | so the device map got corrupted. i said screw it and i just left it for the night.. |
07:52.09 | *** join/#asterisk UlbabraB (n=salama@host241-43.pool8172.interbusiness.it) |
07:52.19 | justnulling2 | fixed meetme but just got /usr/sbin/safe_asterisk: line 42: 9373 Segmentation fault (core dumped) ${ASTSBINDIR}/asterisk ${CLIARGS} ${ASTARGS} >&/dev/${TTY} </dev/${TTY} |
07:52.27 | Assid | today i fixed that.. fixed that swap issue.. and am HOPEFUL of the zaptel |
07:52.29 | *** join/#asterisk Zeeek (n=icechat5@pdpc/supporter/active/Zeeek) |
07:52.44 | *** join/#asterisk kilobit (n=seth@210.193.58.33) |
07:53.12 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
07:54.28 | *** join/#asterisk Gunnar (n=gunnar@bkkb-gw.bitcon.no) |
07:54.30 | Zeeek | wake up and smell the nespresso (tm) |
07:54.36 | kilobit | hi, im getting alot of "WARNING[18268]: chan_sip.c:4826 check_auth: Stale nonce received from" message... sip users can't register... any ideas? |
07:54.56 | Assid | nespresso? |
07:55.59 | Assid | gonnna tryu and install asterisk on the other box with EXACT same hardware |
07:56.01 | Assid | 02:55:43 (1.17 MB/s) - `asterisk-1.2.5.tar.gz' saved [10546846/10546846] |
07:57.40 | linxroute | sorry if may i ask , does anyone here using the g729 "open source " |
07:57.52 | Qwell | linxroute: no, it isn't legal to use it |
07:58.02 | edwar64896 | yeah I used it for a while |
07:58.09 | edwar64896 | legal smeagal ;-) |
07:58.24 | linxroute | you meant for a business or production box |
07:58.35 | linxroute | how about just for testing |
07:58.38 | edwar64896 | nope. my use was a home box. |
07:58.51 | linxroute | cos i was kind of wondering |
07:58.56 | edwar64896 | Be warned, its a real pain in the ass. |
07:58.59 | linxroute | if there any .. |
07:59.04 | edwar64896 | you have to go through hoops with intel to get the license |
07:59.12 | edwar64896 | you might as well just pay Digium for a couple of channels. |
07:59.16 | linxroute | compatibale issue |
07:59.19 | edwar64896 | it's a lot less hassle and it works straight away. |
07:59.34 | linxroute | cos i find the quality seems to be not very good |
07:59.48 | linxroute | i dont know how about the digium codec |
07:59.52 | edwar64896 | its probably the best lossy codec avaialble at the moment. |
07:59.52 | Assid | dammit |
07:59.53 | linxroute | does it better ? |
07:59.53 | Assid | Mar 20 02:59:20 box kernel: Zapata Telephony Interface Registered on major 196 |
07:59.53 | Assid | Mar 20 02:59:20 box kernel: Zaptel Version: 1.2.4 Echo Canceller: KB1 |
07:59.53 | Assid | Mar 20 02:59:20 box kernel: Registered tone zone 0 (United States / North America) |
07:59.57 | Assid | this box works fine |
08:00.08 | linxroute | thanks edward |
08:00.21 | linxroute | have u ever tried the " original" codec |
08:00.22 | Assid | EXACT same hardware |
08:00.26 | linxroute | from Digium ? |
08:00.31 | edwar64896 | [mark] no problem |
08:01.26 | linxroute | hello |
08:01.27 | Assid | io_error: that works fine on another box .. with the EXACT same hardware |
08:01.28 | io_error | Assid: Yep, hardware, figures |
08:02.37 | *** join/#asterisk atif_ (n=atif@202.92.16.30) |
08:03.03 | atif_ | hello there, can some one help me regarding OPTIONS message support in Asterisk.... |
08:03.25 | atif_ | asterisk is replying at udp src port instead of contact information provided in VIA headers |
08:03.26 | *** join/#asterisk xbmodder_lappy (i=nobody@unaffiliated/xbmodder) |
08:03.33 | xbmodder_lappy | what are the downsides to ilbc? |
08:04.31 | Assid | how the hell do i get those guys to give me another box? |
08:05.06 | io_error | Assid: tell them the motherboard in that box is fried and to replace it. |
08:05.26 | Assid | well... accoridng to these guys.. if the machine boots up.. mobo is fine |
08:05.39 | io_error | Assid: cancel your contract for cause |
08:05.40 | Assid | one sec.. |
08:05.51 | Assid | im regetting the kernel .config from that one |
08:05.54 | Assid | and recreating it |
08:06.15 | Assid | without modifications |
08:06.33 | io_error | Assid: you've got a broken APIC or something probably, in any case if it works on another identical (hw/sw) box, then it's hardware |
08:07.45 | Assid | yep |
08:07.56 | Assid | thats why i took the .config from the other box and am recompiling |
08:08.09 | Assid | if it works.. it works |
08:08.13 | Assid | if it doesnt |
08:08.14 | io_error | Assid: who is this server hosting company? |
08:08.20 | Assid | im gonna get the hard drives exchanged |
08:08.25 | Assid | ev1server |
08:08.30 | Assid | s |
08:08.31 | io_error | ah, that explains it |
08:08.40 | io_error | Never, ever do business with ev1servers |
08:08.43 | Qwell | I use ztdummy on my ev1 server |
08:09.05 | Qwell | io_error: there is nothing wrong with ev1 |
08:09.07 | Assid | yeah.. my other box works fine too |
08:09.21 | io_error | Qwell: it sure sounds like there is. |
08:10.45 | Assid | while thats compiling im gonna go get ready |
08:12.03 | io_error | The best hardware is your own hardware. :) |
08:12.27 | Assid | i know people who have issues with that |
08:12.38 | Assid | that intel one i showed you earlier.. uhci usb |
08:12.43 | Assid | that had a hell lotta problems too |
08:13.06 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
08:13.22 | *** join/#asterisk stefan (i=stefan@home.stefan.id.au) |
08:13.58 | Zeeek | use loudspeaker |
08:14.23 | tsume | io_error: and what is your own hardware called? :) |
08:14.57 | *** join/#asterisk TonyM (n=TonyM@adsl-solo-80-168-225-161.claranet.co.uk) |
08:16.46 | BugKham | does it take long to start an E1? |
08:18.02 | io_error | tsume: Which one? The one on my desk, the one under my desk, or the one next to my desk? |
08:19.33 | *** join/#asterisk mrtwister|mobile (n=andrius@cable-10-68.cgates.lt) |
08:19.59 | tsume | io_error: any? |
08:20.27 | *** join/#asterisk tuxinator_linux (n=tuxinato@netblock-68-183-112-100.dslextreme.com) |
08:20.44 | io_error | tsume: and what do you want to know about it? |
08:23.29 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
08:25.45 | tsume | io_error: you make them? |
08:28.24 | io_error | sometimes. |
08:30.47 | io_error | I need a new headset. |
08:31.11 | Zeeek | plantronics |
08:31.21 | io_error | heh :) |
08:31.32 | io_error | it's 2:30 in the morning, where am I going to get one of those at this time of day? |
08:31.51 | Zeeek | fedex |
08:31.56 | io_error | sheesh |
08:32.41 | *** join/#asterisk sulex (n=sulex@pdpc/supporter/active/sulex) |
08:36.03 | io_error | ah, everything seems to be working. |
08:36.15 | *** join/#asterisk Heartwich (n=Miranda@130.228.38.63) |
08:37.22 | *** join/#asterisk fenlander (n=fenlande@82.152.81.57) |
08:37.32 | io_error | I love it when a plan comes together. |
08:37.59 | Zeeek | always have two headsets |
08:38.11 | *** join/#asterisk ribbo (n=riaan@196.36.161.235) |
08:38.11 | Zeeek | great for debugging, too |
08:38.18 | io_error | Zeeek: heh, it's the mic, I have to have it almost IN my mouth for it to pick anything up |
08:38.28 | Zeeek | yes that's the first to go |
08:38.52 | Zeeek | but look, I had a sound problem a while back, change heaset and it was ok. Then I changed back and it was still ok! |
08:39.05 | Zeeek | so it wasn't the headset but the connector on the PC! |
08:39.19 | io_error | 393612 |
08:39.24 | io_error | er |
08:39.38 | io_error | wrong number :) |
08:39.54 | Zeeek | 303 is FWD? 612 is the old minneapolis area code |
08:40.01 | Zeeek | 393 |
08:40.05 | io_error | 393 = fwd |
08:40.08 | Zeeek | 612 was thre echo test? |
08:40.08 | io_error | at least on my * |
08:40.15 | io_error | 613 is the echo test |
08:40.15 | Zeeek | or the time |
08:40.24 | Zeeek | date/time |
08:40.34 | Zeeek | 666 is the echo test on mine |
08:40.45 | io_error | Hm, ok taht's a little better. |
08:40.57 | io_error | the mic needed to be jacked up in the mixer |
08:41.41 | tsume | 99999999999999999999999999999999 is mine |
08:41.43 | tsume | j/k :) |
08:41.46 | io_error | heh :) |
08:41.50 | Assid | also get this: end_request: I/O error, dev fd0, sector 0 |
08:41.50 | Assid | end_request: I/O error, dev fd0, sector 0 |
08:42.05 | io_error | Assid: that means ther'es no floppy disk in the floppy disk drive, it's harmless :) |
08:42.23 | io_error | unless you DO have a floppy disk in the drive, then it's a problem :) |
08:42.26 | Assid | just saying i got wayyyyyy too many errors taking place |
08:42.30 | Zeeek | heh |
08:43.07 | Assid | wish i could reboot my life |
08:43.09 | Assid | make things easier |
08:43.41 | Assid | its like anything possible that can go wrong is going wrong |
08:43.41 | Zeeek | Assid just don't re-format first |
08:45.50 | trixter | reformatting is fun |
08:45.54 | trixter | I do it several times a day |
08:46.06 | Zeeek | why keep running windows then? |
08:46.14 | trixter | never said I was |
08:46.16 | Zeeek | nyuk, nyuk |
08:46.18 | io_error | dd if=/dev/urandom of=/dev/hda |
08:46.38 | trixter | add a bs=1024 or something to make it work better |
08:46.51 | trixter | actually doing 1M chunks would prolly be better than 1k |
08:46.53 | Assid | rtc: lost some interrupts at 1024Hz. |
08:46.57 | Assid | still getting it :( |
08:49.33 | Zeeek | Montpellier |
08:50.48 | Delvar | hi, anyone got teh URL for the digium g729 .so's and the register app? |
08:51.17 | trixter | ftp://ftp.digium.com ? |
08:52.20 | trixter | ftp://ftp.digium.com/pub/asterisk/g729/ more specifically |
08:52.31 | Delvar | yeah thanks i go tit :) |
08:52.35 | Delvar | got it* |
08:52.38 | brookshire | it's under documentation on g729 page |
08:52.39 | brookshire | http://www.digium.com/en/supportcenter/documentation/viewdocs/G729 |
08:53.04 | Delvar | ther used to be links form teh digium site, but they dont seem to be there any more |
08:53.36 | Zeeek | <PROTECTED> |
08:54.31 | brookshire | http://kb.digium.com/entry/17/5/ |
08:54.33 | brookshire | there is another one |
08:55.25 | Zeeek | <PROTECTED> |
08:56.19 | Zeeek | you get there from digium products -> G.729 codec -> ralated information, Documentation -> Readme |
09:01.05 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
09:03.54 | *** join/#asterisk PoWeRKiLL (n=PoWeRKiL@80.178.85.79.adsl.012.net.il) |
09:05.18 | Heartwich | how is the best way to get a status for the extensions in asterix? |
09:05.42 | Zeeek | you mean? |
09:06.03 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
09:06.14 | Heartwich | i want to list the status for the phones connected to asterisk. is they ringing, busy aso. |
09:06.24 | Heartwich | for use in php. |
09:06.38 | *** join/#asterisk astra^^ (n=muhajir_@59.145.104.74) |
09:06.39 | Zeeek | use the manager interface |
09:07.17 | Heartwich | by making an socket connection to it? |
09:07.26 | Zeeek | http://www.google.fr/search?q=asterisk+manager+interface&start=0&ie=utf-8&oe=utf-8&client=firefox-a&rls=org.mozilla:en-US:official |
09:07.35 | Zeeek | read a few of these |
09:07.56 | io_error | Heartwich: show channels ? |
09:08.05 | Zeeek | you can just use php on apache or another http server |
09:08.23 | Zeeek | or use FOP which does exactly what you said |
09:08.28 | io_error | yeah, FOP is nice :) |
09:08.34 | Zeeek | FOP uses the manager interface |
09:08.46 | Heartwich | well, i want to the same that fop does, just without flash. |
09:09.08 | Zeeek | studying how FOP works might help |
09:11.15 | *** join/#asterisk scanna (n=scannach@81-174-16-211.f5.ngi.it) |
09:17.14 | Assid | io_error: i interchanged the harddrive |
09:17.21 | Assid | but the one that was working |
09:17.24 | Assid | is still working |
09:21.11 | Assid | damn |
09:21.14 | Assid | he just had to leave |
09:21.38 | BugKham | why do I keep getting this -- B-channel 0/xx successfully restarted on span 1"? |
09:22.08 | edwar64896 | [Bugkham] this is your PRI resetting. |
09:22.17 | edwar64896 | you can change the default timeout for PRI resets in your zapata.conf file. |
09:22.26 | edwar64896 | its perfectly normal behaviour |
09:23.28 | BugKham | edwar64896: so it has no effect on the current connections/calls? |
09:23.34 | edwar64896 | nope |
09:23.56 | edwar64896 | you'll notice that any b channel in use doesn't get touched |
09:24.51 | *** join/#asterisk y-man (n=y-man@udp115909uds.hawaiiantel.net) |
09:24.54 | BugKham | edwar64896: okay |
09:31.53 | *** join/#asterisk sundancer (n=marko@shyana.perkmandlc.org) |
09:32.59 | sundancer | Hi.. can anyone tell me what software other than bristuff also supports HFC (cologne) PCI cards |
09:33.23 | sundancer | Because in bristuff i cant assign MSN number i want to use with asterisk |
09:33.39 | *** join/#asterisk terracon (n=tc@CPE0050da608e99-CM0012254076d6.cpe.net.cable.rogers.com) |
09:33.42 | sundancer | I have two MSN numbers but HFC card takes over both of them |
09:33.45 | sundancer | And i dont want that |
09:33.51 | astra^^ | how will i make * accept calls from an ser sip server and reroute to an outbound proxy |
09:34.58 | Greek-Boy | where can one find recorded sounds? |
09:35.55 | x86 | <PROTECTED> |
09:36.03 | Greek-Boy | lol |
09:36.03 | Greek-Boy | k |
09:36.38 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
09:36.40 | Greek-Boy | is it a bad idea to use synthesis to create voice files. (text 2 speech type of thing) |
09:38.50 | BugKham | err with 1.2.5, there's no applications "Cut" or "SubString" |
09:39.20 | BugKham | what do you guys use? executing a system 'cut"? |
09:40.16 | Juggie | you just use the variable name |
09:40.37 | Aurs | (Deprecated, use ${variable:a:b} instead) |
09:40.37 | Juggie | ${EXTEN:2:5} |
09:40.47 | BugKham | ohh right |
09:40.57 | BugKham | thanks |
09:41.03 | Aurs | show application substring |
09:41.32 | BugKham | Aurs: -> Your application(s) is (are) not registered |
09:41.40 | Aurs | ok. hehe |
09:41.48 | Aurs | on 1.0.9 i get this: (Deprecated, use ${variable:a:b} instead) |
09:42.03 | Aurs | among other things |
09:42.24 | *** join/#asterisk areski (n=areski@polar.es6.egwn.net) |
09:43.02 | *** part/#asterisk edwar64896 (n=edwar648@72.83.233.220.exetel.com.au) |
09:43.27 | Zeeek | substring and x:y do not replace CUT |
09:43.49 | luckyduck | hi, is it possible to pass arguments to an agi script which i call inside of an callfile? |
09:43.55 | luckyduck | i cant find the right syntax |
09:44.09 | *** join/#asterisk exten123 (n=exten@60.49.6.190) |
09:44.40 | luckyduck | i can call the agi app using the Application keyword |
09:44.55 | luckyduck | the name of the agi script can be passed using the data keyword |
09:45.05 | *** join/#asterisk A-Tuin|work (n=A-Tuin@212.41.185.81) |
09:45.16 | astra^^ | how will i make * accept calls from an ser sip server and reroute to an outbound proxy |
09:45.27 | luckyduck | i cant find a way to pass arguments to the agi-script |
09:45.28 | luckyduck | any hints? |
09:47.03 | Aurs | luckyduck: show application agi |
09:47.12 | *** join/#asterisk oej (n=oej@bkkb-gw.bitcon.no) |
09:49.20 | *** join/#asterisk xterminus (n=cmauch@00104bc8bd59.click-network.com) |
09:51.05 | luckyduck | Aurs: thx, will try to do it that way |
09:51.21 | Aurs | <PROTECTED> |
09:51.32 | Aurs | np |
09:52.39 | *** join/#asterisk Shoragan (n=shoragan@d072.apm.etc.tu-bs.de) |
09:57.15 | *** join/#asterisk snip3r (n=sniper@195.246.199.136) |
09:57.26 | snip3r | hi all |
09:57.46 | snip3r | Zeeek: I wonder if you remember my problem |
09:58.08 | snip3r | do you have a couple of minutes to figure out what happened? |
09:58.14 | Zeeek | god no, what was it? |
09:58.19 | snip3r | :D |
09:58.48 | snip3r | the point is that if a client is behind NAT, * doesn't even TRY to send the RTP |
09:59.14 | snip3r | if it's outside NAT, everything is OK |
09:59.43 | luckyduck | Aurs: yeah, i saw it. didnt noticed the show command before. works like charm, thanks |
10:00.13 | snip3r | I've checked this with tcpdump and saw this errrr... odd behavior :) |
10:01.04 | Zeeek | I'm afraid I don't have a clue for you at all |
10:01.16 | snip3r | uh |
10:01.18 | *** join/#asterisk puzzled (n=yeahrigh@puzzled.xs4all.nl) |
10:01.28 | *** join/#asterisk pif (n=ldm@zenon.apartia.fr) |
10:01.34 | snip3r | the issue is that NATed hosts never get audio |
10:01.56 | Zeeek | I haven't seen that here |
10:02.06 | snip3r | the usual fixes (nat=yes etc) are setup |
10:02.21 | *** join/#asterisk nagl (n=nagl@rih.zid-nw.wu-wien.ac.at) |
10:02.49 | snip3r | I've posted a call setup trace at cozmo.hu/log.txt |
10:03.12 | snip3r | it's a bit old, however |
10:03.14 | snip3r | :) |
10:03.29 | snip3r | but the situation hasn't changed since |
10:04.34 | snip3r | so: if I try to connect with a client behind NAT, I get no audio |
10:05.00 | snip3r | if I call from a public IP, works like a charm |
10:05.01 | Zeeek | sometimes it's a matter of the NAT router just not working right |
10:05.11 | snip3r | that's the point |
10:05.21 | Zeeek | in which case nothing to do but replace it - what is the router? |
10:05.44 | snip3r | I've tcpdump'd a call to a NAT device and * doesn't try to send the RTP at all! |
10:06.00 | Zeeek | answer? |
10:06.02 | snip3r | the router is a cheapo X-Micro one |
10:06.11 | snip3r | doing port restricted cone |
10:06.36 | Zeeek | look no further - change the router and don't look back |
10:06.44 | snip3r | :) |
10:07.10 | snip3r | I've got a Linksys WRT54GP2 at hand |
10:07.31 | Zeeek | use that |
10:07.47 | snip3r | but the really weird part is that I managed to use * behind this router |
10:08.47 | snip3r | and now I'm confused 'cause I did very little config to * to achive full NAT penetration a week ago |
10:09.20 | snip3r | and now it doesn't want to to the same after several days of debugging |
10:09.30 | snip3r | to do the same, sry |
10:09.58 | snip3r | anyway, trying to change the router |
10:11.20 | Zeeek | k |
10:11.20 | *** join/#asterisk stefan (i=stefan@home.stefan.id.au) |
10:19.21 | RoyK | ~nickometer snip3r |
10:19.25 | RoyK | :) |
10:20.30 | dpryo | ~nickometer RoyK |
10:20.33 | *** join/#asterisk Curus (n=Curus@x1-6-00-12-17-df-1b-be.k182.webspeed.dk) |
10:20.41 | trixter | there you have it |
10:20.44 | Curus | Is it possible to ask a queue how many people are currently waiting? |
10:20.47 | trixter | an authoritative rsponse |
10:21.35 | *** join/#asterisk backblue (n=igor@82.102.1.42) |
10:22.43 | trixter | Curus: yes its possible, depending on various factors |
10:22.54 | Zeeek | shit, curl actually works! |
10:23.12 | Curus | trixter: How? |
10:23.32 | Curus | Preferably something I can look at in extensions.conf |
10:23.48 | trixter | I never said that was one of the factors |
10:26.59 | astra^^ | do we have to buy g723 codec for givin routes to g723.. |
10:27.09 | astra^^ | * supports g723 right? |
10:28.43 | RoyK | passthrough |
10:28.44 | RoyK | only |
10:28.56 | astra^^ | ok do we get to buy g723 |
10:28.59 | RoyK | or perhaps with that transcoder card that came out |
10:29.09 | RoyK | there is no software solution for g.723.1 |
10:29.11 | RoyK | for asterisk |
10:29.15 | astra^^ | ok |
10:29.18 | RoyK | but why do you need it? |
10:29.23 | RoyK | g.729a does the job |
10:29.38 | *** join/#asterisk joelsolanki (i=joelsola@202.160.161.93) |
10:29.39 | RoyK | difference is only like 10% more data |
10:29.42 | RoyK | when conting overhead |
10:29.51 | RoyK | counting..... |
10:30.45 | astra^^ | i have g729 al so |
10:30.45 | RoyK | g.729a is 8kbps + 16kbps overhead. g.723.1 is 5.3kbps + 16kbps overhead..... |
10:30.58 | RoyK | rather patch up asterisk to use larger packets than the usual 20ms |
10:31.02 | RoyK | there is a patch |
10:31.03 | RoyK | somewhere |
10:31.08 | astra^^ | ok .. . |
10:33.31 | Zeeek | can | be escaped in a function ? |
10:36.45 | Zeeek | worse yet, set(name=Hey|Now) will set it to Hey |
10:36.59 | Zeeek | Moral of the story, don'(t use | as a delimiter |
10:37.39 | *** join/#asterisk rogier (n=rogier@16-65-dsl.ipact.nl) |
10:38.17 | RoyK | Zeeek: usually set(Zeeek=quite\|stupid) will work :P |
10:38.32 | Zeeek | you've shown yourself amply on the ML |
10:38.51 | RoyK | que? |
10:39.20 | X-Rob | RoyK, I think he hates you. |
10:39.26 | *** join/#asterisk bmg505 (n=leon@dsl-165-157-13.telkomadsl.co.za) |
10:39.42 | Zeeek | bed time in .au, no? |
10:39.50 | RoyK | .no |
10:39.51 | X-Rob | 8:30pm, it's only early. |
10:39.58 | X-Rob | hint: /ctcp x-rob time |
10:40.03 | Zeeek | that's past my bedtime |
10:40.11 | snip3r | 8:30 PM? |
10:40.14 | Zeeek | yep |
10:40.17 | snip3r | :D |
10:40.26 | Zeeek | I'm only 5 |
10:40.31 | snip3r | huh |
10:40.39 | snip3r | that's weird |
10:40.48 | Zeeek | Harvard has accepted me for next year |
10:40.51 | snip3r | he's 5 and does * better than me |
10:40.56 | Zeeek | I'll be 6 then |
10:41.03 | Zeeek | muhahah |
10:41.11 | snip3r | Zeeek: what is your IQ? |
10:41.16 | Zeeek | 42 |
10:41.24 | snip3r | *10^3? |
10:41.33 | Zeeek | or ASC('@') + 2; |
10:41.37 | snip3r | uhh |
10:41.44 | Zeeek | oopos no good |
10:41.54 | snip3r | 'xcuse me? |
10:42.35 | Zeeek | asc('@') is 64 in hex, 40 |
10:42.45 | snip3r | :) |
10:42.53 | *** join/#asterisk g0mb0 (n=g0mb0@external.micom.mng.net) |
10:42.56 | Zeeek | so what about this router |
10:42.56 | snip3r | that's sg I already know :) |
10:43.05 | snip3r | changing it soon |
10:43.25 | snip3r | but first I started an * install on one of my linux boxes |
10:43.26 | g0mb0 | Can Asterisk support R2 digital compelled signaling with Sangoma card? |
10:45.52 | Mavvie | Our PABX supporter says that the Alcatel 4400 takes time from the network. Anybody here knows how that works? |
10:46.00 | Mavvie | the phone network, not the computer netowkr |
10:46.22 | snip3r | what is your question? |
10:46.37 | snip3r | the 4400 is pretty big |
10:46.47 | Assid | umm . is 726 enabled in asterisk by default |
10:47.00 | snip3r | check sip.conf |
10:47.17 | snip3r | and look for the row: 'allow=' |
10:47.52 | snip3r | if I recall correctly, pcmu, pcma and gsm are enabled by default |
10:48.05 | Assid | show translation shows it |
10:48.07 | Assid | thats why |
10:48.08 | snip3r | but if you get your hands on a CLI |
10:48.15 | snip3r | yep |
10:48.17 | snip3r | :) |
10:48.20 | Zeeek | show translation is just a chart |
10:48.27 | snip3r | show codecs? |
10:48.57 | ambriento | g0mb0, yes it can. |
10:49.03 | g0mb0 | Hi all, Can Asterisk support R2 digital compelled signaling with Sangoma card? |
10:49.04 | Zeeek | same thing, a chart |
10:49.11 | ambriento | g0mb0, yes it can. |
10:49.23 | Zeeek | the little pbx that could |
10:49.25 | Assid | i have allow all |
10:49.34 | Assid | for my sip in my context |
10:49.37 | RoyK | g0mb0: see http://soft-switch.org/ |
10:49.50 | g0mb0 | I see |
10:49.54 | Assid | but cant seem to make a call with 726-24 |
10:50.54 | snip3r | Assid: are you sure that both endpoints support .726? |
10:51.07 | snip3r | Linksys/Sipura stuff? |
10:51.26 | Assid | snip3r: yes.. linksys pap2 |
10:51.32 | snip3r | both? |
10:51.40 | Assid | err.. connecting to asterisk |
10:52.01 | snip3r | anyway, you should issue a 'sip debug peer [peername]' command |
10:52.16 | snip3r | asterisk -r on the same machine |
10:53.48 | snip3r | Assid: check this out: http://www.voip-info.org/wiki-ITU+G.726 |
10:54.09 | snip3r | ^^ this site rocks |
10:56.47 | *** join/#asterisk snip3r (n=sniper@195.246.199.136) |
10:57.32 | RoyK | Assid: iirc asterisk only supports g.726-32 |
10:58.47 | Assid | okay heres something interesting |
10:58.48 | Assid | Capabilities: us - 0x1f07ff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc|jpeg|png|h261|h263|h263p), peer - audio=0x51d (g723|ulaw|alaw|g726|g729|ilbc)/video=0x0 (nothing) |
10:58.57 | Assid | peer supports ilbc? |
10:59.27 | Assid | i dont see ilbc in the list |
11:00.07 | snip3r | in which list? |
11:00.22 | Assid | in the pap2 web interface |
11:00.38 | Assid | okay now time to test a few calls |
11:01.00 | Assid | anyone got free calling? |
11:01.48 | snip3r | yep, that's odd |
11:02.39 | Assid | how the hell does one conference in this pap2? |
11:07.36 | *** join/#asterisk fulgas (n=fulgas@209.8.233.252) |
11:07.52 | *** join/#asterisk michael-i (i=user@141.41.38.185) |
11:09.43 | BugKham | any reasons why would my ISDN PRI-E1 does not support "Playback(file,noanswer)? |
11:10.02 | BugKham | err no "would" |
11:15.48 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
11:19.06 | *** join/#asterisk Qorky (n=spam@202.173.160.26) |
11:20.19 | Curus | BugKham: Did you remember to answer the call before trying to play? |
11:23.47 | twisla | are the application names in extensions.conf CaseSensitive ? |
11:26.04 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
11:26.36 | Qorky | wheres a howto install the cvs-head please? |
11:28.01 | kaldemar | http://www.asterisk.org/download <-- take a look under "SVN repository" |
11:30.32 | Qorky | I must be behind the times. I dont have svn |
11:33.18 | Frogzoo | Qorky: subversion - lets you check in a bunch of files under one change |
11:33.35 | X-Rob | BugKham, 'Playback(file,noanswer)' isn't valid |
11:34.07 | X-Rob | especailly on an E1 |
11:36.20 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
11:36.30 | BugKham | X-Rob: so, it's valid on a T1? |
11:36.35 | X-Rob | No |
11:36.46 | X-Rob | it's valid on a SIP or IAX device with the 'r' flag on it |
11:36.55 | X-Rob | just about everything else you have to Answer first |
11:37.26 | *** join/#asterisk stuartcw (n=chatzill@softbank221025056004.bbtec.net) |
11:38.15 | BugKham | X-Rob: I was about to put an announcement to a caller without having them paid for the call |
11:38.22 | X-Rob | Can't do it |
11:44.21 | Curus | Some providers accept it |
11:44.28 | Curus | Most don't |
11:46.36 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
11:46.56 | BugKham | Curus: okay |
11:47.57 | BugKham | Curus: I'm wondering if it's the same way as putting truetones for mobile phones |
11:54.32 | *** join/#asterisk in-side (n=bsd-desk@213.58.69.127) |
11:54.34 | in-side | HI |
11:54.41 | in-side | I have a strange problem here |
11:54.51 | in-side | I would like to know if anybody has a glue for it :( |
11:55.15 | *** join/#asterisk linstar (n=achu@220.225.191.18) |
11:55.17 | in-side | what happens is my * box in a normal call send byes ok to my sip gw |
11:55.19 | linstar | hi |
11:55.44 | linstar | is there is any good documentation about connecting two asterisk servers using sip? |
11:55.46 | in-side | but when I try to use the autohangup in function dial() |
11:55.59 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
11:56.10 | in-side | it gimme a delay sending the bye or simple ignores it |
11:56.32 | astra^^ | WARNING[25769]: chan_sip.c:2530 sip_write: Asked to transmit frame type 4, while native formats is 1 (read/write = 4/4) |
11:56.37 | astra^^ | y do i get this msg |
11:56.48 | linstar | is there is any good documentation about connecting two asterisk servers using sip? |
11:57.09 | in-side | does any one has a clue why when a uac hangup I can receive every bye and not receive when asterisk does the hangup ? |
11:57.36 | in-side | linstar: you mean in reduntancy ? |
11:58.03 | *** part/#asterisk UlbabraB (n=salama@host241-43.pool8172.interbusiness.it) |
11:58.23 | linstar | in-side : I found some documentation about connecting two asterisk servers I mean peer connection |
11:58.56 | linstar | in-side: Asterisk - dual servers |
11:59.03 | linstar | configuration |
12:00.47 | astra^^ | WARNING[25769]: chan_sip.c:2530 sip_write: Asked to transmit frame type 4, while native formats is 1 (read/write = 4/4) |
12:00.55 | linstar | <PROTECTED> |
12:01.03 | astra^^ | wht mgt b the problem |
12:03.00 | xterminus | is it possible for iax phones (idefisk,etc) to make sip address calls (eg sip:extension@domain.name) through asterisk? |
12:05.57 | xterminus | when i try - asterisk doesn't seem to even try to make the call... in logs all i see is "Mar 20 04:04:46 NOTICE[22966]: chan_iax2.c:7053 socket_process: Rejected connect attempt from 10.0.0.254, who was trying to reach '9586111@mutual.bcwireless.net'" for example |
12:06.15 | *** join/#asterisk lorinc (n=ang@caracas-2449.adsl.interware.hu) |
12:07.54 | *** join/#asterisk chris_ast (n=Administ@59.93.56.163) |
12:08.22 | chris_ast | anyone there |
12:08.38 | RoyK | <PROTECTED> |
12:08.52 | chris_ast | hi RoyK |
12:09.13 | *** join/#asterisk core-ix (n=ivo@pirus.securax.be) |
12:09.26 | linstar | do anybody familiar with Asterisk - dual servers? |
12:09.59 | chris_ast | RoyK, I have a doubt in Astersik AGI and PHP, can you help me |
12:11.57 | linstar | is anybody familiar with Asterisk - dual servers? |
12:13.12 | xterminus | linstar, ask your question (i dont run dual servers - but i'm pretty good at asterisk in general) |
12:14.39 | RoyK | chris_ast: i don§t use php with agi. sorry |
12:15.12 | RoyK | linstar: what do you want to do? |
12:15.20 | RoyK | failover? load balance? |
12:15.21 | *** join/#asterisk eliel (n=eliel@200.123.183.89) |
12:15.28 | RoyK | sandwitch the servers? |
12:15.39 | xterminus | agi + perl/bash here - dont use php for much of anything |
12:17.10 | *** join/#asterisk saftsack (n=saftsack@p54A7FDDF.dip.t-dialin.net) |
12:17.18 | *** join/#asterisk saftsack (n=saftsack@p54A7FDDF.dip.t-dialin.net) |
12:19.10 | *** join/#asterisk zotz (n=zotz@24.231.32.85) |
12:19.26 | in-side | linstar: sorry I wasn't here |
12:19.41 | in-side | linstar: what is your problem ? |
12:20.57 | michael-i | anyone have a quick introduction to writing/compiling your own channel driver? If I finish mine up and put it into /channels will it be compiled? |
12:22.58 | *** join/#asterisk sanee (n=sanee@82.117.210.45) |
12:23.43 | xterminus | anyone know of a website/something where you can get it to call an arbitrary sip number (for testing incoming calls?) |
12:25.14 | Curus | xterminus: Your favourite phone should do that, if it supports IP dialing |
12:26.14 | linstar | in-side : I have created two asterisk servers and now want to connect them |
12:26.24 | *** join/#asterisk fourcheeze (n=rich@82.153.215.21) |
12:26.38 | fourcheeze | anyone know of an ATA that does SIP and Skype ? |
12:26.41 | xterminus | Curus, yah - but if its on the inside of a nat network, it really cant test for incoming nat traversal very well |
12:27.02 | *** join/#asterisk UlbabraB (n=salama@host241-43.pool8172.interbusiness.it) |
12:27.19 | linstar | in-side : If I dial one extension of the 2nd server from the 1st server I want it to work |
12:27.39 | linstar | in-side : Is it possible with sip configuration? |
12:32.35 | *** join/#asterisk basti__ (n=basti@dsl-220-253-65-42.NSW.netspace.net.au) |
12:33.05 | xterminus | linstar: with asterisk, use iax.conf to define a context for the remote asterisk server |
12:34.55 | astra^^ | WARNING[25769]: chan_sip.c:2530 sip_write: Asked to transmit frame type 4, while native formats is 1 (read/write = 4/4) |
12:34.57 | xterminus | linstar, then dial those numbers in extensions.conf like Dial(IAX2/hostname/{EXTEN}@remotecontext) |
12:35.31 | chris_ast | xterminus, I have a very small AGI in PHP and I need to know how can I read a argument passed to it? |
12:37.16 | linstar | xterminus : I have no hardware installed |
12:37.35 | linstar | <PROTECTED> |
12:38.41 | linstar | xterminus : I mean I have configured the asterisk servers with sip only |
12:40.04 | xterminus | linstar, concept is the same |
12:40.26 | xterminus | lilo, dialing command will be a little different is all |
12:40.28 | Zeeek | "asterisk is now deprecated. We recommend you purchase a Nortel pbx and quit fooling around" |
12:41.19 | xterminus | chris_ast, not sure - my experience is limited to arguments with websites - not php (looking) |
12:41.37 | linstar | xterminus : can I configure it same in sip.conf and extensions.conf? |
12:41.58 | linstar | xterminus : Can you give me good URL for reference? |
12:42.01 | Zeeek | chris_ast aren'(t all args passed on stdin ? |
12:42.42 | astra^^ | WARNING[25769]: chan_sip.c:2530 sip_write: Asked to transmit frame type 4, while native formats is 1 (read/write = 4/4) |
12:43.39 | xterminus | linstar, i dont have a url on hand other than the asterisk bible (voip-info.org) |
12:44.16 | kaldemar | linstar: http://www.voip-info.org/wiki-Asterisk+-+dual+servers |
12:44.23 | *** join/#asterisk lunaphyte_ (n=lunaphye@70.90.148.1) |
12:45.23 | Zeeek | I never figured out how switch works |
12:45.40 | Zeeek | or rather how to limit it |
12:46.01 | Zeeek | but calling the other server as a "user" works fine |
12:47.15 | xterminus | Zeeek, it looks like with php, you have to establish handes to deal with STDIN |
12:47.28 | xterminus | , |
12:47.28 | xterminus | STDOUT |
12:47.28 | xterminus | , and STDERR - easist way to deal with those variables would be to read them into a hash probably |
12:47.28 | xterminus | whoops |
12:47.48 | xterminus | xchat wierdness |
12:48.00 | Zeeek | yeah there is an example of almost every language (well three or four) |
12:48.26 | Zeeek | you could also use ENV type args |
12:49.22 | *** join/#asterisk coppice (n=chatzill@91.203.17.210.dyn.pacific.net.hk) |
12:49.35 | xterminus | bah - hashes are better (of course - this is coming from a perl fan =) |
12:53.27 | *** join/#asterisk dlynes (n=dlynes@S010600c09f9a0fc4.vc.shawcable.net) |
12:54.37 | *** join/#asterisk Tili (n=Tili@193.172.20.10) |
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13:01.37 | tzafrir | how do I authenticate SIP calls by the IP of the calling gateway? |
13:02.00 | tzafrir | let's assume I can trust the IP address |
13:02.08 | tzafrir | (not to be spoofed) |
13:02.31 | Zeeek | if it was spoofed, they'd never get an answer :) |
13:03.03 | *** join/#asterisk _Paulo_ (n=Paulo@200-168-112-132.dsl.telesp.net.br) |
13:03.48 | tzafrir | I can change the default context for all non-identified calls |
13:04.56 | tzafrir | But then I can't tell between two specific gateways |
13:05.16 | *** join/#asterisk [ProB]CrazyMan (n=crazyman@p549F0A92.dip0.t-ipconnect.de) |
13:05.17 | _Paulo_ | coppice, how are you? |
13:05.24 | coppice | OK |
13:05.41 | Zeeek | good |
13:05.43 | *** join/#asterisk medusaXX (i=proxy@p54A9A5E7.dip0.t-ipconnect.de) |
13:06.01 | [ProB]CrazyMan | coppice: if my faxes look like : http://www.roterschnee.org/failure.tif is this an libtiff issue ? |
13:06.10 | _Paulo_ | coppice, I think need some help with libmfcr2 |
13:06.31 | *** join/#asterisk Creperum (n=ilya@mail.tex.kiev.ua) |
13:07.48 | _Paulo_ | coppice, I was trying to send a clear back right after answer, but could not figure out where should I insert this. |
13:08.33 | _Paulo_ | [ProB]CrazyMan, this is with rx_fax or tx_fax? |
13:08.55 | [ProB]CrazyMan | _Paulo_: rx_fax |
13:09.00 | coppice | if you want to clear back after answer just drop the call |
13:09.24 | *** join/#asterisk tdonahue (n=tdonahue@208.51.101.201) |
13:09.30 | _Paulo_ | coppice, there is a provision in some pbx in brazil |
13:09.54 | _Paulo_ | coppice, to avoid collect calls they send a clearback after answer |
13:10.27 | _Paulo_ | I want to send the clearback without drop the call |
13:11.17 | coppice | yeah, I know about that trick |
13:11.17 | _Paulo_ | [ProB]CrazyMan, what version of libspandsp are you using? |
13:12.11 | _Paulo_ | coppice, I tried to drop the cal inside chan_unicall, but I cant re-answer, because the call is in incorrect state |
13:12.13 | coppice | [ProB]Crazyman: rxfax didn't produce that TIFF file |
13:12.17 | [ProB]CrazyMan | _Paulo_: spandsp-0.0.2pre25 |
13:13.36 | [ProB]CrazyMan | coppice: not? |
13:14.10 | coppice | Software (305) ASCII (2) 27<Adobe Photoshop CS Windo ...> |
13:14.23 | [ProB]CrazyMan | coppice: yes thats right .. |
13:14.31 | [ProB]CrazyMan | its not the orginal tif file .. |
13:14.35 | _Paulo_ | coppice, so, is libmfcr2 the right place to do this? |
13:14.46 | coppice | so its worthless crap, that is wasting my time |
13:14.53 | [ProB]CrazyMan | wait .. |
13:17.08 | coppice | _Paulo_ I think mfcr2.c needs to go into a special timed release-that-is-not-really-a-release state. You can easily make the current mfcr2 code tolerate these short clears, by adjusting a timer who's name I have forgotten. I haven't yet provided a means to generate them |
13:17.33 | *** join/#asterisk brookshire (n=mbrooks@gateway.digium.com) |
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13:18.31 | _Paulo_ | coppice, I've done that, tolerate these short clears... with mfcr2->clear_back_persistence_check = 1000 |
13:18.54 | _Paulo_ | coppice, What I wanto to do now is generate these myself. |
13:19.39 | coppice | I'll get around to doing it at some point, but I'm ripping the unicall stuff apart at the moment to restructure things |
13:21.42 | *** join/#asterisk fugitivo (n=ajf@201.255.177.90) |
13:21.43 | _Paulo_ | coppice, I wold like to help you |
13:21.50 | fugitivo | hello |
13:21.52 | coppice | [ProB]CrazyMan: that looks like things fell apart due to timing slips |
13:22.01 | [ProB]CrazyMan | hm |
13:22.20 | _Paulo_ | [ProB]CrazyMan, use the last IAXmodem stuff |
13:22.24 | [ProB]CrazyMan | coppice: so what could I do against timig slips ? |
13:22.40 | coppice | fix them. it is nothing to do with me |
13:23.14 | [ProB]CrazyMan | timing slips -> bristuff related ? |
13:25.01 | _Paulo_ | [ProB]CrazyMan, timing slips can be hardware related. |
13:25.31 | _Paulo_ | [ProB]CrazyMan, bad network interface cards |
13:25.38 | _Paulo_ | [ProB]CrazyMan, bad zaptel hardware |
13:25.50 | *** join/#asterisk illuy (n=assdf@85-65-123-85.barak-online.net) |
13:25.54 | _Paulo_ | [ProB]CrazyMan, shared interrupts |
13:26.23 | _Paulo_ | [ProB]CrazyMan, too much interrupts |
13:26.36 | [ProB]CrazyMan | _Paulo_: there is just one quadbri card inside |
13:26.59 | _Paulo_ | [ProB]CrazyMan, By the way, app_rxfax works fine for me with a digium TE110P |
13:27.54 | _Paulo_ | [ProB]CrazyMan, I had timing problems with app_txfax, but I'm using IAXmodem+hylafax now and it works fine. |
13:29.47 | _Paulo_ | [ProB]CrazyMan, for my needs, iaxmodem+hylafax proved to be a better fit then app_txfax, anyway, because I dont need to reinvent the wheel (the error handling). |
13:30.50 | *** join/#asterisk Skymarshal (n=Skymarsc@p54AF2C85.dip0.t-ipconnect.de) |
13:31.13 | _Paulo_ | coppice, what "set_mf_signal()" does? |
13:31.55 | coppice | if you haven't worked that out, I don't think you are going to get very far with solving your problem |
13:31.57 | *** part/#asterisk linstar (n=achu@220.225.191.18) |
13:32.21 | Skymarshal | I have an AGI problem. I want to use "STREAM FILE marryme 15000 3" but it doesn't work (CLI says that it is playing the file but I can not hear it). I can use "EXEC Playback marryme" but not the "STREAM FILE" command. Any idea why? |
13:32.34 | _Paulo_ | coppice, I tried to put a set_mf_signal(uc, ch, mfcr2->back_abcd_clear_back) after the set_abcd_signal(uc, ch, mfcr2->back_abcd_answer) |
13:33.27 | coppice | seems like you don't understand how MFC/R2 works |
13:33.57 | _Paulo_ | coppice, Yes, you are right |
13:34.15 | coppice | well, that makes it kinda hard to do development :-) |
13:35.05 | coppice | clue: once the call is answered, the MF signals are not used any more |
13:35.52 | _Paulo_ | coppice, where should I be looking? |
13:37.16 | coppice | after |
13:37.18 | coppice | <PROTECTED> |
13:37.19 | coppice | you need a timer to wait for a few hundred ms, then you need |
13:37.21 | coppice | <PROTECTED> |
13:37.22 | coppice | and another few hundred ms wait, then |
13:37.24 | coppice | <PROTECTED> |
13:37.44 | _Paulo_ | coppice, thanks. |
13:43.09 | *** join/#asterisk [ProB]CrazyMan (n=crazyman@p549F11B7.dip0.t-ipconnect.de) |
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13:55.50 | _Paulo_ | coppice, Is uc_schedule_event the rigth way to wait a few ms inside libmfcr2? |
13:56.29 | coppice | yep. if you look at all the other timers, they use that |
13:56.49 | *** join/#asterisk Mauro__ (n=mauro@oliver.altascumbres.cl) |
13:56.54 | Mauro__ | Hi |
13:57.04 | Mauro__ | I have a noob question :P |
13:57.32 | Mauro__ | what is the easiest way to develop a web app with asterisk? |
13:57.44 | Zeeek | using a bounty |
13:57.50 | MikeJ__ | HEH!! |
13:58.26 | Mauro__ | :P |
13:58.40 | _Paulo_ | coppice, thanks again. |
13:59.03 | MikeJ__ | Mauro__, pick your fav web dev language.... develop |
13:59.30 | *** join/#asterisk L|NUX (n=linux@202.5.145.58) |
13:59.43 | Mauro__ | php is fine? |
14:01.17 | *** join/#asterisk octothorpe_ (n=octothor@unaffiliated/octothorpe) |
14:01.35 | *** join/#asterisk apardo (n=apardo@87.218.45.153) |
14:02.02 | MikeJ__ | sure |
14:02.10 | MikeJ__ | depends on what you need to do |
14:02.19 | MikeJ__ | there is a php asterisk mod somewhere |
14:02.22 | *** join/#asterisk nextime (n=nextime@213-140-6-103.ip.fastwebnet.it) |
14:02.29 | MikeJ__ | ast_php or somthing like that |
14:02.31 | *** join/#asterisk sergeus (n=s@195.112.98.13) |
14:02.31 | MikeJ__ | never used it.. |
14:06.14 | astra^^ | WARNING[25769]: chan_sip.c:2530 sip_write: Asked to transmit frame type 4, while native formats is 1 (read/write = 4/4) |
14:06.35 | kardecallan | I have MFC-5C class, I want to integrate with asterisk. Is there anybody that can help me? |
14:06.47 | astra^^ | i get tis error wen a call is placed frm a ser .. to * we are using g729 at both ends |
14:07.05 | *** join/#asterisk X-Rob (n=rob-x@dsl-202-173-151-24.qld.westnet.com.au) |
14:07.27 | *** join/#asterisk digg10 (n=john@206-248-152-244.dsl.teksavvy.com) |
14:09.33 | _Paulo_ | coppice, I looked at all the other timers, all set the callback function. Is it mandatory? |
14:10.11 | _Paulo_ | (my brain hurts from reading C code) |
14:10.51 | Mauro__ | where can I read the asterisk api documentation? |
14:11.41 | *** join/#asterisk PumpkinPie (n=a@unaffiliated/PumpkinPie) |
14:11.44 | PumpkinPie | Hello |
14:11.48 | xterminus | is it possible for iax phones (idefisk,etc) to make sip address calls (eg sip:extension@domain.name) through asterisk? |
14:12.41 | *** join/#asterisk Utah_Dave (n=boucha@0-1pool149-100.nas31.salt-lake-city1.ut.us.da.qwest.net) |
14:13.12 | PumpkinPie | I have voicepulse service but not 'connect voicepulse' service. Can I still use asterisk ? |
14:13.51 | kaldemar | Mauro__: you can find some at voip-info.org |
14:14.42 | Mauro__ | thanks |
14:15.07 | _Paulo_ | kardecallan, coppice wrote libunicall |
14:15.18 | MikeJ__ | Mauro__, fore real api, there is doxygen in the code, it's onthe asterisk.org site |
14:15.34 | *** join/#asterisk zotz (n=zotz@24.231.32.85) |
14:17.40 | Mauro__ | the real api is written in C? |
14:17.54 | MikeJ__ | y |
14:23.40 | PumpkinPie | can I replace hardware based voip products with software? |
14:24.17 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
14:24.21 | _Paulo_ | PumpkinPie, software runs in hardware. |
14:24.30 | *** part/#asterisk babyhuey (n=justin@ip-131-123-81-11.housing.res.kent.edu) |
14:24.54 | _Paulo_ | PumpkinPie, you can replace almost any appliance with a PC |
14:25.39 | kardecallan | _Paulo_ I need to write it? |
14:26.24 | _Paulo_ | kardecallan, are you into #asteriskbrasil.org? |
14:26.45 | PumpkinPie | Paulo: im talking about asterisk to repalce my sipura device specifically |
14:27.18 | Zeeek | PumpkinPie yes |
14:27.58 | Creperum | are there some apps for using * as a telemarketing tool or CATI? |
14:28.14 | PumpkinPie | Zeeek: I think voicepulse gathers the serial number off of my sipura device so it knows which is my account? etc? |
14:28.17 | _Paulo_ | PumpkinPie, replace a "no mobile parts" appliance with a PC has some tradeofs |
14:28.41 | PumpkinPie | how do I configure the serial number with asterisk |
14:29.04 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
14:29.04 | Zeeek | use voicepulse connect and IAX |
14:29.18 | PumpkinPie | i dont want to use 'voicepulse connect' |
14:29.40 | PumpkinPie | theres no point in me asking if it can replace my hardware if I use 'voicepulse connect' |
14:29.43 | *** join/#asterisk Emrah (n=user@adslgva0491.worldcom.ch) |
14:29.53 | Emrah | Hello! |
14:30.19 | Zeeek | I don't use vp SIP so I don't know about serial numbers. MOst SIP providers don't do that AFAIK |
14:30.25 | Emrah | I'm wondering, do you know how it is possible to generate a call from asterisk and starting an application when the remote party peak up the phone? |
14:30.37 | Zeeek | but just so you understand, one can use an IAX phone with voicepulse connect so there was nothing funny about my answer |
14:30.47 | Emrah | I mean from a way or another, I'd like to receive a call, like a call back application, and be able to access my voicemail |
14:31.49 | PumpkinPie | i dont care what you can or cant use with 'voicepulse connect' |
14:31.49 | PumpkinPie | thats not what I have |
14:31.49 | kmilitzer | Can someone tell me how I can find out which codecs are actually installed and usable in my asterisk? |
14:31.58 | Zeeek | I do,'t know any reason why voicepulse normal service wouldn't work |
14:32.08 | Zeeek | but then I don't have that service |
14:32.37 | Emrah | kmilitzer: show codecs |
14:32.46 | Emrah | or something like that let me try |
14:32.54 | Creperum | ppl, my client wants to build a callcentre for outgoing calls - he wants to call his clients periodically - how can it be done with *? are there some scripts? |
14:33.08 | PumpkinPie | it identifies my sipura unit with its serial number yes? or with what? and how do I setup that serial number with asterisk? |
14:33.21 | kmilitzer | This shows G.729A which I haven't installed (that's the version where u need the license for, right?) |
14:33.28 | jsharp | Creperum: Yes and yes. |
14:33.46 | Zeeek | show codecs just shows a table of them - not nec installed |
14:33.46 | Emrah | kmilitzer: in your * cli ti sow codecs |
14:33.55 | Creperum | jsharp: ok, where they can be found? |
14:34.09 | Zeeek | in modules maybe? |
14:34.13 | PumpkinPie | how do I make asterisk appear/work as a replacement of my sipura unit ? |
14:34.20 | digg10 | hello |
14:34.39 | RoyK | ~seen zoa |
14:34.41 | jbot | zoa <n=kkk@pirus.securax.be> was last seen on IRC in channel #asterisk, 4d 21h 51m 24s ago, saying: 'it looks kinda suspicious :p'. |
14:34.43 | Emrah | sorry for my wrong answer if it is one |
14:35.05 | jsharp | Creperum: Depends on how complex you want the system to be. Asterisk has some basic autodial out functions. But if you want something like a predictive dialer, you'll need some additional software. |
14:35.08 | Emrah | Anyone has an idea about a way to make Asterisk calling a number from the cli or the manager and then starting an application? |
14:35.30 | Zeeek | Emrah if it's voicemail, just put the voicemail number in the call file |
14:35.32 | *** join/#asterisk coppice (n=chatzill@21.202.17.210.dyn.pacific.net.hk) |
14:35.47 | Emrah | What do you mean the call file? |
14:36.02 | Creperum | jsharp: i know only about .call files... is there more complex soft? |
14:36.24 | Zeeek | Emrah I must have mixed up your question with someone lese's |
14:36.27 | jsharp | Yes. There are companies who have build full-scale telemarketing level predictive dialers. |
14:36.39 | jsharp | And call centre packages. |
14:36.43 | jsharp | For Asterisk. |
14:37.16 | Creperum | jsharp: of course, it's proprietary soft.. |
14:37.40 | jsharp | I think there's a GPL package. |
14:37.58 | *** join/#asterisk io_error (n=error@mdsnwikwbas08-pool7-a82.mdsnwikw.tds.net) |
14:38.07 | Creperum | jsharp: please, can you name any? |
14:38.17 | digg10 | how can i put some conditions in the dialplan? |
14:38.26 | *** join/#asterisk austinnichols101 (n=austinni@70.46.69.131) |
14:38.45 | digg10 | i also need to query other databases while in the dialplan |
14:38.57 | *** join/#asterisk exten123 (n=exten@202.133.101.88) |
14:39.14 | *** join/#asterisk cuco (n=diego@local.xorcom.com) |
14:39.22 | Tili | hey is there any idea of enabling H323 channel to let Asterisk act as 3G gateway |
14:40.51 | HamYaI | is FC5 released today? |
14:42.20 | exten123 | ya |
14:42.35 | io_error | I'm already running it :) |
14:43.14 | exten123 | io_error, any thing that need to change in order run asterisk? or just like install in core 4? |
14:43.46 | io_error | Should be the same as always. |
14:44.10 | exten123 | io_error, what advantages u give after using fc5? |
14:44.23 | io_error | Oh, it's nice :) |
14:44.39 | PumpkinPie | how do I make asterisk appear/work as a replacement of my sipura unit ? |
14:45.13 | *** join/#asterisk core-ix (n=ivo@pirus.securax.be) |
14:45.59 | exten123 | PumpkinPie, buy single channel fxo port card. and install in 1 of ur pc that run linux and install asterisk inside that pc |
14:46.00 | HamYaI | io_error: I can't see in the download site, where did you get it from? |
14:46.52 | PumpkinPie | exten123 cant I not use hardware? |
14:47.22 | PumpkinPie | I want to replace my sipura with software |
14:47.42 | *** join/#asterisk tmjb (n=tmjb@www.grappoloin.com) |
14:47.48 | io_error | PumpkinPie: just set up a SIP trunk to your provider |
14:47.55 | *** join/#asterisk k31th (n=keith@87.117.194.66) |
14:48.11 | snip3r | Zeeek: are you around? |
14:48.20 | Zeeek | just leavinjg |
14:48.22 | snip3r | :) |
14:48.23 | k31th | bit of a longshot this but does anyone use speex ? |
14:48.28 | Zeeek | changed router? |
14:48.43 | snip3r | I've installed * on another box and it works like a charm |
14:49.00 | Zeeek | hmmmmm odd. But good in a way |
14:49.02 | tmjb | hello,could some one recomend me what cisco phone is the best for Asterix PBX tnx |
14:49.09 | Skid | 7940/60 ? |
14:49.12 | Skid | they both work well for me |
14:49.15 | snip3r | tried the previous one with the Linksys router, but without any luck |
14:49.18 | Zeeek | Sipura 940 |
14:49.26 | snip3r | LOL |
14:49.28 | snip3r | :) |
14:49.34 | *** join/#asterisk hfern (n=hfern@h-64-105-50-78.dllatx37.dynamic.covad.net) |
14:49.40 | snip3r | Sipura is Cisco, in a certain sense |
14:49.50 | Skid | ya, *cough* |
14:50.17 | backblue | Sipura is cisco? *g* |
14:50.31 | tmjb | Sipura is Cisco ?? |
14:50.35 | snip3r | ... |
14:50.38 | Skid | no. |
14:50.42 | snip3r | Linksys acquired Sipura |
14:50.51 | k31th | sipura / linksys / cisco |
14:50.53 | Skid | anyway, imo go with 7940/60 with sip 7.4 |
14:50.54 | snip3r | Linksys is a division of Cisco |
14:50.57 | Skid | 7.5 == buggy |
14:51.04 | PumpkinPie | where can I find info on 'sip trunk' setup ? |
14:51.10 | backblue | its not the same |
14:51.11 | snip3r | 7960 has a nice huge screen |
14:51.15 | k31th | the sipura phones look the same as the cisco ones work the same less features basically |
14:51.17 | backblue | as linksys its not cisco |
14:51.20 | Skid | i've heard there's a 7970 sip image too |
14:51.23 | Skid | whicih i need to get hold of |
14:51.28 | exten123 | any one got any idea make asterisk have a future like live call monitoring another party conversations? does u guy thing meetme can dun this, or actually got others existing function work for that. |
14:51.29 | backblue | even if they are all in the same company |
14:51.30 | Skid | ust cant be arsed :) |
14:51.48 | io_error | HamYaI: download.fedora.redhat.com |
14:51.51 | snip3r | hold of what? |
14:52.01 | snip3r | ^^ Skid? |
14:52.13 | Skid | 7970 sip |
14:52.16 | snip3r | IC |
14:52.22 | Skid | someone mentioned it in here two weeks back |
14:52.26 | snip3r | um |
14:52.50 | Skid | one thing that really annoys me, about cisco phones are the logo displays tend to work on and off |
14:52.59 | Skid | if i reboot, it'll be unable to locate http server |
14:53.02 | Skid | reboot again, fine |
14:53.07 | Skid | reboot later, same :P |
14:53.08 | Skid | etc etc |
14:53.20 | snip3r | I need something like a 48V adapter or a PoE converter for the 7960 |
14:54.16 | snip3r | do you know an on-line shop that sells Cisco PSU's? |
14:54.21 | Skid | erm |
14:54.25 | Skid | there's some chap on ebay that sells them new |
14:54.30 | Skid | i bought a few from him |
14:54.36 | snip3r | great |
14:54.38 | Skid | like 15 gbp |
14:54.56 | Skid | depends on how many phones you have, rather than going for a poe switch i guess |
14:55.03 | snip3r | I've got a pair of these nice phones, but need some adapters |
14:55.27 | snip3r | hardly can wait to get my hands on 'em |
14:55.40 | Skid | :) |
14:56.08 | Skid | i need to sort out an alternative provider now, the morons who i currently have support is a waste of time |
14:56.17 | Skid | took me 2 weeks to get 3 new lo-call numbers |
14:56.25 | Skid | and now they wont sodding work with their shitty control panel |
14:56.49 | Skid | I peer with about 3 i think, so i must try them first for the amount of waffling my girlfriend does heh |
14:57.27 | tmjb | is cisco 7970 or some other 100% compatible with asterisk? |
14:57.43 | Skid | 7940/60's are |
14:57.45 | Skid | (SIP) |
14:57.49 | Skid | 7912 too |
14:57.55 | Skid | we've usd all those on our asterisk box |
14:58.05 | Skid | right from 1.0.7 |
14:58.09 | austinnichols101 | snip3r: voipsupply.com |
14:58.25 | tmjb | tnx Skid |
14:58.25 | *** join/#asterisk MRH2 (n=Mr_happy@fcirc-adsl.demon.co.uk) |
14:58.29 | *** join/#asterisk oej (n=oej@bkkb-gw.bitcon.no) |
14:58.29 | Skid | np |
14:58.32 | Skid | 12's are cheap |
14:58.35 | Skid | (obv) |
14:58.43 | Skid | 40's are cheaper than 60's bu tonly have 2 lines |
14:58.55 | Skid | suppose it depends what your needs are though |
14:59.08 | austinnichols101 | having 2 lines may not be an issue because you only use one for multiple calls in a basic setup |
14:59.12 | Skid | again, there's people that sell 10/100 lots on ebay refurb's |
15:02.14 | *** join/#asterisk nfi|ermes (n=ermsewrk@217.220.121.62) |
15:02.16 | astra^^ | <PROTECTED> |
15:02.16 | astra^^ | <PROTECTED> |
15:02.16 | astra^^ | <PROTECTED> |
15:03.20 | kmilitzer | astra^^: What should that tell us? |
15:03.49 | *** join/#asterisk wunderkin (n=kev@ip68-226-113-228.ph.ph.cox.net) |
15:04.02 | rpm | brrr... |
15:04.22 | Skid | kmilitzer: can't you read? :P the sky is yellow is means |
15:05.43 | MRH2 | Hi anyone know how long it normally takes for patches to be reviewed (not my patch just a bug i would like to have a fix for) :) |
15:05.50 | astra^^ | kmilitzer: :) |
15:05.50 | kmilitzer | Skid: Sorry, not enough coffee, now that you say it, I can see it too ;) |
15:05.53 | astra^^ | done |
15:06.00 | astra^^ | i forgot to put { |
15:06.03 | astra^^ | heheheh |
15:06.11 | astra^^ | Thanx anyway |
15:10.11 | *** join/#asterisk Zeeek (n=IceChat7@pdpc/supporter/active/Zeeek) |
15:16.41 | Nivex | oooh! http://lwn.net/Articles/167897/ Will that deprecate ztdummy for things like MeetMe? |
15:18.38 | *** join/#asterisk miztic (n=gerard@rarcoa.com) |
15:19.07 | kmilitzer | Any idea how I can measure the jitter of a call/connection? |
15:19.35 | *** join/#asterisk azzie (n=az@azzie.net) |
15:20.04 | blitzrage | kmilitzer: iax2 jb debug |
15:20.46 | kmilitzer | blitzrage: ... for SIP calls? ;) |
15:20.59 | *** join/#asterisk shido6 (n=shido6@d221-68-217.commercial.cgocable.net) |
15:21.41 | vgster | does anyone run chan_fax? |
15:22.31 | exten123 | does digium analog card support FSK caller id? |
15:23.21 | MikeJ[Laptop] | vgster, we run chan_fax |
15:23.27 | MikeJ[Laptop] | exten123, yes |
15:23.46 | vgster | i get an error when i build it |
15:23.48 | *** join/#asterisk sergeus (n=s@195.112.98.13) |
15:24.03 | blitzrage | kmilitzer: not sure for sip -- I don't see anything for it |
15:24.04 | MikeJ[Laptop] | your the second one who has told me that, and I can't replicate it |
15:24.17 | MikeJ[Laptop] | what kind of box, what asterisk version, what error? |
15:25.06 | MikeJ[Laptop] | I think a change in trunk asterisk broke it... |
15:25.11 | *** part/#asterisk io_error (n=error@mdsnwikwbas08-pool7-a82.mdsnwikw.tds.net) |
15:25.24 | vgster | MikeJ[Laptop] > did you get any errors when you built it? |
15:25.37 | MikeJ[Laptop] | no, but have not done against truck |
15:25.39 | MikeJ[Laptop] | trunk |
15:25.46 | kmilitzer | blitzrage: it would be really cool to meassure the jitter between two SIP-Systems ... |
15:25.47 | MikeJ[Laptop] | give me the info.... |
15:25.53 | tzanger | blitzrage: actually iax2 show netstats |
15:25.55 | blitzrage | MikeJ[Laptop]: do you happen to know? |
15:25.56 | MikeJ[Laptop] | OS, asterisk version, error... |
15:25.58 | tzanger | iax2 jb debug is a little more... verbose |
15:26.03 | blitzrage | tzanger: right -- that too, I forgot about that |
15:26.07 | MikeJ[Laptop] | know what? |
15:26.08 | tzanger | MikeJ[Laptop]: what problem is this? |
15:26.18 | MikeJ[Laptop] | huh? |
15:26.24 | MikeJ[Laptop] | I'm lost |
15:26.26 | blitzrage | MikeJ[Laptop]: if there is a "netstats" style thing for SIP? |
15:26.32 | blitzrage | tzanger: you'd probably know |
15:26.33 | *** join/#asterisk angler (n=angler@pdpc/sponsor/digium/angler) |
15:26.36 | MikeJ[Laptop] | oh, the rtcp stuff |
15:26.42 | MikeJ[Laptop] | did that ever get commited? |
15:26.48 | MikeJ[Laptop] | that |
15:26.53 | exten123 | MikeJ[Laptop], it auto support? |
15:26.56 | MikeJ[Laptop] | that's where it is.. not sure if it got in or not |
15:26.57 | tzanger | no netstats for sip yet |
15:27.01 | tzanger | at least none that I"m aware of |
15:27.02 | blitzrage | not that I'm aware of ... but yah, that'd be required to measure the jb in SIP right? |
15:27.05 | exten123 | MikeJ[Laptop], do I need configure any thing? |
15:27.29 | MikeJ[Laptop] | exten123, don't even know if it got committed... it was in the bug tracker a long time ago |
15:27.35 | MikeJ[Laptop] | search for RTCP |
15:27.41 | MikeJ[Laptop] | see what happened to it |
15:27.52 | blitzrage | I'm going to find it now |
15:28.05 | Juggie | MikeJ[Laptop], it still in progress so far as i know |
15:28.09 | Juggie | oej still wants it |
15:28.13 | blitzrage | http://bugs.digium.com/view.php?id=2863 |
15:28.17 | blitzrage | still open |
15:28.28 | MikeJ[Laptop] | wow.. old bug |
15:28.35 | blitzrage | aye |
15:28.37 | oej | And a branch to test :-) |
15:29.05 | Juggie | he speaks |
15:29.06 | blitzrage | oej made the last post on March 9th |
15:29.10 | blitzrage | all hail oej |
15:29.14 | blitzrage | hail! |
15:29.17 | *** join/#asterisk h4mm3r` (n=h4mm3r@85-18-14-10.fastres.net) |
15:29.20 | oej | And it's part of test-this-branch |
15:29.26 | oej | Which all of you tested during the weekend! |
15:29.29 | oej | Right? |
15:29.46 | blitzrage | oej: I didn't sleep at all -- I was up with test-this-branch all night... he just wouldn't sleep |
15:29.48 | oej | I guess I have to code the missing part of RTCp |
15:30.02 | oej | Where we take some data into the dialplan |
15:30.06 | Juggie | we dont pay you for nothing :) |
15:30.10 | oej | Or a cdr variable |
15:30.23 | blitzrage | oej: Sweden got a mention this morning in a report I was watchin on CPAC about the Canadian economy, and apparently Sweden figured something out about money :) |
15:30.24 | *** join/#asterisk mhnoyes (n=mhnoyes@user-2ivfist.dialup.mindspring.com) |
15:30.26 | oej | Oh, you don't pay me nothing for nothing? |
15:30.40 | blitzrage | I prefer to pay nothing for the world |
15:30.45 | oej | Oh, if we figured out something about money in Sweden, I missed it since I am in Norway |
15:30.46 | Juggie | oej, i told you, setup a paypal account, i'll send beer. |
15:30.55 | tsume | heh |
15:30.59 | oej | And these guys figured out something about oil... |
15:31.14 | tsume | and 75 for the people in town ;) |
15:31.24 | blitzrage | tsume: you don't charge enough :D |
15:31.34 | tsume | blitzrage: I know ;( |
15:31.44 | tsume | blitzrage: I have contracts, so it doesn't matter :) |
15:31.45 | blitzrage | tsume: I'm the same way though |
15:31.50 | blitzrage | tsume: same |
15:33.23 | *** join/#asterisk iq (n=iq@71-214-5-20.omah.qwest.net) |
15:34.40 | *** part/#asterisk cuco (n=diego@local.xorcom.com) |
15:36.54 | Zeeek | no matter how much I charge they always want to help |
15:37.03 | Zeeek | which makes it that much harder |
15:37.04 | *** join/#asterisk stkn (n=stkn@gentoo/developer/pdpc.active.stkn) |
15:37.40 | PumpkinPie | tsume: can I replace my sipura hardware with asterisk, without buying any more hardware? |
15:37.58 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
15:37.58 | *** mode/#asterisk [+o anthm] by ChanServ |
15:38.22 | Zeeek | PumpkinPie what is your sipura hardwireare btw? an ATA? |
15:38.54 | PumpkinPie | I think so yes |
15:38.56 | PumpkinPie | spa-2000 |
15:39.35 | Zeeek | and is there some kind of restriction in your vp agreement saying you can only use their hardware or sipura ATA? |
15:39.36 | MikeJ[Laptop] | asterisk does not have a magic software fxo port to plug your lines int |
15:39.38 | MikeJ[Laptop] | into |
15:40.03 | MikeJ[Laptop] | so you can not replace an ata with asterisk without using hardware... |
15:40.16 | PumpkinPie | I dont need to plug my phone into it at all |
15:40.23 | MikeJ[Laptop] | ummm |
15:40.30 | MikeJ[Laptop] | so you don't use the ATA then? |
15:40.41 | PumpkinPie | I dont have to, no |
15:41.01 | PumpkinPie | theoretically it shouldn't be hard to make software appear as the spa-2000 unit |
15:41.10 | Zeeek | PumpkinPie maybe tell us what exact setup you'd want if you could use asterisk and vp? I'm not sure I understand? I thought you just had a phone you wanted to replace |
15:41.21 | PumpkinPie | im not sure why you need a fxo |
15:41.28 | PumpkinPie | not my phone |
15:41.30 | MikeJ[Laptop] | ok.. totally lost me |
15:41.33 | PumpkinPie | I dont need the phone |
15:41.43 | PumpkinPie | I want to replace the spa-2000 unit with a software solution |
15:41.49 | MikeJ[Laptop] | you have an ata.. that you don't use, that you want to replace with asterisk, to do nothign? |
15:41.54 | Zeeek | what's connected to the sipura? |
15:41.55 | *** part/#asterisk mhnoyes (n=mhnoyes@user-2ivfist.dialup.mindspring.com) |
15:42.03 | in-side | Hi |
15:42.06 | Skid | w 14 |
15:42.10 | PumpkinPie | my phone right now |
15:42.17 | in-side | does anyone here had worked with agi setcallback ? |
15:42.28 | MikeJ[Laptop] | PumpkinPie, well.. you can't do that without hardware |
15:42.31 | Zeeek | so you want to plug your sipura into asterisk and have asterisk talk to vp ? |
15:42.45 | PumpkinPie | no I dont want to plug my phone into it |
15:42.48 | in-side | any one here are used to deal with perl agi ? |
15:42.50 | PumpkinPie | I dont need a phone |
15:42.56 | MikeJ[Laptop] | PumpkinPie, so what is it going to do |
15:43.00 | MikeJ[Laptop] | just voip to voip? |
15:43.05 | MikeJ[Laptop] | or voip to ivr |
15:43.12 | MikeJ[Laptop] | no analog phone? |
15:43.15 | PumpkinPie | answer the call and play a sound.. hang up |
15:43.15 | Zeeek | the point is, I think vp will let asterisk connect |
15:43.21 | PumpkinPie | no analog or digital phone |
15:43.24 | Zeeek | for whatever you want to do with that |
15:43.26 | Hmmhesays | god I hate this place |
15:43.29 | MikeJ[Laptop] | PumpkinPie, yes, with a voip provider, you can probably do that |
15:43.32 | in-side | I can get setcallback working in perl agi |
15:43.43 | in-side | it result 1 no error |
15:43.46 | MikeJ[Laptop] | Hmmhesays, but you love me, right? |
15:43.47 | PumpkinPie | I want to replace the spa-2000 unit with a software solution... |
15:43.51 | in-side | but doesn't execute the sub |
15:44.01 | in-side | does anyone have an ideia why that is happening ? |
15:44.09 | MikeJ[Laptop] | PumpkinPie, yes, most likely you can. |
15:44.10 | Zeeek | what is the functionality you want to replace with asterisk? |
15:44.23 | MikeJ[Laptop] | do you have passwords and such for your voip account |
15:44.31 | MikeJ[Laptop] | Zeeek, he said, play sound, hangup |
15:44.39 | PumpkinPie | how do I configure asterisk to replace my spa-2000 unit? |
15:44.54 | MikeJ[Laptop] | PumpkinPie, you read the stuff on the wiki |
15:44.55 | Zeeek | yeah you can do that |
15:45.00 | PumpkinPie | i dont care if it plays a sound or not.. its just an example |
15:45.02 | MikeJ[Laptop] | set up your sip config |
15:45.02 | PumpkinPie | how do I configure asterisk to replace my spa-2000 unit? |
15:45.02 | in-side | :S |
15:45.09 | MikeJ[Laptop] | ~docs |
15:45.11 | jbot | [docs] 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 |
15:45.13 | Zeeek | have you built asterisk yet PumpkinPie |
15:45.15 | MikeJ[Laptop] | you read taht |
15:45.22 | PumpkinPie | I looked at the sip config and I see lots of crap I dont have |
15:45.35 | MikeJ[Laptop] | PumpkinPie, who is the voip provider |
15:45.35 | Zeeek | nah |
15:45.37 | *** join/#asterisk SplasPood (n=jwb@206.252.198.100) |
15:45.41 | Zeeek | voicepulse |
15:45.49 | in-side | great no perl guys here? |
15:45.52 | MikeJ[Laptop] | they have sample asterisk configs on their site |
15:46.02 | MikeJ[Laptop] | in-side, more of an opal guy |
15:46.03 | Zeeek | unless they have a weird proxy system or a license agreement, it'll be possible |
15:46.12 | in-side | opal? |
15:46.16 | MikeJ[Laptop] | voicepulse is asterisk friendly |
15:46.30 | Zeeek | well, connect is, I'm not so sure about VP vanilla |
15:46.32 | MikeJ[Laptop] | pearl, opal... insert other semi precious stone here |
15:46.39 | PumpkinPie | MikeJ: that is for their service specifically designed for it... called 'voicepulse connect' |
15:46.40 | MikeJ[Laptop] | hmm |
15:46.42 | MikeJ[Laptop] | true |
15:46.43 | in-side | rotlfl |
15:46.43 | PumpkinPie | thats not what I have... |
15:46.49 | in-side | damn... |
15:46.54 | MikeJ[Laptop] | dunno then |
15:46.58 | Zeeek | PumpkinPie still, it should be easy to do |
15:47.05 | MikeJ[Laptop] | in-side, sure there is somwhere |
15:47.08 | MikeJ[Laptop] | what's the prob |
15:47.11 | Zeeek | usually the phone/ATA has more shit in the config than asterisk needs |
15:47.12 | in-side | stupid dcumentation |
15:47.14 | MikeJ[Laptop] | I can hobble my way a bit |
15:47.15 | *** join/#asterisk epablo (n=epablo@200.109.73.215) |
15:47.29 | in-side | I can't figure out why setcallback simple refuses to work :S |
15:47.33 | epablo | hi people.. how's it going? |
15:47.35 | PumpkinPie | how do I get login information etc? |
15:47.36 | *** part/#asterisk chris_ast (n=Administ@59.93.56.163) |
15:47.45 | Zeeek | by looking at the ATA web interface |
15:47.53 | in-side | PumpkinPie: please go read the documentation man :( |
15:48.14 | PumpkinPie | in-side when I clicked on the documentation link it just went to some website asking me to buy stuff |
15:48.18 | *** join/#asterisk asterboy (n=kevin@S010600485480f4be.ed.shawcable.net) |
15:48.50 | Zeeek | *you have to read the doc for your sipura and find the web interface. Look at that stuff and you'll be ok |
15:49.02 | in-side | PumpkinPie: welcome to internet... |
15:49.29 | *** join/#asterisk gbodemantv (n=gbodeman@216.142.38.154) |
15:49.40 | gbodemantv | hi all |
15:49.44 | epablo | I need to use the manager to capture SIP peer login info: when a user logs in,logs out or timesout. What event should I suscribe? |
15:49.57 | gbodemantv | anyone in here use REALTIME configs |
15:50.16 | gbodemantv | trying to figure out if I can user calling groups, AMP, and FOP with it |
15:50.27 | twisted[asteria] | if you mean realtime as in DB driven, yeah. if you mean realtime as in bill mahr, hell no |
15:50.59 | MikeJ[Laptop] | twisted[asteria], liar.. you love the bill mahr... cmon.. admit it |
15:51.10 | twisted[asteria] | MikeJ[Laptop], lol |
15:51.14 | MikeJ[Laptop] | morning |
15:51.20 | twisted[asteria] | or something like it. |
15:51.28 | gbodemantv | twisted: :) yeah the REALTIME DB |
15:51.31 | twisted[asteria] | it's the first day of spring, and it's nasty out. |
15:52.12 | Zeeek | tomorrow is fiorst day, no? |
15:52.24 | in-side | perl agi is so damn well documented :( (NOT!) |
15:52.30 | asterboy | we killed a whale and are feasting on whale blubber in our igloo. |
15:52.49 | twisted[asteria] | Zeeek, nope. |
15:52.51 | twisted[asteria] | Zeeek, http://www.holidaysmart.com/seasons.htm |
15:52.52 | twisted[asteria] | ;) |
15:53.05 | twisted[asteria] | today is the spring equinox, and thus, the first day of spring |
15:53.13 | gbodemantv | any suggestions on where to start |
15:53.23 | Zeeek | All my life I believed it was the 21st <( |
15:53.25 | MikeJ[Laptop] | in-side, use mod_perl |
15:53.48 | in-side | MikeJ[Laptop]: what for? |
15:53.56 | in-side | use mod_perl with asterisk ? |
15:53.57 | MikeJ[Laptop] | the perl agi stuff is... |
15:53.59 | MikeJ[Laptop] | yeah |
15:54.03 | in-side | really? |
15:54.09 | MikeJ[Laptop] | one sec |
15:54.27 | in-side | ok |
15:54.40 | PumpkinPie | all I need is the serial number and the mac address off the sipura so I can emulate the device? |
15:54.40 | in-side | does it fast as the other? |
15:54.42 | MikeJ[Laptop] | errr |
15:54.45 | MikeJ[Laptop] | res_perl |
15:54.50 | MikeJ[Laptop] | sorry.. wrong software |
15:55.04 | epablo | perl agi has a decent man |
15:55.21 | mishehu | hmm... anybody know of any good way to bill calltime for faxes sent thru hylafax using iaxmodem when using one instance of iaxmodem for multiple entities? |
15:55.24 | in-side | epablo: really? |
15:55.28 | in-side | rotfl.. |
15:55.54 | MikeJ[Laptop] | http://www.pbxfreeware.org/archives/2005/06/res_perl_welcom.html |
15:55.56 | in-side | I just stacked up here with setcallback |
15:56.04 | _Paulo_ | mishehu, use the hylafax logs instead of * |
15:56.11 | MikeJ[Laptop] | I was never a big agi fan.. |
15:56.17 | in-side | it is not supposed it to be executed after call ends ? |
15:56.32 | in-side | I need to execute a db query after asterisk set over the call |
15:56.36 | epablo | in-side: i've managed to make my stuff work.. ;) |
15:56.52 | in-side | epablo: well I not called it well documentated :) |
15:56.54 | in-side | anyway |
15:56.58 | in-side | question is... |
15:57.10 | in-side | I need to execute a query after call ends |
15:57.26 | in-side | but I have to be shure that asterisk has already handle with all bye stuff |
15:57.31 | in-side | how can i do it? |
15:57.40 | in-side | I was supposing i had to use setcallback |
15:58.23 | in-side | MikeJ[Laptop]: thanks but I will check it later |
15:58.34 | *** join/#asterisk stefan (i=stefan@home.stefan.id.au) |
15:58.35 | epablo | i did that with perl agi. |
15:58.47 | in-side | ok I'm using perl agi too |
15:58.54 | in-side | problem is that perl is executing the query |
15:58.59 | mishehu | _Paulo_: was hoping to streamline it to one single database if possible. |
15:58.59 | in-side | after the first bye leg |
15:59.02 | Zeeek | I'm sleepy and hungry. Can asterisk help ? |
15:59.07 | in-side | not after the leg two |
15:59.31 | in-side | how can I be shure to only execute a command in asterisk after a bye have been sent ? |
15:59.54 | gbodemantv | twisted: |
16:00.08 | _Paulo_ | mishehu, I have a patch so hylafax autenticates via mysql |
16:00.10 | mishehu | especially since hylafax logs are flat text files |
16:00.21 | gbodemantv | any resource you can recommend for REALTIME config |
16:00.23 | asterboy | I have a SIP line registered on my Polycom phone and a speed dial using "sip:extension@localhost", but only get the fast beep when trying to dial it. What is the format for a sip URI call? |
16:00.26 | kmilitzer | What's the planned release date of asterisk 1.4 ? |
16:00.34 | mishehu | _Paulo_: for hylafax 4.2.x ? got a url for hte patch? |
16:00.37 | epablo | in-side:are you trying to implemente a retry or something similar' |
16:00.55 | in-side | epablo: no I doing the handle of accounting to ser |
16:01.05 | in-side | yap but it is similar |
16:01.18 | _Paulo_ | mishehu, I wrote it. e-mail me and I send it to you. |
16:01.26 | in-side | problem is I have to be shure to execute it only when the last leg had dealed with bye |
16:01.30 | in-side | at asterisk side |
16:01.30 | asterboy | thought it was, "sip:exten@server" |
16:02.11 | asterboy | Anyone make a SIP to SIP call with Polycom phones? |
16:02.13 | epablo | in-side: so you wan't to run it on hangup of both channels? |
16:02.23 | astra^^ | Mar 20 10:01:39 NOTICE[25958]: chan_sip.c:6278 check_auth: stale nonce received from '1001 <sip:1001@64.246.52.52>' |
16:02.28 | in-side | MikeJ[Laptop]: thanks for the tip bbut res_perl is worse documented stuff I saw in my life |
16:02.39 | asterboy | OR how do you setup extension to extension calling? |
16:02.39 | in-side | epablo: exaclty |
16:02.50 | astra^^ | am gettin an error what does tat mean |
16:02.52 | astra^^ | Mar 20 10:01:39 NOTICE[25958]: chan_sip.c:6278 check_auth: stale nonce received from '1001 <sip:1001@64.246.52.52>' |
16:02.52 | *** join/#asterisk Jedirl (n=hhgds4@154.Red-217-127-168.staticIP.rima-tde.net) |
16:02.53 | in-side | after all legs been down on that transaction |
16:02.55 | Jedirl | Hello |
16:03.04 | Jedirl | I'm having problems with my AGI extensions |
16:03.06 | in-side | if possible keep state of some inside vars of call |
16:03.12 | in-side | like sip call id |
16:03.31 | epablo | in-side: I think you can do it when receving the exit code for the Dial.. |
16:03.45 | asterboy | Anyone have a Polycom to Polycom setup? |
16:03.54 | in-side | hmm |
16:03.57 | Jedirl | asterisk doesn't keep executing all the flow in the extension when the channel is hung-up |
16:04.00 | in-side | can you dram me it ;) |
16:04.07 | in-side | I think I already had it |
16:04.10 | mishehu | _Paulo_: You've Got Mail<tm> |
16:04.14 | in-side | I was trying to use setcallback |
16:04.21 | epablo | in-side: setup an AGI.. inside just do the dial.. and catch the return code.. Then do your stufff |
16:04.21 | in-side | watching callstatus also |
16:04.23 | astra^^ | what does stale nonce mean? |
16:04.55 | in-side | at this moment all stuff is working except the catch code |
16:05.01 | Jedirl | epablo: that's what I've done but my post-dial stuff never gets executed |
16:05.14 | in-side | should it should be something like... |
16:05.24 | in-side | agi(script1) |
16:05.29 | in-side | agi(scrip2) ? |
16:05.35 | fulgas | hey in-side :) |
16:05.39 | asterboy | Can anyone please verify for me the SIP URI format? |
16:05.41 | in-side | hey fulgas |
16:06.23 | epablo | in-side: it could work.. but I was thinking of doing it all inside script1 |
16:06.43 | in-side | at this moment I have all inside one script |
16:06.54 | in-side | problem is the asterisk execute it after the first bye |
16:06.56 | asterboy | can I use say, "sip:upstairs@192.168.1.2", (where upstairs is a registered line)? |
16:06.56 | in-side | :S |
16:07.15 | in-side | and problem is i need bith :( |
16:07.17 | in-side | both |
16:07.22 | epablo | Jedirl: I really haven't tried it.. but it should work |
16:07.40 | epablo | <PROTECTED> |
16:07.40 | asterboy | holy fuck, must be on ignore. |
16:08.15 | Jedirl | epablo: it doesn't |
16:08.20 | asterboy | nobody here uses sip or polycom? |
16:08.26 | Zeeek | yes, both |
16:08.31 | asterboy | jbot, do I exist? |
16:08.32 | jbot | ACTION does I exist. |
16:08.39 | in-side | ya maybe it is abetter ideia |
16:08.39 | Zeeek | but not polycom-polycom which you specifically requested |
16:08.42 | in-side | using two |
16:08.47 | *** join/#asterisk caio1982_ (i=caio1982@CAcert-br/caio1982) |
16:08.49 | in-side | it will force it run |
16:08.55 | in-side | jsut after hangup |
16:09.01 | *** part/#asterisk caio1982_ (i=caio1982@CAcert-br/caio1982) |
16:09.35 | asterboy | Zeek, how do you talk from one phone to the other? |
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16:09.43 | asterboy | like exten to exten |
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16:11.58 | Zeeek | asterboy thru asterisk - that's one of the things it does well |
16:13.07 | asterboy | Yes, I'm trying to do it through *. |
16:13.07 | Zeeek | our phones are in different locations behind different NAT so there's no reason to try |
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16:13.14 | RoyK | ops |
16:13.14 | in-side | cool... |
16:13.16 | Zeeek | I just dial say 2002 and asterisk handles it |
16:13.16 | Jedirl | uhm |
16:13.16 | Jedirl | I'm lost |
16:13.50 | mishehu | that was a nice little split there. |
16:13.50 | Jedirl | why my extension flow doesn't keep executing when the phone hangups? |
16:13.50 | epablo | in-side: did it work? |
16:13.50 | Jedirl | I can't catch the ANSWERED and DIALSTATUS and all that |
16:13.50 | asterboy | So you register one of the lines as 2002 and dial it from another phone...should be the same as what I'm trying to do. |
16:14.04 | in-side | didn't test it yet |
16:14.04 | in-side | I will drop a feedback |
16:14.04 | in-side | in a moment |
16:14.04 | mishehu | jsharp: heees? |
16:14.30 | Zeeek | asterboy this stuff is documented in a great artcile here: http://www.onlamp.com/pub/a/onlamp/2003/07/03/asterisk.html |
16:14.34 | asterboy | I'll test with numbers instead of Letters. |
16:14.47 | asterboy | thanks Zeeek, I'll digest that. |
16:15.13 | Jedirl | anyone could take a look at my extensions.conf? http://pastebin.com/612559. When Dial ends with a hangup, the rest of the extension doesn't get executed |
16:15.35 | jsharp | mishehu: Laughs. It just made me snicker. |
16:15.37 | Juggie | Jedirl, its not supposed to. |
16:15.44 | Jedirl | uh? |
16:15.57 | Juggie | well its hangup |
16:16.00 | Jedirl | so how can I get ANSWEREDTIME, DIALSTATUS and that? |
16:16.01 | Juggie | what are you expecting |
16:16.11 | Juggie | when a hangup occurs it goes to the h context. |
16:16.14 | iCEBrkr | yo yo yo |
16:16.30 | Jedirl | uhm? |
16:16.56 | mishehu | jsharp: I didn't realize heee was a way to laugh, but sure! |
16:17.00 | Jedirl | Juggie: where can I see an example of what you are saying? |
16:17.24 | *** join/#asterisk stoffell (n=stoffell@fw.catsanddogs.com) |
16:17.30 | Juggie | Jedirl, add 'exten=> h,1,Noop(Hello i was hungup)' |
16:17.36 | Juggie | to your context |
16:17.40 | Juggie | and you'll see what i mean |
16:17.43 | Juggie | ALSO |
16:17.55 | Juggie | there is another option, where you can have dial continue after a hangup. |
16:18.13 | Juggie | but i would only use that if you intend to do another subsequent dial |
16:18.35 | Jedirl | uhmmm |
16:18.37 | Jedirl | OK |
16:19.07 | Juggie | did you see it trap the hangup? |
16:19.22 | asterboy | Zeeek, not much at that link. Falling back on the * manual here: http://www.digium.com/handbook-draft.pdf |
16:19.26 | Jedirl | let me check |
16:19.47 | Zeeek | asterboy well I thought it explained how to dial a number and connect to a phone ;) |
16:20.17 | Jedirl | great Juggie, that's what I need |
16:20.22 | asterboy | Zeeek, it just shows a summary for me. |
16:20.22 | Juggie | you can use dial with option 'g' if you want to continue without triggering a hangup |
16:20.24 | Zeeek | you just need extensions that make the connection as in 2000,1,Dial(SIP/FUBad,45,t) |
16:20.39 | Juggie | but jerdirl, i woudnt use it unless absolutely necesairy |
16:20.40 | Zeeek | with the vmail logic if you want it |
16:20.42 | *** join/#asterisk trbldwine (n=trbldwin@adam.ur.northwestern.edu) |
16:20.42 | Jedirl | Juggie, the vars i've set are the same in the "h" extension? |
16:20.46 | Juggie | eg stacking dials together |
16:20.49 | Juggie | Jedirl, yes |
16:20.54 | Jedirl | then I'll use it |
16:21.06 | Zeeek | you can also have alpha extensions like Sales,1,Dial(SIP/Sales123) |
16:21.08 | Juggie | try Noop(${DIALSTATUS}) |
16:21.13 | Jedirl | :D |
16:21.14 | Jedirl | GREAT |
16:21.16 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
16:21.19 | Juggie | in exten=>h,1... |
16:21.20 | asterboy | Zeeek, that example helps...give's me a bone to chew on....giving it a try |
16:21.21 | Jedirl | it's easier than I though |
16:21.35 | Juggie | yep, you seem to understand programming so youl'l be fine |
16:21.39 | Juggie | check out www.voip-info.org |
16:21.43 | Juggie | lots of information there. |
16:21.45 | Zeeek | asterboy look up macro-oneline on the net |
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16:22.11 | Zeeek | http://www.google.com/search?q=asterisk+macro-oneline&start=0&ie=utf-8&oe=utf-8&client=firefox-a&rls=org.mozilla:en-US:official |
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16:24.55 | asterboy | Found an example: exten => 2030,1,Macro(oneline,SIP/2030) |
16:26.45 | asterboy | This exten => _1888NXXXXXX,1,Dial(SIP/${EXTEN}@sipphone) |
16:27.08 | *** join/#asterisk Octothorpe (n=octothor@unaffiliated/octothorpe) |
16:27.20 | asterboy | at least tells me my Dial( format is not "sip:${EXTEN..." |
16:27.46 | asterboy | looks like it should just be "${EXTEN}@sipphone" |
16:28.07 | Zeeek | I think I mùentioned to you about 9 months ago that ou should read the docs on the dialplan |
16:28.54 | Zeeek | the format for Dial is explained in show appication dial, in the asteriskdocs.org site and on the wiki |
16:32.11 | asterboy | ya the docs do give very good examples...been through the a dozen times...must not be intuative enough for me. |
16:32.35 | Zeeek | how intuitive is this: TECHNOLOGY/channel ? |
16:32.48 | asterboy | doesn't tell me much |
16:32.58 | asterboy | I need specifics. |
16:32.59 | Zeeek | then you need a dictionary as well :) |
16:33.23 | Zeeek | you need ti deduce the specifics frm the generalities |
16:33.38 | Zeeek | so you probably know that SIP is a technology? |
16:33.39 | asterboy | that I am not very good at. |
16:34.22 | asterboy | yes, however, what is the cormat for channel? I've seen sip:exten@server. |
16:34.22 | asterboy | but that does not work |
16:34.23 | Jedirl | Juggie |
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16:34.27 | Jedirl | I need the 'g' option but it doesn't seem to work |
16:34.44 | asterboy | I've seen SIP/exten@server, but for me that does not work. |
16:34.51 | asterboy | there is something missing, not sure what. |
16:34.53 | Zeeek | asterboy you've never seent hat in a dial command have you? |
16:35.05 | Zeeek | sip: is a SIP URL |
16:35.06 | Jedirl | well I think I need the 'g' option because I retry with another dialstring if the first's dialstatus was CONGESTION or CHANUNAVAIL |
16:35.12 | asterboy | URI |
16:35.36 | asterboy | but when dealing with general contexts, I'm left to try it when the other does not wok. |
16:35.42 | Zeeek | whatever, it won't go in the dial |
16:35.45 | asterboy | s/wok/work/ |
16:35.57 | Juggie | Jedirl, sounds like you probally do, if you dial w the g option it should go back into the second dial? |
16:36.31 | asterboy | It's very frustrating...if it did work I wouldn't be asking so I can understand what replaces general assumptions. |
16:36.56 | Zeeek | asterboy I understand that English may not be your mother tongue but you really need to read or re-read the docs mentioned because you're too far away from undertsanding the most basic principles of it all. And type show application dial at the cli and read and study that |
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16:37.23 | Jedirl | I dial Zap/g1/962510XYZ|30|g |
16:37.31 | Jedirl | but it still goes to the 'h' extension |
16:37.31 | Jedirl | :?? |
16:37.43 | asterboy | To my thinking, if I have a registered SIP line, why can't I dial extension@server? |
16:37.54 | asterboy | should be that simple...but obviously I'm missing something. |
16:38.42 | thieumS | hi, what 's happening if trunk=yes on A and nothing on B on a friendy IAX2 A-B interco ? |
16:39.08 | Juggie | Jedirl, paste bin the output for me. |
16:39.24 | Octothorpe | ~pb |
16:39.29 | jbot | it has been said that pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
16:39.57 | jsharp | thieumS: If you have trunk=yes on one side, but no the other, stuff breaks in ugly ways. |
16:40.21 | thieumS | that's what i thought |
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16:41.55 | *** join/#asterisk gaspiz (n=gaspiz@86.34.6.164) |
16:41.55 | sevard | I was looking at installing ARI. I'm running * on my wrt, problem is there is no PHP for WhiteRussian. I'm trying to think of a solution but coming up with nothing. |
16:41.56 | *** join/#asterisk SplasPood (n=jwb@206.252.198.100) |
16:42.33 | gaspiz | hi does anyone of you know an issue with phpagi and newer php versions? |
16:43.00 | Jedirl | Juggie I've pasted it to you by /msg because it contains my phone number... :D thanks |
16:43.03 | gaspiz | I installed a newer version of php and deadagi is now crashing |
16:44.10 | Jedirl | gaspiz: php 5? |
16:44.47 | iCEBrkr | sevard: fastAGI? |
16:45.24 | iCEBrkr | sevard: oh, nevermind me. oops |
16:45.28 | sevard | :) |
16:45.37 | sevard | I'm attempting to think of another solution |
16:45.38 | gaspiz | jedirl: yes |
16:45.40 | eliel | hello oej |
16:45.41 | sevard | attempting to think |
16:45.43 | sevard | key words :S |
16:45.57 | oej | Hello |
16:46.39 | eliel | oej: what do you think about pach (6735)? |
16:46.56 | Jedirl | gaspiz: try enabling Zend 1 compatibility in php.ini; anyway PHP5 is WAY different from PHP4, many scripts aren't compatible |
16:46.59 | oej | eliel: Sorry, no time right now |
16:47.09 | eliel | oej: ok no problem! :-) |
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16:48.00 | sevard | iCEBrkr: :( |
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16:52.32 | mutilator | http://time3.livejournal.com/62990.html |
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16:56.30 | gaspiz | jedril: I tried using a newer php 4.4.x and still having problems |
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16:57.02 | Jedirl | gaspiz: check which problems... "having problems" is not a very descriptive error |
16:58.48 | *** part/#asterisk hfern (n=hfern@h-64-105-50-78.dllatx37.dynamic.covad.net) |
16:59.41 | backblue | do you guys dont use iax2 trunks with domains suport? |
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17:03.40 | *** join/#asterisk RoyK (n=roy@host-81-191-115-203.bluecom.no) |
17:03.47 | stack_ | So my company decided to get a PRI for our telephone lines with asterisk as the PBX. I don't know too much about PRI and the company that is setting it up for us what's us to pick the options (Start Signal, Signal Protocol, Outpulse, etc...) for our line, which I know nothing about. We are getting a Digium TE110P. What options should I be looking for? |
17:04.17 | jsharp | Are you in the US? |
17:04.20 | stack_ | yes |
17:04.43 | jsharp | Tell em you want a PRI that follows NI-2 ISDN. |
17:04.44 | _Paulo_ | mishehu, I sent a mail to you |
17:04.53 | *** join/#asterisk unmanaged (n=unmanage@64.89.118.139) |
17:05.05 | jsharp | If they stare blankly at you, get a new telco. |
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17:05.30 | Hmmhesays | ni2 5ess, dms, who cares |
17:05.47 | gaspiz | jedril: running Deadagi... script.php returned 0 |
17:05.51 | stack_ | jsharp, ok that's just one of the options on the page. For example "Line Coding: D4/AMI or B8ZS" |
17:06.01 | jsharp | B8ZS |
17:06.07 | Hmmhesays | esf |
17:06.07 | gaspiz | jedril: actually not calling the php script |
17:06.13 | stack_ | Framing: SF or ESF |
17:06.16 | jsharp | ESF |
17:06.25 | stack_ | Wiring 2 wire or 4 wire? |
17:06.29 | jsharp | 4 |
17:06.57 | stack_ | Start Signal: Loop, Ground or E&M? |
17:07.02 | Hmmhesays | none |
17:07.11 | jsharp | Uh. That's not a PRI if they're asking for that kind of signalling. |
17:07.18 | Hmmhesays | thats for a cas/robbed bit t1 |
17:07.24 | jsharp | Yeah, what he said. |
17:07.27 | stack_ | Sorry, that's down in the Trunk Group Information |
17:08.06 | RoyK | ~seen zoa |
17:08.31 | jbot | zoa <n=kkk@pirus.securax.be> was last seen on IRC in channel #asterisk, 5d 25m 14s ago, saying: 'it looks kinda suspicious :p'. |
17:08.41 | stack_ | what's SS7? Sorry, my boss is doing the ordering, he just has me do the setup... |
17:08.56 | jsharp | ~ss7 |
17:08.58 | jbot | ss7 is probably can be used in conjunction with ss7box.com - see the website. |
17:09.34 | jsharp | Don't worry about ss7. Unless you're going to be a carrier, you'll not need it. |
17:09.44 | stack_ | k |
17:10.14 | Jedirl | ss7 with asterisk |
17:10.17 | Jedirl | anyone? :D |
17:10.21 | stack_ | Jack Type: Smartjack or RJ48C |
17:10.30 | jsharp | I'd recommend Smartjack. |
17:10.43 | jsharp | So they can do loop testing in the event the circuit goes down. |
17:10.50 | stack_ | k |
17:11.13 | stack_ | Under ISDN Details: Primary D channel: NFAS or FAS |
17:11.14 | Jedirl | anyone made chan_ss7 work? |
17:11.26 | jsharp | Are you just getting 1 PRI? |
17:11.29 | *** join/#asterisk Holos (n=asdf@204.101.26.106) |
17:11.34 | *** join/#asterisk a1fa (n=a1fa@207.210.210.202) |
17:11.35 | a1fa | hm |
17:11.37 | jsharp | Or will you grow to lots more? |
17:11.37 | a1fa | this is so fucked up |
17:11.40 | a1fa | Total SIP Message Sent: 19 |
17:11.47 | a1fa | i dont see any SIP msgs reach the server |
17:11.50 | a1fa | which makes me believe |
17:11.55 | a1fa | the ISP is blocking SIP traffic |
17:12.05 | a1fa | i have sip debug ip |
17:12.08 | a1fa | enabled |
17:12.12 | a1fa | and i dont see any |
17:12.13 | stack_ | jsharp: we'll be using about half of it now and plan to grow completely into it with a year |
17:12.17 | a1fa | sip msgs coming in |
17:12.21 | jsharp | Okay. Go with FAS, then. |
17:12.31 | jsharp | You'd go with NFAS if you were bringing in a bunch of PRIs. |
17:12.42 | stack_ | ok |
17:12.45 | a1fa | Total SIP Message Sent: 56 |
17:12.47 | a1fa | wow |
17:12.50 | a1fa | this is insane |
17:12.51 | Holos | I have asterisk-sounds-1.2.1 and vm-options.gsm says "press 4 to change your password" but 4 is to enter a temporary greeting. I checked sounds-1.2.5 and it seems to be the same sound file. Anyone have a corrected file? |
17:13.08 | stack_ | So I imagine this is PRI over T1, do I need to worry about these T1 options? |
17:13.36 | jsharp | Probably not. |
17:14.10 | a1fa | anybody ever experienced same problems? |
17:14.25 | stack_ | jsharp: awesome, thanks... I'll probably have more quetions but this should be all I need for now |
17:14.38 | jsharp | K |
17:15.02 | a1fa | this is insane |
17:15.08 | a1fa | god damn GRANDSTREAM |
17:15.13 | a1fa | and their nasty ass interface |
17:16.40 | a1fa | so much for that |
17:16.44 | a1fa | <-- idiot |
17:16.56 | a1fa | i downed a wan interface on the router |
17:17.00 | a1fa | oh well |
17:17.04 | jsharp | Whups. |
17:17.08 | a1fa | upsy :P |
17:17.20 | a1fa | fuck |
17:17.31 | a1fa | its not coming back up |
17:17.33 | a1fa | oh well |
17:18.03 | austinnichols101 | a1fa: potty mouth :) |
17:18.24 | a1fa | ah yes |
17:18.29 | a1fa | austinnichols101 : i am still having a same issue |
17:18.35 | a1fa | it works for 5 minutes |
17:18.39 | a1fa | then asterisk stops recieving packets |
17:18.40 | jbalcomb | anyone know why i see (oui Unknown) so much in my tcpdump of phone traffic? |
17:19.02 | *** join/#asterisk chrismog_ (n=chrismog@mog.traxtech.net) |
17:19.57 | Holos | How do I disable my temporary message in Voicemail? |
17:20.23 | *** join/#asterisk r_evolution (i=_evoluti@208.251.203.246) |
17:21.06 | jbalcomb | Holos: go into the menu to record a temporary greeting and press 2 to erase it |
17:21.39 | *** join/#asterisk nDuff (n=ccd@64.128.31.220) |
17:21.55 | jbalcomb | Holos: -> VM -> 0 -> 4 -> 2 |
17:22.31 | Holos | jbalcomb: For some reason the newer sounds didn't get installed and I'm missing all the prompts for that. Thanks, |
17:22.52 | *** join/#asterisk Trazz (n=traderz@65.114.86.29) |
17:22.53 | Trazz | i am calling out and my call is cut off before i can get to the voice mail.. what setting do i need to change to allow the call to ring longer? |
17:23.18 | jbalcomb | Holos thats odd. im running sound 1.2.1 |
17:23.27 | Hmmhesays | the timeout of your dial command |
17:23.59 | nDuff | Is there a conventional way to set caller ID strings for folks with non-DID numbers? I'm trying to figure out how to format the caller ID string in sip.conf for users who are accessible only via extension. I'm currently pondering something like ("Person Name" <2345678901+EXTN>) |
17:23.59 | Holos | jbalcomb: Me too.. I just did a make install in the sounds-1.2.1 file and it's still missing. No worries, I'm switching to my spare server (1.2.5) this week anyways, I just copied over the missing files. |
17:24.12 | asterboy | Dial(Technology/resource[&Tech2/resource2...][|timeout][|options]) |
17:24.55 | jbalcomb | Holos right on. i'm working on setting up a 1.2.5 system right now. got my new dell 2850. :) |
17:25.49 | a1fa | fuck |
17:25.51 | a1fa | ;( |
17:25.53 | a1fa | not to self |
17:25.57 | a1fa | dont bring down wan interface |
17:26.14 | *** join/#asterisk Winkie (n=urmom@cpc2-stre1-6-0-cust119.bagu.cable.ntl.com) |
17:26.24 | austinnichols101 | a1fa: want to run through it again? |
17:26.33 | a1fa | i cant |
17:26.36 | a1fa | i am cut off |
17:26.45 | a1fa | i have to wait a few hours |
17:26.46 | Winkie | sup gents, i'm having problems tracking a call that goes into a queue, the manager events returned hold no reference to what actual call is ringing the agents |
17:26.55 | Winkie | anyone got a clue how to better track queue events? |
17:28.23 | a1fa | Flash Operator, Winkie |
17:29.52 | Winkie | a1fa: i'm sorry? |
17:30.08 | Winkie | ah right i see |
17:30.25 | Winkie | that's only going to pull information from the manager interface though, and there's a critical piece of information we don't get that i need |
17:32.36 | Trazz | where can i change the timeout on the dial out command? |
17:32.49 | backblue | anyone know anykind of alarms in zaptel? i need notification if one span its down. anyone knows how? |
17:32.58 | backblue | layer1 and layer2 notifications |
17:33.00 | Zeeek | trazz in the Dial() parameters |
17:33.04 | *** join/#asterisk mogorman (n=mogorman@gateway.digium.com) |
17:33.46 | backblue | hi mogorman |
17:34.02 | mogorman | hi blackblue |
17:34.40 | Trazz | thanks |
17:35.35 | tuxinator_linux | mog has so many names |
17:35.48 | mog_work | just mog |
17:36.55 | asterboy | never forget the value of SIP DEBUG. |
17:37.03 | Winkie | backblue: i think you're just looking for yellow / red alarms? |
17:37.43 | asterboy | my sip dial()s are not working cause of a 407 Proxy authentication required. |
17:37.51 | asterboy | now that does not show in the manual. |
17:38.10 | *** join/#asterisk RoyK (n=roy@host-81-191-147-248.bluecom.no) |
17:39.08 | austinnichols101 | freePBX is really starting to look nice! |
17:40.06 | RoyK | .... .. |
17:40.37 | backblue | Winkie: i dont get yellow / red alarms when one span its down. |
17:40.58 | backblue | austinnichols101: why? |
17:42.13 | RoyK | why? |
17:42.36 | austinnichols101 | backblue: the modules stuff. They just added a disa module between 2.0.0 and 2.0.1. It's very nice to have disa as a destination and it's very cool to see how it can be extended |
17:43.28 | austinnichols101 | I expect to see a lot of add-in modules |
17:43.38 | *** part/#asterisk Utah_Dave (n=boucha@0-1pool149-100.nas31.salt-lake-city1.ut.us.da.qwest.net) |
17:44.40 | backblue | austinnichols101: pbx without disa, its not pbx! :o |
17:44.59 | austinnichols101 | I had done my own, but it's nice to be able to just set them up from the GUI |
17:45.20 | austinnichols101 | one less thing to bother with |
17:45.49 | austinnichols101 | disa + nextel 'free' incoming = sticking it to the man, bigtime! |
17:47.14 | a1fa | incoming minutes should always be free |
17:47.18 | a1fa | you crazy amerikanz! |
17:48.02 | austinnichols101 | agreed |
17:49.34 | a1fa | i love european prepaid gsm cellphones |
17:49.41 | a1fa | free incoming calls |
17:49.59 | a1fa | c0.05/minute |
17:50.04 | a1fa | awezome! |
17:50.09 | austinnichols101 | a1fa: at least we can keep our voip phones online for more than 5 minutes at a time :) |
17:50.16 | a1fa | LOL |
17:50.17 | a1fa | f0X u! |
17:50.30 | austinnichols101 | lol |
17:50.37 | Jedirl | uhm |
17:50.42 | Jedirl | in USA you pay for incoming calls? |
17:51.11 | austinnichols101 | yup |
17:51.14 | austinnichols101 | most providers |
17:51.18 | Juggie | on cell phone yes, depending on the provider |
17:51.33 | austinnichols101 | you actually have to sign up for a 'free' incoming plan |
17:51.37 | *** join/#asterisk umay (n=chris@70-101-61-50.dsl2-plymouth.roc.ny.frontiernet.net) |
17:51.48 | austinnichols101 | instead of 'caller pays', we're on 'everyone pays' |
17:52.40 | Jedirl | uhmmm |
17:52.41 | Jedirl | jejeje |
17:52.48 | justinu | welcome to the USA |
17:52.54 | justinu | where everyone gets fucked |
17:53.06 | austinnichols101 | justinu: welcome to the USA, please bend over |
17:53.29 | LoRez | it's typical of capitalism. you don't vote with your dollars so they keep bending you over. |
17:53.50 | asterboy | like brokeback mountain? |
17:53.57 | asterboy | except no spit |
17:54.13 | austinnichols101 | yes, just like brokeback but with cellphones |
17:54.27 | *** join/#asterisk jpm_SD (n=jpm@207-40-115-38.sugardog.com) |
17:54.53 | justinu | <PROTECTED> |
17:54.55 | Jedirl | hehehehe |
17:55.25 | *** join/#asterisk generalhan (i=general_@ip67-90-64-2.z64-90-67.customer.algx.net) |
17:55.33 | generalhan | whats going on everyone !? |
17:56.10 | iDunno | well, th earth is rotating at approximately a 24 hour period around it's axis... |
17:56.18 | justinu | Lorez: you're right. what's important here is what ringtone you have, or whether your phone has enough cameras |
17:56.21 | iDunno | and a 1 year period around the sun |
17:56.23 | justinu | not price/service |
17:56.27 | generalhan | apporx |
17:56.35 | generalhan | lol |
17:56.55 | *** join/#asterisk ToTo (n=ToTo@host62-142.pool874.interbusiness.it) |
17:57.09 | austinnichols101 | 'friends and family' and 'IN' plans must look ridiculous from outside the US |
17:57.14 | LoRez | justinu: I'm all too familiar with the situation :) luckily I haven't had to pay for cell service for the past 11 years. |
17:57.27 | justinu | lorez: my work has been taking care of my bill for a while also |
17:57.33 | generalhan | my boss is looking for a wireless conference room phone to accomidate 12 users around the table; anyone had any good experiences with any ? or heard anything good about some ? |
17:57.46 | justinu | wireless conference room phone? never heard of such a thing |
17:57.51 | generalhan | bah |
17:57.53 | justinu | i can suggest the polycom soundstations tho |
17:57.58 | austinnichols101 | polycom |
17:58.02 | justinu | perhaps w/ an ATA + Wifi Bridge |
17:58.06 | justinu | *shrug* |
17:58.09 | generalhan | hmm |
17:58.51 | stack_ | jsharp: you still around? |
17:58.58 | justinu | lorez: in the US, they call people "consumers" |
17:59.02 | justinu | not customers |
17:59.05 | justinu | it's pretty damn sad |
17:59.11 | justinu | no one cares tho |
17:59.25 | *** join/#asterisk malverian[work] (n=pawalls@pawalls.teamgleim.com) |
17:59.31 | generalhan | justinu: http://www.officeworld.com/Worlds-Biggest-Selection/PCY220007880001/05Q4/ something like this ? |
17:59.46 | salviadud | justinu |
17:59.48 | austinnichols101 | generalhan: polycom makes a voip conference phone |
17:59.51 | salviadud | did you like Fight Club? |
17:59.52 | generalhan | i just want to be sure that the quality isnt gonna suck on that phone because of the wireless vs wired |
17:59.53 | justinu | interesting, didn't know they made a wireless product |
17:59.54 | *** join/#asterisk Zeeek_ (n=IceChat7@80.125.80.38) |
17:59.56 | austinnichols101 | just add wireless bridge and stir |
17:59.59 | justinu | salviadud: yeah, great movie |
18:00.14 | justinu | generalhan: polycom makes great stuff, you can't go wrong |
18:00.19 | salviadud | you know how i can take out some audio samples from a DVD? |
18:00.27 | salviadud | could mplayer do such a thing? |
18:00.35 | *** join/#asterisk Netgeeks (n=chris@68-185-24-8.static.mdfd.or.charter.com) |
18:00.36 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
18:00.36 | *** mode/#asterisk [+o anthm] by ChanServ |
18:00.38 | justinu | i dunno... you need to rip the audio tracks off the disk |
18:00.46 | justinu | but i'm not up on the latest techniques |
18:01.16 | stack_ | We are getting a PRI over T1 setup and they are asking how we want the T1 setup. The last option is "Dialtone" and it can be either Precise, SCC or None... anyone know the answer :) |
18:01.26 | justinu | never heard of that |
18:01.36 | justinu | why the hell would you need dialtone over a PRI? |
18:01.41 | justinu | i'd opt for none |
18:02.01 | stack_ | Oh, sorry, not the T1, this is the trunk group... |
18:02.10 | austinnichols101 | stack: fyi: my carrier could only deliver callerid NUMBER + NAME over NI2. NI1 could only give callerID number |
18:02.12 | justinu | still |
18:02.31 | justinu | i don't know why you'd need dialtone... unless you were doing centrex or something |
18:03.24 | stack_ | the other options are for Signal Protocol, Outpulse and Start Signal, if that helps (you'll have to pardon me, I'm pretty green when it comes to Telco stuff, I'm just the UNIX need here) |
18:03.46 | justinu | all that crap isn't applicable to a PRI |
18:03.51 | justinu | it's for CAS T1s |
18:03.52 | Luke-Jr | iDunno: the Earth does not rotate... |
18:04.21 | *** part/#asterisk Zeeek_ (n=IceChat7@80.125.80.38) |
18:04.24 | *** join/#asterisk Zeeek_ (n=IceChat7@80.125.80.38) |
18:04.59 | stack_ | justinu: Im just a little confused... what's the point of the dialtone on the app? wouldn't we need one? |
18:05.48 | *** join/#asterisk ToTo (n=ToTo@host62-142.pool874.interbusiness.it) |
18:05.48 | justinu | no |
18:06.03 | justinu | no need for dialtone on a PRI |
18:06.03 | stack_ | justinu: why is that? |
18:06.07 | stack_ | ok |
18:06.12 | justinu | because it uses enbloc dialing |
18:06.18 | justinu | and inband tones are legacy on ISDN |
18:06.31 | asterboy | holy crap, asterisk docs search won't allow "sip proxy" cause its shorter than 5 characters |
18:06.41 | stack_ | justinu: ok, gotcha... this stuff make my head hurt a little but i'm getting it, thanks |
18:07.32 | *** join/#asterisk cripito (n=ncripito@ip67-154-143-190.z143-154-67.customer.algx.net) |
18:07.37 | cripito | hi |
18:09.03 | *** join/#asterisk stkn (n=stkn@gentoo/developer/pdpc.active.stkn) |
18:09.06 | Luke-Jr | hi |
18:12.01 | justinu | stack: the telco isn't helping the situation by asking you for settings that don't apply to your circuit type! |
18:12.56 | *** join/#asterisk tdonahue (n=tdonahue@208.51.101.201) |
18:16.15 | *** join/#asterisk stoffell (n=stoffell@d51A4D1F8.access.telenet.be) |
18:18.02 | jpm_SD | I thought that is what the Telco was for - asking irrelevant questions? at least that is what Sprint does. |
18:18.58 | Luke-Jr | they're always looking for some excuse to drop or charge for support |
18:19.39 | Luke-Jr | my latest one was "oh, we don't support Asterisk" followed soon after by "we don't support a manually configured PAP2 either" |
18:20.01 | Luke-Jr | of course, I was never getting a single packet for calls, so my end didn't matter at all |
18:20.23 | Luke-Jr | in fact, I'm still not... iConnectHere/deltaThree hasn't managed to fix my account yet |
18:20.39 | Winkie | damnit, no queue name is ever passed back |
18:20.42 | Luke-Jr | (needless to say, I'm planning to port my number away from them) |
18:20.51 | Winkie | i guess nobody making app_queue ever thought anyone would use > 1 queue >:( |
18:21.07 | Luke-Jr | so modify it |
18:21.11 | Luke-Jr | you've got source |
18:21.16 | Winkie | i am doing, but it's still very annoying |
18:21.53 | Luke-Jr | last time I went to modify something like that, I found I didn't need to: it already had support for what I wanted, just completely undocumented |
18:22.00 | Winkie | haha what was that? |
18:22.04 | *** join/#asterisk spunz_ (n=spunz@h081217096096.dyn.cm.kabsi.at) |
18:22.07 | *** join/#asterisk bweschke (n=bweschke@c-68-36-51-213.hsd1.nj.comcast.net) |
18:22.28 | Luke-Jr | Winkie: 'jump' statement in AEL |
18:23.11 | jpm_SD | Luke-Jr -- Making the World a better place. |
18:23.17 | Luke-Jr | lol |
18:23.46 | Luke-Jr | well, I could have gone farther and fixed the bugs in AEL instead of just using old extensions.conf for the two macros that the bugs broke... =p |
18:24.13 | Winkie | i haven't found a need to use AEL yet, but i'm finding it hard to work with asterisk's call reporting, tracking calls is quite tricky |
18:24.23 | Luke-Jr | AEL isn't a need; it's a want =p |
18:24.23 | *** join/#asterisk SexyKen (n=Ken@c-24-5-129-114.hsd1.ca.comcast.net) |
18:24.41 | SexyKen | Hey guys -- Is call blocking (IE: *67) possible with VOIP |
18:24.42 | Winkie | at the moment to get the queue name i have to check up on QueueMemberStatus and AgentCalleds, when AgentCalled should say the queue name surely |
18:24.54 | *** part/#asterisk rjt69 (n=rjt@wsip-68-15-224-183.om.om.cox.net) |
18:25.33 | mutilator | with a name like SexyKen whats it matter? |
18:25.58 | stoffell | Luke-Jr, you mean AEL is good/better to create complete dialplans, or only partially ? |
18:26.11 | Luke-Jr | stoffell: when it works, it's better |
18:26.37 | jpm_SD | The question would then be -- How often does it "work" |
18:26.58 | stoffell | Luke-Jr, okay, looking into AEL then :) |
18:26.59 | Luke-Jr | stoffell: nice not to have to repeat the extention every line and keep track of priorities |
18:26.59 | SexyKen | My name shouldn't matter. But some people are obviously lacking a certain ability to respond, I suppose. |
18:27.17 | Luke-Jr | not to mention switch statements and the like |
18:27.18 | justinu | lol |
18:27.35 | stoffell | oh, i see Luke-Jr, good tip |
18:27.41 | justinu | with statements like that... those who can respond probably won't |
18:27.48 | Luke-Jr | jpm_SD: for-loops are broken in my installed version |
18:27.50 | stoffell | justinu, lol |
18:28.40 | Luke-Jr | and functions need to use | instead of commas |
18:29.07 | *** join/#asterisk sssk (n=sssk@s55935276.adsl.wanadoo.nl) |
18:29.15 | Luke-Jr | the only stuff that broke for me really was my str_replace and iterator functions |
18:29.22 | stoffell | Luke-Jr, it seems very well documented on voip-info, nice |
18:29.49 | Luke-Jr | s/functions/macros |
18:30.16 | SexyKen | Is caller ID blocking possible with Asterisk? |
18:30.19 | Winkie | i wonder how many people actually use asterisk's CDR for billing |
18:30.32 | Luke-Jr | Winkie: what else would you use? |
18:30.35 | Nugget | Just about anything is possible with Asterisk. |
18:30.36 | Luke-Jr | SexyKen: yes |
18:30.48 | Luke-Jr | SexyKen: voice obfuscation is probably possible too if you want that |
18:31.51 | mutilator | make SexyKen's voice sound sexy? |
18:31.51 | SexyKen | Luke-Jr, what would the process be? I know how to set my CID< and I know if I dont set the CID that my provider will use theirs |
18:31.51 | Nugget | rigging up asterisk to punch people in the face if they ask questions that are covered by the faq is even possible. just a Simple Matter of Programming(tm). |
18:31.52 | Luke-Jr | SexyKen: don't use such a provider, or set a fake CID |
18:31.52 | Winkie | Luke-Jr: well CDR provides virtually no information, so i'm using the manager interface + CDR events, but even then the information provides is woefully inadequate |
18:31.53 | Luke-Jr | to actually block it, there's some setting for it |
18:31.57 | Winkie | there's no proper way to track calls and to be honest the syntax is a little iffy |
18:31.58 | justinu | or set the presentation indicator to privacy |
18:32.08 | justinu | asterisk CDRs suck |
18:32.09 | Winkie | plus who knows i could nicely crash asterisk by deadlocking the manager interface D: |
18:32.21 | Luke-Jr | Winkie: CDR provides account, destination number and seconds connected-- all the info needed |
18:32.23 | stoffell | Winkie, we also use manager to track calls in an own cdr |
18:32.25 | Winkie | justinu: totally, we're trying to hack stuff up with a bunch of ResetCDRs and ForkCDRs etc |
18:32.31 | justinu | yep, it's a big joke |
18:32.37 | Winkie | Luke-Jr: if you think that's all the info needed you've never worked anywhere, sry :( |
18:32.42 | justinu | agreed |
18:32.50 | Luke-Jr | no telcos anyway |
18:32.59 | Winkie | nowhere with more than 2 phones |
18:33.10 | Winkie | i'm having to hack things in to do basic tracking of calls in queues |
18:33.11 | justinu | lol |
18:33.15 | Luke-Jr | just because a place has phones doesn't mean people pay for using them =p |
18:33.16 | medusaXX | does asterisk support video with sip? |
18:33.21 | Winkie | yes |
18:33.30 | medusaXX | nice |
18:33.35 | medusaXX | do you know some clients for windows? |
18:33.38 | Winkie | Luke-Jr: tell me how account, destination number and seconds connected is enough for tracking transfers :) |
18:33.44 | Winkie | i don't use windows i'm afraid |
18:34.05 | justinu | eyebeam is a SIP client w/ video |
18:34.06 | Luke-Jr | Winkie: ah, didn't consider transfers; maybe call #? |
18:34.19 | Winkie | Luke-Jr: queues |
18:34.23 | Winkie | agents |
18:34.24 | Winkie | etc |
18:34.26 | Winkie | I can go on and on |
18:34.39 | justinu | freeswitch will have much better CDRs |
18:34.42 | Winkie | there's no guarenteed way to check on how Agent/1001 was transformed into Local/whatever@wherever etc |
18:34.44 | justinu | since it tracks the entire history of a call |
18:34.56 | Winkie | meh |
18:35.02 | Winkie | i'm more interested in asterisk fixing theirs |
18:35.09 | justinu | meh - don't hold your breath |
18:35.24 | *** join/#asterisk noky (n=Noky@200.69.211.18) |
18:35.26 | Luke-Jr | Winkie: sounds like a hack.. Agent/1001 should just be connected to the call, not transformed |
18:35.31 | noky | hi |
18:35.31 | *** join/#asterisk Qwell[] (i=north@unaffiliated/qwell) |
18:35.35 | Winkie | there's a fair amount more coders now though |
18:35.45 | Winkie | Luke-Jr: welcome to chan_agent and AgentCallbackLogin |
18:35.58 | justinu | there's too many political problems, imo |
18:36.19 | Winkie | possibly |
18:36.29 | Winkie | I need to submit a couple of patches but i'm too lazy to figure out how to do it yet |
18:36.32 | justinu | i'm not sure they'd even admit that the CDR structure is lacking |
18:36.46 | *** join/#asterisk MGSsancho (n=user@adsl-67-127-164-145.dsl.irvnca.pacbell.net) |
18:36.51 | *** join/#asterisk brodiem (i=1000@cpe-66-69-222-36.austin.res.rr.com) |
18:37.02 | Luke-Jr | Winkie: IIRC, first you need to sign a copyright release form or such |
18:37.17 | Luke-Jr | hence why I avoid edting * code =p |
18:37.25 | justinu | luke-jr: agreed |
18:37.30 | Winkie | Luke-Jr: any idea why? digium retains copyright on your code or something? |
18:37.37 | Luke-Jr | Winkie: yeah, I think so |
18:37.40 | justinu | sign away your first born |
18:37.42 | Winkie | well isn't that infuriating |
18:37.43 | mog_work | digium does not |
18:37.49 | Nugget | digium needs the ability to release your code under their commercial license, not the gpl. |
18:37.53 | Winkie | ah i see |
18:37.58 | mog_work | you grant digium a license |
18:38.01 | mog_work | to use your code |
18:38.01 | Nugget | they don't require you to hand over copyright, just licensing flexibility. |
18:38.02 | *** join/#asterisk joe4319 (n=yjoe@65.222.176.9) |
18:38.04 | Luke-Jr | Nugget: maybe I don't approve of non-GPL'd usage? |
18:38.05 | mog_work | you retain copyright |
18:38.06 | Winkie | that's fine then |
18:38.18 | Winkie | Luke-Jr: then that's your problem |
18:38.18 | Nugget | Luke-Jr: then don't contribute to asterisk. |
18:38.18 | Winkie | (and you're an idiot ;) ) |
18:38.24 | Luke-Jr | Winkie: not at all, it's *'s problem |
18:38.29 | justinu | heh |
18:38.31 | Winkie | Luke-Jr: no it's not |
18:38.35 | Winkie | they have picked their license |
18:38.36 | stoffell | Luke-Jr, if everyone would think that way.. :( |
18:38.42 | jpm_SD | This could go on all day... |
18:38.50 | Luke-Jr | stoffell: now I know why FreePBX was forked, I think |
18:38.50 | mog_work | yeah |
18:38.51 | Winkie | it could but it won't because i'll just start insulting everyone 8) |
18:38.54 | mog_work | it tends to as well |
18:38.57 | mog_work | its openpbx |
18:38.58 | stoffell | indeed, on to some practical stuff? dundi experts here? ;) |
18:39.00 | stoffell | Luke-Jr, indeed.. |
18:39.01 | mog_work | freepbx is amp |
18:39.02 | justinu | openpbx != freepbx |
18:39.04 | Luke-Jr | ah |
18:39.14 | mog_work | and its not even openpbx |
18:39.18 | mog_work | as that is a perl pbx |
18:39.21 | mog_work | its openpbx.org |
18:39.25 | tsume | flash is dumb, heh |
18:39.31 | mog_work | but i digress |
18:39.34 | Luke-Jr | so does openpbx actually have development? |
18:39.40 | mog_work | what do you need to know about dundi stoffell |
18:39.41 | justinu | a bit |
18:39.47 | mog_work | they have mostly kept up with bugs |
18:39.47 | *** join/#asterisk MGSsancho (n=user@adsl-67-127-164-145.dsl.irvnca.pacbell.net) |
18:39.55 | mog_work | but there are a few things there that are not in asterisk |
18:39.57 | mog_work | and vice versa |
18:40.07 | Luke-Jr | can't Asterisk changes be merged directly? |
18:40.13 | stoffell | mog_work, i'm just reading up on it, want to get a test-setup working this week |
18:40.14 | mog_work | no |
18:40.19 | mog_work | as they changed api names |
18:40.20 | mog_work | etc |
18:40.23 | Luke-Jr | oh |
18:40.30 | Luke-Jr | that was a bit dumb :\ |
18:40.31 | stoffell | mog_work, any good references besides TFOT and dundi.com ? |
18:40.31 | brodiem | is there ANY way I can get firmware updates from Polycom (IP301)? It seems crazy that I have to become a certified reseller to get them |
18:40.36 | justinu | that was a big mistake |
18:40.37 | mog_work | not that i know of |
18:40.43 | mog_work | i have learned from doing |
18:40.47 | stoffell | brodiem, through the guys that selled them to you.. |
18:40.50 | mog_work | voip-info has an example as well |
18:40.50 | justinu | brodiem: talk to whomever sold you the phone |
18:40.56 | *** join/#asterisk bsdfreak (n=alex@breakbeats.okkernoot.net) |
18:40.56 | joe4319 | Can anyone help with a problem dialing out on a TDM400P? I get out 50% of the time. |
18:41.06 | mog_work | add some waits jo |
18:41.11 | mog_work | joe4319, |
18:41.29 | mog_work | instead of dial(zap/g1/${EXTEN}) |
18:41.31 | jpm_SD | joe4319, what happens the other 50%? |
18:41.40 | mog_work | do dial(zap/g1/www${EXTEN}) |
18:41.45 | joe4319 | I get a Verizon message that says enter a calling card |
18:41.47 | *** join/#asterisk brodiem (i=1000@cpe-66-69-222-36.austin.res.rr.com) |
18:41.48 | stoffell | mog_work, i can understand, but do the phones still 'need' their own fixed server then? |
18:41.49 | mog_work | that will allow for 300 ms wait before it dials |
18:41.52 | ian_k | Do we sell the octasic cards yet? |
18:42.00 | mog_work | no there are ways around |
18:42.15 | brodiem | the guys I bought it from (antonline.com) just referred me to Polycom's resource center page.. they said they are just "online" resellers |
18:42.26 | justinu | lol |
18:42.26 | *** part/#asterisk ian_k (n=ian@gateway.digium.com) |
18:42.39 | justinu | you could complain to polycom |
18:42.43 | justinu | about them not being authorized reseller |
18:42.49 | brodiem | talking to polycom is like talking to a wall |
18:43.03 | jpm_SD | brodiem, but less responsive. |
18:43.04 | brodiem | but anyway, they told me that their resllers weren't allowed to give our the firmware either |
18:43.10 | justinu | that's not true |
18:43.14 | Luke-Jr | justinu: um... you don't need to be authorized to resell stuff =p |
18:44.01 | asterboy | just look at ebay |
18:44.17 | brodiem | anyone know who sells them at a nice price and will offer the firmware? |
18:44.24 | brodiem | I need about 20 of them |
18:44.34 | Luke-Jr | I got my PAP2-NA fourth-hand |
18:44.35 | brodiem | antonline sells them w/ POE cable for $117 |
18:44.36 | stoffell | mog_work, you have any tips on recources on the complete "cloud-alike" way to setup *, dundi and sip phones? |
18:44.53 | brodiem | that's the IP301 btw |
18:45.04 | mog_work | stoffell, dundi + iax +regexten + roundrobin dns = HAPPY |
18:45.09 | joe4319 | Thx mog_work. I think that did the trick. |
18:45.13 | mog_work | but aside from that i dont have any real tips |
18:45.15 | *** join/#asterisk Shoragan (n=shoragan@d072.apm.etc.tu-bs.de) |
18:45.30 | stoffell | mog_work, okay, it's start! thanks! |
18:45.30 | Luke-Jr | mog_work: does anyone actually use dundi worth setting it up? =p |
18:45.33 | mog_work | yup joe4319 there is no way for it to know if it has gotten dial tone from telco yet so it just starts dialing |
18:45.49 | mog_work | umm yes Luke-Jr like you wouldnt even know |
18:46.01 | Luke-Jr | mog_work: I wouldn't. I'm not part of the dundi network |
18:46.08 | joe4319 | Why can't it detect a dial tone? |
18:46.15 | mog_work | there is much more to dundi than e164 network |
18:46.15 | stoffell | i also believe 'many many' people are using dundi.. (and not only public.....) |
18:46.23 | Luke-Jr | I just know that eg, enum had virtually no benefit |
18:46.23 | mog_work | exactly stoffell |
18:46.33 | mog_work | thats where it is huge |
18:46.44 | *** join/#asterisk chapeaurouge (n=chap@user-85-201-82-146.tvcablenet.be) |
18:46.44 | hardwire | mogmogmog |
18:46.46 | stoffell | correct me if I'm wrong, but the possibilities this has in an enterprise ... oh boy.. |
18:46.51 | *** join/#asterisk epablo (n=epablo@200.109.73.215) |
18:46.53 | mog_work | joe4319, it just doesnt attempt to , you could probably write a soft dsp to do it |
18:46.58 | mog_work | yeah stoffell has the idea |
18:47.06 | mog_work | hardwirehardwirehardwire |
18:47.19 | stoffell | lol |
18:47.37 | epablo | How to i get the asterisk manager to showme all events? or to sendme all events' |
18:47.42 | a1fa | ;( |
18:47.45 | stoffell | avaya can only "drea" of 1 virtual pbx :D |
18:47.49 | stoffell | "dream" |
18:48.06 | mog_work | epablo, you can set the events manager sees in manager.conf |
18:48.10 | brodiem | ok how about this, can someone give me the sip firmware for polycom phones? :) |
18:48.18 | noky | hi, i'm trying to set asterisk's cdr with MySQL database, i installed the asterisk-addon and I configurate cdr_mysql.conf. I create a mysql database with the table 'cdr'... But it doesn't work... |
18:48.23 | noky | 2006-03-20 15:14:28 ERROR[4995]: cdr_addon_mysql.c:438 my_load_module: Failed to connect to mysql database asterisk on 127.0.0.1. |
18:48.26 | noky | any ideA? |
18:48.33 | stoffell | brodiem, buy the 501, i've got a release of that one :) |
18:48.34 | jsharp | Is mysql started? |
18:48.39 | stoffell | brodiem, doesn't the phone has sip installed already? |
18:48.48 | epablo | mog_work: I've been looking at it.. but I cant find how to make the call |
18:48.48 | jsharp | Can you connect to it with the mysql client? |
18:48.52 | mog_work | well it obviously isnt connecting noky you proably have it misconfigured |
18:48.55 | noky | yes |
18:49.03 | noky | i have the table sip_buddies working OK |
18:49.04 | mog_work | epablo, ? |
18:49.07 | brodiem | stoffell I have 1.6.2 now, but I wanted to update it and have the ability to keep it up to date |
18:49.09 | mog_work | originate? |
18:49.16 | noky | mysql is started |
18:49.20 | *** join/#asterisk livesNbox (n=livesNbo@68-76-129-3.ded.ameritech.net) |
18:49.31 | mog_work | you can turn on debug noky to see connect attempt |
18:49.37 | mog_work | but if i had to guess its auth error |
18:49.48 | epablo | mog_work: I set up a user.. and can connect with telnet, but I need to se events related to SIP register |
18:50.15 | livesNbox | Hey guys -- I'm trying to figure out how to let agents log into the queues -- I am using the AgentCallbackLogin command and it asks me for my agent number followed by # sign -- so I put it in.. and then nothing happens.. about 10 seconds later, the line hangs up. |
18:50.37 | mog_work | its there epablo you just need to parse |
18:50.45 | noky | 2006-03-20 15:18:19 WARNING[4995]: res_config_mysql.c:553 parse_config: MySQL RealTime: No database socket found, using '/tmp/mysql.sock' as default. |
18:51.00 | noky | my mysql.sock is in /tmp/mysql.sock :( |
18:51.07 | epablo | mog_work: but it doesn't come out on the screen there is nothing to parse :S |
18:51.48 | mog_work | well sip registration might not have a manager event associated with it |
18:51.53 | mog_work | to which you will need to make on |
18:51.54 | mog_work | e |
18:51.57 | mog_work | but i thought it did |
18:51.57 | *** join/#asterisk Gunnar (n=gunnar@bkkb-gw.bitcon.no) |
18:53.05 | *** join/#asterisk dja (n=alden@zaphod.math.ohio-state.edu) |
18:54.00 | epablo | ok.. let me test other events |
18:54.22 | epablo | mog_work: thanks |
18:55.23 | mog_work | no prob |
18:55.30 | *** part/#asterisk epablo (n=epablo@200.109.73.215) |
18:59.32 | *** join/#asterisk glm2k (n=glm@rrcs-24-199-11-46.west.biz.rr.com) |
18:59.54 | SexyKen | Hey guys -- how would I change this: http://pastebin.ca/46357 to include a *67 ability to block caller id on the call? |
18:59.58 | a1fa | keep-allive setting to :5s |
19:01.41 | tzanger | SexyKen: you'd have |
19:02.05 | tzanger | exten => *67,1,SetDB(BLOCKCID=true) and *67,2,Goto(s,1) |
19:02.31 | tzanger | then change s,2, to gotoif(GetDB(BLOCKCID)=true?s,4) |
19:02.35 | tzanger | and make the current 2 a 3 |
19:02.48 | tzanger | you'd have to put a little magic in there though to erase the db key afterward |
19:02.51 | SexyKen | Well -- I dont mean to sound...stupid...but I have no idea what you just said? |
19:03.25 | tzanger | SexyKen: or have something like _*67.,1,SetDb(BLOCKCID=true) and continue on, erasing the key afterward |
19:03.50 | *** join/#asterisk saftsack (n=oliver@IP-213188106101.dialin.heagmedianet.de) |
19:03.54 | SexyKen | Any chance you could through that into a pastebin the way it should look? I'm sorry if I'm difficult, I am speaking a foreign language with asterisk tho :-( |
19:04.24 | saftsack | hi, im searching for moh music. does anyone of you know a good url? |
19:06.06 | *** join/#asterisk pb_ (n=pb@cpc1-cmbg6-0-0-cust434.cmbg.cable.ntl.com) |
19:06.32 | Katty | is there a way to make VI spit out a certain line of a document? |
19:06.39 | Nugget | google for "royalty free music" |
19:06.49 | Katty | or cat. |
19:07.08 | *** join/#asterisk Broom (n=none@12.174.235.14) |
19:07.28 | Broom | hello all, i was wondering if anyone could guide me to information on how to turn up the volume of tue MOH |
19:07.29 | Broom | ? |
19:07.29 | *** join/#asterisk epablo (n=epablo@200.109.73.215) |
19:07.31 | I-MOD | Katty: you could use a combination of tail an dhead |
19:07.37 | *** part/#asterisk epablo (n=epablo@200.109.73.215) |
19:07.41 | I-MOD | s/an dhead/and head |
19:07.46 | I-MOD | s/an dhead/and head/ |
19:07.50 | I-MOD | .... |
19:08.06 | Katty | nevermind |
19:08.12 | Katty | it's :n |
19:08.14 | *** part/#asterisk Peaceful (n=Peaceful@70.98.162.62) |
19:08.44 | Katty | I-MOD: and what on earth are you talking about? |
19:09.34 | Katty | justinu: hiya. |
19:10.39 | justinu | capitalizing vi is blashphemy :P |
19:10.43 | I-MOD | nvm, i'm just 10 different kinds of retarded all the time |
19:10.49 | justinu | blasphemy |
19:10.50 | saftsack | Nugget, didnt find any good sources :( |
19:11.57 | *** join/#asterisk twisla (i=twisla@lutin.jard.in) |
19:13.40 | *** join/#asterisk epablo (n=epablo@200.109.73.215) |
19:14.11 | epablo | is there a way of getting asterisk to save the local tree on a DB? |
19:14.23 | justinu | katty: try this command: sed -n {45p} file.txt |
19:14.27 | justinu | would print line 45 only |
19:15.21 | epablo | Has anyone done a asterisk - ser, setup? |
19:16.13 | epablo | i want to transperantlly setup SER to loadbalance my asterisk servers. Is this posible? |
19:16.15 | justinu | ~seen r_evolution |
19:16.21 | jbot | r_evolution is currently on #asterisk (1h 55m 58s), last said: 'but soon!'. |
19:18.03 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
19:18.30 | *** join/#asterisk ApEtc (i=apetc@ip70-162-216-7.ph.ph.cox.net) |
19:19.41 | *** join/#asterisk VxJasonxV (n=jason@unaffiliated/VxJasonxV) |
19:20.18 | sundancer | Noobish question here.. how to configure dialplan if to give me external line via Zap interface when i dial 9 and then external number |
19:20.45 | iCEBrkr | sundancer: Dial(Zap...) |
19:20.58 | sundancer | Where do i specify external number? |
19:21.02 | *** join/#asterisk MattH (n=MattH@63.174.244.195) |
19:21.15 | iCEBrkr | sundancer: Use ${EXTEN} |
19:21.19 | MattH | Hi... can anyone explain why _X.,1,Zapateller is not catching inbound calls that don't have a route? |
19:21.25 | Netgeeks | exten => _9.,1,Dial(Zap/G1/${EXTEN:1}) |
19:21.32 | SexyKen | tzanger any luck? |
19:21.32 | iCEBrkr | sundancer: It's like dialing another SIP phone, but you're using the zap device instead |
19:21.38 | Netgeeks | assuming you have your zap channels set up as group=1 |
19:21.56 | iCEBrkr | MattH: ZapTeller has nothing to do with 'route' or whatever you're talking about :P |
19:22.05 | Broom | Netgeeks: doesn't it needs to be _9.| so it strips off the 9? |
19:22.18 | sundancer | Yup, thanx.. but i dont understand what is ${EXTEN} variable here |
19:22.32 | Netgeeks | ${EXTEN:1} stips off the 9 |
19:22.36 | Broom | perfect |
19:22.39 | sundancer | Ahh i get it |
19:22.56 | sundancer | ${EXTEN:10} would cut first 10 digits ? |
19:23.03 | MattH | iCEBrkr, right understood :) is there a reason you (or anyoen else) can think of that _X. wouldn't match a call coming in? |
19:23.16 | epablo | <PROTECTED> |
19:23.18 | Netgeeks | ${EXTEN} contains the full number that was matched in that extension line.... Yes, ${EXTEN:10} would strip the left 10 digits |
19:23.20 | sundancer | Thanx! |
19:23.26 | Broom | cool |
19:23.36 | Netgeeks | MattH: yes |
19:23.40 | iCEBrkr | MattH: Yea, cuz nothing matches it. |
19:23.42 | Broom | Net: you happen to know how to spike up the volume of the MOH? |
19:23.52 | Netgeeks | no number comes in, therefore asterisk would look for the s extenstion |
19:24.03 | MattH | yeah I guess so... bah not getting DNIS |
19:24.06 | iCEBrkr | Netgeeks: Not always true :P |
19:24.23 | Netgeeks | one number comes in... for example 1 wouldn't match _X. because _X. says there is at least two numbers in the string.. |
19:24.53 | iCEBrkr | It depends on your provider. For instance, VoicePulse delivers calls with my full number, while my friend delivers only the last 4 digits if my phone number... |
19:25.54 | Netgeeks | I run a provider that delivers a random length of your number sometimes from the left side, sometimes from the right, just to keep you on your toes ;) |
19:26.00 | iCEBrkr | LOL |
19:26.04 | Qwell[] | Netgeeks: See msg. :) |
19:26.05 | Netgeeks | We don't have any customers.. they leave right away |
19:26.11 | MattH | I'm using asterlink |
19:26.16 | jsharp | Makes customer support easy. |
19:26.18 | MattH | they claim to deliver the number, but I'm not seeing it |
19:26.33 | *** part/#asterisk epablo (n=epablo@200.109.73.215) |
19:26.35 | iCEBrkr | MattH: You should see what's going on with your inbound calls via the CLI |
19:26.39 | iCEBrkr | MattH: 'set verbose 9' |
19:27.17 | MattH | yeah I do see. .and I see no inbound DNIS |
19:27.45 | iCEBrkr | Asterisk doesn't complain about not knowing what to do with XXX-XXX-XXXX? |
19:27.50 | iCEBrkr | or whatever the error message is? |
19:27.51 | MattH | yes it does :) |
19:27.56 | MattH | and it doesn't match on _X. |
19:28.09 | iCEBrkr | You're still not understanding |
19:29.14 | *** join/#asterisk spatulamaan (n=ggilmore@ip66-107-33-196.z33-107-66.customer.algx.net) |
19:29.29 | MattH | I SEE s@from-trunk doesn't exist |
19:30.01 | iCEBrkr | yea, ok.. So it's looking for extension 's' in context 'from-trunk' |
19:30.23 | iCEBrkr | So it'll never 'match' _X. |
19:30.44 | MattH | right... so that's on the provider's end? why would it try to send it to s@from-trunk?! everyone else sends it to just from-trunk |
19:30.52 | *** join/#asterisk kend (n=chatzill@host-64-65-199-187.man.choiceone.net) |
19:30.57 | *** join/#asterisk nshm (n=shmyrev@b.gz.ru) |
19:31.01 | MattH | can I do s,1,include=>from-pstn ? |
19:32.06 | kend | Reading the astGUI install notes, and they mention using the "o" flag in extensions.conf, and then show a line line this: exten => _901144XXXXXXXXXX,1,Dial(${TRUNKX}/${EXTEN:1},55,tTo) -- where do I even look up what the "tTo" on the end means? |
19:32.23 | *** join/#asterisk fifer (n=20f04395@c-24-20-155-56.hsd1.wa.comcast.net) |
19:32.37 | iCEBrkr | kend: On the Wiki |
19:32.40 | iCEBrkr | ~wiki |
19:32.41 | iCEBrkr | ~docs |
19:32.43 | 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 |
19:32.50 | jsharp | tT = transfer allows, o says "set outbound callerid to the same as inbound callerid" |
19:32.53 | kend | Thanks -- but what do I look for? I mean, I have no idea what it *is*. |
19:32.54 | iCEBrkr | kend: Look up the Dial() application |
19:33.01 | kend | Ah! Thanks. |
19:33.17 | iCEBrkr | kend: Each flag is documented in the on the Dial() page |
19:33.27 | fifer | in * 1.2.x how are lines in a single context in the dial plan handeled when they are the same exten and priority, is ony the first one taken or the last? |
19:33.37 | kend | Hadn't realized it was a Dial() thang. Makes sense, now. *feels dumb* |
19:34.01 | iCEBrkr | ,1,Dial() should have tipped ya off :D |
19:34.19 | fifer | I know this question makes no sense with hand written dialplan, but I'm actualy trying to get AMP to do something and want to override something it controles in a context |
19:34.25 | iCEBrkr | fifer: Why would you have the same priority? |
19:34.36 | kend | Yeah, yeah... I'm tired. It's Monday. for i in `cat /usr/local/excuses`; do echo $i; done |
19:34.45 | iCEBrkr | fifer: Waste of time man.. AMP will overwrite it later. |
19:34.52 | iCEBrkr | kend: :) |
19:35.00 | *** part/#asterisk kardecallan (n=kardecal@ns1.pcma.com.br) |
19:35.11 | fifer | Good question, because I'm including lines in an included custom file |
19:35.21 | iCEBrkr | ~amp |
19:35.26 | jbot | well, amp is "amp is, like, NOT supported here! people using it should join #freepbx (the new name of amp)" |
19:35.26 | *** join/#asterisk sfChrisJacob_ (n=sfChrisJ@sf-nat.sourcefire.com) |
19:35.26 | iCEBrkr | err hrrm. |
19:35.27 | fifer | they are not overwritten, just dont know how they are handled. |
19:35.37 | MattH | blah there we go |
19:35.39 | iCEBrkr | fifer: But the main file will be over-written |
19:36.04 | iCEBrkr | if Amp over looks includes (which I doubt) you'll be fine. |
19:36.15 | fifer | I know, I know, the question is about * not amp |
19:36.19 | sfChrisJacob_ | Hey all, anyone know of a wireless headset (with remote answer) that will work with the Polycom IP 500? |
19:36.22 | *** join/#asterisk Lino` (n=Lino@i577BF45F.versanet.de) |
19:36.23 | fifer | It is about how the extensions.conf is parsed |
19:36.52 | fifer | These are includes AMP has in there to allow you to customize things. |
19:37.11 | noky | hi |
19:37.20 | fifer | Botom line, I was just wondering how similure lines were dealt with. |
19:37.20 | sfChrisJacob_ | Polycom had a pdf fow awhile with compatible units, but I cant find it anymore... and google is not proving to be as helpful as it could be... |
19:37.21 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
19:39.56 | a1fa | Mar 20 19:26:20 NOTICE[10294]: chan_sip.c:11364 sip_poke_noanswer: Peer '201' is now UNREACHABLE! Last qualify: 168 |
19:40.00 | a1fa | god fucking damn it |
19:40.04 | a1fa | it was reachable for an hour |
19:40.10 | *** join/#asterisk juanjoc (n=juanjoc@200.73.189.82) |
19:40.30 | a1fa | Total RTP Packet Loss: 127 |
19:40.35 | a1fa | Registered: Yes |
19:40.41 | a1fa | this stupid shit still think its registred |
19:41.10 | *** join/#asterisk PMantis (n=pmantis@cpe-66-66-115-197.rochester.res.rr.com) |
19:41.14 | Katty | what's a /nice/ song with a good solo? |
19:41.18 | Katty | a good geetar solo |
19:41.27 | Netgeeks | Hiya Katty! |
19:41.27 | austinnichols101 | stairway to heaven |
19:41.29 | a1fa | basement jaxxx |
19:41.34 | a1fa | fuck bitches |
19:41.42 | Katty | hiya Netgeeks! |
19:41.47 | Katty | austinnichols101: i've already got that one |
19:42.09 | austinnichols101 | peter frampton - do you feel like I do |
19:42.09 | Netgeeks | Are you looking for acoustical guitar or electric? what flavor of music? |
19:42.18 | xbmodder_lappy | ? |
19:42.44 | xbmodder_lappy | Anglea? |
19:42.52 | a1fa | Katty : got a license :L |
19:42.53 | vuud | Katty: anything from G3 |
19:43.17 | xbmodder_lappy | missouri, that sucks |
19:43.20 | xbmodder_lappy | _sucks_ |
19:43.36 | _Paulo_ | Katty, Bad to The Bone, ZZ Top |
19:43.41 | a1fa | lol |
19:43.50 | *** join/#asterisk pb_ (n=pb@cpc1-cmbg6-0-0-cust434.cmbg.cable.ntl.com) |
19:43.57 | *** join/#asterisk dja (n=alden@zaphod.math.ohio-state.edu) |
19:43.59 | a1fa | _Paulo_ she is not running a strip joint |
19:44.11 | a1fa | bad to the bone.. lol |
19:44.16 | [TK]D-Fender | Katty : Final Countdown (Europe), Here I Go Again ; Still oF the Night (Whitesnake) |
19:44.19 | *** join/#asterisk sigwerk (n=sigwerk@cyclone.sigterm.net) |
19:44.28 | austinnichols101 | D-Fender: the 80s are over |
19:44.32 | [TK]D-Fender | LIES! |
19:44.46 | *** join/#asterisk saftsack (n=oliver@IP-213188106101.dialin.heagmedianet.de) |
19:44.50 | austinnichols101 | I've got some 'Winger' here for you |
19:44.56 | saftsack | hi i couldnt find any moh files in google :( |
19:45.03 | sfChrisJacob_ | no love on the Polycom headsets huh? |
19:45.07 | [TK]D-Fender | austinnichols101 : I already play "Easy Come Easy Go", and "Seventeen" :) |
19:45.19 | austinnichols101 | nice |
19:45.52 | stoffell | sfChrisJacob_, plantronics works nice with polycom |
19:46.09 | saftsack | [TK]D-Fender, do you have moh files? |
19:46.23 | [TK]D-Fender | saftsack : Like most everyone else, yes. |
19:46.23 | sfChrisJacob_ | do you know if polycom/plantronics allow for remote answer? |
19:46.46 | saftsack | do you know a good source for them? |
19:46.55 | stoffell | sfChrisJacob_, you mean hitting 'answer' on the headset or wire ? |
19:46.56 | sfChrisJacob_ | I am looking for wireless, and I seem to think this is possible without the lifting contraption... |
19:46.57 | [TK]D-Fender | sfChrisJacob_ : Yes & no, thats in SIP-B to be implemented in * for this summer (expected) |
19:47.10 | sfChrisJacob_ | hitting answer on the headset |
19:47.19 | [TK]D-Fender | sfChrisJacob_ : Oh that... No.... |
19:47.33 | [TK]D-Fender | I have the plantronics lifters for my CSR's.. PITA... |
19:47.40 | astra^^ | any open source billing software for*? |
19:48.16 | sfChrisJacob_ | [TK] so the hand set needs to be physically lifted in order to answer? |
19:48.34 | stoffell | sfChrisJacob_, i only use the "wired ones" but they don't have remote answer |
19:48.56 | sfChrisJacob_ | stoffell, I see... thanks for the info |
19:49.03 | [TK]D-Fender | sfChrisJacob_ : for a headset no, but I extended your meaning a little. You just need to press the headst button. But that means you have to be in range of the phone |
19:49.05 | saftsack | [TK]D-Fender, can you give me a file? :) |
19:49.24 | [TK]D-Fender | saftsack : * comes with 3-4 samples already.... |
19:49.35 | saftsack | where to find these samples? |
19:50.08 | [TK]D-Fender | saftsack " its in the "add-ons" file I believe |
19:50.23 | saftsack | in the source directory? |
19:50.36 | [TK]D-Fender | Just go look.... |
19:50.54 | sfChrisJacob_ | [TK]d-fender, Yeah... so say the receptionist is in the next room making coffee and she hears the phone ring... she can press a button on the wireless headset and get the call.... |
19:51.39 | a1fa | anybody else having broblems with CrapPhone100 |
19:51.51 | a1fa | this nat is bothering me soo much |
19:52.01 | [TK]D-Fender | sfChrisJacob_ : nope.... |
19:52.34 | sfChrisJacob_ | damn.... ok... got it... need to press the answer button on the phone... |
19:52.42 | a1fa | it seems, that everytime it unregisters |
19:52.47 | a1fa | it cant re-register |
19:52.49 | [TK]D-Fender | sfChrisJacob_ : you need either a lifter or to be next to the phone. I would suggest you get an ATA and a plantronics headset phone and ring them in parallel |
19:52.51 | a1fa | stupid netgear |
19:53.11 | a1fa | UNREACHABLE |
19:53.15 | a1fa | i hate this word |
19:54.30 | *** join/#asterisk Shoragan (n=shoragan@d072.apm.etc.tu-bs.de) |
19:57.44 | *** join/#asterisk jtodd (n=jtodd@ti.fox-den.com) |
19:59.07 | r_evolution | man... everyone went strangely quiet |
19:59.34 | xbmodder_lappy | no |
19:59.37 | xbmodder_lappy | not meh |
20:01.44 | jsharp | Then I realized I'd make more money just broadcasting the revolution. |
20:01.51 | r_evolution | ... |
20:01.53 | r_evolution | har har |
20:02.25 | *** join/#asterisk hfb (n=hfb@pool-71-116-252-188.lsanca.dsl-w.verizon.net) |
20:04.30 | Netgeeks | Darn, Katty is MIA and i found her the best guitar solo |
20:04.58 | *** join/#asterisk Inkubot (n=inkubot@200.74.170.218) |
20:05.03 | Inkubot | hi |
20:05.06 | r_evolution | you guys are just mean... |
20:05.10 | xbmodder_lappy | i'd like to die... |
20:05.22 | xbmodder_lappy | Inkubot, it ryhmes! |
20:05.23 | xbmodder_lappy | :-D |
20:05.32 | Inkubot | i've got some SpeedTouch devices (thomson). |
20:05.35 | *** join/#asterisk tzafrir (n=tzafrir@local.xorcom.com) |
20:05.44 | Inkubot | Asterisk doesn't recognize the FLASH key |
20:05.55 | Inkubot | it thinks that is a Hung UP |
20:05.57 | Inkubot | or something |
20:06.23 | xbmodder_lappy | Inkubot, what ATA? |
20:06.23 | a1fa | yup yuypyppy |
20:06.24 | a1fa | y |
20:06.33 | a1fa | no body buy BUDGETPHONE |
20:06.36 | a1fa | fuck grandstream] |
20:06.39 | a1fa | omfg |
20:06.44 | Inkubot | xbmodder_lappy ST-190 and a ST-780(wl) |
20:06.48 | jbalcomb | i thought it was budge not budget? |
20:07.10 | xbmodder_lappy | Inkubot, what brand? |
20:07.31 | Inkubot | damn.. what is brand! |
20:08.00 | Inkubot | sorry, English it is not my language |
20:08.02 | xbmodder_lappy | oh |
20:08.03 | xbmodder_lappy | lol |
20:08.06 | Inkubot | :P |
20:08.06 | xbmodder_lappy | SpeedTouch |
20:08.09 | xbmodder_lappy | is the brand |
20:08.12 | Inkubot | yeps |
20:08.13 | xbmodder_lappy | I don't know about them. |
20:08.32 | xbmodder_lappy | I know that grandstream doesn't support FLASH |
20:08.34 | a1fa | jbalcomb : assbite, you got the idea |
20:08.44 | xbmodder_lappy | and that sipura does... |
20:08.57 | xbmodder_lappy | thats my limited knowledge of these evil things called ATAs |
20:09.00 | Inkubot | emmm.. i also have grandstream devices.. it's works fine |
20:10.23 | Abydos313 | man i ordered my spa3k from digitnetworks.com and they charged my card on the 15th and just sent device now. what a buch of fuktards. i'll never use them again |
20:10.30 | *** join/#asterisk starlein (i=star@fo0bar.de) |
20:11.01 | tzanger | yeah a whole 3 business days, gosh you must really be in a bind |
20:11.03 | Abydos313 | anyoe else ever use them? is their regular business practice for them? |
20:11.35 | *** part/#asterisk sfChrisJacob_ (n=sfChrisJ@sf-nat.sourcefire.com) |
20:11.39 | Abydos313 | no bind, it's just bullshit to charge a card and not send product out to customer for 5 days |
20:11.40 | starlein | does anyone know why i get with enabled qualify in iax2 show peer ... Status: UNKNOWN |
20:11.49 | tzanger | I could see you upset if it happened habitually but it's a first time you ordered from them and there was a discontinuity between when they charged and shippped... jeez |
20:12.03 | justinu | IBM put an auth on my credit card for a laptop they won't ship for another two weeks |
20:12.16 | Abydos313 | my customers would never put up with that in my business |
20:12.19 | r_evolution | kill them justin. |
20:12.22 | Abydos313 | haha |
20:12.26 | justinu | i just want the fucking laptop! :) |
20:12.32 | tzanger | Abydos313: sometimes that stuff happens. As I said if it is a habitual practise of a company I would avoid them, but sometimes shit just happens. No need to be a prick about it |
20:12.32 | r_evolution | ok well |
20:12.35 | r_evolution | get the fucking laptop |
20:12.36 | r_evolution | THEN |
20:12.38 | r_evolution | kill them justin. |
20:12.47 | justinu | heh |
20:12.55 | Abydos313 | how is asking if someone else has issues with them being a prick? |
20:12.55 | r_evolution | ;) |
20:13.46 | r_evolution | you bastards... distracted me... |
20:14.02 | r_evolution | put the wrong IP in for connection ;x |
20:14.20 | tzanger | Abydos313: are you obtuse? |
20:14.22 | tzanger | "what a buch of fuktards. i'll never use them again |
20:14.23 | tzanger | " |
20:14.26 | tzanger | is not being a prick? |
20:14.37 | Abydos313 | ok that was alittle over the top..heh |
20:15.06 | r_evolution | you know... any time anyone asks about someone else being obtuse |
20:15.10 | r_evolution | i cant help but think of triangles |
20:15.21 | generalhan | lol |
20:15.54 | r_evolution | wonder why that is... :-\ |
20:16.23 | justinu | my triangle is obscene |
20:16.23 | tzanger | r_evolution: :-) |
20:16.25 | justinu | not obtuse |
20:17.04 | r_evolution | oh my godness gracious! |
20:17.05 | r_evolution | hah |
20:17.10 | r_evolution | throwback expressions |
20:17.21 | *** join/#asterisk razu_ (n=razu@80-235-91-173-dsl.prn.estpak.ee) |
20:17.32 | justinu | .ee? |
20:17.50 | r_evolution | O_o |
20:17.51 | r_evolution | are you making noises at me justin? |
20:17.54 | a1fa | aiiight |
20:17.57 | a1fa | everybody stop for a second |
20:18.02 | justinu | (12:17:26) razu_ [n=razu@80-235-91-173-dsl.prn.estpak.ee] entered the room. |
20:18.04 | justinu | what's .ee? |
20:18.05 | a1fa | nat=yes;qualify=2000 |
20:18.08 | r_evolution | oh |
20:18.14 | a1fa | keep-alive=10 (on the phone) |
20:18.16 | justinu | estonia? |
20:18.20 | a1fa | register-every=3600s |
20:18.21 | r_evolution | maybe? |
20:18.30 | a1fa | and it goes unavailable |
20:18.35 | a1fa | UNREACHABLE |
20:18.57 | justinu | connectivity problem then |
20:19.02 | starlein | does anyone know why i get with enabled qualify settings in "iax2 show peer ..." Status: UNKNOWN ??? |
20:19.12 | tzanger | alephcom: 3600s is 1h is it not? |
20:19.12 | a1fa | justinu : i can get to it via http:// |
20:19.15 | a1fa | i can get to the phone |
20:19.19 | a1fa | i was thinking dns issues |
20:19.22 | tzanger | your NAT holes would close up far sooner than that |
20:19.25 | a1fa | so i used static IPs |
20:19.26 | tzanger | try 60s not 3600ds |
20:19.29 | tzanger | er 3600s |
20:19.30 | juanjoc | Has anybody been experiencing memory corruption problems when using the latest app_txfax with spandsp? |
20:19.45 | a1fa | tzanger : i tried that too.. didnt help.. what happens, it goes unreachable, and never re-registers |
20:20.02 | tzanger | hmm it's like the phone is convinced it doesn't have to re-register |
20:20.17 | a1fa | i have the newest firmware |
20:20.20 | justinu | i suppose it could be DNS |
20:20.23 | a1fa | i guess the most stable |
20:20.30 | a1fa | does anybody know a public dns? |
20:20.32 | justinu | rule it out by switching to numeric IP address |
20:20.32 | a1fa | that i can use? |
20:20.34 | *** join/#asterisk dja (n=alden@zaphod.math.ohio-state.edu) |
20:20.35 | justinu | 4.2.2.2 |
20:20.36 | r_evolution | i know in the case of some of our customers... they STAY unknown... |
20:20.40 | a1fa | justinu : i did it.. it didnt work |
20:20.46 | justinu | not DNS then |
20:20.48 | a1fa | justinu : i will do it again |
20:20.52 | jbalcomb | a1fa haha.. assbite, that's a good one. |
20:20.55 | justinu | you can use 4.2.2.2 |
20:21.01 | jbalcomb | i always use 4.2.2.2 |
20:21.06 | r_evolution | because the firewall that they happen to be located behind isn't very friendly :) |
20:21.15 | jbalcomb | justinu i don't tell anyone about it though... |
20:21.18 | a1fa | 4.2.2.2? |
20:21.19 | justinu | lol |
20:21.35 | Abydos313 | that number is used by alot of techs.. cisco owns it i think |
20:21.35 | a1fa | dnsauth1.sys.gtei.net. |
20:21.39 | jbalcomb | apple owns 4.x.x.x/8 |
20:21.39 | razu_ | justinu : ee is estonia :) |
20:21.50 | justinu | ok |
20:21.51 | a1fa | i want to own |
20:21.55 | justinu | i thought 4.2.2.2 was verizon |
20:22.01 | a1fa | 3.1.33.7 |
20:22.05 | jbalcomb | i'm pretty sure its apple |
20:22.07 | a1fa | its my lifes dream |
20:22.10 | Abydos313 | we could look it up |
20:22.11 | jbalcomb | 4.2.2.1 and 4.2.2.2 |
20:22.15 | a1fa | whois 4.2.2.2 |
20:22.24 | a1fa | OrgName: Level 3 Communications, Inc. |
20:22.26 | jbalcomb | Abydos313 you do it, cause i don't know how |
20:22.40 | a1fa | level3 owns it |
20:22.42 | Abydos313 | dnsstuff.com |
20:22.46 | a1fa | CIDR: 4.0.0.0/8 |
20:22.58 | jbalcomb | which one does MIT own? |
20:23.05 | jbalcomb | they have a /8 as well |
20:23.08 | *** join/#asterisk Shoragan (n=shoragan@d072.apm.etc.tu-bs.de) |
20:25.34 | a1fa | this is insane |
20:25.42 | a1fa | maybe the ISP is randomly dropping packets |
20:25.59 | justinu | set up a ping |
20:26.33 | a1fa | yah |
20:26.42 | a1fa | what sip runs via tcp? |
20:26.50 | justinu | you can run sip over TCP |
20:26.56 | justinu | not directly to asterisk tho, without patches |
20:27.01 | a1fa | sucks |
20:27.21 | justinu | a lot of times tcp can be worse than udp |
20:27.26 | a1fa | yeah |
20:27.28 | a1fa | sure is |
20:27.40 | justinu | now SCTP is what we need |
20:30.45 | a1fa | the router has gone mad now |
20:30.50 | a1fa | netgear |
20:30.55 | a1fa | she cant ping it anymore |
20:31.17 | justinu | anyone ever have issues with mixmonitor recording one side of the conversation much louder than the other? |
20:31.24 | justinu | playing with the gains doesn't help |
20:31.50 | justinu | even tho the gain adjustments are working, the local side of the conversation in the monitored calls is always very loud |
20:32.09 | jbalcomb | justinu could be attenuation |
20:32.37 | jbalcomb | justinu can you set an option in the mixmonitor to do it's own adjustments? |
20:32.45 | justinu | i've had some really bizzare experiences w/ this install |
20:32.50 | justinu | ztmonitor also shows very high TX levels |
20:33.05 | justinu | however, if I set txgain -50, you can't hear shit on the calls, but ztmonitor still shows high TX levels |
20:33.17 | jbalcomb | justinu as does mine. i assume for now that mine is because of the gxp-2000 super gain mic. |
20:33.47 | jbalcomb | justinu now that is wierd. are you seeing any difference at all in ztmonitor? |
20:34.02 | justinu | no |
20:34.09 | jbalcomb | justinu do you have one of those 1ghz test tone numbers? |
20:34.15 | justinu | this is a sangoma system... i need to talk to them about this |
20:34.19 | justinu | something is screwy |
20:34.23 | *** join/#asterisk RoyK (n=roy@216-99-29.0506.adsl.tele2.no) |
20:34.27 | jbalcomb | justinu any chance you are monitoring the wrong channel? |
20:34.30 | justinu | it didn't act like this when I originally set it uop |
20:34.32 | justinu | no chance |
20:34.55 | justinu | however, I did update everything recently... asterisk 1.2.5, zaptel 1.2.4, wanpipe 2.3.3 current |
20:34.57 | RoyK | ~seen zoa |
20:35.16 | jbot | zoa <n=kkk@pirus.securax.be> was last seen on IRC in channel #asterisk, 5d 3h 51m 59s ago, saying: 'it looks kinda suspicious :p'. |
20:35.17 | jbalcomb | justinu how about a make clean ; make install on zaptel to see if it's jacked somehow? |
20:35.17 | justinu | but i've since reverted back to all the original software revs |
20:35.17 | jbalcomb | justinut i don't know wanpipe |
20:35.17 | justinu | and the behavior is still there |
20:35.25 | *** part/#asterisk eliel (n=eliel@200.123.183.89) |
20:35.56 | jbalcomb | justinu maybe the upgrade/revert busted some lib or config? |
20:36.01 | justinu | possible |
20:36.17 | *** join/#asterisk LostFrog (n=reallyno@dsl093-100-093.wdc2.dsl.speakeasy.net) |
20:36.18 | *** join/#asterisk mattodude (n=matt@gateway.digium.com) |
20:36.36 | jbalcomb | justinu have you looked around for file or links being used that point to 1.2.5 stuff? |
20:36.49 | *** join/#asterisk Assid (n=assid@59.183.55.15) |
20:36.51 | a1fa | ok |
20:36.53 | justinu | not specifically... i'm mostly interested in the kernel modules |
20:36.54 | Assid | heya |
20:36.59 | a1fa | justinu : i think i know whats the problem |
20:37.00 | a1fa | SPI |
20:37.01 | justinu | i should look thru there |
20:37.04 | a1fa | Stateful Packet Inspection |
20:37.07 | Assid | um. is there a way to relaxdtmf for incoming iax calls? |
20:37.08 | a1fa | on that netgear router |
20:37.11 | justinu | ah, could very well be alfa |
20:37.11 | a1fa | hahhaha |
20:37.20 | a1fa | its actually simple flood protection |
20:40.37 | *** join/#asterisk nsillik (n=nsillik@avenger.cis.temple.edu) |
20:41.26 | nsillik | has anyone here had much luck getting outbound calling to landlines working with Gizmo from asterisk? I can call other gizmo numbers, but not the pstn |
20:43.23 | *** join/#asterisk Delmar (n=delmar@203-114-178-231.inspire.net.nz) |
20:44.55 | *** join/#asterisk Flauto (n=zhao@adsl-75-3-130-186.dsl.chcgil.sbcglobal.net) |
20:45.09 | Flauto | Mar 20 14:26:01 WARNING[30440]: chan_sip.c:6840 check_via: 'Server217' is not a valid host |
20:45.13 | Flauto | what is this for? |
20:45.17 | LostFrog | Is there any way to get the status of a sip peer from manager? Like we do from sip through hints? |
20:46.45 | Flauto | any idea? |
20:46.57 | Delmar | Flauto, check sip.conf for the section relating to Server217 |
20:47.13 | Flauto | hmm... |
20:47.38 | Delmar | Flauto, perhaps u have typo'd the host= |
20:48.05 | Delmar | Flauto, after u fix it do "sip reload" at console |
20:48.13 | Flauto | thanks |
20:48.19 | Flauto | let me see |
20:48.21 | justinu | lostfrog: there are peerstatus change events, in the manager, afaik |
20:49.21 | Katty | there's a linux terminal program that will take an ip and tell me what ports it's listening on |
20:49.31 | Katty | someone refresh my memory as to the name of it |
20:49.34 | Netgeeks | nmap |
20:49.37 | Katty | thanks. |
20:51.32 | Katty | my brain just sucks sometimes. |
20:51.56 | tsume | Hey hot babe! :D |
20:52.07 | tsume | Katta-kun1! oi! :D |
20:52.07 | Katty | take that comment and shove it |
20:52.48 | tsume | *slapped |
20:54.46 | LostFrog | justinu: So, I have to listen to Evens.. :( |
20:55.46 | justinu | ? |
20:56.34 | Hmmhesays | Katty: mine to |
20:56.36 | Hmmhesays | *too |
20:57.29 | Hmmhesays | oh therapy can you please fill the void, am I retarded or am I just overjoyed |
21:00.34 | *** join/#asterisk dsfr (n=dsfr@pdpc/sponsor/digium/dsfr) |
21:03.37 | a1fa | lol |
21:04.10 | a1fa | so shitty |
21:04.20 | a1fa | now i cant get a WAN IP via Cable Modem |
21:04.28 | a1fa | so i cant do anything |
21:04.43 | justinu | cable modems suck |
21:04.47 | a1fa | sure does |
21:07.38 | tsume | Katty: will you bear my children? |
21:07.59 | a1fa | cable is cool |
21:08.08 | a1fa | did you know you can get 30mibts down that crazy cable? |
21:08.14 | tsume | cable is definitely cool, you can uncap the modems |
21:08.38 | a1fa | tsume : how do you uncap a modem? |
21:08.39 | a1fa | :P |
21:08.51 | LostFrog | illegaly. |
21:09.03 | Inkubot | damn FLASH key |
21:09.04 | Inkubot | :\ |
21:09.08 | tsume | a1fa: all cable implementations available use docsis |
21:09.08 | Inkubot | and damn ATA |
21:09.22 | tsume | alephcom: you don't need to know how ;) its for specials like I to know ;) |
21:10.11 | Katty | Hmmhesays: bwha? |
21:10.18 | Katty | Hmmhesays: what's going on with you? therapy? |
21:12.17 | *** join/#asterisk zotz (n=zotz@24.231.32.85) |
21:13.20 | Hmmhesays | Katty well I got really drunk this weekend and tore FZ a new one |
21:13.42 | Hmmhesays | i suppose that could be considered therapy |
21:13.44 | Inkubot | can someone giveme an account on an Asterisk BOX to test a thing ? |
21:13.51 | Hmmhesays | testing what |
21:13.56 | Inkubot | an ATA |
21:14.02 | Hmmhesays | paypal me a 10 |
21:14.06 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
21:14.06 | *** mode/#asterisk [+o russellb] by ChanServ |
21:14.11 | Inkubot | pfff... |
21:14.15 | a1fa | keep talking |
21:14.16 | a1fa | tsume |
21:14.20 | a1fa | docsis |
21:14.27 | tsume | docsis :) yes |
21:14.39 | tsume | a1fa: you don't need to know about it ;) |
21:14.40 | Hmmhesays | aight paypal me 5 bucks and one is all yours |
21:15.03 | jbalcomb | Inkubot i'll do it for $.99 |
21:15.08 | a1fa | tsume : what you got? |
21:15.09 | Hmmhesays | hahaa |
21:15.21 | Inkubot | eiiiiiiii guys i want to try just a thing.. no more than a minute |
21:15.22 | a1fa | how do you connect to it |
21:15.22 | astra^^ | what is the difference betwenn g729 and g729a |
21:15.30 | tsume | a1fa: its a trade secret ;) |
21:15.34 | Hmmhesays | g729 is annex A |
21:15.40 | Hmmhesays | g729a even |
21:15.57 | astra^^ | hw do i knw which g729 i am using |
21:16.16 | *** part/#asterisk LostFrog (n=reallyno@dsl093-100-093.wdc2.dsl.speakeasy.net) |
21:16.26 | astra^^ | i bgt a g729 licence y'day |
21:16.39 | astra^^ | is it g729a or g729 |
21:16.43 | Hmmhesays | wouldn't matter to you anyway |
21:16.48 | a1fa | tsume : i am sure the telco would know you uncapped your modem instantly |
21:17.31 | Hmmhesays | g729a is less complex than g729 |
21:17.34 | tsume | nope |
21:17.43 | tsume | they can only tell if they are able to connect to the modem |
21:18.09 | astra^^ | so wen i get g729 it supports both? |
21:18.33 | a1fa | so it would raise supicions if they cant connect |
21:18.36 | Hmmhesays | g729 a and b are stream compatible with g729 |
21:18.43 | a1fa | and they can immediatley assume, you fuxerod the modem |
21:18.47 | Hmmhesays | ~google |
21:18.49 | jbot | it has been said that google is a search engine found at http://www.google.com/ |
21:19.19 | astra^^ | ok |
21:19.23 | *** join/#asterisk dsfr (n=dsfr@pdpc/sponsor/digium/dsfr) |
21:19.43 | iq | astra^^: did you find out the number of legs it covers :) ? |
21:19.46 | astra^^ | but tellme i jst bgt a new g729 frm digium |
21:20.01 | Hmmhesays | why do you care? |
21:20.05 | Hmmhesays | ;) |
21:20.18 | astra^^ | and weather it is g729a or g729 |
21:20.20 | iq | astra^^: call Digium. They have good support |
21:20.28 | astra^^ | please.... |
21:20.31 | astra^^ | :( |
21:20.38 | russellb | You do not get g729b from Digium. |
21:20.47 | iq | astra^^: did you pay for it ;) ? |
21:20.55 | Katty | Hmmhesays: FZ? tore? new one? mew?! |
21:21.11 | astra^^ | yes i did |
21:21.13 | Katty | Hmmhesays: speak kat you freak! |
21:21.16 | astra^^ | 10$/channel |
21:21.21 | Hmmhesays | astra^^ you know what is really irritating? |
21:21.30 | astra^^ | am sorry |
21:21.35 | Hmmhesays | http://www.digium.com/en/products/voice/g729codec.php <-- |
21:21.49 | Katty | (= |
21:21.55 | Hmmhesays | we're all n00bs at some point, but come on man, read a little |
21:22.25 | astra^^ | ok what do i do to get g729a |
21:22.48 | iDunno | # come on let me hold you |
21:22.51 | iDunno | # touch you |
21:22.54 | iDunno | # feel you |
21:22.57 | iDunno | # always |
21:22.58 | iDunno | :) |
21:23.02 | iq | :) |
21:23.07 | iq | lol |
21:23.09 | Hmmhesays | feel the lurve bi@atch |
21:23.16 | Hmmhesays | lol |
21:23.23 | Hmmhesays | hot |
21:23.52 | Hmmhesays | was it good for you? |
21:24.35 | *** join/#asterisk ManxPower (n=ewieling@dpc6745150107.direcpc.com) |
21:24.37 | Hmmhesays | lol |
21:24.58 | *** join/#asterisk crich1999 (n=crich@port-212-202-198-154.dynamic.qsc.de) |
21:25.47 | a1fa | whats'up with all these describes |
21:29.38 | *** join/#asterisk sanee (n=sanee@82.117.210.45) |
21:32.08 | *** part/#asterisk UlbabraB (n=salama@host241-43.pool8172.interbusiness.it) |
21:35.03 | *** join/#asterisk Seldon1975 (n=someone@199.243.101.131) |
21:35.20 | Seldon1975 | Q:what's red and invisible? |
21:35.57 | iDunno | nothing. |
21:36.03 | Seldon1975 | A: no tomatoes |
21:37.07 | tuxinator_linux | aren't we all |
21:37.53 | X-Rob | do some coding, you lazy buggers. |
21:38.00 | Netgeeks | !! |
21:38.09 | Seldon1975 | hno point: our project was cancelled |
21:38.28 | tuxinator_linux | X-Rob: I am.. doing that... coding |
21:38.37 | Netgeeks | function hello_world(); |
21:38.47 | Seldon1975 | while(true); |
21:38.54 | jsharp | profit(); |
21:38.56 | jsharp | endwhile() |
21:39.16 | tuxinator_linux | you know... coding is not as fun when you do it on a schedule |
21:39.18 | Seldon1975 | 10 PRINT("I AM COOL"); |
21:39.20 | Seldon1975 | 20 GOTO 10 |
21:39.32 | Seldon1975 | tuxinator_linux: amen |
21:40.01 | justinu | > v v ,,,,,"Hello"< >48*, v v,,,,,,"World!"< >25*,@ |
21:40.25 | tuxinator_linux | I used to go into computer stores... back in the day.... and quickly write programs in qbasic.... 'You've been hacked'... or something to that affect.... and then watch them try to figure it out. |
21:40.52 | Seldon1975 | haha that was a lot of fun |
21:40.55 | Netgeeks | by the way, you didn't need that ; at the end of line 10, Seldon... |
21:41.20 | Seldon1975 | Netgeeks: sorry, I sold my C64 about 20 years ago... |
21:41.44 | tuxinator_linux | most of us were born about 20 years ago |
21:41.44 | *** join/#asterisk MattH (n=MattH@63.174.244.195) |
21:41.45 | Netgeeks | hehe, yeah, I know, I'm even trying to remember if you needed the (). I don't think so |
21:41.53 | Seldon1975 | now I just use an emulator occasionally to play Defender of the Crown |
21:42.03 | MattH | Hi... why would iax2 show channels show 99ms of jitter, when a ping between me and the other iax server comes up with 13ms? is there really that much packet loss? |
21:42.15 | justinu | jitter != latency or packet loss |
21:42.18 | MattH | 00006/00017 00021/00020 00127ms 0122ms 0162ms ulaw |
21:42.34 | tuxinator_linux | back to work for me |
21:42.49 | MattH | justinu, understood... but my ping times are a solid 13ms.. not going up and down huge amounts |
21:43.36 | justinu | i'm not convinced those stats can be trusted either, matt |
21:43.43 | MattH | ahh hrmm ok.. |
21:43.49 | ManxPower | justinu, is that RPG-II? |
21:43.50 | ManxPower | it's been 15 years since I did anything in RPG-II |
21:43.58 | justinu | ManxPower: befunge ;) |
21:44.08 | MattH | well my audio is pretty nasty to this host... to whem I have pretty good solid 13ms pings.. no drops... no jumps... yet the audio is bad.. and asterisk is showing the above |
21:44.11 | MattH | any thoughts? |
21:44.21 | justinu | matth: increase the frequency/size of the pings |
21:44.30 | MattH | justinu, I did a ping -f! |
21:44.30 | justinu | i use ping -i0.02 -s180 -c 5000 |
21:44.37 | MattH | and then all came back fine.. will try larger ones |
21:44.55 | justinu | that's pretty close to the kind of traffic an RTP stream is |
21:45.04 | MattH | k |
21:45.31 | MattH | 221 packets transmitted, 219 received, 0% packet loss, time 5472ms |
21:45.31 | MattH | rtt min/avg/max/mdev = 13.061/13.998/16.703/0.688 ms, pipe 2 |
21:45.50 | justinu | looks good, your problem must be somewhere else |
21:46.11 | MattH | yeah.. why would asterisk show high jitter issues, yet ping times would be fine? that's wher eI am drawing a blank |
21:46.23 | MattH | 1745 packets transmitted, 1743 received, 0% packet loss, time 43174ms |
21:46.23 | r_evolution | just != jitter O_O |
21:46.23 | MattH | rtt min/avg/max/mdev = 13.017/13.912/29.220/0.811 ms, pipe 3 |
21:46.25 | justinu | fucked up QoS? |
21:46.30 | r_evolution | justin !=jitter even |
21:46.36 | ManxPower | MattH, Well, RTP is not ICMP for one thing. |
21:46.53 | justinu | maybe something is deliberately causing jitter on RTP or IAX packets |
21:46.54 | *** join/#asterisk tmccrary (n=tmccrary@68.78.185.254) |
21:46.55 | MattH | I'd think QoS.. but I have other iax streaming going to external hosts and they are ok |
21:47.06 | ManxPower | Also Asterisk's jittter buffer (what little there is of it) has known issues. |
21:47.08 | tmccrary | Mar 20 16:43:45 WARNING[17480]: chan_zap.c:10868 setup_zap: Ignoring signalling |
21:47.08 | tmccrary | Mar 20 16:43:45 WARNING[17480]: chan_zap.c:10868 setup_zap: Ignoring switchtype |
21:47.12 | MattH | could be... but again... other iax2 connections sound fine |
21:47.20 | tmccrary | is that normal for a PRI connection with asterisk on a TE110P? |
21:47.21 | ManxPower | MattH, what happens if your turn OFF the jitter buffer |
21:47.25 | justinu | i'd use ethereal to analyze the RTP stream |
21:47.27 | MattH | I dunno.. let me try |
21:47.36 | justinu | tmccrary: it's normal when you do a reload chan_zap.so |
21:47.42 | tmccrary | oh ok, thanks |
21:48.16 | MattH | interestingly.... it sounds better! |
21:48.31 | MattH | wow that was odd |
21:48.32 | justinu | heh |
21:48.39 | MattH | it's POTS quality |
21:48.39 | ManxPower | tmccrary, you cannot change those things when doing a reload, you can only change them my stopping and starting Asterisk or by unloading and reloading chan_zap.so |
21:49.22 | MattH | well now that's interesting |
21:49.27 | MattH | so it's a jitterbuffer bug? |
21:49.28 | *** join/#asterisk alexis101 (n=alex@70.54.204.92) |
21:49.37 | alexis101 | hello all |
21:50.05 | alexis101 | I was wondering if any of you know how to implement *67 |
21:50.07 | ManxPower | MattH, pretty common, actually. |
21:50.17 | ManxPower | My satillite connection is going up and down more often than linda lovelace. |
21:50.26 | a1fa | ManxPower :lol |
21:50.30 | MattH | hehe interesting |
21:50.41 | a1fa | ManxPower : how does that work? do you dial-in somewhere? |
21:50.51 | tzanger | alexis101: I explained this earlier this aft I thought |
21:51.10 | ManxPower | Damn rain |
21:51.24 | alexis101 | ok i will check the logs |
21:51.25 | jsharp | mmm Ku band rain fade |
21:51.26 | tmccrary | Is there like a zap debug command? |
21:51.28 | ManxPower | a1fa, Um, no. I use a dish |
21:51.31 | *** join/#asterisk RoyK (n=roy@216-99-29.0506.adsl.tele2.no) |
21:51.42 | ManxPower | tmccrary, what information are you trying to see? |
21:51.51 | Nugget | I hear ManxPower is quite a dish. |
21:51.55 | tmccrary | Is the channel is busy or what |
21:52.00 | ManxPower | jsharp, I shall celebrate the say I can get a T-1 in here. |
21:52.06 | ManxPower | say == day |
21:52.40 | jsharp | Just a T1? No T3? OC-3? |
21:53.09 | a1fa | i dont have those problems |
21:53.11 | jsharp | To the house? |
21:53.19 | tmccrary | tmccrary has an OC-256 |
21:53.40 | ManxPower | jsharp, My DirectTV has a signal strength of 93 right now, where the DirecWay dish has a signal strength of 41 |
21:53.43 | *** join/#asterisk shimi (n=shimi@unaffiliated/shimi) |
21:53.47 | blitzrage | I have an OC-512 to my house |
21:53.51 | ManxPower | Of course I paid the extra $50 to get the larger dish too |
21:53.58 | tzanger | exten => _NXXXXXX,1,do.what.you.need.for.all.calls.here |
21:53.58 | tzanger | exten => _NXXXXXX,n,GotoIf($[${BLOCKCID} = 1]?foo) |
21:53.58 | tzanger | exten => _NXXXXXX,n,Set(CID(name)=SOME NAME) |
21:53.58 | tzanger | exten => _NXXXXXX,n,Set(CID(num)=8012345678) |
21:53.58 | tzanger | exten => _NXXXXXX,n(foo)do.more.of.what.you.need.for.all.calls.here |
21:54.00 | tzanger | exten => _NXXXXXX,n,Dial(Tech/${EXTEN}@peer,timeout,options) |
21:54.03 | tzanger | alexis101: ^^ |
21:54.09 | jsharp | DirecTV and Direcway on the same dish? |
21:54.10 | tzanger | and then simply do this |
21:54.12 | ManxPower | jsharp, I live on the top of a mountian, 11 miles from the nearest CO |
21:54.12 | justinu | is directway 2 way over satellite, or do you have a modem? |
21:54.17 | ManxPower | jsharp, hell no! |
21:54.25 | jsharp | Direcway is 2 way. |
21:54.25 | ManxPower | DirecWay is 2-way |
21:54.28 | shimi | Anyone perhaps has a pointer for setting up asterisk to call back upon request, preferrably to a given number, then execute something like running DISA ? |
21:54.29 | justinu | ic |
21:54.37 | blitzrage | tzanger: CID == CALLERID I think right? |
21:54.43 | jsharp | Sounds like the monkey who installed your direcway dish didn't point it right. |
21:54.44 | tzanger | exten => _*67.,1,Set(BLOCKCID=1) |
21:54.44 | tzanger | exten => _*67.,n,Goto(${EXTEN:3},1) |
21:54.46 | blitzrage | or is CID valid too? |
21:54.49 | ManxPower | jsharp, 1) you cannot put both on the same dish and 2) I'm riding on the owner's DirecWay connection, but have my own DirecTV dish |
21:54.49 | *** join/#asterisk bazz (n=nick@fw.marklogic.com) |
21:54.57 | PoWeRKiLL | how can I create a new cdr column in the db and set it from extension.conf ? |
21:55.02 | tzanger | blitzrage: yeah, that's just off the top of my head, there may be some minor changes to make Asterisk "grok" it |
21:55.24 | Octothorpe | tzanger: pastebin is your friend |
21:55.24 | Octothorpe | ~pb |
21:55.25 | jbot | [pb] a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
21:55.26 | blitzrage | tzanger: understood -- I wish CALLERID function was just CID actually |
21:55.26 | ManxPower | jsharp, DirecWay is VERY picky about being just perfect aim. |
21:55.30 | tzanger | Octothorpe: I'm familliar with pastebin |
21:55.35 | tzanger | blitzrage: me too :-) |
21:55.36 | ManxPower | We will be moving the dish to get a clearer view of the sat |
21:55.53 | blitzrage | there's nothing wrong with 4-5 lines of pasting |
21:55.58 | tzanger | Personally I am not that keen on _*67. as a pattern match, I like my dialplans to match more strictly |
21:55.59 | blitzrage | its when its obscene that its a problem |
21:56.18 | MattH | ewww.. so now I'm just getting bad lag |
21:56.28 | tzanger | so personally I'd have _*67NXXXXXX, _*67NXXNXXXXXX, and the _*670 and *671 variants :-) |
21:56.37 | bazz | what i'd like to do: have asterisk call my voicemail, dial a few numbers to get to the record a message prompt, have asterisk play a sound file over the line (to record it), dial a few more #'s to confirm, and hang up. (a) can i do this (b) how hard is it (c) if i can do this, where should i begin reading to figure out how |
21:56.45 | jsharp | Ah. DirecTV and Direcway are on different sats. I thought they were all serviced off the same satellite. |
21:57.26 | ManxPower | jsharp, They might be on the same sats, but they are different companys, do not offer combined billing and do not offer combined dishes |
21:57.29 | Hmmhesays | i'm thinking about getting a razr |
21:57.33 | Hmmhesays | are they any good? |
21:57.46 | jsharp | Oh. Gotcha. |
21:57.58 | ManxPower | Hmmhesays, shaving is overrated |
21:58.11 | justinu | motorola phones suck |
21:58.11 | Hmmhesays | i meant phone |
21:58.29 | a1fa | lilo |
21:58.40 | jpm_SD | Hmmhesays, Verizon reports a 46 percent return rate on that phone... take that for what it's worth. |
21:58.46 | jsharp | I've used standalone direcway systems and they all made me want to stab myself with a rusty trout. |
21:58.48 | a1fa | lilo u crazy ass :) |
21:58.54 | tzanger | 46%? holy crap |
21:58.56 | ManxPower | But Jessica Simpson has that phone! |
21:59.09 | ManxPower | jsharp, Yeah. |
21:59.24 | a1fa | what phone? |
21:59.28 | shimi | Anyone perhaps has a pointer for setting up asterisk to call back upon request, preferrably to a number given by the caller, then when the calledback number answers, execute something like running DISA to let them dial on my expense? |
21:59.35 | tzanger | directway for interactive? isn't the lag pretty decent? |
21:59.46 | justinu | 1.5 seconds or so, iirc |
21:59.53 | *** join/#asterisk digime (n=digime@ip68-101-196-93.sd.sd.cox.net) |
22:00.13 | ManxPower | jsharp, but at this point I cannot afford the $500 - $700/month for a T-1, and there is no cable or DSL here. |
22:00.13 | ManxPower | once we get a few more seasonals here I'll be able to offset the cost be selling interent and phone service to them |
22:00.44 | ManxPower | and since cell phones don't work very well up here (pretty much anything but Verizon does not work) I'm pretty sure they will want POTS service |
22:00.46 | jsharp | The latency bounced all around the place cause of the TDM multiplexing they use on the enduser-to-satellite connection. |
22:00.57 | ManxPower | tzafrir, ping me |
22:01.01 | justinu | there's no existing pots service? |
22:01.03 | jsharp | It was all sorts of jittery. |
22:01.11 | *** join/#asterisk Dr-Linux (n=Linux@host202-147-168-130.lhr.dancom.net.pk) |
22:01.27 | ManxPower | justinu, there is pots service here, but nobody wants to pay $40/month for a service they will only use on the weekend. |
22:01.28 | Hmmhesays | yeah their like on the 3rd generation of that phone htough |
22:01.38 | Octothorpe | shimi: check nerdvittles.com |
22:01.58 | justinu | makes sense |
22:02.12 | ManxPower | I figure $25/month for combined interent and telephone. |
22:02.17 | Octothorpe | shimi: it is something like "phone home" or something like that, they have a whole tutorial on their site |
22:02.19 | jpm_SD | Hmmhesays, as I said.. take it for what it's worth... They may be fine now, but I'd be leary. |
22:03.32 | *** join/#asterisk maxx4life (n=maxx4lif@71-35-210-12.slkc.qwest.net) |
22:03.34 | tmccrary | I setup a PRI connection between a TE110P and a traditional phone system. But I keep getting: SIP/2.0 486 Busy here when I try to call it. |
22:04.01 | bazz | anyone? |
22:05.12 | shimi | hmmm... web activated DISA... psshh :) |
22:05.39 | *** join/#asterisk afrosheen (n=test@txprotoa2.august.net) |
22:06.18 | tzanger | tmccrary: can you get it to work without the SIP |
22:06.30 | tzanger | i.e can you place calls through the PRI with Local/ or something? |
22:06.31 | *** join/#asterisk amdtech (n=stdamd11@ab1-1-246.shsu.edu) |
22:07.04 | tmccrary | How would I go about doing that? |
22:07.21 | tmccrary | or do you mean with a traditional phone? |
22:07.32 | astra^^ | Mar 21 03:29:07 WARNING[32272]: codec_g729.c:170 g729tolin_framein: Out of G.729 Decoder Licenses! |
22:07.35 | tzafrir | ManxPower, here |
22:07.52 | tmccrary | I think this may be an issue with the comdial phone system, i was just curious if this is a common issue or not |
22:08.19 | astra^^ | i am gettin tis error .. |
22:08.21 | astra^^ | Mar 21 03:29:07 WARNING[32272]: codec_g729.c:170 g729tolin_framein: Out of G.729 Decoder Licenses! |
22:08.31 | astra^^ | i have a g729 licence |
22:08.32 | justinu | pretty clear to me |
22:08.46 | astra^^ | i hve a licence i bgt yday |
22:08.54 | jsharp | How many licenses? How many calls do you have going? |
22:08.56 | [TK]D-Fender | astra^^ : Its in USE and a device is ask to use one more than you have free. |
22:09.32 | astra^^ | i gt only 1 licence |
22:10.45 | shimi | Octothorpe, unfortunately it does calling via PHP script - that I already know how to do :) I need something that accepts a call from one of my DIDs (or whatever), then asks for a phone number, then waits a few seconds, dials to it, and connects that call with DISA. :) |
22:11.06 | astra^^ | wid 1 licence we can place only 1 call right |
22:11.06 | *** join/#asterisk FuriousGeorge (n=Brian@ool-43536ea8.dyn.optonline.net) |
22:12.00 | [TK]D-Fender | astra^^ : Correct |
22:12.33 | astra^^ | ok bt nw my client placed only one call frm a switch .. for testing .. and i gt tis error |
22:12.56 | tmccrary | Patents suck |
22:13.13 | tmccrary | Do you have missing buttons on your keyboard or something? |
22:13.36 | astra^^ | no why did u ask that |
22:13.41 | tmccrary | Or perhaps missing fingers |
22:13.45 | astra^^ | :' |
22:13.49 | tmccrary | bc joo tk lk ths |
22:13.54 | tzanger | hahahaha |
22:13.58 | tzanger | I hate when people do that |
22:13.58 | [TK]D-Fender | tmccrary : NO, just not his native language. |
22:14.02 | astra^^ | hmmm... not funny |
22:14.09 | [TK]D-Fender | tmccrary : if u cn rd th tn u cn pgm n c ;) |
22:14.24 | [TK]D-Fender | tmccrary : if u cn rd ts tn u cn pgm n c ;) |
22:14.28 | tzanger | I don't know if that's because of a native language or simply because they're so used to SMSing from phones without T9 |
22:15.32 | astra^^ | lol very funny.. ha ha ha.. and 3 more ha.. |
22:15.48 | tzanger | astra^^: once license = 1 transcode at a time |
22:15.58 | tzanger | astra^^: 2 licenses = 2 transcodes at the same time |
22:15.58 | tzanger | etc |
22:16.12 | astra^^ | thank tou choo much |
22:16.25 | astra^^ | :>> |
22:16.28 | tmccrary | astra^^: When you wrote that ha ha ha part, I pictured that kid from indiana jones and the temple of doom |
22:16.41 | digime | anyone know where the asterlink guys are? |
22:16.48 | tzanger | #openpbx likely |
22:16.55 | digime | oh yeah |
22:17.04 | justinu | #asterlink? |
22:17.37 | digime | no they are not in their channel |
22:17.45 | digime | who are people using these days for incoming DID? |
22:19.21 | r_evolution | well that works... werd. |
22:19.42 | r_evolution | TK = funny guy today ;) |
22:20.03 | digime | who are people using for incoming DID? |
22:20.11 | denon | nufone |
22:20.15 | [TK]D-Fender | r_evolution : nice to see SOMEBODY got it ;) |
22:20.39 | FuriousGeorge | he all |
22:20.39 | r_evolution | oh yes... may i add how much i enjoyed the ';') |
22:20.52 | FuriousGeorge | HEY |
22:21.00 | ManWithYellowBat | ? |
22:21.02 | tzanger | hahahha |
22:21.16 | ManWithYellowBat | where'd that damn dirty ape go |
22:21.36 | *** join/#asterisk ronn (n=zakforev@84-45-132-117.no-dns-yet.enta.net) |
22:22.10 | asterboy | dam corn, unpredictable areodynamics |
22:22.17 | r_evolution | ew |
22:22.19 | r_evolution | you're gross |
22:22.35 | FuriousGeorge | lol |
22:22.42 | xbmodder_lappy | weeee |
22:23.38 | Mauro__ | any good opensource softphone for linux/bsd? |
22:23.50 | FuriousGeorge | Mauro__: no |
22:23.51 | FuriousGeorge | exiga |
22:23.52 | FuriousGeorge | is new |
22:23.58 | Mauro__ | :D |
22:24.02 | FuriousGeorge | havent tried |
22:24.03 | *** part/#asterisk a1fa (n=a1fa@207.210.210.202) |
22:24.18 | *** join/#asterisk RoyK (n=roy@216-99-29.0506.adsl.tele2.no) |
22:24.38 | r_evolution | note to self |
22:24.45 | r_evolution | do not listen to jungle whilst coding... |
22:24.54 | r_evolution | type too fast... make sloppy mistakes |
22:24.56 | tuxinator_linux | jungle? |
22:25.00 | r_evolution | yessir |
22:25.02 | r_evolution | drum and bass |
22:25.04 | r_evolution | edm |
22:25.26 | r_evolution | think Roni Size... Dieselboy... Concord Dawn... Black Sun Empire... Evol Intent... etc |
22:25.27 | *** join/#asterisk spatulamaan (n=ggilmore@ip66-107-33-196.z33-107-66.customer.algx.net) |
22:25.51 | *** part/#asterisk spatulamaan (n=ggilmore@ip66-107-33-196.z33-107-66.customer.algx.net) |
22:26.06 | r_evolution | what in the hell... |
22:26.07 | jpm_SD | photek |
22:26.08 | tuxinator_linux | skin a wizard? |
22:26.09 | r_evolution | skins a wizard? |
22:26.15 | r_evolution | photek is decent |
22:26.19 | r_evolution | that sounds dirty... |
22:26.37 | jpm_SD | I hear Wizards make good cloaks. |
22:26.57 | *** join/#asterisk doolph (n=doolph@201.227.72.230) |
22:26.58 | FuriousGeorge | it does sound dirty doesnt it |
22:27.07 | FuriousGeorge | i dont think it is though |
22:27.16 | doolph | sup |
22:27.36 | doolph | any astguiclient experimented user? |
22:27.54 | r_evolution | are we experimenting on people now? |
22:27.57 | r_evolution | sweet. |
22:28.12 | tuxinator_linux | where do I sign up r_evolution for the experiments |
22:28.19 | doolph | Laughing Out Loud |
22:28.37 | *** join/#asterisk ManxPower (n=ewieling@dpc6745150107.direcpc.com) |
22:28.45 | jpm_SD | You know.. you can just LOL.. we all know what it means now. |
22:28.58 | r_evolution | ... |
22:29.00 | r_evolution | BURN! |
22:29.04 | doolph | it wasnt me |
22:29.04 | r_evolution | you dont sign me up |
22:29.30 | r_evolution | that is classic... |
22:29.31 | r_evolution | <doolph> Laughing Out Loud |
22:29.31 | r_evolution | * ManxPower has joined #asterisk |
22:29.31 | r_evolution | <jpm_SD> You know.. you can just LOL.. we all know what it means now. |
22:29.35 | r_evolution | i love it. |
22:29.42 | *** part/#asterisk tmccrary (n=tmccrary@68.78.185.254) |
22:29.43 | *** join/#asterisk hfb (n=hfb@pool-71-116-252-188.lsanca.dsl-w.verizon.net) |
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22:29.57 | doolph | was the script |
22:30.01 | doolph | it auto fillup |
22:30.57 | *** join/#asterisk Mitja (n=Mitja@cpe2-25-116.cable.triera.net) |
22:31.02 | *** part/#asterisk Mitja (n=Mitja@cpe2-25-116.cable.triera.net) |
22:31.09 | r_evolution | lol |
22:31.12 | r_evolution | hrrmmm |
22:31.13 | *** join/#asterisk Op3r (n=op3r@202.71.189.90) |
22:31.15 | r_evolution | LOL |
22:31.17 | r_evolution | hrmm |
22:31.18 | r_evolution | O_o |
22:31.21 | r_evolution | LoL |
22:31.23 | r_evolution | hrmmm |
22:31.27 | Op3r | Hi can I ask questions regarding vicidial? |
22:31.33 | Op3r | and how it works? |
22:31.41 | r_evolution | you can always ask them... but it doesnt mean someone will answer them |
22:31.52 | doolph | Op3r it's predictive dialer |
22:32.01 | Op3r | doolph: I know |
22:32.03 | *** join/#asterisk darkskiez (n=darkskie@bb-87-81-62-203.ukonline.co.uk) |
22:33.19 | doolph | then |
22:33.19 | Op3r | because here is the setup |
22:33.19 | Op3r | we connected analog phones to the quintums |
22:33.19 | Op3r | sometimes vicidial can connect to the phones |
22:33.19 | doolph | I think they use sip now |
22:33.19 | Op3r | but sometimes if a user just hung up the phone without login off first when they try to login it wont connect to the phones anymore |
22:33.19 | Op3r | the hardphones's protocol was sip |
22:34.17 | Op3r | I was wondering whats happening |
22:34.20 | Op3r | :( |
22:34.54 | Op3r | I just learned voip for about 2 weeks I am supposed to be the comp admin not the pbx admin L( |
22:36.59 | Primer | Anyone know if it's possible to map a button on a 7960 to "login/logout of a support queue" on asterisk? |
22:38.03 | Hmmhesays | sure, make your voicemail button hit that extension |
22:38.13 | Primer | well, we also need voicemail |
22:38.20 | Primer | so it can't be that button |
22:38.24 | Mauro__ | *77 |
22:38.25 | Mauro__ | :D |
22:38.36 | Primer | Mauro__: callate |
22:38.41 | Mauro__ | :) |
22:39.19 | *** join/#asterisk nagl (n=nagl@vie-086-059-104-148.dsl.sil.at) |
22:39.37 | r_evolution | just because i really enjoyed that |
22:39.38 | r_evolution | <doolph> Laughing Out Loud |
22:39.38 | r_evolution | <jpm_SD> You know.. you can just LOL.. we all know what it means now. |
22:39.40 | r_evolution | :-D |
22:39.43 | Op3r | does anyone knows any great vicidial how to;s and tutorial? |
22:39.57 | doolph | what |
22:45.45 | *** join/#asterisk ToTo (n=ToTo@host62-142.pool874.interbusiness.it) |
22:46.47 | *** join/#asterisk tmccrary (n=tmccrary@68.78.185.254) |
22:46.55 | tmccrary | what is the a bchannel and dchanne;? |
22:47.33 | bweschke | tmccrary: b channel carries voice - d channel carries signaling for a group of b channels in a PRI group |
22:47.53 | tmccrary | ah |
22:47.55 | tmccrary | thanks |
22:48.30 | *** join/#asterisk Strom_C (i=strom@66.159.243.60) |
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22:54.18 | *** join/#asterisk Nodren (n=nodren@64.193.95.10) |
22:54.22 | Hmmhesays | i subscribed to guitarone today |
22:54.26 | Hmmhesays | 12 bucks a year, cheap |
22:55.11 | Nodren | what distro would anyone recommend for running asterisk with a digium TDM400P |
22:55.25 | justinu | whatever you feel comfortable with |
22:55.37 | *** join/#asterisk naturalblue (n=Administ@87.192.100.109) |
22:55.41 | Hmmhesays | i recommend "insert *nix flavor you feel comfortable with here" ___________ |
22:55.55 | Nodren | well i've used debian and rehat based distros before |
22:56.03 | Nodren | i've had nothing but pure trouble trying to install this on centos |
22:56.12 | Hmmhesays | so don't use cent |
22:56.17 | justinu | centos works for me |
22:56.29 | Nodren | i mean freebsd? ubuntu? debian sarge? |
22:56.39 | justinu | fbsd is not a linux disto |
22:56.40 | justinu | distro |
22:56.47 | Hmmhesays | sarge works fine |
22:56.48 | Nodren | then not fbsd :P |
22:56.51 | Hmmhesays | ubuntu works fine |
22:56.59 | Nodren | ubuntu works great? |
22:57.01 | Nodren | i might try that. |
22:57.12 | Hmmhesays | hell fbsd works fine for some people |
22:57.35 | [TK]D-Fender | Nodren : Ubuntu is missing a LOT of packages required to build * and its components. probably better off with a more "complete" distro |
22:58.04 | Hmmhesays | i disagree [TK]D-Fender |
22:58.25 | Hmmhesays | you get apt-get install every package you need in ubuntu |
23:00.24 | [TK]D-Fender | Hmmhesays : I never said you coudn't do it, rather that you'll have plenty to do before you canhope to be up and running. |
23:00.53 | Hmmhesays | i just did one a couple days ago, 5 minutes of apt'ing and you're done |
23:01.07 | *** part/#asterisk amdtech (n=stdamd11@ab1-1-246.shsu.edu) |
23:01.22 | [TK]D-Fender | Hmmhesays : True, but thats for the more experienced user who know whre to geb everything from and what to get. |
23:01.27 | Abydos313 | amportal howto has a full list of requirements before * should be installed |
23:01.45 | [TK]D-Fender | Hmmhesays : just not as friendly "out of the box" as most other common ones |
23:01.57 | Strom_C | AMP is pure assrape. Don't use it. |
23:02.12 | Abydos313 | you can use the howto to list the packages required |
23:02.15 | justinu | lol |
23:02.25 | Abydos313 | see it's good for something |
23:02.39 | Hmmhesays | [TK]D-Fender you are right, its no debian |
23:03.58 | *** join/#asterisk VahramI (n=nospam@83.139.4.112) |
23:03.58 | *** join/#asterisk MacWeenie (n=chatzill@82-35-73-28.cable.ubr02.dals.blueyonder.co.uk) |
23:04.29 | *** join/#asterisk lunaphyte (n=lunaphyt@pool-71-115-128-96.gdrpmi.dsl-w.verizon.net) |
23:04.47 | shimi | I've set up asterisk with a script that copies a call request to a /var/spool/asterisk/outgoing. console says that all the commands were executed, yet it seems that the outgoing call request is not performed. what can it be? |
23:06.04 | VahramI | collegues, who has good practice writing cdrs by Agi script? |
23:06.39 | mishehu | hmm... I forgot, do polycom ip 50x's get the sntp setting from dhcp or from the config files? |
23:06.52 | MstlyHrmls | yes :-) |
23:07.09 | [TK]D-Fender | mishehu : Both |
23:07.13 | MstlyHrmls | if it's in DHCP, it will use that, but you can also set it in the config files (IIRC, anyways. I just use DHCP) |
23:07.27 | shimi | how do I designate ANY ZAP interface when setting up an outgoing call ? |
23:08.02 | shimi | I mean I have Zap with 31 channels, how do I tell asterisk to just use a free one? Zap/what ? |
23:08.19 | mishehu | [TK]D-Fender: know what the line in the configuration files is (and which file) ? I don't have anything in MAC-phone.xml, and the phones are hardcoded with IP information, and while in the settings menu on the phone itself it shows the ip of the ntp server, the web interface shows "clock" and the wrong offset. |
23:08.35 | mishehu | shimi: define a group |
23:08.44 | shimi | hey, mishehu! :) |
23:08.47 | shimi | sup dude? |
23:09.03 | mishehu | shimi: busy as all hell. you still hanging around #israel? |
23:09.26 | shimi | rarely |
23:09.29 | [TK]D-Fender | mishehu : its not in the phone, its in sip.cfg somewhere |
23:09.34 | shimi | I have ZAP/g0 - should I use that? |
23:09.52 | [TK]D-Fender | mishehu : just search for SNTP or GMT |
23:10.06 | mishehu | shimi: *nod* |
23:10.08 | shimi | uhm, I will just try, heh :) |
23:10.48 | mishehu | shimi: I'm only on #israel efnet, but I rarely even speak there these days. hard to take 3 classes at the college, run a small biz fulltime, and still have time to waste. ;-) |
23:11.22 | *** join/#asterisk Lurkan (n=Lurkan@201.152.101.6) |
23:11.29 | shimi | oh well :) |
23:11.36 | shimi | ok, it looks better now, it's not complaining |
23:11.45 | shimi | though it says that it couldn't complete the call :\ |
23:13.24 | *** join/#asterisk heart (n=zippetto@lugbari/people/heart) |
23:13.59 | Flauto | hey people |
23:14.11 | Flauto | i have a stupid question here |
23:14.54 | Flauto | how can i set up to answer a call by auto attentant and to give it only two chances of dialing an extension and then just hung up |
23:15.09 | justinu | create a variable |
23:15.11 | justinu | increment it |
23:15.12 | justinu | test it |
23:15.34 | Flauto | i have a variable |
23:15.41 | Flauto | but it goes in circles |
23:15.41 | *** join/#asterisk dimmik (n=dimmik@static217244.dsl.hol.gr) |
23:17.02 | Flauto | justinu, do you mind me to show you my settings privately? |
23:17.26 | r_evolution | INFINITE LOOP PWNS YOU! |
23:17.27 | r_evolution | ;) |
23:17.36 | Flauto | right |
23:17.42 | justinu | i'm not all that good at IVR stuff in the dialplan |
23:17.50 | Flauto | okay |
23:18.00 | dimmik | hi everyone. I am trying to set * to use g711 for the local phones and g729 when talking to a sip trunk. I am trying to avoid transcoding. Is this possible? |
23:18.00 | justinu | just giving you a suggestion |
23:18.00 | Flauto | r_evolution, would you help? |
23:18.06 | justinu | use noop to display your variable |
23:18.09 | justinu | figure out why it's not incrementing |
23:18.26 | Strom_C | dimmik, um, you have to transcode if you're going from 711 to 729 |
23:19.03 | dimmik | why so? both the sip trunk and the phone are capable of using g729 |
23:19.15 | Flauto | no, at the end of the variable, it is waiting for the caller to dial an extension |
23:19.40 | justinu | you need to create a counter variable |
23:19.49 | Strom_C | dimmik...so you want the phone to talk 729 to asterisk if the call goes out over a trunk, yet 711 if the call is internal? |
23:19.54 | Flauto | how would i do it? |
23:19.54 | justinu | so you can track how many times that particular channel has been thru the menu |
23:19.59 | dimmik | btw, I am not using Tt etc in the dial cmd |
23:20.14 | justinu | smething like Set(COUNT=1) |
23:20.26 | justinu | or Set(COUNT=${COUNT}+1)? |
23:21.08 | dimmik | Strom_C : I want the phone to talk g729 to the trunk without * touching the media |
23:21.10 | *** join/#asterisk AlexCTI (n=alex@adsl-072-156-253-012.sip.mco.bellsouth.net) |
23:21.10 | Flauto | where should i put it |
23:21.38 | justinu | right before it plays the menu to ask for an extension |
23:21.40 | AlexCTI | Hi. |
23:21.42 | justinu | then test it |
23:21.44 | Strom_C | dimmik, so have the phone talk 729 to asterisk and then asterisk will function as a pass-through |
23:21.45 | dimmik | Strom_C : Basically, making * act as a sip proxy |
23:21.50 | *** join/#asterisk riddlebox (n=james@24-171-11-166.dhcp.stls.mo.charter.com) |
23:21.56 | Flauto | okay |
23:22.00 | dimmik | Strom_C : exactly |
23:22.21 | Strom_C | dimmik, I wasn't asking you, I was telling you :) |
23:22.21 | justinu | dimmik, you use the allow, and disallow statements in sip.conf to control what codec stuff uses |
23:22.28 | *** join/#asterisk lunaphyte (n=lunaphyt@pool-71-115-128-96.gdrpmi.dsl-w.verizon.net) |
23:22.46 | dimmik | Strom_C : I tryied it without success |
23:23.01 | Strom_C | what happened? |
23:23.05 | Strom_C | define "without success" :) |
23:23.14 | dimmik | that was for justinu :) |
23:23.43 | Strom_C | then why did you direct the statement at me? |
23:23.44 | justinu | codec negotiation is kinda lame |
23:23.50 | justinu | it might be that you can't do what you want |
23:23.54 | dimmik | well, I set the defaults to disallow=all and allow=g729 and allow g711 |
23:24.09 | justinu | in that case, if the phone supports g729, it'll choose that first |
23:24.16 | dimmik | and for the sip trunk just disallow=all and allow=g729 |
23:24.20 | Strom_C | dimmik, why do you want g711 if you're restricting everything to g729? |
23:24.37 | dimmik | I want g711 for internal calls |
23:25.06 | Strom_C | how many simultaneous calls are you planning on having your asterisk box handle? |
23:25.36 | dimmik | justinu: this is correct, still I wanted to have g711 for internal calls |
23:25.45 | justinu | i understand what you want |
23:25.48 | justinu | i don't know the answer tho |
23:25.54 | justinu | it may not be posible |
23:26.01 | [TK]D-Fender | Strom_C : then just set the phones to G711 and te SIP peer to G729. End of story. * will transcode anything goin over that trunk. |
23:26.02 | justinu | or you may need some trickery in your dialplan |
23:26.04 | dimmik | Strom_C : just a few, transcoding is not possible though |
23:26.06 | Strom_C | dimmik, how many simultaneous calls are you planning on having your asterisk box handle? |
23:26.13 | Strom_C | dimmik, why is transcoding not possible? |
23:26.33 | Strom_C | are you too cheap to buy the licenses or something? |
23:26.38 | dimmik | i have a box with geode cpu |
23:26.48 | [TK]D-Fender | That'd do it :) |
23:27.19 | Strom_C | dimmik, get a real box :) |
23:27.24 | *** join/#asterisk websae (n=websae@h69-129-132-18.69-129.unk.tds.net) |
23:27.24 | dimmik | hehe |
23:27.48 | tsume | geode is good for sliim devices |
23:27.53 | tsume | not for servers |
23:27.55 | dimmik | Still, codec negotiation seems wrong in * |
23:28.47 | justinu | you'd need to dynamically set the codec basedon the destination of the call |
23:28.52 | Strom_C | exactly |
23:28.56 | Strom_C | and that's a bit mad |
23:29.02 | websae | anyone have a good suggestion (cost efficient) for a company that has 9 phone lines (8 voice and 1 FAX), they have a traditional PBX system for PSTN lines, they are going to be switching over to VoIP instead of SBC...what's a good interface to bring in the VoIP lines to this PBX that they already have, any suggestions, anyone :)? |
23:29.14 | dimmik | Is this possible? |
23:29.29 | justinu | i don't think so |
23:29.30 | Strom_C | websae, rip out the analog lines, install a PRI. |
23:29.33 | justinu | not out of the box |
23:29.49 | mishehu | analog lines... cacka poo poo |
23:29.58 | websae | they don't want a PRI |
23:29.59 | justinu | websae: you want something like a Lucent TNT |
23:30.03 | dimmik | this will require a reinvite, though |
23:30.10 | websae | they wa show...oh |
23:30.17 | websae | whoops...sorry |
23:30.21 | justinu | or a 12 port SIP to FXO gateway |
23:30.39 | Strom_C | jesus, don't you people know about the rule of two-wire conversion? :) |
23:30.52 | websae | okay... |
23:31.01 | websae | 12 port SIP to FXO..would work |
23:31.08 | websae | i could just wire that in right? |
23:31.08 | Qwell[] | Strom_C: When going from two-wire, to ethernet, reuse existing cables? ;) |
23:31.20 | Strom_C | websae, why do they want to hang on to their legacy PBX? |
23:31.46 | justinu | websae: yeah, it would basically just drop right in |
23:31.47 | Flauto | did not work |
23:31.58 | websae | they have PSTN phones (FXS)....they don't want to buy 20 new ones |
23:32.02 | justinu | an asterisk machine w/ a TDM2400 would also work |
23:32.16 | justinu | or some other high density analog telephony board |
23:32.19 | Strom_C | websae, asterisk machine with a quad-span T1 card, PRI, and two channel banks |
23:32.52 | websae | justinu...know of any good 12 port SIP to FXO? |
23:33.00 | Qwell[] | websae: Asterisk :P |
23:33.04 | mog_work | damn |
23:33.06 | mog_work | beat me to it Qwell |
23:33.08 | Qwell[] | mog_work: damn? |
23:33.09 | Qwell[] | oh :p |
23:33.14 | mog_work | i was gonna say 2400p and asterisk |
23:33.16 | justinu | websae: i have no hands on experience with such thing |
23:33.19 | Qwell[] | I'm looking at skinny |
23:33.23 | Qwell[] | ha! |
23:33.27 | justinu | um, I said asterisk and 2400 |
23:33.34 | Qwell[] | It crashes my phone when I lift up the handset :P |
23:33.40 | websae | so setup an asterisk box with a 2400 |
23:33.40 | Strom_C | websae, you can kludge something together or you can do it right. It would be much better to give them a unified solution than to kludge yet another part onto their existing system |
23:33.45 | mog_work | chan_skinny Qwell ? |
23:33.46 | Qwell[] | it registers though...yay me! |
23:33.49 | Qwell[] | mog_work: of course |
23:33.54 | mog_work | good start |
23:33.56 | Qwell[] | heh |
23:33.59 | mog_work | lots of fun stuff to debug |
23:34.05 | mog_work | you are pretty lucky |
23:34.13 | Qwell[] | how so? |
23:34.18 | mog_work | you get all the fun stuff |
23:34.20 | websae | channel banks are expensive |
23:34.22 | mog_work | ^_^ |
23:34.25 | Qwell[] | oh, you can do it if you'd like. ;) |
23:34.27 | Qwell[] | I'm a nice guy |
23:34.27 | justinu | you don't need channel banks |
23:34.33 | mog_work | bye |
23:34.39 | Qwell[] | heh |
23:34.59 | *** join/#asterisk Andr3w_ (n=Andrew@stjhnf0122w-142162049036.pppoe-dynamic.nl.aliant.net) |
23:35.24 | websae | just asterisk and a TDM 400 |
23:35.25 | Strom_C | websae, if they have 20 existing stations, a channel bank is going to be as cost-effective per station as a multi-port FXS gateway |
23:35.33 | justinu | TDM400, you'll need 3 of them |
23:35.41 | websae | *2400 |
23:35.49 | justinu | he wants to keep the existing PBX |
23:36.00 | Strom_C | screw keeping the existing PBX |
23:36.20 | justinu | that's his decision |
23:36.23 | Strom_C | you don't graft kludges onto existing hardware if you're going to do the install with any sense |
23:36.24 | dimmik | just get 6 pap2 and get rid of it |
23:36.26 | justinu | i'm not going to try and sway him |
23:36.51 | websae | okay...get rid of exisitng PBX |
23:37.00 | websae | and get channel banks then? |
23:37.04 | websae | which ones? |
23:37.04 | *** join/#asterisk harryvv (n=none@S010600a0c93f6f7e.vs.shawcable.net) |
23:37.04 | Strom_C | websae, do this |
23:37.11 | Strom_C | - asterisk box |
23:37.12 | justinu | now you need two TDM2400s, or a dual span T1 card + channel banks |
23:37.16 | Strom_C | - quad-span T1 card |
23:37.20 | Strom_C | - channel bank |
23:37.26 | Strom_C | - replace POTS lines with PRI |
23:37.36 | Strom_C | - renegotiate long-distance contract |
23:37.43 | Strom_C | you will save them plenty of money |
23:37.46 | websae | i ahve termination |
23:37.55 | websae | i have sip origination adn termination |
23:38.04 | websae | i don't need PRI |
23:38.06 | *** join/#asterisk A-Tuin (n=a-tuin@steves.ip.v4.me.uk) |
23:38.15 | Strom_C | PRI will be less expensive than nine analog lines |
23:38.27 | websae | i have a VoIP carrier |
23:38.31 | *** part/#asterisk PMantis (n=pmantis@cpe-66-66-115-197.rochester.res.rr.com) |
23:38.32 | websae | that uses SIP... |
23:38.38 | Strom_C | you're going voip only? you're taking a big risk... |
23:38.40 | justinu | are you happy with their quality? |
23:38.42 | Strom_C | who is the carrier? |
23:38.44 | justinu | because yes, that's risky |
23:38.53 | websae | yes i am very happy with their quality |
23:38.56 | harryvv | I dont know why it is but at cirtain times I have to shut down asterisk then when restarting it get a errr about outch! error while writing audio data broken pipe. zaptel/wcfxo would not load then I would recompile them. this is probebly a old issue and has been corrected with the latest patches. |
23:39.37 | websae | so i don't need PRI |
23:39.44 | Strom_C | websae, WHO IS THE CARRIER? |
23:39.48 | harryvv | Also, is it typical to see two instances of asterisk running? I have been seeing this as of late. |
23:39.55 | *** join/#asterisk unmanaged (n=unmanage@64.89.118.139) |
23:39.59 | websae | ComSolo |
23:40.29 | Strom_C | are you installing a dedicated T1 for the voip traffic? |
23:40.43 | websae | they have biz class roadrunner connection |
23:40.48 | Strom_C | what speed? |
23:40.56 | websae | 5mbs i think |
23:40.59 | Strom_C | symmetric? |
23:41.01 | websae | 1.5mb |
23:41.11 | Strom_C | wait wait wait wait |
23:41.12 | websae | they have aplenty of bandwidth |
23:41.25 | harryvv | hi Strom_C |
23:41.26 | Strom_C | you're going to run their voice AND data traffic over a 1.5 megabit connection?? |
23:41.31 | Strom_C | are you mad? |
23:41.33 | websae | 1.5 up |
23:41.35 | harryvv | I would not |
23:41.36 | justinu | i wouldn't run shit over a roadrunner connection |
23:41.37 | justinu | they suck |
23:41.44 | *** join/#asterisk Snake-Eyes (n=blog@203.220.55.70) |
23:41.49 | websae | their fiber is quite nice :) |
23:41.52 | Strom_C | and over roadrunner no less |
23:42.01 | unmanaged | in the current SVN release of asterisk, show application voicemail says that you can put an option for gain on rec of VM but I can't seem to get it to work... |
23:42.06 | Strom_C | can we check this guy into the asylum now? :) |
23:42.15 | justinu | you oughta let him dig his own grave |
23:42.24 | Strom_C | true, I should |
23:42.28 | harryvv | Are there any quick test tools on the market to do a snap shot of network bandwith for medium sized networks? |
23:42.35 | websae | so i need a asterisk box with what in it now? |
23:42.42 | websae | for the 20 analog phones |
23:42.49 | websae | ? |
23:42.54 | Strom_C | I told you already |
23:42.55 | justinu | single span T1 card + channel bank |
23:42.57 | justinu | or TDM2400 |
23:43.02 | dimmik | btw I thing I found something regarding codec negotioation : http://bugs.digium.com/view.php?id=4825 |
23:43.07 | unmanaged | t1 card and channel bank |
23:43.11 | websae | i don't need a t1 card |
23:43.19 | unmanaged | err |
23:43.26 | harryvv | web, how many channels are you going to be using? |
23:43.34 | websae | 20 analog phones |
23:43.34 | justinu | you might be able to get the T1 card + Channel bank for less |
23:43.38 | justinu | the TDM2400 card is like 2500 bucks |
23:43.38 | websae | 9 sip channels |
23:43.40 | unmanaged | you can get 20 ata's |
23:43.52 | harryvv | web, comming into your office? |
23:43.56 | Strom_C | websae, trust me from experience. You are going to severely regret running all their voice and data traffic over a single cable modem connection. |
23:43.59 | justinu | TE110P is 450 |
23:44.11 | justinu | channel bank? not sure... maybe 200-300 off ebay |
23:44.19 | websae | maybe i'll get another dsl connection |
23:44.35 | justinu | get a T1, and a router w/ good QoS policies |
23:44.36 | harryvv | Strom_C, did you recomend to him a seperate IP address and dedicated bandwith to voice traffic? |
23:44.36 | Strom_C | websae, are you an employee or just the consultant? |
23:45.21 | Strom_C | harryvv, no, I'm recommending that he get a PRI for the voice traffic |
23:45.30 | harryvv | yea good idea. |
23:45.38 | justinu | that's what I sell my customers |
23:45.44 | websae | part owner |
23:45.50 | harryvv | pri her cost 600-1200 dollars per month. |
23:46.11 | websae | justinu: what do you sell your customers? |
23:46.18 | justinu | PRIs and SIP phones |
23:46.22 | Strom_C | websae, listen to me. Don't be a cheapskate. You do not want to sacrifice reliability for a few hundred dollars a month. |
23:46.56 | harryvv | yes, realability issues will sink you in no time. no one wants poor voice connection. |
23:46.59 | *** join/#asterisk voipuser_au (n=voipuser@static-114.241.240.220.dsl.comindico.com.au) |
23:47.06 | blitzrage | anyone else seem to be able to get to ftp.digium.com? |
23:47.12 | blitzrage | nevermind -- I'll just use wget and the http link |
23:47.32 | harryvv | im connected |
23:47.35 | harryvv | blitzrage |
23:47.53 | *** join/#asterisk sigwerk (n=sigwerk@cyclone.sigterm.net) |
23:47.57 | blitzrage | wierd... couldn't get anything from the ftp a few mins ago -- ahhh well :) |
23:48.09 | harryvv | go for a walk |
23:48.10 | harryvv | ;) |
23:48.40 | harryvv | We are experaincing some of our nicest weather. One month ago it was a rain record. 40 days of rain almost in a row. |
23:49.23 | tuxinator_linux | blitzrage is going crazy |
23:49.32 | *** join/#asterisk octothorpe_ (n=octothor@unaffiliated/octothorpe) |
23:49.36 | Flauto | justinu, did not work |
23:50.04 | harryvv | BTW, shaw digital telephone and rogers telephone are competing in this market. Its a little discouraging to sell to people when thay say thay may want to go with them. Canadians overall are conservative and go with name brands. |
23:50.25 | *** join/#asterisk brc_ (n=brian@pdpc/supporter/basic/brc) |
23:50.55 | blitzrage | tuxinator_linux: going? |
23:51.10 | blitzrage | harryvv: agreed |
23:51.17 | brc_ | ~seen Corydon-w |
23:51.30 | jbot | corydon-w is currently on #asterisk, last said: 'Damn fork bombs... ;-)'. |
23:51.30 | brc_ | ~seen Corydon76-home |
23:51.31 | jbot | corydon76-home is currently on #asterisk (7h 38m 19s), last said: 'for unlimited data, including SMS'. |
23:51.54 | brc_ | hey Corydon-w you around? |
23:53.47 | *** join/#asterisk jasonpr2 (n=jason@64.78.192.164) |
23:55.29 | jasonpr2 | anyone know of any good tutorials on setting up Manager? |
23:55.47 | Strom_C | jasonpr2, yes, I've got the world's simplest tutorial |
23:55.48 | harryvv | blitzrage outside of digium, is there any asterisk support networks? say a company that does tier I-III support? |
23:55.53 | Strom_C | 1. Delete manager |
23:56.04 | Strom_C | 2. Learn how to operate config files :) |
23:56.23 | harryvv | company would ask...who is going to support our gear..I would. thay ask, what if your killed..then I did not have a good answer. |
23:56.24 | jasonpr2 | I need to do a lot of dynamic stuff |
23:56.43 | Strom_C | erm...by manager, you mean AMP, right? |
23:57.13 | jasonpr2 | ?? I'm a little new to asterisk. I just want access to the socket |
23:57.23 | Strom_C | never mind then - I was wrong |
23:57.28 | jasonpr2 | I looked at AMP it was pretty ugly |
23:57.50 | jasonpr2 | way to many dependacies |
23:58.05 | Strom_C | harryvv, is there a directory somewhere of voip / telecom consultants? |
23:58.12 | Katty | hihi |
23:58.28 | blitzrage | harryvv: how do you mean? |
23:58.30 | harryvv | Strom_C not that i know of. I think most are independent. |
23:58.32 | blitzrage | Strom_C: there is a listing on the wiki |
23:58.40 | Strom_C | blitzrage, link? |
23:59.15 | blitzrage | Strom_C: www.voip-info.org |
23:59.18 | blitzrage | not sure where it is on the site |
23:59.22 | blitzrage | should be easy to find though |
23:59.35 | Strom_C | i know the wiki, but I didn't see a page along those lines, which is why I asked |
23:59.41 | harryvv | blitzrage say a support company that has several asterisk consultant working under one roof and at a rate the customer can afford. no way in hell is a small company going to pay 175 per hour for support..thay will go some where else. |